| 00:01.09 | *** join/#asterisk finchy (n=finchy@66.236.227.227.ptr.us.xo.net) |
| 00:02.46 | brookshire | 8 eastern |
| 00:02.51 | brookshire | mog |
| 00:02.52 | brookshire | :) |
| 00:03.46 | mog_work | support goes 7 to 7 central which is how i corrected myself brookshire |
| 00:11.26 | *** join/#asterisk riddlebox (n=james@24-207-158-49.dhcp.stls.mo.charter.com) |
| 00:12.39 | SpaceBass | anyone have a IP5000 wifi phone? |
| 00:13.12 | riddlebox | has anyone integrated an IP Office with asterisk? |
| 00:13.22 | Qwell[] | riddlebox: IP office? |
| 00:14.25 | riddlebox | Qwell[], its an avaya product, I was just wondering it has the ability of creating an IP trunk and communicating with other IP Office's over it |
| 00:14.41 | Qwell[] | What technology? |
| 00:15.07 | riddlebox | umm you mean what is the codec? |
| 00:15.18 | Qwell[] | no, what technology does it use |
| 00:15.35 | Qwell[] | vpn? sip? You've given 0 details |
| 00:16.11 | riddlebox | in the ip office you just tell the ip trunk the address of the other server, let me get my work laptop out to look |
| 00:16.50 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 00:16.56 | SpaceBass | arrruuggg I want a WIP330.... cannot find any vendors with one |
| 00:16.58 | [av]bani | yay |
| 00:17.11 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:18.50 | riddlebox | Qwell[], I do see it has H.323 IP trunking |
| 00:19.32 | harlequin516 | Okay the voicemail thing isn't working for me.. When I dial 8500, from kphone, I get the voicemail menu but I cannot login. Asterisk CLI outputs : Incorrect password '' for user 'sham' (context = default), but I dial 1234 not 'sham', is this a bug? |
| 00:19.48 | harlequin516 | sham is my sip username |
| 00:20.08 | r0d3nt | harlequin516, check your DTFM settings |
| 00:20.18 | SpaceBass | harlequin516 asterisk@home? |
| 00:20.29 | *** join/#asterisk Graphinboyy (n=piespy@user-24-214-132-43.knology.net) |
| 00:21.01 | harlequin516 | Nope gentoo portage compiled asterisk 1.2.4 |
| 00:21.14 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 00:21.24 | SpaceBass | didn't think so since you were using 8500 and not *97 or something...just asking |
| 00:21.29 | harlequin516 | dtmfmode=rfc2833 in my sip.conf |
| 00:21.38 | SpaceBass | try =inline |
| 00:22.04 | harlequin516 | really? I thought rfc2833 is preferred... |
| 00:22.19 | SpaceBass | depends on the phone and the results you are getting |
| 00:23.14 | *** join/#asterisk kisu (n=daniel@cielkisu.tb.as8758.net) |
| 00:23.18 | harlequin516 | Hmm, Well I tried with inline, same result.. |
| 00:23.31 | harlequin516 | Is there a setting for kphone to do one or the other? |
| 00:23.42 | SpaceBass | not sure about kphone |
| 00:24.10 | harlequin516 | What's the best sip phone nowadays, is there something better than kphone? |
| 00:24.16 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 00:24.33 | harlequin516 | Linphone didn't work for me at all. |
| 00:25.22 | harlequin516 | I wish these programs were a little more transparent. |
| 00:26.51 | Qwell[] | harlequin516: tried twinkle? |
| 00:26.55 | Qwell[] | I hear it's good |
| 00:27.25 | _Soul_ | greetings |
| 00:27.42 | _Soul_ | has anybody managed to exchange sip calls with voipbuster ? |
| 00:28.22 | _Soul_ | im not asking if you managed to use voipbuster as your termination provider, but calling some voipbuster user sip url |
| 00:28.36 | _Soul_ | or some voipbuster user calling your sip url |
| 00:29.46 | *** join/#asterisk iGotNoTime (n=joshua@cpe-65-189-240-199.woh.res.rr.com) |
| 00:30.44 | delta34ooo | question, in v1.2 is there a way to display the name of the person u dialed rather then the number you press for a cisco phone, so if i dialed 0, it will displayed I called the Operator |
| 00:30.59 | harlequin516 | hmm twinkle lemme see |
| 00:31.09 | *** join/#asterisk dextro (n=dextro@cpe-70-116-10-201.austin.res.rr.com) |
| 00:31.11 | *** part/#asterisk sfirefinch (n=finchy@66.236.227.227.ptr.us.xo.net) |
| 00:31.36 | SpaceBass | delta34ooo it should work if you could transfer the call |
| 00:33.59 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 00:34.28 | justinu | ~seen thetatag |
| 00:34.33 | jbot | justinu: i haven't seen 'thetatag' |
| 00:35.05 | justinu | ~seen _sam |
| 00:35.06 | jbot | _sam <n=sam@65-100-5-175.eugn.qwest.net> was last seen on IRC in channel #debian, 229d 1h 12m 32s ago, saying: 'say in lieu of keyboard-interactive'. |
| 00:35.13 | justinu | ~seen _sam-- |
| 00:35.15 | jbot | _sam-- is currently on #asterisk. Has said a total of 124 messages. Is idling for 23h 42m 20s, last said: 'duplex- : no.'. |
| 00:37.53 | justinu | ~seen r_evolution |
| 00:37.56 | jbot | r_evolution <i=_evoluti@208.251.203.246> was last seen on IRC in channel #asterisk, 5d 22h 37m 43s ago, saying: 'OUT!'. |
| 00:38.02 | justinu | where the fuck is everyone? |
| 00:38.08 | Qwell[] | hiding |
| 00:39.35 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:40.58 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 00:43.09 | *** join/#asterisk St1ckm4n (n=shortes9@68.178.74.166) |
| 00:43.48 | St1ckm4n | anyone here very familiary with the FOP? |
| 00:43.58 | St1ckm4n | *familiar |
| 00:45.51 | SpaceBass | St1ckm4n i've used it |
| 00:46.10 | riddlebox | Qwell[], do you know anything about setting up a h.323 gateway? |
| 00:46.15 | Qwell[] | no |
| 00:46.33 | riddlebox | ok, I found some stuff on voip-info.org |
| 00:47.37 | *** join/#asterisk nexgen (n=me@adsl-70-135-6-65.dsl.tulsok.sbcglobal.net) |
| 00:48.00 | harlequin516 | Qwell: building twinkle now |
| 00:48.39 | nexgen | anyone have any idea why I would be getting "chan_zap.c: Ignoring caller_id" in my logs, and no Caller ID is not working |
| 00:49.04 | Qwell[] | nexgen: Did you do caller_id=yes, or something? |
| 00:49.10 | Qwell[] | I'm pretty sure it |
| 00:49.12 | Qwell[] | s without the underscore |
| 00:49.47 | nexgen | where at? in my dialplan? |
| 00:49.52 | Qwell[] | zapata.conf? |
| 00:49.57 | nexgen | hmm |
| 00:49.59 | Qwell[] | or zaptel.conf maybe...dunno |
| 00:50.00 | nexgen | lemme see |
| 00:50.24 | Qwell[] | propably the former |
| 00:50.36 | Qwell[] | s/op/ob/ |
| 00:50.57 | St1ckm4n | SpaceBass, We just upgraded our FOP to the latest version but I keep getting client/server version mismatch |
| 00:51.06 | SpaceBass | hummm |
| 00:51.07 | nexgen | I have hidecallerid=no |
| 00:51.09 | St1ckm4n | I moved the original op_server.pl is there any other way of stopping it from running |
| 00:51.10 | SpaceBass | i'e never upgraded it |
| 00:51.11 | SpaceBass | sorry |
| 00:51.34 | St1ckm4n | that ok |
| 00:51.43 | St1ckm4n | I'm sure I'll figure it out |
| 00:52.01 | *** part/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 00:52.35 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 00:54.10 | *** join/#asterisk b66mer (n=b66mer@204.9.61.37) |
| 00:55.50 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-94-53.aubnin.fios.verizon.net) |
| 01:04.14 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 01:06.47 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 01:08.43 | *** join/#asterisk Jon335 (i=Jon335@ottawa-hs-209-217-84-152.d-ip.magma.ca) |
| 01:12.41 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 01:13.44 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:14.38 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 01:16.03 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 01:19.48 | *** join/#asterisk _Soul_ (n=Soul@87-196-33-121.net.novis.pt) |
| 01:21.47 | *** join/#asterisk Jon335 (i=Jon335@ottawa-hs-209-217-84-152.d-ip.magma.ca) |
| 01:22.36 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 01:23.34 | websae | sure is quiet in here |
| 01:23.45 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 01:23.51 | websae | everyone on the phone again |
| 01:24.05 | websae | is that why there are 1000 concurrent calls going through my switch |
| 01:25.03 | brockj49464 | Anybody know if multiple registered peers to the same IP are matched correctly in 1.2.6? |
| 01:29.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:32.19 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 01:36.51 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 01:37.56 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 01:43.37 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 01:43.44 | nick125 | hey everyone |
| 01:47.47 | *** join/#asterisk ateoh211 (n=kalupson@c-68-33-231-126.hsd1.md.comcast.net) |
| 01:49.10 | ateoh211 | hi, I'm brand new to asterisk...I installed xorcom under debian...I set it up on a local net and was able to make and recieve calls via iax softphones...pretty cool |
| 01:49.33 | mog_work | yay |
| 01:50.01 | ateoh211 | I'm wondering if someone can point me to a howto for starters...the thing I would like to use is conference calls(meetme) |
| 01:50.40 | SpaceBass | ~wiki |
| 01:50.48 | SpaceBass | jbot where is the wiki |
| 01:50.51 | jbot | SpaceBass: what are you talking about? |
| 01:51.00 | SpaceBass | i dont know apparently... |
| 01:51.32 | ateoh211 | xorcom aparently comes with 300 as a conference room by default, but I recieve a female voice message that it is not a valid conference room |
| 01:51.44 | SpaceBass | ateoh211 http://www.voip-info.org/wiki/ |
| 01:52.17 | Hmmhesays | so SpaceBass is your name supposed to be bass like the fish or bass like the guitar |
| 01:52.42 | ateoh211 | ok, thanks SpaceBass...I'll be back with more specific questions ;) |
| 01:52.49 | SpaceBass | like the guitar |
| 01:52.55 | Hmmhesays | you play huh? |
| 01:53.02 | SpaceBass | used to quite a bit more than I do now |
| 01:53.05 | SpaceBass | you play guitar, right? |
| 01:53.25 | Hmmhesays | attempt to anyway |
| 01:53.30 | SpaceBass | cool! |
| 01:53.39 | nick125 | Hey, I got a question, I was wondering what you guys would suggest for a linux SIP softphone that can do DTMF correctly and can transfer call (xtensoftphone can't transfer calls)? |
| 01:53.53 | SpaceBass | i travel so much for work these days and don't really have anyone at home to play with....let it lapse for a while and am working to get my chops back currently |
| 01:54.06 | SpaceBass | nick125 google idefisk (think thats it) |
| 01:54.23 | Hmmhesays | same here, i just got done doing speed drills for the last 90 minutes |
| 01:54.43 | nick125 | ooo |
| 01:54.52 | nick125 | *downloads* |
| 01:55.06 | Hmmhesays | http://66.173.103.100:4080/pm.jpg |
| 01:56.05 | Hmmhesays | i finally have a band to play with again, so that helps |
| 01:56.16 | SpaceBass | Hmmhesays nice! |
| 01:56.23 | SpaceBass | i miss gigging so much! what kind of music? |
| 01:56.27 | brockj49464 | Anybody know if multiple registered peers to the same IP are matched correctly in Asterisk 1.2.6? |
| 01:56.32 | nick125 | ./idefisk: error while loading shared libraries: libexpat.so.1: cannot open shared object file: No such file or directory < :( |
| 01:56.45 | SpaceBass | nick125 never tried it on linux |
| 01:56.50 | Hmmhesays | Rock, some country rock |
| 01:56.55 | SpaceBass | Hmmhesays fun! |
| 01:56.56 | Hmmhesays | we're playing lit up by buckcherry there I think |
| 01:57.12 | Hmmhesays | nick125 apt-get install libexpat |
| 01:57.36 | file | Hmmhesays!!! |
| 01:57.39 | Hmmhesays | too bad you don't live here' we are actually short a bassist right now |
| 01:57.40 | Hmmhesays | hey file |
| 01:57.44 | SpaceBass | damn! |
| 01:57.45 | file | how goes? |
| 01:57.59 | Hmmhesays | it goes, just got done strumming on my guitar |
| 01:58.09 | Hmmhesays | now i'm contemplating going to find some honey's |
| 01:58.33 | SpaceBass | funny....i was just contemplaying going to find some beers |
| 01:58.44 | Hmmhesays | it is 35 cent tap night |
| 01:58.52 | SpaceBass | i am so fucking sick of traveling...need to get home and upgrade my asterisk box and chill in front of my tv for a while |
| 01:58.59 | [av]bani | \o/ |
| 02:00.47 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 02:01.09 | CrashHD | if autoload=no does each app module need to be loaded? |
| 02:03.08 | mog_work | yes |
| 02:03.13 | CrashHD | fun heh |
| 02:03.14 | mog_work | if you want to use it |
| 02:03.17 | CrashHD | hey mog |
| 02:03.20 | mog_work | and it comes up in that order |
| 02:03.20 | CrashHD | about earlier |
| 02:03.23 | mog_work | yup |
| 02:03.35 | CrashHD | I'm a little lost as to what I can do with this information |
| 02:03.37 | CrashHD | and what it means |
| 02:03.43 | CrashHD | it's iax2 that is causing the failure |
| 02:03.44 | nick125 | why doesn't idefisk like me :/ |
| 02:03.52 | CrashHD | I just don't why or what to do about it...? |
| 02:04.34 | mog_work | can you do a pastebin |
| 02:04.34 | *** join/#asterisk talljon84 (n=talljon8@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 02:04.34 | mog_work | of thread apply all bt |
| 02:04.34 | CrashHD | bt or bt full? |
| 02:04.35 | CrashHD | ok |
| 02:04.35 | mog_work | bt full will be fine |
| 02:04.35 | mog_work | whatever im just curious |
| 02:05.07 | talljon84 | Is anyone aware of a VoIP provider that will allow multiple outgoing calls at a decent rate? I'd love to create a 'line pool' that * could use as needed to make outbound calls if existing trunks (Zap or SIP) are full. |
| 02:06.02 | CrashHD | telcomone.com |
| 02:06.09 | Hmmhesays | gentoo doesn't like you |
| 02:06.12 | CrashHD | I recently signed up with them |
| 02:06.15 | CrashHD | 1.1 rate |
| 02:06.19 | CrashHD | * based |
| 02:06.33 | CrashHD | been good to me thus far |
| 02:06.36 | Hmmhesays | most termination providers will |
| 02:07.02 | CrashHD | hmm |
| 02:07.03 | CrashHD | oops |
| 02:07.05 | CrashHD | www.telcommone.net |
| 02:07.07 | CrashHD | there we go |
| 02:07.30 | CrashHD | mog_work: this thing has died probably 10 times in the last 4 hours |
| 02:07.37 | CrashHD | hasn't done this until recently :( |
| 02:08.00 | mog_work | which version of asterisk are you using? |
| 02:08.06 | CrashHD | 1.2.4 |
| 02:08.08 | CrashHD | although |
| 02:08.12 | CrashHD | I thought I did a make upgrade |
| 02:08.15 | CrashHD | and a make install |
| 02:08.15 | mog_work | and the other endpoint? |
| 02:08.20 | Hmmhesays | everyone an their brother are starting up an ip phone company now |
| 02:08.40 | CrashHD | 1.0.9 |
| 02:08.40 | mog_work | i have to go to dinner |
| 02:08.46 | CrashHD | but I don't control the 1.0.9 |
| 02:08.47 | mog_work | but put svn 1.2 on |
| 02:08.52 | mog_work | and tell me |
| 02:08.56 | mog_work | what happens |
| 02:09.07 | CrashHD | subversion? |
| 02:09.14 | talljon84 | CrashHD: thanks a ton |
| 02:09.47 | CrashHD | talljon84: no worries, decent termination providers are few and far between these days, sharing is caring lol |
| 02:12.24 | talljon84 | CrashHD: It indicates that it's 1.1 cent /min for termination but it also mentions a $0.33 cent account. Do you know if that's just an activation charge by chance? |
| 02:13.55 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:15.54 | nick125 | Okay, incoming sound works, it's just outbound |
| 02:17.14 | nick125 | fun |
| 02:19.52 | *** join/#asterisk Dabian (n=M0RTEN@fsf/member/dabian) |
| 02:19.57 | St1ckm4n | anyone know of a way to disconnect a manager session from asterisk CLI? |
| 02:20.55 | nick125 | okay, now no sound is coming though.. |
| 02:21.19 | nick125 | is there any known issues with IAX2 and MoH? |
| 02:22.17 | Dabian | nick125 : Is the moon made of green cheese? |
| 02:22.37 | nick125 | depends on who you ask ;) |
| 02:22.39 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 02:22.40 | Dabian | :o) |
| 02:24.40 | Shaun2222 | wiki is driving me crazy with all the crap in it, isnt their a good official up2date manual some where? |
| 02:25.13 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 02:25.23 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 02:25.31 | PakiPenguin | morning |
| 02:26.00 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 02:27.48 | nick125 | anyone else here using idefisk and have MoH working correctly |
| 02:28.00 | nick125 | when I try to listen to my MoH in idefisk, I get nothing |
| 02:28.25 | *** join/#asterisk meshuga (i=meshuga@c-67-160-86-86.hsd1.wa.comcast.net) |
| 02:28.33 | meshuga | so is callwaiting=no gone in sip.conf now? |
| 02:28.41 | meshuga | it doesnt appear to work for me |
| 02:30.27 | Darwin35 | 1.2.6 sucks |
| 02:30.40 | Dabian | sucks |
| 02:31.11 | Dabian | ? |
| 02:31.30 | Darwin35 | having issues with it and realtime |
| 02:32.46 | russellb | pebkac |
| 02:33.36 | talljon84 | haha |
| 02:34.48 | *** join/#asterisk froguz (i=froguz@67-135-222-201.adsl.terra.cl) |
| 02:36.07 | Darwin35 | wow they are making it so you can turn off call waiting per phoone |
| 02:36.45 | Darwin35 | they need to make it a percall turn off also |
| 02:36.51 | nick125 | Okay, so, any ideas on a good linux SIP phone? IAX doesn't like me too much.. |
| 02:37.03 | Darwin35 | iax rocks |
| 02:37.12 | froguz | can AsterFax recieve a fax and convert it to tiff or pdf? |
| 02:37.12 | Darwin35 | kphone |
| 02:37.22 | nick125 | Kphone, last time I checked, DTMF did not work..xtensoftphone you can't do transfers in |
| 02:37.54 | Darwin35 | linphone |
| 02:38.15 | froguz | nick125, you can do blind transfers using xten |
| 02:38.18 | Dabian | Whats the best codec? |
| 02:38.25 | justinu | lpc10 |
| 02:38.25 | nick125 | froguz: how? |
| 02:38.35 | Darwin35 | ilbc |
| 02:38.40 | nick125 | gsm |
| 02:38.42 | froguz | pressing pound |
| 02:38.46 | Darwin35 | speex |
| 02:38.49 | Dabian | if bandwidth is no consideration? |
| 02:38.54 | nick125 | ulaw |
| 02:39.00 | nick125 | for quality |
| 02:39.05 | Dabian | g711u? |
| 02:39.14 | froguz | and then the extension you want to transfer to |
| 02:39.20 | Dabian | (or g711a ?) |
| 02:39.24 | nick125 | froguz: I got to try that |
| 02:39.26 | nick125 | g711u |
| 02:39.30 | Dabian | ok |
| 02:39.59 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:40.00 | froguz | i don't know if is it enabled by default in features.conf, but you can do that |
| 02:41.07 | *** join/#asterisk testshifter (n=testshif@203.172.17.212) |
| 02:41.24 | froguz | Dabian, did you mean better in bandwidth or better in audio quality? |
| 02:41.37 | Dabian | best audio |
| 02:41.46 | testshifter | HELP! SER + Asterisk |
| 02:41.49 | nick125 | froguz: It doesn't seem to work.. |
| 02:41.52 | froguz | ulaw, i think |
| 02:42.15 | froguz | did you looked at features.conf? |
| 02:42.15 | testshifter | kamusta kayong lahat! |
| 02:42.21 | nick125 | froguz: yes, and its enabled |
| 02:42.47 | testshifter | how to confugure SIP Express Router and Asterisk.. Newbie here! |
| 02:42.52 | nick125 | let me just try restarting asterisk just to make srue |
| 02:43.14 | testshifter | how to confugure SIP Express Router and Asterisk.. Newbie here! |
| 02:43.25 | Darwin35 | RTFM |
| 02:43.25 | *** join/#asterisk Strom_C (n=Strom@66.159.243.59) |
| 02:43.38 | testshifter | how to confugure SIP Express Router and Asterisk.. Newbie here! |
| 02:43.49 | testshifter | WHERE IS THE RTFM? |
| 02:43.52 | testshifter | any ref? |
| 02:43.54 | froguz | press the # key during a conversation, you'll hear "transfer", then press the extension you want to transfer |
| 02:44.14 | Darwin35 | RTFM = Read the Fricking Manual |
| 02:44.37 | testshifter | what is a good manual to read |
| 02:44.40 | testshifter | any suggestions? |
| 02:44.57 | nick125 | froguz: That's odd..it doesn't seem to work :( |
| 02:44.59 | Darwin35 | the ser manual and the asterisk wiki |
| 02:45.49 | froguz | nick125 did you put the t or T (or both) parameters in your dialplan? |
| 02:45.57 | testshifter | if i configure ser do i need a hardware or device??? |
| 02:46.07 | nick125 | froguz: ohh...umm...well...umm..no |
| 02:46.16 | Darwin35 | just the hardware it runs on |
| 02:46.18 | nick125 | Where exactly would that go though? |
| 02:46.54 | Dabian | There went nbd .. wonder if his client will reset his nick. |
| 02:46.57 | testshifter | so only the server? |
| 02:47.13 | testshifter | no need to buy physical routers?? |
| 02:47.34 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 02:47.34 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 02:47.41 | Darwin35 | have you even spent time figuring this all out before doing |
| 02:47.44 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 02:48.11 | Darwin35 | you need to read the web sites and all the help/coonfigure info they have before comming here |
| 02:48.23 | testshifter | If i will be installing ser+linux, do i need physical router? or just the server alone works!!! |
| 02:48.30 | testshifter | If i will be installing ser+linux, do i need physical router? or just the server alone works??? |
| 02:48.30 | froguz | nick125, for example Dial(SIP/${EXTEN:1},50,Tt) |
| 02:48.57 | Darwin35 | man you need to read the ser website it tells you whgat you need |
| 02:48.58 | Dabian | Ok .. I assume that G711a is alaw and inferior then. Thanks! |
| 02:49.16 | Darwin35 | so does the asterisk website |
| 02:49.34 | nick125 | froguz: would it be possible to do that in a WaitMusicOnHold? (for testing..) |
| 02:50.11 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 02:50.46 | nick125 | froguz: Also, how would I be able to do that for incoming calls? |
| 02:51.31 | testshifter | sorry for the ignorance, im just a student trying to learn things up! |
| 02:51.59 | Darwin35 | well you should always read the websites and gather information first\ |
| 02:52.09 | Darwin35 | they teach you that in school |
| 02:52.29 | Darwin35 | its called doing research before asking |
| 02:52.42 | testshifter | i had and i just need to know the architecture coz im new to the Telephony world! |
| 02:52.59 | Darwin35 | well the sites tell you what you need |
| 02:53.26 | Darwin35 | and they give you links to sites to help you |
| 02:53.42 | testshifter | thanks though! |
| 02:53.53 | testshifter | signing off! |
| 02:54.06 | konfuzed | hhhmmmm |
| 02:55.06 | froguz | nick125, you don't need to do anything special for doing blind transfer on incoming calls |
| 02:56.35 | froguz | just make sure you have t and/or T in your Dial command for outgoing calls (i think t is for allowing to the called party to make transfers and T for the calling party, or vice versa) |
| 02:58.37 | Dabian | Trying it out on the GUI and check what string it generates might be helpfull? |
| 02:59.01 | nick125 | froguz: That seems to work, but, I want to do some more testing with music on hold, how would I do that? |
| 02:59.05 | nick125 | I don |
| 02:59.15 | nick125 | I don't think you can pass flags the same way |
| 03:00.54 | froguz | nick125, when you press the pound key, the other side will hear the music on hold inmediatly, until the extension you have transfered answers |
| 03:01.11 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 03:01.55 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 03:03.22 | litage | how can you tell Asterisk which RTP ports to use? |
| 03:03.40 | justinu | rtp.conf |
| 03:04.28 | litage | thanks justinu |
| 03:05.40 | froguz | nick125, read this http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf and look for the blindxfer, it's all too clear |
| 03:05.57 | nick125 | Yep, I got that working, thanks :D |
| 03:07.39 | froguz | http://www.voip-info.org/wiki/view/PBX+CallTransfer just in case you want to read more |
| 03:09.13 | Dabian | Where do I learn about PBX, and what do I want PBX hardware for? |
| 03:10.57 | froguz | Dabian, look for the book "asterisk the future of telephony" pdf. you should read voip-info.org too |
| 03:11.02 | froguz | nites everybody |
| 03:15.59 | mishehu | heh, I doubt I could consider any one specific application as the future of telephony. |
| 03:21.56 | *** join/#asterisk rufoz (i=rufoz@200.226.56.54) |
| 03:26.10 | exten123 | Can we rename IAX2/bah in CDR channel column to others charcters instate of IAX2? |
| 03:30.11 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 03:30.59 | CrashHD | is there a way to track down which applications come from which modules? |
| 03:31.28 | litage | what's the difference between an auto-attendant, ivr, and menu? |
| 03:32.21 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:32.45 | CrashHD | all similar? |
| 03:32.50 | CrashHD | all the same really |
| 03:33.07 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 03:33.49 | SwK | that depends on who you ask |
| 03:34.03 | *** join/#asterisk Souvent22 (n=chatzill@c-69-143-189-36.hsd1.va.comcast.net) |
| 03:34.05 | litage | SwK: why's that? |
| 03:34.07 | SwK | auto-attendant/menu are all really the same |
| 03:34.14 | Souvent22 | what's the differnce between a SIP peer, and a SIP user ? |
| 03:34.25 | SwK | in the industry IVR typically referse to something with a DB backend |
| 03:34.45 | SwK | ie: IVR == that automated pile of crap you use to call the bank and get your balance |
| 03:34.58 | file[laptop] | Your balance is $0.00 |
| 03:35.01 | file[laptop] | ut roh! |
| 03:35.03 | SwK | but in asterisk IVR/Auto-Attendant/Menu are all used interchangably |
| 03:35.30 | SwK | file more like "You ballance is.... Bitch you need to go to the bank and pay up" |
| 03:35.45 | file[laptop] | SwK: eep |
| 03:35.48 | CrashHD | ast_func_read: Function GROUP_COUNT not registered |
| 03:35.54 | CrashHD | I'm working on slimming down my asterisk |
| 03:36.08 | CrashHD | but I can't find a resource to determine which .so files contain which functions |
| 03:36.09 | CrashHD | any help? |
| 03:36.24 | file[laptop] | usually by their name you can guess |
| 03:36.57 | *** join/#asterisk delink (i=delink@unerreicht.delink.net) |
| 03:37.02 | loko | has anyone tried running asterisk in Xen? |
| 03:37.11 | delink | haha |
| 03:37.13 | CrashHD | load=app_groupcount.so ; Group Management Routines |
| 03:37.17 | CrashHD | I have it loaded |
| 03:37.20 | delink | people were just talking about that on another channel i am in loko |
| 03:37.20 | CrashHD | but not working |
| 03:37.24 | file[laptop] | app is not a function |
| 03:37.37 | file[laptop] | func_ are functions |
| 03:37.38 | loko | delink yea me too in wplug |
| 03:37.49 | loko | oh isee it in ohio now lol |
| 03:37.54 | delink | loko: yup :) |
| 03:38.04 | CrashHD | only 3 func* files in the whole modules dir |
| 03:38.12 | CrashHD | would one of those contain what I need? |
| 03:38.21 | delink | could anyone enlighten me as to the process of submitting a feature patch to asterisk? |
| 03:39.08 | Darwin35 | wplug is run by a bunch of self centerd linux users. Who are judgemental . |
| 03:39.09 | litage | SwK: ah i see. so they're used interchangeably. i was under the impression that an IVR was an app that allowed users to speak and [attempted] to translate a word/phrase of theirs into an action. Eg: you say "sales" and you're transferred to that particular dept, rather than having to push a button |
| 03:39.30 | CrashHD | interactive voice response |
| 03:39.34 | CrashHD | requires a menu |
| 03:39.36 | CrashHD | to go through |
| 03:39.42 | CrashHD | it all gets mushed together |
| 03:41.21 | SwK | litage: the term IVR originate way before anyone have the speech recognition stuffs that are deployable no |
| 03:41.24 | SwK | w |
| 03:41.24 | loko | Darwin35 get over it |
| 03:41.50 | litage | ah i see. thanks for clearing that up, guys |
| 03:42.18 | loko | Darwin35 where do you live now? |
| 03:42.22 | eipi | i have no audio in this scheme... anyone can help me? VOIP wireless phone <-> a hotspot router <-> internet <-> wrt54gs <-> asterisk 1.2.6 linuxbox |
| 03:42.34 | Darwin35 | Living and working in Denver |
| 03:42.42 | loko | ok |
| 03:42.50 | loko | did you go out there with kryme |
| 03:42.55 | Darwin35 | and the Freebsd user group here is much friendlier |
| 03:43.04 | loko | compared to? |
| 03:43.07 | SwK | eipi: NAT on both ends? |
| 03:43.19 | eipi | yes |
| 03:43.20 | Darwin35 | the wplug group |
| 03:43.29 | SwK | eipi: good luck |
| 03:43.31 | loko | wplug = linux users group, not freebsd users group |
| 03:43.32 | *** join/#asterisk Can0Beans (n=Fart@pool-71-162-14-35.pitbpa.fios.verizon.net) |
| 03:43.41 | Darwin35 | they welcome anyone and block/kickout noone |
| 03:44.02 | Darwin35 | its suppost to be a linux/unix users group |
| 03:44.15 | loko | what about jack |
| 03:44.21 | Darwin35 | our group deals in linux/freebsd/solaris |
| 03:44.33 | Darwin35 | jack is still in PA |
| 03:44.43 | loko | yes i know |
| 03:44.50 | loko | i meant concerning block/kickout |
| 03:45.06 | Can0Beans | didn't WPLUG have a freebsd committer in it's ranks at one time? |
| 03:45.25 | eipi | swk: all hotspots i think that work with nat? |
| 03:45.31 | Darwin35 | he should have never been block from the group and the way beth did it was wrong |
| 03:45.47 | Can0Beans | he was block? |
| 03:45.53 | Darwin35 | thats one of the reasons i left the group |
| 03:45.56 | loko | you just said they block/kickout noone |
| 03:46.05 | loko | now you said they blocked him |
| 03:46.24 | Darwin35 | they kicked him out and told him not to come back |
| 03:46.29 | SwK | eipi: SIP is sorta like active FTP ... nat screws with it or you need a media proxy thats on a public IP |
| 03:46.51 | Darwin35 | and he paid the fee they never refunded me or him |
| 03:47.13 | Can0Beans | That almost sounds criminal |
| 03:47.16 | Darwin35 | i paid min also only to have him tossed 1 week ltr |
| 03:47.43 | Can0Beans | Darwin35, is your keyboard missing letters? |
| 03:48.37 | Darwin35 | not just using shorthand |
| 03:48.52 | eipi | swk: then i have to put my *box in dmz? |
| 03:49.28 | SwK | eipi thats the best thing to do |
| 03:50.35 | riksta | eipi: just forward port 5060 4569 and 10,000 to 20,000 to your asterisk box (all udp) |
| 03:50.37 | CrashHD | anyone notice the line 61 errors for the init script in 1.2.6? |
| 03:50.48 | eipi | ok, thnkx... and iptables always on :D |
| 03:51.06 | CrashHD | ~urnary |
| 03:51.26 | eipi | riksta: i already have before ask here... but no way, can you help? |
| 03:51.35 | *** part/#asterisk Can0Beans (n=Fart@pool-71-162-14-35.pitbpa.fios.verizon.net) |
| 03:51.37 | riksta | eipi: i just told you all you need to know |
| 03:52.07 | eipi | but what about the phone? |
| 03:52.21 | eipi | the server side i know that's correct, but in the phone? |
| 03:52.40 | riksta | same thing, if you get no audio the RTP packets arent getting through( thats the ones from 10000 to 20000) |
| 03:53.37 | riksta | eipi: http://www.voipuser.org/forum_topic_1022.html try this |
| 03:55.06 | SpaceBass | eipi start by limiting the rtp ports in rtp.conf to only like 10 ports... then forward those ports (UDP !!) on the linksys to your * box |
| 03:55.24 | SpaceBass | eipi then in the extension make sure nat=always (i think, or =yes) |
| 03:55.28 | riksta | theres no point in limiting it to 10 imo |
| 03:55.37 | riksta | its the same configuration for 10 or 10000 |
| 03:55.44 | riksta | nat=yes |
| 03:55.50 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 03:55.59 | SpaceBass | riksta makes it a little easier to test...and I've seen the linksys routers have problems with massive port ranges |
| 03:56.10 | SpaceBass | riksta otherwise, you are right...not much point |
| 03:56.22 | riksta | SpaceBass: that's true, now that you mention it, ive seen bugs in linksys firmware too |
| 03:56.29 | SpaceBass | still...I'd never punch 10,000 holes in my nat router :) |
| 03:56.51 | riksta | SpaceBass: its safe enough to do |
| 03:57.41 | SpaceBass | eipi if those settings still don't work, it could be either the hotspot's ISP or your ISP blocking something |
| 03:59.09 | riksta | anyone know what its like to do SIP over openvpn ? |
| 03:59.28 | Dabian | Yes, I met a guy the other day who did that proffessionally. |
| 03:59.28 | SpaceBass | riksta i would suspect it works fine...i need to play with OpenVPN...still using MS PPTP |
| 03:59.44 | Dabian | PPTP is bad, I heard. |
| 03:59.46 | riksta | Dabian: what is the latency like |
| 03:59.59 | SpaceBass | Dabian not bad per se...but its not the best...there are some known voulnerbilities |
| 04:00.09 | riksta | i have an openvpn network here i should try it one day |
| 04:00.19 | Dabian | SpaceBass : I mean for VoIP. |
| 04:00.27 | SpaceBass | I have OpenVPN on my IPcop router...I should try it |
| 04:00.54 | riksta | the windows client is a bit of a bitch |
| 04:00.58 | SpaceBass | Dabian its great for me...I never have problems... I even used it to connect to my * box while on a SAS flight |
| 04:01.07 | SpaceBass | riksta windows client for OpenVPN? |
| 04:01.09 | riksta | SpaceBass: PPTP has nat problems |
| 04:01.12 | riksta | SpaceBass: yes |
| 04:01.24 | rpm | openswan or racoon pwnz. |
| 04:01.35 | Dabian | SpaceBass : You use real SIP VoIP without stun? |
| 04:01.37 | eipi | spacebass: but in my phone i have to configure a stun server? |
| 04:01.38 | SpaceBass | I've been meaning to switch to l2tp or what ever |
| 04:02.04 | SpaceBass | Dabian I've never really figured out what stun is for |
| 04:02.04 | eipi | i tried with or without with no result... no audio |
| 04:02.17 | riksta | eipi: voip-info.org you need to start reading |
| 04:02.23 | Dabian | SpaceBass : I can find link for the RFC if you like. |
| 04:02.47 | eipi | riksta, i already read, but i cant do many tests at office because i dont have two networks... locally works perfectly |
| 04:02.54 | SpaceBass | stun is basically a 3rd party registration service, right? |
| 04:03.17 | eipi | no, its like a address proxy |
| 04:03.22 | eipi | an |
| 04:03.35 | SpaceBass | eipi, I tried at a public hotspot once with a wifi phone... remember it being a bit of a challenge |
| 04:03.48 | *** part/#asterisk talljon84 (n=talljon8@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 04:04.04 | riksta | i have connected to my * box from 1000s of wifi and wired networks around the world and never once had to use STUN or any kind of trickery at all |
| 04:04.19 | eipi | spacebass... you say that i do port forwarding from router to *, and point the wifi phone to the router ip, and that's all? |
| 04:04.45 | SpaceBass | i have a static IP for my * box (still behind nat)...I am not really that concerned about security on it...perhaps I should be...but it works for me |
| 04:05.05 | SpaceBass | eipi thats what I do...simple approach and seems to work |
| 04:05.15 | eipi | ok, ill try again tomorrow |
| 04:05.31 | eipi | but i already tried without results... and nat=yes at general in sip.conf |
| 04:05.53 | SpaceBass | eipi try nat=always in the specific exten |
| 04:06.39 | eipi | ok |
| 04:07.06 | *** join/#asterisk hansin321 (n=chatzill@c-67-174-182-21.hsd1.co.comcast.net) |
| 04:07.24 | SpaceBass | good luck |
| 04:07.27 | SpaceBass | I'm hitting the sac |
| 04:07.29 | SpaceBass | sack |
| 04:07.52 | eipi | ;) |
| 04:08.36 | CrashHD | what .so file contains GROUP()? |
| 04:11.08 | CrashHD | ahh oops |
| 04:11.12 | CrashHD | functions are case sensitive |
| 04:11.13 | CrashHD | duh |
| 04:11.14 | CrashHD | lol |
| 04:13.01 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 04:13.21 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 04:18.00 | CrashHD | anyway to count total calls in a group category? |
| 04:18.33 | CrashHD | so if I do set(group(INBOUND)=${CALLERIDNUM}) |
| 04:18.43 | CrashHD | I would like to count the group and cat as well as total in cat |
| 04:18.45 | CrashHD | possible? |
| 04:22.34 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 04:26.48 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 04:31.56 | CrashHD | so group_count will not accept a reg ex ?? |
| 04:31.59 | Darwin35 | man nanpa is a pain |
| 04:32.21 | Darwin35 | having to reorder my *XX to match thier conf sucks |
| 04:33.04 | CrashHD | ahh nm group_match_count is what I need |
| 04:37.36 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 04:38.15 | exten123 | Do you ppl know How can we know when new version will release everytime? |
| 04:38.35 | Qwell | exten123: You can't |
| 04:38.47 | Qwell | There is a release when a release is needed |
| 04:39.11 | exten123 | who deside the release? |
| 04:39.24 | Darwin35 | yes when they announce it on the website and here |
| 04:40.07 | wasim | and on -announce |
| 04:40.17 | exten123 | do there got any mile stone for there future release? |
| 04:40.25 | exten123 | Wasim, what mean? |
| 04:41.35 | wasim | exten123: http://lists.digium.com/mailman/listinfo/ |
| 04:42.16 | exten123 | wasim,thanks |
| 04:48.17 | kimosabe | how much are you all paying for a t-1 pri with 50 did right know ?? |
| 04:49.43 | CrashHD | ok weird question |
| 04:49.55 | CrashHD | I have answer() and some playbacks |
| 04:49.58 | CrashHD | but they aren't playing |
| 04:50.32 | CrashHD | in the following context: http://pastebin.com/628181 |
| 04:50.36 | CrashHD | any ideas fella's? |
| 04:50.43 | CrashHD | if I comment out the gotoif line they play fine |
| 04:51.04 | CrashHD | it should only play if the criteria is met |
| 04:51.06 | CrashHD | which I meet |
| 04:51.18 | CrashHD | and see in my verbose logs that the system is executing the playback functions |
| 04:51.22 | CrashHD | just nothing heard |
| 04:51.29 | CrashHD | any ideas? this is driving me nuts lol |
| 04:53.57 | CrashHD | *crickets* |
| 04:54.00 | CrashHD | must be dinner time |
| 04:54.00 | CrashHD | lol |
| 04:57.18 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 04:59.41 | CrashHD | so much fun |
| 05:05.02 | kimosabe | some one here that can tell me how to resolve qos isues on difrent link sites please |
| 05:06.42 | *** join/#asterisk BugKham (n=lamer@202.8.86.163) |
| 05:07.50 | *** join/#asterisk Tili (i=Tili@219.137.201.63) |
| 05:07.55 | CrashHD | kim? |
| 05:08.02 | CrashHD | more info |
| 05:08.23 | Tili | why doesn't asterisk support CNG |
| 05:08.33 | Dabian | How did it go? |
| 05:08.43 | Tili | is it possible to have silence detection in asterisk and then send CNG packets instead |
| 05:09.16 | VeNoMouS_ | wtf is cng |
| 05:09.24 | wasim | compressed natural gas |
| 05:09.29 | VeNoMouS_ | lol |
| 05:09.30 | Tili | Comfort Noise Generation |
| 05:09.40 | VeNoMouS_ | Tili stop making shit up |
| 05:09.46 | Tili | wasim: having trouble with gas prices han |
| 05:10.08 | wasim | Tili: diesel is killing me, we have 3 cars, each car does 3000 per week in petrol/diesel |
| 05:10.12 | Tili | VeNoMouS_: what? it is true. There is no Voice Actvity Detection |
| 05:10.33 | VeNoMouS_ | heh i know |
| 05:10.40 | VeNoMouS_ | there is actually |
| 05:10.49 | encode | rofl @ compressed natural gas |
| 05:10.53 | Tili | wasim: yeah, every morning I see long lines of cars on one of gas stations here in China. i think they have timings for providing petrol |
| 05:11.11 | luke-jr_ | wasim: do a veggie oil conversion |
| 05:11.13 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 05:11.28 | wasim | luke-jr_: biodiesel plant is in process of being made |
| 05:11.31 | Tili | VeNoMouS_: yeah, cuz its killing me. it keeps on giving sending empty voice packets which is not good for network. need to do something about it |
| 05:11.32 | VeNoMouS_ | <PROTECTED> |
| 05:11.49 | luke-jr_ | wasim: veggie oil is readily available |
| 05:11.59 | wasim | luke-jr_: we're working with algae and mustard oil strands |
| 05:12.15 | wasim | luke-jr_: veggie oil is also more expensive than petrodiesel here |
| 05:12.41 | luke-jr_ | wasim: most restaurants will give it to you gratis ;) |
| 05:12.51 | VeNoMouS_ | fuck this t4 is doing my head in for this tiff crap |
| 05:12.56 | wasim | luke-jr_: not in pk, even used veggie oil is a commodity here |
| 05:12.59 | Hmmhesays | this weezer song is a bitch |
| 05:13.26 | luke-jr_ | wasim: that sucks |
| 05:13.34 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 05:13.50 | bkw__ | blah |
| 05:13.59 | Qwell | eh? |
| 05:14.09 | orlock | wasim: actually, diesel has a higher amount of joules per weight/volume compared to petrol |
| 05:14.10 | orlock | it should be cheaper to run, maybe |
| 05:15.13 | wasim | orlock: yeah, and its cheaper 60% of petrol cost here too |
| 05:15.22 | Tili | wasim: but we found more gas in Pakistan. so we are safe for next 2 decades or so |
| 05:15.35 | wasim | orlock: but the stupid hog of a pajero still wants 90 liters every 4 days |
| 05:16.42 | *** join/#asterisk angom_h (n=angom@red-corp-200.79.154.31.telnor.net) |
| 05:17.20 | justinu | 3000 what per week? |
| 05:18.34 | Tili | yeah what 3000. it must be pak rupee. 50 USD. |
| 05:18.38 | justinu | ah |
| 05:18.46 | justinu | i get gas once a month, if that |
| 05:18.50 | Tili | wasim: move to UAR |
| 05:18.51 | Tili | UAE |
| 05:19.24 | Tili | UAE is like 50 DHs for petrol in a week. |
| 05:19.35 | justinu | so what does petrol cost in pk? per liter? |
| 05:20.07 | luke-jr_ | maybe $40 |
| 05:20.13 | justinu | we're paying about 3USD per gallon (3.78L) |
| 05:20.27 | Tili | i am away from Pakistan for past few weeks but i think it was something 80 cents a litre |
| 05:20.32 | Tili | 80 USD Cents |
| 05:20.44 | Tili | may be 90 |
| 05:20.50 | justinu | so maybe a bit more expensive than here |
| 05:21.03 | Tili | where are u located justinu |
| 05:21.07 | justinu | los angeles |
| 05:21.49 | Tili | i have heard that they found petrol in sea in Pakistan. but USA stopped building rig there as it was oil flowing from Kuwait to that place under ground somehow. |
| 05:21.57 | bkw__ | justinu, I was there in feb |
| 05:22.01 | bkw__ | for all of 28 hours |
| 05:22.03 | justinu | have any fun? |
| 05:22.17 | Tili | bkw__: lost all the money or made some |
| 05:22.23 | Tili | oh sorry |
| 05:22.28 | justinu | tili: oil is flowing from kuwait to the sea near pakistan? |
| 05:22.40 | justinu | that's a long way |
| 05:22.47 | Tili | yeah |
| 05:22.49 | Tili | it was not in Pakistan |
| 05:22.54 | Tili | it was in arabean sea |
| 05:22.55 | justinu | oh |
| 05:22.59 | bkw__ | I was there to install new gear |
| 05:23.02 | bkw__ | at one whilshire |
| 05:23.07 | bkw__ | er wilshire |
| 05:23.12 | bkw__ | damn I can't type tonight |
| 05:23.12 | *** part/#asterisk St1ckm4n (n=shortes9@68.178.74.166) |
| 05:23.13 | justinu | does that mean asterlink has a west coast pop now? |
| 05:23.23 | bkw__ | we have had gear there for over a year now |
| 05:23.28 | justinu | yeah, but does it work now? |
| 05:23.28 | bkw__ | its not used for Asterlink stuff yet |
| 05:23.37 | bkw__ | yes its not used for that "yet" |
| 05:23.39 | *** join/#asterisk testshifter (n=testshif@203.172.17.212) |
| 05:23.44 | bkw__ | soon grashopper |
| 05:23.49 | justinu | looking forward to that |
| 05:27.35 | Qwell | bkw__: You were here, and didn't stop by? :p |
| 05:28.07 | bkw__ | ya |
| 05:28.14 | Qwell | I see how it is |
| 05:29.34 | *** join/#asterisk docE (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 05:32.50 | testshifter | any howto for beginner? |
| 05:33.14 | wasim | ~voip-info |
| 05:33.16 | jbot | hmm... voip-info is 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 |
| 05:33.22 | Dabian | testshifter : why? why? |
| 05:33.47 | testshifter | because im planning to test/install |
| 05:35.15 | Dabian | moon: Yeah .. but you gotta make sure you got the money to pay for both your books, the lectures, the room you must live in, and your food, clothes etc. |
| 05:35.56 | wasim | beer |
| 05:36.10 | testshifter | Do i need to buy additional devices or just PCs will do?? |
| 05:36.10 | Dabian | Exactly! |
| 05:36.15 | b66mer | if my DIDs are 9700-9710 can I no longer do 9 for outbound calls? |
| 05:36.35 | Dabian | testshifter : Have you read hackers howto? |
| 05:36.39 | wasim | b66mer: contexts |
| 05:36.47 | testshifter | not yet.. where is it located? |
| 05:36.51 | testshifter | the hackers howto?? |
| 05:36.55 | b66mer | thats what I thought! thanks! |
| 05:36.57 | Dabian | google "howto become a hacker" |
| 05:37.29 | testshifter | hmmnnn.. asterisk docs? |
| 05:37.41 | wasim | ~docs |
| 05:37.42 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 05:38.36 | testshifter | ~docs |
| 05:38.37 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 05:38.59 | Dabian | moon: I am real sleepy. I guess I must try some commands another day (maybe later today) and then do qos-stat and stuff. I still don't understand how many pipes there are .. and how to find out which pipe is better .. but I guess you shape how big it is. |
| 05:41.34 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:45.25 | *** join/#asterisk _Soul_ (n=Soul@87-196-33-121.net.novis.pt) |
| 05:47.17 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 05:52.51 | Faithful | does GSM have a G code ? like G729? |
| 05:54.24 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 05:56.19 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 05:56.42 | wasim | Faithful: yes, GSM |
| 05:59.21 | VeNoMouS_ | lol |
| 06:00.03 | VeNoMouS_ | man i think the only way around this corrupt tif shit from rxfax is to rewrite spandsp to read the page length and if its 0 rewrite the tif page index |
| 06:00.18 | VeNoMouS_ | FUN!@!$#@!!! |
| 06:01.45 | *** join/#asterisk forao (n=dfasdfs@ool-4354d6b4.dyn.optonline.net) |
| 06:02.17 | VeNoMouS_ | s/work/world/ |
| 06:04.27 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 06:05.57 | Faithful | wasim: enlightening... |
| 06:06.25 | Faithful | I am shocked then that a sipura-3000 does not support GSM |
| 06:19.50 | iGotNoTime | does anyone know where the TFTP root directory is, I am on my 7th page of google results :( |
| 06:20.03 | Qwell | iGotNoTime: /tftproot/? |
| 06:20.23 | iGotNoTime | no wonder it is not on google :( |
| 06:20.52 | iGotNoTime | thank you qwell |
| 06:21.07 | Qwell | find / -name '*tftp*' |
| 06:21.18 | Qwell | -type d, for bonus points |
| 06:22.22 | VeNoMouS_ | <iGotNoTime> does anyone know where the TFTP root directory is, I am on my 7th page of google results :( |
| 06:22.24 | VeNoMouS_ | lol |
| 06:22.25 | iGotNoTime | i wrote that down! |
| 06:22.26 | VeNoMouS_ | where eva u set it |
| 06:22.51 | iGotNoTime | I didn't set it yet, it is default install right now |
| 06:24.31 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 06:24.43 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:25.08 | kmilitzer | Morning everyone ... |
| 06:25.08 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 06:29.31 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 06:31.04 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-106.claranet.co.uk) |
| 06:35.03 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 06:35.35 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 06:36.55 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 06:37.52 | Shaun2222 | with asterisk what is really required in /etc/asterisk for it to work. |
| 06:38.02 | Shaun2222 | make samples installed about 50 diffrent confs |
| 06:38.14 | Shaun2222 | i want the basics only. |
| 06:39.06 | harlequin516 | You really only need to configure the features you want. (and disable the modules you don't want) |
| 06:39.10 | Qwell | gentoo trunk # du -hs /etc/asterisk/ |
| 06:39.10 | Qwell | 244K /etc/asterisk/ |
| 06:39.18 | Qwell | I'm gonna go ahead and say that it doesn't really matter... |
| 06:39.37 | Shaun2222 | Qwell: talking about files not size. |
| 06:39.49 | Shaun2222 | the samples just have too much crap |
| 06:39.56 | Shaun2222 | makes it hard to understand it all |
| 06:40.59 | Shaun2222 | does asterisk read configs based on name or does it just go and read anything in that folder or with the .conf extention? |
| 06:42.11 | Qwell | by name |
| 06:42.20 | Shaun2222 | ok |
| 06:42.34 | Shaun2222 | and what configs are required as minimal? |
| 06:44.26 | mcnobody | Does anyone know Digium TE410P E1 RJ45 pinout? |
| 06:44.42 | Qwell | mcnobody: It isn't RJ45 |
| 06:44.50 | Qwell | It's E1 |
| 06:45.06 | SwK | mcnobody its a standard pinout for T1/E1 CPE applications |
| 06:45.30 | mcnobody | SwK ok. so pins 1,2 and 4,5 are used |
| 06:45.33 | SwK | yes |
| 06:45.50 | SwK | i forget which is tx and rx |
| 06:46.08 | mcnobody | 1,2 are RX |
| 06:46.47 | mcnobody | If I'm right.... =) |
| 06:50.00 | Shaun2222 | what is pbx_gtkconsole.so for? |
| 06:50.10 | Shaun2222 | gtkconsole sounds like it would be for xwindows |
| 06:50.32 | wasim | SwK: http://ss7box.com/userguide.html look at the bottom of the page |
| 06:52.26 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 06:53.27 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:01.36 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 07:08.45 | Shaun2222 | anybody know what this means... "WARNING[10325]: pbx.c:3740 ast_merge_contexts_and_delete: Requested contexts didn't get merged" |
| 07:17.25 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 07:18.14 | konfuzed | some days it rains and some days it pours - my so called buddy just confirmed arrangements with my other so called buddy (currently working an MS Support Line) to put our weblink to our paid support site on support.microsoft.com so that my supposed buddy can tell "unsupportable by microsoft"-callers to call us instead. |
| 07:18.19 | wiseguy_ | hello, i'm using junghanns quadBRI card, ant i'm getting messages 'Ignoring callwaiting SETUP...' |
| 07:18.22 | konfuzed | like I dont have enough to worry about |
| 07:18.56 | konfuzed | now they wanaa send me verybody thats too stupid for microsoft to support |
| 07:19.17 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:19.17 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 07:19.22 | brookshire | wow.. that's possible? |
| 07:20.07 | konfuzed | sure the link placement is very good marketing but i dont think I ever want to answer the phone again |
| 07:21.36 | konfuzed | brookshire, i think thats likely only plausible cause one of my two buddies has been answering that support line for more than 6 months |
| 07:21.59 | konfuzed | with friends like that who needs enemies ;^) |
| 07:22.53 | konfuzed | im currently working on migrating all users I touch over to debian servers and [*]ubuntu workstations |
| 07:24.28 | konfuzed | no doubt im supposed to spit out a free Distributed Virtual Call Centre |
| 07:35.10 | konfuzed | brookshire, hey can you answer stupid microsoft questions? |
| 07:35.27 | *** join/#asterisk apardo (n=apardo@87.218.44.120) |
| 07:36.03 | konfuzed | we could put you in the support queue and pay by the ticket ;^) |
| 07:39.04 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:58.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:04.13 | *** join/#asterisk skeffling (n=chatzill@andrew.1ec.aaisp.net.uk) |
| 08:04.42 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:04.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:05.18 | Shaun2222 | in the iax.conf under each persion you put in their you need a directive called type |
| 08:05.27 | Shaun2222 | i see in most examples they use friend as the type |
| 08:05.35 | Shaun2222 | what are the valid types and whats the purpose of them? |
| 08:06.42 | *** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it) |
| 08:08.13 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:11.26 | *** join/#asterisk Eggplant (i=No@dsl-745.cascadeaccess.com) |
| 08:14.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:16.25 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 08:17.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:20.18 | *** join/#asterisk yoko (i=yokomo@ppp-69-222-79-185.dsl.euclwi.ameritech.net) |
| 08:21.15 | konfuzed | ~type |
| 08:21.36 | konfuzed | well sometimes that works |
| 08:22.05 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:22.05 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:26.14 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:26.14 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:27.12 | *** join/#asterisk apardo (n=apardo@87.218.45.103) |
| 08:27.29 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
| 08:30.31 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 08:32.04 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:32.04 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:43.46 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 08:45.07 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:46.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:49.12 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 08:49.50 | *** join/#asterisk FITA1 (n=fita1@80.77.12.2) |
| 08:52.44 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 08:56.00 | tsume | win 7 |
| 08:56.04 | tsume | whoops :) |
| 08:56.22 | iDunno | close :) |
| 08:57.10 | *** join/#asterisk Astinus- (n=abba@213.167.111.138) |
| 08:59.53 | *** join/#asterisk apardo (n=apardo@87.218.44.114) |
| 09:10.00 | *** join/#asterisk GolobTGG (n=GolobTGG@193.2.154.246) |
| 09:11.11 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) |
| 09:11.39 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 09:27.33 | *** join/#asterisk apardo (n=apardo@87.218.45.10) |
| 09:30.10 | *** join/#asterisk fulgas (n=fulgas@209.8.233.251) |
| 09:30.12 | *** join/#asterisk xtr (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 09:33.35 | *** join/#asterisk batman2 (n=asdfd@ip70-181-90-193.oc.oc.cox.net) |
| 09:33.46 | batman2 | hello |
| 09:33.53 | batman2 | I need help with astesrisk@home, I will pay you for your time. |
| 09:36.51 | luke-jr_ | batman2: wrong channel |
| 09:36.58 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 09:39.38 | wasim | batman2: $2000 per hour or any part thereof ... |
| 09:41.57 | wasim | k/14:40 <batman2> sure no problem |
| 09:42.04 | wasim | un huh ... |
| 09:43.08 | astra^^ | hai i need some help |
| 09:43.11 | *** join/#asterisk Modcuts (n=bob@proporta.gotadsl.co.uk) |
| 09:43.16 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 09:43.34 | *** join/#asterisk batman2 (n=asdfd@ip70-181-90-193.oc.oc.cox.net) |
| 09:44.58 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 09:45.34 | astra^^ | how do i forward the calls commin from a sip server to the same ip |
| 09:45.48 | astra^^ | to same server |
| 09:45.56 | astra^^ | with an tech prefix |
| 09:47.01 | nettie | Hi guys, I have a couple of polycoms phones on my lan, connected to my asterisk server at the colo facility. The asterisk server is then connected to my voip provider using sip protocol. The problem is that when I make a call to the pstn network I dont hear any ring/busy tones and so on.. anyone know what could be wrong please? calling lan to lan is fine. thanx in advance |
| 09:47.44 | fourcheeze | nettie: have you tried a different client? |
| 09:47.56 | fourcheeze | even a softphone to see if that has the same problem |
| 09:47.59 | nettie | ok |
| 09:48.01 | nettie | trying now |
| 09:48.32 | astra^^ | how do i forward the calls commin from a sip server to the same ip |
| 09:49.09 | fourcheeze | astra^^: how do you mean? |
| 09:50.13 | astra^^ | ie :i am getting calls from a sip server to my asterisk |
| 09:50.42 | astra^^ | and asterisk should forward calls with a tech prefix to the same ip |
| 09:51.18 | fourcheeze | you mean the accounts are tech1, tech2 etc |
| 09:51.40 | fourcheeze | or rather the extensions are |
| 09:51.43 | fourcheeze | so you want |
| 09:52.03 | nettie | fourcheeze exaclty the same with xten too |
| 09:52.04 | fourcheeze | exten => _tech.,1,Dial(SIP/someone@someotherserver.com,60,t) |
| 09:52.20 | fourcheeze | nettie: try a different sip provider |
| 09:52.34 | nettie | uhmm |
| 09:52.41 | fourcheeze | if it works with them then you need to talk to your sip provider and ask how they are sending call progress information |
| 09:53.17 | nettie | uhmm I rememeber I read something regarding inband call progress with polycom |
| 09:53.19 | nettie | uhm |
| 09:53.54 | nettie | on the asterisk console I can see the handover |
| 09:54.13 | *** join/#asterisk puzzled (n=yeahrigh@puzzled.xs4all.nl) |
| 09:54.21 | nettie | it's actually passing the call |
| 09:54.23 | nettie | uhmm |
| 09:56.03 | astra^^ | fourcheeze: as before we get usualcalls from 216 |
| 09:56.10 | astra^^ | from an ip |
| 09:56.19 | astra^^ | wjith a prefix say 123 |
| 09:56.21 | Modcuts | whats peoples views on using speex over g729? |
| 09:56.38 | zoa | its better, cheaper |
| 09:56.41 | zoa | but less supported |
| 09:56.44 | nettie | fourcheeze progressinband=no didnt help.. so .. no idea.. I'll try a diff carrier to figure out if it works or not |
| 09:57.09 | astra^^ | and now i should forwared the calls comming from the sip server back to the same server witha new prefix |
| 09:57.30 | nettie | fourcheeze because the mobile phone I'm calling doesnt even recognize the polycom hangup |
| 09:57.37 | fourcheeze | astra^^: just make sure that the incoming calls are in some context |
| 09:57.46 | fourcheeze | and then you can dial wherever you like |
| 09:58.07 | fourcheeze | zoa: I'm not sure I find speex "better" |
| 09:58.14 | fourcheeze | it doesn't sound as good to my ears |
| 09:58.27 | fourcheeze | maybe I have bad ears |
| 09:59.13 | nettie | fourcheeze: yes.. with the other carrier rings |
| 09:59.18 | nettie | fourcheeze: damn! |
| 09:59.28 | fourcheeze | thought as much |
| 09:59.47 | Modcuts | zoa: cheers, thinking of changing our system on speex as i'm getting quality complents |
| 10:00.40 | zoa | it will not make an audible difference for most people |
| 10:00.40 | fourcheeze | Modcuts: complaints about g729? |
| 10:00.40 | zoa | + you need to configure it |
| 10:00.40 | zoa | and that is poorly documented |
| 10:00.40 | fourcheeze | astra^^: so if you have calls from that server in a context you can just do something like |
| 10:01.12 | fourcheeze | astra^^: exten => _tech.,1,Dial(newprefix${EXTEN:4}@otherserver,60) |
| 10:01.23 | astra^^ | yes right |
| 10:01.24 | fourcheeze | or something like that |
| 10:01.50 | X-Gen | Modcuts: dont u need plenty of cpu to handle speex ? |
| 10:02.01 | astra^^ | actually right now the situation is that i am gettin call from an ip and i am fwding the call to different ip |
| 10:02.02 | X-Gen | like...plenty plenty plenty |
| 10:02.11 | nettie | fourcheeze: anything else I can try? |
| 10:02.21 | fourcheeze | nettie: not off the top of my head |
| 10:02.27 | astra^^ | i have a context to recieve calls |
| 10:02.33 | fourcheeze | I'm not an expert on polycoms, as my next question will make clear.... |
| 10:02.34 | astra^^ | from an ip |
| 10:02.36 | nettie | fourcheeze: I tried with other 2 carriers, 1 is fine the other has the same problem |
| 10:02.50 | nettie | fourcheeze: that's a polycom related issue then? |
| 10:02.55 | fourcheeze | not necessarily |
| 10:02.57 | astra^^ | now i need to fwd the calls comin to the context to that same ip with a new prefix |
| 10:03.16 | fourcheeze | nettie: is asterisk between the polycoms and the sip provider? |
| 10:03.17 | *** join/#asterisk Lino` (i=Lino@i577BC54E.versanet.de) |
| 10:03.27 | nettie | fourcheeze: yes |
| 10:03.57 | fourcheeze | see if the provider has a guide on configuration with asterisk |
| 10:04.13 | fourcheeze | astra^^: the same IP as what? |
| 10:04.26 | nettie | fourcheeze I'll check.. |
| 10:04.39 | fourcheeze | OK, I got a polycom 601 |
| 10:04.46 | fourcheeze | seems to only have a bootloader |
| 10:04.53 | astra^^ | fourcheeze: ie wher i am getting calls from (SIP Server) . |
| 10:04.55 | fourcheeze | looking here: |
| 10:04.56 | fourcheeze | http://www.freedomphones.net/polycom/files/ |
| 10:05.01 | fourcheeze | which of those files do I want? |
| 10:05.16 | fourcheeze | ok, so box A calls box B |
| 10:05.26 | fourcheeze | astra^^: and box B wants to send the call back to A |
| 10:05.30 | fourcheeze | with a different prefix |
| 10:05.36 | astra^^ | yep right |
| 10:05.44 | fourcheeze | I think that's what my example does |
| 10:06.06 | fourcheeze | so tech wants to become something else |
| 10:06.16 | fourcheeze | support let's say |
| 10:06.45 | fourcheeze | exten => _tech.,1,Dial(support${EXTEN:4}@boxa,60) |
| 10:06.50 | astra^^ | :) |
| 10:07.22 | fourcheeze | assuming "boxa" resolves to that box |
| 10:07.27 | fourcheeze | you can use the box IP there if you want |
| 10:07.37 | fourcheeze | or are there many box As? |
| 10:07.54 | Modcuts | xgen: maybe i should stick g729, i'm not sure if it's codec or external provider quality issues |
| 10:08.21 | astra^^ | 1 box a |
| 10:08.36 | fourcheeze | ok, so does that work? |
| 10:10.04 | astra^^ | what is the domain wher i can plaste the conf? |
| 10:10.09 | astra^^ | is it pastebin |
| 10:10.35 | *** join/#asterisk apardo (n=apardo@87.218.44.175) |
| 10:10.41 | *** join/#asterisk _Soul_ (n=Soul@87-196-33-121.net.novis.pt) |
| 10:11.10 | fourcheeze | @pb |
| 10:11.12 | fourcheeze | ~p |
| 10:11.14 | fourcheeze | ~pb |
| 10:11.15 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:11.26 | astra^^ | ya right |
| 10:12.58 | fourcheeze | anyone know which firmware to use for a polycom 601? |
| 10:13.41 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:19.18 | astra^^ | fourcheeze:http://pastebin.com/628448 tis is my present conf |
| 10:19.30 | astra^^ | can u please look into it |
| 10:23.04 | nettie | fourcheeze found a manual.. didnt help |
| 10:23.10 | nettie | fourcheeze same issues |
| 10:23.23 | nettie | fourcheeze I'll google a bit around to see if I can see soemthing |
| 10:23.42 | fourcheeze | astra^^: ok, I looked at it |
| 10:24.00 | fourcheeze | nettie: yeah, or maybe phone the provider |
| 10:24.06 | astra^^ | what are the changes to be made |
| 10:25.26 | astra^^ | fourcheeze:so that it meets the present requirement |
| 10:26.16 | RoyK | zoa: ping |
| 10:26.29 | Dabian | [olli] : For me? |
| 10:30.17 | astra^^ | http://pastebin.com/628448 |
| 10:35.06 | joelsolanki | Hello all |
| 10:35.09 | astra^^ | fourcheeze: any idea can u please check and tell me what are the changes .211563 should be prefid and sent bck to that host ip |
| 10:35.20 | astra^^ | http://pastebin.com/628448 |
| 10:35.38 | fourcheeze | astra^^: I don't see how your existing setup works at all |
| 10:35.51 | joelsolanki | I want to change the field in mysql for asterisk-addson which logs cdrs in mysql. |
| 10:35.53 | *** join/#asterisk apardo (n=apardo@87.218.44.118) |
| 10:36.00 | joelsolanki | is this possible by just changing the field. |
| 10:36.15 | joelsolanki | or do i need to change the code in asterisk-adds on ? |
| 10:36.18 | joelsolanki | any hints plz |
| 10:36.19 | astra^^ | existing set up is just mere forwarding calls comin in from a server to a different server |
| 10:37.05 | astra^^ | now i need a set up as calls comin from serverA to Same server A with 211563 prefix |
| 10:37.17 | astra^^ | i get calls with 123 prefix from server A |
| 10:37.46 | astra^^ | change the prefix from 123 to 211563... |
| 10:37.58 | astra^^ | and sent back to server A |
| 10:38.04 | astra^^ | any idea |
| 10:38.11 | astra^^ | please |
| 10:40.49 | fourcheeze | I think I already told you |
| 10:40.56 | fourcheeze | tell me why that doesn't work |
| 10:41.13 | joelsolanki | ? |
| 10:41.22 | astra^^ | it will work can u please paste it in the bin please |
| 10:43.03 | astra^^ | http://pastebin.com/628448 |
| 10:46.32 | fourcheeze | can't you do that? |
| 10:46.52 | astra^^ | i lost what u typed last |
| 10:46.58 | astra^^ | am sorry |
| 10:48.14 | RoyK | ops. asterisk crashed..... |
| 10:48.22 | joelsolanki | ? |
| 10:48.27 | RoyK | http://bugs.digium.com/view.php?id=6831 |
| 10:48.42 | joelsolanki | any plz provide hints |
| 10:49.25 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 10:49.48 | RoyK | joelsolanki: just use the userfield.... |
| 10:50.29 | joelsolanki | Royk: means ? |
| 10:51.00 | joelsolanki | Royk: right now i m using asterisk-addson to log the cdrs in mysql. but i want to rename the fields |
| 10:51.04 | hackeron | hey, how do I get asterisk to allow a group of people to press a flashing button on the phone to pick up another ringing phone? |
| 10:51.15 | RoyK | Set(CDR(userfield)=yourtext) |
| 10:51.16 | joelsolanki | Royk: i tried to rename but after that it doesnt log the call in mysql |
| 10:51.25 | RoyK | hehe |
| 10:51.31 | joelsolanki | :) |
| 10:51.33 | RoyK | why the fsck do you want to rename them???? |
| 10:52.02 | RoyK | asterisk uses hardcoded field names |
| 10:52.06 | joelsolanki | i m setting up the billing system. and my boss needs that this parameters should be renamed for clarity. |
| 10:52.14 | joelsolanki | yes i understand but.... |
| 10:52.15 | RoyK | create a view for your boss |
| 10:52.15 | joelsolanki | :( |
| 10:52.26 | RoyK | or a view for asterisk to work with |
| 10:52.34 | joelsolanki | view ? |
| 10:52.37 | RoyK | given you've got mysql 5, that'll work well |
| 10:52.55 | joelsolanki | i dont have mysql 5. i have older version :( |
| 10:53.08 | RoyK | then your fucked |
| 10:53.10 | RoyK | :P |
| 10:53.11 | X-Gen | mysql supports views ?!?! *gasp* they have come a long way |
| 10:53.34 | RoyK | X-Gen: mysql 5 has ome quite a bit further than the 4.x crap |
| 10:53.40 | joelsolanki | Royk: will the Set(CDR(userfield)=yourtext) will work ? |
| 10:53.54 | RoyK | joelsolanki: not for renaming fields, no |
| 10:54.16 | joelsolanki | oh shit |
| 10:54.32 | fourcheeze | astra^^: http://pastebin.com/628498 |
| 10:54.36 | RoyK | joelsolanki: upgrade to mysql 5, use postgresql or something other real, or hit your boss on the head :) |
| 10:54.37 | X-Gen | db2 rahter |
| 10:54.43 | fourcheeze | astra^^: replace newprefix with you prefix |
| 10:54.59 | joelsolanki | hehe :) i will test the view in mysql5 |
| 10:55.13 | batman2 | which SIP program has on hold feature? |
| 10:55.14 | joelsolanki | Royk: anyway what is view ? |
| 10:55.53 | wasim | joelsolanki: fjords, lots of them ... |
| 10:56.00 | fourcheeze | can anyone advise on a polycom firmware? |
| 10:56.05 | fourcheeze | like which file I want? |
| 10:56.13 | joelsolanki | ok |
| 10:56.19 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 10:56.23 | RoyK | joe: it's a stored query you can treat as a table |
| 10:56.36 | RoyK | joelsolanki: that was for you |
| 10:56.37 | fourcheeze | RoyK: that's starting to sound interesting |
| 10:56.40 | RoyK | joelsolanki: http://dev.mysql.com/doc/refman/5.0/en/create-view.html |
| 10:56.42 | joelsolanki | oh ok. |
| 10:56.49 | fourcheeze | does this mean I can map realtime stuff onto my own schema? |
| 10:56.56 | *** join/#asterisk Hermis (n=guitarug@85.21.204.146) |
| 10:56.57 | RoyK | yes |
| 10:57.00 | fourcheeze | ahhh |
| 10:57.01 | joelsolanki | Royk: thanks. let me study it. :( |
| 10:57.05 | RoyK | fourcheeze: that's the whole point of views :) |
| 10:57.06 | fourcheeze | might have to play with that |
| 10:57.07 | joelsolanki | :) |
| 10:57.20 | RoyK | you just create a view for asterisk to use |
| 10:57.23 | Hermis | Is Asterisk support SSC 7 signalling over E1? |
| 10:57.28 | hackeron | anyone? - what is it even called when you pick up other people's phone? |
| 10:57.36 | RoyK | Hermis: bug wasim about it |
| 10:57.39 | Hermis | sorry CCS 7 |
| 10:57.43 | wasim | Hermis: oui |
| 10:57.46 | astra^^ | fourcheeze:am gettin this message |
| 10:57.51 | astra^^ | Mar 29 04:56:43 WARNING[1059]: app_dial.c:979 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 10:58.04 | wasim | Hermis: you have three options, cosini, xygnada and chan_ss7 |
| 10:58.14 | wasim | Hermis: cosini is the most mature, but most expensive also |
| 10:58.16 | viperdude | hi guys is there anyway to play a file to a called party before the caller gets bridged? |
| 10:58.23 | wasim | Hermis: xygnada is new, and works beautifully |
| 10:58.33 | wasim | Hermis: chan_ss7 is prenatal stage at this point |
| 10:58.34 | astra^^ | ouch ... |
| 10:58.38 | astra^^ | it hurts |
| 10:58.40 | fourcheeze | astra^^: looks like you're missing a SIP/ |
| 10:58.55 | astra^^ | am dailling froma softphone x-lite |
| 10:59.09 | Hermis | 2wasim I need to use E1 in Russia with CCS7 will it work? |
| 10:59.50 | fourcheeze | astra^^: SIP/ as the first bit of your Dial() |
| 11:00.16 | fourcheeze | astra^^: http://pastebin.com/628506 |
| 11:01.46 | wasim | Hermis: yes it does |
| 11:02.28 | *** join/#asterisk ^rage^ (n=cih@194.84.1.237) |
| 11:02.35 | ^rage^ | re |
| 11:02.45 | astra^^ | fourcheeze:chears... u are gr8 |
| 11:02.53 | fourcheeze | yeah, I know |
| 11:03.01 | astra^^ | thank dude |
| 11:03.09 | astra^^ | thank you very much |
| 11:03.23 | astra^^ | even though i get a error message |
| 11:03.35 | astra^^ | Mar 29 05:01:53 NOTICE[1080]: chan_sip.c:9524 handle_response_invite: Failed to authenticate on INVITE to '"muhajir" <sip:1000@64.246.52.52>;tag=as6322f9ff' |
| 11:06.07 | ^rage^ | hey! |
| 11:06.23 | fourcheeze | astra^^: looks like you need a user for the server |
| 11:06.29 | ^rage^ | * can support SS-7 signaling system? |
| 11:07.54 | Dabian | nbd: I guess I can translate the syntax from iptables to ebtables .. but I don't understand how you figure out which devices to use (unless you use the name of the file) |
| 11:08.04 | Dabian | wrong window |
| 11:08.08 | Dabian | Soorry! |
| 11:10.45 | subdolus | ^rage^ pretty much |
| 11:11.06 | subdolus | has support for FXO/FXS cards |
| 11:12.05 | *** join/#asterisk Ahrimanes (n=michael@195.137.237.81) |
| 11:12.24 | Ahrimanes | hm, is there any special configuration needed to allow users to forward voicemail messages to other users? |
| 11:15.08 | astra^^ | fourcheeze:user for my asterisk server or the server which i am forwarding to ? |
| 11:15.37 | fourcheeze | does the server you're forwarding to know to accept incoming calls from the other server? |
| 11:15.45 | astra^^ | yes |
| 11:15.53 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 11:16.00 | astra^^ | but tey are saying they do not have any authentication |
| 11:16.11 | astra^^ | no user name and pwd |
| 11:16.36 | fourcheeze | you'll have to figure that one out |
| 11:16.58 | astra^^ | yep... trying to figure out |
| 11:17.02 | *** join/#asterisk _Soul_ (n=Soul@87-196-33-121.net.novis.pt) |
| 11:17.07 | astra^^ | thanks once again |
| 11:17.26 | astra^^ | no problemif i ask more doubts later on ..right... :) |
| 11:18.21 | mut | i have some fxs cards hooked into a pbx |
| 11:18.30 | mut | is there a way to detect faxes |
| 11:18.41 | mut | and if detected send it out to an fxo line |
| 11:18.50 | wasim | fax() |
| 11:19.07 | wasim | actually, fax,1,() |
| 11:19.24 | wasim | 'tis been a long time since i played with paper thingies |
| 11:19.25 | mut | um |
| 11:20.08 | mut | whers that documented |
| 11:21.07 | wasim | http://www.voip-info.org/wiki-Asterisk+fax |
| 11:25.58 | nettie | hey guys anyone know what couldbe the problem please? |
| 11:25.59 | nettie | Mar 29 13:20:18 WARNING[17954]: chan_sip.c:9552 handle_response_invite: Forbidden - wrong password on authentication for INVITE to |
| 11:26.08 | nettie | im sure the password is correct |
| 11:26.30 | nettie | because asterisk registers to the voip provider with that password |
| 11:27.06 | wasim | sip debug it |
| 11:27.11 | mut | so wasim.. |
| 11:27.19 | mut | i'ed have to record the fax and send it back out |
| 11:27.28 | mut | cause if i answer the line the fax will start sending stuff correct? |
| 11:27.46 | wasim | eh? no |
| 11:27.53 | mut | so i can answer() |
| 11:28.11 | mut | then in my fax exten i can dial zap/1 ${EXTEN} ? |
| 11:28.17 | mut | or how does the number get passed? |
| 11:28.26 | wasim | answer() fax,1,Dial(zap/1/3429348) |
| 11:28.28 | wasim | what number? |
| 11:28.35 | mut | the number the fax machine dialed |
| 11:28.52 | wasim | you picked it up in the answer() |
| 11:28.55 | mut | right |
| 11:29.03 | mut | how does it get passed to the fax exten tho |
| 11:29.10 | mut | need to set a global? |
| 11:29.10 | wasim | var |
| 11:29.14 | mut | or does one exist? |
| 11:29.33 | *** join/#asterisk eipi (n=eipi@OL17-54.fibertel.com.ar) |
| 11:30.05 | RoyK | mut: don't trust wasim, muslim terrorist! |
| 11:30.10 | mut | :P |
| 11:30.42 | wasim | ooh eclipse ... |
| 11:30.53 | fourcheeze | Ooooh clouds... |
| 11:31.37 | nettie | sip debug doesnt show anything interesting :( |
| 11:31.58 | RoyK | wasim: is it total down there? |
| 11:33.39 | *** join/#asterisk duckz (n=duckz@193.192.47.26) |
| 11:34.45 | *** join/#asterisk smeevil (n=smeevil@gateway.office.sod.nl) |
| 11:34.51 | smeevil | hello |
| 11:34.59 | *** join/#asterisk txtNation (n=Kazuki@82-33-205-227.cable.ubr11.newt.blueyonder.co.uk) |
| 11:35.14 | txtNation | Anyone had any luck compiling Asterisk on a Solaris 10 SPARC 64 machine? |
| 11:35.47 | Dabian | txtNation : Hard to say, unless there are some listening in now, that actually did it themselves. |
| 11:35.53 | Ahrimanes | txtNation: what os? |
| 11:36.15 | Dabian | Ahrimanes : "*Solaris 10* sparc 64 machine" ... |
| 11:36.27 | Ahrimanes | ah |
| 11:36.29 | Dabian | :D |
| 11:36.45 | Ahrimanes | no, i think someone i know did it on a freebsd sparc 64 tho |
| 11:36.59 | txtNation | I'm fairly new to Solaris, so I'm not too sure on what to look for. |
| 11:37.09 | txtNation | Unless in my sleep-deprived state, I'm missing a flag for a 64-bit compile. |
| 11:37.24 | Dabian | txtNation : You could dump solaris and pour some freebsd on the disk? |
| 11:37.28 | wasim | dammed mice, first they were after the moon, and now the sun too ... |
| 11:37.41 | txtNation | Dabian, got to stick with Solaris 10 for compatiblity reasons. |
| 11:38.04 | Ahrimanes | wasim: huh? |
| 11:38.10 | Dabian | txtNation : You'll get your donkey busted if you waste the Solaris? |
| 11:38.18 | txtNation | Dabian, precisely. :P |
| 11:38.25 | Dabian | ok :) |
| 11:38.26 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 11:38.52 | txtNation | "Ah, you've fubared our $15k piece of equipment. Now how would you like to be castrated?" |
| 11:39.10 | GolobTGG | hi all, does anyone have any thoughts about cisco callmanager (express) vs asterisk? our company will be deploying voip and some people here are cisco fans |
| 11:39.19 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:39.24 | txtNation | Dabian, got another question. |
| 11:39.41 | txtNation | We've got a Digium TDM400P, 1 incoming and 3 extensions. |
| 11:39.48 | txtNation | Can we just utilize softphones to allow for more extensions? |
| 11:40.09 | wasim | oui |
| 11:41.05 | Dabian | txtNation : As our friendly muslim terrorist say, i don't see why not. I am prolly not the right person to ask about that. I seldom use softphones. |
| 11:41.18 | txtNation | Ditto, but we're waiting on an order of phones. ¬_¬ |
| 11:41.26 | Dabian | txtNation : Besides, I don't have a Digium TDM400P ;-) |
| 11:41.40 | txtNation | I'm not trained to deal with VOIP or Asterisk in the least, yet somehow it got deferred to me. |
| 11:41.43 | txtNation | ;_; |
| 11:41.54 | GolobTGG | phones are just softphones in a nice hw package anyway... |
| 11:42.13 | wasim | or not so nice ... case in point barbietone ... |
| 11:42.41 | GolobTGG | point taken |
| 11:43.04 | txtNation | Oh yeah, it's the eclipse. |
| 11:43.44 | Ahrimanes | heavy cloud coverage here.. so cant see it |
| 11:44.06 | txtNation | I'm in South West England, a day without cloud coverage is a rarity. |
| 11:44.25 | Ahrimanes | same here in denmark atm it seems |
| 11:45.05 | txtNation | Oh for the love of God, now they want to me to implement Bulk SMS Wap Push. |
| 11:45.12 | iDunno | the ecclipse has been and gone in England, hasn't it... it was at 10.45 or something? |
| 11:45.41 | smeevil | could someone please tell me why i do not hear anything when using musiconhold ? in asterisk i see : -- Executing Answer("SIP/668-440c", "") in new stack , -- Executing MusicOnHold("SIP/668-440c", "") in new stack ,-- Started music on hold, class 'default', on channel 'SIP/668-440c' .....but there is no sound |
| 11:45.50 | txtNation | iDunno, see your name. :P |
| 11:46.40 | smeevil | in musiconhold.conf i have mode=mp3 directory=/usr/share/asterisk/mohmp3 (which contains some mp3 files) and random=yes |
| 11:46.40 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:46.56 | caio1982 | ~seen coppice? |
| 11:47.08 | jbot | coppice <n=chatzill@3.143.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 3d 4h 43m 1s ago, saying: 'why don't you just get some speakerphones?'. |
| 11:47.08 | wasim | smeevil: read perms? |
| 11:47.08 | smeevil | hmmm |
| 11:47.08 | smeevil | might be , hold on |
| 11:47.53 | caio1982 | hmm :( |
| 11:48.04 | smeevil | wasim, nope permissions are correct |
| 11:48.16 | smeevil | wasim, only see one warning Mar 29 14:47:28 WARNING[9189]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
| 11:48.39 | wasim | smeevil: yeh, you need a timing device, like a zap card or ztdummy |
| 11:49.09 | smeevil | i have capi , (hfc-s) how can i use those ? |
| 11:49.27 | smeevil | wasim, i mean, they are working :) but how can i tell musiconhold to use them for timing ? |
| 11:49.29 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 11:50.36 | RoyK | zoa: hello? |
| 11:50.51 | caio1982 | has someone here ever seen "switching equipment congestion" using unicall with mfc/r2 protocol? i used a radcom performer to test the signalling and the machine just freezes (asterisk process) until the error appears in the log |
| 11:51.27 | caio1982 | maybe radcom is sending too much data over the network to my machine, i dont know... |
| 11:52.15 | wasim | smeevil: it should automatically, afaik |
| 11:52.24 | wasim | smeevil: try adding a ztdummy also |
| 11:53.16 | txtNation | Memory: 8184M real, 6834M free, 195M swap in use, 8476M swap free - Yargh. :) |
| 11:53.37 | smeevil | that did work yes, though still no music. if i use MP3Player , then it does work |
| 12:05.19 | eipi | i think that's a common question: how i can change the pager email from? |
| 12:10.18 | fourcheeze | what's the default gui login for a polycom? |
| 12:11.32 | _4d4m_ | hi all.. we have an isdn30 and a traditional pbx with 20 phones, and are looking at voip-enabling. we want to integrate our old system with our IP network rather than replace the old one. An * server will then be hosted remotely to manage all calls/extensions. Any suggestion on the harwdare we should be looking at to handle the integration? Any help appreciated. |
| 12:13.48 | Dabian | pbx is like a switchboard, right? |
| 12:13.59 | wasim | _4d4m_: a digium or sangoma e1 card, a pc |
| 12:14.33 | fourcheeze | some utp cables |
| 12:14.35 | fourcheeze | phones |
| 12:14.39 | fourcheeze | coffee |
| 12:14.41 | wasim | _4d4m_: get a 2e1 card, plug the ISDN30 into 1 port, and the PBX into the other port |
| 12:14.43 | fourcheeze | more coffee |
| 12:15.38 | _4d4m_ | thanks all.. am looking at some of the product info online now.. |
| 12:16.51 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 12:20.10 | *** join/#asterisk billings (n=billings@pdpc/supporter/active/billings) |
| 12:20.41 | *** join/#asterisk |cleric| (n=dacleric@p5482BBCA.dip0.t-ipconnect.de) |
| 12:21.03 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 12:21.44 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
| 12:25.00 | skeffling | Anyone know what "Write to 80 failed: Unknown error 500" and Short write: 0/15 (Unknown error 500) means? -this happens jsut before yellow alarms detected |
| 12:25.19 | skeffling | ...using a TE410P |
| 12:25.58 | nettie | fourcheeze I'm having authentication issues on outgoing calls. I'm sure username and password are fine because I can register properly. Do you know if there's a debug switch to see what's really happening ?? any idea? |
| 12:26.23 | fourcheeze | sip debug peer |
| 12:26.35 | fourcheeze | and put the name of your peer after that |
| 12:27.29 | _4d4m_ | wasim: so E1 -> PC w/ TE210 + * -> old PBX. This would allow us to effectively manage ingress/egress without needing to modify the old architecture seamlessly? |
| 12:29.20 | nettie | fourcheeze did that.. cant see anything interesrting |
| 12:29.20 | nettie | uhmm |
| 12:29.37 | fourcheeze | you could a tcpdump |
| 12:29.45 | *** join/#asterisk sambal (n=ivo@sd511723c.adsl.wanadoo.nl) |
| 12:29.45 | fourcheeze | ehtereal |
| 12:29.46 | fourcheeze | whatever |
| 12:30.18 | *** join/#asterisk cfh (n=luca@82.193.23.6) |
| 12:30.25 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
| 12:31.10 | smeevil | what is the best way to debug musiconhold ? |
| 12:31.23 | RoyK | http://koti.mbnet.fi/peku3/celebrities.gif |
| 12:31.41 | wasim | _4d4m_: yeah |
| 12:31.47 | nettie | fourcheeze I just ee a 403 forbidden message |
| 12:31.53 | Ahrimanes | anyone feel like patching format_mp3 so that it can do streams as well? |
| 12:31.53 | wasim | _4d4m_: ofcourse my service charges also! |
| 12:32.11 | wasim | Ahrimanes: app_ices? |
| 12:32.12 | nettie | also 401 unauthorized |
| 12:32.14 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 12:32.27 | Ahrimanes | wasim: oh it's there? in addons? |
| 12:32.59 | sambal | did anyone ever used chan_ss7? |
| 12:33.20 | wasim | sambal: partially, but its not ready for a real network yet |
| 12:33.39 | x86 | morning |
| 12:35.20 | fourcheeze | RoyK: wtf is dakota fanning? |
| 12:35.21 | Ahrimanes | sambal: hm app_ices relies on external application like moh used to do with mpg123 ? |
| 12:35.58 | tzanger | hmm |
| 12:36.03 | tzanger | what ring # on the polycoms is silent? |
| 12:37.34 | _4d4m_ | wasim: cheers mate.. much appreciated! |
| 12:38.18 | x86 | what does app_ices do exactly? where does it stream to? |
| 12:38.27 | x86 | a phone? |
| 12:38.59 | Ahrimanes | x86: whoever is at the other end of the channel you start app_ices on |
| 12:39.32 | x86 | so why use it instead of something like Playback or Background? |
| 12:40.07 | Ahrimanes | to stream from an icecast server? |
| 12:40.28 | x86 | ah, it acts as a receiver then |
| 12:40.39 | fourcheeze | on the polycoms in the sip register section, what does it want in the "address" box? |
| 12:40.44 | x86 | or, proxy almost ;) |
| 12:40.55 | Ahrimanes | x86: it receives a stream from an icecast server and plays it to the caller |
| 12:41.28 | fourcheeze | there seem to be too many places to put a registrar/proxy in |
| 12:44.18 | pigpen2 | Question: When I check my voicemail from my polycom 601 (via speaker phone) the voicemail attendant says "assword." Any way to increase the lead time of this prompt? I am doing a deployment for a Church.....hehe |
| 12:45.36 | fourcheeze | pigpen2: rerecord and click your fingers before the word starts |
| 12:45.38 | nettie | pigpen2 lol!!!! |
| 12:45.38 | x86 | hahaha |
| 12:46.07 | pigpen2 | yeah....that is what I was thinking... |
| 12:46.28 | pigpen2 | at least the people here have a good sense of humor (Baptists) |
| 12:47.03 | pigpen2 | hmm...maybe I can play a 1 sec of silence to bring the speaker phone up... |
| 12:47.04 | x86 | at least they're good for something :P |
| 12:48.01 | *** join/#asterisk pengyong (n=lala@222.185.17.29) |
| 12:48.20 | ManxPower | ugh. mornings. evil. |
| 12:48.30 | pigpen2 | Get your coffee.... |
| 12:48.59 | nettie | fourcheeze everything seems to be fine .. but I still cant dial out.. everytime Forbidden.. do you think could be a specific asterisk configuration issue with that ISP? |
| 12:49.21 | fourcheeze | nettie: what exactly are you doing at that point? |
| 12:49.40 | smeevil | grmbl, why does MP3Player work fine, and MusicOnHold only gives silence, no matter if it gets mp3s or raw files..... |
| 12:49.47 | nettie | hey ManxPower gmorning |
| 12:50.13 | nettie | ManxPower all nat issues have been solved |
| 12:50.16 | nettie | :) |
| 12:50.26 | ManxPower | nettie, how did you solve them? |
| 12:50.46 | nettie | ManxPower I think I had a bad sip.conf |
| 12:51.06 | nettie | I didnt figured out exactly but I wrote a new one from scratch |
| 12:51.16 | nettie | and all the problems went away |
| 12:51.37 | pigpen2 | gotta love sip |
| 12:51.39 | *** join/#asterisk CleanerX (n=nix@p54A3AEDF.dip0.t-ipconnect.de) |
| 12:51.59 | nettie | now I'm bitching with "call progress" seems my carrier is not passing it correctly.. |
| 12:52.20 | ManxPower | nettie, analog or digital? |
| 12:52.33 | *** join/#asterisk coppice (n=chatzill@35.201.17.210.dyn.pacific.net.hk) |
| 12:52.36 | nettie | fourcheeze suggested me to try with a different carrier.. I tried and it works .. so I think that could be a specific conf issue |
| 12:52.58 | nettie | polycom->ASTERISK->SIPPROVIDER |
| 12:53.47 | ManxPower | oh! a sip provider. |
| 12:54.03 | ManxPower | nettie, make sure you have an /etc/asterisk/indications.conf |
| 12:54.22 | nettie | uhmm |
| 12:54.26 | nettie | indications.. |
| 12:54.30 | nettie | let's see :) |
| 12:54.53 | nettie | it's there.. country code is wrong.. says us :) |
| 12:55.07 | nettie | the default one.. |
| 12:55.14 | nettie | do you think will make differences? |
| 12:56.01 | ManxPower | nettie, no. |
| 12:56.18 | nettie | well |
| 12:56.24 | nettie | those are there |
| 12:56.26 | nettie | uhmm |
| 12:57.25 | ManxPower | nettie, also be sure not to use the "r" option to Dial |
| 12:58.21 | nettie | not using it.. |
| 12:58.22 | nettie | [outbound-enerjetica] |
| 12:58.22 | nettie | exten => _0.,1,Dial(SIP/enerjetica-out/${EXTEN:1}) |
| 12:58.22 | nettie | exten => _0.,2,Congestion() |
| 12:58.22 | nettie | exten => _0.,102,Congestion() |
| 12:58.39 | tzanger | why does everyone use congestion like that... |
| 12:58.45 | tzanger | it's so... unfriendly |
| 12:59.02 | nettie | tzanger I'm open to suggestion :) |
| 12:59.26 | tzanger | nettie: Dial, use 'g', then wait 30 seconds, then indicate congestion |
| 12:59.41 | smeevil | is it possible to use musiconhold for sip channels ? |
| 12:59.42 | tzanger | also dial jumps to n+101 with option j and for many reasons, not just congestion |
| 12:59.47 | ManxPower | smeevil, yes |
| 12:59.52 | smeevil | hmmm |
| 13:00.02 | tzanger | so parse out the DIALSTATUS and play the correct indication (busy, congestion, SIT, etc.) |
| 13:00.19 | smeevil | ManxPower, trying to figure out why everything seems to be fine, but only thing i hear when MusicOnHold runs is silence |
| 13:00.25 | tzanger | nettie: I have macro for that actually... so my Dial() is usually Dial(,g) and then Macro(handle-hangup) |
| 13:00.33 | ManxPower | there are samples of using DIALSTATUS in macro-stdexten in extensions.conf.sample |
| 13:00.38 | tzanger | but that's beside the point... you're not getting proper tones from your provider |
| 13:00.47 | nettie | exacly |
| 13:00.48 | nettie | eheh |
| 13:01.15 | nettie | I'll definitely move in the "fine tuning area" as soon as I get this thing do basic stuff properly :) |
| 13:01.42 | tzanger | I was just commenting on how people almost universally configure the PBX to blast congestion at every opportunity |
| 13:01.50 | nettie | eheh |
| 13:02.05 | ManxPower | we all hate our users, that's why., |
| 13:02.10 | nettie | I appreciated your input .. |
| 13:02.12 | tzanger | besides being annoying 99.99999999% of people I deal with say "all I get is busy" |
| 13:02.16 | tzanger | ManxPower: haha |
| 13:02.26 | nettie | yeah it's to give them less options |
| 13:02.27 | nettie | ehehe |
| 13:02.32 | Dabian | :-) |
| 13:02.34 | tzanger | IT'S NOT BUSY, IT'S A FAST BUSY. IT'S CONGESTION. |
| 13:03.07 | ManxPower | "Is it a busy or a fasy busy?" *blank stare* *silence* |
| 13:03.28 | nettie | the othe rissue I have is related to authentication of outgoing calls |
| 13:03.49 | Dabian | *pull shotgun* * B L A M * |
| 13:03.52 | nettie | it keeps saying the password is wrong and I get 401 and 404 error codes |
| 13:03.59 | tzanger | ManxPower: yep |
| 13:04.11 | nettie | of course it registers properly with the supplier password. |
| 13:04.11 | coppice | tzanger: is the congestion caused by H5N1? |
| 13:04.21 | tzanger | coppice: no, death is caused by H5N1 |
| 13:04.28 | tzanger | and coughing up feathers |
| 13:04.34 | nettie | that's pretty strange too |
| 13:04.56 | tzanger | nettie: get a sip debug pastebinned |
| 13:05.13 | tzanger | nettie: 404 is "can't get there from here" and 401 is "you're not allowed to get there from here" |
| 13:05.14 | nettie | sure |
| 13:05.23 | nettie | seems I get both |
| 13:05.24 | nettie | ehehe |
| 13:05.35 | nettie | and that's only with a particular carrier.. |
| 13:05.40 | tzanger | both seem to indicate that you're either in the wrong context or not supplying the number correctly |
| 13:05.42 | *** join/#asterisk hwt (n=hwt@82.117.37.14) |
| 13:05.52 | tzanger | I'd call that particular carrier and bitch at them :-) |
| 13:06.01 | nettie | ehehe |
| 13:06.21 | hwt | hey, can someone point me in direction to a few _very compact_ extensions.conf examples? |
| 13:06.31 | tzanger | hwt here's a very compact one: |
| 13:06.32 | wasim | hwt: .,1,Hangup() |
| 13:06.36 | tzanger | exten => _X.,1,Hangup |
| 13:06.49 | tzanger | wasim: won't work, you didn't use _ to indicate pattern matching :-) |
| 13:06.51 | *** join/#asterisk Z0m81e (n=support@66.77.187.81.in-addr.arpa) |
| 13:06.52 | wasim | great minds think alike ... |
| 13:06.54 | hwt | or, a bundle of configuration files without much. |
| 13:07.26 | tzanger | hwt: what are you trying to accomplish? That's a more useful question and one we can help with |
| 13:07.47 | hwt | tzanger: okay. i want to be able to dial between phones on the lan (just experimenting now). |
| 13:07.59 | tzanger | hwt: ok... well you can do that with Dial() and nothing more |
| 13:08.00 | hwt | tzanger: they are added to sip.conf and is able to register and dial 1000 and 500. |
| 13:08.02 | Z0m81e | hello all, can anyone tell me why our asterisk console spams Mar 29 14:05:15 NOTICE[29351]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 0! |
| 13:08.27 | hwt | tzanger: and now i want to be able to dial between them. i've added a few simple dialplans in the [local] context, and still get 404. |
| 13:08.46 | fourcheeze | is there some secret to getting the polycoms to use g729? |
| 13:08.48 | hwt | tzanger: exten => 5678,1,Dial(SIP/5678,10) |
| 13:08.51 | fourcheeze | I've selected it as "first" |
| 13:08.53 | tzanger | hwt: what does your sip.conf look like for [1000] and [500]? (use pastebin.ca) |
| 13:08.55 | hwt | tzanger: and more. |
| 13:09.22 | tzanger | ManxPower: nettie's pastebin is at http://pastebin.com/628668 |
| 13:09.25 | smeevil | could anyone please tell me where to look when you see -- Executing SetMusicOnHold("SIP/668-a29b", "default") in new stack , -- Executing WaitMusicOnHold("SIP/668-a29b", "30") in new stack, but hear nothing |
| 13:09.27 | tzanger | nettie: line 3 is you rproblem |
| 13:09.34 | tzanger | it's pretty straightforward :-) |
| 13:09.55 | nettie | leme read |
| 13:09.59 | hwt | tzanger: http://pastebin.ca/47434 <-- sip.conf |
| 13:10.17 | nettie | tzanger nah |
| 13:10.29 | nettie | tzanger that's just a client which is not defined |
| 13:10.39 | nettie | tzanger ignore it please |
| 13:10.47 | hwt | tzanger: http://pastebin.ca/47435 <-- extensions.conf |
| 13:11.41 | tzanger | hwt: what context is [5678] and [5679] in? There is no context= line in their config, which means (unless I'm mistaken) that they end up in [default], which is not what you want |
| 13:12.00 | tzanger | nettie: hmm |
| 13:12.06 | hwt | tzanger: i just specifu [context] above those lines? |
| 13:12.09 | *** join/#asterisk stoffell (n=stoffell@d5153FC35.access.telenet.be) |
| 13:12.34 | tzanger | hwt: no, you say context=somewhere |
| 13:12.42 | tzanger | where somewhere is [somewhere] in extensions.conf that you want them |
| 13:12.44 | astra^^ | hello all |
| 13:12.56 | tzanger | hwt: in this case (for simplicity), say context=local for each |
| 13:12.59 | astra^^ | Mar 29 05:01:53 NOTICE[1080]: chan_sip.c:9524 handle_response_invite: Failed to authenticate on INVITE to '"muhajir" <sip:1000@64.246.52.52>;tag=as6322f9ff' |
| 13:13.01 | tzanger | hwt: and issue a sip reload |
| 13:13.11 | astra^^ | i get tis message wen i dail any number |
| 13:13.31 | astra^^ | http://pastebin.com/628674 => sip and ext cof in heree |
| 13:13.31 | hwt | tzanger: okay, that worked. thanks. |
| 13:14.00 | hwt | tzanger: what is the Right Way of doing it? |
| 13:14.07 | hwt | tzanger: since you specify "for simplicity". |
| 13:14.19 | tzanger | hwt: basically the sip user (friend is both user&peer) needs to know where to dump their call request in the dialplan. without that it assumes [default] and you never want that |
| 13:14.29 | astra^^ | i am trying to fowared the calls comin in with the extension 123 to the same server with different prefix |
| 13:14.35 | tzanger | hwt: I would do something like context=officephones or something |
| 13:14.44 | astra^^ | anyone please ..http://pastebin.com/628674 |
| 13:14.49 | tzanger | actually hwt |
| 13:15.08 | tzanger | context=officephones |
| 13:15.20 | tzanger | then have a [office_extensions] context that does nothing but |
| 13:16.07 | tzanger | exten => _XXXX,1,Macro(OfficeExtension,${EXTEN}) |
| 13:16.15 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:16.19 | pigpen2 | I have about 50 polycom 601's...is there a way to get each phone to read one xml directory file? |
| 13:16.27 | hwt | tzanger: that seems compact, yes. nice, thanks. |
| 13:16.47 | tzanger | then make a [macro-OfficeExtension] macro which sets up to do whatever you want... dial, drop to voicemail after so many seconds, allow *72-style call forwarding, whatever |
| 13:17.02 | tzanger | but your [officephones] context would |
| 13:17.06 | tzanger | include = office_extensions |
| 13:17.12 | tzanger | include = international |
| 13:17.37 | tzanger | which means that anyone dumped into the [officephones] context can call office extensions and also dial out internationally |
| 13:17.44 | astra^^ | http://pastebin.com/628674 => sip and ext cof in heree |
| 13:17.57 | *** part/#asterisk X-Gen (n=x-gen@dsl-145-231-103.telkomadsl.co.za) |
| 13:18.01 | hwt | tzanger: where do i specify the macro? |
| 13:18.08 | tzanger | nettie: it really looks like your credentials are wrong for that provider |
| 13:18.39 | tzanger | hwt: in extensions.conf. check out the asterisk handbook draft on digium's site, and perhaps check out blitzrage's (and others') book: Asterisk: The Future of Telephony |
| 13:18.44 | tzanger | all of this is explained very well |
| 13:18.57 | hwt | tzanger: thanks a lot. |
| 13:19.14 | tzanger | np. you'll hvae a killer dialplan in under a week |
| 13:19.26 | caio1982 | coppice: hey, i just sent an email to you. i thought you wouldnt be around, since we're hmmm 12 hours away from each other |
| 13:19.27 | tzanger | ManxPower has some wicked standard extension macros on his site |
| 13:20.00 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 13:20.22 | fourcheeze | tzanger: which site is that? |
| 13:20.35 | tzanger | astra^^: you're trying to forward any call starting with 123 to some OTHER server? |
| 13:20.38 | nettie | tzanger that's unlikely.. |
| 13:21.17 | tzanger | nettie: that's kind of what it looks like. the far side is saying "whozat?" and you're saying "I'm a friend of Fat Tony" and the other side's saying "Well we gots a problem.. Fat Tony's got no friends" |
| 13:21.35 | tzanger | fourcheeze: fnords.org I think. google for manxpower and fnords |
| 13:21.40 | nettie | tzanger I can register successfully with the same user and pass |
| 13:21.41 | tzanger | or just ask him, he's here |
| 13:21.45 | astra^^ | tzanger: no to the same server wher i get te call from |
| 13:21.49 | tzanger | nettie: doesn't necessarily mean you're allowed ot pass calls |
| 13:21.51 | nettie | tzanger I'll try with "Fat Albert" :) |
| 13:21.56 | tzanger | nettie: :-) |
| 13:22.12 | tzanger | astra^^: so this is on server A and you want to get a call and send it to server B |
| 13:22.12 | fourcheeze | nettie: there's username and authentication username, make sure they are both what they shoudl be |
| 13:22.56 | tzanger | astra^^: it looks like that should work. you may want to split it up and see if *'s having a problem with mixing the two together. |
| 13:23.05 | nettie | fourcheeze fourcheeze I have username= and fromuser= defined |
| 13:23.18 | tzanger | astra^^: i.e. _123.,1,Set(NUM=${EXTEN:3}) |
| 13:23.26 | coppice | caio1982: when you get congestion all the time, the usual reason is the other end os not configured properly, and cannot handle any calls |
| 13:23.28 | astra^^ | server a with prefix 123 ==>asterisk sents call with prefix 211563===>server A |
| 13:23.33 | tzanger | astra^^: i.e. _123.,2,Set(NEWNUM=211563${NUM)) |
| 13:23.46 | tzanger | astra^^: i.e. _123.,2,Dial(SIP/${NEWNUM}@xxx.xxx.xxx.xxx,60) |
| 13:23.52 | tzanger | er make that last one ,3, |
| 13:23.59 | caio1982 | coppice: even when it's for R2 signalling only? |
| 13:24.03 | tzanger | astra^^: uh |
| 13:24.07 | tzanger | astra^^: why the hell are you using dial then? |
| 13:24.18 | astra^^ | but i get an error while i place a call: |
| 13:24.19 | astra^^ | Mar 29 07:21:19 NOTICE[1333]: chan_sip.c:9524 handle_response_invite: Failed to authenticate on INVITE to '"muhajir" <sip:1000@64.246.52.52>;tag=as152bfcbd' |
| 13:24.37 | tzanger | astra^^: why not just exten => _123,1,Goto(SOME_CONTEXT,211563${EXTEN:3},1) |
| 13:24.47 | tzanger | astra^^: why not just Goto? Why try to call back in to the same box? |
| 13:24.51 | coppice | caio1982: it makes no difference whether it is R2, ISDN or something else. if it cannot handle calls you will get some form of congestion signal |
| 13:25.14 | astra^^ | exten => _123.,1,Dial(SIP/211563${EXTEN:3}@207.173.206.116,60) |
| 13:25.26 | caio1982 | coppice: i suspect it is a radcom's fault but now i have to prove that, but anyway i wanted to be sure unicall/r2 is okay... later i'll test it a bit more, thanks :) |
| 13:25.27 | ManxPower | don't dial by ip address!!!!!!! |
| 13:25.28 | tzanger | astra^^: is this box 207.173.206.116? |
| 13:25.42 | astra^^ | tismy context wher take example :207 ip is the sip server wher i get call from |
| 13:26.13 | astra^^ | i am forwarding calls comin to this ipwith some other prefix |
| 13:26.18 | Z0m81e | Does anyone know why our * server spams this: Mar 29 14:05:15 NOTICE[29351]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 0! |
| 13:26.18 | stoffell | ManxPower, can you share your site's url ? or is it fnords.org ? |
| 13:26.35 | tzanger | astra^^: sure, but why Dial() yourself? You already have the call, just Goto() the correct context/extension |
| 13:26.40 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F437C.dip0.t-ipconnect.de) |
| 13:26.40 | Dabian | fnordz? |
| 13:27.02 | ManxPower | stoffell, I don't have a site anymre |
| 13:27.07 | coppice | ManxPower: I wonder what interesting number dialing 127001 or 19216811 might reach :-) |
| 13:27.34 | stoffell | oooh, okay, thanks ManxPower ;) |
| 13:28.11 | astra^^ | tzanger:can u please make the changes to be made in the pastebin :http://pastebin.com/628674 |
| 13:28.22 | caio1982 | coppice: does your r2 code comes with any ISDN error message or something? that isdn message when testing r2 confused me too |
| 13:28.32 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:29.00 | Splat | what codecs do people suggest using? |
| 13:29.13 | RoyK | people suggest lots of things |
| 13:29.13 | tzanger | astra^^: no, I just told you what to do: _123.,1,Goto(211563${EXTEN:3},1) |
| 13:29.14 | [ProB]CrazyMan | how gets the src value set ? I get there from my dect phones an horribly sign Þ |
| 13:29.26 | tzanger | astra^^: if you are wanting it in a different context, then Goto(context,211563...) |
| 13:29.29 | RoyK | Splat: i suggest using the one that works best for you |
| 13:29.51 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:30.01 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:30.03 | coppice | ISDN errors are pretty much a superset of all other telephony errors. I use the ISDN codes for all signalling protocols. This error (congestion) has been reported as a tone signal by the far end |
| 13:30.41 | ManxPower | just remember if you forget the priority in the goto Bad Things will happen |
| 13:31.08 | tzanger | ManxPower: heh |
| 13:31.10 | Dabian | (muahahaha) |
| 13:32.21 | caio1982 | coppice: okay, thanks again |
| 13:32.40 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:33.18 | x86 | [TK]D-Fender: you around? |
| 13:33.47 | [TK]D-Fender | no |
| 13:33.51 | x86 | :P |
| 13:34.00 | x86 | heh |
| 13:34.03 | Dabian | he |
| 13:34.04 | x86 | whats up man? |
| 13:34.05 | Dabian | hehe |
| 13:34.11 | Splat | RoyK: any there any specific codecs you'd suggest trying? I'm currently using either ulaw or alaw which is great and all.. but if I want to allow 2 maybe 3 at the most (and vary rarely would it hit 3) outgoing calls at the same time then I suspect I won't have the bandwidth to run ulaw or alaw for all 3 calls.. |
| 13:34.11 | stoffell | any opinions on creating "larger' numbering plans? (multiple sites, 40-100, with max. 300 nrs per site) |
| 13:34.18 | nettie | fourcheeze the username and authentication username you referring in your message are: username and fromuser ? |
| 13:34.25 | [TK]D-Fender | blarg.... I want this week over, I want my vacation... I want my MTV..... |
| 13:34.52 | *** join/#asterisk linstar (n=achu@220.225.191.18) |
| 13:35.11 | x86 | [TK]D-Fender: isnt there a song in there somewhere? |
| 13:35.13 | x86 | hehe |
| 13:35.14 | [TK]D-Fender | stoffell : 4 digit extensions, 1st digit implying which site. |
| 13:35.16 | linstar | when I make a call to another extension it won't work |
| 13:35.28 | RoyK | Splat: g.729 then..... |
| 13:35.29 | x86 | some 80's song about MTV |
| 13:35.29 | [TK]D-Fender | x86 : kudos to you for that. |
| 13:35.30 | linstar | showing Destroying Call in CLI |
| 13:35.38 | linstar | any help to solve this? |
| 13:35.49 | x86 | [TK]D-Fender: bruce springsteen? *thinks* |
| 13:35.55 | stoffell | [TK]D-Fender, ack, but when i reach more then 10 sites, i'm in trouble, yes? (some sites are 'big', some are 'very' small) |
| 13:36.44 | fourcheeze | nettie: no, in asterisk there's the user in [] and then there's username |
| 13:36.46 | [TK]D-Fender | stoffell : you could always just dial the other site as a "gateway" with a single access # and then get 2nd dialtone.... |
| 13:36.56 | fourcheeze | nettie: that's if I'm remembering ordinary sip.conf |
| 13:36.58 | [TK]D-Fender | x86 : Dire Straits! |
| 13:37.18 | x86 | [TK]D-Fender: hey at least i picked up on it at all ;) |
| 13:37.18 | stoffell | [TK]D-Fender, hm, okay, thanks! will look into that possibility. |
| 13:37.23 | x86 | [TK]D-Fender: no one else did ;) |
| 13:37.30 | fourcheeze | x86: how can you confuse springsteen and dire straits? |
| 13:37.37 | linstar | fourcheeze : I can't make calls between sip extensions |
| 13:37.40 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 13:37.49 | x86 | fourcheeze: *shrugs* |
| 13:37.50 | *** join/#asterisk Assid (n=assid@59.183.43.24) |
| 13:37.53 | Assid | heya |
| 13:38.02 | Assid | when i try and load ztdummy.. i get this error: WARNING: Error inserting rtc (/lib/modules/2.6.16.1/kernel/drivers/char/rtc.ko): No such device |
| 13:38.04 | linstar | fourcheeze : gettting error in CLI as Destroying Calls |
| 13:38.12 | fourcheeze | x86: Money for nothing was a great song |
| 13:38.16 | linstar | fourcheeze : any help to solve this? |
| 13:38.18 | x86 | [TK]D-Fender: can you check something out for me with my dialplan? |
| 13:38.24 | fourcheeze | why me? |
| 13:38.29 | [TK]D-Fender | x86 : pastebin away |
| 13:38.29 | x86 | fourcheeze: money for nothing and your chicks for free :P |
| 13:38.41 | x86 | [TK]D-Fender: https://office.shellshark.net:7960/extensions.conf |
| 13:38.50 | fourcheeze | tunnel of love is one of my faves |
| 13:38.51 | Assid | anyone know whats up |
| 13:38.55 | fourcheeze | might have to download that |
| 13:39.26 | smeevil | going crazy from the musiconhold silence |
| 13:39.49 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:39.51 | [TK]D-Fender | x86 : Ok, what about it? |
| 13:40.08 | x86 | [TK]D-Fender: when i dial *1 and a 10 digit phone number, it does the ENUM dial, falls back to the regular SIP/PSTN gateway dial, records the call like it should, but when i hang up, it dials again... |
| 13:40.28 | fourcheeze | linstar: got someting in your extensions.conf to let you do that? |
| 13:40.45 | nettie | fourcheeze the name inside the [] could be whatever you want |
| 13:41.04 | fourcheeze | nettie: IME not always |
| 13:41.21 | txtNation | Yargh, anyone know how to compile Asterisk on a Solaris 10 64-Bit SPARC box? |
| 13:41.41 | linstar | fourcheeze : No changes have made in extensions.conf |
| 13:41.41 | x86 | [TK]D-Fender: first time CLI says Spawn extension (macro-app-rad ...), second time it says Spawn extension (rad ...) |
| 13:42.23 | nettie | fourcheeze well I changed it anyway to test.. didnt work |
| 13:42.32 | [TK]D-Fender | x86 : Your [macro-enum-dial] has no "1" priority. |
| 13:42.50 | x86 | [TK]D-Fender: for instance, if local device 103 calls outside PSTN phone number 2125552424, it will make the call, then 2125552424 hangs up before 103 does, asterisk makes another outbound call from 103 |
| 13:43.02 | x86 | [TK]D-Fender: hmm, i stole that macro from the wiki somewhere... |
| 13:43.11 | x86 | [TK]D-Fender: does it need at least a 1 priority somewhere? |
| 13:43.41 | [TK]D-Fender | x86 : Thats the rule... |
| 13:43.47 | x86 | ah :) |
| 13:44.13 | linstar | fourcheeze : it was working fine before 10 min and now go down |
| 13:44.13 | x86 | so what's n do, just go in order of apperance unless there is a label? |
| 13:45.28 | [TK]D-Fender | x86 : Almost. It goes to the next #'s sequence. Labels are used to track the jump points for your goto's to the auto-renumbered "n"'s |
| 13:45.44 | linstar | fourcheeze : I had stopped asterisk and restarted it again but nothing works |
| 13:46.05 | hwt | where do i set the asterisk-sounds and voicemail language to norwegian? |
| 13:46.12 | hwt | i want it to be in norwegian everywhere. |
| 13:46.21 | x86 | [TK]D-Fender: cool |
| 13:46.39 | x86 | [TK]D-Fender: ok another thing.... |
| 13:46.44 | hwt | alsa.conf? |
| 13:46.58 | x86 | [TK]D-Fender: that macro-enum-dial has an endless loop i cant pinpoint |
| 13:47.02 | iDunno | hmmm. |
| 13:47.10 | x86 | [TK]D-Fender: see anything that jumps out at you? |
| 13:47.19 | backblue | txtNation: it does not compile? |
| 13:47.30 | backblue | what its the error? |
| 13:48.26 | hwt | it says -- Playing 'vm-login' (language 'no') |
| 13:48.36 | hwt | and i have /var/lib/asterisk/sounds/no/ |
| 13:49.22 | linstar | Can anybody help me to solve the CLI error Destroying Connection? |
| 13:50.03 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 13:50.09 | Z0m81e | Does anyone know why our * server spams this: Mar 29 14:05:15 NOTICE[29351]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 0! |
| 13:50.23 | x86 | [TK]D-Fender: if the number is listed in an E164 database (like e164.org), i can dial it about 75% of the time... numbers not listed (that should fall back to the regular SIP<-->PSTN trunk) end up in an endless loop in macro-enum-dial |
| 13:50.24 | [TK]D-Fender | x86 : Not offhand... it makes me dizzy |
| 13:50.29 | hwt | tzanger: ? |
| 13:50.59 | x86 | Z0m81e: sounds like you need a timing source :P |
| 13:51.17 | x86 | Z0m81e: you have zaptel hardware or just doing SIP/IAX2/H323 trunks? |
| 13:51.26 | Z0m81e | x86: humm, possible that machine only handles IAX traffic |
| 13:51.34 | backblue | Z0m81e: ztdummy |
| 13:51.34 | x86 | you need ztdummy |
| 13:51.45 | Z0m81e | x86, backblue, cheers :) |
| 13:51.45 | *** join/#asterisk _andre (n=andre@fosforo.k8.com.br) |
| 13:51.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:51.49 | linstar | <PROTECTED> |
| 13:51.52 | backblue | np |
| 13:52.02 | _andre | good morning |
| 13:52.25 | _andre | is this what you guys use to send queue logs to a DB? http://lists.digium.com/pipermail/asterisk-users/2005-July/109892.html |
| 13:52.40 | _andre | or is there a better alternative? |
| 13:53.47 | *** join/#asterisk HamYaI (n=HamYai@125.24.12.236) |
| 13:53.47 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 13:54.18 | HamYaI | anyone has this problem while puuting the call on hold? => chan_sip.c:3444 process_sdp: Unable to lookup host in c= line |
| 13:54.41 | HamYaI | I had this problem since 1.2.5 |
| 13:54.56 | SplasPood | Hrm did something change with the nufone sip settings recently? I continually get a server error on outbound calling |
| 13:56.08 | x86 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=9503239627 |
| 13:56.15 | x86 | NSFW |
| 13:57.49 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:58.26 | *** join/#asterisk Dovid (n=Dovid@89-138-76-126.bb.netvision.net.il) |
| 14:02.04 | Dovid | . |
| 14:02.11 | Dovid | no one here ? |
| 14:02.48 | Katty | we're all napping. |
| 14:04.02 | Dovid | lol |
| 14:04.05 | Dovid | cmon wake up |
| 14:04.35 | Katty | newp. |
| 14:04.37 | Katty | shan't. |
| 14:05.42 | Dovid | lol |
| 14:05.55 | x86 | hehehe |
| 14:06.58 | coppice | everyone is sleeping |
| 14:07.11 | coppice | there is a forest of thorns around the castle |
| 14:07.24 | coppice | and we are waiting for prince charming to arrive |
| 14:07.33 | [TK]D-Fender | Katty: mew. |
| 14:08.16 | mut | O_O |
| 14:08.44 | x86 | [TK]D-Fender: do you have E164 setup? :P |
| 14:08.59 | linstar | any help to solve the CLI error "Destroying call" ? |
| 14:09.29 | [TK]D-Fender | x86 : nope. |
| 14:11.14 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 14:11.20 | *** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 14:11.24 | linstar | <PROTECTED> |
| 14:11.41 | stoffell | is it legal to use the asterisk logo in a visio/powerpoint slide of your own? |
| 14:12.31 | *** join/#asterisk b66mer (i=fwuser@blackhole.c5i.com) |
| 14:13.01 | coppice | I wonder how it feels to be pampled? |
| 14:13.20 | Dovid | stofell: I think to the letter of the law it is, post ur question on the users lst |
| 14:13.57 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:14.01 | x86 | [TK]D-Fender: you totally should... you'd save a bundle on outbound calls and people who have E164 would save a bundle calling you :) |
| 14:14.31 | x86 | [TK]D-Fender: bypassing the PSTN for PSTN-bound calls is sweet :P |
| 14:14.46 | linstar | any help to solve the CLI error "Destroying call" ? |
| 14:15.21 | fourcheeze | can the polycom lights be reconfigured to represent presence ? |
| 14:15.44 | Assid | word up my crazy cheese eating kats |
| 14:15.58 | Katty | katty is vegan. |
| 14:16.00 | MikeJ[Laptop] | fourcheeze, yes |
| 14:16.03 | Katty | thank youverymuch. |
| 14:16.09 | adelas | hey guys which word seems better for this sentence, The play is very insightful upon human attributions/attributes |
| 14:16.11 | fourcheeze | MikeJ[Laptop]: how? |
| 14:16.24 | Katty | coppice: i dunno, but you could ask my ex. |
| 14:16.24 | MikeJ[Laptop] | don't recall how to set it up onthe polycomm |
| 14:16.41 | MikeJ[Laptop] | but in dialplan, that is what those hint things are for |
| 14:16.47 | fourcheeze | Katty: weren't you just vegetarian a few weeks back? |
| 14:17.00 | fourcheeze | MikeJ[Laptop]: I can do the presence thing on the on-screen display |
| 14:17.01 | Katty | fourcheeze: i was considering it. |
| 14:17.01 | MikeJ[Laptop] | I am pretty sure there is a wiki article on it |
| 14:17.06 | [TK]D-Fender | x86 : I don't have an analog line at home, I'm running a DID from my work whose PRI I use for all calls. as my home is on their setup, people can also call my work's 1-800 # and dial my be ext #. Phone costs ME nothing. |
| 14:17.10 | coppice | being vegetarian has nothing to do with her coming from Vega |
| 14:17.13 | Katty | fourcheeze: i've not made up my mind yet...but that doesn't mean i've switched. |
| 14:17.17 | Katty | coppice: :>> |
| 14:17.36 | MikeJ[Laptop] | coppice, correct! |
| 14:17.39 | fourcheeze | I find it hard to justify vegetarianism |
| 14:17.49 | fourcheeze | except on ecological grounds |
| 14:17.55 | MikeJ[Laptop] | coppice, I think Katty just bit shifted you??? |
| 14:18.02 | [TK]D-Fender | fourcheeze : You need to have a free line key. Add a contact and enable "buddy-watch" on it. add the hint into your dialplan and you're set. |
| 14:18.03 | fourcheeze | and I was never brave enough to be a vegan |
| 14:18.26 | Katty | fourcheeze: have you seen ground beef? |
| 14:18.28 | Katty | fourcheeze: ugah. |
| 14:18.29 | MikeJ[Laptop] | fourcheeze, you are either born in Vega or not :P |
| 14:18.33 | [TK]D-Fender | Warning to vegetarians : You are what you eat... |
| 14:18.36 | Katty | stuff makes me sick. |
| 14:18.42 | Katty | just the /smell/ is revolting |
| 14:18.43 | coppice | vegetarians have longer than average lives. vegans have significantly shorter than average lives |
| 14:18.58 | Katty | coppice: but less chances for cancer. |
| 14:19.10 | iDunno | hmmm. |
| 14:19.10 | MikeJ[Laptop] | it's because of the painfully hot summers in Vega? |
| 14:19.25 | fourcheeze | Katty: Katty what's so bad about ground beef? |
| 14:19.37 | coppice | Katty: i think that is unproven, but there are plenty of long term figures on life expectancy |
| 14:19.39 | Katty | fourcheeze: all the grease, for one thing. |
| 14:19.45 | MikeJ[Laptop] | fourcheeze, I am pretty sure it has somthing to do with the beef part |
| 14:19.51 | fourcheeze | [TK]D-Fender: where does buddywatch get enabled? |
| 14:19.55 | *** part/#asterisk linstar (n=achu@220.225.191.18) |
| 14:20.04 | Katty | and maybe that god awful smell. |
| 14:20.06 | fourcheeze | Katty: I tend to feel if you don't like beef you're not going to like it ground |
| 14:20.17 | Katty | obviously. |
| 14:20.18 | MikeJ[Laptop] | hehe |
| 14:20.23 | iDunno | what awful smell?! |
| 14:20.26 | fourcheeze | [TK]D-Fender: ok I found buddy watch |
| 14:20.28 | x86 | [TK]D-Fender: you could save the people who call you and save your work money though ;) |
| 14:20.33 | fourcheeze | I'm watching but it's not appearing on a button |
| 14:20.37 | Katty | iDunno: that awful smell of really greasy beef cooking. |
| 14:20.39 | MikeJ[Laptop] | iDunno, probably the rotting flesh one |
| 14:20.55 | [TK]D-Fender | fourcheeze :Are you provisioning your phone? |
| 14:20.58 | coppice | ground beef often tastes like its made with pure ground. i'm not sure there's a lot of beef in it |
| 14:20.59 | fourcheeze | Katty: what about non=greasy beef |
| 14:21.04 | iDunno | ahh - yes, that's not good. |
| 14:21.06 | Katty | fourcheeze: that's not possible. |
| 14:21.06 | fourcheeze | [TK]D-Fender: I'm not sure |
| 14:21.16 | [TK]D-Fender | fourcheeze : You need to set a speed-dial index for it to get assigned to a free line-key |
| 14:21.18 | fourcheeze | ok, so you don't like beef |
| 14:21.20 | Katty | fourcheeze: beef has grease, end of story. |
| 14:21.24 | fourcheeze | Katty: there's other kinds of meat |
| 14:21.26 | Katty | fourcheeze: obviously. |
| 14:21.30 | fourcheeze | what about fish? |
| 14:21.30 | Katty | fourcheeze: i like veggie burgers. |
| 14:21.38 | Katty | fourcheeze: especially the ones from Denny's. |
| 14:21.39 | fourcheeze | see, the clue's in the name there |
| 14:21.45 | iDunno | mmhmm ;) |
| 14:21.53 | [TK]D-Fender | Katty : Veggie burgers are typically LOADED with fat... just not ANIMAL. |
| 14:21.55 | Katty | fourcheeze: i think you're missing the whole i don't eat animals part :P |
| 14:22.09 | MikeJ[Laptop] | fourcheeze, where you from? |
| 14:22.12 | fourcheeze | just trying to work out what it is about animals |
| 14:22.12 | iDunno | Katty: well, eating whole animals could take *days* anyways ;) |
| 14:22.15 | Katty | [TK]D-Fender: but they're not greasy and not smelling of ground beef. |
| 14:22.19 | iDunno | (depending on the animal :) |
| 14:22.20 | fourcheeze | MikeJ[Laptop]: all the way from the U of K |
| 14:22.26 | MikeJ[Laptop] | heh |
| 14:22.30 | [TK]D-Fender | "There's room enough for all God's creatures..... right next to the mashed potatoes :D" |
| 14:22.35 | MikeJ[Laptop] | let me think... |
| 14:22.53 | MikeJ[Laptop] | nope.. all of the comparable resturaunts I can think of are american |
| 14:22.58 | coppice | beef, pork, chicken, duck, rabbit, pigeon, frog, horse, donkey, crocodile, kangaroo, emu. such variety |
| 14:23.05 | MikeJ[Laptop] | denny's, big boy, perkins.... |
| 14:23.08 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:23.21 | MikeJ[Laptop] | anyone know a UK parallell? |
| 14:23.27 | Katty | mushrooms! |
| 14:23.34 | Katty | portabella mushroom steaks are /pantpantpant/ |
| 14:23.36 | coppice | little chef |
| 14:23.47 | MikeJ[Laptop] | they have little chef? |
| 14:23.47 | Katty | and oddly good with bbq sauce. |
| 14:23.52 | fourcheeze | little chef |
| 14:23.57 | iDunno | mushrooms are fantastic :) |
| 14:24.02 | Katty | aren't they? |
| 14:24.02 | fourcheeze | little chef is dead here |
| 14:24.08 | fourcheeze | well almost |
| 14:24.11 | fourcheeze | they went bust |
| 14:24.11 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:24.13 | MikeJ[Laptop] | dennys is dead here... |
| 14:24.17 | fourcheeze | because no-one wanted to eat their crap |
| 14:24.22 | MikeJ[Laptop] | but that is just cuz the food is grose... |
| 14:24.24 | iDunno | they go very well with bacon and eggs and sausages and black pudding :) |
| 14:24.31 | MikeJ[Laptop] | but for some reason.. they keep open |
| 14:24.31 | fourcheeze | [TK]D-Fender: ok, got the button working |
| 14:24.40 | iDunno | fourcheeze: they were over-priced, it was not suprising ;) |
| 14:24.45 | fourcheeze | [TK]D-Fender: can the button show presence? |
| 14:24.49 | coppice | really. a couple of years ago when I toured around the UK for a week I was surprised to find Little Chef had become fairly civilised |
| 14:24.50 | fourcheeze | I've got my buddy on it |
| 14:24.55 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
| 14:25.19 | MikeJ[Laptop] | fourcheeze, WHOA! |
| 14:25.43 | coppice | MikeJ: the food is goose? i love roast goose |
| 14:25.51 | Dovid | damn |
| 14:25.59 | fourcheeze | that tends to rule out little chef |
| 14:25.59 | [TK]D-Fender | fourcheeze : Yes. we've alre4ady answered that one... |
| 14:26.01 | Dovid | tis all gettin me hungry |
| 14:26.08 | MikeJ[Laptop] | not goose silly |
| 14:26.13 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 14:26.15 | MikeJ[Laptop] | disgusting |
| 14:26.26 | coppice | it would rule out Denny's too. very similar establishments |
| 14:26.31 | fourcheeze | [TK]D-Fender: checklist: buddy - yep; watched - yep; speed dial index - yep ; |
| 14:26.48 | [TK]D-Fender | fourcheeze : Dialplan "hint"? |
| 14:26.56 | fourcheeze | got that |
| 14:27.07 | fourcheeze | it works in the buddy list |
| 14:27.11 | fourcheeze | just not on the button |
| 14:27.17 | [TK]D-Fender | fourcheeze : Presence feature enabled in provisioning file? |
| 14:27.20 | coppice | and then there's all the wonder yummy creatures of the sea |
| 14:27.27 | coppice | bombay duck |
| 14:27.36 | iDunno | hmm - I want some decent applejuice |
| 14:27.38 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 14:27.44 | coppice | or is that mumbai duck these days |
| 14:27.47 | [TK]D-Fender | coppice : Ducks are typically fresh-water :) |
| 14:28.10 | coppice | bombay duck isn't :-\ |
| 14:28.11 | fourcheeze | absolutley great |
| 14:28.21 | fourcheeze | and even better than that |
| 14:28.25 | fourcheeze | the goose fat we got off it |
| 14:28.33 | fourcheeze | kept us going for about a month |
| 14:28.49 | MikeJ[Laptop] | I should go shave |
| 14:28.49 | fourcheeze | got about 2 litres of it |
| 14:28.52 | MikeJ[Laptop] | it's spring |
| 14:28.53 | coppice | had goose twice in the last week. and 5 beijing ducks the week before. i need to cut down on fat intake, but its hard when i keep travelling |
| 14:29.29 | MikeJ[Laptop] | coppice, are you still as skinny as your pictures? if so, I don't think you are in much risk of weight gain |
| 14:29.47 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:29.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:29.51 | fourcheeze | nothing wrong with fat |
| 14:29.56 | coppice | my pictures aren't skinny. i weight nearly 100 |
| 14:29.59 | fourcheeze | our cave-dwelling ancestors would have taken all they can get |
| 14:30.05 | MikeJ[Laptop] | 100 what? |
| 14:30.06 | fourcheeze | 100 what - grams? |
| 14:30.11 | *** part/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:30.17 | coppice | well it ain't pounds |
| 14:30.21 | fourcheeze | stone? |
| 14:30.24 | fourcheeze | tonnes? |
| 14:30.24 | MikeJ[Laptop] | the one with your wife and kids on your site is the only one I have ever seen |
| 14:30.37 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:30.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:30.45 | fourcheeze | ounces |
| 14:30.46 | coppice | i weighed nearly 100 of some obscure unit then, too |
| 14:30.56 | MikeJ[Laptop] | hmmm |
| 14:31.01 | CleanerX | hey guys - visit europe ! ;-) |
| 14:31.05 | MikeJ[Laptop] | I think units are important |
| 14:31.21 | coppice | tens and hundred are important too |
| 14:31.22 | file | oxygen is important too |
| 14:31.26 | MikeJ[Laptop] | my kid comes up to me... and says she weighs 50 |
| 14:31.29 | fourcheeze | units are only important because the USA won't use sensible ones |
| 14:31.38 | MikeJ[Laptop] | well.. |
| 14:31.44 | MikeJ[Laptop] | units are always important.. |
| 14:31.56 | MikeJ[Laptop] | because it could be grams or kg... |
| 14:31.59 | MikeJ[Laptop] | how do you know |
| 14:32.04 | file | don't forget stones |
| 14:32.13 | russellb | i like slugs and furlongs |
| 14:32.14 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:32.16 | *** part/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 14:32.22 | MikeJ[Laptop] | russellb, you like slugs? |
| 14:32.27 | coppice | bloody MS have broken the centimetre option in their latest FAX reading. I'm giving up and changing spandsp to use inches in the TIFF files. |
| 14:32.29 | russellb | the unit, yes |
| 14:32.44 | MikeJ[Laptop] | heh |
| 14:32.49 | file | russellb, !!! |
| 14:32.56 | russellb | file: !!!!!! |
| 14:33.01 | MikeJ[Laptop] | russellb, you like the unit? |
| 14:33.06 | russellb | damnit! |
| 14:33.08 | bkw__ | oh guys don't go there |
| 14:33.09 | bkw__ | damn |
| 14:33.13 | bkw__ | :P |
| 14:33.15 | coppice | the unit? as in "a slug of whiskey" |
| 14:33.19 | file | bkw__, it's official - you corrupted MikeJ |
| 14:33.19 | bkw__ | not even I was going to go there |
| 14:33.22 | MikeJ[Laptop] | I was talking about the tv show.. |
| 14:33.28 | MikeJ[Laptop] | where was your mind... |
| 14:33.37 | bkw__ | take a guess |
| 14:33.38 | russellb | liar |
| 14:33.42 | file | in the gutter obviously |
| 14:33.42 | MikeJ[Laptop] | hehe |
| 14:33.47 | MikeJ[Laptop] | ummmm |
| 14:34.12 | *** join/#asterisk shiznatix (n=shiznati@213-35-236-110-dsl.end.estpak.ee) |
| 14:34.17 | shiznatix | hello everyone! |
| 14:37.20 | shiznatix | I just installed Kiax and I am able to make calls to SIP devices with it no problem but I am unable to dial from a SIP to Kiax |
| 14:38.00 | fourcheeze | [TK]D-Fender: ok I seem to have <feature feature.1.name="presence" feature.1.enabled="1" ...> in my provisioning |
| 14:38.03 | MikeJ[Laptop] | what is kiax? |
| 14:38.15 | shiznatix | its a IAX2 softphone for linux |
| 14:38.35 | MikeJ[Laptop] | oh |
| 14:38.41 | MikeJ[Laptop] | well.. |
| 14:38.57 | MikeJ[Laptop] | so you are talking through somthing that speexs iax and sip I assume? |
| 14:39.06 | MikeJ[Laptop] | perhaps asterisk given the channel we are in |
| 14:39.15 | shiznatix | perhaps :) |
| 14:39.21 | shiznatix | ya It's all through asterisk |
| 14:39.25 | shiznatix | asterisk gives me this error: |
| 14:39.39 | MikeJ[Laptop] | ahhh |
| 14:39.44 | shiznatix | Unable to create channel of type 'IAX' (cause 66 - Channel not implemented) |
| 14:39.45 | MikeJ[Laptop] | see.. now, real information |
| 14:39.51 | MikeJ[Laptop] | hmmmm |
| 14:39.59 | russellb | IAX2! |
| 14:40.01 | shiznatix | then it says: == Everyone is busy/congested at this time (1:0/0/1) |
| 14:40.04 | MikeJ[Laptop] | heh |
| 14:40.10 | russellb | s/IAX/IAX2/ ! |
| 14:40.14 | MikeJ[Laptop] | russellb, you just spoil all my fun |
| 14:40.16 | MikeJ[Laptop] | :( |
| 14:40.17 | sambal | is there any CLI sip or iax2 client to test dial plans? |
| 14:40.19 | russellb | :-p |
| 14:40.19 | shiznatix | oh shiznazzle |
| 14:40.26 | shiznatix | haha ok lemme test that |
| 14:40.29 | MikeJ[Laptop] | sambal, yes |
| 14:40.37 | sambal | how is it called? :) |
| 14:40.47 | MikeJ[Laptop] | what platform? |
| 14:40.52 | sambal | linux |
| 14:41.00 | *** join/#asterisk |Vulture| (n=Vulture@82.115.205.68.cfl.res.rr.com) |
| 14:41.06 | MikeJ[Laptop] | well.. you can use the sound card stuff in asterisk |
| 14:41.09 | sambal | command line |
| 14:41.25 | MikeJ[Laptop] | or use testcall from iax client |
| 14:41.44 | shiznatix | Hey thanks guys, that worked just fine |
| 14:45.33 | HamYaI | anyone using Asterisk Perl Library here? |
| 14:46.10 | MikeJ[Laptop] | :( |
| 14:46.17 | MikeJ[Laptop] | no more RoyK |
| 14:46.29 | MikeJ[Laptop] | and he is my very favorite troll |
| 14:46.49 | HamYaI | MikeJ[Laptop]: r u using AGI at all? |
| 14:48.38 | MikeJ[Laptop] | personally, no |
| 14:48.40 | MikeJ[Laptop] | you? |
| 14:49.05 | HamYaI | yeah |
| 14:49.17 | HamYaI | i'm using it |
| 14:49.24 | MikeJ[Laptop] | congrats!!! |
| 14:49.34 | MikeJ[Laptop] | how about sip? |
| 14:49.51 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 14:51.00 | HamYaI | MikeJ[Laptop]: what about it? |
| 14:51.09 | file | SIP it! |
| 14:51.28 | MikeJ[Laptop] | you use it? |
| 14:51.29 | HamYaI | MikeJ[Laptop]: I'm having problems passing a call thru zap channels |
| 14:51.36 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 14:51.41 | HamYaI | MikeJ[Laptop]: sip works fine |
| 14:51.46 | caio1982 | lol |
| 14:52.11 | MikeJ[Laptop] | congrats!!! |
| 14:52.43 | mut | mmmmmmmmmm |
| 14:52.48 | mut | mcdonalds steak bagel! |
| 14:52.50 | MikeJ[Laptop] | sorry to hear about your zap probs |
| 14:52.55 | file | MikeJ[Laptop], you're sure congratulating people a lot |
| 14:53.01 | MikeJ[Laptop] | well... |
| 14:53.11 | MikeJ[Laptop] | I am not sure what the guy is asking me really |
| 14:53.13 | file | mut: here in Atlantic Canada McDonalds will actually have McLobster some months of the year... it's disturbing |
| 14:53.19 | MikeJ[Laptop] | he just seems to be telling me a lot of stuff |
| 14:53.23 | MikeJ[Laptop] | vaugely.. |
| 14:53.30 | mut | lobster is teh suck |
| 14:53.31 | MikeJ[Laptop] | so not sure how to react really |
| 14:53.43 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 14:53.47 | file | http://members.shaw.ca/jdkeller/halifax/pages/_DSC00083.html |
| 14:53.49 | file | looks like that |
| 14:53.53 | mut | be like 5% lobster and 95% tuna parts |
| 14:54.04 | coppice | file: they have buns replaced by rice here at the moment |
| 14:54.28 | file | there's some sense of security for when you travel... that almost every McDonalds is the same |
| 14:54.41 | mut | heh |
| 14:54.56 | file | as long as the people don't screw up and make your fries 90% salt |
| 14:55.07 | coppice | except in the philipinnes where they permanently have chicken and steamed rice :-) |
| 14:55.10 | file | er wait, Laguardia! |
| 14:55.13 | opc0de | hey can anyone tell me how to define a channel as outgoing only in asterisk? I have in zapata.conf "group=1; channel=1-4; callgroup=1; pickupgroup=1", but the problem is, I don't want channel 4 to answer any calls, only place outgoing calls |
| 14:55.24 | FITA1 | hi I am using eicon diva 4BRI card, I have configured the eicon-diva-server and complied chan-capi-HEAD , The problem is that When asterisk starts I can call-out using CAPI but after some duration of time CAPI reports unable to create channel of type capi and does not initiates call, anyone can help??? |
| 14:55.26 | opc0de | so I want it to be part of the outgoing call group, but not incoming.. |
| 14:56.53 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-231-103.telkomadsl.co.za) |
| 14:57.30 | mut | anyone ever had to deal with innovative or first data for credit card processing? |
| 14:57.39 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 14:57.58 | [TK]D-Fender | file : FACT : That until the Gulf war no country with a McDonald's ever attacked another country with one... |
| 14:58.08 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:58.09 | file | [TK]D-Fender, ooh |
| 14:58.16 | file | good to know |
| 14:58.25 | x86 | [TK]D-Fender: really? |
| 14:58.27 | mut | thats pretty much impossible anymore isn't it? |
| 14:58.31 | x86 | [TK]D-Fender: i never knew that |
| 14:58.54 | mut | and before the gulf war.. what was there? world war ii? |
| 14:59.03 | x86 | uh |
| 14:59.05 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:59.12 | x86 | vietnam, korea, gernada.... etc ;) |
| 14:59.24 | mut | ah yea forgot about viet and korea |
| 14:59.28 | mut | gernada doens't count |
| 14:59.52 | mut | my brain was fried talking to first data and innovative and verisign yesterday |
| 14:59.58 | mut | i was on the phone for 6 hrs yesterday |
| 15:00.06 | mut | took a half hour break in the middle |
| 15:00.10 | CoffeeIV_ | If I am recording with the Record() in hte dialplan, is there any way to allow the user to press any digit at any time to exit it and continue in the dialplan, or is it only * and # ? |
| 15:00.21 | mut | trying to figure out why master debit cards won't process thru our online card gateway |
| 15:00.42 | mut | the final 'resolution' was the innovative guy was supposed to call me back within the 9 o clock hour today |
| 15:00.54 | mut | clock just struck 10 on my watch |
| 15:01.01 | Dovid | anyolol |
| 15:01.14 | X-Gen | sucker |
| 15:01.21 | Dovid | mut: i use some one else, dont know how thier gateway works. been happy with them |
| 15:01.29 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:01.35 | mut | he had to do more 'research' |
| 15:01.44 | mut | o had all 3 companies on the phone at once yesterday |
| 15:01.44 | Dovid | lol |
| 15:01.49 | Dovid | goto luv newbies |
| 15:01.55 | X-Gen | Dovid: whats the service cost you ? u pay per month or per transaction ? |
| 15:02.07 | mut | called FD they conferences verisign, then i called inno and conferenced them into the call |
| 15:02.23 | mut | then the verisign guy hung up |
| 15:02.26 | Dovid | bpth |
| 15:02.29 | Dovid | both8 |
| 15:02.33 | mut | there was a frickin 20 minute HOLD TIME |
| 15:02.34 | Dovid | both* |
| 15:02.37 | mut | and the guy hangs up |
| 15:02.41 | Dovid | how sweet |
| 15:02.49 | *** join/#asterisk heka (n=Mango@80.80.174.140) |
| 15:03.08 | x86 | [TK]D-Fender: you can call UK for free? |
| 15:03.18 | x86 | [TK]D-Fender: arent you the one that tested my UK number before? |
| 15:03.37 | mut | o i was hot |
| 15:03.41 | mut | i had a migraine all nite |
| 15:03.47 | x86 | [TK]D-Fender: i'm having problems registering it with e164.org, and i'm thinking it no longer works... care to give me a quick call? |
| 15:03.51 | [TK]D-Fender | x86 : Free for ME, yeah |
| 15:03.52 | mut | i went to bed like 2 hrs early cause i couldn't stand to be awake anymore |
| 15:04.05 | Dovid | lol |
| 15:04.09 | [TK]D-Fender | x86 : not free for my company :) |
| 15:04.17 | Dovid | mut: u on the asterisk users lis ? |
| 15:04.21 | Dovid | list* |
| 15:04.38 | x86 | [TK]D-Fender: i wont answer, just want to see if it rings |
| 15:04.47 | x86 | [TK]D-Fender: +44 871 309 4409 |
| 15:05.02 | heka | anybody can help me with jitter buffer patch for sip? |
| 15:05.14 | mut | no |
| 15:05.22 | mut | y |
| 15:05.28 | heka | should I do jb-enable = yes on global setting or for each user? |
| 15:05.33 | [TK]D-Fender | x86 : Sorry, no can do for now.... |
| 15:05.40 | x86 | [TK]D-Fender: mmk |
| 15:05.42 | Dovid | mut: there is some one there by the name of doug that had problems, some one sugested that he have a bottle of booze for just in case |
| 15:05.53 | Dovid | i think u whould join the club. it allways helps |
| 15:06.13 | brad_mssw | anyone else notice the iaxy's really really really suck ? |
| 15:06.14 | mut | i would but i've got myself a lil ulcer |
| 15:06.26 | mut | would probly kill me |
| 15:06.28 | mut | :P |
| 15:06.32 | *** join/#asterisk Jon335 (i=Jon335@ottawa-hs-209-217-119-86.d-ip.magma.ca) |
| 15:06.37 | mut | spose it'de put me as ease tho |
| 15:07.04 | mut | keeled over in pain, rush to the hospital, INJECT MORPHINE! |
| 15:07.11 | brad_mssw | I mean, if you turn on qualify, it reachable then unreachable, reachable, then unreachable ... then even when it is reachable, it doesn't work half the time ... (and yes, I've tried it on different switches, and a Linksys PAP2 SIP adapter on the same switch never has glitches) |
| 15:07.38 | FITA1 | <PROTECTED> |
| 15:09.24 | Dovid | lo |
| 15:09.25 | Dovid | lol |
| 15:09.35 | Dovid | so u can allways aim for the whacky weed |
| 15:09.37 | Dovid | ;):):) |
| 15:10.19 | x86 | anyone else in the UK or can call the UK for free? |
| 15:10.30 | x86 | i wont answer just want to see if it rings |
| 15:10.42 | opc0de | can anyone tell me how to define an outgoing-only channel? |
| 15:11.18 | *** join/#asterisk Dovid (n=Dovid@89-138-76-126.bb.netvision.net.il) |
| 15:12.32 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:15.12 | Assid | can someone help me on this: exten => s,n,Set(MONITOR_FILENAME=${CALLERIDNUM}) |
| 15:15.31 | opc0de | assid: what's the problem? |
| 15:15.47 | Assid | it doesnt set the filename apparenly |
| 15:15.50 | opc0de | I think you might need to change n to 1 |
| 15:15.57 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 15:16.09 | Assid | before answer? |
| 15:16.19 | opc0de | exten => s,1,Answer |
| 15:16.28 | opc0de | then you can put the rest |
| 15:16.49 | Assid | thats where i had it |
| 15:17.03 | opc0de | and you can try exten => s,n,NoOp(MONITOR_FILENAME=${CALLERIDNUM}) to see what it prints on the console |
| 15:17.31 | fourcheeze | on the snom 360 you can select an outgoing line using the cursor key on the display |
| 15:17.48 | fourcheeze | only seems to let you choose lines1-4 |
| 15:17.55 | fourcheeze | any idea how to choose 5+ ? |
| 15:18.00 | Assid | okay how do i make it agent-1001-callerid-YMDH:i |
| 15:18.16 | GerbilNut | I didn't even know you could do that on the snoms... that's pretty sweet |
| 15:19.14 | *** join/#asterisk pengyong (n=lala@222.185.18.93) |
| 15:19.39 | opc0de | Assid: you can concatenate by just agent-1001-${CALLERIDNUM} I don't know the function for getting the current date, but I'm sure it's there under the function/application/variable list |
| 15:20.20 | Assid | err.. how about agentid? |
| 15:21.48 | opc0de | don't know |
| 15:30.09 | opc0de | can anyone tell me how to _not_ have asterisk answer a line? I have a channel in a separate context, which has no answer directive, yet asterisk still picks up on this line |
| 15:30.17 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 15:30.37 | eric_hill | opc0de: You can't really define an "outgoing only" channel. You don't really have control over that. You can ignore inbound calls though... |
| 15:30.53 | eric_hill | What's the channel type? |
| 15:31.09 | opc0de | eric_hill: FXO |
| 15:31.28 | opc0de | I thought by putting this channel in another context and not defining any extensions for it, asterisk won't pick up the line |
| 15:31.33 | eric_hill | exten => s,1,Hangup // Does this work? |
| 15:31.45 | opc0de | lemme try, although I think that'll still pick the line up. |
| 15:32.55 | *** join/#asterisk Chopinhauer (n=Chopinha@morgoth.karwasz.org) |
| 15:33.33 | Assid | opc0de : -- Executing Set("SIP/301-0d72", "MONITOR_FILENAME=AGENTBYCALLERID_"Satish" <301>-301-20060329-210019") in new stack |
| 15:33.33 | Assid | <PROTECTED> |
| 15:33.46 | Z0m81e | fourcheeze: You can select the other lines in the web interface |
| 15:34.12 | opc0de | Assid: that looks pretty good |
| 15:34.19 | Assid | doesnt work |
| 15:34.23 | heison | I'm trying to call a specific extension on a telephone number, is it possible to do it all in one shot? i.e. how can I setup Asterisk to call 2345678 x1234 directly? Can I put commas after the telephone number to generate the waits? |
| 15:34.36 | fourcheeze | Z0m81e: ahh yes |
| 15:34.43 | fourcheeze | not so good when you're just about to make a call though |
| 15:35.01 | *** join/#asterisk ro0t2 (n=hack@60-240-240-183.tpgi.com.au) |
| 15:35.24 | Z0m81e | fourcheeze: probably not :) but thats how you do it... Perhaps you'd have to try it, if you press the button under reg scroll down then hit the tick does that do it? |
| 15:35.36 | fourcheeze | yeah, just foudn that one |
| 15:35.39 | opc0de | eric_hill: hmm, using hangup does work.. interesting |
| 15:35.43 | fourcheeze | however it doesn't use the friendly names then |
| 15:35.53 | *** join/#asterisk smeevil (n=smeevil@217-19-24-16.dsl.cambrium.nl) |
| 15:35.54 | smeevil | hello |
| 15:36.14 | eric_hill | heison: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial --- See the D(digits) variable |
| 15:36.41 | heison | eric_hill: thx! |
| 15:36.45 | eric_hill | aye |
| 15:36.56 | Z0m81e | fourcheeze: likewise on mine using 4.5 firmware I think |
| 15:36.57 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 15:38.11 | smeevil | could someone tell me what is wrong when asterisk does not detect dtmf tones from any softphone (sip) |
| 15:38.22 | Z0m81e | fourcheeze: it uses accountname@registrar if you set your account name to be something sensible and the auth username to your username that might be better |
| 15:38.40 | Z0m81e | fourcheeze: or not my registration just failed :) |
| 15:39.02 | eric_hill | smeevil: What kind of softphones? Also, what encoding? g711? g729? Etc... |
| 15:39.57 | smeevil | sjphone and estara |
| 15:40.01 | smeevil | checking codecs hold on |
| 15:40.23 | smeevil | g723 and g711 |
| 15:40.26 | Assid | argh.. its not letting me set the monitor_filename |
| 15:40.37 | Assid | nor is it using it |
| 15:41.07 | smeevil | sending dtmf as rfc2833 |
| 15:41.25 | Z0m81e | smeevil: is * set to rfc2833 or inband? |
| 15:41.41 | smeevil | where can i check that ? |
| 15:41.50 | eric_hill | http://www.voip-info.org/wiki-Asterisk+sip+dtmfmode |
| 15:41.57 | smeevil | ty |
| 15:42.04 | Z0m81e | it would be in the context for your line |
| 15:45.01 | Assid | umm can someone help me with this monitor filename |
| 15:45.08 | Z0m81e | fourcheeze: 1 other thing, bind the lines to the function keys then just press the key of the line you want to dial |
| 15:45.15 | smeevil | do i understand it correctly that i have to put dtmfmode=inband in the [general] part of me sip.conf ? |
| 15:45.31 | Z0m81e | smeevil not if your phone is sending it rfc they must agree |
| 15:45.31 | fourcheeze | Z0m81e: I tried to do that but it dialed out on line 1 anyway |
| 15:45.49 | Z0m81e | fourcheeze: hmm i'm sure mine works when setup like that, let me test it a mo |
| 15:46.34 | eric_hill | smeevil: Yes, but you need to use dtmfmode=rfc2833 |
| 15:47.20 | jaike | dtmfmode=rfc2833 works with most softphones |
| 15:48.05 | ro0t2 | gah ! been having problems with iax channels all because of 3 lines in the conf file......what a waste of 9 hours |
| 15:48.20 | heison | eric_hill: I see DTMF digits being sent, but it may have been sent too early.... |
| 15:48.32 | heison | <PROTECTED> |
| 15:48.33 | heison | <PROTECTED> |
| 15:49.09 | ro0t2 | for some reason if i enable jitterbuffer i get a NOTICE[1613]: sched.c:286 ast_sched_del: Attempted to delete nonexistent schedule entry 122! but turn it off and everything is peachy |
| 15:49.45 | smeevil | hmm changed it to rfc2833 , made sure the softphones do sent that as well. but still digits are not being picked up |
| 15:50.39 | Z0m81e | ro0t2: do you have an interface for timing? like a zaptel or ztdummy? earlier somoene pointed me to that for a similar error |
| 15:51.03 | eric_hill | heison: How about using a M(acro) and having the macro Wait for a few seconds, then playback the digits. |
| 15:51.45 | Katty | file :< |
| 15:52.01 | Katty | scott :<< |
| 15:53.07 | ro0t2 | Z0m81e: i suspect that could be the problem...i have an X100P in the box....but is not plugged in (to the line) im gonna test this now that you mention this as i suspect it may be the cause.....but im quite happy it is working now :DDD |
| 15:54.06 | heison | eric_hill: you mean using application sendDTMF? |
| 15:54.39 | eric_hill | right. |
| 15:54.55 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 15:55.03 | eric_hill | (FWIW, I've never done what you're doing - I'm just guessing things you can try...) |
| 15:55.27 | ro0t2 | zap and sip channels were working fine....but i didnt actually think to check the sip channels til just before, although the error didnt look network related...it looked like a possible cause (that is i assumed SIP was fucked too) |
| 15:55.47 | eric_hill | heison: Better idea... |
| 15:55.56 | opc0de | can anyone give me some advice? I have an [incoming] context for handling incoming calls, and an [internal] context for my SIP phones.. what is the standard method for allowing people dialing in to connect to internal extensions, as well as allow people in the [internal] context to contact each other? |
| 15:56.41 | opc0de | should I put something like exten => ${USER1},1,Goto(internal,${USER1},1) ? and then in [internal] have something like "exten => ${USER1},1,Macro(stdexten,${USER1},SIP/${USER1}) ? |
| 15:56.45 | eric_hill | heison: Use the A(...) command to play an announcement to the called party first, and use an empty silence.gsm file for the correct duration. |
| 15:56.58 | opc0de | so basically duplicating the users extension definition in both places? |
| 15:57.31 | Assid | okay can someone please help me with this monitor_filename |
| 15:57.37 | eric_hill | Then your command becomes Dial(blah/blah,60,A(silence-5)D(12345) |
| 15:57.39 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:57.56 | eric_hill | Assid: Not sure what you're trying to do... my scrollback buffer isn't that big. |
| 15:58.15 | Assid | exten => s,n,Set(MONITOR_FILENAME=${AGENTBYCALLERID_${CALLERIDNUM}}-${CALLERIDNUM}-${TIMESTAMP}) |
| 15:58.16 | heison | eric_hill: that may be easier... let me try that |
| 15:58.42 | Assid | eric_hill: the agent id isnt being saved.. and the file itself isnt being renamed |
| 15:58.54 | Assid | it still using the old unique format |
| 15:59.20 | Assid | not only did i reload extensions.. i even restarted asterisk.. no help |
| 15:59.42 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:00.04 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.150) |
| 16:00.04 | *** join/#asterisk leicaWRK (n=leica@lfw505.securepod.com) |
| 16:00.15 | leicaWRK | hello? |
| 16:00.20 | leicaWRK | anybody around? |
| 16:00.26 | eric_hill | When you NoOp that same line, what is the output? I.e. does the filename have any invalid characters? |
| 16:00.38 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 16:00.42 | salviadud | i just got here |
| 16:00.55 | heison | eric_hill: it did A(x) properly but ignored D() |
| 16:01.12 | heison | eric_hill: only saw -- Zap/5-1 answered SIP/2508-7581 |
| 16:01.12 | heison | <PROTECTED> |
| 16:01.27 | heison | eric_hill: let me try the Macro way |
| 16:01.30 | leicaWRK | sorry first time setting up asterisk here, so sorry if I ask basic questions... |
| 16:01.30 | eric_hill | heison: Damn. :) |
| 16:01.35 | brodiem | is it possible to do the equivalent of a UserEvent using the Manager API? |
| 16:01.40 | ro0t2 | opc0de: you could use an include statement include => context_name ? |
| 16:02.01 | ro0t2 | would include all the extentions from another context in the other one.... |
| 16:02.38 | leicaWRK | anybody know why one phone would be assigned an IP with a mask of 0.0.0.0 and status as "unmonitored"? |
| 16:02.44 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 16:03.19 | brodiem | leicaWRK, set qualify=yes in sip.conf for monitoring... the netmask is probably just what it received from the dhcp server if you're using dhcp |
| 16:03.33 | leicaWRK | aha, thanks |
| 16:03.38 | opc0de | ro0t2: the problem with using the include is that I allow my [internal] context access to make outgoing calls/long distance calls.. so if I include internal, then I allow all incoming users to dial outgoing calls |
| 16:03.46 | eric_hill | Assid: Is this something like what you're trying to do? http://www.voip-info.org/wiki/view/Asterisk+Bounty+record+call+queues+with+detailed+filename |
| 16:03.59 | leicaWRK | yeah it's odd that one of the dhcp clients would get a mask of 0.0.0.0 but not the others |
| 16:04.22 | Assid | similar i guess |
| 16:04.32 | brodiem | leicaWRK, if the phone is still able to talk to * I wouldn't worry about it |
| 16:04.32 | Assid | even if i drop the whole agent thing |
| 16:04.37 | Assid | the file name still doesnt change |
| 16:04.46 | leicaWRK | yeah it talks fine |
| 16:04.52 | leicaWRK | cool |
| 16:04.57 | eric_hill | Assid: Can you get the filename then rename it after the call is completed? |
| 16:05.21 | opc0de | ro0t2: what I'd like is to have a context which defines my sip phones and include that from my [incoming] context so that dialing in users can reach internal phones, but disallow dialing in users from being able to call out |
| 16:05.30 | Assid | thats the thing.. monitor_filename is supposed to do that for me |
| 16:05.47 | [TK]D-Fender | opc0de : pastebin your extensions.conf |
| 16:06.32 | opc0de | [TK]D-Fender: right now it's a realy mess |
| 16:06.35 | eric_hill | Assid: You haven't specified a path for the file. Try prepending /tmp/monitor-${AGENTBYCALL....... to your filename |
| 16:06.36 | opc0de | s/realy/real |
| 16:06.36 | brodiem | Question: I know events are generally sent from * to the Manager API, but is there a way to use the Manager API to send an event instead? |
| 16:06.49 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 16:06.53 | [TK]D-Fender | opc0de : that bad? |
| 16:07.32 | opc0de | [TK]D-Fender: I'm just trying to find out the standard way of separating incoming context from internal context, while allowing people dialing into the incoming context to connect with uses on the internal context, without having to duplicate the extension definitions in both contexts |
| 16:08.00 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 16:08.13 | smeevil | @.@ |
| 16:08.27 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 16:09.46 | brodiem | maybe if I explain this more it'll help... |
| 16:10.03 | [TK]D-Fender | opc0de : I got that.... so pastebin it all and I'll see what I can do to help. |
| 16:10.04 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-180-235.austin.res.rr.com) |
| 16:10.22 | warthawg | i officially give up on trying to make zultys phones work with asterisk |
| 16:10.52 | warthawg | though i think the problem has to do with the phone registration not properly setting the userid |
| 16:11.16 | warthawg | tcpdump logs available for anyone interested |
| 16:11.23 | brodiem | The DND dialplan does a "UserEvent" so that FOP receives notification of DND being enabled/disabled so that it can update its status for that extension. I'm using an snom360 phone that supports action URLs, so I want to use the phone's built-in DND button instead of needing to dial an ext to turn on/off DND. So, I'm trying to make the action URL for the phone's DND button update FOP's DND status, so how could I trigger that UserEvent fr |
| 16:14.48 | *** join/#asterisk Assid (n=assid@59.183.8.196) |
| 16:14.59 | Assid | sorry |
| 16:15.00 | Assid | got cut |
| 16:15.03 | Assid | eric_hill: doesnt work |
| 16:15.30 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 16:16.18 | opc0de | [TK]D-Fender: hmm, I found an example extensions.conf file, it looks like it might do what I want.. |
| 16:18.10 | ro0t2 | alright, have a question, i have an IAX provider that only allows ilbc and g729.....i have a sipura-2000 ATA which supports g729 and is set to prefered codec of ulaw but the "use only prefered codec" tab is unchecked...when making a call i get this set_format: Unable to find a codec translation path from g729 to ulaw....if i manually configure the ATA to g729 the call will go through im just curious of a better way because its a pain |
| 16:19.07 | opc0de | [TK]D-Fender: yeah, I think I got it.. I created an [ext-local] context and put my SIP phone extensions in it, then I put an "include => ext-local" inside my [incoming] context, and then I created an [internal] context and put "include => outbound-long-distance; include => ext-local" into it, and assigned the context to my sip phones in zapata.conf to be "internal" |
| 16:19.29 | [TK]D-Fender | Thats the way to do it. |
| 16:20.10 | eric_hill | Assid: Sorry, I've got no other ideas. Maybe browse the source and see what it's doing??? |
| 16:20.29 | opc0de | good, I knew there was a better way to do it than putting a bunch of goto's inside my [incoming] context, duplicating all my extensions. just had to get my head around the way the contexts work |
| 16:23.14 | *** join/#asterisk enots (i=dimka@freelsd.net) |
| 16:24.07 | *** join/#asterisk dVoka (n=dVoka@CPE-65-29-147-86.wi.res.rr.com) |
| 16:28.29 | *** join/#asterisk LeeForkenbrock (n=LeeForke@ip67-95-66-69.z66-95-67.customer.algx.net) |
| 16:29.40 | dVoka | I'm affiliated with 10 small businesses (10-25 ppl per biz) and am considering putting them all on one asterisk system at my office. Am thinking of integrating Asterisk into their current "ma bell" pbx & providing them with outbound voip. So ... one asterisk system at my office, and perhaps one multi-port fxo/fxs adapter integrated into current system. Is this possible or am I dreaming? |
| 16:29.57 | Hmmhesays | i hate it when idiots buy things against all of my recommendations |
| 16:30.16 | Hmmhesays | its like "seriously you tool" i know what i'm talking about |
| 16:31.33 | [TK]D-Fender | Hmmhesays : You can't save them all... I learned this long ago, and facing it early lengthens your life... |
| 16:31.57 | salviadud | lots of tools out there |
| 16:32.40 | salviadud | dvoka, completely possible dude |
| 16:33.13 | salviadud | it might have to be a big server though... |
| 16:33.40 | salviadud | and it'll probably chew up some bandwith |
| 16:35.21 | dVoka | salviadud - thanks man! I agree on the server & bandwidth ... will build out a rack if I move forward & have redundant lines with 90k upload allocated to each line. Am thinking the hard part might be integrating the voip equipment into the "ma bell" stuff - particularly if the phone co leases to the biz. |
| 16:35.56 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 16:37.57 | *** join/#asterisk willcampos123 (n=willcamp@198.87.100.3) |
| 16:38.09 | dVoka | salviadud: the person I'm working with doesn't actually want spa3000 units on each phone, but rather ... a "call hunt" type feature so that two of 10 lines would be voip - the "spa3000" type unit would actually be connected to current "ma bell" PBX. |
| 16:39.00 | salviadud | interesting, i happen to own a spa 3000 |
| 16:39.19 | salviadud | i guess it really depends on how limited the ma bell pbx is |
| 16:39.35 | willcampos123 | Hello, is there any way to avoid sip REALTIME to do continues queries over the same table trying to authenticate a user? Let's say have the real time setup to query just the ipadd, or host, or name? |
| 16:39.42 | salviadud | you could register the spa3000 with asterisk as 2 seperate sip channels |
| 16:40.06 | LeeForkenbrock | I'm a little confused about something. I could sound totally stupid. But, my Telco company, in the past was unable to send caller id from a LD T1 circuit. Reason why is beyond me. So, they would send me something called enhanced DNIS in the format *ANI*DNIS* and I would parse out the ANI with AGI. Well, i'm setting up a new T1 with them and they say they are 100% sending it this same way with this one....but I jus |
| 16:41.06 | dVoka | salviadud: thanks |
| 16:46.48 | wasim | LeeForkenbrock: don't leave us in suspense |
| 16:47.01 | iDunno | mmhmm |
| 16:47.47 | LeeForkenbrock | wasim: suspense? |
| 16:47.56 | wasim | <PROTECTED> |
| 16:48.18 | LeeForkenbrock | wasim: oh, it must have chopped it off...is there a max message length? |
| 16:48.41 | LeeForkenbrock | but I just get the 10 digit DNS, but on the other hand, the callerid var is being set. Is that format a standard and Asterisk automatically parses it in newer version for me, or does this mean they might be sending caller id correctly now and DNIS normally. |
| 16:49.41 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 16:49.43 | kardecallan | _Paulo_ I have Installed the unicall module, but when asterisk starts the channels are showed as Idle, while in the CO a showed as blocked. |
| 16:49.49 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 16:49.59 | *** join/#asterisk file (n=jcolp@mctnnbsa24w-142167058031.pppoe-dynamic.nb.aliant.net) |
| 16:50.07 | xermesx | hi all |
| 16:50.13 | Katty | file :> |
| 16:50.18 | file | Katty! |
| 16:50.26 | xermesx | i d like to install asterisk cvs in my rhel es 3 |
| 16:50.37 | Chopinhauer | I have a strange problem with a Wildcard X101PÂ : after hanging up a call Asterisk closes the channel (show channels shows no channels) but the card remains Off-Hook. Do you know what the problem could be (I am under linux 2.6.15/amd64/zaptel 1.2.4)? |
| 16:50.49 | xermesx | so i need bristuff-0.2.0-RC8f-CVS . right ? |
| 16:51.23 | backblue | xermesx: use 0.3.x... |
| 16:51.33 | nettie | damn.. I'm still stucked with call progress |
| 16:51.41 | backblue | 0.2.0 will not work in cvs branch |
| 16:51.49 | xermesx | thx backblue |
| 16:52.20 | kardecallan | _Paulo_, I send you a sample extracted from asterisk log. |
| 16:52.20 | kardecallan | Mar 29 13:37:33 WARNING[3563] chan_unicall.c: MFC/R2 UniCall/14 Block |
| 16:52.21 | kardecallan | Mar 29 13:37:33 WARNING[3563] chan_unicall.c: MFC/R2 UniCall/14 1101 -> [1/40000000/Idle /Idle ] |
| 16:52.21 | kardecallan | Mar 29 13:37:51 WARNING[3577] chan_unicall.c: MFC/R2 UniCall/18 <- 1001 [1/40000000/Idle /Idle ] |
| 16:52.21 | kardecallan | Mar 29 13:37:51 WARNING[3577] chan_unicall.c: Unicall/14 event Far end unblocked |
| 16:52.52 | kardecallan | Can you help me? |
| 16:53.51 | konfuzed | iDunno, nah its not even lunch yet |
| 16:54.02 | xermesx | backblue, i can find no more cvs version of asterisk to download |
| 16:54.33 | tzanger | xermesx: we use subversion now |
| 16:54.40 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 16:56.07 | iDunno | konfuzed: it's nearly 1800 BST :) |
| 16:57.19 | konfuzed | iDunno, no it isnt, its 11:58 EST 8^) |
| 16:57.21 | brodiem | The DND dialplan does a "UserEvent" so that FOP receives notification of DND being enabled/disabled so that it can update its status for that extension. I'm using an snom360 phone that supports action URLs, so I want to use the phone's built-in DND button instead of needing to dial an ext to turn on/off DND. So, I'm trying to make the action URL for the phone's DND button update FOP's DND status, so how could I trigger that UserEvent fr |
| 16:58.01 | kardecallan | Is there anybody that can help me? |
| 16:58.39 | iDunno | konfuzed: that's not a sensible timezone, though ;) |
| 16:59.01 | konfuzed | ok its 1:29 Newfy Standard Time |
| 16:59.09 | konfuzed | 8^) |
| 16:59.25 | iDunno | one of our webapps doesn't quite understand them ;) |
| 16:59.37 | iDunno | (where doesn't quite actually translates to "doesn't at all") |
| 17:00.23 | konfuzed | your web app probably speaks British instead of Newfy |
| 17:02.09 | konfuzed | cause newfy is really derived from Irish and well it seems the Irish and British have never really understood each other. |
| 17:02.59 | konfuzed | maybe if you taught your web app the Rosetta Stone ;^) |
| 17:05.35 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 17:05.48 | xermesx | if i use asterisk 1.26, with bristuff 0.3.0-PRE-1f and florz's patch, shoul my old CVS config files still work ??? |
| 17:06.24 | *** part/#asterisk dVoka (n=dVoka@CPE-65-29-147-86.wi.res.rr.com) |
| 17:09.01 | salviadud | you guys know about a sip client i can install on a mobile phone? |
| 17:09.08 | salviadud | java-based maybe... |
| 17:09.56 | iDunno | konfuzed: heh - that might work... alternatively, I could just teach the Developers about timezones ;) |
| 17:11.52 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 17:12.30 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 17:13.20 | konfuzed | iDunno, well you could point out that the timed server is very effective and accurate |
| 17:16.44 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 17:16.50 | *** join/#asterisk bytewarrior (n=bytewarr@p54A46777.dip.t-dialin.net) |
| 17:17.04 | bytewarrior | hi |
| 17:17.05 | *** join/#asterisk vykarian (n=stefano@200.138.30.10) |
| 17:17.10 | vykarian | hi guys |
| 17:17.22 | project_2501 | konfuzed: I know lots of newfy's |
| 17:17.43 | vykarian | does anyone have a guide/tutorial/manual for Asterisk integration with analog PABX? I already have a SIP trunk up and running.. |
| 17:18.20 | *** join/#asterisk jbroome (n=jbroome@63-168-10-93.celito.net) |
| 17:18.45 | bytewarrior | does anybody know about the grandstream gxp-2000 sending SIP REGISTER requests with a missing byte (CR or LF, not sure)? |
| 17:18.57 | pigpen2 | Hi all...I got an urgent one: I have a Digium 4 port PRI w/echo cancelation... with 2 PRI's connected. |
| 17:19.14 | konfuzed | my mothers family had a Reunion in Montreal and 250 Newfys descended from across north america |
| 17:19.33 | salviadud | Newfys? |
| 17:19.44 | pigpen2 | when I call in, with the call passing to a Digium 2431, to an fxs port, the dial tones like "1 , 2, 3" come through as "Bee-eep" |
| 17:19.48 | jbroome | people from newfoundland |
| 17:19.51 | pigpen2 | with the "-" being a pause. |
| 17:19.58 | konfuzed | salviadud, too bad your salvia is a dud |
| 17:20.04 | pigpen2 | So the digit comes through as two digits. |
| 17:20.16 | salviadud | actually, its supposed to be salviadude |
| 17:20.22 | konfuzed | dud |
| 17:20.25 | konfuzed | ;^) |
| 17:20.31 | salviadud | yet some irc servers don't handle the last e |
| 17:20.41 | pigpen2 | I have "relaxdtmf=yes" and "faxdetect=no" |
| 17:20.43 | pigpen2 | ideas? |
| 17:20.47 | salviadud | hehe |
| 17:20.55 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
| 17:20.56 | saftsack | hi |
| 17:21.03 | saftsack | is AMP free software? |
| 17:21.06 | pigpen2 | I have isolated the issue to an issue with the PRI... |
| 17:21.10 | salviadud | don't be smokin' salvia, you might realize you were once a buddhist monk in 1863 |
| 17:21.44 | pigpen2 | saftsack, I think it is gpl, so yes. |
| 17:22.00 | saftsack | thanks :) and can i run it on another machine than on my *? |
| 17:22.13 | pigpen2 | I haven't tried. |
| 17:22.47 | saftsack | do you have it run on your * server directly? |
| 17:22.53 | [TK]D-Fender | saftsack : It expects the * files to be in their normal place otherwise it won't work. |
| 17:23.21 | saftsack | can i apply amp on a normal * installation? |
| 17:23.33 | pigpen2 | yes. |
| 17:23.40 | pigpen2 | but it will re-write your configs. |
| 17:23.46 | saftsack | what configs? |
| 17:23.55 | pigpen2 | any current configs.... |
| 17:24.01 | saftsack | thats very bad :( |
| 17:24.06 | saftsack | i dont want to have a configuration tool |
| 17:24.13 | pigpen2 | then don't use amp |
| 17:24.14 | saftsack | i just want to have a user web based tool |
| 17:24.32 | saftsack | for hearing vmails and viewing to missed calls, etc. |
| 17:25.25 | pigpen2 | shit...I just want my dtmf tones to pass correctly. |
| 17:25.31 | pigpen2 | hmm...don't use amp for that... |
| 17:25.34 | mut | anyone know how far those tdm2400 |
| 17:25.38 | mut | push signal? |
| 17:26.00 | pigpen2 | I am running one about 120ft of cat6 |
| 17:26.34 | saftsack | pigpen do you know better solutions? |
| 17:26.41 | mut | oh |
| 17:26.45 | saftsack | is ARI good for this? |
| 17:26.47 | mut | hm |
| 17:26.59 | mut | k well a channel bank would probly still be better for my use then |
| 17:27.10 | mut | adtrans run 16000ish feet |
| 17:27.30 | bytewarrior | my ISP's SIP server claims the REGISTER request of my gxp-2000 is not rfc3261 compliant. I read the rfc, it is probably because there is a byte missing at the end of the request. |
| 17:27.33 | bytewarrior | any ideas? |
| 17:28.36 | pigpen2 | saftsack, I am trying to remember what Asterisk at home uses for viewing vm's and such...check it out... |
| 17:29.22 | pigpen2 | mut, I haven't pushed it any further....and until I get my dtmf tones...I cannot do any playing... |
| 17:29.46 | pigpen2 | ManxPower, r u around O' all seeing Eye...? |
| 17:30.17 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 17:30.37 | pigpen2 | I guess the "Eye" is shut. |
| 17:30.53 | wasim | viking alert! |
| 17:31.29 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 17:34.44 | Hmmhesays | i hate this guy |
| 17:36.01 | file | ctooley, you! |
| 17:37.06 | Assid | anyone have MONITOR_FILENAME wworking? |
| 17:38.28 | jaike | Assid: its always been working |
| 17:38.50 | *** join/#asterisk ToTo (n=ToTo@host123-121.pool8258.interbusiness.it) |
| 17:38.51 | CoffeeIV_ | Assid: it works for me |
| 17:39.02 | ctooley | file, you! |
| 17:39.19 | Assid | it just refuses to work for me |
| 17:40.11 | Assid | http://pastebin.com/629190 |
| 17:40.18 | file | ctooley, how goes? |
| 17:40.30 | ctooley | file, it goes. stuff happens |
| 17:40.37 | file | ctooley, sounds exciting |
| 17:41.14 | Assid | jaike , CoffeeIV_: it shows the monitor file name being set.. but the file always comes up as the default |
| 17:41.26 | ctooley | file, how's stuff there? |
| 17:41.30 | file | ctooley, great |
| 17:43.04 | *** join/#asterisk xtr (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 17:43.11 | jaike | Assid: do you set the variable in the dialplan? Set(MONITOR_FILENAME= |
| 17:43.29 | Assid | yes |
| 17:43.37 | Assid | thats how it shows in the verbose |
| 17:44.03 | jaike | can you add your dialplan in the pastebin |
| 17:44.33 | bytewarrior | does anybody know where I can get some help regarding my grandstream gxp-2000? |
| 17:44.40 | jaike | i only have the filename. i dont include the path. might help |
| 17:44.44 | Assid | http://pastebin.com/629197 |
| 17:44.54 | *** join/#asterisk bmg505 (n=leon@165.146.41.101) |
| 17:44.55 | Assid | i tried with filename only first |
| 17:44.57 | Assid | didnt work |
| 17:46.21 | brodiem | is anyone farmiliar with sending manager API events (i.e. UserEvent) without doing it in a dial plan? |
| 17:46.59 | Nugget | moo |
| 17:47.25 | justinu | brodiem: how else would you send them? |
| 17:47.35 | brodiem | justinu, that's what I need to find out :) |
| 17:48.31 | wunderkin | Assid, what is happening? the file isn't being recorded, or not using that filename? |
| 17:48.45 | brodiem | justinu, it's for snom 360 phones. They support action URLs for the phone's DND button. So I need the script that the action URL points to send a UserEvent so that FOP will see DND was enabled/disabled so that it can update the display reflecting that |
| 17:49.05 | nettie | Hi again guys.. I'm wondering if there's something I could check on my asterisk configuration regarding "call progress". When I call a number on the PSTN using my SIP VOIP carrier I dont hear the phone ringing. I can only hear the voice of the called party when they pickup. If they hangup or refuse the call I dont get the busy tone.. so they phone stucks there till asterisk timeout comes in. Anyone have any suggestion please? Thanx in adva |
| 17:49.07 | brodiem | i don't know of any other way for FOP to get DND events |
| 17:49.36 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 17:50.00 | Assid | wunderkin: not usinhg that filename |
| 17:50.03 | justinu | well, it appears you can't send userevents from the AMI itself |
| 17:50.39 | Assid | <PROTECTED> |
| 17:50.42 | Assid | thats how i get it |
| 17:50.44 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 17:50.54 | brodiem | I can't think of any othe way to do DND where the phone's display actually reflects that DND is on/off |
| 17:51.03 | wunderkin | Assid, i dont think ive ever set a monitor_filename outside of the normal directory, try without, and then try making sure that it can actually write there |
| 17:51.17 | brodiem | and it's much more convenient just to have a dnd on/off button on the phone itself instead of needing to dial extensions to enable/disable |
| 17:51.22 | wunderkin | try without specifying the directory i mean |
| 17:51.30 | Assid | tried that too |
| 17:51.31 | Assid | didnt work |
| 17:51.52 | Assid | -- Executing Set("SIP/301-f20e", "MONITOR_FILENAME=1143674363.6-301-20060329-231925") in new stack |
| 17:52.25 | wunderkin | ive never had a decimal in a monitor_filename either |
| 17:52.32 | Assid | doesnt matter |
| 17:52.34 | Assid | it doesnt work |
| 17:52.44 | Assid | either which way |
| 17:52.54 | wunderkin | dont know what to tell you, ive never had a problem with it either |
| 17:53.06 | brodiem | Assid, what is the problem exactly? |
| 17:53.13 | jaike | ive seen this before...think its got something to do with queues.conf |
| 17:53.16 | Assid | brodiem: cant set the monitoring filename |
| 17:53.36 | Assid | the file uses the default agent-1001-1143674365-9.gsm |
| 17:54.09 | brodiem | Assid, are you setting the monitor based on the SIP account, the queue, or the agent? |
| 17:54.45 | *** join/#asterisk stoffell (n=stoffell@d5153FF4E.access.telenet.be) |
| 17:55.56 | GerbilNut | anyone in here configured DUNDi between two systems via IAX? |
| 17:56.10 | Assid | im setting it in the dialplan .. i officially wanna save it as agentnumber-callerid-timestamp |
| 17:56.12 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 17:56.14 | kippi | hey |
| 17:56.17 | jaike | Assid: you have monitor-format= in queues.conf? |
| 17:56.18 | Assid | but i cant get the agentnumber either |
| 17:56.32 | Assid | jaike: yes.. as i said: agent-1001-1143674365-9.gsm works fine |
| 17:56.33 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 17:56.36 | kippi | what is the recomended limit for registering SIP handsets with asterisk |
| 17:56.54 | Assid | it saves to that file name format no problems.. i just cant override it |
| 17:56.54 | brodiem | Assid, do you have recording turned on in agents.conf? |
| 17:57.08 | Assid | brodiem: yes.. agent-1001-1143674365-9.gsm WORKS fine |
| 17:57.18 | Assid | just cant change the name |
| 17:57.58 | brodiem | I didn't think it was possible to change that filename |
| 17:58.17 | wunderkin | yeah you can |
| 17:58.50 | wasim | MONFILE |
| 17:59.10 | FarrisG | I've read the wiki pages on door phones, but I have sort of a different situation I'm hoping maybe someone can help with. We have an exisiting analog intercom phone for our door. The door unit has a button that sounds buzzer on the inside, and the receptionist has a button which turns on the intercom to speak with the person outside. I'd like to either use this existing phone and tie into an FXO line on our * server, or replace it with another simple |
| 17:59.12 | jaike | i remember |
| 17:59.21 | jaike | try recordagentcalls=no in agents.conf |
| 17:59.28 | FarrisG | I don't mind doing a little hacking around with the hardware, especially if it'll allow me to use the existing door intercom |
| 17:59.33 | brodiem | have you tried disabling recordings in agents.conf and just use your dialplan monitor instead? |
| 17:59.34 | jaike | i think its the agent app thats calling monitor |
| 17:59.44 | Assid | jaike: it will not record then |
| 17:59.56 | jaike | it will...because you have monitor-forma in queues.conf |
| 18:00.00 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
| 18:00.02 | jaike | just give it a try |
| 18:00.56 | brodiem | Assid, set record_in and record_out to Always in sip.conf? |
| 18:00.56 | Assid | brodiem: problem is the file name.. not the recording |
| 18:00.56 | Assid | recording takes place fine.. just the filename cant be set |
| 18:01.52 | Assid | jaike: as i said.. nothing to do with the file name.. put it off.. it didnt record |
| 18:02.00 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-174-24.nwrk.east.verizon.net) |
| 18:03.08 | brodiem | I know for me using recordings in agents.conf seems to take over MONITOR_FILENAME setting previously in the dialplan.. |
| 18:03.14 | b66mer | SIP firewall question for y'all... I setup a gizmo sip connection to my asterisk... I can call out with no problems (from asterisk to gizmo#s)... when I call in to the asterisk... my console says playing the main menu, but I hear nothing from my gizmo client |
| 18:03.18 | b66mer | any ideas? |
| 18:03.23 | brodiem | turning it off in agents.conf keeps what I use for MONITOR_FILENAME |
| 18:03.51 | Assid | hold on. |
| 18:04.01 | Assid | where does your file go? |
| 18:04.10 | jaike | brodiem: same here |
| 18:04.12 | Assid | in /var/spool/asterisk/monitor ? |
| 18:04.15 | brodiem | yes |
| 18:04.16 | pigpen2 | anyone around to help me on a dtmf issue on a pri? |
| 18:04.29 | Assid | i got savecallsin for another directory |
| 18:04.32 | justinu | just ask |
| 18:04.42 | Assid | and i see 1 file in /var/spool/asterisk/monitor as the fileformat i want |
| 18:05.03 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 18:05.03 | Assid | or rather near to wwhat i want |
| 18:05.13 | Assid | any way to get the agentnumber as well? |
| 18:05.35 | brodiem | Assid, since you'd be setting your MONITOR_FILENAME before an agent is assigned to the call I doubt it |
| 18:05.51 | jaike | Assid: i wanted to do the same |
| 18:05.52 | pigpen2 | justinu, I asked about an hour ago...but basically on a 4 port digium with echo cancelation, when calls come in, being passed to an fxs on a 2431, the tones are coming though as "bee-eeep" vs. "beep" |
| 18:06.03 | pigpen2 | I have narrowed it down to an issue with the pri. |
| 18:06.19 | brodiem | otherwise, ${AGENTBYCALLERID_${CALLERIDNUM}} will return the agent logged into ext $CALLERIDNUM |
| 18:06.20 | LostFrog | Is there a way to use queues in such a way as to ring directly to an available agent instead of puting the caller on moh first? |
| 18:06.21 | jaike | but yes, since its set before a call is assigned to an agent, not possible |
| 18:06.25 | pigpen2 | I have faxdetect=no and relaxdtmf=yes |
| 18:06.42 | justinu | what's a 2431? |
| 18:07.08 | LostFrog | Can you Dial an agent? |
| 18:07.16 | jaike | was hoping agents.conf would have a monitor_filename= setting |
| 18:07.32 | pigpen2 | Digium TDM2400P with 12FXS/4 FXO |
| 18:07.40 | Assid | hrmm.. okay i needed to disable recording for it to work |
| 18:07.41 | justinu | oh |
| 18:07.42 | pigpen2 | With integrated echo cancelation. |
| 18:07.51 | Assid | one sec. gonna try setting the savecallsin |
| 18:07.59 | jaike | setting recordagentcalls=yes always sets the filename to agent-uniqueid |
| 18:08.00 | pigpen2 | But like I said...the issue is with the pri side...not the 2400 |
| 18:08.08 | justinu | how did you determine that? |
| 18:08.30 | brodiem | on the subject of agent recording... do you guys see those recordings using ARI? |
| 18:08.43 | pigpen2 | Well...I mapped an extension to the zap port on the 2400 and dialed it from a sip phone...with no troubles. |
| 18:08.54 | Assid | jaike: apparenly so |
| 18:09.01 | justinu | how about recording the audio off the PRI call with ztmonitor? |
| 18:09.03 | Assid | isnt there a way to mix/match both |
| 18:09.07 | pigpen2 | dial in via the pri, I get "bee-eep" |
| 18:09.07 | justinu | verify the tones are broken |
| 18:09.28 | Assid | so i can come to know the agent AND the caller id |
| 18:09.30 | SplasPood | Anyone here familiar with iTalkBB ? |
| 18:09.37 | brodiem | or what do you guys use for a gui to manage the recordings and match them with the CDRs? |
| 18:09.52 | pigpen2 | justinu, errr....you lost me...use ztmonitor to regord the call directly via the pri? |
| 18:09.54 | jaike | Assid: if you find the solution..would like to know..hehe |
| 18:10.00 | *** join/#asterisk ToTo (n=ToTo@host123-121.pool8258.interbusiness.it) |
| 18:10.19 | justinu | yeah... ztmonitor can record to a file... you'd be recording the TDM stream before it even got into asterisk |
| 18:10.20 | pigpen2 | I have been monitoring the analog side with a butt set. 1 comes in as 11, 2 is 22 |
| 18:10.37 | pigpen2 | so it may be a telco issue? |
| 18:10.45 | justinu | well, that's what you'd determine with ztmonitor |
| 18:10.51 | justinu | (i doubt that) |
| 18:10.59 | brodiem | Assid only thing I could think of is a small script that periodically reads your CDRs, and renames the recording filenames accordingly |
| 18:11.01 | pigpen2 | justinu, you are a cool person. |
| 18:11.05 | pigpen2 | I will do it. |
| 18:11.06 | justinu | :) |
| 18:11.16 | pigpen2 | bbiab... |
| 18:11.55 | Assid | hrmm |
| 18:12.03 | Assid | btw: thanks jaike; brodiem |
| 18:12.18 | brodiem | np |
| 18:14.59 | *** join/#asterisk eipi (n=eipi@OL17-54.fibertel.com.ar) |
| 18:15.51 | *** join/#asterisk ebag (n=gabe@adsl-69-239-166-49.dsl.renocs.pacbell.net) |
| 18:17.20 | *** join/#asterisk Muecke77 (n=muecke77@p54A9F9AF.dip.t-dialin.net) |
| 18:17.33 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 18:18.28 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 18:18.37 | *** join/#asterisk meshuga (i=meshuga@c-67-160-86-86.hsd1.wa.comcast.net) |
| 18:18.53 | *** part/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 18:20.24 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 18:21.53 | pigpen2 | justinu, still around? |
| 18:22.05 | justinu | yes |
| 18:22.11 | pigpen2 | ok..I did a record... |
| 18:22.35 | pigpen2 | now, the "beeps" seem whole, in the recording (ie: text) |
| 18:22.52 | pigpen2 | ###################################* Rx: 28028 (28 ###################################* Rx: 28028 (28 ###################################* Rx: 28028 (28 ###################################* Rx: 28028 (28 ###################################* Rx: 16868 (28 |
| 18:22.56 | pigpen2 | oops. |
| 18:23.02 | justinu | oh... there's a way to make it record audio |
| 18:23.11 | pigpen2 | really? |
| 18:23.13 | justinu | ztmonitor -f i think |
| 18:23.28 | pigpen2 | I did ztmonitor 3 -vv -f filename |
| 18:23.37 | pigpen2 | what format? |
| 18:23.42 | justinu | so that doesn't actually record? |
| 18:23.44 | pigpen2 | maybe I just need to grab the file off... |
| 18:23.50 | justinu | what's in filename? |
| 18:23.50 | pigpen2 | I haven't opened it...hehe |
| 18:23.54 | justinu | program PCM ulaw |
| 18:23.56 | justinu | probably |
| 18:24.12 | pigpen2 | empty file actually... |
| 18:24.22 | justinu | hmm... |
| 18:24.39 | pigpen2 | but the output was cool. |
| 18:24.56 | pigpen2 | and the "beeps" look like "full" beeps. |
| 18:25.09 | justinu | yeah - you're not the first one to have double DTMF issues w/ * |
| 18:25.18 | justinu | i doubt it's the telco that's screwing it up |
| 18:25.25 | pigpen2 | ok... |
| 18:25.28 | pigpen2 | I am glad to hear. |
| 18:25.35 | pigpen2 | so now the fun part. |
| 18:25.41 | pigpen2 | any ideas? |
| 18:26.43 | pigpen2 | This last time I actually was recording a voicemail |
| 18:26.50 | pigpen2 | which I have emailed to me... |
| 18:27.00 | pigpen2 | want to hear what I am talking about? |
| 18:27.11 | justinu | i get the gist |
| 18:27.16 | pigpen2 | k |
| 18:27.34 | justinu | you said that when you call the FXS via a SIP channel, it's ifne? |
| 18:27.36 | justinu | it's fine |
| 18:27.46 | pigpen2 | correct. |
| 18:27.58 | pigpen2 | it is only when it transverses the pri |
| 18:28.07 | justinu | what if you call a PRI channel w/ SIP? |
| 18:28.09 | justinu | same problem? |
| 18:28.52 | pigpen2 | Well, this last test was from an outside cell phone, into the system, recording to a voicemail. |
| 18:28.55 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 18:29.08 | ManxPower | DTMF problems on PSTN ports are usually traced to one of several problems. |
| 18:29.17 | pigpen2 | if I dial from a sip phone, to a did mapped on the pri, then back in, I get the trunicated "beep" |
| 18:29.30 | pigpen2 | POTS or PRI? |
| 18:29.48 | ManxPower | 1) rxgain or txgain is out of whack 2) relaxdtmf is enabled or 3) (and this is usually only with IVRs) you need to increase the length of the DTMF tones (configurabile in 1.2.x) |
| 18:30.07 | pigpen2 | hmm...looks like a man with a plan... |
| 18:30.13 | justinu | certainly things to check |
| 18:30.17 | pigpen2 | Ok...currently rx/tx is 0.0 |
| 18:30.21 | ManxPower | pigpen, REMEMEBER, if you are not using ulaw or alaw then you should expect tones to be garbled. |
| 18:30.29 | *** join/#asterisk TheoC (n=theochao@68-191-219-240.dhcp.dntn.tx.charter.com) |
| 18:30.43 | pigpen2 | relaxdtmf was not defined, but I defined it as yes. |
| 18:30.49 | pigpen2 | yes...ulaw. |
| 18:30.56 | ManxPower | don't define it to yes. doing that can cause the problem |
| 18:31.20 | pigpen2 | eitherway...ulaw vs. the issue is irrelivant...as I have it happen when I dial via a cell over the pri, to a voicemail box. |
| 18:31.24 | pigpen2 | Ok..I will turn it back off. |
| 18:31.33 | pigpen2 | or not define it I mean. |
| 18:31.54 | ManxPower | pigpen, your cell phone does not use ulaw or alaw |
| 18:32.28 | pigpen2 | not yet :) |
| 18:32.33 | TheoC | I'm trying to setup up call parking on my polycom 501 phone and I'm able to get Park to show up as a menu option while on a call - but I can't figure out how to make that work. Does anyone know how that works? Or is it possible to assign a button to mean "transfer to ext 70"? |
| 18:32.36 | ManxPower | it never will either. |
| 18:32.38 | pigpen2 | sorry , I got it backwards... |
| 18:32.41 | *** join/#asterisk nshm (n=shmyrev@217.67.124.2) |
| 18:32.46 | ManxPower | uses up far too much bandwidth on the cell network |
| 18:32.53 | pigpen2 | yeah... |
| 18:33.10 | pigpen2 | Ok..I will kill the relaxdtmf...and try again... |
| 18:33.32 | nshm | Hey all, sorry for being a bit offtopic, but does someone know open source SIP client that is able to work through Linux TAPI? |
| 18:33.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:33.58 | pigpen2 | ManxPower, can I just reload, or do I need to restart Asterisk? |
| 18:34.06 | justinu | restart to be safe |
| 18:34.06 | ManxPower | pigpen, no idea |
| 18:34.15 | justinu | some settings take affect on reload, some dont |
| 18:34.17 | pigpen2 | Finding time to restart Asterisk is a pain with active calls... |
| 18:34.53 | pigpen2 | Ok..I managed to find it with only 1 channel. |
| 18:34.56 | pigpen2 | in use that is... |
| 18:34.58 | pigpen2 | testing. |
| 18:34.58 | zoa | restart when convenient |
| 18:35.21 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 18:35.36 | *** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
| 18:37.24 | *** join/#asterisk saftsack (n=saftsack@p54A7E5F5.dip.t-dialin.net) |
| 18:38.25 | *** join/#asterisk laichzeit (n=User01@dsl-145-171-00.telkomadsl.co.za) |
| 18:39.21 | laichzeit | anyone have an idea why asterisk would destroy a call for no apparant reason? |
| 18:39.50 | inv_arp[work] | laichzeit: err no |
| 18:39.52 | justinu | that message is normal |
| 18:40.31 | laichzeit | only thing I pick up in the logs is: Scheduling destruction of call 'aab83132-b7b5-d911-8cc1-00c12602ed28 thales' in 15000 ms |
| 18:40.54 | ManxPower | laichzeit, many things are "calls". VM notify, options, registrations |
| 18:41.32 | laichzeit | ManxPower, well an outgoing call over pstn, one minute you're speaking, next minute its dead. |
| 18:42.03 | ManxPower | laichzeit, what type of phone device? What type of PSTN interface? |
| 18:42.18 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 18:42.21 | ManxPower | of course, using busydetect or callprogress would also cause that problem |
| 18:42.40 | laichzeit | busydetect, hmm.. ok. |
| 18:42.43 | GerbilNut | anyone in here configured DUNDi between two systems via IAX? |
| 18:43.08 | laichzeit | its a IP-300 phone |
| 18:43.35 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:43.47 | Zodiacal | qwell you around? |
| 18:43.52 | _Paulo_ | ~pb |
| 18:43.55 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 18:43.55 | pigpen2 | Ok..guys...no change with the dtmf issue.... |
| 18:44.07 | Zodiacal | qwell what firmware are you using with ip communicator? |
| 18:44.25 | Zodiacal | qwell and what device do you have it set to using? i.e. 79XX? |
| 18:44.27 | laichzeit | and TDM4000 FXS |
| 18:44.29 | ManxPower | pigpen, increase or decrease your gains |
| 18:44.38 | justinu | pigpen: what call flow are you having trouble with? |
| 18:44.38 | pigpen2 | which way.... |
| 18:44.47 | ManxPower | pigpen, no way to tell |
| 18:44.47 | justinu | navigating thru an IVR? |
| 18:44.53 | pigpen2 | k |
| 18:45.00 | ManxPower | I would start by lowering your rxgain |
| 18:45.04 | justinu | actually, since your meter looked pegged on ztmonitor... probably lower |
| 18:45.12 | justinu | perhaps 4 units at a time |
| 18:45.16 | *** join/#asterisk simulated (n=simulate@adsl-070-155-044-222.sip.bct.bellsouth.net) |
| 18:45.30 | pigpen2 | justinu, well, if I need to pass any dtmf tones through the pri, it chops the tone in half. |
| 18:45.44 | justinu | k |
| 18:45.47 | ManxPower | pigpen, define "chop in half" |
| 18:45.49 | simulated | is anyone else having some issues with the latest SVN ? I get an error during compilation telling me my LibPRI is out of date, and breaks on compiling zaptel |
| 18:45.53 | pigpen2 | No matter if it goes into the ivr, or out to a medical ditictation software. |
| 18:46.01 | ManxPower | simulated, did you update your libpri? |
| 18:46.08 | pigpen2 | ManxPower, "Bee___eep" |
| 18:46.09 | simulated | yeah, latest svn |
| 18:46.22 | pigpen2 | On a butset, it sees for "1" = 11 |
| 18:46.34 | pigpen2 | So it sees two digits for each number pressed/ |
| 18:46.37 | ManxPower | I had that also |
| 18:46.44 | simulated | libpri revision 320 |
| 18:46.53 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 18:47.00 | ManxPower | turned out my rxgain was too high and the received DTMF was echoing and causing double detection |
| 18:47.03 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:47.12 | ManxPower | simulated, wait a day or so and try again |
| 18:47.12 | pigpen2 | ah...drop to -4 then? |
| 18:47.24 | simulated | ManxPower hehe... interesting option :) |
| 18:47.26 | justinu | pigpen: go lower until your users can't hear to well |
| 18:47.31 | simulated | wish i could go with that hehe |
| 18:47.33 | justinu | then go back up just enough that the volume is ok |
| 18:47.38 | elg | incoming to asterisk from a sip ATA, then outgoing to a sip provider, if I'm having DTMF problems asterisk isn't really in the loop right? because of the native bridge? |
| 18:47.41 | pigpen2 | k |
| 18:47.44 | ManxPower | simulated, SVN IS the developement version, it WILL be broken sometimes. |
| 18:47.47 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 18:47.48 | caio1982 | has anyone here got "-- Unicall/1 protocol error. Cause 32773" some time? |
| 18:47.52 | ManxPower | if you don't like that, use the tarball releases |
| 18:47.59 | justinu | pigpen: units of 4 seems to work out |
| 18:48.29 | ManxPower | elg, asterisk is still in the SIP SIGNALING loop, just not the RTP AUDIO loop. |
| 18:48.58 | *** part/#asterisk nshm (n=shmyrev@217.67.124.2) |
| 18:49.02 | elg | right, but rfc8233 is part of the rtp loop right? |
| 18:49.09 | justinu | yes |
| 18:49.12 | justinu | 2833 |
| 18:49.16 | ManxPower | I don't know. |
| 18:49.20 | elg | right, oops |
| 18:49.28 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 18:49.30 | justinu | RFC2833 payload is carried as part of the RTP stream |
| 18:49.37 | elg | ok, thanks |
| 18:49.57 | mroth_imm | anyone have any experience on fedora core 3 and irqbalance |
| 18:49.59 | pigpen2 | Well, I dropped it to -3.0...and the pause in the middle is less.... |
| 18:50.01 | ManxPower | Bell has totally botched a number port. |
| 18:50.44 | simulated | It's not that I dont like it hehe... ive always ran cvs/svn as i could remember, never had major compilation issues such as this... but ill give the tarballs a shot |
| 18:50.53 | TheoC | Does anyone know if it's possible to program a polycom key to simulate a series of key presses (ie '#70')?? |
| 18:51.20 | ManxPower | TheoC, you did not read about that when you read the ADMIN manual? |
| 18:52.49 | TheoC | Well, I see that you can program a key to act as any other key (change the 2 to be 5 or 0 to be redial or whatever) but can you set a button to act as #key then 7key then 0key? |
| 18:53.09 | ManxPower | TheoC, look at speed dials in the manual |
| 18:53.28 | ManxPower | which may be part of the Directory, I don't recall |
| 18:54.34 | *** join/#asterisk batphone (n=will@69.15.174.114) |
| 18:54.34 | TheoC | ok |
| 18:54.37 | batphone | http://pastebin.com/629340 |
| 18:54.43 | batphone | if anyone has a minute to check this out |
| 18:55.11 | simulated | hehe batphone youre here all the time :) |
| 18:55.17 | justinu | pigpen2: sounds good, keep going |
| 18:55.31 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 18:55.32 | batphone | what can i say, its a way of life for me these days |
| 18:56.01 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 18:57.32 | x86 | anyone using ENUM? |
| 18:57.42 | ManxPower | x86, yes |
| 18:57.45 | ManxPower | or was at least |
| 18:57.56 | x86 | ManxPower: can you try calling me using it/ |
| 18:58.06 | ManxPower | x86, you are not in my private ENUM system |
| 18:58.22 | x86 | you dont do lookups to a public E164 server? |
| 18:58.33 | ManxPower | x86, hwll no. |
| 18:58.35 | ManxPower | hell no. |
| 18:58.38 | x86 | why? |
| 18:58.44 | ManxPower | why would I want to do that? |
| 18:58.50 | x86 | uh |
| 18:58.52 | simulated | x86 you got access to it? |
| 18:58.56 | simulated | hehe |
| 18:58.57 | x86 | to save costs on outbound calls ;) |
| 18:59.02 | ManxPower | and route my calls over someone's horrid little crappy internet connection |
| 18:59.15 | simulated | x86 you needa use SS7 ;) |
| 18:59.22 | ManxPower | x86, Saving less than $10/month just isn't worth it. |
| 18:59.27 | x86 | ManxPower: well, if that's where it's going anyway... ;) |
| 18:59.39 | ManxPower | and if calls are not perfect, my users scream |
| 19:00.00 | ManxPower | x86, You do realize that I do almost no VoInternet, right? |
| 19:00.06 | ManxPower | Almost all my calls are VoWAN/LAN |
| 19:00.11 | x86 | ah |
| 19:00.21 | ManxPower | internet is too unreliable. |
| 19:00.21 | x86 | so why are you doing ENUM at all? |
| 19:00.42 | ManxPower | x86, so I don't have to update the dialplan on all the servers I admin every time I add another server. |
| 19:00.56 | x86 | ah, good idea :) |
| 19:01.04 | batphone | >( |
| 19:01.04 | ManxPower | once all my servers are running 1.2 I'll try out setting up a private DINDi thing |
| 19:01.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:01.39 | simulated | dundi :) |
| 19:02.22 | ManxPower | but really, we are routing fewer and fewer of our calls over or own private WAN anyway. Just not reliable enough. |
| 19:03.47 | ManxPower | we have one office that has to reboot their Asterisk server once a week or it stops working. |
| 19:04.07 | simulated | ManxPower: ok it was my mistake, i entered /usr/src/asterisk instead of the 1.2 directory... So dont lose your faith in SVN :) |
| 19:04.24 | *** part/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 19:04.38 | simulated | ManxPower: believable, happens once in a while... they must be running like A@H or something |
| 19:04.39 | ManxPower | simulated, I lost my faith in the developement version a long time ago. |
| 19:04.49 | ManxPower | simulated, heck no. |
| 19:04.51 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 19:04.52 | simulated | :) Ive always had good success with it |
| 19:04.53 | ManxPower | TDM400P cards |
| 19:04.57 | simulated | the svn |
| 19:05.11 | simulated | hrmm... they run on a generic clone PC ? |
| 19:05.13 | ManxPower | simulated, I have plenty of stress and work in my life, I don't need more. |
| 19:05.20 | ManxPower | simulated, intel reference board. |
| 19:05.36 | simulated | but was a brand new box built from scratch or recycled? |
| 19:05.48 | ManxPower | pretty much any server we put a TDM400P into has to be rebooted regularly |
| 19:06.01 | ManxPower | simulated, we don't build corporate phone systems from recycled parts. |
| 19:06.26 | eric_hill | ManxPower: Sounds more like a power supply issue to me. Have you tried a power conditioner? We've had REALLY good luck with those on everything from servers to copiers... |
| 19:06.46 | ManxPower | eric_hill, the UPSs should be conditioning the power. |
| 19:07.04 | eric_hill | No, they don't. |
| 19:07.32 | eric_hill | Most UPS boxes will brownout before cutting over to battery, unless you're buying REALLY expensive active/active UPS systems. |
| 19:07.38 | *** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:07.38 | *** join/#asterisk Assid (n=assid@59.183.60.214) |
| 19:07.41 | Assid | back |
| 19:07.45 | Assid | <PROTECTED> |
| 19:07.51 | eric_hill | Check out the APC line of Voltage Regulators: http://www.apcc.com/products/category.cfm?id=12&subid=57 |
| 19:07.53 | ManxPower | eric_hill, at least some of our UPSs should be active. |
| 19:08.03 | ManxPower | Hell, the cisco switchs alone take 15amps |
| 19:08.04 | Assid | i get it till there.. but it doesnt wanna authenticate my pass |
| 19:08.23 | ManxPower | eric_hill, but I'll put it on the list of things to try. We basically stopped buying tdm400Ps |
| 19:08.29 | eric_hill | That's what I thought until I started looking at low-level specs. APC doesn't do active/active until you get into their stackable summetra series. |
| 19:08.33 | simulated | what card did you switch over to? |
| 19:08.46 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 19:08.50 | simulated | yeah, symmetra is expensive as hell too |
| 19:09.06 | ManxPower | simulated, TExxP w/adtran channel bank if we need analog, but we no longer install analog phones or analog lines. |
| 19:09.32 | ManxPower | we are also starting to install tellabs echo cancelers instead of using the Digium ones. |
| 19:09.47 | simulated | interesting |
| 19:09.49 | eric_hill | Yea, we went with the off the shelf SmartUPS (1500-ish) and a LineR 1200VA regulator behind it. Switching on and off AC shows -zero- changes in voltage to the computer. |
| 19:10.20 | ManxPower | granted, I don't mess with tiny systems anymore. |
| 19:10.36 | oej | ManxPower: Ping |
| 19:10.43 | *** join/#asterisk |cleric| (n=dacleric@p5482BBCA.dip0.t-ipconnect.de) |
| 19:14.10 | *** join/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
| 19:16.13 | Assid | stupid disa.. doesnt wanna work |
| 19:16.44 | Assid | i get upto executing disa in new stack.. but dont get a fresh dialtone |
| 19:17.35 | *** join/#asterisk chr|s_ (n=chris@217.171.52.76) |
| 19:19.02 | ManxPower | Assid, using a SIP phone? |
| 19:19.14 | Assid | yep |
| 19:19.40 | ManxPower | did you try an Answer before the DISA. You should not need it, but it can't hurt to try it. |
| 19:19.40 | *** join/#asterisk DrData (n=michael@p54B244D6.dip.t-dialin.net) |
| 19:20.07 | Assid | well.. i get upto -- Executing DISA("SIP/3001-ad81", "no-password|default") in new stack |
| 19:23.08 | TheoC | Does anyone know how to use the parking feature on the polycom 501? |
| 19:23.16 | FarrisG | Can anyone recommend a good door phone? |
| 19:24.06 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-180-235.austin.res.rr.com) |
| 19:24.34 | ManxPower | Assid, do you have a [default] context in extensions.conf? |
| 19:25.36 | ManxPower | back in the days of 0.65 I could never get DISA to work with SIP. |
| 19:25.51 | ManxPower | Eventually I learned that I have never really needed DISA |
| 19:29.27 | CoffeeIV_ | is there an UnSet() dialplan function to match Set() ? |
| 19:30.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 19:31.41 | fnordian | Set(VAR=) |
| 19:32.31 | *** join/#asterisk essaredee (i=srd@dhcp37.frictious.net) |
| 19:33.48 | essaredee | from reading the voip-info wiki it looks like fax support is shoddy, I was wondering if sending fax from FXO to FXS would be workable |
| 19:34.54 | essaredee | on the same machine |
| 19:36.51 | CoffeeIV_ | You mean you want to receive your faxes on a POTS line, and auto-detect that they are faxes and then direct the call at an extension that has a real fax machine on it ? That works |
| 19:37.02 | Assid | ManxPower: i was just playing with default.. i had it at the context i needed.. but didnt work |
| 19:38.09 | *** join/#asterisk citats (n=james@69.54.200.117) |
| 19:38.39 | essaredee | sort of. I'd have a dedicated fax line, even tho it's not really nessecary, have the line routed through asterisk |
| 19:39.52 | CoffeeIV_ | essaredee: that will work. In my experience asterisk will reliably receive faxes itself, also, and deliver them by email -- example dialplans are available that do that on voip-info and in asterisk@home |
| 19:40.29 | essaredee | nod, I want to use a real fax machine, I just don't want it directly hooked up to the POTS :) |
| 19:40.34 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:41.37 | essaredee | plus I figure it'd also be an added benefit of having the line available to one of the phones if all other lines are engaged, etc |
| 19:42.41 | essaredee | am trying to setup a decent dial plan to determine whether to go through pots or through an IAX/SIP provider. say it's betwen 6pm and 6am go through pots if it's a non-intl call, during the day go through the IAX |
| 19:43.11 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 19:43.35 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 19:45.09 | *** join/#asterisk backblue (n=moo@87-196-42-46.net.novis.pt) |
| 19:45.39 | *** join/#asterisk Cadu20 (n=Cadu20@200.102.53.174) |
| 19:48.27 | GerbilNut | gotoiftime |
| 19:49.53 | essaredee | I've been using contexts for that mostly |
| 19:50.32 | essaredee | like, I set up a context so if someone phones between 9pm and 8am the next day it rings for a few minutes and sends them to voicemail |
| 19:50.45 | essaredee | doesn't send it to the phone unless they know the secret code |
| 19:50.59 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 19:51.06 | nestAr | yello' |
| 19:51.11 | essaredee | hi |
| 19:52.03 | nestAr | I was reading the Queue page on voip-info, and it makes mention of being able to allow single digit extensions to be used by people on hold in a Queue, but it doesn't give any info about configuring such options |
| 19:52.04 | pigpen2 | ManxPower, I had to drop my rx/tx gain down to -8 and all is well....thanks.! |
| 19:52.08 | nestAr | anyone know how that works? |
| 19:57.40 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 19:57.50 | FuriousGeorge | my iax peers become unreachable for no apparent reason. the only wait to get them back seems to be to restart asterisk. i posted on the mailing list but it seems no one knows. http://pastebin.ca/47498 |
| 19:58.15 | *** join/#asterisk [Airwolf] (n=airwolf@groeneboord.xs4all.nl) |
| 20:00.13 | nestAr | nevermind, i'm a moron who can't read examples. |
| 20:01.29 | justinu | will this work: PRI into asterisk, sipura ATA connected to fax machine, sipura connected to Asterisk via local ethernet. sipura running g711... will faxes be reliable? |
| 20:02.16 | SpaceBass | justinu dont know about the pri part...but the rest should work |
| 20:02.38 | backblue | justinu: yes, if you use ulaw or alaw only |
| 20:02.47 | justinu | yeah, there's no reason to use anything else in this config |
| 20:02.48 | backblue | pri bri whatever |
| 20:02.53 | backblue | just dial it out |
| 20:03.28 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 20:04.18 | *** join/#asterisk silasmarner (n=silas@dharma.summersault.com) |
| 20:06.21 | Juggie | greetings |
| 20:07.48 | silasmarner | Anyone up for trying to help solve a mystery with our newly-launched Asterisk instance? |
| 20:08.07 | Katty | hey Juggie (= |
| 20:08.23 | *** part/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 20:08.35 | backblue | silasmarner: what? |
| 20:08.51 | Katty | :> |
| 20:09.04 | silasmarner | 1.2.4 on FreeBSD. After a seemingly arbitrary period of time, the system will still answer calls, but will not respond to DTMF tones in the attendant. |
| 20:09.10 | silasmarner | Restarting the server fixes it. |
| 20:09.16 | `Sauron | Damn my friend for getting me to try EVE Online |
| 20:09.23 | `Sauron | I was up until 3:30 last night |
| 20:09.35 | *** join/#asterisk darkskiez (n=darkskie@194.164.233.141) |
| 20:09.35 | CoffeeIV_ | if I'm inside a Wait() in the dialplan, and some presses keys, will it "break out" of the wait and go to the right extension ? |
| 20:09.40 | Guggemand | `Sauron quit while you still can |
| 20:09.55 | `Sauron | iDunno |
| 20:10.24 | Katty | punner. |
| 20:10.39 | docelm0 | MEW MEW MEEEEEEEEEEEEEEEWWWWWWWWWWWWWWWWWWWWWWWWWWWWWWW |
| 20:10.53 | tzanger | you sound like my cat Jake |
| 20:11.08 | docelm0 | haha |
| 20:11.08 | silasmarner | CoffeeIV: I believe Wait will not break on tones; you probably want WaitExten |
| 20:11.09 | Abydos313 | the last one reminds more of a cow |
| 20:11.14 | docelm0 | Just trying to be like Katty |
| 20:11.19 | `Sauron | cats mew |
| 20:11.21 | `Sauron | cows moo |
| 20:11.23 | *** join/#asterisk saftsack (n=saftsack@p54A7E5F5.dip.t-dialin.net) |
| 20:11.35 | Katty | docelm0: yeah but i'm not /that/ obnoxious. |
| 20:11.39 | Abydos313 | cat meow is what i thought and cows moo |
| 20:11.41 | silasmarner | backblue: any thoughts? |
| 20:11.45 | docelm0 | Your female.. Need I say more? |
| 20:11.47 | tzanger | yeah and some cats just MEEERRROWWWWWWWRRRROOOOWWRRRREEWWWWRREEOOOOOWWWWRRRR |
| 20:11.54 | tzanger | those are the kind that usually get a shoe thrown at them |
| 20:11.54 | CoffeeIV_ | silasmarner: that's exactly what I want, thanks a ton |
| 20:11.58 | docelm0 | `Sauron, dude.. did I type moooooooooooooooooooooooo? No.. |
| 20:12.11 | `Sauron | 14:11 <Abydos313> the last one reminds more of a cow |
| 20:12.13 | `Sauron | 14:11 <`Sauron> cows moo |
| 20:12.21 | `Sauron | codelm0: Eat shit. |
| 20:12.22 | Abydos313 | same here |
| 20:12.34 | docelm0 | ahh my bad.. |
| 20:12.37 | Abydos313 | i figured it was just mispelled mew/moo |
| 20:12.40 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 20:12.50 | backblue | silasmarner: i know a very easy solution, rm -rf / && install linux |
| 20:12.58 | backblue | why do you need freebsd? |
| 20:13.08 | Abydos313 | backblue you talking about the windows virus? |
| 20:13.12 | silasmarner | Thanks, that won't work for me, but I appreciate your suggestion. |
| 20:13.16 | silasmarner | is this the best channel for discussing asterisk questions/issues, or is it just general chat? |
| 20:13.58 | backblue | silasmarner: asterisk have better suport in linux, so if you want to use with no kind of that problems, use linux. |
| 20:14.17 | backblue | i dont see why do you need freebsd, and performance its better in linux. |
| 20:14.27 | backblue | sorry i cant help you in freebsd issues |
| 20:14.53 | silasmarner | That's fine, but it may not be a FreeBSD issue. Everything else is working fine, so it would be a shame to start over. |
| 20:15.49 | backblue | silasmarner: i have 1.2.4 in linux, and everything work fine. |
| 20:17.51 | TheoC | does anyone use polycom phones and call parking? |
| 20:17.53 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 20:19.33 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-83-26.dsl.chcgil.sbcglobal.net) |
| 20:21.50 | [av]bani | how does one do a blind xfer with polycom 601? |
| 20:25.42 | Flauto | would anyone here tell me how to setup asterisk exten for sipura spa 3000 to pass throgh pstn call to asterisk? |
| 20:26.27 | TheoC | [av]bani: if you're on a call and push the transfer key - there's a blind key - push that and then enter the number to transfer to |
| 20:26.52 | *** join/#asterisk Pryk (n=tmalkut@host-ip2-24.crowley.pl) |
| 20:29.39 | [TK]D-Fender | [av]bani : Press [Transfer] and then the right soft key becomes [blind] press that before beginning your target |
| 20:30.32 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 20:31.45 | [av]bani | any way to make transfers blind by default? |
| 20:31.56 | [av]bani | cisco, grandstream do blind by default |
| 20:32.01 | [av]bani | only polycom seems to default to attended |
| 20:33.16 | *** join/#asterisk linlin (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 20:34.17 | linlin | whats a cheap provider of incoming toll free numbers to my pbx |
| 20:34.49 | tzanger | so people have been saying those ustarcomm wifi phones aren't too shabby |
| 20:34.51 | *** join/#asterisk _DAW (n=bob@adsl-150-58-20.msy.bellsouth.net) |
| 20:34.56 | *** part/#asterisk Chopinhauer (n=Chopinha@morgoth.karwasz.org) |
| 20:36.15 | GerbilNut | tzanger, i heard the Hitachi-Cable WirelessIP-5000 were good |
| 20:36.22 | *** join/#asterisk darkskiez (n=darkskie@194.164.233.141) |
| 20:36.32 | *** join/#asterisk denon (i=root@synapse.subneural.net) |
| 20:36.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:39.01 | tzanger | we're looking for bluetooth on the wifi phones but haven't found one with that yet |
| 20:39.55 | GerbilNut | yeah, good luck with that |
| 20:39.59 | tzanger | heh |
| 20:40.53 | GerbilNut | I can get you t he Hitachi at a decent price though if they decide to not wait for the bluetooth |
| 20:40.56 | *** join/#asterisk darby_t (i=darby_t@dle159.neoplus.adsl.tpnet.pl) |
| 20:41.06 | tzanger | got a link on the hitachi ones? |
| 20:42.16 | GerbilNut | http://www2.hitachi-cable.co.jp/apps/hnews.nsf/0/6dace07217a4b20049256e7a00828406?OpenDocument |
| 20:42.20 | Qwell[] | bluetooth on a wifi phone? |
| 20:42.32 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 20:42.33 | Qwell[] | Sounds like a buzzword waiting to happen |
| 20:42.53 | hypa7ia | web 2.0 enabled, buzzword compliant |
| 20:42.57 | justinu | i can tell you the zyxel sucks |
| 20:43.13 | *** join/#asterisk Maxxed (n=root@cpe-72-177-150-20.houston.res.rr.com) |
| 20:43.19 | Maxxed | oi' :) |
| 20:43.24 | Qwell[] | Nugget: You need to converge it...that's the only way you can leverage the utilization factor |
| 20:43.40 | Maxxed | hey can sombody point me in the right direction inregards to intercepting dtmf tones during a call |
| 20:43.42 | *** join/#asterisk backblue (n=moo@87-196-33-13.net.novis.pt) |
| 20:43.50 | Strom_C | Is IAX2 fixed in SVN Trunk? |
| 20:44.26 | Maxxed | the idea in whole is, while a call is in progress, the user can press say.. the pound key, and then anothe phone will ring and beable to listen in |
| 20:44.42 | Maxxed | not sure exactly how to pick up those tones during a call |
| 20:44.43 | *** join/#asterisk _deg_ (n=deg@200.250.222.8) |
| 20:44.50 | justinu | qwell forgot to shift the paradigm outisde the box first |
| 20:45.29 | Qwell[] | justinu: I heard a few really good ones yesterday...I don't recall what they were though |
| 20:45.30 | Maxxed | a reverse zapbarge i guess sorta kinda |
| 20:45.31 | Maxxed | heh |
| 20:45.59 | justinu | Qwell[]: mesmerizing, isn't it? |
| 20:46.12 | tzanger | GerbilNut: thanks for the link... what's pricing like? |
| 20:46.24 | GerbilNut | looking at about $300 plus shipping each |
| 20:46.33 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 20:46.41 | GerbilNut | Retail on them is about $355 |
| 20:46.44 | blitzrage | its lump, its lump, its in my head |
| 20:47.37 | Maxxed | anybody? |
| 20:47.49 | Maxxed | not even sure were to look ;\ |
| 20:48.01 | caio1982 | _Paulo_: hey |
| 20:48.23 | Maxxed | maing |
| 20:48.26 | Maxxed | heh |
| 20:50.41 | caio1982 | _Paulo_: you there? i want to make sure you're using my .deb packages for unicall/mfcr2 and want to know if they're working fine... i'm testing them, and getting lots of signalling problems with an E1 |
| 20:52.11 | Maxxed | well.. in what direction? |
| 20:52.22 | Maxxed | maybe twards the dtmf tone detection during a call? |
| 20:52.25 | iDunno | FORWARDS! |
| 20:52.31 | iDunno | always go forwards... |
| 20:52.32 | Maxxed | and not into the spiky pit of doom |
| 20:52.39 | Maxxed | what about sideways |
| 20:52.46 | Maxxed | or up |
| 20:52.49 | Maxxed | shuv up |
| 20:52.50 | iDunno | going backwards can lead you in to all sorts of trouble ;) |
| 20:52.50 | Maxxed | heh |
| 20:53.03 | Maxxed | um, like running backwards thru a cord feild naked? |
| 20:53.09 | Maxxed | corn* |
| 20:53.29 | Maxxed | nothing but trouble there |
| 20:53.29 | Maxxed | belive me! |
| 20:53.40 | Strom_C | why run backwards? you'll vomit |
| 20:53.51 | Qwell[] | heh |
| 20:53.54 | *** join/#asterisk Assid (n=assid@59.183.5.147) |
| 20:53.56 | GerbilNut | tzanger, message me if they appear interested in the phones |
| 20:54.00 | Assid | umm |
| 20:54.00 | [av]bani | ... |
| 20:54.02 | Assid | this is freaky |
| 20:54.06 | Maxxed | n'deed |
| 20:54.10 | Maxxed | so the dtmf thing! |
| 20:54.14 | Qwell[] | bonus points for Strom_C, for describing 25 pair cable :p |
| 20:54.14 | Assid | it doesnt wait for the person to type in the pin.. just straight to not valid |
| 20:54.21 | Maxxed | anybody have any idea how to detect tones during a call? |
| 20:54.28 | Qwell[] | Strom_C: now, is that tip, or ring? |
| 20:54.53 | Strom_C | that would be the ring colors |
| 20:54.55 | Strom_C | er |
| 20:54.59 | Strom_C | tip colors |
| 20:55.14 | Maxxed | red right ring |
| 20:55.18 | tzanger | GerbilNut: well what kind of price do they carry? Just be interested in a couple for testing |
| 20:55.22 | Maxxed | tip green left |
| 20:55.25 | _DAW | Hello. I am having a problem with hdlc on a TE110P. WARNING: Error inserting zaptel (/lib/modules/2.6.9-34.EL/extra/zaptel.ko): Unkn |
| 20:55.25 | _DAW | own symbol in module, or unknown parameter (see dmesg) |
| 20:56.15 | GerbilNut | tzanger, they retail for $355, i'd be willing to do $315-300 plus shipping, and if you order in quantity, even lower |
| 20:56.23 | _DAW | it only happens when I uncomment the line #define CONFIG_ZAPATA_NET |
| 20:56.56 | _DAW | in zconfig.h that is |
| 20:57.38 | tzanger | GerbilNut: ok, I'll catch you on here or email me actually akohlsmith@benshaw.com so I can contact you if I can get the powers that be to try 'em |
| 20:57.47 | [av]bani | wifi phones? ew |
| 20:58.08 | Strom_C | Maxxed, BRGY is so...1960s |
| 20:58.23 | GerbilNut | in my city wifi phones have a huge market, because the city is in the midst of finishing up a city wide mesh wifi network |
| 20:58.36 | Maxxed | Strom_C: hah! true, true ;p |
| 21:00.48 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 21:02.50 | r_evolution | hey maxxed... whats your dtmf issue? i meant to ask you but people seem to enjoy talking to me |
| 21:03.04 | GerbilNut | tzanger, e-mail sent |
| 21:03.10 | [av]bani | how to make polycoms default to blind xfer? |
| 21:03.21 | r_evolution | detecting the tones? shouldnt asterisk do that automatically? |
| 21:03.33 | r_evolution | or am I reading the question differently than you're asking it? |
| 21:03.36 | ghotiboy1 | anyone here use AsterFax? |
| 21:03.43 | *** join/#asterisk Mauro__ (n=mauro@oliver.altascumbres.cl) |
| 21:03.47 | Mauro__ | Hi |
| 21:04.09 | tzanger | GerbilNut: perfect, thanks |
| 21:05.57 | *** part/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 21:07.43 | Maxxed | well heres the idea |
| 21:07.58 | Maxxed | il keep it simple as to not compl..bah.. ok here we go |
| 21:08.13 | Maxxed | for example! if i am talking to someone |
| 21:08.20 | Maxxed | a call in progress |
| 21:08.39 | Maxxed | i would like to beable to hit a button or two on the phone and have asterisk record the call |
| 21:09.16 | Maxxed | now, i can handle the record thing (i think) |
| 21:09.26 | Maxxed | its just picking up that tone as the call is going on |
| 21:09.30 | Assid | do polycoms have any dtmf issues? |
| 21:09.39 | Maxxed | is there some dtfmpickup_cmd or somethin |
| 21:09.55 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 21:09.59 | Maxxed | assid: i use cisco stuff, *shrugs* |
| 21:11.18 | Assid | hrmm |
| 21:12.18 | jaike | Assid: none that i know off..we use 301s |
| 21:12.31 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 21:12.48 | qseek | hi all |
| 21:13.25 | *** join/#asterisk doolph (n=doolph@201.227.72.230) |
| 21:13.32 | doolph | anyone can helpme with sipura 3000? |
| 21:13.43 | doolph | how can I forward my pstn to an extension? |
| 21:15.20 | mjmac | does someone from atacomm hang out in here? |
| 21:16.31 | *** join/#asterisk ToTo (n=ToTo@host123-121.pool8258.interbusiness.it) |
| 21:16.52 | mjmac | just wondering if i really can't buy a bare tdm400p... i have one that has gone wonky. pretty sure it's the card itself, as opposed to the modules, since i have the problem (garbled sound) on both fxs and fxo interfaces |
| 21:16.55 | jaike | Maxxed: Dial with w or W |
| 21:17.08 | Maxxed | dial with w ? |
| 21:17.29 | jaike | to start recording in the middle of a call |
| 21:17.34 | modulus_ | werd |
| 21:17.43 | Maxxed | i dont follow jaike? |
| 21:17.46 | Maxxed | i mean thats the idea |
| 21:17.51 | Maxxed | but dial with W ? |
| 21:18.07 | jaike | options |
| 21:18.09 | mog_work | mjmac, you can probably rma the base board |
| 21:18.15 | jaike | w - Allow the called party to enable recording of the call by sending |
| 21:18.16 | doolph | anyknow know how to forward my pstn line to an asterisk extension? |
| 21:18.20 | jaike | the DTMF sequence defined for one-touch recording in features.conf. |
| 21:18.46 | mjmac | btw. has anyone else had this problem with a tdm400p? i have a fairly old rev (don't know which offhand, bought it in 2004). worked fine for a long time. |
| 21:18.47 | Maxxed | jaike: that sounds what im lookin for |
| 21:18.52 | mjmac | mog_work: maybe |
| 21:18.56 | jaike | W - calling party |
| 21:19.00 | mog_work | thats what i would do mjmac |
| 21:19.45 | mjmac | think they'd still take it after so long? i don't have a support contract or anything... i guess i should try. i'm using an ATA as a temp. stand-in, but i lost my fax line. :/ |
| 21:21.25 | mog_work | digium has 2 year warranty on all hw |
| 21:27.47 | *** join/#asterisk MacDome (n=eseidel@A17-255-97-70.apple.com) |
| 21:29.06 | *** join/#asterisk MacDome (n=eseidel@A17-255-97-70.apple.com) |
| 21:29.29 | *** join/#asterisk trelane_ (n=trelane@mail.allthingsit.com) |
| 21:29.50 | trelane_ | when a call goes into congestion is there a way to hold onto the call long enough to hangup on a zap channel and retry the call? |
| 21:30.15 | trelane_ | iow does it jump to n+101? |
| 21:31.24 | bkw_ | don't bother saying Hi now |
| 21:31.29 | bkw_ | just bust in and ask your question.... |
| 21:31.58 | bkw_ | you can going on in the dialplan.. or jump and retry the call again |
| 21:32.01 | *** join/#asterisk ComputerWarm (n=dan@HS196-230-97.nt.net) |
| 21:32.03 | blitzrage | HI! |
| 21:32.09 | ComputerWarm | Hello question how do i send a sms message with asterisk? |
| 21:32.17 | bkw_ | ComputerWarm, You don't |
| 21:32.26 | tzanger | bkw_: hey stranger |
| 21:32.26 | ComputerWarm | I thought Asterisk could handle that? |
| 21:32.33 | bkw_ | no it can do fixed line SMS |
| 21:32.46 | Splatty47 | I have a weird problem - I have just managed to connect two Snom 360 phones to asterisk as extension 300 and 301. But when I try to call one from the other - it tells me the number is not inthe speed dial system! any ideas ? |
| 21:32.49 | ComputerWarm | oh it can`t do a voip like sms |
| 21:32.52 | bkw_ | its not the same as cellular SMS |
| 21:33.10 | GerbilNut | trelane, i believe Congestion will go to n=101 |
| 21:33.11 | ComputerWarm | oh is there anything that i can get that can do cellular sms? |
| 21:33.13 | GerbilNut | n+101 that is |
| 21:33.20 | Nugget | a cellular phone. |
| 21:33.35 | ComputerWarm | thats the only way |
| 21:33.36 | tzanger | I was reading a little about land-line SMS with * on the mailing list |
| 21:33.39 | tzanger | didn't get anywhere just yet |
| 21:34.09 | Katty | i prefer saying hi, and forgetting to mention my questions ;) |
| 21:34.33 | *** join/#asterisk MacDome (n=eseidel@A17-255-97-70.apple.com) |
| 21:34.35 | qseek | well computerwarm u can do landline sms which would commuincate to a cellular network |
| 21:34.43 | qseek | but u would need to interface with a sms gateway |
| 21:35.00 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 21:35.37 | *** join/#asterisk MacDome (n=eseidel@A17-255-97-70.apple.com) |
| 21:36.05 | *** join/#asterisk scrambray8927 (n=scrambra@12.104.121.147) |
| 21:36.57 | ComputerWarm | qseek ok so i couldn`t just send it from asterisk to a voip long distance provider? |
| 21:37.00 | ComputerWarm | that wouldn`t work |
| 21:37.21 | qseek | no that would not work |
| 21:37.32 | qseek | u would need an application server which handles sms |
| 21:37.54 | ComputerWarm | so you know where i can get more information on sms messaging and what all i need? |
| 21:38.23 | tzanger | ComputerWarm: the wiki has some stuff but to be honest I got lost in it all |
| 21:38.27 | tzanger | I'm not very familliar with all of it |
| 21:38.36 | tzanger | I want Telus Mobility's SMSC number but nobody seems to know it |
| 21:41.51 | ComputerWarm | well if i figure it out tzanger i will let you know |
| 21:42.20 | ComputerWarm | oh if anyone need usa / canada termination via sip let me know |
| 21:42.26 | ComputerWarm | I found a good provider |
| 21:42.30 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 21:42.54 | inv_arp[work] | ComputerWarm: who? |
| 21:43.07 | ComputerWarm | AirStar Communications Network |
| 21:43.13 | inv_arp[work] | site? |
| 21:43.28 | ComputerWarm | they really don`t talk about it there but www.airstarcommunications.com |
| 21:45.22 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:46.04 | octothorpe | so . . . what makes them "good". Rates? Call Quality? |
| 21:46.11 | ComputerWarm | both |
| 21:46.40 | octothorpe | example rates (origination / termination) cust for did, etc . . . |
| 21:46.57 | ComputerWarm | all they do is termination As far as i know |
| 21:46.58 | octothorpe | *cost |
| 21:47.07 | ComputerWarm | here contact support@airstarcommunications.com on msn |
| 21:47.26 | ComputerWarm | he will help you more i am not interested in being a sales person lol |
| 21:48.58 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 21:54.43 | qseek | computerwarm : i was reading up on it too and only found these guys Bayham Systems |
| 21:54.43 | *** join/#asterisk Dovid (n=Dovid@89-138-76-126.bb.netvision.net.il) |
| 21:54.52 | scrambray8927 | Anyone have experience with VoIP Termination services (aka "raw" VoIP connectivity) with a major company such as AT&T(SBC), Sprint, Verizon, or even a Cable company such as Comcast? |
| 21:55.06 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:56.41 | *** join/#asterisk jijgeh (i=jijgeh@0-2pool130-217.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 21:56.43 | *** join/#asterisk Crontibs (n=frankie@ool-43525f0d.dyn.optonline.net) |
| 21:57.41 | jijgeh | anyone here running asterisk in a production environment? |
| 21:57.51 | Dovid | lots of us are |
| 21:58.21 | jijgeh | I want to build an Asterisk server to support approx. 100 phones in a small office... I need to know what sort of system requirements I am looking at |
| 21:58.31 | X-Rob | 100 phones is _not_ a 'small office' |
| 21:58.35 | jijgeh | ok |
| 21:58.57 | jijgeh | then what sort of system am I looking at for about that many? |
| 21:59.06 | Dovid | jijgeh: what are you looking to do ? |
| 21:59.28 | jijgeh | I just want to have mostly internal calling with about 8 lines out |
| 21:59.39 | jaike | jijgeh: load will most likely depend on the max number of simultaneous calls |
| 22:00.52 | Dovid | what codecs etc. |
| 22:00.52 | ManxPower | tzanger, try this +15149931123 |
| 22:00.52 | jijgeh | if most of the phones are VOIP, what sort of load do you think approximately 25 calls at a given time would entail |
| 22:00.52 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 22:00.52 | jijgeh | SIP |
| 22:00.52 | jijgeh | with GSM |
| 22:00.52 | X-Rob | RoyK, much better than your usual picine abuse, thanks. |
| 22:00.52 | jaike | recording? |
| 22:00.52 | jijgeh | no recording |
| 22:01.06 | Dovid | confrencing ? |
| 22:01.14 | ManxPower | tzanger, or the phone should have the SMSC number programmed in it somewhere under message settings |
| 22:01.14 | jaike | simple PC will do...1GB ram |
| 22:01.20 | jijgeh | conferencing would be good, but not very often |
| 22:01.53 | jaike | as long as u dont use g729..or recording (soxmix+shell eats up a lot of system resources) |
| 22:02.15 | jijgeh | what sort of processor? 2GHz machine work? |
| 22:02.27 | jaike | that should do |
| 22:02.35 | jaike | pure * server? |
| 22:02.37 | jijgeh | what impact would this sort of implementation have on the network |
| 22:02.38 | jijgeh | yes |
| 22:02.53 | jaike | 64k per call x 20..1.2mbps max |
| 22:03.23 | scrambray8927 | just looking for an estimate - how many outbound calls could an asterisk box with a 3ghz processor, 512MB ram on a Cable modem handle at once? |
| 22:03.46 | angom_w | jijgeh: http://www.packetizer.com/voip/diagnostics/bandcalc.html |
| 22:04.02 | jaike | 3ghz with only 512mb ram? |
| 22:04.03 | Strom_C | scrambray8927, what speed connection? |
| 22:04.17 | scrambray8927 | jaike yessir |
| 22:04.40 | jaike | will depend on your cable bandwidth |
| 22:05.09 | X-Rob | scrambray8927, your limitation is your internet connection |
| 22:05.21 | jaike | a call usually takes up 87kbps, 64kbps + tcp overhead |
| 22:05.22 | scrambray8927 | Strom_C 1.5Mb down 1.5Mb up |
| 22:05.26 | X-Rob | anything faster than a piii 500 is going to be a limitation of your internet connection |
| 22:06.14 | jaike | safe to compute at around 1.2mb only..at 1.5, your latency will be a factor |
| 22:06.21 | scrambray8927 | thanks |
| 22:07.06 | jijgeh | thanks for the input! |
| 22:08.19 | FarrisG | if you make changes in zapata.conf, do you have to restart or just reload? |
| 22:08.28 | ManxPower | http://www.globedotnet.ch/products/sms_en.asp |
| 22:08.42 | *** part/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 22:08.44 | ManxPower | FarrisG, that depends on the asterisk version and what you changed. |
| 22:09.20 | X-Rob | changed your zap? Reboot the machine! In fact, switch it off, pull out the CPU and replace it - EACH TIME YOU CHANGE zapata.conf! |
| 22:09.32 | ManxPower | YEAH! |
| 22:09.35 | FarrisG | ManxPower: It's an older version, 1.0.x something. And I just changed the rx and tx gain on two zap channels |
| 22:10.00 | ManxPower | FarrisG, for 1.x you need to either stop and then start asterisk or unload chan_zap.so and load chan_zap,so |
| 22:10.11 | ManxPower | for 1.2 most zap changes will be applied on a reload |
| 22:10.51 | opc0de | hey can anyone tell me how to get asterisk to display what commands it's executing as it goes along the dialplan? I have a rule in one of my contexts like "exten => i,1,Playback(pbx-invalid)" yet even when I dial an invalid extension, it doesn't get matched |
| 22:11.26 | ManxPower | opc0de, you mean like "asterisk -rvvv" |
| 22:12.13 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 22:12.23 | ManxPower | opc0de, exten => i is normally only works for IVR types of things |
| 22:12.39 | opc0de | it's for when someone dials an invalid extension |
| 22:13.09 | opc0de | I'm already in the console with -vvvvvc, but when I dial an invalid extension in this context, I simply get a fastbusy signal.. I wanted to hear a "you have pressed an invalid extension" sound |
| 22:13.28 | ManxPower | Nodren, it's for when someone dials an invalid extension after a background or waitexten |
| 22:13.31 | ManxPower | <PROTECTED> |
| 22:13.45 | opc0de | ah I didn't see that part |
| 22:13.47 | ManxPower | not if you dial an invalid exten from a zap port or a sip device |
| 22:14.11 | opc0de | okay, so I have no option but the fast busy signal |
| 22:14.55 | ManxPower | opc0de, not really. do something like exten => _X.,1,Whatever |
| 22:15.06 | ManxPower | it should match anything that doesn't already match. |
| 22:15.11 | opc0de | ah yeah, good idea, thanks |
| 22:15.25 | opc0de | I'm still learning |
| 22:15.43 | opc0de | strange that they didn't say anything about only using i after background or waitext in the asterisk documentation project book |
| 22:18.14 | Dovid | live and learn |
| 22:18.14 | opc0de | hmm, that seems to match every extension though, not just invalid ones.. I thought when there'sa more specific matching rule, the more specific rule takes precedence |
| 22:18.27 | Dovid | yes |
| 22:18.31 | Dovid | wats ur problem ? |
| 22:18.43 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 22:18.49 | Dovid | if u have _N800 |
| 22:18.52 | Dovid | or _NXXX |
| 22:18.59 | Dovid | _N800 takes president |
| 22:19.00 | opc0de | cause I have "[from-internal] include => ext-local; exten => _X.,1,Playback(pbx-invalid); exten => _X.,2,Hangup" |
| 22:19.23 | Dovid | ok |
| 22:19.25 | Dovid | u have |
| 22:19.28 | opc0de | where ext-local contains extensions such as 300,301,302,303,etc.. with this new rule, if I dial one of these valid extensions, ie 300, it says "sorry invalid etension" |
| 22:19.49 | Dovid | ok |
| 22:19.55 | Dovid | this is because it works as a flow |
| 22:20.11 | Dovid | the include will have to be b4 the exten => _N.,1 |
| 22:20.15 | opc0de | it is |
| 22:20.20 | opc0de | it's at the top of the context |
| 22:20.23 | Dovid | hmm |
| 22:20.28 | opc0de | lemme try reload again |
| 22:20.32 | Dovid | try to puttin it in without the include and see what happens |
| 22:20.37 | Dovid | could be ur include isnt working |
| 22:21.04 | opc0de | if I take out the _X.,1,.... and leave in the include, I can reach all the extensions, so I believe the include is working |
| 22:21.16 | Dovid | hmm |
| 22:21.30 | Dovid | try putting it all in the same context see what hapens |
| 22:22.07 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:22.47 | Dovid | can be a bug |
| 22:23.20 | Dovid | do a reload |
| 22:23.25 | Dovid | or stop and restart asterisk |
| 22:23.53 | opc0de | okay, this is strange |
| 22:23.56 | Dovid | ? |
| 22:24.19 | opc0de | I just took all the lines from ext-local context and put them in this [from-internal] context, right before the "exten => _X.,1,Playback(pbx-invalid)" and it worked |
| 22:24.26 | opc0de | if I take the lines out and instead use the include, it doens't work |
| 22:24.43 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 22:24.51 | Dovid | paste ur config on pastebin.com and i will look at it |
| 22:24.58 | opc0de | ok |
| 22:26.13 | opc0de | what's the pastebin url? |
| 22:26.27 | iDunno | ~pb |
| 22:26.28 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:26.28 | tecnico | any way to print in the console the time that a user registers ? All I see is " -- Registered IAX2 ... " |
| 22:26.29 | Dovid | pastebin.com |
| 22:26.37 | opc0de | thanks |
| 22:26.47 | Shaun222 | any working software phones for linux with iax2 support? |
| 22:26.59 | tecnico | Shaun222: idefisk |
| 22:27.01 | Nivex | Shaun222: kiax |
| 22:27.10 | Shaun222 | thanks |
| 22:28.41 | Dovid | did u pastebin it ? |
| 22:28.52 | opc0de | http://pastebin.com/629834 |
| 22:29.21 | opc0de | if I take out the stuff from ext-local and paste it instead of the include, it works |
| 22:31.13 | Dovid | ok |
| 22:31.26 | *** join/#asterisk darkskiez (n=darkskie@194.164.233.141) |
| 22:31.29 | Dovid | this is wierd |
| 22:31.33 | Dovid | post it on the list |
| 22:31.40 | opc0de | yeah, seems like a bug |
| 22:32.01 | opc0de | i'm just about to compile 1.2.6, i'll try it out first before posting |
| 22:32.06 | Dovid | kk |
| 22:32.10 | opc0de | this is with 1.2.5 |
| 22:37.46 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 22:38.40 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 22:38.58 | FuriousGeorge | can anyone tell me why im constantly losing my connection with my iax peers on this one box: http://pastebin.ca/47498 |
| 22:39.25 | FuriousGeorge | my provider drops out too |
| 22:39.28 | FuriousGeorge | and hes not dynamic |
| 22:39.34 | FuriousGeorge | the box is though |
| 22:40.12 | FuriousGeorge | the otherones are as well and they lose their peers as often. this didnt correpond to an ip change |
| 22:41.52 | terrapen | anybody played with an IP600 sidecar? |
| 22:42.01 | terrapen | are they useful? |
| 22:44.31 | brad_mssw | terrapen: sup dude |
| 22:44.40 | Dovid | ~google |
| 22:44.42 | jbot | [google] a search engine found at http://www.google.com/ |
| 22:45.03 | Dovid | ~ |
| 22:46.11 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 22:48.35 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 22:49.26 | FuriousGeorge | what could it bee about this one box that its peers spontaneously become unreachable |
| 22:49.33 | FarrisG | I got my door phone working! Using an FXO channel. Problem is, it's either really quiet or just a bunch of noise. Should I just use an external amplifier instead of trying to tune the rx/tx gain in zapata.conf? |
| 22:49.59 | FuriousGeorge | FarrisG: voltage issues? |
| 22:50.33 | inv_arp[work] | any good external fxo's out there these days |
| 22:50.50 | FarrisG | FuriousGeorge: What sort of voltage issues? The door phone itself is passive, and is connected directly to an FXO line. |
| 22:50.50 | FuriousGeorge | i got a viking doorphone that if i turn the volume up too high on the unit it disconnects when ringing |
| 22:51.08 | Maxxed | you guys know of anyway to press a key during a call and have a script fire off ? |
| 22:51.12 | FuriousGeorge | you see the led dim, it gets quieter, and cuts out |
| 22:51.45 | FarrisG | FuriousGeorge: Hmmm... No led on mine, like I said, it isn't powered. It relies on the source to amplify the rx/tx. |
| 22:51.55 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:51.59 | Maxxed | ie. as im talking to sombody, i would like to have the ability to press *1 and that would run a script on the asterisk box |
| 22:52.28 | Maxxed | i think we might hire digium for this one |
| 22:52.49 | mog_work | as much as i love work / money |
| 22:52.53 | mog_work | you can already do that |
| 22:52.59 | mog_work | with the features stuff |
| 22:53.03 | Maxxed | how automon works, xcept no recording, |
| 22:53.09 | Maxxed | hah ;) |
| 22:53.10 | mog_work | you can program dial plan options with that stuff |
| 22:53.16 | Maxxed | well could you point me in the right direction |
| 22:53.28 | mog_work | features.conf |
| 22:53.28 | Maxxed | useing dial() or what? |
| 22:53.29 | FarrisG | Another question... Why does "Show channels" sometimes not show any information about the outbound party when it shows a SIP agent connected through a Zap channel? |
| 22:53.41 | mog_work | open it up rock out |
| 22:53.44 | mog_work | but if i ever see you |
| 22:53.47 | mog_work | you so owe me a coke |
| 22:53.49 | Maxxed | so i can intercept dtmf tones duing a call |
| 22:53.49 | FuriousGeorge | should these 4 boxes i interface with eachother all listen for iax2 connections on different ports? i have serious reliablility issues with my dynaimc ips |
| 22:53.53 | mog_work | yes |
| 22:53.54 | Maxxed | hah ;) |
| 22:54.01 | Maxxed | i'd be delighted |
| 22:54.11 | Maxxed | il paypal ya 75 cents ;p |
| 22:54.17 | Maxxed | i've done it before to a few guys here |
| 22:54.17 | Maxxed | heh |
| 22:54.27 | qseek | doesnt NVBackgrounddetect do that? |
| 22:54.44 | qseek | i thought i read that on the wiki |
| 22:54.45 | mog_work | lol |
| 22:54.52 | mog_work | dont let paypal steal your money |
| 22:55.16 | Maxxed | ah there not that bad |
| 22:55.25 | mog_work | i mean 75 cent transaction has to cost you 25 cents |
| 22:55.51 | Maxxed | yeah, that is a lil shaft |
| 22:55.54 | FuriousGeorge | mog_work: did the changes in 1.2.6 have anything to do with iax2 and unreachable peers? |
| 22:56.15 | mog_work | i just run trunk |
| 22:56.19 | mog_work | ^_^ |
| 22:56.28 | FuriousGeorge | i see |
| 22:56.37 | qseek | mog_work if you run trunk do u know if they released the app_amd |
| 22:56.39 | qseek | with it? |
| 22:56.43 | qseek | in 1.2.6 |
| 22:56.47 | mog_work | app_amd is in trunk |
| 22:56.50 | mog_work | it wont be in 1.2 |
| 22:56.57 | mog_work | it will be in 1.4 |
| 22:57.04 | FuriousGeorge | summer, right? |
| 22:57.06 | mog_work | its easy to back port to 1.2 though |
| 22:57.07 | qseek | so if i was to get it and compile it ..would that work |
| 22:57.08 | mog_work | yes FuriousGeorge |
| 22:57.15 | mog_work | yeah |
| 22:57.18 | mog_work | nothing to tricky |
| 22:57.21 | mog_work | to my knowledge |
| 22:57.26 | qseek | mog_work : ok i will give it a shot |
| 22:57.32 | qseek | when is 1.4 going to be released |
| 22:57.45 | qseek | oh ok |
| 22:57.49 | qseek | ignore that i read it |
| 22:57.50 | qseek | summer |
| 22:58.06 | FuriousGeorge | anything significant being done with iax2 as realted to "dynamic ip'ed" peers? as a whole im finding it strangely unreliable |
| 22:58.19 | FuriousGeorge | when it works it works great, but when it doesnt, no route to host |
| 22:58.27 | qseek | has anyone worked on interfacing asterisk with an IMS |
| 22:58.57 | *** join/#asterisk nickswanjan_ (n=chatzill@69-168-106-108.sbtnvt.adelphia.net) |
| 22:59.00 | Dovid | what is IMS ? |
| 22:59.15 | iDunno | instant messaging service? |
| 22:59.19 | qseek | no |
| 22:59.20 | iDunno | (I guess) |
| 22:59.32 | iDunno | Internet Mail Service? |
| 22:59.39 | angom_w | Internet Multimedia Service ? |
| 23:00.07 | mog_work | integrated messaging system |
| 23:00.49 | mog_work | and there already is some sugar and zimbra integration |
| 23:00.50 | qseek | neat mog_work |
| 23:01.50 | qseek | that would be neat..if an interface/app could be defined..then interfacing with wireless devices would make life much easy |
| 23:02.45 | qseek | mog_work : so could i bug u if i had any issues with app_amd |
| 23:03.01 | Shaun222 | i'm getting this error... "Mar 30 07:52:41 WARNING[12260]: pbx_config.c:1700 pbx_load_module: Invalid priority/label 'Background' at line 14" |
| 23:03.09 | mog_work | i am usually around for bugging |
| 23:03.12 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:03.13 | mog_work | as mog_work or mog_home |
| 23:03.15 | Shaun222 | anybody know why, the docs talk about using Background |
| 23:03.17 | qseek | ok thanks |
| 23:03.32 | harryvv | hi mog |
| 23:03.37 | mog_work | hi harryvv |
| 23:03.59 | Maxxed | mog_work: [applicationmap] is what im looking for ey? |
| 23:04.03 | opc0de | Dovid: I just tried 1.2.6, same problem |
| 23:04.04 | harryvv | I dont know if this is possible but can a vm be played while in a two way convo? |
| 23:04.04 | mog_work | yup |
| 23:04.14 | Shaun222 | n/m i figured it out... |
| 23:04.15 | Dovid | then post to users list |
| 23:04.15 | mog_work | ? |
| 23:04.16 | Maxxed | mog_work: weeee dogggy! :) |
| 23:04.21 | mog_work | its cool stuff |
| 23:04.23 | Maxxed | mog_work: thx for the help! |
| 23:04.28 | mog_work | i have a playback Maxxed of static |
| 23:04.30 | Dovid | Shaun222: why it does or dosent ? |
| 23:04.30 | mog_work | that i can turn on |
| 23:04.32 | *** join/#asterisk MacDome (n=eseidel@A17-255-97-70.apple.com) |
| 23:04.33 | mog_work | if i want to hang up |
| 23:04.35 | Maxxed | mog_work: and yes, yes i do owe u a soda ;) |
| 23:04.52 | mog_work | im holding you to it |
| 23:04.58 | Maxxed | :) |
| 23:05.23 | Shaun222 | Dovid: i typo'd the conf.. |
| 23:05.25 | Shaun222 | was my problem |
| 23:06.01 | Dovid | Shaun222: i am lost, why wouldnt u use it ? |
| 23:06.18 | Shaun222 | no, i am... i was just receiving a error and didnt know why |
| 23:07.18 | harryvv | mog, you have heard of one of our BC farries was lost a few days ago? I have been modeling in 3d what it looks like. The situation was very lucky that only two lives were lost since it can normally handel 700 people. |
| 23:07.47 | Shaun222 | man i'm having a hell of a time just trying to get internal extentions working.. (softphoneA can dial softphoneB |
| 23:08.01 | mog_work | how did it sink? |
| 23:08.20 | mog_work | exten => 100,1,dial(sip/softphonea) |
| 23:08.25 | mog_work | and then the 200 and b |
| 23:08.27 | mog_work | thats it |
| 23:08.29 | harryvv | the ferry went off course and hit a rock. In one hour it sank. |
| 23:08.36 | mog_work | wow |
| 23:08.39 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 23:08.50 | mog_work | i imagine it must be pretty chilly this time of year too |
| 23:08.52 | Strom_C | Shaun222, pastebin the relevant bits of your dialplan and sip.conf |
| 23:08.58 | harryvv | mog, here is what it looks like in 3d members.shaw.ca/glyfx3d/qotn.jpg I have been working on this for two days now. |
| 23:09.35 | mog_work | thats pretty big, i take its jsut a people farry? |
| 23:09.41 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:09.48 | mog_work | no cars? |
| 23:09.51 | mog_work | and other such junk |
| 23:09.59 | qseek | later all |
| 23:10.03 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 23:10.04 | mog_work | bye |
| 23:10.12 | Strom_C | it's a FERRY |
| 23:10.14 | mog_work | ooh someone from nortel |
| 23:10.15 | Strom_C | not a FARRY |
| 23:10.17 | Strom_C | jeez |
| 23:10.18 | mog_work | im sorry |
| 23:10.23 | Shaun222 | http://pastebin.com/629888 |
| 23:10.26 | mog_work | i am ashamed |
| 23:10.27 | ebag | nortel.... grrrr |
| 23:10.32 | Shaun222 | actually the softphones are iax2 |
| 23:10.37 | harryvv | yes, takes cars to |
| 23:10.41 | mog_work | well replace sip for iax |
| 23:10.59 | Shaun222 | did |
| 23:11.03 | Shaun222 | i put iax2 |
| 23:11.04 | mog_work | yeah thats fine |
| 23:11.07 | Shaun222 | that could be my problem? |
| 23:11.07 | mog_work | just have to dial the name |
| 23:11.11 | mog_work | instead of 100 |
| 23:11.14 | mog_work | but it should work |
| 23:11.16 | mog_work | if they register |
| 23:11.16 | Strom_C | Shaun222, no wonder you're having difficulty. your priorities should be 1, not 1001 |
| 23:11.17 | mog_work | to server |
| 23:11.23 | mog_work | oops |
| 23:11.28 | mog_work | i totally missed that |
| 23:11.29 | mog_work | yeah |
| 23:11.36 | mog_work | if you want extension to be 1001 |
| 23:11.53 | Strom_C | exten => 1001,1,Dial(IAX2/shaun) |
| 23:12.03 | Strom_C | exten => 1002,1,Dial(IAX2/steve) |
| 23:12.21 | mog_work | http://pastebin.com/629889 |
| 23:12.45 | Qwell[] | mog_work: I can make a call from my 7960 :D |
| 23:12.55 | mog_work | WOW |
| 23:13.00 | Qwell[] | with dtmf even |
| 23:13.13 | mog_work | you are a mad haxor |
| 23:13.14 | Qwell[] | still gotta work on incoming... |
| 23:13.22 | mog_work | bah all we need is outbound |
| 23:13.25 | Qwell[] | heh |
| 23:13.34 | Qwell[] | lets call it a feature |
| 23:14.17 | Shaun222 | Strom_C: thanks.. |
| 23:14.37 | Shaun222 | must be a good day here.. i've been in here for days and the only person who seams to know anything has been Qwell[]:... |
| 23:14.56 | Strom_C | Shaun222, it's always exten => extension,priority,action |
| 23:15.02 | Shaun222 | Strom_C, mog_work thanks again :) |
| 23:15.31 | *** join/#asterisk fugitivo (n=fugitivo@201.255.183.220) |
| 23:15.47 | fugitivo | hello |
| 23:15.59 | Shaun222 | Strom_C: i see, ok, well now i guess i need to get voicepulse hooked into this thing and do more testing... |
| 23:16.34 | Shaun222 | Strom_C: you have any experience setting up redundant asterisk servers |
| 23:16.47 | Strom_C | no, unfortunately |
| 23:16.59 | Strom_C | but you should probably get the one working before you bite off more than you can chew |
| 23:18.01 | inv_arp[work] | Shaun222: redundant.. hmm heartbeat,NFS,drdb should do it |
| 23:19.19 | Maxxed | omfg you guys asterisk is so fugin badass |
| 23:19.20 | Maxxed | heh |
| 23:19.34 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:19.36 | blitzrage | lol |
| 23:23.22 | Shaun222 | inv_arp[work]: why NFS? |
| 23:24.04 | Shaun222 | inv_arp[work]: just for the configs and what not? i would imagine asterisk cant share it's db between multiple running servers. |
| 23:24.20 | nickswanjan_ | does anyone know if you can specify tos in skinny.conf like you can in sip.conf? |
| 23:24.25 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 23:27.06 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 23:27.31 | austinnichols102 | Getting a ZT_SPANCONFIG failed on span 1: with a TE110P |
| 23:27.43 | blitzrage | sounds like a configuration error |
| 23:28.03 | terrapen | well, i'm placing the order for the foneBRIDGE |
| 23:28.07 | Strom_C | did you chec;k your conf8g file f0r typoz? |
| 23:28.13 | terrapen | hope this thing works well |
| 23:28.30 | austinnichols102 | blitzrage: yeah, invalid argument (22) |
| 23:28.45 | nickswanjan_ | e.g. tos=184, tos=lowdelay, etc... |
| 23:28.47 | blitzrage | you have a typo in zaptel.conf it sounds like |
| 23:28.50 | Strom_C | austinnichols102, pastebin your zaptel.conf |
| 23:28.53 | austinnichols102 | k |
| 23:28.54 | blitzrage | zapata.conf if oyu get it when you start asterisk |
| 23:28.59 | Shaun222 | what is type for in the iax.conf, i see it set to friend or peer all the time but have no idea what those mean |
| 23:29.15 | blitzrage | Shaun222: |
| 23:29.16 | Strom_C | shaun: peer == outgoing, user == incoming, friend == both |
| 23:29.16 | blitzrage | ~docs |
| 23:29.18 | jbot | [docs] 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 |
| 23:29.37 | Shaun222 | i see |
| 23:29.41 | Strom_C | yes... #asterisk is for when RTFM doesn't work |
| 23:29.59 | blitzrage | yah but if you don't know what peer and user and friend is, you didn't RTMF |
| 23:30.04 | Strom_C | it's also for cupcakes |
| 23:30.04 | austinnichols102 | strom_c: or when you're just too damn lazy |
| 23:30.05 | twisted[asteria] | no |
| 23:30.09 | twisted[asteria] | #asterisk is for humor |
| 23:30.14 | blitzrage | twisted[asteria]: true! |
| 23:30.18 | twisted[asteria] | there is no FM |
| 23:30.26 | Strom_C | #asterisk is for DEAD HOOKERS |
| 23:30.30 | twisted[asteria] | uh |
| 23:30.39 | blitzrage | twisted[asteria]: liar |
| 23:31.15 | austinnichols102 | strom_c: http://pastebin.ca/47526 |
| 23:32.14 | Strom_C | *blink* |
| 23:32.15 | Strom_C | wow |
| 23:32.24 | *** join/#asterisk RoyKa (n=roy@28.80-203-106.nextgentel.com) |
| 23:32.38 | Strom_C | you've got a PRI? |
| 23:33.05 | nickswanjan_ | dead hookers = DSCP 38 = high drop probability, class 4 |
| 23:33.41 | austinnichols102 | yes |
| 23:34.21 | Strom_C | austinnichols102, ok...what signaling format does your PRI use? |
| 23:34.32 | austinnichols102 | national (NI2) |
| 23:34.43 | Strom_C | ok, so why isnt that specified? |
| 23:35.00 | austinnichols102 | don't you do that under channels in zapata.conf? |
| 23:35.25 | Strom_C | I'm going mad :) |
| 23:35.27 | Strom_C | one sec |
| 23:36.34 | austinnichols102 | zttool doesn't even show the card |
| 23:37.11 | Strom_C | does it show up under lspci? |
| 23:37.23 | austinnichols102 | checking |
| 23:37.42 | Strom_C | you've only got seven b-channels? |
| 23:38.04 | *** join/#asterisk Catalyst3301 (n=NNSCRIPT@c-66-56-35-93.hsd1.ga.comcast.net) |
| 23:38.11 | austinnichols102 | nope - not under lspci |
| 23:38.18 | Strom_C | well that would be your problem then |
| 23:38.25 | austinnichols102 | yes - 7B, 1D and the rest data (via an adtran) |
| 23:38.25 | austinnichols102 | yup |
| 23:38.37 | Catalyst3301 | Are there asterisk binaries I can go get? I cannot find the kernel sources I need to compile it myself |
| 23:38.46 | Strom_C | Catalyst3301, what distro? |
| 23:38.51 | austinnichols102 | any thoughts on how to figure that out (centos4) |
| 23:39.00 | Catalyst3301 | CentOS4, but I think its a little different |
| 23:39.08 | Strom_C | austinnichols102, shut down, reseat the card, try again |
| 23:39.08 | Catalyst3301 | Its a VPS/VDS |
| 23:39.14 | austinnichols102 | k |
| 23:39.14 | Catalyst3301 | kernel ver gives 2.6.8-022stab067.1-enterprise |
| 23:39.32 | Strom_C | Catalyst3301, isnt there a kernel headers and kernel source package you can install? |
| 23:40.01 | Catalyst3301 | Ive been looking for it. Google doesnt turn up anything |
| 23:40.17 | Strom_C | doesnt centos have a package management system? |
| 23:40.31 | justinu | yum install kernel-devel |
| 23:40.34 | Catalyst3301 | The company tech who manages the box seems to think SWsoft holds the kernel sources |
| 23:40.43 | Catalyst3301 | Wow, Thanks. |
| 23:40.47 | Strom_C | what justinu said |
| 23:40.58 | Catalyst3301 | I figured it was something simple |
| 23:41.08 | Catalyst3301 | But the tech rep I was talking to had no idea.. |
| 23:41.31 | Catalyst3301 | No Match for argument: kernel-devel |
| 23:41.32 | Catalyst3301 | hrm |
| 23:42.41 | Catalyst3301 | I just looked at the list, there isnt a package in there for the kernel. |
| 23:42.58 | justinu | there is on my centos machines |
| 23:43.00 | hfb | Hi Strom_C |
| 23:43.10 | Strom_C | this may be proof that centos blows donkeys for quarters, but I can't be sure |
| 23:43.13 | Strom_C | hi hfb |
| 23:43.18 | justinu | but my machines run 2.6.9-22 |
| 23:43.31 | justinu | not 2.6.8-blah-d-blah |
| 23:43.32 | hfb | Strom_C, You like the food at Denny's? |
| 23:43.43 | Catalyst3301 | Well, would a VPS has a specially made kernel? |
| 23:43.50 | justinu | i dunno what VPS is |
| 23:43.57 | Catalyst3301 | argh |
| 23:44.04 | justinu | i run centos 4.2 |
| 23:44.04 | Strom_C | hfb, it's tolerable |
| 23:44.09 | justinu | zero problems with that distro, btw |
| 23:44.46 | Catalyst3301 | well, I come back to my original question, are there binaries out there for centos? |
| 23:44.50 | Strom_C | VPS == virtual private server |
| 23:45.53 | justinu | the problem you have is you need your kernel modules to match the exact kernel you're running |
| 23:47.38 | Catalyst3301 | erm |
| 23:47.40 | Catalyst3301 | Okay |
| 23:47.53 | Catalyst3301 | So I just gotta keep looking for the sources right? |
| 23:47.57 | *** join/#asterisk phpmattk_ (n=phpmattk@ip-216-7-118-114.fireserve.net) |
| 23:48.04 | Catalyst3301 | or install another kernel.. |
| 23:49.21 | justinu | yeah, pretty much |
| 23:50.23 | justinu | yum list | grep kernel |
| 23:50.27 | justinu | does that come back with anything? |
| 23:50.39 | justinu | or maybe your distro doesn't use yum, i dunno |
| 23:51.25 | Shaun222 | any of you guys using voicepulse connect? |
| 23:51.31 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:51.43 | hfb | Strom_C, I'm going to assume that you were the guy that ended up giving a talk at Denny's on Saturday for sfvlug, yes? |
| 23:51.51 | Strom_C | yes |
| 23:52.17 | terrapen | i need to find a good resource with info about distributing various Asterisk components (voicemail, IVR, queues, etc.) between different boxes |
| 23:52.38 | Strom_C | I wonder why you didnt just ask me that outright instead of asking if I liked Denny's food |
| 23:53.10 | Shaun222 | Catalyst3301: with VPS's you usually cant install your own kernel... |
| 23:53.31 | Shaun222 | Catalyst3301: do you know what VPS software your provider is running? |
| 23:53.39 | Catalyst3301 | Virtuozzo |
| 23:53.39 | jeffgus | Strom_C, he was trying to be sneaky |
| 23:53.44 | Catalyst3301 | Ive done this before |
| 23:53.48 | hfb | I probably shoud have, but I didn't |
| 23:53.57 | Catalyst3301 | I had asterisk up and running with voicepulse connect |
| 23:53.59 | Catalyst3301 | Perfectly |
| 23:54.00 | jeffgus | Strom_C, good impromptu talk BTW |
| 23:54.13 | Strom_C | thanks |
| 23:54.19 | Catalyst3301 | Shaun222: I have |
| 23:54.26 | Strom_C | I personally think it kind of sucked, but if you enjoyed it... :) |
| 23:54.37 | Shaun222 | voicepulse's site gives me 2 servers for outgoing, but their config in their knowledge base doesnt use either of them... |
| 23:55.00 | jeffgus | Strom_C, it's not like you had time to evaluate the audience and prepare the talk :) |
| 23:55.11 | Strom_C | true |
| 23:55.13 | Strom_C | but still |
| 23:55.20 | Strom_C | I hold myself to high standards :) |
| 23:55.28 | Catalyst3301 | Shaun222: erm, well use the ones they give you, unless they dont work. |
| 23:55.50 | Catalyst3301 | Shaun222: I was about to follow the info in the email they sent and have it working fine. |
| 23:55.53 | jeffgus | Strom_C, do you do a lot of asterisk consulting? |
| 23:55.54 | Catalyst3301 | able* |
| 23:55.54 | Shaun222 | Catalyst3301: did you use their prewritten config? |
| 23:56.00 | Catalyst3301 | Yea |
| 23:56.03 | Strom_C | jeffgus, I try |
| 23:56.06 | jeffgus | Strom_C, or it a kinda sideline thing? |
| 23:57.45 | Catalyst3301 | argh |
| 23:57.50 | Catalyst3301 | Man this is killin me. |
| 23:57.57 | Catalyst3301 | I have done this before.. |
