| 00:00.02 | PatrickDK | 8 |
| 00:00.03 | smash- | wtf |
| 00:00.10 | smash- | how come there is like 100 in the cat3 cables here |
| 00:00.16 | PatrickDK | 8 in network |
| 00:00.16 | smash- | im so confused |
| 00:00.17 | tclark | shmaltz: sorta do your nned generla info & some specfic frimware setting on that adit 600 |
| 00:00.22 | Uther_P | help me!! |
| 00:00.23 | smash- | i know 8 in cat5 |
| 00:00.34 | smash- | anyway |
| 00:00.36 | PatrickDK | you can it it anywhere from 2pair 4pair 12pair 25pair 50pair +++ |
| 00:00.36 | smash- | im off work |
| 00:00.40 | PatrickDK | it doesn't matter |
| 00:00.43 | smash- | o |
| 00:00.51 | Uther_P | I updated zaptel, libpri and asterisk... and the farkin zaptel crap wasn't working... I reverted BACK to what I had already and it still wont work |
| 00:00.59 | shmaltz | tclark, nope I need just general (capacity and reliablity) info |
| 00:01.07 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
| 00:01.10 | tclark | ok they are gear !! |
| 00:01.18 | Uther_P | it gives me this error when modprobing wcfxs "ZT_CHANCONFIG failed on channel 1: No such device or address (6) ; FATAL: Error running install command for wctdm |
| 00:01.23 | *** part/#asterisk blueskiesokie (~blueskies@65.242.87.151) |
| 00:06.43 | Uther_P | does zaptel normally say this when you compile "*** Uh-oh, you have stale module entries. You messed with SUBDIRS" ? |
| 00:07.00 | p1tst0p | lo, trying to use sjphone on Pocket PC, an i just keep getting, 407, proxy auth required.. any ideas ? followed the sjphone guide on wiki. |
| 00:13.38 | p1tst0p | hmm anyone know what, Got SIP response 481 "Call Does Not Exist" back from 192.168.0.10 |
| 00:13.46 | p1tst0p | *means in my logs |
| 00:15.57 | terrapen | hrmmm |
| 00:16.11 | terrapen | i need to figure out how to get a remote phone (think: at home) onto the office phone system |
| 00:16.14 | terrapen | without using a VPN |
| 00:16.41 | terrapen | i can get the the office Asterisk installation to dial the phone using Dial() |
| 00:16.51 | shmaltz | MDT is -GMT by how much? |
| 00:16.51 | terrapen | but i'm not sure how i am going to let the phone dial office extensions |
| 00:16.59 | terrapen | without calling into the main number first |
| 00:17.09 | terrapen | i could do some kind of lame-o caller*id authentication |
| 00:17.12 | terrapen | but that's pretty weak |
| 00:17.18 | paulc_ | shmaltz: 6 hours? or 7.. PST = -8 and I think mountain time is 1 over? |
| 00:17.36 | terrapen | i need something like certificates |
| 00:18.42 | terrapen | anyone make a hardware SIP proxy? |
| 00:19.16 | shmaltz | paulc_, thanks |
| 00:19.22 | paulc_ | :) |
| 00:24.50 | *** join/#asterisk YoYo (YoYo@dilbert.psknet.com) |
| 00:26.37 | *** join/#asterisk fgravato (~frankie@ool-44c02d18.dyn.optonline.net) |
| 00:29.59 | JohnnyC | Whats the best way to dial an IP adress with a IP hardware Phone ? |
| 00:30.03 | JohnnyC | using a macro ? |
| 00:31.14 | KalD|WORK | hey guys - I'm making a call w/ Zap and it always dials out on the 5th channel on the PRI - is there any reason why it wouldnt just dial out on the 1st channel? |
| 00:31.43 | niZon | JohnnyC: Edit the phone's internal dialplan maybe? |
| 00:31.44 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 00:32.11 | TomL | trunk channel selection isn't done by the phone |
| 00:32.33 | DenisL | terrapen: Why not just setup a VPN connection between home and office and connect it to the Asterisk server that way? |
| 00:32.35 | TomL | KalD|WORK: maybe your zapata.conf isn't quite right |
| 00:32.40 | JohnnyC | niZon: edit the internal dialplan ... its a very simple phone I dont think its able to have a dialplan ?! |
| 00:34.04 | KalD|WORK | TomL pri1 starts at 1 goes to 23, pri2 starts at 24 - 47, pri3 49 - 71, pri 4 73 - 95 correct? |
| 00:34.06 | dwmw2_gone | kphone and linphone both have been persuaded to play real sound instead of white noise by byteswapping appropriately. But still I only _receive_ sound with either of them; anything I send doesn't get there. |
| 00:34.28 | JohnnyC | I wanted the capability to dial an IP was ouside editing phones configurations |
| 00:34.39 | brycec | Could someone help me diagnose/solve a problem? When I call in from an external line into one of the POTS lines, Ast will answer but randomly and never in the same place it will just hangup |
| 00:34.44 | xkev | kald sounds about right |
| 00:35.01 | KalD|WORK | then dial(zap/g3/num) correct? |
| 00:35.01 | xkev | ..mine are that way |
| 00:35.11 | xkev | where do you define group= |
| 00:35.15 | TomL | KalD|WORK: there's a lot more to it than that, I would hope |
| 00:35.19 | brycec | Is there some way to get more verbose information for zaptel channels? |
| 00:35.30 | xkev | did you set channel => 1-23 ? |
| 00:35.41 | xkev | group = 1 |
| 00:35.42 | xkev | channel => 1-23,25-47 |
| 00:35.44 | xkev | ^ mine |
| 00:35.52 | Goldenear | Why can't I transfer native bridged calls ? Do only calls going via * can be tranfered/parked ? |
| 00:36.08 | xkev | CLI> show channels <- make sure 1-4 aren't in use already too (stuck zombie or something) |
| 00:36.09 | KalD|WORK | group = 1, channel => 1-23 group = 2, channel => 25-47 |
| 00:36.11 | KalD|WORK | etc |
| 00:36.39 | xkev | do the pris do different things? why 4 groups |
| 00:36.54 | KalD|WORK | each PRI interface is connected to a different PBX |
| 00:37.00 | xkev | ah k |
| 00:37.15 | xkev | well, you shouldn't start at chan 5 unless 1-4 are in use |
| 00:37.15 | TomL | hmm 4 groups, dialing starts at trunk 5... coincidence? |
| 00:37.27 | KalD|WORK | for this one we are calling into deadlogic hardware and always get incoming/outgoing on channel 5 |
| 00:37.51 | KalD|WORK | 1-4 are not in use =( |
| 00:37.55 | JohnnyC | how can I dial an IP address ? |
| 00:38.01 | JohnnyC | anyone has an idea ? |
| 00:38.04 | TomL | you checked "show channels"? |
| 00:38.07 | KalD|WORK | JohnnyC - what protocol? |
| 00:38.14 | JohnnyC | SIP |
| 00:38.17 | KalD|WORK | TomL, yeah - everything is empty (all channels) |
| 00:38.21 | xkev | zap show channel 4, zap show channel 5. any difference? |
| 00:38.26 | KalD|WORK | JohnnyC, dial(sip/user@ipaddress) |
| 00:38.38 | JohnnyC | Kaid shoudl I make a macro ? |
| 00:38.45 | JohnnyC | I have soft phones and hardware phones |
| 00:39.05 | KalD|WORK | well - ok I should say this: it is the 5th channel on span 3 so the channel is 77 |
| 00:39.10 | JohnnyC | can I dial just the IP address instead of user@ipaddress ? |
| 00:39.22 | xkev | ok, you get my drift though |
| 00:39.30 | KalD|WORK | JohnnyC, yes |
| 00:39.34 | Beirdo | oooh, my fixed wireless terminal has left Memphis, should be delivered tomorrow 5pm (estimated) |
| 00:39.36 | KalD|WORK | xkev - I'm lookin now |
| 00:40.00 | Beirdo | the fun will be getting Bell Mobility to activate it on their cell network :) |
| 00:40.07 | JohnnyC | any idea were I can find this ? |
| 00:40.43 | bjohnson | terrapen: a few options are outlined on the tips and tricks wiki page under user authentication |
| 00:41.21 | KalD|WORK | xkev, same no difference between 77 and other channels |
| 00:41.30 | KalD|WORK | JohnnyC, find what? |
| 00:41.44 | JohnnyC | a macro to dial Sip IP |
| 00:41.47 | JohnnyC | to IP |
| 00:41.53 | xkev | kald hrm |
| 00:42.13 | KalD|WORK | How do you tell Dial to start at the end or beginning of the PRI channels for outbound calls? |
| 00:42.33 | KalD|WORK | or can you set that? i.e. all incoming are high outgoing low |
| 00:42.35 | xkev | I haven't seen such an option, I assumed it was front-loading only |
| 00:42.51 | xkev | ..but that is a common feature in telco land |
| 00:43.52 | xkev | might set debug 1, or pri debug and look for failures causing it to roll down the list |
| 00:44.02 | xkev | verbose 4, all that crap |
| 00:44.57 | KalD|WORK | omg.. isam! I have never used 'set debug' |
| 00:45.43 | PTG123 | anyone know anyone looking for work that likes talking on the phone and wants to telecommute? :) |
| 00:46.00 | xkev | hiring for a phone sex line? :P |
| 00:46.10 | PTG123 | hah no mortgage stuff :) need an assistant |
| 00:46.17 | PTG123 | pays really well :) |
| 00:46.24 | KalD|WORK | ok w/ debug I get this: -- Moving call from channel 77 to channel 73 |
| 00:46.24 | KalD|WORK | Mar 22 16:44:48 WARNING[213005]: chan_zap.c:7012 pri_fixup_principle: Whoa, there's no owner, and we're having to fix up channel 77 to channel 73 |
| 00:46.37 | xkev | buh |
| 00:49.13 | brycec | Can anyone help me with spontaneous hagups on Zaptel channels?? |
| 00:50.11 | KalD|WORK | xkev, w/ pri debug span I get my call going out chan 5 ... (span 4 so it's really channel 77) |
| 00:50.14 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 00:51.04 | KalD|WORK | hmm yet I make another call and it is hitting channel 2 |
| 00:51.12 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
| 00:51.23 | xkev | kald you got me |
| 00:51.44 | sudhir492 | something weird is happening to my new Asterisk box |
| 00:52.06 | sudhir492 | inspite of installing g729 license, I get the following message: chan_sip.c:2994 process_sdp: No compatible codecs! |
| 00:55.26 | KalD|WORK | JohnnyC, make your sip clients register w/ asterisk w/ their exten - then you can do Dial(sip/exten) |
| 00:55.55 | JohnnyC | oki |
| 00:56.15 | JohnnyC | my doubt is how can I dial an IP in an IP Phone |
| 00:56.29 | KalD|WORK | you cant |
| 00:56.38 | KalD|WORK | unless you do some dtmf stuff =) |
| 00:57.39 | KalD|WORK | i.e Read(IPV41,3) Read(IPV42,3) Read(IPV43,3) Read(IPV44,3) then dial(sip/${IPV41}.${IPV42}.${IPV43}.${IPV44}) |
| 00:58.07 | KalD|WORK | tho dont do that cuz it might work but most likely not |
| 00:58.30 | JohnnyC | hehe |
| 00:59.05 | *** join/#asterisk FxMulder (~me@209.159.235.241) |
| 00:59.19 | KalD|WORK | i.e. 10#0#0#120# would dial sip/10.0.0.120 =) but that is all the help I'll give you =) there is a better way I'm sure - why dial ip from an ip phone anyways? |
| 00:59.41 | *** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) |
| 01:03.09 | *** join/#asterisk kleper (~kleper@200.30.69.177) |
| 01:03.12 | kleper | hi |
| 01:04.37 | brycec | Can anyone help me with spontaneous hagups on Zaptel channels?? PLEASE |
| 01:05.55 | mstocco | Power is out in Beverly Hills in case anyone has servers in LA area |
| 01:05.59 | tzanger | brycec: TURN BUSYDETECT OFF |
| 01:06.12 | fgravato | hey tzanger |
| 01:06.16 | fgravato | long time no see |
| 01:06.17 | fgravato | heh |
| 01:06.24 | tzanger | fgravato: hello |
| 01:06.28 | tzanger | isn't it kind of late out there |
| 01:07.32 | tzanger | busydetect needs to be renamed |
| 01:07.37 | tzanger | in fact I think I'm gonna write a patch for it right now |
| 01:07.40 | tzanger | -busydetect |
| 01:07.47 | tzanger | +randomly_disconnect_my_calls |
| 01:08.05 | brycec | lol |
| 01:08.13 | brycec | where do I define it? |
| 01:08.18 | tzanger | zapata.conf |
| 01:08.25 | tzanger | or is it zaptel.conf I can never remember |
| 01:08.28 | tzanger | the one in /etc/asterisk |
| 01:08.32 | brycec | zapata is under ast |
| 01:09.13 | terrapen | hmmm |
| 01:09.37 | brycec | What's the default setting of busydetect? So far, it's been commented out. |
| 01:09.53 | terrapen | tzanger, would you feel comfortable setting up a customer with a phone system that uses a VoIP provider exclusively for PSTN access? |
| 01:10.09 | terrapen | i've been using sixTel and NuFone here at work |
| 01:10.09 | tzanger | terrapen: if they understood that there WAS NO 911, then sure |
| 01:10.11 | tzanger | I have several that way |
| 01:10.13 | terrapen | and they do a good job |
| 01:10.31 | terrapen | tz, my plan is to put a POTS phone in their office for fax and 911 |
| 01:10.44 | terrapen | maybe i could put an FXO card in the * server |
| 01:10.50 | terrapen | and make 911 available through that POTS line |
| 01:11.08 | kleper | how can connect a fsxo gateway to asterisk? |
| 01:11.25 | terrapen | tzanger: have your clients complained about performance at all? i can deal with the occasional jitter but i worry about the reliability of these providers |
| 01:11.33 | PTG123 | why not use a voip provider that supports e911? |
| 01:11.33 | terrapen | what the hell is a fsxo gateway? |
| 01:11.55 | kleper | o sorry is a FXS |
| 01:11.59 | terrapen | <PROTECTED> |
| 01:12.00 | kleper | or FXO |
| 01:12.03 | kleper | or FXSO |
| 01:12.13 | terrapen | klep, read the wiki |
| 01:12.15 | tzanger | terrapen: the odd jitter, the odd dropped call but they have been VERY happy |
| 01:12.22 | terrapen | ~wiki |
| 01:12.28 | terrapen | err |
| 01:12.33 | terrapen | ~jbot wiki |
| 01:12.35 | terrapen | ugh |
| 01:12.52 | terrapen | tz: i'm going to explain to them that they will save a tremedous amount of money |
| 01:12.59 | brycec | well any jitter or any dropped call is not good for me |
| 01:13.01 | tzanger | terrapen: yup |
| 01:13.03 | terrapen | but they may (very rarely) experience jitter or a dropped call |
| 01:13.20 | terrapen | tz: how do you handle the payments for their voip service? |
| 01:13.27 | terrapen | most of the providers like pre-pay |
| 01:13.36 | terrapen | i guess i could teach them how to add money to the account |
| 01:13.44 | tzanger | terrapen: I basically tell them it's a cell with really good rates and excellent quality |
| 01:15.33 | terrapen | which provider did you set them up with? |
| 01:15.40 | tzanger | nufone of course |
| 01:15.43 | tzanger | and myself for local PRI hopoff |
| 01:15.47 | fgravato | nice |
| 01:15.51 | terrapen | i use nufone at home and it works great but i'm probably not going to recommend them for my customers |
| 01:15.51 | opus_ | who was the guy in here calling pakistan? |
| 01:16.05 | terrapen | i can put up with the nufone customer service |
| 01:16.09 | fgravato | voipjet isn't to bad |
| 01:16.10 | terrapen | but im not sure that my customers will |
| 01:16.15 | fgravato | pretty decent rates |
| 01:16.18 | fgravato | to europe |
| 01:16.40 | fgravato | sixtel aka iax.cc 50/50 |
| 01:16.46 | kleper | the wiki of the voip-info.org??? |
| 01:16.47 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
| 01:17.01 | fgravato | to this day prefer nufon |
| 01:17.02 | fgravato | e |
| 01:17.18 | terrapen | fgravator: 50/50 what? |
| 01:17.26 | fgravato | sixtel |
| 01:17.28 | fgravato | 50 upt |
| 01:17.28 | terrapen | i use sixtel and nufone |
| 01:17.36 | fgravato | 50 some lagging |
| 01:17.40 | fgravato | or erroring out |
| 01:17.49 | terrapen | see, i have a super-solid ping to sixtel |
| 01:17.51 | terrapen | like 13ms |
| 01:17.56 | terrapen | and steady. |
| 01:17.59 | fgravato | kinda helps |
| 01:18.04 | fgravato | :-) |
| 01:18.10 | terrapen | i will set them up with sixTel and maybe a NuFone backup |
| 01:18.26 | fgravato | i'm using Nufone along with Coloco |
| 01:18.33 | tzanger | I have a sixtel number but I have not been overly happy with it |
| 01:18.34 | fgravato | at home |
| 01:18.38 | terrapen | i'm going to recommend Cisco 7960's to them |
| 01:18.48 | terrapen | or, if they balk, Polycom IP500s |
| 01:19.14 | terrapen | If they still balk, I will tell them to get a Rhino channel bank |
| 01:19.17 | tzanger | nufone I've been extremely happy with |
| 01:19.28 | terrapen | and we will put POTS phones on their desk |
| 01:19.38 | Beirdo | nufone I'm just waiting for my DID that I emailed for yesterday :) |
| 01:19.46 | Beirdo | but I like :) |
| 01:19.49 | tzanger | Beirdo: :-) I only use them for termination |
| 01:19.51 | terrapen | tz: these people will need a 210 area code DID tho |
| 01:19.55 | fgravato | tzanger is actually person that recommended them to me |
| 01:20.06 | *** join/#asterisk pigpigpig (~pig@165.21.246.202) |
| 01:20.12 | nestAr | my sixtel account is ok.. i don't actuall use it too much though.. |
| 01:20.22 | terrapen | i've never had a problem with nufone, aside from the occasional jitter and the non-existant customer service |
| 01:20.27 | terrapen | but you get what you pay for |
| 01:20.31 | terrapen | and you dont pay much @ nufone |
| 01:20.32 | nestAr | lol |
| 01:20.35 | tzanger | terrapen: jitter's not on their network as far as I can tell |
| 01:20.38 | file | customer service? what's that?!? |
| 01:20.42 | terrapen | tz: its probably not |
| 01:20.47 | nestAr | i've been using the companies pri.. |
| 01:20.51 | fgravato | hahah nice file |
| 01:20.56 | nestAr | making long distance calls on the company dime |
| 01:20.59 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 01:21.00 | terrapen | i am willing to put up with no customer service |
| 01:21.05 | terrapen | but my clients are not |
| 01:21.11 | tzanger | terrapen: you become their customer service |
| 01:21.14 | terrapen | oh, check this out |
| 01:21.23 | *** join/#asterisk RazaMetaL (~razametal@pc.gsalas.manta.telconet.net) |
| 01:21.25 | terrapen | i'm going to set my client up with a special extension |
| 01:21.33 | terrapen | when they dial that, it interfaces them into my RT system |
| 01:21.34 | RazaMetaL | hi all .. .greetings from ecuador <g> |
| 01:21.36 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:21.36 | tzanger | but I tell you, I have never had a problem with them... jerjer's been personally helping test the new jitter buffer |
| 01:21.41 | terrapen | so they can submit a helpdesk request |
| 01:21.45 | terrapen | and i will bill them |
| 01:21.57 | tzanger | yup |
| 01:22.04 | fgravato | nice tzanger |
| 01:22.19 | terrapen | I'll make them dial an authorization code first |
| 01:22.24 | terrapen | which only the boss will have |
| 01:22.37 | terrapen | so if they dial that and leave a message, it counts as billable |
| 01:22.52 | fgravato | tzanger -- is Jerjer gonna deploy that on Nufone or that just private thing? |
| 01:23.00 | tzanger | fgravato: it is in -HEAD right now |
| 01:23.13 | tzanger | I think he's waitin gon a few little bugs before it's on switch-1 and -2 |
| 01:23.13 | terrapen | tz: do your clients run -HEAD? |
| 01:23.23 | tzanger | terrapen: oh hell no. |
| 01:23.28 | tzanger | terrapen: I run -HEAD on their machines |
| 01:23.28 | terrapen | what do you run? |
| 01:23.39 | terrapen | i mean |
| 01:23.40 | tzanger | they have NO idea what they're running, there's a box in the closet that the phones are wired to, that's all they know |
| 01:23.45 | terrapen | what did you set up on their machines |
| 01:23.49 | tzanger | -HEAD |
| 01:23.50 | terrapen | tz: that's what i mean |
| 01:23.53 | terrapen | err meant |
| 01:24.03 | terrapen | i want the most reliable setup for them |
| 01:24.12 | terrapen | im wondering if i should have Asterisk restart nightly |
| 01:24.18 | tzanger | terrapen: ??? why |
| 01:24.19 | terrapen | i dont want a bunch of calls from these people |
| 01:24.26 | *** join/#asterisk ToyMan (~stuq@user-0cevdks.cable.mindspring.com) |
| 01:24.29 | terrapen | i need something uber-fucking-stable |
| 01:24.35 | fgravato | Terrapen |
| 01:24.36 | tzanger | my asterisk boxes stay up until I update -HEAD which is usually between 1 day and 3 weeks |
| 01:24.40 | fgravato | follow whats on the Wiki |
| 01:24.42 | terrapen | personally, i've never had to restart my Asterisk installation at work |
| 01:24.49 | fgravato | there's good writeup on there |
| 01:25.01 | terrapen | fgrav: under which topic, do you know? |
| 01:25.25 | tzanger | I've never had to restart asterisk for any technical reason... always because I want to update or have made some change that required a restart |
| 01:25.32 | terrapen | same here |
| 01:25.41 | tzanger | and I've been running * for over a year |
| 01:25.42 | terrapen | i saw some poster on the wiki saying he does it 2x a day |
| 01:25.52 | terrapen | but he is probably a retard |
| 01:25.52 | tzanger | terrapen: he's fucked up that is why :-) |
| 01:26.00 | tzanger | I personally dislike the wiki a great deal |
| 01:26.04 | terrapen | hah |
| 01:26.12 | terrapen | the wiki was very useful to me at first |
| 01:26.19 | terrapen | but there is a lot of BS on there, for sure |
| 01:26.23 | fgravato | http://www.voip-info.org/wiki-Asterisk+administration |
| 01:26.31 | *** join/#asterisk che (~che@che.user) |
| 01:26.39 | tzanger | it's impossible to maintain |
| 01:26.43 | tzanger | it's impossible to fidn anything |
| 01:26.54 | tzanger | I appreciate the effort whoever runs it has put in to it, it is a thankless job |
| 01:27.02 | tzanger | but it's not a good resource, I am very sorry to say that |
| 01:27.29 | dwmw2_gone | the beauty of the wiki is that the barrier to contributions is very low |
| 01:27.30 | fgravato | granted i rarely restart asterisk unless i just did update from cvs |
| 01:27.35 | tzanger | and I don't have a better answer for how to do it, which is why I don't bitch about it |
| 01:27.40 | dwmw2_gone | the problem with the wiki is that the barrier to 'contributions' is very low |
| 01:27.45 | tzanger | dwmw2_gone: that's also the VERY BAD thing about the wiki |
| 01:27.59 | dwmw2_gone | tzanger: indeed |
| 01:28.09 | dwmw2_gone | RtpPacket.h:35:2: error: #error RTP only works with little endian -- fix. |
| 01:28.09 | dwmw2_gone | bah |
| 01:28.12 | *** join/#asterisk IQ (~IQ@70-59-164-139.omah.qwest.net) |
| 01:28.28 | che | well why not have 2 branches of wikis? |
| 01:28.36 | fgravato | anyone have issues with wrt54g and opening up ports for rtp for sip clients |
| 01:28.51 | fgravato | i'm about to trash this wrt54g for linux box and iptables |
| 01:28.52 | che | one stable that is controlled... with a high barrior of merging new infos in |
| 01:28.59 | terrapen | when you set up VLANs on a switch, is that on a port-by-port basis or a MAC address basis? |
| 01:29.00 | che | and an unmaintained one where people have easier access to |
| 01:29.01 | *** join/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 01:29.17 | dwmw2_gone | p'raps I should just install asterisk and use its console mode? |
| 01:29.22 | tzanger | terrapen: I don't use it |
| 01:29.24 | tzanger | at all |
| 01:29.29 | terrapen | ie., do you say "Ports 1-4 are their own VLAN" or "MAC xx:xx:xx.. and yy:yy:yy.. are their own VLAN" |
| 01:29.34 | tzanger | terrapen: oh |
| 01:29.36 | che | this way you have a collection of reliable information and on the other hand dont turn away people to contribute ;) |
| 01:29.37 | tzanger | I have a VLAN for that |
| 01:29.40 | tzanger | it's port-based |
| 01:29.44 | terrapen | tz: what kind of switch do you use for your clients? |
| 01:29.53 | terrapen | tz: so you do two ethernet drops per desk then? |
| 01:29.56 | tzanger | oh any old POS, I don't do SIP phones |
| 01:29.59 | *** join/#asterisk Rodms (~Rodrigo@200164134065.user.veloxzone.com.br) |
| 01:29.59 | tzanger | T100P+channel Bank |
| 01:30.06 | terrapen | oh |
| 01:30.08 | tzanger | I don't do SIP |
| 01:30.11 | tzanger | it's brain damaged |
| 01:30.11 | terrapen | ok |
| 01:30.16 | tzanger | IAX or nothin |
| 01:30.19 | TechDawg | Can anyone recommend a good device that will connect a POTS line to a SIP device and transport that call to our Asterisk server? |
| 01:30.20 | fgravato | tzanger : Great term |
| 01:30.23 | terrapen | im trying to figure out how i'm going to wire these guys |
| 01:30.36 | terrapen | ideally, i want one ethernet drop per desk |
| 01:30.36 | fgravato | spa3000 |
| 01:30.41 | fgravato | techdawg |
| 01:30.43 | tzanger | terrapen: yeah unfortunately there aren't many good choices for hardphones |
| 01:30.48 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 01:31.00 | tzanger | I tend to use analog phones or PRI and ocnnect it to their existing KSU |
| 01:31.04 | TechDawg | And the manufacturer being fgravato? |
| 01:31.06 | tzanger | that way they get to keep their phones |
| 01:31.09 | Sedorox | terrapen: phone with two ports? |
| 01:31.20 | fgravato | TechDawg Sipura SPA3000 |
| 01:31.29 | terrapen | sederox: but how would you keep someone else's computer from hogging bandwidth on the switch? |
| 01:31.37 | fgravato | QOS |
| 01:31.39 | fgravato | terrapen |
| 01:31.39 | Sedorox | true |
| 01:31.52 | tzanger | fgravato: I use my world-renowned rc.tc script |
| 01:31.54 | terrapen | fgrav: ok, so which switch would you buy (for 16 ports) |
| 01:31.56 | FuriousGeorge | ahoy. just tried to get that "voip telephony with asterisk" book from amazon, and they said i had to wait till may to get it. anyone know where i can get it ASAP |
| 01:31.58 | tzanger | www.mixdown.ca/~andrew/dump/rc.tc |
| 01:32.04 | tzanger | it works amazingly well |
| 01:32.07 | FuriousGeorge | woulndt be at a borders or barnes and noble would it? |
| 01:32.23 | shmaltz | anybody here got slackware running with 2.6 kernel? |
| 01:32.27 | tzanger | shmaltz: I do |
| 01:32.35 | terrapen | i'm trying to keep my costs down and i would rather avoid buying a full-retail Cisco 16 port switch |
| 01:32.44 | terrapen | surely there has to be something cheaper that can do decent QoS |
| 01:32.46 | shmaltz | tzanger, should I go this route with if I want HT? |
| 01:33.09 | shmaltz | or is 2.4 good enough? |
| 01:33.45 | tzanger | shmaltz: it works fine for me |
| 01:34.07 | shmaltz | tzanger, what? HT under 2.4? or 2.6 on slackware? |
| 01:34.21 | tzanger | 2.6 on slackware |
| 01:34.38 | fgravato | terrapen i think asus makes managed switch for under 300 bucks that does qos and some neat stuff |
| 01:34.41 | tzanger | I run both 2.4 and 2.6 on slackware withou no issues on either |
| 01:34.53 | terrapen | ok |
| 01:35.17 | fgravato | but go with crisco switch dude |
| 01:35.28 | fgravato | if your budget allows it |
| 01:35.35 | shmaltz | tzanger, you use HT on 2.4 and no issues? |
| 01:35.56 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 01:36.17 | FuriousGeorge | anyone know if it would be possible to get the book "voip telephony with asterisk" at a local borders or barnes&noble (i.e. not online) |
| 01:37.13 | tzanger | 2.4 doesn't know much about HT |
| 01:37.20 | tzanger | so it just runs pretty much UP |
| 01:38.23 | fgravato | FuriousGeroge dont bother with those books out |
| 01:38.25 | fgravato | better off |
| 01:38.26 | blitzrage | FuriousGeorge: you don't want it |
| 01:38.30 | fgravato | reading the Docs |
| 01:38.34 | fgravato | project Blitz |
| 01:38.37 | fgravato | maintains |
| 01:38.40 | blitzrage | FuriousGeorge: that book == wiki |
| 01:38.41 | fgravato | better documents |
| 01:38.49 | fgravato | hey leif |
| 01:38.54 | blitzrage | fgravato: yo |
| 01:39.24 | blitzrage | read the wiki, and use google to search the mailing lists. That's the documentation you get. |
| 01:39.26 | fgravato | the books out for asterisk are crap.. better off saving $$$ |
| 01:39.34 | jesster | what is a 'moderated conference bridge' ? |
| 01:39.36 | fgravato | and reading the Document project mailing list |
| 01:39.39 | fgravato | and wiki |
| 01:39.45 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 01:40.08 | odie_flocon | Hey all. |
| 01:41.41 | DEEZED | heh i was just about to buy that book off of amazon |
| 01:42.03 | DEEZED | ill stick to the wiki.. its nice |
| 01:42.05 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 01:42.52 | shmaltz | anybody here been testing chanspy? |
| 01:43.04 | shmaltz | it's been reopened, as bug 3686 |
| 01:43.29 | shmaltz | make that 3836 |
| 01:53.02 | anthm | make your paypal donations to anthmct@yahoo.com |
| 01:53.12 | fgravato | hey whats up anthony |
| 01:54.02 | elriah | Is there anyway to play a gsm file on the console? |
| 01:55.14 | TechDawg | Okay, good night all, thanks for the input fgravato. |
| 01:55.16 | *** part/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 01:59.15 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 01:59.16 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 01:59.22 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:59.50 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || Everyone thank anthm -> http://bugs.digium.com/bug_view_page.php?bug_id=0003836 |
| 02:00.31 | brc_ | s/thank/paypal |
| 02:00.33 | brc_ | ~anthm |
| 02:00.36 | terrapen | holy crap |
| 02:00.39 | brc_ | yeah! |
| 02:00.41 | brc_ | I know |
| 02:00.44 | terrapen | this ASUS smart switch is only $110 |
| 02:00.46 | terrapen | WOW |
| 02:00.52 | terrapen | that blows the pants off of Cisco |
| 02:01.07 | fgravato | hey brc :-) |
| 02:01.21 | fgravato | terrapen its pretty good switch can't beat the price |
| 02:01.26 | *** join/#asterisk IQ (~IQ@70-59-164-139.omah.qwest.net) |
| 02:01.27 | Rodms | Has anyone successfully configured QoS in linux for asterisk? |
| 02:01.32 | terrapen | i wonder if it does VLANs |
| 02:01.38 | terrapen | fgrav: awesome recommendation! |
| 02:01.45 | jesster | what is a 'moderated conference bridge' ? |
| 02:01.46 | terrapen | fgrav: check out neweggs prices |
| 02:01.59 | terrapen | hrmmm no mention of VLANs :( |
| 02:02.16 | fgravato | asus |
| 02:02.20 | fgravato | site mentions vlans |
| 02:02.26 | fgravato | http://usa.asus.com/products/networks/switch/x1024i/overview.htm |
| 02:03.50 | terrapen | well, thats a different switch |
| 02:04.11 | terrapen | i love their web-based config thing...that looks cool |
| 02:04.18 | fgravato | hrmm looking at the switch |
| 02:04.25 | PTG123 | anyone have a real land line that does caller id so i know what my cnam is? |
| 02:04.26 | fgravato | seems to look like the same ones |
| 02:04.29 | fgravato | dell sells |
| 02:04.35 | terrapen | yeah |
| 02:04.52 | *** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 02:05.03 | terrapen | fgravato: doesn't look like newegg carries the smart model with VLANs |
| 02:05.16 | pdracevich | easy why of getting a remote sip client to connect? |
| 02:07.38 | TomL | define "easy" |
| 02:07.53 | TomL | the documented way isn't easy enough? |
| 02:08.07 | pdracevich | I have been looking :( |
| 02:09.05 | pdracevich | can you possibly point me the right way. |
| 02:10.06 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 02:10.19 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 02:10.29 | terrapen | fgrav: this is strange |
| 02:10.37 | terrapen | the documentation says it does VLANs |
| 02:10.50 | terrapen | the GigaX1024P |
| 02:11.31 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 02:12.11 | TomL | who would bother building a layer 2 switch that doesn't do VLANs? what would be the point? :P |
| 02:12.52 | mogorman | anyone have experience with importvar? |
| 02:12.54 | fgravato | hrmm weird dude |
| 02:12.59 | RazaMetaL | hi |
| 02:13.18 | RazaMetaL | i´ve installed asterisk now .. |
| 02:13.43 | terrapen | if this switch did PoE, i would have an orgasm |
| 02:13.44 | RazaMetaL | on a Denbian with kernel Linux 2.6.8-2-686 |
| 02:13.48 | terrapen | but alas, it doesn't |
| 02:13.59 | RazaMetaL | Mar 22 21:11:17 WARNING[2098]: Unable to open '/dev/phone0' |
| 02:14.00 | RazaMetaL | Mar 22 21:11:17 ERROR[2098]: Unable to register channel '/dev/phone0' |
| 02:14.00 | RazaMetaL | Mar 22 21:11:17 WARNING[2098]: chan_phone.so: load_module failed, returning -1 |
| 02:14.00 | RazaMetaL | Mar 22 21:11:17 WARNING[2098]: Loading module chan_phone.so failed! |
| 02:14.24 | RazaMetaL | dmesg shows : |
| 02:14.25 | RazaMetaL | Linux telephony interface: v1.00 |
| 02:14.25 | RazaMetaL | ixj: found Internet PhoneJACK PCI at 0xde00 |
| 02:14.29 | Damin | ~seen kram |
| 02:14.33 | jbot | kram is currently on #asterisk. Has said a total of 7 messages. Is idling for 7h 23m 18s |
| 02:14.42 | RazaMetaL | lsmod | grep ixj |
| 02:14.42 | RazaMetaL | ixj 432068 0 |
| 02:14.42 | RazaMetaL | phonedev 5344 1 ixj |
| 02:15.05 | RazaMetaL | any idea ? |
| 02:15.25 | *** join/#asterisk rjburkh (~chatzilla@dialup-4.233.118.150.Dial1.LosAngeles1.Level3.net) |
| 02:15.26 | RazaMetaL | seems like asterisk can´t load the module chan_phone.so |
| 02:16.17 | RazaMetaL | Mar 22 21:16:05 WARNING[2154]: Unable to open '/dev/phone0' |
| 02:16.28 | RazaMetaL | and.. /dev/phone0 does not exists .. |
| 02:17.23 | rjburkh | hello everyone |
| 02:17.52 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 02:19.49 | *** join/#asterisk _daver_ (~daver@ns1.tmok.com) |
| 02:21.10 | elriah | How well does the asterisk fax integration work? |
| 02:21.49 | syslod | fax intergration? RECV works OK. Sending not so OK. |
| 02:22.18 | syslod | Is there a var for sip peer????? |
| 02:22.29 | elriah | I just want to receive faxes and them email them ast tiff documents, nothing more. |
| 02:22.39 | *** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 02:22.42 | syslod | That works good in *. Check out spandsp |
| 02:22.48 | elriah | Thanks - |
| 02:22.55 | syslod | np |
| 02:22.56 | bjohnson | elriah: and also use the telco line for voice? |
| 02:23.10 | elriah | Yes. |
| 02:23.12 | syslod | faxdetect |
| 02:25.14 | FuriousGeorge | if im setting up an asterisk system using only softphone and internet voip dialtone providers, what exactly do i have to do to zaptel.conf. the docs dont say, but they do say to use ztdummy, ehich ive compiled and loaded |
| 02:25.35 | FuriousGeorge | p.s. i have no digium hardware |
| 02:25.37 | elriah | How reliable is faxdetect? |
| 02:25.45 | FuriousGeorge | or any voip sepcific hardware |
| 02:25.55 | *** join/#asterisk bparker (bparker@cable-71-8-65-183.mtv.al.charter.com) |
| 02:26.57 | elriah | One last question - is there a way for me to play a sound from the /sounds dir from the asterisk console? |
| 02:28.18 | Chuji | ~ztdummy |
| 02:28.19 | jbot | ztdummy is probably zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs) |
| 02:28.27 | FuriousGeorge | is the answer to my ?: "nothing" |
| 02:28.29 | Chuji | FuriousGeorge : Only need it if you are doing meetme |
| 02:28.59 | FuriousGeorge | Chuji: doest that ztdummy use meetme or something. i am very new to this |
| 02:28.59 | Chuji | FuriousGeorge : Otherwise, no, you don't need zap at all |
| 02:29.10 | jakepdev | is libpri 1.0.0 the newest? |
| 02:29.12 | FuriousGeorge | for the timer |
| 02:29.14 | Chuji | ztdummy provides timing for meetme |
| 02:29.38 | FuriousGeorge | and meetme is? conferencing? |
| 02:29.57 | mstocco | jakepdev: no |
| 02:30.27 | Chuji | Yes sir |
| 02:30.49 | jakepdev | I keep on getting an error on * install "you need newer libpri" |
| 02:30.53 | FuriousGeorge | chuji: muchisimas gracias |
| 02:31.47 | RazaMetaL | FuriousGeorge: de donde eres ? yo soy de Ecuador |
| 02:31.51 | jakepdev | mstocco - 1.06 is the latest stable? |
| 02:31.57 | syslod | elriah: Its seems to work well. |
| 02:32.03 | blitzrage | wow... who wrote that answer for ztdummy, because it's wrong |
| 02:32.06 | mstocco | jakepdev: 1.0.6 is the latest |
| 02:32.15 | jakepdev | mstocco - tnx |
| 02:32.17 | blitzrage | it uses usb-uhci, no usb-ohci (and that's only for 2.4.x kernels) |
| 02:32.19 | FuriousGeorge | razametal: mis padres son espanoles pero yo vivio in NJ EE.UU. |
| 02:33.55 | shmaltz | ~nj |
| 02:33.56 | jbot | nj is, like, home to the Sopranos |
| 02:34.15 | syslod | Is there a sip peer var? Or other way of conditionally setting caller id depending on sip account? |
| 02:34.48 | debaser | how dare you cry, angel gone. |
| 02:34.55 | *** join/#asterisk Elshar (~Elshar@ip205-68.oregonfast.net) |
| 02:34.55 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 02:34.57 | bjohnson | syslod: yes |
| 02:35.02 | Chuji | blitzrage : It's probably pretty old |
| 02:35.05 | bjohnson | syslod: sip.conf settings |
| 02:35.08 | debaser | cheat and lie, cheat and lie. angel gone. |
| 02:35.16 | bjohnson | syslod: or set it just before a dial |
| 02:35.35 | elriah | Hey all - where would I set the message waiting tone? A temporary fix until I get my MWI lights working on my polycom's. |
| 02:35.45 | *** join/#asterisk Rival (~rival@66.177.249.219) |
| 02:36.25 | syslod | bjohnson: I am trying to emulate local id and outbound id. sip.conf allows one of the setting. So 109 John Holmes is local to the system and 8882332333 Big Johns callerid gets put depending on where they are dialing. |
| 02:37.26 | FuriousGeorge | ~nj |
| 02:37.27 | jbot | nj is, like, home to the Sopranos |
| 02:37.30 | FuriousGeorge | ~ny |
| 02:37.31 | jbot | ny is probably a place where they make the best pizza, the best hot dogs, and the nicest hookers |
| 02:39.01 | syslod | ~ga |
| 02:39.02 | jbot | rumour has it, ga is "go ahead" |
| 02:39.04 | Sedorox | ~pa |
| 02:39.05 | jbot | i heard pa is PAY ATTENTION!!!!! |
| 02:39.08 | syslod | ~va |
| 02:39.09 | jbot | rumour has it, va is one of Debian's servers which has crashed lately (if you're in search for Branden Robinson's "X strike force" pages, try http://samosa.debian.org/~branden) VA Linux Systems. |
| 02:39.27 | FuriousGeorge | lol, its fun to abuse the jbot |
| 02:39.35 | FuriousGeorge | ~newark |
| 02:39.37 | elriah | The message waiting stutter tone, is it automatic? |
| 02:40.01 | FuriousGeorge | <jbot> newark is home of crack |
| 02:40.08 | syslod | Crap... Anyone else doing internal callerpres and different outbound? |
| 02:40.19 | bjohnson | syslod: I'd set it manually before the dial out |
| 02:40.52 | syslod | bjohnson: I'd like to do that but I need to be able to tell which sip account so I know what Callerid to use. |
| 02:42.45 | bjohnson | maybe set callerid in sip.conf and then check callerid before resetting for dial out and change it based on the current value? |
| 02:43.08 | bjohnson | sounds like you'll want a db lookup or you ould have a pile of gotoifs |
| 02:43.57 | FuriousGeorge | `ak |
| 02:43.59 | bjohnson | that's an odd request though |
| 02:44.01 | syslod | I need to set a different callerid for each sip. It sucks that it doens't work like a mitel or panasonic. They would kinda let you set callerid for internal and allow you to select id on outbound. |
| 02:44.04 | FuriousGeorge | ~ak |
| 02:44.23 | shmaltz | ~ny |
| 02:44.24 | jbot | somebody said ny was a place where they make the best pizza, the best hot dogs, and the nicest hookers |
| 02:44.33 | syslod | Not odd at all we are competing against systems we installed years ago and that would a real feature most people can't do without now. |
| 02:45.15 | bjohnson | odd in that I have never heard of it and can't perceive a ractical use of it .. but I'm mostly used to smaller systems |
| 02:45.36 | FuriousGeorge | ~nh |
| 02:45.37 | jbot | rumour has it, nh is neo-hippie, ask me about it |
| 02:45.45 | syslod | bjohnson: When a inside call is made is it right to display the PSTN callerid? I would suspect most ppl would want there directory name and extension. |
| 02:45.57 | bjohnson | ahhh .. new york .. home of those nice hookers |
| 02:46.06 | bjohnson | other places they are just so mean |
| 02:46.18 | Qwell | nothing beats a friendly hooker |
| 02:46.37 | bjohnson | syslod: that is normal and is the default for asterisk |
| 02:46.51 | bjohnson | syslod: you just set it in sip.conf |
| 02:47.03 | bjohnson | or in the phone config if it has that option |
| 02:47.06 | syslod | bjohnson: Just as a question how are you handling it on the smaller systems? Callerid in SIP is Job Bob 109? and you globally define outbound as PSTN? |
| 02:47.14 | bjohnson | yes |
| 02:47.28 | bjohnson | one callerid for all outbound |
| 02:47.56 | bjohnson | which is the primary pstn number |
| 02:48.03 | syslod | Thats not real typical in our world. We have all PRI customers. They are use to having internal caller id and outbound caller id for sales, tech, etc all being different of course. |
| 02:48.13 | elriah | How would I play a gsm file from my linux console? What app? I'm running debian sarge. |
| 02:48.15 | syslod | U using digital service? |
| 02:48.24 | YoYo | app_dial.c:578 wait_for_answer: Unable to forward voice |
| 02:48.33 | YoYo | 30 seconds later, it works fine |
| 02:48.37 | syslod | elriah: Freshmeat has lots of players. |
| 02:48.44 | bjohnson | syslod: I've never encountered that. Pretty much all places have a central pstn number and you go through the ivr for tech, sales, etc |
| 02:49.01 | syslod | bjohnson: Are they digital or analog? |
| 02:49.04 | bjohnson | elriah: soxplay is popular |
| 02:49.12 | bjohnson | analog you can't set |
| 02:49.24 | bjohnson | I set it for digital voip outgoing calls |
| 02:49.40 | elriah | tnx |
| 02:50.19 | syslod | Our guys are used to building groups. They are looking to cut IVR times. |
| 02:51.31 | IQ | Hi, anyone got NuFone configuration info? |
| 02:51.37 | syslod | The mitel/panas/sylantros have some nice CLIP features. |
| 02:51.41 | bjohnson | isn't it on their website? |
| 02:51.47 | syslod | IQ: It should be on there site. |
| 02:51.59 | syslod | Worked first time for me. |
| 02:52.02 | IQ | already on the website - still looking for it |
| 02:52.09 | syslod | Did you sign up? |
| 02:52.17 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
| 02:52.19 | Rival | sucks i can get mine able to make outgoing calls but cant recieve calls rings busy |
| 02:52.33 | IQ | yeah, I did - and it was working - I'm setting it up at home now |
| 02:52.36 | Rival | anyone have info on setting up for teliax |
| 02:53.00 | bjohnson | http://www.voip-info.org/wiki-Asterisk+settings+nufone |
| 02:53.05 | bjohnson | that might have it |
| 02:53.16 | IQ | thanks :) |
| 02:53.20 | Qwell | Does anybody know of something that can take a phone number, and tell you what (if anything) it says in words? heh |
| 02:53.22 | bjohnson | IQ: well crap .. copy it from where it is working |
| 02:53.42 | bjohnson | Qwell: 3 options |
| 02:53.48 | IQ | bjohnson: can't log on to my VPN :( something wrong with keyfob |
| 02:53.49 | bjohnson | Qwell: festival |
| 02:53.55 | syslod | Qwell: I've seen that before on USLECs site. Vanity numbers in google should find something. |
| 02:53.57 | Qwell | bjohnson: no...I mean... |
| 02:54.03 | Qwell | what he said |
| 02:54.06 | bjohnson | Qwell: a bunch of gotoif that play individual files |
| 02:54.17 | YoYo | Qwell: like phonespell.org ? |
| 02:54.19 | elriah | swift is way better than festival. $29. |
| 02:54.24 | Qwell | YoYo: sounds like it, lemme look |
| 02:54.47 | Qwell | YoYo: yep, thanks |
| 02:55.33 | Qwell | bjohnson: Thanks though |
| 02:57.06 | IQ | phonespell.org is interesting - just found out my work number is 544-TRIP :) |
| 02:57.16 | Rival | anyone here using teliax? |
| 02:57.29 | Qwell | IQ: heh |
| 02:57.36 | Qwell | IQ: my old phone number was mypiggy ;/ |
| 02:57.48 | IQ | Qwell: loool |
| 03:01.41 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfllf.dialup.mindspring.com) |
| 03:05.25 | FuriousGeorge | will there be any appreciable voice quality difference b/w a synchronos and an asynchronos broadband connection? |
| 03:05.50 | *** join/#asterisk mwcnetwork (~mwcnetwor@user-0c93oob.cable.mindspring.com) |
| 03:06.03 | mwcnetwork | hello |
| 03:06.07 | FuriousGeorge | dsl specifically. assuming a max of 800mb down and 200 up |
| 03:07.03 | mwcnetwork | I downloaded Asterisk with the help of someone I met on #Asterisk. It seems to have installed correctly but all I get is a busy signal. |
| 03:07.13 | mwcnetwork | How do I dial my cell phone from my computer? |
| 03:07.35 | TomL | what's your trunking? |
| 03:07.43 | TomL | you have an IP trunk or PSTN? |
| 03:07.44 | mwcnetwork | I really do not know. |
| 03:07.48 | *** join/#asterisk jhowardPA (~jhoward@12.25.177.120) |
| 03:07.58 | TomL | well, you have to connect asterisk to the phone network somehow |
| 03:07.59 | mwcnetwork | I am on a cable modem |
| 03:08.01 | TomL | it ain't magic |
| 03:08.03 | FuriousGeorge | im asking b/c for digital telephony you need duplex sound for it to sound natural. but i dont know if that would translate |
| 03:08.12 | FuriousGeorge | to adsl vs sdsl |
| 03:08.30 | mwcnetwork | I have a LAN setup after the cable modem |
| 03:08.39 | mwcnetwork | I have Vonage setup above the dlink router |
| 03:08.44 | jhowardPA | Hello folks. I'm curious as to whether or not Asterisk would be up to the needs of my company. Anyone care to help me figure it out? |
| 03:08.52 | TomL | you have to configure asterisk to use your vonage account |
| 03:09.04 | FuriousGeorge | mwcnetwork: i thouht vonage wouldnt work w/ asterisk. |
| 03:09.36 | fgravato | furious somepeople have had it working |
| 03:09.41 | mwcnetwork | So how do I configure asterisk for vonage? |
| 03:09.42 | fgravato | i did this past summer using |
| 03:09.47 | fgravato | vonage softphone option |
| 03:09.52 | FuriousGeorge | for starters, you could circumvent their "softphone plan" pretty easily |
| 03:09.53 | fgravato | and bit of debuging |
| 03:09.56 | mikegrb | jit will |
| 03:10.04 | jhowardPA | I've got some 100 extensions internally, and 2 T1's feeding them. |
| 03:10.04 | mikegrb | jhowardPA: it will |
| 03:10.38 | jhowardPA | Our current PBX is a crappy old Panasonic switch. How many concurrent connections can one Asterisk box handle? |
| 03:10.40 | FuriousGeorge | fgravato: really, you think if u didnt have the softphone plan u could still do it? |
| 03:10.47 | mwcnetwork | That sounds like what I want to do- and to answer the question above- yes, from what I have read I think Asterisk will be great for small and medium business. |
| 03:10.50 | *** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218) |
| 03:10.57 | FuriousGeorge | could solve some peoples local number portability problems |
| 03:11.13 | jhowardPA | We've also got some 300 remote locations I'd be interested in supporting... |
| 03:11.14 | IQ | jhowardPA: how many lines do u have on Panasonic? |
| 03:11.19 | mwcnetwork | How do I make Asterisk treat Vonage like the trunk? |
| 03:11.55 | Sedorox | asterisk will work with vonage huh? |
| 03:11.56 | Sedorox | hmmmm |
| 03:12.09 | mwcnetwork | jhoward- I am learning Asterisk so I can offer it as a network support service for my clients. |
| 03:12.10 | jhowardPA | IQ: Well, there's the hundred extns internally, and the 2 T1's make 46 external. |
| 03:12.38 | fgravato | actually if you used digium x100p card and ata186 could still have vonage |
| 03:12.39 | IQ | jhowardPA: by 300 remote locations you mean 300 extensions or 300 locations having multiple extensions? |
| 03:12.44 | mwcnetwork | I have two router/firewalls above my asterisk box. |
| 03:12.52 | fgravato | if you don't use softphone |
| 03:13.11 | jhowardPA | IQ: 300 stores across the US, all of which would be nice to have setup with a VoIP softphone or two. |
| 03:13.51 | mwcnetwork | Does someone know of a good article about Vonage and Asterisk? |
| 03:14.00 | IQ | jhowardPA: connected to one central location? or having their own servers? |
| 03:14.05 | jhowardPA | I'm trying to push this with my bosses, because they don't want to spend much on a new phone system.. they'd be happy to spend a bit on a setup/support contract. |
| 03:14.44 | jhowardPA | IQ: The former, via VPN - but not many would be in use at once. I'd plan on taking maybe 2-3 calls via softphone per hour. |
| 03:15.07 | mwcnetwork | jhoward- Asterisk is new enough- don't let people fool you- this is sort of bleeding edge stuff. |
| 03:15.27 | IQ | mwcnetwork: really :O ? |
| 03:15.30 | jhowardPA | Primarily, I'd want to support our 2 T1's for voice, and internal (100 in-office extensions) calling. |
| 03:15.38 | mwcnetwork | Still it has great potential- and I would ask questions about the number of man hours installing and managing asterisk |
| 03:15.54 | *** join/#asterisk Legend (~Legend@24.244.142.133) |
| 03:16.23 | IQ | mwcnetwork: we dont do it number of man hours... we count it in number of man seconds. it doesnt take hours to set up asterisk. that is if you know what you are doing |
| 03:16.30 | jhowardPA | mwcnetwork: You've got a point. And I am well aware of it. I'd like to be able to demo it, or use it for one small department before I even consider using it in a larger context. |
| 03:16.50 | IQ | mwcnetwork: and no one is elling anything here. why would people fool someone ? |
| 03:17.09 | IQ | mwcnetwork: no one is selling anything |
| 03:17.17 | TrevorSHarrison | here's a stupid question... how do I get * to load / enable my Zap channels? I've got my zapata.conf setup with a few channels, but when I dial(Zap/1/blah) it says its unable to create a channel of type 'zap' |
| 03:17.17 | mwcnetwork | I am just using figures of speech- this forum is for help with understanding Asterisk |
| 03:17.27 | *** join/#asterisk alakdan (~alakdan@210.213.196.113) |
| 03:17.33 | mwcnetwork | But I am a consultant who will support Asterisk given time. |
| 03:18.08 | jhowardPA | IQ: I'm not mening to come off as anxious, I'm evaluating my options. Asterisk looks sexy, and I'm hoping to get some ideas on where to start my research. |
| 03:18.10 | RazaMetaL | (i) |
| 03:18.11 | IQ | jhowardPA: get a regular machine with 2 or 4 GB ram and give it a try - dont invest too much if you just want to see demo |
| 03:18.13 | jhowardPA | "meaning" |
| 03:18.25 | RazaMetaL | mknod /dev/phone0 c 100 0 |
| 03:18.36 | RazaMetaL | now */ is running .. :) |
| 03:18.42 | jhowardPA | IQ: Which analog hardware is recommended? |
| 03:18.42 | mwcnetwork | jhoward- last time I was on this channel a very helpful man from Pakistan talked me through an Asterisk Install. |
| 03:19.15 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 03:19.16 | mwcnetwork | Why I am back here is all I get when I dial is a busy signal |
| 03:19.21 | IQ | jhowardPA: http://www.digium.com/ |
| 03:19.28 | IQ | mwcnetwork: pakistanis are alwas good ;) |
| 03:19.33 | mwcnetwork | I would like to dial my cell phone or some other phone |
| 03:19.37 | jhowardPA | mwcnetwork: I'm sure I can figure the install/setup out, possibly with help from you fine folks, but I don't know what kind of capacity a server can handle. |
| 03:19.48 | jhowardPA | IQ: 2-4GB, for how many lines? |
| 03:20.32 | jhowardPA | IQ: saw the digium.com site, I'll look around there more (though I think I've scoured it pretty well). Do they have usage specs on there? |
| 03:20.33 | bjohnson | jhowardPA: I think current max count is at about 1500 concurrent calls on one box |
| 03:20.35 | mwcnetwork | jhoward- I recommend any PIII decent system- and I also recommend that it be used solely for asterisk so you can troubleshoot the network effectivel |
| 03:20.48 | jhowardPA | bjohnson: what kind of box? |
| 03:20.55 | mwcnetwork | keep it simple |
| 03:21.01 | IQ | jhowardPA: they got everything there - if you can't find something do google |
| 03:21.13 | bjohnson | there was a site .. and I can't remember the name |
| 03:21.15 | *** join/#asterisk khaladan (~gnewf@GroupMackenzie.s11-1-0-16-0.ar3.SEA1.gblx.net) |
| 03:21.21 | bjohnson | asterisksomething |
| 03:21.23 | mwcnetwork | I found some good 1u deals on ebay today sub $600 |
| 03:21.34 | IQ | jhowardPA: and then you will say * is not reliable :) |
| 03:21.37 | bjohnson | jhowardPA: probably more capacity than you need anyway |
| 03:21.49 | jhowardPA | IQ: cool, thanks. I figured someone here would have some real-life experience with it. :\ |
| 03:21.50 | bjohnson | the wiki has some examples of more normal systems |
| 03:22.15 | alakdan | anyone from hawaii? |
| 03:22.31 | khaladan | question: we have two offices that both have a seperate pbx (panasonic i think?); we wanna allow employees in one office to call the other office using normal extensions (and not over the PSTN); we have a point-to-point T1 line between the two offices; what would be the best way to make something like this happen? |
| 03:22.31 | jhowardPA | bjohnson: I've got an E3500 sitting around I could use for it, but I'd rather use a cheap P4 or something. ;) |
| 03:22.35 | IQ | jhowardPA: My home machine has 2GB - if you want to evaluate something then give it a fair shot... 2GB to 4GB is not too much |
| 03:22.42 | bjohnson | http://www.voip-info.org/wiki-Asterisk+hardware+recommendations |
| 03:23.14 | jhowardPA | IQ: I'm not saying it's too much. I've got plenty of ram. I'm curious about some real-life examples. |
| 03:23.18 | mwcnetwork | Has anyone set up vonage as the source for asterisk |
| 03:23.26 | bjohnson | khaladan: put * in front of the pbx or on an extension (at both offices) |
| 03:23.31 | IQ | jhowardPA: sorry - no examples :( |
| 03:23.46 | jhowardPA | Just because I can setup the box doesn't mean I want to figure out how to route a hundred test-calls through it simultaneously. ;) |
| 03:23.49 | IQ | jhowardPA: atleast I dont know of any benchmark |
| 03:23.56 | khaladan | bjohnson: what kind of hardware would i need for that? (anything?) |
| 03:24.32 | mwcnetwork | I would settle for one call right now... :) |
| 03:24.53 | bjohnson | http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 03:25.01 | bjohnson | that's what I was thinking of |
| 03:25.04 | jhowardPA | IQ: I'll just have to rig it up and see... and maybe call the Digium guys for some advice. Thanks for the info, though. |
| 03:25.07 | IQ | try asterisk@home or asteriskwin32 |
| 03:25.27 | IQ | jhowardPA: ya call digium, they're very helpful |
| 03:25.34 | jhowardPA | bjhonson: Awesome, thanks! |
| 03:25.45 | jhowardPA | IQ: Thanks, I shall. |
| 03:26.01 | TrevorSHarrison | mwcnetwork: yep, using vonage |
| 03:26.10 | bjohnson | http://www.astertest.com/ |
| 03:26.13 | mwcnetwork | trevor- help! |
| 03:26.14 | bjohnson | that one too |
| 03:26.18 | jhowardPA | Take it easy folks, thanks for the help! |
| 03:26.21 | *** part/#asterisk jhowardPA (~jhoward@12.25.177.120) |
| 03:26.41 | Sedorox | 0_o |
| 03:26.45 | bjohnson | khaladan: if only doing a couple of calls at a time .. an xbox would be enought power |
| 03:26.46 | nestAr | hrmph |
| 03:26.49 | TrevorSHarrison | mwcnetwork: how far have you gotten? |
| 03:26.53 | alakdan | this is wierd, a friend tried calling a toll free number from hawaii (we are currently subscribed to nufone) but can not get through, a friend tried calling from new york and its ok. Got any ideas? |
| 03:26.57 | nestAr | openclose.agi doesn't seem to actually be setting the variable |
| 03:27.14 | mwcnetwork | trevor - I am installed- softwarewize on a linux Fedora Core 2 box |
| 03:27.17 | khaladan | bjohnson, well--besides the computer. i got that taken care of. do i need anything like an ATA? or some other kind of converter? |
| 03:27.29 | bjohnson | khaladan: and a fxo if connecting to and exsting ATA port or a fxs if connecting to an analog line in port or a T1 if connecting to the pbx that way |
| 03:27.42 | TrevorSHarrison | have you got calls between your local extensions working already? |
| 03:27.51 | bjohnson | khaladan: you have to figure out what your pbx will allow |
| 03:27.53 | mwcnetwork | Trevor? ?? |
| 03:27.58 | IQ | Anyone using NuFune with X-Lite ? |
| 03:28.03 | TrevorSHarrison | mwcnetwork: yes? |
| 03:28.03 | mwcnetwork | All I get is a busy signal |
| 03:28.05 | khaladan | it's an older pbx |
| 03:28.37 | mwcnetwork | If I use "dial 1000" or "dial 2000" I get a busy signal |
| 03:28.46 | bjohnson | alakdan: is not available in hawaii? |
| 03:28.52 | TrevorSHarrison | mwcnetwork: work on calling between 2 local xlite extensions. that way you can eliminate vonage from the picture. |
| 03:28.59 | mwcnetwork | I was hoping to use "dial npa - nxx" |
| 03:29.15 | alakdan | bjohnson: yeah sort of |
| 03:29.18 | mwcnetwork | sounds good but I have not got a clue |
| 03:29.29 | khaladan | so, if i set up the panasonic pbx to send some extensions down a phone wire -> ATA -> FXO -> T1 -> Asterisk -> ? |
| 03:30.01 | bjohnson | khaladan: I don't know your pbx .. you'll have to figure it out |
| 03:30.07 | khaladan | ok, sorry |
| 03:30.21 | bjohnson | khaladan: if you can get it to * you can go over the internet to the other asterisk |
| 03:30.28 | khaladan | ok |
| 03:30.29 | alakdan | bjohnson: cant call the number from hawaii. |
| 03:30.30 | TrevorSHarrison | mwcnetwork: well, at least now you don't have to worry about vonage... just look at the docs on how to add an softphone (ie. the xlite), and get it to work where you can here the demo voice stuff. |
| 03:30.35 | *** join/#asterisk w0w0 (~w0w0@80.26.162.27) |
| 03:30.45 | bjohnson | alakdan: is likely a US/48 toll free number then |
| 03:30.48 | khaladan | bjohnson, i'm guessing I'll need an fxs... |
| 03:31.08 | bjohnson | khaladan: depending on how you can connect to the pbx |
| 03:31.09 | mwcnetwork | I did hear some demo things coming out of asterisk- |
| 03:31.20 | alakdan | bjohnson: 1 877 xxx xxxx number |
| 03:31.30 | bjohnson | khaladan: if the extensions can use regluar analog phones .. then a fxo can plug into one |
| 03:31.44 | khaladan | yea the phones we have now are analog ones |
| 03:31.58 | mwcnetwork | I would like asterisk to serve as my own private pbx at home |
| 03:32.07 | TrevorSHarrison | mwcnetwork: basic question re: vonage. do you have a softphone account with them, or just the normal basic hard phone? |
| 03:32.08 | alakdan | bjohnson: what does it mean US/48? and if its a US/48 number it can not be accessed from hawaii? |
| 03:32.11 | khaladan | so, our pbx -> fxo -> asterisk |
| 03:32.21 | bjohnson | khaladan: if you have an extra extension a fxo can plug directly into it |
| 03:32.34 | bjohnson | khaladan: yep .. and the reverse at the other end |
| 03:32.40 | khaladan | i think we have plenty of extra extensions |
| 03:32.50 | nestAr | anyone using openclose.agi ? |
| 03:32.52 | khaladan | thanks so much |
| 03:32.53 | mwcnetwork | I heard I did not need one since I was going to do it through asterisk- I have the 24.99 unlimited plan |
| 03:33.09 | bjohnson | khaladan: hook up as many as you want concurrent calls |
| 03:33.29 | TrevorSHarrison | yeah... you can't with that. vonage won't let anything but their hardware adapter connect to that plan. |
| 03:33.29 | khaladan | thanks!! |
| 03:33.31 | bjohnson | khaladan: your internet connection might limit the max number of those too |
| 03:33.41 | khaladan | probably 4 concurrent calls |
| 03:33.55 | mwcnetwork | What do I need to do? |
| 03:33.57 | khaladan | we have a direct t1 between offices |
| 03:33.59 | bjohnson | a dsl line should handle that |
| 03:34.05 | TrevorSHarrison | mwcnetwork: you can get an additional softphone line for $10'ish that will allow you to connect with any sip device |
| 03:34.30 | bjohnson | khaladan: you doing a lot of data transfer? |
| 03:34.35 | TrevorSHarrison | but the crappy thing is that it has its own minute limit... ie. its not unlimited like your main line... it also has its own phone number. |
| 03:34.44 | bjohnson | vonage blows |
| 03:34.47 | mwcnetwork | There *must* be a way around it! |
| 03:34.49 | khaladan | bjohnson: some data transfer, anyway |
| 03:34.57 | *** join/#asterisk TSWoodV (~woodt@216.230.39.168) |
| 03:35.07 | khaladan | bjohnson: but 4 concurrently is sufficient in any case |
| 03:35.11 | bjohnson | khaladan: a t1 seems expensive |
| 03:35.33 | TrevorSHarrison | mwcnetwork: dunno... its just not worth it to me... when I get this all setup, I'll probably switch over to broadvoice or some other sip provider thats not such a butt-head. |
| 03:35.36 | bjohnson | mwcnetwork: tell vonage to piss off .. you're going to a real voip provider |
| 03:35.47 | khaladan | bjohnson, suggest dsl instead? not my decision in any case :) |
| 03:35.49 | TrevorSHarrison | mwcnetwork: I'm only using our existing vonage line because we already had them |
| 03:36.21 | bjohnson | khaladan: well .. if not much data transfer and only 4 concurrent calls .. why have a t1 |
| 03:36.47 | mwcnetwork | So vonage is trouble? The cancel is only $36 so when I am ready it will not be a problem |
| 03:36.52 | khaladan | bjohnson, i think maybe it was so that it could be a direct connection between the offices, for less latency(?)... not sure if dsl can offer that |
| 03:36.55 | bjohnson | khaladan: what internet connection is there? |
| 03:37.15 | TrevorSHarrison | mwcnetwork: trouble for asterisk, y. I use it at home where I've just got a plain phone and its okay. |
| 03:37.18 | bjohnson | less latency for what? |
| 03:37.20 | khaladan | bjohnson, the internet connection is another T1. we have four T1s in total.. |
| 03:37.46 | bjohnson | khaladan: geez .. sounds like you got way more bandwidth than you use |
| 03:37.51 | khaladan | probably |
| 03:37.52 | TrevorSHarrison | mwcnetwork: but broadvoice's byod (bring your own device) plan is much better as far as signup fees go |
| 03:37.58 | khaladan | well sometimes we do transfer large files |
| 03:38.06 | khaladan | we store all the files in this office |
| 03:38.37 | mwcnetwork | I saw lingo had a byod plan as well.. |
| 03:38.38 | khaladan | two T1s for connections to other offices, a T1 for voice only, and another for i don't-remember-what |
| 03:38.57 | mwcnetwork | T1 can be frac-T voice and data |
| 03:39.02 | bjohnson | khaladan: your voip connection of 4 calls should be able to use the internet t1. the direct t1 is not needed for voip .. would maybe increase reliability |
| 03:39.20 | bjohnson | but wouldn't have much effect if using same isp at both ends anyway |
| 03:39.24 | mwcnetwork | QoS is delivered over Pri/T1 |
| 03:39.49 | khaladan | bjohnson: well, i don't know all the stats on utilization right now. our offices are growing.. maybe ppl are just being optimistic |
| 03:40.03 | mwcnetwork | So to be the maverick then- how do I use the softphone in Asterisk? |
| 03:40.09 | bjohnson | likely costing $500 a month |
| 03:40.32 | mwcnetwork | Rocket out of Los Angeles has Wireless T1 for $99 |
| 03:40.46 | *** join/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 03:40.59 | BuckRogers | bjohnson good night see you in the morning |
| 03:41.05 | bjohnson | bye |
| 03:41.19 | DEEZED | is there any good asterisk forums? |
| 03:41.27 | BuckRogers | o one thing |
| 03:41.38 | BuckRogers | we got the te405p working noprob today |
| 03:41.41 | opus_ | is there a model better then the SPA-3000? |
| 03:41.55 | opus_ | whats the brand of choice :) |
| 03:42.17 | BuckRogers | bjohnson thanks for your help with it, it was a prob in the zapconf |
| 03:42.29 | BuckRogers | any how good night |
| 03:44.31 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 03:45.11 | bjohnson | opus_: for what use? |
| 03:45.21 | khaladan | bjohnson, so for my purposes I would have FXOs at both ends, right, and not FXO -> asterisk -> t1 -> asterisk -> FXS |
| 03:45.30 | bjohnson | the spa 3000 is a good choice if looking for one fxo and one fxs port |
| 03:46.12 | bjohnson | khaladan: yes .. fxo at both ends connecting to your analog extension ports on your pbx |
| 03:46.37 | khaladan | what's a good 4-port FXO? Wildcard TDM400P w/ 4 FXO modules? |
| 03:46.58 | bjohnson | khaladan: you can buy a couple of cheap x100p pci cards off ebay for about $20 each to do proof of concept but you will want the digium cards for production use |
| 03:47.06 | khaladan | ok |
| 03:47.08 | Beirdo | question... |
| 03:47.20 | mwcnetwork | yes bierdo? |
| 03:47.23 | Beirdo | is the order you put the "allow=" line in important? |
| 03:47.31 | bjohnson | some people say yes |
| 03:47.32 | Beirdo | i.e. does it set the preference? |
| 03:47.35 | bjohnson | some people say no |
| 03:47.50 | bjohnson | I haven't looked at the code to find out for sure |
| 03:47.54 | Beirdo | heh |
| 03:48.06 | stdio | khaladan: we like the tdm400p.. just got it and have 3 fxo modules in it |
| 03:48.06 | bjohnson | it is rumoured to be preset in the code |
| 03:48.23 | mwcnetwork | Ok, I started asterisk so I can run and configure while talking in the forum |
| 03:48.27 | khaladan | stdio, good to know |
| 03:48.37 | bjohnson | I better go study for my exams tomorrow |
| 03:48.50 | bjohnson | bye |
| 03:49.00 | stdio | khaladan: plus, there is *lots* of documentation out there for it... |
| 03:49.28 | mwcnetwork | I noticed the xlite in the config files |
| 03:49.33 | DEEZED | stdio... any good asterisk forums? or is voip-info.com the best stash of info |
| 03:49.43 | mwcnetwork | How do I configure it? |
| 03:50.07 | Beirdo | hmm |
| 03:50.13 | Beirdo | it does seem to matter |
| 03:50.36 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 03:50.57 | Beirdo | I put g726 first, it's using that instead of ulaw which used to be first |
| 03:51.22 | Sedorox | Beirdo: any luck yet? |
| 03:51.34 | bjohnson | Beirdo: try reversing that order |
| 03:51.45 | Beirdo | Sedorox: no |
| 03:51.49 | nestAr | anyone have a working openclose.agi ? |
| 03:51.52 | Beirdo | I sent them bitchmail agian |
| 03:52.06 | nestAr | mine doesn't seem to be setting a status |
| 03:52.09 | Sedorox | if I don't hear back by thursday. I'm gonna start to get piccy... |
| 03:52.10 | Sedorox | pissy* |
| 03:52.15 | Sedorox | is it isn't working by monday |
| 03:52.16 | stdio | DEEZED: i'd imagine the asterisk-users mailing list would be one of the best... not exactly a forum, but you're likely to get just as quick of an answer... |
| 03:52.17 | Sedorox | Buh Bye |
| 03:52.25 | Beirdo | bjohnson I think the order you define them in the allow= is the order they get used in (preference) |
| 03:52.29 | Sedorox | 'cause it'll be over a month at that point |
| 03:52.41 | stdio | DEEZED: http://lists.digium.com/mailman/listinfo/asterisk-users |
| 03:52.42 | DEEZED | ok cool |
| 03:52.50 | DEEZED | thx |
| 03:52.54 | stdio | np |
| 03:53.00 | *** part/#asterisk ctooley (~ctooley@rrcs-24-153-228-2.sw.biz.rr.com) |
| 03:54.31 | Half_Dome | newbie needs some direction. Can I ask a beginner's question? |
| 03:54.53 | nestAr | don't ask to ask |
| 03:54.55 | nestAr | just ask |
| 03:55.01 | Half_Dome | I have 4 POTS coming into my home office. I'm setting up a remote office and would like to use these same lines in it. Am I looking at two Asterisk servers and one FXO and one FXS PCI card? |
| 03:55.03 | khaladan | bjohnson, will i have to have Asterisk configured to be able to talk to my specific PBX or will the FXO convert the data over the line to some standard protocol? |
| 03:55.10 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:56.09 | stdio | Half_Dome: you could vpn the two with freeswan (openswan), then use sip phones to connect to a common sip server located at the home office. |
| 03:56.37 | nestAr | don't even need a vpn |
| 03:57.07 | stdio | nestAr: what, a sip proxy on the one side? |
| 03:58.19 | opus_ | Whoah |
| 03:58.23 | nestAr | connect the phones at the remote office directly to the * box at the main office |
| 03:58.51 | stdio | nestAr: with what.. sip? |
| 03:58.55 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
| 03:59.03 | Half_Dome | So, this SIP server will be an Asterisk box with a FXO 4 port card? |
| 03:59.26 | khaladan | couldn't i just buy 4 X100P FXO PCIs, instead of a quad pack thing |
| 03:59.47 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 04:00.44 | nestAr | stdio: yes |
| 04:01.04 | *** join/#asterisk hypa7ia (~leigh@HSE-Montreal-ppp143275.sympatico.ca) |
| 04:01.16 | nestAr | ugh |
| 04:01.24 | nestAr | why doesn't this stupid openclose thing work? |
| 04:02.06 | stdio | nestar: I imagine he'd use a nat appliance at the "remote" end. Is that going to nat those sip packets correctly? What about back through the other nat device on the other side.... |
| 04:02.24 | nestAr | in my world, nat doesn't exist |
| 04:03.02 | PTG123 | nat works great in my world |
| 04:03.10 | PTG123 | :) |
| 04:03.17 | nestAr | i have no need for nat |
| 04:03.21 | nestAr | thank god |
| 04:04.39 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 04:04.51 | PTG123 | nat is your friend :) |
| 04:04.57 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 04:05.03 | nestAr | NOOOOO!!!1 |
| 04:06.52 | *** join/#asterisk WasPhantom (~neil@203-86-192-98.tasman.net) |
| 04:07.39 | WasPhantom | hey all..... I'm currently working on using asterisk to talk to the INOC-DBA system, and within which, you dial a code ( in this case, 8) ASN, and then * EXTN, however, I'm having no luck with the extension.... |
| 04:07.39 | *** part/#asterisk RazaMetaL (~razametal@pc.gsalas.manta.telconet.net) |
| 04:07.44 | Vco | anyone have pointers to find out why g729 will work XTEN<->XTEN but not BT-100<->BT-100? |
| 04:07.58 | WasPhantom | I'm registered correctly to the server, as I can dial the ASN only, but I have no luck when I add an extension..... |
| 04:09.24 | Vco | was getting inband errors and fixed them, now i see errors on the console about dropping frames since we already have them... |
| 04:09.39 | Vco | i do a show g729 0/0 |
| 04:10.05 | mwcnetwork | i just dialed 500 |
| 04:10.14 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 04:10.15 | Vco | try the same settings with XLite....and seems to work... |
| 04:10.25 | *** part/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 04:10.25 | mwcnetwork | how do I set up a phone number in 'local'? |
| 04:11.59 | dmccollum | I was told that there are some models of the Cisco 7940/7960's that don't support SIP. Is this true? I thought they all could have their firmware upgraded to support SIP. |
| 04:12.13 | PTG123 | yah they dont' unless you upgrade them |
| 04:12.19 | PTG123 | most of the time they come without sip license |
| 04:12.21 | PTG123 | used |
| 04:12.29 | mwcnetwork | in order to add xlite client, edit /etc/asterisk/sip.cong |
| 04:12.32 | mwcnetwork | .conf |
| 04:12.32 | PTG123 | and they are a pain in the butt to upgrade |
| 04:12.33 | PTG123 | i just did one |
| 04:12.38 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 04:12.58 | BrianR___ | dmccollum: requires firmware upgrade - very much a pain in the ass for certain earlier revisions of the firmware... The upgrade path usually involves several intermediate versions. |
| 04:13.17 | dmccollum | yea, I've done a Cisco Call Manager install before, so I'm certain that I could setup the tftp firmware upgrade. |
| 04:13.32 | t0p | Hi, friends |
| 04:13.39 | BrianR___ | dmccollum: Also, a few very old firmwares had bogus firmware loaders which made getting any other firmware on the unit difficult. (Ie, firmware loader runs out of memory or crashes while downloading new image) |
| 04:13.49 | t0p | may I ask a newbie question here? |
| 04:13.59 | PTG123 | BrianR: actuallyu i figured out a full proof way to do it without stepping up one at a time |
| 04:14.00 | dmccollum | Cool. thanks for the answer. Just as long as they can be upgraded is my main concern. |
| 04:14.09 | BrianR___ | So your milage may vary. I was able to upgrade all of my Cisco 79xx phones to a useful firmware. |
| 04:14.11 | mwcnetwork | yes go ahead |
| 04:14.13 | PTG123 | BrianR___: it involves some bogus files which i have no one my tftp for future upgrades |
| 04:14.17 | PTG123 | :) |
| 04:14.23 | mwcnetwork | I have added 1000 and 2000 as users |
| 04:14.37 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 04:14.42 | mwcnetwork | sip show users lists them |
| 04:14.57 | t0p | Thanks, does Asterisk work with Dialogic D/160 or D/41 |
| 04:15.04 | BrianR___ | I have about a dozen files on my tftp for stepping various phones through the various versions. An ancient phone plugged in reboots about 8 times before finally completing all of the upgrades.. |
| 04:15.38 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
| 04:15.49 | PTG123 | BrianR :) true |
| 04:16.09 | PTG123 | Can we use ftp on these things, or only tftp? |
| 04:16.12 | BrianR___ | Anyway, you do need a cisco support account to download firmware images, but any support account will work. |
| 04:16.13 | PTG123 | b/c isn't tftp pretty insecure? |
| 04:16.31 | BrianR___ | PTG123: Using tftp for read-only access to firmware blobs is just fine. |
| 04:16.44 | PTG123 | well how do you make it so they can only ftp THIER firmware |
| 04:16.44 | BrianR___ | Besides, the ftp client in many of the cisco firmwares is broken. |
| 04:16.48 | PTG123 | so they can't see others pwrds |
| 04:17.07 | BrianR___ | PTG123: There's no good solution for that problem, unfortunately. |
| 04:17.16 | BrianR___ | PTG123: Well.. Aside from not putting the passwords in the config files.. |
| 04:17.20 | mwcnetwork | any dial xxxx leads to a busy signal |
| 04:17.27 | mwcnetwork | other than 500, 600 |
| 04:17.46 | PTG123 | BrianR: on a local lan i could make a custom tftp daemon that matches nat addy |
| 04:17.47 | WasPhantom | ok.... when you dial a number*exten, how should it look in a debug? |
| 04:17.48 | WasPhantom | <PROTECTED> |
| 04:17.54 | PTG123 | on tftp can you make it so list files doesnt work? |
| 04:18.02 | WasPhantom | thats what I get, but the number I'm dialing, is SIP/3856*878 |
| 04:19.25 | WasPhantom | <PROTECTED> |
| 04:19.30 | BrianR___ | PTG123: The tftp daemon on my machine can run an external program to do filename remapping... You could do checks there. |
| 04:19.40 | *** join/#asterisk qwerp (~abc@219.93.57.58) |
| 04:19.47 | qwerp | harlo... |
| 04:20.15 | qwerp | is there anyway i can play a beeping sound to a channel that is offhook? |
| 04:20.17 | PTG123 | brianr: hmm might be easier just to modify source on unix tftp |
| 04:20.18 | Vco | you did set context=local in sip.conf and have the bits under the same context in extensions.conf? |
| 04:20.32 | PTG123 | so unless people brute force mac addies |
| 04:20.33 | PTG123 | its safe |
| 04:20.54 | BrianR___ | Well.. There's no directory list command in tftp... |
| 04:21.16 | PTG123 | ah |
| 04:21.18 | PTG123 | then thats not so bad |
| 04:21.20 | PTG123 | kinda safe |
| 04:21.31 | BrianR___ | so it's a guessing game in either case. |
| 04:21.48 | qwerp | any way i can do that? |
| 04:22.17 | dmccollum | Is there anyway to setup asterisk so that you don't need to dial a number to get an outside line? I can see my wife complaining about that one when I finally get this installed. |
| 04:22.32 | *** join/#asterisk rious (~rious@adsl-69-208-72-102.dsl.klmzmi.ameritech.net) |
| 04:22.50 | Vco | you think thats *All* she'll complain about? ;) |
| 04:23.15 | mishehu | dmccollum: the sky is the limit. |
| 04:24.05 | Sedorox | slePP: Can I PM you? |
| 04:24.17 | dmccollum | well I'm confident I can make the rest pretty wife friendly. |
| 04:24.45 | Vco | I'm just using ours for calls to/ from japan to her family, and for voicemail rightnow... |
| 04:24.58 | WasPhantom | this is just really weird heh |
| 04:25.26 | *** join/#asterisk Rick_Hunter (~rhunter@04-073.008.popsite.net) |
| 04:25.43 | Vco | i guess she's spooked by the idea of me getting a tdm400 card since everyhting would go through the server after that.. |
| 04:28.37 | WasPhantom | exten => _8.,9,Dial(SIP/${EXTEN:1}@inoc-dba,90,r) |
| 04:29.14 | WasPhantom | that is the dial line in my outgoing call.... so when I dial 83856*878 it seems that the call is being sent to the right host, but as 3856878, rather than 3856*878 |
| 04:32.18 | *** join/#asterisk BIZH0P (~bizh0p@12.207.10.46) |
| 04:34.19 | elriah | Hey guys, where in asterisk is the actual voicemail conf that holds the names of all the vm-???.gsm files in sounds? |
| 04:35.57 | *** join/#asterisk zhier (~nick@219.136.15.39) |
| 04:36.04 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 04:38.49 | elriah | Hey guys, where in asterisk is the actual voicemail conf that holds the names of all the vm-???.gsm files in sounds? |
| 04:39.07 | docelmo | I believe its in the source. |
| 04:39.19 | elriah | ahh.. tnx |
| 04:39.54 | zhier | i want to answer the incoming call.but how?thanks |
| 04:40.09 | docelmo | What type of incoming call? |
| 04:40.33 | docelmo | either way doesnt matter its pretty much all done the same way. |
| 04:40.47 | docelmo | But are you answering on a DID or just answering in general? |
| 04:40.52 | zhier | sip |
| 04:41.09 | zhier | i have no any hardware |
| 04:41.26 | docelmo | in sip.conf make sure you specify context=default or something then in your extensions.conf do something like this: |
| 04:41.30 | WasPhantom | so, does anyone have any idea on my issue? ( By issue, I probably really mean inability to RTFM correctly) |
| 04:41.32 | docelmo | [default] |
| 04:41.56 | docelmo | exten => s,1,answer (Provide Answer) |
| 04:42.10 | docelmo | exten => s,2, What do you wanna do with the call.. |
| 04:42.13 | docelmo | so on and so forth |
| 04:42.29 | WasPhantom | as a refresher, and apologies to those who have read this all before.... |
| 04:42.31 | docelmo | if you wanna answer a DID and do something spacific.... |
| 04:42.42 | docelmo | exten => DID,1,Answer |
| 04:42.45 | zhier | oh,thanks,can i dial and answer the phone on the same pc? |
| 04:42.46 | docelmo | yada yada yada |
| 04:43.03 | WasPhantom | exten => _8.,9,Dial(SIP/${EXTEN:1}@inoc-dba,90,r) |
| 04:43.03 | WasPhantom | that is the dial line in my outgoing call.... so when I dial 83856*878 it seems that the call is being sent to the right host, but as 3856878, rather than 3856*878 |
| 04:43.09 | docelmo | should.. Dont understand your exact question |
| 04:43.50 | zhier | can i dial and answer the same phone on the same pc? |
| 04:44.01 | docelmo | as your * server yes |
| 04:44.44 | zhier | oh!but i can't do this ,i don't know why |
| 04:44.50 | docelmo | I dont believe * will pass your * try a # instead |
| 04:45.10 | docelmo | change your SIP ports |
| 04:45.19 | docelmo | 5060 on * and 5061 on the client |
| 04:45.33 | docelmo | Personally I use multiple computer and Linksys ATA's in my home |
| 04:46.02 | zhier | oh i'll try |
| 04:46.07 | zhier | thanks |
| 04:46.12 | docelmo | No biggie.. |
| 04:46.26 | docelmo | So whats the adverage going rate for Domestic termination right now? |
| 04:47.21 | WasPhantom | a # seems to do the same.... |
| 04:48.14 | *** part/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 04:48.26 | docelmo | hmm.. Dont know what to tell you. Never really tried to pass a alpha numeric string to someone via asterisk. It may not be possible. |
| 04:48.38 | docelmo | Check the WIKI for special characters in the Dial Application |
| 04:48.52 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 04:50.29 | elriah | Will the voicemail system play .wav replacements of the vm-??.gsm's in sounds? |
| 04:51.01 | docelmo | theoretically yes. Personally I never got it to work. I used sox to convert it. |
| 04:51.27 | elriah | Well, I noticed that sox creates gsm's with some background hissing. |
| 04:51.34 | elriah | so I was shooting for wavs. |
| 04:51.38 | docelmo | depends on your flags |
| 04:51.49 | elriah | Give it to me! What flags did you use? |
| 04:51.58 | docelmo | I forget what I use.. I got mine from the Yellow * book |
| 04:52.04 | docelmo | works very well |
| 04:52.13 | elriah | I have that book. I'll look it up. |
| 04:52.49 | docelmo | Look in the middle to back it will show a sox command. But one of the values doesnt work as that book was printed when sox was changed |
| 04:53.09 | elriah | -q1 |
| 04:53.10 | zhier | where can i buy the Yellow * book? |
| 04:53.15 | docelmo | yes that one |
| 04:53.17 | elriah | amazon has it, really good book. |
| 04:53.27 | elriah | do I replace -q1 with something else or just leave it off? |
| 04:53.30 | docelmo | Barnes and noble. A friend just told me he got it there |
| 04:53.34 | docelmo | leave it off |
| 04:53.39 | elriah | There's also an e-book. |
| 04:53.44 | dmccollum | Has anyone tried to show the xml weather feed on the LCD of a Cisco 7940/60? |
| 04:53.45 | elriah | Yea, that's what I did, got hissing. |
| 04:54.04 | docelmo | elriah call 800-481-5076 all voice prompts were converted with sox |
| 04:54.05 | elriah | Maybe the hissing won't be noticeable on the phone, I've been playing it on the console through my pc speakers. |
| 04:54.38 | zhier | e-book? where is the url? |
| 04:55.17 | docelmo | dmccollum, nope. But I have been thinking about trying it for an upcoming project I am working on |
| 04:55.19 | elriah | I called, those prompts have that same hissing I was talking about. Maybe I'm just being picky. |
| 04:55.29 | docelmo | I dont notice them.. hmm |
| 04:55.45 | elriah | http://www.signate.com/book.php |
| 04:56.01 | zhier | thanks |
| 04:56.05 | elriah | Well, maybe it's my waves are so clear ... |
| 04:56.08 | elriah | heh |
| 04:56.11 | docelmo | I have been looking for a windows version of an application to save gsm format. But so far have been unsuccessful |
| 04:56.17 | elriah | doesn't matter if the vm won't play the .wav's. |
| 04:56.19 | pigpen | If I were to hook my vonage ata to a sipura fxo...then have * grap on to it ...how bad would it sound? |
| 04:56.31 | pigpen | just temp...until my pri is delivered... |
| 04:56.54 | docelmo | pig just pump vonage directly into your * box as a sip client |
| 04:57.11 | docelmo | Search the Asterisk-users list's.. I saw it there last week I belive |
| 04:57.27 | pigpen | vonage only works with the softphone stuff..not the primary account |
| 04:57.34 | docelmo | ohh |
| 04:57.41 | pigpen | yeah..I was bummed. |
| 04:57.52 | docelmo | Im waiting on my 3 DS3's to be installed so I can get * up and running |
| 04:58.06 | pigpen | only 3? |
| 04:58.22 | docelmo | 3 in the us and 2 DS3's of E1's in the UK |
| 04:59.22 | pigpen | how many phones off one asterisk box? |
| 04:59.36 | docelmo | dont know.. Im planning to cluster |
| 04:59.43 | docelmo | 10 * boxes to start |
| 04:59.57 | pigpen | hmm...I would love to see your config files... |
| 05:00.04 | dmccollum | Can't you use the Vonage main number off the ATA box they give you into the X100? |
| 05:00.24 | pigpen | It looks like I may be thrown into large deployemnts soon... |
| 05:00.30 | docelmo | all configs are going to be done in mysql |
| 05:00.37 | pigpen | cool... |
| 05:00.41 | docelmo | And I am using a front end router |
| 05:00.45 | docelmo | well sip |
| 05:00.53 | pigpen | what front end router? |
| 05:00.57 | docelmo | SER |
| 05:01.05 | elriah | the audio file vm-INBOX.gsm, what is it saying? sounds like 'view' |
| 05:01.08 | pigpen | yep..that was I was thinking.. |
| 05:01.29 | docelmo | Then again I may just use a F5 I dont know yet. Looking at a couple configs |
| 05:02.02 | docelmo | I just bought 2 AS5850's for my DS3's I am planning to roll out service in beta late next month |
| 05:02.02 | pigpen | Are you using a custom datbase/config or a project? |
| 05:02.03 | JohnnyD | Hi, has anyone got the new Web-Meetme working? |
| 05:02.38 | pigpen | hmm...AS5850's...nice. |
| 05:02.46 | docelmo | I am using Realtime built in but some things modified for how I want them. ie reporting.. There is none... |
| 05:02.49 | pigpen | kicks the crap out of the AS5300's |
| 05:03.14 | docelmo | I got one for the US and one for my connection in the UK and I have SS7 connecting it all |
| 05:03.34 | pigpen | Cool..does Realtime support Postgresql? |
| 05:03.54 | docelmo | I will have LNP and Number lookups for caller id soon.. I dont know.. I use MySQL |
| 05:04.07 | pigpen | yeah..we pretty much use Postgres... |
| 05:04.42 | docelmo | Pig, where do you work? |
| 05:04.50 | pigpen | SA, TX |
| 05:05.16 | pigpen | I am one of 3 owners in a linux consulting company / ISP / Dev work...etc... |
| 05:05.43 | pigpen | yeah..not everyone knows how to handle the 5300/5850's... |
| 05:05.51 | pigpen | good equipment... |
| 05:05.58 | docelmo | hehe.. Im a cisco engineer :) |
| 05:05.58 | drumkilla | pigpen: it supports anything under odbc |
| 05:06.13 | pigpen | drumkilla: cool...I am trying to find it... |
| 05:06.19 | pigpen | CCIE...cool... |
| 05:06.29 | pigpen | I was going for it..but just got too busy... |
| 05:07.03 | docelmo | no.. CCNA/CCNP Not IE yet.. I wish |
| 05:07.29 | pigpen | ah...ok..I had those and lost them already...along with citrix, M$, etc... |
| 05:07.44 | pigpen | I have no desire anymore for certifications... |
| 05:08.01 | bjohnson | khaladan: the fxo will convert the voice to analog signals (you said your pbx extensions were analog phones) |
| 05:08.03 | pigpen | So what is the site for Realtime...? |
| 05:08.07 | docelmo | me either.. With this new venture.. I dont have time to deal with them.. |
| 05:08.10 | bjohnson | khaladan: the x100p are shit |
| 05:10.09 | docelmo | I have heard alot of up's and downs on the wild cards.. I found just shoving cisco in front does the job pretty well |
| 05:10.34 | pigpen | sure....and with DS3's...hey... |
| 05:10.48 | docelmo | Well.. They cost almost nothing |
| 05:11.03 | docelmo | Thats why I am wondering what current termination rates run |
| 05:11.15 | pigpen | for DS3's? |
| 05:11.21 | docelmo | cause I am gonna under cut most pricing |
| 05:11.24 | docelmo | yep |
| 05:11.36 | docelmo | unlimited incoming and / minute outgoing |
| 05:11.49 | pigpen | Are you a voip provider? |
| 05:12.03 | docelmo | in about 30 or so days |
| 05:12.11 | Sedorox | docelmo: what area you service? |
| 05:12.14 | pigpen | cool...v |
| 05:12.38 | docelmo | I am waiting for my cage, ds3's and internet to be dropped. I just ordered my gear. Should be here sometime in the next 2 weeks |
| 05:12.48 | docelmo | for DID's or termination? |
| 05:12.49 | pigpen | We are looking to do the same thing...but mostly for businesses on a virtual hosting areana... |
| 05:13.04 | pigpen | docelmo: Where are you located? |
| 05:13.08 | docelmo | NYC |
| 05:13.11 | docelmo | 111 8th |
| 05:13.13 | pigpen | ah...cool. |
| 05:13.20 | pigpen | I am in San Antonio, TX |
| 05:13.20 | docelmo | I am physically in Tampa, FL |
| 05:13.29 | docelmo | I spent 2 months there. USAF! |
| 05:13.29 | Shido6 | ZzZZz |
| 05:13.43 | Shido6 | cool docelmo, ds3's |
| 05:13.49 | pigpen | Randolph, Lackland or Kelly? |
| 05:14.01 | docelmo | hehe.. Forgot there were 3.. :) Lackland |
| 05:14.04 | docelmo | I did basic there |
| 05:14.09 | pigpen | cool... |
| 05:14.25 | docelmo | Shido you own NuFone right? |
| 05:15.50 | Qwell | Shido6 works for nufone? :) |
| 05:16.02 | docelmo | I dont know.. I dont keep up.. |
| 05:16.03 | *** join/#asterisk stdio (~stdio@pcp09745793pcs.lncstr01.pa.comcast.net) |
| 05:16.05 | elriah | What the hell is up with the tt-monkey sound in sounds? |
| 05:16.08 | elriah | Weasels? |
| 05:16.10 | Qwell | I'm trying to figure out why a woman just answered my newly assigned tollfree DID :P |
| 05:16.25 | docelmo | dunno.. |
| 05:16.28 | pigpen | well...did she sound hot? |
| 05:16.39 | Qwell | pigpen: umm, I freaked out, and hung up |
| 05:16.49 | pigpen | well pay attention next time... |
| 05:16.56 | Qwell | at first, I thought I dialed the wrong number, and felt stupid |
| 05:17.07 | Qwell | then I checked...and checked...and checked...and I had it right |
| 05:17.22 | docelmo | RESPORG's.. Gotta love em |
| 05:17.31 | stdio | elriah: try "weasel-eaten-phonesys" |
| 05:17.43 | Qwell | elriah: or teletubbies-murder |
| 05:17.52 | stdio | oh my |
| 05:18.27 | elriah | I don't see those two ... |
| 05:18.41 | Qwell | elriah: got the latest asterisk-sounds from cvs? |
| 05:19.18 | elriah | No - using 1.0.5, it comes in a nice, clean debian package. |
| 05:19.31 | *** join/#asterisk cripito (~saul@c-65-34-156-173.se.client2.attbi.com) |
| 05:19.43 | Qwell | minus half the sounds... |
| 05:20.16 | docelmo | Hay Shido what do you guys charge for termination? |
| 05:20.21 | elriah | heh .. I have 1.0.7, I pulled it for the new free hold music. |
| 05:20.22 | Shido6 | 2 cents/minute |
| 05:20.25 | elriah | I'll go check them out. |
| 05:20.32 | *** join/#asterisk pratik (~pratik@202.149.48.209) |
| 05:21.11 | pratik | gm to everyone |
| 05:21.26 | Qwell | Shido6: Do you work for nufone or something? |
| 05:21.31 | *** join/#asterisk Mazda-MX5 (~leo@220-130-142-43.HINET-IP.hinet.net) |
| 05:21.37 | Mazda-MX5 | hi ,all |
| 05:21.40 | pratik | is it posible to listen to the calls m,ade by asterisk |
| 05:21.45 | PTG123 | heh stop whining about your did qwell :) |
| 05:21.51 | stdio | pratik: 12:20 am EST us :) |
| 05:21.52 | Qwell | PTG123: It's b0rked :P |
| 05:21.52 | PTG123 | that lady stole it from you fair and square |
| 05:21.55 | Qwell | heh |
| 05:22.05 | PTG123 | heh |
| 05:22.10 | PTG123 | i'll give you a 800 did :) |
| 05:22.12 | pratik | stdio:well 10.30 india |
| 05:22.21 | stdio | pratik: heh :) |
| 05:22.22 | *** part/#asterisk mwcnetwork (~mwcnetwor@user-0c93oob.cable.mindspring.com) |
| 05:22.30 | pratik | just kiddin |
| 05:22.31 | Qwell | PTG123: I'm probably gonna end up giving this account to my mother-in-law now |
| 05:22.40 | PTG123 | heh |
| 05:22.41 | stdio | pratik: that must mean it's wednesday :) |
| 05:22.47 | Mazda-MX5 | I have stupid question,Asterisk for SIP is SIP server or SIP proxy ? or both ? |
| 05:22.47 | PTG123 | we'll sell her one too ;) |
| 05:22.49 | pratik | yes ti is |
| 05:22.51 | Qwell | PTG123: :P |
| 05:22.54 | PTG123 | Mazda-MX5: server |
| 05:23.05 | PTG123 | well i guess you could treate it like a proxy |
| 05:23.09 | PTG123 | so lets say both ;) |
| 05:23.10 | docelmo | Both |
| 05:23.19 | docelmo | you can proxy or pass data |
| 05:23.21 | Mazda-MX5 | PTG123 > can not setting for proxy ? |
| 05:23.22 | docelmo | your choice |
| 05:23.26 | stdio | Mazda-MX5: don't call it a proxy, they'll hang you for that. *sigh* |
| 05:23.28 | pratik | stdio:tell me is there any way to listen to the calls made by asterisk |
| 05:23.42 | PTG123 | if you are using it right, you are sort of proxying :) |
| 05:23.45 | pratik | i have a sipura phone attached with asterisk |
| 05:23.51 | PTG123 | re-invite rocks to give you the best connection |
| 05:23.59 | stdio | pratik: I want to say yes... assuming you have a sound card on the console... but I don't know how. |
| 05:23.59 | Mazda-MX5 | thank you PTG, stdio ~ |
| 05:24.17 | PTG123 | but the nice thing is if your not compatible codec wise, it can step in and transcode |
| 05:24.20 | pratik | so i want to listen to the calls made by tghis sipura phone |
| 05:24.30 | *** join/#asterisk rjburkh (~chatzilla@dialup-4.231.174.218.Dial1.LosAngeles1.Level3.net) |
| 05:24.32 | elriah | what is that tt-monkeys used for anyway? |
| 05:24.46 | stdio | pratik: if you figure out, I wouldn't mind knowing too... we have a ton of spa-841's. |
| 05:25.09 | PTG123 | you can record them |
| 05:25.12 | PTG123 | but no listen to them live |
| 05:25.19 | stdio | pratik: but the fact that it's sip shouldn't matter .. you should be able to listen to any call i think. |
| 05:25.24 | pratik | well the sound card i have it in a different pc but in the same network |
| 05:25.32 | pratik | will that work |
| 05:25.46 | Mazda-MX5 | aother 1 question , Asterisk "must" need database or not ? |
| 05:25.49 | stdio | elriah: pure fun. no other purpose. |
| 05:26.14 | stdio | Mazda-MX5: nope. you can use a db for extensions and other stuff if you want.... |
| 05:26.17 | *** join/#asterisk Newbie___ (some@218.111.159.51) |
| 05:26.29 | elriah | ahh... |
| 05:26.42 | Mazda-MX5 | thank you |
| 05:26.58 | Newbie___ | hi, anyone successfully using asterisk to connect to inphonex.com ? |
| 05:27.06 | stdio | Mazda-MX5: the advantage being, that you can then dynamically change extensions without having to restart asterisk or reload extensions, and you can slap a slick web interface on it and do all the other cool things that db servers can inherently do... |
| 05:28.14 | stdio | pratik: don't think so... PTG123's saying you can't even listen to them... only record... |
| 05:28.54 | stdio | elriah: lots of those sounds seem to be just for fun. |
| 05:29.23 | PTG123 | i haven't seen a live listen, although its possible :) |
| 05:29.27 | PTG123 | but i don't think it exists |
| 05:29.31 | Mazda-MX5 | well ,cdr need database, but I want not install ant database in platform. can i setting cdr to needless database ? |
| 05:30.03 | drumkilla | Mazda-MX5: you do not need a database by default |
| 05:30.23 | Mazda-MX5 | thank you , I will see conf docs again ~ |
| 05:30.56 | pratik | PTG123:well listening to the calls , nothing much iun detailk is given in the wiki as well |
| 05:32.18 | pratik | and another problem i am facing is i am not able to recieve incoming fwd calls from anywhere |
| 05:34.20 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 05:35.02 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 05:36.16 | *** join/#asterisk buu_ (~buu@ip67-95-66-69.z66-95-67.customer.algx.net) |
| 05:37.07 | buu_ | I've got kind of a bizarre AGI thing going on. I'm deep in the middle of a script, I just finished a "WAIT FOR DIGIT " type loop, and the next time I do "STREAM FILE", it instantly returns without streaming the file, but if I repeat that exact command again, it works perfectly |
| 05:37.13 | buu_ | Anyone have the vaguest clue where I could look? |
| 05:37.36 | docelmo | Did you write the script? |
| 05:37.43 | buu_ | Yes |
| 05:38.03 | docelmo | well minor detail in your script? |
| 05:38.12 | buu_ | What? |
| 05:38.18 | docelmo | I know when I have problems with AGI its usually me causing the problems |
| 05:38.38 | buu_ | Well, I'm sure it's me, but I have no idea where to look |
| 05:38.49 | docelmo | What was the script coded in? |
| 05:38.51 | buu_ | I tried reading from stdin before I print the first STREAM FILE, but nothing is there |
| 05:38.54 | buu_ | Perl |
| 05:39.11 | docelmo | outa my league.. I can read and understand but dont know any perl |
| 05:39.25 | buu_ | =/ |
| 05:39.33 | buu_ | I doubt it's a perl issue |
| 05:39.50 | docelmo | I dont know.. I do all of my AGI in php |
| 05:39.50 | buu_ | It kind of feels like there is an extra input hanging around.. |
| 05:39.53 | buu_ | I'm so sorry. |
| 05:40.02 | buu_ | Bad boss? |
| 05:40.11 | docelmo | I am the boss.. I know php fluently |
| 05:40.13 | stdio | might want to see how another command reacts... does it fail silently the first time, too? |
| 05:40.18 | docelmo | I just prefer to code in it.. |
| 05:40.42 | docelmo | I dont like the fact they changed PHPAGI.. I had to adapt 20 scripts to the new version.. |
| 05:41.39 | zhier | how can buy the VoIP Telephony with Asterisk e-Book? |
| 05:42.05 | docelmo | dunno? |
| 05:42.08 | buu_ | stdio: Yeah, every command fails silently |
| 05:42.12 | buu_ | At this particular spot |
| 05:42.24 | buu_ | Everything else works perfectly everywhere else =] |
| 05:42.34 | buu_ | It kind of seems like theres an extraneous digit hanging about |
| 05:42.43 | zhier | how can i buy the VoIP Telephony with Asterisk e-Book? |
| 05:42.55 | docelmo | http://www.osoft.com/store/productdetails.php?pid=39 |
| 05:43.26 | stdio | buu_ it's not the script... |
| 05:43.51 | stdio | buu_: yep, it's gotta be something looks for more input somewhere... |
| 05:43.57 | stdio | might want to pastebin it |
| 05:44.07 | stdio | although i shudder... |
| 05:44.31 | docelmo | ahh well all.. off to bed I go.. |
| 05:45.08 | zhier | what time? |
| 05:45.40 | buu_ | stdio: Eh, I suppose |
| 05:45.49 | buu_ | It's mildly long though, I don't really have a test case |
| 05:46.29 | stdio | buu_: yeh, not much we can do :( At least you have a rough idea of the problem now, though |
| 05:46.36 | buu_ | ... |
| 05:46.57 | buu_ | I do "wait for digit" |
| 05:46.59 | buu_ | read from stdin.. |
| 05:47.24 | stdio | is this in a macro? |
| 05:47.27 | JohnnyD | exit |
| 05:47.30 | buu_ | Um. No |
| 05:47.35 | buu_ | Because perl doesn't have macros |
| 05:47.36 | stdio | perl...? |
| 05:47.38 | buu_ | Most of the time |
| 05:47.41 | stdio | heh |
| 05:47.51 | stdio | they call them there things functions :) |
| 05:47.52 | *** part/#asterisk JohnnyD (~passionfr@203-217-21-234.perm.iinet.net.au) |
| 05:47.56 | buu_ | ... |
| 05:48.08 | buu_ | I can't think of a single language where macros and functions are equivalent |
| 05:48.18 | stdio | arright. procedures. |
| 05:48.32 | buu_ | Fortran?! |
| 05:48.35 | stdio | heh. |
| 05:48.40 | stdio | pascal :) |
| 05:48.43 | buu_ | ah |
| 05:48.46 | buu_ | Almost as bad I suppose |
| 05:48.49 | stdio | i dunno. it's late. |
| 05:48.54 | buu_ | One of those "dark age" languages |
| 05:49.09 | stdio | basic isn't functional, so it doesn't even qualify... |
| 05:49.11 | stdio | logo! |
| 05:49.13 | stdio | :) |
| 05:49.41 | buu_ | ick! |
| 05:49.55 | stdio | don't like the turtle, eh? |
| 05:50.41 | stdio | it's a good way to learn trig. |
| 05:51.39 | Nugget | LEFT 90 |
| 05:51.41 | Nugget | FORWARD 10 |
| 05:52.29 | *** join/#asterisk loud (~ariel@201.139.192.101) |
| 05:52.35 | *** join/#asterisk jedaustin (~chatzilla@host4.twingeckos.net) |
| 05:53.17 | jedaustin | I finally got my asterisk box to make outbound calls on Broadvoice.. they sound like crap from my Sipura841 phone.. is there a best protocol to select with this thing? |
| 05:54.01 | loud | you sure have a decent connection ? |
| 05:54.14 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
| 05:54.17 | jedaustin | loud: DSL |
| 05:54.45 | jedaustin | loud: supposed to be up to 1.5m bit |
| 05:55.08 | loud | sipura directly to bv |
| 05:55.55 | jedaustin | loud: the phone can connect directly to broadvoice? |
| 05:56.50 | loud | you can buy a BYOD (generic) and try |
| 05:57.09 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 05:57.11 | Newbie___ | hi, i got an account from inphonex.com and would like to make use of my * to call, any ideas ? |
| 05:57.20 | loud | im sure it shouldnt be a problem at all. |
| 06:00.11 | Newbie___ | i tried using examples from voip-info.org and i keep getting Unable to find SIP channels |
| 06:04.04 | zhier | and i want to know how i can get VoIP Telephony with Asterisk e-book? |
| 06:05.15 | zhier | who can tell me.thsnks |
| 06:05.29 | elric | google can. |
| 06:05.43 | zhier | i want to download it |
| 06:06.04 | *** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
| 06:06.14 | elriah | Any better * beeps than beep.gsm? |
| 06:06.27 | elriah | http://www.signate.com/book.php |
| 06:06.29 | elric | http://www.signate.com/book.htm |
| 06:06.31 | elriah | for the e-book |
| 06:06.40 | zhier | and i go to signate web, but i can't get it |
| 06:07.20 | elric | ah |
| 06:07.26 | elric | well buy it online |
| 06:07.39 | elric | http://www.osoft.com/store/productdetails.php?pid=39 |
| 06:07.43 | elric | and then download |
| 06:08.05 | elriah | Or get the paperback from amazon. Good read. Explains about the phone systems in general too. |
| 06:08.14 | Newbie___ | i got mine from osoft.com |
| 06:10.40 | Mw3 | damn, they dont ship to hungary :( |
| 06:10.51 | zhier | but when i go to the url "http://www.osoft.com/store/productdetails.php?pid=39", i can't find the url for download! |
| 06:11.25 | elric | zhier, you have to pay money |
| 06:11.27 | elric | to download |
| 06:12.01 | zhier | oh,but i don't know how to pay money. |
| 06:12.13 | elric | credit card |
| 06:12.44 | *** join/#asterisk cybercobra (~chris@h-67-101-210-127.snfccasy.dynamic.covad.net) |
| 06:12.51 | zhier | oh,and i know |
| 06:13.01 | *** part/#asterisk cybercobra (~chris@h-67-101-210-127.snfccasy.dynamic.covad.net) |
| 06:13.53 | qwerp | harlo |
| 06:14.06 | qwerp | anyone here know how to use super valet parking? |
| 06:14.06 | Supaplex | hi |
| 06:14.40 | kks | Anyone can help me to debug my pri call http://www.pastebin.com/261628, i'm suffering one minute incomming call drop. I really appreciate help from someone because i have no idea what is the debug msg indicating. |
| 06:15.11 | *** join/#asterisk goatmilk (~goatmilk@cae168-249-184.sc.rr.com) |
| 06:15.37 | qwerp | anyone? please guide me on super valet parking? |
| 06:16.47 | Supaplex | if we don't answer, we don't know, or you questions isn't a "smart question". (or smart enough). I still consider myself a * newfie, or I'd say something. |
| 06:17.26 | qwerp | wukie.. |
| 06:18.12 | *** join/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 06:18.45 | *** part/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 06:19.10 | *** part/#asterisk CaT[tm] (~cat@nessie.weebeastie.net) |
| 06:21.18 | *** join/#asterisk Hydroxide (user@Hydroxide.developer.debian) |
| 06:22.14 | Hydroxide | hi, I'm using a CVS HEAD snapshot from Dec 16 2004, and I can't figure out how to tell it to look in /usr/share/asterisk/sounds for my sounds instead of /var/lib/asterisk/sounds |
| 06:22.25 | Hydroxide | it was quite easy to do in asterisk.conf with 1.0.5 |
| 06:27.08 | elriah | Anybody here use festival? |
| 06:27.42 | Hydroxide | yeah. I have had it cause asterisk to hang hard needing a kill -9. that's why I'm trying out the lastest Debian-packaged CVS HEAD version to see if it has that bug fixed |
| 06:27.58 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:28.09 | Hydroxide | if not I'd be fine using 1.0.x |
| 06:28.40 | Hydroxide | (those version numbers were asterisk version numbers, not festival ones) |
| 06:30.34 | elriah | Hydroxide: How's the speech quality? |
| 06:30.59 | goobster | I haven't been able to get festival to work with asterisk |
| 06:31.13 | Hydroxide | usable for development or hobbyist/noncommercial purposes, but definitely robotic-sounding |
| 06:31.17 | elriah | I've been using swift, the quality is really good. |
| 06:31.41 | elriah | It's very monotone, but is acceptable I think - swift, that is, I haven't heard festival. |
| 06:31.57 | Hydroxide | elriah: URL for swift, please? and does it already work with asterisk? |
| 06:32.09 | elriah | It's $$ |
| 06:32.11 | Hydroxide | ah |
| 06:32.32 | Hydroxide | well, I can't compare the two then ;) |
| 06:32.33 | elriah | $29.00 for the engine, but if you want to have * call it realtime, you need a port licens, it's expensive. |
| 06:32.42 | elriah | You can download the demo. cepestal.com? |
| 06:32.52 | bkw_ | new ultimate task... Cybersex via SMS!!! muhahahaha :P |
| 06:32.56 | Hydroxide | nah, no motivation to do so right now |
| 06:33.00 | Hydroxide | thanks though |
| 06:33.06 | elriah | cepstral.com |
| 06:33.11 | bkw_ | elriah, WRONG |
| 06:33.13 | elriah | There's a demo online. |
| 06:33.17 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 06:33.19 | bkw_ | you don't have to have a lic for realtime |
| 06:33.20 | bkw_ | *SMACK* |
| 06:33.27 | Hydroxide | right now I'm trying to figure out how to tell asterisk to look for sounds in /usr/share/asterisk/sounds instead of /var/lib/asterisk/sounds |
| 06:33.31 | *** join/#asterisk mwgbc (mwallacegb@adsl-69-109-181-140.dsl.pltn13.pacbell.net) |
| 06:33.41 | bkw_ | use the src luke |
| 06:33.50 | elriah | bkw: That's not what I read... Plus I bought the $29.00 license... It only reads one thing at a time, in a queue type fasion. |
| 06:34.05 | elriah | so if there are multiple callers, the second caller has to wait for swift to finish. |
| 06:34.06 | bkw_ | depends on how you code it |
| 06:34.06 | bkw_ | ;) |
| 06:34.15 | bkw_ | again depends on how you code it |
| 06:34.19 | Hydroxide | bkw_: shouldn't it be documented? I was able to do it in 1.0.x in asterisk.conf I think |
| 06:34.35 | elriah | Yea, I tried that agi sample script - it doesn't allow for interruptions if a tone is pressed. |
| 06:34.36 | bkw_ | Hydroxide, same place if ya did it there |
| 06:34.46 | Hydroxide | bkw_: what do you mean? |
| 06:34.58 | elriah | Press 1 for this, 2 for that, the user has to wait for the entire thing to make a choice. |
| 06:35.01 | Hydroxide | bkw_: I didn't have to modify any source in 1.0.x |
| 06:35.04 | bkw_ | see this |
| 06:35.04 | bkw_ | astvarlibdir => /var/lib/asterisk |
| 06:35.05 | qwerp | bkw_: valet parking is write by u, can i ask some question about it? |
| 06:35.11 | stdio | bkw_: yep, change it in asterisk.conf - but you have to chage the entire var lib dir.. |
| 06:35.13 | bkw_ | I DID NOT WRITE IT |
| 06:35.16 | Hydroxide | bkw_: right, I've already changed that to /usr/share/asterisk |
| 06:35.23 | bkw_ | then use a symlink |
| 06:35.24 | Hydroxide | bkw_: and restarted asterisk |
| 06:35.28 | qwerp | bkw_: sorry.. |
| 06:35.29 | bkw_ | cd /var/lib/asterisk |
| 06:35.32 | Hydroxide | bkw_: and it still doesn't work |
| 06:35.40 | stdio | it's still looking in /var/lib/asterisk? |
| 06:35.41 | bkw_ | mv sounds sounds.old |
| 06:35.44 | BoRiS | bkw!!!!!!!!! |
| 06:35.52 | Hydroxide | bkw_: I use /var/lib/asterisk/sounds for my locally installed sounds and /usr/share/asterisk/sounds for sounds installed through the Debian packaging system |
| 06:35.53 | bkw_ | ln -s /usr/share/asterisk/sounds . |
| 06:35.58 | Hydroxide | which is the way it should work |
| 06:36.02 | bkw_ | here is what i do |
| 06:36.06 | bkw_ | cd /var/lib/asterisk/sounds |
| 06:36.14 | bkw_ | ln -s /usr/share/asterisk/sounds local |
| 06:36.19 | bkw_ | then playback(local/sound) |
| 06:36.32 | Hydroxide | but doesn't that break all the standard applications such as Directory? |
| 06:36.42 | bkw_ | no |
| 06:36.49 | bkw_ | you're not thinking... so lets move along |
| 06:36.53 | bkw_ | BoRiS, how are you |
| 06:37.06 | riksta | lol bkw_ :D |
| 06:37.10 | Hydroxide | bkw_: no need to be rude ... I am definitely thinking. I've even tried stracing the binary, which had really interesting results |
| 06:37.18 | BoRiS | bkw: Not bad........looking at ChanSpy right now <g> |
| 06:37.27 | bkw_ | Hydroxide, no you're over thinking... thats what I should have said |
| 06:37.39 | bkw_ | and I wasn't being rude.. trust me.. if I were you would know it |
| 06:37.48 | bkw_ | I told you how to solve the problem in two ways |
| 06:37.54 | *** join/#asterisk pascals (~248d34d6@ip503c8584.speed.planet.nl) |
| 06:38.02 | bkw_ | I usually give up after one |
| 06:38.06 | pascals | Good morning |
| 06:38.08 | bkw_ | :P |
| 06:38.15 | bkw_ | oh wait that didn't come out right |
| 06:38.21 | riksta | hahaha |
| 06:38.23 | bkw_ | BoRiS, 996 |
| 06:38.31 | pascals | Anyone have a polycom ip 300/500/600 phone? |
| 06:38.36 | riksta | that's what ya wife says bkw_ :) |
| 06:38.40 | BoRiS | ok...give me a sec :) |
| 06:38.41 | Beirdo | bkw_: any reason app_dbodbc isn't included by default? :) |
| 06:38.46 | bkw_ | i'm gay boi |
| 06:38.47 | bkw_ | I don't have a wife |
| 06:38.54 | bkw_ | Beirdo, its lame.. thats why ;) |
| 06:38.56 | riksta | lol |
| 06:38.57 | BoRiS | His wifes name is .... |
| 06:38.58 | bkw_ | I did that as a learning project |
| 06:39.01 | Hydroxide | bkw_: well, the Directory application tries to play the sound dir-intro-fn. if dir-intro-fn.gsm is in the local/ subdirectory, how will Directory find it? |
| 06:39.19 | crash3m_ | anyone have an IP300 -w- firmware version 1.4.1.0040 handy that could test something for me? |
| 06:39.21 | bkw_ | Hydroxide, no you leave the default sounds where they are |
| 06:39.22 | Hydroxide | bkw_: does it search all subdirectories? that would allow name colisions |
| 06:39.28 | Beirdo | well, it seems to me that's the only way to pull stuff outta a real DB, no? |
| 06:39.32 | bkw_ | no |
| 06:39.36 | bkw_ | Hydroxide, on |
| 06:39.37 | bkw_ | no |
| 06:39.38 | Hydroxide | bkw_: the default sounds are in /usr/share/asterisk/sounds |
| 06:39.45 | bkw_ | no |
| 06:39.47 | Hydroxide | bkw_: since I installed through Debian packages |
| 06:39.50 | pascals | crash3m_ do you have the polycom firmware.zip file? |
| 06:39.51 | bkw_ | in /var/lib/asterisk/sounds |
| 06:39.59 | bkw_ | you create a dir or a symlink called local |
| 06:40.00 | bkw_ | then |
| 06:40.04 | bkw_ | you keep local sounds there |
| 06:40.07 | crash3m_ | pascals: no, but you can get it on freedomphones.net/polycom/ |
| 06:40.17 | bkw_ | just put the sounds where you want them |
| 06:40.21 | bkw_ | and symlink it |
| 06:40.23 | bkw_ | it really doesn't matter |
| 06:40.31 | pascals | crash3m_ thank YOU! |
| 06:40.38 | crash3m_ | pascals: no problem |
| 06:40.45 | Hydroxide | bkw_: the default sounds are not and have never been in /var/lib/asterisk on my system, since I installed through Debian packages, and I want them to live where the Debian package management system will upgrade them when I upgrade my system |
| 06:40.53 | crash3m_ | pascals: been searching for them long? |
| 06:40.55 | pascals | I spent half a day looking for that, last monday |
| 06:41.13 | Hydroxide | bkw_: it really shouldn't be so hard to have asterisk look elsewhere ... if it's not possible then it's at least a minor bug |
| 06:41.24 | bkw_ | no its not a bug thats for sure |
| 06:41.30 | bkw_ | I bet debian patched it |
| 06:41.45 | bkw_ | let me look i bet it takes me 2 seconds to find this |
| 06:41.54 | Hydroxide | oh I'm sure I can find it too |
| 06:41.55 | crash3m_ | pascals: lol, I swear polycom makes them impossible to locate |
| 06:42.06 | bkw_ | asterisk.h:#define AST_SOUNDS AST_VAR_DIR "/sounds" |
| 06:42.14 | bkw_ | TADAAAAAAA |
| 06:42.19 | pascals | crash3m_ I really don't understand a policy like that... |
| 06:42.22 | bkw_ | it will be in your var/sounds |
| 06:42.32 | bkw_ | if it was changed in asterisk.conf |
| 06:42.36 | bkw_ | or in the debian package |
| 06:42.39 | bkw_ | you'll have to patch it |
| 06:42.58 | pascals | The only people interested would be the phone users, and any competitor interested enough would be able to get a copy anyway |
| 06:43.23 | Hydroxide | wildcard:~# grep varlibdir /etc/asterisk/asterisk.conf |
| 06:43.23 | Hydroxide | astvarlibdir => /usr/share/asterisk |
| 06:43.23 | Hydroxide | wildcard:~# ls -l /usr/share/asterisk/sounds/beep.gsm |
| 06:43.23 | Hydroxide | -rw-r--r-- 1 root root 726 2004-12-16 20:50 /usr/share/asterisk/sounds/beep.gsm |
| 06:43.36 | bkw_ | as it should be |
| 06:43.38 | Hydroxide | even with a properly configured astvarlibdir setting in asterisk.conf, it can't find beep when I do Playback(beep) |
| 06:43.39 | bkw_ | thats what you want right? |
| 06:43.50 | crash3m_ | pascals: which model of polycom do you have? |
| 06:43.51 | Hydroxide | it is, but it doesn't find the sounds. it does find my agi-bin directory |
| 06:43.51 | bkw_ | show me what the CLI output says |
| 06:43.53 | rvhi | anyone has problem with polycom phone registration with the server? |
| 06:44.15 | bkw_ | crank up the logger.conf console line to include debug,notice,warning |
| 06:44.17 | rvhi | i have a few phones, all same setting, register and expire=600 in phone.xml config |
| 06:44.28 | rvhi | some sends register in 300 sec |
| 06:44.42 | rvhi | some sends 1800 sec |
| 06:44.42 | Beirdo | is there any external way to load up/manipulate the asterisk database that DBput/DBget use? |
| 06:44.42 | Hydroxide | already does, as well as error ... one sec |
| 06:44.44 | rvhi | some sends 600 sec |
| 06:45.16 | pascals | rvhi: I don't see my phones register very often, but that doesn't seem to affect proper function |
| 06:45.41 | pascals | rvhi: Mind you, I have dhcp set to always issue the same IP to the same phone |
| 06:46.12 | Hydroxide | Mar 23 01:45:48 DEBUG[9115]: pbx.c:1259 pbx_extension_helper: Launching 'Playback' |
| 06:46.15 | Hydroxide | Mar 23 01:45:48 WARNING[9115]: file.c:475 ast_openstream: File beep does not exist in any format |
| 06:46.18 | Hydroxide | Mar 23 01:45:48 WARNING[9115]: file.c:779 ast_streamfile: Unable to open beep (format ulaw): No such file or directory |
| 06:46.21 | Hydroxide | Mar 23 01:45:48 WARNING[9115]: app_playback.c:83 playback_exec: ast_streamfile failed on SIP/51-fe8b for beep |
| 06:46.35 | Hydroxide | and I showed you already that it is there |
| 06:46.48 | elriah | I only caught part of that, did you find a better 'beep'? |
| 06:46.57 | Hydroxide | an strace showed it looking in /var/lib/asterisk/sounds for that file but not at all in /usr/share/asterisk/sounds |
| 06:47.15 | Hydroxide | elriah: no, I'm using it as a generic test sound to debug sound file location issues |
| 06:47.24 | elriah | Ahh |
| 06:47.56 | bkw_ | Hydroxide, I see whats up |
| 06:48.01 | bkw_ | looking at it more |
| 06:48.13 | Hydroxide | oh? |
| 06:48.45 | *** part/#asterisk mwgbc (mwallacegb@adsl-69-109-181-140.dsl.pltn13.pacbell.net) |
| 06:49.43 | bkw_ | AST_SOUNDS isn't used ANYWHERE |
| 06:49.56 | Hydroxide | hehe, wow |
| 06:50.39 | bkw_ | thats dumb |
| 06:51.26 | terrapen_ | anyone use SBC DSL? |
| 06:51.30 | bkw_ | you using cvs-stable? |
| 06:51.55 | mishehu | terrapen_: I do, unfortunately. |
| 06:52.08 | terrapen_ | ugh |
| 06:52.12 | terrapen_ | i need high speed internet |
| 06:52.18 | mishehu | Stupid Bastard Cocksuckers |
| 06:52.18 | terrapen_ | that doesnt cost $60/month |
| 06:52.19 | bkw_ | I think ast_fileexists is at fault |
| 06:52.22 | terrapen_ | or require contracts |
| 06:52.32 | mishehu | terrapen_: good luck. |
| 06:52.38 | mishehu | doesn't exist as far as I know. |
| 06:52.44 | mishehu | at least not in the usa |
| 06:52.46 | elriah | What'st he exten => 8500(???) to pass the callerid? |
| 06:52.48 | terrapen_ | RoadRunner is like $40/mo but i dont want cable TV |
| 06:53.31 | terrapen_ | i'm following the Dave Ramsey approach and cutting all non-essential indulgences from my life |
| 06:53.36 | terrapen_ | i dont need cable |
| 06:53.36 | bkw_ | go lok at ast_buildfilename |
| 06:53.39 | Hydroxide | this is a cvs head snapshot from dec 16, which is the latest I can find a Debian package for. I'm perfectly willing to try switching to cvs-stable or any other version, esp. if you're going to apply a patch now |
| 06:53.43 | Hydroxide | ah, okay |
| 06:53.45 | terrapen_ | i'm going to sell my payphone |
| 06:53.45 | Hydroxide | let me download cvs-stable |
| 06:53.53 | terrapen_ | and sell this bosch washer/dryer |
| 06:54.28 | bkw_ | Hydroxide, show me your exact extensions.conf entry |
| 06:54.28 | terrapen_ | i wish my neighbors had wifi |
| 06:54.28 | terrapen_ | i would be content leeching off of them |
| 06:54.32 | bkw_ | Hydroxide, no |
| 06:54.34 | bkw_ | use cvs-head |
| 06:54.37 | bkw_ | thats what i'm working with |
| 06:54.37 | Hydroxide | exten => 4,1,Playback(beep) |
| 06:54.39 | Hydroxide | ok |
| 06:54.39 | bkw_ | just checking to see |
| 06:54.44 | Beirdo | sell your payphone? |
| 06:54.48 | terrapen_ | ya |
| 06:54.50 | mishehu | pay your cellphone? |
| 06:54.50 | Beirdo | didn'y you JUST get it? |
| 06:54.52 | terrapen_ | i don't need it |
| 06:54.55 | terrapen_ | yes. |
| 06:54.56 | bkw_ | did you restart asterisk after you set the new vardir? |
| 06:55.00 | terrapen_ | its not even unwrapped, beirdo |
| 06:55.01 | Beirdo | sheesh |
| 06:55.11 | Hydroxide | bkw_: should I download a newer version than dec 16 2004? it would not be in a debian package, but I will do it if there is a useful reason to |
| 06:55.15 | Hydroxide | bkw_: yes, several times |
| 06:55.18 | terrapen_ | im determined to get out of most of my debt in a year |
| 06:55.20 | Beirdo | what kinda geek buys a toy then wants to immediately get rid of it |
| 06:55.31 | terrapen_ | i have like $4100 in debt |
| 06:55.42 | mishehu | use the payphone to reclaim some of your debt? |
| 06:55.51 | terrapen_ | well, plus $20,500 in student loans but they are so low-interest that they don't matter much |
| 06:56.00 | mishehu | get a payphone to sell, so you can pay your cellphone... |
| 06:56.06 | mishehu | makes sense |
| 06:56.08 | terrapen_ | beirdo: remorseful geeks like mine |
| 06:56.15 | terrapen_ | oh, and the cell phone |
| 06:56.19 | terrapen_ | i will keep that i guess |
| 06:56.26 | terrapen_ | but i should nix it |
| 06:56.27 | Beirdo | too bad you ain't in Toronto |
| 06:56.28 | terrapen_ | and just do VoIP |
| 06:56.32 | terrapen_ | its $50/mo |
| 06:56.40 | terrapen_ | at least that actually |
| 06:56.43 | terrapen_ | $700-800/yr |
| 06:56.51 | Beirdo | yeah |
| 06:56.55 | Beirdo | keep the payphone |
| 06:56.57 | bkw_ | Hydroxide, ya do that.. latest CVS |
| 06:56.58 | bkw_ | check it |
| 06:57.00 | Beirdo | ditch the cellphone |
| 06:57.00 | bkw_ | I have to go to bed |
| 06:57.07 | terrapen_ | beirdo, i dont know how to program the payphone |
| 06:57.08 | bkw_ | in two min |
| 06:57.15 | terrapen_ | it needs some funky-ass windows app to set it up |
| 06:57.22 | terrapen_ | and i will need to connect it to a POTS line for htis |
| 06:57.26 | terrapen_ | err this |
| 06:57.29 | Beirdo | I'm sure you can get it working :) |
| 06:57.31 | terrapen_ | i think i will just ebay it back |
| 06:57.43 | Beirdo | heh. Have fun then :) |
| 06:57.50 | terrapen_ | beirdo, but if i spend that time on doing a consulting gig instead... |
| 06:57.57 | terrapen_ | beirdo, wanna buy it? |
| 06:58.07 | Hydroxide | bkw_: thanks. I'll compile and test it out. should I let you know how it goes in some manner? |
| 06:58.14 | Beirdo | I have no use for it without a house :) |
| 06:58.23 | Beirdo | and I don't wanna pay shipping |
| 06:58.27 | Beirdo | nor customs |
| 06:58.36 | terrapen_ | i just had this huge...what is the word...awakening...moment of clarity... that i want to be completely debt-free |
| 06:58.40 | bkw_ | Hydroxide, I'm always here |
| 06:58.48 | Hydroxide | bkw_: okay then. enjoy bed. |
| 06:58.51 | Hydroxide | and thanks again |
| 07:02.24 | SexyKen | Anyone know what this does: switch => Realtime/default@extensions |
| 07:03.22 | rjburkh | hello again, please what does 'modprobe: can't locate module zaptel' mean? |
| 07:07.20 | riksta | rjburkh: are you joking or what? |
| 07:10.19 | cobryce | rjburkh: Probably that your zaptel module isn't installed. |
| 07:10.43 | Beirdo | anyone wanna try sending me a test fax? |
| 07:10.55 | cobryce | area code? |
| 07:11.00 | Beirdo | 416 |
| 07:11.26 | cobryce | sry |
| 07:11.30 | Beirdo | I'm not sure if I caught all the places... |
| 07:11.36 | Beirdo | I'll try it from work tomorrow :) |
| 07:11.49 | Beirdo | anyonw have fax transmission behaving? |
| 07:12.36 | cobryce | So I realise that a SIP "address" can simply be given as exten@server, but what would the syntax be for an IAX2 address? |
| 07:12.58 | Beirdo | IAX2/user@server/extension |
| 07:13.49 | cobryce | lol |
| 07:14.00 | cobryce | Not quite what I meant |
| 07:14.17 | cobryce | But close enough :) |
| 07:14.19 | rjburkh | you guys are good |
| 07:15.56 | cobryce | yup, that's us |
| 07:16.19 | rjburkh | want to try another? |
| 07:16.25 | cobryce | sure... |
| 07:16.33 | rjburkh | linux-2.4/include/linux/kernel.h:60: invalid suffix on integer constant |
| 07:17.19 | cobryce | Is that an error or just a warning? |
| 07:18.17 | rjburkh | it the first line of error in my attempt to compile zaptel-1.0.7 |
| 07:18.33 | cobryce | How recent are your kernel headers? |
| 07:19.50 | *** join/#asterisk BlueMuscle (~jfarland@dsl081-037-032.lax1.dsl.speakeasy.net) |
| 07:20.15 | BlueMuscle | I asked last night and nobody was sure, but is anyone aware of a bug or other reason that my 'show queue' would show 1681339180 callers waiting? |
| 07:20.35 | Qwell | BlueMuscle: Did you get famous overnight? :) |
| 07:20.42 | BlueMuscle | Yes. |
| 07:20.45 | rjburkh | do you mean kernel headers as in linux-2.4.20-8 |
| 07:20.48 | BlueMuscle | Yes I did. |
| 07:21.09 | BlueMuscle | I started an American Idol voting system. |
| 07:21.14 | BlueMuscle | Not. |
| 07:22.03 | modulus_ | linux sucks |
| 07:22.10 | BlueMuscle | No, but seriously, it has done this a few times, and occasionally the 'max' value will also change to some high number and then at some point the 'holding' can exceed the 'max' (they both get up in the millions, sometimes one a million or two above the other) and therefore it stops allowing callers. |
| 07:22.41 | BlueMuscle | Just thought I'd check back in again to see if anyone new had popped in that may have experienced this. |
| 07:25.27 | SexyKen | does anyone know what class 5 switching is? |
| 07:27.40 | Qwell | SexyKen: Try here maybe. http://wiki.cs.uiuc.edu/cs427/Class+5+Switch+Introduction |
| 07:28.41 | Qwell | I know for sure brettnem knows |
| 07:30.39 | *** part/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 07:33.02 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 07:33.19 | *** join/#asterisk fenlander (~irc@82.152.81.57) |
| 07:45.41 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 07:45.43 | rjburkh | cobryce, thanks for the tips. I better go to sleep on it. |
| 07:45.45 | Alexi1 | hi |
| 07:54.05 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 07:54.46 | ta[i]nted | anyone here familiar with digium's 729 codec? |
| 07:55.55 | modulus_ | nope it costs money |
| 07:56.04 | modulus_ | hi tainted |
| 07:56.14 | ta[i]nted | hey what's up |
| 07:56.16 | ta[i]nted | how's your app going |
| 07:56.31 | modulus_ | it's still running |
| 07:56.34 | langals | Hi there. When I run asterisk in the console I get the following error: "WARNING[1670]: chan_oss.c:269 sound_thread: Read error on sound device: Resource temporarily unavailable". This is repeated about 10 times. Does anyone have any idea what this is about? |
| 07:56.50 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 07:56.54 | modulus_ | tainted, got a big buyer for phone cards |
| 07:57.15 | ta[i]nted | how big |
| 07:57.29 | modulus_ | couple hundred thousand cards |
| 07:58.33 | ta[i]nted | that's pretty big |
| 07:58.45 | modulus_ | i'd tell you the buyer but it's a secret |
| 07:58.56 | ta[i]nted | isn't it always |
| 07:59.01 | modulus_ | you'd never believe that they'd purchase so many calling cards |
| 08:00.08 | ta[i]nted | cool man |
| 08:00.17 | RoyK | they need it for all those 0900 cdalls..... |
| 08:00.28 | modulus_ | tainted, you buy g729 license? |
| 08:00.42 | ta[i]nted | yea |
| 08:00.54 | modulus_ | how's the quality? |
| 08:01.02 | ta[i]nted | dunno yet |
| 08:01.31 | ta[i]nted | will let u know in a couple of days |
| 08:12.50 | *** join/#asterisk Andrey_Kirov (~Andrey_Ki@16-51-customer.kirov.mtsnet.ru) |
| 08:12.50 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 08:25.02 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 08:27.05 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 08:27.15 | mutilator | O_o |
| 08:27.20 | mutilator | mornin all |
| 08:27.32 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:28.31 | Makenshi | morning |
| 08:32.08 | Delvar | morning |
| 08:35.42 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) |
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| 08:50.01 | *** join/#asterisk brc-tux (~brc-tux@pD9E98EA7.dip0.t-ipconnect.de) |
| 08:50.18 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:51.40 | Makenshi | are there any known issues running asterisk on x86_64 cpus? |
| 08:51.51 | RoyK | not that I know of |
| 08:52.28 | *** join/#asterisk jojoba (~jojoba@220.248.36.42) |
| 08:53.23 | jojoba | hi, could any one help me? |
| 08:53.48 | RoyK | I'm using several servers with intel em64t (athlon64 licenced stuff) |
| 08:54.27 | Makenshi | yip, same here |
| 08:54.48 | Makenshi | just that i replaced my * server, and theres some strange behaviour going on |
| 08:54.56 | Makenshi | i installed lates stable yesterday |
| 08:54.59 | RoyK | what? |
| 08:55.04 | Makenshi | *latest |
| 08:55.04 | RoyK | what strange stuff? |
| 08:55.22 | Makenshi | with sip calling, a lot of inbound and outband calls are failing |
| 08:55.28 | Makenshi | *outbound |
| 08:55.37 | RoyK | what does sip debug say? |
| 08:55.41 | RoyK | pastebin it..... |
| 08:58.42 | Makenshi | come on pastebin.. |
| 09:01.07 | Makenshi | while im waiting for that |
| 09:01.21 | Makenshi | :) |
| 09:01.26 | Supaplex | hehe :) |
| 09:01.32 | Supaplex | just about eh? |
| 09:01.37 | Makenshi | the server is multihomed, one public, one private |
| 09:01.43 | Supaplex | reminds me of an old, but good tagline. |
| 09:01.56 | Makenshi | i configured sip.conf just to listen on the public network facing interface |
| 09:02.07 | Supaplex | <PROTECTED> |
| 09:02.22 | Makenshi | i did a packet capture and its showing packets from, and to(!) it's private facing interface |
| 09:02.36 | Makenshi | from my sip provider |
| 09:05.23 | Makenshi | http://pastebin.ca/8042 and http://pastebin.ca/8043 |
| 09:08.57 | RoyK | better paste the whole setup and teardown |
| 09:11.48 | p1tst0p | -- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98 |
| 09:11.52 | Makenshi | it seems like it's trying to register with the first interface only, which is on a private network |
| 09:11.58 | p1tst0p | anyone know how i can resolve that ^ |
| 09:12.08 | Makenshi | even though i told it to bind to the address of the public facing interface |
| 09:12.28 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 09:12.44 | p1tst0p | i am using a Avaya 4602 phone.. and i keep seeing that, which in turn render's the phone un usable after a few minute's. |
| 09:15.54 | DannyF | morning folks |
| 09:16.45 | RoyK | Makenshi: why multihomed? |
| 09:16.49 | Makenshi | RoyK, is there a configuration directive to specify what ip address asterisk should use for registering with other sip proxies? |
| 09:17.09 | Makenshi | RoyK, authentication services are only available on the private network |
| 09:17.18 | Makenshi | (ie, ldap, radius) |
| 09:17.23 | RoyK | ok |
| 09:18.45 | *** join/#asterisk ckruetze (~nospam@i3ED61FB8.versanet.de) |
| 09:19.38 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@wm20hb.34.ADSL.NetSurf.Net) |
| 09:20.11 | SPoon_TSX | Hello everyone out there.I got a quick questions. Do I need to have a Sound Card in order to make MusicOnHold work? |
| 09:20.22 | Makenshi | aha |
| 09:20.25 | Makenshi | i think i know the problem |
| 09:20.28 | Makenshi | (kills self) |
| 09:20.34 | RoyK | Makenshi: what? |
| 09:20.56 | Makenshi | i havent changed the default route from the internatal gateway >< |
| 09:21.01 | RaYmAn-Bx | <PROTECTED> |
| 09:21.11 | RoyK | Makenshi: idiot! |
| 09:21.14 | Makenshi | yip! |
| 09:21.37 | Makenshi | got there in the end |
| 09:21.44 | RoyK | default gateway: wgere tge other guys are. local networks: rip/ospf/static routes |
| 09:22.22 | SPoon_TSX | RaYmAn-Bx: Thanks. Then I think I might not have the mpg123 install properly then. Since everytime when I get the channel on hold, it say something like starting the music on hold but it will just show the message that it was stopped in 1 second. |
| 09:22.58 | Makenshi | well, its internal interface is running dhcp |
| 09:23.11 | Makenshi | no excuse though |
| 09:23.49 | SPoon_TSX | btw, I am just wondering is it possible to make multipule SIP client as a group? Just like how we did on Zap Channel where you can assign a group? |
| 09:24.23 | RaYmAn-Bx | SPoon_TSX: you might want to look at making sure you have the right version of mpg123 |
| 09:24.42 | Makenshi | that explains all the problems :) |
| 09:26.31 | RoyK | Makenshi: servers on dhcp? idiot! |
| 09:26.32 | RoyK | :) |
| 09:26.36 | SPoon_TSX | oic |
| 09:27.23 | Makenshi | RoyK, yes, static addresses set using dhcp |
| 09:28.46 | Makenshi | very useful to keep configuration parameters up to date |
| 09:28.54 | Makenshi | rather than having to change it on a whole bunch of servers |
| 09:29.13 | *** join/#asterisk zhier (~nick@219.136.15.39) |
| 09:29.17 | *** part/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 09:29.33 | RoyK | sure |
| 09:29.35 | SPoon_TSX | Hello everyone, just wondering can I assign SIP client to a group? |
| 09:29.50 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:31.02 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 09:31.03 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 09:31.58 | SPoon_TSX | ...? |
| 09:36.42 | *** join/#asterisk CP-Alex (~sales@220.227.236.3) |
| 09:37.01 | *** part/#asterisk brc-tux (~brc-tux@pD9E98EA7.dip0.t-ipconnect.de) |
| 09:37.05 | CP-Alex | Hello All |
| 09:37.11 | RoyK | SPoon_TSX: ..--.. |
| 09:38.09 | SPoon_TSX | RoyK: hahaa.. btw, do you know if I can group multipules SIP extension and do something lke a rollover extension list? |
| 09:38.22 | CP-Alex | can anyone tell me what is the integer based amaflags? my system show 3 in mysqldb. |
| 09:38.39 | *** join/#asterisk meppl (~mephisto@p3E9E2B95.dip.t-dialin.net) |
| 09:38.42 | meppl | guten morgen |
| 09:39.47 | RoyK | Guten Morgen, herr meppl |
| 09:39.54 | meppl | guten morgen royk |
| 09:40.09 | RoyK | SPoon_TSX: I don't understand what you're trying to do...... |
| 09:41.29 | langals | Hi there. When I run asterisk in the console I get the following error: "WARNING[1670]: chan_oss.c:269 sound_thread: Read error on sound device: Resource temporarily unavailable". This is repeated about 10 times. Does anyone have any idea what this is about? |
| 09:41.36 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 09:41.54 | SPoon_TSX | RoyK: okay. What I want is I want to ring a group of extension which is a SIP Client, if the 1st one is not picking up the call, it will try to ring the second one. |
| 09:42.10 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 09:42.51 | RoyK | SPoon_TSX: use a queue :) |
| 09:43.47 | Andrey_Kirov | langals: Your sound card used by over process (artsd for example) |
| 09:44.24 | SPoon_TSX | RoyK: Queue? Is it a Dial Cmd? |
| 09:44.35 | RoyK | show application queue |
| 09:44.44 | RoyK | see queues.conf |
| 09:45.18 | langals | Andrey_Kirov: So is it not really a problem then? |
| 09:46.55 | jojoba | Is anyone know why asterisk return SIP 200OK message immdediatly after dial FXO channel? |
| 09:47.08 | jojoba | I configured asterisk to tranfer all SIP call to FXO channel |
| 09:48.24 | Druken | because fxo picks up the right after dial... unless you have callprogress=yes set |
| 09:48.37 | Druken | and even then, call progress isn't the greastest... |
| 09:49.05 | Fraeggl | hi all |
| 09:49.23 | Fraeggl | do i have to always register sip phones ? |
| 09:49.24 | Druken | or, let me rephrase that... FXO is considered anwered immediately after dial |
| 09:49.26 | jojoba | thanks, let me try |
| 09:49.28 | Fraeggl | even if i use staic ips ? |
| 09:50.07 | jojoba | realy? |
| 09:50.12 | langals | Would anyone be able to help me get Meetme working? I am having a bit of a problem |
| 09:50.12 | Druken | Fraeggl: all my phones register on their own... and they are all on statics |
| 09:50.58 | Druken | langals: do you have a digium card in your server? |
| 09:51.20 | Fraeggl | Druken: so, show ip registry is not empty ? |
| 09:51.49 | langals | no - I am trying to use Ztdummy |
| 09:52.03 | Druken | langals: k :) just checking :) |
| 09:52.22 | langals | I installed the Zaptel package after I installed the Asterisk package - is this a problem |
| 09:52.37 | Druken | Fraeggl: my registry's are empty, but that is because i use realtime |
| 09:53.03 | Druken | langals: yes.. you have to rebuilt asterisk with zaptel, i belive |
| 09:53.18 | Andrey_Kirov | langals: No |
| 09:53.20 | Fraeggl | Druken: how can i know then the phone registered sucesfully ? is it enough if i shows in show peers ? |
| 09:53.44 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:53.45 | Andrey_Kirov | langals: I installed Zaptel after Asterisk and i hav no broblem |
| 09:53.45 | langals | I will show you the error I get..... |
| 09:53.47 | Druken | Fraeggl: how many phones are you talking? |
| 09:53.48 | christo | morning all |
| 09:54.19 | Fraeggl | Druken: just 2, with static ips... should be a test-setup, but cant get it working... |
| 09:54.27 | langals | I can dial between 2 softphones no problem, but when I try and dial into a conference I get the follow error (from SIP debug)..... |
| 09:54.50 | Fraeggl | Druken: they appear in sip show peers, but i cant dial each other (i think the dialplanshould be ok) |
| 09:55.12 | langals | pbx.c:1291 pbx_extension_helper: No application 'Meetme' for extension (from-sip, 1234, 1) |
| 09:55.21 | Druken | Fraeggl: if they are in sip peers, then they registered |
| 09:55.30 | christo | I broke something quite beautifully yesterday. Now I can't get one of our * servers to dial another. The system attempts to set up a sip channel, then dies with "chan_sip.c:6811 handle_response: Forbidden - wrong password on authentication for INVITE.." Does anybody recognise that error message? |
| 09:55.37 | Druken | langals: modules.conf |
| 09:55.44 | christo | perhaps it's the way the users/peers are set up in the sip.conf at either end? |
| 09:56.16 | langals | what [...] in extensions should the conference extension be under - is [from-sip] correct? |
| 09:56.28 | Fraeggl | Druken: but if i do a "sip show peer 21" e.g. the Fucc Contact field stays empty, could this be the problem ? |
| 09:56.40 | Andrey_Kirov | langals: You don't load app_meetme module |
| 09:57.02 | langals | When and how should I load this module? |
| 09:57.08 | Fraeggl | Druken: i try to dial them with "Dial(SIP/21)" in the dialplan.. |
| 09:57.19 | Andrey_Kirov | langals: Promt in your asterisk console "show modules" |
| 09:57.58 | langals | what is the meetme module called? |
| 09:58.22 | Andrey_Kirov | langals:check file /usr/lib/asterisk/modules/app_meetme.so |
| 09:58.48 | Druken | app_meetme.so MeetMe conference bridge 0 |
| 09:58.54 | Druken | that's what yer missing :) |
| 10:00.05 | Druken | can someone remind me why the hell i'm up at 5am ? |
| 10:00.32 | Pj386 | Druken: 'cause at 4am you said "Hmm... coffee" |
| 10:00.43 | Druken | nah.... |
| 10:00.47 | langals | I cannot seem to find app_meetme.so in that folder |
| 10:01.04 | Fraeggl | Druken: thx for your help so far, could you please send me yous sip.conf and perhaps the output of "sip show peer <onephone>" ? |
| 10:01.19 | Druken | i think it was more because the wife woke my ass up at 4am, getting ready for work... |
| 10:01.45 | Druken | Fraeggl: as i told you, my sip show peer doesn't show anything |
| 10:01.54 | Druken | ad my sip.conf is blank |
| 10:01.58 | Andrey_Kirov | langals: Can you find this file in you asterisk source directory? |
| 10:02.03 | Druken | i use realtime for my sip |
| 10:02.12 | langals | will look for it |
| 10:02.25 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 10:02.27 | Fraeggl | Druken: realtime, ok, will look out for that, thx |
| 10:03.25 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 10:03.42 | Andrey_Kirov | Can I add args to the macro in asterisk 1.07? |
| 10:03.50 | Andrey_Kirov | or only cvs version? |
| 10:06.43 | langals | Andrey_Kirov: I cannot seem to locate that file anywhere |
| 10:08.06 | Andrey_Kirov | langals: Your asterisk compiled without this module |
| 10:09.26 | *** join/#asterisk montag___ (~montag@nat-psv.sssup.it) |
| 10:09.55 | Andrey_Kirov | langals: hmm, i look at MakeFile at /<asterisk source>/apps |
| 10:10.06 | montag___ | i've buyed a Voismart Sip Phone 302 (a rebranded Planet VIP-152T), i've a proble during call, sometime the phone do a reset, any tips ? |
| 10:10.10 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 10:10.19 | *** join/#asterisk Madd0 (madd0@m31.net81-65-66.noos.fr) |
| 10:10.49 | Andrey_Kirov | langals: This module can't compile without zaptel :) sorry |
| 10:11.07 | Andrey_Kirov | langals: just rebuild asterisk now |
| 10:12.40 | langals | Andrey_Kirov: Sorry, I am fairly new to linux. To recompile Asterisk do I just do the same as initial compile / install? |
| 10:13.43 | langals | As in "make clean; make install"? |
| 10:13.55 | RoyK | make clean install |
| 10:13.56 | RoyK | :P |
| 10:14.29 | langals | thanks |
| 10:15.05 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 10:16.18 | p1tst0p | -- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98 <-- i get this if i add "Mailbox=" on my extention, the phone is an Avaya 4602 IP phone, with SIP image. |
| 10:18.22 | p1tst0p | after a couple of mins, phone refuses to work ';) im new to Asterisk, and i can fix it by removing the "Mailbox=", but i want to have voicemail, what will removing "Mailbox=" not allow me too do ? |
| 10:18.49 | *** join/#asterisk gst (~gst@wireless.sysfrog.org) |
| 10:21.59 | Mavvie | http://telephonyonline.com/finance/news/nominum_enum_ip_032205/ <- slightly interesting for here. |
| 10:23.42 | *** join/#asterisk __a (user@193.140.215.2) |
| 10:24.14 | pratik | hello everyone i m still facing problem with my incoming FWD |
| 10:24.32 | pratik | the out going calls is not a problem |
| 10:24.37 | __a | i'm writing an asterisk app, and would like to dial a number within the application |
| 10:24.54 | __a | i'm using ast_spawn_extension but that overwrites my apps CDR |
| 10:25.17 | __a | any idea what to use to have a new CDR for the outgoing call from within the app? |
| 10:25.49 | PoWeRKiLL | __a forkcdr |
| 10:26.39 | pratik | i have checked my extensions.conf and the iax.conf many times but still i am not able tyo figure otu the problem |
| 10:30.56 | langals | Andrey_Kirov: I recompiled and now app_meetme.so is installed. But still throwing a whole lot of errors |
| 10:33.14 | pratik | any clue why is it not working |
| 10:38.48 | Madd0 | hi, what kind of hardware would I need if I want the phone to ring in one location, then forward the call over the Internet to answer using a softphone (or VoIP phone) at another location? |
| 10:39.01 | Madd0 | I'm new to voip and I'm just starting to understand how this works... |
| 10:40.00 | pratik | well any FWD experts here |
| 10:44.24 | Andrey_Kirov | langals: What kind of error> |
| 10:44.25 | Andrey_Kirov | ? |
| 10:47.43 | pascals | What I am doing wrong with Record(/tmp/new:gsm); Wait,1; Playback(/tmp/new) - sometimes it records, sometimes it leaves an empty file. |
| 10:48.31 | pascals | Actually, it is about 50/50 |
| 10:48.35 | langals | Sorry - app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device |
| 10:49.31 | langals | and "chan_zap.c:763 zt_open: Unable to open '/dev/zap/pseudo': Nos such device |
| 10:49.33 | pratik | pascals:tell me one thing can i record and listen to the calls whcih i make through asterisk |
| 10:50.08 | langals | ..and chan_zap.c:6700 chandup: Unable to dup channel: No such device.... |
| 10:50.30 | langals | The chan_zap errors come before the app_meetme error |
| 10:50.34 | pascals | pratik: you should be able to, use Monitor() |
| 10:50.54 | pratik | i used it in the extensions.conf |
| 10:51.22 | pascals | pratik: I don't know - haven't played with it yet |
| 10:51.35 | christo | when should I use 'restart now' and when should I just use 'reload' ? |
| 10:51.36 | pratik | i have a sipura phone attached with my asterisk , and i ewantr to listen to the calls which i make through it |
| 10:55.44 | christo | guys, if I change my dialplan and sip.conf and other bits and pieces, should I just reload asterisk, or actually restart it with 'restart now' |
| 10:55.55 | pascals | reload |
| 10:56.13 | christo | when is 'restart now' required? |
| 10:56.48 | pascals | when the manual says it is ;-) |
| 10:58.42 | Andrey_Kirov | langals: zaptel driver is properly installed? |
| 10:59.38 | Andrey_Kirov | langals: lsmod | grep zaptel |
| 11:00.48 | langals | where must I execute that command? Does not seem to work |
| 11:00.52 | christo | pascals - aye |
| 11:01.04 | langals | I don't have any Zaptel software, so am trying to use ZtDummy |
| 11:01.26 | langals | Would i still need the Zaptel driver? |
| 11:01.33 | Andrey_Kirov | langals: ZtDummy only? |
| 11:01.34 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
| 11:02.12 | langals | yes - i believe that is possible |
| 11:03.46 | langals | basically that command you gave me returns nothing |
| 11:04.07 | Andrey_Kirov | langals: that's right |
| 11:04.28 | SexyKen | Hey guys - I need a good way to run multiple companies from a single Asterisk server. |
| 11:04.49 | Mavvie | SexyKen: jail them |
| 11:05.12 | SexyKen | For intance, Company A, B and C may all have an extenion 200, but each one will go to a different person (phone). |
| 11:05.27 | Mavvie | SexyKen: properly context them |
| 11:05.33 | SexyKen | Mavvie - Jail them? Then the ports get fubarred. I want a single Asterisk system. |
| 11:05.54 | SexyKen | Mavvie - Are there any documents out there that would walk through this? |
| 11:06.04 | Mavvie | SexyKen: none which I can give you. |
| 11:06.29 | Andrey_Kirov | langals: hmm. I am not sure what it work only with ztdummy |
| 11:06.42 | Andrey_Kirov | langals: ztdummy module is loaded? |
| 11:06.55 | langals | mmm....will look |
| 11:07.25 | langals | Do you know what the ztdummy module is called? |
| 11:07.29 | Makenshi | you need zaptel to use ztdummy |
| 11:07.38 | Makenshi | the module is called ztdummy |
| 11:07.44 | Makenshi | lsmod should show something like this.. |
| 11:07.56 | Makenshi | Module Size Used by |
| 11:07.56 | Makenshi | ztdummy 5472 0 |
| 11:07.56 | Makenshi | zaptel 198568 5 ztdummy |
| 11:08.18 | langals | Where do I specify that the module should be loaded? |
| 11:08.23 | Makenshi | you will need to alter the makefile when building zaptel because ztdummy is not built by default |
| 11:08.39 | Makenshi | langals, it depends upon your system |
| 11:08.40 | langals | I did that - I uncomment #ztdummy |
| 11:08.46 | Makenshi | mine is /etc/modules.conf |
| 11:09.34 | Makenshi | could be /etc/modprobe.conf |
| 11:09.38 | Makenshi | or you can load it by hand also |
| 11:09.52 | Makenshi | i use the init script (zaptel.init) |
| 11:09.56 | langals | found modules.conf in asterisk folder |
| 11:10.36 | Makenshi | not that one :) |
| 11:11.14 | langals | found /etc/modules.conf |
| 11:11.24 | Andrey_Kirov | langals: make ztdummy.o, make install, modprobe ztdummy |
| 11:11.43 | langals | has a line at the bottom - post-install ztdummy /sbin/ztcfg |
| 11:12.28 | christo | guys, I'm trying to work out the exact difference between 'restart now' and 'reload' at the CLI. I can't find this in the docs somehow. I want to know more than just 'reload reloads configs' and 'restart restarts asterisk'. |
| 11:12.29 | langals | must I put that in the the modules.conf file?, or execute from the command line? |
| 11:13.49 | Delvar | christo: restart cills all calls, reload doesnt |
| 11:13.53 | Andrey_Kirov | langals: execute from command line |
| 11:13.56 | Delvar | cills = kills |
| 11:14.07 | langals | should I be in the Zaptel directory? |
| 11:14.11 | Andrey_Kirov | yes |
| 11:15.26 | christo | Delvar - mmmkay |
| 11:15.28 | *** join/#asterisk tandrews (~tandrews@mail.grok.co.za) |
| 11:15.51 | christo | Delvar - does reload not reset the zap channels? |
| 11:16.16 | tandrews | hi * |
| 11:16.42 | Delvar | christo: as far as i know, it doesn't. to change config on zaptel you usualy have to restart |
| 11:17.09 | Druken | christo: yes, to reset zaptel configs you must restart |
| 11:17.29 | pratik | can any one help me out with my FWD issue |
| 11:18.57 | langals | when I go go - modprobe ztdummy it comes up with "/lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters |
| 11:19.57 | Druken | langals: let me give ya a lil advice, go on ebay, and get a 6 dollar FXO card, and use it simply for a timing device |
| 11:20.08 | Druken | i personally don't trust ztdummy :) |
| 11:20.52 | langals | is that the Zaptel card? Is it a network card? |
| 11:21.02 | Druken | it's a zaptel card |
| 11:21.49 | Druken | http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&category=61841&item=5762599065&rd=1 |
| 11:21.57 | Druken | there ya go, $1.00 :) |
| 11:21.59 | Druken | hehehe |
| 11:22.03 | SexyKen | No one knows of any good gui's for asterisk? |
| 11:22.39 | Delvar | SexyKen: 'good'? i know of a couple |
| 11:23.30 | Druken | define gui :) |
| 11:23.39 | Delvar | hehe |
| 11:23.44 | Druken | there are soo many things you may want to see |
| 11:23.57 | Delvar | well there is http://sourceforge.net/projects/amportal/ , its free, but looks crap, never used it properly tho |
| 11:24.13 | Druken | i use FOP :) |
| 11:24.16 | pratik | can any one help me out with my FWD issue |
| 11:25.13 | Delvar | then there is www.bicom.us, NOT free, looks a lot better, works ok, totaly propriatry and everything is done via AGI with very little you can do via modifying extensions.conf |
| 11:25.19 | mstocco | SexyKen: or you can do what I did and write your own |
| 11:25.47 | SexyKen | mstocco - What does your GUI do? |
| 11:26.08 | SexyKen | Delvar - Yea -- Bicom SUCKS. |
| 11:26.25 | pratik | mstocco:do you have your astguiclient set up |
| 11:26.27 | mstocco | SexyKen: manages callcenters |
| 11:26.40 | SexyKen | mstocco, Does it run on static or realtime? |
| 11:27.31 | mstocco | pratik: I wrote mine in coldfusion and yes, I have two centers running right now |
| 11:28.00 | mstocco | SexyKen: from asterisk's point of view it is static |
| 11:28.05 | *** join/#asterisk memic (~memic@chicago089.server4free.de) |
| 11:28.08 | pratik | wow thats great |
| 11:28.17 | SexyKen | mstocco, I see - but you use AGI's to make it database based? |
| 11:28.46 | mstocco | pratik: one in Hollywood Florida and one in Beverly Hills California |
| 11:28.57 | ennuyeux72 | SexyKey: what sort of Bicom problems have u come across |
| 11:28.59 | pratik | mstocco:i have almost setup the astguiclient but then i am not sure how to proceed |
| 11:29.57 | pratik | i have proceeded from the site astguiclient.sourceforge.net/scratch_install |
| 11:29.58 | mstocco | SexyKen: I use two AGI scripts written in python |
| 11:30.00 | SexyKen | mstocco, Does your syste run multile companies under one instance of asterisk? For instance, 3 companies can have an extension '200' but depending on what company it's assigned to, it'll ring a different person. |
| 11:30.43 | mstocco | SexyKen: If my customer wants something like that I would set it up , yes |
| 11:31.17 | pratik | mstocco:well can u set it up for me, |
| 11:31.25 | SexyKen | ennuyeux72, Bicom charged me an outrageous amount of money then gave me a client that was based on the manager api and had to run off their servers, then they installed a local version. It crashed my server bi-hourly and didn't do havlf the shit it was supposed to. |
| 11:31.56 | pratik | i have got ill i get the screen where i can add the phone numbers , add a server and all |
| 11:32.17 | SexyKen | mstocco, How quick could you get a working system setup that will host 3 companies and would support 'extension roaming' and have a decent cdr viewer and user management gui. |
| 11:32.18 | mstocco | pratik: what I wrote is not based off of astguiclient |
| 11:32.28 | pratik | ok |
| 11:33.13 | pratik | bcos what i wanted was that we have a server in UK and i eanted to connect that server to asterisk |
| 11:33.26 | Druken | all my shit is custom dialplan stuff, everything taken from the database.. but that's because i hate changing the dialplan :) |
| 11:34.40 | SexyKen | mstocco, you there? |
| 11:34.44 | mstocco | SexyKen: define user management? |
| 11:34.59 | mstocco | SexyKen: South Florida |
| 11:35.37 | SexyKen | Add Extensions to selelcted companies, remove them, edit caller id per extenson, assign user a agent#, edit call forwarding or call abilities etc. |
| 11:36.05 | SexyKen | And edit a callers ability to dial out, limit to certain extensions or only local within certain area codes etc. |
| 11:37.01 | SexyKen | mstocco, seem like a lot? |
| 11:37.25 | Druken | i would say that would take a lil to make.. hehe |
| 11:37.26 | mstocco | SexyKen: not really |
| 11:37.34 | Druken | do-able.. |
| 11:37.39 | mstocco | yup |
| 11:37.49 | SexyKen | mstocco, How much of it would you consider actually already done? |
| 11:37.58 | *** join/#asterisk vaewynAFK (freeman@mail.deltamach.com) |
| 11:38.28 | ennuyeux72 | SexyKen: that doesn't sound like a pleasurable experience |
| 11:39.06 | SexyKen | ennuyeux, Yes, we went through about 30 days of trying to get all the bugs fixed and they just fucked shit up more and more so I requested a refund. |
| 11:39.09 | SexyKen | They denied. |
| 11:39.11 | mstocco | no, actually most of it it is done, I would have to tweak it to work with his multi-company idea |
| 11:39.12 | vaewynAFK | anyone have their zaptel.conf hanging around for NI2? |
| 11:39.18 | SexyKen | And their reason is 'we delivered the product' |
| 11:39.48 | SexyKen | mstocco, Do you have time in your schedule to do this? Quote me if you will :-) |
| 11:40.38 | vaewyn | arggh... setting up these T cards is the first time I have hit soething in * that is IMO grossly underdocumented :} |
| 11:41.01 | mstocco | SexyKen: not much in the way of time this week but... |
| 11:41.02 | Druken | SexyKen: wouldn't you have an onsite server to host a companies internal PBX ? that could end up being an awefull lot of bandwidth otherwise |
| 11:41.45 | SexyKen | Druken - It's a remotely housed server and we do about 3-4k/minutes a week right now...if not more....and it works fine. |
| 11:41.50 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 11:41.50 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:42.04 | Makenshi | Druken, the audio wouldn't pass through the remote server |
| 11:42.21 | Makenshi | it would only be used for signalling |
| 11:42.29 | Makenshi | unless you dialed out through the remote server |
| 11:42.30 | Druken | Makenshi: that all depends if you have reinvite enabled |
| 11:42.40 | Makenshi | Drunken, true |
| 11:42.47 | Druken | and if the two phones can communicate to each other |
| 11:43.14 | Makenshi | i think a stateful sip proxy on the network would do the trick |
| 11:44.01 | Druken | probably |
| 11:44.03 | Makenshi | if only someone would include linksys's ip_conntrack_sip module into the kernel.. |
| 11:44.17 | Makenshi | i haven't gotten around to trying it myself |
| 11:44.27 | Makenshi | (linux kernel that is) |
| 11:45.37 | Druken | why do i think there's a missing t at the end of Makenshi ? :) |
| 11:47.15 | mikegrb | Druken: because there is? |
| 11:48.16 | Druken | :) |
| 11:48.28 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:55.25 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 11:56.49 | *** join/#asterisk pratik (~pratik@202.149.48.210) |
| 11:57.21 | Makenshi | Druken, i have no idea |
| 11:57.23 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 11:57.43 | Makenshi | But if the irc server supported unicode i could use the proper characters :) |
| 11:58.08 | Druken | hehe |
| 11:58.14 | jakepdev | is there a link that shows how to update libpri? I d/l the gz and extracted it - make clean/make install... what else? |
| 11:58.21 | Druken | not many will allow you to use unicode |
| 12:00.33 | jakepdev | druken - did you install * from CVS? |
| 12:04.50 | RoyK | æøåß??ß |
| 12:07.23 | Druken | jakepdev: yes.. a very long time ago :) |
| 12:07.46 | jakepdev | druken - np |
| 12:07.57 | Druken | why do you ask ? |
| 12:08.14 | jakepdev | just can't figure out how to get the new libpri in there |
| 12:08.30 | Druken | RoyK: alot of people think it's missing... but in reality it's not true |
| 12:08.31 | Druken | :) |
| 12:08.36 | jakepdev | i did - make clean - make install - what else is there? |
| 12:09.05 | Druken | i can't see anything else, cept maybe rebuilt asterisk it'self ? |
| 12:09.30 | Druken | or even rebuild it.. hehe damn brain of mine never works right |
| 12:09.39 | jakepdev | that's just it - when I go to build *, it tells me i need a newer libpri |
| 12:09.59 | Druken | that intresting |
| 12:12.32 | jakepdev | is there a command that will tell me if the new libpri actually got loaded? |
| 12:12.59 | Druken | dunno |
| 12:14.04 | *** join/#asterisk n1gg4s (~bruno@200.236.162.2) |
| 12:16.12 | n1gg4s | I'm having trouble transfering data between kphone clients |
| 12:16.59 | n1gg4s | I get the error message "Call Failed: Not found" |
| 12:17.04 | n1gg4s | could anyone give me a hand? |
| 12:19.16 | *** join/#asterisk skrusty (muad@217.79.111.73) |
| 12:19.26 | skrusty | hi |
| 12:28.41 | *** join/#asterisk Muy_Loco (~muy_loco6@rrcs-24-73-107-138.se.biz.rr.com) |
| 12:30.26 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 12:31.02 | *** join/#asterisk shamid (~shamid@pk-isb-trg-sc01-019.speedcast.com) |
| 12:31.17 | shamid | hi everyone |
| 12:31.45 | shamid | anyone know how to configure dailup modem with asterisk as FXO/FXS |
| 12:32.36 | Druken | uhmm... you can't? |
| 12:33.22 | shamid | becuase i know one person is doing that, but he is selling this solution |
| 12:33.37 | shamid | i want if someone in this group ever tried to do that |
| 12:34.49 | Druken | a MODEM can only be used if it's a certain type |
| 12:35.06 | Druken | and it's not used in a modem sence, it's done with ZAPTEL drivers |
| 12:35.48 | shamid | yes, but how to configure a dialup modem as a ZAPTEL drivers |
| 12:35.53 | Druken | and can only be FXO, unless you purchase a TDM card from digium with FXS modules |
| 12:36.18 | Druken | ~wiki |
| 12:36.20 | shamid | yes, i want to test it as FXO, if you can help me please to |
| 12:36.38 | shamid | i am using US Robotics Voice and Fax modem |
| 12:37.08 | shamid | can you forward some usefull link which guide me to do this |
| 12:37.11 | Druken | your not listening.. a generic modem cannot be used, it must be a certain type |
| 12:37.25 | n1gg4s | anyone know how to configure SIP with asterisk ? |
| 12:37.55 | Druken | shamid: look on ebay for a X100p card |
| 12:38.02 | Druken | you can get them for like 10 bux... |
| 12:38.19 | RoyK | that's the copies |
| 12:38.26 | RoyK | not the Real Ones From Digium |
| 12:39.08 | shamid | yes, Druken i really thankful to you for this but i want to try modem, if you can help me to configure modem as FXO |
| 12:39.20 | Druken | RoyK: he's obviously a cheap ass, so i figure they are better for him :) |
| 12:39.46 | RoyK | CONFIG_PRINTK_TIME=y |
| 12:39.50 | RoyK | new kernel option :) |
| 12:40.13 | Druken | w00t!... wuts it do? hahahahaha |
| 12:41.10 | jakepdev | ~help |
| 12:43.01 | p1tst0p | -- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98 <-- anyone know why i get this if i add "Mailbox=" on my extention, the phone is an Avaya 4602 IP phone, with SIP image. |
| 12:44.05 | p1tst0p | then after a couple of mins, phone refuses to work ';) im new to Asterisk, and i can fix it by removing the "Mailbox=", but i want to have voicemail, what will removing "Mailbox=" not allow me too do ? |
| 12:44.16 | Druken | p1tst0p: the mailbox= line is just for the voicemail light on your phone |
| 12:44.53 | Druken | it won't remove any functionaity except the little light flashing when you have voicemail.. |
| 12:45.04 | Zeeek | make sure the line reads mailbox=name@email_context |
| 12:49.57 | *** join/#asterisk Hydroxide (user@Hydroxide.developer.debian) |
| 12:50.18 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 12:50.40 | Druken | Zeeek: wouldn't it default to the default if it's not set? |
| 12:51.00 | Zeeek | why not use the precise form, that way no questions later? |
| 12:51.08 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 12:51.09 | Druken | very true |
| 12:51.26 | Zeeek | mine are never in '"default" |
| 12:51.35 | Druken | i should know better |
| 12:52.01 | Druken | yeah, i run like 3 business pbx's on my system, all with seperate mailboxes :) |
| 12:52.52 | Darwin35 | how many users on each pbx ? |
| 12:53.20 | Darwin35 | and what type of server |
| 12:54.16 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 12:54.22 | p1tst0p | Druken, cool, is there a way to add a tone when u pick the phone up, to say theres a message, instead of a light ? i think the mailbox= is an issue with this specific phone you see |
| 12:55.21 | Druken | p1tst0p: i don't think so.. not without the mailbox= line... even with FXS ports it flashes the light, and gives a studder tone |
| 12:58.59 | Muy_Loco | excuse me, anyone know why asterisk would give a "Rejected connect attempt from 192.168.10.106, request '1234567@outgoing' does not exist" when I try to make an outgoing call (incoming works though) |
| 12:59.26 | Druken | because your dialplan is not setup properly |
| 12:59.31 | Muy_Loco | oh.... |
| 12:59.36 | Muy_Loco | sounds about right... |
| 12:59.40 | Druken | go over your outgoing context and fix it :) |
| 12:59.47 | Muy_Loco | lol, thanx |
| 13:00.09 | Muy_Loco | I used the sample files provided by voicepulse connect! and they work halfway, lol |
| 13:00.31 | *** join/#asterisk tessier (~treed@210.245.100.18) |
| 13:00.37 | Druken | use them as guidelines :) |
| 13:00.47 | Muy_Loco | oh ok... will do... |
| 13:03.09 | p1tst0p | Druken, hmm i may have to leave Mailbox indication out for this type of phone then, checked the Wiki, and someone else experienced the same issue as i described when adding mailbox indicator to this phone type ;~( |
| 13:06.08 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 13:09.49 | Muy_Loco | heh.. anyone wanna help this complete newb (me) write a dial plan? I just need to see if I can dial out to a land line... please? |
| 13:13.55 | pascals | can I make asterisk do something when a file exists and something else if it doesn't? |
| 13:14.25 | pascals | There doesn't seem to be a 'file exists' test |
| 13:16.22 | *** join/#asterisk denon (denon@synapse.subneural.net) [NETSPLIT VICTIM] |
| 13:16.22 | *** mode/#asterisk [+o denon] by ChanServ |
| 13:16.24 | *** join/#asterisk Hydroxide (user@Hydroxide.developer.debian) |
| 13:16.50 | n1gg4s | to place kphone (sip) functioning I need to only modify sip.conf? |
| 13:17.00 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk __a (user@193.140.215.2) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk fenlander (~irc@82.152.81.57) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk YoYo (YoYo@dilbert.psknet.com) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk cftbl (hector@dipsy.tch.org) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk roamer323 (~sing@67.71.60.211) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk ArkyLady (ArkyLady@h248.76.255.206.cable.htsp.cablelynx.com) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk astlog (astlog@cpe-24-58-84-250.twcny.res.rr.com) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) [NETSPLIT VICTIM] |
| 13:18.06 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) [NETSPLIT VICTIM] |
| 13:18.47 | Zeeek | Starter tutorial: |
| 13:18.47 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 13:18.47 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 13:18.47 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 13:18.47 | Zeeek | THE reference of the moment: |
| 13:18.48 | Zeeek | http://www.asteriskdocs.org |
| 13:20.33 | Alexi1 | * works fine with fedora 3 doen't it ? |
| 13:20.49 | Alexi1 | because fedora 3 uses a 2.6 |
| 13:21.11 | Alexi1 | and my boss says thet it's not possible :( |
| 13:22.40 | Zeeek | 2.6 has issues but it works. Look at the wiki and the mailing list |
| 13:23.02 | Zeeek | ~google asterisk fedora |
| 13:23.48 | Zeeek | pretty good shot since I wouldn't know fedora 3 from my aunt emma |
| 13:23.50 | jakepdev | pascals - try this - http://voip-info.org/wiki-Asterisk+cmd+System |
| 13:26.17 | *** join/#asterisk gonzo- (~gonzo@portacare.portaone.com) |
| 13:26.25 | Alexi1 | ok |
| 13:27.14 | *** join/#asterisk Xride (~xride@xforce.dk) |
| 13:27.41 | langals | Could someone help me get ZtDummy working for Meetme - I am trying to get conferencing working |
| 13:28.50 | *** join/#asterisk fugitivo (~ajf@201.255.109.193) |
| 13:29.06 | fugitivo | good morning |
| 13:30.00 | Xride | fugitivo: hello |
| 13:30.08 | Zeeek | it's 2:30 PM here |
| 13:30.40 | Xride | Zeeek: are you in europe? |
| 13:30.50 | Zeeek | yes |
| 13:31.00 | fugitivo | its 10:30am here :) |
| 13:31.43 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 13:32.01 | Xride | that gotta be something like greenland or far east canada |
| 13:32.13 | clive- | does anyone have any ideas why a native transfer wouldnt work in iax2? |
| 13:32.34 | Zeeek | well good morit's illegal to transfer natives these days |
| 13:33.17 | roamer323 | zeeek - but them natives carries sip and iax2 gateway passports |
| 13:33.27 | fugitivo | anyone is using zaptel drivers in gentoo with devfs? |
| 13:34.04 | fugitivo | oh, it doesnt work with devfs |
| 13:34.09 | fugitivo | but gentoo needs devfs |
| 13:34.34 | clive- | zeek...lol |
| 13:35.16 | Zeeek | why linux distros have cute Apple-like names? |
| 13:35.25 | Zeeek | fedora, my hat |
| 13:35.31 | trym | ass in a hat |
| 13:35.38 | trym | asshat |
| 13:36.21 | langals | does anyone know how I would check whether I have the UHCI USB controller that is need for Ztdummy? |
| 13:36.39 | roamer323 | langals - lsmod |
| 13:37.11 | roamer323 | ~lsmod |
| 13:37.46 | *** join/#asterisk NewSole (david@i216-58-19-5.avalonworks.net) |
| 13:37.49 | langals | UHCI USB is not listed there |
| 13:38.43 | rious | lspci |
| 13:39.21 | dwmw2_gone | The election is over... why are people still talking about the donkey hats? |
| 13:39.46 | dwmw2_gone | I thought it was just Democratic party merchandising? |
| 13:40.10 | langals | tried lspci - not listed there either |
| 13:40.11 | roamer323 | langals - do you see uchi or usbcore? if not - you may need to reconfig your kernel. |
| 13:40.24 | rious | langals: lspci will tell you about your hardware, you'll have to look it up to see if it is right, or just try loading the uhci module |
| 13:40.59 | rious | if you have usb in kernel, not module, that doesn't work right ? |
| 13:40.59 | jakepdev | I put in libpri 1.0.7, but * still says it needs a newer libpri to compile. any ideas? |
| 13:41.26 | rious | did you make install ? |
| 13:41.34 | jakepdev | make clean - make install |
| 13:41.45 | jakepdev | zaptel - goes in fine |
| 13:41.56 | langals | How would I try load the uhci module? |
| 13:42.10 | rious | which kernel ? |
| 13:42.22 | rious | 2.6 modprobe uhci-hcd |
| 13:42.48 | langals | I am using Redhat 9 |
| 13:43.48 | rious | langals:uname -a |
| 13:44.00 | jakepdev | ~lart RoyK |
| 13:44.11 | langals | 2.4.20-8 |
| 13:44.39 | fgravato | is it worth putting in Digium TDM card vs using Ztdummy |
| 13:44.50 | jakepdev | fg - depends on your app |
| 13:45.27 | rious | modprobe usb-uhci ? |
| 13:45.36 | p1tst0p | where do i specify how many rings before it cuts to VoiceMail ? |
| 13:46.13 | Dandan | I do not know |
| 13:47.08 | jakepdev | <p1tst0p> - look in extensions.conf |
| 13:47.15 | langals | came up with /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be casued by incorrect module parameters, including invalid IO or IRQ parameters |
| 13:48.33 | rious | that doesn't sound good |
| 13:48.40 | langals | no - it does not |
| 13:49.34 | jakepdev | <p1tst0p> - should be the timeout param in the Dial cmd |
| 13:52.21 | *** join/#asterisk hemant (hemant@61.0.57.40) |
| 13:53.03 | langals | rious - do you have any suggestion what I should do? |
| 13:53.18 | langals | Would it be best to get one of those cards? |
| 13:55.29 | dwmw2_gone | langals: do you actually have a UHCI controller in your system? |
| 13:56.12 | elriah | Hi all - what's the preferred way to handle call detail reporting? Postgres, mysql, etc... |
| 13:56.44 | RoyK | elriah: whatever suits your needs. |
| 13:56.56 | RoyK | elriah: mssql, if you have to |
| 13:57.12 | elriah | Do you do CDR? What db to you use? |
| 13:59.53 | RoyK | mysql |
| 14:01.33 | langals | dwmw2_gone - it doesn't seem as though I do have that controller on my system |
| 14:01.47 | dwmw2_gone | do you have an OHCI controller instead? |
| 14:02.04 | elriah | RoyK Did you have to add/compile the mysql libs or are you using the odbc stuff built in to *? |
| 14:02.06 | dwmw2_gone | a USB controller ought to be listed in the output of 'lspci' |
| 14:02.46 | langals | it does not seem to be |
| 14:03.49 | langals | OHCI does not seem to be listed there either |
| 14:04.31 | RoyK | elriah: I use the stuff from asterisk-addons |
| 14:04.40 | RoyK | elriah: why do you ask? they all work... |
| 14:05.07 | langals | The following is listed: Host bridge, PCI bridge, Ethernet Controller x 2, ISA bridge, IDE interface, Multimedia audio controller |
| 14:06.10 | dwmw2_gone | langals: that implies that you don't actually have USB. Are there USB sockets? |
| 14:06.22 | *** join/#asterisk fugitivo (~ajf@201.255.108.80) |
| 14:06.31 | p1tst0p | jakepdev, cheers dude, found that |
| 14:06.44 | jakepdev | <p1tst0p> - np |
| 14:06.50 | `Sauron | sauron@mordor:~> cat /proc/bus/usb/devices |
| 14:06.54 | fugitivo | langals: try lspci -vv to find your usb controller |
| 14:07.01 | `Sauron | That'll tell you if you have USB stuff |
| 14:07.09 | p1tst0p | is there a way i can dial into my voicemail from my extern number ? |
| 14:07.16 | fugitivo | langals: is it a VIA usb controller? |
| 14:07.25 | jakepdev | <p1tst0p> - from PSTN? |
| 14:07.26 | dwmw2_gone | p1tst0p: yes, if you set your dialplan up accordingly. |
| 14:07.30 | langals | Yes - there are - but they are not being used, so maybe the drivers are not installed |
| 14:07.46 | fugitivo | langals: is a VIA usb controller? |
| 14:08.01 | p1tst0p | well, im registered with SipGate.co.uk, they give you a PSTN number, which is pointed at my * box |
| 14:08.33 | langals | There is IDE interface: VIA Technologies, Inc. VT..... PIPC Bus Master IDE (rev 06) |
| 14:08.40 | elriah | RoyK: Well, I'm trying to keep from having to recompile just to add the mysql-libs. I'm using the debian 1.0.5 package and it's really stable and easy to deploy. |
| 14:09.00 | fugitivo | langals: you should get USB Controller: VIA Tech.... .... |
| 14:09.16 | `Sauron | hum |
| 14:09.17 | `Sauron | that's fun |
| 14:09.23 | fugitivo | langals: do this, lspci -vv |grep USB |
| 14:09.23 | `Sauron | texas AG is suing vonage |
| 14:09.27 | `Sauron | Fun Fun |
| 14:09.30 | jakepdev | <p1tst0p> - yep - just find the appropriate context for your provider in your dialplan (extensions.conf) |
| 14:09.53 | p1tst0p | jakepdev, yep i got htat |
| 14:09.58 | p1tst0p | *that |
| 14:09.58 | RoyK | ~using packages |
| 14:09.59 | jbot | it has been said that using packages is not recommended. get the asterisk source from http://www.asterisk.org/index.php?menu=download and compile them |
| 14:10.10 | langals | that comes up with nothing |
| 14:10.25 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 14:10.30 | ManxPower | ~docs |
| 14:10.31 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:10.31 | elriah | RoyK: Yea, I've seen that. But like I said, it's really stable and easy to deploy. |
| 14:10.32 | ManxPower | ~mailinglist |
| 14:10.33 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 14:10.33 | dwmw2_gone | out of interest, why is it not recommended to use packages? |
| 14:10.36 | fugitivo | langals: then for some reason, linux doesn't detected your USB controller |
| 14:10.41 | jakepdev | <p1tst0p> - do you want it to dial your exten then goto vm if no answer? |
| 14:10.43 | dwmw2_gone | I was vaguely planning to package Asterisk in Fedora Extras |
| 14:11.03 | langals | what do you suggest I do? |
| 14:11.15 | elriah | I can have a running system up in about 45 minues with just a net-install debian cd. |
| 14:11.20 | elriah | Answering calls. |
| 14:11.26 | p1tst0p | jakepdev, got that bit working, what i would like to do is, say im out of my office, and i want to check my voicemail from outside, can i do that ? |
| 14:11.28 | elriah | Amd O |
| 14:11.31 | fugitivo | langals: in this kind of problem, google is your best friend |
| 14:11.33 | elriah | And I'm new to * |
| 14:12.09 | jakepdev | <p1tst0p> - how do you want to get into vm from the outside - press a number - etc? |
| 14:12.15 | langals | so let me get this clear what I want to do - install a USB UHCI driver |
| 14:12.16 | dwmw2_gone | p1tst0p: set up some way that you can (authenticate and) get to VoicemailMain() from the outside. |
| 14:12.32 | fugitivo | langals: yes, but without any USB controller, the module won't load |
| 14:12.39 | elriah | Ahh - just realized there is a cdr.csv.. nice.. this will make my life easier.. |
| 14:12.49 | dwmw2_gone | p1tst0p: one option is to just fix your dialplan so that incoming calls from the outside go directly there... you probably don't want to do it quite like that though |
| 14:13.13 | dwmw2_gone | langals: what type of machine is this? What motherboard? |
| 14:13.13 | langals | so I need to get a usb controller then |
| 14:13.24 | fugitivo | langals: you don't have any? |
| 14:13.27 | dwmw2_gone | langals: if you have USB sockets on it, it's a fairly safe bet there's a USB controller. |
| 14:13.27 | elriah | How do I turn on Call Detail Record so my cdr.csv starts getting populated? |
| 14:13.33 | dwmw2_gone | is USB disabled in the BIOS? |
| 14:13.45 | dwmw2_gone | that might hide it from the PCI bus and explain why Linux doesn't see it. |
| 14:14.01 | dwmw2_gone | there may be a way to turn it back on from Linux, or you could just re-enable it in the BIOS |
| 14:14.05 | langals | Intel Celeron 2.4Ghz.... |
| 14:14.13 | dwmw2_gone | that's a CPU not a motherboard |
| 14:14.14 | langals | Linux Redhat 9 |
| 14:14.16 | p1tst0p | jakepdev, yeh i guess, say i ring my office, on the PSTN number provided by sipgate, i guess i can get it to recognise my mobile number ? and press for voicemail box? |
| 14:14.19 | dwmw2_gone | there's no USB on that I promise you |
| 14:14.23 | langals | ok. hang on |
| 14:15.08 | langals | How do I check the motherboard? |
| 14:15.23 | dwmw2_gone | p1tst0p: or do something like accepting '*' while it's ringing, regardless of the source, and take you there. |
| 14:15.39 | p1tst0p | dwmw2_gone yeh that would be nice |
| 14:15.45 | fugitivo | langals: just, see if you have some place to plug an usb device |
| 14:15.52 | dwmw2_gone | langals: run 'dmidecode' |
| 14:16.00 | langals | ja - I do have USB ports |
| 14:16.15 | fugitivo | langals: then, reboot your machine, and check the BIOS |
| 14:16.33 | fugitivo | langals: if usb is enable, there's another problem |
| 14:16.54 | jakepdev | <p1tst0p> - you can check the CALLERID variable and do a GotoIf |
| 14:16.57 | RoyK | elriah: CDR is default on unless you turn it off |
| 14:17.21 | fugitivo | i never had problems with usb in linux |
| 14:17.37 | RoyK | jakepdev: er. callerid can be checked with the normal pattern checking |
| 14:17.42 | RoyK | or what do you mean? |
| 14:17.53 | _Sam-- | hey i have a PRI on NI2, but i dont seem to be getting any names on the caller id -- what should i check? |
| 14:18.27 | jakepdev | <RoyK> - he said he wants to goto voicemail admin based on if it recognizes his phone number |
| 14:18.50 | jakepdev | <RoyK> - his phone number should be in CALLERID |
| 14:19.38 | jakepdev | <RoyK> - GotoIf will allow him to respond based on the caller id info |
| 14:19.49 | *** join/#asterisk jlewis (~jlewis@solo.atlantic.net) |
| 14:20.09 | RoyK | jakepdev: exten => blah/his_number,1,VoiceMailMain |
| 14:20.11 | *** join/#asterisk mesi (~player@dsl-082-083-055-218.arcor-ip.net) |
| 14:20.12 | pascals | Which extensions.conf editor would you guys suggest? I've grown a bit tired of its silly exten => xx,y prefixing. |
| 14:20.12 | RoyK | right? |
| 14:20.29 | pascals | ... very 80's basic. |
| 14:21.01 | pascals | No, actually, 80's basic let you skip line numbers, so you could insert some later |
| 14:21.27 | mesi | pascals: Yes, perhaps you can use some kind of compiler. |
| 14:21.49 | pascals | mesi: that was what I was asking for - people must use those already |
| 14:22.25 | mesi | pascal: I'm not sure. You shouldn't do too much with extension.conf. Better write C modules using asterisk's module api. |
| 14:22.28 | p1tst0p | jakepdev, could you get the system to recognise me, and give me a dial tone ? therefor i could call out on SIP then couldnt i from my mobie for isntance ! |
| 14:22.32 | jakepdev | <RoyK> - don't know - if it works that way also - then that is an alternative |
| 14:22.36 | langals | fugitivo - how do I check bios (sorry for the basic question) |
| 14:22.51 | RoyK | jakepdev: it's cleaner and documented as 'ex girlfriend logic' |
| 14:22.56 | pascals | C modules to script the dialplan? Are you serious? |
| 14:23.20 | RoyK | jakepdev: exten => _X./${EXGF},1,Goto(buggeroff) |
| 14:24.00 | mesi | pascals: No, to implement what you want to implement. E.g. I was implementing acallback extension, it calls back the callerid when the line washung up before asterisk could answer. This would better have been a seperate application. |
| 14:24.31 | fugitivo | langals: eeerr, reboot, and press del or f2, it depends on your computer, just look at the screen |
| 14:25.09 | pascals | No, that is not what I am doing. This is just basic stuff, like conditional announcements |
| 14:25.26 | jakepdev | <p1tst0p> - before I answer again - I'll give Roy a chance as he might have a better way |
| 14:25.29 | jlewis | when setting up a set of iax user/peer entries, are there any reasons for/against using the same [name] for the pair of entries for sending calls to / taking calls from a particular server? |
| 14:28.02 | langals | can't find usb - I am looking under integrated periperals |
| 14:28.27 | *** join/#asterisk cereal_ (~nico@gifu.newel.net) |
| 14:28.30 | cereal_ | hi |
| 14:28.31 | wildgoose | anyone here on sipgate.co.uk? I am just getting an engaged tone when I ring my external number? Is anyone else working at the moment? |
| 14:29.01 | Makenshi | wildgoose, i use sipgate |
| 14:29.09 | cereal_ | Im looking for a good tool to analyse cdr_csv files any advice ? |
| 14:29.10 | wildgoose | can you call yourself at the mooment? |
| 14:29.13 | pigpen | Has anyone used a SBE, Inc T1 card? http://www.sbei.net/Products/WAN/wanPCI-CxT1E1.htm |
| 14:29.15 | Makenshi | wildgoose, yes |
| 14:29.42 | wildgoose | Hmm, I'm getting a busy tone, but I just changed my number to a new one, then fiddled with the asterisk settings.... Could be anything... |
| 14:29.53 | wildgoose | I can ring out ok though! |
| 14:30.05 | jakepdev | <p1tst0p> - ok looks like he isn't going to answer - I'd say just prompt for an extension then include your outdialing context. |
| 14:30.11 | wildgoose | I have changed asterisk back to how it was before so I suspect a sipgate issue...? |
| 14:30.34 | jakepdev | <p1tst0p> - er. prompt for SIP number |
| 14:31.23 | langals | fugitivo - usb does not seem to be there |
| 14:31.27 | wildgoose | By the way, sipgate.co.uk have just released a whole bunch of new tel numbers, so have a peek if you want something more memorable... |
| 14:32.04 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 14:32.32 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 14:32.58 | Makenshi | wildgoose, perhaps it takes a little time |
| 14:33.06 | Makenshi | i haven't changed my number since i got it |
| 14:33.10 | Makenshi | it's +441213146461 |
| 14:33.19 | Makenshi | registered with e164.org too :> |
| 14:34.00 | p1tst0p | my sipgate number was active within 24 hours i think |
| 14:35.36 | RoyK | bbl. reboot. testing yellowdog linux :) |
| 14:35.53 | dwmw2_gone | RoyK: do you run asterisk on it? |
| 14:36.10 | dwmw2_gone | or indeed any SIP client? |
| 14:36.14 | *** join/#asterisk mmckernan (mmckernan@c211-28-35-204.sunsh1.vic.optusnet.com.au) |
| 14:36.33 | RoyK | dwmw2_gone: on what? yellowdog? |
| 14:36.36 | dwmw2_gone | yeah |
| 14:36.43 | RoyK | dwmw2_gone: I just want to see if this powerbook can run linux :) |
| 14:36.45 | dwmw2_gone | I've been trying to get a SIP client working on Fedora/PPC |
| 14:36.55 | RoyK | well |
| 14:36.56 | dwmw2_gone | nothing much seems to work well. |
| 14:36.59 | RoyK | we'll see :) |
| 14:37.01 | RoyK | bbl |
| 14:37.27 | Makenshi | does anyone know any writings on integrating sip with a public key infrastructure? |
| 14:37.29 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 14:38.39 | wildgoose | Makenshi: how does registering at e164.org help me? |
| 14:39.14 | p1tst0p | jakepdev, is it possible to edit the Voicemail section, to add an option to login and check voicemail ? |
| 14:39.22 | nestAr | nice.. |
| 14:39.23 | nestAr | asterisk*CLI> show version |
| 14:39.23 | nestAr | Asterisk built by root@asterisk on a i686 running Linux |
| 14:39.24 | *** join/#asterisk kahuna_ (~sootroom@rtl-2.i2k.com) |
| 14:40.03 | che | nestAr: you shouldnt compile software as root in general. (2 cent) |
| 14:40.24 | nestAr | nickel |
| 14:40.54 | che | only the make install step requires root ;) |
| 14:41.05 | *** join/#asterisk Tjardick (~tjardick@13.140-136-217.adsl.skynet.be) |
| 14:41.25 | nestAr | if i manage to break anything compiling asterisk as root.. well, the only thing i'll break is asterisk.. |
| 14:41.34 | nestAr | since it's the only thing running on this box |
| 14:41.53 | Makenshi | wildgoose, those people who run phone exchanges that lookup using e164 can call you for free using the internet rather than pstn |
| 14:42.06 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 14:42.11 | cereal_ | Im looking for a good tool to analyse cdr_csv files any advice ? |
| 14:42.58 | che | nestAr: well that maybe true for asterisk ;) |
| 14:43.14 | che | nestAr: theoretically it can wreck your whole sys though to compile root. |
| 14:43.25 | che | nestAr: compile as even. |
| 14:43.41 | che | nestAr: worst case scenario atleast ;) |
| 14:43.57 | nestAr | [09:40] * nestAr doesn't care |
| 14:44.35 | *** join/#asterisk jakepdev (~jakepdev@pool-68-163-51-30.phil.east.verizon.net) |
| 14:44.51 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 14:46.50 | dwmw2_gone | first they send the registration email from an invalid address |
| 14:47.20 | dwmw2_gone | now it doesn't let me log in to resend it. Neither does the 'Forgot your password?' link do anything useful |
| 14:47.34 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 14:47.41 | wildgoose | my email came from support.sipgate.co.uk which does have an IP registered... |
| 14:47.42 | n1gg4s | I do not obtain to initiate colloquy between usuarios in kphone! they anyone knows why? |
| 14:48.09 | dwmw2_gone | oh, maybe I'm looking at the wrong mail in the log then |
| 14:48.46 | wildgoose | any recommendations for a nice DECT handset which works slightly better with VOIP? |
| 14:48.56 | wildgoose | ie has buttons which might be mapped to useful featuers |
| 14:49.35 | Hmmhesays | wow that is the strangest thing i've ever heard |
| 14:49.54 | Hmmhesays | nuts |
| 14:50.00 | dwmw2_gone | wildcard0: you're right. It was greylisted for 5 minutes for having spamassassin points. It was libretel who are still trying to send me something from root@www2.libretel.com -- evidently for something else I tried to sign up for |
| 14:52.20 | *** join/#asterisk Tili (~Tili@202-133-65-15-dialup.sat.net.pk) |
| 14:54.18 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 14:54.42 | langals | Has anyone out there use app_conference and can give me a comparison with Meetme |
| 14:55.17 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 14:59.32 | fugitivo | what's the lastest version of zaptel? |
| 15:01.53 | ManxPower | fugitivo: 1.0.7 and CVS-HEAD |
| 15:02.15 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 15:02.27 | fugitivo | ManxPower: thanks |
| 15:02.50 | ManxPower | fugitivo: But that does not matter. You should ALWAYS use the same version of Asterisk, Zaptel and libpri. |
| 15:03.04 | langals | fugitivo - I think I need to install a usb controller - how do I do this? |
| 15:03.56 | NewSole | anyone know of a good accounting package that is free or pay that is quick to setup.... |
| 15:04.00 | ManxPower | langals: Are you sure app_conference needs zaptel? |
| 15:04.24 | ManxPower | langals: What is your specific problem? |
| 15:04.41 | langals | I am trying to use Meetme |
| 15:05.05 | langals | I thought of app_conference as an option if I couldn't get meetme working |
| 15:05.06 | ManxPower | langals: Then you need a zaptel timer. ztdummy (requires USB hardware of the correct type), or zaprtc |
| 15:05.19 | ManxPower | langals: I don't know. app_conference is not part of Asterisk |
| 15:05.31 | langals | so I could use zaprtc instead of usb for ztdummy? |
| 15:05.49 | ManxPower | langals: Yes. ZapRTC does not work with SMP on kernel 2.4 |
| 15:06.07 | *** join/#asterisk blackjack (~dermot@82.141.226.201) |
| 15:06.32 | langals | Then it won't work because I am running kernel 2.4 |
| 15:06.44 | blackjack | hi all. I have a linphone related problem. Thought I would ask here where experts might be found? |
| 15:06.45 | ManxPower | If you are not running SMP it will work. |
| 15:06.55 | langals | what is SMP? |
| 15:07.04 | ManxPower | blackjack: I don't know anyone that uses linphone. |
| 15:07.08 | NewSole | Multi proccessors |
| 15:07.18 | ManxPower | langals: multiple processors |
| 15:07.31 | langals | no - not running multiple processors |
| 15:07.32 | bile_one | blackjack, I use linphone |
| 15:07.36 | mishehu | I've used linphone for testing purposes. |
| 15:07.41 | langals | where do I get zaprtc? |
| 15:07.42 | ManxPower | langals: then ZapRTC might work for you. |
| 15:07.52 | mishehu | ManxPower: there, now you know 2 people who have used linphone. ;-) |
| 15:07.52 | ManxPower | ~google site:lists.digium.com zaprtc |
| 15:08.02 | ManxPower | RTFG |
| 15:08.13 | NewSole | lol |
| 15:08.13 | *** join/#asterisk olivier_ (~olivier_@82.127.99.32) |
| 15:08.19 | mishehu | RTFHHGTTG |
| 15:08.22 | bile_one | haa haa haa RFTG! |
| 15:08.32 | NewSole | Manx |
| 15:09.31 | Nugget | heh |
| 15:09.31 | NewSole | I need some help on finding an Accounting package.... you know any good ones |
| 15:09.39 | ManxPower | NewSole: No. |
| 15:09.47 | ManxPower | Why would I need an accounting package? |
| 15:09.57 | ManxPower | I just outsource all my bookkeeping. |
| 15:10.06 | NewSole | just thought you might know of a free or pay one |
| 15:10.20 | NewSole | for asterisk |
| 15:10.22 | ManxPower | My accounting package is called Susan. |
| 15:10.30 | ManxPower | NewSole: Perhaps YOU should RTFG? |
| 15:10.40 | blackjack | bile_one/lishehu: linphonec reads commands from stdin (blocking read). If we modify it to read from a named pipe (blocking read) no more called party audio. Reading from the named pipe (non-blocking read), audio works in both directions. Reading from named pipe (non-blocking read with a sleep so that we're not in a tight loop), audio for the called party missing again. |
| 15:11.44 | BrianR___ | Anyone know if asterisk supports setting the call-by-call services stuff on ISDN? |
| 15:11.44 | bile_one | ManxPower, he should UTFG. |
| 15:12.15 | ManxPower | NewSole: Maybe you mean a BILLING software for Asterisk CDRs? There are one or two I think, but most people write their won. |
| 15:12.43 | bile_one | blackjack, I use linphone but have not customized it. Sorry |
| 15:12.46 | NewSole | this is my problem.... |
| 15:13.06 | NewSole | I was looking for termination that was good... but cheap.... |
| 15:13.18 | NewSole | I fond one... but |
| 15:13.20 | ManxPower | NewSole: Since billing is so unique to each company, each company writes their won. |
| 15:13.45 | mishehu | blackjack: same as bile_one. |
| 15:13.55 | NewSole | I have to buy large blocks.... and I was looking for a billing system I could use |
| 15:14.12 | ManxPower | As I said, each company writes their own. |
| 15:14.16 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 15:14.33 | NewSole | k |
| 15:16.02 | bile_one | ManxPower, that is sick |
| 15:16.17 | ManxPower | bile_one: A LOT of tshirthell's shirt are sick. |
| 15:16.23 | ManxPower | some of them are even sick in a good way |
| 15:16.33 | ManxPower | But not many of them. |
| 15:17.02 | mishehu | is that an "I survived Red Lake and all I got was this lousy t-shirt" t-shirt? |
| 15:17.08 | ManxPower | This one is funny: http://www.tshirthell.com/store/product.php?productid=422 |
| 15:18.21 | mishehu | do they have any asterisk-related t-shirts? |
| 15:18.22 | mishehu | heh |
| 15:18.32 | ManxPower | Of this one: http://www.tshirthell.com/store/product.php?productid=374 |
| 15:18.40 | ManxPower | Of == Or |
| 15:18.56 | ManxPower | I prefer t-shirthumor.com most of the time |
| 15:20.51 | ManxPower | I want to get this one: http://www.t-shirthumor.com/Merchant2/merchant.mvc?Screen=PROD&Product_Code=pltr&Category_Code=sanr |
| 15:20.52 | tzanger | heh I like that one |
| 15:21.24 | Nugget | I like some of the http://bustedtees.com/ shirts |
| 15:21.33 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 15:21.41 | tzanger | I think one of the funniest I've seen was a baby t-shirt that said "who's my daddy?" |
| 15:21.54 | SexyKen | Hey guys -- if wget isn't installed on a server what's another option to use for dling from http? |
| 15:21.58 | tzanger | ManxPower: haaaaaaaaaahahahahhaa |
| 15:22.04 | christo | I saw a baby bib thing which said 'make clean' :) |
| 15:22.11 | Delvar | SexyKen: telnet |
| 15:22.19 | blackjack | bile_one/mishehu: what about other linux sip clients in case we can't solve this problem? |
| 15:22.20 | trym | make me |
| 15:22.23 | Nugget | that's a great shirt |
| 15:22.31 | SexyKen | Delvar,how? |
| 15:22.47 | Delvar | SexyKen: telnet someserver.com 80 |
| 15:22.54 | tzanger | christo: yes I saw that too |
| 15:23.22 | ManxPower | T-Shirt Hell has a baby shirt that says "I'm the reason Daddy drinks." |
| 15:23.30 | tzanger | hahaha |
| 15:23.38 | tzanger | my son had one that said "geek in training" |
| 15:23.40 | bile_one | blackjack, kphone and gaimphone for starters |
| 15:23.54 | ManxPower | kphone doesn't even support OOB DTMF |
| 15:24.04 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 15:24.05 | SexyKen | Dont work, Delvar, gives error. |
| 15:24.22 | Delvar | SexyKen: what error? |
| 15:24.29 | bile_one | blackjack, take a look here. http://www.voip-info.org/wiki-Linux#comments |
| 15:24.46 | SexyKen | 'Unkown Server Error' |
| 15:25.25 | SexyKen | wait |
| 15:25.29 | SexyKen | now I'm connected |
| 15:25.34 | SexyKen | How do I dl the ifle? |
| 15:25.35 | SexyKen | http://belnet.dl.sourceforge.net/sourceforge/egroupware/eGroupWare-1.0.00.006-1.tar.gz |
| 15:25.56 | Nugget | for the pilots in the channel: http://www.pilotwear.com/product_info.php/products_id/307 |
| 15:26.19 | tzanger | nah |
| 15:26.23 | tzanger | use exchange4linux |
| 15:26.30 | tzanger | they just released their v3.0 stuff, it looks awesome |
| 15:26.40 | tzanger | the 2.5x worked well but was slow... this is much faster |
| 15:27.11 | Delvar | SexyKen: if it connects then times out try typing 'GET some/url.html HTTP/1.1' (without quotes) |
| 15:28.06 | SexyKen | Delvar, I'm trying to download this file: |
| 15:28.07 | SexyKen | http://belnet.dl.sourceforge.net/sourceforge/egroupware/eGroupWare-1.0.00.006-1.tar.gz |
| 15:28.09 | SexyKen | What do I type for that |
| 15:28.24 | Delvar | oh to download a file i duno |
| 15:28.30 | Delvar | i thought you were just testing |
| 15:30.56 | *** join/#asterisk jixi (~damien@193.190.210.151) |
| 15:32.09 | langals | manPower - if i am using zaptelrtc, do I need to use ztdummy with this? |
| 15:32.34 | Delvar | SexyKen: w3m is another app for downloading, not used it tho |
| 15:32.41 | jixi | hello, I want to set up a queue for which all members are phoned in a defined order. I tried the "roundrobin" strategy, but it seems that if the first member doesn't answer, it will start at the second one on next call. Is there a way to force calls to always start at member 1? Thanks. |
| 15:33.29 | *** join/#asterisk __a (user@193.140.215.2) |
| 15:33.54 | __a | guys, how do i call an app from within another app? any examples doing this? |
| 15:34.28 | __a | i.e. i'm writing an app and would like to call Dial() from it, as if it is called via extensions.conf |
| 15:35.04 | *** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 15:35.14 | *** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 15:37.03 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:37.03 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:37.09 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:37.14 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:37.14 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:37.24 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:37.28 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:37.28 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:37.39 | langals | I seem to have a linux installation which has USB ports, but does not have a usb controller installed - would someone be able to give me some advice on how to get a usb controller installed |
| 15:38.13 | Hmmhesays | that's odd |
| 15:38.15 | Makenshi | langals, what distribution is it? |
| 15:38.26 | dwmw2_gone | langals: did you look in the BIOS? |
| 15:39.17 | langals | Redhat 9 |
| 15:39.20 | *** join/#asterisk Rick_Hunter (~rhunter@06-127.008.popsite.net) |
| 15:39.25 | langals | looked in BIOS and did not seem to be there |
| 15:39.33 | dwmw2 | nothing about enabling/disabling USB? |
| 15:39.57 | langals | let me have another look - will you wait for me to reboot? |
| 15:40.03 | dwmw2 | ok |
| 15:40.04 | *** join/#asterisk pycsusz (~pycsusz@pluto.euronetrt.hu) |
| 15:40.20 | pycsusz | Hi everybody! |
| 15:40.29 | nestAr | anyone in here using openclose.agi ? |
| 15:40.52 | nestAr | hi |
| 15:41.14 | pycsusz | I need some help about to make conference calling with asterisk |
| 15:41.19 | langals | dwmw2 - ok - i am now in BIOS setup |
| 15:41.22 | pycsusz | somebody can help me? |
| 15:41.31 | langals | where should I go? |
| 15:41.43 | dwmw2 | go where the option to enable USB is. |
| 15:41.52 | dwmw2 | do you have an 'onboard peripherals' menu? |
| 15:43.07 | nestAr | or intergrated peripherals |
| 15:43.15 | nestAr | as some call it |
| 15:43.27 | langals | found it! - it is ask me whether I want to enable 2, 4 or 6 usb ports - does it make a difference? |
| 15:43.44 | nestAr | how many you want to use/] |
| 15:44.08 | PoWeRKiLL | I have a bug when calling from a * to another * box via IAX the billsec start also when it's ringing on the zap channel |
| 15:44.12 | PoWeRKiLL | any idea? |
| 15:44.41 | langals | Thanks for the help - hopefully ztdummy will work now |
| 15:44.53 | *** join/#asterisk Bacon (~Bacon@thorin.nplus1.net) |
| 15:45.22 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 15:45.27 | dwmw2 | not entirely sure why ztdummy needs USB... |
| 15:45.42 | *** join/#asterisk Lee__ (~Lee__@ool-44c26142.dyn.optonline.net) |
| 15:46.17 | cjk | hi, what prog do you suggest for preconfiguring/modifiying a grandstream firmware |
| 15:46.56 | langals | neither am I - I cannot see the link between the 2 |
| 15:47.42 | Delvar | dwmw2: it needs a reliable timing device, and some usb chips have such a timmer. so they just use that |
| 15:48.01 | dwmw2 | ah. |
| 15:48.08 | Hmmhesays | ~seen katty |
| 15:48.13 | jbot | katty is currently on #asterisk. Has said a total of 158 messages. Is idling for 20h 9m 50s |
| 15:48.16 | *** part/#asterisk __a (user@193.140.215.2) |
| 15:48.37 | dwmw2 | if langals has a VIA board I suspect that's OHCI and hence probably doesn't work, if ztdummy needs uhci |
| 15:48.44 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 15:48.51 | syslod | Hello. |
| 15:48.54 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 15:49.22 | fishboy1669 | hi guys hows things |
| 15:49.33 | *** join/#asterisk lespiggot (~les@217.206.141.130) |
| 15:49.40 | syslod | Having problems with not detecting handups. |
| 15:49.44 | syslod | hangups |
| 15:50.41 | *** join/#asterisk zapfhc (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 15:50.53 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:51.11 | zapfhc | Mar 23 15:51:27 WARNING[1053]: chan_zap.c:922 zt_open: Unable to specify channel 1: No such device or address |
| 15:51.17 | zapfhc | help? |
| 15:51.19 | zapfhc | :| |
| 15:51.46 | langals | dwmw2 - I managed to load the usb-uhci module and ztdummy module (listed if I go lsmod), but conferencing still causing issues.... |
| 15:52.07 | langals | It logs into the conference, and say that I am the first person, but then cuts me off |
| 15:53.08 | fishboy1669 | sys what interface |
| 15:53.16 | Lee__ | can someone recommend an origination/termination service? Right now I only know of VoicePulse |
| 15:53.41 | fishboy1669 | zap what interface? |
| 15:53.48 | BrianR___ | Anyone know if it's possible for Asterisk to get/set the SID on calls over a PRI? |
| 15:54.27 | syslod | zap |
| 15:54.36 | syslod | Its a T1 going to a CAC AB I |
| 15:54.42 | langals | If I run sip debug, I am getting various warning messages.... |
| 15:54.54 | fishboy1669 | sorry sys im x100p user |
| 15:55.29 | *** join/#asterisk sariabod (~sariabod@ip21.farheap.net) |
| 15:55.52 | langals | WARNING [1744]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/2). Any idea what the problem is? |
| 15:56.24 | syslod | Is there any cmd to see what channels are in use? |
| 15:56.25 | *** join/#asterisk blackadder (~sburley@163-177.adsl.totalweb.net.uk) |
| 15:56.36 | drumkilla | syslod: show channels |
| 15:56.40 | blackadder | hi guys been a while |
| 15:57.23 | blackadder | this maybe a stupid comment but has the asterisk capi_channel been broke recently? |
| 15:57.26 | syslod | drumkilla: thks |
| 15:57.51 | christo | I'm telnetting to an asterisk server just to try out some commands.. do I just type each line followed by 'return', or should I send escaped newline and carriage return characters (\n\r) explicitly? |
| 16:00.39 | jontow | christo.. the manager port? 5038? |
| 16:00.40 | ManxPower | langals: to see the names/format numbers type "show codecs" |
| 16:00.48 | pycsusz | Somebody knows something about grandstream bt-100's conference button? |
| 16:01.07 | Fraeggl | does someone know if it is possible to register sip-phones with static, non host=dynamic ip-addresses ? |
| 16:01.11 | jontow | pycsusz; that is at least a mystery to me as well. |
| 16:01.14 | langals | manpower - done that |
| 16:01.15 | ManxPower | pycsusz: Other than the fact that it doesn't work and the fact is documented by Grandstream in their product sheets? No. |
| 16:01.25 | Fraeggl | i think this registration is essential ? |
| 16:01.25 | langals | What am i looking for? |
| 16:01.39 | ManxPower | Fraeggl: Yes, but it's not called "registration" |
| 16:02.26 | Fraeggl | i get errors like "Peer '22' is trying to register, but not configured as host=dynamic" |
| 16:02.37 | Fraeggl | but they show up in show ip peers |
| 16:03.03 | ManxPower | Fraeggl: A device MAY NOT register unless you have host=dynamic. |
| 16:03.06 | Fraeggl | if i use defaultip=... the error disapears, but phoning doesnt work either ;) |
| 16:03.08 | christo | jontow - yes |
| 16:03.11 | *** join/#asterisk adjacent (~scott@68.115.123.35) |
| 16:03.33 | ManxPower | All registration does is tell Asterisk that the IP of the device is. If the ip never changes then just tell your client not to register and set host= to the ip of the device. |
| 16:04.00 | Fraeggl | ManxPower: thx, so its suff to have it in 'sip show peers', 'sip show registry' is allowed to be empty ? |
| 16:04.16 | Essobi | What's some hardphones that support IAX2? |
| 16:04.24 | pycsusz | ManxPower: That's all? |
| 16:04.29 | Essobi | I'm going to talk my boss into buying a few to test |
| 16:05.26 | ManxPower | Fraeggl: "sip show registery" shows what remote servers Asterisk is registered TO |
| 16:06.01 | Essobi | Excuse me.. what's some "good" IAX hardphones.. |
| 16:06.16 | fishboy1669 | hi manx hows things |
| 16:06.20 | Fraeggl | ManxPower: ah, thx, but still its not working :( im a newbie ;) |
| 16:06.59 | ManxPower | Fraeggl: then it's not working for some other reason |
| 16:07.43 | Fraeggl | one final (no :) ) question: my dial-plan is eg 'exten => 21,1,Dial(SIP/21,20,tr)', so i would hope to have the phone ring, when i dial 21 on the other phone (22), but nothing happens ? |
| 16:07.50 | Fraeggl | would this plan be ok ? |
| 16:08.20 | Fraeggl | the phones are found by asterisk (sip show peer 21 / 22) |
| 16:09.16 | syslod | Does anyone have a sample or a way to do internal callerid and external caller id? |
| 16:09.56 | jontow | just a theory kinda question.. what're the benefits of running comedian mail backed by a mysql database? |
| 16:10.09 | jontow | or postgresql even |
| 16:11.03 | syslod | I'm doomed until I can do this... Any idea on how hard it would be or where to look to add either a var for the sip account id or a system that will return the output? |
| 16:11.22 | langals | If I get the following message in sip deug: WARNING: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 64/64) - has this something to do with the codec? |
| 16:12.07 | tzanger | johnnyb: I can't think of any |
| 16:12.10 | tzanger | added complexity :-) |
| 16:12.20 | tzanger | unless you've got a SHITLOAD of voicemailboxes but even then... |
| 16:12.36 | jontow | syslod; get crazy with a macro that matches s/XXXX (where XXXX is the pattern for your internal calls) and does SetCallerId.... on it, and then outside numbers, pattern-match those to do another SetCallerId() |
| 16:12.43 | tzanger | slePP: ftp3.ca.postgresql.org is slowwwwwwwwwwwwwwww |
| 16:12.53 | jontow | well, im gonna have 300-500 |
| 16:12.55 | Moc____ | so use ftp4 ;) |
| 16:13.08 | tzanger | Moc____: slePP runs ftp3 I think :_) |
| 16:13.13 | Alexi1 | bye |
| 16:13.15 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
| 16:13.21 | _Sam-- | from my softphone extension, if i dial any other internal extensions or external numbers, it takes like 5 seconds before the console (on the same LAN) sees the SIP call info and connects the call....all my extensions are configured the same, but everyone else here connects in like .1 seconds....ive tired everything, but im not sure what to check |
| 16:14.10 | _Sam-- | like i have music on hold setup on an extension to answer immediately and play....if i call it from my softphone, it takes like 10 seconds before i hear it....on another softphone, same lan, same config, it plays in like .1 seconds |
| 16:15.08 | Moc____ | ha hehe |
| 16:16.35 | Fraeggl | could someone perhaps tell me which fields in 'show peer <phone>' for a sip-phone are essential for a working phone ? |
| 16:16.53 | Fraeggl | esp i'm missing output for "Full Contact" |
| 16:17.03 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
| 16:19.00 | olivier_ | <Fraeggl> i'm not a sip guru, but you should try "Sip debug" to see the pb |
| 16:19.11 | *** join/#asterisk StealthMethod (~nelsonx@adsl-070-148-141-009.sip.mia.bellsouth.net) |
| 16:20.41 | _Sam-- | what would i check if my caller ID is just sending the number, no name ? i verified with the PRI provider that they are sending name&number across |
| 16:21.13 | Fraeggl | when doing "sip debug" "sip reload" all the output looks sensible.... hard to say for me tough ;) |
| 16:21.36 | *** join/#asterisk Dibbler (~Dibbler@zidane.pi-net.net) |
| 16:23.23 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfjsi.dialup.mindspring.com) |
| 16:23.51 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 16:24.37 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 16:25.51 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 16:25.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:31.50 | *** join/#asterisk mxmasster (~maxc@rottie.media.net) |
| 16:31.51 | mxmasster | hi all |
| 16:31.53 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.client.comcast.net) |
| 16:32.03 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 16:32.35 | mxmasster | how do i implement a dial by name directory in asterisk? |
| 16:32.43 | afrosheen | Ok, we've got a huge delay with our meetme conference, and the conference is between local extensions and people dialing in via an iax trunk. We have a zaptel card installed so it should get proper timing. How can I kill the delay? |
| 16:33.26 | BuckRogers | Good morning |
| 16:33.48 | afrosheen | not yet |
| 16:33.50 | afrosheen | :) |
| 16:34.26 | skrusty | anyone here fancy coming to an asterisk/voip event in the uk? if so, join #asterisk-uk :) |
| 16:34.59 | BuckRogers | I was thinking about it but the dollar value agianst the euro just dont pay |
| 16:35.37 | afrosheen | the euro against the british pound isn't much better |
| 16:36.05 | BuckRogers | really i thought it was about on target but im no expert |
| 16:36.31 | afrosheen | it's only on target if it's 1:1 |
| 16:36.44 | BuckRogers | hahaha if only that was true |
| 16:37.00 | *** join/#asterisk _omer (dfsdf@202.147.174.177) |
| 16:37.03 | *** join/#asterisk lilwookie (~bender@modemcable215.87-81-70.mc.videotron.ca) |
| 16:37.06 | afrosheen | british pound is about 1.5:1 against the dollar, probably 1.3 or 1.2:1 against the euro |
| 16:37.07 | _omer | hi... |
| 16:37.12 | *** join/#asterisk emacsen (hidden-use@gw.coderyte.net) |
| 16:37.20 | afrosheen | anyway |
| 16:37.22 | BuckRogers | the euro value reflects the collective gnp of the EU members where as the pound reflects the uk |
| 16:37.33 | emacsen | Is there a way to layer encryption from gateway to gateway or SIP phone to gateway? |
| 16:37.39 | _omer | ooops......I think I am in a wrong room... |
| 16:37.42 | afrosheen | yeah the uk refused to join the EU because they'd have to severely de-value their currency |
| 16:37.47 | _omer | It was related to asterisk......;) |
| 16:38.07 | BuckRogers | and most of their members are corupt with forien policy |
| 16:38.10 | afrosheen | emacsen: sure, if you like overhead you can feed it over an ipsec tunnel or a vpn |
| 16:38.24 | *** join/#asterisk Maxxed (Maxxed@65.67.149.242) |
| 16:38.24 | afrosheen | everyone is corrupt with foreign policy :p |
| 16:38.27 | Lee__ | emacsen: not unless you know the endpoints of each gateway |
| 16:38.32 | Maxxed | hello :) |
| 16:38.35 | emacsen | afrosheen: yeah seems like that's the only way ATM. |
| 16:38.46 | emacsen | afrosheen: it's a shame there's not something cleaner |
| 16:38.55 | Maxxed | i am looking for a pinout of the tdm400p |
| 16:39.04 | Lee__ | with email there's STARTTLS but not all gateways support it so it'll fall back to unencrypted |
| 16:39.05 | afrosheen | pinout? |
| 16:39.06 | _omer | I want to call at my Asterisk from my Cisco ATA ......how to do that? both are at public ip address. |
| 16:39.17 | Maxxed | i have 2 fxo modules, i want to "plug" in 2 analog lines |
| 16:39.22 | cjk | anyone here who has some experience with grandsteram and cfg.txt customization |
| 16:39.22 | Maxxed | well, my analog lines are cat3 |
| 16:39.24 | Lee__ | it'd be cool if there was something like STARTTLS for SIP phones :) |
| 16:39.28 | Maxxed | the connectors on the card are cat5 |
| 16:39.37 | Maxxed | 66block i supose |
| 16:39.39 | afrosheen | Maxxed: they're actually both |
| 16:39.44 | Maxxed | oh really? |
| 16:39.53 | afrosheen | that's why they have that wide and tall notch in them |
| 16:39.55 | Maxxed | so i can stuff a cat3 phone line in there? |
| 16:40.00 | afrosheen | yeah try it |
| 16:40.01 | Maxxed | ah :) |
| 16:40.06 | Maxxed | well damn i am just impressed now |
| 16:40.07 | Maxxed | heh |
| 16:40.29 | Maxxed | hell i never hurd of it utill a week ago |
| 16:40.32 | afrosheen | now that everyone is awake, here's my original big problem |
| 16:40.34 | afrosheen | Ok, we've got a huge delay with our meetme conference, and the conference is between local extensions and people dialing in via an iax trunk. We have a zaptel card installed so it should get proper timing. How can I kill the delay? |
| 16:40.44 | BrianR___ | Any isdn gurus around? |
| 16:40.47 | Maxxed | hey it worked! |
| 16:40.49 | Maxxed | thanks guys |
| 16:40.51 | ManxPower | afrosheen: take off enter/exit sounds option |
| 16:41.09 | afrosheen | that's it? |
| 16:41.11 | afrosheen | |q? |
| 16:41.17 | _omer | any one give me a chance to say thanks....;) |
| 16:41.18 | ManxPower | afrosheen: Try it |
| 16:41.21 | emacsen | afrosheen: is it possible the codec is too expensive? |
| 16:41.26 | afrosheen | no |
| 16:41.38 | ManxPower | afrosheen: there was a discussion about enter/exit sounds causing excessive delay in meetme |
| 16:41.43 | afrosheen | I'll try the quiet option, I'm gonna laugh if it works |
| 16:41.53 | ManxPower | afrosheen: You are not on the mailing lists, are you? |
| 16:41.55 | *** part/#asterisk lilwookie (~bender@modemcable215.87-81-70.mc.videotron.ca) |
| 16:42.08 | ManxPower | If you were, you might have seen the exact same suggestion this morning or last night |
| 16:42.10 | afrosheen | ManxPower: no but I read some thread on google that mentioned it |
| 16:42.20 | afrosheen | but there was no followup saying 'thanks it worked' |
| 16:42.21 | Juggie | why didnt you try it then? |
| 16:42.52 | _omer | How to call at my Asterisk through Cisco ATA186? both are at public IP addresses.. |
| 16:45.41 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 16:45.49 | Fraeggl | i someone here perhaps using an allnet 7950 sip phone together with asterisk ? |
| 16:46.05 | Fraeggl | cant get mine working.. |
| 16:46.52 | *** join/#asterisk viKing78 (~AdamHerbe@cerberus.franklinamerican.com) |
| 16:48.17 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:49.32 | *** join/#asterisk Ruben_Quinones (~ruben@66-50-56-199.prtc.net) |
| 16:50.02 | *** join/#asterisk RoyKa (~roy@143.80-202-166.nextgentel.com) |
| 16:50.45 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 16:50.48 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 16:51.15 | SexyKen | How can I add a '#' to the beginning of a line in vi? |
| 16:51.33 | smash- | ummm |
| 16:51.35 | eKo1 | You press the # key? |
| 16:51.46 | SexyKen | You have never used vi eh? |
| 16:51.47 | eKo1 | in insert mode |
| 16:51.48 | che | SexyKen: you press the insert key to be able to type. then just type ;) |
| 16:51.49 | Ruben_Quinones | press i, then the # |
| 16:51.51 | _Sam-- | press I |
| 16:52.04 | smash- | read the help me |
| 16:52.09 | smash- | there is mad commands |
| 16:52.12 | smash- | ;q |
| 16:52.13 | _Sam-- | then wq! |
| 16:52.13 | smash- | quits |
| 16:52.17 | Qwell | vimtutor is a great command on most Linux distros |
| 16:52.19 | smash- | aster u press esacpe |
| 16:52.33 | _Sam-- | or pico |
| 16:52.35 | smash- | pico is ok |
| 16:52.37 | smash- | vi is ok |
| 16:52.42 | eKo1 | vim >> pico |
| 16:52.48 | smash- | bleh |
| 16:52.51 | SexyKen | Thanks :-) |
| 16:52.52 | smash- | which ever |
| 16:52.55 | smash- | its a editor |
| 16:53.01 | smash- | sexy u get it all done? |
| 16:53.09 | Nugget | pico is easy to learn and difficult to use. vi is difficult to learn and easy to use. Since you learn once and use forever it's a simple decision. |
| 16:53.10 | FuriousGeorge | hi all. when i registered w/ sipphone.com for an incomming # i got 3 servers from them. a domain, a proxy, and a server. which one does sip.conf want for "hast" |
| 16:53.14 | FuriousGeorge | "host" |
| 16:53.15 | eKo1 | mcedit is better than pico for simple dumb editing. |
| 16:53.22 | _Sam-- | hey why would i get this error after ztcfg -v shows all my channels: Mar 23 11:52:13 NOTICE[19534]: app_dial.c:960 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 16:53.31 | smash- | i kinda forgot out to use vi but its not that bad |
| 16:53.33 | smash- | like 7 buttons u use |
| 16:53.33 | Ruben_Quinones | pico is much better than vi... vi is hell... pico is heaven |
| 16:53.38 | smash- | i j and k |
| 16:53.39 | smash- | or something |
| 16:53.47 | Qwell | smash-: hjkl? |
| 16:53.52 | smash- | yeah those ones |
| 16:53.52 | smash- | lol |
| 16:53.53 | Nugget | pico is crap, ok? |
| 16:54.03 | `Sauron | Whatever nugglet. |
| 16:54.04 | Qwell | nano is where its at :p |
| 16:54.04 | eKo1 | nano >> pico |
| 16:54.08 | smash- | haha |
| 16:54.11 | smash- | notepad |
| 16:54.12 | Makenshi | vi>* |
| 16:54.14 | smash- | is my favorite |
| 16:54.15 | `Sauron | pico > nugget |
| 16:54.18 | Qwell | ed>vi |
| 16:54.19 | smash- | notepad!! |
| 16:54.23 | Nugget | using pico is like trying to mow the lawn with toenail clippers. sure, it's a simple interface, but it can't do anything worthwhile. |
| 16:54.25 | _Sam-- | sauron, arent you from #php? |
| 16:54.27 | eKo1 | sed > ed |
| 16:54.29 | smash- | i do everything through a ssh on a windows laptop |
| 16:54.31 | `Sauron | Sam: I am. |
| 16:54.36 | smash- | so i lov notepad |
| 16:54.37 | `Sauron | Oooh. Punny. |
| 16:54.42 | Makenshi | vi>!vi |
| 16:54.45 | smash- | lol |
| 16:54.45 | eKo1 | smash-: get vim on windows |
| 16:54.45 | _Sam-- | i think you did some work for me years ago |
| 16:54.58 | `Sauron | Depends. What's your name? |
| 16:54.59 | smash- | echo "theinfo" >> file.name |
| 16:55.03 | _Sam-- | i am, sam |
| 16:55.10 | _Sam-- | kneedraggers.com |
| 16:55.14 | Qwell | It's "Sam I am" |
| 16:55.24 | `Sauron | hum |
| 16:55.29 | `Sauron | Maybe, I don't remember. |
| 16:55.34 | smash- | man im so bored at work |
| 16:55.37 | `Sauron | You used to host with Carl? |
| 16:55.38 | smash- | i goto mexico |
| 16:55.40 | smash- | at 3am |
| 16:55.40 | _Sam-- | you and derick from efnet #Php helped out in 2001 or so |
| 16:55.44 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 16:55.52 | `Sauron | Oh. Right. |
| 16:56.46 | _Sam-- | i could use some more help...zoa yesterday helped me get a new PRI / Zap stuff up and running |
| 16:57.00 | _Sam-- | but today incoming calls get a busy, and outgoing calls say cant get channel |
| 16:57.12 | _Sam-- | ztcfg -v shows everything fine |
| 16:57.31 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 16:57.54 | `Sauron | Can't help you there. |
| 16:57.59 | `Sauron | Not a Zap man. |
| 16:58.05 | Hymie | anyone know why callerid info would always appear two hours out of sync, for sip calls?? |
| 16:58.58 | smash- | hrm |
| 16:58.59 | Sedorox | wrong timezone somewhere? |
| 16:59.00 | smash- | only sip calls |
| 16:59.10 | smash- | how so 2 hours outta sync |
| 16:59.15 | *** join/#asterisk viKing78 (~AdamHerbe@cerberus.franklinamerican.com) |
| 16:59.21 | Hymie | Sedorox: the box has the right timezone... and emails are fine outgonig as well, so it's not that at least |
| 16:59.22 | smash- | like you get caller id info's for people who called 2 hours ago? |
| 16:59.28 | Hymie | smash-: yeah |
| 16:59.33 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:59.33 | Hymie | outgoing sip, from asterisk |
| 16:59.39 | Sedorox | to what? |
| 16:59.41 | smash- | what hardware |
| 16:59.43 | Hymie | always shows two hours early |
| 16:59.44 | cpatry | someone knows how to fix that: |
| 16:59.45 | cpatry | deptaudio@asterisk:~/papers/clod$ sox readym.wav -r 44100 -c 1 -s -w readym.gsm |
| 16:59.45 | cpatry | sox: Failed reading readym.wav: WAVE: RIFF header not found |
| 16:59.53 | Hymie | linksys voip router |
| 17:00.08 | Hymie | timezone is set right in it, too.. and altering its timezone has no effect |
| 17:00.09 | smash- | eww |
| 17:00.12 | Sedorox | does the voip router have the right timezone? |
| 17:00.14 | Maxxed | kneedraggers.com!? |
| 17:00.20 | Hymie | Sedorox: yeah |
| 17:00.31 | Sedorox | hmmm |
| 17:00.34 | Sedorox | not sure... |
| 17:00.35 | smash- | how much u pay for voip router? |
| 17:00.42 | Hymie | $120 or so, I think |
| 17:00.43 | _Sam-- | thats me |
| 17:00.43 | Hymie | CDN |
| 17:00.47 | _Sam-- | i rock a few R6s on the track |
| 17:00.57 | _Sam-- | where is your home track? |
| 17:00.59 | Hymie | Sedorox: any idea if there is a place to set sip timezone independantly? |
| 17:01.12 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 17:01.27 | Hymie | Sedorox: other than voicemail.conf, I can't find any inof about doing so, on the wiki |
| 17:01.31 | Maxxed | houston, tx |
| 17:01.49 | _Sam-- | nice, what year is your r6? |
| 17:01.52 | Maxxed | i scored my lil bro a tt600 last month |
| 17:01.53 | Maxxed | 02 |
| 17:01.58 | smash- | ha |
| 17:02.08 | smash- | i got a 600rr |
| 17:02.11 | *** join/#asterisk NetOfSickCoder (~NetOfSick@200.121.129.178) |
| 17:02.11 | Maxxed | im really a honda guy |
| 17:02.12 | Hymie | smash-: they're really the only way to do sip reliably over a home internet connection.. as they prioritize voip traffic, and require no additional computer / etc on to run |
| 17:02.12 | Maxxed | oh my |
| 17:02.17 | Maxxed | yeah see, im after a 6rr |
| 17:02.19 | Sedorox | Hymie: MAYBE in sip.conf.. but I haven't seen anything there... |
| 17:02.23 | _Sam-- | sorry to hear that...i bleed blue. |
| 17:02.25 | Maxxed | aw, makes my balls hurt looking at those |
| 17:02.26 | Hymie | Sedorox: I looked :/ |
| 17:02.27 | Maxxed | heh |
| 17:02.29 | _Sam-- | we run a yamaha supported AMA team |
| 17:02.37 | _Sam-- | ive been a die hard yam fan forever |
| 17:02.47 | smash- | its raining |
| 17:02.48 | smash- | outside here |
| 17:02.49 | Maxxed | dont get me wrong, i like a yamie, but deep down, im a honda fan |
| 17:02.51 | smash- | i just got it |
| 17:02.54 | smash- | 2 weeks ago |
| 17:02.59 | Fraeggl | hmm... stupid question... do i need a soundcard (/dev/dsp) in the asterisk server to work ?? |
| 17:03.01 | Maxxed | oh u lucky sob :p |
| 17:03.04 | smash- | less then 500 miles on it |
| 17:03.14 | *** join/#asterisk adjacent_ (~scott@office.bftwave.com) |
| 17:03.24 | _Sam-- | maxxed, had you ever heard of kneedraggers? |
| 17:03.24 | smash- | Fraeggl no |
| 17:03.28 | viKing78 | Anybody ever hooked up * to a Tadiran PBX? |
| 17:03.32 | Sedorox | Hymie: sorry... dunno |
| 17:03.38 | Maxxed | yes i have |
| 17:03.44 | viKing78 | How'd it go? |
| 17:03.45 | Maxxed | i ran across the site a few times |
| 17:04.03 | _Sam-- | is your r6 a trackbike or a streetbike? |
| 17:04.04 | Maxxed | i seen the url, n was like? hey thats familair |
| 17:04.04 | Maxxed | heh |
| 17:04.17 | Maxxed | its my daily rider, but i like to get er out on the track |
| 17:04.21 | Fraeggl | thx smash- ,, |
| 17:04.41 | Maxxed | i want to score my self a 6rr, n track my r6 all the way |
| 17:04.53 | smash- | rr;s are sweet |
| 17:04.59 | smash- | way better then f4 |
| 17:05.04 | Maxxed | oh yeah |
| 17:05.08 | Maxxed | by faaaar |
| 17:05.14 | smash- | i wanted a rc51 |
| 17:05.18 | smash- | but the rr's are just so sweet |
| 17:05.18 | Maxxed | 2cyl ;) |
| 17:05.19 | `Sauron | I'm thinking of more interesting ways to kill myself than on the track. |
| 17:05.36 | Maxxed | yeah, i feel in love with the 6rr right off the bat |
| 17:05.37 | smash- | i dun even ride fast |
| 17:05.40 | Maxxed | heh |
| 17:05.47 | Maxxed | oh now ur lie'n ;) |
| 17:05.56 | Maxxed | i couldnt keep it under 3 digits |
| 17:05.57 | Maxxed | heh |
| 17:06.02 | *** part/#asterisk emacsen (hidden-use@gw.coderyte.net) |
| 17:06.03 | smash- | lol |
| 17:06.07 | `Sauron | My problem with wheeled vehicles, is that the speed at which I scare myself, is above the speed that would kill me... |
| 17:06.22 | smash- | i dont try and scare my self |
| 17:06.26 | smash- | ive had fast cars all my life |
| 17:06.27 | `Sauron | And luck only gets you out of _so_ many accidents. |
| 17:06.28 | Maxxed | cut me a deal ;) |
| 17:06.48 | _Sam-- | configure my extensions.conf and we may have a deal. |
| 17:06.51 | Nugget | yay fast cars. |
| 17:06.52 | Maxxed | heh |
| 17:07.02 | Maxxed | i might beable to do somthing, wha cha trying to do |
| 17:07.09 | `Sauron | smash: Ever seen a fullsize pickup do 4-wheel slides around corners in the city? |
| 17:07.17 | smash- | no |
| 17:07.26 | NetOfSickCoder | i have a quesion is possible connect two FXS gateway with asterisk? |
| 17:07.28 | smash- | seen some 240's |
| 17:07.46 | _Sam-- | well right now, i have to wait for someone else to help get my PRI back up and running....after that i need some help with a dialplan...need a menu system (1 for sales etc..), need a directory, and a bunch of stdextensions |
| 17:08.11 | `Sauron | I can within a fairly short amount of driving, tell you exactly HOW far you can push a car/truck/whatever... |
| 17:08.13 | RoyKa | I think I know where Mark picked up C and learned how to program. See http://www.es.ioccc.org/2004/anonymous.c |
| 17:08.13 | Maxxed | ah |
| 17:08.36 | *** join/#asterisk klasstek (~nunyobiz@sta-206-168-218-206.rockynet.com) |
| 17:08.40 | NetOfSickCoder | ? |
| 17:08.43 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 17:08.44 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-12-139.w81-248.abo.wanadoo.fr) |
| 17:08.52 | *** part/#asterisk klasstek (~nunyobiz@sta-206-168-218-206.rockynet.com) |
| 17:09.02 | _Sam-- | sauron: have you ever taken any vehicles around a race track? |
| 17:09.15 | _Sam-- | that is a good way to find how far you can push your stuff. |
| 17:09.20 | `Sauron | Sam: Nah. that costs money :) |
| 17:09.30 | `Sauron | 'sides, I hate 'merrican race tracks. |
| 17:09.46 | Maxxed | hate? |
| 17:09.52 | Maxxed | all that now? |
| 17:09.52 | Maxxed | heh |
| 17:10.02 | `Sauron | Shrug |
| 17:10.04 | Nugget | what is the factor common to all american race tracks? |
| 17:10.10 | _Sam-- | no runoff |
| 17:10.12 | tzanger | Nugget: pavement? |
| 17:10.14 | `Sauron | Insert "strongly dislike" if it makes you feel better. |
| 17:10.16 | *** part/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 17:10.26 | Katty | hmm |
| 17:10.31 | `Sauron | Nugget: They're all either circles, or some derivative of a not-complex circular shape. |
| 17:10.38 | `Sauron | Yawn. |
| 17:10.40 | Nugget | not really. |
| 17:11.00 | `Sauron | s/not/non |
| 17:11.08 | Nugget | sure, Indy is, and talladega is pretty damn boring in its default configuration... but there's good tracks like laguna seca too, and that one in atlanta. |
| 17:11.20 | Nugget | about all america lacks is a nice monte-carlo style street course |
| 17:11.33 | `Sauron | Alternately, the Monaco F1 course. |
| 17:11.42 | smash- | ~pri |
| 17:11.44 | jbot | well, pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 17:12.30 | Maxxed | what do you guys recomend as far as a good helmet goes |
| 17:12.44 | Maxxed | iv been under a shoi for a while now, not so bad |
| 17:12.52 | _Sam-- | the top of the line shoei is as nice as it gets ...X11 |
| 17:13.21 | Maxxed | comfortable? nice ventalation |
| 17:13.35 | Maxxed | i am in texas, and it gets to be one hot mofo down here |
| 17:13.42 | smash- | were can u ride with no helmet? |
| 17:13.44 | _Sam-- | yes and yes...but it is a racing type helmet with tight cheekpads and a tight fit |
| 17:13.53 | _Sam-- | we have no helmet law here in Delaware,....and i know there's no law in FL |
| 17:14.00 | `Sauron | Maxxed: Be like the dumb yuppies around here and ride helmetless. |
| 17:14.11 | Maxxed | nah im cool |
| 17:14.17 | smash- | well |
| 17:14.19 | `Sauron | Then be like the dumb yuppies around here and die. |
| 17:14.27 | smash- | i only drive like |
| 17:14.31 | Maxxed | i have my m endorcment, and here in texas if u have ur M endorce, u can ride w/o helmet |
| 17:14.31 | smash- | 1.3 miles |
| 17:14.34 | Maxxed | but i like my head :p |
| 17:14.41 | Maxxed | cant mess my face up, thats my meal ticket |
| 17:14.42 | Maxxed | heh |
| 17:14.52 | *** join/#asterisk Rick_Hunter (~rhunter@04-177.008.popsite.net) |
| 17:14.57 | afrosheen | sure you're not an english major? |
| 17:15.18 | Maxxed | iv had some nasty laydowns, helmets have saved my arse a maaaaanya times |
| 17:15.30 | johnnyb | Has anyone here compiled asterisk w/ GCC 4.0 w/ its autovectorizer? |
| 17:15.33 | smash- | y you lay it down? |
| 17:15.43 | Katty | that's a fun word. |
| 17:15.46 | Beirdo | but it's your right to be a dumbass and wear no helmet :) |
| 17:15.47 | Katty | autovectorizerimication |
| 17:16.06 | n1gg4s | when use the command "/usr/sbin/safe_asterisk" it shows to the message |
| 17:16.09 | n1gg4s | "Asterisk ended with exit status 1 |
| 17:16.09 | n1gg4s | Asterisk died with code 1. |
| 17:16.09 | n1gg4s | Automatically restarting Asterisk. they anyone knows because? |
| 17:16.15 | Maxxed | its allways been a car, driver comes off into my lane, i have under car and curb as an option, well, there u have it, mess |
| 17:16.16 | Maxxed | heh |
| 17:16.27 | Beirdo | ~seen voipjet |
| 17:16.28 | jbot | voipjet <~helios@ottawa-hs-64-26-155-97.s-ip.magma.ca> was last seen on IRC in channel #asterisk, 13d 22h 19m 43s ago, saying: 'New to test the new Server'. |
| 17:16.54 | Maxxed | and i do a good bit of stuntin too, 90mph wheelies (my mom hates me) and all that jazz |
| 17:16.56 | _Sam-- | sauron have you done any cool php/sql/asterisk integrations? |
| 17:17.03 | Maxxed | so im good about trashing pretty bikes |
| 17:17.03 | Maxxed | heh |
| 17:17.16 | Maxxed | hince! why i am in the market for a nice new helmet |
| 17:17.22 | _Sam-- | like how could i use our distributor information from an SQL table in a softphone for asterisk? |
| 17:17.27 | `Sauron | Sam: Nope. |
| 17:17.40 | `Sauron | I might, some day - but I've got other projects that are more interesting at this point. |
| 17:17.40 | _Sam-- | aside from exporting it as a csv |
| 17:17.45 | *** join/#asterisk J[SS] (~jeremy@chaoscon.user) |
| 17:18.14 | _Sam-- | do you think it would possible using php to have a webpage display phone numbers that our users could click on and that a softphone (or something else) would call? |
| 17:18.26 | zoa | Sam yes |
| 17:19.42 | cjk | anyone here who has some experience with grandsteram and cfg.txt customization? |
| 17:20.26 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 17:23.28 | *** join/#asterisk jsharp (~jsharp@65.90.64.82) |
| 17:24.25 | christo | is it possible to forcibly end a call just by sticking cd_time_end=NOW() in the cdr? |
| 17:25.19 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 17:26.22 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 17:26.36 | *** join/#asterisk PTG123 (~PTG123@66.213.239.122) |
| 17:27.40 | *** join/#asterisk DrFrancky (~chaos@pirus.securax.be) |
| 17:27.52 | _Sam-- | hey dr |
| 17:27.59 | DrFrancky | _Sam--: ye sam |
| 17:28.03 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:28.11 | _Sam-- | hi, i need more of your talents |
| 17:28.22 | *** join/#asterisk Cherebrum (~jgarland@216.32.77.10) |
| 17:28.30 | DrFrancky | :-)) |
| 17:28.34 | DrFrancky | i am here |
| 17:32.14 | *** join/#asterisk nirs (~nirs@62.90.49.115) |
| 17:32.19 | nirs | hey all |
| 17:32.39 | nirs | has anyone got access to the ITU-T Q.931 documents ? |
| 17:33.07 | *** join/#asterisk adjacent_ (scott@nc-65-40-81-77.sta.sprint-hsd.net) |
| 17:33.34 | Cherebrum | Anyone mess with the Cisco 7960 phones? |
| 17:33.37 | Cherebrum | I just got one |
| 17:33.45 | *** join/#asterisk lilwookie (~bender@modemcable215.87-81-70.mc.videotron.ca) |
| 17:33.46 | Cherebrum | and upgraded it to 7.4 firmware |
| 17:33.48 | lilwookie | hi folks |
| 17:34.25 | *** join/#asterisk luisgrin (~luis@209.99.227.220) |
| 17:34.49 | lilwookie | is there a way to set S100I to factory defaults? |
| 17:35.03 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:35.05 | fgravato | iaxy? |
| 17:35.18 | lilwookie | yeah |
| 17:35.26 | fgravato | hrmm good question |
| 17:35.37 | luisgrin | did somebody work with ivr and database? i need advice where to find demos etc |
| 17:35.39 | lilwookie | for some reason the box I have isnt asking for DHCP requies |
| 17:36.00 | fgravato | ahh -- got one thats doing that just now |
| 17:36.02 | luisgrin | i need few analog lines i have a x100p |
| 17:36.06 | kfuq-lap | <PROTECTED> |
| 17:36.08 | fgravato | plus for some reason it overheats |
| 17:36.09 | kfuq-lap | http://www.oag.state.tx.us/oagNews/release.php?id=849 |
| 17:36.17 | fgravato | guessing cause of the power supply |
| 17:36.19 | kfuq-lap | Lawsuit against Vonage first in nation to address 9-1-1 access concerns |
| 17:36.50 | lilwookie | Hrrrm Dang it :) |
| 17:37.39 | fgravato | iopoijkl;kl;jkjklyuiop[yuiopol;' |
| 17:37.42 | fgravato | opps |
| 17:37.51 | lilwookie | anyone now how to reset an IAXY to factory defaults .. is it even possible? |
| 17:38.11 | fgravato | hrmm you try powering it down |
| 17:38.18 | fgravato | and sniff it with ethereal |
| 17:38.24 | fgravato | see if does anything |
| 17:39.35 | lilwookie | yeah |
| 17:39.55 | lilwookie | it doesnt seem to do anything |
| 17:40.06 | fgravato | hrmm guess rma |
| 17:40.09 | fgravato | to digium |
| 17:40.17 | lilwookie | I think it was provinsioed badly |
| 17:40.44 | fgravato | well there's reset |
| 17:40.45 | fgravato | switch |
| 17:40.48 | fgravato | on the unit |
| 17:40.54 | fgravato | i have open one |
| 17:40.56 | fgravato | right now |
| 17:41.09 | fgravato | guess hold down the reset |
| 17:41.12 | fgravato | while power it up |
| 17:41.22 | lilwookie | yeah I have tried it.. powered on while holding.. and such combos |
| 17:41.23 | lilwookie | :) |
| 17:42.12 | jsharp | Its an IAXY! Its a toaster! |
| 17:42.23 | Nugget | it's a floorwax and a dessert topping! |
| 17:42.37 | ManxPower | Today god hates me.\ |
| 17:42.41 | fgravato | nice paper wait |
| 17:42.43 | fgravato | weight |
| 17:42.48 | ManxPower | Well he hates me every day. It's just that today he's doing something about it. |
| 17:44.13 | johnnyb | ManxPower, what's going on? |
| 17:44.23 | Cherebrum | My Cisco 7960(7.4 firmware) isn't requesting the RINGLIST.DAT file from the TFTP server |
| 17:45.04 | ManxPower | johnnyb: I ran out of heart burn meds, a customer's phone system crashed (not Asterisk), and the bitch at the ormond office keeps calling in whining. |
| 17:45.06 | johnnyb | Cherebrum, maybe it thinks your ringtones are l8me |
| 17:45.43 | Cherebrum | it doesn't know of my ring tones |
| 17:45.49 | Cherebrum | it didn't read the ringlist.dat file |
| 17:45.53 | G0shen | Vonage made slashdot once again, but this time they are being sued over no 911 |
| 17:46.18 | *** part/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 17:46.39 | *** join/#asterisk SpaceBass (~sp@24.125.33.214) |
| 17:46.48 | SpaceBass | word |
| 17:46.49 | Ruben_Quinones | ManxPower... In those Cases... I tell the client to disconnect all the equipment and trow it through the window... |
| 17:47.15 | SpaceBass | im having trouble getting an incoming call to ring a call group |
| 17:47.35 | ManxPower | Ruben_Quinones: power cycling the old phone system worked. |
| 17:47.42 | SpaceBass | i tried dial(SIP/702) but it doesnt seem to like that... and the group is not just sip phones |
| 17:48.09 | viKing78 | Anybody ever hooked up * to a Tadiran PBX? |
| 17:48.36 | *** join/#asterisk Lee__ (~Lee__@ool-44c26142.dyn.optonline.net) |
| 17:48.38 | _omer | hi |
| 17:48.45 | SpaceBass | so what kind of extension is a group? |
| 17:49.05 | Lee__ | do extentions have to be four digits? |
| 17:49.05 | _omer | My asterisk is configured...now I want to call it through my Cisco ATA...anyone please give me some idea.. |
| 17:49.11 | Cherebrum | viKing78 : What kind og interfaces are available? |
| 17:49.14 | SpaceBass | Lee__ they can be any lenght |
| 17:49.27 | ManxPower | Spacebar: you can't do CALL GROUPS with SIP. Only PICKUP GROUPS. |
| 17:49.31 | SpaceBass | _omer asterisk@home |
| 17:49.36 | Shido6 | <PROTECTED> |
| 17:49.39 | Shido6 | poor guy |
| 17:49.41 | SpaceBass | ManxPower what do i use to dial a group? |
| 17:49.46 | Shido6 | cant change ur ringtones, Cherebrum ? |
| 17:49.59 | _omer | SpaceBass: Asterisk@home? |
| 17:50.03 | ManxPower | You can fake a SIP CALL GROUP by doing Dial(SIP/happy&SIP/grumpy&SIP/dopey) |
| 17:50.10 | SpaceBass | _omer that was a question.. sorry... are you using asterisk at home? |
| 17:50.24 | SpaceBass | ManxPower ahhhh that will work for what I need, thanks |
| 17:50.25 | _omer | yes.... |
| 17:50.37 | SpaceBass | _omer pick up the phone and dial 1234 |
| 17:50.43 | _omer | my asterisk and CiscoATA..both are at public IP Addresses... |
| 17:50.53 | SpaceBass | _omer you should get a voice recording |
| 17:50.59 | _omer | I want to call at my Asterisk from Cisco ATA.... |
| 17:51.16 | SpaceBass | _omer not sure I follow... what do you want the Asterisk box to do when it answers? |
| 17:52.43 | _omer | I need to call my asterisk from my Cisco ATA over the IP.....??? anyone? |
| 17:54.26 | *** join/#asterisk dano_ (~dano@mail.crosscountrycourier.com) |
| 17:55.06 | *** join/#asterisk Skid (~cm@skid.user) |
| 17:55.09 | ManxPower | _omer: You configure the ATA |
| 17:55.19 | _omer | yes...that's the what I want to know... |
| 17:55.22 | ManxPower | How you do that? Well there are about 400 million pages that tell you. |
| 17:55.34 | ManxPower | ~google site:lists.digium.com configure Cisco ATA |
| 17:55.35 | Skid | hi.. is it possible for me to use our Asterisk server from at home, we're behind NAT - I've heard both it's possible and it's not? |
| 17:55.45 | ManxPower | ~google site:lists.digium.com configure Cisco ATA site |
| 17:55.48 | dano_ | Does asterisk support the phone status lights on mitel 5220's? voip-info.org didn't shed any specific light on this question. |
| 17:56.01 | _omer | wow! |
| 17:56.09 | _omer | thanks I read them out...... |
| 17:56.15 | ManxPower | In other words, RTFG first. |
| 17:56.35 | Delvar | lol |
| 17:56.35 | afrosheen | or FSG |
| 17:56.48 | Lee__ | http://www.fuckinggoggleit.com is my favorite |
| 17:56.58 | _omer | :-/ |
| 17:57.35 | mogorman | aww it doesnt resolve |
| 17:58.28 | tzanger | mogorman: justfuckinggoogleit.com |
| 17:59.04 | mogorman | that resolves |
| 17:59.07 | mogorman | thats awesome |
| 17:59.14 | Nugget | yeah, that site is great. |
| 17:59.39 | Nugget | you can even use it as a query proxy: http://www.justfuckinggoogleit.com/?q=zaprtc |
| 18:00.09 | mogorman | lol |
| 18:02.07 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-50-212.ukonline.co.uk) |
| 18:03.56 | Katty | dododododoooo! i just love on hold music. |
| 18:04.30 | __Sparks_ | Hi, - I seem to be having a problem where if there is no audio being sent one way in a call (Say someone calls and gets the voicemail, then starts leaving a message) after 30 seconds of one way speech, the call gets cut off - This is suing SipGate, so I am unsure if it is them cutting the call, or my asterisk box! - any ideas!? |
| 18:04.39 | Skid | hi.. is it possible for me to use our Asterisk server from at home, we're behind NAT - I've heard both it's possible and it's not? -- does it require patching? |
| 18:05.02 | __Sparks_ | Skid - can you port forwward? |
| 18:05.10 | Skid | yep |
| 18:05.16 | __Sparks_ | then you can do it :) |
| 18:05.16 | Nugget | one side being behind nat is usually manageable, but if both sides are behind nat you're pretty much screwed. |
| 18:05.18 | Skid | I heard that i might have to run asterisk internally ? |
| 18:05.21 | Darwin[laptop] | how do you interface Manx and asterisk and a spiura unit |
| 18:05.32 | Skid | nah it's only home NAT'd cable -> me network |
| 18:06.03 | Nugget | there's a page in the wiki about nat and asterisk. I'm confident it will answer your questions. |
| 18:06.08 | __Sparks_ | Skid - My server is on a private IP address, with potrs forwaded from my router - works fine for me! |
| 18:06.48 | __Sparks_ | Back to my problem - the last thing Asterisk reports when the call is dripped is " User hung up" |
| 18:07.07 | __Sparks_ | Would that be SipGate terminating the call then? |
| 18:07.56 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 18:10.44 | fenlander | __Spakrs_: know what your problem is - it is an issue between the cisco gateway and asterisk - calls are dropped after 30s of no RTP from asterisk |
| 18:12.03 | fenlander | __Sparks_: while leaving voicemail asterisk is silent, hence no RTP so the cisco gateway drops the call thinking the end point has gone away |
| 18:16.48 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 18:16.52 | smash- | hey |
| 18:17.06 | Beirdo | ~seen JerJer[mobile] |
| 18:17.09 | jbot | jerjer[mobile] <~jj@feth100-fw.fament.net> was last seen on IRC in channel #asterisk, 1d 5m 48s ago, saying: 'SexyKen: sure'. |
| 18:17.09 | smash- | what is a good router for use with asterisk |
| 18:17.15 | smash- | im kinda confused about which router to gt |
| 18:17.16 | smash- | to get |
| 18:17.21 | *** join/#asterisk linenoise (~linenoise@cerberus.franklinamerican.com) |
| 18:17.22 | smash- | for 2 voice t1's |
| 18:17.23 | Darwin[laptop] | kram you alive |
| 18:17.24 | Beirdo | hmm, he's not here to complement :) |
| 18:17.32 | __Sparks_ | fenlander - thanks for that, is there any way round it! |
| 18:17.37 | smash- | http://www.cisco.com/en/US/products/hw/routers/ps259/products_data_sheet09186a0080194e20.html |
| 18:17.42 | smash- | is it gonna be a card like that |
| 18:17.53 | Darwin[laptop] | if it has where did he put it |
| 18:18.07 | fenlander | __Sparks_: the real fix is a proper rtcp implementation for asterisk, but I have a patch that fixes it for now if you want it |
| 18:18.23 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 18:19.05 | bannerman | I changed my phones from IAX2 to SIP, fixed them in extensions and stuff, put them in the right context.. but now when I do "#<exten>" to transfer, it just sends the beeps over the line instead of picking up the transfer |
| 18:19.14 | bannerman | and hi everyone :) |
| 18:20.14 | bile_one | bannerman pastebin your extension.conf, sip.conf files |
| 18:20.19 | bannerman | alright |
| 18:20.43 | Cherebrum | smash-: You don't need a router for voice T1s |
| 18:20.55 | nestAr | so.. has anyone written an IVR adventure game for * yet? |
| 18:20.59 | Cherebrum | smash-: just get a pair of T1 PCI cards for your asterisk box |
| 18:21.07 | *** part/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 18:21.09 | Cherebrum | nestAr: I'm trying to |
| 18:21.16 | nestAr | sweet |
| 18:21.22 | Cherebrum | nestAr: I'm trying to find developers to write it actually |
| 18:21.32 | Cherebrum | I'm taking bids |
| 18:21.33 | Cherebrum | :) |
| 18:21.56 | Cherebrum | mine is going to be multiplayer |
| 18:21.57 | Cherebrum | :) |
| 18:22.25 | jsharp | Roll in some speech recognition. |
| 18:23.03 | Darwin[laptop] | sphinx |
| 18:23.08 | Darwin[laptop] | is about it |
| 18:23.39 | jsharp | Sphinx kind of sphucked last time I played with it. |
| 18:24.01 | bannerman | bile_one: http://pastebin.ca/8062 |
| 18:24.13 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 18:24.20 | bannerman | I think I put all of the pertinent info in there |
| 18:24.30 | bile_one | bannerman, give me a second or two to review them |
| 18:25.17 | bannerman | thanks |
| 18:27.25 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 18:28.29 | DrWho17 | looking for TNT Sip <-> Asterisk users here, or the guy who made the wiki entry |
| 18:29.26 | bile_one | bannerman, I don't see a transfer context nor do I see a "t" in your dial plan please see the following: http://www.voip-info.org/wiki-Asterisk+cmd+Transfer |
| 18:29.34 | bannerman | thanks bile |
| 18:29.50 | bannerman | sorry to be such a newb :_/ |
| 18:30.31 | bile_one | bannerman, also you should think about making an incoming context that will handle any pattern of extensions you have. |
| 18:30.36 | bannerman | have to say I've never heard of such a thing :-P I do read the wiki much, but until you know what you're looking for it's difficult to find it. and once you know what you're looking for, it kinda seems to fall into lpace anyway :) |
| 18:30.59 | bannerman | I will figure out what you mean by that, and look into doing it :) thanks |
| 18:31.46 | harryvv | btw was having some issus with connecting with voipjet yesterday. I was going to include a failover link to say iax.cc with a voice anoucment in between them in the event that does happen. What is the typical timout period for a failed connection to a voip service before such a failover would occure? |
| 18:32.09 | bile_one | bannerman I can paste you an example hold on |
| 18:32.20 | harryvv | Tried doing that yesterday and it did not fail over to my zap dialout |
| 18:32.21 | bannerman | bile_one: that would be great |
| 18:34.18 | dano_ | anyone else have problems with the first part of voice prompts (like the agentlogin() app) getting cut off on sip phones? |
| 18:34.58 | Maxxed | wtf? |
| 18:34.58 | Maxxed | root@NRG-PBX ~# modprobe zaptel |
| 18:34.59 | Maxxed | while this kernel is version 2.4.29-3tr. |
| 18:34.59 | Maxxed | modprobe: insmod /lib/modules/2.4.29-3tr/misc/zaptel.o failed |
| 18:34.59 | Maxxed | modprobe: insmod zaptel failed |
| 18:35.13 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-136-89.buff.east.verizon.net) |
| 18:35.22 | SuPrSluG | hello |
| 18:35.37 | Maxxed | modprobe: insmod /lib/modules/2.4.29-3tr/misc/zaptel.o failed |
| 18:35.43 | Maxxed | modprobe: insmod /lib/modules/2.4.29-3tr/misc/zaptel.o failed |
| 18:35.44 | Maxxed | ? |
| 18:36.00 | Maxxed | i have a digi tdm400p |
| 18:36.07 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 18:37.21 | _omer | extensions.conf ....is this file handles only internal calls?? |
| 18:37.39 | SuPrSluG | i'm thinking of getting a few Polycom 300's for a customer. anyone use them. any horror stories I should know before buying them? |
| 18:37.49 | jhiver | what's the term for the time that is spent trying to connect your call? |
| 18:37.51 | *** join/#asterisk mxmasster|work (~Max@rottie.media.net) |
| 18:37.52 | mxmasster|work | hi all |
| 18:37.54 | jhiver | ASN? ASR? Forgot! |
| 18:38.19 | RoyK | AA |
| 18:38.28 | jhiver | AA? What's it mean? |
| 18:38.37 | bile_one | bannerman look at this: http://pastebin.ca/8064 |
| 18:39.10 | bannerman | bile_one that makes a lot of sense |
| 18:39.22 | bannerman | bile_one -- although -- before, when I hit "#", it would go silent until I was done dialing |
| 18:39.25 | RoyK | yellowdog linux on powerbook :) |
| 18:39.26 | RoyK | this rock |
| 18:39.27 | RoyK | s |
| 18:39.36 | bannerman | now, it jsut sends the "#" tone across the line |
| 18:39.42 | bannerman | is that because I don't haev anywhere for it to go? |
| 18:39.57 | smash- | ~ft1 |
| 18:40.04 | bile_one | do you have a transfer context? |
| 18:40.06 | dano_ | SuPrSluG: I'm using 500's with good results |
| 18:40.28 | bile_one | bannerman yes that is correct |
| 18:40.36 | bannerman | ok, thanks. I'll get to work |
| 18:40.38 | SuPrSluG | me too. would like hear about 300's? |
| 18:41.00 | SuPrSluG | or if there are better phones for the buck. |
| 18:41.24 | bile_one | bannerman, your current setup is sending the call to transfer, which in your example is supposed to be a context. |
| 18:41.38 | dano_ | 300s use the same firmware as the 500s so beyond that it's all physical feature differences & fewer "lines" available...I believe. |
| 18:41.58 | bile_one | bannerman what phone are you using? |
| 18:42.01 | SuPrSluG | dano_:thanx |
| 18:42.06 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 18:42.40 | SuPrSluG | so i'll go w/ them. unless you know of a better phone for the lower end. |
| 18:42.42 | mxmasster|work | hi all |
| 18:42.43 | bannerman | bile_one: ariaVoice Atlas |
| 18:42.46 | dano_ | and no speakerphone which is sucky. |
| 18:43.01 | bannerman | bile_one: it has an IAX firmware, which I thought would be spiffy, but it doesn't work good |
| 18:43.02 | *** join/#asterisk a1fa (~a1fa@ip70-178-46-30.ma.dl.cox.net) |
| 18:43.03 | a1fa | yo |
| 18:43.08 | a1fa | anybody have sipura-2000? |
| 18:43.12 | mxmasster|work | quick question... on the "s" extension, when the digit timeout is reached how do i forward the call to an extension? |
| 18:43.12 | bannerman | the sip firmware seems to work better. |
| 18:43.55 | dano_ | SuPrSluG: not that I'm aware of. Polycom's soundpoint line is very good. |
| 18:43.56 | a1fa | for some reason, line 1 authenticates, while line two doesnt |
| 18:44.04 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 18:44.05 | Beirdo | a1fa: I have a 2100, same difference |
| 18:44.12 | bile_one | Yes I agree, IAX over ZAP has a huge amount of echo too. Hopefully it iwll be fixed one day |
| 18:44.20 | a1fa | Beirdo : you got broadvoice,right? |
| 18:44.26 | Beirdo | no |
| 18:44.43 | smash- | hey does anyone know the name of wildcard that has a QOS proc on the card? |
| 18:44.48 | a1fa | Beirdo : for some odd reason.. it wont register my 2nd line |
| 18:44.49 | smash- | er brand |
| 18:44.51 | bile_one | VOnage is being sued in Texas too for 911 service |
| 18:44.53 | a1fa | i am getting pissed |
| 18:44.53 | Shido6 | heh |
| 18:44.59 | Shido6 | dont get pissed |
| 18:45.02 | a1fa | bile_one : why? |
| 18:45.07 | a1fa | Shido6 : tru |
| 18:45.07 | smash- | for not having 911 |
| 18:45.12 | smash- | when soemone needed it |
| 18:45.16 | harryvv | anyone who goes with vonage is dumb there termination points are back east. |
| 18:45.22 | a1fa | it says limited 911 |
| 18:45.23 | a1fa | :) |
| 18:46.02 | bile_one | Vonage failed to clearly inform customers that its service excludes 911 services |
| 18:46.11 | Beirdo | bile_one: the plaintiffs will lose, their customer agreement specifically says that the 911 is excluded |
| 18:46.19 | Beirdo | quite clearly |
| 18:46.22 | Beirdo | I've read it |
| 18:46.22 | a1fa | yup |
| 18:46.24 | harryvv | what about primus do thay include 911? |
| 18:46.31 | a1fa | fuck 911 |
| 18:46.32 | Beirdo | likely not |
| 18:46.37 | a1fa | just call local poo-ulice! |
| 18:46.44 | a1fa | poo-lice! |
| 18:46.53 | harryvv | alfa, if somone is breaking into your home and are armed wouldnt you want it? |
| 18:46.56 | topping | i've had vonage for over a year and I got slapped in the face with 911-this and 911-that tons of times |
| 18:47.06 | topping | this is like having to warn people that the coffee is hot |
| 18:47.06 | a1fa | harryvv : HAVE A GOD DAMN SPEED DIAL FOR 911 THEN |
| 18:47.12 | a1fa | #91 |
| 18:47.12 | smash- | man dont anyone have a cell phone |
| 18:47.12 | topping | americans are stupid is the probel |
| 18:47.15 | topping | problem |
| 18:47.15 | a1fa | instead of 911 |
| 18:47.17 | harryvv | enough alfa |
| 18:47.23 | a1fa | ;) |
| 18:47.25 | a1fa | enough said |
| 18:47.32 | smash- | hey |
| 18:47.34 | a1fa | they deserved to get hurt ;) |
| 18:47.42 | harryvv | Well you did a bad job expressing your self. |
| 18:47.48 | DrWho17 | topping: what kind of number does vonage hand out? If your neighbor across the stree tries to call you will he get billed as long distance? |
| 18:47.50 | a1fa | tru |
| 18:47.52 | smash- | so im needing a little info would be glad if someone could help me, Im trying to find T1 Voice routers. |
| 18:47.55 | SpaceBass | i always hear the cell argument, and its a poor one... if my wife is home alone and someone breaks in, she is not going downstairs to find her cell phone in her purse which isnt charged |
| 18:48.07 | bile_one | harryvv, all VoIP solutions have trouble with 911 services. Asterisk can use a local POTS line but that is for the LOCAL box not for me if I am in St. Louis, and my box is in New Orleans. |
| 18:48.19 | topping | DrWho17: i always wanted service in a metro area so never a problem with area code |
| 18:48.21 | harryvv | I know |
| 18:48.29 | a1fa | there is always a real number that is binded to 911 |
| 18:48.34 | harryvv | unless the service is terminated locally |
| 18:48.35 | a1fa | just ask what is the real # |
| 18:48.42 | a1fa | end of story.. setup a speed dial |
| 18:48.47 | a1fa | or have a prepaid cellphone |
| 18:48.49 | smash- | i have 46 real pots in my building |
| 18:48.51 | a1fa | they cost $20 |
| 18:48.52 | smash- | with 100 voip's |
| 18:48.54 | a1fa | and they have 911 |
| 18:48.56 | Beirdo | heh |
| 18:48.58 | SpaceBass | a1fa that seems to be the best solution |
| 18:49.00 | a1fa | they cost $20 / life-time |
| 18:49.01 | smash- | so just no one can dail 911 on a softphone |
| 18:49.03 | DrWho17 | topping: yes, hrm I mean how are people who call you billed? |
| 18:49.05 | Beirdo | or just get a cellphone and not activate it |
| 18:49.08 | smash- | so no one better try and use a computer to call 911 |
| 18:49.11 | smash- | while someones having a heart attack |
| 18:49.13 | SpaceBass | and from what I hear 911 doesnt mind test calls |
| 18:49.15 | smash- | they better call a manager |
| 18:49.18 | Beirdo | they legally MUST allow 911 on all cellphones activated or not |
| 18:49.20 | a1fa | Beirdo : pre-payed cell phone, activated or not.. you can always call 911 |
| 18:49.21 | DrWho17 | does Vonage respect local calling rules |
| 18:49.23 | BrianR___ | The 911 location problem could be solved pretty easily. A good interim solution would be mapping 911 to a 10 digit emergency number with in-band signalling of the location. |
| 18:49.24 | DrWho17 | just wondering about that |
| 18:49.28 | topping | DrWho17: dunno, never asked |
| 18:49.33 | G0shen | Map 911 to dial local dispatch |
| 18:49.43 | a1fa | YUP |
| 18:49.43 | smash- | but 911 can trace your call |
| 18:49.45 | Beirdo | just get a damn craptacular cellphone for 911 |
| 18:49.46 | Beirdo | :) |
| 18:49.46 | smash- | which is good |
| 18:49.47 | DrWho17 | BrianR___: well, you need a live hookup into a routing database |
| 18:49.47 | a1fa | no |
| 18:49.51 | topping | DrWho17: everyone i know has nationwide plans and rarely has a home phone |
| 18:49.52 | a1fa | 911 cant trace your call |
| 18:49.54 | smash- | yes Beirdo |
| 18:49.56 | G0shen | You don't need cell service to call 911 |
| 18:50.03 | a1fa | if you have un-acitvated cellphone :) |
| 18:50.06 | a1fa | i got two of them |
| 18:50.06 | smash- | they can lookup the number |
| 18:50.08 | G0shen | any cellphone can dial 911, and the providers have to honor it |
| 18:50.09 | DrWho17 | if the end user is going to be responsible for changing it on a web interface that is a bit of a nightmare |
| 18:50.10 | smash- | of the call i mean a1fa |
| 18:50.11 | Beirdo | you need battery life and for the phone to see a cell tower |
| 18:50.13 | Beirdo | that's it |
| 18:50.15 | smash- | and respond to the location |
| 18:50.17 | smash- | if its a pot |
| 18:50.26 | DrWho17 | they can put in false locations which will break 911 |
| 18:50.26 | G0shen | so gather up some old cellphones, charge them up, and slide the battery off the contacts |
| 18:50.31 | a1fa | someone help me with sipura 2000 and broadvoice |
| 18:50.39 | a1fa | i have 4 accounts with them |
| 18:50.44 | G0shen | a1fa: sipura 2000 directly to broadvoice? |
| 18:50.48 | a1fa | yup |
| 18:50.49 | Beirdo | buy an old cellphone off ebay and some zinc-air batteries :) |
| 18:50.54 | a1fa | i had to kill my asterisk box |
| 18:50.57 | topping | having voip only without a cell is somewhat irresponsible if you are accident-prone |
| 18:50.57 | SpaceBass | Beirdo thats the key... who wants to have to worry about battery and cell towers in an emergency |
| 18:51.03 | BrianR___ | DrWho17: Someone who calls an emergency service from a non e911 capable line can always lie about their location. That's not an excuse. |
| 18:51.06 | SpaceBass | sorry... can't call the ambulance, no signal |
| 18:51.11 | G0shen | a1fa: did you select sipura as your client? they send you a setup file if you do |
| 18:51.16 | *** join/#asterisk sd-tux (user2267@emasq.stusta.mhn.de) |
| 18:51.28 | Beirdo | so don't go with a VoIP provider and expect it to be better. |
| 18:51.32 | a1fa | G0shen : i have to lines |
| 18:51.33 | bile_one | this will help muddy the waters: http://www.911dispatch.com/information/voip.html |
| 18:51.36 | DrWho17 | BrianR___: well, but the 911 switchboard has the facility to see the real number |
| 18:51.37 | a1fa | they sent me config for line 1 |
| 18:51.44 | a1fa | but i manually configured line 2 |
| 18:51.45 | Beirdo | Oh damn, my DSL went down, now I can't call 911.... that's just dumb |
| 18:51.45 | topping | but then, we've survived millions of years without 911 (and phones) |
| 18:51.46 | G0shen | a1fa: ahh |
| 18:51.48 | SpaceBass | Beirdo true... I'm using ptsn so its not an issue for me yet |
| 18:52.11 | DrWho17 | 911 here is routed at the local switch here with dedicated trunks to each call center |
| 18:52.12 | BrianR___ | Beirdo: That's a non-argument. Typically if your DSL line has been cut, the POTS line it rides on is also dead. |
| 18:52.22 | Beirdo | I've lived 31 years and only ever had to call 911 once. |
| 18:52.42 | G0shen | Find a local number for 911 and map that to 911 in your dialplan |
| 18:52.43 | Beirdo | BrianR___: heh, DSL goes down all the time when the DSLAM buggers up, etc |
| 18:52.45 | a1fa | i call 911 almost every day |
| 18:52.48 | harryvv | Beirdo yea thats true. Before I sell a system to a company i want to know which dsl/cable service thay sell. If it has a bad rep of going down alot I would say sorry cannot sell voip service but can sell a pbx with pstn termination. |
| 18:52.53 | topping | my neighbors are married gay terrorists and they call 911 to report bombs every day |
| 18:52.53 | a1fa | crazy mexicans get drunk and beat on their wife |
| 18:53.00 | bile_one | and this one: http://www.911voip.org/ |
| 18:53.36 | a1fa | you can cancel your phone service, and still be able to dial 911 |
| 18:53.37 | a1fa | BTW!!! |
| 18:53.49 | a1fa | as long as they didnt disconnect the wires from your phone dist box |
| 18:53.54 | a1fa | which they usually dont |
| 18:54.04 | a1fa | they just dont route your calls xcept than 911 |
| 18:54.05 | topping | bile_one: are these sites put out by the same people that brought you homeland security and apple pie? |
| 18:54.06 | jhiver | well, maybe it's due to poor standards of living or other factors rather than being mexican :-/ |
| 18:54.07 | Beirdo | true |
| 18:54.18 | Beirdo | and 611 (repair service) likely |
| 18:54.22 | a1fa | home land security < gay |
| 18:54.24 | a1fa | try |
| 18:54.25 | topping | talk about FUD! |
| 18:54.31 | G0shen | a1fa: cellphones are the same way...you can call 911 if you don't have service |
| 18:54.49 | SpaceBass | how can i read in digits to a variable? |
| 18:54.55 | BrianR___ | In-band playback of your location and callback number addresses the majority of temporary signalling problems... |
| 18:54.56 | a1fa | G0shen : 611 also |
| 18:54.58 | G0shen | Qwest disconnects you line once you discontinue service I beleive |
| 18:55.01 | bile_one | topping, I have not a clue, but it is an important issue when having to deal with lots of cunsumer based applications |
| 18:55.02 | SpaceBass | ie if i want to read them in and then dial them on another trunk? |
| 18:55.08 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 18:55.11 | harryvv | Thats what the 911 fees are for on the cell bill |
| 18:55.18 | *** join/#asterisk marno (~marno@213-182-127-196.teleos-web.de) |
| 18:55.19 | G0shen | you loose your dialtone completely when you disconnect from Qwest |
| 18:55.24 | marno | hi |
| 18:55.33 | harryvv | Goshen thanks that is nice to know. |
| 18:55.35 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:56.02 | marno | i've got a problem and i don't know why |
| 18:56.13 | BrianR___ | In my area, when I ended POTS service the line still had battery/ground, but no dialtone. |
| 18:56.15 | G0shen | harryvv: we keep our old cellphone at our headboard fully charged with the battery slid off so it doesn't discharge |
| 18:56.15 | marno | in my extensions.conf i use sipgetheader |
| 18:56.23 | a1fa | OMG |
| 18:56.24 | marno | then i get pbx_extension_helper: No application 'SIPGetHeader' |
| 18:56.25 | G0shen | it doesn't have service, but you can still dial 911 with it |
| 18:56.26 | a1fa | i am getting pissed |
| 18:56.28 | a1fa | at broadvoice |
| 18:56.42 | drumkilla | marno: that application is only in CVS HEAD |
| 18:56.45 | G0shen | a1fa: you call them yet? |
| 18:56.49 | jesster | anyway i can change my 2833 DTMF options to only advertise 100 0-15 instead of 100 0-16 (the 16 is Flash) |
| 18:57.04 | marno | drumkilla, ok thats the problem :) |
| 18:57.12 | Lee__ | does asterisk load the zaptel kernel module when needed or do I have to do it manually? |
| 18:57.21 | drumkilla | Lee__: manually |
| 18:57.24 | DrWho17 | Lee__: depends on your init script |
| 18:57.36 | marno | the problem is, i have 50 local telephone numbers on one sip account |
| 18:57.36 | Lee__ | I'm using the Debian packages in Sarge |
| 18:57.41 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 18:57.43 | harryvv | Goshen sorry to tell you this but all MnHi Nicad Li have a cirtain drain rate every day. The only batteries that dont are alcaline. |
| 18:57.54 | marno | i will have to use the from-line |
| 18:58.04 | Beirdo | harryvv: and zinc-air |
| 18:58.14 | marno | but how can i read the the dialed-number? |
| 18:58.24 | G0shen | harryvv: sure, but you know how long it lasts once fully charged and not on the phone? months |
| 18:58.30 | Beirdo | don't let the air in, it's not activated. once the air is in there it has a shelf-life |
| 18:58.34 | harryvv | Beirdo, 2 years ago read about zinc air batteries..what if there was a pin hole in its plastic covering? |
| 18:58.38 | G0shen | I check it once a month, and charge it about every 6 months |
| 18:58.54 | harryvv | Goshen unless there is some internal resistance. |
| 18:59.04 | *** join/#asterisk heison (~heison@w3.somanetworks.com) |
| 18:59.12 | Beirdo | harryvv: jeez. then bang on the door of the neighbour's house |
| 18:59.13 | Lee__ | how do I find the name of the module for a digium FXO PCI card? |
| 18:59.20 | marno | no idea? |
| 18:59.24 | Beirdo | mankind lived a LONG time without 911 |
| 18:59.41 | harryvv | Beirdo hehe not for the people who live remotly |
| 18:59.43 | denon | mankind also used to die at the age 30 |
| 18:59.45 | SpaceBass | how can i read in digits to a variable? |
| 18:59.45 | SpaceBass | ie if i want to read them in and then dial them on another trunk? |
| 18:59.51 | jhiver | Beirdo: yeah. Actually I was thinking the other day we might as well go back to living in caves |
| 18:59.58 | denon | people used to die of the flu |
| 19:00.33 | a1fa | Line 2 Status |
| 19:00.33 | a1fa | Hook State: On Registration State: Failed |
| 19:00.33 | a1fa | bastards |
| 19:00.33 | a1fa | everything is set right |
| 19:00.33 | G0shen | a2fa: you call them yet? |
| 19:00.33 | a1fa | i am about to call them |
| 19:00.34 | a1fa | and bitch at them |
| 19:00.39 | G0shen | ask nice first ;) |
| 19:00.42 | a1fa | they dont even have my area code # |
| 19:00.45 | *** join/#asterisk widowlicker (~Naturalbl@62.77.178.121) |
| 19:00.47 | a1fa | (Arkansas) |
| 19:00.49 | widowlicker | hi there |
| 19:00.55 | a1fa | i bought 7 accounts from them |
| 19:00.59 | a1fa | :) |
| 19:01.04 | G0shen | they just added mine, when I asked about it |
| 19:01.18 | a1fa | they added your 2nd line? |
| 19:01.18 | SpaceBass | do AGIs have to peral only? or can they be a shell script? |
| 19:01.19 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 19:01.25 | G0shen | it was funny, they didn't have my local area, so I signed up and requested LMP, and they had my area 6 days later |
| 19:01.28 | widowlicker | any one know how to set up a call through a modem |
| 19:01.34 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 19:01.39 | G0shen | a1fa: no my area code |
| 19:01.41 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 19:01.44 | jaxxan | hey ya'll |
| 19:01.47 | G0shen | my calls go through asterisk |
| 19:01.49 | SpaceBass | widowlicker check e-bay for a x100p |
| 19:01.57 | a1fa | just do caller ID override |
| 19:02.06 | G0shen | widowlicker: www.digitnetworks.com |
| 19:02.14 | SpaceBass | a1fa what VOIP carriers support caller id override? |
| 19:02.19 | a1fa | vonage |
| 19:02.20 | a1fa | does |
| 19:02.29 | Maxxed | x100p's are a dime a dozen on ebay |
| 19:02.29 | SpaceBass | really? cool |
| 19:02.30 | a1fa | packet8, does, methinks |
| 19:02.32 | SpaceBass | wonder if BV does |
| 19:02.35 | G0shen | you can't set your own callerid with vonage can you? |
| 19:02.41 | harryvv | yea callerid=mickey mouse |
| 19:02.45 | widowlicker | will a normal voice/data/fax modem not work |
| 19:02.53 | Maxxed | if your like me however and dont have any room (1U rack) go with the TDM400P |
| 19:02.54 | terrapen | dammit, high speed internet is such a rip-off |
| 19:02.54 | a1fa | G0shen : i dont think they BYOD |
| 19:02.55 | G0shen | BV doesn't allow you to set your own caller id, you call in and have them set it |
| 19:03.07 | G0shen | vonage doesn't have BYOD no |
| 19:03.07 | a1fa | thats what i am going to do right now |
| 19:03.08 | terrapen | if i cancel my digital cable, my RoadRunner service goes up to $70/mo |
| 19:03.11 | SpaceBass | widowlicker it might- from what i understand- if its a voice modem supported by linux telephony project |
| 19:03.27 | *** part/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 19:03.32 | a1fa | G0shen : did they add the 2nd line into your webconfig? |
| 19:03.44 | SpaceBass | terrapen Comcast does the same thing |
| 19:03.45 | Maxxed | has anybody used * with time warners digital phone service? |
| 19:03.46 | G0shen | the funny thing is...when I call qwest numbers it says Broadvoice INC, when I call the university of utah it is the right caller ID |
| 19:03.55 | widowlicker | it is showing up when i do lspci |
| 19:03.59 | Maxxed | just curious |
| 19:04.01 | G0shen | a1fa: I dropped my first line once they got my area code |
| 19:04.09 | G0shen | a1fa: because it was free to change |
| 19:04.15 | G0shen | if..I dropped the old number |
| 19:04.26 | a1fa | i have 7 accounts with them |
| 19:04.32 | G0shen | nice :) |
| 19:04.40 | a1fa | all residental :) |
| 19:04.52 | terrapen | it reaally sucks |
| 19:04.56 | G0shen | I will change over my business line when they LMP my number |
| 19:04.57 | terrapen | i want to pay $40/mo for high speed |
| 19:04.59 | terrapen | and no more |
| 19:05.06 | terrapen | i dont want cable tv |
| 19:05.07 | Maxxed | who dosent ;) |
| 19:05.07 | G0shen | terrapen: good luck :) |
| 19:05.13 | terrapen | or "digital phone" |
| 19:05.14 | a1fa | err.. it really doesnt say "residenta" |
| 19:05.15 | Maxxed | heh |
| 19:05.19 | a1fa | it doesnt say anything |
| 19:05.21 | a1fa | they are stupid |
| 19:05.37 | G0shen | a1fa:? |
| 19:05.41 | Maxxed | im so broke, i cant afford to pay attention, but best belive i gots to have my broadband |
| 19:05.49 | Maxxed | il sell freakin cans to keep my cable |
| 19:05.49 | Maxxed | heh |
| 19:05.51 | G0shen | lol |
| 19:05.58 | a1fa | G0shen ; their plans are not business/residental oriented |
| 19:06.06 | a1fa | well, they have two business plans |
| 19:06.10 | *** join/#asterisk mrtwister (~mrtwister@cable7107.tele2internet.lt) |
| 19:06.20 | a1fa | err, one, but others are not "explict" residental plans |
| 19:06.47 | jaxxan | anyone know if the Cisc 7935 works with Asterisk ? |
| 19:06.54 | Maxxed | it does |
| 19:06.57 | Maxxed | like a champ |
| 19:07.04 | jaxxan | excellent |
| 19:07.04 | BuckRogers | if it runs sip u should be good |
| 19:07.07 | BuckRogers | skinny too |
| 19:07.20 | terrapen | some of my friends are so lucky...they can steal internet from their neighbors wifi |
| 19:07.28 | Maxxed | lucky sob's |
| 19:07.28 | jaxxan | i was tossing the 7935 and the polycom soundstation 4000 around |
| 19:07.34 | Maxxed | i live in the gehtto |
| 19:07.40 | BuckRogers | thats not stealing if the neighbor does nothing to prevent it |
| 19:07.41 | Maxxed | like i am the only hotspot for 20 miles |
| 19:07.42 | Maxxed | heh |
| 19:07.44 | G0shen | terrapen: put up a dish and point it at a local hotspot ;) |
| 19:07.45 | *** join/#asterisk miguellinux (~miguellin@64.76.202.2) |
| 19:08.38 | G0shen | you get lots of gain with a parabolic antenna |
| 19:08.38 | BuckRogers | g0shen: i got one with an amp that we play with for longdistance links |
| 19:08.39 | BuckRogers | goes great with an omni |
| 19:08.49 | BrianR___ | use a usb wifi adaptor mounted at the focus point of a dish so you don't have to worry about part 15 as much :) |
| 19:08.54 | BuckRogers | within FCC limits |
| 19:09.49 | Maxxed | they couldnt prove shit, but they knew it was me |
| 19:09.50 | Maxxed | heh |
| 19:10.02 | BrianR___ | My town has aparently taken the 10 digit numbers for police/fire emergency off their web site... I wonder if the numbers still work.. |
| 19:10.09 | a1fa | broadnigga on the phone |
| 19:10.13 | a1fa | i am going to bitch them out |
| 19:10.24 | BuckRogers | BrianR did you adjust for the distance from dish to emitter to take into acount the change of source rf |
| 19:10.58 | BuckRogers | BrianR__:call the precent |
| 19:11.02 | BrianR___ | BuckRogers: The actual emitter needs to be at the focus point, so it does require a little twiddling.. |
| 19:11.24 | BuckRogers | i see |
| 19:12.00 | BrianR___ | BuckRogers: But if you're using a dish originally designed to hold a LNB, you're going to need a bit of a spacer anyway... |
| 19:12.07 | Maxxed | root@NRG-PBX ~# modprobe zaptel |
| 19:12.07 | Maxxed | */lib/modules/2.4.29-3tr/misc/zaptel.o: kernel-module version mismatch |
| 19:12.07 | Maxxed | */lib/modules/2.4.29-3tr/misc/zaptel.o was compiled for kernel version 2.4.27-4tr |
| 19:12.07 | Maxxed | while this kernel is version 2.4.29-3tr. |
| 19:12.07 | Maxxed | modprobe: insmod */lib/modules/2.4.29-3tr/misc/zaptel.o failed |
| 19:12.07 | Maxxed | modprobe: insmod zaptel failed |
| 19:12.15 | Maxxed | i did a make clean; make install :\ |
| 19:12.25 | a1fa | broadvoice support is so fucked |
| 19:12.28 | Maxxed | what args do i pass along to compile for my kernel ver ? |
| 19:12.28 | BrianR___ | Maxxed: Built against wrong kernel headers. |
| 19:12.40 | a1fa | they dropped mine call |
| 19:12.47 | Maxxed | BrianR___: I got that much, what should I change? |
| 19:12.51 | G0shen | a1fa: I love broadvoice support, they have been great so far |
| 19:13.11 | Lee__ | anyone here using Debian and the zaptel-source package with module-assistant? |
| 19:13.18 | Maxxed | BrianR___: downgrade the kernel? |
| 19:13.23 | BrianR___ | Maxxed: If you don't have the parameters for the dish you'll need to measure. |
| 19:13.23 | a1fa | G0shen ; i am on hold for ever, and then they dropped me |
| 19:13.38 | a1fa | Control the type of phone calls you can make. BroadVoice may reset the settings when the account is not in good standing, e.g. Non-Payment. |
| 19:13.41 | BrianR___ | Maxxed: Make sure zaptel gets compiled against the headers that go with the kernel you plan to run. |
| 19:13.42 | a1fa | i like this new option |
| 19:13.45 | G0shen | a1fa: did you call them on your broadvoice phone? I usually get through to them in 1-3 minutes |
| 19:14.04 | G0shen | a1fa: they answered on a Saturday, which earned many points in my book too |
| 19:14.06 | DrWho17 | Maxxed: sounds like your kernel source doesn't match your running kernel |
| 19:14.10 | a1fa | yeah |
| 19:14.11 | DrWho17 | do a uname -a |
| 19:14.14 | a1fa | once a while |
| 19:14.18 | a1fa | they will answer immediatley |
| 19:14.28 | BrianR___ | Maxxed: Or take a best guestimate. If you have the LNB that was originally supposed to go with the dish you can usually figure out what the intended focus point was and mount your USB wifi adaptor accordingly. |
| 19:14.35 | a1fa | i like the account portal.. they added new options |
| 19:14.35 | Maxxed | DrWho17: yeah, thats what im thinkin |
| 19:14.44 | DrWho17 | that's all it can be |
| 19:14.51 | terrapen | this stupid time warner salesperson threatened that if i wanted to reconnect my cable service, that it would be much more expensive |
| 19:14.55 | Maxxed | BrianR___: wifi what now ? wrong nick bud ;) |
| 19:14.56 | DrWho17 | if you did a make clean before compiling again, wrong kernel headers |
| 19:14.58 | G0shen | a1fa: yea, the broadvoice account portal is really nice |
| 19:14.59 | terrapen | i told her, "No problem, I'll just move to Grande" |
| 19:15.08 | terrapen | (Grande Communications is their competitor) |
| 19:15.28 | a1fa | the best so far |
| 19:15.42 | Maxxed | DrWho17: yeah, i got some funky headers, were are they located? the kernel source? and what whould u recomend me do as far as overwriting ? |
| 19:15.58 | riksta | Maxxed: what distr |
| 19:15.58 | riksta | o |
| 19:16.10 | Maxxed | DrWho17: this looks like a mess up on the trustix guys |
| 19:16.20 | Maxxed | riksta: Trustix Linux 2.2 |
| 19:16.32 | riksta | Maxxed: hmm i don't really know anything about that |
| 19:16.40 | marno | i installed the cvs-version, but there is the same problem.... No application 'SIPGetHeader' |
| 19:16.44 | Maxxed | riksta: pretty much a bare bones linux distro, kinda feels like redhat |
| 19:16.48 | G0shen | Maxxed: just upgrade to 2.6 :) |
| 19:16.58 | Maxxed | Goshen: is it stable? |
| 19:17.07 | Lee__ | do zaptel drivers compile against 2.6? |
| 19:17.07 | DrWho17 | Maxxed: well, I'm not sure what trustix uses for package management |
| 19:17.12 | Maxxed | Goshen: iv been out of the loop for a goooood long while |
| 19:17.18 | G0shen | I use 2.6.10 to run my clinic server, web server, asterisk...yea it rock solid |
| 19:17.19 | a1fa | they need to change their hold music |
| 19:17.20 | DrWho17 | Maxxed: if you manage the box you should know |
| 19:17.22 | a1fa | it is driving me nuts |
| 19:17.31 | Maxxed | DrWho17: RPM |
| 19:17.32 | G0shen | I think they are up to 2.6.11 stable now |
| 19:17.47 | DrWho17 | Maxxed: well just update the kernel and kernel-headers and kernel-sources |
| 19:17.48 | G0shen | a1fa: yea, their hold music is the same every time, makes me laugh |
| 19:18.00 | *** join/#asterisk AgiNamu (~Bob@12.172.224.49) |
| 19:18.02 | DrWho17 | Trustix doesn't use up2date or whatever though probably |
| 19:18.06 | Maxxed | DrWho17: ok, sounds easy enuff :) thanks! |
| 19:18.13 | AgiNamu | Well, hello there. |
| 19:18.18 | DrWho17 | although on FC4 test 1 I couldn't compile asterisk |
| 19:18.25 | Maxxed | DrWho17: they have some variation of up2date, suwp |
| 19:18.25 | DrWho17 | but I think that was a GCC 4 thing |
| 19:18.32 | AgiNamu | FC4? didnt they just get FC3 out the door? :P |
| 19:18.45 | Maxxed | DrWho17: or swup, somthing like that |
| 19:18.46 | DrWho17 | AgiNamu: heh FC4 test 1 was released a month ago |
| 19:18.58 | DrWho17 | or so |
| 19:19.01 | Maxxed | im going to just upgrade the kernel n what not, see how that fairs me |
| 19:19.10 | DrWho17 | well you need to do more then that |
| 19:19.18 | DrWho17 | that's probably why you have the mismatch now |
| 19:19.25 | DrWho17 | between the os includes and the kernel |
| 19:19.26 | Maxxed | DrWho17: ey ? |
| 19:19.52 | Maxxed | DrWho17: so the kernel, kernel-headers and kernel-sources wont cure what ales me ? |
| 19:20.03 | DrWho17 | yes, they should |
| 19:20.04 | marno | i installed the cvs-version, but there is the same problem.... No application 'SIPGetHeader' |
| 19:20.05 | Maxxed | DrWho17: doing that and recompiling i would think would do the trick? |
| 19:20.06 | DrWho17 | get them all |
| 19:20.11 | Maxxed | yeah :) |
| 19:20.14 | DrWho17 | yes |
| 19:20.22 | Maxxed | ok ok, off with me now, lets see how quick i can trash this machine :p |
| 19:20.29 | Maxxed | thanks a bunch for the help |
| 19:20.46 | G0shen | Maxxed: good luck |
| 19:20.53 | Maxxed | ima need it ;) |
| 19:21.02 | DrWho17 | marno: hrm, mine has it |
| 19:21.12 | DrWho17 | asterisk -r |
| 19:21.16 | DrWho17 | show applications |
| 19:21.24 | DrWho17 | at least it's in my list of applications |
| 19:22.00 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
| 19:22.19 | *** join/#asterisk mrtwister (~mrtwister@cable7107.tele2internet.lt) |
| 19:22.25 | DrWho17 | <PROTECTED> |
| 19:22.25 | DrWho17 | Asterisk CVS-HEAD-03/18/05-13:30:06 b |
| 19:24.11 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 19:26.40 | a1fa | dang it |
| 19:26.43 | a1fa | line 2 did not work |
| 19:26.52 | *** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net) |
| 19:27.38 | RoyK | booooooooring |
| 19:28.06 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 19:28.39 | a1fa | tru |
| 19:28.41 | a1fa | i am pissed |
| 19:28.49 | terrapen | hrmm |
| 19:29.00 | terrapen | i think i need to set these guys up with one POTS line |
| 19:29.03 | terrapen | through an FXO |
| 19:29.05 | terrapen | for their 911 |
| 19:29.23 | terrapen | to shield my ass from a lawsuit |
| 19:29.23 | a1fa | terra |
| 19:29.24 | a1fa | no |
| 19:29.26 | terrapen | ? |
| 19:29.28 | a1fa | get them a cellphone |
| 19:29.33 | a1fa | prepaid cellhpone from walmart |
| 19:29.37 | a1fa | it has 911 |
| 19:29.40 | a1fa | it only costs $20 |
| 19:29.42 | terrapen | no, its a client of mine |
| 19:29.46 | a1fa | so |
| 19:29.49 | a1fa | get him a cell phone |
| 19:29.52 | a1fa | emergency cellphone |
| 19:29.55 | a1fa | he will be proud |
| 19:30.00 | terrapen | here's the thing, when you are frantic, you don't think "USE THE CELLPHONE TO DIAL 911" |
| 19:30.04 | terrapen | you just pick up the phone and dial |
| 19:30.09 | a1fa | nah |
| 19:30.10 | a1fa | not me |
| 19:30.12 | terrapen | i know this. last week i was involved in a shooting. |
| 19:30.18 | a1fa | lol |
| 19:30.18 | terrapen | it was really crazy |
| 19:30.20 | a1fa | ghetto |
| 19:30.21 | a1fa | ! |
| 19:30.24 | a1fa | phunkster |
| 19:30.27 | a1fa | i am used to shooting |
| 19:30.29 | terrapen | the person calling 911 actually dialed '811' twice |
| 19:30.30 | a1fa | and bombing |
| 19:30.34 | a1fa | hhehe |
| 19:30.37 | a1fa | what a fool |
| 19:30.39 | a1fa | i mean tool |
| 19:30.39 | terrapen | and then i gave her the local number to the police |
| 19:30.44 | terrapen | and she misdialed that, too |
| 19:30.54 | terrapen | a1fa, when someone is bleeding all over the place, you get a little shaken |
| 19:31.01 | terrapen | i completely understand her mistakes |
| 19:31.10 | terrapen | i was a little freaked out, too |
| 19:31.20 | Grooby | anyone experiencing bad voice quality w/ broadvoice? |
| 19:31.43 | terrapen | i will use a phone splitter to share the POTS line between the FXO and the fax machine |
| 19:31.58 | terrapen | the FXO will do outbound calls on that line only |
| 19:32.38 | terrapen | the only thing i want is the ability for the FXO (asterisk?) to forcibly disconnect the fax |
| 19:32.45 | terrapen | in the event of a 911 dial-out |
| 19:34.54 | *** join/#asterisk clive- (~pirch@rrba-146-101-246.telkomadsl.co.za) |
| 19:36.07 | Shido6 | back |
| 19:37.51 | AgiNamu | So what's all involved in getting 911 support? |
| 19:38.15 | terrapen | i don't *want* my 911 calls to go over ther inetnet |
| 19:38.24 | terrapen | i just want asterisk to send them over POTS |
| 19:39.16 | DrWho17 | AgaNamu: some kind of interconnect agreement with every LEC whose area you are provide VoIP service? |
| 19:39.28 | a1fa | terrapen : nah.. i've seen heads blown |
| 19:39.50 | AgiNamu | DrWho17 -- yea, well, i wanted to avoid that |
| 19:40.44 | DrWho17 | AgiNamu: well, I think some are forwarding the 911 extension to the normal telephone number |
| 19:40.56 | bannerman | terrapen: that's what I'm planning to do as well |
| 19:41.02 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 19:41.04 | AgiNamu | Yea, I can look up that number and just redirect. |
| 19:41.12 | AgiNamu | but.... that wont send location information eh? |
| 19:41.15 | DrWho17 | 911 -> normal 911 number |
| 19:41.16 | AgiNamu | ewll, obviously it wont |
| 19:41.22 | DrWho17 | AgiNamu: no |
| 19:41.23 | AgiNamu | yea, the PSAP or something\? |
| 19:41.26 | DrWho17 | but that's ok |
| 19:41.49 | Lee__ | when I type 'zap show channels' the only channel it shows is labled pseudo. Is this normal? |
| 19:41.52 | harryvv | http://www.oag.state.tx.us/oagNews/release.php?id=849 Lasuit against Vonage |
| 19:41.59 | Lee__ | I have a wildcard X100P and the modules are loaded |
| 19:42.00 | AgiNamu | harryvv, that, that's retarded. |
| 19:42.10 | harryvv | yea it is |
| 19:42.13 | Uther_P | I have a sip provider, my asterisk server is behind a nat... can anyone tell me why asterisk is registering with the provider as the internal address, even though I have the EXTERNIP variable set in the sip.conf? |
| 19:42.14 | AgiNamu | dumbasses didnt bother setting up their 911 service, then they complain? screw em. natual selecton. |
| 19:42.20 | jontow | hmm, yuck |
| 19:42.26 | a1fa | fffjjjjjjxjjxed |
| 19:42.27 | a1fa | :) |
| 19:42.31 | a1fa | fiiiiiixxxeeeeeeeeeeeed! |
| 19:42.37 | AgiNamu | Vonage should countersue for them being dumbasses. But I guess in texas, that's the norm? ;) |
| 19:42.38 | a1fa | back to work |
| 19:42.49 | smash- | hrm |
| 19:42.51 | smash- | i gues |
| 19:42.54 | smash- | agi |
| 19:43.06 | DrWho17 | AgiNamu: it's also the headquarters of some big LEC's |
| 19:43.09 | jontow | this is what i have to work with, i've gotta make *'s voicemail application 'feel like' : |
| 19:43.09 | AgiNamu | you guess? what other way is there? |
| 19:43.14 | jontow | http://mno.bsd.st/~jontow/apex.map.txt |
| 19:43.25 | AgiNamu | I dislike vonage, but im happy they get to sort all these issues out for the rest of us. |
| 19:43.51 | DrWho17 | Uther_P: did you set the NAT setting on your asterisk sip.conf |
| 19:44.14 | spackle | Will Vonage get blamed when the ISP is down too? |
| 19:44.18 | Uther_P | yes, NAT=yes; EXTERNIP=myoutsideaddress |
| 19:44.38 | DrWho17 | spackle: probably |
| 19:44.54 | Uther_P | but on the provider's side, its still registering with the internal address |
| 19:45.43 | G0shen | AgiNamu: I agree its nice to have a big voip provider stomping out problems, like that isp that was blocking them |
| 19:45.59 | spackle | rediculout. Look at the stack it takes for Vonage to work! Cable/DSL->ISP->Vonage. It's not your father's Ma bell anymore. |
| 19:46.36 | Uther_P | heh |
| 19:46.40 | DrWho17 | Uther_P: don't know, the few I have behind NAT worked fine, after switching Nat=yes |
| 19:46.45 | Uther_P | that is not your daddy's shotgun technology |
| 19:46.48 | DrWho17 | previously they were presenting the internal ip |
| 19:47.15 | AgiNamu | yea, but blaming the ISP is easier. "Can you get to our website?" "no." "go screw yourself" :) |
| 19:47.20 | *** join/#asterisk jeffik (~jeffik@CPE0050bac711e3-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 19:47.24 | smash- | LOL~ |
| 19:47.32 | smash- | ~PRI router |
| 19:47.47 | smash- | ~AGI |
| 19:47.48 | jbot | [agi] the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 19:47.51 | AgiNamu | anyone have any experience with the VoIP 911 providers? |
| 19:48.21 | DrWho17 | that website posted previously is pretty informative |
| 19:48.24 | AgiNamu | i guess i can just redirect to a PSAP for now. |
| 19:48.31 | DrWho17 | www.911voip.org |
| 19:48.34 | AgiNamu | which website? |
| 19:48.34 | DrWho17 | check their FAQ |
| 19:49.20 | ManxPower | How I handle 911 for my VoIP users: "I'm sorry, but 911 is not available from the phone you are calling from. We told you this when we handed you the VoIP phone. Please hang up and use a land line to dial 911" |
| 19:49.51 | AgiNamu | yea, but its a nice feature for us to offer. |
| 19:50.07 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
| 19:50.13 | harryvv | Manx one idea is get each police,fire,ambulance 7 digit phone numbers and do a 911 extension with a ivr statung press 1 for police 2 for ambulance ect. |
| 19:50.16 | ManxPower | For an office, I always install PSTN service locally anyway. |
| 19:50.38 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 19:50.42 | clive- | does anyone know what this means: Ooh, voice format changed to 256 |
| 19:50.42 | harryvv | I know our local rcmp have a standard 7 digit call in. |
| 19:50.53 | G0shen | rcmp? |
| 19:51.02 | sudhir492 | Asterisk crashes sometimes on H323 calls |
| 19:51.05 | ManxPower | clive-: it means the codec changed to number 256. "show codecs" will tell you what 256 is |
| 19:51.05 | harryvv | Royal Canadian Mounted Police |
| 19:51.32 | AgiNamu | is there a PSAP database or something? |
| 19:51.34 | *** join/#asterisk cjk (~cjk@80.92.75.119) |
| 19:51.35 | clive- | Manx, thanks, trying to solve a native transfer issue here |
| 19:51.50 | Uther_P | hehe, I think I just figured out why it was still registering with internal... my sip provider, which is also my t1 provider, used an internal address of 192.168.22.212 for the sip server... my LOCALNET setting was 192.168.0.0/255.255.0.0, so even though I had EXTERNIP set, since the target was within the LOCALNET, it didn't use it... sound reasonable? |
| 19:51.59 | cjk | so hi its me again, anyone here who got some experience in grandstream firmware preconfiguration (cfg.txt) |
| 19:52.07 | ManxPower | Uther_P: Yes. Welcome to NAT. |
| 19:52.26 | AgiNamu | no, welcome to Shitty Internet Protocol. |
| 19:52.58 | clive- | hmmm, does eveyone get this ooh message ...in my codecs thingy 256 = G729 |
| 19:53.19 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 19:53.21 | AgiNamu | yea, i hate that message |
| 19:53.27 | AgiNamu | Thats why I wanted a console filter. |
| 19:53.37 | Uther_P | heh, I know about nat, but I didn't realize that the externip option depended on the target being outside a certain block.. I didn't HAVE the localnet defined.. that was just the default |
| 19:53.42 | AgiNamu | "Ooh! you're making a call!" "Ooh! It's using a codec." "Ooh! Asterisk is running" |
| 19:53.48 | clive- | Agi, so thats normal then this Ooh thingy...ok,,, |
| 19:53.53 | clive- | :) |
| 19:54.25 | clive- | so now still puzzled why my native bridge is not hapenning |
| 19:55.02 | AgiNamu | whats the setup? |
| 19:56.00 | clive- | ipphone---asterisk1---asterisk2---cisco---TDM |
| 19:56.09 | TomL | I don't suppose there's been any jitter buffering introduced for SIP since CVS-HEAD-12/21/04-19:04:45? |
| 19:56.26 | clive- | I am trying to get asterisk1 out the path |
| 19:56.48 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 19:57.20 | *** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
| 19:57.28 | clive- | actually, ..maybe this nufone rating thingy I have installed is messing with the native transfer |
| 19:57.49 | xkev | do I need to escape + in $[${var} : \d+] ? |
| 19:57.59 | AgiNamu | rating thinguy?? |
| 19:58.59 | goobster | I have a question - |
| 19:59.09 | goobster | Does Asterisk support g729? |
| 19:59.42 | AgiNamu | yes. |
| 19:59.50 | AgiNamu | you must buy a license from digium. www.digium.com |
| 19:59.57 | harryvv | was going to say that :) |
| 19:59.59 | goobster | ahh - that's what I suspected |
| 20:00.01 | harryvv | what does it cost |
| 20:00.02 | goobster | thanks! |
| 20:00.09 | AgiNamu | $10 a channel |
| 20:00.13 | harryvv | And are there any real advantages |
| 20:00.26 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 20:00.30 | AgiNamu | real advantages? |
| 20:00.33 | AgiNamu | being legal. |
| 20:00.35 | AgiNamu | thats it? |
| 20:00.36 | harryvv | of g729 |
| 20:00.36 | AgiNamu | support? |
| 20:00.41 | AgiNamu | the codec? |
| 20:00.44 | xkev | low bandwidth, good quality |
| 20:00.48 | AgiNamu | Yea, great quality at way less bandwidth. |
| 20:00.52 | spackle | harryvv: the codec can compress to 8kbps. |
| 20:00.53 | harryvv | over that of the other codecs? |
| 20:00.53 | xkev | eats cpu though |
| 20:00.58 | harryvv | ahh |
| 20:01.02 | nirs | hey all |
| 20:01.04 | nirs | anybody home ? |
| 20:01.10 | harryvv | xkev thats expected |
| 20:01.11 | xkev | e.g. some polycoms can't conference g729 cuz it's too much for them |
| 20:01.15 | spackle | harryvv: Very nice through a WAN or low bandwidth connection. |
| 20:01.22 | AgiNamu | yea, it's a lot of CPU\ |
| 20:01.23 | Shido6 | brb |
| 20:01.25 | harryvv | like wireless wan |
| 20:01.26 | clive- | g927 is a great codec |
| 20:01.39 | *** join/#asterisk hugeguy (~atle@217.170.130.47) |
| 20:02.30 | sivana | what's this talk about on-board echo cancellation on TE411P? |
| 20:05.39 | RoyK | te411p? |
| 20:05.45 | RoyK | te410p? |
| 20:06.23 | sivana | no |
| 20:06.33 | sivana | TE410P can be upgraded to TE411P |
| 20:06.36 | sivana | supposedely |
| 20:06.48 | RoyK | url? |
| 20:07.03 | sivana | mailing list |
| 20:07.10 | sivana | here's the part I caught: |
| 20:07.13 | sivana | " For $2500 you can buy the TE411P which is a 4 port T1/PRI card that has onboard echo cancellation. Or you can send in your TE410P and $1000 and they will upgrade it for you." |
| 20:07.32 | sivana | I'm here to find out more about that statemetn :) |
| 20:07.43 | RoyK | too bad I don't have any echo problems, then :P |
| 20:07.53 | RoyK | also, that $1000 extra isn't worth it |
| 20:07.59 | cpatry | TE411P ? new one? |
| 20:07.59 | RoyK | sangoma cards do that a lot better |
| 20:08.05 | RoyK | with lower interrupt load |
| 20:08.06 | sivana | yes, agreed |
| 20:08.13 | sivana | we're a reseller for Sangoma |
| 20:08.28 | sivana | I'm just trying to find out if anyone here have heard of that |
| 20:08.29 | RoyK | ;) |
| 20:08.39 | cpatry | sivana: ya've the link for that emaiL? |
| 20:08.48 | clive- | sinana its a wicthdoctor |
| 20:08.50 | sivana | especially since I can get a hardware based echo can for $2200 USD |
| 20:08.59 | RoyK | fscking $2500 is a lot of money for a shite card from digium |
| 20:09.12 | RoyK | sivana: the normal 104 card. does that have echo cancellation? |
| 20:09.18 | sivana | ~seen mboehm |
| 20:09.20 | jbot | i haven't seen 'mboehm', sivana |
| 20:09.28 | Maxxed | depmod: *** Unresolved symbols in /lib/modules/2.4.29-3tr/misc/torisa.o |
| 20:09.28 | Maxxed | depmod: *** Unresolved symbols in /lib/modules/2.4.29-3tr/misc/wcfxs.o |
| 20:09.28 | Maxxed | depmod: *** Unresolved symbols in /lib/modules/2.4.29-3tr/misc/wct4xxp.o |
| 20:09.28 | Maxxed | depmod: *** Unresolved symbols in /lib/modules/2.4.29-3tr/misc/wcte11xp.o |
| 20:09.28 | Maxxed | depmod: *** Unresolved symbols in /lib/modules/2.4.29-3tr/misc/zaptel.o |
| 20:09.28 | Maxxed | depmod: *** Unresolved symbols in /lib/modules/2.4.29-3tr/misc/ztd-eth.o |
| 20:09.30 | Maxxed | depmod: *** Unresolved symbols in /lib/modules/2.4.29-3tr/misc/ztdynamic.o |
| 20:09.32 | Maxxed | damnit |
| 20:09.34 | Maxxed | heh |
| 20:09.36 | sivana | none of them do |
| 20:09.41 | sivana | Maxxed: pastebin.ca |
| 20:09.42 | RoyK | ~pastebin |
| 20:09.43 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 20:09.53 | Maxxed | ya i dig ya |
| 20:09.53 | G0shen | MAxxed: did you recompile your kernel yet? |
| 20:09.56 | Maxxed | yep |
| 20:09.57 | nirs | hey all, anyone has experience with BackGroundDetect ? |
| 20:10.00 | ManxPower | RoyK: I'll sell you a 4 port Digium card for only US$2000! |
| 20:10.03 | G0shen | to 2.6? |
| 20:10.03 | RoyK | Maxxed: apt-get install module-init-tools |
| 20:10.09 | Maxxed | cat /proc/version |
| 20:10.09 | Maxxed | Linux version 2.4.29-3tr (root@trustix-22.trustix.net) (gcc version 3.3.4 (Trustix)) #1 Fri Feb 18 18:22:04 CET 2005 |
| 20:10.25 | RoyK | ManxPower: and the sangoma 104 card costs 1699,- |
| 20:10.30 | Maxxed | apt-get ? |
| 20:10.35 | G0shen | Maxxed: thought you were going to 2.6 |
| 20:10.45 | RoyK | looks like it |
| 20:10.49 | Maxxed | nuhuh? was i supose to? |
| 20:10.49 | ManxPower | RoyK: The 4-port Digium card is only $1500 |
| 20:10.59 | RoyK | 2.6 needs module-init-tools instead of modutils |
| 20:11.16 | RoyK | ManxPower: 1599 from digium, and it stresses the system a lot more than sangoma's cards |
| 20:11.20 | sivana | ManxPower: have you heard of the TE411P (with built-in echo can) |
| 20:11.38 | sivana | on-board echo can |
| 20:11.40 | Maxxed | was compiled for kernel version 2.4.27-4tr, this kernel is version 2.4.29-3tr |
| 20:11.50 | drumkilla | well, how much has sangoma contributed to Asterisk? |
| 20:11.55 | drumkilla | that's about all I'll say on that subject ... |
| 20:11.59 | RoyK | Maxxed: use a kernel from kernel.org |
| 20:12.00 | ManxPower | sivana: I heard that it's just wishful thinking. |
| 20:12.10 | Maxxed | RoyK: ok, il give it a shot |
| 20:12.11 | drumkilla | ManxPower: it's reality :) |
| 20:12.14 | sivana | drumkilla: I have no idea, none if they're a hardware maker |
| 20:12.33 | BrianR___ | I was looking at the sangoma cards.. I hear they have DMA? |
| 20:12.42 | BrianR___ | The sangoma cards are cheaper too, aren't they? |
| 20:12.46 | sivana | no |
| 20:12.46 | drumkilla | sivana: exactly - but without Digium, there would be no Asterisk |
| 20:12.52 | RoyK | drumkilla: well. as long as sangoma cards are a lot better, it doesn't really matter |
| 20:13.01 | sivana | drumkilla: well, you can't expect Digium to be the only hardware maker |
| 20:13.01 | ManxPower | drumkilla: It's not a reality until it's on Digium's online store. |
| 20:13.05 | RoyK | I'll be ditching asterisk for aefirion :) |
| 20:13.21 | ManxPower | BrianR___: Digium's TE* cards do DMA. |
| 20:13.28 | drumkilla | sivana: I'm not saying that |
| 20:13.31 | ManxPower | RoyK: Is that the Irish Asterisk clone? |
| 20:13.45 | RoyK | ManxPower: irish? |
| 20:13.49 | sivana | I don't know, I look for hardware that works :) |
| 20:14.03 | ManxPower | RoyK: Gaelic has a lot of ae and i's |
| 20:14.08 | BrianR___ | Aren't they both based very closely off the zapata design? |
| 20:14.16 | sivana | BrianR___: yes, they are |
| 20:14.25 | *** join/#asterisk hugeguy (~atle@217.170.130.47) |
| 20:14.34 | sivana | BrianR___: Sangoma has taken it further, and developed better drivers (supposedly) |
| 20:14.37 | RoyK | ManxPower: dunno where they found the name. it's an asterisk fork meant to do things better |
| 20:14.59 | Darwin[laptop] | ok |
| 20:15.04 | sivana | BrianR___: I have Digium hardware, I haven't tried Sangoma yet |
| 20:15.26 | BrianR___ | Are there any important differences between the TE410 and the sangoma A104, aside from the digium card being single voltage? |
| 20:15.30 | sivana | BrianR___: supposedly, they've done more work on the card and the drivers, but it's all hear-say right now |
| 20:16.18 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 20:16.59 | bkw_ | leeeeeeeeeeeeeeeeets see |
| 20:17.00 | BrianR___ | I have the TE410P and it works just fine. Was expensive though. |
| 20:17.00 | sivana | I'd like to see Digium spend more time on their hardware |
| 20:17.20 | Katty | what is the name of the open office package for apt-get installing purposes? |
| 20:17.32 | Katty | or at least writer. |
| 20:17.40 | sivana | bkw_: you there? |
| 20:17.40 | BrianR___ | Also fits only in a 3.3v slot, but that's not much of an issue these days. |
| 20:17.42 | Hmmhesays | apt-cache search openoffice |
| 20:17.47 | Katty | Hmmhesays: thx |
| 20:17.49 | bkw_ | no problems with digium hardware here |
| 20:17.51 | jaxxan | anyone done any work with metrics and statistics for call center phones similar to ACD ? |
| 20:17.59 | Hmmhesays | np |
| 20:18.15 | sivana | bkw_: have you heard of the TE411P? no product maybe? |
| 20:18.18 | sivana | new |
| 20:18.29 | bkw_ | what is it gonna be? |
| 20:18.36 | *** join/#asterisk brimstone (~brimstone@216.207.244.170) |
| 20:18.44 | jaxxan | i have a small call center with limited statistics, but management wants more, they crave more.... and they're talking about call manager. I'd rather not go that route. |
| 20:18.45 | drumkilla | bkw_: it's the board with the onboard echo can |
| 20:18.47 | bkw_ | oh isn't that the one with the hardware echocan add on? |
| 20:18.49 | drumkilla | bkw_: they had it at von |
| 20:18.52 | bkw_ | ya ya |
| 20:18.56 | zoa | yeah |
| 20:18.59 | sivana | supposedly |
| 20:18.59 | zoa | we toyed with it |
| 20:19.00 | bkw_ | but its not a user upgradeable part |
| 20:19.07 | Katty | Hmmhesays: you're just so handy |
| 20:19.14 | bkw_ | the firmware on the card has to be updated |
| 20:19.15 | brimstone | does anyone know if the iaxy supports mwi? |
| 20:19.15 | sivana | is it available yet? |
| 20:19.16 | jaxxan | if i had ACD like statistics to show them, i could shut em up. |
| 20:19.21 | ManxPower | RoyK: It prolly means "crazy programmer" in Gaelic. Sort of like "yate" means "crazy programmer" in Romanian |
| 20:19.22 | drumkilla | brimstone: yes, it does |
| 20:19.26 | Katty | bkw_: moo |
| 20:19.28 | bkw_ | sivana, I think he said production was delayed a bit over some small part |
| 20:19.30 | bkw_ | Katty, yo |
| 20:19.35 | sivana | ok |
| 20:19.43 | sivana | bkw_: so it's going to be shortly? |
| 20:19.47 | Katty | bkw_: my employeer just bought me a vonexus server :< |
| 20:19.49 | bkw_ | sivana, i'm sure |
| 20:19.53 | brimstone | drumkilla: any idea why this phone doesn't get the mwi but it gets stutter tone? |
| 20:19.57 | bkw_ | Katty, what is that? |
| 20:20.10 | Katty | bkw_: voip software that runs on windows :/ |
| 20:20.15 | drumkilla | brimstone: heh, nope |
| 20:20.16 | Katty | bkw_: server software |
| 20:20.21 | brimstone | drumkilla: ok, thanks! |
| 20:20.26 | bkw_ | EWWWWWWWWWWWWWWWWWWWWWWWWWWWWW |
| 20:20.29 | Katty | bkw_: save me! |
| 20:20.31 | bkw_ | Katty, did you smack him around? |
| 20:20.36 | Katty | bkw_: i scowled |
| 20:20.48 | drumkilla | Katty: if you must, you could use asterisk on cygwin |
| 20:20.52 | drumkilla | but I would still cringe |
| 20:20.56 | Katty | drumkilla: no solutions pls |
| 20:20.59 | Katty | drumkilla: just complaining |
| 20:21.04 | drumkilla | ha, well sorrrrry |
| 20:21.08 | bkw_ | haha |
| 20:21.10 | Katty | kthx, all better |
| 20:21.11 | bkw_ | I don't blame her |
| 20:21.14 | bkw_ | I would complain too |
| 20:21.17 | bkw_ | EVIL EVIL EVIL |
| 20:21.22 | Katty | i'm still setting up asterisk |
| 20:21.23 | drumkilla | definitely evil |
| 20:21.28 | Katty | i intend to woo everyone into asterisk |
| 20:21.38 | Hmmhesays | heh |
| 20:21.41 | Hmmhesays | i see how you are |
| 20:21.48 | bkw_ | or setup a windows box and make them think its that vonexsus crap |
| 20:21.48 | Katty | do you...do you really |
| 20:21.55 | Katty | Hmmhesays: sniffle. |
| 20:22.03 | bkw_ | false front .... but really run asterisk |
| 20:22.04 | bkw_ | muhahah |
| 20:22.06 | Hmmhesays | evil vixen! |
| 20:22.12 | Katty | bkw_: or maybe i'll just complain about viruses and spyware all the time |
| 20:22.19 | Katty | bkw_: not to mention everything else ;) |
| 20:22.33 | *** join/#asterisk mog_home3 (~mog_home@digium.com) |
| 20:22.35 | bkw_ | viruses? spyware? ... whats that? |
| 20:22.47 | bkw_ | whats this malware you speak of? |
| 20:23.20 | mog_home3 | malware is everywhere... |
| 20:23.37 | bkw_ | NO its not.. I don't have any |
| 20:23.43 | mog_home3 | heh |
| 20:23.44 | bkw_ | ZERO |
| 20:23.48 | bkw_ | ZILCH |
| 20:23.49 | mog_home3 | yours is called windows bkw |
| 20:23.55 | bkw_ | windows? |
| 20:23.56 | bkw_ | I don't run that |
| 20:24.00 | mog_home3 | what |
| 20:24.01 | Maxxed | fawk me |
| 20:24.08 | bkw_ | mog_home3, OS X baby |
| 20:24.11 | mog_home3 | bkw_ i have seen that on your box on multiple occasions |
| 20:24.15 | mog_home3 | when you upgrade |
| 20:24.15 | Maxxed | geehs |
| 20:24.19 | Maxxed | geesh* |
| 20:24.27 | bkw_ | mog_home3, no windows here baby.. I use a mac |
| 20:24.29 | Maxxed | i really need to read the stdout of stuff more often |
| 20:25.00 | xkev | pcadach thx |
| 20:27.00 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-233-206.dsl.scarlet.be) |
| 20:27.46 | nirs | hey all, anyone has experience with BackGroundDetect ? |
| 20:29.16 | *** join/#asterisk bah (048830696@AC8ACB74.ipt.aol.com) |
| 20:29.48 | brimstone | and there he goes |
| 20:30.28 | nirs | bkw, you there ? |
| 20:30.53 | *** join/#asterisk anachron (~caseystro@168.158.222.20) |
| 20:31.22 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 20:32.01 | bkw_ | ok.. FOOT.. DOWN... NEXT!!! |
| 20:32.14 | DannyF | bkw_, ;) |
| 20:32.51 | Beirdo | ~seen slePP |
| 20:32.53 | jbot | slepp is currently on #asterisk (20h 36m 46s) |
| 20:32.54 | DannyF | urk have a remote * thats not calling home any more yay |
| 20:33.00 | mog_home3 | lol |
| 20:33.24 | *** part/#asterisk anachron (~caseystro@168.158.222.20) |
| 20:33.34 | nestAr | hrmmm.. |
| 20:33.43 | nestAr | number portability in progress |
| 20:33.45 | nestAr | wheeee |
| 20:33.51 | DannyF | tick tack |
| 20:34.47 | spackle | Question: if I have two * boxen, and I want one to be an extension of the other, is there a way to make anything I dial on A automatically get dialed on B over IAX? |
| 20:35.20 | TedC | is queue wraptime calculted from when the call is picked up, or from when it's hung up? |
| 20:35.37 | nestAr | TedC: wrap up time doesn't work for me.. |
| 20:35.57 | nestAr | but in a working enviroment, it should be from hang up |
| 20:36.02 | TedC | nestAr: Yeah, I'm suspecting it's not working here either. Have you figured out a way to implement it? |
| 20:36.04 | nestAr | otherwise it's not much use. |
| 20:36.09 | TedC | right |
| 20:36.13 | nestAr | i haven't |
| 20:36.19 | nestAr | i've given up on it.. |
| 20:36.22 | TedC | I was thinking of possibly something in the dialplan. |
| 20:37.12 | *** join/#asterisk kcir2 (~kcir@ariadne.sanguinary.net) |
| 20:37.16 | kcir2 | so like |
| 20:37.29 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || http://www.digium.com -> The real hardware prices are listed! |
| 20:37.31 | TedC | Ideally, if wraptime was going to work, it would take account of outbound calls as well. |
| 20:37.32 | *** join/#asterisk c00w (~sean@cpc1-staf1-3-0-cust86.brhm.cable.ntl.com) |
| 20:37.34 | c00w | hello |
| 20:38.55 | kcir2 | Mar 23 15:38:24 ERROR[7218]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device or address |
| 20:39.03 | TedC | Alternately, it might be doable in an AGI, although I'm nott too familitar with AGI right now |
| 20:39.05 | kcir2 | but ztcfg -vv shows a channel 1 clearly |
| 20:39.06 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 20:39.57 | kcir2 | and other times it loads without error |
| 20:41.06 | *** join/#asterisk jedaustin (~chatzilla@austin-j.its.dist.maricopa.edu) |
| 20:42.09 | rephorm | i'm setting up an IVR, and am having trouble with ResponseTimout not working as it should (the call drops as soon as my Background finishes, without doing a response timer) |
| 20:42.15 | rephorm | the context is at http://pastebin.ca/8073 |
| 20:42.31 | jedaustin | Ah.. finally got inbound and outbound calling working with Broadvoice :) One problem.. audio quality sucks |
| 20:43.03 | rephorm | everything works fine if I dial an extension while the message is playing |
| 20:44.12 | BrianR___ | http://www.fonefinder.com/Introduction.html has a good description of in-band emergency signalling... |
| 20:44.15 | jedaustin | Any tips for improving audio quality between asterisk and broadvoice? |
| 20:44.28 | mog_home3 | :q |
| 20:44.28 | ManxPower | rephorm: look at the priorities. Asterisk will stop at the first gap in priorities. |
| 20:45.04 | rephorm | ManxPower: yeah, but it should wait for the amount of time set by ResponseTimeout() after it runs out of priorities, no?? |
| 20:46.34 | *** join/#asterisk wildcard0 (~generic@S0106006097e16040.vc.shawcable.net) |
| 20:48.20 | rephorm | hmm. happens with the demo also. odd. |
| 20:48.38 | *** join/#asterisk gmcinnes (~gmcinnes@67.71.63.9) |
| 20:50.07 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 20:50.12 | ariel_ | hello everyone |
| 20:50.16 | gmcinnes | anyone around? It's quiet in here. |
| 20:50.47 | heison | anyone used fastSMS with Asterisk? |
| 20:51.39 | MikeJ[Laptop] | is anyone sucessfully using 2 B chan transfer on pri? |
| 20:52.17 | kram | mikej: i think it's implemented for 5E |
| 20:53.55 | gmcinnes | is anyone doing ivr stuff? |
| 20:54.41 | *** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net) |
| 20:54.43 | MikeJ[Laptop] | y, I was trying to make sense of what options are needed |
| 20:54.48 | Shido6 | boink |
| 20:54.52 | rephorm | gmcinnes: yes |
| 20:55.06 | MikeJ[Laptop] | a-ha : mattf-> I just changed it so that the "transfer" keyword in zapata.conf enables/disables 2BCT on channels. |
| 20:56.15 | gmcinnes | rephorm: I have issues. I don't know how to stream multiple files consecutively and catch multi digit inputs. |
| 20:56.44 | BrianR___ | 2BCT works in which asterisk version? |
| 20:57.16 | BrianR___ | trying to prevent tromboning in this key system integration... I may wind up having to use '#' transfer... |
| 20:57.22 | rephorm | gmcinnes: by consecutively you mean one after the other, right? you use several consecutive Background() commands |
| 20:57.31 | *** join/#asterisk NewSole (david@i216-58-19-5.avalonworks.net) |
| 20:57.42 | gmcinnes | rephorm: More specifically my problem is that the "say date" command is in the middle, and it doesn't return any digits. |
| 20:57.50 | gmcinnes | rephorm: is there a way around that? |
| 20:57.51 | _Sam-- | hey does anyone know what i should try to get caller id name coming through? i get caller ID number, but no name....PRI provider says they are sending the name too |
| 20:58.00 | MikeJ[Laptop] | BrianR___, it's in head, trying to get it working niw |
| 20:58.25 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 20:58.34 | rephorm | gmcinnes: ahh. yeah, not sure how you can get DTMF while its doing a SayDate, sorry. |
| 20:58.59 | BrianR___ | MikeJ[Laptop]: Interesting. |
| 20:59.22 | MikeJ[Laptop] | it does not seem to be working at the moment.. |
| 20:59.28 | gmcinnes | rephorm: gah. I'm going to have to build my own if it becomes a problem. |
| 20:59.46 | BrianR___ | Not sure if this stupid norstar key system supports 2BT on any PRI signalling scheme except SL-1, which Asteridks doesn't support either, so it might not help me anyway. But I'm glad to hear we'll be able to use it on PSTN calls soon. |
| 20:59.49 | MikeJ[Laptop] | I need to go look at the patches, cuz there are refs to a config file option that is not documented |
| 21:00.11 | *** join/#asterisk loick (~loick@APuteaux-151-1-19-24.w82-124.abo.wanadoo.fr) |
| 21:00.13 | kram | brianr: one advantage of buying from digium is tech support that knows what a 2BCT is :) |
| 21:00.32 | MikeJ[Laptop] | hehe |
| 21:00.46 | drumkilla | free tech support at that |
| 21:01.09 | BrianR___ | kram: I already bought my gear from digium... |
| 21:01.10 | rephorm | gmcinnes: does ResponseTimeout() work properly for you? (it isn't wokring here) |
| 21:01.12 | *** join/#asterisk file[laptop] (~file@mctn1-3636.nb.aliant.net) |
| 21:01.15 | MikeJ[Laptop] | o yeah... mental note, add wiki article on zaptel --> nortel meridian (option 11) |
| 21:01.20 | kram | good :) |
| 21:01.30 | BrianR___ | I think we're giving them some $700 to answer my questions during this project too :) |
| 21:01.34 | mog_home3 | we are the best... |
| 21:01.37 | mog_home3 | at least i think so |
| 21:01.43 | rephorm | gmcinnes: its dropping the call immediately after the last priority without waiting for the response |
| 21:01.46 | _Sam-- | i tried to get my card from digium...they suggested i buy it from a distributor |
| 21:01.56 | _Sam-- | maybe they knew i was going to need alot of support. |
| 21:02.06 | MikeJ[Laptop] | same support |
| 21:02.07 | mog_home3 | lol |
| 21:02.07 | drumkilla | _Sam--: are you outside of the US? |
| 21:02.19 | _Sam-- | nope...although some might not consider delaware part of it |
| 21:02.20 | gmcinnes | rephorm: Seems to work ok for me, but I haven't used it much. |
| 21:02.26 | mog_home3 | and we support our distriubtors |
| 21:02.29 | terrapen | has anybody ever made up a "Why Asterisk?" document for non-technical users? |
| 21:02.30 | mog_home3 | if you buy your card from x |
| 21:02.36 | mog_home3 | we help |
| 21:02.37 | drumkilla | _Sam--: ha, well like MikeJ[Laptop] said, you still get the Digium support |
| 21:02.38 | terrapen | i need something that I can modify and give to this client |
| 21:03.02 | terrapen | People live in Delaware? |
| 21:03.08 | mog_home3 | just sheep |
| 21:03.10 | _Sam-- | nah but its a good place to locate a server |
| 21:03.11 | kram | _sam-- when you buy from a disty, you get support from digium *and* the disty |
| 21:03.15 | terrapen | I thought it was only P.O. Boxes |
| 21:03.21 | kram | there's also the yahoo store if you want to buy direct |
| 21:03.25 | jontow | terrapen; the signate book seems to do something similar to that.. and there is a document that sort of explains itself located..somewhere :) "non-technical review of the asterisk...." |
| 21:03.35 | drumkilla | kram is my hero |
| 21:03.50 | terrapen | jon: yes, that's what i need |
| 21:03.58 | mog_home3 | drumkilla is my hero and thus kram by refrence ^_^ |
| 21:04.16 | BrianR___ | Wish you folks would hurry up and get 1.2 out so I don't have to backport the '##' transfer feature :( |
| 21:04.20 | terrapen | when you guys do consulting, how much do you typically mark up hardware? |
| 21:04.29 | _Sam-- | so with all the free support, should i call digium and ask them what i need to do to get inbound caller id name to show up? :) |
| 21:04.39 | drumkilla | BrianR___: you don't want a broken 1.2 do you? A number doesn't make it stable :) |
| 21:04.42 | terrapen | sam--: thats' |
| 21:04.44 | terrapen | err |
| 21:04.47 | terrapen | that's pretty simple |
| 21:04.48 | kram | sam: of course |
| 21:04.48 | MikeJ[Laptop] | terrapen, as much as I can :) |
| 21:04.53 | _Sam-- | i get inbound caller id number, but no name |
| 21:05.09 | terrapen | mike: i'm selling the Cisco 7960 for US$550 |
| 21:05.13 | terrapen | but that includes setup |
| 21:05.16 | drumkilla | _Sam--: 256-428-6161, or support@digium.com |
| 21:05.17 | terrapen | and upgrading the firmware |
| 21:05.24 | _Sam-- | thank you |
| 21:05.27 | BrianR___ | drumkilla: No no.. I want everyone to go on a big stimulant binge and fix all the bugs... |
| 21:05.35 | drumkilla | BrianR___: haha |
| 21:05.44 | mog_home3 | hey |
| 21:05.44 | _Sam-- | if i call ext500 on my asterisk demo, can i get support? :) |
| 21:05.45 | MikeJ[Laptop] | terrapen, well... I won't buy at those prices, but if you found somone who will.. more power too ya |
| 21:05.52 | mog_home3 | that information is to be secret drum |
| 21:06.00 | MikeJ[Laptop] | stop tickling :) |
| 21:06.04 | drumkilla | _Sam--: actually, yeah - just dial ext 6161 when you get to the menu |
| 21:06.11 | terrapen | mike, of course i wouldn't buy at them, either :) |
| 21:06.15 | mog_home3 | NOOOOO |
| 21:06.20 | terrapen | but this customer has no clue how to set up a Cisco phone |
| 21:06.22 | drumkilla | mog_home3: ? |
| 21:06.24 | mog_home3 | he knows the way through the infrstructure |
| 21:06.24 | terrapen | thus, the markup |
| 21:06.39 | drumkilla | mog_home3: haha ... sorry, I thought that was the correct way to do it :) |
| 21:06.43 | mog_home3 | im joking |
| 21:06.47 | mog_home3 | right way |
| 21:06.49 | drumkilla | ok :) |
| 21:06.52 | mog_home3 | is to call 1877-linux-me |
| 21:06.52 | MikeJ[Laptop] | you can absolutly do that kind of markup, but you need to share the profit with me |
| 21:06.58 | bkw_ | terrapen, 550? |
| 21:06.58 | vaewyn | drumkilla: btw... wanted to catch you on here and says thanks for pandering the the peanut gallery at VON |
| 21:07.00 | drumkilla | mog_home3: I could give them all your direct extension! |
| 21:07.01 | bkw_ | dude are you nutz |
| 21:07.07 | mog_home3 | heh |
| 21:07.09 | mog_home3 | i dont have one |
| 21:07.09 | drumkilla | vaewyn: no problem! |
| 21:07.16 | terrapen | bkw, how much would you charge someone? |
| 21:07.21 | mog_home3 | mwahahaha |
| 21:07.23 | harryvv | kram you may or may not know if this is a existing issues but is there a problem with comedian mail having voice prompt problems? Ie it would say enter your mail box number normally but it the issue since installing on a second system is the words would but cut out and the next word would start. Example PleaEnter youMail box |
| 21:07.27 | bkw_ | I can get them for 250-275 with power |
| 21:07.28 | vaewyn | bkw_: sex with pistachios... (f'in nuts) |
| 21:07.39 | MikeJ[Laptop] | bkw_, he is selling them for that much |
| 21:07.43 | MikeJ[Laptop] | not buying |
| 21:07.51 | MikeJ[Laptop] | and getting some idiot to bite |
| 21:07.52 | bkw_ | so |
| 21:07.56 | bkw_ | thats MAD |
| 21:07.57 | bkw_ | haha |
| 21:08.00 | bkw_ | if stupid people ay.. let em |
| 21:08.06 | MikeJ[Laptop] | yeah... |
| 21:08.09 | outtolunc | ay |
| 21:08.12 | MikeJ[Laptop] | he just needs to share |
| 21:08.23 | drumkilla | i'm a poor college student! |
| 21:08.28 | bkw_ | my phonenumber has 429 in it.. which spells GAY |
| 21:08.29 | harryvv | Mabey my asterisk install is the only one that has this issue. Its very anoying |
| 21:08.31 | file[laptop] | who bought an Ipod |
| 21:08.32 | xkev | I'm having issues where about a half dozen of my 35 polycom IP 600s have locked up once or twice over the last week, using 1.4.1 and 2.6.1 boot |
| 21:08.32 | file[laptop] | :p |
| 21:08.32 | Nugget | heh |
| 21:08.34 | xkev | anyone else? |
| 21:08.39 | mog_home3 | lol |
| 21:08.43 | mog_home3 | you are so funny bkw_ |
| 21:08.53 | bkw_ | my cell has 9378 which is WEST |
| 21:08.55 | file[laptop] | silly drumkilla |
| 21:09.11 | ManxPower | xkev: 1.0.x or CVS-HEAD? |
| 21:09.15 | harryvv | drumkilla the students here in bc are litterly starving..the premier yanked all the subsidized collage funding for students. |
| 21:09.21 | xkev | manx head |
| 21:09.29 | bkw_ | ManxPower, I doubt the phone locking up has anything to do with CVS-HEAD vs STABLE |
| 21:09.30 | ManxPower | xkev: I'm sorry to hear that. |
| 21:09.34 | drumkilla | harryvv: :( |
| 21:09.35 | xkev | I'm not |
| 21:09.37 | xkev | :) |
| 21:09.46 | ManxPower | bkw_: There were a lot of SIP changes in CVS-HEAD recently. |
| 21:09.53 | harryvv | drumkilla he has done alot of diservice to this provinace. |
| 21:09.56 | bkw_ | so none that would cause a phone to lock up |
| 21:10.06 | bkw_ | unless the phone has bugs |
| 21:10.09 | xkev | I am on march 11, just before the experimental config options stuff |
| 21:10.10 | ManxPower | bkw_: The two servers I accidently ran CVS-HEAD on would not stay up for even 1 day. |
| 21:10.17 | MikeJ[Laptop] | so your cell # is Gay West ? |
| 21:10.26 | bkw_ | thats funny I have it up all the time.. no crashes |
| 21:10.31 | bkw_ | ok that sounds wrong |
| 21:10.38 | ManxPower | bkw_: Well, it could be argued that NOTHING should EVER make the phone lock up. |
| 21:10.42 | Nugget | /topic #asterisk <bkw_> thats funny I have it up all the time |
| 21:10.50 | ManxPower | bkw_: Dunno. But that's happened. |
| 21:10.57 | bkw_ | ManxPower, you're so down on cvs-head but I have never had any show stoppers that I couldn't fix or work around. |
| 21:10.58 | xkev | manx, I have had only one crash problem, and it's in chan_sip or otherwise not checking corrupting data before doing strcmp() stuff |
| 21:11.23 | ManxPower | bkw_: I'm only down on CVS-HEAD if someone runs it in production. |
| 21:11.32 | bkw_ | we do it all the time |
| 21:11.35 | bkw_ | so does digium :P |
| 21:11.36 | *** part/#asterisk loick (~loick@APuteaux-151-1-19-24.w82-124.abo.wanadoo.fr) |
| 21:11.42 | drumkilla | go stable! hehe |
| 21:11.46 | terrapen | there used to be a site where you could put in your number and it would tell you what it might spell |
| 21:11.49 | bkw_ | ya non-working stable |
| 21:11.55 | bkw_ | drumkilla, broke it |
| 21:11.59 | drumkilla | did not! |
| 21:11.59 | bkw_ | haha |
| 21:12.00 | file[laptop] | darumkilla |
| 21:12.05 | bkw_ | darum? |
| 21:12.12 | file[laptop] | yup |
| 21:12.14 | drumkilla | I haven't done a thing that wasn't done to head first :p |
| 21:12.19 | `Sauron | terrapen: http://www.phonespell.org/ |
| 21:12.19 | file[laptop] | he drinks tons of rum when he's doing stable stuff |
| 21:12.22 | bkw_ | drumkilla, no COMMENT |
| 21:12.23 | file[laptop] | so that is why it is broken |
| 21:12.37 | terrapen | There are no words in 859-3107 |
| 21:12.38 | ManxPower | file[laptop]: I thought he loaded up the bong before doing a release. |
| 21:12.38 | terrapen | sigh |
| 21:12.47 | file[laptop] | release, uh huh |
| 21:12.49 | `Sauron | Wow. |
| 21:12.52 | MikeJ[Laptop] | o screw stable... down with drumkilla....!!!!! |
| 21:12.54 | bkw_ | "<drumkilla>I haven't done a thing that wasn't done to head first :p" |
| 21:13.02 | MikeJ[Laptop] | :) |
| 21:13.03 | vaewyn | eww |
| 21:13.04 | vaewyn | :} |
| 21:13.04 | terrapen | waha look what my toll free spells |
| 21:13.05 | terrapen | toot-17-poop |
| 21:13.12 | bkw_ | hahah |
| 21:13.17 | bkw_ | mike no its no gay west |
| 21:13.31 | MikeJ[Laptop] | isn't that san fran? |
| 21:13.41 | bkw_ | ya |
| 21:13.47 | MikeJ[Laptop] | ouch |
| 21:13.48 | drumkilla | dirty, evil people |
| 21:13.54 | MikeJ[Laptop] | I diserved that I think :) |
| 21:13.55 | bkw_ | lets have a flash back of how I drove in SFO |
| 21:14.04 | bkw_ | drumkilla, care to give a play by play |
| 21:14.09 | ManxPower | bkw_: let's not. |
| 21:14.10 | drumkilla | bkw_: like a mad man! |
| 21:14.11 | vaewyn | I choose mine for the numbers rather than the spellings :} 867-5309 for instance :} |
| 21:14.14 | bkw_ | traffic was so entertaining. |
| 21:14.28 | file[laptop] | I loved when drumkilla lost it |
| 21:14.32 | *** join/#asterisk Goldenear (~Goldenear@d149.dhcp212-198-168.noos.fr) |
| 21:14.34 | `Sauron | Doesn't spell anything. |
| 21:14.36 | MikeJ[Laptop] | bkw_ driving and kram walking in san jose I hear could be a bad combination :) |
| 21:14.39 | `Sauron | well |
| 21:14.43 | `Sauron | barely anything |
| 21:14.49 | ManxPower | Anyone going to VON Europe? |
| 21:14.57 | file[laptop] | it's so sexy! |
| 21:15.04 | xkev | how big will unique-id get? |
| 21:15.05 | drumkilla | ManxPower: I think I may be |
| 21:15.14 | MikeJ[Laptop] | damn... 2bct only works on cpe side, and I had to make * net side... well that sucks |
| 21:15.21 | drumkilla | file[laptop]: haha ... yeah, I did go crazy after driving in circles for a few hours |
| 21:15.22 | xkev | I'm up to 1400 calls since restart, so it's getting larger |
| 21:15.28 | file[laptop] | drumkilla: indeed |
| 21:15.32 | drumkilla | I threw the map to the bag of the van |
| 21:15.33 | vaewyn | I don't know what bkw_ was up to in San Jose... he managed to duck out every time I looked for him at the booth :P |
| 21:15.39 | drumkilla | back* |
| 21:15.43 | bkw_ | vaewyn, I was there |
| 21:15.44 | Goldenear | hi. Are they any Asterisk or IAX devellopers here ? |
| 21:15.47 | _Sam-- | hmmm...whoever told me to call tech at digium.... |
| 21:15.49 | bkw_ | just look for the flame |
| 21:15.53 | _Sam-- | im not sure if i should thank them or not |
| 21:15.53 | MikeJ[Laptop] | maybe I will just get a couple quad pri cards and run everything front end through * |
| 21:15.54 | ManxPower | drumkilla: That would be cool. I just finished my registration. |
| 21:15.59 | _Sam-- | they dont seem too happy to help me |
| 21:16.01 | vaewyn | bkw_: yeah... I know... but you di duck out every time... it was amazing |
| 21:16.08 | jaxxan | how do you do a call count ? |
| 21:16.11 | terrapen | vaewyn, somebody told me that you have a nice wifi phone |
| 21:16.20 | _Sam-- | mog_home3 : the guys name thats helping me is matt |
| 21:16.26 | vaewyn | terrapen: :} why yes I do... :P |
| 21:16.27 | _Sam-- | i asked if it was you, but he said now |
| 21:16.28 | _Sam-- | no |
| 21:16.30 | bkw_ | vaewyn, were you that really strange guy that bugged everyone? |
| 21:16.32 | file[laptop] | there's lots of Matts |
| 21:16.35 | kram | there are 5 matts at digium |
| 21:16.35 | drumkilla | 5 of them |
| 21:16.38 | mog_home3 | that is not me |
| 21:16.39 | terrapen | vaewyn, which one is it? |
| 21:16.39 | _Sam-- | thats what this one said |
| 21:16.41 | vaewyn | bkw_: ohh like that narrows it down :} |
| 21:16.41 | mog_home3 | its mattr |
| 21:16.42 | bkw_ | vaewyn, show me a picture |
| 21:16.45 | mog_home3 | i can talk to you sam if you want |
| 21:16.49 | bkw_ | so I know if I talked to you |
| 21:16.56 | vaewyn | terrapen: hitachi cable WIP 5000 |
| 21:17.07 | _Sam-- | he was as nice as he could be...he got on my console, didnt see any name coming across in incoming call... |
| 21:17.09 | Goldenear | I'm would like to understand something about Asterisk IAX2 codec negotiation, could somebody help me, I think I've found a bug. |
| 21:17.09 | ManxPower | drumkilla: Now I just need to the T-shirts with my resume on it made up! |
| 21:17.17 | _Sam-- | and i said, well now you know what im saying..."well we cant help you then" |
| 21:17.18 | terrapen | vae: is it reliable and how is the sound quality? |
| 21:17.21 | *** part/#asterisk rvhi (~rv@66.175.65.89) |
| 21:17.24 | mog_home3 | one sec |
| 21:17.27 | Nugget | ManxPower: if you need help translating your resume into dutch, lemme know. :) |
| 21:17.27 | ManxPower | Goldenear: Did you check the mailing lists first? |
| 21:17.30 | ManxPower | ~mailinglist |
| 21:17.31 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 21:17.32 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 21:17.36 | Goldenear | Sure |
| 21:17.42 | Cherebrum | Programmers Dare Not Throw Salty Pretzels Away |
| 21:17.47 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 21:17.47 | Goldenear | and I search on google also |
| 21:17.58 | ManxPower | Nugget: The resume writing service used so many marketing words I felt sick, but if it works..... |
| 21:18.05 | Nugget | heh |
| 21:18.14 | `Sauron | Nugget: You speak dutch? |
| 21:18.16 | terrapen | how do you keep jbot from privmsging |
| 21:18.19 | Nugget | my resume is cheeky and irreverent. that's always worked well for me. |
| 21:18.24 | terrapen | and have it speak to the channel instead |
| 21:18.28 | Goldenear | ManxPower: but no way to have any answer ... |
| 21:18.32 | Nugget | anyone deterred by my resume isn't someplace I'd want to work |
| 21:18.43 | ManxPower | Goldenear: post it to the asterisk-dev mailing list |
| 21:19.07 | ManxPower | Nugget: At this point I'm almost ready to apply to escort services. |
| 21:19.17 | ManxPower | I may not be much to look at, but I can be VERY creative. 8-) |
| 21:19.31 | Nugget | heh. well, give bkw_ a call at GAY-WEST :) |
| 21:19.38 | Goldenear | do I have to subscribe to the mailing list to post on it ? |
| 21:19.41 | `Sauron | hehe |
| 21:19.45 | ManxPower | Goldenear: yes. |
| 21:20.11 | ManxPower | Nugget: Naw. I'm a rebel. I think sex should be free. |
| 21:20.11 | Goldenear | ok so I will :) |
| 21:20.12 | *** join/#asterisk tessier (~treed@210.245.99.31) |
| 21:20.22 | vaewyn | bkw_: reason we were looking for you is checking on if anyone has done SIP takeover to get high availability setups |
| 21:20.22 | ManxPower | ..er...escort should be free, that is. |
| 21:20.29 | terrapen | well, at least he doesn't have 429-5897 |
| 21:20.43 | terrapen | or 429-5683 |
| 21:20.52 | vaewyn | bkw_: ie... 2 machines... both processing the call status data... one takes over MAC/IP when other fails... |
| 21:21.01 | bkw_ | oh fun |
| 21:21.06 | bkw_ | thats hot!!! |
| 21:21.10 | vaewyn | hehehe I guess that is a no :} |
| 21:21.18 | bkw_ | correct |
| 21:21.20 | bkw_ | thats a NO |
| 21:21.21 | bkw_ | haha |
| 21:21.22 | ManxPower | I guess I should start packing. |
| 21:21.35 | smash- | ~pri |
| 21:21.36 | jbot | pri is, like, Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 21:21.47 | vaewyn | hmm... so next question... who do we start paying to code it... and how big of a bounty do we need :P |
| 21:22.05 | jaxxan | bkw_: is there any work in progress somewhere for metrics and statistics for asterisk? |
| 21:22.30 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:22.55 | harryvv | jbot needs to be slightly corected. 56 of the 64 is usable content transferable. |
| 21:22.56 | jbot | that's too long, harryvv |
| 21:23.32 | smash- | Hey what kinda handoff do i want for te410p card PRI or ethernet? |
| 21:24.03 | bkw_ | OMG he just didn't |
| 21:24.06 | _Sam-- | mog: call back, thanks! |
| 21:24.19 | MikeJ[Laptop] | smash-ummmmmmm uhhhhhhh |
| 21:24.24 | _Sam-- | lol, ok. |
| 21:24.34 | Juggie | bkw, do u know app_meetme? |
| 21:24.48 | MikeJ[Laptop] | smash- you need to get somone to help you bad |
| 21:24.49 | smash- | help me out |
| 21:24.49 | bkw_ | Juggie, jes.... we don't use it.. we wrote our own |
| 21:24.50 | kram | smash: it basically depends on which side you're coming in on |
| 21:25.03 | cjk | hi its me again, anyone here who got some experience in grandstream firmware preconfiguration (cfg.txt), i will ask every 2 hours till someone can help |
| 21:25.09 | kram | The TE410P takes in T1's or E1's or PRI's and then it could come out your ethernet interface on the PC you put it in |
| 21:25.28 | smash- | what do i need to ask for from t1 provider |
| 21:25.32 | mog_home3 | hey sam |
| 21:25.34 | mog_home3 | make a call out |
| 21:25.35 | Juggie | bkw_, app_conference, or something else? |
| 21:25.37 | _Sam-- | okie |
| 21:25.38 | mog_home3 | sorry for cutting you off |
| 21:25.38 | bkw_ | T1 with PRI signalling |
| 21:25.43 | _Sam-- | no worries, THANK YOU for your help. |
| 21:25.45 | smash- | thanks |
| 21:25.49 | jaxxan | there's something else to use besides meetme ? |
| 21:25.52 | bkw_ | Juggie, app_confcall |
| 21:25.52 | mog_home3 | we live to serve |
| 21:25.55 | smash- | bkw the pri signalling will plug right into the te410p |
| 21:25.57 | _Sam-- | outgoing call in progress now |
| 21:25.58 | kram | no worries, smash |
| 21:26.00 | _Sam-- | working fine |
| 21:26.10 | ManxPower | there! done packing |
| 21:26.12 | Juggie | bkw_, is it released publically? |
| 21:26.18 | bkw_ | Juggie, no |
| 21:26.19 | smash- | or do i need a router from the pri to the te410p |
| 21:26.22 | bkw_ | you'll have to beg anthm for it |
| 21:26.31 | nestAr | anyone having problems with CheckGroup in the lastest cvs? |
| 21:26.35 | vaewyn | reminds me... T setups are the piss poorest documentation I have found so far in *... but everyone probably knows that already :P |
| 21:26.37 | Juggie | bkw, are u still working on meetme2? or is that an early version of meetme2? |
| 21:27.48 | bkw_ | you mean you need docs? |
| 21:27.49 | *** join/#asterisk brimstone (~brimstone@216.207.244.170) |
| 21:27.51 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:28.01 | PBXtech | doesnt hylafax have an irc channel? |
| 21:28.03 | bkw_ | t's are easy to setup |
| 21:28.10 | bkw_ | if the telco pulls their head out |
| 21:28.12 | ManxPower | smash-: There are two options. Would you like to hear them? |
| 21:28.19 | smash- | yes please |
| 21:28.21 | jaxxan | I take it the lack of response to metrics and statistics mean no, there is no work in progress and i'll have to either do it myself or wait 12 years. |
| 21:28.55 | vaewyn | bkw_: yeah ... fairly easy... but it would be nice to say that PRI you can just ignore 1/3 of the options and such |
| 21:29.02 | ManxPower | smash-: This is what I PREFER: Telco brings in the T-1 into some device that splits out an ethernet port for IP (connect to your router) and a DXS-1 for voice to plug into your Digium card. |
| 21:29.20 | vaewyn | bkw_: handing these docs to a newbie is fairly rough for the Ts |
| 21:29.33 | *** join/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 21:29.46 | ManxPower | smash-: The way I don't like: Telco brings the T-1 to you and does NOT split out the voice and data and then you need to handle that all inside Asterisk with HDLC kernel modules and stuff like that for the data. |
| 21:30.12 | dave_mwi | if I do a comparison between a datetime and a string of the same format, will it actually funciton correclty like if datetime1 > datetime2? |
| 21:30.16 | Juggie | bkw, does app_confcall = the app_meetme2 the new meetme you were working on? |
| 21:30.24 | terrapen | is there an inexpensive box that can split out the data and voice? |
| 21:30.28 | vaewyn | ohh @#$@# someone put the remote site TE405P into a 500mhz p3... now i gotta beat someone |
| 21:30.31 | terrapen | ie., T1 in, and ethernet and T1 out |
| 21:30.39 | ManxPower | terracon: My telco provides that box. |
| 21:30.47 | ManxPower | I think they use an Adtran 850 |
| 21:31.12 | terrapen | T1 comes in from the telco and the box splits it to ethernet (data) and another T1 (voice) |
| 21:31.36 | Juggie | thats a pretty weak data connection |
| 21:31.44 | pdracevich | moring, all I was wondering if you are able to point me in the right way, I am wanting to run ser for my extenal people to connect via sip, and have web interface's etc, and be able to route calls to asterisk, |
| 21:31.51 | ManxPower | terracon: Correct. That is they way I like to do it. HOWEVER, the Digium cards and zaptel can handle all that without splitting. |
| 21:31.53 | pdracevich | how do i connect ser to asterisk |
| 21:31.53 | terrapen | that thing looks pretty expensive, Manx |
| 21:32.11 | ManxPower | terracon: As I said, my telco owns and manages the Adtran |
| 21:32.15 | smash- | ok so ManxPower im getting 2 data t1's and 2 voice t1's. It would go like 2xvoicet1->ethernetdevices>dxs-1>digiumcard? |
| 21:32.21 | BrianR___ | Heh.. I plan to build two switches for my office voip setup, but I never thought it was important to preserve calls in progress if one of them crashed and burned.. |
| 21:32.26 | smash- | i understand the t1 data's |
| 21:32.31 | ManxPower | But then we have our telco by the balls. We are their 2nd largest customer. |
| 21:32.33 | *** join/#asterisk darby_t (~tom@dnf172.neoplus.adsl.tpnet.pl) |
| 21:32.45 | terrapen | seems like the hardest part about getting a Voice T1 or Voice+Data T1 working with Asterisk would be dealing with the telco, who is not Asterisk-knowledgable |
| 21:32.46 | BrianR___ | Redial exists for a reason :) |
| 21:33.03 | smash- | manx can i msg you |
| 21:33.06 | ManxPower | terracon: No, the hardest part is getting a telco that will do it that way. |
| 21:33.27 | ManxPower | smash-: Only if you send me $120 via paypal first. Otherwise talk on the channel so others can learn too. |
| 21:33.32 | dave_mwi | does anyone know off hand about that comparison: datetime1 > datetime2 and will it return the correct value? |
| 21:33.36 | smash- | ok |
| 21:33.48 | MikeJ[Laptop] | Manx, nice |
| 21:33.51 | smash- | k did u see what i said last? |
| 21:34.01 | terrapen | i need to learn how to order a voice PRI from my telco and have them move our number from our existing voice T1 |
| 21:34.03 | ManxPower | <smash-> i understand the t1 data's |
| 21:34.08 | smash- | before that one |
| 21:34.14 | terrapen | we have a voice T1 which goes to an Adtran 644 or something |
| 21:34.20 | ManxPower | <smash-> ok so ManxPower im getting 2 data t1's and 2 voice t1's. It would go like 2xvoicet1->ethernetdevices>dxs-1>digiumcard? |
| 21:34.23 | smash- | <PROTECTED> |
| 21:34.25 | terrapen | and it has 23 FXS ports |
| 21:34.39 | terrapen | but i want to go all-digital and put the T1 straight into my * server |
| 21:34.44 | ManxPower | smash-: if you have a Data only T-1 then just hook them into a router and don't involve asterisk. |
| 21:34.55 | smash- | yeah i know that |
| 21:34.56 | smash- | for data |
| 21:34.58 | smash- | lets forget them |
| 21:35.02 | smash- | say i got 1 voice t1 |
| 21:35.04 | terrapen | smash, what is the problem then? |
| 21:35.05 | epoch | what's a "voice T1?" |
| 21:35.09 | smash- | PRI |
| 21:35.10 | epoch | is that PRI, or something different? |
| 21:35.13 | ManxPower | smash-: for Voice you want T-1/PRI -> Digium Card/Asterisk |
| 21:35.16 | epoch | ok, then call it PRI :P |
| 21:35.25 | terrapen | there is some kind of non-PRI voice T1, i think |
| 21:35.27 | ManxPower | epoch: A PRI is just a specially configured T-1 |
| 21:35.27 | terrapen | i think we have one. |
| 21:35.32 | epoch | exactly |
| 21:35.40 | BrianR___ | Still trying to figure out how to order a second PRI and have my DID's routed up both... |
| 21:35.43 | smash- | the PRI plugs right into the DIGIUM card? |
| 21:35.48 | tzanger | smash-: yup |
| 21:35.48 | ManxPower | smash-: Yes. |
| 21:35.48 | smash- | or is there equipment |
| 21:35.51 | vaewyn | You... you can get a 24chan BRI type T... is nasty |
| 21:35.52 | tzanger | got two of 'em |
| 21:36.01 | epoch | smash-: the digium card *is* the equipment :) |
| 21:36.05 | smash- | i c |
| 21:36.09 | _Sam-- | brian: i thought there is a way to share 2 PRIs on one D channel |
| 21:36.10 | BrianR___ | Verizon's web site has no information about PRI ordering, so you gotta talk to an account rep who knows very little about it either... |
| 21:36.13 | epoch | 24 channel BRI? |
| 21:36.13 | ManxPower | smash-: As long as your telco hands the T-1 off to you as a RJ48C (DXS-1) |
| 21:36.13 | smash- | im just thinking like on data side there is a router |
| 21:36.14 | epoch | wtf |
| 21:36.22 | smash- | o oook |
| 21:36.24 | smash- | manx thanks |
| 21:36.26 | BrianR___ | _Sam--: Yes, NFAS. BUt i want to terminate the two PRI's on different asterisk boxes. |
| 21:36.30 | smash- | thats what i needed to know |
| 21:36.34 | harryvv | data on pri is just that raw data. when getting say caller id into that is provided by the calling parties telco not the provider you are not hooked to? |
| 21:36.35 | vaewyn | The only time you need equipment before your digium card is if they use some freaky interface other than rj45/rj48 |
| 21:36.42 | BrianR___ | _Sam--: I don't care if I waste a channel in that case. |
| 21:36.56 | ManxPower | vaewyn: You mean like coax? |
| 21:37.21 | epoch | a client of mine uses some really old interface... RS-something or other |
| 21:37.24 | vaewyn | ManxPower: yeah... that evil... I have never seen it... but I guess it is popular in europe in a couple places i nthe US |
| 21:37.25 | epoch | I forget which |
| 21:37.35 | BrianR___ | _Sam--: Essentially I want DID's to ring on whichever PRI has free B channels. If a PRI is down, I want it to be considered as having 0 free B channels. |
| 21:37.42 | ManxPower | epoch: Ot |
| 21:37.58 | terrapen | how do you tell if you have a PRI or something else? |
| 21:38.02 | ManxPower | epoch: That sounds like a form of serial interface, commonly called "winchester" |
| 21:38.06 | terrapen | im afraid i may have 'something else' |
| 21:38.08 | ManxPower | terracon: you ask your telco |
| 21:38.16 | vaewyn | terrapen: read your bill :} |
| 21:38.17 | smash- | right now we got something else in our building cat3 |
| 21:38.18 | smash- | or something |
| 21:38.32 | MikeJ[Laptop] | BrianR___, telco can set those up hunting like that, but likely will fill one pri then the other |
| 21:38.34 | ManxPower | terracon: I currently have 2 asterisk systems running on Channelized Voice T-1 |
| 21:38.34 | smash- | its like 100 wires split into a white phonebox type thing |
| 21:38.39 | smash- | with metal tabs on it |
| 21:38.42 | *** part/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 21:38.43 | smash- | like some 1985 shit |
| 21:38.44 | epoch | ManxPower: yeah, that sounds about right |
| 21:38.58 | harryvv | brb |
| 21:39.05 | ManxPower | terracon: it's pretty much EXACTLY like POTS service, one "line" per T-1 channel. |
| 21:39.08 | BrianR___ | MikeJ[Laptop]: That's OK provided it works correctly in the case where either PRI is broken |
| 21:39.14 | epoch | ManxPower: in-line signalling in that case? |
| 21:39.18 | terrapen | maxnx, i *think* thats what we have |
| 21:39.20 | epoch | or is there still a D cahnnel |
| 21:39.21 | ManxPower | epoch: Yes. Called "CAS" |
| 21:39.27 | terrapen | because it goes into this adtran |
| 21:39.31 | ManxPower | epoch: D channel is a PRI only thing. |
| 21:39.33 | epoch | hrm, that's kinda gross |
| 21:39.36 | p1tst0p | hey, is there a simple phonebook/directory included in asterisk |
| 21:39.40 | terrapen | manx: and each POTS line coming out of the adtran has a phone number assigned to it |
| 21:39.42 | epoch | I really like the concept of out-of-band signalling |
| 21:39.45 | jontow | p1tst0p; yes :) |
| 21:39.48 | terrapen | does that sound like a channelized voice t1? |
| 21:39.53 | MikeJ[Laptop] | BrianR___, that is just 2 PRI's with rollover too each other, you could put some did's primary on 1, some primary on another to balance a bit |
| 21:39.56 | ManxPower | terracon: You do not have PRI, you have "CT1" |
| 21:40.01 | terrapen | ugh |
| 21:40.02 | terrapen | ok |
| 21:40.04 | jontow | p1tst0p; for example: |
| 21:40.05 | jontow | exten => 411,1,Directory(default) |
| 21:40.16 | ManxPower | PRIs do NOT have a phone number assigned per channel. |
| 21:40.18 | MikeJ[Laptop] | so you have some channels for outbound on each too, to reduce your trunking between boxes for outbound |
| 21:40.21 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 21:40.22 | vaewyn | So if you have a PRI what is the signalling? cause it isn't like the lines do e&m or such then |
| 21:40.28 | MikeJ[Laptop] | manx, they can. |
| 21:40.31 | vaewyn | D channel handles that |
| 21:40.33 | p1tst0p | jontow, under default context ? |
| 21:40.36 | MikeJ[Laptop] | but usually no. |
| 21:40.40 | jontow | 'default' is the context.. |
| 21:40.50 | ManxPower | vaewyn: It's PRI signaling 8-) |
| 21:40.53 | BrianR___ | MikeJ[Laptop]: Ok... Now I've just gotta figure out the ordering... |
| 21:40.55 | jontow | you put the extension wherever you need it, and using whatever context.. the context reads entries from voicemail.conf to populate the directory. |
| 21:41.05 | jontow | p1tst0p; from the CLI, type: show application directory |
| 21:41.12 | vaewyn | ManxPower: I mean... what do you tell *... I have copied all my configs so I don't remember |
| 21:41.18 | MikeJ[Laptop] | BrianR___, is it LD or local? |
| 21:41.24 | hajekd | Do you know something more about app_voicemail fix in 1.0.6? |
| 21:41.28 | epoch | I'm almost sad I'm quitting this job, I won't have direct access to the PRI or fibre link anymore ;/ |
| 21:41.32 | ManxPower | As soon as my ride is here I'll be leaving to do a CT1 -> PRI conversion |
| 21:41.41 | ManxPower | vaewyn: I would have to look it up and I'm too lazy right now. |
| 21:41.45 | BrianR___ | MikeJ[Laptop]: I'm planning to terminate the two PRI's on seperate asterisk boxes... |
| 21:41.49 | vaewyn | ManxPower: 'clear' maybe? |
| 21:41.58 | MikeJ[Laptop] | yes, pri's from who?> |
| 21:42.02 | ManxPower | vaewyn: for the B channels, yes. |
| 21:42.15 | BrianR___ | MikeJ[Laptop]: They would both be PRI's to our LEC. Both incoming and outgoing calls. |
| 21:42.15 | vaewyn | *nods* that was it then |
| 21:42.20 | ManxPower | BrianR___: I'll be doing a 2 asterisk, 2 PRI install in a couple of weeks. 60 SIP phones |
| 21:42.32 | BrianR___ | ManxPower: Interesting. Let me know how it goes. |
| 21:42.43 | smash- | manx |
| 21:42.44 | BrianR___ | MikeJ[Laptop]: Both PRI's to the same LEC (Verizon). |
| 21:42.54 | smash- | im doing 2 pri's and 20 sip phones |
| 21:43.00 | smash- | and like 60 softphones |
| 21:43.03 | BrianR___ | Just planning some fault tolerance so if we lose an asterisk box, lose a PRI, etc. we're still OK. |
| 21:43.07 | MikeJ[Laptop] | y, just tell them you want 2 pris, that roll over too each other, and to point 1/2 the did's to 1 and 1/2 to the other (to keep it balanced, they will still roll if one fills) |
| 21:43.14 | smash- | brianR |
| 21:43.17 | smash- | thats a good idea |
| 21:43.21 | smash- | i didnt think about using 2 |
| 21:43.21 | Lee__ | are there any command line utilities for testing connectivity to asterisk? |
| 21:43.23 | ManxPower | BrianR___: It will be...interesting. This is the site that wanted Asterisk at the NEW office. The office that was supposed to open in Jan, then April, now end of May. They are scheduled to open about 3 weeks into my 6 week trip to Euroland |
| 21:43.37 | MikeJ[Laptop] | and make sure they knwo that each will run it's own b chan, not b chan per with backup b chan or anything like that |
| 21:43.49 | MikeJ[Laptop] | Lee__, testcall |
| 21:44.15 | ManxPower | So we are installing it in the current office and I'll have the IS people do the MOVE. |
| 21:44.19 | Lee__ | that's in the asterisk source? |
| 21:44.26 | MikeJ[Laptop] | ummm |
| 21:44.32 | MikeJ[Laptop] | no, iaxclient |
| 21:44.36 | BrianR___ | List one as the primary SIP proxy on phones, the other as the secondary. Keep a nearly identical extensions.conf on both boxes (using an include for the sets file, which is actually identical) that uses a macro for creating set extensions which tries to call SIP/extension on both boxes hoping that the phone will actually be registered at one. |
| 21:44.41 | Lee__ | ah |
| 21:44.41 | MikeJ[Laptop] | it's on sourceforge |
| 21:44.55 | ManxPower | BrianR___: That's the plan |
| 21:45.16 | BrianR___ | Voicemail is going to be interesting... |
| 21:45.49 | ManxPower | BrianR___: But we may just not bother. If one PRI goes down, 1/2 the phones go down |
| 21:46.04 | ManxPower | BrianR___: voicemail on a 3rd box? |
| 21:46.31 | BrianR___ | ManxPower: I want to review the source to commedian mail to see if it's safe to share a spool between two asterisk instances... |
| 21:47.02 | ManxPower | BrianR___: if it isn't it should be pretty easy to fix |
| 21:47.02 | BrianR___ | At current, it looks like the message creation logic is fairly lame. Like it can have a race collision against _itself_ running on a single box. :( |
| 21:47.36 | ManxPower | BrianR___: that MAY have been fixed. |
| 21:47.40 | BrianR___ | ManxPower: Or adding some deletion tracing logic to facilitate rsync'ing the spool between servers. |
| 21:49.01 | BrianR___ | bbiab |
| 21:51.27 | terrapen | ok, my telco says we have a channelized t1 |
| 21:51.29 | terrapen | that's no good |
| 21:51.32 | terrapen | i wish we had a PRI |
| 21:51.38 | smash- | ~pri |
| 21:51.39 | jbot | extra, extra, read all about it, pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 21:51.59 | bile_one | ManxPower, what call progress features are available for t1? Meaning, is it possible to track and see if someone has forwarded you from one phone to the next? |
| 21:52.44 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 21:53.23 | *** join/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 21:54.26 | ManxPower | bile_one: Pretty much the same as analog -- none |
| 21:54.40 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:54.46 | tekjacob | Hey all, I set up a new asterisk box.. (which I have done a few times before ) but this time with a TE110P... The PRI is not yet connected. When I call into the system to test I can see it playing files but I hear nothing. Any ideas? |
| 21:55.04 | bile_one | okay I didn't thinks so, all you really get is a valid pick-up to know it has been answered right? |
| 21:55.23 | smash- | Hey manxpower can asterisk manage voip calls through a te410p |
| 21:55.32 | ManxPower | bile_one: on our CT1 we don't even get remote answer supervision, but we could if we used E*M on those channels |
| 21:55.33 | terrapen | SDSL == channelized T1 ??? |
| 21:55.34 | smash- | can u plug data rj45 into it |
| 21:55.39 | ManxPower | smash-: yes, of couse. |
| 21:55.43 | Supaplex | terrapen: haha no. |
| 21:55.45 | terrapen | http://www.adtran.com/adtranpx/Rooms/DisplayPages/LayoutInitial?product=com.webridge.entity.Entity[OID[6604232441BBF747BDA55AC220E7C289]]&Container=com.webridge.entity.Entity[OID[F5C7CEE8D8313E49B4D65B30BDDF4734]] |
| 21:55.45 | ManxPower | terracon: yes and no. |
| 21:55.49 | terrapen | ugly oops |
| 21:55.56 | terrapen | we have an Adtran 624 it seems |
| 21:55.58 | tzanger | terrapen: what the fuck was that |
| 21:56.05 | terrapen | and im not sure if i can put a PRI on it |
| 21:56.07 | ManxPower | terracon: SDSL can be used to transport T-1 or data or whatever |
| 21:56.14 | terrapen | ah |
| 21:56.22 | tzanger | HDSL2 is usually used ot transport T1 these days |
| 21:56.24 | Uther_P | SDSL is analog though |
| 21:56.30 | tzanger | actually HDSL2 is also what SDSL seems to be |
| 21:56.33 | Uther_P | just like adsl |
| 21:56.36 | terrapen | Total Access 600 Series, SDSL |
| 21:56.36 | terrapen | <PROTECTED> |
| 21:56.36 | terrapen | <PROTECTED> |
| 21:56.36 | terrapen | <PROTECTED> |
| 21:56.42 | tzanger | as the chipset in every single SDSL modem I've seen is actually an HDSL2 chipset |
| 21:56.47 | ManxPower | terracon: regardless you NEVER SEE IT, it's still presented to you as a DXS-1 |
| 21:56.49 | tzanger | Brooktree something IIRC |
| 21:56.59 | terrapen | hrmmm |
| 21:57.09 | ManxPower | ..er.. DSX-1 |
| 21:57.09 | bile_one | ManxPower, what about the log file size. Is there still a limit or is that fixed too? |
| 21:57.16 | ManxPower | bile_one: no idea |
| 21:57.21 | terrapen | i wish my telco was asterisk-friendly |
| 21:57.28 | terrapen | "Oh, yes sir, we can do this for you" |
| 21:57.32 | bile_one | ManxPower, thanks I'll see if I can find it. |
| 21:58.20 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 21:59.22 | ManxPower | I may have to stay at a Hostel |
| 21:59.28 | bkw_ | HOUSTON, Texas (AP) -- Texas sued the nation's largest Internet-based phone service provider Tuesday, saying Vonage failed to clearly inform customers they cannot automatically dial 911 when they sign up |
| 21:59.33 | bkw_ | http://www.cnn.com/2005/TECH/internet/03/23/internet.phones.911.ap/index.html |
| 21:59.40 | bkw_ | I think vonage clearly states this during signup |
| 22:00.03 | bkw_ | and sends this really bright coloered paper saying so when you signup |
| 22:00.37 | debaser | bkw_: this involves texans. i'm not too suprised. |
| 22:00.47 | tekjacob | any suggestions for no sound? |
| 22:01.02 | bkw_ | debaser, hahaha true |
| 22:01.10 | bkw_ | its this nice big ass red box too during signup |
| 22:01.14 | Essobi | <PROTECTED> |
| 22:01.20 | bkw_ | Emergency calling service or Dialing 911 requires activation (and that you provide a physical location). The 911 service Vonage offers is very different from traditional 911 and/or E911, which are not available at this time. For more information, we urge you to review the "911 Dialing" section under "Products & Services" and the "Emergency Services - 911 Dialing" section in the Vonage Terms of Service. |
| 22:01.57 | BrianR___ | I just wish I could find a good list of 10 digit numbers of PSAP's in my area... |
| 22:02.20 | debaser | bkw_: it'll get thrown out, probably. |
| 22:03.08 | bile_one | ManxPower, where ar eyou going? |
| 22:03.11 | bile_one | are |
| 22:03.26 | ManxPower | bile_one: Today I'm going to Mandeville, Louisiana. |
| 22:03.31 | ManxPower | But I don't need a hotel for that. |
| 22:03.50 | ManxPower | bile_one: May 19: Stockholm/Von 2005 Europe. |
| 22:03.59 | ManxPower | Also attending Astricon in Madrid |
| 22:04.19 | bile_one | I once drove through there. I left a black pickup in the parking lot, anmd never went back to get it. 1979. |
| 22:05.07 | johnnyb | Is Zap/pseudo-XXXX the channel used for MeetMe? |
| 22:05.44 | ManxPower | bile_one: Where? |
| 22:06.29 | bile_one | Mandeville, If i remember right they have a papermill or lumber yard there |
| 22:07.13 | TomL | i am in qos hell |
| 22:07.26 | bile_one | Oh I hate that place |
| 22:07.43 | ManxPower | bile_one: Ah. |
| 22:07.54 | dersteer | TomL: I've been there before :) |
| 22:08.02 | bile_one | ManxPower I was stationed at Barksdale AFB |
| 22:08.09 | TomL | IAX is echo-y over WAN with jitter buffering; SIP is not but its jittery |
| 22:08.12 | ManxPower | bile_one: 1979? It's changed a bit since then. Now it's a "bedroom community" for New Orleans |
| 22:08.22 | TomL | either I get severe echo, or broken speech |
| 22:08.32 | TomL | nothing in between |
| 22:08.50 | TomL | blend these! |
| 22:09.06 | *** part/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 22:09.18 | ta[i]nted | how do u guys handle redundancy in case asterisk goes down? |
| 22:09.22 | bile_one | TomL I assume you have already turned on echo training correct? |
| 22:09.27 | BrianR___ | If I use 'save dialplan' are macros saved expanded or not expanded? |
| 22:09.28 | TomL | oh yea |
| 22:09.30 | ManxPower | ta[i]nted: we don't |
| 22:09.41 | TomL | using POTS trunking, TDM22B |
| 22:09.41 | ManxPower | BrianR___: I don't think save dialplan even works |
| 22:09.44 | ta[i]nted | ManxPower what if it locks up or goes down |
| 22:09.50 | bile_one | restart |
| 22:09.53 | ManxPower | ta[i]nted: someone restarts it |
| 22:10.01 | bile_one | haa haa haa haa |
| 22:10.02 | ta[i]nted | that is not good |
| 22:10.12 | TomL | the inbound caller on POTS get bad echo if I'm on IAXy, and bad jitter if I'm on Sipura |
| 22:10.27 | ta[i]nted | so no failovers whatsoever? |
| 22:10.38 | TomL | going the other way is golden, either device |
| 22:10.47 | TomL | I get no jitter and no echo on my end |
| 22:12.07 | ManxPower | ta[i]nted: Do you currently run redundant PBXs? |
| 22:12.13 | TomL | the echo goes away with the Sipura, so its not an imbalanced hybrid |
| 22:12.24 | ta[i]nted | i'm not doing any traditional PBXing |
| 22:12.37 | TomL | the Sipura has echo cancelling in the ATA whereas IAXy does not |
| 22:12.58 | TomL | its the jitter buffering that causes the echo -- but without it, its very jittery |
| 22:13.01 | ta[i]nted | i was just looking for some good failover strategies |
| 22:13.30 | ta[i]nted | detecting hung asterisk processes etc.. re-routing to other peers |
| 22:13.45 | ManxPower | ta[i]nted: You have to make sure asterisk doesn't crash. |
| 22:13.57 | ManxPower | For me Asterisk crashes about every 3 months or so. |
| 22:14.19 | ta[i]nted | what kind of call volume do u do though? |
| 22:14.22 | TomL | the ATA in this case reaches the * with POTS inbound via DSL -> Cisco 7204 -> ATM-to-FR internetworking (on same ATM circuit as DSL PVCs) -> Linux router |
| 22:14.24 | ta[i]nted | 3 concurrent calls? |
| 22:14.28 | ManxPower | ta[i]nted: not high -- yes. |
| 22:14.33 | ManxPower | ta[i]nted: up to 8 calls at a time |
| 22:14.43 | ta[i]nted | yea |
| 22:14.48 | TomL | with ATM ... no outgoing service-policies. fuck. |
| 22:14.52 | ManxPower | Actually, no the most recent system does up to 18 channels at a time across 4 ports |
| 22:15.09 | bile_one | My box has been up for 37 days |
| 22:15.10 | ManxPower | plus a TDM card and 2 SIP clients |
| 22:15.27 | TomL | I can rate-limit the cisco's brains out but its still jittery |
| 22:15.42 | ta[i]nted | yea, i'm sure your set up doesn't need any kind of redundancy other than reboots and backup cards |
| 22:15.52 | ManxPower | TomL: QoS is REALLY hard to make work right. |
| 22:15.58 | bile_one | I have 10 Sips, two X100P's , 3 iaxs. |
| 22:16.00 | *** join/#asterisk xarg (~Administr@ool-4354c55c.dyn.optonline.net) |
| 22:16.08 | ta[i]nted | bile_one that's cute |
| 22:16.13 | TomL | ManxPower: no shit :) |
| 22:16.27 | ta[i]nted | bile_one are all of them hooked to your intercom? |
| 22:16.36 | TomL | i've got everything smoothed out in one direction, but I can't for the life of me get it working the other way |
| 22:16.59 | bile_one | Nope. But the sips are 2 sipura 2000 and the rest X-lite |
| 22:17.11 | Uther_P | sipura 2000 boxes are cool |
| 22:17.47 | bile_one | So they host 2 analog phones each |
| 22:18.04 | bile_one | I will see it anyway! |
| 22:18.08 | *** part/#asterisk xarg (~Administr@ool-4354c55c.dyn.optonline.net) |
| 22:18.09 | ManxPower | TomL: I've been trying to put QoS on the corporate WAN for almost 2 years. 1 month ago I got it working. |
| 22:18.21 | ManxPower | Well, got it working and didn't have users screaming at me. |
| 22:18.22 | TomL | I'll hafta draw a diagram in a bit and see if you have any ideas :) |
| 22:18.26 | *** join/#asterisk xarg (~Administr@ool-4354c55c.dyn.optonline.net) |
| 22:18.38 | ManxPower | TomL: The key thing to remember is you can ONLY QoS TRANSMITTED data. |
| 22:18.46 | Supaplex | will the linksys ata run/accept/operate on sipura firmware? (or am I asking a silly question, because it's already sipura firmware to begin with...) |
| 22:18.49 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 22:18.53 | TomL | and on ATM, you specifically CANNOT do that :/ |
| 22:19.05 | ManxPower | TomL: Also remember that QoS for voice does NOT WORK on Frame Relay unless your CIR is the same as your port speed. |
| 22:19.11 | TomL | you can only police incoming rates on an ATM circuit, AFAIK |
| 22:19.19 | ManxPower | TomL: I know nothing about ATM QoS |
| 22:19.28 | TomL | me neither :( |
| 22:19.30 | TomL | heh |
| 22:19.37 | ta[i]nted | ManxPower do u have any tips for slimming down unused asterisk modules? |
| 22:19.40 | ManxPower | TomL: you want LLQ aka "priority", not police. At least on non-ATM stuff. |
| 22:19.53 | ManxPower | ta[i]nted: No. My Asterisk is not too fat. |
| 22:20.08 | ManxPower | ta[i]nted: I noload => the chan_protocolidontwant.so of course. |
| 22:20.11 | TomL | yea, there's no "ip rtp priority" for ATM interfaces |
| 22:20.21 | TomL | = I'm fucked |
| 22:20.25 | ManxPower | TomL: hold on. |
| 22:20.30 | Uther_P | yea, there is no QoS on SONET Ring networks |
| 22:20.39 | TomL | this isn't SONET |
| 22:20.40 | bile_one | haa haa haa |
| 22:20.45 | epoch | er, don't you not need to do QoS on ATM? |
| 22:20.47 | bile_one | We are giving up too easy |
| 22:20.54 | TomL | its DS3 |
| 22:20.59 | epoch | like, doesn't ATM do QoS at the lowest layer? |
| 22:21.04 | doughecka | ~seen atacomm |
| 22:21.06 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 48d 20h 24m 13s ago, saying: 'anyone want a IP 3000 conference phone? looking to replace ours with a IP 4000 model. Barely been used, in great condition.... looking for around $500'. |
| 22:21.15 | Supaplex | QoS on ATM makes little sence anyway. The're all small cells of packets. Don't you put QoS a layer or two up in the network stack? (think OSI model here) |
| 22:21.23 | dant | you've PVC would be set up as vbr-rt? |
| 22:21.31 | dant | you're |
| 22:21.36 | dant | bah, can't type |
| 22:21.36 | jontow | alright.. i think i've mostly hacked up this SMDI bullshit |
| 22:21.38 | TomL | I'm only allowed ubr by the telco |
| 22:21.43 | dant | ahh |
| 22:21.54 | epoch | dant: "your" ;) |
| 22:21.55 | TomL | =, again, me fucked |
| 22:22.11 | ManxPower | TomL: http://pastebin.ca/8076 |
| 22:22.12 | jontow | its to the point now where i understand.. one can simply ignore ALL incoming SMDI events.. literally, and just focus on asterisk's view of things.. externnotify = ... and a quick patch to system() a shell script to spawn the correct bullshit :) |
| 22:22.15 | ManxPower | That's how I do it. |
| 22:22.26 | dant | epoch, I said I couldn't type, the fact my english is pnats too is about right :) |
| 22:22.32 | dant | epoch, I blame the jetlag :) |
| 22:22.34 | Uther_P | if you are connect to an ATM, usually the priority is decided by customer and setup by the telco, based on your SLA |
| 22:22.37 | ManxPower | I just make SURE all my VoIP devices use DSCP EF / TOS 0x8B |
| 22:23.06 | TomL | the ATM layer is not carrying voice directly, just IP traffic |
| 22:23.11 | Uther_P | qos is lower level then than, for routers to decide which of your packets or packets from customers of equal priority to go through first |
| 22:23.21 | epoch | dant: hehe |
| 22:23.32 | ManxPower | TomL: You need to talk to your ISP to get QoS set up on your connection. |
| 22:23.49 | Uther_P | you need an SLA |
| 22:23.58 | ManxPower | or your WAN provider, or whatever. |
| 22:23.58 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. The the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller hears ring, then eventually goes dead. |
| 22:24.14 | doughecka | anyone hear from atacomm? |
| 22:24.19 | TomL | in this case, I am the "ISP" |
| 22:24.22 | doughecka | he used to lurk in here |
| 22:24.25 | Uther_P | jesster: sounds like you have call waiting on the line |
| 22:24.29 | TomL | I have control of every router in the link |
| 22:24.31 | ManxPower | Lets see. It's 4:23pm. I have a migration to do at 5pm. I'm 45 mins from the customer. My ride is not here yet. |
| 22:24.34 | dant | TomL, sorry, I've come into this one late |
| 22:24.41 | doughecka | ManxPower, migration? |
| 22:24.53 | jesster | Uther_P: that may be true, however the analog phone does not hear the ring for callwaiting |
| 22:24.53 | ManxPower | doughecka: switch from old PBX to Asterisk |
| 22:24.56 | doughecka | ah |
| 22:24.59 | ManxPower | there he ius |
| 22:25.10 | dant | TomL, you have a ubr pvc between two points and you want to prioritise voice traffic at your interface? |
| 22:25.15 | bile_one | Later Manx |
| 22:25.16 | TomL | ManxPower: "service-policy output" is disallowed on ATM interfaces and sub-interfaces |
| 22:25.48 | TomL | dant: let me redescribe |
| 22:26.50 | TomL | Sip1001 -> DSL (PPPoA) -> Cisco 7204 -> ATM-to-FR internetworking (diff PVC on same ATM circuit) -> Linux router -> * -> POTS trunks |
| 22:27.21 | dant | ok |
| 22:27.39 | *** join/#asterisk madounet (~mad|net@juvenal-3-82-226-155-19.fbx.proxad.net) |
| 22:27.43 | *** join/#asterisk mqht (~mtht@roam.wblib.org) |
| 22:27.51 | TomL | sip is jittery in one direction only, iaxy has severe echo |
| 22:27.51 | mqht | Hi all |
| 22:28.22 | mqht | I am having an issue with AGI, it seems that the script keeps running waiting for input even after a user hangs up.....Any ideas? |
| 22:28.35 | TomL | I don't think the IAXy echo problem can be solved without turning off jitter buffer, which puts it in the same boat as the Sipura |
| 22:28.59 | dant | TomL, where are you unable to set the service-policy output? |
| 22:29.02 | Uther_P | TomL: the echo can be caused in combination with latency + sidetone from the pots and/or pstn |
| 22:29.11 | TomL | on any ATM interface or sub interface |
| 22:29.22 | TomL | Uther_P: the echo is not present on the Sipura |
| 22:29.36 | TomL | i only get echo if I repace the Sipura with an IAXy |
| 22:29.52 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. The the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller hears ring, then eventually goes dead. I've pasted my relevant zapata.conf here http://pastebin.ca/8077 |
| 22:30.02 | dant | TomL, and is it giving an error when you try to set it? |
| 22:30.10 | TomL | yea |
| 22:30.33 | dant | R02(config)#in ATM1/IMA0.1 point-to-point |
| 22:30.34 | dant | R02(config-subif)#service-policy output test |
| 22:30.34 | dant | R02(config-subif)# |
| 22:31.52 | Uther_P | I had this same problem going through 2 sets of analog equipment (sipura -> VoIP -> fxs) and derived the problem to be that the latency was > than the # of samples for the echo cancelation software.... which caused the sidetone to create an amplified echo / reverb effect |
| 22:33.16 | TomL | sure, but what's in policy "test"? |
| 22:33.31 | dant | TomL, bugger all, I just made it to test it :) |
| 22:33.37 | TomL | binelli(config-pmap)#class voice |
| 22:33.37 | TomL | binelli(config-pmap-c)#priority 128 |
| 22:33.37 | TomL | CBWFQ : Not supported on subinterfaces |
| 22:33.37 | TomL | binelli(config-pmap-c)# |
| 22:33.44 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 22:34.13 | xarg | if I am simply looking to hook something up to an analog port on the Partner ACS at my office so that I can have a VoIP extension at a remote location (most likely soft client) and my resources consist of a 500mhz p3 with an ISA Voice modem and $0 can I use asterisk to accomplish what I wish? if a simple voice modem will work is there a list of ones which are compatible? |
| 22:34.35 | doughecka | xarg, nope |
| 22:34.41 | xarg | dang |
| 22:34.45 | doughecka | that wont work, you will need to get a card |
| 22:34.50 | doughecka | the box should work for 1 caller |
| 22:35.36 | doughecka | you can find cards on ebay for cheaper than digium's pricing, but its not supported and doesnt support the people that develope asterisk |
| 22:35.54 | Uther_P | xarg: you can get an fxo card, or you can get an fxo to sip external box |
| 22:35.58 | *** join/#asterisk AgiNamu (~Bob@12.172.224.49) |
| 22:36.19 | DannyF | xarg just get two iaxy's ,) |
| 22:36.28 | DannyF | cheap and still support digium ;) |
| 22:36.39 | Shido6 | 99 |
| 22:36.43 | Shido6 | bucks |
| 22:36.46 | xarg | did anyone read that my budget is $0 |
| 22:36.50 | DannyF | okok relaitoively cheap then ,) |
| 22:36.50 | Shido6 | goes a long way wiff those iaxys |
| 22:36.55 | AgiNamu | get 2 PA168's for the same price, then PayPal $100 to digium. |
| 22:36.59 | TomL | if you're budget is $0, you're fucked |
| 22:37.16 | Uther_P | heh yea, in that case you got nothin |
| 22:37.23 | Uther_P | kick $100 out of your boss |
| 22:37.43 | AgiNamu | what was teh question? :) |
| 22:37.46 | xarg | I am my boss and I have nothing |
| 22:37.51 | DannyF | or xlite and a roll of duct tape ;) |
| 22:38.05 | Supaplex | sorry, noone has contributed to the shoestring transport driver. (tin cans and a string) |
| 22:38.21 | bile_one | ta[i]nted, have you looked at this as a possible solution to your redundantcy solution? http://www.iptel.org/ser/ |
| 22:38.40 | AgiNamu | SER -- spit. |
| 22:38.51 | *** join/#asterisk SpaceBass (~sp@24.125.33.214) |
| 22:38.59 | SpaceBass | anyone ever used a ag-168 ata? |
| 22:39.10 | AgiNamu | Spacebass, yea |
| 22:39.12 | SpaceBass | bought one off e-bay, its brand new but the only instructions are in chinese |
| 22:39.14 | AgiNamu | he PA168 |
| 22:39.18 | AgiNamu | rocks |
| 22:39.19 | SpaceBass | cannot find the default password anywhere |
| 22:39.24 | debaser | xarg: basically, you're SOL. need a car a sailboat to go sailing. |
| 22:39.25 | AgiNamu | 19800211 |
| 22:39.26 | dant | TomL, http://www.cisco.com/warp/public/105/qos_subint.html |
| 22:39.41 | AgiNamu | try that SpaceBass. |
| 22:39.44 | SpaceBass | no dice |
| 22:39.59 | AgiNamu | and 1234 or 12345678 didnt work? |
| 22:40.10 | SpaceBass | 1234! |
| 22:40.12 | SpaceBass | YES! |
| 22:40.13 | SpaceBass | thanks! |
| 22:40.15 | SpaceBass | AH HA! |
| 22:40.18 | *** join/#asterisk sezuan (sezuan@port-212-202-55-249.dynamic.qsc.de) |
| 22:40.19 | AgiNamu | that is probably the user paswword |
| 22:40.25 | AgiNamu | 12345678 is probably the admin password. maybe. |
| 22:40.28 | SpaceBass | thank you so much |
| 22:40.42 | SpaceBass | both seem to work |
| 22:40.47 | AgiNamu | if not, "boot into safe mode" -- hold * while booting. |
| 22:40.57 | AgiNamu | there are 2 levels of password |
| 22:41.03 | AgiNamu | user (cant change much) and admin |
| 22:41.07 | SpaceBass | your right, now that i used 12345678 i can see that |
| 22:41.11 | AgiNamu | You using it for IAX2? |
| 22:41.13 | TomL | dant: wow!!!! |
| 22:41.24 | SpaceBass | aig was planning on sip |
| 22:41.25 | AgiNamu | lol |
| 22:41.29 | AgiNamu | ewww. sip. |
| 22:41.36 | AgiNamu | anyways, 1.42 was just released yesterday. |
| 22:41.37 | *** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net) |
| 22:41.40 | SpaceBass | should I use IAX2? never had any expirence with it |
| 22:41.47 | AgiNamu | IAX2 rocks SIP |
| 22:42.01 | Damin_Mobile | Space: Yes! |
| 22:42.08 | SpaceBass | now... where to set that |
| 22:42.16 | AgiNamu | IAX2 needs a separate firmware. |
| 22:42.16 | Damin_Mobile | iax2 is the shit! |
| 22:42.21 | SpaceBass | lol |
| 22:42.31 | SpaceBass | AgiNamu where do I donwnload |
| 22:42.31 | SpaceBass | ? |
| 22:42.35 | AgiNamu | Onbly v1.38 had al;l the protocls in one single binary |
| 22:42.39 | SpaceBass | couldnt find anything in engrish for this thing |
| 22:43.24 | AgiNamu | but in 1.38, IAX2 only had an option. it didnt actually work. |
| 22:43.30 | AgiNamu | OK, email me: mgg@atrevido.net |
| 22:43.36 | AgiNamu | I'll build you some firmware |
| 22:43.50 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
| 22:43.51 | AgiNamu | heh |
| 22:44.44 | SpaceBass | Beirdo beat you to it... lets see... porn, viagra, free ipods.. that should do it |
| 22:44.45 | SpaceBass | :) |
| 22:44.46 | SpaceBass | kidding! |
| 22:44.55 | AgiNamu | ehe |
| 22:45.00 | AgiNamu | sounds like a good mix. |
| 22:45.04 | mog_home3 | iax2 firmware |
| 22:45.23 | Beirdo | make sure it's the really degraded pr0n |
| 22:45.48 | SpaceBass | oh, of course |
| 22:45.50 | AgiNamu | degraded? like bad quality |
| 22:45.54 | AgiNamu | or did you mean degenerate? :) |
| 22:46.08 | SpaceBass | (not that I know where to find that kind of stuff... i mean... i dont look at it or anything...seriously) |
| 22:46.15 | Supaplex | junk pr0n |
| 22:46.16 | file[laptop] | porn? where? |
| 22:46.20 | SpaceBass | or degrading? |
| 22:46.21 | AgiNamu | hehe |
| 22:46.22 | Beirdo | yeah, degenerate |
| 22:46.30 | Beirdo | damn I suck today |
| 22:46.34 | AgiNamu | how much you pay for the 168 ATA? |
| 22:46.42 | file[laptop] | Beirdo: suck? |
| 22:46.43 | mog_home3 | iax2 firmware? |
| 22:46.44 | Beirdo | you know like "daughters and their dogs". crap like that |
| 22:47.03 | Supaplex | so you're the one with the ideas haha |
| 22:47.07 | *** join/#asterisk fugitivo (~ajf@201.255.104.167) |
| 22:47.07 | AgiNamu | "farmers and their hogs" |
| 22:47.14 | SpaceBass | AgiNamu 47 |
| 22:47.15 | SpaceBass | usd |
| 22:47.19 | SpaceBass | good or bad? |
| 22:47.19 | Beirdo | yeah, that stuff |
| 22:47.26 | bile_one | and goats and donkey's with mexican women |
| 22:47.36 | AgiNamu | Hi fugitivo, we're talking about degenerate, degraded porn. |
| 22:47.36 | AgiNamu | good price |
| 22:47.40 | AgiNamu | where did you buy it from? |
| 22:47.44 | doughecka | the porn? |
| 22:47.49 | SpaceBass | you cannot prove that I have gone to www.daughtors&dogs.com... i use a proxy! |
| 22:47.53 | AgiNamu | www.tortillatossers.com? :P |
| 22:47.54 | SpaceBass | AgiNamu ebay |
| 22:48.04 | AgiNamu | space, but where in the world |
| 22:48.04 | AgiNamu | China? |
| 22:48.14 | SpaceBass | good question, I'd have to check the shipping address |
| 22:48.15 | AgiNamu | how much did you pay in shipping? |
| 22:48.19 | *** join/#asterisk mesi (~player@dsl-082-083-055-218.arcor-ip.net) |
| 22:48.24 | SpaceBass | $10.00 |
| 22:48.27 | AgiNamu | the pa168 has firmware for H323, SIP, IAX2, MGCP, and Net2Phone. |
| 22:48.32 | AgiNamu | wow spac.e |
| 22:48.32 | AgiNamu | not bad. |
| 22:48.47 | SpaceBass | came from florida |
| 22:48.54 | SpaceBass | seller has a few more online still i think |
| 22:48.57 | AgiNamu | wow thatsa good price. |
| 22:49.05 | AgiNamu | i buy em in china\ |
| 22:49.10 | AgiNamu | Gotl ike 50 sitting around my office. |
| 22:49.11 | SpaceBass | really? I might have to scarf up 2 more |
| 22:49.19 | doughecka | what |
| 22:49.22 | AgiNamu | yea, at that price, you should. |
| 22:49.26 | AgiNamu | PA168 ATAs |
| 22:49.32 | doughecka | what are them |
| 22:49.34 | doughecka | :P |
| 22:49.41 | AgiNamu | Supposedluy we're getting some 2FXS 1FXO ATAs sooon |
| 22:49.45 | *** join/#asterisk sabre (~urfos@69.149.209.83) |
| 22:49.46 | AgiNamu | GOOD ATAs with IAX2 support |
| 22:49.47 | ta[i]nted | does musiconhold use app_mp3.so ? |
| 22:49.49 | doughecka | oooooh |
| 22:49.53 | doughecka | I WANT ONE |
| 22:49.56 | SpaceBass | how can I tell which firmware i have now? |
| 22:50.00 | AgiNamu | and codec support for 723, 729, gsm, ulaw, alaw and soon ilbc. |
| 22:50.06 | AgiNamu | when you login to the web admin |
| 22:50.11 | doughecka | sweet |
| 22:50.13 | doughecka | with webgui? |
| 22:50.14 | doughecka | wow |
| 22:50.16 | AgiNamu | it'll saty soeting like "WuChuan v1.41.007" |
| 22:50.27 | AgiNamu | HTTP, Telnet, TFTP, FTP |
| 22:50.42 | bile_one | ta[i]nted, I have a pssoble solution to your back guestion. |
| 22:50.47 | bile_one | possible |
| 22:50.50 | AgiNamu | whats tainted's question? |
| 22:50.51 | SpaceBass | I'ma looking but I ain't seein |
| 22:50.58 | Damin_Mobile | aginamu: ThAR" |
| 22:50.59 | jontow | box = 0; /* Shut up compiler */ |
| 22:51.00 | jontow | hehehe |
| 22:51.07 | bile_one | He asked if we have some kind of backup for asterisk |
| 22:51.10 | AgiNamu | "ThAR"? |
| 22:51.14 | bile_one | Like failover |
| 22:51.16 | AgiNamu | backup? |
| 22:51.25 | bile_one | yep, |
| 22:51.26 | AgiNamu | Oh yea. it's called Linux High Availability. |
| 22:51.31 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 22:51.35 | AgiNamu | just mirror the thing, with IP takover. |
| 22:51.36 | bile_one | Or rsync |
| 22:51.41 | bile_one | yep! |
| 22:51.43 | AgiNamu | rsync does IP takeover?\ |
| 22:51.50 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 22:51.53 | bile_one | nope |
| 22:51.59 | AgiNamu | oh |
| 22:52.15 | bile_one | but you can rsync and network reconfigure |
| 22:52.16 | AgiNamu | actually, I was thinking of PXE boot |
| 22:52.24 | dant | keepalived's vrrp does ip takeover |
| 22:52.25 | AgiNamu | yea, but how do you network configure in 5 seconds? :) |
| 22:52.49 | linenoise | unplug it |
| 22:52.49 | bile_one | type real fast or issue a script |
| 22:53.04 | AgiNamu | yea, anyways, theres tons of stuff to do that |
| 22:53.06 | AgiNamu | nothing to invent there. |
| 22:53.29 | dant | or... |
| 22:53.51 | bile_one | HAve you read the digium list on that. Wheew! |
| 22:53.51 | dant | for phones at least, have a backup gateway |
| 22:53.52 | AgiNamu | I'm surprised digium hasnt gotten into IAX hardware |
| 22:53.59 | AgiNamu | like a nice IAX phone |
| 22:54.00 | AgiNamu | or ATA |
| 22:54.26 | doughecka | they make a great ata |
| 22:54.33 | SpaceBass | the iaxy? |
| 22:54.33 | doughecka | iaxy |
| 22:54.43 | bile_one | haa haa haa |
| 22:54.48 | *** join/#asterisk SkySky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 22:54.49 | doughecka | what |
| 22:54.52 | AgiNamu | well, your definition of "great" is different than mine. |
| 22:54.59 | bile_one | Mine too |
| 22:55.04 | AgiNamu | Telnet only, G711 Only doesnt exactly qualify as great :P |
| 22:55.14 | doughecka | I have been using one on my desk for a few months... |
| 22:55.19 | doughecka | oh, but it works |
| 22:55.21 | AgiNamu | and it's double what the PA168s are. |
| 22:55.21 | SkySky | hi.. does anyone know of some application to test my actual bandwidth limit ? |
| 22:55.28 | AgiNamu | since when does "works" == "great" ? |
| 22:55.29 | doughecka | gimme a website for that iax2 ata |
| 22:55.30 | terrapen | OH BOY! |
| 22:55.34 | terrapen | a phishing scam! |
| 22:55.36 | SpaceBass | SkySky wdc.speakeasy.net |
| 22:55.36 | terrapen | i love these |
| 22:55.52 | nestAr | terrapen: hi, I'm with Regions Bank |
| 22:55.54 | doughecka | wait, the iaxy supports telnet? |
| 22:55.55 | bkw_ | Emergency calling service or Dialing 911 requires activation (and that you provide a physical location). The 911 service Vonage offers is very different from traditional 911 and/or E911, which are not available at this time. For more information, we urge you to review the "911 Dialing" section under "Products & Services" and the "Emergency Services - 911 Dialing" section in the Vonage Terms of Service. |
| 22:55.55 | doughecka | =D |
| 22:55.56 | bkw_ | doh |
| 22:55.58 | terrapen | looks like someone else is using my takedown script too |
| 22:55.59 | bkw_ | OMG OMG OMG |
| 22:56.01 | bkw_ | CHANSPY |
| 22:56.02 | bkw_ | CHANSPY |
| 22:56.04 | nestAr | terrapen: can you give me your account? |
| 22:56.06 | doughecka | aahhh |
| 22:56.06 | doughecka | aahhh |
| 22:56.27 | doughecka | aahhh |
| 22:56.27 | nestAr | :x |
| 22:56.27 | ArkyLady | haha :D |
| 22:56.27 | terrapen | dammit bkw :P |
| 22:56.45 | terrapen | http://84.247.60.119/1/index.htmlk |
| 22:56.46 | terrapen | err |
| 22:56.47 | terrapen | http://84.247.60.119/1/index.htmlk |
| 22:56.49 | terrapen | dammit |
| 22:56.50 | terrapen | http://84.247.60.119/1/index.html |
| 22:56.56 | doughecka | LOLOL |
| 22:56.58 | terrapen | PREPARE LASERS FOR DESTRUCTION |
| 22:57.02 | terrapen | ALL LASERS FULL, CHECK |
| 22:57.04 | Grooby | isn't there new config options i can play with for the new jitter buffer? |
| 22:57.09 | Grooby | or just jitterbuffer=yes |
| 22:57.17 | SpaceBass | AgiNamu would it normally take a few minutes to update firmware? |
| 22:57.31 | AgiNamu | spacebass, yea, the upgrade is slow |
| 22:57.38 | SpaceBass | AgiNamu I downloaded 1.360 and its taking for ever |
| 22:57.42 | AgiNamu | if you use PalmTool, it goes faster. |
| 22:57.44 | SpaceBass | just wanted to try the process |
| 22:57.47 | AgiNamu | 1.36 is ANCIENT |
| 22:57.49 | terrapen | anybody want the skr1pt? |
| 22:57.53 | SpaceBass | oh really? |
| 22:58.02 | stevekstevek | Grooby: see README.jitterbuffer |
| 22:58.04 | doughecka | AgiNamu, what website can I get that ata from? |
| 22:58.09 | Grooby | going there now steve |
| 22:58.14 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 22:58.14 | Grooby | was looking at the wrong place |
| 22:58.15 | Grooby | ;) |
| 22:58.20 | AgiNamu | yea. firmware upgrade is slow. and we're on v1.42 |
| 22:58.29 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 22:58.39 | AgiNamu | if you email me, mgg@atrevido.net, I can sell you a few. but they arent as cheap as spacebass paid. |
| 22:58.48 | Damin_Mobile | AgiNamu: Iax2 ata...where from? |
| 22:58.49 | doughecka | yea, but from where :) |
| 22:58.56 | AgiNamu | PA168 chips |
| 22:59.01 | terrapen | oh rats. |
| 22:59.03 | AgiNamu | I'm in guatemala. we've imported them from guatemal |
| 22:59.05 | terrapen | my home machine is offline |
| 22:59.06 | doughecka | if I want to buy some it will be in large quantity |
| 22:59.07 | doughecka | ah |
| 22:59.09 | AgiNamu | we send up to the states every so often. |
| 22:59.09 | terrapen | no takedown for me |
| 22:59.14 | terrapen | Agi, where in .gt? |
| 22:59.20 | SpaceBass | AgiNamu I e-mailed you re: the firmware dd it make it through? |
| 22:59.21 | AgiNamu | then we'll probablu just ship from china. depending on order. |
| 22:59.27 | AgiNamu | liek what, 20,000 pcs? |
| 22:59.37 | AgiNamu | I'm not at home. on someone elses machine using IRC to kill time. |
| 22:59.43 | SpaceBass | ahh haaa |
| 22:59.45 | AgiNamu | near GT city. why terra? |
| 22:59.46 | SpaceBass | where in the world are you? |
| 22:59.55 | AgiNamu | im in guatemala right now. |
| 22:59.59 | SpaceBass | geographically... |
| 23:00.00 | terrapen | AgiNamu, one of my best friends has a home in Antigua |
| 23:00.01 | SpaceBass | oh |
| 23:00.08 | terrapen | but he lives in the States, too |
| 23:00.09 | AgiNamu | heh, cool |
| 23:00.15 | AgiNamu | wow, he's big./ |
| 23:00.27 | terrapen | agi, we have been talking about starting a WISP down there |
| 23:00.34 | terrapen | how hard is it to get connectivity? |
| 23:00.37 | doughecka | AgiNamu, quantity being like 3 |
| 23:00.37 | AgiNamu | oh really? |
| 23:00.38 | doughecka | :P |
| 23:00.41 | terrapen | ie., 10Mbit, etc |
| 23:00.54 | AgiNamu | doug, yea, email me, and ill pass it off to the business manager. |
| 23:01.04 | AgiNamu | 10MBit? good luck :P. |
| 23:01.09 | AgiNamu | it'll cost you |
| 23:01.15 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 23:01.19 | AgiNamu | Talk to InstaRed, or Telgua. they've got the most connectivity. |
| 23:01.22 | terrapen | what about a T1 |
| 23:01.28 | terrapen | yeah, i've seen InstaRed |
| 23:01.30 | AgiNamu | I think InstaRed ran a T3 from Miami. |
| 23:01.31 | terrapen | are they expensive? |
| 23:01.32 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. The the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller hears ring, then eventually goes dead. I've pasted my relevant zapata.conf here http://pastebin.ca/8077 |
| 23:01.41 | terrapen | over the ocean? |
| 23:01.48 | AgiNamu | and Telgua (horrible monopoly) has a T3s, I think |
| 23:01.52 | AgiNamu | terra, yea. fibre. |
| 23:01.56 | AgiNamu | telgua goes thru mexico, fibre. |
| 23:02.06 | AgiNamu | well, 4 years ago, InstaRed charged $400 for 128k. |
| 23:02.22 | AgiNamu | on top of that, you run into a lot of wireless licensing issues. |
| 23:02.29 | AgiNamu | you ahve to secure radio licenses first. good luck with that. |
| 23:02.36 | AgiNamu | Plus, you're competing pretty hard. |
| 23:02.52 | AgiNamu | Telefonica/Bellsouth have wireless (CDMA + EVDO) as does telgua. |
| 23:02.53 | ta[i]nted | AgiNamu what is Linux High Availability |
| 23:03.05 | elriah | Is there a way to get * to call out? maybe on a schedule? |
| 23:03.05 | SpaceBass | AgiNamu are you in international communications by chance? |
| 23:03.10 | dant | 4 years ago it was $800/month for 64k in the UK |
| 23:03.13 | AgiNamu | it's a [set of?] projects |
| 23:03.16 | ta[i]nted | AgiNamu did u ever get around to doing the res_mono.so ? ;) |
| 23:03.23 | AgiNamu | to enable failover clusters and so on |
| 23:03.34 | AgiNamu | tainted, i had some stuff, but A: Asterisk API changes a lot and B: it's not documented at all. |
| 23:03.46 | AgiNamu | So making a RELIABLE system with res_mono was gonna be a huge undertaking. |
| 23:03.52 | AgiNamu | It'd work if it got some committment from say, Mark. |
| 23:03.59 | AgiNamu | but that's never gonna happen. |
| 23:04.02 | ta[i]nted | bkw_ was interested in it |
| 23:04.13 | buddah | anyone here have experience with quintum tenor DX? |
| 23:04.17 | bkw_ | well |
| 23:04.19 | AgiNamu | yea.. we can do it. it's just a lot of resources. |
| 23:04.24 | bkw_ | res_mono or mono in general is gonna be an issue |
| 23:04.29 | bkw_ | the differences in threading |
| 23:04.31 | bkw_ | cause issues |
| 23:04.34 | bkw_ | that are beyond anything fixable |
| 23:04.45 | AgiNamu | well, you have to include their gc and stuff, afaik |
| 23:04.53 | SpaceBass | AgiNamu while I'm waiting to play with IAX2, can you help me with a SIP set up? |
| 23:04.54 | bkw_ | that don't work |
| 23:04.57 | bkw_ | I tried it |
| 23:05.07 | AgiNamu | you added their includes all over asterisk?? |
| 23:05.20 | bkw_ | yes |
| 23:05.22 | AgiNamu | Well, if Mark wanted to do it, he could talk with the Mono people, and im sure something could get fixed up. |
| 23:05.40 | AgiNamu | hosting mono would be a big win for both. |
| 23:05.40 | bkw_ | the way I was gonna do it was in the non-managed way |
| 23:05.46 | bkw_ | I didn't really understand the way you talked about |
| 23:05.48 | AgiNamu | non managed? |
| 23:05.51 | bkw_ | but i'm not a .net user |
| 23:05.58 | mog_home3 | bleck .net |
| 23:06.00 | AgiNamu | I had a sample running. call hit the extension |
| 23:06.06 | AgiNamu | and it ran the .net code just fine. |
| 23:06.08 | bkw_ | I had it doing that too |
| 23:06.14 | bkw_ | but it hangs |
| 23:06.17 | AgiNamu | where'd you run into problems? |
| 23:06.18 | bkw_ | never returns from the thread |
| 23:06.22 | bkw_ | no matter what I did |
| 23:06.34 | AgiNamu | strange. my test worked fine. |
| 23:06.40 | AgiNamu | hmmph. |
| 23:06.59 | bkw_ | ya what you gave me is where i started |
| 23:07.03 | bkw_ | now if you look after you hangup |
| 23:07.06 | bkw_ | the channel never goes away |
| 23:07.10 | bkw_ | its deadlocked |
| 23:07.10 | AgiNamu | well, those problems , plus what I mentioned ... its a hard issue |
| 23:07.16 | AgiNamu | oh hmm |
| 23:07.21 | *** join/#asterisk DenisL (~denis@68.148.230.233) |
| 23:07.26 | bkw_ | I did 100000 things |
| 23:07.29 | bkw_ | it was fun |
| 23:07.34 | AgiNamu | i decided to just run the C# code somewhere else, and use gSOAP to connect to it. |
| 23:07.34 | bkw_ | I got ast_log to work from mono |
| 23:07.39 | AgiNamu | cool |
| 23:07.40 | elriah | Is there a way to get * to call out? maybe on a schedule? |
| 23:07.50 | bkw_ | sample.call |
| 23:07.59 | jontow | cooooooolll |
| 23:08.01 | jontow | i have SMDI working |
| 23:08.03 | AgiNamu | gSOAP + ASP.NET is damn fast. good enough that I can do a few soap calls in the switching core. |
| 23:08.37 | jontow | just gotta plug it into the switch and let it go wild :) |
| 23:08.38 | elriah | What is SMDI? |
| 23:08.38 | AgiNamu | fast enough that im replacing the dialplan with a few soap calls and some logic. |
| 23:08.38 | SpaceBass | AgiNamu under service type... do i want that set to sipphone or comon? |
| 23:08.38 | jontow | a signalling-type protocol meant for voicemail systems to integrate with telco type switches |
| 23:08.38 | AgiNamu | common. |
| 23:08.38 | DenisL | I put the following: signalling=fxo_ks |
| 23:08.39 | DenisL | callerid="Radio Hack" <112> |
| 23:08.39 | DenisL | mailbox=112 |
| 23:08.39 | DenisL | extension=112 |
| 23:08.39 | DenisL | context=from-internal |
| 23:08.39 | DenisL | channel => 7 in my zapata.conf but zap show channels lists no extensions for any of the zap channels and from-pstn as the context for all channels. Its like zapata.conf is being ignored... What might I be doing wrong? |
| 23:08.41 | jontow | developed by bell when voicemail hardware wasn't the same as the switch :) |
| 23:08.50 | AgiNamu | ~pastebin |
| 23:08.51 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 23:08.53 | jontow | and just happens it is still in use by DMS-10 devices, coppercom switches, etc etc |
| 23:09.12 | SpaceBass | DenisL did you stop and restart asterisk? |
| 23:09.24 | SpaceBass | AgiNamu and then set all the ports to 5060? |
| 23:09.26 | jontow | im deploying asterisk as a voicemail server at a local telco, who uses SMDI to signal for stutter-dialtone on the home user's lines when they have new voicemail |
| 23:09.27 | DenisL | SpaceBass: I did "reload" at the console |
| 23:09.29 | AgiNamu | right |
| 23:09.35 | mxmasster|work | hi all |
| 23:09.38 | SpaceBass | DenisL try restart gracefully |
| 23:09.42 | SpaceBass | AgiNamu even for rtp? |
| 23:09.44 | AgiNamu | in v1.41 and before, the interface is th esame for MGCP, H323, SIP, IAX and N2P |
| 23:09.48 | AgiNamu | so you get a lot of stupid things |
| 23:09.49 | jontow | and with some simple shell scripting and some code-reading/patching.. i've got it simulated/working in my test environment :) |
| 23:09.53 | AgiNamu | well, no, RTP goes on whatever you use |
| 23:09.56 | AgiNamu | like 15000 or something |
| 23:10.08 | AgiNamu | if you dont know SIP, good luck :P. it's a bitch. |
| 23:10.18 | SpaceBass | AgiNamu haven't changed it from the default... |
| 23:10.28 | DenisL | SpaceBass: Same thing after a "gracefull restart" |
| 23:10.36 | SpaceBass | AgiNamu I haven't had problems configuring softphones at all, but this seems a bit different |
| 23:10.47 | SpaceBass | pastbin.ca your zapata.conf |
| 23:10.49 | *** join/#asterisk fgravato (~frankie@ool-44c02d18.dyn.optonline.net) |
| 23:10.52 | AgiNamu | its very different |
| 23:10.57 | AgiNamu | becahse it has settings for H323 and everything |
| 23:10.58 | DenisL | Ok. |
| 23:11.05 | SpaceBass | AgiNamu so i see |
| 23:11.07 | AgiNamu | v1.42 cleans it up |
| 23:11.22 | SpaceBass | but that doenst seem to be available for public download huh ? |
| 23:11.23 | SpaceBass | :) |
| 23:11.49 | SpaceBass | i'm like a kid a christmas... finally got my asterisk box working the way I want it and tired of using softphoens |
| 23:12.00 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. The the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller hears ring, then eventually goes dead. I've pasted my relevant zapata.conf here http://pastebin.ca/8080 |
| 23:12.14 | *** join/#asterisk ratapeluda (~m@80-28-34-225.adsl.nuria.telefonica-data.net) |
| 23:12.19 | AgiNamu | spacebass, it just got finished yesterday |
| 23:12.19 | ratapeluda | hi |
| 23:12.26 | SpaceBass | AgiNamu that might be why |
| 23:12.28 | DenisL | SpaceBass: http://pastebin.ca/8081 |
| 23:12.29 | AgiNamu | so the individual makers of phones and ATAs need to package it up[ |
| 23:12.38 | AgiNamu | I got the source, so i can compile it, so long you tell me what device it is |
| 23:12.46 | AgiNamu | like PA168Q, or V or R or whatever. |
| 23:12.55 | bjohnson | jesster: maybe call witing settings? |
| 23:12.57 | AgiNamu | course, i wont be responsible for bricking your ATA |
| 23:13.02 | elriah | Thanks, I see sample.call. |
| 23:13.06 | AgiNamu | so you might just wanna wait for Atcom to put it up :) |
| 23:13.10 | SpaceBass | not sure I know, all it says is AG-168 made in china |
| 23:13.11 | SpaceBass | :) |
| 23:13.38 | SpaceBass | is there a 1.41 or something that would at least be better than 1.36 in the interem? |
| 23:13.53 | *** part/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 23:13.54 | jesster | bjohnson: the analog phone does not ring if that were the case, and asterisk gets a busy from the channel bank |
| 23:14.23 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 23:14.24 | bjohnson | Spacebar: if it does sip .. just set it up as a sip phone for now |
| 23:14.48 | SpaceBass | bjohnson trying, it doesnt seem to register |
| 23:14.59 | SpaceBass | bjohnson I'm perfectly fine with sip for a home setting like this |
| 23:15.10 | jontow | 5line patch to make SMDI work :))) plus a pair of simple shell scripts |
| 23:15.32 | ratapeluda | I'm using asterisk to make calls through a sip proxy but sound quality is very bad only one-way (outside) I'm using x-lite as a client. any suggestion? thxs! |
| 23:15.51 | DenisL | SpaceBass: Any ideas? |
| 23:16.07 | SpaceBass | bjohnson since yesterday- you might be interested to know- I made great strides :) |
| 23:16.25 | SpaceBass | DenisL it looks normal... do you have 8 cards? |
| 23:17.02 | DenisL | SpaceBass: I have two TDM400P boards, one with four fxo modules and one with three fxo modules and one fxs module... |
| 23:17.15 | SpaceBass | gotcha |
| 23:17.19 | AgiNamu | SpaceBass, AG I think means AtCom |
| 23:17.22 | ratapeluda | :( help me.. please |
| 23:17.22 | AgiNamu | so look for atcom PA168 |
| 23:17.30 | AgiNamu | google around |
| 23:17.33 | SpaceBass | AgiNamu thanks! |
| 23:17.46 | SpaceBass | AgiNamu what about local type: is that account or phone number or... ? |
| 23:18.46 | AgiNamu | local type is usually account |
| 23:18.50 | DenisL | SpaceBass: I just don't understand why any of my settings in that config file appear to be applying. |
| 23:19.06 | DenisL | SpaceBass: Or not being applied rather. |
| 23:19.25 | SpaceBass | DenisL I didn't see anything in your zapata.conf that looked different than the default |
| 23:19.29 | SpaceBass | send the right one? |
| 23:19.35 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:20.06 | Shido6 | boink |
| 23:20.08 | DenisL | SpaceBass: The very bottom channel (channel 7) I've specified a different context. and an extension. |
| 23:21.43 | AgiNamu | denis, ztcfg -vv works ok? |
| 23:21.43 | SpaceBass | ahhh leme recheck |
| 23:21.43 | bile_one | DensiL you have to restart asterisk when you make changes to a ZAP device |
| 23:21.43 | DenisL | AgiNamu: Yes, that works ok. |
| 23:21.44 | DenisL | bile_one: I have done a gracefull restart as per SpaceBass' suggestion. |
| 23:21.44 | AgiNamu | just do "stop now" and then restart asterisk |
| 23:22.34 | *** part/#asterisk linenoise (~linenoise@cerberus.franklinamerican.com) |
| 23:22.40 | AgiNamu | i wonder if my uuid patch will ever get in |
| 23:22.48 | SpaceBass | DenisL at the CLI try zap show channels |
| 23:22.51 | AgiNamu | i hate maintaining a whole bunch of patches :P |
| 23:22.56 | SpaceBass | and see if shows the right context |
| 23:23.00 | *** join/#asterisk sung (~sung@fluorine.idge.net) |
| 23:23.14 | DenisL | AgiNamu: I've restarted that way. Same issue |
| 23:23.40 | DenisL | SpaceBass: I did zap show channels, and it shows everything in the from-internal Context... Hence my original question, why changes aren't showing up... |
| 23:23.56 | DenisL | I'm very new to this so could be a stupid syntax mistake I made somewhere... |
| 23:24.09 | AgiNamu | denis, start with asterisk -vvc |
| 23:24.17 | AgiNamu | and look for any bad linews |
| 23:24.19 | SpaceBass | like me spelling pstn ptsn for 2 days and not seeing the difference |
| 23:24.22 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 23:25.12 | SpaceBass | DenisL for one thing, you are setting the context to from-internal and thats already the context it is in |
| 23:25.24 | SpaceBass | so the settings are being- in effect- reset |
| 23:25.36 | SpaceBass | bjohnny feel free to correct my verbiage any time |
| 23:25.39 | AgiNamu | pastebin it and lets have alook |
| 23:25.47 | SpaceBass | http://pastebin.ca/8081 |
| 23:25.53 | AgiNamu | correct verbiage? yea, tell verbiage to reply to peoples' emails!!!! |
| 23:26.24 | SpaceBass | lol |
| 23:26.45 | debaser | i take it their phone never appeared? |
| 23:27.29 | SpaceBass | AgiNamu cross your fingers... updating to 1.41xxxx from atcom |
| 23:27.46 | AgiNamu | debaser: their USB phone? |
| 23:27.50 | AgiNamu | I havee 50 of those. |
| 23:28.03 | AgiNamu | But paid a LOT less than the $130 AUD they wanted!!! |
| 23:28.18 | jontow | http://mno.bsd.st/~jontow/smdi-0a.tgz |
| 23:28.24 | jontow | thats the current status :) |
| 23:28.28 | SpaceBass | AgiNamu still curious what you do all over the world.... |
| 23:28.39 | AgiNamu | im creating a "voip in a box" system |
| 23:28.47 | AgiNamu | i.e., you want to be SpaceBass VoIP Inc |
| 23:28.54 | AgiNamu | you give us a bit of money, and we put you in business. |
| 23:29.00 | *** join/#asterisk PTG1234 (PTG123@66.213.239.122) |
| 23:29.00 | SpaceBass | ahhh |
| 23:29.02 | AgiNamu | one of the issues I ad to fix was SIP |
| 23:29.05 | AgiNamu | and the answer was to use IAX |
| 23:29.13 | PTG1234 | anyone here run tftpd on unix? |
| 23:29.14 | AgiNamu | cause SIP is a pos |
| 23:29.16 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 23:29.20 | PTG1234 | SIP IS AWESOME! :) |
| 23:29.20 | SpaceBass | lol... from what I am learning seems like a good choice |
| 23:29.26 | Exstatica | i'm having so many issues with realtime |
| 23:29.31 | AgiNamu | yae. it means writing firmware and stuff |
| 23:29.38 | AgiNamu | so it's "harder" than using SIP evices |
| 23:29.40 | AgiNamu | but its much more reliable. |
| 23:29.51 | AgiNamu | it also means having to scale asterisk rather than use SER |
| 23:29.59 | AgiNamu | \but it think having asterisk cluster is a better idea than SEr anyways. |
| 23:30.02 | SpaceBass | ultimatly realibality is a big selling point |
| 23:30.07 | AgiNamu | SEr seems like a cheap hack for asterisk shortcomings. |
| 23:30.10 | debaser | AgiNamu: not the usb phone, the lan phone |
| 23:30.20 | SpaceBass | SEr? |
| 23:30.22 | AgiNamu | debaser, screw their lan phone. nothing insteresting there. |
| 23:30.25 | AgiNamu | Sip Express router |
| 23:30.29 | SpaceBass | gotcha |
| 23:30.30 | PTG1234 | ok your provider most likely uses sip, by you using sip it means you can preserver the re-invite and connect directly to their provider, which means no latency.. why would you use iax? |
| 23:30.37 | PTG1234 | sip ->iax -> iax makes less sense then |
| 23:30.40 | PTG1234 | sip -> sip -> sip |
| 23:30.41 | SpaceBass | speaking of phones, anyone ever used an i.picasso? |
| 23:30.56 | AgiNamu | Asterisk has some ... not amazing ... code that doesnt go fast with 1000+ users |
| 23:31.02 | AgiNamu | so having 15,000 registered to a single machine doesnt work |
| 23:31.10 | AgiNamu | hell, with the default code, asterisk can barely LOAD 15,000 users. |
| 23:31.18 | debaser | ser and * have different intended uses. |
| 23:31.19 | bile_one | Good night all. |
| 23:31.33 | AgiNamu | PTG, actually, i've got SIP, H323, or IAX2 connects. |
| 23:31.39 | AgiNamu | BUT, almost all my clients are NAT'd. |
| 23:31.45 | AgiNamu | so there's no transfers gonna happen anyways |
| 23:31.51 | _Vile | SS7 Question for anyone who can respond: Are Intermachine (LIS, IMTs) trunks interlinked with A links to the LEC or could I use SIP-7 to handle ISUP and get by without having to do A links to the LEC? |
| 23:32.13 | AgiNamu | in fact, some clients have special jitter needs, so I'll jitterbuffer for them |
| 23:32.24 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ |
| 23:32.30 | AgiNamu | since im colo'd with my provider, im in the same switch |
| 23:32.34 | AgiNamu | 0.7ms latency |
| 23:32.38 | *** part/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 23:32.45 | AgiNamu | that way, I can controll the jitterbuffer completely. |
| 23:32.46 | Druken | AgiNamu: nice.. hehe |
| 23:32.55 | AgiNamu | (talking about sometimes 50ms or more of jitter, 10% packet loss) |
| 23:33.26 | Druken | 10% packet is loss horrible.... you have customers who get that ? |
| 23:33.40 | PTG1234 | agi: transfer happens fine with NAT, i use it now |
| 23:33.42 | AgiNamu | yep |
| 23:33.51 | AgiNamu | PTG wow. I can't even get SIP to work with NAT |
| 23:33.57 | PTG1234 | agi: what type of firewall? |
| 23:34.01 | AgiNamu | some of my clients are double and triple NAT'd too. |
| 23:34.07 | PTG1234 | agi: i do around 10 phones inside one nat firewall, and have no issues |
| 23:34.07 | AgiNamu | firewall/router? Idiotic dumbass ISP |
| 23:34.14 | PTG1234 | agi: double nat works fine too |
| 23:34.14 | terrapen | good lord |
| 23:34.16 | AgiNamu | if you control the firewall, that's great. |
| 23:34.20 | SpaceBass | i can't get sip to work nat either |
| 23:34.21 | terrapen | romania has the most annoying phone-ringing sound |
| 23:34.28 | AgiNamu | but when you have buckfuttingly stupd ISPs who actually HATE SIP |
| 23:34.31 | PTG1234 | agi: well the problem is outdated firewalls tend to cause issues |
| 23:34.32 | AgiNamu | it's different. |
| 23:34.36 | AgiNamu | exactly. |
| 23:34.41 | PTG1234 | agi: i am in a new clients right now, having that issue |
| 23:34.47 | AgiNamu | so when I say it's not reliable, i mean that i never know if someone will have an issue. |
| 23:34.50 | terrapen | aginamu, are you a voip provider in .gt? |
| 23:34.50 | SpaceBass | getting a POS wifi sip phone that doesnt support WPA, so I had to set up a WEP access point on the other side of my firewall |
| 23:34.51 | AgiNamu | with IAX, it just doesnt happen. |
| 23:34.51 | PTG1234 | agi: polycoms work fine, softphones work fine, ciscos won't work |
| 23:34.56 | PTG1234 | agi: so i am making them replace firewall |
| 23:34.56 | AgiNamu | terrapen, in GT, all over. |
| 23:34.59 | terrapen | nice |
| 23:34.59 | SpaceBass | tried passing only 5060 and setting up nat... no dice |
| 23:35.02 | terrapen | what company? |
| 23:35.05 | _Vile | I think it may be illegal for ISPs to block SIP ports these days, I remember something about that |
| 23:35.05 | AgiNamu | Telefinity |
| 23:35.08 | AgiNamu | (shit site up) |
| 23:35.11 | AgiNamu | Vile, not in GT |
| 23:35.16 | Nugget | I want to post to the asterisk-users list, but I can't decide if I want to ask what the best free softphone is or if I want to explain why I chose postgresql over mysql. what do you guys think? |
| 23:35.18 | _Vile | in US |
| 23:35.21 | AgiNamu | and in Costa Rica, they're trying to make VoIP *illegal* |
| 23:35.27 | fgravato | still need RTP Ports |
| 23:35.28 | fgravato | open |
| 23:35.29 | fgravato | also |
| 23:35.30 | AgiNamu | yea, in US, ti's a completely different market. ULAW, SIP, no problem! |
| 23:35.32 | _Vile | state owned telco? |
| 23:35.32 | fgravato | not just Sip |
| 23:35.47 | SpaceBass | fgravato any idea on which ones or range? |
| 23:35.48 | AgiNamu | fgravato, not ports opened. no NAt issues with IAX. single socket. |
| 23:35.49 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 23:36.03 | AgiNamu | oh oh you were talkin to spacebass :) |
| 23:36.08 | SpaceBass | :) |
| 23:36.10 | AgiNamu | spacebass, whatever you set for RTP |
| 23:36.16 | AgiNamu | 15000 or 50000 or something liek that. |
| 23:36.20 | AgiNamu | vile, was a state own |
| 23:36.43 | AgiNamu | now its the ex-state owned + Telefonica + 1 non-Telco ISP |
| 23:36.44 | *** join/#asterisk oo (~marko@marko.horde) |
| 23:36.49 | SpaceBass | AgiNamu how do I know what its using? I have a stop and start |
| 23:37.28 | SpaceBass | i can open that range on my firewall no problem, but the pa168 wants a specific port |
| 23:38.04 | bjohnson | * rtp os set to 10000 to 20000 by default |
| 23:38.12 | bjohnson | it is set in one the .conf files |
| 23:38.21 | file[laptop] | rtp.conf ironically |
| 23:38.28 | SpaceBass | that's what I'm seein... a range of 10000 to 20000 |
| 23:38.37 | SpaceBass | pretty broad range |
| 23:39.09 | fgravato | you can narrow it down i think |
| 23:39.21 | fgravato | in the spa config |
| 23:39.31 | fgravato | you can adjust the range to match what's on * |
| 23:39.45 | AgiNamu | spacebass, that's what RTP port is. |
| 23:39.48 | AgiNamu | on the ATA settings. |
| 23:39.50 | AgiNamu | I think. |
| 23:39.54 | AgiNamu | i dont use sip :P |
| 23:40.03 | SpaceBass | AgiNamu you think its the range? |
| 23:41.11 | *** join/#asterisk sbarrius (~sbarrius@c-24-15-201-23.client.comcast.net) |
| 23:41.31 | sbarrius | whats up guys... |
| 23:41.33 | AgiNamu | i dunno.. never used those things as SIP devices. |
| 23:41.53 | SpaceBass | :) |
| 23:42.10 | sbarrius | any you guys use broadvoice, any one happy with them? |
| 23:42.41 | SpaceBass | sbarrius I have their byod plan for like $5.00/month... been using it to do "follow-me" to my cell and for incoming calls from Washington DC |
| 23:42.44 | SpaceBass | so far, so good |
| 23:42.58 | SpaceBass | but not sure I'd want them as my primary line based on what I've heard |
| 23:43.35 | sbarrius | yeah...I added an 800 number and it knocked out my primary |
| 23:44.00 | sbarrius | now my phones rings when you dial the 800 number, and when you dial the primary nothing |
| 23:44.39 | sbarrius | but when the 800 number rings asterisks try to send to sip phone and the logs say it trying to bridge but nothing happens |
| 23:44.56 | sbarrius | SpaceBass - who do you use? |
| 23:45.02 | SpaceBass | for? |
| 23:45.12 | sbarrius | primary carrier |
| 23:45.15 | SpaceBass | local pots |
| 23:45.20 | sbarrius | :) |
| 23:45.33 | SpaceBass | have two fxo's for now |
| 23:45.37 | sbarrius | do you use the # feature for follow me? |
| 23:46.03 | SpaceBass | not yet, just have a IVR extension... press 1 for voice mail, press 2 for my cell |
| 23:46.06 | SpaceBass | not really follow me |
| 23:47.14 | sbarrius | Im doing follow me... the only thing that sucks is if it trys my cell and I dont pickup me cell voice mail takes it, I want it to ring until I hit # |
| 23:47.32 | sbarrius | if I dont hit # then the asterisk vm takes it, slick huh? |
| 23:48.04 | SpaceBass | yeah |
| 23:48.24 | SpaceBass | ok... how the F can I tell what RTP port * is using from that 10000 to 20000 range? |
| 23:48.35 | SpaceBass | short of breaking out a sniffer |
| 23:48.38 | AgiNamu | spacebass, in the sip.conf, you can narrow it down. |
| 23:48.40 | sbarrius | but you can only do the # feature with zaptel device... I think |
| 23:48.45 | AgiNamu | and a sniffer is like, essential, for SIP :) |
| 23:49.02 | SpaceBass | AgiNamu ethereal here I come |
| 23:49.24 | sbarrius | is that your outgoing port range? |
| 23:49.42 | AgiNamu | just wait till you try IAX |
| 23:49.46 | AgiNamu | it'll just work :) |
| 23:49.47 | SpaceBass | AgiNamu as simple as rtp=15000 ? |
| 23:49.47 | gambolputty | Is SRTP planned for * anytime soon? |
| 23:49.53 | SpaceBass | AgiNamu I'm looking forward to it, believe me |
| 23:51.02 | sbarrius | AgiNamu is that the outbound port range? |
| 23:52.33 | SpaceBass | looks like I'm going to have to wait for IAX |
| 23:52.36 | SpaceBass | sip ain't happening |
| 23:52.45 | sbarrius | does anyone else have any recomendations for a carrier |
| 23:53.55 | sbarrius | another bitch about broadvoice... no 24 hour support |
| 23:54.07 | fgravato | sbarius = nufone |
| 23:55.00 | sbarrius | do you like them fgravato? |
| 23:55.12 | Exstatica | i read about sip caching, where it can cache the sip users, but i can't figure out where to add that |
| 23:55.30 | AgiNamu | ex - what do you mean by cache? |
| 23:55.49 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 23:56.00 | fgravato | nufone is great |
| 23:56.03 | fgravato | 0 problems |
| 23:56.09 | fgravato | support is tought at times but |
| 23:56.11 | bjohnson | and isn't accepting new accounts |
| 23:56.27 | bjohnson | sbarrius: get a per minute provider to use as backup |
| 23:56.30 | Exstatica | NOTE: As of CVS-HEAD 3/16/05, if you enable RealTime caching in your sip.conf, Voicemail MWI works and so does 'sip show peers'. |
| 23:56.35 | sbarrius | why aren't they accepting new accounts |
| 23:56.43 | bjohnson | like nufone, voipjet, livevoip, teliax, or a hundred others |
| 23:57.09 | bjohnson | sbarrius: don't know .. I guess they have enough for their system right now |
| 23:57.13 | TomL | RealTime caching? what's that? |
| 23:57.17 | AgiNamu | ex, oh, realtime chaching. |
| 23:57.17 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.we.client2.attbi.com) |
| 23:57.17 | sbarrius | bjhonson which would you use? |
| 23:57.24 | AgiNamu | thats for DB caching. |
| 23:57.27 | Exstatica | yeah |
| 23:57.30 | TomL | blah |
| 23:57.30 | bjohnson | all of them |
| 23:57.40 | Exstatica | what is the command command for it? |
| 23:57.49 | AgiNamu | no clue. i dont touch realtime. |
| 23:57.52 | AgiNamu | im writing my own. |
| 23:57.52 | bjohnson | throw down $10 or $30 on a few and try them out |
| 23:58.02 | sbarrius | I need a reliable incoming carrier for my biz |
| 23:58.11 | bjohnson | oh .. incoming |
| 23:58.16 | AgiNamu | sbarrius, but you dont want to commit? |
| 23:58.40 | Supaplex | AgiNamu: only if you have a ring ;) |
| 23:58.46 | sbarrius | i already pay for broadvoice biz package |
| 23:58.48 | AgiNamu | hHA |
| 23:58.59 | AgiNamu | oh, youre just looking for a coupla lines |
| 23:59.22 | sbarrius | im not happy with broadvoice right now |
| 23:59.44 | sbarrius | dont really want to go pots... |
| 23:59.54 | Supaplex | pots is the pits? |