00:00.03 | hfb | "ManxPower vile, You have to buy new handsets. PBX companies make their handsets incompatable with ANYTHING else so their customers are locked into buying their overproced phones." |
00:00.06 | hfb | Argh. |
00:00.31 | ManxPower | Hmm? |
00:00.37 | ManxPower | Ah. |
00:00.48 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
00:01.00 | hfb | Wow something you posted 2 years ago. :) |
00:01.09 | ManxPower | sofh, Um, all bandwidth measurements are ONE WAY |
00:01.16 | justinu | there's a company that sells very expensive SIp gateway for definitey, avaya, nec d-term, nortel phones |
00:02.08 | justinu | settle down, beavis |
00:02.49 | FuriousGeorge | your nothing without me |
00:02.59 | FuriousGeorge | *you're |
00:03.04 | ManWithYellowBat | you = pwnd. |
00:03.17 | ManWithYellowBat | MONKEY >:O |
00:03.28 | FuriousGeorge | Sapien |
00:03.37 | ManWithYellowBat | boo |
00:03.52 | FuriousGeorge | you know i'm 5 to 7 times stronger than the average man dont you |
00:04.00 | ManWithYellowBat | you're not a gorilla |
00:04.06 | ManWithYellowBat | you're a very small monkey |
00:04.19 | FuriousGeorge | damn straight im a carnivour |
00:04.19 | ManWithYellowBat | and i have opposable thumbs :) |
00:04.23 | FuriousGeorge | and a bad mofo |
00:04.38 | FuriousGeorge | dude, i got opposable thumbs on me feet |
00:04.56 | FuriousGeorge | that means i can punch you and kick you with all four appendages |
00:05.00 | ManWithYellowBat | haha |
00:05.36 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
00:06.18 | giggles | anyone know how to program a mediatrix 2102? |
00:06.30 | FuriousGeorge | ManWithYellowBat: that was me: http://www.cbsnews.com/stories/2005/03/07/national/main678634.shtml |
00:06.34 | justinu | monkey feet |
00:06.36 | fugitivo | anyone with experience with R2 + unicall ? |
00:09.05 | justinu | do any IP phones support ipv6? |
00:09.20 | buu | Argh. Fufkcing hold music |
00:10.28 | buu | There has to be some way to easily change which music it's playing. |
00:10.31 | ManWithYellowBat | George :http://www.cfhf.net/lyrics/images/super-mario.jpg |
00:10.33 | ManWithYellowBat | that's me :) |
00:10.46 | ManWithYellowBat | oop... http://www.cfhf.net/lyrics/images/super-mario.jpg |
00:10.58 | buu | Anyone? |
00:11.29 | justinu | buu: put different mp3's into /var/lib/asterisk/moh |
00:11.32 | justinu | or whatever the directory is |
00:11.38 | buu | justinu: And how do I stop it playing the current one? |
00:11.42 | buu | What if I remove one? |
00:11.55 | justinu | moh reload |
00:11.58 | justinu | delete it |
00:11.59 | xevo | PoE doesn't require different patch cables, does it? |
00:11.59 | buu | moh ? |
00:12.03 | justinu | mod reload from the CLI |
00:12.05 | justinu | xevo: no |
00:12.08 | r_evolution | hey buu... you should also use lame and sox to convert it to a raw file : ) |
00:12.13 | justinu | however, if we're talking polycom |
00:12.15 | buu | r_evolution: Um. Why? |
00:12.21 | xevo | they're just trying to upsell me then |
00:12.23 | justinu | only the 601 supports native 802.11af |
00:12.39 | r_evolution | because it will free up some of your processor load |
00:12.39 | justinu | the 301/501 need a special cable/adapter |
00:12.51 | xevo | oh really? |
00:12.53 | buu | r_evolution: Oh, well, eh |
00:12.54 | xevo | odd |
00:13.08 | buu | r_evolution: More concerned with making it work than making it work fast. |
00:13.23 | r_evolution | well... part of making it work is making it work with more users, right? :) |
00:13.33 | justinu | xevo: i know... i just ordered 20 IP301's w/ the special connector |
00:13.37 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
00:13.51 | xevo | how much were each of the connectors? |
00:13.59 | justinu | like 30 bucks |
00:14.03 | xevo | ouch |
00:14.05 | justinu | yeah |
00:14.10 | justinu | customer's money tho... |
00:14.12 | justinu | their problem |
00:14.16 | xevo | hehe |
00:14.34 | *** part/#asterisk dec (n=tom@ppp151-30.lns3.adl2.internode.on.net) |
00:14.46 | xevo | Do 501s come with power adapters? |
00:14.58 | r_evolution | wha justin? the power injectors? |
00:17.32 | justinu | yeah, the injectors |
00:17.41 | justinu | 501s can be ordered with the injectors |
00:17.53 | dja_ | Hi. Any hints on how to minimize echo? I'm going from AnalogPhone->ATA->Asterisk->VOIPProvider->PSTN->AnalogPhone. The echo is driving me crazy. :) |
00:18.08 | justinu | dja: on your ATA, turn down your gains |
00:18.53 | dja_ | justinu: thanks -- I'll give that a try. |
00:18.58 | justinu | good luck |
00:23.32 | xevo | So it's just a special cable if they have 802.3af switches? |
00:24.43 | asterboy | Anyone with some insight into why I can call out no problem, but when I try to call in, no voice! |
00:25.21 | asterboy | ztmonitor registers the * side, but nothing coming from the pots side. |
00:25.56 | asterboy | Works when I use a Digium FXS card, but not for my Polycom SIP setup. |
00:26.13 | asterboy | So I'm sure the FXO ZAP Channel is fine. |
00:26.20 | justinu | xevo: a cable with a small box in the middle of it |
00:26.42 | justinu | box has an LED that lights up when it gets power from the switch, but isn't plugged into the phone |
00:27.03 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
00:27.15 | asterboy | moved the FXO cards to a new server...same. |
00:27.19 | *** join/#asterisk outtolunc (n=me@c-24-23-162-126.hsd1.ca.comcast.net) |
00:27.22 | xevo | ahh ok |
00:27.35 | *** part/#asterisk outtolunc (n=me@c-24-23-162-126.hsd1.ca.comcast.net) |
00:30.21 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
00:36.47 | Zodiacal | anyone know what could cause constant static on an fxo module? a reboot fixes it, but only for about 12 hours, the it reapears on a random fxo module in my system |
00:37.00 | Zodiacal | the = then |
00:37.18 | justinu | irq conflict |
00:37.20 | Zodiacal | nope |
00:37.40 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
00:37.43 | justinu | since the module is random, it's gotta be a software problem |
00:38.23 | Zodiacal | justinu any ideas where to begin figuring out where? |
00:39.03 | justinu | figure out what happens when it goes bad |
00:39.08 | justinu | look at your messages.log |
00:39.10 | justinu | full log |
00:39.26 | justinu | check /proc/interrupts when it's working, recheck when it's borked |
00:39.31 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
00:40.01 | X-Rob | Zodiacal, does loading and unloading the module fix it? |
00:40.28 | Zodiacal | i'll have to try again when it happens |
00:40.30 | Zodiacal | but i don't think so |
00:40.41 | Zodiacal | err, can't remember ;P |
00:40.53 | Zodiacal | its happend 3 times so far.. |
00:41.04 | Zodiacal | it will probably happen again at noon |
00:41.24 | Zodiacal | i'll try those suggestions, its hard to figure out exactly when it happens tho |
00:41.35 | Zodiacal | i will have to constantly pickup the line every 5 mins or so |
00:48.29 | *** join/#asterisk NeonLevel (n=NeonLeve@200.52.142.186) |
00:49.12 | *** join/#asterisk nahirean (n=nahirean@67.132.43.2) |
00:49.44 | NeonLevel | good day everyone, anyone has setup a sip provider on a diferent sip port than 5060? i cannot get this working. i've read it and i think it may be the port= parameter but it wont accept it. has someone has done this? thanks in advance |
00:50.03 | justinu | bindport= |
00:52.14 | ManxPower | NeonLevel, I believe port= is the DESTINATION port for that device. |
00:52.38 | NeonLevel | ManxPower: that is exactly what im trying to do |
00:53.00 | NeonLevel | the destination according to my provider it would be 5070 |
00:53.03 | NeonLevel | no 5060 |
00:53.09 | ManxPower | do you have a remote SIP device that's listening on port 5070? |
00:53.21 | NeonLevel | is a SIP terminator to the PSTN |
00:53.31 | NeonLevel | it's a big provider here in my country |
00:54.00 | nahirean | hello, I am using a .call file to trigger a call in the spooler.. however, once it references the context - it keeps looping to the dial string, and doesnt pass beyond that even when I pick up the call.. may I post a 6 line context, perhaps someone can take a look? |
00:54.01 | ManxPower | so you have a [provider] section in sip.conf with a port=5070 in it? |
00:54.11 | justinu | call a SIP uri: like sip:user@voipprovider.com:5070 |
00:54.15 | NeonLevel | that is corret |
00:54.35 | NeonLevel | i haven't thought of that.... |
00:54.39 | NeonLevel | it might work |
00:55.00 | NeonLevel | but since i find the parameter port, i'd be struggling with it |
00:55.07 | NeonLevel | sorry to bother you guyes with it |
00:55.16 | nahirean | there are no goto commands in the context, i dont know why it's looping |
00:56.14 | nahirean | it follows the context correctly until the call is answered, then it repeats the context |
00:57.29 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
00:58.08 | synthetiq | anyone here use spandsp ? ive compiled is and everythiung and when i use asterisk -vvvvvv it stops loading at app_rxfax.so ...any ideas why? |
00:58.31 | asterboy | How do the channels #s get assigned to wcfxo cards? Are they whatever you set in zaptel.conf? |
00:59.37 | asterboy | I'm getting channels 1 and 2, but my 3rd card won't load even though it shows in interrupts |
01:00.26 | NeonLevel | justinu: how would be the syntax for a sip uri, with a number to call, username, password, host, port ? |
01:02.24 | NeonLevel | anyone??? |
01:02.34 | r_evolution | Neon try registering the provider in the sip.conf |
01:03.15 | r_evolution | register => :@66.185.167.228 |
01:03.42 | NeonLevel | i did register it, and it registers ok |
01:03.47 | r_evolution | register => SIP:PW@PROVIDER/EXTEN |
01:03.57 | r_evolution | then use that for outbound calling in the extensions |
01:04.03 | NeonLevel | and the incoming calls are OK |
01:04.06 | *** join/#asterisk asteriskmonkey (n=phil@69.158.146.217) |
01:04.20 | NeonLevel | but when i try to place a call using this SIP channel, it simply wont work |
01:04.35 | asteriskmonkey | anyone know any possible reasons why a tmd2400 card cant be modprobed? |
01:04.36 | r_evolution | how are you formatting it? |
01:04.46 | asteriskmonkey | alsthough i do see an extra ethernet card in lspci |
01:05.20 | r_evolution | exten = _1ZXXXX.,1, Dial(SIP/1${EXTEN:1}@provideraddress) |
01:05.23 | r_evolution | is what i use for long-dis |
01:05.35 | r_evolution | you can actually not use the EXTEN:1 |
01:05.38 | r_evolution | and just use EXTEN |
01:05.43 | r_evolution | and drop the 1 after the SIP/ |
01:05.46 | NeonLevel | this is the line i have in the general sip.conf <<<< register => username:password@200.66.96.57:5070/335004301 and it registers ok |
01:05.56 | r_evolution | ok what about your extensions.conf? |
01:06.06 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
01:06.37 | NeonLevel | extensions.conf <<<< exten => _8.,1,Dial(SIP/${EXTEN:1}@username:password/200.66.96.57:5070||Tr) |
01:07.11 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
01:07.14 | r_evolution | did you register a peer in the sip.conf? |
01:08.12 | NeonLevel | well i did commented the peer, in sip.conf because i thought it won't be necesary dialing with a sip uri |
01:08.37 | NeonLevel | you want me to uncommenting it? |
01:09.10 | r_evolution | you can... it's usually easier that way (for me) |
01:09.24 | r_evolution | b/c then you just put SIP/ etc @ [whateverisbetweenbrackets] |
01:09.26 | NeonLevel | ok hold on |
01:09.47 | asterboy | How are wcfxo cards assigned channels? |
01:10.27 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
01:11.10 | r_evolution | dunno aster... i don't use a tdm card |
01:11.24 | asterboy | They are the fxo. |
01:11.33 | r_evolution | i know... i dont use one |
01:11.48 | r_evolution | i was under the impression that they were activated in the zaptel.conf file |
01:11.56 | Zodiacal | now that would be neat |
01:12.00 | asterboy | thats what I thought. |
01:12.01 | r_evolution | no kidding |
01:12.03 | r_evolution | that would be the hotness |
01:12.36 | asterboy | audio wireless sip fones don't work, so why may video? |
01:12.57 | Zodiacal | which have you tried? |
01:13.16 | Zodiacal | asterboy |
01:13.17 | r_evolution | i dunno... the one our CIO uses seems to work pretty well |
01:13.35 | r_evolution | the ones pulver was selling.. the bcm taiwanese phones |
01:13.43 | asterboy | The Zyxel |
01:13.55 | Zodiacal | isn't the zyxel the cheapest of the bunch |
01:14.05 | r_evolution | Zyxel and UT Starcom |
01:14.15 | asterboy | Starcom has bad delay |
01:14.18 | r_evolution | the grandstream of WiSIP phones ;x |
01:14.43 | Zodiacal | i just want ViWiSiP too much to ask? |
01:14.45 | asterboy | I haven't a hope of landing this contract using * |
01:14.51 | r_evolution | way too much |
01:15.02 | Zodiacal | once my fios gets lit up... that would rock |
01:15.07 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
01:15.26 | asterboy | Echo on the line, so I reduce TX gain, then it's too low a volume...catch 22 |
01:16.08 | asterboy | Can call into ZAP FXO from SIP phone but can't hear calling party. |
01:16.32 | asterboy | Can't figure how the fucking thing assigns channel numbers to try other FXO cards. |
01:16.51 | asterboy | Still need to figure out call transfer and call forward. |
01:18.07 | Zodiacal | anyone know if its posible some how to use a web cam and a sip hardphone ? |
01:18.18 | asterboy | Time to look at other solutions...OpenPBX any good? |
01:18.52 | r_evolution | open PBS? |
01:18.54 | Zodiacal | openPBS isn't that redudent? |
01:18.55 | Zodiacal | :P |
01:19.05 | r_evolution | planning on running our own Public Broadcasting System? |
01:19.33 | asterboy | lol |
01:19.43 | asterboy | short for PuBeS |
01:19.53 | asterboy | open pubes |
01:20.10 | Zodiacal | r_evolution ever hear of such a setup? pc with webcam and hardphone/sip ? |
01:20.17 | Zodiacal | to make a video call. can Sip do things like that? |
01:20.20 | Zodiacal | and asterisk? |
01:20.46 | r_evolution | dunno... i know * possibly has some video support |
01:22.19 | r_evolution | show video codecs |
01:22.20 | r_evolution | Disclaimer: this command is for informational purposes only. |
01:22.20 | r_evolution | <PROTECTED> |
01:22.20 | r_evolution | <PROTECTED> |
01:22.20 | r_evolution | -------------------------------------------------------------------------------- |
01:22.20 | r_evolution | <PROTECTED> |
01:22.22 | r_evolution | <PROTECTED> |
01:22.24 | r_evolution | <PROTECTED> |
01:22.26 | r_evolution | see |
01:22.29 | r_evolution | i havent messed with it at ALL though |
01:23.02 | Zodiacal | someone on the wiki said that "Cisco VT Advantage" can do it |
01:23.04 | Zodiacal | hrmm. |
01:23.47 | Zodiacal | i guess just setting up some kind of webcam software to turn on when the phone rings and to send the user params is all thats nessisary |
01:24.38 | Zodiacal | hehe |
01:32.47 | *** join/#asterisk Eitch (i=[U2FsdGV@unaffiliated/eitch) |
01:33.06 | synthetiq | anyone here use spandsp ? ive compiled is and everythiung and when i use asterisk -vvvvvv it stops loading at app_rxfax.so ...any ideas why? |
01:33.50 | r_evolution | im not using spandsp yet synth ;x |
01:33.52 | r_evolution | but soon! |
01:35.11 | synthetiq | =/ |
01:50.47 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
01:53.46 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
01:56.44 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
02:02.46 | *** join/#asterisk file[laptop] (n=jcolp@142.131.190.116) |
02:03.11 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
02:03.13 | file[laptop] | hello world |
02:03.44 | justinu | it's back |
02:03.55 | justinu | i rm -rf'ed that file |
02:04.06 | file[laptop] | :\ |
02:04.08 | Eitch | LOL |
02:04.13 | Eitch | i liked this new verb |
02:05.19 | file[laptop] | I was going to get a muffin this morning... but they didn't have any good ones |
02:05.51 | Qwell[laptop] | file[laptop], liar! |
02:06.09 | file[laptop] | nope, quite true |
02:06.32 | Qwell[laptop] | I asked them. "Has file been there?" "No." |
02:06.46 | file[laptop] | I went to the little cafe place in the Marriott... |
02:06.48 | file[laptop] | quite good |
02:06.56 | Qwell[laptop] | oh? |
02:19.24 | exten123 | hey guy how to solve this failed to pass IP ACL when trying using Asterisk Manager? |
02:21.15 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
02:25.51 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.97.246) |
02:29.00 | *** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net) |
02:29.50 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lm4.cable.mindspring.com) |
02:32.12 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
02:33.16 | *** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net) |
02:37.45 | willt | Do I have to use the ztdummy module to use the meetme application? What if im using FreeBSD? |
02:38.36 | *** join/#asterisk asteriskmonkey (n=phil@69.158.146.217) |
02:38.49 | asteriskmonkey | anyone know why a tdm2400 serious card would modprobe? |
02:38.53 | asteriskmonkey | it dosnt show up at all |
02:39.00 | Qwell[laptop] | asteriskmonkey, why it *would* modprobe? |
02:39.07 | asteriskmonkey | sorry wouldnt |
02:39.08 | asteriskmonkey | :P |
02:39.15 | Qwell[laptop] | there are various reasons |
02:39.20 | asteriskmonkey | using centos 4.2 |
02:39.24 | Qwell[laptop] | first things first - does lspci show it? |
02:39.29 | asteriskmonkey | no |
02:39.34 | Qwell[laptop] | call Digium |
02:39.39 | asteriskmonkey | shows unkonw device |
02:39.55 | asteriskmonkey | digiums not open right now |
02:40.00 | Qwell[laptop] | email Digium |
02:40.07 | asteriskmonkey | from your experience what is this issue? |
02:40.11 | Qwell[laptop] | got me |
02:40.18 | asteriskmonkey | ive had it in the past but its been due to a bad motherboard |
02:40.53 | Zipper_32 | hmm, my x100p clone is showing up on lspci as: Communication controller: Motorola: Unknown device 5608 .... I have a feeling that's at least the source of one error of mine. |
02:48.23 | ManxPower | contact the vendor you bought it from |
02:51.11 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
02:54.21 | *** join/#asterisk digg10 (n=john@206-248-135-54.dsl.teksavvy.com) |
02:59.34 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
02:59.56 | synthetiq | anyone here use spandsp ? ive compiled is and everythiung and when i use asterisk -vvvvvv it stops loading at app_rxfax.so ...any ideas why? |
03:01.44 | asteriskmonkey | yep its not compiled properly |
03:02.56 | *** join/#asterisk nvicf (n=nvicf@201.250.165.83) |
03:03.04 | nvicf | hello |
03:03.32 | synthetiq | but it did compile so...? |
03:04.03 | nvicf | I have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this? |
03:04.14 | digg10 | ne1 managed to build a web interface for managing the dialplan? |
03:04.29 | synthetiq | web interface |
03:04.32 | synthetiq | hah |
03:04.52 | digg10 | funny? |
03:05.43 | asteriskmonkey | synthetiq: i had the same isse i had to delete all the old files and recompile it |
03:05.54 | digg10 | i need to control the dialplan from browser |
03:06.05 | asteriskmonkey | user a m p then |
03:07.01 | Qwell[laptop] | freepbx.. |
03:07.06 | Qwell[laptop] | ~freepbx |
03:07.32 | asteriskmonkey | ha freepbx wants money so why call if freepbx lol |
03:07.51 | Qwell[laptop] | for a similar reason |
03:08.32 | asteriskmonkey | just write some php and make your dial plan sql driven :) |
03:10.25 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
03:15.02 | exten123 | Why I can't connect Asterisk Manager from others computer? |
03:20.20 | digg10 | why is this irc channel not more active? |
03:20.51 | *** join/#asterisk Psyiode (n=lacigol@205.241.238.186) |
03:21.15 | Psyiode | im having trouble ringing a sip cisco 7940 |
03:21.40 | Psyiode | I've setup queues and agents, and it will ring sip softphones, but the ciscos still wont ring... what am i missing.. |
03:22.50 | digg10 | how can i do time of day routing? |
03:25.18 | synthetiq | gotoiftime |
03:25.20 | synthetiq | exit |
03:25.21 | *** join/#asterisk habakuk (n=chatzill@c-24-6-173-113.hsd1.ca.comcast.net) |
03:25.43 | Psyiode | include => yourcontext|time-time|day-day|*|* that will route the call to yourcontext during the specified time during specified days, do that for all your time and days |
03:25.46 | Psyiode | or here http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+openhours |
03:26.00 | *** join/#asterisk jjg_ (n=doink@dsl081-245-050.sfo1.dsl.speakeasy.net) |
03:26.03 | jjg_ | hi |
03:26.43 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
03:26.58 | jjg_ | i'm having trouble getting my granstream budgetone 100 to register using NAT through a linksys .... the phone is in the DMZ and i am getting 401's over and over, any recommendations? |
03:27.17 | jjg_ | i followed teh instructions on voip-info on setting up the grandstream and sip.conf file |
03:27.57 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
03:28.06 | habakuk | Hey folks. I have a question. I'm using manager app using Originate. The problem is it tends to block until the channel is established. Any ideas on how to configure is so that the channel immediately connects? I was considering using the local channel. Would that solve the problem? |
03:29.21 | habakuk | jjg_ the phone is behind NAT or the asterisk is behind NAT? |
03:29.43 | jjg_ | habakuk , the phone is behind NAT |
03:29.47 | digg10 | the include => line goes in a seperate section, or inside a context? |
03:29.56 | *** join/#asterisk bit123 (n=bit123@203.115.15.252) |
03:30.16 | bit123 | how to do ad-hoc conferencing with asterisk ? |
03:30.27 | Psyiode | digg10: i have it in my default context, which is my first context, which only contains openhour included |
03:30.31 | habakuk | jjg_: so does the phone respond to the 401? is it getting the 401? |
03:30.40 | tengulre | anybody know which website provider software exchange services? |
03:31.14 | jjg_ | habakuk , i don't think so, is there a way to turn on debugging alerts on the phone? |
03:31.50 | habakuk | jjg_: get a cheap hub and a laptop running ethereal |
03:32.13 | nvicf | I have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this?thanks |
03:33.53 | orlock | man |
03:34.00 | orlock | sip/*/grandstreams are loosing me |
03:34.04 | orlock | anybody here used Sail? |
03:35.45 | dippo | no, what is it? |
03:36.01 | dippo | which grandstream phones are you using? |
03:36.05 | orlock | its an interface for managing asterisk via the smeserver interface |
03:36.13 | orlock | gxp2k |
03:37.04 | dippo | i bought a slew of grandstream budgetones for our office |
03:37.08 | dippo | perhaps not a wise decision in retrospect |
03:37.45 | *** join/#asterisk litage (n=nick@203.220.55.70) |
03:37.53 | FuriousGeorge | so i think qualify is causing some of my iax peers to drop out and not come back. specifically my iax provider |
03:37.53 | bit123 | hi, anybodyknows how to do ad-hoc conferencing with asterisk ? |
03:38.05 | FuriousGeorge | you guys think switching to sip will remedy that? |
03:38.12 | dippo | what's qualify? |
03:38.51 | FuriousGeorge | a setting that measures the latency for sip protocol communication and desides whether that peer qualifies to be used |
03:40.12 | nvicf | I have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this?thanks |
03:40.24 | FuriousGeorge | between this, and *'s apparent inability to handle .dynu addresses with any measure of success, im thinking about just restarting all the servers at 4AM every day |
03:41.37 | willt | can't you set qualify=no? |
03:41.51 | nvicf | me? |
03:41.55 | *** join/#asterisk chendy (n=hello_vi@218.80.62.113) |
03:42.05 | orlock | dippo: yeah |
03:42.13 | FuriousGeorge | yeah, but i like that safety net. my boss wont care why the call quality stinks |
03:42.23 | orlock | dippo: when i punch in the auth details, update, reboot... the old auth username is still there! |
03:43.01 | willt | FuriousGeorge: o i c.. I haven't got a chance to play with IAX yet all sip here.. |
03:43.15 | orlock | rt? |
03:43.16 | orlock | whats that? |
03:43.43 | FuriousGeorge | i guess ill talk to my procider about if changing to sip will help |
03:45.04 | *** join/#asterisk Flauto (n=zhao@71.194.38.112) |
03:46.21 | *** join/#asterisk jjg__ (n=doink@dsl081-245-050.sfo1.dsl.speakeasy.net) |
03:46.33 | jjg__ | habakuk i have the hub now, gonna see if i can see the messages |
03:49.38 | *** join/#asterisk escribzz (n=manzzzee@VDSL-130-13-28-196.PHNX.QWEST.NET) |
03:51.57 | *** join/#asterisk bmg505 (n=leon@dsl-146-16-10.telkomadsl.co.za) |
03:52.16 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
03:52.37 | De_Mon | hrm, I recompiled app_meetme with some new code and did a 'reload app_meetme.so', but nothing's changed. Do I have to restart asterisk? |
03:52.41 | *** part/#asterisk Psyiode (n=lacigol@205.241.238.186) |
03:53.50 | *** join/#asterisk MarioGamboa (n=yo@201.133.229.135) |
03:54.00 | MarioGamboa | hi everyone |
03:57.36 | nvicf | que hace marito |
03:58.15 | *** join/#asterisk punkgode (n=punkgode@r200-125-63-195-dialup.adsl.anteldata.net.uy) |
03:59.36 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
04:00.19 | punkgode | hello, anyone knows what this error message is about? "chan_sip.c:12628 reload_config: Can't add wildcard IP address to domain list, please add IP address to domain manually." |
04:00.54 | punkgode | i've triple-checked the configuration files, I can't seem to find the problem |
04:01.10 | rpm | do you have any wildcard ip's? |
04:01.16 | rpm | x.x.x.* |
04:01.20 | punkgode | no |
04:01.32 | rpm | paste your sip.conf |
04:02.15 | punkgode | k |
04:04.55 | justnulling2 | <PROTECTED> |
04:05.14 | giggles | anyone know how to program a mediatrix 2102? |
04:05.42 | habakuk | giggles: yeah throw it in the trash |
04:06.20 | punkgode | rpm, http://debian-uy.pastebin.com/604789 |
04:06.27 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
04:06.32 | habakuk | giggles: actually you can configure it via SNMP |
04:06.48 | punkgode | rpm, i've removed the lines that started with ";" |
04:06.52 | giggles | snmp has been disabled |
04:10.18 | giggles | there appears to be a faciltity to download a config file in xml, anyone know the format? |
04:13.52 | punkgode | rpm, well, it was the bind address, sorry to bother |
04:14.04 | FuriousGeorge | i thought iax was supposed to be good with nat? |
04:14.23 | habakuk | FuriousGeorge: what's the issue? |
04:14.30 | FuriousGeorge | i got a bunch of boxes behind nat and dynamic ips, some cant register with eachother. others have their peers drop out |
04:14.37 | sevard | Does anyone have a HOP8T? |
04:14.42 | *** part/#asterisk punkgode (n=punkgode@r200-125-63-195-dialup.adsl.anteldata.net.uy) |
04:14.55 | FuriousGeorge | part of the problem is that asterisk looks up the servers by ip not by url |
04:15.01 | FuriousGeorge | the ip is dynamic the url isnt |
04:15.16 | FuriousGeorge | ok, fine, that sucks. but then the ip changes and they cant seem to find eachother |
04:15.37 | FuriousGeorge | add that to the fact that im noticing my iax provider peer drops out and the only solution is to restart the server |
04:15.44 | FuriousGeorge | i guess restarting networking would probably help too |
04:15.49 | FuriousGeorge | let me try that now |
04:16.22 | *** join/#asterisk litage (n=nick@202.168.41.172) |
04:17.45 | habakuk | FuriousGeorge: so you are saying your server is on a dyn ip? or your provider is? |
04:18.50 | FuriousGeorge | my server is. i run a service that monitors the ip and links it to a url even when it changes |
04:18.52 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
04:19.10 | FuriousGeorge | but nhow for instance, a reboot didnt get my other dynamic ip peers back. it will fix my provider dropping out |
04:19.20 | FuriousGeorge | this is all adding up to very frustrating |
04:19.23 | *** part/#asterisk escribzz (n=manzzzee@VDSL-130-13-28-196.PHNX.QWEST.NET) |
04:19.38 | FuriousGeorge | could you imagine if ssh just kinda sorta stopped working from time to time |
04:20.28 | FuriousGeorge | when 1.2 was comming out i thought this junk would be addressed, but it seems to be getting worse not better. just my opinion |
04:24.01 | FuriousGeorge | BOX1: 4 iax2 peers [2 online, 2 offline, 0 unmonitored BOX2: 4 iax2 peers [3 online, 1 offline, 0 unmonitored] BOX3: 4 iax2 peers [4 online, 0 offline, 0 unmonitored] BOX4: 4 iax2 peers [1 online, 0 offline, 3 unmonitored] |
04:24.20 | FuriousGeorge | these computers are all attempting to log into eachother, wtf gives with that |
04:24.39 | FuriousGeorge | how is one supposed to debug that |
04:25.42 | habakuk | FuriousGeorge: so, you are using something like dyndns.org right? Why not add an iax2/extension reload to your ifup script? |
04:26.10 | FuriousGeorge | ifup=cron? |
04:26.28 | habakuk | and then have a sed script that updates extensions.conf , iax.conf as well |
04:26.37 | FuriousGeorge | in any case it wont help |
04:26.47 | FuriousGeorge | update how? |
04:26.54 | FuriousGeorge | search by ip? |
04:27.05 | habakuk | well with dyndns you install a client on each server right? |
04:27.08 | *** join/#asterisk Eroick (n=chatzill@Ottawa-HSE-ppp269684.sympatico.ca) |
04:27.11 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
04:27.13 | habakuk | to update dyndns servers right? |
04:27.22 | FuriousGeorge | habakuk: no, on the lan's gateway,firewall box |
04:27.56 | FuriousGeorge | and its not like im using a microsoft wireless router or anything, this is a wired linux box running iptables at every location. makes no difference |
04:28.01 | willt | im seeing this when leaving voicemail: WARNING[1348]: file.c:981 ast_writefile: No such format 'gsm,wav' |
04:28.16 | FuriousGeorge | what surprises me is that no one else seems to have this problem |
04:28.16 | willt | and voice mail doesn't work |
04:28.18 | Eroick | Ok, so is this the idea behind Asterisk? I can have someone dial into my computer over Skype/Gizmo/Other VoIP and then program it to do whatever I want? Is there a tutorial to jump in? |
04:28.26 | FuriousGeorge | not skype |
04:28.33 | FuriousGeorge | ~docs |
04:28.36 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
04:29.08 | habakuk | FuriousGeorge: so you don't have a ddns client?? you lost me |
04:29.31 | FuriousGeorge | the gateway on the 4 respective networks runs the clients |
04:29.42 | FuriousGeorge | for the dyndns service |
04:29.45 | habakuk | move it to the servers.. problem solved |
04:29.51 | FuriousGeorge | why? |
04:29.59 | FuriousGeorge | pinging the url gives the right ip |
04:30.05 | FuriousGeorge | its asterisk that cant seem to handle the nat |
04:30.08 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
04:30.52 | FuriousGeorge | im only using iax b/c i heard it was better with nat |
04:31.11 | FuriousGeorge | considering my clients all use sip its starting not to make sense |
04:31.54 | habakuk | sip behind NAT is a pain. I actually prefer SIP to iax, but it is a pain with NAT |
04:32.43 | FuriousGeorge | as opposed to what im experiencing which is what? a minor discomfort. the only thing im sure of is that my sip clients, be it eyebeam, snom phones, sipura phones, or ANYTHING NOT USING IAX doesnt have this problem |
04:32.47 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
04:34.56 | orlock | Hmm.. |
04:35.00 | orlock | non * sip question - |
04:35.09 | orlock | I've got a cisco 7940 at home, running sip firmware |
04:35.21 | orlock | every so often it stops making/receiving calls |
04:35.23 | orlock | reboot doesnt fix it |
04:35.25 | habakuk | FuriousGeorge: yeah go with sip then. sorry without understanding what you are trying to do, its difficult to figure out what's causing this |
04:35.27 | orlock | but then it will work again |
04:36.13 | FuriousGeorge | habakuk: there is nothing really complicated going on. i have asterisk boxes and i want them to be able to communicate with eachother, so they have corresponding firends in iax.conf |
04:36.14 | habakuk | FuriousGeorge: btw did IAX reload work? I'm assuming you are using IAX registration |
04:36.31 | FuriousGeorge | i set up everything right, and it will work at random 75% of the time with no ryme or reason |
04:36.39 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
04:36.45 | FuriousGeorge | iaxreload helps a bit, but some peers never come back |
04:37.05 | orlock | and my * setup, i cant seem to register with upstream |
04:37.15 | habakuk | FuriousGeorge: heh.. not good. |
04:37.35 | FuriousGeorge | i guess ill turn off the web proxy on the gatways before i give up |
04:37.59 | FuriousGeorge | (or the peers come back a day later, but by then another 2 have dropped out for good) |
04:38.00 | habakuk | FuriousGeorge: any indication if it is the client side or the server side that causing the issue? |
04:38.42 | FuriousGeorge | both, because asterisk is both the client and the server. box a is both a client and a server to box b. |
04:38.54 | *** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-28-100.houston.res.rr.com) |
04:39.39 | *** part/#asterisk MarioGamboa (n=yo@201.133.229.135) |
04:39.47 | habakuk | of course... but the client side seems to have the issue right?.. i.e. not able to find other server |
04:40.33 | *** join/#asterisk litage (n=nick@203.220.55.70) |
04:41.11 | FuriousGeorge | well, no because since they are dynamic they have corresponding registers=> to eachother, but even though those dont come back after ip change, as long as the peer side doesnt screw up, they can still call between eachother |
04:41.16 | habakuk | FuriousGeorge: the reason why is I'm building an IAX client, and want to make sure I don't have these problems on the server side... i.e. if I design my client right, hopefully I won't have this problem |
04:41.43 | FuriousGeorge | i dont use any iax clients but i have a feeling they wouldnt have the same issue |
04:42.39 | habakuk | yeah they would if not designed right |
04:46.49 | FuriousGeorge | sure, and sip clients would probably do the same thing if not designed right, but they dont, which tells me there is something wrong with * |
04:47.32 | De_Mon | I'm dialing the exten => 50,1,Playback(agent-loginok) |
04:47.39 | habakuk | FuriousGeorge: yeah sounds like it |
04:47.45 | De_Mon | and all I hear on the phone is 'ent logged in' |
04:48.47 | De_Mon | I was just padding the playback with a wait(1), but it doesn't work in all cases... There's gotta be a more global solution |
04:49.36 | Z-Knight | anyone have a VIA based motherboard? If so, do you still have to do the PROC=i586 when compiling Asterisk 1.2.5? |
04:53.00 | willt | any reason why I should save voicemails in more then one format? |
04:53.19 | De_Mon | willt to avoid transcoding (not really) |
04:53.28 | *** join/#asterisk Eroick (n=chatzill@Ottawa-HSE-ppp269684.sympatico.ca) |
04:54.19 | Eroick | I dont understandhow other people can dial into my PBX from something like Gizmo if they are on a different computer on a different network. |
04:55.08 | Eroick | ~docs |
04:55.32 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
04:55.32 | willt | :s |
04:55.49 | *** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
04:56.37 | Eroick | what softphone is recomened? |
04:58.18 | Z-Knight | probbaly the xlite softphone |
04:58.29 | *** join/#asterisk jjg_ (i=jjg@dsl081-245-050.sfo1.dsl.speakeasy.net) |
04:58.34 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
04:58.41 | Octothorpe[away] | ~idefisk |
04:58.44 | jbot | from memory, idefisk is a great iax2 softphone for asterisk. See http://www.asteriskguru.com/tools/idefisk_beta.php |
04:58.51 | FuriousGeorge | heres another mystery: box1.dynu.com = 70.118.26.5 but even when i restart asterisk it looks to the old address. where the hell is asterisk cacheing this and how can i stop it |
05:00.03 | FuriousGeorge | -- Registered IAX2 to '70.118.26.5', who sees us as 72.68.119.22:1026 with no messages waiting |
05:00.09 | FuriousGeorge | that makes 0 sense to me |
05:00.21 | nvicf | I have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this?thanks |
05:00.27 | FuriousGeorge | so now of course the other box has no freakin idea |
05:00.29 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
05:01.10 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
05:06.16 | FuriousGeorge | can anyone tell me why, at random, one of my peers choses to use a port other than 4569? |
05:06.44 | SplasPood | hrm.. new 7960/40 firmware |
05:08.09 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
05:16.41 | orlock | HHmm |
05:16.42 | orlock | <PROTECTED> |
05:16.48 | orlock | What could cause that? i can ping it |
05:17.56 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
05:18.30 | orlock | Peer 'vic.nexvoice.net.au' is now REACHABLE! (6ms / 3000ms) |
05:18.31 | orlock | hmm |
05:18.50 | wasim | did your dns flake out? |
05:18.58 | orlock | nuh |
05:19.09 | wasim | or possible some netfilter inroute? |
05:19.12 | orlock | and theres basically just fibre between them and us |
05:19.14 | orlock | nope |
05:19.20 | wasim | hmm |
05:19.46 | wasim | maybe they got upset at the protean run chase ... |
05:19.49 | jjg_ | anybody ever get "port unreachable" ICMP errors from a pc running xlite in response to 407s from *? |
05:20.04 | FuriousGeorge | i was able to get all my peers working by setting host=boxname.dynuservice.com. the only problem is that now all the boxes complain all day that Peer "SoandSo" is not dynaimic |
05:20.12 | FuriousGeorge | and the registrations are being refued |
05:22.55 | willt | FuriousGeorge: Why can't you get a static? |
05:23.03 | *** join/#asterisk frk2 (n=frk2@202.5.145.13) |
05:23.23 | frk2 | hey guys! |
05:23.27 | frk2 | jbalcomb- you home? :) |
05:24.17 | FuriousGeorge | willt why should i have to. i set it up right to begin weith and i dont feel like spending the extra 800 dollars a year for 4 static ips |
05:24.42 | willt | who is you internet provider? |
05:24.52 | FuriousGeorge | verizon |
05:25.09 | FuriousGeorge | and it would come out to more like 1000 per year extra for 4 static ips |
05:25.10 | frk2 | why am i the ONLY one having issues with the GXP 2000 |
05:25.16 | willt | yikes |
05:25.25 | willt | what kind of service do you get from them? |
05:25.47 | FuriousGeorge | 1.5/768 mbps |
05:25.55 | frk2 | my phone hangs NIGHTLY :( |
05:25.58 | FuriousGeorge | what i might do i switch to cabelvision which gives semi-static ips |
05:26.17 | willt | how much is your service right now cost? |
05:26.28 | SwK_ | anyone noticed a memory leak lately? |
05:26.35 | FuriousGeorge | 50 per network |
05:26.42 | FuriousGeorge | static ip would be 70 |
05:26.51 | FuriousGeorge | difference of 20 X 4 X 12 mo = $960 |
05:26.53 | willt | oh you have more then one ip? |
05:27.04 | willt | i mean more then one connectino |
05:27.21 | FuriousGeorge | yeah, and they wont connect to eachother |
05:27.34 | wasim | SwK_: can't remember |
05:27.36 | FuriousGeorge | because of nat and dynamic ips. at least not consistantly |
05:27.51 | FuriousGeorge | well, we'll see now that i dropped the registers and changed host=url |
05:27.58 | SwK_ | trunk is leaking like a bitch on my sparc |
05:28.01 | willt | a lot of time they just block dsl -> dsl connections |
05:28.05 | FuriousGeorge | hopefully i will still need to restart but it will at least be reliable like it is right now |
05:28.16 | FuriousGeorge | unfortunately the CLI wont stop complaining about it |
05:28.26 | willt | just seems like a lot of money. is covad out your way? |
05:28.54 | SwK_ | 4.5minutes of CPU time 85000 calls and its using over 90meg of memory and not giving any of it back when the calls go away |
05:29.20 | FuriousGeorge | willtt yewah i think they are. dont they sell proprietary voip solutions? |
05:29.38 | willt | yeah but they sell dsl and it's pretty good |
05:29.47 | frk2 | anybody using grandstream's gxp 2000? |
05:29.52 | wasim | SwK_: not good |
05:29.58 | willt | maybe I can sell it to you at my cost :) |
05:30.28 | FuriousGeorge | willt: i aprreciate it, but its not my isp's fault |
05:30.42 | wasim | SwK_: transcoding? |
05:31.04 | SwK_ | wasim: i'm basically sending it 250calls/sec right now with sipp, calls get answered and wait() is called in the dialplan til sipp sends it a bye 1 sec later then the calls go away |
05:31.20 | FuriousGeorge | adn im sure a static ip from covad is gonna be more than a dynamic one from verizon |
05:31.22 | wasim | SwK_: we've seen it stablizie about 45M for 4 e1's, about 10k calls per hour |
05:31.36 | wasim | ofcoruse it crashes, but thats a different story |
05:31.38 | SwK_ | <PROTECTED> |
05:31.59 | SwK_ | 135K calls since restart |
05:32.26 | SwK_ | it just topped 100M on size and 99M on res |
05:32.59 | SwK_ | this box will run for a while like this tho (it has 8G or ram in it) |
05:33.15 | SwK_ | hmmm |
05:33.16 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
05:33.18 | SwK_ | soemthing aint right |
05:33.25 | willt | FuriousGeorge: example they retail TeleSoho starting at $59.95 comes with 1 or 4 statics can't remember |
05:33.28 | SwK_ | <PROTECTED> |
05:33.34 | wasim | eww |
05:33.36 | SwK_ | it just jumped to 160+ me |
05:37.17 | willt | SwK: What are you using to stress test that? |
05:38.12 | SwK_ | sipp |
05:38.23 | SwK_ | sipp.sf.net |
05:38.27 | willt | oh sorry im blind |
05:38.37 | Abydos313 | anyone here ever work with a 3com mbx 100? |
05:38.47 | SwK_ | is that anything like a nbx? |
05:38.52 | SwK_ | heh |
05:38.53 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
05:39.05 | Abydos313 | no idea, haven't actually seen the unit yet |
05:39.24 | SwK_ | 3com's IP-PBX? |
05:39.28 | Abydos313 | yes |
05:40.04 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
05:40.05 | Abydos313 | i have a buddy that needs to move it to another office. he was curious if that unit moves easily and doesn't lose config |
05:40.50 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
05:40.59 | Abydos313 | the manual doesn't specify any special shutdown or reboot procedures |
05:41.33 | SwK_ | i should keep its config i've never really used them |
05:41.47 | SwK_ | i've seen them but thats about it |
05:41.50 | Abydos313 | that is what i was thinking. don't see why not |
05:42.06 | SwK_ | how else would it survive a power failure |
05:42.19 | Abydos313 | that is exactly what i said |
05:42.20 | SwK_ | i would just mke sure that any configs have been written out on it |
05:43.15 | Abydos313 | i'll have to go thru it and checkout the dialplan and config. manual talks about backing up database so i take it the config is all held there |
05:43.17 | SwK_ | several phone systems work much like cisco routers, you can make changes on them, but they dont commit it to flash or non-volitile storage unless you "save translation" or "write mem" or whatever the command for that unit is |
05:43.21 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
05:43.34 | Abydos313 | ok, kewl so like cisco ios |
05:43.35 | SwK_ | omg its mikej |
05:43.41 | SwK_ | I dunno |
05:43.45 | Abydos313 | thx SwK_ |
05:43.47 | SwK_ | check the manual |
05:44.19 | Abydos313 | all web based config according to manual. but i'd guess there is shell access also. but i'll see |
05:44.36 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:55.08 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
05:58.21 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
06:12.08 | *** join/#asterisk jasonpr (n=jasonpr@c-24-10-236-54.hsd1.ut.comcast.net) |
06:13.48 | jasonpr | I head about an asterisk interface that an application can connect via a socket. Then the application can execute commands. Does anyone know that this app is? |
06:13.49 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
06:14.17 | brookshire | manager |
06:15.41 | jasonpr | Asterisk Manager. Does it allow you to do a MeetMe? |
06:16.02 | *** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116) |
06:16.08 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
06:21.19 | MikeJ[Laptop] | it's brooks!!!! |
06:21.49 | tarheelcoxn | I'm having trouble recording my voicemail greeting from the phone I've got. the volume level is loooooowwwwww. call volume levels are fine. Any ideas what might be going on? |
06:22.16 | tarheelcoxn | alternatively, where do I stick the audio file if I just want to record one and plop it in? |
06:28.48 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
06:34.15 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
06:34.22 | *** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116) |
06:41.08 | *** join/#asterisk ptblank (n=MURDER1@yorbalnd-cuda2-68-70-91-158.lmdaca.adelphia.net) |
06:45.07 | brookshire | hey! |
06:45.26 | brookshire | it's mike j |
06:45.30 | brookshire | :D |
06:49.21 | tarheelcoxn | okay so it's in /var/spool/asterisk/voicemail/default/700 ... I guess I'll go record replacement greet.gsm and unavail.gsm files... |
06:52.04 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
06:54.29 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
06:55.02 | RoyK_Stockholm | hej |
06:55.21 | mogorman | ? |
06:55.34 | jasonpr | I'm trying to set up my dial plan but the Background function drops the call imediatly after the file is done playing. Isn't there suposed to be a 10 timeout before it contiues? |
06:58.41 | tsume | manager sucks pretty bad |
06:58.47 | tsume | so does any implementation out there |
06:58.49 | X-Rob | jasonpr, set(TIMEOUT(whatever)=foo) |
06:59.00 | tsume | but what can you expect, they are made by peopel who don't now much |
06:59.08 | X-Rob | jasonpr, -- show function TIMEOUT |
06:59.53 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
07:05.26 | jasonpr | is there a good c/c++ api or other interface that I could use to connect to asterisk and listen for evens and execute commands? |
07:05.37 | mogorman | ~agi |
07:05.46 | jbot | i heard agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
07:05.46 | russellb | denied |
07:05.52 | russellb | ah. there it is |
07:06.18 | mogorman | heh looks like i win again russellb |
07:09.49 | *** join/#asterisk lorinc (n=ang@caracas-1029.adsl.interware.hu) |
07:12.08 | *** join/#asterisk Surye (n=Surye@ip68-8-242-220.sd.sd.cox.net) |
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07:12.44 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
07:13.16 | Surye | Hey, do you need a FXO card, or will a voice modem work for some/all functionality for the POTS line? |
07:13.27 | Qwell[laptop] | You need an FXO |
07:13.35 | Qwell[laptop] | rj11 port does not an FXO make |
07:17.39 | justnulling2 | how do i disable "SRV mapped to host" msg? |
07:20.30 | *** join/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p143.telkom-ipnet.co.za) |
07:20.51 | Supercross | good morning |
07:20.58 | Supercross | how is everybody doing? |
07:21.01 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
07:21.18 | mogorman | fantastic |
07:21.22 | tarheelcoxn | Supercross: confused |
07:21.31 | Supercross | lol |
07:21.33 | tarheelcoxn | but loving asterisk! |
07:21.36 | Supercross | join the club |
07:21.41 | Supercross | ya asterisk is great |
07:21.51 | Supercross | i am needing a bit of help with mysql and asterisk |
07:21.54 | tarheelcoxn | I wish I could figure out what's going on with the volume |
07:21.55 | Supercross | can anybody help? |
07:22.05 | Supercross | volume? |
07:22.15 | tzafrir | not before you give some details |
07:22.34 | Supercross | i am trying to setup my queue.conf into a sql table |
07:23.11 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
07:23.54 | Supercross | the doc's on the wiki say i must enter the string, realtime_family=family name |
07:24.07 | Supercross | but it doesnt work |
07:24.31 | Supercross | no queue's are loaded |
07:24.51 | Supercross | the family name refers to the name in extconfig file |
07:25.04 | *** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it) |
07:25.38 | tarheelcoxn | Supercross: when I record my greet and unavail from the phone, the volume is reeeaally low |
07:25.46 | tarheelcoxn | but the volume is fine when making calls |
07:26.00 | Supercross | ummm, that is weird |
07:26.09 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
07:26.13 | Supercross | what format are you saving them to? GSM? |
07:26.19 | tarheelcoxn | I've been playing with sox trying to just record my ow |
07:26.23 | tarheelcoxn | own* |
07:26.40 | tarheelcoxn | whatever asterisk is doing byy default... which... |
07:26.47 | tzafrir | tarheelcoxn, are you sure they are the right format? |
07:26.48 | tarheelcoxn | the directory has three files for each |
07:26.50 | Supercross | it will be gsm then |
07:27.04 | tarheelcoxn | greet.WAV greet.gsm greet.wav |
07:27.05 | tzafrir | what does 'file' give you for them? |
07:27.09 | Z-Knight | Can someone answer a couple of questions about the zaptel/ztdummy modules in linux and how do you add those at boot? |
07:27.26 | tzafrir | Z-Knight, what distro? |
07:27.32 | Z-Knight | centOs |
07:27.34 | Z-Knight | 4.2 |
07:27.44 | tarheelcoxn | tzafrir: file greet.WAV |
07:27.45 | tarheelcoxn | greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz |
07:27.47 | Z-Knight | I compiled zaptel ztdummy |
07:27.57 | Supercross | tarheelcoxn, strange it give you three |
07:27.58 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
07:28.04 | Z-Knight | but they don't load on boot |
07:28.11 | tarheelcoxn | file greet.gsm |
07:28.11 | tarheelcoxn | greet.gsm: data |
07:28.13 | Z-Knight | when I do lsmod | grep ztdummy I get nothing |
07:28.27 | Z-Knight | but when I check /etc/modprobe.conf they are listed there |
07:28.30 | tzafrir | Z-Knight, first-off, does 'modinfo zaptel' and 'modinfo ztdummy' show them? |
07:28.40 | tarheelcoxn | greet.wav |
07:28.40 | tarheelcoxn | greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
07:28.48 | tzafrir | modprobe.conf is irrelevant for autoloading |
07:29.02 | tzafrir | nither zaptel nor ztdummy need any post-install action |
07:29.19 | tzafrir | Feel free to delete the ztcfg lines from there |
07:29.24 | Z-Knight | but, should I see them when I do: lsmod | grep ztdummy? |
07:29.39 | tzafrir | Not to mention that kernel 2.6's tools won't look there, I believe |
07:29.51 | Z-Knight | I have Asterisk@ home.....and I have my separate version of asterisk on a different computer |
07:29.57 | tarheelcoxn | tzafrir: so it's using the .gsm version of each? |
07:30.00 | Z-Knight | the Asterisk@home has those modules listed |
07:30.27 | Z-Knight | I'm fairly new to the modules topic so I might be asking stupid questions |
07:30.50 | tzafrir | Z-Knight, the script /etc/init.d/zaptel looks for MODULES which can be defined in /etc/sysconfig/modules |
07:31.24 | Z-Knight | there is no /etc/sysconfig/modules |
07:31.28 | tzafrir | unrem in the latter file the line referring to ztdummy |
07:31.44 | tzafrir | or the quick&dirty: |
07:32.04 | tzafrir | echo MODULES=ztdummy >>/etc/sysconfig/modules |
07:32.39 | tzafrir | I'm not sure about "pure" centos. |
07:32.58 | Z-Knight | I need to read more about this |
07:33.30 | Z-Knight | I was simply following the basic installation of zaptel using 'make linux26' which incudes ztdummy |
07:33.31 | tzafrir | On Rapid and on latest Debians the zaptel init script does not load modules explicitly. However it checks for a timing source, and if there is none, it loads ztdummy |
07:33.47 | tzafrir | I believe that this approach is saner |
07:33.59 | Z-Knight | so it dynamically loads it |
07:33.59 | Z-Knight | ? |
07:34.47 | tzafrir | modules are always dynamically loaded... |
07:35.44 | Z-Knight | I guess I'm confused a bit, I'm reading the Asterisk book (yes for older version of *) but it mentions having to do a 'modprobe zaptel' and 'modprobe ztdummy' to load those modules |
07:35.59 | Z-Knight | afterwards you can load asterisk |
07:36.15 | tzafrir | 'modprobe ztdummy' will load zaptel with it. |
07:36.25 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
07:36.44 | tzafrir | because the module ztdummy depends on the module zaptel |
07:37.00 | Z-Knight | but you can't simply load: modprobe ztdummy |
07:37.06 | Z-Knight | it complains about not having zaptel |
07:37.12 | tzafrir | And the kernel modules loader has a bit more brains than Asterisk's modules loader |
07:37.13 | Z-Knight | so I have to do 'modprobe zaptel' ifrst |
07:37.21 | tzafrir | Z-Knight, you used insmod |
07:37.53 | Z-Knight | that i don't know....i'm unfamiliar |
07:37.55 | tzafrir | surely it did not complain explicitly about zaptel |
07:38.01 | Z-Knight | this is what I got: |
07:38.09 | Z-Knight | modprobe ztdummy |
07:38.09 | Z-Knight | Notice: Configuration file is /etc/zaptel.conf |
07:38.10 | Z-Knight | line 0: Unable to open master device '/dev/zap/ctl' |
07:38.10 | Z-Knight | 1 error(s) detected |
07:38.12 | Z-Knight | FATAL: Error running install command for ztdummy |
07:38.16 | *** join/#asterisk bails (n=bails@bailsyatton.plus.com) |
07:38.22 | *** join/#asterisk Lino` (n=Lino@i577BD710.versanet.de) |
07:38.34 | Z-Knight | once I did: modprobe zaptel, then I could do modprobe ztdummy without a problem |
07:38.39 | tzafrir | grep ztcfg /etc/modprobe.conf |
07:38.58 | tzafrir | grep ztcfg /etc/modprobe.conf | egrep 'zaptel|ztdummy' |
07:38.59 | Telamon | I've been having a problem with IAX for a while, and can't seem to figure it out. IAX calls (any IAX phone, I've tried 4 different models) that go to internal Asterisk functions (voicemail, sound recorder) work fine, but IAX calls that go to other phones (SIP or IAX) or out the zaptel interface get voice transmit problems (the sound cuts out for a half second every few seconds.) Incoming sound is fine. Any ideas? Digium has been pretty |
07:39.00 | Telamon | <PROTECTED> |
07:39.06 | Z-Knight | [root@localhost 2.6.9-22.EL]# grep ztcfg /etc/modprobe.conf |
07:39.06 | Z-Knight | install tor2 /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg |
07:39.06 | Z-Knight | install torisa /sbin/modprobe --ignore-install torisa && /sbin/ztcfg |
07:39.06 | Z-Knight | install wcusb /sbin/modprobe --ignore-install wcusb && /sbin/ztcfg |
07:39.06 | Z-Knight | install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg |
07:39.06 | Z-Knight | install wctdm /sbin/modprobe --ignore-install wctdm && /sbin/ztcfg |
07:39.08 | Z-Knight | install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp && /sbin/ztcfg |
07:39.10 | Z-Knight | install ztdynamic /sbin/modprobe --ignore-install ztdynamic && /sbin/ztcfg |
07:39.12 | tzafrir | ~pb |
07:39.18 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
07:39.18 | Z-Knight | install ztd-eth /sbin/modprobe --ignore-install ztd-eth && /sbin/ztcfg |
07:39.18 | Z-Knight | install wct1xxp /sbin/modprobe --ignore-install wct1xxp && /sbin/ztcfg |
07:39.19 | tzafrir | sorry, my mistake |
07:39.19 | Z-Knight | install wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg |
07:39.19 | Z-Knight | install wcte11xp /sbin/modprobe --ignore-install wcte11xp && /sbin/ztcfg |
07:39.20 | Z-Knight | install pciradio /sbin/modprobe --ignore-install pciradio && /sbin/ztcfg |
07:39.22 | Z-Knight | install ztd-loc /sbin/modprobe --ignore-install ztd-loc && /sbin/ztcfg |
07:39.24 | Z-Knight | install ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg |
07:39.25 | mogorman | ahhh |
07:39.26 | Z-Knight | so they are there |
07:39.28 | Z-Knight | ok |
07:39.32 | mogorman | pastebin my friend |
07:39.42 | Z-Knight | how do you use pastebin? |
07:40.12 | Telamon | Z-Knight: Go to pastebin.ca and copy in the text you want to paste, then just type in the URL they give you into the channel. |
07:40.14 | tzafrir | 'install ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg' : totally useless. ztdummy needs no config. |
07:40.21 | tzafrir | Anyway, ztdummy should be loaded |
07:40.59 | Z-Knight | but once I reboot, I won't have it show up when I do 'lsmod | grep ztdummy' .... is this a mistake or is this ok? |
07:41.19 | tzafrir | And anyway, this is not the job of modprobe to run ztdummy. You have a zaptel init.d script. This is where ztcfg should be run |
07:41.24 | tzafrir | bah |
07:41.42 | Z-Knight | ok |
07:41.50 | Z-Knight | I think I understand |
07:41.56 | Telamon | Z-Knight: Your Linux distro should have a file that lists all the modules to load at boot, you can just add it to that. |
07:42.17 | Z-Knight | Telamon: isn't that /etc/modprobe.conf? |
07:42.27 | tzafrir | In Debian it is /etc/modules . I'm not aware of any such simple thing on RH |
07:42.43 | tzafrir | Z-Knight, no. That file is configuration for the module loader |
07:42.52 | Telamon | No, modprobe.conf just helps modprobe figure out dependancies, it doesn't actually do any loading. |
07:43.00 | Z-Knight | ok |
07:43.03 | Z-Knight | hmm |
07:43.20 | Z-Knight | centos is basically redhat,...there is not /etc/modules or /etc/sysconfig/modules |
07:43.24 | Telamon | Z-Knight: IE, telling modprobe that when you load wctdm or ztdummy, it has to load the zaptel module first. |
07:44.40 | tzafrir | Z-Knight, so write your own zaptel init script . It should be run before the Asterisk one. See the skeleton init script in /etc/init.d/ |
07:45.11 | tzafrir | It should basically run 'modprobe ztdummy' on 'start' and do nothing otherwise |
07:45.30 | tzafrir | You may need to put some content in it later on. |
07:45.31 | bails | centos /etc/modules.conf |
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07:46.08 | Z-Knight | shouldn't it add one when I do 'make install' on zaptel source? |
07:46.30 | tzafrir | /etc/modules.conf is the old (kernel <=2.4) modutils config file. It can't be CentOS-specific |
07:46.50 | Z-Knight | I have kernel 2.6 with centos4.2 |
07:47.03 | tzafrir | init scripts are quite distro-specific |
07:47.34 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
07:47.42 | Z-Knight | I figured though I would not have to go to trouble of making my own init scripts....I'd think that zaptel would have that configured for me |
07:48.15 | *** join/#asterisk htims (n=htims@Vf3fb.v.pppool.de) |
07:48.21 | tzafrir | Z-Knight, I think that an init script is the way to go. hmm, time to file a bug report |
07:48.22 | tzafrir | ? |
07:48.49 | Z-Knight | it might be too early...maybe I'm doing something wrong |
07:48.55 | X-Rob | Z-Knight, the zaptel.init works fine with centos |
07:49.01 | Z-Knight | did you do your own installation of * or did you do *@home? |
07:49.14 | X-Rob | make sure you create /etc/sysconfig/zaptel and put 'TELEPHONY=yes' and 'MODULES=wctdm' or ztdummy or whateveryou want to load |
07:49.38 | tsume | yuck, the distro which rips rhel and comes with uber late security patches |
07:49.40 | tsume | great |
07:49.48 | tsume | one should be using debian for an asterisk box |
07:49.57 | tzafrir | uber-late? |
07:50.09 | tsume | tzafrir: they were 3 weeks late with a sshd exploit update |
07:50.16 | tzafrir | not that you shouldn't use Debian |
07:50.25 | Z-Knight | X-Rob...are there any specific instructions out there for your own installation of *? I was using one of asteriskguru and it made no mention of making additional scripts/etc....neither does the * book |
07:50.26 | tsume | debian is what most use |
07:50.32 | tsume | also the livecds |
07:50.37 | Z-Knight | Here is the general installation I followed: http://www.asteriskguru.com/tutorials/general_asterisk_installation_compilation.html |
07:50.42 | tsume | its jsut much easier to set up :) |
07:51.06 | tsume | but I guess some people are GUI pussies :) |
07:51.29 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
07:51.54 | tzafrir | what GUI would you recommend that generates debugable dialplan? |
07:52.01 | bails | maybe its because some people dont have all day to sort out apt-get/dpkg nastiness, and i use debian so i'm not slagging it off |
07:52.34 | tzafrir | bails, apt-get install wajig to get a nicer command-line interface |
07:52.47 | bails | an e1 gurus about? |
07:52.57 | bails | any^ |
07:53.13 | tzafrir | wajig search <package>; wajig get <package>; etc |
07:53.37 | bails | I have problems with a te110p |
07:53.52 | tzafrir | or use aptitude/synaptic |
07:53.52 | bails | looks like layer 1 |
07:54.10 | X-Rob | tzafrir, if you use a gui that generates buggy dialplan, then tell the gui maintainers |
07:54.11 | bails | lots of crc4 errors and sync errors |
07:54.24 | X-Rob | AMP/FreePBX doesn't have any wierdism at the moment, we think 8) |
07:54.49 | tzafrir | X-Rob, but what if you put a wrong username or hostname to the trunk? |
07:54.58 | russellb | X-Rob: so what's the story behind renaming AMP? |
07:55.16 | tsume | bails: there is no nastiness. there is only incompetance |
07:55.17 | tzafrir | X-Rob, have you seen the definitions of SIP/IAX trunks in AMP? |
07:55.45 | X-Rob | russellb, fucked if I know, not my call. But I think the idea was to get away from 'Asterisk' management portal, when they want it to be a genereic manager for opbx,freeswitch,whatever you want to plug into it |
07:55.47 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
07:55.51 | tzafrir | You basically need to put there the peer entry, thje user entry and the register line |
07:55.56 | tzafrir | as plain text |
07:56.04 | russellb | X-Rob: i see. I was just curious if you knew. |
07:56.05 | tsume | web based configs are a fkin joke |
07:56.15 | russellb | I personally hate the new name, because obviously, AMP itself is *not* a PBX at all. |
07:56.17 | X-Rob | tzafrir, I'm a FreePBX developer. I'm _aware_ that it sucks, and 'real soon now' there'll be a provider wizard. |
07:56.20 | brookshire | tsume: they have their place :) |
07:56.22 | tsume | IE, Moz keep changing how JS, DOM, other stuff works. |
07:56.43 | tsume | makes it complete hell to actually make a maintainable project, especially with DnD |
07:56.48 | X-Rob | russellb, well, yes, I kinda agree, but it had nothing to do with me. I'm just a grunt developer in .au, not living the high life in canada 8) |
07:57.09 | russellb | X-Rob: ha, it's all good. I still think it's a great effort. I just don't like the name. :) |
07:57.16 | tzafrir | X-Rob, so you have to be able to debug errors yourself. And AMP generates such a verbose output that it is practically impossible to figure out |
07:57.25 | russellb | anyway, back to work ... |
07:57.31 | tzafrir | It is also quite good at masking out the real reason for the problem |
07:57.43 | brookshire | you guys should have just kept the aberviation 'AMP' |
07:57.48 | tsume | heh, bails is very incompetant :) |
07:57.50 | brookshire | and dropped the name ;) |
07:57.57 | X-Rob | 'A management portal'? 8) |
07:58.05 | russellb | sure, why not |
07:58.06 | tsume | (CR)AMP just isn't very good |
07:58.16 | tsume | it really gives me the (CR)AMPs |
07:58.17 | brookshire | SAMP! |
07:58.31 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
07:58.54 | tsume | need a decent client GUI based application |
07:58.56 | tsume | not java |
07:59.08 | tsume | not proprietary platform code |
07:59.09 | tsume | maybe something mono based ;) |
07:59.12 | tsume | and GTK# |
07:59.18 | X-Rob | So your issue with it, tzafrir, is that it's hard to debug username/password issues in AMP, and I agree, that can be a bastard. |
07:59.20 | brookshire | mono is just as bad as java |
07:59.30 | tzafrir | X-Rob, a "wizard" is not what I'm after. AMP clearly lacks decent data structures that could be easily manipulated from scripts |
07:59.31 | tsume | brookshire: in memory, no its not |
07:59.34 | X-Rob | but when I tidy up trunks, then you'll put a username, password, and pick a provider from a drop-down |
07:59.41 | brookshire | java + swt = mono + gtk# |
07:59.46 | brookshire | lol |
07:59.48 | tsume | brookshire: its not such a fkin hog like java |
07:59.49 | X-Rob | tzafrir, it uses mysql. Do whatever the smeg you want to it. |
07:59.55 | tsume | and it uses gtk# by default |
07:59.58 | tsume | winforms coming soon |
08:00.01 | tsume | native on each platform |
08:00.03 | X-Rob | it's all stored in a database, and the database is scraped to generate the dialplan |
08:00.08 | tsume | making it better than java for one |
08:00.10 | brookshire | winforms have been coming soon since 2000 |
08:00.14 | tsume | also C# being more OO based |
08:00.23 | tzafrir | If you have a wizard, then any for any change to it, you'll have to go through the wizard again (think of outlook and of cups's web interface) |
08:00.24 | tsume | brookshire: no, winforms for mono started in 2002 |
08:00.38 | tsume | brookshire: peopel have really been working on gtk# more, making it stable |
08:00.54 | X-Rob | tzafrir, no, you use the wizard to generate the entry. Then if you want to change anything, you go into the 'edit trunk' and all the settings are laid out in front of you. |
08:01.02 | tsume | brookshire: also theres a small fact... Novell/mono project focuses more on the paying customers, mainly SuSE or anyone who pays |
08:01.25 | X-Rob | including a 'I know what the hell I'm doing' box which lets you edit the trunk definition directly (which is what you've got now) |
08:01.37 | brookshire | opengl + gtk+ is the only way to code ;) |
08:01.48 | tsume | not really |
08:01.52 | tsume | GTK# is just fine |
08:02.02 | brookshire | no... gtk# is lame |
08:02.03 | brookshire | :D |
08:02.09 | tsume | that is such a troll :) |
08:02.10 | X-Rob | no, gtk( is better! |
08:02.13 | brookshire | it's a distraction |
08:02.23 | X-Rob | and gtk% is like l33t. |
08:02.26 | tsume | I should make a dialplan gui and sell it |
08:02.29 | tsume | closed source |
08:02.34 | brookshire | x-rob: it's 1338 |
08:02.36 | tsume | or maybe shareware :) |
08:02.40 | X-Rob | 1338? Fwor. |
08:02.43 | tzafrir | X-Rob, again, how do I add a phone from the command-line (or a script)? |
08:03.06 | X-Rob | tzafrir, you populate the mysql database and then call php's config generate. |
08:03.07 | brookshire | 1337+1 |
08:03.14 | tsume | eww! mysql |
08:03.21 | X-Rob | if you _Want_ to do it from the command line, feel free to write some code and I'll stick it in AMP |
08:03.23 | tarheelcoxn | ouch. that joke is painful |
08:03.26 | *** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no) |
08:03.29 | X-Rob | but I don't think it's what average users want |
08:03.49 | X-Rob | I think the issue is, AMP is _not_ for people that know what they're doing. It's for beginners, and it's deliberately designed to hide the complexities from the end user. |
08:04.07 | X-Rob | if they're clueful enough to want to add 100 xtns, then they should be doing it themselves, rather than using AMP |
08:04.15 | iDunno | and annoy the hell out of everyone else ;) |
08:04.26 | tzafrir | X-Rob, The problem is that too many clueless people use it for production |
08:04.39 | brookshire | x-rob: i know someone who has 1,000 lines with amp |
08:04.58 | X-Rob | tzafrir, brookshire, what can I do to make it easier? |
08:05.02 | X-Rob | I really have no idea. |
08:05.31 | X-Rob | We're about to release 2.0, which is all new and funky, but the dialplan it generates is basically the same. |
08:05.47 | X-Rob | I'm ready and willing to take suggestions of ways to make it 'better' |
08:05.51 | brookshire | x-rob: i dunno.. i've never used it.. |
08:06.12 | brookshire | but, i would definately talk with matt o'gorman |
08:06.17 | mogorman | ? |
08:06.31 | brookshire | hah |
08:06.42 | X-Rob | Why am I talking to you, mogorman? |
08:06.47 | mogorman | i dont know |
08:06.56 | X-Rob | brookshire? |
08:07.11 | brookshire | tell x-rob how he can make amp better :) |
08:07.36 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:07.45 | russellb | mogorman knows all |
08:07.53 | mogorman | that is true russellb |
08:08.01 | mogorman | its just getting it out of my tiny brain |
08:08.03 | mogorman | is the trick |
08:08.15 | X-Rob | crowbar, stat! |
08:09.15 | tzafrir | X-Rob, but what I mentioned is that it doesn't do a good job at hiding the complexity: when it breaks: it breaks into pieces. This means error reporting is bad. |
08:09.28 | tzafrir | The error handling of the dial macro is simply ohrrible |
08:09.40 | tzafrir | Not to mention that it is overly complex |
08:09.42 | X-Rob | tzafrir, yes, but a lot of that has been fixed in 2.0 |
08:09.52 | *** join/#asterisk RoyK (n=roy@static-213-115-44-227.sme.bredbandsbolaget.se) |
08:10.02 | X-Rob | overly complex? It _does_ do a shitload of stuff |
08:10.15 | tzafrir | Which most users don't need anyway |
08:11.12 | tzafrir | Most of the applications in the dialplan can't be changed from the GUI |
08:11.32 | X-Rob | call forwarding, call waiting, voicemail no answer, voicemail busy, override caller ID |
08:11.34 | tzafrir | Compare that to the nice "applications" menu in destar |
08:11.47 | X-Rob | tzafrir, yes, and that'll be fixed in 2.1, I totally agree that that is arse. |
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08:12.36 | X-Rob | all applications will be pluggable modules, and you'll be able to disable 'em if you don't use 'em to clean up your dialplan |
08:14.28 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
08:14.34 | X-Rob | so yes, your two complaints so far I think have been addressed? Next? 8) |
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08:16.48 | X-Rob | On the subject of applications, we're also setting up a localisation menu, where you pick 'UK' and it sets up call waiting, forward, etc, numbers as per the standard in that country |
08:16.49 | buu | X-Rob: It's not chartreuse |
08:17.00 | X-Rob | Aaah, damn good point, buu |
08:17.21 | buu | Thanks! |
08:17.23 | X-Rob | I'm afraid that I'll have to take that as an insult, and now argue with you for three hours |
08:17.35 | X-Rob | s/I'm/But I'm/ |
08:17.37 | buu | Excellent. |
08:18.18 | buu | hrm |
08:19.16 | buu | s/hrm/$x/ |
08:19.16 | buu | No? |
08:19.16 | X-Rob | nah |
08:19.16 | X-Rob | it's quite limited |
08:19.16 | X-Rob | s/quite/very/ |
08:19.16 | X-Rob | you can't do backquotes |
08:19.16 | buu | And it seems to work randomly |
08:19.16 | X-Rob | yeah |
08:19.16 | X-Rob | or you just broke it |
08:19.21 | X-Rob | ooh, there we go |
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08:22.08 | RoyKa | hrmf |
08:22.09 | X-Rob | s/(there) (we)/$2 $1/ |
08:22.13 | X-Rob | ooh, there we go |
08:22.14 | X-Rob | s/(there) (we)/$2 $1/ |
08:22.23 | X-Rob | don't be gay, jbot. |
08:22.27 | X-Rob | s/jbot/sparky/ |
08:22.45 | X-Rob | oook |
08:22.55 | X-Rob | s/o+/o/ |
08:23.06 | X-Rob | very limited. |
08:23.09 | RoyKa | :%s/X-Rob//gi |
08:23.18 | X-Rob | RoyKa, heh |
08:23.23 | RoyKa | :) |
08:23.50 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
08:23.59 | Zeeek | hey now |
08:24.04 | RoyK | Zeeek: wot? |
08:24.20 | Zeeek | hej! |
08:24.24 | RoyK | hej hejj |
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08:24.52 | Zeeek | I thought you were Danish or something |
08:25.08 | RoyK | I'm norwegian |
08:25.13 | RoyK | but I'm in a meeting |
08:25.21 | RoyK | and staying over in stockholm for the weekend |
08:25.23 | Zeeek | oh yeah that's right. Same thing |
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08:27.54 | X-Rob_ | Grr. |
08:27.58 | X-Rob_ | or have you got a couple of seconds to spare for a norwegian who doesn't speak english all that well? |
08:28.04 | X-Rob_ | (to RoyK) |
08:28.17 | Zeeek | oh god, the ghetto begins here |
08:29.26 | RoyK | X-Rob_: are there any norwegians that doesn't speak english? |
08:29.35 | X-Rob_ | Actually, he seems to have got better now |
08:29.43 | X-Rob_ | I think he may have just been rusty 8) |
08:29.53 | RoyK | X-Rob_: who is this? |
08:30.02 | X-Rob_ | powerchip on #freepbx |
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08:44.15 | Zeeek | bleh |
08:48.18 | Supercross | hi everybody |
08:48.35 | Supercross | i am needing some help with queues.conf and mysql |
08:48.37 | shiznatix | hi doctor nix |
08:48.40 | shiznatix | hi doctor nic |
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08:49.00 | Zeeek | ask your question, please |
08:49.07 | Zeeek | and drop a quarter in the slot |
08:49.14 | Supercross | lol |
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08:49.34 | Supercross | what line would you put in the queues.conf file to tell it to use the mysql table? |
08:49.45 | Supercross | in the extentions u use the switch => |
08:49.52 | Supercross | what do you use in the queues? |
08:52.10 | Supercross | anybody? |
08:52.35 | X-Rob_ | I'd check /usr/src/asterisk/configs/queues.conf.sample for documentation |
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08:53.29 | Supercross | ok will check |
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09:10.23 | kippi | Hi |
09:11.33 | kippi | Trying to logon a agent, I have added it to queues.conf and agents.conf, when I ring to add the agent it asks for logon, that works fine, then it is asking me for a new extension, what is this? |
09:16.41 | Zeeek | did you reload queues and agents or restart asterisk? |
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09:26.01 | shiznatix | Hello, I am having difficulty with sending faxes on asterisk 1.2.4. Here http://pastebin.com/605040 is my extensions.conf file and the callfile. If anyone can help that would be fantastic |
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09:31.11 | konfuzed | slePP, eh are you alive at this hour |
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09:43.18 | Winkie | asterisk billing support sucks |
09:46.06 | Zeeek | so many things in this universe suck worse |
09:47.15 | frk2 | yup |
09:47.20 | frk2 | like chinese voip phones |
09:47.25 | frk2 | they suck big time |
09:47.36 | Zeeek | I have three, they're not that bad |
09:47.45 | frk2 | wait till you put a 100 of them |
09:47.48 | frk2 | i have 5 too, not bad |
09:47.52 | frk2 | deployed 75 at a client |
09:47.57 | frk2 | and random things start happening |
09:47.59 | Zeeek | why would I want 100 chinese phones? |
09:48.09 | Zeeek | your client is too cheap |
09:48.14 | frk2 | hahahah |
09:48.16 | frk2 | indeed |
09:48.18 | frk2 | have told them so |
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09:50.18 | fourcheeze | frk2: which models are they? |
09:50.50 | *** join/#asterisk snip3r (n=sniper@195.246.199.136) |
09:50.53 | frk2 | atcom at-323 |
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09:53.36 | snip3r | hi all |
09:54.08 | snip3r | anyone having a couple of minutes helping me out with an asterisk/sip/nat issue? |
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10:11.51 | kippi | Zeeek: yea I did a reload on asterisk |
10:15.48 | [ProB]CrazyMan | hello, anybody had some issue like this with spandsp ? http://www.roterschnee.org/failure.tif |
10:16.04 | [ProB]CrazyMan | that faxes doesn't look ok |
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10:18.19 | Zeeek | snip3r just ask and ye shall receive |
10:20.59 | snip3r | Zeeek: thx :) |
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10:22.41 | snip3r | so, I have a simple config: asterisk on a public IP, clients behind NAT, and there's no voice between any client and/or any other endpoint |
10:22.52 | snip3r | protocol is SIP |
10:23.12 | snip3r | nat=yes & canreinvite=no is set in sip.conf |
10:23.36 | snip3r | for all clients and my outbound proxy |
10:23.56 | shiznatix | does anyone have any expierence with sending / recieving faxes with asterisk 1.2.4? |
10:24.23 | snip3r | however, I'm still getting an "attempting native bridge of SIP/blah" message in the CLI |
10:24.58 | frk2 | sniper thats fine... it will never actually bridge |
10:25.20 | frk2 | unless you have no tTowW options in your dial |
10:25.29 | snip3r | frk2: wow. that's good news. |
10:25.34 | snip3r | frk2: nope |
10:25.49 | snip3r | I have a 4-line dialplan |
10:25.56 | [ProB]CrazyMan | shiznatix: my spandsp fax doenst work very well with 1.2.4 |
10:26.01 | frk2 | meaning if your dial is like dial(SIP/100,,) then bridging may happen |
10:26.26 | *** part/#asterisk toot (i=chris@212.20.250.186) |
10:26.56 | snip3r | my dialplan looks like "exten => _06.,Dial,SIP/${EXTEN}@outb" |
10:27.02 | Skid | what about 1.2.5 ? |
10:27.11 | snip3r | where "outb" is my outbound link |
10:27.14 | shiznatix | [ProB]CrazyMan: have you found a solution to this? Also, where do you get a problem? Maybe its my configuation, can you check out what I have at http://pastebin.com/605040 |
10:27.20 | frk2 | it might bridge... unless you have canreinvite=no |
10:27.25 | snip3r | I have |
10:27.41 | frk2 | if it bridges you'll get another message saying so |
10:27.44 | frk2 | so no worries |
10:27.46 | snip3r | both in PSTN gateway config and client config |
10:28.08 | snip3r | then, why I'm not getting any voice? |
10:28.21 | kmilitzer | Hi! Can anybody tell me what I need to test the t.38 capabilities of spandsp and rxfax? If i read it right, the latest version of spandsp and rxfax/txfax can act as t.38 gateway, but I could not find out if I need the svn branch of asterisk or if 1.2.4 works and where exactley to get the latest spandsp 0.3 and the t38-bits.tgz file .... |
10:28.29 | frk2 | dont know |
10:28.31 | frk2 | whats the sip client? |
10:28.35 | snip3r | the most interesting thing is that it worked |
10:28.42 | snip3r | ^^ a moment |
10:28.49 | [ProB]CrazyMan | shiznatix: The faxes I receive doesn't look very well .. |
10:29.00 | snip3r | a Sipura h/w phone and an X-Lite |
10:29.08 | [ProB]CrazyMan | shiznatix: aswell I get connection interruptions |
10:29.14 | snip3r | so, it worked in a previous installation |
10:29.26 | snip3r | with almost no configuration changes at all |
10:29.27 | Zeeek | X-Lite - make sure transmit silence is ON |
10:29.31 | shiznatix | [ProB]CrazyMan: that is not a problem, i just need it to send *anything* even if its horrible quality. |
10:29.34 | snip3r | yep, it's on :)_ |
10:29.47 | snip3r | i've read a couple of articles on this issue |
10:30.02 | Zeeek | sniff it to see what ip address is being sent to asterisk |
10:30.11 | shiznatix | [ProB]CrazyMan: I paste the callfile in the asterisk outgoing folder and it starts the fax but when it tries to write to /tmp/ffax/ it juts hangs and times out |
10:30.12 | snip3r | but thought I'm missing something trivial |
10:30.30 | snip3r | Zeeek: a minute |
10:30.35 | tv_manojkumar | Hi All, I need some help on asterisk realtime setup |
10:32.36 | [ProB]CrazyMan | shiznatix: you have to put somehere a number to dial an destination |
10:32.49 | snip3r | Zeeek: To: <sip:2004@192.168.1.102:5060> |
10:33.02 | snip3r | I'll resolve this and come back again |
10:33.03 | snip3r | :) |
10:33.17 | [ProB]CrazyMan | shiznatix: you mixed txfax and rxfax |
10:33.18 | snip3r | thanks for pointing out the obvious |
10:33.30 | shiznatix | [ProB]CrazyMan: huh ok lemme try it |
10:33.36 | [ProB]CrazyMan | shiznatix: txfax => sending, rxfax -> receiving |
10:33.47 | [ProB]CrazyMan | shiznatix: do you use zap ? |
10:34.11 | shiznatix | i have to get it to work with sip first them zap, bosses orders |
10:34.17 | Zeeek | snip3r but the client is on the same LAN as * or not? |
10:34.25 | tv_manojkumar | Hi CrazyMan |
10:34.39 | [ProB]CrazyMan | shiznatix: do in the callfile sth. like Channel: zap/g1/numbertodial |
10:35.55 | snip3r | Zeeek: * is on a public IP, clients behind NAT on a different public IP |
10:36.05 | shiznatix | [ProB]CrazyMan: Alright, ill try some of this stuf |
10:36.51 | snip3r | x.x.x.100 is asterisk and x.x.x.136 is my nat box |
10:37.18 | snip3r | there're 2 clients behind the nat box, the Sipura phone and an X-Lite |
10:37.23 | Zeeek | well sending the local ip address won't get the audio properly |
10:37.28 | snip3r | sure |
10:37.38 | Zeeek | you could try STUN |
10:38.15 | snip3r | X-Lite uses STUN by default, isn't it |
10:38.16 | snip3r | ? |
10:38.31 | frk2 | but setting nat=yes fixes that |
10:38.48 | frk2 | why dont you just try a echo test first? |
10:38.53 | snip3r | nat=yes is set in all clients' config |
10:39.31 | frk2 | hmmm |
10:39.33 | snip3r | I'm getting no audio and I've tried all sample extensions :( |
10:39.38 | frk2 | maybe nat box is dropping packets? |
10:39.46 | frk2 | does the call connect? |
10:39.51 | snip3r | sure |
10:40.09 | frk2 | is the nat box also a firewall? |
10:40.20 | snip3r | Zeeek pointed out that the phone sent its private IP |
10:40.31 | snip3r | it isn't |
10:40.32 | shiznatix | [ProB]CrazyMan: I get the error 'No channel type registered for zap' |
10:40.39 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
10:40.49 | Zeeek | huh? |
10:40.51 | frk2 | the phone will send its private ip |
10:40.57 | frk2 | asterisk will ignore it if you set nat=yes |
10:41.01 | snip3r | IC |
10:41.01 | [ProB]CrazyMan | shiznatix: how do you make externeal calls ? |
10:41.06 | Zeeek | 192.168..... is public ip? nah! |
10:41.10 | snip3r | nope |
10:41.24 | snip3r | 195.x.x.100 is * |
10:41.32 | snip3r | 195.x.x.136 is the NAT box |
10:41.36 | Zeeek | anyway it's logical tohave nat=yes for a client behind NAT |
10:41.47 | snip3r | yep |
10:41.53 | shiznatix | [ProB]CrazyMan: I don't call outside of the asterisk network |
10:42.18 | shiznatix | [ProB]CrazyMan: And I don't have a fax machine or a phone I am actually trying to dial, I just want it to take the fax in then save it to file |
10:42.28 | snip3r | X-Lite also discovers the type of NAT |
10:42.40 | snip3r | and identifies it as a port restricted cone |
10:42.52 | Zeeek | actually, X-Lite worked for me behind nat right away on FWD |
10:42.57 | snip3r | IC |
10:43.11 | snip3r | as I said, it worked for me too |
10:43.18 | [ProB]CrazyMan | shiznatix: ? how do you want to receive an fax? if you try to let asterisk send to itself an fax (same box) that will not work |
10:43.27 | snip3r | but I removed * to install SER |
10:43.37 | Zeeek | blasphemy! |
10:43.45 | snip3r | 'cause I needed DB auth |
10:43.50 | snip3r | sry :) |
10:44.09 | snip3r | now I realized that I need 'em both |
10:44.19 | shiznatix | [ProB]CrazyMan: Why not? I give asterisk the callfile which says 'start sending the fax' then i want it to just save the fax to a folder on the asterisk server. |
10:44.20 | Zeeek | two wives are better than one |
10:44.27 | snip3r | absolutely true |
10:44.55 | Zeeek | shiznatix what is the original source of the fax? |
10:45.15 | shiznatix | Zeeek: Just a random tiff file |
10:45.32 | frk2 | dude that is blasphemy. no more asterisk help for you |
10:45.33 | frk2 | lol |
10:45.37 | [ProB]CrazyMan | shiznatix: spandsp could not handle that faxes .. (I dont know it, maybe because there is then no timing or so ..) |
10:45.51 | snip3r | :) |
10:46.43 | Aze` | Anyone use misdn with avm ? |
10:47.32 | shiznatix | [ProB]CrazyMan: What do I do then? I just need to be able to send a crappy fax to asterisk with a call file and have asterisk save it to disk. Any way to get that done is a great way |
10:49.40 | Zeeek | why not just email the TIFF? |
10:51.01 | tv_manojkumar | Hi People, I need some help on Asterisk RealTime setup |
10:55.34 | Fiskfan | How many of you here are working daily with asterisk, as a technichan with installation and maintence? |
10:55.46 | wasim | me, me and me |
10:55.55 | astra^^ | me 2 |
10:55.56 | Fiskfan | wasim... schizo? :) |
10:55.58 | *** join/#asterisk juanjoc (n=juanjoc@222-32-235-201.fibertel.com.ar) |
10:55.59 | tv_manojkumar | me 2 |
10:56.04 | wasim | Fiskfan: you have no idea ... |
10:56.13 | Fiskfan | :) |
10:56.27 | kmilitzer | What do I need to test t.38 gateway capabilities of spandsp? |
10:56.51 | tv_manojkumar | Hi People, I need some help on Asterisk RealTime setup |
10:57.06 | astra^^ | wht is it |
10:57.22 | Zeeek | yo wasim |
10:57.47 | wasim | bonjour monsieur zeek, comment ca va? |
10:57.52 | Fiskfan | ah lunch... see you later... |
10:58.23 | tv_manojkumar | astra: Whenevr I want to add a new context I should add it to extensions.conf, can I avoid this and directly add to Mysql |
10:58.49 | tv_manojkumar | If I do not add in Ext.conf then that context is not recognozed by asterisk |
10:59.11 | Zeeek | ça va bien, mais j'ai un peu faim là |
10:59.22 | tv_manojkumar | any solution for this |
11:00.22 | tv_manojkumar | please help me |
11:00.38 | *** join/#asterisk Whisk (n=a@194.130.117.202) |
11:04.48 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
11:06.05 | tv_manojkumar | Hi Whisk,zotz : can u help me on asterisk realtime |
11:07.51 | zotz | i would not be the best, have not done much asterisk lately |
11:07.58 | zotz | what is your question though |
11:08.19 | tv_manojkumar | Whenevr I want to add a new context I should add it to extensions.conf, can I avoid this and directly add to Mysql |
11:08.34 | tv_manojkumar | If I do not add in Ext.conf then that context is not recognozed by asterisk |
11:08.57 | tv_manojkumar | zotz: this is my question |
11:11.35 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
11:12.05 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
11:15.35 | *** join/#asterisk nela (n=nela@xdsl-213-196-230-59.netcologne.de) |
11:15.51 | nela | hi |
11:16.55 | tv_manojkumar | hi nela |
11:17.23 | *** part/#asterisk nela (n=nela@xdsl-213-196-230-59.netcologne.de) |
11:17.55 | *** part/#asterisk tv_manojkumar (n=Administ@59.93.56.163) |
11:20.28 | wasim | i guess they hit it off well ... |
11:23.21 | zotz | so tv left with no answer |
11:23.59 | snip3r | snip3r is back :) |
11:24.19 | snip3r | could anyone take a look at the log http://cozmo.hu/log.txt ? |
11:24.52 | *** join/#asterisk RoyK_Stockholm (n=roy@static-213-115-44-227.sme.bredbandsbolaget.se) |
11:25.16 | snip3r | that's a full call log with no rtp traffic in neither direction |
11:27.09 | snip3r | Zeeek, frk2: could you check this out? |
11:27.56 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
11:29.49 | Zeeek | weird |
11:30.35 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
11:30.37 | fenlander | hmm. so much for astricon earlybird registration starting yesterday... |
11:31.15 | snip3r | the PSTN g/w is working well and as I mentioned, the same setup worked in an earlier installation |
11:32.14 | snip3r | I gotta go but if you can comment on my problem, please write and I'll check it out soon |
11:33.36 | chris_ast | Please help me configure asterisk dialplan |
11:41.04 | *** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116) |
11:41.37 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
11:42.25 | astra^^ | if i recieve a call frm a sip server to * . what are the setting to be at the sip server end to route the cal to my * server |
11:42.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
11:43.22 | chris_ast | astra: what is your sip server, ser? |
11:43.39 | astra^^ | free BBd |
11:43.46 | astra^^ | bsd |
11:44.01 | astra^^ | sorry free BSD |
11:44.15 | kippi | Trying to logon a agent, I have added it to queues.conf and agents.conf, when I ring to add the agent it asks for logon, that works fine, then it is asking me for a new extension, what is this? |
11:45.44 | chris_ast | astra: what is the name of your sip server, free BSD is OS of your machine |
11:47.06 | astra^^ | cris_ast:porta one |
11:47.37 | *** join/#asterisk oracle^ (n=cam@unaffiliated/cameleons) |
11:48.36 | chris_ast | I can tell how we can do it for ser to asterisk |
11:49.24 | Zeeek | kippi I had that happen the other day. I think it means you are logging in it wants to know what your call back is |
11:49.24 | chris_ast | for portaone I have no idea |
11:50.08 | chris_ast | Hi People, I need help configuring my DialPlan |
11:51.21 | Zeeek | spit it out |
11:52.14 | frk2 | damn my grandstream locked up again |
11:52.24 | astra^^ | chris :yes ser to asterisk |
11:52.33 | astra^^ | i have a dial peer in asterisk |
11:54.40 | astra^^ | chris_ast: PORTA ONE was my billin .s/w s |
11:54.47 | chris_ast | You just need to rewritehost name and port |
11:55.18 | astra^^ | name to..? |
11:55.56 | astra^^ | name to peer name of the asterisk..? |
11:56.16 | chris_ast | check http://www.voip-info.org/wiki-Asterisk+at+large |
11:56.41 | chris_ast | see inder Asterisk and SER on same box |
11:56.46 | chris_ast | under |
11:57.11 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
11:57.21 | chris_ast | Hi Zeek, here is my problem |
11:57.30 | *** join/#asterisk dyer (i=dyer@unaffiliated/dyer) |
11:57.39 | Zeeek | we're all ears! |
11:58.58 | frk2 | somebody PLEASE help me with my grandstream :( |
11:59.26 | frk2 | does anybody have a grandstream with asterisk here? |
11:59.34 | Zeeek | I have one in a box, it just stopped working |
11:59.48 | Zeeek | yes we have three, what is the problem |
11:59.48 | frk2 | i wonder if the 100 windows pc on the same network are causing it to screw up |
11:59.55 | frk2 | random lock ups |
12:00.08 | frk2 | using 1.0.2.13 |
12:00.29 | frk2 | the phone voice quality degrades.... and then poof.. its unpingable. hard crash |
12:00.31 | chris_ast | I have around 1000 DID's and I have to setup context for each of it's extension and I want to do that dynamically and I do not want to change extensions.conf alll the time |
12:00.31 | chris_ast | whenever I want to add new context I will do it in Mysql DB but without mentioning that in ext.conf asterisk is not taking it |
12:00.37 | chris_ast | how can we do this |
12:00.57 | frk2 | this happens with the original firmware also |
12:01.03 | chris_ast | Zeeek: that is my question |
12:01.06 | frk2 | i am almost clueless |
12:01.37 | frk2 | i thought grandstreams were good |
12:01.45 | frk2 | or do i have a unique issue? |
12:02.29 | frk2 | Zeeek? are you facing something similar? |
12:03.01 | Zeeek | most people will say bad things about GS phones. My older ones work ok but the new GX ones I've always heard BAD about |
12:03.29 | Zeeek | chris_ast I don't know about that it sounds like you want to look at realtime on the wiki or something |
12:04.00 | frk2 | but you said you have 3 |
12:04.04 | frk2 | do they cause similar issues? |
12:04.53 | chris_ast | zeeek: I looked at realtime but even for that we have to name context in ext.conf and tell it to switch to realtime |
12:05.28 | konfuzed | slePP, are you awaje yet ;^) |
12:05.55 | Zeeek | chris_ast look at the mailing list and see if you find anything there |
12:06.47 | chris_ast | Thanks for the idea but I already added a post there before two days but sadly no good response |
12:06.54 | Zeeek | frk2 I don't have random lockups but I do have one phone that will not stay registered even on the same side of LAN as asterisk |
12:07.19 | Zeeek | chris_ast that means only one thing. You'll have to invent the solution and sell it |
12:07.29 | chris_ast | I am testing my luck here :) |
12:08.06 | Zeeek | your luck doesn't look too good at the moment :) |
12:08.44 | chris_ast | But I was not provided that much time here |
12:09.14 | Zeeek | provided? Oh, you mean no one gives a $hit ? |
12:09.18 | chris_ast | it's true, anyway is my question clear? |
12:09.30 | Zeeek | I think it is |
12:09.49 | frk2 | zeeek exactly. |
12:09.49 | Zeeek | but youhave to find someone who cares about that kind of thing, maybe they solved it |
12:10.27 | frk2 | the phone that doesnt stay registered- does it lock up? or do you have to restart it to get it to register? |
12:10.40 | frk2 | do you have the qualify=yes parameter? |
12:11.19 | Zeeek | yeah nothing helped |
12:11.23 | chris_ast | zeeek where can I get that kind of info, I am a newbie to asterisk and only today I cam to know about this chat |
12:11.29 | Zeeek | no lock up though, just unreachable phone |
12:11.37 | Zeeek | tried all firmwares, no diff |
12:11.44 | frk2 | yes thats what happening to me |
12:11.52 | Zeeek | but I can call out |
12:11.53 | frk2 | just that one phone? |
12:11.54 | powerchip | <PROTECTED> |
12:11.58 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
12:12.19 | frk2 | are you using your phones connected to a pc or standalone? |
12:12.26 | frk2 | maybe i have TWO phones like the one you have :) |
12:12.34 | powerchip | sip |
12:12.44 | kmilitzer | I guess I am too dumb, I cannot find out in Bug 5090 if there is a way to use asterisk as a t.38 gateway or not ... last message from steveu from 03-12-06 09:38 seems to imply that, but I cannot figure out how this can be accomplished ... |
12:12.52 | frk2 | so you DO have qualify=yes? |
12:13.41 | chris_ast | zeeek: where can I get that kind of info, I am a newbie to asterisk and only today I cam to know about this chat |
12:14.50 | wasim | ~wiki |
12:15.11 | wasim | ugh, you silly stupid jbot |
12:15.27 | wasim | chris_ast: www.voip-info.orggggggg |
12:15.28 | Zeeek | frk2 I did when it was behind NAT and NOT when it was on the same side as astrisk |
12:15.41 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
12:16.29 | chris_ast | wasim: I have gone thru that site but my thing is not listed there |
12:16.36 | tzafrir | ~voip-info |
12:16.38 | jbot | [voip-info] the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
12:17.12 | tzafrir | no need to be harsh on the poor bot_ |
12:17.16 | frk2 | hmmmmm |
12:17.40 | frk2 | mine are all on the same side as asterisk |
12:17.45 | chris_ast | can I tell u people what I require exactly? |
12:18.01 | chris_ast | I have gone thru voip-info |
12:18.03 | frk2 | can there be a problem with older hardware revisions? |
12:19.20 | chris_ast | I need a dynamic asterisk setup, I do not want to change extenions.conf all the time instead I change mysql DB |
12:19.36 | kardecallan | Is ther anybody that can help me in configuration of the STUN? |
12:21.17 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
12:22.08 | *** join/#asterisk dragonkh (n=kings@dr4g0nn.gotadsl.co.uk) |
12:22.11 | dragonkh | hi |
12:22.57 | dragonkh | how do I load the zaptel drivers? I installed by TMP400 and installed asterisk |
12:23.27 | dragonkh | if I modprobe zaptel - I still see nothing in zttool |
12:23.40 | tzafrir | modprobe wctdm |
12:23.59 | dragonkh | I get an error with wctdm |
12:24.14 | dragonkh | FATAL: Module wctdm not found. |
12:24.17 | tzafrir | run ztcfg |
12:24.18 | dragonkh | how do I fix that ? |
12:24.22 | chris_ast | tzafrir: please help me on astersik realtime setup |
12:24.25 | tzafrir | Do you get the same error? |
12:24.40 | tzafrir | if so: module was loaded |
12:24.46 | dragonkh | ZT_CHANCONFIG failed on channel 4: No such device or address (6) |
12:24.49 | dragonkh | I get |
12:25.20 | tzafrir | lsmod | grep ^zaptel |
12:25.33 | kippi | If i get a IXAy box, will that let me send faxes? |
12:25.52 | tzafrir | chris_ast, sorry, ask others... |
12:26.03 | dragonkh | zaptel 186372 0 |
12:26.06 | tzafrir | Anyway, you better ask your question anyway |
12:26.10 | dragonkh | crc_ccitt 2176 2 zaptel,hisax |
12:26.15 | tzafrir | so module was not loaded |
12:26.37 | tzafrir | Did you run modprobe wctdm ? |
12:27.02 | kardecallan | hello Zeeek!! |
12:27.09 | dragonkh | modrobe wctdm ==> FATAL: Error running install command for wctdm |
12:27.23 | dragonkh | modrobe wctdm ==> FATAL: Module wctdm not found. |
12:27.46 | dragonkh | where do I get the wctdm ? |
12:27.51 | tzafrir | what version of zaptel is it? |
12:27.59 | dragonkh | how do I tell ? |
12:28.12 | tzafrir | How exactly did you install it? |
12:28.13 | dragonkh | I ran the cvs astinsaller |
12:28.29 | kardecallan | asterisk -V |
12:28.33 | dragonkh | sudo /.astinstall |
12:29.30 | dragonkh | Asterisk CVS-v1-0-01/26/06-14:37:36 |
12:29.39 | wasim | q |
12:29.53 | tzafrir | dragonkh, you're using an ancient version of asterisk and zaptel |
12:30.07 | tzafrir | Any special reason to keep using it? |
12:30.28 | dragonkh | nope - how do I get a modern version then ? |
12:30.51 | tzafrir | http://www.asterisk.org/download |
12:30.53 | dragonkh | I downloaded the astinstall script and it connects to cvs and installs it |
12:31.12 | tzafrir | Where did you get it from? |
12:31.24 | dragonkh | the asterisk site I think |
12:31.26 | dragonkh | in here |
12:31.56 | kardecallan | Is there anybody that can help me? |
12:32.24 | dragonkh | isnt 01/26/06 - 26th Jan 2006 ? |
12:33.08 | kardecallan | I need to install the Stun? |
12:33.18 | tzafrir | dragonkh, right. My mistake. |
12:33.23 | dragonkh | I guess its USA format - 2001 - June 6 ? |
12:33.30 | *** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it) |
12:33.45 | dragonkh | it does say CVS v1 |
12:34.04 | tzafrir | does 'modinfo zaptel' give something? (don't paste it here) how about modinfo wctdm? modinfo wcfxs? |
12:34.33 | dragonkh | tzafrir: so you think the problem is im missing the wctdm driver? |
12:34.35 | tzafrir | hmmm... it pulls the branch v1-0 from the CVS. It is ancient code |
12:34.45 | tzafrir | modprobe fcfxs |
12:34.50 | tzafrir | modprobe wcfxs |
12:35.11 | tzafrir | (and still, you should work with more recent code) |
12:35.32 | dragonkh | all of those commands give not found errors |
12:35.52 | tzafrir | fcfxs is a typo . wcfxs should exist |
12:36.01 | dragonkh | ok let me check |
12:36.49 | kardecallan | by the way, I have installed it, but I need now to configure it, could you help me? |
12:37.15 | dragonkh | nope all the comands complain - Module wctdm not found |
12:37.23 | dragonkh | its clearly missing |
12:38.10 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
12:39.41 | *** join/#asterisk eaglezz (n=eaglez@62.108.213.120) |
12:42.37 | shiznatix | hello, does anyone have a tiffg3 file that I can use to test a fax with? |
12:43.37 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
12:44.08 | kardecallan | dragonhk, you can to go www.asterisk.org and download of the zaptel, to install you run make;make install. |
12:47.30 | Zeeek | dragonkh I had that problem - it's a linux config file |
12:47.41 | Zeeek | modules or something |
12:49.38 | Zeeek | <PROTECTED> |
12:50.13 | Zeeek | or is it modprobe.conf ? |
12:50.29 | Zeeek | post-install wcfxs /sbin/ztcfg |
12:52.26 | fourcheeze | got some issues with ringtone not being generated |
12:52.56 | fourcheeze | it seems that whenever I'm on my second extension of a call when I dial I don't get ringtone |
12:53.17 | *** join/#asterisk pheo (n=pheo@63.Red-80-36-138.staticIP.rima-tde.net) |
12:53.18 | fourcheeze | e.g. in an ivr or when handling a * or 0 drop out from voicemail |
12:53.20 | fourcheeze | any ideas? |
12:53.23 | pheo | hi all |
12:54.54 | shiznatix | how can I make a tiffg3 document without ghostscript? |
12:55.49 | fourcheeze | install ghostscript first?? ;-) |
12:58.11 | shiznatix | har har :) |
13:05.12 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.153) |
13:05.36 | dasenjo | Hi, I got just - in the transalation line for a codec (g729) .. why? |
13:07.06 | Zeeek | because you don't have g729 installed? |
13:07.51 | dasenjo | yes I have codec_g729.so in /usr/lib/asterisk/modules |
13:08.01 | Zeeek | and the license? |
13:10.14 | fourcheeze | how does * work out the times in show translation ? |
13:11.05 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
13:11.09 | Zeeek | I think it's just a table |
13:11.15 | dasenjo | is the "free" one .. downloaded from http://kvin.lv/pub/Linux/Asterisk/ |
13:11.27 | Zeeek | no advice for the free, never used it |
13:11.33 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:12.26 | fourcheeze | I didn't think there was a free one |
13:12.45 | fourcheeze | there's a patent on using g729, so if you live anywhere where patents apply then it's not free |
13:13.37 | dasenjo | I know .. there is no free one .. I found the error .. the module was in /usr/lib/asterisk but strangely didnt load automatically .. |
13:15.24 | astra^^ | hw do we know wher pattent is aplicable |
13:17.15 | fourcheeze | astra^^: no doubt you'll be well versed in your country's patent law |
13:17.18 | fourcheeze | ;-) |
13:19.32 | [ProB]CrazyMan | shiznatix: you can't |
13:31.57 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
13:32.40 | *** join/#asterisk shiznatix (n=shiznati@213-35-233-152-dsl.end.estpak.ee) |
13:34.53 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
13:35.20 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
13:35.56 | *** join/#asterisk nettie (n=esivieri@85-18-54-38.ip.fastwebnet.it) |
13:40.27 | astra^^ | :-/ |
13:40.47 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.209) |
13:40.48 | Kernel_Core | hi all |
13:41.13 | *** join/#asterisk mko-025 (n=korpim@p5498BD34.dip0.t-ipconnect.de) |
13:41.31 | *** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116) |
13:41.34 | astra^^ | fourcheeze: i have a dbt |
13:42.42 | Kernel_Core | Mar 16 07:33:28 WARNING[6451]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x85c83a8', 10 retries! <---- what is wrong ?! ( I am running 1.2.4) |
13:43.59 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
13:44.37 | gambolputty | Anyone having trouble compiling zaptel 1.2.4 with centos 4.2 kernel 2.6.9-34.ELsmp? |
13:45.22 | astra^^ | if a ser fwds call at my * wat are the setting to be made .... |
13:45.53 | astra^^ | i chkd in the sit bt nt so informative.. |
13:45.58 | astra^^ | site i mean |
13:46.37 | astra^^ | SER to Asterisk ... |
13:47.11 | astra^^ | ser--->Asterisk--->host server. |
13:51.17 | Aurs | gambolputty: you have to change a typo in a file in the kernel source |
13:51.50 | Aurs | gambolputty: I have a URL in an email here somewhere.. let me find it for you.. one sec |
13:52.08 | Aurs | https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 - there you are |
13:52.50 | astra^^ | any help on my question |
13:52.55 | astra^^ | :) |
13:54.58 | Aurs | Kernel_Core: http://bugs.digium.com/view.php?id=6445&nbn=15 - something like this? |
13:57.21 | Kernel_Core | Aurs: yea |
13:57.54 | Kernel_Core | Aurs: when asterisks load increase I get such error and after that asterisk stops serving ..... |
13:58.03 | *** join/#asterisk aze (n=lucky@ACayenne-101-1-11-208.w81-248.abo.wanadoo.fr) |
13:58.09 | aze | hi all |
13:58.25 | Zeeek | hello |
13:59.39 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
14:00.06 | Aurs | Kernel_Core: not that it helps you, but I had a similar case on a 1.0.9 this morning |
14:01.05 | Kernel_Core | Aurs: could you fix this problem ? |
14:01.35 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
14:01.38 | Aurs | Kernel_Core: not really. just ignoring it until we make the switch to 1.2.5 |
14:02.07 | Aurs | Kernel_Core: but this happened on a 1.2.4? |
14:02.29 | Kernel_Core | Aurs: yes , exactly |
14:02.32 | Aurs | Kernel_Core: are you using IAX to connect to other asterisk servers? |
14:03.19 | Kernel_Core | Aurs: no , I am useing chan_h323 |
14:03.52 | Kernel_Core | but I think it is generally channel.c problem |
14:04.08 | Kernel_Core | nor SIP IAX Problem |
14:04.10 | Aurs | Kernel_Core: I was googling this earlier today. check this out: http://bugs.digium.com/bug_view_page.php?bug_id=3848 |
14:04.40 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
14:04.55 | astra^^ | i want to fwd calls frm SER to * wher in asterisk will route to the iconnect server.. |
14:05.10 | astra^^ | what are the setting to b made in tis.. |
14:05.21 | astra^^ | asterisk is working fine as of nw |
14:05.24 | Aurs | hmm.. but that bug is probably fixed in 1.2.4, Kernel_Core.. i dunno :P |
14:06.48 | Aurs | astra^^: checked http://www.voip-info.org/wiki-SER+tips+and+tricks ? |
14:07.44 | Curus | Do any of you happen to have Elmeg 290 phones? |
14:08.16 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
14:09.40 | chris_ast | hi all |
14:12.16 | *** join/#asterisk exonic (n=exonic@209.172.11.54) |
14:12.22 | powerchip | how I can make so if Agent take , the call not go to him? |
14:13.26 | *** join/#asterisk KeX-NB (n=KeX-NB@ng1.kurtkrenn.com) |
14:13.27 | powerchip | talk* |
14:13.46 | KeX-NB | hi |
14:14.09 | KeX-NB | borbably not the correct chan, but is someone using ser/openser with tls? |
14:14.26 | *** join/#asterisk Tili (n=Tili@193.172.20.10) |
14:15.15 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
14:17.08 | tdonahue | Curus: if the picture I found of the Elmeg 290 is actually the phone, then it is probably a rebranded Snom 190. We have had mixed reviews of the 190, people either love or hate the phone. |
14:17.15 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
14:18.19 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
14:18.20 | SpaceBass | morning |
14:18.34 | SpaceBass | anyone know why faxes I receive come in blank? |
14:19.11 | chris_ast | SpaceBass: did u check codec it is using? |
14:19.33 | bkw_ | doubt the codec is the issue |
14:19.51 | chris_ast | reliable faxing only with G711 |
14:19.53 | Zeeek | good grief, bkw! |
14:20.07 | Zeeek | does asterlink still exist? |
14:20.24 | SpaceBass | chris_ast, its coming in on my zap trunk |
14:21.29 | SpaceBass | fyi as an aside- i fax out over a sip trunk and it works fine |
14:22.57 | Aurs | bkw_: are you the one who wrote app_dbodbc.c? |
14:23.23 | Zeeek | bkw_ are you the one that wrote NEXT! ? |
14:23.55 | chris_ast | SpaceBass, I have no idea on zap. The best I can tell you is check the codecs |
14:24.03 | Aurs | hmm.. is he the one who is about to write /ignore aurs, Zeeek ? :P |
14:24.26 | Zeeek | what did you do for such mean treatment? |
14:24.37 | chris_ast | Can anyone help me on Asterisk realtime |
14:24.57 | Zeeek | actually I'm an asterlink customer and I wondered if there was anyone at the controls |
14:25.07 | Aurs | dunno |
14:25.08 | Zeeek | having written a coiple of times |
14:25.26 | Zeeek | not that important, just curious |
14:25.29 | Aurs | chris_ast, what do you want help with? |
14:25.38 | chris_ast | can someone please help me configure dialplan dynamically |
14:26.16 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.132.131) |
14:26.39 | bkw_ | yes thats me |
14:26.53 | Fedoracore6 | hai all |
14:26.54 | Zeeek | where've you been hiding all these months/years? |
14:27.00 | bkw_ | right here |
14:27.05 | chris_ast | Aurs, I have around 1000 DID's and I have to setup context for each of it's extension and I want to do that dynamically and I do not want to change extensions.conf all the time |
14:27.42 | chris_ast | whenever I want to add new context I will do it in Mysql DB but without mentioning that in ext.conf asterisk is not taking it |
14:27.42 | Aurs | chris_ast: use _patternX. things |
14:27.43 | Aurs | ah.. for new contexts |
14:28.28 | chris_ast | Aurs, that is the problem contexts will be changing frequently and we cannot change ext.conf manually all the time |
14:28.50 | Aurs | bkw_: ok, what version was it written for? I'm using it (modified) on a 1.0.9 box |
14:29.02 | chris_ast | Aurs: please guide what could be done here |
14:29.29 | willt | chris_ast: probably not what you want but you could write a quick php scipt do grab info out of DB and right it out |
14:29.38 | willt | s/right/write/ |
14:30.01 | Aurs | lol. nice bot |
14:30.17 | willt | jbot rocks! :) |
14:30.22 | chris_ast | so with mysql asterisk we cannot do this |
14:30.36 | chris_ast | I mean the add on we have |
14:31.29 | Aurs | chris_ast: does this help? http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
14:32.03 | Aurs | guess not, since you still have to write the context in the extensions.conf |
14:32.47 | chris_ast | that is the problem, we have to mention each and every context and also give switch to realtime under it |
14:33.39 | chris_ast | Aurs: so practically we add or change did numbers and all the time manually we cannot change ext.conf |
14:33.43 | Aurs | you could do a static realtime on the entire extensions.conf file chris_ast |
14:33.55 | Aurs | but then you will need a extensions reload after any changes |
14:34.41 | Aurs | but that makes it easier to make an automated gui for it |
14:35.02 | chris_ast | jbot_: From php how I could I write rules with priorities to asterisk? |
14:35.21 | Aurs | extensions.conf => mysql,database,table_name |
14:35.35 | Aurs | in extconfig.conf |
14:36.09 | Aurs | chris_ast: check this out: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static |
14:36.34 | chris_ast | Aurs: I will check that link, one sec |
14:36.53 | dragonkh | hi I have asterisk working but when I dial in asterisk says it cant create channel type 'Zap' - and when I pick up the phone I have dialtone and can call out |
14:37.11 | dragonkh | anyone know why this could be ? |
14:37.38 | Zeeek | dragonkh let's see the Dial command you are using |
14:38.16 | dragonkh | Zeeek: er I installed that xorcom rapid asterisk - I'll look in the dialplan one sec |
14:38.58 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
14:39.04 | willt | asterisk doesn't support saving voicemail in mp3 does it? |
14:39.16 | frk2 | gsm is way better man |
14:39.33 | willt | why is it better? doesn't sound better to me |
14:39.38 | [TK]D-Fender | frk2 : Nah.. WAV. |
14:39.39 | *** join/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br) |
14:40.02 | _Paulo_ | ~seen coppice |
14:40.14 | jbot | coppice <n=chatzill@211.155.168.153> was last seen on IRC in channel #asterisk, 2d 4h 39m 12s ago, saying: '"Nun"'. |
14:40.15 | [TK]D-Fender | WAV is inherently better quality and more prolific in its playback capability |
14:40.15 | Aurs | gsm is not better for email attachments :) |
14:40.20 | willt | yes the wav49 sounded pretty good and is about the same size as gsm |
14:40.25 | Zeeek | gsm is better if you want faster loading |
14:40.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
14:40.45 | Zeeek | but windows has a hard time playing gsm natively |
14:40.59 | willt | quicktime plays it |
14:40.59 | asterisk99 | anyone know why I might get the message "chan_zap.c:6285 handle_init_event: Unable to play dialtone on channel 3 |
14:41.00 | asterisk99 | " when I lift handset of FXS attached phone? |
14:41.00 | Zeeek | and now a general OT announcement |
14:41.33 | Zeeek | I bought a book about Solaris 10 which sucked badly. I complained to the publisher and they are sending me a free book of my choice from their catalogue - cool, no? |
14:41.34 | dragonkh | Zeeek: er I dont understand the thing heh - but tell me this - I have zap/1 and zap/4 - the phone is set to zap/1 and so is the trunk - is this how it should be ? |
14:42.01 | Zeeek | ask xorcom how it works |
14:42.06 | willt | Zeeek: what publisher? lol |
14:42.35 | Zeeek | McGraw-Hill - check out "The Complete Guide to Solaris 10" - now that I read the reviews I see why I felt screwed |
14:42.46 | af_ | a good softphone for linux? |
14:42.54 | Zeeek | My complaint letter was already contained in other's reviews! |
14:43.02 | willt | lol |
14:43.14 | Zeeek | hey, the squeaky whell gets the grease |
14:43.22 | Zeeek | s/whell/wheel/ |
14:43.28 | chris_ast | Aurs: I feel that is not the thing we are looking for |
14:43.57 | Zeeek | ~seen TheLight |
14:44.00 | jbot | i haven't seen 'thelight', Zeeek |
14:44.04 | Zeeek | too bad |
14:44.09 | asterisk99 | Zeeek: What was wrong with it? Content missing? The grammer were bad? Full of spelling misteeks? |
14:44.16 | Aurs | chris_ast: ok. why not? you could make a script that automatically inserts the magic into the table and does a extensions reload for you |
14:44.22 | jsharp | ~seen Elvis |
14:44.23 | jbot | elvis <n=sdad@ipd50a583c.speed.planet.nl> was last seen on IRC in channel #debian, 33d 1h 9m 40s ago, saying: 'is there a combined package on debian to install all perl modules?'. |
14:44.38 | Aurs | magic as in "the context" |
14:44.43 | Zeeek | ~seen Elvisleaveth Building |
14:44.44 | jbot | i haven't seen 'elvisleaveth building', Zeeek |
14:44.55 | Zeeek | nyuk, nyuk |
14:45.24 | Zeeek | asterisk99 - it says Solaris 10 but has none of the actual SOlaris 10 differences |
14:45.44 | Zeeek | there is a whole new concept in SOlaris 10 regarding the way the system starts up |
14:46.06 | Zeeek | it'd be like "Complete Guide to Asterisk 1.2" with no reference to dialplans |
14:46.24 | Zeeek | there are 100 negative reviews confirming the same thing |
14:46.37 | chris_ast | Aurs: we do not want to reload many times |
14:46.41 | Zeeek | for some reason I didn't see these on my first visit - because I bought from a local amazon |
14:47.12 | chris_ast | Aurs: I strongly feel there must be a way to tell Asterisk to search for contexts directly from mysql DB |
14:47.17 | asterisk99 | anyone here have a working FXS phone?? |
14:47.21 | Zeeek | ~seen An Elephant Fly |
14:47.23 | jbot | Zeeek: i haven't seen 'an elephant fly' |
14:47.41 | Zeeek | I have phones that work on FXS cards |
14:48.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:48.12 | Zeeek | ~Zeeek |
14:48.14 | jbot | methinks zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
14:48.31 | Zeeek | jeeze what a memory! |
14:48.58 | chris_ast | jbot_: Please tell me how I could write rules with priorities to asterisk from php? |
14:49.00 | asterisk99 | Zeek: I have one that used to work and no longer does... I'm trying to figure out what I did wrong (probably something dumb) .... do you have your immediate=no or immediate=yes? |
14:49.29 | Zeeek | immediate=no - yes tells it to jump in the context and go |
14:49.33 | Zeeek | when picked up |
14:50.00 | asterisk99 | Zeek: You said that? Kewl!! BUt I rather think learning Asterisk isn;t quite as much fun as the other thing ;) |
14:50.22 | Zeeek | no, in fact it's actually a lot like masturbation, but I digress :) |
14:50.22 | asterisk99 | Zeek: A little less messy, tho |
14:50.31 | willt | chris_ast: are you trying to get the contexts in extensions.conf to pull dynamicaly from mysql? |
14:51.25 | Zeeek | gotta run, asterisk99 I don't know what changed for your FXS |
14:51.39 | chris_ast | willt: yes I do want to mention them in extensions.conf but directly tell Asterisk to get them from Mysql DB |
14:51.47 | asterisk99 | Zeek: K ... bye |
14:51.53 | Zeeek | later |
14:51.56 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
14:52.04 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
14:52.08 | kippi | from a * command is there away to set your extension to divert to voicemail or to another number? |
14:52.32 | willt | chris_ast: what do you mean mention them? |
14:52.57 | caio1982 | kippi: diverting like in what situation? |
14:53.12 | chris_ast | for realtime we have give context name in ext.conf and under we have to write switch to realtime |
14:53.12 | kippi | whenever a call comes in |
14:53.33 | chris_ast | willt: I do want to do this for every context |
14:53.56 | willt | chris_ast: this is for your did's? |
14:54.06 | caio1982 | kippi: i guess i didn't get you meaning of "divert" then |
14:54.15 | chris_ast | willt: s for my DID's |
14:54.20 | chris_ast | yes |
14:54.43 | kippi | so that if your away from your desk you can set a divert and it will either to your voicemail or mob |
14:56.05 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
14:56.29 | caio1982 | a call forwarding you mean, right? |
14:56.37 | kippi | yeah |
14:57.52 | caio1982 | kippi: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding |
14:58.02 | kippi | thanks |
14:58.06 | kippi | i'll have a read |
14:58.22 | kippi | once the page loads |
14:58.52 | caio1982 | kippi: yeah, it's also a bit slow today for me |
15:00.37 | *** join/#asterisk Theuni (n=ctheune@alphastar.gocept.com) |
15:00.40 | Theuni | howdi |
15:01.23 | Theuni | I upgraded asterisk last week (from 1.0 to 1.2) and with that change the music on hold became extremely loud (way overdriven actually). Is there a known itch during upgrade there? |
15:04.16 | *** join/#asterisk michael-i (i=michael@141.41.38.185) |
15:07.26 | *** join/#asterisk Deep6 (n=DEEP6@208.38.35.162) |
15:07.58 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
15:08.40 | ManxPower | Theuni, Was there anything mentioned in UPGRADE.txt? |
15:08.52 | tzafrir | dragonkh, any problems? |
15:08.57 | puzzled | hi |
15:13.09 | Kernel_Core | Aurs: I found something...... when asterisks load increase this messages more repeats .... Mar 16 09:06:27 WARNING[12616]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb5885f08', 10 retries! |
15:14.48 | ManxPower | Kernel_Core, that is normal unless it impacts call quality. |
15:15.02 | ManxPower | It seems more common when doing Monitor, ZapScan, ChanSpy, etc. |
15:15.41 | Theuni | ManxPower: only that the format has changed. probably it's a problem of mpg123. I upgraded from 0.59r to 0.59s. That might be the problem. |
15:16.07 | ManxPower | Theuni, only 0.59r is expected to work. |
15:16.44 | Theuni | downgraded already |
15:16.47 | Theuni | works again. thanks anyway. |
15:17.04 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
15:17.29 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
15:17.29 | Kernel_Core | ManxPower:what about this ? Mar 16 09:10:46 WARNING[18766]: file.c:584 ast_readaudio_callback: Failed to write frame |
15:18.22 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfigs.dialup.mindspring.com) |
15:18.27 | tamp4x | anyone here use spandsp? when i load asterisk -vvvvvvvvv it stops loading when app_rxfax.so loads....any ideas why? |
15:18.34 | *** join/#asterisk f7950qs0 (n=redhotte@61.17.213.99) |
15:18.41 | f7950qs0 | hi all |
15:18.45 | f7950qs0 | is fourcheeze here? |
15:19.00 | f7950qs0 | damn I can see in the list if he's there or not |
15:19.08 | *** join/#asterisk cas (n=cas@83.98.233.2) |
15:19.41 | tzafrir | he's here according to the users list |
15:19.50 | chris_ast | Please tell me whether I can match a context using regular expression? |
15:20.02 | tzafrir | also try /msg dpkg seen username |
15:20.48 | Kernel_Core | ManxPower: even when I issue show channels I get this : http://pastebin.com/605508 , active calls should be 1/2 active channels .... but it isnot... |
15:22.51 | f7950qs0 | tzafrir you use xorcom dont you |
15:23.03 | f7950qs0 | i get your msgs in mailing list of xorcom |
15:23.11 | f7950qs0 | rapid |
15:23.19 | *** join/#asterisk MikeJ_ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
15:23.26 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
15:23.50 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfigs.dialup.mindspring.com) |
15:24.21 | Kernel_Core | Mar 16 09:16:50 WARNING[19713]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files |
15:24.21 | Kernel_Core | Mar 16 09:16:50 WARNING[19713]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! |
15:24.28 | Kernel_Core | help me :| |
15:24.33 | f7950qs0 | does anybody know of a soft phone application that can configure one extension which can be assigned to a sip device |
15:26.19 | Kernel_Core | res_agi.c:246 launch_script: unable to create fromast pipe: Too many open files :(( |
15:26.19 | justinu | For the third straight year, the Department of Homeland Security -- which is charged with charting the federal government's cyber security agenda -- earned a grade of "F" for computer security from a key congressional oversight committee, according to a story at Washingtonpost.com. |
15:26.28 | justinu | way to go, guys! |
15:26.47 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
15:27.15 | tzafrir | f7950qs0, yes, I am |
15:27.22 | powerchip | how can i dio so if some client take to one , the not sent more phone to the client? |
15:27.44 | powerchip | do* |
15:27.55 | powerchip | talk* |
15:29.08 | justinu | kernel_core, is that a problem with ulimits perhaps? |
15:29.16 | f7950qs0 | hi justinu |
15:29.21 | justinu | your kernel is either out of resources, or denying you resources |
15:29.26 | [TK]D-Fender | powerchip : You can do a "ChanIsAvail" test to see if they are on a call in the dial-plan OR just limit it on the phone (tell it to allow only 1 call) |
15:30.41 | Kernel_Core | justinu: what is this ? |
15:30.45 | Kernel_Core | how do I check it ? |
15:30.47 | tzafrir | justinu, the legacy of the Symantec chief they had? |
15:30.47 | powerchip | I have try http://pastebin.com/605457 |
15:30.52 | powerchip | but not work:( |
15:31.10 | powerchip | what is wrong..:/ |
15:32.34 | justinu | Kernel_Core: ulimit -a |
15:32.54 | justinu | tzafrir: lol, interesting theory |
15:33.49 | Kernel_Core | justinu: http://pastebin.com/605546 |
15:34.22 | justinu | ok, so you have a 1024 open file limit |
15:34.24 | powerchip | [TK]D-Fender: u know? |
15:34.26 | *** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
15:34.44 | justinu | try changing it with ulimit -n 4096 |
15:34.46 | justinu | or something |
15:34.47 | f7950qs0 | show me one way of using asterisk as a call shop billing software |
15:35.02 | justinu | show me.... subscriber trunk dialing! <poof!> |
15:35.19 | ManxPower | Kernel_Core, yes, I get that too. It does not cause a problem in my experience. I assume you are using 1.2.x |
15:35.20 | f7950qs0 | 7 concurrent calls and i want to meter them |
15:35.24 | *** part/#asterisk cas (n=cas@83.98.233.2) |
15:35.27 | f7950qs0 | they come out of my sip devices |
15:35.38 | Kernel_Core | ManxPower: running 1.2.5 |
15:35.57 | ManxPower | Kernel_Core, what PROBLEMS is it causing? |
15:36.29 | Kernel_Core | ManxPower: calls being rejected by asterisk .... |
15:36.48 | Lloydie-t | I am trying to install res_sqlite3 but get error http://pastebin.ca/45925 Can you help? |
15:37.03 | ManxPower | Kernel_Core, then report it as a bug to bugs.digium.com. If you neglect to include what PROBLEM it is causing the bug will be closed. |
15:37.19 | [TK]D-Fender | powerchip : You aren't use the "ChanIsAvail" application properly. Read the instructions for it. |
15:37.21 | ManxPower | Kernel_Core, those messages are normal, rejecting calls is not. |
15:39.09 | powerchip | [TK]D-Fender: ok |
15:39.23 | ManxPower | Kernel_Core, you must have MANY MANY calls to be running out of FDs |
15:39.47 | Kernel_Core | ManxPower: I am useing chan_h323 ... |
15:39.48 | ManxPower | Kernel_Core, or your AGI is buggy and not exiting correctly |
15:40.00 | vuud | From a SIP phone attached to *, what is the best way to access the "s" extension? |
15:40.04 | Kernel_Core | ManxPower: I am useing astbill for accounting... |
15:40.12 | ManxPower | vuud, there's isn't. |
15:40.33 | vuud | oh wait - how about I put all that as an ext and goto it from the s exten |
15:40.46 | ManxPower | vuud, the "s" extensions is run when asterisk does not receive any information about the destination of the call (usually only when using analog fxo ports) |
15:40.49 | chris_ast | tzafrir, Please tell me whether I can match a context using regular expression? |
15:41.01 | vuud | ManxPower: right, I want to test menus and such |
15:41.16 | ManxPower | vuud, exten => 666,1,Goto(s,1) |
15:41.18 | tzafrir | chris_ast, I can't think of a simple way |
15:41.40 | ManxPower | on MY systems, I do more of exten => s,1,Goto(2101,1) |
15:41.43 | ManxPower | where 2101 is my IVR |
15:42.05 | vuud | ManxPower: yeah, i just started changing to that |
15:42.09 | *** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116) |
15:42.28 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
15:42.35 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
15:42.35 | *** mode/#asterisk [+o anthm] by ChanServ |
15:42.53 | ManxPower | exten => s really should be changed in the source code to be exten => unknown because "s" is confusing. |
15:43.52 | chris_ast | ManxPower, I could not get it clearly |
15:44.10 | Kernel_Core | ManxPower: is it normal ?! frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
15:44.13 | iDunno | s for "starting" ;) |
15:44.20 | [TK]D-Fender | ManxPower : Then we'll need to make a new one for macro's as rell... and so on... |
15:44.24 | ManxPower | Kernel_Core, THAT message is normal. |
15:44.33 | ManxPower | Kernel_Core, it looks like you have three totally different messages. |
15:45.34 | ManxPower | [TK]D-Fender, even in my macros I many times do exten => s,1,Goto(${MACRO_EXTEN},1) so my CDRs look good. |
15:45.36 | Kernel_Core | ManxPower: this message is repeating too fast .... Dropping extra frame of .... |
15:45.43 | Lloydie-t | Any thoughts on http://pastebin.ca/45925 ? |
15:45.52 | ManxPower | Kernel_Core, that is caused by buggy G729 devices. |
15:45.54 | chris_ast | can somwone please help me |
15:46.16 | ManxPower | Kernel_Core, you can just comment out that message in the source code if you want, or just run at a less verbose debug level |
15:46.36 | ManxPower | chris_ast, I did not see a question from you. |
15:47.01 | ManxPower | Lloydie-t, are you compileing asterisk as root? |
15:47.01 | Kernel_Core | ManxPower: OKey |
15:47.29 | Lloydie-t | ManxPower: Yes I am |
15:47.32 | Kernel_Core | ManxPower: hehe asterisk crashed at last !!!!! |
15:47.35 | tzafrir | chris_ast, perhaps you should state the reason for which you need regex matching of context name. As noone here seems to know how to do that |
15:47.44 | ManxPower | Kernel_Core, it will do that if you run out of FDs |
15:47.56 | Kernel_Core | FDs ? |
15:48.01 | justinu | sigh |
15:48.05 | willt | ;) |
15:48.05 | Kernel_Core | File Descriptror ? |
15:48.12 | ManxPower | Lloydie-t, perhaps astxs is not set to be executable. |
15:48.16 | ManxPower | Kernel_Core, correct. |
15:48.20 | justinu | with a name like kernel_Core, you'd think you'd know about the services a kernel provides |
15:48.23 | justinu | at least a little bit |
15:48.24 | Kernel_Core | ManxPower: how do I increase it ? |
15:48.29 | Kernel_Core | :)) |
15:48.30 | justinu | lol, i told you that already. |
15:48.34 | justinu | pay attention |
15:48.54 | chris_ast | here is my need, I have around 1000 DID's and I have to setup context for each of it's extension and I want to do that dynamically and I do not want to change extensions.conf all the time |
15:49.00 | chris_ast | whenever I want to add new context I will do it in Mysql DB but without mentioning that in ext.conf asterisk is not taking it |
15:49.02 | ManxPower | Kernel_Core, justinu already told you how to do that. |
15:49.12 | ManxPower | Each call should use no more than 4 FDs. |
15:49.14 | Kernel_Core | ulimit ... ? |
15:49.20 | ManxPower | if you are using more than that there is a bug somewhere. |
15:49.29 | ManxPower | Kernel_Core, correct, before you start asterisk |
15:49.30 | Lloydie-t | ManxPower: I a linux newbie. How would I set that to be an executable? |
15:49.41 | Hmmhesays | 1000 did's huh? |
15:49.42 | Hmmhesays | fun |
15:49.50 | justinu | 1000 dids? weak |
15:49.53 | justinu | i've got 20k + |
15:49.59 | ManxPower | Lloydie-t, I only help with Asterisk questions. Yours is a basic *nix question. |
15:50.14 | chris_ast | justinu, how are you managing |
15:50.24 | justinu | just fine |
15:50.37 | justinu | extensions reload is starting to take an appreciable amount of time to complete |
15:50.38 | Lloydie-t | No problem, I'll check on google |
15:50.42 | justinu | but it seems to run just ifne |
15:50.43 | Kernel_Core | ManxPower: so so what should I do now .... :| |
15:50.44 | fourcheeze | Lloydie-t: chmod +x somefile |
15:50.46 | puzzled | ManxPower: trade basic *nix knowledge for stroopwaffels and you should soon have that large crate |
15:50.59 | ManxPower | Kernel_Core, I recommend buying a book on basic linux |
15:51.05 | Kernel_Core | :)) |
15:51.17 | ManxPower | puzzled, THAT is the BEST idea I've heard this week. |
15:51.17 | chris_ast | justinu, I do not want to relaod all the time, can't we give everything in mysql DB |
15:51.21 | Kernel_Core | ManxPower: it is long time which I don't use linux tooooooo much ... |
15:51.27 | justinu | chris_ast: yes |
15:51.28 | willt | Kernel_Core: or start using the man pages |
15:51.50 | ManxPower | Patience. You must find your own path to Linux nirvana, Grasshopper. |
15:51.54 | Lloydie-t | Yeah, I got the redhat bible |
15:52.09 | justinu | man pages, there's an idea! |
15:52.38 | chris_ast | justinu, even updating contexts is also a problem |
15:52.38 | Hmmhesays | linux for dummies 2nd edition is a good book |
15:52.49 | ManxPower | You REALLY do need a basic understanding of Linux for you to be able to install, troubleshoot, and manage an Asterisk server. |
15:53.09 | Delvar | can somone help, how can i access ${CDR(billsec)} after a DIAL? i always get 0 back, even though the CDR has a value in the database afterwards. |
15:53.25 | f7950qs0 | can anyone tell me if I can get this work (www.callerid.com) done by asterisk |
15:55.08 | chris_ast | please tell me how can we have contexts dynamically? |
15:55.14 | justinu | realtime |
15:55.18 | Hmmhesays | rtfm dude |
15:55.21 | Hmmhesays | seriously |
15:55.28 | dragonkh | how can I improve the sound quality? people sound far away ?? |
15:55.31 | tamp4x | anyone here use spandsp? when i load asterisk -vvvvvvvvv it stops loading when app_rxfax.so loads....any ideas why? |
15:55.37 | fourcheeze | chris_ast: why is updating contexts a problem? |
15:55.44 | jarrod | dragonkh: thats usually gain levels |
15:55.55 | Hmmhesays | tamp4x: set debug in logger.conf |
15:55.59 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
15:56.01 | Dr-Linux | hi |
15:56.05 | Dr-Linux | error |
15:56.06 | Dr-Linux | Mar 16 07:42:26 WARNING[7029]: channel.c:784 channel_find_locked: Avoided initial deadlock for '0x8d63898', 10 retries! |
15:56.08 | Hmmhesays | theres trouble |
15:56.11 | Dr-Linux | what does this error mean? |
15:56.26 | jarrod | dr-linux: does that hose your box? |
15:56.27 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
15:56.30 | tahorg | ast_rtp_read: Unknown RTP codec 115 received |
15:56.31 | dragonkh | jarrod: where do I set them ? |
15:56.45 | Dr-Linux | jarrod: what is "hose" ? |
15:56.46 | tahorg | anyone can tell me what codec 115 is ? |
15:56.48 | Lloydie-t | fourcheeze: thanks for that. got a new problem though. I'll try and suss it myself though |
15:56.53 | Kernel_Core | note that if you do not set the ulimit -n 100000 or something similar efore you start asterisk you'll run out of FD's around 151 calls. :)) |
15:57.01 | tamp4x | debug => debug is set |
15:57.04 | jarrod | dr-linux: the asterisk box becomes unusable |
15:57.10 | *** join/#asterisk _Sam-- (n=sam@mail.kneedraggers.com) |
15:57.13 | tamp4x | i assume /var/log/asterisk/debug willt ell me |
15:57.26 | Dr-Linux | jarrod: my everything is working fine |
15:57.29 | justinu | Dr-Linux: unless you're having a problem, it's normal to see that occasionally |
15:57.44 | Dr-Linux | but when i dial this extension i see this error |
15:57.47 | Dr-Linux | hi justinu |
15:57.54 | SwK[Work] | anyone know what protocol Mitel 5020 IP phones use? |
15:57.57 | tamp4x | debug tells me nothing hmmhesays |
15:58.04 | jarrod | swk: i believe its proprietary |
15:58.05 | Dr-Linux | justinu: so it's not a bug ? |
15:58.10 | jarrod | so it wont interoperate with * |
15:58.12 | justinu | no |
15:58.26 | Dr-Linux | i just wornder if its makes bad my box |
15:58.38 | SwK[Work] | jarrod: you know I should know this.. mitel and sprints IP-PBX is the same platform with a different label onit heh |
15:58.42 | Dr-Linux | but actually what does this warning MEANS? |
15:59.27 | SwK[Work] | <PROTECTED> |
15:59.28 | *** join/#asterisk AlexCTI (n=alex@pembrkfl-bellsouth-24-53-200-134.miamfl.adelphia.net) |
15:59.31 | chris_ast | Please tell me how I configure conexts in ext.conf dynamically |
15:59.36 | SwK[Work] | damn its amazing what you can find on google |
16:00.07 | jarrod | chris: dynamically? |
16:00.38 | chris_ast | yes I want dynamically |
16:01.23 | chris_ast | jarrod, I do not want to give them in ext.conf but directly from mysql DVB |
16:01.26 | chris_ast | DB |
16:01.39 | jarrod | use realtime |
16:01.44 | AlexCTI | Hi, I need some help, I update my server to the new version and I got this MSG and * doesnt come up Mar 16 11:55:01 WARNING[3185]: loader.c:499 load_modules: Loading module chan |
16:01.44 | AlexCTI | dem.so failed! |
16:02.06 | chris_ast | even with realtime I have to mention them in ext.conf |
16:02.20 | jarrod | and there are two types of realtime I believe... one that loads it into memory upon reload/startup, and one that reads it dynamically |
16:02.36 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
16:02.38 | jarrod | chris_ast: incorrect - you can go to a full sql config |
16:03.21 | chris_ast | jarrod, everthing from sql?how can we do that? |
16:04.25 | jarrod | chris: http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database |
16:05.53 | chris_ast | jarrod: asterisk needs to be reloaded all the time |
16:06.16 | fourcheeze | chris_ast: extensions reload |
16:06.17 | jarrod | External configuration is configured in /etc/asterisk/extconfig.conf allowing you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload. |
16:06.22 | Lloydie-t | still got problem with res_sqlite, tried to suss it for nyself, not sure where to start http://pastebin.ca/45928 |
16:06.47 | f7950qs0 | HI FOURCHEEZE |
16:06.50 | f7950qs0 | oops |
16:06.52 | f7950qs0 | caps on sorry |
16:06.53 | fourcheeze | chris_ast: I'm using realtime myself but I don't go as far as putting my extensions in there |
16:07.11 | fourcheeze | chris_ast: but running extensions reload shoudln't give you too many problems |
16:07.37 | fourcheeze | f7950qs0: erm hello |
16:07.49 | f7950qs0 | there's an identifier machine on www.yes-tele.com and i want that work done through some software |
16:07.56 | chris_ast | jarrod,fourcheeze: even for realtime we need to include context name in ext.conf and under it we have to give switch to realtime, I want to avoid even that |
16:07.57 | f7950qs0 | how? |
16:08.14 | fourcheeze | chris_ast: are you putting your extensions.conf into realtime? |
16:08.18 | jarrod | you don thave to specify in two locations |
16:08.24 | f7950qs0 | my girlfriend has got visas to come to me |
16:08.34 | jarrod | if you specify extconfig to look for extensions in sql via realtime |
16:08.37 | chris_ast | fourcheeze: reload is not a problem but we update or create contexts very frequently |
16:08.46 | f7950qs0 | i'm happy |
16:09.06 | f7950qs0 | o.k. back to work anybody had a look at that site? |
16:09.08 | fourcheeze | f7950qs0: where are you - guantanamo bay ? ;-) |
16:09.23 | SpaceBass | anyone using dring? |
16:09.32 | f7950qs0 | what's that |
16:09.38 | f7950qs0 | i'm in India |
16:09.50 | chris_ast | fourcheeze: extensions.conf is realtime, I mean context rules are defined in mysql db |
16:10.53 | chris_ast | jarrod: even if we specify in extconfig.conf we have include context name and swith to realtime in ext.conf |
16:11.06 | fourcheeze | chris_ast: what does your extconfig.conf look like? |
16:11.42 | *** join/#asterisk jmacz (n=jmacz@201.244.241.189) |
16:11.55 | Hmmhesays | look at me cause i ain't wearing no frown |
16:12.42 | fourcheeze | f7950qs0: it looks like the yes telecom thing is just what you were on about yesterday |
16:12.48 | f7950qs0 | i'm getting tired of my cafe |
16:13.02 | f7950qs0 | is there anyone here providing asterisk consultancy (configuring) services in India? |
16:13.11 | f7950qs0 | a day's work for an expert |
16:13.12 | Hmmhesays | why do you need to be in india? |
16:13.26 | f7950qs0 | because I haven't yet got a green card to be in the US |
16:13.28 | Hmmhesays | thats the beauty of ssh |
16:13.42 | f7950qs0 | the call shop is in India |
16:13.52 | chris_ast | fourcheeze, any suggestion for me |
16:13.55 | Hmmhesays | why do you need someone physically there? |
16:13.55 | asterboy | great more outsourcing. |
16:14.00 | asterboy | no thanks. |
16:14.01 | fourcheeze | f7950qs0: if you set up a basic debian box and get ssh access people can be anywhere |
16:14.10 | f7950qs0 | i dont have to have someone physically here. |
16:14.15 | asterboy | Tired of every call I make giong to India. |
16:14.21 | Hmmhesays | ok what problem are you having? |
16:14.25 | Hmmhesays | or what do you need done? |
16:14.27 | fourcheeze | chris_ast: show us yout extconfig.conf |
16:14.33 | f7950qs0 | :)) india BPO crazy asses |
16:14.40 | willt | ~pastebin |
16:14.41 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
16:14.47 | gniretar_work | no way i'm gonne be involved in another buissness going to India |
16:15.11 | SpaceBass | can someone take a look at my zapata.conf http://pastebin.ca/45929 when I use dring EVERYTHING goes to the zap/2's context |
16:15.40 | asterboy | The US has an 8 Trillion dollar deficit...can't see how they can afford to outsource. |
16:15.45 | f7950qs0 | how much would somebody charge me for this thing? i want metering for my sip devices that's it |
16:15.49 | chris_ast | fourcheeze: here it is |
16:15.56 | chris_ast | extensions => mysql,asterisk,extensions_table |
16:16.11 | Hmmhesays | f7950qs0: metering? |
16:16.23 | fourcheeze | chris_ast: are you using the dynamic or static realtime configuration? |
16:16.33 | chris_ast | dynamic |
16:16.43 | fourcheeze | f7950qs0: what you're talking about is a basic asterisk installation |
16:16.48 | *** join/#asterisk file[laptop] (n=jcolp@142.131.190.116) |
16:16.54 | Hmmhesays | i wouldn't say that quite yet |
16:17.31 | chris_ast | I did all the things shown in voip-info but this thing is not given there |
16:17.40 | f7950qs0 | that's it? dint quite feel that easy to me fourcheeze |
16:17.46 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
16:17.56 | Hmmhesays | f7950qs0: explain what you want done in more detail |
16:17.58 | fourcheeze | f7950qs0: get me a debian box and I'll install asterisk for you |
16:18.05 | TheCompWiz | has anyone tried to use a nortel pbx as a voip termination? |
16:18.08 | fourcheeze | and I'll only charge $100 |
16:18.15 | chris_ast | can I use reges for context name? Is this possible? |
16:18.20 | Hmmhesays | fourcheeze you're going to end up eating those words |
16:18.30 | fourcheeze | maybe :-) |
16:18.42 | fourcheeze | never done it before |
16:18.43 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
16:18.49 | willt | omg lol |
16:19.00 | Hmmhesays | take my advice, find out what he wants done because "metering for sip devices" means nothing |
16:19.11 | Hmmhesays | and never ever charge a flat fee |
16:19.13 | fourcheeze | Hmmhesays: well if it's what he was on about yesterday |
16:19.25 | fourcheeze | it means storing cdr inmysql |
16:19.26 | f7950qs0 | Hmmhesays: i have an internet call shop people come in and use my sipdevices (connected to analog phones) when the call ends i need to meter how long did they call that's it. just wanna know the minutes they utilized |
16:19.37 | *** join/#asterisk salviadud (n=ralfalfa@201.137.161.31) |
16:19.41 | Hmmhesays | so you want to set up call detal records |
16:19.43 | fourcheeze | f7950qs0: if I give you a mysql port somewhere are you able to configure something to talk to it? |
16:19.59 | fourcheeze | f7950qs0: like windows obdc drivers? |
16:20.03 | f7950qs0 | i can use my telephony provider's online webpage to track the minutes but they dont provide instant online billing |
16:20.10 | salviadud | i got this prank call i recored with mixmonitor |
16:20.14 | *** join/#asterisk Altair256 (n=icechat5@mail.clccorp.com) |
16:20.16 | f7950qs0 | i dont know what is windows odbc drivers i'm sorry |
16:20.18 | salviadud | anybody want to hear it? |
16:20.25 | justinu | f7950qs0: install "asterisk-stat" |
16:20.29 | chris_ast | fourcheeze: can u suggest me something on this? |
16:20.37 | Hmmhesays | f7950qs0: thats a fairly simple setup |
16:20.41 | *** join/#asterisk Aurs (i=aurs@hallo.aurs.info) |
16:21.08 | f7950qs0 | practically i dont know anything about asterisk and dont think even wanna know just for the purpose of cafe someday will learn just for learning |
16:21.08 | TheCompWiz | anyone in here work with nortel systems? |
16:21.36 | Hmmhesays | you know anything about linux ? |
16:21.42 | Nugget | Linux is poo. |
16:21.48 | willt | f7950qs0: do they pay you cash or charge with credit card? |
16:21.50 | salviadud | freebsd is poo |
16:21.51 | f7950qs0 | a little bit |
16:21.54 | salviadud | with a devil log |
16:21.56 | salviadud | logo |
16:21.56 | f7950qs0 | they pay me cash |
16:22.35 | Fedoracore6 | hai all i was problem i my updatecode to databases .. any one can help me what my false in this code |
16:22.36 | Fedoracore6 | http://pastebin.com/605643 |
16:22.37 | f7950qs0 | I charge them two cents or less per minute to use my cafe shop's cabins and the phones and they dont fucking pay me sometimes becuase i can't track if the call went through or not |
16:22.56 | salviadud | you dudes know how many formats does mixmonitor support? |
16:23.24 | f7950qs0 | call shop sorry not cafe shop |
16:23.28 | AlexCTI | Someone can help me to fix this error: Mar 16 12:16:06 WARNING[8712]: loader.c:499 load_modules: Loading module chan_modem.so failed! happend after update. |
16:23.32 | Fedoracore6 | cos i press like example 2222 " code for subject computer" but can put in databases |
16:24.24 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
16:25.05 | SpaceBass | can someone take a look at my zapata.conf http://pastebin.ca/45929 when I use dring EVERYTHING goes to the zap/2's context |
16:25.38 | ManxPower | AlexCTI, That is why "make install" warns you about those modules. |
16:27.44 | ManxPower | in fact UPGRADE.txt ALSO talks about those modules. |
16:27.56 | ManxPower | And that issue has come up on this channel many times. |
16:28.02 | ManxPower | and it's been talked about on the mailing lists. |
16:28.15 | snip3r | Hi folks |
16:28.33 | snip3r | wonder if you remember my problem with asterisk |
16:28.56 | salviadud | we're not here to remember |
16:28.59 | salviadud | we forgive and forget |
16:29.11 | ManxPower | well forget at least. |
16:29.20 | salviadud | yeah |
16:31.48 | Fedoracore6 | :) |
16:32.05 | chris_ast | can someone please help me solve my problem? |
16:32.10 | Aurs | Fedoracore6: what are the datatypes on the fields you are updating in mysql? |
16:32.52 | *** join/#asterisk digg10 (n=john@206-248-135-54.dsl.teksavvy.com) |
16:32.58 | chris_ast | Aurs: still I could not get a clue on my problem :( |
16:33.12 | Aurs | chris_ast: the extensions thing? |
16:33.19 | chris_ast | s Aurs |
16:34.19 | Aurs | Fedoracore6: the '$clid' in the sql can cause the error if the clid field is a integer (or some other number-type). that was my thought. |
16:34.30 | digg10 | is it possible to do time_of_day routing with extensions.conf being loaded from the database? |
16:35.03 | Aurs | *food* |
16:36.14 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
16:40.25 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
16:41.07 | Zeeek | oh my |
16:42.25 | Fedoracore6 | i wanna update kodsubjek2 in my code |
16:42.25 | *** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116) |
16:43.23 | *** join/#asterisk thdei (n=DD@nat1.cri74.org) |
16:43.37 | thdei | hello everybody |
16:43.45 | thdei | I have a question about a problem |
16:43.54 | thdei | Sometimes, when I call someone out of my company, i have the message below |
16:43.59 | thdei | -- Called ISDN1/0450047588 |
16:44.01 | TheCompWiz | does anyone have any experience with nortel equipment? |
16:44.07 | thdei | <PROTECTED> |
16:44.15 | thdei | <PROTECTED> |
16:44.22 | thdei | <PROTECTED> |
16:44.28 | thdei | <PROTECTED> |
16:44.35 | thdei | Do you have a idea for me ? |
16:44.54 | SpaceBass | can someone take a look at my zapata.conf http://pastebin.ca/45929 when I use dring EVERYTHING goes to the zap/2's context |
16:45.00 | ManxPower | thdei, what is the value of DIALSTATUS? |
16:45.21 | ManxPower | SpaceBass, very few people use dring so not many people will be able to help you. |
16:45.22 | [TK]D-Fender | TheCompWiz : Sure, try asking the more direct question...... |
16:45.32 | thdei | ManxPower, How a know this ? |
16:45.58 | ManxPower | themacuser, Noop(DIALSTATUS=${DIALSTATUS}) as the priority after the DIAL |
16:46.10 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
16:46.12 | thdei | ok, I try |
16:46.18 | Zeeek | hey ManxPower you coming to Astrocn Eu this time? |
16:46.30 | Zeeek | s/Astrocn/Astricon/ |
16:46.30 | ManxPower | Zeeek, no. |
16:46.38 | TheCompWiz | [TK]D-Fender ... do you know how to configure the meridian box to accept connections from asterisk & use it to dial out? |
16:46.40 | AlexCTI | Manxpower, Thanks I'll read the docs about this |
16:47.01 | TheCompWiz | (in/out rather) |
16:47.03 | ManxPower | Zeeek, too many post-Kartina things to deal with. I may not even have any holiday this summer. |
16:47.14 | SpaceBass | ManxPower, I know...trying to space that apart so as not to be "that guy"...just hoping someone has a clue |
16:47.16 | justinu | TheCompWiz: use a T1 PRI to interconnect your PBXs |
16:47.18 | justinu | if you can. |
16:47.27 | SpaceBass | ManxPower, I'm basically with out fax until I can get it fixed.... |
16:47.28 | Fedoracore6 | Aurs: its i wrong using the $clid |
16:47.32 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
16:47.34 | TheCompWiz | justinu ... can I use an ethernet interface instead? |
16:47.39 | Fedoracore6 | or i must change the other feild |
16:47.42 | SpaceBass | unless, is there a way for asterisk to just recognize fax as soon as it answers the zap trunk? |
16:47.44 | ManxPower | SpaceBass, the mailing list archives were not helpful? |
16:47.52 | ManxPower | SpaceBass, it can do that. |
16:47.55 | ManxPower | see exten => fax |
16:48.07 | ManxPower | also see faxdetect in zapata.conf |
16:48.16 | justinu | TheCompWiz: doubt it... legacy PBXs use analog or T1/E1 interfaces usually |
16:48.33 | SpaceBass | ManxPower, didn't find anything there, but just posted to the forums for A@H on SF |
16:48.35 | f7950qs0 | time for me to leave. wont come to bug you guys again. will come here to hang out though |
16:48.43 | SpaceBass | ManxPower, thanks! |
16:48.54 | ManxPower | SpaceBass, you are using Asterisk@Home? Pervert. |
16:48.58 | TheCompWiz | justinu... this box of mine isn't really "legacy" |
16:49.07 | justinu | meridian is legacy |
16:49.28 | Zeeek | ManxPower have you (sucessfully) relocated? |
16:49.33 | justinu | unless your meridian happens to speak SIP |
16:49.38 | TheCompWiz | supposedly... this pbx box of mine is already voip capable... |
16:49.43 | TheCompWiz | (I'm guessing SIP) |
16:50.00 | justinu | sure... anything is VoIP capable if you plug a media gateway into it. |
16:50.07 | TheCompWiz | heh. |
16:51.08 | Zeeek | there are whole sites about using @hole |
16:51.24 | SpaceBass | ManxPower, i was orignally drawn to A@H out of lack of expirence and the cool wake up call feature...next box I build will be from scratch...but I still like freepbx for adding and managing extensions and users |
16:52.16 | thdei | ManxPower, I have DIALSTATUS=CHANUNAVAIL |
16:52.18 | *** join/#asterisk fugitivo (n=ajf@201.255.177.17) |
16:52.31 | ManxPower | Zeeek, I am, however, willing to consider accepting consuling jobs now. |
16:52.38 | [TK]D-Fender | TheCompWiz : Careful, it may be a proprietary format though. Confim your exact scenario first.... |
16:52.49 | ManxPower | thdei, that means Asterisk cannot complete the call. |
16:52.51 | thdei | But as I say that don't happen each time |
16:52.57 | *** join/#asterisk T-Squared (n=T-Square@hidden.serreyn.com) |
16:53.04 | Zeeek | ManxPower that's good to know! |
16:53.11 | *** part/#asterisk T-Squared (n=T-Square@hidden.serreyn.com) |
16:53.12 | ManxPower | or it means that your PROVIDER can't process the call. |
16:53.27 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
16:53.31 | ManxPower | Zeeek, I'm not cheap. 8-) |
16:53.45 | thdei | ManxPower, do you know what can do that Asterisk cannot complete the call. |
16:54.01 | ManxPower | thdei, no. |
16:54.02 | Zeeek | someone asked me yesterday about recovering from Pickup() when the extension is NOT ringing, e.g., play a message when they arrive at Pickup(). It just hangs up if not ringing. |
16:54.14 | ManxPower | Perhaps your provider only allows you 1 call at a time? |
16:54.21 | ManxPower | perhaps your provider does not have enough lines? |
16:54.47 | fugitivo | anyone had this problem before? |
16:55.02 | thdei | no, I have 3 rnis lines and before, using a 3com system, there was no probleme . |
16:55.03 | fugitivo | chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
16:55.05 | Zeeek | actually the provider usually give a CONGESTION status when it can't happen |
16:55.09 | ManxPower | fugitivo, yes, we have problems with people not telling us their problem. |
16:55.20 | Zeeek | Help me someone please! |
16:55.26 | Zeeek | Newbie question! |
16:55.29 | Zeeek | Can I ask it? |
16:55.31 | justinu | lol |
16:55.32 | Zeeek | Or not? |
16:55.36 | justinu | just ask it |
16:55.38 | justinu | wtf? |
16:55.46 | ManxPower | fugitivo, that means data was lost between the card and the zaptel driver. usually something is locking interrupts for too long. common causes of this can be IDE, SATA, on board Ethernet, RAID, etc. |
16:55.54 | Zeeek | is that your question? You only get one you know! |
16:56.02 | thdei | thnks for your help ManxPower, i will search more |
16:56.03 | ManxPower | anything that wants to take control of the PCI bus to improve performance. |
16:56.05 | justinu | heh |
16:56.13 | fugitivo | ManxPower: great |
16:56.26 | fugitivo | ManxPower: interrupts? |
16:57.15 | Zeeek | Love the mailing list thread "must distinguish between voIP and voPI" |
16:58.37 | fugitivo | ManxPower: and this? sorry, this is my first isdn pri (Don't know what to do with control frame 15) |
16:59.07 | ManxPower | fugitivo, you said you were having HDLC messages, not control frame messages. |
16:59.17 | ManxPower | ignore the control frame messages. |
16:59.22 | jbalcomb | [TK]D-Fender How long do you estimate needing to revamp our dialplan? |
16:59.31 | *** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it) |
16:59.35 | Micetto | hi |
16:59.38 | Micetto | :) |
16:59.44 | justinu | fugitivo: you've either got a hardware problem with your zaptel card, or your T1 is not deriving timing from the loop correctly. |
16:59.44 | Zeeek | :( |
16:59.58 | Micetto | can anyone help me with chan_zap and asterisk to reiceve a fax from isdn interface? |
17:00.12 | Micetto | how can I do this ? |
17:00.24 | SpaceBass | can you technically get a fax over isdn? |
17:00.33 | Micetto | yes |
17:00.33 | justinu | of course |
17:00.37 | justinu | faxes go over ISDN all the time |
17:00.50 | Zeeek | but do they ever come back? |
17:00.56 | Micetto | but I have set "faxdetect=both" |
17:01.06 | SpaceBass | i just didn't know...cool |
17:01.08 | [TK]D-Fender | jbalcomb : within a week. I'm moving next weekend though. |
17:01.41 | Hmmhesays | i guess i have to move |
17:01.42 | Hmmhesays | that sucks |
17:01.43 | Micetto | and if in CLI type "zap show channel 13" |
17:01.43 | Hmmhesays | i hate moving |
17:01.53 | fugitivo | justinu: how can i check if it's a problem with the card? |
17:01.55 | Micetto | I see Fax Handled: no |
17:02.33 | justinu | fugitivo: check to make sure nothing is sharing IRQs w/ your card |
17:03.01 | Micetto | puff.... |
17:03.04 | Micetto | :) |
17:03.22 | Micetto | !@#....with asterisk!!! |
17:04.03 | Micetto | I'm going crazy to get a signal fax from my ISDN card!!! |
17:05.28 | Micetto | help me! :D |
17:05.30 | Micetto | ^_^ |
17:06.12 | tzafrir | Micetto, how about giving some actual details as opposed to ranting |
17:06.24 | justinu | lol |
17:06.30 | Zeeek | more fun to try to guess the problem |
17:06.48 | tzafrir | Version of Asterisk, They type of ISDN channel, error message you get, etc. |
17:07.03 | Micetto | ok |
17:07.19 | Micetto | Asterisk take from bristuff-0.3.0-PRE-1k.tar.gz |
17:07.31 | fugitivo | justinu: could this be the same problem? No D-channels available! Using Primary channel 16 as D-channel anyway! |
17:07.31 | Micetto | Driver that I use is qozap |
17:07.43 | Micetto | with my quadBRI card |
17:07.47 | justinu | fugitivo: yeah, d-channel is completely down now. |
17:07.48 | Micetto | (Cologne Chipset) |
17:08.06 | Micetto | I don't get any error message.... |
17:08.16 | Micetto | My problem is that fax is not handle!!! |
17:08.18 | fugitivo | justinu: that's not a config problem? |
17:08.34 | ManxPower | fugitivo, HDLC errors are come of the hardest problems to solve with Asterisk |
17:08.40 | justinu | fugitivo: is it a new setup? |
17:08.44 | ManxPower | some of that is |
17:08.47 | fugitivo | justinu: yes |
17:08.53 | justinu | fugitivo: so it never worked? |
17:09.02 | fugitivo | justinu: i was working with R2 but never received those errors |
17:09.09 | justinu | fugitivo: look into IRQ conflicts first. |
17:09.15 | justinu | cat /proc/interrupts |
17:09.16 | Micetto | tzafrir: in default context exist a fax extensions that execute a NoOp if a fax is received |
17:09.43 | Micetto | tzafrir: but NoOp it's never been executed |
17:09.59 | SpaceBass | ManxPower, you didn't tell me that faxdetect just doesnt work |
17:10.01 | SpaceBass | :) |
17:10.26 | ManxPower | SpaceBass, it does work |
17:10.38 | ManxPower | it can be confusing, of course. |
17:10.59 | SpaceBass | ManxPower, I'm sure thats the case....I found a little about it on the wiki... |
17:11.50 | SpaceBass | i have it set to =incoming but its just sending it straight into the dialplan for voice calls...from what I'm reading if the fax tone doesnt happen to beep right as the zap channel picks it up then it doesnt know its a fax |
17:11.55 | ManxPower | you, of course, have to answer the line for asterisk to hear the fax. |
17:12.36 | ManxPower | then you need to do something like Ring and then something like background or waitexten for a few seconds |
17:12.36 | SpaceBass | there is an answer in the dial plan |
17:12.49 | SpaceBass | yeah, I'd need to build in a wait |
17:13.12 | tzafrir | Micetto, do you try to use the fax detection option of zap? |
17:13.12 | Zeeek | take a look at this: http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk |
17:13.30 | Micetto | yes |
17:13.33 | Zeeek | it works |
17:13.47 | Micetto | tzafrir: faxdetection=both |
17:13.51 | ManxPower | if you want more control then you can build and install NVFaxDetect, but most people don't need that. |
17:13.57 | *** join/#asterisk enzo123 (n=enzo123@200.sub-70-197-78.myvzw.com) |
17:14.14 | ManxPower | Micetto, you don't want that. Then asterisk will send an OUTGOING fax to the asterisk fax extension instead of the PSTN |
17:14.57 | ManxPower | well COULD, rather than "will" |
17:15.32 | Zeeek | you know what, we get faxes and there's no answer() to be seen anywhere |
17:15.35 | Micetto | ManxPower: and how I set up faxdetection ? which value should I use ? |
17:15.50 | ManxPower | Micetto, usually incoming |
17:15.56 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
17:15.56 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.209) |
17:15.59 | Kernel_Core | hi alll |
17:16.12 | Micetto | ManxPower: ok, now I'm testing |
17:16.14 | ManxPower | Zeeek, playback, background, and other apps answer the line |
17:16.21 | Zeeek | I just have exten => fax,goto(real_fax_area,1) in the beginning of the context |
17:16.41 | Fedoracore6 | i wanna do update system to databases , its i must using the $query_result1 = @mysql_query($query1); |
17:16.50 | ManxPower | Zeeek, that won't work. you don't have a priority |
17:16.58 | Zeeek | the scottstuff site I gave above has details on how to receive faxes and they work |
17:17.02 | Micetto | ManxPower: where I can sure that faxdetection is enabled ? |
17:17.12 | ManxPower | Micetto, zapata.conf |
17:17.22 | Zeeek | yeah, well I'm bad at typing, but it's there and we get all the junk faxes assholes insist on sending us |
17:17.23 | Micetto | ManxPower: ok, and in CLI ? |
17:17.36 | ManxPower | Micetto, I don't know. |
17:17.48 | Micetto | :( |
17:18.08 | Zeeek | show fax detection :) |
17:18.12 | Micetto | #@!#@@!#!!#.... |
17:18.14 | ManxPower | if it's set it's enabled |
17:18.20 | Kernel_Core | ManxPower: I changed my ulimit and /proc/sys/fs/file-max , after 69 calls , I get this error http://pastebin.com/605728 :| |
17:18.28 | SpaceBass | Micetto, I can emphatize...thats about how I feel |
17:18.42 | Zeeek | has anyone read the article I posted above? |
17:18.54 | SpaceBass | from Scott? yeah i had it bookmakred even |
17:19.02 | Zeeek | well it works perfectly |
17:19.08 | Micetto | Zeeek: but it is an old article :P |
17:19.09 | Kernel_Core | ManxPower: after that , users will hear busy ... |
17:19.21 | Zeeek | Micetto so what? It works |
17:19.42 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
17:20.06 | ManxPower | Kernel_Core, and what does ulimit -a say for open files |
17:20.19 | *** part/#asterisk enzo123 (n=enzo123@200.sub-70-197-78.myvzw.com) |
17:20.24 | Micetto | ManxPower: don't work....:'(( |
17:20.33 | justinu | poor manxpower |
17:20.46 | Zeeek | beseiged with questions |
17:20.47 | justinu | mad n00b kung-fu going on |
17:20.56 | Micetto | <PROTECTED> |
17:21.00 | ManxPower | justinu, eventually I get frustrated and stop answering questions |
17:21.01 | Zeeek | questions that have been answered a million times elsewhere |
17:21.06 | justinu | ManxPower: i know the feeling |
17:21.07 | Micetto | 18:19:52 16/03/2006 DEBUG[2659]: channel.c:797 channel_find_locked: Avoiding initial deadlock for 'Zap/1-1' |
17:21.15 | SpaceBass | n00b would be "can I use my usb modem with asterisk as an answering machine?" |
17:21.17 | ManxPower | Micetto, if you flood the channel with crap I'm going to bitchslap you. |
17:21.23 | justinu | lol |
17:21.39 | Micetto | ManxPower: sorry ^_^' |
17:21.45 | ManxPower | ~pb |
17:21.50 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
17:21.51 | Zeeek | how do I know if bitchslap() is installed on CLI? |
17:22.01 | justinu | show application bitchslap |
17:22.03 | justinu | n00b |
17:22.04 | justinu | :P |
17:22.09 | Micetto | eheheh |
17:22.13 | [TK]D-Fender | Zeeek : its all about the arguements ;) |
17:22.23 | Zeeek | I can't read the description, it's in dark green! |
17:22.30 | SpaceBass | LOL |
17:23.21 | ManxPower | Here's the basic idea. If Asterisk is waiting for DTMF then it can detect a fax. If it's not listening for DTMF, then it can't detect a fax. |
17:23.55 | SpaceBass | then wouldn;t it stand to reason that if an IVR answered and faxdetect=incoming it would pick it up? |
17:24.10 | Micetto | ManxPower: ok |
17:24.17 | bkw_ | ManxPower, not entirely true |
17:24.23 | bkw_ | you can play a file and do fax detect |
17:24.27 | Zeeek | SpaceBass not only an IVR but a human as well |
17:24.30 | bkw_ | without the need to wait or even think about DTMF |
17:24.33 | ManxPower | SpaceBass, Yes, that should work fine as long as you have an exten => fax AND you are thinking of "incoming" from the perspective of Asterisk |
17:24.47 | Zeeek | I've answered the phone only to hear the fax tone and then have asterisk hang me up to recv the fax |
17:25.07 | Zeeek | also works in vmail |
17:25.09 | SpaceBass | I've tried both...answering and sending to IVR...in all instances asterisk never detects it |
17:25.18 | ManxPower | bkw_, I was not aware of that. I use NVFaxDetect. |
17:25.26 | bkw_ | faxdetect needs to be on btw |
17:25.26 | SpaceBass | and Ricoh is probably tried of me using their faxback to test :) |
17:25.30 | bkw_ | its not usually on by default |
17:25.32 | Zeeek | are you using genuine Inte^h^h^h Digium hardware? |
17:25.37 | ManxPower | SpaceBass, using ZAP interfaces, right? |
17:25.37 | Micetto | and NVFaxDetect works ? |
17:25.42 | SpaceBass | ManxPower, yep |
17:25.48 | bkw_ | faxdetect=incoming |
17:25.51 | ManxPower | Micetto, yes. |
17:25.55 | bkw_ | yes that should do it |
17:25.58 | Idle | does anyone have a recomended place to buy digium hardware? I need 2 more fxo and 1 fxs module. |
17:26.08 | Zeeek | from digium |
17:26.08 | ManxPower | SpaceBass, BEFORE the channel = lines, right? |
17:26.09 | SpaceBass | Idle, did you check ebay |
17:26.16 | SpaceBass | leme look |
17:26.17 | Zeeek | or voipsupply |
17:26.26 | Idle | SpaceBass: not yet, I would like to buy it new |
17:26.27 | Micetto | so....it's possible that the cause of my problem is that I send a fax from an apple iBook ? |
17:26.32 | Idle | Zeeek: ok |
17:26.46 | ManxPower | Micetto, I don't know. Does the apple iBook send the fax tones when calling? |
17:26.47 | Zeeek | digium has a stroe online |
17:26.52 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
17:27.05 | Zeeek | warning, many fax machines do NOT work with spandsp at all |
17:27.09 | Idle | oh haha, |
17:27.10 | PakiPenguin | evening |
17:27.14 | SpaceBass | ManxPower, yeah....its before the channel => <line> |
17:27.17 | Zeeek | however, fax detection works even when the faxes don't |
17:27.22 | Micetto | ManxPower: I don't know....^_^' |
17:27.24 | Idle | Zeeek: I was lookingat the find a reseller, didn't realize they had a link at the right |
17:27.25 | ManxPower | Zeeek, We have found that recent spandsps work is almost all fax machines. |
17:27.27 | Idle | Zeeek: thanks man |
17:27.36 | fourcheeze | can asterisk hand off SIP calls to other hosts? |
17:27.38 | Zeeek | ManxPower not here |
17:27.48 | Zeeek | although recent = ? |
17:27.56 | ManxPower | Micetto, if the caller doesn't send fax tones then there is no way for Asterisk to know if it's a fax. |
17:28.00 | Zeeek | I'm using asterisk 1.2 and whatever the spandsp was then |
17:28.05 | SpaceBass | what I CAN do is pick up the call and transfer it to my fax extension...that works fine |
17:28.11 | ManxPower | Zeeek, "recent = in the past 3 months" |
17:28.17 | SpaceBass | of course the other problem I have is that faxes are coming in as blank pages |
17:28.18 | Zeeek | nope. Not me |
17:28.19 | *** join/#asterisk flynux (i=v8hy3c1@cl-8.bru-01.be.sixxs.net) |
17:28.21 | SpaceBass | but that seems unrelated |
17:28.22 | fourcheeze | if a call comes into asterisk box A for box B, how do I hand that off so that box A plays no further part - would that be a 302 ? |
17:28.25 | fourcheeze | (sip) |
17:28.26 | ManxPower | SpaceBass, do you year fax tones when you pick up the call? |
17:28.34 | SpaceBass | ManxPower, oh yeah! |
17:28.57 | Micetto | ManxPower: "fax tones" = "beep....beep.....beep....", right? |
17:29.35 | ManxPower | fourcheeze, that will happen by default if all legs of the call are using the same codec, if you don't have oddball options to the Dial command, if no leg of the call is NAT'd, and you don't have canreinvite=no. Remember SIGNALING will still go thru box A, just AUDIO will go direct. |
17:30.29 | SpaceBass | ManxPower, I hear the tones and I've let it beep a few times...Asterisk doesnt even make an attempt (at least according to the CLI) |
17:31.06 | fourcheeze | ManxPower: but how do I actually do that - is it just a Dial() ? |
17:31.09 | ManxPower | exten => s,6,Answer |
17:31.09 | ManxPower | exten => s,7,Ringing |
17:31.09 | ManxPower | exten => s,8,WaitExten(3) |
17:31.14 | Zeeek | SpaceBass you do have app_rx_fax installed? |
17:31.17 | ManxPower | this is a copy from my extensions.conf |
17:31.19 | justinu | anyone running opera and gaim at the same time on windoze? |
17:31.26 | Zeeek | FLOOD! FLOOD! |
17:31.27 | asterboy | holy crap, fixed that pesky problem with rx transmission failure on ZAP FXO...dam rxgain=1.0 was killing it. |
17:31.33 | SpaceBass | actually...my dring solution worked before I upgraded my A@H.... I need to study up on zapata syntax.... see if I can figure it out again |
17:31.43 | Micetto | AH!!!!!! |
17:31.50 | asterboy | dam that sounds geeky when I read what I just wrote. |
17:32.00 | Micetto | but exists a simulator (or configuration checker) for Asterisk ? |
17:32.07 | asterboy | wonder if I could use that as a pickup line? |
17:32.09 | justinu | asterboy: welcome to the club |
17:32.16 | [TK]D-Fender | Micetto : reality.sh <- |
17:32.46 | asterboy | Hey baby, maybe I can stick my FXS into your FXO and make a channel. |
17:32.46 | mutilator | anyone here ever worked in tech support? |
17:32.57 | *** part/#asterisk SWAT (n=SWAT@dsl159-68-100.fastxdsl.nl) |
17:32.58 | justinu | in a different life, yes |
17:32.58 | Zeeek | yes all day |
17:33.01 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
17:33.12 | mutilator | what do you guys do with support tickets that ya don't closed before your workday is over |
17:33.19 | Micetto | [TK]D-Fender: ahahaha |
17:33.28 | mutilator | stay til it's done or pass it off? |
17:33.39 | Zeeek | ask my provider: I'm still waiting for an answer five days later! |
17:33.41 | asterboy | or just drop it to the floor. |
17:34.05 | asterboy | no...pass it to some guy in India working for 2 cents an hour. |
17:34.53 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
17:34.57 | asterboy | 8 trillion and climbing! |
17:35.02 | jsharp | Pick it up tomorrow morning. It'll still be there. |
17:35.12 | Hmmhesays | 2 cents an hour, thats a deal |
17:35.13 | Darwin35 | good morning all my fellow asterisk whores |
17:35.16 | asterboy | lol |
17:35.38 | ManxPower | mutilator, Huh? We have work tickets that are almost a year old. |
17:36.11 | ManxPower | "user wants us to rewrite the Asterisk voicemail system so it's dumbed down enough for the user to understand." |
17:36.27 | mutilator | ManxPower: thats sad.. |
17:36.29 | Hmmhesays | dumded down? |
17:36.39 | Darwin35 | whats a flesh light |
17:36.42 | asterboy | press 1 for dumb down mode |
17:36.50 | ManxPower | The users don't like all those menu options when they are checking their voicemail, sending email via Blackberry, and driving down the freeway at 70mpg all at the same time. |
17:37.04 | asterboy | I like the voicemail that says, "Dial" |
17:37.11 | [TK]D-Fender | 70mpg? WOW, what kind of cars are they driving?! |
17:37.12 | Zeeek | "if you want to hear it, do nothing. Else hangup" |
17:37.27 | Hmmhesays | g must be a new unit of measurement for length |
17:37.33 | asterboy | anyone actually have a phone you can Dial? |
17:37.37 | mutilator | ya |
17:37.41 | tzafrir | gallon? |
17:37.41 | ManxPower | Hmmhesays, yes, it stands for "irc inches" |
17:37.49 | mutilator | wind used to blow through the dorm hallways at 30+mpg |
17:37.51 | [TK]D-Fender | <- putting the "smart" back into smart-ass :) |
17:38.09 | [TK]D-Fender | mutilator : Wow, thats a lot of BEER I'm sure ;) |
17:38.13 | Hmmhesays | [TK]D-Fender took me two hours to get that ugly kid joe solo down |
17:38.20 | tzafrir | But then again, it could be a plain "gee" |
17:38.27 | Hmmhesays | 1 hour to fake it half assed |
17:38.58 | Darwin35 | ok I still need a good realtime persont o help me move my home plan to real time |
17:38.59 | [TK]D-Fender | Hmmhesays : not bad... |
17:39.13 | Micetto | so...another fantastic question for you....!!!! ^.^ |
17:39.41 | konfuzed | slePP, are you round today |
17:39.42 | Hmmhesays | i definatley need to sell the les paul though |
17:39.47 | Micetto | beroNet driver can be used with junghanns cards ? |
17:39.50 | Zeeek | yes and no |
17:39.52 | sevard | weird, when i dial my own extension and press * and enter my voicemail password i get in, but when I dial my *98 extension and enter my extension and password it says login incorrect. |
17:40.48 | [TK]D-Fender | sevard : mayeb you should look and see what the difference is in how VoiceMail is being called.... |
17:41.16 | *** join/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr) |
17:42.01 | *** part/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr) |
17:42.07 | *** join/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr) |
17:42.43 | *** part/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr) |
17:42.59 | justinu | Service Impact Statement |
17:42.59 | justinu | Dropped calls and possible busies and All Circuits Busy recordings due to facilities being down. Fiber cut between Pittsfield and Toledo, Ohio, Trucks are enroute and no ETR. |
17:42.59 | justinu | Affected Locations |
17:43.00 | justinu | Affected Rate Centers: BIRCH RUN FRANKENMTH AUBURN FAIRGROVE BAD AXE BAY CITY ROSEBUSH COLEMAN MIDLAND AKRON SAGINAW FREELAND ST HELEN AUBURN LINWOOD WESTBRANCH OSCODA REESE ST CHARLES VASSAR MIDLAND |
17:43.20 | mutilator | wha no? |
17:43.49 | puzzled | justinu: no redundancy? |
17:43.58 | mutilator | where ya see that? |
17:44.02 | Zeeek | the trucks are to carry the packets? |
17:44.18 | sevard | [TK]D-Fender: I just saw an error message, vm_execmain: Couldn't read username |
17:44.36 | justinu | zeek: i think the trucks are enroute to repair the fiber cut :) |
17:44.42 | mutilator | i suggest avian carriers |
17:44.43 | justinu | that alert came from Level3 |
17:44.45 | Zeeek | oh, that won't do at all |
17:44.49 | mutilator | least that has an rfc |
17:45.04 | Zeeek | voIP with flu |
17:45.13 | puzzled | lol |
17:45.21 | [TK]D-Fender | sevard : I said how it is being CALLED. Look at the exec line..... |
17:45.21 | mutilator | and i think there is an rfc for and QoS over avian carrier aswel |
17:45.30 | justinu | Summary |
17:45.31 | justinu | UPDATE - Regarding Michigan Facilities Down - The affected OFF NET Transport Fiber technicians are on site excavating, the splicer truck has arrived, no ETR. |
17:45.51 | Zeeek | what will they be having for lunch? |
17:46.15 | mutilator | wonder if i got one from misdig |
17:46.45 | mutilator | <PROTECTED> |
17:46.47 | mutilator | so i'm good |
17:46.50 | Zeeek | Topic "If you are using AAH or AMP, please say so right away" |
17:46.57 | justinu | heh |
17:48.07 | Zeeek | sevard it means vmail is waiting for you to identify your number |
17:48.09 | xachen | AAH and AMP just suck period |
17:48.17 | SpaceBass | i'd disagree |
17:48.28 | Zeeek | your mouth would have to be reallllly small to suck a period |
17:48.29 | justinu | ban |
17:48.40 | konfuzed | geez some people. I just found out this guy had his switch unplugged since sturday and was just bitch at me under his breath like its my fault |
17:48.43 | xachen | tbqh I don't think users should be using a broken down control panel to run software that is way advanced |
17:49.10 | Zeeek | they can do what they want but it's hard to help them sometimes |
17:49.10 | *** join/#asterisk powerchip (i=powerchi@197.80-202-229.nextgentel.com) |
17:49.11 | justinu | konfuzed: people are real assholes |
17:49.27 | xachen | Do you give a 12 year old who has never shot a gun before a 30/30 for their first shot? |
17:49.34 | konfuzed | its nearly enough to make me change carreers |
17:49.38 | *** join/#asterisk heison (n=heison@216.235.9.2) |
17:49.53 | xachen | Then you get #asterisk filled with people asking what a bloody extension is |
17:49.55 | mutilator | xachen: i first shot a muzzle loader.. |
17:49.59 | [TK]D-Fender | Hmmhesays : Funny think, I just took up iaido last week :) |
17:50.01 | SpaceBass | if you look at something like the 2.x versions of A@H...sure its a dummyed-down version and has its faults...but it allows a lot of people to get into asterisk and VoIP easily and learn...and that will lead to more hardware, more providers, more proliferation so the govt and ISPs can't screw us |
17:50.02 | mutilator | knocked my on my fscking ass |
17:50.04 | mutilator | heh |
17:50.08 | xachen | and how to implement one :) |
17:50.09 | konfuzed | im gonna go work in a chocolate factory |
17:50.15 | Zeeek | xachen be fair - it's usually more like "I read the section about contexts" |
17:50.30 | Zeeek | when they really didn't |
17:51.02 | tzanger | wow Vonage is spamming now |
17:51.05 | SpaceBass | xachen, I think there should be a asterisk for newbies or asterisk@home channel....I even tried to start one....i agree there should be a diffrent place for experts vs hobbiests |
17:51.10 | xachen | yeah, you get lazy ppl who can't be bothered to learn a drag and drool control panel either :p |
17:51.18 | heison | hello [TK]D-Fender... problems with a couple new 7940 phones, can't get the stupid thing to talk SIP |
17:51.36 | mutilator | anyone try out them new grandstream gxv-3000's? |
17:51.39 | mutilator | video phones |
17:51.43 | [TK]D-Fender | SpaceBass : A@H isn't even "hobbyist" its "I don't want to learn any other details and just want to practice my clicking finger" |
17:52.01 | [TK]D-Fender | heison : Sorry, never worked on Cisco's personally.... |
17:52.08 | konfuzed | ya know the cacao bean has therapuetic levels of magnesium and MAO uptake inhibitors |
17:52.09 | xachen | yeah and when you get a desync issue |
17:52.11 | xachen | your done :) |
17:52.17 | heison | [TK]D-Fender: okay, thx |
17:52.32 | Hmmhesays | [TK]D-Fender: I use a@h for small biz pbx installs |
17:52.44 | konfuzed | then I could maintain a smile for the assholes complaining theres not enough sugar in their chocolate |
17:52.44 | xachen | I consider * hobbyist if you compile it from scratch, write your own extensions macros agi scripts tec. |
17:52.50 | SpaceBass | [TK]D-Fender, I'd totally disagree with that...I'm hobbiest and A@H was my launching pad |
17:53.04 | Zeeek | look where it got you :) |
17:53.14 | salviadud | we're mostly hobbyists then |
17:53.16 | SpaceBass | I'v learn a fair ammount of dial plan syntax now, but I never would have been able to if I didn't have something that worked out of the box to start with |
17:53.20 | salviadud | i write my own extensions... |
17:53.26 | heison | does anyone here have experience with 7960/7940? I'm trying to convert from MGCP to SIP (following the Cisco docs), and the phone doesn't want to talk SIP, it thinks it's still MGCP and tries to contact the CM for TFTP |
17:53.30 | SpaceBass | Zeeek, thanks for the vote of confidence |
17:53.31 | xachen | yeah, i started from the console :) |
17:53.35 | xachen | started with cvs head |
17:53.40 | Micetto | bye bye |
17:53.44 | Micetto | I'm going away |
17:53.44 | Micetto | :)= |
17:53.48 | Micetto | =:) |
17:53.50 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
17:53.54 | Darwin35 | grrr |
17:53.56 | *** join/#asterisk stoffell (n=stoffell@d51A4D12C.access.telenet.be) |
17:53.56 | Zeeek | Not only did I learn asterisk but I never had installed linux before |
17:54.00 | SpaceBass | by this logic X was the worst thing to happen to unix/linux |
17:54.01 | salviadud | my approach was to SEE a config file, then edit it. now i make my own |
17:54.01 | Hmmhesays | i started my linux and asterisk ventures at the same time |
17:54.15 | Zeeek | Not that I remember 90% of what that took, but I got a lot of help here |
17:54.26 | salviadud | what's wrong with X? |
17:54.27 | exonic | I'm writing a manager API ncurses interface. I am trying to find a way of getting 'detailed info' on channels. |
17:54.41 | SpaceBass | salviadud, its for lazy people who cannot be bothered to use the shell |
17:54.46 | salviadud | don't you like neat graphics? i can't watch pr0n without x dude! |
17:54.49 | Zeeek | I never was able to finish the asterisk make on solaris 10 the other day |
17:54.50 | [TK]D-Fender | SpaceBass : Do you actually use A@H with the knowledge you've gained since? For all the control you give up to it there isn't much "in-between except for special IVR's at best.... |
17:54.56 | Hmmhesays | it would be nice to have a web interface for lcdial |
17:54.56 | justinu | exonic: i got a little bit of help from AMI yesterday |
17:54.59 | stoffell | oej, can i ask you somethin' in private? |
17:55.02 | salviadud | i'm a slackware user, the console is my best friend of course |
17:55.04 | Zeeek | TTY pr0n |
17:55.06 | SpaceBass | and since there is xchat the linux channels get flooded with nubes asking how to run itunes on linux |
17:55.11 | Zeeek | much more left to the imagination |
17:55.11 | salviadud | but when it gets down and dirty |
17:55.11 | justinu | exonic: turns out, if you set async=true in an originate action, it'll return you the channel name, and unique ID |
17:55.17 | Darwin35 | my xkey keyboard stoped working |
17:55.21 | Darwin35 | grr |
17:55.22 | exonic | justinu, hey, how's it going |
17:55.26 | [TK]D-Fender | salviadud : Hallelujah! |
17:55.29 | Darwin35 | I now have to reprogram in |
17:55.35 | exonic | justinu, hmm. good to know |
17:55.35 | SpaceBass | [TK]D-Fender, I still use it b/c my box is working fine and all I ever do is add/remove the occasional trunk and thats just easy with freepbx |
17:55.46 | Hmmhesays | i need my Itunes damnit |
17:55.52 | SpaceBass | [TK]D-Fender, my next box in the future will probably be from scratch, but I'll still run freepbx |
17:55.57 | salviadud | i don't like it when people diss on X |
17:55.59 | justinu | exonic: here's my latest dilemma... i send out the originates, but I don't get the "OriginateSuccessEvent" until the call is actually answered. |
17:56.01 | exonic | justinu, I am I'd like to get detailed channel variables if possible. |
17:56.03 | salviadud | i use fluxbox and make it look cool |
17:56.04 | Hmmhesays | pffftttt |
17:56.05 | oej | stoffell: go ahead |
17:56.08 | justinu | so I can't terminate the call until it's answered :( |
17:56.09 | xachen | fluxbox is the best :) |
17:56.13 | Zeeek | the new PalyStation comes with asterisk pre-installed |
17:56.20 | salviadud | got my own menu, lots of shortcuts |
17:56.21 | jsharp | xachen: That just made me snort my Pepsi. |
17:56.22 | SpaceBass | I'm just saying things like A@H and amp/freepbx have a place and they will ultimatly benifit everyone |
17:56.28 | Hmmhesays | xachen: you see that family guy episode where the crippled prof was talking dirty to his crippled wife? |
17:56.32 | exonic | justinu, hmm.. You can of course track the channel state. 'Up' , 'Ringing' etc |
17:56.33 | xachen | yup :) |
17:56.34 | [TK]D-Fender | SpaceBass : then again what value does that have to truely understanding * when all you're doing is letting it do the whole job for you? Not much of a hobby when you're sitting in the stands while AMP is in the driver's seat. |
17:56.34 | Hmmhesays | I agree with SpaceBass |
17:56.36 | SpaceBass | it will get more people using asterisk and that equals more $ |
17:56.40 | salviadud | and well, if im running asterisk i don't need it. that's all |
17:56.47 | asterboy | if you get it to work. |
17:56.53 | justinu | exonic: yeah - except that I have no way of knowing which channel I created... except by the context, which irritates me |
17:56.54 | konfuzed | Hmmhesays, itunes is just a branded mp3 player but ya got suckered into paying for each tune |
17:56.58 | SpaceBass | how many windows users edit the registery to change their desktop background? |
17:57.04 | Hmmhesays | konfuzed: it was a joke |
17:57.08 | SpaceBass | how man mac users spend all their time in pref files exiting XML? |
17:57.08 | justinu | even then, I can't match each originate action with the "new channel event" |
17:57.14 | exonic | justinu, but I thought async returned the unique id? |
17:57.21 | justinu | it does... when the channel is answered |
17:57.28 | exonic | justinu, ugh! |
17:57.29 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
17:57.39 | justinu | i think i should just modify that to give me the originate success when the call is proceeding |
17:57.42 | konfuzed | Hmmhesays, those itunes people have been triggering me lately ;^) |
17:57.45 | exonic | justinu, have you seen the Dial event, I'm not sure so what to do what that. Perhaps it can help you? |
17:57.50 | SpaceBass | I'm building a * box as a project at work....build from source, but I'm still going to put freepbx on it for day to day stuff |
17:57.54 | [TK]D-Fender | SpaceBass : I didn't say FreePBX was "bad" per-se, just completely bad if you are intending on using it as a launching pad to actually learning *. Maybe as a motivator to say "see we can make it work, so can you, only you'll have to start from scratch because understanding our crap will give you a TUMOR" |
17:58.01 | Hmmhesays | i still download my music illegally like everyone else |
17:58.09 | xachen | speech recongintion could be used to detect panting and change the tone of the Festival config parameters :p |
17:58.14 | Hmmhesays | freepbx pretty much kicks ass |
17:58.18 | Hmmhesays | for guys like me |
17:58.27 | justinu | exonic: i only get a dial event when my originate call hit's a "Dial" command in the dialplan |
17:58.30 | [TK]D-Fender | Hmmhesays : You on the new beta of it? |
17:58.37 | Hmmhesays | yeah, i love it |
17:58.40 | exonic | justinu, yeah, it doesn't hit Dial all the time? |
17:58.46 | Hmmhesays | on the flip side, I know whas going on under the hood too |
17:58.52 | justinu | no, because I originate a call, and send it to an IVR extens |
17:58.57 | exonic | justinu, ahh |
17:59.00 | SpaceBass | [TK]D-Fender, look, I agree that trying to learn traditional (and clean) dial plan syntax from reading amp's conf files is like learning french by reading strero instructions in french.... but it does enable people to get setup out of the box |
17:59.04 | [TK]D-Fender | Hmmhesays : what seperates it from the previous version? (Didn't notice much in the screen-shots) |
17:59.18 | Hmmhesays | oh its way prettier than amp 1.x |
17:59.20 | Hmmhesays | and modular |
18:00.01 | [TK]D-Fender | SpaceBass : completely true. and if that gets them running enough to feel confident in buying the HARDWARE to get started sure, but getting away from AMP is no easy matter, and learning FROM it, less so. |
18:00.04 | asterboy | My Dial plan won't let me simply pickup the phone and start dialing. I have to put something like, "_9." to dial 9 before I get dial tone. How do you change that??? |
18:00.09 | sevard | [TK]D-Fender: I figured out the problem My context was &#^$ed |
18:00.15 | Hmmhesays | amp will confuse the fark out of a nub |
18:00.17 | sevard | [TK]D-Fender: Do I have to create /var/spool/asterisk/voicemail/default/140/INBOX manually? |
18:00.24 | exonic | justinu, yeah, Quite the dilema. I don't think very much thought went into the manager api. |
18:00.26 | [TK]D-Fender | Hmmhesays : Is there a demo page I could cruise for it? Didn't see it a Coalescent's site previously. |
18:00.39 | [TK]D-Fender | sevard : I was pretty sure thats what it was |
18:00.42 | justinu | exonic: heh, nope... i think i'm just gonna have to patch the code to make it work for me. |
18:00.46 | Hmmhesays | yeah you can cruise an install I messed up, it doesn't actually work, but everything is there |
18:00.50 | SpaceBass | [TK]D-Fender, I'd agree with that....when I customize my dialplan its totaly outside of AMP's structure...i just use their 'custom' commands to drop into my own plan |
18:00.53 | [TK]D-Fender | sevard : the box should initialize itself... |
18:01.15 | asterboy | My Dial plan won't let me simply pickup the phone and start dialing. I have to put something like, "_9." to dial 9 before I get dial tone. How do you change that??? |
18:01.18 | Hmmhesays | hold on i'll link you |
18:01.23 | SpaceBass | but the bottom line, amp or no amp....I fucking hate faxing right now and I'm going to go Office Space on the next fax machine I see |
18:01.25 | [TK]D-Fender | SpaceBass : how much "custom" work have you done with it so far? Give me a sample. |
18:01.31 | konfuzed | AMP is the biggest motivator for all those asterisk manager project a likes to have gotten started |
18:01.51 | [TK]D-Fender | SpaceBass : Oh... yeah analog fax + * = pain... just gt POTS for that if you want to remain sane... |
18:01.54 | salviadud | office space, as in the movie? |
18:01.59 | salviadud | that movie rocks |
18:02.08 | SpaceBass | [TK]D-Fender, nothing extravigant.... i have a 2nd incoming context that routes work calls differently than personal...i have gotos that i use for followme dialing when I'm traveling abroad, stuff like that |
18:02.12 | SpaceBass | salviadud, yeah it does! |
18:02.15 | [TK]D-Fender | I *do* however intend on learning A@H + all its components as it will open up a new customer base for me. |
18:02.28 | exonic | I am having incredible trouble w/ faxes too. patched asterisk 1.2.4 w/ T.38 and can receive just fine but sending fails 1/8 |
18:02.31 | asterboy | Any examples showing how to put get an outside line without pressing a key first? |
18:02.39 | SpaceBass | [TK]D-Fender, I think there is a market for A@H installers |
18:02.44 | konfuzed | analog fax stuck on a pstn with plain fax machine |
18:02.50 | salviadud | pattern matching |
18:02.55 | Zeeek | asterboy dial ZAP/1 |
18:03.00 | asterboy | exonic, put the faxes onto HylaFax instead. |
18:03.06 | SpaceBass | asterboy, just don't require it in the dial plan :P |
18:03.09 | Hmmhesays | haha I toasted my apache2.conf on this box too it'll be a second |
18:03.12 | Zeeek | or whatever inferface the line is on |
18:03.12 | asterboy | thx Zeeek , trying... |
18:03.23 | [TK]D-Fender | SpaceBass : Agreed, and for business reasons I should not ignore it. I hate it when business ethics clash with my personal ethics :) |
18:03.44 | SpaceBass | [TK]D-Fender, exactly...for instance, I'm an MCSE |
18:03.48 | [TK]D-Fender | I'm a very "practice what I preach" kind of guy... |
18:03.49 | SpaceBass | I'd shoot myself if I could |
18:03.52 | exonic | asterboy, Hylafax? it's unknown to me. Does it allow SIP to sent faxes ? |
18:04.00 | [TK]D-Fender | SpaceBass : Try target practice first ;) |
18:04.06 | salviadud | MCSE? |
18:04.06 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
18:04.09 | salviadud | what's that? |
18:04.15 | SpaceBass | salviadud, microsoft slut |
18:04.24 | salviadud | hahaha! |
18:04.27 | [TK]D-Fender | salviadud : Microsoft Certified System Engineer. |
18:04.31 | salviadud | you poor bastard |
18:04.47 | SpaceBass | I don't even work in that copacity anymore...but it comes in handy |
18:04.55 | Zeeek | without microsoft there would be no linux |
18:05.04 | konfuzed | ok 4 active lines in the house (not mine but somebody wondering why their phone bill is so high) so keep the analog fax line active and put dsl on it and an x100p for fail over to pstn |
18:05.05 | [TK]D-Fender | or : Multiple-Choice Sysadmin Exam ;) |
18:05.10 | konfuzed | yeah that should do it |
18:05.23 | SpaceBass | LOL |
18:05.36 | [TK]D-Fender | "C" baby! All the way to 80% !!! |
18:05.38 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
18:05.48 | SpaceBass | They don't tell you % anymore...just pass/fail |
18:05.57 | konfuzed | only faxes call that number so it wont have to be forwarded or answered by asterisk at all then |
18:06.00 | salviadud | windows is terrible with filenames, file permissions |
18:06.09 | salviadud | and well... spyware |
18:06.09 | SpaceBass | which sucks b/c I was proud of my % for a week or so that I cuared |
18:06.10 | [TK]D-Fender | God I'm so good its ~SCARY~ |
18:06.28 | asterboy | pwd |
18:06.28 | asterboy | ls |
18:06.37 | Zeeek | cd |
18:06.39 | asterboy | slol |
18:06.42 | asterboy | ifocnfig |
18:06.52 | Zeeek | ipconfig /ALL |
18:06.58 | SpaceBass | I was pretty anti windows and MS when I switched to OS X recently....but it has its pluses |
18:07.12 | salviadud | so, SpaceBass, who do you need to *** to get some source code in microsoft? |
18:07.16 | asterboy | ipconfig! how dare you! |
18:07.27 | salviadud | yeah, wmv videos man |
18:07.33 | salviadud | i use windows for pr0n |
18:07.36 | Zeeek | I have to see my ip address |
18:07.37 | salviadud | and that's it |
18:07.44 | SpaceBass | salviadud, I recently had to sign a NDA with MS...it involved a LOT of lube |
18:08.01 | justinu | but you liked it |
18:08.08 | SpaceBass | osx for porn....all the picts, none of the spyware |
18:08.24 | konfuzed | then keep the main house phone number line active with dsl aswell with x100p and asterisk actually answer that line |
18:08.38 | ManxPower | I thought MS didn't believe in lube. |
18:08.57 | SpaceBass | ManxPower, we had our lawyer negoicate it....my company respects me like that |
18:09.06 | salviadud | who has a xbox 360? |
18:09.16 | SpaceBass | seriously...this was a sit down with MS lawyers and ours just over a NDA that no one cares about |
18:09.28 | konfuzed | and then move the other two lines to a VOIP provider as 2 DIDs and then drop the kids pstn line and the business pstn line |
18:09.38 | Zeeek | the .NET version of asterisk 2.0 ? |
18:10.08 | salviadud | you think asterisk might get succesfully ported to windows? |
18:10.09 | Zeeek | konfuzed if you have only DSl you don't wanna drop the business line |
18:10.16 | konfuzed | i wanted to drop the fax line cause its not really used but |
18:10.27 | Zeeek | only if windows is successfully ported to windows first |
18:10.31 | SpaceBass | * will run on windows with cygwin |
18:10.42 | salviadud | cygwin is huge |
18:10.50 | SpaceBass | i dont care for it myself |
18:11.02 | salviadud | i rather use linux than cygwin |
18:11.07 | Zeeek | but it's still in love with you |
18:11.11 | salviadud | i can't select text on cygwin |
18:11.13 | xachen | cygwin is real slow |
18:11.17 | salviadud | if asterisk would run on putty |
18:11.19 | salviadud | that be awesome |
18:11.37 | mutilator | huh? |
18:11.40 | konfuzed | well i figure the the voip provider DID can be set to forward to house pstn number if asterisk doesnt pick up |
18:11.41 | salviadud | although, that is practically impossible |
18:11.44 | SpaceBass | i have one xp box left here at home...use it for outlook and thats it |
18:12.00 | Zeeek | hardware is so cheap, just put asterisk on a box and talk to that box from whatever system you like, even windows |
18:12.15 | salviadud | yeah, windows is the OS for grandmas, teenage girls and lawyers |
18:12.20 | SpaceBass | thats what I did.... |
18:12.34 | Darwin35 | ok now i am pissed why does realtime vm not work with pgsql |
18:12.39 | SpaceBass | got a 2ghz dell optiplex off ebay for like $75...runs * great! |
18:12.58 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
18:13.00 | salviadud | hey |
18:13.11 | salviadud | i got a prank call i did to a rehab clinic |
18:13.17 | salviadud | mixmonitor rulez! |
18:13.22 | salviadud | who wants it? |
18:13.26 | konfuzed | well crap theres 4 pstn lines in the house one just for fax and one double charged as a business line. i figure leave 2 pstn and move 2 over to VPBX provider and make it a hybrid setup |
18:13.26 | justinu | bueno? |
18:13.51 | salviadud | yes amigo friend platano banana |
18:14.17 | salviadud | well, any takers? |
18:14.31 | salviadud | it's about 1.7 megs big |
18:14.32 | justinu | post the link |
18:14.42 | salviadud | errr, no link |
18:14.49 | salviadud | i got the file |
18:14.55 | justinu | uh, upload it to a webserver |
18:15.13 | konfuzed | cp file ~/pbulic_html/. |
18:15.14 | salviadud | like, which one? |
18:15.20 | Zeeek | http://yousendit.com |
18:15.33 | Zeeek | send yourself the link and post it |
18:16.00 | konfuzed | even pastebin lets you attach files |
18:16.29 | salviadud | can i attach a wav in pastebin? |
18:16.31 | *** join/#asterisk NexGen (n=me@adsl-70-135-6-65.dsl.tulsok.sbcglobal.net) |
18:16.57 | konfuzed | salviadud, cp path/to/files /var/www/htdocs/html/prank/. |
18:17.04 | konfuzed | ;^) |
18:17.22 | konfuzed | attach file is generic |
18:17.43 | NexGen | hey guys, have a small problem, when connected to my cli I get the following: |
18:17.43 | NexGen | Parsing '/etc/asterisk/manager.conf': Found |
18:17.43 | konfuzed | it rarely bothers to chack file types |
18:17.43 | NexGen | Connect attempt from '127.0.0.1' unable to authenticate |
18:17.48 | NexGen | i have verbosity set to 10 |
18:18.04 | SpaceBass | NexGen, got amp/freepbx ? |
18:18.08 | NexGen | but I have no way what is trying to connect on the server, any ideas? |
18:18.08 | pauldy | that just for the remote agi isn't it |
18:18.18 | NexGen | amp |
18:18.22 | SpaceBass | NexGen, thats what it is |
18:18.28 | pauldy | the pl file |
18:18.30 | NexGen | amp trying to connect? |
18:18.47 | SpaceBass | its amp connecting to the manager service |
18:18.48 | Darwin35 | anyone here doing realtime with pgsql ? |
18:19.09 | NexGen | ok so apparently the password is wrong in the manager.conf? |
18:19.10 | SpaceBass | just mysql here |
18:19.31 | SpaceBass | NexGen, yeah, most likely or in amp.con ... check out #freepbx |
18:19.50 | SpaceBass | see [TK]D-Fender I'm not a total waste! :) |
18:20.27 | *** join/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
18:20.28 | pauldy | op_server.pl |
18:20.45 | bancus | If my system clock is running slightly fast, would that cause the audio to skip a lot? |
18:21.03 | SpaceBass | system time? or overclocked? |
18:21.06 | bancus | system time |
18:21.12 | pauldy | amportal.conf |
18:21.13 | Zeeek | nah |
18:21.14 | SpaceBass | can't see why it would |
18:21.14 | bancus | seems to be a problem with my motherboard |
18:21.31 | bancus | If left unchecked, it runs about 30-60 seconds fast every 10 minutes. |
18:21.39 | [TK]D-Fender | .clear |
18:21.49 | pauldy | neat |
18:21.56 | bancus | I have it resyncing every minute until I can fix it, but the workaround wouldn't help within each minute. |
18:21.58 | Zeeek | why not sync it with a time server? |
18:22.05 | Zeeek | o |
18:22.08 | tsume | eww amp |
18:22.11 | tsume | suck0rx |
18:22.29 | pauldy | tsume: why? |
18:22.33 | bancus | But I'm having a weird issue where half of a word or more gets skipped every second or so. |
18:22.42 | bancus | It'll just skip ahead. |
18:22.44 | Zeeek | on the local network? |
18:22.47 | bancus | yeah |
18:22.56 | bancus | Happens when I check my voicemail. |
18:23.14 | bancus | I thought it might be a lag issue I'm having with my upstream SIP provider. |
18:23.16 | salviadud | alright!!! |
18:23.19 | salviadud | http://s49.yousendit.com/d.aspx?id=0UGEMIUD5PLD42370SSWF10WLX |
18:23.19 | *** join/#asterisk iGotNoTime (n=iGotNoTi@cpe-65-189-240-199.woh.res.rr.com) |
18:23.20 | Zeeek | any IRQ issues? |
18:23.20 | bancus | But then it started happening locally too. |
18:23.29 | bancus | I don't believe so, but how do I check? |
18:23.47 | salviadud | basically, im the guy that says hello |
18:23.49 | tsume | pauldy: its not a very good interface, also crashes in linux much |
18:23.50 | Zeeek | I forgot how! |
18:23.53 | bancus | heh |
18:24.08 | Zeeek | ls /int somebody jump in here |
18:24.15 | Zeeek | proc/int |
18:24.19 | Zeeek | something |
18:24.27 | iGotNoTime | salviadud, hello :) |
18:24.31 | pauldy | tsume: it crashes? |
18:24.43 | *** join/#asterisk Lino` (n=Lino@i577BC430.versanet.de) |
18:24.45 | bancus | <PROTECTED> |
18:24.51 | bancus | except with two rs |
18:24.58 | Zeeek | sounds about right |
18:25.09 | bancus | What am I looking for? |
18:25.10 | tsume | pauldy: yeah, well.. its flash. what to expect |
18:25.12 | Zeeek | look but don't paste |
18:25.18 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
18:25.19 | bancus | k |
18:25.20 | iGotNoTime | question.... I have installed * and it is running now.... isn't there a web GUI for it though? I have tried localhost but it says denied. Am I missing something? |
18:25.30 | Zeeek | muhahaha |
18:25.36 | tsume | all these GUI babies make me sick |
18:25.36 | bancus | iGotNoTime: there is no web gui in the default program |
18:25.37 | Zeeek | @hole |
18:25.40 | tsume | learn to use vi and linux |
18:25.45 | Zeeek | vi sucks |
18:25.45 | pauldy | amp is flash? |
18:25.47 | bancus | some 3rd party addons might |
18:25.53 | Zeeek | no FOP is flash |
18:25.58 | salviadud | not vi necesarily |
18:25.59 | tsume | pauldy: oh wait, I'm thinking og flash operator |
18:26.02 | salviadud | you can use pico |
18:26.02 | bancus | Zeeek: what am I looking for? |
18:26.09 | Zeeek | nano is even better |
18:26.14 | pauldy | ahhh ytea the flash operator pannel is flash |
18:26.16 | Zeeek | bancus IRQ shared |
18:26.35 | Zeeek | what is the Trash Operator Panel then? |
18:26.37 | iGotNoTime | tsume, I wish I had your knowledge, but unfortunately it is difficult to lean command line because those who do learn it well you know. :) |
18:26.39 | [TK]D-Fender | iGotNoTime : GUI's are a whole other world...... |
18:26.39 | bancus | How can I tell when something is shared? |
18:26.50 | tsume | iGotNoTime: it takes 5 minutes to set up ;) |
18:26.57 | iGotNoTime | thank you [TK]Fender :) |
18:27.02 | tsume | maybe I should make a GUI based drag n drop and sell it |
18:27.02 | salviadud | hey, you guys hear my prank yet? |
18:27.08 | iGotNoTime | tsume, yes it is running I said that :) |
18:27.08 | tsume | for asterisk |
18:27.18 | tsume | iGotNoTime: I mean fully setup ;) |
18:27.52 | pauldy | I know how to setup via command line and I preffer to use amp just because it makes some of the more complicated stuff rpetty damb simple |
18:27.52 | salviadud | its a prank done with asterisk... might i add |
18:28.05 | Zeeek | bancus : cat /proc/interrupts |
18:28.11 | bancus | Zeeek: done |
18:28.15 | iGotNoTime | I agree pauldy, why waste hours trying to learn all the commands when you can have them placed as a button :) |
18:28.25 | tsume | iGotNoTime: hours? |
18:28.28 | Zeeek | look at each line and see if there is more than one device |
18:28.30 | tsume | iGotNoTime: only takes a few minutes |
18:28.39 | tsume | reading helps |
18:28.43 | bancus | Yeah, trying to listen to my voicemail is almost like listening to a tape and repeatedly hitting FF. |
18:28.43 | iGotNoTime | tsume type help on your command line :) |
18:29.02 | iGotNoTime | there are alot of commands to remember |
18:29.02 | pauldy | tsume: no to learn all the commands you need or might need it can take a lot longer than a few minutes |
18:29.07 | tsume | iGotNoTime: I don't need help, I already know how to do everything in asterisk ;) |
18:29.13 | tsume | pauldy: not really |
18:29.16 | Zeeek | bancus - is there more than one device on lines? |
18:29.18 | bancus | 21 has ehcd_hcd and eth0, 22 has libata and ohcd_hcd |
18:29.21 | pauldy | yea really |
18:29.26 | iGotNoTime | tsume then refer to my above observation ;) |
18:29.37 | tsume | pauldy: when I first started asterisk, I looked at an extensions.conf and started programming |
18:29.49 | iGotNoTime | tsume I have done that |
18:29.52 | pauldy | and you learned everything in the first 5 minutes |
18:29.52 | iGotNoTime | it is setup |
18:29.59 | tsume | pauldy: yes |
18:30.09 | pauldy | hahaha |
18:30.10 | Zeeek | I took a slightly different approach |
18:30.21 | salviadud | asterisk is great... i took the same aproach as tsume |
18:30.24 | Zeeek | I read the old asteriskdocs.org manual about 10 times |
18:30.41 | Zeeek | what I didn't get, I re-read until I figured it out |
18:31.02 | salviadud | the new book from o reilly kicks mayor ass |
18:31.02 | Zeeek | then I imposed my experiments on two innocent women for one year |
18:31.19 | salviadud | it's based on 1.2 |
18:31.36 | salviadud | killer book, good jokes, excellent editing |
18:31.46 | Zeeek | give us a joke from it |
18:31.46 | iGotNoTime | I like that Zeeek, I never said I did not read, and I am sure Pauldy has read as well. That does not make us ignorant because I ask about a GUI though does it? |
18:31.53 | salviadud | download my file! download it in the name of asterisk! |
18:31.56 | salviadud | http://d49.yousendit.com/F/0UGEMIUD5PLD42370SSWF10WLX/narconon.wav |
18:32.00 | Zeeek | I never said that |
18:32.10 | SpaceBass | I think Bill O'Riley should do an Asterisk book....Asterisk, No Spin Zone |
18:32.10 | *** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe) |
18:32.15 | Hmmhesays | lol |
18:32.16 | Zeeek | heh |
18:32.20 | iGotNoTime | SpaceBass, yes he should! |
18:32.20 | SpaceBass | im sure it'd be fair and balenced |
18:32.35 | Zeeek | fox@home |
18:32.52 | Zeeek | Bush Control Panel |
18:32.56 | iGotNoTime | LMAO |
18:33.03 | salviadud | they joke about asterisk couldn't get so popular at the start because it wasn't a way to get pr0n much faster |
18:33.16 | Zeeek | oh, but it is! |
18:33.23 | Zeeek | voice pr0n |
18:33.25 | iGotNoTime | haha |
18:33.36 | SpaceBass | Chapter 3 - Legal Wire Taps using FOP |
18:33.42 | Zeeek | someone has a recording of Allison drunk |
18:33.48 | tsume | hmm |
18:33.51 | SpaceBass | Chapter 4 - if it cashes, blame femao |
18:33.53 | SpaceBass | Zeeek, no shit? |
18:33.55 | Zeeek | saying "ok you stupid asshole, I said hit one" |
18:34.00 | salviadud | the voice of digium? |
18:34.01 | SpaceBass | s/femao/fema |
18:34.07 | salviadud | do you think she's hot? |
18:34.19 | tsume | and not providing person to person connections but service :D |
18:34.21 | SpaceBass | salviadud, I have to assume so, its the only thing holding comedian mail together |
18:34.29 | Zeeek | I'm more attracted by Vulcan women personally |
18:34.41 | salviadud | pointy eers |
18:34.50 | Zeeek | yeah |
18:34.50 | asterboy | For some reason, I have to dial a # to get a dial tone from my FXO port, THEN I can dial to another number...how can I make it so that I just have to dial the number end of story? |
18:34.51 | salviadud | vulcan women are just the same as zelda |
18:35.08 | Zeeek | asterboy what phone? |
18:35.16 | asterboy | Polycom IP600/500 |
18:35.17 | salviadud | maybe you like the fact that they are very logical |
18:35.22 | SpaceBass | salviadud, http://www.theivrvoice.com/ |
18:35.25 | Zeeek | look up digitmaps |
18:35.33 | asterboy | thnx |
18:35.39 | Zeeek | in Polycom |
18:35.45 | salviadud | "would you blow me?" - "yes, it is only logical" |
18:36.13 | Zeeek | ~seen digitmaps in Polycom |
18:36.24 | jbot | i haven't seen 'digitmaps in polycom', Zeeek |
18:37.12 | salviadud | she's canadian? |
18:37.18 | salviadud | she probably says aboot, instead of about |
18:37.26 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-20-208.slkc.qwest.net) |
18:37.33 | salviadud | not that i mind |
18:38.23 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
18:39.44 | salviadud | i think allison might have some trouble actually talking on the phone, they'd probably think its just an automated service |
18:42.09 | *** join/#asterisk ToTo (n=ToTo@host154-207.pool872.interbusiness.it) |
18:42.19 | tsume | how much would someone pay for a full fledged dial plan creator GUI? |
18:42.20 | *** part/#asterisk oracle^ (n=cam@unaffiliated/cameleons) |
18:42.21 | asterboy | http://www.freedomphones.net/polycom/files/docs/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf |
18:42.28 | [av]bani | polycom has no overlap dialing :< |
18:42.28 | tsume | for win,lin,osx ;) |
18:42.30 | asterboy | Admin guide has some info. |
18:42.39 | tsume | 49USD? |
18:42.53 | hypa7ia | tsume: make it flash or ajax, then it's croxxplatform by default :) |
18:43.12 | [av]bani | croxx! |
18:43.37 | hypa7ia | hehe |
18:43.39 | hypa7ia | yay spelling |
18:44.00 | hypa7ia | i have one of the weird swoopy microsoft keyboards at work |
18:44.07 | hypa7ia | totally throws off the spelling |
18:44.19 | Zeeek | asterboy just enter your patterns in the digitmap and the phone will dial as instructed |
18:44.35 | asterboy | checking... |
18:45.07 | bancus | Okay, I found a stopwatch to verify seconds against my clock |
18:45.14 | bancus | "sleep 10" resolves in about 5 seconds |
18:45.21 | bancus | so there's definitely something screwy going on here |
18:45.42 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
18:45.46 | Zeeek | stuff like [23]xxx to dial 2000-3999 right away |
18:45.49 | bancus | yay, bugreport about it |
18:45.52 | Darwin35 | where are the real time guys |
18:46.00 | jsaunders | Can * utilize a multiprocessor environment to any end? |
18:46.04 | jsharp | Yes |
18:46.12 | jsharp | More processors make it taste better. |
18:46.19 | jsaunders | Reheally. |
18:46.34 | hypa7ia | they improve the texture too don't forget |
18:46.44 | jsharp | * is multithreaded, so multiple processors make * happy. |
18:47.21 | jsaunders | So multiple low clocked dual core opterons would be preferable to oh say a single high clocked Athlon. |
18:47.23 | jsaunders | Beauty |
18:47.41 | bancus | just under $300 as I recall |
18:47.49 | bancus | cheapest dual-core proc I could find |
18:48.02 | justinu | beauty, eh? |
18:48.15 | asterboy | ok, so it looks like Polycom is setup with a default digit map that does not allow standard pickup and dial. |
18:48.26 | jsaunders | I'm lookin' at the opteron 240's, two of 'em. |
18:48.31 | Zeeek | what exactly is the result you want |
18:48.33 | asterboy | So I added xxxxxxx|xxxxxxxxxx|1xxxxxxxxxx |
18:48.47 | asterboy | I just want to pickup a line a dial, period. |
18:48.47 | ManxPower | asterboy, wow, that was stupid. |
18:48.59 | asterboy | how so? |
18:49.02 | justinu | hahah |
18:49.06 | jsaunders | heheh |
18:49.10 | ManxPower | asterboy, they are pretty much the same. |
18:49.39 | asterboy | the x's |
18:49.55 | Zeeek | every x is the same |
18:50.03 | justinu | i need my laptop to arrive before I can go on vacation |
18:50.26 | SpaceBass | I need a macbook pro |
18:50.33 | Zeeek | get a Dingleberry |
18:50.49 | asterboy | ok, I'm not understanding the idea then. |
18:50.52 | _Paulo_ | In Brazil collect calls are automated. There are a standard recording like "This is a collect call. To accept the call ...bla..bla..bla". The callee has 5 seconds to let the caller identify himself and accept or drop the call. |
18:51.01 | justinu | i considered the macbook pro |
18:51.02 | justinu | too big tho |
18:51.05 | asterboy | why does the default in the manual list, 9]xxxxxxxx |
18:51.10 | justinu | i ordered a thinkpad X60s |
18:51.30 | asterboy | I don't want 9 to dial out, just the number part. |
18:51.37 | _Paulo_ | Is there any application so I can recognise the collect call recording? |
18:51.37 | *** part/#asterisk mko-025 (n=korpim@p5498BD34.dip0.t-ipconnect.de) |
18:51.44 | Zeeek | how many different types of numbers? |
18:52.29 | asterboy | ie. 123-1234, 123-123-1234 or 1-123-123-1234 |
18:52.48 | asterboy | actually I just want it to dial no matter what number combo. |
18:52.58 | asterboy | Just pickup the god dam line and dial! |
18:53.23 | Zeeek | tune up the crank |
18:53.26 | asterboy | right now, I have to dial something first, then it gives a dial tone, then I can dial whatever I want. |
18:53.38 | Zeeek | that's not the phone, it's asterisk |
18:53.51 | tsume | _Paulo_: 5 seconds is much to transmit enough info ;) |
18:53.52 | asterboy | That's what I thought in the beginning. |
18:54.06 | asterboy | I've tried every combination in front of the dial plan. |
18:54.07 | tsume | _Paulo_: just keep calling back with parts of the sentence :P |
18:54.07 | Zeeek | so this is another case of @home? |
18:54.18 | asterboy | no @home |
18:54.29 | _Paulo_ | tsume, It used to be 10 seconds, they shortened to 5 seconds due to abuse... |
18:54.37 | Zeeek | well someone put the logic to wait for a digit in there |
18:54.40 | tsume | _Paulo_: still easy to abuse :) |
18:54.51 | asterboy | _9.,1,Dial(ZAP/1) - have to dial 9 first obviously. |
18:54.59 | tsume | _Paulo_: portuguese are fast talkers :P |
18:54.59 | Zeeek | yes? |
18:55.08 | asterboy | 1,1,Dial(ZAP/1) - gives me a fast beep |
18:55.13 | tsume | you should hear my motor mouth ;) |
18:55.24 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
18:55.25 | SpaceBass | ok.. SIP calls started coming in and looping, so I did a restart gracefully....never came back, so I bounced the box and now when I connect to the CLI i get an error about no realtime |
18:55.32 | SpaceBass | wtf...i didn't change a thing! |
18:55.39 | Zeeek | asterboy it's impossible for you to have read about dialplans and still be in the place you are now |
18:55.56 | asterboy | I know I'm missing something, just not sure what. |
18:56.14 | Zeeek | each technology has a slightly different Dial syntax |
18:56.27 | [TK]D-Fender | asterboy : if you want it to "just dial" then do "x.T" and it'll through EVERYTHING at it regardless of length. |
18:56.31 | _Paulo_ | tsume, the worse is that the market is distorted... |
18:56.32 | Zeeek | ZAp is ZAP/chan/Number or something like that |
18:56.53 | SpaceBass | ARRRRUUUUGGGGGGGG my entire box just shat itself and died for no reason....nothing changed |
18:56.54 | asterboy | ah |
18:57.05 | Zeeek | exten => 18005551212,1, Dial(ZAP/1/${EXTEN}) |
18:57.22 | ManxPower | Zeeek, don't put a space after the , |
18:57.30 | Zeeek | write a fixed extension like that and get it working. Then abstract it to _1800. |
18:57.30 | _Paulo_ | tsume, in the state owned telco model from the 70s the long distance calls used to subsidize local calls... |
18:57.40 | Zeeek | NO SPACES! I'm a bad typist |
18:57.42 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
18:57.53 | ManxPower | Zeeek, that's why you should COPY & PASTE |
18:58.05 | Zeeek | I was typing but I'm lousy at it :) |
18:58.16 | justinu | like copy/paste has never caused any problems :P |
18:58.17 | Zeeek | plus my mind isn't a perfect memory |
18:58.26 | _Paulo_ | tsume, long distance calls are still expensive despite intense privatization in the late 90s. |
18:58.29 | tsume | _Paulo_: yikes :/\ |
18:58.32 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
18:58.39 | tsume | _Paulo_: voip power :) |
18:58.41 | Zeeek | the POINT is, this is all discussed extensively on the wiki and in asteriskdocs.org docs for years |
18:58.45 | ManxPower | _Paulo_, you must not be in the USA |
18:59.16 | tsume | _Paulo_: voip will take over brazil, then they'll have to look for more ways to make money.. by destroying the rest of the rainforest |
18:59.26 | tsume | obliterating the world, murderers |
18:59.57 | tsume | I wish bush would attack brazil and save the worlds air ;) |
18:59.59 | Darwin35 | ok I have postgress setup and running and all the tables inthe db made |
19:00.11 | Darwin35 | now to get asterisk to work with it right |
19:00.18 | Hmmhesays | this should be interesting |
19:00.26 | Zeeek | asterboy you should take the time to read this: it's old but it has the answers to your question |
19:00.29 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
19:00.33 | Zeeek | <PROTECTED> |
19:00.48 | Zeeek | I said take the time! |
19:00.59 | asterboy | This just sucks: http://www.voip-info.org/wiki-Asterisk+howto+dial+plan |
19:01.02 | tamp4x | debug tells me nothing hmmhesays |
19:01.06 | tamp4x | oops |
19:01.08 | asterboy | can you make it any more vague |
19:01.08 | _Paulo_ | tsume, a squirrel could travel from cost to cost in EUA without touching the ground befor colonization. |
19:01.08 | tamp4x | anyone here use spandsp? when i load asterisk -vvvvvvvvv it stops loading when app_rxfax.so loads....any ideas why? |
19:01.10 | Zeeek | This says it all: "The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls" |
19:01.16 | Zeeek | duh! |
19:01.46 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
19:01.55 | tsume | _Paulo_: 35% in the past 20 years is too much |
19:02.08 | tsume | probably more if I look at the charts |
19:02.11 | Zeeek | then there's this gem: Calling channels with the Dial() application |
19:02.17 | GerbilNut | is it possible to capture everything going to the CLI for a period of time? |
19:02.19 | Zeeek | shit, it's all there |
19:02.34 | Zeeek | everything but the final 42 |
19:02.42 | Zeeek | which you then come here to find :) |
19:03.06 | ManxPower | GerbilNut, you mean like as is configured in logger.conf ? |
19:03.15 | ManxPower | or is it logging.conf? Anyway.... |
19:03.24 | GerbilNut | prolly so |
19:03.54 | Zeeek | ManxPower is your madrid presentation online anywhere? |
19:04.09 | ManxPower | Zeeek, no. I don't even know if I have it. |
19:04.44 | Zeeek | I think I'll be speaking later this year, I thought I might Stea^H^H^H borrow an idea or three |
19:04.48 | ManxPower | Oh! The astricon people got a copy of it, I think. |
19:05.20 | Zeeek | they must have posted the stuff |
19:05.29 | Zeeek | maybe no longer available tho |
19:05.59 | justinu | the function dial_exec_full is a fucking joke |
19:06.05 | oej | I'll make sure the Astricon presentations come back on line |
19:06.15 | *** join/#asterisk Gertrude (n=gert@chickenbones.bflony.adelphia.net) |
19:06.27 | Zeeek | there was some good stuff there |
19:06.48 | Zeeek | Matt from digium had some great intro programming stuff too |
19:06.52 | salviadud | i wish i could go to astricon...some day |
19:06.53 | Zeeek | IIRC |
19:06.54 | Darwin35 | hey Manx |
19:07.29 | justinu | yeah, we were promised an email w/ the presentation info in it |
19:07.32 | justinu | got zip |
19:07.41 | Zeeek | the t-shirt! |
19:07.46 | justinu | i did get spammed about asterisk book camp tho |
19:07.51 | justinu | so they obviously have my address |
19:07.56 | justinu | s/book/boot/ |
19:08.14 | Zeeek | FWD has been intensly spamming me about von. I had to threaten them |
19:08.41 | Zeeek | I told em I was gong to loop through all FWD numbers with a special message |
19:08.50 | justinu | lol |
19:08.51 | SpaceBass | i think my zaptel hardware just died |
19:09.06 | Zeeek | give it a plain pine box, nothing fancy |
19:09.33 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
19:09.35 | Darwin35 | send it back make them replace it |
19:09.44 | Zeeek | the fastest way to kill it is to plug an fxs into the wall |
19:09.48 | justinu | try a different pci slot |
19:09.52 | Darwin35 | call them everyday ill they do |
19:10.13 | SpaceBass | Darwin35, was a x100p clone |
19:10.24 | SpaceBass | $15 off ebay |
19:10.36 | Darwin35 | toss it in the trash |
19:10.38 | Zeeek | always buy two of them on ebay |
19:10.44 | SpaceBass | not going to lose sleep over it, except that its my offical, company paid for work number...and I'm going on vacation tomorrow and need to have my IVR working before I leave |
19:10.49 | Darwin35 | buy in bulk |
19:10.54 | SpaceBass | might just forward everything to a SIP trunk for the week |
19:11.09 | Darwin35 | then get a nmbr threw us |
19:11.29 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
19:11.30 | SpaceBass | im not supposed to be doing this kind of stuff...or my real job...im supposed to be working on my ipod mix for skiing |
19:11.36 | [TK]D-Fender | SpaceBass : Yup.... I'm about to be dependant on things like that. Getting Dry-loop DSL next week :) |
19:13.04 | SpaceBass | when I run ztcfg I get "unable to open /dev/zap/ctl" |
19:13.10 | SpaceBass | does that sound like hardware failure? |
19:13.20 | SpaceBass | dry-loop ? |
19:13.21 | [TK]D-Fender | SpaceBass : modprobe wcfxo |
19:13.38 | [TK]D-Fender | dry-loop DSL is DSL without a voice line behind it (no dial-tone) |
19:13.44 | SpaceBass | wcfxo not found..... |
19:13.48 | SpaceBass | [TK]D-Fender, gotcha.... |
19:13.58 | SpaceBass | i'd LOVE to get rid of my pots lines all togather |
19:14.14 | SpaceBass | i have FiOS for internet connectivity....just need to get my numbers ported |
19:14.43 | SpaceBass | can't do an insmod on wcfxo either |
19:14.58 | hypa7ia | ``i'm getting dsl on dry copper when i move |
19:15.02 | hypa7ia | we have it at my current house |
19:15.05 | justinu | oh, i got my dry DSL service about 2 weeks ago |
19:15.14 | justinu | waaay better than the shit cable modem service I had |
19:15.15 | *** join/#asterisk file[laptop] (n=jcolp@sjcc28x192.sjccnet.com) |
19:15.23 | SpaceBass | i had 'dry' dsl at one point...didn't know it was called that |
19:15.38 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
19:15.39 | SpaceBass | what the hell.... cannot even find the zaptel module anymore |
19:15.41 | hypa7ia | yeah they have started traffic shaping torrents on the cable provider up here |
19:15.43 | hypa7ia | sucks |
19:15.44 | SpaceBass | disapeared |
19:15.50 | [TK]D-Fender | SpaceBass : Because those stuck paying the telo for services they don't want are "all wet" :) |
19:16.06 | [TK]D-Fender | SpaceBass : recompile zaptel and try switching PCI slots. |
19:16.15 | SpaceBass | thats what I was thinking |
19:16.38 | SpaceBass | actually, I'm thinking that I'll finish the 3 hours of work I have, forward that pots number to my sip trunk and not worry about it for 2 weeks |
19:17.17 | file[von] | hiya |
19:17.18 | [TK]D-Fender | SpaceBass : and they can run on that fora while? |
19:17.32 | [av]bani | [TK]D-Fender: ever figure out how to disable polycom log uploads? |
19:17.33 | [TK]D-Fender | file[von] : Getting outt my old plac at the end of next week. |
19:17.38 | SpaceBass | [TK]D-Fender, I'll lose call waiting on on those numbers |
19:17.46 | file[von] | [TK]D-Fender: found somewhere? |
19:17.49 | [TK]D-Fender | [av]bani : No way yet it seems short of changing permissions (yuck) |
19:17.58 | iGotNoTime | [TK]D-Fender, do you have a link to the manual that Zeeder was talking about for * ? |
19:18.03 | [TK]D-Fender | file[von] : I was approved at least 2 weeks ago. |
19:18.24 | file[von] | [TK]D-Fender: excellent |
19:18.30 | [av]bani | [TK]D-Fender: that sucks. and still no overlap dialing either. |
19:18.31 | [TK]D-Fender | iGotNoTime : there is no "manual", but there are some good books, the first of which is .... |
19:18.33 | [TK]D-Fender | ~thebook |
19:18.34 | jbot | rumour has it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
19:18.43 | SpaceBass | now to find the zaptel source on a@H |
19:18.43 | [TK]D-Fender | [av]bani : Still livable in my books.... |
19:18.50 | powerchip | how can I call a ring grup? |
19:18.50 | [av]bani | [TK]D-Fender: still violation of rfcs :)) |
19:18.57 | [TK]D-Fender | SpaceBass : OMG... yeah.. that part could suck :/ |
19:19.10 | [TK]D-Fender | they're RFC's, not RULES ;) |
19:19.11 | SpaceBass | [TK]D-Fender, remember what I said about "when I build my next box...." |
19:19.15 | [TK]D-Fender | NO COMMENT!!!! |
19:19.28 | [av]bani | [TK]D-Fender: they are rules if you want to interoperate (rfc2833) |
19:19.30 | [TK]D-Fender | SpaceBass : When "sooner" meets "later"! |
19:19.39 | [av]bani | [TK]D-Fender: sip and rtp are rfcs also.. they are _rules_ |
19:19.39 | iGotNoTime | thanks :) |
19:19.41 | *** join/#asterisk kuku5 (n=kuku5@c-71-228-21-166.hsd1.il.comcast.net) |
19:19.51 | kuku5 | sup all |
19:19.51 | SpaceBass | ahhhhh /usr/src/zaptel |
19:19.54 | [TK]D-Fender | [av]bani : Yeah... and how far off is * in SIP compliance all by itself? |
19:20.15 | [av]bani | [TK]D-Fender: * does overlap dialing :)) |
19:20.16 | [TK]D-Fender | [av]bani : no, just increasingly "strong" suggestions ;) |
19:20.19 | SpaceBass | nothing a little make clean ; make install shouldn't fix |
19:20.41 | [TK]D-Fender | [av]bani : I'm talking SIP support.... not just an aspect of it in reference to other channel types... |
19:20.51 | SpaceBass | shit! remember what I did now....installed all the centos updates a few days ago...hadn't bounded asterisk since then |
19:21.04 | SpaceBass | damn, make failed |
19:21.09 | [av]bani | [TK]D-Fender: at least * is fixing its problems :)) |
19:21.09 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
19:21.15 | [av]bani | [TK]D-Fender: sip-b this summer, yay |
19:21.29 | [av]bani | [TK]D-Fender: now try to get polycom to implement overlap dial. lollerskates? |
19:21.49 | tamp4x | anyone here use spandsp? when i load asterisk -vvvvvvvvv it stops loading when app_rxfax.so loads....any ideas why? |
19:22.05 | [TK]D-Fender | [av]bani : "supposedly".... that's a BIG plus for inter-op for sure and drags things like the SPA's back into the realm of "suggestable" and elevates Polycom further still... |
19:22.46 | [TK]D-Fender | [av]bani : not forgetting that ther is an ACD patch for Polycom's signalling that could very well be merged in at that point. |
19:22.55 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
19:23.04 | asterboy | done digesting that link Zeek. |
19:23.56 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-118.sd.sd.cox.net) |
19:24.32 | asterboy | Now the only thing I can see that I may need to do is passing my number via the ${EXTEN} variable, into the Dial plan so I don't have to re type it. |
19:25.36 | asterboy | But how do I get past having to dial a # to execute the application? use of "s"? |
19:26.18 | asterboy | nope, just get fast beep. |
19:27.06 | willt[work] | was it ulaw that provided the best sound quality? |
19:27.31 | [TK]D-Fender | willt : the best quality suppoted by the PSTN, yes. |
19:27.54 | *** join/#asterisk ToTo (n=ToTo@host154-207.pool872.interbusiness.it) |
19:28.01 | fourcheeze | [TK]D-Fender: surely you can get better down pstn if you compress? |
19:28.09 | [TK]D-Fender | willt : At the same time as being pretty much the least load on * for converting for that purpose as well... |
19:28.18 | ms345 | Anyone know the proper timing settings for multiple pri's from the teclo? Do I set the span=1,1,0,esf,b8zs for my primary pri_cpe span and span=2,2,0,esf,b8zs for my 2nd pri_cpe span? I assume all my pri_net spans that go to my PBX, routers, and other cpe are span=<span#>,0,0,esf,b8zs. Does that sound right? I'm going off the material here http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax |
19:28.26 | asterboy | If I don't want any numbers, like pressing 9 for outside line, should not "s" execute the application? |
19:28.47 | asterboy | or "_."? |
19:28.51 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
19:28.56 | asterboy | pattern match anything |
19:29.05 | justinu | ms345: i think you're on track |
19:29.06 | [TK]D-Fender | asterboy : You are not running AMP are you? |
19:29.30 | asterboy | ~amp |
19:29.32 | jbot | rumour has it, amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
19:29.40 | ms345 | justinu - thx. I'll give it a try tonight after-hours. |
19:30.01 | asterboy | Since I don't know what AMP is, I'd have to say no. |
19:30.10 | [TK]D-Fender | asterboy : You just want to dial a # to place an outgoing PSTN call without a prefix, right? |
19:30.18 | asterboy | YES! |
19:30.35 | *** join/#asterisk kamranazeem (i=[U2FsdGV@203.135.40.97) |
19:30.38 | [TK]D-Fender | ast, then just use a pattern without a 9 in front. thats what I've recently adapted mine to. |
19:30.49 | [TK]D-Fender | asterboy : pastebin your extensions.conf and I'll take a look for you. |
19:30.55 | asterboy | ok thx |
19:31.07 | kamranazeem | hello all, is this the right place to ask asterisk@home questions ? |
19:31.11 | Darwin35 | thats it I quit .. |
19:31.15 | Darwin35 | I give up |
19:31.35 | file[von] | yay giving up |
19:31.51 | Darwin35 | I am going to startup my own company some how where I dont have to be treated like shit in phone support. |
19:31.52 | justinu | lol |
19:32.02 | kamranazeem | I cannot get a simple setup work, One Asterisk 2.3 on centos, two XP machines. two extensions registered on AMP, cannot get audio on xten-lite |
19:32.18 | justinu | ~amp |
19:32.19 | jbot | somebody said amp was "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
19:32.44 | fourcheeze | ~a@h |
19:32.49 | fourcheeze | hmm |
19:33.06 | kamranazeem | sorry I am a total newbie to IRC as well |
19:33.24 | fourcheeze | kamranazeem try this: /join #freepbx |
19:33.28 | justinu | it's ok, it's just that no one here likes to support AMP |
19:33.31 | justinu | because it's a pain in the ass |
19:33.38 | kamranazeem | ok thanks very much |
19:34.17 | fourcheeze | kamranazeem: when you decide that freepbx doesn't do what you want, come back :-) |
19:34.20 | Hmmhesays | i think i found a bug in freepbx |
19:34.40 | Hmmhesays | dial(SIP/<host>/<user>) isn't valid is it? |
19:34.48 | file[von] | Hmmhesays: 42 |
19:35.15 | Hmmhesays | heh |
19:35.18 | asterboy | http://pastebin.ca/45839 - look at the [poly] context. I've changed it to everything with no success. |
19:35.25 | wunderkin | file[von]: 24 |
19:35.30 | file[von] | wunderkin: negative |
19:35.31 | fourcheeze | can I send an arbitrary SIP header out? |
19:35.34 | asterboy | Ignore the CALLERID part, taken that out. |
19:35.37 | *** join/#asterisk homebrew-hsv (n=homebrew@mail.kancharla.com) |
19:35.37 | kamranazeem | fourcheeze oops. You mean, this chat room is better than freepbx ? |
19:35.50 | Hmmhesays | you can get to this chat room from freepbx |
19:35.56 | fourcheeze | kamranazeem: I mean that asterisk is better than freepbx, so choose your poison |
19:36.16 | asterboy | the "_9." used in conjuction with ignorepat = 9 |
19:36.18 | kamranazeem | ok I want to ask about asterisk |
19:36.28 | [TK]D-Fender | asterboy : so right no you have to dial 9 + something more, and the "more" is wasted and then you get a 2nd dial-tone and have to dial out.... |
19:36.47 | [TK]D-Fender | asterboy : ignorpat only workks on analog FXS on zaptel cars. |
19:36.55 | [TK]D-Fender | asterboy : you are on a SIP phone..... |
19:36.55 | asterboy | yes, to at least get it working. |
19:36.57 | [TK]D-Fender | no good |
19:37.03 | kamranazeem | I cannot get a simple setup work, One Asterisk 2.3 on centos, two XP machines. two extensions registered on AMP, cannot get audio on xten-lite |
19:37.04 | asterboy | didn't know about the only working on FXS. |
19:37.09 | asterboy | Yes SIP phone. |
19:37.13 | asterboy | Polycom IP 600 |
19:37.39 | [TK]D-Fender | asterboy : this is what you want to do : exten => _XX.,1,Dial(Zap/1/${EXTEN}) |
19:37.39 | asterboy | I've also take out the 9 so I can try it without having to dial a prefix first, using "_." |
19:37.40 | kamranazeem | the speakers and mic at both machines are ok, I have checked using the "sound recorder" |
19:37.48 | tzanger | is there a preferred POE switch for Polycom IP501s? |
19:37.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@88.153.166.193) |
19:37.59 | [TK]D-Fender | asterboy : It'll through pretty much everything at your Zap card. |
19:38.05 | [TK]D-Fender | and actually DIAL it for you.... |
19:38.06 | asterboy | excellent! |
19:38.12 | asterboy | trying... |
19:38.33 | justinu | tzanger: any 802.3af switch will work, but you'll need the polycom 802.3af power injector |
19:38.46 | *** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net) |
19:40.18 | [TK]D-Fender | tzanger : its not so much an injector as a splitter from a PoE supplied cable back out again.... as for switch any 802.3af should do. D-Link & NetGear are the cheapest these days.... |
19:40.25 | Hmmhesays | craziness |
19:40.26 | [TK]D-Fender | +/-$500 for 24 ports. |
19:40.41 | asterboy | FINALLY! |
19:41.06 | asterboy | Can you imagine putting that in as an example SOMEWHERE in the docs. |
19:41.11 | justinu | fender: what do you know about this "injector/splitter"? |
19:42.38 | vuud | Can I have * allow direct dial in from SIP phones? Well, can I make it work anyway... |
19:42.44 | asterboy | why do you need 2 "_XX." X's in the pattern matching space? or does it matter? |
19:43.27 | asterboy | Now I understand the ${EXTEN} passing through. |
19:43.28 | octothorpe_ | [TK]D-Fender: what to you mean $500 for 24 ports? could you post a url? |
19:43.43 | asterboy | $500 is kick ass for 24! |
19:43.58 | [TK]D-Fender | asterboy : there is no magic "all answers here" doc... * is just too powerful, but they could give a few better CONCISE samples. |
19:44.02 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:44.28 | asterboy | why do you need 2 "_XX." X's in the pattern matching space? |
19:44.35 | [TK]D-Fender | Oh, I'm sorry... $400 ... http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1205369&CatId=868 |
19:44.36 | fourcheeze | kamranazeem: is the client registering with * ? |
19:45.03 | octothorpe_ | [TK]D-Fender: thanks, that's awesome |
19:45.19 | [TK]D-Fender | oops... Netgear for $335! http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1697254&CatId=868 |
19:45.27 | [TK]D-Fender | Just keeps getting better... |
19:45.30 | asterboy | and PoE |
19:45.31 | asterboy | nice |
19:45.47 | asterboy | thought that was for FXO ports though. |
19:46.15 | [av]bani | i doubt it can provide 15.4W to all ports simultaneously |
19:46.19 | [TK]D-Fender | asterboy : no, thats for powering your phones :) |
19:46.30 | asterboy | yes, see that now. |
19:46.39 | [TK]D-Fender | 369W tot..... not unbelieveable.... |
19:46.53 | [av]bani | you need a big switching PS for >300W |
19:46.55 | [TK]D-Fender | I have lamps bigger than that... |
19:46.56 | asterboy | In pattern matching the "_" is for beginning and "." for ending, hence the double X. |
19:47.03 | asterboy | ok, starting to clue in here. |
19:47.16 | [TK]D-Fender | asterboy : I just make sure not to through single digits at the pstn :) |
19:47.16 | [av]bani | [TK]D-Fender: providing >300W power is altogether different from consuming >300W |
19:47.32 | asterboy | ah, ok. |
19:47.33 | [TK]D-Fender | [av]bani : True there is overhead, but lets not get neurotic.... |
19:47.41 | octothorpe_ | <PROTECTED> |
19:48.13 | [av]bani | [TK]D-Fender: i doubt it can provide 360W, i'm betting peak is more like 150W |
19:48.26 | [TK]D-Fender | octothorpe_ : They are.... the prices are just finally plummeting... |
19:48.30 | [av]bani | [TK]D-Fender: consider that your average desktop phone eats maybe 5W |
19:49.10 | [TK]D-Fender | Oops! Ignor the Netgear one! Only HALF of the ports are PoE (unlike the D-Link DES-1526 I linked previously) |
19:49.36 | [TK]D-Fender | I knew netgear's LOWER products were 50/50, but took a while to find it on the 24 port. |
19:49.44 | [av]bani | yep, says there.. 180W max |
19:49.46 | [av]bani | i was close |
19:49.58 | SpaceBass | does anyone make a sub $200 24port managed switch? |
19:50.27 | [av]bani | http://www.dlink.com/products/resource.asp?pid=403&rid=1492&sec=0 <- max power consumption 180 watts |
19:50.32 | GerbilNut | do incoming calls just come in as asterisk on the phones if they don't have an incoming callerid? |
19:50.56 | SpaceBass | GerbilNut, depends on dial plan....but typically yes |
19:51.00 | *** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-243.dsl.pltn13.pacbell.net) |
19:51.08 | *** part/#asterisk Grizzy (i=Generic@ppp-71-133-231-243.dsl.pltn13.pacbell.net) |
19:51.23 | *** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-243.dsl.pltn13.pacbell.net) |
19:52.42 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
19:52.51 | [TK]D-Fender | [av]bani : then again how many heavy load devices are you planning on loading onto it? |
19:53.07 | Grizzy | Isn't Quell a medication for lice? (/me ducks) |
19:53.36 | Qwell[filetop] | yes |
19:53.41 | Qwell[filetop] | ~qwell |
19:53.43 | jbot | rumour has it, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
19:53.59 | Qwell[filetop] | wait, no |
19:54.02 | Qwell[filetop] | Kwell is |
19:54.08 | Qwell[filetop] | Quell means, basically, to kill |
19:54.15 | Grizzy | aaah, ok. : o ) |
19:54.15 | Hmmhesays | god i love it when customers say they have an e&m wink t1 thats 5Ess |
19:54.33 | Grizzy | I think "to suppress" |
19:55.17 | Grizzy | They know the details, eh? |
19:55.55 | Hmmhesays | anyone on fwd I need to test something on this new install |
19:56.14 | ManxPower | Hmmhesays, I believe FWD has an echo test number |
19:56.28 | *** join/#asterisk clive- (n=pirch@dsl-145-31-108.telkomadsl.co.za) |
19:56.37 | [av]bani | [TK]D-Fender: well, people look at "15.4w per port" and think OMG 360W |
19:56.41 | Hmmhesays | yeah but then i'd have to open a web browswer and find it |
19:56.42 | [av]bani | [TK]D-Fender: it's deceptive at best |
19:56.50 | justinu | there's an available power pool |
19:56.52 | Grizzy | I'll be brave: 741907 |
19:56.54 | SpaceBass | Hmmhesays, use lynx |
19:56.58 | SpaceBass | :D |
19:56.59 | justinu | and the thing will send out SNMP traps if you get close to exceeding it |
19:57.01 | Qwell[filetop] | [av]bani: What did we decide, about POE on the ciscos? Which ones have 802.3af? |
19:57.02 | *** join/#asterisk [Outcast] (n=outcast@222-152-110-218.jetstream.xtra.co.nz) |
19:57.18 | [TK]D-Fender | [av]bani : for $400 I think we can afford to STFU and jsut be happy we don't need a motgage for it ;) |
19:57.27 | [Outcast] | does asterisk support "P-Asserted-Identity |
19:57.35 | tamp4x | anyone here use spandsp? when i load asterisk -vvvvvvvvv it stops loading when app_rxfax.so loads....any ideas why? |
19:57.46 | homebrew-hsv | Does anyone know how to force g.711 pass-through when also allowing g729 between two sip peers? |
19:57.49 | *** join/#asterisk jmacz (n=jmacz@201.244.198.113) |
19:57.51 | [av]bani | Qwell[filetop]: ~phones |
19:58.01 | Qwell[filetop] | I don't know how to use this system :p |
19:58.08 | Qwell[filetop] | no clue where/what the browser is |
19:58.14 | justinu | canadiafied |
19:58.17 | Hmmhesays | 613 is fwd's echo test |
19:59.04 | Hmmhesays | i think vonage just killed my super secret test account |
19:59.19 | Qwell[filetop] | ~phones |
19:59.20 | jbot | extra, extra, read all about it, phones is at http://bani.anime.net/phones/ |
20:00.35 | *** part/#asterisk homebrew-hsv (n=homebrew@mail.kancharla.com) |
20:00.53 | clive- | anyone got any advice about setting up raid for the first time |
20:01.04 | [av]bani | pray? |
20:01.10 | Grizzy | jbot_ softphones |
20:01.17 | clive- | I am bit confused, it wants a stripe size, but I am using raid 1 |
20:01.21 | Grizzy | ~softphones |
20:01.41 | justinu | wrong channel, yo |
20:01.53 | Grizzy | nice chart, the phones one. |
20:02.21 | [av]bani | thanks |
20:02.44 | vuud | I am trying to connect to my astrisk from XLite on the same segment via direct dial. peer debug shows that astrisk is getting the invite, but then getting a SIP/2.0 404 Not Found from the calling client. WTF? |
20:03.02 | Grizzy | It needs an "add" button. : o ) |
20:03.02 | vuud | At least that is how I think it is happening... the sip debug stuff is new to me |
20:03.08 | SpaceBass | anyone played wiht the new WIP-300 wifi phone? |
20:04.41 | Hmmhesays | i love it when customers want ME to call their telco and find out their t1 settings for them |
20:05.01 | justinu | that's typically what avendor does :P |
20:05.06 | tamp4x | wip 300 sucks |
20:05.17 | Hmmhesays | heh, no... i'm support for the vendor |
20:05.18 | justinu | you can't expect lusers to understand any of the telco jargon |
20:05.32 | Hmmhesays | then vendor can do the bitch work |
20:06.02 | rollergrrl | Which distro should I go with for a production Asterisk server? |
20:06.23 | fourcheeze | queue holy war |
20:06.25 | fourcheeze | cue |
20:06.40 | Nugget | It doesn't matter in the least. |
20:06.46 | tamp4x | windows xp |
20:06.51 | jsharp | Whatever you feel comfortable with. |
20:07.05 | rollergrrl | I can't decide on debian or gentoo |
20:07.15 | *** join/#asterisk Noky (n=damian@200.69.211.18) |
20:07.16 | Noky | hi |
20:07.25 | fourcheeze | rollergrrl: just pick one - I use both |
20:07.33 | rollergrrl | I'm just wondering what others are using in production |
20:07.33 | Noky | i want know what is the 'priority' in the dialplan's ? |
20:07.35 | Hmmhesays | i like to run asterisk on your mom |
20:07.54 | fourcheeze | rollergrrl: one downside of gentoo might be excessive cpu usage if you have to update stuff |
20:07.55 | rollergrrl | My mom is dead |
20:08.08 | rollergrrl | ahh true |
20:08.14 | rollergrrl | can't have that |
20:08.25 | Vitux | gentoo isn't bad... |
20:08.32 | Hmmhesays | rollergrrl: thats hot |
20:08.43 | Vitux | runs pretty clean... |
20:08.49 | rollergrrl | Hmmhesays: and I'm not |
20:08.53 | [TK]D-Fender | rollergrrl : Just make sure whatever you use has the required libraries to compile the components you'll require. |
20:09.08 | tamp4x | freebsd 6.0 |
20:09.16 | Vitux | but yah.. updating is a pain in the ass sometimes but if anything I find it easier on gentoo to update then any other distro anyway |
20:09.16 | Noky | hi |
20:09.17 | Grizzy | Mandrake/Mandriva is a nightmare. |
20:09.23 | rollergrrl | Is it very difficult to get spandsp working in debian? |
20:09.29 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
20:09.37 | tamp4x | ahh spandsp |
20:09.39 | Grizzy | freebsd 4.6 and 6.1 work; 6.0 wouldn't even boot. |
20:09.42 | tamp4x | i have toruble with that |
20:09.57 | Vitux | bsd is ok... not much support though last time I used it |
20:10.07 | Vitux | mind you that was a long ass time ago |
20:10.10 | dpryo | rollergrrl: Nope, it's easy |
20:10.17 | Vitux | ubuntu is also decent.. |
20:10.28 | dpryo | :) |
20:10.31 | Grizzy | bsd systems feel faster than linux. |
20:10.33 | [TK]D-Fender | rollergrrl : Debian is a very good base for most things period. Just keep an eye on the versions between * and SpanDSP. its the mix that kills more often than not. |
20:10.39 | tamp4x | if its easy why did spandsp compile ad when i start asterisk asterisk stops loading when app_rx /tx fax is loaded |
20:11.08 | AlexCTI | Justinu: When I did a Update to the new version i have a problem with the module chan_modem.so the UPGRADE.txt said that i need upgrade it to chan_misdn.so can you explian me how to do that? |
20:11.24 | rollergrrl | thanks for the info guys |
20:11.55 | justinu | just get rid of chan_modem.so |
20:12.44 | AlexCTI | just remove? and that's it? do I need again the make install? |
20:12.58 | justinu | no |
20:13.48 | fourcheeze | rollergrrl: I tend to build my own *, even on debian |
20:14.39 | tamp4x | prob spandsp will be easier sicne they have it as a package |
20:14.55 | *** join/#asterisk dokhench (n=no@adsl-065-080-180-134.sip.bna.bellsouth.net) |
20:15.06 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
20:15.07 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
20:15.24 | kuku5 | <PROTECTED> |
20:15.29 | Katty | hi lads. |
20:15.48 | justinu | afternoon |
20:15.58 | file[von] | kuku5: because it's making progress? |
20:16.14 | Hmmhesays | heh |
20:16.20 | jsharp | One channel is passing call progress information to another channel. |
20:17.05 | justinu | set verbose 0, problem solved |
20:17.20 | kuku5 | file[von]: can I stop it ? |
20:17.41 | Hmmhesays | so you can't make calls? |
20:18.22 | SpaceBass | ok...just had my telco forward one pots to one broadvoice number and my other pots line to another bv number, but they both appear to be coming in on the same bv trunk |
20:18.40 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
20:18.51 | Noky | i want know what is the 'priority' in the dialplan's ? |
20:19.06 | [TK]D-Fender | Katty: mew. |
20:19.35 | *** join/#asterisk JerJer[mobile] (n=jj@sjcc28x192.sjccnet.com) |
20:19.49 | [TK]D-Fender | Noky : Yes, I'm sure most of us know what "priorities" are.... |
20:20.00 | SpaceBass | for instance, mine is getting drunk tonight |
20:20.14 | jsharp | A man with a plan. |
20:20.21 | SpaceBass | thus exten =>s,1,Drunk(SpaceBass|scotch) |
20:20.46 | JerJer[mobile] | exten => s,2,DUI |
20:20.56 | SpaceBass | followed by exten => s,2,Bitch(about|job) exten => s,3,Passout() |
20:21.08 | SpaceBass | JerJer[mobile], naw...came too close once, will NEVER risk it again |
20:21.19 | SpaceBass | when I start I either cab it, make someone else drive or just dont leave |
20:21.41 | JerJer[mobile] | lol |
20:21.49 | jsharp | exten =>s,4,Wake() exten=>s,5,StareOverInBed() exten=>s,6,KnawArmOff() |
20:21.58 | jsharp | Gnaw, too. |
20:22.07 | SpaceBass | I gotta catch a flight tomorrow morning |
20:22.25 | SpaceBass | can;'t be too hung over...then again I can drink on the plane |
20:22.58 | justinu | bad move |
20:23.10 | justinu | booze and fliying == not healthy |
20:23.19 | SpaceBass | disagree.... do it every week |
20:23.27 | SpaceBass | now when I'm the one doing the flying...thats another story |
20:23.37 | justinu | there's a certain physiology to it |
20:24.04 | justinu | just drink lots of water with your booze |
20:24.07 | SpaceBass | what altitude? |
20:24.13 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
20:24.22 | Hmmhesays | wow i upgraded * now i can't call out vonage anymore |
20:24.25 | justinu | well, everytime you fly commercial you're at an apparent altitude of about 8500 feet |
20:24.33 | [TK]D-Fender | Katty : Mar 25th is the big day.... |
20:24.50 | Hmmhesays | big day for? |
20:25.02 | [TK]D-Fender | <PROTECTED> |
20:25.03 | SpaceBass | i'm going to be spending the next 10 days sleeping at 9,500 and days mostly over 12k |
20:25.04 | Katty | [TK]D-Fender: :> |
20:25.27 | justinu | heh |
20:25.34 | justinu | american beer is probably ok |
20:25.37 | jbalcomb | too many damned Asterisk on Debian HOWTOs to choose from |
20:25.43 | Katty | i wouldn't know, justinu, i don't drink it. |
20:25.43 | SpaceBass | and i find that gin martinis go down much better at 9k feet! |
20:25.49 | justinu | me neither |
20:25.59 | jbalcomb | Anyone recomend the /best/ guide on installing Asterisk on Debian? |
20:26.08 | [TK]D-Fender | jbalcomb : Just iognore the ones with "apt-get install asterisk" and might do OK :) |
20:26.14 | Hmmhesays | i want a big black jet with a bedroom in it gonna join the mile high club at 37,000ft |
20:26.22 | *** join/#asterisk medusaXX (n=medusaxx@p54A98532.dip0.t-ipconnect.de) |
20:26.23 | justinu | you don't need that |
20:26.36 | SpaceBass | Hmmhesays, did that in college on the way to paris once....comercial flight |
20:26.37 | jsharp | Hmmhesays: Go business class on Virgin Airlines. They've got sleepers. |
20:26.41 | fourcheeze | jbalcomb: just do it, it's not hard |
20:26.41 | [TK]D-Fender | jbalcomb : just install the base from debian, then download * stable from digium's FTP and compile from source |
20:26.44 | jbalcomb | [TK]D-Fender thats no good eh? ;) |
20:26.49 | SpaceBass | also...ammmm...took care of that as a pilot once too |
20:27.10 | justinu | you're a rated pilot? |
20:27.14 | SpaceBass | yeah |
20:27.21 | jbalcomb | XXX rated pilot |
20:27.25 | justinu | i've got a private single, and multi |
20:27.25 | SpaceBass | LOL |
20:27.28 | willt[work] | LOL |
20:27.38 | SpaceBass | just private single here |
20:27.42 | justinu | never did the deed while being PIC |
20:27.54 | SpaceBass | and i fly so much comercially I have a hard time keeping it up ....ok...leme rephrase that |
20:28.17 | medusaXX | hi |
20:28.30 | jbalcomb | i did it in a phone booth in a bar once, rather risque i felt. |
20:28.36 | medusaXX | i always get this error when starting asterisk: moh_register: Unable to open pseudo channel for timing |
20:28.53 | medusaXX | you know what causes that? |
20:29.04 | jsharp | medusaXX: You have a zaptel device installed or ztdummy running? |
20:29.10 | medusaXX | no neither nor |
20:29.16 | jsharp | There ya go. |
20:29.26 | jsharp | Need one or the other to make MOH work worth a crap. |
20:29.42 | medusaXX | uhm ok |
20:30.12 | medusaXX | i thought i would have disabled all modules i dont need, but i missed that one obviously |
20:30.13 | medusaXX | thanks |
20:31.10 | [TK]D-Fender | jbalcomb : nothing to it... |
20:31.10 | SpaceBass | i changed my MOH to Copland's Rodeo....really cracks people up to be put on hold to the "beef, its whats for dinner" comericial |
20:31.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
20:31.48 | digg10 | how can i do macros in extensions from db? |
20:31.49 | jsharp | I inadvertantly changed ours to Lords of Acid. Oops. |
20:31.58 | *** join/#asterisk file[laptop] (n=jcolp@sjcc28x192.sjccnet.com) |
20:33.03 | justinu | darling come here and fu... |
20:33.06 | *** join/#asterisk nroej (n=joern@heaven.cyphertext.de) |
20:33.07 | nroej | hi |
20:33.44 | medusaXX | jsharp: wait :) |
20:33.55 | medusaXX | i think i misunderstood that |
20:34.06 | medusaXX | moh is music on hold, right? |
20:34.12 | jsharp | Yeah. |
20:34.14 | medusaXX | why do i need a zaptel device or a ztdummy for htat |
20:34.27 | justinu | good question... |
20:34.27 | medusaXX | i thought moh would be something zaptel related |
20:34.35 | medusaXX | sorry |
20:34.47 | jsharp | cause it dervives clocking and timing for the audio from zaptel. |
20:34.54 | medusaXX | ahh |
20:35.01 | medusaXX | are there other music on hold modules? |
20:35.08 | *** join/#asterisk the_magic_bean (n=mhermsdo@209.43.15.211) |
20:35.39 | medusaXX | i dont like the idea of adding a module to the kernel |
20:36.15 | digg10 | ne1 knows how to do macros while using sql extensions? |
20:36.33 | jsharp | No other MOH modules. Either zaptel or nothing. |
20:36.48 | jbalcomb | Why isn't asterisk-sounds in the download directory? |
20:36.52 | medusaXX | sad |
20:37.01 | medusaXX | then i need that stupid module |
20:37.15 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
20:37.52 | clive- | medusa, just install it, it works fine, and wont upset your kernel |
20:38.07 | Grizzy | missed the mile high club, though I've got a pilot friend. |
20:38.11 | Noky | [TK]D-Fender |
20:38.16 | Noky | what is the priority? |
20:38.32 | medusaXX | it's just that i need to remember that module every time i recompile my kernel |
20:40.32 | [TK]D-Fender | Noky : Which priority? What ARE you talking about? |
20:41.10 | heison | does anyone know how to make 7960's use RFC2833? |
20:41.29 | heison | after a recent upgrade from 7.5 to 8.2, inband DTMF no longer works... |
20:41.32 | jbalcomb | heison RFC2833 for Cisco is referred to as PVT |
20:42.00 | ManxPower | I REALLY hate it when someone releases a Windows only compiled TCL script and won't release the source |
20:42.23 | jbalcomb | heison or maybe is AVT... |
20:43.35 | justinu | dtmf_inband: 0 |
20:43.39 | justinu | dtmf_outofband: avt |
20:44.21 | jsharp | I have mine set for avt_always |
20:45.10 | jsharp | heison: Are there any new/nifty features to SIP 8.2? |
20:45.20 | jsharp | All I can find are bug fix release notes. |
20:45.42 | heison | i don't know, i was on MGCP and was trying to get the stupid thing to talk SIP, according to Cisco, they recommend 8.2 |
20:45.47 | heison | all my other phones are still 7.5 |
20:45.59 | jsharp | And I can't download 8.2 cause my damn smartnet contract expired. |
20:47.11 | SpaceBass | jsharp, in the same boat |
20:48.42 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
20:49.25 | Peaceful | Does asterisk still require _exactly_ version 0.59r of mpg123, or will newer versions work? (like 0.59s) |
20:49.42 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
20:50.15 | jsharp | Newer should work. |
20:51.06 | Nugget | when did that change? |
20:51.23 | Nugget | It's traditionally required that precise version, nothing newer or older. |
20:51.33 | Noky | [TK]D-Fender |
20:51.38 | Noky | i'm talking about dialplans |
20:51.42 | jsharp | I'm running it with newer with no problems. |
20:51.47 | Noky | extend => name,PRIORITY,... |
20:51.54 | Noky | what is this priority? |
20:51.59 | jsharp | priority is the step number in the dialplan. |
20:52.06 | Nugget | Noky: like line numbers in BASIC. :) |
20:52.13 | jsharp | Priority 1 = step 1. Priority 2 = step 2 |
20:52.13 | justinu | GOSUB WITHOUT RETURN |
20:52.15 | Noky | ahah |
20:52.16 | jsharp | etc etc etc |
20:52.32 | Noky | and... this prioritys must have an order? |
20:52.45 | *** join/#asterisk illuy (n=assdf@85-64-194-107.barak-online.net) |
20:52.47 | jsharp | They go in ascending sequence. |
20:53.02 | Noky | ok |
20:53.04 | Noky | thanks |
20:53.21 | Nugget | and there can't be gaps in the sequence. |
20:54.06 | jsharp | And that's a PITA. Anytime you want to insert a step in the middle of the dialplan, you have to renumber everything. |
20:54.45 | Noky | ahah |
20:55.03 | Noky | ok, thanks for the information :) |
20:55.07 | Noky | xD |
20:55.13 | Noky | j# xD |
20:59.38 | *** join/#asterisk Dr-Linux (n=nothing@host202-147-168-130.lhr.dancom.net.pk) |
21:00.11 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
21:02.28 | *** join/#asterisk illuy (n=assdf@85-64-194-107.barak-online.net) |
21:02.57 | *** join/#asterisk heison (n=heison@216.235.9.2) |
21:05.17 | [TK]D-Fender | Noky : If you are asking that question you need to read a lot more..... |
21:05.23 | [TK]D-Fender | Noky : Try TFOT |
21:05.25 | [TK]D-Fender | ~thebook |
21:05.27 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
21:05.44 | [TK]D-Fender | Nugget : sure there can ;) |
21:05.48 | SpaceBass | can I use my modem and asterisk as an answering machine? |
21:05.58 | Dr-Linux | TFOT is very good book |
21:06.11 | [TK]D-Fender | SpaceBass : Depends on your modem. NEXT!!@!@!@ (c) BKW |
21:06.36 | SpaceBass | what about changing my caller ID to prank my friend, can asterisk and my winmodem do that? |
21:06.37 | SpaceBass | :) |
21:06.41 | SpaceBass | sorry...I'll stop |
21:06.45 | Dr-Linux | anybody knows if SJphone supports g729 ? |
21:08.17 | [TK]D-Fender | Dr-Linux : nope. |
21:08.45 | [TK]D-Fender | Dr-Linux : you won't find downloadable OSS soft-phones with it because of the licensing. Common sense. |
21:10.24 | Dr-Linux | [TK]D-Fender: but license is for g729, so i thought may be SJphone support it, xlite doesn't |
21:11.33 | [TK]D-Fender | Dr-Linux : Ask yourself how they could control it if you just DLed it from anywhere.... |
21:11.55 | Dr-Linux | DLed :S |
21:12.13 | Dr-Linux | hhm.. you are right |
21:12.37 | Dr-Linux | however i have EyeBeam for unlimited users |
21:12.57 | Dr-Linux | that may support g729 |
21:13.19 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
21:14.02 | justinu | eyebeam has g729, indeed |
21:14.11 | markit | hi, today when I receive a call throught FWD, seems I receive it twice... any clue? anyone else with this problem? I've found an old post about other having experienced this, but not solution in that post |
21:14.24 | Peaceful | so, if mpg123 0.59s will really work with asterisk, the "You have the WRONG version of mpg123... you need .59r" message in the makefile should probably be modified |
21:15.28 | _Sam-- | Peaceful : you could always use native moh |
21:16.12 | Peaceful | _Sam--: "moh"? |
21:16.23 | exonic | anyone around here know what the ExtraChannel paremeter is for using Action: Redirect ? |
21:16.46 | _Sam-- | moh = moooosic on hold |
21:17.15 | *** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-164-89.lsanca.fios.verizon.net) |
21:18.41 | Hmmhesays | to fall in love and fall in debt, to alcohol and cigarettes |
21:18.53 | Hmmhesays | and mary jane to keep me insane sniffin someone elses cocaine |
21:19.21 | exonic | Hmmhesays, don't for get the leuds |
21:19.34 | exonic | ludes*.. (spelling?) |
21:19.52 | Gamercjm | For GMS format sound, what does the sample rate need to be at? or does it not matter |
21:21.44 | markit | Gamercjm: aseterisk gsm sounds are at 8000 hz (8Khz) |
21:22.13 | Gamercjm | thanks |
21:23.42 | wasim | and for that matter, are all other formats as well |
21:28.01 | tamp4x | anyone here use spandsp? when i load asterisk -vvvvvvvvv it stops loading when app_rxfax.so loads....any ideas why? |
21:28.22 | stoffell | tamp4x. what version? (of * and spandsp) |
21:30.05 | Darwin35 | sex |
21:30.08 | Darwin35 | ~sex |
21:30.09 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
21:30.58 | Darwin35 | ~seen jbot |
21:31.04 | jbot | jbot <i=ibot@pdpc/supporter/active/TimRiker/bot/apt> was last seen on IRC in channel #debian, 147d 6h 14m 57s ago, saying: 'rumour has it, sarge is Ten-HUT! Fall in! Sarge is the code name for the current stable Debian release, version 3.1, released on June 6th, 2005. Ask me about <install debian>, or i guess sarge is the biggest lump of free as in ... |
21:31.18 | De_Mon | cd? |
21:31.23 | stoffell | lol |
21:31.49 | Darwin35 | cd = change direction |
21:31.57 | De_Mon | ah |
21:32.15 | Darwin35 | or charge date |
21:32.25 | stoffell | lool |
21:33.14 | *** join/#asterisk eivindtr (n=eivindtr@062016241059.customer.alfanett.no) |
21:33.57 | Darwin35 | ok wich one you bitches stole my wallet |
21:34.38 | stoffell | if it's free, it's okay :p |
21:34.42 | Darwin35 | thnks nugget |
21:34.59 | eivindtr | Hi all. I have a system (1.2.1 with Sangoma A101) where I experience a longer delay on inbound than outbound voice (in the area 100-200 ms). Anyone got any ideas what might cause it? |
21:35.10 | tamp4x | stoffell : asterisk 1.2.5 ; spandsp 0.0.2pre25 |
21:35.21 | ManxPower | Darwin35, I'm sure it's nobody that knows you -- if they knew you they would know you are poor 8-) |
21:35.40 | ManxPower | eivindtr, we really can't help you with Sangoma |
21:35.42 | gaupe | stoffell: you've got thomson to update voip-info :) |
21:35.46 | eivindtr | I'm trying to learn about the buffering in zaptel, but I can't really figure it out... |
21:35.48 | stoffell | tamp4x, you used the howto like http://asteriskguru.com/tutorials/spandsp.html |
21:35.50 | Gamercjm | Any one try the: Asterisk Realtime Voice Pitch Changer |
21:35.52 | Darwin35 | heheh thnks manx |
21:36.02 | ManxPower | eivindtr, zaptel basically doesn't buffer |
21:36.04 | stoffell | gaupe, yeah, some1 just was faster :p but no BLF yet.. MWI works though.. |
21:36.10 | justinu | eivindtr: echotraining |
21:36.22 | gaupe | stoffell: will try it later tonight :) |
21:36.48 | eivindtr | justinu: I tried to disable echocancel alltogether, but it didn't really help.. |
21:36.56 | stoffell | gaupe, it's cool, blf is coming, waiting for email from alcatel to get timing :) |
21:36.59 | ManxPower | eivindtr, we really can't help you with Sangoma |
21:37.06 | eivindtr | ManxPower: Thanks :) |
21:37.15 | Darwin35 | Manx pvt me |
21:37.30 | gaupe | stoffell: I was hoping for the norwegian translation to show up too... |
21:37.53 | stoffell | gaupe, no no, all norway people I know, speak very well english :P |
21:38.06 | gaupe | stoffell: I know :) |
21:38.38 | eivindtr | Is it right that the ZT_POLICY_IMMEDIATE is only relevant at the setup of the rtp-stream, or is it used throughout the communication? |
21:38.39 | tamp4x | yes stoffell |
21:38.50 | stoffell | what error you get tamp4x? |
21:38.57 | ManxPower | eivindtr, zaptel doesn't know anythinhg about RTP |
21:39.20 | tamp4x | no error reported stofffel |
21:39.35 | stoffell | tamp4x, check /var/log/asterisk/full ? |
21:39.38 | eivindtr | ManxPower: my bad... I really knew that :/ but the question stands... sortof |
21:39.51 | *** part/#asterisk Gertrude (n=gert@chickenbones.bflony.adelphia.net) |
21:39.58 | Darwin35 | ok we all need to ppol our money together and open a giant voip provider |
21:40.08 | tamp4x | cannot open `/var/log/asterisk/full' for reading: No such file or directory |
21:40.30 | stoffell | tamp4x okay, where did you hide your log then? ;) |
21:40.32 | ManxPower | tamp4x, stop. step away from the asterisk server. Take a class on Linux or Unix |
21:41.22 | tamp4x | the debug log? |
21:41.53 | stoffell | yeah, you should find why it doesn't start * |
21:42.02 | tamp4x | ahh i see [app_rxfax.so]2006-03-16 09:17:04 WARNING[25479] loader.c: libtiff.so.3: cannot open shared object file: No such file or directory |
21:42.43 | stoffell | seems like you don't have libtiff.so.3 , you should install libtiff |
21:43.20 | fourcheeze | Darwin35: ok, I have £5 |
21:43.24 | tamp4x | i have it installed |
21:43.36 | tamp4x | /usr/local/lib/libtiff.so.3 |
21:43.36 | tamp4x | <PROTECTED> |
21:43.50 | fourcheeze | tamp4x: do you have /usr/local/lib in /etc/ld.so.conf ? |
21:43.56 | stoffell | and /usr/local/lib/ is probably not in your ld.so... |
21:44.12 | stoffell | wasn't worth typing that for me :) |
21:44.19 | fourcheeze | :-) |
21:44.54 | *** part/#asterisk nroej (n=joern@heaven.cyphertext.de) |
21:45.11 | tamp4x | yeahaw |
21:45.41 | fourcheeze | tamp4x: did you run ldconfig ? |
21:45.46 | tamp4x | thanks four and stoffell |
21:45.48 | tamp4x | yep |
21:46.04 | stoffell | hehe, now the fun starts, good luck tamp4x ;) |
21:49.52 | exonic | i've had nothing but truoble w/ faxes :) |
21:50.06 | jsharp | Just the fax, ma'am. |
21:50.10 | exonic | thinking of takign it off asterisk all together soon enough |
21:50.17 | stoffell | exonic, i only use it to rx, .. works fine |
21:50.25 | exonic | stoffell, yeah, rx is money |
21:50.34 | exonic | stoffell, tx from a SIP phone on the other hand.... |
21:50.49 | exonic | even with T.38 I havn'thad much success |
21:50.56 | exonic | about 10% fail |
21:51.05 | tamp4x | hmm sending out a fax seems to crash asterisk |
21:51.10 | stoffell | ok, good to know.. |
21:52.09 | exonic | stoffell, how do u send ? |
21:52.12 | exonic | stoffell, land line? |
21:52.14 | *** join/#asterisk drs9 (n=DRS@host86-133-127-224.range86-133.btcentralplus.com) |
21:52.26 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
21:52.26 | stoffell | exonic, yes, hylafax and a good old usrobotics:d |
21:52.44 | stoffell | but, with an analog->isdn to a BRI ofcourse :D |
21:53.11 | exonic | stoffell, does hylafax work w/ any digium hardware? |
21:53.11 | *** part/#asterisk drs9 (n=DRS@host86-133-127-224.range86-133.btcentralplus.com) |
21:53.41 | stoffell | exonic, i use analog modem (serial) that connects to a converter to connect it to BRI card |
21:53.53 | stoffell | but i guess the analog cards can use a modem with hylafax, yes |
21:54.17 | exonic | ahh, I was thinking it'd be awesome if I could use a single port PRI card and use hylafax to get 23 channels for faxing |
21:54.34 | stoffell | exonic, hm, no idea on that.. guess not.. |
21:54.35 | SpaceBass | I still have fax issues....i can get asterisk to recieve fine, if I manually transfer the call |
21:54.42 | SpaceBass | but I all the pages are blank |
21:54.54 | exonic | SpaceBass, sending is trouble |
21:55.01 | exonic | SpaceBass, i'm in the same boat |
21:55.12 | SpaceBass | sending blanks? |
21:55.26 | exonic | SpaceBass, i've had blanks, black smears and half filled black pages... |
21:55.42 | SpaceBass | thats like the 4th unintentional innuendo ive made today |
21:56.01 | SpaceBass | I can send fine usaing an analogue fax connected to an ATA, even over SIP trunks |
21:56.02 | _Sam-- | for the record, USRobotics sucks with hylafax. |
21:56.03 | Gamercjm | anyone try Asterisk Voice Changer 0.3 |
21:56.09 | SpaceBass | its recieving that sucks |
21:56.16 | exonic | asterisk is infertile when it comes to sending :) |
21:56.18 | _Sam-- | multitech modems are recommended for hyla :) |
21:56.19 | SpaceBass | Gamercjm, I've seen it...havent tried it |
21:56.54 | jsharp | Gamercjm: I was playing with it earlier. Quality kinda sucks. Sounds *very* processed. |
21:57.40 | Gamercjm | oh, I was gonna try it out, Would it be even worth testing? |
21:57.58 | _Sam-- | why would you want to sound like a girl? :) |
21:58.01 | exonic | Gamercjm, if you're trying to make a scary movie =) |
21:58.11 | jsharp | Its easy enough to set up. Maybe you'll have better luck. |
21:58.28 | SpaceBass | its even better if you change your CID with nufone or something similar |
21:58.42 | SpaceBass | but be careful...I had that prank backfire last weeked! |
21:58.46 | Gamercjm | Did you have any trouble installing, or did it affect your regular usage |
21:58.51 | jsharp | No and no. |
21:58.52 | Gamercjm | i have nufone |
21:59.14 | *** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239697.sympatico.ca) |
21:59.32 | Gamercjm | did you install from CVS? |
21:59.46 | Gamercjm | sounds like I would have to reinstall asterisk with the regular patch |
22:00.31 | jsharp | patched my work/compiled tree of 1.2.4, then just re-ran make && make install |
22:00.51 | jsharp | It only compiled app_voicechanger.so, since everything else was already compiled and there were no dependencies. |
22:01.07 | Gamercjm | u used CVS? |
22:01.11 | jsharp | No. |
22:01.14 | jsharp | 1.2.4 from ftp.digium.com |
22:01.28 | Gamercjm | oh |
22:01.40 | *** join/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net) |
22:02.26 | asterboy | When another call comes through on a pots line and I want to flash, how do you administrate those kind of calls? |
22:02.33 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
22:03.26 | *** part/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net) |
22:03.34 | jsharp | Dial a certain DTMF sequence on your phone and the zaptel interface will flash the line. |
22:03.40 | jsharp | I forget the sequence, though. |
22:04.08 | Seldon1975 | * wiki page is down - can someone tell me how to tweak the MOH volume? |
22:04.11 | ManxPower | *0 |
22:04.19 | jsharp | That's it. |
22:04.34 | jsharp | Dial *0 and your zaptel interface will flash. |
22:04.43 | *** join/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net) |
22:06.17 | Gamercjm | :/ |
22:06.18 | Gamercjm | checking for C++ compiler default output file name... configure: error: C++ compiler cannot create executables |
22:06.42 | *** join/#asterisk RoyK (n=roy@213-150-148-98.telenor.se) |
22:07.39 | *** join/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net) |
22:08.12 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
22:09.39 | mroth_imm | anyone have any thoughts on irqbalance...should it be running on a high-volume asterisk server |
22:09.53 | jeffoatrulez | i have a musiconhold() question: if i'm not sending frames from my client, will it not send music back? |
22:10.53 | *** part/#asterisk illuy (n=assdf@85-64-194-107.barak-online.net) |
22:11.00 | Dr-Linux | jsharp: on what phone? *0 |
22:11.04 | jeffoatrulez | it's working for me via a phone, but not through a muted iaxclient implementation. |
22:12.07 | ManxPower | mroth_imm, asterisk does a pretty good job of IRQ balancing |
22:12.26 | mroth_imm | ManxPower: it's the network interface i'm concerned about |
22:12.38 | ManxPower | jeffoatrulez, if you are not sending frames, astersk won't send frames |
22:12.49 | mroth_imm | irqbalance is putting each NIC on its own processor in my SMP system (4 CPUs) |
22:12.51 | ManxPower | that is why Asterisk does not support silence supressiion |
22:13.09 | Dr-Linux | question about sip, if the genral section codec is set to g711 , will that work, if i define anyother codec for indivisual user account? |
22:13.24 | mroth_imm | net result is that CPU 0 handles ALL interrupts from the interface handling the RTP streams and so it runs at a higher percentage than the others |
22:13.42 | mroth_imm | meaning the box will be cpu bound on the processor handling those interrupts |
22:13.54 | jeffoatrulez | manxpower: we do the same with app_conference, though. i don't send frames, but app_conference sends me other folks' frames. |
22:14.09 | De_Mon | voip-info down? |
22:14.17 | mroth_imm | you can use irq affinity to spread the interrupts via /proc/irq/#/smp_affinity |
22:14.35 | jeffoatrulez | i.e. i can listen without sending. or i should say, we send pings, etc. |
22:15.00 | De_Mon | server is up, but wiki not talkin |
22:15.05 | mroth_imm | but irqbalance tweaks those values as well on a ten second interval...so i'm wondering if I can just turn it off without problems |
22:16.00 | jeffoatrulez | i started looking through channel.c and the ast_generator code, but it was non-obvious how to get the generator code to send regardless of incoming frames. |
22:17.18 | asterboy | Missed if there was an answer...phones are off the hook literaly. |
22:17.21 | asterboy | Missed if there was an answer...phones are off the hook literaly. |
22:17.24 | asterboy | When another call comes through on a pots line and I want to flash, how do you administrate those kind of calls? |
22:17.33 | jsharp | Dial *0 |
22:17.45 | jsharp | And your zaptel device will flash your POTS line. |
22:18.04 | asterboy | thx jsharp! |
22:20.04 | *** join/#asterisk tainted_ (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net) |
22:20.37 | SpaceBass | trying to flash pots is miserable |
22:21.00 | SpaceBass | get your telco to call forward on busy to a BroadVoice byod lite account |
22:23.17 | Seldon1975 | hello all, I've tried playing with mpg123's '-g' parameter to decrease the gain, but when reload MOH and I listen to the on-hold music it's just as loud - can someone tell me how to make this quieter? |
22:24.31 | *** join/#asterisk gtodd (n=gtodd@198.62.158.205) |
22:24.43 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
22:24.50 | *** join/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
22:25.29 | bancus | Just for the edification of those who earlier saw me asking if a clock running too fast could affect asterisk playback, the answer is yes. |
22:25.40 | gtodd | I have the AMP GUI running in Asterisk@Home VMWare image ... and an idefisk softphone running on the host system |
22:26.06 | bancus | Upgrading my kernel fixed the problem. |
22:26.11 | *** part/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
22:26.28 | gtodd | * is running on 192.168.40.128 .... but I can't get idefisk to register as a peer or friend to use that system |
22:26.40 | rajiv|work | how many voice trunks are recommended per person at a company ? like 1 trunk for eveyr 5 employees ? |
22:26.42 | SpaceBass | gtodd, host or bridgged netowkring? |
22:26.50 | gtodd | I've set up an extension ... so |
22:26.50 | SpaceBass | same subnet as your box? |
22:26.58 | gtodd | hmm no |
22:27.10 | SpaceBass | trying to even check voicemail while * is running in VMware sucks |
22:27.23 | Seldon1975 | rajiv|work: if you need a rule of thumb, 1 trunk per 5 employees is as good as any... |
22:27.26 | SpaceBass | seriously...ebay a $40 PIII or something...you'll be glad you did |
22:27.41 | SpaceBass | gtodd, can you ssh into your * "box" |
22:27.41 | gtodd | the box is running 172.16.5.* the *@Home is running in 192.168.40.128 ... |
22:27.44 | Seldon1975 | rajiv|work: WE HAVE 7 TRUNKS AND ABOUT 35 EMPLOYEES, SO... |
22:27.48 | Seldon1975 | oops sorry caps |
22:28.01 | SpaceBass | gtodd, set VM to bridged networking (if its not) and put them on the same subnet |
22:28.04 | SpaceBass | brb |
22:28.12 | gtodd | SpaceBass, I have access via vmware so ... |
22:28.13 | gtodd | OK |
22:28.23 | rajiv|work | Seldon1975: thanks |
22:28.39 | Seldon1975 | rajiv|work: yw |
22:28.42 | rajiv|work | also, anyone heard the term "3MG" referring to t1 voice/data? what is it |
22:31.04 | Seldon1975 | rajiv: are you sure they don't mean 3mb |
22:31.14 | Seldon1975 | rajiv|work: as in 3 megabit |
22:31.55 | rajiv|work | that is what i would think, but they wrote 3MG sevearl times |
22:32.38 | Seldon1975 | rajiv|work: hmm, I can only thtink the author is using it to mean megabit. If it's not that I have no idea |
22:33.18 | Seldon1975 | hello all, I've tried playing with mpg123's '-g' parameter to decrease the gain, but when reload MOH and I listen to the on-hold music it's just as loud - can someone tell me how to make this quieter? |
22:33.23 | fourcheeze | is t1 3mbit/sec ? |
22:34.52 | Darwin35 | lick my balls |
22:35.21 | Darwin35 | everything is working but vm with pgsql and realtime |
22:35.38 | Dr-Linux | Darkhalf: :S |
22:35.41 | Seldon1975 | T1 = 1.544 Mbps |
22:35.42 | Darwin35 | now O have to figure out unixodbc |
22:35.55 | Seldon1975 | <PROTECTED> |
22:36.21 | Seldon1975 | or T1c = 3.152 Mbps |
22:36.23 | Dr-Linux | E1 = 2 Mb/ps |
22:36.43 | Seldon1975 | so maybe they were talking about T1c |
22:37.54 | *** join/#asterisk Raszh (n=Spoon@66.253.253.210) |
22:38.07 | Raszh | Has anyone gotten Zaptel 1.2.* to work on Linux 2.4? |
22:39.00 | fourcheeze | Darwin35: unixodbc is easy |
22:39.32 | [TK]D-Fender | Darwin35 : yup, just download compile and run the GUI config and you're up and running fast.... |
22:39.36 | Dr-Linux | Raszh: yes, i'm using |
22:39.47 | [TK]D-Fender | one of the few places I appreciated having a front end :) |
22:39.55 | fourcheeze | never tried a gui |
22:40.03 | fourcheeze | apt-get install unixodbc |
22:40.55 | fourcheeze | then hack on odbc.ini |
22:41.21 | Seldon1975 | isnt ODBC really slow? |
22:41.40 | Dr-Linux | Seldon1975: what's voice channel and data channel? :S |
22:42.03 | Raszh | Dr-Linux: how did you do so without CRC_CCITT support in 2.4? |
22:42.43 | Dr-Linux | Raszh: 2.4 linux kernel right? |
22:42.49 | Raszh | yes |
22:43.08 | Dr-Linux | i'm using RHEL, i just update it from RHN |
22:43.46 | Raszh | and Zaptel 1.2.*? |
22:44.00 | Dr-Linux | Raszh: my both asterisk box have RHEL |
22:44.18 | Dr-Linux | and both have zaptel 1.2.x |
22:44.39 | Raszh | do you have your kernel config file? |
22:44.43 | Dr-Linux | Raszh: RHEL AS 3 has 2.4 kernel |
22:44.52 | *** join/#asterisk jeffoatrulez (n=jeffo@slim-eth0.horizonlive.net) |
22:45.10 | Raszh | for that matter, can you 'locate ccitt' for me and see if you have a module in /lib/modules ? |
22:45.15 | Seldon1975 | Dr-Linux: I don't understand your last question to me |
22:45.24 | jeffoatrulez | i was asking a question about musiconhold a bit ago, but my irc client exploded, so i lost any responses. if anyone still has responses on their screen, could you copy/paste it to me? |
22:45.34 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
22:45.41 | *** part/#asterisk Nodren (n=nodren@64.193.95.10) |
22:45.54 | Dr-Linux | Raszh: sorry right now i'm at home and using win .. i'll do it later |
22:45.58 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
22:46.03 | Dr-Linux | Raszh: what OS you are using? |
22:46.09 | Raszh | Gentoo |
22:46.25 | Raszh | but I'm looking at the linux sources downloaded from kernel.org |
22:46.43 | Raszh | 2.4 doesn't have CRC_CCITT support |
22:46.54 | Dr-Linux | Seldon1975: as i read E110P T1 digium card can do 12 voice channel and 12 data channel, |
22:47.03 | Nodren | anyone ever experience problems compiling zaptel into centos 4.2? |
22:47.16 | Dr-Linux | Seldon1975: i can understand FXO/FXS, PRI/ISDN/T1/E1 |
22:47.33 | Dr-Linux | but i didn't understand voice/data channels? :S |
22:48.08 | Dr-Linux | Raszh: i'm not much good with kernel, but that works for me |
22:48.24 | Nodren | i cant seem to get zaptel to compile, i keep getting errors |
22:48.36 | medusaXX | i installed the ztdummy kernel module but i still receive this error when starting asterisk: moh_register: Unable to open pseudo channel for timing... Sound may be choppy |
22:48.38 | Raszh | either you're mistaken or RH has patched CRC_CCITT into their 2.4 kernels |
22:48.44 | medusaXX | do i need additional configuration after loading the module? |
22:48.46 | Dr-Linux | Raszh: can you tell me when did Zaptel 1.2.* released ? |
22:48.51 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
22:50.08 | tsume | hmm |
22:50.15 | tsume | where can I get genzaptelconf? |
22:50.19 | tsume | I'm missing it on this system |
22:50.45 | ManxPower | tsume, never heard of it |
22:51.12 | tsume | <PROTECTED> |
22:51.13 | ManxPower | maybe you are confused and it's s 3rd party piece of software |
22:51.15 | tsume | this is a fresh install |
22:51.31 | ManxPower | tsume, no, you are using some silly gui for Asterisk |
22:51.33 | ManxPower | ~amp |
22:51.36 | jbot | somebody said amp was "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
22:51.39 | tsume | zapata-channels.conf |
22:51.45 | tsume | ManxPower: no I'm not |
22:51.51 | tsume | ManxPower: hardcore command line here |
22:51.56 | ManxPower | tsume, correct, asterisk does not generate any config files |
22:52.07 | tsume | genzaptelconf takes /etc/zapta.conf and generate the sapta for me |
22:52.10 | Raszh | Dr-Linux, I don'tk know when it was released |
22:52.15 | tsume | I'm just curious where to get it |
22:52.16 | ManxPower | And there are no sample config files with that name |
22:52.34 | tsume | where did this come from? |
22:52.43 | tsume | I never installed any third party software |
22:52.46 | ManxPower | tsume, no idea, but I don't think it came from Digium. |
22:52.49 | medusaXX | i found it... the device nodes were missing |
22:52.54 | tsume | ManxPower: how odd.. |
22:53.04 | ManxPower | tsume, so you downloaded Asterisk and Zaptel and then built from source and installed? |
22:53.19 | *** join/#asterisk talljon84 (n=chatzill@66-168-63-104.dhcp.mdsn.wi.charter.com) |
22:53.26 | tsume | ManxPower: oh, I know where it came from, nevermind |
22:53.33 | ManxPower | tsume, where? |
22:53.38 | tsume | ManxPower: my friend copied configs from the other box he was playing with |
22:53.44 | tsume | he was using amp,, heh.. |
22:54.01 | tsume | all existing "easy" configs suck |
22:54.10 | *** join/#asterisk menger (n=menger@dsl-125-209-136-232.vic.veridas.net) |
22:54.15 | tsume | I should make one, base it on GTK# |
22:54.26 | tsume | a decent one like how oracles db designer is |
22:54.56 | tsume | ManxPower: I was just curious if there was a magical way to configure this TDM2400 card :) |
22:55.05 | tsume | TDM2400P (prototype) |
22:55.10 | justinu | magic... pay someone to do it |
22:55.21 | tsume | justinu: thats what I'm for ;) |
22:55.23 | talljon84 | I installed A@H 2.7 and it worked correctly until I rebooted. Now I am receiving the missing /dev/zap error. I have installed named and dhcpd but everything else is fresh on the box. The udev.rules file has the right lines for zap but I do not understand what is going wrong. Does anyone have ideas? |
22:55.26 | justinu | heh |
22:55.30 | justinu | then you're the magic! :) |
22:55.41 | tsume | justinu: I'm not some idiot linuxer. I'm a hard core BSD guy and will make/use scripts to cut down my time |
22:55.51 | justinu | lol |
22:56.10 | tsume | anything I use will either make me more money, or decrease my time :) |
22:56.10 | justinu | sed, awk, perl, or what? |
22:56.23 | tsume | justinu: whatever works ;) |
22:56.28 | tsume | I know em all ;) |
22:56.31 | justinu | as long as it's not linux |
22:56.52 | tsume | justinu: oh I don't care if its linux, but most linux dweebs sit there and do it by hand to be "leet" *chuckle* |
22:56.58 | justinu | hah |
22:57.26 | tsume | its just halarious, you know. and they use gimp because they hate adobe making money from photoshop *chuckle* |
22:57.31 | justinu | i guess there are a lot of linux n00bs |
22:57.36 | justinu | lol |
22:57.40 | tsume | even though gimp ois more like PS1 |
22:57.49 | justinu | i came from solaris... *shrug* |
22:58.00 | Nodren | anyone want to help me, i'm having a problem compiling zaptel on a 2.6 kernel |
22:58.06 | Nodren | and i'm using the make linux26 |
22:58.11 | justinu | you don't need that anymore |
22:58.13 | justinu | make install |
22:58.21 | Nodren | just make install? |
22:58.23 | justinu | yep |
22:58.24 | *** part/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
22:58.27 | tsume | justinu: I favor sun for improving postgres and releasing the code BSD licensed for sol |
22:58.35 | tsume | *the postgres code |
22:58.40 | justinu | i like java... |
22:59.03 | tsume | justinu: java wont run here ;) on NetBSD servers. not well anyway |
22:59.08 | tsume | I'm poritng mono to netbsd |
22:59.15 | justinu | yeah... they've gotta release an opensource JRU |
22:59.17 | justinu | JRU |
22:59.18 | justinu | err |
22:59.20 | justinu | JRE |
22:59.21 | Nodren | it gives me an error |
22:59.25 | Nodren | when i do just make install |
22:59.41 | justinu | sun is being pretty stubborn about the whole JRE thing |
22:59.46 | justinu | opensource the bastard already |
22:59.49 | tsume | justinu: once they release an opensource JRE, I'll be all over it. Development tools should be free, applications should be allowed not to |
22:59.52 | justinu | yep |
22:59.56 | justinu | i'm with you |
23:00.28 | tsume | I don't like java because you have to pay a license if you want to include it on embedded platforms |
23:00.37 | tsume | I'd rather work on and port mono |
23:00.42 | tsume | maybe opensource the changes |
23:00.58 | Nodren | justinu: i tried just doing a make install of zaptel |
23:01.01 | Nodren | it gave me an error |
23:01.04 | Nodren | said Error 2 |
23:01.32 | justinu | paste the error |
23:01.37 | justinu | hopefully someone will look at it |
23:01.37 | ManxPower | Actual error message: |
23:01.40 | ManxPower | ** (.:11973): CRITICAL **: gtk_pizza_set_size: assertion `pizza != NULL' failed |
23:01.44 | justinu | heh |
23:02.05 | justinu | s/pizza/muffin and file could have written that |
23:02.08 | Nodren | make[2]: *** [/usr/src/asterisk/zaptel/zaptel.o] Error 1 |
23:02.08 | Nodren | make[1]: *** [_module_/usr/src/asterisk/zaptel] Error 2 |
23:02.08 | Nodren | make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' |
23:02.09 | Nodren | make: *** [linux26] Error 2 |
23:02.13 | justinu | not here |
23:02.16 | justinu | ~pb |
23:02.18 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
23:03.27 | mroth_imm | anyway to tell if a kernel installed from an rpm (*not* source) has certain constants set such as CONFIG_IRQBALANCE? |
23:03.39 | justinu | you need the srpm |
23:04.37 | mroth_imm | aaahh...good thinkin'...thanks, i'm a little flustered today : ) |
23:05.09 | mroth_imm | trying to figure out what the need for a daemon that balances irqs is if the kernel is already doing it |
23:05.31 | mroth_imm | irqbalance VS irq affinity |
23:05.52 | Nodren | no suggestions? |
23:06.29 | justinu | nodren, learn how to use pastebin to paste your error output |
23:06.33 | justinu | someone will help you then |
23:06.39 | Nodren | alright i will next time |
23:09.20 | Nodren | http://pastebin.com/606425 |
23:09.49 | [av]bani | http://www.dell.com/downloads/jp/corporate/imagebank/ecc_1.jpg |
23:10.19 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
23:13.00 | justinu | ~centosbug |
23:13.01 | jbot | centosbug is, like, a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
23:13.22 | justinu | that's your problem, nodren |
23:13.28 | Nodren | thanks! |
23:15.21 | tsume | hmm wow |
23:15.32 | tsume | I need to download the HEAD of cvs for zaptel module |
23:15.37 | tsume | to get this TDM2400 card to work :) |
23:16.02 | ManxPower | tsume, you should not need to. |
23:16.04 | ManxPower | 1.2 should support it. |
23:16.17 | tsume | ManxPower: its not working for the channels I have installed |
23:16.28 | ManxPower | tsume, maybe you configured it wrong |
23:16.38 | tsume | 1-12 dont work, the empty slots work. I just get no such device |
23:16.55 | tsume | ManxPower: doubt it, the card is full yconfigured |
23:17.43 | tsume | works fine for theh TDM400 |
23:17.47 | ManxPower | Well THAT sucks. The README in the Zaptel source dir does not mention the TDM2400P at all |
23:17.51 | tsume | so it must be the kernel module |
23:18.11 | Nodren | awesome it worked great |
23:18.11 | ManxPower | tsume, does the kernel module show up in "lsmod"? |
23:18.20 | tsume | yes |
23:18.27 | tsume | else the TDM400 wouldn't be working |
23:18.40 | tsume | zttool reports the card as there |
23:18.42 | tsume | and configured |
23:18.54 | ManxPower | and what does ztcfg -vvv say? |
23:18.59 | tsume | anything trying to open a channel which has an installed module fails with "No such device" |
23:19.20 | tsume | 28 channels configured |
23:19.24 | tsume | I've done all this already ;) |
23:19.25 | ManxPower | 28? |
23:19.37 | ManxPower | the card has 24 channels |
23:19.49 | tsume | TDM2400P + TDM400 |
23:20.05 | ManxPower | and what kernel module do you have loaded for the TDM2400P? |
23:20.08 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
23:20.14 | tsume | 1.2.4 |
23:20.24 | tsume | I'll just build the kernel module out of head right quick ;) |
23:20.26 | shmaltz | hi everyone |
23:20.33 | ManxPower | no. wctdm, wcfxo, etc. |
23:20.45 | ManxPower | go right ahead, I don't think it will fix anything. |
23:20.45 | tsume | oh |
23:21.40 | tsume | wctdm |
23:21.46 | tsume | wctdm24xxp |
23:21.49 | tsume | oh wait here |
23:21.53 | tsume | it ssupposed to be working |
23:21.55 | tsume | *hrm* |
23:21.58 | justinu | so what's so special about a TDM2400P? |
23:22.04 | tsume | I'll try running head |
23:22.09 | tsume | justinu: its a 24port card |
23:22.14 | ManxPower | justinu, It has some nifty cool buzzword technology! |
23:22.19 | justinu | i know what a regular TDM2400 is |
23:22.20 | tsume | 1 module card == 4 lines |
23:22.36 | tsume | justinu: 2400 is a prototype, period |
23:22.41 | ManxPower | tsume, and "lsmod" shows both modules loaded? |
23:22.42 | tsume | its just TDM2400P |
23:22.50 | tsume | this one has echo cancellation built onboard as well |
23:22.57 | tsume | ManxPower: yeah it does, strange, eh? |
23:23.21 | ManxPower | tsume, which driver loads FIRST? |
23:23.35 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
23:23.40 | tsume | hmmm |
23:23.47 | tsume | the TDM2400p |
23:24.04 | ManxPower | so then you configure the tdm2400p modules first in /etc/zaptel.conf |
23:24.08 | tsume | no wait |
23:24.10 | tsume | let me check.. |
23:24.31 | tsume | ManxPower: wht? |
23:24.41 | tsume | ManxPower: I just tell it to config 1-25 ;) |
23:24.42 | ManxPower | The order of the channels configured in /etc/zaptel.conf needs to be the same order as the kernel modules are loaded. |
23:24.46 | tsume | 1-25 |
23:24.59 | justinu | zaptel gets really pissy about stuff not matching just right |
23:25.01 | ManxPower | tsume, what modules do you have on the cards? |
23:25.19 | tsume | FXS 1-25 |
23:25.21 | tsume | 3 FXO |
23:26.07 | ManxPower | so you have signaling=fxs_ks then channel=1-25 then signaling=fxo_ks and then channel=26-28 |
23:26.20 | ManxPower | sorry. |
23:26.27 | ManxPower | so you have signaling=fxo_ks then channel=1-25 then signaling=fxs_ks and then channel=26-28 |
23:26.29 | ManxPower | that is correct. |
23:26.38 | ManxPower | fxo ports use fxs signaling |
23:26.41 | tsume | yeah |
23:26.44 | ManxPower | fxs ports use fxo signaling |
23:29.20 | twisted[asteria] | anyone know how to set ANI with Set(CALLERID(num))? |
23:30.57 | justinu | Set(CALLERID(num) = 12135551212) |
23:31.00 | *** join/#asterisk fjean (n=fjean@201009210199.user.veloxzone.com.br) |
23:31.03 | twisted[asteria] | right, i know that much |
23:31.08 | twisted[asteria] | but that doesn't toggle the ANI flag |
23:31.14 | justinu | oh |
23:31.38 | justinu | is it in the presentation indicator? |
23:31.47 | justinu | SetCallingPres()? |
23:32.01 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
23:32.07 | justinu | screened=yes or something |
23:32.14 | *** join/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
23:32.20 | fjean | hello guys ! ==- anyone sucessfully installed the unicall patch into asterisk 1.2.5 ? |
23:32.39 | bancus | For some reason, it seems like asterisk is not hearing keypresses coming in over sip. Any ideas? |
23:32.49 | I-MOD | dtmf setting |
23:33.01 | I-MOD | in sip.conf |
23:33.20 | X-Rob | ~centosbug |
23:33.27 | jbot | somebody said centosbug was a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
23:33.27 | I-MOD | there are 2 or 3 different settings |
23:33.33 | bancus | it says inband |
23:34.01 | I-MOD | try the other options |
23:34.09 | Dr-Linux | ~RHELbug |
23:34.12 | twisted[asteria] | justinu, heh... well, i just used the deprecated application ;) |
23:34.15 | bancus | k |
23:34.28 | bancus | the wiki says broadvoice needs inband, but that could be outdated, I guess |
23:34.59 | Dr-Linux | hi justin |
23:35.20 | justinu | hello |
23:35.40 | orlock | Hmmm |
23:35.49 | orlock | heya |
23:36.01 | Dr-Linux | justinu: how data channels work with T1 ? :S |
23:36.18 | justinu | you set aside a certain number of timeslots for data |
23:36.22 | Dr-Linux | i wish to read a doc about my question, but no luck to find some in google :( |
23:36.38 | [av]bani | data channels just use timeslots, and no signalling |
23:37.25 | Dr-Linux | yes, but what is this "data" internet from same telco the PRI is coming from? or what |
23:37.32 | justinu | it depends |
23:37.43 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
23:37.44 | Dr-Linux | i have doubts |
23:37.49 | justinu | the telco will MUX the dataconnection onto the T1 on their side |
23:38.19 | justinu | then you demux the data timeslots to some kind of router |
23:38.29 | justinu | and the voice timeslots into some kind of telco switch |
23:38.44 | Dr-Linux | like T1 card has only port, where we plug the PRI from telco that have TS 24 , 1.544 Mbps |
23:38.48 | Dr-Linux | but data :S |
23:39.01 | justinu | well, you need a T1 card that can deal with fractional T1s |
23:39.03 | [av]bani | your router handles it |
23:39.09 | justinu | they make those |
23:39.16 | [av]bani | or oyu can get a t1 mux |
23:39.53 | Dr-Linux | mux ? :( |
23:39.58 | [av]bani | multiplexer |
23:39.59 | orlock | hmm |
23:40.14 | [av]bani | send some channels out 1 T1 port, and other channels out another T1 port |
23:40.29 | [av]bani | or just get a cisco router and handle everything in one box |
23:40.34 | justinu | yeah, adtran TSU100 can do that |
23:40.46 | justinu | T1 out to pbx, V.35 serial to router |
23:40.51 | justinu | but that's oldschool |
23:41.21 | rajiv|work | my company is moving to a new space. i suggeted getting a t1 line instead of the 8 pots lines we have now. it will be frac t3 supposedly with some voice and some data |
23:41.22 | orlock | Stupid question - i need a register=blahblah:blah@voipprodier.com dont i? |
23:41.33 | rajiv|work | so i too would get a t1 card for an asterisk box ? |
23:42.01 | justinu | yeah |
23:42.09 | rajiv|work | recommendations ? |
23:42.21 | justinu | Digium TE110P or Sangoma A101u |
23:42.22 | Dr-Linux | justinu: but what if i use all 23 B channels for voice? i can do that right? |
23:42.25 | rajiv|work | i think also i do nto want to build the * box. who sells pre-built and tested asterisk boxes ? |
23:42.31 | justinu | Dr-Linux: yep, then no data! |
23:43.03 | I-MOD | rajiv|work: i know digium does turnkey systems |
23:43.19 | *** join/#asterisk Meads (i=Meads@host-84-9-28-57.bulldogdsl.com) |
23:43.22 | *** part/#asterisk Meads (i=Meads@host-84-9-28-57.bulldogdsl.com) |
23:44.00 | rajiv|work | their bundles ? |
23:44.19 | Dr-Linux | justinu: data mean here >> internet/bandwidth right? |
23:44.28 | justinu | yes |
23:44.32 | justinu | that's what you're talking about, right? |
23:44.46 | rpm | is this right, i called sprint the other day to get a quote on a PRI, they said they want 500-600$ a month for only 8 available B channels? |
23:45.06 | justinu | that's high |
23:45.06 | [av]bani | yep |
23:45.12 | [av]bani | thats sprint |
23:45.14 | justinu | but sounds ballparkish |
23:45.23 | [av]bani | your ILEC can probably do better |
23:45.38 | rajiv|work | I-MOD: i see only their bundles on the digium site |
23:45.48 | rpm | if i register as a CLEC do you think i can get at-cost rates? |
23:46.01 | [av]bani | no |
23:46.03 | rajiv|work | i just got a quote for $800 / mo for 3mb data + $8 / mon for each line |
23:46.15 | justinu | they'll probably charge a CLEC more! |
23:46.41 | rpm | 800$ a month for 3mb, from who? |
23:46.53 | rajiv|work | conversent, in boston |
23:46.53 | bancus | I-MOD: thanks, rfc mode fixed it |
23:46.56 | *** part/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
23:47.09 | [av]bani | unless you become a facilities based clec, you can only get resale rates, and you can't sell to yourself |
23:47.15 | rpm | rajiv|work: is that two bonded t1's? |
23:47.18 | rajiv|work | rpm: well i should say that it is "3MG" which i'm thinking is 3mb since thats what we talked about. |
23:47.22 | rajiv|work | rpm: frac t3 |
23:47.29 | [av]bani | now, you can setup a separate corporation and resale to yourself, but thats a lot of work |
23:47.33 | rajiv|work | oh. can i plug a frac t3 into a te110p ? |
23:47.36 | justinu | is that including the t3 loop charges? |
23:47.42 | justinu | not without a ds3 mux |
23:47.53 | *** join/#asterisk NeonLevel (n=NeonLeve@200.52.142.186) |
23:48.00 | rajiv|work | justinu: supposedly yes, that includes the loop charges |
23:48.00 | [av]bani | telcos like to do atm on ds3 |
23:48.00 | justinu | you can buy the CAC widebank 28s pretty cheap |
23:48.09 | justinu | rajiv: that's an amazing price |
23:48.17 | rajiv|work | 3 year contract |
23:48.21 | justinu | still |
23:48.31 | [av]bani | justinu: ds3 is pretty cheap actually, its the bw that costs |
23:49.01 | justinu | ds3 loops charges are like 3k a month around here |
23:49.14 | rajiv|work | where is that? |
23:49.17 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
23:49.24 | justinu | Los Angeles |
23:49.32 | rpm | i heard you can get a ds3 for about 400-500$ a month |
23:49.51 | [av]bani | thats just the local loop cost |
23:49.57 | [av]bani | you need services on top of that |
23:50.00 | justinu | i remember them costing a lot more when we installed them at our facilities in calabasas |
23:50.34 | [av]bani | we pay ~$1500 for our ds3 loop, and then services on that are pretty cheap |
23:50.41 | [av]bani | ends up much cheaper than buying T1s |
23:50.59 | [av]bani | 1 meg pvc is only like $15/mo |
23:52.19 | justinu | frame relay or ATM? |
23:52.23 | *** join/#asterisk criptos (n=criptos@201.121.246.67) |
23:52.27 | criptos | Hi everyone |
23:53.07 | [av]bani | atm, though you can get atm->fr conversion too |
23:53.13 | criptos | I have a pap2 at a remote asterisk, and when I dial from the pap to another nother asterisk box, I got no sound.. |
23:53.31 | [av]bani | you can have the telco route frame relay PVCs to your atm |
23:53.33 | criptos | I´m using ulaw all the way, at the iax transport and beweet pap2 and asterisk... |
23:53.38 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
23:54.31 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
23:55.21 | rajiv|work | justinu: good to konw we're getting a deal. |
23:55.40 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
23:55.55 | Zodiacal | anyone know how to use the one touch recording feature? mines setup for *1, but it doesn't do anything |
23:55.59 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
23:56.07 | Zodiacal | it just gives me a fast busy tone.. i tried it during a call and it doesn't do anything it seems |
23:56.23 | I-MOD | rajiv|work: you'll likely have to call into digium and ask sales to find out about turnkeys |
23:56.24 | Zodiacal | i read it suposed to store them in /var/spool/asterisk/monitor but it doesn't seem to |
23:56.43 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |