| 00:00.06 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 00:00.26 | JT | orlock: acetylene is some dangerous stuff |
| 00:00.27 | dotSlashW | I have a voIP line which from a provider. Now it's connected to a voIP gateway, which is connected to the internet, and has an analog port (line 1) connected to my trixbox |
| 00:00.38 | dotSlashW | can I skip using this voIP gateway, and connect my trixbox directly to my voIP provider ? |
| 00:00.40 | orlock | yeah |
| 00:00.45 | dotSlashW | I mean, get the voIP gateway out of the middle |
| 00:00.47 | dual-man | i am getting: unable to re-open dsp device, device or resource busy |
| 00:00.52 | dual-man | what is the problem |
| 00:00.54 | JT | dotSlashW: yeah but this isn't the channel for trixbox |
| 00:00.55 | dotSlashW | handle the line from within trixbox |
| 00:00.58 | dual-man | i am trying to dial the consol |
| 00:01.01 | N9URK | wunderkin -> thanks for the reply -> will this precent the reinvite? exten => _9XXXXXXXXXXX,1,DIAL(sip/teliax2/${EXTEN:1}) |
| 00:01.03 | dotSlashW | is that doable ? |
| 00:01.06 | orlock | a mate at high school did the garbage bag thing once, and his neighbour jumped off his balcony cos he thought a gas line was blowing |
| 00:01.14 | N9URK | wunderkin prevent not precent |
| 00:01.18 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 00:01.25 | Mahmoud | creature_ |
| 00:01.36 | JT | orlock: how high was the balcony? |
| 00:01.40 | wunderkin | no |
| 00:01.47 | dotSlashW | JT sure, how about if I ask the same about asterisk... |
| 00:02.03 | dotSlashW | is that OK ? I mean.. the system is almos the same, right ? |
| 00:02.18 | N9URK | wunderkin -> I have all of my contexts set to canreinvite=yes. Is there somewhere else that might stop the reinvite? |
| 00:02.19 | JT | dotSlashW: yes it will work |
| 00:02.29 | JT | not really, trixbox has a big messy gui on top |
| 00:02.30 | wunderkin | N9URK, i don't know what else to check for |
| 00:02.43 | orlock | JT: only 1st floor.. but it was out bush, pretty big gaps between houses |
| 00:02.43 | dotSlashW | yes, new interface sucks |
| 00:02.56 | N9URK | wunderkin ok thanks for trying |
| 00:03.10 | JT | dotSlashW: it's always sucked |
| 00:03.17 | N9URK | Anyone else have any ideas as to why a sip call might not be reinviting? |
| 00:03.32 | orlock | JT: ever read Michael Yon's stuff? |
| 00:03.39 | JT | orlock: no? |
| 00:04.13 | orlock | JT: ex-special forces guy who is now a journalist/blogger, spends a lot of time in Iraq travelling with the troops |
| 00:04.21 | JT | ah ok |
| 00:04.39 | orlock | JT: he has a book about his earlier years.. he spent a lot of time making bombs as a kid :) |
| 00:04.53 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:04.59 | JT | heh, you get called a terrorist these days for doing that |
| 00:05.17 | *** join/#asterisk SomethingISODD (n=Dan@NTL208H101-91-124.nt.net) |
| 00:05.22 | orlock | yeah.. funny thing is, he was "taught" how to do it proprely later by the military |
| 00:06.36 | k-man_ | jt, you going to cebit in sydney this year? |
| 00:06.46 | JT | probably |
| 00:06.49 | JT | when is it again |
| 00:06.58 | k-man_ | 1-3 may |
| 00:06.58 | SomethingISODD | hello all, anyone here know alot about pwlib? if so can yo u please take a look at this error http://www.pastebin.ca/377749 |
| 00:07.05 | JT | hmm |
| 00:07.12 | k-man_ | jt, registrations are open |
| 00:07.13 | k-man_ | http://www.cebit.com.au/ |
| 00:07.21 | JT | lol waste of time |
| 00:07.28 | k-man_ | jt, yeah... |
| 00:07.31 | k-man_ | jt, it usually is |
| 00:07.31 | JT | you know you can always get in, without registering |
| 00:07.38 | k-man_ | you can? |
| 00:07.40 | k-man_ | how? |
| 00:07.41 | JT | yes |
| 00:07.56 | k-man_ | without paying? |
| 00:08.00 | JT | err, the rows of computers in the exhibition centre foyer for self ticket printing |
| 00:08.03 | JT | dude |
| 00:08.06 | wunderkin | 2 leg discount |
| 00:08.08 | JT | NO-ONE pays except exhibitors |
| 00:08.26 | JT | they just try to trick people so they can grab tonnes of marketing details |
| 00:08.32 | JT | even if people don't go |
| 00:08.33 | k-man_ | jt, oh.. i dunno... afaik, if you don't pre-register, you have to use tome trikery to get in free |
| 00:08.46 | JT | k-man_: that's never ever been the case |
| 00:08.51 | k-man_ | otherwise there is a $25 charge at the door |
| 00:09.00 | JT | they have computers specifically there for you to sign in for free |
| 00:09.03 | JT | that's bullshit |
| 00:09.03 | k-man_ | jt, i dissagree... |
| 00:09.05 | JT | always has been |
| 00:09.13 | JT | they just say that |
| 00:09.15 | k-man_ | jt, oh well, we can check this year |
| 00:09.20 | JT | to trick people into signing on the web site |
| 00:09.34 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:09.34 | JT | k-man_: the last 3 years, they've had computers there to self sign in |
| 00:09.42 | k-man_ | jt, oh there is something there about asterisk |
| 00:09.57 | JT | probably a couple of companies doing solutions or something |
| 00:10.03 | k-man_ | jt, i know the computers of which you speak... just i thought they were for getting in once you had preregistered.. |
| 00:10.22 | k-man_ | jt, its not worth arguing about though.. i have registered, you don;t have to if you don't want to and we can both be happy |
| 00:10.33 | k-man_ | jt, and we can prove each other wrong when we get there this year ;) |
| 00:10.37 | JT | nah they let you fill in all your details there |
| 00:10.44 | JT | i've registered online years ago |
| 00:10.52 | JT | and still get cebit spam every year |
| 00:10.58 | JT | a few times a year |
| 00:11.13 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 00:11.22 | JT | it's not really a consumer show |
| 00:11.25 | JT | that's why it's free |
| 00:11.53 | JT | the exhibitors pay, to try and sell you (hopefully a business person in control of $$$) lots of stuff |
| 00:12.08 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-12-203.hsd1.fl.comcast.net) |
| 00:12.32 | k-man_ | jt, yeah, i have been many times |
| 00:12.42 | k-man_ | jt, in the good old days i used to enjoy it and there was lots of cool free stuff given away |
| 00:12.57 | k-man_ | these days its not like that any more... but i still like to go every now and then |
| 00:13.21 | JT | yeah when they combined with interop they had a free coffee lounge run by souther cross cable, and some places had alcoholic slush puppies, a little excessive ;) |
| 00:13.37 | JT | not worth spending more than a couple hours there |
| 00:13.47 | k-man_ | jt no |
| 00:13.52 | orlock | i only ever went to hassle the vendors with annoying questions |
| 00:13.53 | RoyK | http://www.conservapedia.com/Main_Page oh my gods |
| 00:14.43 | ryanca | I asked earlier, but nobody responded.... When someone calls my DID (which is an IAX2 chan) there is no telco ring on the caller side even if I wait before doing an Answer(). Is there any way to fix that? Same thing with 2 DID providers. |
| 00:15.27 | JT | ryanca: calling from the pstn? |
| 00:15.50 | ryanca | JT: correct. |
| 00:16.15 | ryanca | I mostly tested calling from a cell phone. |
| 00:16.18 | JT | ryanca: your provider should send ringing indications to the caller |
| 00:16.27 | wunderkin | are you calling to a phone or ivr |
| 00:16.55 | ryanca | the initial context is an anut-attendant. |
| 00:17.10 | JT | anut? |
| 00:17.14 | type0 | I remember being at defcon one year and rolling over to the Sun party at the Hard Rock Cafe in vegas, holy shit they had a shitload of free booze |
| 00:17.15 | ryanca | auto |
| 00:17.32 | ryanca | damn laptop keyboard. |
| 00:17.57 | wunderkin | where would the ringing come in then |
| 00:18.41 | ryanca | If I have it Wait() for a few seconds, shouldn't ringing be indicated until I Answer()? |
| 00:18.47 | JT | no |
| 00:19.05 | SomethingISODD | hello all, anyone here know alot about pwlib? if so can yo u please take a look at this error http://www.pastebin.ca/377749 |
| 00:19.10 | RoyK | http://www.conservapedia.com/Kangaroo <-- see origin |
| 00:20.45 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:20.48 | JT | ryanca: it is quite normal when ringing an IVR for there to be no ringing indication before the call is answered by the ivr |
| 00:20.59 | ryanca | So there's no way to give a couple rings before the call is answered (as far as the caller's billing is concerned, I know I can just play a ringing tone) |
| 00:21.19 | JT | you can, but it seems unnecessary |
| 00:21.33 | ryanca | How would I do it? |
| 00:21.42 | JT | PlayTones |
| 00:21.48 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 00:22.23 | ryanca | Right, but if I do that, the call has still been answered as far as pstn is concerned. |
| 00:22.46 | j0 | some callers complain about hearing their own voice... it seems to happen on my grandstream phone as well as my linksys pap2... so is it a problem with asterisk, or my phones? |
| 00:23.15 | ryanca | j0: what outbound channel are you using? |
| 00:23.24 | j0 | ryanca: sip and iax |
| 00:23.31 | JT | ryanca: yes, that's right, PlayTones before Answer might work |
| 00:23.40 | JT | if early media is supported by provider |
| 00:24.15 | ryanca | JT: Hmm. |
| 00:24.15 | JT | ryanca: i personally wouldn't bother and would put them straight in IVR |
| 00:24.34 | JT | people don't like to be delayed when they are going to have to deal with an IVR anyway |
| 00:26.06 | *** join/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 00:26.16 | mtgll | greetings |
| 00:26.18 | ryanca | I suppose. |
| 00:27.02 | mtgll | what else is there to hpec config besides changing the echocancelling line isn zapata.conf not much on the web yet? |
| 00:28.15 | ryanca | It's just a personal system, and I've been trying to decide how I want it to answer... I was thinking have a brief announcement that allows an extension to be dialed before dialing the hardphones in my house. |
| 00:29.32 | *** join/#asterisk fr33bi3 (n=fr33bi3@122.164.147.161) |
| 00:29.51 | ryanca | Trying to decide how to set it up to prevent people from being confused by it. |
| 00:31.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:32.02 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 00:40.34 | *** join/#asterisk christo (i=christo@courgette.jml.net) |
| 00:40.40 | christo | aye |
| 00:43.05 | christo | If I move a call file into the outgoing spool, and set the associated context extens to: exten => s,1,Answer exten => s,2,agi,testapp.pl Is there any difference in how that AGI will operate compared to if the call was incoming? |
| 00:44.04 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:49.52 | *** join/#asterisk BlueSkyFrog (n=bluskyfr@ppp170-170.lns1.per1.internode.on.net) |
| 00:51.49 | BlueSkyFrog | what is asterisk ideal server requirements |
| 00:52.03 | RevPepse | depends on your requirements :) |
| 00:52.36 | BlueSkyFrog | well we are looking to roll it out to about 40 to 200 ip phones |
| 00:54.18 | SomethingISODD | hello all, anyone here know alot about pwlib? if so can yo u please take a look at this error http://www.pastebin.ca/377749 |
| 00:59.04 | *** join/#asterisk netsurfer (n=netsurfe@user-514f09ee.l1.c3.dsl.pol.co.uk) |
| 01:02.48 | *** join/#asterisk mog (i=ejabberd@71.207.215.93) |
| 01:02.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:03.30 | BlueSkyFrog | anyone online to helpme on asterisk |
| 01:05.27 | JT | BlueSkyFrog: a relatively recent machine, with RAID1 and redundant power supplies should do it |
| 01:05.41 | JT | if you're worried about uptime, you could use 2 servers |
| 01:06.57 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 01:07.24 | J4k3 | wow... that nokia e60 is sexy and the price is right |
| 01:07.35 | J4k3 | (shows how overpriced the linksys wip330 is...) |
| 01:07.46 | Qwell | J4k3: link? |
| 01:09.32 | BlueSkyFrog | thanks JT |
| 01:10.06 | masked | J4k3: what radio does it have? |
| 01:10.28 | BlueSkyFrog | what is the difference from asterisk the opern source version compared to asterisk the business version |
| 01:10.39 | J4k3 | http://www.newegg.com/Product/Product.asp?Item=N82E16875205022 |
| 01:10.55 | J4k3 | masked: its a WCDMA+GSM+WIFI+Bluetooth nightmare ;) |
| 01:11.18 | J4k3 | (how the hell do you get wifi and bluetooth to co-exist in the same box nicely? I have a hard enough time using wifi and bluetooth-based data to my CDMA handset at the same time) |
| 01:11.41 | J4k3 | well, wifi and bt-based data to my cdma handset, from my laptop |
| 01:11.43 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 01:11.47 | J4k3 | (I wish my CDMA handset had wifi ;) |
| 01:11.59 | masked | does it have a multiband antenna, chip or? |
| 01:12.18 | J4k3 | no idea |
| 01:12.28 | J4k3 | buy me one and a special nokia screwdriver :) |
| 01:12.40 | J4k3 | or are you talking about my laptop? |
| 01:12.42 | masked | i got them screwdrivers |
| 01:12.46 | masked | nah the phone |
| 01:12.51 | orlock | what are they? |
| 01:12.54 | orlock | torx? |
| 01:12.56 | J4k3 | (laptop is a wistron neweb cm9 a/b/g minipci card, and a bluetooth USB adapter on the back) |
| 01:12.59 | masked | i see the netgear branded wifi phones have a 5.5dBi antenna |
| 01:13.05 | masked | yeah torx |
| 01:13.07 | J4k3 | like... T2.5 or something... |
| 01:13.09 | J4k3 | itty bitty |
| 01:13.17 | J4k3 | then theres motorola and their 5-point phillips style. |
| 01:13.34 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 01:13.39 | J4k3 | netgear's phones appear to be the same chipset as the f1000g I just filled the RMA form out |
| 01:13.42 | J4k3 | at least their $200ish flip phone is |
| 01:14.21 | J4k3 | now... the real question is... does the E60 support any decent protocols for mobile wifi use, or just the typical g711/g729 crap. |
| 01:14.27 | J4k3 | I mean... it *should* have GSM at least ;) |
| 01:14.34 | J4k3 | (considering its primary task at life is GSM) |
| 01:14.44 | masked | i'm only interested if it does mobile ad-hoc |
| 01:14.54 | masked | heh |
| 01:15.17 | masked | does it have the horsepower to do iLBC? |
| 01:15.23 | J4k3 | mobile adhoc is easy... throw an AP on the roof of your car or in a backpack. |
| 01:15.43 | J4k3 | hrm... dunno, ilbc would be best for wifi... |
| 01:15.52 | J4k3 | (works darned well when tested from my laptop) |
| 01:16.10 | masked | yeah works great here on my laptop |
| 01:16.45 | masked | ok i have this dreaded winxp install to do for my brother |
| 01:17.02 | J4k3 | haha |
| 01:17.06 | masked | bbl eta 7 hours |
| 01:17.52 | *** join/#asterisk dj-fu (n=ajc@202-74-195-152.ue.woosh.co.nz) |
| 01:19.47 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 01:20.00 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-73-237.vic.bigpond.net.au) |
| 01:20.02 | iceyp | hey guys, anyone here running freebsd and got mpg123 to run properly and not at 99% CPU |
| 01:20.24 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 01:21.03 | J4k3 | I have E61 the SIP functionality the same as E60. The phone support not only the g711x codecs, support: ilbc, g729, amr. I use my phone ilbc codec. The |
| 01:25.23 | *** join/#asterisk mindstate (n=debianch@pool-71-164-112-7.albyny.east.verizon.net) |
| 01:26.14 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:29.28 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 01:31.32 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:32.47 | iceyp | anyone running bsd with asterisk? |
| 01:33.49 | *** join/#asterisk Giofe (n=opera@190.81.4.169) |
| 01:40.21 | *** join/#asterisk FaithX (n=faithful@ppp236-58.static.internode.on.net) |
| 01:40.55 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 01:40.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:40.55 | Nugget | Yes. I do. |
| 01:40.58 | Nugget | I don't recommend it. |
| 01:41.09 | Nugget | (although it's less terrible now than it was a year ago) |
| 01:41.46 | JT | J4k3: what codecs does it support? |
| 01:41.47 | JT | telnet |
| 01:41.47 | Nugget | telnet is eeeeeeevil! |
| 01:41.52 | JT | triggers are evil! |
| 01:42.10 | Nugget | they can't be that bad, with only one "e" |
| 01:42.34 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:43.27 | JT | never underestimate the power of correct english |
| 01:45.36 | Nugget | I think that using correct english on irc violates the rfc. |
| 01:45.55 | mishehu | never misunderestimate the goodliness of gw bush english. |
| 01:47.29 | *** join/#asterisk The_DoC^ (n=me@pool-71-244-137-146.chi.dsl-w.verizon.net) |
| 01:50.24 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:51.49 | *** join/#asterisk fafnir (n=fafnir@unaffiliated/fafnir) |
| 01:52.04 | fafnir | any providers with 800 numbers with multiple points of ring? |
| 01:52.39 | J4k3 | lets take english 101 with GW Bush and Kevin Federline. |
| 01:53.56 | J4k3 | JT: don't you have an E60 or E90 or some other nokia wifi-enabled phone? |
| 01:56.11 | JT | J4k3: no |
| 01:56.15 | JT | why would i? :P |
| 01:57.10 | J4k3 | oh... I knew somebody mentioned having one that had an extremely short nick |
| 01:57.11 | J4k3 | ;) |
| 01:57.20 | J4k3 | but thats all I can remember |
| 01:59.01 | JT | so what protocols does it support? |
| 01:59.04 | JT | codecs even |
| 01:59.21 | JT | i carry 2 phones, a pretty basic gsm unit, and a pretty basic cdma unit |
| 02:01.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:04.04 | J4k3 | gsm+wcdma+wifi(sip)+bt |
| 02:04.36 | J4k3 | and the sip support looks outstanding |
| 02:04.36 | J4k3 | even gsm and ilbc codecs |
| 02:05.16 | J4k3 | I think I'll buy one from somebody with a decent return policy and pick up a prepaid sim from somebody |
| 02:05.21 | J4k3 | or borrow somebody else's sim |
| 02:06.05 | J4k3 | "hey... your boyfriend is in jail? Can I borrow his SIM? Can I borrow his side of the bed?" |
| 02:06.14 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 02:07.14 | riddlebox | does anyone have multiple phone lines provided by broadvoice? |
| 02:10.54 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 02:17.57 | *** join/#asterisk sklim (n=shankuo@60.49.242.225) |
| 02:18.57 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:18.58 | iceyp | hey guys, anyone here running freebsd and got mpg123 to run properly and not at 99% CPU |
| 02:19.07 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:19.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:19.29 | shmaltz | he everyone |
| 02:19.32 | shmaltz | hi* |
| 02:19.47 | *** join/#asterisk lwh (n=lwh192@rdsl-0270.tor.pathcom.com) |
| 02:20.57 | *** part/#asterisk mog (i=ejabberd@71.207.215.93) |
| 02:21.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 02:22.22 | dual-man | hey everybody i have a question, i just changed my trixbox to a plain asterisk server, i have a t1 card tdm400p that connects into channel banks for local extensions, for some reason I have two extension that don't work, they're dead, and they worked fine earlier today with trixbox |
| 02:22.37 | dual-man | and ideas what the problem could be |
| 02:22.40 | riddlebox | if you want to have say two lines from BV, and have with about 8 phones, what kind of specs on a pc do you need? |
| 02:23.16 | riddlebox | dual-man, what kind of extensions? |
| 02:23.37 | JT | riddlebox: analogue, he mentioned channel bank |
| 02:23.48 | [TK]D-Fender | BV kinda sucks I've had clients ditch them |
| 02:24.17 | riddlebox | [TK]D-Fender, I have used them for a year and a half, and havent had to many problems |
| 02:24.19 | dual-man | when i dial the extension it rings, but no ring on th eother end |
| 02:24.42 | [TK]D-Fender | riddlebox: they've had all sort of RR-DNS issues, flaking out reg servers, etc... |
| 02:25.47 | riddlebox | [TK]D-Fender, honestly, I had some problems connecting once in a while, but since I started using asterisk-1.4, it has stayed connected to them and when I do restart asterisk, it always connects right away |
| 02:26.20 | JT | dual-man: what does asterisk say it's doing? |
| 02:26.36 | dual-man | JT: ringing the extension :D |
| 02:26.37 | *** part/#asterisk Giofe (n=opera@190.81.4.169) |
| 02:26.44 | dual-man | zap/24 ringing |
| 02:26.58 | JT | hmm |
| 02:27.30 | dual-man | i'm plaing with the line build out could that be related? |
| 02:27.41 | dual-man | playing* |
| 02:27.41 | JT | possibly |
| 02:27.54 | JT | you might want to pastebin.ca your zaptel.conf and zapata.conf |
| 02:27.59 | dual-man | ztcfg reloads the zaptel.conf right? |
| 02:28.35 | JT | yes |
| 02:29.28 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:29.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:31.04 | J4k3 | hrm... how many concurrent outgoing calls does a broadvoice 'unlimited' account allow you? |
| 02:31.05 | J4k3 | 1? 2? |
| 02:31.57 | lunaphyte | is there a bit less permissive approach to configuring iptables on an asterisk machine for sip than opening up an inbound range of 10000 udp ports? |
| 02:32.22 | shmaltz | lunaphyte, yes, configure rtp.conf with fewer ports |
| 02:32.29 | shmaltz | but why would you care anyhow? |
| 02:32.47 | lunaphyte | well, i guess just mostly curious. |
| 02:33.24 | lunaphyte | for a server with a generally light load, is there a practical need for a range that large? |
| 02:33.44 | dual-man | JT: can you have a look: http://pastebin.ca/377877 |
| 02:33.57 | shmaltz | lunaphyte, no |
| 02:34.18 | shmaltz | dual-man, theser are FXS cards in the cb? |
| 02:34.46 | JT | dual-man: why on ear is your LBO set to 6? |
| 02:34.50 | JT | earth |
| 02:34.52 | lunaphyte | are there any sort of guidelines i could follow in picking a different value? |
| 02:35.10 | dual-man | JT: o i just did that, tried every setting there :D |
| 02:35.21 | JT | just set it to 0. |
| 02:35.31 | JT | unless your cable is ridiculously long |
| 02:35.34 | shmaltz | LBO shouldn't make much of a difference |
| 02:36.07 | shmaltz | dual-man, run ztcfs -s and then ztcfg -vvvvvv what is the output? |
| 02:36.10 | dual-man | well it is over 150 feet at some points |
| 02:36.26 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 02:36.27 | JT | dual-man: between the channel bank and the asterisk box? |
| 02:36.33 | shmaltz | dual-man, lbo is only for the run between cb and asterisk, not the analog runs |
| 02:36.42 | dual-man | ahhh |
| 02:36.59 | shmaltz | dual-man, run ztcfs -s and then ztcfg -vvvvvv what is the output? |
| 02:37.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:38.17 | dual-man | shmaltz: normal, what it always has been, anything in particular your looking for? |
| 02:38.17 | shmaltz | dual-man just pb it |
| 02:38.17 | J4k3 | wow, nevermind... after reading broadvoice's ToS I don't want to be their customer. |
| 02:38.17 | shmaltz | ~pb |
| 02:38.23 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:38.31 | *** join/#asterisk Stridernzl (n=neville@222-152-248-128.jetstream.xtra.co.nz) |
| 02:38.34 | fall0ut | J4k3: you need origination/termination? |
| 02:38.46 | fall0ut | go to carriers.icall.com |
| 02:38.50 | fall0ut | cheap/fairly reliable |
| 02:39.11 | J4k3 | fall0ut: neither, just curious about their offering... then I Read the tos and started laughing at how big of a ripoff it is about halfway through... :P |
| 02:39.17 | fall0ut | heh |
| 02:39.29 | e-milio | hello all |
| 02:39.59 | shmaltz | helo e-milio |
| 02:40.43 | e-milio | hello shmaltz, having some dificulty here: connect to 'ago://127.0.0.1:4577/cal_log failed: connection refuse |
| 02:40.48 | J4k3 | wow, interesting pricing @ icall, this is worth investigation |
| 02:41.06 | fall0ut | yea |
| 02:41.09 | fall0ut | they're an asterisk/ser shop, too |
| 02:41.10 | shmaltz | e-milio, you sure you got the right channel? |
| 02:41.12 | fall0ut | fwiw |
| 02:41.20 | shmaltz | ~fwiw |
| 02:41.23 | jbot | from memory, fwiw is for {what,whatever} it's worth |
| 02:41.31 | orlock | hmm.... |
| 02:41.37 | dual-man | where is zaptel log output? |
| 02:41.51 | *** join/#asterisk zpertee (n=chatzill@cpe-65-25-121-5.neo.res.rr.com) |
| 02:42.04 | shmaltz | dual-man, dmesg |
| 02:42.08 | e-milio | shmaltz: I ma trying to conf vici-dial but i though i migh find a clue about why * is refusing the connec |
| 02:42.11 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:42.16 | zpertee | is it possible to run shell script from asterisk to shutdown remote computer? |
| 02:42.34 | shmaltz | e-milio, sorry never configured vicidial I can't ehlp you |
| 02:42.36 | zpertee | I already wrote the shell script just need a way to run it |
| 02:42.37 | shmaltz | help* |
| 02:42.50 | shmaltz | zpertee, yes just use System |
| 02:43.16 | shmaltz | ~wiki asterisk |
| 02:43.29 | zpertee | shmaltz: ok |
| 02:47.07 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:48.11 | *** join/#asterisk sfbosch (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 02:48.31 | sfbosch | Has anyone here done bulk provisioning of Polycom phones? |
| 02:48.44 | Damin | Yes. |
| 02:48.52 | lunaphyte | shmaltz: thanks |
| 02:48.53 | sfbosch | I can't get my phone to contact the boot server |
| 02:49.13 | shmaltz | sfbosch, sure did |
| 02:49.15 | sfbosch | I believe I have my dhcpd configured properly |
| 02:49.17 | fall0ut | in the polycom phone some of the older bootroms require option 66 to be ascii instead of IP |
| 02:49.29 | fall0ut | (by default anyway) |
| 02:49.29 | Damin | You need to tell the polycom to use option 66.. |
| 02:49.42 | Damin | It's in the dhcp options.. |
| 02:49.44 | fall0ut | 66/150 |
| 02:49.46 | shmaltz | sfbosch, you using FTP or TFTP? |
| 02:49.47 | sfbosch | Oh... instead of "Custom + Option 66"? |
| 02:49.48 | sfbosch | TFTP |
| 02:49.57 | Damin | Bah.. use FTP.. |
| 02:50.00 | sfbosch | the phone is brand-spanking new - -it's a 650 |
| 02:50.01 | Damin | TFTP sucks ass.. |
| 02:50.22 | *** join/#asterisk josef_k (n=josef_k@201009196241.user.veloxzone.com.br) |
| 02:50.24 | sfbosch | The phone setting is "Custom + Option 66" |
| 02:50.38 | fall0ut | and you're using FTP, right? |
| 02:50.39 | fall0ut | not TFTP? |
| 02:50.43 | sfbosch | no, I'm using tftp |
| 02:50.49 | fall0ut | if TFTP make sure in server menu you set server type to TFTP |
| 02:51.09 | sfbosch | this is something that has to be set in the console of the phone before you even try to provision it? |
| 02:51.16 | fall0ut | to use TFTP |
| 02:51.16 | fall0ut | yes |
| 02:51.27 | fall0ut | polycoms use FTP by default |
| 02:51.32 | sfbosch | Ah |
| 02:51.40 | sfbosch | what do you prefer? |
| 02:51.49 | josef_k | are there a good soul here to help me? It is basic, too basic. Using asterisk 1.2.9, is possible to install in several different directories setting the constant INSTALL_PREFIX in makefile. HOW I can do that in 1.4.0? |
| 02:52.00 | Damin | FTP is better. |
| 02:52.06 | Damin | It works better through NAT.. |
| 02:52.14 | Damin | WEll.. it works through NAT period. :) |
| 02:52.15 | sfbosch | well, I'm doing all the provisioning internally |
| 02:52.29 | sfbosch | (actually, FTP often doesn't work through NAT) |
| 02:52.32 | Damin | Polycom reccomends that you use FTP instead of TFTP. |
| 02:52.36 | fall0ut | yes |
| 02:52.44 | fall0ut | and the truthfully best method is FTP + HTTPS |
| 02:52.54 | Damin | I'd follow their reccomendation! ;) |
| 02:52.57 | sfbosch | okay -- is that a pain to set up? |
| 02:53.01 | Damin | Nope. |
| 02:53.04 | sfbosch | (Where do they recommend that?) |
| 02:53.15 | Damin | In the Polycom Administrators Manual.. |
| 02:53.45 | techie | it's like reading the kamasutra |
| 02:54.13 | sfbosch | Yeah, no kidding :P |
| 02:54.14 | dual-man | JT: how does it look? |
| 02:54.28 | sfbosch | well, my version of the manual does not make any mention of FTP being recommended over TFTP |
| 02:54.40 | sfbosch | It's the most recent one I could get off the Polycom site |
| 02:55.14 | sfbosch | The document is painful to read |
| 02:55.56 | JT | dual-man: umm, a bit too much unnecessary repitition in zapata.conf, but i think it should work |
| 02:56.54 | dual-man | JT: yeah, i know, i want to clean it up |
| 02:56.59 | *** join/#asterisk cnet2 (n=nada@190.10.0.120) |
| 02:57.21 | *** join/#asterisk LanceSnyder (n=LanceSny@adsl-225-89-245.mia.bellsouth.net) |
| 02:57.52 | JT | dual-man: language, signalling, and context don't need to be repeated |
| 02:58.03 | cnet2 | hi, i have asterisk installed, and the CDR doesn-t seem to enable, how do i do that?.. i have some old calls on /var/log/asterisk/cdr-csv/, but i just made a call and the file wasn-t touched |
| 02:58.05 | dual-man | JT: what confuses me is it worked perfectly with trixbox |
| 02:58.34 | JT | are you sure it's not a hardware problem? |
| 02:58.55 | dual-man | well, i guess that is the next step |
| 02:59.03 | dual-man | troubleshooting |
| 02:59.14 | JT | can you make calls from the phones? |
| 02:59.22 | dual-man | no the line is dead |
| 02:59.39 | JT | do they come up in zap show channels? |
| 02:59.55 | JT | anything of interest in zap show channel <no.> |
| 03:03.56 | dual-man | JT: nothing, everything the same as a working channel. Even show phone as onhook |
| 03:03.58 | sfbosch | fall0ut, Damin: it just keeps saying "could not contact boot server" |
| 03:04.14 | sfbosch | and now I've told it what the server IP is and set the server type to TFTP |
| 03:04.45 | *** part/#asterisk sklim (n=shankuo@60.49.242.225) |
| 03:05.25 | JT | dual-man: what about with the phone off hook, does it still say on hook then? |
| 03:06.17 | [TK]D-Fender | sfbosch: And do you have a proper full set of provisioning files in the tftp folder matching the MAc's for your phone & confirmed that the TFTP server is actually running? |
| 03:07.03 | *** join/#asterisk bigred (n=ian@75-1-209-228.lightspeed.snantx.sbcglobal.net) |
| 03:07.43 | *** join/#asterisk foobar778 (i=johhny@ip68-100-210-15.dc.dc.cox.net) |
| 03:08.50 | LanceSnyder | tomorrow = dCAP test! |
| 03:08.56 | LanceSnyder | anyone here take it? |
| 03:08.59 | foobar778 | [TK]D-Fender: found yjhe hangu[ problempigpen too |
| 03:09.57 | bkruse_home | LanceSnyder: dCaP++ <3 |
| 03:10.03 | cnet2 | anyone got any idea why asterisk could be writing the queue, the message and event, but not the cdr_csv _ |
| 03:10.03 | cnet2 | ? |
| 03:10.09 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 03:10.21 | cnet2 | but was writing it before.. |
| 03:10.34 | Strom_M | LanceSnyder: yeah, i've taken it :) |
| 03:10.43 | LanceSnyder | any tips ? |
| 03:10.52 | LanceSnyder | im studying for it right now |
| 03:11.01 | Strom_M | yeah |
| 03:11.05 | LanceSnyder | this test makes me more nervous than the CISSP |
| 03:11.10 | Strom_M | more people fail the practical than the written test |
| 03:11.13 | Strom_M | so: |
| 03:11.16 | Strom_M | - be calm |
| 03:11.33 | *** join/#asterisk lwh (n=lwh192@rdsl-0270.tor.pathcom.com) |
| 03:11.40 | LanceSnyder | im trying to be haha |
| 03:11.49 | Strom_M | - buy yourself some time by writing your configs while asterisk is still compiling |
| 03:11.57 | LanceSnyder | ahh .. genius |
| 03:12.07 | Strom_M | - read all the directions before you do anything |
| 03:12.48 | orlock | JT: http://www.acma.gov.au/WEB/STANDARD//pc=PC_310020 |
| 03:12.53 | bigred | Has anyone done any mass calling by dumping in .call files? |
| 03:13.40 | LanceSnyder | this test just really has me stressed |
| 03:13.47 | Strom_M | why? |
| 03:14.01 | LanceSnyder | becauase it's alot of stuff to cram in a week and then be tested on |
| 03:14.25 | Strom_M | understanding how everything fits together is more important than understanding little details |
| 03:14.31 | Strom_M | you can google for the little details |
| 03:14.36 | LanceSnyder | yes true |
| 03:14.39 | Strom_M | and "show application" is your friend |
| 03:14.44 | LanceSnyder | can i PM you? |
| 03:14.47 | Strom_M | sure |
| 03:17.41 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:19.54 | JT | Strom_M: do you have the DCAP? |
| 03:20.12 | sfbosch | Polycom friends: I've got it so that the phone finds the TFTP server, loads the MAC.cfg, phoneXXX.cfg and sip.cfg |
| 03:20.21 | sfbosch | but now the phone is stuck at the "Welcome!" screen |
| 03:20.32 | JT | orlock: hmm |
| 03:20.34 | Strom_M | JT: yes, I'm certified |
| 03:20.45 | JT | Strom_M: is it a sit in test, or online? |
| 03:22.05 | Strom_M | administered in person |
| 03:22.18 | JT | ah, so probably not available in Australia |
| 03:24.44 | *** join/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 03:24.56 | *** part/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 03:25.16 | sfbosch | Is there a way to get the sip.cfg file off the phone? |
| 03:26.25 | [TK]D-Fender | sfbosch: Nope. You should flush all local settings and be provisioning from scratch off your server |
| 03:26.43 | sfbosch | Okay -- does this mean "erasing the file system"? |
| 03:27.24 | sfbosch | I ask because: the phone is very new, a Polycom IP 650 |
| 03:27.39 | sfbosch | and, of course, because it is hanging at the Welcome! screen :) |
| 03:28.24 | sfbosch | It has SIP version 2.0.2 on it, but I've obtained 2.1.0 from Polycom; I'm just reluctant to the update unless I have a compelling reason to |
| 03:28.32 | [TK]D-Fender | That means its adapting to a new firmware by flushing everything stored locally on the phone and preparing to install a new firmware |
| 03:28.58 | sfbosch | but I don't have a firmware |
| 03:29.00 | [TK]D-Fender | sfbosch: Read the changelog for all the reasons your could need. |
| 03:29.10 | [TK]D-Fender | sfbosch: then what are you provisioning from? |
| 03:29.33 | sfbosch | well, I'm just provisioning with the sip.cfg file -- no sip.ld; I'm using the one on the phone |
| 03:29.41 | sfbosch | I don't have a 2.0.2 image, and that's what's on the phone |
| 03:29.52 | sfbosch | I didn't want to use 2.1.0 unless I had a compelling reason to do so |
| 03:30.05 | [TK]D-Fender | sfbosch: And which template are you using for your phone setup? |
| 03:30.20 | sfbosch | Well, I pulled the one from freedomphones :\ |
| 03:30.32 | sfbosch | I'm a little suspicious of it now that it's not working |
| 03:31.07 | [TK]D-Fender | sfbosch: If it doesn't match the firmware running on the phone then your phone will hang on boot, crash or perform other wonderful annoyances |
| 03:31.15 | sfbosch | Ah |
| 03:31.16 | sfbosch | Okay |
| 03:31.31 | sfbosch | so basically, if Polycom has provided the 2.1.0 bundle, that's what I should use |
| 03:31.36 | sfbosch | right? |
| 03:31.46 | [TK]D-Fender | sfbosch: Yup. |
| 03:31.55 | sfbosch | Thanks, Fender -- now I have something to work on |
| 03:31.59 | sfbosch | I will report my progress in a bit :) |
| 03:32.04 | [TK]D-Fender | sfbosch: the reason there are higher versions is because lower versions DID have problems |
| 03:32.19 | sfbosch | That makes sense |
| 03:32.26 | sfbosch | but sometimes you get upgradeitis, too |
| 03:32.40 | sfbosch | and some people have reported problems with newer firmwares that didn't exist on the older ones |
| 03:32.50 | sfbosch | but hey, it's hard to make a broken setup more broken, so I might as well try this |
| 03:34.04 | [TK]D-Fender | sfbosch: And the reverse is normal. |
| 03:37.02 | *** join/#asterisk codazoda (n=chatzill@63.110.13.126) |
| 03:38.46 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 03:40.19 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:41.40 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 03:41.52 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-49.intrastar.net) |
| 04:00.56 | JacksLivr | hello all, i have a question. in the following line, what does the :0:3 mean? |
| 04:00.56 | JacksLivr | exten => s,4,GotoIf($["${CALLERIDNUM:0:3}" = "877"]?s|20) |
| 04:00.58 | JacksLivr | thanks |
| 04:02.21 | Strom_M | hmmm |
| 04:02.32 | Strom_M | I wonder if there's some trick to getting the tc400b working under 1.2 |
| 04:03.26 | JacksLivr | im guessing the 3 means to grap 3 digits, but i cant find docs on nit anywhere |
| 04:03.39 | JacksLivr | since its matching 877 |
| 04:03.53 | Strom_M | JacksLivr: it means "start at position 0 and read out the next three digits" |
| 04:04.11 | JacksLivr | AHHHHHHHHHHH |
| 04:04.24 | JacksLivr | Strom_M: thanks man |
| 04:05.38 | [TK]D-Fender | JacksLivr: lookup "asterisk variables" on the WIKI |
| 04:05.41 | [TK]D-Fender | ~wikis |
| 04:05.48 | jbot | well, wikis is http://www.voip-info.org |
| 04:06.16 | JacksLivr | yeah, i looked this up on voip-info |
| 04:06.30 | [TK]D-Fender | JacksLivr: then it should be no mystery to you. |
| 04:06.46 | dual-man | JT: problem solved |
| 04:07.26 | JacksLivr | i may be stupid, but i tried to find it |
| 04:07.37 | JacksLivr | i waited about 30 minutes before i asked it |
| 04:07.42 | JacksLivr | sorry |
| 04:07.45 | Strom_M | [TK]D-Fender: go easy on him man; sometimes the wiki isn't helpful |
| 04:08.07 | [TK]D-Fender | JacksLivr: Should have LOOKED for 30 mins on that page.... far more productive than just sitting around waiting :) |
| 04:08.11 | JT | dual-man: yeah? |
| 04:08.22 | [TK]D-Fender | Strom_M: THAT page is, and he says he's been there :) |
| 04:08.32 | [TK]D-Fender | Strom_M: GUILTY!!! |
| 04:08.42 | Strom_M | *shrug* took me all of ten seconds to answer his question instead of admonishing him |
| 04:08.59 | JacksLivr | [TK]D-Fender: please link to the page. |
| 04:09.12 | JacksLivr | fwiw, i did not sit on my hands for 30 minutes |
| 04:09.22 | JacksLivr | i was googling and what not |
| 04:09.36 | [TK]D-Fender | JacksLivr: Of course not.... that might force amputation of the now choked off limb! |
| 04:10.05 | JacksLivr | i try not to ask the n00b questions, I'm not asking you to describe SIP |
| 04:10.13 | [TK]D-Fender | http://www.voip-info.org/wiki-Asterisk+variables |
| 04:10.16 | Strom_M | JacksLivr: try googling "asterisk variables site:voip-info.org" |
| 04:10.42 | [TK]D-Fender | Strom_M: there is a search link right on the WIKI, no need to look further |
| 04:10.59 | Strom_M | *shrug* just trying to help him construct better google queries |
| 04:11.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:11.26 | Strom_M | i figure it's better to teach rather than admonish, but hey, if you think he's going to learn more from you screaming at him, then who am I to say otherwise? |
| 04:12.54 | JacksLivr | search the logs. I have the system completely set up with very very very few quesitons to the channel |
| 04:12.54 | [TK]D-Fender | Strom_M: Np, and I wasn't screaming, you should have gotten the "jest" value from my smilies.... |
| 04:13.04 | JacksLivr | i have the dial plans forking based on callerid and VM conf bridges and everything. |
| 04:13.13 | [TK]D-Fender | Strom_M: And you know I only really stomp on the trolls who inevitably get kicked out anyways.... |
| 04:13.24 | JT | dual-man: how? |
| 04:13.40 | Strom_M | eh, sometimes it comes across a little too harshly, IMHO |
| 04:13.58 | [TK]D-Fender | JacksLivr: using the Search option on the sidebar, lokup "asterisk functions" while you're at it. CallerID is now a FUNCTION, and no longer a variable (1.2+) |
| 04:14.10 | Bobthehunter | lame that our societwaht is it with asterisk that all of a sudden it STOP S sending the voicemail emails |
| 04:14.18 | Bobthehunter | i see nothing in full.log |
| 04:14.22 | [TK]D-Fender | Strom_M: Yeah, I can be read that way sometimes, but hopefully you know me better :) |
| 04:14.30 | [TK]D-Fender | Strom_M: Or will learn to.... |
| 04:15.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:15.28 | Strom_M | yeah, i know you well enough to know that; i'm just looking at it through the lens of someone who doesn't know you terribly well |
| 04:15.42 | [TK]D-Fender | Strom_M: I'll try to use those eye more often.... |
| 04:16.01 | Strom_M | you may see it as joking around, but for someone new here, it can come across as pure admonishment |
| 04:16.44 | [TK]D-Fender | Strom_M: Ok, topic is well said and done. New page. |
| 04:16.50 | Strom_M | *nod* |
| 04:16.54 | [TK]D-Fender | *nod* |
| 04:17.05 | Strom_M | any idea how to get this TC400B card going? documentation is a bit on the slim side :) |
| 04:17.17 | [TK]D-Fender | Strom_M: Not sure about 1.2..... |
| 04:17.37 | [TK]D-Fender | strom I know you need the LATEST zaptel for sure. |
| 04:17.46 | Strom_M | yeah, i'm running latest 1.2 zaptel |
| 04:17.49 | [TK]D-Fender | Strom_M: (stable at least) |
| 04:17.51 | Bobthehunter | anyone ? |
| 04:18.02 | Bobthehunter | i had that problem for years now |
| 04:18.02 | [TK]D-Fender | Strom_M: Not sure if 1.2 works with it yet... |
| 04:18.12 | Strom_M | ah, ok |
| 04:18.30 | [TK]D-Fender | Bobthehunter: Have you checked you mail daemon? |
| 04:18.40 | Strom_M | yeah, the module compiles and loads, but asterisk says "no zaptel transcoder support!" |
| 04:19.01 | [TK]D-Fender | hrm |
| 04:19.02 | Bobthehunter | its not hitting sendmail for whatever reasonim tailing var/log/maillog |
| 04:19.19 | Bobthehunter | Locked path '/var/spool/asterisk/voicemail/default/2/INBOX' |
| 04:19.24 | Bobthehunter | Unlocked path '/var/spool/asterisk/voicemail/default/2/INBOX |
| 04:19.35 | Bobthehunter | then pbx functionr euslts.. then cdr insert |
| 04:19.42 | Bobthehunter | nthing even talks about vm |
| 04:19.54 | [TK]D-Fender | Bobthehunter: Do a sanity check on your voicemail.conf |
| 04:20.02 | Bobthehunter | its realtime |
| 04:20.09 | [TK]D-Fender | Bobthehunter: And then try and runn sendmail as * would completely by hand. |
| 04:20.14 | [TK]D-Fender | Bobthehunter: :/ |
| 04:20.26 | [TK]D-Fender | Bobthehunter: No idea... but check what you can on that |
| 04:20.34 | Bobthehunter | the thing is in FULL.log it doesnt even SHOW anything about that |
| 04:20.39 | JT | Strom_M: you actually bought a TC400B? |
| 04:20.43 | Bobthehunter | its not wanting to call sendmail at all it seems |
| 04:23.45 | Strom_M | JT: no, it's on loan from the digium :) |
| 04:23.59 | JT | ah ok |
| 04:24.05 | JT | it seems pricey |
| 04:24.15 | JT | might be useful if you must transcode g.723 |
| 04:24.25 | Bobthehunter | nothing |
| 04:27.04 | Strom_M | ah, well, off to dinner |
| 04:27.13 | Strom_M | i'll diddle with this later |
| 04:28.03 | Bobthehunter | this is ridiculous.. |
| 04:28.13 | Bobthehunter | can someone tail -f full |
| 04:28.22 | Bobthehunter | and let me know if you see mention of it sending mail |
| 04:31.09 | *** join/#asterisk MonkeyHugs (n=jojo@63.149.122.94) |
| 04:31.50 | Bobthehunter | hmm i dont even see any of the LOG_MESSAGESfdrom app_voicemail.c |
| 04:31.57 | Bobthehunter | so.. its not calling sendmail |
| 04:33.48 | MonkeyHugs | I am attempting to install Zaptel 1.2.14 on a clean install of RHEL 3 running kernel 2.4.21. I am receiving an error during the make process. http://pastebin.ca/377959 Any help would be appreciated. |
| 04:34.24 | *** join/#asterisk antlers (n=antlers@ip68-224-230-141.lv.lv.cox.net) |
| 04:39.26 | [TK]D-Fender | MonkeyHugs: It looks like you don't have the mathcing headers for your kernel |
| 04:41.32 | *** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
| 04:41.42 | MonkeyHugs | Thank you TK |
| 04:41.45 | BrianR___ | I'm trying to troubleshoot a T1 problem.. I'm getting severe audio distortion.... |
| 04:41.46 | Bobthehunter | i fixed fvoicemail |
| 04:41.57 | Bobthehunter | fromstring =mypbx.. |
| 04:42.00 | BrianR___ | Does anyone know if the audio portion of a PRI is covered with HDLC? |
| 04:42.05 | Bobthehunter | but how cn i force domain its sending as |
| 04:42.11 | BrianR___ | Ie, if there's corruption can I see link statistics anywhere, or does it just sound bad? |
| 04:42.17 | Bobthehunter | else i get the systemcrap |
| 04:42.18 | JT | BrianR___: how does zttest fare? |
| 04:43.11 | BrianR___ | JT: 100, 99.98, etc. What does that mean? |
| 04:43.37 | BrianR___ | Isn't that thing just a timing test? |
| 04:44.02 | JT | BrianR___: sounds pretty decent, as long as it never goes under 99.97%, test this especially during audio problems |
| 04:44.09 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:44.13 | JT | it checks zaptel timing |
| 04:44.23 | JT | poor zap timing causes distortion and d channel errors |
| 04:44.37 | BrianR___ | Aah.. I'm not seeing any d channel errors though.. . |
| 04:45.54 | BrianR___ | I have a few T1's on this card.. It's a sangoma a104D. I can place test calls on a few of them using another asterisk box with T1/PRI and they're fine. But calls which go over my PRI to the PSTN have very bad audio quality... Sounnds like at least the TX audio is getting butchered. Has almost a robotic sound. |
| 04:46.12 | JT | well if it doesn't go below 99.97 it's fine |
| 04:46.17 | BrianR___ | I'm wondering if the wiring might be suspect or something. |
| 04:46.26 | JT | that's weird |
| 04:46.29 | JT | unlikely |
| 04:46.34 | JT | if no d channel errors |
| 04:46.37 | BrianR___ | Are the b channels covered with HDLC, or just the D channel? |
| 04:47.01 | BrianR___ | The A104D does hardware hdlc though - I wonder if it is hiding the errors from me somewhere... |
| 04:47.02 | [TK]D-Fender | BrianR___: You need to make sure that the EC is not affecting the dchan |
| 04:47.47 | BrianR___ | TK]D-Fender: How would that explain my poor audio quality? |
| 04:47.48 | [TK]D-Fender | BrianR___: And you may want to down your server, kill wanpipe, then make sure the EC is flushed. Sometimes it can get stuck and lose audio, etc. |
| 04:48.13 | BrianR___ | TK]D-Fender: Egadz! |
| 04:48.29 | BrianR___ | I rebooted the server, but I anm not near it to power cycle... |
| 04:48.40 | BrianR___ | Stupid ipmi is messed up on it. |
| 04:49.42 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 04:49.52 | BrianR___ | this problem is going to be a bummer to troubleshoot, since this is my only PRI circuit into the building and I can't mes with it during business hours. |
| 04:50.57 | *** join/#asterisk shodan- (n=shodan@ip204.96-113-216.pppoe1.joliette.intermonde.net) |
| 04:51.22 | Qwell | speaking of cookies...I have...girl scout cookies |
| 04:51.22 | file | Qwell! |
| 04:51.38 | file | you robbed them didn't you? |
| 04:51.57 | Qwell | kinda |
| 04:52.33 | JunK-Y | can the AADK really deal upto 50 users, with full rtp, like ivr or meetme? |
| 04:53.08 | Qwell | JunK-Y: I think they mean 50 seats, not 50 simultaneous.. don't quote me on that though |
| 04:53.47 | JunK-Y | if you could ask someone in that dept, id like to know how many sim with full rtp. |
| 04:53.58 | JunK-Y | website doesnt specify it. |
| 04:59.09 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 04:59.19 | DocHolliday | does anyone have the locale files for the Cisco 7941? |
| 05:02.42 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 05:02.55 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-49.intrastar.net) |
| 05:05.55 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 05:06.45 | dual-man | how do i shut down an application in debian |
| 05:07.02 | JT | dual-man: so what did you do to fix it? |
| 05:07.05 | JT | dual-man: kill? |
| 05:07.29 | [[blah]asfd | ok, so i am trying to help get asterisk installed on Linux version 2.4.21-47.0.1.EL (brewbuilder@hs20-bc2-2.build.redhat.com) (gcc version 3.2.3 20030502 (Red Hat Linux 3.2.3-56)) |
| 05:07.42 | [[blah]asfd | i am getting errors and looking for help... pasting the errors.... |
| 05:07.57 | JT | ~pb |
| 05:07.58 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 05:08.04 | dual-man | JT: the punch block had a loose connection |
| 05:08.13 | [[blah]asfd | http://pastebin.ca/377986 |
| 05:08.14 | JT | dual-man: told you it was physical :P |
| 05:08.19 | dual-man | JT: how do i use kill to stop an app? |
| 05:08.24 | [[blah]asfd | i am actually am trying to install zaptel. |
| 05:08.31 | JT | dual-man: kill <pid> |
| 05:08.40 | [[blah]asfd | i have installed the kernel src.. I think... |
| 05:09.02 | dual-man | JT: how do i find the pid, i only have the app name |
| 05:09.04 | [[blah]asfd | i have tried installing kernel-devel. up2date doesnt find it |
| 05:09.15 | JT | dual-man: ps aux |
| 05:09.18 | JT | or top |
| 05:09.19 | JT | etc etc |
| 05:09.36 | [[blah]asfd | ps aux | grep app name |
| 05:10.12 | dual-man | thanks |
| 05:11.48 | dual-man | kill pid is not stopping it |
| 05:11.54 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 05:12.26 | [[blah]asfd | kill -9 |
| 05:12.49 | dual-man | what does the -9 stand for? |
| 05:12.57 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:12.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:13.11 | J4k3 | 9 millimeter |
| 05:13.21 | JT | dual-man: signal 9 |
| 05:13.28 | [[blah]asfd | it means kill it |
| 05:13.29 | *** part/#asterisk timphnode (n=tim@adsl-68-91-95-148.dsl.ksc2mo.swbell.net) |
| 05:13.40 | [[blah]asfd | so do a kill -9 pid |
| 05:13.42 | dual-man | thanks :D |
| 05:14.02 | JT | signal 9 is SIGKILL |
| 05:14.12 | JT | default of kill is to send SIGTERM |
| 05:14.15 | Qwell | I renamed it SIGPWN |
| 05:14.18 | JT | which asks the program to terminate |
| 05:14.22 | russellb | Qwell: nice. |
| 05:14.36 | JunK-Y | SIGQWELL |
| 05:14.49 | [[blah]asfd | anyone able to help me get zaptel installed on kernel 2.4? |
| 05:14.53 | [[blah]asfd | RHEL |
| 05:14.56 | [[blah]asfd | version 3 |
| 05:15.02 | [[blah]asfd | I am super stumped |
| 05:15.08 | russellb | it should work ... |
| 05:15.21 | [[blah]asfd | here is the output of my make http://pastebin.ca/377986 |
| 05:15.28 | [[blah]asfd | well a few lines of it at least |
| 05:15.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:16.39 | [[blah]asfd | i am not sure i have the headers and source installed correctly... |
| 05:16.42 | [[blah]asfd | not used to up2date. |
| 05:17.02 | [[blah]asfd | i tried installing kernel-devel. it doesnt seem to exist for 2.4 |
| 05:17.24 | Qwell | no, it doesn't exist for RHEL3 :) |
| 05:17.31 | [[blah]asfd | that answers that question... |
| 05:17.43 | Qwell | kernel-sources |
| 05:17.49 | Qwell | that's as good as you'll get |
| 05:18.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:18.58 | [[blah]asfd | Qwell: The following packages you requested were not found: |
| 05:18.58 | [[blah]asfd | kernel-sources |
| 05:19.04 | k-man_ | [[blah]asfd, why not use a more recent kernel? |
| 05:19.06 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 05:19.13 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:19.46 | [hC] | what do you guys use to convert wav to g729 audio? |
| 05:20.14 | [hC] | I tried gx::transcoder, and it seemed to work, except all my g729 files spit out an error at the end of playing and to the caller it sounds like a loud pop/screech at the end of the audio. |
| 05:20.14 | [[blah]asfd | trying to install a brooktout card for hylafax. they have some limitations with their driver for the card. has to be RHEL3 for the version of the card we bought. |
| 05:23.47 | Qwell | [hC]: If you have g729 licenses, you can use the convert command from the asterisk CLI |
| 05:23.58 | [[blah]asfd | what do these errors mean? http://pastebin.ca/377986 |
| 05:25.13 | russellb | [hC]: if you have codec_g729, asterisk can do it |
| 05:25.46 | russellb | sudo asterisk -rx "convert /path/to/mysound.wav /path/to/mysound.g729" |
| 05:28.31 | [hC] | Qwell: ........ seriously.... Ive been fighting with this this whole time and asterisk has a built in converter? |
| 05:28.42 | [hC] | If i didnt just get a haircut, i'd pull my hair out. |
| 05:29.41 | [hC] | vg1*CLI> convert |
| 05:29.41 | [hC] | No such command 'convert' (type 'help' for help) |
| 05:29.42 | [hC] | Hmm. |
| 05:29.46 | [hC] | 1.2.9.1 |
| 05:30.07 | russellb | yeah, just 1.4 |
| 05:30.21 | russellb | sorry |
| 05:30.29 | [hC] | Ahhh... Hmm. How did people do it pre 1.4? |
| 05:30.38 | Qwell | offtopic, any of you folks know what a vmware team is? |
| 05:30.47 | ManxPower | [hC]: I suspect they purchased a commercial audio software |
| 05:30.52 | russellb | [hC]: don't know of a way |
| 05:31.17 | Qwell | [hC]: use module embedding, and you won't even need to do a make install |
| 05:31.37 | [hC] | It might even be worth installing a copy of 1.4 just to convert these files. Transcoding to wav on this soekris net4801 is killing it. |
| 05:31.51 | [hC] | er, from wav. |
| 05:32.00 | J4k3 | duh |
| 05:32.06 | J4k3 | its a 486 :P |
| 05:32.14 | russellb | Qwell: good idea |
| 05:32.17 | [hC] | Qwell: I have yet to explore 1.4, dont know what module embedding is. I'll go ask google. |
| 05:32.40 | [hC] | J4k3: when it doesnt transcode it does a great job for ~10-50 phones |
| 05:32.41 | russellb | same idea as in the kernel |
| 05:32.54 | russellb | instead of loadable modules, you have them embedded into the main binary |
| 05:32.55 | [hC] | oh, just compile the module from 1.4 and do a load modname.so ? |
| 05:33.02 | russellb | no, that won't work |
| 05:33.07 | [hC] | I didnt think so. |
| 05:33.10 | J4k3 | [hC]: well yeah. one can route many thousand PPS on a 486 no problem |
| 05:33.28 | J4k3 | its just the performance goes to poop when you make the CPU actually think ;) |
| 05:33.54 | [hC] | I'll just install 1.4 in vmware or something and do it in there. |
| 05:35.04 | *** join/#asterisk Maveric (n=maveric@ip68-99-138-154.ph.ph.cox.net) |
| 05:46.08 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:56.34 | *** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
| 05:59.26 | Zipper_32 | Are there any hardware devices that can be used with asterisk to view lines which are currently in use? I've been looking at the Polycom 601 Expansion module; is anybody aware if this does the trick or not? |
| 06:00.02 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 06:01.36 | [TK]D-Fender | Zipper_32: Yes it would |
| 06:02.44 | Zipper_32 | So, hypothetically, if I had a receptionist using that phone, s/he could delegate where a call would go based on who is not on the phone in an office? |
| 06:03.27 | [TK]D-Fender | yes |
| 06:04.01 | Zipper_32 | Thank you, =) |
| 06:05.10 | *** join/#asterisk antlers (n=antlers@ip68-224-230-141.lv.lv.cox.net) |
| 06:05.39 | *** join/#asterisk jaike (n=jaike@125.5.144.90) |
| 06:13.25 | *** join/#asterisk nn (n=joseph@cpe-72-229-104-43.nyc.res.rr.com) |
| 06:13.33 | nn | anyone using callwithus.com for IAX? |
| 06:15.31 | *** join/#asterisk antlers (n=antlers@ip68-224-230-141.lv.lv.cox.net) |
| 06:16.15 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 06:24.44 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:26.41 | *** join/#asterisk CrazyTux (n=CrazyTux@216.142.88.85) |
| 06:27.45 | *** join/#asterisk LanceSnyder (n=LanceSny@adsl-225-89-245.mia.bellsouth.net) |
| 06:28.16 | LanceSnyder | reinvites do what? |
| 06:28.56 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 06:30.02 | eald_home | if yes RTP pass peer to peer, if no pass peer-asterisk-peer |
| 06:30.09 | J4k3 | tells your sip devices to speak to each other directly instead of 'thru' the asterisk box whenever possible. Disabling it causes the traffic to always relay off the * box |
| 06:30.12 | J4k3 | thats my understanding at least |
| 06:30.30 | LanceSnyder | thought so |
| 06:30.46 | LanceSnyder | studying for dCAP, brain is running low |
| 06:30.52 | LanceSnyder | take it in 11 hour |
| 06:30.53 | LanceSnyder | s |
| 06:31.02 | J4k3 | tests are always fun |
| 06:31.15 | LanceSnyder | ehh i dont feel so good about this one |
| 06:31.26 | JT | stressball :P |
| 06:31.28 | *** join/#asterisk nighty^ (n=nighty@211.154.202.98) |
| 06:31.36 | LanceSnyder | hah yeah |
| 06:31.47 | nn | how should i format the IAX2/ "uri" for forwarding? so that it comes in to my machine (1.2.3.4) in context 'main' at extension 's'? |
| 06:33.23 | nn | iax confuses me ;) |
| 06:35.04 | *** join/#asterisk jaike` (n=jaike@203.115.188.120) |
| 06:35.49 | jaike` | can anyone suggest a good canadian sip/iax provider? |
| 06:37.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:42.25 | jql | you checked voip-info, right? |
| 06:42.29 | antlers | does anyone have a working broadvoice configuration they wouldn't mind sharing (scrubbed of course)? thanks! |
| 06:42.35 | jql | no recommendations there, but a list is better than nothing |
| 06:43.11 | nn | IAX2/hostname/extension? |
| 06:43.17 | nn | err |
| 06:43.20 | nn | IAX2/hostname/context ? |
| 06:43.46 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 06:48.59 | kaldemar | nn: depends on if you have defined the peers in iax.conf. but IAX2/username:secret@address/extension@context works always. see the wiki for more information. |
| 06:50.55 | nn | Mar 2 01:50:40 NOTICE[27907]: chan_iax2.c:6910 socket_read: Rejected connect attempt from 66.219.59.39, who was trying to reach '2345678901' |
| 06:52.07 | nn | rather 2345678901@ |
| 06:52.47 | nn | should work with a peer as type=friend, no? |
| 06:53.04 | kaldemar | friend is both peer and user, so yes. |
| 06:53.56 | nn | i'm using IAX2/userid:pass@host/2345678901 as the destination for the DID |
| 06:54.11 | nn | any way to see more detail why it was rejected? |
| 06:54.49 | kaldemar | the other end rejects it because it doesn't have a matching extension in the context where the call ends up in. |
| 06:54.58 | kaldemar | most likely. |
| 06:55.35 | nn | kaldemar: the other end is me :) |
| 06:55.51 | nn | been about 2.5 years since i touched * |
| 06:56.24 | kaldemar | so you check the context parameter in your iax.conf and look at the context in extensions.conf. |
| 06:56.53 | *** join/#asterisk kronic (n=blowfish@28.64.87.203.static.vic.chariot.net.au) |
| 06:57.19 | kronic | what method would someone recommend for receiving a voice stream and producing text from it? |
| 06:57.21 | nn | yea, i have context=in-cwu, and a context [in-cwu] in extensions.conf with that |
| 06:57.46 | kronic | Eg reverse of swift/festival |
| 06:57.58 | kaldemar | with exten => 2345678901... ? |
| 06:58.01 | nn | yea |
| 06:58.17 | nn | exten => 15167340898,1,Answer |
| 06:58.27 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 06:58.33 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 06:59.18 | kaldemar | nn: can you pastebin your iax.conf, the dial string and [in-cwu] from extensions.conf? |
| 06:59.41 | nn | kaldemar: yea it'll take a moment to cleanse them |
| 07:00.22 | *** join/#asterisk aadilismail (i=zeeesh@202.38.55.125) |
| 07:00.23 | aadilismail | hi |
| 07:00.29 | nn | kaldemar: for say SetCallerID, how do i append the original callerid? (such as fwd # originating in-bound call) |
| 07:00.50 | JT | setcallerid is deprecated |
| 07:00.54 | nn | kaldemar: im exploiting my telco giving me a "favourite 5 #'s) |
| 07:01.00 | nn | JT: how so? |
| 07:01.21 | JT | it is no longer the correct way to set callerid since 1.2, removed as of 1.4 |
| 07:01.42 | kaldemar | function CALLERID replaces the application. |
| 07:02.48 | nn | http://pastebin.ca/378069 |
| 07:04.04 | kaldemar | nn: in your dial string, replace myuser with cwu and youll be fine. |
| 07:04.47 | nn | aha |
| 07:04.50 | kaldemar | the username parameter is for outbound calls. |
| 07:05.01 | nn | yea, i was just looking at it and wondering if that was it :) |
| 07:05.45 | nn | woot, got the lovely "Weasels have eaten our phone system" |
| 07:05.57 | Strom_M | those damn weasels |
| 07:06.01 | kaldemar | you could call the cwy box with just IAX2/cwu/exten if the secrets are the same. |
| 07:06.27 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 07:10.43 | nn | now.. to set up outbound calling with a pin ;) |
| 07:12.17 | kaldemar | 'show application authenticate' in asterisk's cli will help. |
| 07:12.37 | nn | having 300 min on my mobile then setting up my * box's DID as calling card (at 0.008$/min incoming after my 3000 included) then calling out over my unlimited outbound == much cheaper |
| 07:12.57 | nn | since i can pick 3 #'s that act as unlimited mobile2mobile #'s |
| 07:13.18 | JT | heh, nice |
| 07:13.28 | JT | DISA is probably what you want |
| 07:13.36 | nn | i dont need anything complex |
| 07:13.42 | JT | if it's cheaper, you can even make your asterisk box ring you back when you call it |
| 07:13.45 | JT | DISA is simple |
| 07:13.54 | nn | JT: i want to encode the pin in the dial string |
| 07:14.12 | JT | why? |
| 07:14.40 | nn | JT: so i can store it in the phone |
| 07:14.54 | JT | disa might still work |
| 07:14.56 | nn | i can set a calling card and then just match all calls |
| 07:15.15 | nn | authenticate seems to have been what i needed |
| 07:15.38 | nn | tho disa might work too |
| 07:18.52 | *** join/#asterisk lorinc (n=ang@pool-3640.adsl.interware.hu) |
| 07:19.26 | JT | here i can just authenticate off callerid |
| 07:19.37 | JT | you can't do that in the US, as it's easily forged |
| 07:19.43 | JT | well callback would work fine |
| 07:19.47 | JT | as it'd only call you anyway |
| 07:32.36 | nn | be back tomorrow from another state |
| 07:35.50 | nn | ahh maybe i should use Authenticate at the start of the call, then use DISA w/o password when '9' is dialed |
| 07:36.18 | JT | i much prefer callerid auth |
| 07:37.07 | nn | thats a bad idea here :) |
| 07:37.13 | nn | i prefer pin |
| 07:37.23 | nn | since phones call the ability to exchange x.509 certs |
| 07:37.24 | JT | although a reckon a better idea than authenticate at the start is for it to match the number you want to dial, THEN ask for pin code |
| 07:37.47 | JT | what's this about certs? |
| 07:38.09 | JT | callerid auth is ok if you use callback |
| 07:39.03 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 07:43.11 | nn | JT: i got it working good enough for now :) |
| 07:43.12 | nn | bbl |
| 07:43.15 | nn | thanks :) |
| 07:44.14 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:49.51 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:54.10 | gfraysse | <PROTECTED> |
| 07:54.35 | gfraysse | <PROTECTED> |
| 07:55.13 | gfraysse | <PROTECTED> |
| 07:55.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:56.22 | *** join/#asterisk yonahw (i=yonahw@84.228.13.75) |
| 08:00.38 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 08:06.03 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:06.23 | *** join/#asterisk J4k3 (i=J4k3@12.197.130.219) |
| 08:07.29 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 08:09.31 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 08:11.22 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 08:12.24 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 08:16.00 | *** join/#asterisk mjun007 (n=mjun007@221.221.145.233) |
| 08:17.45 | *** join/#asterisk mellen (n=tmh@85.235.250.110) |
| 08:19.11 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 08:23.09 | *** join/#asterisk mjun007 (n=mjun007@221.221.145.233) |
| 08:23.27 | *** join/#asterisk Strom_M (n=strom@209.19.56.4) |
| 08:28.48 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 08:30.54 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:44.42 | *** join/#asterisk SoftIce (i=awk@208.98.52.125) |
| 08:45.24 | SoftIce | hi, please can soebody sugest some sort of systemI can use. this is my problem, I come in the morning and Ive lost 5000 minutes because ther was no routing for 12 hours, what type of system is there available to mail me an allert to let me know minutes are not getting routed? |
| 08:45.50 | JT | minutes are not getting routed |
| 08:45.53 | JT | calls? |
| 08:45.57 | SoftIce | yes |
| 08:46.08 | JT | right, calls |
| 08:46.22 | JT | umm you can get it to email you everytime a call fails, but that would be annoying |
| 08:46.33 | JT | would take a little scripting work to make something nicer happen |
| 08:46.34 | SoftIce | iesh, that would be scary :) |
| 08:46.51 | SoftIce | structured how ? |
| 08:48.40 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 08:56.24 | *** join/#asterisk skirmisha (i=507e4918@gateway/web/cgi-irc/ircatwork.com/x-5bd228b56862e5ac) |
| 08:57.01 | Strom_M | SoftIce: look into something called nagios |
| 08:57.03 | skirmisha | hello |
| 08:58.44 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 09:00.37 | SoftIce | thanks |
| 09:01.00 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 09:05.39 | SoftIce | hmm, nagios would be difficult in this sitation |
| 09:06.05 | SoftIce | because if I get a congestion tone or something else |
| 09:06.20 | SoftIce | it wouldn't be able to mail me stating routing to this country isn't working |
| 09:06.57 | fetcher | what's an easy way to detect from an AGI script whether the channel has hung up ? |
| 09:08.06 | skirmisha | does anyone know how can i catch in chan_sip user unregistrations |
| 09:08.59 | *** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 09:09.44 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 09:10.46 | JT | awk: why couldn't you get it to work? |
| 09:10.59 | awk | why couldn't I get what to work? |
| 09:11.17 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 09:11.26 | JT | nagios |
| 09:11.44 | awk | well it isn't ideal for what i'm trying to do |
| 09:11.51 | J4k3 | yay, looks like ram prices went doooooown |
| 09:12.02 | awk | I would need something that has a trigger allert reading cdr or something |
| 09:12.11 | awk | but set on a scale that it isn't every call, etc |
| 09:12.41 | JT | err i'm pretty sure you can set thresholds |
| 09:12.48 | skirmisha | hello |
| 09:12.52 | *** join/#asterisk susinths (n=susinths@ifi-8021x-dhcp276.uio.no) |
| 09:13.27 | awk | with nagios ? |
| 09:13.35 | JT | yes |
| 09:13.40 | JT | it's very comprehensive |
| 09:15.36 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:21.53 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 09:27.16 | skirmisha | does anyone know how can i catch in chan_sip user unregistrations |
| 09:30.27 | *** join/#asterisk mborohov (n=mborohov@BORSKI.MIT.EDU) |
| 09:31.10 | mborohov | hey guys, i have a question. im running asterisk with around 50 users or so, and would like to be able to get rates for voicepulse and track usage for each user. any ideas? |
| 09:32.22 | *** join/#asterisk vlt (n=dm@p54B3350C.dip0.t-ipconnect.de) |
| 09:32.48 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:34.53 | J4k3 | sql is your friend in this case |
| 09:37.26 | vlt | Hello, can I persuade the Queue() application to try to call a member on a Zap (BRI) channel even if it _thinks_ it is in use? The Zap member here can transfer a call to another channel of the legacy ISDN PBX it is connected to ... |
| 09:39.16 | vlt | s/in use/busy/ |
| 09:41.07 | *** join/#asterisk susinths (n=susinths@ifi-8021x-dhcp096.uio.no) |
| 09:41.33 | creativx | haha |
| 09:41.33 | creativx | smartass bot innit |
| 09:42.50 | jm|work | and no need to escape white space :O |
| 09:43.00 | jm|work | s/white\ space/special\ chars/ |
| 09:48.24 | *** join/#asterisk FuraX (n=christop@umb-sls99-003.u-strasbg.fr) |
| 09:52.14 | nfi|ermes | to well bt che core dump is better to compile asterisk with or withouth DONT OPTIMIZE option ?? |
| 09:53.47 | nfi|ermes | i answer to myself: with !! |
| 09:54.47 | vlt | ... after transferring a call (e. g. on Zap/1) from queue to another IDSN PBX channel it could answer calls on Zap/2 while Zap/1 is still used for the first call. The member's name is "Zap/g1/555" and "g1" refers to Zap/1 and Zap/2 ... Any idea? |
| 09:55.13 | JT | vlt: really not sure what you are asking |
| 09:56.00 | vlt | JT: hmmm ... seems I'm unable to write clear sentences these days ... |
| 09:58.20 | vlt | JT: * is connected via "4xS0 card" (thanks for help) to a legacy ISDN pbx on the channels Zap/1 and Zap/2. This is one BRI port defined as "Zap/g1". |
| 09:58.31 | *** join/#asterisk dongc (n=dongc@203.117.206.249) |
| 09:58.42 | vlt | JT: Got it 'til now? |
| 09:58.46 | JT | yep |
| 09:59.04 | vlt | JT: Zap/g1/555 is a member of a queue. |
| 09:59.33 | vlt | JT: The queue is filled from other BRI ports and IAX2 channels. |
| 10:00.32 | vlt | JT: When a call is delivered to Zap/g1/555 and answered, the channel Zap/1-1 for ex. is in use. |
| 10:00.54 | vlt | JT: Zap/2 is still free. Ok? |
| 10:01.22 | JT | 555 is a queue on the LEGACY PBX? |
| 10:02.04 | vlt | JT: 555 is an extension on the legacy pbx. The only queue is on * |
| 10:02.40 | JT | i see |
| 10:02.57 | vlt | JT: When 555 transfers the call _internally_ (on the old pbx) completely bypassing asterisk ... |
| 10:03.05 | *** part/#asterisk dongc (n=dongc@203.117.206.249) |
| 10:03.07 | JT | ok |
| 10:03.20 | vlt | JT: .. it is available again. But * doesn't know. |
| 10:03.51 | JT | why doesn it know? |
| 10:04.07 | vlt | JT: To me it seems so. |
| 10:04.46 | vlt | JT: I _think_ asterisks knows that Zap/1-1 is in use and so Zap/g1/555 is busy. |
| 10:05.00 | vlt | JT: Am I wrong here? |
| 10:05.05 | JT | it would |
| 10:05.32 | vlt | what would what? |
| 10:05.33 | vlt | :) |
| 10:05.43 | JT | it would know the channel is in use |
| 10:06.17 | vlt | JT: Yes, but 555 could answer calls on Zap/2 ... |
| 10:06.25 | *** join/#asterisk f_akmal (n=f_akmal@203-214-99-27.dyn.iinet.net.au) |
| 10:06.25 | JT | ok |
| 10:06.32 | vlt | JT: ... if asterisk only tried to call it ... |
| 10:06.46 | JT | pastebin.ca zaptel.conf and zapata.conf |
| 10:07.48 | f_akmal | hey everyone |
| 10:07.59 | f_akmal | i need someone to help test out my system |
| 10:08.47 | vlt | JT: I can do that ... but if you're not sure that asterisk doesn't try I will do further checks here before consuming more of your time -- maybe I'll need it later ;-) |
| 10:09.07 | JT | asterisk should be |
| 10:09.25 | JT | it's probably a configuration issue |
| 10:12.16 | vlt | JT: http://rafb.net/p/vNFwY983.html |
| 10:13.46 | vlt | JT: And yes, I checked p2mp in the PBX (group 1) and telco (groups 2 and 3). |
| 10:13.50 | Makenshi | Are there any channel banks that support upward of 200 FXO ports? |
| 10:14.10 | JT | vlt: yeah that should call ch 2 if ch 1 is busy if you call g1 |
| 10:14.38 | vlt | JT: Ok, I'll check again ... |
| 10:18.23 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 10:18.29 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 10:24.25 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 10:27.52 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 10:31.11 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 10:33.10 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 10:34.15 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 10:34.25 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 10:43.44 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 10:59.59 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 11:03.17 | skirmisha | does anyone know how can i catch in chan_sip user unregistrations |
| 11:05.01 | *** join/#asterisk slayer192 (n=slayer19@pirus.securax.be) |
| 11:08.28 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 11:11.01 | *** part/#asterisk jaike (n=jaike@203.115.188.120) |
| 11:14.09 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 11:18.10 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 11:20.42 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 11:21.39 | *** join/#asterisk Pon`work (n=jamesm@ip-217.146.113.66.merula.net) |
| 11:31.13 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 11:32.51 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 11:41.35 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 11:43.16 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 11:44.29 | *** join/#asterisk shinux__ (n=shinux@196.201.153.178) |
| 11:51.20 | *** join/#asterisk susinths (n=susinths@1x-193-157-180-143.uio.no) |
| 12:02.34 | Kigh | skirmisha: what do you want to do with the catched unregisters? |
| 12:03.27 | Kigh | does somebody have experience in using a Windows TSP / TAPI solution with asterisk? |
| 12:03.46 | creativx | i tried asttapi |
| 12:03.50 | creativx | but it didnt want to work |
| 12:03.54 | creativx | so i didnt spend more than 10 minutes on that |
| 12:14.59 | *** join/#asterisk clive- (n=pirch@dsl-243-120-241.telkomadsl.co.za) |
| 12:16.43 | Kigh | i see |
| 12:18.39 | Kigh | problem is, our customer has a clean 64-bit environment (winxp 64bit edition) and none of the solutions wanted to run on 64 bit. seems like 64-bit windows does not support 32-bit drivers (if TSP is a driver for windows) |
| 12:20.05 | Kigh | well, star outlook dialer worked but is a) not usable (it executes every call twice and stuff) and b) not a TSP solution (no click&dial in outlook, solution is an outlook toolbar) |
| 12:23.35 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 12:24.22 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 12:25.54 | *** join/#asterisk |Johny| (n=gomesper@bacus.corp.fccn.pt) |
| 12:26.00 | |Johny| | <PROTECTED> |
| 12:26.23 | clive- | I am looking for a good usa toll free did provider |
| 12:27.22 | clive- | any recommendations |
| 12:27.58 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-81.megalan.bg) |
| 12:28.56 | *** join/#asterisk Jeff-Johnson (n=grml@unaffiliated/jeffjohnson) |
| 12:29.18 | Jeff-Johnson | howdy, I don't get an busy tone, if my sip line is busy... there is an sip adapter logged in on the account |
| 12:29.23 | Jeff-Johnson | anbody has an idea? |
| 12:29.29 | Jeff-Johnson | "t" dial option don't makes a difference |
| 12:31.14 | EmleyMoor | Jeff-Johnson: Do you mean when someone tries to call it, they don't get a busy tone? |
| 12:34.21 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:37.40 | Jeff-Johnson | yes |
| 12:37.44 | Jeff-Johnson | EmleyMoor |
| 12:45.23 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 12:49.22 | skirmisha | Kigh: i just want to make a log when user unregister |
| 12:49.53 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 12:50.00 | EmleyMoor | Jeff-Johnson: What kind of kit is the call coming from? |
| 12:50.52 | *** join/#asterisk HexDump (n=a@31.Red-217-126-215.staticIP.rima-tde.net) |
| 12:50.54 | HexDump | hi all |
| 12:51.59 | Jeff-Johnson | EmleyMoor: from landline, in the meatime I now get an busy announcment after 30sec, it has something todo with the freepbx dialplan i think |
| 12:53.21 | *** join/#asterisk coppice (n=chatzill@123.196.17.210.dyn.pacific.net.hk) |
| 12:53.59 | skirmisha | Kigh: any ideas? |
| 12:55.58 | vlt | Hello. Does anyone know if a GXP2000 phone can display some status messages (like "Dialling peer XY") on its diplay and if Asterisk can send those messages over SIP protocol? |
| 12:56.27 | vlt | skirmisha: Don't know if Asterisk can do this. I use tcpdump for similar tasks here. |
| 12:57.00 | skirmisha | vlt currently not, but i need to modify chan_sip tough |
| 12:57.52 | skirmisha | also i have another question |
| 12:58.11 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 12:58.15 | skirmisha | can asterisk route calls based on routing table and not on registration table |
| 12:59.24 | ThoMe | hallo. |
| 13:01.31 | skirmisha | where are u guys |
| 13:07.56 | vlt | skirmisha: Germany ;-) |
| 13:09.14 | *** join/#asterisk fr33bi3 (n=fr33bi3@122.164.147.161) |
| 13:10.04 | *** join/#asterisk dswillia74437 (n=me@71.153.188.97) |
| 13:10.28 | dswillia74437 | anyone here using asterisk in a large call center application? |
| 13:10.58 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:11.03 | ThoMe | vlt: ich auch ;) |
| 13:11.44 | cpm | ich auch was? |
| 13:12.07 | vlt | cpm: Ja, Du auch was. |
| 13:12.09 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:12.19 | cpm | :) |
| 13:12.28 | cpm | ich bin nicht |
| 13:15.00 | *** join/#asterisk vanumo (n=test@80.122.72.250) |
| 13:15.03 | vanumo | hi :-) |
| 13:15.10 | *** join/#asterisk grugnog (n=Grugnog@87.127.31.157) |
| 13:15.20 | grugnog | Hi all |
| 13:15.45 | vanumo | have anybody chan_cellphone in use ? i can make outcalls but without sound ? |
| 13:21.41 | grugnog | Does anyone have a suggestion of the best way to get round the problem of callfile ownership - i.e. asterisk needs to own the files to be able to change the umask, which is difficult is they are placed there by another user (e.g. a webserver) |
| 13:22.18 | grugnog | the only solution I can think of is to run the webserver process under the asterisk user...but that just feels like giving up ;) |
| 13:22.30 | grugnog | any ideas would be much appreciated! |
| 13:24.15 | RoyK | grugnog: sudo, perhaps |
| 13:24.19 | RoyK | sudo and a script |
| 13:24.53 | RoyK | ~rtfm |
| 13:25.06 | jbot | rumour has it, rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 13:25.34 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 13:25.46 | RoyK | jbot: rtfm is also http://uncyclopedia.org/wiki/Rtfm |
| 13:26.07 | jbot | ACTION tells is also http://uncyclopedia.org/wiki/Rtfm to RTFM |
| 13:26.39 | key2 | RoyK :) |
| 13:27.36 | grugnog | RoyK: Thanks |
| 13:27.40 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:27.43 | key2 | he sais |
| 13:27.43 | RoyK | grugnog: np |
| 13:27.47 | key2 | READ THE FUCKIN MANUAL |
| 13:27.48 | key2 | !!! |
| 13:27.50 | key2 | :) |
| 13:28.00 | RoyK | no. repeat the first message |
| 13:28.01 | RoyK | :P |
| 13:28.08 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 13:28.27 | grugnog | I am still stuck actually, since we run Openwall linux...so no sudo |
| 13:28.44 | grugnog | but at least I knew my instincts were right ;) |
| 13:28.48 | bobbytux | lo |
| 13:30.22 | grugnog | su - asterisk -c 'cp /tmp/steam.call /var/spool/asterisk/outgoing' looks hopeful |
| 13:31.44 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 13:33.10 | dswillia74437 | anyone here using asterisk in a large call center application? We are looking at using in our DR area, and I just wanted to verify asterisk "could be" a workable solution, and more importantly we are looking at the possibility of using sip or iax inbound and would like to know if there are any reliable, sound, vendors that would be recommended for about 15 t1's worth of traffic, but only in a DR scenerio |
| 13:34.42 | Damin | dswillia74437: Yes, it can do it if archtected properly. And yes, there are vendors that can easily handle that traffic. |
| 13:36.32 | dswillia74437 | my concern, as will be our CTO's is that the vendor be financially sound, seems a lot of people come and go in this space |
| 13:37.45 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 13:37.48 | *** join/#asterisk MattH (n=MattH@63.174.244.230) |
| 13:38.04 | dswillia74437 | we currently run 1+ million minutes a month on a traditional telephony network (sprint) and we you an Avaya infront of our IVR platform, thhis would replace the avaya part in front of a scaled down IVR platform |
| 13:38.29 | MattH | What would cause an error like chan_agent.c: Extension '688' is not valid for automatic login of agent '9999' |
| 13:38.47 | MattH | does that sound like a context problem.... or am I barking up the wrong tree trying to chase down a context issue? |
| 13:41.05 | *** join/#asterisk xibalba (n=reza@cpe-72-130-190-135.san.res.rr.com) |
| 13:41.08 | xibalba | hello everyone |
| 13:41.25 | xibalba | when i call my asterisk box on an inbound call it says the person is not available infact it never rings, goes right to VM. any thoughts? |
| 13:41.37 | xibalba | i have my inbound route setup as any cid/ any did, going to a specific core extension |
| 13:41.37 | MattH | is your extention registered? |
| 13:41.50 | xibalba | and the extension is setup on x-lite, and i can dial out to my cell phone from that ext. |
| 13:41.59 | MattH | yes but can you dial into it? |
| 13:42.09 | xibalba | dial into extension 200? |
| 13:42.15 | xibalba | dunno, i only have one machine to work with |
| 13:42.37 | MattH | this is going to sound odd.. but try turning call waiting on on the X-Lite phone |
| 13:42.53 | xibalba | k , ill try to find that |
| 13:43.04 | xibalba | i'm so excited, i finally got my sip trunk working with broadvoice |
| 13:43.07 | xibalba | it took allnight |
| 13:43.11 | MattH | should be *70 if you have a trixbox or asterisk@home setup, I believe |
| 13:43.14 | xibalba | but now i can only dial out |
| 13:43.14 | xibalba | lol |
| 13:43.49 | xibalba | "the party you are trying to reach, is not available to take your call" |
| 13:46.10 | xibalba | anything else i should look at? I have an ANY DID / ANY CID inbound trunk setup to just go straight to an extension |
| 13:46.23 | MattH | check your full debug log |
| 13:46.27 | MattH | it shoudl tell you why the call is failing |
| 13:46.37 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:46.39 | xibalba | k |
| 13:47.09 | xibalba | i tried grapping it for Inbound, but didn't find anything in the debug log |
| 13:47.14 | xibalba | I see Outbound going to my cell |
| 13:47.40 | MattH | ok.. when your call comes in you should see it in your full debug logs.. if yu don't maybe it's not an issue on your end. |
| 13:48.03 | xibalba | have you ever used broadvoice? |
| 13:48.12 | xibalba | i've heard they're kind of a crappy carrier |
| 13:48.38 | MattH | did at one point.. and I had some issues with them, yes.. but we did get inbound and outbound working |
| 13:48.43 | awk | dswillia74437: tell me something, how are you informed if some of your calls are not getting routed in some or ther country |
| 13:48.45 | awk | without waking up in the morning and knowing you just lost 10 thousand minutes? |
| 13:48.58 | xibalba | well i'm watching tcpdump on my router, looks like a spike in data from broadvoice when i dial into my box. |
| 13:49.24 | xibalba | nevermind, only one more packet than normal |
| 13:53.02 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 13:56.13 | xibalba | MattH, well when i do the 7777 test the IVR menu comes up just fine. Did you have to do anything with broadvoice to get them to forward the calls inbound to your asterisk box properly? |
| 13:56.44 | MattH | no... but it's possible you aren't registered correctly with them |
| 13:56.45 | xibalba | i'm setup with their BYOD plan....so i wouldn't expect to have to |
| 13:56.49 | MattH | this would cause them to not be able to send calls to you |
| 13:57.45 | xibalba | could you look at http://proxy.rndcomputing.com/~reza/asterisk1.bmp |
| 13:58.10 | xibalba | should be registered correctly right? sorry i'm a total asterisk newb |
| 13:58.15 | xibalba | just picked it up last week |
| 13:58.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:58.38 | *** join/#asterisk kornerr (n=kornerr@gw-e-light-telecom-fr.ll-kem.zsttk.ru) |
| 13:58.51 | *** part/#asterisk kornerr (n=kornerr@gw-e-light-telecom-fr.ll-kem.zsttk.ru) |
| 14:00.41 | xibalba | http://proxy.rndcomputing.com/~reza/asterisk2.bmp too if you dont mind |
| 14:01.32 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 14:01.54 | xibalba | ahh, you know what. Maybe it's my firewall |
| 14:02.04 | xibalba | the only port i punched through was 5060 back to my asterisk box |
| 14:04.57 | tzafrir_laptop | xibalba, it is generally much more productuve to produce text dumps |
| 14:05.13 | xibalba | ok...gotta figure out how to do that |
| 14:05.46 | tzafrir_laptop | what ssh client do you use? |
| 14:05.53 | xibalba | ssh/putty/java ssh |
| 14:05.54 | xibalba | whatever |
| 14:06.02 | xibalba | im running the box inside a vmware img |
| 14:06.10 | tzafrir_laptop | just mark the text. IT is automatically copied to the clipboard |
| 14:06.26 | xibalba | ah, yeh vmware via rdesktop is kind of a lamer |
| 14:06.32 | xibalba | i guess i could ssh into it |
| 14:06.36 | xibalba | heh |
| 14:06.54 | xibalba | ok that's quite a bit easier |
| 14:07.07 | xibalba | been a long night trying to get this broadvoice trunk up |
| 14:07.28 | tzafrir_laptop | putty can also log whatever ig going through it to a log file |
| 14:07.36 | xibalba | yeh i've used that before |
| 14:07.39 | tzafrir_laptop | (most terminal programs can do that) |
| 14:07.41 | xibalba | i dont think they have putty for linux |
| 14:08.02 | tzafrir_laptop | This is usually better for something that is slightly longer than a short snippet |
| 14:08.42 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:08.42 | tzafrir_laptop | In addition, instead of uploading the text to your web server, consider simply using a pastebin, if that shortens the process: |
| 14:08.45 | tzafrir_laptop | ~pb |
| 14:09.00 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:09.01 | xibalba | yup, i've used pb before |
| 14:09.25 | tzafrir_laptop | xibalba, they have puttty for linux. But you can always run ssh from whatever terminal you want |
| 14:09.41 | tzafrir_laptop | (the linux console, xterm, konsole, or whatever) |
| 14:09.46 | xibalba | yup |
| 14:09.52 | xibalba | eterm,aterm |
| 14:09.55 | xibalba | etc.. |
| 14:10.06 | xibalba | still can't figure out why inbound calls aren't coming through |
| 14:10.35 | tzafrir_laptop | inbound calls from what? |
| 14:10.41 | xibalba | my cell phone to my asterisk box |
| 14:11.07 | tzafrir_laptop | through what? through a zaptel card? |
| 14:11.21 | xibalba | sip trunk |
| 14:11.45 | xibalba | ah, maybe i need to port forward more than just 5060 |
| 14:12.18 | tzafrir_laptop | port 5060 should be enough for the sognalling. Without it you'll probably just not hear anything |
| 14:12.29 | tzafrir_laptop | do you see the call in 'show channels'? |
| 14:12.44 | xibalba | well currently with just 5060 being forwarded i can place outbound calls to my cell phone, but when i try to call my asterisk box from my cell phone it just hangs up |
| 14:13.12 | xibalba | no, doesn't look like it shows up |
| 14:13.52 | tzafrir_laptop | actually: it is interesting to see 'sip show users' (just dont post the secrets) - into which context is the call supposed to go? |
| 14:14.11 | xibalba | a |
| 14:14.18 | xibalba | 200 200 from-internal No Always |
| 14:14.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:15.04 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:16.33 | anonymouz666 | which is better format_mp3 or mpg123.063 with ast-1.2.15? |
| 14:16.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:17.07 | [TK]D-Fender | anonymouz666: Better for WHAT..... |
| 14:17.24 | tzafrir_laptop | when you try to call, do you see anything? |
| 14:17.28 | tzafrir_laptop | If not: if you enable 'sip debug', do you see anything? |
| 14:18.08 | xibalba | when i call it, i get call failed on my cell phone |
| 14:18.13 | xibalba | tzafrir_laptop, you want to try calling it? |
| 14:18.26 | anonymouz666 | [TK]D-Fender MOH |
| 14:18.39 | Pon`work | hasn't mpg123 been made redundant? |
| 14:18.40 | tzafrir_laptop | xibalba, actually, no, I should be leaving soom |
| 14:18.43 | xibalba | how do i turn off sip debug |
| 14:18.48 | [TK]D-Fender | anonymouz666: There are advantages to each and its up to you to value them |
| 14:18.49 | tzafrir_laptop | sip no debug |
| 14:18.50 | anonymouz666 | they say, don't use mpg123-0.63 but nobody say why |
| 14:18.50 | xibalba | i see a bunch of data spit out |
| 14:19.02 | xibalba | want to see it ? |
| 14:19.26 | xibalba | it's huge |
| 14:19.51 | xibalba | http://proxy.rndcomputing.com/~reza/sip.txt |
| 14:20.00 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:20.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:21.16 | [TK]D-Fender | anonymouz666: Try Native MoH. Odds are you'll like it just fine. If so it is lighter and integrated. |
| 14:21.35 | xibalba | tzafrir, did you get a look at that? |
| 14:23.57 | [TK]D-Fender | xibalba: Looking for 8589264573 in from-broadvoice (domain 192.168.1.199) |
| 14:23.58 | [TK]D-Fender | Reliably Transmitting (NAT) to 147.135.20.128:5060: |
| 14:23.59 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 14:24.27 | [TK]D-Fender | xibalba: Fix your inbound context, it should have "exten => 8589264573,1,(something here)" |
| 14:24.30 | xibalba | hmm, so maybe i need to setup the user context info with externip=xxx.xxx.xxx.xxx? nat=yes |
| 14:24.51 | [TK]D-Fender | xibalba: No, you have an extensions.conf problem right now, not sip.conf. |
| 14:25.09 | xibalba | hmm, i used freepbx to set this up |
| 14:25.21 | [TK]D-Fender | ~freepbx |
| 14:25.24 | jbot | well, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:25.54 | xibalba | hehe |
| 14:25.56 | xibalba | thanks |
| 14:26.00 | xibalba | at least i know what to look for now |
| 14:26.02 | [TK]D-Fender | xibalba: Then I'm sure you'll be able to follow one of the many guides out there for setting up Broadvoice with it. |
| 14:26.25 | xibalba | do you guys build everything from scratch? |
| 14:26.49 | anonymouz666 | [TK]D-Fender: what about the MOH mp3 files? |
| 14:26.57 | [TK]D-Fender | xibalba: Yes. FreePBX/TrixBox / AsteriskNOW / whatevere != Asterisk. |
| 14:27.04 | [TK]D-Fender | anonymouz666: ? |
| 14:27.26 | anonymouz666 | does the native MoH will play the mp3 files ? |
| 14:27.27 | xibalba | [TK]D-Fender, do you only use asterisk for one company? |
| 14:27.33 | SomethingISODD | hello all, anyone here know alot about pwlib? if so can yo u please take a look at this error http://www.pastebin.ca/377749 |
| 14:27.52 | SomethingISODD | and maybe tell me what i am doing wrong. |
| 14:28.06 | [TK]D-Fender | xibalba: I'm an Asterisk consultant and have done multi-temment setups, home installs, small companies, etc. |
| 14:28.33 | xibalba | [TK]D-Fender, i'm trying to learn as much as possible about it to deploy it for my clients that i consult to for IT services. |
| 14:29.16 | [TK]D-Fender | xibalba: Well FreePBX won't teach you much of anything. Start here : |
| 14:29.18 | [TK]D-Fender | ~docs |
| 14:29.20 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 14:29.20 | [TK]D-Fender | ~book |
| 14:29.21 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:29.21 | [TK]D-Fender | ~wikis |
| 14:29.23 | jbot | hmm... wikis is http://www.voip-info.org |
| 14:29.34 | [TK]D-Fender | xibalba: Start with THE BOOK. |
| 14:29.46 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 14:29.50 | Simplix | hell all |
| 14:29.53 | Simplix | hello |
| 14:29.59 | SomethingISODD | anyone? |
| 14:30.55 | xibalba | [TK]D-Fender, yeh I'm also reading O'Reilly Asterisk |
| 14:30.59 | xibalba | and Trixbox Made Easy |
| 14:31.26 | Simplix | my issue : NOTICE[6788]: chan_sip.c:7085 sip_reg_timeout: -- Registration for 'xxx@gw1.directcentrex.com' timed out, trying again (Attempt #1) indefinitely after an undefined period of time |
| 14:31.48 | anonymouz666 | [TK]D-Fender: does the native moh will play all formats including mp3 without mpg123? |
| 14:31.55 | *** join/#asterisk orkid (n=orkid@dataq2.utias.utoronto.ca) |
| 14:32.07 | xibalba | [TK]D-Fender, so do you always build everything from scratch? Like the voicemail system, etc |
| 14:32.22 | Kigh | does somebody have experience in using a Windows TSP / TAPI solution with asterisk? i'm looking for a working one for 64-bit windows xp |
| 14:32.32 | Kigh | mitcheloc: go compile 64bit :) |
| 14:32.37 | ManxPower | ~mailinglist |
| 14:32.38 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 14:32.51 | [TK]D-Fender | anonymouz666: Yes. Go read about it in THE BOOK |
| 14:32.53 | Pon`work | anonymouz666, Format_MP3 will play mp3's. |
| 14:33.01 | [TK]D-Fender | xibalba: Yes. |
| 14:33.49 | [TK]D-Fender | xibalba: Trixbox made easy is like saying "Idiots for Dummies" practically, heh |
| 14:34.07 | ManxPower | [TK]D-Fender: It looks to me like its Newbie Day |
| 14:34.37 | [TK]D-Fender | ManxPower: And here I thought it was "more snow that you can shake a stick at.... 'cause its BURIED" Day :) |
| 14:34.54 | xibalba | [TK]D-Fender, how much do you usually charge for SOHO PBX setup? |
| 14:35.24 | xibalba | i have one client paying almost $800 a month for 10 lines and a basic PBX w/only 1 voicemail |
| 14:35.28 | [TK]D-Fender | xibalba: Depends on the size and complexity. I usually clock in cheaper than most I'm sure. |
| 14:35.55 | xibalba | i'd like to come in and offer them at least half that with alot more features |
| 14:36.04 | [TK]D-Fender | xibalba: My complete install charge has never reaached $800 TOTAL. |
| 14:36.09 | xibalba | damn |
| 14:36.19 | xibalba | you must use scripts or keep files on hand for all the standard stuff |
| 14:36.22 | ManxPower | I don't do SOHOs. Never ends up being profitable |
| 14:37.03 | ManxPower | SOHOs always want silly things like BLF and cheap phones |
| 14:37.33 | xibalba | BLF? |
| 14:37.41 | ManxPower | Busy Lamp Field |
| 14:39.02 | [TK]D-Fender | ManxPower: You don't do basic Presence? eek. |
| 14:40.00 | ManxPower | [TK]D-Fender: The few times I've done BLF they never ended up using it much. |
| 14:40.20 | SomethingISODD | hello all, anyone here know alot about pwlib? if so can yo u please take a look at this error http://www.pastebin.ca/377749 |
| 14:40.21 | ManxPower | In a SOHO there really isn't a need since there are so few people, with larger offices it doesn't scale. |
| 14:40.26 | [TK]D-Fender | ManxPower: Kind of an "always give" scenario for me. Polycom Presence just kinda rocks.... |
| 14:40.56 | ManxPower | [TK]D-Fender: Ah. If you can convince a customer to use Polycom then you have solved the "we're cheap" issue. |
| 14:41.18 | xibalba | i hear those are pretty bad |
| 14:41.26 | [TK]D-Fender | ManxPower: Yes, if they choose GS or some other crap it becomes "manage your own damn phones.. I'll setup the server". |
| 14:41.45 | [TK]D-Fender | xibalba: Heard what are pretty bad? |
| 14:43.22 | xibalba | polycomms |
| 14:43.57 | ManxPower | HAHAHAHA! The only people that think Polycoms such are the ones that have 1) never used Polycoms or 2) have not learned how to manage and set them up. |
| 14:44.49 | ManxPower | You have to learn quite a lot to get Polycoms setup, but once you learn the required stuff they are a breeze. |
| 14:45.04 | Simplix | anyone have a solution for my problem ? |
| 14:45.44 | ManxPower | Simplix: Your problem is that the far end is not responding |
| 14:46.05 | Simplix | not exactely |
| 14:46.16 | [TK]D-Fender | xibalba: In case you didn't know, Polycom is amongst the most stable flexible, and usable phones on the market as well as being top-tier in audio & manufacturing quality. |
| 14:46.24 | Simplix | ManxPower, with a sofphone i can connect to directcentrex |
| 14:46.34 | ManxPower | Exactly. Asterisk is sending the registation packet and Asterisk does not get a response. |
| 14:46.51 | ManxPower | wunderkin: You are just cursed. |
| 14:46.55 | wunderkin | heh |
| 14:46.58 | Simplix | ManxPower, but asterisk don't connect until i reboot the computer |
| 14:47.03 | [TK]D-Fender | wunderkin: Mornin' whippin' boy! |
| 14:47.09 | wunderkin | baah! |
| 14:47.25 | Simplix | ManxPower, restarting asterisk is not enough |
| 14:47.33 | ManxPower | I still think wunderkin needs to use firmware 1.6.7, Asterisk 1.2.15, and my config files for the polycoms |
| 14:47.56 | wunderkin | ManxPower, well, i did go back to 1.6.7... we are using abe b-1.3... i have not done the last one yet... |
| 14:48.12 | wunderkin | 1.6.7 caused us more reboots.. 2.1.0 fixed most of them.... |
| 14:48.24 | ManxPower | wunderkin: I believe you have bad config files. |
| 14:48.29 | wunderkin | but we still have the key problem thing, so something is wrong somewhere |
| 14:48.30 | ManxPower | we never have phone reboots |
| 14:48.36 | wunderkin | but we had that problem even with a bare 2.1.0 config |
| 14:48.58 | ManxPower | You know what I think is causing the key problems |
| 14:49.08 | wunderkin | i've done that too, still am |
| 14:49.22 | wunderkin | they say it is worse on 2.1.0 than 1.6.7, for some reason |
| 14:49.55 | [TK]D-Fender | wunderkin: You know I'm still available for contracting to help in the complete rebuild :) |
| 14:49.58 | wunderkin | heh |
| 14:50.03 | wunderkin | get out the bulldozer! |
| 14:50.11 | Simplix | ManxPower, i think the far end respond but asterisk don't get the response |
| 14:50.21 | *** join/#asterisk TaiSHi (n=juanma@zion.dattaweb.com) |
| 14:50.23 | TaiSHi | Hello everyone |
| 14:50.34 | [TK]D-Fender | wunderkin: And I *DID* encounter that 1 key-pad entry bug in 2.1.0 you discovered |
| 14:50.40 | wunderkin | yeah... may need to.... trying to see what i can make of all of this first, if anything |
| 14:50.46 | wunderkin | [TK]D-Fender, yup.... :D |
| 14:50.50 | TaiSHi | Could someone explain me a bit more about how does asterisk handle invalid menu entries and timeouts? |
| 14:50.51 | ManxPower | Simplix: Asterisk cannot know that. All Asterisk knows is that it never got a response |
| 14:50.54 | *** join/#asterisk MarkWD (n=Mark@rrcs-67-78-88-186.sw.biz.rr.com) |
| 14:51.03 | TaiSHi | I mean, in TFOT it marks it as 'i' and 't', but are those fixed variables? |
| 14:51.03 | Simplix | ManxPower, sure |
| 14:51.10 | [TK]D-Fender | wunderkin: So while you are still likely crazy.... you have occasionally been RIGHT as well ;) |
| 14:51.20 | ManxPower | TaiSHi: Yes, those are hardcoded |
| 14:51.47 | TaiSHi | ManxPower: and what about the timeout time on 't' variable? |
| 14:51.54 | TaiSHi | ManxPower: Thanks, by the way |
| 14:51.58 | [TK]D-Fender | wunderkin: Plenty of .... CATS tonight ;) |
| 14:52.11 | wunderkin | heh :P |
| 14:52.18 | Simplix | ManxPower, but it work fot weeks and since today it don't work anymore |
| 14:52.35 | ManxPower | TaiSHi: http://pastebin.ca/378436 |
| 14:52.47 | wunderkin | i probably should have used the manx config when i went back to 1.6.7 |
| 14:52.49 | [TK]D-Fender | TaiSHi: Go read this for a whilehttp://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 14:52.50 | ManxPower | TaiSHi: i and t are NOT VARIABLES |
| 14:52.53 | ManxPower | They are extensions |
| 14:52.55 | ManxPower | see my pastebin |
| 14:53.00 | TaiSHi | True, opening now |
| 14:53.07 | TaiSHi | (slow 270Mb connectio >.<) |
| 14:53.21 | ManxPower | TaiSHi: Extensions do not have to be numbers. |
| 14:53.21 | [TK]D-Fender | TaiSHi: and this : http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu |
| 14:53.42 | [TK]D-Fender | TaiSHi: Its also realtively well laid out in THE BOOK |
| 14:53.44 | [TK]D-Fender | ~book |
| 14:53.46 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:54.24 | TaiSHi | [TK]D-Fender: I figured it now I saw the Timeouts commands |
| 14:54.34 | TaiSHi | Thanks both of you |
| 14:55.07 | nfi|ermes | [TK]D-Fender, do you use svn version with gui ? |
| 14:55.27 | [TK]D-Fender | nfi|ermes: I don' to do "GUI's" |
| 14:55.44 | TaiSHi | nfi|ermes: Still getting that weird core dump? |
| 14:55.56 | nfi|ermes | no solved upgrading to svn version |
| 14:56.21 | TaiSHi | Good :) |
| 14:56.21 | ManxPower | I think you'll find the GUI people hang out on #asterisk-gui, #asterisknow, #freepbx |
| 14:56.26 | *** join/#asterisk rmayorga (i=rmyorg@168.243.89.17) |
| 14:56.40 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 14:56.46 | nfi|ermes | i m already there |
| 14:57.02 | nfi|ermes | but in asterisk-gui a bad virus killed everyone :D |
| 14:57.11 | nfi|ermes | nobody writes more |
| 14:57.11 | [TK]D-Fender | nfi|ermes: How's the echo? ;) |
| 14:57.33 | nfi|ermes | funny |
| 15:00.55 | coppice | echo is for wieners (a little in DSP joke there :-) ) |
| 15:02.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:02.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:02.46 | Pon`work | ho ho ho |
| 15:03.20 | [TK]D-Fender | ho-ly shit the white crap keeps on fallin' here... |
| 15:03.32 | [TK]D-Fender | ^%@$#$%^@&*^#$%^@#$&@#%$@&# |
| 15:03.37 | [TK]D-Fender | *BLARG* |
| 15:04.04 | ManxPower | You are in Canada, you should be expecting it. |
| 15:04.19 | coppice | gwano? :-\ |
| 15:05.11 | wunderkin | when it turns to brown or white with brown (or similar) crap.. then be scared |
| 15:06.04 | [TK]D-Fender | ManxPower: Actually we were snow-free through christmas this year.... was goin SOOOO well.... its a big dump today. Phsycho out there. |
| 15:06.26 | TaiSHi | It's summer here, and this morning it was @ 12ºC |
| 15:06.31 | TaiSHi | Dont complain |
| 15:06.34 | *** join/#asterisk bogomil (n=bogomil@194.12.229.90) |
| 15:06.44 | anonymouz666 | 30 C here |
| 15:06.53 | wunderkin | yes, the weather this year, pretty weird... darn global warming.. we had snow in phoenix little while back... |
| 15:06.58 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:06.59 | anonymouz666 | I love AC |
| 15:07.11 | anonymouz666 | thats why tomorrow I will go to the beach |
| 15:07.20 | TaiSHi | :O |
| 15:07.23 | TaiSHi | Hate >.< |
| 15:07.27 | coppice | if you have AC why can't you get it cooler than 30C? |
| 15:07.27 | TaiSHi | I went to beach few weeks ago |
| 15:07.45 | TaiSHi | :O |
| 15:07.49 | anonymouz666 | 30C outside |
| 15:07.50 | anonymouz666 | lol |
| 15:07.51 | Simplix | ManxPower, any idea for my issue ? |
| 15:07.53 | TaiSHi | I've got 18 AC here |
| 15:07.57 | TaiSHi | 16 of them are industrial |
| 15:08.00 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 15:08.11 | TaiSHi | damn computers |
| 15:08.14 | TaiSHi | They heat quite a lot :P |
| 15:08.27 | coppice | its only 25C outside here |
| 15:08.30 | bogomil | hi all I have a freaky question about Asterisk: How to check the status of a call |
| 15:09.13 | ManxPower | It is supposed to be sunny and a high of 68F (20C) today |
| 15:09.31 | coppice | 20C sounds a bit chilly |
| 15:09.33 | ManxPower | Simplix: Yes. Exactly what I already said. |
| 15:09.45 | ManxPower | bogomil: define "status of a call" |
| 15:10.02 | ManxPower | coppice: I would prefer 25C, but I can live with 20C |
| 15:10.03 | bogomil | I want to make a PHP predictive dialer |
| 15:10.27 | bogomil | and I will dial 5 numbers and I want to check the status of 5 calls |
| 15:10.48 | ManxPower | bogomil: What statuses are you trying to detect? |
| 15:11.11 | Pon`work | I like it nice and cold - 5 - 15C is just great |
| 15:11.22 | ManxPower | bogomil: I assume none of the existing predictive dialers will work for you |
| 15:11.27 | bogomil | ManxPower: busy and answering mashine |
| 15:11.28 | coppice | if he's making a predictive dialer the only valid statuses will be "busy" and "bloody annoying" |
| 15:11.44 | ManxPower | bogomil: asterisk does not detect answering machines |
| 15:11.49 | bogomil | I see |
| 15:11.54 | bogomil | then busy only |
| 15:12.08 | ManxPower | bogomil: the value of HANGUPCAUSE or DIALSTATUS will tell you what the result of the call is. |
| 15:12.19 | TaiSHi | How can I make * to choose an operator randomly ? |
| 15:12.20 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 15:12.24 | ManxPower | see "show application dial" in the asterisk CLI |
| 15:12.30 | TaiSHi | (from a group, avoiding my office, of coursE) |
| 15:12.36 | bogomil | ManxPower: thanks |
| 15:12.41 | ManxPower | TaiSHi: see queues on the Wiki |
| 15:13.48 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 15:14.24 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 15:22.16 | *** join/#asterisk ping2921 (n=marc3234@206-248-153-161.dsl.teksavvy.com) |
| 15:23.04 | *** join/#asterisk dreamind (n=dreamind@p54A7C1AE.dip0.t-ipconnect.de) |
| 15:23.05 | dreamind | Hi folks |
| 15:23.18 | dreamind | can anybody tell me if and how its possible to wrap long lines in asterisk config files? |
| 15:25.12 | ManxPower | dreamind: no it is not possible |
| 15:27.31 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:27.57 | dreamind | ManxPower: oh :( |
| 15:28.18 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 15:28.23 | Telemac | Hello |
| 15:28.53 | ManxPower | dreamind: I think the max config file line length is around 250 characters |
| 15:29.49 | dreamind | ok, well my lines are shorter, but I would like to wrap them to make reading easier |
| 15:30.36 | ManxPower | use an editor that supports "soft line wrap" |
| 15:30.54 | dreamind | yep I do, but I disabled that feature... |
| 15:30.55 | nfi|ermes | fd == -1 in astman_append, should not happen |
| 15:31.05 | nfi|ermes | ? |
| 15:31.20 | ManxPower | I use jedit, it can wrap the line in the screen for easy viewing, but does not actually wrap the line when saving |
| 15:31.53 | ManxPower | nfi|ermes: We don't use astman here. |
| 15:32.32 | Juggie | even notepad does that |
| 15:36.46 | *** part/#asterisk bogomil (n=bogomil@194.12.229.90) |
| 15:39.00 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 15:41.40 | Telemac | Hello, I'm trying to use features.conf. I've the "x,1,Set(DYNAMIC_FEATURES=testfeature)", but when I press '*9' as specified in applicationmap, I hears nothing either at caller or at caller. I think I should have missed something |
| 15:41.41 | *** join/#asterisk rmayorga (i=rmyorg@168.243.89.17) |
| 15:41.44 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 15:44.23 | EmleyMoor | Is it possible to get the caller ID of the previous call on a channel into the dialplan? |
| 15:45.30 | [TK]D-Fender | EmleyMoor: Shove it in AstDB. |
| 15:45.55 | EmleyMoor | Hmmm... could do |
| 15:46.03 | *** join/#asterisk Strom_M (n=strom@209.19.56.4) |
| 15:47.36 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:49.00 | Telemac | When I put Playback in applicationmap it doesn't work, I don't really known whether it is features use or just Playback that fails. Is it a "simplier" test for features.conf than Playback one ? |
| 15:49.10 | EmleyMoor | Is there something that I can make happen in the dialplan when a call is answered? |
| 15:49.41 | [TK]D-Fender | Telemac: Paste your dial statement... |
| 15:49.48 | HexDump | could anyone help a newbye with using asterisk with 2 softphones(sjphone)? |
| 15:50.06 | [TK]D-Fender | EmleyMoor: "show application dial" (M) |
| 15:50.33 | HexDump | I have followed several tutorial, and phones register ok, but no way to call each other, asterisk seems to cut the calls :S, can paste sip debug |
| 15:51.30 | *** join/#asterisk sasch (n=sasch@82.107.30.102) |
| 15:51.45 | ManxPower | HexDump: debug is usually only needed after we eliminate everything else. |
| 15:52.03 | ManxPower | pastbin (without debug) the CLI output of a failed call |
| 15:52.12 | HexDump | ManxPower: ok, sorry as I told you I´m a poor newbie :S |
| 15:52.27 | Telemac | [TK]D-Fender: Which Dial statement ? What I'm trying to do is to allow user to trigger action when pressing '*9' on their phone so I use test in features.conf and Set(DYANMIC_FEATURES=xxx) in extensions.conf |
| 15:52.43 | HexDump | ManxPower: if I don´t set "sip debug" to active, I get nothing on the cli |
| 15:52.50 | [TK]D-Fender | Telemac: Show me the DIAL command you issue AFTER that. |
| 15:53.06 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 15:53.57 | sasch | can help me with my problem with codec of swissphone and asterisk |
| 15:54.21 | sasch | in my swissphone I have only this G711A, G711µ and G729 |
| 15:54.25 | Telemac | [TK]D-Fender: "exten => xxx,4,Dial(IAX2/xxx|30)" |
| 15:54.45 | ManxPower | sasch: you cannot use G729 without a commercially licensed G729 codec |
| 15:54.51 | sasch | ok |
| 15:54.56 | sasch | i know that |
| 15:54.56 | ManxPower | Telemac: "show application dial" notice the w and W option |
| 15:55.04 | sasch | but if i want to use g711 |
| 15:55.05 | sasch | ?? |
| 15:55.11 | sasch | it is possible or not ?? |
| 15:55.35 | ManxPower | sasch: then set it to yes. |
| 15:55.39 | TaiSHi | No application 'DigitTimeout' for extension <-- I'm getting that message |
| 15:55.43 | ManxPower | sorry. Yes, you can use G711 |
| 15:55.54 | ManxPower | TaiSHi: what version of Asterisk? |
| 15:55.57 | TaiSHi | 1.4 |
| 15:55.58 | sasch | ok for configure it ?? |
| 15:56.13 | ManxPower | TaiSHi: those were removed in 1.4, use the new methods to set those options. See upgrade.txt |
| 15:56.21 | TaiSHi | Roger that |
| 15:56.22 | TaiSHi | Thanks |
| 15:56.30 | ManxPower | sasch: in sip.conf in the [section] name for that device put disallow=all and allow=ulaw |
| 15:56.37 | sasch | ok |
| 15:56.40 | sasch | i try |
| 15:56.45 | [TK]D-Fender | Telemac: Well your dial pattern is missing the "_" in front, and you need to use one of "tTwW" to enable Dial to use features.conf |
| 15:57.01 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 15:57.02 | HexDump | MainxPower: let me know when you´re a little more free :), don´t wanna rush you. And thanks. |
| 15:57.26 | [TK]D-Fender | TaiSHi: That app is deprecated. "core show functiuon TIMEOUT" |
| 15:57.54 | TaiSHi | Thank you |
| 16:01.15 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 16:02.21 | HexDump | ManxPower: this is what asterisk showed -> http://pastebin.com/892143 |
| 16:02.30 | *** join/#asterisk dswillia74437 (n=swilliam@199.3.247.34) |
| 16:02.32 | HexDump | when making a call |
| 16:03.58 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 16:04.53 | EmleyMoor | Wow, the DB now logs the CALLERID(num) of the last call answered per channel - that will make things interesting |
| 16:05.16 | ManxPower | HexDump: There is no exten => line to match the dialed phone number |
| 16:06.59 | HexDump | ManxPower: I edited extensions.conf |
| 16:07.21 | HexDump | ManxPower: and added them as some tutorials stated |
| 16:07.22 | HexDump | :S |
| 16:07.42 | ManxPower | HexDump: that does not change what I said. What number are you dialing? |
| 16:08.29 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 16:08.52 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 16:09.26 | HexDump | ahhh, sorry. 1234 |
| 16:09.33 | HexDump | just, used a dumb extension num |
| 16:10.52 | ManxPower | HexDump: and what context is exten => 1234,1,whatever in? |
| 16:11.24 | HexDump | tutorial |
| 16:11.49 | ManxPower | HexDump: you have [tutorial] and on the next line exten => 1234,1,whatever ? |
| 16:11.55 | ManxPower | in extensions.conf |
| 16:12.30 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 16:12.30 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 16:12.57 | TaiSHi | == Auto fallthrough, channel 'SIP/600-09c06f58' status is 'UNKNOWN' <-------- after I play a Background() |
| 16:12.58 | HexDump | yes of course |
| 16:13.50 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 16:13.57 | ManxPower | HexDump: and you have a context=tutorial in the sip.conf [whatever] for the phone? |
| 16:14.05 | ManxPower | TaiSHi: dd you read UPGRADE.txt? |
| 16:14.12 | TaiSHi | Yes |
| 16:14.27 | HexDump | ManxPower: let me check that |
| 16:14.28 | TaiSHi | Set(TIMEOUT(digit)=5) |
| 16:14.30 | ManxPower | and it did not mention you need WaitExten now when doing background/playback |
| 16:14.44 | TaiSHi | Wasn't it WaitForExt ? |
| 16:15.01 | ManxPower | TaiSHi: I dunno. "show applications like wait" |
| 16:15.05 | TaiSHi | Anyway, my upgrades.txt didn't said anything about Timeouts, caught it @ asteriskguru |
| 16:15.33 | mercestes | WaitExten? |
| 16:15.36 | TaiSHi | WaitExten |
| 16:15.49 | TaiSHi | No parameters ? |
| 16:16.02 | ManxPower | TaiSHi: I am not going to read the docs for you. |
| 16:16.11 | TaiSHi | I know |
| 16:16.15 | TaiSHi | I'm checking @ *guru |
| 16:16.28 | ManxPower | TaiSHi: don't do that. check the asterisk cli |
| 16:16.56 | ManxPower | if you can't find what you need from the built in CLI and docs, then check outside sites |
| 16:17.07 | Bobthehunter | can one PM me on how to fix BUG 9182 |
| 16:17.09 | Bobthehunter | please |
| 16:17.12 | ManxPower | The wikis frequently are totally wrong |
| 16:17.14 | [TK]D-Fender | TaiSHi: You probably don't see it because DigitTimeout was deprecated in 1.2 and REMOVED in 1.4 You are WAY too far behind. |
| 16:17.18 | HexDump | ManxPower: this is my sip.conf -> http://rafb.net/p/KYz7Z021.html |
| 16:17.35 | HexDump | ManxPower: I think I don´t have here the [tutorial] thing :S |
| 16:17.40 | ManxPower | HexDump: I cannot help you with Trixbox |
| 16:17.49 | TaiSHi | [TK]D-Fender: Reading TFOT u_U |
| 16:17.59 | Bobthehunter | ??? |
| 16:18.00 | TaiSHi | And taking examples from the sites you gave me before |
| 16:18.00 | HexDump | Trixbox? |
| 16:18.20 | queuetue | Wow, this config-by-hand stuff isn't so hard at all. :) |
| 16:18.23 | ManxPower | HexDump: sip_nat.conf and sip_additional.conf are trixbox/freepbx stuff |
| 16:18.49 | kratzers | is there a way to execute an AGI script when a queue member (agent) is being tried? |
| 16:18.49 | HexDump | ManxPower: but I´m only using sip.conf, and this is asterisk doesn´t it? |
| 16:18.53 | [TK]D-Fender | ManxPower: Looks like he's trying to move away from it manually... |
| 16:19.09 | ManxPower | HexDump: any context=tutorial in sip.conf needs a [tutorial] section in extensions.conf |
| 16:19.23 | [TK]D-Fender | kratzers: Have your agents be Local channels and call it before the Dial. |
| 16:19.31 | ManxPower | HexDump: then remove those #include lines so we can see what sip.conf REALLY is. |
| 16:19.37 | Bobthehunter | anyone ? |
| 16:19.40 | ManxPower | we don't know what all those included files are doing. |
| 16:19.40 | kratzers | k, thanks |
| 16:20.01 | HexDump | ManxPower: I see... |
| 16:20.01 | ManxPower | Bobthehunter: if there was a fix it would be in Mantis |
| 16:20.16 | Bobthehunter | thread islocked |
| 16:20.33 | Bobthehunter | one said you need to bind 0.0.0.0.. sovuln must be about internal card.. if its not binded.. |
| 16:20.42 | Bobthehunter | so whats the problem.. ill find a local fix.. |
| 16:20.48 | ManxPower | Bobthehunter: I have no idea what you just said |
| 16:20.55 | *** join/#asterisk cedric_accro (n=cedric@modemcable094.12-201-24.mc.videotron.ca) |
| 16:21.06 | cedric_accro | Hi |
| 16:21.34 | cedric_accro | Need help with a tdm400 rev.I that Fxos don't seem to work |
| 16:21.56 | sasch | <cedric_accro> i have a tdm400p |
| 16:21.57 | HexDump | ManxPower: do you think I should be using a fresh asterisk instalation instead of a pre-fabriqued one? (I'm using xorcom asterisk distribution) |
| 16:22.30 | ManxPower | HexDump: It would allow people here to help you without telling you things that don't work with Xorcom. |
| 16:22.44 | *** part/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 16:22.52 | HexDump | ManxPower: ok then, gonna make a fresh one, don´t wanna drive you mad |
| 16:22.53 | ManxPower | HexDump: have you read The Book? |
| 16:22.57 | ManxPower | ~book |
| 16:22.58 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:23.08 | HexDump | ManxPower: what book? :S |
| 16:23.22 | ManxPower | HexDump: not drive only me mad, drive anyone that tries to help you mad |
| 16:23.35 | HexDump | :) |
| 16:23.51 | HexDump | ok ok. What book was it ManxPower? |
| 16:23.59 | ManxPower | ~book |
| 16:24.00 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:24.01 | ManxPower | that book |
| 16:24.03 | Bobthehunter | ok then waht source part handles the -c part ? |
| 16:24.08 | HexDump | ahh ... |
| 16:24.14 | dual-man | I am getting a no such file or directory error when trying to execute an agi script |
| 16:24.20 | dual-man | fresh asterisk install |
| 16:24.32 | ManxPower | dual-man: paste the exten line that runs the AGI |
| 16:25.01 | dual-man | exten => 6,1,AGI(features.php|${RINGBACK}) |
| 16:25.11 | ManxPower | dual-man: try specifying the path |
| 16:25.14 | HexDump | no, did not read it. Didn´t even know it existed :), gonna have a look at it. Thanx ManxPower, will be back, when I sort my mind out :). |
| 16:25.48 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 16:25.58 | [TK]D-Fender | HexDump: .... |
| 16:26.00 | [TK]D-Fender | ~osmosis |
| 16:26.02 | jbot | extra, extra, read all about it, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 16:26.22 | [TK]D-Fender | FTW! |
| 16:26.53 | dual-man | exten => 6,1,AGI(/var/lib/asterisk/agi-bin/features.php|${RINGBACK}) = same thing |
| 16:27.29 | ManxPower | dual-man: chances are you have a permissions problem or the first like of the AGI points to the wrong place |
| 16:27.42 | HexDump | [TK]D-Fender: :( |
| 16:27.55 | HexDump | [TK]D-Fender |
| 16:28.01 | HexDump | [TK]D-Fender: it hurts man :/ |
| 16:28.43 | ManxPower | [TK]D-Fender: you are well on your way to becoming a BOFH. I'm so proud of you! |
| 16:28.50 | dual-man | Manxpower: as far as i can tell, everything is good |
| 16:29.02 | ManxPower | dual-man: apparently not. |
| 16:29.03 | dual-man | i have execute privalges for all users |
| 16:29.07 | dual-man | on the file |
| 16:29.24 | ManxPower | dual-man: outside of asterisk run this command: /var/lib/asterisk/agi-bin/features.php |
| 16:29.25 | TaiSHi | Auto fallthrough, channel 'SIP/600-082522e8' status is 'UNKNOWN' |
| 16:29.27 | ManxPower | what happens? |
| 16:29.29 | dual-man | and asterisk points to /var/lib/asterisk/agi-bin for agi's |
| 16:29.31 | TaiSHi | STill, even with WaitExten(30) |
| 16:29.52 | ManxPower | TaiSHi: put a copy of that exten block on pastebin.ca |
| 16:29.54 | [TK]D-Fender | ManxPower: Yes my Liege! |
| 16:31.08 | TaiSHi | http://pastebin.ca/378588 |
| 16:31.17 | TaiSHi | Christ... I want to sleep |
| 16:31.22 | TaiSHi | Been awak for 48 hours now >.< |
| 16:31.56 | dual-man | ; /usr/bin/php: bad interpreter: no such file or directory |
| 16:32.07 | ManxPower | TaiSHi: try this http://pastebin.ca/378593 |
| 16:32.08 | [TK]D-Fender | dual-man: HMMMMMM |
| 16:32.13 | ManxPower | dual-man: there is your problem |
| 16:32.13 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:32.15 | TaiSHi | By the way ManxPower, I've tried WaitExten over and below BG |
| 16:32.34 | ManxPower | [TK]D-Fender: damn I'm good 8-) |
| 16:32.49 | dual-man | ManxPower: my php is the probelm? |
| 16:32.49 | sasch | if i want that asterisk in a call send a dtmf for example #21#333333* |
| 16:32.55 | sasch | in wich mode i made it ?? |
| 16:33.04 | ManxPower | TaiSHi: you, of course, do a "reload" in the cli after every change to extensions.conf |
| 16:33.14 | TaiSHi | Of course |
| 16:33.26 | ManxPower | sasch: "show application dial" pay special attention to the D() option |
| 16:33.47 | TaiSHi | D(J)? |
| 16:34.59 | [TK]D-Fender | dual-man: "which php" |
| 16:35.07 | TaiSHi | <PROTECTED> |
| 16:35.18 | ManxPower | TaiSHi: I have no more suggestions |
| 16:36.10 | TaiSHi | Ok |
| 16:36.32 | TaiSHi | Let's do a roll... |
| 16:36.35 | TaiSHi | [TK]D-Fender: :) |
| 16:36.38 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 16:37.20 | funxion | anyone know how I owuld go about detecting dialtone before dialing on an analog line? |
| 16:37.33 | [TK]D-Fender | TaiSHi: pastebin the entire call, and please not you have a "Background" AFTER your Waitexten. That would normally be considered a bad thing. |
| 16:37.53 | TaiSHi | Ok |
| 16:37.57 | dual-man | ManxPower: what must i do to fix? |
| 16:38.59 | TaiSHi | [TK]D-Fender: http://pastebin.ca/378603 |
| 16:43.58 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:43.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:45.15 | Juggie | RUSSELLB! |
| 16:46.03 | russellb | zom ghi |
| 16:47.26 | Qwell[] | zomghi want braaaiiiinnnsss! |
| 16:47.26 | russellb | lawl |
| 16:47.28 | reber | wich are the softphone mainly used with linux and asterisk ? |
| 16:47.28 | *** join/#asterisk watchy (n=watchy@work.gwhsi.com) |
| 16:47.29 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 16:47.33 | watchy | i got a sipura ata that when i dial from a phone it takes a few seconds to dial, is there a way to make it faster? |
| 16:47.33 | Qwell[] | watchy: give the ATA a dialplan |
| 16:47.38 | Qwell[] | It's explained in the admin guide |
| 16:47.41 | watchy | in the ata or in asterisk |
| 16:47.45 | Qwell[] | in the ATA |
| 16:47.47 | watchy | ah ok |
| 16:47.49 | JunK-Y | lo guys |
| 16:48.09 | watchy | ill check it out then when i get on location |
| 16:49.30 | TaiSHi | [TK]D-Fender: Any idea on it? |
| 16:50.18 | *** join/#asterisk dlynes__ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 16:51.17 | SomethingISODD | I am using asterisk 1.4.0 i have pwlib and openh323 install i have ran make opt in the channels/h323 directory and then i ran make install in the main directory and its not creating the chan_h323.so anyideas please |
| 16:51.19 | HexDump | ok, time to leave, thanks to everybody, bye |
| 16:52.13 | *** join/#asterisk eald (n=eald@189.157.162.14) |
| 16:52.30 | iCEBrkr | hey someone call this FWD 47191. I don't even know if my Asterisk box will accept it.. |
| 16:52.37 | TaiSHi | http://pastebin.ca/378593 === http://pastebin.ca/378603 |
| 16:53.05 | [TK]D-Fender | TaiSHi: Looks like you got rid of WaitExten./ |
| 16:53.16 | TaiSHi | I haven't... it's there! |
| 16:53.27 | sfbosch | Hark, Polycom wizards! How do I make idle background images work? |
| 16:53.49 | aydiosmio | has anyone found a creative solution for storing transcriptions of custom audio prompts? like some kind of file metadata? |
| 16:53.51 | [TK]D-Fender | TaiSHi: Look at your pastebin. It obviously not getting called. Sure you RELOADed your changes? |
| 16:54.01 | TaiSHi | Ok |
| 16:54.03 | TaiSHi | [TK]D-Fender: |
| 16:54.08 | TaiSHi | Are you ready to punch me hard in the face? |
| 16:54.09 | aydiosmio | tell it like it is! |
| 16:54.14 | aydiosmio | YEAH PUNCH HIM |
| 16:54.18 | sfbosch | (Fender: I got the bloody phone working! Only worked once I replaced EVERYTHING -- bootrom.ld, sip.ld, and used the template files unadulterated) |
| 16:54.31 | iCEBrkr | aydiosmio: Lumenvox? hehe Voice recognition! |
| 16:54.33 | sfbosch | PUNCH HIM |
| 16:54.38 | TaiSHi | Check carefuly line 6 |
| 16:54.46 | ManxPower | SomethingISODD: H323 with Asterisk is one of the hardest things to install and make work |
| 16:54.56 | aydiosmio | iCEBrkr: no no, I mean documenting the audio |
| 16:55.06 | aydiosmio | so the application can be packaged and documented properly |
| 16:55.08 | [TK]D-Fender | TaiSHi: -- Executing [s@entrantes:3] Set("SIP/600-08a1b218", "TIMEOUT(response)=20") in new stack <- line 6 |
| 16:55.20 | TaiSHi | The other one |
| 16:55.20 | SomethingISODD | ManxPower, i have done it before on the older versions of asterisk, normally as long as i have openh323 and pwlib install it works no issue |
| 16:55.25 | TaiSHi | http://pastebin.ca/378593 this one |
| 16:55.25 | iCEBrkr | aydiosmio: Still not sure what you mean? |
| 16:55.26 | [TK]D-Fender | TaiSHi: Doesn't look like WaitExten.... |
| 16:55.33 | SomethingISODD | but for some reason on 1.4.0 it will nore create the chan_h323 its very allowing |
| 16:55.41 | iCEBrkr | aydiosmio: like greeting.gsm = The message that's played when a caller first calls your system? |
| 16:55.49 | TaiSHi | Yawn |
| 16:55.58 | [TK]D-Fender | TaiSHi: Yes, but your CLI output doesn't match it <- |
| 16:56.03 | SomethingISODD | nore=not |
| 16:56.08 | TaiSHi | Just check the line 6 - carefuly |
| 16:56.13 | dual-man | my php5 is working on my webserver, but i can't get it working for agi-there is no /usr/bin/php file, what can i do to fix this? |
| 16:56.15 | TaiSHi | And then you will kick me :P |
| 16:56.16 | iCEBrkr | [TK]D-Fender: I got some CLI output for ya, buddy |
| 16:56.37 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:56.38 | iCEBrkr | dual-man: how'd you install php? |
| 16:56.43 | iCEBrkr | dual-man: you need php-cli |
| 16:56.46 | aydiosmio | iCEBrkr: more like |
| 16:56.49 | dual-man | apt-get install php5 |
| 16:56.53 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 16:56.54 | aydiosmio | greeting.gsm = The message that's played when a caller first calls. "Welcome to Company, Inc. Please wait for an operator." |
| 16:56.56 | [TK]D-Fender | TaiSHi: Tell you what, put everything you want in a new pastebin and go from there... |
| 16:57.03 | iCEBrkr | dual-man: Check for a CLI php package |
| 16:57.05 | *** join/#asterisk sharp (n=sharp@2001:470:1f01:ffff:0:0:0:1c23) |
| 16:57.21 | iCEBrkr | aydiosmio: Yeah, that's going to be a lot of manual labor to document such a thing |
| 16:57.23 | aydiosmio | it's going to be a pain developign this application and keeping the audio properly documented through a ll the changes |
| 16:57.27 | SomethingISODD | ManxPower, with the steps i took do you know of anything I am missing |
| 16:57.50 | TaiSHi | Bleh |
| 16:57.54 | TaiSHi | I've put 'extem' |
| 16:57.56 | TaiSHi | Instead of exten |
| 16:58.00 | iCEBrkr | aydiosmio: Yeah, it's like my templating system I have. Each template needs to be documented how/when it should be used. PITA! |
| 16:58.08 | iCEBrkr | TaiSHi: Learn to typo, sparky! |
| 16:58.09 | aydiosmio | I mean I can listen to all the audio and write down the transcriptions, but I want a good way to store it (I wish all audio files allowed metadata) |
| 16:58.09 | iCEBrkr | err |
| 16:58.20 | TaiSHi | iCEBrkr: Got some 'clit' output for ya ;) |
| 16:58.36 | TaiSHi | Anyway, lunch time |
| 16:58.40 | TaiSHi | bbiab |
| 16:58.41 | iCEBrkr | TaiSHi: I'd be more than welcome to take that |
| 16:58.42 | iCEBrkr | lol |
| 16:58.42 | aydiosmio | like ID3 in mp3 -- you can store all the lyrics in the mp3 file |
| 16:58.57 | iCEBrkr | aydiosmio: Ahhh, that'd be the way to do it. |
| 16:59.04 | [TK]D-Fender | TaiSHi: Yeah I guess I found your point... in another way :) |
| 17:00.42 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:01.14 | *** join/#asterisk rmayorga (i=rmyorg@168.243.89.17) |
| 17:01.24 | *** join/#asterisk triple-e (n=piaergnj@71-85-147-93.dhcp.stls.mo.charter.com) |
| 17:02.23 | funxion | anyone know how I owuld go about detecting dialtone before dialing on an analog line? |
| 17:05.14 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:05.47 | EmleyMoor | I'm not sure if ChanIsAvail checks that aspect |
| 17:05.59 | EmleyMoor | There is no guarantee it won't break existing calls |
| 17:06.20 | funxion | chanisavail doesnt werk |
| 17:06.25 | funxion | I've tried it |
| 17:06.40 | funxion | was thinking about nvlinedetect |
| 17:06.47 | funxion | anyone ever used that? |
| 17:07.11 | wunderkin | that's not what that is for, it is not possible currently |
| 17:08.01 | *** join/#asterisk Strom_M (n=strom@63.110.13.126) |
| 17:08.19 | iCEBrkr | oops.. func_odbc kills asterisk |
| 17:08.23 | funxion | wunderkin what is not possible |
| 17:08.37 | funxion | to detect dialtone? |
| 17:09.41 | wunderkin | yes\ |
| 17:11.15 | funxion | on the description for nvlinedetect it says it detects dialtone |
| 17:11.32 | funxion | does it not work? |
| 17:12.08 | wunderkin | i was not aware of that |
| 17:13.11 | funxion | http://www.voip-info.org/wiki/view/NVLineDetect |
| 17:16.34 | *** part/#asterisk danp (i=danp@204.118.103.42) |
| 17:16.45 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:21.14 | *** join/#asterisk weazahl (n=weazahl@ppp-70-248-200-150.dsl.ksc2mo.swbell.net) |
| 17:23.06 | weazahl | what are peoples favorite full-featured GUIs for asterisk besides freebpx. it seems freepbx has a nagging littel problem with multiline sip clients that i cant tolerate and i would like to try something else |
| 17:23.53 | *** join/#asterisk xim010 (n=xim010@75-120-202-76.dyn.centurytel.net) |
| 17:25.19 | mafkees | vim |
| 17:27.15 | weazahl | yeah like i want to spend forever writing macros to handle dnd and cw, etc |
| 17:27.17 | sfbosch | How do I get the background idle bitmap working on Polycom phones? |
| 17:27.23 | ManxPower | weazahl: people here don't use GUIs. Perhaps you should ask on one of the asterisk gui channels (see the topic) |
| 17:27.33 | ManxPower | sfbosch: did you read the Polycom Admin Guide? |
| 17:27.40 | sfbosch | Yes |
| 17:27.58 | ManxPower | sfbosch: then you know as much about the subject as anyone else here |
| 17:28.14 | sfbosch | Manx: I have created a compliant bitmap, added the entry to the bitmap. parameter in sip.cfg, and changed the animation entry |
| 17:28.31 | sfbosch | I don't even see an idle image, let alone the one I put in :) |
| 17:28.50 | xim010 | did you reboot everybody involved. |
| 17:35.48 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 17:37.47 | *** join/#asterisk Ch0Hag (n=mking@87.127.170.250) |
| 17:38.10 | Ch0Hag | Hey that's a lot of people. |
| 17:38.10 | *** join/#asterisk fr33bi3 (n=fr33bi3@122.164.147.161) |
| 17:38.20 | Ch0Hag | Almost enough to rival #Perl or #Debian. |
| 17:39.11 | Ch0Hag | So, is there anybody (either here or, at least, generally somewhere in the world) who understands how the name/username and peer/user settings interact? |
| 17:39.37 | Qwell[] | where's [hC]... |
| 17:39.52 | CunningPike | sfbosch: Which entries in sip.cfg did you make? |
| 17:41.30 | sfbosch | CunningPike: Let me check |
| 17:42.33 | Ch0Hag | I'm trying to create a database which contains all iax/sip users/peers in one table and I want to check with someone that my understanding of these settings is correct. |
| 17:43.25 | sfbosch | CunningPike: bitmap.IP_600.82.name="logo_650", <IDLE_DISPLAY ind.anim.IP_600.38.frame.1.bitmap="logo_650" ind.anim.IP_600.38.frame.1.duration="0"/> |
| 17:44.11 | sfbosch | The phone is an IP 650 |
| 17:44.36 | CunningPike | sfbosch: OK - let me check what we did |
| 17:44.43 | sfbosch | Looking at the TFTP logs I can see the phone is loading the image |
| 17:45.05 | mercestes | Ch0Hag: this is just a guess based on contextual experience, but I believe a user "uses" the services of a server whilst a peer offers services to a user, and a "friend" is both. |
| 17:45.32 | mercestes | Ch0Hag: However, I tend to maek username and peername equivalent just to make it easy on myself. |
| 17:45.42 | Ch0Hag | Aye. |
| 17:45.56 | *** join/#asterisk fr33bi3 (n=fr33bi3@122.164.147.161) |
| 17:46.06 | Ch0Hag | I just want to confirm it with someone who knows, who isn't, like me, guessing based on not-enough experience. |
| 17:46.27 | mercestes | It likely varies by device. |
| 17:46.59 | Ch0Hag | Also unless I've got something confused, there are situations where they'd be different. |
| 17:47.17 | mercestes | Yea, I don't think a sip user/peer/friend is the same as an IAX2 user/peer/friend |
| 17:47.25 | Ch0Hag | eg. I log in to chohag@provider, so I need username=chohag and name=provider |
| 17:47.30 | *** join/#asterisk dlynes__ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 17:47.36 | Ch0Hag | mercestes: Aye that also bugs me. |
| 17:48.05 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 17:48.42 | CunningPike | sfbosch: Check your pm |
| 17:49.27 | Ch0Hag | If the wonderfully-clear voip-info.org is anything to go by, the meaning of a sip user is being phased out. |
| 17:52.42 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 17:52.47 | [TK]D-Fender | sfbosch: On a 650 screw the idle image, use the MicroBrowser Idle page. |
| 17:54.17 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 17:54.27 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 17:57.49 | sfbosch | Oh hey, fender |
| 17:58.25 | *** join/#asterisk southtel (n=slester@68-114-21-96.dhcp.gwnt.ga.charter.com) |
| 17:59.05 | sfbosch | D-Fender: MicroBrowser Idle page... I take it that's explained in the Admin guide also? :) |
| 17:59.13 | southtel | Does anyone know what would cause phones to start to "register thrash"? |
| 18:00.02 | southtel | I've got some Polycom's behind a NAT, registering to a public IP, and most of them do fine, but one of the offices' phones will occasionally just start to constantly loose registration. |
| 18:00.05 | [TK]D-Fender | sfbosch: Yes |
| 18:00.38 | [TK]D-Fender | southtel: Maye sure you have "nat=yes" and "qualify=yes" for those phones' entries in sip.conf |
| 18:01.02 | southtel | D-Fender: I do. |
| 18:01.21 | southtel | Same basic config as other offices, but this one just seems to crap out. |
| 18:01.43 | southtel | Previously, I've been able to restart the router to fix the problem, but today that didn't work. |
| 18:02.38 | southtel | I thought it was a cone/symmetric/restricted NAT problem in the router, but since that didn't work, I'm stuck. |
| 18:02.58 | [TK]D-Fender | southtel: Well if that solves the problem, then it certain identifies itself as the problem... |
| 18:03.06 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-141-152-132.pppoe.mtu-net.ru) |
| 18:03.10 | [TK]D-Fender | southtel: Time to come up with "plan B" |
| 18:03.25 | southtel | Plan B: Polish up the old resume |
| 18:04.10 | southtel | D-Fender: Thanks. I've been leaning to a plan b, but I just wanted to ping some people and see if it wasn't something staring me in the face. |
| 18:05.24 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 18:05.52 | Ch0Hag | The username field in an IAX type=peer is supposed to do something, right? |
| 18:06.07 | [TK]D-Fender | southtel: Is replacing the router that complicated? |
| 18:06.12 | sfbosch | Many thanks to CunningPike who has helped me to get my IDLE image working on a Polycom IP650! |
| 18:06.17 | sfbosch | applause |
| 18:06.18 | Ch0Hag | I have a section [bob] with type=peer and username=ben, but the host it connects to says the requested username is bob. |
| 18:06.30 | [TK]D-Fender | sfbosch: And now you can work to UNDO that and use the MicroBrowser for that FTW! |
| 18:06.40 | CunningPike | [TK]D-Fender: heh |
| 18:06.42 | [TK]D-Fender | I have live Queue stats for my CSR's on my IP 6XX's here |
| 18:06.51 | CunningPike | [TK]D-Fender: Nice |
| 18:06.54 | sfbosch | very cool |
| 18:06.57 | [TK]D-Fender | And could add live weather and other useful content |
| 18:07.07 | sfbosch | weather, shmeather |
| 18:07.10 | CunningPike | Live weather is useful? ;) |
| 18:07.18 | sfbosch | I use my Window Browser |
| 18:07.22 | [TK]D-Fender | CunningPike: I don't have a window :( |
| 18:07.30 | sfbosch | you poor soul |
| 18:07.36 | Ch0Hag | Lag 237 -- I see freenode's still as great as ever. |
| 18:07.38 | CunningPike | Poor [TK]D-Fender |
| 18:07.38 | [TK]D-Fender | <- got the "closet" as an office in his co's last move |
| 18:07.41 | Waverly360 | Whoa...live weather on my phone? Management would love that here... |
| 18:08.03 | sfbosch | How do I format the microbrowser page? |
| 18:08.15 | *** join/#asterisk tonycarstens (n=tonycars@209.101.229.196) |
| 18:08.21 | sfbosch | is there a template I can get somewhere? |
| 18:08.26 | ManxPower | Ch0Hag: paste your Dial(... line |
| 18:09.05 | ManxPower | Ch0Hag: I assume you are using IAX. |
| 18:09.09 | Ch0Hag | Um. Actually it's 'Dial(${${ARG1}_MODULE}/${${ARG1}_PEER}|15|g)' |
| 18:09.23 | ManxPower | Ch0Hag: OK, paste the CLI output of that dial line. |
| 18:09.58 | Ch0Hag | <PROTECTED> |
| 18:10.04 | Ch0Hag | <PROTECTED> |
| 18:10.25 | Ch0Hag | Should it be @bob? |
| 18:10.28 | Waverly360 | Anyone know what it means, when I'm trying to connect to IAXModem with cu, and asterisk gives me a "Rejected connect attemp from 127.0.0.1, request 's@phones' does not exist?" |
| 18:10.36 | ManxPower | Ch0Hag: how about Dial(IAX2/ben@bob,15,g) |
| 18:10.37 | Ch0Hag | I thought the @ was unnecessary if no explicit username was supplied. |
| 18:11.02 | sfbosch | D-Fender: The admin guide seems short on details for Microbrowser stuff |
| 18:11.06 | sfbosch | it just shows how to call one |
| 18:11.06 | Ch0Hag | OK that works. |
| 18:11.15 | Ch0Hag | But I don't want to have to put the username in the Dial string. |
| 18:11.15 | ManxPower | Ch0Hag: there is what is supposed to happen and there is what really happens |
| 18:11.32 | Ch0Hag | Manx: Yes I have become aware of the discrepancy after using * for a while. |
| 18:11.38 | Ch0Hag | :) |
| 18:14.06 | ManxPower | Ch0Hag: paste the [bob] section of iax.conf on pastebin.ca |
| 18:14.22 | Ch0Hag | OK |
| 18:14.30 | ManxPower | Ch0Hag: and you have only ONE [bob] section, correct? |
| 18:14.33 | Ch0Hag | But last time I used pastebin for something it crashed. |
| 18:14.45 | Ch0Hag | That is correct. |
| 18:16.11 | Ch0Hag | http://pastebin.ca/378695 |
| 18:17.18 | ManxPower | Ch0Hag: I have no more ideas |
| 18:18.34 | EmleyMoor | Is there a variable equal to the current channel's default context? |
| 18:19.10 | Ch0Hag | So is this a bug? |
| 18:19.35 | Ch0Hag | It would seem illogical to default the username to be equal to the peername but that appears to be what happens. |
| 18:20.32 | [TK]D-Fender | sfbosch: Just shove a page in there and an idel refresh timeout and you're in business. You can even use your web-server to host that image in the mean-time |
| 18:21.03 | Ch0Hag | Wtf? |
| 18:21.08 | Ch0Hag | Now it's stopped happening... |
| 18:21.32 | Ch0Hag | Now the debug trace on the destination show the username is passed as ben. |
| 18:21.59 | sfbosch | D-Fender: as a GIF or BMP? |
| 18:22.04 | ManxPower | maybe you forgot a reload |
| 18:22.09 | Ch0Hag | Does * sit around for a while doing bugger all after reload is issued before it actually decides to use the values it's reloaded? |
| 18:22.21 | sfbosch | D-Fender: is there a sample page somewhere I can check out, just for formatting purposes? |
| 18:22.55 | Corydon-w | Ch0Hag: no, but it does reload each module in order. If any module hangs on reload, each subsequent module reload will be delayed |
| 18:23.13 | Ch0Hag | That's not a (the) problem. |
| 18:23.16 | Ch0Hag | I have verbose == 3 |
| 18:23.35 | Ch0Hag | Maybe I did forget reload but I don't think so. |
| 18:24.08 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 18:25.36 | sfbosch | D-Fender: Okay, I see the information about this in voip-info.org |
| 18:26.43 | [TK]D-Fender | sfbosch: I suggest BMP 4-bit as per spec. |
| 18:27.10 | [TK]D-Fender | sfbosch: Make a BARE XHTML page to start with "hello world" and go from there. |
| 18:27.20 | [TK]D-Fender | sfbosch: Check it out on the WIKI. |
| 18:27.35 | sfbosch | D-Fender: done and done. Thanks for the help! |
| 18:28.35 | [TK]D-Fender | sfbosch: You're going to have a LOT of fun with that phone... |
| 18:29.23 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 18:30.38 | RoyK | ~seen wasim |
| 18:31.06 | jbot | wasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 66d 11h 55m 9s ago, saying: 'hmm .. poor england'. |
| 18:35.34 | *** join/#asterisk DaveCanoe (n=Dave@canoe.wbb.net.cable.rogers.com) |
| 18:35.43 | *** join/#asterisk fr33bi3 (n=fr33bi3@122.164.141.192) |
| 18:36.33 | type0 | What would the equivelent of HP Openview be in Solaris? |
| 18:36.55 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:37.06 | xibalba | what is openview |
| 18:37.26 | *** join/#asterisk fr33bi3 (n=fr33bi3@122.164.141.192) |
| 18:38.08 | ManxPower | type0: you might consider asking that question on a channel where is is not totally off topic |
| 18:39.04 | J4k3 | hp openview.. when the only thing "open" about it is your wallet. |
| 18:39.15 | type0 | hence, my question jake |
| 18:39.17 | type0 | haha |
| 18:39.29 | *** join/#asterisk bkw_ (n=brian@m010f36d0.tmodns.net) |
| 18:40.07 | *** join/#asterisk hellojoe (n=hijoe@natint3.juniper.net) |
| 18:40.56 | cpm | J4k3, what are you doing slumming down here? |
| 18:41.28 | *** join/#asterisk Opperior (n=chatzill@24.61.165.73) |
| 18:41.46 | type0 | i was looking at mirc.com the other day |
| 18:42.34 | type0 | he donated all the profits of registration for a month |
| 18:42.37 | type0 | and it was like 300,000$ |
| 18:42.49 | type0 | I bet registrations makes him at least 150k a year |
| 18:43.52 | Waverly360 | Anyone know what would cause my IAXModem to return a no dial tone? |
| 18:44.45 | hellojoe | Guys, I am using a Dial command patching to a macro (call screen). Is there a way to set some variable and inherit it in the called context? |
| 18:45.14 | hellojoe | There seems to be no way of setting a variable in the macro (Dial) and get the values back in the calling context |
| 18:45.24 | ManxPower | hellojoe: see README.variables Pay special attention to the variable MACRO_CONTEXT |
| 18:45.47 | hellojoe | thanks ManxPower |
| 18:45.50 | ManxPower | hellojoe: by default macros should be seen by the calling channel |
| 18:46.16 | hellojoe | no, that's NOT true for the Dial(Macro) |
| 18:46.25 | hellojoe | its true for the macros otherwise |
| 18:46.33 | hellojoe | should I try : exten => s,n,Set(${ARG1}=$[${${ARG1}} + 1]) |
| 18:46.50 | hellojoe | perhaps it will set the value of the ARG that I pass to the macro |
| 18:47.05 | ManxPower | hellojoe: you mean M(whatever) on the Dial line, right? |
| 18:47.27 | hellojoe | right |
| 18:47.38 | ManxPower | hellojoe: try Set(MY_VAR=BARNEY). Keep things simple when debugging |
| 18:47.55 | hellojoe | exten => ${FWD-HOME-NUMBER},n,Dial(${PHONE-1}|${INCOMING-RING-SECONDS}|gmM(from-iax-fwd-screen^${SCREEN-FILE}^${UNIQUEID}^${CALLERIDNUM})) |
| 18:48.10 | hellojoe | MY_VAR is not reported back to the calling context, tried that :-( |
| 18:48.31 | *** join/#asterisk drhbn (n=joseph@S01060050da7ae68c.ed.shawcable.net) |
| 18:48.45 | ManxPower | hellojoe: M() is a lot less useful than it seems. I found that out the hardway. |
| 18:48.50 | *** part/#asterisk drhbn (n=joseph@S01060050da7ae68c.ed.shawcable.net) |
| 18:49.05 | hellojoe | yeah.... |
| 18:49.20 | hellojoe | I am hoping exten => s,n,Set(${ARG1}=$[${${ARG1}} + 1]) does the trick |
| 18:49.23 | ManxPower | I discovered that you can't accept more than 1 DTMF inside of a M() |
| 18:49.24 | hellojoe | i am trying it out right now |
| 18:49.45 | hellojoe | I will post the results ... stay tuned if you are interested |
| 18:51.24 | *** join/#asterisk drhbn (n=joseph@S01060050da7ae68c.ed.shawcable.net) |
| 18:53.56 | drhbn | How to configure Digium S101i adapter to work with "pickupexten *8" ? |
| 18:53.57 | drhbn | I have few Sipura adapters and "*8" work OK but my new Digium S101i |
| 18:53.57 | drhbn | refuses to cooperate. |
| 18:54.46 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 19:01.43 | ManxPower | drhbn: No configuration is required. IT should work automatically. |
| 19:02.08 | ManxPower | Your issue is not with the IAXy, but somewhere else. For example are you SIPuras in a different context than your IAXy? |
| 19:02.41 | ManxPower | Did you put the correct pickupgroup and callgroup in iax.conf? |
| 19:04.11 | file | chan_iax2 doesn't support that stuff |
| 19:04.20 | ManxPower | file: Ah! |
| 19:04.25 | ManxPower | drhbn: you are out of luck then |
| 19:04.52 | *** join/#asterisk andyjh (n=harrella@72-164-254-211.dia.static.qwest.net) |
| 19:05.30 | ManxPower | file: could he emulate it via features.conf? |
| 19:05.47 | file | I do not think so |
| 19:06.41 | ManxPower | Hoe about exten => *8,1,Pickup |
| 19:07.05 | file | Pickup is a different application |
| 19:07.45 | ManxPower | I'll bet drhbn is another person that will only ever buy 1 IAXy 8-| |
| 19:10.53 | EmleyMoor | Am I still connected here? |
| 19:12.07 | J4k3 | no |
| 19:12.21 | J4k3 | you see that guy running away from your house with wire cutters? blame him! |
| 19:12.44 | EmleyMoor | J4k3: Actually, I think the fault is about 15 miles SE of here |
| 19:12.55 | J4k3 | EmleyMoor: ugh, fiber? |
| 19:13.34 | drhbn | callgroup=1 |
| 19:13.34 | drhbn | pickupgroup=1 |
| 19:13.35 | EmleyMoor | No - just lost connectivity between the ISP's equipment and a machine in their office |
| 19:14.37 | drhbn | all context iax.conf and sip.conf have callgroup=1 and pickupgroup=1 |
| 19:15.00 | ManxPower | drhbn: did not you see file's comment that IAX does not support those features? |
| 19:15.53 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:16.55 | *** join/#asterisk rat1101 (n=jircii@c-68-57-135-14.hsd1.va.comcast.net) |
| 19:17.09 | mendol | how can I add pstn number to user extension so he will present with this pstn number when he calls? |
| 19:21.19 | sivana | can anyone confirm if Digium hardware works in a Mac? |
| 19:22.31 | russellb | assuming it's a mac running linux |
| 19:22.32 | russellb | then yes |
| 19:22.37 | drhbn | I was looking at: http://www.voip-info.org/wiki/view/Asterisk+PBX+functions and general support (for all channels) have Call Pickup: Supported |
| 19:22.42 | sivana | russellb: macos |
| 19:22.46 | russellb | then no. |
| 19:22.51 | sivana | shoots |
| 19:22.51 | J4k3 | mac is either a standard pc with a crappy drm chip, or a ppc. |
| 19:23.59 | *** join/#asterisk deb_user (n=Hypnotis@albuquerque.agroinnovations.com) |
| 19:24.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:25.21 | deb_user | i have a problem...all of a sudden incoming sip calls aren't routing properly to my machine |
| 19:25.39 | Ch0Hag | VoIP is incredibly hard to debug with one handset. |
| 19:26.21 | Ch0Hag | Does anybody fancy some reading? |
| 19:26.32 | Ch0Hag | I want a second opinion on a basic SIP explanation. |
| 19:27.01 | sivana | J4k3: I've never heard Apple say it was open |
| 19:27.33 | *** join/#asterisk ZaVoid (n=colin@c-67-165-25-195.hsd1.ct.comcast.net) |
| 19:27.52 | Opperior | Is a Mac running Linux still a Mac? hmm... |
| 19:27.58 | ManxPower | drhbn: According to "file", who actually works with the Asteirsk source code, IAX2 does not support call pickup groups |
| 19:28.43 | ZaVoid | sup all |
| 19:28.54 | Ch0Hag | Rainclouds. |
| 19:32.02 | sfbosch | This IP 650 is driving me crazy! I have this in my phone's digit map: [2-9]11|[2-9]xx|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT |
| 19:32.12 | sfbosch | but I cannot make calls to 3 digit extensions at all |
| 19:32.21 | sfbosch | the phone prompts for more digits |
| 19:32.23 | drhbn | Thank ManxPower: so in this respect SIP is better than IAX2 |
| 19:32.35 | ZaVoid | i'm getting like 5-10 seconds extra duration after my asterisk box gets the BYE.... its strange |
| 19:33.13 | sfbosch | I have also tried [2-9]xxT |
| 19:33.19 | [TK]D-Fender | sfbosch: Screw that : x.T|*.T|#.T <- the only dialplan you'll ever need |
| 19:33.27 | sfbosch | lol |
| 19:33.49 | [TK]D-Fender | sfbosch: And make sure you set the "remoeendOfDial="0"" and "impossiblematchhandling="2" |
| 19:33.50 | ManxPower | sfbosch: the call is rejected? Or do you have to press "SEND" |
| 19:34.03 | sfbosch | It does the same thing whether I press send or not |
| 19:34.13 | ManxPower | sfbosch: and what is that? |
| 19:34.21 | sfbosch | "Enter more digits" |
| 19:34.42 | ManxPower | sfbosch: [TK]D-Fender's suggest config and mine are the opposite. |
| 19:34.54 | ManxPower | I suggest digit impossible match of 1 |
| 19:34.59 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 19:35.07 | sfbosch | It's currently 0, which is the default |
| 19:35.11 | sfbosch | I will try 1 |
| 19:35.22 | ManxPower | sfbosch: and you have an exten => 211,1,Whatever, of course? |
| 19:35.27 | [TK]D-Fender | ManxPower: My style is "STFU and take whatever I give you and let ASTERISK deal with it." |
| 19:35.36 | sfbosch | lol |
| 19:35.38 | Waverly360 | Are there any IAXmodem gurus in the house? |
| 19:35.47 | hellojoe | ManxPower: The experiment failed |
| 19:35.47 | ManxPower | [TK]D-Fender: my style is to not require a timeout and not require the user to press SEND |
| 19:35.49 | EmleyMoor | Something funny has happened to my laptop - I will have to reboot it |
| 19:35.56 | sfbosch | ManxPower: I have 3 digit extensions beginning with 2, yes |
| 19:36.10 | ManxPower | sfbosch: what is the actual exten line for that? |
| 19:36.10 | hellojoe | there is no way to figging set a variable in Dial Macro such that the variable is exposed to the dialing party |
| 19:36.35 | hellojoe | i meant dialing context |
| 19:36.56 | [TK]D-Fender | ManxPower: Sure,you can do that, but it usually leads to things like dialing prefixes for outbound access, fixed * dialplan options a single entries, etc... I like the "KISS" theory. |
| 19:37.19 | type0 | Sun Microsystems has the best customer support I've ever had |
| 19:37.29 | EmleyMoor | Does 270 extensions, 812 priorities, 55 contexts sound too outrageous for a dialplan? |
| 19:37.35 | ManxPower | As I said, [TK]D-Fender's dial plan design is the exact opposite of mine. |
| 19:37.36 | Ch0Hag | I like the KISS practice better than the theory. |
| 19:37.37 | type0 | 1500 dollar order and these people are jumping through 1000 hoops an hour |
| 19:38.02 | [TK]D-Fender | EmleyMoor: Depends what you need it to do. |
| 19:38.26 | [TK]D-Fender | EmleyMoor: I could easily say its reasonable, but I'd have to see YOUR case to judge it. |
| 19:38.33 | EmleyMoor | Well, as long as it doesn't sound generally excessive, that's all |
| 19:38.51 | sfbosch | ManxPower: The extension I am trying to call is "200" |
| 19:38.54 | EmleyMoor | I will have to reboot the laptop now... |
| 19:39.02 | ManxPower | My dialplan is really quite simple. extensions are 4 digits and start with 2-7. dialing local is 9 + 7-digits, dialing long distance is 9+1+10 digits, emergency is 911 or 9911, star codes are *xx |
| 19:39.14 | ManxPower | sfbosch: I asked you to PASTE it |
| 19:39.16 | Ch0Hag | If you register (SIP) to a remote host, whether or not they attempt to authenticate incoming calls is entirely up to them, right? |
| 19:39.37 | sfbosch | ManxPower: sorry, that wasn't clear |
| 19:39.41 | ManxPower | Ch0Hag: yes, but they usually will |
| 19:39.46 | Ch0Hag | ie. if their system places calls anonymously, the only way to change that is to get them to change policy. |
| 19:39.51 | sfbosch | ManxPower: what do you want me to paste? |
| 19:40.09 | ManxPower | the like for extension 200 from extensions.conf. Just the first line should be fine. |
| 19:40.14 | Ch0Hag | manx: They will because they will set the system up to do so, or they will because the system will do so on its own? |
| 19:40.18 | sfbosch | okay |
| 19:40.19 | ManxPower | like == line |
| 19:40.53 | Ch0Hag | I ask because I don't like things like 'insecure=very' on my servers. |
| 19:40.55 | ManxPower | Ch0Hag: sip does not require authentication |
| 19:41.10 | Ch0Hag | (Or insecure=invite as it is now) |
| 19:41.14 | ManxPower | Ch0Hag: I don't use ITSPs that require insecure=very |
| 19:41.20 | ManxPower | They do not get my business |
| 19:41.49 | Ch0Hag | I see. |
| 19:42.21 | ManxPower | Ch0Hag: generally ITSPs that require insecure=very send the SIP userid as the callerid of the caller. |
| 19:42.48 | ManxPower | i.e. if I place a call to you from telephone number 5045551212 then the itsp will send the userid as 5045551212 |
| 19:43.52 | sivana | can the * appliance support 8 fxo? |
| 19:44.09 | ManxPower | "appliance"? |
| 19:44.16 | sfbosch | ManxPower: I am ashamed -- I see why it wasn't working now |
| 19:44.24 | sfbosch | ManxPower: It helps if the extension actually exists |
| 19:44.28 | sivana | ManxPower: aadk |
| 19:44.46 | ManxPower | sfbosch: My trouble shooting methods can be pretty sneaky at times. |
| 19:44.48 | Ch0Hag | And insecure=invite? |
| 19:44.53 | Ch0Hag | Do you avoid those providers too? |
| 19:45.02 | ManxPower | sivana: perhaps you can contact the vendor and ask them. |
| 19:45.03 | sfbosch | ManxPower: But the behaviour is different from my 501 when I dial an extension that doesn't exist, where it just fails silently |
| 19:45.07 | ManxPower | Ch0Hag: correct. |
| 19:45.07 | Opperior | sivana: I have used 2 4-port fxo cards, though it is not recommended. Digium has a 24-port card, though |
| 19:45.22 | ManxPower | sfbosch: that depends on the digit impossible match setting |
| 19:45.24 | sivana | Opperior: you used it with their appliance? |
| 19:45.34 | sfbosch | ManxPower: ah |
| 19:45.52 | Opperior | oh, you're talking about the pre-built asterisk box. No, never used that |
| 19:45.58 | ManxPower | sivana: I strongly doubt anyone here knows what an aadk is |
| 19:46.07 | wunderkin | heh |
| 19:46.08 | sivana | ManxPower: http://www.digium.com/en/products/hardware/aadk.php |
| 19:46.12 | J4k3 | is there any law (in the USA) about sending completely false Caller ID info? ;) |
| 19:46.14 | wunderkin | aardvark? |
| 19:46.18 | ManxPower | Asterisk At Dutch Komrads |
| 19:46.24 | ManxPower | sivana: ASK DIGIUM! |
| 19:46.40 | sivana | ManxPower: :) |
| 19:47.00 | ManxPower | J4k3: I don't believe so, but there are laws against fraud |
| 19:47.06 | cpm | J4k3, I think so |
| 19:47.13 | sivana | ManxPower: you didn't seem to know what it was, I was just FYI you |
| 19:47.21 | Opperior | ok, so that 1 yeas and 1 no. Anyone else? |
| 19:47.22 | cpm | however, if it's your number, it's not false, it's kinda wierd, |
| 19:47.26 | sivana | clearly you don't know the answer, however there are Digium folks here. :P |
| 19:47.33 | ManxPower | sivana: it is best to ask the vendor about their own hardware. |
| 19:47.36 | cpm | something to do with that telecommunications act from the 80s |
| 19:47.40 | sivana | you are correct :) |
| 19:47.46 | sfbosch | On the TDM-400 cards -- is there a way to make the echo cancellation converge more quickly? It's not bad now, but when you answer the phone you get considerable echo, and it's distracting |
| 19:47.46 | cpm | that old |
| 19:47.52 | wunderkin | Opperior, maybe? |
| 19:48.08 | sfbosch | I would say it takes about 3 - 4 seconds before the echo is gone, but if you're answering the phone a lot, it's bothersome |
| 19:48.17 | Opperior | no fence riding |
| 19:48.18 | ManxPower | sfbosch: see fxotune |
| 19:48.26 | sfbosch | Ah! Thank you |
| 19:48.34 | cpm | meaning, you can set up a number of fax lines, and set the ANI to reflect your inbound fax, so that everything is still *you* even though the true ANI is incorrect |
| 19:48.35 | sfbosch | (groan -- too much to learn) |
| 19:49.04 | BrianR___ | Is there any way to get the HDLC statistics from a Sangoma A104D? I think I may be getting CRC errors but don't know where they would be logged. |
| 19:49.30 | cpm | but, if you are setting the ANI to someone you hate, and then calling up the cop-shop and mouthing off to the officer on duty, that's a no no |
| 19:49.44 | ManxPower | cpm: I think that would fall under fraud |
| 19:50.00 | techie | highly illegal I'd say |
| 19:50.05 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a7430e722a02e26d) |
| 19:50.07 | *** join/#asterisk foobar778 (i=johhny@ip68-100-210-15.dc.dc.cox.net) |
| 19:50.14 | ManxPower | or whatever the same thing is in the analog world. i.e. mouthing off at the police and have a fake id of the person you don't like. |
| 19:50.15 | cpm | ManxPower, true, however I am really certain, that there is something specific to ANI/CallerID about that |
| 19:50.37 | J4k3 | well... I set a bunch of extensions to an entirely false number (NPA-555-1234) basically because the extensions *have* no real number. |
| 19:50.39 | casey | is there a way to make penalty not dynamically change based on calls answered? |
| 19:50.46 | cpm | That's a no no |
| 19:50.49 | Opperior | besides, iirc, police, 911 and other such services do not rely on caller id for just that reason |
| 19:50.57 | ManxPower | J4k3: better to set the callerid to be that of the main phone number for that office. |
| 19:51.01 | foobar778 | [TK]D-Fender; u there |
| 19:51.05 | cpm | what ManxPower said |
| 19:51.25 | BrianR___ | The PSAP here appears to be using callerid, at least when deciding what number to call back. |
| 19:51.41 | sfbosch | Sounds like cpm is setting up a fax broadcaster to flood us all with junk |
| 19:51.48 | BrianR___ | Our office creates a lot of 911 hangups :( |
| 19:51.50 | sfbosch | :) |
| 19:51.54 | ManxPower | That's what we do. SetVar=DID=<main number of office> and then my outbound dialing macro will check to see if that variable exists and do what it needs to do |
| 19:52.18 | foobar778 | flash operator panel anyone use it having issue transferring calls |
| 19:52.21 | ManxPower | BrianR___: put in a Wait(1.5) before dialing 911 |
| 19:52.43 | BrianR___ | ManxPower: I'm tempted. Our lawyer says not to. |
| 19:52.51 | Bobthehunter | anyway to recompile asterisk only. .make bin only ? |
| 19:53.08 | BrianR___ | Just getting rid of the rest of the key system telephones so I can remove the "9" prefix for outbound calls. |
| 19:53.10 | J4k3 | so |
| 19:53.22 | J4k3 | double route the 911 calls or something... have them transfer to your boss's desk *AND* 911 simultaniously. |
| 19:53.22 | *** join/#asterisk diamon (n=diamon@c-66-176-91-114.hsd1.fl.comcast.net) |
| 19:53.24 | J4k3 | ;) |
| 19:53.33 | BrianR___ | We also have extensions that start with "11" |
| 19:53.37 | J4k3 | eww |
| 19:53.48 | ManxPower | BrianR___: well THAT was poor planning |
| 19:53.54 | BrianR___ | J4k3: It now runs a script that pages reception and IT whenever someone dials 911... Got tired of the police showing up and walking around the whole office. |
| 19:53.56 | Bobthehunter | at least its not 011 |
| 19:53.57 | Bobthehunter | lol |
| 19:54.00 | BrianR___ | ManxPower: Previous IT guy wasn't a big thinker... |
| 19:54.00 | sfbosch | ManxPower: would you show me your Polycom digitmap? |
| 19:54.29 | ManxPower | sfbosch: http://www.fnords.org/~eric/polycom-config-examples/ |
| 19:54.44 | BrianR___ | When I first became responsible for phones, I discovered that 911 was actually routed out an OUTWATS T1... |
| 19:54.44 | sfbosch | ManxPower: gracias |
| 19:54.56 | BrianR___ | I don't think it actually worked at all. |
| 19:55.17 | type0 | why is your office calling 911 so much? |
| 19:55.42 | BrianR___ | type0: morons are careless, we have a '9' prefix for outside lines, about 100 extensions that start with "11"... |
| 19:56.28 | BrianR___ | With 9,1NXXXXXXXX dialing when missing the second digit of the area code there's a high probability (nearly 40%) that the third digit will be a 1, causing a 911 call too. |
| 19:57.00 | Bobthehunter | coz make install copies the sounds files over and over again.. too slow |
| 19:57.12 | Bobthehunter | so is there a bin only method ? |
| 19:57.17 | type0 | You could redirect 911 to an extension, "If this is an emergency, please press 1. Otherwise, get your life together" |
| 19:57.20 | BrianR___ | Anyway.. DOes someone here know how to look at hdlc stats? |
| 19:57.35 | BrianR___ | type0: Corporate counsel says that's a liability no-no :( |
| 19:57.43 | ManxPower | playback(now-dialing-911); wait(1) |
| 19:58.02 | type0 | umm, how would it not be a liability to be calling the PSAP 10 times a day |
| 19:58.12 | ManxPower | If you are on a PRI, that is STILL less delay than an analog system dialing 911 |
| 19:58.31 | BrianR___ | ManxPower: Probably true... |
| 19:59.13 | type0 | that's a good idea |
| 19:59.22 | type0 | "Now connecting you to Emergency Services, please wait" |
| 19:59.51 | type0 | I dont think that would be illegal or a liability |
| 19:59.57 | diamon | Trick will be in getting someone to sign off on that, in case in a real emergency those seconds count... |
| 20:00.13 | BrianR___ | We probably have 2 or 3 911 hangups a week... |
| 20:00.41 | type0 | at some point, they're getting a call from the city |
| 20:00.48 | BrianR___ | It's usually already ringing at the PSAP before the caller realizes they've fucked up. |
| 20:00.53 | Ch0Hag | Whoever started the trend of '9' for an outside line should probably be murdered in his bed. |
| 20:00.55 | type0 | I think that'll be the kick in the ass for them |
| 20:01.03 | Ch0Hag | And if he's already dead, exhumed and put to bed first. |
| 20:01.08 | BrianR___ | type0: Yeah.. The psap director has already called a few times :( |
| 20:01.14 | ManxPower | Ch0Hag: not at all. |
| 20:01.26 | type0 | well.. let legal know they're calling |
| 20:01.39 | ManxPower | First of all, dialing 9 for an outside line was around LONG before 911 was around |
| 20:01.41 | type0 | make something up like they're going to make a police substation in your office if you dont quit calling so much |
| 20:02.00 | ManxPower | so if you are going to kill someone, kill the person that picked 911 for emergency services |
| 20:02.17 | Ch0Hag | Oh he's up for it too. |
| 20:02.18 | BrianR___ | The biggest problem is when people accidentally dial and then don't stay on the line... |
| 20:02.22 | diamon | I dropped the 9 for an outside line. If they're not dialling an extension, it's got to be outside. I left 9 in there so if they do it there's no error, but I let my shop direct-dial. |
| 20:02.23 | Ch0Hag | 999 is way better. |
| 20:02.40 | BrianR___ | "112" is probably best. |
| 20:02.41 | Ch0Hag | At least - now that we no longer have rotary-dial phones, it is. |
| 20:02.46 | ManxPower | my users almost never accidently dial 011 |
| 20:02.46 | type0 | little kids press buttons on phones too much |
| 20:02.49 | type0 | for 999 |
| 20:02.53 | type0 | I have the number 777-7777 |
| 20:02.56 | Opperior | I think it started with rotery phone. Something to do with making it fast to dial but hard to do by accident |
| 20:03.01 | type0 | I have the entire 7XXX block |
| 20:03.09 | type0 | you wouldnt believe the amount of kids that call that |
| 20:03.13 | type0 | "HI |
| 20:03.26 | type0 | or people calling to see if it was a real number |
| 20:03.41 | Ch0Hag | Put it through to the telemarketer torture script. |
| 20:03.44 | Opperior | like people who call 666-6666 to see if the devil answers :P |
| 20:03.58 | type0 | I dont think they issue 666 in any area code |
| 20:03.59 | Ch0Hag | Opperior: Oh now that's a perfect candidate for the Evil Menu System. |
| 20:04.07 | type0 | just like there's not a 13th floor in most places |
| 20:04.25 | Opperior | actually, 666 is an exchange in New Hampshire |
| 20:04.29 | cpm | sfbosch, all my departments have outbound faxes, but the ani is set back to the main fax number, so that in bound goes through distro, like it's supposed to. |
| 20:04.33 | type0 | who has 6666 |
| 20:04.47 | diamon | Speaking of my shop, for some reason I cannot get my Cisco 7940/60 phones to light the message waiting indicator on them. I have the extensions using ext#@default (and had previously thought @device was the problem), but a restart of the phone and a few other checks shows no indicator. Anything silly I should look at offhand? |
| 20:04.59 | type0 | I bet it rings to nowhere |
| 20:05.01 | type0 | people are calling like |
| 20:05.06 | type0 | "YOU ARE THE TELCO ANTICHRIST" |
| 20:05.08 | type0 | REPENT! |
| 20:05.45 | type0 | In other news, 5E's are really nice looking |
| 20:05.54 | type0 | I was in my cellular providers switch room yesterday |
| 20:06.06 | type0 | how many voice circuits can you put on an OC-48 |
| 20:06.09 | diamon | I'm probably doing something silly, but who knows. * 1.2.12 on a CentOS 4 box. |
| 20:06.14 | type0 | cause I'm pretty sure its more than the entire population of alaska |
| 20:06.31 | diamon | type0: OC48? Assuming 64kbit/channel? |
| 20:06.39 | type0 | yeah |
| 20:07.14 | Ifaistos | Does anyone have any experience in g729 protocol ? |
| 20:08.08 | type0 | A telco/isp here called GCI just bought the cellular provider they were leasing their PCS license to |
| 20:08.17 | type0 | so naturally, they run a oc-48 to the 5E |
| 20:08.19 | diamon | type0: *IF* I did my math right, it's a 2.5Gb channel (assuming perfection), and could do just under 42k channels. |
| 20:08.52 | wunderkin | not enough, and it needs to be done yesterday |
| 20:08.55 | CunningPike | BrianR___: Tell your lawyers to get a clue - that's exactly what we do (we play a message while they're paused) and we're a local govt |
| 20:09.04 | type0 | 42,000 concurrent channels with 10% use would translate to 420,000 |
| 20:09.24 | Opperior | what would *you* do if you had 420,000 channels |
| 20:09.25 | CunningPike | BrianR___: If the 2 seconds is the difference between life and death, they're gonners anyway |
| 20:09.32 | wunderkin | heh |
| 20:09.35 | Qwell[] | Opperior: I'd mux it as data, and download pr0n |
| 20:09.35 | type0 | I would make a giant conference |
| 20:09.40 | Qwell[] | That's just me though |
| 20:09.42 | *** join/#asterisk koos (n=koos@e176185126.adsl.alicedsl.de) |
| 20:10.00 | type0 | speaking of multiplexors |
| 20:10.02 | type0 | I hate coastcom |
| 20:10.16 | type0 | biggest piece of shit mux on the planet. |
| 20:10.49 | BrianR___ | CunningPike: I'd rather be prepared than count on 911 for anything that's time-critical, but sheeple are not so bright :( |
| 20:11.29 | BrianR___ | Anyway, getting rid of the stupid 9 prefix should eliminate 90% of our 911 hangup calls. |
| 20:12.07 | Opperior | at least once the users are broken of the habbit of dialing 9 |
| 20:12.34 | BrianR___ | I was thinking of having a 9[023456789] rule in the dialplan that does something like milliwatt() |
| 20:12.39 | BrianR___ | or maybe megawatt() :) |
| 20:12.49 | diamon | BrianR: Sheeple... Wow, there's a word I haven't heard in a bit. :) |
| 20:12.50 | wunderkin | gigawatt |
| 20:13.20 | wunderkin | sheep people? |
| 20:13.35 | ZaVoid | meh so anyone have any idea what would cause an answer super vicsion screw upin 1 direction? a calls b... a hangs up.. its fine... if b hangs up.. a doesn't get supervsion for 10 seconds... a is a client.. in the middle asterisk gets the bye |
| 20:13.45 | BrianR___ | Although I'm sure the lawyers should be complaining about ear damage from handset shrapnel, etc. |
| 20:14.07 | diamon | Yep. Sheeple == sheeplike people. Silly folks who follow the herd and don't ask questions. |
| 20:14.22 | *** join/#asterisk gatuno (n=gatuno@230.red-82-158-212.user.auna.net) |
| 20:14.23 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 20:14.28 | CunningPike | Baaaaaaaa |
| 20:14.53 | vooduhal | Are these delay times average on a local lan segment if the phone data is the only traffic on it? http://www.pastebin.ca/378804 |
| 20:15.22 | type0 | how often does google update their google earth maps? |
| 20:15.30 | ZaVoid | often |
| 20:15.38 | Qwell[] | whenever they get new data |
| 20:15.54 | wunderkin | google maps is the most up to date for phoenix that i've ever seen.. |
| 20:16.24 | type0 | not for alaska.. haha |
| 20:16.30 | type0 | I still see a blank mountain with nothing on it |
| 20:16.31 | type0 | .. |
| 20:16.46 | diamon | vooduha1: I find my phones don't have the brightest TCP stacks ever. It works mostly, but the latency does tend to vary. Let me see what mine say. |
| 20:16.50 | Qwell[] | no, that's correct |
| 20:16.57 | Qwell[] | that's pretty much all alaska is :p |
| 20:17.00 | BrianR___ | Google maps is OK around here. Sometimes the M$ live local thing has higher resolution maps, but the google ones are often newer. |
| 20:17.06 | vooduhal | Those are all Polycom 501s. |
| 20:17.32 | BrianR___ | The 501's stack is mediocre. If there's a lot of junk traffic showing up on the RTP port it can fall over... |
| 20:17.33 | vooduhal | And could that be the cause of high amounts of echo? |
| 20:17.35 | wunderkin | vooduhal, yeah i could guess without even looking at it.. those are qualify times... it is a sip packet back and forth.. not an icmp ping |
| 20:17.47 | BrianR___ | vooduhal: Higher latency will make echo seem worse. |
| 20:18.41 | diamon | vooduha1: Echo I've always found to be the PSTN device's fault. Skip, scintillation, and dropped sounds is usually the phone of * server. |
| 20:18.52 | BrianR___ | If you expect call quality on par with regular pbx phones the polycom ones need to go on their own vlan. |
| 20:19.15 | mitcheloc | how does ms get better resolution maps then google? i think they are pulling strings with the government |
| 20:19.21 | vooduhal | Well, we've never had any real echo off the A104D, so that's why I'm wondering. |
| 20:19.42 | diamon | BrianR: How's that? What do they do, or what do they have a problem with? I've got a polycom conference phone. No problem yet, but I'd be happy to head any off before I get a late-night call... |
| 20:20.35 | ManxPower | echo is caused by the far end analog loop. Not by latency on the IP side. |
| 20:23.03 | BrianR___ | diamon: Ok. Create an extension for something like echotest that will go on and on playing audio forever. Dial up that extension on a dozen or so call appearances on your polycom. Without hanging up, power-cycle the polycom. The phone will be pretty much unusable until all of the sip sessions time out and stop sending rtp. |
| 20:23.15 | BrianR___ | diamon: Also, lots of broadcast traffic on the segment can cause trouble too. |
| 20:23.18 | vooduhal | ManxPower, we've had echo caused by extremely high amounts of jitter. |
| 20:23.46 | BrianR___ | Latency/jitter doesn't make echo - it just makes it more prominent. |
| 20:23.57 | diamon | BrianR: Huh, so it's an even-worse SIP or TCP stack than the 79**'s, eh? |
| 20:25.49 | ManxPower | vooduhal: no. Echo is caused by the analog 2-wire loop. It can also be cause by speakerphones |
| 20:26.25 | ManxPower | By the time the call is IP the echo should have been removed by the echo canceler |
| 20:27.23 | diamon | I had echo on some SIP gateway 8-port POTS device, turns out it only does echo cancellation tests when powered on. |
| 20:27.37 | |ryan| | Would anybody be willing to see if they can get a call through to my Asterisk via sip:828240813@callwithus.com ? |
| 20:28.25 | *** join/#asterisk brea (n=rollergr@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:28.58 | brea | Is asterisk 1.4 stable enough to base new projects off of it? |
| 20:29.20 | |ryan| | I'm trying to figure out wheter or not a problem iw eith my DID |
| 20:29.30 | Ch0Hag | http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication lists (possibly correctly) the order in which IAX authentication attempts are made. |
| 20:29.37 | Ch0Hag | Is there an equivalent list anywhere for SIP? |
| 20:30.09 | deb_user | could somebody tell me why, even though my sip show registry and sip show peers checks out ok, I can't receive calls on a SIP DID? |
| 20:30.35 | Ch0Hag | deb_user: What does the console say when the call is made? |
| 20:30.41 | ManxPower | deb_user: how about "sip show users" |
| 20:31.06 | Ch0Hag | manx: You probably know. |
| 20:31.21 | deb_user | manx: it shows up on sip show users too |
| 20:31.33 | ManxPower | Ch0Hag: I've never needed to care, it just works when I set it up |
| 20:31.36 | deb_user | ch0hag: i don't see that the call is trying to connect at all |
| 20:31.48 | ManxPower | deb_user: turn on sip debug and try an inbound call |
| 20:31.58 | Ch0Hag | deb_user: If that's the case (with verbose set to 2 or 3) you probably need to fix your supplier. |
| 20:32.02 | deb_user | manxpower: i've done that too, and still see no output |
| 20:32.03 | Ch0Hag | And/or firewall. |
| 20:32.14 | Ch0Hag | Manx: Well yes, it works. |
| 20:32.14 | deb_user | supplier? you mean the did supplier? |
| 20:32.16 | ManxPower | deb_user: then the call is not getting to you |
| 20:32.19 | Ch0Hag | I want to know why it works. |
| 20:32.46 | deb_user | manxpower: that's what I think...i was having the same problem on another paid DID line, and they said my server was rejecting the calls, but I didn't see anything at all coming in |
| 20:33.26 | deb_user | ch0hag: firewall problem seems unlikely since it was working yesterday and I made no changes (was working for months before that) |
| 20:33.27 | ManxPower | deb_user: an incoming call might not show up without sip debug, but ALL sip data will show up with sip debug enabled |
| 20:33.36 | deb_user | one sec.. |
| 20:33.39 | deb_user | lemme try again |
| 20:33.40 | type0 | I sure am glad I dont have to head to the mountain today, its -52 with no windchill |
| 20:33.43 | type0 | fu-ck that. |
| 20:33.51 | deb_user | i'll just turn it on for that peer |
| 20:34.28 | type0 | do you know what defcon status your balls head into when its -52f |
| 20:34.28 | ManxPower | deb_user: no, turn it on for everything. |
| 20:34.36 | ManxPower | if the incoming call is not matching a peer..... |
| 20:35.21 | deb_user | manxpower: i'm going to paste the output to pastebin |
| 20:35.22 | deb_user | one sec |
| 20:37.20 | vooduhal | So a 108ms delay on the same LAN segment is acceptable? |
| 20:37.30 | deb_user | manxpower: http://pastebin.ca/378833 |
| 20:37.37 | deb_user | this is the output i got when I dialed in |
| 20:37.39 | Qwell[] | 250ms round-trip is considered "acceptable" |
| 20:37.42 | deb_user | but... |
| 20:37.49 | deb_user | i'm not sure if its related to the incoming call or not |
| 20:38.03 | ManxPower | vooduhal: I can't imagine ever having 108ms ping times on a local lan |
| 20:38.20 | sfbosch | ManxPower: Does an FXO module need to be connected to a working line in order for fxotune to work? |
| 20:38.23 | vooduhal | I didn't think sip show peers was showing ping times. |
| 20:38.49 | sfbosch | ManxPower: because I have only one line plugged in; the other two FXO modules aren't connected |
| 20:38.51 | ManxPower | sfbosch: yes, as it tests your SPECIFIC line |
| 20:39.20 | ManxPower | vooduhal: Oh! it shows response time to a SIP options packet. It's a pretty useless measurement. |
| 20:39.32 | sfbosch | ManxPower: will it fail on an unconnected FXO module and roll over? |
| 20:39.37 | vooduhal | Ah... |
| 20:39.45 | ManxPower | sfbosch: as in it tests that specific pair of copper from you to the telco office. |
| 20:40.02 | deb_user | would making changes to the cdr manager affect my sip inbound calls? |
| 20:40.02 | ManxPower | sfbosch: I have no idea, I've never tested it without all lines plugged in |
| 20:40.05 | sfbosch | ManxPower: Right -- the trouble is that I have only one line to test with and three FXO modules on this card |
| 20:40.34 | ManxPower | sfbosch: |
| 20:40.54 | ManxPower | sfbosch: I have no idea what it would do. I stopped using analog cards at least 2 years ago |
| 20:40.59 | Opperior | ManxPower: I have. It just skips them with an error |
| 20:41.15 | sfbosch | Opperior: How long before it does that? |
| 20:41.22 | Opperior | immediately for me |
| 20:41.40 | sfbosch | Opperior: weird |
| 20:41.53 | sfbosch | Opperior: asterisk1 zaptel-1.2.14 # ./fxotune -i 4 |
| 20:41.53 | sfbosch | Tuning module 1 |
| 20:41.53 | sfbosch | Skipping non-TDM / non-FXO |
| 20:41.54 | sfbosch | Failure! |
| 20:41.55 | sfbosch | Tuning module 2 |
| 20:41.55 | sfbosch | <PROTECTED> |
| 20:41.56 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:42.03 | sfbosch | That's what I see -- module 2 is not connected |
| 20:42.06 | *** join/#asterisk progmonkey (n=asdfkle@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:42.36 | progmonkey | Is there a way to do a file exist for the Read() command? |
| 20:42.58 | sfbosch | hmn -- I see output from fxotune in /var/log/messages |
| 20:43.08 | sfbosch | Mar 2 13:42:57 asterisk1 -- Setting echo registers: |
| 20:43.08 | sfbosch | Mar 2 13:42:57 asterisk1 -- Set echo registers successfully |
| 20:43.12 | progmonkey | I know about playbackstatus for Play() but that doesn't work with read |
| 20:44.05 | *** join/#asterisk MarkWD (n=Mark@rrcs-67-78-88-186.sw.biz.rr.com) |
| 20:44.14 | brea | progmonkey: nope |
| 20:44.18 | ManxPower | progmonkey: Does READ play any sound files? |
| 20:44.29 | koos | nokia 770/800 has this telepathy framework to talk via google. Can I configure gtalk to use my n800 like a SIP phone (ie. w/o the google roundtrip)? |
| 20:44.52 | progmonkey | ManxPower: yes Read plays a sound file and then accepts dtmf |
| 20:45.07 | ManxPower | progmonkey: don't use the playback feature of read then. Use playback, then a read |
| 20:45.16 | ManxPower | it would do exactly the same thing |
| 20:45.18 | [TK]D-Fender | koos: You don't have to do anything. Its telepathic and thus already knows how... |
| 20:45.55 | progmonkey | ManxPower: what if they hit a button during the play? |
| 20:46.32 | ManxPower | progmonkey: based on the output of "show applicaiton read" it looks like they can't press a key during READ's playback of the file either. |
| 20:46.35 | koos | [TK]D-Fender: oh, but can I dial my own Asterisk extensions from my nokia device? |
| 20:46.45 | ManxPower | <PROTECTED> |
| 20:46.58 | ManxPower | notice the word "before" |
| 20:47.15 | brea | DTMF will interrupt a read |
| 20:47.18 | progmonkey | ManxPower: max digits, skips to the end of the play |
| 20:47.38 | progmonkey | ManxPower: Read I mean |
| 20:47.39 | ManxPower | progmonkey: perhaps you just need to make sure the file exists |
| 20:47.56 | progmonkey | ManxPower: outside of asterisk you mean? |
| 20:48.07 | *** join/#asterisk oQPa (n=uawename@143.Red-83-41-117.dynamicIP.rima-tde.net) |
| 20:48.23 | Opperior | sfbosch: if might be that I don't have my unused channels defined in my zapta.conf |
| 20:48.29 | *** join/#asterisk backblue (n=moo@87-196-11-24.net.novis.pt) |
| 20:49.03 | ManxPower | progmonkey: correct |
| 20:49.55 | progmonkey | ManxPower: well thx anyways |
| 20:50.08 | progmonkey | you too brea |
| 20:52.04 | diamon | So, any suggestions on what to look at to figure out why my "Message Waiting Indicator" (MWI light) on my Cisco 7960/7940's refuses to go on? I have 'mailbox=###@device' and 'mailbox=###@default', even rebooted the phones both with and without messages, and I can't get the darn thing to light when there's messages present. * 1.2.12 on CentOS 4. |
| 20:52.54 | ManxPower | and if you are really desperate for this feature, then you can write an AGI to test |
| 20:53.18 | deb_user | does * 1.4 have any known issues with incoming sip calls? |
| 20:53.29 | deb_user | i've started to have problems since upgrading |
| 20:53.34 | ManxPower | diamon: it would be mailbox=voicemailfromvoicemailconf@voicemailcontestfromvoiicemailconf |
| 20:54.27 | ManxPower | i.e. if you have a voicemail.conf that contained a mailbox of 1234 in the context of [corporate] then in sip.conf, under the section for that [device] you would have mailbox=1234@corporate |
| 20:54.29 | *** join/#asterisk lacrimosa (n=food@client-82-199-198-254.speedy.sellinet.net) |
| 20:55.09 | diamon | So the likely problem is the @default portion then, the extension is correct... |
| 20:55.32 | ManxPower | diamon: are all your mailboxes in the [default] section of voicemail.conf? |
| 20:55.51 | lacrimosa | hi, is there a h264-based video/VoIP client, compatible with Asterisk ? |
| 20:55.53 | diamon | Yes. |
| 20:56.26 | ManxPower | mailbox=###@default should work. All you would need to do is a "reload" command in the Asterisk CLI |
| 20:57.39 | diamon | Maybe that's what I'm missing then... Sadly, I know a ton about hardware and OS's, and a lot about linux in general, but my knowledge of * is just past the 'dangerous' phase. I keep looking to strange possibilities when I should double-check the simple stuff. |
| 20:57.58 | *** join/#asterisk darylvoip (n=darylvoi@c-67-165-25-195.hsd1.pa.comcast.net) |
| 20:58.03 | diamon | When I run that, does the user need to do anything on their phone, or will it just light up? |
| 20:58.22 | ManxPower | diamon: MWI is totally independent from the phone. |
| 20:58.30 | ManxPower | it is totally asteirsk's job |
| 20:59.03 | diamon | That's what I'd thought. I'd given a passing check to the phone's config, but for every operation except MWI, it works like a champ. |
| 20:59.33 | diamon | Ok, reload issued, seems happy. Let me see if there's any change... |
| 20:59.45 | ManxPower | just do a "reload app_voicemail.so" after any changes to /etc/asterisk/voicemail.conf |
| 21:00.26 | diamon | manx: Done, but it didn't report anything, just came back to the prompt. Is that expected? |
| 21:00.51 | ManxPower | diamon: it should have said something like: |
| 21:00.51 | ManxPower | <PROTECTED> |
| 21:00.52 | ManxPower | <PROTECTED> |
| 21:01.20 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:01.37 | diamon | Hmm. It did not. And yet, voicemail works if you dial it or leave a message, it's just not doing the MWI. |
| 21:01.50 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 21:02.13 | diamon | Ok, AFK a minute while I check the phones with eyeball mk I. |
| 21:04.25 | MonkeyHugs | It did not come back with a response because verbose is probably set to 0 |
| 21:04.37 | *** part/#asterisk lacrimosa (n=food@client-82-199-198-254.speedy.sellinet.net) |
| 21:05.26 | diamon | Nope, no lights for MWI. |
| 21:05.39 | diamon | Monkey: Is that something I set in voicemail.conf, or elsewhere? |
| 21:06.17 | diamon | Oh, foo, or do I just need to add -v when I connect to asterisk... |
| 21:06.21 | MonkeyHugs | at the cli type set verbose 3 or higher. the higher the verbose the more output you will see |
| 21:06.51 | *** part/#asterisk koos (n=koos@e176185126.adsl.alicedsl.de) |
| 21:06.59 | MonkeyHugs | or the v for each level of verbose you want |
| 21:07.00 | diamon | Was 1, now 3. |
| 21:07.18 | diamon | Ok, reload of voicemail did give info now. |
| 21:07.30 | MonkeyHugs | exp. -vvv for a verbose of 3 |
| 21:08.12 | diamon | Ok, so now I'm up to V 3. Still no MWI lit. Is there a way to toggle it from the CLI just to test? |
| 21:08.46 | Qwell[] | Cisco bug |
| 21:08.55 | Qwell[] | try searching the various mailing lists and forums |
| 21:09.44 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:09.45 | diamon | Qwell: Been doing that, but nothing useful so far. |
| 21:10.09 | Qwell[] | .google cisco mwi (0/0) |
| 21:10.47 | diamon | Actually, I just did the simple search for 'asterisk cisco mwi', and by god, there it is, and yes, I'm using that dratted 8.2 |
| 21:11.20 | sfbosch | Setting up a Polycom phone to do auto answer (for paging) -- do I even need to add anything to sip.cfg? There's already a "ringtype" tag in it |
| 21:11.26 | MonkeyHugs | diamon: are you using an ftp server for your configs or just static settings on the phone? |
| 21:11.31 | diamon | ARRGH. Oh well. If that's all it needs, I won't argue. 8.3 is supposed to fix it, I think folks are saying... Bummer nobody follows up on their "let me try this fast and see" sorts of posts. |
| 21:11.53 | diamon | Monkey: tftp. Rolling out a new firmware won't break my heart, I'll just have to track down 8.3 somewhere. |
| 21:11.56 | ManxPower | sfbosch: the wiki has info on it |
| 21:12.45 | sfbosch | The wiki has really old information on it. It refers to an ipmid.cfg file that isn't used anymore |
| 21:12.51 | sfbosch | the ringtype tag is in the sip.cfg file now |
| 21:13.04 | sfbosch | and the tags are not documented in the Admin Guide :( |
| 21:13.08 | diamon | Qwell: Gah. Thanks a lot for the comment, I've been looking into this for days with no luck. I didn't think such a simple search string would get anything useful back, so here I was doing fancy searches... |
| 21:13.30 | *** join/#asterisk egypcio (n=egypcio@200.194.102.234) |
| 21:13.31 | *** part/#asterisk ZaVoid (n=colin@c-67-165-25-195.hsd1.ct.comcast.net) |
| 21:15.12 | sfbosch | and it has this line in it: <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/> |
| 21:16.16 | *** join/#asterisk docelm0 (n=vircuser@c-68-82-181-225.hsd1.de.comcast.net) |
| 21:16.24 | *** join/#asterisk Strom_M (n=strom@63.110.13.126) |
| 21:17.00 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 21:17.20 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:18.04 | sfbosch | hang on, found the reference in the Admin Guide... |
| 21:18.24 | sfbosch | hidden in the "Sound Effects" section |
| 21:23.36 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:26.11 | *** join/#asterisk J4k3 (i=J4k3@143.sub-70-216-99.myvzw.com) |
| 21:26.35 | diamon | Ok, now to see if new firmware is what it needs. Silly bugs, but thank goodness for upgradable firmware! |
| 21:28.01 | diamon | Anyone know if 8.3 or 8.6 are ok from personal experience? :) Sudden paranoia... |
| 21:28.16 | Qwell[] | diamon: Ciscos SIP images suck in general |
| 21:28.27 | Opperior | paranoia is healthy, in moderation |
| 21:28.32 | diamon | Yeah, but at least they mostly work, more-or-less. |
| 21:28.45 | SomethingISODD | question is there anyway to send a error code back if there is no answer? right now all i am getting is normal call clearing |
| 21:28.54 | diamon | Opp: Exactly. Figured I'd ask quick, and barring anything else, POW, off they go anyway. |
| 21:28.54 | Opperior | at least that's what the voices tell me |
| 21:29.00 | SomethingISODD | i would like to give me an error 19 or 21 or 34 |
| 21:29.26 | diamon | Opp: Gotta watch the voices. The suggestions may be neat, but they're usually not so fun... |
| 21:30.23 | J4k3 | haha asterisk has a milliwatt signal generator :) |
| 21:30.25 | J4k3 | err ? :) |
| 21:31.04 | tzafrir_laptop | J4k3, I recall somewhat also wrote a miliwatt tone tester utility |
| 21:31.11 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1167878982.dsl.bell.ca) |
| 21:31.21 | tzanger | J4k3: yes |
| 21:34.03 | sivana | bah |
| 21:37.58 | SomethingISODD | is there anyway to see how many active channels there is and the duration of the call |
| 21:38.51 | Opperior | been trying to figure out this one: when I have any feature enabled in features.conf, whatever key it starts with, I can no longer send that as a dtmf tone during a call. |
| 21:38.54 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 21:39.08 | Defend | somethingisodd show channels |
| 21:39.26 | SomethingISODD | Defend, ya in older versions there was show active if i remember |
| 21:39.55 | SomethingISODD | and it shows actual duration |
| 21:39.55 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 21:39.55 | Opperior | e.g. I have automon => *1. Now I can't send a * dtmf dduring a call, which makes navigating some voice mail systems difficult |
| 21:39.56 | Defend | show channel <channel> will show duration |
| 21:40.06 | Ch0Hag | Can some kind sould read http://monnsta.net/~mking/user-vs-peer and point out any inaccuracies. |
| 21:40.23 | Opperior | sip/iax/zapel show channels will list the active ones, iirc |
| 21:40.47 | Ch0Hag | I intend to use it as a starting point for a writeup on a basic Asterisk setup. |
| 21:41.41 | diamon | And now to 8.6... |
| 21:42.13 | SomethingISODD | Defend, thanks let me try that i have been using show channels |
| 21:42.25 | Ch0Hag | The most important feature of said writeup is that is will be accurate and unambiguous, unlike just about everything else I've found. |
| 21:42.27 | brea | Anyone know of a document that helps you convert 1.2 apps to 1.4? |
| 21:42.39 | Defend | np it gives you the specifics about the active channel |
| 21:42.44 | brea | Something that points out the changes |
| 21:43.41 | SomethingISODD | thats alot more information. |
| 21:44.49 | Bobthehunter | wow asterisk filtering mailling lists lol |
| 21:44.51 | Mavvie | hmm... is there a way to catch a * inside an extension? |
| 21:45.35 | ManxPower | Mavvie: Um put a * in the exten pattern |
| 21:45.52 | *** join/#asterisk DaneM (n=DaneM@adsl-75-25-36-35.dsl.chi2ca.sbcglobal.net) |
| 21:45.56 | ManxPower | exten => *69,1,AGI(sex.agi) |
| 21:46.21 | DaneM | Hello. I have a question: can a TDM400P support a 2-line (4 lead) phone line? |
| 21:46.33 | SomethingISODD | ManxPower, thats a interesting agi script lol |
| 21:46.34 | Mavvie | ManxPower: no, inside: exten => 3018*436,1,macro(foo,$EXTEN). |
| 21:46.45 | Qwell[] | DaneM: no, but you can make a "special" cable for it |
| 21:46.47 | Mavvie | ManxPower: and then that 3018 and 436 are random numbers. let me check. |
| 21:46.51 | ManxPower | Mavvie: THERE IS NO REASON THAT WOULD NOT WORK |
| 21:46.53 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 21:46.57 | DaneM | how so? |
| 21:47.08 | ManxPower | damn caps lock |
| 21:47.13 | Mavvie | ManxPower: with _X.*X. ? |
| 21:47.18 | Qwell[] | DaneM: it's just 4 wires instead of 2... make a little Y adapter dealie |
| 21:47.38 | ManxPower | Mavvie: no. You cannot have any more digits after a . wildcard |
| 21:47.45 | Mavvie | ManxPower: that's what I mean. |
| 21:47.47 | DaneM | ...and connect each wire from the Y adapter into a different port on the modem? |
| 21:48.18 | ManxPower | you could do exten => _XXX*XXX,1,Dial( |
| 21:48.19 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:48.19 | Qwell[] | "modem"? |
| 21:48.21 | ManxPower | Mavvie: it has nothing to do with a * |
| 21:48.21 | DaneM | well, on the TDM400P |
| 21:48.21 | Mavvie | ManxPower: and then one line per possible number of X's. |
| 21:48.27 | Mavvie | before and after the *. |
| 21:48.33 | Qwell[] | DaneM: yeah, something like that should work |
| 21:48.37 | Mavvie | say 20 x 4 lines. |
| 21:48.42 | DaneM | sweet. Thanks |
| 21:49.06 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 21:49.09 | ManxPower | Mavvie: why not just _X.,1,Whatever |
| 21:49.10 | Mahmoud | hello |
| 21:49.29 | Mavvie | ManxPower: before calls to numbers in that format have to handled by a different macro. |
| 21:49.44 | Mahmoud | i'm trying to deliver my users' status to my SIP phone, but i'm kinda lost |
| 21:50.02 | Mahmoud | i edited the extentions.conf and added "hint" but unsure if it's the right way or not |
| 21:50.38 | Mahmoud | exten => 222,hint,111 <--- using this, i send 222's status to the 111 extention.. am i right? |
| 21:50.44 | ManxPower | Mavvie: your issue us that you can't have digits after the . patern match, not that you have a * in the extension |
| 21:50.55 | Mavvie | ManxPower: true. |
| 21:51.08 | ManxPower | Mahmoud: no. exten => 222,hint,SIP/111 if the SIP userid is 111 |
| 21:51.09 | wunderkin | you can override other matches before that one |
| 21:51.11 | *** join/#asterisk kotique[male] (n=burumba@host-86-106-210-108.moldtelecom.md) |
| 21:51.13 | kotique[male] | hi |
| 21:51.33 | wunderkin | heh, mold telecom |
| 21:51.49 | Mahmoud | ManxPower, oh it was a typo.. |
| 21:52.00 | Mahmoud | ManxPower, userid = username for authentication right? |
| 21:52.09 | ManxPower | Mahmoud: that is why you copy and paste |
| 21:52.25 | ManxPower | Mahmoud: no, userid is what you would put in the Dial( line |
| 21:52.32 | Mahmoud | ManxPower, the problem is that i want to deliver user status from every one to every one.. it gets hard :/ |
| 21:54.13 | ManxPower | Mahmoud: your phone needs to have a line button for each one you want to monitor |
| 21:54.14 | *** join/#asterisk [shodan] (n=shodan@ip067.99-113-216.pppoe4.joliette.intermonde.net) |
| 21:54.14 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 21:54.28 | Mahmoud | ManxPower, i didn't get your point.. do you mean that it's physically hard to implement such thing, and that's why asterisk solves it in a more manual way? |
| 21:55.32 | kotique[male] | hey guys, I have such an expression in AEL dialplan: TMP=${EVAL(${cutpattern})}; {cutpattern} is set to ${NEWNUMBER}, and ${NEWNUMBER} is set to 14000000000. guess what I'm having in TMP ? yeah, 140300000000861 :) why ?!! |
| 21:55.35 | SomethingISODD | question in the cdr_custom Can Asterisk put out fractional seconds ? |
| 21:56.14 | kotique[male] | on next run I'm gettin 140300000003618 :) |
| 21:56.30 | ManxPower | Mahmoud: not hard to do in asterisk at all, just lots of exten lines for hints |
| 21:57.11 | ManxPower | Mahmoud: and you would usually do exten => 222,hint,SIP/222 then tell your phone device to do a SIP SUBSCRIBE for 222 so it can see the status of 222 |
| 21:57.43 | Mahmoud | 222 is the extention that's being advertised.. right? |
| 21:58.06 | CunningPike | Mahmoud: We include a file containing a hint for all our endpoints, so that it's easy to set them up for anyone |
| 21:58.20 | kotique[male] | :-/ |
| 21:58.23 | ManxPower | correct. advertized to any device that requests notification |
| 21:58.55 | Mahmoud | ManxPower, oh.. so when we say SIP/<same advertised phone> it will result in advertising to all phones? |
| 21:59.04 | *** join/#asterisk m0nst3r (n=m0nst3r@c-71-206-116-36.hsd1.mi.comcast.net) |
| 21:59.25 | ManxPower | Mahmoud: NO! IT does not SEND notifications unless the phone requests the information. The phone has to initate the request |
| 21:59.39 | Mahmoud | sure, the request is needed |
| 21:59.58 | Mahmoud | <advertise this>,hint,SIP<to this phone> <---- this is what i was thinking about |
| 22:00.16 | ManxPower | Mahmoud: what if you had 500 people in your company. Do you really want a massive desk of little lights to monitor all 500 phones? |
| 22:00.21 | *** join/#asterisk dual-man (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 22:00.21 | ManxPower | That is why this method does not scale. |
| 22:00.24 | *** part/#asterisk m0nst3r (n=m0nst3r@c-71-206-116-36.hsd1.mi.comcast.net) |
| 22:00.40 | Mahmoud | yeah.. |
| 22:00.57 | dual-man | i having trouble to get asterisk to accept a extension after Playtones(busy), any ideas why that might be? |
| 22:02.02 | ManxPower | dual-man: and it works just fine if you don't use playtones? |
| 22:02.09 | dual-man | yup |
| 22:02.16 | kotique[male] | hey wait, i wanna have a desk with 500 little lights too. where can I apply for this job? |
| 22:02.20 | ManxPower | dual-man: what DTMF mode are you using? |
| 22:02.33 | dual-man | uhm, dunno :D |
| 22:02.51 | dual-man | how do i find out? |
| 22:03.19 | russellb | where can I find a SIP phone that supports 500 subscriptions? |
| 22:03.49 | dual-man | ManxPower: this is on an zap extension |
| 22:04.03 | ManxPower | dual-man: Next time mention what kind of device you are using. |
| 22:04.32 | Mahmoud | ManxPower, hmmm my understanding for the command was wrong :/ |
| 22:05.04 | Mahmoud | ManxPower, I was thinking that it's implemented like this: <advertisethis>,hint,SIP/<tothisperson> |
| 22:06.15 | Mahmoud | ManxPower, while the right one is: <queries_to_this_extn>,hint,SIP/<get_the_status_of_this_user> |
| 22:06.20 | reber | i installed asterisk and i can't get x-lite to connect localy to my server. Any help ? |
| 22:07.25 | kotique[male] | fixed my problem by replacing that string with Set(NEWNUMBER=${EVAL(${cutpattern})}); |
| 22:07.49 | kotique[male] | thanks to all ppl who helped me |
| 22:08.01 | reber | i get always the same window that appears. I just put a very basic configuration file. Any help maybe ? |
| 22:08.53 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:11.12 | ManxPower | reber: we have no idea what you are talking about. |
| 22:11.25 | MonkeyHugs | What do I specify in Zapata.conf to setup one of my ports on a digium t1/pri card? |
| 22:11.50 | ManxPower | MonkeyHugs: that totally depends on what your T-1 is set up as. |
| 22:11.55 | *** join/#asterisk J4k3 (n=J4k3@12.45.185.225) |
| 22:13.09 | MonkeyHugs | What I am doing is coming in from the telcom as PRI and trying to send it out to a fax server as t1 signaling. |
| 22:13.29 | ManxPower | MonkeyHugs: so you need to set up the port for PRI? |
| 22:14.30 | MonkeyHugs | Trying to set up the port that is acting as the network side of a t1 (RBS) |
| 22:14.43 | reber | ManxPower, i configure asterisk with a basic setup (only one extension), and i'm trying to connect x-lite (a softphone) to asterisk, but nothing happens (a quick message about a firewall ... But i have no firewall and using asterisk and the sofphone locally) ... Any ideas ? |
| 22:14.50 | ManxPower | MonkeyHugs: I assume the port is going into a channel bank? |
| 22:15.06 | ManxPower | reber: none at all |
| 22:15.59 | MonkeyHugs | channel bank --pri--> asterisk gateway --t1 (RBS)--> Brooktrout TR114 t1 fax card |
| 22:16.06 | syzygyBSD | anyone have an idea of why when i have the music on hold option in the dial command it rings once then hangs up? |
| 22:16.13 | *** join/#asterisk tecolote99 (n=ruben@63.110.13.126) |
| 22:16.22 | ManxPower | MonkeyHugs: does the fax card expect fxo signalling or fxs signalling? |
| 22:16.59 | ManxPower | you would use either signalling=fxo_ks or signalling=fxs_ks and then channel => startchannel - endchannel |
| 22:17.21 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-46-49.vic.bigpond.net.au) |
| 22:17.26 | MonkeyHugs | it would be receiving faxes and sending faxes |
| 22:17.28 | ManxPower | where start and end are the starting channel numbers and the ending channel numbers. The actual value would depend on which port you have connected to the fax card |
| 22:17.51 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 22:17.53 | ManxPower | MonkeyHugs: I would ASSUME then it wants fxs_ks, but there is no way to tell |
| 22:18.27 | reber | ManxPower, ok. Got any basic configuration files i could try maybe ? |
| 22:18.37 | thekidrio | manxpower, heh i knew a fella in SWG who had Manx Power |
| 22:18.42 | ManxPower | reber: I would never ever use a softphone. |
| 22:19.00 | ManxPower | SWG = SouthWest Georgia? |
| 22:19.02 | thekidrio | softphones are good enough for testing basic connections |
| 22:19.08 | thekidrio | swg= star wars galaxies haha |
| 22:19.25 | *** join/#asterisk stridernzl (n=neville@125-237-97-222.jetstream.xtra.co.nz) |
| 22:19.25 | diamon | Ok, firmware 8.6 seems like it may be working for the MWI, I'm seeing users getting their voicemail in the asterisk CLI. Now to call some of them and ask... |
| 22:20.11 | ManxPower | The difference between a softphone and hardphone is like the difference between a Barbie Easy Bake Oven and a kitchen oven. |
| 22:20.11 | syzygyBSD | lol, southwest georgia.. I would have never come up with that, but I am in the northwest |
| 22:20.19 | reber | thekidrio, exactly what i try to do :/ |
| 22:20.25 | Qwell[] | ManxPower: both will burn you if you try to use them? |
| 22:20.42 | syzygyBSD | those easy bake ovens are dangerous |
| 22:20.44 | ManxPower | Qwell: I don't think the easy bake can burn anything. |
| 22:21.05 | syzygyBSD | sounds like ManxPower has never had an easy bake oven, or a daughter |
| 22:21.07 | *** join/#asterisk hellojoe (n=hijoe@nat-service4.juniper.net) |
| 22:21.24 | ManxPower | syzygyBSD: children are evil. |
| 22:21.35 | diamon | Odd... when doing a 'sip show peers' I'm seeing both port 5060 and 5061. I don't have any references to port 5061 that *I* know of... Status shows OK (xxxms) though. |
| 22:21.39 | syzygyBSD | so are old people |
| 22:21.50 | ManxPower | They start out that way. Good parenting can usually change that, but not always |
| 22:22.01 | syzygyBSD | original sin |
| 22:22.03 | ManxPower | diamon: that is the REMOTE port and means pretty much nothing |
| 22:22.08 | thekidrio | ManxPower: they were just recalled haha (easy bakes) |
| 22:22.35 | thekidrio | kids were burning themselves |
| 22:22.35 | ManxPower | syzygyBSD: "sin" has nothing to do with it. |
| 22:22.35 | thekidrio | proof we are getting dummer |
| 22:22.42 | syzygyBSD | lol, sin is evil... |
| 22:22.43 | thekidrio | dumber too |
| 22:22.44 | diamon | Ok, I'm happy then. Status of OK made me feel decent about it, but figured I'd check quick. |
| 22:22.50 | syzygyBSD | thekidrio: I think dummer makes your point better |
| 22:23.02 | thekidrio | hehe yeah |
| 22:23.58 | reber | "ManxPower> syzygyBSD: children are evil." yeap but noddy is worse ... |
| 22:24.28 | *** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net) |
| 22:25.13 | syzygyBSD | reber: thank you for destroying the rest of my life by making me look up what noddy was... |
| 22:25.21 | reber | heh |
| 22:26.15 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-bbe420bb1c33b5a6) |
| 22:27.44 | *** join/#asterisk vlt|home (n=daniel@dslb-088-073-251-027.pools.arcor-ip.net) |
| 22:28.13 | dual-man | ManxPower, i've tried setting relaxdtmf=yes but i still can't dial an extension on playtones(busy) with a zap extension |
| 22:28.14 | reber | syzygyBSD, i can't avoid to have it every morning god makes, think at that ;) |
| 22:28.28 | mercestes | what is noddy? |
| 22:28.42 | reber | mercestes, no.. ermm, don't... |
| 22:28.42 | syzygyBSD | mercestes: ignorance is bliss |
| 22:28.44 | reber | ever ... |
| 22:28.53 | mercestes | ... |
| 22:29.05 | syzygyBSD | lol, now he is going to google it for sure |
| 22:29.13 | reber | hehe |
| 22:29.19 | mercestes | just tell me. :( |
| 22:31.06 | syzygyBSD | anyone have a sample musiconhold.conf for 1.2.* that they could show me? |
| 22:31.14 | brea | Anyone know what causes: Module 'app_blah.so' did not register itself during load ? |
| 22:33.06 | SplasPood | syzygyBSD: there's one included in the 1.2 source tarball |
| 22:33.09 | SplasPood | syzygyBSD: in configs/ |
| 22:33.19 | SplasPood | syzygyBSD: along with configs for everything else.. |
| 22:33.27 | dual-man | does anyone know why asterisk does not accept dtmf digit while playtones(beep) is going, from a zap extension? |
| 22:33.54 | syzygyBSD | dual-man: yes |
| 22:34.17 | dual-man | syzygyBSD: ok, why? |
| 22:34.18 | Qwell[] | dual-man: playtones doesn't accept digits |
| 22:34.21 | Qwell[] | use Background() |
| 22:34.28 | syzygyBSD | ^ |
| 22:34.44 | dual-man | Qwell, i've used playtones before, worked fine |
| 22:34.55 | Qwell[] | Not with dtmf you haven't |
| 22:36.32 | ManxPower | playtones should just start the tone and go onto the next priority |
| 22:36.57 | wunderkin | Oh no you didn't, girlfriend |
| 22:37.20 | ManxPower | it will continue the tone until you stop the tones using stoptones or whatever the correct app is |
| 22:37.30 | dual-man | so how can i create a busy signal with background? |
| 22:37.51 | ManxPower | stopplaytones is the correct one |
| 22:38.10 | ManxPower | dual-man: do an Answer() before the playtones |
| 22:40.12 | dual-man | MAnxPower still nothing |
| 22:40.56 | diamon | Ok, for the record, switching to the P0S3-08-6-0 firmware did fix the MWI light on my 7940/7960's, but some phones did not reconfigure properly, and required a manual reset (*-6-settings) locally to continue. Jury's still out if they're going to recover after a reset or if I'll be very sad, but I'll see shortly. |
| 22:41.53 | hellojoe | I am going nuts with the variable setting with the Dial Macro? Spend over 4 hours and no luck.... damn! |
| 22:42.03 | Mahmoud | is there any command to always report "online" hint status? |
| 22:42.38 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:42.42 | hellojoe | Wish there was a away to set a value of the variable within the macro and export it out to the calling context. Works for regular macro but the Dial macro doesn't work |
| 22:43.14 | hellojoe | ManxPower: Any suggestions? I will spoil the rest of my day if I don't get it done quickly :-( |
| 22:43.48 | ManxPower | hellojoe: post an example on the mailing list and ask if anyone has a workaound. Then go home and have a couple of beers |
| 22:44.05 | hellojoe | ahh... not a bad idea... |
| 22:44.08 | brea | AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "My Application"); won't compile even though I'm including modules.h, any ideas? |
| 22:44.12 | hellojoe | thanks ManxPower |
| 22:48.32 | syzygyBSD | hellojoe: are you setting the global variable or just the local variable |
| 22:49.02 | vlt|home | Hello. Anyone else experienced the "ppp0 on NAT machine (Sarge w/ 2.6.8) is sending sip/udp packets FROM the old IP after ppp0 is reconnected" problem? Atm I fear every new morning hoping that my workaround if-up.d/script that sets some ip_conntrack_udp_timeout values to 0, waits 10 seconds and then resets them has worked. Any known better solution for this? |
| 22:49.27 | hellojoe | just the local |
| 22:49.32 | Mahmoud | ManxPower, is it normal to have the status not changing for minutes and minutes unless the client is restarted? |
| 22:50.02 | Mahmoud | ManxPower, using eyeBeam, the status never changes unless the client is just started |
| 22:50.56 | syzygyBSD | hellojoe: http://www.voip-info.org/wiki-Asterisk+variables read that, specifically the inheritance of Channel Variables |
| 22:52.49 | hellojoe | thanks syzygyBSD |
| 22:53.07 | etfonhome | 1.4.1 coming anytime soon? |
| 22:53.17 | hellojoe | I read this... I am able to pass the variable to the macro but whatever I set in macro doesn't go back to the calling context and that's my problem |
| 22:54.39 | syzygyBSD | well, my suggestion is to use the DB as a work around, prefix the variable name with the unique call identifier or something |
| 22:55.31 | syzygyBSD | I don't know what you are trying to do so I can't give specifics, and extensions.conf would help but boss is bringing beers soon |
| 22:56.06 | Vec | Not sure if anyone has checked out www.asternic.org/stats/demo/, it looks great but it seems I can't find any info about it, anyone know if its GNU or what ? |
| 22:56.53 | Ch0Hag | Why is the commented field in the configuration database supposed to be an integer not a boolean? |
| 22:59.27 | syzygyBSD | I use integers instead of boolean's for most things I do. Gives more flexibility later, could be commented, being commented, the number of times commented... etc |
| 22:59.40 | Ch0Hag | That doesn't make any sense. |
| 22:59.53 | Ch0Hag | Either something is commented and ignored, or it is not and is parsed. |
| 23:00.01 | Ch0Hag | Hence - boolean. |
| 23:00.10 | Ch0Hag | Booleans are what computers are built on. |
| 23:00.10 | syzygyBSD | true, but I have no clue what DB you are talking about... or the field |
| 23:00.23 | Ch0Hag | It's not the DB, it's asterisk. |
| 23:00.38 | Ch0Hag | It executes: SELECT * FROM config WHERE filename='iax.conf' and commented=0 ORDER BY ... |
| 23:00.54 | Ch0Hag | Where it should be 'and not commented' or whatever the syntax is. |
| 23:01.23 | syzygyBSD | ahh, ok, well when you write it like that I know exactly what you mean |
| 23:01.24 | Ch0Hag | So when I created a table with a commented field of type boolean, the statement returned an error. |
| 23:01.49 | syzygyBSD | Ch0Hag: what DB |
| 23:01.49 | Ch0Hag | Namely, ERROR: operator does not exist: boolean = integer |
| 23:01.54 | Ch0Hag | Postgres |
| 23:02.08 | syzygyBSD | hmm, thought it could evaluate boolean to 1, 0 |
| 23:02.40 | Ch0Hag | It can if I make it, but how does the DB know true == 1? |
| 23:02.47 | Nugget | in pgsql boolean is 'T' or 'F' |
| 23:03.32 | syzygyBSD | wonder what field/database I was using where booleans were 1,0 |
| 23:03.33 | Ch0Hag | I'm going to have to either create a view to munge the field to an int or store an int where I should store a boolean. |
| 23:03.44 | Nugget | or you can "WHERE field IS TRUE" or "WHERE field IS FALSE" |
| 23:03.44 | Ch0Hag | And storing the wrong kind of data really winds me up. |
| 23:04.39 | Nugget | if the code was using "WHERE IS TRUE" or "WHERE IS FALSE" it would work on mysql or pgsql. |
| 23:04.41 | *** join/#asterisk Aurs (n=Aurs@c6870BF51.dhcp.bluecom.no) |
| 23:04.46 | Nugget | that's the cleaner fix |
| 23:05.06 | Ch0Hag | Well yes - there's also fixing Asterisk. |
| 23:05.29 | Aurs | doesn't leavewhenempty=strict and joinempty=strict work on permanent queue members? (member=SIP/123 in queues.conf)? |
| 23:05.47 | Qwell[] | Aurs: I'd think so |
| 23:05.49 | syzygyBSD | ahh, twas mysql.. dirty different handleing of booleans |
| 23:06.05 | Nugget | mysql blows goats. |
| 23:06.09 | Aurs | Qwell[]: correction... member=Local/123@context in queues.conf |
| 23:06.18 | syzygyBSD | humans too? |
| 23:06.21 | Ch0Hag | Nugget: And enjoys it. |
| 23:06.29 | Ch0Hag | Does Postgres have a ternary operator? |
| 23:06.34 | syzygyBSD | sounds like the goats are getting a great deal |
| 23:06.47 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 23:06.49 | Qwell[] | completely random and offtopic: I could *totally* go for chili dogs tonight |
| 23:06.55 | Nugget | that sounds good. |
| 23:06.56 | Qwell[] | just putting it out there |
| 23:07.04 | Qwell[] | Nugget: I got a sudden craving :P |
| 23:07.08 | type0 | chili dogs are good with nathans hotdogs |
| 23:07.20 | Nugget | I bought hot dogs today, but have no chili. |
| 23:07.26 | Qwell[] | type0: I don't eat meatlike substance named after people. |
| 23:07.32 | Nugget | steaks are marinating for grilling already anyway. |
| 23:07.36 | type0 | lies |
| 23:07.42 | type0 | I bet you've eaten a meat named after a man |
| 23:07.44 | syzygyBSD | Nugget: http://www.postgresql.org/docs/8.0/interactive/datatype-boolean.html |
| 23:07.52 | type0 | or WOman for that matter |
| 23:07.56 | file | Qwell[]: let's go for it |
| 23:07.58 | syzygyBSD | looks like postgres can handle those ints just fine... |
| 23:08.01 | Qwell[] | file: for...what? |
| 23:08.05 | Nugget | syzygyBSD: no, note the ticks. |
| 23:08.06 | file | Qwell[]: chili dogs |
| 23:08.07 | type0 | killin the aunt jememia pancakes |
| 23:08.10 | Qwell[] | oic, yeah, okay |
| 23:08.17 | Nugget | to compare a bool in pgsql it has to be '1' and not 1 |
| 23:08.23 | Qwell[] | type0: pancakes != meat |
| 23:08.29 | Qwell[] | or, rather, meatlike |
| 23:08.31 | type0 | your pancakes are meat? |
| 23:08.34 | Qwell[] | hotdogs, however, are meatlike |
| 23:08.42 | Qwell[] | meatesque, as it were |
| 23:09.06 | syzygyBSD | "boolean uses 1 byte of storage" wtf... can I ask why it uses 8 bits of storage? |
| 23:09.17 | Ch0Hag | CASE WHEN commented THEN 1 WHEN NOT commented THEN 0 END AS commented |
| 23:09.29 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 23:09.49 | diamon | And yay for the MWI, confirmed operational and the reset phones are OK, but they did need a manual reset. |
| 23:10.12 | diamon | Some did anyway, but not all of them. Not clear why. |
| 23:10.44 | chrisknight | Can anyone help me reset a cisco 7960 phone back to factory defaults? I have tried everything... |
| 23:10.58 | syzygyBSD | the reset button on the back? |
| 23:11.12 | ManxPower | chrisknight: ask diamon |
| 23:11.24 | diamon | Aha! I can do this! How bad off is the phone? |
| 23:11.35 | syzygyBSD | chrisknight: http://www.sokol-associates.com/?q=node/51 |
| 23:11.41 | chrisknight | reset button... If this thing has a reset button, im going to go crazy.. hehehe |
| 23:11.48 | thekidrio | haha |
| 23:11.50 | syzygyBSD | ya, sorry |
| 23:11.52 | diamon | It's a keypad entry. |
| 23:11.59 | chrisknight | well, its locked. and its not the default "cisco" password |
| 23:12.17 | chrisknight | **# settings, gives me nothing. |
| 23:12.25 | diamon | Yes, that URL is exactly what you need for a factory reset. Normal reset is *-6-Settings. |
| 23:12.32 | chrisknight | holding # during boot does not do anything either |
| 23:13.01 | chrisknight | *-6-settings reboots the phone, right? |
| 23:13.17 | syzygyBSD | it should |
| 23:13.27 | chrisknight | then holding # during reboot then 123456789*0# |
| 23:13.31 | syzygyBSD | all at the same time |
| 23:13.32 | diamon | chris: Do *-6-Settings, and the INSTANT the screen blanks and before the LED's start blinking, press and hold *, and KEEP holding it until the three LED's flicker along the bottom one after the other. |
| 23:13.46 | syzygyBSD | * or #? |
| 23:13.50 | chrisknight | ok, just a sec |
| 23:14.07 | diamon | Once they blink, do 123456789*0#. It'll prompt you to choose 1 or 2, clear the net config as well of course. |
| 23:14.17 | diamon | Sorry, dammit, #. |
| 23:14.47 | diamon | Press and hold #, not *, and keep holding it. |
| 23:14.56 | chrisknight | I did # as soon as the screen went blank, headset blinked, then mute, then the phone boots normally while Im still holding # |
| 23:15.18 | thekidrio | <PROTECTED> |
| 23:15.18 | thekidrio | <PROTECTED> |
| 23:15.18 | thekidrio | <PROTECTED> |
| 23:15.18 | thekidrio | <PROTECTED> |
| 23:15.18 | thekidrio | <PROTECTED> |
| 23:15.19 | thekidrio | <PROTECTED> |
| 23:15.21 | thekidrio | <PROTECTED> |
| 23:15.29 | thekidrio | anyone able to see why thats being hung up? |
| 23:15.42 | ManxPower | my eyes! my eyes! |
| 23:15.44 | thekidrio | is it something on the iax providers side? |
| 23:15.46 | chrisknight | Driving my nuts.... |
| 23:15.48 | ManxPower | ~pb |
| 23:15.49 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:15.49 | diamon | Hrrm. Try *-6-Settings and then hit # the instant you let go of the other keys? Timing maybe? |
| 23:16.01 | chrisknight | tried that too... hmmm.. |
| 23:16.08 | thekidrio | heh yeah sorry about that manx, its on one line in my term heh |
| 23:16.46 | chrisknight | it never gets to the speaker butting lighting up, just boots right away |
| 23:16.58 | diamon | chrisknight: Then try powering it off entirely, then hold # and plug it back in, and try just keeping it pressed. |
| 23:16.58 | ManxPower | thekidrio: Put Noop(HANGUPCAUSE is ${HANGUPCAUSE} as the priority after the Dial |
| 23:17.18 | ManxPower | well oop(HANGUPCAUSE is ${HANGUPCAUSE}) |
| 23:17.23 | ManxPower | *sigh* |
| 23:17.33 | chrisknight | I think maybe something is wrone with the phone, I tried to unplug it too with # |
| 23:17.37 | ManxPower | thekidrio: and if you flood the channel again you will be put on the /ignore list |
| 23:17.49 | thekidrio | heh |
| 23:18.01 | thekidrio | i said sorry you monkey |
| 23:18.27 | diamon | chris: If even that doesn't work, and the speaker LED never even flicks, your firmware might be mis-loaded. Try feeding it a new one if possible, then re-try the reboot/reset again once it has the new version. |
| 23:18.39 | chrisknight | if I press settings, (3) network config, then (8) unlock config it askes me for a password. Cisco or cisco does not work. |
| 23:19.01 | chrisknight | oh man... might be over my head |
| 23:19.31 | diamon | chris: Then it was set by someone in the past. The 'press #' thing even resets that AFAIK, so the phone may be a problem. |
| 23:19.55 | diamon | Does it talk to your TFTP server at all, or can it not even get on the network? |
| 23:20.28 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
| 23:20.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:20.41 | chrisknight | im thinking so... It gets an ip address from my DHCP (172.16.16.31/24) let me look & see where it thinks the tftp server is... |
| 23:20.42 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:20.54 | diamon | Heh, I set qualify=3000 finally for my phones with this new firmware, they all show 145 to 195ms latency. :) Ah well, so long as they work, I'm happy! |
| 23:21.36 | ManxPower | diamon: I don't use qualify anymore unless that is NAT involved. It is just not reliable enough |
| 23:21.46 | chrisknight | it thinks the tftp server is at 207.90.66.something |
| 23:22.09 | Aurs | chrisknight: have you set boot server option in your dhcp? |
| 23:22.11 | diamon | chris: Good. Most are set to DHCP, as the MAC is what's used to define the config anyway. You might be able to give it a newer version of firmware (I've just started using 08-6-0 myself, avoid 8.2 through 8.4, Message LED bug). |
| 23:22.44 | diamon | Max: I turned it on just to see what the phones are doing, and to make sure they stay online while I check this new firmware out. It's more of a paranoia thing. |
| 23:23.05 | thekidrio | <PROTECTED> |
| 23:23.14 | thekidrio | though i see you answered me already |
| 23:23.26 | ManxPower | diamon: if the phone misses responding to even 1 SIP OPTIONS packet (which is what qualify= does) then asterisk will consider the phone offline for a time. |
| 23:23.47 | chrisknight | Im very new to asterisk, and IP phones in general but not new to networking. Where do I get the image? and does asterisk run tftp? I could set 1 up I guess. How does the phone know to load it? |
| 23:24.07 | ManxPower | thekidrio: I don't see the Noop line being executed. |
| 23:24.15 | diamon | Manxpower: I'll keep that in mind, thanks. |
| 23:24.21 | *** join/#asterisk foobar778 (i=johhny@ip68-100-210-15.dc.dc.cox.net) |
| 23:24.21 | Nugget | chrisknight: that's different for every phone, and asterisk plays no part in it. |
| 23:24.37 | Nugget | if your ip phone uses tftp for config you'll have to handle it all yourself. |
| 23:24.48 | ManxPower | chrisknight: Welcome to the world of Cisco. If you want a legal copy of the firmware you need to purchase it. |
| 23:25.07 | diamon | chris: Yeah, the config is specific to each phone. You're in luck though; I run several shops with 79x0's and can probably help a good bit. |
| 23:25.21 | chrisknight | hmmm... its getting a dhcp lease from my router but all these other 207 addresses for tftp & dns are not from my network |
| 23:25.26 | diamon | Note my * vs # problem before though, so know my silliness quotient might be pretty high today. :) |
| 23:25.42 | thekidrio | i pasted that before i saw you responded |
| 23:25.57 | diamon | chris: Have you got *any* other 79x0's configured at that location? |
| 23:26.31 | diamon | Manx: It can be downloaded after 6 months or something, you can get versions right from them. |
| 23:26.36 | Nugget | http://hastalasiesta.org/stuffs/nakkeTattis.jpg |
| 23:27.32 | chrisknight | no, I just brought this phone (only ip phone here) home from work, which used to belong to someone else. I have some linksys phones (3) on the way though. |
| 23:27.54 | diamon | And to be honest, if their firmware in the past (7.5) was working, and 8.2 was not, even if 8.6 *was* a pay-only download, I'd shamelessly use it to fix their bug. :) |
| 23:28.48 | chrisknight | diamon: firmware says p0m3-06-2-00 |
| 23:29.24 | diamon | Oh wow, that's madly out of date. Current that I know of is p0m3-08-6-00 |
| 23:29.49 | chrisknight | do I need different firmware to run sip? |
| 23:30.43 | diamon | I think 6.2 might need to get an upgrade to 7.something to be able to upgrade higher. Hmmm. To be honest, I'm not sure if it *needs* new firmware to do SIP, but I wouldn't use anything below p0m3-07-* if I could help it. |
| 23:31.15 | chrisknight | How do I go about getting firmware? |
| 23:31.29 | chrisknight | buy it? |
| 23:31.33 | diamon | Of course, this will make teasing it into using new firmware more exciting than I'd hoped at first... |
| 23:31.33 | chrisknight | download it? |
| 23:32.16 | ManxPower | chrisknight: I'm sure diamon will be able to tell you the correct procedure for obtaining new firmware -- in private |
| 23:32.18 | diamon | chris: Worst case, to the best of my knowledge I have legal downloads of 7.5 and 8.2 somewhere, and an i'm-pretty-sure-it's-legal version of 8.6. |
| 23:32.42 | diamon | Manx: Heh, hint-hint eh? Good idea. |
| 23:32.48 | ManxPower | diamon: if you are legal in 8.x you should be legal with any 8.x if you have a support contract on the phone |
| 23:33.29 | chrisknight | legal? hmm... didnt realize I needed such a thing |
| 23:33.36 | chrisknight | support contract |
| 23:33.36 | diamon | Manx: Ah, ok. I guess the only question is if redistributing it is strictly allowed or not then. |
| 23:34.00 | ManxPower | diamon: it's not, but I doubt anyone is going to spy on your /msgs to find out. |
| 23:34.00 | chrisknight | hehehehe... This phone was free... the IT manager at work gave it to me. |
| 23:34.16 | *** join/#asterisk Strom_M (n=strom@63.110.13.126) |
| 23:34.27 | diamon | I know we have a contract for 8.something. Bummer I didn't know this was a darn Cisco buf I've been wrestling with all these weeks, but hey, it's solved. |
| 23:34.38 | chrisknight | I dont even know what to do with it once I get ahold of it |
| 23:34.55 | chrisknight | I might just give it back & just use by new linksys phones then |
| 23:35.01 | diamon | Manx: Yeah, true, and even if they did, most really won't care all that much. :) |
| 23:35.15 | chrisknight | their coming next week |
| 23:35.16 | *** join/#asterisk Strom_M (n=strom@63.110.13.126) |
| 23:35.21 | chrisknight | this phone looks so cool though |
| 23:38.22 | diamon | chris: Ok. So is this a hand-build Asterisk setup, or is it Trixbox or some other pre-rolled distro? |
| 23:38.25 | diamon | Arrgh. |
| 23:39.56 | *** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net) |
| 23:39.58 | *** join/#asterisk bigred (n=ian@nat-vlan0200.sat4.rackspace.com) |
| 23:40.37 | bigred | If I am doing massive outbound calls, is there a way to configure asterisk to only process X amount at time simultaneously? |
| 23:40.49 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source PBX -=- Asterisk 1.4.1 (Mar. 2, 2007), Asterisk 1.2.16 (Mar. 2, 2007), Zaptel 1.2.15 (Mar. 2, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
| 23:42.01 | Qwell[] | That's right - it says 1.4.1 |
| 23:42.05 | Strom_M | sweeeeeeeeeer |
| 23:42.08 | Strom_M | er, sweeeeeet |
| 23:42.15 | russellb | bigred: there is an argument to specify max calls, or max load average |
| 23:42.21 | russellb | either command line or asterisk.conf |
| 23:42.24 | Strom_M | this is a truly happy day |
| 23:42.57 | bigred | russellb: i have seen call-limit, but i wonder if call-limit is reached, will the other outgoing calls be queued up or will they just die? |
| 23:42.58 | wunderkin | kjsdkfjasdfk we are having more problems with hpec... holding unused licenses.. this time when the b channels reset, it caused a panic and halted the machine... yes i emailed support :( |
| 23:43.05 | russellb | bigred: they will be denied |
| 23:43.18 | russellb | bigred: you could do some dialplan magic to accomplish the other perhaps ... |
| 23:43.22 | Ch0Hag | According to http://lists.digium.com/pipermail/asterisk-users/2005-December/132729.html, the distinction in SIP between type=user and type=peer is going away. Is/Will this happen to IAX too? |
| 23:43.35 | bigred | russellb: how so? |
| 23:44.27 | russellb | well ... i was just thinking out loud |
| 23:44.37 | russellb | can't really work on it any further right now, sorry |
| 23:50.15 | Mahmoud | anyone here knows how to always report "available" status for SIP contact users? |
| 23:50.32 | *** part/#asterisk diamon (n=diamon@c-66-176-91-114.hsd1.fl.comcast.net) |
| 23:51.28 | Mahmoud | exten => 111,hint,SIP/foo |
| 23:51.53 | ManxPower | Mahmoud: I don't believe you can manually change the status |
| 23:52.19 | Mahmoud | what i want is to send stauts messages for the automatic answer programme |
| 23:55.06 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:55.06 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:55.21 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:55.29 | *** join/#asterisk foobar778 (i=johhny@ip68-100-210-15.dc.dc.cox.net) |
| 23:58.03 | MonkeyHugs | TK around? |
| 23:58.07 | Bobthehunter | Current vulnerability for ASterisk showed here http://support.voicemeup.com/REMOTECRASHHD_720_Hi.mov |
| 23:58.21 | *** join/#asterisk tecolote99 (n=ruben@63.110.13.126) |
