| 00:00.13 | x86 | generalhan: sip.conf, same as where bindaddr is set ;) |
| 00:00.21 | generalhan | x86: see i had it set to 0.0.0.0 but i changed it thinking that was the issue, but it wasnt |
| 00:00.32 | x86 | JT: it can't resolve the destination hostname |
| 00:00.37 | x86 | sopo2k4: check DNS |
| 00:01.06 | x86 | generalhan: yeah leave it as 0.0.0.0, and make your realms the hostname of the machine (which should definitely be different) |
| 00:01.15 | *** join/#asterisk pejo_ (n=pete-joh@triton.dsv.su.se) |
| 00:01.17 | generalhan | x86: LOL, ok could it be that im not setting domain/realm on either server ? i never had to set that before |
| 00:01.39 | x86 | well by default i think it uses your server's hostname |
| 00:01.43 | x86 | are they the same hostname? |
| 00:01.46 | generalhan | no |
| 00:01.51 | x86 | then that should be good |
| 00:02.01 | generalhan | well wth then ?? this is slightly aggrivating |
| 00:02.09 | jarrod | does asterisk have a way of specifying the source ip address of secondary ips on nics for responses? |
| 00:02.58 | x86 | jarrod: your question is unclear |
| 00:03.20 | jarrod | i want the sip process to listen and respond on a secondary (alias) ip address |
| 00:03.31 | jarrod | is that possible |
| 00:03.38 | JT | it will probably listen if bound to all |
| 00:03.53 | JT | probably will only respond on that ip if requests come in on that ip |
| 00:04.38 | jarrod | but, given the connectionless nature of udp, it will respond with a source of the secondary (alias) ip? |
| 00:05.14 | JT | that would be logical |
| 00:05.23 | JT | if the request came in on that ip |
| 00:05.27 | JT | but feel free to test |
| 00:05.40 | x86 | jarrod: you can set bindaddr to only the secondary IP, the primary IP wont be bound at all |
| 00:08.09 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:21.09 | jarrod | that doesnt provide a stable solution, because the virtual ip needs to be able to be migrated from one server to another, without hitch, for this to work as planned |
| 00:21.33 | jarrod | if you start a server on an ip that does not exist it pseudo binds to 'all', then once that ip is assigned it stops listening on whatever port |
| 00:22.33 | JT | asterisk isn't a ha solution, you will need to work out how to do the ha bit |
| 00:22.45 | jarrod | should be :( |
| 00:22.56 | jarrod | i guess if it executed the init script, AFTER the IP was assigned |
| 00:23.02 | jarrod | and stopped it when it was removed, it would work |
| 00:25.49 | fujin | Use heartbeat. |
| 00:25.53 | fujin | Don't be retarded ;) |
| 00:26.08 | fujin | Heartbeat v1 is perfect for asterisk-ha. has been working perfectly here |
| 00:27.14 | fujin | jarrod: it's quite simple to change a virtual IP from one server to another |
| 00:27.21 | Hmmhesays | any electronic freaks in here? |
| 00:27.26 | fujin | just spoof an ARP, this will tell the switches that the port that the IP is on has relocated |
| 00:27.50 | Hmmhesays | say I have an 18V psu at 1 amp, to drop it to 14v I would need a 4ohm resistor right? |
| 00:27.51 | JT | what is an electronic freak? |
| 00:27.56 | JT | like a terminator cyborg |
| 00:28.04 | fujin | I'm not sure this is the correct place to ask |
| 00:28.12 | JT | Hmmhesays: how much current would you draw? |
| 00:28.30 | fujin | lol no, a 4ohm resistor won't drop 18v to 14v |
| 00:28.42 | fujin | resistance != voltage |
| 00:28.44 | snuff-work | V=IR |
| 00:28.53 | fujin | point |
| 00:29.23 | Hmmhesays | yes V=IR, at 1amp a 1ohm resistor will drop 1volt across it |
| 00:29.49 | snuff-work | well remember u have a 18W sorce |
| 00:29.59 | snuff-work | P = IV |
| 00:30.00 | jarrod | i am using heartbeat fujin, and it is moving the ip |
| 00:30.13 | Hmmhesays | where did the 18W source come from? |
| 00:30.16 | jarrod | fujin: the problem is asterisk responding on the new virtual ip assigned to it |
| 00:30.26 | snuff-work | 18V in 1 amp.. so.. 18x1 = 18W |
| 00:30.59 | snuff-work | which means u'd want a 5W resistor |
| 00:31.15 | snuff-work | which are generally ceramic if i remember rightly |
| 00:31.15 | fujin | jarrod: not really a problem at all |
| 00:31.23 | fujin | just configure asterisk to *always* listen on the virtual IP |
| 00:31.40 | Hmmhesays | where did you come up with 5W |
| 00:31.41 | jarrod | i do that, but when the ip is assigned after asterisk is started |
| 00:31.48 | fujin | you're doing it wrong ;) |
| 00:31.54 | jarrod | oh, how should i :-D |
| 00:31.58 | jarrod | im using ha2 |
| 00:32.03 | fujin | That's probably half of the problem |
| 00:32.13 | fujin | ha2 is overly complicated for an asterisk setup, imho |
| 00:32.23 | fujin | I'm using ha1, I can provide my haresources/ha.cf if you need. |
| 00:32.25 | jarrod | i like xml :-D |
| 00:32.28 | jarrod | please do |
| 00:32.33 | fujin | Hold ;) |
| 00:32.37 | jarrod | and any script it uses? |
| 00:32.39 | jarrod | i guess ill downgrade |
| 00:33.13 | snuff-work | Hmmhesays.. if its going from 18V in 1 amp.. to 14V in 1 amp.. means u have a difference of 4 W in power to dissapate |
| 00:33.40 | snuff-work | therefore u'll probably want to be on the safe side and use a 5W resistor |
| 00:33.42 | fujin | ha.cf -> http://rafb.net/p/28BKc444.html |
| 00:34.13 | fujin | haresources -> http://rafb.net/p/FGHGRm97.html |
| 00:34.20 | fujin | just put those files in /etc/ha.d on both servers |
| 00:34.40 | fujin | obviously mine are named asterisk01/02, the management ip's are .2 and .3, the virtual ip is .1 |
| 00:35.11 | jarrod | so the asterisk is haresources is /etc/init.d/asterisk startup? |
| 00:35.20 | fujin | correct |
| 00:35.24 | jarrod | dude |
| 00:35.26 | jarrod | that looks so much easier |
| 00:35.26 | Hmmhesays | snuff-work: then were do ohms come into that equation |
| 00:35.28 | jarrod | what is up with ha2 |
| 00:35.29 | fujin | asterisk, atftpd proxyman etc are all starting when it detects the fail |
| 00:35.30 | *** join/#asterisk MrMister2 (n=mrmister@89-180-74-85.net.novis.pt) |
| 00:35.33 | fujin | ha2 is *overly* complex |
| 00:35.43 | fujin | more suited for clusters than 2box hot/cold or hot/hot. |
| 00:35.46 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 00:35.53 | Hmmhesays | cause every resistor i'm looking at have an ohm/watt rating |
| 00:36.00 | jarrod | that is somewhat upsetting |
| 00:36.01 | jarrod | heh |
| 00:36.11 | VJFROMGT | i notice that the extension.conf file refer to trunk by a number, which file defines what trunk is what number? |
| 00:36.12 | fujin | you need all that freaky XML to get ha2 going, lol, I couldn't be bothered. |
| 00:36.31 | fujin | don't get me wrong, I love it too |
| 00:36.32 | JT | Hmmhesays: using resistors to drop the voltage for any real amount of current is dumb |
| 00:36.33 | jarrod | i use it everywhere |
| 00:36.48 | JT | too much heat, the resistors get big and expensive fast |
| 00:36.55 | fujin | capacitattooorrr |
| 00:37.03 | jarrod | i guess my assumption that ha2 could do everything ha1 could do, but better, was WRONG :) |
| 00:37.03 | JT | fujin: ? |
| 00:37.18 | fujin | lol |
| 00:37.22 | fujin | just throwin a spanner in the works |
| 00:37.23 | fujin | my bad ;) |
| 00:37.27 | jarrod | BRB |
| 00:37.33 | jarrod | i gotta uinstall ha2 |
| 00:37.34 | jarrod | UGGHH |
| 00:37.40 | fujin | lol |
| 00:37.43 | Hmmhesays | I don't think 18v to 14v is much |
| 00:37.51 | fujin | I should probably document my findings for heartbeating asterisk |
| 00:38.18 | JT | Hmmhesays: the voltage is not as important as how much current you're pulling |
| 00:38.36 | JT | Hmmhesays: it's a stupid inefficient unregulated way to do it, anyway |
| 00:38.39 | JT | Hmmhesays: what's it for? |
| 00:38.59 | Hmmhesays | well I can't find a 14v power supply for my wireless ear monitor base station |
| 00:39.09 | Hmmhesays | closest I can find is 15v |
| 00:39.12 | Hmmhesays | at 500ma |
| 00:39.14 | JT | wireless ear monitor? |
| 00:39.26 | JT | do you have a url |
| 00:39.37 | Hmmhesays | http://www.musiciansfriend.com/product/Nady-PEM500-UHF-Personal-Ear-Monitor-System?sku=277129 |
| 00:40.06 | JT | Hmmhesays: the transmitter? |
| 00:40.09 | Hmmhesays | yes |
| 00:40.19 | JT | did it not come with a power supply? |
| 00:40.20 | Hmmhesays | the original psu was 14v @500ma |
| 00:40.24 | Hmmhesays | its been long lost |
| 00:40.30 | JT | then stick 15v in |
| 00:40.34 | JT | buy the 15v unit |
| 00:40.44 | JT | oh no, not a volt of difference |
| 00:40.50 | Hmmhesays | is that not going to make a difference? |
| 00:40.55 | JT | nope |
| 00:41.02 | Hmmhesays | can you explain to me why? |
| 00:41.08 | *** join/#asterisk atomicd (n=atomicda@74-206-0-80.static-ip.m.telepacific.net) |
| 00:41.13 | JT | it would have internal voltage regulators |
| 00:41.25 | JT | especially since that 14v supply was probably a pile of cheap junk |
| 00:41.25 | Hmmhesays | it is kind of a cheap transmitter |
| 00:41.36 | JT | voltage regulators are kind of cheap |
| 00:41.49 | Hmmhesays | i see |
| 00:41.51 | JT | cheaper than warranty claims |
| 00:42.00 | JT | it's just good electronics design practice |
| 00:42.49 | *** join/#asterisk jarrod (i=anon@theos.org) |
| 00:43.06 | JT | it can probably handle at least 5v overvoltage |
| 00:43.09 | JT | maybe more |
| 00:43.15 | JT | 1v is nothing |
| 00:43.23 | *** join/#asterisk minkus (n=minkus@pool-71-182-32-236.clrkwv.east.verizon.net) |
| 00:43.37 | atomicd | Quick question... does Asterisk have an audible "on hold" reminder? |
| 00:43.47 | JT | mind you, if the supply is unregulated, at no load a 15v supply is probably more like 17v |
| 00:44.06 | jarrod | fujin |
| 00:44.14 | jarrod | what is this: ping 192.168.108.210 192.168.108.254 |
| 00:46.41 | Nugget | What do you mean "what is this?" |
| 00:49.29 | fujin | jarrod: that tells it to ping the AS5400 and the 2600 to see if the connectivity is up |
| 00:49.36 | fujin | just more connectivity tests |
| 00:49.43 | fujin | the two boxes are connected by a crossover cable, I forgot to mention |
| 00:49.52 | jarrod | hmm, the processes are started, but its not assigning the ips to the interface on the primary, nor did it start * |
| 00:50.18 | killfill | Is it possible to make agents recieve more than 1 call at a time in a queue? my agents cannot.. :S |
| 00:50.23 | [ViAjErO] | Is there a way that some analog port in a TDM22B card from digium doesn't give tone upon pick up the line ? some power or misconfiguration issue ? |
| 00:51.12 | fujin | jarrod: have you got your authkeys setup? |
| 00:51.31 | jarrod | yes, they both match |
| 00:51.39 | jarrod | auth 1, 1 sha blah |
| 00:51.42 | jarrod | on both |
| 00:51.51 | fujin | and those ha.cf haresources are exactly the same on both boxes? |
| 00:51.58 | jarrod | identical |
| 00:52.01 | fujin | k |
| 00:52.29 | fujin | do asterisk stop on both |
| 00:52.30 | DrukenHME | the search foobared on voip-info ? |
| 00:52.37 | fujin | and then /usr/lib/heartbeat/hb_takeover all |
| 00:52.44 | jarrod | oh, i had the processes stopped on both when i started? |
| 00:52.59 | jarrod | (asterisk processes) |
| 00:53.05 | jarrod | i figured it would manage those for me |
| 00:53.57 | fujin | generally I stop them all |
| 00:54.26 | fujin | yeah, if you do asterisk stop |
| 00:54.34 | fujin | and then start up the heartbeat init script |
| 00:54.40 | fujin | it should check in a couple of secs and start up your primary |
| 00:54.47 | fujin | check the syslog |
| 00:54.55 | jarrod | http://ipeng.net/pastebin/16/ |
| 00:55.35 | jarrod | how long does it generally take for it to bind the IP |
| 00:55.35 | jarrod | ? |
| 00:55.36 | fujin | yep, ok |
| 00:55.41 | fujin | now do /usr/lib/heartbeat/hb_takeover all |
| 00:55.44 | fujin | to make it speed up |
| 00:55.53 | fujin | you should see it talk to the other one |
| 00:55.58 | jarrod | ok, i did it |
| 00:56.03 | jarrod | where? |
| 00:56.15 | fujin | http://rafb.net/p/Rqji3h37.html |
| 00:56.17 | fujin | in your syslog ;) |
| 00:56.19 | fujin | like that ^^ |
| 00:56.41 | jarrod | maybe it just takes a minute? |
| 00:57.02 | jarrod | Jul 23 19:54:13 ss1a heartbeat[8482]: info: pid 8482 locked in memory. |
| 00:57.08 | jarrod | thats the last entry i have |
| 00:57.08 | fujin | yes |
| 00:57.12 | fujin | did you run hb_takeover all like I said? |
| 00:57.17 | jarrod | sure idd |
| 00:57.18 | jarrod | did |
| 00:57.25 | jarrod | ill try again |
| 00:57.30 | fujin | strange |
| 00:57.53 | jarrod | it says ping heartbeat started, wonder what the problem could be |
| 00:58.03 | fujin | doesn't really look like a problem |
| 00:58.05 | fujin | just chill for a sec |
| 00:58.08 | jarrod | i wonder if there are any ha2 remnants throwing me off |
| 00:58.09 | fujin | see if it starts up |
| 00:58.16 | jarrod | ha2 took ~2min |
| 00:58.37 | fujin | did you remove all of ha2? |
| 00:58.38 | jarrod | i see it listening on the udp ports |
| 00:58.40 | jarrod | yea |
| 00:58.47 | jarrod | then i install ha over it |
| 00:58.53 | fujin | and you've got your eth1 hooked up with crossover to the other box? |
| 00:59.19 | fujin | and your boxes are asterisk01/asterisk02 ? |
| 00:59.22 | jarrod | well, i use eth2, and they are both plugged into switches on the same broadcast domain |
| 00:59.25 | jarrod | ss1a/ss1b |
| 00:59.30 | fujin | well |
| 00:59.32 | jarrod | i modified the cf accordingly |
| 00:59.33 | fujin | modify the 'node' lines |
| 00:59.35 | fujin | k |
| 00:59.41 | fujin | and you changed bcast to eth2 |
| 00:59.42 | fujin | ? |
| 00:59.59 | jarrod | yup yup, and both nodes resolve properly on each |
| 01:00.12 | jarrod | respawn hacluster |
| 01:00.15 | jarrod | hmm |
| 01:00.23 | fujin | I'm not sure if it works without a crossover or serial cable |
| 01:00.36 | jarrod | well its based on ip connectivity |
| 01:00.37 | fujin | have never done it that way ;[ |
| 01:00.52 | jarrod | i have a cross over between them for DRBD |
| 01:00.53 | jarrod | but |
| 01:01.16 | jarrod | this way protects me in case switches die, they are both plugged into separate switches, but can ping each other |
| 01:01.29 | jarrod | hmm |
| 01:01.32 | jarrod | maybe i can enable debug |
| 01:01.46 | fujin | you do have heartbeat started on both boxes aye? |
| 01:01.51 | fujin | looks like they aren't failing correctly |
| 01:02.08 | fujin | do /usr/lib/heartbeat/hb_standby on your primary |
| 01:02.09 | jarrod | yes, and i can see them both pinging |
| 01:02.21 | jarrod | [8:01pm][root@ss1a:/etc/ha.d]# /usr/lib/heartbeat/hb_standby |
| 01:02.21 | jarrod | 2007/07/23_20:02:15 Going standby [all]. |
| 01:02.25 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-a0d77766a3cca928) |
| 01:02.56 | jarrod | im going to stop it all and restart |
| 01:04.39 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:04.47 | fujin | if you did standby |
| 01:04.50 | fujin | what did the second ones syslog say? |
| 01:05.21 | jarrod | im not seeing anything past the locked in memory |
| 01:05.32 | jarrod | ps auwx shows the heartbeat processing pinging each other, and my gateway |
| 01:05.38 | jarrod | on both boxes |
| 01:05.40 | jarrod | here, ill restart |
| 01:06.56 | jarrod | on the:ok i enabled debu |
| 01:07.31 | jarrod | it assigns the virtual ip to the interface specified in bcast? |
| 01:07.38 | fujin | no |
| 01:07.43 | fujin | it aliases the primary interface I think |
| 01:07.52 | jarrod | how does it know which is primary |
| 01:09.00 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 01:09.24 | fujin | not sure |
| 01:09.43 | fujin | probably from haresources |
| 01:10.00 | jarrod | hmm |
| 01:10.17 | jarrod | does the host need to be the same as # hostname ? |
| 01:12.43 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 01:15.16 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:17.09 | fujin | sorry? |
| 01:17.13 | fujin | what do you mean |
| 01:17.19 | jarrod | nothing, answered my own question |
| 01:17.56 | jarrod | dang man |
| 01:17.58 | jarrod | im clueless |
| 01:18.12 | jarrod | i did hb_takeover all |
| 01:18.17 | jarrod | and it gave me a few things |
| 01:18.34 | jarrod | i wonder how it decides which interface gets the alias IP |
| 01:18.45 | tzafrir_laptop | ZX81, what did you ask just before you left? |
| 01:24.32 | jarrod | fujin! |
| 01:24.34 | jarrod | it worked! |
| 01:25.26 | jarrod | hmm, standby didnt transfer |
| 01:26.04 | [ViAjErO] | shared interrupts could cause strange behavior on a digium tdm22b card ? |
| 01:26.22 | ZX81 | tzafrir_laptop yesterday? |
| 01:26.25 | [ViAjErO] | <PROTECTED> |
| 01:26.30 | ZX81 | about codec 126 |
| 01:26.34 | saftsack | ooh |
| 01:26.53 | saftsack | not good. try to take another pci slot for your fxo/fxs card |
| 01:27.03 | ZX81 | I was seeing it when polarity reversed on a hangup - had a really strange line (analogue line but provides DDIs) |
| 01:27.08 | Juggie | better yet, disable onboard usb |
| 01:27.09 | [ViAjErO] | saftsack: that answer is for me ? |
| 01:27.12 | saftsack | yes |
| 01:27.17 | [ViAjErO] | ok thank you |
| 01:27.29 | Juggie | [ViAjErO], disable anything you arnt using, like, onboard usb. |
| 01:27.31 | ZX81 | had to be connected to a FXS socket instead of an FXO -> both sides providing dialtone to each other! |
| 01:27.52 | saftsack | Juggie: i think he needs etherneed so i think he has to change the slot |
| 01:27.56 | [ViAjErO] | saftsack and Juggie : could I use a better NIC (i have a sis900), to avoid sharing IRQ's on it ? |
| 01:28.08 | saftsack | no. dont you have a free pci slot? |
| 01:28.33 | [ViAjErO] | saftsack: yes .. i have another .. |
| 01:28.41 | Juggie | saftsack, i'm aware, but every bit helps :) |
| 01:28.44 | [ViAjErO] | i'll try this and disabling usb |
| 01:28.51 | [ViAjErO] | thank you ... |
| 01:28.53 | saftsack | so take this .. in most cases the third one is "alone" |
| 01:29.23 | saftsack | Juggie: thats true ;) |
| 01:29.28 | [ViAjErO] | safstack : i have one of mi analog port without tone upon pickin up .. the another is fine |
| 01:29.45 | [ViAjErO] | i'll try this |
| 01:29.52 | [ViAjErO] | thank you a lot guys |
| 01:29.56 | saftsack | wait ... |
| 01:30.18 | saftsack | do you get no tone if you enable a second one, or just on this specific port? |
| 01:30.27 | saftsack | enable => pick up |
| 01:30.29 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 01:31.36 | [ViAjErO] | saftsack: i have 1 and 2 ports as analog internal .. 3 and 4 as incoming . If I put a phone on port 2 i get tone and I'm able to dial everywhere, but in port one I didnt get tone when I pick up the phone |
| 01:32.42 | [ViAjErO] | saftsack: is a TDM22B 2 fxs & 2 fxo |
| 01:32.43 | saftsack | ah ok. so furthermore to the portchange you can try to remove module on port 2 and see if there is tone then on port one |
| 01:32.59 | sopo2k4 | whats the default SIP Port |
| 01:33.02 | sopo2k4 | to add to my router? |
| 01:33.12 | saftsack | [ViAjErO]: i know ;) no other combination possible |
| 01:33.15 | *** join/#asterisk az^^za (n=azza@202.183.121.172) |
| 01:33.33 | saftsack | after this test swap both modules |
| 01:33.41 | saftsack | sopo2k4: voip-info.org |
| 01:33.48 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 01:33.54 | Hmmhesays | well I got a 15v 1000ma psu |
| 01:33.58 | [ViAjErO] | saftsack: I see .. i'm a newbie ... but i'm trying to get this working ... |
| 01:34.01 | Hmmhesays | hopefully it won't smoke my transmitter |
| 01:34.05 | saftsack | good luck :) |
| 01:34.08 | [ViAjErO] | ;) |
| 01:34.11 | JT | ZX81: sounds like a good way to blow something up |
| 01:34.31 | [ViAjErO] | i'll be back ... thank you saftsack & Juggie |
| 01:34.35 | saftsack | kk |
| 01:35.05 | [hC] | anyone seen where sending 'h' as an option to Dial (to not send caller id) blocks the number but NOT the name? |
| 01:35.08 | sopo2k4 | is there a default SIP Port? |
| 01:35.09 | *** join/#asterisk NirS_ (n=Nir@84.94.120.181.cable.012.net.il) |
| 01:35.21 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 01:35.30 | az^^za | howdy all, complete noob here... trying to decide on a corporate PBX. Will Asterisk interact with a remote Cisco Call Manager, or Alcatel or similar PBX? |
| 01:35.38 | skymeyer | sopo2k4: 5060 |
| 01:36.08 | sopo2k4 | cheers |
| 01:36.14 | [hC] | az^^za: it can in some capacity or another, yes. |
| 01:36.25 | saftsack | sopo2k4: plz look at voip-info.org |
| 01:36.29 | saftsack | this is a everyasked question |
| 01:36.34 | [hC] | az^^za: wether it be using SIP (voip) or connecting the two with a 'fake' PRI, but youll find everything you need to know at www.voip-info.org |
| 01:36.43 | [hC] | az^^za: its a wiki with a retarded amount of info on asterisk |
| 01:37.05 | az^^za | excellent, for us retards ;) |
| 01:37.50 | saftsack | http://www.voip-info.org/wiki/view/Asterisk+firewall+rules |
| 01:38.33 | az^^za | does it only run on Red Hat 7.3? |
| 01:39.25 | ZX81 | JT: reckon |
| 01:39.30 | ZX81 | :) |
| 01:40.04 | JT | ZX81: if they both ring at the same time... |
| 01:40.14 | JT | hell, even too disimilar -48VDC sources |
| 01:43.07 | Aces1Up | when communicating with the ami what does the ActionID do exactly? is it something i make up in my script? |
| 01:45.45 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4eb951b7229f2298) |
| 01:47.47 | ZX81 | JT: just using it as inbound trunks - doesn't work as outbound |
| 01:47.55 | ZX81 | have tdm2400p as outbound |
| 01:48.19 | JT | ZX81: what is with this crackpot analogue lines everywhere setup? ;) |
| 01:51.46 | ZX81 | :) I know - its a town up north in New Zealand - the exchange can't provide ISDN but the customer wanted DDI numbers - so the telco set up this weird stuff |
| 01:52.19 | ZX81 | they also hang up on a call after 3 minutes if they don't get a polarity reversal at the start :) |
| 01:52.30 | JT | how do they signal the did? |
| 01:52.40 | ZX81 | they pick up the line and dial an extension |
| 01:52.41 | ZX81 | :) |
| 01:52.43 | ZX81 | 3 digits |
| 01:52.49 | ZX81 | as if the exchange was a phone |
| 01:52.50 | ZX81 | :) |
| 01:53.03 | JT | when you pick up the line, they send a 3 dtmf code? |
| 01:53.20 | ZX81 | the signalling is fxs - so its as if they were a phone |
| 01:53.26 | ZX81 | so they pick up and dial 3 digits |
| 01:53.51 | JT | that's screwed up |
| 01:53.54 | ZX81 | hell yeah |
| 01:53.58 | JT | yet they provide talk battery current? |
| 01:54.02 | ZX81 | yeah |
| 01:54.03 | ZX81 | :) |
| 01:54.13 | ZX81 | can't think of any way to get callerid from them though |
| 01:54.22 | *** join/#asterisk nath0099 (i=James@77-96-249-156.cable.ubr02.maid.blueyonder.co.uk) |
| 01:54.23 | ZX81 | normally an analogue phone doesn't provide it! |
| 01:54.56 | ZX81 | I should have known something was weird when they said they had 8 analogue lines and 24 phone numbers :) |
| 01:55.13 | JT | hmm |
| 01:55.22 | JT | what handles the termination if this setup now? |
| 01:55.37 | ZX81 | you mean before Asterisk? |
| 01:55.43 | ZX81 | was a fujitsu pbx |
| 01:56.04 | JT | hrm ok |
| 01:56.07 | ZX81 | so now they have 4 inbound lines and 4 outbound |
| 01:56.15 | JT | did it have a custom card to handle this |
| 01:56.18 | ZX81 | couldn't send calls out via the fxs |
| 01:56.22 | ZX81 | yeah must have done |
| 01:56.35 | ZX81 | heh and all cabling on a krone block thingy |
| 01:56.45 | ZX81 | I'd brought RJ cables |
| 01:56.54 | fujin | sounds nasty |
| 01:56.57 | ZX81 | indeed |
| 01:58.30 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:58.31 | JT | krone ftw :) |
| 01:58.59 | fujin | I'm glad i'm operating at the layer where I don't have to fuck with kroning or anything. |
| 01:59.04 | fujin | have cable monkey :) |
| 01:59.28 | JT | kroning is fine as long as it doesn't involve crawling |
| 02:00.00 | sopo2k4 | if my voip provider is going to forward the incoming calls to my asterisk server, what port would i have to open in order for my server to receive the call? |
| 02:01.34 | JT | udp 5060, 10000-20000 |
| 02:04.21 | Sci_05 | what is the command to show the variables again? |
| 02:06.01 | Sci_05 | core show function...thats what it was |
| 02:07.21 | az^^za | so is there any reason other than money for us not to buy cisco call manager? |
| 02:08.32 | Sci_05 | ya you r stuck with what they say and can only do how they say it and what they say it can do |
| 02:08.57 | JT | buying cisco is like writing a blank cheque |
| 02:09.05 | JT | and they treat their customers with contempt |
| 02:09.06 | sevard | hahaha |
| 02:09.20 | Sci_05 | lol that is the perfect way to expain cisco JT |
| 02:09.35 | az^^za | so they have gotcha's |
| 02:09.57 | Sci_05 | I have a client who bought a cisco call manager setup and spent 120K on and and an additional 80k on support and the damn thing still doesn't work right |
| 02:10.12 | *** join/#asterisk jerlique (n=jerlique@lnk2.adl.adsl.esc.net.au) |
| 02:10.27 | jerlique | How would I get INFO added to the list of abilities in "Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
| 02:10.28 | jerlique | " |
| 02:11.30 | JT | az^^za: yes, i guess you could call all that a "gotcha" |
| 02:11.32 | az^^za | so as far as asterisk is concerned... it's possible to run menu's "press 1 to do blah" eta.. and it can detect that in a call center all operators are busy and to take some kind of action? |
| 02:11.56 | Sci_05 | anyone know off hand what they replaced the "System" command to? I want to be able to call a few commands outside of asterisk but all the docs say to use "n,System" and I am not seeing system in the core show function... |
| 02:12.01 | az^^za | btw: this is more or less for a commercial helpdesk |
| 02:12.20 | JT | yes, IVR, and queues |
| 02:12.21 | jerlique | az^^za; yes it can do that |
| 02:13.13 | az^^za | thanks jerlique |
| 02:13.29 | az^^za | and JT |
| 02:13.37 | ZX81 | hey a quick question |
| 02:13.50 | ZX81 | when sending out a mail via voicemail |
| 02:14.08 | ZX81 | how do I cause it to use a particular smtp server instead of localhost |
| 02:14.44 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 02:25.22 | *** join/#asterisk zpertee (n=chatzill@cpe-65-189-209-131.neo.res.rr.com) |
| 02:25.35 | zpertee | has anyone used zapmicro hardware? |
| 02:26.56 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:29.24 | snuff-work | mailcmd=/usr/sbin/sendmail -t |
| 02:29.27 | snuff-work | in voicemail.conf |
| 02:35.53 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 02:46.12 | *** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net) |
| 02:56.09 | *** join/#asterisk implicit (n=bayan@vc240151.vpn.uci.edu) |
| 03:00.27 | *** join/#asterisk levi_home (n=levi@levi.dsl.xmission.com) |
| 03:13.11 | *** join/#asterisk \malex\ (i=24pULTGv@unaffiliated/malex/x-000000001) |
| 03:13.58 | \malex\ | is it known when asterisk.org will be back online? |
| 03:18.45 | x86 | when did it go down? |
| 03:19.18 | \malex\ | sometime before i asked :) |
| 03:20.54 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:22.17 | Hmmhesays | well the 15v works |
| 03:22.35 | JT | surprise surprise ;) |
| 03:25.40 | *** join/#asterisk jsin (n=jason@cpe-75-184-119-6.indy.res.rr.com) |
| 03:28.51 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 03:38.58 | *** part/#asterisk \malex\ (i=24pULTGv@unaffiliated/malex/x-000000001) |
| 03:40.04 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 03:50.14 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 03:52.13 | *** join/#asterisk bmg505 (n=leon@196.209.183.97) |
| 03:57.07 | *** join/#asterisk joshr (n=joshr@65.103.116.63) |
| 04:11.52 | *** join/#asterisk walter_rodrigues (n=walter@201-048-147-003.static.ctbctelecom.com.br) |
| 04:13.27 | walter_rodrigues | need help installing TDM24XXP with 8 FXO + 16 FXS ... it doesn't install definetely not...Walter Rodrigues Filho - Brazil |
| 04:17.07 | walter_rodrigues | dmesg relays this.... wctdm24xxp: Unknown symbol pci_module_init |
| 04:18.30 | Strom_M | what version of zaptel are you building? |
| 04:20.22 | snuff-work | sounds like u have 2.6.22 kernel |
| 04:24.44 | *** join/#asterisk AdamB0122 (n=Adam@207.200.28.175) |
| 04:24.46 | AdamB0122 | Hey everyone |
| 04:26.08 | walter_rodrigues | sorry for the delay...zaptel 1.4.4 |
| 04:26.31 | walter_rodrigues | kernel 2.6.22.1-27.fc7 |
| 04:27.22 | walter_rodrigues | I am being beaten like a street dog by this TDM 24XXP...:( |
| 04:27.55 | walter_rodrigues | snuff...seems you sniffed my problem..huh? |
| 04:38.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:39.01 | AdamB0122 | anyway to find out info about a T1 without calling the phone company? |
| 04:39.40 | [TK]D-Fender | AdamB0122, could you be a little more vague please? I think I still know the general topic you're speaking about..... |
| 04:39.55 | AdamB0122 | sorry, Like, I'm trying to configure my T1 card |
| 04:40.05 | AdamB0122 | and I dont know things like framing, or coding |
| 04:40.19 | AdamB0122 | or which side is doing timing, ect |
| 04:40.23 | [TK]D-Fender | AdamB0122, Why not? Don'y you have a copy of the circuit order? |
| 04:40.58 | AdamB0122 | Pretty much no. I have a tele cabinet, with crap going everywhere |
| 04:41.05 | [TK]D-Fender | AdamB0122, And you didn't order it yourself? |
| 04:41.10 | [TK]D-Fender | I meant PAPER |
| 04:41.17 | AdamB0122 | no, i didn't |
| 04:41.25 | AdamB0122 | it was here before I was @ this company |
| 04:41.26 | [TK]D-Fender | AdamB0122, You know, like the contract that describes the service your signing up for... |
| 04:41.38 | AdamB0122 | and of course, the guy before me, didn't keep track of jack shit |
| 04:41.41 | [TK]D-Fender | AdamB0122, and THEY don't have it either? |
| 04:41.42 | Strom_M | AdamB0122: step 1: call the phone company |
| 04:41.53 | AdamB0122 | yea |
| 04:41.58 | AdamB0122 | thats what I figured i'd have to do |
| 04:42.21 | AdamB0122 | but i wanted to know if their was any magical way to avoid doing such as i wanted to get alot of work done tonight, and its like midnight here |
| 04:42.42 | Strom_M | AdamB0122: how much do you know about the circuit? |
| 04:42.52 | AdamB0122 | Unfortualtely not much |
| 04:42.58 | Strom_M | is it ISDN or CAS? |
| 04:43.07 | AdamB0122 | ISDN |
| 04:43.12 | Strom_M | ok |
| 04:43.18 | Strom_M | well, that makes life easier |
| 04:43.40 | AdamB0122 | actually one sec |
| 04:43.40 | Strom_M | try receiving timing, esf/b8zs, d-channel on 24, NI2 switchtype |
| 04:44.09 | AdamB0122 | One of the other Sysop's (the only one here longer then I) just got online, lemme go see if he has it |
| 04:48.38 | walter_rodrigues | Please...could somebody hint me on potential problems envolving kernel 2.6.22.1-27.fc7 and installing TDM24XXP? |
| 04:51.30 | AdamB0122 | for receiving timing, is that a 0 or 1 |
| 04:51.49 | Strom_M | 1 |
| 04:53.36 | AdamB0122 | ZT_SPANCONFIG failed on span1: invalid arg |
| 04:54.02 | Strom_M | pastebin teh config |
| 04:55.00 | AdamB0122 | ./etc/zaptel.conf > http://rafb.net/p/5TFdY361.html |
| 04:56.05 | AdamB0122 | ./etc/asterisk/zapata.conf > http://rafb.net/p/UI2I0d53.html |
| 04:56.38 | AdamB0122 | (its a T1 from XOCommunications, if that helps at all) |
| 05:01.21 | bkruse_home | Strom_M: pastebin teh! |
| 05:01.23 | bkruse_home | :D |
| 05:01.55 | AdamB0122 | (those are my config, did you want something differently?) |
| 05:02.55 | snuff-work | walter_rodrigues, svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 |
| 05:03.18 | snuff-work | that shoudl then allow u to compile a new zaptel for your 1.4 asterisk |
| 05:04.35 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 05:05.33 | Strom_M | AdamB0122: unload your drivers and reload them |
| 05:06.24 | AdamB0122 | gr, whats the command to check their removed, I just modprobe -r'ed, but i cant remember how to check |
| 05:07.10 | snuff-work | lsmod ? |
| 05:07.53 | AdamB0122 | yea, and what the hell |
| 05:08.01 | AdamB0122 | zaptel 186532 10 wcte11xp,zttranscode,ztdummy,wctdm,wcfxo,wct1xxp,wct4xxp,tor2 |
| 05:08.05 | JT | ... |
| 05:08.06 | JT | wtf |
| 05:08.12 | JT | you don't need all that crap loaded |
| 05:08.16 | JT | stupid distro |
| 05:08.35 | snuff-work | the init script trys to load em all i know that much |
| 05:08.48 | AdamB0122 | yea |
| 05:09.28 | AdamB0122 | zaptel 186532 3 wcte11xp |
| 05:09.46 | AdamB0122 | i've removed everything but the TE120p mod |
| 05:10.01 | Strom_M | wcte11xp is for the TE110P |
| 05:10.08 | Strom_M | the TE120P uses wcte12xp |
| 05:10.24 | AdamB0122 | gr. damn quickstart guide |
| 05:11.15 | AdamB0122 | ok |
| 05:11.19 | AdamB0122 | its not using wcte12xp |
| 05:11.46 | AdamB0122 | now* |
| 05:12.20 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 05:12.21 | AdamB0122 | last time i worked with a T1 card though, when I loaded the mod using modprobe, the red light on the actual T1 card started blinking. This one is currently not doing so. |
| 05:12.37 | Strom_M | it /is/ a te120p, right? |
| 05:12.46 | AdamB0122 | yes |
| 05:12.51 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 05:12.56 | Strom_M | does it show up when you run lspci -bv? |
| 05:12.59 | AdamB0122 | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-58188221440.htm?gclid=COObqpC3v40CFRU6OAodlWy8Lw |
| 05:13.04 | AdamB0122 | that one, bought it 3 days ago |
| 05:13.40 | AdamB0122 | 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) |
| 05:13.48 | Strom_M | alright |
| 05:13.57 | Strom_M | ztcfg -vv |
| 05:13.58 | Strom_M | what happens |
| 05:14.09 | AdamB0122 | oh damn, no errors |
| 05:14.17 | AdamB0122 | i get a chanmap of 01-24 |
| 05:14.24 | Strom_M | nice |
| 05:14.24 | AdamB0122 | then a changing signalling on 1-24 |
| 05:14.29 | Strom_M | is it in red alarm now? |
| 05:14.32 | AdamB0122 | and then an end... i can nopaste if you like |
| 05:14.34 | AdamB0122 | brb |
| 05:14.50 | AdamB0122 | yes |
| 05:14.51 | AdamB0122 | it is |
| 05:14.59 | delmar | hey everyone. I have an interesting problem with DISA. I have contaxt 'local' under which the many other contexts are included. the local phones/extensions are all part of [local] and can all dial each other and dial out fine... there is NO problems with this at all.. it all works... DISA however, cant dial ANY of the local 3digit extensions at all, but can dialout to anything else. |
| 05:15.13 | Strom_M | AdamB0122: cool. |
| 05:15.19 | AdamB0122 | Very ^.^ |
| 05:15.37 | Strom_M | also, NI2 is switchtype=national in zapata.conf |
| 05:15.44 | AdamB0122 | ok |
| 05:15.48 | delmar | so a call can come in via the cellular GSM gateway.. get DISA access.. then dialout via VoIP ok.. and everything else.. but DISA wont dial a local extension... it just drops the call.. no console output that is useful or anything... |
| 05:15.54 | red9012 | how do I handle fax with asterisk? |
| 05:15.55 | delmar | anyone have an idea? |
| 05:16.01 | Strom_M | signalling should be pri_cpe also, AdamB0122 |
| 05:16.02 | delmar | red9012, spandsp |
| 05:16.06 | red9012 | is there a reliable solution? |
| 05:16.36 | delmar | red9012, analog or digital? |
| 05:16.44 | AdamB0122 | Strom_M > Done as well |
| 05:17.00 | delmar | red9012, i had nothing but issues with X100 and TDM400 FXO. |
| 05:17.31 | delmar | red9012, better of using 3rd party application with a modem and faxability. get Asterisk to ignore faxability and let the other device answer. |
| 05:17.36 | delmar | red9012, works a charm. |
| 05:17.46 | Strom_M | AdamB0122: now, try loading asterisk |
| 05:17.49 | AdamB0122 | worked |
| 05:17.58 | AdamB0122 | and Zap show channels shows me 23 channels |
| 05:18.16 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 05:18.21 | Strom_M | now....does the span come up, and can you place and receive calls? |
| 05:19.19 | AdamB0122 | Sorry, the span? (sorry, Im new to T1's) |
| 05:19.33 | delmar | red9012, the main issue I had was.. I could tweak the hell out of the TX / RX gains until I had it perfect for Fax, but then I was running into echo issues again. |
| 05:19.38 | AdamB0122 | But i cannot recieve a call |
| 05:19.56 | delmar | red9012, i rather have voice working good and tell whoever wants to send me a fax to piss off. |
| 05:20.03 | Strom_M | AdamB0122: what happens when you type "pri show span 1" at the CLI? |
| 05:20.34 | AdamB0122 | shows me information on what looks like the t1 |
| 05:20.37 | delmar | red9012, instead I used the faxability detection in Asterisk to ignore the incoming fax call, and let my box running Relayfax answer it. |
| 05:20.55 | AdamB0122 | http://rafb.net/p/HNCstG48.html |
| 05:20.59 | Strom_M | AdamB0122: do you see "provisioned, up, active"? |
| 05:21.14 | AdamB0122 | No, porvision,ed in alarm, down, active |
| 05:21.30 | delmar | red9012, search the wiki for Spandsp if u want to try to build fax send/receive into Asterisk. I had fun with it at least. |
| 05:21.36 | Strom_M | is the T1 plugged into the card: |
| 05:21.36 | Strom_M | ? |
| 05:21.46 | AdamB0122 | yes |
| 05:21.56 | Strom_M | is the card in red alarm? |
| 05:22.09 | AdamB0122 | ( i assume you mean the blinking red light, ) yes |
| 05:22.20 | Strom_M | what kind of cable are you using |
| 05:22.29 | AdamB0122 | Standard Cat5e |
| 05:22.39 | Strom_M | what's on the other end? |
| 05:22.40 | AdamB0122 | i can go run a cable test on it, just to make sure |
| 05:22.49 | AdamB0122 | right into the giant grey XOcommunications box |
| 05:22.56 | delmar | AdamB0122, what about the T1/E1 jumper on the card? |
| 05:23.03 | Strom_M | does that box have an HDSL card in it? |
| 05:23.28 | AdamB0122 | delmar > shutting down box to double check, hopefully i wasn't that stupid, |
| 05:23.41 | AdamB0122 | Strom_M > I'm not 100% sure what that would look like, so i dont knw |
| 05:23.42 | delmar | its a common thing actually |
| 05:24.02 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 05:24.03 | Strom_M | AdamB0122: it'll be a card with visible status lights in the XO smartjack |
| 05:24.03 | delmar | we are all E1 here... as are alot of the world. |
| 05:24.25 | Strom_M | delmar: it sounds like adam is in north america |
| 05:24.27 | Strom_M | therefore T1 |
| 05:24.27 | AdamB0122 | One moment, taking laptop off 2nd monitor to walk back there |
| 05:24.33 | delmar | yup. |
| 05:24.40 | delmar | should be T1 in the USA for sure. |
| 05:25.23 | delmar | E1 u get 30 channels :P |
| 05:25.23 | snuff-work | heh if u deploy a E1 in the states would you be considered unpatriotic? |
| 05:25.38 | *** join/#asterisk AdamB0122 (n=Adam@207.200.28.175) |
| 05:25.39 | AdamB0122 | whopps |
| 05:25.41 | delmar | snuff-work, nah. just too sensible.. hence it will never happen |
| 05:25.47 | AdamB0122 | and yes, I'm in USA |
| 05:25.55 | delmar | AdamB0122, was set to E1 huh? :P |
| 05:26.04 | AdamB0122 | duno, opening box now |
| 05:26.43 | delmar | strangely tho.. I see alot of people in the USA getting their first card and the damn thing was set to E1. |
| 05:26.46 | AdamB0122 | on the card of i've E1 = on, T1 = off |
| 05:26.56 | delmar | oops |
| 05:26.59 | AdamB0122 | and the jumper is only on ONE pin |
| 05:26.59 | Strom_M | AdamB0122: for T1 the jumper should be open |
| 05:27.04 | Strom_M | yeah, so it's set to T1 |
| 05:27.06 | AdamB0122 | so its an open, or off connection |
| 05:27.11 | delmar | oh ok |
| 05:27.19 | Strom_M | let's work backwards and see if the T1 is provisioned at the smartjack |
| 05:27.22 | delmar | sounds like u are on T1 then.. so its no that. |
| 05:27.31 | Strom_M | is there a card in the smartjack which has lights on it? |
| 05:27.57 | delmar | with I had a T1 card to play with the channel bank I was given |
| 05:28.18 | delmar | why are T1 cards so damn expensive. they are just a bloody ethernet card with different programming in the micro !!!. |
| 05:28.19 | delmar | grrr |
| 05:28.25 | AdamB0122 | Strom_M > I'm at the grey XO box now, and i dont see anything labed smartjack |
| 05:28.33 | JT | ethernet card.... riight |
| 05:28.39 | Strom_M | delmar: not quite |
| 05:28.48 | JT | delmar: supply and demand |
| 05:29.00 | JT | they are not ethernet cards anyway |
| 05:29.03 | Strom_M | AdamB0122: the smartjack is the jack your T1 comes out of from the XO box |
| 05:29.40 | JT | it's a dumb name americans use instead of T1 or G.703 or whatever a T1 is ;) |
| 05:30.00 | AdamB0122 | ran a cable test, the Cat5e line is good, just to make sure |
| 05:30.08 | AdamB0122 | then no, no lgihts |
| 05:30.13 | AdamB0122 | not even LED's for lights |
| 05:30.24 | *** join/#asterisk Jedi-Jiji (n=Jedi-Jij@82.252.35.4) |
| 05:30.25 | Strom_M | can you describe the box? |
| 05:30.41 | AdamB0122 | uh. i can send a pic, if that'd be easier |
| 05:30.44 | Strom_M | yes |
| 05:30.45 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:30.52 | AdamB0122 | one moment, lemme grab my conn cable |
| 05:32.57 | AdamB0122 | http://img530.imageshack.us/img530/3417/img012lv5.jpg |
| 05:33.37 | AdamB0122 | http://img360.imageshack.us/img360/9486/img013st7.jpg |
| 05:33.39 | Strom_M | can you take a wider angle? |
| 05:33.51 | AdamB0122 | ya, one sec |
| 05:34.04 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 05:34.07 | *** join/#asterisk astserdev (n=core@dns1.muppidis.com) |
| 05:34.53 | *** join/#asterisk babu_ast (n=babu@dns1.muppidis.com) |
| 05:35.51 | AdamB0122 | http://img118.imageshack.us/img118/4972/img015kc9.jpg |
| 05:36.45 | JT | wow, those photos really aren't ideal |
| 05:36.56 | JT | fuzzy and can't see any labels or connectors straight on |
| 05:37.06 | AdamB0122 | yea, sorry, my phone kinda sucks |
| 05:37.07 | denon | camera phone :) |
| 05:37.09 | denon | hah |
| 05:37.15 | AdamB0122 | yea, its the best i got right now |
| 05:37.40 | AdamB0122 | ... *tears up* |
| 05:37.47 | AdamB0122 | what about my LikaM7! |
| 05:37.51 | JT | hah, few mm bigger screen, take that Strom_M |
| 05:37.53 | denon | Nikon..psha |
| 05:37.59 | AdamB0122 | lol |
| 05:38.09 | Strom_M | JT: it's not the size, it's how you use it!!! |
| 05:38.09 | AdamB0122 | I have a Sony something or other |
| 05:38.10 | JT | canon slrs are toys by comparison ;) |
| 05:38.14 | AdamB0122 | dunno really |
| 05:38.23 | Strom_M | JT: which lenses do you have? |
| 05:38.44 | JT | Strom_M: mostly just use the 18-200mm VR lense |
| 05:38.58 | JT | but also got an old manual focus 300mm mirror lense to play around with |
| 05:39.11 | denon | JT: I'm guessing that you're kidding? :) |
| 05:39.25 | Strom_M | i've got the 18-70 f/3.5-4.5 kit lens, and a 50mm f/1.4 :) |
| 05:39.30 | denon | have you seen the mark III? |
| 05:39.48 | JT | denon: not really, they consumer and prosumer SLRs aren't build nearly as strong as equivalent nikons |
| 05:39.49 | AdamB0122 | Strom_M > do those pics help any? |
| 05:39.54 | JT | also the UI and screen design is poor |
| 05:39.56 | Strom_M | AdamB0122: no, not really |
| 05:40.10 | denon | JT: 10fps, with 110 burst |
| 05:40.15 | AdamB0122 | dang. anything i can look for, that'd help? |
| 05:40.19 | denon | does your fisher price^H^H^H^H^H^H nikon do that? |
| 05:40.25 | Strom_M | AdamB0122: status lights? |
| 05:40.29 | JT | denon: MarkIII is not prosumer or consumer |
| 05:40.32 | JT | that's pro |
| 05:40.41 | denon | you said canon slr were all toys :) |
| 05:40.47 | JT | and has a price tag to match |
| 05:40.52 | AdamB0122 | one sec |
| 05:41.02 | AdamB0122 | ahh |
| 05:41.06 | AdamB0122 | status lights are on the top |
| 05:41.11 | Strom_M | ah good |
| 05:41.13 | Strom_M | what do you see |
| 05:41.15 | JT | eh, look at the differences in build quality between a D70s and a 350d, you'll see what i mean |
| 05:41.16 | denon | JT: well, in the world of cameras, I dont think nikon has anything to compare with the markIII |
| 05:41.23 | denon | at any cost |
| 05:41.33 | JT | denon: how many MP? |
| 05:41.39 | AdamB0122 | the box is a Adit 600 |
| 05:41.41 | denon | MP means nothing .. |
| 05:41.42 | AdamB0122 | 3 cards |
| 05:41.48 | AdamB0122 | FXS cards |
| 05:42.02 | AdamB0122 | first card has 1 green light |
| 05:42.10 | Strom_M | FXS != T1 |
| 05:42.11 | JT | denon: it's not everything, it still means something |
| 05:42.17 | denon | well, about 10.7M |
| 05:42.21 | JT | leica make digital cmaeras now |
| 05:42.22 | AdamB0122 | and the other two cards have all eight lights that are yellow |
| 05:42.27 | AdamB0122 | Strom_M > yea, I didn't think so either |
| 05:42.36 | JT | and seitz make a 160MP camera |
| 05:42.46 | JT | so there's plenty higher than a MArkIII |
| 05:43.01 | denon | I said nikon :) |
| 05:43.24 | denon | http://www.usa.canon.com/consumer/controller?act=ModelInfoAct&fcategoryid=139&modelid=14999 |
| 05:43.31 | JT | so what, it does 10fps bursts, pretty impressive, but meh |
| 05:43.41 | denon | it really is spectacular, look through the features |
| 05:44.01 | JT | when their slow website finally loads |
| 05:44.14 | denon | loads fine for me - though I gave you the link to the usa mirror |
| 05:44.17 | denon | perhaps the pond is lagged |
| 05:44.49 | JT | hardly |
| 05:44.57 | denon | 63 zones metering, 19 AF points .. |
| 05:45.03 | Strom_M | guys guys |
| 05:45.05 | denon | hmm? it loaded instantly here .. |
| 05:45.21 | JT | over 1Tbit/s to the US, we are not lagged, it's their server and clientside heavy rendering |
| 05:45.28 | JT | denon: you can have 3848430307034709 AF areas |
| 05:45.31 | JT | you don't need that many |
| 05:45.44 | AdamB0122 | http://img360.imageshack.us/img360/3066/img020gl3.jpg |
| 05:46.00 | AdamB0122 | thats the status lgihts, and 1/2 of me since the damn box was put so high above the ground |
| 05:46.16 | JT | haha it's an ADit600 channel bank |
| 05:46.32 | Strom_M | AdamB0122: yeah, thats a channel bank; that's not ISDN PRI service |
| 05:46.35 | JT | AdamB0122: it's not ISDN. |
| 05:46.38 | denon | well, for whatever reason, the server must have gotten lagged the moment after I loaded the page then |
| 05:46.39 | AdamB0122 | fuck |
| 05:46.43 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 05:46.44 | denon | anyway - its a nice camera, whether ya like canon or not |
| 05:46.49 | AdamB0122 | No way. |
| 05:46.51 | denon | Im sure nikon makes nice cams too |
| 05:46.53 | denon | nuff said |
| 05:47.15 | JT | AdamB0122: CAS T1 |
| 05:48.10 | *** join/#asterisk babu_ast (n=babu@dns1.muppidis.com) |
| 05:48.11 | *** join/#asterisk phigan (n=phig@ip68-109-169-37.ph.ph.cox.net) |
| 05:48.22 | phigan | hi |
| 05:48.25 | JT | denon: my whole point was the comparison in build quality, and usability for the consumer and prosumer dslrs |
| 05:49.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:49.24 | Strom_M | JT: http://www.flickr.com/photos/stromcarlson/877971916/in/set-72157600967906271/ |
| 05:49.27 | Strom_M | f/1.4 :) |
| 05:49.43 | AdamB0122 | ok |
| 05:50.07 | JT | Strom_M: nice, manual settings, handheld? |
| 05:50.10 | AdamB0122 | sorry, I was told that from another person, and I didn't know wtf i was looking for |
| 05:50.22 | JT | what lense do you have that's f/1.4 |
| 05:50.27 | Strom_M | JT: aperture priority |
| 05:50.37 | Strom_M | the nikon 50mm AF f/1.4 lens |
| 05:50.43 | JT | ah ok |
| 05:50.48 | Strom_M | and yes, handheld |
| 05:50.48 | JT | you buy it new or second hand? |
| 05:50.54 | Strom_M | thats the only new thing in my kit |
| 05:51.07 | JT | hrm |
| 05:51.07 | Strom_M | the camera and the other lens were secondhand and refurb, respectively :) |
| 05:51.25 | Strom_M | the f/1.4 was a gift from my dad |
| 05:51.25 | JT | i can't sing the praises of the 18-200mm more highly for use when out and about :) |
| 05:51.45 | AdamB0122 | to change this to a CAS setup, what all do i need to do, change emf to cas in the zaptel.conf |
| 05:51.57 | JT | emf? |
| 05:52.58 | AdamB0122 | I'm shooting in the dark, mixed between googling and trying to follow a not-so-helpful "how to" |
| 05:53.30 | AdamB0122 | aspan=1,1,0,esf,b8zs to something like span=1,1,0,cas,b8z |
| 05:53.41 | JT | b8zs |
| 05:53.49 | AdamB0122 | yea, missed a key |
| 05:53.54 | JT | and signalling would be fxs_ks |
| 05:53.57 | JT | and no d channel |
| 05:53.58 | JT | etc |
| 05:54.37 | Strom_M | AdamB0122: no, esf is your framing |
| 05:54.42 | Strom_M | you'll leave that as esf |
| 05:54.49 | Strom_M | assuming your t1 is esf |
| 05:54.55 | Strom_M | if it's d4, you set that to d4 |
| 05:55.01 | AdamB0122 | ok |
| 05:55.06 | *** join/#asterisk tuzhila (i=tuzhila@84.47.128.99) |
| 05:55.12 | tuzhila | hi all |
| 05:55.39 | AdamB0122 | do i still use a switchtype? |
| 05:55.47 | AdamB0122 | no |
| 05:55.54 | JT | http://www.voip-info.org/wiki/index.php?page=Zaptel.conf+span+syntax |
| 05:55.58 | JT | no switchtype |
| 05:56.44 | phigan | anyone ever get "callerid.c: Caller*ID failed checksum" ? |
| 05:56.57 | phigan | my callerid worked for a while, then suddenly stopped working |
| 05:57.42 | delmar | JT, so I just got off the phone with the old man.. the short answer is there is very little cost difference at all.. between a decent ethernet card and a T1 card, and they are WAY WAY over priced for what they are. |
| 05:57.53 | phigan | there's a phone connected to the zaptel card, it gets cid info just fine each time. the * will pass correct cid once every 10 or so calls |
| 05:58.11 | phigan | or more |
| 05:58.12 | JT | delmar: i'm not sure who your old man is, and i'm not sure how it matters |
| 05:58.12 | denon | phigan: please tell me you dont have an x100p |
| 05:58.16 | JT | but as i said before |
| 05:58.24 | JT | denon: supply and demand |
| 05:58.32 | delmar | JT, oh.... www.dynamics.co.nz |
| 05:58.37 | JT | unless the demand goes up, the price will stay home |
| 05:58.39 | delmar | JT, hardware dev. |
| 05:58.57 | JT | delmar: cool, well i don't need him to tell me the obvious, sorry to be blunt |
| 05:59.03 | JT | gar |
| 05:59.03 | phigan | denon: i'm sure that's what it is. just the one line card? |
| 05:59.09 | JT | unless the demand goes up, the price will come DOWN |
| 05:59.15 | denon | phigan: did you get it for like $3 off ebay? |
| 05:59.21 | snuff-work | mm.. telstra are going to ditch isdn in a few years |
| 05:59.25 | JT | i blased that up again |
| 05:59.28 | JT | snuff-work: haha what bs |
| 05:59.33 | JT | sounds like marketing crap |
| 05:59.37 | phigan | hehe, no, a friend that works with * gave it to me |
| 05:59.42 | tzafrir | delmar, the design of tor2 is free. make your own :-) |
| 05:59.55 | denon | phigan: probably is -- those are really crappy cards, but more to the point, callerid is also flaky on them |
| 06:00.01 | denon | so you're probably not going to be happy with it .. ever. |
| 06:00.09 | snuff-work | jt.. i'm sure if ur big enough.. after that time u can still have whatever u want.... but i'd think mostly marketing too |
| 06:00.23 | phigan | it's weird that it went from working 9 out of 10 times to only 1 out of 10 times |
| 06:00.33 | delmar | JT, well I wanted to understand the core differences between a regular ethernet card and a T1, and the operational aspects are obviously significantly different, the component differences are minor apparently....so I go back to my original question... as to why the damn things are costing so much :( |
| 06:00.34 | JT | snuff-work: what do they propose will replace isdn? |
| 06:00.45 | denon | JT: cablemodems and mux :) |
| 06:00.51 | delmar | tzafrir, tor2 ? |
| 06:00.51 | phigan | is there anything that i can check or do? |
| 06:00.53 | tzafrir | phigan, which card, which country? |
| 06:00.58 | denon | phigan: buy a tdm400 card |
| 06:00.59 | JT | delmar: supply and demand, i'll say it for the third time now |
| 06:01.27 | JT | delmar: it's how almost the entire economy works with goods and services |
| 06:01.28 | phigan | tzafrir: I guess the x100p? US |
| 06:01.33 | delmar | JT, sadly, its as simple as that yes. |
| 06:01.45 | AdamB0122 | anyway to tell what "ZT_SPANCONFIG failed on span 1: Invalid argument (22)" is actually failing on in span1? |
| 06:01.54 | JT | delmar: it's not worthwhile for companies to drop prices unless they ship more units |
| 06:01.59 | tzafrir | delmar, of the zapata telephony project. Still available in the sources of zaptel |
| 06:02.15 | tzafrir | delmar, also look up astfin.org |
| 06:02.20 | delmar | tzafrir, sorry... what is it and what does it do ? |
| 06:02.20 | tzafrir | (if it's up) |
| 06:02.25 | JT | delmar: t1 card |
| 06:02.41 | tzafrir | I thought you wanted a T1 card you can make cheaply... |
| 06:02.49 | delmar | tzafrir, to work with Asterisk? |
| 06:02.50 | phigan | or, is there any sort of patch that will pass the CID info it gets regardless of if it passes checksum? |
| 06:03.18 | tzafrir | AdamB0122, it means that /etc/zaptel.conf does not match your current system (compare to cat /proc/zaptel/* ) |
| 06:03.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:03.23 | JT | delmar: YES FOR ASTERISK, now look it up before asking more obivous questions :) |
| 06:03.56 | AdamB0122 | on span1 i have ztdummp |
| 06:03.58 | tzafrir | either /etc/zaptel.conf is badly written or relevant module not loaded, usually. What card do you have? |
| 06:04.01 | AdamB0122 | er |
| 06:04.15 | AdamB0122 | damn init script |
| 06:04.22 | AdamB0122 | killed my modprobeing |
| 06:04.23 | AdamB0122 | one sec |
| 06:04.40 | AdamB0122 | with modprobe, is there anyway to just drop All modules for a certain item? |
| 06:04.54 | AdamB0122 | ie: drop all zaptel mods, because its got like 10 of them right now |
| 06:05.16 | tzafrir | AdamB0122, xpp/utils/genzaptelconf will detect your configuration nicely |
| 06:05.31 | tzafrir | AdamB0122, xpp/utils/genzaptelconf -sdvM |
| 06:05.35 | AdamB0122 | genzaptelconf didn't work very well |
| 06:05.45 | tzafrir | Which version? |
| 06:05.51 | AdamB0122 | one sec |
| 06:05.53 | tzafrir | That needs fixing |
| 06:05.55 | AdamB0122 | lemme find out |
| 06:06.04 | phigan | tzafrir: any idea about the cid checksum failure problem? |
| 06:06.06 | AdamB0122 | I have a WCTE120P |
| 06:06.43 | AdamB0122 | i dont have a version for genzaptelconf |
| 06:06.59 | AdamB0122 | but it does this |
| 06:07.00 | AdamB0122 | zaptel 186532 7 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 06:07.05 | AdamB0122 | to my zaptel card, lol |
| 06:07.15 | AdamB0122 | and never loads the right driver for the WCTE120P |
| 06:09.51 | AdamB0122 | http://rafb.net/p/mOmJAD79.html thats the zaptel.conf that it generates for me |
| 06:10.21 | AdamB0122 | http://rafb.net/p/UCotFZ68.html thats the zapata.conf that it generates |
| 06:12.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:12.45 | jerlique | I'm having problems with * listening to DTMF from a channel bank, the sip debug says unauthorised, any hints? |
| 06:13.45 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:18.35 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 06:19.15 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:22.16 | AdamB0122 | ok |
| 06:22.20 | AdamB0122 | I'm getting somewhere |
| 06:22.33 | phigan | what's the correct syntax to set callerid? Set(CALLERID(all)="Name" <number>) or =Name <number>) ? |
| 06:22.36 | AdamB0122 | on a T1 with a channelbank, How can i check the status? |
| 06:22.49 | AdamB0122 | I've gotten the ztcfg -vvvv to work the way its supposed to, no errors |
| 06:23.16 | AdamB0122 | 24 channels configured, all FXO Kewlstart |
| 06:25.13 | AdamB0122 | however, when i call in, I get about 10 second of silence, and then the phone hangs up |
| 06:26.52 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:27.54 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4eb951b7229f2298) |
| 06:30.03 | JT | Name <number> |
| 06:31.18 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 06:31.47 | AdamB0122 | how can I see whats making the Alarms "red" in zap show status |
| 06:33.32 | AdamB0122 | hm |
| 06:34.08 | bakermd | I dont get it... Unable to open /var/spool/asterisk/outgoing/sample.call: Permission denied, deleting |
| 06:34.23 | bakermd | Permission denied? Then how were you able to delete it?? |
| 06:35.12 | phigan | damn. I'm using LES.Net, and when I set my outgoing CID it just goes across as "number <number>" instead of "name <number>" :/ |
| 06:37.18 | bakermd | Whats the quality with les.net? |
| 06:38.01 | phigan | i haven't used it much, but good so far |
| 06:38.11 | bakermd | cool |
| 06:38.44 | bakermd | I hate this - I search for my issue on Google and get 100 sites that are all a mirror of the same damn forum thread that offers no assistance |
| 06:38.46 | AdamB0122 | when i do a cat /proc/zaptel/* |
| 06:38.47 | phigan | it's being annoying atm, sip registration times out, and iax2 calls are seeming to take a year to connect out |
| 06:38.56 | AdamB0122 | I see IRQ misses: 61 |
| 06:38.58 | phigan | bakermd: i know the feeling |
| 06:38.58 | AdamB0122 | what does that mean |
| 06:39.07 | AdamB0122 | same here. |
| 06:39.46 | phigan | Span 1: WCFXO/0 "Generic Clone Board 1" |
| 06:42.19 | JT | phigan: you usually can't set callerid name over the pstn |
| 06:42.30 | phigan | les.net is voip |
| 06:43.09 | JT | phigan: it connects to the pstn. |
| 06:43.40 | phigan | well, i'm setting the number. But it's setting the number for the name as well |
| 06:43.50 | JT | ok, and the problem is? |
| 06:44.04 | phigan | it's not setting the name i'm giving it |
| 06:44.08 | creativx | morning JT |
| 06:44.14 | JT | i just told you it won't happen, phigan . |
| 06:44.20 | JT | hey creativx |
| 06:44.26 | phigan | oh name? |
| 06:44.36 | phigan | i missed that, i thought you were saying i couldn't set callerid at all |
| 06:44.42 | AdamB0122 | ugh |
| 06:44.52 | bakermd | phigan: I've never had a provider that allowed setting name, only number |
| 06:44.52 | AdamB0122 | what woudl cause a red status on a TE12xP T1 card? |
| 06:45.03 | AdamB0122 | I've googled it , and nothing is helping |
| 06:45.14 | JT | it's not possible because the provider pulls it from a database at the other end |
| 06:45.17 | JT | the callerid name |
| 06:45.28 | bakermd | Ours show up as "Level3" on the other end |
| 06:45.29 | phigan | bakermd: Even if you set it on the provider's site, or do you mean setting it with your own equipment? |
| 06:45.54 | bakermd | Our providers do not give access to a portal to set it, that I am aware of. |
| 06:46.13 | phigan | there's a place for CallerID: on my les.net profile |
| 06:46.28 | phigan | err no, in my trunk config |
| 06:46.34 | JT | phigan: can you please read what i typed? you can almost NEVER set callerid NAME over the PSTN |
| 06:46.37 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
| 06:46.39 | JT | trunk config? |
| 06:46.47 | bakermd | That sounds like Trixbox |
| 06:47.22 | phigan | .. i called it that for lack of a better term.. on les.net's website |
| 06:47.31 | phigan | there's a section called peers/trunks |
| 06:47.41 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 06:47.45 | JT | oh, nice, they misuse terms too |
| 06:47.47 | phigan | i geuss maybe peer config? sorry |
| 06:47.49 | phigan | guess |
| 06:50.29 | AdamB0122 | gr |
| 06:50.31 | AdamB0122 | Wildcard TE12xP Card 0 RED 61 0 0 |
| 06:50.37 | AdamB0122 | I have that when i do a zap show status |
| 06:51.01 | AdamB0122 | and I've googled TE12xP alarm, and just about any other combonation of problems with a TE12xP T1 |
| 06:51.20 | AdamB0122 | needless to say, I can't find anything online |
| 06:52.16 | JT | what are you trying to do? disconnect the ADit 600 and plug the T1 from the telco directly into your box, or connect the Adit 600 to your box? |
| 06:52.31 | AdamB0122 | Connect Adit600 into the box |
| 06:52.45 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:52.47 | *** join/#asterisk friedrich| (n=friedric@e177240141.adsl.alicedsl.de) |
| 06:52.49 | AdamB0122 | the Adit600 went to our old phone system, which I'm replacing with this one |
| 06:52.49 | JT | oh, it would be useful if you clearly mentioned this |
| 06:53.03 | JT | everyone thought you were connecting to the telco |
| 06:53.04 | AdamB0122 | Sorry, its really late here, so i'm missing important things from time to time |
| 06:53.14 | JT | so you are just trying to hook up some analogue handsets? |
| 06:54.03 | AdamB0122 | I'm trying to setup a asterisk PBX so i can have computers with softphones can use the phone system |
| 06:54.13 | bakermd | JT: I am going insaine.. you got any ideas? Unable to open /var/spool/asterisk/outgoing/sample1.call: Permission denied, deleting |
| 06:54.31 | AdamB0122 | go look at the permissions on sample1.call? |
| 06:54.33 | bakermd | owner is asterisk:asterisk |
| 06:54.37 | JT | AdamB0122: ok, i give up, seriously. not worth my effort until you tells us what you are trying to do clearly |
| 06:54.41 | bakermd | I did a chmod 666 on it |
| 06:54.51 | AdamB0122 | ok1 |
| 06:55.06 | AdamB0122 | I currently have Telco > Channel bank > Old crappy phone system |
| 06:55.08 | JT | AdamB0122: softphones are great, but what on earth do they have to do with T1s etc |
| 06:55.21 | AdamB0122 | I want to setup a Telco > channel bank > Asterisk PBX |
| 06:55.25 | JT | great, i mean that sarcastically btw |
| 06:55.33 | JT | that's completely wrong. |
| 06:55.39 | AdamB0122 | fuck. |
| 06:55.41 | AdamB0122 | lol |
| 06:55.44 | JT | i have no idea what you've plugged into where |
| 06:55.54 | JT | what cables you've managed to force into what holes |
| 06:56.13 | JT | but if you want the telco to connect to asterisk, you will not use the channel bank at all |
| 06:56.17 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:56.29 | creativx | heheh |
| 06:57.01 | JT | there is an SHDSL modem / "smartjack" somewhere that connects 1 or 2 pairs in the street from the telco, has power, and has an RJ-45 output, find it |
| 06:57.08 | JT | it should be connecting to the channel bank |
| 06:57.18 | JT | unless you unplugged it |
| 06:57.30 | AdamB0122 | no, I haven't touched that side of the channel bank |
| 06:57.35 | bakermd | How can asterisk complain that it is unable to read a .call file due to permissions... and then turn around and delete the file. So you are not privileged enough to read it, yet you can delete it. Makes no sense at all |
| 06:57.56 | creativx | bakermd: sure it does |
| 06:57.59 | creativx | fine grain access control |
| 06:58.18 | bakermd | I have no response at all for that. |
| 06:58.22 | bakermd | ;-) |
| 06:58.25 | JT | bakermd: ls -la sample.call |
| 06:58.36 | bakermd | -rwxrwxrwx 1 asterisk asterisk 149 Jul 24 11:18 sample1.call |
| 06:58.41 | creativx | so i assume that when * says "i am deleting it" it actually happens |
| 06:58.49 | bakermd | Oh yeah |
| 06:58.57 | creativx | because deleting is not the same as deleted |
| 06:59.03 | bakermd | Right |
| 06:59.12 | bakermd | But it is successfull in the deleting operation |
| 06:59.21 | JT | AdamB0122: well i suggest you locate the modem |
| 06:59.31 | bakermd | It deletes the file, and then my will to live ;-) |
| 06:59.41 | creativx | welcome to the world of computers bakermd |
| 07:00.01 | bakermd | lol - I've been in it since I was 12. Nothing new here |
| 07:00.53 | bakermd | I really love asterisk, but damnit it needs to work! I've used the same setup on other PBX's - dont know why I am having an issue now |
| 07:04.08 | *** join/#asterisk MrMister2 (n=mrmister@89.181.104.76) |
| 07:04.13 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 07:04.22 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:101:ec0d:68a9:dfa4:dcd8) |
| 07:04.31 | bakermd | Inbound and Outbound working fine. |
| 07:04.37 | bakermd | .call files failing |
| 07:05.20 | tzafrir_laptop | found a problem with genzaptelconf's modules detection on some systems due to the stupid aliaes of automatic run of ztcfg |
| 07:05.51 | tzafrir_laptop | Problem is that some platforms don't have modprobe -i, and hence I can't use it |
| 07:06.37 | tzafrir_laptop | bakermd, can asterisk write to the directory? this is what you need to unlink a file |
| 07:07.42 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:07.44 | bakermd | tzafrir_laptop: Yes, ownership on directory is asterisk:asterisk |
| 07:10.21 | AdamB0122 | I've got a cable that comes in from the celing of this room, two cables of this cable are plugged into a punch block |
| 07:10.34 | AdamB0122 | the punch block goes to a large grey device (i sent pictures of this.) |
| 07:10.48 | AdamB0122 | This Block has two RJ45 (Cat-5) connections |
| 07:10.51 | JT | it has a name |
| 07:11.02 | JT | Adit 600, if that's what you mean |
| 07:11.05 | JT | it's a channel bank |
| 07:11.07 | AdamB0122 | Unfortuately, i do not know it, other then abit600 |
| 07:11.11 | AdamB0122 | ok |
| 07:11.20 | AdamB0122 | From the channel bank |
| 07:11.26 | AdamB0122 | i have two ehtercnet cables |
| 07:11.41 | AdamB0122 | One cable goes to another white box below it, with four cards in it |
| 07:11.46 | AdamB0122 | These are our Data T1's |
| 07:12.02 | AdamB0122 | The other cable from the channel bank, plugs into our current phone system. |
| 07:12.08 | *** join/#asterisk menil (n=meni@bzq-179-153-130.static.bezeqint.net) |
| 07:12.18 | JT | gar |
| 07:12.20 | JT | what a mess |
| 07:12.21 | AdamB0122 | That is the cable i currently have plugged into the Asterisk PBX |
| 07:12.24 | AdamB0122 | yea |
| 07:12.30 | JT | have a clear pic of all tyhe cables on the adit 600? |
| 07:12.36 | AdamB0122 | one sec |
| 07:12.51 | JT | and the white box maybe |
| 07:13.20 | AdamB0122 | One moment, lemme go take some more. |
| 07:13.34 | JT | this current pbx |
| 07:13.42 | JT | what other cables are plugged into it? |
| 07:14.13 | *** join/#asterisk Strider86 (n=m_atta_r@82.147.198.212) |
| 07:15.30 | Aces1Up | i'm sure someone here has run a calling card business, anyone know what licenses are required. or anything? |
| 07:15.38 | *** join/#asterisk NirS_ (n=Nir@84.94.120.181.cable.012.net.il) |
| 07:15.56 | JT | Aces1Up: licenses? |
| 07:16.17 | Aces1Up | permits? |
| 07:16.24 | JT | for what? |
| 07:16.51 | Aces1Up | providing a calling card service |
| 07:17.07 | JT | Aces1Up: you could be anywhere on the planet, also |
| 07:17.18 | Aces1Up | in america |
| 07:17.24 | JT | the usa, right |
| 07:17.30 | JT | i have no idea |
| 07:17.47 | JT | not sure why you'd need in licenses specific to doing calling cards |
| 07:18.02 | Aces1Up | not sure why either :) |
| 07:18.19 | Aces1Up | you never know |
| 07:18.56 | JT | just look up telecommunications laws |
| 07:19.06 | JT | maybe all you might need to do is keep good logs, i dunno |
| 07:22.06 | phigan | g'night guys. |
| 07:22.44 | bakermd | Calling cards are primarily Mafia run, so I doubt they require good logs |
| 07:23.59 | AdamB0122 | ok |
| 07:24.00 | JT | bakermd: maybe in america |
| 07:24.03 | AdamB0122 | Here is the "big picture" |
| 07:24.04 | AdamB0122 | http://imageshack.us/?x=my6&myref=http://load.imageshack.us/ |
| 07:24.09 | JT | and even so, the mafia do not have exceptions |
| 07:24.19 | JT | with all this anti terrorism stuff |
| 07:24.36 | JT | AdamB0122: you mean no picture |
| 07:24.44 | AdamB0122 | pretty much |
| 07:24.46 | AdamB0122 | but one sec |
| 07:24.50 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:24.54 | AdamB0122 | http://imageshack.us/?x=my6&myref=http://imageshack.us/ |
| 07:24.59 | JT | ... |
| 07:25.04 | JT | that is not the correct url |
| 07:25.06 | AdamB0122 | damn imageshack |
| 07:25.13 | JT | get real webhosting ;) |
| 07:25.18 | AdamB0122 | bah |
| 07:25.20 | AdamB0122 | yea |
| 07:25.24 | AdamB0122 | lemme go put this on my server |
| 07:27.16 | AdamB0122 | http://digirev.us/stupidphone/ |
| 07:27.22 | AdamB0122 | 25 = big picture |
| 07:27.40 | AdamB0122 | 22 = outside cable > punch block |
| 07:27.53 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
| 07:28.08 | AdamB0122 | 24 = Channel bank (as good as I could get it) |
| 07:28.30 | AdamB0122 | 23 = white box that splits the blue cable from 24 into dual-T1's for our office |
| 07:28.57 | AdamB0122 | on 24, there is a white cable, thats what went to the old phone system. |
| 07:29.05 | *** join/#asterisk vgster (n=vgster@h146154.navonline.net) |
| 07:29.20 | AdamB0122 | Hopefully those help a bit, its about the best i can do with a camera phone :/ |
| 07:29.24 | *** join/#asterisk bspasic (n=bDOTspas@webserver.cardisoft.gr) |
| 07:29.36 | JT | the white box is probably a modem/smartjack |
| 07:29.39 | AdamB0122 | if you have any questions on where stuff does, lemme know |
| 07:29.43 | JT | but an american could better confirm |
| 07:29.49 | JT | orange boxes, what are they? |
| 07:29.59 | AdamB0122 | Punch Blocks |
| 07:30.04 | AdamB0122 | with an orange cover thing |
| 07:30.04 | creativx | tidy wiring |
| 07:30.19 | AdamB0122 | lol, all i have to say is it wasn't me |
| 07:30.19 | JT | where's the pbx? |
| 07:30.28 | AdamB0122 | Old or new? |
| 07:30.33 | bspasic | hi guys, can anyone help me set the extension.conf so when I make originate call from manager API to get autoanswer or answer after ring 0 on the source channel |
| 07:30.37 | creativx | sure blame someone else ;) |
| 07:30.47 | AdamB0122 | rofl |
| 07:30.51 | JT | AdamB0122: er? |
| 07:30.59 | JT | AdamB0122: i thought there was only 1 pbx |
| 07:31.20 | creativx | bspasic: you want autoanswer on the source channel, which would be ex a local phone? |
| 07:31.25 | AdamB0122 | JT : there is the "old phone system" which we all hate, because its expensive to get new phones, its like a IVX128. |
| 07:31.32 | bspasic | yes, local phone |
| 07:31.39 | AdamB0122 | Jt then there is the new Asterisk-based PBX that I can trying to get to work. |
| 07:31.44 | bspasic | this is valid only for the local extensions |
| 07:31.45 | creativx | bspasic: autoanswer is a vendor specific feature. what phone? |
| 07:31.56 | JT | AdamB0122: so there's only one pbx |
| 07:32.01 | bspasic | am using softphone, x-lite |
| 07:32.07 | JT | the other is a pc that doesn't work yet |
| 07:32.10 | creativx | bspasic: x-lite does not support that feature. |
| 07:32.13 | AdamB0122 | ok yes. |
| 07:32.14 | JT | AdamB0122: where is the pbx? |
| 07:32.39 | AdamB0122 | its not in that picture, i can go grab a pic of that if you'd like. |
| 07:32.58 | JT | i want to know what it connects to |
| 07:32.58 | bspasic | what softphone supports this future? |
| 07:33.16 | AdamB0122 | White cable goes to a Patch Panel |
| 07:33.23 | AdamB0122 | which was wired by a horriable company |
| 07:33.30 | creativx | bspasic: havent heard of any that do. its not in the SIP rfc |
| 07:33.35 | AdamB0122 | and will only result in a bunch of "omfg, wtf?" comments |
| 07:34.15 | AdamB0122 | from that patch panel, it plugs into the IVX128 via huge connector, (much like the orange one in pic 24) |
| 07:34.43 | JT | AdamB0122: what ELSE connects to the PBX? |
| 07:34.50 | bspasic | aha, thanks, but i see on many forums that the autoanswer can be configured, just can't make it work |
| 07:35.10 | bakermd | drwx-w---- How do I chmod a file to equal this? |
| 07:35.12 | creativx | bspasic: x-lite has a feature to autoanswer ALL incoming calls yes. but that is not equal to the answer-after: SIP header |
| 07:35.14 | bakermd | (brain cramp) |
| 07:35.17 | AdamB0122 | Two more of the Orange huge connectors, which go BACK to the Patch Panel on a different layer, and connects the the current Cat-3 Wired phones |
| 07:35.47 | JT | AdamB0122: what on earth connects to the big cables from the Adit 600? |
| 07:35.55 | AdamB0122 | ( which we'll be selling once asterisk is working ) |
| 07:36.08 | AdamB0122 | The Orange Cable on the Adit600? |
| 07:36.14 | bspasic | so there is no way of making call and the first ring to be autoanswered? |
| 07:36.16 | JT | yes |
| 07:36.19 | JT | it is big, i assume |
| 07:36.23 | AdamB0122 | yes. |
| 07:36.30 | AdamB0122 | good 3-4" connector. |
| 07:36.39 | AdamB0122 | and it does to the Punch block, in picture 22 |
| 07:36.52 | AdamB0122 | which as you can see, has only 2 connections going IN |
| 07:37.01 | AdamB0122 | (the little blue cables) |
| 07:37.36 | JT | AdamB0122: the adit 600, which one is it in the big picture? |
| 07:38.16 | AdamB0122 | one moment |
| 07:38.49 | creativx | bspasic: not with x-lite |
| 07:39.11 | creativx | bspasic: it works on some hardphones, like ip10s. but the sip firmware generally sucks in those, so stay clear |
| 07:39.11 | AdamB0122 | refresh and open 25-2 |
| 07:39.11 | bspasic | ok, thanks |
| 07:39.47 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 07:39.53 | JT | AdamB0122: thought so, so what is the cream thing to the top left? |
| 07:40.04 | waptaxi | hi! I have Panasonic TDA connected with Asterisk through E1. Now I can dial local Panasonic extension with X-Lite, hear busy tone and press 5 to begin spy on this extension. But how can I simulate this behaviour with dialplan? Dial(Zap/g1/121) will not return unless channel is answered ot busy, and I can't send any digits |
| 07:40.36 | AdamB0122 | It says IDSN, but it doesn't goto anything, and there's never been anything plugged into it since we moved into this building |
| 07:40.47 | JT | ok |
| 07:41.10 | AdamB0122 | to my knowledge, it was probably someone's who was here before us, and they just didn't remove it |
| 07:41.12 | creativx | damnit i need to set up more fax modems.. sigh |
| 07:41.29 | JT | AdamB0122: are you seriously telling me only a couple of pairs in that fat orange cable from the Adit 600 are used? |
| 07:41.41 | AdamB0122 | Yea |
| 07:41.49 | AdamB0122 | Two of them.... |
| 07:42.31 | AdamB0122 | Two little blue cables from the Telcom line in, to that punch block |
| 07:42.38 | JT | AdamB0122: how many T1s from the telco do you have? do you have combined data and voice T1s? |
| 07:42.44 | AdamB0122 | no |
| 07:42.48 | AdamB0122 | 2 dedicated DataT1's |
| 07:42.59 | AdamB0122 | and 1 Voice T1 |
| 07:42.59 | AdamB0122 | both are dedicated to their purpose |
| 07:42.59 | *** join/#asterisk kje (n=kje@62-99-209-38.c-vzollerg.xdsl-line.inode.at) |
| 07:43.26 | kje | seen chris |
| 07:43.52 | JT | i am confused, you say terminals from the fat telco connector cable on the Adit 600 connect to the telco, yet what do the cat5 cables do? |
| 07:44.29 | AdamB0122 | the White and Blue Cat5 Cables? |
| 07:44.35 | JT | yes |
| 07:44.44 | JT | they can't all be connecting to the telco |
| 07:44.47 | JT | that makes no sense |
| 07:44.53 | AdamB0122 | no |
| 07:44.59 | JT | as the telco connector should have local FXS ports on it |
| 07:45.08 | JT | nothing from the telco connector should connect to the telco |
| 07:45.11 | AdamB0122 | White Cable goes to PBX |
| 07:45.18 | JT | they should connect to analogue lines at your location |
| 07:45.32 | AdamB0122 | Blue Cable goes the the White Box below it, which split into the 2 DataT1's |
| 07:45.57 | AdamB0122 | The top of the Channelbank has the FXS ports on it |
| 07:46.09 | AdamB0122 | or its got 3 cards in it anyway |
| 07:46.24 | JT | and where do they come out? the telco connector |
| 07:46.28 | *** join/#asterisk CoolGuy21 (n=Tilt@cpe-76-81-1-73.socal.res.rr.com) |
| 07:46.43 | AdamB0122 | I dont know, the cards pluginto the channel bank |
| 07:46.52 | CoolGuy21 | how do i check what zaptel version i have? |
| 07:46.58 | bakermd | my god - the permissions on a call file have to be juuuuust right.. not too much, not too little |
| 07:47.04 | JT | the 2 data T1s have nothing to do with the channel bank, i assume |
| 07:47.21 | AdamB0122 | They plug into the channel bank, via the blue cable |
| 07:47.31 | AdamB0122 | one sec.. lemme do a visio drawing to clear it up a bit |
| 07:47.43 | JT | AdamB0122: well the telco connector has the FXS ports, there should be no T1 over the telco connector |
| 07:48.40 | AdamB0122 | uh, I dont quite understand |
| 07:49.25 | creativx | its times like these im glad i only have to worry about that single cat6 cable going into my asterisk server :) |
| 07:49.35 | JT | on the white box, CPE 2 and CPE 3 are the data T1s, right? |
| 07:49.45 | AdamB0122 | yes |
| 07:50.01 | AdamB0122 | Img uploaded |
| 07:50.17 | AdamB0122 | img020.jpg, its the top of the adit600, a bit blurry, but its hard to reach way up there |
| 07:51.08 | AdamB0122 | first card is the VoiceT1 I'd assume, since its different then the other Two DataT1'a |
| 07:51.09 | [hC] | i thought there was an argument you could pass to dial() to NOT send callerid? |
| 07:51.13 | AdamB0122 | 's |
| 07:51.24 | JT | AdamB0122: one last time, do you know EXACTLY what the lines coming from the big orange cable from the Adit 600 are patched to? |
| 07:51.58 | JT | i already saw the top of the adit 600 earlier |
| 07:52.00 | jerlique | I'm having problems with * listening to DTMF from a channel bank, the sip debug says "unauthorised", any hints? |
| 07:52.18 | AdamB0122 | the big orange calbe has a giant connector on it, , and on the other end of the giant connector is the punch block, with the two little cables going to it. |
| 07:52.38 | Strider86 | hi .. i am a student interning at a networking solutions company and my boss wants me to try this asterisk .. now i need some help in it who can help me directly?? .. thanx |
| 07:52.50 | JT | AdamB0122: do you know where those cables go? |
| 07:53.00 | AdamB0122 | the Two small cables? |
| 07:53.02 | creativx | Strider86: try the internets |
| 07:53.19 | AdamB0122 | Strider86 > yea, there's lots of walkthroughs that can show you the basics of stuff on the web |
| 07:53.34 | JT | AdamB0122: yes. |
| 07:54.22 | AdamB0122 | JT > They goto the celing. Beyond that I have no idea, we dont have another telecom room in this building though, so if their connecting to anything else, I have no idea. |
| 07:54.44 | JT | AdamB0122: i am confused because i thought you said they connected to the telco, which makes no sense |
| 07:55.03 | JT | i think i understand the setup |
| 07:55.09 | JT | tell me |
| 07:55.11 | AdamB0122 | I assume they goto telco (by this you mean XO Communications) |
| 07:55.50 | JT | there's 3 RJ-45s conne on the white box. one for voice T1, 2 for data. |
| 07:55.59 | JT | WHAT ELSE connects to the white box |
| 07:56.15 | JT | and please don't assume you know where wires go if you don't, it causes a lot of confusion |
| 07:56.18 | AdamB0122 | one moment, let me go double check, rather then trying to rmember at 3am. |
| 07:57.51 | AdamB0122 | ok |
| 07:58.02 | AdamB0122 | 1) fuck me, I've confused the shit out of you |
| 07:58.17 | AdamB0122 | sorry about that |
| 07:58.23 | AdamB0122 | Telecom goes INTO this box |
| 07:58.33 | AdamB0122 | (FAR right cable on 023) |
| 07:58.45 | AdamB0122 | then, on the left side |
| 07:58.52 | AdamB0122 | i have Blue, which is VoiceT1 |
| 07:59.01 | AdamB0122 | a Grey, which is DataT1a |
| 07:59.13 | AdamB0122 | another grey, which is the ground for the box |
| 07:59.22 | JT | that makes 4 cables |
| 07:59.24 | AdamB0122 | and a third Grey, which is DataT1B |
| 07:59.25 | JT | what about power? |
| 07:59.40 | JT | 5 cables |
| 07:59.46 | JT | + power? |
| 07:59.46 | [hC] | how might i block outgoing caller id (*67) by using the dialplan? |
| 08:00.15 | JT | [hC]: Dial/technology/*67${EXTEN} ? |
| 08:00.26 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 08:00.29 | coldsteal | hello |
| 08:00.45 | Strider86 | hey .. 1 more question sorry for troubling u popl .. for downloading through CVS i have this cmd here export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot .. but it gives me arror that unknown host cvs.digium.com .. is it problem with proxy or internet or has the host been changed? |
| 08:00.52 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:00.56 | [hC] | Huh. I dont think I can send calls out my PRI that way. Maybe I/ll try. I was actually looking for a way to null caller id values before handing off to the provider. |
| 08:01.27 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:01.39 | AdamB0122 | there is nothing that looks like obvious power |
| 08:01.44 | JT | [hC]: is *67 a prefix? |
| 08:01.52 | coldsteal | okay so im trying t get my friend to connect to my asterisk box and when he does and i call his ext or he calls mine we cant hear each other our mics or speakers dont work... |
| 08:01.53 | AdamB0122 | but, the unit DOES have power, (i opened it, and there is tons of LEDS in it) |
| 08:01.56 | JT | AdamB0122: it has to be powered somehow |
| 08:01.57 | [hC] | JT: Typically, yes, for blocking outbound caller id. |
| 08:02.16 | JT | AdamB0122: oh, mind reading off stuff from inside, or photo |
| 08:02.18 | [hC] | JT: All I want to do is 'not send caller id' on the next call, using the dial plan. |
| 08:02.18 | JT | anyway |
| 08:02.25 | JT | pretty sure that's the smart jack, AdamB0122 |
| 08:02.43 | AdamB0122 | Yea, I can get a photo of the cards on the inside if you'd like |
| 08:02.50 | JT | [hC]: prefix could work, but sending a null callerid will work on PRI |
| 08:03.01 | JT | AdamB0122: it will be a triple SHDSL modem unit |
| 08:03.18 | AdamB0122 | ok |
| 08:03.34 | JT | it won't work without power though ;) |
| 08:03.42 | AdamB0122 | I dont know how to explain that one |
| 08:03.48 | [hC] | JT: I just tried doing a simple SetCallerID("") and i still got caller id. |
| 08:04.06 | JT | [hC]: that command was deprecated in 1.2, removes in 1.4 |
| 08:04.06 | AdamB0122 | It does have power somehow, I just dont know how. |
| 08:04.17 | AdamB0122 | If I plug the cable thats the voice T1 from there |
| 08:04.23 | [hC] | JT: I'm using 1.2 .. I t should still work. |
| 08:04.35 | AdamB0122 | into the AsteriskPBX |
| 08:04.49 | JT | best to move to the new syntax anyway, [hC] |
| 08:04.50 | AdamB0122 | and then setup the zapata / zaptel in a PRI setup |
| 08:04.58 | AdamB0122 | It should work |
| 08:05.00 | coldsteal | http://rafb.net/p/7seZhm60.html |
| 08:05.01 | JT | [hC]: is it sending according to pri intense debug? |
| 08:05.11 | AdamB0122 | or thats the "proper setup? |
| 08:05.20 | JT | AdamB0122: you shouldn't do that, |
| 08:05.25 | tuzhila | -- Called 47182877379666336199417493354@298685513150 |
| 08:05.25 | tuzhila | Jul 24 11:37:04 WARNING[16372]: chan_sip.c:9761 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Are" <sip:2986855@sipnet.ru>;tag=as0ab8dcee' |
| 08:05.25 | tuzhila | <PROTECTED> |
| 08:05.25 | tuzhila | <PROTECTED> |
| 08:05.26 | AdamB0122 | ok. |
| 08:05.30 | JT | tuzhila: stop it. |
| 08:05.33 | JT | ~pb |
| 08:05.33 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 08:05.33 | tuzhila | please, tell me, why i have got such error: |
| 08:05.37 | AdamB0122 | yea, dont do that again. |
| 08:05.39 | AdamB0122 | nopaste. |
| 08:05.41 | JT | tuzhila: please don't paste here |
| 08:05.45 | AdamB0122 | rafb.net or w/e its called |
| 08:05.46 | JT | pastebin.ca |
| 08:05.46 | tuzhila | ok |
| 08:06.02 | tuzhila | tell me, why i've got this error? |
| 08:06.07 | JT | no, i'm busy |
| 08:06.20 | [hC] | JT: Ok, So now I'm using Set(CALLERID()="") and now i see correct caller id, whereas before it showed my pilot numer. |
| 08:06.39 | JT | [hC]: you should avoid using quotation marks |
| 08:06.49 | AdamB0122 | JT > If I've got the * Box plugged into the that white box, where the T1 is, what method should i use to setup the * system |
| 08:07.10 | AdamB0122 | so i dont have to bug you anymore, now that my brain properly wrapped around wtf is going on with that back closet. |
| 08:07.26 | tuzhila | hey, help me. please |
| 08:07.40 | AdamB0122 | tuzhila look at your error. |
| 08:07.43 | tuzhila | what does such errorr mean? |
| 08:07.45 | coldsteal | anyone? |
| 08:07.47 | AdamB0122 | .... |
| 08:07.56 | *** part/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 08:07.58 | coldsteal | okay so im trying t get my friend to connect to my asterisk box and when he does and i call his ext or he calls mine we cant hear each other our mics or speakers dont work... |
| 08:07.59 | AdamB0122 | "Forbidden - wrong password on authentication for INVITE" might be a pretty big clue. |
| 08:08.02 | coldsteal | http://rafb.net/p/7seZhm60.html |
| 08:08.02 | creativx | tuzhila: here's a wild guess: wrong password |
| 08:08.09 | AdamB0122 | lol |
| 08:08.20 | AdamB0122 | Google is your friend tuzhila |
| 08:08.22 | [hC] | JT: So I see whats happening here. When i send no caller id, it sends my PRI pilot number as caller id, instead of 'nothing' |
| 08:08.26 | [hC] | (the telco is doing this) |
| 08:09.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:09.16 | NirS | anybody has an idea how Asterisk calculates the proper Digest-Response for SIP registration ? |
| 08:09.40 | creativx | [hC]: my itsp does the same |
| 08:09.49 | JT | AdamB0122: i have worked out how all that crap in the cupboard works |
| 08:10.00 | [hC] | creativx: find a way around that? Im about to just try another one of my itsp's. |
| 08:10.02 | JT | i was about to say, but got distracted |
| 08:10.13 | JT | also tuzhila was annoying me |
| 08:10.15 | JT | heh |
| 08:10.21 | creativx | [hC]: nope. my itsp forces me to use the cids that is valid for me, e.g one of my 100 numbers |
| 08:10.23 | AdamB0122 | yea. lol |
| 08:10.33 | AdamB0122 | omfg |
| 08:10.34 | JT | AdamB0122: the Adit 600 must be kept in line |
| 08:10.38 | JT | if you don't |
| 08:10.45 | Uatec | [hC], my telco are just weird, they allow me to set all the clid on online, just the last 3 digits on another and none on two more |
| 08:10.45 | JT | you will lose 2-4 phone lines |
| 08:10.48 | tuzhila | JT, sorry |
| 08:11.05 | AdamB0122 | ok |
| 08:11.12 | AdamB0122 | I'd like to keep phone lines |
| 08:11.24 | JT | AdamB0122: those wires going into the ceiling are those lines |
| 08:11.38 | JT | the Adit 600 is pulling 2-4 timeslots off |
| 08:11.53 | JT | it's over equipped, considering it has 3 FXS card, probably only 1 is in use |
| 08:12.05 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 08:12.07 | [hC] | Haha. I just sent it out another ITSP and presumably got their pilot number as my CID. |
| 08:12.08 | JT | AdamB0122: the rest of the CAS timeslots go to the pbx |
| 08:12.20 | JT | AdamB0122: you must determine what timeslots are left for you to work with |
| 08:12.28 | JT | to configure asterisk correctly |
| 08:12.53 | JT | the data t1s have nothing to do with this and connect to the white box because they need SHDSL modems too |
| 08:13.06 | JT | no-one delivers real T1 circuits anymore ;) |
| 08:13.27 | coldsteal | !help |
| 08:13.34 | coldsteal | ~help |
| 08:13.54 | JT | ~question |
| 08:13.55 | jbot | methinks question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 08:14.13 | AdamB0122 | lol |
| 08:14.15 | AdamB0122 | Ok. |
| 08:14.28 | JT | (real T1s have no range) |
| 08:15.09 | JT | AdamB0122: either look at the config of the pbx, ask the telco, or look at the Adit 600 config |
| 08:15.12 | JT | heh |
| 08:15.20 | JT | for determining timeslots |
| 08:15.33 | JT | or trial and error, but that's not desirable |
| 08:15.34 | AdamB0122 | hm. since I dont have access to either of the config |
| 08:15.38 | AdamB0122 | guess I'm calling telco |
| 08:15.46 | JT | AdamB0122: i think the Adit 600 is all dip switches |
| 08:15.50 | AdamB0122 | yea, trial and error would be lovely if i could avoid it |
| 08:16.04 | AdamB0122 | yea, there's a ton of switches on the top |
| 08:16.08 | JT | :) |
| 08:16.18 | AdamB0122 | lemme google reading those first |
| 08:16.28 | coldsteal | okay so im trying t get my friend to connect to my asterisk box and when he does and i call his ext or he calls mine we cant hear each other our mics or speakers dont work... here is paert of my extensions.confhttp://rafb.net/p/7seZhm60.html |
| 08:16.46 | JT | the photo is too fuzzy for me to read the switches |
| 08:16.54 | AdamB0122 | yea |
| 08:17.01 | coldsteal | JT: hello |
| 08:17.11 | AdamB0122 | and there is no way I'm gonna get a better shot, we dont have any stronger cameras here |
| 08:17.15 | JT | coldsteal: over what tecnology, what network setup, etc |
| 08:17.16 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:17.35 | coldsteal | JT: what do you mean |
| 08:17.37 | JT | AdamB0122: more light ;) |
| 08:17.52 | coldsteal | its Asterisk and softphones |
| 08:17.56 | JT | coldsteal: zap, h.323, sccp, mgcp, isdn2, iax2, sip |
| 08:18.00 | coldsteal | sip |
| 08:18.02 | JT | coldsteal: Internet, nat, etc |
| 08:18.07 | JT | explain your setup. |
| 08:18.11 | coldsteal | internet |
| 08:18.13 | JT | ... |
| 08:18.15 | JT | explain |
| 08:18.18 | JT | not one word answer |
| 08:18.22 | Uatec | do you know what is a really crap phone? |
| 08:18.33 | Uatec | the sip client, built into the xda exec on windows mobile 6 |
| 08:18.34 | JT | Uatec: granstream! |
| 08:18.35 | creativx | Uatec: yes I do |
| 08:18.40 | creativx | swissvoice ip10s!! |
| 08:18.50 | Uatec | aastra 9122i |
| 08:18.51 | Uatec | eurgh |
| 08:19.00 | creativx | lets invert the question |
| 08:19.06 | creativx | do you know what is a really good phone? |
| 08:19.07 | Uatec | some place on tv last night was using the aastra 9122is |
| 08:19.12 | Uatec | eurgh |
| 08:19.13 | Uatec | umm |
| 08:19.21 | Uatec | iam using linksys spa922s |
| 08:19.23 | Uatec | they're pretty good |
| 08:19.29 | Uatec | there are a few little niggles i have with them |
| 08:19.30 | Uatec | but they're not bad |
| 08:19.35 | creativx | are they stable enough to be used in a business enviroment |
| 08:19.43 | Uatec | i am using them in a business environment |
| 08:19.47 | Uatec | they are definately stable |
| 08:19.50 | *** join/#asterisk CM3_1_2_632 (n=CM3_1_2_@n219076080194.netvigator.com) |
| 08:20.00 | CM3_1_2_632 | hello |
| 08:20.00 | Uatec | they don't do distinctive ring as i expected :( |
| 08:20.08 | creativx | whats distinctive ring |
| 08:20.15 | coldsteal | i have a Asterisk server in my basement and its conected to broadvoice and i have a laptop thats conected via a sip softphone to my Asterisk box internaly i would like my friend acrost the internet to connect to my Asterisk box at ext 21 i have fowarded udp port 5060 |
| 08:20.19 | creativx | different tones for different callers? heh |
| 08:20.28 | Uatec | indeed |
| 08:20.37 | creativx | i see |
| 08:20.42 | Uatec | but i didn't want it for different callers... i wanted to emulate different lines on the same phone |
| 08:20.49 | coldsteal | JT: does that explain it? |
| 08:20.51 | creativx | we were actually considering removing all ring tones |
| 08:20.53 | Uatec | we have a mainline and a support line |
| 08:20.56 | Uatec | removing? |
| 08:20.57 | Uatec | Why? |
| 08:21.01 | creativx | silence in the office |
| 08:21.07 | creativx | instead let it just be a popup on the screen |
| 08:21.09 | JT | coldsteal: you need to forward udp 10000-20000 also |
| 08:21.13 | creativx | lower the stress level |
| 08:21.17 | Uatec | creativx, what if you're not at your pc |
| 08:21.17 | coldsteal | JT: i did |
| 08:21.18 | Uatec | ? |
| 08:21.20 | JT | coldsteal: and you must set externip= and localnet= on asterisk |
| 08:21.27 | creativx | if youre not at your pc you have no need to answer the phone |
| 08:21.28 | JT | coldsteal: oh, you just didn't emntion it, but ok |
| 08:21.36 | JT | coldsteal: you must set canreinvite=no |
| 08:21.37 | JT | in sip.conf |
| 08:21.40 | Uatec | what if you're just having a conversation with someone else |
| 08:21.41 | Uatec | ? |
| 08:21.44 | Uatec | anyway, it's your call |
| 08:21.51 | creativx | Uatec: we have great moh ;) |
| 08:22.03 | Uatec | you can provision the spa922s with an xml file on a tftp server |
| 08:22.08 | Uatec | simple as |
| 08:22.31 | coldsteal | what would i put in externip= |
| 08:22.41 | Uatec | creativx> lol |
| 08:22.49 | JT | your external internet accessible IP address, coldsteal |
| 08:23.19 | Uatec | OMFG |
| 08:23.23 | Uatec | this phone is crap |
| 08:23.28 | creativx | Uatec: we're just trying to think in new ways of using telephony |
| 08:23.35 | creativx | its all just audio in/out anyways |
| 08:23.36 | coldsteal | okay i did all of that |
| 08:23.39 | Uatec | "Cannot complete the call. The signal my be unavailable or the phone number may not be valid." |
| 08:23.40 | Uatec | WTF? |
| 08:24.01 | Uatec | creativx, yeah, that's what we're doing. but i think we all here agree that phones ring for a reason |
| 08:24.16 | Uatec | i don't think the first one did, but it was a very quick addition to the concept of a telephone |
| 08:24.18 | Uatec | for a reason |
| 08:24.51 | Uatec | hey, you know what is weird? |
| 08:25.04 | Uatec | some of my users are reporting a delay follows by a click, before their call is connected |
| 08:25.19 | Uatec | when picking up incoming calls |
| 08:25.26 | Uatec | incomming over misdn |
| 08:25.48 | JT | b410p? |
| 08:25.54 | Uatec | yeah |
| 08:26.06 | JT | just checking ;) thought that was the case |
| 08:26.37 | Uatec | :| |
| 08:26.38 | creativx | Uatec: the reason is not the issue.. the issue is that its a stress factor with ringing phones :) |
| 08:27.02 | Uatec | put a ring tone of twittering birds on it |
| 08:27.18 | Uatec | how many calls are you looking of handling per day on your silent phone system? |
| 08:27.23 | coldsteal | JT: i did what u said i shoud |
| 08:27.34 | creativx | our normal call volume Uatec.. which I have no idea what is |
| 08:28.05 | Uatec | oh |
| 08:28.35 | creativx | our clients are real estate agents |
| 08:28.49 | JT | coldsteal: okay |
| 08:28.51 | creativx | so they have lots of dumb questions they like to call about |
| 08:28.55 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 08:29.06 | coldsteal | JT: idk if that will fix the soind and stuff |
| 08:29.17 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 08:29.17 | coldsteal | *sound |
| 08:29.18 | JT | coldsteal: why don't you find out? |
| 08:29.43 | coldsteal | lol well my friends soft phoe client isnt running |
| 08:29.48 | *** join/#asterisk gzero (n=gzero@81.175.82.2) |
| 08:29.56 | JT | coldsteal: if it still doesn't work, you will have to come back ant put your configs up on pastebin.ca for us to see |
| 08:29.59 | Uatec | well we have 300+ calls per day, and it's not an issue with stress |
| 08:30.03 | Uatec | and we deal with.... |
| 08:30.04 | JT | and any console messages |
| 08:30.04 | Uatec | users |
| 08:30.06 | Uatec | eurgh |
| 08:30.24 | creativx | Uatec: real estate agents are worse than users.. trust me |
| 08:30.25 | creativx | =) |
| 08:30.35 | creativx | i would assume we have somewhat around that volume too |
| 08:30.50 | coldsteal | JT: parts of my extensions.conf and sip.conf http://rafb.net/p/7seZhm60.html |
| 08:31.28 | coldsteal | idk i just copyany pasted them to make a new ext so idk if something is wrong |
| 08:31.46 | Uatec | we have some estate agents on our books |
| 08:31.48 | JT | coldsteal: your sip.conf is missing externip and localnet |
| 08:31.55 | Uatec | fortunately i'm a software developer, not a support lackey |
| 08:32.05 | JT | coldsteal: can you stop saying "idk" please ;) |
| 08:32.05 | creativx | hehe |
| 08:32.16 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-240.net.novis.pt) |
| 08:32.29 | coldsteal | JT: thats not my full sip.conf |
| 08:32.37 | JT | coldsteal: then what's the point? |
| 08:32.41 | JT | if it's not the full thing |
| 08:32.55 | Uatec | that's only about 150 incomming calls a day |
| 08:33.07 | coldsteal | JT: lol okay |
| 08:33.19 | Uatec | JT, have you had experience with that problem? with the b410p? |
| 08:33.24 | coldsteal | JT: if i cant get it to work ill come back |
| 08:33.30 | JT | Uatec: yes, it doesn't work with bristuff |
| 08:33.35 | Uatec | ... |
| 08:33.43 | JT | as far as has been tested |
| 08:33.47 | JT | misdn is utter rubbish |
| 08:33.50 | JT | it's a valid concern |
| 08:33.52 | Uatec | it's not rubbish |
| 08:33.54 | Uatec | it's just not good |
| 08:33.55 | JT | it is |
| 08:34.08 | JT | dtmf recognition, who needs that |
| 08:34.17 | JT | proper nt mode support, nah, what idiot would need that ;) |
| 08:34.26 | JT | it's rebranded isdn 4 linux |
| 08:34.32 | Uatec | just cos you don't doesn't mean that nboody does |
| 08:34.36 | JT | because everyone hated isdn4linux so much |
| 08:34.47 | JT | because it sucked |
| 08:34.54 | Uatec | so it's better than isdn4linux |
| 08:34.55 | Uatec | the point is |
| 08:34.58 | Uatec | we have a b410p |
| 08:35.01 | JT | it IS isdn4linux |
| 08:35.03 | Uatec | which was designed to work with misdn |
| 08:35.05 | JT | they just renamed it |
| 08:35.43 | Uatec | so that's what i'm using |
| 08:35.58 | Uatec | lol, a user just came in saying "my laptop has run out of battery and it wont charge up" |
| 08:36.08 | JT | and you wanted to know what the problem with it was, and i told you ;) |
| 08:36.08 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 08:36.15 | Uatec | actually, the 240v power cable has bare wires in it |
| 08:36.17 | CM3_1_2_632 | Uatec: lmao |
| 08:36.21 | JT | nice |
| 08:36.31 | Uatec | it shorted out the plug we plugged it in to |
| 08:36.39 | Uatec | i'm surprised it didn't short out his office |
| 08:36.41 | Uatec | probably did |
| 08:36.57 | Uatec | JT, that's not the problem |
| 08:37.03 | Uatec | the problem is within misdn |
| 08:37.10 | JT | another problem is the company that makes the b410p can't even buy a circuit to test it on |
| 08:37.13 | Uatec | misdn wasn't designed to have a pause before calls |
| 08:37.15 | JT | but that's another matter |
| 08:37.19 | Uatec | they have a test circuit |
| 08:37.47 | JT | Uatec: if a piece of hardware only works with rubbish software, the hardware is only as good as that software |
| 08:37.52 | JT | Uatec: no, they have a simulator |
| 08:38.03 | Uatec | oh |
| 08:38.03 | Uatec | well |
| 08:38.04 | JT | Uatec: you cannot buy etsi bri in the us |
| 08:38.18 | Uatec | my boss has bought a b410p |
| 08:38.24 | *** join/#asterisk guillote_GNU (n=guillote@host111.190-30-66.telecom.net.ar) |
| 08:38.26 | Uatec | and it wasn't cheap |
| 08:38.36 | Uatec | and we're not getting rid of it |
| 08:38.38 | *** join/#asterisk kkn088 (n=kikoun@84.4.50.39) |
| 08:38.38 | Uatec | becuase we're using it |
| 08:38.42 | Uatec | and it works well enough most of the time |
| 08:38.45 | JT | and you're stuck with it, yes i've heard this before |
| 08:38.51 | Uatec | i only have 2 problems with it |
| 08:38.55 | kkn088 | hi |
| 08:39.01 | JT | you'll find almost all of your problems are related to misdn |
| 08:39.02 | Uatec | round robin dialling doesn't work on all channels (amateur bug) |
| 08:39.16 | JT | file bugs on the digium bug tracker |
| 08:39.31 | Uatec | i have |
| 08:39.43 | Uatec | i'm in open communication with their support about it too |
| 08:39.43 | JT | group dialling doesn't work at all on NT ports |
| 08:39.50 | Uatec | but their support is... crap |
| 08:39.54 | Uatec | i know it doesn't work |
| 08:40.00 | JT | the readme says "no-one will ever need this" or something like that, the misdn readme |
| 08:40.00 | Uatec | i had to write a macro to do the job instead |
| 08:40.11 | JT | the guys who write misdn are morons |
| 08:40.13 | JT | beronet |
| 08:40.21 | Uatec | so i hear |
| 08:40.36 | darkskiez | dont be mean |
| 08:40.37 | Uatec | OMFG, why is it so hard to find files |
| 08:40.39 | Uatec | i search for a file |
| 08:40.45 | Uatec | and i get 101 sites which have it |
| 08:40.46 | JT | darkskiez: ? |
| 08:40.47 | Uatec | but wont show it to me |
| 08:40.51 | Uatec | they will just show me the difference |
| 08:40.54 | Uatec | on the versions they have |
| 08:41.00 | Uatec | or notes somebody made about it |
| 08:41.04 | Uatec | but never the actual god damned file |
| 08:41.07 | darkskiez | JT: meanie |
| 08:41.23 | JT | darkskiez: truth hurts |
| 08:41.33 | darkskiez | doesnt hurt me |
| 08:41.39 | JT | heh |
| 08:43.48 | AdamB0122 | hey JT |
| 08:43.56 | JT | hi |
| 08:44.15 | AdamB0122 | ok, I've setup my zap files to match this |
| 08:44.16 | AdamB0122 | http://rafb.net/p/2vIpJV95.html |
| 08:44.21 | AdamB0122 | except a real context |
| 08:44.29 | *** join/#asterisk lsodi (n=lsodi@195.80.124.193) |
| 08:44.46 | AdamB0122 | It goes from Telco > White Box > Cannelbank > AsteriskPBX |
| 08:44.52 | JT | wrong signalling |
| 08:44.56 | JT | fxsks |
| 08:44.58 | AdamB0122 | ok |
| 08:45.31 | Uatec | WTF? you can send text down an open misdn channel |
| 08:45.33 | Uatec | WTF for? |
| 08:46.06 | lsodi | greetings, has any one used mor + hud/hud lite, or is there better call manager software than HUD? |
| 08:46.10 | JT | Uatec: how? |
| 08:46.12 | AdamB0122 | ok |
| 08:46.22 | AdamB0122 | Its switch to that signalling |
| 08:46.42 | Uatec | from the cli you can do misdn send display misdn/1-1 "Hello world." |
| 08:46.46 | AdamB0122 | Now, when I call in, I dont get anything on the asterisk PBX, and i dont get anything in the phone |
| 08:46.52 | AdamB0122 | so i assume my channels are still off |
| 08:46.57 | JT | Uatec: probably for bri phones |
| 08:47.11 | JT | AdamB0122: have you checked if the pri is UP? |
| 08:47.31 | AdamB0122 | does that could a a PRI, or a Zap? |
| 08:47.43 | AdamB0122 | oh hm. |
| 08:47.48 | AdamB0122 | Zap show status alarm's red |
| 08:48.00 | Uatec | :\ |
| 08:48.04 | Uatec | i called myself over misdn |
| 08:48.14 | Uatec | i.e. out on online, in on another |
| 08:48.17 | AdamB0122 | anyway to tell what its throwing that for? |
| 08:48.25 | Uatec | i didn't get anything back on the cli when i sent myself a message |
| 08:48.27 | Uatec | LAME |
| 08:49.44 | creativx | send yourself an email instead and save the hassle |
| 08:50.13 | *** join/#asterisk voltagex (n=voltagex@121-79-12-198-dsl.ispone.net.au) |
| 08:50.49 | voltagex | hi, how can I get asterisk to support text messages in SIP? I'm getting SIP/2.0 415 Unsupported Media Type |
| 08:51.13 | Uatec | creativx, i could just think it |
| 08:51.14 | AdamB0122 | JT : since its a FXS type, wouldn't it show up in zap show status rather then pri show status? |
| 08:51.38 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 08:52.13 | creativx | Uatec: thats too simple! |
| 08:54.54 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 08:54.57 | AdamB0122 | JT / Anyone : Will the a Zap T1 card throw the RED code if the channels aren't configured right? |
| 08:55.09 | AdamB0122 | IE: if channel 1 is for something else, or something like that |
| 08:55.14 | salvatore2 | 5+ |
| 08:56.27 | ghenry | Recommendations: Polycom IP330 vs Aastra 55i IP vs Snom 320 ?? Aastra if I want a big screen, otherwise Polycom? |
| 08:56.32 | AdamB0122 | hm... I've got it set to use only 1 channel right now, channel 10. pretty sure it'd be an open channel |
| 08:58.09 | AdamB0122 | anyone know what could cause that? |
| 09:02.06 | Uatec | creativx, i am a big fan of simple |
| 09:02.09 | lsodi | Adam: is line correctly bulid up, 1,2 tx and 4,5 rx in RJ45 |
| 09:02.11 | Uatec | i used to have a car |
| 09:02.16 | Uatec | now i have one like in the flintstones |
| 09:02.29 | Uatec | less parts, you see4 |
| 09:02.31 | Uatec | -4 |
| 09:03.16 | AdamB0122 | lsodi > yes, I ran it through a cable tester to be sure |
| 09:04.08 | creativx | Uatec: well thats a good approach being a developer |
| 09:04.30 | JT | AdamB0122: yes i meant t1, not pri, sorry |
| 09:04.37 | JT | and red means Layer 1 failure |
| 09:04.43 | JT | no physical link |
| 09:04.57 | AdamB0122 | hm |
| 09:04.59 | AdamB0122 | thats odd. |
| 09:05.06 | AdamB0122 | Its plugged into the channel back right now |
| 09:05.10 | CoolGuy21 | hi |
| 09:05.11 | JT | your cable could be wrong |
| 09:05.30 | CoolGuy21 | i installed rhino using the instructions im getting this running ztcfg |
| 09:05.31 | CoolGuy21 | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 09:05.36 | AdamB0122 | checked it twice already... its just a standard Cat5e line right, one that a laptop could connect through? |
| 09:05.51 | JT | AdamB0122: is that the same cable the pbx used? |
| 09:06.30 | AdamB0122 | no, this one is a standard cat5 cable i had from my office, I've left all the old wiring in place |
| 09:06.52 | Uatec | creativx, i'm not sure if you're being sarcastic or serious |
| 09:07.03 | JT | AdamB0122: wrong cable. |
| 09:07.11 | JT | AdamB0122: you need a T1 crossover cable. |
| 09:07.32 | AdamB0122 | From the channel bank to to * box? |
| 09:07.36 | JT | yes |
| 09:07.47 | JT | no wonder it doesn't work |
| 09:07.57 | AdamB0122 | does it use Standard Cat1E cable, with just a wierd pin alignment? |
| 09:08.10 | AdamB0122 | or does it use some other type of cabling? |
| 09:08.19 | JT | you can use cat 5 |
| 09:08.23 | JT | 1 to 2 |
| 09:08.24 | JT | 4 to 5 |
| 09:08.35 | Uatec | you can't google for cat1e |
| 09:08.40 | AdamB0122 | yea |
| 09:08.41 | JT | err |
| 09:08.42 | Uatec | you just get loads ofp eople who like cats |
| 09:08.44 | JT | ignore that |
| 09:08.48 | JT | 1 to 4 |
| 09:08.51 | JT | 2 to 5 |
| 09:08.55 | JT | 4 to 1 |
| 09:08.57 | JT | 5 to 2 |
| 09:09.07 | AdamB0122 | whoaa |
| 09:09.08 | AdamB0122 | wtf |
| 09:09.16 | AdamB0122 | pin 6: none? |
| 09:09.20 | JT | look at the article on t1 crossover cable on voip-info.org |
| 09:09.26 | AdamB0122 | yea, I've got it up |
| 09:09.30 | JT | pin 3, 6, 7 and 8 are not used |
| 09:09.41 | JT | just like on ethernet, only 2 pairs are requred |
| 09:09.47 | JT | the rest are unused |
| 09:09.54 | AdamB0122 | retarded.lol |
| 09:09.58 | AdamB0122 | ok, lemme go fix that cable |
| 09:09.59 | JT | differential bidirectional serial communications |
| 09:10.07 | JT | why is it retarded? it makes perfect sense |
| 09:10.08 | AdamB0122 | afk a moment |
| 09:10.22 | AdamB0122 | it makes sense |
| 09:10.26 | AdamB0122 | i just dont like it :p |
| 09:10.39 | JT | ethernet cables are retarded too then |
| 09:10.43 | JT | they only use 4 wires |
| 09:10.44 | AdamB0122 | this is true |
| 09:10.57 | JT | tx- tx+ rx- rx+ |
| 09:10.59 | AdamB0122 | I dont understand the point of "Cat5" when it only uses 4 pairs |
| 09:11.14 | JT | why? |
| 09:11.15 | [hC] | cat5 is a type of cabling |
| 09:11.22 | [hC] | its not only used for ethernet. |
| 09:11.22 | AdamB0122 | yea |
| 09:11.34 | AdamB0122 | but everyone claims you need cat5 for RJ45 connectors |
| 09:11.45 | JT | you do, for good ethernet performance |
| 09:11.51 | [hC] | well you need the proper twist that cat5 provides |
| 09:12.01 | JT | attenuation, crosstalk, twistrates, impedence |
| 09:12.03 | AdamB0122 | but thats because of sheilding, and twist, not really the 4 extra cables |
| 09:12.12 | AdamB0122 | anyway. afk, going to fix this cable |
| 09:12.20 | [hC] | the 4 extra cables twisted the way they are, are what makes the difference |
| 09:12.27 | JT | AdamB0122: cat 3 can be 1 pair or 500 pairs, i don't think you understand |
| 09:12.57 | JT | there is no shielding in cat 3 or cat 5 cables |
| 09:16.25 | Uatec | i need two computers off my boss |
| 09:16.35 | Uatec | but i wont get them |
| 09:16.43 | Uatec | he'll say "use virtual PC" |
| 09:17.00 | Uatec | IF MY PC CAN'T RUN VISTA IT WONT BE ABLE TO RUN XP *AND* VISTA |
| 09:17.14 | Uatec | and i can't put specialist isdn hardware in a virtualpc |
| 09:17.53 | JT | heh |
| 09:18.02 | JT | isdn on windows, eww |
| 09:21.38 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 09:21.50 | Uatec | no, i have no intention of running misdn on windows |
| 09:22.25 | ptiggerdine | that's just nasty.... |
| 09:22.40 | Uatec | i could use misdn under cygwin :D |
| 09:22.43 | Uatec | wouldn't that make you happy JT? |
| 09:22.58 | JT | yes, but you'd have to ring digium tech support for help |
| 09:23.07 | AdamB0122 | ok |
| 09:23.15 | AdamB0122 | now my Wildcard TE12x_) alarms Ok |
| 09:23.29 | JT | using the right cables does wonders |
| 09:23.40 | AdamB0122 | yea |
| 09:23.44 | AdamB0122 | starting simple switch on zap 1 |
| 09:23.56 | AdamB0122 | from-pstn |
| 09:23.57 | AdamB0122 | awesome |
| 09:24.10 | AdamB0122 | hm, odd. |
| 09:24.32 | AdamB0122 | ok |
| 09:24.39 | AdamB0122 | anyway to make the system do an outbound call? |
| 09:24.45 | AdamB0122 | just to see if i can get my phone to ring |
| 09:26.36 | JT | heh |
| 09:27.07 | AdamB0122 | I know i could setup outbound contents, is there a simple CLI command just to get it to initiate a call? |
| 09:27.20 | AdamB0122 | i thought there used to be a dial command |
| 09:27.32 | AdamB0122 | dial sip/8105 was my best friend. |
| 09:27.34 | JT | there is a dial commend |
| 09:27.40 | JT | it's a pretty major command |
| 09:27.59 | AdamB0122 | uhhh. |
| 09:28.12 | AdamB0122 | its not "dial" is it? |
| 09:28.20 | AdamB0122 | because i get no such command |
| 09:29.20 | JT | it is. |
| 09:29.25 | JT | ~thebook |
| 09:29.26 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:30.12 | AdamB0122 | grunts. lame 5 char limit |
| 09:31.55 | creativx | Uatec: i was being serious. being a developer and managing to focus on simple solutions often works very well. not implying that the solution needent be less complex, but the end result. |
| 09:32.15 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 09:34.01 | JT | AdamB0122: what limit |
| 09:34.37 | AdamB0122 | on thebook's site |
| 09:34.44 | AdamB0122 | searching is limited to >5 chars |
| 09:34.54 | JT | ok.... |
| 09:34.56 | JT | it's a pdf |
| 09:34.58 | JT | but okay |
| 09:35.05 | AdamB0122 | o,0 |
| 09:35.22 | AdamB0122 | lol |
| 09:35.26 | AdamB0122 | completely missed that download link |
| 09:39.08 | AdamB0122 | hm |
| 09:39.12 | AdamB0122 | intersting issue |
| 09:39.18 | AdamB0122 | Ok |
| 09:39.21 | AdamB0122 | I called in |
| 09:39.25 | AdamB0122 | and zap-1 picked up |
| 09:39.28 | AdamB0122 | all that good stuff |
| 09:39.36 | AdamB0122 | and it even rang my softone |
| 09:39.47 | AdamB0122 | but when i picked up, |
| 09:40.09 | AdamB0122 | the phone hung up, and my softphone paused, and then said it could not connect |
| 09:40.24 | AdamB0122 | and now its playing "vm-reachoper" to zap/2-1 over and over for some reason |
| 09:42.25 | AdamB0122 | alright.... its 5am. time to goto bed. |
| 09:42.47 | AdamB0122 | thanks a TON everyone |
| 09:45.59 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:49.48 | dominic1 | Hello guys, how is it possible to add general entries in the mysql database? |
| 09:51.43 | dominic1 | for my sip peers |
| 09:54.38 | *** join/#asterisk dreamind (n=dreamind@84.167.176.211) |
| 09:54.41 | dreamind | hi folks |
| 09:54.54 | dreamind | can anybody help me with set(CALLERID(num) = ...) and sip channels? |
| 09:55.32 | dreamind | I have just tested setting the number on my inbound zap to for example 0000 - but the number stays the one being sent by the zap device |
| 09:59.14 | Uatec | Isn't it about lunch time yet? |
| 09:59.29 | Uatec | creativx, i understand entirely |
| 09:59.37 | Uatec | i am a big fan of simplicity |
| 09:59.44 | Uatec | especially when it comes to user interface |
| 09:59.50 | Uatec | UI design is an art |
| 09:59.50 | dreamind | Uatec: yes its lunch time ;) |
| 09:59.56 | Uatec | it's not lunch time :( |
| 09:59.58 | Uatec | it's only 11am |
| 10:00.04 | Uatec | Time for Elevenses!!! |
| 10:00.09 | dreamind | here its exactly 12am |
| 10:00.16 | Uatec | OMFG |
| 10:00.20 | Uatec | my PC's clock is wrong |
| 10:00.29 | Uatec | i keep changing it |
| 10:00.32 | Uatec | but somebody keeps resetting it |
| 10:00.55 | Uatec | i'll bloody change it |
| 10:01.15 | Uatec | OBAN! |
| 10:01.54 | dreamind | hm, can anybody tell my why the callerid(num) is still wrong? |
| 10:02.15 | Uatec | typical |
| 10:02.26 | Uatec | time.microsoft.com is 4 minutes different from pool.ntp.org |
| 10:02.50 | tzafrir_laptop | Uatec, pool.ntp.org is a bunch of servers |
| 10:03.33 | Uatec | i know |
| 10:03.37 | Uatec | but they're all the same |
| 10:03.45 | Uatec | where as time.microsoft.com is different |
| 10:03.47 | tzafrir_laptop | dreamind, what example 0000? |
| 10:04.14 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 10:05.14 | dreamind | well just figured it out |
| 10:05.25 | dreamind | Set(CALLERID(num) = 0000) doesnt work |
| 10:05.34 | dreamind | Set(CALLERID(num)=0000) works |
| 10:05.42 | creativx | beware spaces |
| 10:05.50 | creativx | i think its written on the wiki |
| 10:05.58 | creativx | that it can give you interesting results :) |
| 10:06.55 | creativx | man i love computards.. the boss is back from 2 weeks of vacation.. his xp has not been used in 2wks either.. first thing that happens, nothing works. great |
| 10:10.10 | dreamind | hrhr |
| 10:11.19 | salvatore2 | is it possible to adjust volume on asterisk |
| 10:14.40 | creativx | dreamind: well i shouldnt complain.. its the first problem in 2 years of constant stable operation |
| 10:15.45 | tuzhila | no, it is possible to adjust volume on clients |
| 10:17.10 | dominic1 | how can I set general settings in a realtime configuration? |
| 10:18.21 | tuzhila | dominic1, use astbill |
| 10:20.37 | dominic1 | nice answer I don't need a software for kids and gamer |
| 10:20.42 | dominic1 | thank you |
| 10:24.13 | tuzhila | oh, you sucker, then use CLI |
| 10:24.44 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 10:26.39 | tuzhila | do you hear me? dominic1? |
| 10:29.52 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 10:30.31 | dominic1 | no |
| 10:32.09 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:35.53 | *** join/#asterisk incorrect (n=incorrec@host217-39-160-222.in-addr.btopenworld.com) |
| 10:36.10 | incorrect | hello, what ports should be open? |
| 10:36.55 | andrewg_fm | without any further information, I'd imagine all of them would have to be |
| 10:37.16 | incorrect | ok maybe i should ask what ports should i expect to be listening? |
| 10:37.35 | andrewg_fm | it would depend I would imagine |
| 10:37.41 | andrewg_fm | on what protocols etc you'd like to use |
| 10:37.46 | incorrect | ah |
| 10:38.58 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 10:41.13 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 10:42.50 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 10:45.38 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 10:48.31 | incorrect | lol my voip telephone provider is off line |
| 10:48.32 | incorrect | sigh |
| 10:51.23 | creativx | impressive |
| 10:51.56 | incorrect | i couldn't figure out what on earth was going on |
| 10:52.39 | incorrect | my mysql db had gone splat, the defaults on the asterisk user were set so it couldn't use the /dev/zap devices and then the voip provider was dead |
| 10:55.15 | lsodi | can any one point me to call forward macro example, voip-info.org dont have good examples ( *45<number_to _forward_calls> #45 takes forwarding off) |
| 10:59.00 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 11:00.58 | *** join/#asterisk Paul_UK (n=foo@email.seatwave.com) |
| 11:01.18 | Paul_UK | hey guys, whats the standard codec that asterisk defaults too? |
| 11:01.32 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 11:03.08 | *** join/#asterisk vgster (n=vgster@h146154.navonline.net) |
| 11:05.56 | creativx | Paul_UK: whatever you tell it to basically |
| 11:06.09 | creativx | as with asterisk, theres very rarely a "default" setting |
| 11:07.25 | Paul_UK | creativx: ok thanks |
| 11:10.33 | creativx | and also it depends on the other end |
| 11:15.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:25.36 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 11:33.09 | *** join/#asterisk Strider86 (n=m_atta_r@82.147.198.212) |
| 11:34.22 | Strider86 | someone tell me .. i need to test asterisk in a purely network environment, no analog phones or PSTN are involved .. do i need any hardware for it or just the asterisk software is enough? |
| 11:35.01 | Strider86 | i read on one of the walkthroughs which i dont remember which one now that v only need the hardware for connecting to a analog pbx or something .. |
| 11:35.49 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:36.01 | *** join/#asterisk af_ (n=getsmart@81-174-8-254.dynamic.ngi.it) |
| 11:37.25 | *** join/#asterisk jxd (n=jxd@125-229-196-147.dynamic.hinet.net) |
| 11:39.10 | Paul_UK | strider, you just need a PC, put linux on it, then install asterisk and then plug in 2 softphones which are installed on 2 other PC's, grab some headsets and you are good to go |
| 11:40.06 | jxd | having problems with my asterisk config -> TDM400P single FXS + X100P FXO... cards are installed, ztcfg looks good, zaptel looks good, the zap channels show up properly on * and have defined contexts setup in the extensions.conf yet asterisk never shows a detected call nor picks up or provides a dialtone even to my regular phone plugged straight into the TDM400P. Any advice? |
| 11:40.14 | Strider86 | ahan .. wht abt other phones .. i have in my office Mitel IP phones .. for them i need an FXO or something?? |
| 11:43.47 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 11:44.00 | creativx | Strider86: IP phones with no ethernet? |
| 11:44.13 | creativx | what protocol does these mitel phones use |
| 11:45.00 | JT | eww mitel ip phones |
| 11:45.02 | Strider86 | it is connected to ethernet .. umm not sure about the protocol .. |
| 11:45.12 | Strider86 | heh |
| 11:45.50 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 11:46.58 | Strider86 | so .. to use ip-phones with asterisk .. i need wht?? currently a ip-phone system of mitel exists in my office and i need to test asterisk first with it .. |
| 11:47.21 | creativx | Strider86: first figure out what protocol those mitel ip phones uses |
| 11:47.37 | creativx | sip perhaps |
| 11:47.46 | creativx | maybe they need a firmware upgrade |
| 11:48.13 | Strider86 | yea sip exactly .. .. |
| 11:49.16 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:49.39 | creativx | so they use SIP today? |
| 11:50.25 | salvatore2 | how can i install a stun server? |
| 11:51.00 | Strider86 | well i am a n00b abt ip-telephony .. i am interning at a networking solutions company and the guy i am working with wants me to test asterisk .. and i dont know much .. but he did tell me tht we need to make asterisk work with SIP phones .. |
| 11:51.18 | creativx | that is pretty straight forward |
| 11:51.23 | creativx | get yourself a unused mitel phone |
| 11:51.30 | creativx | set up a separate subnet for the asterisk and the phone |
| 11:51.39 | creativx | read the wiki, and you should be up and going. |
| 11:52.28 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:52.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:53.58 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 11:55.10 | Strider86 | just wht i wanted to know is that i dont need separate hardware like digium cards rite?? i need them only if i want to use analog phones with it? |
| 11:55.54 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 11:56.02 | JT | they connect to analogue or digital circuits or phones |
| 11:56.03 | lirakis | morning |
| 11:56.05 | JT | ~thebook |
| 11:56.06 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 11:56.15 | cy- | heh |
| 11:57.12 | lirakis | i know "the book" is a fantastic reference.. does anyone know when it might come out in a new version .. covering 1.4+ ? |
| 11:57.38 | *** join/#asterisk Cheetah (n=cheetah2@main-gw.bense.de) |
| 11:57.41 | Cheetah | hey fellas |
| 11:57.52 | JT | lirakis: it's still in production |
| 11:58.30 | sysreq | JT: august 1st, supposedly? (amazon says so) |
| 11:58.34 | lirakis | JT: what do you mean by that? .. they are curently working on the next version? .. |
| 11:58.38 | lirakis | ah.. cool |
| 11:58.51 | Cheetah | I've got a little problem with a Digium TE120P. Everything works fine but sometimes it happens that a few numbers of the extensions are missing on incoming calls. As if the calling person picked up and dialed too slowly. |
| 11:58.56 | JT | lirakis: yes, 2nd edition |
| 11:59.03 | lirakis | JT: excellent! |
| 11:59.18 | Cheetah | is there a way to make sure that we get the whole extension before asterisk begins to run through the dial plan? |
| 11:59.35 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 12:00.02 | Cheetah | say, we've got 1234-XXX as phone numbers, where XXX is the extension. Some calls get dropped because asterisk obviosly can't find 1234-2 (with the rest of the extension missing) |
| 12:00.28 | JT | Cheetah: incoming, outgoing, what? |
| 12:00.35 | Cheetah | incoming, from phone network |
| 12:00.37 | *** part/#asterisk incorrect (n=incorrec@host217-39-160-222.in-addr.btopenworld.com) |
| 12:00.45 | JT | that's just too weird |
| 12:00.54 | JT | definitely a digital pri circuit? |
| 12:00.57 | Cheetah | yeah, E1 |
| 12:01.10 | Cheetah | as if the call notification comes in too early and asterisk doesnt get the whole extension |
| 12:01.14 | JT | there is a d channel involved? ;) |
| 12:01.57 | Cheetah | obviously, thast the control channel of the E1 |
| 12:02.07 | JT | Cheetah: run a pri intense debug and watch the SETUP messages of incoming calls to see what data they are actually providing |
| 12:02.09 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 12:02.25 | JT | no, a D channel is the control channel of a PRI, E1 is just the physical transport |
| 12:02.35 | Cheetah | it worked fine with our old ASCOM phone switch, so i guess its a configuration issue |
| 12:02.50 | JT | watch the pri intense debug |
| 12:03.05 | Cheetah | alright.. we've got a PRI via E1 then :D bchans are 1-15,17-31 and dchan is 16 |
| 12:03.06 | JT | and pastebin.ca your extensions.conf and zapata.conf |
| 12:03.23 | Cheetah | i can't reproduce it with my phone :D it happens when customers call now and then |
| 12:03.36 | JT | annoying |
| 12:03.43 | Cheetah | it is |
| 12:03.50 | JT | let's paste bin those files anyway ;) |
| 12:04.05 | Cheetah | -.- |
| 12:04.33 | Cheetah | wow |
| 12:04.38 | Cheetah | now that's spammy |
| 12:04.56 | Cheetah | how am I supposed to filter out the needed info with lots of peopl eplacing and receiving calls every moment :D |
| 12:05.18 | JT | just extensions.conf and zapata.conf right now |
| 12:05.34 | JT | no point with the pri debug until a defective call occurs |
| 12:06.00 | Cheetah | it looks like people pick up BEFORE they dial |
| 12:06.05 | Cheetah | that is, not dial, then pick up |
| 12:06.14 | *** join/#asterisk bacs (n=bacs@flunge.gladserv.com) |
| 12:06.20 | Cheetah | resulting in the number getting slowly transmitted while it gets typed |
| 12:06.24 | JT | this makes no sense |
| 12:06.41 | JT | incoming calls over a PRI are sent as SETUP messages over the D channel |
| 12:06.57 | JT | after the telco switch has received enough data on the other side to route the call |
| 12:06.58 | Cheetah | zaptel.conf: http://pastebin.ca/631844 |
| 12:06.59 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:07.15 | Cheetah | the thing is that we have a whole number block.. say 1234-XXX |
| 12:07.26 | Cheetah | XXX can be anything from 000 to 999 |
| 12:07.39 | JT | sure |
| 12:07.56 | JT | don't really need zaptel.conf, but fyi, it appears to be in order |
| 12:08.04 | Cheetah | yeah, i figured that |
| 12:08.17 | Cheetah | extensions.conf is very very long an contains sennnsitive informaion.. what exactly do you need? |
| 12:08.37 | JT | the whole section that deals with incoming calls on the pri |
| 12:08.48 | JT | and anything included or referenced to by it |
| 12:09.38 | Cheetah | ah, that is simple |
| 12:09.39 | Cheetah | http://pastebin.ca/631846 |
| 12:10.03 | Cheetah | it basically hands off the numbers to a different section so we don't have twice the amount of configuration stuff |
| 12:10.43 | JT | although weird it'd doing that on a PRI, you should NOT match calls that way if you can avoid it |
| 12:10.49 | *** join/#asterisk lopuh666 (n=igor@motorola154-31.ip.PeterStar.net) |
| 12:10.50 | JT | . is a very naughty match |
| 12:10.58 | Cheetah | yeah i know :D |
| 12:10.59 | JT | _1234XXX, |
| 12:11.15 | lopuh666 | hi everybody |
| 12:11.24 | JT | Cheetah: can i see zapata.conf? |
| 12:11.24 | lopuh666 | can you help me |
| 12:11.28 | Cheetah | there is a little problem, because there are a few short numbers, like 12340 |
| 12:11.48 | lirakis | i am having trouble in my cdr's. Basically .. i get a call in.. it goes to an IVR .. the caller presses some number 1,2,3 ... and then that number shows up in my cdr's .. i cant figure out how to change that... :\ |
| 12:11.50 | JT | make patterns for them too. |
| 12:12.23 | lopuh666 | give me please some links |
| 12:12.36 | lopuh666 | where i can find about |
| 12:12.43 | JT | ~thewiki |
| 12:12.44 | jbot | from memory, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 12:13.07 | lopuh666 | difference between asterisk 1.2 and 1.4 |
| 12:13.18 | creativx | find the release readme for 1.4 lopuh666 |
| 12:13.21 | JT | UPGRADE.txt in the source of 1.4 |
| 12:13.30 | creativx | or that file.. |
| 12:13.31 | creativx | :) |
| 12:13.45 | lopuh666 | whats new and so one |
| 12:13.58 | lopuh666 | UPGRADE.txt? |
| 12:14.10 | Cheetah | http://pastebin.ca/631849 is zapata.conf |
| 12:14.20 | lopuh666 | where i can find this file? |
| 12:15.10 | JT | lopuh666: source of 1.4 |
| 12:15.29 | JT | Cheetah: ok, doesn't appear to be any on crack options there |
| 12:15.55 | Cheetah | well, asterisk works for weeks now without any trouble |
| 12:16.05 | Cheetah | this is just an annoying problem ;) |
| 12:16.12 | creativx | b |
| 12:16.16 | JT | it is trouble |
| 12:16.22 | JT | and shouldn't be happening |
| 12:16.32 | JT | i'm inclined to blame your telco |
| 12:16.48 | Cheetah | jt, it worked fine with our ASCOM switch before :D |
| 12:16.55 | JT | in the meantime, fix your pattern matches and watch pri intense debug for relevant calls |
| 12:16.57 | Cheetah | so I guess it must be something I can tweak on the server |
| 12:17.15 | JT | Cheetah: repeating this over and over isn't going to make me change my recommendation |
| 12:17.28 | JT | Cheetah: what happens when a call comes in and the number is short? |
| 12:18.30 | Cheetah | exactly the same |
| 12:18.34 | JT | ? |
| 12:18.42 | Cheetah | hang on |
| 12:19.07 | lopuh666 | can you give me link where it can be |
| 12:19.10 | lopuh666 | ? |
| 12:19.23 | Cheetah | http://pastebin.ca/631854 |
| 12:19.24 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:19.24 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:19.25 | *** join/#asterisk davixx (n=davixx@85.69.122.52) |
| 12:19.25 | JT | lopuh666: just download the source and extract it yourself |
| 12:20.07 | lopuh666 | i downloaded but there no something that i searching |
| 12:20.28 | lopuh666 | =( |
| 12:20.41 | JT | Cheetah: doesn't look exactly the same, look like the call fails due to asterisk hanging it up |
| 12:20.51 | jxd | [repeat] having problems with my asterisk config -> TDM400P single FXS + X100P FXO... cards are installed, ztcfg looks good, zaptel looks good, the zap channels show up properly on * and have defined contexts setup in the extensions.conf yet asterisk never shows a detected call nor picks up or provides a dialtone even to my regular phone plugged straight into the TDM400P. Any advice? |
| 12:21.31 | *** join/#asterisk friedrich| (n=friedric@e177240136.adsl.alicedsl.de) |
| 12:21.38 | JT | jxd: pastebin.ca zaptel.conf zapata.conf |
| 12:21.38 | davixx | Hi... does some one can help me to perform, via a command line tool, to register to an asterisk box, make a call, play a wave and hangup, and disconnect from asterisk ? |
| 12:22.16 | lopuh666 | i need to know only what's new in asterisk 1.4 |
| 12:22.49 | Cheetah | jt, asterisk isnt supposed to pick up the call if the number is incomplete. now the question is WHY we get those incomplete numbers. defective calls look exactly like incomplete numbers in the log. how does asterisk handle if the user on the other end is a slow typer and we get number by number once it is clear that 1234 is our prefix? |
| 12:23.15 | JT | Cheetah: i don't think you're understanding it |
| 12:23.26 | JT | Cheetah: your telco is DEFECTIVE for sending these calls at all |
| 12:23.32 | JT | this problem may have always occured |
| 12:23.34 | lopuh666 | ...(((( |
| 12:23.43 | JT | but their exchanges are setup in a stupid manner |
| 12:23.54 | Cheetah | JT, but if the telco fixes it, how is a number like 1234-0 supposed to work? thats the only exception here |
| 12:24.13 | JT | jxd: i said pastebin.ca, NOT SPAM ME IN PM |
| 12:24.36 | JT | jxd: i have your pm msgs on ignore for a few minutes |
| 12:24.52 | Cheetah | even if I fix my dialplan to only accept 1234XXX, it wont make a difference |
| 12:25.03 | JT | 12340 |
| 12:25.15 | JT | will match it |
| 12:25.16 | jxd | JT: oh ok how do I pastebin.ca? |
| 12:25.23 | jxd | jT: sorry about that |
| 12:25.32 | JT | you type it into a web browser, it's quite obvious from there |
| 12:25.49 | jxd | jt thx |
| 12:26.23 | Cheetah | JT, exactly. but is there something like a command to tell the digium card to wait longer till the extension arrived? or does the protocol not allow numbers to be completeld after the connection to the telco is established? |
| 12:26.47 | *** join/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 12:26.50 | JT | Cheetah: seriously, watch pri intense debug |
| 12:26.57 | JT | PRI is NOT AN ANALOGUE PHONE |
| 12:27.04 | Cheetah | like we get 1234-2 asterisk should say "uhh, lets wait a bit till we get the rest" and not "we don't have 1234-2 here." |
| 12:27.06 | lopuh666 | hey! anybody |
| 12:27.07 | JT | numbers do not get send digit at a time |
| 12:27.15 | Cheetah | no, but ISDN supports slow-dialing |
| 12:27.19 | JT | unless your telco is seriously screwed |
| 12:27.25 | Rienzilla | Hello everyone |
| 12:27.31 | Cheetah | ISDN supports that, though |
| 12:27.33 | andrewg_fm | lopuh666: http://www.google.com.au/search?q=asterisk+1.4+changelog&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a |
| 12:27.40 | JT | the callED and callING number are both in one frame, the SETUP frame |
| 12:27.42 | Cheetah | PRI is ISDN basically |
| 12:27.52 | JT | pri is one variety of ISDN |
| 12:28.37 | Cheetah | JT, but am I wrong if I say that ISDN supports that digit by digit thing for outgoing calls? how is determined if the number is complete? |
| 12:28.53 | JT | Cheetah: but it should never ever be used for incoming calls |
| 12:28.58 | JT | it makes no logical sense |
| 12:28.59 | Cheetah | ah |
| 12:29.04 | Cheetah | that was my question |
| 12:29.11 | jxd | JT: http://www.pastebin.ca/631857 |
| 12:29.11 | JT | the telco only routes a call once it knows where to send it |
| 12:29.29 | lopuh666 | what's dufference bettween versions asterisk 1.4 and 1.2....whats new// |
| 12:29.56 | Cheetah | JT, so I guess they are only supposed to accept 1234-0 and pass it to us OR a complete 1234-XXX number, right? |
| 12:30.26 | JT | right |
| 12:30.36 | Cheetah | so that a (intentionally) dialed 1234-1 would result in a error within the telco and we should never see it |
| 12:30.41 | JT | the people who setup their exchange are clearly idiots |
| 12:30.45 | JT | right |
| 12:30.49 | Cheetah | ah thanks |
| 12:31.00 | Cheetah | i just hope there is something like a technical hotline |
| 12:31.00 | Cheetah | :D |
| 12:31.03 | JT | but hmm |
| 12:31.11 | JT | in theory you have more DIDs than you paid for? |
| 12:31.23 | Cheetah | yeah |
| 12:31.26 | JT | dodgy varying length DIDs, but nonetheless |
| 12:31.28 | Cheetah | i mean, everything else makes no sense |
| 12:31.45 | Cheetah | weird enough that the whole PRI worked with our old ASCOM switch |
| 12:31.49 | lopuh666 | i use asterisk v1.2.22 and want to know what new in 1.4 |
| 12:32.15 | JT | jxd: what is the exact problem you're trying to solve? |
| 12:32.29 | JT | Cheetah: it's possible you never saw the errored incoming calls |
| 12:32.36 | JT | rejected by your switch |
| 12:33.17 | *** join/#asterisk mtaht4 (n=m@cpe-74-76-23-86.nycap.res.rr.com) |
| 12:33.18 | Cheetah | odd |
| 12:33.26 | JT | jxd: is it a problem with both cards or just the TDM400P? |
| 12:33.39 | jxd | JT： When asterisk runs, i cant do anything with it... both cards... wont detect the line nor answer etc |
| 12:33.45 | Rienzilla | Hey... would it somehow be possible to connect an asterisk pbx to a ventrilo server? (for example by writing a new channel module) in order to use a sip client to talk to people on a ventrilo server? |
| 12:33.55 | Rienzilla | or maybe somebody has done this already? |
| 12:33.56 | Cheetah | JT, thanks for the help. I guess I need to find a phone/person who can trigger a defective call on purpose and see what the logs say. :) |
| 12:34.03 | Rienzilla | (couldnt find info on google) |
| 12:34.08 | JT | jxd: have you setup extensions.conf? |
| 12:34.21 | jxd | yes a very simple one... ill pastebin it |
| 12:34.36 | JT | Cheetah: right, i'd be very interested to see what pri messages you see in such calls |
| 12:34.53 | JT | if it's just a short number in SETUP or there's additional messages for additional digits |
| 12:35.12 | JT | and my best advice would be to harrass your telco to fix their defective telephone exchange |
| 12:35.23 | *** join/#asterisk RSAMan (n=aa@196.210.155.3) |
| 12:35.43 | lopuh666 | ? |
| 12:35.53 | jxd | JT: updated now http://www.pastebin.ca/631863 |
| 12:35.58 | Cheetah | JT, i guess that telling my telco to fix their systems is the bigger problem. I usually hang out on freenode, so I'll let you know if I know more :) |
| 12:36.13 | lopuh666 | no one know? |
| 12:36.19 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:36.20 | *** join/#asterisk Tili (n=tili@78.16.221.87.dynamic.jazztel.es) |
| 12:36.27 | JT | jxd: exten => s,1,Wait,1 |
| 12:36.32 | JT | this is not correct syntax |
| 12:36.43 | JT | Cheetah: cool |
| 12:36.44 | jxd | JT:oh, i just copied and pasted the demo |
| 12:36.52 | JT | s,1,Wait(1) |
| 12:36.59 | RSAMan | greetings |
| 12:37.17 | RSAMan | can you please tell me where my sip.conf file is located |
| 12:37.29 | RSAMan | i cant seem to find it.. |
| 12:37.37 | jxd | JT: should asterisk be showing it detects ringing tho regardless? im wondering if its an issue running it on amd64 |
| 12:37.41 | RSAMan | following the guide , but must have missed something |
| 12:38.01 | [TK]D-Fender | JT : its perfectly valid..... |
| 12:38.22 | JT | [TK]D-Fender: didn't realise, it's just not optimal then |
| 12:38.23 | [TK]D-Fender | RSAMan: usually /etc/asterisk |
| 12:38.44 | JT | andrewg_fm: eh? |
| 12:38.54 | [TK]D-Fender | JT : Irrelevent. I don't LIKE that style personally, but its still used somewhat widely. |
| 12:38.56 | RSAMan | kk got it |
| 12:39.00 | RSAMan | real stupid question |
| 12:39.01 | RSAMan | sorry |
| 12:39.09 | JT | jxd: it should, if you are ringing the fxo |
| 12:39.10 | [TK]D-Fender | RSAMan: no, not a biggie |
| 12:39.37 | jxd | JT: yah i was... and when i hook up a phone directly to the FXS and pick it up, no dialtone, and no indication of any change on * |
| 12:39.48 | JT | [TK]D-Fender: it's dog's breakfast, i reserve my right to ask people to neaten that up |
| 12:39.51 | [TK]D-Fender | jxd: did you make sure to plug in the molex connector to your card? |
| 12:39.59 | JT | jxd: is the molex connector connected to the tdm400p? |
| 12:40.25 | jxd | JT: i am not sure what a molex connector is... the card came with a single onboard card plugged in and that is the only port that lights up |
| 12:40.40 | JT | jxd: FXS ports do not work without the power connector plugged in. |
| 12:41.09 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 12:41.18 | JT | jxd: so with the console set to a verbosity of at least 10, it shows up NOTHING when you call the FXO? |
| 12:41.24 | jxd | JT:I used to use these same cards on another PC a while back without any issues and there was no additional connector there, the problem is i dont have that box anymore to compare settings |
| 12:41.39 | JT | jxd: just plug the connector in. |
| 12:41.52 | jxd | JT: i had it set at 7. There is no connector tho, there never was |
| 12:41.53 | JT | jxd: best to follow installation instructions than argue here ;) |
| 12:42.05 | JT | jxd: the TDM400P has a connector |
| 12:42.06 | jxd | JT: the # of vs = verbosity i assume |
| 12:42.28 | JT | right |
| 12:42.28 | jxd | JT: no, mine does not. I think the connector is optional |
| 12:42.30 | JT | or set verbose 10 |
| 12:42.44 | JT | http://kb.digium.com/entry/1/85/ |
| 12:42.47 | jxd | JT: mine was ordered with only a single piggybag card |
| 12:42.57 | jxd | JT: and ordered directly from digium new |
| 12:43.03 | JT | jxd: the connector is on the main card, not a daughterboard |
| 12:43.18 | JT | it HAS a molex connector unless it's a manufacturing defect |
| 12:43.51 | JT | have you even got the computer case open? |
| 12:44.16 | jxd | JT: oh okay, maybe i just didnt know what a molex connector is or possibly forgot that i had used that port before, i just know there was no additional parts from the other pc... ill try that |
| 12:44.41 | JT | jxd: assumptions are generally not a good idea |
| 12:44.47 | jxd | JT: No i just had the card in my hands before posting tho so I know there was no additional hardware, but, there may be a plug-in-port like u said for power |
| 12:45.10 | JT | it's just a plug |
| 12:45.17 | jxd | JT: Ill try that thanks~：） I still wonder about the X100P too tho |
| 12:45.17 | JT | not additional hardware |
| 12:45.26 | jxd | JT: yes i understand that now, so its quite likely |
| 12:45.34 | JT | jxd: so what shows on the console when you call the fxo? |
| 12:45.35 | jxd | JT: its definitely NOT plugged in |
| 12:45.42 | JT | ;) |
| 12:45.48 | jxd | JT: it shows nothing |
| 12:45.50 | jxd | no console change |
| 12:45.58 | JT | maybe it's wired incorrectly |
| 12:46.11 | JT | also, X100Ps are utter junk, but it should work better than thart |
| 12:46.13 | JT | that |
| 12:46.33 | jxd | yah, if i could at least get my TDM400P working id consider buying another piggy-back module and saving myself a port anyways |
| 12:46.55 | jxd | JT: ill try it with power later as I have to step out, thanks for the advice |
| 12:47.18 | JT | jxd: also for your first dialplan i suggest something much simpler |
| 12:47.20 | JT | (no ivr) |
| 12:47.32 | JT | Answer |
| 12:47.40 | JT | Playback(tt-monkeys) |
| 12:47.42 | JT | for the fxo |
| 12:47.46 | jxd | JT: ok |
| 12:47.54 | JT | and something simple on the outgoing too |
| 12:47.59 | JT | like Answer and Dial |
| 12:48.30 | jxd | JT: Yah i agree, i just copied and pasted original stuff to make sure i didnt screw anything up |
| 12:48.42 | jxd | JT: thanks again, ill mess around more and report my findings, see u |
| 12:49.41 | jxd | JT: and sorry for the spam |
| 12:49.44 | *** part/#asterisk jxd (n=jxd@125-229-196-147.dynamic.hinet.net) |
| 12:50.06 | creativx | i like playback(tt-weasels) |
| 12:50.19 | creativx | i actually forgot that inside our dialplan some weeks after it went into production |
| 12:50.26 | creativx | and occured some 4-5 odd times a day |
| 12:50.31 | JT | monkeys is my favourite testing audio file |
| 12:51.00 | JT | both are great to play pranks on friends with |
| 12:51.15 | JT | call files to call them, play the file and record the result ;) |
| 12:55.17 | sopo2k4 | anyone here use voiptalk iaxtalk? |
| 12:58.26 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.193.91) |
| 12:59.12 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 13:04.42 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 13:07.19 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 13:07.48 | EricL | When I upgraded to 1.4.8, for some reason, the CDR stopped recording to MySQL. |
| 13:07.49 | alin` | how can I start in asterisk SLATrunk application ? |
| 13:08.09 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 13:08.23 | EricL | I don't see MySQL in my 'make menuselect'. I didn't make nay config changes from 1.4.4 to 1.4.8, any ideas what happened? |
| 13:09.05 | sopo2k4 | blah |
| 13:10.30 | sopo2k4 | is there a common answer as to why the asterisk isnt receiving voice from the person it has dialed end? |
| 13:10.38 | sopo2k4 | dialed to* |
| 13:10.56 | JT | sopo2k4: that'd depend on your setup |
| 13:11.10 | sopo2k4 | let me pastebin |
| 13:11.29 | JT | sip, zap, iax2, nat, softphones, bri phones ? :P |
| 13:12.06 | alin` | can somebody explain me how can I use the SLA, please? |
| 13:12.37 | sopo2k4 | sip |
| 13:13.02 | JT | over what from what to what |
| 13:13.21 | Corydon76-home | alin`: there's a PDF in the doc/ directory |
| 13:13.26 | JT | you need to be more descriptive, or you cannot be helped |
| 13:13.38 | JT | but my guess is nat or firewall issues |
| 13:13.39 | sopo2k4 | http://pastebin.com/d3e23971e |
| 13:14.17 | sopo2k4 | the receiving end starts ringing so thats working, however cant hear anything on the VOIP side |
| 13:14.43 | Corydon76-home | Sounds exactly like a NAT issue |
| 13:14.46 | JT | both ways? |
| 13:14.53 | sopo2k4 | yup |
| 13:15.00 | sopo2k4 | ive got port 5060 opend. |
| 13:15.13 | JT | and 10000-20000 of course |
| 13:15.24 | sopo2k4 | let me try that :P |
| 13:15.47 | sopo2k4 | this could be a problem, not sure if my router allows port ranges to be opened |
| 13:15.53 | Corydon76-home | SIP only uses 5060 for control. Media (audio) is sent using the other ports, individually allocated per call |
| 13:16.02 | JT | what a pile of junk, if it doesn't |
| 13:16.05 | sopo2k4 | yup |
| 13:16.08 | sopo2k4 | my thoughts exactly |
| 13:16.08 | JT | bin it if it doesn't |
| 13:16.09 | sopo2k4 | lol |
| 13:16.20 | sopo2k4 | is it not possible to make it use a static port |
| 13:16.22 | JT | if you can't open ranges of ports, it's useless |
| 13:16.26 | sopo2k4 | as ill be the only one using it? |
| 13:16.28 | JT | or disable the firewall |
| 13:16.30 | JT | nope |
| 13:16.32 | Corydon76-home | sopo2k4: No, it is not |
| 13:16.38 | JT | one each eay, and they change with each call |
| 13:16.38 | sopo2k4 | ok |
| 13:16.49 | sopo2k4 | mite have to disable the fwall then |
| 13:17.15 | blitzrage | you can control the range of ports Asterisk uses in rtp.conf though |
| 13:17.19 | [TK]D-Fender | sopo2k4: Read this, NOW. ... |
| 13:17.21 | [TK]D-Fender | ~sipnat |
| 13:17.21 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:17.41 | JT | still, if he can't open ranges of ports, limited usefulness, blitzrage ;) |
| 13:17.51 | sopo2k4 | ok mate, ill read that |
| 13:17.52 | sopo2k4 | :P |
| 13:18.00 | sopo2k4 | im still waiting on my IAX information tho |
| 13:18.07 | sopo2k4 | think ill set it all up properly using IAX |
| 13:18.21 | JT | no reason sip shouldn't work |
| 13:19.17 | sopo2k4 | SetCallerID doesnt work with my SIP provider tho thats why im waiting for IAX :P |
| 13:19.36 | blitzrage | JT: I didn't read that far up :) |
| 13:19.55 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:20.00 | JT | sopo2k4: should be Set(CALLERID(num)=) these days |
| 13:20.24 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:20.29 | *** join/#asterisk vgster (n=vgster@91.103.131.98) |
| 13:20.37 | sopo2k4 | ill give it a try |
| 13:20.39 | JT | sopo2k4: setting callerid to numbers you don't own is a peculiarity that only north american telcos seem to allow |
| 13:21.02 | sopo2k4 | i wanted to use asterisk for outbound calls only |
| 13:21.10 | sopo2k4 | and keep my bt phone number.... |
| 13:21.20 | sopo2k4 | by using the set cid to my bt number |
| 13:21.28 | JT | british telecom? |
| 13:21.30 | sopo2k4 | yup |
| 13:21.39 | JT | they probably don't allow it |
| 13:21.49 | JT | most telcos around the world do not permit callerid spoofing |
| 13:21.59 | JT | only ones in usa/canada seem to do it |
| 13:22.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:23.08 | sopo2k4 | surely if you own the number your spoofing as then it should be allowed? |
| 13:23.23 | sopo2k4 | well shouldnt be a problem* |
| 13:23.31 | JT | sopo2k4: how on earth do they know you own the number? they're 2 seperate accounts |
| 13:23.49 | JT | it's just not a feature you should be depending on |
| 13:23.58 | *** join/#asterisk skyphyr (n=alanj@host81-151-96-83.range81-151.btcentralplus.com) |
| 13:24.06 | sopo2k4 | well if they looked in my account details they would see my contact number the same as the outboung cid |
| 13:24.27 | JT | you're thinking very naievely |
| 13:24.37 | JT | the itsp connects to the telco |
| 13:24.37 | sopo2k4 | im new to this. |
| 13:24.54 | sopo2k4 | ok |
| 13:24.59 | JT | the telco still doesn't know that one customer of that itsp happens to own a number that its trying to set caller id to |
| 13:25.10 | JT | you just should not expect it to work |
| 13:25.22 | JT | americans are simply spoilt in that it often works there |
| 13:25.36 | sopo2k4 | how things are going right now im not expecting it too :P |
| 13:25.49 | sopo2k4 | but would be a bonus |
| 13:26.06 | JT | usually you're only meant to set callerid to a number you have like a particular DID or what not on a PRI |
| 13:27.38 | sopo2k4 | i assumed that :P |
| 13:28.32 | *** join/#asterisk NirS (n=Nir@84.94.120.181.cable.012.net.il) |
| 13:29.07 | mocker | damn spoiled americans/ |
| 13:29.49 | alin` | Corydon76-home: yes, there is a PDF file, but it is not well explained. |
| 13:30.06 | alin` | Jul 24 18:19:44] WARNING[25112]: pbx.c:1783 pbx_extension_helper: No application 'SLATrunk' for extension (line, 111, 1) |
| 13:30.06 | alin` | <PROTECTED> |
| 13:30.21 | alin` | 'SLATrunk' is not a registered application |
| 13:30.31 | alin` | how can I register it? |
| 13:30.31 | Corydon76-home | alin`: you need zaptel |
| 13:30.49 | Corydon76-home | Compile and install zaptel, then rebuild Asterisk |
| 13:31.02 | alin` | Corydon76-home: but I use just 2 SNOM phones connected to asterisk |
| 13:31.11 | Corydon76-home | Compile and install zaptel, then rebuild Asterisk |
| 13:31.12 | sopo2k4 | whats your favourite * distro? |
| 13:31.22 | sopo2k4 | *nix* |
| 13:31.31 | alin` | What is zaptel good for? |
| 13:31.36 | Corydon76-home | Timing |
| 13:31.43 | alin` | ??? |
| 13:31.48 | Corydon76-home | Just do it |
| 13:32.01 | Corydon76-home | SLA requires timing |
| 13:32.23 | JT | someone should write a list of all the software functions that require zap timing |
| 13:32.26 | JT | it seems to be growing |
| 13:32.35 | alin` | ok, but I thought that ZAPTEL is needed just for ZAPTEL CARDS |
| 13:32.42 | Corydon76-home | JT: it's because SLA is built on top of meetme |
| 13:33.03 | [TK]D-Fender | JT : You mean all THREE of them? :) |
| 13:33.05 | alin` | so what link is between ZAPTEL AND MEETME? |
| 13:33.09 | Corydon76-home | alin`: you can do whatever you like, but Zaptel is required for SLA |
| 13:33.18 | alin` | aaaa, ok |
| 13:33.21 | [TK]D-Fender | alin`: You need a TIMING source for * based conferencing. |
| 13:33.24 | sopo2k4 | to solve the |
| 13:33.27 | sopo2k4 | port 10000-20000 |
| 13:33.43 | sopo2k4 | The DMZ feature allows you to specify one computer on your network to be placed outside of the NAT firewall. using this feature would solve my problems correct? |
| 13:34.05 | JT | sopo2k4: shouldn't have to go that far |
| 13:34.05 | [TK]D-Fender | sopo2k4: If you're too lazy to specify more specific port ranges, sure |
| 13:34.06 | *** join/#asterisk AdamPal (n=adam@194.164.230.200) |
| 13:34.20 | JT | i didn't know sla uses meetme |
| 13:34.25 | sopo2k4 | fender, my router doesnt allow me to open ranges. |
| 13:34.26 | [TK]D-Fender | sopo2k4: But that starts to become a security risk |
| 13:34.31 | Corydon76-home | JT: yep |
| 13:34.44 | JT | sopo2k4: what is this pile of junk? |
| 13:34.48 | sopo2k4 | Belkin |
| 13:34.54 | sopo2k4 | let me find what model |
| 13:34.55 | sopo2k4 | hold |
| 13:34.55 | [TK]D-Fender | QUALITY |
| 13:35.14 | [TK]D-Fender | sopo2k4: don't bother. Just DMZ it to start and figure out the rest after |
| 13:35.36 | sopo2k4 | ok mate :) |
| 13:36.03 | JT | an old $0 desktop pc running linux will do a better job than that belkin ;) |
| 13:36.20 | sopo2k4 | my ubuntu has all the updates that it notified me about so that should be ok |
| 13:36.36 | sopo2k4 | for a temp fix anyway |
| 13:36.59 | sopo2k4 | think ill get a linksys |
| 13:37.02 | *** join/#asterisk Fulk (n=test@87-194-176-39.bethere.co.uk) |
| 13:37.24 | *** part/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 13:41.07 | alin` | Corydon76-home: I am installing Zaptel now... |
| 13:42.31 | AdamPal | Hi there - I'm trying to set up a very simple 'first time asterisk setup' with one sip phone talking to another sip phone. Both sip phones and my asterisk server all have separate public, unfirewalled IPs, and I'm a bit stuck... I'm receiving error "Registration from '<sip:adam@194.164.230.199>' failed for '194.164.230.200' - Username/auth name mismatch. ASTERISK - 194.164.230.199, PHONE1 - 194.164.230.200 ... |
| 13:43.54 | *** join/#asterisk galeras (n=root@201.245.103.169) |
| 13:44.11 | sopo2k4 | ive DMZ'd my ubuntu and i still cant hear the other line |
| 13:44.14 | sopo2k4 | :s |
| 13:45.59 | Uatec | do people find that CDR is sometimes just a little bit weird? |
| 13:46.20 | Uatec | i've got a call that came in from misdn to one destination number |
| 13:46.34 | Uatec | i know it went to that destination number because of the Dial(...) data that was used |
| 13:46.48 | Uatec | but it's recorded in cdr as them having dialed a different number |
| 13:47.03 | creativx | impressive. the company who sells plantronic headsets cant even manage to transfer me to a damn sales person without the call dying |
| 13:47.07 | *** join/#asterisk davidsf (n=davidsfe@bl8-162-117.dsl.telepac.pt) |
| 13:47.18 | davidsf | Hi |
| 13:47.19 | Uatec | sopo2k4, DMZ is not a verb |
| 13:47.25 | Uatec | It's a TLA. |
| 13:47.40 | sopo2k4 | well you understood what i meant? |
| 13:47.49 | *** join/#asterisk myiagy (i=myiagy@189.4.123.131) |
| 13:47.50 | davidsf | someone please can help me? i have a problem with the dial command CMD |
| 13:48.25 | Uatec | Only because I'm hyperintelligent. |
| 13:48.25 | sopo2k4 | ;p |
| 13:48.30 | alin` | So meetme does not work without Zaptel? |
| 13:48.39 | sopo2k4 | should have put Fender/JT: before |
| 13:48.45 | davidsf | how can i sen a beep sound in asterisk every 60 seconds?? |
| 13:49.03 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 13:49.11 | creativx | davidsf: after dial() ? |
| 13:49.22 | alin` | davidsf: good question |
| 13:49.26 | davidsf | yes! |
| 13:49.34 | creativx | or just an extension where you hear a beep every 60 seconds |
| 13:49.42 | alin` | davidsf: please tell me also when you find out |
| 13:49.48 | davidsf | i whant to play a beep every 60 seconds.. |
| 13:50.07 | alin` | davidsf: during the call |
| 13:50.11 | davidsf | yes! |
| 13:50.17 | creativx | modify the tone table and set that extension to an obscure language |
| 13:50.37 | davidsf | where i can find a tone table' |
| 13:50.37 | creativx | dial plays the ringing tones based on the tone configs, doesnt it |
| 13:50.39 | davidsf | ? |
| 13:51.14 | Strom_M | no...dial should not play ringing tones unless you specifically tell it to |
| 13:51.17 | davidsf | no is a sound of asterisk, on sounds paste.. |
| 13:51.39 | Strom_M | it's the called party's responsibility to send ALERTING or PROCEEDING back to your end |
| 13:51.45 | davidsf | yes but i tell it to it bu it doesn't work.. |
| 13:51.49 | AdamPal | Hi there - I'm trying to set up a very simple 'first time asterisk setup' with one sip phone talking to another sip phone. Both sip phones and my asterisk server all have separate public, unfirewalled IPs, and I'm a bit stuck... I'm receiving error "Registration from '<sip:adam@194.164.230.199>' failed for '194.164.230.200' - Username/auth name mismatch. ASTERISK - 194.164.230.199, PHONE1 - 194.164.230.200 ... |
| 13:51.55 | Uatec | so everybody finds CDR behaves perfectly normally? |
| 13:52.54 | [TK]D-Fender | AdamPal: bad user/pass like it says... |
| 13:53.24 | alin` | *CLI> [Jul 24 18:44:00] WARNING[6724]: pbx.c:1783 pbx_extension_helper: No application 'SLATrunk' for extension (line, 111, 1) |
| 13:53.25 | alin` | <PROTECTED> |
| 13:53.30 | alin` | I compiled Zaptel |
| 13:53.46 | alin` | however, SLA does not work yet... |
| 13:53.55 | AdamPal | [TK]D-Fender: I see that.. how do I set a correct user/pass ? |
| 13:54.09 | [TK]D-Fender | AdamPal: "vi /etc/asterisk/sip.conf" |
| 13:54.27 | AdamPal | [TK]D-Fender: Thats where I'm editing... I've set it correctly as far as I see. type=friend, username=adam, secret=test |
| 13:55.13 | [TK]D-Fender | AdamPal: remove "username=", and make sure the user you are filling in on your client is the same as the [whatevershere] |
| 13:56.04 | AdamPal | type=? |
| 13:56.19 | [TK]D-Fender | AdamPal: "type=firend" |
| 13:56.23 | [TK]D-Fender | friend* |
| 13:56.28 | *** join/#asterisk guilherme-jorge (n=guilherm@host170.190-30-12.telecom.net.ar) |
| 13:56.33 | AdamPal | Now I have [adam], type=friend, context=from-sip, host=dynamic, secret=test |
| 13:56.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:56.40 | AdamPal | No error at all now, just client registration failed |
| 13:57.10 | alin` | So nobody can help me to start the SLA? :( |
| 13:57.45 | AdamPal | Ah no, error is back again |
| 13:57.52 | guilherme-jorge | Somebody knows a AGI script in Perl, to integrate Asterisk+Jabber?? I'ld like to send message to Jabber when I get responde NOANSWER, for example... Any idea? |
| 13:58.30 | guilherme-jorge | I found AGI script to do this in PHP, but not in Perl |
| 13:59.08 | *** join/#asterisk davixx (n=davixx@85.69.122.52) |
| 14:00.01 | AdamPal | Are there any other changes I need to make to sip.conf ? |
| 14:00.25 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 14:00.39 | AdamPal | asteriskdocs.org is down |
| 14:01.50 | sweeper | voip-info.org \o\ |
| 14:03.32 | AdamPal | heh |
| 14:03.42 | AdamPal | now it just says 'wrong password' when it definitely isnt |
| 14:03.58 | [TK]D-Fender | AdamPal: PASTEBIN is your friend |
| 14:04.00 | [TK]D-Fender | ~pb |
| 14:04.01 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:04.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 14:05.20 | *** join/#asterisk wunderkin (n=wunderki@ip68-2-62-143.ph.ph.cox.net) |
| 14:05.34 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:06.15 | coolbeans | Hi all. When using mysql static configs and changing voicemail passwords, does asterisk update the database with the new password? I'm having an issue where the password is changed, but the database doesn't seem to update. |
| 14:06.49 | davidsf | how can i send a beep every 60 seconds during the call????? |
| 14:07.02 | davidsf | Anyone??? |
| 14:07.07 | [TK]D-Fender | davidsf: You can't |
| 14:07.15 | davidsf | why? |
| 14:07.30 | davidsf | ? |
| 14:07.35 | [TK]D-Fender | davidsf: Not unless you manually do a 3-way conference with some dialplan to do it, but nothing automatic |
| 14:07.54 | davidsf | really??? are you sure??? |
| 14:08.12 | [TK]D-Fender | davidsf: Why can't my suitcase fly by itself? Because nobody MADE IT POSSIBLE TO. |
| 14:08.44 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:08.44 | davidsf | dial command has a options as a warning option??? |
| 14:09.05 | *** join/#asterisk joshr (n=joshr@63-211-239-34.teliax.com) |
| 14:09.07 | davidsf | ?? |
| 14:09.30 | [TK]D-Fender | davidsf: sorry, I don't speak that dialect of gibberiish.... |
| 14:09.42 | davidsf | exten => 200,2,Set(TIMEOUT(absolute) = 390) |
| 14:09.42 | davidsf | exten => 200,3,Set(LIMIT_PLAYAUDIO_CALLEE=yes) |
| 14:09.42 | davidsf | exten => 200,4,Set(LIMIT_WARNING_FILE=beep) |
| 14:09.42 | davidsf | exten => 200,5,Set(LIMIT_TIMEOUT_FILE=beep) |
| 14:09.42 | davidsf | exten => 200,6,Dial(SIP/${EXTEN},20,L(390000:58000:60000)r) |
| 14:10.38 | *** join/#asterisk Zhadnost (n=tom@serbacoatings.demon.co.uk) |
| 14:10.39 | davidsf | sorry my english |
| 14:10.53 | davidsf | but asterisk is universal!! |
| 14:11.01 | davidsf | isn't? |
| 14:11.15 | davidsf | anyone?? |
| 14:11.18 | coolbeans | Hi all. When using mysql static configs and changing voicemail passwords, does asterisk update the database with the new password? I'm having an issue where the password is changed, but the mysql database table doesn't seem to update. |
| 14:11.33 | *** join/#asterisk nephfl (n=traveler@adsl-070-147-105-151.sip.gnv.bellsouth.net) |
| 14:12.12 | [TK]D-Fender | davidsf: What kind of pathetic claim is "but asterisk is universal!!"? |
| 14:12.22 | *** join/#asterisk Tili (n=tili@78.16.221.87.dynamic.jazztel.es) |
| 14:12.30 | [TK]D-Fender | davidsf: is it supposed to make you COFFEE too?! (The fact that mine DOES is besides the point) |
| 14:12.46 | Zhadnost | Does anyone know if there was a big difference between 1.2 and 1.4 versions of Pickup? |
| 14:12.47 | nephfl | hello, does anyone happen to know how to get buddies to work on the polycom 60!? |
| 14:12.54 | creativx | im impressed you even have a suitcase, [TK]D-Fender |
| 14:13.33 | davidsf | coffe is a good idea!! :D |
| 14:13.38 | [TK]D-Fender | creativx: Its so I can store the body parts of the newbs I'm forced to dismember for transport ;) |
| 14:14.04 | creativx | heheh |
| 14:14.19 | davidsf | ok, if you can help please fly with your suitcase away from her.. |
| 14:14.21 | davidsf | :P |
| 14:14.21 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:14.38 | davidsf | anyone? |
| 14:14.52 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 14:15.11 | Qwell | Bueller? |
| 14:15.23 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 14:15.27 | Mercestes | Farris? |
| 14:15.40 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 14:15.47 | Hymie | any idea why placing the funky MESSAGE_WAITING .... silence line in my own globalconf file yields no joy in removing the mwi sound from my polycom? |
| 14:17.12 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 14:17.13 | creativx | Mercestes: isnt it ferris |
| 14:17.23 | davidsf | L(x[:y][:z]) |
| 14:17.24 | davidsf | Limits the call to x milliseconds, warning when y milliseconds are left and repeating |
| 14:17.24 | davidsf | every z milliseconds until the limit is reached. The x parameter is required; the y and z |
| 14:17.47 | davidsf | silly peoplE!!!! |
| 14:18.06 | coolbeans | Hi all. When using mysql static configs and changing voicemail passwords, does asterisk update the database with the new password? I'm having an issue where the password is changed, but the mysql database table doesn't seem to update. |
| 14:18.14 | coolbeans | In 1.2.x |
| 14:19.10 | nephfl | so, does anyone know a reference for getting buddies to work? |
| 14:19.15 | Mercestes | creativx, you are correct |
| 14:19.38 | zeeesh | how to get "computer serial or identification number"? |
| 14:19.54 | Mercestes | nephfl, which phones? |
| 14:19.59 | *** join/#asterisk Jon335_ (n=Jon335@unaffiliated/jon335) |
| 14:20.25 | nephfl | polycom 601 |
| 14:20.31 | Zhadnost | quit |
| 14:20.43 | *** join/#asterisk TedNJ37 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 14:20.56 | TedNJ37 | Hi guys. I am a newbie in this. I have a small box which handles about 10 extensions, I want to be able to link them to a landline, is there any way for me to allow extensions to dial out through my landline if I connect the phone plug to the modem of the box? |
| 14:20.57 | red9012 | how can I handle fax with asterisk? |
| 14:21.41 | coolbeans | TedNJ37: No. |
| 14:21.55 | coolbeans | TedNJ37: Get a X100P card, $9-15 dollars ebay. |
| 14:22.13 | red9012 | ted-- yes, using dial command and x100p card |
| 14:22.16 | [TK]D-Fender | TedNJ37: No, you can't use just any cheap-shit modem to access the PSTN |
| 14:22.17 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 14:22.41 | red9012 | the x100p will work just fine. |
| 14:22.55 | wunderkin | Hymie, http://www.pastebin.ca/631955 |
| 14:23.45 | [T]ank | something odd is happening to one of my servers... after about 4 days my inbound calls start acting up. the caller does not hear my recorded messages. if i do a full reload then all is well again for about 4 days. any ideas what could be causing this? version 1.2.19 |
| 14:23.50 | coolbeans | Hi all. When using mysql static configs and changing voicemail passwords, does asterisk update the database with the new password? I'm having an issue where the password is changed, but the mysql database table doesn't seem to update. |
| 14:24.03 | coolbeans | In 1.2.18 |
| 14:24.08 | [T]ank | te410p t1 card also with 3 pris |
| 14:25.28 | Mercestes | coolbeans, does your asterisk user to the Mysql database have update privileges? |
| 14:26.03 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 14:26.23 | Sci_05 | can someone check this out and tell me why my asterisk box isn't calling the System command at the end. http://pastebin.com/m1263ee67 |
| 14:26.25 | guilherme-jorge | Somebody knows a AGI script in Perl, to integrate Asterisk+Jabber?? I'ld like to send message to Jabber when I get responde NOANSWER, for example... Any idea? |
| 14:26.39 | guilherme-jorge | I found AGI script to do this in PHP, but not in Perl |
| 14:27.25 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 14:28.59 | TedNJ37 | Thanks Coolbeans |
| 14:29.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:29.14 | lirakis | has anyone been to an "asterisk world" during the VON convention? |
| 14:29.31 | lirakis | my company is sending some people to VON .. and i am trying to decide if i should go to VON .. or Asterisk World |
| 14:29.52 | coppice | go to the beach and relax |
| 14:29.55 | *** join/#asterisk saftsack (n=saftsack@pD9E05D88.dip.t-dialin.net) |
| 14:30.24 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 14:31.08 | *** join/#asterisk Zhadnost (n=tom@serbacoatings.demon.co.uk) |
| 14:32.15 | *** part/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 14:33.22 | myiagy | i'm looking for a way to secure my conversations, out of luck googling for it.. the suggestion i got most is to use vpn, but that doesnt guaranty security inside of the VPN. i found something about SRTP, but no documentation about it.. |
| 14:33.50 | myiagy | does anyone have a suggestion what may i search to find a way to encrypt my voice packets? and the sip packets too if possible |
| 14:34.04 | coolbeans | A ipsec tunnel? |
| 14:34.14 | Sci_05 | vtun? |
| 14:34.28 | coolbeans | Or just tunnel with an ssh wrapper? |
| 14:35.00 | coolbeans | myiagy: It would be an ordeal for someone to reconstruct an actual conversation, considering the nature of routing and the packets' construction. |
| 14:35.17 | coolbeans | If security is that big of a concern, you should probably have dedicated, private connectivity ... |
| 14:35.58 | AdamPal | Anyone know of a good gui softphone for linux? |
| 14:36.16 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 14:36.35 | myiagy | i see.. i'll go look for ipsec and vtun then.. thank you |
| 14:36.43 | ramindia | hey can some one tell me.. how can i send the recordings to other server only recordings of all calls |
| 14:36.53 | coolbeans | myiagy: What's the reason for needing such security? |
| 14:36.54 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:36.54 | [TK]D-Fender | AdamPal: Ekiga |
| 14:37.03 | coolbeans | ramindia: rsync |
| 14:37.04 | AdamPal | I tried that |
| 14:37.09 | AdamPal | It keeps coming "Security Check Failed" |
| 14:37.12 | AdamPal | Whenever I try and make a call |
| 14:37.13 | lirakis | AdamPal: xten is a good softphone for linux too |
| 14:37.16 | AdamPal | I have no real options to configure either |
| 14:37.19 | lirakis | AdamPal: it isnt open source |
| 14:37.19 | ramindia | not offline real time |
| 14:37.20 | [TK]D-Fender | AdamPal: You've set it up wrong then |
| 14:37.25 | lirakis | AdamPal: but it works fine |
| 14:37.25 | myiagy | coolbeans the client is demanding it.. in my understanding, he has confidential conversations |
| 14:37.29 | AdamPal | [TK]D-Fender: There is nothing to set up really? |
| 14:37.38 | lirakis | myiagy: you should try zfone |
| 14:37.39 | [TK]D-Fender | AdamPal: Not much.... |
| 14:37.50 | [TK]D-Fender | AdamPal: If you can't figure that out, its not the phone... |
| 14:37.58 | myiagy | zfone, ok, i'll look for that too |
| 14:38.00 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:38.01 | AdamPal | Figure what out? |
| 14:38.04 | [TK]D-Fender | AdamPal: pastebin your sip.conf minus only passwords |
| 14:38.08 | AdamPal | okay |
| 14:38.12 | ramindia | coolbeans: iam looking on real time.. not offline |
| 14:38.17 | [TK]D-Fender | AdamPal: How to get Ekiga setup right |
| 14:38.19 | coolbeans | myiagy: The act of reconstructing a conversation from a VOIP connection is tremendous. You would have to someone captuare *all* the packets, then decode and reassemble them. If someone was going to put that much effort into it, it would be easier just to bug the guys office. |
| 14:38.46 | [TK]D-Fender | coolbeans: Actually there all sorts of Wireshark filters for this..... |
| 14:38.47 | coolbeans | A sniffer on the LAN where the phone is would be the best bet for capturing all the packets. |
| 14:39.20 | myiagy | what about this? |
| 14:39.21 | myiagy | http://www.oxid.it/ca_um/topics/voip.htm |
| 14:39.29 | lirakis | myiagy: http://www.e164.org/wiki/AsteriskSRTP |
| 14:39.30 | coolbeans | [TK]D-Fender: Right, but getting it to an actual audiable medium would be an effort. My point was, there's a lot easier way to hear what's up that this route. |
| 14:39.58 | [TK]D-Fender | coolbeans: From what I've read I spits out a ready-to-play sound file... |
| 14:40.02 | [TK]D-Fender | it* |
| 14:40.16 | *** join/#asterisk Strom_M (n=strom@h72-2-22-215.bigpipeinc.com) |
| 14:40.54 | coolbeans | [TK]D-Fender: That's interesting. I stand corrected, then. But someone would still have to sniff off the packets which would require LAN access on either the phone side or the soft switch side. You just can't pull packets off the internet. |
| 14:41.23 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:41.24 | [TK]D-Fender | coolbeans: I never said it wouldn't likely be an inside job ;) |
| 14:41.44 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 14:41.45 | *** mode/#asterisk [+o angler] by ChanServ |
| 14:41.48 | myiagy | coolbeans well, if i use ipsec for security, then anyone on the inside of the vpn would be able to sniff it, wouldnt they? |
| 14:41.53 | Zhadnost | Shame, after upgrading to 1.4.8, I'm damned if I can get it to work. |
| 14:41.59 | coolbeans | So physical security + common sense would be better than trying to encrypt packets. Even with a VPN, a sniffer would still prevail at either endpoint. |
| 14:42.02 | myiagy | they are concerned about internal escurity too |
| 14:42.41 | myiagy | what do you mean by physical security? how can i use that with voip? |
| 14:42.48 | Zhadnost | I remember a "shocking report" that I read suggesting that 80% of companies who use VoIP-like services don't encrypt the voice data. |
| 14:42.57 | AdamPal | Heres a new one: Jul 24 18:33:08 WARNING[6160]: chan_sip.c:3602 process_sdp: Unknown SDP media type in offer: video 5006 RTP/AVP 31 |
| 14:43.04 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 14:43.32 | coolbeans | myiagy: physical, in that the hardware is secure, i.e., nobody can walk in off the street, plug in a sniffer, and grab network traffic. |
| 14:43.37 | Zhadnost | lo tecnico |
| 14:43.41 | coppice | encrypt the voice == speak in code |
| 14:43.46 | jbroome | windtalkers! |
| 14:43.55 | *** join/#asterisk CVirus (n=GoD@62.135.96.251) |
| 14:43.58 | coolbeans | coppice: lol, that's another cheap, easy way to do it. |
| 14:44.19 | myiagy | coppice :P |
| 14:44.41 | Zhadnost | Does anyone here use Pickup with 1.4.8? |
| 14:44.54 | myiagy | coolbeans but that would only protect from the outside right? what about the company employees who use the network |
| 14:44.55 | CVirus | Can the extensions.conf be as simple as this http://rafb.net/p/xgRakr43.html ? |
| 14:45.14 | coolbeans | myiagy: with iptunnels and ipsec, you could originate the VPN from a local asterisk box. But there's still no encryption to/from the phones. |
| 14:45.18 | CVirus | or do I need any of the other default values ? |
| 14:45.23 | Zhadnost | CVirus> don't see why not |
| 14:45.26 | CVirus | Zhadnost: thanks |
| 14:45.38 | coolbeans | My point, really, I would just buy a $15 "bug" and slide it under the guys desk or something. |
| 14:45.44 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:45.49 | Zhadnost | CVirus> You should be able to get it working with just the last 2 lines depending on what else is set up. |
| 14:45.49 | coolbeans | ... before I would try to "hack" voice packets back together. |
| 14:46.19 | coolbeans | brb |
| 14:46.22 | myiagy | i see.. well i'll throw the customer the options and see what he says.. thanks |
| 14:49.06 | variable_office | anybody know what the maximum transmission range for pots is? ie. fxs to phone |
| 14:50.06 | coppice | a real exchange line will usually do about 10km |
| 14:50.09 | Fulk | variable_office, as in physical cable length? |
| 14:50.20 | variable_office | Fulk, yes |
| 14:50.21 | Fulk | very long |
| 14:51.00 | Fulk | it's not like ethernet which has a 100m max |
| 14:51.05 | variable_office | what about off of an asterisk box? |
| 14:51.33 | Fulk | variable_office, how far do you want to run it? |
| 14:51.44 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 14:52.23 | variable_office | at least a 3/4 of a mile, id like 1 + miles though |
| 14:52.26 | *** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net) |
| 14:52.36 | *** join/#asterisk saftsack (n=oliver@p54A7E1DC.dip.t-dialin.net) |
| 14:52.42 | coppice | any line should do that |
| 14:52.58 | Fulk | yes, but what's the power output from the Asterisk box |
| 14:53.03 | variable_office | even if i just stuck a linksys ata on the line, it would work fine? |
| 14:53.03 | coppice | some PBX lines take compromises that limit them to a couple of km |
| 14:53.29 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:53.42 | coppice | but most cards should be good for a few km |
| 14:53.42 | variable_office | Fulk, what do you mean power output? like in mA / |
| 14:54.02 | variable_office | is there anywhere to check these statistics, before i go buying? |
| 14:54.44 | Fulk | I'd check with the card manufacturer |
| 14:54.44 | coppice | basically if the card puts out 48V it should be OK. of these nasty PBX type cards that use 24V will be limited |
| 14:54.45 | Fulk | just to be sure |
| 14:55.07 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 14:56.18 | msetim | hi guys |
| 14:56.33 | msetim | how can I know if a meet is locked or not |
| 14:56.38 | msetim | meetme * :) |
| 14:57.24 | *** join/#asterisk Zhad (n=tom@cpc1-sout6-0-0-cust691.sotn.cable.ntl.com) |
| 14:59.39 | *** join/#asterisk CuriosCat (i=stian@ninja.noc.host.net) |
| 14:59.41 | CuriosCat | Hi. |
| 14:59.45 | Fulk | hi |
| 14:59.58 | blitzrage | hoi |
| 15:00.18 | Fulk | salut |
| 15:00.21 | CuriosCat | So, I have a POTS line, a VOIP phone and a Linux box and a Digium FXS card. |
| 15:00.25 | CuriosCat | This should be interesting. :) |
| 15:00.35 | Fulk | sounds good |
| 15:00.54 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 15:01.21 | Mercestes | Got a good start |
| 15:01.40 | [TK]D-Fender | CuriosCat: Except for the PSTN line you have nothing to take in... |
| 15:02.18 | CuriosCat | Correct. |
| 15:02.21 | Fulk | yes, if you want to connect your POTS line you need an FXO card |
| 15:02.36 | CuriosCat | err, that's what I have. Sorry. |
| 15:02.49 | CuriosCat | I still get FXO and FXS mixed up. |
| 15:03.13 | CuriosCat | I have a TDM400P with one FXO module. |
| 15:04.27 | jarrod | "rtc: lost some interrupts at 1024Hz. |
| 15:04.34 | jarrod | im getting that error when loading ztdummy |
| 15:06.31 | [TK]D-Fender | CuriosCat: Ok, better |
| 15:07.07 | *** join/#asterisk Chuji (n=brian@mail.point3media.com) |
| 15:07.34 | Chuji | Anyone know how to set the digit timeout value in Zaptel for the simpleswitch? |
| 15:07.54 | CuriosCat | TK: Starting from scratch -- I had the zaptel drivers working at one point, but I had to move the card to a different server in order to get access to a phone line. |
| 15:08.47 | *** join/#asterisk karrotx (n=karrotx@ip67-91-24-2.z24-91-67.customer.algx.net) |
| 15:09.12 | karrotx | my asterisk server is sending reverse dns requests every 2-3 seconds to my dns server |
| 15:09.20 | karrotx | does it require reverse dns to operate |
| 15:09.23 | karrotx | and can i turn it off? |
| 15:10.25 | Chuji | karrotx: Nah asterisk wants DNS. You can run a proxy only DNS server on your asterisk box if you want. That's what I do at home so it doesn't become dependant on the internet |
| 15:10.46 | karrotx | i have no problem with that |
| 15:10.57 | karrotx | it's just spamming my dns server for reverse entries for the local ip |
| 15:11.26 | neverblue | any VOIP providers in the channel? |
| 15:11.28 | CuriosCat | Set up a reverse zone for the local IP :) |
| 15:12.04 | neverblue | looking to try out a service |
| 15:12.04 | karrotx | CuriosCat: we have 250 phones; it would become rather tedious |
| 15:12.10 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:15.12 | CuriosCat | $ORIGIN 0.168.192.in-addr.arpa. |
| 15:15.13 | darkskiez | karrotx: include "/etc/bind/zones.rfc1918"; in your named.conf.local - or similar depending on distro |
| 15:15.30 | CuriosCat | $GENERATE 1-254 $ IN PTR phone-$.local. |
| 15:15.38 | CuriosCat | add NS and SOA records and you're done |
| 15:15.41 | CuriosCat | literally four lines. |
| 15:16.12 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 15:16.17 | karrotx | CuriosCat: they're not all phones |
| 15:16.27 | karrotx | phones are on the same network as workstations |
| 15:16.33 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:16.50 | karrotx | but i guess it's not a bad idea |
| 15:21.02 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:21.36 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 15:23.13 | CuriosCat | karrotx: Well, client-$ then. You get the idea :) |
| 15:24.28 | CuriosCat | eww, make install relies on access to digium's ftp server? |
| 15:24.39 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 15:24.48 | Qwell[] | CuriosCat: you can disable the sounds in menuselect, or use a release tarball |
| 15:25.04 | IOscanner | Has anyone setup a USB sound device to connect an overhead speaker system? |
| 15:25.35 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 15:26.01 | CuriosCat | qwell: It works, I was just a bit surprised to see wgets in a make install :P |
| 15:26.25 | Primer | Anyone know how to dial a sip:user@domain directly from a polycom soundpoint IP 430? |
| 15:26.35 | *** join/#asterisk CunningPike (n=arodgers@209.17.159.211) |
| 15:26.53 | Primer | I'm wondering if somehow it's been disabled on this phone |
| 15:26.54 | Primer | nm |
| 15:26.56 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 15:29.04 | AdamPal | Hi there, I have taken the default asterisk config, and added the following to sip.conf: [adam], type=friend, calledid=Adam <adam>, host=dynamic, nat=1, mailbox=adam, secret=test |
| 15:29.16 | AdamPal | I'm getting "Security Error" from Ekiga when trying to dial but it DOES register correctly |
| 15:31.46 | *** join/#asterisk Peri (n=redanti@xtreme-14-56.dyn.aci.on.ca) |
| 15:32.09 | Peri | hi, is there any way to limit the number of incoming calls per sip peer? |
| 15:32.14 | *** join/#asterisk af_ (n=getsmart@81-174-46-138.dynamic.ngi.it) |
| 15:33.48 | Zhad | quit |
| 15:35.01 | *** join/#asterisk astserdev (n=core@59.160.62.22) |
| 15:35.14 | *** join/#asterisk nain (i=nain@203.81.197.56) |
| 15:35.51 | nain | Hi Every body |
| 15:37.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:38.30 | variable_office | how many sip ulaw channels could a 533 mhz box take approximately? |
| 15:38.46 | Mercestes | <PROTECTED> |
| 15:39.00 | blitzrage | 10-20 probably |
| 15:39.01 | Qwell[] | variable_office: Anywhere between 1 and 5000 |
| 15:39.26 | blitzrage | assuming no transcoding... probably quite a few |
| 15:39.38 | variable_office | blitzrage, 10-20 you think? thats not too bad! |
| 15:39.44 | nain | I am using asterisk 1.4 and using A200r Sangoma Card, But zap channels is not detecting Callerid when call arrive on zap channel? |
| 15:39.47 | variable_office | na, straight ulaw, ulaw in, ulaw out |
| 15:39.47 | astserdev | did anybody tested this 5000 ? |
| 15:39.51 | blitzrage | variable_office: with transcoding... probably like 2-3 |
| 15:40.10 | blitzrage | variable_office: might be able to get more... hard to say... but 10-20 seems like a reasonable number.... |
| 15:40.16 | blitzrage | I might be guessing slightly high though |
| 15:40.22 | astserdev | :) |
| 15:40.23 | blitzrage | try it with sipp! |
| 15:40.30 | blitzrage | ~sipp |
| 15:40.31 | jbot | Single In-Line Pin Package: The last "standard" PC RAM configuration before they started making SIMMsA lot like SIMMs, but they have little pins instead of contacts. SIPPs are to VLB what SIMMs are to PCI.. A suicide tool for geeks |
| 15:40.35 | blitzrage | ugh! |
| 15:41.41 | blitzrage | jbot: no, sipp s a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/ |
| 15:42.03 | Strom_M | jbot: no, sipp is a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/ |
| 15:42.03 | jbot | Strom_M: okay |
| 15:42.15 | blitzrage | missing i! |
| 15:42.26 | *** join/#asterisk minkus (n=minkus@static-141-153-94-2.clrk.east.verizon.net) |
| 15:42.27 | blitzrage | Strom_M: show off |
| 15:42.37 | nain | Can any one tell me why sangoma is not detecting Callerid " SET(CALLERID(NUM)=${CALLERIDNUM}) |
| 15:42.46 | variable_office | blitzrage, thanks, i googled it right away anyways |
| 15:42.47 | Strom_M | nyaah nyaah nya nyaah nyaah |
| 15:42.57 | variable_office | blitzrage, i also found -> http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 15:42.59 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:43.00 | blitzrage | variable_office: no problem, thought I'd add it for future use in this channel |
| 15:43.11 | blitzrage | variable_office: thanks for googling and being self sufficient! |
| 15:43.18 | blitzrage | you just might make it kid |
| 15:43.55 | Strom_M | blitzrage: do you want to see some choice photos from the superlol training facility I'm teaching at this week? |
| 15:44.18 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 15:44.24 | *** join/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
| 15:44.32 | blitzrage | Strom_M: sure! |
| 15:44.39 | Strom_M | figure 1: http://www.flickr.com/photos/stromcarlson/877122825/in/set-72157600967906271/ |
| 15:44.57 | Strom_M | figure 2: http://www.flickr.com/photos/stromcarlson/877969480/in/set-72157600967906271/ |
| 15:45.18 | centrex | Strom_M, nice use of the cat V..... |
| 15:45.31 | jarrod | does the redhat cluster suite cost $$$ ? |
| 15:45.33 | nain | Any one have used Sangaoma A200r (2 Port FXO Card)??? |
| 15:45.46 | coolbeans | Anyone have a clue why in 1.2.18, when using static real-time and mysql, voicemail passwords aren't updated in the DB when changed with app_voicemail? It changes them in asterisk, but never updated to the db. Of course, a restart of app_voicemail restores whatever passwords are in the db. Any help would be appreciated. |
| 15:46.22 | Strom_M | centrex: yeah, i'm simultaneously amused and horrified |
| 15:48.09 | coolbeans | Best asterisk cluster = mysql + rsync |
| 15:48.11 | *** join/#asterisk CVirus (n=GoD@62.135.96.251) |
| 15:48.41 | CuriosCat | Hrm |
| 15:49.11 | CVirus | I added conf => 1234 to my meetme.conf and exten => 500,1,Meetme(1234) to my extensions.conf and when I dial 500 .. it says invalid conference number ! |
| 15:50.36 | Dr-Linux | How can i playback a any sound file while call is bridged, like, "you have 1 min left ans so on" ? |
| 15:51.00 | CuriosCat | Well, zttool recognizes my FXO module now. That took a little work, but it seems to be ok :) |
| 15:51.16 | CuriosCat | /etc/zaptel.conf was a little daunting until I remembered I just need one line :P |
| 15:52.13 | CuriosCat | I guess now I attack Asterisk itself :) |
| 15:52.20 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:54.22 | blitzrage | Strom_M: LOL!!! |
| 15:54.47 | Dr-Linux | Strom_M: any suggestoin for me? |
| 15:55.06 | CVirus | the answer to my question is to modprobe the ztdummy module ... forgot to do that |
| 15:55.28 | CuriosCat | I had to modprobe zttdm |
| 15:55.37 | tzafrir_laptop | zttdm? |
| 15:55.43 | CuriosCat | should probably put those in pmodprobe.conf so it keeps working after a reboot, actually |
| 15:55.52 | CuriosCat | err |
| 15:55.53 | CuriosCat | wctdm |
| 15:56.02 | dlynes_laptop | nain: What about the A200? |
| 15:56.16 | *** join/#asterisk kleofas (n=kleofas@router.dir.pl) |
| 15:56.22 | tzafrir_laptop | CuriosCat, actually chances are those are loaded at bot before the zaptel script is ever called. What distro you use? |
| 15:56.41 | dlynes_laptop | Dr-Linux: That's part of the dial command...check the options list for Dial() |
| 15:57.10 | CuriosCat | tzafrir: I'm using Fedora 7, but I'm building my own zaptel drivers and asterisk binaries, I'm not using a package. |
| 15:57.22 | Dr-Linux | dlynes_laptop: yes, my i wanna play warning message to the caller, my own sound file |
| 15:57.31 | Mercestes | CuriosCat, So are you inherently curious or do you collect small figurines and statuettes? |
| 15:57.38 | tzafrir_laptop | CuriosCat, still |
| 15:57.46 | CuriosCat | Mercestes: The nick is from "Curiosity killed the Cat" |
| 15:58.07 | CuriosCat | tzafrir: Does make install modify Fedora boot scripts? |
| 15:58.18 | Mercestes | CuriosCat, Ah, so you just can't spell, ok. |
| 15:58.26 | dlynes_laptop | Dr-Linux: So modify that portion of app_dial.c to change the file that it's using, or rename the existing file to something else, and then drop your file in as the old name instead |
| 15:58.27 | nain | Hi Any one have used Sangoma A200r FXO module ? |
| 15:58.28 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 15:58.32 | CuriosCat | Mercestes: You can't spell either. You royally screwed up "Mercedes". |
| 15:58.47 | dlynes_laptop | nain: I ALREADY ASKED YOU WHAT ABOUT IT |
| 15:58.48 | CuriosCat | And unlike "curious", "curiosity" only contains one u. |
| 15:59.08 | Mercestes | CuriosCat, Correct. But your nick isn't Curiosity cat. .. |
| 15:59.16 | Mercestes | CuriosCat, So I thought you might collect curios. |
| 15:59.29 | hi365 | just wondering if AgentCallbackLogin() can acept a queue as an agrument that it can dynamicaly login to? |
| 15:59.49 | tzafrir_laptop | CuriosCat, no. make config does |
| 15:59.55 | Dr-Linux | dlynes_laptop: maybe there is some patch .. but don't know the name |
| 16:00.01 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 16:00.11 | dlynes_laptop | Dr-Linux: Just rename the old file, and name your file the same as the existing file |
| 16:00.20 | dlynes_laptop | Dr-Linux: very simple fix, without needing a patch |
| 16:00.37 | dlynes_laptop | Dr-Linux: unless of course you want to adjust the times, and/or put in more than one file |
| 16:02.59 | Dr-Linux | dlynes_laptop: yeah, but difficult to find what's the file name in dial.c app |
| 16:03.10 | CuriosCat | Mercestes: Let me explain a bit about IRC history to you |
| 16:03.19 | CuriosCat | there's this thing called a nick length limitation |
| 16:03.23 | Mercestes | CuriosCat, Please don't. |
| 16:03.30 | CuriosCat | on EFnet, and before that, on IRCnet, it was nine characters. |
| 16:03.40 | CuriosCat | How would you abbreviate "Curiosity killed the Cat" to fit in nine characters? |
| 16:03.40 | Mercestes | *sighs* |
| 16:03.52 | Strom_M | CurKilCat |
| 16:04.03 | Mercestes | CuriosCat, http://internetarguing.ytmnd.com/ |
| 16:04.04 | CuriosCat | I like my version better, Strom :) |
| 16:04.23 | CuriosCat | Mercestes: If you didn't want an Internet argument, why did you start a pedantic quarrel with someone you don't know? |
| 16:04.42 | Mercestes | CuriosCat, I don't like children. You shouldn't say that. |
| 16:04.59 | Mercestes | not in that way anyways. |
| 16:05.17 | CuriosCat | Mercestes: I don't like dorks. Welcome to my ignore list. |
| 16:05.19 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:05.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:05.40 | Mercestes | Hrm. A new record. |
| 16:06.09 | CuriosCat | tzafrir: Gotcha. |
| 16:07.44 | CuriosCat | Well, with gcc-c++ in place, let's see if Asterisk will compile now :) |
| 16:07.54 | coolbeans | Anyone have a clue why in 1.2.18, when using static real-time and mysql, voicemail passwords aren't updated in the DB when changed with app_voicemail? It changes them in asterisk, but never updated to the db. Of course, a restart of app_voicemail restores whatever passwords are in the db. Any help would be appreciated. |
| 16:12.58 | *** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net) |
| 16:15.47 | jarrod | is there a app or module in asterisk i can reload so it will reload zaptel? i just upgraded zaptel, but i dont want to restart asterisk completely |
| 16:17.09 | astserdev | jarrod, i think you need to restart asterisk |
| 16:17.27 | dlynes_laptop | jarrod: unload chan_zaptel.so, and then load chan_zaptel.so |
| 16:17.37 | dlynes_laptop | jarrod: make sure you don't use the reload command |
| 16:18.03 | tzafrir_laptop | chan_zap.so |
| 16:18.22 | dlynes_laptop | jarrod: erm...what tzafrir_laptop said...i haven't had any coffee yet this morning |
| 16:18.26 | tzafrir_laptop | dlynes_laptop, is that safe if you have PRI? |
| 16:18.36 | [TK]D-Fender | jarrod: "reload chan_zap.so" <- 1 step |
| 16:18.40 | dlynes_laptop | tzafrir_laptop: yep...I've done it on my pri before |
| 16:19.03 | tzafrir_laptop | reload chan_zap.so won't change settings of existing channels |
| 16:19.26 | dlynes_laptop | [TK]D-Fender: does reload work properly with zap? I've used it with something less complicated like app_voicemail.so, and it doesn't reload all the settings |
| 16:19.48 | tzafrir_laptop | it works "properly" in the sense that it does no harm |
| 16:20.02 | tzafrir_laptop | But there are some settins it doesn't apply |
| 16:21.23 | *** join/#asterisk Flauto (n=zhao@71.194.141.225) |
| 16:21.24 | tzafrir_laptop | I wonder how I can automate this. How long I need to wait between the unload and load |
| 16:21.40 | Flauto | hello people |
| 16:21.50 | Dr-Linux | aww |
| 16:21.53 | Flauto | i have a problem with installing asterisk 1.4.8 |
| 16:22.01 | Dr-Linux | dlynes_laptop: did you see the "L" option in Dial() ? |
| 16:22.07 | Dr-Linux | i don't think |
| 16:22.21 | Flauto | when i ./configure, it says configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 16:22.36 | tzafrir_laptop | Flauto, which distro? |
| 16:22.48 | Flauto | fedora core 6 |
| 16:23.18 | dlynes_laptop | Flauto: Hey...long time, no see |
| 16:23.33 | Flauto | hello dlynes, yes, i was in china for almost 3 months |
| 16:23.37 | Flauto | how are you doing |
| 16:23.43 | Flauto | just got back last week |
| 16:23.51 | dlynes_laptop | Flauto: Great...met a super sweet Cantonese girl a few months ago |
| 16:23.52 | Flauto | now, i am trying to update my asterisk |
| 16:23.59 | Flauto | really |
| 16:24.06 | dlynes_laptop | Yeah...she's quite hot :p |
| 16:24.07 | Flauto | are you still going out with her? |
| 16:24.14 | Flauto | hehe |
| 16:24.15 | dlynes_laptop | Yeah...engaged to her |
| 16:24.25 | Flauto | man, congratulations |
| 16:24.29 | dlynes_laptop | Thanks |
| 16:24.34 | dlynes_laptop | I don't believe in wasting time |
| 16:24.42 | Flauto | haha |
| 16:24.43 | dlynes_laptop | Or letting someone else steal her away from me :) |
| 16:25.06 | Flauto | maybe because it is about the right time |
| 16:25.33 | Flauto | have not touched my asterisk for more than 3 months |
| 16:25.42 | dlynes_laptop | Yeah...we're both getting older, and we both want kids, so trying to rush it along a little bit faster |
| 16:25.55 | Flauto | good |
| 16:26.12 | Flauto | my wife and i are planning to have a kid next year |
| 16:26.27 | dlynes_laptop | You saw the new versions of asterisk available, right? |
| 16:26.42 | msetim | how can I know if a meet is locked/unlocked, asterisk manager can give it |
| 16:26.50 | Flauto | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 16:26.54 | Flauto | what is this for |
| 16:27.14 | dlynes_laptop | Flauto: It's probably producing something in the output that configure doesn't like |
| 16:27.26 | Flauto | what should i do |
| 16:27.31 | dlynes_laptop | Flauto: Could be one of those fedora/redhat/centos eccentric type things |
| 16:27.49 | dlynes_laptop | Flauto: Try a different version of fedora, or try a different distro if you want to fix the problem faster |
| 16:27.50 | Flauto | i was using 1.4.1 and it was okay |
| 16:28.02 | dlynes_laptop | Flauto: which one is bugging out? |
| 16:28.10 | Flauto | 1.4.8 |
| 16:28.17 | tzafrir_laptop | yum install gcc ? yum install cpp ? |
| 16:28.23 | dlynes_laptop | Flauto: gimme a sec |
| 16:28.32 | Flauto | thanks, tzafrir |
| 16:28.34 | Flauto | i will try that |
| 16:29.02 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:29.10 | Flauto | tzafrir, it is saying nothing to do for both |
| 16:29.17 | Flauto | i do have them installed |
| 16:29.52 | tzafrir | FlatFoot, glibc-devel ? look into config.log |
| 16:29.53 | msetim | somebody knows??? |
| 16:30.10 | dlynes_laptop | Flauto: I'm downloading it now to try it on slackware 10.22222222 |
| 16:30.13 | dlynes_laptop | grrr |
| 16:30.20 | dlynes_laptop | stupid lllllllaptop keyboarde |
| 16:30.40 | Flauto | i have never used slackware |
| 16:30.52 | tzafrir_laptop | msetim, I don't really understand your question |
| 16:31.08 | dlynes_laptop | Flauto: yeah, but maaaaybe the same problem you're encountering on fedora core 7, i will also experience on slackware 10.2 |
| 16:31.21 | dlynes_laptop | Flauto: what version do you get when you do a gcc --version? |
| 16:31.28 | dlynes_laptop | Flauto: what do you get from uname -a? |
| 16:31.32 | tzafrir_laptop | you ask if you can tell through the manager interface if a meetme (?) room(?) is locked or unlocked? |
| 16:31.59 | Flauto | Linux EncoreNetwork 2.6.20-1.2962.fc6 #1 SMP Tue Jun 19 19:27:14 EDT 2007 i686 i686 i386 GNU/Linux |
| 16:32.27 | Flauto | oh, you know wht, i think i updated the kernel |
| 16:32.37 | Flauto | i was using 2.6.18.... |
| 16:32.41 | Flauto | and it was working |
| 16:32.41 | dlynes_laptop | Flauto: well, for what it matters, I'm using 2.6.21 |
| 16:33.00 | msetim | tzafrir_laptop, I have a conference (meetme) and I would like to know if it is locked ( meetme lock <meetme_number> ) |
| 16:33.20 | Flauto | i think for fedora, 2.6.20 is the most up to date one |
| 16:33.33 | Flauto | maybe i can go back to 2.6.18 |
| 16:34.06 | dlynes_laptop | Flauto: Only time I ever use a prepackaged kernel is when I first install linux |
| 16:34.10 | Flauto | i did not have problem compiling zaptel and normally, that would be more problematic |
| 16:34.20 | dlynes_laptop | Flauto: After it gets installed, I always compile my own kernel |
| 16:34.28 | Flauto | oh |
| 16:34.35 | Flauto | i don't know how to do it |
| 16:34.36 | Flauto | hehe |
| 16:34.44 | *** join/#asterisk neoxo_tech (n=Michel@64.254.239.194) |
| 16:35.17 | neoxo_tech | greets, all |
| 16:35.51 | MrTelephone | hey are you there russell? |
| 16:36.29 | neoxo_tech | I'm having a seg-fault error with the mISDN driver in Asterisk with a B410P ... anyone here familiar with it? |
| 16:37.00 | Flauto | dlynes, i will try it on 2.6.18 |
| 16:37.11 | Flauto | do you use fwd? |
| 16:37.35 | Flauto | my number is 652969 |
| 16:37.56 | Flauto | okay, need to fix some food |
| 16:37.58 | Flauto | hungry |
| 16:38.00 | Flauto | talk to you later |
| 16:38.28 | dlynes_laptop | Flauto: cd /usr/src && ncftp ftp://ftp.kernel.org/pub/linux/kernel/v2.6/linux-2.6.21.6.tar.bz2 && tar jxvf linux-2.6.21.6.tar.bz2 && cd linux-2.6.21.6 && make xconfig |
| 16:38.52 | dlynes_laptop | Flauto: pretty straight forward :) |
| 16:39.34 | *** part/#asterisk stubert (i=stu@techtools.actusa.net) |
| 16:39.42 | dlynes_laptop | Flauto: but, I would change the makefile after you finish with xconfig, so that it dumps the kernel into its own dedicated directory, and then add that new kernel to your boot manager, instead of replacing the existing entry |
| 16:40.29 | dlynes_laptop | Flauto: look for INSTALL_PATH |
| 16:42.49 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 16:43.18 | CuriosCat | I shouldn't need libpri with just an FXO card, right? |
| 16:43.25 | jarrod | that worked guys |
| 16:43.26 | jarrod | thanks |
| 16:43.54 | Hmmhesays | what up folks |
| 16:44.24 | Flauto | dlynes, would you login to my computer? |
| 16:44.30 | Flauto | you know what you are doing |
| 16:44.33 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:44.49 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:45.18 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 16:48.14 | *** join/#asterisk Mrtaz (n=wcl@h-68-167-99-74.cmbrmaor.covad.net) |
| 16:49.20 | Mrtaz | hey all, im having a problem on incoming calls being very quiet and almost unable to be heard, using asterisk 1.4.8 and a TDM2400P board, calls are PTSN lines |
| 16:49.30 | Mrtaz | what can I do to fix it? |
| 16:53.05 | [TK]D-Fender | CuriosCat: Nope. |
| 16:53.31 | [TK]D-Fender | Mrtaz: txgain & rxgain in zapata.conf. Go read the sample configs |
| 16:53.45 | Mrtaz | so I did this, I increased them to 5.0 each |
| 16:53.57 | Mrtaz | how much can I go before the calls are distorted? |
| 16:54.39 | Mrtaz | the problem is even worse when trying to use 3 way calling with parties, usually the 3rd party cant hear the first |
| 16:54.50 | Flauto | okay, i am doing it now |
| 16:55.28 | CuriosCat | thanks fender |
| 16:55.30 | Flauto | dlynes, i have it downloaded |
| 16:56.46 | Flauto | dlynes, got this. make[1]: *** No rule to make target `scripts/kconfig/.tmp_qtcheck', needed by `scripts/kconfig/qconf.o'. Stop. |
| 17:00.28 | *** join/#asterisk tioan (n=kvirc@pd95b1d9d.dip0.t-ipconnect.de) |
| 17:00.30 | tioan | hello |
| 17:05.57 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 17:06.11 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 17:06.25 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 17:08.12 | [T]ank | I am trying to learn how to use GROUPCOUNT to limit the number of calls to a phone when the phone is added to a queue dynamically. this is what I tried: http://pastebin.ca/632158 My result with no other calls to that phone was this: http://pastebin.ca/632159 could anyone give me a hand? |
| 17:09.14 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 17:14.28 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-220-132.cablep.bezeqint.net) |
| 17:15.48 | fetcher | Anyone running HUD-Lite on AsteriskNow? |
| 17:15.54 | *** join/#asterisk taqua2008 (n=perdue@66.118.69.58) |
| 17:16.23 | fetcher | wondering about installing the needed dependencies via rPath / Conary |
| 17:18.24 | hi365 | im trying to use agentcallbacklogin im getting: |
| 17:18.25 | hi365 | Jul 24 20:17:27 VERBOSE[3387] logger.c: -- Executing AgentCallbackLogin("SIP/201-09897e70", "371||201") in new stack |
| 17:18.25 | hi365 | Jul 24 20:17:27 WARNING[3387] chan_agent.c: Extension '201' is not valid for automatic login of agent '371' |
| 17:18.36 | hi365 | whats wrong with 201?? |
| 17:22.42 | Mercestes | fetcher: #asterisknow |
| 17:22.53 | Sci_05 | ok anyone know about the bounty for the FOIP stuff? Does it have to be T.38 or are they just looking for faxing over asterisk? |
| 17:23.22 | Qwell[] | FoIP != FoVoIP |
| 17:24.14 | Sci_05 | Qwell right |
| 17:24.34 | hi365 | what is considered a valid extension for AgentCallbackLogin? |
| 17:24.37 | Sci_05 | but for the bounty does it have to be T.38 of are they looking for just getting FoVOIP to work with asterisk? |
| 17:24.47 | coppice | The thing I know about bounties for FoIP is nobody will ever pay them :-) |
| 17:24.58 | *** join/#asterisk tako-san (n=Tako-san@24.108.162.254) |
| 17:25.12 | Sci_05 | hy do you say that coppice ? |
| 17:25.15 | Sci_05 | hy=why |
| 17:25.38 | coppice | historical precedent. |
| 17:26.12 | coppice | now, if you can get FoVoIP to work solidly, you won't need bounties. You will be able to make a fortune as a magician |
| 17:26.52 | *** join/#asterisk robin_z (n=robin@rszemeti.gotadsl.co.uk) |
| 17:27.25 | robin_z | Hi guys, can anyone just confirm ... the setting in sip_nat.conf, they are totally irrelevant to IAX connections aren;t they? |
| 17:27.48 | [TK]D-Fender | hi365: Bad dialplan. |
| 17:28.01 | [TK]D-Fender | robin_z: ummm.. DUH?! |
| 17:28.04 | [TK]D-Fender | ;) |
| 17:28.14 | [TK]D-Fender | robin_z: IAX2 doesn't care much about NAT |
| 17:28.17 | Corydon76-work | coppice: I prefer IP over Avian Carrier for sheer inefficiency |
| 17:28.32 | robin_z | I have an IAX account with a provider .. and they keep telling me to "check you have sip_nat.conf set up right" .... |
| 17:28.59 | coppice | someone was recently trying to put a fun demonstrator together of VoIP over cans and string |
| 17:29.12 | Corydon76-work | robin_z: I wasn't aware there was such a file |
| 17:29.12 | robin_z | [TK]D-Fender, thats exactly what I thought .. I was just checking ... before I tell them to get some clue. |
| 17:29.19 | [TK]D-Fender | robin_z: Ask them to reroute the plasma flow through the EPS conduits, and reverse polarity.... |
| 17:29.33 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 17:29.52 | Strom_M | coppice: i had a joke I used to tell about voice over tin cans that included a fictitious Digium four-port tin can interface card called the "TC400B" |
| 17:30.25 | Strom_M | but now that that's a real product that merely makes you /sound/ like you're on a tin can, the joke doesn't work so well anymore :) |
| 17:30.31 | robin_z | bearing in mind I connect from * /// through NAT and to them, they also seem to think I should worward the IAX port from the external IP .. through the firewall and to my * box ... |
| 17:30.53 | robin_z | that also sounds like bollox to me, as it will appen automagically as the outgoing connection sets itself up |
| 17:31.18 | Corydon76-work | It's baloney if you're registering your IAX connection to them |
| 17:31.33 | robin_z | exactly what I thought |
| 17:31.38 | Corydon76-work | If you're not registering, it's the only way to get the packets through |
| 17:31.54 | robin_z | this is for outgoing ONLY |
| 17:32.08 | Corydon76-work | Then yes, it's baloney |
| 17:32.13 | robin_z | phew |
| 17:32.36 | coppice | Its funny seeing how people react to that bounty page for T.38. Various people have got excited by it, without having a clue what is involved |
| 17:32.43 | Strom_M | an incompetent ITSP!? |
| 17:32.46 | Strom_M | who knew |
| 17:34.02 | *** join/#asterisk mtaht4 (n=m@cpe-74-76-23-86.nycap.res.rr.com) |
| 17:34.52 | tako-san | [TK]D-Fender: You got a sec? |
| 17:35.09 | [TK]D-Fender | tako-san: If thats all you require ;) |
| 17:35.22 | tako-san | [TK]D-Fender: Times up, huh? :) |
| 17:35.40 | [TK]D-Fender | tako-san: Just ask... |
| 17:35.58 | tako-san | [TK]D-Fender: I am still having the issue where when trying to make an outbound call the caller will get a dial tone. |
| 17:36.20 | tako-san | [TK]D-Fender: I have inserted "ww" in the trunk configuration but it does not seem to be helping. |
| 17:36.27 | tako-san | [TK]D-Fender: Other suggestions? |
| 17:36.34 | Strom_M | tako-san: show me your Dial() line |
| 17:36.36 | [TK]D-Fender | tako-san: pastebin what you're doing, and the CLI of what happens |
| 17:36.39 | *** join/#asterisk whatwherewhen (i=whatwher@196.211.34.3) |
| 17:36.40 | [TK]D-Fender | ~pb |
| 17:36.41 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:37.00 | tako-san | [TK]D-Fender: Alrighty. |
| 17:37.38 | whatwherewhen | hi there anyone know what the latest version book is and if there is one covering all the aspects of the new dialplan applications? |
| 17:37.45 | hi365 | [TK]D-Fender: I have this in my dial plan: exten => s,n,AgentCallbackLogin(${AMPUSER}|s|${CALLERID(num)}@from-internal) |
| 17:37.52 | whatwherewhen | in asterisk 1.4 that is? |
| 17:38.06 | Strom_M | whatwherewhen: new edition will be out shortly |
| 17:38.18 | tako-san | [TK]D-Fender: Zaptel.conf? zapata.conf? what all do you want in the pastebin? |
| 17:38.19 | hi365 | [TK]D-Fender: tried doing from @local or with no context. it seems to work for a while then i get the invalid extnesion error |
| 17:38.20 | [TK]D-Fender | hi365: Executing AgentCallbackLogin("SIP/201-09897e70", "371||201") in new stack |
| 17:38.27 | whatwherewhen | any dates yet?: strom_M? |
| 17:38.35 | [TK]D-Fender | hi365: this does NOT have the context in this new "version" you've decided to show me. |
| 17:38.45 | [TK]D-Fender | hi365: Try not to mix your apples & oranges. |
| 17:38.54 | Strom_M | whatwherewhen: i think they're saying next month |
| 17:39.01 | hi365 | [TK]D-Fender: thats right |
| 17:39.16 | [TK]D-Fender | hi365: So show me the NEW code & NEW error. |
| 17:39.20 | whatwherewhen | where can i find the most up to date for now?:StromM? |
| 17:39.40 | [TK]D-Fender | hi365: And of course everything to backup why you think it should be working. |
| 17:40.00 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 17:40.02 | hi365 | [TK]D-Fender: exten => s,n,AgentCallbackLogin(${AMPUSER}|s|${CALLERID(num)}@from-internal) |
| 17:40.07 | hi365 | Jul 24 20:36:55 VERBOSE[3692] logger.c: -- Executing AgentCallbackLogin("SIP/201-096abc38", "371|s|201@from-internal") in new stack |
| 17:40.07 | hi365 | Jul 24 20:36:55 WARNING[3692] chan_agent.c: Extension '201' is not valid for automatic login of agent '371' |
| 17:40.09 | [TK]D-Fender | whatwherewhen: "show applications" <------------- |
| 17:40.20 | [TK]D-Fender | hi365: PASTEBIN. |
| 17:40.24 | [TK]D-Fender | hi365: ALL of it. |
| 17:40.30 | hi365 | [TK]D-Fender: k |
| 17:40.32 | whatwherewhen | k cheers |
| 17:41.53 | Mrtaz | so any other suggestions for increasing call decible level besides rxgain/txgain for a tdm2400p board? is there anything else I can do? |
| 17:41.53 | tako-san | [TK]D-Fender: Here are my zaptel.conf and zapata.conf files "http://pastebin.ca/632188" |
| 17:42.25 | [TK]D-Fender | tako-san: pastebin the CLI output of your call attempt |
| 17:42.37 | [TK]D-Fender | Mrtaz: No. |
| 17:43.04 | neverblue2 | any VOIP providers in the channel today? |
| 17:43.10 | hi365 | [TK]D-Fender: it should all be there (warning: lots of freepbx stuff there, although not related to what im doing) |
| 17:43.13 | hi365 | http://pastebin.ca/632192 |
| 17:43.25 | neverblue2 | looking for service |
| 17:43.25 | Mrtaz | [TK]D-Fender: what is an acceptable max level for rxgain/txgain? |
| 17:43.32 | denon | 0 |
| 17:43.36 | denon | :) |
| 17:44.15 | [TK]D-Fender | hi365: I'm missing the relevant dialplan.... |
| 17:44.39 | hi365 | [TK]D-Fender: do you mean the macro? |
| 17:44.59 | [TK]D-Fender | hi365: 201@from-internal |
| 17:45.07 | [TK]D-Fender | hi365: The whole CONTEXT |
| 17:45.44 | *** join/#asterisk red9012 (n=marc3234@206-248-133-111.dsl.teksavvy.com) |
| 17:45.45 | hi365 | [TK]D-Fender: you dont want to see it :) let me give you the log when without using a context |
| 17:45.58 | [TK]D-Fender | hi365: Yes, I DO want to see it. |
| 17:46.08 | red9012 | how can I handle faxing using asterisk? |
| 17:46.11 | [TK]D-Fender | hi365: I just asked for it, and shouldn't have even HAD to. |
| 17:46.36 | tzanger | red9012: use t38modem and h323, or callweaver |
| 17:46.39 | [TK]D-Fender | hi365: God helps those who help themselves.... I am far LESS forgiving ;) |
| 17:46.59 | hi365 | [TK]D-Fender: tell me about it! |
| 17:47.00 | hi365 | lol |
| 17:48.32 | tako-san | [TK]D-Fender: http://pastebin.ca/632198 |
| 17:49.20 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 17:49.45 | [TK]D-Fender | tako-san: Dial("IAX2/310-2", "ZAP/g0/ww4778420|300|") in new stack |
| 17:50.05 | [TK]D-Fender | tako-san: You don't have the EXTRA delay between the PIN like I showed you, nor the "9" you said you needed yesterday |
| 17:50.17 | hi365 | [TK]D-Fender: http://pastebin.ca/632201 |
| 17:50.42 | tako-san | [TK]D-Fender: Yesterday? I don't think we have talked since Friday or perhaps Saturday? |
| 17:50.46 | *** part/#asterisk galeras (n=root@201.245.103.169) |
| 17:51.01 | tako-san | [TK]D-Fender: And what do you mean by PIN? |
| 17:51.54 | [TK]D-Fender | hi365: I don't see 201 as valid in there... |
| 17:51.57 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 17:52.15 | *** join/#asterisk Victor_Yure (n=aaaa@postfix.tradein.com.br) |
| 17:52.20 | [TK]D-Fender | tako-san: yesterday you said you had to dial "9", and a pin (access code or something, and them your destination # IIRC |
| 17:52.32 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:52.35 | [TK]D-Fender | tako-san: yesterday = any variable time in the recent past ;) |
| 17:52.37 | tako-san | [TK]D-Fender: No PINS needed on this setup. |
| 17:52.47 | tako-san | [TK]D-Fender: Perhaps you are getting me confused with someone else? |
| 17:53.01 | [TK]D-Fender | tako-san: Possible.. |
| 17:53.02 | tako-san | [TK]D-Fender: And we dont dial 9 to get an outside line either. |
| 17:53.11 | [TK]D-Fender | tako-san: Similar nick & needs |
| 17:53.14 | tako-san | [TK]D-Fender: Sorry for any confusion |
| 17:53.19 | tako-san | [TK]D-Fender: Could be |
| 17:53.23 | [TK]D-Fender | tako-san: -no biggie. |
| 17:53.37 | [TK]D-Fender | tako-san: So what exactly DO you need? |
| 17:54.11 | hi365 | [TK]D-Fender: what is it exactly that your looking for in the dial plan? |
| 17:54.26 | [TK]D-Fender | hi365: Show me where 200@from-internal is valid. |
| 17:54.37 | tako-san | [TK]D-Fender: Outbound calls are randomly returned to a dialtone. You suggested inserting a ww or 2 |
| 17:54.43 | hi365 | [TK]D-Fender: ill try. 1 min |
| 17:54.44 | tako-san | sorry a w or 2 |
| 17:55.04 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 17:55.21 | tako-san | [TK]D-Fender: I have done that in the trunk configuration. Would it make a difference in the outbound route instead? or did i misunderstand you completely? |
| 17:56.10 | [TK]D-Fender | tako-san: What kind of interface are you using, and exactly where you you think you need to insert this delay? |
| 17:56.25 | hi365 | [TK]D-Fender: now i get it, your right (as usual) |
| 17:57.39 | tako-san | [TK]D-Fender: I am using freepbx to do some of the configuration (as per the clients request). I inserted the "ww" in the oubound dial prefix of my main zap trunk. I can hear there is a delay now but I am still having the same problem. |
| 17:57.58 | tako-san | [TK]D-Fender: So I was curious what other area I could look in to that might be causing that return to dial tone. |
| 17:58.04 | robin_z | sigh ... |
| 17:58.19 | robin_z | so I had GREAT results with this new provider, Gradwell.com |
| 17:58.25 | robin_z | for about 2 weeks |
| 17:58.33 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 17:58.38 | robin_z | now I seem to have a lot of calls that fail to connect |
| 17:59.00 | robin_z | and quite often I lose the outgoing audio, I can hear the other party, they can;t hear me |
| 17:59.08 | [TK]D-Fender | tako-san: your gain is psycho low from what I recall... that may be part of it. |
| 17:59.30 | tako-san | [TK]D-Fender: Really. That could be a possible cause of the problem? |
| 17:59.51 | Strom_M | tako-san: when it "returns to dial tone" where is the dial tone coming from? |
| 18:00.45 | tako-san | Strom_M: It is hard to distinguish between the dial tone of the PBX and that of the telco. I would be happy to hear of a way to determine which one I am hearing. :) |
| 18:01.29 | Strom_M | tako-san: when you get the second dial tone, type "show channels" at the CLI and see if you're currently bridged to the zaptel channel |
| 18:01.53 | tako-san | Ok thanks. |
| 18:02.02 | *** join/#asterisk Vulpyne (n=vulpyne@sta-207-174-202-66.rockynet.com) |
| 18:02.42 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 18:02.42 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 18:02.42 | *** join/#asterisk gzero (n=gzero@81.175.82.2) [NETSPLIT VICTIM] |
| 18:02.43 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) [NETSPLIT VICTIM] |
| 18:02.43 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-249-118-204.dsl.ipltin.ameritech.net) [NETSPLIT VICTIM] |
| 18:02.43 | *** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) [NETSPLIT VICTIM] |
| 18:02.43 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 18:02.44 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) [NETSPLIT VICTIM] |
| 18:02.44 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) [NETSPLIT VICTIM] |
| 18:02.44 | *** join/#asterisk ruskie (i=ruskie@sourcemage/mage/ruskie) |
| 18:02.44 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 18:02.44 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 18:02.44 | *** mode/#asterisk [+o file] by irc.freenode.net |
| 18:04.12 | IOscanner | Has anyone setup a USB sound device to connect an overhead speaker system? |
| 18:04.40 | [TK]D-Fender | IOscanner: If you can set it up with ALSo I suppose it'll work. |
| 18:04.42 | Rienzilla | Hey... would it somehow be possible to connect an asterisk pbx to a ventrilo server? (for example by writing a new channel module) in order to use a sip client to talk to people on a ventrilo server? |
| 18:04.43 | [TK]D-Fender | ALSA* |
| 18:05.00 | tako-san | Strom_M: It would appear the zaptel channel is still bridged http://pastebin.ca/632215 |
| 18:05.10 | Strom_M | tako-san: alright, so you're getting a telco dial tone |
| 18:05.17 | tako-san | Right |
| 18:05.17 | Strom_M | now comes the fun part |
| 18:05.22 | Strom_M | do you have a buttset handy? |
| 18:05.53 | tako-san | Strom_M: Unfortunately I am not on-site at the moment. And I do not have a set handy though I can get my hands on one. |
| 18:06.05 | tako-san | Strom_M: What do I need to do? |
| 18:06.13 | tako-san | Strom_M: I will take notes now and do it when I get on-site |
| 18:06.14 | Strom_M | tako-san: you need to clip onto the circuit and see what's actually happening |
| 18:06.24 | Strom_M | monitor while the buttset is on-hook |
| 18:06.28 | *** part/#asterisk Vulpyne (n=vulpyne@sta-207-174-202-66.rockynet.com) |
| 18:07.00 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 18:07.00 | tako-san | Strom_M: I got the general idea. Is there something specific I am looking for? Or just make general observations? |
| 18:07.06 | Strom_M | make observations |
| 18:07.10 | tako-san | ok |
| 18:07.21 | Strom_M | figure out at which point the circuit resets to dial tone |
| 18:07.22 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 18:07.24 | Katty | weeee!!! |
| 18:07.32 | *** join/#asterisk saftsack (n=oliver@p54A7EE06.dip.t-dialin.net) |
| 18:07.35 | tako-san | ok |
| 18:08.20 | IOscanner | Yeah I think so too, but I was looking to see if someone has done it before. |
| 18:10.01 | Fulk | shiny |
| 18:10.06 | Fulk | is it brushed aluminium? |
| 18:10.34 | [TK]D-Fender | Fulk: Brushed would make it DULL <- |
| 18:11.09 | Qwell[] | maybe it's brushed stainless steel? |
| 18:11.09 | Fulk | maybe it's been chavved up, and is transparent with neon lights :P |
| 18:11.15 | Qwell[] | or brushed chrome |
| 18:12.03 | Fulk | I have to make do with a white box |
| 18:12.07 | Fulk | how 90's |
| 18:12.19 | Fulk | actually, it's more yellowy than white, the case is that old |
| 18:12.30 | Qwell[] | stop smoking in your server room :p |
| 18:12.45 | Fulk | server room, it's sitting on the floor |
| 18:12.51 | Fulk | :-( |
| 18:15.20 | Strom_M | http://www.stromcarlson.com/misc/lolte410p-small.jpg |
| 18:16.17 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 18:16.18 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 18:16.18 | *** join/#asterisk gzero (n=gzero@81.175.82.2) [NETSPLIT VICTIM] |
| 18:16.18 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) [NETSPLIT VICTIM] |
| 18:16.18 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-249-118-204.dsl.ipltin.ameritech.net) [NETSPLIT VICTIM] |
| 18:16.18 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 18:16.19 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) [NETSPLIT VICTIM] |
| 18:16.19 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) [NETSPLIT VICTIM] |
| 18:16.19 | *** join/#asterisk ruskie (i=ruskie@sourcemage/mage/ruskie) [NETSPLIT VICTIM] |
| 18:16.19 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 18:16.19 | *** mode/#asterisk [+o file] by irc.freenode.net |
| 18:17.40 | Strom_M | i guess that must have been one hell of a lol |
| 18:17.52 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 18:18.52 | *** join/#asterisk E1ven (n=chatzill@SQ7/ProjectLead/E1ven) |
| 18:19.17 | kombi | people, is it thinkable to build a queue before a conference room? |
| 18:19.46 | Strom_M | ...why? |
| 18:20.05 | kombi | special project.. |
| 18:20.20 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:20.25 | Qwell[] | Strom_M: nice |
| 18:20.30 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 18:20.32 | kombi | only 3 people max in the room where the 3rd keeps changing |
| 18:20.55 | Strom_M | Qwell[]: :D |
| 18:21.06 | Qwell[] | took me a minute to figure out the joke :p |
| 18:21.33 | Strom_M | the joke is that the training facility here doesn't know the difference between 3.3 and 5v PCI slots |
| 18:21.44 | mocker | Hmm, going to need a replacement for iaxtel.com if it doesn't ever come back. :( |
| 18:22.27 | kombi | StromM: oh, the "why" wasn't for me then, sorry.. |
| 18:22.38 | Qwell[] | kombi: yes it was |
| 18:22.45 | Strom_M | kombi: yes it was |
| 18:22.59 | kombi | Qwell: why, thanks for the great input you both |
| 18:25.36 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 18:25.47 | E1ven | Hrmm.. Is there a way to say "Every X calls, do foo?" It seems like I could use a GotoIf, and use modulo on against the EPOCH, or against a counter, but if there any examples that are out there, I'd love to see them. |
| 18:26.18 | blitzrage | E1ven: just set a global var as a counter and use that (or, use astdb) |
| 18:27.38 | kombi | modulo agains epoch sounds good too if you need time intervalls rather than number of calls |
| 18:28.12 | E1ven | It doesn't need to be exact, I just want to do some Load Balancing. |
| 18:28.24 | E1ven | Fair enough. Thanks ;) |
| 18:28.49 | kombi | load balancing round robin style? |
| 18:28.53 | E1ven | Yeah. |
| 18:29.01 | kombi | bad idea.. |
| 18:29.04 | E1ven | It's cheezy, but it'll work. |
| 18:29.10 | Hmmhesays | I walk an endless mile |
| 18:30.33 | kombi | is Fender on vacation? |
| 18:30.40 | Hmmhesays | heh |
| 18:31.28 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:31.40 | *** join/#asterisk saftsack (n=oliver@p54A7DEF5.dip.t-dialin.net) |
| 18:31.44 | hi365 | is it posible to have hint for non-sip extensions in 1.2? |
| 18:31.44 | davixx | how to transform a number to a SIP URL |
| 18:33.07 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 18:35.10 | kombi | as much as I dislike Fender's way of never giving straight answers, it is quiet here without him |
| 18:35.49 | [TK]D-Fender | hi365: yes. SIP, IAx2, Zap, etc. |
| 18:35.57 | kombi | whoops..;) |
| 18:36.13 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 18:36.18 | [TK]D-Fender | davixx>how to transform a number to a SIP URL <- huh?! |
| 18:36.19 | *** part/#asterisk Victor_Yure (n=aaaa@postfix.tradein.com.br) |
| 18:36.23 | *** join/#asterisk coolfreecode (n=jimmy@190.41.82.1) |
| 18:36.24 | kombi | hey Fender! |
| 18:37.09 | davixx | [TK]D-Fender, how to ask my sip proxy to call a "real" number ? i have to give it a sip:URI no ? |
| 18:37.30 | *** join/#asterisk zpertee (n=chatzill@cpe-65-189-209-131.neo.res.rr.com) |
| 18:37.36 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:37.52 | zpertee | has anyone used zapmicro fxs/fxo pci cards? |
| 18:37.59 | [TK]D-Fender | davixx: What SIP "proxy"? * is NOT a proxy. and what is a "REAL NUMBER"? |
| 18:38.11 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 18:39.20 | davixx | [TK]D-Fender, using pjsua i register to an asterix box, and i can make a call, but to make a call i have to give the SIP URL of the remote contact |
| 18:39.20 | kombi | Fender, can you think of a way to put a queue before a conference? |
| 18:39.32 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:39.57 | Katty | should i get a t1 card that's pci, or pci express? |
| 18:40.20 | dmz | why do you want a queue before a conference? |
| 18:40.22 | Katty | is there any real advantage gained with pci express? besides a smidgen of speed, obviously. |
| 18:40.37 | Hmmhesays | depends on what you are using it for goofball |
| 18:40.45 | kombi | Katty: are there t1 cards for PCIe yet? |
| 18:41.04 | Katty | kombi: no, just pci and pci express. |
| 18:41.06 | [TK]D-Fender | kombi: How/why? |
| 18:41.13 | Katty | kombi: at leaset on voip0supply, anyway |
| 18:41.23 | kombi | dmz: special project requires it |
| 18:41.42 | dmz | is it possible to have 2 asterisk boxes with incoming sip connections going to them randomly and have some way for them both to know who is connected to which so if someone wanted to call another person it would know the stat of the connections & peers on the other box & bridge there if necessary? |
| 18:41.47 | [TK]D-Fender | kombi: What does the Queue do? |
| 18:41.59 | kombi | Katty: I'd say some performance advantage at extreme loads |
| 18:42.10 | Katty | yeah we don't have extreme loads. |
| 18:42.11 | dmz | kombi (good nick, i have a vw kombi:) |
| 18:42.44 | kombi | Fender: tell people they are pos x until eventually they are allowed in |
| 18:42.53 | dmz | kombi, you can easily put a queue before a conference. just have the context of the queue drop them into your conference |
| 18:42.56 | kombi | dmz: south africa? |
| 18:43.34 | dmz | kombi, no us: http://en.wikipedia.org/wiki/Volkswagen_Type_2 |
| 18:43.57 | kombi | dmz: a type 2, kewl.. does it drive well? |
| 18:45.29 | [TK]D-Fender | kombi: What will decide to let them in? |
| 18:45.55 | *** join/#asterisk galeras (n=root@201.245.103.169) |
| 18:45.56 | kombi | Fender: next in line, or, former caller hung up |
| 18:46.21 | [TK]D-Fender | kombi: if being next in line gets you in, then EVERYBODY gets in... just in fast order :) |
| 18:46.30 | *** join/#asterisk phessler_ (n=phessler@gir.theapt.org) |
| 18:46.52 | [TK]D-Fender | kombi: You can do this with a static dialplan "agent" where you script the check. |
| 18:47.07 | kombi | Fender: lol.. understood, some gate necessary.. |
| 18:47.19 | phessler_ | hi, I'm running into a weird problem. when asterisk Answers() an analog line, you hear a dialtone in the background. if asterisk passes the call to a SIP client, no problems |
| 18:47.24 | phessler_ | details at http://pastebin.com/d42585efc |
| 18:48.00 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:48.08 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 18:48.08 | kombi | Fender: what logic would the check follow? One below theshold -> allow one in? |
| 18:48.45 | Hymie | phessler_: sounds like you're answering the wrong line |
| 18:48.57 | Hymie | phessler_: or, you have a crossed wire |
| 18:49.05 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 18:49.10 | phessler_ | hmm |
| 18:49.26 | kombi | Fender: Anyway, some serious coding ahead.. |
| 18:49.32 | phessler_ | how can I tell if I am answering the correct line? |
| 18:49.41 | mocker | phessler_: You wind up talking to someone? |
| 18:49.45 | mocker | :) |
| 18:49.49 | phessler_ | hah |
| 18:50.09 | phessler_ | well, asterisk plays the correct sound file, and I can hear that |
| 18:50.27 | kombi | phessler_: you should get a hint from CLI too |
| 18:50.50 | phessler_ | when I type in an extension, it only acknowledges the 2nd digit, and restarts the prompt. at this point, no dial tone and everything Just Works |
| 18:50.56 | ccesario | hiii somebody have ideia about this error ? check_auth: username mismatch, have <8299>, digest has <8212> ... ? |
| 18:51.03 | phessler_ | CLI only refeers to Zap/10-1 |
| 18:51.17 | phessler_ | simple switch, Answer, Playback, Hungup, etc |
| 18:51.26 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:51.51 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 18:51.55 | [TK]D-Fender | kombi: its up to you to define why you're not letting everyone in.... |
| 18:52.22 | phessler_ | the pastebin above has the output from the console during one of these calls |
| 18:53.07 | *** join/#asterisk sashion (n=sdgsdg@dsl-244-213-32.telkomadsl.co.za) |
| 18:53.07 | kombi | Fender: true, I'd have to get the number of callers in a conference from manager and decide accordingly |
| 18:53.10 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 18:53.23 | sashion | What happened to asterisk-ss7 ? |
| 18:55.23 | [TK]D-Fender | kombi: Something like that. |
| 18:55.37 | [TK]D-Fender | kombi: Not terribly difficult |
| 18:55.56 | kombi | not terribly easy too..(; |
| 18:56.00 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:56.45 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:57.39 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.202.140) |
| 18:57.59 | dmz | kombi, it's in perfect shape :) |
| 18:58.44 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 18:58.53 | kombi | like to hear that.. |
| 19:05.01 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 19:05.01 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- *Critical Updates* Asterisk 1.2.22 and 1.4.8 released (July 17, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 19:05.21 | kombi | cannot make head or tail of it.. don't see a reason for * to do lookups at all |
| 19:05.23 | *** join/#asterisk logyati (n=logyati@201.29.26.188) |
| 19:05.28 | logyati | hello |
| 19:05.33 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 19:05.49 | logyati | does asterisk support *.ogg for voice menus? |
| 19:06.43 | pigpen | I am working with Realtime. I found this, "The REALTIME() function is now available in version 1.4 and app_realtime has been deprecated in favor of the new function." |
| 19:06.48 | kombi | logyati: to my knowledge it does not |
| 19:07.01 | Qwell[] | it can in 1.4 |
| 19:07.16 | logyati | hmmm |
| 19:07.23 | Qwell[] | and 1.2 for that matter |
| 19:07.29 | pigpen | I have found several references to RealTime() being used in the dialplan, but I cannot find the new REALTIME function documentations for dialplan usage. Any ideas where I can find this? |
| 19:07.45 | logyati | thats bad, since ogg is a free format |
| 19:07.50 | Strom_M | pigpen: core show function REALTIME |
| 19:07.56 | Qwell[] | logyati: Why is that bad? |
| 19:07.59 | *** join/#asterisk AdamB0122 (n=Adam@207.200.28.175) |
| 19:08.14 | pigpen | ah..thank you. |
| 19:08.34 | kombi | Qwell + Strom: are you attached? |
| 19:08.42 | logyati | for exemple, i use linux, and i wanna build a voice menu to my asterisk... gnome sound recorder uses ogg and wav format |
| 19:08.52 | Strom_M | attached? |
| 19:09.07 | pigpen | Strom_M, I think he means as "at the hip" |
| 19:09.17 | Strom_M | logyati: wav, 16-bit, 8khz, mono |
| 19:09.28 | logyati | Qwell[], wich formats does * suport |
| 19:09.33 | Qwell[] | logyati: So how is it bad that it supports ogg? |
| 19:09.50 | logyati | its good, support to ogg |
| 19:09.51 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:09.52 | kombi | pigpen: lol.. |
| 19:10.01 | Qwell[] | so then what is bad? |
| 19:10.05 | logyati | its bad it exist only in 1.4 |
| 19:10.09 | [TK]D-Fender | logyati: use * to make your recordings. |
| 19:10.13 | logyati | it should be in 1.2 too hehe |
| 19:10.18 | Qwell[] | I just said it was |
| 19:10.48 | *** join/#asterisk BZBW (n=wlwzhang@static-72-72-74-210.bstnma.east.verizon.net) |
| 19:11.54 | logyati | [TK]D-Fender, how? something like rec()? |
| 19:12.09 | [TK]D-Fender | logyati: "show application record" |
| 19:12.12 | kombi | Record() |
| 19:12.16 | logyati | [TK]D-Fender, im searching the book, but i cant find this application |
| 19:12.24 | logyati | [TK]D-Fender, ty |
| 19:12.28 | kombi | type that in CLI |
| 19:12.42 | [TK]D-Fender | logyati: "show applications" <- and then "show application [appname]" Do this for ALL of them. |
| 19:13.29 | [TK]D-Fender | logyati: When you're done with that, "show functions" and "show function [FUNCTION-IN-ALL-CAPS]" |
| 19:14.02 | logyati | [TK]D-Fender, oh, i didnt know this usefull comand |
| 19:15.03 | kombi | nice name for a band |
| 19:16.23 | [TK]D-Fender | kombi: What is? |
| 19:16.36 | kombi | above line |
| 19:16.45 | kombi | maybe not.. |
| 19:16.46 | [TK]D-Fender | kombi: Which? |
| 19:17.43 | kombi | there once was "I've got a fuzzbox and I'm gonna use it", in my head I just substituted fuzzbox with "useful command" |
| 19:18.23 | pigpen | Ok, with: exten => 300,1,REALTIME(sipusers|accountcode|204) |
| 19:18.40 | pigpen | I am getting: app_realtime.c:189 realtime_exec: The RealTime application has been deprecated in favor of the REALTIME dialplan function. |
| 19:19.18 | pigpen | Hmm, maybe it is just a general bitch. |
| 19:20.21 | Strom_M | pigpen: uh |
| 19:20.28 | Strom_M | use the function, not the application :) |
| 19:21.34 | *** join/#asterisk ManxPower (n=manxpowe@032-393-989.area5.spcsdns.net) |
| 19:22.24 | pigpen | yeah..I am going to use it in a Set command....like: |
| 19:22.25 | pigpen | exten => 300,1,SET(SIMNUM=${REALTIME(sipusers|accountcode|204)}) |
| 19:22.30 | pigpen | better? |
| 19:22.57 | Katty | ideas, anyone? |
| 19:23.02 | Mercestes | Katty, steaks |
| 19:23.04 | [TK]D-Fender | Katty: Gotta Be KD! |
| 19:23.11 | Katty | [TK]D-Fender: KD? |
| 19:23.20 | Katty | Mercestes: meh, steak :< |
| 19:23.22 | [TK]D-Fender | Katty: .....Kraft Dinner..... |
| 19:23.33 | Katty | [TK]D-Fender: ewww. |
| 19:23.41 | Katty | [TK]D-Fender: no thanks, i'll just make something for real :P |
| 19:24.31 | kombi | the missing link: how do I acually put logic into an extension BEFORE it picks up? |
| 19:24.55 | [TK]D-Fender | Katty: Yeah... a nice slab of steak on the BBQ.... now were talking.... |
| 19:25.08 | [TK]D-Fender | Katty: Say..... have we de-vegen'd you yet? ;) |
| 19:25.16 | [TK]D-Fender | vegan* |
| 19:25.19 | Katty | i haven't been vegan for... |
| 19:25.21 | Katty | 7 months. |
| 19:25.27 | Katty | no, 6 months. |
| 19:25.28 | [TK]D-Fender | kombi: don't ANSWER first :) |
| 19:25.40 | kombi | Katty, you are disco, where do we show up? |
| 19:25.48 | [TK]D-Fender | Katty: Congratulations on becoming a Born Again Carnivore :D |
| 19:25.52 | Katty | disco does not parse, kombi |
| 19:25.53 | kombi | Fender: my speciality it seems..;) |
| 19:26.41 | kombi | Katty: good one.. s/disco/grand/ |
| 19:27.12 | Katty | oh. |
| 19:27.17 | Katty | my house isn't open to strangers ;) |
| 19:27.18 | Corydon76-work | Uh, just because she's not vegan doesn't mean she's not vegetarian |
| 19:27.22 | Katty | the doggy will probably eat you. |
| 19:27.46 | kombi | Katty: don't worry, at least I'm a 1000 miles away |
| 19:27.50 | Katty | :P |
| 19:27.56 | Mercestes | Katty: Oh, you said *VEGAN*. For a second there this converstaion was really interesting |
| 19:28.25 | Katty | Mercestes: what were you thinking? |
| 19:28.34 | Mercestes | Katty: Of yoru first time. |
| 19:28.39 | sevard | Mercestes: I love you, man. |
| 19:28.47 | Mercestes | sevard: I love you too! |
| 19:28.50 | Katty | we're not discussing that in #asterisk |
| 19:28.52 | sevard | ONARRZZZ |
| 19:28.54 | sevard | +B |
| 19:28.59 | kombi | now, what's the matter with you here.. |
| 19:29.01 | Mercestes | Katty: I penciled myself in btw, I hope you don't mind. |
| 19:29.08 | Mercestes | Katty: Can I msg you? |
| 19:29.13 | Katty | Mercestes: uhh, sure. |
| 19:29.14 | sevard | Mercestes: MAY |
| 19:29.22 | Mercestes | s/can/may/ |
| 19:29.31 | Katty | sevard: no, it's july. |
| 19:29.34 | Katty | sevard: :P |
| 19:29.37 | sevard | you're dumb. |
| 19:29.56 | Katty | i'm not dumb. |
| 19:30.16 | Katty | still working on smart, perhaps... but certainly not dumb (= |
| 19:30.17 | [TK]D-Fender | yeah... she's even a brunette! |
| 19:30.22 | Katty | YEAH |
| 19:30.24 | Mercestes | Corydon76-work. I wax everywhere. |
| 19:30.24 | [TK]D-Fender | :O |
| 19:30.37 | cpm | <PROTECTED> |
| 19:30.48 | Corydon76-work | Mercestes: that's who your date will be, when you show up |
| 19:31.02 | Katty | so about dinner... |
| 19:31.07 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:31.11 | blitzrage | ManxPower: you can start using 1.4.x now :) |
| 19:31.18 | Katty | blitzrage: ! |
| 19:31.23 | Katty | blitzrage: what did you have for dinner last night? |
| 19:31.26 | blitzrage | Katty: !!! |
| 19:31.32 | Mercestes | Corydon76-work, In your dream.s |
| 19:31.38 | blitzrage | Katty: hrmmm... not sure I ate dinner last night |
| 19:31.43 | kombi | Katty: saucage sandwich + beer |
| 19:31.45 | Katty | blitzrage: gasp! |
| 19:31.49 | blitzrage | I do that a lot... |
| 19:31.51 | Katty | blitzrage: for shame! |
| 19:31.56 | blitzrage | for shame indeed! |
| 19:32.00 | Katty | blitzrage: better make it an alarm on your blackberry. |
| 19:32.13 | Corydon76-work | Katty: he was too busy reading Harry Potter... |
| 19:32.19 | blitzrage | haha... I probably should (although I don't have a BB, I have a Nokia E61i) |
| 19:32.22 | Katty | oh, well that explains everything. |
| 19:32.32 | blitzrage | s/ready/read/ |
| 19:32.40 | Katty | i like harry potter stuff. |
| 19:32.48 | blitzrage | I read philosophy and cosmology |
| 19:32.49 | Katty | especially that cute little halloween costume they have out now! |
| 19:33.26 | Katty | kombi: sausage sammich? |
| 19:33.39 | blitzrage | sandridge! |
| 19:33.45 | pigpen | Ok..so now I have: exten => 300,1,SET(MOMMY=${REALTIME(extensions|exten|9119|context)}) , Then it dumps every item associated with this query. |
| 19:33.50 | Katty | kombi: you mean like, breakfast sausage on white bread? |
| 19:34.02 | pigpen | How could I get it to return just, lets say, the context column? |
| 19:34.09 | Katty | kombi: that sounds kinda blah, really. |
| 19:34.22 | Mercestes | Yea ,use wheat |
| 19:34.23 | kombi | Katty: actually, serious solid german wurst on black bread.. |
| 19:34.44 | tzanger | heh |
| 19:34.46 | tzanger | speaking of wurst |
| 19:34.51 | blitzrage | pigpen: you could use func_odbc... |
| 19:34.55 | Katty | i have no clue what 'wurst' is... |
| 19:34.58 | blitzrage | tzanger: yer the wurst! |
| 19:35.02 | pigpen | blitzrage, no. |
| 19:35.04 | Katty | kombi: is it like italian sausage? |
| 19:35.11 | [TK]D-Fender | blitzrage: hukt on fonix werkt 4 u! |
| 19:35.14 | blitzrage | I think wurst is a german term.... |
| 19:35.18 | blitzrage | [TK]D-Fender: sometimes! |
| 19:35.19 | kombi | Katty: german for sausage, yeah, kind of |
| 19:35.41 | pigpen | From what I understand realtime will do what I need, just trying to get the syntax right. |
| 19:35.47 | pigpen | with limited documentation. |
| 19:35.59 | kombi | pigpen: we have all been there.. |
| 19:36.17 | pigpen | yep. |
| 19:36.20 | Katty | bratwurst? |
| 19:36.35 | Corydon76-work | pigpen: what's wrong with func_odbc? |
| 19:36.38 | kombi | that is good too, don't put that on bread though, more on the grill |
| 19:36.40 | tzanger | my msn name says "if puns were deli meat, this would be the wurst." |
| 19:36.41 | pigpen | at least now I am getting some info back...now to just trim it down. |
| 19:36.43 | Rienzilla | hm can anyone help me with an asterisk issue? I have 2 voip phones connected to an asterisk pbx. They can dial and phone to the outside world fine, but when I try to call one phone with the other there seems to be only one sided communication (one side can't hear the other). Any ideas? |
| 19:36.52 | tzanger | yes wurst is sausage |
| 19:37.07 | pigpen | Corydon76-work, with a direct connect to the db? Why use it? |
| 19:37.10 | Corydon76-work | All this talk of sausages... |
| 19:37.19 | blitzrage | lol |
| 19:37.20 | pigpen | I have realtime querying postgres directly. |
| 19:37.28 | kombi | I love the subject.. |
| 19:37.38 | Katty | hm, bratwurst. |
| 19:37.40 | Corydon76-work | pigpen: because func_odbc allows you to customize your query directly |
| 19:37.45 | E1ven | I'm getting quite a few crashes from MonitorMix on hangup- Is there something I can do to avoid it dying? |
| 19:37.48 | blitzrage | and because it's the COOLEST! |
| 19:37.51 | Katty | kombi: what do you do if you don't have a grill? |
| 19:37.54 | pigpen | Dam. I am getting hungry. Thanks all. |
| 19:38.00 | kombi | pan will do |
| 19:38.05 | Katty | pan fry? >.< |
| 19:38.11 | kombi | could do |
| 19:38.16 | Katty | sounds greasy. |
| 19:38.20 | kombi | grill's better though |
| 19:38.30 | kombi | grease is what we live on.. |
| 19:38.39 | Katty | i don't :P |
| 19:38.56 | kombi | that's 'cause you're a girl |
| 19:39.02 | Katty | pfft. |
| 19:39.04 | Corydon76-work | Mmmm, a heart attack special... |
| 19:39.11 | kombi | sorry, didn't say that.. |
| 19:39.34 | Katty | 99 cent heart attack, now available at the drive up window! |
| 19:39.44 | kombi | sneaked out there, didn't mean it either |
| 19:40.08 | kombi | pedals! 350KVolts! |
| 19:40.16 | kombi | clear! |
| 19:40.37 | Corydon76-work | "paddles" |
| 19:40.37 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 19:40.47 | [TK]D-Fender | bbbbbbzzzzzzZZZZZZZORTCHHHH!!!!!! |
| 19:40.51 | kombi | sorry, it's paddles? |
| 19:41.01 | sashion | someone smell burning flesh? |
| 19:41.03 | kombi | 450, clear! |
| 19:41.06 | Corydon76-work | Pedals are things you push with your feet |
| 19:41.25 | logyati | [TK]D-Fender, i recorded from asterisk, ty... but now i have an weird problem... calls from pstn have too low volume :( |
| 19:41.26 | kombi | i know..;) |
| 19:41.27 | Corydon76-work | and petals are parts of a flower |
| 19:41.36 | logyati | does it has a volume control or something? |
| 19:41.43 | [TK]D-Fender | logyati: fix your card gains. |
| 19:41.53 | kombi | logyati: you can normalize with sox |
| 19:41.56 | Katty | [TK]D-Fender: you are such a riddicurus person. |
| 19:42.04 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 19:42.16 | Katty | [TK]D-Fender: that sounds like something my mother's quaker parrot would sqwak while they're watching tv. |
| 19:42.17 | [TK]D-Fender | Katty: U can has graham-her! |
| 19:43.18 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 19:43.27 | CuriosCat | "The following |
| 19:43.28 | CuriosCat | minimal configuration defines an FXO port with FXS signaling: |
| 19:43.32 | CuriosCat | ...why would I want to do that? |
| 19:44.16 | tzanger | aha |
| 19:44.23 | tzanger | kontact and kopete are taking up ALL my memory |
| 19:44.26 | tzanger | 1G RAM and 0.5G swap |
| 19:44.27 | tzanger | wtf |
| 19:44.39 | EricL | Where do I tell Asterisk to look for call files in /var/spool/asterisk/outgoing? |
| 19:44.49 | EricL | Its not picking up the files in that directory. |
| 19:45.02 | kombi | EricL: that's were you move them |
| 19:45.20 | kombi | syntax right? |
| 19:45.58 | EricL | I took the syntax and copied it from the Polycom Auto Answer config wiki page. |
| 19:46.05 | EricL | I guess its right. |
| 19:46.45 | EricL | Should I pb it? |
| 19:46.58 | Katty | EricL: it just will |
| 19:47.04 | Katty | EricL: mine does, anyway |
| 19:47.12 | kombi | EricL: should be ok if it is a .call file |
| 19:47.22 | Katty | EricL: as soon as a file gets droped, it goes.. kinda like the queue folder for email. |
| 19:47.22 | EricL | pbx_spool is loaded, but its 'use count' is 0. |
| 19:47.41 | EricL | I have had files sitting in there for 30 minutes and nothing. |
| 19:48.01 | sashion | EricL: You sure asterisk can read them - ie: permissions ? |
| 19:49.06 | *** join/#asterisk Vulpyne (n=vulpyne@sta-207-174-202-66.rockynet.com) |
| 19:49.16 | EricL | (I know its wrong but), Asterisk is running as root. |
| 19:49.27 | Vulpyne | Hello. Is there anywhere I can read about porting codec modules from 1.2.x to 1.4.x? |
| 19:49.55 | kombi | EricL: maybe look up call files on voip-info and try one of those |
| 19:51.10 | sashion | EricL: did you create your call file outside of /var/spool/asterisk/outgoing and the mv it in ? |
| 19:51.21 | [TK]D-Fender | Vulpyne: Don't believe you can..... |
| 19:51.36 | kombi | sashion: that might be it.. |
| 19:51.36 | Vulpyne | [TK]D-Fender: Why not? |
| 19:51.43 | [TK]D-Fender | Vulpyne: Coding differences. |
| 19:51.58 | Vulpyne | [TK]D-Fender: Yeah, I meant porting - not just compiling it for 1.4. :) |
| 19:52.16 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 19:52.18 | EricL | sashion: Yep. |
| 19:52.27 | EricL | It doesn't even work with them chmod'd 0777. |
| 19:52.29 | [TK]D-Fender | Vulpyne: You mean you want to know HOW to code one for 1.4? |
| 19:52.40 | Vulpyne | I've gotten to the point where the module I'm attempting to port will compile, register itself, but there are some semantic changes in the interface for the callbacks. |
| 19:52.48 | *** part/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 19:52.56 | Vulpyne | And it's hard to figure out how that changed just by looking at the source ot other codec modules for 1.4. |
| 19:52.58 | [TK]D-Fender | EricL: They need to be dated in the past, and you should MV them there, not CP. |
| 19:52.59 | Vulpyne | [TK]D-Fender: Yeah. |
| 19:53.15 | [TK]D-Fender | Vulpyne: thats a question for the -dev channel |
| 19:53.20 | easimon | hi everyone. i upgraded from 1.0.something to 1.4.8+bristruff+florz+chan_capi recently, having a avm fritz card (capi) on the outside and a hfc-card (NT mode) on the inside. the same configuration worked fine with asterisk 1.0, but now i cannot receive any calls anymore. the phone connected to the hfc card rings only once and then asterisk drops the call with cause 18 (no equipment connected). does anybody have an idea how to fix that? |
| 19:53.21 | sashion | EricL: so you created your .call file in say /root then mv'd it to /var/spool/asterisk/outgoing ? |
| 19:53.24 | EricL | [TK]D-Fender: Are they not dated in the past after sitting there? |
| 19:53.29 | Vulpyne | [tk: Ahh. I wasn't aware of that, but I'll go there. :) Thanks. |
| 19:53.35 | [TK]D-Fender | EricL: depends... |
| 19:53.43 | *** part/#asterisk Vulpyne (n=vulpyne@sta-207-174-202-66.rockynet.com) |
| 19:53.45 | EricL | And I do move them, I create the files in /var/tmp and then mv *.call /var/.../outgoing/ |
| 19:53.46 | *** join/#asterisk JD_2007 (n=Abc@cpe-75-82-48-60.socal.res.rr.com) |
| 19:54.19 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.23 and 1.4.9 released (July 24, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 19:54.32 | sashion | 1.4.9 ? |
| 19:54.34 | sashion | geez |
| 19:54.42 | Mercestes | easimon, I think the supported upgrade path from 1.0 to 1.4.8 is to install from scratch. |
| 19:55.08 | Qwell[] | Mercestes: s/8/9/ |
| 19:55.19 | easimon | Mercestes: in spite of not knowing this - i installed from scratch. |
| 19:55.52 | EricL | Let me try throwing a sleep(5) in there before I do the mv. |
| 19:55.52 | Mercestes | Qwell[]: woot! |
| 19:55.53 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:55.57 | Mercestes | Qwell[]: I bought an account last week. |
| 19:56.11 | Qwell[] | nice, what server you using? |
| 19:56.14 | Mercestes | easimon, Ah, you said "upgrade." |
| 19:56.17 | Mercestes | Lightninghoof |
| 19:56.21 | Qwell[] | lame |
| 19:56.24 | Mercestes | ? |
| 19:56.26 | Qwell[] | but I think that's shadowburn |
| 19:56.32 | easimon | Mercestes: it's a whole new disk with a new linux distribution and a fresh asterisk installation... just the hardware is the same |
| 19:57.06 | Mercestes | easimon, I'm not real sure then. Sounds like a modules/driver issue tho |
| 19:57.32 | Mercestes | Qwell[] Where are the 133t servers then? I kinda like Lightninghoof but I'm a total n00b |
| 19:57.37 | Qwell[] | Mercestes: dunno |
| 19:57.39 | *** part/#asterisk JD_2007 (n=Abc@cpe-75-82-48-60.socal.res.rr.com) |
| 19:57.42 | Qwell[] | lame because it isn't baelgun :p |
| 19:57.43 | Mercestes | Lvl 17 NE dru tho. |
| 19:57.55 | Mercestes | Qwell[]: A or H? |
| 19:58.09 | Qwell[] | A, but yeah, lightninghoof is fine :p |
| 19:58.20 | Mercestes | I might drop by and give it a gander. |
| 19:58.24 | easimon | Mercestes: i don't know, that's why i ask - an interesting detail is, that i can do outgoing calls without any problems, just incoming calls dont work. my phone even rings once on an incoming call. |
| 19:58.30 | Mercestes | I was hoping H tho..I want a BE War |
| 19:58.39 | EricL | Even putting a sleep on it so it gets moved into the outgoing directory with a time in the past doesn't help. |
| 19:58.40 | Qwell[] | Mercestes: H, Spirestone |
| 19:58.48 | Qwell[] | Mercestes: There's a BE guild :P |
| 19:58.51 | EricL | Is there anyway to force Asterisk to grab whats in the directory? |
| 19:58.56 | Mercestes | Nice, you on Spirestone? |
| 19:58.57 | easimon | looks like there's some kind of timeout for the phone answering to the SETUP message... |
| 19:59.03 | Qwell[] | not really |
| 19:59.07 | easimon | and my phone might be too slow |
| 19:59.09 | Mercestes | oh. |
| 19:59.42 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 20:02.01 | Qwell[] | Mercestes: see msg |
| 20:03.10 | kombi | hmm, how do I best tell the number of people in a conference? I though of manager but there doesn't seem to be an action that fits.. |
| 20:06.27 | sashion | EricL: by default, asterisk will grab any file and read it... |
| 20:06.47 | sashion | what does your call file look like ? |
| 20:07.44 | EricL | I just put it up in a pb, but firefox crashed, let me put it back up. |
| 20:07.45 | *** join/#asterisk SexyKen (n=sexy@c-76-21-43-222.hsd1.ca.comcast.net) |
| 20:07.53 | SexyKen | Does Asterisk support Shared Lines? |
| 20:08.09 | EricL | sashion:http://paste.ubuntu-nl.org/31148/ |
| 20:09.16 | [TK]D-Fender | SexyKen: No. |
| 20:09.26 | [TK]D-Fender | EricL: and your dialplan please... |
| 20:10.08 | SexyKen | Will it ever? :-( |
| 20:10.40 | pigpen | Well, I got realtime in the dialplan to produce some info: |
| 20:10.41 | [TK]D-Fender | SexyKen: "load res_psychic.so" <-------- |
| 20:10.41 | pigpen | exten => 300,1,SET(MOMMY=${REALTIME(extensions|exten|9119)}) |
| 20:10.55 | EricL | [TK]D-Fender: Am I supposed to put something in my dialplan for .call files to work? |
| 20:11.13 | pigpen | It is returning the value as "id=24" which is the id number. |
| 20:11.24 | kombi | EricL: no, but the reason might be in there |
| 20:11.28 | [TK]D-Fender | EricL: I'd like to see if things match like they're supposed to. |
| 20:11.29 | pigpen | anyone know how I might grab a value from a different column? |
| 20:11.31 | SexyKen | My balls itch anyway. |
| 20:12.14 | *** join/#asterisk ServerGod (n=ppetroff@70.97.159.120) |
| 20:12.38 | ServerGod | anyone have luck with opensolaris 10 and asterisk? |
| 20:12.42 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 20:12.59 | EricL | [TK]D-Fender: Do you want the dialplan from "show dialplan" or the extensions.conf ? |
| 20:14.03 | [TK]D-Fender | EricL: extensions.conf |
| 20:16.36 | galeras | is genzaptel supported here? |
| 20:17.18 | shido6 | :) |
| 20:17.33 | *** join/#asterisk nickrooster (n=nbaldrid@commonwealth01.commund.com) |
| 20:18.05 | [TK]D-Fender | ~8ball is genzaptel supported here? |
| 20:18.05 | jbot | Yes. |
| 20:18.06 | *** join/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 20:18.15 | [TK]D-Fender | the ball never lies.... |
| 20:18.20 | Mercestes | ~8ball does Katty like me? |
| 20:18.20 | jbot | Absolutely. |
| 20:18.24 | Mercestes | YES!!!! |
| 20:18.26 | Strom_M | ~8ball will I win the lottery? |
| 20:18.27 | jbot | Unsure. |
| 20:18.36 | Strom_M | that's hot |
| 20:18.37 | [TK]D-Fender | ~8ball is Mercestes delusional? |
| 20:18.38 | jbot | Please ask again. |
| 20:18.48 | andresmujica | hi !! any pointers to ericsson bp250 and asterisk integration ???? i've found nothing at voip-info .... |
| 20:18.49 | [TK]D-Fender | ~8ball is Mercestes delusional? |
| 20:18.50 | jbot | I'm not sure. |
| 20:18.54 | Mercestes | ha! pwned! |
| 20:19.07 | [TK]D-Fender | Mercestes: You're not off the hook yet! |
| 20:19.11 | Mercestes | hehe |
| 20:19.18 | Mercestes | idc, Katty likes me. |
| 20:19.44 | Corydon76-work | ~8ball does Mercestes need a boyfriend? |
| 20:19.45 | jbot | Please ask again. |
| 20:19.54 | EricL | http://www.pastebin.ca/632358 |
| 20:20.01 | Corydon76-work | ~8ball does Mercestes need a boyfriend? |
| 20:20.01 | jbot | Please ask again. |
| 20:20.56 | sevard | ~8ball bonars? |
| 20:20.56 | jbot | Absolutely. |
| 20:21.00 | sevard | awesome. |
| 20:21.19 | Corydon76-work | ~8ball does Mercestes need a boyfriend? |
| 20:21.19 | jbot | Please ask again. |
| 20:21.59 | Corydon76-work | Something's wrong with the 8-ball. That was a clear "Yes" |
| 20:22.04 | Mercestes | lmao |
| 20:22.05 | Mercestes | rofl |
| 20:22.07 | Mercestes | PWNED! |
| 20:22.20 | Mercestes | ~8ball Does Cory want my sexy body? |
| 20:22.20 | jbot | Are you smoking crack? |
| 20:22.24 | Mercestes | ... |
| 20:22.28 | [TK]D-Fender | pwned |
| 20:22.30 | Mercestes | Yea |
| 20:22.32 | Mercestes | hard. |
| 20:23.13 | andresmujica | hi !! any pointers to ericsson bp250 and asterisk integration ???? i've found nothing at voip-info .... |
| 20:23.54 | Corydon76-work | andresmujica: is that a proprietary handset? |
| 20:24.27 | [TK]D-Fender | andresmujica: Stop asking the same thing over & over. We heard you the first time. As for connecting the two, it obviously dependon what kind of interfaces youi have available on it. |
| 20:24.34 | [TK]D-Fender | Corydon76-work: its a PBX |
| 20:24.56 | Mercestes | ericsson makes a PBX? |
| 20:24.56 | Corydon76-work | andresmujica: you need a PRI interface on it |
| 20:25.37 | Corydon76-work | Yikes. The BP250 is limited to 60 maximum calls |
| 20:25.51 | *** join/#asterisk oej_ (n=olle@apollo.webway.se) |
| 20:27.57 | EricL | Any ideas? |
| 20:28.01 | andresmujica | thks.. i'm trying to look for some info about it... anyone knows if it would be possible to use e&m ??? and yeap.. it seems that thing is cap to 60 calls.... |
| 20:28.13 | andresmujica | and point taken about the double question... |
| 20:28.42 | Corydon76-work | Yes, it's possible to use E&M on Asterisk |
| 20:29.38 | nickrooster | Hi all - anyone had an issue with snom phones sending a 302 redirect causing all sip phones to freak out and lose registration? We have 18 snom phones with qualify=yes in sip.conf and twice today so far, all peers have gone unavailable and could not re-register until a server reboot |
| 20:29.49 | *** join/#asterisk taqua2008 (n=perdue@66.118.69.58) |
| 20:30.32 | *** join/#asterisk Webspot (n=Webspot@unaffiliated/webspot) |
| 20:30.43 | *** join/#asterisk rpm (n=russell@66.183.28.233) |
| 20:30.50 | [TK]D-Fender | EricL: Are you trying to do basic paging? |
| 20:31.08 | EricL | [TK]D-Fender: Yep. I just want it to be an all-call one way intercom. |
| 20:31.18 | *** part/#asterisk galeras (n=root@201.245.103.169) |
| 20:31.25 | [TK]D-Fender | EricL: then you should be using the Page application, not call files. |
| 20:31.33 | [TK]D-Fender | EricL: Your approach is pbackwards |
| 20:31.40 | [TK]D-Fender | BBIAB, heading home. |
| 20:31.44 | *** join/#asterisk wothinn (i=root@vs1.svartalfheim.net) |
| 20:31.56 | pigpen | Please note, the page app will not span past 22 sip phones with 4 digit extensions. |
| 20:32.00 | Webspot | Hi. I am trying to test setting up asterisk for the first time. I just want to do a basic Ekiga app to asterisk server over SIP. I've managed to execute a few commands, such as waiting for 8 seconds then answering. But when I try to play a sound, nothing plays. Any ideas? |
| 20:32.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 20:33.00 | EricL | pigpen: So if that's the case and I have more than 22 phones, am I doing it correctly? |
| 20:33.04 | Corydon76-work | Lack of audio with SIP is usually caused by NAT or firewall issues |
| 20:33.21 | Webspot | It's internal though. Would that make a difference? |
| 20:33.41 | Webspot | No firewall too |
| 20:33.41 | pigpen | Well, I didn't see how you are doing it now, but I have several deployments that exceed 150 |
| 20:33.47 | Corydon76-work | Depends on network architecture |
| 20:34.00 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 20:34.00 | Webspot | Ah right |
| 20:34.20 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 20:34.40 | *** part/#asterisk Webspot (n=Webspot@unaffiliated/webspot) |
| 20:34.53 | pigpen | EricL, I would love for the page app to handle more, but I have ran into the fact that the max string length in asterisk is 200`ish characters. |
| 20:35.19 | pigpen | I would -love- to use the page app. |
| 20:35.46 | Corydon76-work | pigpen: 255 characters, exactly |
| 20:36.14 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:36.43 | EricL | pigpen: So how do you handle an "all call" intercom? |
| 20:36.43 | pigpen | Ah, thanks, I forgot the exact. |
| 20:36.58 | pigpen | I use an agi script that dumps everyone into a meetme. |
| 20:37.24 | pigpen | but, with 150'ish phones, heh..it takes about 5 - 9 seconds for everyone to get there. |
| 20:37.25 | EricL | That's exactly what I am trying (unsuccessfully) to do. |
| 20:37.45 | EricL | I only have about 25 phones right now, but we're growing pretty quickly. |
| 20:37.47 | Corydon76-work | pigpen: if you look in the source, in pbx/pbx_config.c, function pbx_load_config, the size of "realvalue" determines the maximum number of characters. |
| 20:38.03 | pigpen | Corydon76-work, no shit. Hmm.... |
| 20:38.05 | Corydon76-work | pigpen: you can increase that up to 8000 characters |
| 20:38.17 | *** join/#asterisk JD_2007 (n=Abc@cpe-75-82-48-60.socal.res.rr.com) |
| 20:38.21 | EricL | I am getting stuck because the call files I am attempting to use to dump everyone into the MeetMe isn't working. |
| 20:38.44 | *** join/#asterisk `paul (n=aldee@124.107.13.212) |
| 20:38.46 | pigpen | hmm..that would handle over 800. |
| 20:38.59 | Corydon76-work | pigpen: or sorry, it's actually realext |
| 20:39.14 | pigpen | Corydon76-work, think it would be "faster" than using the old agi? |
| 20:39.30 | Corydon76-work | Extensions are always faster than AGI |
| 20:39.43 | `paul | im looking at the CDR data in mysql how does one know the length of a successful call excluding the queues etc... |
| 20:39.57 | pigpen | hmm..I may need to give that a try. |
| 20:40.13 | pigpen | EricL, I would try that. The page app is so much easier. |
| 20:40.45 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 20:40.47 | EricL | But wouldn't that require a rebuild of Asterisk ot change that realext value? |
| 20:40.54 | pigpen | yup. |
| 20:41.07 | pigpen | no worries here. we run gentoo. :) |
| 20:41.14 | EricL | I run Gentoo also. |
| 20:41.21 | EricL | But I build Asterisk from source. |
| 20:41.23 | pigpen | My wife says, "Take out the trash!" |
| 20:41.34 | pigpen | I say, "Wait, it has to compile!" |
| 20:41.39 | EricL | I would still like to figure out why the .call files aren't working. |
| 20:41.57 | pigpen | EricL, probably permissions. |
| 20:42.13 | pigpen | if the file permissions are wrong, it will ignore them. |
| 20:42.44 | EricL | I chmod it 0777 and still nothing. |
| 20:43.45 | delmar | hey everyone. I have a problem with DISA that I can't figure out. All phones/devices that are members of context [local] can call each other (3digit extensions) and dial out via Zap or VISP etc.. that part work without issue. However, DISA can't dial any local extensions at all. I even created a new special context [disa-out] and added some dialplan to it for the local extensions, and set that to the DISA context to use.. still |
| 20:43.45 | delmar | it wont dial the local extensions.... it just drops the caller with no useful console output. any ideas anyone? |
| 20:44.57 | De_Mon | ever since I upgraded from asterisk 1.2.14 to asterisk 1.4.6 my Aastra phone refuses to operate |
| 20:45.15 | De_Mon | I'm getting constant REGISTAR requests from the phone but nothing shows up in asterisk |
| 20:45.18 | De_Mon | http://pastebin.ca/632401 |
| 20:46.20 | wothinn | Anyone familiar with the OpenBSD dhcpd? I need to get it to send option 66 to my Polycom phone at server-name and next-server both appear to not be what I need. |
| 20:46.38 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 20:46.45 | blitzrage | anyone know if System() can return the result of the command you are running? |
| 20:47.23 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:48.42 | Katty | blitzrage: i can get callerid and date/time info |
| 20:48.56 | blitzrage | eh? |
| 20:48.56 | Katty | blitzrage: but that's cause its stuck in a variable :/ |
| 20:49.12 | blitzrage | ya, I'm running System(hostname), and want to get the result of 'hostname' |
| 20:49.16 | blitzrage | but I don't think I can do that in 1.4 |
| 20:49.21 | JD_2007 | http://www.freebsd.org/cgi/man.cgi?query=dhcp-options&sektion=5 |
| 20:49.25 | blitzrage | ${SYSTEMNAME} will probably work for me |
| 20:50.45 | logyati | hey guys, im forwarding calls from openser to asterisk. I dont know where is my mistake, i need a tip... when i try to call a pstn number it says "user not found" unless i put the content of [default] inside [incoming] context!! its wrong right? i want call from sip using default context! please look my sip.conf and extensions.conf http://www.pastebin.ca/632408 |
| 20:50.59 | *** part/#asterisk nickrooster (n=nbaldrid@commonwealth01.commund.com) |
| 20:52.38 | CuriosCat | Hrm. |
| 20:54.01 | wothinn | JD: Good pointer. Thanks... I missed the reference to that manpage. |
| 20:54.44 | jkiff | logyati: You should include the default context in the incoming one with "include=>". |
| 20:55.05 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 20:55.09 | CuriosCat | So..I tried the example in the documentation from Digium, but instead of getting the "echo" application I was expected, I get something that sounds kind of ..but not quite..like a modem tone when I call in. |
| 20:55.26 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:57.21 | *** join/#asterisk Op3r (n=Op3r@125.212.63.101) |
| 20:59.26 | EricL | It's not the permission and its not the timestamp. Does anyone have any idea what the issue could be with the .call files |
| 20:59.43 | ServerGod | anyone have luck with opensolaris 10 and asterisk? |
| 21:00.04 | ServerGod | besides solarisvoip? |
| 21:00.17 | [TK]D-Fender | EricL, pastebin it all again, but remember, this isn't the way to do paging.... |
| 21:00.58 | EricL | [TK]D-Fender:I was just told that page_app can only support about 20 people without modifying the source code. |
| 21:01.48 | logyati | jkiff, no no, i dont want the call passing through [incoming] context, i want sip call only in default context |
| 21:01.55 | EricL | [TK]D-Fender: http://www.pastebin.ca/632358 |
| 21:02.15 | logyati | [TK]D-Fender, master ^^ i need your help with another noob question |
| 21:02.29 | logyati | [TK]D-Fender, im forwarding calls from openser to asterisk. I dont know where is my mistake, i need a tip... when i try to call a pstn number it says "user not found" unless i put the content of [default] inside [incoming] context!! its wrong right? i want call from sip using default context! please look my sip.conf and extensions.conf http://www.pastebin.ca/632408 |
| 21:04.43 | [TK]D-Fender | logyati, need more backup... |
| 21:08.00 | *** join/#asterisk icel (n=icel@63.78.162.77) |
| 21:08.29 | icel | does anyone know how to just reload voicemail in * 1.4? |
| 21:08.59 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 21:09.14 | logyati | [TK]D-Fender, well, as i know, asterisk will use a extensions.conf context for each context in sip.conf for example, right? looking my sip.conf you will see that allowguests=yes is inside context default. So, i expected that when i make a sip call, it goes thought [default] (ONLY) in extensions.conf. Am i right? |
| 21:09.23 | EricL | icel: module reload app_voicemail |
| 21:09.39 | icel | EricL: thanx |
| 21:09.48 | EricL | icel: np |
| 21:10.18 | [TK]D-Fender | logyati, un-authed calls should go through [default] |
| 21:10.35 | [TK]D-Fender | icel: module reload app_voicemail.so |
| 21:10.49 | icel | thx Fender |
| 21:11.19 | logyati | [TK]D-Fender, yes, but isnt going! it says "user not found" unless i put a "include => default" inside [incoming] |
| 21:12.44 | [TK]D-Fender | logyati, I still need to see more.... |
| 21:13.04 | logyati | [TK]D-Fender, did you see my extensions.conf and sip.conf? |
| 21:13.09 | [TK]D-Fender | logyati, Yes |
| 21:13.28 | karrotx | so asterisk checks every ip address to make sure the forward and reverse dns match? |
| 21:13.30 | karrotx | that's pretty wacco |
| 21:14.52 | logyati | asterisk is acting as if un-authed calls come from incoming context, inside sip.conf, unstead default context... thats da problem |
| 21:15.48 | logyati | [TK]D-Fender, if i change at sip.conf all "incoming" names to "foo-bar", asterisk will use context [foo-bar] in extensions.conf to handle un-authed calls |
| 21:16.52 | [TK]D-Fender | logyati, stop thinking, and start PASTEBIN-ing./ |
| 21:17.43 | De_Mon | logyati do you have a sip.conf entry for openser? |
| 21:18.08 | logyati | de_mon, no cos i use un-authed calls |
| 21:18.28 | logyati | [TK]D-Fender, i dont know what more i can paste |
| 21:19.30 | [TK]D-Fender | logyati, When you find a clue, let me know. |
| 21:20.14 | *** join/#asterisk joe-f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 21:20.33 | fujin | rafb.net/paste |
| 21:20.50 | joe-f | anyone know of a good dedicated server hosting provider based in New York City, that's peered with Level 3? (voxbone's DID stuff comes from NY L3) |
| 21:22.10 | EricL | [TK]D-Fender: I have no idea where to go with this .call file stuff not working. |
| 21:23.02 | [TK]D-Fender | EricL, show absolutely everything related. |
| 21:23.21 | logyati | [TK]D-Fender, ok, this is my output of CLI with 10 verbose http://www.pastebin.ca/632450 |
| 21:23.43 | [TK]D-Fender | logyati, keey trying.... |
| 21:23.48 | [TK]D-Fender | keep* |
| 21:25.38 | logyati | [TK]D-Fender, http://www.pastebin.ca/632454 zaptel.conf and zapata.conf |
| 21:25.52 | [TK]D-Fender | logyati, clearly worthless..... |
| 21:26.24 | [TK]D-Fender | logyati, You've got a problem with where the calls are going and you're not even looking at the CALL. |
| 21:26.26 | logyati | [TK]D-Fender, lol, i didnt changed any other asterisk file!!! |
| 21:26.36 | logyati | hmmm |
| 21:26.39 | logyati | wait |
| 21:27.04 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 21:27.06 | *** join/#asterisk davixx (n=davixx@85.69.124.15) |
| 21:28.29 | logyati | [TK]D-Fender, ngrep of * sip port http://www.pastebin.ca/632458 |
| 21:28.34 | davixx | Hi. I have an "SIP/2.0 401 Unauthorized" when my client try to connect... i don't understand why.... but i have somes others client which arrives to connect... how to check ? |
| 21:28.47 | EricL | [TK]D-Fender: What is related? Do I need to throw my entire asterisk config dir up somewhere? |
| 21:29.14 | [TK]D-Fender | logyati, getting warmer.... |
| 21:30.11 | [TK]D-Fender | EricL, file dumps from console, "ls" dumps, cli output at high verbose, etc |
| 21:30.48 | *** part/#asterisk JD_2007 (n=Abc@cpe-75-82-48-60.socal.res.rr.com) |
| 21:30.53 | EricL | The CLI output at verbosity level 10, doesn't show anything but the AGI script exiting with a status of 0. |
| 21:31.11 | logyati | [TK]D-Fender, i dont know where to look :( |
| 21:31.21 | Mercestes | EricL, Throw some AGI-Noops in there for verbosity |
| 21:31.51 | EricL | Mercestes: Fair enough, what is it that I should be printing out? |
| 21:31.58 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 21:32.00 | logyati | [TK]D-Fender, ooohh i know |
| 21:32.05 | logyati | [TK]D-Fender, wait |
| 21:32.35 | Mercestes | Agi_Exec,NoOp(EricL still loves da cock at line 12). I dunno....whatever you want man. output some variables or something. |
| 21:33.09 | Mercestes | Use it to trace out your AGI program flow. |
| 21:33.26 | Mercestes | Start with a NoOp, then put some NoOps at some key positions. |
| 21:33.27 | EricL | The AGI program works fine, it generates the call files and places them in proper directory. |
| 21:33.41 | stubert | Since upgrading to 1.4.8 I'm seeing chan_sip.c errors in the logs. "We could NOT get the channel lock for" and "SIP transaction failed"... Anything known about this? |
| 21:33.47 | EricL | Asterisk just isn't picking up the call files once they are there. |
| 21:34.38 | Mercestes | EricL, Permissions? |
| 21:34.40 | logyati | [TK]D-Fender, sip debug!!! http://www.pastebin.ca/632464 |
| 21:34.43 | Mercestes | EricL, sure it's the right directory? |
| 21:34.50 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-389cda526b9fb0db) |
| 21:34.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:34.51 | Mercestes | EricL, are you moving them or copyiing them?? |
| 21:35.11 | [TK]D-Fender | logyati, Looking for 025614265 in incoming (domain caerj.proderj.rj.gov.br) |
| 21:35.18 | [TK]D-Fender | logyati, SIP/2.0 404 Not Found |
| 21:35.27 | EricL | Merecs: I have made the perms 0777, I am positive I am moving them to: /var/spool/asterisk/outgoing |
| 21:35.50 | [TK]D-Fender | logyati, Found peer 'incoming' |
| 21:35.59 | EricL | How can I be sure that /var/spool/asterisk/outgoing is the outgoing spool directory? I don't see a config setting for that. |
| 21:36.10 | Mercestes | EricL, Don't use move, use copy. |
| 21:36.25 | [TK]D-Fender | Mercestes, BACKWARDS |
| 21:36.34 | logyati | [TK]D-Fender, should i create another peer?? at this point i dont know what to do |
| 21:36.47 | Mercestes | [TK]D-Fender, Oh.... |
| 21:36.49 | [TK]D-Fender | logyati, is is MATCHING your peer. |
| 21:36.56 | Mercestes | [TK]D-Fender, that's probably why my call files don't work |
| 21:37.04 | [TK]D-Fender | logyati, so it is NOT going through as UNAUTHED |
| 21:37.13 | [TK]D-Fender | logyati, Now go look at it! |
| 21:37.14 | Mercestes | EricL, nevremind, use move, not copy |
| 21:37.15 | Mercestes | lol |
| 21:38.07 | logyati | [TK]D-Fender, now im confused, cos i added this peer incoming cos of calls pstn-to-sip |
| 21:38.23 | EricL | Mercestes: I already am using mv. |
| 21:38.35 | logyati | [TK]D-Fender, seems that i dont even know what im doing, can you clear me? |
| 21:38.43 | *** join/#asterisk djs_2_6 (n=DJS@cpe-075-182-081-167.nc.res.rr.com) |
| 21:39.35 | [TK]D-Fender | logyati, I hate to suggest this, but put your peer auth in your dial statement, and remove the entry from sip.conf (comment out at least). then all calls should be forced un-auth'd |
| 21:39.54 | Mercestes | EricL, Not sure. What is the spool dir in asterisk.conf? |
| 21:40.25 | logyati | [TK]D-Fender, hmmmmm |
| 21:40.29 | logyati | [TK]D-Fender, got it |
| 21:40.49 | EricL | Nothing with the word outgoing in it. |
| 21:41.05 | Mercestes | hrm |
| 21:41.26 | Mercestes | EricL, look for astspooldir => |
| 21:43.10 | EricL | <PROTECTED> |
| 21:43.22 | EricL | Is the outgoing directory supposed to be a subdir of that? |
| 21:44.45 | *** join/#asterisk ITiliti (n=IceChat7@72.54.46.18) |
| 21:45.55 | ITiliti | hello all. |
| 21:46.13 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 21:46.13 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 21:46.24 | EricL | It works...The directory was in the wrong place. |
| 21:46.35 | ITiliti | I am trying to figure out how to write the the DID that is getting called on our * box to the CDR datatbase. Any help? |
| 21:48.22 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 21:52.29 | joe-f | How can I allow dialing a conference # over the audio playback? I'm using meetme and it's disconnecting me if I don't wait until the audio is done playing. |
| 21:53.28 | x86 | http://pastebin.ca/632484 |
| 21:53.40 | joe-f | My log reads: -- Executing [...@voxbonecontext:3] BackGround("SIP/81.201.84.29-08201820", "welcome-to-kb") in new stack |
| 21:53.42 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 21:54.33 | joe-f | WARNING[1317]: pbx.c:2494 __ast_pbx_run: Invalid extension '1234#', but no rule 'i' in context 'voxbonecontext' |
| 21:54.45 | joe-f | any idea on how to allow dialing a conf # over the BackGround audio? |
| 21:54.54 | *** join/#asterisk Stridernzl (n=neville@125-239-163-97.jetstream.xtra.co.nz) |
| 21:54.57 | *** join/#asterisk minkus (n=minkus@pool-71-182-32-236.clrkwv.east.verizon.net) |
| 21:55.26 | Katty | nite guys (= |
| 21:55.33 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 21:55.36 | kombi | will gotoif accept boolean operators? |
| 21:56.06 | kombi | ||, && ? |
| 21:56.07 | hrmphh | can someone recommend a T1 PRI card? box will need to support 12 channels on an integrated T1 |
| 21:57.03 | minkus | kombi: the [ ] expression accepts them |
| 21:57.16 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 21:57.18 | kombi | minkus: thanks! |
| 21:58.18 | minkus | kombi: the operators are & and | not && and || |
| 21:58.27 | tako-san | hrmphh: Sangoma makes a number of good T1 cards. Rhino is another alternative though I have never used them myself. |
| 21:59.00 | hrmphh | yeah i use digium now and the card is 100% a piece of shit |
| 21:59.03 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 21:59.03 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 21:59.04 | hrmphh | ill never buy digium again |
| 21:59.57 | Strom_M | hrmphh: por que? |
| 22:00.12 | fujin | get some real hardware |
| 22:00.20 | hrmphh | umm random static |
| 22:00.26 | fujin | we run an as5400 here, I can put 8 E1's into it. |
| 22:00.31 | fujin | currently have two going into it |
| 22:00.43 | hrmphh | yeh |
| 22:00.45 | hrmphh | cisco is nice |
| 22:01.08 | hrmphh | ive used ccm in the past w/nice integrated routers |
| 22:01.15 | hrmphh | plop in vwic and done |
| 22:01.33 | hrmphh | assuming router has enough dsp :) |
| 22:02.31 | joe-f | anyone know what "no rule 'i' in context" means? |
| 22:02.42 | joe-f | im being disconnected because of that error.. |
| 22:03.05 | hrmphh | hmm do i need onboard echo cancel on this guy |
| 22:03.10 | hrmphh | or does the telco do that for these lines? |
| 22:03.15 | hrmphh | (ISDN PRI) |
| 22:03.25 | minkus | joe-f: that means that the extension that you are trying to use in that context is invalid |
| 22:03.31 | *** join/#asterisk mindCrime (n=chatzill@adsl-221-69-155.rmo.bellsouth.net) |
| 22:04.20 | minkus | joe-f: * tries to jump to extension 'i' when you try to dial or goto an invalid extension |
| 22:04.38 | joe-f | i'm using web-meetme to handle my conference #'s |
| 22:04.53 | joe-f | and it works fine, if i wait till the BackGround audio file plays |
| 22:09.10 | ITiliti | If I am using the dial command to play a wav file, how can I have it stop playing it? |
| 22:09.53 | ITiliti | I am p[lying a ringer through a paging system in the ceiling, and I want it to play until someone picks up the phone, and then have it stop playing upon someone picking it up... |
| 22:12.26 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:16.32 | brodiem | anyone know how I can get an extension state from the dial plan? Trying to create a dial plan for call waiting. I know of using the API's ExtensionState but want to use dial plan |
| 22:19.43 | matt_ | hello, i just restarted azureus and some of the text is in some funny font |
| 22:19.51 | Qwell[] | matt_: umm |
| 22:19.59 | Qwell[] | Try over there --> |
| 22:20.24 | matt_ | lol oops :) |
| 22:20.41 | blitzrage | brodiem: see http://www.asterisk.org/node/48360 |
| 22:21.06 | blitzrage | brodiem: and http://www.asterisk.org/node/48325 |
| 22:21.30 | blitzrage | i.e. DEVSTATE() is probably the function you are looking for |
| 22:26.27 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:26.37 | syzygyBSD | how can I force a hangup of a call? |
| 22:26.47 | blitzrage | soft hangup |
| 22:27.41 | syzygyBSD | what if that doesn't work? |
| 22:27.57 | blitzrage | stop now |
| 22:27.59 | blitzrage | :) |
| 22:28.07 | syzygyBSD | lol |
| 22:28.17 | syzygyBSD | ya... I know that one too, but this server is too busy for that |
| 22:29.04 | brodiem | blitzrage thanks, I actually read that article about a week ago and forgot all about it :) |
| 22:29.09 | blitzrage | :D |
| 22:29.45 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:30.07 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 22:31.02 | syzygyBSD | oh.. so the channel is hung up but it is still in the queue.. how can i restart a queue |
| 22:32.06 | ZX81 | heh: http://www.sineapps.com/news.php?rssid=1783 |
| 22:34.34 | AdamB0122 | Hey everyone |
| 22:34.34 | AdamB0122 | http://rafb.net/p/dgIrBF77.html |
| 22:36.22 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:36.25 | AdamB0122 | I've got a T1 coming off of a channel bank, into the asterisk box |
| 22:36.37 | AdamB0122 | I can see that the asterisk box picks the call up on Zap1-1 |
| 22:36.55 | AdamB0122 | but for some reason, then also sees something on Zap/2-1, and starts using that channel for some reason |
| 22:37.30 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:37.32 | AdamB0122 | and it says that its playing the the ivr-2|s|1, and playing "main" but nothing is playing through the phone |
| 22:38.17 | *** join/#asterisk [Outcast] (n=bill@219-89-206-239.adsl.xtra.co.nz) |
| 22:38.32 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:38.51 | *** join/#asterisk tako-san (n=Tako-san@S01060015e96b9a45.gv.shawcable.net) |
| 22:39.12 | *** join/#asterisk bkruse_home (i=kruz@nat/digium/x-54854d550cdf605a) |
| 22:39.16 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:39.37 | *** part/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 22:43.58 | Strom_M | ok, I feel like a nub asking this, but here goes anyway |
| 22:44.25 | Strom_M | I've got a server with two TE410P cards and one TDM880B |
| 22:44.40 | Strom_M | when going off hook on the TDM880B, I get this: |
| 22:44.57 | Strom_M | WARNING[3286]: chan_zap.c:6613 handle_init_event: Unable to play dialtone on channel 200 |
| 22:45.18 | Strom_M | i'm sure it's probably something simple I'm overlooking, but intarweb searches aren't revealing much |
| 22:46.47 | Strom_M | I do have talk battery |
| 22:47.09 | Qwell[] | Strom_M: You lost me at 'ok' |
| 22:48.12 | Strom_M | boners. |
| 22:51.34 | ZX81 | I guess have a look at tone_zone_play_tone |
| 22:51.47 | ZX81 | and see what conditions it returns res < 0 |
| 22:52.33 | ZX81 | what version? |
| 22:53.05 | ZX81 | hmmm |
| 22:53.08 | ZX81 | http://www.asterisk.org/doxygen/1.4/chan__zap_8c.html doesn't work |
| 22:53.24 | ZX81 | hmm http://www.asterisk.org/doxygen doesn't work |
| 22:54.23 | AdamB0122 | gr |
| 22:54.31 | AdamB0122 | Outbound calling works fine |
| 22:54.43 | ZX81 | doxygen is down... |
| 22:54.58 | ZX81 | Strom_M: so if it comes back up I'll try give you a hand |
| 22:57.02 | Strom_M | ZX81: sorry, back at console now :) |
| 22:57.10 | Strom_M | asterisk 1.4 branch as of this afternoon |
| 22:57.29 | ZX81 | yeah but there's no doxygen so I can't find tone_zone_play_zone |
| 22:57.30 | ZX81 | oh |
| 22:57.38 | ZX81 | maybe check if you have defaultzone and loadzone |
| 22:57.46 | Strom_M | oh! haha, durhhhhh |
| 22:58.06 | x86 | hah |
| 22:58.22 | x86 | funny, coming from a dCAP instructor ;) |
| 22:58.56 | Strom_M | no, I forgot to write that into the script I wrote to automatically generate the configs for me based on my responses to a half dozen questions |
| 22:59.09 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 23:00.53 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 23:01.54 | DrukenLPY | is there a variable for the current users username? |
| 23:02.35 | Strom_M | "username" in what sense" |
| 23:02.36 | Strom_M | ? |
| 23:02.44 | x86 | SIP user? |
| 23:02.56 | Strom_M | x86: haha, yeah, that was it. loadzone. |
| 23:03.02 | Strom_M | braaaaaaaaaaaaaaaaaain faaaaaaaaaaaaaaaaaaaaart |
| 23:03.15 | x86 | lolz |
| 23:04.16 | DrukenLPY | yeah, the sip username |
| 23:04.20 | DrukenLPY | basically for voicemail |
| 23:05.11 | `paul | in CDR (mysql) a call is being passd thru several agents in queue how do i count the number of calls(unique) made... (not the number of entries in the CDR) |
| 23:05.19 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 23:05.40 | Strom_M | DrukenLPY: why not just use callerid number? |
| 23:05.42 | plasmid | what's the fastest way to view how much memory my workstation has? |
| 23:06.09 | Strom_M | look at the spec sheet you've already glued to the side of it |
| 23:06.21 | DrukenLPY | Strom_M: well... i dunno... cause i don't have my system setup to use CID? hehehe |
| 23:06.44 | Strom_M | you don't? |
| 23:06.52 | DrukenLPY | not for voicemail no... |
| 23:06.58 | DrukenLPY | i used to use accountcodes... |
| 23:07.11 | Strom_M | do the phones have caller ID number? |
| 23:07.18 | DrukenLPY | but now i have multipul "peers" assigned to a single accountcode |
| 23:07.20 | Strom_M | and are the mailboxes the same as that caller ID number? |
| 23:07.28 | Strom_M | spelling: multiple :) |
| 23:07.51 | DrukenLPY | no, mailboxes match the peers id number, so it's username |
| 23:12.23 | Strom_M | in theory, you could just use ${CHANNEL} in conjunction with CUT() |
| 23:13.17 | DrukenLPY | nah... too messy... i'll look over my tables and see what common fields i have... must be a way to do it... i just gotta figure out which :0 |
| 23:13.29 | *** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 23:16.25 | AdamB0122 | ugh |
| 23:16.27 | AdamB0122 | io'm confused |
| 23:16.32 | AdamB0122 | I can call out fine |
| 23:16.44 | AdamB0122 | I'm using a T1 card connected into a Channelbank |
| 23:16.48 | AdamB0122 | but when I call in |
| 23:16.51 | JT | plasmid: "free" |
| 23:17.02 | ZX81 | Strom_M: was that the problem? |
| 23:17.26 | AdamB0122 | the CLI tells me i've got something coming in on Zap/1-1, goes through a set of things, but then jumps to Zap/2-1 |
| 23:17.31 | AdamB0122 | http://rafb.net/p/dgIrBF77.html |
| 23:17.35 | AdamB0122 | ^ Log |
| 23:18.12 | AdamB0122 | and i never hear anything, and right as I see things with Zap/2-1 start in the CLI, the call drops |
| 23:18.30 | JT | AdamB0122: did you write that dialplan? |
| 23:19.07 | AdamB0122 | no |
| 23:19.16 | JT | i didn't think so |
| 23:19.17 | AdamB0122 | lemme poke around in the dialplan and see if somehow its being told to do something stupid |
| 23:19.21 | JT | it looks like a mess |
| 23:19.26 | JT | start from basics |
| 23:22.58 | AdamB0122 | good god. |
| 23:23.14 | AdamB0122 | well, this dialplan is pretty much screwed up |
| 23:23.52 | JT | AdamB0122: have you worked out which channels are available to your pbx yet? |
| 23:24.16 | AdamB0122 | not yet |
| 23:24.32 | AdamB0122 | and because the old phone system company did things, XO communications doesn't know |
| 23:24.36 | AdamB0122 | and that company went under |
| 23:24.51 | AdamB0122 | so i'll have to figure out dip switches or guess and check |
| 23:24.58 | JT | i think the dip switch config on the adit would be helpful |
| 23:25.08 | JT | otherwise check every channel dialling out individually on them |
| 23:25.19 | AdamB0122 | Yea. |
| 23:25.26 | AdamB0122 | still have to give the dial to work as well |
| 23:25.44 | JT | give the dial to work? |
| 23:25.47 | AdamB0122 | all i gotta say, is wtf is the point of trixbox? its done nothing but be a pain in the ass. |
| 23:25.50 | AdamB0122 | get* |
| 23:26.03 | AdamB0122 | in asterisk, my dial command isn't loading, so i've got a missing module somewhere |
| 23:26.09 | AdamB0122 | gotta google that one |
| 23:26.29 | JT | thought you said you could dial out fine |
| 23:26.45 | AdamB0122 | from a sip phone, where asterisk just pics a working channel |
| 23:26.52 | AdamB0122 | not from the *CLI |
| 23:26.54 | JT | use zap groups |
| 23:27.01 | JT | i never make calls from the cli |
| 23:27.23 | JT | you need a sound card to do that, none of my asterisk boxes have sound cards |
| 23:27.38 | AdamB0122 | mine doesn't |
| 23:29.45 | AdamB0122 | hm. |
| 23:29.52 | AdamB0122 | ok, for now, I know channel 1 is good. |
| 23:30.01 | delmar | hrm. Is DISA really touchy about pattern matching and such? I seem to be able to get some patterns to match fine and others not at all. |
| 23:30.20 | AdamB0122 | JT > If a call is coming IN on the T1, does it do a sort-order, (start with channel 1, and move up untill it gets an available slot)? |
| 23:30.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:31.16 | JT | delmar: i don't think that's a function of disa |
| 23:31.46 | JT | AdamB0122: yes, whether it comes in ascending or descending order depends on telco |
| 23:32.09 | AdamB0122 | JT > its definately ascending |
| 23:33.00 | delmar | ok this is crazy.. |
| 23:33.23 | delmar | I had disa set to use [local] but for testing i changed it to [disa-out] and created some test strings in there. |
| 23:33.46 | delmar | so i just gave up on it.. switched back to [local] and tested for the hell of it.. now it works |
| 23:33.54 | delmar | unreliable imo. |
| 23:34.01 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:37.54 | delmar | yep. DISA 50% fail so far without changing anything or reloading. it either dials the number and works, or it just hangs up without anything useful on the console. |
| 23:39.11 | delmar | ah i think I have an idea whats going on. |
| 23:39.20 | delmar | must be poor dtmf detection.... |
| 23:39.37 | AdamB0122 | ok |
| 23:39.40 | delmar | noticed it doesnt hear some digits when logging in sometimes.. must be that. |
| 23:39.44 | AdamB0122 | I'm getting somewhere |
| 23:39.58 | AdamB0122 | i changed my dialplan, just to do a basic Dial(SIP/140) |
| 23:40.10 | AdamB0122 | so when an inbound call comes in, it just rings my extension |
| 23:40.20 | AdamB0122 | My phone rings, but the second i pick up, the call is dropped |
| 23:41.42 | *** join/#asterisk sharp (n=sharp@dsl092-238-219.phl1.dsl.speakeasy.net) |
| 23:42.00 | snuff-work | could be no codec translation path.. that will make call die on pickup |
| 23:42.11 | *** join/#asterisk TedNJ37 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 23:42.22 | TedNJ37 | Hi guys. I have a problem. I am not able to ping my asterisk box by name, only by IP. What am I doing wrong? |
| 23:42.39 | AdamB0122 | you have a dns server not working |
| 23:43.02 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 23:43.05 | AdamB0122 | exteneral or local? |
| 23:43.12 | TedNJ37 | Internal. |
| 23:43.30 | TedNJ37 | I'll google it and see how I can get it to work. |
| 23:43.39 | AdamB0122 | could be that its just not set to broadcast |
| 23:44.38 | TedNJ37 | How do I check that? |
| 23:45.14 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 23:47.07 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 23:47.58 | AdamB0122 | cant remember |
| 23:48.08 | AdamB0122 | Not really a asterisk issue though, thats linux |
| 23:48.18 | AdamB0122 | try in #linux or w/e your distro is |
| 23:48.26 | AdamB0122 | Hm. |
| 23:49.36 | *** join/#asterisk Strom_C (n=strom@h72-2-22-215.bigpipeinc.com) |
| 23:49.40 | AdamB0122 | I'm not fluent in dialplans anymore (its been like 8-9 months since I've touched an Asterisk box) |
| 23:49.41 | AdamB0122 | http://rafb.net/p/7FxIKg45.html |
| 23:50.03 | AdamB0122 | After I've dialed the 140 extension, what do i need to do to connect the inbound Zap/1-1 call to SIP/140? |
| 23:50.06 | TedNJ37 | Thanks. |
| 23:50.45 | ZX81 | AdamB0122: just that should work assuming your phone is not doing g729 etc |
| 23:50.48 | AdamB0122 | As of right now, it dials my extension, but once its finished dialing, the incoming zap/1-1 just gets disconnected |
| 23:51.09 | AdamB0122 | my phones just a standard Xlite phone |
| 23:51.14 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 23:51.15 | AdamB0122 | and thats what I thought. |
| 23:51.26 | JT | xlite is not really a standard phone ;) |
| 23:51.38 | AdamB0122 | trying Express talk |
| 23:52.02 | AdamB0122 | (yea, I dont have any sip hardphones) |
| 23:52.12 | ZX81 | er ok, no0b question, where does the make progdocs put the files? |
| 23:52.13 | AdamB0122 | same thing |
| 23:52.14 | JT | all softphones suck, make sure you get some real phones eventually |
| 23:52.23 | AdamB0122 | We will be |
| 23:53.12 | AdamB0122 | Boss wants to see the benefits of this system before dishing out the dough for all the phones |
| 23:53.17 | DrukenLPY | JT: hear hear, REAL DESKTOP PHONES!!!! |
| 23:53.18 | ZX81 | ah |
| 23:53.19 | ZX81 | doc/api |
| 23:53.20 | ZX81 | :) |
| 23:53.45 | AdamB0122 | hm |
| 23:54.33 | ZX81 | AdamB0122: just try: |
| 23:54.36 | ZX81 | exten => s,1,Answer |
| 23:54.45 | ZX81 | <PROTECTED> |
| 23:54.51 | ZX81 | <PROTECTED> |
| 23:55.27 | JT | the parantheses after Echo are optional ;) |
| 23:55.57 | ZX81 | heh but look nicer |
| 23:56.03 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:56.03 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 23:56.09 | AdamB0122 | inside, dialing 7777, that works fine. |
| 23:56.13 | AdamB0122 | from the outside |
| 23:56.19 | JT | ZX81: by that token you should be putting them after Answer too |
| 23:56.22 | AdamB0122 | i see the Executing answer (zap/1-1) ect ect |
| 23:56.25 | AdamB0122 | the background |
| 23:56.35 | JT | wifi + voip, not a fan |
| 23:56.36 | AdamB0122 | and the exceuting echo, but i do not hear anything |
| 23:56.51 | JT | does it hang up? |
| 23:56.54 | JT | try talking. |
| 23:57.03 | ZX81 | JT: funny I didn't - normally I do :) |
| 23:57.07 | ZX81 | or Answer(2) |
| 23:57.14 | AdamB0122 | yes |
| 23:57.18 | AdamB0122 | it does hang up |
| 23:57.25 | JT | something is defective |
| 23:57.30 | AdamB0122 | and it does not echo anything |
| 23:57.39 | ZX81 | AdamB0122: what kind of card? |
| 23:58.05 | AdamB0122 | TE120P T1 card |
| 23:58.39 | ZX81 | T1 settings maybe |
| 23:58.48 | ZX81 | here you can get that if you don't have crc4 |
| 23:58.48 | AdamB0122 | when i do lsmod, I have zaptel numbers 8 zttranscode,wcte12xp |
| 23:58.52 | AdamB0122 | what is zttranscode |
| 23:58.52 | ZX81 | but we're E1 |
| 23:59.08 | ZX81 | for the transcoder card thingy |
| 23:59.45 | AdamB0122 | turned on crc4 for kicks |
| 23:59.52 | JT | don't turn on crc4 |
