00:01.12 | Druken | TUplink: iaxtel.com |
00:01.33 | TUplink | when you do a iax2 show registery |
00:02.02 | TUplink | 69.73.19.178:4569 xxxx <Unregistered> 60 Timeout |
00:02.14 | TUplink | the 69.73.19.178 |
00:02.26 | TUplink | is the ip the same |
00:02.31 | Druken | Host Username Perceived Refresh State |
00:02.32 | Druken | 65.39.205.121:4569 632421 <Unregistered> 60 Request Sent |
00:02.47 | TUplink | thats fwd |
00:02.59 | TUplink | no its not |
00:03.00 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
00:03.51 | tzanger | ReD-MaN: well you would have entered it into the dialplan |
00:04.00 | infinity1 | tzanger: i'm trying to install your zaptel packages... having a little hickup. |
00:04.04 | infinity1 | <PROTECTED> |
00:04.18 | tzanger | ... my zaptel packages?? |
00:04.39 | JASON-0 | Is asterisk supposed to bring up the D-Channel on my PRI ? |
00:04.48 | tzanger | JASON-0: if you tell it to, yes |
00:05.01 | infinity1 | tzanger: the ones from rapid.dotsrc.org |
00:05.07 | tzanger | ... ?? |
00:05.16 | JASON-0 | tzanger: Zaptel is installed, but the D-Channel isn't coming up. How do I configure Asterisk to bring it up? |
00:05.37 | tzanger | JASON-0: have you told zapata.conf it's on a PRI |
00:05.46 | JASON-0 | yes |
00:06.04 | JASON-0 | maybe I did it wrong though :P But I think i did it right |
00:06.11 | JASON-0 | switchtype = 5ess |
00:06.11 | JASON-0 | signalling = pri_cpe |
00:06.11 | JASON-0 | group = 0 |
00:06.11 | JASON-0 | channel => 1-23 |
00:06.17 | infinity1 | tzanger: oh ..:) ..tzafrir's ..not yours. |
00:06.33 | tzanger | JASON-0: looks good. zaptel.conf has bchan=1-23, dchan=24 ? |
00:06.39 | JASON-0 | tzanger: yes |
00:06.47 | *** join/#asterisk e-Hernick (n=ncc@modemcable120.39-131-66.mc.videotron.ca) |
00:06.50 | JASON-0 | span=1,0,0,esf,b8zs |
00:06.50 | JASON-0 | bchan=1-23 |
00:06.50 | JASON-0 | dchan=24 |
00:06.51 | tzanger | is the link up? |
00:06.57 | JASON-0 | the link is up yes |
00:06.58 | tzanger | you will want that 1,1,0 too |
00:07.01 | JASON-0 | but a d alarm |
00:07.10 | tzanger | there's no such thing as D alarm |
00:07.29 | TUplink | mine is trying the same ip that yours is working on |
00:07.47 | JASON-0 | tzanger: There is a DChannel Alarm |
00:07.58 | tzanger | JASON-0: there is no such thing as a D channel alarm |
00:08.00 | JASON-0 | On my Telco switch |
00:08.04 | Druken | TUplink: can you dial out iaxtel ? or is that the problem? |
00:08.15 | TUplink | both dont work |
00:08.21 | tzanger | JASON-0: you have a green light on the card? |
00:08.23 | TUplink | i can dialout fwd out |
00:08.25 | JASON-0 | tzanger: the telco side is showing the |
00:08.32 | JASON-0 | tzanger: yes, green light on card |
00:08.34 | Druken | try to call 700-725-9124 over iaxtel |
00:08.41 | TUplink | it dont register |
00:08.46 | tzanger | JASON-0: plug a loopback in to the smartjack, do they see it looped? |
00:08.49 | TUplink | give me a min |
00:09.04 | Druken | don't need to register to dial numbers, need to register to receive calls |
00:09.34 | JASON-0 | tzanger: its a sangoma card.. where is the smartjack? |
00:09.45 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
00:09.56 | tzanger | JASON-0: the smartjack is on the wall, it's what the telco puts there for you to plug in to |
00:11.41 | TUplink | Drunken its trying to call |
00:12.05 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
00:12.31 | TUplink | <PROTECTED> |
00:12.31 | TUplink | <PROTECTED> |
00:12.31 | TUplink | Oct 23 20:11:55 WARNING[527]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/Iaxtel/2 (type = 6, subclass = 1, ts=3, seqno=0) |
00:12.31 | TUplink | <PROTECTED> |
00:12.31 | TUplink | <PROTECTED> |
00:12.36 | JASON-0 | tzanger: I have not tried that but the T1 works with other T1 equipment. When I connect it to asterisk the link comes up but the d channel does not come up. |
00:12.56 | tzanger | JASON-0: that's odd. I'd call Sangoma support in the a.m. |
00:13.03 | tzanger | they should just come up |
00:13.04 | tzanger | mine did |
00:13.07 | tzanger | my digium cards do too |
00:13.08 | Druken | TUplink: i'm begining to think iaxtel is foobared |
00:13.12 | tzanger | oh wait |
00:13.17 | tzanger | you said switchtype's 5ess |
00:13.20 | tzanger | what is it emulating |
00:13.20 | TUplink | but explain FWD to |
00:13.28 | tzanger | VERY RARELY is a 5ESS actually signaling using 5ESS |
00:13.29 | TUplink | FWD dose the same thing |
00:13.39 | tzanger | 99% of the time it's dms100 or ni2 or something |
00:13.48 | tzanger | ask them what their switch is emulating, not what the switch is |
00:13.51 | Druken | TUplink: let me get my FWD number... |
00:13.55 | JASON-0 | ok I'll check that |
00:13.57 | tzanger | what is their switch signaling |
00:14.04 | JASON-0 | give me a few mins |
00:14.34 | TUplink | FWD should be send out 1800 calls use to |
00:14.36 | Druken | TUplink: 632421 |
00:15.15 | *** join/#asterisk __Elf (n=irc@rivendell.glassfish.net) |
00:15.23 | tzanger | dms100's kind of old but still used a lot |
00:15.24 | TUplink | Drunkin just ringing |
00:15.33 | tzanger | Bell Canada typically uses ni2 (national) |
00:15.58 | Druken | TUplink: it's not hittin my server then |
00:16.01 | tzanger | as does telus, group and UUnet (but that's a LONG time ago) |
00:16.02 | JASON-0 | tzanger: to reload zapata.conf what do I do? Just to make sure it's reloading |
00:16.12 | tzanger | restart asterisk |
00:16.19 | Druken | UUnet... haven't seen that in a while... |
00:16.21 | tzanger | switchtype is in asterisk not zaptel so you don't need to reload the kenrel module or execute zapate.conf |
00:16.25 | ReD-MaN | tzanger: is this what you were talking about? |
00:16.25 | tzanger | Druken: yeah no shit |
00:16.27 | ReD-MaN | [outrt-001-9_outside] |
00:16.27 | ReD-MaN | include => outrt-001-9_outside-custom |
00:16.27 | ReD-MaN | exten => _9.,1,Macro(dialout-trunk,1,${EXTEN:1},) |
00:16.27 | ReD-MaN | exten => _9.,2,Macro(outisbusy); No available circuits |
00:16.37 | tzanger | our first ever T1 was a 40km loop length from uunet |
00:16.37 | JASON-0 | zapata.conf is asterisk isnt it ? |
00:16.40 | tzanger | cost us a fucking fortune |
00:16.43 | tzanger | JASON-0: yes. |
00:16.49 | JASON-0 | just reload asteris ? |
00:16.54 | tzanger | ReD-MaN: ok that's the macro that dials it, now go into that macro, it will eventually get to a Dial() command |
00:17.01 | tzanger | JASON-0: restart, not reload |
00:17.21 | JASON-0 | ok |
00:17.34 | TUplink | Druken why not? |
00:17.53 | TUplink | do you have guest for you IAX setup |
00:18.05 | TUplink | so that i could try to IAX you |
00:18.07 | Druken | TUplink: yes sir |
00:18.29 | TUplink | ill add a regester line in my IAX |
00:18.34 | TUplink | and c if that registers |
00:18.58 | Druken | well, you'd just dial 7057259124@fishy.abss.ca |
00:19.18 | file | sounds a little fishy |
00:19.31 | TUplink | but that dont test te reg |
00:19.47 | Druken | ha ha ha file... hehe |
00:20.21 | Druken | TUplink: well, no.. you wouldn't register to a guest... a guest is for stray incoming calls |
00:20.33 | TUplink | ok.... |
00:20.44 | TUplink | well.... i dont know what to do? |
00:20.57 | Druken | someone just called.... |
00:20.59 | TUplink | im going to creat another FWD account and c if that dose any thing |
00:21.15 | Druken | was that you file?? :) |
00:21.19 | file | no |
00:21.22 | IronHelix | fwd over iax has been having trouble lately |
00:21.26 | file | I'm on my cell calling my ISP |
00:21.33 | e-Hernick | hey |
00:21.35 | ReD-MaN | tzanger: exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial |
00:21.35 | e-Hernick | what's up with AEL |
00:21.49 | tzanger | ReD-MaN: where are you getting this dialplan from? |
00:21.50 | JASON-0 | tzanger: i changed it to national and its the same.. I'm not seeing anything about it looking at zapata.conf in the full logs.. Is there an asterisk command that will tell me if it sees it? |
00:22.05 | tzanger | there's a lot of abstraction there and it looks like you're missing something in setting it up |
00:22.27 | ReD-MaN | dunno.. just found that in my extensions.conf file |
00:22.47 | ReD-MaN | under the macro-dialout-trunk |
00:23.24 | Druken | hmm.... tommy huff.... |
00:23.26 | ReD-MaN | I haven't modified a lot, just using what genzaptelconf made, with some modifications in AMP |
00:24.46 | *** join/#asterisk _DAW (n=_DAW@adsl-6-122-39.msy.bellsouth.net) |
00:25.04 | TUplink | Druken ??? |
00:25.10 | TUplink | Tommy Would be me |
00:25.41 | ReD-MaN | i also see this in the logs: Executing Dial("SIP/2600-72e8", "ZAP/g1/905741xxxx") in new stack |
00:26.58 | Druken | TUplink: oh... ok, well then you called me... |
00:27.31 | Druken | hey... a toronto guy |
00:27.40 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
00:28.07 | ReD-MaN | who's a toronto guy |
00:28.07 | *** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net) |
00:28.19 | Druken | ReD-MaN: edit your zapata.conf file and make sure your zap device belongs to group 1 |
00:28.40 | __Elf | is iaxtel working properly at the moment? |
00:28.41 | Druken | ReD-MaN: well... perhaps not toronto... but close enough |
00:28.55 | ReD-MaN | signalling=fxs_ks |
00:28.55 | ReD-MaN | rxwink=300 |
00:29.02 | ReD-MaN | is that right for Canada? lol |
00:29.12 | Druken | should be fine |
00:29.33 | *** join/#asterisk Inv_Arp (i=junya@adsl-144-134-75.mia.bellsouth.net) |
00:29.42 | ReD-MaN | ; Span 1: WCFXO/0 "Generic Clone Board 1" |
00:29.43 | ReD-MaN | signalling=fxs_ks |
00:29.43 | ReD-MaN | ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1 |
00:29.43 | ReD-MaN | context=from-pstn |
00:29.43 | ReD-MaN | group=0 |
00:29.46 | ReD-MaN | channel => 1 |
00:29.59 | ReD-MaN | Druken: Hamilton :) |
00:30.10 | ReD-MaN | Work in Mississauga |
00:30.11 | Inv_Arp | hell0 to all my fellow floridians |
00:30.15 | Druken | ReD-MaN: change that to group=1 |
00:30.18 | tzanger | ReD-MaN: that's all fine and working if your incoming calls work |
00:30.20 | tzanger | you have a dialplan issue |
00:30.21 | tzanger | that's all |
00:30.39 | tzanger | you need to dig through and find out what the OUT_* vars are if they're set right |
00:30.42 | Druken | tzanger: nah.. he's dialing g1, but no devices in group 1 |
00:30.51 | tzanger | Druken: oh? |
00:30.52 | ReD-MaN | well incoming calls get answered, but if you try to dial an SIP extension, it says the person is busy.. goes right to vm |
00:30.58 | tzanger | Druken: I missed that part |
00:31.02 | Druken | :) |
00:31.04 | tzanger | ahh group=0 |
00:31.05 | tzanger | I see it now |
00:31.15 | Druken | that's why we all look over the information |
00:32.29 | TUplink | Druken ou know whats wierd it dosent even tell me my pass is wrong |
00:33.09 | ReD-MaN | omg... it worked! :) |
00:33.18 | Druken | how about that... :) |
00:33.23 | Chuji | heh |
00:33.26 | ReD-MaN | changed it back to ZAP/g0 |
00:33.31 | ReD-MaN | and then it actually dialed out |
00:33.45 | ReD-MaN | now, one more question.. is there a way to change after how many rings before it answers? |
00:33.59 | TUplink | Wait() |
00:34.07 | ReD-MaN | my wife wants to keep using an analog set alongside this box.. |
00:34.21 | tzanger | ReD-MaN: you just Wait() before answering |
00:34.44 | Druken | ReD-MaN: uhmm.... are you going to use an ivr? |
00:35.38 | JASON-0 | tzanger : I had a noload = chan_zap |
00:35.39 | ReD-MaN | well what I was hoping to do, was not have the asterisk box answer the line right away.. all I have right now is a softphone.. don't have any ATA's yet |
00:35.44 | tzanger | JASON-0: hahahaha |
00:35.48 | tzanger | JASON-0: that'll do it |
00:35.53 | JASON-0 | tzanger: removed that and everything is fine now.. thanks for the help |
00:35.54 | JASON-0 | :) |
00:36.06 | ReD-MaN | so when my wife is home, she can still use a normal phone to answer the line. |
00:36.11 | Druken | ReD-MaN: get an ata as soon as possible :) |
00:36.18 | tzanger | JASON-0: I'd still verify the dchan switchtype emulation, it's unusual to actually use 5ess |
00:36.29 | ReD-MaN | Druken: I would, but they seem so expensive lol |
00:36.43 | ReD-MaN | I would only need one for my four handset cordless phone system tho |
00:36.46 | Druken | expensive? no more expensive than a telephone... 100 bux... |
00:37.12 | tzanger | I typically recommend the TDM11B or some variant instead of X101Ps and ATAs |
00:37.39 | tzanger | but tha'ts just me |
00:37.47 | Druken | i wouldn't reccomend the use of a TDM ever.... |
00:37.55 | ReD-MaN | Druken: expensive when you normally don't have to pay for stuff ;) mebbe I will sell off one of my extra Cisco switches and buy a decent ATA or something |
00:37.59 | Druken | i have a TDM and it's a peice of crap |
00:38.11 | tzanger | Druken: mine work *great* |
00:38.20 | Druken | what kinda cisco switchs? |
00:38.35 | ReD-MaN | 3548 |
00:38.49 | ReD-MaN | or a 2950 |
00:38.51 | Druken | tzanger: mine craps out all the time... only thing i use it for is my fax machine... otherwise i use ata;s |
00:38.59 | tzanger | Druken: using CVS HEAD? |
00:39.05 | Druken | yes sir |
00:39.12 | tzanger | hmm |
00:39.17 | tzanger | I don't know what to tell ya then |
00:39.23 | tzanger | mine work *great* |
00:39.27 | tzanger | and it's on an old dell P2 |
00:39.29 | Druken | just that i got a lemon |
00:39.31 | Druken | :) |
00:39.35 | tzanger | RMA the fucker |
00:39.56 | Druken | can't be bothered.... the ata's work just fine... |
00:40.01 | tzanger | heh |
00:40.16 | Druken | i just won't buy any other TDM's |
00:40.53 | ReD-MaN | hmm.. dialin gets answered, accepts the extension, but goes right away to saying the person at extension 2600 is unavailable.. weird |
00:42.12 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
00:42.39 | Druken | ReD-MaN: is that a 48 port switch ? |
00:42.49 | ReD-MaN | the 3548 yup |
00:42.53 | ReD-MaN | 2950 is a 24 port |
00:43.30 | Druken | nice... :) |
00:43.42 | Druken | i had a small cisco fry on my today... :( |
00:43.56 | ReD-MaN | switch? |
00:44.42 | ReD-MaN | ok this is weird.. I can call from my X-Lite softphone, call out no problem.. try calling into it, and says it is busy |
00:45.01 | ReD-MaN | almost like * doesn't know I am connected |
00:45.06 | morale | w |
00:45.34 | morale | here can i find a decent iax/sip provider in alberta? areacode 403 |
00:45.39 | Druken | yeah a switch... just a lil 5 port... but it still added to my day of my entire network crashing |
00:46.02 | *** join/#asterisk shakuhashi (n=shaku@200.163.5.67) |
00:46.23 | ReD-MaN | Druken: I hear ya. I now have 7 cisco's.. always gotta have a spare or two ;) |
00:48.07 | Druken | well, i had 3 servers crash... the hydro dropped out last night at the office, and the switch kicked the bucket |
00:48.21 | Druken | kicker is... one of the servers that crashed was 4 provinces over in alberta |
00:48.37 | ReD-MaN | ouch |
00:49.07 | tzanger | four provinces over? you're an easterner then |
00:49.13 | tzanger | NB? |
00:49.18 | Druken | no... i just can't count :) |
00:49.28 | tzanger | neither can I |
00:49.34 | tzanger | four provinces over is Quebec |
00:49.43 | tzanger | I forgot Saskatchewan |
00:50.05 | tzanger | yeah I know you're out there :-) |
00:50.14 | ReD-MaN | ON here |
00:50.15 | tzanger | although I thought you were newf |
00:50.18 | tzanger | ON here too |
00:50.33 | tzanger | Moc's PQ |
00:50.35 | ReD-MaN | omg.. calling my cell phone just caused it to reboot.. nice work Palm |
00:50.37 | Druken | i'm from good old barrie ontario :) |
00:50.39 | tzanger | sivana's ON |
00:50.46 | tzanger | ReD-MaN: what phone? |
00:50.51 | ReD-MaN | Treo 600 |
00:50.58 | tzanger | barrie? you're not far from sivana then, he's in NB |
00:51.06 | tzanger | I'm 1.5hrs WNW of Pearson |
00:51.15 | tzanger | ReD-MaN: I'm waiting for the Treo650 |
00:51.19 | file[laptop] | Pearson, I know that airport well |
00:51.20 | Druken | hehehe if you drive like 200km/h |
00:51.23 | ReD-MaN | my work is 5 mins from Pearson lol |
00:51.31 | ReD-MaN | the plains fly over our office |
00:51.47 | tzanger | Druken: well he does drive quickly |
00:51.54 | tzanger | you're not really far from me either, probably 2h |
00:51.55 | ReD-MaN | tzanger: work phone.. we bought over 200 of them.. before we knew how crap they were |
00:51.56 | Druken | hehehe so i've heard |
00:52.04 | ReD-MaN | now Rogers is going to upgrade us to 650s |
00:52.14 | tzanger | ReD-MaN: yeah the 600s have their bugs but I thought most of those were corrected in the 650, with the addition of a few other bugs :-) |
00:52.22 | tzanger | ReD-MaN: yeah I'm with Telus and Bell |
00:52.26 | tzanger | neither have the 650 yet |
00:52.26 | ReD-MaN | especially now that RIM and Palm have signed to have Blackberry services on Treo 650 + |
00:52.37 | tzanger | ReD-MaN: yeah, I have always HATED the blackberries |
00:52.40 | file[laptop] | we have no Mike out here :D |
00:52.43 | tzanger | proprietary, slowass pieces of shit |
00:52.54 | tzanger | now bring some of that tech over to Palm and we got something |
00:52.56 | ReD-MaN | I have Telus for personal service, Rogers for work |
00:53.30 | ReD-MaN | we call BB's Crackberries at work |
00:53.38 | ReD-MaN | cause every one who has one is ALWAYS doing something on it |
00:54.07 | file[laptop] | I remember when I was waiting at an airport, guy beside me was playing with his... trying to get it hooked up for data service to his laptop |
00:54.10 | file[laptop] | failed horribly |
00:54.23 | tzanger | :-) |
00:54.25 | tzanger | I hate them |
00:54.29 | tzanger | nice transflective screens though |
00:54.33 | tzanger | but they need a touch screen |
00:54.36 | tzanger | holy hell do they need touch screen |
00:54.53 | ReD-MaN | heh.. one guy at work.. every time he presses his space bar, the thing reboots.. so he has to send emails with a - instead of a space |
00:55.03 | file[laptop] | LOL |
00:55.08 | rking | that-sucks. |
00:55.44 | Druken | i just use webmail... where can you go these days where a computer isn't already at? |
00:56.10 | wunderkin | a cardboard box |
00:56.10 | file[laptop] | laptop+cellphone=goodness |
00:56.50 | Druken | laptop + aircard == heavenly |
00:57.15 | file[laptop] | Druken: bluetooth capable cellphone and laptop :D |
00:57.33 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
00:57.56 | Druken | that works too... but ya still can't get calls and be online at the same time |
00:58.18 | Druken | with an air card, ya may be able to use voip with g729 |
00:58.20 | tehdely | news at 11: for anyone around the other night when i was trying to solve my popping/crackling problem |
00:58.26 | tehdely | upgrading to bleeding-edge zaptel from cvs fixed it |
00:58.29 | tehdely | oh yes it most certainly did |
00:58.29 | Druken | might be a lil shitty, but i bet it would work |
00:59.20 | *** join/#asterisk Inv_arp (i=junya@adsl-144-134-75.mia.bellsouth.net) |
01:00.12 | orlok | I wonder how many people have been killed by Cisco 7940/7960's |
01:00.14 | Druken | ok, all these canadians in here, who has a 1u keyboard 19" rolling shelf they want to sell me for cheap ? |
01:00.39 | *** join/#asterisk file (n=jcolp@mctnnbsa31w-142166113136.nb.aliant.net) |
01:00.45 | file | bleh |
01:00.53 | file | internet connection be b0rken |
01:01.51 | ReD-MaN | 1u keyboard 19" rolling shelf? |
01:02.01 | ReD-MaN | I have some, dunno if they are 19" though |
01:02.24 | ReD-MaN | they were from some weird rack at work.. thought they would fit my 19" rack, but they are too wide |
01:03.17 | Druken | well, if they won't fit your 19" i doubt they would fit mine... |
01:04.06 | *** join/#asterisk orbi (n=dantate@pcp08696782pcs.500ash01.tn.comcast.net) |
01:05.10 | orbi | I just upgraded a Cisco 7960 phone to the latest SIP firmware, made a typo and got a Protocol Application Invalid error. I cant find any definitive answers on ye olde google, but can someone tell me if i just made a perfectly good phone into a pretty paperweight? |
01:05.41 | Qwell | orbi: just reboot it, and it should get the right firmware, no? |
01:06.04 | orbi | <PROTECTED> |
01:06.16 | orbi | but if i watch the TFTP server, it never attempts to pull any of the right files |
01:06.25 | orbi | never tries to pull OS79XX.TXT for example |
01:06.26 | Qwell | "right files" or "any files"? |
01:06.42 | orbi | it tries to pull its own config, and a .tlv file, but thats it. |
01:06.50 | ReD-MaN | mmmm Cisco voip phone.. what I wouldn't give for one of those |
01:07.01 | Qwell | ReD-MaN: I'll sell you mine :p |
01:07.03 | *** join/#asterisk ejo1 (n=ejo1@209.32.147.246) |
01:07.10 | ReD-MaN | almost had Cisco to deploy at work... until they decided to go with Mitel yuck |
01:08.01 | orbi | im just now starting to play with Asterisk |
01:08.19 | orbi | we use Cisco Callmangler at work, and resell it |
01:08.27 | orbi | but they're interested in asterisk for smaller offices |
01:08.41 | orbi | i'ev got a ton of newbie questions, but i'm not going to espouse them here. |
01:08.45 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166113136.nb.aliant.net) |
01:08.54 | orbi | Thats what bugging twisted is for. |
01:08.54 | orbi | ;) |
01:09.11 | tehdely | we have a cisco phone at work |
01:09.14 | tehdely | but it has the SCCP image |
01:09.21 | tehdely | and i do not feel like dicking with chan_sccp |
01:09.35 | tehdely | i think i'll go spend $200 on a support contract to get the SIP image |
01:09.36 | tehdely | ;) |
01:09.59 | ReD-MaN | heh.. I have access to all Cisco downloads.. but not much to put them on lol |
01:10.05 | Aughey | how many X100P cards can I put in a machine reliabily? I can get 4 X100P cards for much less than a single tdm04 card setup. |
01:10.45 | Druken | i managed to get 3 working properly |
01:10.46 | Qwell | reliably? 0 |
01:11.02 | Druken | but that's 2 x100p's and 1 TDM |
01:11.12 | orbi | I was going to use sccp or skinny, but people swear up and down that SIP is better |
01:11.17 | Aughey | did you try 3 x100p? |
01:11.17 | orbi | do you guys find that true? |
01:11.42 | tehdely | orbi: i don't know enough about the implementation of either to say |
01:11.48 | tehdely | but SIP is definitely a better choice if you're working w/ open systems |
01:11.50 | Druken | Aughey: nope... but if i can use the 3 cards like i do, i'm sure you could do 3 x100p's |
01:11.54 | tehdely | like asterisk, or any other SIP IP PBX |
01:11.57 | Druken | they all use an IRQ |
01:11.58 | tehdely | SCCP is proprietary |
01:12.05 | tehdely | and there is no open reference implentation, or really a reference at all |
01:12.12 | tehdely | so everyone's impl except for cisco's is bound to be half-assed |
01:12.15 | tehdely | which i hear chan_sccp is :/ |
01:12.33 | Druken | night |
01:13.28 | tehdely | nn |
01:13.42 | ReD-MaN | hmmm. I wonder if Mitel phones do SIP |
01:13.57 | Qwell | ReD-MaN: are they even IP phones? |
01:15.22 | ReD-MaN | yup |
01:15.32 | ReD-MaN | in the middle of deploying em at work |
01:15.53 | ReD-MaN | I was given their crappy softphone to use since I am mobile user |
01:16.27 | tehdely | another crazy boogyman protocol from the crypt: unistim |
01:16.32 | tehdely | anyone here ever buggered around with * and nortel phones |
01:16.35 | tehdely | i'd like to hear how that went |
01:18.06 | JASON-0 | I am using the AMP portal and i've setup an inbound DID but when I call it I get the following error in the logs.. - Extension '(did here)' in context 'default' from '(clid here)' does not exist. Rejecting call on channel 0/14, span 1 |
01:22.34 | JASON-0 | anyone have any idea? |
01:24.55 | Aughey | do you have a default context in your extensions.conf file? |
01:24.59 | tehdely | JASON-0: probably want to create an extension for it then ;) |
01:25.01 | tehdely | [globals] |
01:25.05 | tehdely | MYCOOLDID=3125551212 |
01:25.13 | tehdely | exten => ${MYCOOLDID},1,DoStuff |
01:25.55 | tehdely | just guessing though |
01:25.59 | tehdely | i am actually teh n0b |
01:26.01 | tehdely | ^0 |
01:27.16 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
01:44.04 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
01:44.26 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
02:08.15 | *** join/#asterisk littleball (n=littleba@bb220-255-134-71.singnet.com.sg) |
02:08.43 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
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02:13.26 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
02:13.33 | Goshen | Where does monitor store its files? |
02:13.47 | Goshen | seems like it was /var something |
02:14.14 | Goshen | ahh, /var/spool/asterisk/monitor |
02:17.28 | *** join/#asterisk aka-Debs (n=dbusch78@70.89.213.201) |
02:18.34 | lunk | w |
02:19.35 | |Vulture| | Goshen: you can specify |
02:20.15 | Goshen | |Vulture|: Thank you, I just wanted to back up some recordings |
02:20.50 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
02:21.01 | |Vulture| | Goshen: we use the monitor() function on all of our calls and it works well, make sure you specify the mux string as a global |
02:21.45 | *** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com) |
02:21.49 | Katty | hihi. |
02:24.52 | *** part/#asterisk squeegy (n=mike@cpe-24-33-74-234.cinci.res.rr.com) |
02:38.06 | Vco | any pointers on where i can unfux0ur this? "Everyone is busy/congested at this time (1:1/0/0)" |
02:38.12 | Vco | incoming SIP DID.. |
02:38.14 | Vco | no joy |
02:39.04 | X-Rob | Vco - about 10 lines back is the actual error. |
02:39.22 | X-Rob | usually 'no extension xxx in context yyy' |
02:39.42 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
02:39.42 | Vco | well..thats the thing |
02:39.46 | shido6 | boink... |
02:39.51 | Vco | the phone is ringing.. |
02:39.56 | Vco | i answer it and it goes busy |
02:39.58 | *** join/#asterisk paxr0 (n=paxr0@200-85-200-140.bk3-dsl.surnet.cl) |
02:39.59 | shido6 | answer it |
02:40.00 | Vco | on the phone |
02:40.07 | Vco | the incomgin call still rings |
02:40.30 | X-Rob | sip debug |
02:42.02 | paxr0 | i correct this line? exten => s,28,NoOp(${CHANNEL STATUS ZAP/1}) |
02:42.21 | paxr0 | y have a FXO in Zap/1-1 |
02:45.43 | Vco | yay...miles of debug output |
02:45.53 | phrog123 | if I put context=something in zapata.conf in the channel for ZAP/1 and in extensions.conf, I do not create a context called something, does this guarantee that * will not answer zap/1? |
02:46.04 | phrog123 | if I forego of doing context= in zapata.conf, then this is as if context=default? |
02:46.10 | phrog123 | finally, if I put context=something after channel=>1, then it is as-if context=default |
02:49.49 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
02:50.12 | *** join/#asterisk JASON-0 (n=jason@jason.unitz.ca) |
02:50.38 | JASON-0 | Hello, I have AMP and I've added extensions busy when I call them, they are all busy and I get forwarded to vmail.. I'm not sure why.. |
02:51.10 | shido6 | brb |
02:51.18 | shido6 | Jason dont freak out - brb |
02:51.22 | shido6 | i'll help ya out in a minute |
02:51.26 | JASON-0 | thank you :D |
02:53.55 | hugo-v6 | does someone know if there exists a better solution than suEXEC from apache to wrap cgis? |
02:54.51 | Inv_arp | Ariel_: around? |
02:58.35 | *** join/#asterisk hcir (n=hcir@rdbck-static-532.palmer.mtaonline.net) |
03:02.42 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
03:04.02 | JASON-0 | Hi shido6, did you still want to help me :P |
03:05.35 | shido6 | sure |
03:05.37 | shido6 | whats up? |
03:06.38 | JASON-0 | Every extention I dial I get busy or vmail. I am able to dial out on trucks but no extensions. I have used AMP to add the extension. |
03:06.58 | hugo-v6 | hmmm dan 5am and no sleep at all. |
03:07.04 | hugo-v6 | at least its my bd. |
03:07.14 | hugo-v6 | s/dan/damn/ |
03:10.21 | glm2k | bd? |
03:10.30 | glm2k | happy birthday then? |
03:11.16 | blitzrage | JunK-Y: lol -- you have a Junky tshirt! |
03:11.28 | JASON-0 | shido6: Any idea? |
03:12.04 | file[laptop] | blitzrage: !!! |
03:12.08 | X-Rob | JASON-0 - your dialparties.agi is broken. |
03:12.17 | file[laptop] | blitzrage: did you make it okay? |
03:12.27 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
03:12.32 | X-Rob | JASON-0 - also, AMP support is on #amportal (hint) |
03:12.44 | JASON-0 | X-Rob: sorry |
03:13.11 | *** join/#asterisk concept10 (n=concept1@c-67-166-167-125.hsd1.tx.comcast.net) |
03:13.18 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
03:13.19 | shido6 | ok |
03:13.21 | shido6 | what phones? |
03:13.25 | X-Rob | you're, uh, meant to join the #amportal channel. |
03:13.25 | shido6 | ip phones or analog |
03:13.35 | JASON-0 | Cisco 7960 |
03:13.38 | shido6 | nice |
03:13.43 | shido6 | sip show peers shows them registered? |
03:13.57 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166113136.nb.aliant.net) |
03:14.13 | JASON-0 | yes, sure does |
03:14.28 | shido6 | any "x" 's next to the line names on the criscos? |
03:14.43 | blitzrage | file[laptop]: nope |
03:14.54 | file[laptop] | blitzrage: so they really turned you away? |
03:14.59 | blitzrage | file[laptop]: yep |
03:15.25 | JASON-0 | shido6: no |
03:15.32 | file[laptop] | blitzrage: I prefer you without 'da beard |
03:15.42 | hugo-v6 | thanx glm2k |
03:15.57 | blitzrage | file[laptop]: that was kind of the point :) |
03:16.21 | Corydon76-home | pretty pretty blue eyes |
03:16.42 | shido6 | kewl |
03:16.48 | shido6 | now follow the CLI error |
03:16.53 | shido6 | and paste them at pastebin.ca |
03:16.57 | shido6 | erro(s) |
03:17.01 | JASON-0 | http://pastebin.ca/26437 |
03:17.15 | JASON-0 | I think I found the problem.. X-Rob was right |
03:17.27 | X-Rob | shido6 - it's AMP, can't call extensions, dialparties.agi is broken. |
03:17.31 | shido6 | ooooh busy phones |
03:17.33 | X-Rob | that's always the problem 8) |
03:17.46 | shido6 | u want my dialparties.agi? |
03:17.48 | JASON-0 | Can't locate Asterisk/AGI.pm |
03:17.56 | shido6 | erf |
03:17.57 | X-Rob | shido6 - it's missing a prerequisite. |
03:18.00 | JASON-0 | I'm midding Asterisk::AGI |
03:18.04 | JASON-0 | oops.. missing |
03:18.04 | shido6 | asterisk perl? |
03:18.05 | Qwell | blitzrage: The tartan room! heh |
03:18.07 | orlok | woo |
03:18.11 | JASON-0 | yes :P |
03:18.13 | shido6 | then download asterisk perl |
03:18.14 | JASON-0 | thanks guys |
03:18.14 | orlok | got the sip images onto this here 7940 |
03:18.15 | shido6 | dagnabbit |
03:18.16 | orlok | :) |
03:18.17 | shido6 | and install it |
03:18.20 | shido6 | perl Makefile.PL |
03:18.23 | JASON-0 | thank you shido.. I will :P |
03:18.23 | shido6 | make install |
03:18.32 | JASON-0 | Thanks X-Rob |
03:19.14 | X-Rob | w00 |
03:19.23 | shido6 | my site will be up shortly |
03:19.25 | X-Rob | Opteron? |
03:19.27 | shido6 | no |
03:19.28 | shido6 | X2 |
03:19.31 | X-Rob | bah |
03:19.38 | blitzrage | Qwell: hehehehe :) |
03:19.39 | X-Rob | the one that takes 400w at idle? |
03:19.49 | X-Rob | I'll pass on that. I like my power bill only being 2 digits long, thanks. |
03:19.59 | orlok | man |
03:20.08 | orlok | imagine upgrading a whole server room to those suckers |
03:20.14 | orlok | imagine the aircon upgrade you would need |
03:20.19 | shido6 | fine pass it up :) give it back |
03:20.24 | orlok | would probably cost more than the CPU's :) |
03:20.35 | X-Rob | But I'll have a dual core opteron anytime, thanks 8) |
03:20.43 | shido6 | thanks |
03:20.46 | JASON-0 | It works now :D :D :D |
03:20.48 | JASON-0 | thanks guys |
03:21.11 | shido6 | I will ship my first one next week |
03:21.29 | shido6 | i have 4600+'s working |
03:22.08 | Qwell | blitzrage: so, where was his shoe? |
03:23.23 | Vco | ok.. |
03:23.35 | Vco | i'm thining this DID issue is something else.. |
03:24.02 | Vco | have some UK ones that work fine..same config...it's 2 DID's for Tokyo from 2 different providers... |
03:29.51 | *** part/#asterisk mcadory (n=mcadory@208.149.64.28) |
03:31.07 | *** join/#asterisk Frank999 (n=frank999@ppp-70-227-85-31.dsl.sfldmi.ameritech.net) |
03:36.40 | Inv_arp | Ariel_: around? |
03:38.26 | blitzrage | Qwell: I think it was in a bush somewhere :) |
03:41.29 | hypa7ia | blitzrage: the unidentified person in the photos with you and mark and allison is qwell |
03:41.44 | JunK-Y | blitzrage: u were able to decipher my shit btw? |
03:42.46 | Qwell | hypa7ia: where are those? |
03:43.10 | hypa7ia | http://leifmadsen.com/gallery/astricon_2005 |
03:43.16 | Qwell | ahh |
03:44.09 | JunK-Y | some are at |
03:44.12 | JunK-Y | ~astricon2005 |
03:44.13 | jbot | rumour has it, astricon2005 is at http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album02 |
03:44.18 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
03:48.12 | FuriousGeorge | you know what bothers me. when you a dial(sip/user,30) followed by a voicemail(u${USEREXTEN}), what if the call is answered and the answering party hangs up first. why shouldnt it still go on to the next priority |
03:48.37 | FuriousGeorge | iow, the calling party would still go to voicemail. seems like it should |
03:50.05 | IronHelix | dial(sip/user,30,g) will do that |
03:50.10 | IronHelix | g: When the called party hangs up, exit to execute more commands in the current context. |
03:55.23 | FuriousGeorge | IronHelix: i recind my complaint |
03:55.28 | IronHelix | :) |
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03:58.12 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
03:58.58 | *** part/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
03:59.34 | *** join/#asterisk e-Hernick (n=ncc@modemcable120.39-131-66.mc.videotron.ca) |
04:01.18 | *** join/#asterisk ejo1 (n=ejo1@209.32.147.246) |
04:01.50 | encode | ooh, lots of pretty astricon pictures |
04:02.07 | encode | what happens at astricon? obviously its a conference about asterisk |
04:02.13 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
04:02.14 | Qwell | encode: lots of drinking |
04:02.20 | encode | lol |
04:04.33 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
04:05.21 | blitzrage | hypa7ia: oh nooooooo way!!! |
04:05.30 | blitzrage | Qwell: holy shit, I just realized *now* who you are!!!!!! |
04:05.33 | Qwell | haha |
04:05.52 | Qwell | yeah, I don't think I ever actually intoduced myself... |
04:05.53 | blitzrage | next Astricon, I'm making sure there is a spot to put IRC names on the badges |
04:05.59 | Qwell | blitzrage: yes, thats important |
04:06.04 | blitzrage | Qwell: you probably did, but I'm terrible with names |
04:06.36 | Qwell | blitzrage: makes sense why I bought you a drink now, eh? :p |
04:06.57 | blitzrage | Qwell: uhhhhh yah :) |
04:07.02 | blitzrage | Qwell: I feel like a fool now :) |
04:07.09 | Qwell | blitzrage: its all good, heh |
04:07.24 | Qwell | I thought you were one of the Matts when you first got to the tartan room... |
04:07.32 | blitzrage | Qwell: lol |
04:07.38 | Qwell | in my defense, I was a bit drunk by that time, heh |
04:07.50 | blitzrage | Qwell: hehehehehe, who wasn't? :) |
04:07.53 | file[laptop] | blitzrage: your name is cooler then Matt though |
04:08.02 | blitzrage | file[laptop]: well yah, thats true :) |
04:09.16 | blitzrage | but I *am* a r0ck star... so it makes sense |
04:09.55 | Vco | that doesn't look like your arm tho |
04:10.11 | Corydon76-home | The Leif doesn't fall far from the (Asterisk) tree |
04:11.08 | blitzrage | Corydon76-home: I don't really know what that means... :) |
04:11.19 | blitzrage | Corydon76-home: oh I get it now if I say my name wrong |
04:11.24 | blitzrage | lol |
04:11.28 | blitzrage | Leif == Life, not Leaf |
04:11.48 | Corydon76-home | blitzrage: it's meant to confuse you while I take other measures to seduce you... :-P |
04:12.25 | blitzrage | Corydon76-home: gah! |
04:12.44 | Corydon76-home | Everybody can be seduced... |
04:13.10 | Corydon76-home | The only question is, how difficult? |
04:13.11 | blitzrage | Corydon76-home: not true |
04:13.33 | Corydon76-home | blitzrage: there's always chloroform... :-P |
04:14.10 | blitzrage | Corydon76-home: I suppose thats true... not much you can do to fight that |
04:14.57 | Qwell | eh...chloroform isn't difficult to avoid. |
04:15.14 | Qwell | learn the smell of it, and if you ever smell it, just hold your breath :p |
04:15.27 | Corydon76-home | file[laptop]: only mildly? |
04:15.33 | file[laptop] | only mildly. |
04:15.35 | file[laptop] | try harder next time |
04:16.09 | Corydon76-home | file[laptop]: it doesn't get much harder than it is now... |
04:16.19 | file[laptop] | that's sad |
04:16.57 | Corydon76-home | file[laptop]: Ask Coil about it sometime... |
04:17.02 | blitzrage | can any else get to creativecommons.org ? |
04:17.56 | Qwell | no, but I get an incredibly low ping to them |
04:18.05 | Qwell | 9ms on cable? |
04:18.08 | encode | page doesnt seem tobe loading |
04:18.14 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
04:18.14 | *** mode/#asterisk [+o denon] by ChanServ |
04:18.26 | file[laptop] | omg denon |
04:18.27 | encode | i get "operation timed out" from our proxy |
04:18.30 | file[laptop] | nini all |
04:18.37 | Corydon76-home | g'night, file |
04:18.43 | encode | cya file |
04:19.01 | Qwell | oh, I'm on my server still...wtf |
04:19.50 | blitzrage | yah, doesn't load for me, must be down or something |
04:21.07 | blitzrage | ~seen jerjer |
04:21.10 | jbot | jerjer <n=JerJer@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk, 1d 11h 1m 39s ago, saying: 'but playback will answer the channel if its not already up'. |
04:21.33 | Qwell | blitzrage: working now |
04:21.35 | blitzrage | anyone have Jeremy's IM? (if he uses IM that is) |
04:21.37 | blitzrage | Qwell: thanks |
04:26.54 | Corydon76-home | blitzrage: AIM:Corydon76-home is his IM... |
04:27.31 | Corydon76-home | Or maybe just AIM:Corydon76 ;-) |
04:28.48 | blitzrage | heh |
04:29.14 | blitzrage | plus I'm a bit disillusioned with the USA today |
04:29.18 | Corydon76-home | Gee, I dunno why... |
04:29.43 | Corydon76-home | It's not like Canada doesn't also watch its citizens... |
04:29.58 | wunderkin | blitzrage: you'll probably end up talkin to some 70 year old pervert :D |
04:30.19 | Qwell | wunderkin: in the internet age, that is inevitable |
04:30.25 | Corydon76-home | wunderkin: No, that's file. He only pretends to be 19 online... |
04:31.00 | blitzrage | lol |
04:31.15 | wunderkin | .oh maybe hes 76 :D |
04:31.22 | Corydon76-home | Uh huh |
04:32.04 | Icemaann | 76? |
04:32.05 | Icemaann | lol |
04:32.15 | wasim | don't forget silly c888 |
04:33.36 | Corydon76-home | Icemaann: bitch. :-P |
04:33.57 | glm2k | 76 going 77? or 75 going 76? |
04:34.18 | Icemaann | i have impressive ping times to sellvoip (the ips listed on their site), but ive never used them |
04:34.33 | Qwell | Icemaann: What is impressive? |
04:34.39 | Corydon76-home | Born in 76. Which would make me one thousand, nine hundred, twenty nine years of age... |
04:35.12 | glm2k | hmmm, that means you're either a ponty eared, blue skinned elf or a warlock |
04:35.34 | Corydon76-home | glm2k: and I have a bridge I'd like to sell you... |
04:35.44 | glm2k | hehehe |
04:35.54 | Icemaann | Qwell: 16 ms |
04:36.23 | glm2k | Icemaann: wow. i get 76 ms |
04:37.27 | Icemaann | glm2k: did u look at the ips in the FAQ? i get 76 to 1 and only 16 to the other |
04:38.37 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
04:38.43 | Icemaann | 7 hops for me |
04:39.21 | LostFrog | kram: may I just say thank you for asterisk and gaim.. you are a god. |
04:41.35 | glm2k | 13 hops for one and 18 for the other |
04:41.50 | Vco | anyone else here use voxbone? |
04:42.13 | Qwell | voxbone sounds like a really bad porno or something |
04:42.17 | Vco | yea..i know |
04:42.32 | glm2k | hehe |
04:42.37 | Vco | Voip Over Interracial Porno |
04:42.43 | Vco | or some shit |
04:43.24 | LostFrog | Very Overrated Internet Porn |
04:43.39 | *** join/#asterisk wjn78 (n=wjn78@adsl-68-123-232-96.dsl.irvnca.pacbell.net) |
04:44.38 | wjn78 | did I miss anything? |
04:44.39 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
04:45.24 | Vco | yea, Microsoft Just bought Digium |
04:45.55 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
04:45.55 | *** mode/#asterisk [+o denon] by ChanServ |
04:46.04 | wjn78 | lol that would be bad |
04:46.07 | wjn78 | I like Asterisk |
04:46.53 | Jabroni | anyone knows if there a way to get the response time for IAX2/SIP peers via the manager api ?? |
04:47.13 | LostFrog | Ack.. those are fighting words, Vco. |
04:47.29 | wjn78 | LostFrog its a joke because I use to work at Microsoft |
04:48.18 | Icemaann | this conversation about goiax on -users is interesting. |
04:53.43 | *** part/#asterisk Hobbes` (n=calvin@59.92.143.37) |
05:04.04 | Vco | bah...verbose cli display and sip debug just blur together now.... |
05:09.46 | wjn78 | Vco how long have you been working on Asterisk? |
05:21.49 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
05:22.42 | Vco | **shrug**, maybe year or 2 now...more seriously for the last 6-8 months |
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05:46.27 | genmud | escribzz what part of phoenix you in? |
05:46.49 | escribzz | Genmud: North Phoenix |
05:46.56 | genmud | Coo |
05:47.00 | escribzz | Where are you at? |
05:47.03 | genmud | I am like 13th and Union Hills |
05:47.35 | escribzz | I'm at 23rd & Happy Valley lol just down the road |
05:47.42 | genmud | yea |
05:47.44 | genmud | that is funny |
05:47.56 | genmud | where do you work for? |
05:48.03 | *** join/#asterisk Tili (n=Tili@218.19.1.157) |
05:48.33 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET) |
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05:55.27 | Qwell | oej: morning |
05:58.15 | paxr0 | i have a question, im tryng to use DIALSTATUS before Originate, the print of NoOp(${DIALSTATUS}), why i can know if the called answer , or this variable it only por use in AGI ? anybody can suggest me ? |
05:58.37 | paxr0 | the print of NoOp(${DIALSTATUS}) before Originate is "" |
05:59.13 | paxr0 | im tryng to use DIALSTATUS in dialplan for outcall.... |
05:59.22 | blitzrage | oej: morning! |
05:59.39 | Qwell | paxr0: What version of asterisk? |
05:59.44 | Qwell | DIALSTATUS is fairly new, isn't it? |
06:00.31 | paxr0 | hummm |
06:00.39 | paxr0 | 1.0.9 |
06:00.40 | Qwell | Don't quote me on that |
06:00.58 | Qwell | paxr0: read doc/README.variables in the asterisk source for 1.0.9 |
06:01.10 | blitzrage | night all |
06:01.12 | paxr0 | ok thanks |
06:01.14 | paxr0 | night |
06:01.50 | infinity1 | wow. fun day updating voip-info |
06:07.16 | littleball | hello, any relationship between extension h and T? |
06:07.28 | Qwell | littleball: just that they happen after certain events |
06:09.00 | littleball | Qwell, because i need to trigger DeadAGI at the end of call. It seems sometimes, DeadAGI is triggerd by T and other times, it is triggered by h |
06:09.22 | Qwell | t is timeout. If its waiting for something, and it never happens, thats where it goes |
06:09.22 | littleball | Qwell, i need to make sure the DeadAGI only run onetime |
06:09.28 | Qwell | if the call is hung up, it goes to h |
06:09.51 | littleball | after T, does it to to h? |
06:09.59 | *** join/#asterisk power1 (n=marktren@rndf-146-4-251.telkomadsl.co.za) |
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06:15.10 | wunderkin | T is absolute timeout |
06:15.37 | *** part/#asterisk santiago (n=santiago@208.195.215.158) |
06:16.57 | LostFrog | test it by putting a NoOp in each priority. |
06:17.04 | LostFrog | And watching the log. |
06:19.04 | infinity1 | tzafrir_laptop: thanks! |
06:19.15 | infinity1 | tzafrir_laptop: -Brendon |
06:24.21 | *** join/#asterisk mkl1525 (n=daniel@84.19.198.194) |
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06:37.20 | LostFrog | I should be able to provide settings to a snom phone using tftp, shouldn't I? |
06:39.23 | Inv_arp | LostFrog: is that what the directs say? |
06:39.33 | Inv_arp | directions* |
06:40.07 | *** join/#asterisk e-Hernick (n=ncc@modemcable120.39-131-66.mc.videotron.ca) |
06:44.05 | LostFrog | I think I found what I needed. |
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07:00.51 | argos73 | any echo gurus around? (not a usual "why does my x100p have echo" question) |
07:04.32 | Guggemand | argos73 try asking what you wanna know, maybe soneone have the answer :) |
07:04.43 | argos73 | heh - that makes too much sense... :) |
07:04.48 | argos73 | here ya go |
07:06.38 | argos73 | given a T1 going into an adtran 750 channel bank, with some channels passed to the FT1 port to a digium T1 card, echo is still a problem. phones are SIP hardphones. can the same be expected if I was to bring a PRI directly into the digium card? |
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07:15.18 | giesen | anyone alive that could lend a hand and perhaps a thought? |
07:15.33 | giesen | I get - wrong password on authentication for INVITE |
07:15.46 | giesen | when I dial between to cisco 7940's |
07:16.01 | giesen | but I can dial between an Xlite soft phone and one of the 7940's no problem. |
07:16.25 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
07:16.30 | mkl1525 | Hi, trying to setup asterisk with a hfc-s card. When starting with asterisk -vvv -c it quits with "chan_modem.c:394 modem_setup: Unable to autodetect modem." so it seems that it tries to open some modem is this the right way for a zapata card or is there some misconfiguration? |
07:17.27 | X-Rob | mkl1525 - you're not using chan_modem |
07:17.31 | X-Rob | that's not the error you care about. |
07:18.23 | *** join/#asterisk cj- (n=jrocha@CPE0040ca7bd059-CM001225db967c.cpe.net.cable.rogers.com) |
07:19.14 | cj- | Hey guys, is Asterisk seg faulting on rxfax a known issue? |
07:19.23 | X-Rob | yes 8) |
07:19.27 | X-Rob | usually because you haven't run ldconfig |
07:19.45 | X-Rob | check where spandsp.so is, and make sure it's in /etc/ld.so.conf and then run ldconfig |
07:20.35 | cj- | Already done :( |
07:20.43 | X-Rob | check /var/log/asterisk/full |
07:21.46 | cj- | Do I have to run in debug or soemthing to get that log? |
07:21.56 | X-Rob | enable it in /etc/asterisk/logger.conf |
07:22.26 | cj- | aha... Brilliant... LEt's see if I get anything now.. (reproduces crash) |
07:24.06 | mkl1525 | X-Rob, this means in my configuration I wouldn't need chan_modem? Tried to comment chan_modem out in modules.conf but starting asterisk still fails with "WARNING[4542]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_modem_bestdata.so: undefined symbol:ast_unregister_modem_driver" |
07:25.04 | X-Rob | mkl1525 - put noload=>chan_modem.so in modules.conf |
07:25.31 | X-Rob | chan_modem_bestdata.so? |
07:25.33 | X-Rob | just delete it |
07:25.43 | X-Rob | rm /usr/lib/asterisk/modules/chan_modem_bestdata.so |
07:25.56 | X-Rob | it's some old module that's been left lying around |
07:26.43 | cj- | Nothing useful in full :( |
07:27.16 | cj- | Essentially it seems to be related to large tif files... I think. Although it's nothing I'd call 'massive'.. It's 400k |
07:27.19 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
07:27.28 | X-Rob | cj- oh. I thought it was an onload crash |
07:27.29 | cj- | (I'm using txfax to send it back to myself) |
07:27.33 | X-Rob | make sure you're using the correct libtiff |
07:27.37 | X-Rob | that causes that problem |
07:28.00 | cj- | Which one is the correct one? :) I'm running 3.7.3 |
07:28.10 | X-Rob | it's on the wiki. voip-info.org |
07:28.13 | af_ | what may I put as "Application" in originate? (manager interface) |
07:28.24 | cj- | *checks* |
07:28.39 | X-Rob | af - on the command line, 'show applications' |
07:29.06 | *** join/#asterisk paxr0 (n=P@200-85-200-140.bk3-dsl.surnet.cl) |
07:29.14 | af_ | I see, so any of the cli commands? |
07:29.21 | af_ | that's cool |
07:29.34 | af_ | mhhh, what could be used for? |
07:31.08 | cj- | oh shit I'm a moron |
07:31.17 | cj- | I updated my libtiff during a recent emerge :( |
07:32.25 | *** join/#asterisk eivindtr (n=wingnut-@194.248.208.94) |
07:33.17 | X-Rob | ooh, I know this one. Gentoo sucks! |
07:33.47 | Igbothom_III | rofl |
07:34.25 | Igbothom_III | vacuum cleaners do, as to their salespeople |
07:35.01 | orlok | Hey |
07:35.07 | orlok | is anybody using a cisco 7912? |
07:35.31 | orlok | I've just managed to upgrade a 7940 to a sip image, but i forget now how the hell i made it get the new image! |
07:35.41 | Igbothom_III | tftp |
07:36.19 | cj- | lol I've actually only been using Gentoo for like 2 weeks now (in a larger role that I previously used it for)... Since I'm a FreeBSD guy I sort of like it... I'm chalking this one up to me being careless :) |
07:36.36 | Igbothom_III | FreeBSD is nice |
07:36.47 | mkl1525 | X-Rob, thanks for the help but no it complains about chan_modem_aopen.so and if I move this "/usr/lib/asterisk/modules/chan_modem_i4l.so: undefined symbol: ast_unregister_modem_driver" so it seems as there is still a reference to chan_modem although I inserted noload => chan_modem.so in modules.conf - any further suggestions? |
07:36.51 | cj- | Okay, I just compiled 3.5.7... now to test :) |
07:37.09 | X-Rob | mkl1525 - you can happily delete any chan_modem* files |
07:37.11 | X-Rob | you don't need 'em. |
07:37.23 | orlok | Igbothom_III: yes, i know that bit.. i'm wondering how i make the phone reqyest the image? |
07:37.30 | orlok | Igbothom_III: or do i specify a file via dhcp? |
07:41.39 | cj- | hmm, that seems to have fixed the crashing... It still doesn't accept the transmission though :( |
07:42.28 | *** join/#asterisk wjn78 (n=wjn78@adsl-68-123-233-181.dsl.irvnca.pacbell.net) |
07:42.41 | cj- | Although I'm doing some weird things with my fax |
07:43.31 | cj- | I'm basically working on a email -> fax gateway... I have tif's working properly, but now I got greedy and want gifs. So I'm converting the gifs to tifs but ... oh it crashed again... when it gets the large tif it craps out :( (the whole server dies) |
07:43.48 | orlok | what gif-tiff util? |
07:43.49 | Igbothom_III | pdf is so much nice |
07:43.52 | Igbothom_III | nicer |
07:43.54 | orlok | imagemagic? |
07:43.57 | orlok | yes, PDF's rock |
07:44.19 | Igbothom_III | lol |
07:44.20 | cj- | I tried a few... I used the gif2tiff util but that wouldn't detect any gifs properly |
07:44.35 | argos73 | orlok: IIRC, my 7960 requests OS79XX.TXT from tftp, which contains the requested software version (POS3-06-1-00). if the phone is not running that version, it grabs it via tftp and updates. |
07:44.38 | cj- | so I used a program called a2ping.pl which is a front end to several utils |
07:44.47 | Igbothom_III | I'm not wearing my "deCSS" t-shirt, but I *did* support the guy |
07:44.50 | orlok | argos73: yeha, thats what the 7940 did |
07:44.57 | cj- | I have my incoming faxes convert from tif to pdf already |
07:45.02 | cj- | I'm trying to do the opposite |
07:45.05 | orlok | tcpdumping tftp, havent seen any requests for it |
07:45.08 | cj- | My outgoing have to end up as a tif :) |
07:45.22 | orlok | Igbothom_III: i have to wear this. Work. |
07:45.29 | Igbothom_III | tif is just sooooooo big |
07:45.35 | argos73 | orlok: "for it" - OS79XX.TXT or POS3.... ? |
07:45.39 | Igbothom_III | where ewe work? |
07:45.58 | cj- | txfax won't send in anything other than tif though |
07:46.01 | cj- | (TMK?) |
07:46.17 | *** join/#asterisk kks (n=kks@202.73.8.130) |
07:46.57 | orlok | argos73: i get requests for SEP000E833CB763.cnf.xml |
07:47.35 | cj- | Oh well, time for bed :( |
07:47.36 | orlok | oh gno, you know my mac address, dont maxxor me! |
07:47.37 | orlok | ;) |
07:47.44 | orlok | haxxor, even |
07:47.47 | argos73 | hehe |
07:48.09 | orlok | 7940 is working well |
07:48.13 | orlok | but my boss took it home |
07:48.36 | orlok | Igbothom_III: place that specialises in stuff for the graphic arts industry |
07:49.03 | argos73 | the xml request should happen after the image request.. |
07:49.17 | argos73 | i think |
07:49.18 | orlok | argos73: yeah, i thought so |
07:49.23 | argos73 | lemme look |
07:49.29 | Igbothom_III | orlok; cool, what drugs in particular? |
07:49.40 | orlok | heheh |
07:50.03 | orlok | hmm.. hye, another aussie ;) |
07:50.09 | X-Rob | what? |
07:50.09 | X-Rob | wherE? |
07:50.10 | argos73 | my SIP...xml file does have a line in it - "image_version: P0S3-06-1-00" |
07:50.15 | orlok | hey, wtf, you are on #centos as well! :) |
07:50.29 | Igbothom_III | yup |
07:50.43 | argos73 | random memories tell me that the "image_version" line is important |
07:50.43 | X-Rob | woo aussies! |
07:50.45 | X-Rob | woo! |
07:50.49 | X-Rob | we r0xx0r. |
07:50.56 | Igbothom_III | I'm an Australyanmate! |
07:50.56 | orlok | beer oclock |
07:50.59 | orlok | and we all hit irc |
07:51.05 | Igbothom_III | yup |
07:51.18 | X-Rob | I've been here all damn day |
07:51.30 | Igbothom_III | you hear that "beer o'clock" is now in the Macquarie Dictionary? :) |
07:51.42 | Igbothom_III | I've been onsite for half of it |
07:51.51 | Igbothom_III | office for the rest |
07:52.16 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
07:54.33 | paxr0 | asterisk not load app_dial.so, i think i need this for DIALSTATUS anybody can suggest ? |
07:54.44 | X-Rob | you need app_dial |
07:54.48 | X-Rob | that's the most important application |
07:55.01 | paxr0 | :( |
07:55.13 | Igbothom_III | anyone else having issues with their IAX connexion to fwd? Mine remains <unregistered> |
07:55.57 | paxr0 | asterisk send errors to the log whe i try to load |
07:56.12 | X-Rob | which errors? |
07:56.15 | X-Rob | ~pb |
07:56.16 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
07:56.20 | X-Rob | ^^^ put them there |
07:57.37 | paxr0 | Oct 24 09:22:15 WARNING[3443]: /usr/lib/asterisk/modules/app_dial.so: undefinedsymbol: ast_bridge_call Oct 24 09:22:15 WARNING[3443]: Loading module app_dial.so failed! |
07:58.00 | X-Rob | paxr0 - recompile everything. |
07:58.06 | X-Rob | your modules are out of date. |
07:58.29 | paxr0 | ok thanks :) |
07:58.42 | X-Rob | paxr0 - btw, use CVS HEAD |
07:58.47 | X-Rob | it's much better than 1.0 |
07:59.00 | *** join/#asterisk rene- (n=rene-@201.144.60.6) |
08:00.08 | paxr0 | ok thanks, really i will follow ur suggest |
08:01.03 | *** join/#asterisk Guest^DJ (i=me@60.48.51.109) |
08:01.21 | Guest^DJ | hi all, when a user call into *, how can i give the user a dial tone ? |
08:01.38 | X-Rob | Guest^DJ - the magic command is DISA |
08:02.12 | Guest^DJ | X-Rob: DISA? where can i find more info ? |
08:02.35 | X-Rob | 'show application DISA' |
08:02.44 | X-Rob | on your asterisk console |
08:03.17 | littleball | hi, is the default context must be defined in the extensions.conf ? |
08:03.30 | X-Rob | littleball - try again? |
08:03.34 | Guest^DJ | X-Rob: thanks |
08:03.38 | *** join/#asterisk Miggidy (i=user@dsl-202-72-180-171.wa.westnet.com.au) |
08:04.02 | littleball | X-Rob, must i define the default context in the extensions.conf? |
08:04.25 | X-Rob | littleball - no. The default context is defined in your sip.conf or iax.conf or zapata.conf |
08:04.28 | X-Rob | uh |
08:04.30 | X-Rob | let me rephrase that |
08:04.35 | X-Rob | the _context_ of the client is.. |
08:04.44 | X-Rob | if there's no context specified, it uses [default] in extensions.conf |
08:05.13 | littleball | X-Rob, what i mean is the [default] compulsary? |
08:05.55 | shido6 | how do you replace all occurances of a string in vi |
08:05.56 | shido6 | ? |
08:05.57 | X-Rob | well, no |
08:06.10 | X-Rob | but if you dont' specify a context and there's no default, I'm not sure what happens |
08:06.18 | X-Rob | shido: ':%s/string1/string2/g' |
08:06.27 | X-Rob | that's through the entire file |
08:06.33 | X-Rob | drop the % for just that line |
08:06.38 | littleball | X-Rob, if [default] is not in the extensions.conf, and if there is no context specified in the sip.conf etc, what will happen |
08:06.45 | X-Rob | I don't know. |
08:06.46 | X-Rob | don't do it 8) |
08:07.05 | littleball | X-Rob, yes, i just drop it. |
08:07.07 | littleball | thanks |
08:07.23 | littleball | [default] |
08:07.23 | littleball | exten=>i,1,Hangup |
08:07.26 | littleball | is this enouth? |
08:07.27 | glm2k | Igbothom_III: iax2 thru fwd status UNREACHABLE :( |
08:07.35 | littleball | s/enouth/enough |
08:07.47 | Igbothom_III | glm2k; yeah, seems they are having issues |
08:07.53 | Igbothom_III | mentioned on their forum too |
08:08.06 | shido6 | X-rob |
08:08.19 | shido6 | I want to replace /nagios/cgi-bin with /cgi-bin |
08:08.20 | shido6 | in vi |
08:08.21 | X-Rob | littleball - um. should be exten => _X.,1,Playback(weasels) then _X.,2,Hangup |
08:08.29 | shido6 | so do i use the '/' you mentioned? |
08:08.37 | X-Rob | so if anyone ever mentions hearing 'weasles have eaten our phone system!' you know there's a problem |
08:09.16 | X-Rob | shido6 - ':%s/nagios\///g' |
08:09.20 | littleball | ok |
08:09.24 | littleball | thanks |
08:10.51 | littleball | X-Rob, weasels is already included in the system? |
08:11.21 | X-Rob | I think so |
08:11.26 | X-Rob | check /var/lib/asterisk/sounds |
08:11.35 | X-Rob | you may need to download asterisk-sounds from cvs |
08:11.37 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:12.46 | littleball | X-Rob, weasels is not there. but there is beep |
08:13.00 | X-Rob | whatever you want 8) |
08:13.13 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
08:13.19 | *** join/#asterisk TMirage (n=mirage@cust.12.229.adsl.cistron.nl) |
08:13.25 | kks | if two clients in the same LAN registered at a public asterisk server and want to call each other. Will the asterisk server detect those clients are from same LAN, like MSN do??? |
08:13.57 | X-Rob | kks - it will try to bridge the connection between the phones no matter what. |
08:14.14 | X-Rob | (unless you specify no briding, by using tT or something els eon the dial string, or various other reasons) |
08:17.39 | glm2k | argos73: lol. yep. just a few minutes alright. |
08:17.54 | kks | X-Rob, thanks |
08:18.11 | glm2k | argos73: use the 24 bucks for something else |
08:18.30 | argos73 | glm2k: it's a pair of connectors with a capacitor in the middle.... $25 is crazy |
08:18.37 | glm2k | aye |
08:19.27 | glm2k | argos73: you's paying for the robot's labor hehe |
08:19.39 | glm2k | er, you're |
08:19.40 | argos73 | of course, i'll need to pick up a $3 SV to SV cable from walmart to hack apart... :) |
08:19.43 | orlok | and nicer connectors |
08:19.45 | orlok | and a clean job |
08:19.51 | orlok | and not worrying about breaking solder join |
08:19.53 | orlok | s |
08:20.01 | argos73 | heh - my soldering is picture-perfect... :) |
08:20.04 | glm2k | heh, it all depends on your soldering skill. |
08:22.23 | *** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) |
08:26.15 | mkl1525 | X-Rob, thanks for the help so far after moving all chan_modem modules asterisk is starting again and works. But I have another problem: my configuration is sip-phones <-> asterisk <-> elmeg pbx <-> ISDN provider, I'm able to call from my sip phones to the outside world but asterisk doesn't react on incoming calls so I'll have to assign a msn is this done in capi.conf or another file? |
08:26.54 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
08:28.18 | *** join/#asterisk brasco (n=root@83.137.128.7) |
08:28.35 | X-Rob | mkl1525 - sorry, out of my league there. |
08:30.14 | mkl1525 | X-Rob, don't worry - you helped me a lot thanks for this! |
08:37.22 | paxr0 | X-Rob, u think an update to asterisk using CVS and make update work fine? |
08:37.39 | X-Rob | paxr0 - yep. |
08:38.00 | paxr0 | thanks a lot |
08:40.36 | encode | hi - i'm trying to get asterisk to register with a voip provider (engin in australia), and i'm getting in my log the following message: "SIP response 423 "Interval Too Brief"" |
08:40.39 | encode | any ideas? |
08:41.28 | X-Rob | encode - odd. |
08:41.57 | littleball | hello, what is the function of the context field in zapata.conf? |
08:42.08 | mkl1525 | So does anybody know where I have to configure the msn on which asterisk should listen if I use a hfc-s card in combination with the bristuff drivers? |
08:43.18 | littleball | hi, mkl1525, what is hfc-scard? |
08:43.55 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
08:44.01 | christo | morning all |
08:44.30 | festr__ | hello, i've problem with sending calleridpresentation over IAX2 channel, is it possible this? anyone is using this? |
08:44.45 | mkl1525 | littleball, it's a isdn card using a hfc-s chip to process the isdn data |
08:45.31 | littleball | mkl1525, like digium card? |
08:46.05 | encode | X-Rob: got any more insight than just odd? |
08:48.43 | X-Rob | encode, I think it's saying 'you're trying to log in too rapidly' |
08:48.57 | X-Rob | but I don't use engin |
08:48.58 | X-Rob | sorry 8) |
08:49.04 | X-Rob | faktortel is the best I've found |
08:49.09 | X-Rob | as they actively support asterisk |
08:49.19 | encode | but...how can i be trying to log on too fast?? |
08:49.24 | encode | never mind... |
08:49.27 | encode | theres an engin forum |
08:49.37 | encode | which would be great, if i could sign up |
08:49.56 | encode | stupid website wont let me log on with either my phone number of my account number |
08:50.02 | encode | both of which are supposed to work |
08:50.07 | encode | anyway, bbl |
08:50.29 | kll | how do I get asterisk going with some form of zaptel dummy on sparc/linux? zaptel won't compile properly.. any pointers? |
08:50.37 | X-Rob | kll - you can't. |
08:50.49 | X-Rob | ztdummy only works on x86 |
08:50.59 | mkl1525 | littleball, don't know if this is comparable to the digium cards |
08:51.52 | kll | X-Rob: so any corresponding for sparc? I've heard of zaprtc but I'm unable to find much more info on it.. |
08:54.29 | X-Rob | rtc only works on intel, too |
08:54.37 | X-Rob | unfortunately, you're outta luck for timing on sparc |
08:54.46 | X-Rob | you got a PCI bus in the spac? |
08:54.48 | X-Rob | sparc? |
08:58.02 | christo | I have a media gateway receiving a call over E1. It then routes the call over IAX to an application server which runs an AGI. For some reason the call just cuts off in the middle and I can't work out why |
08:58.06 | christo | can anybod think of a reason for this? |
08:58.32 | christo | It seems to happen randomly - sometimes not at all and always at different places |
08:59.59 | christo | could it be the quality of the streamed sound file? I mean if it's too quiet, or if there is a gap, the IAX connection might just drop it |
09:04.59 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
09:06.26 | kll | X-Rob: yupp, got one PCI... |
09:08.54 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
09:10.46 | *** join/#asterisk trym (n=trym@cD9088B17.sdsl.catch.no) |
09:30.34 | *** join/#asterisk |baby| (n=lere@62-43-207-229.user.ono.com) |
09:31.06 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
09:35.31 | *** join/#asterisk SlickRick (n=dams@hjp.envia-tel.net) |
09:37.06 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
09:38.30 | *** join/#asterisk DeeJay01 (n=icechat5@S0106006067664673.wp.shawcable.net) |
09:38.47 | *** part/#asterisk rene- (n=rene-@201.144.60.6) |
09:40.34 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
09:49.04 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.15) |
09:49.08 | MuppetMaster | Hello |
09:49.17 | MuppetMaster | Anyone had a chance to download and test http://www.astjab.org? |
09:52.37 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
10:05.41 | *** join/#asterisk jbenson (n=jbenson@genpubad.gotadsl.co.uk) |
10:08.05 | mkl1525 | because I don't get any notice of incoming isdn calls, is there a way to look at the isdn channels, which data is transmitted on them? |
10:08.13 | mkl1525 | using zaphfc driver |
10:08.52 | jbenson | Hi all. Does anyone use the ZyXEL P662HW ADSL router please, with the bandwidth management option? We are trying to convince ZyXEL that the SIP support just doesn't work. |
10:12.51 | *** join/#asterisk cinzas (n=ashes@62.48.254.104) |
10:12.53 | cinzas | Howdy |
10:13.34 | MuppetMaster | Howdy |
10:18.35 | *** join/#asterisk ful|work (n=fulgas@209.8.233.227) |
10:19.00 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
10:19.06 | |baby| | alguien habla español? xD |
10:19.13 | *** join/#asterisk dfordivam (n=dfordiva@59.176.52.35) |
10:19.16 | MuppetMaster | solo un poquito |
10:19.24 | MuppetMaster | Comprende mucho, hablo poquito |
10:19.27 | *** part/#asterisk dfordivam (n=dfordiva@59.176.52.35) |
10:19.27 | cinzas | |baby|: Portugues. Pero te comprendo |
10:19.50 | |baby| | jeje, gracias! |
10:19.53 | cinzas | I'm having some strange behavior with asterisk 1.0.8 |
10:20.36 | |baby| | mientas estoy transfiriendo una llamada usando "atxfer" como puedo cancelarla y recuperar la llamada? |
10:21.48 | X-Rob | si |
10:22.03 | cinzas | This is my diaplan |
10:22.05 | cinzas | exten => s,1,Answer() |
10:22.05 | cinzas | exten => s,2,SetLanguage(pt) |
10:22.05 | cinzas | exten => s,4,SayDigits(${LINHA}) |
10:22.05 | cinzas | exten => s,5,wait(2) |
10:22.05 | cinzas | exten => s,6,Playback(linha) |
10:22.06 | cinzas | exten => s,7,SayDigits(${LINHA}) |
10:22.24 | cinzas | And this is what happens |
10:22.27 | cinzas | -- Executing SayDigits("Local/1234@testes-ea9a,1", "100") in new stack |
10:22.27 | cinzas | <PROTECTED> |
10:22.27 | cinzas | <PROTECTED> |
10:22.27 | cinzas | <PROTECTED> |
10:22.36 | MuppetMaster | |babu|: Have a look here (as I have never used atxfer myself) http://www.voip-info.org/wiki-Asterisk+config+features.conf |
10:22.56 | MuppetMaster | I do that with a flash hook on an analog line on a Sipura ATA, works the same. |
10:22.58 | cinzas | Sometimes it stays correct, but sometimes "asterisk" starts playing the messages in english |
10:23.33 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
10:23.37 | cinzas | |baby|: No te puedo ayudar, pues nunca trabajei com atxfer |
10:23.56 | |baby| | cinzas no pasa nada ;) |
10:26.34 | cinzas | Anyone knows why it changes my Language from pt to en ? |
10:27.27 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.15) |
10:28.30 | paxr0 | i have recompiled asterisk for a upgrade from CVS, all appear good ....but has not created chan_zap.so anybody can suggest me why my system haven t chan_zap.c? |
10:28.45 | |baby| | cinzas haz un ln a "digits" en ingles al pt... y listo |
10:29.25 | X-Rob | paxr0 - you need to download zaptel (and libpri, if needed) from cvs and install that too |
10:29.52 | cinzas | |baby|: Si. Esso as una possibilidad. Pero my sistema tiene que funcionar en varias lenguas. |
10:30.08 | cinzas | |baby|: y me gustaria saber lo que passa :( |
10:30.10 | paxr0 | ok X-Rob thanks a lot again :) |
10:30.45 | cinzas | I just doesnt understande why it changes the Language ... |
10:31.04 | |baby| | cinzas si t sirve de consuelo a mi tambien me ocurre lo mismo que a ti... |
10:31.18 | cinzas | HeHe |
10:32.38 | cinzas | Return Code |
10:32.38 | cinzas | Returns -1 if the channel was hung up, or if the file does not exist. Returns 0 otherwise. |
10:32.46 | cinzas | <PROTECTED> |
10:33.12 | cinzas | The file exists ... and the channel exists too |
10:36.51 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
10:41.32 | shido6 | any nagios nuts |
10:48.42 | *** join/#asterisk Hotfurry (n=bharatsa@210.211.246.47) |
10:48.50 | Hotfurry | hello there |
10:49.29 | Hotfurry | i am facing the problem of relocation error when installing the mysql support to the Asterisk.. |
10:49.57 | Hotfurry | does anybody know about the installing the mysql support for Asterisk |
10:50.03 | Hotfurry | please tell me |
10:50.48 | puppet | its just to compile it |
10:51.01 | puppet | of couse u need the mysql libs |
10:51.05 | wasim | Hotfurry: asterisk_addons |
10:53.01 | paxr0 | # apt-cache search asterisk|grep sql |
10:53.02 | paxr0 | asterisk-sqlite - sqlite support for the Asterisk PBX |
10:53.06 | paxr0 | :) |
10:55.16 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
10:56.23 | cinzas | Hotfurry: what error ? |
10:58.37 | puppet | ~pastebin |
10:58.39 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
10:58.49 | puppet | Hotfurry: paste your error there |
11:09.42 | Hotfurry | ok guys |
11:09.47 | Hotfurry | let me paste the error |
11:10.08 | Hotfurry | asterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_copy_string |
11:10.16 | Hotfurry | this is the error |
11:10.18 | Hotfurry | i am getting |
11:10.51 | Hotfurry | can anybody help me figure out whats the error because of |
11:12.18 | Ahrimanes | seems it's unable to find the asterisk include files? |
11:16.19 | cinzas | hotfurry when it happens ? |
11:18.32 | *** join/#asterisk wmandra (i=wmandra@pcp04943183pcs.verona01.nj.comcast.net) |
11:19.59 | Hotfurry | cinzas: this happens when the asterisk is getting loaded.. |
11:20.44 | Hotfurry | the res_config_mysql.so is present in the modules directory... |
11:20.52 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
11:21.08 | Hotfurry | i installed it from the CVS... |
11:21.12 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
11:23.00 | cinzas | do a make clean |
11:23.04 | cinzas | and recompile it |
11:23.23 | cinzas | lunch time |
11:26.33 | Hotfurry | i tried doing that |
11:26.42 | Hotfurry | but still I am getting the same errot |
11:26.44 | Hotfurry | error |
11:27.18 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
11:27.23 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
11:27.30 | drumkilla | what version of asterisk are y0ou using |
11:27.35 | Inkubot | good morning |
11:31.38 | Hotfurry | Asterisk CVS-HEAD-05/12/05-02:17:19, Copyright (C) 1999 - 2005 Digium. |
11:35.39 | drumkilla | ast_copy_string probably came after 5/12/05 |
11:35.42 | drumkilla | you need to update asterisk. |
11:39.45 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
11:40.01 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
11:40.02 | puzzled | morning all |
11:46.03 | *** join/#asterisk scubasteve (n=steve@cpe-071-065-212-199.nc.res.rr.com) |
11:46.33 | mkl1525 | can anybody give me the name of the voicemail box tool that added the boxes on the filesystem? |
11:46.46 | scubasteve | Hey guys! I need some help troubleshooting a SIP registration problem. I've got two phones outside of my network (one at a family member's house and one in my camper outside, bridged wireless) and both are having issues. |
11:47.04 | oej | Outside of a NAT, scubasteve? |
11:47.10 | scubasteve | This started last week when I noticed the Linksys adapter at my mom's was causing stale nonce errors every few seconds. |
11:47.25 | scubasteve | oej, yes, but I have a linux firewall that's been fine and unchanged for quite some time |
11:47.45 | scubasteve | My SIP softphone at work was doing the same thing last week... gave up after a while and poof it started working. |
11:47.53 | scubasteve | Installed HEAD over the weekend and it's doing it agian. |
11:48.07 | scubasteve | One adapter gets stale nonce errors all day long, the other doesn't cause * to complain at all. |
11:48.15 | scubasteve | It just registers and registers over and over ... |
11:49.01 | scubasteve | I'd like to dump the sip debug to pastebin and see what you folks think |
11:50.22 | scubasteve | http://pastebin.ca/26472 |
11:54.05 | *** join/#asterisk paxr0 (n=C@200-85-216-219.bk4-dsl.surnet.cl) |
11:59.49 | scubasteve | Name/username Host Dyn Nat ACL Port Status |
11:59.49 | scubasteve | anybody? |
12:01.25 | drumkilla | ok, well, you can probably safely ignore the stale nonce messages ... |
12:01.34 | drumkilla | as for registering over and over, that is what is supposed to do |
12:01.44 | scubasteve | drumkilla, well... I would if the linksys at my mom's house would register. |
12:01.46 | drumkilla | it re-registers within the negotiated registration timeout |
12:02.11 | scubasteve | Drumkilla, I see nonce warnings every 10 seconds or so. |
12:02.12 | drumkilla | ah, well I didn't catch that part :) |
12:02.18 | drumkilla | that it won't reg ... |
12:02.23 | drumkilla | silly SIP |
12:02.32 | scubasteve | And the one on pastebin... doesn't reg at all and causes no warnings or anything on the console. |
12:03.48 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
12:04.05 | scubasteve | It looks like the linksys at mom's house isn't doing the nonce warnings this morning...it is doing the same as the other adapter I put on pastebin |
12:04.52 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
12:05.42 | *** join/#asterisk kshumard (n=root@gateway.digium.com) |
12:07.46 | *** join/#asterisk Lathos42 (n=Lathos42@adsl-69-208-243-229.dsl.lgtpmi.ameritech.net) |
12:09.56 | wmandra | morning all |
12:10.05 | paxr0 | morning |
12:10.07 | scubasteve | Hey, does the latest CVS HEAD have the new iax2 jb in it? |
12:10.11 | scubasteve | Or is that 1.2 ? |
12:13.26 | Guggemand | i would guess the latest CVS HEAD is somewhere after 1.2 beta1 |
12:16.21 | *** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net) |
12:16.34 | docE | YA BUDDY How's the hurricane?!?!?! |
12:17.54 | scubasteve | cool. thanks. |
12:18.40 | docE | Your not even here.. go sit in a corner.. |
12:23.01 | Inkubot | anyone use flash operator panel ? |
12:23.11 | *** join/#asterisk hatems (n=hatems@196.203.15.218) |
12:23.12 | docE | not here.. |
12:23.17 | hatems | hi all |
12:23.24 | docE | hola |
12:24.00 | *** join/#asterisk |baby| (n=jrp@62-43-207-229.user.ono.com) |
12:24.00 | X-Rob | Inkubot - try #amportal - AMP contains FOP |
12:24.05 | mkl1525 | there is CALLERID and CALLERIDNUM are there also variables like CALLEDID or CALLEDIDNUM? |
12:24.21 | X-Rob | mkl1525 - CALLEDID == EXTEN |
12:24.30 | X-Rob | You're matching that. |
12:24.42 | docE | ya more or less.. |
12:24.43 | Inkubot | X-Rob: thnks |
12:24.59 | docE | Unless your coding a c app then you get all the variables you need.. |
12:25.21 | |baby| | transfering calls, no ringing sent to caller |
12:25.21 | |baby| | I'm transfering an extension or number to, |
12:25.21 | |baby| | doesn't hear any ringing. When |
12:25.33 | |baby| | Anybody have any ideas how I can resolve this? |
12:25.53 | docE | add a r to your dialplan |
12:26.14 | docE | DIAL(Tech/${EXTEN},timeout,r) |
12:26.46 | tzanger | I'm pretty sure you shouldn't need 'r' in that case... what technology? |
12:27.05 | docE | You shouldnt but he asked how to make it produce a ring.. :) |
12:27.17 | tzanger | that's not a very helpful answer... |
12:27.52 | |baby| | -- Executing Dial("Local/100@from-internal-3598,2", "SIP/100|50|trR") in new stack |
12:27.52 | |baby| | -- Called 100 |
12:27.52 | |baby| | -- SIP/100-3a02 is ringing |
12:28.00 | tzanger | ok it's sip |
12:28.32 | tzanger | what's the 'R' for? |
12:28.42 | tzanger | I don't see an 'R' flag in my dial help |
12:28.53 | docE | ya I dont remember that one either |
12:29.11 | tzanger | unless he meant Ttr |
12:29.18 | tzanger | instead of trR |
12:29.20 | |baby| | this position the r... and does not generate tone when transferring using to atxfer |
12:30.10 | docE | True.. would allow transfers and make it ring.. but if its not then I dunno.. |
12:30.19 | X-Rob | baby - that's not how attended transfers work. |
12:30.20 | tzanger | |baby|: what is the 'R' for |
12:30.28 | docE | Man is it howling outside.. |
12:30.34 | tzanger | docE: where are you? FL? |
12:30.38 | X-Rob | they'll be listening to music on hold while you're transferring. |
12:30.39 | docE | Hurricane party in Tampa! |
12:30.51 | docE | I thought that was m |
12:31.26 | tzanger | yeah that is m |
12:31.33 | X-Rob | no, but he's doing an attended transfer. |
12:31.37 | X-Rob | (this is what he said) |
12:31.48 | tzanger | yeah attended transfer I thought was different |
12:31.55 | |baby| | http://archives.free.net.ph/message/20050930.235856.7d2bcce3.en.html |
12:32.01 | |baby| | it happens to me just like this person, you excuse my english |
12:32.29 | X-Rob | |baby| - your musiconhold is broken. |
12:32.57 | docE | My MOH sucks.. elevator music.. |
12:32.59 | tzanger | I have not used atxfr myself yet |
12:33.03 | docE | Im putting in some light trance.. |
12:33.11 | tzanger | I think it was blitzrage who had spanish flea for his MOH |
12:33.38 | docE | figures.. :) Hes crazy like that.. T were you at the conference? |
12:33.49 | tzanger | no I wasn't :-( |
12:33.56 | docE | man did you miss out.. |
12:34.00 | tzanger | yeah I bet |
12:34.16 | docE | Most of the IRC crew hung out.. Was a real good experience to meet and greet |
12:34.19 | X-Rob | |baby| - attended transfer does not ring. It puts the caller on hold. This is what attended transfer does. fix your musiconhold. |
12:34.40 | tzanger | awesome |
12:34.51 | tzanger | I go to the toronto asterisk user's meetings on occassion |
12:35.11 | docE | Well Im off to work on my companies new PBX I finally talked them into asterisk after a year.. |
12:36.14 | |baby| | X-Rob how it's possible that a ringtone doesn't sound when I pull an extension number? |
12:36.29 | tzanger | |baby|: you're not listening |
12:36.46 | X-Rob | tzanger - english as a third language. |
12:36.47 | X-Rob | be nice. |
12:36.50 | tzanger | |baby|: attended transfers put the person you're transferring on hold, and they should hear your music on hold |
12:36.53 | tzanger | I am being nice |
12:36.56 | X-Rob | you are |
12:37.02 | X-Rob | I thought you were going to go off at him |
12:37.02 | tzanger | |baby|: you need to set up your music on hold |
12:37.02 | X-Rob | 8) |
12:37.08 | tzanger | X-Rob: :-) |
12:38.22 | |baby| | X-Rob speak spanish? noo?:( |
12:38.35 | |baby| | tzanger ? speak spanish? :P |
12:38.40 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
12:38.47 | X-Rob | |baby| -- I'm in Australia. No habla espanol. |
12:38.58 | |baby| | ok |
12:39.01 | X-Rob | (???) |
12:39.01 | |baby| | no problem |
12:39.05 | fourcheeze | got interesting router problems |
12:39.10 | cinzas | |baby|: If you want i can trasnlate |
12:39.16 | fourcheeze | took a couple of Snoms out to a customer's place last week |
12:39.18 | X-Rob | cinzas - go go! |
12:39.24 | |baby| | cinzas bien! |
12:39.27 | fourcheeze | they register fine |
12:39.29 | paxr0 | muy bien |
12:39.34 | paxr0 | jeje |
12:39.38 | fourcheeze | but when I place a call the router drops the ADSL line |
12:39.46 | fourcheeze | anyone else had anything like that? |
12:39.50 | cinzas | ¤0.5 for letter |
12:39.51 | cinzas | hehehe |
12:39.51 | tzanger | |baby|: unfortunately not. I am learning Romanian though |
12:40.05 | X-Rob | fourcheeze - yeah. Buggy firmware in the router. |
12:40.24 | fourcheeze | X-Rob: ok, firmware is supposed to be up to date, router is a fairly old Vigor |
12:40.38 | fourcheeze | is the only option a new router? |
12:40.42 | |baby| | cinzas diles que cuando yo tecleo # (atxfer key) el usuario escucha la musica (musiconhold) pero yo al marcar una extension para transferir la llamada no escucho el tono de llamada, ni es escucho nada! en cambio si suena la extension a la que llamo |
12:41.12 | cinzas | Ok |
12:41.52 | fourcheeze | is this an issue of SIP getting through the router or would a different phone help things at all (can't see how it would myself) |
12:42.38 | X-Rob | fourcheeze - get a linksys WRT54GS with the adsl modem thing built in |
12:42.41 | cinzas | Afeter pressing the # key (atxfer key) the user listens to music on hold, but when he starts dialing the extension, so he can transfer the call, there is no dial tone |
12:42.42 | X-Rob | they are _the_ best. |
12:43.01 | |baby| | cinzas graciasss!! :D |
12:43.05 | fourcheeze | X-Rob: is that wireless? They have wireless already on a different router |
12:43.06 | X-Rob | cinzas - that's correct? |
12:43.06 | cinzas | |baby| se marcas bien la extension, la llamada es transferida ? |
12:43.22 | fourcheeze | the strange thing is that a similar Vigor router in the office works fine |
12:43.26 | cinzas | X-Rob: yeap |
12:43.27 | fourcheeze | has anyone else had problems with Vigor? |
12:43.43 | X-Rob | cinzas - that's the way dialling works. As soon as you start dialing, the dialtone goes away |
12:44.06 | *** join/#asterisk [ASK]ithrynn (n=nick@host217-34-132-179.in-addr.btopenworld.com) |
12:44.13 | |baby| | cinzas si, puedo hablar con la extension remota, pero si no consigo contactar con la extension se cuelga la llamada. |
12:44.22 | cinzas | X-Rob: No theres is no translation error. He say's that there is no dial tone. |
12:44.23 | [ASK]ithrynn | lo |
12:44.40 | X-Rob | well that's right? Does he want the dial tone to keep going while he's dialling? |
12:44.41 | cinzas | X-Rob: Wait. i think he is confused :) |
12:44.44 | fourcheeze | X-Rob: if I set up a vpn from the customer's office to our asterisk server would that likely remove the problem? |
12:44.52 | X-Rob | fourcheeze - possibly. |
12:45.04 | X-Rob | But you may find that the same thing happens |
12:45.14 | X-Rob | a whole pile of UDP traffic comes through, and the router/modem has a hernia. |
12:45.26 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
12:45.35 | fourcheeze | so sending that via vpn would disguise it all |
12:45.45 | cinzas | X-Rob: Ok. He saying that if the user doesn't answer the phone, he can't go back and retrieve the call |
12:45.47 | fourcheeze | or at least should |
12:46.11 | fourcheeze | I suppose it might depend on the vpn technology |
12:46.13 | cinzas | Maybe he's doing a blind transfer |
12:46.27 | X-Rob | cinzas - aaah. OK. I don't know how you do that. I thought that if you hang up the call, the attended xfer should call you back. I don't know. Usually, it's better to use the phone's transfer. |
12:46.38 | wiizard | what would cause my extensions to report they are busy when i try to dial them |
12:46.45 | cinzas | Ok x-rob |
12:46.57 | |baby| | cinzas k dice? :P |
12:47.10 | X-Rob | wiizard - are you using AMP? |
12:47.13 | wiizard | yep |
12:47.14 | cinzas | |baby|: A ver .. |
12:47.14 | X-Rob | or Asterisk@Home? |
12:47.20 | X-Rob | Your dialparties.agi is broken |
12:47.20 | wiizard | amp |
12:47.22 | wiizard | asterisk |
12:47.27 | wiizard | similar thing ent it? |
12:47.36 | wiizard | whats the crack with asterisk@home |
12:47.39 | cinzas | |baby|: Estas usando SIP o H323 ? Los telefonos estan conectados a asterisk como SIP users ? |
12:47.48 | X-Rob | wiizard - I just committed a fix to dialparties.agi to tell you exactly what _is_ broken |
12:47.55 | |baby| | cinzas, SI, son SIP concretamente PAP2 de Linksys |
12:47.56 | X-Rob | but that'll be in AMP ..10 |
12:48.00 | X-Rob | which is due out in a couple of days |
12:48.08 | cinzas | |baby|: Los telefonos no tienen la tecla de transfer ? |
12:48.20 | cinzas | |baby|: O lo tenes que hacer com la # ? |
12:48.20 | |baby| | tecla d transfer k es? la R ? |
12:48.25 | cinzas | Sí |
12:48.29 | |baby| | es que la R no funciona |
12:48.33 | X-Rob | until then, you'll need to actually run /var/lib/asterisk/agi-bin/dialparties.agi and see what the error is |
12:48.34 | |baby| | la pulso y no hace nada |
12:48.35 | cinzas | hmmm |
12:48.56 | |baby| | hay que activar algo en la configuracion para que funcione la R? |
12:49.09 | nfi|ermes | if i would like to take a call arrived to another extension(phone), for example dialing *57, sould be enough an "Answer" command ? |
12:49.16 | cinzas | Lo que quieres es hacer la transferencia de la llamada. Y si no hai nadien en el otro lado, queires voltar a hablar con el utilizador |
12:49.19 | cinzas | es esso ? |
12:49.32 | |baby| | asi es |
12:49.38 | |baby| | transferencia asistida |
12:49.39 | mutilator | <PROTECTED> |
12:49.56 | cinzas | Pues, todoavia no ententei. Da-me um minuto para que le vea |
12:50.03 | |baby| | pulsar R o # o lo que sea, marcar extension, hablar con la extension y colgar para transferir. en caso de que no me conteste, poder volver a la llamada |
12:50.06 | X-Rob | nfi|ermes -- 'show applcation pickup' on your asterisk console |
12:50.15 | |baby| | ok |
12:50.17 | |baby| | gracias!! |
12:52.10 | nfi|ermes | X-Rob, i found very few informationm in the consolle, but thanks for your hel; now i ll go to study PickUp command |
12:52.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:52.23 | nfi|ermes | hel=help |
12:52.28 | X-Rob | I usually use ** |
12:52.39 | nfi|ermes | ok |
12:52.43 | X-Rob | exten => _**.,1,Pickup(${EXTEN:2}) |
12:52.47 | tzanger | cinzas: what was the problem? |
12:52.55 | nfi|ermes | thx so much |
12:54.40 | synthetiq | if u dial ** that will pick up every extension you dialed beginign with the first 2 letters? |
12:55.47 | X-Rob | No. |
12:55.52 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
12:56.24 | synthetiq | im interested into finding out how i can get devstate and call pick up to work by pressing the flashign button on snom 360 is someone is on the line |
12:56.58 | X-Rob | can't get call pickup yet |
12:57.12 | X-Rob | you can get state with CVS and 'HINT' |
12:57.25 | synthetiq | yea but uspposedly there is a way to do it with a directed call pick up |
12:57.37 | X-Rob | with an ancient version of chan_sip |
13:01.00 | *** join/#asterisk jake1932 (n=jake1932@pool-68-236-60-180.phil.east.verizon.net) |
13:02.18 | cinzas | X-Rob: Something like this. I've answered a call. i want to transfer the call. I press de # key |
13:02.41 | cinzas | X-Rob: But i've dialed the wrong number. |
13:03.00 | cinzas | X-Rob: How can i pickup the call again ? |
13:03.08 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
13:04.15 | |baby| | transfer assisted (atxfer) |
13:04.24 | brad_mssw | If a number i've dialed a remote number, and they want you to hit the '#' sign after dialing say a conference room number, how do you do that with asterisk. Hitting # initiates the internal blind transfer feature |
13:04.29 | brad_mssw | of asterisk |
13:07.39 | *** join/#asterisk ejo1 (n=ejo1@209.32.147.246) |
13:08.25 | encode | what settings should i check if incoming calls (via DID) dont seem to go anywhere? i just get entries in my log about congestion and absolute timeout |
13:10.43 | docE | Anyone know the CVS tag for 1.0.9 stable? |
13:11.29 | JamesDotCom | not v1-0-9 or something obvious? |
13:12.02 | docE | I tried 1.0.9 and got nothing.. I tried 1-0-9 and said it was an invalid format |
13:12.20 | ReD-MaN | anyone ever encountered where if you dial another extension is goes direct to vm saying the person is on the phone? |
13:12.23 | JamesDotCom | try what i said then |
13:12.43 | *** join/#asterisk iCEBrkr (n=icebrkr@242858hfc41.tampabay.res.rr.com) |
13:13.27 | Katty | morning. |
13:14.26 | wiizard | right ive got a question |
13:14.36 | wiizard | which is more reliable amp or asterisk@home |
13:14.42 | wiizard | or are they infact the same |
13:14.48 | docE | same |
13:14.48 | brad_mssw | docE: Just go to the CVSweb interface at : http://cvsweb.digium.com/index.cgi/asterisk/ Then drop down the box with 'All tags / default branch' .. it shows _all_ the CVS tags for asterisk! |
13:14.55 | wiizard | yeah thought as much |
13:15.04 | docE | thanks brad. |
13:15.39 | brad_mssw | but v1-0-9 was correct ... |
13:16.04 | encode | Spawn extension (from-internal, *43, 3) exited non-zero on 'SIP/203-d102' <-- is it bad or ok that spawn extension exited non-zero? |
13:16.17 | jake1932 | wiizard: they are not the same - AMP is a web interface included in @home |
13:16.30 | docE | ya forgot the v |
13:16.39 | wiizard | ok |
13:16.41 | jake1932 | <PROTECTED> |
13:16.42 | wiizard | so if i have asterisk |
13:17.05 | wiizard | and if ive installed the Asterisk management portal i technically have asterisk@home |
13:17.17 | docE | Now if I could find a reliable H323 for stable.. Then I would be set! |
13:17.42 | jake1932 | wiizard: no - you'd have asterisk + AMP |
13:18.06 | jake1932 | wiizard: i'm pretty sure asterisk@home has a page that tells you what's included |
13:18.35 | brad_mssw | docE: thought openh323 was fairly stable ... just didn't support pass-thru or something, so it would transcode communicating h323->h323 |
13:18.36 | SlickRick | hello everybody |
13:18.51 | jake1932 | http://asteriskathome.sourceforge.net/features.html |
13:19.24 | SlickRick | Can I somehow strip some digits from the beginning and from the end of a number? |
13:19.27 | wiizard | hmmm looks like asterisk@home isnt as feature packed |
13:20.05 | SlickRick | I tried something like ${EXTEN:1:-4} but that didn't work. |
13:20.09 | docE | Honestly IMO A@H sucks.. |
13:20.21 | docE | Nice for beginners.. but thats bout it. |
13:20.50 | wiizard | i |
13:20.53 | jake1932 | i spent way to much time on @home before realizing I need to understand what's going on |
13:20.55 | wiizard | shame AMp is broke |
13:21.06 | *** join/#asterisk paxr0 (n=c@216-155-78-128.bk1-dsl.surnet.cl) |
13:21.43 | Katty | jake1932: beep! |
13:22.15 | jake1932 | Katty: beep beep |
13:24.31 | Katty | jake1932: you set off my hilight. |
13:24.56 | jake1932 | Katty: thank you |
13:26.47 | jake1932 | Katty: good morning |
13:27.02 | Katty | ^_- |
13:27.28 | tzanger | Katty: he thinks you want him. :-p |
13:27.36 | tzanger | wel all know you love me though |
13:27.43 | Katty | oh, that. |
13:27.56 | Katty | you've both clearly insaned. |
13:28.25 | tzanger | DRUNK LIKE A FOX!! |
13:28.45 | docE | Im almost waterlogged like one |
13:29.35 | jake1932 | late sunday night session at the local pub? |
13:30.44 | tzanger | heh |
13:30.55 | *** join/#asterisk marc324 (n=marc3234@206-248-133-144.dsl.teksavvy.com) |
13:31.58 | mkl1525 | when I get a call from external the leading 0 is missing in the shown number and so I can't use the recall function of my sip phone is there an option that causes this behavior (or anything to avoid this)? |
13:36.24 | *** join/#asterisk paxr0 (n=c@216-155-71-234.bk1-dsl.surnet.cl) |
13:39.07 | *** join/#asterisk Tili (n=Tili@211.147.234.5) |
13:39.27 | synthetiq | i have a end;ess loop of goto's going how do i kill it? |
13:39.55 | IronHelix | mkl125- edit the context whereby calls come in to make the first command SetCIDNum(0${CALLIERIDNUM}) |
13:40.04 | IronHelix | soft hangup the channel? |
13:41.43 | *** join/#asterisk SERGEUS (i=sergey@195.112.98.13) |
13:42.06 | SERGEUS | Hi! is there expirienced AGI users? :) |
13:42.16 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
13:42.40 | SERGEUS | I have a problem with AGI, - probably my missunderstanding.. |
13:43.23 | *** join/#asterisk _T3_ (n=rposada@53.228.uio.satnet.net) |
13:45.11 | *** join/#asterisk iCEBrkr (n=icebrkr@242858hfc41.tampabay.res.rr.com) |
13:45.12 | jake1932 | SERGEUS: ask |
13:46.16 | SERGEUS | when i perfom command "WAIT FOR DIGIT -1" |
13:46.22 | SERGEUS | everything is ok |
13:46.47 | SERGEUS | * blocks and wait till i push any button |
13:47.11 | SERGEUS | but when |
13:47.46 | *** join/#asterisk Teeli (n=Tili@211.147.234.5) |
13:47.54 | SERGEUS | WAIT FOR DIGIT 10 |
13:49.13 | jake1932 | that's 10 milliseconds |
13:49.48 | Delvar | do you want to do a read digits loop? |
13:51.21 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
13:53.11 | encode | ok, asterisk is really starting to bug me...i have a sipura 3000 and a softphone, both of them registered to my asterisk box - the softphone can dial out fine, the handset connected to the sipura gets the congested message |
13:53.25 | encode | they're both trying to dial the same number, even in the logs |
13:53.46 | encode | what on earth is going on? |
13:54.12 | jake1932 | encode: are they both using the same context? |
13:54.37 | SERGEUS | pardon |
13:54.43 | SERGEUS | ohhhh |
13:54.47 | SERGEUS | i got it |
13:54.52 | encode | jake1932: yes, from-internal |
13:54.55 | SERGEUS | i thougt |
13:55.02 | SERGEUS | it was 10 seconds |
13:55.20 | encode | so the sipura phone always gets "Got SIP response 404 "Not Found" back from 202.61.13.40" when dialling |
13:55.38 | SERGEUS | that's why * returned control immediateley... |
13:55.38 | bjohnson | encode: you're just talking about the fxs on the spa correct? |
13:55.40 | jake1932 | encode: try sip debug? |
13:55.55 | encode | bjohnson: yes, analog handset connected to the fxs |
13:55.58 | encode | jake1932: ok |
13:56.04 | jake1932 | encode: then you'll get more detail as to what's going on |
13:56.04 | SERGEUS | jake1932: THANK YOU VERY MUCH :) |
13:56.12 | jake1932 | SERGEUS: np |
13:56.22 | SlickRick | Could I ask again? |
13:56.28 | encode | jake1932: what should i look for? |
13:56.47 | SlickRick | Is there any way how to strip the last 4 digits from a number? |
13:56.50 | jake1932 | encode: differences between the two scenerios |
13:56.59 | bjohnson | encode: what does sip show registry display? |
13:57.24 | mkl1525 | has anybody a snom (360) phone and got the mwi (message waiting indicator) working? when I press the key asterisk doesn't note anything - any help? |
13:58.19 | jake1932 | SlickRick: should be ${var:-4} |
13:58.21 | *** join/#asterisk aRJAy (n=aRJAy@adsl-130-112.swiftdsl.com.au) |
13:58.31 | jake1932 | no |
13:58.34 | jake1932 | that's wrong |
13:58.42 | SlickRick | jake1932: no, that returns the only the last 4 digits. |
13:58.49 | bjohnson | encode: nm the registry thing. sip show peers |
13:58.57 | SlickRick | but I want everything except the last 4 digits. |
13:58.58 | bjohnson | encode: it's likely an authentication issue |
13:59.19 | bjohnson | SlickRick: read the varieable wiki page |
13:59.48 | bjohnson | SlickRick: it's something like $var:0:4 |
13:59.54 | encode | bjohnson: ok to paste here? or is there a pastebin somewhere |
14:00.02 | aRJAy | any fellow aussies out there? :) |
14:00.04 | jake1932 | ~pastebin |
14:00.06 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
14:00.38 | bjohnson | SlickRick: I can't rmember if the second parameter is the length to take or the number off the right end |
14:01.08 | *** join/#asterisk evangelion (n=manzy_ze@213.199.26.99) |
14:01.13 | evangelion | hi all =) |
14:01.26 | jake1932 | it's the length from the offset |
14:01.28 | encode | bjohnson: http://pastebin.ca/26480 |
14:01.50 | encode | extension 203 is sipura, 204 is softphone |
14:01.51 | evangelion | does any of you use asterisk-realtime? |
14:02.26 | festr__ | i've problem with sending calleridpresentation over IAX2 channel, is it possible this? anyone is using this? |
14:02.28 | *** join/#asterisk azzie (n=az@azzie.net) |
14:02.51 | aRJAy | I'm looking to hook myself up with VOIP. Am aware of Asterisk and have only just heard of Asterisk@Home. What's Asterisk@Home like? |
14:02.58 | aRJAy | Worth the effort? |
14:03.02 | jake1932 | no |
14:03.15 | aRJAy | jake... why so? |
14:03.33 | SlickRick | jake1932: I got that already, but is there no way to get everything except the last 4 digits? in the variable wiki page I found only examples with fixed string lengths. |
14:03.36 | jake1932 | do you know a linux distro already? |
14:03.42 | aRJAy | nope |
14:03.55 | SlickRick | and ${EXTEN:0:4} doesn't work with realtime. |
14:03.57 | aRJAy | fresh machine completely |
14:04.09 | paxr0 | any asterisk@home user? i need know zaptel modules precompiled in this distro |
14:04.20 | *** join/#asterisk aor (n=bob@209-220.246.81.adsl.skynet.be) |
14:04.22 | jake1932 | aRJAy: xorcom.com |
14:04.31 | aor | Hi everybody |
14:04.48 | jake1932 | aRJAy: and check the docs |
14:04.50 | jake1932 | ~docs |
14:04.51 | jbot | hmm... docs is 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 |
14:04.51 | aor | does someone know if it is possible to do an include with realtime ? |
14:04.52 | paxr0 | jake1932, rapid dont have dial and chan modules precompiled? |
14:05.15 | bjohnson | encode: increase verbosity and watch the spa for authentication errors |
14:05.17 | jake1932 | paxr0: are you sure? |
14:05.24 | aor | i need this do to dynamicaly dial plan changes according to date/time |
14:05.28 | *** join/#asterisk Toadee (n=pixie@wblv-146-199-121.telkomadsl.co.za) |
14:05.35 | Toadee | greets |
14:05.41 | aRJAy | jake1932: will check out that site now.. |
14:05.42 | aRJAy | tah |
14:05.49 | jake1932 | paxr0: it worked great for me - and was simple to install |
14:05.49 | Toadee | quick (complicated?) question |
14:05.59 | SlickRick | oh, I take it back, it works with fixed string lengths. |
14:06.04 | paxr0 | yeah i think , and debian style |
14:06.13 | paxr0 | but i have problems with modules |
14:06.56 | jake1932 | paxr0: only reason i switched to debian + asterisk was I was ready to try an install myself on a new machine |
14:07.07 | jake1932 | paxr0: but rapid worked fine |
14:07.07 | encode | bjohnson: the softphone seems to be using NAT and some other stuff (which i havent filled out in the settings) wheras the sipura isnt |
14:07.15 | Toadee | i have a factory and showroom connected by a wireless network and a pabx in the factory, what is the preferred method to connect the showroom to the pabx |
14:07.47 | mutilator | a wire? |
14:07.57 | Toadee | erm no! |
14:08.11 | jake1932 | heh - cans and string were abandoned a while ago |
14:08.23 | evangelion | does any of you use asterisk-realtime? |
14:08.24 | mutilator | not where i work |
14:08.43 | *** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
14:09.08 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
14:09.24 | jake1932 | mutilator: are you using chan_can_and_wire? |
14:09.54 | jake1932 | uck - getting worse |
14:10.04 | jake1932 | and katty didn't even beep me for that |
14:10.04 | aor | evangelion, yes I do, well I try :-) |
14:10.05 | FuriousGeorge | does anyone use sipphone (gizmo) accounts with * |
14:10.21 | mutilator | just so everyone knows |
14:10.22 | FuriousGeorge | i find that mine tend to not want to register |
14:10.30 | mutilator | i'm depressed right now |
14:10.32 | mutilator | :'( |
14:10.33 | FuriousGeorge | but then they do and its ok for a while |
14:10.42 | mutilator | i fscking slept under my office desk last night |
14:10.45 | bjohnson | encode: is the SPA going through NAT to get to the * server? |
14:10.46 | mutilator | got about 2 hrs of sleep |
14:11.04 | Katty | beep! |
14:11.10 | jake1932 | haha |
14:11.11 | rocket | Toadee: not that I have done it with asterisk .. but you could get a pair of wrt54g's and install custom firmware on them that lets you bridge the two together .... this is if you have a small installation of phones you need connected back .. |
14:11.37 | Toadee | yeah that's it |
14:11.40 | *** join/#asterisk gabb0 (n=gabb0@131.202.90.23) |
14:11.50 | Toadee | thanks for the nudge in the right direction |
14:11.59 | evangelion | aor: how does it works? |
14:12.03 | Katty | how does one relocate? |
14:12.07 | bjohnson | Toadee: another idea |
14:12.11 | Katty | there seems to be this big loop of doom |
14:12.21 | jake1932 | just do it |
14:12.25 | Katty | just..how? |
14:12.27 | mutilator | over wifi you need atleast 60% link quality to get one good call |
14:12.28 | rocket | Toadee: look at dd-wrt for the firmware |
14:12.38 | jake1932 | different town? |
14:12.38 | Katty | mishehu: how does one relocate to chicago! |
14:12.44 | bjohnson | Toadee: make a * server just for the showroom. allows you to keep local calls between showrooms extensions local. Then trunk between * servers |
14:12.57 | aor | evangelion well it does not work too bad, but some features are not available (or I wasn't able to let them work) |
14:13.04 | FuriousGeorge | Katty: depends how big the things you need to take with you are |
14:13.08 | bjohnson | Toadee: wifi will limit how many concurrent calls you can make |
14:13.16 | rocket | Toadee: a wired connection will definately be better though |
14:13.23 | Katty | FuriousGeorge: i'll need a moving truck, for sure. |
14:13.35 | FuriousGeorge | bigger than a u-haul |
14:13.37 | mutilator | i have about 100 phones on wifi right now |
14:13.47 | Katty | FuriousGeorge: i'm more caught up in the problem of how do you get a job out there so you can move out there |
14:13.54 | bjohnson | mutilator: on one AP? |
14:13.57 | Toadee | bjohn, ta |
14:13.57 | Katty | FuriousGeorge: except you need to be there for an interview.. |
14:13.58 | mutilator | and in my experience i tell customers no more than 2 lines period and 2 lines are iffy |
14:14.04 | evangelion | aor: wich features (i know i'm buggy)? |
14:14.05 | mutilator | bjohnson: network covering nothern michigan |
14:14.07 | evangelion | =) |
14:14.10 | FuriousGeorge | mutilator: you know 100 people who can cycle a wireless connection all by themselves! |
14:14.23 | bjohnson | mutilator: so not really equivalent to his showroom and factory example |
14:14.24 | mutilator | not ap technology |
14:14.31 | blitzrage | Katty: don't do it! |
14:14.33 | mutilator | well 2 are |
14:14.35 | FuriousGeorge | Katty: offer video conferencing when you send your resume |
14:14.46 | Katty | blitzrage: why not? |
14:14.55 | Katty | FuriousGeorge: hmm. |
14:14.59 | blitzrage | Katty: not sure... I've been moving a lot in the last 5 years and I'm sick of it :) |
14:15.03 | Katty | FuriousGeorge: i'd rather not video conference an interview |
14:15.04 | aor | evangelion : well for example, for pickups, it seems that the application don't look in the db to retrive the pu group so you can't do pickups |
14:15.32 | mutilator | bjohnson: i've also setup in office configurations |
14:15.34 | aor | evangelion : also some parts must be done in static files, like the zap configuration |
14:15.35 | FuriousGeorge | Katty: shoot, if your trying to get a job in telecom it would probably look good to have your own toll free video conferencing DID |
14:15.48 | blitzrage | lol |
14:15.50 | Katty | FuriousGeorge: i'm not trying to get a job in telecom silly |
14:15.51 | blitzrage | FuriousGeorge: true :) |
14:15.55 | evangelion | aor: thanks so much |
14:15.58 | blitzrage | lol |
14:16.05 | mutilator | it's 'reliable' with a great connection |
14:16.33 | FuriousGeorge | Katty: i tend to stereotype people in #asterisk. what are you getting a job in |
14:16.37 | Katty | if anything i've the most experience in regular windows tech stuff and web design. |
14:16.42 | Katty | mcse stuffs. |
14:17.23 | jake1932 | it's proabaly a little difficult to get a job with the number of MCSEs today then it were a few years back |
14:17.23 | blitzrage | Katty: should be able to find a job for that no problem :) |
14:17.38 | blitzrage | jake1932: really? stuff like that I see jobs for ALLLLLLL the time at places |
14:17.42 | *** join/#asterisk riksta (n=rick@62.6.163.90) |
14:17.45 | blitzrage | at least in Canada |
14:17.51 | FuriousGeorge | i dunno if chicago is like newark, but here those types of people are everywhere. ive heard people say that you "lift a rock and find 10" :) but that just shows theyre making a living |
14:17.54 | aRJAy | jake1932: interesting site: www.xorcon.com |
14:18.25 | jake1932 | xorcom |
14:18.41 | Katty | jake1932: i will go to chicago if i want to. end of story |
14:18.41 | docE | I got an interesting question for someone.. :) I am setting up a PBX for my office. When I call another extension it works fine. But when I try to dial outside the office it gives me 484 Proxy Auth Required. |
14:18.42 | jake1932 | Katty: you can do it! |
14:18.43 | paxr0 | very easy way to make work debian & asterisk :) |
14:18.45 | docE | Any ideas? |
14:19.17 | FuriousGeorge | Katty: i was encouraging you. im saying there is demand and a lot of people to hook up with, in the IT sense |
14:19.26 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
14:19.31 | Katty | FuriousGeorge: yeah, i just can't find them. |
14:19.53 | Katty | FuriousGeorge: hopelessly unqualified :< |
14:20.23 | blitzrage | docE: that should be normal, then your PBX should try to send authorization based on the information (nonce) that is sent in the Proxy Auth |
14:20.42 | FuriousGeorge | the unqualified paradox: you cant get qualified till you get a job, and you cant get a job unless youre qualified |
14:20.52 | encode | ok, after reading through those debug results |
14:21.03 | encode | the sipura is just getting a 404 not found for the number |
14:21.16 | encode | but the softphone is gettin 407 proxy authentication required |
14:21.17 | docE | blitz yes.. But when I call extensions -> extension in the office its fine. When I call extension -> SIP PSTN it says 484.. Its in the same context.. I dont get it. |
14:21.22 | Katty | FuriousGeorge: and throw the whole can't do an interview unless you're in the area and can't be in the area unless you have an apartment and can't have an apartment unless you make enough money to pay for one |
14:21.44 | FuriousGeorge | life sucks sometimes :) |
14:22.03 | Katty | mishehu will help me! |
14:22.07 | aRJAy | jake1932: is that what you're using? |
14:22.08 | Katty | or maybe hug me |
14:22.12 | aRJAy | xorcom ? |
14:22.17 | jake1932 | <PROTECTED> |
14:22.21 | tzanger | xorcom.com doesn't resolve for me |
14:22.27 | tzanger | er xorcon |
14:22.35 | tzanger | xorcom does :-) |
14:22.39 | encode | how can i get asterisk to send the proxy authentication instead of the handset? |
14:22.54 | jake1932 | aRJAy: using debian + asterisk HEAD now |
14:23.02 | aRJAy | Katty: and what happened? :) |
14:23.11 | aRJAy | asterisk HEAD? :) |
14:23.12 | Katty | aRJAy: i imploded. |
14:23.18 | jake1932 | aRJAy: yes |
14:23.24 | *** join/#asterisk ejo1 (n=ejo1@209.98.205.179) |
14:23.25 | aRJAy | Katty :) |
14:23.35 | Katty | aRJAy: but probably not knowing what the hell i was doing had something to do with that ;) |
14:23.47 | Katty | linux? what's linux?! |
14:23.50 | aRJAy | Katty: OIC :) |
14:23.53 | n0rf- | HEAD.. the version that broke so many of the former |
14:23.59 | aRJAy | Line Ucks |
14:24.04 | bjohnson | encode: you can't |
14:24.33 | bjohnson | encode: why doesn't the spa find the server? |
14:24.56 | aRJAy | well.. I have a 'little' experience with linux, but certainly not a confident user |
14:25.22 | aRJAy | Is Asterisk 'that' hard to set up?? |
14:25.27 | bjohnson | no |
14:25.39 | Toadee | ta for the help |
14:25.39 | jake1932 | aRJAy: especially on debian - really simple |
14:25.43 | bjohnson | especially if you use atome |
14:25.46 | aRJAy | Does it really need a 3rd party to come along and make a special installer for it and rebrand it? |
14:26.04 | jake1932 | apt-get install asterisk |
14:26.08 | bjohnson | I don't understand the question |
14:26.11 | Katty | jake1932: eww! |
14:26.15 | Katty | jake1932: don't do that! |
14:26.20 | jake1932 | (of course that's not a current version) |
14:26.21 | Katty | jake1932: you'll get an insanely old version! |
14:26.29 | encode | bjohnson: i've no idea - the sipura was working fine for a different voip provider |
14:26.46 | bjohnson | encode: are they on the same subnet? |
14:26.52 | jake1932 | but at home isn't the most current either |
14:27.00 | encode | bjohnson: yes |
14:27.24 | aRJAy | jake1932: if the Asterisk install is simple, why would one deviate from the core s/w and install process? |
14:27.36 | bjohnson | read the wiki page for a spa 2000 and it links to a howto. the fxs on the 3000 sets up exactly the same as the fxs on 2000 |
14:27.44 | bjohnson | encode: you've got some setting wrong |
14:27.48 | jake1932 | aRJAy: use the recommended way on the digium site - that's the best way |
14:27.48 | Vco | 'make install' isnt' exactly a ball buster |
14:28.09 | Nugget | It can be tricky! On some systems you have to type gmake install! :) |
14:28.14 | encode | but some setting in asterisk? or sipura? |
14:28.25 | tzanger | "not exactly a ball-buster" haaaaaaaaaaaahahahahaha |
14:28.42 | n0rf- | aRJAy: just don't use HEAD if you're not that comfy eventually ripping it all back out and reverting to a release version.. which might happen after you searched for a fault on your side that was in fact just bad luck :) |
14:28.56 | aRJAy | jake1932: so, your suggestion is now to just get the normal Asterisk instead of xorcom? |
14:28.59 | ejo1 | Any digium employees online here now? |
14:29.09 | jake1932 | aRJAy: either |
14:29.11 | docE | ERIC! |
14:29.31 | ejo1 | Hey!, I can't get through to support since 3pm on Friday |
14:29.34 | jake1932 | aRJAy: xorcom is 1-2-3 simple |
14:29.34 | Vco | No, they're just hitting the local bars as they open PST |
14:29.35 | evangelion | does res_data project die? |
14:29.40 | aRJAy | n0rf-: sounds like it's a bit finiky. |
14:29.41 | Vco | actually that would be 7.30 am.... |
14:29.52 | Vco | that is a little early.. |
14:29.56 | jake1932 | aRJAy: the asterisk install over a distro could take a little patience |
14:30.04 | aRJAy | jake1932: k... xorcom is well supported? upgrade/update path, etc. ?? |
14:30.13 | ejo1 | I get dead air, on hold? for 40 minutes, then disconnected |
14:30.16 | bjohnson | encode: if no NAT involved, likely a SPA setting issue |
14:30.28 | docE | their screwing with you |
14:30.30 | jake1932 | aRJAy: once it's on your system, you should be able to upgrade to a newer version |
14:30.33 | encode | ok, got it |
14:30.33 | aRJAy | jake1932: I get your point :) |
14:30.35 | ejo1 | Must be |
14:30.42 | docE | They know its you calling |
14:30.46 | docE | :) |
14:30.48 | *** join/#asterisk kippi (n=kippi@host86-133-85-206.range86-133.btcentralplus.com) |
14:30.50 | kippi | hey, |
14:30.54 | aRJAy | So, xorcom is not a 'fork' as such... |
14:30.59 | encode | i needed to add "fromuser=<user here>" |
14:31.02 | encode | to the outgoing section |
14:31.16 | encode | thanks for your help |
14:31.35 | jake1932 | aRJAy: rapid (xorcom) includes the debian and asterisk (and a few other tools) - i don't think of it as a fork |
14:31.44 | ejo1 | Installing a TE410P in an SMP machine. I get the message in the linux console about "putting 1 in register 31 on span 1" |
14:31.47 | ejo1 | <PROTECTED> |
14:31.53 | kippi | when I have a voicemail message and I do *97 to pick up my voicemail it asks for my password. Once I have put my password in it says login in correct. Anyone got any ideas on this? |
14:31.58 | bjohnson | xorcom is a packaging to make install on a new server faster |
14:32.23 | jake1932 | kippi: check voicemail.conf - do you have the proper password? |
14:32.42 | encode | now i only have my digital receptionist not responding to keypresses when dialed from DID issue |
14:32.46 | encode | but that can wait |
14:32.48 | bjohnson | encode: weird. I've never needed that |
14:33.07 | aRJAy | jake1932, bjohnson: k... I guess when I see that they have their own hardware coming out (not sure if it's available at this point, is it?) then it makes me a little dubious.. |
14:33.13 | encode | its bedtime - 12:35 here |
14:33.15 | aRJAy | instinct |
14:33.17 | encode | nite folks |
14:33.27 | aRJAy | encode: say hi to ya brother for us |
14:33.33 | kippi | jake1932: I can logon using the web interface |
14:33.38 | encode | which brother would that be? |
14:33.44 | bjohnson | aRJAy: kind of like Lindows? |
14:33.45 | aRJAy | decode :P |
14:33.50 | encode | oh lol |
14:33.52 | Vco | your brother from anotha mother |
14:33.56 | paxr0 | rofl |
14:34.10 | encode | i dont have a brother whose name starts with de |
14:34.15 | encode | but mine starts with en |
14:34.16 | aRJAy | together, they make codec :) |
14:34.18 | mutilator | or sister from another mister! |
14:34.24 | aRJAy | how sweet ;) |
14:34.24 | encode | (well, D and N, but yeah) |
14:34.49 | Vco | i met his cousin vocode once... |
14:34.57 | aRJAy | Vco: heh |
14:34.59 | *** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net) |
14:34.59 | Vco | couldn't understand a word he said |
14:35.20 | n0rf- | ah, the voder family |
14:35.20 | aRJAy | So then.. this voip stuff is really new to me.. just been reading up on it a little.. |
14:35.25 | aRJAy | heh |
14:35.35 | jake1932 | aRJAy: you can use regular asterisk hardware/software with rapid |
14:36.07 | aRJAy | I have an ADSL connection, a spare machine (P3 550) and I wanna have some fun.. |
14:36.24 | kippi | voicemail.conf seems to have the user login on there |
14:36.31 | jake1932 | aRJAy: asterisk is not the recommended method |
14:36.40 | vader-wrk | does anyone in here use asterisk with IP faxing? |
14:36.51 | aRJAy | jake1932: why so ? |
14:37.15 | jake1932 | aRJAy: maybe we have different ideas of fun |
14:37.16 | vader-wrk | im looking to hook my main fax line in so it goes through ip to my ip capable fax machine |
14:37.21 | bjohnson | aRJAy: for a home system, there are easier methods |
14:37.36 | mutilator | erm |
14:37.44 | bjohnson | aRJAy: * is more suited a business style phone system |
14:37.53 | mutilator | trying to compile zaptel 109 |
14:38.07 | mutilator | tells me i need my kernel source installed |
14:38.10 | mutilator | but its there |
14:38.11 | bjohnson | aRJAy: most people using * at home are tying into other remote systems |
14:38.13 | mutilator | /usr/src/linux] |
14:38.55 | aRJAy | jake1932: perhaps we do have a different definition of fun... |
14:39.19 | aRJAy | I like the idea of having growing flexibility... |
14:39.29 | n0rf- | bjohnson: or they're just going for UMS the geeky way :)) |
14:39.33 | vader-wrk | does anyone have their asterisk system tied into a paging system of any sort or is there paging capabilities to asterisk? |
14:39.38 | aRJAy | So, I tend to get something that's more than what I "need" |
14:41.07 | hypa7ia | vader-wrk: http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom |
14:41.13 | aRJAy | i believe that there are relatively cost-effective ways of getting into VOIP with existing home telephones, etc... |
14:42.24 | mutilator | grr |
14:42.28 | evangelion | where can i find some docs on tables structure to use with realtime? |
14:42.28 | mutilator | damn zaptel won't compile |
14:44.14 | aRJAy | I have an ADSL connection (512/128), one phoneline coming into the house with 3 sockets around the house... I want all incoming calls to come through either the regular telephone number which woud then route to the regular telephones... Calls going out I'd like to go through my ADSL line (or drop down to POTS if the system fails, etc.. |
14:44.37 | *** join/#asterisk Defraz (n=t0tal@24-119-12-238.cpe.cableone.net) |
14:45.18 | aRJAy | Voice mailbox, web-browser interface, ease of use... are all important.. |
14:45.31 | aRJAy | not sure of the hardware I need... Sipura? |
14:45.46 | aRJAy | I need an ATA (I believe).. |
14:45.53 | aRJAy | I already have the computer |
14:46.14 | Qwell | evangelion: You just add a column for each config setting |
14:46.19 | Defraz | Sipura is a good ATA |
14:46.29 | Defraz | makes a good ATA |
14:46.43 | aRJAy | the 3000 ? |
14:46.45 | {zombie} | aRJAy: use a sipura3000 to handle the PSTN line, and hang one of your 3 phones off |
14:47.04 | jake1932 | can't get rid of the echo on my 3000 |
14:47.08 | aRJAy | {zombie}: tah |
14:47.09 | {zombie} | and either rewire the other two to hang off the back of the sipura, or get another sipura 2000/pap2 to handle those two |
14:47.12 | wmandra | arjay: or you could just get a TDM-400P or dev kit from digium |
14:47.23 | {zombie} | or ignore those sockets and use real IP phones, or whatever |
14:47.28 | Vco | of course if you're goign to run pstn adn a few phones, you may as well get a TDM |
14:47.30 | evangelion | Qwell: thank you |
14:47.33 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.15) |
14:48.14 | MuppetMaster | Hello |
14:48.20 | MuppetMaster | Anyone run Asterisk on AMD 64-bit? |
14:48.23 | aRJAy | I don't want to go out and buy new phones.. use existiing phones to start off with.. cost is an issue.. |
14:49.08 | marc324 | keep what you have. |
14:49.26 | file[laptop] | MuppetMaster: yes works fine |
14:49.31 | *** join/#asterisk stefanro (n=stefan@ip51cf43f8.direct-adsl.nl) |
14:50.55 | aRJAy | I found a SPA3000 for AUD$153 and Digium TDM400b for AUD$110 |
14:50.56 | MuppetMaster | Is Asterisk optimized for 64-bit? |
14:51.09 | Qwell | I wonder how many calls a 4 way X2 Opteron could handle... |
14:51.30 | marc324 | is there a reason why most use xeon as servers? |
14:51.44 | {zombie} | aRJAy: does that price for the tdm include any modules? |
14:51.45 | Aughey | MM: It's mostly your compiler that will generate more optimized code |
14:51.47 | {zombie} | or is that the bare card? |
14:52.00 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
14:52.17 | Qwell | {zombie}: for 4 modules, it'll be between 290 and 330 or so, depending on which modules you get |
14:52.25 | aRJAy | {zombie}: well.. that's another thing I don't know... what modules would I need with my setup? and why? |
14:52.36 | Aughey | MM: The algorithms could be written to take advantage of special processors, but for the most part it won't matter much just dealing with audio |
14:52.42 | wmandra | arjay, 1 fxo & 1 fxs |
14:52.44 | MuppetMaster | Ok, thanks! |
14:52.46 | Qwell | aRJAy: 1 FXO, and 1-3 FXS. |
14:52.58 | marc324 | a x100p and a tdm400p |
14:52.58 | blitzrage | marc324: probably because Asterisk is usually a CPU intensive application (especially with transcoding) -- so thats probably a good reason to have Xeons if you want to try and put a number of calls on one server |
14:53.01 | Qwell | 1 if you want all 3 phones to be the same "line", 3 otherwise |
14:53.13 | Vco | why would you futz it up with a x100p? |
14:53.14 | aRJAy | marc324? |
14:53.32 | mishehu | ugh. this is annoying... a system of mine can't even seem to keep one single g729 channel open without dropping packets... and I only have about 47 ms of ping between the two... (and the other end is on a frac ds3 circuit) |
14:53.33 | *** part/#asterisk cinzas (n=ashes@62.48.254.104) |
14:53.41 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.15) |
14:53.49 | wmandra | arjay: as long as you don't mind all of the anaolog phones being on the same extension you can get away with 1 fxs, if you want each phone to have it's own extension you need 1 fxs per analog phone |
14:54.07 | marc324 | blitzrange-- most amd processors outperform xeons. |
14:54.24 | rocket | are sipura3000 normally locked .. or if you find em on ebay etc. will they be able to be easily reprogrammed? |
14:54.32 | iCEBrkr | yeah yeah yeah.. and OS/2 out performs Windows.. |
14:54.34 | iCEBrkr | Blah blah blah |
14:54.37 | aRJAy | So, I'd need a 4 port unit with 1 x FXO and 3 x FXS (one of which could be the fax) |
14:54.55 | blitzrage | marc324: yah... but you'll get less problems with Intel chipsets (and I love AMD stuff) |
14:55.16 | marc324 | oldmyth. |
14:55.18 | aRJAy | Is that about riht? |
14:55.21 | aRJAy | right |
14:55.28 | Qwell | aRJAy: pretty much |
14:55.28 | wmandra | arjay: right |
14:55.38 | aRJAy | wmandra: thank you.. |
14:55.44 | blitzrage | marc324: well, tzanger says he has problems with AMD stuff |
14:55.46 | aRJAy | Qwell: tah :) |
14:56.01 | blitzrage | marc324: and I think he's a smart cookie :) |
14:56.01 | Qwell | tzanger just has problems. ;) |
14:56.11 | mishehu | blitzrage: could it just be the amd motherboards most people use? I saw some comments about this on saturday's /. article about the new dual core overly power hungry xeons... |
14:56.26 | blitzrage | mishehu: it is entirely possible |
14:56.26 | aRJAy | So then, it's now a matter of chosing between the SPA3000 and Digium 400b/p |
14:56.36 | blitzrage | Qwell: lol |
14:56.41 | blitzrage | Qwell: no comment :) |
14:56.43 | marc324 | marketing does work |
14:56.45 | Qwell | aRJAy: with the sipura, you'd only have 1 fxs |
14:56.58 | mishehu | blitzrage: one guy was saying that it is more crucial to follow the memory recommendations of the mobo manufacturer with amd opteron equipment than xeon equipment. |
14:57.05 | wmandra | arjay: I would start with the developers kit from digium, it gives you 1 fxo and 1 fxs |
14:57.19 | wmandra | arjay: you can always add additional fxs modules later |
14:57.21 | aRJAy | Qwell: oic :) that makes the choice slimmer |
14:57.22 | aRJAy | heh |
14:57.24 | aRJAy | k |
14:57.27 | Vco | OR fxo |
14:57.30 | Vco | ;) |
14:57.36 | rocket | wmandra: does the devel kit allow you to pass through the call to the pstn if the server is down? |
14:57.51 | iCEBrkr | I've always had CPU issues with AMD. Intels just work. |
14:58.05 | mkl1525 | Hi, I get the follow warning message quite often (every 5 seconds) " chan_zap.c:7545 zt_pri_error: PRI: !! Facility message shorter than 14 bytes" anybody knows what goes wrong? |
14:58.25 | iCEBrkr | Of course, I have more experience with matching chipsets with Intel CPU's than I do with AMD, so that may have something to do with it |
14:58.29 | *** join/#asterisk Little-L (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk) |
14:58.46 | aRJAy | iCEBrkr: I've never had any issue with AMD CPUs. I've always stayed away from them for the same reason. 99.99% of software/solutions are developed on Intel |
14:58.48 | marc324 | icebrkr-- can you be more specific. what os/app was causing problem? |
14:58.56 | wmandra | rocket: no it doesn't |
14:59.04 | n0rf- | yay, amd vs intel jihad \o/ |
14:59.05 | Qwell | aRJAy: that statement is very untrue |
14:59.12 | aRJAy | Qwell: oh really? |
14:59.13 | blitzrage | jihad! :) |
14:59.15 | aRJAy | lol |
14:59.21 | rocket | wmandra: that would make the WAF much better if it did ... |
14:59.33 | aRJAy | blitzrage: I'll tip a bucket of holy oil on you ;) |
14:59.39 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
14:59.49 | Vco | screw the cpu religion crap, it's 10am |
14:59.51 | Vco | and it's monday |
14:59.52 | iCEBrkr | marc324: Um, windows.. In general. Things won't boot consistantly. They randomly reboot.. THey lock up for no apparent reason.. The list just goes on and on. |
14:59.53 | power1 | Hey all, ive got a strange problem, I have just set up and outgoing sip trunk using inphonex as the voip provider, I can make calls and can connect to the party and can chat for about 20 secs, then we are cut off abruptly...any ideas? |
14:59.56 | blitzrage | aRJAy: hrmmm... if you're a hot chick, then maybe we can work something out |
14:59.57 | aRJAy | blitzrage: it'll be US$54.4/barrel ;P |
15:00.09 | blitzrage | aRJAy: but I'm going to assume not since we're in #asterisk :) |
15:00.20 | blitzrage | aRJAy: woh... thats cheap / barrel :) |
15:00.21 | iCEBrkr | aRJAy: I try to keep an open mind.. But any time I venture down the road of AMD, it sucks. So I go with what works for me. :P |
15:00.35 | wmandra | rocket: all the dev kit is, is a TDM400P with 1 fxo and 1 fxs daughter card installed |
15:00.35 | aRJAy | blitzrage: let's just stick to a handshake and a *cough* on that |
15:00.42 | iCEBrkr | I'm the same way with Seagate vs. Maxstor. I'm a Segate person. |
15:00.48 | *** join/#asterisk b0xii (i=b0xii@pool-70-110-83-70.dfw.dsl-w.verizon.net) |
15:00.56 | blitzrage | aRJAy: fair enough :) |
15:01.01 | aRJAy | heh |
15:01.14 | rocket | wmandra: ok .. thanks for the info |
15:01.17 | wmandra | rocket: i don't even think the cisco vic cards will allow you to do any king of pass-though if the server is down either |
15:01.50 | rocket | wmandra: was just looking that the sipura3000 lets ya do that .. |
15:02.49 | aRJAy | much diff between the 400b and the 400p? |
15:03.30 | wmandra | rocket: yes true, but it only gives you 1 fxs port |
15:03.52 | rocket | true .. but I guess in my case that is not currently an issue |
15:03.57 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
15:04.04 | KriS83 | Hi,... |
15:04.33 | rocket | wmandra: as I only have one phone in my house currently .. and would probably pick up an sip phone for other extensions later .. |
15:04.37 | mmmToop | Hi...quick question...can an agent dial when logged in with the AgentLogin() cmd? |
15:04.48 | wmandra | arjay: the tdm-40b has 4 fxs modules installed, the 400p is a blank card |
15:05.00 | rocket | wmandra: or an iax based phone if there are any that are good |
15:05.07 | KriS83 | If I needed a 4 Port ISDN Card, could anybody give a suggestion? |
15:05.19 | aRJAy | There is a friend of mine who has a company with one advertised telephone number and a fax number. He's got 5 lines coming into the office to cope with more calls going out/in. How would VOIP help him? |
15:05.28 | docE | Is there anyway to turn of digest authentication? |
15:05.41 | aRJAy | wmandra: ahhh... k :) |
15:05.49 | wmandra | arjay: a tdm-31b would give you 1 fxo port and 3 fxs ports |
15:06.13 | *** part/#asterisk Crontibs (n=Cronpars@office-nat.choopa.net) |
15:07.35 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com) |
15:08.22 | Vco | 4 port isdn? |
15:08.35 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
15:09.34 | KriS83 | Vco, Yes.. well all we want to be ablt to do is handle 4 ISDN Lines |
15:09.37 | Vco | http://www.westbaseuk.com/products.php?cat=24&SID= |
15:09.54 | Vco | just happen ed to see today..never used tho |
15:09.56 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
15:10.07 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
15:10.54 | *** join/#asterisk garthvs (n=chatzill@196.31.11.194) |
15:10.59 | KriS83 | Well we use a AVM B1 already.. but that card can only handle 2 Lines |
15:11.17 | Vco | they have the quad too |
15:11.21 | Vco | where are you using it? |
15:11.27 | aRJAy | wmandra: I can't seem to see this tdm-31b you speak of |
15:11.28 | Vco | (globally speaking) |
15:11.29 | KriS83 | Germany ;) |
15:11.55 | Vco | ah...wondering what works or doesnt' work in japan |
15:11.59 | Vco | :| |
15:12.20 | Aughey | arjay: it's a convienent part number. It's the base tdm card with the combination of fxo and fxs ports you want |
15:12.23 | KriS83 | anyone have an idea what card I could use with zaptel? |
15:12.39 | KriS83 | I mean the AVM Cards work with CAPI |
15:12.56 | aRJAy | Aughey: ahhh.. I see now... |
15:12.57 | aRJAy | :) |
15:13.03 | KriS83 | but from what I have heared zaptal supported cards are better |
15:13.57 | aRJAy | ok.. sleeeep time here.... nice to chat to y'all |
15:13.59 | wmandra | arjay: here is a good link that will help you in getting started with ast... http://www.automated.it/guidetoasterisk.htm |
15:14.09 | tzanger | Qwell: I'm having problems with what? |
15:14.13 | aRJAy | I'm sure I'll be back wit more questions.. |
15:14.26 | aRJAy | wmandra: thanks.. I'll bookmark that now. |
15:14.34 | wmandra | arjay: yw |
15:15.30 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
15:17.00 | KriS83 | noone that could suggest me one? |
15:17.08 | aRJAy | right... cheers all ~~ |
15:17.10 | aRJAy | :) |
15:17.14 | aRJAy | -=out=- |
15:17.31 | Qwell | tzanger: nothing :) |
15:20.10 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.195) |
15:20.30 | mmmToop | bye |
15:24.05 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
15:25.17 | *** join/#asterisk brookshire (n=pfffft@gateway.digium.com) |
15:25.54 | AgiNamu | Hey, anyone have any experience with Cornet or Dataprobe T1 fail switches? |
15:26.52 | Vco | ok..the word corn and probe just arent meant to go in same sentence |
15:29.16 | *** join/#asterisk Prego (n=dante_em@66.237.242.41.ptr.us.xo.net) |
15:29.34 | AgiNamu | lol |
15:29.42 | AgiNamu | hey, if it keeps my T1 up... |
15:29.49 | Vco | http://news.cincypost.com/apps/pbcs.dll/article?AID=/20050930/EDIT/509300303/1003 |
15:29.59 | Vco | sorry i had been reading this the other day and having a good laugh |
15:30.14 | docE | Prego its IN there! |
15:30.42 | AgiNamu | heh, i lived in guatemala for a few years |
15:30.55 | AgiNamu | no fuycking way -- Daniel Todd wrote that!!!! lol |
15:30.57 | AgiNamu | i know that guy |
15:31.21 | *** join/#asterisk paxr0 (n=jmardoin@200-85-216-205.bk4-dsl.surnet.cl) |
15:31.21 | AgiNamu | shit what a small world. |
15:31.24 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
15:32.00 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
15:32.13 | blitzrage | morning Mr. Weschke |
15:32.30 | bweschke | good morning |
15:33.38 | AgiNamu | vco thats hilarious. made my morning. |
15:34.11 | *** join/#asterisk eivindtr (n=wingnut-@062016241059.customer.alfanett.no) |
15:34.12 | bjohnson | aRJAy: how many people work at your friend's 5 line office and do any of them travel? |
15:34.54 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
15:38.31 | KriS83 | What would be to competative of digium to a AVM C4 Card? |
15:39.00 | wiizard | X-Rob u still about? |
15:39.13 | *** join/#asterisk kiwnix (n=egarcia@82.158.153.207) |
15:44.46 | blitzrage | ~thebook |
15:44.47 | jbot | somebody said thebook was Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk |
15:45.10 | evangelion | can i load sip users on start and then switch them realtime? |
15:46.05 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
15:46.12 | synthetiq | is it possible to hold sip phone registrations in realtime db and not on an asterisk server (not the sip.conf but the actual phone reg) ??? |
15:46.23 | blitzrage | ~thebook |
15:46.24 | jbot | well, thebook is Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk for purchase, or FREELY AVAILABLE under the Creative Commons license at http://www.asteriskdocs.org |
15:47.00 | docE | synth, yes.. If you populate your information for your peers/users in the db then all of your registrations will be kept there |
15:48.34 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
15:53.07 | KriS83 | Which ISDN card would suite for doing the following: incomming call -> IVR -> forward to local MSN Don't need more than 4 - 6 Lines |
15:53.49 | JunK-Y | ~astricon2005 |
15:53.50 | jbot | somebody said astricon2005 was at http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album02 |
15:54.55 | *** join/#asterisk svenna_ (n=svenna@p548D1D75.dip0.t-ipconnect.de) |
15:55.04 | synthetiq | yes docE but how do i go about doing that so far to my knowledge that functioanlity has not been implemented |
15:56.15 | jake1932 | KriS83: http://www.junghanns.net/en/produkte.html |
15:56.45 | mkl1525 | When a call comes in via zap interface and I do a NoOp(${CALLERIDNUM}) I get 1753754444 although normally there is a leading zero (01753754444) have tried with and without nationalprefix=0 in zapata.conf but that didn't help - any suggestions where to get the 0 in front of the number? |
15:57.13 | jake1932 | use setvar with 0${var} |
15:58.45 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl9e.dialup.mindspring.com) |
15:58.51 | AgiNamu | anyone here have experience with T1 failover switches? |
16:01.36 | RoyK | hm. just a thought... if not having a jitterbuffer for SIP, will forwarding the call through iax2 to a separate box with iax2 jb enabled have any effect on the jitter? |
16:01.54 | mkl1525 | jake1932, thanks wil try it |
16:02.00 | jake1932 | np |
16:02.14 | evangelion | Can i load realtime data on start\reload or i must wait for a new registration from the ata? |
16:03.05 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
16:03.26 | jake1932 | mkl1525: should actually be Setvar(CALLERIDNUM=0${CALLERIDNUM}) |
16:05.01 | mkl1525 | jake1932, have tried it with SetCIDNum(0${CALLERIDNUM} and seems to work, but would that work with foreign callers too that have a 00? |
16:05.39 | jake1932 | mkl1525: it will append the zero to everything that passes that line |
16:06.06 | jake1932 | mkl1525: if you want to filter international calls, you'll need to check for that in an expression |
16:07.01 | mkl1525 | ok thanks for your help! |
16:07.23 | *** join/#asterisk kiwnix (n=egarcia@82.158.153.207) |
16:10.58 | *** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net) |
16:12.29 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
16:14.00 | nfi|ermes | <PROTECTED> |
16:14.00 | nfi|ermes | <PROTECTED> |
16:14.20 | nfi|ermes | where does "128" comes out ? |
16:14.25 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
16:18.21 | *** join/#asterisk lmergen (n=Leon@grib.btcnet.nl) |
16:18.51 | lmergen | hello, I'm trying to get asterisk to recognize my newly bought TDM400P card |
16:19.02 | lmergen | I configured zaptel drivers as should, I think |
16:19.15 | lmergen | FXO ports to the outside, so I used fxsks (kickstart) as protocol |
16:19.34 | lmergen | no startup problems, kernel message logs goes fine |
16:19.40 | lmergen | ztcfg -vvv shows 4 channels |
16:19.41 | Aughey | have you run ztcfg -vv? |
16:19.45 | lmergen | yes |
16:20.04 | lmergen | anyway, in my zapata.conf, under [channels] I configured the 4 channels I want to use |
16:20.08 | lmergen | signalling = fxs_ks |
16:20.12 | lmergen | and well |
16:20.16 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
16:20.16 | *** join/#asterisk fugitivo (n=ajf@209.13.241.231) |
16:20.16 | lmergen | when starting up asterisk |
16:20.18 | fugitivo | hello |
16:20.23 | lmergen | it just doesn't register the channel "Zap" |
16:20.39 | Aughey | did you compile asterisk from source? |
16:20.42 | *** part/#asterisk SERGEUS (i=sergey@195.112.98.13) |
16:20.43 | marc324 | ne1 can recommend a athlon64 board? |
16:20.44 | lmergen | yes I did |
16:21.14 | Aughey | did you compile and install the zaptel libraries before compiling asterisk? |
16:21.20 | lmergen | no I didn't |
16:21.24 | Aughey | do that. That'll fix it |
16:21.30 | lmergen | \o/ |
16:21.30 | lmergen | tx |
16:21.33 | lmergen | will see :) |
16:21.35 | Aughey | compile and install zaptel. then recompile asterisk |
16:21.39 | Aughey | (and install) |
16:21.59 | Aughey | I had that exact same problem. |
16:22.33 | Vco | Asterisk is bascially wondering "What is this 'Zaptel' you speak of?" |
16:26.08 | lmergen | myeah i understand that now |
16:26.16 | lmergen | bit since asterisk lacks a `configure` script |
16:26.22 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
16:26.22 | lmergen | i figured it always installed everything |
16:26.34 | Aughey | it installed everything it knew of |
16:26.42 | fugitivo | anyone using asterisk+h323? |
16:28.49 | pif | hi, I'd like to have the telco dialtone, how should I Dial() ? |
16:29.03 | pif | I'm using bristuff |
16:29.49 | malverian[work] | When I do a Dial(SIP/123&SIP/456) it should ring both of them even if one is not available right? |
16:29.58 | malverian[work] | As long as one is available it should ring? |
16:30.49 | *** part/#asterisk Primer (n=primus@sh.nu) |
16:31.04 | pif | no, if one is busy voicemail shortcut it |
16:32.36 | drumkilla | malverian[work]: yes, that is correct |
16:32.54 | malverian[work] | pif, Yeah, that's what I'm seeing, but it seems like bad behavior... |
16:33.09 | malverian[work] | pif, It wasn't always like that I don't think.. is there some way to circumvent? |
16:33.23 | malverian[work] | drumkilla, I'm seeing the behavior pif mentioned. |
16:33.57 | drumkilla | well if that's true, then it is a bug |
16:34.00 | malverian[work] | For example.. here I know SIP/128 is available.. I can dial it manually and it works fine. |
16:34.02 | drumkilla | cvs head? |
16:34.19 | malverian[work] | drumkilla, From 10/19 |
16:34.45 | drumkilla | try the latest |
16:34.47 | malverian[work] | Anyway, if I do a Dial(SIP/128&SIP/119) and 119 is in "dnd" it says "All channels are busy/congested" |
16:34.52 | drumkilla | then, if it still doesn't work, please file a bug report |
16:34.56 | *** join/#asterisk niZx (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
16:34.58 | malverian[work] | Will do. |
16:35.03 | drumkilla | thanks |
16:35.08 | pif | malverian[work] : use queues |
16:35.28 | pif | they ignore redirects and busies |
16:35.42 | *** part/#asterisk santiago (n=santiago@208.195.215.158) |
16:36.17 | ManxPower | *shivver* *shivver* |
16:36.22 | ManxPower | winter has come to the south |
16:36.51 | pif | what arg should I pass to Dial to hear the ISDN dialtone? |
16:36.53 | tzanger | ManxPower: you just admitted in -users that you used -HEAD and -1.2beta |
16:36.58 | ManxPower | malcolmd, Are you SURE 128 IS available. |
16:37.12 | ManxPower | tzanger, I confess! I confess! |
16:37.16 | tzanger | hahaha |
16:37.31 | Renacor | anybody know where I can find an app that has reports for calls/queues, etc for asterisk? |
16:37.34 | ManxPower | tzanger, We tried it out after the hurricane to fix some call dropping problems. Only using it on 2 systems |
16:37.48 | tzanger | what's your impression? |
16:37.54 | ManxPower | tzanger, It's beta. |
16:38.00 | ManxPower | Several annoying things/issues. |
16:38.22 | ManxPower | I think they have been fixed in CVS-HEAD. |
16:38.28 | VxJasonxV | Renacor, voip-info.org has a list with a ton of CDR (and more) analyzers |
16:38.47 | tzanger | ManxPower: what do you do to test these systems? I'm just curious, I just run them and see what crops up |
16:38.48 | ManxPower | Specifially priorityjumping was off by defauilt, also in CVs-HEAD enumlookup REQUIRES a + prefix on the number to look up. |
16:39.18 | tzanger | yeah that priorityjumping thing was a mistake commit |
16:39.35 | ManxPower | tzanger, I usually do this 1) deploy to home Asterisk server, 2) deploy to the developement Asterisk server, 3) deploy on production servers IN ORDER OF SIZE of the server. |
16:39.56 | tzanger | I hear ya |
16:40.02 | tzanger | so nothing specific for call generation/load testing? |
16:40.10 | ManxPower | i.e. the 4 person office gets upgraded before the 60 person office. |
16:40.14 | VxJasonxV | Renacor, http://voip-info.org/wiki/view/Asterisk+GUI |
16:40.15 | *** part/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
16:40.19 | ManxPower | tzanger, not really. We don't have load issues. |
16:40.22 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
16:40.23 | VxJasonxV | whoops |
16:40.28 | VxJasonxV | wrong application to ^W :P |
16:41.19 | lmergen | Aughey, worked like a charm, tx ;) |
16:41.26 | Aughey | no problem |
16:41.29 | tzanger | ok |
16:41.57 | ManxPower | tzanger, we don't run huge systems. |
16:42.06 | tzanger | me either |
16:42.17 | ManxPower | Real Estate companies tend to have many small offices, rather than few large offices. |
16:42.23 | malverian[work] | Hmm.. nothing changed in CVS since the 19th that would affect this. |
16:42.48 | malverian[work] | Just some changes to the Dial() documentation (and the addition of a spelling error "defiend") |
16:43.27 | ManxPower | malcolmd, have you confirmed you can call both phones individually? |
16:44.42 | ManxPower | ..er... malverian[work] have you confirmed you can call both phones individually? |
16:44.50 | malcolmd | oi :) |
16:45.08 | ManxPower | Somehow I suspect malcolmd would not have that problem 8-) |
16:45.21 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
16:46.52 | malverian[work] | ManxPower, One I can, one I can't. |
16:47.19 | malverian[work] | ManxPower, The strange thing is that sip debug on the available peer shows nothing being sent to it. |
16:47.26 | malverian[work] | ManxPower, Other than OPTIONS for qualify. |
16:47.31 | malverian[work] | ManxPower, Nothing when I try to dial it. |
16:48.27 | malverian[work] | "sip show peer 128" shows -- Status : OK (41 ms) |
16:48.34 | ManxPower | Well, if you try calling SIP/123 and it's not registered and SIP/456 and it's in DND, then all channels will be congested. |
16:48.46 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
16:49.00 | ManxPower | file[laptop], thanks! |
16:49.05 | malverian[work] | Okay, I'll use the real numbers.. |
16:49.06 | ManxPower | It was 90F when I left TX |
16:49.10 | malverian[work] | SIP/128 is registered, as you can see. |
16:49.22 | malverian[work] | SIP/112 is in DND |
16:49.34 | ManxPower | Can you Dial(SIP/128)? |
16:49.36 | malverian[work] | SIP/310 is not registered |
16:49.38 | malverian[work] | ManxPower, Yes. |
16:49.46 | ManxPower | weird |
16:49.57 | malverian[work] | ManxPower, But when I do Dial(SIP/128&SIP/112&SIP/310) it doesn't pickup. |
16:50.09 | malverian[work] | Says all are congested. |
16:50.28 | malverian[work] | I even rebooted the phone (Snom) and it reregistered. |
16:51.05 | malverian[work] | ManxPower, the actual dial I'm using has about 30 numbers in it.. It's possible that Asterisk is not doing bounds checking on arguments passed to Dial |
16:51.19 | malverian[work] | And it's somehow screwing up the stack. |
16:51.25 | harryvv | malverian[work] what does your extension.conf looks like. |
16:51.32 | Vco | <malverian[work]> is this sip phone to sip phone? |
16:51.40 | ManxPower | malverian[work], Possible, I guess, but I doubt it. What happens if you dial a shorter list? |
16:51.46 | Vco | i'm having similar problem with an incoming DID |
16:51.49 | Vco | well..2 DID's |
16:51.56 | ManxPower | And why not use a queue? |
16:52.12 | ManxPower | you can do a ringall for queues |
16:52.20 | Vco | is a queue resolving the issue or just ignoring a problem? |
16:52.23 | malverian[work] | ManxPower, I haven't looked into them actually.. maybe I should ;) |
16:52.32 | malverian[work] | Vco, Really? |
16:52.39 | ManxPower | malverian[work], most of the docs for queues are complicated and annoting. |
16:52.54 | ManxPower | But a simple queue can be very simple to set up |
16:53.15 | pif | stroke your own queue and be happy |
16:53.27 | tzanger | stroke it to the left |
16:53.30 | tzanger | stroke it to the right |
16:53.43 | pif | make it shiny |
16:54.12 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
16:54.27 | malverian[work] | Vco, I'm doing mine through a special macro. |
16:54.35 | rocket | I was just curious how many people are using asterisk in their homes? and what they mainly use if for? I know it can do a ton .. I am just trying to see what the most useful features are |
16:54.59 | puppet | woot i got bluetooth working ;D |
16:55.01 | ManxPower | tzanger, it does appear that forwarding voicemails is broken in 1.2beta |
16:55.01 | *** join/#asterisk justinu (n=j2@72.18.13.48) |
16:55.03 | malverian[work] | Vco, Basically I'm defining ring groups in an included configuration file and then using a special "groupring" macro that allows you to specify overflow extension and eventually what voicemail to go to if no one ever answers. |
16:55.06 | puppet | its hot ;D |
16:55.15 | puppet | no more cellphone taxes ;D |
16:56.50 | malverian[work] | Vco, And yes.. I'm using SNOM phones for them, but it also occurs through our PRI. |
16:57.03 | Vco | hmm..mines different i guess... |
16:57.09 | Vco | but equally annoying |
16:57.30 | puppet | anyone know if its possible to get DTMF to work threw GSM? |
16:57.44 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
16:58.02 | file | puppet: sure, use an out of band dtmf method |
16:58.09 | file | like rfc2833 or info on SIP |
16:58.32 | ManxPower | puppet, yes, but not inband DTMF |
16:59.16 | puppet | ManxPower: is inband im talking about |
16:59.22 | puppet | ManxPower: outband works ok |
16:59.42 | puppet | but i want inband :/ so i dont have to specify each damn number i should be able to call "enter phonenumber to call" |
16:59.50 | ManxPower | puppet, you cannot get reliable DTMF over ANY compressed codec. This is not an Asterisk issues, this is just the way codecs are designed. |
17:00.08 | puppet | How does banks etc. work? |
17:00.11 | ManxPower | puppet, you can do that. |
17:00.14 | puppet | you can call thoose form cellphone |
17:00.24 | ManxPower | puppet, Um, cell phones use out of band DTMF |
17:00.39 | puppet | ah like file said :P |
17:00.53 | file | crazy eh? |
17:00.54 | ManxPower | the carrier converts the OOB DTMF from the phone to DTMF audio when it passes it off the PSTN |
17:01.00 | puppet | yes ;P |
17:01.04 | ManxPower | Just like Asterisk does. |
17:01.14 | ManxPower | and all voIP carriers do |
17:01.52 | ManxPower | some carriers do it wrong, but..... |
17:02.35 | *** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net) |
17:02.51 | puppet | but is there anyway to reach the out of band dtmf? if chan_bluetooth is recoded some? |
17:02.51 | orbi | any idea on how to generate a CTLFile.tlv for a Cisco 7960 phone? |
17:03.34 | Vco | i get incoming DID via SIP, call hits the server, "exten => tokyo1,1,goto(business,s,1)" it goes to the right context and starts the steps... |
17:04.05 | *** join/#asterisk cfrank (n=cfrank@bi01p1.co.us.ibm.com) |
17:04.08 | Vco | but just cycles around and around in the context, dying off and respawning.. |
17:05.55 | Vco | if i direct the call to a SIP phone, call comes in , i answer it on the phone, i hear busy on the handset, but the calling party still hears ringing.. |
17:06.04 | Vco | i hang up and it starts to ring again.. |
17:06.54 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
17:07.16 | Vco | but according to debug, - Got SIP response 486 "Busy" |
17:07.39 | tzanger | god damn P4 2.8GHz is expensive |
17:07.56 | Vco | Everyone is busy/congested at this time (1:1/0/0), but this is the line that was ringing.. |
17:08.00 | *** join/#asterisk asteriskmonkey (n=phil@HSE-Windsor-ppp211407.sympatico.ca) |
17:08.12 | asteriskmonkey | hey all |
17:09.13 | asteriskmonkey | with a single pri is there a way of splitting of the channels and assigning them context id's? |
17:09.20 | tzanger | asteriskmonkey: huh? |
17:09.48 | asteriskmonkey | nm: just figured it out :) |
17:10.26 | asteriskmonkey | another quest5ion though for those that have experince with the cdr system, how do you get it to record accountcodes on incoming calls |
17:11.24 | malverian[work] | Can you include one queue in the other and add to it? |
17:14.55 | *** part/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net) |
17:15.05 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
17:15.08 | jalsot | hi |
17:15.35 | jalsot | does anybody use muxmon? for me it doesn't want to do it's job |
17:15.40 | malverian[work] | Eg.. "foo" has members SIP/110 and SIP/112. I want "bar" to have whatever "foo" has but also SIP/114 |
17:16.51 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
17:16.57 | hardwire | fo to the mo |
17:17.59 | Vco | cdr is for wussies |
17:18.01 | Vco | ALT-PRTSCR |
17:18.04 | Vco | ya baby |
17:19.27 | lmergen | hey, if I want to associate some external functionality (say, manipulating a database) with Asterisk when a caller presses 2 for example, would there be built-in support in asterisk in some way, or shall I start writing my own extension ? :) |
17:20.03 | brookshire | agi or realtime |
17:20.22 | lmergen | realtime |
17:20.34 | brookshire | well.. both have database support |
17:21.01 | brookshire | realtime basically puts the extensions.conf into a database |
17:21.25 | brookshire | agi lets you extend asterisk so that you can write applications in perl or php much like cgi |
17:21.33 | *** join/#asterisk jets (n=jets@guardian.pmt.org) |
17:21.37 | lmergen | oh, i guess we misunderstood |
17:21.58 | lmergen | what I want is use the extensions.conf, and the for example bind a simple command to a certain action |
17:21.59 | b0xii | anyone here having issues with the goiax DIDs? |
17:22.11 | b0xii | or is it just me? |
17:22.23 | lmergen | and I'm browsing the internet right now (yeah baby!) and only now notice the System() command for dialplans - I assume that will pretty much do |
17:22.24 | lmergen | ? |
17:22.35 | lmergen | (when I connect a proper CGI script to it) |
17:22.47 | brookshire | :) |
17:23.01 | brookshire | i would find a nice agi tutorial :) |
17:23.42 | syle2 | got a question for you VOIp PRI , sip termination dudes, if you got a route that is say 10/6 , does that mean calculate 6 second increments after the 10 original seconds or with the 10 original seconds? cause if duration of call was say 36 with 10 would be 36 but without 10 would be 40 in my calculations |
17:24.14 | lmergen | brookshire, k will do that :) |
17:24.23 | lmergen | brookshire, just checking whether it's possible at all :) |
17:24.46 | harryvv | Has anyone tried out this wireless voip phone before? looks alot like the samsung wirless flip cell phone. |
17:24.48 | brookshire | lmergen: it's just software, of course you can do it |
17:24.48 | harryvv | http://www.voipsupply.com/product_info.php?products_id=1067 |
17:24.49 | brookshire | ;) |
17:25.21 | lmergen | i know, but i was doubting between writing my own extension or that it was already supported... glad to see I can call external CGI scripts, which is just fine by me |
17:26.16 | brookshire | it's AGI in asterisk ;) |
17:26.19 | brookshire | CGI is http |
17:27.21 | blitzrage | AGI tutorial |
17:27.26 | blitzrage | ~thebook |
17:27.27 | jbot | [thebook] Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk for purchase, or FREELY AVAILABLE under the Creative Commons license at http://www.asteriskdocs.org |
17:27.35 | *** join/#asterisk escribzz (n=escribzz@71.36.229.227) |
17:27.40 | blitzrage | check chapter... 7 I think |
17:29.35 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
17:29.48 | blitzrage | jtodd: Mr. John! (err... Todd :)) |
17:30.09 | justinu | everytime I hear John Todd, I think of Ferrari's Jean Todt. |
17:32.51 | malverian[work] | Anyone? |
17:37.20 | Aughey | Here's a quick question: I'm using a PSTN line, and I'd like incoming calls to first ring the secretary phone for 10 seconds before being answered by the automated system. But I don't want Asterisk to actually answer the channel until the secretary picks up or it gets answered by the automated system. If I just Dial(Sip/200) without doing an Answer beforhand, will that work? |
17:37.48 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
17:38.39 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
17:39.20 | fugitivo | Aughey: Dial(Sip,200,10) that's 10 seconds |
17:40.01 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
17:42.52 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
17:43.14 | trelane | mark's over a day idle |
17:43.23 | trelane | and my iaxy is toast and noone in customer service is responding |
17:43.43 | brookshire | trelane: did you call support? |
17:43.49 | trelane | with what? |
17:43.53 | trelane | my iaxy=my phone line |
17:44.01 | brookshire | ahh.. email support@digium.com |
17:44.01 | fugitivo | softphone? |
17:44.02 | trelane | well it's the extension that rings and is dialable anyway |
17:44.05 | trelane | brookshire, already did |
17:44.15 | trelane | fugitivo, soundcard isn't set up for it |
17:44.25 | *** join/#asterisk scubasteve (n=steve@cpe-071-065-212-199.nc.res.rr.com) |
17:44.38 | fugitivo | cellphone? |
17:44.45 | scubasteve | Does anyone know how to make * not native bridge for SIP? |
17:45.04 | file | you mean not reinvite? |
17:45.11 | trelane | fugitivo, don't have one. Trust me if there was a great solution I'd have it |
17:45.15 | scubasteve | I just upgraded to head.. everything appears fine .. but calls between SIP devices have no audio |
17:45.16 | trelane | file, want to do me a favor? |
17:45.23 | file | trelane: like? |
17:45.26 | jets | BROOKS! |
17:45.28 | scubasteve | file: -- Attempting native bridge of SIP/ScubaSteve-20c6 and SIP/Tiffany-5076 |
17:45.36 | file | scubasteve: that's normal. |
17:45.48 | scubasteve | file: I have no audio on calls between SIP devices. |
17:45.58 | malverian[work] | app_queue doesn't work how it's intended I don't think... |
17:46.00 | fugitivo | scubasteve: nat between them?? |
17:46.03 | file | scubasteve: separate problem |
17:46.05 | scubasteve | file: I have audio on outbound calls through * and calls to AGI and vm.. |
17:46.07 | *** join/#asterisk durex (n=ironman@weber.anpa.org.br) |
17:46.18 | scubasteve | file: No nat between them. Nothing has changed except for a new HEAD (today) |
17:46.36 | file | "now the days are gone, now you're on your own" |
17:46.37 | scubasteve | file: I want to see if I can tell * to stay in the media path |
17:46.39 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
17:46.43 | file | scubasteve: canreinvite=no |
17:46.47 | scubasteve | hm |
17:46.50 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
17:47.09 | durex | hi. in this context: exten => _5561.,1,Dial(SIP/${EXTEN}@bdf_out-bsb,30) , the number dialled will be 5561XXXXXXXXX@bdf_out-bsb . How to dial JUST XXXXXXXXX@bdf_out-bsb, and not the 5561? |
17:47.20 | file | durex: ${EXTEN:3} |
17:47.22 | file | er |
17:47.24 | file | durex: ${EXTEN:4} |
17:47.30 | durex | file thank u |
17:47.32 | scubasteve | file: I set canreinvite=no and it still bridges.. |
17:47.32 | durex | where I found doc about it? |
17:47.39 | file | scubasteve: bridging is normal |
17:47.44 | file | scubasteve: repeat after me, "BRIDGING IS NORMAL" |
17:47.56 | file | canreinvite just stops asterisk from going directly |
17:47.59 | scubasteve | file: no sound is not normal :-) |
17:48.16 | file | rtp debug |
17:48.36 | scubasteve | rtp debug returns nothing. |
17:48.50 | file | you put canreinvite=no in each peer/friend entry? |
17:48.52 | file | and did a sip reload? |
17:49.02 | scubasteve | file: yes |
17:49.13 | file | do a sip debug and pastebin it |
17:49.22 | Katty | save me from the windows servers! |
17:49.23 | Katty | save me! |
17:50.32 | malverian[work] | !!!! |
17:50.38 | file | !!!!!!!! |
17:50.45 | malverian[work] | From "show application queue" |
17:50.47 | malverian[work] | <PROTECTED> |
17:50.58 | malverian[work] | However, instead of doing what it says.. it just NEVER retries, and NEVER exits.. |
17:51.12 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
17:51.47 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
17:51.54 | malverian[work] | I also set retry=0 timeout=5 in queues.conf and it it just retries infinitely |
17:51.56 | shmaltz | anybody here know where I can check the status of CallForwarding on Zap channels (using the default built in extensions of *72 on the zap channel) |
17:52.50 | b0xii | if anyone here has a goiax account can you please test my setup by calling 878201001165 I'd appreciate it. |
17:53.49 | bjohnson | the LAN equivalent to rebooting |
17:54.16 | Katty | bjohnson: :< |
17:55.20 | malverian[work] | Seriously.. am I doing something wrong? |
17:55.39 | *** join/#asterisk w0w0 (n=w0w0@130.Red-83-44-178.dynamicIP.rima-tde.net) |
17:56.09 | malverian[work] | I have a queue "support" with members SIP/112 and SIP/114, i set timeout to 5 seconds and retry to 0 |
17:56.23 | malverian[work] | I run Queue(support,t) and the application never returns. |
17:58.51 | malverian[work] | And if I run Queue(support,nt) it also never returns. |
17:58.55 | malverian[work] | Anyone else have this issue with CVS HEAD? |
17:59.59 | malverian[work] | Asterisk does print this out: -- Nobody picked up in 5000 ms |
18:00.07 | malverian[work] | But it still never returns from the Queue() application. |
18:01.40 | malverian[work] | Please, I'm about to lose my fucking mind... |
18:02.07 | Katty | malverian[work]: gosh. |
18:02.16 | Katty | malverian[work]: are the color metaphores really required? |
18:03.02 | malverian[work] | It's extremely frustrating that I'm having to use this because of some stupid bug in asterisk, and when I try to use this, the documentation is not accurate. |
18:05.22 | *** join/#asterisk evangelion (n=manzy_ze@3ffe:80ee:1e96:c:20e:35ff:fe28:c2b1) |
18:05.54 | malverian[work] | I'm really beginning to regret recommending using Asterisk to my employer. |
18:06.04 | malverian[work] | We should have just shelled out the 5,000 for a comdial machine. |
18:06.59 | *** join/#asterisk JASON-0 (n=jason@jason.unitz.ca) |
18:07.23 | JASON-0 | Hi, I'm just wondering if its normal for the sound quality to be a little choppy sometimes in the voicemail and in conversations.. |
18:07.24 | mutilator | well for one.. why are you using head for your pbx? |
18:07.28 | trelane | malverian[work], I've got comdial at work, if this is the only issue you have... |
18:08.08 | *** join/#asterisk loick (n=loick@APuteaux-151-1-54-57.w82-120.abo.wanadoo.fr) |
18:08.10 | malverian[work] | trelane, This isn't the only one. |
18:08.26 | malverian[work] | trelane, It's things piled on top of things piled yet on top of others. |
18:10.20 | malverian[work] | I've had to edit far more code than I'd have prefered to, just to get things as simple as call parking working in a sane manner. |
18:10.54 | asteriskmonkey | silly question if anyone can help... my voicemail keeps writing the files as root how do i change it back to the asterisk user? |
18:10.56 | trelane | malverian[work], the comdial transfers calls at random, puts voicemails on phones which are not even mentioned in the voicemail routing system, the time is never right, voicemail messages disappear, and then there are the silent calls of death |
18:11.22 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
18:12.22 | malverian[work] | We have a 15 year old comdial system that is working flawlessly if it weren't for hardware failure.. |
18:12.55 | trelane | this is one of the new ones, installed last year |
18:12.56 | malverian[work] | It's so old they don't even have replacement parts anymore. Our sync card died so we get frame slips all the time now. |
18:13.05 | malverian[work] | Which is why we're replacing it. |
18:13.27 | malverian[work] | The point is, things that work out of the box on comdial have taken silly hacks to get working at all. |
18:13.38 | trelane | have you tried 1.0 stable? |
18:13.41 | asteriskmonkey | ANYONE KNOW WHY a comedian mail system would be writing the files as root and how to fix it? |
18:13.51 | malverian[work] | trelane, Yes. |
18:14.00 | trelane | broken in stable and in head? |
18:14.18 | malverian[work] | trelane, On that, I can't even get auto-answer with intercoms working on my SNOM phones. I had to write some code and submit to CVS to get it to work. |
18:14.33 | malverian[work] | Luckily they accepted my patch and it's in the trunk now. |
18:14.39 | malverian[work] | But it's just one thing after another. |
18:14.55 | asteriskmonkey | twisted , drumkilla anyone of you around? |
18:15.09 | trelane | malverian[work], keep writing code and submitting bugs |
18:15.34 | malverian[work] | trelane, I don't have time for that. Our phone server is failing and I need a _working_ machine. |
18:15.43 | shmaltz | anybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan? |
18:16.03 | malverian[work] | trelane, Now to be fair, Asterisk is a nice piece of software. And I know I'm using CVS, so bugs are to be expected. |
18:16.48 | trelane | malverian[work], I'm not necessarily defending asterisk, but there's very few things I can say about Comdial without using language I wouldn't use in front of say, my mother. |
18:17.30 | malverian[work] | I think the root of my problems is russel's "massive cleanups" on the 19th |
18:18.02 | trelane | I didn't do it ;) |
18:18.06 | trelane | good :) |
18:18.47 | malverian[work] | ast_freak, You're probably running asterisk as root. |
18:19.29 | malverian[work] | asteriskmonkey, , You're probably running asterisk as root. |
18:19.52 | asteriskmonkey | yes i am :) thanks noticed that in my top program ... how do i change what user it runs as |
18:20.03 | malverian[work] | asteriskmonkey, -U <user> -G <group> |
18:20.21 | malverian[work] | asteriskmonkey, `asterisk -h` for more information. |
18:20.25 | asteriskmonkey | k |
18:21.00 | sahafeez | question, i am trying to get asterisk not to try to load mysql support. is that in modules.conf |
18:21.07 | hohum | support for polycom and cisco hand sets seems good |
18:21.14 | hohum | dunno about any other IP handset |
18:21.21 | hohum | and dunno about MGCP either |
18:21.28 | hohum | all my phones run SIP |
18:21.49 | hohum | malverian: what type of phone did you say you were using again? |
18:22.12 | malverian[work] | SNOM 320 |
18:22.15 | marc324 | is openssh secure? |
18:22.17 | sahafeez | using, polycom here. works fine. pain to setup the phones on the polycom side |
18:22.44 | malverian[work] | I'm using a pretty heavily modified version of app_valetparking to get my orbits working. |
18:23.20 | malverian[work] | Is there any kind of regression testing that gets performed on Asterisk builds? |
18:23.48 | shmaltz | anybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan? |
18:25.21 | *** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com) |
18:26.06 | sahafeez | how do i disable asterisk trying to load mysql realtime on startup? |
18:27.27 | *** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net) |
18:27.56 | orbi | Does anyone know where i can find instructions on flashing a Cisco 7912 phone to SIP firmware? I can't find instructions on google anywhere. |
18:28.51 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
18:29.05 | groogs | p |
18:29.34 | wmandra | orbi, try http://voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
18:30.00 | *** join/#asterisk arp2 (i=infinite@obscure.metachar.net) |
18:31.43 | orbi | wmandra: that's good for The 7940/60s but doesnt seem to work for the 7912 |
18:32.17 | wmandra | is there even a SIP image available for the 7912 |
18:32.50 | orbi | yeah |
18:33.47 | wmandra | try http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a0080443a25.html |
18:35.02 | asteriskmonkey | My asterisk dosnt seem to want to start with the -U -G args |
18:35.44 | orbi | nada :( |
18:35.48 | asteriskmonkey | do i have to put it like this asterisk -U asterisk -G asterisk or like this asterisk -U <asterisk> ? |
18:36.13 | *** join/#asterisk MUD (n=MUD@206-248-138-115.dsl.teksavvy.com) |
18:36.48 | malverian[work] | -U asterisk -G asterisk |
18:36.58 | asteriskmonkey | ok i try again |
18:36.59 | malverian[work] | Make sure the user and group exist and have ownership of the directories necessary.. |
18:37.12 | malverian[work] | asteriskmonkey, Eg /var/lib/asterisk /var/spool/asterisk , etc |
18:37.14 | wmandra | orbi: or try http://adisworld.oodi.ca/2005/10/10/using-cisco-ip-phones-with-asterisk/ at the bottom of the page there are instructions for the 7905 which should be similar to the install in the 7912 |
18:39.09 | shmaltz | anybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan? |
18:41.01 | asteriskmonkey | malverian : those 2 directories are owned by asterisk th one dir thos /var/spool/asterisk/voicemail/default is owned by root though |
18:41.04 | asteriskmonkey | is that correct? |
18:41.40 | malverian[work] | They should all be owned by asterisk |
18:41.43 | malverian[work] | Recursively. |
18:41.54 | malverian[work] | Also.. /var/run/asterisk (presumably) and /var/log/asterisk need to be owned by asterisk. |
18:41.55 | fugitivo | anyone using E1 with MFCR2 ? |
18:43.04 | *** join/#asterisk r0d3nt (i=r0d3nt@66.0.156.250) |
18:43.06 | riksta | can someone please tell me about this error "Oct 23 20:05:31 WARNING[18453]: channel.c:1178 ast_waitfordigit_full: Unexpected control subclass '8' |
18:43.09 | asteriskmonkey | malverian: all the directories are owned by asteisk except thta vm dir, asterisk is running as root atm as i cant get it to start as asterisk |
18:43.16 | asteriskmonkey | anywhere else i should look? |
18:44.47 | malverian[work] | chown that directory to asterisk. |
18:44.51 | malverian[work] | and then run it as asterisk |
18:46.17 | znoG | guys, when a user tries to dial out, i want to ask for a pin and either allow them out or not based on their entered pin. Is it best to do it with a AGI? |
18:47.33 | asteriskmonkey | malvarian: like this chown asterisk:asterisk vm? |
18:47.45 | shmaltz | anybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan |
18:47.50 | fugitivo | znoG: no |
18:48.10 | znoG | fugitivo: then? how would you do it? |
18:48.17 | fugitivo | znoG: Authenticate() |
18:48.29 | znoG | using authenticate? |
18:48.30 | znoG | ah yes. |
18:48.34 | znoG | just found that on Google :) |
18:48.46 | fugitivo | :) |
18:49.29 | shmaltz | znoG, how do you plan on controlling the pin? |
18:49.33 | malverian[work] | asterisk chown -R |
18:49.41 | malverian[work] | UGH... |
18:49.47 | malverian[work] | Why is app_queue broken.. |
18:49.49 | malverian[work] | This is stupid. |
18:49.56 | znoG | shmaltz: i'll have to see how Authenticate works first, i thought of just keeping it in a DB, or it just depends on how Authenticate works |
18:50.16 | shmaltz | znoG, authencticate is a mess |
18:50.29 | shmaltz | you should either write your own dialplan to handle it |
18:50.31 | fugitivo | znoG: well, authenticate is just a simple function, if you want a pin for each user, you'll need to find another way |
18:50.41 | bjohnson | znoG: there is info on the wiki about different ways to do user authentification |
18:51.02 | asteriskmonkey | ahah :) for those that dont know how to chown --- chown -R asterisk:asterisk /var/spool/asterisk/voicemail/ |
18:51.13 | Aughey | just use a variable for the Authenticate argument. Load the variable with the user-dependent password |
18:51.22 | shmaltz | or you could use the new VMAuthenticate to use voicemail.conf files to reequest the pin |
18:51.54 | bjohnson | you can do that with the old voicemail, but you have to go into vm first |
18:52.10 | znoG | shmaltz: or AGI, right? |
18:52.33 | shmaltz | znoG, AGI is only an option as a last resort |
18:52.39 | bjohnson | why would you use AGI unles you HAVE to? |
18:52.47 | shmaltz | I personaly never use AGI for such apps |
18:53.12 | znoG | see, not everyone in voicemail.conf will have outgoing access.. |
18:53.15 | shmaltz | I only use AGI for intergration with external DBs for IVR systems |
18:53.28 | znoG | i'm looking at http://www.voip-info.org/wiki-Asterisk+user+authentication |
18:53.40 | bjohnson | iirc the vm system will allow the user to change their own pwd |
18:53.54 | shmaltz | znoG, so you do it in the dialplan, use the variablies that vmauthenticate returns to figure out to which DISA to drop the user |
18:54.11 | shmaltz | ~iirc |
18:54.12 | jbot | from memory, iirc is "if I recall correctly" |
18:54.29 | shmaltz | bjohnson, exactly |
18:54.42 | shmaltz | thats why I think that authenticate is a mess |
18:54.48 | VxJasonxV | Anyone know how to clear up these 'Maximum retries exceeded on call %hash%@ipaddress' warnings? |
18:54.50 | asteriskmonkey | ah daminit damnit damit i cant get asterisk to run as root .. is there anywhere else i can find out why not |
18:55.07 | znoG | the idea is that a user dials 12341234 and once dialed, it asks for a pin. If successful, it dials the number, else it says "sorry" and hangs up. |
18:55.33 | stevek | VxJasonxV: it means that the other side is dead/unreachable: make the other side respond, and those messages will go away. |
18:55.51 | shmaltz | VxJasonxV, yeah, make sure that you have qualify=yes in sip.conf, that way asterisk will know they are not reachable and not try. provided that its a udp timout issue behind nat |
18:56.18 | shmaltz | znoG, so this i show you do it: |
18:56.18 | VxJasonxV | stevek, and shmaltz I haven't made any calls though... |
18:56.35 | VxJasonxV | It happens any time I start/reload the asterisk process. |
18:56.39 | shmaltz | VxJasonxV, then take out the register lines |
18:57.04 | VxJasonxV | I only have it registering to gizmo, and that succeeded |
18:57.22 | shmaltz | znoG, create the 12341234 exten |
18:57.23 | shmaltz | have asterisk ask for a pin using the vmauthenticate cmd |
18:57.31 | znoG | shmaltz: right |
18:57.48 | shmaltz | I'm not done yet |
18:57.50 | shmaltz | wait |
18:58.04 | znoG | shmaltz: 12341234 is any random number (ie. any number starting with 4 longer than 4 digits should be asked for a pin on every attempt to dial) |
18:58.24 | znoG | so i guess when you said "create the 12341234 exten" you also meant i can create _4XXXXXX |
18:58.27 | shmaltz | znoG, obvoiuslyh |
18:59.27 | shmaltz | using the AUTH_MAILBOX variable route the call to a macro that either denies or allows that user to complete the call |
18:59.37 | VxJasonxV | hmmm |
18:59.43 | znoG | nice |
18:59.46 | VxJasonxV | well, the register was removed. I reloaded, and it's still coming up |
18:59.48 | znoG | thanks shmaltz |
18:59.53 | shmaltz | np |
19:00.28 | mistral | anyone know anything about using the sipura 3000 with asterisk ? |
19:00.43 | shmaltz | VxJasonxV, then you have a SIP client that is trying to connect to asterisk and asterisk has problems comunicating with it |
19:00.43 | znoG | shmaltz: out of curiosity, can the "wait for pin" be customized? in the sense that I want to play a strange tone .gsm (users are used to that with the current system) |
19:00.52 | shmaltz | mistral, shoot |
19:00.55 | VxJasonxV | Ahhhh, it's probably that ip phone |
19:01.23 | mistral | its got an fxo port but for the life of me i cannot get it to ring asterisk |
19:01.33 | shmaltz | znoG, yes, just use the playback command to play what ever you want, and then use the vmauthenticate with the s option |
19:01.58 | shmaltz | mistral, you have to configure a dialplan in the sipura box that it should use |
19:02.23 | znoG | shmaltz: ah yes.. the only problem I see is if I want a user to have a password but no mailbox |
19:02.24 | mistral | yeah its easyier said than done. Are you fimiluar with it ? |
19:02.28 | shmaltz | I always make it into a hotline and configure the s extension in asterisk to do what I want |
19:02.40 | bjohnson | shmaltz: which versions include vmauthenticate? just head? |
19:02.56 | shmaltz | znoG, just specify the mailbox with vmauthenticate |
19:03.08 | shmaltz | that way the user will only be asked for the password |
19:03.28 | shmaltz | bjohnson, AFAIK, only HEAD |
19:03.55 | shmaltz | mistral, yes I am |
19:04.01 | shmaltz | mistral, RTFM |
19:04.10 | mistral | yeah i have done 3 times now :D |
19:04.17 | bjohnson | or set up a second exten to catch users without a vm and do a different auth system for them |
19:04.30 | *** join/#asterisk durex (n=ironman@weber.anpa.org.br) |
19:04.41 | bjohnson | read the bottom part of the spa 3000 wiki page |
19:04.44 | shmaltz | mistral, how do you think I know, I don't remember it by heart, and I don't have access to a sipura box at the moment |
19:04.49 | bjohnson | use the forwaring thing |
19:04.51 | durex | hi. how to make all connections pass trhough my asterisk? |
19:05.06 | bjohnson | durex: conifrgure them that way |
19:05.11 | bjohnson | and don't use reinvite |
19:05.19 | shmaltz | durex, by wearing slippers |
19:05.20 | durex | bjohnson but how? |
19:05.32 | shmaltz | RTFM durex |
19:05.42 | shmaltz | ~docs |
19:05.48 | jbot | i heard docs is 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 |
19:05.49 | bjohnson | durex: you're gonna have to ask a more specific question |
19:06.02 | bjohnson | durex: I don't know if you can't connect at all or you're disconnecting part way through a call |
19:06.19 | *** join/#asterisk acertain (n=acertain@200.119.18.102) |
19:06.46 | bjohnson | eg * can establish a call and then back out of the data flow so the end devices talk to each other directly |
19:07.21 | durex | yes, i don't want that end devices talk to each other directly. It will complicate my voip rules. |
19:08.02 | shmaltz | durex, configure your sip settings to have canreinvite=no |
19:08.19 | znoG | shmaltz: the annoying thing is the boss just wants users to enter one single pin (no username or mailbox number first)... |
19:08.24 | shmaltz | durex, but I can't see how this will complicate your voip rules |
19:08.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:08.36 | znoG | shmaltz: ie. everyone has their own pin, but they should be able to dial from any phone using one pin, no mailbox number |
19:08.44 | shmaltz | znoG, so you supply the usename based on caller id or whatever |
19:08.58 | znoG | shmaltz: right, but as i said any user should be able to dial out from any phone using a pin |
19:09.10 | znoG | shmaltz: i thought about caller ID, but the user can move around |
19:09.11 | shmaltz | znoG, http://www.voip-info.org/wiki-asterisk+cmd+VMAuthenticate |
19:09.40 | shmaltz | oh, hold on |
19:10.54 | *** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net) |
19:11.05 | shmaltz | znoG, for the moment the only thing I can think of: then the mb will have to match the password |
19:12.00 | orbi | anyone have any ideas why i could pick up the phone, get a dialtone but not be able to dial any extensions (including the demo extension) - if i attempt to dial the dialtone stays w/o registering an input. Skinny Protocol. If i hang up i get the on hook notice, waitfordigits <0 |
19:12.15 | malverian[work] | Hmm.. I think Queue would work better as a channel driver... |
19:12.19 | shmaltz | znoG, or the only way to really do it is to use something like authenticate, but then the users wont be able to change their passwords by themselfs |
19:12.40 | malverian[work] | Then you can just do Dial(Queue/support) or something.. |
19:12.48 | malverian[work] | It has so much duplicated code from app_dial it's silly... |
19:12.52 | shmaltz | malverian, nah |
19:13.02 | malverian[work] | shmaltz, Why not? |
19:13.05 | VxJasonxV | is there some - vs. _ wonkeyness when it comes to context= and context section names? |
19:13.07 | shmaltz | the way it's now is much more flexiable |
19:13.29 | VxJasonxV | for example, subdomains can't have _'s, so you have to translate -'s into _'s? |
19:13.34 | malverian[work] | shmaltz, Give me an example of something that couldn't be handled as a channel driver? |
19:13.43 | *** join/#asterisk trentster (n=marktren@rndf-146-5-208.telkomadsl.co.za) |
19:13.46 | shmaltz | you can always use dial(local/whatever) which can point to an extensions doing the queue |
19:13.52 | *** join/#asterisk r0d3nt|m (i=r0d3nt@66.0.156.250) |
19:14.31 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
19:14.37 | shmaltz | give me any reason it should be a part of a channel driver? |
19:14.48 | malverian[work] | I'm not sure why I brought that up, but I do have a complaint with the queues.. |
19:14.53 | malverian[work] | Or perhaps it's ignorance.. |
19:15.06 | shmaltz | shoot |
19:15.08 | malverian[work] | What's the best way to have a queue (eg. "support") that has about 10 numbers in it. |
19:15.20 | malverian[work] | And then "support-overflow" that will include those 10 numbers and have like 5 more |
19:15.24 | shmaltz | you mean 10 agenst/reps? |
19:15.27 | malverian[work] | Without having to just copy and paste the contents. |
19:15.54 | malverian[work] | shmaltz, Well, I don't want to overcomplicate it. I just want to use SIP/xxx |
19:16.11 | shmaltz | ok, but what do you mean by copy and paste? |
19:16.24 | malverian[work] | Eg... |
19:16.26 | malverian[work] | [support] |
19:16.31 | malverian[work] | member=>SIP/112 |
19:16.39 | malverian[work] | .... (10 or so more) |
19:16.44 | malverian[work] | [support-overflow] |
19:16.48 | malverian[work] | include => support |
19:16.53 | *** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net) |
19:16.55 | malverian[work] | member=>SIP/192 |
19:17.00 | malverian[work] | Something like that.. is that possible? |
19:20.23 | shmaltz | malverian, I have no clue it it's possible or not, (the include is for sure not possible), but you could try member=Local/extenpointingtosupportqueue |
19:20.40 | malverian[work] | Ahh.. true. |
19:20.52 | trentster | whats the command to debug asterisk again "asterisk -xxxxxxxxx? |
19:21.05 | shmaltz | asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvcr |
19:21.06 | malverian[work] | vvvv |
19:21.08 | orbi | i found the answer on how to get a SIP load on a Cisco 7912 and that's just a pain in the ass you wouldn't believe |
19:21.09 | shmaltz | c for console |
19:21.17 | shmaltz | and r for remote connection |
19:21.25 | orbi | so im sticking with Skinny |
19:21.31 | shmaltz | orbi, tell me about it |
19:22.11 | trentster | shmaltz, thanks |
19:22.19 | sahafeez | any issues with head from today. |
19:22.44 | shmaltz | sahafeez, tell me |
19:23.26 | orbi | ok, i pick up the damn phone, it starts simple switch, but if i dial a number it doesnt do anything |
19:23.37 | orbi | i hang up and it's still waitfordigit <0 |
19:23.53 | orbi | its like im missing a "Send" button somewhere. |
19:24.04 | shmaltz | try the # key after you dial |
19:24.18 | orbi | I did |
19:24.20 | orbi | no dice |
19:24.29 | orbi | there are no softkeys or anything |
19:25.18 | Katty | justrightkeys. |
19:25.22 | malverian[work] | Meh... |
19:25.27 | bjohnson | znoG: not much security involved if all it takes is to guess a password |
19:25.29 | malverian[work] | I'm just going to fix app_dial, F this :-P |
19:25.33 | bjohnson | not a username/password pair |
19:25.54 | sahafeez | sorry what i was asking any big issues with head from today. i was going to update |
19:25.55 | orbi | i've been stuck at this point for 2 days now |
19:26.04 | orbi | :/ |
19:26.27 | orbi | I was going to try the chan_sccp module |
19:26.34 | orbi | but it crashes when the phones try to register :P |
19:26.51 | *** join/#asterisk kippi (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com) |
19:27.27 | *** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net) |
19:27.37 | kippi | hey |
19:28.00 | sahafeez | how do i disable asterisk from trying to load sql support on startup? |
19:28.09 | sahafeez | i have been looking thru the realtime docs but.. |
19:29.58 | kippi | I have GXP-200 handset but when I logon to my voicemail and enter my password for the mailbox it dosn't want to take it. It seems to take 2 digits and then take the next two. Any ideas what this could be? |
19:32.04 | znoG | bjohnson: i agree, that's how it is now though, boss seems to like it this way |
19:32.58 | shmaltz | znoG, you can then do hardcoded dialplan without either the authenticate or vmauthenticate to accomplish that |
19:33.22 | shmaltz | znoG, beware though that some phones keep these numbers for a redial |
19:33.36 | bjohnson | ha |
19:33.39 | bjohnson | they all would |
19:33.54 | shmaltz | bjohnson, not the Polycom or cisco |
19:34.15 | shmaltz | once the stream is esteblished the keys are not rememberd |
19:34.16 | n0rf- | sahafeez: check out modules.conf. add noload => [modulename] there |
19:34.43 | shmaltz | most system phones (toshiba, avaya, panasonic) don't remember paswords either for the redial |
19:35.08 | sahafeez | n0rf- did that. not there |
19:35.22 | shmaltz | sahafeez, of course it's there |
19:35.40 | znoG | shmaltz: the phones here at least don't seem to remember keys after the stream is established |
19:36.25 | znoG | shmaltz: what do you mean do a hardcoded dialplan? how do I do a simple check to see if the users' "pin" exists in a text file? (this is why I thought using AGI would do the trick, as I just need to find out if a users' pin exists in a file, and since it's not a username/password match, as long as it's in the file is good enough for me) |
19:36.51 | shmaltz | znoG, why use a text file? |
19:36.54 | shmaltz | use the DP |
19:36.59 | sahafeez | shmaltz: no, its not. not that i see. |
19:37.22 | orbi | where can i find instructions on making a buttomtemplate file? |
19:37.25 | jalsot | does anybody use muxmon? for me it doesn't want to do it's job |
19:37.25 | znoG | shmaltz: global variables?! or asterisk's internal DB? |
19:37.45 | sahafeez | shmaltz: if i have autoload, then i need to spec, which file not to load. i do not know what the file is.. |
19:37.58 | sahafeez | called |
19:38.08 | shmaltz | znoG, dialplan, use the dialplan |
19:38.26 | znoG | shmaltz: ok, which function exactly to do the pin checking? |
19:38.32 | shmaltz | sahafeez, I'm working on it hold on |
19:38.41 | shmaltz | znoG, GotoIf |
19:38.49 | znoG | shmaltz: if you mean DBGet/DBPut then ok, makes sense. |
19:38.56 | znoG | (although they are now deprecated) |
19:39.34 | *** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com) |
19:39.40 | znoG | shmaltz: and where would I store the list of allowed pins? |
19:39.44 | shmaltz | znoG, you can use that as well, but you could use hardcoded as well |
19:39.54 | shmaltz | sahafeez, its app_realtime.so |
19:39.57 | bjohnson | only if users can't change them |
19:40.07 | VxJasonxV | hmmm |
19:40.11 | shmaltz | bjohnson,, they can't |
19:40.24 | VxJasonxV | the mysql database meant to hold my CDR (and more?) has no tables |
19:40.33 | sahafeez | shmaltz: thanks. brain dead. i was looking for sql type stuff |
19:40.42 | shmaltz | VxJasonxV, RTFM its on the wiki |
19:40.44 | VxJasonxV | is this an installation step I missed? or is the program supposed to do that itself if it doesn't find them? |
19:40.46 | VxJasonxV | ok |
19:42.00 | sahafeez | shmaltz: still tries to load the mysql stuff |
19:43.07 | *** join/#asterisk carrar (i=tim@osburn.com) |
19:43.09 | n0rf- | sahafeez: there are several applications that can use mysql |
19:43.28 | shmaltz | sahafeez, what is the message you getting? |
19:44.04 | sahafeez | figured it out. i just add noload for the name in the error. makes sense now. that you. |
19:44.06 | *** join/#asterisk kiwnix (n=egarcia@82.158.153.207) |
19:44.07 | sahafeez | thank you even |
19:44.29 | n0rf- | :) |
19:44.59 | carrar | Any one in here have a recommended Asterisk server vender who uses Tyan boards? |
19:45.02 | sahafeez | I added: noload=app_realtime.so |
19:45.02 | sahafeez | noload=cdr_addon_mysql.so |
19:45.27 | sahafeez | i want to setup this up at some point but not untill i get everything working without sql |
19:45.31 | sahafeez | then i will move it. |
19:46.02 | *** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com) |
19:46.03 | shmaltz | gtg |
19:46.07 | shmaltz | c ya |
19:47.21 | znoG | shmaltz: and where would I store the list of allowed pins? |
19:48.04 | shmaltz | znoG, in the dialplan (/etc/asterisk/extensions.conf) |
19:48.15 | shmaltz | ok, gtg |
19:48.17 | shmaltz | c ya guys |
19:48.18 | shmaltz | bye |
19:49.42 | znoG | i still didn't get what he meant by storing the list of pins IN the extensions.conf file |
19:50.56 | bjohnson | znoG: do a bunch of GotoIf's |
19:51.09 | bjohnson | znoG: one right after the other |
19:51.23 | znoG | haha thats awful |
19:51.31 | znoG | better go with the DB route i think |
19:51.34 | bjohnson | the whole concept is awful |
19:51.45 | znoG | yea, username/pass would be best |
19:51.49 | znoG | using VMauthenticate |
19:51.53 | bjohnson | you could do it by including those gotoif's from another file |
19:52.25 | bjohnson | if user's need to be able to change their own passwd via the phone system, voicemail is the only way to currently do it |
19:52.55 | bjohnson | well, no you couldn't allow that |
19:52.58 | znoG | or using AGI :) |
19:53.04 | bjohnson | the system wouldn't know which pwd to change |
19:53.12 | znoG | no, that's right |
19:53.29 | znoG | i'm not going to allow them to change their pass, so its ok |
19:53.34 | orbi | Guys, any idea why (oh, why) a phone using the skinny protocol wouldnt recognize digits are being dialed? |
19:53.55 | bjohnson | then a straight text file formated as a context would be easiest to maintain |
19:54.24 | bjohnson | orbi: I don't know skinny, but dmtf sounds like your problem |
19:54.47 | bjohnson | must be some way to set it on the phones AND in *. Make sure they are both using the same standard |
19:54.59 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
19:56.30 | znoG | i think i'll just have a DB family called PINS, with keys for each pin and a boolean value associated to each one, which shows if they are enabled or not |
19:56.53 | orbi | bjohnson: hmmm, good thought. |
19:57.08 | znoG | bjohnson: this way somebody could add new pins using their phone (an admin, of course) |
19:58.17 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
20:00.07 | carrar | Who sells well built servers to use with Asterisk? (hardware) |
20:02.25 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
20:03.14 | *** join/#asterisk jtdintulsa (n=jtdintul@lancer.mbo.net) |
20:05.28 | orbi | bjohnson: isn't all DTMF the same DTMF? |
20:05.36 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
20:06.03 | sahafeez | reading the asterisk BOOK. the examples do not match current. |
20:06.04 | trelane | carrar, only one I've really had any dealings with is ATAcomm (www.atacomm.com), he's in here pretty frequently |
20:06.06 | asteriskmonkey | carrar: what do you need |
20:06.32 | bjohnson | orbi: no |
20:06.39 | bjohnson | orbi: not in sip or iax anyway |
20:08.39 | sahafeez | in the book there is a file called, enter-ext-of-person. i do not see it or something like it in /var/lib/asterisk/sounds |
20:09.18 | znoG | then create it |
20:09.54 | harryvv | career you can try Arial. He has sold over 100 asterisk pbx and configured them. |
20:10.08 | carrar | ok |
20:10.26 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
20:10.30 | harryvv | career, what is this going to be used for? |
20:10.39 | sahafeez | yes, i know that. was looking for canned sounds now. i guess what i am asking is they pulled stuff from the install between the book and HEAD |
20:10.44 | glm2k | sahafeez: that sound file is in my /va/rlib/asterisk/sounds dir |
20:10.52 | glm2k | er, var/lib |
20:10.59 | carrar | well I just need the hardware big enough to handle two quad cards |
20:10.59 | sahafeez | glm2k: what version |
20:11.07 | glm2k | 1.0.8 HEAD |
20:11.21 | harryvv | I see |
20:11.29 | carrar | the software and configuring I will do |
20:11.34 | Corydon-w | 108 has never been HEAD |
20:11.34 | sahafeez | root@voice-gateway:/var/lib/asterisk/sounds# pwd |
20:11.35 | sahafeez | root@voice-gateway:/var/lib/asterisk/sounds# ls -la enter* |
20:11.42 | harryvv | Carrar, how many channels would you be piping though that? |
20:11.50 | glm2k | Corydon-w: then HEAD at around that time |
20:12.01 | carrar | 184? |
20:12.01 | glm2k | circa 2005/08 |
20:12.14 | sahafeez | ah, i am 10/5 |
20:12.17 | carrar | 23*8 |
20:12.17 | Corydon-w | What's the output of 'show version'? |
20:12.20 | harryvv | is that sip or sip/zap or what? |
20:12.23 | sahafeez | its not there |
20:12.32 | carrar | T1's |
20:12.38 | carrar | of voice |
20:12.42 | carrar | PRI's |
20:12.50 | glm2k | Corydon-w: Asterisk built by jplc@buran.netfone2x.com on a i686 running Linux |
20:12.59 | hohum | I wish Digium made DSP boards |
20:13.10 | hohum | instead of leaving it up to the Host CPU |
20:13.11 | tzanger | hohum: wy |
20:13.19 | tzanger | hohum: that's the entire point of Zaptel |
20:13.20 | hohum | I'd be willing to pay extra for the hardware |
20:13.26 | hohum | I realize that |
20:13.26 | Corydon-w | glm2k: it's been signicantly altered if it's giving you that version string |
20:13.29 | tzanger | hohum: go get yourself a dialogic board then :-) |
20:13.33 | tzanger | and you WILL pay extra |
20:13.34 | glm2k | Corydon-w: aye |
20:13.42 | hohum | but have you ever tried to use an asterisk box as a TDM gateway? |
20:13.46 | hohum | it doesn't work too well |
20:13.58 | sahafeez | is there an app that will play each .gsm file so i can here each one |
20:13.59 | hohum | IMHO |
20:14.37 | hohum | and I would run Dialogic if Asterisk supported PRI on any other type of board besides Zaptel |
20:14.53 | glm2k | egad, please. no Dialogic |
20:14.58 | tzanger | hohum: actually it works pretty damn well for me, and nufone's network is *all* asterisk-to-TDM, and it works very well for them |
20:15.24 | tzanger | what specific problems do you have? |
20:15.31 | *** join/#asterisk bronc (i=bronc@phalse.2600.COM) |
20:15.33 | tzanger | or are you trying to terminate a quadspan of PRIs on a Duron900 or something? |
20:15.48 | hohum | no, dual Xeons at the time |
20:16.00 | hohum | but then again it's been a while since I've tried it with Asterisk |
20:16.06 | bronc | is it just me, or is the iaxclient library not compile any longer? |
20:16.07 | hohum | I'm just talking in general |
20:16.21 | tzanger | hohum: generally speaking, I disagree with you. :-) |
20:16.25 | hohum | okay |
20:16.38 | hohum | and you're entitled to that opinion, I'm not trying to take that away from you |
20:16.51 | hohum | :) |
20:16.56 | tzanger | :-) I'm just saying that asterisk as a TDM gateway seems to work pretty damn well for people I work with |
20:17.10 | hohum | I don't doubt that |
20:17.19 | tzanger | for large large rollouts I think I'd recommend unlocked MaxTNTs terminating DS3s myself since you could do voice, data and fax on them |
20:17.41 | hohum | never played with thouse |
20:17.48 | carrar | they double as heaters |
20:17.50 | hohum | lately I've been tinkering with Brooktrout boards |
20:17.54 | bronc | ahh |
20:17.58 | bronc | the Makefile is horked |
20:18.05 | tzanger | yes they do spit out a fair amount of heat |
20:18.06 | tzanger | and noise |
20:18.12 | bronc | it never looks to see if you're on FreeBSD |
20:18.14 | bronc | just linux |
20:18.16 | hohum | 8 span PRIs, and 1 port DS3 cards that fit nicely into a Compact PCI chasis |
20:18.17 | tzanger | and I consume on average an extra two burgers if I'm lugging the shit around |
20:18.21 | bronc | while the included libs are freebsd aware |
20:18.26 | tzanger | brooktrout boards? |
20:18.27 | bronc | what nice code... |
20:18.30 | hohum | yeah |
20:18.41 | hohum | http://www.brooktrout.com (I think, let me double check) |
20:18.53 | hohum | yeah |
20:18.55 | hohum | that's it |
20:19.00 | hohum | they make compact PCI boards |
20:19.09 | hohum | that have a SIP stack |
20:19.33 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
20:19.47 | hohum | and some of their products are pretty high density |
20:19.56 | tzanger | which board? |
20:20.01 | hohum | uhm |
20:20.35 | tzanger | I don't see anything (quick glance mind you) relating to DS3 or even T1 |
20:20.44 | hohum | hold |
20:20.49 | tzanger | TR1034 is fax and T1 |
20:21.09 | hohum | http://www.brooktrout.com/products/tr2500/ |
20:21.26 | tzanger | ahh there we are |
20:21.49 | tzanger | that's a pretty hefty card |
20:22.11 | tzanger | I bet it's got a pretty heft price too |
20:22.22 | hohum | I don't know how much we're paying for them |
20:22.35 | hohum | they work well though |
20:23.20 | tzanger | I bet |
20:23.30 | *** join/#asterisk swankier (n=swank@S010600080219cb14.vc.shawcable.net) |
20:23.32 | brad_mssw | why is SIP recommended over IAX for faxing? |
20:23.50 | swankier | Hello channel |
20:23.52 | hohum | they're a hell of a lot less expensive for the port density than alot of things |
20:24.02 | swankier | has there been any status change on zaptel drivers for OpenBSD |
20:24.03 | swankier | ? |
20:24.14 | trelane | brad_mssw, I don't think iax support has been finished for faxing |
20:24.21 | *** join/#asterisk eKo1 (n=fake@65.169.159.98) |
20:24.29 | hohum | and compact PCI cards really aren't *that* costly |
20:25.32 | tzanger | hohum: there's always the "compared to what" factor |
20:25.43 | *** join/#asterisk TedC (n=ted@gray.impulse.net) |
20:25.45 | swankier | anyone? |
20:25.48 | |dennis| | Question: IF i do not do a make asterisk-samples, i will not get any of the config files. What i want to know is which config files are needed just to get asterisk running with a few sips and a tdm04b card.. |
20:25.50 | hohum | I don't know, I don't play with media gateways much |
20:25.51 | tzanger | swankier: no idea |
20:25.56 | hohum | er |
20:25.57 | hohum | sorry |
20:26.00 | hohum | TDM gateways |
20:26.12 | hohum | so I haven't seen much |
20:26.12 | tzanger | |dennis|: first answer: stop overoptimizing |
20:26.24 | hohum | I've seen really expensive Cisco solutions |
20:26.24 | tzanger | |dennis|: second answer: stop overoptimizing |
20:26.32 | tzanger | |dennis|: third answer: get it working, THEN wonder |
20:26.33 | hohum | in the form of a 250,000 dollar AS5850 |
20:26.44 | *** join/#asterisk bmg505 (n=leon@rndf-146-37-221.telkomadsl.co.za) |
20:26.57 | hohum | which can hold up to 96 PRIs and I think like 12 DS3s |
20:27.00 | tzanger | |dennis|: seriously... a hundred kilobytes of text in /etc is NOT going to kill you. Get it working, THEN start pissing around getting it small |
20:27.38 | tzanger | hohum: yeah that's big.. I generally just fill a half rack with two MaxTNTs, a terminal server and a router/pop box |
20:27.41 | |dennis| | tzanger: hmm.....ok...was not really worried about storage...but about what the default config files might leave open in terms of config issues...like connections externally and so on..but yeah...you are right..thanks.. |
20:27.52 | swm_ | XDMCP is a BITCH to setup on lastest Slackware! But the Latest slackware supports all 4 processors in my server! woo hoo |
20:28.08 | bmg505 | join ##slackware |
20:28.13 | trelane | swm_, umm any linux will support 4 processors |
20:28.15 | tzanger | |dennis|: it's all pretty locked down by default |
20:28.20 | bmg505 | mmmh ja noob alert :) |
20:28.30 | hohum | that's a lucent product? |
20:28.33 | swm_ | Linux Slackware 8.x would only find two processors |
20:28.37 | tzanger | swm_: uh, you mean the latest default kernel. Slackware supported multiprocessor as soon as the kernel did |
20:29.03 | |dennis| | tzanger: thanks... |
20:29.35 | tzanger | hohum: well after amalgamations and corporate purchases and all that, yes, I think so |
20:30.28 | swm_ | Oh I guess that was the situtation. |
20:30.41 | swm_ | I dont know why they put KDE in the /opt directory now :( |
20:30.52 | swm_ | I can only get KDM to work and not XDM to work |
20:31.59 | hohum | I'm working on implementing a very simple B2BUA |
20:32.08 | hohum | which supports RADIUS |
20:32.23 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
20:34.29 | swankier | I will purchase an AS5850 fully loaded for anyone who can help me with a wcfxo driver in OpenBSD |
20:34.32 | swankier | ..... promise |
20:34.34 | swankier | :) |
20:34.52 | hohum | why OpenBSD? |
20:35.07 | swankier | because it is my preferred platform :) |
20:35.54 | tzanger | a fully loaded AS5850 |
20:35.58 | tzanger | that's a lot of machine |
20:36.02 | bronc | damn who wrote iaxclient |
20:36.10 | bronc | it only compiles on Linux |
20:36.15 | swankier | actually, we have a 5850 |
20:36.18 | swankier | it's just not fully loaded |
20:36.25 | X-Rob | Why don't you put a bounty up for it? Will probably be cheaper than $30,000. |
20:36.25 | bronc | they claim it compiles on like windows, freebsd, solaris, etc, etc |
20:36.29 | bronc | and i just tried |
20:36.38 | bronc | on all of them |
20:36.38 | marc324 | ne1 runs * on amd processor? |
20:36.39 | bronc | lol |
20:36.57 | swankier | X-Rob: I could hire a developer for most of a year and have it developed as well... |
20:37.00 | bronc | even with linux compat mode on freebsd it horks |
20:37.03 | swankier | but I'm a small shop |
20:37.05 | swankier | I can't afford that right now :( |
20:37.16 | swankier | (for $30,000 that is) |
20:37.50 | hohum | 30k developer that knows C? |
20:37.50 | hohum | I'd like to see that |
20:39.13 | bronc | hire someone from Ukraine or India |
20:39.18 | bronc | I do it all the time |
20:39.53 | hohum | *shrug* |
20:40.03 | marc324 | *all the time* |
20:40.17 | bronc | ya |
20:40.24 | bronc | for Perl, C/C++ and Java or .net crap |
20:40.26 | X-Rob | marc324 - yeps. Wups, he's just hired another one. |
20:40.34 | X-Rob | uh-oh, there he goes again. |
20:40.37 | X-Rob | It's like a nervous twitch. |
20:40.53 | marc324 | bronc--$/hour ? |
20:41.00 | bronc | depends on the job |
20:41.07 | X-Rob | swankier - well, don't offer $30k worth of hardware. |
20:41.11 | bronc | i can get a dedicated developer for a month for $2k |
20:41.16 | swankier | X-Rob: that was a joke ;) |
20:41.20 | bronc | doing web application stuff |
20:43.33 | tzanger | you own't get me for a month for $2k that's for damn sure |
20:44.39 | hohum | a week maybe :) |
20:44.55 | tzanger | well maybe ten days. :-) |
20:46.58 | swankier | so the general consensus is... no zaptel drivers on OpenBSD, right? |
20:47.28 | bronc | i hear the drivers kinda work on *bsd |
20:47.32 | bronc | but not very well |
20:47.49 | hohum | some of them work on FreeBSD IIRC |
20:47.55 | hohum | but not all of them |
20:47.56 | bronc | i saw at astercon that they are making drivers in 1.2 and 2.0 for Freebsd 6 |
20:47.56 | hohum | YMMV |
20:48.56 | hohum | if a driver works in FreeBSD though it isn't guaranteed (and probably won't) work in OpenBSD or NetBSD |
20:49.19 | Igbothom | as long as you're not chasing your own tail |
20:49.35 | bronc | most likely they have the drivers now working with linux compat |
20:49.57 | bronc | so you figure you could get it for work on *bsd |
20:49.57 | hohum | really? |
20:50.01 | bronc | that's my guess |
20:50.01 | marc324 | p4 630 vs opteron 142 -? |
20:50.12 | bronc | since they said they are working on bsd aware drivers for freebsd 6 |
20:50.53 | eKo1 | marc324: opteron |
20:50.57 | hohum | FreeBSD is my platform of preference but I use what ever OS is best suited to the application which I'm trying to run |
20:51.06 | Igbothom | marc324; opteron (just) in this case |
20:51.12 | hohum | so in the case of Asterisk, I run Linux |
20:51.24 | Igbothom | both are at the bottom end of their ranges |
20:52.30 | hohum | when I need a good PC-based router (sick, I know) I use OpenBSD |
20:53.20 | marc324 | p4 630 vs athlon64 3000+? |
20:53.43 | Igbothom | closer again, depending on your needs |
20:54.01 | Igbothom | if it is heavy FPU usage, the P4, if not, the Athlon64 |
20:54.14 | VxJasonxV | Garrr, freaking Grandstream budgetone phone. Will the asterisk console really not give me a specific reason why my budgetone couldn't register? |
20:54.24 | VxJasonxV | I get a generic registration failed, but no reason. |
20:54.34 | *** join/#asterisk [hC] (n=hardcore@8.10.2.42) |
20:54.47 | hohum | Vx: tethereal/ngrep |
20:55.08 | hohum | watch what the Asterisk box tells the phone |
20:56.08 | VxJasonxV | hmmmm |
20:56.15 | VxJasonxV | I think I have one of those installed |
20:56.51 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
20:56.51 | *** mode/#asterisk [+o denon] by ChanServ |
20:57.25 | harryvv | mark342 im running the opteron42 |
20:58.01 | marc324 | what board? |
20:58.30 | Rowter | for massive calling .call files would work ok? |
20:58.57 | synthetiq | what file do u edit to test if u can go over a t1 |
20:59.03 | Igbothom | as for Opterons, have a look at the Sun Fire X2100, X4100 and X4200 systems - rather nice |
20:59.09 | drumkilla | Rowter: manager interface is probably a better way to go |
20:59.36 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
20:59.53 | harryvv | marc324 the msi k8t neo master 2 far |
21:00.05 | harryvv | but |
21:00.25 | shido6 | ... |
21:00.31 | harryvv | there are better boards |
21:00.40 | harryvv | some limitations to this one. |
21:00.47 | Igbothom | that's my issue with AMD - the mainboards |
21:00.59 | harryvv | Igbothom this one works fine |
21:01.04 | Igbothom | with Intel CPUs it is easy - buy an Intel board, they are rock solid |
21:01.26 | harryvv | I would trust this server as a asterisk box. |
21:01.37 | Igbothom | it has a decent CPU :) |
21:01.39 | harryvv | It was designed for 3d animation |
21:02.19 | synthetiq | how do you generate fak phone calls in asterisk |
21:02.43 | synthetiq | digiumw as working on my machine 2 days ago and the guy left whatever making test calls still |
21:02.53 | Igbothom | the Sun boxes look nice - all Opteron-based |
21:03.42 | harryvv | Igbothom interesting that sun would go that route |
21:04.18 | Igbothom | they have had some Opteron boxes for a while, but this is the first in-house design based on Opteron - the rest were just re-badged (OEM) |
21:05.20 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
21:06.23 | bjohnson | synthetiq: look for .call files |
21:06.28 | bjohnson | or call digium |
21:07.42 | VxJasonxV | Hey hohum, would you happen to know if a virtual server would prevent you from using either of those tools? |
21:08.02 | harryvv | tyan tiger great board. |
21:08.14 | hardwire | heh |
21:08.14 | hohum | you have to be logged into root to sniff packets |
21:08.15 | hardwire | growl. |
21:08.21 | hohum | s/into/in as/ |
21:10.45 | Rowter | drumkilla, why you think so? |
21:13.48 | orbi | anybody idea why a phone using the skinny protocol wouldnt recognize digits are being dialed? |
21:13.59 | orbi | the phone acts like im not even pressing a button |
21:15.13 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
21:16.37 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
21:17.12 | paryl | i'm trying to figure out why this isn't working... |
21:17.15 | paryl | exten => ${ARG1}/${MACRO_EXTEN},1,VoicemailMain(${ARG1}) |
21:17.15 | paryl | exten => ${ARG1}/${MACRO_EXTEN},2,Hangup |
21:17.15 | paryl | exten => s,1,Dial(SIP/${ARG1},20) |
21:17.15 | paryl | exten => s,2,Voicemail(su${ARG1}) |
21:17.15 | paryl | exten => s,3,Hangup |
21:17.34 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
21:17.42 | paryl | i'm wanting it to work so that if you dial your own extension you're automatically dumped into voicemail |
21:20.07 | darkskiez | paryl: cant you do the voicemail matching in the main context not in the macro? |
21:20.52 | paryl | dark: well, i guess, but isn't that the whole point of a macro? to cut down on lines that you have to type? |
21:21.22 | Igbothom | :) |
21:23.43 | *** join/#asterisk kippi (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com) |
21:23.47 | kippi | hey |
21:25.24 | kippi | is there away you can phone a queue |
21:25.25 | kippi | ? |
21:26.06 | fugitivo | make an extension with the queue |
21:26.57 | kippi | hmm i keep on getting a 499 error when trying to ring any extension |
21:29.14 | fugitivo | paste your Dial line |
21:30.14 | *** join/#asterisk iCEBrkr (n=icebrkr@242858hfc41.tampabay.res.rr.com) |
21:30.21 | *** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net) |
21:30.22 | cripito | hi |
21:30.50 | Aughey | (sorry if someone already answered this) Here's a quick question: I'm using a PSTN line, and I'd like incoming calls to first ring the secretary phone for 10 seconds before being answered by the automated system. But I don't want Asterisk to actually answer the channel until the secretary picks up or it gets answered by the automated system. If I just Dial(Sip/200) without doing an Answer beforhand, will that work? |
21:31.22 | iCEBrkr | Aughey: Asterisk has to answer it to do the transfer/call handling |
21:31.38 | sahafeez | do you have to define the parking extention (700) in extensions.conf |
21:32.14 | Aughey | So if someone called, and hung up on the first ring, Asterisk would have still answered it and they would get a connection charge |
21:32.15 | cripito | exten => ???, 1, Dial(SIP/200, 10) |
21:32.28 | iCEBrkr | Aughey: Most likely, yup |
21:32.36 | Aughey | hrm |
21:32.46 | sahafeez | also, does the include => parkedcalls have to be in every context you want to use it? |
21:32.50 | cripito | and aughey yes |
21:32.54 | orbi | anybody idea why a phone using the skinny protocol wouldnt recognize digits are being dialed? |
21:33.10 | Aughey | It would be nice if you could ring an extension, and when it's picked up go to the next extension line and only then do the answer and actually establish the connection between channels |
21:33.13 | Igbothom | orbi; dtmf settings? |
21:34.00 | orbi | Igbothom: I didn't think cisco phones had multiple DTMF settings? |
21:34.04 | cripito | try the cmd i send u aughey. and put the answer behind that command... |
21:34.13 | cripito | also u can do someother things |
21:34.22 | Igbothom | orbi; dunno, don't have any here |
21:34.33 | iCEBrkr | Aughey: Asterisk would have to keep the line-voltage pulsing to mimick a ring in order to do that via PSTN |
21:34.43 | orbi | Igbothom: is there a place in Asterisk to change the DTMF Settings? |
21:35.00 | Igbothom | sip.conf |
21:35.15 | orbi | Igbothom: we're using Skinny |
21:35.20 | orbi | :~( |
21:35.34 | Igbothom | aha - then wherever you configure that |
21:35.38 | orbi | ok |
21:35.40 | orbi | tankoo |
21:35.52 | Aughey | but the device can detect a ring. It wouldn't be able to detect a hang-up until it didn't detect a ring within a rings length |
21:37.12 | Aughey | because when it detects a ring, it goes to the s extension. You answer it with an Answer command. But you don't have to answer it. |
21:38.19 | Aughey | so your first command in the s extension would be a DialButDoNotActuallyConnect command to an extension. On success or failure of that command the Answer would be done connecting the channels |
21:38.40 | harryvv | I thought that asterisk only offers blind calls. obviosly that is not the case. |
21:41.38 | znoG | is there a way to keep a counter updated in a Dialplan? ie. i want to ask for a password 3 times, then hang up... |
21:42.17 | denon | so, stupid Q, but a 7960 tags everything to * with 802.1p right? |
21:43.18 | *** join/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de) |
21:43.28 | orbi | nothing i find in the Skinny.conf indicates where i'd modify DTMF settings |
21:43.36 | orbi | any ideas on a reference for Skinny.conf? |
21:43.38 | drumkilla | denon: i thought 802.1p was only used in conjunction with 802.1q ... |
21:43.58 | denon | drumkilla: not sure .. |
21:44.07 | denon | was just setting up qos on these workgroup switches |
21:44.30 | denon | trying to figure out the best way to do it .. 802.1p or dscp |
21:44.36 | drumkilla | hm, well i guess not |
21:44.37 | denon | or by IP port, but that could get ugly |
21:44.47 | Laibsch | Hi. I understand this question might not be very well liked among you, but what is a good GUI for configuring asterisk on a Debian system to get a system up and running in the shortest time frame? |
21:44.48 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt) |
21:45.08 | Laibsch | GUI can of course be web server based. |
21:45.29 | Igbothom | Laibsch; AMPortal |
21:45.29 | Aughey | vi |
21:46.04 | Laibsch | Igbothom: Thank you for the hint. Do you happen to know if that is available for Debian? |
21:46.09 | Igbothom | yup |
21:46.14 | denon | drumkilla: I usually just prioritize based on MAC, and give it the MACs of the phones .. .but these switches wont do that |
21:46.40 | swm_ | anyone know of any voip software for the symbian series 60 phones? |
21:46.42 | Laibsch | http://www.sineapps.com/news.php?rssid=1002 -> seems to be. Cool. |
21:46.50 | swankier | hmm... I'm talking to consultants to have them develop and OpenBSD zaptel driver. |
21:46.59 | orbi | Igbothom: Is there a good reference for the parameters available in skinny.conf to your knowledge? |
21:47.03 | swankier | I'm just gauging interest in it right now |
21:47.06 | swm_ | anyone know of any voip software for the symbian series 60 phones? |
21:47.10 | denon | drumkilla: so you dont think I can rely on 802.1p? |
21:47.16 | Igbothom | dunno - I don't use Cisco phones here |
21:47.19 | drumkilla | denon: I have no idea what the phone does. |
21:47.51 | swm_ | Nokia phone, cell phone, but has a OS where you can load programs onto it, also has bluetooth for connectivity.... |
21:47.55 | drumkilla | it supports 802.1q, so i'd think it supported p as well ... |
21:48.11 | drumkilla | it has been a while since I have looked at that stuff |
21:48.11 | denon | any special configs to make it happen? |
21:48.16 | denon | or it just magically work? |
21:48.37 | drumkilla | it's probably an option in the phone, if it has it |
21:48.50 | drumkilla | sounds like a job for ethereal! |
21:49.41 | justinu | i know I want a Nokia E61` |
21:49.57 | drumkilla | denon: i don't see an option for it in my 7960 |
21:50.37 | denon | hmm .. these are gig-e switches, maybe I should just assume there wont be an issue |
21:50.37 | denon | hehe |
21:52.12 | justinu | the 7960s will use something called CDP (cisco discovery protocol) to figure out which vlan to tag traffic with |
21:52.36 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
21:52.44 | harryvv | nokia will be comming out with a wimax phone in about two yeas. |
21:55.20 | denon | justinu: yeah, but these arent ciscos switches, so no cdp |
21:55.23 | denon | and its all on one vlan |
21:55.27 | denon | just qos prioritizing |
21:57.52 | justinu | yeah |
21:58.28 | justinu | i think you can run cdp off a server, if you want to |
21:59.57 | JunK-Y | ~astricon2005 |
21:59.58 | jbot | astricon2005 is probably at http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album02 |
22:00.10 | darkskiez | it may be possible to write your own CDP daemon for other switches. |
22:00.57 | *** part/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com) |
22:01.06 | *** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com) |
22:01.25 | *** join/#asterisk glomph (n=black@85.133.18.154) |
22:01.55 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
22:02.36 | *** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com) |
22:10.20 | *** part/#asterisk mogorman (n=mogorman@gateway.digium.com) |
22:10.44 | *** part/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de) |
22:18.16 | asteriskmonkey | is there anywhere i can set what user comedian mail writes the files as? |
22:20.18 | darkskiez | patch app_vm.c |
22:20.19 | darkskiez | :) |
22:20.43 | asteriskmonkey | darkskiez: littme more info man :) |
22:20.52 | asteriskmonkey | you saying there is a patch file i have to compile? |
22:21.08 | darkskiez | asteriskmonkey: no, you have to write some code and modify asterisk to do that. |
22:21.29 | darkskiez | asteriskmonkey: its adding approx two lines of code i believe for a quickhack(tm) |
22:21.42 | asteriskmonkey | crap, what i dont understand is why after i upraded it writes wavs as root not asterisk anymore |
22:21.43 | FuriousGeorge | i noticed today that my fxo sometimes fail to detect a hangup |
22:22.02 | asteriskmonkey | tried chowning everything asterisk wont start as user asterisk... |
22:22.11 | darkskiez | asteriskmonkey: probably because you are not running asterisk as user asterisk anymore. |
22:22.28 | darkskiez | asteriskmonkey: checked the permissions on the /dev bits and bobs? |
22:22.52 | *** join/#asterisk logicalonline (n=Ken@209.242.52.25) |
22:22.52 | asteriskmonkey | darkskiez: ah no not yet.. will that cause it not to start |
22:23.14 | logicalonline | hello, has anyone gotten the intercom button to work on the aastra 480i? |
22:23.27 | darkskiez | asteriskmonkey: probably, what is the error it does starting as not-root |
22:23.42 | asteriskmonkey | it dosnt give me one it simple goes back to cli |
22:24.42 | FuriousGeorge | i designed my dialplan so that if one channel was busy it would use another technology to dialout. obviously i prioritize by whats cheapest. i noticed today that my fxo werent doing so hot at detecting a hangup, and would just keep going between the call out macros on iax2 and zap/g1 (neither succeding). im looking at my logs and i notice this: Ooh, voice format changed to 4 Dunno what to do with control type 15 |
22:24.52 | FuriousGeorge | wow that was a lot, sorry about that |
22:25.12 | asteriskmonkey | darkskiez: it dosnt give me one.. just goes back to cli is there a command that i can type to get an error msg? |
22:25.14 | *** part/#asterisk puppet (i=puppet@1av10.nu) |
22:25.16 | FuriousGeorge | but what's control type 15? |
22:25.50 | darkskiez | asteriskmonkey: su asterisk; asterisk -cvvvvvvvvdddddddd |
22:27.36 | X-Rob | darkskiez - uh. You are aware that verbose and debug only do up to 4, right? |
22:27.41 | X-Rob | anything beyond that == 4. |
22:27.55 | FuriousGeorge | http://pastebin.ca/26544 <---- anyway thats my logs. if you look you see it does go between my macros but isnt calling out like it should when zap is busy |
22:27.58 | darkskiez | X-Rob: my keyboard repeat is too high. |
22:28.13 | VxJasonxV | hmmm |
22:28.27 | VxJasonxV | something is causing the asterisk console to be colorless |
22:28.31 | VxJasonxV | while being run in screen :( |
22:28.50 | X-Rob | screen is filtering out the colours |
22:28.56 | X-Rob | set your TERM to be ANSI before you start screen |
22:28.58 | X-Rob | NEXT!!! |
22:29.01 | VxJasonxV | thanks :) |
22:29.50 | asteriskmonkey | darksiez: seems to work now... i ran it from the user asterisk and runnign as asterisk now.. wierd.. couldnt get it to go under root... |
22:29.54 | VxJasonxV | X-Rob, hmmm, ANSI or ansi ? |
22:30.03 | X-Rob | ansi, lower case. |
22:30.12 | VxJasonxV | apparently that didn't do it either. |
22:30.23 | VxJasonxV | I just did: expert TERM=ansi && asterisk -vvvvddddc |
22:30.37 | X-Rob | Yes |
22:30.42 | [hC] | export |
22:30.44 | [hC] | not expert |
22:30.45 | VxJasonxV | oops |
22:30.46 | X-Rob | then TERM=ansi screen -R |
22:30.57 | X-Rob | notice how I said 'before you start screen'? |
22:30.57 | VxJasonxV | well, regardless, it didn't error [hC], so your point is moot :) |
22:30.59 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
22:31.09 | VxJasonxV | I typed it correctly in the term, just not here :P |
22:31.13 | [hC] | ah |
22:31.14 | [hC] | :) |
22:31.15 | VxJasonxV | indeed I did not. |
22:31.46 | *** join/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de) |
22:31.49 | darkskiez | editline doesnt seem to work in remote attachment on osx :( |
22:31.55 | *** join/#asterisk Weezey (i=WeezeyD@206.210.109.233) |
22:32.12 | Weezey | anyone have WRT54GP2-NA firmware? |
22:32.34 | Laibsch | Igbothom: Did you ever install AMPortal on a Debian machine? The debs seem to be totally borked. |
22:32.37 | VxJasonxV | hmmmmmmm |
22:33.00 | VxJasonxV | X-Rob, is there anything else a noob may have looked over? :P |
22:33.14 | Igbothom | Laibsch; Xorcom Rapid is Debian based, and it was installed on that while I was playing with it |
22:33.21 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
22:33.44 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
22:35.16 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
22:35.35 | Laibsch | Igbothom: For example they try to overwrite files in /etc/asterisk/. The current debs from http://rapid.dotsrc.org/ will not install on a Debian testing system. Xorcom Rapid sounds nice. |
22:36.05 | Igbothom | it did, but the config files are a bit... non-standard |
22:36.58 | Laibsch | What do you mean "it did"? |
22:37.14 | *** part/#asterisk logicalonline (n=Ken@209.242.52.25) |
22:37.27 | Laibsch | The server I just tried to set up flat out refuses to install those debs. |
22:39.35 | *** join/#asterisk cp5 (n=samy@adsl-69-110-135-211.dsl.irvnca.pacbell.net) |
22:39.40 | fifer | I'm having a logger issue. |
22:39.54 | Weezey | heh, I thought that said, I have a longer issue |
22:40.04 | cp5 | what signalling types in zaptel/zapata are robbed bit signalling? i can't seem to find any documentation on it |
22:40.06 | Weezey | and I was like, what are you impaling chicks now? |
22:40.13 | fifer | I have setup log rotation but after the logs are rotated they are not written to by the logger anymore unless I do a logger reload |
22:40.15 | fifer | Any ideas? |
22:40.40 | fifer | I'm seeing this on two very diferent * setups, one Debian the other RHEL |
22:40.53 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
22:42.24 | *** join/#asterisk fugitivo (n=ajf@209.13.241.231) |
22:42.57 | *** join/#asterisk syle (n=blag@unaffiliated/syle) |
22:43.53 | *** join/#asterisk sapo_original (n=Fenix@200.138.76.58) |
22:44.02 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
22:45.18 | Laibsch | Igbothom: Does Xorcom Rapid use the debs from http://rapid.dotsrc.org/? I doubt it. |
22:45.31 | Igbothom | as do I :) |
22:45.56 | Laibsch | Can you please explain what you meant by "it did"? What is it? |
22:46.24 | justinu | cp5: e&m, fxo, fxs, etc. |
22:46.45 | sapo_original | hi guys |
22:46.47 | cp5 | justinu, grazi |
22:47.09 | sapo_original | anybody knows dan, the onwer the project diax? |
22:47.30 | justinu | cp5: basically anything other than ISDN PRI/SS7 |
22:48.03 | *** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net) |
22:48.20 | cp5 | justinu, awesome, thanks. is e&m ALWAYS robbed bit? |
22:48.25 | justinu | correct |
22:48.32 | cp5 | ah |
22:50.19 | ender | what is 'robbed bit' ? |
22:51.16 | justinu | it's when you take bits away from the audio path and use them as signalling bits |
22:53.00 | sapo_original | ??? |
22:54.36 | lters | it is a technology for tl signalling |
22:56.04 | *** join/#asterisk fugitivo (n=ajf@209.13.241.231) |
22:58.37 | ender | interesting. We're using em_w to communicate w/ a Fujitsu PBX |
23:00.14 | |baby| | alguien habla español? |
23:02.24 | *** part/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com) |
23:03.12 | eKo1 | |baby|: Si. |
23:04.16 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
23:04.18 | eKo1 | robbed bit == inband |
23:04.27 | |baby| | eKo1 sabes si existe el patch bristuff para el ultimo asterisk del CVS? |
23:05.30 | znoG | |baby|: si, que pasa? |
23:06.40 | |baby| | eso, k me falla el atxfer cuando pulso la tecla para transferencia asistida, y marco la extension... si no contesta se cuelga la llamada en proceso y no puedo recuperarla |
23:07.03 | |baby| | uso el bristuff-0.2.0-RC8f-CVS |
23:07.13 | eKo1 | No uso bri |
23:07.32 | |baby| | ok :( |
23:07.36 | |baby| | y como haces para transferir llamadas? |
23:08.02 | *** join/#asterisk screenname (n=appdev@pcp0011162839pcs.pennington.tn.nash.comcast.net) |
23:08.06 | eKo1 | pues, los teléfonos tienen un botón que dice 'transfer' |
23:08.21 | |baby| | yo uso PAP2 con un telefono normal |
23:08.26 | |baby| | la tecla R no funciona tampoco |
23:08.28 | eKo1 | yo lo oprimo, marco el número, y viola, se tranfiere la llamada. |
23:08.41 | znoG | probaste con la tecla numeral, baby? |
23:08.53 | |baby| | znoG #? |
23:08.56 | znoG | sep |
23:09.04 | |baby| | si, pero yo tengo en features.conf |
23:09.10 | eKo1 | puede ser que el pap2 no puede hacer transferencias |
23:09.12 | |baby| | puesto atxfer => # |
23:09.22 | *** part/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de) |
23:09.28 | |baby| | y cuando la marco, funciona... |
23:09.44 | |baby| | pero si la extension a la que llamo no contesta... no puedo recuperar la llamada :( se cuelga! |
23:10.33 | Vco | umm.....so...Never had a working H323 channel before... |
23:10.42 | Vco | now it worked right out of the box... |
23:10.53 | justinu | neat |
23:10.59 | Vco | how the hell do i restrict access? |
23:11.33 | justinu | never worked on h323 before |
23:11.44 | FuriousGeorge | isnt there some way to allow zap channels to initiate a transfer with a # |
23:11.56 | Vco | flash |
23:11.59 | Vco | no? |
23:12.07 | cripito | wow!! #asterisk in spanish? |
23:12.12 | FuriousGeorge | flash isnt a pound |
23:12.25 | |baby| | cripito yeahh! XD |
23:12.53 | justinu | nosotros tenemos mas influencia con sus hijos que tu tiene |
23:12.54 | FuriousGeorge | ay dios mio, el asteric bilinguo |
23:12.59 | Vco | yea..just pound...ext....pound |
23:13.00 | justinu | pero los queremos |
23:13.09 | cripito | :) bueno si la cosa es en espa~Nol ok |
23:13.10 | cripito | :P |
23:13.11 | justinu | creado y gegalo de los angeles, juana's addiccion! |
23:13.14 | FuriousGeorge | tu tienes* justinu |
23:13.36 | justinu | i'm surprised that was my only mistake |
23:13.44 | viLeR | yo no sabia que había tanto hispanoparlante por aca |
23:13.52 | |baby| | joer... |
23:13.52 | justinu | no habla espanol |
23:13.54 | cripito | pues parece q hay una buena cantidad :)) |
23:13.56 | |baby| | no si ahora todos hablremos español |
23:13.57 | FuriousGeorge | y yo tampoco |
23:13.58 | |baby| | XDDDD |
23:14.00 | Vco | but seriously, this H323 allows anyone to call through the server |
23:14.05 | Vco | which i kinda dont' like... |
23:14.41 | cripito | well definitelly the spanish comunity in the asterisk is more big that the ppl think at the first time ;) |
23:14.44 | cripito | any way... |
23:15.16 | FuriousGeorge | Vco: de ahora en adelante no se hable el ingles aca. guerra popular! |
23:15.26 | FuriousGeorge | we can be zapatistas |
23:15.49 | |baby| | jaja |
23:16.00 | |baby| | nadie usa atxfer para transferir llamadas??? :( |
23:16.29 | FuriousGeorge | i use regular transfer on zap and sip channels |
23:16.42 | FuriousGeorge | was just wondering aloud if there was some way to initiate a transfer using # on zap |
23:16.49 | cripito | me 2.. |
23:16.54 | Vco | yea.. |
23:17.00 | Vco | # dial ext and # again |
23:17.08 | FuriousGeorge | |baby|: por que? |
23:17.14 | cripito | basically the phones have support for that... |
23:17.35 | FuriousGeorge | cripito: Vco: the phones need to support # transfer? |
23:17.51 | FuriousGeorge | as in # flashes the line when its off the hook? |
23:17.54 | Vco | far as i know it was just handled by the fxs |
23:18.09 | FuriousGeorge | that cant be |
23:18.09 | |baby| | el telefono lo soporta, porque la llamada se transfiere si el otro contesta... |
23:18.09 | justinu | could be |
23:18.15 | |baby| | el problema es cuando no contesta nadie, que la llamada se corta |
23:18.22 | Vco | like, it's a zap thing..is it not? |
23:18.29 | |baby| | con attended transfer unicamente |
23:18.32 | |baby| | *2 |
23:18.37 | |baby| | ;atxfer => *2 ; Attended transfer |
23:18.41 | justinu | anyone need or is interested in RTCP support in asterisk? |
23:18.48 | eKo1 | maybe you should flash them before to make sure they are there |
23:19.03 | Vco | or you mean attended xfer? |
23:19.39 | cripito | george incluso in soportar el # |
23:19.42 | FuriousGeorge | i think means "not a blind transfer" |
23:19.45 | cripito | :) |
23:19.52 | *** join/#asterisk mattHelm (n=root@adsl-68-90-140-81.dsl.fyvlar.swbell.net) |
23:20.00 | Vco | i just use parking |
23:20.03 | cripito | casi todo nuevo sip phone have un key for that |
23:20.14 | Vco | or blindxfer |
23:20.29 | cripito | i mean almost all the new sip phones have support for blind transfer or attended transfer |
23:20.45 | Vco | thats for a sip phone tho,,,not a zap |
23:21.05 | cripito | in zap.... basically the features.... |
23:21.25 | cripito | u can program the features to work with the phones... |
23:21.53 | FuriousGeorge | |baby|: o sea: si tu inicias a un xfer, y tu cuelgas la lina, disconectas a la persona que estas tratando de transferir |
23:21.54 | cripito | or look in the *XX in the chan_zap.c |
23:22.23 | cripito | :D attd transfer right? |
23:22.23 | mattHelm | Anyone want to entertain a "no d-channels" question? |
23:22.38 | justinu | is there any way to get the latest polycom sip image and boot loader? it's not on the freedomphones link |
23:23.19 | justinu | i don't really care about the sip image, but i'd like the new boot loader so I can do HTTPs provisioning |
23:23.25 | Vco | hmm..k..specified h323 to only listen on internal interface |
23:24.00 | Vco | would proabably be a good plan to have that by default |
23:24.22 | |baby| | FuriousGeorge no, si yo inicio una atxfer, y la persona de la extension que llamo no me contesta... no puedo recuperar la llamada de nuevo, se cuelga |
23:24.24 | Vco | rather than anonymous-assclown=sure why not |
23:27.11 | FuriousGeorge | |baby|: ah. no es posible pasar cuantos segundos quieres que intenta el atxfer=> 1...${EXTEN},30) |
23:27.16 | FuriousGeorge | ? |
23:28.07 | |baby| | creo k no |
23:29.05 | FuriousGeorge | pues la verdad es que no se decirte. siempre he usado el xfer normal del zap, o implementado por el cliente |
23:29.16 | FuriousGeorge | en sip. |
23:30.21 | cripito | baby basicamente te pasa porque tu ya colgaste... |
23:30.28 | cripito | no hay retorno atras |
23:30.44 | cripito | es blind |
23:31.31 | |baby| | cripito no, yo no cuelgo el telefono |
23:32.22 | |baby| | yo marco *3 (musicohold a la persona que me llama), me da tono... marco la extension y si la extension no contesta... y la llamada se corta y yo no puedo recuperarla |
23:32.32 | |baby| | si marco *3 de nuevo, o cualquier cosa... no vuelvo a la llamada |
23:32.56 | |baby| | en cambio si la extension me contesta, hablo con la extension y cuelgo .. la llamada se transfiere |
23:33.00 | marc324 | what motherboard to get for opteron cpus? |
23:33.02 | cripito | interesante |
23:33.18 | cripito | pastebin tu features.conf |
23:33.57 | lters | justinu, I hope u find the firmwares. |
23:34.17 | lters | justinu, cause I will need them too... |
23:34.27 | cripito | yeap .. keep us posted ;) |
23:35.39 | |baby| | [featuremap] |
23:35.39 | |baby| | blindxfer => *3 ; Blind transfer |
23:35.39 | |baby| | disconnect => * ; Disconnect |
23:35.39 | |baby| | automon => *1 ; One Touch Record |
23:35.39 | |baby| | atxfer => # ; Attended transfer |
23:35.49 | |baby| | tengo las teclas cambiadas # y *3 |
23:38.19 | cripito | <PROTECTED> |
23:38.19 | |baby| | si, en realidad para atxfer estoy marcando # |
23:39.10 | justinu | iters: no luck yet |
23:39.28 | mattHelm | <PROTECTED> |
23:39.35 | mattHelm | What is the cause of that usually? |
23:39.45 | justinu | you're not receiving any T1 signal |
23:40.04 | mattHelm | Telco's fault? |
23:40.10 | justinu | possible |
23:40.22 | justinu | can you loop up your niu towards yourself to check your cabling? |
23:40.42 | mattHelm | No D-channels available! Using Primary on channel anyway 24 |
23:40.57 | justinu | without a T1 carrier, you can't have a D channel, so makes sense |
23:41.04 | mattHelm | Hmmm. I'm from the old school of external CSU/DSU |
23:41.20 | justinu | new circuit? |
23:41.35 | mattHelm | No, has been mostly working for a few months. |
23:41.42 | justinu | ok, anything that you know of changed? |
23:41.44 | mattHelm | A few dropped calls every day. |
23:41.48 | mattHelm | Nothing changed. |
23:41.50 | *** join/#asterisk paxr0 (n=Walter@200-126-112-99.bk8-dsl.surnet.cl) |
23:42.04 | justinu | worth a call to the telco then |
23:42.11 | justinu | they can usually get things going pretty quickly |
23:42.47 | mattHelm | Thanks, that's what I thought but don't know how to tell what's going on the the digium card. |
23:43.01 | justinu | seems like you got the right info from it somehow |
23:43.54 | mattHelm | <PROTECTED> |
23:52.33 | *** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
23:52.58 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
23:53.34 | *** join/#asterisk wmandra (i=wmandra@pcp04943183pcs.verona01.nj.comcast.net) |
23:55.52 | n3u7 | yst another night trying to get asterisk installed |
23:56.04 | n3u7 | www.darkphiber.ca/asterisk |
23:56.06 | drumkilla | n3u7: make install!!! |
23:56.32 | n3u7 | drumkilla:I'm working on ztool right now |
23:56.37 | drumkilla | you need newt |
23:56.39 | n3u7 | *zttool |
23:56.46 | encode | X-Rob: are you around? |
23:56.50 | n3u7 | it boots and quits |
23:57.08 | n3u7 | which is an improvment over ubunto , knoppix and sarge |