| 00:00.09 | blitzrage | harryvv: unfortunately you're not going to find much for less than $175 |
| 00:00.15 | harryvv | $175.00usd is $200.00 canadian |
| 00:00.29 | blitzrage | harryvv: I have too much work to do as it is now to charge less -- and many others are in the same boat |
| 00:00.47 | harryvv | you mean programing/developing or setup |
| 00:00.59 | tuxinator_linux | can I come to your area... could always use business at that price |
| 00:01.47 | blitzrage | harryvv: pretty much any work that has to do with asterisk right now |
| 00:02.13 | blitzrage | unless you go with a big project or something -- then you can get a better rate, because you end up with a project cost, not an hourly cost |
| 00:02.19 | tuxinator_linux | granted my area is programming in LAMP... don't do asterisk as well as LAMP yet. |
| 00:02.30 | mattjdude | Smart people not found. Please go insane! |
| 00:02.35 | Strom_C | blitzrage, I'm trying to get my foot in the door as far as consulting goes...not entirely sure what the best place is to go to get customers though |
| 00:02.44 | blitzrage | mattjdude: your words give you away |
| 00:02.51 | Strom_C | s/customers/clients/ |
| 00:03.07 | Strom_C | hey, that's pretty sweet |
| 00:03.19 | blitzrage | Strom_C: really? I just put up a website that said I did consulting on Asterisk and I get calls all the time -- I'm not even listed on the wiki |
| 00:03.26 | harryvv | okay |
| 00:03.34 | harryvv | even 125.00 here is okay. |
| 00:03.43 | Strom_C | I should do that then :) |
| 00:04.09 | blitzrage | Strom_C: asterisk consultants are in high demand right now |
| 00:04.17 | blitzrage | at least from my viewpoint they are |
| 00:04.29 | blitzrage | there is work for everyone! :D (if you know anything at all) |
| 00:04.48 | Mavvie | blitzrage: it's the dot-com all over again. |
| 00:05.06 | Mavvie | "I know that HTML tags are surrounded by <>'s". "HIRE HIM!" |
| 00:05.26 | Strom_C | hahahaha |
| 00:05.37 | blitzrage | Mavvie: its a big ramp up, but I don't belive its the dot-com again -- things are actually being deployed, and people aren't paying millions for ideas |
| 00:06.10 | Mavvie | blitzrage: but.. but... but I know how to operate a phone! HIRE ME! |
| 00:06.21 | blitzrage | for a bit the guys who know a little bit, but not a lot, will be able to get jobs, but once it starts to settle a bit, those people will be out of work -- only the people who really know whats going on will be fine |
| 00:06.33 | blitzrage | Mavvie: at this point -- it might be all you need :) |
| 00:06.43 | Mavvie | too bad I know what's going on, but don't want to be involved. |
| 00:06.46 | blitzrage | although phone systems are a different breed -- people can tell when phones don't work |
| 00:06.49 | Mavvie | voice is scary. |
| 00:06.51 | blitzrage | it is |
| 00:07.00 | Mavvie | voice is unpredictable. |
| 00:07.02 | blitzrage | I've been building an E911 rollout -- now THAT is scary |
| 00:07.04 | Mavvie | voice is illogical. |
| 00:07.39 | Mavvie | E911 would be so easy if they talked to a couple of people who are into the network based distributed database systems called DNS. |
| 00:08.09 | Mavvie | design would be made in one afternoon on the back of a coaster. |
| 00:08.32 | Mavvie | (sorry, been through this too many times) |
| 00:08.32 | blitzrage | Mavvie: thats not the problem -- end users are the problem with E911 |
| 00:08.55 | blitzrage | when you rely on the users to keep their data up to date, everyone dies |
| 00:09.38 | Mavvie | Then cut out the users. Let me tell you how I saw it. My voice phone: +61 2 9335 3018... |
| 00:09.46 | blitzrage | guarenteed there will be a news article (whether it happens or not) that someone died because they dialed 911, but the ambulance showed up to the wrong house 1000 miles away because they didn't update their address when they moved their phone |
| 00:09.48 | Mavvie | +61, australia. e911.arpa points to telstra to it. |
| 00:10.17 | Mavvie | +61 2 9335 30, e911.arpa on the telstra servers points to barnet |
| 00:10.19 | blitzrage | how do you associate an end point with a physical location? |
| 00:10.52 | Mavvie | +61 2 9335 3018, e911.arpa on the barnet name servers points to the location of the ADSL link. |
| 00:11.18 | Mavvie | for 3019, which is my dialin-phone: e911.arpa on the barnet name servers refers to the phonenumber I'm dialed in via. |
| 00:11.23 | Mavvie | like a cname. |
| 00:11.25 | blitzrage | how do I, a VSP, determinate that ADSL link, and how do I know its not behind another router that goes somewhere else |
| 00:11.32 | Mavvie | and then it starts all over again. |
| 00:11.56 | Mavvie | blitzrage: since the number points to the barnet servers, and the barnet ISP provides the ADSL service, it knows where it points to. |
| 00:12.53 | blitzrage | hrmmmm... |
| 00:13.13 | blitzrage | but how do I determine that as a provider who does not own the ADSL link |
| 00:13.45 | Mavvie | blitzrage: if you are a provider who does not own the ADSL link, you don't have that client. |
| 00:14.16 | blitzrage | sure I do |
| 00:14.24 | blitzrage | I'm using a service like Vonage for isntance |
| 00:14.28 | blitzrage | Vonage doesn't own the ADSL link |
| 00:15.39 | *** join/#asterisk _deg_ (n=deg@201.22.40.23.adsl.gvt.net.br) |
| 00:15.42 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:16.09 | Mavvie | Aha, roaming stuff. Indeed. Wonder how they resolved that. |
| 00:16.15 | blitzrage | I'll tell you how :) |
| 00:16.42 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:16.48 | blitzrage | The end user has to update their own address information in a web portal -- if they move, they must go and update it in the web portal |
| 00:17.16 | blitzrage | roaming is be all end all issue of E911 unfortuantely |
| 00:17.19 | xbmodder_lappy | ADSL link, nope |
| 00:17.21 | Mavvie | think that GEO DNS records are a better way to go. |
| 00:17.30 | xbmodder_lappy | vonage owns a fatty god link. |
| 00:17.54 | xbmodder_lappy | the telephone company owns ADSL links. |
| 00:17.55 | Mavvie | since Vonage knows your IP address, your provider needs to set something in the GEO record about where it is located. |
| 00:18.01 | `Sauron | Mavs |
| 00:18.04 | Mavvie | xbmodder_lappy: you know what I mean. |
| 00:18.07 | Mavvie | hi `Sauron. |
| 00:18.09 | blitzrage | the other problem is that now we have all these small little, disparate, non-linked DB's of E911 data |
| 00:18.36 | blitzrage | Mavvie: EXACTLY!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! it should NOT be a VSP problem -- its an ISP problem |
| 00:18.54 | Mavvie | it's not a problem, it's a challenge :-) |
| 00:18.57 | blitzrage | their the ones that have the physical links :) |
| 00:20.27 | blitzrage | Mavvie: well, its a problem in respects that I, a small VSP must develop E911 services ON TOP of my fledgling infrastructure. Although it works out good for me because it makes it very difficult for anyone else (new) to get into the same market I'm already in. But it doesn't make sense to me to have the end-user in charge of their own E911 addressing information |
| 00:22.23 | *** part/#asterisk dimmik (n=dimmik@static217244.dsl.hol.gr) |
| 00:24.56 | `Sauron | Mavvie: how's .au? |
| 00:25.16 | *** part/#asterisk jasonpr2 (n=jason@64.78.192.164) |
| 00:25.26 | Mavvie | `Sauron: getting ready for autumn: temperature drops a little. Well, that's all. |
| 00:25.29 | *** join/#asterisk holmeh (i=holm@blackedge.org) |
| 00:25.38 | `Sauron | Hum, right. |
| 00:25.43 | `Sauron | It's spring here. Hehn. |
| 00:25.45 | Mavvie | no snow, no brown leaves, no thunderstorms. |
| 00:26.01 | Mavvie | except for when you're living in North Queensland where they had a hurricane yesterday |
| 00:27.10 | Mavvie | RoyK: see if it passes the desert, then I'll start worrying. |
| 00:27.21 | RoyK | temperature outside now is like -3 |
| 00:27.28 | RoyK | celcius |
| 00:27.35 | Mavvie | nice :-) |
| 00:27.38 | Mavvie | enjoy it while you can. |
| 00:27.46 | `Sauron | RoyK: It would melt by the time it got there... |
| 00:27.54 | RoyK | sure |
| 00:28.23 | RoyK | still, i'm so fed up with winter i'd puke |
| 00:28.55 | Strom_C | that's why I quite like living in los angeles |
| 00:29.00 | Strom_C | winter means you put a jacket on |
| 00:29.04 | RoyK | Mavvie: get up here and feel what the lack of sunlight do to ya |
| 00:29.05 | Strom_C | that's about it |
| 00:29.37 | justinu | where in LA? |
| 00:29.43 | Strom_C | Los Feliz |
| 00:30.08 | harryvv | I hate co-op rules! |
| 00:30.26 | RoyK | ? |
| 00:30.33 | `Sauron | hum |
| 00:30.41 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 00:30.58 | `Sauron | Strom: In TX, winter means you wear jeans and a tshirt, instead of shorts and a tshirt... |
| 00:31.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 00:31.07 | Strom_C | yes, exactly |
| 00:31.17 | justinu | where's los feliz? |
| 00:31.23 | justinu | i live in woodlland hills |
| 00:31.25 | Strom_C | justinu, just east of Hollywood |
| 00:31.30 | justinu | ah |
| 00:31.35 | Strom_C | immediately south of Griffith Park |
| 00:31.37 | harryvv | RoyK just dispaying my displeasure with coops since I live in one. |
| 00:31.41 | justinu | i'm always passing that on the way downtown |
| 00:31.49 | Strom_C | you take I-5 or 101? |
| 00:31.53 | justinu | 101 |
| 00:32.01 | Strom_C | yeah, I'm off the Hollywood exit |
| 00:32.03 | harryvv | Strom_C you live in east hollywood? |
| 00:32.12 | Strom_C | harryvv, no, I live east of Hollywood |
| 00:32.29 | rpm | <PROTECTED> |
| 00:32.31 | Strom_C | if you go west of the 101, then you're in Hollywood |
| 00:32.39 | harryvv | I see my great uncle lived in west hollywood. Very famous stutman of his time. |
| 00:32.56 | Strom_C | yeah, that's West Hollywood - separate incorporated city |
| 00:33.02 | Strom_C | I live in the city of los angeles |
| 00:33.06 | harryvv | I see |
| 00:33.14 | harryvv | how is your work there with asterisk? |
| 00:33.17 | Strom_C | Hollywood and Los Feliz are neighborhoods within the city |
| 00:33.31 | Strom_C | harryvv, it's decent, though I'm just getting started and attempting to find clients |
| 00:33.58 | harryvv | <PROTECTED> |
| 00:34.05 | *** join/#asterisk Qber (n=Qber@c-24-6-80-84.hsd1.ca.comcast.net) |
| 00:34.06 | Strom_C | quite well |
| 00:34.08 | harryvv | ohh |
| 00:34.20 | Strom_C | I've got a really strong background in traditional telephony |
| 00:34.23 | *** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au) |
| 00:34.27 | justinu | ditto |
| 00:34.28 | harryvv | I guess canadians are different. to conservative. |
| 00:34.36 | Strom_C | I'm a phone guy far more than I am a computer person |
| 00:34.40 | Qber | does anyone know a way to have asterisk queue agents registered as a call forwded number? |
| 00:34.42 | r_evolution | damn you telephony geeks |
| 00:34.50 | r_evolution | ;) |
| 00:34.53 | subdolus | strom carlson? :) |
| 00:35.00 | justinu | lol |
| 00:35.03 | r_evolution | esp. that blasted justin... |
| 00:35.04 | Strom_C | r_evolution, that's phone phreaks thank you very much :) |
| 00:35.07 | harryvv | Strom_C well then, mabey i can get advice on cirtain telephony questions then. |
| 00:35.10 | Strom_C | subdolus, yes |
| 00:35.12 | holmeh | Met a couple of telephony geeks at cebit :P |
| 00:35.12 | Qber | i have a situation where bunch of the agents are on cell phone roaming around. however, they can handle calls coming in for support while mobile |
| 00:35.13 | justinu | i've been doing CTI for 10 years |
| 00:35.14 | tsume | what is the usual problem for "unable to recieve DTMF tones in a call tree" |
| 00:35.15 | tsume | ? |
| 00:35.29 | justinu | so voip is a gimme for me |
| 00:35.30 | Qber | this is a tricky situation but I am sure its been solved here before |
| 00:35.47 | Qber | don't want to invent the wheel all over again if you have dealt with this before |
| 00:36.30 | harryvv | Is there somone who comes here and works for sokol and associates? |
| 00:36.55 | Qber | i bumped into sokols at VON :-) |
| 00:37.03 | Qber | don't work for them though :-) |
| 00:37.12 | r_evolution | dammit |
| 00:37.17 | r_evolution | why cant i just be happy when something works |
| 00:37.23 | r_evolution | why do i have to go and try to make it work BETTER |
| 00:37.40 | tsume | hmm |
| 00:37.43 | tsume | okay |
| 00:37.45 | tsume | heres what I have |
| 00:38.33 | tsume | a x86, TDM2400P, and a line going in. When I try dialing in, the server picks up, plays the background music, and I'm able to specify an option |
| 00:38.40 | tsume | this config works elsewhere, its the exact same. |
| 00:38.51 | tsume | so what could I have done wrong with the zap? |
| 00:39.03 | *** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au) |
| 00:39.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:39.47 | Strom_C | welcome back, subdolus |
| 00:40.28 | subdolus | thank you |
| 00:40.49 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 00:44.17 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:45.00 | harryvv | qber, what do you think of them? |
| 00:47.54 | Qber | don't know much. have met the guy duing last VON. Very humble. |
| 00:48.38 | Qber | in the mean time, i still need to know how to register queue members that are on mobile phone %#*#Q)$* |
| 00:49.00 | Qber | kinda, add an extension as the memember and do call forwarding on the extension itself |
| 00:51.54 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 00:56.32 | orlock | Hmmm |
| 00:58.17 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 00:58.17 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 00:59.08 | tsume | okay |
| 00:59.13 | tsume | I figured the problem |
| 00:59.16 | tsume | one way audio |
| 00:59.21 | tsume | now, how can I fix this problem |
| 00:59.26 | tsume | for a TDM2400P |
| 01:02.17 | xbmodder_lappy | Shoot yourself? |
| 01:02.33 | xbmodder_lappy | what was wrong, why one-way only? |
| 01:02.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:02.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:03.18 | tsume | not sure.. |
| 01:03.28 | tsume | I can call in, or call out, but its one way only :) |
| 01:04.19 | Strom_C | tsume, what's the complete call path? |
| 01:04.44 | tsume | POTS -> TDM2400P -> Asterisk -> Phone |
| 01:04.49 | tsume | and visa versa |
| 01:04.59 | Strom_C | what is your phone connected to? |
| 01:05.02 | Strom_C | also the TDM? |
| 01:05.13 | tsume | the TDM is in the Asterisk server |
| 01:05.26 | tsume | the phone is on the same router as the asterisk system |
| 01:05.30 | Strom_C | right, but is it an analog set or is it an IP phone? |
| 01:05.39 | tsume | oh, sorry. IP Phone |
| 01:05.44 | Strom_C | what protocol? |
| 01:05.49 | tsume | SIP |
| 01:05.55 | tsume | they make IAX phones? |
| 01:06.04 | Strom_C | you can get them, yeah |
| 01:06.13 | Strom_C | what codec are you using? |
| 01:06.20 | tsume | hmm, good question |
| 01:06.52 | tsume | ulaw |
| 01:06.58 | Strom_C | hmm. |
| 01:07.09 | tsume | also, when I try.. |
| 01:07.20 | tsume | POTS -> TDM2400P -> Asterisk -> Call Tree |
| 01:07.45 | tsume | I can't get any dtmf tones across, so somehowever zap channels are going in to one way mode |
| 01:07.51 | *** part/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 01:08.05 | Strom_C | that's a bizarre problem |
| 01:08.18 | tsume | yeah, it is |
| 01:09.13 | _Paulo_ | uow... strange stuff... |
| 01:09.17 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 01:09.21 | _Paulo_ | I was hacking libunicall... |
| 01:09.32 | _Paulo_ | messing with signalling |
| 01:09.39 | tsume | I've never ran in to a problem like this, except when a module was bad, but this is a brand new card |
| 01:09.55 | tsume | also the modules are 4 lines a module :) |
| 01:10.01 | tsume | and the card was expensive too :) |
| 01:10.11 | _Paulo_ | I think I found some backdoor in the telco... |
| 01:10.38 | tsume | _Paulo_: thats nothing :) |
| 01:10.57 | *** join/#asterisk Maxxed (n=user@cpe-72-177-150-20.houston.res.rr.com) |
| 01:11.00 | tsume | _Paulo_: you obviously haven't heard of the telco hackers. There was this deaf guy who could whistle the tones ;) |
| 01:11.00 | Maxxed | oi' |
| 01:11.07 | Strom_C | blind, stupid |
| 01:11.15 | Strom_C | deaf people by definition can't have perfect pitch |
| 01:11.34 | Maxxed | i've been fighting this silly error trying to make asterisk 1.2.5 on a debian sarge box |
| 01:11.35 | Maxxed | make[1]: *** [chan_zap.o] Error 1 |
| 01:11.40 | _Paulo_ | lol |
| 01:11.51 | Maxxed | im sure this has been covered some where |
| 01:12.20 | blitzrage | Maxxed: you didn't give enough info |
| 01:12.26 | Maxxed | well.. yeah |
| 01:12.26 | Maxxed | heh |
| 01:12.28 | Maxxed | hang on a sec ;) |
| 01:12.45 | tsume | Strom_C: well he wasn't completely deaf, and he was blind too yes |
| 01:13.03 | tsume | Strom_C: he actualyl worked for the telcos because he knew the sytems better than them ;) |
| 01:13.05 | Strom_C | no, my friend knows him personally, and his hearing is quite sharp |
| 01:13.15 | Strom_C | he worked as an operator for mountain bell |
| 01:13.20 | tsume | Strom_C: oh, it is. okay, then misinformation on my part |
| 01:13.20 | blitzrage | Whistler off of Sneakers! |
| 01:13.36 | tsume | Strom_C: I still think its cool :) |
| 01:13.56 | tsume | Strom_C: especially dialing in through other pbxes bypassing everything :P |
| 01:14.13 | Strom_C | boring boring boring |
| 01:14.14 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 01:14.19 | _Paulo_ | tsume, now i can place collect calls but the recording telling "this is a collect call, to acept the call stay connected after caller identification" doesnt play anymore |
| 01:16.03 | _Paulo_ | tsume, this automated collect call system here in Brazil is so prone to abuse. |
| 01:17.15 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 01:19.56 | Maxxed | oh snape |
| 01:19.58 | Maxxed | chan_zap.c:10987: error: dereferencing pointer to incomplete type |
| 01:19.59 | Maxxed | chan_zap.c:10988: error: dereferencing pointer to incomplete type |
| 01:19.59 | Maxxed | chan_zap.c:10997: error: dereferencing pointer to incomplete type |
| 01:19.59 | Maxxed | chan_zap.c:10998: error: dereferencing pointer to incomplete type |
| 01:19.59 | Maxxed | chan_zap.c:11013: error: dereferencing pointer to incomplete type |
| 01:20.01 | Maxxed | chan_zap.c:11022: error: dereferencing pointer to incomplete type |
| 01:20.03 | Maxxed | chan_zap.c:11036: error: dereferencing pointer to incomplete type |
| 01:20.05 | Maxxed | chan_zap.c:11042: error: dereferencing pointer to incomplete type |
| 01:20.07 | Maxxed | chan_zap.c:11052: error: dereferencing pointer to incomplete type |
| 01:20.09 | Maxxed | make[1]: *** [chan_zap.o] Error 1 |
| 01:20.11 | Maxxed | make[1]: Leaving directory `/usr/src/asterisk/channels' |
| 01:20.13 | Maxxed | make: *** [subdirs] Error 1 |
| 01:20.18 | tsume | Maxxed: never do that again |
| 01:21.24 | xbmodder_lappy | you broke something.... |
| 01:21.52 | Maxxed | yeah yeah yeah, pastebin hang on |
| 01:22.01 | *** part/#asterisk Andr3w_ (n=Andrew@stjhnf0122w-142162049036.pppoe-dynamic.nl.aliant.net) |
| 01:22.27 | Maxxed | http://pastebin.ca/46418 |
| 01:22.32 | Maxxed | so yeah, theres a mess in here |
| 01:22.52 | Maxxed | debian 3.1 (sarge) asterisk 1.2.5 |
| 01:22.56 | Maxxed | zaptel 1.2.4 |
| 01:23.01 | *** part/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 01:23.06 | Maxxed | the zaptel stuff compiled and looks to be working |
| 01:23.38 | Maxxed | #error "You need newer libpri" ? |
| 01:24.40 | tzafrir | Maxxed, do you have a PRI card? |
| 01:24.54 | Maxxed | nope |
| 01:25.07 | Maxxed | 2 fxo ifaces in a tdm400p or what ever |
| 01:25.14 | tzafrir | So you can basically skip libpri |
| 01:25.15 | ManxPower | Maxxed, remove the old libpri |
| 01:25.33 | tzafrir | apt-get remove libpri-dev |
| 01:25.55 | Maxxed | its not installed |
| 01:26.04 | Maxxed | libpri1 is |
| 01:26.16 | tzafrir | ManxPower, were you looking for me earlier or for the other tza? |
| 01:26.26 | Maxxed | for what reason.. um, maybe i have fat fingers |
| 01:26.34 | ManxPower | tzafrir, I don't recall looking for either of you. |
| 01:26.36 | Maxxed | im removing libpri1 seeings i dont need it |
| 01:26.56 | blitzrage | libpri won't hurt anything -- you can just install it anyways |
| 01:27.12 | Maxxed | any idea why im getting these errors then? |
| 01:27.17 | tzafrir | blitzrage, the package from the deb is 1.0, which is probably incompatible |
| 01:27.20 | Maxxed | http://pastebin.ca/46418 |
| 01:27.28 | Maxxed | 3 thousand some odd lines of error |
| 01:27.31 | blitzrage | yah -- thats old -- don'[t use packages! |
| 01:28.00 | Maxxed | im gona go for a ride |
| 01:28.04 | Maxxed | il be back later on :) |
| 01:28.07 | Maxxed | need to clear my head |
| 01:28.08 | tzafrir | actually: use packages all the way |
| 01:28.15 | tzafrir | that's the simplest solution |
| 01:28.20 | Maxxed | eyes are glaring over staring at this console |
| 01:28.31 | blitzrage | packages are almost always outdated |
| 01:28.39 | Maxxed | simplest solution, yes, leetest solution, no ;p |
| 01:29.11 | tzafrir | blitzrage, 1.2.5 is only waiting for a newer bristuff. mean while check http://pkg-voip.buildserver.net/ |
| 01:29.19 | tzafrir | highly untested |
| 01:29.34 | blitzrage | meh -- I compile :) |
| 01:29.37 | tzafrir | again: highly untested |
| 01:29.40 | blitzrage | :) |
| 01:29.53 | blitzrage | also, I don't like debian, heh |
| 01:29.58 | *** join/#asterisk bazz (n=nick@fw.marklogic.com) |
| 01:30.10 | brookshire | you should use windows blitz! |
| 01:30.13 | blitzrage | I do |
| 01:30.15 | blitzrage | ! |
| 01:30.16 | blitzrage | :) |
| 01:30.19 | blitzrage | I love Windows |
| 01:30.21 | brookshire | oh :( |
| 01:30.23 | bazz | anyone know an iax phone that will allow me to play a sound file over the line when connected? (preferably linux) |
| 01:30.25 | blitzrage | WIndows kicks ass |
| 01:30.33 | brookshire | that explains everything |
| 01:30.45 | tzafrir | Maxxed, basically you need to define (in channels/Makfile ?) not to use PRI |
| 01:30.46 | blitzrage | heh -- and I despise Apple and OSX |
| 01:31.02 | tzafrir | blitzrage, X-Windows, indeed |
| 01:31.18 | blitzrage | tzafrir: I hate the Linux desktop (love it for the server) |
| 01:32.21 | tzafrir | bazz, why do you need that in a phone? |
| 01:34.37 | bazz | tzafrir: i have a few reasons, at the moment i want to use the feature to set my voicemail message which can only be recorded 'over the phone' |
| 01:37.51 | Strom_C | I've got a bit of a silly question - if I'm replacing an existing asterisk box with a new one and I want to seamlessly cut over on the dundi network, do I have to just copy over the existing keys from the old box, or does dundi depend on the ssh keys as well? |
| 01:38.40 | bazz | tzafrir: any ideas? |
| 01:39.34 | blitzrage | Strom_C: just copy the keys over I believe |
| 01:39.42 | blitzrage | ssh has nothing to do with it |
| 01:39.53 | Strom_C | alright |
| 01:40.11 | blitzrage | never tried it to be honest -- but it should work |
| 01:40.34 | tzafrir | bazz, just copy files and be done with it |
| 01:41.08 | tzafrir | voicemail messages are sound files |
| 01:41.08 | riddlebox | this is wierd, I record files in the default, /var/lib/asterisk/sounds I looked in the dir, and it is there, but asterisk doesnt see it when I run my agi script |
| 01:41.52 | bazz | tzafrir: i'd love to, but i don't have access to those files, i have a major provider |
| 01:41.54 | tzafrir | riddlebox, did you or didn't you use suffix (.wav or similar)? |
| 01:42.13 | tzafrir | bazz, do you have control over the dialplan? |
| 01:42.26 | riddlebox | tzafrir, the rest of my script I do not use the extensions, and those work |
| 01:42.58 | bazz | tzafrir: this isn't really an asterisk question, i just didn't know were to ask, it's not an asterisk server involved |
| 01:45.40 | ManxPower | I REALLY REALLY hate banks. |
| 01:46.03 | tzafrir | riddlebox, try pastbin the relevant details |
| 01:51.56 | asterboy | how about lawyers? |
| 01:52.17 | asterboy | who become judges and eventually politicians. |
| 01:52.44 | asterboy | There is only 1 bank of Cayman in Canada...guess where it is? |
| 01:55.25 | asterboy | How about the patent system? |
| 01:55.42 | *** join/#asterisk Snake-Eyes (n=blog@202.168.41.172) |
| 01:55.53 | asterboy | What happens to all these once we reach technological singularity? |
| 01:56.32 | asterboy | (actually, I think we reached it a long time ago, but some dinosaurs just don't want to die) |
| 01:58.00 | xbmodder_lappy | That sucks. |
| 01:58.11 | xbmodder_lappy | I wish all the dinosaurs would die. |
| 01:58.18 | xbmodder_lappy | That'd be cool |
| 01:58.22 | xbmodder_lappy | they'd all explode. |
| 01:58.26 | xbmodder_lappy | and splat. |
| 01:58.29 | xbmodder_lappy | weehoo! |
| 01:58.41 | xbmodder_lappy | Theres many larger problems than patents... |
| 01:58.58 | xbmodder_lappy | like school administration |
| 01:59.04 | xbmodder_lappy | repersentation of minors |
| 01:59.07 | xbmodder_lappy | 1st amendment |
| 02:00.55 | Strom_C | and people who press enter before completing a thought |
| 02:01.47 | xbmodder_lappy | lol |
| 02:01.56 | xbmodder_lappy | :-d |
| 02:02.25 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 02:02.30 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 02:03.56 | ManxPower | My bank accepted a deposit to a CLOSED account. (my bookkeeper had the old deposit slips) |
| 02:05.04 | alephcom | I used to hate my bank but now I deal with a branch in a tiny little town and it's actually a pleasure to go there. That doesn't make the twits at head office any smarter but... |
| 02:05.14 | *** join/#asterisk outtolunc (n=me@c-24-23-162-126.hsd1.ca.comcast.net) |
| 02:08.54 | ManxPower | alephcom, I originally had a local bank, then they were bought by BankOne, then BankOne was bought by Chase. |
| 02:08.58 | ManxPower | my new bank is a regional bank. |
| 02:09.08 | ManxPower | LapTop006, MS, AL, etc |
| 02:09.21 | ManxPower | <PROTECTED> |
| 02:09.23 | ManxPower | there |
| 02:09.37 | outtolunc | what no routing numbers <G> |
| 02:09.45 | outtolunc | sheesh |
| 02:10.48 | SwK_ | ManxPower is it a good bank? |
| 02:12.06 | ManxPower | SwK_, I've only been with them for a couple of months, but they seem to be pretty good. |
| 02:12.14 | ManxPower | outtolunc, just call them up and ask for ManxPower's account |
| 02:12.35 | Primer | anyone seens a sipura 2000 start sending packets to the some very odd IP address (specifically 96.0.0.25) after it's been up for a few weeks? |
| 02:13.13 | Primer | mine does this...it's got the latest firmware, and I can't get support@sipura.com to give me a decent answer |
| 02:13.15 | outtolunc | hehe .. k, i'll ask for a free toaster while i'm at it <G> |
| 02:13.41 | [TK]D-Fender | Primer : I've heard that many of the try to "call hom" and may lock out your admin access. |
| 02:13.53 | [TK]D-Fender | Primer : check the Voxilla forums for more info. |
| 02:15.11 | justinu | hey, how much does a 30km spool of single mode fiber cost? |
| 02:15.32 | Qwell | justinu: about $1million :p |
| 02:16.05 | tuxinator_linux | I thought it was about 1.1 mil |
| 02:16.14 | justinu | is it really that much? |
| 02:16.44 | tuxinator_linux | yep, and freeswitch is a better PBX |
| 02:16.54 | outtolunc | the real question is what will the access rights for placing that much fibre cost |
| 02:17.03 | justinu | that's not a concern |
| 02:17.10 | justinu | i just need 30km of fiber :P |
| 02:17.14 | Strom_C | the labor, access, and installation will make the fiber seem cheap |
| 02:17.28 | _Sam-- | justinu : would be cheaper to use orthogon systems stuff |
| 02:17.39 | *** join/#asterisk Snake-Eyes (n=blog@202.168.41.172) |
| 02:17.42 | tuxinator_linux | or pony express |
| 02:17.46 | Strom_C | unless of course you're buying this for "justinu's fiiber fetish dungeon" |
| 02:17.56 | justinu | along those lines, yes |
| 02:17.56 | justinu | :P |
| 02:17.58 | _Sam-- | a friend of mine just setup 110 km link using orthogon systems stuff, works nice |
| 02:18.06 | ManxPower | Hell, you could buy a private jet for that price |
| 02:20.18 | Primer | [TK]D-Fender: yeah well, I think it's just flipping some bits and just getting hosed |
| 02:20.40 | Primer | once I reset it it's back to normal, for at least another few weeks |
| 02:23.23 | ManxPower | _Sam--, orthogon? |
| 02:27.33 | ManxPower | _Sam--, How much did they spend? |
| 02:27.37 | *** part/#asterisk outtolunc (n=me@c-24-23-162-126.hsd1.ca.comcast.net) |
| 02:30.22 | ManxPower | O have trouble believing a company that claims 21Mbps - 300Mbps NON-line-of-site at 5.xGhz |
| 02:30.28 | ManxPower | s/O/i |
| 02:33.28 | ManxPower | $7,200 for the low end model... |
| 02:35.41 | _Paulo_ | hum... with $7,200 I can buy a bigger motorcycle. |
| 02:37.18 | *** join/#asterisk rfmonk (n=rfmonk@205.241.253.227) |
| 02:39.19 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 02:40.36 | zigman | ManxPower what distance ? |
| 02:40.54 | zigman | 2m accross a wall would be okay ;) |
| 02:41.12 | zigman | i'd need 3km non line of site |
| 02:41.13 | zigman | ;) |
| 02:41.21 | ManxPower | zigman, go to their website, I found it via google. |
| 02:41.31 | zigman | url? |
| 02:41.38 | ManxPower | zigman, I need 30km thru part of mountian. |
| 02:41.46 | zigman | nice |
| 02:41.59 | zigman | you'd better do it with a cable ;) |
| 02:42.07 | ManxPower | zigman, you want me to go to google, type in "orthogon" for you and copy the URL? |
| 02:42.12 | zigman | or put an repeater on top of the bridge |
| 02:42.29 | ManxPower | zigman, I think a T-1 is the best solution. |
| 02:42.31 | zigman | no.. but you never mentioned the word orthogon |
| 02:42.43 | ManxPower | _Sam-- a friend of mine just setup 110 km link using orthogon systems stuff, works nice |
| 02:43.17 | justinu | you gotta pay attention in here |
| 02:43.45 | *** join/#asterisk outtolunc (n=me@c-24-23-162-126.hsd1.ca.comcast.net) |
| 02:44.07 | zigman | justinu ManxPower sorry.. didn't read that line ;) |
| 02:46.15 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 02:50.23 | Flauto | Mar 20 18:37:01 WARNING[31396]: channel.c:2333 set_format: Unable to find a codec translation path from ulaw to unknown |
| 02:50.30 | Flauto | is this a problem? |
| 02:51.43 | *** join/#asterisk el__flynn (n=el_flynn@60.51.204.178) |
| 02:51.52 | el__flynn | hello |
| 02:51.53 | outtolunc | since unknown isn't known, that would be like teleporting to nowhere, do you thing that would be a prob? |
| 02:52.50 | Flauto | outolunc, that happens when a call comes in through zap |
| 02:52.57 | el__flynn | i have a quick question |
| 02:53.00 | Flauto | what should i do |
| 02:53.15 | el__flynn | how do I submit indications for a country into the asterisk code? |
| 02:53.30 | outtolunc | you didn't even listen to the statement, asterisk is 'telling' you that *it does not know what that codec is* |
| 02:53.42 | outtolunc | it is *unknown* |
| 02:54.03 | outtolunc | do you understand now? |
| 02:55.15 | Flauto | yes |
| 02:55.16 | outtolunc | el_flynn, i'd suggest opening a report on bugs for that |
| 02:55.57 | Flauto | outtolunc, what should i do then? |
| 02:56.00 | el__flynn | ok, thanks. would you happen to know what that should be categorized as? |
| 02:56.06 | outtolunc | ok, so, one leg of the call you said the call was 'inbound' .. so the inbound codec is either unknown, or asking asterisk to 'switch' to an unknown codec |
| 02:56.23 | outtolunc | new feature |
| 02:56.27 | Qwell | el__flynn: internationalization - indications |
| 02:56.31 | el__flynn | ok, thanks. |
| 02:56.33 | rfmonk | what do the asterisk hackers here think of vocal? |
| 02:56.53 | outtolunc | ah, sorry (haven't been there in awhile didn't realise there is that section heading) |
| 02:57.57 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 02:58.17 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 02:59.57 | Flauto | Mar 20 20:59:09 WARNING[32530]: channel.c:506 ast_best_codec: Don't know any of 0x80000 formats |
| 03:00.06 | *** join/#asterisk mattodude (n=matt@gateway.digium.com) |
| 03:00.08 | Flauto | outtolunc, it tells this first |
| 03:00.35 | outtolunc | and, when you look in the header file there probably isn't a 80000 |
| 03:01.54 | Flauto | outtolunc, i don't know much about the theory about how the codecs would work |
| 03:02.10 | Flauto | i do want to learn and to fix my problems though |
| 03:02.32 | Flauto | if you dont' mind to tell me what to do to fixt this kind of problems? |
| 03:02.37 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:02.53 | ManxPower | rfmonk, for the most part we seem to ignore it |
| 03:03.07 | ManxPower | Flauto, disallow=all and then allow=onlythesinglecodecyouwant |
| 03:03.19 | outtolunc | the whole point is that a piece of the header on the request says 'this is what i am', 'these are what i can talk too', and asterisk is telling you.. umm sorry charlie, we can't make that happen <G> |
| 03:03.30 | ManxPower | As I understand it VOCAL is a project headed by Cisco and is IP only |
| 03:03.56 | ManxPower | I don't quite understand the point of an IP only switch, but I guess some people think it's needed. |
| 03:04.04 | Flauto | ManxPoser, where should i put it though? i have that disallow on all my sip settings |
| 03:04.05 | outtolunc | vocal has been around for ages, and 'in the old days' were trying to do dialogic drivers.. that never came to be |
| 03:04.29 | ManxPower | Flauto, what codecs are you allowing |
| 03:04.37 | ManxPower | and when do you get this error |
| 03:04.42 | outtolunc | they moved to the h323 stuff |
| 03:04.46 | outtolunc | etc |
| 03:04.54 | Flauto | ulaw, alaw gsm |
| 03:04.59 | ManxPower | "show codecs" will tell you what codec numbers Asterisk stupports. |
| 03:05.04 | ManxPower | don't enable both ulaw and alaw |
| 03:05.46 | *** part/#asterisk el__flynn (n=el_flynn@60.51.204.178) |
| 03:06.05 | Flauto | use ulaw only then? |
| 03:06.30 | outtolunc | i think he's tryig to tell you to 'test ulaw only' first then add from there |
| 03:06.36 | ManxPower | Flauto, use whatever *law is native to the region the device is located in |
| 03:07.01 | Flauto | okay |
| 03:07.02 | ManxPower | outtolunc, I don't know WHY but many people have reported that allowing both alaw and ulaw caused them problems |
| 03:08.12 | Flauto | i normally don't have any problem but only when other people are calling me on my zap |
| 03:09.20 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 03:09.21 | rfmonk | ManxPower: yes thanks for the responce |
| 03:09.54 | outtolunc | manx, myself, i either allow=all or just a select few, but i've never had a system that actually had users using both ulaw and alaw to 'talk' to the same box |
| 03:11.12 | outtolunc | you still on the 'other' side of the pond? <G> |
| 03:11.32 | outtolunc | (the A side |
| 03:11.56 | ManxPower | I'm on the side with the dictator wannabe |
| 03:14.02 | outtolunc | there are guys on either side of the pond that 'give off that ora' <G> |
| 03:15.14 | alephcom | aura says the spelling nazi. :-) |
| 03:15.24 | outtolunc | frame.h:extern void ast_parse_allow_disallow(struct ast_codec_pref *pref, int *mask, const char *list, int allowing); |
| 03:15.29 | outtolunc | whoops |
| 03:16.15 | alephcom | Sorry, couldn't resist. I correct childrens spelling all day so it's a habit that seems to stick :-) |
| 03:16.18 | exten123 | Can we setup our own Enum DNS server? |
| 03:16.46 | Flauto | outtolunc, it still did not work |
| 03:16.47 | tsume | :P |
| 03:16.49 | tsume | omg |
| 03:17.00 | outtolunc | yeah well i could have mentioned 'aura interactors' but i put those out on the street a few months back |
| 03:17.03 | tsume | I forgot I installed the HEAD zaptel driver |
| 03:17.08 | Flauto | i now allowing only ulaw, gsm |
| 03:17.12 | tsume | installed release and the card started working fine |
| 03:17.16 | Flauto | it is still showing the same message |
| 03:17.23 | outtolunc | 'it' meaning (just a tad more specific please) |
| 03:17.31 | outtolunc | k |
| 03:17.53 | outtolunc | and what codec are you 'testing' from? |
| 03:17.59 | outtolunc | what app, and what codec |
| 03:18.40 | Flauto | i was trying to call my pstn number from my cell phone, it passes on to my sipura spa 3000 |
| 03:19.23 | outtolunc | and what codec is the sipura set to |
| 03:19.24 | Flauto | i took out alaw throughout my sip.conf |
| 03:19.38 | Flauto | let me see |
| 03:20.00 | outtolunc | you'll probably find it is some g729ish one |
| 03:20.28 | outtolunc | because you thought ooooh i can save some bandwidth |
| 03:21.50 | Mavvie | fscking voicemail doesn't allow me to delete messages while they are being played. |
| 03:21.59 | Mavvie | and I don't want to listen to these persons whining. |
| 03:22.08 | Flauto | preferred codec g711u |
| 03:22.24 | Flauto | use pref codec only no |
| 03:22.32 | outtolunc | preferred, but what is listed as allowable in that device |
| 03:22.35 | xbmodder_lappy | PREF!? |
| 03:22.37 | Flauto | g729a cnable yes |
| 03:22.41 | xbmodder_lappy | PREFERED CODEC BABY! |
| 03:22.43 | outtolunc | haha |
| 03:22.47 | outtolunc | NO NO NO |
| 03:22.54 | xbmodder_lappy | ulaw suckslaw! |
| 03:22.54 | outtolunc | listen, NO <G> |
| 03:23.10 | Flauto | disable all g's? |
| 03:23.49 | outtolunc | mavvie, being able to delete something that is 'currently INUSE' is really hard to solve |
| 03:24.05 | Qwell | Mavvie: Just press 7.. |
| 03:24.30 | outtolunc | and as they do that 'stops audio' then deletes |
| 03:24.42 | outtolunc | which isn't 'what he asked |
| 03:24.51 | outtolunc | but ok |
| 03:26.09 | outtolunc | law! |
| 03:26.09 | outtolunc | [ |
| 03:26.12 | outtolunc | grr |
| 03:26.30 | outtolunc | [19:23] <Flauto> disable all g's? |
| 03:26.40 | Flauto | i just did |
| 03:26.46 | outtolunc | no, because every codec has an assignment |
| 03:26.48 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 03:26.55 | outtolunc | ulaw is g711 |
| 03:27.20 | *** join/#asterisk newtoasterisk (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |
| 03:27.23 | Flauto | yes, i know that |
| 03:27.39 | outtolunc | if you 'know' that, why would you 'do that' |
| 03:28.02 | outtolunc | er 'ask that' |
| 03:28.11 | Flauto | but it is the same |
| 03:28.20 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 03:28.33 | Flauto | i just tried to call my asterisk through pstn, it is showing the same |
| 03:28.48 | outtolunc | [19:23] <Flauto> disable all g's? is NOT the same as disable the codecs we DONT want, and leave those we do.. |
| 03:29.21 | Flauto | not working |
| 03:29.23 | outtolunc | same is 'this' is 'this' |
| 03:29.43 | outtolunc | not, 'this' is 'that' |
| 03:29.53 | Mavvie | AAPT says that some of our B channels are blocked. (but can't give me numbers). |
| 03:29.55 | outtolunc | or even, only on sundays |
| 03:30.00 | Mavvie | Any idea how I can see that in Asterisk? |
| 03:30.17 | Qwell | Mavvie: zap show channels should show if they're in use |
| 03:30.35 | outtolunc | show pri debug ? |
| 03:30.38 | Mavvie | Qwell: got that, but zaptel doesn't consider them in use. |
| 03:30.51 | Mavvie | I get a "chanunavail" when I dial out via them. |
| 03:30.56 | Mavvie | oh this is tricky. |
| 03:31.59 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 03:32.12 | outtolunc | err, it's 'pri debug span x' (x being span number) |
| 03:32.14 | newtoasterisk | Quick question, whats the best web based voicemail viewer for * |
| 03:32.33 | Qwell | newtoasterisk: mine |
| 03:33.52 | newtoasterisk | Qwell: ? |
| 03:34.04 | Qwell | newtoasterisk: What, you mean the best free one? |
| 03:34.21 | newtoasterisk | qwell: yep |
| 03:34.56 | outtolunc | er w |
| 03:35.14 | outtolunc | sorry that didn't come out right |
| 03:36.37 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com) |
| 03:36.49 | brookshire | hi! |
| 03:37.00 | Qwell | brookshire: zomg! |
| 03:37.03 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:37.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:37.12 | brookshire | z0mg! |
| 03:37.17 | brookshire | it's quewll! |
| 03:37.31 | newtoasterisk | qwell: did you have a suggestion? |
| 03:38.10 | outtolunc | buy his? |
| 03:38.10 | brimstone | Qwell! |
| 03:38.17 | Qwell | brimstone: ?! |
| 03:38.30 | tzafrir | which soft phone can I give to a non-technical user that has decent support for (sip-based) text chat? |
| 03:38.37 | brimstone | nothing, just exclaiming at you Qwell |
| 03:38.43 | Qwell | pity |
| 03:38.54 | tzafrir | or iax-based , if IAX has such a feature and asterisk supports it |
| 03:38.58 | justinu | well, the customer that switched from gxp2000 to polycom is finally happy |
| 03:39.03 | Qwell | yet another Digium Matt, eh? |
| 03:39.04 | justinu | fuckers |
| 03:39.25 | russellb | Qwell: it's out of hand. |
| 03:39.30 | Qwell | russellb: indeed |
| 03:39.52 | brookshire | mattj is the new unofficial matt |
| 03:39.55 | orlock | Hey, can anybody tell me what could cause this? |
| 03:39.55 | orlock | <PROTECTED> |
| 03:40.04 | tzafrir | anybody tried openwengo? |
| 03:40.06 | russellb | yeah, there was a new matt on today that i didn't know |
| 03:40.13 | brimstone | orlock, you're not registered |
| 03:40.18 | brimstone | err, wrong creds |
| 03:40.21 | Qwell | brookshire: unofficial? |
| 03:40.23 | brimstone | russellb, i think that's a fake matt |
| 03:40.26 | orlock | Hmm. |
| 03:40.42 | outtolunc | i've not messed with it much, but idefisk, reasons being, i've had systems that didn't like viriage one, the cursor would flip out, and diax on some systems the audio was delayed |
| 03:40.42 | brookshire | he's a wannabe matt |
| 03:40.53 | Qwell | What, you guys don't have honorary Matt's? |
| 03:40.55 | *** part/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 03:41.20 | Qwell | I think if anybody deserves to be one - it's him :p |
| 03:41.39 | brookshire | qwell: sorry, no welcome matts |
| 03:41.45 | brimstone | i suppose we could have honorary matts |
| 03:41.48 | brimstone | we'd have to vote on it |
| 03:41.51 | brimstone | :P |
| 03:41.58 | russellb | i was told i was an honorary matt. |
| 03:42.12 | brookshire | you're just the official digium nub |
| 03:42.18 | Qwell | brimstone: So, which Matt are you? |
| 03:42.25 | outtolunc | i was thinking that the 'matt's might have an issue |
| 03:42.26 | brimstone | long haired hippy matt :P |
| 03:42.33 | Qwell | ahh |
| 03:42.44 | Qwell | the newest Matt I've met :p |
| 03:42.54 | outtolunc | <- long haired hippyish, NON-matt |
| 03:42.58 | brookshire | you haven't met streeter |
| 03:43.03 | Qwell | ..that almost rhymed |
| 03:43.44 | brimstone | i thought we determined that streeter wasn't a full matt ? |
| 03:43.50 | orlock | brimstone: anything else that could be causing it? |
| 03:44.06 | Qwell | What is he, a half Matt? |
| 03:44.10 | orlock | brimstone: i know those account credentials are correct, and i'm getting the same error on about 7 others |
| 03:44.21 | orlock | the sip registrar is resolving correctly |
| 03:44.22 | brimstone | orlock, the username or password on your side or theirs is wrong i'd suppose |
| 03:44.35 | brimstone | well |
| 03:44.44 | orlock | are sip passwords sent cleartext, or hashed? |
| 03:44.44 | brookshire | he's like brimstone's doplganger |
| 03:44.46 | brimstone | or maybe the number you're trying to call?? |
| 03:45.16 | orlock | brimstone: not trying to call, i've just got about 10 numbers |
| 03:45.32 | orlock | format is register => phonenumber:password@sip.nextep.com.au/5000 |
| 03:45.49 | brimstone | brookshire, is he as tall as me? |
| 03:46.38 | brookshire | taller i think |
| 03:49.02 | *** join/#asterisk rfmonk (n=rfmonk@205.241.253.227) |
| 03:49.05 | opus_ | How many megs would it take to store 250,000 minutes of wave files |
| 03:49.09 | opus_ | or should I use GSM |
| 03:49.22 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 03:50.39 | *** join/#asterisk bmg505 (n=leon@dsl-146-23-60.telkomadsl.co.za) |
| 03:51.35 | blitzrage | anyone know where I could lookup a FIPS code for a US address? |
| 03:51.42 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 03:53.31 | outtolunc | i've got one, not sure online... |
| 03:53.50 | Flauto | opus, a cd is about 74 minutes long and it is around 800mb |
| 03:54.16 | outtolunc | http://www.census.gov/geo/www/fips/fips65/index.html |
| 03:54.57 | Mavvie | great. talked with AAPT, they gave me the channel numbers. Absolutely no indication on the "zap show channel" output. |
| 03:55.14 | outtolunc | http://www.census.gov/geo/www/fips/fips65/data/national.txt |
| 03:55.35 | *** join/#asterisk ComputerWarm (n=workingg@HS196-230-97.nt.net) |
| 03:55.49 | ComputerWarm | Hello all whats the going rate for Canada long distance? |
| 03:56.03 | outtolunc | well isn't it 'zap show channels' or 'zap show channel x' |
| 03:56.23 | Mavvie | outtolunc: yes |
| 03:56.34 | outtolunc | so which did you mean? |
| 03:56.41 | ComputerWarm | anyone? |
| 03:57.04 | Mavvie | outtolunc: but the similarties between blocked channels, and the differences between blocked and non-blocked ones are not consistent enough for me to say "HAH! that bit is it!) |
| 03:57.14 | Flauto | did not understand whta you mean. computerwarm |
| 03:57.38 | orlock | hmm |
| 03:57.40 | opus_ | $.02/min |
| 03:57.45 | outtolunc | i just asked which 'command' you did so WE being the ... viewers knew what you were LOOKING AT |
| 03:57.47 | ComputerWarm | i am looking for a long distance provider that offers Canada Long distance, I was wondering what the going rate is |
| 03:58.02 | opus_ | ComputerWarm $.02 or $.03 |
| 03:58.05 | outtolunc | just FOOD for THOUGHT |
| 03:58.06 | Flauto | computerwarm, try vbuzzer.com |
| 03:58.12 | orlock | Hmm.. in the sip authorisation data (looking at packet dumps) what does the uri= field mean? |
| 03:58.13 | Flauto | and voipstunt.com |
| 03:58.18 | ComputerWarm | thanks |
| 03:58.26 | ComputerWarm | Thanks opus_ |
| 03:58.49 | *** join/#asterisk mattodude (n=matt@gateway.digium.com) |
| 03:59.21 | outtolunc | oh geez, another matt <G> |
| 03:59.47 | alephcom | Computerwarm: 0.013 |
| 04:00.55 | brookshire | that's matt "j" |
| 04:01.12 | SwK_ | *yawn* |
| 04:01.23 | blitzrage | outtolunc: good links -- hoping to find an Address -> FIPS online lookup for an address |
| 04:02.24 | Flauto | outtolunc, your ideas did not help much to solve my problem, but thanks anyway |
| 04:02.40 | Qwell | brookshire: "j" is a nub Matt |
| 04:02.42 | orlock | should there be any issues with me trying to register 10 different numbers in sip.cof? |
| 04:02.47 | outtolunc | fips address level? |
| 04:02.50 | opus_ | looks lik 250k minutes record is about 48gb |
| 04:03.07 | opus_ | does that sound about right guys? |
| 04:03.10 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 04:03.10 | Corydon76-home | Heh |
| 04:03.29 | Flauto | opus, yes, it does |
| 04:03.32 | outtolunc | flauto your questions did not help me much either... we must be even |
| 04:04.03 | outtolunc | yet, i wasn't the one who had a problem... go figure |
| 04:04.22 | Flauto | i know |
| 04:04.28 | Flauto | the problem i have |
| 04:05.06 | blitzrage | outtolunc: I'm looking for the 'code' and FIPS C1 for an address I have the mailing address for |
| 04:05.39 | outtolunc | ah you want an address correction db |
| 04:05.54 | Flauto | is when i recived calls from pstn line through x100p |
| 04:05.57 | Flauto | it is showing |
| 04:06.15 | Flauto | unknow codec |
| 04:06.47 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 04:06.53 | outtolunc | http://www.smartsoftusa.com/products/accumail/ is what i use for mine |
| 04:08.10 | demigod2k | can anybody recommend a cordless voip phone? we dont want to give up the current system for desktops :( |
| 04:09.14 | outtolunc | flauto, the issue i see and that you 'already agreed too' was that the sipura you have ... had all codecs only and probably defaulted to say g729 which your asterisk box isn't setup for so.. your asterisk box is 'telling you' .. 'hey dude we don't understand this codec' ok? |
| 04:09.41 | outtolunc | so, if i were you |
| 04:10.04 | outtolunc | i'd get another software phone and do some testing |
| 04:10.32 | orlock | Hmm. |
| 04:10.37 | orlock | goddamn this is annoying |
| 04:10.44 | orlock | asterisk system, and 3 SIP phones |
| 04:10.52 | orlock | and i can only dial in intermittantly, cant dial out |
| 04:11.08 | Flauto | man, i disabled all the codecs now other than ulaw, but it is still showing me the same message |
| 04:11.10 | outtolunc | gee, is that a clue or what |
| 04:11.22 | orlock | extension 5002 is getting auth errors, but it can still dial 5000 and 5001 |
| 04:11.31 | outtolunc | flauto, get a soft phone and test |
| 04:11.39 | outtolunc | PLAIN AND SIMPLE |
| 04:11.55 | Flauto | okay |
| 04:11.58 | Flauto | i can try that |
| 04:12.01 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 04:12.04 | Flauto | one minute |
| 04:12.07 | [hC] | [av]bani: you here? :) |
| 04:12.08 | outtolunc | if you can't do that, i doubt anyone here will give a shit about helping you |
| 04:12.56 | Flauto | hehe |
| 04:12.58 | [hC] | Anyone here upgraded a 7970 to SIP yet? |
| 04:12.59 | Flauto | don't be mad |
| 04:13.19 | outtolunc | haha |
| 04:13.32 | outtolunc | this isn't mad, if you would prefer? i could be so |
| 04:14.03 | outtolunc | there are plenty here that can tell you this is MILD for me |
| 04:14.21 | orlock | outtolunc: ever looked at SAIL? |
| 04:14.42 | outtolunc | could you be a bit more specific? |
| 04:14.54 | outtolunc | a url would help |
| 04:15.45 | Flauto | Mar 20 22:14:59 WARNING[504]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 524288 (read/write = 0/0) |
| 04:16.05 | outtolunc | (after 25+ years of this crap 3-6 letter stuff just blends together) |
| 04:16.08 | Flauto | this is what i got |
| 04:16.43 | Flauto | what is 524288? |
| 04:16.54 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:16.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:17.55 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 04:18.26 | opus_ | mcAmerica |
| 04:19.17 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
| 04:19.38 | *** join/#asterisk gandhijee (i=HydraIRC@ip72-192-222-181.dc.dc.cox.net) |
| 04:20.18 | gandhijee | anyone here using the phonom service from cavalier with asterisk |
| 04:20.27 | orlock | outtolunc: http://www.selintra.com/docs/cgi-bin/view/Main/SysKwik |
| 04:22.38 | outtolunc | and you aren't asking your 'sail' questions to them or on thier help forums because? |
| 04:24.35 | outtolunc | i come here when i'm being lazy and IF i ask a question it's to those that i think 'might' know 'off hand' but if i get a ration of shit even i say kiss my ass and move on |
| 04:27.19 | gandhijee | i come when i have random questions |
| 04:27.24 | outtolunc | sadly this happens on various levels |
| 04:27.55 | outtolunc | but a 'random' question you might expect a random answer |
| 04:28.00 | outtolunc | or none |
| 04:28.06 | outtolunc | but no reply |
| 04:28.13 | gandhijee | this is quite true |
| 04:28.27 | outtolunc | or when called on the no reply |
| 04:28.36 | outtolunc | another sad response |
| 04:28.52 | outtolunc | myself, i'm totally put off by that |
| 04:29.32 | outtolunc | if nothing else, anyone who asks me SOMETHING will get some (might even be shitty) response |
| 04:30.18 | gandhijee | i wish cavalier would support SIP for their phonom service |
| 04:30.24 | outtolunc | but i do at least deem 'have you' as a precursor to a f'n question |
| 04:30.57 | gandhijee | i take it someone has pushed your button |
| 04:31.08 | outtolunc | as one butthead here obviously doesn't and i didn't before and never will ask him a F"N thing again |
| 04:31.24 | outtolunc | obviously |
| 04:31.50 | rfmonk | recomendations on a board for 2-5 phone ast box (url works) |
| 04:32.11 | rfmonk | motherboard =) |
| 04:32.12 | outtolunc | i'm not to happy with russell either, but i'm sure that had nothing to do with his leaving |
| 04:32.13 | tuxinator_linux | outtolunc: clean it up... this is a PG channel |
| 04:32.24 | exten123 | do astrisk got such call observation future? |
| 04:32.34 | outtolunc | excuse me? |
| 04:32.44 | Mavvie | wonder how destructive "zap destroy channel" is for the operation of a PRI link. |
| 04:32.45 | Qwell | outtolunc: He said clean it up. |
| 04:32.54 | outtolunc | ah ok |
| 04:33.09 | SwK_ | mavvie probably not all that good cause the channel goes away |
| 04:33.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:33.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:33.23 | Mavvie | SwK_: wonder if it will unbreak my blocked channels. |
| 04:33.34 | outtolunc | then may those of you that ARE here and ARE listening can help those (everyone) with questions |
| 04:33.35 | Mavvie | wait. |
| 04:33.37 | SwK_ | well the channel goes away period |
| 04:33.47 | SwK_ | as in you have to restart asterisk to get it back |
| 04:33.51 | Mavvie | SwK_: it *removes* the channel from the life configuration. |
| 04:33.53 | Mavvie | ouch. |
| 04:33.55 | Mavvie | let's not try that one. |
| 04:34.01 | outtolunc | so i can leave because 'i'm offending you sooooo' |
| 04:34.06 | Qwell | outtolunc: ok |
| 04:34.15 | SwK_ | mavvieL lotta up channels? |
| 04:34.23 | outtolunc | well |
| 04:34.36 | outtolunc | help him and i'll leave |
| 04:34.49 | Qwell | ~door |
| 04:34.51 | jbot | somebody said door was don't let it hit you on your way out |
| 04:34.58 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 04:35.03 | outtolunc | that didn't help him, NOR you |
| 04:35.20 | Mavvie | SwK_: AAPT said I had blocked channels, so I'm trying to find a way to unblock them :-P |
| 04:35.23 | outtolunc | that just put the same old BS sign on your ass |
| 04:35.31 | rfmonk | how about an asterisk channel for newbies? |
| 04:35.33 | Mavvie | SwK_: I know what blocked channels are, but have no idea what causes them. |
| 04:35.38 | outtolunc | how about it |
| 04:35.52 | SwK_ | mavvie: sure fireway, restart asterisk heh |
| 04:36.13 | Mavvie | SwK_: that overcomes the symptons, but not the cause |
| 04:36.58 | outtolunc | note: i only get this way when dickheads here decide they want to press me |
| 04:37.14 | outtolunc | up and until that point i'm helping |
| 04:37.47 | outtolunc | ah i called qwell a dickhead.. damn |
| 04:38.45 | outtolunc | shall we scroll back to the last time he was helping someone... |
| 04:39.09 | rfmonk | *goes back to the woefully inadiquate book* |
| 04:39.17 | giggles | anybody have any experience with voicetronix boards? |
| 04:40.06 | outtolunc | rfmonk i remember you from last time and sadly it was the same crap... are you ever gonna get out of qwell's shadow? |
| 04:40.28 | rfmonk | heh |
| 04:40.48 | rfmonk | thats great, my first irc flame! |
| 04:41.14 | outtolunc | if you thought that was a flame you are new |
| 04:41.29 | outtolunc | anyways |
| 04:41.55 | outtolunc | i tried to tell you last time that i 'used to be here daily' years ago |
| 04:42.09 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 04:42.10 | outtolunc | but none of you 'new guys' gave a crap |
| 04:42.42 | outtolunc | i came back to see if the 'buttheads' had finally left, i see not |
| 04:43.34 | outtolunc | so.. when i'm here, i'll do what i do which is 'first help' but if i come across you buttheads.. it will be 'crap mode' |
| 04:43.38 | outtolunc | it's that simple |
| 04:43.57 | rfmonk | ill try back tomarrow, and jus listen, thx... X) |
| 04:44.01 | *** part/#asterisk rfmonk (n=rfmonk@205.241.253.227) |
| 04:44.15 | JackEStorm | I have finaly given up on SixTel, does any one know who offers IAX/SIP unlimited DID's for US numbers? |
| 04:44.31 | outtolunc | unlimited DID? |
| 04:44.44 | outtolunc | the only one close that 'i know of' is nufone |
| 04:44.47 | outtolunc | for 866 |
| 04:44.48 | outtolunc | 's |
| 04:45.06 | FuriousGeorge | hey all |
| 04:45.08 | outtolunc | but that is 'aquire mode, not calling mode' |
| 04:45.13 | FuriousGeorge | i got kinda a linux administration question |
| 04:45.20 | russellb | i think voicepulse does ... |
| 04:45.24 | FuriousGeorge | i had a server crash and i looked at the logs, and i have no idea why |
| 04:45.27 | outtolunc | meaning you can get a bunch of did's, but the calls are NOT free |
| 04:45.28 | FuriousGeorge | is that par for the course? |
| 04:45.55 | JackEStorm | outtolunc: as in flat monthly rate for DID's, non metered. |
| 04:46.16 | russellb | JackEStorm: check out voicepulse |
| 04:46.20 | outtolunc | as in, i don't get charged ANYTHING but the call rate for a 866 did |
| 04:46.20 | blitzrage | outtolunc: http://www.zipinfo.com/cgi-local/zipsrch.exe |
| 04:46.25 | blitzrage | outtolunc: fyi |
| 04:46.58 | outtolunc | seems familiar .. checking |
| 04:47.50 | outtolunc | hmm i'll check it against accumail |
| 04:48.10 | [hC] | Anyone got a Cisco 7970 speaking SIP to asterisk? |
| 04:48.14 | *** join/#asterisk fogall (n=fogall@customer-200-79-84-78.uninet-ide.com.mx) |
| 04:48.52 | fogall | hi can anyone help me? |
| 04:48.55 | outtolunc | i have a local fips db <G> |
| 04:49.02 | Strom_C | I didn't know there was SIP firmware for the 7970 |
| 04:49.07 | JackEStorm | russellb: they don't sell DID's, only plan packages. |
| 04:49.12 | fogall | i'm newbie in asterisk |
| 04:49.14 | Strom_C | fogall, just ask your question |
| 04:49.23 | outtolunc | fogall, it is truely possible that someone might be able to help you <G> |
| 04:49.44 | russellb | JackEStorm: ah. my bad |
| 04:49.49 | Strom_C | outtolunc, <g> is so 80s-era compuserv it's not even funny :) |
| 04:49.55 | fogall | how i configure asterisk to support SIP on a LAN? |
| 04:50.07 | Strom_C | fogall, it already supports SIP |
| 04:50.13 | outtolunc | gee ya think |
| 04:50.15 | Strom_C | look in sip.conf |
| 04:50.36 | blitzrage | I love <G>! |
| 04:50.38 | outtolunc | strangely, i just can't help myself |
| 04:50.42 | JackEStorm | russellb: thats the problem, voip is great but for what I want bell is still the best after SixTel lost the contract :( |
| 04:50.42 | outtolunc | <G> |
| 04:50.53 | Strom_C | I remember <g> seemed dated way back in 1993 |
| 04:51.14 | blitzrage | I'm totally busting out <G> again |
| 04:51.15 | giggles | JackEStorm: what is your problem with sixtel? |
| 04:51.19 | outtolunc | then you might have issue with me being dated LONG before that |
| 04:51.20 | JackEStorm | I've been with out phone service since late Aug |
| 04:51.59 | JackEStorm | giggles: I need a DID provider, since SixTel is not offering "unlimited" DID's anymore |
| 04:52.14 | outtolunc | some of my 'issue' is that after 25+ years of this crap .. someone might want to listen to me <G> |
| 04:52.25 | fogall | can someone recomend me a good manual? |
| 04:52.28 | outtolunc | but oh f'n well |
| 04:52.33 | russellb | ~docs |
| 04:52.35 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:52.53 | *** join/#asterisk mimecine (n=mimecine@user-0cdfon5.cable.mindspring.com) |
| 04:54.49 | [hC] | Strom_C: there is now. |
| 04:55.09 | Strom_C | oh cool |
| 04:55.19 | mimecine | Sorry for jumping in (I wish I had time to wait around): any one not recommending Athlon 64 for asterisk? |
| 04:55.31 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 04:56.07 | russellb | mimecine: lots of people run asterisk on 64 bit machines |
| 04:57.08 | [hC] | looks like its a bit retarded in that it does not allow you to specify a SIP auth password, it forces you to do it by using host= in asterisk's sip.conf |
| 04:57.13 | mimecine | russelb: with zapata hardware? meant to google for more info, but I'm in the same hurry as everyone else...:) |
| 04:57.15 | [hC] | I cant find the sip auth password line in the xml config anyways. |
| 04:57.24 | russellb | mimecine: yup |
| 04:57.24 | fogall | do i need aditional hardware to use asterisk over a LAN only? i dont wanto to connect to traditional telephony |
| 04:57.30 | Strom_C | fogall, no |
| 04:57.44 | Strom_C | you only need special hardware if you want to connect to analog phones or T1 lines |
| 04:58.05 | fogall | ok |
| 04:58.39 | JackEStorm | Strom_C: s/analog phones or T1 lines/non IP transport devices/ |
| 04:59.03 | outtolunc | hC, 'being retarded' would in that case be a 'system/network' thing, quite honestly i don't see where you get off calling either retarded |
| 04:59.28 | mimecine | russelb: have you heard of anyone doing aah with amd64? Just found a machine at a decent price and was thinking of replacing the crazy old machine with a better one... |
| 05:00.11 | [hC] | outtolunc: what the? Im saying its stupid that the phone doesnt allow you to specify an auth password for invidiual sip identities. |
| 05:00.19 | [hC] | Which leads me to believe you can register only one line on it, via SIP. |
| 05:00.24 | russellb | i'm sure people do. I don't know much about aah. You should probably try #amportal for questions specific to that |
| 05:00.27 | [hC] | (it has 8 of them) |
| 05:01.07 | outtolunc | exactly, so what 'you deem' as 'your world' is ok, but what another asterisk admin (and he just thinks a bit bigger), is 'retarded' |
| 05:01.08 | Juggie | hc, you have to use a dif name/pass for each line |
| 05:01.10 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 05:01.13 | Juggie | * doesnt do multiple registrations |
| 05:01.17 | Juggie | (yet) |
| 05:01.22 | mimecine | russelb: thanks, good idea. was actually looking for #asteriskathome, but couldn't find it... thanks a bunch! |
| 05:01.30 | [hC] | What in the fuck are you talking about |
| 05:01.32 | [hC] | Im not talking about asterisk |
| 05:01.39 | [hC] | I'm talking about the sip firmware on this stupid piece of shit cisco phone! |
| 05:01.40 | russellb | mimecine: you're welcome |
| 05:01.47 | [hC] | Get your head out of your ass and pay attention |
| 05:01.49 | mimecine | I'll linger around a while.. |
| 05:01.53 | outtolunc | haha |
| 05:02.03 | outtolunc | speaking of heads and asses |
| 05:02.04 | Juggie | which model phone jerk |
| 05:02.08 | JackEStorm | [hC]: then your in the wrong chan if you are not talking about asterisk or related directly. |
| 05:02.14 | [hC] | not you jug. |
| 05:02.15 | [hC] | :) |
| 05:02.26 | [hC] | I got the new SIP load for the Cisco 7970 |
| 05:02.33 | [hC] | Which was previously SCCP only |
| 05:02.36 | outtolunc | jC you must mean me, since i have no clue |
| 05:02.41 | [hC] | SCCP worked fine, but i figured id give it a shot. |
| 05:02.42 | russellb | talking about cisco IP phones is way more on topic than many other conversations that go on in here |
| 05:02.43 | Juggie | with the 7960 i was able to do multiple lines. |
| 05:03.04 | [hC] | yeah, the 70 has a new sip image that uses the xml configs exactly like the SCCP load did. |
| 05:03.06 | Qwell | I think the 7970 changed format, no? |
| 05:03.11 | [hC] | Yeah it did |
| 05:03.12 | Qwell | yeah...cheese |
| 05:03.16 | Qwell | cheesy rather |
| 05:03.17 | Juggie | hc i told you cisco phones sucked |
| 05:03.20 | Juggie | and i stand by that :) |
| 05:03.24 | [hC] | I agree that they suck |
| 05:03.26 | outtolunc | which is why someone like me to does multiple users, and radius and all that' total bs' crap is just someone that is totally out there <G> |
| 05:03.26 | [hC] | I dont like them at all |
| 05:03.30 | [hC] | but they make me lots of money. |
| 05:03.30 | Juggie | mitel phones imo! |
| 05:03.33 | Juggie | suppport canadian. |
| 05:03.37 | Qwell | [hC]: Did you see the sample sip conf somebody posted on the chan-sccp-users list? |
| 05:03.42 | [hC] | Yeah, I used that. |
| 05:03.50 | JackEStorm | I like Cisco and Polycom |
| 05:04.03 | [hC] | outtolunc: Im still not sure what you're going on about, but knock yourself out. |
| 05:04.06 | *** join/#asterisk mattodude (n=matt@gateway.digium.com) |
| 05:04.15 | outtolunc | there you have folks, the answer to all yours issues is 'cheese' |
| 05:04.17 | Qwell | mattofake! |
| 05:04.25 | Strom_C | the only mitel system I ever used was horrid :) |
| 05:04.33 | russellb | mattjdude: imposter! |
| 05:04.37 | outtolunc | hC please come try |
| 05:04.57 | giggles | oi |
| 05:05.00 | [hC] | hahaha. |
| 05:05.41 | outtolunc | PLEASE |
| 05:06.00 | [hC] | outtolunc: DUDE! |
| 05:06.06 | [hC] | outtolunc: i dont even know what you are talking ABOUT |
| 05:06.13 | [hC] | now please stop! |
| 05:06.34 | [hC] | Haha |
| 05:06.38 | [hC] | I just got an ad emailed to me from china |
| 05:06.44 | [hC] | for a new IP phone from a company called POSDATA |
| 05:06.56 | [hC] | Im not sure if these guys are familiar with american acronyms or not.... |
| 05:07.05 | [hC] | I wouldnt have put POS at the start of my product name.... ;) |
| 05:07.24 | outtolunc | you tell someone to stfu you better be able to back it up dickhead |
| 05:07.35 | [hC] | Are you from florida? |
| 05:07.40 | [hC] | Im just curious.. |
| 05:07.42 | outtolunc | no minnesota |
| 05:07.51 | [hC] | Ah. Right. |
| 05:08.12 | [hC] | Well as fun as its been trying to figure out what you're talking about im gonna go home now. See ya kids later. |
| 05:08.17 | outtolunc | where are you from? |
| 05:09.02 | outtolunc | i figured as much |
| 05:09.03 | [hC] | Canada |
| 05:09.21 | outtolunc | even before i seen the response |
| 05:09.51 | [hC] | What, that i was from canada? |
| 05:09.52 | Qwell | s/seen/saw/ |
| 05:10.05 | [hC] | bbl |
| 05:10.08 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 05:11.20 | outtolunc | all i ever say to these weenies is 'say it to my face' |
| 05:11.34 | outtolunc | PLEASE |
| 05:12.35 | outtolunc | i must be old if i think 'what falls out your HOLE is your resposibilty' |
| 05:13.01 | russellb | moving on ..... |
| 05:13.19 | outtolunc | please, anyone got a question? |
| 05:13.22 | JackEStorm | outtolunc: no your just fucking the wrong woman the wrong way. |
| 05:13.23 | FuriousGeorge | for some reason my music on hold isnt working |
| 05:13.33 | FuriousGeorge | its probably the new sytax |
| 05:13.43 | blitzrage | how do I encapsulate IAX2 in SIP? |
| 05:13.55 | russellb | FuriousGeorge: the old syntax is still supported, too |
| 05:14.09 | russellb | FuriousGeorge: if you wanted to see if your old config works |
| 05:14.12 | russellb | blitzrage: wtf? :) |
| 05:14.14 | outtolunc | so russellb, how am i supposed to respond to that, myself i thing invite the weenie over for an ass kicking |
| 05:14.31 | blitzrage | russellb: :D |
| 05:14.38 | Qwell | outtolunc: You don't respond to it. |
| 05:14.39 | outtolunc | i suppose you would what, give him $100 |
| 05:14.39 | russellb | outtolunc: i don't know what you're talking about |
| 05:14.51 | outtolunc | if you can't read |
| 05:14.57 | outtolunc | <PROTECTED> |
| 05:15.06 | FuriousGeorge | russellb: if thats the case than i guess its not working b/c thats the config ive always used |
| 05:15.10 | FuriousGeorge | i cahnged it to this http://pastebin.ca/46444 |
| 05:15.13 | FuriousGeorge | with no luck |
| 05:16.02 | russellb | FuriousGeorge: what was your old config |
| 05:16.05 | tzafrir | so, any recommendation of a nice softphone with regards to text chat? |
| 05:16.29 | Qwell | does idefisk do text? |
| 05:16.56 | tzafrir | I'm basically trying to lure our people away from skype |
| 05:16.57 | russellb | FuriousGeorge: I don't think "loud" is a valid mode |
| 05:17.10 | FuriousGeorge | russellb: it just worked before, i had it on quietmp3 earlier |
| 05:17.20 | tzafrir | because I figure skype will never talk to asterisk through a decent interface |
| 05:17.34 | FuriousGeorge | russellb: tbh, i never "set it up" it always just worked |
| 05:17.34 | outtolunc | fg, how old is your base? |
| 05:17.46 | theorem_ | tzafrir - I thought it was doing that fine already. |
| 05:17.49 | FuriousGeorge | so well, in fact, i thought it was my client and asked xten how to turn it off in eyebeam :) |
| 05:17.58 | FuriousGeorge | since then ive just dropped better files in there |
| 05:18.29 | russellb | FuriousGeorge: just try the [default] that is in the sample config |
| 05:18.37 | theorem_ | what is the name of the asterisk process ? |
| 05:18.43 | Qwell | theorem_: asterisk |
| 05:18.47 | theorem_ | hmm, ok |
| 05:18.56 | russellb | believe it or not, folks! |
| 05:19.05 | theorem_ | and .. |
| 05:19.12 | theorem_ | where are config files defaultly held ? |
| 05:19.19 | theorem_ | : /usr/local/etc/asterisk ? |
| 05:19.20 | Qwell | /etc/asterisk/ |
| 05:19.21 | outtolunc | russelb: in the "old days" ;loud => mp3:/var/lib/asterisk/mohmp3 |
| 05:19.23 | Qwell | usually |
| 05:19.27 | Qwell | depends on OS, I guess |
| 05:19.27 | outtolunc | was valid |
| 05:19.30 | theorem_ | Linux system ? debian ? |
| 05:19.40 | Qwell | /etc/asterisk/ is sane |
| 05:19.49 | alephcom | on freebsd they're in /usr/local/etc/asterisk I believe. |
| 05:19.53 | theorem_ | ok |
| 05:20.02 | russellb | outtolunc: ah, ok. well that "loud" is the class name, not the mode name. that's using the mode, mp3 |
| 05:20.04 | theorem_ | to get started I just edit .. which file ? |
| 05:20.10 | outtolunc | oh gee wally lets get into a 'path' arguement |
| 05:20.21 | theorem_ | say to get a simple SIP soft phone going to ... a skype interface .. |
| 05:20.22 | Qwell | theorem_: You're asking some very basic questions there... |
| 05:20.27 | theorem_ | yes |
| 05:20.31 | Qwell | ~wikis |
| 05:20.33 | jbot | hmm... wikis is http://www.voip-info.org |
| 05:20.33 | Qwell | ~docs |
| 05:20.34 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 05:20.35 | theorem_ | still haven;t had time to play |
| 05:20.41 | outtolunc | i was just trying to give feedback as to "WHY" he was talking about it |
| 05:20.50 | outtolunc | but who gives a shit |
| 05:20.54 | outtolunc | it's all bs |
| 05:20.56 | tzafrir | idefisk's manual does not mention anything about text messages. I didn't bother trying it as I don't like non-free programs |
| 05:21.06 | Qwell | tzafrir: It's free :p |
| 05:21.36 | theorem_ | ok Qwell - I'll bite and read up .. is there a getting started right there ? |
| 05:21.37 | asterboy | Does anyone experience strang Caller ID behavior? I'm getting what looks like random CallIDs of live accounts. |
| 05:21.38 | tzafrir | Qwell, you mean the gratis beta. |
| 05:21.39 | theorem_ | " " |
| 05:21.47 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 05:21.48 | Qwell | theorem_: yes |
| 05:21.49 | tzafrir | Will it run on my amd64? |
| 05:21.54 | Qwell | tzafrir: good question |
| 05:23.01 | asterboy | Where it gets the CallIDs is unknown, but crazy that it can pull up an actual listing seeming from mid air. |
| 05:23.07 | tzafrir | Qwell, I don't like non-free programs because I know they bring problems with them. I don't like relying on them |
| 05:23.11 | theorem_ | oh there's a good question |
| 05:23.20 | theorem_ | where does asterisk pull caller ID information from ?? |
| 05:23.25 | FuriousGeorge | russellb: i got the default configuration going. i notice there is a ton of hanging mpg123 processes |
| 05:23.31 | tzafrir | maybe it's just me. But personally I tend to value my opinion. |
| 05:24.28 | theorem_ | outtolunc - oh, yso you pre-program numbers to respond to the callerID function , it's not like you tap into the POTS callerID service for free ? |
| 05:24.54 | blitzrage | theorem_: RPID |
| 05:25.06 | FuriousGeorge | ...but i always compile mpg123 every time i upgraded * so im confused as to why this started happening |
| 05:25.16 | outtolunc | umm theorem_ i'm not the one who started this 'lets pick on the typos' crap |
| 05:25.18 | fogall | thanks for the docs ill try to configure it if a have more questions ill be back |
| 05:25.21 | Qwell | theorem_: callerid is pushed to phones, not pulled by them |
| 05:26.00 | outtolunc | and not all phones are phones |
| 05:26.10 | outtolunc | some are channel banks |
| 05:26.11 | theorem_ | Qwell - that makes a bit more sense .. but how does asterisk get the callerID information ? |
| 05:26.20 | outtolunc | but noone cares |
| 05:26.23 | Qwell | theorem_: depends. |
| 05:26.26 | theorem_ | .. to send to the "phone" |
| 05:26.35 | theorem_ | oh, so it's configurable .. |
| 05:26.37 | russellb | FuriousGeorge: might have something to do with it not getting killed correctly when the config wasn't valid |
| 05:26.38 | Qwell | With POTS lines, it can and will take it, if it's there |
| 05:26.45 | Qwell | with SIP, as russellb said, RPID is an option |
| 05:26.52 | theorem_ | gotcha, ok |
| 05:26.55 | Qwell | erm, I guess it was blitzrage who said RPID |
| 05:27.04 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:27.11 | FuriousGeorge | russellb: ive done this before and it never came back, but ill try. interestingly, all of my other * servers are playing moh correctly |
| 05:27.14 | russellb | heh, i was wondering ... |
| 05:27.18 | Qwell | russellb: heh |
| 05:27.27 | Qwell | russellb: "Wow, I'm smart" |
| 05:27.40 | theorem_ | so, lets say I buy something like VOnage service, hook it into asterisk, vonage supplies the callerID for hte incoming calls , asterisk passes it along ... same with POTS lines .. depends if it's there |
| 05:27.46 | Qwell | pfft |
| 05:27.54 | Qwell | theorem_: Vonage doesn't really work with * |
| 05:28.08 | Qwell | not without some hack, or extra fees |
| 05:28.30 | theorem_ | oh I see .. I've read something on locked ... somethings .. not being able to share etc. |
| 05:28.30 | FuriousGeorge | hey that worked |
| 05:28.33 | FuriousGeorge | russellb: thanks |
| 05:28.41 | Qwell | theorem_: there is a bit more than that, but yes |
| 05:28.41 | russellb | no problem |
| 05:28.44 | outtolunc | oh comeon he already thinks i'm a crackpot |
| 05:28.45 | Abydos313 | i read you need the next plan up to use vonage with * on their site |
| 05:28.46 | FuriousGeorge | that probably explains the crash today too |
| 05:29.01 | FuriousGeorge | russellb: but for the record i was using the same config as always when it stopped working :) |
| 05:29.02 | Qwell | outtolunc: You are a crackpot. |
| 05:29.13 | outtolunc | yep thats me... |
| 05:29.19 | russellb | FuriousGeorge: heh, well, at least it's working now |
| 05:29.30 | Qwell | Abydos313: Are they officially allowing * now? Last I heard, it was a violation of their TOS, even with the softphone account. |
| 05:30.07 | outtolunc | if i could take back every inspiration that i ever caused, i would, just to mess with you guys |
| 05:30.16 | Qwell | inspiration? |
| 05:30.19 | Abydos313 | i could swear i read a complete readme there that said with vonage-plus or something that i could be used with vonage service.. |
| 05:30.36 | outtolunc | i wasn't talking just about asterisk but we could |
| 05:31.06 | outtolunc | we have been through this time and time again, yet you are like... who are you? |
| 05:31.17 | outtolunc | so yeah |
| 05:31.18 | Qwell | outtolunc: A developer and bug marshal. |
| 05:31.21 | Qwell | try again |
| 05:31.36 | outtolunc | when it comes to guys like you... i'm like F you |
| 05:31.54 | outtolunc | i WAS a dev, and never a bug marshall |
| 05:32.04 | outtolunc | your point is |
| 05:32.13 | alephcom | Please, please... |
| 05:32.51 | outtolunc | i point is i've been nothing but honest in anything i do, yet i get shit from even dicks like you |
| 05:32.59 | russellb | outtolunc: chill out. |
| 05:33.03 | outtolunc | so yes, i have issue |
| 05:33.24 | outtolunc | and i'm HONEST enough to tell you TO YOUR face that i am |
| 05:33.26 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 05:33.42 | outtolunc | someone please tell me i'm wrong |
| 05:33.50 | russellb | you're wrong |
| 05:33.52 | russellb | now chill out |
| 05:34.04 | outtolunc | then you also are on the other side |
| 05:34.11 | outtolunc | which i already knew |
| 05:34.20 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 05:34.36 | outtolunc | so what you gonna do now, kick me? |
| 05:34.40 | alephcom | outtolunc: which side am I on? I was just a bystander but this is getting tiring fast. |
| 05:35.23 | outtolunc | alephcom if you thought i was talking to you, you have other issues, that we probably shouldn't bring up |
| 05:35.44 | alephcom | you're talking in a public forum. |
| 05:35.44 | russellb | outtolunc: why do you get like this all time? |
| 05:36.10 | outtolunc | i get like this because i 'try and help people' then someon here starts talkin SMACK |
| 05:36.18 | outtolunc | then i get peeved |
| 05:36.27 | outtolunc | it's THAT simple |
| 05:36.33 | outtolunc | read the f'n logs |
| 05:36.40 | outtolunc | PLEASE |
| 05:36.42 | russellb | well you need to learn how to just brush things off ... |
| 05:36.50 | russellb | you get annoyed way too easily |
| 05:36.51 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 05:37.24 | russellb | and your ranting about being angry is making the channel unpleasant for everyone else |
| 05:37.39 | outtolunc | so when i'm helping people and all is well, then ONE OF YOU, start talking smack... and I GET ANNOYED, it's MY fault |
| 05:37.47 | outtolunc | haha |
| 05:37.49 | outtolunc | <PROTECTED> |
| 05:38.05 | outtolunc | if you aren't gonna help STAY THE F OUT OF IT |
| 05:38.11 | outtolunc | it's that simple |
| 05:38.32 | outtolunc | then noone 'like me' will have to take offence |
| 05:38.43 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 05:39.02 | russellb | take a deep breath or something |
| 05:39.07 | outtolunc | why? |
| 05:39.13 | outtolunc | i didn't start this |
| 05:39.19 | theorem_ | hush hus |
| 05:39.32 | russellb | outtolunc: who did? |
| 05:39.43 | Qwell | $20 says me |
| 05:39.51 | outtolunc | where in there did you get that ANY of you telling me to 'hush' or 'be quiet' was even in the realm of possiblity |
| 05:40.06 | outtolunc | hello? |
| 05:40.19 | alephcom | :-) lol |
| 05:40.34 | outtolunc | if i said it to ANY of you, would you? not!@ |
| 05:40.37 | Qwell | russellb: He was told to "clean it up... this is a PG channel" |
| 05:40.42 | Corydon76-home | Considering that Russell can quiesce you, it's well within the realm of possibility |
| 05:41.17 | outtolunc | he can kick/ban me, but would i do nothing in response... |
| 05:41.19 | outtolunc | haha |
| 05:41.23 | Abydos313 | Qwell why would vonage care if you hooked up thier service to an asterisk box? how would that effect them? |
| 05:41.26 | outtolunc | PLEASE |
| 05:41.38 | russellb | outtolunc: is that a threat? |
| 05:41.45 | Corydon76-home | Actually, the quiesce command is not either a kick or a ban |
| 05:41.46 | outtolunc | fact |
| 05:41.51 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 05:41.55 | Qwell | Abydos313: Because they would have to support it |
| 05:42.01 | outtolunc | all i've ever done was the truth |
| 05:42.05 | outtolunc | ever |
| 05:42.11 | russellb | your attitude is not welcome here. |
| 05:42.27 | outtolunc | and so it begins |
| 05:42.44 | Abydos313 | oh ok , so they just don't people calling up trying to configure to use with thier service. do they use anything proprietary that you know of? |
| 05:42.45 | outtolunc | .. |
| 05:42.48 | Corydon76-home | outtolunc: you're being told to chill by someone who has the power and the standing in the community to tell you. Please stop. |
| 05:42.52 | outtolunc | .. |
| 05:43.10 | *** mode/#asterisk [+b %outtolunc!*@*] by russellb |
| 05:43.21 | Qwell | That's the one |
| 05:43.27 | alephcom | Thank you sir |
| 05:43.46 | Qwell | Abydos313: Just the passwords |
| 05:44.17 | *** join/#asterisk jcollie (n=jcollie@dsl-ppp239.isunet.net) |
| 05:44.25 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:45.07 | Abydos313 | i don't want to use vonage for asterisk i just wanted to know .. you know how it goes |
| 05:46.29 | shido6 | you can use asterisk with vonage |
| 05:46.41 | Qwell | "can" :) |
| 05:46.53 | shido6 | but why? |
| 05:46.55 | Qwell | shido6: we just went over the drawbacks of doing so |
| 05:47.02 | Qwell | You came in a few minutes late |
| 05:47.03 | shido6 | okie dokie |
| 05:47.44 | *** part/#asterisk jcollie (n=jcollie@dsl-ppp239.isunet.net) |
| 05:47.59 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 05:55.48 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 05:56.23 | alephcom | Take care everyone. |
| 05:56.25 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 05:56.35 | FuriousGeorge | ive never witnessed anyone getting kicked from here before |
| 05:57.07 | FuriousGeorge | heck, ive never been kicked and ive been in here drunk plenty of times :) |
| 05:57.19 | Qwell | he wasn't kicked |
| 05:58.09 | asterboy | Hey guys, is there a way to filter off specific messages, but keeping the same verbose level? I don't want to see these messages, they fill up the page: |
| 05:58.13 | asterboy | "- Registered SIP 'Home2' at 192.168.1.19 port 5060 expires 30 |
| 05:58.14 | asterboy | <PROTECTED> |
| 05:58.16 | Strom_C | ah, the fun of building a new asterisk box from scratch |
| 05:59.05 | russellb | asterboy: the only way would be to modify the code |
| 05:59.29 | asterboy | thats what I was fearing. |
| 06:00.09 | asterboy | or pump up the expiry, but then the phones don't ring all extensions on incoming because the session hasn't ended. |
| 06:00.48 | Strom_C | asterboy, really, I find that in practice you just end up ignoring the printout that you're not looking for anyway |
| 06:01.40 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 06:01.42 | bmg505 | morning |
| 06:01.59 | asterboy | Its just a pain to have to glean out and debug |
| 06:02.00 | bmg505 | anybody have got chan_modem to load under 1.2.5? |
| 06:02.10 | Qwell | isn't chan_modem gone? |
| 06:02.23 | bmg505 | its still there but refuses to load |
| 06:02.36 | bmg505 | intel 536ep modem :( |
| 06:03.02 | asterboy | turned down the debug info for now...seems to be good enough and then I can pump it up again later. |
| 06:05.23 | bmg505 | Mar 21 00:03:18 WARNING[12862] loader.c: /usr/lib/asterisk/modules/chan_modem_bestdata.so: undefined symbol: ast_unregister_modem_driver |
| 06:05.24 | bmg505 | Mar 21 00:03:18 ERROR[12862] chan_modem.c: Failed to load driver chan_modem_bestdata.so |
| 06:05.37 | Qwell | bmg505: yeah...it's dead, I'm pretty sure. When you upgrade, the modules dir isn't cleaned out |
| 06:05.58 | bmg505 | its vigin downlaod and its still there in source form |
| 06:06.03 | Qwell | oh |
| 06:06.14 | Qwell | You've never installed * on that box before? |
| 06:06.40 | *** join/#asterisk fogall (n=fogall@customer-200-79-84-78.uninet-ide.com.mx) |
| 06:06.47 | bmg505 | nope first * install as well |
| 06:07.01 | bmg505 | I use it for sip only and 1 incoming line |
| 06:07.17 | fogall | anoyone speak spanish? |
| 06:07.39 | Strom_C | not anywhere near well enough to talk tech :) |
| 06:07.55 | Qwell | fogall: Speak Spanglish |
| 06:08.14 | Strom_C | I can get by for routine conversational stuff though |
| 06:08.51 | fogall | ill do my best speaking english |
| 06:09.36 | bmg505 | fogall: English is not my first lang as well and most peeps on freenode realize that not everyone is perfect in eng :) |
| 06:12.29 | bmg505 | Is there any solution "cheapish" that I can use for fxo line in? |
| 06:12.52 | russellb | the generic x100p |
| 06:12.57 | Strom_C | there's the cheapo clone x100p cards |
| 06:13.06 | Strom_C | but those really blow donkeys for quarters |
| 06:13.08 | fogall | i've been reading some of the docs you provide me, but when i try to configure my sip.conf i got stuck in "register => me@mysipproxy.com/1000" line |
| 06:13.24 | fogall | do i need to configure a sip proxy? |
| 06:13.30 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 06:14.13 | bmg505 | Strom_C as far as I tried is SA I cannot find modems only I can get has intel 536EP chipset on |
| 06:14.21 | bmg505 | s/is/in/ |
| 06:14.52 | bmg505 | regex in channel thats cool :) |
| 06:15.41 | fogall | im trying to configure asterisk to work over a LAN with no exit to Internet and private IP's |
| 06:16.05 | tzafrir | bmg505, try searching for x100p on ebay |
| 06:16.20 | Strom_C | bmg505, you can buy the clone x100p from the states and it will likely work on a south african telephone line |
| 06:16.37 | Strom_C | I believe that the last time I was there the telephone sets were the same |
| 06:16.37 | *** mode/#asterisk [-b %outtolunc!*@*] by russellb |
| 06:16.44 | bmg505 | ok will try that |
| 06:16.54 | bmg505 | we are 100% compatible wit huk standard |
| 06:16.55 | tzafrir | But then again, it's useful as a practice card, but don't plan for it to work in production |
| 06:17.04 | Strom_C | christ, now I want ostrich biltong. DAMN YOU! |
| 06:17.12 | bmg505 | lol |
| 06:17.53 | Strom_C | there's a guy in west los angeles who makes excellent beef biltong and boerwors |
| 06:17.54 | bmg505 | for production what do u suggest? |
| 06:18.08 | tzafrir | bmg505, is an ISDN line an option? cheap supported ISDN cards are available |
| 06:18.27 | bmg505 | ha I have an ISDN will try later today |
| 06:18.34 | Strom_C | tzafrir, you can plug BRIs into asterisk now? |
| 06:18.40 | fogall | i've been reading some of the docs you provide me, but when i try to configure my sip.conf i got stuck in "register => me@mysipproxy.com/1000" line |
| 06:18.42 | fogall | do i need to configure a sip proxy? |
| 06:18.53 | Strom_C | fogall, do you have a sip account with someone? |
| 06:19.00 | fogall | no |
| 06:19.04 | tzafrir | Strom_C, sure. TIMTOWTDI |
| 06:19.09 | Strom_C | then you dont need the register statement |
| 06:19.09 | bmg505 | fogall: i'm new to it and my sip works without proxies |
| 06:19.25 | fogall | ok |
| 06:19.38 | tzafrir | ZapBRI, chan_capi, chan_misdn, chan_visdn, ... |
| 06:19.57 | Strom_C | tzafrir, interesting - I've been contemplating getting an ISDN BRI line but I'd only want to do it if I can have asterisk do all the D-channel stuff directly |
| 06:20.55 | tzafrir | Strom_C, I'm not sure what exactly you refer to, but basically, yes |
| 06:21.13 | Strom_C | tzafrir, well you know what a d-channel is, right? |
| 06:21.39 | FuriousGeorge | is anyone working on itraserver device state awareness |
| 06:22.52 | FuriousGeorge | i suppose ith olle's patch i could just have the servers update their own states with eachother |
| 06:24.13 | tzafrir | Strom_C, I know what it is. But I'm not very familiar with the internals of ISDN |
| 06:24.54 | tzafrir | anyway, you may find that the lower-level driver messes with that |
| 06:25.07 | Strom_C | basically, I want to make sure I can just bring my ISDN line directly into the asterisk box and have the asterisk box handle all the signaling directly to the telco instead of hacing some external device between my asterisk box and the line |
| 06:25.16 | *** join/#asterisk peted20 (n=chatzill@71.39.93.58) |
| 06:26.47 | bmg505 | tzafrir: does * supportmultiple numbers on isdn? |
| 06:27.01 | Strom_C | Just to make sure - if I have a subdirectory in my /etc/asterisk directory called "samples" and I put the original sample config files in it, asterisk will ignore them when I do a reload, right? |
| 06:27.02 | tzafrir | Strom_C, yes |
| 06:27.12 | tzafrir | this should work. |
| 06:27.22 | Strom_C | tzafrir, which card would I need? |
| 06:27.47 | tzafrir | bmg505, sounds like it should work, assuming you mean something like what Strom_C was talking about. |
| 06:28.35 | tzafrir | chan_capi and chan_misdn seem to require capi2-enabled card (such as fritz-avm, I believe) |
| 06:28.59 | tzafrir | bristuff's zapbri can work with HFC-s PCI cards |
| 06:29.10 | bmg505 | ok I'll try it later today, I have an ISDN BRI on site |
| 06:29.24 | bmg505 | bbl |
| 06:29.30 | tzafrir | I'm totally unfamiliar with chan_visdn |
| 06:29.37 | Strom_C | tzafrir, are you doing this with ISDN in north america? |
| 06:30.07 | tzafrir | Strom_C, I'm not in north america, and I'm not that familiar with ISDN... |
| 06:30.20 | Strom_C | crap. |
| 06:30.50 | bmg505 | Strom_C: as far as I know isdn is std its jsut some d chan signalling that is different |
| 06:30.58 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 06:31.25 | tzafrir | Strom_C, right, the "samples" will be ignored. pbx_config looks for files specifically in /etc/asterisk (unless the default location is overiden, yada, yada, yada) |
| 06:31.30 | bmg505 | in i4l has got lots on d chan signalling in it |
| 06:31.42 | Strom_C | well, um, that's the important point - if north american ISDN is different from european ISDN and the cards only speak european ISDN, I'm fucked. |
| 06:32.02 | Strom_C | I really need to get an ISDN book newer than the one I have which was published in 1990 :) |
| 06:32.03 | bmg505 | all the cards I have can speak both its setting |
| 06:32.10 | bmg505 | :) |
| 06:32.13 | bmg505 | cya later |
| 06:32.15 | bmg505 | bbl |
| 06:32.15 | tzafrir | bmg505, chan_modem is deprecated. I suppose it's not recommended |
| 06:32.27 | Strom_C | bmg505, I've successfully done ISDN PRI here in north america, but never have I touched BRI |
| 06:32.56 | tzafrir | BRI, in north america? is there such a beast? |
| 06:33.35 | *** join/#asterisk Qber (n=Qbera@c-24-6-80-84.hsd1.ca.comcast.net) |
| 06:34.09 | Strom_C | there certainly is |
| 06:34.21 | Qber | is it possible to do agent call back with persistant login |
| 06:34.30 | Qber | in asterisk queueus |
| 06:35.17 | Qber | i have agents on mobile phone and always logged in? |
| 06:36.22 | Strom_C | good god man, what are you doing with agents on mobile phones? |
| 06:36.48 | Qber | well, i roaming agents |
| 06:36.58 | Qber | i have roaming agents/sales guys |
| 06:37.07 | Qber | also supporting our customers |
| 06:37.19 | asterboy | Been going through the docs and need something to help me setup Call TRANSFER and CONFERNCE calling. Can someone please point me to the specific doc? I'm trying to do this with SIP on Polycom Phones and ZAP channels. Not sure if it can be done. |
| 06:38.23 | Qber | really need to know how can i implement extensions as queue agents |
| 06:38.58 | Qber | it must have been solved before...i am sure. |
| 06:39.43 | asterboy | Qber, does this help? http://www.voip-info.org/wiki-Asterisk+Agents |
| 06:40.25 | blitzrage | asterboy: if its not on the wiki, you'll probably have to figure it out yourself |
| 06:40.27 | Qber | let me check...thanks asterbiy |
| 06:40.41 | Qber | asterboy.. |
| 06:41.03 | *** join/#asterisk maxx4life (n=max4life@71-35-210-12.slkc.qwest.net) |
| 06:41.27 | asterboy | blitzrage, ya I'm trying...if someone has experience with this though, sure would save me diggin. |
| 06:41.29 | asterboy | Qber.? |
| 06:42.15 | asterboy | When I try to do a transfer I get this message: |
| 06:42.17 | asterboy | <PROTECTED> |
| 06:42.46 | asterboy | Same message if I try to conference. |
| 06:43.48 | *** join/#asterisk Snake-Eyes (n=blog@202.168.41.172) |
| 06:43.49 | asterboy | Don't think it knows how to create the ZAP channel. |
| 06:44.00 | asterboy | I also need to be able to Forward calls. |
| 06:44.34 | asterboy | All are similar in that they need to take an inbound connection and either addto or transferto another ZAP channel. |
| 06:46.02 | bsdfreak | last sixtel |
| 06:48.53 | Strom_C | well that's cute...I seem to have misplaced the ac adapter for my PAP2 |
| 06:50.56 | asterboy | I gotta call Rogers, the CallID on my cell is picking random listings! |
| 06:52.12 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 06:52.12 | asterboy | Strom_C, checks out ebay for a guy who burnt out his PAP2. |
| 06:52.50 | Strom_C | asterboy, by misplaced I mean that it's just gotten lost in the pile |
| 06:53.03 | Strom_C | if I had actually lost it somewhere outside my apartment, then I'd be worried |
| 06:55.52 | asterboy | it's growing legs and crawling about. |
| 06:56.07 | asterboy | I have a burnt out pap2 and thus a spare power adaptor. |
| 06:56.44 | asterboy | Actually, I have two, but the other might just as well be burnt out cause its a revision that can't be unlocked. |
| 06:56.56 | asterboy | Vuck Fonage |
| 06:58.20 | Strom_C | see, I got the free vonage one and unlocked it before it could talk to vonage :) |
| 06:58.35 | *** join/#asterisk insync (n=spam@66-188-89-49.dhcp.mdsn.wi.charter.com) |
| 06:58.56 | insync | hello all |
| 07:00.26 | insync | looking for a bit of advice on te110p with proliant dl145 |
| 07:02.12 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 07:02.26 | Strom_C | insync, if it blows up with sparks and smoke and fire, you didn't install it correctly |
| 07:03.11 | insync | close installed it and machine came to a slower than snail shit boot pace |
| 07:03.46 | insync | is most certainly an interupt problem |
| 07:04.15 | Strom_C | have you tried putting it in different slots? |
| 07:04.35 | insync | has pci riser one full one half |
| 07:04.51 | insync | only "really" fits in one |
| 07:05.20 | insync | can mod support bracket to make it fit though |
| 07:05.33 | insync | but seems to be similar in response |
| 07:09.29 | insync | many devicesare all sharing irq9 and cant seem to force them to go anywhere else |
| 07:10.29 | insync | card is modprobing fine and by the looks it is online but i fear in production i will have drops etc.. |
| 07:11.17 | FuriousGeorge | insync: i dont think there is any guarantee you can get the card on its own irq, depending on how many and what kind of devices you have |
| 07:11.30 | FuriousGeorge | and, probably most importantly, your mb |
| 07:11.47 | Qwell | disable everything onboard that you don't need |
| 07:11.52 | Qwell | usb, serial, etc |
| 07:12.08 | insync | did it |
| 07:12.12 | FuriousGeorge | parallel is pretty useless if you have usb printers |
| 07:12.14 | Qwell | and if it's still sharing an irq, change pci slots |
| 07:12.21 | insync | i have heard of this prob with the digium |
| 07:12.31 | insync | many have went to the sangoma |
| 07:12.45 | Qwell | insync: welcome to plug and play hardware |
| 07:12.45 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 07:12.53 | Qwell | pretty much all pci cards will have this problem |
| 07:13.12 | Qwell | alternatively, use apic |
| 07:13.16 | FuriousGeorge | insync: i run one box with two tdms right now and one shares an irq with eth0, and we get no issues that i can attribute to that |
| 07:14.17 | insync | i have a production machine that is working as well with a shared irq but this is the first time i saw a very drastic performance issue |
| 07:14.55 | FuriousGeorge | cat /proc/interupts see about that |
| 07:15.26 | insync | apic seems to be where i need to go, but it looks as though this mb wants to play nice |
| 07:21.09 | asterboy | How do you record a call with 1 touch dial? |
| 07:21.14 | *** join/#asterisk kos (n=kos@unaffiliated/kos) |
| 07:21.35 | *** join/#asterisk xterminus (n=cmauch@00104bc8bd59.click-network.com) |
| 07:22.15 | asterboy | setup features.conf, but the call does not record when I press the command. |
| 07:23.37 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:24.02 | kmilitzer | Morning everyone ... |
| 07:25.34 | xterminus | is it possible to make sip uri calls from a iax phone? I can call sip uri's fine from xlite, but i cannot from idefisk - all i get is "chan_iax2.c:7053 socket_process: Rejected connect attempt from 10.0.0.254, who was trying to reach '9586111@mutual.bcwireless.net'" |
| 07:27.37 | Qwell | xterminus: umm...no |
| 07:28.24 | xterminus | so i'm stuck with xlite? |
| 07:28.43 | Qwell | or one of the many other SIP softphones |
| 07:29.27 | xterminus | any idea why asterisk wont even try to bridge the iax channel to a sip outbound channel? |
| 07:29.54 | Strom_C | xterminus, because you're not trying to dial through asterisk? |
| 07:30.04 | Strom_C | you're trying to dial sip urls directly |
| 07:30.14 | Strom_C | draw circle, bang head here |
| 07:30.19 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-34-22.neo.res.rr.com) |
| 07:30.22 | *** join/#asterisk Snake-Eyes (n=blog@202.168.41.171) |
| 07:30.29 | sleepy_one | hello everyone :-D |
| 07:30.32 | xterminus | hrm, so asterisk cant proxy iax ? |
| 07:30.40 | Strom_C | asterisk can proxy iax just fine |
| 07:30.48 | Strom_C | but you need to set that up *on asterisk* |
| 07:31.18 | Strom_C | asterisk doesnt just automagically know what to do |
| 07:31.22 | xterminus | so is there a way to tell it to proxy everything |
| 07:31.34 | xterminus | it looks like its refusing now |
| 07:31.40 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 07:32.21 | sleepy_one | I have a TDM400P with 4 FXO modules and the * server keeps getting confused and MOH, voicemail, playback, background, and Zap channel audio disappears after a few minutes on FC4 running 2.6.11-1.1369_FC4smp on a P4 with HT and zaptel 1.2.4 and * 1.2.5 any ideas? suggestions? |
| 07:32.49 | xterminus | yes - i know i can setup a context for mutual.bcwireless.net (and that works), but if i have to add every sip domain on the internet to asterisk... |
| 07:33.22 | sleepy_one | it works if I use ztdummy -- but ztdummy conficts with wctdm ( WC TDM400P ) |
| 07:33.55 | xterminus | sleepy_one, you dont need ztdummy if you have a TDM card |
| 07:34.09 | tzafrir | sleepy_one, rmmod ztdummy |
| 07:34.17 | tzafrir | now try zttest |
| 07:34.27 | tzafrir | do you get anything? |
| 07:34.30 | Strom_C | xterminus, if you're hellbent on dialing sip URLs, why use asterisk? |
| 07:34.37 | Strom_C | why must asterisk proxy for you? |
| 07:34.48 | Strom_C | why not just use a sip softphone and be done with it? |
| 07:35.28 | xterminus | Strom_C, xlite crashes a lot - and the only other (multiplatform) softphone that seems semi-reliable is idefisk |
| 07:35.28 | tzafrir | sleepy_one, besides, why does ztdummy conflict with wctdm? IIRC wctdm simply overrides ztdummy as a timing source |
| 07:35.40 | sleepy_one | Yes, I know guys, thank you. What I'm saying is if use ztdummy * works if I use wctdm ( the TDM400p kernel module ) all audio stops working after a few minutes |
| 07:35.57 | Strom_C | xterminus, get a real phone |
| 07:36.16 | xterminus | Strom_C, thanks for the help |
| 07:36.43 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-157.claranet.co.uk) |
| 07:37.01 | tzafrir | sleepy_one, this is why I asked about the output of zttest, as it is the best indication of the availbility and quality of the timing source |
| 07:37.34 | sleepy_one | I have no idea why it conficts with ztdummy audio works but the Zap lines do not work - no audio from the Zap channels - without ztdummy the Zap channels ( 1 - 4 ) work fine for a few minutes then they go dead and * cannot play audio anymore or anything. No MOH, playback, background, no Zap, nothing it's unusable |
| 07:39.57 | sleepy_one | ./zttest |
| 07:39.58 | sleepy_one | Opened pseudo zap interface, measuring accuracy... |
| 07:40.03 | sleepy_one | this is all I got - it's still running |
| 07:41.31 | sleepy_one | I suspect the host computer or the TDP400p card might be broken / confused |
| 07:44.19 | *** join/#asterisk mattjdude (n=matt@24.96.136.141) |
| 07:44.39 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-34-22.neo.res.rr.com) |
| 07:45.40 | sleepy_one | hello again - after waiting a while I hit CTRL + C and got this: http://pastebin.com/613913 |
| 07:46.25 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:46.35 | asterboy | * must be able to transfer a call from ZAP to SIP and then to another SIP extensions if the pickup decides. |
| 07:47.01 | asterboy | Anyone give some insite on SIP to SIP transfers from incoming ZAP? |
| 07:47.05 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 07:48.50 | littleball | hello, i have a IVR system. when the users call in, my system prompt "Please enter your extension number followed by hash key". the system uses Background() cmd to capture the extension number. I encount one problem in this. After user key in # key, the system should end immediately the waiting and continue next step. In stead, the system always wait for 15 seconds to timeout and then continue next step. How to solve this problem? |
| 07:48.59 | asterboy | man the group is dead tonight. (today if other side of globe) |
| 07:50.24 | asterboy | with the lack of support here littleball, I don't think you'll solve it tonight. |
| 07:50.35 | asterboy | wait till the next bunch come online. |
| 07:50.36 | *** join/#asterisk U-238 (n=U-238@ppp157-212.static.internode.on.net) |
| 07:50.41 | *** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it) |
| 07:50.42 | *** part/#asterisk U-238 (n=U-238@ppp157-212.static.internode.on.net) |
| 07:51.32 | asterboy | littleball, pastebin would be a great help, but I have to go, so hope you get it fixed. |
| 07:52.04 | littleball | thanks asterboy |
| 07:52.25 | asterboy | no prob, I like to help when I can. |
| 07:53.03 | *** join/#asterisk medusaXX (n=medusaxx@p54A9C734.dip0.t-ipconnect.de) |
| 07:55.24 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 07:55.27 | sleepy_one | zttool detects a Wildcard TDM400P REV I Board 1 but zttest produces nothing - has anyone had a problem like this? |
| 07:59.05 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 07:59.11 | tzafrir | sleepy_one, you said it worked at first, right? |
| 08:00.09 | tzafrir | if you stop asterisk, rmmod wctdm, modprobe wctdm, run ztcfg, then do you get some output from zttest? |
| 08:00.27 | sleepy_one | yes |
| 08:01.01 | sleepy_one | when I rmmod all the * modules then modprobe zaptel; modprobe wctdm; ./zttest I get output |
| 08:01.12 | sleepy_one | after a few minutes it does nothing |
| 08:01.30 | tzafrir | Time to ocntact Digium support? |
| 08:02.37 | sleepy_one | indeed |
| 08:02.56 | sleepy_one | too bad they won't be open for several hours :-( |
| 08:07.13 | *** join/#asterisk Aurs (i=aurs@hallo.aurs.info) |
| 08:08.41 | sleepy_one | I restarted * left zttest running in a terminal and so far it is working |
| 08:10.21 | sleepy_one | I wonder if the card is resting in an evil PCI slot or if the chipset is b0rk3n |
| 08:14.59 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:18.22 | sleepy_one | I noticed I get TDM PCI Master abort |
| 08:18.25 | sleepy_one | in dmesg |
| 08:18.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:21.32 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 08:21.56 | *** join/#asterisk Creperum (n=ilya@166.160.skiff.relc.com) |
| 08:25.54 | *** join/#asterisk oej (n=oej@bkkb-gw.bitcon.no) |
| 08:26.58 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 08:27.12 | x86 | morning |
| 08:27.23 | chris_ast | Hi People |
| 08:27.26 | sleepy_one | morning :-D |
| 08:28.01 | Zeeek | hi |
| 08:29.28 | x86 | anyone get SIP extensions working from MySQL / RealTime with 1.2? |
| 08:29.37 | x86 | i'm having a bear of a time |
| 08:29.59 | x86 | i've got asterisk-addons, and i've loaded res_config_mysql.so (preload in modules.conf) |
| 08:30.30 | x86 | and in extconfig.conf, i've got sipusers => mysql,asterisk,sipfriends and sippeers => mysql,asterisk,sipfriends |
| 08:30.36 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 08:30.40 | x86 | it doesnt seem to even hit the database |
| 08:30.55 | x86 | all the flat-file SIP extensions i have defined still work fine though |
| 08:32.10 | sleepy_one | are you sure your mysql connection settings are correct? |
| 08:33.12 | x86 | yeah |
| 08:33.23 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:33.41 | x86 | i see that in /var/log/asterisk/messages, when i do a restart, it says something about being successfully connected |
| 08:33.53 | *** join/#asterisk SHad|Work (n=kvirc@popust.net) |
| 08:33.59 | chris_ast | try to login to mysql from outside other than asterisk |
| 08:34.23 | x86 | like as root? |
| 08:34.32 | SHad|Work | Does anyone know will there be a version of the G729 codec for the 2.6 linux kernel? |
| 08:34.34 | x86 | ok i'm connected :) |
| 08:34.50 | chris_ast | whatever u gave in mysql_conf |
| 08:34.56 | x86 | mysql_conf? |
| 08:35.14 | x86 | res_mysql.conf ? |
| 08:35.16 | chris_ast | res_mysql.conf |
| 08:35.24 | x86 | ok right, yeah it works |
| 08:35.32 | sleepy_one | <PROTECTED> |
| 08:35.44 | Frogzoo | SHad|Work: g729 is proprietary - you won't find a FOSS codec as far as I'm aware (might be legal some countries - dunno) |
| 08:36.06 | SHad|Work | I know |
| 08:36.12 | x86 | sleepy_one: only have res_mysql.conf and cdr_mysql.conf... CDR works great ;) |
| 08:36.17 | SHad|Work | but even the current modules have to be compiled |
| 08:36.59 | chris_ast | Which softphone are you using? |
| 08:37.19 | x86 | who? |
| 08:37.28 | chris_ast | x86 |
| 08:37.49 | x86 | chris_ast: BT101 hardphone and X-Lite, but that doesn't matter |
| 08:38.26 | chris_ast | that does not matter but it is always better to check atleast with two devices |
| 08:38.49 | x86 | mmk |
| 08:38.51 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:38.52 | chris_ast | anyway I have 1.24 and sip, extensions etc are realtime and everything works |
| 08:39.00 | SHad|Work | Frogzoo: from what I gather Digium compiled the G729 modules for a variety od platforms, what is a bit puzzling to me is why there are no modules for the 2.6 kernel version (.ko) |
| 08:39.02 | x86 | can you give me your confs? |
| 08:39.14 | chris_ast | all are oneliners |
| 08:39.15 | x86 | 1.2.4 is also what I'm using |
| 08:39.26 | x86 | chris_ast: please share? :) |
| 08:40.16 | chris_ast | [settings]sipusers => mysql,asterisk,sip_buddies |
| 08:40.24 | chris_ast | extconf |
| 08:40.46 | x86 | you dont have a sippeers? |
| 08:41.04 | x86 | just sipusers? |
| 08:41.16 | x86 | maybe that's what is confusing my asterisk |
| 08:41.25 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:41.35 | chris_ast | giva a shot |
| 08:41.45 | x86 | yeah no good |
| 08:41.58 | x86 | can you please tar up your confs and post them somewhere or email them to me? |
| 08:42.13 | x86 | i mean if you do everything realtime, all your passwords are in your database right? |
| 08:42.21 | Strom_C | are dundi keys tied to the mac address of the machine? |
| 08:42.51 | *** join/#asterisk ptblank (n=MURDER1@yorbalnd-cuda2-68-70-91-158.lmdaca.adelphia.net) |
| 08:46.16 | Frogzoo | SHad|Work: you won't find g729 modules in the kernel base ever - because g729 is subject to license restrictions |
| 08:47.26 | SHad|Work | Frogzoo: I know that, I'm sorry but I think I was a bit misinformed, I was told that the g729 module is a kernel-space module from what I've seen now it seems to be an asterisk module so it doesen't matter which kernel is used by the system |
| 08:47.34 | SHad|Work | thank you anyhow >:)) |
| 08:47.49 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 08:48.26 | Frogzoo | sure |
| 08:50.44 | SHad|Work | on that note, do I really need the module for two phones to driectly talk to eachother with g729? |
| 08:51.15 | Strom_C | won't asterisk do g729 passthrough? I forget |
| 08:51.53 | SHad|Work | well I though so too, but all of my phones connect with uLaw |
| 08:52.04 | Strom_C | SHad|Work, in my experience, g729 is so imperceptibly different from GSM or Speex that it really doesn't make much sense to waste money on it |
| 08:52.24 | SHad|Work | what aboit iLBC? |
| 08:52.33 | Strom_C | they're all equally horrid |
| 08:52.43 | Strom_C | they |
| 08:52.49 | Strom_C | they're just horrid in subtly different ways :) |
| 08:52.56 | SHad|Work | hehe |
| 08:54.33 | SHad|Work | well getting the phones to use any other low bitrate codec would be nice, the allow entries in sip.conf don't seem to affect that at all |
| 08:54.41 | littleball | hello, i have a IVR system. when the users call in, my system prompt "Please enter your extension number followed by hash key". the system uses Background() cmd to capture the extension number. I encount one problem in this. After user key in # key, the system should end immediately the waiting and continue next step. In stead, the system always wait for 15 seconds to timeout and then continue next step. How to solve this problem? |
| 08:55.14 | Strom_C | SHad|Work, do the phones support any other low bitrate codec? |
| 08:55.28 | SHad|Work | they do |
| 08:55.35 | Strom_C | littleball, are the extension numbers all the same length? |
| 08:55.50 | SHad|Work | that's what's so weird, I even set the codec priority list on the phones |
| 08:56.00 | Strom_C | SHad|Work, why do you need such low bandwidth codecs? |
| 08:56.51 | mitcheloc | littleball: i think background has nothing to do with waiting for a user to press the # key, you should take that out of your greeting and then set the responsetimeout to say 3-5 seconds |
| 08:57.17 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 08:57.19 | SHad|Work | I will be connecting asterisk servers over IAX2 through relatively slow DSL connections |
| 08:57.37 | Strom_C | littleball, or just make the extensions all the same length so that it can match on the extension number without having to wait for hash or timeout |
| 08:58.08 | Strom_C | SHad|Work, how slow are we talking here? |
| 08:58.42 | jsaunders | Does anyone know what type (manufacturer/mdel) of fxo/fxs adapter is used by the Shoretel ShoreGear 120/24? |
| 08:58.46 | SHad|Work | 1024/256kbit |
| 08:58.57 | jsaunders | model |
| 08:59.12 | Strom_C | SHad|Work, how many concurrent calls do you expect to handle on each connection? |
| 08:59.26 | SHad|Work | well about 5 would be nice |
| 08:59.37 | SHad|Work | more would be great |
| 08:59.44 | *** join/#asterisk slak- (i=slak@shudup.before.you.get.rewted.biz) |
| 08:59.45 | slak- | hi |
| 08:59.50 | Strom_C | are the softphones connecting to the asterisk boxen via the dsl connections also, or are they on the same LAN segment? |
| 09:00.00 | slak- | do i actually need to build my2.6 sources for zaptel to compile? |
| 09:00.11 | slak- | i cant seem to build zaptel, ive never built my kernel sources |
| 09:00.18 | Strom_C | slak-, yes, unless the kernel headers and so on are installable as a package |
| 09:00.29 | SHad|Work | most of the phones are on the LAN but some off location phones are connected trough that DSL line |
| 09:00.32 | slak- | ok on debian got instrcutionsd? |
| 09:00.39 | slak- | instructions for debian for the kernel headers |
| 09:00.52 | Strom_C | slak-, there's a kernel headers package |
| 09:00.54 | Strom_C | search for that |
| 09:01.16 | Strom_C | install it along with the source (make sure you unzip the source) and you should be good to go |
| 09:01.29 | slak- | <PROTECTED> |
| 09:01.32 | slak- | ok |
| 09:01.35 | slak- | thanks |
| 09:01.37 | Strom_C | SHad|Work, which phone are you using |
| 09:01.51 | SHad|Work | Grandstream GX-2000 |
| 09:01.51 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:02.03 | SHad|Work | Polycom-300,500 |
| 09:02.08 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 09:02.15 | SHad|Work | and xten's softphone |
| 09:03.04 | Strom_C | show me your sip configuration on the asterisk box |
| 09:03.12 | Strom_C | pastebin it |
| 09:03.35 | SHad|Work | pastebin? |
| 09:03.43 | Strom_C | pastebin.ca |
| 09:03.52 | Strom_C | it beats spamming the channel |
| 09:04.20 | SHad|Work | hm not familiar with that |
| 09:04.29 | SHad|Work | but wiat I think I've got the order wrong in there |
| 09:04.32 | SHad|Work | wait even |
| 09:04.54 | SHad|Work | I have a few test setups so I guess I forgot to set the list here |
| 09:05.33 | jsaunders | Anyone famliar w/ ShoreGear? |
| 09:06.10 | slak- | /usr/bin/ld: cannot find -lssl |
| 09:06.13 | slak- | on * build |
| 09:06.16 | slak- | openssl-dev? |
| 09:06.39 | Strom_C | slak-, the asterisk download page lists the packages you need to install |
| 09:06.48 | SHad|Work | heh a stupid mistake Strom_C, thanks for pointing it out :) |
| 09:06.56 | Strom_C | any time :) |
| 09:07.12 | Strom_C | im cutting over from one asterisk box to another and I'm making plenty of stupid mistakes :) |
| 09:08.23 | Ikarus | I have an interesting bug with BRIStuff, if I call a number then hang up (without picking uip the phone at the other end), I get -- Hungup 'Zap/1-1', but the phone keeps ringing |
| 09:10.14 | slak- | zlib1g-dev - compression library - development |
| 09:10.17 | slak- | is that zlib-dev? |
| 09:10.30 | slak- | i cant find the proper package in debian sources |
| 09:10.51 | Strom_C | slak-, close enough |
| 09:11.02 | Strom_C | if you install the package and it doesnt compile, try a different one :) |
| 09:11.52 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 09:12.15 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:12.21 | slak- | does the asterisk-1.2.5.tgz already include sounds and addons? |
| 09:13.02 | MGSsancho | open and find out ;) |
| 09:14.15 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:15.25 | Strom_C | seriously, slak-, we're happy to help, but you can answer a lot of these questions for yourself |
| 09:15.32 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 09:15.40 | Strom_C | it's as if I were to ask you what color the sky was and you kept telling me to look upwards |
| 09:15.49 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F2DA4.dip0.t-ipconnect.de) |
| 09:16.16 | bmg505 | lol @ Strom_C |
| 09:16.30 | slak- | mang i hosed a production system |
| 09:16.36 | slak- | and im working since 8am yesterday |
| 09:16.39 | slak- | please, bare with me |
| 09:16.51 | Strom_C | you hosed a production asterisk system? |
| 09:16.53 | slak- | and, please, dont use linux software raid |
| 09:17.23 | bmg505 | whats wrong wit hsoft raid? |
| 09:17.26 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 09:17.29 | ramtha | hi |
| 09:17.35 | Mavvie | that reminds me, I have to svn update my 1.2 one |
| 09:17.36 | slak- | well it went nuts and corrupted ext3 |
| 09:17.40 | ramtha | how can i cut the last digits of a number? |
| 09:17.41 | slak- | and well i couldnt recover |
| 09:17.47 | slak- | and well, i had to reinstall the box |
| 09:17.51 | Strom_C | slak-, and you didn't have backups? |
| 09:17.54 | slak- | i did |
| 09:17.56 | ramtha | first digits works CALLERIDNUM:1 |
| 09:17.57 | slak- | of my config files |
| 09:18.02 | slak- | and /var |
| 09:18.02 | ramtha | for cut first digit |
| 09:18.11 | ramtha | how can i do this with the last digit? |
| 09:18.17 | Strom_C | ramtha: ${VAR:5:10} |
| 09:18.25 | ramtha | ic |
| 09:18.26 | ramtha | thx |
| 09:18.38 | ramtha | :5 (first):10(last) ? |
| 09:18.39 | Strom_C | slak-, but not of the full system? |
| 09:18.55 | Strom_C | ramtha, yes |
| 09:19.21 | slak- | Strom_C: none |
| 09:19.37 | Strom_C | ouch. |
| 09:19.59 | Mavvie | oh, and /var/lib/agi |
| 09:20.04 | Mavvie | oh, and /var/lib/asterisk/agi-bin |
| 09:20.08 | Mavvie | something like that. |
| 09:21.54 | Zeeek | pimp my asterisk |
| 09:22.25 | slak- | loader.c:325 __load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: |
| 09:22.27 | slak- | damnit |
| 09:22.34 | slak- | i installed asterisk-addons |
| 09:22.37 | slak- | where else is it? |
| 09:22.49 | slak- | cannot open shared object file: No such file |
| 09:23.15 | Strom_C | do you know how to use the locate command? |
| 09:23.39 | slak- | sure i do |
| 09:23.47 | slak- | all my modules are underneath there |
| 09:25.31 | slak- | do i need mysql installed for cdr_mysql to get built? |
| 09:25.48 | Strom_C | um, that might be something to consider :) |
| 09:26.01 | slak- | what if im not running mysql on my * box |
| 09:26.06 | slak- | why would i need to have it installed |
| 09:26.09 | Strom_C | i dont know |
| 09:26.13 | Strom_C | why are you installing it? |
| 09:26.22 | slak- | cos i like the gui cdr in mysql |
| 09:26.23 | [ProB]CrazyMan | you need the mysql_client and mysql_devel |
| 09:26.29 | slak- | thansk prob |
| 09:26.44 | Zeeek | start simple by getting a box running, then worry about mysql etc |
| 09:26.48 | [ProB]CrazyMan | because you need the library to compile |
| 09:26.57 | Strom_C | slak-, seriously, if your mind is dying on you, it might be worth it to take a break, get some rest, then come back when you're feeling fresh |
| 09:27.17 | slak- | i still have a dns server to configure |
| 09:27.19 | slak- | heh |
| 09:27.22 | slak- | and make sure * works |
| 09:27.29 | Zeeek | mine died the first time Allison said "Goodbye" |
| 09:27.31 | slak- | and then a whole bunch of utlities |
| 09:27.34 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:27.40 | Strom_C | slak-, you're panicking. |
| 09:27.41 | Strom_C | stop. |
| 09:28.03 | backblue | morning all! |
| 09:28.39 | slak- | im not panicking ;) its been close to 24hrs since ive been at work (20 exact), i have an 11am dentist appointment, and a 2pm meeting! |
| 09:28.45 | slak- | and its 4:30am :D |
| 09:29.18 | backblue | hehe, its 09:30 am here. |
| 09:29.19 | backblue | :P |
| 09:29.40 | Zeeek | 10h30 here |
| 09:29.43 | backblue | slak-: where are you us? |
| 09:29.47 | slak- | CT |
| 09:29.48 | slak- | USA |
| 09:29.54 | slak- | okay astrisk is running w000p |
| 09:29.58 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 09:30.08 | backblue | Zeeek: .fr? .uk? |
| 09:30.09 | Zeeek | CT id deprecated. PLease live in Delaware |
| 09:30.14 | slak- | now i need to merge back users voicemailes |
| 09:30.17 | slak- | mails |
| 09:30.19 | Zeeek | fr |
| 09:30.24 | slak- | and uhm..sounds |
| 09:30.26 | slak- | crap! |
| 09:30.57 | backblue | i need to code some domain suport, for asterisk |
| 09:31.14 | backblue | yesterday i finish up, the from domain, in chan_sip |
| 09:31.15 | *** join/#asterisk shiznatix (n=Bambr@213-35-232-62-dsl.end.estpak.ee) |
| 09:31.26 | ramtha | hmif i do somethink like this: (${CALLERIDNUM:0:3} callerid= 0049xxxxxxxx gets to 049 .. |
| 09:31.27 | ramtha | ? |
| 09:31.28 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 09:31.40 | ramtha | i thought, i cut the last 3 digits.... |
| 09:31.49 | shiznatix | Good morning all. I was wondering if anyone could tell me how to use zapata to send a fax with spandsp (if spandsp is needed) on Asterisk 1.2.4 |
| 09:32.14 | ramtha | can i do something like if CALLERID=00493221XXXX = |
| 09:32.15 | ramtha | ? |
| 09:32.25 | ramtha | X for every digit? |
| 09:32.32 | ramtha | it works for extension |
| 09:32.39 | backblue | use set() |
| 09:32.39 | ramtha | does it work for CALLERID too? |
| 09:33.09 | backblue | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set |
| 09:33.19 | backblue | Set(CALLERID(number)=000000) |
| 09:33.26 | backblue | Set(CALLERID(name)="The Name") |
| 09:33.45 | slak- | where is voicemail stored in /var if i was to replace it |
| 09:33.47 | backblue | and include the right module |
| 09:33.48 | slak- | or move to another box |
| 09:33.49 | slak- | or restore |
| 09:34.35 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:35.31 | ramtha | ok, let me say it an other way. |
| 09:35.40 | ramtha | i wnat to route a call based in his CALLERID |
| 09:35.55 | ramtha | if its calleridnum: 222XXX do this |
| 09:36.04 | ramtha | if its calleridnum 111XXX do that |
| 09:36.09 | slak- | Strom_C: hey where do i move my voicemail to restore it |
| 09:36.12 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-164-145.dsl.irvnca.pacbell.net) |
| 09:36.19 | Strom_C | slak-, beats me |
| 09:36.25 | Strom_C | check your voicemail.conf |
| 09:36.29 | ramtha | but how do i mach a calleridnum, if i only now the firs 6 digits? |
| 09:36.41 | ramtha | in example, the first 3 digits.. |
| 09:36.42 | backblue | ramtha: _222.,... |
| 09:36.46 | ramtha | ah |
| 09:36.47 | Strom_C | rather |
| 09:36.50 | Strom_C | _222XXXX |
| 09:37.00 | Strom_C | X matches single digit |
| 09:37.05 | Strom_C | . matches infnite digits |
| 09:37.12 | backblue | yes, but he wants the first 3 digits |
| 09:37.22 | Strom_C | well then he only needs _222. |
| 09:37.25 | Strom_C | not _222..... |
| 09:37.44 | backblue | i dont use ..... i used .,... |
| 09:37.49 | backblue | loke for , |
| 09:37.57 | backblue | look |
| 09:37.57 | ramtha | GotoIf($["${CALLERIDNUM}" = "_004932211063."] ? |
| 09:38.00 | ramtha | can this work? |
| 09:38.07 | backblue | ramtha: dont do that |
| 09:38.16 | Zeeek | ugly |
| 09:38.18 | Zeeek | evil |
| 09:38.21 | Zeeek | avoid |
| 09:38.46 | slak- | guys |
| 09:38.51 | slak- | where is my voicemail in /var |
| 09:38.55 | slak- | need to restore it |
| 09:38.56 | ramtha | how can i do this a better way? |
| 09:38.57 | slak- | eyes closing |
| 09:38.58 | backblue | ramtha: [context] |
| 09:39.02 | ramtha | i must screen the callerid |
| 09:39.11 | ramtha | and based on this i must put it in a context |
| 09:39.23 | backblue | exten => _222.,1,Dial(IAX/trunk/${EXTEN}) |
| 09:39.31 | Zeeek | ramtha screen the cid when the call comes in |
| 09:39.49 | backblue | or whatever you want |
| 09:39.50 | x86 | anyone have Realtime working with SIP and extensions? |
| 09:39.53 | Zeeek | s/004932234343 |
| 09:40.01 | ramtha | thats what i do woth thgotoif command i thought.. |
| 09:40.03 | x86 | i'm having a very hard time getting it to work |
| 09:40.14 | backblue | x86: why? |
| 09:40.16 | *** join/#asterisk io_error|laptop (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 09:40.40 | backblue | ramtha: i dont understand your problem, that's so simple... |
| 09:40.53 | backblue | i give you the anwser |
| 09:40.59 | x86 | backblue: well, it connects to my mysql database just fine and registers the configuration engine (according to the logs), but does not show any of my sip extensions from the database, just from sip.conf |
| 09:41.00 | backblue | awnser |
| 09:41.10 | x86 | backblue: even though extconfig.conf points it to mysql |
| 09:41.13 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 09:41.19 | Zeeek | exten => s/0142080940,1,macro(ring-evelyne) ; nelly |
| 09:41.32 | backblue | x86: check for logs when asterisk starts. |
| 09:41.34 | Zeeek | ramtha when the call comes in in the beginning of the context use this: exten => s/0142080940,1,macro(do-something) |
| 09:41.34 | x86 | backblue: and i'm not seeing anything in the logs |
| 09:41.38 | slak- | looks like i lost everyones voicemail |
| 09:41.39 | slak- | ooops |
| 09:41.40 | slak- | ;( |
| 09:41.41 | x86 | backblue: right, logs say it connects fine ;) |
| 09:42.19 | backblue | x86: it should says it parses fine too. are you using realtime, or realtime static? |
| 09:42.40 | backblue | it should only read everything at startup if you use realtime static, i think. |
| 09:42.46 | backblue | i never used realtime yet |
| 09:42.50 | backblue | maybe this week |
| 09:42.59 | ramtha | the exten is my prob.. |
| 09:43.16 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
| 09:43.24 | [hC] | Sup kids |
| 09:43.26 | backblue | i'm trying to fix some code problems in asterisk |
| 09:43.41 | [hC] | [av]bani: awake? |
| 09:43.46 | backblue | incoming external sip domains |
| 09:43.52 | backblue | full domain suport |
| 09:45.13 | backblue | dont you guys have problems with domains? |
| 09:45.25 | backblue | or you guys use allways ser? |
| 09:45.42 | Zeeek | what kind of problems with domains? |
| 09:46.40 | *** join/#asterisk julien[re] (n=julien[r@AStDenis-103-1-7-13.w81-248.abo.wanadoo.fr) |
| 09:46.58 | backblue | Zeeek: if i dial from one of my sip servers, to your sip server, what domain will you have in the from field? |
| 09:47.08 | backblue | it will say "incoming call from ..." ? |
| 09:47.31 | julien[re] | hi all |
| 09:47.35 | julien[re] | i've got a question about PRI |
| 09:47.39 | backblue | asterisk use allways the local fromdomain (configured in sip.conf) |
| 09:47.51 | backblue | to the external and internal calls |
| 09:47.59 | julien[re] | can i install an asterisk between a legacy PBX and a E1 line? |
| 09:48.30 | Ikarus | I have an interesting problem with BRIStuff, if I call a number then hang up (without picking up the phone at the other end), I get -- Hungup 'Zap/1-1', but the phone at the other end keeps ringing for some time (quite long) |
| 09:48.31 | Strom_C | julien[re], explain further what you want to do |
| 09:48.32 | julien[re] | so that calls are routed through SIP, except emmergency and local calls which should go to ISND |
| 09:48.37 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:48.41 | julien[re] | ISDN |
| 09:48.43 | backblue | julien[re]: where will asterisk stand between a E1 line and a legacy pbx? |
| 09:48.50 | julien[re] | yes |
| 09:49.00 | Zeeek | backblue I'll have to try it |
| 09:49.01 | julien[re] | the asterisk will be connected to the telco |
| 09:49.04 | Strom_C | julien[re], that should theoretically be possible |
| 09:49.08 | backblue | Zeeek: :) |
| 09:49.13 | julien[re] | and the legacy pbx will be connected to asterisk |
| 09:49.21 | Strom_C | though one wonders why you're not just replacing the legacy pbx entirely |
| 09:49.21 | backblue | Zeeek: try it |
| 09:49.24 | julien[re] | using the same cable/port as if it was connected to isdn |
| 09:49.43 | backblue | julien[re]: what its the problem of that implementations? |
| 09:49.49 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 09:49.54 | Strom_C | julien[re], I'm not sure you can set up asterisk to pretend to be the switch end of a PRI |
| 09:49.55 | backblue | that's simple |
| 09:50.01 | Zeeek | backblue I'm busy doing a stupid grunt job that will take a few more minutes. Then I'll try it |
| 09:50.22 | julien[re] | in fact i can't remove the legacy pbx |
| 09:50.31 | backblue | Zeeek: ring me, if you need some help. |
| 09:50.39 | julien[re] | because they want to keep features, etc. |
| 09:50.40 | x86 | backblue: realtime |
| 09:50.46 | x86 | backblue: says nothing about parsing |
| 09:50.48 | backblue | julien[re]: you should use both |
| 09:50.50 | julien[re] | and costly proprietary phone wiring |
| 09:50.54 | x86 | backblue: just that it loaded properly |
| 09:51.03 | Mavvie | julien[re]: we have that here. |
| 09:51.17 | julien[re] | ok and how do u configure it Mavvie? |
| 09:51.24 | Mavvie | julien[re]: an Alcatel 4400 with two PRIs into asterisk and then two PRIs towards AAPT. |
| 09:51.28 | backblue | x86: if you are not using static realtime, you should only access to sql backend, if you need, like when comes one call or something like that. |
| 09:51.29 | Mavvie | julien[re]: one as net, one as CPE. |
| 09:51.37 | julien[re] | ok that simple? |
| 09:51.56 | Mavvie | julien[re]: and then for each PRI make a context, and forward that to the other PRI. |
| 09:52.09 | Mavvie | julien[re]: that was the initial setup. |
| 09:52.12 | julien[re] | and u receive number dialed fro mthe alcatel |
| 09:52.26 | Mavvie | julien[re]: yeah: |
| 09:52.32 | Mavvie | [from-a4400] |
| 09:52.38 | julien[re] | which country are you in? |
| 09:52.56 | Mavvie | exten => _.,Dial(Zap/{TRUNK_AAPT}/${EXTEN}) |
| 09:53.02 | Mavvie | that's it |
| 09:53.05 | Mavvie | .au |
| 09:53.15 | Mavvie | and the from-aapt context is the same. |
| 09:53.17 | julien[re] | ok thanks |
| 09:54.34 | x86 | backblue: err? |
| 09:54.51 | shiznatix | how do you exactally use zapata to send faxes? |
| 09:54.53 | x86 | backblue: i'm trying to put my SIP users and extensions in mysql |
| 09:55.09 | x86 | backblue: but it's apparantly not reading the database, although it connects fine |
| 09:58.52 | Zeeek | backblue I called from one server to another and didn't have a problem with the domain |
| 09:59.37 | backblue | Zeeek: i will call you, give me your URI |
| 10:00.44 | *** join/#asterisk Heim|away (n=Heimidal@phpbb/styles/heimidal) |
| 10:01.00 | *** join/#asterisk medusaXX (n=medusaxx@p54A98DD5.dip0.t-ipconnect.de) |
| 10:01.40 | Zeeek | I don't think that'll work since we don't accept calls from "outside" |
| 10:04.33 | kmilitzer | I just asked myself who came up with the bad idea to schedule the Astricon Berlin for June 19th and 20th ... theres the Fifa World Cup in germany and during this time there are a couple of games in berlin ... will be hard/expensive to get a hotel there I guess ... |
| 10:04.57 | MGSsancho | fifa? |
| 10:05.16 | Zeeek | Friends In Fact of Asterisk |
| 10:05.41 | MGSsancho | ahhhh |
| 10:05.46 | MGSsancho | :) |
| 10:06.37 | Zeeek | backblue you have a guest context (or something) |
| 10:07.42 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:08.14 | kmilitzer | MGSsancho: football/soccer world cup |
| 10:08.39 | backblue | Zeeek: your server its not open sip? |
| 10:08.47 | *** join/#asterisk Assid (n=assid@59.183.43.45) |
| 10:09.46 | Zeeek | I don't want people calling me at 3AM? |
| 10:09.58 | Assid | auto voicemail |
| 10:10.07 | Zeeek | we don't have a desire to receive caller from unknown clients/servers |
| 10:10.26 | backblue | wheel, i will explain, users: foo1,foo2 @domain1.com | bar1,bar2 @domain2.com, foo1@domain1.com call bar1@domain2.com, the incoming call will be from foo1@domain2.com, and should be foo1@domain1.com, this is the problem. |
| 10:10.41 | backblue | Zeeek: you should :P |
| 10:10.42 | Zeeek | however, I don't understand what your problem is exactly. If we had an open context, I'm sure we could get the calls |
| 10:11.03 | Zeeek | ok |
| 10:11.19 | Zeeek | IOW, the user name is not keeping the dmain to uniquely identify it |
| 10:11.40 | Zeeek | theobvious conclusion is use unique usernames :) |
| 10:13.00 | Zeeek | is there a web page somewhere that can call an arbitrary URI? |
| 10:14.11 | kos | is asterisk@home suitable for a large organisation? |
| 10:15.02 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:15.16 | sleepy_one | kos, probably not but it is configurable |
| 10:15.17 | Zeeek | no you need asterisk@business |
| 10:15.29 | sleepy_one | lol |
| 10:15.54 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 10:16.08 | kos | Zeeek, only if i get to pay for it. |
| 10:16.16 | backblue | Zeeek: sip in asterisk, have a couple of problems. |
| 10:16.20 | Zeeek | wow, one of my servers is at 2ms from one of the SIP providers |
| 10:16.29 | sleepy_one | nice :-D |
| 10:16.37 | Zeeek | kos - oh but you will pay, I guarantee it! |
| 10:16.38 | backblue | 2ms it's pretty god :D |
| 10:16.54 | sleepy_one | hey guys is there a way to increase the volume on Zap channels? |
| 10:16.57 | Zeeek | unfortunately, I am at 250ms from that server :( |
| 10:17.10 | Zeeek | sleepy_one see zapata.conf |
| 10:17.17 | kos | how much bandwidth would a normal video/voice conversation take? |
| 10:17.37 | Zeeek | damn, it just went to 3ms |
| 10:17.39 | sleepy_one | voice 64kbps uncompressed down to 5-8kbps using GSM |
| 10:17.43 | [hC] | anyone played with the SIP load for the Cisco 7970 yet? |
| 10:17.48 | Zeeek | exit |
| 10:17.51 | Zeeek | oopos |
| 10:17.55 | Zeeek | Not. |
| 10:17.56 | sleepy_one | not yet |
| 10:18.03 | sleepy_one | do you have the firmware? |
| 10:18.04 | *** join/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 10:18.16 | [hC] | yes |
| 10:18.31 | [hC] | it doesnt appear that there is an option for specifying the sip secret in the config file |
| 10:18.44 | [hC] | which means you have to auth via host ip... which is fairly limiting |
| 10:18.49 | [hC] | SCCP seems like a better idea. |
| 10:18.55 | [hC] | but sccp has its couple bugs too |
| 10:20.16 | Zeeek | backblue so user1@dom1 is seen as user1? user1@dom2 is also user1? |
| 10:20.43 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:21.54 | backblue | Zeeek: y |
| 10:22.39 | Zeeek | why do users need un-unique names? |
| 10:23.36 | [hC] | shit i was tired like 4 hours ago, i should have gone to bed then |
| 10:23.39 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
| 10:24.49 | backblue | Zeeek: the problem its not the unique names, sip stuff should have full domain suport |
| 10:25.52 | Zeeek | there's always the bugtracker |
| 10:26.49 | Zeeek | iax2 show peers |
| 10:29.00 | RoyK | ~seen zoa |
| 10:29.10 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 5d 17h 45m 53s ago, saying: 'it looks kinda suspicious :p'. |
| 10:31.34 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 10:31.41 | FreezeS | hi guys |
| 10:32.56 | FreezeS | I'm trying to use switch => IAX2/server and I get Rejected connect attempt from 192.168.105.206, who was trying to reach 'TBD@206180' |
| 10:33.04 | FreezeS | what's with this TBD ? |
| 10:33.18 | Zeeek | I was having that same pb the other day |
| 10:33.44 | Zeeek | first make sure you have user as type on the server you are calling |
| 10:33.55 | FreezeS | hmm, I think it's peer |
| 10:33.57 | *** join/#asterisk shimi (n=moshe@unaffiliated/shimi) |
| 10:34.01 | FreezeS | i'll check |
| 10:34.02 | Zeeek | change it |
| 10:34.03 | shimi | I tried to use Asterisk BLF on my GXP-2000 - and for each extension that I define this, the LED is just constantly ON, regardless of the person being on the phone or not. any ideas? |
| 10:34.46 | FreezeS | now I get registration refused |
| 10:35.11 | Zeeek | what dod yiou change it to? |
| 10:35.15 | FreezeS | user |
| 10:35.28 | FreezeS | from peer |
| 10:35.30 | Zeeek | it has a username and password? |
| 10:35.33 | FreezeS | yes |
| 10:35.41 | FreezeS | actually, from friend |
| 10:35.41 | Zeeek | auth=md5 |
| 10:35.43 | FreezeS | not peer |
| 10:35.50 | Zeeek | friend may have been ok |
| 10:35.59 | FreezeS | ok, so back to friend |
| 10:36.12 | Zeeek | if you look up switch on the digium list, yoiu'll find some very old posts from Mark Spencer |
| 10:36.25 | Zeeek | there is one (i think peer) that just won't work |
| 10:36.32 | FreezeS | Registered IAX2 to '192.168.105.204', who sees us |
| 10:36.38 | FreezeS | so this works |
| 10:36.51 | shimi | btw "sip show subscriptions" gives me a huge list with "User" being the phone that is defined with the BLF, 27 of them! (and only one BLF number defined on the phone) |
| 10:37.04 | FreezeS | I hava a server with an BRA card, and I need to switch it to another server |
| 10:37.12 | Zeeek | I got it working but then it ALWAYS was switching so I changed the idea to use the dialplan to call through the server as in Dial(IAX2/user@server2/2003 |
| 10:37.29 | FreezeS | yeah, that's what I was trying to avoid :) |
| 10:38.34 | Zeeek | well, I did get switch working |
| 10:38.50 | *** join/#asterisk BugKham (n=lamer@ppp-58.10.64.72.revip2.asianet.co.th) |
| 10:38.54 | Zeeek | but I coulnd quite figure out how to make it only swith with certain numbers |
| 10:39.41 | BugKham | Hi, why is my isdn-pri config automatically filling a '0' as a calling out prefix? |
| 10:40.07 | BugKham | is there any way to remove it? |
| 10:40.13 | FreezeS | it's working with Dial, but I was looking for a more elegant way |
| 10:40.30 | FreezeS | Zeeek: so what did you do to make it work ? |
| 10:41.33 | Zeeek | let me see |
| 10:42.24 | Zeeek | switch => IAX2/username:pass@domain.com/context |
| 10:42.32 | FreezeS | :) |
| 10:42.55 | x86 | sweet, i got my test SIP user working from Realtime... but it's odd because he's not showing up in 'sip show peers' or 'sip show users' |
| 10:42.58 | x86 | is that normal? |
| 10:44.25 | Zeeek | FreezeS then in the called box it was host=domain.net auth=md5 type = user |
| 10:45.37 | Zeeek | FreezeS : http://www.marko.net/asterisk/archives/0210/0185.html |
| 10:46.25 | FreezeS | I've made it work :) |
| 10:46.33 | Zeeek | HOW, HOW ? |
| 10:46.35 | FreezeS | immediate=yes :) |
| 10:46.37 | Zeeek | ooh oohh |
| 10:46.41 | FreezeS | this was wring |
| 10:46.42 | FreezeS | wrong |
| 10:46.50 | Zeeek | you mean you had it |
| 10:46.57 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 10:46.58 | FreezeS | it doesn't like switching aparently :) |
| 10:47.10 | FreezeS | so, with immediate=no it works |
| 10:47.30 | Zeeek | well who ever said to put immediate=yes, the batphone ion there? |
| 10:47.48 | FreezeS | it was already there |
| 10:48.04 | Zeeek | what the ghost of Xmases past put it there? |
| 10:48.34 | FreezeS | no, most probably me, when I was tinkering with that server before :D |
| 10:48.49 | Zeeek | don't do that, I formally forbid doing that |
| 10:49.08 | Zeeek | tinkering it a form of masturbation |
| 10:49.11 | shimi | I tried to use Asterisk BLF on my GXP-2000 - and for each extension that I define this, the LED is just constantly ON, regardless of the person being on the phone or not. any ideas? "sip show subscriptions" gives me an item with a "Call ID" however the last line is "0 active SIP subscriptiuon(s)". Where could I be at error? |
| 10:49.55 | FreezeS | masturbation is good sometimes ;) |
| 10:50.31 | Zeeek | depends how long you wait between sessions |
| 10:54.05 | *** join/#asterisk t0ke (n=kaka@40.Red-83-57-222.dynamicIP.rima-tde.net) |
| 10:56.50 | *** join/#asterisk Strom_C (i=strom@66.159.243.60) |
| 10:57.05 | t0ke | anyone have one TOPEX gsm gateway connected to asterisk via E1 interface? |
| 10:59.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:02.55 | *** join/#asterisk _deg_ (n=deg@201.22.40.23.adsl.gvt.net.br) |
| 11:04.36 | *** join/#asterisk BugKham (n=lamer@ppp-58.10.69.110.revip2.asianet.co.th) |
| 11:05.34 | BugKham | when calling out from my E100P, my ${EXTEN} was fed with a leading '0', any idea? |
| 11:06.27 | Strom_C | BugKham, what does your dial statement say? |
| 11:07.25 | BugKham | exten => _02XXXXXXX,1,Dial(Zap/g2/${EXTEN:1},30) |
| 11:07.51 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 11:08.22 | BugKham | I had to deduce a leading '0' from my ${EXTEN} |
| 11:09.12 | Zeeek | looks nice |
| 11:09.23 | BugKham | if not telco will see it as "002XXXXXXX" |
| 11:10.06 | BugKham | Zeek: does the switchtype have an effect on this? |
| 11:10.55 | Zeeek | in the above your dialing 2XXXXX to the telco |
| 11:11.29 | BugKham | yeah, so that telco can get the 02XXXXXXX from my * |
| 11:12.13 | BugKham | that's what has been proved actually |
| 11:12.24 | BugKham | signalling=pri_cpe |
| 11:12.24 | BugKham | switchtype=euroisdn |
| 11:12.44 | BugKham | these are what I'm using in the zaptel.conf |
| 11:12.52 | BugKham | sorry |
| 11:12.56 | BugKham | zapata.conf |
| 11:13.48 | BugKham | I never had this problem on my TDM cards |
| 11:14.29 | BugKham | they said something about Nadi type of sending digits |
| 11:19.57 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:20.42 | Zeeek | anyone here have problems with asterisk and Dell servers? |
| 11:20.48 | kmilitzer | BugKham: What is your pridialplan? |
| 11:21.43 | *** join/#asterisk __Paulo__ (n=pirch@201-13-17-36.dsl.telesp.net.br) |
| 11:22.45 | kmilitzer | BugKham: And what in general is wrong in leavin out the first 0? |
| 11:22.58 | __Paulo__ | ~seen coppice |
| 11:23.02 | jbot | coppice <n=chatzill@91.203.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 21h 26m 33s ago, saying: 'yep. if you look at all the other timers, they use that'. |
| 11:24.08 | BugKham | kmilitzer: I didn't place this parm -> pridialplan in my zapata.conf |
| 11:24.20 | *** part/#asterisk __Paulo__ (n=pirch@201-13-17-36.dsl.telesp.net.br) |
| 11:25.11 | BugKham | tryinto put pridialplan=local now |
| 11:25.19 | Zeeek | anyone know about Junghanns cards and Dell servers? |
| 11:27.06 | BugKham | kmilitzer: pridialplan=local seems to solve my problem |
| 11:28.50 | *** join/#asterisk Strom_C (i=strom@66.159.243.60) |
| 11:31.50 | FreezeS | BugKham: also set the prilocaldialplan=local. I had some problems with that... |
| 11:35.02 | BugKham | FreezeS: okay, thanks |
| 11:36.14 | BugKham | FreezeS: what's your switchtype and signalling? I don't know if they matter |
| 11:39.56 | *** join/#asterisk kreilmeier (n=kreilmei@hq.commoveo.com) |
| 11:40.14 | kreilmeier | Hi all! My name is Michael Kreilmeier - and I am new here. I got a few issues with one of our Asterisk installations. Actually problems I am unable to solve myself. |
| 11:40.22 | kreilmeier | We got many "Coudln't or Didn't get a frame from channel" - messages. Mostly on IAX and SIP, but also on Zap channels. Googling didn't help. The source I don't understand. Has anyone here experience with that issue. Then "Bridge stops brdging follows". |
| 11:40.32 | kreilmeier | Also we got many "Got a FRAME_CONTROL (15) frame on channel IAX2/xxxxx-1" messages. I can't find out what 15 is supposed to mean. |
| 11:40.54 | *** join/#asterisk CleanerX (n=nix@p54A3B19E.dip0.t-ipconnect.de) |
| 11:43.25 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 11:43.28 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 11:43.30 | *** part/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 11:43.33 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 11:44.22 | fourcheeze | which bit of "sip show channels" is a channel? |
| 11:44.40 | fourcheeze | or rather how do I derive a channelid from that? |
| 11:45.54 | mutilator | uh |
| 11:46.00 | mutilator | the stuff under the channel heading.. |
| 11:46.02 | mutilator | like.. |
| 11:46.05 | mutilator | SIP/9898265744-fd3d |
| 11:46.16 | fourcheeze | I don't see that |
| 11:46.24 | mutilator | what do ya see |
| 11:46.35 | fourcheeze | peer/user/callid/seq/form/hold/last message |
| 11:47.05 | mutilator | o |
| 11:47.11 | mutilator | sip show channels |
| 11:47.15 | mutilator | just do a show channels |
| 11:47.24 | fourcheeze | ok |
| 11:47.34 | fourcheeze | yeah, I have channel there |
| 11:47.54 | fourcheeze | but I seem to have more channels shown on sip show channels |
| 11:48.03 | fourcheeze | and some seem to be non-active to I want to kill them off |
| 11:48.32 | mutilator | stuff in sip show channels sill also show registrations |
| 11:48.53 | fourcheeze | I've got some presence lights showing that sholdn't be there |
| 11:49.05 | mutilator | paste what it shows |
| 11:49.40 | fourcheeze | here's one that I think is non-active |
| 11:49.41 | fourcheeze | 192.168.25.66 20060015 45c6df1b5f1 00102/00008 g729 No Rx: INVITE |
| 11:50.21 | mutilator | ah i don't allow reinviting so i don't ever see that |
| 11:50.32 | fourcheeze | I don't allow that either |
| 11:50.35 | mutilator | that should timeout |
| 11:50.46 | fourcheeze | it's been there for about 12 hours |
| 11:50.54 | fourcheeze | so I'd quite like to kill it |
| 11:51.00 | mutilator | ah |
| 11:51.12 | fourcheeze | there's another one the same |
| 11:51.36 | fourcheeze | if I do soft hangup [tab] I get a seg fault too :-( |
| 11:52.16 | mutilator | well is anyone using it? |
| 11:52.18 | mutilator | restart it |
| 11:52.19 | mutilator | :P |
| 11:52.23 | fourcheeze | tried |
| 11:52.27 | fourcheeze | oh |
| 11:52.30 | fourcheeze | you mean restart asterisk |
| 11:52.33 | mutilator | ya |
| 11:52.35 | fourcheeze | yeah, people are using it |
| 11:53.32 | fourcheeze | I'll try a restart when convenient, but I don't think it will ever restart with those calls |
| 11:53.53 | mutilator | wait til theres one person on and boot to the head |
| 11:54.04 | mutilator | :P |
| 11:54.33 | *** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au) |
| 11:55.16 | Strom_C | fourcheeze, just restart...people will call each other back and go "weird, ok, lets continue" :) |
| 11:55.25 | Strom_C | blame the telco |
| 11:55.26 | Strom_C | hah |
| 11:55.52 | mutilator | i tried that once |
| 11:56.03 | mutilator | all 3 customers i killed called in saying their calls were dc |
| 11:56.14 | mutilator | after the finished their other conversations ofcourse |
| 11:58.00 | *** part/#asterisk kreilmeier (n=kreilmei@hq.commoveo.com) |
| 11:59.44 | fourcheeze | nope, I restarted but those invites are still there |
| 11:59.53 | fourcheeze | or maybe they are dying now |
| 12:00.17 | mutilator | i think invite is the init to registration or something |
| 12:00.21 | mutilator | briefly see |
| 12:04.06 | fourcheeze | I love "sip notify reboot snom" |
| 12:05.18 | fourcheeze | or reboot-snom rather |
| 12:06.15 | x86 | does that work for grandstream phones too? |
| 12:06.18 | x86 | or just snom? |
| 12:06.20 | Zeeek | snom sounds like the mucus in the lkeenex after a big sneeze |
| 12:06.22 | *** join/#asterisk oej (n=oej@bkkb-gw.bitcon.no) |
| 12:07.41 | iDunno | that'd be snot. |
| 12:08.17 | Zeeek | snom in countries where a sneeze is Achoum |
| 12:12.12 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 12:12.20 | x86 | how do i interact with the asterisk call manager? |
| 12:12.34 | Zeeek | on what level? |
| 12:12.34 | RoyK | rtfw :P |
| 12:12.53 | x86 | is there some kind of GUI for it? |
| 12:12.58 | x86 | or what can i do with it? |
| 12:13.00 | Zeeek | FOP |
| 12:13.06 | x86 | FOP? |
| 12:13.08 | Zeeek | talk to it using sockets |
| 12:13.17 | *** join/#asterisk SHad|Work (n=kvirc@popust.net) |
| 12:13.38 | SHad|Work | does anyone have any experience with asterisk realtime sip configuration? |
| 12:14.42 | SHad|Work | I just can't get it to read the sip clients from the database like it's ignoring the settings in extconfig.conf |
| 12:15.07 | SHad|Work | and also how come I don't have a /var/log/asterisk/debug ? |
| 12:15.16 | SHad|Work | do I have to enable it somewhere? |
| 12:15.16 | RoyK | logger.conf |
| 12:15.24 | Zeeek | because it isn't given in your loggin.conf? |
| 12:15.39 | RoyK | do you see the sip peers if you do a 'sip show peer xxx load'? |
| 12:15.49 | RoyK | the 'load' at the end tells asterisk to load it from realtime |
| 12:15.57 | RoyK | without that, it only shows cached peers |
| 12:16.00 | Zeeek | anyone use Junghanns and Dell? I have a friend with a big problemo |
| 12:16.09 | SHad|Work | hm |
| 12:16.55 | SHad|Work | no |
| 12:17.00 | SHad|Work | I can't see the peer |
| 12:17.25 | SHad|Work | should the peers from the config file be registered also ? |
| 12:18.52 | SHad|Work | it's a bit weird, I look into the debug log and there's a entry that the realtime table was queried and it says everything is fine but those peers still don't show up in the list |
| 12:19.11 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 12:19.32 | nettie | Hi guys, anyone have experience with asterisk and h323? |
| 12:19.58 | Strom_C | nettie, the moral of the story is "don't use h.323 and all your problems will be solved" |
| 12:20.20 | nettie | ehehehe |
| 12:20.33 | Zeeek | that's one solution |
| 12:22.03 | nettie | the point is that my ISP uses h323 and we've 2 flat channels with the included in the "package". |
| 12:22.42 | nettie | I really would like to use them because of the flat fee we already pay. I noticed there're various implementations |
| 12:23.25 | nettie | oh323 seems to be the most interesting, do you guys know what works and what doesnt work please? |
| 12:24.23 | nettie | of course I'm not referring to the protocols problems/feature which isnt very NAT agnostic and so on.. my questions are related to the stability and feature of the solution |
| 12:24.35 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 12:25.51 | nettie | does sip2h323 transcoding works? |
| 12:26.10 | *** join/#asterisk kakadu (n=blubb@p54B8DFF0.dip.t-dialin.net) |
| 12:26.29 | kakadu | hi |
| 12:26.44 | x86 | SHad|Work: i have the same issue... |
| 12:27.14 | x86 | SHad|Work: Realtime peers dont show up in 'sip show peers' or 'sip show users', but they are able to register and use SIP properly... |
| 12:27.44 | SplasPood | there's an option in sip.conf to have them show up, if I recall |
| 12:28.02 | SHad|Work | hm |
| 12:28.28 | SHad|Work | rtcachefriends=yes |
| 12:28.42 | SplasPood | yea, somethin like that |
| 12:28.48 | SHad|Work | My problem was that they wouldn't even work |
| 12:29.08 | SHad|Work | it does know, but by what chance I have no idea |
| 12:29.24 | Aurs | x86: but the table in your db shows you the registration status. I think? |
| 12:29.33 | Aurs | the sippeers table |
| 12:30.26 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 12:31.29 | SHad|Work | it should, that's why the "rtupdate=yes" options is for and it's set to true when not defined |
| 12:32.47 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 12:38.29 | medusaXX | what is the argument for sip show peer ? |
| 12:38.57 | medusaXX | if i just put there the name from sip show peers, then i get "peer not found" |
| 12:39.45 | FreezeS | is there a way to do Set(CALLERID(num) before going to context pattern matching ? |
| 12:39.54 | Zeeek | sip show peers name |
| 12:40.50 | medusaXX | sipgate/1234565 217.10.79.9 N 5060 Unmonitored |
| 12:40.59 | Zeeek | sip show peer sip<tab> |
| 12:41.05 | medusaXX | this is what show peers without arguement shows |
| 12:41.12 | Zeeek | ^^^^^^^^^^^^^^^^^^ |
| 12:41.35 | medusaXX | i dont understand.. |
| 12:41.43 | fourcheeze | do you have a tab key? |
| 12:41.44 | medusaXX | if i take do sip show peer sipgate/1234565 |
| 12:41.46 | Zeeek | *sip show peer NAME OF PEER |
| 12:42.12 | [ProB]CrazyMan | what could i do when I have timing slips, and the result of zttest is : |
| 12:42.13 | [ProB]CrazyMan | --- Results after 835 passes --- |
| 12:42.13 | [ProB]CrazyMan | Best: 100.000000 -- Worst: 99.975586 -- Average: 99.999504 |
| 12:42.22 | medusaXX | if "sipgate/1234565" is the name of the peer in my case |
| 12:42.25 | medusaXX | then it does not work |
| 12:42.34 | Zeeek | because sipgate is the name not what you put |
| 12:42.41 | medusaXX | ahh |
| 12:42.55 | Zeeek | if you followed instructions you'd have seen that a while back |
| 12:43.21 | medusaXX | i'm sorry. |
| 12:43.29 | Zeeek | 20 pushups, NOW! |
| 12:43.35 | Zeeek | come on get down there |
| 12:44.07 | Skid | christ, sipgate |
| 12:44.09 | Skid | move aaway |
| 12:44.15 | FreezeS | Zeeek: you like girls doing pushups ? |
| 12:44.17 | Zeeek | sipgate is wonderful |
| 12:44.21 | Skid | their service is shit |
| 12:44.28 | Skid | overloaded - I've been speaking to their noc |
| 12:44.28 | *** join/#asterisk _deg_ (n=deg@200.250.222.8) |
| 12:44.29 | Skid | :0) |
| 12:44.30 | Zeeek | medusaXX you are female? |
| 12:44.45 | medusaXX | no i'm not |
| 12:44.47 | Skid | I even offerd them transit heh, got it slapped back at me :0 |
| 12:44.52 | Zeeek | ok, so get down and give me 20 |
| 12:44.57 | medusaXX | lol |
| 12:45.06 | Zeeek | virtual pushups |
| 12:45.07 | FreezeS | medusaXX: in that case, you suck at choosing nicknames |
| 12:45.14 | medusaXX | i know |
| 12:45.23 | Zeeek | haha FreezeS you're in SIberia or what? |
| 12:45.27 | medusaXX | i've used this one for 6 years |
| 12:45.32 | medusaXX | but i'm not gonna change it ;) |
| 12:45.35 | Zeeek | like Medical-USA |
| 12:45.44 | Zeeek | or OffMyMeds-USA |
| 12:45.46 | FreezeS | Zeeek, relatively close |
| 12:46.07 | Zeeek | so your peers are like SIP/COmeradeYuchenko ? |
| 12:46.22 | Zeeek | IAX2/CapatilistSlut |
| 12:46.33 | FreezeS | Zeeek: ever heard about Romania ? |
| 12:46.48 | Zeeek | my grand mother was born there |
| 12:47.00 | Zeeek | but it isn't that cold there afaik |
| 12:47.06 | FreezeS | yeah :) |
| 12:47.14 | FreezeS | now there are about 15 degrees |
| 12:47.14 | Zeeek | not like where I was bork |
| 12:47.25 | Zeeek | 15 is very good! (in °C) |
| 12:47.32 | FreezeS | yeah, I know |
| 12:47.37 | medusaXX | another thing |
| 12:47.45 | medusaXX | how do i detach from the asterisk console? |
| 12:47.50 | FreezeS | CTRL+C |
| 12:47.58 | Zeeek | it's 16° in Bordeaux according to my call in |
| 12:48.01 | medusaXX | and that doesnt kill the daemon? |
| 12:48.04 | FreezeS | no |
| 12:48.11 | medusaXX | okay |
| 12:48.14 | Zeeek | medusaXX you don't |
| 12:48.25 | Zeeek | are you running safe_asterisk ? |
| 12:48.30 | Zeeek | if not you should |
| 12:48.40 | Zeeek | it will run asterisk on TTY9 (I think) |
| 12:48.56 | Zeeek | then you can switch between a normal console and asterisk with the function keys |
| 12:49.17 | _Paulo_ | Yes, its good to run from inittab. |
| 12:49.37 | medusaXX | i have no physical access to the system, normally |
| 12:49.47 | medusaXX | so i prefer attaching to the daemon by using -r |
| 12:49.50 | Zeeek | oddly enough, when I installed 1.2.5 on FreeBSD running asterisk runs it as a daemon |
| 12:49.59 | *** join/#asterisk ToTo (n=root@81-174-33-2.f5.ngi.it) |
| 12:50.08 | Zeeek | medusaXX if you use -r just exit |
| 12:50.13 | FreezeS | I made a script called "ast" that runs asterisk -cfgrvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 12:50.36 | _Paulo_ | I like to run from inittab in console 8 because when I connect via -r I got color output. |
| 12:51.02 | _Paulo_ | I dont know why, when I run as a daemon, using -r doesnot give me colors. |
| 12:51.03 | FreezeS | _Paulo_, if you run safe_asterisk, you also get color output |
| 12:51.06 | Zeeek | not a problem on my BW monitor :) |
| 12:51.07 | FreezeS | even via ssh |
| 12:51.26 | Zeeek | well, it's actually color. Green |
| 12:51.39 | FreezeS | mine is gray |
| 12:51.49 | FreezeS | silver gray |
| 12:54.29 | _Paulo_ | We have a small 12" white phosphor monitor in the server room (its 256 gray tones VGA), but green phosphor I dont see one for about 10 years. |
| 12:55.36 | shiznatix | does anyone know of a very detailed explanation from start to finish about sending faxes with asterisk? |
| 12:55.50 | FreezeS | I haven't been using a CRT for about 2 years :) |
| 12:56.57 | Zeeek | shiznatix http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk |
| 12:57.12 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:57.14 | Zeeek | title: Faxing with Asterisk |
| 12:58.13 | _Paulo_ | shiznatix, what do you want to do? |
| 12:59.06 | _Paulo_ | shiznatix, I would advice going the iaxmodem+hylafax way. |
| 12:59.39 | Zeeek | is there still a page about compatibility with various hardware like Dell servers? |
| 13:00.17 | _Paulo_ | I think redhat has an HCL |
| 13:01.33 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:08.58 | *** join/#asterisk clive- (n=pirch@dsl-146-112-180.telkomadsl.co.za) |
| 13:09.58 | clive- | anyone using software raid 1 together with asterisk on the same box ? |
| 13:10.40 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:11.04 | Curus | Yes? |
| 13:11.12 | Curus | clive-: I bet lots of us are |
| 13:11.25 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 13:11.30 | clive- | curos, i am just wonering about performance issues |
| 13:11.57 | Curus | asterisk isn't exactly I/O heavy |
| 13:12.50 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 13:14.01 | SHad|Work | shouldn't asterisk be able to recode from g726 to ilbc or ulaw? |
| 13:14.32 | SHad|Work | I get ast_channel_make_compatible errors if I have phones with different codecs |
| 13:14.52 | clive- | curos thanks |
| 13:15.06 | [ProB]CrazyMan | what could i do when I have timing slips, and the result of zttest is : |
| 13:15.09 | [ProB]CrazyMan | --- Results after 835 passes --- |
| 13:15.12 | [ProB]CrazyMan | Best: 100.000000 -- Worst: 99.975586 -- Average: 99.999504 |
| 13:16.46 | *** join/#asterisk coppice (n=chatzill@40.206.17.210.dyn.pacific.net.hk) |
| 13:17.32 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:17.38 | [ProB]CrazyMan | hi coppice |
| 13:17.54 | *** join/#asterisk michael-i (i=michael@141.41.38.185) |
| 13:18.00 | [ProB]CrazyMan | coppice: could you give me an advice what I could do against these timing slips ? |
| 13:18.17 | coppice | what hardware are you using? |
| 13:18.29 | [ProB]CrazyMan | junghanns quadbri |
| 13:19.12 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:20.03 | coppice | people have reported problems with the BRI cards. however, i don't use them and i don't know what configuration things you need to get right |
| 13:20.34 | [ProB]CrazyMan | hm ok thx anyway so I need to call kapejod.. |
| 13:20.37 | zigman | [ProB]CrazyMan ... use the rc2q drivers |
| 13:20.47 | zigman | only the modules |
| 13:21.07 | [ProB]CrazyMan | zigman: what ? |
| 13:21.21 | zigman | get latest bristuff (not the l version) |
| 13:21.28 | zigman | patch , compile, install |
| 13:21.36 | zigman | get latest bristuff (version 0.0.2) |
| 13:21.41 | zigman | compile qozap.o |
| 13:22.04 | zigman | copy that file over your original qozap (from bristuff 0.0.3) |
| 13:22.06 | zigman | done |
| 13:22.21 | [ProB]CrazyMan | I currently run at bristuff 0.0.3Pre l |
| 13:22.41 | zigman | http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1k.tar.gz |
| 13:22.45 | [ProB]CrazyMan | and that work ? |
| 13:22.51 | [ProB]CrazyMan | ok I will try that |
| 13:22.52 | zigman | http://www.junghanns.net/downloads/bristuff-0.2.0-RC8q.tar.gz |
| 13:23.21 | shiznatix | _Paulo_: we have spoken about faxing last week. I still just want to somehow send asterisk a fax and asterisk save it to the computer that it is running on as a .tif file |
| 13:23.25 | zigman | insmod qozap from 0.2.0 |
| 13:23.51 | SplasPood | shiznatix: I'm using iaxmodem /w hylafax, seems to be working well |
| 13:25.51 | shiznatix | _Paulo_: I don't have any of these modems or anything. I just have my standard nic card |
| 13:26.31 | shiznatix | _Paulo_: Can I just use hylafax to send a fax to asterisk then it save it to its computer? |
| 13:26.55 | SplasPood | no you use iaxmodem + hylaxfax on the asterisk box to receive the fax |
| 13:27.14 | _Paulo_ | shiznatix, the computer in question is the station or the server? |
| 13:27.37 | [ProB]CrazyMan | zigman: and the qozap from 0.2.0 work with * 1.2.4 ? |
| 13:27.48 | shiznatix | the computer i want to send a fax from is mine. the server is another computer |
| 13:27.48 | FreezeS | I really hate faxes |
| 13:27.54 | FreezeS | e-mail is A LOT better |
| 13:28.01 | shiznatix | no kidding |
| 13:28.17 | tdonahue | shiznatix: i use spandsp and rxfax to receive the fax, then use a custom perl script to mail it to the recipient as a pdf |
| 13:28.19 | ToTo | hi all, i'm looking for a method to add dinamically user profile on a sip.conf file, someone can suggest a way to me? |
| 13:28.34 | FreezeS | ToTo: Asterisk Realtime |
| 13:28.41 | ToTo | FreezeS, tnx |
| 13:28.58 | shiznatix | tdonahue: how do you send the fax? |
| 13:29.11 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:29.16 | shiznatix | Maybe thats my problem, how can i send a fax to asterisk if I don't have a iaxmodem or anything fancy like that |
| 13:29.26 | SplasPood | using a fax machine? |
| 13:29.32 | SplasPood | or a fax modem and a pots line? |
| 13:29.33 | tdonahue | with a fax machine.... you could use make a email to fax gateway and use txfax |
| 13:29.40 | _Paulo_ | shiznatix, what OS are you using in your machine? |
| 13:29.45 | shiznatix | Debian |
| 13:30.06 | SplasPood | you want to send yourself faxes? |
| 13:30.18 | tdonahue | but i have not had the time to write one (nor do i think my boss would allow me to publish it if i had) |
| 13:30.37 | shiznatix | I want to send ANYONE a fax |
| 13:30.47 | _Paulo_ | shiznatix, iaxmodem is pure software. |
| 13:30.57 | SplasPood | you could theoretically use the iaxmodem+hylafax solution to send as well |
| 13:31.04 | shiznatix | I just want a fax to go to asterisk in anyway then asterisk save it or email it or do anything with it, just get it |
| 13:31.08 | _Paulo_ | shiznatix, just compile it and install on your machine |
| 13:31.10 | SplasPood | hylafax has a client/server architecture |
| 13:31.38 | SplasPood | shiz: what you keep describing is receipt of a fax, not sending |
| 13:31.43 | shiznatix | do I have to install this stuff on the asterisk machine or can I do anything from my own machine? i have spandsp isntalled on asterisk but it is a pain to get anyone to install anything else |
| 13:31.43 | _Paulo_ | shiznatix, then you will have a virtual modem. |
| 13:32.02 | coppice | shiznatix: install either a) spandsp/rxfax/txfax or b) spandsp/iaxmodem |
| 13:32.04 | coppice | configure as indicated in many places |
| 13:32.05 | coppice | done |
| 13:32.14 | _Paulo_ | shiznatix, you can install iaxmodem on your machine. |
| 13:32.39 | SplasPood | he keeps talking about receiving faxes on his asterisk, but then says he wants to send |
| 13:32.41 | SplasPood | makes no sense. |
| 13:33.06 | shiznatix | errr |
| 13:33.14 | shiznatix | I just want asterisk to be able to deal with a fax |
| 13:33.23 | Curus | "Deal with"? |
| 13:33.25 | shiznatix | i don't know how to send, i don't know how to recieve, i just want it to work |
| 13:33.26 | [ProB]CrazyMan | he wants to send an fax from * to the same * box |
| 13:33.31 | _Paulo_ | coppice, Hi... |
| 13:33.38 | coppice | hi |
| 13:33.49 | shiznatix | [ProB]CrazyMan: i think you are correct |
| 13:34.02 | SplasPood | shiz: Well if you pay me then I'll make it work and you don't need to "know" :P |
| 13:34.29 | SplasPood | [ProB]CrazyMan: Does that make any sense? |
| 13:34.37 | Curus | SplasPood: You'll just secretly put the paper through a copier and pretend you sent it |
| 13:34.46 | SplasPood | Curus: shh |
| 13:34.58 | Curus | I know consultants, I used to be one |
| 13:34.58 | [ProB]CrazyMan | SplasPood: for me not realy.. but maybe he just want to make some tests ? |
| 13:35.39 | shiznatix | SplasPood, that would be great if i had money |
| 13:35.43 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 13:36.01 | SplasPood | shiznatix: Well you either set things up yourself and thus "know", or you pay someone to "know" for you. |
| 13:36.11 | Zeeek | If a frog had wings, he wouldn't bump is ass every time he jumped |
| 13:36.28 | _Paulo_ | Im having a hard time trying to understand how libmfcr2 stuff works. |
| 13:37.15 | shiznatix | SplasPood, I am aware of this. I would like to 'know' but I don't see how I can just do some easy test |
| 13:37.58 | SplasPood | well asterisk has zero support for sending or receiving faxes out of the box.. |
| 13:38.08 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:38.13 | shiznatix | SplasPood, I need nothing perminant, just the ability for asterisk to recieve a fax and save it to the computer asterisk is running on. I thought I knew when I tried to use rxfax and txfax but when saving the fax it just hangs. That is why I think that sending my fax is where my problem is |
| 13:38.37 | SplasPood | ahh well I don't know tx/rxfax.. but I have iaxmodem+hylafax doing exactly what you describe |
| 13:38.55 | SplasPood | you can replace hylafax with any linux fax software that'll talk to a standard fax modem |
| 13:39.20 | SplasPood | and iaxmodem is VERY easy to setup |
| 13:39.27 | SplasPood | iaxmodem.sf.net |
| 13:39.30 | shiznatix | i just installed hylafax and it asked me all sorts of things like my country code and area code (we don't have area codes here) and this is not what I want because I want to be able to only send on the local network |
| 13:39.30 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 13:39.55 | SplasPood | send? you just said receive. |
| 13:40.58 | shiznatix | is that was hylafax does? |
| 13:41.05 | Zeeek | it is better to give than to receive |
| 13:41.05 | SplasPood | do me a favor |
| 13:41.08 | coppice | I think he is deeply confused :-\ |
| 13:41.08 | SplasPood | define 'send' for me |
| 13:41.21 | _Paulo_ | shiznatix, hylafax is a fax server |
| 13:41.40 | _Paulo_ | shiznatix, it needs at least one modem to work |
| 13:41.59 | shiznatix | send: I have a .tif or .jpg or .png and I want to 'send' this as a fax to asterisk then asterisk save it itsself |
| 13:42.03 | _Paulo_ | shiznatix, to use hylafax with * you use a virtual modem called iaxmodem |
| 13:42.23 | [ProB]CrazyMan | maybe he is in a big company and want only to send faxes internaly ? |
| 13:42.34 | shiznatix | yes |
| 13:42.35 | _Paulo_ | shiznatix, if you use Debian, you can install hylafax-client |
| 13:42.45 | [ProB]CrazyMan | why as fax and not as email ? |
| 13:42.49 | SplasPood | yea |
| 13:43.10 | shiznatix | good question, if i was the boss I would tell you |
| 13:43.29 | SplasPood | so you have a big company |
| 13:43.32 | SplasPood | two or more offices |
| 13:43.32 | [ProB]CrazyMan | why do you want anyway to send this files to asterisk ? |
| 13:43.37 | SplasPood | and you want to fax between them |
| 13:43.40 | SplasPood | correct? |
| 13:44.05 | shiznatix | SplasPood, yes |
| 13:44.12 | SplasPood | ahh ok |
| 13:44.16 | SplasPood | what will be on each end |
| 13:44.20 | SplasPood | how is the fax going to be generated |
| 13:44.44 | shiznatix | In any way but with a fax machine as of right now |
| 13:45.01 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:45.10 | SplasPood | shiz: when its in production tho |
| 13:45.48 | shiznatix | No idea, probably without fax machines but a fax machine might be at the end but nobody knows as of right now |
| 13:46.19 | SplasPood | if fax machines aren't involved at all, why deal with faxing? :P |
| 13:46.25 | _Paulo_ | coppice, I read simplesched.c, but could not figure where the tones scheduled with uc_schedule_event are sent, nor how can I control how long it will play. |
| 13:46.56 | _Paulo_ | coppice, do you have some hint? |
| 13:47.11 | shiznatix | SplasPood, I don't know. I have no say over any of this. I just have to do as I am told and I was told to use faxes |
| 13:47.29 | coppice | look at how the other tones are played. turn on the tone, and start a timer. in the callback routine which services the timer, turn off the tone and take the next step |
| 13:47.34 | shiznatix | SplasPood, There might be a fax machine at one end and just a computer at the other end |
| 13:48.18 | SplasPood | well you need fax modems then... So if you want to test you'll need some form of fax modem.. be it a fax machine, a hardware fax modem and phone line, fax service like efax, or another asterisk+iaxmodem combo |
| 13:48.33 | _Paulo_ | coppice, I see now how Ive made funny things... |
| 13:48.41 | vgster | does anyone have any ideas why one of my asterisk boxes keeps spawning new asterisk processes? |
| 13:48.51 | SplasPood | lonely? |
| 13:49.05 | _Paulo_ | coppice, I think I found some bugs at the telco side. |
| 13:49.23 | SplasPood | vgster: maybe the script your using to start asterisk.. if any.. |
| 13:49.39 | mutilator | i use daemon tools to start my asterisk |
| 13:49.48 | mutilator | just run it with -f so it doesn't fork |
| 13:49.54 | mutilator | otherwise it restarts it a billion times |
| 13:49.56 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 13:50.38 | vgster | odd they have gone now but yesterday there were loads. i use freepbx atm |
| 13:50.42 | vgster | so amportal start |
| 13:50.48 | vgster | which works fine on my other box |
| 13:50.58 | SplasPood | maybe they got bored and went home |
| 13:51.13 | vgster | maybe, looks like its firing a new thread for each call |
| 13:51.52 | Hmmhesays | i like freepbx |
| 13:51.59 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 13:52.25 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 13:52.31 | eliel | hello |
| 13:52.44 | vgster | has no one seen this before? |
| 13:52.47 | _Paulo_ | coppice, can we talk in pvt? |
| 13:53.21 | eliel | is there an easy way to implement user access control on channels in asterisk? |
| 13:54.12 | _Paulo_ | eliel, show application Authenticate ? |
| 13:56.06 | eliel | _Paulo_: hmmm, and the usernames? |
| 13:57.17 | _Paulo_ | eliel, I think if you need something fancier you will have to go the AGI way. |
| 13:57.31 | eliel | _Paulo_: ok, thank you |
| 13:57.52 | _Paulo_ | eliel, I have perl AGI, it works very well. |
| 13:58.51 | eliel | _Paulo_: OK, i will give it a try... |
| 13:59.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:59.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:00.02 | *** part/#asterisk michael-i (i=michael@141.41.38.185) |
| 14:00.15 | _Paulo_ | coppice, why the timers are set to -1 inside the callback functions? |
| 14:00.52 | coppice | -1 means the timer is not active |
| 14:01.49 | fu3|gone | houston sucks |
| 14:02.07 | jsharp | Indeed it does. |
| 14:02.17 | fu3 | lots of beautiful women though |
| 14:02.18 | fu3 | LOTS |
| 14:02.23 | iDunno | what? where? |
| 14:02.24 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
| 14:02.31 | fu3 | in Houston |
| 14:02.33 | iDunno | oh, Houston. that's Miles away. |
| 14:02.39 | fu3 | but otherwise, that city licks. |
| 14:03.07 | *** join/#asterisk lorinc (n=ang@caracas-0641.adsl.interware.hu) |
| 14:03.25 | coppice | lots of beautiful women (if true) is a completely adequate reason for a city to not suck |
| 14:03.47 | fu3 | well.. |
| 14:03.56 | fu3 | no.. Houston sucks. |
| 14:04.02 | fu3 | i'd take the women away with me |
| 14:04.10 | jsharp | Houston sucks to so much of a degree, there's not enough to offset it. |
| 14:06.50 | coppice | dallas is widely considered to suck more than houston |
| 14:06.53 | [ProB]CrazyMan | zigman: still have timing slips |
| 14:07.34 | coppice | austin might suck even more for all I know, but it so boring it hard to stay awake long enough to find out |
| 14:07.44 | _Paulo_ | coppice, I see now... uc_schedule_event return how many timers of this kind are active... I mistake the timer counters with timer values in ms... |
| 14:08.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:08.28 | _Paulo_ | <:-| dumb hat |
| 14:08.30 | jsharp | Austin is good if you're a drunken frat boy or a political vegetarian. |
| 14:08.35 | Winkie | is there any decent documentation on CDR Manager events? |
| 14:08.42 | Zeeek | austin is the heart of the most lively Textas music scene |
| 14:11.59 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 14:12.27 | DeeJay[2] | is it possible to add fields in the cdr over odbc? |
| 14:12.46 | DeeJay[2] | Suppose I would like to add a "hostname" field to know from which server the row comes from.. |
| 14:14.39 | *** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe) |
| 14:15.53 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:16.23 | FreezeS | how can I select a certain ZAP channel when I want to dial out ? |
| 14:16.38 | Zeeek | by reading the dial command doc? |
| 14:16.41 | FreezeS | I have a digium card with 4 FXO ports and I need to select a certain port to dial out |
| 14:16.57 | jsharp | Zap/1, Zap/2, Zap/3, Zap/4 |
| 14:20.02 | FreezeS | as I understood, zap/3 is the third group from zapata.conf |
| 14:20.14 | FreezeS | and there is my card with 4 ports |
| 14:20.28 | FreezeS | if I dial zap/g3 it dials out through a free channel |
| 14:20.43 | FreezeS | but I want to select the channel I'm dialing out through |
| 14:20.57 | jsharp | Zap/3 is zaptel port 3. |
| 14:21.00 | jsharp | Zap/g3 is group 3 |
| 14:21.09 | jsharp | Zap/3 specifies a specific port. |
| 14:21.29 | FreezeS | hmm... |
| 14:21.32 | FreezeS | let's see |
| 14:22.45 | FreezeS | ok, that's it :) |
| 14:22.46 | FreezeS | thanks |
| 14:23.01 | mutilator | man i'de hate to be asian |
| 14:23.09 | mutilator | get a phd and ppl walk around calling you "dr wang" |
| 14:23.13 | FreezeS | because of the short dick ? |
| 14:23.49 | FreezeS | mutilator: you've got an asian coleague you're calling Dr. Wang ? :) |
| 14:24.23 | mutilator | yep, Dr. xinli wang, comp sci phd |
| 14:26.36 | Hmmhesays | i don't want, anybody else, when I think about you I touch myself wohahahah |
| 14:26.48 | *** part/#asterisk Holos (n=asdf@204.101.26.106) |
| 14:27.00 | FreezeS | test |
| 14:27.10 | Hmmhesays | Dr. Wang is possibly one of the greatest names ever |
| 14:27.35 | coppice | Dr Hu might be better |
| 14:27.45 | Hmmhesays | Dr. Phuck hu |
| 14:27.56 | jsharp | Phu Yuk |
| 14:28.53 | Hmmhesays | everybody wang chung tonight |
| 14:29.47 | FreezeS | well, for us romanians, Mu Chin Chai is very funny :) |
| 14:30.15 | FreezeS | or Cha In San |
| 14:30.35 | FreezeS | or Ling Cha Puk |
| 14:35.04 | *** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe) |
| 14:35.28 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 14:38.02 | _Paulo_ | coppice, in the callback function, to turn the off the tone, should I use set_mf_signal(uc, ch, 0) or some other value instead of 0? |
| 14:38.19 | coppice | use zero |
| 14:38.24 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 14:39.47 | _Paulo_ | coppice, I would like to hire someone with your knowledge to do this for me. My head is aching so much! |
| 14:40.31 | tzafrir | I'm banging my head at the wall right now because I can't make a certain asterisk register with iax to another Asterisk server |
| 14:40.47 | tzafrir | I'm trying to figure out what I did wrong |
| 14:41.43 | tzafrir | Both are behind NAT and port-forwarded |
| 14:42.14 | tzafrir | (not the same NAT) |
| 14:43.09 | coppice | _Paulo_ how quickly do you need this? |
| 14:44.31 | _Paulo_ | coppice, I cant go online with this fax service without blocking collect calls, otherwise i will be pretty much abused. |
| 14:44.56 | FreezeS | is there a way to make the call return to the person that transfered it in case it's not answered ? |
| 14:45.16 | _Paulo_ | FreezeS, see the "T" extension |
| 14:45.45 | _Paulo_ | FreezeS, and the "t" extension |
| 14:46.00 | FreezeS | thanks |
| 14:49.02 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 14:51.45 | gaspiz | hi, how can I start asterisk under a diferent user (not root) |
| 14:52.23 | _Paulo_ | coppice, Even if you are buzy right now I think you will get this working faster than I, and i will gladly pay your time. |
| 14:54.17 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:59.32 | *** join/#asterisk ariel (n=ariel@100sdl30m31.codetel.net.do) |
| 14:59.50 | *** join/#asterisk tseno (n=tsenotan@212.56.13.18) |
| 15:01.41 | tseno | i have an asterisk running and i wish to see it work and test some extensions ... but i dont know how to make i call |
| 15:01.48 | tseno | can someone help me |
| 15:01.55 | tseno | with this |
| 15:02.23 | _Paulo_ | coppice, the alternative for me is dialogic boards, wich are a couple thousand mor expensive. |
| 15:02.39 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 15:03.32 | *** join/#asterisk stormfr (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 15:03.33 | coppice | last time I used Dialogic cards for R2, they sucked |
| 15:03.40 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 15:03.40 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 15:04.27 | coppice | they were more than a couple of thousand, too |
| 15:05.02 | _Paulo_ | coppice, well, I could expent this money on services instead of hardware |
| 15:06.02 | _Paulo_ | coppice, are you too buzy right now or just dont have interest in this kind of job? |
| 15:06.04 | austinnichols101 | anyone familiar with the procedure to downgrade 79xx firmware. Is there anything 'extra' you need to do? |
| 15:06.29 | coppice | _Paulo_ its on a list of things to do |
| 15:07.36 | _Paulo_ | coppice, cant I bribe you? :-) |
| 15:08.57 | Hmmhesays | is there anything wrong with calling one macro from another? |
| 15:10.37 | *** join/#asterisk fogall (n=fogall@customer-200-79-84-78.uninet-ide.com.mx) |
| 15:10.46 | _Paulo_ | Hmmhesays, not that I know, but I would pass every ${ARGn} just to avoid bugs |
| 15:11.36 | coppice | Hmmhesays: its OK, as long as they don't call each other rude names |
| 15:11.47 | fogall | im getting a warning msg when i connect to asterisk |
| 15:11.53 | fogall | this is the warning msg |
| 15:11.57 | *** join/#asterisk doolph (n=doolph@201.227.72.230) |
| 15:11.57 | fogall | 060321-090712 WARNING[4080]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x810aa30 (len 515) to 192.168.0.30:5060 returned -1: Bad file descriptor |
| 15:12.08 | fogall | what does it mean? |
| 15:12.46 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
| 15:12.52 | Hmmhesays | i'm trying to decide if I should make this dp into one giant macro or not |
| 15:12.52 | shanky | hi, good afternoon |
| 15:13.36 | fogall | hi, good morning |
| 15:13.46 | fogall | im getting a warning msg when i connect to asterisk |
| 15:13.51 | fogall | 060321-090712 WARNING[4080]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x810aa30 (len 515) to 192.168.0.30:5060 returned -1: Bad file descriptor |
| 15:13.56 | fogall | what does it mean? |
| 15:14.57 | _Paulo_ | coppice, if I send "answer" then "clearback" but dont send answer again, there is a curious efect: the collect call recording is skiped... |
| 15:14.59 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2l.dialup.mindspring.com) |
| 15:15.13 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2l.dialup.mindspring.com) |
| 15:16.09 | coppice | if you clearback and do nothing else the call actually clears. nothing strange about that |
| 15:16.18 | _Paulo_ | coppice, But the call is answered. Weird... |
| 15:16.20 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 15:16.43 | lzhang | is there any other way to hangup a channel if soft hangup isn't working? |
| 15:17.19 | _Paulo_ | coppice, clearback should trigger a drop call event or am i wrong? |
| 15:17.31 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 15:17.32 | ramtha | peace |
| 15:17.52 | ramtha | why das * not dial pri 1000 if the calleridnum is correct? |
| 15:17.57 | coppice | _Paulo_ not really. drop call is something the application initiates |
| 15:17.58 | ramtha | exten => _X.,3,GotoIf($["${CALLERIDNUM}" = "_004932X."]?${EXTEN}|1000) |
| 15:18.21 | *** join/#asterisk mattjdude (n=matt@24.96.136.141) |
| 15:18.53 | gaspiz | because you use _X. |
| 15:19.13 | gaspiz | this matches number only 0123456789 |
| 15:19.39 | RoyK | gaspiz: no... |
| 15:19.46 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 15:19.48 | ramtha | gaspiz: thats evil, i musst user minimum the first 6 digits |
| 15:19.50 | RoyK | gaspiz: a dot at the end means 'and whatever' |
| 15:19.54 | _Paulo_ | coppice, should I call start_call_disconnected after set_abcd_signal(uc, ch, mfcr2->back_abcd_clear_back) ? |
| 15:20.13 | ramtha | if the calleridnum is: 004932211063030 it should jump to pri 1000? |
| 15:20.15 | ramtha | right? |
| 15:20.23 | ramtha | but i did not do that |
| 15:20.33 | ramtha | if i type the whole nummber in there, it works |
| 15:20.51 | ramtha | there must be an error in "_004932X." |
| 15:20.55 | ramtha | but what.. |
| 15:21.01 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 15:21.21 | sevard | Does anyone ever get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) ? |
| 15:21.27 | _Paulo_ | ramtha, use the "i" extension |
| 15:21.44 | jsharp | sevard: Only when * isn't running. |
| 15:21.45 | sevard | When it does exist, asterisk is running and the pid and ctl files are asterisk:"nogroup |
| 15:21.57 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 15:22.01 | _Paulo_ | ramtha, I think this jump berravior is deprecated |
| 15:22.17 | *** join/#asterisk chrismog (n=chrismog@mog.traxtech.net) |
| 15:22.25 | ramtha | _Paulo_: is that really what i want? i mean it must work like this, didn´t it? |
| 15:22.31 | lzhang | sevard: do you have the proper permissions on /var/run/asterisk/ ? |
| 15:23.41 | sevard | lzhang: BLAH i should just spend 5 more minutes figureing thing out from now on. |
| 15:23.57 | mattjdude | you can't put pattern matches in the caller ID matching |
| 15:24.11 | *** join/#asterisk himalrana (n=himal@61.17.213.79) |
| 15:24.21 | ramtha | hmm |
| 15:24.24 | himalrana | hello! |
| 15:24.33 | sevard | I've been reading all day on asterisk sip security and I've come to the conlclusion that there really isn't any. Is that a correct assumption? |
| 15:24.34 | ramtha | how can i do routing based on calleridnum? |
| 15:24.50 | _Paulo_ | ramtha, sorry, I misread your case |
| 15:25.06 | ramtha | i have two nummberblocks |
| 15:25.19 | ramtha | and based on witch block calls, i must do an other routing |
| 15:25.24 | himalrana | is video confernce possible with asterisk? |
| 15:25.44 | RoyK | video call, at least |
| 15:25.47 | RoyK | two parts |
| 15:25.53 | sevard | himalrana: with a video enabled phone and a video codec you may make a video call |
| 15:26.05 | ramtha | so i want to read the head of the block (00493232110XXX / 004932212224XXX) and route it, based on its information |
| 15:26.10 | ramtha | any solutin for that? |
| 15:26.28 | *** join/#asterisk fugitivo (n=ajf@201.255.177.90) |
| 15:27.08 | himalrana | yes but it will not be conference but video call |
| 15:27.14 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 15:28.22 | ramtha | hmm |
| 15:28.25 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 15:28.37 | ramtha | is it really such a special case, waht i want to do? |
| 15:29.04 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 15:29.23 | Abydos313 | good morning everyone |
| 15:29.37 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 15:31.31 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:31.49 | himalrana | i think video conferencing is possible as asterisk does with audio! |
| 15:32.16 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 15:32.28 | *** join/#asterisk snip3r (n=sniper@195.246.199.136) |
| 15:32.52 | Winkie | anyone got a clue how the uniqueid is formatted? |
| 15:32.57 | Winkie | or even where it's defined? |
| 15:33.33 | *** join/#asterisk devilpim (n=Pim@195.135.145.195) |
| 15:33.44 | himalrana | does any one has interest with video conferencing with asterisk? |
| 15:33.53 | brodiem | has anyone used the UIP200? Looking for some reviews/feedback |
| 15:34.15 | devilpim | hi....im having problem with fax |
| 15:34.55 | devilpim | just wondering if there's anyone familiar with faxing |
| 15:35.06 | doolph | what's your problem with fax |
| 15:35.26 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@88.154.4.239) |
| 15:35.48 | devilpim | i have a number - direct number connect to my asterisk box - i am able to send the fax successfully with this number |
| 15:36.41 | devilpim | but now i want to redirect a number -- say i have 0800xxxx and when someone fax to this number, i want it to be fwd to my asterisk box |
| 15:37.33 | devilpim | but hmm...it doesnt go in....it is forwarded to the box...but it seems that it didnt detect that its a fax signal... |
| 15:37.54 | devilpim | it just answer like a normal call |
| 15:38.06 | devilpim | im not sure where i should check.... |
| 15:38.09 | Curus | Is there an easy way to limit maximum call duration? |
| 15:38.24 | tzanger | Curus: AbsoluteTimeout |
| 15:38.28 | Curus | Thanks |
| 15:38.59 | _Paulo_ | doolph, Ive ready everything about * fax |
| 15:39.23 | devilpim | anyways, is there anyway to identify the dialed number? |
| 15:39.24 | Winkie | devilpim: how is it being delivered? |
| 15:39.29 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 15:40.16 | devilpim | hmm..there's a router |
| 15:40.42 | Hmmhesays | ok one giant super macro is not going to work |
| 15:40.44 | Hmmhesays | damnation! |
| 15:42.04 | *** part/#asterisk shanky (i=jramirez@217.11.114.145) |
| 15:42.41 | medusaXX | what is the correct string to match all numbers starting 0900 ? |
| 15:42.49 | medusaXX | if i dont know how long the number can be |
| 15:42.51 | FreezeS | _0900. |
| 15:42.59 | jsharp | Yeah, what FreezeS said. |
| 15:43.03 | RoyK | :) |
| 15:43.30 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 15:45.38 | stormfr | Hello, i have several time a day "Too many open files". Limit at much higher and process running as root. call number go to more than 120 channels with transcoding (no zap just sip or iax) |
| 15:46.28 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 15:47.12 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 15:47.41 | russellb | stormfr: read the section in the README about file descriptors |
| 15:47.58 | FreezeS | chan_zap.c:8511 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 <--- I get a lot of these every 10 minutes or so, but the PRI card is working properly. Should I worry ? |
| 15:48.14 | jsharp | Timing slips? |
| 15:48.39 | FreezeS | what are those ? |
| 15:49.20 | FreezeS | chan_zap.c:8511 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 <--- and some times things like this |
| 15:49.35 | devilpim | question:i want the fax to be send to an extension eg. 123 ...but then....when the fax signal is detected, it will be sent to [fax] extension.....what should i do about this?? |
| 15:49.38 | russellb | FreezeS: is this a Digium card? |
| 15:49.43 | FreezeS | russellb: yes |
| 15:49.49 | russellb | FreezeS: contact support@digium.com |
| 15:49.58 | russellb | they can help you check some common things that cause that to occur |
| 15:50.16 | jsharp | timing slips - when the clock on your PRI card and your PRI source get out of sync if you're not clocking off the PRI. |
| 15:50.25 | FreezeS | thing is, it's working perfectly. No call drops, no echo, nothing unusual, except these messages |
| 15:50.28 | RoyK | stormfr: echo lots > /proc/sys/fs/file-max and 'ulimit -n lots' |
| 15:51.08 | FreezeS | jsharp, so if I'll clock locally, these would dissapear ? |
| 15:52.02 | lzhang | guys, is there any other way to hangup a channel aside from soft hangup? |
| 15:52.21 | lzhang | it seems like I have a hung channel but soft hangup isn't doing the trick |
| 15:52.33 | Hmmhesays | yank your cat 5 cable |
| 15:52.54 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
| 15:52.58 | *** part/#asterisk devilpim (n=Pim@195.135.145.195) |
| 15:53.06 | lzhang | haha right |
| 15:54.00 | Hmmhesays | reboot |
| 15:54.21 | lzhang | yea I don't want to have to restart asterisk just for one hung channel |
| 15:54.27 | lzhang | but I guess I will if I have to |
| 15:54.33 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:54.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:55.11 | *** join/#asterisk fosco (i=fosco@tao.mu) |
| 15:55.14 | fosco | hi |
| 15:55.20 | fu3 | hi |
| 15:56.22 | fosco | for using /etc/security/limits.conf with asterisk, must I use a specific file un /etc/pam.d/ ? |
| 15:56.32 | stormfr | f |
| 15:57.01 | fosco | or just I use /etc/pam.conf ? |
| 15:57.22 | fosco | (for session required pam_limits.so) |
| 15:58.07 | Raszh | how you do your pam configs is up to you |
| 15:58.33 | Raszh | that you're doing so for Asterisk doesn't matter |
| 15:59.05 | tzafrir | fosco, generally you should edit files in pam.d rather than pam.conf . pam.conf is deprecated, IIRC |
| 15:59.24 | malverian[work] | Is there any way to only subscribe to certain events in manager connections? |
| 15:59.37 | fosco | tzafrir: yep, but there is no 'asterisk' file under pam.d |
| 15:59.44 | fosco | so I use 'other' |
| 15:59.45 | wunderkin | malverian[work]: only by types, not by event name |
| 15:59.52 | tzafrir | fosco, because asterisk does not use pam |
| 16:00.26 | fosco | /etc/security/limits.conf is only used with pam I think, no? |
| 16:01.21 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 16:02.33 | *** join/#asterisk mattjdude (n=matt@gateway.digium.com) |
| 16:02.37 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 16:02.49 | tzafrir | does iax.conf have any equivalent to sip.conf's externip/externhost? |
| 16:03.29 | russellb | no, why would it need it |
| 16:04.17 | tzafrir | I have no idea why one server asterisk won't register to another. But both are behind different NATs |
| 16:04.50 | russellb | is the registration message making it to the other server? |
| 16:04.55 | russellb | verified with iax debug? |
| 16:05.40 | austinnichols101 | anyone using cisco 8.2 fw yet? |
| 16:05.42 | tzafrir | I get error messages that registration fails |
| 16:06.00 | tzafrir | on the recieving side |
| 16:06.02 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 16:07.01 | malverian[work] | I need to get my hands on the SIP firmware for my Cisco 7960 phone... but I don't have Cisco account to download it :-/ |
| 16:07.09 | tzafrir | what is the maximal length of a username? There shouldn't be a problem with 12 chars, right? |
| 16:07.15 | malverian[work] | I bought the phone second hand. |
| 16:07.28 | russellb | tzafrir: that should be fine |
| 16:07.36 | russellb | maybe you have a typo somewhere, i don't know |
| 16:07.49 | malverian[work] | Anyone know if there is a way to download it without getting an account? |
| 16:07.58 | russellb | i mean, iax registrations are obviously a pretty common thing. if they were broken, there would be a lot of upset people out there |
| 16:07.58 | malverian[work] | Or at least, without paying more money :-P |
| 16:08.28 | austinnichols101 | malverian[work]: you're asking in the wrong place |
| 16:08.45 | tzafrir | I'm still with 1.2.4 . Anything that critical broken there? |
| 16:08.46 | malverian[work] | I'm referring to legal means of course. |
| 16:09.00 | austinnichols101 | legal = you need to purchase a copy |
| 16:09.09 | austinnichols101 | which means you need to pay money |
| 16:09.12 | malverian[work] | I already bought the phone though, I have no use of the SCCP image :-P |
| 16:09.30 | malverian[work] | Approx price? |
| 16:09.31 | austinnichols101 | you can use the SCCP with asterksk. |
| 16:09.43 | malverian[work] | Yeah, but it's a pain.. and the rest of my phones are SIP. |
| 16:10.02 | austinnichols101 | it's not very expensive. check voip-info.org and there's a list of resellers |
| 16:10.42 | malverian[work] | Alright. |
| 16:13.31 | tzafrir | Is a peer entry relevant to the success of the registration itself? |
| 16:13.58 | austinnichols101 | malverian: http://www.voip-info.org/wiki/index.php?page=Cisco+Phones - they say about $8/year |
| 16:13.58 | tzafrir | Didn't think so, but some of the error messages I saw hinted so |
| 16:14.37 | austinnichols101 | damn: cisco 8.2 firmware only allows one call at a time on the first line |
| 16:14.51 | blitzrage | anyone had these errors before? |
| 16:14.51 | blitzrage | HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 16:14.52 | blitzrage | HDLC Abort (6) on Primary D-channel of span 1 |
| 16:14.52 | malverian[work] | It just let me download the firmware from Cisco.com... POS3-08-2-00.zip ? |
| 16:14.54 | blitzrage | HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 16:14.54 | blitzrage | HDLC Abort (6) on Primary D-channel of span 1 |
| 16:15.02 | austinnichols101 | yup |
| 16:15.04 | blitzrage | oops - sorry for the dupe :) |
| 16:15.11 | austinnichols101 | but 08-2 has issues |
| 16:15.29 | doolph | (Netsplit Detector) Netsplit between clarke.freenode.net and irc.freenode.net - Invincible |
| 16:15.37 | blitzrage | I'm on 7.5 because it works ;) |
| 16:15.42 | malverian[work] | I guess the free account I set up lets me download firmware... interesting. |
| 16:15.46 | austinnichols101 | yes - trying to downgrade now |
| 16:16.00 | malverian[work] | blitzrage, You don't have the problems with dropped registration that are mentioned on voip-info ? |
| 16:16.10 | blitzrage | malverian[work]: nope |
| 16:16.12 | austinnichols101 | I've never had problem w/7.5 |
| 16:16.15 | malverian[work] | Cool. |
| 16:16.29 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
| 16:16.34 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 16:16.46 | djMax | anybody had their customized key settings break when upgrading firmware on a polycom (ip500) |
| 16:16.53 | austinnichols101 | but that doesn't necessarily mean that there aren't problems :) |
| 16:18.01 | *** join/#asterisk coppice (n=chatzill@116.196.17.210.dyn.pacific.net.hk) |
| 16:18.44 | malverian[work] | Is it very difficult to revert the phone to SCCP after installing the SIP firmware? |
| 16:19.02 | djMax | I can press the fourth button on the poly and crash it. Sweet! |
| 16:19.11 | mutilator | just as difficult as making it sip |
| 16:19.28 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:19.44 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 16:20.12 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:20.36 | *** join/#asterisk Bambr (n=Bambr@213-35-232-62-dsl.end.estpak.ee) |
| 16:21.12 | austinnichols101 | supposedly it's just a reflash |
| 16:21.29 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) [NETSPLIT VICTIM] |
| 16:21.29 | *** join/#asterisk flynux (i=v8hy3c1@cl-8.bru-01.be.sixxs.net) [NETSPLIT VICTIM] |
| 16:21.45 | jsharp | Some of the 7940s I've done have been easy. Others have been cast-iron bitches. |
| 16:22.01 | austinnichols101 | damn. can't figure out how to downgrade from 8.2 |
| 16:23.00 | malverian[work] | The Cisco sw-center is a PAIN to navigate... |
| 16:26.31 | chrismog | Is there a good guide on how to work with the digital answering assitant? |
| 16:26.34 | *** join/#asterisk jinxed (n=jf69@CPE0000c0c781ef-CM00111ae6a016.cpe.net.cable.rogers.com) |
| 16:27.09 | djMax | anybody know where to get a list of the "y" values for Polycom keys? |
| 16:27.29 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 16:27.45 | *** join/#asterisk DrRotmos (n=magnus@85.8.2.169.se.wasadata.net) |
| 16:27.56 | a1fa | how many ulaw sip peers can i run stim. on Ultra10 w/ 2x512mb and 440mhz sparc cpu? |
| 16:28.51 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 16:28.53 | jbalcomb | a1fa lots |
| 16:29.17 | jinxed | is it possible to set up asterisk with multiple outbound sip lines and have it pick an available line for outbound calls? |
| 16:29.18 | a1fa | 50? |
| 16:29.22 | a1fa | 100? |
| 16:29.25 | a1fa | 1000? |
| 16:29.30 | Hmmhesays | is anyone using regex in their dp? |
| 16:29.33 | a1fa | i am thinking about 25 |
| 16:29.38 | tzanger | jinxed: yes why not? |
| 16:29.38 | a1fa | tell me if i am wrong |
| 16:29.44 | Hmmhesays | i'm having a hell of a time getting it to work |
| 16:29.48 | jsharp | 25-50, at least. |
| 16:29.51 | DrRotmos | hi, is there any way to execute two things parallel in my dialplan? |
| 16:29.52 | tzanger | a1fa: do your own benchmarking, jeez |
| 16:29.54 | jinxed | i can't find any info on how to set it up |
| 16:30.04 | tzanger | jinxed: you can do it a couple of ways |
| 16:30.26 | tzanger | first way, Dial(SIP/${EXTEN}@peer1) and use gotoif if they return back a "go away" response |
| 16:30.36 | tzanger | the gotoif dials the SIP/${EXTEN}@peer2 and so on |
| 16:30.53 | a1fa | jsharp : too cool |
| 16:31.10 | jbalcomb | a1fa from what i've read and experienced i don't 100+ would give you trouble. It does depend on what else you're doing though. |
| 16:31.15 | mishehu | where can I find a list of all items that make up CALLERID that can be set in the dialplan? |
| 16:31.16 | a1fa | i just got an ultra10 |
| 16:31.17 | tzanger | second way is to do a little DB magic and look up which peer to use, then after the call, update the DB to point to the next one. this balances the call volume out over the peers |
| 16:31.34 | jsharp | alfa: If you get into transcoding, that number will drop since there's no Sparc optimized codecs. |
| 16:31.56 | a1fa | gsm |
| 16:31.58 | jinxed | ok, the second one is what i was looking for, thanks |
| 16:32.01 | a1fa | so no gsm, i guess? |
| 16:32.13 | a1fa | it sucks that you can run more ulaws than gsms |
| 16:32.15 | tzanger | jinxed: after that you start running combinations of that, especially wehre you start looking at call completion stats and start load balancing better based on actual minutes used or call quality or whatever... your "selection" just becomes more complex |
| 16:32.16 | jbalcomb | a1fa only one way to say for sure... |
| 16:32.19 | malverian[work] | Hmmm? ftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/ |
| 16:32.37 | austinnichols101 | what's up a1fa? |
| 16:33.05 | malverian[work] | Is that normal for the firmware image to be in the ftp pub? |
| 16:33.07 | jsharp | alfa: You can run gsm as long as you run gsm in & gsm out. |
| 16:33.09 | a1fa | austinnichols101 : nada.. i am thinking of doing in-house pbx |
| 16:33.18 | a1fa | but i dont have enough of bandwidth in-house |
| 16:33.27 | a1fa | jsharp : ulaw+gsm |
| 16:33.36 | a1fa | gsm in - ulaw out :p |
| 16:33.55 | *** join/#asterisk jets (i=jetsnoc@216.83.66.202) |
| 16:33.55 | a1fa | and vice versa |
| 16:34.19 | jbalcomb | a1fa what do you consider not enough bandwidth to run ulaw? |
| 16:34.28 | jsharp | Now I'm curious. I should light up * on my Ultra 30 and see how it behaves. |
| 16:34.42 | a1fa | jbalcomb : i have about 4mbit download, but 512k upload |
| 16:34.51 | a1fa | jbalcomb : i also have a 100mbit up/down |
| 16:35.01 | a1fa | connection.. but it is in the datacenter |
| 16:35.07 | a1fa | i am using that box for my pbx |
| 16:35.15 | a1fa | i wanted to bring it in da house :P |
| 16:36.04 | jbalcomb | all this Sparc talk makes me think yall might be interested in my 2, 5, or Server 20? |
| 16:36.22 | *** join/#asterisk snip3r (n=sniper@195.246.199.136) |
| 16:36.25 | a1fa | jbalcomb : i got my Ultra10 for free |
| 16:36.33 | a1fa | what you got to offer |
| 16:36.49 | a1fa | hard to beat that :P |
| 16:37.03 | jbalcomb | a1fa i have a big monitor, drives, cd server, and some pill boxes. |
| 16:37.07 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:37.16 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 16:37.33 | jbalcomb | a1fa everything cheap, cheap. trying to sell all my worldly possesion before i leave for Japan |
| 16:38.07 | a1fa | msg me your pricing :P |
| 16:38.11 | a1fa | shipping is what kills you |
| 16:38.20 | a1fa | i dont need monitors |
| 16:38.25 | a1fa | serial baby :P |
| 16:39.06 | *** join/#asterisk amdtech (n=stdamd11@ab1-1-246.shsu.edu) |
| 16:39.14 | jsharp | hooray for serial console.s |
| 16:39.18 | a1fa | yup |
| 16:39.32 | a1fa | how about a terminal server |
| 16:39.39 | a1fa | and manage that via tcp/ip |
| 16:39.50 | a1fa | ssh into them via serial console |
| 16:39.53 | a1fa | hehehe |
| 16:39.57 | a1fa | #@#$>?#@?>$?#@ |
| 16:41.11 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 16:41.23 | fu3 | WHO SOLD YOU THE DRUGS?!@!?!?!?!?!?!?!@!)(*@&!(*@^ |
| 16:42.06 | Darwin35 | ?me looks around drugs where |
| 16:42.11 | Darwin35 | what type |
| 16:43.09 | Darwin35 | ok I need input on ho to make a 2 part call show up correctly in a cdr |
| 16:43.14 | ms345 | anyone know if you can pass ISDN cause codes from one PRI to another? say I have isdn_router----*-----telco and the telco gives me an ISDN CC of 31. Can I send that CC 31 to my isdn_router or do I have to give my router technicians access to * to get the cause codes? Currently a cause code of 16 (normal disconnect) is seen by the isdn_router no matter what the telco sends. |
| 16:43.20 | *** join/#asterisk hardwire (n=nspencer@209.112.194.39) |
| 16:43.49 | Darwin35 | part 1 cll comes in to my box and then part 2 it gets sent back oout to pstn to a cell phone overseas |
| 16:44.22 | Darwin35 | right now the cdr is showing the first part of the call but not the 2nd part |
| 16:45.06 | Darwin35 | and fork_cdr is not working |
| 16:45.43 | twisted[asteria] | so what does it show the call doing? sitting there with it's thumb up it's arse? |
| 16:46.07 | *** join/#asterisk Hmm-work (i=Blorp@66.173.103.100) |
| 16:46.14 | Hmm-work | anyone else using regexp in their dp? |
| 16:46.17 | Darwin35 | it shows the call part 1 and then just hangs there till the call ends |
| 16:46.34 | twisted[asteria] | well, according to you, part 1 is just coming in to your box |
| 16:46.35 | Darwin35 | its not showing the 2nd part of the call |
| 16:46.48 | Hmm-work | well function regex I should say, I can't get it to work right |
| 16:46.56 | twisted[asteria] | it has to show it doing something... some application or something |
| 16:47.06 | *** join/#asterisk mko-025 (n=korpim@p5498B353.dip0.t-ipconnect.de) |
| 16:47.10 | Darwin35 | it comes in and gets callforwareded out to a cell in germany |
| 16:47.39 | twisted[asteria] | via SIP? |
| 16:47.45 | Darwin35 | no iax |
| 16:48.02 | twisted[asteria] | is the call being answered on your box or just 'passing through'? |
| 16:48.13 | Darwin35 | just passed threw |
| 16:48.36 | twisted[asteria] | back out via iax too? |
| 16:48.42 | Darwin35 | twisted are you saying we have to andswer it then forward it ? |
| 16:48.56 | Darwin35 | to my friends cellphone in germany |
| 16:49.19 | twisted[asteria] | how is it getting to the cell phone? another iax connection to a termination gateway or directly to a zap card? |
| 16:49.21 | russellb | "iax too" ... hehehe |
| 16:49.28 | twisted[asteria] | russellb, :P |
| 16:49.48 | Darwin35 | it goes back out via iax to my provider then to pstn |
| 16:50.02 | twisted[asteria] | ah. have you checked to see if iax is handing off the call in a native transfer? |
| 16:51.08 | Darwin35 | not that I know of |
| 16:51.18 | backblue | anyone know, why all sip calls the "." it's erased? |
| 16:51.22 | Darwin35 | I will call the provide r here in a min and check |
| 16:51.30 | backblue | why we cant use "." in sip urls? |
| 16:52.13 | *** join/#asterisk saftsack (n=oliver@p54A7D02C.dip.t-dialin.net) |
| 16:54.12 | Darwin35 | ok no its not doing native transfer |
| 16:54.31 | Darwin35 | man this is getting me a headack |
| 16:54.42 | *** join/#asterisk Micetto (n=bks@adsl-209-66.38-151.net24.it) |
| 16:57.21 | backblue | there is someway to specify the fromdomain in a iax2 trunk? |
| 16:57.27 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 16:57.36 | Peaceful | Can you use macros in iax.conf? |
| 16:58.19 | *** join/#asterisk TedC (n=ted@gray.impulse.net) |
| 17:03.30 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 17:04.07 | websae | does any one use a asterisk origination/termination carrier for business phone lines...i am wondering how your up time is as well as quality |
| 17:05.22 | Hmm-work | ahh i see function regex is broken on the bug tracker |
| 17:05.51 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 17:09.50 | websae | file: how are you doing? |
| 17:09.58 | file[laptop] | marvelous |
| 17:10.13 | websae | I have a question for you....do you know any businesses that use asterisk termination/originiation? |
| 17:10.42 | websae | have a SIP trunk for their phone lines instead of a PRI and what not |
| 17:11.42 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 17:12.53 | justinu | i had considered going that route for some of my customers... but it's too unreliable at this stage |
| 17:13.07 | websae | yeah iwas talking to you yesterday |
| 17:13.12 | websae | it's more cost efficient though |
| 17:13.22 | jsharp | If it traverses the public internet, then you run the risk of it tanking on you at any time. |
| 17:13.35 | file[laptop] | yes, the public internet is out of your control |
| 17:13.39 | websae | i wonder how vonage does it |
| 17:13.40 | websae | lol |
| 17:13.42 | justinu | you can either have cheap, or reliable |
| 17:13.42 | websae | yeah that is true |
| 17:13.49 | amdtech | we don't use a provider, but we did set up a box an hour and a half away to get free long distance to the city |
| 17:13.49 | justinu | vonage is for residential, mostly |
| 17:14.12 | justinu | websae, you can get it to work |
| 17:14.14 | websae | amdtech: how's that working out for you |
| 17:14.20 | justinu | but you need to make sure your ITSP has good routes to your ISP |
| 17:14.28 | justinu | and that your ISP is dependable |
| 17:14.31 | amdtech | works like a charm |
| 17:14.38 | websae | ISP with an SLA |
| 17:14.43 | amdtech | calls sound great, i think we had a hiccup or two when we first got it |
| 17:14.50 | justinu | there will be hiccups |
| 17:14.52 | amdtech | but it's stable and unless the net goes out, we're good to go |
| 17:15.57 | Peaceful | websae: We ran SIP over two private (data) T1's to an asterisk server at an ISP that had tied into the PSTN. Unfortunately, the ISP was very unreliable. We ended up getting away from them, buying some digium cards for our asterisk server and improving our reliability a ton. |
| 17:16.29 | amdtech | it really just depends on the termination location and your internet's stability |
| 17:16.50 | amdtech | we've got a 40 meg connection, and i'm pushing for them to reserve our next upgrade bandwidth strictly for sip |
| 17:16.56 | justinu | I'm using a high end ITSP over a nice internet connection |
| 17:16.58 | justinu | and it's flawless |
| 17:17.00 | *** join/#asterisk fulgas (n=fulgas@207.226.175.2) |
| 17:17.09 | justinu | but it was carefully chosen/designed |
| 17:17.31 | justinu | this is for another customer |
| 17:17.37 | justinu | not a PBX customer |
| 17:18.09 | Hmmhesays | function regex is broken and i'm not sure how to fix it |
| 17:18.15 | *** join/#asterisk Axel69 (n=alexlsf@200.62.38.91) |
| 17:21.36 | chrismog | How can I make asterisk put an area code infront of a 7 digit number? Say if the user dials "456-7890" asterisk will send "123-456-7890" to my iax2 trunk? |
| 17:21.55 | justinu | 123${EXTEN} |
| 17:22.01 | chrismog | ah, ok |
| 17:22.25 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F32E0.dip0.t-ipconnect.de) |
| 17:22.54 | doolph | why when I call to my asterisk I don't hear any ringback? |
| 17:23.08 | justinu | lots of reasons |
| 17:23.15 | justinu | explain your configuration more |
| 17:23.38 | doolph | Ok I have an DID from a company (356-0001) |
| 17:23.45 | justinu | sip? |
| 17:23.48 | doolph | Yes |
| 17:24.03 | doolph | when I call to that number from another number |
| 17:24.11 | doolph | I don't hear nothing |
| 17:24.16 | justinu | ok, check to see that your ast box is sending 180 ringing in response to the invite |
| 17:24.19 | doolph | but it's ringing in 356-0001 |
| 17:24.36 | doolph | how |
| 17:24.48 | justinu | use sip debug from the console |
| 17:24.58 | Hmmhesays | is there any better functions to check if a string starts with a certain set of digits? |
| 17:25.11 | doolph | Ok |
| 17:25.22 | Hmmhesays | regex is broken |
| 17:26.07 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:26.18 | doolph | justinu 180 ringing... where's that I am in a debug |
| 17:26.33 | doolph | asterisk1*CLI> |
| 17:26.34 | doolph | <-- SIP read from 201.227.72.230:62249: |
| 17:26.34 | doolph | SIP/2.0 180 Ringing |
| 17:26.35 | asterboy | justinu, you have the Polycom phone right? |
| 17:26.38 | asterboy | iirc |
| 17:27.01 | justinu | i have a few |
| 17:27.05 | doolph | but that ringing is to my exten |
| 17:27.08 | doolph | right? |
| 17:27.10 | justinu | doolph: that looks like they're sending you a 180 ringing |
| 17:27.12 | doolph | how to the caller |
| 17:27.16 | justinu | we want to see a call in the OTHER direction. |
| 17:27.34 | asterboy | justinu, have you setup conference calls and call transfer? |
| 17:27.39 | justinu | transfer yes |
| 17:27.48 | justinu | and i've done conferencing also |
| 17:28.19 | doolph | the caller is not having ringbacks :( |
| 17:28.27 | justinu | i understand that |
| 17:28.42 | asterboy | excellent, I'm stumbling on setting up those features...from what I have read, * can do it OR the Polycom phone can do it internally. |
| 17:28.45 | justinu | you need to place a call to your DID while you have sip debug turned on |
| 17:28.52 | justinu | sip debug peer <name of peer> helps |
| 17:29.05 | justinu | asterboy: that's right |
| 17:29.11 | justinu | for conferencing at least |
| 17:29.27 | asterboy | I have ZAP channels, so which is the least complicated to setup? |
| 17:29.39 | asterboy | best maybe to get the phones to do it? |
| 17:30.01 | chrismog | So, what if I want people to just be able to dial their local # (456-7890) but also be able to dial another area code too |
| 17:30.08 | justinu | unless you need more than 3 people, just let the phone do it |
| 17:30.09 | chrismog | I think my rules are messed up |
| 17:30.11 | doolph | justinu what should I check? |
| 17:30.26 | justinu | doolph: check to see you respond to the incoming invite with a 180 ringing |
| 17:31.00 | asterboy | Do you know of any examples or specific docs I can read before I ask hand holding questions? |
| 17:31.16 | chrismog | ^ yeah I need some of those too on how to setup extensions :/ |
| 17:31.18 | justinu | you could read the polycom admin and users manual |
| 17:31.30 | justinu | or maybe the polycom pages on the wiki |
| 17:31.36 | asterboy | They don't give very much info on the specifics of the setup. |
| 17:31.49 | justinu | there's no setup for transfer or conferencing |
| 17:31.52 | justinu | you just do it |
| 17:32.01 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:32.05 | justinu | however, i've noticed a bug with attended transfer in asterisk/polycom |
| 17:32.13 | asterboy | ah, ok...then I'm doing something wrong. |
| 17:32.34 | doolph | justinu can you go to paste bin and check it any error ? http://pastebin.ca/46505 |
| 17:32.34 | justinu | if you do an attended transfer, and the person you're transfering to doesn't answer, but you proceed with the transfer... the original caller won't hear any call progress tones during the transfer |
| 17:32.44 | justinu | just silence, until they pick up, or their voicemail picks up |
| 17:32.56 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
| 17:33.17 | justinu | doolph: looks like you're sending the 180 ringing |
| 17:33.25 | justinu | doolph: i would complain to your ITSP, and send them that pastebein |
| 17:33.36 | justinu | it's not your fault |
| 17:34.03 | doolph | Ok |
| 17:35.15 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 17:35.35 | asterboy | justinu, here is what I get when I try to do the transfer: Mar 21 10:34:27 NOTICE[23629]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 17:36.12 | justinu | ok, that's a whole different problem |
| 17:36.21 | asterboy | After establishing a call with my other party, I hit "transfer" and the new 7 digit telephone # |
| 17:36.22 | justinu | has nothing to do with your polycom setup |
| 17:36.34 | justinu | something is wrong in the dialplan, or with your * config |
| 17:36.41 | *** join/#asterisk Alric (n=nbowyer@avantacom.com) |
| 17:37.25 | doolph | mmm |
| 17:37.37 | doolph | justinu how can my istp can fix it ? |
| 17:37.49 | justinu | by playing the ringback tone to the PSTN call |
| 17:37.55 | justinu | like they're supposed to |
| 17:38.20 | doolph | I have tested connecting directly |
| 17:38.24 | doolph | with a ata 186 |
| 17:38.31 | doolph | it works with a fix |
| 17:38.38 | doolph | in connectmode |
| 17:38.42 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:39.01 | justinu | i dunno what that means, sorry |
| 17:40.11 | Axel69 | hi guys..... how do i use a extencion as a trunk |
| 17:41.05 | Axel69 | i mean a quintum to be my gw with analog lines but in a private ip so it has to register to the asterisk as a extension |
| 17:46.47 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 17:50.24 | *** join/#asterisk W8TAH (n=Tim@static-acs-24-239-210-31.zoominternet.net) |
| 17:50.46 | W8TAH | Does this room also answer questions about asterisk@home? |
| 17:50.52 | a1fa | no |
| 17:50.58 | a1fa | fuck AAH |
| 17:51.01 | a1fa | :P |
| 17:51.06 | W8TAH | sorry for the noise |
| 17:51.11 | *** part/#asterisk W8TAH (n=Tim@static-acs-24-239-210-31.zoominternet.net) |
| 17:51.12 | a1fa | j/j |
| 17:51.15 | a1fa | i was joking |
| 17:51.17 | a1fa | damn dude |
| 17:51.17 | a1fa | :P |
| 17:52.30 | *** join/#asterisk W8TAH (n=Tim@static-acs-24-239-210-31.zoominternet.net) |
| 17:53.26 | a1fa | just ask |
| 17:53.29 | Nivex | W8TAH de N8VNR: I think there is a different room for that. Not sure the name offhand. |
| 17:53.36 | a1fa | their official channel is #AAH |
| 17:53.40 | W8TAH | oh -- ok -- sorry guys |
| 17:53.44 | *** part/#asterisk W8TAH (n=Tim@static-acs-24-239-210-31.zoominternet.net) |
| 17:53.45 | a1fa | and #FreePBX |
| 17:53.47 | a1fa | damn it |
| 17:54.09 | Nivex | some people haven't mastered the fine art of lurking I guess :) |
| 17:55.23 | a1fa | yeya |
| 17:55.23 | a1fa | :p |
| 17:55.25 | russellb | lurking rocks |
| 17:55.26 | a1fa | lurking? |
| 17:55.36 | a1fa | how about peeping tom? |
| 17:55.58 | Qwell[] | I stalk |
| 17:56.13 | a1fa | true |
| 17:56.15 | a1fa | :P |
| 17:56.37 | a1fa | if you "stall" does that mean you stall while taking a piss? |
| 17:57.03 | *** join/#asterisk W8TAH (n=Tim@static-acs-24-239-210-31.zoominternet.net) |
| 17:57.24 | a1fa | its time for gyros |
| 17:57.33 | a1fa | anybody join me for gyros for lunch? |
| 17:57.36 | djMax | anybody having working key remappings on Polycoms? |
| 17:58.02 | justinu | i'll join |
| 17:58.12 | W8TAH | I need some help configuring for calls inside my network only -- i want to use this as an intercom system for our school |
| 18:02.39 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
| 18:03.11 | W8TAH | what protocol should i be choosing for gizmo softphones? |
| 18:03.18 | Qwell[] | sip |
| 18:03.23 | W8TAH | thank you |
| 18:03.33 | dahunter3 | How can I increase the volume of a recorded digital receptionist? |
| 18:03.36 | *** join/#asterisk ryback (n=shigueta@69.65.149.190) |
| 18:04.42 | W8TAH | I have had several recomendations on the Gizmo softphones -- is that the best one to use, or is there a better (free / opensource) one? |
| 18:04.47 | W8TAH | this will primarially be for windows |
| 18:05.08 | ryback | hi, i´m a newbie on asterisk, i just installed asterisk@home on a PC with 3 FXO cards; i´d like to configure it to use 3 POTS lines to dial/receive calls and then configure extensions to transfer calls... how should i do that? |
| 18:06.12 | a1fa | ryback : load zaptel driver |
| 18:06.24 | a1fa | add Zap Cards to your zap.conf |
| 18:06.31 | a1fa | add extensions to extensions.conf |
| 18:06.43 | a1fa | some1 correct me if i am missing anything |
| 18:06.48 | a1fa | W8TAH : SIP? |
| 18:07.11 | backblue | ryback: pay me, and i will do that. |
| 18:07.24 | W8TAH | a1fa, i have a new a@h server -- so its an open book right now |
| 18:07.31 | W8TAH | i want whatever will work best |
| 18:07.53 | W8TAH | as i read the docs, there are 4 choices |
| 18:08.38 | W8TAH | Qwell, told me that sip was appropriat for the gizmo softphones, and im installing one now to begin testing with |
| 18:09.55 | a1fa | why dont you get real phones for it> |
| 18:10.06 | a1fa | backblue : how much do you charge :P |
| 18:10.09 | W8TAH | for now this project has to be zero budget |
| 18:10.18 | a1fa | igor? russian? |
| 18:10.19 | asterboy | "The old musiconhold.conf syntax has been deprecated!" |
| 18:10.26 | asterboy | where are the new samples? |
| 18:10.27 | W8TAH | it is designed to be an intercom system between our classrooms and to office |
| 18:10.31 | a1fa | asterboy : make samples |
| 18:10.39 | a1fa | W8TAH : cool.. |
| 18:10.58 | W8TAH | i have a grant for the microphones etc |
| 18:10.59 | asterboy | thought I did that, but didn't get a musiconhold.sample |
| 18:11.07 | asterboy | not sure why |
| 18:11.19 | a1fa | :( |
| 18:11.21 | W8TAH | but, i can get all the mics for the whole building for what about 3 phones will cost -- |
| 18:11.23 | asterboy | try again |
| 18:11.29 | a1fa | backblue |
| 18:11.30 | a1fa | yo |
| 18:11.35 | backblue | a1fa: no |
| 18:11.40 | backblue | portuguese |
| 18:11.42 | a1fa | backblue : ah |
| 18:11.51 | a1fa | backblue : how much do you charge /h? |
| 18:12.03 | backblue | i dont, i was kidding |
| 18:12.30 | backblue | i work for a company |
| 18:12.35 | a1fa | kewl |
| 18:12.36 | a1fa | :) |
| 18:12.37 | dahunter3 | Does anyone know how to increase the output volume of a recorded digital receptionist so that it sounds louder to callers/ |
| 18:12.51 | a1fa | dahunter3 : incrase the sample |
| 18:13.03 | a1fa | dahunter3 : playback()? |
| 18:13.09 | a1fa | or background() |
| 18:13.49 | asterboy | doh, should have backed up before make samples. |
| 18:14.01 | asterboy | now I have to rename old files back. |
| 18:14.02 | dahunter3 | a1fa: So, I can just pass an extra argument to background that controls the level? If so, that's great, then I just need to fight with asterisk@home to make the change. |
| 18:16.37 | dahunter3 | a1fa: I don't see any extra argument to background. How do I increase the sample? |
| 18:17.19 | *** join/#asterisk stoffell (n=stoffell@d5153FC33.access.telenet.be) |
| 18:17.41 | djMax | any guesses on what "log to standard output" means in the context of a polycom phone? |
| 18:17.56 | W8TAH | is there documentation someplace for using the gizmo phones with asterisk / a@h or is there a better softphone to use? its wanting me to create an account on their network |
| 18:18.05 | jets | i'm writing an ivr in an app because i dont want to do it in the dial plan and AGI won't be much of a possibility... what an odd morning. |
| 18:22.19 | jbalcomb | djMax STDOUT? perhaps something to do with syslog? |
| 18:23.00 | asterboy | wouls be better if make samples put sip.sample instead of replaceing sip.conf with sip.conf.old |
| 18:23.14 | djMax | hmmm, perhaps. Thanks, I'll check that. |
| 18:24.04 | justinu | dahunter3: is it just the prompts that are too low? |
| 18:24.12 | dahunter3 | justinu: Yes. |
| 18:24.48 | dahunter3 | justinu: Normal conversations are fine, so I didn't want to mess with rx/txgains... |
| 18:25.03 | justinu | i don't think you can boost the volume of the prompts without modifying the files w/ a wave editor |
| 18:25.51 | dahunter3 | justinu: Okay, well I guess I'll start doing that: thanks for the heads up. |
| 18:25.54 | *** join/#asterisk Eggplant (i=No@dsl-836.cascadeaccess.com) |
| 18:25.55 | justinu | np |
| 18:27.33 | tecnico | in the console, how can I re-register to a server ? iax2 show registry shows no error, but the other server still doesn't know where I am. |
| 18:30.41 | asterboy | or better yet, a samples directory under /etc/asterisk...then a guy can cherry pick and keep his setup clean |
| 18:31.30 | Peaceful | What's the standard way of dialing multiple extensions at once (aka a ring group) in asterisk 1.2? The entry for Dial() in the O'Reilly book doesn't address that... |
| 18:33.43 | astra^^ | wher is the log of * stored |
| 18:34.01 | backblue | astra^^: in the logs directory. |
| 18:34.04 | Peaceful | astra^^: /var/log/asterisk |
| 18:34.17 | astra^^ | ok |
| 18:35.14 | *** join/#asterisk JulianoSMM (n=oz@201.14.218.39) |
| 18:36.58 | Peaceful | ah, there it is. Chapter 5: Dialplan Basics. Use '&' to join multiple extensions. |
| 18:38.48 | Darwin35 | ok found ou t we should be using the dial(local/ for forking a call and the cdr recording it all |
| 18:40.08 | astra^^ | y is tat wen i put 2 prefix my call doesnt go tru . |
| 18:41.16 | Darwin35 | dod you have your dial plan setup for it |
| 18:41.21 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 18:42.07 | astra^^ | it is |
| 18:42.09 | astra^^ | exten => _00.,3,SetCallerID(9000000000) ;(4996806524) |
| 18:42.10 | astra^^ | exten => _00.,4,Dial(SIP/${EXTEN:1}@mypbx1) |
| 18:42.41 | tzanger | astra^^: you need to get a basic grasp on what it is you're doing |
| 18:42.49 | tzanger | what do you think the :1 on ${EXTEN:1} does? |
| 18:43.27 | astra^^ | ohhh... holly shit.. You are right ... |
| 18:43.34 | astra^^ | what a fool am i |
| 18:44.32 | tzanger | astra^^: not a fool, just need help seeing things |
| 18:44.44 | astra^^ | thanx dude.. |
| 18:44.54 | astra^^ | you people rock.. |
| 18:46.33 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 18:47.47 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 18:47.54 | r_evolution | hey hey the gangs all here. |
| 18:48.24 | justinu | werd |
| 18:48.26 | asterboy | justinu, I should be able to call my ZAP channel, pick up on my Polycom SIP extensions and then transfer or conference the call to another SIP or ZAP channel right? |
| 18:48.33 | justinu | yep |
| 18:48.34 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 18:48.50 | brodiem | does anyone know of IP phones that have amplified headset jacks so that external amps aren't needed for headsets? |
| 18:48.52 | r_evolution | so justin... i got the 7 digit dialing worked out... and speed dial... |
| 18:48.53 | asterboy | but I have something screwy in my config preventing that. |
| 18:48.56 | Nodren | i'm having a problem with asterisk on ubuntu, can anyone help? |
| 18:48.57 | justinu | whether it goes to SIP or ZAP is determined by the number you transfer to /dialplan |
| 18:49.04 | justinu | r_evolution: noyce |
| 18:49.11 | r_evolution | so i guess it's time to see what else i can do |
| 18:49.14 | justinu | hehe |
| 18:49.15 | Nodren | i compiled and installed zaptel and when i try to modprobe it says module not found |
| 18:49.24 | r_evolution | im thinking the ATAs we've sent out to our customers *MIGHT* be smart enough to differentiate between a fax call |
| 18:49.28 | r_evolution | and a voice call |
| 18:49.31 | justinu | which ATAs? |
| 18:49.34 | asterboy | On Polycom I hit "Transfer" it puts the call on hold, gives me tone and then I dial a SIP or ZAP #. |
| 18:49.37 | justinu | sipura 2100 supports t.38 |
| 18:49.41 | brodiem | Nodren, did you 'make install'? |
| 18:49.43 | r_evolution | ahh but level 3 does not |
| 18:49.48 | Nodren | yes |
| 18:49.48 | r_evolution | not from what the guy told me |
| 18:49.50 | justinu | nope, but I've run fax over L3 |
| 18:49.52 | justinu | g711 |
| 18:49.53 | justinu | works |
| 18:49.57 | r_evolution | yeah that's what im getting at... |
| 18:50.05 | r_evolution | i'm saying... if we make G729 the primary... |
| 18:50.07 | r_evolution | and allow G711 |
| 18:50.13 | Nodren | make install worked with out errors |
| 18:50.18 | r_evolution | b/c i've noticed when it faxes... it bitches about no compatible codecs |
| 18:50.22 | r_evolution | ya herd. |
| 18:50.35 | brodiem | Nodren, see if the module exists at /lib/modules/<kernelver>/misc |
| 18:50.42 | r_evolution | some woman called this morning and said her ENTIRE business rested on her ability to fax. |
| 18:50.47 | justinu | picking codecs based on whether it's a fax call or not is going to be tricky |
| 18:51.00 | *** join/#asterisk cji (i=3000@66.80.146.7) |
| 18:51.08 | justinu | SIP can change codecs in mid call, but good luck making that work |
| 18:51.38 | Nodren | aha, brodiem i think i found my problem, the modules went to the wrong directory |
| 18:51.46 | Nodren | but they compiled under the right source |
| 18:52.10 | MikeJ[Laptop] | so terminate the fax locally to pstn then |
| 18:52.17 | r_evolution | easier than fitting 350 people on a T1 |
| 18:52.21 | MikeJ[Laptop] | problem solved |
| 18:52.28 | justinu | that's about all you could do, at this point in time |
| 18:52.34 | justinu | but fax is still gonna be iffy |
| 18:52.42 | W8TAH | Thanks Folks for all the help -- my system is now working |
| 18:52.45 | justinu | because the link from you to the customer is out of your control |
| 18:52.51 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 18:52.55 | W8TAH | im moving to a limited deployment |
| 18:52.57 | brodiem | is there no way to make sipura ata's use anything other than g711 for fax? |
| 18:52.58 | _Paulo_ | I'm connecting to a cisco ata 186, no nat, and got this: SIP/2000-bc11 is circuit-busy |
| 18:53.15 | _Paulo_ | I new to Sip |
| 18:53.20 | _Paulo_ | any hints? |
| 18:53.26 | r_evolution | eh... faxing is iffy to begin with |
| 18:53.30 | r_evolution | at least when it comes to SIP |
| 18:53.39 | r_evolution | and *something* if it IS poor quality |
| 18:53.45 | r_evolution | will allow me to delay until the end of the year |
| 18:53.53 | r_evolution | which is when L3 is supposed to start using T.38 |
| 18:54.24 | r_evolution | hey paulo... is the box plugged in and registering? |
| 18:54.32 | Nodren | brodiem, i copied the modules into the right folder for my kernel version, but its still saying not found |
| 18:54.50 | r_evolution | cant terminate the fax call to pstn for 350 different people faxing to and from different locations :) |
| 18:54.53 | brodiem | Nodren, try insmod /path/to/module |
| 18:54.58 | _Paulo_ | r_evolution, yes, its pluged. |
| 18:54.58 | Nodren | ok |
| 18:55.04 | r_evolution | is it registering |
| 18:55.12 | _Paulo_ | r_evolution, but I saw no register message |
| 18:55.13 | r_evolution | can you go type sip show peer 2000 |
| 18:55.15 | brodiem | Nodren, also, if the modules were found in a directory for a kernel version you aren't using, then it's likely that the modules were compiled against the wrong kernel source |
| 18:55.32 | r_evolution | then it's not registering... which would probably be why there's no route |
| 18:56.23 | brodiem | Nodren, make sure /lib/modules/<kernelver>/build points to the correct source |
| 18:56.26 | *** join/#asterisk robb_ (i=robb@spoon.netsoc.tcd.ie) |
| 18:56.28 | _Paulo_ | r_evolution, http://pastebin.ca/46521 |
| 18:56.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:57.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:57.15 | r_evolution | so it has registered at some point |
| 18:57.26 | r_evolution | put qualify=yes under the definition |
| 18:57.35 | _Paulo_ | r_evolution, I'm using * for a fax service. |
| 18:57.36 | r_evolution | so it'll keep a track on it... instead of UNMONITERED |
| 18:57.45 | r_evolution | make sure you stick to G711 then |
| 18:57.45 | asterboy | when recording with Zapbarge, how do you playback the file? I'm getting: File /tmp/test-in.wav does not exist in any format |
| 18:57.48 | asterboy | ?? |
| 18:58.18 | asterboy | Playback(/tmp/test-in.wav) |
| 18:58.37 | Nodren | ok heres a serious question, why does every linux distro and every computer i attempt to install the zaptel drivers on to communicate with my TDM400P not work |
| 18:58.40 | stoffell | asterboy: tri Pla...(/tmp/test) |
| 18:58.41 | Nodren | i thought this was a highly supported card |
| 18:58.47 | Nodren | and i cant get anywhere on any system |
| 18:59.01 | r_evolution | maybe you're not installing the drivers correctly? |
| 18:59.01 | stoffell | asterboy, i mean : (/tmp/test-in) |
| 18:59.11 | asterboy | ok, thats what I thought. |
| 18:59.13 | asterboy | thx |
| 18:59.22 | brodiem | Nodren, probably because the zaptel driver isn't loaded :) |
| 18:59.26 | r_evolution | ^ |
| 18:59.27 | *** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd6-CM014350000760.cpe.net.cable.rogers.com) |
| 18:59.31 | Nodren | i've followed the tutorials offered on asteriskguru.com and on voip-info.org and i've tried from asterisk@home |
| 18:59.34 | Nodren | nothing's worked |
| 18:59.41 | justinu | _Paulo_: that's not the real error |
| 18:59.42 | r_evolution | my point exactly |
| 18:59.43 | Nodren | and i can lsmod and see the zaptel module is in the list |
| 18:59.50 | justinu | look in the log before the circuit busy message to find the real error |
| 19:00.07 | _Paulo_ | justinu, any hint? |
| 19:00.14 | r_evolution | speaking of which it's time to delete the debug log again justin O_O |
| 19:00.17 | asterboy | yep, that worked! (dam I wish things were a little more intuative) |
| 19:00.18 | justinu | i just gave you the hint |
| 19:00.22 | brodiem | Nodren, so what happens when you load the driver for your card? |
| 19:00.39 | _Paulo_ | ok... |
| 19:00.39 | justinu | r_evolution: eventually, you won't want to be logging debug messages |
| 19:00.46 | justinu | r_evolution: just warnings, notifies, perhaps |
| 19:00.54 | Nodren | well on centos the wctdm module wouldnt load but the wcfxs did |
| 19:01.05 | r_evolution | eh... im only logging debugs while im adding new features |
| 19:01.08 | MikeJ[Laptop] | ~centos |
| 19:01.09 | jbot | centos is probably better than Fedora Core except for that silly bug, see ~centosbug for details |
| 19:01.10 | Nodren | on ubuntu the wcfxs doesnt even get compiled, but the wctdm works |
| 19:01.11 | justinu | right |
| 19:01.13 | _Paulo_ | justinu, == Everyone is busy/congested at this time (1:0/1/0) |
| 19:01.16 | Nodren | i already fixed the centos bug. |
| 19:01.19 | MikeJ[Laptop] | ~centosbug |
| 19:01.21 | jbot | well, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 19:01.21 | justinu | _Paulo_: keep going back |
| 19:01.23 | Nodren | got past that just fine |
| 19:01.30 | Nodren | it wasnt a compiling issue |
| 19:01.31 | MikeJ[Laptop] | :D |
| 19:01.33 | Nodren | it compiled ok |
| 19:01.40 | r_evolution | eventually it'll be good to go and i can go about my merry way |
| 19:01.43 | Nodren | it was just actually getting zaptel working after compiling it |
| 19:01.46 | *** join/#asterisk Lino` (n=Lino@i577BD7DA.versanet.de) |
| 19:02.31 | Nodren | i keep getting this exact same error ZT_CHANCONFIG on channel 1 : No such device or address (6) |
| 19:02.37 | Nodren | no matter what i try |
| 19:02.51 | r_evolution | hey justin... you explore that FTP anymore? |
| 19:02.57 | *** join/#asterisk GoRK (n=GoRK@209.40.175.194) |
| 19:02.59 | Nodren | i've ruled out everything but the card itself as an issue |
| 19:03.22 | brodiem | Nodren, but it loads the fxs driver? |
| 19:03.37 | Nodren | when i tried the fxs driver on centos yes it worked w/o errors |
| 19:03.43 | Nodren | well loaded |
| 19:03.49 | Nodren | but when trying ztcfg |
| 19:03.51 | Nodren | i got that same error |
| 19:03.59 | Nodren | that i'm getting now with wctdm on ubuntu |
| 19:04.23 | brodiem | and you have an fxo module on the card correct? |
| 19:04.53 | Nodren | heh even that's vague, its the TDM400P... the site we bought it from said it had FXO and FXS modules |
| 19:05.03 | justinu | you should be able to ID that visually |
| 19:05.18 | Nodren | well its got 4 ports |
| 19:05.24 | Nodren | how do i tell if they are FXS or FXO? |
| 19:05.25 | brodiem | yeah, red=fxo, green=gxs |
| 19:05.28 | justinu | look at the card itself... i think they're different colors |
| 19:05.29 | justinu | ah yes |
| 19:05.31 | brodiem | er fxs |
| 19:05.39 | GoRK | when an attended transfer is performed via the 'transfer' feature of a SIP phone is there a way to have a tone played to the party the transfer is going to? my phones are polycom 601's if it's a phone configuration thing |
| 19:05.50 | justinu | hey gork |
| 19:05.51 | asterboy | I recall a program to stitch those in and out wav files together, but can't remember the name...anyone have a hint? |
| 19:06.05 | justinu | gork: I think that's a bug in asterisk, i experience it myself with polycom |
| 19:06.16 | justinu | asterboy: there's tons of them |
| 19:06.23 | justinu | i use something old called cooledit for windws |
| 19:06.27 | Nodren | the only color i can locate is the green led next to each port |
| 19:06.37 | justinu | nodren: take the cover off your PC and look at the card |
| 19:06.38 | brodiem | Nodren, you have to look at the card itself |
| 19:06.40 | asterboy | Was hoeping for something on the linux side |
| 19:06.41 | GoRK | justinu: does it work correctly with other sip->sip transfers or star-code transfers? |
| 19:06.42 | _Paulo_ | justinu, http://pastebin.ca/46523 |
| 19:06.48 | justinu | gork: it works fine with blind transfers |
| 19:06.54 | r_evolution | there's a LOT of good stuff for putting wav files together |
| 19:06.54 | chrismog | How can I tell asterisk what my local prefix is, and have it be smart enough to add it to numbers if said number is only 7 digits long (but not add it to numbers that are 10 digits long) ? |
| 19:06.55 | justinu | not sure about * code |
| 19:06.56 | r_evolution | soundforge |
| 19:07.02 | _Paulo_ | should i turn sip debug on? |
| 19:07.07 | asterboy | but I thought there was something that came with the * distro |
| 19:07.09 | r_evolution | if you dig on cooledit... then Adobe took that over and made it Audition |
| 19:07.19 | GoRK | justinu: no i mean the transfer works fine but the callee can't tell when the transfer happens so they don't know when to say 'hello' |
| 19:07.19 | justinu | hmm |
| 19:07.40 | Nodren | the card doesnt have any distinguishing red or green colors |
| 19:07.42 | Nodren | just a blue card |
| 19:07.42 | justinu | gork: what do you mean? doesn't their phone ring? |
| 19:07.55 | justinu | nodren: look at the modules plugged into the card |
| 19:07.57 | brodiem | Nodren that makes no sense |
| 19:08.22 | Nodren | there are no modules plugged into it |
| 19:08.25 | r_evolution | yes it does Brod |
| 19:08.27 | r_evolution | he bought the card |
| 19:08.30 | r_evolution | not the modules |
| 19:08.32 | brodiem | Nodren that is your problem :) |
| 19:08.33 | Nodren | bleh |
| 19:08.34 | justinu | lol |
| 19:08.35 | r_evolution | therein lying the problem |
| 19:08.37 | justinu | lmao! |
| 19:08.42 | justinu | that sucks |
| 19:08.42 | Nodren | my freaking boss and his effort to save money |
| 19:08.43 | Nodren | ok |
| 19:08.43 | asterboy | r_evolution, could I bother your for a pastebin of your setup? I'm trying to get my TRANSFER and CONFERENCING working but am getting an error message...need a working example to digest. |
| 19:08.46 | Nodren | thanks guys. |
| 19:08.47 | Peaceful | chrismog: on the seven-digit extension pattern, just have it dial 123${EXTEN}, where 123 is your local prefix |
| 19:08.55 | justinu | rofl |
| 19:08.56 | brodiem | haha |
| 19:09.05 | r_evolution | wha? aster what are you talking about? |
| 19:09.17 | GoRK | justinu: sorry my description is a bit off -- call comes in, operator answers, presses transfer, calls the extension, extension answers and operator announces the transfer, operator presses transfer and call is transferred to the extension -- problem is there is no indication that the extension is now talking to the incoming call and the operator is not on the line any more; the caller knows because they stop hearing hold music |
| 19:09.18 | asterboy | You have a Polycom phone? |
| 19:09.21 | r_evolution | no |
| 19:09.24 | chrismog | Peaceful: i don't have a seven-digit extension pattern. I'll pastebin what I have. |
| 19:09.35 | *** part/#asterisk amdtech (n=stdamd11@ab1-1-246.shsu.edu) |
| 19:09.36 | asterboy | oh, thought you did. |
| 19:09.37 | r_evolution | I have a bunch of freaking customers right now... but you were asking for something to combine wav files |
| 19:09.39 | asterboy | nvr mind. |
| 19:09.43 | r_evolution | ouch... that's got to HURT man |
| 19:09.47 | _Paulo_ | chrismog, you will have to create one... |
| 19:09.47 | justinu | _Paulo_: we'll need a sip debug on peer 2000 to diagnose further |
| 19:10.00 | r_evolution | dude's been trying to figure out why he can't get the fxo or fxs ports to work |
| 19:10.04 | r_evolution | for probably WEEKS |
| 19:10.08 | _Paulo_ | brb |
| 19:10.10 | r_evolution | if he's used a bunch of different distros |
| 19:10.16 | asterboy | yes, I thought there was something in * to do it, but it looks like freshmeat will have lots. |
| 19:10.19 | r_evolution | and it's because he didnt have any modules on the card |
| 19:10.26 | r_evolution | ouch. |
| 19:10.28 | chrismog | http://pastebin.ca/46524 |
| 19:10.33 | justinu | gork: i see... on traditional PBXs i've used... they don't play any tone... the transfering person just says "here's the call now" |
| 19:10.34 | r_evolution | p.s. who's the salesman that let that one get away? |
| 19:10.54 | justinu | gork: sometimes the display of the phone changes to show you're now connected to a Tie or CO line |
| 19:11.19 | justinu | yeah, nodren has been thru hell and back because of an incomplete TDM2400 card! |
| 19:11.22 | justinu | lol |
| 19:11.25 | r_evolution | OUCH! |
| 19:11.26 | justinu | er TDM400 |
| 19:11.26 | GoRK | justinu: yeah if the display changed that would be ok but no idea how to make that happen.. could possibly be a use for the rpid patch if i could somehow make that indication happen |
| 19:11.31 | r_evolution | thats all im saying |
| 19:11.40 | Peaceful | chrismog: $434{EXTEN:1} should be 434${EXTEN:1} ... I believe. |
| 19:11.52 | brodiem | does anyone know of IP phones that have amplified headset jacks so that external amps aren't needed for headsets? |
| 19:12.12 | _Paulo_ | justinu, http://pastebin.ca/46525 |
| 19:12.36 | GoRK | justinu: on traditional pbx's too there's often a click or something subtle that people do not realize they rely on but with this there's not even a subtle click or background noise change |
| 19:13.08 | justinu | gork: not on my nec NEAX systems |
| 19:13.13 | justinu | they're all digital too |
| 19:13.22 | r_evolution | same for the comdial pbx im replacing here |
| 19:13.29 | justinu | _Paulo_: line 41 |
| 19:13.32 | r_evolution | someone says I have a call for you... they'll be there when i hang up |
| 19:13.33 | justinu | SIP/2.0 404 Not Found |
| 19:13.36 | r_evolution | then bam... there they are |
| 19:13.41 | justinu | that is why your "circuit busy" |
| 19:13.48 | justinu | s/your/you're |
| 19:14.00 | GoRK | well non-digital anyway; at any rate it would be a nice feature; these people are used to key systems so it may just be something they have to deal with |
| 19:14.00 | stoffell | GoRK, you have 2 options to 'solve' your problem |
| 19:14.31 | stoffell | GoRK, the thomson st2030 phone plays a tone to indicate the transfer is done. or you can use the built-in asterisk transfer function |
| 19:14.38 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 19:14.59 | CrashHD | what should I do about dtmf digits not being picked up very well with iax (711)? |
| 19:15.07 | justinu | gork: call transfering in asterisk is really fucked up |
| 19:15.13 | justinu | gotta learn how the masquarading works |
| 19:15.31 | _Paulo_ | justinu, this web based config from the cisco ata is awful. |
| 19:15.32 | GoRK | stoffell: i will see about the internal transfer function |
| 19:15.47 | justinu | _Paulo_: i have no experience with cisco ATAs, sorry |
| 19:15.53 | a1fa | justinu : use that flash operator to transfer calls :P |
| 19:16.13 | r_evolution | pound is your friend :-D |
| 19:16.17 | chrismog | Peaceful: Humm, well it is still isn't working as I would like. I want to replicate the "dial 9 for an outside line" functionality. If I dial a 9 now it will always put the areacode on the front. |
| 19:16.38 | GoRK | justinu: yeah looks like it; ill see what i can see; maybe i can shoehorn at least an extra packet in to send an extra packet with called party id when the transfer is completed; visual indication is better than nothing |
| 19:16.46 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 19:16.50 | GoRK | wow that was redudnant |
| 19:17.10 | a1fa | what was |
| 19:17.11 | r_evolution | i think you extra packet'd us to death :( |
| 19:17.12 | a1fa | :P |
| 19:17.16 | justinu | gork: call transfer w/ sip is almost redicuously complex... you might talk to oej about it |
| 19:17.33 | a1fa | iax r00000lez! |
| 19:17.40 | a1fa | what is iax btw.. hahaha |
| 19:17.41 | a1fa | :P |
| 19:17.54 | a1fa | j/k |
| 19:17.58 | a1fa | dont answer tjat |
| 19:17.59 | GoRK | justinu: ok |
| 19:18.08 | a1fa | GoRK : call transfer "works" |
| 19:18.17 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 19:18.21 | GoRK | haha |
| 19:18.23 | a1fa | its magic |
| 19:18.28 | a1fa | kind of like magic dust |
| 19:19.40 | GoRK | pretty soon im going to get me some magic dust and jump off the roof if i get one more 'why cant we just say call on line 6' |
| 19:19.54 | a1fa | lol |
| 19:20.10 | a1fa | i had a guy want 4 SIP Lines on a Cable Modem |
| 19:20.10 | justinu | lmao |
| 19:20.21 | a1fa | then he complained how he cant send faxes |
| 19:20.32 | a1fa | and how lame the voice quality is |
| 19:20.42 | asterboy | justinu, how do you get Polycoms to pass the "#" in the digit map? |
| 19:20.44 | a1fa | and then, he got pissed off during the black friday for BroadVoice |
| 19:20.54 | a1fa | and broke the PBX |
| 19:21.00 | justinu | asterboy: not sure about that |
| 19:21.13 | r_evolution | hey a1fa... you must've missed my bitch earlier... |
| 19:21.19 | a1fa | ahhaha |
| 19:21.21 | a1fa | which one? |
| 19:21.22 | stoffell | asterboy, you could modify the dialplan on the polycom |
| 19:21.27 | r_evolution | i had a woman tell me her ENTIRE business... rested solely on her being able to fax... |
| 19:21.28 | asterboy | In order to turn call forward on at the ZAP channel, I need to issue a 72# |
| 19:21.45 | a1fa | r_evolution : well, same here |
| 19:21.45 | r_evolution | when with the old switch (which WASN'T asterisk) she hadn't been faxing for about 4 months |
| 19:21.47 | a1fa | same problem |
| 19:21.52 | r_evolution | ^ |
| 19:22.14 | a1fa | the guy was freaking out because his fax was forced to 56k and he couldnt send fax via SIp |
| 19:22.21 | a1fa | bastard |
| 19:22.23 | r_evolution | lol |
| 19:22.26 | r_evolution | d'oh! |
| 19:22.33 | a1fa | so i forced that bitch down to 9200Bps |
| 19:22.46 | a1fa | and it faxed ok, until broadvoice crapped out |
| 19:22.56 | W8TAH | its funny how people assume that if you make dire predictions or bully them IT people will either work harder or pull some magic out thin air so everythign is perfect again |
| 19:22.57 | asterboy | ah yes, good suggestion...just put an extension in and have it dial for me. |
| 19:23.12 | stoffell | asterboy, you can look here: http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501 |
| 19:23.24 | stoffell | asterboy, search for "One final thing to modify or delete is the digitmap" |
| 19:23.33 | r_evolution | hey asterboy... don't you include Tt in the dial string in order to enable |
| 19:23.37 | r_evolution | transfer? |
| 19:23.58 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 19:23.59 | lzhang | I love it when people get belligerent, it just relieves me of any responsibility I feel I have to help them |
| 19:24.09 | r_evolution | I usually like to make dire predictions when people aren't doing what i want |
| 19:24.11 | r_evolution | :) |
| 19:24.14 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 19:24.15 | W8TAH | Yes -- |
| 19:24.19 | r_evolution | also known as big fat lies |
| 19:24.42 | asterboy | r_evolution, I tried that, but I'm getting this message: http://pastebin.ca/46524 |
| 19:24.59 | asterboy | oops not that message, this one: |
| 19:25.04 | asterboy | Mar 21 12:24:29 NOTICE[30048]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 19:25.39 | asterboy | justinu suggested that its something wrong in config or * has an issue. |
| 19:26.06 | r_evolution | yeah i cant really help you much with Zaptel stuff dude... everything i do is IP based |
| 19:26.06 | asterboy | I'll turn on sip debug for a peer and try to capture more info. |
| 19:26.07 | justinu | yeah, your zap dial string is probably screwed |
| 19:26.32 | justinu | asterboy paste your dialplan |
| 19:26.38 | asterboy | ok, doing. |
| 19:26.38 | justinu | or the section that tries to dial out on zap |
| 19:27.41 | a1fa | jesus, here it comes |
| 19:28.06 | a1fa | damn dude |
| 19:28.07 | a1fa | i am cold |
| 19:28.07 | r_evolution | oh fux |
| 19:28.10 | a1fa | its about to snow |
| 19:28.16 | r_evolution | i think thats hell freezing over |
| 19:28.16 | justinu | lol, snow |
| 19:28.21 | r_evolution | hold me :( |
| 19:28.26 | justinu | it's cold here |
| 19:28.29 | justinu | 60 degrees |
| 19:28.33 | a1fa | justinu : where@? |
| 19:28.34 | r_evolution | O_O |
| 19:28.37 | justinu | los angeles |
| 19:28.40 | r_evolution | it's gotten chilly here too :-\ |
| 19:28.41 | a1fa | damn |
| 19:28.47 | a1fa | i am down south |
| 19:28.50 | iq | PG-13 please |
| 19:28.51 | a1fa | and its damn cold |
| 19:28.56 | r_evolution | you too huh a1fa? |
| 19:29.01 | justinu | down south where? |
| 19:29.05 | justinu | argentina? |
| 19:29.09 | a1fa | AR.US |
| 19:29.12 | a1fa | no |
| 19:29.14 | cji | I'm trying to setup a cisco 7960 with asterisk@home and when the cisco phone tries logging into the tftp server I get the following errors in my asterisk log: |
| 19:29.14 | justinu | arkansas? |
| 19:29.16 | a1fa | yup |
| 19:29.17 | r_evolution | Arkansas!! |
| 19:29.20 | justinu | hmm |
| 19:29.20 | Alric | Its a little cold in TX :) |
| 19:29.22 | *** join/#asterisk clive- (n=pirch@dsl-145-24-171.telkomadsl.co.za) |
| 19:29.26 | Alric | At least for late March it is... |
| 19:29.26 | cji | Rejecting Device [device name]: Device not found |
| 19:29.28 | a1fa | its going to snow tomorrow |
| 19:29.29 | cji | any ideas? |
| 19:29.41 | [Airwolf] | I'm having a little problem with a voipbuster account. I just made a sip config like I always did for voipbuster and I can call out, but my asterisk box just hangs up incoming calls. I pasted the debug and config here http://pastebin.com/614781 |
| 19:29.41 | r_evolution | wow... that's vauge... |
| 19:29.49 | a1fa | :P |
| 19:29.53 | [Airwolf] | And I was wondering if anyone has a idea to solve it. |
| 19:30.24 | clive- | hi guys, what kernel sources do I need on centos for zaptel to compile, its giving wacky results |
| 19:30.37 | Alric | kernel-devel |
| 19:30.48 | Alric | or kernel-smp-devel, if you're using the smp kernel... |
| 19:31.18 | justinu | clive: see ~centosbug |
| 19:31.18 | clive- | alric thanks, missed that smp part |
| 19:31.20 | justinu | ~centosbug |
| 19:31.22 | jbot | well, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 19:31.32 | r_evolution | ~justinu |
| 19:31.37 | r_evolution | damn... still doesnt work... |
| 19:31.41 | clive- | thanks justinu |
| 19:31.42 | r_evolution | i wanted it to define you justin |
| 19:31.43 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 19:31.45 | justinu | no one put me in there |
| 19:32.01 | justinu | jbot, justinu is some d00d |
| 19:32.02 | jbot | justinu: okay |
| 19:32.11 | r_evolution | ~justinu |
| 19:32.13 | jbot | from memory, justinu is some d00d |
| 19:32.13 | jsharp | alfa: * is multithreaded out of the box, so it automatically takes advantage of SMP. |
| 19:32.18 | r_evolution | right on. |
| 19:32.35 | r_evolution | ~bestquote |
| 19:32.38 | r_evolution | :( |
| 19:32.44 | justinu | jbot, no justinu is some other d00d |
| 19:32.45 | jbot | okay, justinu |
| 19:33.00 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 19:33.04 | r_evolution | i wonder... |
| 19:33.04 | r_evolution | hmm |
| 19:33.05 | kink0 | good night |
| 19:33.12 | r_evolution | jbot, bestquote is <doolph> Laughing Out Loud |
| 19:33.14 | jbot | okay, r_evolution |
| 19:33.14 | r_evolution | <jpm_SD> You know.. you can just LOL.. we all know what it means now. |
| 19:33.15 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 19:33.19 | r_evolution | ~bestquote |
| 19:33.20 | jbot | i guess bestquote is <doolph> Laughing Out Loud |
| 19:33.20 | vader-- | hello |
| 19:33.26 | r_evolution | damn. |
| 19:33.31 | vader-- | does anyone know off hand what the star code is to find out what line you are calling from is? |
| 19:33.47 | asterboy | Here is me DialPlan: http://pastebin.ca/46529 |
| 19:33.56 | cji | does anyone have experience with the cisco 7960 and asterisk@home? I followed the steps in the handbook but I'm getting "registration rejection" errors on the phone itself, and and error saying it's rejecting the registration because of "Device not found" in the asterisk full log. |
| 19:34.00 | kink0 | what about to use h323 with asterisk ? is recomendable ? must I forgot it ? must I implement in separate machine ? |
| 19:34.02 | vader-- | i have a few phone lines i don't know what the phone number is |
| 19:34.16 | vader-- | i need to figure out what the numbers are |
| 19:36.19 | a1fa | jsharp : wow dude.. that is good to know |
| 19:37.26 | asterboy | vader, 311 work? |
| 19:37.37 | asterboy | sip debug peer showed me this: |
| 19:38.01 | asterboy | Looking for [transfer#] in Home2 (domain 192.168.1.8) |
| 19:38.30 | asterboy | Where transfer# is the number 'm trying to transfer to, and does *not* exist in my extensions.conf |
| 19:38.53 | stoffell | asterisk business edition supports up to 120 calls, does that mean zap calls or 'just calls' ? |
| 19:38.55 | *** part/#asterisk DrRotmos (n=magnus@85.8.2.169.se.wasadata.net) |
| 19:39.00 | asterboy | So I'll try adding it for now, and then make a pattern match later. |
| 19:39.16 | [Airwolf] | I'm having a little problem with a voipbuster account. I just made a sip config like I always did for voipbuster and I can call out, but my asterisk box just hangs up incoming calls. I pasted the debug and config here http://pastebin.com/614781 |
| 19:39.18 | [Airwolf] | And I was wondering if anyone has a idea to solve it. |
| 19:39.24 | asterboy | 120 zap calls? thats is kinda crazy isn't it? |
| 19:39.52 | stoffell | yeah, just wondering what that number is matching, the total nr of calls then? |
| 19:40.02 | r_evolution | shiiiit i hope to god it means more than 'just calls' |
| 19:40.16 | r_evolution | because im damn sure gonna run a LOT more than 120 calls through this box :-D |
| 19:40.23 | asterboy | Well, I'd like to see the how crambed the box would be for handling 120 ZAP calls. |
| 19:40.35 | r_evolution | that would be some insanity aster. |
| 19:40.41 | stoffell | r_evolution, yeah, thought so.. |
| 19:40.43 | asterboy | lol |
| 19:41.00 | r_evolution | i bet the processor would fucking grow legs |
| 19:41.04 | r_evolution | come OUT of the machine |
| 19:41.06 | r_evolution | and beat your ass |
| 19:41.14 | r_evolution | for running 120 zaptel channels O_O |
| 19:41.15 | asterboy | You could cook eggs on the case a peek calling. |
| 19:41.23 | stoffell | not sure r_evolution, 120 zaptel channels is doable i think |
| 19:41.33 | r_evolution | i dont do zap so i dunno :) |
| 19:41.39 | r_evolution | i know 120 SIP is doable |
| 19:41.48 | r_evolution | and imma see how much i can get on here before the serverscreams |
| 19:41.55 | stoffell | digium card with 4x pri is 4x30 in europe, so.. :) |
| 19:41.57 | asterboy | especially if your G729 |
| 19:42.02 | cpm | it's a bit hard on the pci bus to handle that many clocking calls, I think. |
| 19:42.58 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-154.dynamic.qsc.de) |
| 19:43.29 | shido6 | you can always overclock the proc and cool down the box with nitrogen or water |
| 19:43.46 | stoffell | cpm, TE411P should be able to do it easily |
| 19:44.04 | shido6 | keep the cards cool, too |
| 19:44.16 | shido6 | if you blow too much traffic through them they will melt components |
| 19:44.43 | shido6 | 8 months of heavy traffic |
| 19:44.48 | stoffell | hm, good to know shido6 |
| 19:44.49 | doolph | anyone here can help me solve ring back problem? |
| 19:45.17 | shido6 | the rack cooling and the xeon case cooling sometimes doesnt do it :) |
| 19:45.22 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:45.32 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 19:45.33 | *** join/#asterisk Village (n=Village@wan2.liquidcore.com) |
| 19:46.00 | Village | Hiya. Anyone know if it's possible to create groups for sip channels and not just zap? |
| 19:46.00 | *** join/#asterisk nDuff (n=chatzill@user-0ccss1b.cable.mindspring.com) |
| 19:46.42 | brodiem | does anyone know of IP phones that have amplified headset jacks so that external amps aren't needed for headsets? |
| 19:47.25 | Village | That's scary brodiem..I just got off the phone a second ago with someone and was discussing that same thing. |
| 19:47.33 | brodiem | lol |
| 19:47.52 | Village | Unfortunately I didn't have any words of wisdom for him on the issue. |
| 19:47.59 | brodiem | who do you work for? |
| 19:48.05 | Village | He wanted a headset for an Aastra 480i CT |
| 19:48.08 | russellb | what if ... you were talking to each other! |
| 19:48.21 | brodiem | I called aastra about it about an hour ago |
| 19:48.21 | Village | I don't work for a reseller or anything. |
| 19:48.28 | Village | What did they say? |
| 19:48.30 | brodiem | since I have a bunch of their phones already |
| 19:48.45 | brodiem | they said their phones need amplified headsets |
| 19:49.10 | Village | That's what my sales rep. said too..but I wasn't sure if he just wanted to make extra money on an amp. |
| 19:49.18 | Village | Amps are pricey. |
| 19:49.40 | Village | http://www.voipsupply.com/index.php?cPath=97_309_331 |
| 19:49.53 | brodiem | the problem here is we have amps and headsets already, but the boss doesn't want amps due to needing to plug into an AC outlet, and the fact that they take up too much desk space =/ |
| 19:49.56 | nDuff | How can I generate DTMF tones on an outgoing call? |
| 19:50.17 | brodiem | since we're doing PoE he doesn't want the phone to rely on AC power for anything |
| 19:50.21 | Village | I'm in agreement with you, I think it's inelegant. |
| 19:50.29 | brodiem | and since it's a call center environment, batteries in the amps won't work |
| 19:50.45 | brodiem | It *looks* like the snom phones have an amplified jack |
| 19:50.51 | nDuff | (I've been asked by one of my users to implement functionality to generate and record scheduled calls out to 3rd-party conference calls, potentially needing to send DTMF data (ie. conference number and PIN) to join. If anyone has done this before, I'd be interested to hear it. Otherwise, I'm looking at generating .call files that initiate an outgoing call with variables set for conference#,... |
| 19:50.52 | nDuff | ...PIN, etc. in a context with appropriate logic). |
| 19:50.55 | Village | Haven't used any of the snom stuff. |
| 19:51.05 | nDuff | The Snom 360s are excellent phones. |
| 19:51.09 | brodiem | I just don't like the fact that they're based out of berlin with no US contacts |
| 19:51.16 | nDuff | Haven't tried the headset jack, though. |
| 19:51.28 | brodiem | The cisco 79XX phones also look to have amplified jacks but they're so pricey |
| 19:51.42 | Village | Yeah, I couldn't justify the extra cost for those either. |
| 19:51.46 | stoffell | brodiem, also 7960 might have amplified jacks |
| 19:51.50 | Village | Wouldn't mind having one though. |
| 19:52.07 | brodiem | stoffell, yeah it's just the cost that doesn't justify it |
| 19:52.08 | astra^^ | how do i check loss of packages in * |
| 19:52.25 | brodiem | I have a grandstream GXP2000 and the headset works great without an amp on that |
| 19:52.34 | nDuff | ugh. |
| 19:52.36 | Village | We're using a bunch of the GXPs as well. |
| 19:52.52 | stoffell | brodiem, but you can't use a plantronics then, can you ... |
| 19:52.57 | brodiem | Village, how are they working for you? We got one in to test it but just wasn't overly impressed with it to have to deal with 20 more of them =/ |
| 19:52.58 | _Paulo_ | justinu, The cisco ata doesnt save the settings from my IP... |
| 19:53.00 | Village | I've done outgoing call generation nDuff..but not with variables |
| 19:53.11 | Village | Just straight number dial out stuff |
| 19:53.26 | nDuff | Village: the variables I think I can handle... it's the sending DTMF *after* the call is established I'm not so sure about. |
| 19:53.45 | clive- | justinu , made the change to that spinlock file and installed the sources, and still having erros in compilling zaptel....any pointers |
| 19:53.47 | brodiem | stoffell, we have hello direct sets, it seemed to work fine both amped and not amped |
| 19:53.49 | stoffell | brodiem, also check out the ST2030 (thomson), a real alternative to the GXP's |
| 19:53.56 | _Paulo_ | justinu, the screen shows the new values, just to fool you. It saves only from the same lan. |
| 19:53.57 | tzanger | stoffell: got a link? |
| 19:54.03 | justinu | clive: not off hand... make clean? |
| 19:54.12 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 19:54.17 | nDuff | gah, n/m... a bit more googling found an answer to that one. |
| 19:54.20 | tzanger | I am really pleased with the ip501s right now but those aren't "budget" by any means |
| 19:54.20 | Village | They work "okay". We've had some call quality issues with them, and I'm eagerly waiting for new firmware. They have issues, to be sure, but the cost is right. |
| 19:54.31 | _Paulo_ | justinu, problem solved. |
| 19:54.31 | clive- | I get this error: error: syntax error before "zone_lock" |
| 19:54.40 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
| 19:54.40 | stoffell | tzanger: http://www.voip-info.org/wiki/index.php?page=Thomson%20ST2030 |
| 19:54.43 | brodiem | Village, The speakerphone is definitely shotty on them |
| 19:54.45 | _Paulo_ | justinu, thank you |
| 19:54.48 | astra^^ | <PROTECTED> |
| 19:54.51 | Village | brodiem, agreed |
| 19:54.55 | astra^^ | y do i get tis error |
| 19:55.26 | brodiem | stoffell, it's sharp lookin |
| 19:55.26 | Seldon1975 | can someone please remind me of the freedomfiles url to polycom firmware update binaries: |
| 19:55.26 | justinu | _Paulo_: glad to help |
| 19:55.26 | nDuff | ...albeit an answer only valid on Asterisk 1.0.9 and newer (while I'm still on 1.0.7)... gah! |
| 19:55.26 | Seldon1975 | http://www.,freedomfiles.. something |
| 19:55.45 | tzanger | yeah not a bad looking phone but I wonder how comfortable it is |
| 19:55.45 | stoffell | brodiem, i'm using both gxp-2000, st2030 and polycom 501 |
| 19:56.09 | W8TAH | Thanks to ALL -- It works!!!! |
| 19:56.40 | Village | Okay, now that this place is hoppin'..any of you guys know if I can setup a group for SIP channels and not just ZAP. I need to rollover outgoing SIP calls across multiple providers when a limit is reached. |
| 19:56.42 | stoffell | i currently prefer the st2030 and polycom501, but the 501 is by no means "much better" then the st2030 imho |
| 19:56.53 | justinu | plenty of asterisk success stories here today :P |
| 19:56.57 | Village | ha |
| 19:57.05 | W8TAH | :) |
| 19:57.05 | stoffell | tzanger, if you're in belgium you can always come and have a feel on the phone ;) |
| 19:57.05 | brodiem | stoffell, Does it supprot PoE? |
| 19:57.13 | stoffell | brodiem, yes, on-board |
| 19:57.19 | brodiem | stoffell, interesting.. |
| 19:57.32 | tzanger | stoffell: :-) I'm in Canada |
| 19:57.42 | stoffell | yeah, i've got 1 in use, will be ordering the next 15 very soon |
| 19:57.45 | brodiem | stoffell, I have a gxp2000, polycom301 and aastra 480i here, none of them seem to have everything we need |
| 19:57.54 | justinu | what do you need? |
| 19:57.56 | stoffell | brodiem, what you 'miss' on those? |
| 19:58.15 | Hmmhesays | ~[tk]fender |
| 19:58.20 | brodiem | it needs to support PoE and have an amplified headset jack... the gxp2000 does but just don't like the phone |
| 19:58.22 | Hmmhesays | ~seen [tk]fender |
| 19:58.24 | jbot | i haven't seen '[tk]fender', Hmmhesays |
| 19:58.30 | Hmmhesays | ~seen [tk]-fender |
| 19:58.32 | jbot | i haven't seen '[tk]-fender', Hmmhesays |
| 19:58.33 | *** join/#asterisk backblue (n=moo@87-196-13-23.net.novis.pt) |
| 19:58.40 | justinu | ~seen [tk]d-fender |
| 19:58.41 | jbot | [tk]d-fender <n=joe@66.11.164.239> was last seen on IRC in channel #asterisk, 17h 44m 48s ago, saying: 'Primer : check the Voxilla forums for more info.'. |
| 19:58.54 | brodiem | XML browser, SMS support, etc is a bonus for agent login/logout visuals though |
| 19:59.05 | Village | I love the gxp2000 configuration and layout, but voice quality is sub-par. Aastra 480i is pretty good, but the firmware scares me..I've toasted a couple phones doing updates. The Polycom's are good, but I find setup to be overly complicated. |
| 19:59.32 | justinu | it's a shame about the audio quality issues on the gxp |
| 19:59.33 | stoffell | Village, also, if you touch/hold the handset cord of a GXP, the other sides hears a humming noise :( |
| 19:59.49 | Village | stoffell, is that a bug or a feature? |
| 19:59.49 | justinu | ferrite cores around the cables might cure that |
| 19:59.55 | clive- | justinu : that centosbug thingy, does one have to compile anything after making changes to the spinlock.h file ? |
| 20:00.06 | stoffell | Village, the older the model is, the more it's a feature :) |
| 20:00.11 | justinu | clive: just zaptel |
| 20:00.12 | Village | haha |
| 20:00.16 | stoffell | justinu, have been trying, doesn't help alot :( |
| 20:00.21 | justinu | hmm |
| 20:00.29 | justinu | those phones are a recipe for frustrations |
| 20:00.36 | clive- | zaptel doesnt want to compile.:( |
| 20:00.36 | justinu | i know... i had a customer who tried them |
| 20:00.39 | astra^^ | <PROTECTED> |
| 20:00.41 | stoffell | justinu, it's not on 'all' GXP's, but they all have it, some more then others.. |
| 20:01.03 | stoffell | brodiem, depending on your budget, you should go for polycom/thomson or snom.. |
| 20:01.04 | justinu | yeah, there's definitely inconsistant QC on the phones |
| 20:01.06 | justinu | some are better than others |
| 20:01.12 | stoffell | justinu, ack! |
| 20:01.14 | brodiem | stoffell, the polycom doesn't have an amplified headset jack |
| 20:01.19 | *** join/#asterisk tracinet (n=tracinet@64.139.137.94) |
| 20:01.20 | brodiem | at least the 301 doesn't |
| 20:01.20 | Village | With the latest firmware I had to create a script to reboot the GXP's every 12 hours or so due to screen blanking problems. |
| 20:01.36 | Village | I really wish they'd hurry up with the new release. |
| 20:01.46 | asterboy | Should this dial "72#" when I do 811? |
| 20:01.47 | asterboy | <PROTECTED> |
| 20:02.04 | brodiem | stoffell, the snom 320 looks like it may be a good option, but having some trouble finding some reviews/feedback etc |
| 20:02.16 | stoffell | brodiem, neither the 501 .. also doubt the ST2030 has it (i used a plantronics 261 on it, but i think this has amp) |
| 20:02.21 | *** part/#asterisk W8TAH (n=Tim@static-acs-24-239-210-31.zoominternet.net) |
| 20:02.23 | justinu | snom kinda sucks |
| 20:02.37 | stoffell | brodiem, indeed, snom costs, and difficult to get hold on |
| 20:02.49 | Village | We should assemble every in chat and start our own VoIP company. |
| 20:02.56 | tracinet | <PROTECTED> |
| 20:02.56 | tracinet | hello all - quick question regarding analog zap channels - getting |
| 20:02.56 | tracinet | Mar 21 14:58:37 NOTICE[8501]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... |
| 20:02.56 | tracinet | Mar 21 14:58:39 NOTICE[8501]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)... |
| 20:02.56 | tracinet | every time a call comes in - is that normal? |
| 20:03.23 | tracinet | never see that on calls from different protocols |
| 20:03.50 | [Airwolf] | I'm having a little problem with a voipbuster account. I just made a sip config like I always did for voipbuster and I can call out, but my asterisk box just hangs up incoming calls. I pasted the debug and config here http://pastebin.com/614781 |
| 20:03.52 | [Airwolf] | And I was wondering if anyone has a idea to solve it. |
| 20:04.01 | asterboy | justinu, how do I get the dialplan to do dial 72# ?? |
| 20:04.03 | brodiem | stoffell, see that's the problem. The docs on the st2030 don't say anything about the technical specs on the headset jack and neither do most other manufacturers, so it's like trial and error and I can't just keep buying random phones to see the headset functionality =/ |
| 20:04.19 | justinu | Dial(Zap/g1/72#) ? |
| 20:04.23 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 20:04.28 | asterboy | ya thats what I though. |
| 20:04.38 | Village | Someone just came to me with a very odd 480i issue. After talking to someone who is using a 480i for a few minutes, their voice degrades into a "slow motion" effect. Anyone ever hear of anything quite like that. I can tell from the sound of it that it won't be a fun one to debug. |
| 20:04.42 | asterboy | whats the "g" in there? |
| 20:05.39 | tracinet | anyone use TDM400 cards? |
| 20:05.47 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 20:05.57 | justinu | group1 |
| 20:06.22 | tracinet | just wondering if "Got event 18 (Ring Begin)... " is because of the verbosity level and completely normal when a zap call comes in |
| 20:06.35 | justinu | it's normal |
| 20:06.39 | tracinet | thanks dude |
| 20:06.48 | ryback | i have 3 X100P FXO installed on a PC. I edited zaptel.conf to include fxsks=1-3, but when running ztcfg -vvvv I get "ZT_CHANCONFIG failed on channel 2: Invalid argument (22)"... What does this mean? |
| 20:06.50 | justinu | village: that sounds like an RTP timing issue |
| 20:06.59 | justinu | village: make sure the phone is set to use 20ms RTP packets |
| 20:07.00 | asterboy | gotcha |
| 20:07.19 | Village | You can specify your group in the zapata.cfg file asterboy |
| 20:07.31 | Village | Thanks justinu..will check that now. |
| 20:08.06 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:08.13 | [hC] | [av]bani: here today? |
| 20:08.18 | astra^^ | <PROTECTED> |
| 20:08.26 | stoffell | brodiem, not even in the admin guide.. if you want i can contact thomson to get more headset info |
| 20:08.26 | justinu | ~vad |
| 20:08.28 | jbot | vad is probably Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 20:08.29 | [hC] | Anyone played with the Cisco 7970 SIP image yet? |
| 20:08.37 | [Airwolf] | ast_freak, disable noice supression in Xlite |
| 20:09.05 | Village | asterboy, http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+channels |
| 20:09.15 | Village | More information on ZAP groups ^ |
| 20:09.32 | Village | Speaking of groups, does anyone know if you can group SIP channels and not just ZAP? |
| 20:09.39 | astra^^ | vad is the problem..? |
| 20:09.55 | brodiem | stoffell, no worries I cna call them, btw: do you know who sells them and for how much? |
| 20:10.19 | stoffell | brodiem; i do:) but there's a list on the st2030 page on voip-info |
| 20:10.21 | astra^^ | * does not support it thats why it drops the packets |
| 20:10.23 | brodiem | stoffell, actually.. doesn't look like there's a phone number on their site |
| 20:11.34 | stoffell | indeed, not easy to get hold of someone there :) |
| 20:17.15 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 20:19.44 | Hmmhesays | mmm hot pockets |
| 20:19.56 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 20:20.11 | clive- | the centos bug thingy is giving me tough time..:( |
| 20:20.34 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:23.39 | *** join/#asterisk medusaXX (n=medusaxx@p54A98DD5.dip0.t-ipconnect.de) |
| 20:24.17 | *** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 20:27.19 | loonacy | Anyone know of a good logging program for Asterisk? I'd like to timestamp all output in the logs. |
| 20:29.09 | fourcheeze | how about logging to syslog? |
| 20:29.11 | [Airwolf] | loonacy, logger.conf ? |
| 20:30.10 | asterboy | thx Village |
| 20:30.19 | asterboy | Well I'm off to demo my * box. |
| 20:30.39 | stoffell | g'luck asterboy |
| 20:30.39 | asterboy | Hope I get the contract. Digium will be getting an order if I do. |
| 20:30.51 | asterboy | thx |
| 20:31.10 | asterboy | Would have liked to have had Transfer and Conferencing working. |
| 20:31.17 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 20:31.30 | clive- | ~centosbug |
| 20:31.31 | jbot | rumour has it, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 20:31.31 | asterboy | I'll work on that later and just say I didn't have time to configure it yet. |
| 20:31.34 | r_evolution | hey aster... what do you mean conferencing? |
| 20:31.40 | r_evolution | like... meetme? |
| 20:31.44 | stoffell | hehe |
| 20:31.51 | asterboy | I have the Polycom IP600 phones. |
| 20:32.01 | ryback | i have 3 X100P FXO installed on a PC. I edited zaptel.conf to include fxsks=1-3, but when running ztcfg -vvvv I get "ZT_CHANCONFIG failed on channel 2: Invalid argument (22)"... What does this mean? |
| 20:32.04 | r_evolution | yeah but you should still be able to set it up in * |
| 20:32.08 | asterboy | Wanted to be able to press conference, dial a number and get another party on the phone. |
| 20:32.11 | asterboy | same with Transfer. |
| 20:32.17 | r_evolution | oh so you mean 3ways |
| 20:32.21 | clive- | blaa, time to swicth off and try tomorrow |
| 20:32.22 | asterboy | but I don't have something setup right in my dialplpan. |
| 20:32.39 | *** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it) |
| 20:32.44 | asterboy | http://pastebin.ca/46529 |
| 20:32.59 | asterboy | If you find something I can do, let me know. |
| 20:33.56 | r_evolution | no... i see a bunch of different lines setup... are you intentionally blocking them from accessing one another? |
| 20:34.00 | asterboy | I get this message when I try it: Mar 21 13:33:52 NOTICE[562]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 20:34.04 | asterboy | <PROTECTED> |
| 20:34.12 | asterboy | no |
| 20:34.23 | r_evolution | well you've got them all under different contexts |
| 20:34.27 | asterboy | could be just my n00b fingers |
| 20:34.50 | asterboy | that is so I can group hunt when a call comes in. |
| 20:35.11 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:35.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:35.12 | asterboy | like Home and Home2 are on Line 1 of the phones, I want them to both ring when a call comes in. |
| 20:35.31 | r_evolution | yeah... well what im saying is this... |
| 20:35.42 | r_evolution | are you using a 4 FXO TDM card or what? |
| 20:36.01 | r_evolution | <PROTECTED> |
| 20:36.21 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 20:36.30 | asterboy | 1 FXO Wildcard X101P and 2 clones in 3 pci slots, channels 1-3 |
| 20:36.39 | asterboy | any help appreciated |
| 20:36.49 | r_evolution | look at the PM |
| 20:37.08 | asterboy | PM? |
| 20:37.14 | r_evolution | private msg |
| 20:37.15 | Nodren | i have a question, is fxo the module for incoming phone lines or is that fxs? |
| 20:37.22 | r_evolution | fxo |
| 20:37.34 | r_evolution | fxs is what you plug the phone into that you want to ring |
| 20:37.41 | Nodren | so if i'm setting up a box with 4 lines on a TDM400P and all the phones are ip phones |
| 20:37.51 | Nodren | i want 4 FXO modules? |
| 20:37.51 | r_evolution | yeah |
| 20:37.55 | Nodren | thanks! |
| 20:37.55 | r_evolution | if you want four lines coming in |
| 20:37.59 | Nodren | yes i do |
| 20:38.00 | Nodren | :D |
| 20:38.04 | r_evolution | k |
| 20:47.46 | justinu | hehe, wb nodren |
| 20:48.09 | r_evolution | he's on fire today justin |
| 20:48.16 | r_evolution | he's after the FXO modules to make the card work :) |
| 20:49.06 | Nodren | yeah i need to get this project done |
| 20:49.23 | Nodren | my normal job is PHP coding, and my boss took me off a time sensative project to set up asterisk |
| 20:49.33 | Nodren | so i'm eager to get back to the original project |
| 20:50.32 | ryback | i have 3 X100P FXO installed on a PC. I edited zaptel.conf to include fxsks=1-3, but when running ztcfg -vvvv I get "ZT_CHANCONFIG failed on channel 2: Invalid argument (22)"... What does this mean? |
| 20:52.20 | *** join/#asterisk rfmonk (n=rfmonk@71-35-163-161.tukw.qwest.net) |
| 20:53.54 | *** join/#asterisk luckyduck (i=lucky@gentoo/developer/luckyduck) |
| 20:58.03 | _Paulo_ | somebody knows how I make a hotline using a cisco ata 186? |
| 20:58.58 | _Paulo_ | I tried "H**123*123*123*23#" at DialPlan in the cisco ata, but it doesnot work. |
| 20:59.20 | _Paulo_ | in fact dialing an IP like that doesnot work. |
| 21:04.14 | Axel69 | hi guys |
| 21:04.18 | Axel69 | i have a little problem |
| 21:04.36 | r_evolution | welcome to the club... everyone who comes in here has a little problem :) |
| 21:04.42 | Axel69 | in the sip.conf when i define more tha 2 codecs it doesn't work |
| 21:04.50 | r_evolution | that's an odd problem |
| 21:04.57 | r_evolution | which are you defining? |
| 21:05.07 | Axel69 | G729 and G723 |
| 21:05.23 | r_evolution | do you have licenses for 729 |
| 21:05.28 | Axel69 | yes |
| 21:05.38 | r_evolution | so if you JUST use 729 it works |
| 21:05.42 | r_evolution | and if you JUST use 723 it works? |
| 21:06.17 | Axel69 | yes |
| 21:06.26 | ryback | i have 3 X100P FXO installed on a PC. I edited zaptel.conf to include fxsks=1-3, but when running ztcfg -vvvv I get "ZT_CHANCONFIG failed on channel 2: Invalid argument (22)"... What does this mean? |
| 21:06.31 | Axel69 | if i define another codec it doesn'; work |
| 21:06.42 | ryback | anyone? |
| 21:06.48 | r_evolution | k hold axel im on the phone |
| 21:06.55 | r_evolution | what other codec are you definig? |
| 21:06.57 | r_evolution | defining* |
| 21:07.26 | Axel69 | allow=alaw (g711a) |
| 21:07.27 | Axel69 | allow=ulaw (g711u) |
| 21:07.38 | Axel69 | when i define the 4 codecs...goes crazy |
| 21:08.32 | Axel69 | it rings...when the other side pick up the phone my side keeps ringing and the other side it does'n hear anything |
| 21:08.40 | r_evolution | weird. |
| 21:08.46 | r_evolution | try not defining 723 |
| 21:08.53 | r_evolution | as in disallowing it |
| 21:09.00 | Axel69 | ok |
| 21:09.04 | Axel69 | i will try that now |
| 21:12.10 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 21:12.14 | *** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no) |
| 21:13.41 | Seldon1975 | Nodren: ahahaha same as me |
| 21:16.08 | *** join/#asterisk Dr-Linux (n=Linux@host202-147-168-130.lhr.dancom.net.pk) |
| 21:16.26 | Seldon1975 | all of you... are gay |
| 21:16.35 | justinu | am not |
| 21:16.39 | Seldon1975 | r2 |
| 21:16.52 | Seldon1975 | times ten |
| 21:17.21 | Dr-Linux | hi |
| 21:17.35 | Luke-Jr | ... |
| 21:18.04 | Axel69 | it works |
| 21:18.05 | Axel69 | great |
| 21:18.15 | Corydon-w | Well, some of us are |
| 21:18.58 | Seldon1975 | not all? |
| 21:19.06 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:19.12 | Qwell[] | Seldon1975: most? |
| 21:19.58 | Dr-Linux | justinu: my all softphone are on same pvt network, but registerd with remote asterisk box, so if they will talk each other, what will be the voice quality difference |
| 21:20.01 | Dr-Linux | ? |
| 21:20.20 | justinu | they won't talk to each other |
| 21:20.23 | *** join/#asterisk _octothorpe (n=octothor@unaffiliated/octothorpe) |
| 21:20.30 | *** join/#asterisk redondos (n=redondos@190.48.44.119) |
| 21:20.35 | justinu | they'll talk to each other using asterisk as a proxy |
| 21:20.43 | _Sam-- | what about canreinvite = yes |
| 21:20.46 | Dr-Linux | justinu: yes |
| 21:20.55 | justinu | canrinvite won't work if they're behind a nat |
| 21:21.09 | Luke-Jr | nor can you record the call if you use canreinvite |
| 21:21.17 | Dr-Linux | canreinvite is "yes" for all of them |
| 21:21.59 | _Sam-- | i never knew anything behind nat couldnt use canreinvite, but then again i dont know alot of things still. |
| 21:22.10 | Seldon1975 | does anyone know a good Atari2600 emulator for the Polycom501 embedded O/S? |
| 21:22.15 | justinu | lol |
| 21:22.25 | Seldon1975 | :D |
| 21:22.26 | *** join/#asterisk jpm_SD (n=jpm@207-40-115-38.sugardog.com) |
| 21:22.33 | justinu | sounds pretty gay to me |
| 21:22.44 | Seldon1975 | no! it would be totally sweet |
| 21:22.48 | Seldon1975 | come on |
| 21:23.06 | Dr-Linux | justinu: what's the logic on same pvt network, some users needs "qualify=yes" some do not need? :S |
| 21:23.06 | justinu | the phone takes 3 minutes to boot, how the fuck is it going to run an atari 2600 game? |
| 21:23.44 | Seldon1975 | you know you'd love to play Tank Battle while talking to telemarketers |
| 21:23.44 | justinu | probably depends on whether the phone does keep-alives by default or not |
| 21:23.46 | Dr-Linux | justinu: pretty gay :P |
| 21:24.06 | justinu | i don't talk to telemarketers |
| 21:24.10 | justinu | they talk to my torture script |
| 21:24.27 | _Sam-- | what, if caller ID is unknown, you make them say something before you answer? |
| 21:24.28 | Seldon1975 | justinu: they implemented dumb features like BuddyWatch and completely ignored the Telephone Tetris players market |
| 21:24.42 | justinu | i don't answer |
| 21:24.43 | Dr-Linux | justinu: i just observed SJphone do not need qualify=yes |
| 21:24.47 | justinu | the IVR script picks it up |
| 21:24.58 | justinu | if they know the sekrit handshake, they can ring thru |
| 21:25.07 | Seldon1975 | justinu: is that the one that plays Hammond-organ lift music |
| 21:25.12 | _Sam-- | they only get IVR if callerid = unknown? |
| 21:25.21 | justinu | that, or certain blacklisted numbers |
| 21:25.47 | _Sam-- | how is your 33km fiber connection going? :) |
| 21:25.56 | justinu | i couldn't get a price quote on the spool |
| 21:26.03 | justinu | useless fuckers in #asterisk |
| 21:26.07 | _Sam-- | seriously, why not use some of the new wirless stuff |
| 21:26.14 | justinu | it wasn't really for telecom |
| 21:26.24 | _Sam-- | like i said i have a friend going 60mbps over the orthogon systems stuff @ 111km |
| 21:26.24 | justinu | trying to do some phsyics experiements |
| 21:26.31 | Luke-Jr | woohoo I'm about to drop iConnectHere |
| 21:26.38 | justinu | i need lightguide cable |
| 21:26.42 | justinu | 30km of it |
| 21:26.43 | justinu | :P |
| 21:26.54 | Luke-Jr | I wonder if I can demand refunds for the past month after the number is ported |
| 21:27.00 | *** join/#asterisk rfmonk (n=rfmonk@dsl231-054-135.sea1.dsl.speakeasy.net) |
| 21:27.02 | Luke-Jr | since they didn't have my number working |
| 21:27.24 | jpm_SD | You can demand.. sure. Will they pay - that is the real question. |
| 21:27.28 | Luke-Jr | heh |
| 21:27.53 | x86 | can anyone get to http://telasip.com ? |
| 21:27.54 | redondos | Heya. When recording IVRs, what's the rule of thumb for knowing how long every menu should be recorded in an instance, or separated into pieces? |
| 21:27.57 | Luke-Jr | or rather, how much will they pay =p |
| 21:27.57 | x86 | they seem to be down... |
| 21:28.04 | *** join/#asterisk harlequin516 (n=sham@65.39.84.194) |
| 21:28.06 | redondos | I don't know if I'm being clearenough. |
| 21:28.26 | Luke-Jr | I managed to figure out that even though they don't have my # working, it will forward calls to the unavailable forwarding number |
| 21:28.27 | x86 | my local number was just LNP ported over to them today, but it's not working, and with sip debug on my asterisk server, i'm not seeing any attempt at all |
| 21:28.33 | Luke-Jr | but they charge per minute for forwarding |
| 21:28.36 | _Sam-- | 500 meters of fiber = about 1000 bucks |
| 21:28.47 | _Sam-- | justinu : http://store.yahoo.com/wcsc/fibopcabdual.html |
| 21:29.06 | Luke-Jr | x86: I can get there |
| 21:29.12 | justinu | cool, thanks! |
| 21:29.26 | x86 | err now it works |
| 21:29.29 | _Sam-- | how would you run it? above ground? |
| 21:29.30 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 21:29.33 | _Sam-- | on telephone poles? |
| 21:29.34 | jpm_SD | redondos, I generally try to make each menu one sound file. trying to piece something together seems like too much work for me. |
| 21:29.53 | redondos | jpm_SD: Thanks for the input. |
| 21:30.13 | redondos | jpm_SD: Doesn't it get very tedious when you have to re-arrange your IVRs, or you just don't do that? |
| 21:31.03 | justinu | it doesn't even need to be run |
| 21:31.09 | justinu | i just need 30km of spooled cable |
| 21:31.14 | *** join/#asterisk ramo (i=ramo@59.92.131.200) |
| 21:31.14 | justinu | the length is what's important |
| 21:31.17 | _Sam-- | seems like an expensive experiment |
| 21:31.20 | justinu | yeah |
| 21:31.21 | jpm_SD | redondos, also.. backgrounding a bunch of little sound clips just bloats the dial plan. Hrm.. When we decide to change prompts I just pay Allison to make a new recording |
| 21:31.26 | Dr-Linux | _Sam--: hows your wife? :) |
| 21:31.44 | jpm_SD | or set of records if the situation requires. |
| 21:32.36 | jpm_SD | Also, I think it makes things sound more natural and less like you built the prompt out of a bunch of little sound bites. Personal perference there. |
| 21:33.14 | _Sam-- | Dr-Linux : she's doing not too bad....pregnant and cranky. |
| 21:33.21 | redondos | Cool. BTW, do you mind telling me how much someone earns for making such recordings? (Allison in your case. |
| 21:33.24 | redondos | ) |
| 21:35.15 | jpm_SD | 120 words (5x20word prompts)= 50 bucks... |
| 21:35.16 | Dr-Linux | _Sam--: cranky? |
| 21:35.19 | *** join/#asterisk Whisk (n=whisk@whisk.gotadsl.co.uk) |
| 21:35.23 | _Sam-- | Dr-Linux : irritable |
| 21:35.36 | Dr-Linux | _Sam--: she gonna have your baby? |
| 21:35.43 | jpm_SD | You can buy prompt from Allison from digium. |
| 21:35.49 | _Sam-- | unless she is the virgin mary, yes. |
| 21:36.42 | Dr-Linux | _Sam--: great, will it be a first baby? |
| 21:37.10 | _Sam-- | only if you dont count our jack russell, he has been our baby for the last 5 years. |
| 21:37.23 | _Sam-- | probably takes more work than a regular baby |
| 21:39.52 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 21:42.54 | Dr-Linux | _Sam--: so what you do while your wife is not in action for you due to her pragnancy? |
| 21:43.29 | jbalcomb | feed the duck |
| 21:43.47 | Seldon1975 | go for runs |
| 21:43.52 | Seldon1975 | watch gardening videos |
| 21:44.48 | websae | anyone here use astbill? |
| 21:45.21 | Seldon1975 | regarding voice recordings, I actually use the free speech synthesis engine at http://www.bell-labs.com/project/tts/voices.html to good effect |
| 21:45.34 | *** join/#asterisk Strom_C (i=strom@66.159.243.60) |
| 21:45.34 | Dr-Linux | i don't understand these 2 * app SendText() and SendURL() ? :S |
| 21:46.15 | Qwell[] | haha |
| 21:46.19 | Qwell[] | "If you plan to enter text which our system might consider to be obscene, check here to certify that you are old enough to hear the resulting output." |
| 21:46.34 | Qwell[] | That is the coolest checkbox...like...EVER |
| 21:47.13 | Seldon1975 | heh |
| 21:47.32 | Seldon1975 | you can spell rude works phonetically |
| 21:47.33 | Dr-Linux | Seldon1975: how can i save the recorded file? :S |
| 21:48.22 | Seldon1975 | Dr-Linux: after you click 'synthesize' you get a download prompt |
| 21:48.25 | Seldon1975 | no? |
| 21:48.33 | Seldon1975 | check your MIME settings of your browser |
| 21:48.51 | Dr-Linux | :S |
| 21:49.05 | Dr-Linux | Seldon1975: i wanna try an IVR :S |
| 21:49.18 | Dr-Linux | i don't know hows "woman" wocie there :S |
| 21:49.53 | *** join/#asterisk rfmonk (n=rfmonk@71-35-163-161.tukw.qwest.net) |
| 21:51.00 | Seldon1975 | it may take a few tries to spell what you want said phonetically |
| 21:51.16 | Seldon1975 | but when you get that right it sounds perfect |
| 21:51.51 | Seldon1975 | the site seems sluggish atm - I think we're all hammering it |
| 21:52.12 | Dr-Linux | Seldon1975: is there any other free prompts available like Allison :S |
| 21:52.34 | Dr-Linux | Seldon1975: that's not working for me |
| 21:52.41 | Dr-Linux | i'll check tomorrow in the office |
| 21:52.46 | Seldon1975 | try: http://www.bell-labs.com/project/tts/voices-java.html |
| 21:52.50 | Seldon1975 | for more control |
| 21:53.10 | x86 | anyone use TelaSIP? |
| 21:53.51 | x86 | i only want to use them for origination services, but they never told me my username or password, nor the gateway I should be using... |
| 21:54.05 | r_evolution | you might wanna call them in that case x86 |
| 21:54.18 | x86 | that would work, except they dont answer :( |
| 21:54.30 | r_evolution | some carriers don't use a username or password... they just fwd to a specific IP |
| 21:54.33 | x86 | my local number just LNP ported to them today, and I cant get ahold of anyone :( |
| 21:54.42 | r_evolution | oh shit |
| 21:54.44 | r_evolution | thats no fun |
| 21:54.47 | r_evolution | email? |
| 21:54.55 | x86 | i tried that too |
| 21:55.02 | r_evolution | Drive there? |
| 21:55.06 | x86 | someone in here recommended them, i dont understand why ;) |
| 21:55.14 | x86 | they are states away from me heh |
| 21:55.28 | r_evolution | so? |
| 21:55.30 | r_evolution | you have a car |
| 21:55.31 | Dr-Linux | justinu: really eyeBean is very good soft client |
| 21:55.32 | r_evolution | gas i presume |
| 21:55.39 | justinu | eyebeam is the best softphone, imo |
| 21:55.51 | r_evolution | shhh |
| 21:55.55 | r_evolution | dont tell my company that justin |
| 21:56.00 | r_evolution | they already want to NOT use hardphones |
| 21:56.04 | Dr-Linux | how can i avail eyeBeam video facility :S |
| 21:56.24 | Dr-Linux | r_evolution: lolz |
| 21:56.46 | r_evolution | LAUGH OUT LOUD! ZZZ |
| 21:56.46 | Whisk | i'm getting a problem where i'm seeing "Maximum trunk data space exceeded" spamming to the console on one box on the end of an iax trunk after a while - the trunk has about 25-35 calls constantly - anyone any ideas on how to fix this? |
| 21:56.48 | r_evolution | O_o |
| 21:58.00 | Dr-Linux | i also can't find call forwarding option in eyeBeam :S |
| 21:58.43 | Dr-Linux | justinu: SJphone is much better than xlite |
| 21:58.44 | x86 | Dr-Linux: handle that on Asterisk |
| 21:58.52 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 21:58.57 | justinu | xlite is old |
| 21:59.15 | justinu | not under development anymore |
| 21:59.17 | sevard | Does anyone know what the variable $MA sipura 2002 echos to? or how I could echo that variable? |
| 21:59.36 | x86 | Dr-Linux: i've got a nice little setup where users can dial *72 to blind forward all calls to another number, and *74 to try both the softphone, and then transfer out to another number if the softphone is not answering |
| 21:59.38 | tecnico | my * keeps loosing registration to my provider.... trunk.13784 , anybody with the same problem ? |
| 21:59.47 | Dr-Linux | x86: i don't know how to do it on asterisk, bcoz i don't know putDB and DelDB sutff :( |
| 22:00.05 | x86 | Dr-Linux: want my conf? |
| 22:00.15 | tecnico | how can I re-register to my server other than restarting or reloading iax2 ? |
| 22:00.21 | Dr-Linux | x86: yes, if you can |
| 22:00.36 | sevard | $MA == mac address |
| 22:02.18 | *** join/#asterisk securez (n=securez@121.Red-80-33-36.staticIP.rima-tde.net) |
| 22:02.26 | securez | Hello |
| 22:03.47 | securez | I'm a newbye with asterisk, so i want to get one functional, and make some test with a FXO port, as i can see a cheap FXO port can be a soft modem, only X100P clones work, or other soft modems? |
| 22:04.03 | securez | i have a soft modem with Lucent chip |
| 22:04.59 | jets | there are some intel chipsets that have been known to work as a x100p clone |
| 22:05.06 | jets | http://www.voip-info.org/ has had that info in the past |
| 22:05.54 | jets | i wouldn't use it in production. |
| 22:05.58 | jets | ever. ;) |
| 22:07.36 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 22:08.40 | justinu | the x100p clone's are so cheap you might as well just buy one |
| 22:10.13 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:10.32 | brc_ | they aren't clones |
| 22:11.03 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 22:11.27 | *** join/#asterisk maxx4life (n=max4life@71-35-210-12.slkc.qwest.net) |
| 22:12.07 | Connor | Hey guys.. question.. I want to setup a pre-queue.. I want to queue up calls and then send them down a pri to another phone system.. I want to limit the number of calls the other phone system gets to about 2 or 4 calls.. How can I do this?? I'm thinking I'll need to use chanlocal or something and groupcheck |
| 22:15.36 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:16.47 | *** part/#asterisk kakadu (n=blubb@p54B8DFF0.dip.t-dialin.net) |
| 22:17.59 | redondos | OK, I just recorded this file but when asterisk is supposed to play it back, nothing is heard. No sound, but the connection remains established. Here;s the file: http://www.redondos.biz/files/test.gsm |
| 22:18.17 | redondos | Please take a look at it and tell me if it's incompatible with *? |
| 22:18.42 | redondos | My partner recorded it on his computer using SoundForce. I wouldn't even get near that thing, hate the windows. |
| 22:19.50 | harryvv | I have created a second ivr for recording for another incomming sip line but any time I press the extention get a fast busy tone and no cli responce. Checked the dialplan on the ip500 web page and added the extention and still no luck. What may i be missing ? |
| 22:20.09 | sevard | With TFTP is it possible to get a list of files in a directory? or is that just not something tftp does. As far as I can remember you can put/get |
| 22:20.15 | Seldon1975 | this is awesome, I just registered at dailywtf.com with the username 'bobafett' - I can't believe it wasn't taken! |
| 22:20.27 | redondos | sevard: not possible. |
| 22:20.46 | Seldon1975 | TFTP = GET or PUT only |
| 22:20.47 | jets | redondos: what does the cli say? it says it is playing it using playback or background? |
| 22:21.29 | redondos | jets: background |
| 22:21.45 | redondos | Why might it be saying that? |
| 22:21.49 | securez | jets: thank, i'll get one for testing |
| 22:21.53 | sevard | redondos: I didn't think so. |
| 22:22.03 | jets | redondos: and it doesn't say an error about unknown format, or file not found? |
| 22:22.58 | sevard | redondos: with TFTP if you try to GET a file that doesn't exist it will just tiemout, correct? |
| 22:22.58 | sevard | timeout* |
| 22:24.03 | redondos | sevard: no, it will say that the file wasn't found. |
| 22:24.12 | redondos | jets: Nope, nothing about that. Weird, huh? |
| 22:24.20 | jets | hmmm |
| 22:24.26 | jets | set debug 50 |
| 22:24.55 | Mavvie | on hold music of digium has stopped when I got a second call... |
| 22:25.03 | Mavvie | wonder if that is intentional |
| 22:25.39 | harryvv | back |
| 22:25.46 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 22:25.50 | harryvv | anyone ever made two ivrs before? |
| 22:26.34 | sevard | redondos: my sip provider gave me a sipura 2002 that downloads a config via tftp, except it's not provisioning itself. I'm trying to find out if it's my fault. I'm nmaping the tftp server now. |
| 22:28.35 | jets | harryvv: two ivrs?? |
| 22:28.51 | Dr-Linux | harryvv: how two ivrs? :S |
| 22:30.08 | *** join/#asterisk rfmonk (n=rfmonk@71-35-163-161.tukw.qwest.net) |
| 22:35.43 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 22:36.13 | *** join/#asterisk heka (n=heka@82.114.68.126) |
| 22:36.40 | sevard | This is kind of a dumb way to provision an ATA. If a provider provisions it via the ATA going out and getting the config with TFTP, eg tftp://<ip>/ata/ata$MA.cfg and that sets it up, that would contain the subscriber name and number and password... |
| 22:36.50 | harryvv | jets yes. One for residence and another for bussiness. |
| 22:37.08 | sevard | All one would have to do is get a MAC address and then they can make free phonecalls. |
| 22:37.18 | Luke-Jr | Is it possible to specify two IPs for a IAX2/SIP context? |
| 22:37.43 | *** join/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 22:38.45 | justinu | TFTP in general is a dumb way to provision an ATA |
| 22:38.51 | justinu | how's it going to pass thru NAT? |
| 22:39.06 | harryvv | anyway i goto go. |
| 22:39.21 | harryvv | Dr-Linux yea, you can make two ivrs. |
| 22:39.58 | heka | hello, wich codec is the best when dealing with big lattency? |
| 22:39.58 | sevard | justinu: what do you mean? FTP can pass through nat, the server isn't NATd the client is |
| 22:39.59 | harryvv | You can assign umpteen sip DIDs to each ivr. Thats probebly how the retail sip providers do it. |
| 22:39.59 | justinu | i said TFTP, not FTP |
| 22:40.05 | sevard | TFTP can't pass through NAT? |
| 22:40.26 | justinu | i had a lot of trouble making that work |
| 22:40.29 | harryvv | sev tftp is the primary way to upgrade cisco routers |
| 22:40.45 | sevard | harryvv: I understand that |
| 22:40.47 | harryvv | so it should pass |
| 22:41.00 | Dr-Linux | harryvv: tht's what we gonna do, but i don't think it will be difficult to do? |
| 22:41.09 | GoRK | tftp can pass through nat with help; though the nat device and/or the tftp server have to know that they are going through nat |
| 22:41.22 | harryvv | im outa here. |
| 22:41.31 | sevard | On my linux shell I try to tftp <ip> and get ata/ata00000000.cfg (real mac address) and it times out |
| 22:41.33 | justinu | HTTP provisioning is the way to go |
| 22:41.42 | GoRK | linux routers have ip_conntrack_tftp and cisco routers have payload inspection |
| 22:42.23 | GoRK | but by default you will likely have problems with tftp clients behind most consumer nat devices without using some specific tftp servers |
| 22:44.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:45.34 | Dr-Linux | <harryvv> You can assign umpteen sip DIDs << ? |
| 22:45.42 | Dr-Linux | what's upmteen ? |
| 22:45.54 | Strom_C | many |
| 22:46.01 | Strom_C | lots |
| 22:46.05 | Strom_C | a large amount |
| 22:46.11 | websae | tons |
| 22:46.11 | Qwell[] | exactly several |
| 22:46.24 | sevard | hahaha |
| 22:46.28 | Strom_C | hahahaa |
| 22:46.29 | Dr-Linux | :S |
| 22:47.48 | sevard | Hey Strom |
| 22:48.12 | *** part/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 22:48.17 | Strom_C | hi |
| 22:48.30 | sevard | what's popalackin fo rizzle |
| 22:48.45 | Dr-Linux | how this app works >> sendtext() ? |
| 22:48.54 | Qwell[] | Dr-Linux: show application sendtext |
| 22:48.57 | Strom_C | oh, i'm shizzy in my chizzy in my apartmizzsldfkasdfasdhfasdrasdvdfbg |
| 22:50.17 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 22:50.24 | Dr-Linux | Qwell[]: thanks, and is there anyway to send a MASS message/call/voicemail to all active extensions? |
| 22:51.12 | sevard | Strom_C: Excellent. |
| 22:51.47 | sevard | Strom_C: I've been researching SIP security today and I've come to the conclusion that there...really isn't any. |
| 22:51.50 | Dr-Linux | like if i add a new feature in my * box, and i want to let all let know about this feature, so how can isend them all a message? |
| 22:51.53 | Strom_C | well, duh |
| 22:52.13 | sevard | "duh" ? :/ |
| 22:52.15 | Qwell[] | Dr-Linux: email |
| 22:52.25 | justinu | there's SIPS |
| 22:52.35 | Dr-Linux | Qwell[]: that i know |
| 22:53.11 | Dr-Linux | but asterisk won't send this email |
| 22:53.48 | austinnichols101 | dr-linux: why not just give them a call :) |
| 22:53.53 | *** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 22:54.08 | GoRK | anyone here have problems with qos'ing rtp audio with cisco and asterisk? I am having the problem that the cisco is not matching the packets with 'match protocol rtp audio' but is matching them with 'match protocol rtp' |
| 22:54.10 | sevard | justinu: SIPS? |
| 22:54.17 | GoRK | i wonder if it is a bug in asterisk or cisco |
| 22:54.18 | Dr-Linux | austinnichols101: call to everyone? if you have 100 users? |
| 22:54.29 | justinu | sevard: SIP over TLS |
| 22:55.05 | GoRK | very few things support SIPS unfortunately |
| 22:55.15 | *** part/#asterisk _deg_ (n=deg@200.250.222.8) |
| 22:55.36 | sevard | You couldn't put a router that encapsulates packets |
| 22:55.40 | sevard | I supposed that'd be a VPN |
| 22:55.49 | GoRK | VPN's work; that is how i do it heh |
| 22:57.02 | justinu | snom is the only phone I know of that does |
| 22:59.48 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 23:00.09 | Dr-Linux | justinu: do you like tatoos? :) |
| 23:01.20 | justinu | nope |
| 23:01.49 | blitzrage | tattoo's rock |
| 23:02.00 | Strom_C | depends on the tattoo |
| 23:02.09 | Strom_C | some tattoos look good, but most are pretty ugly |
| 23:02.24 | blitzrage | true, depends what you get |
| 23:02.33 | Dr-Linux | justinu: i hardly seen 1 or 2 guys over here having tatoos |
| 23:02.40 | sevard | I've only seen a couple tats that look good |
| 23:02.58 | justinu | you gotta love the guys who get all tatted up, and then wear wife beater shirts for the rest of their lives so they can show everyone how cool thy are |
| 23:03.03 | blitzrage | I have a calabi-yau shape on my back :) |
| 23:03.04 | Dr-Linux | justinu: i have seen many in US porn movies :P |
| 23:03.09 | sevard | Strom_C: why don't you care about security? :( |
| 23:03.33 | Strom_C | how the hell did you get that out of me telling you that it was obvious SIP has no security? |
| 23:03.40 | blitzrage | I'm the exact opposite, I have tats in places where I can keep them covered, and very few people even know I have them |
| 23:03.49 | sevard | I'm just pooking you for information |
| 23:03.49 | orlock | justinu: almost as funny as people who wear suits every single day cos they think wearing a tie and shit equates to being smart |
| 23:03.53 | Strom_C | I care about security, but SIP is not secure |
| 23:04.02 | sevard | Strom_C: but you use SIP, correcdt? |
| 23:04.02 | Strom_C | well poke all you want but don't put words in my mouth, dork |
| 23:04.06 | sevard | heh |
| 23:04.15 | Strom_C | I use SIP on my LAN |
| 23:04.22 | sevard | do you VPN your SIP calls? |
| 23:04.32 | sevard | What about your voip calls? |
| 23:05.02 | Strom_C | my SIP calls dont go out over the internet, so who cares? all my interexchange traffic is IAX |
| 23:05.15 | justinu | that's better? |
| 23:05.25 | Strom_C | well, not by much |
| 23:05.32 | sevard | Strom_C: I don't know much about IAX. I read it's just as open |
| 23:05.39 | justinu | it's authenicated, but not encrypted |
| 23:06.00 | sevard | well sip is authentiated, but as far as I can tell the password are sent plain-text |
| 23:06.11 | sevard | which is just _awesome_ |
| 23:06.47 | justinu | no, sip uses the same auth as http |
| 23:06.53 | justinu | challenge response |
| 23:06.57 | _Paulo_ | if it use a secure auth method, why bother? I dont expect voip being more secure than PSTN |
| 23:07.26 | Strom_C | exactly - PSTN isn't inherently secure - if you need a secure communications channel, you can go to the extra trouble of setting one up |
| 23:09.06 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 23:10.23 | _Paulo_ | good night people. time to ride. |
| 23:10.36 | kink0 | what h323 is recomended for asterisk ? will be able to use g729 with h323 ? |
| 23:10.48 | Strom_C | kink0, h323 is a nightmare. |
| 23:11.23 | X-Rob | h323 gives you caner. |
| 23:11.24 | X-Rob | cancer |
| 23:11.40 | kink0 | Strom_C, then what must I do if I need to support h323 peers ? |
| 23:11.42 | Strom_C | h323 puts babies in wood chippers |
| 23:12.08 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 23:12.14 | kink0 | yeah, but my peer is ussing just h323, so I am doing my first try with h323 just now |
| 23:12.26 | Strom_C | kink0, what h323 peers are you supporting and why can they not switch to something more reasonable? |
| 23:12.28 | X-Rob | "<sevard> well sip is authentiated, but as far as I can tell the password are sent plain-text" I love people who make gigantic statements like that which is incredibly easy to test. Lets do a sip trace. Ooh, uh. That's challenge-responce. |
| 23:12.50 | X-Rob | BUT IT'S STILL SENT PLAIN TEXT, DAMMIT! |
| 23:13.07 | kink0 | Strom_C, not sure about what hard/soft they are ussing, but the sends all signalling ussing h323 and not SIP |
| 23:13.43 | sevard | whooa dude, chillax. |
| 23:14.12 | kink0 | may be they just use Cisco, and uses h323 for other compatibilities with other local gw or other remote peers |
| 23:14.27 | Strom_C | kink0, http://www.voip-info.org/wiki/view/Asterisk+H323+channels |
| 23:14.36 | Strom_C | I wish you the best of luck. |
| 23:14.56 | sevard | I'm notannouncing as "hey guys as far as i can tell it goes like this" i'm saying as far as I dunderstand I have found it to be such and I present said information to be evaluated by people who know much more than I, as I am learning. |
| 23:16.36 | justinu | your passwords aren't in the clear, so don't worry |
| 23:17.11 | Strom_C | besides, if someone is stealing my SIP passwords, I've got bigger problems to worry about |
| 23:17.44 | kink0 | Strom_C, I have compiled suscefully and appears stable now... just I am installing some h323 client to test it. |
| 23:18.12 | kink0 | stable = fews hours running only, from compilation hours ago. |
| 23:20.22 | *** join/#asterisk jskcrtech (n=j@30-pool1.ras14.floca.alerondial.net) |
| 23:22.45 | ryback | I have zaptel.conf configured for 3 FXO cards, now what should i configure to start receiving calls? |
| 23:22.49 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 23:23.16 | Strom_C | extensions.conf, obviously |
| 23:24.09 | ryback | ok, but before that, should i configure zaptel.conf only or zapata.conf also? |
| 23:24.13 | Qwell[] | both |
| 23:25.38 | ryback | i have on zapata.conf language=en context=from-pstn signalling=fxs_ks rxwink=300 and callerid, callwaiting, echo cancel lines... am i missing something? |
| 23:26.04 | Qwell[] | I don't know. What is it (not) doing? |
| 23:26.52 | sevard | Strom_C: how come i never see you in here :P |
| 23:26.55 | ryback | well, i haven´t tested anything yet... i´m just starting with asterisk so i´m trying to understand what should i coonfigure |
| 23:27.09 | Strom_C | sevard, because I get busy sometimes? |
| 23:27.12 | Qwell[] | ryback: well, try it. see if its broke |
| 23:27.29 | sevard | Strom_C: Screw your busy body bull crap |
| 23:27.35 | websae | ryback: is that your last name? |
| 23:27.41 | Strom_C | do I know you from somewhere? I don't recognize the handle |
| 23:27.42 | websae | my best friend's last name is ryback |
| 23:27.44 | websae | interesting |
| 23:27.50 | Qwell[] | websae: /whois much? |
| 23:28.23 | sevard | Strom_C: Try dropping the last three characters off my nick. |
| 23:28.33 | Strom_C | well I thought so but the two IPs are different |
| 23:28.34 | ryback | my setup is a PC with 3 FXO and would like to use softphone to test before buying sip phone |
| 23:28.34 | jskcrtech | ryback: read a good asterisk book http://voipspeak.net/index.php?/content/view/33/2/ |
| 23:28.41 | Strom_C | on #la2600 you're at ucsc.edu |
| 23:28.44 | sevard | Strom_C: I have shells everywhere |
| 23:28.49 | Strom_C | blah blah blah |
| 23:28.52 | sevard | :) |
| 23:29.04 | justinu | la 2600... do you know eric (jgalt)? |
| 23:29.04 | *** join/#asterisk harlequin516 (n=sham@65.39.84.194) |
| 23:29.17 | ryback | websae: no it´s not my lastname... i use it as casey ryback from steven seagal's character |
| 23:29.17 | Strom_C | justinu, oy. |
| 23:29.20 | Strom_C | don't get me started. |
| 23:29.26 | justinu | lol |
| 23:29.33 | justinu | i went to one of those meetings |
| 23:29.44 | justinu | at some place that works for NASA |
| 23:29.46 | harlequin516 | What's the name of the thing that lets me share my phone line for dialout with others freely in a p2p fashion? |
| 23:29.53 | Strom_C | justinu, yes, that's the meeting I run |
| 23:29.53 | justinu | aviation contractor or something |
| 23:30.03 | Hmmhesays | paranoia paranoia everybody's coming to get me |
| 23:30.10 | justinu | the organizer was no-show that night |
| 23:30.21 | Strom_C | I was probably out of town |
| 23:30.22 | ryback | Quell[] how should i configure an extension? |
| 23:30.31 | Qwell[] | ryback: manually? |
| 23:31.21 | harlequin516 | Anyone know what I am talking about? |
| 23:31.41 | ryback | I installed asterisk@home so I have a web interface also |
| 23:32.00 | harlequin516 | p2p dialout network for asterisk? |
| 23:32.10 | russellb | dundi |
| 23:32.13 | jskcrtech | ryback: see http://asteriskathome.sourceforge.net/handbook/ |
| 23:32.24 | [ProB]CrazyMan | stupid question... if I want to have an extension 7223 and want to catch also XXXXX7223 and YYYYY7223 do I have to mak exten => _.7223, ... ? |
| 23:32.40 | harlequin516 | I thought dundi was only to replace enum? |
| 23:33.03 | jskcrtech | [ProB]CrazyMan: only if you want 1111111111117223 etc etc etc to goto 7223 |
| 23:33.50 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 23:34.30 | ryback | jskcrtech: thanks, i´m there... just a question, for softphone what type of extension should i use? SIP, IAX2, ZAP or Custom? |
| 23:34.48 | Qwell[] | I'd like to see a zap softphone |
| 23:34.59 | russellb | harlequin516: no ... but maybe you're thinking of fwdout, which uses dundi |
| 23:35.07 | jskcrtech | ryback: sip |
| 23:35.15 | [ProB]CrazyMan | jskcrtech: yes, because have to remove the 0049 and local prefix to forward to internal fone |
| 23:36.33 | ryback | jskcrtech: what's outbound CID? |
| 23:36.52 | jskcrtech | outbound caller id |
| 23:37.00 | jskcrtech | just set it to the extension |
| 23:37.01 | harlequin516 | russellb Thanks |
| 23:37.09 | harlequin516 | Thats wahat I was looking for |
| 23:39.13 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 23:39.16 | *** join/#asterisk Mw3_ (i=mw3@national.t-error.hu) |
| 23:41.02 | *** join/#asterisk trixter_ (n=trixter@65.172.209.246) |
| 23:41.56 | russellb | harlequin516: no problem |
| 23:42.19 | Strom_C | here's a dumb one: is there anything special I have to do to ${DATETIME} to get it to not be empty? For some reason it's blank on my asterisk box. I have the local time zone set correctly... |
| 23:45.46 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
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| 23:57.11 | Strom_C | this is the calmest I've ever seen this channel |
| 23:57.15 | Strom_C | freaky. |
| 23:58.09 | Strom_C | woohoo! |
| 23:58.12 | Strom_C | I'm a torso! |
| 23:58.55 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
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