| 00:01.15 | lunaphyte | ManxPower: why do you say that? |
| 00:02.22 | *** join/#asterisk isamar (n=isamar@voice.maxirede.net) |
| 00:02.32 | *** join/#asterisk javb (n=javb@tdev213-87.codetel.net.do) |
| 00:02.43 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 00:03.50 | javb | did s sip trunk beetween two Asterisk, both are ok in 'sip show peeers' we have call from asterisk 1 to asterisk 2, but no from asterisk 2 to one, and im getting the error "failed to authenticate" i have "unsecure = very" i`m using asterisk 1.2 .. any ideas what could it be ? |
| 00:04.52 | mosty | javb, can you pastebin the sip debug log from each machine? |
| 00:04.58 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:10.12 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 00:11.51 | javb | mosty: good call from asterisk 2 to asterisk 1: http://pastebin.com/m3d6eed7e ... bad call form asterisk 1 to asterisk 2: http://pastebin.com/m7c6b4143 |
| 00:13.09 | ManxPower | ~trunk |
| 00:13.10 | jbot | well, trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 00:15.07 | *** join/#asterisk VitoCorleon (n=Owner@rrcs-76-79-244-73.west.biz.rr.com) |
| 00:15.20 | javb | mosty, there? |
| 00:15.24 | VitoCorleon | hey guys, what ports should i have forwarded in my router? |
| 00:17.30 | tzanger | a trunk is a single voice channel between two pieces of switching equipment? |
| 00:17.41 | mosty | javb, i'm looking, but also doing other things at the moment |
| 00:17.48 | tzanger | I thought a trunk was a single connection carrying multiple voice channels between two pieces of switching equipment |
| 00:17.53 | drmessano | ~trunk |
| 00:17.54 | jbot | it has been said that trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use ... |
| 00:18.03 | lunaphyte | when people use the term sip trunk, what should they really be saying? |
| 00:18.11 | tzanger | sip conversation, probably |
| 00:18.11 | drmessano | sip peer |
| 00:18.20 | drmessano | There is no such thing as a Sip trunk |
| 00:18.23 | drmessano | Sip doesn't trunk |
| 00:18.27 | mosty | "sip call" |
| 00:18.37 | husimon | is it possible to copy the astdb to another to a backup asterisk box and have all the keys work properly? |
| 00:19.00 | mosty | husimon, probably |
| 00:19.12 | husimon | i guess there is only one way to find out |
| 00:19.20 | trippss | i'm curious, what exactly is the (assuredly good) reasoning why you don't want to use "r" in outbound trunk dial plans? |
| 00:19.30 | husimon | i'm setting up call forwarding in the astdb as flags but my backup box has to get that db as well |
| 00:19.56 | mosty | trippss, because if the destination is busy, you might here ringing and then the engaged tones, which is confusing |
| 00:20.00 | mosty | hear, even |
| 00:20.23 | mosty | husimon, maybe you should use an sql database, and func_odbc or AGI |
| 00:20.34 | tzanger | the only reason I have ever found for using 'r' is when dialing a cell phone |
| 00:20.39 | *** part/#asterisk TrXuk (n=trx@cpc5-flee1-0-0-cust532.glfd.cable.ntl.com) |
| 00:20.44 | tzanger | so you don't hear "the customer you are dialing is not availbale" |
| 00:21.00 | trippss | mosty: gotcha - wouldn't a busy signal generate the proper signaling to either open the media stream and/or generate a busy signal locally? |
| 00:21.04 | mosty | tzanger, or when you dial multiple destinations at the same time |
| 00:21.28 | husimon | mostly well if works then I dunno, it's simpler then an sql i think |
| 00:21.30 | mosty | trippss, yes but the r option will start the ringing tones before it knows the other end is busy |
| 00:21.30 | tzanger | I never had trouble with that |
| 00:22.23 | mosty | javb, it looks like you have the wrong password |
| 00:22.45 | trippss | here's my issue - calling some phones with certain telcos (seems to be cingular and AT&T in my case in particular) it never rings until the other end picks up, vm picks up or otherwise the media stream starts. So there may be 15-30 seconds of dead silence until something happens |
| 00:22.53 | husimon | does anyone remember what that alternative to rsync was? i thought it was psync or something |
| 00:25.47 | *** join/#asterisk L4m3r (n=l4m3r@about/essy/warning/L4m3r) |
| 00:27.15 | jblack | husimon: scp? |
| 00:28.09 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:29.38 | lunaphyte | when i'm passing calls from the fxo port on an spa3102 to asterisk, as i understand it - if i want to pass caller id, i have to use a dial plan (on the spa) of (S0<:s>). but i think that means that the spa will call asterisk using whatever the cid number is, right? how do i tell asterisk to answer that if the extension it is calling is always changing? |
| 00:29.43 | *** part/#asterisk linux_galore (n=Richard@dsl-220-253-76-44.NSW.netspace.net.au) |
| 00:30.25 | mosty | lunaphyte, use _X. |
| 00:30.57 | lunaphyte | ohhhh. |
| 00:31.10 | husimon | jblack, no there was something that i saw someone using to send data from the primary to secondary asterisk box |
| 00:31.21 | husimon | it might be a wrapper around rsync that adds more functionality |
| 00:31.49 | mosty | lunaphyte, i don't know much about the spa3102, but i doubt it works in the way you expect |
| 00:31.50 | plik | more functionality to rsync?? Like what? |
| 00:32.47 | lunaphyte | mosty: well, neither do i.. :) but how do you mean? |
| 00:34.22 | drmessano | lunaphyte... no no |
| 00:34.37 | mosty | lunaphyte, sip supports callerid, the spa3102 could do the equivalent of Dial(SIP/youraccount) |
| 00:35.27 | Robba | anyone here from Australia? |
| 00:36.10 | mosty | yes |
| 00:37.51 | *** join/#asterisk PepOSX (n=angeldav@190.72.144.246) |
| 00:38.01 | lunaphyte | drmessano: no? |
| 00:39.08 | drmessano | Hang on.. there is a dialplan for that |
| 00:40.03 | lunaphyte | hey, this is you, isn't it? http://freepbx.org/support/documentation/howtos/howto-spa-3102-and-freepbx |
| 00:40.26 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 00:40.33 | jblack | Based upon the work of other people, and some testing here.... I am going to go out on the ledge and state "Sipphone and Asterisk have incompatible SIP implementations." |
| 00:41.08 | Defraz | Has anyone gotten a world wide packets Lightning Edge 46 VoIP Useragent working with Asterisk? |
| 00:41.41 | jblack | One of the two is breaking the rules. based on the non-asterisk hard/software that does work with sipphone, I believe there to be a bug in *'s SIP invite implementation. |
| 00:41.42 | Robba | has anyone here got a TE122P card working with optus multiline? |
| 00:42.39 | mosty | robba, not that combination specifically, but where are you stuck? |
| 00:43.02 | husimon | btw jblack, it was csync2 that I was trying to remember. |
| 00:43.12 | jblack | husimon: Ok. |
| 00:43.14 | Robba | Our Optus Multiline ISDN is delivered via E1 over SHDSL |
| 00:43.28 | Robba | it seems to connect and the light on the card is green |
| 00:43.46 | Robba | but when calling the associated number asterisk doesn't display anything |
| 00:44.19 | JT | i've done optus SHDSL delivered E1 PRIs before |
| 00:44.23 | JT | they "just work" ;) |
| 00:45.24 | mosty | set verbose 10 & set debug 10? |
| 00:45.36 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:45.37 | mosty | and enable verbose logging to the console in logger.conf ? |
| 00:45.45 | trippss | mosty: so to address this issue, what's the worst thing that can happen (i.e., using tr for outbound calls); if the other side is busy, then would the call ring first and then go busy or continue ringing over the busy signal? |
| 00:46.44 | trippss | motsy: in my case this should trigger the outisbusy macro which should generate a busy signal. confusing possibly but better than 30 seconds of silence before someon picks up |
| 00:46.59 | JT | nah |
| 00:47.04 | mosty | trippss, sometimes you get two different ringing sounds, or a ring then an engaged signal. definitely better than 30s of silence |
| 00:47.07 | JT | all you need is pri intense debug span 1 |
| 00:47.09 | trippss | s/motsy/mosty |
| 00:47.21 | nny_1 | i think thew problem with the term sip trunk is that it relies on the definition of a telecom trunk to convey the same thing for SIP. Since in asterisk a trunk is an IAX2 stream, it would mean you are saying a "SIP IAX2 stream" which makes no sense... |
| 00:47.36 | trippss | i suppose i'll give it a shot for a week or two and see how it works out |
| 00:47.39 | nny_1 | the* |
| 00:47.49 | JT | no, only an IAX2 connection in trunking mode is sort of a trunk |
| 00:48.12 | Robba | currently its set to 13 |
| 00:48.34 | mosty | Robba, what does "pri show span 1" show? pastebin the output |
| 00:49.02 | JT | Robba: pri intense debug span 1 is of most importance |
| 00:49.13 | nny_1 | er yeah thats what i meant :) |
| 00:49.27 | JT | Robba: also whats in your zapata and zaptel.conf files? |
| 00:49.36 | nny_1 | what do you guys think of this case for asterisk |
| 00:49.37 | nny_1 | http://www.itx-warehouse.co.uk/ProductImages/Casetronic%20Travla%20C158%20Black.jpg |
| 00:50.18 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:50.34 | Robba | whats pastebin? |
| 00:50.39 | nny_1 | ~pb |
| 00:50.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:50.45 | jblack | heh. |
| 00:50.48 | nny_1 | :P |
| 00:50.49 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 00:50.56 | JT | Robba: use the rafb.net one |
| 00:51.07 | JT | i have issues viewing the pastebin.* ones atm |
| 00:51.09 | mosty | jblack, my custom func_channel backport worked |
| 00:51.17 | jblack | mosty: Great |
| 00:51.45 | nny_1 | I am actually working on setting up a SIP channel from Junction Network as an extra channel in case all the local lines are being used right now |
| 00:51.49 | *** join/#asterisk real0ne (i=real0ne@adll-180-244-206-196.marocconnect.net.ma) |
| 00:53.07 | nny_1 | anyone know of other providers for SIP or IAX2 that's better than 2.9 per minute national? |
| 00:53.21 | Robba | http://rafb.net/p/DBUEfO82.html |
| 00:54.18 | plik | 2.9 waht per min national in which country? |
| 00:55.02 | Robba | is that right? |
| 00:55.10 | JT | Robba: i take it both sent AND received calls fail |
| 00:55.16 | Robba | indeed |
| 00:55.27 | JT | Status: Provisioned, Down, Active |
| 00:55.45 | JT | please rafb.net /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 00:55.49 | Robba | ok i'm new to this |
| 00:56.05 | Robba | just bare with me |
| 00:56.09 | JT | ok |
| 00:56.20 | JT | i suspect you may need a PRI crossover cable |
| 00:56.27 | JT | but let me check configs first |
| 00:56.31 | Robba | i have an e1 crossover in place |
| 00:56.36 | nny_1 | plik: US sorry |
| 00:56.44 | Robba | want to ssh to it? |
| 00:56.46 | nny_1 | plik: a little ethnocentric on that one |
| 00:56.52 | JT | if you want |
| 00:57.02 | Robba | cause i assume you know what you would be looking for |
| 00:57.12 | Robba | is it ok to PM you? |
| 00:57.22 | JT | yes |
| 00:57.42 | plik | nny_1: not sure off hand (I'm uk) but I'd have thought you could do better |
| 00:59.01 | plik | les.net do $0.015/Minute |
| 01:00.55 | *** join/#asterisk Grnd-Wire (n=grundofw@75.147.178.170) |
| 01:01.03 | Grnd-Wire | greetings everyone! |
| 01:01.09 | plik | hi |
| 01:01.41 | Grnd-Wire | Can someone confirm for me that this is a valid statement: |
| 01:01.42 | Grnd-Wire | ExecIf($["${DB(VOICEMAIL/greeting)}" = "1"]|Playback|${SOUND_PATH}/day-primary.ulaw) |
| 01:02.15 | Grnd-Wire | VOICEMAIL/greeting is indeed set to 1 when I do "database show voicemail" |
| 01:08.03 | jblack | That playback looks suspicious to me. |
| 01:08.34 | jblack | Do you mean ExecIf(....,Playback(${SOUND_PATH}/day.ulaw)) |
| 01:09.49 | nny_1 | plik: thanks that looks pretty good |
| 01:12.29 | nny_1 | anyone know a possible way to setup a linux box to use non static IPS? Something similar to netbios, if say, I was sending a system to an "unknown" network... |
| 01:12.41 | nny_1 | not required, just kicking ideas around |
| 01:12.42 | JT | you mean dhcp? |
| 01:12.45 | nny_1 | lol |
| 01:12.46 | nny_1 | well yeah |
| 01:13.01 | JT | yes of course linux supports dhcp :) |
| 01:13.01 | Grnd-Wire | jblack: hmm.. I'm pretty sure that is correct for an ExecIF statement.. |
| 01:13.02 | nny_1 | I guess I could set up bind on the box |
| 01:13.19 | nny_1 | JT: lol more for finding devices with changing IPs... like netbiox in win |
| 01:13.21 | jblack | Grnd-Wire: If you're happy. |
| 01:13.23 | nny_1 | netbios* |
| 01:13.34 | nny_1 | no crap bind wouldn't work either |
| 01:13.43 | mosty | nny_1, you can do netbios lookups on linux |
| 01:13.44 | nny_1 | guess I could use nmbd |
| 01:13.47 | Grnd-Wire | jblack: You put the application as one parameter to "ExecIf", and the next parameter is all of the parameters to pass in. |
| 01:14.11 | mosty | nny_1, of course you'd be a fool not to just setup dhcp with static mappings, and a small dns server for your lan |
| 01:14.14 | Grnd-Wire | nny_1: Look into... hmm.. What is apple's protocol again? |
| 01:14.16 | nny_1 | mosty: yeah thats what I was thinking, just wasn't sure if there was something linux native.. heck that would work anyways |
| 01:14.19 | Grnd-Wire | rendezvous? |
| 01:14.22 | Grnd-Wire | oh.. bonzai! |
| 01:14.25 | nny_1 | mosty: this isn't on my network |
| 01:14.36 | Grnd-Wire | I believe there is Linux support for bonzai |
| 01:14.53 | nny_1 | hmm well crap nm |
| 01:15.01 | mosty | there's that avahi thing |
| 01:15.03 | nny_1 | I couldn't put the netbios name of the sever in the phones anyways |
| 01:15.13 | nny_1 | maybe snoms would, but not polycoms afaik |
| 01:15.44 | mosty | nny_1, sounds like a waste of time, just get the admin to give you a static ip |
| 01:15.49 | nny_1 | just brainstorming, it's not a huge deal, I can just question the client to find out what their network scheme is |
| 01:16.10 | *** join/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 01:16.13 | Grnd-Wire | nny_1: heh.. You don't know yet? Just tell them you need a static IP :) |
| 01:16.29 | nny_1 | Grnd-Wire: this is for pre built systems |
| 01:16.43 | nny_1 | Grnd-Wire: obviously normally this is a non issue |
| 01:17.36 | Grnd-Wire | hmm ok |
| 01:18.15 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 01:18.18 | nny_1 | Grnd-Wire: selling SOHO phone systems, so this more for your "I have no clue what my IP subnet is" |
| 01:18.36 | nny_1 | Grnd-Wire: alternatively I will have a howto for them to check on a windows machine |
| 01:18.46 | nny_1 | and just hope/assume that the 250-254 range is unused |
| 01:19.06 | mosty | nny_1, i bet you will run into troubles |
| 01:19.13 | nny_1 | mosty: yeah no kidding |
| 01:19.14 | Grnd-Wire | nny_1: err.. You presuming there is nothing else on the network? You should run your own DHCP server then.. |
| 01:19.44 | nny_1 | Grnd-Wire: no I am going to make them "responsible" for the IP scheme, but a lot of purchasers have no idea what that is |
| 01:20.08 | Grnd-Wire | hmm ok |
| 01:20.16 | nny_1 | at least with the small biz clients I have here, they don't know crap about their networks, and since they are SOHO they don't have an IT guy |
| 01:20.52 | nny_1 | in other markets I assume they either do have a "computer guy" who usually removes spyware //geek squad //firedog// etc and even they *don't* know |
| 01:21.43 | nny_1 | the systems I sell here I have complete control over.. usually I mix asterisk with Clarkconnect if possible and make the PBx the edge device, if needed, for remote ISP etc |
| 01:21.45 | nny_1 | SIP* |
| 01:21.48 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 01:22.38 | nny_1 | and CC has Intrusion Detection/ Prevention, nice web interface for the firewall, LAMP etc etc etc and nice repos with minimal changes outside of security updates |
| 01:22.51 | mosty | nny_1, charge them support for figuring that stuff out for them |
| 01:22.52 | nny_1 | they even have asterisk in the repos now, but I still copile |
| 01:22.55 | nny_1 | compile |
| 01:23.09 | nny_1 | mosty: yeah that's gonna be included in the price, 1 hour setup, 1 hour "support" |
| 01:23.33 | mosty | on setups like that, 2 hours is very optimistic |
| 01:23.36 | nny_1 | mosty: it's good stuff actually, I am writing a script that queries the installer to change things like voicemail email address, users, SIP etc etc |
| 01:23.44 | nny_1 | mosty: not with what I am doing |
| 01:24.44 | nny_1 | mosty: not doing ANYTHING custom outside of what is needed to define the user i.e caller id, voicemail, everything else is pre built |
| 01:24.49 | nny_1 | it's pretty damn cool actually |
| 01:25.06 | nny_1 | I can teach a monkey how to install it, and it's NOT freepbx or trixbox which I won't use |
| 01:25.37 | nny_1 | OTOH I am offering customization, and we have 24/7/365 support packages |
| 01:27.01 | nny_1 | I am really pissing off the local nortel/ mitel suppliers.. they can't touch asterisk for pricing or functionality |
| 01:27.12 | *** join/#asterisk weazahl (n=jeremy@adsl-66-143-53-16.dsl.ksc2mo.swbell.net) |
| 01:27.18 | nny_1 | even the local telco is trying to compete against me |
| 01:27.25 | javb | if did MusicOnhold() ... how can make the cmd to be playing music for an specific ammount of seconds? |
| 01:29.50 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 01:30.12 | nny_1 | javb: do you mean after the MOh is done the call re rings the party or just goes to silence>? |
| 01:30.35 | rbd | hey guys....I'd like to use SIP_HEADER to get the most recent (e.g. topmost) Via: header in the current sip message...does it natually do this, or is there some set/array notation I can use to do this? |
| 01:31.30 | nny_1 | afaik sip_header is read only |
| 01:31.51 | nny_1 | we wait |
| 01:31.53 | weazahl | riddlebox: you around? |
| 01:31.54 | nny_1 | mis read the q sorry |
| 01:32.02 | Inssomniak | this X100P card, so is no longer made? |
| 01:32.17 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 01:32.21 | nny_1 | Inssomniak: you can find them on ebay etc, usually knock offs, I have one here for testing |
| 01:32.21 | riddlebox | weazahl, yeah |
| 01:32.39 | Inssomniak | nny_1, a guy on ebay is claiming "authentic", so Im not sure |
| 01:33.05 | nny_1 | Inssomniak: well I have one here that is the winmodem version, they said authentic too, but I don't think it is |
| 01:33.10 | mosty | Inssomniak, even the "authentic" ones are very crappy |
| 01:33.14 | nny_1 | indeed |
| 01:33.17 | Inssomniak | ok |
| 01:33.34 | nny_1 | you can score a digium or even a knock off of digium card and one module for around 100 |
| 01:33.38 | nny_1 | well |
| 01:33.41 | nny_1 | digium is actually 175 |
| 01:33.44 | nny_1 | ish |
| 01:34.01 | Inssomniak | Im not sure I bought the right device for what I want it was an spa 3102 but dont they have built in routers? |
| 01:34.28 | J4k3 | you're prolly better off with a 1 fxo/1 fxs ata |
| 01:34.44 | nny_1 | Inssomniak: no thats just a VOIP adapter |
| 01:34.47 | J4k3 | than dicking with a x100p or paying out the ass for a real pci card. |
| 01:34.48 | *** join/#asterisk BeeBuu (n=beebuu@219.132.190.48) |
| 01:34.51 | nny_1 | well ATA* |
| 01:35.00 | mosty | Inssomniak, what are you trying to do? |
| 01:35.01 | nny_1 | hmm why FXS? |
| 01:35.13 | Inssomniak | I thought the 3102 was an FXO and an FXS ATA |
| 01:35.27 | nny_1 | 2 FXS 1 FXO |
| 01:35.32 | Inssomniak | I need to connect my analog POTS line and my analog phones to asterisk |
| 01:35.39 | nny_1 | but yeah mosty's question is best, what's your goal |
| 01:36.15 | nny_1 | Inssomniak: how many incoming lines and how many lines do you want on the analog phones? |
| 01:36.48 | Inssomniak | I just want one analog phone connected, but I want to recieve calls from my VOIP line as well as my POTS line on it |
| 01:36.55 | mosty | Inssomniak, the 3102 can do that for you, i believe |
| 01:37.23 | Inssomniak | with asterisk in the middle acting as a voicemail/ivr box |
| 01:37.39 | Inssomniak | (turn my analog phone into an "extension" |
| 01:42.27 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 01:42.37 | J4k3 | stay off the POTS |
| 01:42.43 | J4k3 | DARE to stay POTS free |
| 01:45.19 | nny_1 | Heh I wish I had more non POTS systems |
| 01:45.23 | Inssomniak | probaby shoudla bought the diginum card but they are so expensive |
| 01:45.29 | nny_1 | most of my installs are under T1 capacity |
| 01:45.40 | nny_1 | Inssomniak: there are other options |
| 01:46.29 | nny_1 | Inssomniak: http://www.voiplink.com/OpenVox_Cards_Asterisk_FXS_FXO_T1_E1_Linux_VoIP_s/112.htm' |
| 01:46.35 | nny_1 | but |
| 01:46.37 | nny_1 | !openvox |
| 01:46.40 | nny_1 | ~openvox |
| 01:47.00 | nny_1 | not sure if the bot has an opinion, but they are cheap versions of the digium cards... |
| 01:47.37 | Inssomniak | the spa-3102 seems mickey mouse to me, its like there are 2 things now doing the work |
| 01:47.41 | [TK]D-Fender | SPA-3102 = 1 FXS, 1 FXO |
| 01:47.51 | nny_1 | nah the SPA is good |
| 01:48.10 | nny_1 | Some people would say a SPA is a better alternative to the FXO cards |
| 01:48.22 | Inssomniak | what are the advantages to having the PCI card? |
| 01:48.30 | Inssomniak | instead of another gateway like the SPA? |
| 01:48.38 | [TK]D-Fender | Inssomniak, For single line home us, just go with the SPA |
| 01:48.46 | Shaun2222 | any of you guys know of any IAX2/SIP clients for the iphone that work well? |
| 01:49.03 | [TK]D-Fender | Inssomniak, PCI card solutions are for business use (better clarity, faxing, etc) |
| 01:49.08 | Inssomniak | so theoretically the SPA box with asterisk PBX will do all the same stuff as having the PCI card |
| 01:49.18 | tzanger | it looks like the openvox 4-port cards are exact knockoffs of the TDM4xx cards. the FXO is just the winmodem. can't tell with the t1 cards |
| 01:49.24 | tzanger | they look like they could be hte original tormentia cards |
| 01:49.37 | nny_1 | tzanger: yeah I haven't used them ,just know they are there |
| 01:50.09 | tzanger | the site says they'll bus master, so they're definitely NOT tormentia cards |
| 01:50.12 | nny_1 | I am still selling digium cards, although the sangoma with the hardware echo cancel looks good.. digium recently came out with a PCI-x card that has hardware echo cancel |
| 01:50.22 | tzanger | I can't really see the cards though to take a good look at 'em |
| 01:50.51 | nny_1 | I have a module here that came with an IP04 blackfin box... |
| 01:50.59 | nny_1 | works and looks just like the digium ones |
| 01:51.12 | tzanger | yeah |
| 01:51.52 | nny_1 | for now if i need FXO/FXS I sell digium, although I working on trying out other brands |
| 01:51.56 | plik | anyone know if the SPA3102 can handle CLI? |
| 01:52.30 | [TK]D-Fender | plik, CLI? |
| 01:52.49 | JT | <PROTECTED> |
| 01:52.51 | JT | not pci-x |
| 01:53.07 | plik | CLID - calling line identity from a POTS analogue line |
| 01:53.18 | tzanger | 1x pcie cards look so silly |
| 01:53.19 | nny_1 | JT: er yeah PCI-e |
| 01:53.44 | [TK]D-Fender | plik, CallerID? |
| 01:53.48 | plik | yah |
| 01:53.48 | JT | plik: yes, probably unless your POTS line uses an esoteric CLI signalling method |
| 01:53.52 | tzanger | "whaddayamean my $thousands card connects to the PC through 1 0.5" connector?! |
| 01:53.53 | [TK]D-Fender | plik, Yes it does |
| 01:53.58 | *** part/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 01:54.34 | JT | also, forget about distinctive ringing with ATAs |
| 01:54.39 | plik | uk CAllerID is a bit different to most places but I'd hope the uk version supports it ... no mention of it on linksys's site though |
| 01:54.42 | plik | thanks |
| 01:56.18 | plik | am I right in thating the PAP2 or whatever only has FXS ports, no FXO? |
| 01:56.29 | mosty | plik, yes |
| 01:56.36 | plik | thanks |
| 01:56.47 | weazahl | riddlebox: i got 2 magix at auction |
| 01:57.30 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:58.09 | weazahl | riddlebox: each is about exactly half of my clients system. i got me a new toy. about $60 both, shipped. |
| 02:00.12 | *** join/#asterisk saftsack (n=saftsack@p4FC75E9F.dip.t-dialin.net) |
| 02:00.17 | nny_1 | what is magix? |
| 02:00.23 | *** part/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 02:00.27 | MaliutaWrk | thank god someone has a clue ... http://www.voipchoice.com.au/why-isnt-skype-listed-on-voip-choice.html |
| 02:00.54 | riddlebox | weazahl, nice |
| 02:01.21 | riddlebox | weazahl, now you have spare parts if they need something |
| 02:01.27 | weazahl | ever find the manuals? |
| 02:01.29 | nny_1 | I imagine that picture on voipchoices banner is what my funeral will look like |
| 02:01.40 | nny_1 | I love how graphic designers think |
| 02:01.40 | riddlebox | umm hold on, you might be able to get it from avaya.com |
| 02:01.48 | weazahl | riddlebox: when i pull the ds1 card. in goes a 408 |
| 02:02.14 | Inssomniak | so.. tell me again, I can get my father in law a ATA adapter, and he can connect to my asterisk box, and he then could use VOIP to call me locally or be an "extension" I can pick up and dial 200 and reach him? |
| 02:02.19 | weazahl | if i own one i can get it? |
| 02:02.46 | Agrajag- | if i have one Dial command with a timeout of 5 seconds, then another straight after it (dials more people) - how can i make the 2nd Dial command only execute if the 1st one returned NOANSWER ? |
| 02:02.49 | JT | Inssomniak: sure |
| 02:02.56 | Grnd-Wire | weazahl: You need Lucent/Avaya stuff? |
| 02:03.27 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 02:04.11 | riddlebox | weazahl, what version is the magix? |
| 02:04.36 | weazahl | riddlebox: ok, i have a 4 ports fxo in the machine for the vm. what about the fxs ports? 16 or 20 is what this phone system can handle. what kind of sip interface |
| 02:04.45 | plik | Inssomniak: and if you're not careful with your dialplan, he can leave you with a big phone-bill too ;) |
| 02:04.47 | weazahl | hold on. |
| 02:05.12 | Inssomniak | plik, well I dont have an outgoing VOIP provider, just an incoming DID |
| 02:05.18 | riddlebox | weazahl, so now you want to take fxs from asterisk and connect them to the magix as lines? |
| 02:06.02 | plik | that'snow... but surely you'll want to experience the full joys of voip goodness? |
| 02:06.07 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 02:06.16 | Nivex | huzzah! beta2! |
| 02:06.33 | riddlebox | weazahl, http://support.avaya.com/japple/css/japple?PAGE=ProductArea&temp.productID=107609&temp.bucketID=159898&temp.releaseID=129533&x=14&y=6 the bottom one is as good as I can get for ya, you can check the other versions and see if there is one in there |
| 02:06.52 | Inssomniak | Im sure voip is great.. but in my situation is not very cost effective to use |
| 02:07.05 | weazahl | danm cant find the auction right now. |
| 02:07.06 | Grnd-Wire | riddlebox: What is he looking for? |
| 02:07.32 | weazahl | magix programming manual |
| 02:07.46 | Grnd-Wire | weazahl: R4 |
| 02:07.47 | Grnd-Wire | ? |
| 02:07.57 | weazahl | not really sure |
| 02:08.15 | *** part/#asterisk nny_1 (n=Scott@64.20.141.61.dyn-e-pool14.pool.hargray.net) |
| 02:08.18 | Grnd-Wire | weazahl: oh, ok.. Someone started talking about Magix, so it was making my ears burn :P |
| 02:10.44 | weazahl | im interfacing asterisk as to a merlin magix. i need to know how to program the magix |
| 02:11.21 | Grnd-Wire | weazahl: How are you integrating though? |
| 02:11.50 | Grnd-Wire | weazahl: You want the * to act like a voicemail? |
| 02:12.24 | weazahl | Grnd-Wire: the * is going to emulate merlin mail, and add all the features... this has been done before |
| 02:13.08 | weazahl | also, because of the cost of voice here, we can use sip to terminate calls at about 1/3dr the cost of now. and get more bandwidth for net. |
| 02:13.10 | Grnd-Wire | weazahl: yup.. I've done it twice, and I've got a Partner ACS sitting next to me - Partner Mail uses different codes than Merlin Mail. |
| 02:13.30 | Grnd-Wire | weazahl: So do you even know how to setup a Merlin Mail? |
| 02:13.36 | weazahl | nope. |
| 02:14.05 | Grnd-Wire | <sigh> www.tek-tips.com has a massive amount of information.. Let me see if I can find a link. |
| 02:14.13 | weazahl | i have the extensions list for them, asterisk, email server, etc. |
| 02:14.16 | Grnd-Wire | Sadly, the manuals SUCK if you don't already know what you're doing.. |
| 02:14.33 | weazahl | i wont really need to do much on the merlin |
| 02:14.47 | weazahl | the asterisk box is just going to sneak in between |
| 02:15.17 | Grnd-Wire | weazahl: umm.. ok? So you don't know anything about the Magix, but you think you know how to integrate them? |
| 02:15.27 | weazahl | need to turn off the ds1, and enable more analogue ports to its trunk. |
| 02:15.40 | J4k3 | weazahl: only 1/3rd? I dropped my telco bill from about $310/mo to $15/mo with VoIP :) |
| 02:15.45 | Grnd-Wire | weazahl: There are many, many limitations with those systems - something we're not used to with Asterisk |
| 02:15.51 | weazahl | i do real good winging it. |
| 02:16.18 | weazahl | i know. the * is just going to provide dialtone for the magix |
| 02:16.21 | J4k3 | well, its more like $22/mo or so realistically... but still |
| 02:16.57 | Grnd-Wire | weazahl: So how are you plugging things in.. |
| 02:17.13 | Grnd-Wire | weazahl: hmm.. ok - In doing that you can't route a call to a specific phone (no trunking/direct inward dialing) |
| 02:17.14 | weazahl | they are paying about $2500 for phone and data, 328kb data... |
| 02:17.26 | weazahl | sure can. |
| 02:17.33 | weazahl | have tons of ports. |
| 02:17.52 | Grnd-Wire | weazahl: haha.. So you're going to assign private lines to each extension? |
| 02:18.12 | weazahl | for some, yes |
| 02:18.39 | Grnd-Wire | weazahl: You are better off learning how to use the 100DCD card that is in that system.. |
| 02:18.45 | weazahl | 40 stations, it has 4, will have 5 408s |
| 02:19.01 | weazahl | ok, im listening |
| 02:20.15 | Grnd-Wire | weazahl: That T1 board can be programmed to do PRI trunking.. Essentially talking to another phone switch over standard protocols.. Since you can send digits (caller ID as well as DNIS) .. not only can you send phone calls directly to an extension on the Asterisk machine (or whatever you program).. |
| 02:20.40 | Grnd-Wire | but you can tell Asterisk to send calls OUT the Zap channel group - and the Magix will send it to the CO.. |
| 02:20.58 | weazahl | oh, so i could use the ds1? |
| 02:21.18 | Grnd-Wire | weazahl: but.. you're in for a very long road if you don't know any part of what I am talking about.. PRI programming on a magix is about as much fun as a root canal. |
| 02:21.27 | weazahl | how do i interface that to the asterisk box? what card in the *? |
| 02:21.42 | Grnd-Wire | weazahl: Digium has cards.. I really like the Rhino R1T1 though.. |
| 02:21.51 | Grnd-Wire | weazahl: About $360 or so. |
| 02:22.43 | weazahl | see, that is what i need to pay someone like you to do, that part. |
| 02:22.59 | weazahl | that is the beauty of remote administration |
| 02:23.41 | Grnd-Wire | weazahl: yeah, well I wouldn't be the one to do it.. You need to checkout www.tek-tips.com .. There is a Legend/Magix forum there. Many people post there and you can see their contact info from their posts. |
| 02:24.18 | weazahl | so i need to read up on pri programming though. |
| 02:24.39 | Grnd-Wire | weazahl: yup.. and you get 23 conversations all happening over two pairs of wires.. |
| 02:24.55 | Grnd-Wire | Once it is programmed properly, nothing will work that well - and you'll actually look like you know what you're doing. |
| 02:25.52 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 02:28.23 | weazahl | sweet. well i can get the VM in there with 4 ports in the box real fast. |
| 02:29.59 | Grnd-Wire | weazahl: yes.. You use FXO ports on the asterisk.. plug it into four analog station ports on the magix.. |
| 02:31.18 | BBHoss | Grnd-Wire, so you like the Rhino T1 cards? |
| 02:31.32 | weazahl | the dialplans for asterisk to emulate MM are documented. |
| 02:31.37 | Grnd-Wire | BBHoss: I've got several in my labs.. |
| 02:31.50 | Grnd-Wire | weazahl: yup - you'll want to make some modifications, but the core stuff is there.. |
| 02:32.00 | BBHoss | Grnd-Wire, they are a good bit cheaper than digium cards, are there any trade-offs? |
| 02:32.33 | Grnd-Wire | BBHoss: hmm.. 5 year warranty, and the fact that their support team is GOOD, and will login to your machine to troubleshoot issues with you for the full five year warranty.. |
| 02:32.40 | Grnd-Wire | Even echo problems! |
| 02:32.42 | BBHoss | sounds great |
| 02:32.48 | Grnd-Wire | oh wait.. You wanted reasons to NOT buy a Rhino? :) |
| 02:32.49 | riddlebox | Grnd-Wire, PRI on a magix isnt too bad |
| 02:33.07 | riddlebox | well if you've done it a couple of times |
| 02:33.08 | BBHoss | actually, yeah usually those are the most important :) |
| 02:33.13 | Grnd-Wire | riddlebox: It wasn't nearly as easy as just doing E&M TIE over T1 |
| 02:33.46 | riddlebox | Grnd-Wire, true DID's are the worst if you dont have a numbering scheme that is easy to match with your extensions |
| 02:33.49 | Grnd-Wire | BBHoss: I honestly don't have any.. they don't have the type of chipset incompatibility issues as other brands.. their support will actually talk to you 30 days after the purchase. |
| 02:34.01 | BBHoss | cool |
| 02:34.13 | BBHoss | and the setup is the same as a digium card right? |
| 02:34.17 | Grnd-Wire | riddlebox: ya.. Which hopefully he will contact someone who knows Magix .. They'll keep him out of trouble, and make it freakin' rock.. |
| 02:34.19 | riddlebox | I may be getting a 4 port fxs card from my boss, he ordered the wrong card |
| 02:34.23 | weazahl | Grnd-Wire: thanks for all your help. you have me pointed the right direction now. |
| 02:34.32 | riddlebox | Grnd-Wire, I have already offered |
| 02:34.33 | Grnd-Wire | weazahl: sure.. |
| 02:34.50 | weazahl | riddlebox: looks like those are the guides i need. thanks |
| 02:35.16 | riddlebox | Grnd-Wire, are you an avaya business partner? |
| 02:35.22 | Grnd-Wire | riddlebox: no |
| 02:35.48 | riddlebox | I work for one, I think we are number 10 in illinois |
| 02:36.17 | Grnd-Wire | riddlebox: heh.. Then you are indeed qualified to offer.. |
| 02:36.49 | weazahl | riddlebox: im in missouri, boonville |
| 02:37.24 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 02:39.54 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 02:42.03 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 02:50.14 | *** join/#asterisk sacitec (n=tobi@189.129.149.83) |
| 02:53.17 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 02:53.26 | weazahl | Grnd-Wire: can i use that pri to do all the switching on asterisk? i.e. have the magix simply connect requested stations to a channel on the * box. park, hold, transfer is all done on *. |
| 02:53.45 | jameswf-home | trixbox ad supported gui :) http://img82.imageshack.us/img82/9436/spoofrs4.jpg |
| 02:54.01 | weazahl | this can be accomplished cant it? the easy way to do it possibly? |
| 02:56.04 | weazahl | wow, that makes this easy. |
| 02:56.27 | Grnd-Wire | weazahl: When done properly, yes - you can do that sorta stuff. |
| 02:57.35 | weazahl | hybrid pbx mode if i had to guess |
| 02:59.34 | weazahl | Grnd-Wire: Behind switch mode! |
| 03:01.21 | kyron | jameswf-home, LOL |
| 03:01.40 | Grnd-Wire | weazahl: hmm.. Behind switch mode doesn't really make alot of a difference.. That's for a Centrex setup. |
| 03:02.20 | Grnd-Wire | weazahl: The decision is Key or Hybrid/PBX .. and you're going to want to use hybrid PBX anything a T1/PRI trunk is involved. |
| 03:03.13 | *** join/#asterisk thomas_newbie__ (n=thomas@CPE0014bf493235-CM00140493ede8.cpe.net.cable.rogers.com) |
| 03:03.57 | weazahl | ok, ill read the book tommorow thanks to riddlebox for pointing me to it |
| 03:05.26 | sacitec | good night, i'm working with SIP remote extensions, g729, no NAT. I'm expecting problems with calls cut ( http://www.pastebin.org/17712 ) I've read that this issue is when asterisk stop reciving RTP packages after 20 seconds |
| 03:05.52 | [TK]D-Fender | sacitec, Disable VAD/CNG |
| 03:06.12 | sacitec | working with digium g729 comercial codec |
| 03:07.16 | tzanger | grr |
| 03:07.20 | tzanger | kid has a crossword to do |
| 03:07.27 | tzanger | wifey and I spent hte last hour on it instead of the boy |
| 03:07.32 | tzanger | I hate crosswords |
| 03:10.03 | tzanger | ugh this dsp stuff is killing me |
| 03:14.09 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 03:14.15 | jameswf-home | lol telemarketer just called.. I kept saying what they kept repeating them selves.... :)) |
| 03:14.36 | kyron | LOL |
| 03:15.09 | kyron | well, we keep getting the annoying "You've won a trip to the Carabeans" ...don't know what that one is all about.. |
| 03:15.19 | kyron | "Press 9 to collect your prize" |
| 03:15.26 | jameswf-home | it was a vacation something |
| 03:15.36 | kyron | ah...you 2 heh... |
| 03:16.05 | jameswf-home | When Its a guy i usualy accuse him of having an affair with my wife... that call can go for like an hour |
| 03:16.38 | kyron | BTW, installing * at home and was wondering, if I want to be able to put someone on HOLD and pick the call from another phone, how should I go about it, one phone is a poly 320 the others are all regular phones connected to a Mediatrix 1104... |
| 03:16.46 | Nivex | "The number you have reached: 9 1 1 is no longer in service. No further information about: 9 1 1 is available" |
| 03:16.49 | kyron | jameswf-home, LLOOOLLL |
| 03:16.56 | kyron | heheheh |
| 03:18.17 | kyron | sob0l, is parking the call the way to go? (have to figure out the mechanism)... Don't even know if a regular phone can "park" a call.. |
| 03:18.52 | kyron | The only alternative I see to this is SLA...which is not what I would want...unless I could do some sort of "SLA only when I press Hold" :P |
| 03:23.07 | kyron | Hey, what happens if I say "FreeSWITCH" in this channel? |
| 03:23.15 | kyron | ~FreeSWITCH |
| 03:23.26 | jbot | it has been said that freeswitch is an open source soft switch that is *not* a fork of asterisk http://www.freeswitch.org/interview2.htm |
| 03:23.26 | jameswf-home | ~freeswitch |
| 03:23.27 | jbot | freeswitch is, like, an open source soft switch that is *not* a fork of asterisk http://www.freeswitch.org/interview2.htm |
| 03:23.27 | jameswf-home | doh |
| 03:23.36 | jameswf-home | ~fork |
| 03:23.37 | jbot | spoon! |
| 03:23.47 | kyron | LOOOOOOOL |
| 03:24.12 | jameswf-home | ~spoon |
| 03:24.13 | jbot | There is no spoon. Ok, perhaps you meant: Parallel execution of command batches for SMP machines. URL: http://www.farcaster.net/xris/spoon/ |
| 03:24.26 | Inssomniak | I never thought to ask, but asterisk will forward caller-ID from my POTS line to my extensions of whos calling? |
| 03:24.28 | kyron | WOW, didn't know _that_ one! |
| 03:24.49 | kyron | eharris, doesn't exist apparently |
| 03:25.43 | jameswf-home | ~seen eharris |
| 03:26.09 | jbot | eharris is currently on #asterisk-doc (9h 45s) #asterisk (9h 45s). Has said a total of 31 messages. Is idling for 4h 32m 2s, last said: 'although it does seem to be in svn trunk, doesn't look like its in 1.4.x'. |
| 03:26.09 | BBHoss | kyron, i never could wrap my head around FreeSWITCH's config files |
| 03:26.10 | jameswf-home | xml bah |
| 03:26.20 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 03:26.42 | BBHoss | i mean its not a bad thing that its xml, but I never could get a straight answer out of anyone of how to set it up |
| 03:26.55 | kyron | jameswf-home, I am confused, I never wrote that, I wrote "eh, doesn't..." |
| 03:26.57 | kyron | weird... |
| 03:26.59 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 03:27.09 | Nivex | Victory is mine! |
| 03:27.20 | kyron | my xchat history agrees with me O_o |
| 03:27.36 | kyron | BBHoss, that can't be good... |
| 03:27.36 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:28.07 | BBHoss | they don't have near the community of asterisk though, so its hard to blame |
| 03:28.58 | *** join/#asterisk zobia (n=laurashr@222.212.64.63) |
| 03:29.12 | zobia | hello everyone |
| 03:29.33 | zobia | i am having problem with sccp phone with asterisk 1.4.17 |
| 03:29.45 | zobia | the sccp phone kept registering over and over |
| 03:29.58 | zobia | even it registered well. few mins later it will do it again |
| 03:30.22 | zobia | i just upgrade from 1.2.24 to 1.4.17 , 1.2.24 does not have the same problem |
| 03:30.24 | zobia | any idea? |
| 03:32.23 | kyron | BBHoss, guess the first one to come out with "something" gets the bid... kinda like Bill Gates, he was the "first" so all other projects, even if better, never got off to a start (OS/2, from what I heard, it was technically much better than Microsoft's Windows) |
| 03:33.07 | BBHoss | heh speaking of OS/2, i found an OLD copy of Warp version 3 the other day :) |
| 03:33.43 | BBHoss | zobia, is there no possible way to get a sip firmware for that phone? |
| 03:35.33 | kyron | BBHoss, I giggled when I saw my bank's automated machine boot up and show a big OS/2 Warp logo (actually, I was satisfied to see they weren't uselessly spending $$$ porting to Windows on such a closed network...that would be quite a waste of $$$) |
| 03:36.01 | BBHoss | now if it was linux it would have been sweet |
| 03:36.08 | BBHoss | or like openbsd or something |
| 03:36.12 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 03:36.22 | [TK]D-Fender | tzanger, Yo... just mapped up where you were staying.. you kept saying "Verdun", but its actually on Ile-Des-Soeurs. So you Brossard ISN"t such a bad idea, but a car is still not a bad deal. 220 vs 10 @ 20$ easily, and you're going to want to get up early. |
| 03:36.22 | *** join/#asterisk simb2 (n=mooserfu@i-195-137-39-237.freedom2surf.net) |
| 03:36.58 | tzanger | [TK]D-Fender: ahh, sounds like fun... |
| 03:37.11 | kyron | ok, night time (early today..) laters! |
| 03:37.14 | tzanger | yeah ile-des-soeurs is where that place is |
| 03:37.30 | tzanger | which is easier, travelling to ile-des-soeurs from montreal, or from brossard? |
| 03:37.32 | [TK]D-Fender | tzanger, Something like that.... you're just off-shore I thin they conglomerated it for naming... but yeah, you're practically working ON the bridge :) |
| 03:37.34 | kyron | WTF... |
| 03:37.34 | tzanger | I imagine they both suck |
| 03:37.43 | kyron | I still say Digium should move north |
| 03:37.47 | [TK]D-Fender | tzanger, Probably about as bad either way |
| 03:37.53 | tzanger | ahh |
| 03:38.05 | kyron | tzafrir, that's where I used to work (Prima--Elix--now Bell I guess) |
| 03:38.06 | kyron | :P |
| 03:39.25 | *** join/#asterisk techie (n=techie@adsl-76-240-176-254.dsl.lsan03.sbcglobal.net) |
| 03:40.42 | simb2 | hello |
| 03:41.50 | sacitec | [TK]D-Fender: i already disabled silence suppression on hardphone(SPA901) and still same situation, also i'm having one way audio, but i'm having no NAT. My * asterisk box is on DMZ so as hardphone(SPA901) on remote network |
| 03:42.28 | [TK]D-Fender | sacitec, DMZ is not good. |
| 03:42.58 | [TK]D-Fender | sacitec, Phone should not have ports forwarded to it, and * need a bunch of settings if it isn't on a public IP. |
| 03:42.58 | [TK]D-Fender | ~sipnat |
| 03:42.59 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:42.59 | sacitec | just port forwarding for both sides |
| 03:42.59 | [TK]D-Fender | ^^^^ read up |
| 03:43.14 | [TK]D-Fender | sacitec, you should NOT be port forwarding both sides. Go read the guide |
| 03:43.21 | zobia | BBBoss . it's possible to make it sip. but we already use it in 1.2 as sccp |
| 03:43.35 | sacitec | what's a good option for STUN server, running on centos 5 |
| 03:44.23 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 03:44.28 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 03:44.37 | BBHoss | zobia, IMHO, sccp support in * suckzorz, move to SIP at any cost, and avoid headaches :) |
| 03:45.15 | zobia | BBHoss: thank you for your advice. i will try |
| 03:45.17 | simb2 | We have a generally working asterisk system (Cisco 7940 sip <--> asterisk 1.4.11 <--> SIP trunk), but some calls (less than half, say) are dropping to one-way audio after a few minutes. No specific times, people or numbers that I can find, so I can't make it happen on demand... Any ideas what I ought to look at to troubleshoot further, please? |
| 03:46.00 | BBHoss | simb2, can you reproduce it? |
| 03:46.27 | simb2 | not at will |
| 03:46.30 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:47.09 | simb2 | is there anything (SIP, RTP?) that gets re-negotiated mid-call? |
| 03:47.18 | BBHoss | maybe a sip reinvite |
| 03:48.05 | BBHoss | is asterisk behind NAT? |
| 03:48.25 | simb2 | yes |
| 03:48.42 | BBHoss | what kind of NAT |
| 03:49.06 | BBHoss | or router type |
| 03:49.28 | simb2 | Er, ports forwarded through a Watchguard Firebox |
| 03:50.00 | BBHoss | try putting a canreinvite=no in your sip.conf for each 7940 |
| 03:50.32 | simb2 | I believe I have canreinvite=no in sip general settings - to push all calls through asterisk for CDR purposes (will just check that) |
| 03:51.45 | BBHoss | also make sure you have the NAT settings correct, externip, localnet, etc |
| 03:51.54 | simb2 | or maybe I wont as my vpn access needs reworking. :/ |
| 03:52.25 | BBHoss | where does the vpn fit in this |
| 03:52.26 | tzanger | [TK]D-Fender: thanks for the assistance... I now have to make a decision regarding the hotel :-) |
| 03:52.34 | Inssomniak | Can a fax come in on a POTS line, asterisk sees it as a fax, and sent it out a specific "extension" (which would have an ATA on it), (does that even work?) |
| 03:52.48 | simb2 | I had some problems with NAT when setting it up and spent a while playing with externip and such. After incoming and outgoing calls work OK I assumed that must be right... all calls seem to connect fine initially |
| 03:53.19 | BBHoss | Inssomniak, yeah |
| 03:53.38 | BBHoss | Inssomniak, turn faxdetect=incoming on in zapata.conf |
| 03:54.04 | BBHoss | <PROTECTED> |
| 03:54.09 | BBHoss | wrong link |
| 03:54.15 | BBHoss | http://www.voip-info.org/wiki-Asterisk+fax#Zapfaxdetection |
| 03:54.17 | BBHoss | there |
| 03:54.20 | Inssomniak | thx! |
| 03:54.38 | *** join/#asterisk AndyGraybeal (n=andy@node191.37.251.72.1dial.com) |
| 03:54.39 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 03:54.54 | tzanger | [TK]D-Fender: coming down 10 or 20 would be th ebetter commute? |
| 03:55.00 | BBHoss | basically asterisk will throw the call into your incoming context, with the "fax" number |
| 03:55.07 | [TK]D-Fender | tzanger, the price is right, it is "close", and you're screwed either way... I'd say you aren't too far out... |
| 03:56.05 | [TK]D-Fender | tzanger, you mean coming in to town? I'm not sure how to do it south of montreal. |
| 03:56.14 | [TK]D-Fender | tzanger, What time would you be passing through? |
| 03:56.28 | tzanger | no, I mean from the hotel to the island every morning and back at night; it's probably 6 of one, a half dozen of the other |
| 03:56.34 | tzanger | probably onsite at 8am |
| 03:59.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:59.19 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:00.30 | Inssomniak | Now Im kinda worried., I send a lot of faxes. sending a fax -> ATA -> thru asterisk -> PSTN.. is it hit and miss? |
| 04:02.24 | [TK]D-Fender | Inssomniak, You put sip in the middle. |
| 04:02.27 | [TK]D-Fender | ~wglwat |
| 04:02.28 | jbot | methinks wglwat is well, good luck with all that |
| 04:02.29 | [TK]D-Fender | ^^ |
| 04:02.48 | Inssomniak | eh? |
| 04:02.49 | [TK]D-Fender | Inssomniak, and included a "mystery" means of reaching the PSTN as well |
| 04:03.22 | Inssomniak | I reach the PSTN directly from a spa 3102 in my house |
| 04:03.26 | JT | err |
| 04:03.29 | JT | you meant to say |
| 04:03.41 | JT | "I reack the PSTN via a POTS line connected to an ATA" |
| 04:03.43 | JT | reach |
| 04:03.51 | kyron | tzafrir, where are-you going on IDS? |
| 04:04.21 | tzafrir | IDS? |
| 04:04.24 | Inssomniak | yea, something like that. so I dont reach the PSTN thru a voip provider |
| 04:04.27 | kyron | îles des soeurs |
| 04:04.49 | [TK]D-Fender | Inssomniak, Forget that then, to much crap going on will screw your faxes. |
| 04:04.50 | JT | Inssomniak: there are a billion ways to reach the pstn |
| 04:05.10 | JT | Inssomniak: POTS, PRI, channelised T1, BRI, STM-1, VoIP |
| 04:05.13 | JT | GSM |
| 04:05.14 | JT | etc |
| 04:05.18 | Inssomniak | POTS sorry |
| 04:05.22 | kyron | if you're donwtown and going to IDS, there is a bus that will get you there in no time quite cheaper than a cab. You can easily cab back off hours. |
| 04:05.22 | Inssomniak | lol Im still new |
| 04:05.29 | simb2 | could I get asterisk to send emails at the end of phone calls? e.g. "reminder: bill 12 minutes on the phone to 555-12345 (A. name)" |
| 04:05.45 | tzafrir | kyron, you meant tzanger , I guess |
| 04:05.52 | [TK]D-Fender | simb2, Yes. |
| 04:06.00 | kyron | tzafrir, ah craps...yeah |
| 04:06.01 | simb2 | and would I need to work from the manager interface, or could it trigger from the dialplan, or some other way? |
| 04:06.05 | Inssomniak | Ill try to reword it :) |
| 04:06.14 | kyron | tzanger, wht I just told tzafrir |
| 04:06.16 | kyron | ;) |
| 04:06.27 | tzanger | eh? |
| 04:06.37 | tzafrir | <kyron> tzafrir, where are-you going on IDS? |
| 04:06.51 | [TK]D-Fender | simb2, so many ways to do it. You can put it at the end of your dialplan as part of the cleanup, etc. |
| 04:06.55 | tzanger | kyron: I'm going to Touchtunes (3 commerce drive or something like that) |
| 04:06.57 | Inssomniak | fax outgoing -> ATA -> SIP ->Asterisk -> SPA 3102 -> POTS |
| 04:07.06 | tzanger | kyron: trying to determine which cheap hotel to go to |
| 04:07.37 | JT | Inssomniak: all on a lan? |
| 04:07.50 | Inssomniak | yea JT it never leaves my house |
| 04:08.00 | kyron | downtown might not be cheap but I was saying you can easily commute using the bus, I used to work there and there is one going there quite often during regular hours |
| 04:08.01 | tzanger | was thinking motel rideau |
| 04:08.21 | kyron | have to go... maybe catch up tomorrow ;) |
| 04:08.23 | JT | Inssomniak: it might work, or it might not |
| 04:08.26 | tzanger | but that's in brossard... |
| 04:08.29 | tzanger | kyron: thanks! |
| 04:08.33 | JT | i mean being on the lan means it stands a chance ;) |
| 04:08.53 | kyron | tzanger, prolly best bet is downtown, cauz verdun transit might be hell ;) |
| 04:08.56 | BBHoss | Inssomniak, it should work well enough, just make sure you use ulaw |
| 04:08.57 | simb2 | [TK]D-Fender: hmm, ok. Is the call duration anywhere accessible, do you know? |
| 04:09.08 | *** join/#asterisk hades123 (n=wqwsqww@d57-199-17.home.cgocable.net) |
| 04:09.11 | tzanger | kyron: :-) $40/night though is what I heard in broken english |
| 04:09.16 | tzanger | if I was a "worker" |
| 04:09.19 | simb2 | or will I have to track it / pick it up from the CDR logs |
| 04:09.20 | BBHoss | simb2, you could pull it out of the astdb |
| 04:09.35 | BBHoss | from cdr |
| 04:09.37 | Inssomniak | is there any other options or suggestions I can explore besides tying my fax to the POTS line before the FXO box? |
| 04:10.13 | BBHoss | Inssomniak, you can hook it up to an ATA, you may have a few problems, then you might not |
| 04:10.31 | *** join/#asterisk nixbox (i=oh@24.175.74.160) |
| 04:10.34 | nixbox | hi |
| 04:10.44 | hades123 | did any body test the performance of asterisk on multi core, and multi procs system ? does it scale well? |
| 04:11.11 | nixbox | i want to setup a VoIP gateway using asterisk. I need to know the requirements for that, both hardware and software (anything other than asterisk)? |
| 04:11.37 | JT | nixbox: the gateway will be doing what, exactly? |
| 04:11.51 | mosty | hades123, i hit deadlock in asterisk before i hit the limits of smp |
| 04:11.55 | mosty | deadlocks |
| 04:12.12 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 04:12.29 | mosty | nixbox, depends what you want to be a gateway for. if it's pure voip then you just need a pc with some network cards |
| 04:12.29 | JT | asterisk likes deadlocks |
| 04:12.38 | *** join/#asterisk b11d|bbl (n=no@234-200-29-134.hcc.mnscu.edu) |
| 04:12.50 | b11d|bbl | hello chaps |
| 04:12.56 | russellb | i'm happy to fix any deadlock that people find |
| 04:12.59 | b11d\ | damnit |
| 04:13.08 | russellb | they're fun to fix, and we have tools that make them easy to debug in recent versions |
| 04:13.37 | mosty | russellb, what tools? my main problem is i need some good load testing utilities for iax and sip |
| 04:13.54 | nixbox | mosty, i live in US, i have found a provider who can give me a number in my home country, so that when people in my home country call me, i can receive calls in the US, but they are charging for providing hardware which is a voip gateway, how can i replace that hardware with asterisk? what sort of setup should i do? |
| 04:14.29 | mosty | nixbox, if the provider talks iax or (more likely) sip then you don't need any special hardware in the asterisk box |
| 04:14.32 | hades123 | russellb: I was asking : did any body test the performance of asterisk on multi core, and multi procs system ? does it scale well? |
| 04:14.41 | hades123 | can you help with that? |
| 04:14.51 | russellb | no, i can't really help with that ... |
| 04:15.08 | hades123 | oh bummer |
| 04:15.12 | russellb | mosty: well, debugging and fixing deadlocks doesn't really have anything to do with load testing tools :) |
| 04:15.16 | hades123 | :'( |
| 04:15.27 | russellb | i was just making a general comment that if you have a deadlock, i would be happy to fix it |
| 04:15.47 | mosty | russellb, hehe well my problem is that my deadlock appears under high load which i'm unable to produce (yet) in a test environment |
| 04:15.52 | nixbox | mosty, hmmm can i find any pointers as to how to setup asterisk for that purpose, moreover how can i make my life easy by somehow interfacing a phone with the asterisk PC so that i can receive calls on the phone? |
| 04:16.17 | russellb | mosty: ah ... well, the easiest thing to do usually is to use asterisk as your load generating platform |
| 04:16.26 | mosty | nixbox, you would probably want to get an ATA to plug in an analogue telephone, or even better a real sip phone |
| 04:16.29 | russellb | mosty: even if you have to use multiple servers to load up the one server under test |
| 04:16.49 | mosty | russellb, that's what i will be doing, i just don't have enough free servers to do that yet |
| 04:17.02 | russellb | gotcha. |
| 04:17.26 | hades123 | ok ok , one tiny question ... does astrisk PBX have to work as B2BUA , or can we have media go between the two internal callers |
| 04:17.27 | nixbox | mosty, if my provider does not speak SIP, at most what sort of hardware would i be needing for asterisk to be setup? |
| 04:17.33 | hades123 | bypassing |
| 04:17.43 | hades123 | asterisk once call is established |
| 04:17.57 | JT | hades123: asterisk always acts as a B2BUA |
| 04:18.01 | mosty | nixbox, impossible to tell. find out more about their hardware |
| 04:18.02 | BBHoss | nixbox, what country |
| 04:18.02 | JT | however end points can send RTP directly |
| 04:18.27 | hades123 | you mean it;s more on the SIP phone side |
| 04:18.32 | hades123 | itself |
| 04:18.33 | nixbox | BBHoss, US |
| 04:18.52 | hades123 | not asterisk? |
| 04:18.56 | BBHoss | nixbox, what country do you want the telephone number in |
| 04:19.07 | nixbox | BBHoss, Pakistan |
| 04:19.11 | BBHoss | ok hang on |
| 04:19.17 | nixbox | BBHoss, ok |
| 04:19.59 | JT | hades123: err what? |
| 04:20.12 | drmessano | jblack, you around? |
| 04:20.25 | BBHoss | nixbox, didww.us has pakistan dids, $22.00 for unlimited |
| 04:20.33 | jameswf-home | drmessano: did you see the pic i made |
| 04:20.37 | drmessano | Nope |
| 04:20.42 | drmessano | been at WALLLY WORLDDDD |
| 04:20.46 | drmessano | Link me |
| 04:20.47 | b11d | jameswf-home.. can you help me with that zap problem again? |
| 04:20.54 | nixbox | BBHoss, ok thanks, let me take a look |
| 04:21.01 | hades123 | JT, never mind my friend, I am trying to see if asterisk can handle 400 - 500 concurrent calls |
| 04:21.04 | hades123 | on one box |
| 04:21.12 | hades123 | I have read alot and can't seem |
| 04:21.13 | drmessano | HAHAH!!!!! FTW!!! |
| 04:21.18 | hades123 | to fiind an answer |
| 04:22.30 | mosty | hades123, depends what you're doing with those calls |
| 04:22.33 | nixbox | BBHoss, i should be selecting the Phone-to-IP PBX option, right? |
| 04:23.04 | hades123 | mosty: 90% calls between internal (fxs ) stations |
| 04:23.06 | hades123 | sorry |
| 04:23.10 | hades123 | SIP |
| 04:23.16 | hades123 | but internal |
| 04:23.17 | BBHoss | nixbox, just go to buy did |
| 04:23.26 | mosty | hades123, little/no transcoding? |
| 04:23.41 | hades123 | mosty: Almost none |
| 04:24.18 | BBHoss | nixbox, its 2 channels, unlimited |
| 04:24.25 | mosty | hades123, asterisk might be ok, openser + asterisk could definitely |
| 04:24.33 | hades123 | mosty: it's a big building , poeple calle ach other most of the time |
| 04:24.43 | hades123 | then may be 5 % of the calls go to pstn |
| 04:25.11 | mosty | hades123, what codec internally, and how would you connect to the pstn? |
| 04:25.26 | mosty | and do you have to fight through nat? |
| 04:25.59 | hades123 | mosty: PSTN will be a T1/E1 line , NO NAT no users outside that building will need VOIP |
| 04:26.32 | hades123 | mosty: codec, anything, it's 100mbit conneciton to each user , so the easiest on the CPU |
| 04:26.39 | mosty | well then you can use canreinvite=yes for your sip-sip calls, and reduce load |
| 04:26.56 | mosty | doesn't matter what codec you use if it's the same on all phones |
| 04:27.39 | hades123 | ok, now re-invites, if during th call, the person want to do call forward |
| 04:28.04 | hades123 | would that station still be be send signaling to the asterisk |
| 04:28.09 | hades123 | box for that ? |
| 04:28.33 | nixbox | BBHoss, once i get the DID, what is the next set of steps? |
| 04:29.28 | hades123 | mosty: in other words, what do you loos when you use re-invites? |
| 04:29.59 | mosty | hades123, you don't have a record of all call durations on asterisk, and it breaks with nat |
| 04:30.22 | mosty | because the phones can speak to each other directly, asterisk doesn't know how long two phones are connected to each other |
| 04:30.48 | hades123 | that I understand, and it's totally fine. |
| 04:30.53 | JT | mosty: it matters what codec you use |
| 04:30.57 | JT | no reason not to use g.711 |
| 04:31.01 | JT | g.729 sounds like shit\ |
| 04:31.14 | JT | well all compressed codecs do |
| 04:31.17 | x86 | mosty: problem with re-invites is the server has no control... supervisor can no longer spy on employees, you can not have accurate CDR records, etc, etc... |
| 04:31.31 | x86 | it becomes no longer manageable when you do sip-to-sip |
| 04:31.34 | JT | yeah forget abour reinvites |
| 04:31.36 | JT | pointless |
| 04:31.44 | hades123 | I don't need spying |
| 04:31.47 | hades123 | I don't need CDR |
| 04:31.50 | hades123 | it's all internal |
| 04:31.58 | x86 | sounds like a porn |
| 04:32.06 | mosty | hades123, well you can always play with turning canreinvite on later |
| 04:32.20 | nixbox | BBHoss, there? |
| 04:32.22 | mosty | JT, g729 sounds ok on some devices |
| 04:32.26 | hades123 | it's like a hotel actually |
| 04:32.40 | hades123 | guests call operator/ laundry service |
| 04:32.45 | hades123 | crap like that |
| 04:33.11 | hades123 | and every blue moon, the guest calls outside |
| 04:33.46 | hades123 | so I want those small internal calls, to bypass asterisk, however, still be able to do basic stuff like, hold/ transfer |
| 04:35.34 | hades123 | i was that boring !!! :) I can hear u snore guys |
| 04:38.18 | nixbox | mosty, once i get a DID number for my country, what is the next set of steps i should do to configure Asterisk as a VoIP gateway? Pointers to any URLs? thanks |
| 04:40.32 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:42.16 | jblack | nixbox: Setup up the account in either sip.conf or iax.conf. Then, setup phones (either hard or soft) in sip.conf. Then, set up routing for extensions in your dialplan configuration |
| 04:43.08 | drmessano | there you are |
| 04:43.19 | nixbox | jblack, thanks |
| 04:43.58 | *** join/#asterisk ajunge (n=ajunge@200.119.238.199) |
| 04:44.12 | jblack | drmessano: You were looking for me? |
| 04:44.29 | jblack | I wasn't doing anything. You coulda called. |
| 04:46.27 | JunK-Y | cause it will go down tomorrow? :P |
| 04:46.46 | JT | mosty: it will never ever sound as good as g.711 |
| 04:46.50 | JT | that's impossible |
| 04:47.13 | mosty | i didn't say it was possible |
| 04:47.35 | mosty | nixbox, read the book |
| 04:47.37 | mosty | ~book |
| 04:47.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 04:47.40 | jblack | Nothing's impossible. |
| 04:47.43 | mosty | nixbox, of just buy an ATA |
| 04:47.58 | mosty | s/of/or/ |
| 04:48.02 | JT | jblack: magic is |
| 04:48.47 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:50.49 | jblack | JT: Magic can't be disproved for exactly the same reasons god can't be disproved. |
| 04:52.12 | jameswf-home | wow ummmm no |
| 04:52.33 | jameswf-home | s/[a-z]/x/g |
| 04:52.44 | jameswf-home | damn |
| 04:52.58 | jblack | Absolutely true. |
| 04:53.11 | drmessano | brb.. trying to multi-not-task |
| 05:04.10 | JT | jblack: err, facts need to be PROVEN, not disproven |
| 05:04.11 | JT | crazy |
| 05:04.36 | jblack | No. Just freshman philosophy. |
| 05:06.37 | jblack | For the least, be very, very careful of what you call a fact. |
| 05:08.09 | jblack | The best science we have indicates that everything we see, hear, and understand are our perception of a crapshoot. |
| 05:08.31 | JT | the existence of god stands very far outside the the relm of something which can possibly be called a fact |
| 05:08.54 | JT | similar to ideas about compressed codecs magically sounding better than uncompressed codecs |
| 05:09.11 | mosty | JT, nobody said they did |
| 05:09.13 | *** join/#asterisk erojasv (n=erojasv@190.43.244.244) |
| 05:09.15 | jameswf-home | ~fax |
| 05:09.16 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 05:09.23 | jameswf-home | ~faxing |
| 05:09.24 | jbot | i guess faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo |
| 05:09.32 | erojasv | ~iax2 |
| 05:09.33 | jbot | extra, extra, read all about it, iax2 is http://www.voip-info.org/wiki-IAX |
| 05:09.44 | jameswf-home | ~god |
| 05:09.44 | jbot | hmm... god is a llama, or real unless declared integer |
| 05:09.55 | *** join/#asterisk mf2 (n=imcbride@shell.convoke.com) |
| 05:10.30 | jameswf-home | ~llama |
| 05:10.31 | jbot | i heard llama is the incarnation of god on earth, or http://images2.jokaroo.net/flash/llamasong.swf |
| 05:11.01 | JT | mosty: then jblack says nothing's impossible ;) |
| 05:12.56 | jblack | Don't blame me for quantum theory! I'd prefer a deterministic existance too. |
| 05:13.25 | jameswf-home | ~jblack |
| 05:13.26 | jameswf-home | ~impossible |
| 05:18.56 | UnixDog | <PROTECTED> |
| 05:19.54 | jameswf-home | ~clear |
| 05:20.05 | *** part/#asterisk techie (n=techie@adsl-76-240-176-254.dsl.lsan03.sbcglobal.net) |
| 05:20.12 | jameswf-home | jbot: roll over |
| 05:20.13 | jbot | ACTION stuffs over in a hamster ball and rolls it down a steep, snow-covered mountain |
| 05:23.31 | b11d | goodnight all |
| 05:23.37 | b11d | DOH' |
| 05:35.31 | jameswf-home | lol http://www.youtube.com/watch?v=5fda4_wo6JI |
| 05:35.31 | *** join/#asterisk nixbox (i=oh@24.175.74.160) |
| 05:36.02 | nixbox | do i need a sound card in my asterisk PC if i would be receiving calls on some other pc on the network? |
| 05:36.09 | JT | no |
| 05:37.01 | nixbox | ok good |
| 05:37.34 | nixbox | JT, and if i receive calls on an analog telephone connected to an asterisk pc via ATA? |
| 05:38.09 | mosty | nixbox, have you read the book yet? |
| 05:38.28 | nixbox | mosty, no, i was just curious :P |
| 05:38.35 | nixbox | mosty, i have just downloaded it |
| 05:39.00 | mosty | maybe you should browse through it before asking really basic questions here |
| 05:39.56 | nixbox | mosty, ok, sorry for that. |
| 05:41.22 | mosty | it's quite a good book btw |
| 05:46.04 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 05:46.42 | JT | nixbox: if audio is going in and out of your sounds card, then you need it |
| 05:46.44 | JT | otherwise not |
| 05:46.48 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 05:48.26 | drmessano | back, I guess |
| 05:49.26 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7f513103c7166273) |
| 05:50.58 | drmessano | !ping |
| 05:55.07 | drmessano | jblack |
| 05:56.57 | jblack | yeah? |
| 05:57.48 | jblack | drmessano: WHAT!?!? |
| 05:59.24 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 06:01.50 | drmessano | Ummm... hang on |
| 06:02.04 | jblack | Heh. |
| 06:02.23 | jblack | "Come here come here come here!" "Ok, I'm here". "Thank you for coming. Please hold" |
| 06:03.36 | drmessano | I got Gizmo going to my IVR now |
| 06:03.44 | jblack | sipphone? |
| 06:03.58 | jblack | Nice! Did it involve magic? |
| 06:04.18 | drmessano | HA |
| 06:04.19 | drmessano | ues |
| 06:04.20 | drmessano | yes |
| 06:04.29 | drmessano | Lemme whittle it down |
| 06:04.32 | drmessano | I added 4 or 5 settings |
| 06:06.20 | *** join/#asterisk hmm-home (n=Neg@24-119-176-74.cpe.cableone.net) |
| 06:07.27 | drmessano | Ok |
| 06:07.46 | jblack | ok |
| 06:09.30 | drmessano | insecure=port, invite fixed part of it |
| 06:09.59 | jblack | Ok |
| 06:10.26 | jblack | makese sense why a pap2 wouldn't care about those |
| 06:10.33 | drmessano | But |
| 06:10.37 | drmessano | Its going to s |
| 06:11.17 | jblack | It's not going to an extension? |
| 06:11.24 | drmessano | newp |
| 06:12.13 | jblack | so we'll need distinct dialplan contexts for each sipphone line |
| 06:12.33 | jblack | I don't get it. I thought you tried insecure=port,invite yesterday, if not the day before |
| 06:13.44 | drmessano | WELL |
| 06:14.12 | drmessano | Did try it |
| 06:14.21 | drmessano | Maybe we're fighting service issues too? |
| 06:14.35 | jblack | It's always possible, yeah... |
| 06:14.55 | jblack | There's one way to make sure it's just a case of insecure=port,invite. |
| 06:15.06 | jblack | I haven't changed anything here. if that's really the issue, then ... |
| 06:15.38 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 06:16.00 | jblack | want to try dialing 17 472 768 946 ? |
| 06:17.12 | drmessano | gimme 2 mins.. im trying to see if I can get his how I want it |
| 06:17.47 | jblack | Take 5. I want a smoke |
| 06:18.08 | *** join/#asterisk AndyGraybeal_ (n=andy@node82.35.251.72.1dial.com) |
| 06:21.58 | jblack | back |
| 06:24.53 | *** join/#asterisk UnixDog (n=unixdog@ppp-69-238-217-105.dsl.irvnca.pacbell.net) |
| 06:30.47 | nixbox | i have just edited extensions.conf, when i do "dialplan reload" on CLI, it says no such command? |
| 06:31.33 | mosty | it's called "extensions reload" in some versions of asterisk |
| 06:31.41 | BBHoss | you using 1.2? |
| 06:32.53 | nixbox | yeah 1.2.13 |
| 06:33.00 | nixbox | extensions reload worked |
| 06:33.13 | BBHoss | why not 1.4, 1.2 is in a security-fix only mode |
| 06:33.13 | nixbox | whats the alternative for "dialplan show" |
| 06:34.32 | mosty | show dialplan |
| 06:34.47 | nixbox | BBHoss, running debain, got asterisk using apt-get |
| 06:34.55 | BBHoss | nixbox, not a wise idea |
| 06:35.00 | mosty | 1.2 is still more reliable than 1.4 |
| 06:35.19 | BBHoss | either way, running from apt is risky |
| 06:35.31 | nixbox | ahan |
| 06:36.51 | mosty | bbhoss: risky how? debian has a good security team |
| 06:37.01 | alrs | nixbox: Asterisk as packaged in Debian works great. |
| 06:37.06 | BBHoss | great example, the version you're using isnt patched for the chan_sip.c exploit |
| 06:38.00 | BBHoss | also AST-2007-027 |
| 06:39.23 | BBHoss | its not that debian is insecure, its that they are not released as often |
| 06:40.05 | alrs | BBHoss: this chan_sip exploit? http://www.debian.org/security/2007/dsa-1358 |
| 06:46.28 | scooby2 | whats the best way to do this? If the extension is between 800 and 899 i want a different context. Would something that be possible with an IF? |
| 06:46.36 | mosty | BBHoss, it's risky to run testing or unstable, sure |
| 06:47.08 | BBHoss | well 1.2.26.2-netsec was released the 22nd, so you are using a release thats nearly 15 months old |
| 06:47.12 | mosty | scooby2, you could use GotoIf, or you could use extension patterns with Goto |
| 06:47.41 | mosty | BBHoss, 15 months old + security backports |
| 06:50.11 | alrs | BBHoss: The version of 1.2 in Etch dates back to November 29 |
| 06:52.35 | scooby2 | mosty: with that I am still stuck how to determine if $EXT is >=800 and <900 |
| 06:52.40 | BBHoss | ok then there have been only 2 advisories sense then |
| 06:52.47 | BBHoss | since |
| 06:53.37 | mosty | scooby2, _8XX |
| 06:53.53 | scooby2 | doh |
| 06:54.13 | nixbox | i have setup X-lite softphone with asterisk, i am trying to test the echo test configuration straight out of the book, the phone registers with asterisk as i can see it on the asterisk CLI, but when i call 500 (the echo test application extension) the display shows calling, and then nothing happens |
| 06:54.25 | BBHoss | dont get me wrong, i love debian, i've just always been told to use svn or tarballs instead of apt-get |
| 06:56.08 | nixbox | on the CLI it says, -- Executing Verbose("SIP/qasim-0818c4c0", "1|Echo test application") in new stack |
| 06:56.46 | BBHoss | do you have the sound settings (mic, volume) etc correct? |
| 06:56.53 | nixbox | yeah |
| 06:57.06 | nixbox | in any case shouldnt it somehow show the call to be connected? |
| 06:57.13 | nixbox | because the display says "calling" |
| 06:57.17 | BBHoss | hmm |
| 06:57.18 | nixbox | and it stays as is |
| 06:57.33 | BBHoss | type sip show channels |
| 06:57.57 | nixbox | 0 active sip channels |
| 06:58.09 | BBHoss | maybe something else is fscked |
| 06:58.29 | BBHoss | turn on sip debug, then pastebin the results of a call |
| 06:58.31 | BBHoss | ~pb |
| 06:58.32 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 06:58.48 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:59.08 | nixbox | BBHoss, i just got some warnings |
| 06:59.25 | BBHoss | with sip debug on, you will get a lot more than warnings |
| 06:59.44 | nixbox | BBHoss, something like unable to spawn mp3player |
| 06:59.51 | BBHoss | ahh |
| 07:00.21 | BBHoss | either a permissions problem or mp3player isnt installed |
| 07:00.43 | BBHoss | i didnt think it used that to play the intro though |
| 07:00.46 | BBHoss | pastebin it |
| 07:01.01 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 07:02.13 | nixbox | BBHoss, http://pastebin.ca/887324 |
| 07:02.29 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 07:02.35 | mosty | BBHoss, the problem with using regular asterisk releases is that there are often changes in functionality within major versions |
| 07:02.44 | BBHoss | no shit |
| 07:03.19 | BBHoss | nixbox, is that all you got from the console? |
| 07:03.35 | BBHoss | do a set verbose 10 |
| 07:03.39 | hades123 | tspeaking about that the official release says it's tested for up to 200 or 250 concurrent calls |
| 07:03.44 | BBHoss | try it again and post the whole thing |
| 07:03.46 | *** join/#asterisk bmg505 (n=leon@196.209.181.41) |
| 07:03.47 | hades123 | but doesn't say on what machine specs |
| 07:04.39 | hades123 | does any one know what were the test config. |
| 07:04.58 | hades123 | (codecs/ transcoding/ machine specs, etc) |
| 07:05.38 | BBHoss | well they don't want to tell you that, otherwise there would be no incentive to buy ABE :) |
| 07:07.07 | BBHoss | but it probably means it can handle that if the system can handle it, with guarantee of no deadlocks |
| 07:07.41 | nixbox | BBHoss, http://paste.uni.cc/18240 |
| 07:07.51 | *** join/#asterisk Xen^ (i=L_NUX@unaffiliated/lnux/x-10290) |
| 07:08.34 | hades123 | BBHoss, I don't see what's the relation between giving me the test information, and me buying it |
| 07:08.53 | hades123 | if it's anything, it will deter me from bying, because I have less information |
| 07:09.02 | hades123 | which is actually the case right now |
| 07:09.18 | hades123 | I wanna buy it, however I have no clue, till now, what machine specs, can run what |
| 07:09.21 | BBHoss | hades123, they have a special verification and testing system for ABE, which is the most significant difference |
| 07:09.35 | BBHoss | hades123, how many users will you possibly have? |
| 07:09.43 | hades123 | 1500 |
| 07:09.46 | BBHoss | heh |
| 07:09.53 | hades123 | concurrent calls |
| 07:09.56 | hades123 | 200 - 400 |
| 07:10.08 | BBHoss | what you want to do is use OpenSER to load-balance asterisk |
| 07:10.09 | hades123 | 50 PSTN |
| 07:10.11 | hades123 | 350 internal |
| 07:10.16 | BBHoss | hmm |
| 07:10.24 | BBHoss | no transcoding for internal? |
| 07:10.29 | hades123 | nope |
| 07:10.33 | BBHoss | well then |
| 07:10.51 | BBHoss | you may be able to get away with that with 1 box |
| 07:10.59 | BBHoss | well, ill say its probable |
| 07:11.13 | hades123 | I was saying earlier I am considering, using re-invites |
| 07:11.15 | BBHoss | i mean all the asterisk server will be doing is 50, so... |
| 07:11.20 | hades123 | to get the RTP away from asterisk |
| 07:11.20 | BBHoss | exactly |
| 07:11.31 | hades123 | however, no one was able to tell me |
| 07:11.41 | hades123 | if in th emiddle of the conversation |
| 07:11.47 | hades123 | somebody tried to call forward |
| 07:11.58 | hades123 | well asterisk still be able to deal with that |
| 07:12.00 | BBHoss | i would suggest calling digium sales, i'm sure they will be MORE than happy to TELL you |
| 07:12.25 | BBHoss | but im always skeptical of everything |
| 07:12.33 | nixbox | brb |
| 07:12.39 | hades123 | I am in the process of collectin as much information as I can, I will eventually contact digium |
| 07:12.51 | BBHoss | but i know if they sell 4 port E1/T1 PRI cards, then one box should be able to do it |
| 07:13.07 | hades123 | that's what I told my self |
| 07:13.23 | hades123 | I am ready to go as far as a dual , quad core |
| 07:13.28 | hades123 | 3.0 GHZ Xeons |
| 07:13.35 | BBHoss | lemme look at some data |
| 07:13.38 | hades123 | the V8 |
| 07:13.42 | hades123 | as they call it |
| 07:13.45 | hades123 | sure, thanks |
| 07:14.07 | BBHoss | oh and BTW if you can't get a straight answer out of digium, hit me up i can probably get it from someone |
| 07:14.07 | mosty | dual 3Ghz dual xeons can handle 4 E1 lines |
| 07:14.15 | *** join/#asterisk s0lid (n=s0lid@210.213.199.147) |
| 07:14.35 | BBHoss | hades123, this might be what you're looking for: Compiling Asterisk from Source |
| 07:14.37 | BBHoss | damn |
| 07:14.41 | BBHoss | http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning |
| 07:15.25 | hades123 | I read it of course :) |
| 07:15.50 | hades123 | however, no solid data, merely people reporting what they achieved |
| 07:15.59 | hades123 | could be right, could be bogus |
| 07:16.46 | BBHoss | i know for a fact that * can handle 50 pstn connections |
| 07:17.03 | hades123 | yes , I know, but besid eit |
| 07:17.13 | hades123 | 1500 phones sending registrations |
| 07:17.23 | hades123 | and another 200 - 300 calls |
| 07:17.28 | hades123 | internal |
| 07:18.28 | BBHoss | yeah but if all you're doing is reinvites then that basically negates those other calls |
| 07:18.43 | BBHoss | now if all the phones try to register all at once, that might be a problem |
| 07:18.52 | BBHoss | not 100% sure |
| 07:18.54 | hades123 | no no, |
| 07:19.17 | hades123 | do you know , if re-invites |
| 07:19.22 | hades123 | diable any features other than |
| 07:19.24 | hades123 | CDR |
| 07:19.39 | hades123 | I mean , call forward, hold |
| 07:19.43 | hades123 | ext.. |
| 07:19.48 | hades123 | etc.* |
| 07:19.50 | BBHoss | you can do those on the phone |
| 07:20.07 | BBHoss | polycom supports forwarding hold and conferencing independently |
| 07:20.26 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:20.33 | hades123 | hmmm, but besides that |
| 07:20.40 | hades123 | when a phone is in the -re-invite state |
| 07:20.53 | hades123 | and needs to do something that asterisk has to be involved with ti |
| 07:20.58 | BBHoss | like what? |
| 07:21.22 | hades123 | conferencing |
| 07:21.26 | hades123 | with a pstn |
| 07:21.27 | hades123 | line |
| 07:21.39 | hades123 | or forward to a pstn line |
| 07:21.43 | hades123 | for instance |
| 07:21.45 | BBHoss | like a meetme conference? |
| 07:22.06 | hades123 | lets make it simple , forward to a pstn line |
| 07:22.24 | hades123 | in this case, asterisk has to pickup |
| 07:22.24 | BBHoss | well lets say joe wants to forward all his calls to his cell phone |
| 07:22.28 | BBHoss | right? |
| 07:22.32 | hades123 | no |
| 07:22.46 | hades123 | he lets say an operator sis tlaking to a sales person |
| 07:22.50 | hades123 | anor a manger |
| 07:23.09 | hades123 | and hte manager says for instance |
| 07:23.14 | hades123 | dial sam and |
| 07:23.22 | hades123 | for me |
| 07:23.35 | hades123 | and stay with us to take notes |
| 07:23.48 | hades123 | so now the operator will call sam on pstn line |
| 07:24.12 | hades123 | after putting manage ron hold |
| 07:24.16 | hades123 | then join them together |
| 07:24.40 | BBHoss | no, asterisk will only be utilized for the outbound trunk, the conferencing can be done on phone |
| 07:25.05 | hades123 | hmm |
| 07:25.29 | hades123 | the problem I am bieng restricted to a phone set like this |
| 07:25.31 | BBHoss | the exact way you would do this on a polycom is to put the manager on hold, dial sam, then press the conference button |
| 07:25.40 | mosty | for up to 3 callers |
| 07:25.44 | BBHoss | yep |
| 07:26.06 | BBHoss | if you want bigger than that, you can use meetme, or my personal favorite app_conference |
| 07:26.09 | hades123 | I was thinking about SNOM |
| 07:26.13 | hades123 | for the phone |
| 07:26.17 | BBHoss | snoms can do two at least |
| 07:26.35 | BBHoss | hang on ill check, i have one right here |
| 07:26.38 | *** join/#asterisk nixbox (i=oh@cpe-24-175-74-160.tx.res.rr.com) |
| 07:26.41 | nixbox | back |
| 07:26.58 | nixbox | any ideas BBHoss? |
| 07:27.13 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 07:27.23 | hades123 | nixbox: I was keeping BBHoss busy |
| 07:27.48 | BBHoss | hades123, snom 320 can conference 3 people, including itself |
| 07:28.56 | hades123 | I hope the 300 can do the same |
| 07:28.59 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:29.00 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:29.05 | BBHoss | i wouldnt mess with it |
| 07:29.21 | hades123 | you don't like the snom 300 ? |
| 07:29.24 | BBHoss | it's sound quality isn't as good as the 320, especially not on speaker |
| 07:30.00 | BBHoss | sounds really tinny and cheap, IMHO |
| 07:30.15 | hades123 | what about the Equivalent polycom to snom 300? |
| 07:30.27 | hades123 | is it better. |
| 07:30.31 | hades123 | ? |
| 07:30.33 | BBHoss | yeah |
| 07:30.46 | BBHoss | i have used polycom 301, 500, and 600s |
| 07:30.50 | BBHoss | they are all great |
| 07:31.04 | BBHoss | the 430 looks quite attractive though |
| 07:31.58 | adeel | i've used the polycom 320/330, 601, and 650's |
| 07:32.01 | adeel | they're pretty nic |
| 07:32.07 | hades123 | I Am looking at it now |
| 07:32.11 | hades123 | looks nice |
| 07:32.17 | BBHoss | can you believe i got a 600, brand new, for 60 bucks? |
| 07:32.21 | hades123 | u know what BBHoss |
| 07:32.32 | adeel | BBHoss, from where? |
| 07:32.34 | hades123 | you should go ask polycom for money now |
| 07:32.46 | hades123 | because u might have just made them sell 1500 phones |
| 07:32.47 | BBHoss | liquidation sale from a distributor |
| 07:32.49 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
| 07:32.52 | adeel | sweet |
| 07:33.16 | BBHoss | i also got a te210p for $285 |
| 07:33.36 | BBHoss | they had some xeons with 32 megs of l3 and 8 megs of l2 cache for like 25 bucks a piece! |
| 07:33.37 | hades123 | voip is a very expensive addiction |
| 07:33.42 | BBHoss | i thought it was a scam |
| 07:33.42 | drmessano | lol |
| 07:33.59 | drmessano | Voip is a conspiracy |
| 07:34.09 | BBHoss | funny thing is, i've never bought anything from them |
| 07:34.09 | drmessano | It doesnt REALLY work.. but it's fun to try |
| 07:34.17 | BBHoss | drmessano, lol |
| 07:34.27 | hades123 | :D |
| 07:34.32 | jblack | hades123: Tell me about it. All told, I spent $800 in my first month. |
| 07:34.40 | adeel | i'm getting very low call volume when bridging 2 zap calls together (1 is an incoming call, and the other is an outgoing call) and not sure how/why i'm having this problem |
| 07:34.53 | BBHoss | hades123, i wish i had a distributor that sold them, then i could just sell them to you :) |
| 07:34.56 | adeel | BBHoss, what distributor? |
| 07:35.00 | BBHoss | alliance |
| 07:35.04 | jblack | I got tricked by how cheap service is. |
| 07:35.06 | alrs | polycom 320 video <- http://www.tipandring.org |
| 07:35.07 | drmessano | You'll never be happy until you keep spending money, all the way up to Ciscos... |
| 07:35.10 | drmessano | But then |
| 07:35.17 | drmessano | You wasted on shit |
| 07:35.22 | drmessano | So you buy cheaper phones |
| 07:35.25 | jblack | I'm happy with what I have now. |
| 07:35.36 | adeel | drmessano, what are good CIsco phones to get for asterisk? |
| 07:35.47 | drmessano | No such thing, adeel, no such thing |
| 07:35.49 | BBHoss | the 320s are single port though right? |
| 07:35.53 | alrs | yes |
| 07:36.06 | adeel | drmessano, really? i have a client who's hell bent on getting cisco phones |
| 07:36.13 | BBHoss | yeah most rollouts i do are 2 port |
| 07:36.15 | drmessano | A good Cisco phone to use on Asterisk is a Linksys |
| 07:36.17 | BBHoss | no new wires |
| 07:36.20 | alrs | adeel: modify your clients behavior |
| 07:36.26 | BBHoss | heh |
| 07:36.29 | adeel | but i don't have any experience with cisco phones/provisioning |
| 07:36.31 | BBHoss | client reload |
| 07:36.33 | alrs | BBHoss: the 330 is the same thing with the built-in switch |
| 07:36.40 | BBHoss | alrs, exactly |
| 07:36.42 | adeel | alrs, yeah, i've been trying to get him to consider the Polycom 650's instead |
| 07:36.51 | drmessano | Tell your client that Cisco committed suicide and Polycom is filling in for them |
| 07:36.55 | BBHoss | 650s are overkill IMO |
| 07:37.18 | adeel | i like the 650's...they're only 20-30 bucks more than the 601's, and you get that pretty lcd |
| 07:37.26 | BBHoss | except maybe for secretary and executives |
| 07:37.29 | drmessano | Better yet, tell them that Cisco phones get "shorts" in them |
| 07:37.34 | BBHoss | heh |
| 07:37.37 | drmessano | People hate "shorts" |
| 07:38.17 | mosty | adeel, does your client realise that they won't get all the normally supported features when using cisco phones with asterisk? |
| 07:38.35 | BBHoss | anybody here use SIPp? |
| 07:38.37 | adeel | mosty, yep |
| 07:38.53 | hades123 | is $80 a good price for a polycom 320 ? |
| 07:38.56 | alrs | BBHoss: I have, but it was 1Q 2007 |
| 07:39.00 | alrs | hades123: yes |
| 07:39.06 | BBHoss | tell him polycoms are to asterisk as ci$co is to ci$co call manager :) |
| 07:39.13 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 07:39.14 | hades123 | alrs: Thanks |
| 07:39.16 | alrs | hades123: though you will have to buy a power adapter if you don't have 802.3af poe |
| 07:39.23 | mosty | adeel, so why do they want cisco phones? |
| 07:39.27 | BBHoss | oh you can bet hell want that |
| 07:40.00 | BBHoss | hades123, get a PoE switch that does vlans then separate ALL voice traffic from the other network |
| 07:40.06 | adeel | mosty, i have no flipping idea |
| 07:40.06 | hades123 | alrs: oh that's the trick |
| 07:40.09 | BBHoss | give it QoS and Diffserv |
| 07:40.10 | hades123 | no power adaptor |
| 07:40.20 | adeel | will cisco's even work without their Call manager thing? |
| 07:40.25 | BBHoss | yeah |
| 07:40.32 | drmessano | Sadly, yes |
| 07:40.40 | adeel | is provisioning a pain? |
| 07:40.47 | alrs | adeel: yes |
| 07:40.53 | mosty | adeel, the models that support SIP will work. i know nothing about provisioning cisco phones |
| 07:40.53 | drmessano | Cisco phones work better without power and ethernet |
| 07:40.59 | BBHoss | heh |
| 07:41.02 | hades123 | BBHoss, the design was to have internet and voice on the same network, hwoever, QoS qill be enabled on the switch |
| 07:41.08 | adeel | drmessano, haha |
| 07:41.12 | BBHoss | hades123, no i mean with a vlan |
| 07:41.13 | hades123 | there will be a sperate network for IPTV hwoever |
| 07:41.33 | BBHoss | ahh, sounds interesting |
| 07:41.48 | adeel | BBHoss, what's the benefit of isolating the voice traffic from the data? |
| 07:42.05 | mosty | adeel, QoS |
| 07:42.06 | BBHoss | its more easily prioritized |
| 07:42.22 | BBHoss | plus there are security features |
| 07:42.31 | drmessano | The same people that want Cisco are the reason Volvo is still in business |
| 07:42.35 | BBHoss | especailly if you use a mac-list on the dhcp server |
| 07:42.38 | adeel | if you're on a small network, that doesn't use UDP at all, wouldn't it just be easier to prioritize all UDP packets instead of the hassle of a vlan? |
| 07:43.02 | hades123 | drmessano: don't compare cisco to volvo :D |
| 07:43.15 | drmessano | "No, you're not going to run into an 18-wheeler, flip off a bridge, and land in a lake.. buy a cheaper car" |
| 07:43.18 | BBHoss | adeel, well its easier to not even bother with it |
| 07:43.51 | adeel | interesting |
| 07:44.05 | hades123 | either just o make them shout |
| 07:44.10 | hades123 | who cares about phones really |
| 07:44.21 | adeel | does * still have the ~120 simultaneous call limit? |
| 07:44.27 | drmessano | lol |
| 07:44.34 | drmessano | That hard coded damn limit |
| 07:44.37 | drmessano | :( |
| 07:44.40 | BBHoss | heh |
| 07:44.41 | drmessano | What limit? |
| 07:44.43 | hades123 | I am gonna convince them to go back to the tin can solution |
| 07:44.54 | BBHoss | i think it had a lot to do with deadlocks, but apparently they've fixed a bunch |
| 07:45.07 | adeel | is it an * limit, or an os limit, or a phsyical cpu/ram limit? |
| 07:45.10 | BBHoss | and with 1.6 they are using hash tables for SIP, which is a really good idea |
| 07:45.33 | BBHoss | i plan on testing 1.6 very soon |
| 07:45.37 | drmessano | I wonder where they got the idea for HASH tables from.. |
| 07:46.00 | adeel | i hate being sick |
| 07:46.06 | BBHoss | we don't smoke hash down here |
| 07:46.11 | BBHoss | dope coke and meth :) |
| 07:46.13 | hades123 | adeel: justout of a nasty flu |
| 07:46.25 | hades123 | 10 days, sneezing like a mofo |
| 07:46.29 | adeel | hades123, i'm on the verge of a nasty cough, and i can't afford any downtime |
| 07:47.26 | BBHoss | Hmm, everyone around me is sick somewhat, but not me! |
| 07:47.43 | hades123 | then you infected them |
| 07:47.50 | BBHoss | exactly |
| 07:47.55 | BBHoss | the rage virus :) :) |
| 07:48.17 | *** join/#asterisk [hC] (n=hardcore@24.85.160.5) |
| 07:48.27 | hades123 | I Am gonna go watch TV |
| 07:48.38 | hades123 | just installed one dish |
| 07:48.41 | BBHoss | come back anytime :) |
| 07:48.43 | hades123 | that can recive 4 Sats |
| 07:48.48 | BBHoss | damn |
| 07:48.51 | hades123 | from no TV at all |
| 07:48.53 | hades123 | to 1200 channels |
| 07:48.56 | BBHoss | whats your location? |
| 07:49.04 | hades123 | Ontario , Canada |
| 07:49.10 | hades123 | I get BEV |
| 07:49.12 | hades123 | and DN |
| 07:49.13 | BBHoss | ahh |
| 07:49.22 | [hC] | So, i'm calling voicemailmain(@some.awesome.context) - and asterisk is not picking up the password out of that context, even though it says an error like : Incorrect password '12345' for user '104' (context = some.awesome.context) |
| 07:49.28 | [hC] | makes no sense... |
| 07:52.13 | BBHoss | are periods allowed in contexts? |
| 07:53.00 | [hC] | yeah i just checked that |
| 07:53.03 | [hC] | doesnt seem to be the case. |
| 07:53.26 | BBHoss | i know the book says a to z, underscore, and hyphen |
| 07:53.48 | [hC] | checking length now. |
| 07:53.58 | BBHoss | 79 characters max |
| 07:54.07 | BBHoss | 79 chars and 1 null |
| 07:54.10 | [hC] | yeah it wasnt that long |
| 07:54.12 | [hC] | still not working |
| 07:54.23 | [hC] | i have two contexts defined |
| 07:54.28 | [hC] | [domain.com] |
| 07:54.33 | [hC] | with 101-110 defined as vmboxes |
| 07:54.37 | [hC] | then [some.awesome.context] |
| 07:54.42 | [hC] | with 101-110 defined again |
| 07:54.57 | [hC] | even though the error says "context = 'some.awesome.domain' - its verifying against the first context |
| 08:08.42 | yang | I am wondering are the span lines defined correctly ? http://openpaste.org/en/4923/ |
| 08:08.43 | [hC] | this is screwed up. asterisk says the right context in the error, but is clearly not authenticating from it. |
| 08:13.08 | yang | and in zapata.conf I have such definition channel => 1-2,4-5,7-8 |
| 08:15.52 | creativx | what if they only drop the "don't" part |
| 08:17.01 | drmessano | uh |
| 08:17.11 | jblack | Yeah, that's what I mean. |
| 08:17.12 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 08:17.23 | drmessano | Of course it is |
| 08:17.26 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 08:18.23 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 08:19.11 | jblack | Imagine if they pull off not being evil for the next 30 years, until brin and co retire.... only to be replaced with.... I dunno... darth vader, bill gates, steve jobs.. some truly nasty devil incarnate. |
| 08:19.17 | [hC] | so i found the problem |
| 08:19.19 | [hC] | searchcontexts=yes |
| 08:19.20 | [hC] | BAD |
| 08:19.24 | [hC] | in voicemail.conf |
| 08:19.33 | [hC] | ignores contexts and just matches the first one it finds. |
| 08:20.03 | *** join/#asterisk xtr (i=01928375@216.19.191.191.novuscom.net) |
| 08:20.05 | drmessano | Oh this is good |
| 08:20.46 | drmessano | Firefighter.. who can't speak anyway.. in a building... wearing his SCBA.... yelling into a crappy portable... |
| 08:21.21 | drmessano | HAMIBLAGA FLOO CAMAND SEMA FAN UPA SNABLE HAGIBA |
| 08:21.28 | drmessano | "You need the fan? |
| 08:21.39 | drmessano | "SLAMY FLA" |
| 08:21.42 | *** part/#asterisk SteveTotaro (n=root@pool-70-22-26-147.balt.east.verizon.net) |
| 08:21.50 | *** join/#asterisk SteveTotaro (n=root@pool-70-22-26-147.balt.east.verizon.net) |
| 08:24.13 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 08:25.31 | yang | hey jblack ! Could you help me around zaptel ? |
| 08:25.41 | jblack | dont have one |
| 08:25.51 | yang | ok |
| 08:26.43 | yang | and in zapata.conf I have such definition channel => 1-2,4-5,7-8 , but the error is then Unable to register channel '1-2,4-5,7-8' |
| 08:26.55 | yang | For each PCI card 2 channels |
| 08:35.56 | yang | Well I am a bit confues about the groups/channels http://openpaste.org/en/4924/ |
| 08:39.15 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 08:40.08 | kaldemar | yang: do you have zaptel modules loaded and ztcfg run? |
| 08:40.45 | *** join/#asterisk _gm (n=mustafa@58.27.172.111) |
| 08:40.50 | _gm | hi |
| 08:41.23 | _gm | asterisk realtime works like charm but problem is with patterns ,, i am using asterisk realtime ldap driver, |
| 08:41.33 | _gm | pattern matching is not working |
| 08:41.45 | _gm | any idea? |
| 08:45.29 | yang | kaldemar: yes, everything http://openpaste.org/en/4925/ |
| 08:48.51 | yang | and I need some sort of Point to Point ISDN configuration ... |
| 08:49.07 | yang | regarding the number routing |
| 08:49.23 | yang | This is what we have so far on our vlines - Point to Point option |
| 08:50.30 | _gm | anyone here can tell me if pattern matching works with ldap realtime driver? |
| 08:50.56 | *** join/#asterisk SparFux (n=raoul@e182021075.adsl.alicedsl.de) |
| 08:51.41 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 08:54.57 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 09:11.28 | *** join/#asterisk Modcuts (n=Bob@lan.proporta.com) |
| 09:12.37 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 09:14.56 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 09:15.40 | yang | kaldemar: still around ? |
| 09:16.53 | FlatFoot | morning all |
| 09:17.24 | FlatFoot | contact between 2 *'s which would you prefer SIP or IAX ? |
| 09:18.27 | yang | I guess SIP is a better standard nowdays? |
| 09:19.02 | BBHoss | iax |
| 09:19.05 | BBHoss | fo sho |
| 09:19.24 | FlatFoot | mixed thoughts eh ! |
| 09:19.30 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:19.39 | yang | Well, Its just what I heard, don't listen to me |
| 09:19.39 | BBHoss | no, IAX is designed to link two * boxes together |
| 09:19.47 | BBHoss | you can even do encryption |
| 09:20.00 | FlatFoot | i got a prob with * 1.4.17 connecting to 1.2.x on IAX having one way audio with the occasional two way |
| 09:22.03 | FlatFoot | could be a routing prob which is being looked at by the Mr Route ( Tech ) so i can't say for def what it is at the moe |
| 09:29.13 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 09:30.55 | yang | traceroute |
| 09:33.06 | JT | forget about iax |
| 09:33.09 | JT | too many problems |
| 09:33.11 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 09:33.18 | BBHoss | what? |
| 09:33.18 | JT | especially between different versions of asterisk |
| 09:33.54 | BBHoss | i've never had any trouble with it |
| 09:34.01 | JT | FlatFoot: is there any reason to not use sip? |
| 09:34.05 | JT | BBHoss: well you're lucky |
| 09:34.24 | BBHoss | not really |
| 09:34.39 | FlatFoot | JT no can do sip or iax |
| 09:36.36 | FlatFoot | JT ah well off to the wiki for the info tara for a bit |
| 09:37.12 | JT | yeah iax is mostly hype |
| 09:37.20 | JT | there are a couple of corner cases where it's useful |
| 09:37.24 | JT | otherwise just go sip |
| 09:38.02 | BBHoss | corner cases? |
| 09:38.07 | JT | yes |
| 09:38.40 | JT | corner cases. |
| 09:38.40 | BBHoss | what about the fact that it only requires one port? |
| 09:38.45 | BBHoss | works with NAT much better than iax |
| 09:38.46 | JT | big deal |
| 09:38.53 | JT | that makes it less scalable |
| 09:39.05 | JT | and sip works through most NATs fine when setup correctly |
| 09:40.23 | BBHoss | also trunking uses less bandwidth |
| 09:40.31 | JT | again hype |
| 09:40.39 | JT | and trunking massively reduces scalability |
| 09:40.40 | BBHoss | you can do aes128 encryption with it |
| 09:40.49 | JT | that's cool |
| 09:40.55 | JT | but you can also do vpn tunnels |
| 09:41.10 | BBHoss | lemme guess, it reduces scalability too? |
| 09:41.16 | JT | obviously |
| 09:41.31 | JT | encrypting traffic reduces scalability, that's computer basics |
| 09:41.58 | JT | but trunking is implemented poorly |
| 09:42.05 | JT | and will choke with a few dozen concurrent calls |
| 09:42.55 | JT | also reliance on zap timing is a joke |
| 09:44.06 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 09:44.07 | BBHoss | well, i guess you have a point, i've just always preferred IAX for linking * boxes |
| 09:44.12 | [hC] | I am abandoning IAX for SIP in the near future |
| 09:44.22 | [hC] | I took the approach of 'if IAX was designed for this, i should use it' |
| 09:44.27 | BBHoss | what do you mean reliance on zap timing? |
| 09:44.40 | JT | IAX trunking requires timing from Zaptel |
| 09:44.41 | [hC] | but ive recently had to question its ability to run >30 concurrent calls |
| 09:44.48 | JT | it needs zap hardware to work properly |
| 09:44.55 | JT | ztdummy to work less properly |
| 09:45.05 | BBHoss | ahh, yeah zaptel is never good :) |
| 09:46.27 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:46.54 | JT | asterisk has no native voip protocol |
| 09:47.16 | JT | natively it's a big connector with SLIN inside much of it |
| 09:47.30 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:47.47 | [hC] | I hate how i cant run in full g729 native mode, due to the slin dependency |
| 09:47.48 | JT | slin is fine when you need it :) |
| 09:47.54 | JT | well |
| 09:48.04 | JT | if you are doing something with the audio |
| 09:48.11 | JT | you need it to go to a baseline |
| 09:48.11 | [hC] | meetme! |
| 09:48.16 | JT | heh |
| 09:48.16 | [hC] | yeah |
| 09:48.25 | [hC] | i understand it |
| 09:48.27 | JT | as if it would be easy to natively mix g.729 |
| 09:48.30 | BBHoss | app_conference is a good alternative |
| 09:48.44 | [hC] | you still need to transcode to something, you cant natively mix g729 |
| 09:48.59 | [hC] | well, its not easy to. and its not implemented if it IS possible. |
| 09:49.32 | [hC] | I do need to check out speex again, though |
| 09:49.42 | [hC] | last time i played with it in *, it sounded like garbage. |
| 09:50.01 | BBHoss | i like speex, i use it on my intra-company trunks |
| 09:50.45 | [hC] | the key i suppose is finding a codec that the handsets all support, even better if that codec can do narrow and wide band |
| 09:50.53 | [hC] | and so i can eliminate transcoding |
| 09:51.03 | [hC] | hence g729. |
| 09:51.11 | BBHoss | dunno why companies won't put speex in everything |
| 09:51.26 | JT | cpu. |
| 09:51.39 | JT | the fact that no commercially available softswitches support it |
| 09:52.05 | JT | little reason for phone makers to spend big bucks integrated speex |
| 09:53.16 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:55.40 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:56.05 | Daviey | it'll come, no doubt |
| 09:56.36 | JT | really? what will cause the push? i'm skeptical |
| 09:56.57 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:00.08 | Daviey | JT: * is now starting to get the respect of handset manufacturers - if * supports speex, and speex is a free codec - i can't see why they wouldn't |
| 10:00.16 | Daviey | IMO |
| 10:00.29 | JT | asterisk doesn't even support speex out of the back iirc |
| 10:00.41 | [hC] | I think its a licensing issue |
| 10:00.42 | Daviey | no, it's in -addons isn't it |
| 10:00.50 | BBHoss | yeah, licensing |
| 10:00.51 | Alexandre_fr | /jojn #voip-users-conference |
| 10:00.52 | JT | yeah speex is not free |
| 10:01.00 | [hC] | I would like a codec that spans narrow and wide bands on the handset and the switch |
| 10:01.00 | BBHoss | yes it is |
| 10:01.05 | JT | err |
| 10:01.15 | BBHoss | its more free than GPL |
| 10:01.18 | JT | maybe i'm thinking of ilbc |
| 10:01.21 | BBHoss | its bsd licensed |
| 10:01.26 | JT | one of them where it's not free for commercial use |
| 10:01.28 | Daviey | hahahah |
| 10:01.38 | Daviey | Speex is very much free |
| 10:01.41 | BBHoss | that includes g729 |
| 10:01.49 | JT | g.729 is not free for ANY use |
| 10:02.01 | yang | What is the parameter to enable Asterisk BLF (Busy Light field), which grandstream phones have. |
| 10:02.04 | BBHoss | for experimentation, i thought it was free |
| 10:02.09 | JT | ilbc which is what i was thinking of, you can use for free non commercially |
| 10:02.17 | Daviey | yang: gandstream :( |
| 10:02.54 | yang | i mean, in the asterisk config |
| 10:02.56 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:03.48 | JT | but yeah, speex is free, but not included by default i thought |
| 10:03.49 | Daviey | probably not included because they won't disclaim authorship of their code to digium :) |
| 10:03.54 | JT | rofl |
| 10:04.00 | BBHoss | you have to have libspeex |
| 10:10.59 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 10:12.27 | *** join/#asterisk cjk_ (n=ldidelot@d90-129-39-85.cust.tele2.lu) |
| 10:12.33 | cjk_ | hi |
| 10:12.40 | BBHoss | sup dog |
| 10:14.48 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 10:17.11 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 10:19.30 | *** join/#asterisk BruceLeroy (n=herman@adsl-69-234-203-190.dsl.irvnca.pacbell.net) |
| 10:19.32 | BruceLeroy | Hello |
| 10:20.29 | BruceLeroy | What does eten => _P mean? |
| 10:21.00 | BruceLeroy | And what does "exten => _A" mean? |
| 10:22.57 | BruceLeroy | anyone awake? |
| 10:23.03 | Daviey | Oww, a Sangoma A101D just arrived |
| 10:26.38 | *** join/#asterisk BruceLeroy (n=herman@adsl-69-234-203-190.dsl.irvnca.pacbell.net) |
| 10:26.52 | BruceLeroy | anyone here? |
| 10:27.47 | BBHoss | no, never |
| 10:27.55 | BruceLeroy | quit |
| 10:37.39 | yang | Is this zaptel config in sync - http://openpaste.org/en/4926/ ? |
| 10:38.58 | cjk_ | hi, lets assume that I dial SIP/U1&SIP/U2&SIP/U3 now U2 refuses the call which sends back a busy. how can i get that BUSY in the next step. If I dial only SIP/U2 I have it in DIALSTATUS but I need the DISTATUS per peer |
| 10:45.42 | *** join/#asterisk mattzerah (n=matt@121.50.220.20) |
| 10:47.15 | *** join/#asterisk BruceLeroy (n=BruceLer@adsl-69-234-203-190.dsl.irvnca.pacbell.net) |
| 10:47.26 | BruceLeroy | hello |
| 10:47.59 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 10:48.27 | _gm | hi |
| 10:48.35 | _gm | anyone here can tell me if pattern matching works with ldap realtime driver? |
| 10:48.45 | *** join/#asterisk [hC] (n=hardcore@24.85.160.5) |
| 10:49.24 | BruceLeroy | Does anyone know what this means 'exten => _P' and/or 'exten => _A' |
| 10:49.24 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:50.44 | BruceLeroy | how do you get you whois to show '@gentoo/developer/fordfrog'? |
| 10:54.05 | JT | Daviey: why afraid? |
| 10:54.32 | Daviey | JT: fairly expensive bit of kit |
| 10:54.52 | JT | true |
| 10:54.54 | JT | but worth it |
| 10:54.55 | Daviey | RRP ~$1200 :O |
| 10:55.04 | JT | err in what country? |
| 10:55.15 | Daviey | uk |
| 10:55.33 | JT | pretty sure it's a USD$900 or so product these days |
| 10:55.39 | Daviey | oh |
| 11:05.24 | *** join/#asterisk anonymiss (n=user@c-68-84-36-113.hsd1.pa.comcast.net) |
| 11:06.13 | *** join/#asterisk Weetos (i=willy@mail.catalise.fr) |
| 11:06.46 | Daviey | JT: and relax, it's fitted |
| 11:06.57 | Daviey | did need a hammer :O |
| 11:07.43 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 11:08.15 | anonymiss | what are some good kernel options for asterisk systems? |
| 11:08.32 | anonymiss | like preemption and config_hz=1000 ? |
| 11:08.45 | jblack | I'd go with a high hz value, yeah. |
| 11:09.01 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:10.43 | Daviey | 100 hz is better for ztdummy.. but not much else AFAIK |
| 11:11.01 | Daviey | 1000* rather |
| 11:12.07 | anonymiss | thanks |
| 11:12.08 | anonymiss | :) |
| 11:12.49 | anonymiss | what about kernel preemption though? would it have any effect on *? |
| 11:13.07 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 11:14.01 | *** join/#asterisk RoyK (n=roy@ip-187-5-149-91.dialup.ice.no) |
| 11:14.09 | jblack | preemption is intended for desktops, not servers. |
| 11:14.18 | waKKu | morning folks.. |
| 11:14.20 | Weetos | anonymiss> interesting question |
| 11:14.40 | jblack | anonymiss: Anyways, I'd leave preemption off, especially since you have such a high hz. |
| 11:14.44 | anonymiss | jblack: what makes a desktop a desktop is that you want latency over throughput |
| 11:15.22 | waKKu | does someone there using pickupgroup with IAX in asterisk 1.4 ??? I'm trying to use it, but have no more ideas why is it not working... My softphone is idefisk, and configs and CLI is here: http://pastebin.ca/886298 ... any ideas ? |
| 11:15.33 | JT | preempt is good for asterisk |
| 11:15.41 | JT | it needs to be responsive like a desktop |
| 11:16.15 | jblack | Oh, ok. I'd think it would hurt, since * is already using realtime, and you'd want to avoid preempting it. But ok |
| 11:16.40 | Weetos | I know that some game servers needs to be configured like desktops, but I didn't know about asterisk |
| 11:17.12 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 11:17.31 | msetim | I'm looking for how to identify when a call get busy line or secretary... |
| 11:17.51 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 11:18.06 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 11:18.09 | jblack | anonymiss: Anyways, accoring to JT, leave it on. |
| 11:19.06 | anonymiss | thanks for the help everyone |
| 11:20.08 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:26.22 | msetim | someone can help me? |
| 11:34.16 | sergee | jblack: aloha! :) |
| 11:34.25 | jblack | hey sergee |
| 11:34.27 | *** join/#asterisk darthkzm (n=darthk@77.240.56.17) |
| 11:35.02 | jblack | drmessano and I have been looking at something weird for the last few days. |
| 11:35.16 | sergee | ghosts? |
| 11:35.26 | darthkzm | hi everyone |
| 11:35.31 | jblack | I'd say it qualifies as a ghost. |
| 11:36.04 | darthkzm | anyone know what version of * has full rtcp support? |
| 11:36.23 | jblack | Sipphone and * don't get along well. Sipphone's sip connections can authenticate to various devices, but when * is concerned, no one is there. |
| 11:37.12 | jblack | I lost interest in it quickly, but it's driving drmessano bonkers. |
| 11:37.35 | jblack | darthkzm: I could be wrong, but I believe not. |
| 11:38.35 | FabiOne | hi all |
| 11:38.50 | FabiOne | i'min trouble about asterisk |
| 11:38.58 | jblack | In fact, I'm probably wrong. It looks like it went in in 1.4 |
| 11:39.15 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:39.25 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 11:40.04 | FabiOne | after i installed bristuff, zaptel && asterisk the pc at the boot hang at "loading kernel modules..." for about 5 mins |
| 11:40.26 | FabiOne | after it say [failed] |
| 11:40.34 | FabiOne | [fail] |
| 11:40.42 | FabiOne | and continue the boot |
| 11:41.22 | jblack | I'm not having a very accurate day. |
| 11:41.32 | FabiOne | the strange thing is that zaptel and asterisk work well |
| 11:42.07 | FabiOne | can be a permisison problem? |
| 11:43.08 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:43.25 | dominic1 | [Feb 1 11:50:23] WARNING[7950]: res_odbc.c:149 ast_odbc_smart_execute: SQL Execute returned an error -1: 42704: Error while executing the query; |
| 11:43.25 | dominic1 | FEHLER: Type »lo« does not exist (68) |
| 11:44.11 | dominic1 | I always get this error. Anybody knows what I can do? I am using postgres and odbc |
| 11:44.22 | dominic1 | I get this error in the table voicemailmessages |
| 11:46.26 | *** join/#asterisk Perun (i=perun@2001:6f8:1316:1234:218:f3ff:fe99:4b33) |
| 11:46.36 | Perun | hi |
| 11:46.43 | jblack | hi |
| 11:46.48 | darthkzm | jblack: thanks!!!! |
| 11:47.00 | darthkzm | i know it went in at some point |
| 11:47.01 | Perun | whats better for asterisk and hfc cards, zaptel or misdn? |
| 11:47.04 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 11:47.06 | jblack | darthkzm: You're welcome!! |
| 11:47.09 | jblack | What did I do? |
| 11:47.29 | darthkzm | i meant rtcp support :) |
| 11:47.35 | jblack | Oh, yeah. |
| 11:48.03 | darthkzm | i was hoping it went in on 1.2 as well |
| 11:48.38 | darthkzm | got one of those gui * versions i need rtcp support for |
| 11:49.11 | jblack | I'll pull a log off the rcs to see if I can find a date |
| 11:50.16 | jblack | Went in mid 2006 |
| 11:50.52 | jblack | actually, there was stuff for rtcp as early as mid 2005 |
| 11:51.37 | jblack | I just have trunk here. I don't have 1.4's svn |
| 11:52.13 | mvanbaak | real men run trunk in production |
| 11:52.26 | jblack | Pardon, I don't have 1.2's svn |
| 11:53.50 | mvanbaak | I have them all, 1.0, 1.2, 1.4, 1.6, trunk |
| 11:53.58 | msetim | Do you know a api to asterisk manager in c? |
| 11:54.14 | mvanbaak | msetim: did you try the wiki ? |
| 11:54.37 | Perun | is there any howto how to setup 2 hfc cards (NT and TE mode) with zaptel and asterisk? |
| 11:55.45 | msetim | yes... I don't want to write my own socket in c... I'm looking for an api like phpagi or java-asterisk |
| 11:56.02 | jblack | hmm. there's socket++ for c++ |
| 11:56.05 | mvanbaak | msetim: I dont see it there |
| 11:56.16 | mvanbaak | only a c++ and a c# example |
| 11:57.22 | msetim | yes :( |
| 11:57.47 | jblack | Did you take a look at libgnet ? |
| 11:57.49 | BBHoss | C# lol |
| 11:57.54 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 11:58.12 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 11:59.17 | JT | Perun: bristuff is better than misdn |
| 11:59.22 | JT | and comes with sample configs |
| 12:00.06 | Perun | ok |
| 12:00.23 | Perun | JT: you know a howto for 2 hfc cards and asterisk? |
| 12:00.26 | jblack | msetim: http://www.gnetlibrary.org/docs/ if you want to look at gnet. It's got a pile of useful stuff in it |
| 12:01.09 | JT | Perun: there is sufficient sample configs |
| 12:01.17 | JT | a howto isn't really needed |
| 12:04.16 | msetim | jblack, cool! I will take a look :) |
| 12:04.38 | Perun | JT: where I can find the sample configs? |
| 12:05.08 | JT | Perun: once you install bristuff |
| 12:08.52 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:09.54 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:13.21 | mvanbaak | the configs are in the .tgz you download from them |
| 12:17.42 | yang | Which string is required to apply at Dial command to make a forwarding after 10 seconds to SIP/XX |
| 12:20.53 | yang | Right now i am using such a string |
| 12:20.54 | yang | exten => 5863178,1,Dial(SIP/78,30,rt) |
| 12:20.54 | yang | exten => 5863178,n,Hangup() |
| 12:21.02 | *** join/#asterisk vrtk (n=bb@189.21.178.20) |
| 12:21.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:21.08 | yang | and i would like to dial SIP/80 after 10 sec. |
| 12:21.10 | jblack | yang: Really, it's time you read the book. I suggested it to you last week. |
| 12:21.28 | yang | jblack: i am on page 25 |
| 12:21.40 | jblack | There's an example that's very similiar in chapter 5 or 6. |
| 12:21.45 | yang | ok |
| 12:21.48 | yang | I ll have a lok |
| 12:21.56 | jblack | I'll find it for you |
| 12:22.52 | jblack | Look at page 132 |
| 12:23.20 | yang | thanks |
| 12:23.40 | jblack | read back a couple pages, forward a couple pages, and you shoudl be able to get how to do it |
| 12:25.46 | yang | Well the page is about configuring the dialplan to call between two phones, I can't see forwarding |
| 12:26.15 | yang | I can allready make it to ring two phones at the same time, i need a 10 seconds delay |
| 12:26.17 | cjk_ | hi, lets assume that I dial SIP/U1&SIP/U2&SIP/U3 now U2 refuses the call which sends back a busy. how can i get that BUSY in the next step. If I dial only SIP/U2 I have it in DIALSTATUS but I need the DISTATUS per peer |
| 12:26.39 | jblack | yang: Read it more closely. |
| 12:27.11 | jblack | cjk_: You'll have to dial them independantly if you want to get useful statuses. |
| 12:27.25 | jblack | yang: Fine. |
| 12:27.33 | jblack | s,1,Dial(firstnum,10) |
| 12:27.40 | jblack | s,n,Dial(secondnum,120) |
| 12:27.44 | cjk_ | jblack, thats not an option |
| 12:27.45 | yang | thanks |
| 12:29.48 | jblack | cjk_: Ok... Um... You can write and submit a patch that provides a way to access the individual dial statii. :) |
| 12:30.13 | cjk_ | jblack, ok that was my guess |
| 12:34.03 | *** join/#asterisk ArchSSM (n=tommy@host-81-191-139-130.bluecom.no) |
| 12:34.15 | FlatFoot | ~book |
| 12:34.16 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 12:34.27 | FlatFoot | jbot thanks very much |
| 12:34.27 | jbot | FlatFoot: no problem |
| 12:37.03 | dominic1 | anybody storing his voicemessages in a postgres database with asterisk 1.4? |
| 12:38.49 | yang | well |
| 12:40.43 | dominic1 | I have the problem that I always get the following error: retrieve_file: SQL Get Data error! |
| 12:40.43 | dominic1 | [SELECT * FROM voicemessages WHERE dir=? AND msgnum=?] |
| 12:41.00 | *** join/#asterisk L4m3r (n=l4m3r@about/essy/warning/L4m3r) |
| 12:42.26 | FlatFoot | dominic1 you might want to try [SELECT * FROM voicemessages WHERE dir='?' AND msgnum='?'] make it SQL Safe# |
| 12:43.29 | dominic1 | Yes, but this statement is asterisk internal |
| 12:43.42 | dominic1 | I get this error on the asterisk CLI |
| 12:44.22 | FlatFoot | dominic1 ah ok can you see the data being passed to SQL ? |
| 12:44.39 | FlatFoot | has it got any nasty spaces ? those ones you can't always see |
| 12:46.02 | dominic1 | I know that the data is beeing passed to sql, cause I see it in my editor. But in the cli I can not see any statement except the SELECT after the INSERT |
| 12:46.20 | dominic1 | I don't know if that is really a problem, cause the date is stored in database |
| 12:48.01 | *** join/#asterisk hmodes (i=hmodes@B1-66ER.matrix.gs) |
| 12:49.18 | *** join/#asterisk patrick-- (n=dynamite@gate.devnull.biz) |
| 12:50.09 | patrick-- | Hey, im trying to replace an ISDN- PBX with an asterisk server. we have about 40 phones and 4 outgoing channels. would a beroNet - BN8S0 suite that purpose? |
| 12:50.24 | patrick-- | or even 2 of them |
| 12:51.25 | J4k3 | are they exclusively local calls? |
| 12:51.44 | JT | patrick--: so would a junghanns, or a sangoma |
| 12:52.27 | patrick-- | JT: i need to supply the phones with power and therefore chose a beronet to be the best choice combined with an external adaptor |
| 12:52.38 | patrick-- | J4k3: no, also external calls should be made |
| 12:53.09 | J4k3 | wow those BRI cards are expensive... |
| 12:53.38 | patrick-- | well would that do the purpose? 2 beronet 8 port BRI cards ? |
| 12:53.47 | patrick-- | 40 internal phones 4 external lines |
| 12:53.47 | J4k3 | patrick--: are you trying to run ISDN PHONES off your *? |
| 12:53.56 | patrick-- | yes |
| 12:54.07 | JT | patrick--: junghanns can do the exact same thing |
| 12:54.15 | patrick-- | but whats the difference? |
| 12:54.18 | J4k3 | that sounds expensive and... expensive |
| 12:54.20 | JT | not much |
| 12:54.36 | JT | but junghanns writes the superior bri software |
| 12:54.38 | JT | bristuff |
| 12:54.51 | J4k3 | for the price of the cards, you could buy bling-bling IP phones. |
| 12:55.53 | patrick-- | J4k3: but 40 of them? |
| 12:56.35 | J4k3 | 40 ports sounds expensive |
| 12:56.41 | J4k3 | or can euro-isdn phones share lines? |
| 12:56.45 | J4k3 | physical lines |
| 12:56.57 | patrick-- | im not sure |
| 12:56.58 | yang | Regarding BLF keys on Grandstream phones...I can use them, but they don't blink red when the call is being used on another phone * |
| 12:57.04 | patrick-- | 4 lines can be used simultaneously |
| 12:57.43 | J4k3 | patrick--: if you can get decent internet connectivity there, I'd seriously consider IP phones and an ITSP |
| 12:57.49 | J4k3 | unless you're getting the BRIs really really cheap |
| 12:58.04 | patrick-- | thing is we have the ISDN phones |
| 12:58.17 | J4k3 | do they have any worth for resale? |
| 12:58.18 | JT | err, patrick--, bri allows for 2 simultaneous calls, not 4 |
| 12:58.36 | J4k3 | well, its prolly 2-lines to the handsets, 4 'out' of the pbx. |
| 12:58.38 | patrick-- | JT: we have 2 external connects |
| 12:58.46 | patrick-- | meaning 2 x 2 |
| 12:58.55 | JT | each phone takes 2 bris? |
| 12:59.27 | patrick-- | i dont get that question... why would 1 phone take 2 BRI's ? |
| 12:59.36 | JT | i don't know |
| 12:59.37 | JT | what does |
| 12:59.39 | J4k3 | theres a 'pbx' in between all this |
| 12:59.43 | JT | <PROTECTED> |
| 12:59.43 | JT | 23:52 < patrick--> meaning 2 x 2 |
| 12:59.51 | JT | a phone can't do 4 calls can it? |
| 12:59.54 | JT | at once |
| 12:59.58 | J4k3 | yes, but theres 40 phones |
| 12:59.58 | patrick-- | indeed |
| 13:00.02 | patrick-- | well |
| 13:00.07 | patrick-- | we have 4 external lines |
| 13:00.08 | J4k3 | 40 phones <-> pbx <-> 2xBRI |
| 13:00.17 | J4k3 | thats my guess |
| 13:00.27 | patrick-- | yupp pbx = * |
| 13:00.28 | J4k3 | I have a sneaky suspition the phones themselves prolly don't speak BRI |
| 13:00.28 | patrick-- | well |
| 13:00.29 | patrick-- | should be |
| 13:00.42 | JT | you can get bri isdn phones |
| 13:00.45 | patrick-- | they all work that way now |
| 13:00.48 | patrick-- | there is a hardware pbx |
| 13:00.50 | J4k3 | I know you can |
| 13:00.50 | JT | but if it's a proprietary key system |
| 13:00.53 | JT | they probably dont |
| 13:01.26 | patrick-- | the phones all do BRI |
| 13:01.38 | patrick-- | Phones -> * -> External Lines |
| 13:01.54 | J4k3 | two words: train wreck |
| 13:02.02 | JT | err |
| 13:02.07 | JT | i've actually done |
| 13:02.21 | JT | key system phones > key system > bri to * > telco |
| 13:02.27 | JT | and i've done both bri to telco |
| 13:02.30 | JT | then changed to pri |
| 13:02.30 | J4k3 | yeah |
| 13:02.32 | JT | worked fine |
| 13:02.33 | JT | :) |
| 13:03.11 | J4k3 | the only realistic way to do this is to keep your existing pbx running. trying to support 40 isdn bri handsets off * is going to cost you a fortune. |
| 13:03.29 | J4k3 | at least thats my interpretation of the situation |
| 13:04.05 | JT | sangoma have made the only cards that scale for such a solution |
| 13:04.29 | J4k3 | more than $3k? |
| 13:04.36 | J4k3 | $3k should get you 40 nice ip phones |
| 13:04.38 | JT | dunno |
| 13:04.46 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 13:10.28 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 13:11.00 | jmls | is there any way of telling if asterisk is being started or reloaded from within a .conf file ? |
| 13:11.17 | jmls | (is there a .conf file that is read only when * is started ?) |
| 13:11.50 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:12.10 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 13:12.29 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 13:17.07 | *** join/#asterisk _gm (n=mustafa@58.27.172.111) |
| 13:17.14 | _gm | hi all |
| 13:17.19 | _gm | anyone here can tell me if pattern matching works with ldap realtime driver? |
| 13:17.41 | _gm | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions has example with patterns in mysql |
| 13:17.55 | _gm | now i have tried both ldap and mysql driver |
| 13:18.10 | _gm | but pattern matching doesnt seem to work |
| 13:18.36 | J4k3 | wtf, microsoft bid $44B for yahoo |
| 13:18.43 | J4k3 | omgwtf$$$bbq |
| 13:19.12 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:19.20 | Zeeek | Yo Berlin May 26-27th 2008 a rilly big shoe about asterisk |
| 13:20.18 | Zeeek | http://www.asterisk-tag.org/wiki/Hauptseite |
| 13:20.34 | Zeeek | Microsoft just bought Digium |
| 13:20.50 | Zeeek | Ebay bought Fonality |
| 13:21.25 | [TK]D-Fender | Dogs & cats living together! |
| 13:22.01 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 13:22.20 | Zeeek | Lamb and lion lying in the same bed |
| 13:22.35 | Zeeek | My $£*%£ IP500 had to be rebotted again!!!!!!! |
| 13:22.47 | jblack | Breaking news! One evil company buys another! News at 11 |
| 13:22.52 | Zeeek | My most expensive phone and it can't say awake |
| 13:23.28 | Daviey | O_o |
| 13:23.29 | Zeeek | anyone in Germany or neighboring countries here? |
| 13:24.32 | defswork | Zeeek: heading into poland again ? |
| 13:24.40 | Zeeek | never been there |
| 13:25.01 | defswork | Zeeek: I mean Germany - are you giving us some forewarning ? |
| 13:25.15 | Daviey | Zeeek: I'm in Europe if thats enough |
| 13:25.29 | Zeeek | Yes, I'm rolling win an on Asterisk Appliance in May |
| 13:25.31 | Zeeek | http://www.asterisk-tag.org/wiki/Hauptseite |
| 13:25.48 | Zeeek | back ina few I have to go clear my weapons |
| 13:26.42 | defswork | Adhearsion workshop would be nice to go to |
| 13:26.46 | Daviey | Zeeek: I wonder if it's worth an English only person going. |
| 13:28.01 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:36.23 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 13:38.18 | mpwizard | How do I extract the user part from the URI in the dialplan? |
| 13:41.01 | styelz | mpwizard: see the cut() function |
| 13:41.12 | styelz | might help |
| 13:42.37 | styelz | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut |
| 13:42.55 | styelz | CUT() |
| 13:44.28 | mpwizard | styelz: ty. |
| 13:45.09 | FlatFoot | mpwizard: are you on the island ? |
| 13:45.09 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.135.85) |
| 13:45.30 | mpwizard | FlatFoot: Noooo... Sweden |
| 13:45.58 | FlatFoot | ah , just read the blurb about .nu |
| 13:46.26 | mpwizard | FlatFoot: huh? |
| 13:47.13 | FlatFoot | mpwizard: you show as having a .nu domain and i was reading about that because i haven't seen one before |
| 13:48.29 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:48.51 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@octroi.enst-bretagne.fr) |
| 13:49.09 | mpwizard | FlatFoot: okey... the word "nu" means "now" in swedish. So many in Sweden use .nu domains for that reason. |
| 13:49.51 | FlatFoot | mpwizard: ah , it's a small island just off New Zealand made of coral . Just interested in odd things |
| 13:52.43 | Zeeek | Daviey everything is in English or almost |
| 13:53.23 | *** join/#asterisk jeremy_g (i=smokeNsh@m83-178-243-187.cust.tele2.se) |
| 13:53.27 | *** join/#asterisk Weetos (i=willy@mail.catalise.fr) |
| 13:53.29 | shido6 | try googling Michael Tsarion, if you're into odd things. |
| 13:53.38 | jeremy_g | hi |
| 13:54.23 | jeremy_g | VAR1=x209 <-- how can i remove x from that |
| 13:54.31 | jeremy_g | ${VAR1:1} |
| 13:54.44 | jeremy_g | will this evaulate to VAR1 containing 209 |
| 13:56.01 | davevg-btwtech | jeremy_g, use noop to find out what it does :) |
| 13:56.21 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 13:56.50 | styelz | jeremy_g: http://www.voip-info.org/wiki/view/Asterisk+variables |
| 13:57.32 | jeremy_g | davevg-btwtech, styelz: thanks but i know this, have gone through this already. |
| 13:58.03 | FlatFoot | shido6: now thats ODD some reading for the weekend |
| 13:58.21 | shido6 | well |
| 13:58.21 | styelz | why are you asking then |
| 13:58.44 | shido6 | if u can sit through his lectures you'll have a few pages filled with links to more odd things. |
| 13:59.18 | *** join/#asterisk qdk_ (n=qdk@ip18.rev112.brygge.net) |
| 13:59.19 | shido6 | and if u get through them all... well... ...... if you are happy right now then dont read any further :) |
| 13:59.35 | yang | Regarding BLF keys on Grandstream phones...I can use them, but they don't blink red when the call is being used on another phone, any idea ? |
| 13:59.41 | FlatFoot | shido6 is he a bit of a downer ? |
| 14:00.00 | shido6 | no. |
| 14:00.03 | shido6 | Ignorance is bliss. |
| 14:00.17 | FlatFoot | shido6 ahh , ok |
| 14:01.02 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 14:01.11 | shido6 | i need some new mp3's |
| 14:01.23 | shido6 | know any good trance albums? |
| 14:01.23 | FlatFoot | want some SKA ? |
| 14:01.28 | *** join/#asterisk nighty^ (n=nighty@p2007-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 14:01.31 | shido6 | fine.... |
| 14:01.42 | shido6 | nothing to put me to sleep |
| 14:03.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:05.31 | x86 | ugh |
| 14:05.34 | x86 | snow sucks |
| 14:05.55 | anonymouz666 | snow? |
| 14:05.58 | anonymouz666 | it's summer |
| 14:06.05 | anonymouz666 | ;) |
| 14:07.31 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 14:10.45 | jeremy_g | davevg-btwtech, styelz: thanks, problem solved |
| 14:12.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:12.35 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 14:14.51 | sysadmin-lb22 | hi All ..a general voip question about the SIP packet..if a sip packet is sent from asterisk to a public GW ..the To: field is 3453534534@myasterisk.com instead of 342342342@gwIP |
| 14:15.10 | sysadmin-lb22 | is this normal....or is there something wrong with my config... |
| 14:16.53 | dominic1 | since today I got the following error on the cli: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission 3c43abcfc9b8-yfy15mrqcy05 for seqno 2 (Critical Response) |
| 14:16.58 | dominic1 | what does that mean? |
| 14:17.35 | [TK]D-Fender | dominic1: Means your endpoint isn't talking back to * |
| 14:17.54 | *** part/#asterisk jmls (n=jmls@81.138.42.77) |
| 14:20.02 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 14:20.26 | *** join/#asterisk L4m3r_ (n=l4m3r@about/essy/warning/L4m3r) |
| 14:20.34 | dominic1 | that's strange I pickup up the phone, dial a number, my first asterisk sends that over iax to my asterisk on the pstn line and I get a connection, after the cli shows IAX2/mytrunk answered SIP/myaccoung, the error is logged, but I am speaking to the person on the other side |
| 14:20.50 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:21.38 | dominic1 | can you tell me what I can do against this error? |
| 14:22.30 | [TK]D-Fender | dominic1: I suggest you enable SIP debug and show some comprehensive output next time. |
| 14:22.45 | [TK]D-Fender | dominic1: You don't enough details for us to say anything. |
| 14:23.01 | dominic1 | Okay, thank you very much |
| 14:23.02 | *** join/#asterisk SteveTotaro (n=root@pool-70-22-26-147.balt.east.verizon.net) |
| 14:23.41 | dominic1 | Is it normal that I get errors in the console, whe I use odbc as voicemailstorage? |
| 14:23.58 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 14:24.26 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 14:24.42 | dominic1 | <PROTECTED> |
| 14:27.04 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 14:27.47 | *** part/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 14:32.34 | mvanbaak | ~book |
| 14:32.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 14:35.20 | styelz | dominic1: make sure /var/spool/asterisk/voicemail/default/dob/INBOX exists and is writable by the user that asterisk runs as |
| 14:37.24 | dominic1 | yes, it exists and is writable |
| 14:37.27 | dominic1 | writeable |
| 14:38.06 | *** join/#asterisk techie (n=techie@adsl-76-240-176-254.dsl.lsan03.sbcglobal.net) |
| 14:39.06 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:39.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:39.10 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 14:39.36 | styelz | dominic1: looks like you have a msg0001.txt file left over, but the .wav is no longer there |
| 14:39.55 | styelz | try deleting the msg0001.txt is .wav doesnt exist |
| 14:40.21 | styelz | s/is/if |
| 14:40.21 | dominic1 | there is nothing in the directory |
| 14:40.43 | dominic1 | If I want to check my voicemails with voicemailmain, it doesn't work |
| 14:40.56 | dominic1 | [Feb 1 15:39:27] WARNING[9111]: file.c:568 ast_openstream_full: File /var/spool/asterisk/voicemail/default/dob/INBOX/msg0000 does not exist in any format[Feb 1 15:39:27] WARNING[9111]: file.c:871 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/dob/INBOX/msg0000 (format 0x8 (alaw)): No such file or directory[Feb 1 15:39:27] WARNING[9111]: app_voicemail.c:4630 play_message: Playback of message /var/spool/asterisk/voicemail/default/dob |
| 14:41.28 | styelz | odd |
| 14:41.45 | dominic1 | but there is data in the database |
| 14:42.34 | styelz | maybe reload asterisk |
| 14:42.45 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 14:43.20 | *** join/#asterisk AndyGraybeal_ (n=andy@node246.34.251.72.1dial.com) |
| 14:43.29 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:43.39 | styelz | not sure what your problem is |
| 14:43.45 | davevg-btwtech | dominic1, the data in the database probably is not synced up with what is actually recorded, did the voicemail get deleted when the db was unavailable? |
| 14:44.16 | dominic1 | didn't use voicemail before |
| 14:44.20 | dominic1 | today is the first time |
| 14:44.24 | davevg-btwtech | dominic1, probably just need to delete that voicemail record from the db |
| 14:44.29 | dominic1 | I don't think the db is out of sync |
| 14:44.54 | styelz | yea, if there are no .wav / .txt files the db would be empty for that user |
| 14:45.24 | nixguy | how can i verify if my 0h323 driver is properly loaded in my asterisk 1.2? |
| 14:45.43 | nixguy | show channeltypes gives me devicestate no |
| 14:45.47 | nixguy | whatever that means... |
| 14:45.58 | styelz | nigxguy: show modules like 323 ? |
| 14:46.43 | styelz | modules show like 323 . i mean |
| 14:46.43 | nixguy | styelz: it says 1 module loaded |
| 14:46.45 | nixguy | use count 0 |
| 14:46.51 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:46.52 | nixguy | gues that means its loaded but not used?! |
| 14:47.02 | nixguy | thnx then one step closer... |
| 14:47.31 | *** join/#asterisk ToTo (n=ToTo@209.8.41.159) |
| 14:51.57 | drmessano | I just got this Trixbox CD in a box of cereal.. what now? |
| 14:52.06 | styelz | lol |
| 14:52.21 | styelz | add milk and sugar? |
| 14:52.32 | FlatFoot | drmessano , put your mug o tea on it |
| 14:52.41 | drmessano | lol |
| 14:52.48 | styelz | hehe |
| 14:52.55 | [TK]D-Fender | drmessano: Its too late for the cereal.. its already infected... burn it. Burn it with FIRE. |
| 14:53.04 | drmessano | ROFLL |
| 14:53.13 | FlatFoot | LOVFL |
| 14:53.25 | cpm | burn it with love! |
| 14:53.37 | drmessano | If I send in 3 UPC's and $1.95 S&H I can get the limited edition Trixbox Pro Cd |
| 14:54.36 | drmessano | Not nearly as cool as the Sgt Slaughter or "The Fridge" GI Joe I got for UPCs |
| 14:55.00 | *** join/#asterisk PepOSX (n=angeldav@190.79.246.105) |
| 14:55.31 | dominic1 | How can I reset all the messages for a user? |
| 14:55.42 | dominic1 | he always tells me that there are three messages |
| 14:55.49 | dominic1 | but there aren't any messages |
| 14:56.00 | drmessano | WHOA...... [Slashdot] Microsoft Bids $44.6 Billion For Yahoo |
| 14:56.23 | drmessano | Saw that coming |
| 14:56.26 | tzanger | I think it's hilarious how these web companies are worth more than manufacturers of real material |
| 14:56.57 | tzanger | it's amazing the value in virtual shit |
| 14:57.01 | tzanger | or rather the perceived value |
| 14:57.05 | drmessano | MS + Yahoo would be interesting for it's reach.. of course, Yahoo is nothing more than Soccer Moms and spammers anymore |
| 14:57.13 | styelz | of paper |
| 14:57.50 | FlatFoot | all this auctioning of stuff that is either on or off , is weird |
| 14:57.52 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 14:58.27 | tzanger | there are a lot of technicla groups in yahoo groups, pisses me of |
| 14:58.28 | tzanger | er off |
| 14:59.03 | drmessano | It will be interesting to see if they unify their IM on XMPP |
| 14:59.22 | drmessano | Trying to slight Google by "Going Open" too |
| 14:59.45 | drmessano | not "slight", but "Flank" |
| 15:00.28 | coppice | XMPP? compatibility is very important to MS. wherever the see it, they work hard to break it. |
| 15:00.44 | tzanger | coppice: :-) |
| 15:01.33 | tzanger | I'm going to the library today to try an dlocate some "signal processing for dummies" books |
| 15:01.34 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:03.55 | drmessano | Well, Google is XMPP, AIM is not far from it.. and having MSN Messenger and Yahoo Messenger on the same network is only going to mean so much |
| 15:04.05 | drmessano | They |
| 15:04.33 | drmessano | They need to go after that juicy chunk of AIM users |
| 15:05.27 | drmessano | Best way would be to change the target demographic of Yahoo by lowering it's age some, and letting Yahoo work with AIM so you can switch and still add your friends |
| 15:06.08 | *** join/#asterisk tobias (n=tobias@66-233-119-44.ral.clearwire-dns.net) |
| 15:06.15 | drmessano | Either way, I just want everything on XMPP :) |
| 15:06.45 | agx | when i get the console spammed with "P[ 0] Unhandled Message: prim 282 len 64 from addr 52020200, dinfo 500 on this port" does it means i've to change the ISDN BRI from PTMP to PMP ? |
| 15:06.46 | coppice | tzanger: not exactly for dummies, but try "Understanding Digital Signal Processing" by Rick Lyons |
| 15:07.06 | *** join/#asterisk PJ2 (n=xx@213-176-182.netrunf.cytanet.com.cy) |
| 15:07.17 | PJ2 | hi |
| 15:07.50 | PJ2 | can anyone help me with asterisknow? |
| 15:08.10 | PJ2 | oops goin to the other chan :) |
| 15:08.27 | tzanger | coppice: I will try to find that, thank you |
| 15:09.11 | tzanger | coppice: I've got "Advanced Digital Signal Processing and Noise Reduction, 2nd ed" |
| 15:09.36 | PJ2 | hmm no one in asterisknow chan, but my question might be the same with asterisk |
| 15:09.51 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
| 15:10.05 | coppice | tzanger: "noise reduction" in the title sounds narrowly focussed |
| 15:10.16 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1096745276.dsl.bell.ca) |
| 15:10.25 | tzanger | coppice: indeed. |
| 15:10.38 | tzanger | my library doesn't have the book you reccommend |
| 15:10.47 | tzanger | looking around to see what else they have handy |
| 15:11.41 | tzanger | "DSP apps using C and the TMS320C6x" good start |
| 15:11.50 | tzanger | signal processing for telecom and multimedia |
| 15:12.40 | PJ2 | is it possible to use simple PC modem as analog line interface with asterisk? |
| 15:15.52 | lirakis_work | PJ2: no |
| 15:16.09 | PJ2 | hmm :s |
| 15:16.14 | lirakis_work | PJ2: totally different things |
| 15:16.43 | PJ2 | ok so is it fxs card that i need for that? |
| 15:17.20 | jake[work] | or FXO depending on phone or line that you're connecting to |
| 15:17.29 | PJ2 | hmm |
| 15:17.41 | PJ2 | whats the difference? |
| 15:17.46 | jake[work] | FXS = phone, FXO = line |
| 15:18.05 | PJ2 | oh ok |
| 15:18.46 | Zeeek | O = To connect to the (foreign excgange) Office. S to connect tot he Sucker that bought that Polycom |
| 15:18.47 | iCEBrkr | ASTERISK SUX!! |
| 15:18.57 | iCEBrkr | G'mornig all |
| 15:19.07 | niekie | No it doesn't. |
| 15:19.13 | niekie | It rules. |
| 15:19.28 | Weetos | asterisk actually rocks |
| 15:19.29 | *** join/#asterisk tsabi (n=tsabi@gw.creditexpress.hu) |
| 15:19.33 | niekie | You must be doing something wrong :) |
| 15:20.10 | coppice | it doesn't all out suck. it merely sips |
| 15:20.21 | Zeeek | ooooooooooo |
| 15:20.22 | niekie | Haha. |
| 15:20.28 | lunaphyte_ | drmessano: it sounded like you might have been looking for a clue to share you thought you had re: an spa3102 and passing incoming calls through * with cid (something about a dialplan?) - any luck? |
| 15:20.42 | iCEBrkr | I dunno, considering my glorified answering machine has been my primary phone since 2003.... |
| 15:20.42 | styelz | my vacum cleaner sux |
| 15:20.47 | drmessano | Yeah, hang on |
| 15:20.55 | drmessano | Somewhere I have it |
| 15:21.02 | cpm | yeah, so does my roomba |
| 15:21.10 | iCEBrkr | astroomba? |
| 15:21.14 | Zeeek | I now have a phone that dials a number via IAX or SIP depending on a prefix you use |
| 15:21.15 | *** join/#asterisk angryuser (i=nononon@df01t2-212-195-198-128.d4.club-internet.fr) |
| 15:21.24 | mf2 | w |
| 15:21.32 | iCEBrkr | Zeeek: Congrats :) |
| 15:21.33 | Zeeek | 0.0 0.0 0.0 |
| 15:21.47 | PJ2 | so with a fxo card working and asterisk running i could give my voip users possibility to call external nums and vice-versa? |
| 15:21.51 | Zeeek | but I still have no one to call :( |
| 15:21.52 | iCEBrkr | Zeeek: I did the whole 'dial 9' for a VoIP line thing. |
| 15:22.09 | styelz | hehe |
| 15:22.10 | jake[work] | pj2 - yes |
| 15:22.18 | PJ2 | cool :D |
| 15:22.19 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:22.21 | Zeeek | this is an ip phone that does both and connects to the servers via one of the two protocols. Not the same thing |
| 15:22.38 | iCEBrkr | Zeeek: Oh, that's kinda gay then. |
| 15:22.46 | Zeeek | BI you mean |
| 15:22.49 | iCEBrkr | lol |
| 15:23.05 | iCEBrkr | It swings both ways. SIP and IAX |
| 15:23.09 | iCEBrkr | : o |
| 15:23.10 | jake[work] | wow - WGA must still be on strike |
| 15:23.47 | Zeeek | they are thats why the jokes ar so bad in here |
| 15:23.47 | PJ2 | and do u know if the ones sold on digiumcards.com come with builtin support in asterisknow? |
| 15:23.49 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.114) |
| 15:23.54 | styelz | since light travels faster than sound some people may appear bright until they speak |
| 15:23.55 | iCEBrkr | PJ2: AsteriskNow is independent of the hardware.. AFIK |
| 15:24.04 | Zeeek | as am I |
| 15:24.14 | FlatFoot | styelz: vgood |
| 15:24.14 | AndyGraybeal_ | are there handheld sip or voip phones that operate on the wireless network? (like a cell phone, but only for WIFI?) |
| 15:24.18 | drmessano | lunaphyte_, do you have the SPA-3102 regged to Asterisk? |
| 15:24.20 | PJ2 | hmm |
| 15:24.26 | iCEBrkr | AndyGraybeal_: Yeah |
| 15:24.30 | Zeeek | AndyGraybeal_ yes |
| 15:24.33 | PJ2 | it dosnt come with drivers preinstalled? |
| 15:24.40 | Zeeek | ECHO SUPPRESSION IS OFF |
| 15:24.41 | AndyGraybeal_ | awesome, are their headsets for such items? |
| 15:24.44 | iCEBrkr | PJ2: Drivers? |
| 15:24.44 | lunaphyte_ | drmessano: currently, yes. |
| 15:24.55 | drmessano | (S0<:s>) <--- That doesn't work? |
| 15:24.59 | PJ2 | lol comin from windows environment.. |
| 15:25.08 | zobia | hello any one or developer can help me with teh sccp with 1.4.17? |
| 15:25.13 | lunaphyte_ | drmessano: the fxo side and fxs side are registering independently. |
| 15:25.24 | drmessano | Ok, thats good |
| 15:25.25 | agx | there is a link pointing to the changes inside asterisk 1.4.18 ? |
| 15:25.30 | lunaphyte_ | drmessano: i wasn't able to get it to work as of last night, no. |
| 15:25.51 | drmessano | Whats your dial string now? |
| 15:25.53 | AndyGraybeal_ | iCEBrkr and Zeeek; what are some of these devices? can you point me to a webpage? does polycom have such a thing? are they rugged enough for hard usage? |
| 15:26.26 | iCEBrkr | PJ2: Assuming AsteriskNow is justa n00b's implementation of Asterisk, it should handle FXO cards just fine. |
| 15:26.33 | iCEBrkr | AndyGraybeal_: They're not that great.. Actually. |
| 15:26.41 | FlatFoot | agx , UPGRADE.txt |
| 15:26.49 | jake[work] | i heard they pretty much all suck |
| 15:26.49 | AndyGraybeal_ | iCEBrkr: ah oaky |
| 15:26.57 | jake[work] | better to get an ATA and analog phone |
| 15:27.04 | jake[work] | but I could be wrong |
| 15:27.27 | AndyGraybeal_ | hm...... i don't need to use cellular waves... |
| 15:27.50 | cpm | jake[work], ??? work? jake? |
| 15:27.56 | BruceLeroy | hi |
| 15:27.57 | iCEBrkr | I just use a Sipuara and a cordless |
| 15:27.59 | AndyGraybeal_ | it would be more expensive too... i would only want to use the handheld phones inside our busines |
| 15:28.01 | jake[work] | always working on something |
| 15:28.07 | AndyGraybeal_ | iCEBrkr: aaah gotcha |
| 15:28.08 | BruceLeroy | Does anyone know what this means 'exten => _P' and/or 'exten => _A' |
| 15:28.09 | iCEBrkr | I wanted a coredless SIP phone, but they apparently all suck |
| 15:28.12 | drmessano | So use an ATA and a cordless, AndyGraybeal_ |
| 15:28.16 | zobia | hello. any developer here knows sccp ? |
| 15:28.19 | cpm | iCEBrkr, yes, they do. |
| 15:28.38 | [TK]D-Fender | BruceLeroy: means "P" or "A" |
| 15:29.06 | iCEBrkr | niekie: Most softphones suck ass too |
| 15:29.15 | iCEBrkr | niekie: The only one I liked was TalkExpress or whatever it's called. |
| 15:29.19 | niekie | iCEBrkr: that's why I'm not using one :) |
| 15:29.37 | cpm | at one point, I had a iaxy/wrt54/TA-312 field phone running on a pair of lantern batteries, it worked okay. But was a bit clunky |
| 15:29.40 | niekie | Just a DECT one, nothing special VoIP about it. |
| 15:29.46 | niekie | Connected to a VoIP capable router. |
| 15:30.05 | iCEBrkr | ExpressTalk |
| 15:30.38 | jameswf | tzafrir check /msg |
| 15:30.39 | *** part/#asterisk ming_zym (n=ming_zym@124.14.236.114) |
| 15:30.52 | iCEBrkr | http://www.nch.com.au/talk/be.html |
| 15:30.52 | lunaphyte_ | drmessano: i've been alternating between 2 different strings - (S0<:1160>) works, but all calls show up as 1160 in * - and (S0<:s>), which i haven't yet been able to get working. |
| 15:31.02 | iCEBrkr | It's a really nice softphone |
| 15:31.05 | *** join/#asterisk ManxPower (n=manxpowe@238.sub-70-222-242.myvzw.com) |
| 15:31.06 | iCEBrkr | It's not clunky |
| 15:31.11 | drmessano | What is your PSTN answer delay? |
| 15:31.48 | lunaphyte_ | 3 |
| 15:33.29 | drmessano | Try 5 |
| 15:33.32 | drmessano | 3 is too short |
| 15:34.00 | iCEBrkr | Live is to-to-to-short. |
| 15:34.20 | FlatFoot | ok is there a ringback func anywhere in * ???? |
| 15:34.29 | lunaphyte_ | ok. i'll have to wait until after work, it's all at home. |
| 15:34.52 | drmessano | CID info is passed between rings 1 and 2 |
| 15:35.13 | iCEBrkr | FlatFoot: Ringback as in when you transfer a call and the transfered-to extension doesn't answer? |
| 15:35.13 | *** join/#asterisk L4m3r_ (n=l4m3r@about/essy/warning/L4m3r) |
| 15:35.25 | jameswf | 10 hours have passed since i posted that picture and my boss hasent been called wooohoooo |
| 15:35.29 | drmessano | So you need at least the length of ring 1 + a second to parse it |
| 15:35.32 | drmessano | lol |
| 15:35.33 | *** join/#asterisk neoalex (n=chatzill@h-68-167-22-138.nycmny83.covad.net) |
| 15:35.43 | iCEBrkr | drmessano: Yes. CallerID info is sent after ring 1. |
| 15:36.08 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 15:36.08 | lunaphyte_ | ok |
| 15:36.15 | drmessano | I wasn't asking |
| 15:36.18 | drmessano | :) |
| 15:36.20 | iCEBrkr | I saw a ?? |
| 15:36.21 | neoalex | hi guys I'm having a problem with chan_mobile, namely when I try to make a call or when I answer a received call I get a segmentation fault and asterisk crashes alltogeather |
| 15:36.22 | iCEBrkr | :D |
| 15:36.27 | neoalex | any ideas? |
| 15:36.38 | iCEBrkr | See! Asterisk sucks! |
| 15:36.47 | drmessano | I found 5 seconds worked, lunaphyte_ |
| 15:36.56 | lunaphyte_ | cool, i'll give it a try. |
| 15:37.58 | iCEBrkr | neoalex: Dunno if this will help, but have you 'core set verbose 9' |
| 15:38.22 | iCEBrkr | I dunno if I should blame VoicePulse or my shitty cable provider |
| 15:38.22 | iCEBrkr | [Feb 1 10:35:33] NOTICE[8122]: chan_iax2.c:8101 __iax2_poke_noanswer: Peer 'vpconnect-t02' is now UNREACHABLE! Time: 51 |
| 15:38.25 | iCEBrkr | ALL |
| 15:38.26 | iCEBrkr | DAY |
| 15:38.26 | iCEBrkr | LONG |
| 15:38.37 | iCEBrkr | UNREACHABLE! REACHABLE! UNREACHABLE! |
| 15:39.05 | jameswf | neoalex: no bugs reported try rebuilding it. if it happens again and you can clearly reproduce it file a bug |
| 15:39.14 | dennis- | iCEBrkr: same here |
| 15:39.56 | drmessano | iCEBrkr: Are you using Qualify? |
| 15:40.00 | neoalex | well that's the thing it's the chan_mobile from trixbox repositories, I didn't build it myself, but that works fine on a different machine with the same exact dongle |
| 15:40.12 | drmessano | If so, turn it off.. then you wont be reminded heh |
| 15:40.26 | drmessano | trixbox? |
| 15:40.36 | neoalex | yes I know, not my choice |
| 15:41.06 | iCEBrkr | drmessano: Yea |
| 15:41.19 | FlatFoot | iCEBrkr , as you ring an external , they are engaged so press 5 or the like and * checks for them to hangup then init the call between you and them ! or am i gonna have to write that into the dialplan / |
| 15:41.23 | iCEBrkr | drmessano: I don't mind being reminded.. I mind when I can't make a reliable call. |
| 15:41.26 | drmessano | If I didn't think you ****** would flood me out, I would add "trixbox?" trigger on *trixbox* |
| 15:41.40 | iCEBrkr | Trixbox? |
| 15:41.43 | iCEBrkr | F Trixbox. |
| 15:41.52 | drmessano | no no |
| 15:41.57 | drmessano | trixbox? |
| 15:42.40 | drmessano | ¿trixbox? |
| 15:43.00 | jameswf | dongle is one of my favorite pc terms |
| 15:43.04 | iCEBrkr | Asterisk 1.4.6 built by root @ yurmom.cyberdyne.org on a i686 running Linux on 2007-10-08 15:05:38 UTC |
| 15:43.05 | drmessano | Thats the UTF-8 version |
| 15:43.06 | kyron | Q (lost in history...): Installing * at home (also applies to office actually) and was wondering: if I want to be able to put someone on HOLD and pick the call from another phone, how should I go about it, one phone is a poly 320 the others are all regular phones connected to a Mediatrix 1104... |
| 15:43.25 | drmessano | Press "Hold" |
| 15:43.39 | drmessano | Er ok |
| 15:43.40 | drmessano | Parking |
| 15:43.43 | iCEBrkr | kyron: You can have parking lots |
| 15:43.54 | kyron | jameswf, I find it's reminiscent of a male's distinctive anatomy. |
| 15:43.57 | iCEBrkr | Default park times are a bit short though |
| 15:44.13 | kyron | will I be able to park and unpark calls with "regular" phones? |
| 15:44.14 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 15:44.14 | jameswf | neoalex: file a bug report with trixbox, as it is a contained system no one here can really help they need to update |
| 15:44.16 | casix | hello |
| 15:44.17 | ManxPower | kyron: "hold" is a Key System concept. In PBXs you PARK the call. |
| 15:44.28 | iCEBrkr | kyron: Yea, Asterisk handles all of that |
| 15:44.35 | kyron | ManxPower, yeah, so I thought ;) |
| 15:44.50 | *** join/#asterisk ddunavant (n=David@pool-96-231-69-97.washdc.east.verizon.net) |
| 15:44.59 | jameswf | dongle is like those terds that get stuck in an animals butt hair |
| 15:45.00 | kyron | iCEBrkr, so * will capture inband signaling and take care of it all ? |
| 15:45.06 | zobia | hello if i dial a number ,but after i hangup another side is still ringing for long time. anyone knows why this happeneed? |
| 15:45.06 | drmessano | kyron, how's that book reading coming along? |
| 15:45.08 | kyron | jameswf, LOL |
| 15:45.10 | Zeeek | Park *THIS* |
| 15:45.47 | kyron | drmessano, think I downloaded it somewhere :P I just wanted to get fammiliar with the _real_ term (parking the calls) |
| 15:45.51 | iCEBrkr | kyron: Last I tired it. Yes. |
| 15:46.02 | kyron | ok, cool, I'll check it out |
| 15:46.19 | iCEBrkr | Mr. Smith, you have a call on park 1. |
| 15:46.23 | drmessano | "I got akerisk running, but I have one question. Can I call other people or do they need Akerisk too, like both of us needing MySPACE IM to call?" |
| 15:46.29 | zobia | any one can help me out? |
| 15:46.35 | drmessano | ~idk |
| 15:46.38 | iCEBrkr | drmessano: LOLS |
| 15:46.50 | drmessano | ~book |
| 15:46.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 15:46.58 | drmessano | Its more than just a freakin download |
| 15:47.06 | kyron | drmessano, LOL |
| 15:47.13 | casix | I've make a sip user to connect two asterisks. In the asterisk that is receiving the call I've configured the user as a type=user. But it doesn't work. But if I configure it as type=friend it works. what is wrong?? the type=user is for receiving calls, no? |
| 15:47.47 | kyron | drmessano, what, you mean I can't learn through osmosis? |
| 15:47.59 | ManxPower | casix: type=user is for phone->asterisk calls, type=peer is for asterisk->phone calls, and friend is both |
| 15:48.37 | [TK]D-Fender | kyron: Funny you should mention that... |
| 15:48.38 | [TK]D-Fender | ~osmosis |
| 15:48.39 | jbot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 15:48.47 | jameswf | ~texting |
| 15:48.48 | jbot | IDK my BFF Jill |
| 15:48.53 | drmessano | kyron: No offense, but I love guys like you.. When someone has to run to google to learn how to reboot, or jump on IRC for help when the boss asks how to put a call on hold, it just raises what I can ask for when I apply for a job :) |
| 15:49.01 | kyron | [TK]D-Fender, you own dude! |
| 15:49.10 | drmessano | So, dude, I love you |
| 15:49.16 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
| 15:49.47 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:49.55 | drmessano | you're my, idk, bff kyron |
| 15:50.10 | kyron | drmessano, yeah, and I'm usually the first one to scream out RTFM |
| 15:50.16 | jameswf | omfg what is this linux like omfg you mean no hello kitty wallpaper omfg how do you work wihout a mouse omfg omfg |
| 15:50.56 | iCEBrkr | jameswf: lol |
| 15:50.57 | drmessano | kyron: sure, sure.. how's that PDF comin along? ;) |
| 15:50.59 | casix | ManxPower: yes but if a asterisk recieve a call why if I configure it with type=user don't work but if I configure it with type=friend it works. It would have to work, no? |
| 15:51.05 | kyron | jameswf, ironically, the mouse is the first thing I unplug when giving the BASH class, showing the students how it's such a waste of time when you have a half-well designed WM |
| 15:52.00 | kyron | drmessano, will you leave me alone if I rewrite chan_sip to be less convoluted? |
| 15:52.07 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:52.11 | drmessano | lol |
| 15:52.15 | drmessano | Yes.. completely |
| 15:52.27 | kyron | ok, so I don't have time to do (now...but seriously) that but _-will-_ read the book |
| 15:52.31 | ManxPower | You'll pry cut and paste, cut and paste, cut and paste out of my cold dead hands, cold dead hands, my cold dead hands |
| 15:52.32 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 15:52.46 | kyron | ManxPower, O_o... |
| 15:53.04 | kyron | ManxPower, you the type that uses Ctrl-C/V to cut/paste?... |
| 15:53.05 | [TK]D-Fender | kyron: There are at least 2 other chan_sip replacements being "worked on". Take a number |
| 15:53.11 | jameswf | you can cut and paste in nano |
| 15:53.27 | kyron | jameswf, pffff...lame.. use vi |
| 15:53.35 | jameswf | Maybe I will make a hello kitty theme patch for freepbx :) |
| 15:53.37 | kyron | nano's annoying |
| 15:54.08 | kyron | [TK]D-Fender, won't waste my time there then ;) |
| 15:54.09 | errr | jameswf: what about hanna montana? |
| 15:54.24 | jameswf | ohhh yeah mmm jail bait :) |
| 15:54.28 | drmessano | if there was a freaking batman skin for FreePBX, damn.. |
| 15:54.31 | drmessano | I would LOVE that |
| 15:54.37 | errr | lol |
| 15:54.44 | jameswf | kiddie porn is only okay if its animae |
| 15:54.52 | drmessano | Batman FTMFW |
| 15:55.05 | ManxPower | kyron: you know that ctrl-c and ctrl-v don't copy and paste on unix |
| 15:55.11 | jameswf | or is it kitty porn |
| 15:55.16 | ManxPower | on windows, I use those, of course. |
| 15:55.21 | Qwell | jameswf: animeow |
| 15:55.23 | drmessano | ManxPower: He hasn't gotten that far in his dummies book |
| 15:55.24 | [TK]D-Fender | errr: tahts just another sign that Hillary Dufff is getting old... |
| 15:55.25 | Qwell | (TM) |
| 15:55.38 | lunaphyte_ | oooh, like www.livenudecats.com !? |
| 15:55.39 | Zeeek | In about an hour, you can join us in the *sterisk Lounge for a cool one and a talk about VoIP: http://voipusersconference.org |
| 15:55.57 | ChkDigit | "Fritz the Cat" has to be the best kitty p0rn goin' |
| 15:56.05 | kyron | ManxPower, you're so yesterday, ever noticed that Ctrl-insert and shift-insert are much faster and ergonomic to use... |
| 15:56.08 | jameswf | you know what happens to disney stars who get old..... 2 words Britney Spears |
| 15:56.20 | Qwell | kyron: highlight, middle-click |
| 15:56.29 | Zeeek | IRC #voip-users-conference will fill the void left by the WGA strike |
| 15:56.37 | kyron | Qwell, in some circumstances, yes. |
| 15:56.49 | kyron | jameswf, lol |
| 15:56.53 | ManxPower | Zeeek: well at least for the comedy writers. |
| 15:57.19 | kyron | Qwell, even my GF gets frustrated when switching back to do some translation under windows |
| 15:57.21 | Zeeek | exactly. Jay leno's audience listens to the VUC every Friday! |
| 15:57.32 | Qwell | kyron: don't do that then |
| 15:57.48 | ChkDigit | Anyone play with an unruly Mediatrix 1204 ATA? I'm having it spew out eFAILURE_REASON_UNREACHABLE to syslog on incoming calls from cell phones, but not from landlines. |
| 15:57.49 | Zeeek | 63 minutes |
| 15:58.47 | ChkDigit | That cell phone calls act differently than others is a total WTF to me. |
| 15:58.51 | *** part/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 15:59.15 | kyron | Qwell, I don't do the "Windows" stuff (ie: install illegal version + downlaod illegal software + winder why my machine is so slow and Internet connection is suddenly so ...slow) |
| 16:01.23 | ManxPower | I use Windows instead of linux on my personal machine for 2 reasons, one of them is not a common reason |
| 16:02.24 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 16:03.36 | lunaphyte_ | ManxPower: you are a sadomasochist? |
| 16:03.40 | *** join/#asterisk datachomper (n=russ@75.146.194.61) |
| 16:03.53 | datachomper | Did anybody elses Polycom 550's randomly stop working this weekend? |
| 16:04.09 | ManxPower | lunaphyte: Getting linux to work well on any laptop I have owned was a complicated and non-fun thing to do. |
| 16:04.33 | drmessano | As soon as Adobe Audition runs on Linux, we'll talk |
| 16:04.34 | ManxPower | and really, %90 of my work is done via SSH anyway. |
| 16:04.35 | drmessano | lol |
| 16:04.38 | lunaphyte_ | agreed. it's always improving though. |
| 16:04.58 | ManxPower | lunaphyte: I do tend to use cross-platform software, even on Windows. |
| 16:05.08 | lunaphyte_ | yeah, same here. |
| 16:05.33 | jblack | The screenshot looks like audacity |
| 16:05.39 | ManxPower | GAIM/Pidgin, jEdit, Firefox, Thunderbird, Open office, etc. |
| 16:05.40 | drmessano | Ehhh |
| 16:05.41 | drmessano | no |
| 16:05.50 | ManxPower | PuttySSH for SSH |
| 16:05.58 | drmessano | Audacity is a Yugo, Audition is an Escalade |
| 16:06.01 | lunaphyte_ | i try to use open sores software as much as possible. |
| 16:06.04 | lunaphyte_ | :p |
| 16:06.11 | *** join/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 16:06.34 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:06.43 | jake[work] | both will get you there - but the ride is so much better |
| 16:06.46 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:07.03 | zobia | @Qwell: i used exten =>_X!,1,hangup in a context named "notcoverate" to hangup all the bad extensions numbers, but do include=> this in the default context. but eventrullyi found each good extensions also got delayed becasause this .X! do you think i made anything wrong? |
| 16:07.12 | [TK]D-Fender | lunaphyte : Don't forget to refill your penicillin prescription! |
| 16:07.16 | drmessano | ewww |
| 16:07.20 | kyron | man_o_magic, ATI video card^ |
| 16:07.21 | kyron | ? |
| 16:07.23 | zobia | @Qwell: sorry it named "notcoverage" |
| 16:07.24 | kyron | oops |
| 16:07.29 | drmessano | Audacity wouldn't handle 5 minutes at my work |
| 16:07.30 | kyron | ManxPower, ATI video card? |
| 16:07.39 | drmessano | It's not even worth mentioning |
| 16:07.43 | ManxPower | lunaphyte: I have some audio/video software that is not open source. TMPGEnc stuff for conversion and DVD burning |
| 16:07.59 | Daviey | drmessano: Audacity isn't _that_ bad |
| 16:08.03 | ManxPower | kyron: No idea what video card, I use a laptop |
| 16:08.04 | lunaphyte_ | i'm allergic. |
| 16:08.13 | Qwell | ManxPower: I tried to use the demo of that a few times.. is the paid version any good? |
| 16:08.16 | drmessano | Actually, yes it is |
| 16:08.19 | Qwell | the demo was terrible |
| 16:08.30 | drmessano | Audacity is very feature lacking |
| 16:08.34 | kyron | ManxPower, ah, ok, you'd be a candidate for Windows indeed (or kubuntu :P ) |
| 16:08.38 | ManxPower | Qwell: It's the best thing I've tried, but that's not saying much -- the entire class of software sucks. |
| 16:08.40 | drmessano | Again, not made at all for professional use |
| 16:08.47 | Qwell | ManxPower: mencoder |
| 16:08.58 | Daviey | no, audacity has quite a lot of features - but really lacks a decent GUI |
| 16:09.14 | drmessano | Daviey: I have used it.. It wont cut it |
| 16:09.19 | ManxPower | Qwell: all open source stuff I tried gave me horrible results, most of them don't support AVI well, nor WMV |
| 16:09.21 | [TK]D-Fender | Anyone know a good OSS Audio/MIDI recording studio? |
| 16:09.39 | zobia | <PROTECTED> |
| 16:09.43 | [TK]D-Fender | Audacity only does audio, Rosegarden only does MIDI |
| 16:09.48 | ManxPower | Qwell: I suspect at lower bit rates OS stuff might be good, but I was wanting 2Mbps A/V streams. |
| 16:10.14 | drmessano | [TK]D-Fender: I think Reaper runs uner Wine ;) |
| 16:10.24 | kyron | and use cinelerra for video |
| 16:10.35 | [TK]D-Fender | drmessano: I'm fully happy running a windows one. |
| 16:10.36 | errr | [TK]D-Fender: rosebud maybe? |
| 16:10.41 | drmessano | Reaper then |
| 16:10.47 | drmessano | It's pretty solid |
| 16:10.48 | errr | oh rosegarden, but you mentioned it |
| 16:10.50 | errr | nm |
| 16:10.56 | ManxPower | Qwell: at 2Mbps everything I tried either barfed or gave results little better than 512Kbps |
| 16:11.02 | drmessano | It's Justin Frankel's new project, if you're not familiar |
| 16:11.05 | Daviey | cinelerra = v. dated and nasty attitude from the dev. |
| 16:11.07 | [TK]D-Fender | REAPER, Music Software from Winamp Creator, Hits 1.x and No Longer Free |
| 16:11.10 | [TK]D-Fender | bleh |
| 16:11.15 | Daviey | Kino is starting to look good |
| 16:11.16 | drmessano | $50 for "personal use" |
| 16:11.20 | drmessano | It's at 2.x now |
| 16:11.20 | lunaphyte_ | it's too bad, kind of. i actually don't mind quite a few windows programs out there. it's having to run the os underneath them that sucks. |
| 16:11.26 | Daviey | All video editing is crap on Linux, sadly |
| 16:11.34 | drmessano | He has dual licensing for personal |
| 16:11.38 | kyron | Daviey, being here, you should be used to that :P |
| 16:12.04 | ManxPower | Oh, I also use VideoReDo Plus, mostly for commercial detection. |
| 16:12.05 | drmessano | I would replace all my DAW's with Linux if there was something that would cut it |
| 16:12.20 | man_o_magic | Hey guys, I fubar'ed my asterisk server, not a lot, just a little bit. It runs and everything, but I can't get console anymore (with asterisk -r). It says it's missing .ctl file in /var/run. Anybody has a clue what to do? |
| 16:12.38 | Daviey | man_o_magic: run asterisk :) |
| 16:12.47 | man_o_magic | it is already |
| 16:12.52 | Daviey | man_o_magic: sure? |
| 16:12.53 | kyron | drmessano, daw? |
| 16:13.01 | man_o_magic | 100% |
| 16:13.10 | Daviey | man_o_magic: as root? |
| 16:13.11 | kyron | man_o_magic, ps ax|grep asterisk |
| 16:13.11 | ManxPower | man_o_magic: no control file = asterisk not running. not being able to access the control file = permissions problem |
| 16:13.17 | kyron | man_o_magic, ps axu|grep asterisk |
| 16:13.25 | kyron | man_o_magic, added the u :P |
| 16:13.51 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-194-159.bflony.east.verizon.net) |
| 16:14.00 | Daviey | man_o_magic: you have to respond faster if you want our attention |
| 16:14.05 | Daviey | We have low attention spans |
| 16:14.08 | man_o_magic | root@pbx:~ $ ps axu|grep aster |
| 16:14.10 | man_o_magic | asterisk 2720 0.0 0.3 33632 14116 ? S 00:45 0:03 /usr/sbin/httpd |
| 16:14.11 | man_o_magic | asterisk 2721 0.0 0.3 33560 13484 ? S 00:45 0:02 /usr/sbin/httpd |
| 16:14.13 | man_o_magic | asterisk 2722 0.0 0.3 33620 13536 ? S 00:45 0:02 /usr/sbin/httpd |
| 16:14.14 | man_o_magic | asterisk 2723 0.0 0.3 33524 13828 ? S 00:45 0:03 /usr/sbin/httpd |
| 16:14.16 | man_o_magic | asterisk 2724 0.0 0.3 33524 13864 ? S 00:45 0:03 /usr/sbin/httpd |
| 16:14.17 | SuPrSluG | hello |
| 16:14.18 | man_o_magic | asterisk 2725 0.0 0.3 33592 13532 ? S 00:45 0:02 /usr/sbin/httpd |
| 16:14.19 | man_o_magic | asterisk 2726 0.0 0.3 33548 13476 ? S 00:45 0:02 /usr/sbin/httpd |
| 16:14.20 | man_o_magic | asterisk 2727 0.0 0.3 33596 13920 ? S 00:45 0:03 /usr/sbin/httpd |
| 16:14.21 | kyron | KICK HIM! |
| 16:14.22 | man_o_magic | root 2902 0.0 0.0 4500 632 ? S 00:45 0:00 /bin/bash /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 16:14.23 | man_o_magic | asterisk 2908 0.1 0.3 30508 11324 ? Sl 00:45 1:07 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 16:14.24 | Daviey | erm |
| 16:14.25 | kyron | PLEASE! |
| 16:14.25 | man_o_magic | asterisk 2993 0.0 0.0 4496 568 ? S 00:46 0:00 -bash -c cd /var/www/html/panel && /var/www/html/panel/safe_opserver & |
| 16:14.26 | errr | please no flooding.. |
| 16:14.26 | Daviey | ~pb |
| 16:14.27 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:14.27 | man_o_magic | asterisk 2994 0.0 0.0 4448 1084 ? S 00:46 0:00 sh /var/www/html/panel/safe_opserver |
| 16:14.28 | Nivex | PASTEBIN! |
| 16:14.29 | man_o_magic | asterisk 2996 0.1 0.2 12368 8640 ? S 00:46 0:56 /usr/bin/perl -w /var/www/html/panel/op_server.pl |
| 16:14.29 | kyron | someone! |
| 16:14.31 | man_o_magic | root 6134 0.0 0.0 3896 684 pts/0 S+ 11:13 0:00 grep aster |
| 16:14.36 | *** mode/#asterisk [+b %man_o_magic!*@*] by twisted |
| 16:14.37 | drmessano | WTF |
| 16:14.37 | SuPrSluG | pastebin |
| 16:14.39 | drmessano | STOP |
| 16:14.44 | drmessano | heh |
| 16:14.45 | jake[work] | haha |
| 16:14.46 | Daviey | Christ |
| 16:14.46 | ManxPower | kill it! kill it! |
| 16:14.47 | twisted | man_o_magic: use pastebin. |
| 16:14.50 | kyron | ~kick man_o_magic |
| 16:14.51 | jbot | ACTION kicks man_o_magic |
| 16:14.52 | *** join/#asterisk cappslocke|work (n=cappsloc@72.16.231.34) |
| 16:15.00 | *** mode/#asterisk [-b %man_o_magic!*@*] by twisted |
| 16:15.11 | twisted | no need to kick, i squealched |
| 16:15.13 | ManxPower | twisted: Isn't that "use pastebin or we'll stick a knife in your heart and twist it"? |
| 16:15.18 | drmessano | Pasting 5 lines, uncool.. pasting 20, tool |
| 16:15.29 | errr | lol |
| 16:15.31 | [TK]D-Fender | twisted: Thanks.... saves me the effort:) |
| 16:15.32 | kyron | twisted, wow, a term I haven't head in a while... SSB power :P |
| 16:15.48 | man_o_magic | sorry, I didn't know |
| 16:15.55 | twisted | [TK]D-Fender :) |
| 16:15.57 | kyron | man_o_magic, you didn't use my command line correctly |
| 16:15.57 | twisted | kyron: hehe |
| 16:16.07 | kyron | man_o_magic, I had a |grep in there |
| 16:16.18 | Daviey | lets not try again, eh :) |
| 16:16.27 | man_o_magic | tell me what to do again |
| 16:16.29 | errr | kyron: he did too |
| 16:16.34 | errr | man_o_magic: you did it right |
| 16:16.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:16.50 | twisted | just don't paste into the channel. |
| 16:16.51 | errr | just dont flood the room next time :) |
| 16:16.51 | drmessano | Crank down the RF gain |
| 16:17.02 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 16:17.04 | kyron | Q about book: is there much fluff in there in the forms of "this is a cool and we're coll" or does it get to the point quickly...I am starting to read it and have a queezy feeling about the 600+ pages.. |
| 16:17.29 | Daviey | kyron: year the first 300 pages are story time :) |
| 16:17.34 | Daviey | yeah* |
| 16:17.37 | drmessano | Nothing like shortcuts |
| 16:17.42 | kyron | man_o_magic, I am sorry, you did it right, errr is right |
| 16:17.44 | drmessano | "Can I get the cliff notes" |
| 16:17.49 | twisted | lol |
| 16:18.05 | man_o_magic | so why is it not connecting? |
| 16:18.16 | drmessano | I would do that for lmadsen.. |
| 16:18.19 | errr | man_o_magic: try doing it as root |
| 16:18.26 | drmessano | TFOT: The Cliff Notes |
| 16:18.32 | Qwell | tl;dr |
| 16:18.38 | twisted | Qwell :) |
| 16:18.38 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 16:18.42 | drmessano | "1. open Internet Explorer to Trixbox.org" |
| 16:18.43 | man_o_magic | errr, i am logged in as root |
| 16:18.43 | kyron | man_o_magic, join #trixbox |
| 16:18.46 | drmessano | etc |
| 16:18.57 | kyron | drmessano, lol |
| 16:19.12 | drmessano | kyron man, i'm telling you |
| 16:19.21 | drmessano | You got my job so secure |
| 16:19.24 | kyron | daven, seriously |
| 16:19.26 | kyron | damned |
| 16:19.35 | drmessano | "Can I just read one chapter and I R ADMIN?" |
| 16:19.35 | kyron | Daviey, seriously... story time |
| 16:19.39 | *** join/#asterisk bts3685 (n=sanerb@69.17.28.131) |
| 16:20.02 | errr | drmessano: isnt there a cartoon like that? |
| 16:20.14 | drmessano | Has to be.. |
| 16:20.18 | kyron | drmessano, I just spent the past 2 months reading about 10 books on HPC computing and parallel processing and see a pattern of fluff... |
| 16:20.21 | kyron | that's all |
| 16:20.45 | errr | drmessano: I R Baboon or something like that :) |
| 16:21.08 | man_o_magic | /join #trixbox |
| 16:21.10 | SuPrSluG | i'm having an issue with my voicemail being truncated. I watch as the message is recorded and all seems good. But, when the message is sent or played back it never more than 1 sec. long. Any ideas why this is happening. Permis |
| 16:21.13 | drmessano | I just spent the last 9 days reading 3 encyclopedias on "Advanced Russian Spacecraft Design", so i'm browsing comic books |
| 16:21.19 | drmessano | See, I can do that too :) |
| 16:21.25 | FlatFoot | errr , http://simdes.org/img/ir1600.png |
| 16:21.39 | *** part/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 16:21.43 | FlatFoot | I R Baboon , very weird cartoon |
| 16:21.48 | errr | FlatFoot: rofl, I love that show =) |
| 16:21.49 | kyron | drmessano, job security comes with Windows admins, not Ux admins |
| 16:21.55 | drmessano | ahh |
| 16:21.59 | *** join/#asterisk ctp_ (n=ctp@brsg-d9bed8ae.pool.mediaWays.net) |
| 16:22.00 | *** join/#asterisk UnixDog (n=unixdog@ppp-69-238-217-105.dsl.irvnca.pacbell.net) |
| 16:22.27 | ctp_ | hi folks. which livecd do you recommend to take first steps with asterisk? |
| 16:22.35 | Qwell | ctp_: debian |
| 16:22.35 | errr | kyron: yeah since most colleges pump out windows admins faster than most people can read a linux for dummies book |
| 16:22.39 | kyron | ctp_, LFS |
| 16:22.51 | drmessano | errr: or an Asterisk Cliff Notes book? |
| 16:22.53 | Daviey | ctp_: Anything Deb based.. Debina, Ubuntu et all |
| 16:22.59 | errr | lol |
| 16:23.18 | ctp_ | Qwell: distro of my choice ;-) but i mean without installing all the services to get an overview about asterisk |
| 16:23.36 | errr | ctp_: fedora has a live cd and has asterisk 1.4 in yum repos |
| 16:23.37 | Daviey | ctp_: why not just install? |
| 16:23.45 | kyron | drmessano, irony: ctp_, read da book |
| 16:23.51 | kyron | ~book |
| 16:23.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:23.56 | kyron | mwehehehe |
| 16:24.08 | Daviey | Ubuntu has a live cd - and asterisk in the repo's - and it's not fscking rpm based |
| 16:24.16 | drmessano | ok, while I am so enjoying this spirited chat, I need to finishing splicing this DNA and go work on my cancer cure.. See ya after I get back from Area 51! |
| 16:24.23 | errr | drmessano: nothing wrong with rpm. |
| 16:24.27 | errr | Daviey* |
| 16:24.41 | FlatFoot | errr , Hong Kong Phooey ... Rules |
| 16:24.44 | Daviey | errr: err, there is |
| 16:24.47 | kyron | drmessano, actually, have you ever read a Microsoft product book? |
| 16:24.49 | Daviey | ^ see what i did there |
| 16:25.01 | drmessano | I have actually |
| 16:25.05 | errr | FlatFoot: yeah I tvio that :) |
| 16:25.09 | drmessano | I've read a lot of books |
| 16:25.15 | drmessano | One 1 was a dummies |
| 16:25.19 | drmessano | Only* |
| 16:25.20 | ctp_ | i've found this one: http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM. any of them you recommend? |
| 16:25.32 | FlatFoot | errr , its on about 3am here in the uk . Good for beer nights |
| 16:25.34 | *** join/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
| 16:25.39 | kyron | didn't you feel numbed in the brain after the first self-glorifying chapter? |
| 16:25.48 | zobia | DEBUG_THREADS |
| 16:25.49 | zobia | DONT_OPTIMIZE |
| 16:25.53 | drmessano | Not really.. |
| 16:26.11 | zobia | hello how to make debug_threads and don_optimize when compile 1.4/ |
| 16:26.12 | zobia | ? |
| 16:26.15 | [TK]D-Fender | CTP : NONE |
| 16:26.26 | putnopvut | zobia: make menuselect -> Compiler flags |
| 16:26.39 | fetcher | Are there any 2FXS + 1FXO ATA's on the market? Or 2/2? |
| 16:26.40 | drmessano | I'm actually a very fast learning and I absorb material very well.. helps to be receptive to it and not look for Cliff Notes :) |
| 16:26.48 | drmessano | Crap, and a bad typist |
| 16:26.55 | drmessano | s/learning/learner |
| 16:26.57 | Daviey | FlatFoot: 3am in the UK, O RLY |
| 16:27.14 | drmessano | My brain moves too fast for my fingers.. I need a buttonless keyboard |
| 16:27.16 | kyron | drmessano, hehehe |
| 16:27.17 | Daviey | doh, /me apologies |
| 16:27.18 | [TK]D-Fender | fetcher: Not really |
| 16:27.37 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 16:27.50 | drmessano | Anyway.. I'm out |
| 16:27.52 | zobia | putnopvut: i use make menuselect , it said Install ncurses to use the menu interface! and i already yum install ncurses |
| 16:27.59 | kyron | drmessano, ok, nuff book throwing I go read |
| 16:28.02 | fetcher | how about a USB FXO? |
| 16:28.02 | kyron | laters |
| 16:28.10 | [TK]D-Fender | zobia: You need the "-devel" as well |
| 16:28.17 | putnopvut | zobia: ^^^^ |
| 16:28.28 | fetcher | normally a TDM400 would be the obvious thing, but this is for an embedded box that lacks PCI slots :( |
| 16:28.30 | [TK]D-Fender | fetcher: No |
| 16:28.41 | zobia | [TK]D-Fender: putnopvut: thank you , let me try |
| 16:28.44 | [TK]D-Fender | fetcher: 2X2 = 2x SPA-3102 |
| 16:29.59 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 16:30.10 | zobia | [TK]D-Fender: putnopvut: i yum install ncurses-devel. still same message ask for ncurses |
| 16:30.28 | putnopvut | You need to rerun the configure script. |
| 16:30.30 | Qwell | just a note, but if the above is for debugging chan_sccp, it won't work. |
| 16:30.43 | Qwell | since it's built out of tree |
| 16:30.54 | Zeeek | in just 30 minutes: VoIP Users Unite |
| 16:30.59 | FlatFoot | Daviey: yeah one of the cartoon network channels used to show about 3am , mind you i gave sky up a while ago so it might have changed |
| 16:31.29 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-212-195-198-128.d4.club-internet.fr) |
| 16:31.45 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 16:31.50 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:31.58 | ManxPower | I thought it was VoIP Users Untie |
| 16:32.12 | zobia | putnopvut:let me try again |
| 16:32.51 | [T]ank | is there documentation from digium on what is required to do asterisk professionally? ie a reseller license or agreement or fees. things like that? |
| 16:33.03 | zobia | putnopvut:i try /configure again and make menuselect. same messages |
| 16:33.13 | Qwell | zobia: make dist-clean |
| 16:33.48 | [T]ank | Daviey: doing research for my company |
| 16:34.03 | [T]ank | you are correct... we are not ready. |
| 16:34.11 | ManxPower | [T]ank: Asterisk is open source, you can do anything the GPL says you can do. HOWEVER, the names "Digium" and "Asterisk" do have some restrictions on their use. Contact Digium for details. |
| 16:34.29 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:34.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:34.42 | *** join/#asterisk the_5th_wheel (n=edd@dsl-242-161-240.telkomadsl.co.za) |
| 16:34.43 | zobia | @Qwell: i need to do ./configure make dist-clean then make install? |
| 16:34.54 | Qwell | make dist-clean, then follow all other steps |
| 16:35.00 | [T]ank | ManxPower: thank you |
| 16:35.00 | *** join/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 16:35.05 | the_5th_wheel | hi. does anyone know what causes this message? i seem to miss alot of calls since this popped up |
| 16:35.08 | the_5th_wheel | <PROTECTED> |
| 16:35.31 | ManxPower | the_5th_wheel: try using G1 instead of g1 in your outbound Dial() lines |
| 16:35.48 | zobia | @Qwell:ok , let me try |
| 16:35.57 | ManxPower | or whatever ground number you use. |
| 16:36.06 | Qwell | ground? group |
| 16:36.18 | ManxPower | I can't type today. |
| 16:37.51 | tzanger | coppice: I'm reading that book you recommended; I think problem #1 with me is that I've *always* looked at audio as discrete... everyone keeps talking about continuous and discrete and the huge differences between them and I don't "get it" -- the book is making it a little clearer |
| 16:38.11 | *** part/#asterisk bts3685 (n=sanerb@69.17.28.131) |
| 16:38.44 | tzanger | basically with discrete signals and and the sample time, you can make a good guess as to what the frequency components are, but you can't tell for sure because you might be looking at signals above the nyquist frequency... but if you're doing that you're screwed to begin with |
| 16:38.55 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 16:39.29 | zobia | @Qwell:thank you , it works |
| 16:39.54 | zobia | @Qwell: this can let it through out a coredump when there's crash of asterisk? |
| 16:40.15 | *** join/#asterisk murdmath (n=vircuser@209.181.82.1) |
| 16:41.18 | zobia | @Qwell: hello. i upgrade the my asterisl to 1.4 as you suggested. but i have new problem for my sccp, can you help? |
| 16:42.03 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:43.31 | the_5th_wheel | ManxPower: nothing changed. how can that be an issue/ |
| 16:43.34 | russellb | zobia: using chan_skinny? |
| 16:43.43 | Qwell | russellb: he says chan_sccp.. |
| 16:43.45 | man_o_magic | @all, would somebody be willing to help me? I still can't get console for asterisk. The server IS running -- can make calls from my IP phone |
| 16:43.49 | zobia | useing chan_sccp |
| 16:43.54 | russellb | use chan_skinny |
| 16:44.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:44.45 | zobia | @russellb: let me find . if i use chan_skinny do i need to change my config of existing sccp.conf? |
| 16:44.51 | Qwell | zobia: yes |
| 16:45.20 | ManxPower | the_5th_wheel: if a call comes in on channel 1 at the same time you are trying to send a call out on channel 1. the calls collide. |
| 16:45.25 | zobia | @Qwell: thanks. is it big changes? |
| 16:45.28 | Qwell | yes |
| 16:45.34 | Qwell | it's completely different |
| 16:45.47 | ManxPower | G1 says "start at the highest numbered channel, not the lowest numbered channel |
| 16:45.58 | [TK]D-Fender | man_o_magic: "asterisk -r" as root doesn't work? |
| 16:46.06 | zobia | @Qwell:god. chan_skinny is better than chan_sccp? |
| 16:46.18 | Qwell | much better |
| 16:46.36 | man_o_magic | [TK]D-Fender: no it doesn't... :( |
| 16:46.43 | Zeeek | IRC #voip-users-conference and http://VoipUsersConference LIVE in 10 minutes. Be there. Please? Ok. |
| 16:46.48 | Zeeek | dot org |
| 16:46.53 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 16:47.45 | [TK]D-Fender | man_o_magic: Go prove that the PID is where its supposed to be |
| 16:48.34 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 16:48.43 | Zeeek | http://VoipUsersConference.org is open now |
| 16:50.18 | anonymouz666 | Zeeek: [TK]D-Fender will play guitar as MOH |
| 16:50.37 | Qwell | anonymouz666: format_midi? |
| 16:50.46 | anonymouz666 | heh |
| 16:51.47 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:52.19 | tzanger | coppice: this book is good... it tells me how to "say" the equations, something I've never known |
| 16:52.32 | the_5th_wheel | ManxPower: i cant find where that is defined |
| 16:52.38 | tzanger | e.g. The term X1(m) is read as "the spectral sequence X sub one of m." |
| 16:52.54 | the_5th_wheel | ManxPower: and there shouldnt be any outgoing calls on the isdns |
| 16:53.08 | ManxPower | the_5th_wheel: where what is defined? |
| 16:54.58 | the_5th_wheel | ths g1 thing |
| 16:56.14 | neoalex | just curious, is there any way to receive faxes with chan_mobile ? |
| 16:56.16 | [TK]D-Fender | the_5th_wheel: your Dial statments |
| 16:56.19 | ManxPower | group= in /etc/asterisk/zapata.conf |
| 16:56.28 | Qwell | neoalex: you *could*, I suppose... but no |
| 16:56.36 | [TK]D-Fender | neoalex: it has nothing to do with faxing |
| 16:56.37 | ManxPower | neoalex: how would you send a fax from a movile phone? |
| 16:56.47 | ManxPower | and a mobile phone |
| 16:56.54 | Qwell | ManxPower: you can receive a fax call |
| 16:57.02 | ManxPower | or more specifically from a BLUETOOTH headset |
| 16:57.03 | Qwell | can place one too, just pass it through |
| 16:57.12 | neoalex | ManxPower: you could use the bluetooth phone as a modem and then you could send faxes I think |
| 16:57.22 | ManxPower | Qwell: I guess chan_mobile does not do what I think it does. |
| 16:57.29 | neoalex | but yes... that's what I thought it's only being seen as a handset |
| 16:57.34 | ManxPower | neoalex: you have done so before? |
| 16:57.39 | Qwell | ManxPower: it can do both fxo and fxs type of connections |
| 16:57.56 | neoalex | no, but someone has: http://navasgrp.home.att.net/tech/cingular/fax.htm |
| 16:57.56 | Qwell | headset ~= fxs, phone ~= fxo |
| 16:58.02 | the_5th_wheel | ManxPower: http://pastebin.div0.co.za/results/G2AA6GCF6.html <-- this is my zapta.conf. i still dont see what needs to be changed |
| 16:58.03 | neoalex | so in theory it's possible |
| 16:58.15 | Qwell | neoalex: there *is* a fax profile in bluetooth, you *could* implement that in chan_mobile |
| 16:58.23 | Qwell | it will not be easy, but it is certainly possible |
| 16:58.35 | [TK]D-Fender | the_5th_wheel: your Dial statments <------------ |
| 16:58.39 | ManxPower | the_5th_wheel: you change it in extensions.conf on the damn Dial line you use to dial out. |
| 16:59.12 | ManxPower | Qwell: Next thing you'll tell me is that Microsoft is buying Yahoo or some sillyness like that. |
| 16:59.15 | ManxPower | 8-) |
| 16:59.18 | fetcher | how the Grandstream HT-503? Any issues with its FXO port? |
| 16:59.26 | Qwell | fetcher: it's grandstream |
| 16:59.29 | ManxPower | ~gs |
| 16:59.29 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:59.36 | neoalex | Qwell: you mean rewrite chan_bluetooth to offer fax service not just handsfree service |
| 16:59.46 | Qwell | neoalex: chan_mobile, but yes |
| 17:00.18 | neoalex | ok... well I'm not that desperate yet |
| 17:00.28 | *** join/#asterisk rikardok (n=asdjna@210.16.52.230) |
| 17:00.31 | *** join/#asterisk chavigny (n=nrp@c-67-171-147-26.hsd1.or.comcast.net) |
| 17:00.44 | ^Migs^ | I'm setting up some VoIP intercoms for a school. I'll need between 30-40. What's a good brand/model? |
| 17:00.44 | Qwell | neoalex: there is also a printer profile |
| 17:00.46 | chavigny | hi:> |
| 17:02.04 | FlatFoot | any one use FireBox f/walls ? |
| 17:02.21 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 17:02.49 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:05.15 | rikardok | anone know how to achieved this?? "ast2<---->iax2<--->ast1---->sip/zap---->outgoing calls" |
| 17:08.43 | x86 | rikardok: that's about the simplest setup there is ;) |
| 17:09.08 | *** join/#asterisk rajiv_ (n=rajiv@gentoo/developer/rajiv) |
| 17:09.11 | x86 | rikardok: first, peer the two asterisk servers via iax.conf |
| 17:09.48 | x86 | rikardok: then, make sure that whatever context on ast1 that you put the peer definition for ast2 in has access to your outbound dial plan |
| 17:09.54 | x86 | rikardok: easy cheesy ;) |
| 17:12.46 | *** part/#asterisk Weetos (i=willy@mail.catalise.fr) |
| 17:16.32 | rikardok | got that chief |
| 17:17.07 | [TK]D-Fender | ^Migs^: "voip intercoms"? |
| 17:18.09 | rikardok | then the dialplan of my ast2 would be exten=> _XXXXXXXXXXX,1,Dial(IAX2/${EXTEN}@iaxpeeraccount) |
| 17:22.51 | tzanger | nice, my dsl provider is going to get a 100meg xconnect to thinktel |
| 17:23.06 | *** part/#asterisk hendrixski (n=hendrixs@cpe-74-65-1-222.rochester.res.rr.com) |
| 17:24.53 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:27.45 | FlatFoot | tara all off for foster training |
| 17:30.21 | *** part/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 17:31.22 | *** join/#asterisk AndyGraybeal_ (n=andy@node246.34.251.72.1dial.com) |
| 17:31.33 | zobia | @russellb |
| 17:31.53 | zobia | @russellb: his i really got a big problem |
| 17:32.26 | zobia | i got message mutex '&chan->lock' freed more times than we've locked! and it's realtime system , i can notmake change to chan_skinny right now. |
| 17:32.36 | zobia | @russellb: can you help? |
| 17:38.44 | *** join/#asterisk philippel (n=p_lindhe@c-98-203-245-82.hsd1.wa.comcast.net) |
| 17:38.46 | scooby2 | can you do something like this? Set(DEST_COMPANY=${IF($[ ${ARG2} = _8XX]?company2:company1)}) |
| 17:39.17 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 17:39.21 | b11d | hello lads |
| 17:39.55 | scooby2 | trying to change DEST_COMPANY if their extension is in the 800's |
| 17:40.28 | philippel | can anyone here lend a hand getting Asterisk built with cdr_tds for an MSSQL backend CDR connection? I've got FreeTDS installed and working fine, I can connect to the database but when I try make menuselect it claims there are conflicts for this cdr choice? Asterisk 1.4 |
| 17:40.48 | *** join/#asterisk mustiy (n=hey@mailhost1.met-chem.com) |
| 17:42.23 | *** join/#asterisk wglenncamp (n=wglennca@adsl-155-210-131.owb.bellsouth.net) |
| 17:43.26 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:43.47 | wglenncamp | I need a polycom expert. I have a call group with 5 phones. When an inbound Zap call comes into the group, and a person is already on a call (on line 1 on an IP501), the call is cut out for a second on that phone. Almost like a call waiting, but not... Any ideas? |
| 17:46.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:49.01 | wglenncamp | Is there a way to set silent ring for line 2 if there is a call on line 1? |
| 17:49.39 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:49.47 | [TK]D-Fender | wglenncamp: Yes, you disable the CW beep in provisioning |
| 17:50.26 | wglenncamp | I see, but it's not beeping. That's where I am stumped. |
| 17:50.43 | wglenncamp | Where is that setting? Do you know off hand? |
| 17:51.35 | [TK]D-Fender | wglenncamp: Easily searchable in your admin guide. |
| 17:51.56 | [TK]D-Fender | wglenncamp: Is "line 2" a seperate registration? |
| 17:52.06 | wglenncamp | No. Same reg for all lines |
| 17:52.13 | [TK]D-Fender | wglenncamp: Or are you referring to jsut another line key associated with your first? |
| 17:52.40 | wglenncamp | No, each phone has 3 lines (same reg). |
| 17:52.44 | [TK]D-Fender | wglenncamp: Sound should be cut for a second for CW unless you've already disabled it. |
| 17:53.49 | wglenncamp | There isn't a sound. If the user is already in a call, and the phone rings again, it cuts out for a second. The user says they hear the ringing, but I haven't witnessed it first hand yet. |
| 17:54.06 | wglenncamp | Not a beep. Ringing.. Like what the phone is doing |
| 17:54.23 | wglenncamp | (Have I confused you enough yet?) :) |
| 17:57.07 | puzzled | hi |
| 17:58.03 | puzzled | what's the name of the application again that allows you to get dtmf digit input? I know WaitExten but iirc there's another |
| 17:58.18 | wglenncamp | okay, it looks like se.rt.1.callWait = 6.. Would I remove that from the .cfg file or set it to 0? |
| 17:59.06 | *** join/#asterisk ddunavant (n=David@pool-96-231-69-97.washdc.east.verizon.net) |
| 17:59.14 | [TK]D-Fender | wglenncamp: Thats what the admin guide is for... |
| 17:59.26 | [TK]D-Fender | puzzled: "core show application read" |
| 17:59.36 | wglenncamp | What the heck you think I'm reading?!?! It doesn't say.. |
| 18:00.04 | puzzled | [TK]D-Fender: doh, thanks |
| 18:01.41 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 18:01.46 | teknoprep | anyone here using voicepulse? |
| 18:01.56 | teknoprep | i keep getting dropped registration to there nyc or sfo servers |
| 18:02.10 | teknoprep | and it will not re-register unless i restart asterisk |
| 18:04.02 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 18:11.51 | *** join/#asterisk WorgiL (n=usta@85.106.181.223) |
| 18:12.50 | WorgiL | can anyone help me http://paste.ubuntu-nl.org/54366/ |
| 18:12.53 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 18:12.53 | WorgiL | thanks |
| 18:14.16 | WorgiL | my asterisk not running now can anyone help me pls ? |
| 18:16.17 | [TK]D-Fender | WorgiL: Go restart * |
| 18:16.57 | WorgiL | ../etc/init.d/asterisk restart |
| 18:16.57 | WorgiL | Stopping Asterisk PBX: asterisk. |
| 18:16.57 | WorgiL | Starting Asterisk PBX: asterisk. |
| 18:17.16 | WorgiL | .../etc/init.d/asterisk status |
| 18:17.16 | WorgiL | Asterisk PBX is stopped |
| 18:17.31 | scooby2 | check the logs? |
| 18:17.50 | WorgiL | how can i look scooby2 ? |
| 18:18.01 | scooby2 | usually in /var/log/asterisk |
| 18:18.08 | scooby2 | tail /var/log/asterisk/messages |
| 18:18.26 | [TK]D-Fender | WorgiL: start it MANUALLY and see what happens |
| 18:18.26 | WorgiL | [Feb 1 17:17:03] NOTICE[4360] cdr.c: CDR simple logging enabled. |
| 18:18.27 | WorgiL | [Feb 1 17:17:03] NOTICE[4360] loader.c: 165 modules will be loaded. |
| 18:18.27 | WorgiL | root@telekom:/usr/src/freepbx-2.3.1# |
| 18:18.34 | [TK]D-Fender | ~freepbx |
| 18:18.35 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:18.36 | [TK]D-Fender | ^^^^^^^ |
| 18:19.04 | scooby2 | freepbx != asterisk |
| 18:19.14 | WorgiL | [TK]D-Fender how can i do it ? |
| 18:19.46 | [TK]D-Fender | WorgiL: "asterisk -gvvvvvvc" |
| 18:20.57 | scooby2 | what am I doing wrong? trying to set DEST_COMPANY based on extension... Set(DEST_COMPANY=${IF($[ ${EXT} = _8XX]?company2:company1)}) |
| 18:21.10 | WorgiL | [TK]D-Fender, not started http://paste.ubuntu-nl.org/54368/ |
| 18:22.24 | [TK]D-Fender | WorgiL: try "noload => res_musiconhold.so" in modules.conf and see if that helps |
| 18:22.46 | [TK]D-Fender | scooby2: the fact that you can't compare to a PATTERN line that. |
| 18:22.57 | scooby2 | thats what i figured |
| 18:23.31 | [TK]D-Fender | scooby2: You can test for length and the first char naturally. |
| 18:23.46 | russellb | zobia: chan_sccp is not our code, so no, i can not help you |
| 18:25.07 | philippel | any takers to help getting cdr_tds.c built on 1.4? I'm thinkkng I must be missing something obvious? |
| 18:25.57 | WorgiL | [TK]D-Fender, not started looking same error http://paste.ubuntu-nl.org/54369/ |
| 18:26.37 | *** join/#asterisk Tond (n=t@CPE0014bf30c190-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 18:26.45 | [TK]D-Fender | WorgiL: Well thats it then, can't say what off, and FreePBX isn't supported here. Please use their support channel. |
| 18:27.18 | WorgiL | [TK]D-Fender, is errror about freepbx ? |
| 18:27.30 | Tond | Hi a quick question not directly related to asterik. I ahve Mysql installed nad with it came a system account called mysql. Should i ever change the password on that account? can't cause any system security threat? |
| 18:31.30 | fiXXXerMet | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf shows a few dialplan examples and they have a "description" field. When I do dialplan show context , I don't get any type of description... How do I define one? |
| 18:37.19 | [TK]D-Fender | fiXXXerMet: Forget the idea of "descriptions" |
| 18:37.42 | [TK]D-Fender | Tond: You're right... go ask in #mysql |
| 18:38.02 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 18:41.58 | *** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar) |
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| 18:46.22 | *** join/#asterisk stuff1 (n=user@mail.win-ent.com) |
| 18:46.47 | stuff1 | hey guys, when i try to use the ${GLOBAL(VAR)} to dial a phone |
| 18:47.15 | stuff1 | it doesn't work, it says it requires a technology/number |
| 18:47.38 | jake[work] | what's in ${GLOBAL(VAR)} ? |
| 18:47.40 | stuff1 | but my VAR=Zap/3, is there something wrong with that? |
| 18:48.16 | outtolunc | VAR=Zap/3/8005551212 |
| 18:48.22 | outtolunc | or +1 |
| 18:48.24 | stuff1 | i tried using just Dial(${VAR}), but that gave the same results as Dial(${GLOBAL(VAR)} |
| 18:48.32 | [TK]D-Fender | stuff1: why don't you pastebin all of the for use to see? Including the CLI output at verbose 10 |
| 18:48.33 | [TK]D-Fender | ~pb |
| 18:48.34 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:48.35 | [TK]D-Fender | ^^^^ |
| 18:48.41 | jake[work] | you're missing the number - see example above |
| 18:48.42 | stuff1 | k |
| 18:49.06 | Inssomniak | So i just came home with two of these linksys pap2t things... these work OK? |
| 18:49.10 | [TK]D-Fender | stuff1: And your dialplan as well... |
| 18:49.14 | stuff1 | k |
| 18:49.21 | [TK]D-Fender | Inssomniak: Pretty much |
| 18:52.33 | stuff1 | http://pastebin.ca/887929 -> dialplan |
| 18:53.31 | stuff1 | http://pastebin.ca/887930 -> extensions |
| 18:53.37 | stuff1 | http://pastebin.ca/887930 -> error |
| 18:54.43 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 18:55.49 | jake[work] | exten => 301,1,Dial(${GLOBAL(LOCAL2)}) |
| 18:55.58 | jake[work] | you're missing the number to dial |
| 18:56.02 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 18:56.06 | jake[work] | you have the first part |
| 18:56.10 | jake[work] | (technology) |
| 18:56.52 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:57.49 | jake[work] | when you dial 301, i'm assuming you want to dial a phone number? but it's not in the global |
| 19:00.06 | Inssomniak | is there a pap2t/asterisk howto anywhere? |
| 19:00.52 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 19:01.24 | zobia | @Qwell: do you know what's channel.c: Dropping duplicate answer! means? |
| 19:01.53 | stuff1 | in 1.2 i could define a phone with a global var and dial it the same |
| 19:02.07 | stuff1 | so when i dial 301 i am trying to reach the phone on Zap/4 |
| 19:02.35 | stuff1 | replacing the ${GLOBAL(LOCAL2)} with Zap/4 works perfectly |
| 19:02.58 | jake[work] | ok - it's an FXS |
| 19:03.05 | zobia | @russellb: do you know channel.c: Dropping duplicate answer!? |
| 19:03.06 | stuff1 | yes |
| 19:03.43 | stuff1 | and if you look at the outbound context it was ${GLOBAL(OUTBOUND)}/${EXTEN:1} |
| 19:04.06 | stuff1 | and that didn't work either, but replacing the GLOBAL(OUTBOUND) part with Zap/2 worked perfectly |
| 19:04.37 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 19:04.37 | jake[work] | http://www.voip-info.org/wiki-Asterisk+Dialplan+Globals |
| 19:04.54 | jake[work] | i don't use the globals, but it doesn't look like you're setting them properly |
| 19:05.39 | zobia | any ones how to config 2 lines 7910 in sccp.conf? |
| 19:06.00 | jake[work] | should be in a context called [globals] |
| 19:06.23 | jake[work] | (not [general] |
| 19:07.22 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 19:07.44 | [TK]D-Fender | ^^^^ |
| 19:13.31 | wglenncamp | ah.. fixed! Finally.. UNDOCUMENTED in the admin guide though... |
| 19:13.33 | wglenncamp | fyi: http://forums.digium.com/viewtopic.php?p=38401&highlight=&sid=3627ec33fbeadc320d2fdc07e0a04daa |
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| 19:29.17 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-133-53.dsl.teksavvy.com) |
| 19:30.54 | philippel | any takers on getting etheir cdr_odbc or cdr_tds builing on 1.4? I've got freetds working from the box to the backend MSSQL server but asterisk doesn't want to build it and menuselect claims un-met dependencies so I must be missing something dumb? |
| 19:32.45 | JenniferAkemi | hi :) I'm just starting out this asterisk journey. i'm reading the future of telephony ebook and figured i'd hang out here while doing so to soak up anything that's flying around |
| 19:34.18 | [TK]D-Fender | JenniferAkemi: Shit is scheduled to hit the fan in about 2 hours.... |
| 19:35.23 | JenniferAkemi | how do you mean? |
| 19:36.17 | JenniferAkemi | quick google reveals schmooze? |
| 19:36.20 | JenniferAkemi | is that what you are talking about? |
| 19:37.51 | JenniferAkemi | oh nm. |
| 19:37.59 | JenniferAkemi | you're makihng fun me of :P |
| 19:38.07 | [TK]D-Fender | JenniferAkemi: Comedic response to your soaking up whats flying around :) |
| 19:38.26 | JenniferAkemi | yeah i figured it out... a little late but what can i say... it's friday :P |
| 19:38.33 | JenniferAkemi | (at least i hope it is) |
| 19:38.38 | hmmhesays | JenniferAkemi, be prepared to be berated on IRC especially by that [TK]D-Fender guy |
| 19:38.43 | JenniferAkemi | hehe |
| 19:38.45 | [TK]D-Fender | JenniferAkemi: Is here... and TFG |
| 19:38.46 | JenniferAkemi | i'm getting that |
| 19:39.02 | [TK]D-Fender | hmmhesays: SHUP YUO ;0 |
| 19:39.24 | hmmhesays | I challenge you to a duel sir! |
| 19:39.32 | [TK]D-Fender | hmmhesays: ... lol :) |
| 19:39.58 | hmmhesays | that probably sounded horribly geeky to anyone who didn't get the crossroads reference |
| 19:40.16 | [TK]D-Fender | hmmhesays: BTW... I've got a group I now jam with (accoustic only right now), and learned *shudder* some Jimmey Buffet. Took me like 1 minute flat :p |
| 19:41.35 | hmmhesays | LOL, nothing wrong with a little jimmy buffet. "She's a real beauty a mexican cutey, how she got there I haven't a clue......" |
| 19:41.38 | hmmhesays | something like that |
| 19:43.29 | hmmhesays | JenniferAkemi, and since its friday you're going to see some guitar talk flying around. |
| 19:43.30 | [TK]D-Fender | hmmhesays: we did a bunch from Lynyrd Skynyrd, Blues Traveller, Kansas, and a bunch of others. I'm bringing in some Bon Jovi & Bryan Adams for next Sunday's meet-up |
| 19:43.44 | hmmhesays | ohhhh what blues traveller? |
| 19:43.54 | hmmhesays | hook has to be one of my favorite songs by them |
| 19:45.12 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
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| 19:45.39 | *** join/#asterisk MindTheGap (n=MindTheG@c9503f78.bhz.virtua.com.br) |
| 19:47.35 | hmmhesays | I'm not so sure about bryan adams |
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| 19:49.03 | JenniferAkemi | guitar and asterisk goes together? |
| 19:49.22 | JenniferAkemi | the only guitar i know is the hero kind |
| 19:49.40 | hmmhesays | JenniferAkemi, guitar didn't go together with *nix, *voip and general computer geekery I would have gone insane a long time ago |
| 19:50.26 | hmmhesays | I laugh at my roomate, he slows down the songs on ghIII so he can (nail those tough solo's) |
| 19:50.55 | jameswf | ping tzafrir |
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| 19:55.06 | JenniferAkemi | i don't fit the assumptions that this author made at all |
| 19:55.24 | [TK]D-Fender | hmmhesays: Considering my soloing speed & sweep picking, I'd chump him out easy :p |
| 19:56.02 | JenniferAkemi | i should have some linux admin experience and be new to telecom, but in fact i am a total linux newb and have been working on telephone switches for 10 years :) |
| 19:56.06 | [TK]D-Fender | hmmhesays: Did Runaround, and I'm taking in "All for You". Don't know Hook yet, but have seena lot of it on Youtube |
| 19:56.17 | JenniferAkemi | this will be interesting. |
| 19:56.20 | hmmhesays | JenniferAkemi, what kind? |
| 19:57.01 | hmmhesays | [TK]D-Fender, my sweep picking in non existant at the moment, I am still working on john petrucci's rock discipline almost regligiously though |
| 19:57.17 | [TK]D-Fender | hmmhesays: Yeah, his psycho exercises rock! :p |
| 19:57.32 | JenniferAkemi | harris |
| 19:57.58 | [TK]D-Fender | hmmhesays: He's a great guitarist technically speaking, but he's completely cold and lifeless... just like Yngwie Malmsteen |
| 19:58.41 | hmmhesays | [TK]D-Fender, yeah and it is the technical side I need to develop, because I spent years playing stevie ray type stuff |
| 19:58.51 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-234-119.d4.club-internet.fr) |
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| 19:59.30 | Inssomniak | ok I must be an idiot |
| 19:59.34 | hmmhesays | [TK]D-Fender, yeah "this knob controls the rotation of the earth, this one controls the heat of the sun. I haven't got to use that one yet" |
| 19:59.50 | hmmhesays | JenniferAkemi, harris switches to asterisk huh? Apples to oranges |
| 20:00.11 | Inssomniak | I set up this pap2t thing, and its working, I can call a phone plugged into it by extension number from a softphone, but I dont know how to dial an extension from a phone on the pap2t |
| 20:01.00 | JenniferAkemi | well i'm hoping the basic telephony concepts will still apply |
| 20:01.00 | hmmhesays | JenniferAkemi, most do |
| 20:01.00 | hmmhesays | You'll find out in a hurry if not. |
| 20:01.14 | JenniferAkemi | i never really had to worry about processor and scaling and all that though |
| 20:01.21 | JenniferAkemi | if you had the ports you had the power |
| 20:01.57 | hmmhesays | JenniferAkemi, PM me I don't want to say what I need to in this channel haha |
| 20:02.10 | hmmhesays | hmm that sounded bad, but its not. |
| 20:04.47 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
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| 20:06.25 | hmmhesays | Ugh the more I look at the 1.4 ami the more of a pita it is |
| 20:06.26 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-199-107.hsd1.ut.comcast.net) |
| 20:07.20 | hmmhesays | [TK]D-Fender, I still haven't gotten IM to work on this IP 601 |
| 20:08.46 | [T]ank | This is my sip peer set up. http://pastebin.ca/887994 what I am finding is that the fromuser setting is overriding the callerid set in my extensions.conf. How can I get around this? |
| 20:08.47 | tzafrir_home | jameswf, pong |
| 20:09.22 | hmmhesays | Set the callerid in the dialplan? |
| 20:09.58 | [T]ank | fromuser= overrides the callerid set in the dialplan |
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| 20:11.37 | hmmhesays | where are you terminating to? |
| 20:13.15 | [T]ank | the way that my stuff is set up is that I have one server (A) connecting to another where the T1s are (B). my phone connects to (A) where the outbound dial is made. Then the connection goes to (B) where zap dials out to the T1. my sip.conf i pasted is from server (A). That is working, however the callerid is set as the sip peername rather than my telephone number which is set in the extensions.conf. according to everything I have rea |
| 20:14.12 | ZaVoid | is there anyway to record how many times an extension has been used/called/activated? |
| 20:14.19 | ZaVoid | without doing a sql insert into my db that is |
| 20:14.50 | [T]ank | ZaVoid: you can query the cdr.csv or if you are using a cdr database, query that. |
| 20:15.02 | ZaVoid | using my own DB for cdr's |
| 20:15.10 | *** part/#asterisk RoyK (n=roy@ip-187-5-149-91.dialup.ice.no) |
| 20:15.13 | ZaVoid | but this is for stuff that doesn't write a cdr |
| 20:15.26 | ZaVoid | lets say [T]ank you dial 4444 and it plays tank.wav |
| 20:15.50 | ZaVoid | i can record that by doing a sql insert into a table.. but thats rather annoying from a few pov's |
| 20:16.08 | *** join/#asterisk timeshell (n=Khoja@gw.lusi.on.ca) |
| 20:16.13 | [T]ank | so you are looking for a command that will just output how many times this happens? |
| 20:16.30 | timeshell | Greetings. |
| 20:16.57 | ZaVoid | yeah basically |
| 20:17.07 | [T]ank | i dont think anything like that exsists. |
| 20:17.15 | ZaVoid | thats what i thought |
| 20:17.16 | timeshell | Question: Can one use a single sip connection between two asterisk servers and use that single sip connection for both servers to call each other's extensions? |
| 20:17.17 | [T]ank | i may be wrong.. i dont know of anything. |
| 20:17.37 | timeshell | Or is a registered sip connection from each server to each server required. |
| 20:17.37 | hmmhesays | [T]ank, it doesn't seem you need to set fromuser in your setup |
| 20:17.57 | [T]ank | i did try to take it out and at that point the connection stopped working. |
| 20:18.01 | [T]ank | are you using sip peers? |
| 20:18.02 | ZaVoid | yeah and you can't use variables in fromuser i've found |
| 20:18.13 | ZaVoid | i got one sip carrier that NEEDS valid ANI passed |
| 20:18.35 | ZaVoid | so i send them one ANI... but then i can't send valid ani to that carrier :( |
| 20:19.18 | *** join/#asterisk ManxPower (n=manxpowe@238.sub-70-222-242.myvzw.com) |
| 20:20.06 | *** join/#asterisk CrashSys (n=kumba@t1.databalance.com) |
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| 20:22.09 | hmmhesays | ZaVoid, you could write it to astdb |
| 20:22.18 | hmmhesays | why don't you want to sql it? |
| 20:22.54 | ManxPower | "sql" sounds too close to "squirrel"! |
| 20:23.19 | hmmhesays | not if you pronounce it sequel |
| 20:23.21 | ZaVoid | because my DB is getting beat up enuff with with every call i pass |
| 20:23.32 | ZaVoid | i finally cleaned it up enuff so its only 3 read queries and 1 write per call |
| 20:23.35 | ZaVoid | used to be like 10 |
| 20:24.16 | ZaVoid | and in my example say 4444 plays tank.wav... do i really need that info in the db? stored in an ever growing table :) |
| 20:24.24 | hmmhesays | ZaVoid, do you need something that survives a restart? |
| 20:24.37 | hmmhesays | What are you doing with that info? |
| 20:24.40 | J4k3 | dirty squirrel |
| 20:24.58 | pc500 | Has anyone intergrated asterisk into other third party pbxes for seamless systems? What about inter-tel systems? |
| 20:24.58 | ZaVoid | no doesn't hae to survice a restart |
| 20:25.06 | ZaVoid | basically i got enhanced services for my customers rights |
| 20:25.13 | ZaVoid | i want to see how often they actually use em |
| 20:25.15 | [T]ank | removing fromuser= gives me the following on the server I am connecting to: |
| 20:25.16 | [T]ank | [Feb 1 13:22:52] WARNING[27362]: chan_sip.c:8336 check_auth: username mismatch, have <telco_test4>, digest has <telco_test1> |
| 20:25.16 | [T]ank | [Feb 1 13:22:52] NOTICE[27362]: chan_sip.c:13710 handle_request_invite: Failed to authenticate user "asterisk" <sip:asterisk@10.60.10.30>;tag=as62a3a5d6 |
| 20:25.16 | [T]ank | Telco-Test2*CLI> |
| 20:25.48 | hmmhesays | ZaVoid, just write a variable every time it is called an increment it by 1 |
| 20:25.59 | ZaVoid | ahhh |
| 20:26.03 | ZaVoid | i didn't think about that |
| 20:26.06 | hmmhesays | paypal me a 20 and i'll give you a fancy sub |
| 20:26.11 | ZaVoid | thats a good idea hmmhesays |
| 20:26.16 | ZaVoid | fancy sub? |
| 20:26.26 | hmmhesays | a fancy subroutine in the dialplan to call to do that |
| 20:26.30 | ZaVoid | nah i can do it |
| 20:26.36 | ZaVoid | just didn't think about it |
| 20:26.36 | hmmhesays | lol, sorry had to try :D |
| 20:26.38 | ZaVoid | that makes sense |
| 20:26.40 | ZaVoid | haha good try |
| 20:26.51 | hmmhesays | You want my input on how to display it? |
| 20:26.57 | florz | I haven't read all the backlog, but I guess that no, you can't do that. |
| 20:27.22 | florz | You can't do atomic updates to variables, can you? |
| 20:31.21 | ManxPower | florz: See "trymacro", I think |
| 20:32.29 | florz | ManxPower: where would I look? =:-) |
| 20:33.20 | timeshell | How does one allow anonymous calls into asterisk? |
| 20:34.12 | ManxPower | florz: might be in 1.6+ |
| 20:34.26 | yang | Regarding BLF keys on Grandstream phones...I can use them, but they don't blink red when the call is being used on another phone, any idea ? |
| 20:34.28 | ManxPower | timeshell: anyone can call into your asterisk by defauly, at least for sip. |
| 20:34.52 | florz | ManxPower: IC, google didn't show any results for "trymacro asterisk" ... |
| 20:34.56 | ZaVoid | 206 active SIP channels <-- when i do sip show channels.. i get this... however there is really 30 calls on the box active... i hate that.. in every asterisk version i've ever seen |
| 20:35.02 | ZaVoid | like ghosts hanging around |
| 20:35.10 | timeshell | I'm getting "failed to authenticate user" when trying to call from a user on one asterisk server to another user on another. However, the reverse call works |
| 20:35.50 | ManxPower | florz: I only vaguely recall this is a newer feature, and (apparently) don't remember the correct name. |
| 20:35.54 | ManxPower | standby |
| 20:36.55 | hmmhesays | ZaVoid, because asterisk calls any transaction a channel |
| 20:38.54 | ZaVoid | eh |
| 20:39.12 | ZaVoid | xxx.xxx.xxx.xxx yyyyyyyy dba37dd2-80 00101/00105 unkn Yes Rx: INVITE |
| 20:39.17 | ZaVoid | yeah but even that? |
| 20:39.19 | ZaVoid | recieved invite? |
| 20:39.30 | ZaVoid | or same thing.. 19d61d0f3e9 00102/00002 unkn No (d) Rx: BYE |
| 20:40.09 | *** part/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 20:41.22 | *** join/#asterisk imcdona (n=imcdona@m660e36d0.tmodns.net) |
| 20:41.45 | pc500 | Has anyone intergrated asterisk into other third party pbxes for seamless systems? What about inter-tel systems? |
| 20:42.31 | ZaVoid | like on a nortel? |
| 20:42.34 | imcdona | you can use a t1 crossover if it doesn't support sip |
| 20:42.58 | mustiy | Guys whats the best suggested free softphone that can be connected to asteriks? |
| 20:43.41 | Inssomniak | this pbx stuff is some of the coolest stuff since they invented the phone |
| 20:44.11 | pc500 | imcdona - Is does do that, but I don't have the cards. But it does SIP. I don't know about on a trunking basis, but I can hook phones up that way. |
| 20:44.26 | pc500 | ZaVoid - It would probably be similar to link to a nortel if it's an IP-based pbx, yes. |
| 20:46.30 | ZaVoid | so just sip connect them |
| 20:46.44 | ZaVoid | put the non aterisk in as a peer |
| 20:46.45 | ZaVoid | done |
| 20:47.19 | pc500 | ZaVoid - So each extension on asterisk is seen as a seperate phone to the other pbx? |
| 20:47.42 | ZaVoid | depends on what your trying to do |
| 20:47.47 | ZaVoid | just pass calls between the two systems? |
| 20:47.52 | pc500 | enable direct-extension dialing, pass calls. |
| 20:47.55 | ZaVoid | or you want each to see sperate phones? |
| 20:47.58 | pc500 | One large system throughout the enterprise. |
| 20:48.09 | ZaVoid | so say 1/2 phones on one and 1/2 on the other |
| 20:48.12 | pc500 | yes |
| 20:48.16 | pc500 | more like 7/8 and 1/8th |
| 20:48.17 | pc500 | but yes |
| 20:48.18 | pc500 | same concept |
| 20:48.28 | pc500 | What I don't want, is to call an extension and it be just like I dialed the outside # of the other PBX. |
| 20:48.28 | ZaVoid | just setup dialplans to go to the peer for the non asterisk system and point to that peer |
| 20:48.37 | ZaVoid | then on the inbound on the other system route the digits the way you would normally |
| 20:48.54 | ManxPower | the easiest way to do that is the simplest. assign ranges of extensions to each server. |
| 20:49.13 | pc500 | So how do I maintain the dialplan integration between the two? |
| 20:49.28 | pc500 | Is it psosible to jsut pass off 4xxx to the other server, or will you have to add logic every time? |
| 20:50.03 | ManxPower | pc500: for 2 servers with a well planned dialplan, you it by hand, it's fast and simple. |
| 20:50.06 | hmmhesays | does switch still work in the dialplan? if so use it |
| 20:50.30 | ManxPower | a couple of lines of dialplan to route all 4xxx calls to the other server and all 3xxx calls local, or however. |
| 20:50.31 | *** join/#asterisk d-k-t (n=dt@125.120.139.9) |
| 20:50.40 | pc500 | ManxPower - BUt I can't just blindly forward all 5xxx to the other PBX and have it route to the right place? |
| 20:51.03 | ManxPower | pc500: are all 5xxx extensions on the other PBX? |
| 20:51.09 | pc500 | Sure, I can make it so |
| 20:51.19 | pc500 | Normally it's 5xxx per office |
| 20:51.23 | pc500 | But we're running out of numbers... |
| 20:51.30 | pc500 | But It'll soon be 51xx 52xx, etc. |
| 20:51.33 | ManxPower | then just blindly route them to the other PBX and let that PBX sort it out. |
| 20:52.02 | pc500 | Ok, I didn't know if SIP call routing worked in that manner (much like networking / IP routing). |
| 20:52.09 | pc500 | you just pass it on and it's the other sides problem then? |
| 20:52.10 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 20:52.16 | ManxPower | exten _5XXXX,1,Dial(SIP/${EXTEN}@otherpbx) |
| 20:52.16 | *** join/#asterisk WAudette (n=chatzill@75.148.48.213) |
| 20:52.32 | ManxPower | pc500: in your case, pretty much |
| 20:52.54 | pc500 | You mean it gets more complicated? :) |
| 20:53.08 | ManxPower | pc500: now if you wanted to be able to have any extension use any server to call anywhere and get calls, it gets a million times more complicated. |
| 20:53.20 | pc500 | Some fault tolerance woudl be nice, but not necessary. |
| 20:53.28 | pc500 | We have a large private WAN interconnecting offices. |
| 20:53.34 | pc500 | So all calls go IP-based. |
| 20:53.37 | ManxPower | pc500: exactly what we have |
| 20:54.07 | ManxPower | pc500: once you get it working there are things you can do to help with fault tollerance. |
| 20:54.30 | pc500 | ZaVoid - Pokes? :) |
| 20:54.44 | ZaVoid | asterisk has fault tolerance other the load balancing ManxPower ? |
| 20:55.08 | ManxPower | if you use SIP between the two servers and no transcoding, I don't see any reason phones at different locations could not reinvite and get asterisk out of the audio path. with DNS SRV records or redundant entries for the phones' server config you could do even better, but the basic design is good |
| 20:55.39 | pc500 | However asterisk would still need to perform that reinvite... |
| 20:56.03 | kyron | AHHA!! *'s build process does not honor --prefix=$PATH ! |
| 20:56.09 | ZaVoid | pc500 are you worried about the asterisk box crashing? |
| 20:56.11 | ZaVoid | for internal cals? |
| 20:56.30 | pc500 | ZaVoid - General stability problems and points of failure, that seems like hte most likely cause. |
| 20:56.31 | ManxPower | you can have an identical server, if the first one goes down, plug in the 2nd one, move the T-1 wires to the new server, done. |
| 20:56.36 | pc500 | ZaVoid - Circuit outages are also possible |
| 20:56.51 | pc500 | Can we use BRI/analog? |
| 20:56.57 | ManxPower | that gives you "fault shortening", which is all you can really expect on a budget anyway. |
| 20:57.04 | pc500 | We have certain areas where a PRI is $800/mo and 24 pots lines are $400. |
| 20:57.17 | ManxPower | pc500: I don't think you'll find anyone here that would recommend analog |
| 20:57.27 | pc500 | What about BRI? |
| 20:57.34 | pc500 | It at least retains digital signalling just like a PRI, riight? |
| 20:57.49 | ManxPower | pc500: Many CLECs give you better pricing and will put remote numbers on PRIs if you want. |
| 20:57.59 | ManxPower | pc500: are you in the USA or Canada? |
| 20:58.05 | [TK]D-Fender | YAY! : http://xkcd.com/378/ |
| 20:58.11 | pc500 | ManxPower - USA, tariffed poorly in some states. |
| 20:58.29 | pc500 | ManxPower - other areas it's $300/mo. |
| 20:58.34 | ManxPower | pc500: then you have a better chance of removing the appendix from a sick horse than get BRI working in the USA. |
| 20:58.35 | pc500 | (and we have pris) |
| 20:59.00 | pc500 | ManxPower - Why is this? conceptually, ISDN is jsut 2 channels? Shouldn't it be the same functionality? |
| 20:59.22 | ZaVoid | arizona.. pacwest is your clec pc500 ? |
| 20:59.26 | ManxPower | I worked with a company and CLEC where all the company's numbers for all offices rang on a PRI at HQ, then the calls were sent out over the WAN to the remote phones via Asterisk |
| 20:59.28 | pc500 | ZaVoid - Qwest/Idaho |
| 20:59.34 | ZaVoid | ah |
| 20:59.39 | ZaVoid | uyeah i can see tarrifs being bad there |
| 20:59.43 | pc500 | ZaVoid - ISP is based out of Phoeniz, AZ. |
| 20:59.48 | ManxPower | pc500: I would be suprized of more than 10 people use USA BRIs with Asterisk. |
| 20:59.49 | ZaVoid | i see that :) |
| 21:00.15 | pc500 | ManxPower - That is my ultimate goal. Was this done with asterisk? |
| 21:00.18 | ManxPower | so your community support will be ziltch |
| 21:00.26 | pc500 | ManxPower - RIght now we have 4 pri's at only location that takes in 800# calls and distributes. |
| 21:00.27 | ManxPower | pc500: correct. |
| 21:00.46 | Inssomniak | is there any particular codecs that echo less than others? |
| 21:01.03 | pc500 | However I don't know if asterisk is appropriate for that office do to the advanced call, reporting, and queueing needs. |
| 21:01.07 | pc500 | Manily reporting is the big one. |
| 21:01.27 | ManxPower | If your WAN is stable, you could do something similar. now each office DID have an analog pots direct from the telco line or two for fax and cc machines and also for red 911 phones on the walls |
| 21:01.30 | [TK]D-Fender | Inssomniak: Codecs don't cause echo |
| 21:01.34 | Inssomniak | oh |
| 21:01.44 | pc500 | However I want to use it at some smaller offices who just need dialtone for 20-40 people and connectivity to the central system, yet aren't small enough for me to just opush all there telephony over the WAN. |
| 21:01.48 | ManxPower | pc500: but that is a very complex project |
| 21:02.24 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:02.25 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:02.34 | lmadsen | hrmmm... possible bug? [Feb 1 16:00:56] WARNING[14992]: res_odbc.c:149 ast_odbc_smart_execute: SQL Execute returned an error -1: 22001: [FreeTDS][SQL Server]String or binary data would be truncated. (62) |
| 21:02.35 | lmadsen | [Feb 1 16:00:56] WARNING[14992]: res_odbc.c:149 ast_odbc_smart_execute: SQL Execute returned an error -1: 01000: [FreeTDS][SQL Server]The statement has been terminated. (55) |
| 21:02.35 | lmadsen | [Feb 1 16:00:56] WARNING[14992]: app_voicemail.c:1397 store_file: SQL Execute error! |
| 21:02.36 | lmadsen | [INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext) VALUES (?,?,?,?,?,?,?,?,?,?)] |
| 21:02.43 | lmadsen | oops... sorry, that was only supposed to be the last line |
| 21:02.54 | ManxPower | You would think that an ASCEND SoHo BRI router/ATA/NT1 could be paired with Asterisk but that would take a lot more coding info than I have |
| 21:02.56 | timeshell | What is the appropriate sip type to use in sip.conf to set up a connection for another asterisk server? |
| 21:03.01 | pc500 | Historically I've had local termination in all the offices, but we have at least 1.5 megabit going everywhere, private lien T1s, very reliable... maybe I should just look at using one central system for everyone. |
| 21:03.09 | lmadsen | I don't get the same issue when recording the persons name or the busy msg |
| 21:03.18 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 21:03.36 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:03.48 | ManxPower | pc500: You might be able to up the bandwidth with savings from reduction in lines. |
| 21:04.04 | ManxPower | pc500: you could keep a few analog lines around for when the T-1s are down. |
| 21:04.19 | pc500 | They just won't have incoming unless calls are forwarded or something. |
| 21:04.28 | *** join/#asterisk FlatFoot (n=chatzill@80.88.218.4) |
| 21:05.36 | ManxPower | pc500: your asterisk server should be able to reroute automatically |
| 21:05.53 | pc500 | I'm not worried too much about data because there is multiple paths |
| 21:05.54 | WAudette | I am using Asterisk 1.2.x on CentOS and have an issue when calling from a custom extension to another system with that happens to have the same defined extension via sip. The call fails with a busy signal because the asterisk PBX on the far end attempts to authenticate the call. |
| 21:05.56 | pc500 | It isn't going down. |
| 21:06.06 | ManxPower | on many of my servers, if the local PSTN call failed, it sends the call to a server in another city |
| 21:06.25 | FlatFoot | evening all |
| 21:06.40 | ManxPower | (on another carrier, actually) |
| 21:07.18 | JenniferAkemi | can't you buy a PRI from another provider in the states? |
| 21:07.25 | WAudette | ManxPower: You are talking about outbound calls right? |
| 21:07.36 | WAudette | JenniferAkemi: Yes. |
| 21:07.54 | pc500 | When available... for a reasonable rate. |
| 21:08.05 | pc500 | Is 90 ms system-wide ok if I apply QoS? |
| 21:08.17 | pc500 | (that's worst-case scenario). |
| 21:08.40 | WAudette | Corydon76: Are you available? |
| 21:08.58 | JenniferAkemi | that sounded expensive but i just realized why, i'm used to buying stuff in a telco building |
| 21:09.41 | ManxPower | WAudette: outbound and inbound |
| 21:09.43 | JenniferAkemi | i guess when you're talking about a site it's not necessarily in the same building with a ton of other providers |
| 21:10.14 | ManxPower | pc500: it's not so much the latency that will be an issue, it is JITTER (which QoS should take care of) |
| 21:10.46 | ManxPower | 9/100's of a second network delay will not be noticable. |
| 21:10.57 | WAudette | Cool, how do you capture the inbound? I know I can do it via network switching if my providers support BGP or OSPF even... never done it at the PSTN level. Stands to reason their would be a way hough. |
| 21:10.59 | WAudette | though* |
| 21:11.35 | ManxPower | WAudette: we were talking about a wan outage preventing the central asterisk server from routing calls to an asterisk server at a remote office |
| 21:12.06 | WAudette | Oh I see. |
| 21:12.12 | ManxPower | if the asterisk server in HQ fails to send the call over the WAN, it can automatically failover to using the PSTN to send calls to the remote office's Asterisk server's POTS port(s) |
| 21:12.20 | WAudette | That's doable... |
| 21:12.36 | ManxPower | all calls will have to go to the same person, but they can transfer them to the correct extension |
| 21:12.59 | ManxPower | WAudette: MUCH simpler than many solutions, not as good as real fault tollerance |
| 21:13.01 | WAudette | Or to an IVR? |
| 21:13.03 | JenniferAkemi | couldn't you hae the pots ports go to an ivr which prompts for an ext |
| 21:13.24 | ManxPower | JenniferAkemi: people get really freaked out when they get an IVR when they are not expecting it, but yes, you could do that. |
| 21:13.40 | JenniferAkemi | i'd imagine they might get more freaked out when they call bob and get joanne though ;) |
| 21:13.44 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 21:13.53 | JenniferAkemi | especially if joanne didn't know the wan was down |
| 21:13.58 | JenniferAkemi | and was like... uh this is joanne! |
| 21:14.02 | JenniferAkemi | not bob! |
| 21:14.08 | ManxPower | Naw, call bob and get "Fnordic Law, how may I direct your call?" |
| 21:14.39 | JenniferAkemi | i hate IVRs |
| 21:14.48 | JenniferAkemi | and i REALLY hate the ivrs that do voice recognition |
| 21:15.09 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 21:15.22 | FlatFoot | JenniferAkemi , agreed we waited for nearly 8 years before we had to implement 1 |
| 21:15.42 | JenniferAkemi | yeah. even after saying that i wrote one |
| 21:15.54 | JenniferAkemi | sometimes you have no choice |
| 21:16.12 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 21:16.18 | FlatFoot | voice rec BT in the UK started that , then you end up talking to india anyway |
| 21:16.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:17.45 | *** join/#asterisk mihinomenest (n=argh@66.255.220.17) |
| 21:17.56 | ManxPower | "Please dial or speak your selection now." <blaring speakers overhead> "Flight 342 is now boarding" </blaring speakers overhead> "9 is not a valid option. 9 is not a valid option. 9 is not a valid option. Goodbye!" |
| 21:18.03 | WAudette | So is anyone familiar with my issue? |
| 21:18.19 | ManxPower | The Marquis DeSade would be proud. |
| 21:18.20 | FlatFoot | ManxPower , surely if you replicate the ext.conf on server b can you not send the called ext over to it and redirect as first required ? |
| 21:18.35 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 21:18.51 | ManxPower | FlatFoot: assuming you have PSTN DID service, yes. |
| 21:18.52 | WAudette | FlatFoot: Thanks for responding. |
| 21:19.23 | WAudette | oh that wasn't to me. |
| 21:19.25 | FlatFoot | is it state side ? cos you can get that as default in UK |
| 21:19.32 | WAudette | It did relate though... <grin> |
| 21:19.35 | hmmhesays | ok back to my IM sip conundrum |
| 21:19.57 | FlatFoot | WAudette , was sort of related |
| 21:20.22 | FlatFoot | an aside , where are people tonite ? |
| 21:20.23 | WAudette | lol... but not. |
| 21:20.43 | jameswf | << working its only 2:40 pm here |
| 21:20.48 | WAudette | Busy living away from computers I suppose. |
| 21:20.49 | FlatFoot | WAudette , well it's all * based talk so sort of |
| 21:21.07 | WAudette | jameswf: good to see you in here. |
| 21:21.12 | FlatFoot | 2:40pm wheres that then ? nearly saturday here |
| 21:21.28 | jameswf | Arizona (US) |
| 21:21.31 | WAudette | 1:21 Pacific here. |
| 21:21.50 | jameswf | we have the superbowl sunday.... I am too pore for bad seats |
| 21:21.53 | FlatFoot | Arizona , is that mainly desert ? |
| 21:22.26 | jameswf | arizona is a desert but also phoenix is the nations 5th largest city |
| 21:22.28 | WAudette | jameswf: I have this ticket I was hoping the asterisk guys might be able to help with http://freepbx.org/trac/ticket/2486 |
| 21:22.53 | *** join/#asterisk csm4ch (n=caciano@189.32.68.220) |
| 21:22.58 | WAudette | Maybe there is a better way to hand off a call that will prevent the odd authentication issue. |
| 21:22.58 | FlatFoot | jameswf , pheonix didn't they divert a river to feed that with water ? |
| 21:23.04 | ManxPower | Ah yes, the superbowl, the ultimate evolution of what started out as african tribes at war. |
| 21:23.04 | J4k3 | that only means phoenix will have the most dead people if the water stops flowing. |
| 21:23.12 | J4k3 | and in phoenix, it could easily occur. |
| 21:23.35 | jameswf | Amaizingly arizona has been in a drought for like 10 years yet we use water like its going out of style.. |
| 21:23.40 | bsdwarrior | im a .call file in using set : userfield=test but nothing shows up in the database in the cdr table. |
| 21:23.45 | ManxPower | WAudette: try the CORRECT channel. |
| 21:23.45 | J4k3 | and goddamn arizona is full of rednecks. I thought texas was white trash hell til I visited AZ. |
| 21:23.47 | *** join/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 21:24.01 | WAudette | ManxPower: Hey... this is a collaborative question. |
| 21:24.10 | J4k3 | now, new mexico wins the award |
| 21:24.10 | WAudette | I was trying to help the devs out. |
| 21:24.11 | ManxPower | J4k3: MOST of the country is full of rednecks. |
| 21:24.12 | FlatFoot | J4k3 , where do you reside then ? |
| 21:24.15 | jameswf | texas is worse then phoenix... well unless you hit certain areas |
| 21:24.22 | *** join/#asterisk my007ms (i=master@botmaster.x86.be) |
| 21:24.28 | JenniferAkemi | why do you say that BRI is hard to get working? admittedly i have no idea what i'm doing yet (but i am reading this book, and it just said that anotehr way to connect is via BRI and you can get the V410P card for it) |
| 21:24.30 | J4k3 | ManxPower: good call... pennsylvania is suprisingly bad too. |
| 21:24.34 | JenniferAkemi | is the problem with ordering a BRI in the states? |
| 21:24.37 | JenniferAkemi | i know theyr'e available here |
| 21:24.41 | JenniferAkemi | (here beind canada) |
| 21:24.42 | WAudette | ManxPower: really I just wanted to know if it happed for you guys too. |
| 21:24.45 | JenniferAkemi | being even |
| 21:24.47 | J4k3 | jameswf: texas is about a million times bigger than phoenix, I'd guess :) |
| 21:24.50 | J4k3 | (size-wise) |
| 21:24.53 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 21:24.57 | jake[work] | a problem - try a major PIA |
| 21:25.02 | J4k3 | people in cities are generic. Theres the sheep, the loud sheep, and the wolves. |
| 21:25.03 | jameswf | all southern states are sort of redneckish as you head east in the southern states you get hillbillyish |
| 21:25.05 | jake[work] | (at least in Phila) |
| 21:25.13 | ManxPower | JenniferAkemi: Telcos do not want to sell BRI, all the existing products only support the EuroISDN version of the ISDN protocol, no support for USA varients |
| 21:25.15 | J4k3 | in the country you have rednecks and everyone else. |
| 21:25.44 | *** join/#asterisk mltlnx (n=mltlnx@maa5f36d0.tmodns.net) |
| 21:25.48 | JenniferAkemi | oh |
| 21:25.51 | iratik | Setting up asterisk from scratch for the first time........ i have [icall] configured in sip.conf, so to dial a number out on icall ... Dial(sip/${EXTEN},55,o) ? |
| 21:25.53 | jameswf | We have a BRI card comming out soon.... with onboard EC... |
| 21:25.53 | JenniferAkemi | i wonder what we use in Canada :) |
| 21:26.08 | ManxPower | JenniferAkemi: I believe Digium's first and only ISDN BRi card (which has not been out long) is supposed to support USA ISDN BRI. |
| 21:26.17 | FlatFoot | jameswf texas is about 7 times the size of the UK , makes us feel small |
| 21:26.26 | FlatFoot | even though i am 6'5" |
| 21:26.30 | FlatFoot | :p |
| 21:26.34 | JenniferAkemi | is that theone i sad? the B410 or something |
| 21:26.37 | ManxPower | iratik: read extensions.conf.sample AGAIN |
| 21:26.45 | ManxPower | JenniferAkemi: I believe so |
| 21:26.48 | JenniferAkemi | argh. sorry for typos. |
| 21:26.53 | iratik | SIP/icall/${EXTEN} ? |
| 21:26.55 | jameswf | we were acrualy going to vote to sell texas back to mexico... |
| 21:27.02 | JenniferAkemi | i love having my desk in front of these windows, but it's so COLD my hands get stiff and I type badly |
| 21:27.06 | man_o_magic | Sorry to interrupt, I am looking for a gui to configure asterisk. Something web-based. Does anyone have recommendations? |
| 21:27.15 | FlatFoot | jameswf , well its close enough |
| 21:27.15 | bsdwarrior | @home |
| 21:27.22 | bsdwarrior | tkd-fender you around |
| 21:27.26 | jameswf | jbot tell man_o_magic about freepbx |
| 21:27.38 | ManxPower | man_o_magic: they all suck, but look at the /topic for the info you want |
| 21:27.40 | FlatFoot | ~freepbx |
| 21:27.41 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:28.09 | *** join/#asterisk khronos (n=khronos@c-71-57-140-136.hsd1.fl.comcast.net) |
| 21:28.24 | JenniferAkemi | are many people here from Canada? i'm just curious |
| 21:28.31 | iratik | I'm not from canada |
| 21:28.36 | khronos | Anybody help a newbie to a2billing? |
| 21:28.42 | jameswf | caniduh eh |
| 21:28.59 | FlatFoot | JenniferAkemi , my uncle lives in ca |
| 21:29.10 | jameswf | JenniferAkemi: to be fair you must repeat your question in french |
| 21:29.15 | JenniferAkemi | i noticed that tkd-fender was on a bell.ca host but only after he left |
| 21:29.15 | bsdwarrior | lol |
| 21:29.33 | JenniferAkemi | it would have to be bad french |
| 21:29.40 | bsdwarrior | whats the problem ? |
| 21:29.46 | JenniferAkemi | Est-ce qu'il y a les personnes Canadien ici? |
| 21:29.49 | J4k3 | JenniferAkemi: I added a 20A branch to my office to support a small space heater. |
| 21:29.53 | J4k3 | for just that reason |
| 21:30.05 | J4k3 | half this office is glass, and not very well insulated glass... |
| 21:30.09 | Strom_C | une carte postal de poutine vous attend |
| 21:30.28 | FlatFoot | oi whats that daft language ? |
| 21:30.34 | JenniferAkemi | i work from home and i'm reluctant to fork over the cash for electricity. i was thinking of knitting a pair of fingerless gloves instead ;) |
| 21:30.38 | iratik | you are waiting for a post card for something |
| 21:30.48 | bsdwarrior | anyone know how to get the userfield with cdr to work ? |
| 21:30.49 | JenniferAkemi | a post card of poutine |
| 21:30.49 | JenniferAkemi | heh |
| 21:30.51 | WAudette | ManxPower: My question is someone generic though... |
| 21:30.53 | iratik | qu'est-ce qu'un poutine |
| 21:30.54 | J4k3 | but yeah, I'm from texas... and I'm suprised how 'generic' we are... but I Think its the massive "yankee influx" of the last 30 years. |
| 21:31.00 | JenniferAkemi | poutine is manna |
| 21:31.07 | iratik | c'est comme une putain ? |
| 21:31.11 | JenniferAkemi | haha no |
| 21:31.17 | FlatFoot | bsdwarrioir what version of * |
| 21:31.28 | JenniferAkemi | poutine is french fries + cheese curds + gravy |
| 21:31.55 | bsdwarrior | flatfoot 1.2.14 |
| 21:32.02 | ManxPower | After Katrina I lived in Atlanta TX for 3 months. Horrible. |
| 21:32.03 | WAudette | When dialing from a non-unified dialing plan pbx to another, say a fiends system via sip the far end (ie friends system) hangs up. |
| 21:32.14 | J4k3 | my girlfriend is from arkansas... the average white "non-racist" person there is more racist than the folks we consider 'damned racist' down here. Drive west and you hit new mexico - where you watch drunk indians hit the crackpipe as you drive down the highway... |
| 21:32.28 | J4k3 | atlanta is in an awful spot |
| 21:32.38 | ManxPower | The only place you could buy an ethernet switch was the local farm supply place and they had to order it. |
| 21:32.53 | J4k3 | I've stopped there on many occasions on trips to arkansas... they're like a cross between texas, lousianna and arkansas... |
| 21:33.02 | J4k3 | dumb, toothless racists. |
| 21:33.11 | J4k3 | (texas, lousiana and arkansas) |
| 21:33.16 | ManxPower | (and the scary thing is the tech was trying to get some cisco phones working with Asterisk). |
| 21:33.22 | ManxPower | I grabbed Toto and ran. |
| 21:33.32 | iratik | j'habite dans missouri |
| 21:33.32 | WAudette | It only happens if both systems happen to have the same extension number on both the sending and receivning ends. The far end Asterisks attempts to authenticate the senders SIP extension to the which of course fails. |
| 21:33.48 | FlatFoot | bsdwarrioir , http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr |
| 21:33.52 | ManxPower | WAudette: see fromuser= in sip.conf.sample |
| 21:34.12 | WAudette | ManxPower: Ok, checking it out now. |
| 21:34.22 | WAudette | So you've seen this before I take it? |
| 21:34.28 | J4k3 | ManxPower: haha... walmart carries switches and routers these days |
| 21:34.40 | WAudette | lol... scarry eh! |
| 21:35.06 | jameswf | a city size can be determined by the number of walmarts it has |
| 21:35.12 | bsdwarrior | flatfoot, only problem now is im doing this with the manager |
| 21:35.28 | J4k3 | marshall = home of the GPS Jitter (tm) |
| 21:35.29 | FlatFoot | J4k3 , we use 5 port switches that cost us £2.50 gbp |
| 21:35.39 | J4k3 | something there makes my GPS wig the fark out |
| 21:35.55 | FlatFoot | they work best behind our radio network cos they don't care and just send data |
| 21:36.05 | man_o_magic | @ManxPower, thanks. @FlatFoot, I was looking for something other than freepbx, @all, do yall just manually create your dialplans and stuff? |
| 21:36.07 | *** part/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 21:36.26 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 21:36.32 | jake[work] | <PROTECTED> |
| 21:36.33 | FlatFoot | bsdwarrior , not used manager much , juts used to start calls from my customer database interface |
| 21:36.47 | J4k3 | FlatFoot: I'm starting to hate auto-mdix switches... they seem to screw up when connected together |
| 21:36.54 | J4k3 | crossover or straight cable, doesn't matter, they refuse to link |
| 21:36.54 | *** join/#asterisk Daejeo (n=chatzill@211.211.234.81) |
| 21:37.00 | FlatFoot | man_o_magic , yep with a little help from all in this room |
| 21:37.01 | J4k3 | or they'll do something cute like, not link with full duplex. |
| 21:37.03 | iratik | I know this is somewhere in extensions.conf.sample... just can't find where..... What is the difference between ${EXTEN:1} and ${EXTEN} ? |
| 21:37.32 | angryuser | 4444 = exten 444 = exten:1 |
| 21:37.33 | jake[work] | http://www.voip-info.org/wiki/index.php?page=Visual+Dialplan+for+Asterisk |
| 21:37.34 | Daejeo | anyone know fxs port impedance for UK? |
| 21:37.40 | jake[work] | maybe try that - never used it myself |
| 21:37.40 | FlatFoot | J4k3 , if you can find them in the states go for Dynamode . cheap cheerful and don't care |
| 21:37.46 | angryuser | strips one letter from beggining |
| 21:37.52 | iratik | angryuser: so EXTEN:X strips X letters from the beginning? |
| 21:37.54 | Daejeo | anyone knows fxs port impedance for UK? |
| 21:37.58 | angryuser | yes |
| 21:38.33 | iratik | thats what i was going to guess... thanks ----- |
| 21:38.36 | iratik | where is the doc for that? |
| 21:38.46 | FlatFoot | J4k3 , those cheap things we use don't hold arp tables so don't care about network inturuptions |
| 21:39.26 | angryuser | <iratik> wiki |
| 21:39.34 | JenniferAkemi | whoid ${EXTEN:3} = 4 if ${EXTEN} = 4444 ? |
| 21:39.42 | angryuser | ~wiki |
| 21:39.44 | JenniferAkemi | and whoid = would ? |
| 21:40.11 | jblack | Holy shit. Welcome to Amerika. From Missisipi Bill 282 "An act to prohibit certain food establishments from serving food to any person who is obese" |
| 21:40.38 | angryuser | <JenniferAkemi> dont know, try with the noop after |
| 21:41.21 | angryuser | <JenniferAkemi> yes sorry wron answer |
| 21:41.45 | JenniferAkemi | np, i saw the stripping thing after i typed my question, so you basically answered it while i was typing |
| 21:41.57 | JenniferAkemi | i need to learn to type faster or something :) |
| 21:41.59 | FlatFoot | J4k3 , Where are u in Texas |
| 21:42.14 | J4k3 | FlatFoot: east texas, near Crockett |
| 21:42.30 | iratik | In the command Dial("IAX2/513-28", "SIP/icall/4178237644|300|"), what does the |300| mean? |
| 21:42.36 | J4k3 | jblack: 'define obese'. Should we send out an urban passification unit?! :D |
| 21:42.55 | FlatFoot | J4k3 , is that the chilly side ? ;) |
| 21:43.04 | angryuser | <J4k3> Fat |
| 21:43.04 | J4k3 | well |
| 21:43.27 | hmmhesays | My fingers never keep up with what I'm thinking |
| 21:43.36 | J4k3 | well, I'm 'fat' but I can run a 5.5 minute mile and not puke. |
| 21:43.45 | iratik | SIP/icall/4178237644|55|o , what does the |55|o mean? |
| 21:43.49 | FlatFoot | hmmhesays , your lucky you can think |
| 21:43.51 | angryuser | <iratik> time limit |
| 21:43.59 | iratik | 55 ms? |
| 21:44.06 | FlatFoot | my brain stopped that years ago , tooooooooo stressful |
| 21:44.20 | J4k3 | 225# @ 6'3"... maybe eating fast food should require a license? :) |
| 21:44.50 | FlatFoot | J4k3 17 stone @ 6'5" can't be arsed to run |
| 21:44.52 | angryuser | or maybe it is session ident, pastebin all |
| 21:44.55 | iratik | thanks <angryuser> btw |
| 21:44.58 | J4k3 | and its not like you can keep high school kids from buying beer... it'll just let skinny people eat free when fat people pay for them to go get them both food. |
| 21:46.38 | angryuser | <J4k3> they just should do more sport, and forget about fastfood, it is not that difficult to loose some kilos |
| 21:46.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:46.56 | *** join/#asterisk mltlnx (n=mltlnx@maa5f36d0.tmodns.net) |
| 21:47.47 | J4k3 | angryuser: if you're truely obese, you gotta start real slow... |
| 21:47.49 | FlatFoot | angryuser , you must be from europe ( kilo's ) |
| 21:48.16 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:48.21 | J4k3 | angryuser: I know a guy who eats less than 1700 calories/day and still continues to gain weight |
| 21:48.29 | J4k3 | he's a good 400# |
| 21:48.37 | *** part/#asterisk mltlnx (n=mltlnx@maa5f36d0.tmodns.net) |
| 21:48.43 | hmmhesays | 1700 calories thats quite a bit |
| 21:48.59 | angryuser | <J4k3> nutrition problem |
| 21:49.21 | *** join/#asterisk real0ne (i=real0ne@adll-180-244-206-196.marocconnect.net.ma) |
| 21:49.25 | ZaVoid | hell at mcdonalds 1700 calories is a breakfast drink lol |
| 21:49.32 | angryuser | <J4k3> he should go and see the specialist |
| 21:49.49 | WAudette | ManxPower: I see the context... so when I use this command where would I insert the context? Dial("SIP/201-08920920", "sip/<cleansed>@sip.<cleansed>.com||tr") Or does it even get inserted here+ |
| 21:50.07 | hmmhesays | I really wish jack johnson would just die |
| 21:50.24 | J4k3 | angryuser: he is... apparently he has a thyroid issue |
| 21:50.57 | angryuser | <J4k3> yes that is annoing thing ;( |
| 21:51.10 | FlatFoot | angryuser, my dad flatfloot ses i can ask u wher u r in france |
| 21:51.42 | angryuser | north , city of Amiens |
| 21:52.01 | FlatFoot | angeryu |
| 21:52.08 | *** join/#asterisk `Sean (i=Un1x@CPE001c351a764e-CM0014e8869416.cpe.net.cable.rogers.com) |
| 21:52.15 | MooingLemur | does anyone know if it's possible to use a fractional voice/data T1 on asterisk, ignoring the data part, and what type of signalling would that T1 usually carry? |
| 21:52.35 | FlatFoot | angryuser, oops, not far from us in kent |
| 21:53.08 | b11d | Anyone here good at troubleshooting zap problems? I get no audio Zap<->Zap through a channel bank, but I do get audio from sip<->zap. |
| 21:53.11 | iratik | Why does this not work...... I can't dial out and i don't know what i'm doing wrong ------- (Not using freepbx anymore, installed from scratch) http://pastie.caboo.se/146389 ...... can anyone take a look? |
| 21:53.22 | angryuser | <FlatFoot> kent whre is that? |
| 21:54.23 | FlatFoot | angryuser, channel tunnel (u.k) :p |
| 21:54.56 | angryuser | ah i see, i would like to take eurostar one day ;) |
| 21:55.52 | angryuser | have you heard about new train? Paris - London 2 hrs? |
| 21:55.55 | FlatFoot | angryuser, go straight to london don't bother with ashford |
| 21:56.40 | FlatFoot | angryuser, yeh just opened 240kmh |
| 21:57.06 | bsdwarrior | in a .call file Set(CDR(userfield)=1234) does not work. set: userfield=1234 doesnt work. anyone know what the problem is? |
| 21:57.12 | angryuser | no it is faster?? normal tgv goes 300 km/h |
| 21:57.47 | FlatFoot | angryuser, we were running at 240 when i worked at the tunnel |
| 21:58.05 | angryuser | ah ok in the tunnel, yes |
| 21:58.09 | lirakis_work | later all |
| 21:58.12 | FlatFoot | angryuser , we still have tooo many corners in the track to go that fast |
| 21:58.14 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 21:58.20 | [TK]D-Fender | iratik, You've done something tragically silly with your peer. Go read it till your eyes bleed. |
| 21:58.35 | iratik | qualify=no? |
| 21:59.03 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 21:59.09 | FlatFoot | [TK]D-Fender there is a theme of personal pain with your responses ;P |
| 21:59.21 | [TK]D-Fender | iratik, read the whole thing, and don't ask at every turn. lets see what you find in about 5 mins if you aren't 100% sure earlier |
| 21:59.29 | iratik | k |
| 21:59.59 | [TK]D-Fender | FlatFoot, they quality of MY mercy is non-existant :) |
| 21:59.59 | bsdwarrior | tkd-fender bail me out here, in a .call file Set: userfield=1234 does not work. in the database the field is blank |
| 22:00.03 | angryuser | <iratik> dont forget fromdomain option for ur peer, some providers require that |
| 22:00.12 | FlatFoot | [TK]D-Fender lol |
| 22:00.39 | FlatFoot | [TK]D-Fender are you related to that german bloke ;) |
| 22:00.46 | hmmhesays | [TK]D-Fender, is there any debugging info I can get from the IP 601 to try and figure out why it isn't displaying my instant messages? |
| 22:00.53 | [TK]D-Fender | bsdwarrior, Because thats a channel variable, that has nothing to do with setting a CDR value inherently |
| 22:01.02 | [TK]D-Fender | hmmhesays, Not a clue. |
| 22:01.09 | angryuser | <iratik>and you forget your register string |
| 22:01.30 | hmmhesays | [TK]D-Fender, you're supposed to be a polycom jedi! |
| 22:01.37 | b11d | hmmhesays... nice to see you |
| 22:01.45 | angryuser | <iratik> and your config is crap, sorry ;) |
| 22:01.53 | [TK]D-Fender | hmmhesays, and as I've said countless time, I never worked on its IM capabilities. |
| 22:02.06 | iratik | angryuser: sorry.. never done this before |
| 22:02.07 | FlatFoot | angryuser don't mess about eh ! |
| 22:02.08 | hmmhesays | holy crap b11d |
| 22:02.09 | hmmhesays | how goes it? |
| 22:02.10 | b11d | :) |
| 22:02.12 | [TK]D-Fender | angryuser, And you're assuming he NEEDS one. |
| 22:02.26 | b11d | good thanks... got married! |
| 22:02.42 | FlatFoot | b11d , now the trouble starts |
| 22:03.02 | angryuser | <[TK]D-Fender> well everybody need's a config even my grandmother |
| 22:03.18 | [TK]D-Fender | angryuser, ? |
| 22:03.29 | hmmhesays | married, crazy |
| 22:03.31 | [TK]D-Fender | angryuser, I'm talking about the register statement. |
| 22:03.35 | b11d | haha yeah tell me about it FlatFoot.. |
| 22:03.42 | b11d | its nice though.. i enjoy it |
| 22:04.03 | b11d | how have you been? |
| 22:04.15 | angryuser | <[TK]D-Fender> well i saw in step 2 he call out with it |
| 22:04.25 | [TK]D-Fender | angryuser, So? |
| 22:04.30 | b11d | TK.. I still have that weird zap problem.. any idea on where to look next? |
| 22:05.09 | angryuser | <[TK]D-Fender> register et you see the state, maybe pass is wrong, or pb like that |
| 22:05.16 | CrashSys | Anyone ever tried to do a 5TB MD Raid-5? |
| 22:05.17 | [TK]D-Fender | b11d : no idea, last we checked anything that doesn't bridge (even through a suppsedly non-bridged local channel) doesn't work. |
| 22:05.22 | CrashSys | just out of curiousity |
| 22:05.41 | [TK]D-Fender | angryuser, No, you simply have no understanding of what registration is for. |
| 22:05.45 | [TK]D-Fender | ~sipregister |
| 22:05.45 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 22:06.02 | b11d | TK.. thats right. Hmm.. :( Oh well! |
| 22:06.05 | angryuser | <[TK]D-Fender> nothing new for me |
| 22:06.08 | [TK]D-Fender | angryuser, Hence it has nothing to do with outbound calls at all |
| 22:06.26 | b11d | you'd think that since SIP<->ZAP worked, and echo works, pushing it through a local channel would work.. |
| 22:06.42 | [TK]D-Fender | b11d : Yeah, I'm at a loss from there... |
| 22:06.56 | b11d | I guess i'll start over from scratch.. |
| 22:07.00 | b11d | sometimes that helps |
| 22:07.32 | angryuser | <[TK]D-Fender> ok here is the situation, how do you auto failover to /analog/misdn if inet fails? |
| 22:07.35 | JenniferAkemi | [TK]D-Fender: are you in montreal? |
| 22:07.50 | *** join/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 22:08.02 | *** part/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 22:08.04 | angryuser | faster way for me is registration down |
| 22:08.07 | *** join/#asterisk clayc (n=feedthef@c-71-197-237-55.hsd1.wa.comcast.net) |
| 22:08.07 | [TK]D-Fender | angryuser, Put your second dial right after your first |
| 22:08.37 | [TK]D-Fender | angryuser, Register has no impact on your peer knowing if it will succeed/fail. |
| 22:08.53 | [TK]D-Fender | JenniferAkemi, Yup |
| 22:09.02 | clayc | hey, I'm having trouble having my ata register with my asterisk server |
| 22:09.06 | clayc | could anyone give me any tips? |
| 22:09.36 | FlatFoot | b11d , wait till the first daughter arrives then thers trouble , when she hits 10 |
| 22:09.58 | b11d | i'll murder every guy who comes within 10 feet of her! |
| 22:10.01 | b11d | :) |
| 22:10.02 | iratik | i think its a problem with the router |
| 22:10.09 | JenniferAkemi | [TK]D-Fender: that's great :) |
| 22:10.12 | angryuser | <[TK]D-Fender> if state of peer becomes unreachable , it changes te priority faster , if you do not register, you got 5 sec timeout |
| 22:10.19 | hmmhesays | b11d, I bet she thinks that hot |
| 22:10.23 | iratik | i got the register string in there... i forgot that icall has a sample config |
| 22:10.33 | b11d | probably :P |
| 22:10.49 | iratik | .... do i have to have my router 5060 port forwarded to this pbx if i want to make _outgoing_ calls? |
| 22:11.07 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 22:11.39 | FlatFoot | iratik , shouldn't have to , i have had a case whre that stopped voice traffic |
| 22:11.42 | [TK]D-Fender | iratik, Register = irrelevant to your issue |
| 22:11.54 | [TK]D-Fender | iratik, Your router is not the problem either. |
| 22:12.26 | *** join/#asterisk Vec (n=Vec@87-194-2-194.bethere.co.uk) |
| 22:12.33 | FlatFoot | iratik what phone ? i have to make my snoms expiry 60secs to keep traffic through some f/walls |
| 22:12.57 | [TK]D-Fender | iratik, I already told you I saw the problem and you did something clearly very wrong. Don't tell me your eyes have bled out already... |
| 22:13.11 | FlatFoot | iratik and get * to poke the phones to keep reg alive |
| 22:13.16 | Vec | Hi is anyone aware of a way to get asterisk Voicemail to e-mail voicemail's in MP3, without modifying the source ? |
| 22:13.25 | [TK]D-Fender | FlatFoot, Wrong tree... |
| 22:13.39 | FlatFoot | [TK]D-Fender Wrong Tree ? |
| 22:14.13 | clayc | anyone? |
| 22:14.16 | [TK]D-Fender | FlatFoot, the one you're barking up... |
| 22:14.41 | [TK]D-Fender | clayc, enable sip debug and see whats going on. Pastebin is your friend.... |
| 22:14.49 | FlatFoot | [TK]D-Fender ah OK ( uk phrase ) had to fetch beer must have missed summit |
| 22:15.42 | bsdwarrior | tkd-fender, I get the logic. so its not possible ? |
| 22:15.42 | iratik | FlatFoot: phoner |
| 22:16.07 | *** join/#asterisk clayc (n=feedthef@c-71-197-237-55.hsd1.wa.comcast.net) |
| 22:16.18 | FlatFoot | iratik , hmmm not come across those |
| 22:16.18 | [TK]D-Fender | bsdwarrior, Yes, you can set the userfield, and NO you cannot do it by any other means than in the dialplan. |
| 22:16.26 | iratik | FlatFoot: software sip phone |
| 22:16.28 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 22:16.29 | bsdwarrior | tkd-fender, Im pushing a bunch of calls via the manager. I want to then later on check the status of those calls, and if it was a no answer, busy, etc push the call again. Thats what I need to accomplish. (im using php) |
| 22:16.35 | [TK]D-Fender | iratik, wrong tree... fix your PEER <------ |
| 22:16.44 | iratik | eh? really? |
| 22:16.53 | [TK]D-Fender | iratik, I've told you 3 times already |
| 22:16.54 | hmmhesays | Bingo Got my IM |
| 22:16.59 | FlatFoot | iratik , ak ok don't use softph only big plastic things ( the wife likes them ) |
| 22:17.14 | clayc | Really destroying SIP dialog '1803fa728fe769d50@192.168.0.251' Method: REGISTER |
| 22:17.34 | clayc | SIP/2.0 404 UA Not Found |
| 22:17.37 | clayc | and that :) |
| 22:17.52 | [TK]D-Fender | clayc, so me the PROBLEM. pastebin the entire communication from start to finish |
| 22:18.15 | iratik | OMG |
| 22:18.20 | iratik | [TK]D-Fender: thank you |
| 22:18.37 | iratik | i thought you were talking about the trunk! |
| 22:18.40 | FlatFoot | [TK]D-Fender an aside are you near that big lake up from NY ? can't remember the name |
| 22:18.46 | [TK]D-Fender | iratik, things work so much better when you actually ALLOW a codec, now doesn't it? :p |
| 22:18.58 | iratik | well there is that too |
| 22:19.09 | angryuser | <[TK]D-Fender> have you encountered a proble when sip peer go unreachable after 2-3 hours? |
| 22:19.11 | *** join/#asterisk nixbox (n=oh@cpe-24-175-74-160.tx.res.rr.com) |
| 22:19.15 | nixbox | hi all |
| 22:19.34 | [TK]D-Fender | FlatFoot, Being in Quebec, not that close. |
| 22:19.55 | [TK]D-Fender | FlatFoot, 45 min driv to border, how much further to wherever you're thinking, I don't know |
| 22:20.01 | FlatFoot | [TK]D-Fender ah just you said the other nite you are 45mins from NY |
| 22:20.21 | [TK]D-Fender | FlatFoot, and NY has a bunch of big lakes. |
| 22:20.26 | FlatFoot | [TK]D-Fender sorry thought NY state was at the top |
| 22:20.33 | angryuser | <[TK]D-Fender> with external provider |
| 22:20.35 | [TK]D-Fender | FlatFoot, and Google Maps would give you a rather decent answer. |
| 22:20.44 | FlatFoot | [TK]D-Fender yeah true ta mr |
| 22:20.45 | clayc | ok |
| 22:20.56 | [TK]D-Fender | FlatFoot, I'm in CANADA. You know... the bigger land-mass ABOVE the USA? |
| 22:20.58 | clayc | i stuck it on pastebin |
| 22:21.15 | FlatFoot | [TK]D-Fender ah is that what it s |
| 22:21.21 | b11d | yeah.. America's Hat.. |
| 22:21.21 | FlatFoot | called ;) |
| 22:21.26 | nixbox | i have got a DID number and am able to receive calls on my PC via asterisk (running on another PC), the calling part can hear me, but I cannot hear them, what could be wrong? |
| 22:21.43 | nixbox | s/part/party |
| 22:22.28 | [TK]D-Fender | b11d, we're bigger, and we're on top.... if this were prison, you'd be out bitch :D |
| 22:22.28 | jjshoe | nixbox network configuration. |
| 22:22.28 | [TK]D-Fender | our* |
| 22:22.28 | b11d | you are right TK.. America is Canadas underwear. |
| 22:22.28 | FlatFoot | [TK]D-Fender LOVFL |
| 22:22.28 | clayc | haha |
| 22:22.28 | joe | [TK]D-Fender: hahaha |
| 22:22.28 | bsdwarrior | tkd-fender do you do any programming ? |
| 22:22.28 | [TK]D-Fender | bsdwarrior, yes |
| 22:22.39 | b11d | bsdwarrior... you work on the BSD port of zaptel right? |
| 22:22.41 | clayc | tkd-fender, you dirty canuck, you mind checking out my pastebin post? |
| 22:22.42 | clayc | :) |
| 22:22.59 | b11d | any known issues with no audio right now |
| 22:23.00 | b11d | ? |
| 22:23.04 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 22:23.10 | joe | [TK]D-Fender: are you still running your systems on centos? |
| 22:23.22 | FlatFoot | [TK]D-Fender is it true ? do all ca's end a sentence with EH! |
| 22:23.23 | bsdwarrior | b11d - no, I wish I was that cool. |
| 22:23.34 | b11d | oh.. |
| 22:23.41 | b11d | i thought you did.. oh well.. thanks anyways :) |
| 22:23.43 | [TK]D-Fender | clayc, Considering there are hundreds of pastebin sites and I wouldn't even dream of sifting through all of the posts on EACH of them... NO |
| 22:23.47 | bsdwarrior | how do you check that status of a call from a script then at a later time ? |
| 22:23.57 | clayc | ahh, which were you referring to? |
| 22:24.00 | [TK]D-Fender | clayc, Maybe its be useful if you gave us the LINK. |
| 22:24.07 | joe | hehe |
| 22:24.08 | nixbox | jjshoe, apparently i have no firewall blocking any sort of traffic, and the asterisk PC is accessible directly via the Internet, there is no NAT involved |
| 22:24.09 | clayc | http://pastebin.com/m57555c61 |
| 22:24.11 | clayc | sorry :) |
| 22:24.17 | [TK]D-Fender | joe, usually. |
| 22:25.01 | [TK]D-Fender | joe, and your sip.conf please... |
| 22:25.08 | [TK]D-Fender | clayc, rather... |
| 22:25.12 | joe | :) |
| 22:26.09 | [TK]D-Fender | bsdwarrior, if the call is in progress you have to check your channel for an idea of "progress". Otherwise its up to CDR and any other logging you feel like adding into your dialplan. |
| 22:26.22 | joe | [TK]D-Fender: I've been playing around w/ the epel rpm candidates but unless digium decides on a sound license they'll be unofficial for ever! But iirc you deploy from source |
| 22:27.24 | FlatFoot | l |
| 22:27.35 | FlatFoot | oops wrong screen |
| 22:27.42 | iratik | on the .so files in /usr/lib/asterisk/modules ... what does the "*" mean when you do ls -l in that directory? |
| 22:27.45 | bsdwarrior | tkd-fender, im running a php script as a daemon. thats why it would be nice to find out what happened to the call in cdr db. but There is not a unique way to do so |
| 22:28.18 | [TK]D-Fender | bsdwarrior, Sure, set the userfield |
| 22:28.19 | CrashSys | voicemail.conf should be 664 right? |
| 22:28.25 | *** join/#asterisk anthm][ (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:29.12 | bsdwarrior | tkd-fender, lol you told me I can only set it in the dialplan. im using the manger |
| 22:29.13 | bsdwarrior | manager |
| 22:29.25 | clayc | http://pastebin.com/m31781da7 |
| 22:29.33 | [TK]D-Fender | bsdwarrior, "thats nice". Now think about HOW that should be using the dialplan... |
| 22:30.12 | FlatFoot | [TK]D-Fender are you actually british ? |
| 22:30.41 | FlatFoot | [TK]D-Fender lots of UK type thoughts and phrases ;P |
| 22:30.51 | [TK]D-Fender | FlatFoot, no, born and live in Quebec |
| 22:31.10 | *** join/#asterisk beighto (n=chatzill@c-76-105-46-200.hsd1.ca.comcast.net) |
| 22:31.22 | lmadsen | [TK]D-Fender: frenchy! |
| 22:31.46 | [TK]D-Fender | lmadsen, va-t'ens tabarnac, tu me faites enneui! |
| 22:31.47 | [TK]D-Fender | :p |
| 22:31.55 | FlatFoot | [TK]D-Fender me uncle lived there for quite a few year ( 40 ) reckoned it was so good he went back after 2 years back here |
| 22:32.04 | lmadsen | I don't know what enneui means :) |
| 22:32.33 | beighto | I am getting a SIP response 302 "Moved Temporarily" back from ... and then Now forwarding Zap/7-1 to Local... what does this mean? Was that phone set to forward? |
| 22:32.43 | lmadsen | beighto: yes |
| 22:32.50 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:32.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:33.14 | beighto | lmadsen: easy enough, thanks... I would have checked myself but I am about 1000 miles away from the phone |
| 22:33.24 | CrashSys | Well damn... I didn't know Fender was canadian... guess i'll have to start saying "Fender for Prime Minister" |
| 22:33.26 | lmadsen | beighto: or you could have read the SIP RFC and found the code :) |
| 22:33.36 | clayc | i know it has to be something really stupid |
| 22:33.42 | clayc | I had it working before... |
| 22:33.52 | beighto | lmadsen: I barely know what I am doing as is |
| 22:34.08 | lmadsen | beighto: even more of a reason to read the RFC! |
| 22:34.25 | beighto | lol |
| 22:34.33 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:34.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:35.53 | timeshell | What's a suitable softphone to use with Asterisk that supports video other than CounterPath's? CounterPaths products dont' seem to support my camera :( |
| 22:36.11 | [TK]D-Fender | timeshell, Ekiga |
| 22:36.13 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 22:36.37 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 22:36.40 | timeshell | For windows... |
| 22:36.41 | lmadsen | timeshell: really? although the camera should be supported by the driver on the system and not the softphone... at least that's what i would think |
| 22:37.15 | kyron | lmadsen, true only if your softphone uses the "system" api... ;) |
| 22:37.34 | timeshell | lmadsen: That's what I thought too, but on some googling, it appears I'm not the only one having trouble getting some camera's to work with CounterPath's phones |
| 22:37.36 | lmadsen | I see... I would have thought it would I guess |
| 22:37.44 | lmadsen | timeshell: interesting |
| 22:38.09 | [TK]D-Fender | timeshell, Ekiga... |
| 22:38.21 | lmadsen | timeshell: http://www.gnomemeeting.org/index.php?rub=5&path=windows/windows |
| 22:38.26 | timeshell | Yes, TKD, thanks... downloads |
| 22:38.28 | timeshell | downloading* |
| 22:38.32 | hmmhesays | well my poly gets IM now |
| 22:38.40 | lmadsen | hmmhesays: neato! |
| 22:38.43 | timeshell | http://support.counterpath.com/viewtopic.php?t=5813&postdays=0&postorder=asc&start=0&sid=f585e64fc365a529ae2fa5fe538b8ee2 |
| 22:38.46 | lmadsen | my polycom takes 10 mins to reboot now :) |
| 22:38.50 | [TK]D-Fender | hmmhesays, indeed, great to hear |
| 22:39.08 | kyron | lmadsen, da hell you do to it?? |
| 22:39.17 | kyron | (to make sure I dont :) |
| 22:39.21 | lmadsen | kyron: updated to the latest firmware |
| 22:39.26 | lmadsen | took 60 seconds on 1.6.x |
| 22:39.26 | mvanbaak | anyone tried them darn MS phones with asterisk ? |
| 22:39.31 | kyron | boot rom? |
| 22:39.41 | lmadsen | kyron: bootrom I think is 4.x? |
| 22:39.44 | lmadsen | If I remember right |
| 22:39.49 | lmadsen | on an IP501 |
| 22:40.09 | kyron | hmm...bidding on one atm.. sip 3? |
| 22:40.13 | lmadsen | works fine... just takes forcockingever to reboot :) |
| 22:40.37 | FlatFoot | lmasden , nice use of tomesis |
| 22:41.14 | lmadsen | ya, bootrom 4.0.0.0423, bootblock 2.5.0, and 2.2.0.0047 application |
| 22:41.31 | lmadsen | FlatFoot: :) |
| 22:41.53 | hmmhesays | yeah now the only problem is its a pita to actually get to them |
| 22:42.02 | lmadsen | aye |
| 22:42.02 | hmmhesays | the messages button doesn't not have an IM option |
| 22:42.16 | lmadsen | I want a mini keyboard sidecar for mine :) |
| 22:42.25 | lmadsen | now that's be a wicked addon |
| 22:42.28 | lmadsen | that'd* |
| 22:42.33 | [TK]D-Fender | hmmhesays, don't not use double negatives! |
| 22:42.42 | timeshell | any mirrors for ekiga? |
| 22:42.45 | hmmhesays | yeah theres that typing thinking problem again |
| 22:42.49 | timeshell | really slow download |
| 22:42.55 | hmmhesays | the messages button doesn't have an IM option |
| 22:42.57 | *** part/#asterisk man_o_magic (n=chatzill@12.119.107.70) |
| 22:43.22 | [TK]D-Fender | timeshell, link works FINE : http://www.ekiga.org/index.php?rub=5&path=windows/windows |
| 22:43.39 | [TK]D-Fender | hmmhesays, you disabled it with "onetouchvoicemail" no doubt |
| 22:44.51 | hmmhesays | [TK]D-Fender, hmm no the message summary is displayed when I press messages |
| 22:45.15 | timeshell | slow |
| 22:45.24 | timeshell | 11KB/sec |
| 22:45.27 | [TK]D-Fender | timeshell, its just you then.. I got over 150k/s |
| 22:45.47 | timeshell | I'm on a 13 meg Fibre. NOT me. |
| 22:45.47 | [TK]D-Fender | hmmhesays, make sure the feature is enabled. |
| 22:45.53 | kyron | lmadsen, so...which part was "latest firmware"...cuz I am running sip 3 but not the same rom (on a 320) |
| 22:46.00 | [TK]D-Fender | timeshell, if I can get it at 150k/s, yes it is. |
| 22:46.00 | clayc | any ideas tkd-fender? |
| 22:46.04 | hmmhesays | when the message feature is not enabled it doesn't even accept IM |
| 22:46.07 | lmadsen | kyron: I guess I meant latest at the time :) |
| 22:46.11 | [TK]D-Fender | timeshell, Off my 5mbit DSL |
| 22:46.27 | kyron | lmadsen, ;) |
| 22:46.34 | [TK]D-Fender | clayc, where's the sip.conf entry I asked for? |
| 22:46.38 | timeshell | TKD: Well good for you. Obviously we are located in different place on the globe. |
| 22:46.38 | hmmhesays | feature.2.name="messaging" feature.2.enabled="1" |
| 22:46.41 | lmadsen | I'm very tired, so not much I say today is gonna make a whole lot of sense |
| 22:46.43 | clayc | its there, in the link |
| 22:46.48 | clayc | at the bottom of my pastebin |
| 22:46.56 | Maxous | Will a phone work behind nat through the internet? |
| 22:47.12 | timeshell | Maxous: Usually |
| 22:47.26 | lmadsen | all 5 on my desk do |
| 22:47.38 | timeshell | All 5 on my computer do :D |
| 22:47.40 | Maxous | Phone->(NAT)Router->{Internet}<-Router(NAT)<-Asterisk |
| 22:47.56 | Maxous | Do all of the features work? |
| 22:48.07 | lmadsen | everything I use does |
| 22:48.12 | Maxous | Cool. |
| 22:48.31 | lmadsen | you have to configure asterisk / FW|NAT at the other end correctly though |
| 22:48.41 | Maxous | Understood. |
| 22:48.56 | lmadsen | client side shouldn't require anything changed though really |
| 22:49.00 | Maxous | But both the Phone and Asterisk can be behind NAT. |
| 22:49.04 | timeshell | What country are you in TKD? |
| 22:49.06 | Maxous | Wow, that's cool. |
| 22:49.07 | lmadsen | I never had to forward ports or anything crazy for the phones |
| 22:49.20 | [TK]D-Fender | clayc, You made one pastebin that you've linked here. Its not in it |
| 22:49.27 | [TK]D-Fender | timeshell, Canada |
| 22:49.33 | Maxous | On the asterisk side i'm sure you would have to configure port forwarding. |
| 22:49.35 | lmadsen | Maxous: asterisk you have to configure 'externip' and 'localnet', plus forward the appropriate ports to asterisk |
| 22:49.50 | FlatFoot | TK is up for president next week ;p |
| 22:49.53 | [TK]D-Fender | Maxous, Go read : |
| 22:49.55 | [TK]D-Fender | ~sipnat |
| 22:50.04 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:50.05 | timeshell | How many hops on a tracert to www.ekiga.org? |
| 22:50.23 | Maxous | <PROTECTED> |
| 22:50.32 | timeshell | I start losing pings at hop 10 at oc192.m160.core.science.belnet.net [193.191.1.1] |
| 22:50.39 | FlatFoot | ~sausage |
| 22:50.40 | jbot | somebody said sausage was ground up animal parts stuffed into an sphincter, grilled so that you don't gag |
| 22:50.59 | lmadsen | ~sausage_party |
| 22:51.05 | [TK]D-Fender | timeshell, bungles up around17 |
| 22:51.11 | FlatFoot | ~beer |
| 22:51.11 | jbot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 22:51.15 | timeshell | 17 hops?? |
| 22:51.20 | lmadsen | mine dies at 12 |
| 22:51.26 | timeshell | I get there in 14 |
| 22:51.35 | lmadsen | but that just means hop 13 isn't responding to ICMP |
| 22:51.40 | [TK]D-Fender | timeshell, Whatever, the download link works fine for me :) |
| 22:52.31 | Maxous | [TK]D-Fender: Would you VLAN the WAN connection going to the asterisk? |
| 22:52.44 | [TK]D-Fender | Maxous, nope. just follow the guide |
| 22:53.02 | FlatFoot | 13 hops from UK |
| 22:53.07 | lmadsen | o.O |
| 22:54.22 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 22:54.47 | FlatFoot | jbot you awake ? |
| 22:55.31 | Corydon76-dig | ~botsnack |
| 22:55.31 | jbot | thanks, Corydon76-dig |
| 22:55.40 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 22:55.48 | FlatFoot | ~beer |
| 22:55.48 | jbot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 22:56.27 | FlatFoot | jbot can't handle his drink |
| 22:57.25 | FlatFoot | ~botsnack |
| 22:57.25 | jbot | thanks, FlatFoot |
| 23:02.47 | timeshell | ekiga is in Belgium. Are they affected by those cut undersea cables? |
| 23:03.44 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:04.34 | mvanbaak | no |
| 23:04.35 | FlatFoot | jbot do you get wacked on jbotsnacks ? |
| 23:04.35 | jbot | FlatFoot: thanks |
| 23:08.12 | adeel | if i have a ring group with multiple extensions in it, and the first extension has call forwarding enabled on the phone (polycom 601) will * keep going through the ring group, or will it honor the call forwarding request and complete the call, exiting out of the ring group? |
| 23:10.07 | mvanbaak | try it |
| 23:14.13 | b11d | damn this ZAP issue.. |
| 23:14.19 | b11d | damn it straight to ZAP hell.. |
| 23:14.30 | b11d | thank you |
| 23:14.46 | niekie | You're welcome :-) |
| 23:14.58 | FlatFoot | hell is good , they only w4nk in heaven |
| 23:15.06 | hades123 | ZAP hell, doens't sound like a nice place to be |
| 23:15.25 | b11d | no its not.. its a frustrating, dark place with little info on how to escape it |
| 23:16.21 | hades123 | it's better than asterisk hell I bet |
| 23:16.28 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 23:16.30 | FlatFoot | wassup zap ? |
| 23:16.33 | b11d | no asterisk is playing nice.. zap isnt. |
| 23:16.44 | b11d | or maybe they are both ganging up on me |
| 23:17.16 | russellb | contact tech support of your vendor |
| 23:17.16 | niekie | Nah, it's gotta be ZAP. |
| 23:17.38 | b11d | everything works except I get no audio between zap channels on the same channel bank.. |
| 23:17.39 | b11d | thats it. |
| 23:17.43 | b11d | sip<->zap works fine |
| 23:17.57 | russellb | what kind of card is it? |
| 23:18.04 | b11d | two analog phones on the same channel bank cannot hear each other.. |
| 23:18.10 | b11d | I have a Sangoma A104d T1 card |
| 23:18.15 | russellb | then contact their support |
| 23:18.17 | b11d | with a Rhino CB-24 FXS Channle Bank |
| 23:18.22 | b11d | why? it seems to be working fine.. |
| 23:18.29 | russellb | doesn't sound like it's working to me :) |
| 23:18.42 | b11d | I can call from the Channel Bank to a SIP phone across that T1 card and it works though |
| 23:18.52 | hmmhesays | finally got some coffee |
| 23:18.57 | hades123 | sounds like an oxymoron to me |
| 23:19.04 | hades123 | no audio , but it works fine |
| 23:19.09 | b11d | oh.. yeah :) |
| 23:19.10 | b11d | haha |
| 23:19.22 | b11d | I just dont really see how it could be the T1 card is all.. |
| 23:19.27 | hmmhesays | north dakota sucks |
| 23:19.35 | b11d | come to Northern Minnesota hmmhesays.. its SO much better |
| 23:20.00 | hades123 | any hwere is better |
| 23:20.06 | hades123 | than frozen hell I live in now |
| 23:20.23 | FlatFoot | move to uk get done sideways with the rough edge of a pineapple |
| 23:20.37 | hades123 | i have few inches of snow infront of my house |
| 23:20.40 | hades123 | can't go anywhere |
| 23:21.05 | hades123 | well only to the end of the driveway |
| 23:21.05 | FlatFoot | hades123 send me your snow i like it ;P |
| 23:21.12 | russellb | b11d: add tT options to Dial() and I bet it will work (because the calls won't get natively bridged down to zaptel / the card anymore) |
| 23:21.19 | hades123 | FlatFoot: No you don't |
| 23:21.37 | b11d | ok.. one sec. |
| 23:21.54 | FlatFoot | ain't seen snow (real) for 20 years |
| 23:23.03 | hades123 | lol there is a website that lists top 20 oxymorons, number 1 is : Microsoft Works |
| 23:23.24 | b11d | still doesnt work russellb.. |
| 23:23.28 | Strom_C | wow! that joke is so totally not 15 years old |
| 23:23.56 | b11d | Dial(Zap/30,20,tT) right? |
| 23:23.59 | hmmhesays | the classics are the best |
| 23:24.06 | hades123 | Storm_C: I am ony 2 years old |
| 23:24.10 | hmmhesays | is there a group missing there? |
| 23:24.16 | b11d | bacon and eggs walk into a bar.. the bartender turns to them and says "Hey! We dont serve breakfast here!" |
| 23:24.27 | hmmhesays | holy crap |
| 23:24.59 | hades123 | Storm_c: and you thought my jokes are bad |
| 23:25.16 | Strom_C | who is storm? |
| 23:25.31 | plik | he;s a vegetarian |
| 23:25.33 | hades123 | strom, my mistake |
| 23:26.03 | FlatFoot | isn't he code for summit ? |
| 23:26.10 | drmessano | Two priests walk into a bar.. the third one ducks |
| 23:26.19 | hades123 | lol |
| 23:26.49 | FlatFoot | two nuns in the shower ones says wheres the soap the other says yes it does |
| 23:26.59 | drmessano | LOL |
| 23:27.24 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 23:27.55 | b11d | i dont get it |
| 23:27.58 | hades123 | anybody have an effective way to end headaches |
| 23:28.03 | b11d | marijuana |
| 23:28.05 | *** join/#asterisk hmm-home (n=Neg@24-119-176-74.cpe.cableone.net) |
| 23:28.05 | FlatFoot | beer |
| 23:28.06 | plik | amputation? |
| 23:28.07 | drmessano | cut off your head |
| 23:28.09 | hades123 | other than a gun to my head |
| 23:28.23 | b11d | my friends brother shot himself yesterday! :( |
| 23:28.26 | FlatFoot | BEER |
| 23:28.26 | drmessano | Stab yourself in the foot |
| 23:28.33 | drmessano | You'll forget about the headache |
| 23:28.36 | hmm-home | couple of good looking chicks on cash cab right now |
| 23:28.44 | FlatFoot | b11d shit sorry to hear m8 |
| 23:29.07 | hades123 | damn all your solutions .. I will keep my head - ache |
| 23:29.10 | hades123 | thanks |
| 23:29.21 | hades123 | b11d: Yah, thats sad , sorry |
| 23:29.25 | hmm-home | b11d: fatality? |
| 23:29.32 | b11d | yeah |
| 23:29.33 | husimon | or stupidality? |
| 23:29.43 | b11d | haha yeah it WAS stupid.. but he did it. |
| 23:29.44 | hmm-home | s/fatality/fataly/ |
| 23:29.58 | hmm-home | gotta ask cause it could have been an accident |
| 23:29.59 | FlatFoot | only a brave man can kill themselves |
| 23:30.04 | b11d | yeah.. no it was on purpose. |
| 23:30.20 | husimon | of all the ways to go, i'd choose another one |
| 23:30.20 | hmm-home | I can see suicide only under certain conditions |
| 23:30.23 | FlatFoot | takes guts to pull the trigger |
| 23:30.23 | hades123 | was he depressed |
| 23:30.39 | hmm-home | FlatFoot: not if you are clinically depressed |
| 23:30.39 | b11d | yeah.. he was on the meds pretty hard too |
| 23:30.48 | husimon | i think i'd prefer drowning |
| 23:31.04 | FlatFoot | hmm-home true mental probs don't equate |
| 23:31.42 | hmm-home | every time suicide comes up, I think all those panties that I could have had thrown at me and all the guitar I will have missed out on |
| 23:31.55 | FlatFoot | husimon drowning takes too long |
| 23:31.56 | husimon | heh |
| 23:32.07 | husimon | flatfoot i dunno i've heard it's a pretty peaceful way to die |
| 23:32.21 | husimon | i guess c02 is even easier |
| 23:32.25 | husimon | if no one stops you |
| 23:32.37 | hmm-home | once you let that water into your lungs and you start to get euphoric from hypoxia |
| 23:32.39 | hades123 | you have 7 seconds of excruciating pain when the water hits your longs |
| 23:32.42 | hades123 | lungs* |
| 23:33.00 | husimon | better the missing with a gun |
| 23:33.08 | husimon | the=then |
| 23:33.11 | hmm-home | or worse living through the gun shot |
| 23:33.12 | FlatFoot | sex / heart attack now that dying |
| 23:33.29 | hades123 | sex / heart attack/ hell |
| 23:33.34 | plik | FlatFoot: depends who with ;) |
| 23:33.40 | hmm-home | sex and gunshot, now thats dying |
| 23:33.43 | FlatFoot | plik true |
| 23:33.54 | hmm-home | have her blow you away at the peak |
| 23:34.07 | husimon | she would have to be pretty crazy to do that |
| 23:34.13 | FlatFoot | ah now we are into japanese sex rituals |
| 23:34.41 | hmm-home | we've all seen csi |
| 23:35.07 | FlatFoot | hmm-home just finished watching csi |
| 23:35.08 | hades123 | death: such an interesting subject |
| 23:35.22 | FlatFoot | prefer criminal minds though |
| 23:35.33 | hmm-home | I don't watch it anymore had an ex I associate it with |
| 23:35.37 | hmm-home | but I used to |
| 23:35.39 | FlatFoot | hades123 thought provoking |
| 23:36.03 | hmm-home | i have to finish my facebook app now |
| 23:36.26 | FlatFoot | hmm-home why ? i have banned that at my work f/wall |
| 23:36.30 | hades123 | hmm-home: you write facebook app? |
| 23:36.35 | hmm-home | yeah |
| 23:36.45 | hmm-home | it displays users myspace photo albums on your facebook profile |
| 23:37.06 | hades123 | hmm-home: here is an idea for you, there was that app that lets you know who visted your profile etc.. |
| 23:37.21 | hmm-home | my thoughts 1. facebook is much less annoying than myspace. 2. if I never have to log into myspace again it will be too soon |
| 23:37.21 | hades123 | the guy stopped supporting it, and it went to ZAP hell |
| 23:37.29 | hmm-home | hades123: I could do that |
| 23:37.31 | b11d | ZAP hell sucks |
| 23:37.40 | hmm-home | donate me some dinero |
| 23:37.52 | FlatFoot | ZAP hell = proper sex |
| 23:38.02 | hades123 | hmm-home: honest, alot of people is waiting on an applicaiton like that |
| 23:38.09 | russellb | b11d: buy a card from a vendor capable and interested in fixing your problem :) |
| 23:38.14 | hmm-home | there isn't one already? |
| 23:38.14 | hades123 | not the one you have to click to tell th eperson you visited |
| 23:38.25 | hmm-home | you just want to see who viewd you |
| 23:38.27 | hmm-home | *viewed |
| 23:38.36 | b11d | i run a sangoma a104d card, it works tits.. |
| 23:38.47 | hades123 | hmm-home: one sec i WIll show you |
| 23:39.05 | hmm-home | put some nekkid ladies up around it, i will be more interested |
| 23:39.41 | mvanbaak | b11d: we have great succes with the sangoma a104d |
| 23:39.45 | b11d | i love it |
| 23:39.46 | hmm-home | www.vividguitars.com <-- they will be mine maybe nsfw |
| 23:40.20 | mvanbaak | <--- works at home so opens the link anywayz ;) |
| 23:40.25 | FlatFoot | night all time to watch mad karate bloke in china |
| 23:40.47 | mvanbaak | nice ones there hmm-home |
| 23:40.54 | b11d | night FlatFoot |
| 23:40.56 | hmm-home | mvanbaak: yeah I will have the brianna banks one |
| 23:41.09 | mvanbaak | uhhuh |
| 23:41.18 | plik | cya FlatFoot |
| 23:41.35 | plik | http://www.improveverywhere.com/2008/01/31/frozen-grand-central/ |
| 23:41.51 | plik | ^^^ awesome bunch of craxy-heads :) |
| 23:42.15 | d-k-t | anyone here using *BE on RHEL 5 or CentOS 5? |
| 23:42.44 | hmm-home | yes |
| 23:42.54 | hmm-home | wait BE? |
| 23:43.01 | hmm-home | no |
| 23:43.03 | hades123 | hmm-home: it's called Track Bot |
| 23:43.07 | d-k-t | Business Edition |
| 23:43.10 | hmm-home | link me to what it did |
| 23:43.16 | hmm-home | you can add me to facebook if you want |
| 23:43.25 | hades123 | http://mcmaster.facebook.com/apps/application.php?id=17227527808&b&ref=pd |
| 23:43.37 | hmm-home | and you say it doesn't work anymore? |
| 23:43.54 | hades123 | out of the application page: Announcement (aka Why Track Bot is Screwed) |
| 23:44.19 | russellb | d-k-t: if you're having an issue, please contact support@digium.com |
| 23:44.24 | d-k-t | plik, grand canal runs past about 100m from here |
| 23:44.42 | hades123 | hmm-home: the guy had 10 million hit crashed his servers. |
| 23:44.51 | hades123 | hmm-home: don't forget my commission |
| 23:44.51 | d-k-t | russellb, not got that far yet, waiting for the software and servers to arrive |
| 23:45.01 | russellb | ah.. |
| 23:45.40 | d-k-t | russellb, just trying to plan in advanced and noticed the 'only supported if you use RHEL4/Fedora3/4 or the packaged distro' bit |
| 23:46.05 | hmm-home | hades123: it only crashed yesterday |
| 23:46.32 | hades123 | hmm-home: Nooooo , it's beeen like this for ages now |
| 23:46.46 | d-k-t | plik, and it's probably frozen too |
| 23:46.48 | hmm-home | look at the comments man, there were just new ones posted yesterday |
| 23:47.10 | hades123 | I know, that's how popular it was |
| 23:47.19 | hades123 | go all all the way back in the comments, it; over 1000 comments |
| 23:47.33 | b11d | plik.. that was nuts.. |
| 23:47.34 | *** join/#asterisk AndyGraybeal (n=andy@node87.36.251.72.1dial.com) |
| 23:48.06 | hades123 | hmm-home: dude I know for a fact it's down for couple of month now at least |
| 23:48.37 | hmm-home | hades123 add me to your facebook |
| 23:49.47 | plik | yeah :) |
| 23:50.20 | hades123 | hmm-home: dude, my facebook, has tons of private shit :) there is no way in hell I am letting you see those pics :D |
| 23:50.51 | hades123 | (joke) |
| 23:51.23 | hades123 | hmm-home: if you made money of this app, I want 5% |
| 23:51.26 | hades123 | :D |
| 23:52.20 | hmm-home | lol |
| 23:52.56 | *** join/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 23:53.33 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:54.24 | Inssomniak | Im trying to debug why a spa-3102 wont register with my box, where are the logs usually dumped to? |
| 23:56.32 | b11d | I notice my no audio problem is one way now.. sip -> zap doesnt work, but zap -> sip does... |
| 23:57.04 | mvanbaak | nat ? |
| 23:57.22 | hades123 | the usuall suspect |
| 23:57.38 | mvanbaak | for audio problems with sip? yes |
| 23:59.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:59.26 | b11d | no.. |
| 23:59.29 | b11d | its on a closed system.. |
| 23:59.32 | hades123 | I wanna go buy a Cisco 871 router , I just configured one, made me feel how idiotic is my linksys |
| 23:59.48 | b11d | sip phone -> switch <- asterisk -> t1 -> channel bank <-- analog phone |
