| 00:01.22 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 00:06.43 | *** join/#asterisk simonkern (n=simonker@p54AAA96B.dip0.t-ipconnect.de) |
| 00:06.47 | simonkern | hi |
| 00:08.29 | simonkern | short question: If I want to use chan_mobile, is it necessary to install asterisk svn trunk, or is a normal 1.4 install ok? |
| 00:09.56 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:11.58 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 00:14.45 | *** join/#asterisk NirS (i=Nir@87.68.60.4.cable.012.net.il) |
| 00:17.14 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 00:18.13 | RyanW | after spending lots of $ on various models i've come to the conclusion that 802.11 phones suck. Is there any 802.11 phone that lasts more then 4 hours and doesn't screw up after 30 minutes? |
| 00:18.41 | JT | that's pretty much what we would've said after a few seconds ;) |
| 00:18.54 | JT | all 802.11 phones suck |
| 00:19.04 | ManxPower | RyanW: If you had asked here you would have saved all that much money. |
| 00:19.07 | Innatech | I'll sell you a Polycom for $802.11 ;P |
| 00:19.16 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:19.27 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 00:20.00 | ManxPower | RyanW: If you are not trying to prove your endowment via cool gadgets, a good cordless phone with an ATA should work pretty well for you. |
| 00:20.20 | Innatech | in all seriousness, I haven't heard anyone speak highly of any wifi phone yet. They're all buggy early revision messes. Go ATA + consumer cordless. |
| 00:25.02 | *** part/#asterisk pressureman (n=pressure@210.48.105.162) |
| 00:26.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:28.24 | RyanW | Thanks guys, I'm after an 802.11 pager, any ideas ? |
| 00:28.43 | *** join/#asterisk jtoy_ (n=jtoy@c-24-60-25-28.hsd1.ma.comcast.net) |
| 00:28.54 | JT | Innatech: also the technology sucks |
| 00:29.04 | JT | RyanW: if it must be reliable, forget 802.11 |
| 00:29.07 | JT | too finicky |
| 00:29.29 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 00:29.51 | RyanW | I've rolled out 40 AP's to cover a customers premises and deployed an * PBX on their wired wetwork and now they want the ability to send txt messages to mobile staff |
| 00:29.56 | Innatech | or, if you don't have a nearby port for the ATA, use a standalone bridge and plug whatever you want into it. |
| 00:30.11 | Innatech | txt messages? |
| 00:30.15 | RyanW | sms |
| 00:30.22 | Innatech | Since when does * send SMS to SIP endpoints? |
| 00:30.29 | ManxPower | Innatech: IT doesn't |
| 00:30.41 | Innatech | Ah. OK then. |
| 00:30.57 | RyanW | sipsack allows me to send sms's to sip devices |
| 00:31.03 | RyanW | or whatever its called |
| 00:31.12 | Innatech | oh. Interesting. New to me.... |
| 00:31.20 | ManxPower | But "SMS" is really a generic term for "sending a text message to a cell phone" regardless of if the actual transport is SMS or not. |
| 00:31.36 | ManxPower | RyanW: I'm really glad I don't have your job. |
| 00:31.37 | Innatech | yeah, I thought it was cell-specific. |
| 00:31.55 | RyanW | I've got it working, just not reliably, the 802.11 phones only respond to arp half the time if at all and wont stay associated to the AP |
| 00:32.20 | JT | no surprises there |
| 00:32.27 | ManxPower | RyanW: that is pretty typical of WiFi SIP phones. They also don't roam between APs very well at all |
| 00:32.40 | ManxPower | Did you do ANY research before promising this? |
| 00:32.42 | Innatech | well....you could assign them all static IPs (nightmare) and manually associate the MACs (bigger nightmare). That won't solve association tho. |
| 00:32.52 | RyanW | whats the battery life of something like an ipac with 802.11 ? |
| 00:33.00 | Innatech | * won't solve AP association, that is. |
| 00:33.26 | Innatech | DO NOT GO THAT ROUTE. NONONONONO. No. |
| 00:33.32 | Innatech | No iPAQs. |
| 00:34.17 | Innatech | My 4705 was one of the most dissappointing devices I've ever spent money on. |
| 00:34.38 | RyanW | i guess i could go with consumer cordless phone and prank call the phone with the callerid set to the txt message |
| 00:35.40 | ManxPower | Don't DECT phones support most of these features? |
| 00:35.48 | Innatech | dunno.... |
| 00:35.57 | ManxPower | Also, of course, you can't use 2.4 Ghz cordless phones because of your Wifi network |
| 00:36.06 | Innatech | I thought DECT was just some kind of fancy multipath noise reduction. |
| 00:36.20 | RyanW | has anyone made a conventional cordless phone with sip on the base instead of pstn ? |
| 00:36.52 | ManxPower | RyanW: you sure do have a fetish for SIP |
| 00:37.21 | JT | RyanW: you can get commerical DECT phones too |
| 00:37.27 | RyanW | less peices of equipment in theory = less points of failure |
| 00:37.29 | JT | some might be able to send messages |
| 00:37.49 | ManxPower | RyanW: more cutting edge = more headaches |
| 00:37.49 | JT | but if you must page reliably, get a pager transmitter and proper pagers |
| 00:38.07 | Innatech | RyanW: yes. Aastra 480i CT. SIP phone w/up to 4 cordless extensions. |
| 00:38.18 | coppice | A SIP fetish must be a truly kinky thing :-\ |
| 00:38.22 | RyanW | they already have a paging system, its just not compatable with their new software |
| 00:38.31 | Innatech | RyanW: But I'm not sure if you can treat the extensions as separate endpoints or not. |
| 00:38.37 | ManxPower | Sorry, but I don't want cutting edge, I want low hassle systems that Just Work |
| 00:39.02 | RyanW | looks like i'll build a computer interface for their old paging system |
| 00:39.12 | Innatech | can you use a dedicated FXO to send DTMF into their extant paging system? |
| 00:39.29 | coppice | ManxPower: I know lots of systems that just about work. wanna try them? |
| 00:39.39 | JT | paging over wifi is a nightmare, high band vhf at high power, not so much |
| 00:39.57 | RyanW | not sure how their existing paging system works, i went down the road of trying 802.11 first seeing we have saturated coverage |
| 00:40.49 | JT | pagers usually use high band vhf |
| 00:42.38 | Innatech | heh: http://www.halfbakery.com/idea/WiFi_20Alert_20Pager |
| 00:42.50 | ManxPower | coppice: I think I use many of them already. USA Banking System, cell phones, the internet 8-) |
| 00:43.18 | coppice | ManxPower: sounds like you have in depth expertise already |
| 00:43.24 | Innatech | RyanW: how about this? http://www.tellus.com.tw/FLEX%20Alphanumeric%20Pager.html |
| 00:43.38 | Innatech | WiFi based paging system. |
| 00:43.43 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 00:44.18 | JT | i thought were meant to be NOT suggesting bad ideas, Innatech ;) |
| 00:44.47 | RyanW | thanks, Innatech |
| 00:45.08 | Innatech | Heh. I'm trying to segrate the bad ideas from *, since they seem to be a job requirement for RyanW. |
| 00:45.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:45.19 | Innatech | *segregate |
| 00:45.47 | Innatech | that vendor does look scary, tho. >shrug< |
| 00:45.58 | JT | Innatech: wifi = bad idea for anything important |
| 00:46.10 | Innatech | I tend to agree. |
| 00:46.27 | ManxPower | I must admit that my cat is more technical than most of my users. They tend not to request the tech product of the week. |
| 00:47.01 | coppice | Innatech: why do you call a flex pager WiFi based paging? Different protocols. Different frequencies. Nothing in common at all |
| 00:47.37 | Innatech | coppice: I plead ignorance. That site has their breadcrumbs set up such that those pages are shown as Wireless LAN equipment. |
| 00:48.13 | coppice | that cheating. anyone will accept ignorance as a plea on IRC |
| 00:48.22 | Innatech | coppice: "Product> Wireless Lan >FLEX Alphanumeric Pager" |
| 00:48.59 | Innatech | kinda bastardish of them I guess, if they've nothing to do with each other. |
| 00:49.15 | coppice | that looks strange, but hit the button for the PDF, and its a normal flex pager - 12x, 28x, and 93x megs |
| 00:49.21 | Innatech | ah. |
| 00:49.38 | coppice | I'm amazed anyone still makes those. |
| 00:50.04 | coppice | hardly anyone but motorola made them in their heyday (if you consider they had a heyday) |
| 00:50.32 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 00:52.18 | Innatech | Well--there's also the BB 7270. SIP endpoint with RIM software and a WiFi radio-- no cell radios. If nothing else, I'd imagine the batter would be better and the support should beat out vendors less committed to wireless. They'll also cost plenty tho and its still wifi (so the quality of your mesh will still be an issue.) (For RyanW) |
| 00:52.59 | Innatech | http://na.blackberry.com/eng/devices/device-detail.jsp?navId=H0,C65,P324 |
| 00:53.28 | coppice | a pager for wifi is gonna have a rather "non traditional" battery life for a pager |
| 00:53.42 | Innatech | yeah, for real. |
| 00:54.14 | Innatech | cell phone-style battery life at best. |
| 00:55.38 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:56.14 | Innatech | That BB is 802.11b only, BTW. |
| 00:56.51 | coppice | many hand held devices still are |
| 00:57.22 | *** part/#asterisk jtoy_ (n=jtoy@c-24-60-25-28.hsd1.ma.comcast.net) |
| 01:01.30 | RyanW | Thanks everyone, finally something i can show my boss to get him to beleive me that wireless sucks. |
| 01:01.43 | vn | I wonder if there are ip phones with bluetooth2? |
| 01:02.14 | vn | could be nice if I could hear the ring tone or communicate directly with the phone just by putting my hearing aids to telephone mode... |
| 01:02.18 | vn | wirelessly :D |
| 01:02.58 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 01:05.13 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:16.47 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:17.30 | *** join/#asterisk ManxPower (n=manxpowe@032-442-097.area7.spcsdns.net) |
| 01:21.27 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:23.30 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 01:28.29 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:31.45 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 01:36.47 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 01:40.02 | fujin | anyone using * with a cisco as5400? |
| 01:45.58 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 01:46.41 | MACscr | Where are sip peer settings stored and how can i add monitoring to it? |
| 01:48.08 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:01.03 | *** join/#asterisk kimosabe (n=kimosabe@189.175.33.209) |
| 02:01.43 | kimosabe | im conecting 2 sipura boxes back to back a spa3000 and a 2100 in the dial plan can i use a domain name instead of an ip ? |
| 02:10.00 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 02:11.41 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 02:13.14 | techman97_andy | evening all - I have a PRI hooked into a TE212P, working fine. Question though - if I happen to call a number that is disconnected, busy, etc - right now I get a fast busy and "trunks are busy" message because my telco just dumps the call with a disconnect code. How can I trap that and speak a real message back to the user? |
| 02:15.12 | *** join/#asterisk MrMister2 (n=mrmister@89-180-14-63.net.novis.pt) |
| 02:21.22 | snuff-work | techman97_andy, probably using just 'g' in Dial() |
| 02:21.33 | techman97_andy | http://freepbx.org/trac/ticket/1674?format=rss |
| 02:21.35 | techman97_andy | I think I found it |
| 02:21.49 | snuff-work | then u can go ${DIALSTATUS} == bleh playback(no-number) |
| 02:22.04 | snuff-work | i dont deal with freepbx |
| 02:24.18 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 02:24.50 | MrTelephone | does anyone know if there will be support for reinvite based on destination? |
| 02:25.19 | fujin | anyone use siproxd? |
| 02:27.11 | techman97_andy | snuff-work - I used the logic from that article to make it work in my extensions.conf |
| 02:27.13 | techman97_andy | that's all |
| 02:27.20 | techman97_andy | and...it works! |
| 02:33.57 | fujin | anyone here running a sip proxy of some sort? |
| 02:34.17 | fujin | I've read that asterisk isn't a sip proxy, per se |
| 02:38.52 | *** join/#asterisk Putzz (n=me@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 02:39.17 | ManxPower | It isn't a SIP proxy in any shape or form |
| 02:39.43 | ai-a | a sip server. |
| 02:40.01 | ManxPower | It is what is called in SIP terms "Back to Back User Agent" |
| 02:42.15 | *** part/#asterisk ManxPower (n=manxpowe@032-442-097.area7.spcsdns.net) |
| 02:44.55 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 02:45.26 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
| 02:46.21 | snuff-work | ser is more of a sip proxy.. |
| 02:46.57 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 02:52.18 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:56.17 | MACscr | <PROTECTED> |
| 03:00.38 | JT | fujin: OpenSER is a sip proxy |
| 03:04.41 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:07.37 | Sci_05 | what is it just says chan univavable? same thing? |
| 03:08.30 | fujin | is it possible to temporarily disconnect or kill a user from the asterisk console |
| 03:08.35 | JT | Sci_05: i cannot understand that question, are you refering to something said previously? |
| 03:10.29 | kimosabe | has any oe done any hotlines with the sipuras devices i need a hand please |
| 03:10.42 | Sci_05 | no i was working on a system today where I had a sip phone registered correctly and everything, but every time I would call it I would get chanunivavable. Have not had time to look at it but I was thinking it might be like the 404 error |
| 03:11.28 | JT | Sci_05: oh ok, i had no idea you were responding to MACscr |
| 03:12.25 | MACscr | Sci_05 : let me check |
| 03:13.02 | Sci_05 | thats all I got was chanunivable, nothing else, I had the context correct (as far as I know) and everything....but got notta. |
| 03:13.16 | MACscr | SIP/2.0 404 Not Found |
| 03:14.59 | MACscr | Hmm, im looking through all the logging after i set: sip debug |
| 03:15.09 | MACscr | And did a test call, but i cant find any errors besides taht |
| 03:16.55 | Sci_05 | hmmm ok I will have to digg into it tomorrow morning |
| 03:17.46 | JT | MACscr: err, your 404 error, are calls otherwise working? |
| 03:18.03 | MACscr | This is a brand new box im setting up |
| 03:18.19 | MACscr | My extension and my connection to my sip provider are registred |
| 03:18.22 | MACscr | registered |
| 03:18.29 | fujin | so anyone? a way to kill a user from * temporarily? |
| 03:18.34 | fujin | like kill until they re-register |
| 03:18.47 | MACscr | I dont have a DID yet, so i havent tried incoming yet |
| 03:20.06 | Sci_05 | MACscr: you looking for a provider? |
| 03:20.52 | MACscr | I already ordered my DID's from callcentric, just waiting for them to provision |
| 03:21.01 | MACscr | Got my UK DIDs from gradwell |
| 03:21.05 | *** join/#asterisk keith80403 (n=keith804@71-218-231-185.hlrn.qwest.net) |
| 03:21.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:21.44 | Sci_05 | well if you need any outbound pstn termination let me know ;-) |
| 03:23.24 | aptura | My voip providers have been up and down as of late. |
| 03:24.23 | aptura | having a interesting week changing out some 2 volt batteries at 2,600 Ah's |
| 03:24.24 | aptura | brb |
| 03:26.35 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 03:26.47 | kiwoneka | good evening to all |
| 03:27.20 | *** join/#asterisk ManxPower (n=manxpowe@015-836-877.area5.spcsdns.net) |
| 03:34.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:36.14 | *** join/#asterisk sgtpepper (n=ncorrare@168-171-126-200.fibertel.com.ar) |
| 03:36.26 | sgtpepper | any good post-pay billing engine for asterisk |
| 03:36.40 | sgtpepper | I need to bill against the cdr already recorded from asterisk |
| 03:39.20 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 03:39.30 | linagee | can you connect a data T1 to a data T1 socket? |
| 03:39.55 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 03:40.39 | MACscr | Well, i got my incoming calls to work, woo hoo =P |
| 03:40.57 | MACscr | Now just have to figure out how to fix outgoing |
| 03:43.57 | *** join/#asterisk pigpen2 (n=pigpen@fw.seamans.cc) |
| 03:44.08 | *** join/#asterisk honeysting (n=shams@206-248-138-47.dsl.teksavvy.com) |
| 03:44.41 | pigpen2 | hi all, I need some RT asterisk guidance. |
| 03:44.55 | kiwoneka | what do i have change to enable 'sip notify polycom-check-cfg' for my ip650s |
| 03:45.02 | pigpen2 | I already have a running setup on my dev box. (running 1.4.2) |
| 03:45.13 | ManxPower | kiwoneka: that information is on the Wiki |
| 03:45.28 | pigpen2 | I deployed 1.4.5, with all the same settings/files/databases/etc... and I am getting: |
| 03:45.28 | pigpen2 | config.c:1228 find_engine: Realtime mapping for 'voicemail' found to engine 'pgsql', but the engine is not available |
| 03:45.39 | kiwoneka | i apologize, which one? |
| 03:46.26 | *** join/#asterisk Downchuck (n=chatzill@c-24-22-20-80.hsd1.mn.comcast.net) |
| 03:46.33 | ManxPower | pigpen2: changes are some library was not installed on one of the system (postgress-devel?) maybe |
| 03:46.44 | ManxPower | changes = chances |
| 03:47.34 | pigpen2 | ah..maybe. |
| 03:47.37 | Downchuck | i've got wxcommunicator running from my machine through openser to asterisk.. but i can't for the life of me get ekiga or x-lite to even touch the server. |
| 03:47.51 | ManxPower | pigpen2: Well *something * is different |
| 03:48.03 | pigpen2 | yeah...and there isn't much to it. |
| 03:48.04 | Downchuck | i've got wireshark.. the data is nearly identical.. but on the end-server, with ethereal, i see nothing from ekiga/x-lite |
| 03:48.10 | Downchuck | any clues? |
| 03:48.13 | pigpen2 | thus why I was so suprised to see the error. |
| 03:48.40 | pigpen2 | ManxPower, keeping busy...it has been awhile... |
| 03:48.40 | [TK]D-Fender | ~wikis |
| 03:48.40 | jbot | wikis is, like, http://www.voip-info.org |
| 03:48.55 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.16 | ManxPower | pigpen2: my project for part of last week |
| 03:49.31 | ManxPower | http://www.fnords.org/~eric/macro-std-exten-v2.inc |
| 03:51.18 | *** join/#asterisk honeysting (n=shams@206-248-138-47.dsl.teksavvy.com) |
| 03:51.44 | pigpen2 | ManxPower, wow. That is just....beautiful. |
| 03:52.31 | *** join/#asterisk bmg505 (n=leon@196.209.177.165) |
| 03:52.32 | pigpen2 | question...I see that you start new exten's with 1 then after you use "n"....is this a new feature for avoiding the priority numbering? |
| 03:53.27 | ManxPower | "n" was introduced in 1.2 that and labels exten => _XXXX,n(label-here),etc |
| 03:54.30 | Downchuck | gosh this sip programs are just the most frustrating thing since sendmail |
| 03:54.35 | pigpen2 | so instead of 1, 2, 3, 4....it starts with 1, n, n, n |
| 03:54.48 | ManxPower | Downchuck: softphones give VoIP a bad reputation |
| 03:54.55 | ManxPower | pigpen2: Yup. |
| 03:55.08 | pigpen2 | Ugh...what didn't I know this earlier... |
| 03:55.09 | honeysting | new to this channel, any simiar channel for trixbox, anyone knows? |
| 03:55.11 | Downchuck | ManxPower: so do wifi phones |
| 03:55.15 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 03:55.29 | techman97_andy | honeysting: #trixbox or #freepbx |
| 03:55.42 | honeysting | thx |
| 03:55.44 | ManxPower | Downchuck: Yeah, but everyone knows that |
| 03:55.47 | Downchuck | :P |
| 03:56.04 | Downchuck | i dont have a hard-line phone to test my install with atm |
| 03:56.56 | pigpen2 | ManxPower, I wonder if asterisk has issues with postgres 8.2.x |
| 03:56.57 | Downchuck | must be my mistake :P |
| 03:57.03 | pigpen2 | my current running system is running 8.1 |
| 03:57.11 | ManxPower | Downchuck: The Wiki has lots of bad information, but I believe that most if the config info for various softphones is mostly accurate |
| 03:57.21 | CrashSys | I keep getting channel open messages on my sangoma card... http://pastebin.ca/622832 Does that mean that the lines are open (like disconnected?) or is this a driver-thing? |
| 03:57.26 | [TK]D-Fender | ManxPower, Whats the point of _XXXX in your macro... its not like your do anything DIFFERENT int there because of the exten. Should just be more "s" or something.... |
| 03:57.44 | ManxPower | [TK]D-Fender: CDRs |
| 03:57.45 | Downchuck | ManxPower: I've had them working before, i'm just baffled, how wireshark can show the packets going out, and tethereal never receives them.. |
| 03:58.00 | Downchuck | the firewall is allowing udp and tcp in through those ports |
| 03:58.13 | [TK]D-Fender | ManxPower, You aren't really dialing taht inside the macro.. shouldn't matter... |
| 03:58.14 | *** part/#asterisk honeysting (n=shams@206-248-138-47.dsl.teksavvy.com) |
| 03:58.54 | ManxPower | [TK]D-Fender: Huh? I do lots of dialing |
| 03:59.21 | ManxPower | That one macro can pretty much dial an unlimited number of destinations |
| 03:59.52 | [TK]D-Fender | ManxPower, exten => s,n(check-cfu),GotoIf($[${LEN(${CFU_DEST})} = 0]?${MACRO_EXTEN},1) |
| 04:00.16 | [TK]D-Fender | ManxPower, is how you seem to get to _XXXX. so basically CDR records the last exten # you were on, not tthe starting one? |
| 04:00.48 | *** join/#asterisk honeysting (n=shams@206-248-138-47.dsl.teksavvy.com) |
| 04:01.13 | ManxPower | [TK]D-Fender: We don't really do stuff with CDRs, but as I understand it, the CDR is at least generated for a Dial. |
| 04:02.21 | *** part/#asterisk honeysting (n=shams@206-248-138-47.dsl.teksavvy.com) |
| 04:02.39 | ManxPower | At some point we will prolly do stuff with CDRs, but as I undestand it in -trunk there is code to make Macro generate correct CDRs when in macros |
| 04:04.08 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:04.19 | Downchuck | pigpen: 170 for young elvis, 300 for old elvis |
| 04:05.34 | kiwoneka | good eveing, thanks that works great |
| 04:06.13 | ManxPower | [TK]D-Fender: Apparently one of my clients is installing generators at their largest offices --- except for the office where HQ is located (as well as the main NOC for the compan) |
| 04:07.03 | [TK]D-Fender | ManxPower, Who needs a chest when you've got feet, huh? |
| 04:08.30 | ManxPower | [TK]D-Fender: exactly |
| 04:09.24 | CrashSys | I guess sangoma's wiki is down :() |
| 04:10.40 | pigpen2 | ManxPower, ok..I got it...res_postgres.conf had bad permissions. |
| 04:10.47 | pigpen2 | ie: the dumbass factor. |
| 04:11.33 | ManxPower | 8-) |
| 04:13.22 | vn | aka pebkac |
| 04:16.26 | pigpen2 | something. |
| 04:16.50 | pigpen2 | more like, I did this 6 months ago and forgot to document things.... |
| 04:17.07 | pigpen2 | ie: the standard way for most technical people... |
| 04:18.35 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 04:21.20 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 04:21.54 | *** join/#asterisk duckz (n=duckz@141.85.3.18) |
| 04:22.04 | vn | I always forget to document |
| 04:22.05 | vn | heh |
| 04:22.38 | CrashSys | I keep getting channel open messages on my sangoma card... http://pastebin.ca/622832 Does that mean that the lines are open (like disconnected?) or is this a driver-thing? |
| 04:22.53 | De_Mon | vn document your reminder to document this time and maybe you woln't forget next time |
| 04:23.25 | yakkop | hey: anyone know what happened to AST_MAX_MANHEADER_LEN? I used to have to increase it's value, butnow can't find it... |
| 04:26.38 | vn | De_Mon: easy to say but not always to do... :p |
| 04:27.47 | ManxPower | yakkop: try #asterisk-dev |
| 04:29.16 | ManxPower | yakkop: Try during Digium business hours (9am-5pm CDT, which I think is GMT/UTC -5 |
| 04:30.20 | yakkop | humm... did you ever have to increase it? |
| 04:30.59 | ManxPower | not me |
| 04:32.03 | De_Mon | vn nothing worth doing is ever easy |
| 04:32.23 | De_Mon | -5 is EST |
| 04:32.38 | De_Mon | your thinking -6 |
| 04:32.53 | ManxPower | De_Mon: We are not in Standard Time, we are in Daylight Savings time |
| 04:33.20 | De_Mon | edt is -4? |
| 04:33.22 | vn | my dst tz name is cdt? |
| 04:33.24 | vn | didnt know |
| 04:33.24 | ManxPower | And since UTC does not do Summer Time (as they call it over there).... |
| 04:34.08 | ManxPower | vn: it will be CST in the fall |
| 04:34.09 | De_Mon | I knew that est to edt was an hour different, but somehow didn't make the connection that it changed the GMT/UTC time.. |
| 04:34.19 | De_Mon | I guess I assued UTC followed DST rules too ;) |
| 04:34.36 | ManxPower | Hence the "Universal" part of the name |
| 04:34.38 | vn | I'm in EST... -5 |
| 04:35.32 | De_Mon | date -u says im -4 hours (EDT) |
| 04:35.51 | De_Mon | just when I thought I knew everything you have to bring up something stupid like this! |
| 04:36.12 | ManxPower | 8-) |
| 04:37.19 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 04:37.48 | docelmo | need someone's honest opinion.. What are the benefits of using 1.4 over 1.2? I cant see any.. |
| 04:39.05 | ManxPower | docelmo: Many of 1.4's features are design changes for some parts of Asterisk. Much of that is with the goal of stability. However, I feel that 1.4 is too new to run on any production box I manage. |
| 04:40.48 | De_Mon | ManxPower do you use realtime priority for asterisk? |
| 04:41.36 | De_Mon | I turned it off in 1.2 because of weird behavior, and its still doing weird things on me in 1.4.6 echo where there shouldn't be any echo, completely locking up the machine... weird stuff like that |
| 04:41.40 | docelmo | ManxPower thanks.. Thats what I wanted to hear |
| 04:41.58 | De_Mon | docelmo 1.4.6 has whisper mode for chanspy |
| 04:42.34 | De_Mon | docelmo and meetme login without verification |
| 04:43.12 | De_Mon | and I don't think func_odbc is part of 1.2 either (backport exists tho?) |
| 04:44.09 | yakkop | what about the variable length DTMF -- is that in 1.2 also -- wasn't before.... |
| 04:44.09 | Juggie | correct. |
| 04:44.11 | De_Mon | hrmmm res_snmp is screwed in this compile. it works just enough to make me waste hours trying to figure out why its not working |
| 04:45.08 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:49.20 | *** join/#asterisk kimosabe (n=kimosabe@189.175.33.209) |
| 04:49.31 | kimosabe | i sthere any one here that can help me out a bit |
| 04:52.25 | De_Mon | kimosabe depends on what you ask |
| 04:55.04 | kimosabe | de mon i have two spa3000 back to back with one sipura 2100 as hot lines fxo to fxs i enabled ip dialing how can i make it work with a url ? |
| 04:57.00 | bakermd | Hey all - got kind of an emergency - the box is playing "The number you have dialed is not in service" - how can I see some debug that will lead me somewhere? |
| 04:57.11 | [TK]D-Fender | kimosabe, What are you daisy chaining ATA's? thats like filtering all of your phone calls through 10 exchanges, tin-cup&string, inducted through a fish tank, and back to SIP again.... |
| 04:57.38 | JT | make sure there's a wifi sip link too |
| 04:57.44 | CrashSys | d-fender: high-Def Audio! |
| 04:57.47 | [TK]D-Fender | bakermd, * doe not play such a message itself. Details are sorely lacking for us to tell you anything. |
| 04:58.26 | De_Mon | bakermd does it do that for any call? |
| 04:58.29 | bakermd | Yeah, this is actually a customer's trixbox, so I'll have to go elsewhere... |
| 04:58.52 | kimosabe | tkd fender yes im forwarding lines from an office to a rural office |
| 04:59.08 | De_Mon | bakermd and its a ITSP that your dialing, right? |
| 04:59.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:00.45 | *** join/#asterisk apardo (n=apardo@55.145.217.87.dynamic.jazztel.es) |
| 05:01.17 | yakkop | hey: anyone know what happened to AST_MAX_MANHEADER_LEN? I used to have to increase it's value, butnow can't find it... |
| 05:05.12 | [TK]D-Fender | kimosabe, But why are you CHAINING them? |
| 05:05.33 | [TK]D-Fender | kimosabe, just have the SPA-2100 talk direct to *. |
| 05:05.40 | [TK]D-Fender | kimosabe, its insane to chain them. |
| 05:06.38 | [TK]D-Fender | kimosabe, Acually.... its SIP > 2100-FXS > 3000-FXO > *. May as well remove ALL the hardware |
| 05:07.33 | De_Mon | pish, the hardware makes it more reliable |
| 05:15.18 | *** join/#asterisk Downchuck (n=chatzill@c-24-22-20-80.hsd1.or.comcast.net) |
| 05:22.33 | webavant | I need an app that will create a virtual line-out in my windows sound control panel devices |
| 05:24.01 | *** join/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 05:24.45 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 05:25.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:31.29 | *** join/#asterisk apardo (n=apardo@55.145.217.87.dynamic.jazztel.es) |
| 05:35.03 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:35.59 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 05:36.10 | *** join/#asterisk NirS_ (i=Nir@87.68.60.4.cable.012.net.il) |
| 05:37.08 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:42.25 | *** join/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 05:44.08 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:45.32 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:49.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:02.30 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:03.12 | *** join/#asterisk grEvenX (n=even@ti500720a080-0755.bb.online.no) |
| 06:08.29 | JT | yawn |
| 06:12.33 | jarod14 | JT : http://en.wikipedia.org/wiki/Yawn |
| 06:12.33 | jarod14 | really intresting is'nt it |
| 06:12.41 | JT | yep |
| 06:15.23 | *** join/#asterisk WIRAC (n=edin@cust.citosec.806583-33.bih.net.ba) |
| 06:15.32 | WIRAC | hi |
| 06:15.52 | WIRAC | I need help about asterisk installation |
| 06:16.04 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:16.15 | WIRAC | cause I have problem with CRC check |
| 06:16.30 | WIRAC | Is there anybody who can help me |
| 06:16.34 | *** join/#asterisk jakehow (n=jakehow@user-387hjea.cable.mindspring.com) |
| 06:16.37 | *** join/#asterisk Jabeeds (n=jabeeds@116.240.138.77) |
| 06:16.54 | jakehow | could anyone recommend cheap but reliable rackmount servers in US |
| 06:17.00 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 06:19.04 | JT | hosting or just servers? |
| 06:20.23 | tzafrir | WIRAC, please be more specific |
| 06:20.28 | WIRAC | Ok thx |
| 06:20.32 | WIRAC | Well I |
| 06:20.33 | WIRAC | downloaded |
| 06:20.38 | WIRAC | 32 bit x86 |
| 06:20.44 | WIRAC | package from website |
| 06:20.51 | WIRAC | thanI made linux r PtH |
| 06:20.53 | WIRAC | boot |
| 06:20.53 | tzafrir | binary package? |
| 06:20.56 | tzafrir | asterisknow? |
| 06:20.58 | WIRAC | yes |
| 06:21.06 | WIRAC | I start installation |
| 06:21.14 | tzafrir | there's a separate channel for that. |
| 06:21.22 | WIRAC | which |
| 06:21.34 | tzafrir | At the moment I believe it is quite limited in its built-in support for PRI |
| 06:21.39 | tzafrir | #asterisknow |
| 06:21.44 | WIRAC | ok |
| 06:21.55 | WIRAC | well problem is that my instllation stops at 99% |
| 06:22.11 | tzafrir | ah, that kind of CRC |
| 06:22.16 | WIRAC | yeah |
| 06:22.27 | WIRAC | sth like some serious error |
| 06:22.37 | WIRAC | and it restarts my comp |
| 06:22.45 | tzafrir | have you tried the md5sum check? |
| 06:22.58 | WIRAC | when I enter linux mediacheck |
| 06:23.00 | WIRAC | command |
| 06:23.01 | tzafrir | of the CD, at installation startup? |
| 06:23.06 | WIRAC | it repllies |
| 06:23.13 | WIRAC | that it is possible that checksum |
| 06:23.17 | WIRAC | was not added on cd |
| 06:23.33 | tzafrir | anyway, try #asterisknow or #rpath . This is really not an Asterisk issue |
| 06:23.37 | WIRAC | ok |
| 06:23.39 | WIRAC | thx anyway |
| 06:23.50 | tzafrir | And that checksum is md5, rather than CRC |
| 06:24.31 | WIRAC | ok |
| 06:24.43 | JT | jakehow: ? |
| 06:24.58 | jakehow | JT: sorry.. hardware |
| 06:25.06 | JT | oh ok |
| 06:25.15 | JT | can't go past IBM xSeries :) |
| 06:28.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:30.41 | *** join/#asterisk Jabeeds (n=jabeeds@116.240.138.77) |
| 06:32.23 | Jabeeds | Can someone please tell me, is there any way to have extension.conf forward ANY and ALL contexts to a DB without individually specifying the context in extensions.conf |
| 06:32.50 | Jabeeds | Using realtime of course. |
| 06:33.24 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:33.59 | De_Mon | Jabeeds yes, you put the whole dialplan in a database and dont use extensions.conf at all |
| 06:34.13 | De_Mon | very inefficient tho |
| 06:36.17 | snuff-work | there is not a way around that jabeeds.. least not that i know |
| 06:36.19 | Jabeeds | Do you mean it is inefficient is respect to the amount of db queries that will occur? |
| 06:37.04 | snuff-work | you should always put something thats called very often like a macro in extensions.conf.. since it will go faster |
| 06:37.57 | De_Mon | Jabeeds yes 4 queries per extension priority executed I believe it was |
| 06:38.48 | Jabeeds | what I am trying to do is to create a sort of front end that will enable a person to assign different 'rights' as it were to different users |
| 06:39.18 | Jabeeds | the only way i could think of to do this was through different context for each subset of privs |
| 06:39.39 | Jabeeds | with a db backend of course |
| 06:39.54 | Jabeeds | Is there any other way one could achieve such a task? |
| 06:40.10 | De_Mon | what sort of rights are you tring to grant? |
| 06:41.16 | De_Mon | Jabeeds or use func_odbc to query the rights and then use dialplan logic to determine if those rights allow access to each context |
| 06:41.21 | Jabeeds | basically rights to call different destinations. Ie. User 12 cannot call user 11 or internationally, but user 14 can call user 11 but not user 16 and can call internationally |
| 06:42.28 | De_Mon | that granular eh? a func_odbc call at the start of each of those would be faster |
| 06:43.55 | De_Mon | a macro that dials a number but performs a db lookup for the caller and the callee if results are returned dial, otherwise do something else |
| 06:44.31 | De_Mon | 1 macro 1 func_odbc call, all your calling attempts |
| 06:44.51 | Jabeeds | thanks for your help, ill get reading about func_odbc ;-) |
| 06:45.05 | De_Mon | its oh so sexy |
| 06:48.29 | *** join/#asterisk Downchuck (n=chatzill@c-24-22-20-80.hsd1.mn.comcast.net) |
| 06:51.09 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 06:52.05 | Downchuck | i spent more than 6hrs to realize that my isp's firewall was misconfigured :-/ |
| 06:52.25 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 06:52.52 | rad07 | anybody |
| 06:53.45 | Downchuck | i'm useless, but if you reask the question |
| 06:53.49 | Downchuck | i can tell you for sure i dont know |
| 06:54.11 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:54.50 | rad07 | hi, I might be just steps before a successfull connection of SPA 3102. I am doing some testing and I cannot connect it with PSTN or my phone line. |
| 06:55.29 | rad07 | I am calling from outside line and Asterisk i.e SPA doesnt pick up the phone |
| 06:55.30 | *** join/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 06:57.37 | Downchuck | your outbound working? |
| 07:01.59 | MACscr | Does asterisk have md5 support by default? |
| 07:03.29 | rad07 | Downchuck: I prepare sip.conf and extensions.conf for your review: www.rentalvista.com/sipandextensions.txt |
| 07:03.48 | rad07 | neither outbound or inbound is working |
| 07:03.58 | rad07 | I was following some tutorials |
| 07:04.20 | rad07 | Where do I define number of rings when it answers the call from SPA |
| 07:05.14 | Downchuck | you're probably going to want a register = line in sip.conf |
| 07:05.29 | Downchuck | but i'm a n00b, so caution |
| 07:05.39 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 07:06.00 | rad07 | did you see my settings |
| 07:07.11 | *** join/#asterisk ta^3 (n=tacvbo@189.136.37.170) |
| 07:07.23 | rad07 | I didn't set disallow=all and allow=ulaw in sip.conf. I thing this is default |
| 07:09.00 | Downchuck | i see your context / extensions .. i just dont know if asterisk needs to register with anything |
| 07:09.18 | Downchuck | or if the handset registers with asterisk correctly |
| 07:09.44 | Downchuck | sorry i can't help out more.. remember to check out the asterisk logs |
| 07:09.50 | rad07 | do you have experience with ATAs |
| 07:09.55 | Downchuck | no |
| 07:10.04 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-206-171-23.perm.iinet.net.au) |
| 07:10.25 | rad07 | do you know where to set number of rings before asterisk answers the phone |
| 07:10.28 | Downchuck | i've stayed as far from telephony as i could.. and that took some effort working in phone rooms |
| 07:11.50 | Downchuck | i'd think it's per ms not ring |
| 07:12.22 | rad07 | and where would that be |
| 07:12.40 | Downchuck | http://whirlpool.net.au/forum-replies-archive.cfm/780411.html |
| 07:12.54 | Downchuck | guess i'm wrong |
| 07:13.18 | Downchuck | google it a bit, it's out there |
| 07:13.37 | flenders | rad07: search google for SPA 3000 and asterisk |
| 07:14.01 | rad07 | it has been 2 months since I am seeking for help |
| 07:14.06 | rad07 | no one to answer |
| 07:14.14 | Downchuck | rad07: i know the feeling. |
| 07:14.28 | rad07 | -I cannot make a stupid test |
| 07:15.12 | rad07 | flenders: can you see my sip and extensions settings: |
| 07:15.29 | rad07 | www.rentalvista.com/sipandextensions.txt |
| 07:15.31 | flenders | is your asterisk working? |
| 07:15.44 | rad07 | yes |
| 07:15.58 | rad07 | I can let you check or correct settings |
| 07:16.08 | flenders | ok, hangon |
| 07:17.08 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.189.162) |
| 07:18.13 | *** join/#asterisk patrick- (i=patrick@eos.openroot.de) |
| 07:18.24 | patrick- | Hey all, im having heavy problems compiling mISDN |
| 07:19.09 | patrick- | make[2]: *** Keine Regel, um »modules« zu erstellen. Schluss. |
| 07:19.49 | JT | i'm having trouble reading that |
| 07:20.12 | patrick- | No Rule to create "modules" |
| 07:20.34 | flenders | rad07: this works on SPA3000 |
| 07:20.36 | flenders | rad07: http://pastebin.ca/622944 |
| 07:21.39 | rad07 | flenders: I have tried multiple spa and asterisk settings |
| 07:21.56 | flenders | rad07: this one works at my house |
| 07:22.14 | rad07 | I have spa 3102. It could be that I miss something fundamental. I have telnet ready for someone to review my settings |
| 07:22.34 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 07:22.37 | flenders | telnet?? |
| 07:22.47 | Downchuck | :-) |
| 07:23.03 | flenders | you could have screwed up on the SPA UI |
| 07:24.48 | flenders | try changing the exten to 's' on [line1] |
| 07:25.32 | flenders | rad07: ^^^^^^^^^^^^^ |
| 07:27.59 | rad07 | http://www.rentalvista.com/ |
| 07:28.11 | rad07 | flenders: can you check my linsys SPA settigns |
| 07:28.18 | rad07 | I allow it to be seen |
| 07:28.41 | patrick- | can someone tell me why the compile of mISDN fails with: no rule to make modules ? |
| 07:29.07 | flenders | rad07: yeah, pm me details |
| 07:29.54 | tzafrir | patrick-, I really have no clue. But if you patebin the full log (or specifically: a few lines above that error), and give some details about your platform: - ditsro, kernel and such, someone might be able to help you |
| 07:29.56 | tzafrir | ~pb |
| 07:29.56 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 07:30.13 | patrick- | ok tzafrir : one second |
| 07:31.55 | patrick- | http://pastebin.ca/622954 |
| 07:32.08 | patrick- | kernel and distro are specified at the bottom of the paste |
| 07:35.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:38.01 | patrick- | tzafrir: u didnt fall asleep, did you? :) |
| 07:38.20 | tzafrir | patrick-, I'm not really an misdn guru |
| 07:38.27 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:38.52 | tzafrir | Just a general first-level support at the channel for things I don't know well ;-) |
| 07:38.56 | JT | patrick-: what bri card? |
| 07:39.03 | patrick- | Fritz Card PCI |
| 07:39.27 | tzafrir | you're using Debian, so I'll make an exception |
| 07:39.34 | patrick- | :D |
| 07:39.51 | JT | patrick-: good news, you can use bristuff instead of horrible misdn, i believe |
| 07:39.56 | tzafrir | why do you use use 2.4.27-2 ? Why not get 2.4.27-3 from updates? |
| 07:40.11 | tzafrir | JT: from Fritz? |
| 07:40.27 | JT | tzafrir: you can't? |
| 07:40.45 | tzafrir | bristuff has only drivers for HFC cards |
| 07:40.57 | JT | the question is more why patrick- hasn't upgraded to 2.6 |
| 07:41.03 | JT | running historic kernel |
| 07:41.07 | *** join/#asterisk gzero (n=gzero@81.175.82.2) |
| 07:41.23 | tzafrir | also: apt-get install kernel-headers-`uname -r` |
| 07:41.38 | patrick- | JT: thing is, I'm not too fermilliar with this whole asterisk / IDN / dialup stuff |
| 07:41.38 | patrick- | cause i want to stick with sarge |
| 07:41.49 | tzafrir | (that's for sarge . for newer distros it's: linux-headers-`uname -r`) |
| 07:41.56 | tzafrir | Sarge has 2.6 |
| 07:42.24 | patrick- | 07:40 < JT> the question is more why patrick- hasn't upgraded to 2.6 |
| 07:42.24 | patrick- | 07:41 < patrick-> cause i want to stick with sarge |
| 07:42.27 | patrick- | sorry |
| 07:42.28 | *** join/#asterisk E-bola (n=sdfsdf@212.242.95.146.customer.cybercity.dk) |
| 07:42.35 | patrick- | kernel-headers-2.4.27-2-386 ist schon die neueste Version. |
| 07:42.39 | patrick- | its up2date |
| 07:42.39 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 07:43.09 | JT | you shouldn't be using up2date on debian i'd thing |
| 07:43.12 | JT | think |
| 07:43.13 | michael-i | speaking of historic :) can Asterisk be compiled to run on a 486? I have a stripped down version of Asterisk which exits on signal 4 (invalid instruction) on one of the platforms I'm targeting |
| 07:43.15 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 07:43.43 | patrick- | so JT what exactly would you recommend right now? |
| 07:43.53 | JT | kernel 2.6 |
| 07:44.34 | E-bola | Are anybody uysing AgentCallbackLogin ? I simply cant get it working |
| 07:44.35 | tzafrir | also, a 686 kernel is generally recommende |
| 07:44.35 | *** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:44.45 | *** join/#asterisk shinao1 (n=shinao1@41.205.184.60) |
| 07:44.47 | tzafrir | apt-get install kernel-image-2.6-686 |
| 07:44.56 | E-bola | i just want people to be able to dial an extension, type in their password, and then be joined to the queue |
| 07:44.58 | tzafrir | apt-get install kernel-image-2.6-686 kernel-headers-2.6-686 |
| 07:45.37 | tzafrir | unless you have a P5, or an amdk6 or something |
| 07:45.45 | patrick- | second |
| 07:47.02 | E-bola | anybody here using agents? |
| 07:48.31 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-25-110.red.bezeqint.net) |
| 07:48.34 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 07:49.19 | patrick- | back with 2.6.18-4-686 |
| 07:49.20 | patrick- | :) |
| 07:49.52 | patrick- | even though he couldnt find kernel-headers-2.6-686 |
| 07:51.13 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 07:52.19 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 07:52.30 | patrick- | tzafrir: |
| 07:52.51 | Zeeek | hello |
| 07:53.36 | tzafrir | patrick-, 2.6.18-4-686 is Etch, not Sarge |
| 07:54.11 | patrick- | meaning i run an etch kernel |
| 07:54.33 | patrick- | well whatever |
| 07:54.38 | patrick- | how do I proceed? |
| 07:54.50 | patrick- | i just want to get asterisk to work with my BRI |
| 08:00.12 | patrick- | tzafrir: may i pm? |
| 08:00.52 | tzafrir | patrick-, what sources do you have in your sources.list? |
| 08:01.07 | tzafrir | packages.debian.org is slow today |
| 08:01.11 | Dovid | I am going thru a demo code. i see: DeadAGI(agi://127.0.0.1/ |
| 08:01.18 | Dovid | where is the agi locatied ? |
| 08:01.23 | Dovid | located* |
| 08:01.27 | patrick- | http://nopaste.php-q.net/312030 |
| 08:03.31 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 08:03.31 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 08:04.36 | *** join/#asterisk NirS (i=Nir@87.68.60.4.cable.012.net.il) |
| 08:05.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:08.58 | E-bola | Can you use BLF to indicate if an agent is logged in or not?= |
| 08:09.30 | tzafrir | patrick-, replace "stable" with "sarge" or "oldstable". |
| 08:09.43 | tzafrir | Though I hope it is not too late |
| 08:13.24 | patrick- | the I do what= |
| 08:14.19 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 08:17.33 | *** join/#asterisk wunderkin- (i=wunderki@ip68-2-61-64.ph.ph.cox.net) |
| 08:27.28 | *** join/#asterisk Grizzy (i=Generic@adsl-75-36-154-166.dsl.pltn13.sbcglobal.net) |
| 08:31.13 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:38.22 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:42.41 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 08:43.30 | E-bola | hmmm kind of dead today in here |
| 08:44.43 | creativx | not really |
| 08:45.17 | *** join/#asterisk MrMister2 (n=mrmister@89-180-181-41.net.novis.pt) |
| 08:45.26 | E-bola | mm i'd say so |
| 08:45.38 | E-bola | atleast nobody is answering me :P |
| 08:47.20 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 08:48.27 | redax | hm... |
| 08:48.40 | redax | good morning, |
| 08:49.13 | redax | my Digium B410P stopped working for this morning, again... restarting asterisk+misdn drivers solved the prob. |
| 08:50.37 | redax | do you have any idea? |
| 08:51.32 | redax | last time I solved this problem with changing the digium card to Patton external ISDN to SIP gw. |
| 08:52.08 | Grizzy | Some year they'll have nanotech voltage trace dust to spray on your device. |
| 08:55.31 | redax | Grizzy: seems like all of my trunks are busy/circuit busy... |
| 08:55.53 | redax | meanwhile there's no real channels. |
| 08:56.24 | redax | now I turned on misdn trace. hope that helps... |
| 08:57.06 | Grizzy | Sorry you're having trouble, and sorry if I misled you; I'm no asterisk expert. |
| 08:58.03 | Grizzy | Nor have I ever laid eyes or hands on a digium device. |
| 09:01.23 | redax | hehh. I've very bad opinion about misdn driven hw... |
| 09:01.50 | creativx | word on the street is that misdn is for those with low blodo pressure |
| 09:01.55 | creativx | damnit! i mean.. blood |
| 09:02.04 | *** join/#asterisk Aurs (n=Aurs@ti500720a080-0755.bb.online.no) |
| 09:02.18 | NirS | anyone with FreeRadius experience ? |
| 09:02.22 | redax | that's why I bought digium card, because I sucked deep with other 4port ISDN... |
| 09:02.50 | redax | and what happens... this B410P has the very same misdn driver |
| 09:03.07 | Grizzy | Is there an older ISDN card driver? |
| 09:03.22 | redax | creativx: hehh. I had a low bloodpressure... |
| 09:03.29 | redax | not now. :) |
| 09:03.38 | redax | Grizzy: they're even worst |
| 09:04.26 | Grizzy | It'll help re-grow the hair you're pulling out, too. |
| 09:04.39 | redax | NirS: I've used freeradius back in '99 - 2000 |
| 09:04.48 | redax | ;-) |
| 09:05.22 | NirS | redax, ever used rlm_perl with it ? |
| 09:06.01 | redax | NirS: sorry, never. |
| 09:06.34 | *** part/#asterisk patrick- (i=patrick@eos.openroot.de) |
| 09:07.17 | Dovid | I am going thru a demo code. i see: DeadAGI(agi://127.0.0.1/ |
| 09:07.24 | Dovid | where is the agi file located ? |
| 09:07.52 | creativx | hehe |
| 09:09.26 | redax | Grizzy: patton has nice devices |
| 09:12.03 | vgster | Hello, does anyone have distinctive ring tones working with aastra 53i phones? |
| 09:15.22 | JT | Grizzy: not older, better |
| 09:15.34 | JT | bristuff is available, but not for the digium B410P |
| 09:15.47 | JT | misdn is the old horrible isdn4linux renamed |
| 09:19.26 | JT | Grizzy: bri to sip gateway will probably be the proper name |
| 09:20.24 | Tond | .. |
| 09:20.34 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:20.46 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 09:21.20 | JT | Tond: what are you dotting about? |
| 09:21.25 | Grizzy | unless it's BRI to IAX on ethernet. |
| 09:21.37 | JT | yes well that will never happen |
| 09:21.56 | JT | no-one makes bri to iax gateways as far as i know, i doubt anyone would |
| 09:22.42 | Grizzy | ISDN always seems to be some sort of deep despiration; too far from the central office .... |
| 09:22.52 | JT | err what? |
| 09:24.10 | Grizzy | Nobody seems to use ISDN unless their loop length is way too long. |
| 09:24.23 | JT | you seem to be sorely mistaken |
| 09:24.28 | JT | what country are you from? |
| 09:24.43 | Grizzy | T1's, yes. |
| 09:24.44 | JT | pots works better with really long lines |
| 09:24.48 | *** join/#asterisk juuva (i=juuva@peili.org) |
| 09:25.02 | Grizzy | USA, California. |
| 09:25.18 | JT | well bri is not popular in north america |
| 09:25.29 | JT | it's superior to pots generally |
| 09:25.36 | JT | it's just a mini PRI |
| 09:25.46 | JT | PRIs are ISDN too |
| 09:26.17 | Grizzy | People use ISDN when the loop is so long that the POTS signal is badly attenuated. |
| 09:26.38 | JT | that makes no sense |
| 09:26.42 | JT | it's a digital service |
| 09:26.52 | Grizzy | right. |
| 09:26.55 | JT | hence caps the line length |
| 09:27.05 | JT | to be under a certain level of attenuation |
| 09:27.29 | Grizzy | so the bits can be reconstructed into nice, loud analog POTS at the subscriber's place. |
| 09:27.53 | JT | too bad if your telco only uses it for crappy purposes |
| 09:28.09 | JT | and it doesn't work well on bad lines :) |
| 09:28.25 | Grizzy | it's the american way. All hail AT&T. |
| 09:29.46 | JT | isdn bri is generally preferable pots |
| 09:29.55 | JT | proper call supervision, 2 channels, digital... |
| 09:31.07 | Grizzy | much nicer, but it's tarriffed at 5c/minute/connection, here. |
| 09:31.29 | JT | again, that's not a problem with the technology |
| 09:32.40 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 09:32.45 | Grizzy | I think you're misunderstanding me. |
| 09:33.04 | *** join/#asterisk appelza (n=pieter@dsl-244-195-116.telkomadsl.co.za) |
| 09:34.03 | appelza | Hi guys, I'm trying to install asterisk using BRI, I've compiled and installed Zaptel with the BRI patch, but asterisk errors out in compile: chan_zap.c:2382: error: 'BRI_NETWORK_PTMP' undeclared (first use in this function) |
| 09:34.12 | appelza | Please help :] |
| 09:36.38 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 09:38.12 | *** join/#asterisk porche (n=porche@88.239.79.61) |
| 09:38.17 | porche | hi all |
| 09:38.18 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 09:38.57 | porche | i have got a question: is there a numbering plan database per country? i am after the mobile number prefixes |
| 09:39.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:42.08 | Grizzy | worry about number portability |
| 09:42.56 | porche | yes, it's another issue for sure, but in general mobile prefixes are usually the same |
| 09:50.03 | creativx | well |
| 09:50.06 | creativx | mine starts with 99 |
| 09:50.19 | porche | good |
| 09:51.47 | JT | Grizzy: what's to misunderstand about < Grizzy> Nobody seems to use ISDN unless their loop length is way too |
| 09:51.50 | JT | <PROTECTED> |
| 09:51.53 | JT | ? |
| 09:55.37 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 10:06.19 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:26.36 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 10:32.04 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 10:34.18 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 10:37.14 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 10:40.19 | Dovid | !seen me |
| 10:40.29 | *** part/#asterisk porche (n=porche@88.239.79.61) |
| 10:40.58 | *** join/#asterisk yassaccan (n=yassacca@admin238.hgo.se) |
| 10:43.15 | *** part/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 10:53.40 | *** join/#asterisk badcfe (i=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 10:54.59 | badcfe | hello. when i do two consecutive calls to Dail and the second one gets answered, then CDR(answered) still contains the data for the first Dial attemt. how can i reset so i get correct information for the last issued Dial? |
| 10:56.29 | *** join/#asterisk FlatFoot (i=FlatFoot@80.88.192.83) |
| 10:56.38 | FlatFoot | mornin all |
| 10:57.59 | FlatFoot | anyone recommend a good billing engine ( web based ) ? |
| 10:58.32 | *** join/#asterisk blue (n=blue@213.173.227.213) |
| 11:10.17 | *** join/#asterisk colde (n=colde@pdpc/supporter/active/colde) |
| 11:15.34 | *** join/#asterisk salvatore2 (n=cn@teknopet.com) |
| 11:15.37 | salvatore2 | hello |
| 11:15.42 | salvatore2 | i am having a problem with asterisk |
| 11:15.57 | salvatore2 | my asterisk server doesn't register with another server |
| 11:16.09 | salvatore2 | even though the settings are correct, it says 401 unauthorized in debug |
| 11:16.11 | ai-a | salvatore2: logs / sip / iax settings.. |
| 11:16.19 | cpm | sounds like a lack of discipline |
| 11:16.25 | salvatore2 | <--- SIP read from 194.221.62.198:5060 ---> |
| 11:16.26 | salvatore2 | SIP/2.0 401 Unauthorized |
| 11:16.59 | salvatore2 | register command is: register=> user:password@194.221.62.198 |
| 11:17.02 | ai-a | salvatore2: use pastebin website, and upload your asterisk-cli outputs with high debug, and your sip debug.. and so on. |
| 11:17.11 | salvatore2 | okay just a second sorry |
| 11:17.14 | ai-a | salvatore2: also, you shouldnt splace your IP address here. |
| 11:17.24 | ai-a | loads of people iax hacking your box now. |
| 11:17.37 | ai-a | search / replace all sensitive information. |
| 11:19.50 | salvatore2 | http://www.pastebin.ca/623116 |
| 11:21.10 | salvatore2 | my user and password is correct 100% |
| 11:24.01 | *** join/#asterisk sigmounte (n=sigmount@lns-bzn-50f-81-56-234-199.adsl.proxad.net) |
| 11:24.34 | salvatore2 | any ideas |
| 11:24.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:29.31 | ai-a | i would say your login user/pass are not correct. |
| 11:30.04 | ai-a | 401 = invalid login... so.. something is invalid. |
| 11:30.20 | ai-a | its not your server your connecting to, so unless they have issues (contact them) your using invalid cridentials. |
| 11:30.29 | ai-a | *credentials |
| 11:30.52 | ai-a | i thought you ment you had two asterisk servers and were trying to connect them together. |
| 11:31.33 | ai-a | also,, are you register'ing this account ? |
| 11:37.28 | FlatFoot | anyone recommend a good billing engine ( web based ) ? |
| 11:44.43 | *** join/#asterisk jsbach (n=jsbach@fokus6150.fokus.fraunhofer.de) |
| 11:45.18 | jsbach | hi , just installed asterisk and trying to get sip.conf working. |
| 11:45.44 | jsbach | alice gets registered calls bob, for that she gets a 404 Not Found. |
| 11:46.20 | jsbach | now as far as i understand all users must be inputted to sip.conf which is the case for my tests... so what is wrong? |
| 11:47.07 | jsbach | i googled this, there are pretty much questions about this, none of them really answered.. any ideas ? |
| 11:48.57 | jsbach | hello there? |
| 11:49.03 | jsbach | anyone up? |
| 11:49.59 | salvatore2 | ai-a, are you still there |
| 11:50.16 | ai-a | yep |
| 11:50.55 | salvatore2 | so |
| 11:50.58 | salvatore2 | my username is correct |
| 11:51.09 | salvatore2 | actually this setup was working 1 hour ago |
| 11:51.16 | salvatore2 | somehow it started rehecting me |
| 11:51.24 | ai-a | jsbach: need to see your sip setup and are these users logged in.. and what are they calling.. and the cli output... upload to a pastebin site |
| 11:51.28 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 11:53.28 | salvatore2 | ai-a, now strangely it is working again |
| 11:53.31 | jsbach | ai-a, i am trying to make use of first dialplan from the "asterisk, the future of telephony" b ook. |
| 11:53.44 | salvatore2 | it was always working with my sip softphone, even when i wrote you |
| 11:53.54 | jsbach | alice and bob are both in sip.conf.. whereas alice is registered(what you call online), and she uses xlite as phone.. |
| 11:54.00 | ai-a | jsbach: and i dont have that book. |
| 11:54.13 | ai-a | salvatore2: i would say that they had a server issue. |
| 11:54.29 | ai-a | registeration failure somewhere.. and now its back up again... thats what you get for free / cheep servers. |
| 11:54.58 | jsbach | ai-a, does not matter about the book.. the dialplan is simple s,1,Answer() , s,2,Playback(hello-world) , s,3,Hangup() |
| 11:54.59 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 11:55.08 | ai-a | jsbach: and the LOG say ? |
| 11:55.31 | Uatec | hi |
| 11:55.41 | jsbach | ai-a, now i even do not get to the dialplan at all.. i enabled the log the invite comes to the asterisk machine and it replies with 404 Not found to the caller (alice) |
| 11:55.59 | ai-a | jsbach: your wasting time... as i said,, upload your asterisk-cli output, and your sip settings,, to a pastebin site, and someone will have a look. |
| 11:56.19 | ai-a | if you just carry on explaining the problem,, your going to miss the mistake that someone can find for you. |
| 11:56.30 | jsbach | ai-a, ok i do it, gimme a paste-bin site.. |
| 11:56.42 | ai-a | http://paste.css-standards.org/ |
| 11:57.31 | jsbach | ai-a, hold on for a sec, i am re-making the test scenario and will be pasting it in 2 mins. |
| 11:57.40 | ai-a | fine. |
| 11:58.02 | ai-a | basicly idont have time for a converstaion.. if you just said . blah doesnt work.. here is my output -> .. <- and waited.. you would get better response.. |
| 11:58.11 | ai-a | i have my own work to do. |
| 11:59.46 | jsbach | ai-a, i understand |
| 11:59.48 | jsbach | http://paste.css-standards.org/19581 |
| 12:00.00 | jsbach | the sip traffic it is now i am sending my sip.conf |
| 12:02.25 | ai-a | and where has bob registered ? |
| 12:03.19 | jsbach | ai-a, http://paste.css-standards.org/19583 |
| 12:03.21 | jsbach | sip.conf |
| 12:03.50 | jsbach | bob is not registered, that's the part, alice calls bob, bob is not available but his extension will be executed... |
| 12:03.58 | jsbach | an answering machine |
| 12:04.07 | ai-a | eh, no. |
| 12:04.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:04.22 | Qwell[] | hilarious |
| 12:04.23 | ai-a | bob doesnt have an extention,, bob is a sip registeration. |
| 12:04.32 | ai-a | *extension. |
| 12:04.32 | Qwell[] | css-standards.org isn't CSS compliant |
| 12:04.40 | ai-a | lol qwell. |
| 12:05.06 | ai-a | jsbach: do you have an extensions.conf associated with this example ? |
| 12:05.14 | jsbach | ai-a, afaik , i specify an extension with bob, see context=incoming.. |
| 12:05.25 | jsbach | ai-a, yes i have an extension.conf |
| 12:05.26 | ai-a | thats a CONTEXT,, not extension. |
| 12:05.27 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 12:05.28 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:05.34 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 12:05.42 | ai-a | how does screaming bob map to a mailbox number ? |
| 12:05.47 | ai-a | what mailbox number is bob ? |
| 12:06.07 | jsbach | ai-a, huh? i thought that should map to the [incoming] in the extension.conf |
| 12:06.15 | ai-a | thats a context. |
| 12:06.31 | jsbach | ai-a, i see.. i mixed sth there i think then. |
| 12:06.32 | ai-a | do you have a mailbox.conf too ? |
| 12:06.40 | jsbach | ai-a, let me have a look |
| 12:06.52 | ai-a | without all the data, how am i suppost to guess the problem ? |
| 12:07.03 | jsbach | no , but i can find via slocate from the default inst |
| 12:07.40 | jsbach | ai-a, it is documented so, that you get the impression that not two users have to registered.. this is what book says which you dont have |
| 12:07.43 | ai-a | so,, from what you've given me,, how do i know what exp bob is ? |
| 12:07.59 | ai-a | sure, nobody needs to register if you want it to go to voice mail. |
| 12:08.16 | *** join/#asterisk saftsack (n=saftsack@83-131-189-52.adsl.net.t-com.hr) |
| 12:08.32 | jsbach | ai-a, yes that is for instance how it works with other pbx like things too(like sems) |
| 12:08.44 | ai-a | jsbach: can you please show me ext..conf |
| 12:08.53 | jsbach | ai-a, yes of course |
| 12:08.56 | jsbach | hold it.. |
| 12:09.00 | *** join/#asterisk friedrich| (n=friedric@e177250225.adsl.alicedsl.de) |
| 12:09.38 | jsbach | http://paste.css-standards.org/19585 |
| 12:09.41 | creativx | achtung i love pivot reports |
| 12:09.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-53-222.socal.res.rr.com) |
| 12:10.04 | ai-a | lol |
| 12:10.04 | ai-a | okay.. |
| 12:10.25 | ai-a | jsbach: how far into the book is this example ? |
| 12:11.09 | ai-a | do you understand the concept of context, extention, iax/sip registeration, user login? |
| 12:11.11 | jsbach | ai-a, chapter 5 , dialplan basics, page 83 , though first edition, under "our first dialplan" |
| 12:11.14 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:11.31 | jsbach | ai-a, yes i do |
| 12:11.38 | ai-a | what are you calling when callining bob ? |
| 12:11.43 | ai-a | bob@pbx ? |
| 12:12.23 | ai-a | bob@semiconductor.jsbach in this case,, right ? |
| 12:12.36 | jsbach | ai-a, yes you are right it is my fqdn |
| 12:12.51 | ai-a | ok, well, if bob isnt there, it will just give up. |
| 12:13.02 | ai-a | are you sure the book is stating it will go to voicemail ? |
| 12:13.28 | jsbach | ai-a, hold on |
| 12:13.42 | ai-a | is this an online book ? |
| 12:14.00 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:14.00 | jsbach | ai-a, yes yes |
| 12:14.03 | ai-a | i see it.. will check it;) |
| 12:14.07 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 12:14.08 | ai-a | my bad for never reading books haha. |
| 12:14.22 | twitchnln | good morning everyone |
| 12:14.23 | jsbach | from jim van meggelen , jared smith & leif madsen |
| 12:14.25 | ai-a | oh, this is that long winded blah blah book that makes you fall asleep. |
| 12:14.53 | jsbach | ai-a, hehe.. that's true,, in some cases i like that.. but in some it is quite unclear.. ;) |
| 12:15.21 | ai-a | nar, it just hops around like a horny rabbit, with no reason, expecting you to understand what hes thinking. |
| 12:15.51 | ai-a | well i see "our first dialplan".. |
| 12:16.02 | ai-a | dial any number and it says "hello world" |
| 12:16.04 | jsbach | ai-a, as you pointed out about mailbox redirection, i started to think you might be rite.. |
| 12:16.21 | ai-a | i sometimes am. |
| 12:16.46 | jsbach | ai-a, it says something like "if we're GOING TO ANSWER THE CALL, play a sound file and then hung up,...." , this pretty much means that bob should be also registered.. |
| 12:17.01 | ai-a | jsbach: consider this bit of logic... something must state if you want or dont want voicemail on that extention.. if you cant find it.. its not got voice mail. |
| 12:17.08 | jsbach | ai-a, but again it doesn't say concretely, caller and callee must be registered.. |
| 12:17.18 | ai-a | if you call user@sip directly.. you need to set up extensions to handle this. |
| 12:17.39 | ai-a | are you still on page 83 ? |
| 12:17.56 | ai-a | the context [incomming] play hello world, hang up.. one ? |
| 12:18.05 | ai-a | from alice dial 1.. and thats it. |
| 12:18.14 | jsbach | ai-a, so are you saying that alice calls bob, sip.conf points bob's extension, bob's extension says, ok, if bob is unavailible please read his voicemail.conf |
| 12:18.15 | jsbach | ? |
| 12:18.22 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 12:18.26 | jsbach | ai-a, you are fast, yes i am at 83 |
| 12:19.37 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:20.02 | ai-a | its funny,, i cant find anything that states the called party must be connected... but you take that as an assuption. |
| 12:20.21 | jsbach | ai-a, that 's why i accused the book as being unclear ;) |
| 12:20.39 | ai-a | well, haha, if you dont have your home phone connected,, i dont expect to get though to you. |
| 12:20.51 | ai-a | if your asleep i dont expect you to hear me. |
| 12:21.00 | ai-a | bit of common sense there. |
| 12:22.02 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:22.05 | ai-a | also,, jsbach: where does the book state you to call bob@... anyway ? |
| 12:22.18 | jsbach | ai-a, see from my side, asterisk is enough clever that bob has a context, sip.conf finds that in extensions.conf and then executes it whatever it is |
| 12:22.26 | ai-a | it talks consistantly about extensions being mapped to a sip register. |
| 12:22.55 | ai-a | no its not bobs context. |
| 12:23.15 | ai-a | when bob dials digits,, it examines the context bob uses to know what to do about these digits. |
| 12:24.04 | ai-a | if bob has context=default [default] exten _9X!,1,Dial(Zap/1,${EXTEN:1}) will redirect 9012345 to the zap device for handling the call. |
| 12:24.26 | ai-a | [default] exten = 555,Dial(SIP/alice) will dial alice if bob dials "555" |
| 12:25.00 | ai-a | bit of syntax issues with my examples, but get the idea ;) |
| 12:25.23 | ai-a | best off,, continuing with the book and you'll learn. |
| 12:25.39 | jsbach | ai-a, no probs about the syntax even if zaptel section was unclear, i have the idea.. just trying to be clear myself here the difference between context and extensions.. |
| 12:26.43 | ai-a | ok, using just sip.conf and extension.conf users dont really have extensions... you just have context that handle the digits the phone dials. |
| 12:27.17 | ai-a | to associated an extention just have it when someone in the context dials the specific digits, it goes to that sip. |
| 12:27.35 | ai-a | exten = 555,1,Dial(SIP/alice) <- alice now has ext 555 |
| 12:27.45 | ai-a | exten = 666,1,Dial(SIP/alice) <- alice now has ext 666, as well as 555 before. |
| 12:28.00 | jsbach | contexts are named groups of extensions.. simply put , they keep different parts of the dialplan from interacting with one another. |
| 12:28.11 | ai-a | yes |
| 12:28.23 | badcfe | hello. when i do two consecutive calls to Dail and the second one gets answered, then CDR(answered) still contains the data for the first Dial attemt. how can i reset so i get correct information for the last issued Dial? |
| 12:28.27 | ai-a | but i try to look at it as rules for digits, rather than groups of extensions. |
| 12:29.20 | [TK]D-Fender | badcfe: "show application ResetCDR" |
| 12:29.23 | ai-a | exten = 555X,1,Dial(SIP,alice) alice gets triggered / called, when any phone number starting with 555.. so 555 183734 will call her. |
| 12:29.36 | badcfe | [TK]D-Fender: thanks. that gets me ahead |
| 12:29.38 | [TK]D-Fender | ai-a: ... NO |
| 12:30.03 | [TK]D-Fender | ai-a: tthe last pile of samples you've given were actually just about ALL wrong :/ |
| 12:30.13 | ai-a | ;) thanks. |
| 12:30.26 | [TK]D-Fender | ai-a>if bob has context=default [default] exten _9X!,1,Dial(Zap/1,${EXTEN:1}) will redirect 9012345 to the zap device for handling the call. |
| 12:30.27 | [TK]D-Fender | [08:24]<ai-a>[default] exten = 555,Dial(SIP/alice) will dial alice if bob dials "555" |
| 12:30.32 | [TK]D-Fender | *gasp* |
| 12:31.04 | [TK]D-Fender | ai-a: that first one : parameter mismatch. He'll get empty dialtone after a long WAIT. |
| 12:31.05 | Qwell[] | [default] exten = 555,Dial(SIP/alice) will also dial Alice if Joe Hax0r dials 555 |
| 12:31.15 | Qwell[] | well, assuming there was a priority |
| 12:31.19 | [TK]D-Fender | ai-a: Second could really use a PRIORITY |
| 12:31.25 | ai-a | yer, syntax was wrong. |
| 12:31.32 | [TK]D-Fender | Qwell[]: Mornin' |
| 12:31.35 | ai-a | im typing it, im bit off ;) |
| 12:31.44 | Qwell[] | [TK]D-Fender: hell, it isn't even morning yet, as far as I'm concerned |
| 12:32.00 | jsbach | isn't this rite? if alice gets called, the context=[incoming] field will found in sip.conf, then [incoming] will be searched in extensions.conf? |
| 12:32.01 | juuva | It's midday |
| 12:32.07 | jsbach | and executed afterwards ? |
| 12:32.13 | ai-a | but my point was,, dont thing of a context holding extension groups,, but rules for handling digits called. |
| 12:32.23 | *** join/#asterisk DaveCanoe (n=Dave@H12.C16.B96.tor.eicat.ca) |
| 12:33.34 | ai-a | jsbach: context=[incoming] is for asterisk to know which context to refer to when alice dials digits. |
| 12:33.41 | ai-a | not for when someone calls alice. |
| 12:33.53 | jsbach | okkay, i see know.. |
| 12:34.04 | jsbach | now , i meant |
| 12:34.06 | jsbach | :P |
| 12:34.25 | [TK]D-Fender | jsbach: should be : context=incoming |
| 12:34.26 | ai-a | so you have have complete different rules for alice from bob. |
| 12:35.05 | [TK]D-Fender | ai-a: adn NO, stop calling it "dials digits", that is a VERY incorrect outlook. SIP phones only pass an entire NUMBER in 1 shot. |
| 12:35.22 | [TK]D-Fender | jsbach: You'd do well to forget that "concept" |
| 12:35.32 | ai-a | okay, i'll get back to work then. |
| 12:35.47 | jsbach | hehe |
| 12:35.49 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 12:36.08 | jsbach | [TK]D-Fender, jep, i see. |
| 12:36.40 | [TK]D-Fender | the dialplan is not examined every time you push a button on a phone before you have completed your dial. |
| 12:37.05 | salvatore2 | if you use a sip ata, that examines its own dialplan every time you push a button |
| 12:37.36 | [TK]D-Fender | salvatore2: Some not even so much. I've seen SIP devices with NO dialplan that just wait for timeout / # to terminate |
| 12:37.50 | salvatore2 | disgusting... |
| 12:38.20 | salvatore2 | best ata ever seen is pap2 |
| 12:38.37 | [TK]D-Fender | salvatore2: personally I use a very open dialplan like that on all of my deployments. |
| 12:39.09 | [TK]D-Fender | PAP2 is "ok", SPA-2102 is better, and MediaTrix kills them both. |
| 12:39.17 | mvanbaak | I have x-lite on a windows xp computer. I can call it and the xlite can make calls |
| 12:39.21 | salvatore2 | i need to check mediatrix |
| 12:39.23 | salvatore2 | never heard of it |
| 12:39.27 | salvatore2 | is it much better than pap2? |
| 12:39.37 | mvanbaak | but the weird thing is, it takes about 10 seconds after I pickup the xlite before the call is actually connected |
| 12:39.56 | mvanbaak | anyone any idea what it could be ? |
| 12:40.00 | [TK]D-Fender | mvanbaak: makes no sense... |
| 12:40.13 | mvanbaak | my idea |
| 12:40.29 | mvanbaak | to make it more clear |
| 12:40.35 | salvatore2 | lol |
| 12:40.36 | salvatore2 | no |
| 12:40.36 | mvanbaak | the softphone is internal number 10 |
| 12:40.45 | mvanbaak | my softphone (ekiga) is number 13 |
| 12:40.48 | [TK]D-Fender | salvatore2: indeed it is. both the other 2 support T.38, faster processor. Mediatrix supports simultaneous G.729 on ALL calls through it (2 x conference) |
| 12:40.57 | mvanbaak | if number 13 calls number 10 xlite rings |
| 12:41.09 | mvanbaak | when I hit the green ok button it takes like 10 seconds before we are connected |
| 12:41.10 | [TK]D-Fender | salvatore2: PAP2 is the BOTTOM of the line. |
| 12:41.23 | salvatore2 | maybe, i can buy it for 39bucks here |
| 12:41.30 | salvatore2 | and it does everything |
| 12:41.32 | [TK]D-Fender | salvatore2: Still plenty decent for basic use, but I still prefer a SPA-2102 over it in a second. |
| 12:41.56 | salvatore2 | there was a rumor, i've read a story telling that pap2t can support g729 2 channels |
| 12:41.59 | salvatore2 | but it doesn't actually |
| 12:42.06 | [TK]D-Fender | salvatore2: G.729 on *1* channel only, no T.38 support, no internal router..... its the BOTTOM end. |
| 12:42.20 | salvatore2 | how much is the mediatrix |
| 12:42.39 | mvanbaak | same with dialing with xlite |
| 12:42.51 | mvanbaak | I put in 13 and hit the green button |
| 12:43.02 | [TK]D-Fender | salvatore2: MediaTrix = $$ but supports G.729 on all channels, can do a transparent proxy in front of ANYTHING...... very powerful, but rarely needed |
| 12:43.02 | mvanbaak | it takes like 10 seconds before it actually starts to ring |
| 12:43.14 | mvanbaak | what can that be ? |
| 12:43.20 | [TK]D-Fender | mvanbaak: show the CLI output of your call and your dialplan. |
| 12:43.30 | salvatore2 | mvanbaak, do you use a outbound proxy? |
| 12:43.40 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:44.33 | salvatore2 | do you guys have experience with voicetrading.com ? |
| 12:45.23 | mvanbaak | [TK]D-Fender: http://pastebin.ca/623202 |
| 12:45.54 | mvanbaak | salvatore2: both softphones and asterisk are on the same internal network |
| 12:46.18 | [TK]D-Fender | mvanbaak: try ditching the "r" |
| 12:46.29 | mvanbaak | registrar server and proxy are both set to 'office.terrazur.nl' which resolve to 192.168.1.1 on our network |
| 12:46.48 | mvanbaak | it resolves to the outside of our setup when you are not on our internal network |
| 12:46.54 | mvanbaak | split-horizon dns setup |
| 12:47.09 | salvatore2 | what is the wW for? |
| 12:47.13 | salvatore2 | in your dialplan |
| 12:47.21 | mvanbaak | one touch monitoring |
| 12:47.38 | salvatore2 | ah ok |
| 12:47.43 | Sci_05 | salvatore2: are they any good (voicetrading.com)? |
| 12:47.45 | mvanbaak | [TK]D-Fender: ok, i'll try that |
| 12:47.54 | mvanbaak | but can that really be the problem ? |
| 12:47.58 | salvatore2 | Sci_05, not very cheap but very high quality |
| 12:48.01 | salvatore2 | even caller id |
| 12:49.13 | Sci_05 | hmmm alwasy looking for another backup carrier :) |
| 12:49.39 | Sci_05 | the prices seem to be good for the US at least |
| 12:52.41 | salvatore2 | yeah |
| 12:52.50 | salvatore2 | real good quality |
| 12:52.58 | salvatore2 | also icallglobe.com |
| 12:52.59 | salvatore2 | good quality |
| 12:53.56 | mvanbaak | hhmm |
| 12:54.08 | mvanbaak | maybe remove the outbound proxy from xlite config ? |
| 12:57.03 | [TK]D-Fender | mvanbaak: only need "domain" |
| 12:57.11 | *** join/#asterisk JulHer (n=julio@244.Red-217-125-14.staticIP.rima-tde.net) |
| 12:57.23 | [TK]D-Fender | mvanbaak: And set an IP if its fixed.... don't waste time on DNS |
| 12:57.31 | [TK]D-Fender | mvanbaak: a slow DNS could be a problem. |
| 12:58.23 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 12:58.30 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 12:58.32 | waKKu | folks.. where asterisk saves the recorded calls with automon ? |
| 13:00.52 | juuva | waKKu: /var/spool/asterisk/monitor ? |
| 13:01.05 | Zefk | hi, I'm looking for a good headset (Plantronics ?!) for Polycom IP 430 (for a contact center) ... can anyone advise me ? thx |
| 13:01.18 | creativx | Zefk: plantronics are good. |
| 13:01.33 | creativx | i like the voyager usb 510's |
| 13:01.35 | waKKu | juuva hmm... so isnt really working ;/ |
| 13:02.37 | Zefk | I'm looking for a Plantronics model that work very good with IP 430 |
| 13:02.44 | [TK]D-Fender | juuva: Plantronics M22 Amp + H261 Binaural Polaris quick-disconnect headset |
| 13:02.54 | [TK]D-Fender | Zefk: rather |
| 13:03.05 | creativx | is there something you dont know [TK]D-Fender? |
| 13:03.20 | [TK]D-Fender | creativx: Plenty |
| 13:03.28 | JT | i don't see how a usb headset will be at all useful for an ip phone ;) |
| 13:03.38 | [TK]D-Fender | JT : For the IP 650 ;) |
| 13:03.41 | juuva | waKKu: not sure, just guessing also /var/lib/asterisk/sounds is possible |
| 13:03.50 | JT | [TK]D-Fender: really? |
| 13:03.54 | creativx | JT: don't be difficult ;) |
| 13:03.58 | [TK]D-Fender | JT : maybe :) |
| 13:04.18 | [TK]D-Fender | juuva: No.... the monitor spool folder is where it should go. |
| 13:04.21 | Sci_05 | I have the playtronics 510s with a polycom 501 and blackberry and it works damn well |
| 13:04.37 | JT | Zefk: using PoE? |
| 13:05.01 | twitchnln | anybody got any idea why call quality would be degraded when using headset attached to mitel dual mode phone |
| 13:05.12 | Zefk | I saw that there are some Plantronics models that are unamplified ... that means the phone should have an amplif. Does IP 430 have the amplif for the headset ? |
| 13:05.19 | Zefk | Yes ... I will use poE |
| 13:05.28 | JT | Zefk: you need to get an amp |
| 13:05.36 | [TK]D-Fender | Zefk: You don't want an un-amped headset on a Polycom.... too damn wimpy... |
| 13:05.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:05.42 | *** join/#asterisk shinao1 (n=shinao1@41.205.186.49) |
| 13:06.06 | [TK]D-Fender | Zefk: the exact setup is what I'm running in my own call center on IP 600's |
| 13:06.19 | Zefk | And of course I need a price betwenn 100 and 150 USD |
| 13:06.41 | [TK]D-Fender | Zefk: You need to ditch your budget a bit on this one... |
| 13:07.15 | Sci_05 | arggggg stupid dlink!!!!! Always check the codecs on a dvg-1402s after you set it up if you can't make calls....its doesn't keep setting sometimes!! arg |
| 13:07.56 | JT | you can make calls with d-links? :o |
| 13:08.24 | waKKu | cool... automon works now :) |
| 13:08.57 | Sci_05 | ya I got one of thoes dvg 1402s (router with 2 sip ports), just had to configure the ports you know the CORRECT way and it seems to work just fine |
| 13:09.07 | waKKu | but.. how can i do to record it on mp3 format ? ? (automon) |
| 13:09.16 | JT | waKKu: you don't |
| 13:09.42 | Sci_05 | its the same thing that vontage uses only has the sip part of the router open to be able to configure to any server :) |
| 13:09.44 | waKKu | hm.. what u do ? crontab a convertion ? |
| 13:09.57 | mvanbaak | [TK]D-Fender: I cant set an ip |
| 13:10.06 | mvanbaak | [TK]D-Fender: asterisk box here is on our local lan |
| 13:10.16 | mvanbaak | but we work from various places all the time |
| 13:10.27 | mvanbaak | so when not in the office the ip should be our external ip |
| 13:10.45 | [TK]D-Fender | mvanbaak: Ah |
| 13:10.56 | mvanbaak | that's why I did this trick with a hostname that resolves to our public ip from the outside and to the internal ip here on our lan |
| 13:11.24 | mvanbaak | too bad they are not using linux |
| 13:11.33 | mvanbaak | there I could script /etc/hosts |
| 13:11.40 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:12.10 | [TK]D-Fender | mvanbaak: You could to an "on boot script that changes Windows HOSTS file |
| 13:12.49 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 13:13.59 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:14.52 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.134.53) |
| 13:24.26 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 13:27.48 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 13:28.04 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:29.14 | mocker | Good ol' sunrocket. |
| 13:30.03 | creativx | you have a sun rocket? |
| 13:30.49 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:32.27 | mocker | creativx: Luckily, no. |
| 13:33.53 | *** join/#asterisk Paul_UK (n=foo@email.seatwave.com) |
| 13:34.18 | Paul_UK | hey guys, there is an rpm for fedora which i can install fine on centos 5. but are there any issues with installing the latest tarball with centos 5? |
| 13:35.00 | waKKu | hm.. ppl... have some way to use hylafax without a modem card ? :) |
| 13:35.07 | mocker | Paul_UK: I always use source.. |
| 13:35.09 | mocker | waKKu: iaxmodem |
| 13:35.16 | mocker | Bam! |
| 13:35.25 | Paul_UK | mocker, you have installed centos with source fine? |
| 13:35.36 | waKKu | mocker hmmm.. let me check ;) |
| 13:35.41 | mocker | Paul_UK: I'm still on CentOS 4.4, but yeah.. |
| 13:35.58 | Paul_UK | mocker, ok you dont know off hand which dependencies asterisk needs? |
| 13:36.10 | mocker | Paul_UK: I never use packages for Asterisk (which is weird, because in general I'm pro package management) |
| 13:36.29 | jarod14 | <PROTECTED> |
| 13:36.32 | JT | iaxmodem + spandsp ( +libtiff) + hylafax |
| 13:37.20 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:37.31 | Paul_UK | mocker, well i've just installed centos 5 without any base or anyting lol, lets see if the tarball compiles lol |
| 13:37.44 | *** part/#asterisk michael-i (n=michael-@141.41.40.55) |
| 13:39.53 | waKKu | mocker dynamic or static libs ? what do advice me ? :) |
| 13:42.05 | mocker | Paul_UK: yum group install "Development Tools" |
| 13:42.06 | mocker | :) |
| 13:42.30 | mocker | waKKu: Umm, http://www.the-asterisk-book.com/unstable/ |
| 13:42.40 | mocker | See section 7 |
| 13:43.15 | Paul_UK | mocker, so i need all of them lol? |
| 13:43.24 | waKKu | oka |
| 13:43.30 | mocker | Paul_UK: No.. ;) |
| 13:43.38 | Paul_UK | mocker, well then i'd rather not do it |
| 13:43.48 | mocker | Paul_UK: Have fun. :) |
| 13:43.57 | Paul_UK | mocker haha i'll try lol |
| 13:44.08 | Paul_UK | but -devel doesnt have any place in a production machine |
| 13:45.05 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 13:45.08 | JT | eh |
| 13:45.13 | JT | compile asterisk |
| 13:45.25 | JT | don't use packaged, especially rpm |
| 13:45.30 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 13:45.33 | hi365 | hello. every time i try to register a client remotly i get "SIP/2.0 401 Unauthorized " |
| 13:45.34 | ZaVoid | good morning |
| 13:45.45 | mocker | hi365: fix password? |
| 13:45.46 | ZaVoid | anyone know if i have to do a start/stop if i change res_pgsql.conf |
| 13:45.57 | hi365 | mocker: double and triple checked |
| 13:46.29 | mocker | hi365: fix username? |
| 13:46.41 | hi365 | also double checked |
| 13:47.25 | Paul_UK | JT, why not to use rpm? |
| 13:49.05 | mocker | hi365: Weird.. |
| 13:49.09 | mocker | hi365: Just one user or all? |
| 13:49.20 | [TK]D-Fender | hi365: one of them is wrong. |
| 13:49.26 | [TK]D-Fender | hibecause * is not lying. |
| 13:49.31 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:49.43 | hi365 | mocker: anyone. i had the problem with a pap2. now im having the problem with a grandstream. |
| 13:50.00 | hi365 | i though upgrading asterisk might help, but 1.4.5 is giving me the same greif |
| 13:50.29 | mocker | hi365: Might try something simple like a SIP softphone first.. |
| 13:50.42 | hi365 | will do |
| 13:51.05 | mocker | hi365: You were easy to convince. :) |
| 13:51.47 | hi365 | mocker: would the qualify setting have anything to do with it? |
| 13:53.39 | *** join/#asterisk ESCulapio_ (n=elvyn@66.44.88.200.l.sta.codetel.net.do) |
| 13:53.58 | [TK]D-Fender | hi365: 401 = bad user/pas/unauthed IP, etc. |
| 13:54.38 | [TK]D-Fender | hi365: nothing to do with registration, qulify, NAT or anything else. Either the auth is wrong or the location/domain banned |
| 13:56.26 | hi365 | [TK]D-Fender: where do i set the autherised ip's? |
| 13:56.45 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 13:56.59 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:57.00 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:57.26 | [TK]D-Fender | hi365: usually you DON'T, you simply let anyone in. then again, you haven't shown us your configs or the failed attempt.... |
| 13:58.42 | *** join/#asterisk Op3r (n=op3r@125.212.122.209) |
| 13:59.11 | hi365 | mocker: same thing with a sip client |
| 13:59.22 | mocker | hi365: Probably time for pastebin |
| 13:59.24 | mocker | ~pastebin |
| 13:59.25 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 13:59.33 | hi365 | yup, getting it ready |
| 14:01.53 | *** join/#asterisk MadLooaaa (n=igor@motorola154-31.ip.PeterStar.net) |
| 14:03.00 | MadLooaaa | can somebody help me with configuring asterisk with http proxy? |
| 14:04.13 | MadLooaaa | hey.. please... |
| 14:04.15 | *** join/#asterisk UVSoft (n=UVSoft@motorola154-31.ip.PeterStar.net) |
| 14:04.25 | [TK]D-Fender | MadLooaaa: Asterisk has nothing to do with the "web" |
| 14:04.59 | MadLooaaa | I know |
| 14:05.33 | MadLooaaa | but our proxy server doesn't let packets go directly to the SIP provider |
| 14:05.56 | [TK]D-Fender | MadLooaaa: What "proxy"? |
| 14:06.04 | *** join/#asterisk kova (n=Koen@tech.quentris.com) |
| 14:06.21 | MadLooaaa | local proxy server |
| 14:06.22 | kova | anybody here uses chanskype? |
| 14:06.36 | [TK]D-Fender | MadLooaaa: ..meaning what exactly? |
| 14:06.37 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 14:06.44 | [TK]D-Fender | kova: Virtually no-one. |
| 14:07.22 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:07.42 | flujan | hi guys, I am trying to use realtime queues with asterisk |
| 14:07.50 | kova | no one? why not? |
| 14:08.30 | flujan | the problem lies in the monitor-format option. How can I set up it using realtime? I have try to create a monitor-format varchar column without success... |
| 14:08.40 | flujan | someone is using this aproach? |
| 14:08.49 | rene- | i tried to |
| 14:08.55 | rene- | flujan |
| 14:09.02 | rene- | i didnt ran into that |
| 14:09.13 | rene- | but yoy might want to setup that in the code that calls the queue |
| 14:09.19 | kova | flujan, can 't help you. only did iax realtime |
| 14:09.30 | rene- | Set(MonitorFormat=blah) in the dialplan |
| 14:09.39 | MadLooaaa | [TK]D-Fender: what do you mean? |
| 14:09.42 | [TK]D-Fender | MadLooaaa: Youa ren't registered so no private chat for you |
| 14:10.01 | [TK]D-Fender | MadLooaaa: What is this "proxy" you are using, what is the problem exactly. |
| 14:10.09 | UVSoft | [TK]D-Fender: I suppose MadLooaaa means that he has a proxy server and wants to connect to a remote SIP provider through it.... |
| 14:10.23 | MadLooaaa | yeah.. ) |
| 14:10.27 | UVSoft | but doesnt know howto) |
| 14:10.40 | *** join/#asterisk lsodi (n=lsodi@195.80.124.193) |
| 14:11.25 | [TK]D-Fender | MadLooaaa: Why proxy between * and this provider? |
| 14:11.56 | MadLooaaa | the other traffic is automatically blocked by our firewall |
| 14:12.14 | MadLooaaa | so I should use proxy |
| 14:12.16 | [TK]D-Fender | MadLooaaa: You aren't registered so no private chat for you <------------------ |
| 14:12.37 | flujan | rene-: I can set up it before the Queue command? |
| 14:12.41 | [TK]D-Fender | MadLooaaa: you have no control over your firewall? |
| 14:12.42 | UVSoft | hey i'm interested in * through proxy too |
| 14:12.58 | rene- | are you using local channels? if you are it is best before the Dial() |
| 14:13.03 | [TK]D-Fender | UVSoft: Considering you're both at the same HOST, I'm not surprise.... |
| 14:13.05 | UVSoft | could anyone explane me howto make asterisk use it |
| 14:13.11 | rene- | if not then you can probably use it before Queue |
| 14:13.30 | rene- | i am not sure if that would work. can you probably try it? |
| 14:13.36 | [TK]D-Fender | -->|MadLooaaa (n=igor@motorola154-31.ip.PeterStar.net) has joined #asterisk |
| 14:13.37 | [TK]D-Fender | -->|UVSoft (n=UVSoft@motorola154-31.ip.PeterStar.net) has joined #asterisk |
| 14:13.38 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:13.47 | UVSoft | )) |
| 14:13.49 | UVSoft | yep |
| 14:14.07 | UVSoft | and i am registered) |
| 14:14.29 | kova | if no one uses chanskype, is there any other solution to connect ast to skype? |
| 14:14.34 | [TK]D-Fender | UVSoft: use /msg chat, not DCC |
| 14:14.46 | flujan | rene-: ok will give it a try. |
| 14:14.47 | [TK]D-Fender | kova: Nothing free, nor practical. |
| 14:15.10 | rene- | if you are using Local channels as agents it surely would work |
| 14:15.29 | kova | ok, what about connecting to gtalk? is that working for you guys? |
| 14:15.36 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.59 | Paul_UK | do i assume to compile asterisk, i need the zaptel, libpri and addons? |
| 14:16.56 | kova | Paul_UK, you don't need these necessarily |
| 14:17.13 | kova | depends on what you want to do |
| 14:17.42 | [TK]D-Fender | kova: Why don't you use SIP phones like everyone else? |
| 14:17.58 | Paul_UK | kova: I only ask because I get this checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no and configure: error: *** termcap support not found |
| 14:17.59 | [TK]D-Fender | kova: instead of trying to hack in proprietary solutions one after the other... |
| 14:18.10 | kova | I do always add zaptel for the zt_dummy device, which can be needed for timing (e.g when you use conferencing) |
| 14:18.13 | [TK]D-Fender | Paul_UK: You're missing libtermcap & devel |
| 14:19.15 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:20.10 | kova | [TK]D-Fender, gtalk is not proprietary, but based on jabber and xmpp |
| 14:20.31 | Qwell[] | it's just jabber+jingle, really |
| 14:20.46 | kova | [TK]D-Fender, I agree skype is, unfortunately |
| 14:20.46 | [TK]D-Fender | kova: it can be made to work, but you're switching from asing one kind of "troubel" to another... |
| 14:20.55 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:20.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:22.08 | hi365 | mocker: [TK]D-Fender: here is the pasbin of everything sip: http://pastebin.ca/623300 |
| 14:22.25 | *** join/#asterisk drgalaxy (n=drgalaxy@adsl-70-238-195-120.dsl.lbcktx.sbcglobal.net) |
| 14:23.13 | kova | so no one uses chan_gtalk? hard to believe all that development is a waste of time ... |
| 14:23.30 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:26.23 | mocker | hi365: Try changing your username from Pesi to 201 or something like that.. |
| 14:26.50 | hi365 | mocker: Pesi is the freepbx extneion name. i was ONLY using numbers |
| 14:28.24 | *** part/#asterisk drgalaxy (n=drgalaxy@adsl-70-238-195-120.dsl.lbcktx.sbcglobal.net) |
| 14:28.36 | mocker | hi365: What's the username you put into your softphone? |
| 14:29.10 | hi365 | mocker: im back on the hard phone. let me just pastbin the configs |
| 14:30.36 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 14:30.45 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 14:32.39 | hi365 | mocker: http://www.imagehosting.com/show.php/916530_2000.JPG.html |
| 14:33.50 | mocker | Have you tried putting the username into the Authenticate ID as well? |
| 14:33.56 | mocker | haven't played w/ grandstreams.. |
| 14:37.34 | *** join/#asterisk FlatFoot (i=FlatFoot@80.88.192.83) |
| 14:38.03 | FlatFoot | can anyone help with a make res_config_mysql.so error please ? |
| 14:38.22 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 14:38.22 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:39.09 | Mercestes | ~gs |
| 14:39.10 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:39.27 | Mercestes | FlatFoot: What distro and what v.? |
| 14:39.33 | *** join/#asterisk menil (n=meni@212.179.153.130) |
| 14:39.54 | FlatFoot | running 1.0.11.1 on debian |
| 14:40.14 | Qwell[] | FlatFoot: upgrade to something released in the last 2 years first |
| 14:40.18 | Mercestes | Debian is special. |
| 14:40.26 | FlatFoot | special ?????? |
| 14:40.28 | tzafrir | FlatFoot, what's the problem? |
| 14:40.42 | FlatFoot | i am trying to install addons for MySql cdr |
| 14:41.07 | tzafrir | pastebin the trace |
| 14:41.17 | tzafrir | Which Debian, BTW? |
| 14:41.22 | tzafrir | Sarge or Etch? |
| 14:41.28 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 14:41.54 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:41.55 | anonymouz666 | is it possible to Dial(SIP/blah) and at same time calls a Queue(duh) |
| 14:41.56 | FlatFoot | Sarge |
| 14:41.59 | l2cache | Does anyone know why when I upgraded asterisk 1.2 to 1.4 the Master.csv file is no longer being updated?? |
| 14:42.03 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:42.12 | Qwell[] | l2cache: Did you enable cdr_csv? |
| 14:42.24 | l2cache | Yes, I believe so |
| 14:42.38 | anonymouz666 | two dials at the same time.... one calling a peer and another executing a queue() |
| 14:43.07 | tzafrir | l2cache, is the module cdr_csv.so loaded? cdr_custom.so ? |
| 14:43.13 | FlatFoot | tzafrir: http://www.pastebin.ca/623334 |
| 14:43.25 | hi365 | mocker: didnt help. i had the same issues with a pap2. its a really bummer not to be able to use asterisk over the internet |
| 14:43.42 | tzafrir | FlatFoot, apt-get install build-essential |
| 14:43.47 | l2cache | checking.... :) |
| 14:44.01 | tzafrir | FlatFoot, This will get you gcc and co. |
| 14:44.27 | tzafrir | FlatFoot, while you're at it: apt-get install libmysqlclient-dev |
| 14:44.28 | FlatFoot | tzafrir, ok i'm doing that now |
| 14:44.51 | FlatFoot | i did the libmy.... but not the dev version |
| 14:44.57 | twitchnln | any idea how a 300mhz geode running gentoo would do with asterisk? |
| 14:45.19 | tzafrir | twitchnln, If you don't put too many channels on it |
| 14:45.20 | l2cache | my modules.conf file is the same on both of my servers. One running 1.2 and the other running 1.4 Should I check for the modules in /usr/lib/asterisk/modules? |
| 14:45.39 | tzafrir | l2cache, show modules |
| 14:45.46 | l2cache | ahhh.. txh |
| 14:45.54 | tzafrir | and right, also the modules directory |
| 14:46.12 | twitchnln | tzafrir: was thinking that it would have 3 trunks (vonage, vitelity, and pstn) and 4 sip phones |
| 14:46.31 | l2cache | it is in the show modules listing |
| 14:47.15 | *** join/#asterisk jivesuperfresh (n=jivesupe@pool-71-163-173-14.washdc.fios.verizon.net) |
| 14:47.41 | tzafrir | twitchnln, hmmm... a compressed codec may be an issues. some 6 or so concurrent calls (uncompressed) might work fine. But I'm just throwing numbers of the top of my head. |
| 14:48.00 | l2cache | to do the upgrade, I cleared out the modules directory, and recompiled. Should I start a new Master.csv file? |
| 14:48.30 | twitchnln | tzafrir: i was thinking that i disallow all codecs and then allow g729, so it shouldn't have any transcoding issues |
| 14:48.30 | Paul_UK | hey guys, just out of interest, why would i need these for asterisk? : newt newt-devel ncurses ncurses-devel bison openssl-devel |
| 14:49.25 | hi365 | anyone know why i would be getting "SIP/2.0 401 Unauthorized" for a remote extensions? |
| 14:49.28 | tzafrir | l2cache, no. No need for a new file. Make sure that the file and directory are writable to Asterisk |
| 14:49.43 | [TK]D-Fender | hi365: We told you the reason, you've masked too much and are not listening. |
| 14:50.13 | FlatFoot | tzafrir, that's all done but now i get a huge output with error's |
| 14:50.42 | tzafrir | Paul_UK, newt-devel: for astman . ncurses: to support the CLI (editline) IIRC. Or is it menuselect? |
| 14:50.50 | [TK]D-Fender | Paul_UK: menuselect uses a bunch, ssl I think is because of IAX's encryption |
| 14:50.52 | Qwell[] | menuselect |
| 14:51.00 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:51.12 | [TK]D-Fender | Wow.... I wawn't far off the mark! |
| 14:51.18 | [TK]D-Fender | wasn't* |
| 14:51.18 | FlatFoot | tzafrir, http://www.pastebin.ca/623346 |
| 14:51.20 | tzafrir | bison is used for some parsing codes |
| 14:51.47 | tzafrir | res_config_mysql.c:41:30: error: asterisk/channel.h: No such file or directory |
| 14:52.09 | tzafrir | FlatFoot, do you have asterisk installed from source? Or asterisk-dev package? |
| 14:52.26 | FlatFoot | installed from a xorcom rapid |
| 14:52.34 | Paul_UK | ok guys, thanks for the help, i'll just install it for the hell of it, so i dont have issues.. but seems i've got what i need to install it |
| 14:52.35 | hi365 | [TK]D-Fender: no, you gave two posibilites. you said "Either the auth is wrong or the location/domain banned" I double ant tripple checked the user/pass. so its not that. I dont know of any setting to allow/deny remote clients from connecting. If you dont have anything helpful to add, why dont you just shut up? |
| 14:52.45 | tzafrir | FlatFoot, so why not just install asterisk-mysql ? |
| 14:52.54 | tzafrir | apt-get install asterisk-mysql |
| 14:53.10 | FlatFoot | tzafrir thats the kind of thing i've been searching for |
| 14:53.18 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:53.39 | [TK]D-Fender | hi365: you obliterated the authname/ip coming in. If it was right, it would be working. we are trying to help and you are being less than cooperative. |
| 14:53.53 | FlatFoot | tzafrir, how bloomin easy was that . What a PLANK i am . Cheers for the help |
| 14:54.05 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:54.55 | ZaVoid | anyone know if i can add secondary data source in res_pgsql.conf ? |
| 14:55.05 | hi365 | [TK]D-Fender: so your only motive is to get other peoples username's and passwords? Is THAT why your here? why would you need to see all my secerets? How would that help? |
| 14:55.36 | hi365 | [TK]D-Fender: and please, speak for your self/ your not the spokesman here. no need to use the word "we" when your refering to your self |
| 14:55.38 | lsodi | greetings, I'm quite new to asterisk so pleas dont hit hard if I ask too stupid questions. |
| 14:56.09 | Juggie | hi365, [TK]D-Fender, has been a member of the * community for a long time, and has helped people in #asterisk without ever asking for anything in return for a long time |
| 14:56.21 | Juggie | he does not care about your sip/iax accounts, and is only trying to help you |
| 14:56.47 | ZaVoid | i agree [TK]D-Fender is a great guy! |
| 14:56.59 | ZaVoid | so fender any ideas about multiple entries in res_pgsql.conf ? |
| 14:57.14 | lsodi | in zaptel.conf to define span, first value is span num, is this value for oredr,like: sapan1, span2 etc? |
| 14:57.28 | hi365 | Juggie: im sure, but being rude and insulting isnt going to help anyone. If i done understand something, there is nothing wrong with asking again. and there is no need to get all personal if someone didnt get whay you said the first time around! |
| 14:57.46 | hi365 | why cant the room just keep thing imple and freindly? |
| 14:57.52 | hi365 | imple=simple |
| 14:57.54 | Juggie | ZaVoid, i dont think it supports fall over no |
| 14:58.07 | ZaVoid | damnation |
| 14:58.16 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 14:58.24 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 14:58.38 | [TK]D-Fender | hi365: I asked to see the SIP debug because that would probably be quite clear as to where the error is and you're making it all personal and thinking I care about your passwords, etc, not like they aren't encoded anyways. |
| 14:59.26 | [TK]D-Fender | hi365: So if you want to go on being paranoid and thinking we're out to get you, go right ahead, plenty of other people to help around here. |
| 14:59.26 | Juggie | yeah, there are no clear text passwords in the sip debug |
| 14:59.36 | hi365 | [TK]D-Fender: all i did was remove the ip addresses form the debug and replace them with placeholders. no biggi there, UNLESS your ego is the only thing on the line |
| 15:00.03 | [TK]D-Fender | hi365: You seem to have all the answers. I cannot help you. You know it all already. |
| 15:00.12 | hi365 | btw, its called being responsible. it has nothing to do with being paranoid |
| 15:01.07 | hi365 | of course not. your not interested in helpong. you gave me two posibilites befor. neither of them solved the problem. am i not entitled to ask other people. PERHAPS someone know something that you dont? |
| 15:01.08 | Paul_UK | hi365, you arent alone, they are out to get ME too! |
| 15:01.31 | l2cache | I still cannot get the asterisk 1.4 upgraded server to log the cdrs to /var/log/asterisk/cdr-csv/Master.csv I have the cdr-csv Module. And 'show modules' shows the cdr_csv.so What else is there to check? |
| 15:02.03 | *** join/#asterisk af_ (n=getsmart@81-174-44-88.dynamic.ngi.it) |
| 15:02.18 | hi365 | Paul_UK its not a matter of "get me". i asked a question and got a posible answer. when that didnt lead anywhere, i re-asked. maybe someone else is more knowladgeable |
| 15:03.19 | Juggie | you must be under the impression that we are paid support |
| 15:03.25 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:06.18 | tzanger | Juggie: hahaha |
| 15:06.57 | codefreeze | l2cache: is it logging, then, to cdr-custom? |
| 15:06.58 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 15:07.39 | *** join/#asterisk yannj_fr (n=yannj@82.227.103.140) |
| 15:08.13 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 15:09.07 | l2cache | it is logging to cdr.dat in cdr-custom...not the cdr-csv/Master.csv file though |
| 15:10.23 | Paul_UK | anyone here use usb headsets with asterisk? |
| 15:10.39 | Paul_UK | im trying to find a decent one, for use in a call centre |
| 15:10.52 | [TK]D-Fender | Paul_UK: Taht somewhat implies soft-phones, and they work... not much more to say |
| 15:10.57 | Paul_UK | theres nothing in plantronics that has both ears as it were, only 1 |
| 15:11.06 | Paul_UK | ok sorry wasnt clear.. usb and both ears then lol |
| 15:11.20 | [TK]D-Fender | Paul_UK: Couldn't say any of them as being "nice", definately not something I'd want to do to a person tethered to a phone as their job.... |
| 15:11.24 | Paul_UK | seeing as i got trixbox (i know swear word) working with both sjphone and x-lite |
| 15:11.26 | twitchnln | Paul_UK: I found that plantronics usb work well... |
| 15:11.57 | [TK]D-Fender | Paul_UK: is USB really a requirement? |
| 15:12.21 | cpm | too much lag in the plantronics usb |
| 15:12.29 | cpm | sounds really good though. |
| 15:12.35 | tzafrir | USB is a plus ;-) |
| 15:12.49 | *** join/#asterisk JulHer (n=julio@244.Red-217-125-14.staticIP.rima-tde.net) |
| 15:12.51 | Juggie | my plantronics usb headsets are all great. |
| 15:12.55 | Juggie | i have one on right now :) |
| 15:13.22 | twitchnln | Paul_UK: I've also had good luck with logitech usb headsets |
| 15:13.38 | Paul_UK | twitchln, but the plantronics that i see now, only have 1 ear piece and not 2 :( |
| 15:13.43 | l2cache | any ideas for my logging issue? this has broken my reporting site |
| 15:13.48 | Juggie | Paul_UK, http://www.plantronics.com/north_america/en_US/products/cat640035/cat1430032/prod440044 |
| 15:13.51 | l2cache | i need the Master.csv |
| 15:13.57 | Juggie | i have 2 of those, they are good. |
| 15:14.24 | Juggie | the software is optional too, which means i dont use it. |
| 15:14.25 | Paul_UK | Juggie, i have the DSP-400 and its great |
| 15:14.34 | twitchnln | Paul_UK: are your calls coming in stereo? |
| 15:14.38 | Paul_UK | but was trying to find something more lightweight |
| 15:15.25 | Juggie | http://www.plantronics.com/north_america/en_US/products/cat1200043/cat380046 |
| 15:15.28 | Juggie | they have a ton fo them |
| 15:15.52 | l2cache | the calls would have to come in dual-mono |
| 15:16.04 | Juggie | l2cache, yes, but it helps agents hear in a call center |
| 15:16.17 | creativx | the mp3s you can listen to are in stereo atleast |
| 15:16.38 | l2cache | I think they're awesome. But asterisk wont send that calls in stereo to them.. How is the delay with them? |
| 15:16.52 | Juggie | i cant speak for every plantronics but mine is fine. |
| 15:17.04 | l2cache | how much for the one you have? |
| 15:17.50 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-112-25.w86-211.abo.wanadoo.fr) |
| 15:19.15 | Juggie | l2cache, i dont know, work bought it |
| 15:20.04 | mocker | Qwell[]: Sucks, giving VoIP even more of a bad name. |
| 15:20.09 | l2cache | so the cdr.dat file in cdr-custom is logging... all of my set(CDR(accountcode)=files) that were working in 1.2 are no longer working along with Master.csv |
| 15:22.23 | codefreeze | l2cache: cdr.dat? By default, it should be Master.csv; cdr-custom/Master.csv to be exact....? |
| 15:22.49 | l2cache | by default it is ... cdr-custom/cdr.dat and cdr-csv/Master.csv |
| 15:23.14 | l2cache | the cdr.dat is not default...but the Master.csv has always been in cdr-csv |
| 15:23.44 | codefreeze | l2cache: First, run a make menuselect, and see if all your backends are recognized. There could be a problem there. |
| 15:24.06 | codefreeze | l2cache: then, check your config files; there may have been changes since 1.2. |
| 15:24.29 | l2cache | run that in my extracted asterisk-1.4 directory in /usr/src? |
| 15:24.54 | codefreeze | l2cache: Look at the console logs; see if there's any ERROR or warnings when the modules were loaded. |
| 15:25.12 | codefreeze | l2cache: Yes, if that's where you compiled your asterisk |
| 15:25.14 | *** part/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 15:26.10 | codefreeze | NirS: Stranging a programmer sounds pretty.... weird... |
| 15:26.27 | codefreeze | NirS: sounds pretty.... strange! |
| 15:26.29 | creativx | still sounds better than familarizing one |
| 15:26.30 | NirS | I meant strangling a programmer, it came out wrong |
| 15:26.47 | codefreeze | NirS: :) |
| 15:26.53 | l2cache | should there be any issues running that make menuselect in a production box? |
| 15:27.33 | codefreeze | No; but really! you shouldn't be building on a production box at all! |
| 15:27.55 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:27.55 | l2cache | this is the one I upgraded recently...and now the loggin isnt working |
| 15:28.09 | l2cache | oops.. |
| 15:28.15 | l2cache | loggin* |
| 15:28.16 | Mercestes | Nirs: coming out is always wrong. |
| 15:28.18 | Mercestes | >.> |
| 15:28.24 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:28.25 | l2cache | logging* |
| 15:28.42 | l2cache | so I will be fine running that? |
| 15:31.00 | Mercestes | qwell[]: I think I will be purchasing today since my trial runs out...today |
| 15:31.44 | l2cache | code? |
| 15:34.22 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 15:35.26 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 15:36.31 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 15:36.41 | *** join/#asterisk drgalaxy (n=drgalaxy@adsl-70-238-195-120.dsl.lbcktx.sbcglobal.net) |
| 15:36.44 | Paul_UK | Juggie, do ths calls come in dual-mono? |
| 15:37.03 | Juggie | i'm not sure what you mean |
| 15:37.10 | Paul_UK | Juggie, sound in both ears |
| 15:37.12 | Juggie | if your using a softphone and a usbheadset, of course they do |
| 15:37.31 | Paul_UK | Juggie, ok, heh, just getting paranoid to what l2cache was saying |
| 15:37.32 | [TK]D-Fender | Paul_UK: No, they should be dual stereo, its jsut the softphone that should mirror the output |
| 15:37.46 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 15:37.46 | [TK]D-Fender | Paul_UK: meaning if you use it for msuic, etc outside of that it should be stereo. |
| 15:38.55 | l2cache | I said it would be in both ears. but thats not stereo....dual mono |
| 15:39.24 | Paul_UK | l2cache, doesnt matter, as long as they can hear it lol |
| 15:39.46 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 15:40.28 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:40.53 | Paul_UK | oh and i've just installed asterisk lol |
| 15:42.27 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 15:46.49 | *** join/#asterisk pourriture (n=pourritu@mail.cshorecomputing.com) |
| 15:47.19 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 15:48.11 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 15:48.47 | coolbeans | Hi all. Is there a sip.conf setting that allows a peer to receive a second call when it's being used? For some reason, a bunch of our peers just send back a "Sip busy" message instead of taking the call. (Polycom/Aastra phones). It's been working but we just went to db driven sip.conf (asterisk static realtime) and I'm thinking we missed a setting. |
| 15:49.23 | creativx | call-limit |
| 15:49.35 | creativx | is the magic keyword |
| 15:50.32 | coolbeans | It's set to default "0" now. I've never set it before and it's worked?? Should I set it to a higher number? |
| 15:50.46 | creativx | what version asterisk |
| 15:50.46 | coolbeans | It's not even in the sip.conf peer entry at all. |
| 15:50.48 | coolbeans | 1.2.18 |
| 15:50.55 | pourriture | I am a debian user interested in learning about asterisk ... my usual approach with servers is to use the debian packages for everything except the primary service the box is to offer and then I build that from the latest sources. I see, however, that asterisk.org mentions the debian etch package and the debain VOIP team. Is the package a good way to go, or will I miss all the fun? |
| 15:51.22 | creativx | coolbeans: im not sure when it changed from incominglimit to call-limit |
| 15:51.28 | creativx | but i recall it working on my 1.2.10 box |
| 15:51.55 | creativx | that is, it might have worked, but it killed all the extensionstatus events for that phone |
| 15:52.05 | coolbeans | Could it be progressinband? |
| 15:52.08 | Mercestes | coolbeans: Is there a call perline limit in your sip.cfg or phone.cfg on the polycoms? |
| 15:52.18 | creativx | which made people here a bit angry because nobody knew what was happening on their phones |
| 15:52.40 | [TK]D-Fender | pourriture: Screw packaging for *. just download the source and save yourself a lot of greif. |
| 15:52.51 | [TK]D-Fender | pourriture: All the pre-req's are listed on asterisk.org |
| 15:53.27 | pourriture | [TK]D-Fender: that was my gut instinct ... I was suprised to see the packages mentioned so prominently, that is why I asked |
| 15:53.31 | coolbeans | Mercestes: Nope. hrm.... |
| 15:54.10 | [TK]D-Fender | pourriture: * isn't something you want accidentally "upgraded".... things can go bad |
| 15:54.18 | coolbeans | It's working fine on the polycom's, it seems to just be the aastra's, but nothing has changed except going from flatfile sip.conf to a mysql db. |
| 15:54.21 | Mercestes | coolbeans: "Sip busy" sounds like a polycom response...not a sip.conf setting |
| 15:54.34 | Paul_UK | hey guys, im gonna look in the forums, but since there are knowledgable peps here :) What options do I have with asterisk and a gui? like druid and freepbx ? |
| 15:54.43 | Mercestes | Then it sounds like Aastra is literally responding with "leave me alone, I'm busy." |
| 15:54.52 | [TK]D-Fender | coolbeans: pastebin your configs, and the full cli output of the call with SIP debug & verbose 10 |
| 15:54.55 | Paul_UK | i see that freepbx wont work with 1.4 until aug |
| 15:55.13 | pourriture | [TK]D-Fender: thanks for the advice .... I am sure I will have questions for you in about 2 days when I can form an intelligent one :) |
| 15:55.27 | jsbach | how can i turn off the 407 response, everytime i send an invite? |
| 15:55.38 | [TK]D-Fender | pourriture: we'll be around |
| 15:56.09 | coolbeans | This is what the console reports: Got SIP response 486 "Busy Here" back from <ip address> |
| 15:56.23 | jsbach | [TK]D-Fender, any ideas there ? |
| 15:56.59 | Sci_05 | pourriture: I would still from source, I run all debian and never install their packages for asterisk or asterisk additions. I do do the apt-get build-source asterisk to get all the packages so everything will build correctly but after that install asterisk and zaptel from source |
| 15:57.02 | [TK]D-Fender | coolbeans: sounds like the phone isn't accepting more channels. |
| 15:57.30 | creativx | licencing on phone? |
| 15:57.34 | creativx | call waiting disabled? |
| 15:57.59 | coolbeans | [TK]D-Fender: yep, my assumption as well but the phone configs haven't changed in months. hrm.... |
| 15:58.07 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 15:58.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:00.43 | tzafrir | Sci_05, svn-buildpackage ... |
| 16:01.27 | Paul_UK | hmm, let me ask a different way :) has anyone used freepbx, asterisknow and druid ? |
| 16:02.03 | pourriture | Sci_05: doesn't that leave you several versions back ... like 1.2.13 for etch? |
| 16:02.42 | Sci_05 | damn didn't have coffee yet....I ment apt-get build-dep asterisk, then build and install asterisk and zaptel from source |
| 16:02.57 | jeremy_g | so sexy boys whats up |
| 16:03.12 | jeremy_g | how cute are your incoming calls |
| 16:03.23 | pourriture | gotcha |
| 16:03.47 | Sci_05 | ya if you did a build-source it would probably old and nasty |
| 16:04.01 | coolbeans | Aha!!!! Looks like the AAstra phone(s) can only accept one (1) g729 call at a time. |
| 16:04.09 | coolbeans | Changed to ulaw, all is well. |
| 16:04.48 | tzafrir | Sci_05, hence my suggestion to use the up-to-date package from pkg-voip (rebuild it yourself with svn-buildpackage, of course) |
| 16:05.23 | pourriture | Sci_05: I like that lots ... wife is going ga-ga over apt-get build-dep too |
| 16:06.07 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 16:06.19 | *** join/#asterisk techie (n=techie@ppp-69-228-12-207.dsl.frsn01.pacbell.net) |
| 16:06.21 | [TK]D-Fender | coolbeans: Which model(s)? |
| 16:06.31 | Sci_05 | I have had it where building the package doesn't come out just right and messes things up, that and I guess I am just stuck in my ways...if its important to the system install from source |
| 16:07.19 | coolbeans | [TK]D-Fender: 410's |
| 16:08.31 | [TK]D-Fender | coolbeans: Never head of then.... |
| 16:08.35 | [TK]D-Fender | heard* |
| 16:08.53 | [TK]D-Fender | them* |
| 16:08.55 | [TK]D-Fender | kjshdasdlsdfyasyigfduiofdgkhasgd |
| 16:14.22 | coolbeans | Sorry, 480i's |
| 16:14.34 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 16:16.15 | twitchnln | anybody got a simple script i can run as a cron job to email me cdr's? |
| 16:16.25 | Strom_C | ~cohujibuggle |
| 16:16.25 | jbot | cohujibuggle is, like, gublgubbglggugglbuglgbugblgbgbgbgbglbglgbulgblugbgubgublgbglulllbgbb |
| 16:17.01 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:19.39 | drgalaxy | any recommendations for a 2fxo/2fxs SIP gateway for a remote office connecting to my main asterisk box? I need to have it failover to local lines if the 'net is down |
| 16:21.19 | [TK]D-Fender | drgalaxy: If you're talking failover, I might suggest just 2x SPA-3102's |
| 16:21.22 | nDuff | I just upgraded my production system from Asterisk 1.4.1 to 1.4.7.1, along with upgrading to zaptel to 1.4.3 and libpri to 1.4.1. I'm now having sporadic cases of extremely poor line quality on outgoing calls (going through a PRI via a Sangoma card with wanpipe 3.1.2.p7). Any ideas as to what may be causing this? |
| 16:22.01 | rene- | nDuff: was it happening with the prev version? |
| 16:22.06 | nDuff | rene-: no. |
| 16:22.18 | rene- | can you go back? |
| 16:22.23 | drgalaxy | [TK]D-Fender: thanks, I'll take a look. do those work well with asterisk? |
| 16:22.25 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 16:22.31 | [TK]D-Fender | drgalaxy: yup |
| 16:22.52 | nDuff | rene-: not trivially. this is gentoo, and the overlay I was getting the ebuilds from doesn't carry the old versions anymore. |
| 16:23.06 | rene- | damn |
| 16:23.23 | rene- | dunno about gentoo |
| 16:23.46 | rene- | i am getting zaptel driver lockups with the latest asterisk/zaptel |
| 16:24.02 | rene- | when (re)loading the driver |
| 16:24.09 | rene- | and those are nasty since they take down the whole machine |
| 16:29.37 | tzafrir | rene-, with what driver? |
| 16:29.47 | rene- | 1.4.3 |
| 16:29.49 | tzafrir | and what kernel? What distro? |
| 16:30.29 | tzafrir | driver: that is - what card? |
| 16:30.34 | rene- | centos 4.4 kernel 2.6.9-42 |
| 16:30.45 | rene- | wct4xxp with echo cancel |
| 16:31.18 | rene- | it doesny have tjhe ztcfg -s that locked the machine before |
| 16:31.27 | tzafrir | if you unload is there a problem? if you load, is there a problem? |
| 16:31.35 | tzafrir | also: can you try zaptel 1.4.4? |
| 16:31.37 | rene- | problem is at unload load |
| 16:31.38 | *** join/#asterisk honeybeebuzz (n=admin@206-248-138-47.dsl.teksavvy.com) |
| 16:31.48 | rene- | when i start the machine and only do a load everthing is ok |
| 16:32.00 | rene- | sure, i will upgrade today |
| 16:32.06 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 16:32.19 | rene- | since i have some other issues with the box i have a cron script that start |
| 16:32.37 | rene- | that restart the asterisk app and the zap drivers every morning |
| 16:33.43 | tzafrir | Though I can't think of a specific issue that should have required that upgrade |
| 16:34.32 | rene- | it started when i moved to 1.4.7.1 |
| 16:34.40 | rene- | wasnt doing it when using 1.4.7 |
| 16:34.46 | rene- | or earlier versions |
| 16:35.34 | *** part/#asterisk honeybeebuzz (n=admin@206-248-138-47.dsl.teksavvy.com) |
| 16:35.41 | jsbach | i have two users registered on my asterisk machine, whenever A calls B, A gets a "404 Not Found" |
| 16:36.48 | jeremy_g | ****tip of the day!one way not to screw your current asterisk install****" |
| 16:36.51 | jeremy_g | do not download asterisk-addons and make && make install on a running system, the modules it install may screw up the asterisk when you do the next reload" |
| 16:37.48 | codefreeze | l2cache: sorry for the delay. Yes! you can run "make menuselect" without harming your env. It sets up background vars for your next make. It'll tell you what backends are available, etc. |
| 16:41.46 | *** join/#asterisk NirS (i=Nir@87.68.60.4.cable.012.net.il) |
| 16:41.50 | l2cache | its ok...i ran it and it showed up cdr-csv just fine |
| 16:41.53 | l2cache | I am lost on this |
| 16:42.21 | [TK]D-Fender | jsbach: pastebin the failed call attempt at verbose 10 & SIP debug enabled |
| 16:42.24 | [TK]D-Fender | ~pb |
| 16:42.24 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 16:42.45 | codefreeze | l2cache: what CDR backends are you using? Just the cdr-csv? no DB's? |
| 16:43.01 | l2cache | no DBs...correct |
| 16:43.38 | jsbach | [TK]D-Fender, ok, i am doing it.. |
| 16:45.57 | jsbach | [TK]D-Fender, do you want to also have an ngrep? |
| 16:46.03 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 16:46.18 | codefreeze | OK, l2cache: check cdr.conf, did you uncomment the [csv] category? Do you even have a [csv] category? |
| 16:46.27 | *** part/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 16:46.39 | jsbach | [TK]D-Fender, http://pastebin.ca/623456 |
| 16:46.41 | [TK]D-Fender | jsbach: Shouldn't need |
| 16:46.57 | l2cache | lol...just [general] no other context |
| 16:47.01 | l2cache | i think thats it right? |
| 16:47.10 | [TK]D-Fender | Looking for bob in incoming (domain semiconductor.jsbach) |
| 16:47.11 | [TK]D-Fender | Reliably Transmitting (no NAT) to 10.147.67.130:1176: |
| 16:47.13 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 16:47.40 | codefreeze | l2cache: you can compare with the configs/example.cdr.conf file, or whatever it's called... |
| 16:47.42 | [TK]D-Fender | jsbach: You have no "exten => bob,1....." in [incoming]. this is purely a dialplan error |
| 16:48.21 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 16:48.26 | JerJer | does anyone happen to have a SunRocket ATA laying around ? |
| 16:48.31 | *** join/#asterisk nirz (i=nir@bzq-88-152-101-90.red.bezeqint.net) |
| 16:48.50 | JerJer | i am blogging how to hijack the sunrocket DNS to repoint to another IP address |
| 16:49.07 | jsbach | [TK]D-Fender, i have a dialplan which are identical for both users, [incoming] s,1,Answer s,2,Play(hello-world) s,3,Hangup() |
| 16:49.36 | [TK]D-Fender | jsbach: You completely misunderstand the purpose of "s". that does NOT apply to SIP calls as some sort of "catch-all" |
| 16:49.42 | [TK]D-Fender | ~stdextens |
| 16:49.43 | jbot | "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 16:49.43 | jsbach | i just want to see the interaction with the asterisk pbx first |
| 16:49.59 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 16:50.02 | [TK]D-Fender | jsbach: You are dialing "bob" and you don't HAVE a "bob". |
| 16:50.12 | jsbach | [TK]D-Fender, i see .. the prob is, i just got that from the book of (Jim Van Meggelen).. |
| 16:50.31 | [TK]D-Fender | jsbach: "s" has nothing to do with TARGETED extens. |
| 16:50.32 | Hmmhesays | I writing a dialplan that will dial an a user and playback for them who is calling |
| 16:50.54 | Hmmhesays | and allow them to transfer these people to different extensions based on their callerid |
| 16:51.02 | [TK]D-Fender | jsbach: next I highly recommend you don't consider dialing NAMES as extensions. NUMBER them. |
| 16:51.13 | jsbach | ok, so i should have an extension like exten => bob, 1, Answer() bob,2,Play(hello-world), bob,3,hangup() ? |
| 16:51.37 | jsbach | [TK]D-Fender, can you give an example? |
| 16:51.42 | [TK]D-Fender | jsbach: yes, that would WORK, but I HIGHLY recommend you NUMBER the extens so you can dial things from a NORMAL phone. |
| 16:51.54 | l2cache | codefreeze: thanks! I cant believe my boss copied all the configs directly over, overwriting the 1.4 examples. |
| 16:52.10 | [TK]D-Fender | jsbach: "exten => 100,1,SayDigits(100)" |
| 16:52.20 | [TK]D-Fender | jsbach: you dial 100, and it says it back to you |
| 16:52.47 | [TK]D-Fender | jsbach: "exten => _4XX,1,SayDigits(${EXTEN})" |
| 16:52.55 | Corydon76-work | I prefer MorseCode(100) |
| 16:53.00 | [TK]D-Fender | jsbach: that will let you dial from 400-499 and read back the digits. |
| 16:53.37 | Corydon76-work | dit-dah-dah-dah-dah dah-dah-dah-dah-dah dah-dah-dah-dah-dah |
| 16:54.22 | jsbach | [TK]D-Fender, but in sip you can have alphanumeric chars.. which means you are not bordered with numbers.. |
| 16:54.39 | codefreeze | l2cache: a natural mistake! who would change a config file? Actually, it's simpler than even that. You just use the same configs, and only investigate if there's a prob. |
| 16:55.00 | l2cache | Well I appreciate the help. Can't believe this was overlooked. |
| 16:55.23 | mocker | Woo, first pass at converting voip-info to a plucker palm document turned out alright. |
| 16:55.28 | [TK]D-Fender | jsbach: aND YOU WILL BE stuck USING SOFT-PHONES, SINCE NO-ONE IN THE REAL WORLD CAN DIAL A name IN AN IVR, ETC. |
| 16:55.30 | mocker | Portable voip-info! |
| 16:55.31 | mocker | ;) |
| 16:55.40 | l2cache | Im guessing i have to restart asterisk for it to start logging to master.csv |
| 16:55.44 | l2cache | I did a reload already |
| 16:55.57 | jsbach | i guess there would be a wildcard then like exten => $user, 1, DialOn(SIP/$user_placeholder) |
| 16:55.58 | [TK]D-Fender | jsbach: aND WHEN YOU ONLY HAVE 10 PEOPLE TO DIAL, WHY BE FORECED TO REMEMBER A long NAME TO TYPE? a NIFTY IDEA, BUT UTTERLY WORTHLESS |
| 16:56.15 | [TK]D-Fender | jsbach: nO, THERE IS NO SUCH WILDCARD FOR text NAMES. |
| 16:56.20 | [TK]D-Fender | darn caps... |
| 16:56.37 | codefreeze | l2cache: good guess. You might want to check all the other configs while you're at it, and save yourself a sleepless night or two in the next few weeks! |
| 16:56.38 | jsbach | [TK]D-Fender, i think you underestimate the power of soft phones or sip servers |
| 16:56.42 | wwalker | I have "one way audio". If I call from the phone attached to an SPA2102, the call works both ways. If I call into the phone from the outside (anywhwere but this office) the call is one sided (audio From the SPA2102 is not heard at the other end). watching the asterisk console show the correct IP, so although there is a router in between, there is no NAT. Ideas? |
| 16:56.57 | wunderkin- | [TK]D-Fender, yOUR EMPHASIS is BACKWORDS TODAY :D |
| 16:57.23 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 16:57.24 | [TK]D-Fender | jsbach: I think you misunderstand the PSTN world. Can you dial BOB from a touch-tome phone or are you looking at * and SIP as a way of communicating only ith your firends with a soft-phone? |
| 16:57.45 | wwalker | since I get the SIP registration and RTP toward the phone, it seems that it's not a network problem. |
| 16:59.36 | [TK]D-Fender | wwalker: check your extern IP and localnet settings (being multi-LAN + WAN) |
| 16:59.42 | *** join/#asterisk juanjoc (n=juanjoc@host191.190-30-20.telecom.net.ar) |
| 16:59.47 | jsbach | [TK]D-Fender, i do think voip is coming along stronger than pstn does now. soon you would be able to call bob from your home phone without noticing it that runs behind a sip proxy ;) |
| 17:00.30 | [TK]D-Fender | jsbach: You keep believing that... more power to you. Oh and get me the name of your dealer, you're clearly on some really good stuff there ;) |
| 17:01.49 | jsbach | [TK]D-Fender, everytime i hear the skepsis about the voip, i hear Bill gates saying "no one in this world needs more than 640kb memory" |
| 17:01.50 | jsbach | lol |
| 17:02.12 | Nugget | except Bill Gates never actually said that. |
| 17:02.47 | De_Mon | how do I get a device queue member that thinks its 'not in use' to say something else? |
| 17:02.48 | [TK]D-Fender | jsbach: I never said VOIP was bad. I just said that your concept of doing away with NUMBERS for extensions, etc is so remarkably far out as to have no basis in realistic practicality. |
| 17:03.20 | [TK]D-Fender | De_Mon: like? |
| 17:03.42 | jsbach | [TK]D-Fender, take that as my unexperience in asterisk... |
| 17:03.51 | De_Mon | [TK]D-Fender 'In use' would be good |
| 17:04.24 | [TK]D-Fender | jsbach: thats not even an Asterisk based commend, thats SIP/VoIP/PSTN reality. * as a tool or conveyor of any such tech is besides the point :) |
| 17:04.55 | *** join/#asterisk waverly360 (n=waverly@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 17:05.57 | De_Mon | [Jul 17 12:36:14] WARNING[5932]: app_queue.c:2646 try_calling: The device state of this queue member, SIP/test, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 17:06.18 | De_Mon | for that particular message I think the state is supposed to be 'Ringing' |
| 17:06.24 | jsbach | [TK]D-Fender, besides from the usage of numbers discussion , do you know how to disable the annoying 407 thing? tried AutoCreatePeer=yes but it doesn't help... |
| 17:06.36 | De_Mon | UPGRADE.txt doesn't seem to address this tho |
| 17:07.02 | De_Mon | jsbach I'm in favor of non-numerical extensions too |
| 17:07.40 | jsbach | De_Mon, welcome to club.. ;) |
| 17:08.03 | [TK]D-Fender | jsbach: You mean jsut accept un-auth'd calls? |
| 17:08.15 | jsbach | i simply dont see the point to call 100 to be able to speak with bob (??) |
| 17:08.47 | Andy_G | anyone here aclec? |
| 17:08.49 | jsbach | [TK]D-Fender, as far as i googled, it is to disable the "proxy authorization required" for each invite... |
| 17:08.50 | Andy_G | er |
| 17:08.51 | Andy_G | CLEC |
| 17:09.08 | [TK]D-Fender | jsbach: because not all phones let you dial alpha-numeric. You need to rejoin reality... |
| 17:09.44 | *** join/#asterisk FonalityKris (i=sbk@bricks.of.yay.get.smuggled.org) |
| 17:09.53 | coppice | any reasonable person knows all phone numbers should be in chinese |
| 17:09.58 | De_Mon | [TK]D-Fender So, we should only imagine a world that uses phones we have NOW? |
| 17:10.00 | [TK]D-Fender | jsbach: And what is your GOAL with the lack of 407 (which is a security thing) |
| 17:10.16 | FonalityKris | Does anyone happen to have CISCO 7970 SIP Firmware? |
| 17:10.20 | De_Mon | [TK]D-Fender what about video? can we add video to phones even though not all phones let you see video? |
| 17:10.24 | jsbach | [TK]D-Fender, i understood, but it's still too strict backwards... |
| 17:10.27 | [TK]D-Fender | De_Mon: there is a difference between wishing things were different, and building your system for a world we won't see for DECADES at best. |
| 17:10.43 | [TK]D-Fender | jsbach: What is the actual problem with *'s behavior? |
| 17:11.20 | jsbach | [TK]D-Fender, it is enough secure once to be authorized through registration according to rfc3261.. so i dont need more |
| 17:11.41 | jsbach | [TK]D-Fender, so i could do also *,1,Answer() , you mean? |
| 17:12.47 | [TK]D-Fender | jsbach: is this a real load problem? how big a setup are you planning on running or are you simple another Gentoo-ricer looking to optimise that last 3ms of call setup delay? |
| 17:12.58 | [TK]D-Fender | jsbach: No, * = ASTERISK |
| 17:13.00 | *** join/#asterisk nirz (i=nir@bzq-88-152-101-90.red.bezeqint.net) |
| 17:14.09 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 17:14.27 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:14.28 | jsbach | [TK]D-Fender, i am up to make some benchmarking and those 3ms are actually big probs if your asterisk have been abused by say 500 users... |
| 17:14.36 | *** join/#asterisk ToyMan (n=Stuart@cpe-68-175-3-144.hvc.res.rr.com) |
| 17:14.59 | De_Mon | jsbach lol you fail |
| 17:15.12 | [TK]D-Fender | jsbach: And is that the size you're scaling to? In tems of simultaneous invites? |
| 17:15.52 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 17:16.04 | [TK]D-Fender | jsbach: Frankly as of that point you should be looking at (Open)SER anyways. You want a better SIP proxy.... then use a damn proxy :) |
| 17:16.16 | [TK]D-Fender | Asterisk is a... |
| 17:16.19 | [TK]D-Fender | ~b2bua |
| 17:16.20 | jbot | it has been said that b2bua is a back 2 back user agent |
| 17:16.22 | dijungal | hello... i am looking for an opensourse app to monitor my queues in asterisk... any ideas? |
| 17:16.45 | [TK]D-Fender | dijungal: go check the WIKI, plenty listed there. |
| 17:16.53 | dijungal | k |
| 17:17.03 | jsbach | [TK]D-Fender, well it depends, if you take a conference application that will be enough to see with that amount of users what happens with the performance at the asterisk.. |
| 17:17.04 | jm|home | anyone having problems with UK 0800 numbers due to e164? |
| 17:17.19 | jm|home | <PROTECTED> |
| 17:17.29 | jm|home | (public.sip.magrathea.net) |
| 17:17.34 | jsbach | [TK]D-Fender, no worries i am also using some ser instances ;) |
| 17:18.16 | [TK]D-Fender | jsbach: Fear not... one day you will graduate from theoretical to practical (or will be flattened by a rogue bus driver / meteorite) ;) |
| 17:18.42 | jsbach | [TK]D-Fender, i wish i could see your point, but i dont |
| 17:19.05 | De_Mon | jsbach you are not alone |
| 17:19.26 | jsbach | De_Mon, for the second time.. welcome to club ;) |
| 17:19.38 | De_Mon | this club sucks |
| 17:19.50 | [TK]D-Fender | jsbach: You worry about very big things... * isn't that great for huge conferences, or as the front end for auth'd calls. Usually those running SER in front proxy on the call to * un-authed internally (* is never exposed to the outside world) and is used as a termination or application server only. |
| 17:20.10 | [TK]D-Fender | jsbach: this bypasses the need to accoutns / auth / etc. |
| 17:20.10 | De_Mon | jsbach say that one more time and I'm gonna kill myself! then you'll be all alone!!! |
| 17:20.55 | [TK]D-Fender | jsbach: Though frankly if you think 407 auth load is an issue, just wait till you see the REAL fun thats right around the corner ;) |
| 17:20.56 | jsbach | De_Mon, dont do that.. |
| 17:21.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:21.58 | jsbach | [TK]D-Fender, besides of load, i dont think it is even necessary to use an extra "security" challange like 407.. come on, get a rfc 3261.. that will guide you to the sun ;) |
| 17:22.23 | [TK]D-Fender | jsbach: Oh, there you go thinking * actually has a complete SIP stack! ;) |
| 17:22.28 | errr_ | when I make an outbound call from my sip phone out the ZAP/g0 trunk how can I find out which zap chan its using? |
| 17:22.33 | [TK]D-Fender | lol @ chan_sip <-------------- |
| 17:23.32 | [TK]D-Fender | errr_: set verbose to 10 and you'll see it in CLI |
| 17:23.44 | *** join/#asterisk jsmith (n=jsmith@h46055150.area3.spcsdns.net) |
| 17:23.45 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 17:23.47 | jsbach | [TK]D-Fender, frankly i am expecting it to be as long as it offers me a shared sip library.. or do you expect it to implement something like "rm" in your whatever *x system? |
| 17:23.51 | [TK]D-Fender | errr_: Also if you dump your active channels you can see what was bridged. |
| 17:24.12 | [TK]D-Fender | jsbach: "rm"? |
| 17:24.17 | jsbach | man rm |
| 17:24.43 | [TK]D-Fender | jsbach: basically *'s SIP implementation is notably lacking and wasn't made to scale. |
| 17:24.50 | errr_ | [TK]D-Fender: if I dump the active channels will people get disconnected? |
| 17:25.01 | *** join/#asterisk ToyMan (n=Stuart@cpe-68-175-3-144.hvc.res.rr.com) |
| 17:25.06 | jsbach | and i saw the sip.c file - quite long too.. |
| 17:25.11 | [TK]D-Fender | jsbach: You can set "insecure=port,invite" and so on to remove the extra auth's IIRC |
| 17:25.25 | [TK]D-Fender | errr_: "show channels concise" |
| 17:25.27 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:25.33 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 17:26.10 | errr_ | [TK]D-Fender: perfect, thanks |
| 17:26.16 | [TK]D-Fender | errr_: np |
| 17:28.36 | jsbach | [TK]D-Fender, ok i see thanx.. i found somewhere Autocreatepeer=yes.. but it is appearently sth different (?). |
| 17:29.33 | [TK]D-Fender | jsbach: wellt aht should create an authable entry of some sort... not sure on the details personally (never said I was PRO at that). |
| 17:29.44 | *** join/#asterisk holiday_42 (n=chatzill@spike.wcta.net) |
| 17:29.50 | [TK]D-Fender | jsbach: but I do know you could run it "open" on the back side of your SER setup |
| 17:30.28 | [TK]D-Fender | jsbach: and you ARE certainly aiming "big", so its been an interesting chat for sure... |
| 17:30.45 | jsbach | [TK]D-Fender, jah, okay i see.. |
| 17:31.14 | jsbach | [TK]D-Fender, i am just starting with some MRF .. does it ring bells there? |
| 17:31.47 | [TK]D-Fender | jsbach: Definately out of my league... |
| 17:32.08 | [TK]D-Fender | jsbach: I've LOOKED at SER, haven't even gotten my hands truely dirty yet. |
| 17:32.51 | [TK]D-Fender | jsbach: Took a little bit to stretch your initial small world test and "big quesiotns" into what sounds like a jsutifiably huge setup. |
| 17:33.27 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:33.38 | jsbach | [TK]D-Fender, it is about media servers etc.. i read about nice media processing of asterisk.. |
| 17:33.49 | jsbach | [TK]D-Fender, i dunno of course if you agree on that ? |
| 17:34.03 | [TK]D-Fender | jsbach: can you give a more specific scenario? |
| 17:34.29 | jsbach | like annoucements, voicemails conference .. etc => media server |
| 17:34.35 | *** part/#asterisk holiday_42 (n=chatzill@spike.wcta.net) |
| 17:34.38 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 17:34.54 | jsbach | [TK]D-Fender, what i am interested in at asterisk: rtp session setups |
| 17:35.19 | [TK]D-Fender | jsbach: Yes, that is in fact what * is used for mostly. |
| 17:35.54 | jsbach | [TK]D-Fender, so i am "ferkling" around with * for the moment and take a deep look to the media processing afterwards |
| 17:36.19 | [TK]D-Fender | jsbach: often even termination is delegated to large carriers (Level3,e tc), or PRI gateways like AudioCodes Mediant, etc |
| 17:36.45 | [TK]D-Fender | jsbach: and left out of *'s hands. |
| 17:37.01 | [TK]D-Fender | jsbach: but back-end VM, conf, etc, is what * is good for. |
| 17:39.33 | jsbach | [TK]D-Fender, ok i see.. |
| 17:40.29 | jsbach | [TK]D-Fender, as last i just wonder why do i get a 404 to a subscribe , after registration.. do i have to include subscribe=yes for alice in sip.conf ? |
| 17:40.48 | [TK]D-Fender | jsbach: What are you subscribing to? |
| 17:40.57 | [TK]D-Fender | jsbach: VM? Presence? |
| 17:41.07 | jsbach | for presence.. or i try to.. |
| 17:46.16 | [TK]D-Fender | jsbach: You'll need to set up HINT's in your dialplan for that to work. |
| 17:46.36 | [TK]D-Fender | "exten => bob,hint,SIP/bob" |
| 17:47.00 | [TK]D-Fender | jsbach: however this is going to be NASTY with SER in front whre * might not know what other calls "bob" may be having. |
| 17:47.13 | twitchnln | on a multitenant * setup, how can i get daily cdr's in email? |
| 17:47.29 | [TK]D-Fender | jsbach: Also given that you will NEED "autocreatepeer" for that because you need a trackable entry |
| 17:47.30 | twitchnln | anybody got a script? |
| 17:47.47 | [TK]D-Fender | twitchnln: This is up to YOU andexternal scripting. nothing to do withg * |
| 17:51.52 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:52.41 | *** join/#asterisk jsmith (n=jsmith@h46055150.area3.spcsdns.net) |
| 17:52.41 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 17:56.58 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:59.42 | jsbach | [TK]D-Fender, bye! |
| 18:00.39 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:05.37 | FonalityKris | <PROTECTED> |
| 18:07.15 | russellb | what? |
| 18:07.26 | msetim | hi guys, where can i post a bug from http manager? |
| 18:07.38 | russellb | msetim: bugs are posted to http://bugs.digium.com |
| 18:07.51 | msetim | rudholm, thanks |
| 18:09.10 | rene- | hey, can anyone please give me a test tdm call? it seems that my international carrier is not routing calls to me, i am at 52 998 2874123 |
| 18:10.53 | waKKu | damn!! folks.. what need I set on "username" and "password" for winprint hylafax ??? i saw on winprint page that it isnt needed, but seems it be necessary on new version .. |
| 18:11.41 | rudholm | msetim: no problem :) |
| 18:13.39 | MrMister2 | '«p'+eflç~ |
| 18:13.56 | MrMister2 | ASFAsf>Z>aZ |
| 18:14.00 | MrMister2 | ~« |
| 18:14.01 | MrMister2 | « |
| 18:14.01 | MrMister2 | «' |
| 18:14.02 | MrMister2 | º'* |
| 18:14.03 | MrMister2 | -'* |
| 18:14.03 | MrMister2 | -º+ |
| 18:14.03 | MrMister2 | º* |
| 18:15.06 | jm|home | very pretty. |
| 18:15.41 | MrMister2 | oops. sorry. cat jumped on the keyboard :( |
| 18:16.11 | mvanbaak | hhmm |
| 18:16.22 | JerJer | isn't there software that detects cat typing :) |
| 18:16.32 | MrMister2 | LOL |
| 18:16.39 | mvanbaak | extensions.conf [globals] overwrites everything in the extensions.ael globals {} part when issueing 'reload' on the CLI ? |
| 18:16.46 | mvanbaak | is this expected behaviour ? |
| 18:16.57 | mvanbaak | let me make it more clear |
| 18:17.11 | mvanbaak | extensions.conf only has an empty [globals] thing |
| 18:17.27 | jsmith | mvanbaak: Most likely, as the extensions.ael gets converted internally into the old dialplan language |
| 18:17.28 | mvanbaak | also a [hinst] for my subscription stuff |
| 18:17.47 | mvanbaak | my extensions.ael has globals{} where I define my outgoing trunks |
| 18:18.05 | mvanbaak | on a system start everything is fine |
| 18:18.15 | Strom_C | MrMister2: try typing ~cohujibuggle |
| 18:18.19 | mvanbaak | but after running 'reload' all my outgoing calls look like this: |
| 18:18.35 | mvanbaak | .... Dial(/31${EXTEN:1}) |
| 18:18.44 | mvanbaak | it's missing the IAX2/provider |
| 18:18.53 | mvanbaak | which is in a global var in extensions.ael |
| 18:19.01 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 18:19.06 | Strom_C | mvanbaak: what happens if you remove the empty [globals] context from extensions.conf? |
| 18:19.07 | mvanbaak | hhmm |
| 18:19.47 | mvanbaak | Strom_C: now it only has a [general] and a [hints] part |
| 18:19.54 | mvanbaak | same |
| 18:20.47 | Strom_C | i know it's not really an answer to your question, but you may be better off using astdb |
| 18:21.06 | mvanbaak | can I list the global vars set by extensions.ael on the cli ? |
| 18:22.11 | mvanbaak | hhmm |
| 18:22.18 | mvanbaak | reload |
| 18:22.20 | Strom_C | *shrug* |
| 18:22.23 | mvanbaak | and it's dead |
| 18:22.32 | mvanbaak | stop now => nothing happens |
| 18:22.36 | mvanbaak | I have to pkill -9 it |
| 18:22.41 | Strom_C | which version of asterisk? |
| 18:22.46 | mvanbaak | trunk |
| 18:22.54 | Strom_C | odd |
| 18:22.59 | mvanbaak | yeah |
| 18:23.03 | Strom_C | i hope you're not using that in production ;) |
| 18:23.09 | mvanbaak | ehm..... |
| 18:23.12 | mvanbaak | why not ? |
| 18:23.17 | Mercestes | well for one.... |
| 18:23.18 | Strom_C | ... |
| 18:23.23 | Mercestes | you have to pkill -9 it |
| 18:23.23 | Strom_C | because it's the development branch |
| 18:23.30 | Strom_C | things are frequently broken |
| 18:23.33 | centrex | Trunk is the code that is constantly being worked on an updated. Sometimes it's broken. |
| 18:23.45 | [hC] | Thats like running your daily driver go-to-work car on 200 shot nitrous and asking why that might be a bad idea :) |
| 18:23.57 | Strom_C | you might see messages on the commits list like "Who broke IAX?" |
| 18:23.59 | Mercestes | [hC], ...oh...I shouldn't do that? |
| 18:24.04 | centrex | And pkill also isn't exactly the safest command to use on a production server.... |
| 18:24.09 | [hC] | Mercestes: well.. YOU can... but... other people may not want to. |
| 18:24.10 | [hC] | :) |
| 18:24.19 | Mercestes | :D oh good. I'd miss my N0s |
| 18:24.24 | mvanbaak | guys guys |
| 18:24.26 | mvanbaak | easy |
| 18:24.32 | mvanbaak | it's production, for my home office |
| 18:24.33 | mvanbaak | :) |
| 18:24.45 | mvanbaak | customers are still on 1.2 |
| 18:24.49 | JerJer | sweeeeet - master password for Sunrocket devices |
| 18:24.55 | Strom_C | oooh |
| 18:24.57 | JerJer | http://gizmopasswords.blogspot.com/ |
| 18:24.58 | centrex | well pkill won't hurt anything usually, I've just had a few accidents with it personally =) |
| 18:24.58 | jsmith | JerJer: Oh? |
| 18:24.59 | mvanbaak | ROFL JerJer |
| 18:25.22 | Trevor_b | [TK]D-Fender: You happen to get a chance to try DSP on the lastest versions of asterisk again? |
| 18:25.37 | JerJer | someone just commented on my blog post about the topic |
| 18:25.49 | JerJer | http://tinyurl.com/yskdk8 |
| 18:26.05 | Trevor_b | JerJer: Didnt they just close doors, or declare a bankruptcy chapter? |
| 18:26.27 | *** join/#asterisk Zig5000 (n=zig5000@89-179-8-155.broadband.corbina.ru) |
| 18:26.58 | JerJer | i have heard some say bankruptcy, but I haven't seen any paperwork backing that claim up |
| 18:27.21 | dijungal | /exit |
| 18:27.23 | *** part/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 18:27.51 | Strom_C | wasn't sunrocket the company that said "give us one thousand dollars and we'll give you phone service for all eternity"? |
| 18:28.24 | Zig5000 | Hello. Sorry my English I am russian. I have trouble with asterisk. Zaptel channels reinitialize, but I don't know why? |
| 18:28.41 | mascool | is there anyway I can alter the EXTEN variable ? |
| 18:28.56 | JerJer | mascool: Goto ?! perhaps |
| 18:29.08 | [TK]D-Fender | Trevor_b: What DSP? |
| 18:29.19 | JerJer | ${EXTEN} is read-only, i am going to presume |
| 18:29.23 | Mercestes | Zig5000, are you a russian female? |
| 18:29.39 | JerJer | no no no.... a hot russian female |
| 18:29.48 | anonymouz666 | hahaha |
| 18:29.58 | Mercestes | doesn't even have to be hot. Just russian and female. |
| 18:30.03 | Zig5000 | <Mercestes> No I am russian male. =))) |
| 18:30.04 | JerJer | lol |
| 18:30.16 | JerJer | Privet |
| 18:30.25 | Trevor_b | [TK]D-Fender: Sorry SpanDSP, we spoke about you having issues loading it in the latest 1.2 series I beleive. |
| 18:30.27 | Mercestes | Zig5000, Oh, sorry, I don't know the answer to your question then |
| 18:30.45 | mascool | then is there any way I can pass an altered extension to a2billing.php ? |
| 18:30.52 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 18:30.56 | Mercestes | mascool, goto |
| 18:31.05 | Zig5000 | <Mercestes> ok. |
| 18:31.18 | [TK]D-Fender | Trevor_b: I have problems on ALL versions since 1.2.9.1 |
| 18:31.19 | mascool | hmm |
| 18:31.28 | mascool | goto what ? :) |
| 18:31.36 | Mercestes | Zig5000, How fast are they reinitializing? Is it constantly or just sometimes? sometimes * automatically resets the spans to correct common errors in your zap interfaces. |
| 18:31.39 | mvanbaak | gheh |
| 18:31.41 | mascool | I need to strip the first 5 digits off any extension |
| 18:31.45 | mvanbaak | looks like it's kaboom |
| 18:31.51 | Strom_M | mascool: use substrings then |
| 18:31.54 | mascool | and pass that as the new extension |
| 18:31.57 | Zig5000 | <Mercestes> Every 15 minutes |
| 18:32.08 | Mercestes | Zig5000, Could be a faulty PRI> does it drop calls when it does this? |
| 18:32.16 | jsmith | Zig5000: See the "priresetinterval" setting in Zapata.conf |
| 18:32.22 | mascool | a2billing.php routes calls based on prefix, but calls are being sent to me prefixed |
| 18:32.28 | jsmith | Zig5000: It defaults to resetting the *IDLE* B-channels every hour |
| 18:32.32 | Zig5000 | <Mercestes> I know bug if use Dial(Zap/g1/XXXX,120,D(XXXX)); |
| 18:32.33 | Mercestes | mascool: goto altered extensions |
| 18:32.43 | mascool | so I need to strip that tech prefix and just send the number with the area code prefix |
| 18:32.50 | Strom_M | mascool: use substrings then |
| 18:33.01 | mascool | Strom_C and then what ? |
| 18:33.12 | Zig5000 | <Mercestes> Asterisk drop all calls in channels E1 |
| 18:33.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 18:33.15 | mascool | ${EXTEN:5} |
| 18:33.28 | mascool | how do I pass that as ${EXTEN} ? |
| 18:33.38 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 18:33.41 | Zig5000 | priresetinterval default is ? |
| 18:33.42 | Trevor_b | [TK]D-Fender: Thats what I was looking for. Going to be replacing the junker fax server soon and deciding on what to do. Thinking maybe I buy a trixbox appliance and rebuild it with my TDM2400 and custom install (hate TB with a passion after working with them). |
| 18:33.49 | Strom_M | EXTEN will always contain the name of the current extension, mascool |
| 18:34.23 | Strom_M | either store it into a different variable that you can manipulate as you see fit, or jump to an extension which meets your criteria |
| 18:34.26 | [TK]D-Fender | Trevor_b: What I might suggest is getting Trixbox ONLY for the purpose of getting an * / SpanDSP build thaqts stable, then ripping out FreePBX. |
| 18:34.29 | Trevor_b | [TK]D-Fender: Probably spend a week f'in with SpanDSP and asterisk, see if i can get the currents to run. If not I may have time to look into the code itself. |
| 18:34.30 | Zig5000 | And I don't know when priresetinterval drop all calls |
| 18:34.35 | mvanbaak | brb, switching to my lappy so I can code and watch movie |
| 18:34.54 | jsmith | Zig5000: It shouldn't. If it's dropping calls, it's not priresetinterval |
| 18:35.33 | Trevor_b | [TK]D-Fender: More meant the new TB Appliance box. Although yeah, thats the current system is a A*H (preTB) running smooth as can be. Just having the cables run the room sucks, so i wanted to rack it with the rest of the hardware. |
| 18:36.11 | Trevor_b | Ill keep you up to date about my progress, probably wont be until week after next though, vacation time. |
| 18:36.23 | Zig5000 | <jsmith> Yes this is no priresetinterval. Before initializa channels asterisk print. S-Frame. channel NN is down |
| 18:36.23 | mascool | thanks Strom_M |
| 18:37.03 | jsmith | Zig5000: Yeah, then you need to figure out whether it's Asterisk taking the channel down, or the equipment on the other end of the link |
| 18:38.21 | Zig5000 | In other side E1 PSTN operator |
| 18:38.39 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:38.43 | Zig5000 | provider |
| 18:41.11 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:42.32 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 18:44.29 | *** join/#asterisk NirS (i=Nir@87.68.60.4.cable.012.net.il) |
| 18:46.17 | *** join/#asterisk PioneerVM4 (n=IceChat7@24-151-65-253.dhcp.nwtn.ct.charter.com) |
| 18:47.19 | *** join/#asterisk plla (n=nekomimi@200.31.103.86) |
| 18:47.37 | plla | Hello. |
| 18:47.55 | plla | I have a question. |
| 18:48.31 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 18:48.33 | plla | Can I play a background music while I play another music file on the foreground with Asterisk? |
| 18:49.08 | PioneerVM4 | I have a SIP Trunk -> Asterisk -> Two SIP Digital2Analog boxes at 2 locations -- when a call comes in I use Dial(loc1&loc2|15) to dial the phones (which works) -- however it seems if i answer a call on one phone and get another call the other phone won't ring (since one is being used), is that correct? |
| 18:52.05 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 18:52.10 | [TK]D-Fender | PioneerVMno. |
| 18:52.15 | Mercestes | plla: Trying to mix techno tracks with asterisk? |
| 18:52.47 | plla | nope, my client wants to play soft music while a prerecorded voice says the IVR. |
| 18:53.46 | [TK]D-Fender | plla: Remix them yourself |
| 18:53.49 | plla | Two options, or I mix the voices manually or Asterisk does it for me. I was wondering if Asterisk could do it. |
| 18:53.57 | [TK]D-Fender | plla: No, * can't |
| 18:53.59 | Mercestes | [TK]D-Fender, that will result in music breakage in a multi-level IVR |
| 18:54.03 | Hmmhesays | what is the point of tryexec |
| 18:54.07 | BSD_Tech | ok I did it tinybsd+asterisk-1.4.7.1+zaptel-1.4.3+sangoma-3.1.2+rhino-1.1.1+soundfiles onto a 256 ide dom module with 64megs of spage to spare |
| 18:54.13 | [TK]D-Fender | plla: *'s dialplan processing is incredibly linear |
| 18:54.36 | [TK]D-Fender | Mercestes: You say that.... as though I actually GIVE A %^#$ ;) |
| 18:54.52 | Mercestes | [TK]D-Fender, just pointing it out. |
| 18:55.10 | Mercestes | plla: Asterisk "waits" at each application in the dialplan. It will not continue until Background is done. |
| 18:55.20 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:55.38 | Mercestes | plla: You could do it with some meet me magic I bet, and use a third sip channel though. |
| 18:56.03 | _ys | Hello |
| 18:56.05 | Mercestes | plla: Just arbitrarily dump them into the conference, channel 1 is incoming, channel 2 is asterisk, channel 3 is constant, nice, soft music. |
| 18:56.21 | plla | Hmm, that's an interesting idea. |
| 18:56.24 | Mercestes | plla: You can be convicted of voodoo in atleast 3 states for that though. |
| 18:57.20 | Zig5000 | <Zig5000> Where I may view r4362 issue from ChangeLog of zaptel |
| 18:57.38 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.189.162) |
| 18:57.40 | Qwell[] | Zig5000: http://svn.digium.com/view/asterisk/ |
| 18:57.43 | PioneerVM4 | should i have QUALIFY=YES turned off for my softphones/PAP2T box |
| 18:58.02 | PioneerVM4 | saw some discussion where someone inferred it should be off, but didnt say why |
| 18:58.08 | PioneerVM4 | (IE: qualify=no) |
| 18:58.14 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 18:59.31 | Hmmhesays | is there any way beside privacy flag in the dial app to be able to answer a call keep the calling party still ringing? |
| 19:00.47 | Zig5000 | <Qwell[]> I not found this revision on http://svn.digium.com/view/asterisk/ |
| 19:04.06 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:04.07 | PioneerVM4 | anyone know a good public stun server |
| 19:04.52 | *** join/#asterisk evool (i=unknown@evool.com) |
| 19:04.57 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 19:05.19 | [TK]D-Fender | PioneerVM : no need |
| 19:05.48 | PioneerVM4 | i think we've been thru that before |
| 19:05.53 | evool | hello, i'm new to asterisk, had a question about SIP incoming channels. I set up a context in sip.conf, and was trying to setup a wildcard to answer in extensions.conf, but nothing seems to work |
| 19:06.00 | PioneerVM4 | my setup won't work without stun |
| 19:06.10 | [TK]D-Fender | PioneerVM : because....? |
| 19:06.23 | PioneerVM4 | because it won't -- i don't know the technical details |
| 19:06.34 | *** join/#asterisk NirS_ (i=Nir@87.68.60.4.cable.012.net.il) |
| 19:06.37 | evool | do i have to explicitly specify each DID to answer in extensions.conf? |
| 19:06.38 | PioneerVM4 | however, I've been thru it here and then everyone goes "oh well you do need stun" |
| 19:06.45 | [TK]D-Fender | PioneerVM : not one installation I've even heard of required it..... |
| 19:06.47 | PioneerVM4 | i'd love to not need it, however it won't work without it |
| 19:06.50 | Mercestes | The "i don't know the technical details" is more likely the cause of your reliance on stun than any other technical reality you are facing |
| 19:07.16 | PioneerVM4 | ive had long discussions and then the people here, possibly you DF, agreed i did need stun |
| 19:07.27 | [TK]D-Fender | PioneerVM : Never. |
| 19:07.30 | PioneerVM4 | trust me i would rather not need stun |
| 19:07.39 | PioneerVM4 | ok, let me know how to remove it and i gladly will. |
| 19:07.45 | [TK]D-Fender | PioneerVM : So how about you describe your environment quickly and we'll see whats going on... |
| 19:07.46 | Mercestes | Congratulations....you don't need stun. |
| 19:08.00 | Mercestes | That will be $150 please. |
| 19:08.08 | Zig5000 | <evool> what context in sip.conf and extensions.conf |
| 19:08.18 | PioneerVM4 | SIP Peer -> Asterisk server behind Cisco PIX firewall |
| 19:08.28 | PioneerVM4 | Asterisk Server -> home PAP2T-NA box |
| 19:08.30 | [TK]D-Fender | PioneerVM : OH GOD |
| 19:08.39 | PioneerVM4 | PAP2T-NA box is behind dynamic IP Linksys |
| 19:08.42 | [TK]D-Fender | PioneerVM : You need to forward a lot of ports the hard way. |
| 19:08.49 | PioneerVM4 | yea no kidding |
| 19:08.53 | *** join/#asterisk alexhopper (n=a27386@142.167.40.33) |
| 19:09.03 | evool | Zig5000: in SIP.conf, the SIP server is register =>, and later defined as [vitelity-incoming] context=from_vitelity |
| 19:09.04 | PioneerVM4 | (Linksys Router) |
| 19:09.05 | [TK]D-Fender | PioneerVM : PIX is the problem, and * doesn't support STUN, and ITS the one with the problem. |
| 19:09.17 | PioneerVM4 | well, it all works when I use stun |
| 19:09.21 | [TK]D-Fender | PioneerVM : Your phone behind Linksys isn't the problem. |
| 19:09.23 | PioneerVM4 | so, I do need stun |
| 19:09.28 | Mercestes | No. |
| 19:09.37 | Mercestes | He said you need to forward alot of ports the hard way |
| 19:09.39 | PioneerVM4 | well, let me know how to make it work WITHOUT getting rid of pix |
| 19:09.41 | evool | Zig5000: in extensions.conf [from_vitelity], i've tried using s,1,Answer, and _NXXNXXXXXX,1,Answer without success |
| 19:09.50 | PioneerVM4 | i've already done that, i told asterisk to use a certain port range |
| 19:10.01 | PioneerVM4 | to get SIP peer to work to asterisk |
| 19:10.09 | PioneerVM4 | however, i won't forward ports openly on a dynamic IP |
| 19:10.17 | Mercestes | I have this crescent wrench....and a phillips head screw, and I want to remove the screw....*Without* ditching the crescent wrench and getting the right tool for the job. |
| 19:10.20 | PioneerVM4 | my home box is dynamic IP, i won't blindly port forward |
| 19:10.26 | evool | Zig5000: if I specify the actual DID number, and then _NXXNXXXXXX,1,Goto(tag), it works fine, but i can't seem to match on a wildcard |
| 19:10.48 | PioneerVM4 | This isn't a case of "right tool for the job" |
| 19:10.52 | Mercestes | Oh yes it is. |
| 19:10.54 | ccesario | Hey, exists any similar program "ChannelRedirect." in asterisk-1.2.18 ? |
| 19:11.07 | PioneerVM4 | the PIX is embeded in the system, im not going to change it just to make VOIP work on a dynamic address |
| 19:11.08 | [TK]D-Fender | evool: "_NXXNXXXXXX"" *is* a wildcard! |
| 19:11.09 | Zig5000 | <evool> you may use _. for wildcard |
| 19:11.10 | Mercestes | I *GUARANTEE* you if you call up Cisco and go "voip" they'll go "uh, we dont' support that, upgrade your Pix." |
| 19:11.21 | PioneerVM4 | well, it works with Stun |
| 19:11.36 | Mercestes | well, good luck with all that then. |
| 19:11.36 | evool | [TK]D-Fender: i know, which is why i am confused why the Goto works, but the Answer does not |
| 19:11.41 | Zig5000 | No _NXXNXXXXXX is not wildcard _X. or _. is real wildcard |
| 19:11.48 | Mercestes | Don't come in here sawing off our left hand, our right hand, and then ask for applause. |
| 19:11.50 | PioneerVM4 | so back to my original question -- i need a good public stun server |
| 19:11.59 | Mercestes | Try google. |
| 19:12.07 | Mercestes | google: free stun server |
| 19:12.18 | [TK]D-Fender | Zig5000: "_." is a HORRIBLE wildcard that can have all sorts of consequensces due to *'s stantard extensions. |
| 19:12.39 | PioneerVM4 | i love this place, i ask a simple question, get lambasted telling me im doing it wrong and i dont need it... then im told to buy new hardware, then im told its my fault for asking in the first place |
| 19:13.04 | Mercestes | lambasted? |
| 19:13.05 | PioneerVM4 | "where can i buy a candy bar?" "you don't need a candy bar" |
| 19:13.05 | Mercestes | ....that's new |
| 19:13.12 | [TK]D-Fender | PioneerVM : insisting on using a bad tool for the job isn't really smart... |
| 19:13.13 | Mercestes | You *don't* need a candy bar. |
| 19:13.25 | PioneerVM4 | well maybe i like candy bars |
| 19:13.28 | twitchnln | PioneerVM4: what about stun.softjoys.com ? |
| 19:13.41 | evool | maybe you should eat fruit instead |
| 19:13.44 | PioneerVM4 | lol |
| 19:13.45 | Zig5000 | <[TK]D-Fender> if you use _. in right contexts you shouldn't have a problems |
| 19:13.48 | Mercestes | It's like Manxpower said earlier..."I want to make a flux capacitor. I found a paper clip, a dead cockroach, and a few legos under my refridgerator. How do I make a flux capacitor?" |
| 19:13.58 | [TK]D-Fender | Zig5000: You should never need to. |
| 19:14.09 | PioneerVM4 | i don't think thats quite the same. |
| 19:14.15 | Mercestes | Oh yes it is. |
| 19:14.19 | PioneerVM4 | ahhh, no its not |
| 19:14.23 | *** join/#asterisk simonkern (n=simonker@p54AAAC86.dip0.t-ipconnect.de) |
| 19:14.23 | Mercestes | w/e |
| 19:14.23 | PioneerVM4 | i didnt ask how to do something |
| 19:14.25 | simonkern | hi |
| 19:14.25 | [TK]D-Fender | PioneerVM : I was wondering what that burning smell was... |
| 19:14.26 | PioneerVM4 | i asked where to find something |
| 19:14.31 | evool | for some reason _X won't answer in [from_vitelity] context, but if I specify the number itself, it answers |
| 19:14.34 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 19:14.40 | [TK]D-Fender | PioneerVM : Well you've been pointed along the way.... |
| 19:14.41 | Mercestes | http://internetarguing.ytmnd.com/ |
| 19:14.59 | [TK]D-Fender | evool: PASTEBIN is your friend.... |
| 19:15.01 | [TK]D-Fender | ~pb |
| 19:15.02 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 19:15.05 | Zig5000 | <[TK]D-Fender> Yes. But I need to use XXXXXXXXXXXX numbers where count of X I don't know |
| 19:15.18 | evool | one second |
| 19:15.19 | PioneerVM4 | i asked where a good stun server was, i didnt say "i am using the COBOL language and a Apple II, show me how to make a stun server" |
| 19:15.25 | Zig5000 | <[TK]D-Fender> This is PIN codes |
| 19:15.38 | simonkern | does anyone have experiences with chan_mobile, because I've downloaded the asterisk-addon package via svn, but if I load chan_mobile, asterisk crashes |
| 19:15.56 | [TK]D-Fender | Zig5000: Why DON'T you know? How many different numbers are they sending, and I didn't say you couldn't doa variable length on, just not "_.". |
| 19:16.25 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 19:16.42 | Mercestes | Zig5000, exten => _X,1, and exten => _X.,1 is the proper way to do what you want to do. Never use _. ever. |
| 19:16.57 | Mercestes | Zig5000, it's one extra line. not that big of a deal. |
| 19:17.25 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 19:17.56 | [TK]D-Fender | yup |
| 19:19.07 | clyrrad | Is this valid? Set(${CALLERIDNUM}=${number}) - What I mean is am I able to manually set a variable like CALLERIDNUM? |
| 19:19.19 | Mercestes | clyrrad, what version of asterisk? |
| 19:19.36 | clyrrad | Mercestes: 1.2.11 |
| 19:19.56 | [TK]D-Fender | clyrrad: ICK. And unless the result of ${calleridnum} is useful as a VARIABLE NAME, I'd bet on "NO" |
| 19:20.04 | Mercestes | clyrrad, in 1.2 that is valid. in 1.4 you need Set{CALLERID(numanumadance)=${number}) |
| 19:20.17 | [TK]D-Fender | And no, it is NOT valid ;) |
| 19:20.18 | Mercestes | clyrrad, yea, drop the ${} around CALLERIDNUM |
| 19:20.37 | [TK]D-Fender | Mercestes: And you were close on yours ;) |
| 19:20.41 | clyrrad | Are we able to write to variables like that - or they are read only? I guess is my question..... |
| 19:20.46 | Zig5000 | <Mercestes> Ok thanks for consultation |
| 19:20.54 | Mercestes | yea, typo on the { lol |
| 19:21.02 | [TK]D-Fender | clyrrad: Set(CALLERID(num)=${number}) |
| 19:21.05 | Mercestes | clyrrad, Yes, you can write to calleridnumber |
| 19:21.09 | evool | http://paste.debian.net/32931 |
| 19:21.20 | Mercestes | clyrrad, It ignores everything past "num" so...lol |
| 19:21.32 | Mercestes | I used to put "number" but...then I got freaky with it |
| 19:21.38 | clyrrad | [TK]D-Fender: ok I will try your syntax |
| 19:21.51 | [TK]D-Fender | evool: exten => _X,1,Answer <- this only answers if the exten is a SINGLE DIGIT |
| 19:22.04 | Hmmhesays | bah creating an IVR inside of a macro sucks |
| 19:22.07 | evool | [TK]D-Fender, i've tried _NXXNXXXXXX and that doesn't work either |
| 19:22.19 | [TK]D-Fender | evool: exten => _NXXNXXXXXX,n,Goto(default,s,1) <---- this doesn't have a #1 priority and will NEVER get executed. |
| 19:22.55 | [TK]D-Fender | NEXT!!@!@@ (c) BKW |
| 19:23.25 | clyrrad | [TK]D-Fender: yup that worked well - thanks :) - Curious, my syntax as it was worked on variables I defined manually, is there some difference when writing to Asterisk defined variables? |
| 19:23.51 | [TK]D-Fender | clyrrad: review them... what you put was not sane :) |
| 19:23.58 | evool | k, updated priority on it to 1 |
| 19:24.04 | evool | but it's still not answering |
| 19:25.27 | clyrrad | [TK]D-Fender: hrm....... so it was the way I named the variable then? CALLERIDNUM vs CALLERID(num) |
| 19:25.28 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 19:25.53 | [TK]D-Fender | clyrrad: no, you REFERENCED that variable instead of specifying it. |
| 19:26.13 | [TK]D-Fender | clyrrad>Is this valid? Set(${CALLERIDNUM}=${number}) <- bad period. |
| 19:26.42 | [TK]D-Fender | Set(CALLERIDNUM=${number}) <- valid 1.0 syntax, and SHOULDN'T be use for 1.2+ |
| 19:26.54 | [TK]D-Fender | (CAN'T for 1.4) |
| 19:27.18 | clyrrad | I see - how do you specify that in 1.4 out of curiosity? |
| 19:27.24 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.10) |
| 19:27.56 | Inkubot | hi * |
| 19:28.05 | Inkubot | how can i send a register => without a password ? |
| 19:28.15 | [TK]D-Fender | Inkubot: Skip it. |
| 19:28.17 | *** join/#asterisk NirS (i=Nir@87.68.60.18.cable.012.net.il) |
| 19:28.31 | [TK]D-Fender | clyrrad: The way Mercestes and I jsut otld you. |
| 19:28.38 | Inkubot | i'm doing that... and the damn softswitch still reject my request |
| 19:28.39 | [TK]D-Fender | [TK]D-Fender>clyrrad: Set(CALLERID(num)=${number}) |
| 19:28.54 | [TK]D-Fender | Inkubot: beter details = better answers |
| 19:29.01 | Inkubot | i know.. |
| 19:29.08 | Inkubot | give me a minute.. |
| 19:29.20 | clyrrad | [TK]D-Fender: ah - ok so this works on 1.2.11 as well - I just tested it and it works - thanks for input and advice :) |
| 19:29.39 | *** join/#asterisk jsmith (n=jsmith@h46055150.area3.spcsdns.net) |
| 19:29.39 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 19:29.57 | Inkubot | i'm registering asterisk 1.4.7.1, using SIP protocol, the softswitch that it's rejecting my request it's a SoftX3000 from Huawei.. |
| 19:30.23 | Inkubot | DD"wrong password on authentication for REGISTER" |
| 19:32.05 | [TK]D-Fender | Inkubot: PM your register |
| 19:32.26 | *** join/#asterisk saftsack (n=saftsack@83-131-205-107.adsl.net.t-com.hr) |
| 19:32.37 | [TK]D-Fender | evool: PASTEBIN a failed call at verbose 10, SIP debug enabled |
| 19:32.43 | Inkubot | register => 5920311:@10.2.0.10:5060 |
| 19:33.00 | Inkubot | without de : |
| 19:33.16 | Inkubot | register => 5920311@10.2.0.10:5060 |
| 19:33.55 | [TK]D-Fender | Inkubot: Guess you need a pass |
| 19:34.28 | Inkubot | but with a softphone or a PAP2 i don't need it |
| 19:34.51 | Inkubot | works with the username.... |
| 19:35.47 | [TK]D-Fender | Inkubot: You mean you can have a softphone with no password register? |
| 19:36.02 | Inkubot | [TK]D-Fender: that's right |
| 19:36.10 | [TK]D-Fender | :/ |
| 19:36.14 | Inkubot | Huawei xD |
| 19:36.18 | [TK]D-Fender | Inkubot: And what about placing calls? |
| 19:36.29 | mascool | can anyone tell me why goto(${EXTEN:5}) won't match exten => _40.,1,Answer when ${EXTEN:5} = 40blahblah ? |
| 19:36.52 | Inkubot | let me explain the hole thing (i will try my best with english) |
| 19:37.13 | [TK]D-Fender | mascool: "show application goto" |
| 19:37.19 | [TK]D-Fender | Inkubot: No need. |
| 19:37.44 | [TK]D-Fender | Inkubot: You tell me it work without any pass, fine, I'll accept that and concede that the answer is beyond my ability to advise you. |
| 19:40.17 | Inkubot | i'm trying to do a gateway for this Softswitch, from SS7 to SIP and viceversa, i can receive calls from the SoftX300 and send it to the PSTN, i can send calls from PSTN (ss7) to the phones registered in the softswitch, but when i'm sending calls from PSTN (ss7) to PSTN (using the SoftX300 to reach the PSTN) the softswitch drops my call and give a sip error 503. I check the packets with wireshark, and i see no diference in the SIP header. So, i'm trying to |
| 19:40.40 | Hmmhesays | ok this sucks, you can't channelredirct until dial has exited |
| 19:40.42 | Hmmhesays | what good is that |
| 19:41.15 | [TK]D-Fender | Hmmhesays: When would you like to do that? |
| 19:41.24 | Inkubot | in the sip header there is a field... Supported |
| 19:41.34 | Inkubot | Asterisk always send Supported: replaced |
| 19:41.47 | Inkubot | and i think that the softswitch is expecting Supported: 100rel |
| 19:44.04 | *** join/#asterisk NirS_ (i=Nir@87.68.0.17.cable.012.net.il) |
| 19:46.09 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 19:46.29 | *** join/#asterisk sigmounte (n=sigmount@lns-bzn-50f-81-56-234-199.adsl.proxad.net) |
| 19:46.34 | Hmmhesays | [TK]D-Fender: i'm building my own privacy manager |
| 19:46.43 | Hmmhesays | so when dial is called I execute a macro |
| 19:47.07 | Hmmhesays | then based on what the called party dials in the macro I redirect the calling channel with app channelredirect |
| 19:47.09 | [TK]D-Fender | Hmmhesays: Ok, you can set the reject reason in there and the original channel will not be counted as being answered.... |
| 19:47.31 | [TK]D-Fender | Hmmhesays: use the macro about var it creates to do the redirect |
| 19:47.34 | Hmmhesays | in MACRO_RESULT |
| 19:47.38 | [TK]D-Fender | yup |
| 19:47.55 | [TK]D-Fender | Hmmhesays: so let it fall back to the original channel |
| 19:48.23 | Hmmhesays | so I use app channelredirect to direct the channel to the context,exten,prio based on what I dial |
| 19:48.44 | Hmmhesays | but, the kicker is I don't want to hang up my called channel |
| 19:48.51 | Hmmhesays | I want to be able to direct it somewhere else, such as chanspy |
| 19:48.56 | Hmmhesays | etc |
| 19:49.03 | Hmmhesays | so I can listen to the voicemail the user is leaving while they are leaving it |
| 19:50.21 | PioneerVM4 | if my asterisk box has multiple IP addresses, is there a way to tell asterisk to bind to one of them |
| 19:50.32 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 19:50.37 | [TK]D-Fender | Hmmhesays: You don't redirect to chanspy..... unless you're looking to redirect the answering end... |
| 19:51.10 | [TK]D-Fender | PioneerVM : "bindaddr=1.2.3.4" under [general] |
| 19:51.27 | PioneerVM4 | thx |
| 19:51.37 | Hmmhesays | yes, that is what I mean |
| 19:52.01 | PioneerVM4 | general in extensions.conf? |
| 19:52.07 | [TK]D-Fender | Hmmhesays: Hmm... not sure how to save that channel... I thikn * kills it on macro end... |
| 19:52.10 | PioneerVM4 | (there are multiple [generals]) |
| 19:52.21 | Hmmhesays | thats why i'm having trouble with channel redirect, the called party has to be hung up for the channelredirect to happen with the calling party channel |
| 19:52.22 | [TK]D-Fender | PioneerVM : no, this is sip.conf. extensions has nothing to do with SIP ports. |
| 19:52.29 | PioneerVM4 | ok |
| 19:53.01 | [TK]D-Fender | Hmmhesays: Hows this : spawn a call file to ring-back for spy and in there pass the chan to spy on <---- |
| 19:53.33 | [TK]D-Fender | Hmmhesays: 1-shot deal, it you ignore, it goes away.... like a quick "blip" |
| 19:53.42 | PioneerVM4 | df: that solved problem -- i added a second IP to my box and i couldnt register anymore |
| 19:53.49 | PioneerVM4 | df: used bindaddr and it fixed it |
| 19:53.51 | [TK]D-Fender | Hmmhesays: All before the macro extit. |
| 19:54.16 | [TK]D-Fender | Hmmhesays: And add a "wait" in front for good measure. |
| 19:54.32 | [TK]D-Fender | Hmmhesays: that way you don't need to postdata. set the CID to "chanspy", etc... |
| 19:54.56 | Hmmhesays | I can't have a ring back |
| 19:55.02 | Hmmhesays | its gotta be the channel that is already ope |
| 19:55.06 | Hmmhesays | *open |
| 19:55.11 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 19:55.12 | Hmmhesays | I'll figure it out |
| 19:56.15 | [TK]D-Fender | Hmmhesays: Your calling channel IS still open. pas that in a var to a call-file that targets your CALLED phone. When you terminate your called end, it will call you BACK. |
| 19:56.23 | *** join/#asterisk pixelate (n=kd@dsl-242-60-62.telkomadsl.co.za) |
| 19:57.18 | Hmmhesays | [TK]D-Fender: can't do it that way |
| 19:57.20 | [TK]D-Fender | Hmmhesays: it will them call chanspy against that channel for you |
| 19:57.25 | [TK]D-Fender | Hmmhesays: why not? |
| 19:57.25 | Hmmhesays | I MUST keep the called channel open |
| 19:57.32 | pixelate | anyone knows of an asterisk deployment involving the use of a WYSE Thin Client? |
| 19:57.35 | Hmmhesays | because no end user is going to like that |
| 19:57.41 | [TK]D-Fender | Hmmhesays: not AFAICS |
| 19:57.47 | Hmmhesays | we're talking about neophytes here |
| 19:57.53 | Hmmhesays | any extra steps are not going to be wanted |
| 19:57.56 | [TK]D-Fender | Hmmhesays: Oh, LIKING IT?! You picky little ^#%$^ ! |
| 19:58.13 | [TK]D-Fender | Hmmhesays: I hand you a workable solution and nag nag nag is all I get! ;) |
| 19:58.42 | twitchnln | pixelate: i've got * running on a neoware thin client under gentoo... seems fine, as long as no transcoding is going on |
| 19:58.56 | [TK]D-Fender | Hmmhesays: * has no clean way to break that dial bridge to spy nicely... thus is life. |
| 19:59.08 | jsmith | pixelate: Well, sort of... I've got plenty of clients using Wyse thin clients and Asterisk, but no audio is being done on the thin clients |
| 19:59.22 | Hmmhesays | [TK]D-Fender: I had already thought of that solution |
| 19:59.23 | [TK]D-Fender | Hmmhesays: only extra step is to pick up the phone when it blips you! |
| 19:59.27 | Hmmhesays | and rejected it |
| 19:59.36 | [TK]D-Fender | Hmmhesays: TFB <- |
| 19:59.43 | [TK]D-Fender | Hmmhesays: pass it on :) |
| 19:59.46 | Hmmhesays | [TK]D-Fender: you can channelredirect the calledparty out of the macro |
| 20:00.12 | Hmmhesays | it works, but i'm getting ast variable errors doing it |
| 20:00.20 | [TK]D-Fender | Hmmhesays: YUM |
| 20:00.49 | Hmmhesays | but [TK]D-Fender: you can break the dial bridge that wya |
| 20:00.51 | Hmmhesays | *that way |
| 20:01.15 | [TK]D-Fender | Hmmhesays: And let A fall off, and B move along w/ the channelt o spy on? |
| 20:01.57 | Hmmhesays | holy shit I got it to work |
| 20:02.04 | [TK]D-Fender | z0mg |
| 20:02.19 | pixelate | @@ |
| 20:02.42 | [TK]D-Fender | Hmmhesays: next they'll ask how to channel redirect to steal back the call during VM ;) |
| 20:02.56 | Hmmhesays | [TK]D-Fender: I already have the solution for that |
| 20:03.05 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:03.16 | [TK]D-Fender | Hmmhesays: time to whip out the duct tape & Crzy glue! |
| 20:03.31 | syzygyBSD | how can I debug the reason why a PRI is in a red alarm state? |
| 20:03.44 | Hmmhesays | however this is odd, if I use ExecIF to call my channelredirect I get all kinds of crazy errors |
| 20:04.00 | Hmmhesays | but if I just call exten => s,1,ChannelRedirect() then it works |
| 20:04.18 | *** join/#asterisk amadeupname (n=amadeupn@rrcs-24-173-134-222.se.biz.rr.com) |
| 20:05.05 | *** join/#asterisk luke-jr_work (n=luke-jr@adsl-76-194-177-181.dsl.ksc2mo.sbcglobal.net) |
| 20:05.11 | luke-jr_work | Any way to decrease the MOH volume? |
| 20:05.39 | [TK]D-Fender | luke-jr_work: SOX |
| 20:05.53 | jsmith | syzygyBSD: A red alarm is usually a wiring problem |
| 20:06.13 | syzygyBSD | ya... sadly we have rewired everything... or they say they have |
| 20:06.13 | luke-jr_work | [TK]D-Fender: so not with native MOH? |
| 20:06.28 | [TK]D-Fender | luke-jr_work: Nope. Goot do the files up. |
| 20:06.29 | jsmith | syzygyBSD: What are you plugging into the PRI port? |
| 20:06.35 | [TK]D-Fender | gotta* |
| 20:06.49 | amadeupname | is it possible for two instances of asterisk running under chroots on the same box to access different ports on a 2 port T1 card? |
| 20:07.05 | syzygyBSD | the T1 pri |
| 20:07.31 | luke-jr_work | [TK]D-Fender: wtf is a .sln, do you know? :p |
| 20:07.42 | jsmith | syzygyBSD: OK... did you try a T1 crossover (not ethernet crossover) between the smart-jack and the T1 card? |
| 20:07.48 | luke-jr_work | amadeupname: I imagine it depends on the card |
| 20:07.53 | [TK]D-Fender | luke-jr_work: Signed Linear |
| 20:08.00 | syzygyBSD | it was working up to yesterday at 8am .... |
| 20:08.02 | luke-jr_work | [TK]D-Fender: thx |
| 20:08.25 | Hmmhesays | ok, the debug info gives me no reason why i'm getting this error |
| 20:08.28 | luke-jr_work | does mode=files do subdirs? |
| 20:08.32 | amadeupname | more over will this break zaptel |
| 20:08.37 | luke-jr_work | eg, recursively |
| 20:09.10 | jsmith | syzygyBSD: Ah... very strange... either Zaptel isn't configured right, or someone cut your T1 cable :-( |
| 20:09.29 | amadeupname | jsmith: could be a bad end |
| 20:09.30 | [TK]D-Fender | luke-jr_work: Dunno... reduct to 1 file in your base folder, and find out :) |
| 20:09.34 | syzygyBSD | :( Zaptel configuration hasn't changed, maybe the card just went bad? |
| 20:09.43 | jsmith | amadeupname: Not if it was working yesterday |
| 20:09.57 | jsmith | syzygyBSD: It's possible... does zttool show anything interesting? |
| 20:09.58 | amadeupname | jsmith: yes if someone tripped over the cable |
| 20:10.19 | luke-jr_work | [TK]D-Fender: what sample rate etc are .sln assumed to be? :) |
| 20:10.20 | amadeupname | or lightening |
| 20:10.27 | syzygyBSD | jsmith: nope, just a red alarm. Like it isn't getting any info at all. Like a severd cable... |
| 20:10.31 | amadeupname | 8k most likely |
| 20:10.54 | amadeupname | syzygyBSD: got multiple ports on the card? |
| 20:11.04 | [TK]D-Fender | luke-jr_work: slin = 8khz base-line IIRC |
| 20:11.04 | luke-jr_work | 8-bit or 16-bit? |
| 20:11.06 | jsmith | syzygyBSD: Can you build a loopback plug and plug that into the T1 card? |
| 20:11.08 | syzygyBSD | nope, it is a TE110p |
| 20:11.13 | jsmith | syzygyBSD: That would help narrow down the problem |
| 20:11.15 | [TK]D-Fender | luke-jr_work: 16 bit I THINK. |
| 20:11.35 | Hmmhesays | question, is it ok to use goto's inside of a macro? |
| 20:11.49 | Hmmhesays | if you are going to a dst inside the macro |
| 20:12.22 | jsmith | syzygyBSD: http://kb.digium.com/entry/1/95/ |
| 20:12.24 | syzygyBSD | jsmith: good idea. I will have the tech on site do that |
| 20:14.03 | *** join/#asterisk NirS (i=Nir@87.68.0.17.cable.012.net.il) |
| 20:16.59 | twitchnln | my day is over, everyone have a good one |
| 20:17.02 | *** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 20:21.57 | Hmmhesays | yes [TK]D-Fender: you can channelredirect to break the bridge |
| 20:22.08 | Hmmhesays | AND keep both channels open |
| 20:22.15 | [TK]D-Fender | Hmmhesays: freaky... ok, fine, sure.... why not :) |
| 20:22.41 | Hmmhesays | hey it works |
| 20:24.22 | Inkubot | WOW!!! It's works!! |
| 20:24.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:24.46 | Inkubot | i comment a line in chan_sip.c |
| 20:26.17 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:28.29 | [TK]D-Fender | ok, time to head home, bbiab |
| 20:29.04 | *** join/#asterisk NirS_ (i=Nir@87.68.0.17.cable.012.net.il) |
| 20:31.19 | *** join/#asterisk pixelate (n=kd@dsl-242-60-62.telkomadsl.co.za) |
| 20:31.57 | pixelate | need advise to deploy asterisk over a WAN |
| 20:36.20 | Hmmhesays | ok, chanspy is not doing what I want |
| 20:36.21 | Hmmhesays | ARG |
| 20:37.01 | clyrrad | I have a .wav file with Audio Format: CCITT u-Law, anything special I need to do to get Asterisk to play it? |
| 20:37.37 | jsmith | clyrrad: It needs to be 16-bit, 8000 Hz for Asterisk to be able to play it |
| 20:37.55 | *** join/#asterisk NirS (i=Nir@87.68.0.17.cable.012.net.il) |
| 20:37.57 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 20:38.02 | clyrrad | jsmith: ah ok this one is 8bit - that must be way..... How can I convert it? |
| 20:38.15 | jsmith | 8-bit should be OK too, I think |
| 20:38.25 | jsmith | What happens if you try to play it from Asterisk? |
| 20:38.29 | clyrrad | jsmith: hrm.... Asterisk wont play it |
| 20:38.49 | clyrrad | sais the file does not exist in any format |
| 20:39.31 | jsmith | sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql |
| 20:39.45 | jsmith | You could try that, and see if that gets it into something that Asterisk likes better. |
| 20:40.26 | clyrrad | trying......... |
| 20:42.52 | clyrrad | jsmith: I get "sox: Input and Output rates must be different to use resample effect" |
| 20:43.53 | jsmith | clyrrad: You are using the resample effect... if you typed that correctly... replate foo-in.wav with the name of your file, and foo-out.wav with a new filename |
| 20:44.15 | Hmmhesays | is anyone using chanspy? |
| 20:44.21 | clyrrad | jsmith: yep indeed that is what I did |
| 20:44.42 | *** join/#asterisk saftsack (n=saftsack@89-172-132-34.adsl.net.t-com.hr) |
| 20:44.45 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 20:46.03 | clyrrad | jsmith: it just created me a file thats 44 bytes and gives me the error I mentioned above |
| 20:46.46 | jsmith | Very strange... maybe the .wav header doesn't properly describe the audio content... but i"m just guessing at this point. |
| 20:46.57 | jsmith | clyrrad: You could try pulling it into Audacity or something like that as well |
| 20:47.31 | clyrrad | jsmith: yes - I agree it does not like the header |
| 20:47.43 | clyrrad | jsmith: wonder a way to convert it to make it work properly |
| 20:49.06 | Hmmhesays | chanspy doesn't seem to be accepting input |
| 20:49.12 | Hmmhesays | however the dtmf is getting to asterisk |
| 20:50.11 | clyrrad | jsmith: Audacity plays the file with out a problem |
| 20:50.31 | Hmmhesays | is chanspy broken in 1.4.4? |
| 20:50.43 | *** join/#asterisk zpertee (n=chatzill@cpe-65-189-209-131.neo.res.rr.com) |
| 20:50.59 | jsmith | clyrrad: Can you use Audactiy to save it out in a different format? |
| 20:51.22 | zpertee | how do I know how powerful of a pc that I need to run asterisk? I'll have approximately 10 users |
| 20:52.04 | jsmith | zpertee: A 1 GHz machine should be fine for 10 users |
| 20:52.29 | zpertee | jsmith: ok thanks for the information |
| 20:52.58 | clyrrad | jsmith: ok trying now :) |
| 20:54.10 | clyrrad | jsmith: woot! that worked :) |
| 20:55.31 | Hmmhesays | anyone? |
| 20:55.40 | jsmith | clyrrad: Cool |
| 20:56.35 | clyrrad | jsmith: thanks for letting me know about Audacity - nice lill app - cheers! |
| 20:56.53 | jsmith | clyrrad: Glad I could help |
| 20:56.54 | *** join/#asterisk Op3r (n=Op3r@125.212.122.209) |
| 21:00.27 | Hmmhesays | [Jul 17 16:57:20] ERROR[11266]: app_dial.c:1526 dial_exec_full: Could not stop autoservice on calling channel |
| 21:00.33 | Hmmhesays | what exactly is that error |
| 21:00.44 | Hmmhesays | well what does it mean |
| 21:02.08 | waKKu | folks.. can I register an extension to associate with hylafax and transfer a incoming call to it ? |
| 21:02.28 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 21:03.13 | *** join/#asterisk jgoddess (n=womkim@g-cipher.net) |
| 21:04.37 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:04.59 | jgoddess | hello anybody in here noticed a bug with v1.4.7 and the manager API file being parsed but not opening a port? |
| 21:05.34 | Hmmhesays | ok chanspy doesn't seem to take audio while the voicemail is being recorded |
| 21:05.36 | Hmmhesays | wtf is up with that |
| 21:05.43 | zpertee | does anyone have any idea as to what it usually costs to have someone install a pbx for small business? |
| 21:07.43 | Hmmhesays | depends on how many users, how much hardware |
| 21:09.15 | *** join/#asterisk apardo (n=apardo@55.145.217.87.dynamic.jazztel.es) |
| 21:09.28 | zpertee | ok. A local small company wants me, a senior in high school, to install asterisk pbx for them and I have no idea how much to charge |
| 21:10.06 | NoCarrier | 1 year's beer money |
| 21:10.10 | jgoddess | well we are a hosting company and for admin work we usually charge 150.00 dollars an email |
| 21:10.13 | jgoddess | hour |
| 21:10.15 | jgoddess | =P |
| 21:10.17 | jgoddess | erg |
| 21:10.59 | zpertee | ok I'll just make something up as I go |
| 21:11.12 | Hmmhesays | ok |
| 21:11.23 | Hmmhesays | can anyone tell me why chanspy doesn't work while I'm recording a voicemail |
| 21:12.14 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 21:12.17 | *** join/#asterisk NirS_ (i=Nir@87.68.0.17.cable.012.net.il) |
| 21:12.59 | [TK]D-Fender | Hmmhesays, instead of following to VM, fall into a playback, can you hear anything? I suspect you can only spy on BRIDGED channels |
| 21:14.03 | Zodiacal | anyone know why i can't hear people but they can hear me? im using polycom a polycom 601... once i transfer the call to another extention and then they transfer it back to me i can hear them. this happens about 1 out of 3 calls.. |
| 21:14.15 | Zodiacal | it used to work fine |
| 21:14.18 | Zodiacal | for about a year |
| 21:14.20 | Hmmhesays | [TK]D-Fender: you know it is odd, I can hear everything while the vm prompts are playing |
| 21:14.21 | Zodiacal | then this started happening |
| 21:14.30 | Hmmhesays | [TK]D-Fender there is a flag to only spy on bridged channels |
| 21:14.44 | Hmmhesays | <PROTECTED> |
| 21:14.45 | Hmmhesays | <PROTECTED> |
| 21:14.58 | Hmmhesays | that tells me that chanspy should spy on a channel that is not bridged |
| 21:15.11 | Mercestes | <PROTECTED> |
| 21:15.16 | Mercestes | and I meant every word of it. |
| 21:15.29 | Hmmhesays | as soon ass voicemail starts to record I lose the audio on the spy |
| 21:15.40 | Hmmhesays | *as even |
| 21:16.05 | Mercestes | in case anyone is wondering, I have like 3 keyboards on my desk and sometimes I lay my arm on the numberpad <enter> key while I'm reading this screen. |
| 21:16.24 | [TK]D-Fender | Hmmhesays, perhaps its the way that it dumps audio.... if you TALK during the VM playback, can you hear both sides? |
| 21:17.27 | Hmmhesays | [TK]D-Fender: yes only when vm starts to record the message I lose the audio |
| 21:17.44 | Zodiacal | any ideas? |
| 21:17.46 | [TK]D-Fender | Hmmhesays, dang... |
| 21:17.54 | Zodiacal | i tried updating the polycom firmware to v212 and still does it :( |
| 21:18.31 | Hmmhesays | [TK]D-Fender: do you have a 1.2 box you can try it on? |
| 21:18.36 | Hmmhesays | or something newer than 1.4.4 |
| 21:19.12 | *** join/#asterisk NirS (i=Nir@87.68.0.17.cable.012.net.il) |
| 21:21.08 | [TK]D-Fender | Hmmhesays, unfortunately not handy... |
| 21:21.37 | Hmmhesays | [TK]D-Fender: i'll update my box then |
| 21:23.10 | Hmmhesays | looks like the fixed a bunch of stuff in trunk |
| 21:23.16 | Hmmhesays | but haven't been merged into 1.4 yet |
| 21:24.05 | file | if it's a bug it is fixed in the branch it is farthest applicable to, and then merged up |
| 21:26.10 | Hmmhesays | gotcha |
| 21:26.29 | Hmmhesays | there are some new chanspy options in trunk also it looks like |
| 21:26.53 | *** join/#asterisk el_critter (n=chatzill@190.74.124.133) |
| 21:28.47 | k31th | when i setup a phone in sip.conf i have to give the IP of the device? |
| 21:29.39 | el_critter | hi |
| 21:29.49 | el_critter | what's de difference between 1.4 and 1.2? |
| 21:30.22 | Mercestes | el_critter, try google asterisk changelog or read the changelog.txty |
| 21:30.31 | Hmmhesays | [TK]D-Fender: you got box you can test on period? |
| 21:30.38 | Hmmhesays | I'm curious if I'm the only one with this prolem |
| 21:30.42 | Mercestes | el_critter, that's kinda like asking what's the difference between a Prius and Bigfoot. |
| 21:30.55 | [TK]D-Fender | Hmmhesays, My own, yes |
| 21:31.06 | Hmmhesays | you want to try chanspy on a voicemail call for me? |
| 21:31.09 | *** join/#asterisk pariah (n=admin@unaffiliated/pariah) |
| 21:31.10 | Hmmhesays | tell me if it works for you? |
| 21:31.24 | el_critter | Mercestes: I don't think changelog will tell me de difference between two branches. |
| 21:31.32 | Mercestes | Oh? |
| 21:31.40 | *** part/#asterisk drgalaxy (n=drgalaxy@adsl-70-238-195-120.dsl.lbcktx.sbcglobal.net) |
| 21:31.45 | Mercestes | Did you READ it? |
| 21:32.05 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 21:32.40 | VOiCi | hey, I just bought an x100p card and was wishing to have an Asterisk PBX but just realized that my phone line is a voip line..is there anything i can do^ |
| 21:32.51 | pariah | could a cheap x100p be the reason things don't get hung up all the time? i will make a call from sip -> zap -> PSTN and it will work fine....but after a couple minutes of doing that i will try to go PSTN -> zap -> SIP and i will get a busy signal. could this be because i have x100p clones that i bought for $20? |
| 21:33.35 | VOiCi | anyone? |
| 21:33.38 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:33.45 | Hmmhesays | this doesn't work in trunk either |
| 21:33.46 | Hmmhesays | WTF |
| 21:33.51 | el_critter | Mercestes: if you tell me the difference I'll teach you how to be kind to people ;) |
| 21:34.51 | el_critter | Mercestes: just kiding, I thought they were branches like linux kernel numeration (2.14, 2.16, etc.) |
| 21:35.47 | el_critter | Mercestes: So changelog can't tell differences on those cases. |
| 21:36.17 | Mercestes | The changelog for asterisk defines exactly what's changed with each revision. |
| 21:36.49 | Mercestes | there are many differences, code updates, security updates, syntax reconstruction, variable updates, function adds, etc in 1.4. |
| 21:37.16 | Mercestes | I'm not certain exactly what kind of answer you are looking for but it is in the changelogs. |
| 21:37.26 | el_critter | so you keep 1.2 for stability? |
| 21:37.32 | Mercestes | some do. |
| 21:37.49 | Mercestes | The attitude here is "If your just now starting asterisk, you might as well go with 1.4." |
| 21:37.53 | tzafrir_laptop | VOiCi, a voip line?? |
| 21:38.20 | tzafrir_laptop | pariah, if all else fails, use busydetect |
| 21:38.31 | Mercestes | we keep 1.2 so we don't *introduce* instability to working 1.2 systems. People building new systems are advised to go to 1.4 |
| 21:39.25 | el_critter | Mercestes: Ok the last one was the answer I was looking for. I thought you were developing 1.2 and 1.4 in paralell |
| 21:39.37 | anonymouz666 | RetryDial() only works with BUSY? |
| 21:39.55 | tzafrir_laptop | el_critter, both are actually in maintinance mode right now |
| 21:40.57 | Mercestes | el_critter, No, 1.2 was updated into 1.4 and we continued the "legacy" system and the "new" system. |
| 21:41.15 | Mercestes | for people who refuse to upgrade. |
| 21:41.20 | Mercestes | like me. |
| 21:41.47 | pariah | tzafrir_laptop: what is this busydetect? |
| 21:41.48 | VOiCi | tzafrir_laptop, i mean, the line enters my modem and gives a signal to the phone |
| 21:41.53 | el_critter | ok, that's common. I understand now. Thanks a lot for all your help!!! |
| 21:43.00 | tzafrir_laptop | pariah, detecting that a line has hung up by the "busy" tone |
| 21:43.15 | tzafrir_laptop | pariah, where are you from? |
| 21:43.24 | anonymouz666 | I got -- Playing 'demo-congrats' in CLI I used RetryDial(demo-congrats,...) but I can't listen nothing |
| 21:43.39 | *** join/#asterisk shinao1 (n=shinao1@41.205.188.87) |
| 21:43.53 | anonymouz666 | when the peer is UNAVAILABLE |
| 21:44.00 | VOiCi | is there anything i can do tzafrir_laptop, i mean there is no way i can setup something with a fxo and fxs port if my phone line is from my modem .. |
| 21:44.25 | tzafrir_laptop | VOiCi, can you connect a simple analogphoneto that line? If so, that X100P should do |
| 21:44.40 | VOiCi | nah, it gotta go through my modem first..... |
| 21:44.51 | pariah | tzafrir_laptop: i am from the states. so busy detect will hang up a busy line? |
| 21:45.19 | tzafrir_laptop | VOiCi, what modem, exactly? Do you refer to an X100P card or to somethingelse? |
| 21:45.27 | VOiCi | well, my internet modem |
| 21:45.37 | VOiCi | my phone line goes through my internet modem |
| 21:46.21 | tzafrir_laptop | pariah, chances are you can use ks signalling to get notified ofhangup |
| 21:46.42 | tzafrir_laptop | anyway , I'm off to bed, good night |
| 21:46.50 | VOiCi | so, there is nothing i can do tza^ |
| 21:48.07 | tzafrir_laptop | what do you have in zaptel.conf? |
| 21:48.25 | tzafrir_laptop | fxsks=1 or fxsls=1 ? |
| 21:48.44 | VOiCi | my problem is mostly about my line. like i have no analog line, PSTN |
| 21:49.13 | tzafrir_laptop | VOiCi, sorry, confused you with pariah |
| 21:49.20 | VOiCi | ok np |
| 21:49.33 | tzafrir_laptop | anyway, why insist on an analog connection if you don't have any |
| 21:49.35 | tzafrir_laptop | ? |
| 21:50.24 | VOiCi | well, because i wanted to get an asterisk pbx up.. |
| 21:50.48 | tzafrir_laptop | connectit to the world through voip? |
| 21:51.04 | k31th | humm when i try dial an extension i get the following error: pbx_extension_helper: Cannot find extension context 'sip' |
| 21:51.05 | tzafrir_laptop | VOiCi, there's something in your exeplanation I miss |
| 21:51.08 | k31th | bad config? |
| 21:51.17 | *** join/#asterisk seele_ (n=seele@64.76.191.12) |
| 21:51.23 | seele_ | please help I have a queue configured with callback agents, when the queue is full (20 -30 callers) the agents does not receive any call or the calls is received but is hang ... how can i solve this?? |
| 21:51.23 | bkruse | gus, you here? |
| 21:51.52 | tzafrir_laptop | k31th, show dialplan that_context and you'll notice it has no extension 'sip' |
| 21:51.57 | VOiCi | i might have problems understanding some concepts...my line is for voip, so there is no analog from the external world, so |
| 21:52.06 | VOiCi | i cannot use my fxo card |
| 21:52.25 | tzafrir_laptop | VOiCi, don't you have an analog phone at home? |
| 21:52.42 | VOiCi | yeah i have, but i gotta put my phone line in my fxo card dont i^ |
| 21:53.13 | tzafrir_laptop | So connect the FXO card to the same line you normally connect your phone |
| 21:53.34 | Hmmhesays | da da da da da hey |
| 21:53.35 | VOiCi | ok done |
| 21:54.17 | anonymouz666 | anyone in here already touch someday in RetryDial() ? |
| 21:56.29 | De_Mon | anonymouz666 huh? |
| 21:56.53 | anonymouz666 | exten => _11,3,RetryDial(demo-thanks|5|3|SIP/415|60|m) |
| 21:57.03 | anonymouz666 | 415 returns CONGESTION |
| 21:57.11 | anonymouz666 | CLI says demo-thanks is playing |
| 21:57.12 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:57.17 | anonymouz666 | but i don't listen the demo-thanks sounds |
| 21:57.22 | Hmmhesays | ok, I guess you can't chanspy on a channel while it is recording either |
| 21:57.24 | Hmmhesays | what is up with that |
| 21:57.30 | anonymouz666 | only the ringing tone from the damn softphone |
| 21:57.45 | Mercestes | Hmmhesays, Yes you can. There is a specific setting/thing you have to do to chanspy a recorded channel tho. |
| 21:57.54 | Mercestes | Like, transmitsilenceduringrecord or something |
| 21:58.04 | Mercestes | It's on the wiki under asterisk application chanspy |
| 21:58.07 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 21:58.09 | *** part/#asterisk pariah (n=admin@unaffiliated/pariah) |
| 21:58.41 | anonymouz666 | hehe |
| 21:58.47 | anonymouz666 | better get ride of it |
| 21:59.09 | De_Mon | anonymouz666 it works for me, what version do you use? |
| 21:59.14 | Hmmhesays | hmmm I'll check it out hold on |
| 22:00.38 | anonymouz666 | damn I am not so smart. the return code is CHANUNAVAIL. I don't think retrydial() understand this return name. |
| 22:01.02 | Hmmhesays | Mercestes: i see it |
| 22:01.04 | Hmmhesays | but its not working |
| 22:01.13 | De_Mon | anonymouz666 you shouldn't hear any ringing with the m option |
| 22:01.14 | Mercestes | Works for me. (tm) |
| 22:01.19 | bkruse | anonymouz666: you can match it? |
| 22:01.21 | Mercestes | try a full reboot |
| 22:02.00 | De_Mon | anonymouz666 what version of asterisk? |
| 22:02.02 | Mercestes | assuming you changed something since the last reboot of course. |
| 22:02.06 | bkruse | anonymouz666: with an inline if ${var} = ${CHANUNAVAIL} ? blah : blah; |
| 22:02.25 | De_Mon | bkruse wtf is that, ael? |
| 22:02.29 | Hmmhesays | whoops |
| 22:02.31 | Hmmhesays | I be retarded |
| 22:02.36 | Hmmhesays | it is working now |
| 22:02.42 | De_Mon | you retard! |
| 22:02.47 | Hmmhesays | haha |
| 22:02.52 | Hmmhesays | so this is pretty sweet |
| 22:02.54 | bkruse | De_Mon: lol, no, not at all |
| 22:03.01 | bkruse | its neither |
| 22:03.08 | Hmmhesays | I have my awesome modified privacy manger with live voicemail |
| 22:03.09 | bkruse | its not even proper syntax for bash! and thats hard to do. |
| 22:03.35 | De_Mon | ok |
| 22:03.43 | bkruse | De_Mon: javascript, bash, or php? |
| 22:03.45 | Mercestes | sweet |
| 22:03.58 | bcnl | does anyone else here use Aastra phones and occasionally get a 'Got SIP response 405 "Method Not Allowed" back from aaa.bbb.ccc.ddd' message? |
| 22:04.06 | bcnl | it happens mid call |
| 22:04.14 | bcnl | and the caller gets hung up on |
| 22:04.25 | bkruse | var De_Mon = (_$('de_mon').value) ? 'awesome' : 'nub'; |
| 22:06.11 | anonymouz666 | I think it's broken eyeBeam. You never can set the crappy softphone as busy. Always accept a new call. I just want to callwaiting=no. |
| 22:09.04 | De_Mon | bcnl I do, and no |
| 22:09.11 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- *Critical Updates* Asterisk 1.2.22 and 1.4.8 released (July 17, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 22:10.35 | *** join/#asterisk Molotov (n=joe@unaffiliated/wiby) |
| 22:11.54 | Hmmhesays | now only if chanspy had the X option in 1.4 |
| 22:13.41 | *** join/#asterisk yannj_fr (i=yannj@82.227.103.140) |
| 22:13.46 | yannj_fr | Hi all |
| 22:14.30 | yannj_fr | Is there some people interested in doing a large Asterisk benchmark? |
| 22:14.33 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 22:15.48 | anonymouz666 | russellb: critical? |
| 22:15.48 | Qwell[] | critical |
| 22:16.43 | russellb | anonymouz666: details are already on asterisk.org |
| 22:16.50 | russellb | i'm working on getting all details posted to the web site and mailing lists |
| 22:17.00 | Qwell[] | russellb: not on my mirror :( |
| 22:17.05 | Qwell[] | or, cache...hmm, not logged in...right |
| 22:17.18 | Qwell[] | there we go :D |
| 22:17.20 | bkruse | yannj_fr: tyes |
| 22:17.21 | bkruse | yes |
| 22:17.25 | bkruse | im interested |
| 22:17.27 | anonymouz666 | the topic is always faster than the website |
| 22:18.02 | Qwell[] | anonymouz666: the topic takes 2 seconds to update |
| 22:18.40 | anonymouz666 | heh |
| 22:22.17 | bcnl | De_Mon: strange I get them in spurts |
| 22:22.21 | bcnl | no real rhyme or reason |
| 22:22.51 | yannj_fr | Me idea would be to write a test protocol, and then to compare on the maximum of server |
| 22:23.27 | yannj_fr | if a lot of us does it, we would be able to have idea about dimensionning |
| 22:23.41 | adorah | hi does anyone poses the IP PHONE with infineon chipset from Wu Chuan? |
| 22:24.56 | Hmmhesays | so how hard would it be to add the X option of chanspy to 1.4? |
| 22:26.19 | Molotov | How is Asterisk support on FreeBSD? I know it has been spotty, for someone unfamiliar with asterisk would it be better to stick to linux? |
| 22:26.46 | Qwell[] | Molotov: depends on how familiar you are with freebsd, I suppose. |
| 22:27.19 | Molotov | decently familiar, but I think that means Id prefer to just use linux until Im familiar enough with asterisk to debug it on BSD |
| 22:27.22 | Molotov | thank you |
| 22:29.09 | yannj_fr | For the one who are interested in trying to do a large asterisk benchmark, please send me an email to : yann.jouanin@intelunix.fr |
| 22:29.26 | yannj_fr | (Time to sleep in France......) |
| 22:33.23 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:36.37 | *** join/#asterisk IPmonger (n=ipmonger@h-68-164-180-2.atlngahp.covad.net) |
| 22:37.13 | *** part/#asterisk IPmonger (n=ipmonger@h-68-164-180-2.atlngahp.covad.net) |
| 22:48.19 | *** join/#asterisk davidj (n=david@204-181-48-184.skybest.com) |
| 22:50.09 | *** join/#asterisk DaPrivateer (i=Privatee@66.92.79.20) |
| 22:52.52 | *** join/#asterisk CVirus (n=GoD@62.135.96.171) |
| 22:53.56 | Hmmhesays | well I successfully backported the X option of chanspy out of trunk |
| 22:54.20 | [hC] | What should i be able to set caller id number to on an outgoing pri call to have it show up as unknown? |
| 22:54.22 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 22:54.26 | [hC] | If i dont specify caller id it shows up as my pilot number. |
| 22:56.41 | ManxPower | chan_spy has caused problems since THE DAY IT WAS RELEASED. |
| 22:56.52 | Hmmhesays | its working fine now |
| 22:57.37 | carrar | it's been replaced with chan_voyeur |
| 22:57.44 | JT | [hC]: i don't understand what you're asking, could you rephrase? |
| 22:57.54 | anonymouz666 | ManxPower: it works for me |
| 22:58.00 | Strom_C | chan_thatweircreepwiththevideocamera |
| 22:58.01 | [hC] | JT: I think ive solved my own question. sec. |
| 22:58.28 | JT | Strom_C: chan_upskirtcam |
| 22:58.29 | ManxPower | anonymouz666: how many spys do you do per day? |
| 22:58.34 | *** join/#asterisk nahirean (n=FixBayon@unaffiliated/nahirean) |
| 22:58.42 | anonymouz666 | not much |
| 22:58.47 | anonymouz666 | not many |
| 22:59.06 | anonymouz666 | I don't like to hear other people |
| 22:59.06 | anonymouz666 | lol |
| 22:59.07 | anonymouz666 | hauhauhau |
| 22:59.13 | ManxPower | Try spying every call |
| 22:59.16 | Strom_C | ~cohujibuggle |
| 22:59.16 | jbot | [cohujibuggle] gublgubbglggugglbuglgbugblgbgbgbgbglbglgbulgblugbgubgublgbglulllbgbb |
| 22:59.31 | Hmmhesays | i'm doing this for a live voicemail app |
| 22:59.44 | anonymouz666 | ManxPower: what happens? |
| 22:59.44 | ManxPower | or spy a couple of hundred calls per day |
| 23:00.01 | ManxPower | you'll get unable to obtain channel lock, then Bad Things start to happen. |
| 23:00.14 | Hmmhesays | i want to know if I need to worry about this [Jul 17 18:56:02] ERROR[20299]: app_dial.c:1526 dial_exec_full: Could not stop autoservice on calling channel |
| 23:00.34 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 23:01.10 | anonymouz666 | vi ../apps/app_chanspy.c and fix it |
| 23:03.12 | VOiCi | how to know which channel your fxo port is on\ |
| 23:05.02 | [hC] | Is it possible to send arguments to a goto? or just a macro? |
| 23:05.14 | Hmmhesays | your channel variables are available |
| 23:06.13 | ManxPower | exten => 123,1,Set(FRED=Loves Barney) |
| 23:06.19 | ManxPower | exten 123,n,Goto(321,1) |
| 23:06.29 | *** part/#asterisk davidj (n=david@204-181-48-184.skybest.com) |
| 23:06.32 | ManxPower | exten 321,1,Noop(${FRED}) |
| 23:07.08 | [hC] | dont i need to set FRED with __FRED? |
| 23:07.28 | ManxPower | no because there is never a child channel spawned. |
| 23:08.08 | ManxPower | If you did a Dial(Local/321@extensions) then you would have to set __FRED for it to be available, but this is not an issue for gotos |
| 23:09.09 | *** join/#asterisk Downchuck (n=chatzill@c-24-22-20-80.hsd1.or.comcast.net) |
| 23:09.36 | Downchuck | I'm trying to figure out where my outgoing packets are being translated -- the Via header is getting toyed with |
| 23:10.00 | VOiCi | I just plugged my FXO card (x100p) and setted up AsteriskNOW, now im trying to setup and FXO echo but i cannot find on which channel my fxo card is, i tried 1-4 and got ZT_CHANCONFIG failed on channel x : No such device or addresse (6) , anyone&^ |
| 23:10.16 | ManxPower | Downchuck: nat=yes, externip/externhost, localnet, and SIP aware NAT routers |
| 23:10.53 | Hmmhesays | is there any way to bridge two channels in the dialplan without a conference room? |
| 23:11.13 | JT | Dial |
| 23:11.45 | Downchuck | ManxPower: yes, there's a nat, and it's swapping y internal ip to an external.. but why do people do such horrible things to networks |
| 23:12.16 | JT | Downchuck: "people" ? |
| 23:12.27 | Downchuck | sorry.. "they" |
| 23:12.28 | VOiCi | can anyone help me with that problem i just posted^ |
| 23:12.38 | Downchuck | I'm trying to figure out which router it is that's so SIP aware |
| 23:12.45 | Downchuck | so i can stab it |
| 23:13.02 | JT | if you've got a cisco doing NAT, bin it ;) |
| 23:13.46 | Downchuck | trying to convince my ISP that it's still their fault |
| 23:13.57 | Downchuck | turns out they were standing in the way for the last 2 months |
| 23:14.09 | Hmmhesays | can you bridge two existing channels without dial? |
| 23:14.28 | Downchuck | "There aren't any SIP specific rules on the firewall either. I have double checked the firewall rules." |
| 23:15.26 | ManxPower | Downchuck: Cisco boxes will screw up SIP |
| 23:15.35 | ManxPower | unless you turn off sip-fixup |
| 23:15.48 | [hC] | you should really never use fixup for anything. ever. |
| 23:15.56 | Downchuck | I sure didn't.. it's a large facility |
| 23:16.36 | Downchuck | sip-fixup settings a more globalish option than say, the independent firewall rules? |
| 23:16.44 | Downchuck | or you mean.. sip-fixup is on by default |
| 23:16.49 | Downchuck | because cisco is fun |
| 23:17.23 | ManxPower | Downchuck: yes, if it is supported by that verison of the IOS |
| 23:18.09 | JT | Downchuck: you're not using cisco crap between asterisk and your isp are you? |
| 23:18.27 | Downchuck | no, but I don't know what's between the data center and my machine |
| 23:18.51 | Downchuck | odds are the firewall is cisco.. just because |
| 23:19.08 | JT | because they are knobs? :) |
| 23:19.10 | anonymouz666 | "cisco crap" |
| 23:19.20 | JT | crap as in shit |
| 23:20.11 | JT | anonymouz666: does that clear up the confusion? |
| 23:20.13 | ManxPower | Ciscos are GREAT, as long as you don't expect them to do all that complicated stuff the sales people talk about. |
| 23:20.26 | anonymouz666 | cisco are great. |
| 23:20.27 | JT | ManxPower: and as long as you never use a PIX |
| 23:20.34 | JT | anonymouz666: like a hole in the head |
| 23:20.39 | ManxPower | JT: I don't consider a PIX to be "Cisco" |
| 23:20.59 | JT | pix has the "sip fuckup" option among other things |
| 23:21.11 | JT | also, they phones are unimpressive |
| 23:21.13 | JT | their |
| 23:21.25 | ManxPower | I suspect Cisco bought the company, renamed the product PIX and slapped a Cisco name on it. Much like Cisco Call Manager |
| 23:21.26 | JT | and the way they treat customers is crap |
| 23:21.30 | JT | same with firmware |
| 23:21.31 | JT | heh |
| 23:21.54 | anonymouz666 | cisco have the best routers, switch, and already was the most valuable company in the world. |
| 23:22.02 | anonymouz666 | i don't have nothing against cisco |
| 23:22.10 | JT | anonymouz666: ok, let's argue about the point instead of spewing marketing bullshit |
| 23:22.28 | Downchuck | nevar! |
| 23:22.30 | ManxPower | We use all Cisco 2621 routers and are slowly moving all the switches to Cisco Catalyst 550X switches |
| 23:22.31 | JT | ManxPower: i believe the sip fuckup option only with cisco phones |
| 23:22.43 | VOiCi | How to know which channel your FXO port is on??? |
| 23:22.52 | ManxPower | JT: I seem to vaguely recall it does most any SIP |
| 23:23.07 | ManxPower | VOiCi: Channels MUST start at 1 |
| 23:23.14 | JT | ManxPower: yes, it only works properly with cisco phones, fucks all the rest up ;) |
| 23:23.24 | ManxPower | JT: Ah. |
| 23:23.33 | Downchuck | I believe it looks for \r\n |
| 23:23.37 | Hmmhesays | there should be an app_bridge and app_unbridge |
| 23:23.40 | JT | that's what i've been told by people who have evil PIXs anyway ;) |
| 23:23.44 | Downchuck | because I haven't gotten the same results using socat via linux machines |
| 23:24.05 | Downchuck | still, something hates me |
| 23:24.25 | ManxPower | We use just the basic packet filtering on the Cisco routers |
| 23:24.49 | JT | and newsflash: cisco are not in the world |
| 23:24.51 | ManxPower | Our network has massive holes in it from a security standpoint |
| 23:24.56 | JT | can't stand cisco fanboys ;) |
| 23:25.04 | JT | gar |
| 23:25.09 | JT | my english is not with me |
| 23:25.16 | JT | and newsflash: cisco are not the bestin the world |
| 23:25.23 | JT | there are better than cisco |
| 23:25.26 | Downchuck | Yeah I replicated the error.. I'm so awesome |
| 23:25.43 | Downchuck | If the packet uses \n... it don't touch it.. but \r\n.. it sure does |
| 23:25.46 | JT | and more appropriate solutions for most businesses |
| 23:26.36 | ManxPower | People Who Generate Revenue insist on taking their laptops home, plugging into their home router, and also use Verizon PCMCIA internet service (yes, EVEN while connected to the corporate network) |
| 23:26.50 | ManxPower | Since they are the People Who Generate Revenue nothing will ever be done about it |
| 23:28.24 | ManxPower | Also people that find proxy servers out on the internet that we have not managed to block do not even get a reprimand in their personel file. |
| 23:28.29 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:28.50 | JT | sorry, what is it that these people are doing? |
| 23:29.08 | JT | ah, security holes |
| 23:29.21 | ManxPower | JT: Usually trying to get to some website blocked by the corporate proxy |
| 23:29.32 | JT | why does it block websites? |
| 23:29.52 | ManxPower | JT: Why should it not block access to webmail services that do no virus scanning? |
| 23:30.35 | JT | because you can do virus scanning yourself? |
| 23:30.39 | ManxPower | User education has failed miserably everytime it has been tried. |
| 23:30.48 | ManxPower | JT: How exactly do you suggest we do that? |
| 23:30.59 | JT | there's no way you can possibly block all webmail, so that's pretty much a useless security measure |
| 23:31.02 | JT | scanning all traffic |
| 23:31.12 | ManxPower | What products will allow us to do that? |
| 23:31.23 | JT | for windows, GFI |
| 23:31.27 | JT | there are quite a few |
| 23:31.28 | ManxPower | It also does not deal with the cell internet service |
| 23:31.37 | Hmmhesays | what is a "marked user" in meetme? |
| 23:31.39 | ManxPower | JT: Oh. You trust virus scanners? |
| 23:31.41 | JT | also, there's realtime virus scanners for windows |
| 23:32.14 | JT | ManxPower: yes, i trust that all email coming into our office is scanned by 4 different virus scanner, and an exploit scanner |
| 23:32.17 | ManxPower | About once per week we find a virus, trojan, or malware that Norton AV missed |
| 23:32.23 | JT | and web traffic is scanned by 2 scanners |
| 23:32.28 | JT | and there is realtime AV too |
| 23:32.34 | JT | norton is shit |
| 23:32.42 | ManxPower | JT: I agree actually |
| 23:33.17 | ManxPower | I use trendmicro on my personal machine |
| 23:33.29 | JT | nod32, bitdefender, and a couple of others for email and web scanning |
| 23:33.42 | ManxPower | JT: At least we are getting the agents off the corporate network. They own their own machine |
| 23:34.09 | ManxPower | JT: What is the IT Staff/User ratio at your company? |
| 23:34.33 | ManxPower | This client has 3 IT staff for 400 users (prolly 400 users by now) |
| 23:35.00 | JT | i am the IT staff, but maybe a dozen users |
| 23:35.11 | waKKu | asterisk + hylafax + iaxmodem ownz :D |
| 23:35.14 | Downchuck | ManxPower: internal IT? |
| 23:35.18 | waKKu | works like a charm |
| 23:35.36 | ManxPower | Downchuck: define "internal it" |
| 23:35.46 | Downchuck | ManxPower: they go into the office |
| 23:35.52 | Downchuck | vs external IT.. they know how to use shell |
| 23:36.07 | ManxPower | Downchuck: that is ALL IT staff. Does not include the 2 consultants (of which I am one) |
| 23:36.24 | Downchuck | internal IT.. what's vnc? external IT... on call, on cellphone |
| 23:36.58 | Hmmhesays | is there anyway I can just play a single beep to meetme? |
| 23:37.08 | Downchuck | thanks for help btw.. I e-mailed the data-center.. again. :-) and if they still refuse, i'll just use a non-standard port |
| 23:37.15 | ManxPower | And really, the "IT staff" is the IT manager, the PC helpdesk person and the telecom guy |
| 23:37.17 | Downchuck | DNS SRV records are widely implemented |
| 23:37.49 | Downchuck | ManxPower: you label consultants.. "IT consultants"? |
| 23:38.07 | ManxPower | Downchuck: no, I label consultants that deal with IT stuff be "IT consultants" |
| 23:38.17 | Downchuck | k |
| 23:38.38 | ManxPower | Compared to a consultant for electrical problems, which would be "electrical consultant" |
| 23:39.29 | JT | "internal it" it's pretty obvious, internal to a company |
| 23:39.34 | Downchuck | specialization |
| 23:39.49 | ManxPower | I think I have solved the incentive for users to use cell internet while at work |
| 23:39.51 | JT | also, you don't really think there's only consultants in IT do you, Downchuck? |
| 23:39.58 | Downchuck | JT: :P |
| 23:42.04 | Downchuck | i didn't know people could work without computers |
| 23:42.20 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 23:42.20 | Hmmhesays | rock my live voicemail is working |
| 23:47.40 | Hmmhesays | this is kickass |
| 23:52.41 | Trevor_b | Hmmhesays: live voicemail? |
| 23:56.30 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:56.36 | Hmmhesays | Where you can listen to the caller leaving the voicemail and decide if you want to pick it up |
| 23:58.31 | Downchuck | i just pretend to be voicemail |
| 23:58.38 | Downchuck | works. |
