| 00:00.42 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:01.13 | jaytee | EmleyMoor, is Ekiga 2 working? |
| 00:03.05 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 00:03.38 | *** join/#asterisk sprite-- (n=sprite@12.228.1.97) |
| 00:04.19 | *** join/#asterisk Olobola (i=Olobola@101.sub-75-210-251.myvzw.com) |
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| 00:22.29 | *** join/#asterisk chouchen (n=chatzill@61.153.22.67) |
| 00:25.55 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:30.19 | *** join/#asterisk doug_correa (n=douglas@201.21.133.128) |
| 00:30.32 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 00:39.15 | freemind_ | anyone knows a good termination VoIP provider for up 1000 channels |
| 00:39.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 00:39.38 | FruitBasket | vitelity, nexvortex.. |
| 00:39.56 | *** part/#asterisk korihor (n=korihor@201.210.239.172) |
| 00:40.37 | freemind_ | FruitBasket: thanks. Any other^ |
| 00:40.40 | freemind_ | ?? |
| 00:41.33 | FruitBasket | broadvox |
| 00:41.39 | FruitBasket | haven't used them though. |
| 00:44.06 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 00:44.55 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:45.06 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:45.06 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:45.21 | Linuturk | hey folks. I've got a bit of a problem. I had to restart my asterisk server, and I've noticed that the ntp service is failing at boot (centos 4.4). I've also noticed that if I restart one of my sip phones, the time and date is wrong. when I run data at a prompt though,the date is correct |
| 00:45.27 | Linuturk | ideas? |
| 00:46.25 | Linuturk | grabs his towel and tries not to panic |
| 00:48.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-2e07e6f17c7771d5) |
| 00:50.29 | Linuturk | wel, wait a sec. I just restarted another phone and it is right |
| 00:50.43 | Linuturk | I'm so confuzzled |
| 00:52.35 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:52.35 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:53.15 | Linuturk | jeeze, that was stressful. for some reason it's got the right time now |
| 00:53.43 | Linuturk | now to see why these faxes aren't working |
| 00:54.53 | nhuisman_work | is there a way in asterisk to display mac addresses of sip clients? |
| 00:55.10 | nhuisman_work | besides the obvious, arp every single one |
| 01:01.18 | lanning | asterisk runs at a higher level (IP), besides ARP is only good for the local subnet. |
| 01:05.03 | nhuisman_work | lanning, yeah just curious |
| 01:05.14 | nhuisman_work | i used awk, xargs, ping, arp, and grep |
| 01:08.27 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 01:09.21 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:09.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:14.42 | *** join/#asterisk invalidrecord (n=fares@87.113.88.31.plusnet.pte-ag2.dyn.plus.net) |
| 01:17.00 | *** join/#asterisk chendy (n=chatzill@58.61.40.134) |
| 01:20.14 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 01:24.18 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 01:25.33 | *** join/#asterisk Sargun (n=Sargun@atarack/staff/sargun) |
| 01:25.44 | Sargun | Can you get a higher sampling rate than 8 Khz? |
| 01:25.47 | *** join/#asterisk andresmujica (n=andresmu@201.244.108.160) |
| 01:29.43 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:31.11 | lanning | Sargun, not if you want to be compatible with anything else. (nothing will play a higher sample rate) |
| 01:36.28 | Sargun | What's a good software VoIP client that does G.722? |
| 01:42.26 | carrar | eyebeam does 722 I think |
| 01:44.36 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 01:44.46 | carrar | oh crap |
| 01:44.48 | carrar | he's back |
| 01:44.52 | carrar | I better shutup |
| 01:45.00 | carrar | :) |
| 01:45.27 | [TK]D-Fender | figures carrar will last 3 minutes tops... |
| 01:45.27 | carrar | hahah |
| 01:45.27 | [TK]D-Fender | :p |
| 01:48.09 | carrar | Sargun: http://forums.counterpath.com/viewtopic.php?t=14076&view=previous&sid=23276032b90e986e3461c4fc80212cb8 |
| 01:48.16 | carrar | G.722 will be available in upcoming builds of Bria and eyeBeam. |
| 01:48.30 | carrar | that was in 2003 |
| 01:49.39 | [TK]D-Fender | carrar: "Are we there yet..." |
| 01:49.53 | [TK]D-Fender | carrar: By-and-large the world doesn't really care so much I guess |
| 01:50.03 | carrar | damn them! |
| 01:50.06 | carrar | err |
| 01:50.08 | carrar | darn them |
| 01:52.39 | *** join/#asterisk jayyers (n=jayyers@c-71-59-10-252.hsd1.ga.comcast.net) |
| 01:53.06 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 01:53.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:56.05 | jayyers | im currently using switchvox free and 2 voip providers: gafachi and nexvortex. when making or recieving calls with these providers i can hear any audio both ways, if i dial an internal extention i can hear audio but if going through the voip providers i cant hear any audio, on both ends.....can anyone give me any pointers? |
| 01:57.18 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 01:57.57 | [TK]D-Fender | ~sipnat |
| 01:57.58 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:58.03 | [TK]D-Fender | ^^^^ |
| 01:58.20 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 01:58.40 | Akiyuki | ~cluebat akiyuki |
| 01:58.41 | jbot | ACTION pulls out a ClueBat (tm) and thwaps akiyuki. |
| 01:58.45 | Akiyuki | has trouble falling asleep |
| 02:00.10 | jayyers | btw i have the pbx behind a router and have dmz pointing to the pbx....any help would be great |
| 02:00.17 | edibrac | would raid 0 cause intermittant "PRI got event: HDLC Abort" errors? |
| 02:00.34 | edibrac | actually i could have sworn i read somewhere to avoid raid 0 for asterisk |
| 02:00.41 | edibrac | but i can't find it now |
| 02:02.43 | [TK]D-Fender | jayyers: I jsut linked you to the guide you should be following |
| 02:02.52 | [TK]D-Fender | jayyers: Port forwaarding alone is not enough |
| 02:03.31 | jayyers | ok thanks will look |
| 02:05.01 | carrar | avoid raid 0 regardless |
| 02:05.14 | carrar | heh |
| 02:05.42 | carrar | unless you mix it up with 1 or 0+1 |
| 02:05.43 | nhuisman_work | edibrac, no I wouldn't htink so, I think those are irq timing issues |
| 02:06.32 | nhuisman_work | i guess irq timing means nothing, laugh, but its something to do with the bus the card is on. |
| 02:07.28 | carrar | RAID card IRQ shouldn't really have any effect as to what type of RAID is being used |
| 02:07.52 | carrar | but could conflicted with other irq's |
| 02:07.52 | nhuisman_work | oh boy, I just screwed myself |
| 02:07.54 | nhuisman_work | thanks symlinks |
| 02:07.55 | nhuisman_work | thanks so much |
| 02:08.04 | [TK]D-Fender | Anyone with a brain would avoid RAID 0 for anything at all important. You're DOUBLING the chance of a critical failure |
| 02:09.58 | jayyers | is anyone familiar with Avaya Ip Office? |
| 02:10.20 | carrar | I have a Avaya that peers with a asterisk box |
| 02:10.21 | *** join/#asterisk quentusrex (n=quentusr@c-24-19-34-21.hsd1.wa.comcast.net) |
| 02:10.29 | quentusrex | how do I debug a phone registration on asterisk? |
| 02:10.33 | carrar | well, customers avaya |
| 02:10.38 | quentusrex | I'm in the asterisk -r interface. |
| 02:11.10 | carrar | They had to release some new avaya code to fix some bugs they had |
| 02:14.07 | jayyers | because we currently use avaya for our call center and it has a nice feature that when a user loads their avaya software it shows how many calls are waiting in the "Support Queue" and was wondering if there was a similar solution that integrates with asterisk |
| 02:16.52 | [TK]D-Fender | jayyers: Several monitoring apps already and easy enouhgh to write your own |
| 02:17.06 | [TK]D-Fender | quentusrex: "sip set debug on" |
| 02:17.11 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:17.52 | *** join/#asterisk irisht (n=irisht@cpe-66-68-47-247.austin.res.rr.com) |
| 02:18.29 | quentusrex | [TK]D-Fender: what settings to have to configure to get a remote extension working with nat? |
| 02:18.46 | jayyers | [TK]D-Fender: any suggestions on free open source windows apps that will do this? |
| 02:18.54 | [TK]D-Fender | ~sipnat |
| 02:18.55 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:19.05 | [TK]D-Fender | jayyers: atke a look at hudlit |
| 02:19.14 | [TK]D-Fender | jayyers: take a look at hudlite |
| 02:19.57 | kerframil | ~help |
| 02:20.51 | jayyers | ill looked at that before and didnt think it would do what i wanted but will take a close look, thanks |
| 02:21.59 | troubled | hey guys, a quick question. Is there anyway at all to get some of the wideband codecs that x-lite 3.0 supports to work with any version of asterisk and not be ultimately downsampled to 8khz? |
| 02:22.09 | edibrac | yeah I know you should avoid raid 0 - it was setup accidentally, and I caught it |
| 02:22.09 | [TK]D-Fender | jayyers: Might... don't know for sure, but it coveres a number of things. |
| 02:22.13 | troubled | or is asterisk an old dynasaur that will remain 8khz for eternity? |
| 02:22.30 | russellb | asterisk 1.4 supports passthrough of g.722 |
| 02:22.37 | russellb | and recording and playback |
| 02:22.43 | [TK]D-Fender | troubled: 1.6 supports it throughout |
| 02:22.45 | russellb | asterisk 1.6 also includes transcoding support |
| 02:23.17 | carrar | 1.6 has sharp edges that will make you bleed!!! :) |
| 02:23.27 | troubled | [TK]D-Fender: ah great. Still on an old debian stable 1.2 install here. 1.6 stable? |
| 02:24.09 | russellb | 1.6 is new. |
| 02:24.14 | troubled | russellb: passthrough is really all I was wondering about for sip conferences. I have a digium TDM400p card, but I wouldnt care about minor transcoding when I use the land line to talk |
| 02:24.18 | russellb | but a number of people are using it without problems |
| 02:24.32 | russellb | then yeah, 1.4 will do just fine for you, then |
| 02:24.50 | russellb | assming that you mean 2-party calls when you say "SIP conferences" |
| 02:24.52 | troubled | was more just wondering about trying to match skype's 16khz fidelity |
| 02:25.03 | carrar | russellb, any planes for T.38 transcoding? :) |
| 02:25.07 | russellb | yes. |
| 02:25.09 | carrar | woo |
| 02:25.14 | troubled | russellb: well, potentially 3 people, but all using x-lite with a 16khz codec hopefully |
| 02:25.16 | russellb | at some point in the relatively near future for 1.6 |
| 02:25.17 | [TK]D-Fender | troubled: thing is wiuth 1.4 you CAN'T transcode to get to PSTN |
| 02:25.39 | [TK]D-Fender | troubled: So 1.4 is out. |
| 02:25.39 | russellb | wonders if he ever posted a codec_g722 backport anywhere ... |
| 02:25.51 | troubled | [TK]D-Fender: I wouldn't be using it for pstn, just for sip conferences between 3 of us. skype is....not great some days :) |
| 02:26.08 | troubled | i dont mind trying 1.6 though as its not a production box or anything |
| 02:26.49 | troubled | btw, what was all that FBI noise about < 1.6 being susceptable to vishing? |
| 02:27.18 | troubled | i couldnt tell if it was just standard voip service callerid altering or if it was sip account hijacking |
| 02:27.21 | carrar | PHEAR the FBI |
| 02:27.27 | troubled | :) |
| 02:27.36 | quentusrex | [TK]D-Fender: I'm getting: registration failed: 408 request timeout on the softphone. |
| 02:27.54 | russellb | troubled: check out blogs.digium.com for more info on that |
| 02:28.22 | troubled | russellb: thanks |
| 02:28.25 | [TK]D-Fender | quentusrex: packets clearly aren't making it back. |
| 02:29.06 | quentusrex | [TK]D-Fender: is there a way to have sip debug only show for a particular ip, or for a particular extension? |
| 02:29.22 | russellb | sip set debug ip <addr> |
| 02:29.22 | russellb | i think |
| 02:29.26 | russellb | it's there in some form or another |
| 02:30.07 | [TK]D-Fender | quentusrex: First verify that your sip.conf is correct. |
| 02:30.15 | [TK]D-Fender | quentusrex: PASTEBIN is your friend... |
| 02:30.17 | [TK]D-Fender | ~pb |
| 02:30.18 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 02:31.43 | quentusrex | I have the localnet=192.168.100.0/255.255.255.0 |
| 02:31.59 | quentusrex | and externip=*.*.*.* were the stars are my actual externip |
| 02:32.15 | quentusrex | everything works for the local extensions, the remote one is having trouble. |
| 02:33.07 | drmessano | is waiting for the paste |
| 02:33.42 | troubled | russellb: i must say, I was not looking to upgrading my configs to 1.4 syntax, let alone 1.6 :) |
| 02:33.57 | carrar | 1.2 4 ever? |
| 02:34.11 | drmessano | What is "1.2"? |
| 02:34.19 | troubled | carrar: well, I tend to prefer to follow debian stable, so I would expect 1.4 for lenny |
| 02:34.25 | troubled | drmessano: "old" :) |
| 02:34.38 | drmessano | There was a 1.2????? |
| 02:34.52 | carrar | haha |
| 02:34.54 | troubled | debian even has a broken app for tuning my land line for my digium card I believe |
| 02:35.11 | carrar | I used to run 1.0 |
| 02:35.13 | troubled | so I never could get a nice echo canceled line since the app that ships with debian etch is borked |
| 02:35.33 | drmessano | troubled: Many of us prefer source since using your distos old idea of a current version is often a bad idea |
| 02:35.34 | carrar | * has come a long ways |
| 02:35.40 | troubled | i probably should just use cvs though |
| 02:36.48 | troubled | drmessano: ya, I know what you mean. Originally when I got the digium card, I got a nice wallet cdrom with source and all the info to install, which I did. But this install I just needed a quick install so I went with debian to find out they dont exactly test everything with hardware when they package stuff :) |
| 02:36.55 | [TK]D-Fender | quentusrex: And there is no reason I should take it on faith that any of the parms are in the right place. |
| 02:37.02 | [TK]D-Fender | quentusrex: PASTEBIN |
| 02:37.14 | drmessano | He just went into FreePBX looking for help lol |
| 02:37.22 | troubled | asterisk is still cvs I take it? or are we svn finally? |
| 02:37.25 | [TK]D-Fender | moves on to better things |
| 02:37.29 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 02:37.34 | troubled | if anyone says cvs, im gonna cry :( |
| 02:37.39 | drmessano | svn |
| 02:37.51 | troubled | oh thank god :) |
| 02:37.57 | [TK]D-Fender | troubled: Seriously... get out from under that rock... the maggots will eat you! |
| 02:38.02 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 02:38.08 | troubled | I can do svn with my eyes closed, but cvs gives me nightmares ;) |
| 02:38.29 | drmessano | So do people who describe their config files |
| 02:38.41 | troubled | heh, you dont want to see mine then ;) |
| 02:38.54 | drmessano | Thats the whole point |
| 02:39.03 | drmessano | SEEing them.. not describing them |
| 02:39.08 | troubled | I didnt even use ael for extensions |
| 02:39.31 | drmessano | Neither do 99% of us |
| 02:39.39 | jayyers | can anyone make any suggestions on a good asterisk port thats easy to use but also very flexible. ive tried switchvox free which is great but limits # of extentions and concurrent calls.... |
| 02:39.40 | *** join/#asterisk quentusrex23 (n=nobody@c-24-19-34-21.hsd1.wa.comcast.net) |
| 02:40.01 | carrar | jayyers, pay for the license |
| 02:40.06 | carrar | it works great |
| 02:40.12 | [TK]D-Fender | jayyers: "port" is not a good term. |
| 02:40.13 | troubled | [TK]D-Fender: looking at the topic, I see libpri mention 1.4.7, I gather I need to match that with asterisk 1.4? or is all that safe to mix and match with 1.6? Because I really would like to get past this 8khz issue once and for all |
| 02:40.15 | carrar | I use switchvox for several clients |
| 02:40.32 | quentusrex | I'm working on the pastebin... |
| 02:40.39 | [TK]D-Fender | troubled: asterisk.org |
| 02:40.45 | quentusrex | I've got the connection through a few ssh tunnels... |
| 02:41.07 | troubled | [TK]D-Fender: hint taken ;) thanks for all the help. time to catch up on * again :) *waves* |
| 02:41.55 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 02:42.03 | jayyers | [TK]D-Fender: lol my bad...."built" on asterisk |
| 02:42.36 | [TK]D-Fender | jayyers: GUI's only restrict what you can do... I usually advise against them |
| 02:46.21 | russellb | even though you use one yourself |
| 02:46.22 | jayyers | [TK]D-Fender: would like initial gui to help get it functional, then once functional would like to get more familiar with conf files by editing configs that i kno work, rather than blindly troubleshooting shit i dont fully understand |
| 02:46.56 | drmessano | jayyers: If your intention is to move to hand edits, using the GUI created config files is NOT the way to go |
| 02:47.05 | carrar | Why can't I just click START and everything works!!! |
| 02:47.07 | thehar | my experience is that "guis" create non-needed stuff in config files and therefore make them more confusing |
| 02:47.17 | thehar | carrar: YES WHY NOTS |
| 02:47.43 | drmessano | Much of it will be generated in a way efficient to the application generating it |
| 02:47.51 | drmessano | Not for human consumption |
| 02:48.01 | *** join/#asterisk Segnale007 (n=Pietro@host218-255-dynamic.8-79-r.retail.telecomitalia.it) |
| 02:48.55 | [TK]D-Fender | jayyers: Because once you start with a GUI it pretty much owns your ass and you find yourself backed into all sorts of corners and working within a system you can't get out of. |
| 02:49.10 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 02:50.17 | drmessano | If you're not doing anything fancy, using a GUI is fine.. You're no less restricted than any Off the shelf PBX out there from a major vendor, but you dont have the ad nauseum hand editing abilities of a raw config.. Depends on the need. |
| 02:50.42 | jayyers | drmessano: i tried starting from basic asterisk install and couldnt even get a call to terminate or originate, but got everthing else working. when trying switchvox for the first time i got the calls to terminate/originate within 10 mins.....but stuck with other issues....that i cant fix because of the inability to edit the configs |
| 02:51.09 | drmessano | jayyers: and you need to decide which is more important at that point |
| 02:51.21 | thehar | then pay for switchvox |
| 02:51.45 | [TK]D-Fender | jayyers: You shouldn't HAVE to touch the configs with a GUI. Thats the point |
| 02:52.04 | [TK]D-Fender | jayyers: Rock, hardplac. Hardplace, meet rock. |
| 02:52.05 | *** join/#asterisk james (i=james@freenode/staff/njan) |
| 02:52.12 | [TK]D-Fender | jayyers: There |
| 02:52.16 | drmessano | jayyers: GUI Asterisk install (FreePBX, Switchvox) is 5x better than say a 3COM, but you don't have the full editing capabilites to do things outside the box that asterisk does natively |
| 02:52.18 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 02:52.49 | [TK]D-Fender | jayyers: We're here to help if you're actually looking to try again. Here's a like for some inspiration as to how simple a system can be : |
| 02:52.52 | [TK]D-Fender | ~jerjerguide |
| 02:52.53 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 02:52.54 | [TK]D-Fender | ^^^ |
| 02:53.06 | jayyers | so if it was up to u guys you would jus do standard asterisk install |
| 02:53.18 | carrar | I always use from source |
| 02:53.30 | russellb | jayyers: keep in mind that you're asking some of the most hardcore asterisk users out there |
| 02:53.32 | carrar | unless the customer wants to make changes, which then I will probably use switchvox |
| 02:53.55 | jaytee | I'm so hard core I have * tattooed on my ass! |
| 02:53.56 | drmessano | jayyers: depends on the need |
| 02:53.59 | carrar | they seem to like "gui's" |
| 02:54.01 | russellb | jaytee: :) |
| 02:54.03 | jayyers | i realize that but i also dont want to take up your time with stupid n00b questions |
| 02:54.19 | jaytee | waves at russellb |
| 02:54.21 | thehar | then apt-get install asterisk-gui |
| 02:54.33 | jaytee | your blog is way out of date! been busy? |
| 02:54.35 | carrar | haha |
| 02:54.42 | russellb | hey jaytee !! |
| 02:55.00 | [TK]D-Fender | jayyers: this IS #stupidnoobquestions |
| 02:55.12 | jaytee | russellb, how've things been down Huntsville way? |
| 02:55.12 | carrar | hahah |
| 02:55.15 | russellb | jaytee: yes, very busy ... been working on some secret stuff, which is why the blog has been quiet, heh |
| 02:55.21 | [TK]D-Fender | jayyers: No #tellmeI'mmakingtherightchoiceregardless |
| 02:55.23 | carrar | Where is that short url? :) |
| 02:55.26 | carrar | shirt |
| 02:55.28 | thehar | mmm secret skype stuff?! |
| 02:55.30 | thehar | i hope so |
| 02:55.31 | jaytee | really cool secret stuff I bet |
| 02:55.32 | russellb | Huntsville is good ... way colder than Alabama should be |
| 02:55.44 | russellb | skype isn't a secret, heh |
| 02:55.49 | russellb | i haven't been working on that |
| 02:55.50 | thehar | release date is! |
| 02:55.52 | jaytee | it was cold all weekend and monday and today it got up to 50 and melted everything |
| 02:56.31 | thehar | http://www.xmission.com/~mp/tmp/alta.JPG alta this afternoon << here in utah |
| 02:56.41 | drmessano | 1.6 is gonna be a hair puller for me |
| 02:56.47 | jaytee | the older I get the closer to the equator I want to move |
| 02:56.53 | jayyers | ok so im going to try installin from asterisk from scratch but be warned i may spend many days on here requesting the help from the asterisk gods :) |
| 02:56.57 | carrar | tk, http://www.osburn.com/asterisk-shirt.png |
| 02:56.57 | jaytee | drmessano, nope, not for you |
| 02:56.59 | drmessano | It's like moving into a neighborhood with new construction going on |
| 02:56.59 | carrar | ah thememories |
| 02:57.16 | drmessano | You like your new house, but damn, that new one across the way looks better |
| 02:57.29 | drmessano | But yet, its almost the same |
| 02:57.40 | drmessano | 1.6.0 <> 1.6.1 <> 1.6.2 |
| 02:57.42 | jaytee | hahaha |
| 02:57.47 | russellb | drmessano: pretty much. |
| 02:58.00 | jaytee | so many colors! I just can't decide!!! |
| 02:58.09 | drmessano | You get all pissed off because the new ones have sunrooms |
| 02:58.18 | drmessano | or they added electroflush toilets |
| 02:58.21 | drmessano | :( |
| 02:58.30 | jaytee | "Try blue! It's the new red!" |
| 02:58.57 | drmessano | If I bought a house, and the new one across the street had the same plan + electroflush toilets, I would move |
| 02:59.04 | jaytee | those toilets at O'Hare that have the automatic ass-gaskets scare the hell out of me. |
| 02:59.05 | drmessano | Thats a deal breaker |
| 03:00.20 | drmessano | But then again.. I guess being one .1 off is better than all that "Oh you think your house is cool, the ones down the street in the trunk neighborhood are 10x better" |
| 03:00.30 | drmessano | GFY.... thats all I am saying to that |
| 03:01.41 | drmessano | I guess I got my answer to the "Wheres Oslec?" question for 1.6 |
| 03:02.00 | jaytee | really? what was the answer? no? |
| 03:02.06 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-f4b2ef7eb0c6bf93) |
| 03:02.40 | drmessano | Looks like its being integrated |
| 03:03.08 | drmessano | * drivers/dahdi/Kbuild, drivers/dahdi/dahdi_echocan_oslec.c (added), README: An experimental OSLEC echocan module. |
| 03:03.29 | drmessano | Release notes for 2.1.0 Dahdi |
| 03:04.06 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 03:04.06 | jaytee | "find the outside of the envelope, see where that ol' demon lives" |
| 03:04.25 | drmessano | I don't know if tzafrir_laptop or tzafrir is working on it, or both |
| 03:04.40 | jaytee | hehe |
| 03:04.43 | drmessano | Sometimes I wonder as to the level of assimilation |
| 03:05.00 | jaytee | clones or twins? |
| 03:05.25 | drmessano | russellb got married, became a bot.. tzafrir lives inside a laptop.. Qwell throws asterisknow out there and goes on a two month vacation |
| 03:05.29 | drmessano | :( |
| 03:05.34 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 03:05.41 | drmessano | I smell conspiracy |
| 03:05.44 | rue_mohr | so the aastra i33 is really a 9 line phone |
| 03:05.51 | rue_mohr | ? |
| 03:06.09 | rue_mohr | it just has 3 that are dedicated to lines... |
| 03:06.25 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 03:06.35 | rue_mohr | I'm having a time understanding the context of lines on ip phones |
| 03:06.46 | [TK]D-Fender | rue_mohr: I doubt is supports that many simultaneous reg's |
| 03:07.04 | [TK]D-Fender | rue_mohr: and the term "line" should not be thrown around so carelessly |
| 03:07.11 | rue_mohr | "there aren't" is a good start, but the line buttons dont help |
| 03:07.11 | drmessano | I suspect that Mark Spencer is running a giant PBX coding botnet.. One day someone will stick him in the check with something sharp and the inner doors will open, exposing a 4 inch high 500 IQ alien mastermind |
| 03:07.19 | drmessano | chest |
| 03:07.23 | [TK]D-Fender | rue_mohr: In proper terms, lines = unique registrations. |
| 03:07.41 | rue_mohr | [TK]D-Fender, so we got 2 i33's and 4 polycom 600's |
| 03:07.56 | rue_mohr | thought we would try out a few of the cheaper ones :) |
| 03:08.02 | [TK]D-Fender | rue_mohr: Others tend to think of support of SIMULTANEOUS CALLS as "lines" |
| 03:08.18 | rue_mohr | is the manual for the 600 over 1000 pages too? |
| 03:08.50 | [TK]D-Fender | rue_mohr: IP 600 = 6 line phone. each reg takes up at LEAST 1 line key, and each line key is capable of supporting up to 8 calls |
| 03:09.06 | rue_mohr | the users will be doing 1 call at a time, in all reality, so as long as there is an indicator with a button, dont' much matter to me |
| 03:09.29 | [TK]D-Fender | rue_mohr: So you could use 1 reg on a IP 600 spanning 6 keys @ 8 calls per key allowing you to potentially juggle 48 calls |
| 03:09.29 | rue_mohr | [TK]D-Fender, I'll get it yet :) |
| 03:10.08 | rue_mohr | see, ALL the people I'm working with are limited to the understanding that each co line has a button with an "in use" led |
| 03:10.12 | [TK]D-Fender | rue_mohr: It is typically best to forget the term "lines" and just look at "simultaneous calls" or a mix of how many unique reg's * calls |
| 03:10.45 | [TK]D-Fender | rue_mohr: take that idea of theirs, put it in a brown paper bag, light it on fire and stomp the shit out of it |
| 03:10.50 | rue_mohr | however I need to mimmic an archaic system, and I'm finding that more challenging than the rest |
| 03:11.03 | [TK]D-Fender | rue_mohr: You are asking for a world of pain |
| 03:11.10 | rue_mohr | I do have to bow to the uesrs |
| 03:11.27 | [TK]D-Fender | rue_mohr: What you are looking for is "key system" functionality. * was not made for this. * Does not support SLA |
| 03:11.51 | [TK]D-Fender | rue_mohr: there is a difference between bowing and bending over to get ass-raped. |
| 03:11.56 | rue_mohr | granted, but I'm confident I can make it, from the users perspective, mimmic it |
| 03:12.26 | [TK]D-Fender | rue_mohr: thats what is called "false confidence" or "blind hope". It usually comes at the end of a hot poker |
| 03:12.34 | rue_mohr | ok come now |
| 03:12.48 | [TK]D-Fender | rue_mohr: Think you're the first we've counselled on this? |
| 03:13.01 | [TK]D-Fender | rue_mohr: Far from. The outcome is pretty much invariable. |
| 03:13.09 | [TK]D-Fender | rue_mohr: But... |
| 03:13.11 | [TK]D-Fender | ~wglwat. |
| 03:13.12 | jbot | wglwat is probably well, good luck with all that |
| 03:13.25 | rue_mohr | tell me this, can I take one of the fxo channels and have a "line" key light up if that fxo channel is in use? |
| 03:13.48 | [TK]D-Fender | rue_mohr: Because the closest thing we've got a a really dirty hack that is barely workable under miniimal conditions, and not up to what a key system really offers. |
| 03:14.18 | [TK]D-Fender | rue_mohr: Yes, you can get the light.. just not the full "Oh hey, grab line 3" and "hey, I held a call on my phone, go grab it |
| 03:14.19 | drmessano | The idea is to move away from key systems |
| 03:14.27 | drmessano | Teach people what 2008 looks like |
| 03:14.30 | rue_mohr | [TK]D-Fender, understood |
| 03:14.38 | [TK]D-Fender | rue_mohr: the key system way is dead-end little crap thinking.... |
| 03:14.38 | ricko73 | parkandannounce |
| 03:14.46 | rue_mohr | [TK]D-Fender, agreed |
| 03:14.56 | rue_mohr | I want us to stop selling them |
| 03:15.03 | rue_mohr | (panasonic tda30) |
| 03:15.10 | drmessano | So dont build them with asterisk |
| 03:15.15 | drmessano | Just move away, and up |
| 03:15.25 | [TK]D-Fender | rue_mohr: Anyway, I've said my piece. You want to explore how fast you run out of road, you're more than welcome. |
| 03:15.46 | rue_mohr | this is a prolem with all the people I'm gonna be presenting these systems with |
| 03:15.48 | [TK]D-Fender | rue_mohr: first step to changing your life : CHANGE YOUR LIFE. |
| 03:16.22 | drmessano | rue_mohr: Thats a cop out.. Make them a better product |
| 03:16.37 | rue_mohr | yes, I'm trying to pillow the blow a little for people who say "JUST GET US A NORTEL 616, THEY CAN DO ANYTHING" |
| 03:16.42 | drmessano | rue_mohr: The whole "They're used to a key system" thing is old and busted |
| 03:16.54 | jaytee | dump the pillow and hit them over the head with a sledgehammer |
| 03:16.56 | [TK]D-Fender | rueyou must... unlearn |
| 03:16.59 | rue_mohr | well, can you tell me HOW I upgrade the users? |
| 03:17.17 | [TK]D-Fender | rue_mohr: by doing wht the rest of us do. PARK CALLS. |
| 03:17.26 | drmessano | rue_mohr: you want to send us the money afterwards? |
| 03:17.47 | [TK]D-Fender | rue_mohr: * doesn't care about lines. * cares about CALLS. |
| 03:17.53 | rue_mohr | [TK]D-Fender, I dont have a problem getting them to transfer cals |
| 03:18.01 | drmessano | rue_mohr: Teaching them this is like teaching them a new mail client, or a new security system with cardswipes, etc |
| 03:18.04 | ricko73 | rue_mohr: users are upgraded through a process called firing |
| 03:18.08 | drmessano | Its called sales and training |
| 03:18.10 | rue_mohr | hah |
| 03:18.11 | ricko73 | ;) |
| 03:18.13 | [TK]D-Fender | rue_mohr: Good, then parking is a few DTMF away |
| 03:18.40 | rue_mohr | so your saying I should tell the bosses that, with the new phonesystem their getting, they need to fire everyone in the company? |
| 03:18.50 | [TK]D-Fender | rue_mohr: [transfer] 700 (listen to lot), hangup. |
| 03:18.54 | drmessano | Keep making shit look like old shit, and someone will come in and sell them the real thing, and followup with the training necessary to support the sale |
| 03:18.59 | [TK]D-Fender | rue_mohr: and the yell it out. |
| 03:19.09 | drmessano | So keep making it easier for the rest of us, please |
| 03:19.18 | rue_mohr | chill guys |
| 03:20.05 | jaytee | maybe if we got someone like russellb to work on a project where we gut the code in 1.4 or 1.6 to remove most of the functionality but give you the BLF and key system features we could sell it to others. |
| 03:20.09 | rue_mohr | I feel the same way when I'm dealing with offices where people shout out "CALL FOR YOU ON LINE 2" across the room instead of pushing transfer |
| 03:20.16 | jaytee | We could call it AsterGump |
| 03:20.37 | rue_mohr | hah |
| 03:20.41 | drmessano | rue_mohr: Thats crap.. I had a 65 yr old receptionist in my old job that couldnt use the computer, but parked calls all day long |
| 03:20.53 | rue_mohr | drmessano, I KNOW! |
| 03:20.55 | [TK]D-Fender | jaytee: I'm not sure you realize the irony of that... |
| 03:20.58 | drmessano | So WTF? |
| 03:21.39 | rue_mohr | here is a question, paging, I havn't seen it in the i33 manual yet, can you give me a crash course on how it works? |
| 03:21.53 | jaytee | [TK]D-Fender, it wouldn't be the first time I missed something |
| 03:21.58 | rue_mohr | call with autopickup to speaker? |
| 03:22.29 | rue_mohr | oh there is the first mention of it I'v seen |
| 03:22.40 | [TK]D-Fender | jayHe was the one who made the hack-job that 1.4 called "SLA' |
| 03:22.46 | rue_mohr | page 431, man I'm not even half way through |
| 03:22.46 | *** join/#asterisk yardB (n=yardie@c-68-44-45-241.hsd1.nj.comcast.net) |
| 03:23.02 | [TK]D-Fender | jaytee: Meant well, but the idea falls very flat |
| 03:24.23 | drmessano | SLA is porting bad user habit to asterisk to make it more palatable... a noble effort, but I am sure it wasn't done with the same excitement as a new feature, but more as a bridge to the holdouts |
| 03:24.23 | *** join/#asterisk sucituanbo (n=fail@c-24-21-121-148.hsd1.wa.comcast.net) |
| 03:25.21 | rue_mohr | possable to have fixed call park keys and use the indicators to show if its in use? |
| 03:25.41 | rue_mohr | * without custom C programming |
| 03:26.07 | [TK]D-Fender | rue_mohr: Already done |
| 03:26.11 | yardB | question here: can anyone recommend a service provider that can terminate SIP calls to PSTN? |
| 03:26.14 | rue_mohr | cool |
| 03:26.41 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 03:26.55 | ricko73 | yardB: there are several. without knowing where you are, no one can recommend a provider |
| 03:27.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:27.09 | ricko73 | (well we could recommend a provider, but who knows how well they will work) |
| 03:28.11 | yardB | that the 64K question |
| 03:28.29 | drmessano | Upgrade to 128K |
| 03:28.42 | rue_mohr | ya you need a bigger computer dude |
| 03:28.44 | ricko73 | actually has 2 Commodore 64's |
| 03:28.54 | mmattice | anybody using lumenvox? |
| 03:28.57 | drmessano | runs a C-64 RAID... 256K |
| 03:29.33 | drmessano | Still trying to port Asterisk to the C-64.. I got as far as the menuselect screen |
| 03:30.10 | rue_mohr | good luck with those codecs |
| 03:30.13 | rcy | rue_mohr: why dont you not want to just use parked calls? |
| 03:30.28 | rue_mohr | I'll see how far I can swing the users |
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| 03:30.56 | rue_mohr | some of these people would be upgraded by getting a nortel system |
| 03:31.13 | rue_mohr | goofy 3 line analog phones |
| 03:31.50 | rue_mohr | ooo I see |
| 03:32.02 | rcy | any new system is going to mean some change for them, and work needs to be done to educate them about it, but if its done well, they should love it, and not want to go back to the old goofy system |
| 03:32.15 | rue_mohr | you hook all the leds to extensions you toggle the busyness of |
| 03:32.17 | drmessano | rcy: EXACTLY!!!!! |
| 03:32.37 | drmessano | rcy: You don't dumb down NEW, you sell NEW and train up |
| 03:32.53 | rcy | rue_mohr: you'll be swimming upstream trying to make asterisk behave like crappy legacy systems |
| 03:32.55 | rue_mohr | yes, and some cannot comprehend some things, I'v sat down with a few of them and its like smashing your head on the brick wall to try to get through |
| 03:33.22 | drmessano | rue_mohr: You're not the first person to claim "My users are worse than yours".. it's only a 30 yr old argument |
| 03:33.28 | rcy | rue_mohr: yeah, i hear that. not sure what you can do about it though |
| 03:33.33 | *** join/#asterisk BeeBuu (n=beebuu@218.13.82.213) |
| 03:33.42 | rue_mohr | maybe if I quietly dont give them what they expect, they will be so awed by the new phones they wont notice |
| 03:34.08 | drmessano | rue_mohr: It's the point of having expertise and training.. if it was so easy, they would buy it at Wal Mart |
| 03:34.19 | rue_mohr | I mean really, the bottom line is, it rings, they pick it up and say hello |
| 03:34.44 | rue_mohr | this first one is fun cause there are 4 lines with 5 different call directions |
| 03:34.47 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:35.00 | rcy | rue_mohr: there are only a handful of things people do with phones. enumerate the use cases and say for each case "on the old system you did X, on the new one do Y". like a cheat card for people who really will only learn the bare minimum anyway |
| 03:35.04 | drmessano | Bottom line is, PBX is the backbone of any company and should offer a balance of increased efficiency and learnable complexity |
| 03:35.10 | rue_mohr | I'm thinking maybe "new feature of the week" for them |
| 03:35.18 | drmessano | Saying "They pick up and talk" is missing the big picture |
| 03:35.25 | drmessano | Like saying a computer is "For typing memos" |
| 03:35.33 | rue_mohr | * is much much more complex than they can handle |
| 03:35.40 | rcy | im happy if i can have some of our people transfer a call. i dont even bother telling them about a lot of other features cause i know they wont remember, or care, or anything |
| 03:35.45 | drmessano | rue_mohr: No its not, its as complex as YOU make it |
| 03:35.51 | rue_mohr | exactly |
| 03:35.56 | drmessano | rue_mohr: YOU are the one keeping them from learning it, not them |
| 03:36.05 | rue_mohr | and I'm trying to find a compatability level |
| 03:36.05 | rcy | for those that will utilize them, i point them at documentation, or tell them about it. |
| 03:36.10 | rue_mohr | which is rather low |
| 03:36.16 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 03:36.22 | drmessano | rue_mohr: YOU build a useable system and balance the features, and it will work |
| 03:37.07 | jameswf | drmessano: http://trixbox.org/devblog/save-money-trixbox-appliance-through-dec-31 stock up |
| 03:37.07 | jameswf | drmessano: it will run HappyClownPBX |
| 03:37.07 | drmessano | rue_mohr: Sounds like you're trying to dumb it down to get out of the work of making it functional and then training on it |
| 03:37.09 | rcy | but yeah, drmessano's point is worth repeating. our first asterisk attempt was quite frustrating for all, cause the dial plan was very poorly planned. its simpler now, and people like it, i think |
| 03:37.11 | rue_mohr | you shoudl see the time I had today setting up a nortel flash for a company to do their auto answering, that bit was simple, had the lady record the propmts and all was good, BUT then I had to teach two of the guys how to use their voicemail...) |
| 03:37.30 | *** part/#asterisk yardB (n=yardie@c-68-44-45-241.hsd1.nj.comcast.net) |
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| 03:38.24 | rue_mohr | drmessano, have you ever needed to supply an asterisk system to people who had 4 phones on each desk, one for each line, and swore everything was just fine and did not need to change? |
| 03:38.25 | drmessano | rue_mohr: Then hire someone who can do sales and training.. |
| 03:38.28 | jameswf | curious who is calling in tomorrow... |
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| 03:39.12 | joako | Has anyone ever gotten Line 2 to work on a Granstream ATA? |
| 03:39.13 | drmessano | rue_mohr: I have dealt with end users for years and years.. I am well aware of how they operate, and how "my users are so much harder to deal with than yours"... trust me, they're NOT.. we've ALL dealt with users and you're not as unique as you think |
| 03:39.16 | rcy | rue_mohr: what would you like to see then? configure asterisk to work just like the current system, so you can change it and not retrain? |
| 03:39.29 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 03:40.05 | rue_mohr | rcy, if I'm not carefull, then they are gonna say the system is **** cause they dont understand it and will refuse to |
| 03:40.09 | rue_mohr | try |
| 03:40.39 | drmessano | rue_mohr: Then thats a failure on your part |
| 03:40.50 | rcy | rue_mohr: the users ive dealt with generally are fine with adopting new stuff, but are quite intolerant when things break, understandably |
| 03:40.57 | rcy | so make sure it works solid |
| 03:41.11 | rcy | and just be patient in training them on it, and ask for their patience |
| 03:41.12 | drmessano | Make sure you understand it, and train them |
| 03:41.38 | rcy | all users are resistant to change |
| 03:42.07 | jameswf | resistance is futile |
| 03:42.20 | drmessano | all users are the same users.. no one has "more difficult than yours" users |
| 03:42.52 | rue_mohr | this user, in this example is my bos |
| 03:43.02 | rcy | yeah, a "my users are so bad" pissing match is hardly productive |
| 03:43.30 | rue_mohr | I dont want to have to sell panasonic systems where OUR tech cupport spends 7 hours arguing with each other on how to make 2 differnt lines go to different ivrs |
| 03:43.55 | rcy | rue_mohr: does your boss understand that this different system is, well, a different system? |
| 03:44.01 | drmessano | rcy: Indeed. its an excuse to continue bad habits, or dumbing down a new PC to look like 95 because youre too lazy to teach them new features |
| 03:44.12 | rue_mohr | I'v only said it like a million times |
| 03:44.39 | rue_mohr | I know an office where every time windows was upgraded one of the people would throw up her hands and demand retraining |
| 03:45.16 | drmessano | rue_mohr: So you retrain her |
| 03:45.20 | drmessano | rue_mohr: Done |
| 03:45.42 | rue_mohr | our tech support for the panasonic systems does not understand call path |
| 03:45.42 | drmessano | Maybe you need to just get them Skype and both you and then users will be happy |
| 03:45.52 | rcy | we geeks can adapt to radically different stuff really quickly. users are different, so her retraining expectation is not that ridiculous |
| 03:45.54 | drmessano | It sounds like maybe YOU dont understand Asterisk enough to reliably support it |
| 03:46.17 | rcy | there really is only one option |
| 03:46.34 | rcy | leave the system alone, or be prepared to support and train users on new, better systems |
| 03:46.37 | rcy | ok, thats 2 :) |
| 03:46.39 | rue_mohr | I'm prepared to buy a support contract |
| 03:46.58 | rue_mohr | if I get stuck |
| 03:47.08 | rcy | rue_mohr: i dont envy your position though, sounds like yer neck is on the line |
| 03:47.16 | rue_mohr | yea, it is |
| 03:47.16 | drmessano | rcy: Get a Grandstream PBX = 3 |
| 03:48.14 | rcy | im pretty ignorant about all this phone stuff really. i just got tasked with setting up an asterisk based voip system at our facility, in a place where everything is changing all around all the time anyway |
| 03:48.25 | rue_mohr | asterisk, as sip is able to offer all the things to all our customers they want, thats why i'm risking my neck to get into it |
| 03:48.38 | rcy | i got to learn and break things all around my coworkers... if it was a "real" work environment, id probably be out of work |
| 03:48.51 | drmessano | rue_mohr: So stop making it look like a 10 yr old PBX and give them what it can do |
| 03:48.58 | drmessano | rue_mohr: and TRAIN them |
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| 03:49.26 | rue_mohr | drmessano, my challenge it to not overwhelm them |
| 03:49.42 | rue_mohr | and do work out how to make 3 lines and a fax number work for 4 companies |
| 03:50.14 | rue_mohr | "we dont need more lines, we can just use distinctive ring on another one" |
| 03:50.32 | rue_mohr | "if we have call forward no answer on this number, and forward it to that one..." |
| 03:50.48 | drmessano | rue_mohr: Blah blah blah.. you're preaching to the choir.. It's part of selling anything new |
| 03:50.49 | rue_mohr | 'you cant do that....' |
| 03:51.19 | rcy | rue_mohr: yeah, not sure what problem you are trying to have solved here |
| 03:51.34 | rue_mohr | specifically, none |
| 03:51.35 | drmessano | rcy: Getting out of hard work, it sounds like |
| 03:51.40 | rue_mohr | the babbel is helping me |
| 03:51.45 | rcy | you are expecting that everyone is going to hate it, which is a bad attitude to go into it with |
| 03:51.50 | rue_mohr | I'm reading the i33 manual |
| 03:52.12 | rue_mohr | I'v seen bad outcomes when people have tried what I'm trying |
| 03:52.23 | rue_mohr | but the gains are really good if I get it right |
| 03:52.25 | rue_mohr | REALLY |
| 03:52.35 | drmessano | You've already decided it's too overly complex for "your users", but the guy down the street will happily sell them the same thing and magically they will "get it" when he does.. |
| 03:52.37 | rcy | rue_mohr: think about a clean, understandable, functional way to lay things out for people. write end user documentation for it, implement it, make sure it works, and train them on it |
| 03:53.03 | rcy | rue_mohr: to make them really happy, ask them things they like and dont lie about the current system. if they had their dream system, what kinds of things wolud they have in it |
| 03:53.12 | rcy | chances are you can do that with asterisk... they will love you |
| 03:53.18 | drmessano | My point being, somehow millions of asterisk installs have gone in, and users have been upgraded.. |
| 03:53.20 | rcy | and will be willing to reprogram their fingers |
| 03:53.27 | rue_mohr | just not having to sit on a line while tech supprot spends 5 hours working out how to make two different lines ring a group of phones and go to a line specific mailbox... |
| 03:53.32 | drmessano | Your situation is no more unique |
| 03:53.47 | drmessano | rue_mohr: LEARN asterisk |
| 03:53.53 | drmessano | rue_mohr: Thats not a 5 hour job |
| 03:54.08 | rue_mohr | I'm not a complete noob you know |
| 03:54.10 | drmessano | rue_mohr: It sounds to me like YOU are the problem and not the users |
| 03:54.16 | drmessano | rue_mohr: Doesnt sound like it |
| 03:54.22 | rue_mohr | my house runs asterisk with a T1 channelbank |
| 03:54.42 | rue_mohr | all analog sets, but I have some basics |
| 03:54.43 | drmessano | But yet its gonna take 5 hours to ring 2 phones from 1 line and use a shared mailbox? |
| 03:54.48 | [TK]D-Fender | rue_mohr: And there are a million people with cars who should never be allowed to drive. |
| 03:54.55 | jaytee | your house has a channel bank? |
| 03:54.59 | jaytee | how many kids you got? |
| 03:55.07 | drmessano | If you dont know, say you dont know.. we can smell BS a mile away |
| 03:55.11 | rue_mohr | I'm good with writing extensions.conf |
| 03:55.30 | [TK]D-Fender | [22:53]<rue_mohr>just not having to sit on a line while tech supprot spends 5 hours working out how to make two different lines ring a group of phones and go to a line specific mailbox... <- then this must be a joke |
| 03:55.43 | rue_mohr | jaytee, its a 3 room house, I rent out the other rooms, the system was put in so that people could get calls at 1am without everyone ringing |
| 03:55.45 | drmessano | If you're not confident and need help, say so.. but dont tell us it takes 5 hours to set up one simple function, then claim to not be a noob.. not sure what youre trying to prove here, but its self defeating |
| 03:56.04 | rue_mohr | [TK]D-Fender, no, it was the panasonic tech support |
| 03:56.13 | loather-work | ok, so i ordered a second PRI from my provider to expand the number of B channels i'm receiving |
| 03:56.17 | drmessano | You seem scared of setting up this system and are using "difficult users' as an excuse |
| 03:56.20 | jaytee | so 3 different people need 23 b channels? thats frikken nuts |
| 03:56.23 | loather-work | i have a sangoma board with two T1 ports |
| 03:56.32 | rue_mohr | they dont understand call path, so even describing what you want a call to do is like forign to them |
| 03:56.36 | [TK]D-Fender | rue_mohr: exten => 1234,1,Dial(SIP/1&SIP/2&SIP/3,20) exten => 1234,2,Voicemail(5000@default,u) <- yippy-kay-yay. |
| 03:56.41 | loather-work | and i'm unsure how to configure it to recognize the second T1 when I have it installed. |
| 03:56.43 | jaytee | I have an OC48 here in my one bedroom apartment |
| 03:56.54 | rue_mohr | no, I have a T1 between the * machine and the channelbank |
| 03:56.56 | [TK]D-Fender | rue_mohr: There.. I saved you 5 HOURS of work. I'll only charge you for one ;) |
| 03:57.15 | rue_mohr | co line to the channelbank, off to asterisk, back to channelbank and to the sets |
| 03:57.22 | [TK]D-Fender | rue_mohr: rue_mohr T1 is a signalling, not a "thing". |
| 03:57.31 | rue_mohr | [TK]D-Fender, I know how to write extensions.conf! |
| 03:57.31 | [TK]D-Fender | rue_mohr: I can't hold a T1 in my hand. |
| 03:57.41 | [TK]D-Fender | rue_mohr: You said it before, not me.... |
| 03:57.51 | [TK]D-Fender | rue_mohr: Anyway.. |
| 03:57.58 | drmessano | You said |
| 03:58.01 | rue_mohr | thats why 5 hours of tech support with panasonic was SO frustrating |
| 03:58.06 | [TK]D-Fender | drmessano: I think I see some context in there. |
| 03:58.09 | drmessano | just not having to sit on a line while tech supprot spends 5 hours working out how to make two different lines ring a group of phones and go to a line specific mailbox... |
| 03:58.17 | rue_mohr | cause I knew exactly how to do it in asterisk |
| 03:58.42 | rue_mohr | how about telling a customer adding voicemail to their system (2 channels) will be $2000 |
| 03:59.02 | rcy | if i understand correctly then, rue_mohr's motivation for ditiching the pana system is that support sucks, his hands are tied. with * its simple. but there is the problem that users wont like this free software solution |
| 03:59.11 | ricko73 | when did this turn into the Panasonic support group? |
| 03:59.14 | [TK]D-Fender | rue_mohr: Guess teaching them how to PARK should be easy then! |
| 03:59.20 | rue_mohr | how about you customer having to spend $250+$60/mo for a device that plays on hold music? |
| 03:59.25 | drmessano | rcy: and I dont think he can support it, which it sounds like is the issue.. |
| 03:59.51 | rcy | i think he's just nervous about the users response to the change |
| 03:59.58 | rue_mohr | I beleive if I work at it I can |
| 04:00.07 | drmessano | rcy: Projecting |
| 04:00.13 | [TK]D-Fender | rue_mohr: see that "blind faith" comment I made earlier |
| 04:00.21 | rue_mohr | I know I will occasionally get stuck on something, this 1000+ manual is telling me that |
| 04:00.25 | [TK]D-Fender | rue_mohr: how many *cough* .... LINES worth? |
| 04:00.28 | rcy | a lot of us have experienced that, and can safely say, its a surmountable problem. dont worry rue_mohr, if you set things up well, itll be fine |
| 04:00.59 | [TK]D-Fender | rue_mohr: And waht phones are you looking to use? |
| 04:01.00 | drmessano | Its like anything else... if you learn it, you can confidently train the users on it |
| 04:01.08 | rue_mohr | [TK]D-Fender, the ones we just recieved |
| 04:01.14 | drmessano | .... |
| 04:01.16 | [TK]D-Fender | rue_mohr: ... |
| 04:01.18 | rue_mohr | you got me to get polycom 600's |
| 04:01.21 | drmessano | Ba-dump-CHING |
| 04:01.43 | [TK]D-Fender | rue_mohr: Don't recall ever recommending those unless you got a great deal on them |
| 04:01.46 | rue_mohr | but I subbed two low-demand users with aastra i33's |
| 04:01.57 | [TK]D-Fender | rue_mohr: how many *cough* .... LINES worth? <--- |
| 04:02.02 | rue_mohr | we only paid $260 ea for them |
| 04:02.13 | [TK]D-Fender | rueyou got BURNED then |
| 04:02.20 | rue_mohr | did I mention the panasonic phones are $280 ea? |
| 04:02.24 | rue_mohr | isdn... |
| 04:02.36 | [TK]D-Fender | rue_mohr: Shit looks pretty good.... when compared to CRAP |
| 04:02.56 | jaytee | I want to build up my voice network so that I have 3 or 4 of every kind of phone made because I have way too much free time to just standardize on one brand with a couple models |
| 04:03.02 | [TK]D-Fender | rue_mohr: http://www.telephonydepot.com/Catalog/Polycom-IP-Phones;jsessionid=0a01025a1f43fa4ad3712f2143c6863b8b80ac117973.e3eSbNySbxiNe34Pa38Ta38Ochr0 |
| 04:03.06 | rue_mohr | am I painting a descent picture of what I'm having to put up with currently? |
| 04:03.07 | brut- | whats the difference between asterisk 1.4 and 1.6 anyway? |
| 04:03.23 | jaytee | brut- , .2 |
| 04:03.28 | rue_mohr | [TK]D-Fender, usa, border fees will kill any savings |
| 04:03.28 | brut- | I really couldn't find a doc on asterisk.org that says "this is what 1.4 is, this is what 1.6 is" |
| 04:03.29 | [TK]D-Fender | rue_mohr: $237 for an IP 601. Expandable higher model for less money. And even then far more than most users need |
| 04:03.40 | rue_mohr | I'm in cad over here |
| 04:03.46 | brut- | thanks jaytee, that helps :P |
| 04:04.02 | jaytee | brut-, 1.4 is stable 1.6 is new |
| 04:04.02 | [TK]D-Fender | brut-: "upgrade.txt" readme" "changelog". the obvious commentary in the SAMPLE configs |
| 04:04.12 | jaytee | 1.6 has support for SIP TCP. 1.4 does not |
| 04:04.21 | rue_mohr | I'm on the west end of canada, the only place I could get phones within canada was the far east end |
| 04:04.22 | drmessano | $138.50 for the Aastra from telephony depot |
| 04:04.25 | [TK]D-Fender | rue_mohr: Like we weren't past par recently... |
| 04:04.29 | drmessano | and you pay $260? |
| 04:04.32 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 04:04.34 | [TK]D-Fender | rue_mohr: IP 601 is overkill for most. |
| 04:04.40 | rue_mohr | 600 |
| 04:04.50 | rue_mohr | cause they have 6 line buttons |
| 04:05.07 | rue_mohr | and I didn't ahve a manual to read to know if the other ones could be programmed as lines |
| 04:05.18 | rue_mohr | which in hindsight... |
| 04:05.24 | rue_mohr | but too late for that now |
| 04:05.28 | drmessano | http://www.telephonydepot.com/Catalog/Aastra-Phones/Aastra-9143i |
| 04:05.31 | [TK]D-Fender | "blindsight" |
| 04:05.45 | rue_mohr | yaya. I'm gonna make mistakes, this is my first pass |
| 04:06.02 | rue_mohr | I just need to make the mistakes surviveable |
| 04:06.13 | [TK]D-Fender | drmessano: meh... a 9133 refresh... |
| 04:06.19 | rue_mohr | so far "remember to order computer in time" is on the top of my list |
| 04:06.25 | drmessano | But you're no noob, and seem to be insistant on that.. which is making it nearly impossible to get any ideas into your head because you got it all worked out |
| 04:06.31 | [TK]D-Fender | drmessano: overpriced for their 2nd rate call handling, puny screen, etc |
| 04:06.45 | drmessano | [TK]D-Fender: I could care less, was making a price comparison |
| 04:06.49 | [TK]D-Fender | drmessano: but makes ancient PBX owners feel more "at home" |
| 04:06.50 | rue_mohr | yes, but they all ooood over how they looked like nortel sets |
| 04:06.51 | rue_mohr | :/ |
| 04:07.03 | brut- | [TK]D-Fender, I didn't find anything relevant in any of those docs that say what the difference is between the 2..., each one obviously has changes for that branch, but nothing that compares the 2 |
| 04:07.06 | brut- | I'll check the wiki though |
| 04:07.15 | drmessano | rue_mohr: You're already starting off on the wrong foot here |
| 04:07.24 | rue_mohr | I still have feet? |
| 04:07.37 | drmessano | rue_mohr: yes, they're in a jar somewhere |
| 04:07.46 | [TK]D-Fender | brut-: Should your new 2009 Ford Mustang have a "whats new since 1967" chapter? |
| 04:07.50 | subdolus | raises the jar |
| 04:07.58 | drmessano | subdolus FTW |
| 04:07.59 | [TK]D-Fender | brut-: Because who feels like comparing ancient history? |
| 04:08.25 | brut- | [TK]D-Fender, _both_ are still offered and updated, thats why I'm comparing |
| 04:08.28 | jameswf | let me be the first to say WOW!! www.blackbirdhome.com |
| 04:08.38 | rue_mohr | I'm gonna see if I can get one of these sets set up for the sip account out of my house from work |
| 04:08.40 | [TK]D-Fender | brut-: Got an "Upgrade guide from DOS 2.11 to Windows Advanced Server 2008" leaflet hanging around? |
| 04:08.42 | drmessano | I want a browser for white people |
| 04:08.55 | brut- | perhaps if 1.4 wasn't still on the _FRONT_ page of asterisk.org, I wouldn't ask |
| 04:08.57 | rue_mohr | I think I see everything I need to set |
| 04:08.58 | [TK]D-Fender | brut-: No... 1.2 is **EOL** |
| 04:09.05 | brut- | I said 1.4 |
| 04:09.08 | brut- | not 1.2 |
| 04:09.10 | [TK]D-Fender | brut-: it is NOT getting bug fixes, only SECURITY |
| 04:09.12 | drmessano | raises the "Upgrade guide from DOS 2.11 to Windows Advanced Server 2008" leaflet |
| 04:09.27 | rue_mohr | whats this "blue screen of death?" |
| 04:09.42 | rue_mohr | dos didn't do that... |
| 04:09.59 | drmessano | I want to make CD's with the "whole internet" on them.. which basically autorun http://www.google.com |
| 04:10.03 | [TK]D-Fender | brut-: All the docs from 1.4 to 1.6 are in the doc folder in 1.6 along witht he clear changelogs, etc. |
| 04:10.34 | brut- | [TK]D-Fender, in the 1.6 zip, aye, I just found it after some google-fu |
| 04:10.39 | rue_mohr | good point versions, what version you reccomend I use, least surprises and all? |
| 04:10.45 | brut- | and via the svn |
| 04:10.58 | [TK]D-Fender | rue_mohr: 1.4 for now |
| 04:11.02 | rue_mohr | k |
| 04:11.11 | [TK]D-Fender | rue_mohr: and http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-320 is plenty for 95% of users |
| 04:11.15 | brut- | more difficult than it needed to be imho, but I'm picky like that when it comes to docs... :P |
| 04:11.20 | brut- | thanks for the help [TK]D-Fender |
| 04:11.48 | rue_mohr | [TK]D-Fender, I was confused about "line buttons" |
| 04:11.50 | [TK]D-Fender | rue_mohr: I use an IP320 at my desk instead of an IP 600 like I used to have because the difference didn't matter to my life. |
| 04:11.54 | rcy | whats the cheapest, most featureless handset you can get? |
| 04:11.59 | [TK]D-Fender | rue_mohr: Ask next time |
| 04:12.20 | [TK]D-Fender | rcy: some cheap POS you buy at your drugstore |
| 04:12.23 | rue_mohr | they were only an extra $100 on the aastra sets |
| 04:12.25 | rue_mohr | ea |
| 04:12.41 | [TK]D-Fender | rue_mohr: and as you can see, the 320 clocks in at < $90 USD |
| 04:13.03 | rcy | [TK]D-Fender: well, with ip |
| 04:13.09 | jaytee | [TK]D-Fender, ever used one of the sidecars for polycoms? |
| 04:13.12 | rue_mohr | and I suspect like the aatra set you can program the extra buttons up as "line" keys |
| 04:13.13 | rcy | [TK]D-Fender: maybe my drugstore has such things, havent looked for awhile |
| 04:13.47 | [TK]D-Fender | jaytee: My admin assistant has a fully loaded IP 601 w/ 3 |
| 04:13.47 | rcy | im rocking a set of grandstream budge tones these days, but they are too fancy for me |
| 04:13.53 | rue_mohr | "Asterisk 1.2.15, Copyright (C) 1999 - 2006 Digium, Inc. and others.", hmmmm... |
| 04:14.09 | [TK]D-Fender | rcy: Why looking for LESS? |
| 04:14.20 | rcy | cause afaik they are still something like 80 bucks |
| 04:14.33 | drmessano | SPA-941s are $83 |
| 04:14.37 | drmessano | and are awesome phones |
| 04:14.37 | rcy | hmm |
| 04:14.42 | rcy | good to know |
| 04:15.08 | [TK]D-Fender | rcy: http://www.telephonydepot.com/Catalog/Linksys-Phones/Linksys-SPA901 |
| 04:15.19 | rcy | im using salvaged dlink 1120 ata boxes and pots phones too |
| 04:15.34 | rue_mohr | I'v worked out my software sip problems at home are due to a sucky sound card driver |
| 04:15.42 | rcy | i have a zero dollar budget at a nonprofit computer recycler, so we use what we can find and hack into the system |
| 04:15.50 | rue_mohr | rcy, how are those working out? |
| 04:15.53 | [TK]D-Fender | drmessano: SPA-942 is a pretty good value these days. |
| 04:16.06 | rcy | oh, that spa901 looks decent |
| 04:16.07 | [TK]D-Fender | drmessano: when I got my 941 about 2-3 years ago it was $150 |
| 04:16.12 | rue_mohr | I got transfering working at the hosue |
| 04:16.19 | [TK]D-Fender | rcy: the correct answer is "cheap POS". |
| 04:16.27 | rcy | [TK]D-Fender: perfect for a public phone in the warehouse |
| 04:16.38 | [TK]D-Fender | rcyWhy are you looking to spend real money on a phone thats no better than an ATA + POTS? |
| 04:16.39 | rcy | dont need any ability but to ring and make calls |
| 04:16.54 | drmessano | [TK]D-Fender: They've even come down $15 in the last 3 months |
| 04:16.58 | [TK]D-Fender | rcy: You're better off with an ATA + analog phone |
| 04:17.02 | rcy | [TK]D-Fender: because ata's arent exactly dropping from the sky either |
| 04:17.06 | rcy | [TK]D-Fender: yes that is true |
| 04:17.12 | drmessano | ATA's are dirt cheap |
| 04:17.16 | drmessano | $40 |
| 04:17.22 | rue_mohr | one day I'd like my asterisk system to have 2 or more lines |
| 04:17.34 | drmessano | rue_mohr: you said it had a T1? |
| 04:17.34 | [TK]D-Fender | rcy: http://www.telephonydepot.com/Catalog/Linksys-Analog-Adapters/Linksys-PAP2T-NA |
| 04:18.04 | rcy | [TK]D-Fender: i have a bunch of non -na pap2t's that i need to spend a bit of time unlocking |
| 04:18.12 | [TK]D-Fender | rcy: rcy + a $10 cheap-shit phone you won't care that they abuse. And the ATA can support *2* phones |
| 04:18.17 | drmessano | PAP2s, not 2Ts |
| 04:18.19 | rcy | [TK]D-Fender: but yeah, if i was going to throw down a couple 20 dollar bills, those would be the way to go |
| 04:18.37 | [TK]D-Fender | rcy: better than buying a dead-end POS like th 901 |
| 04:18.40 | rcy | i was just wondering if there existed $20 ip phones |
| 04:18.56 | [TK]D-Fender | rcy: the "Don't care if they abuse" factor should not be extended that far. |
| 04:19.01 | rue_mohr | drmessano, T1 betweent eh asterisk machine and the channelbank |
| 04:19.05 | [TK]D-Fender | rcy: Sure... in China |
| 04:19.10 | rcy | im happy with my crappy dlink ata and my rotary phones |
| 04:19.21 | rue_mohr | the channelbank has a fxo card and a few fxs cards |
| 04:19.24 | rcy | so yeah, good to know, im on the right track with that. ill stay away from ip phones for those public phones |
| 04:19.35 | drmessano | rue_mohr: You bought a T1 card and a channelbank vs using a couple ATAs or multi-FXS card? |
| 04:19.36 | rue_mohr | got the rotary to work? |
| 04:19.44 | rcy | rue_mohr: yep. |
| 04:19.46 | rue_mohr | drmessano, $free |
| 04:19.59 | [TK]D-Fender | rcy: I bought 2 Uniden UIP-200's back in the day for those "high-rape-risk" areas. BIG regret |
| 04:20.30 | rcy | yeah my analog stations just work. the budge tone crap is just that |
| 04:20.35 | [TK]D-Fender | rcy: So don't buy dead-end crap. Even an ATA+phone > SPA-901 |
| 04:20.36 | rcy | you get what you pay for i guess |
| 04:20.46 | rcy | [TK]D-Fender: k. nuff said. thanks. |
| 04:20.47 | [TK]D-Fender | ~ygwypf |
| 04:20.48 | jbot | rumour has it, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 04:20.53 | [TK]D-Fender | ^^ |
| 04:20.54 | [TK]D-Fender | indeed |
| 04:21.58 | rue_mohr | by the way, I dont know if I said I was successfull in hooking my asterisk machine into my home automation system to put the message waiting lamps on the front door |
| 04:22.04 | rue_mohr | I was |
| 04:22.12 | rue_mohr | still am come to think of it |
| 04:22.26 | thehar | DEAR XO: fix my trunk group. |
| 04:23.11 | rcy | rue_mohr: i can recieve calls and dial out with the rotary phone, through the dlink 1120s, but thats about it. the bell ringer sounds a bit pathetic |
| 04:23.19 | [TK]D-Fender | rue_mohr: My * used to make me COFFEE. Coffee > MWI :) |
| 04:23.37 | rcy | i used to flush the toilet with my cell phone |
| 04:24.00 | [TK]D-Fender | rcy: I thought about that... but decided if I was ever that lazy, someone should shoot me. |
| 04:24.35 | rue_mohr | rcy, when you put it like that it sounds usefull incase you think you forgot |
| 04:24.39 | rue_mohr | hmm |
| 04:24.43 | drmessano | rcy: the weak sound of the bell ringer is your ATA dying |
| 04:24.46 | drmessano | lol |
| 04:24.49 | rue_mohr | so maybe I should jig up my tea maker? |
| 04:24.56 | jaytee | my asterisk server uses AGI scripts with System() to run an RF remote controller adapter card that's linked in with a GPS transmitter to drive and control the lawnmower. |
| 04:25.07 | drmessano | HAW |
| 04:25.16 | drmessano | "I HIT 5.. WTF.. I HIT 5!!!!" |
| 04:25.19 | [TK]D-Fender | jaytee: You ARE the shiznit y0! |
| 04:25.21 | rue_mohr | jaytee, giz, hope its low delay |
| 04:25.21 | drmessano | "NOOOOOO" |
| 04:25.26 | [TK]D-Fender | drmessano: lol |
| 04:25.26 | rcy | drmessano: i thought it was just not being able to supply enough power |
| 04:25.46 | rue_mohr | I might have the lawnmower 802.11 by mid next summer |
| 04:25.50 | drmessano | rcy: Not supplying enough power = pulling too much current |
| 04:25.55 | rue_mohr | the new drive trains are working out good |
| 04:25.59 | drmessano | rcy: I've lost ATA's that way.. |
| 04:26.13 | drmessano | rcy: REN is not a suggestion, it's the law.. lol |
| 04:26.36 | rue_mohr | oh serious question, the tdm800 card I got, How do I know what module is doing which for what port? |
| 04:26.42 | jaytee | rue_mohr, 802.11g I hope! b is too slow, especially when cornering which is where bandwidth is critical |
| 04:26.45 | [TK]D-Fender | drmessano Its all marketing. Who'd buy into "Suggestions of physics"? |
| 04:26.47 | rue_mohr | I dont think I saw it in the manual |
| 04:27.01 | rcy | heh, yeah, i should not ring that station. but i keep it around for laughs, and its nice that you can pick it up and dial out |
| 04:27.15 | rue_mohr | jaytee, its a really slow mower, right now its driven by a qbasic program( wanders aimlessly) |
| 04:27.15 | [TK]D-Fender | rue_mohr: red = FXO |
| 04:27.29 | rue_mohr | k, thats the card |
| 04:27.34 | drmessano | ... |
| 04:27.38 | rue_mohr | how do I track that to a port ont eh back? |
| 04:27.43 | rue_mohr | 1-8 |
| 04:27.53 | rue_mohr | I ahve 4 fxo and 2 fxs |
| 04:28.05 | [TK]D-Fender | rue_mohr: read the manual that tels you how the ports map to the physical order of module jacks on the back. |
| 04:28.11 | rue_mohr | k |
| 04:28.13 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 04:28.45 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 04:30.20 | rue_mohr | does it use zapata.conf for the tdm800? |
| 04:30.31 | [TK]D-Fender | ... |
| 04:30.36 | [TK]D-Fender | its a ZAPTEL CARD. |
| 04:30.51 | rue_mohr | digium card |
| 04:30.53 | [TK]D-Fender | rue_mohr: so no... it uses mgcp.conf |
| 04:31.00 | jaytee | rofl |
| 04:31.13 | rue_mohr | frowns and scratches head |
| 04:31.18 | drmessano | wait |
| 04:31.21 | drmessano | duh |
| 04:31.22 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 04:31.26 | [TK]D-Fender | ~iwmwb |
| 04:31.27 | jbot | I WANT MY WEEKEND BACK! |
| 04:31.32 | ricko73 | rue_mohr: it's called sarcasm |
| 04:31.35 | *** join/#asterisk jicksta_ (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 04:31.42 | drmessano | genmgcpconf doesnt work |
| 04:31.48 | rue_mohr | I had to work 5.5 hours on sat. |
| 04:32.04 | [TK]D-Fender | drmessano: OMG, the "Go" button just disappeared off my desktop, what do I do?! |
| 04:32.09 | drmessano | wonders if rue_mohr will learn to not claim he's not a newb around those with sense |
| 04:32.27 | ricko73 | [TK]D-Fender: replaced no doubt by the 'easy' button |
| 04:32.37 | rue_mohr | zaptel chipset on digium card? |
| 04:32.40 | rue_mohr | :) |
| 04:32.43 | rue_mohr | symantics |
| 04:32.45 | jaytee | wait a second here! let me get this straight. the guy has a friggin channel bank with a T1 in his house connected up to Asterisk yet he doesn't know if the tdm800 card uses zapata.conf? BUSTED!!!!! |
| 04:32.47 | [TK]D-Fender | rue_mohr: I did 12 hours on Sat between PC profile migrations and a Lotus Domino rebuild/upgrade. Join the club |
| 04:33.02 | jaytee | (loud buzzer sound) Sorry, no prize for you |
| 04:33.07 | *** join/#asterisk dazoe (n=dazoe@64.68.188.167) |
| 04:33.16 | ricko73 | aaawwwe |
| 04:33.16 | drmessano | Yes, and 5 hours to do a DIAL & & then send to voicemail? |
| 04:33.23 | drmessano | DOUBLE BUSTAD |
| 04:33.28 | rue_mohr | I got to run back and forth across town dealing with emergency tel and data wiring issues |
| 04:33.32 | [TK]D-Fender | drmessano: No, that was OTHER techs... and with another PBX..... |
| 04:33.33 | jaytee | POSER!!! |
| 04:33.46 | [TK]D-Fender | jaytee: ouch |
| 04:33.58 | rue_mohr | my main job is pulling and terminating wire |
| 04:33.59 | jaytee | [TK]D-Fender, not you! |
| 04:34.07 | dazoe | I'm having problems with MusicOnHold, i'm not sure where to start looking... |
| 04:34.07 | drmessano | [TK]D-Fender: I know.. and this has never happened to me before either.. must be the cold air |
| 04:34.14 | jaytee | we call them attic monkeys |
| 04:34.28 | rue_mohr | dazoe, musiconhold.conf? |
| 04:34.29 | drmessano | [TK]D-Fender: or the pool water.. you know, shrinkage |
| 04:34.30 | ricko73 | is shocked |
| 04:34.52 | [TK]D-Fender | jaytee: that I know. Besides.... noone can make you feel inferior without your consent. And you know the best part of my God Complex? No peer pressure :p |
| 04:35.00 | thehar | haha |
| 04:35.05 | jaytee | hahahaha |
| 04:35.13 | drmessano | rue_mohr has a serious case of asterisktile dysfunction.... |
| 04:35.22 | [TK]D-Fender | jaytee: but damn... its lonely at the top... |
| 04:35.28 | ricko73 | I certainly would have guessed that rue_mohr was more of a C developer than a wire guy |
| 04:35.34 | [TK]D-Fender | drmessano: Just can't get an analog up... |
| 04:35.46 | jaytee | [TK]D-Fender, and people look at me funny when say "I've got an Asterisk problem I need to speak to God about" |
| 04:35.48 | drmessano | ricko73: No, he doesnt develop in C, he WROTE C |
| 04:35.51 | rue_mohr | heh, i was a programmer, market crashed |
| 04:35.53 | dazoe | rue_mohr: yeah i looked at that... and i have mp3's in the mohmp3 dir |
| 04:36.05 | thehar | permissions? |
| 04:36.07 | ricko73 | rue_mohr == algore? |
| 04:36.20 | rue_mohr | dazoe, if you dial verbose and debug up on the console do you see if crying bout anything? |
| 04:36.31 | drmessano | ricko73: Only if Al Gore invented the Blackberry too, like John McCain |
| 04:36.45 | [TK]D-Fender | dazoe: Perhaps you could actually describe the problem. |
| 04:36.47 | jaytee | I'd just like to go on record that I did not invent the internet........ I did however invent wire. |
| 04:36.50 | dazoe | rue_mohr: i see start musiconhold and right after stop |
| 04:37.12 | joako | dazoe: asterisk 1.2 used mohmp3, 1.4 uses /var/lib/asterisk/moh by default |
| 04:37.12 | drmessano | I'm no newb, but I gotta tell ya.. I don't think your horn has enough fluid, and I think the transmission needs to be recharged |
| 04:37.14 | [TK]D-Fender | jaytee: I didn't invent the internet. I jsut made it BETTER. |
| 04:37.25 | [TK]D-Fender | waits for the TM police to show up |
| 04:37.46 | jaytee | [TK]D-Fender, the BASF of networking and voip |
| 04:38.09 | drmessano | wonders if his $2500 matched tires were a ripoff |
| 04:38.11 | dazoe | joako: yeah i checked the dirs and stuff, i have the config set to mohmp3 and my mp3 files are there |
| 04:38.17 | loather-work | http://pastebin.ca/1281351 <-- i have a PRI connected to my asterisk box. I'd like to add a second channel span with 24 B channels to the existing PRI. Would these be the right edits in order to accomplish that? |
| 04:38.30 | joako | dazoe: did you install asterisk-extras for MP3 support? |
| 04:38.43 | [TK]D-Fender | dazoe: * does not support MP3 by default. this requires the asterisk-addons package |
| 04:38.50 | joako | dazoe: Try this: http://app5.netjdn.com/~joako/sounds/SampleAudioSource.ulaw.wav rename it to .ulaw instead of .ulaw.wav |
| 04:39.03 | [TK]D-Fender | dazoe: Go install that if you haven't already. |
| 04:39.23 | thehar | loather-work: you'd need to specify a d chan for the 2nd pri |
| 04:39.23 | [TK]D-Fender | dazoe: you can confirm via "core show modules like format" |
| 04:39.35 | loather-work | thehar: it's a single PRI spanning two T1s |
| 04:39.45 | dazoe | maybe that's it, i need add on |
| 04:39.52 | jaytee | wow, the guy in the picture on Polycom's home page looks like Ben Stein after AIDS and chemotherapy |
| 04:40.20 | joako | Anyone here know that Pat Fleet recorded a complete set of Asterisk sounds? |
| 04:40.27 | [TK]D-Fender | loather-work: Depends how your telco set it up |
| 04:40.37 | [TK]D-Fender | loather-work: Most won't want to use NFAS |
| 04:40.45 | [TK]D-Fender | loather-work: (Shared D) |
| 04:40.58 | [TK]D-Fender | loather-work: they usually just overflow to your 2nd PRI |
| 04:41.11 | loather-work | yeah, i want to have a shared D. |
| 04:41.17 | [TK]D-Fender | loather-work: And it does increase the risk of calls dropping |
| 04:41.37 | [TK]D-Fender | loather-work: thats what SPANMAP is for. Go read the sample configs really closely |
| 04:41.44 | rue_mohr | [TK]D-Fender, well hope you dont hate me yet, have a good night |
| 04:41.57 | [TK]D-Fender | rue_mohr: Not yet... plenty of time for that later ;) |
| 04:42.05 | thehar | hate is strong with this one. |
| 04:42.22 | rue_mohr | ;) |
| 04:42.24 | [TK]D-Fender | rue_mohr: Try coming in here before getting you head buried too deep, ok? :) |
| 04:42.35 | rue_mohr | oh ok |
| 04:42.46 | [TK]D-Fender | me fries thehar with force-lightning |
| 04:42.51 | thehar | eeeep |
| 04:42.53 | jaytee | who is Pat Fleet? |
| 04:42.53 | thehar | dodges |
| 04:42.56 | rcy | rue_mohr: whats your timeline on getting the system online? im gonna be living in your neck of the woods come january |
| 04:43.08 | rcy | id be happy to help out, where i can, though im a noob too :) |
| 04:43.09 | rue_mohr | rcy, you moving over!? |
| 04:43.22 | rcy | yeah, for awhile anyway |
| 04:43.31 | rue_mohr | well, the deadline for getting it all set up and going is week before last. |
| 04:43.43 | [TK]D-Fender | rcy: Misery meet company! Now co-Misery-ate ;) |
| 04:43.50 | rcy | hehe |
| 04:43.53 | rue_mohr | rcy, you have a place to stay? |
| 04:44.02 | drmessano | Dont do it |
| 04:44.05 | [TK]D-Fender | This isn't #ASL |
| 04:44.05 | rcy | rue_mohr: yeah, ill be couch surfing around |
| 04:44.06 | jaytee | i get the feeling those two will be sharing more than a room come February |
| 04:44.07 | [TK]D-Fender | get a room! |
| 04:44.10 | thehar | haha jaytee |
| 04:44.15 | rue_mohr | pffft |
| 04:44.22 | drmessano | rue_mohr will try to trade kinky sex for dialplans |
| 04:44.24 | rcy | haha |
| 04:44.33 | jaytee | rue_mohr, any relation to rue_paul? |
| 04:44.43 | drmessano | Go ahead and order the "exit only" shirt.. NOW |
| 04:44.47 | rue_mohr | best I rent the room out to someone I can regretlessly overcharge |
| 04:44.51 | rue_mohr | no |
| 04:44.54 | rcy | will hack for a place to stay |
| 04:45.01 | loather-work | [TK]D-Fender: thanks for the pointer in the right direction. I'm editing the configs again. |
| 04:45.04 | thehar | what if he wants it as an "entrance"? |
| 04:45.05 | drmessano | rcy: and get hacked? |
| 04:45.17 | [TK]D-Fender | loather-work: You sure the telco has set this up? |
| 04:45.35 | rue_mohr | I am Rue Nahc Mohr, conquerer of that which is easily and quickly conquered and that nobody else wants, or hasn't shown any interest in yet |
| 04:45.43 | loather-work | [TK]D-Fender: it's not set up yet. |
| 04:45.45 | [TK]D-Fender | loather-work: Normally not a good thing to risk for a 4% gain. |
| 04:45.50 | drmessano | "<rue_mohr> I don't know anything about asterisk, my sexy rcy, but let me show you what I do know....." |
| 04:46.01 | drmessano | Bow-wow-chiga-bow-wow |
| 04:46.03 | dazoe | my problem was MusicOnHold uses mpg123 which was not installed... |
| 04:46.15 | [TK]D-Fender | rue_mohr: that was actually funny. |
| 04:46.17 | loather-work | http://pastebin.ca/1281357 <--- This look any better? |
| 04:46.24 | rue_mohr | ah, well glad to have helped |
| 04:46.34 | joako | jaytee: Pat Fleet is the lady that used to do alot of telco recordings: http://www.youtube.com/watch?v=D2cN2CMMnVw |
| 04:46.36 | [TK]D-Fender | dazoe: Depends... you really should AVOID mpg123 in the first place. |
| 04:46.46 | [TK]D-Fender | dazoe: "mode=files" <- use * native MoH |
| 04:46.55 | *** join/#asterisk sasargen (n=chatzill@173.100.55.82) |
| 04:46.56 | [TK]D-Fender | dazoe: and go install asterisk-addons as we've advised |
| 04:47.21 | jaytee | [TK]D-Fender, so if I want to replace my Nortel CAP which is an M3904 what do you think of a 650 with 1 sidecar for speed dial/BLF for the most commonly used numbers? |
| 04:47.49 | [TK]D-Fender | loather-work: Looks kinda right, but I've never actually done this myself. I've read it over a few times a while back though. |
| 04:48.03 | loather-work | ok, when the time comes to light it up i'll give that a shot. |
| 04:48.10 | [TK]D-Fender | jaytee: For phone #'s? Iw ouldn't waste the money on it. |
| 04:48.22 | PlItS | Hi to all again .... any one can give me an idea or literature on how to integrate sugarCRM with asterisk ... had it in my trixbox BOX now with a asterisk box cant seem to get around it ? |
| 04:48.24 | [TK]D-Fender | jaytee: thats a LOT of money for something you aren't going to use Presence on |
| 04:48.30 | loather-work | reason i'm doing it this way is the last time i had the telco make changes to the span they totally blew it and like half the DIDs weren't being routed properly. |
| 04:48.31 | [TK]D-Fender | jaytee: Seriously... don't |
| 04:48.54 | jaytee | so just a 650 with no sidecar and have them just transfer |
| 04:49.11 | loather-work | so i figured increasing the number of channels would be a safer bet than adding a second PRI they could screw up again |
| 04:49.15 | [TK]D-Fender | jaytee: better to do a dial-plan speed-dial type setup |
| 04:49.26 | [TK]D-Fender | jaytee: Maybe not even a 650 |
| 04:49.59 | loather-work | basically, i don't want to get the 4AM phone call from the customer service department complaining that none of the phone calls are coming through :( |
| 04:50.06 | [TK]D-Fender | loather-work: Sounds like a 1 shot mistake that once fixed life goes on. NFAS risk is forever. |
| 04:50.21 | [TK]D-Fender | loather-work: And if 1 goes downn, at least you're not toast |
| 04:50.36 | loather-work | what's the risk with NFAS? |
| 04:50.50 | [TK]D-Fender | loather-work: PRI with your D chan goes down, you lose EVERYTHING |
| 04:50.51 | jaytee | [TK]D-Fender, most of our lines are DID so the CAP doesn't get an enormous volume of calls during the day and we also have the autoattendant with dial by name voice recognition so I'm thinking just a 550 or a 650. |
| 04:51.16 | loather-work | fortunately the CPE and NE are in the same facility, a cabinet away |
| 04:51.19 | [TK]D-Fender | jaytee: How often will they juggle 4 calls? |
| 04:51.28 | loather-work | so if one goes down the other's going down too |
| 04:51.33 | jaytee | [TK]D-Fender, not that often |
| 04:51.40 | jaytee | maybe once or twice a day |
| 04:51.44 | [TK]D-Fender | loather-work: jsut because they are close doesn't mean it won't mean nasty down-time |
| 04:52.06 | [TK]D-Fender | jaytee: and IP 301 could do that ;) |
| 04:52.08 | loather-work | yeah, the provider only has the one piece of equipment |
| 04:52.18 | loather-work | so if it goes down it'll be nasty downtime either way :\ |
| 04:52.26 | [TK]D-Fender | jaytee: or 430,450, 5XX, or 6XX :) |
| 04:52.36 | jaytee | [TK]D-Fender, hmmm, I'd use a 330 but I'm staying away from the 301 |
| 04:52.41 | [TK]D-Fender | loather-work: Just laying that out there for you to consider.... |
| 04:52.55 | [TK]D-Fender | loather-work: for a "burstable resource like that I really wouldn't. |
| 04:53.12 | jaytee | I'd pretty much made a standard setup for 330's for the majority of phones and a few 550's for secretarys |
| 04:53.15 | [TK]D-Fender | loather-work: You've got what you need to know now though. best of luck which ever way you go |
| 04:53.35 | *** join/#asterisk CunningPike_ (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:53.49 | loather-work | [TK]D-Fender: thanks, and i really do appreciate the input. it's good to have a second set of eyes look at it |
| 04:53.54 | [TK]D-Fender | jaytee: For the biggest, a 650 would not be out of order... but I wouldn't pay for a side-car unless you are considering maybe 2 of them for presence. |
| 04:54.06 | [TK]D-Fender | jaytee: general speed-dials is kinda... cheap |
| 04:54.21 | [TK]D-Fender | jaytee: Now if you asked that about an AASTRA... that would be another matter |
| 04:54.34 | [TK]D-Fender | jaytee: Aastra's attendant module is GODLY |
| 04:55.04 | [TK]D-Fender | jaytee: http://www.telephonydepot.com/Catalog/Aastra-Phones/Aastra-560M-Expansion-Module |
| 04:55.12 | jaytee | [TK]D-Fender, I don't think I'll be using Presence functionality at that position. Their job is just to answer the main line and transfer to whomever when necessary. |
| 04:55.25 | [TK]D-Fender | jaytee: *60*, backlit, PRESENCE, state-based. F_ING AWESOME |
| 04:55.48 | rue_mohr | rcy, your always welcome to crash here, I have a sleeping bag and foamie, there are lots of network jacks in the livingroom |
| 04:55.53 | [TK]D-Fender | jaytee: Ditch the rubber shit buttons and we have a winner! |
| 04:56.02 | rcy | rue_mohr: thanks |
| 04:56.10 | [TK]D-Fender | rue_mohr: Already talking about jacking in the livingroom... perv |
| 04:56.14 | jaytee | [TK]D-Fender, OMG! you're actually recommending a phone other than Polycom? I can feel the shock waves rippling through the industry. Polycom execs leaping out of 8th story windows |
| 04:56.44 | [TK]D-Fender | jaytee: No.. the CONSOLE. the console is GODLY.. the phone it attaches to... IS NOT :) |
| 04:57.16 | jaytee | [TK]D-Fender, seems there's always a downside to the Aastras. |
| 04:57.17 | [TK]D-Fender | jaytee: Careful second guessing me... you're odds are nigh! |
| 04:57.23 | rue_mohr | rcy, you know my number, my extension is 1. well and 2. and come to think of it also 5 |
| 04:57.29 | [TK]D-Fender | jaytee: You haven't heard my rant on them a dozen times already? |
| 04:57.52 | jaytee | yeah, I knew about the shitty rubber buttons but I didn't know you liked the console |
| 04:58.15 | dazoe | [TK]D-Fender: i looked in debian's repo and didn't find an addons package... |
| 04:58.31 | jaytee | [TK]D-Fender, and if recall correctly you thought the base was too light in weight |
| 04:59.17 | joako | dazoe: Switch to SUSE... they have an asterisk-addons package :) |
| 04:59.31 | [TK]D-Fender | jaytee: Aastra 5i series : Rubber shit buttons. cryptic button icons, poor LCD viewing angle, no weight in the base OR handset, tinny speakerphone, and iffy handset, pixel based LCD usied in retard char-matrix model, crappy call-handling for lines / appearances. Thats a STRT |
| 04:59.39 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-7-74.phlapa.east.verizon.net) |
| 04:59.53 | [TK]D-Fender | jaytee: I had a 57i CT at my desk. It made me want my bed-side Polycom IP 301 instead <- |
| 05:00.00 | [TK]D-Fender | jaytee: this was their FLAGSHIP |
| 05:00.07 | jaytee | [TK]D-Fender, ok! you sold me, I'll buy one tomorrow :-) |
| 05:00.22 | [TK]D-Fender | C4!!!! |
| 05:00.24 | [TK]D-Fender | sinks Aastra's battle-ship |
| 05:00.34 | joako | jaytee: Polycoms are great, and now that they are no longer firmware Nazis there's no reason not to |
| 05:00.56 | jaytee | I'm resistant to change and I've developed a wicked fetish for Polycoms. I hate bringing any other brand into the mix. |
| 05:00.57 | [TK]D-Fender | joako: In what way were they before, and are not now? |
| 05:01.09 | [TK]D-Fender | jaytee: I gave them a shot. they failed. |
| 05:01.28 | [TK]D-Fender | jaytee: I SHOULD give Linksys another try as apparently they have come a long way in their firmwares. |
| 05:01.43 | [TK]D-Fender | jaytee: And good for advising outside NA |
| 05:01.53 | joako | [TK]D-Fender: Before you could not get the newest firmware from Polycom, now they post the newest firmware on their site for anyone to download |
| 05:02.09 | jaytee | [TK]D-Fender, are you saying you gave Aastra a shot or that you no longer like Polycom? |
| 05:02.18 | [TK]D-Fender | joaReally? So the latest nice & upfront without the reseller login? |
| 05:02.30 | jaytee | yep, they put it all out there now |
| 05:02.36 | joako | [TK]D-Fender: YES. At first I thought it was a bug on their site |
| 05:02.40 | [TK]D-Fender | jaytee: No... I'm still burnt by Aastra. I said give LINKSYS another shot. |
| 05:02.55 | [TK]D-Fender | jaytee: not that I'm likely to choose them over Polycom on this side of the planet. |
| 05:02.55 | jaytee | ah, but you still prefer Poly's to others |
| 05:03.05 | [TK]D-Fender | jaytee: Polycom > All... |
| 05:03.33 | rue_mohr | [TK]D-Fender, would you remind us of your horror story with aastra? |
| 05:03.34 | [TK]D-Fender | jaytee: but its good to have a basis for fall-backs elsewhere |
| 05:03.38 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 05:03.40 | joako | [TK]D-Fender: I think Linksys (and Sipura before them) are pretty nice, obviously they aren't Polycom, but what gets me is they will not provide the provisioning tools to the public |
| 05:03.44 | [TK]D-Fender | rue_mohr: look up |
| 05:03.55 | jaytee | One of the guys in my Advanced Asterisk class works for Polycom. He was talking about some really good stuff with some of the devs while I was there. |
| 05:04.01 | joako | rue_mohr: I gave Aastra a chance a while back and their firmware was very imature |
| 05:04.16 | [TK]D-Fender | joako: Indeed.... Linksys tends to try to lock in to VSPs |
| 05:04.37 | joako | http://spc.pifiu.com has the tool, but not the latest version |
| 05:05.01 | [TK]D-Fender | joako: the 5i Series 1.4 crashes a fair bit. A mass-page w/ presence will take down a phone in a jiffy |
| 05:05.25 | *** join/#asterisk ultrav1olet (n=telnet@89.20.117.109) |
| 05:05.36 | [TK]D-Fender | joako: I had the CT hoping the DECT would be able to work independant. Not at all the case... |
| 05:05.42 | rue_mohr | were you pushing it at all? or just basic call answering |
| 05:05.45 | dazoe | how do i get asterisk to play musiconhold when someone dials in and then dials an extension? |
| 05:05.45 | joako | [TK]D-Fender: I tried a 480i and the NAT support was very poor. Do you recall the name of the SIP implementation? |
| 05:06.13 | rue_mohr | dazoe, sounds like you want to wrap it into the dialplan? |
| 05:06.26 | [TK]D-Fender | joako: Can't say that I ever know... I did set up a few 480i's once. There were more solid than the 5i's |
| 05:06.27 | rue_mohr | background interrupt? |
| 05:06.28 | ultrav1olet | 'm an absolute newbie in asterisk and I'm trying to add an ability of calling out using a SIP account provided by out telephone company. When I try to make a call I see this message in my asterisk log:[2008-12-09 18:54:27] WARNING[13464]: chan_sip.c:12177 handle_response_invite: Received response: "Forbidden" from '"Username" <sip:sip_account_name@192.9.200.4>;tag=as2fe7f606' |
| 05:06.40 | joako | [TK]D-Fender: Yea, thats another thing. It sort of can operate idependantly but not really... doubt it would take them much to do it, but the firmware updates never came and when they did they didn't deliver what was promised |
| 05:06.40 | rue_mohr | use jepordy music! |
| 05:07.14 | ultrav1olet | Does anyone have a clue what might be wrong? |
| 05:07.22 | [TK]D-Fender | joako: Till then my "Aastra flagship" sits out in the warehouse UNDER a desk collecting dust as dumb DECT receiver |
| 05:07.38 | dazoe | rue_mohr: could you give me any more help other than use jepordy music? |
| 05:07.41 | [TK]D-Fender | joako: sad |
| 05:07.42 | joako | [TK]D-Fender: Now I recall.. CallCtrl... well the version of CallCtrl they use is outdated and they would not update it |
| 05:08.05 | [TK]D-Fender | joako: well its not just the SIP stack... its the rest of the dumb-ass framework around it. |
| 05:08.39 | rue_mohr | dazoe, I dont understand when you want it to play, sounds like there is nowhere in there to put it |
| 05:08.39 | [TK]D-Fender | joako: Aastra's use of soft-keys is GODLY. it hurts that this part is so cool, but the rest makes me cringe |
| 05:09.05 | rue_mohr | and i'm only half assed listening cause i'm catching up on this months emails |
| 05:09.35 | joako | [TK]D-Fender: I think the problem, at least on the 480i is that it was practially the same thing as a mid-90's Nortel analog phone with SIP slapped on it |
| 05:09.35 | [TK]D-Fender | dazoe: Did you install what we told you to? |
| 05:09.48 | [TK]D-Fender | joako: it IS the same phone :0 |
| 05:09.59 | [TK]D-Fender | joako: look at the PT 390 analog phone |
| 05:10.17 | [TK]D-Fender | joako: I bought one when is tarted with * about 5 years ago and ran that w/ ADSI |
| 05:10.35 | dazoe | rue_mohr: someone calls in and hears "Extension please?" they dial 100 then "Pleas hold while i try that extension." "ring ring ring" instead of the ringing i want them to hear the music... |
| 05:10.53 | [TK]D-Fender | joako: Now mind you there are worse phones out there (9XXX series!) |
| 05:10.53 | dazoe | [TK]D-Fender: i said i didn't find any addons package for debian |
| 05:10.56 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 05:11.06 | joako | [TK]D-Fender: Grandstream 100! |
| 05:11.21 | [TK]D-Fender | dazoe: go BROWSE... and when in doubt "fuck packages, packaged * sucks" |
| 05:11.54 | [TK]D-Fender | joako: Yeah, thats just about the very bottom of the barrel. So Aastra has "potential". Thats what disappoints. |
| 05:12.00 | drmessano | Use source.. packages suck |
| 05:12.01 | jaytee | ooooh, compilers are scary!!! |
| 05:12.07 | rue_mohr | dazoe, hmm |
| 05:12.09 | [TK]D-Fender | joako: I hear Linksys has done a fair bit of maturing. |
| 05:12.10 | rue_mohr | I'm sure thats easy |
| 05:12.11 | jaytee | snicker |
| 05:12.18 | drmessano | jaytee: Not if you wrote C, like rue_mohr |
| 05:12.35 | jaytee | drmessano, nope but I invented B |
| 05:12.39 | [TK]D-Fender | drmessano: Waiter.... another drink please? :) |
| 05:12.51 | rue_mohr | hmm, where might that be configured |
| 05:12.52 | drmessano | [TK]D-Fender: Indeed ;) |
| 05:12.53 | jaytee | lol |
| 05:12.55 | joako | [TK]D-Fender: I think most phones have... hell even the new Grandstream GXP-280 is pretty damn decent |
| 05:12.56 | dazoe | rue_mohr: i think i found it... in the dial command |
| 05:12.59 | ultrav1olet | anyone? |
| 05:13.03 | joako | IMO Grandstream tries hard with their firmware |
| 05:13.06 | jaytee | I get the waiter reference hahahaha |
| 05:13.19 | [TK]D-Fender | joakI GXP-280... thats like... high-end crap, right? ;) |
| 05:13.26 | drmessano | "I'll be right out with your clean fork sir, as soon as I reboot the router" |
| 05:13.35 | jaytee | rofl |
| 05:13.52 | joako | [TK]D-Fender: 65 bucks and the phone feels less than a toy than the GXP-2000 |
| 05:14.07 | drmessano | jumps in his $200,000 VoIP van |
| 05:14.08 | [TK]D-Fender | joako: not much of a challenge there... |
| 05:14.10 | joako | [TK]D-Fender: I guess they found some lead in the back of a chinese factory to put in them now |
| 05:14.17 | drmessano | jaytee: Lead the way! |
| 05:14.57 | [TK]D-Fender | joako: Never really saw the 280 before. And now that I have... it does still look like a cheap piece of shit :) |
| 05:15.04 | drmessano | haw |
| 05:15.31 | [TK]D-Fender | joako: 2010 = I look like a polycom, but I can spontaneously combust! *poof* |
| 05:15.50 | jaytee | "My name is Matt Foley, I'm an Asterisk developer, recently divorced AND I LIVE IN A VAN DOWN BY THE RIVER" |
| 05:16.03 | drmessano | HA |
| 05:16.17 | joako | [TK]D-Fender: I'm not saying Grandstream is the best thing on earth, but their firmware is stable, they provide the provisioning tool and each new generation they release is better than the last |
| 05:16.39 | [TK]D-Fender | joako: Thats only a testament to how much they need to improve ;) |
| 05:16.53 | jaytee | wow, stable Grandstream firmware? umm, did anyone just feel the ground shaking? |
| 05:17.03 | [TK]D-Fender | watches the walls bleed |
| 05:17.33 | joako | jaytee how is it unstable? |
| 05:18.00 | jaytee | I really love a firmware that tells me the silence suppression is off when it's actually not and I have to blow into the mouthpiece or make constant noise to keep the MOH from dropping out. |
| 05:18.06 | drmessano | Hasnt the firmware always been the problem? |
| 05:18.12 | drmessano | grandstreamsucks.org? |
| 05:19.16 | drmessano | http://www.grandstreamsucks.com/ <--- Hated Grandstream so much, he spent almost 1/4 the price of one of their phones to register the domain |
| 05:19.21 | drmessano | I say that means something |
| 05:19.37 | jaytee | is that your site! |
| 05:20.01 | drmessano | Hell no, I wouldn't waste the $10 |
| 05:20.11 | drmessano | But polycommunist.org is open :) |
| 05:20.11 | jaytee | whose is it? |
| 05:20.26 | drmessano | I dunno |
| 05:21.55 | joako | What do you guys think of Audiocodes? They blatantly violate the GPL. |
| 05:22.10 | drmessano | I can't believe I am giving Comcast $15 a month for a static IP |
| 05:22.12 | drmessano | bastards |
| 05:22.28 | [TK]D-Fender | drmessano: I've paid that before.. |
| 05:22.38 | jaytee | I'd love to see that site have a picture of chinese kids scavenging in trash heaps with text overlay that says, "Yeah, we're communists and someday we'll take over your country and run your imperialist asses into the ground with our cheap crap. As a matter of fact we're here at the dump searching for parts for your new phone right now!" |
| 05:22.40 | [TK]D-Fender | drmessano: Mind you I now have a /29 for 5$... |
| 05:22.42 | drmessano | [TK]D-Fender: I am only paying $60/mo for the service lol |
| 05:22.57 | drmessano | 14 down/2 up |
| 05:23.07 | [TK]D-Fender | drmessano BASTARD :p |
| 05:23.16 | [TK]D-Fender | drmessano: I'd pay that gladly for 2/2 |
| 05:23.18 | drmessano | At.. home lol |
| 05:23.39 | jaytee | I can't believe how......well, I can believe how screwed up AT&T is. their 800 customer service number has been unavailable for 2 days. |
| 05:23.48 | joako | I wish I had a fast upload. Fastest I can get is 6/0.5 |
| 05:24.04 | jaytee | I can't even cancel my service with the bastards. I suppose if I don't pay my bill it might get their attention. |
| 05:24.14 | joako | Well I could get Comcast but considering they don't even offer HDTV I woulder how could that could be |
| 05:24.21 | [TK]D-Fender | I WAS stable at 5/.8 now I'm at 3/.8. I'd still happily toss for 1.5/1.5 |
| 05:24.32 | drmessano | Buddy of mine switches from AT&T residential DSL 1.5MB/256k to Knology Business 6 down/768k up with a static IP.. shaved $20 off his bill too |
| 05:24.36 | drmessano | Business class rocks |
| 05:24.38 | [TK]D-Fender | is condisering bonded ADSL |
| 05:24.42 | drmessano | and we get free drinks |
| 05:25.49 | jaytee | I think the only difference between your liver and a piece of swiss cheese is that the swiss cheese isn't a light shade of green |
| 05:26.24 | drmessano | joako: comcast does offer HDTV |
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| 05:26.54 | jaytee | I get HD channels from Comcast right now. One of these days I'm going to have to get me an HDTV as well. :-) |
| 05:27.30 | joako | drmessano: Not in my town. They don't offer HDTV or their phone service (not that I would use it) |
| 05:28.00 | drmessano | Which market? |
| 05:28.09 | joako | Dunnellon, FL |
| 05:28.12 | kb3ien | bonding adsls works for me! although right now i'm on a single one... :( |
| 05:28.20 | joako | You probably never heard of it... |
| 05:28.40 | joako | Close to Ocala, FL... which you probably never heard of, either |
| 05:29.14 | drmessano | I have heard of Ocala |
| 05:29.17 | jaytee | I've heard of Ocala, but I can't remember in reference to what or whom |
| 05:29.49 | drmessano | Near the Rainbow river state park |
| 05:29.54 | drmessano | is google whore |
| 05:30.37 | drmessano | Do they still have that good pizza place over there on Pennyslvania avenue? |
| 05:31.27 | kb3ien | i seem to have told asterisk that i wanted to store my voicemails in a database... *grr* |
| 05:32.01 | jaytee | now why'd ya go and do a dumb stunt like that? |
| 05:32.09 | joako | drmessano: Yea Im right next to the rainbow river.... do you know what that pizza place is called? |
| 05:32.37 | drmessano | joako: Vaguely.. it was <some guys name>'s pizza |
| 05:33.03 | drmessano | tony, joe, frank, ed, bob, guido, guiseppe.. |
| 05:33.06 | drmessano | I forget |
| 05:33.17 | jaytee | only 2 more days |
| 05:33.37 | joako | Actually I was thinking of Williams St.... AFAIK on Penn. Ave there is no pizza place |
| 05:33.53 | drmessano | joako: Maybe it's gone now.. this has been years ago, or not |
| 05:33.53 | joako | Do you happen to recall what it was next to? |
| 05:34.17 | drmessano | joako: It was kinda around some other stores, but by itself.. so not really, but yeah |
| 05:35.00 | drmessano | joako: The guy that ran it was a transplant from up north |
| 05:35.02 | joako | drmessano: Maybe you are just going crazy (j/k).. there's nothing interesting around here |
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| 05:35.30 | drmessano | joako: No, I am just generically describing every pizza place in North America |
| 05:37.30 | drmessano | joako: With 98% accuracy, I can pick a major road on in town on google, and pretty much tell you it has a pizza place called <some dudes name>'s pizza, with so-so sauce, and some random topping they do just as good as the other 2 million places, but (oh man, they had the best _____) |
| 05:37.45 | jaytee | hehe |
| 05:37.55 | jaytee | I like Papa Murphy's Take and Bake |
| 05:38.16 | [TK]D-Fender | drmessano: Actually I've got a really awesome pizza place near me here ;) |
| 05:38.16 | drmessano | So my hobby: letting someone tell me where they live, so I can tell them about that pizza place, thanks to google maps |
| 05:38.32 | [TK]D-Fender | drmessano: I only order out once a year maybe, but tis worth it every time |
| 05:38.55 | drmessano | I got a place near me run by a guy... from up north.. that used to be called Tony's pizza... |
| 05:39.09 | drmessano | With so-so sauce.. and the best sliced meatballs...... lol |
| 05:39.25 | drmessano | Fitting my pattern well |
| 05:39.36 | drmessano | Oh, and its near a main road, but not on it.. |
| 05:39.44 | drmessano | But close enough... |
| 05:39.52 | jaytee | but I understand fully the concept of "Generica" where if you get on the interstate and drive for six hours and get off at any urban or suburban exit, there'll be a TGI Fridays or an Applebee's, an OSCO or CVS pharmacy and several other national chain stores in a strip mall and you won't know what state you're in without looking on the map or asking someone. |
| 05:40.23 | drmessano | jaytee: I'll meet you over there at the Waffle House by the interstate in 15 mins |
| 05:40.26 | drmessano | See ya there, buddy! |
| 05:40.48 | jaytee | did you know that La Quinta is spanish for "next to Denny's" ? |
| 05:40.54 | drmessano | ROFLLLLLL!!!!!! |
| 05:41.02 | drmessano | Wait.. |
| 05:41.08 | drmessano | Is there some significance to that? |
| 05:41.26 | drmessano | Because in Augusta, the Denny's is next to the La Quinta |
| 05:41.37 | drmessano | O.o |
| 05:41.37 | joako | jaytee: Thought La Quinta was Spanish for "Free WiFi" (That's what the bilboard said on the interstate) |
| 05:41.51 | joako | In Gainesville, there is no Denny's |
| 05:42.27 | drmessano | The other day I had to set up a customer on the wireless at one of their locations.. he sets up the laptop, starts a remote session... I go in, check his wireless |
| 05:42.46 | jaytee | I've been in almost every state in the US and wherever I saw a Dennys there was a La Quinta Inn next to it. I actually started looking hard whenever I drove trying to find one of them without the other next to it. |
| 05:42.48 | drmessano | Hes got a La Quinta and a T-Mobile hotspot |
| 05:43.03 | drmessano | Im like "Dude, you need to go to the restaurant..i can tell youre at the office" |
| 05:43.04 | drmessano | "How?" |
| 05:43.20 | drmessano | "Because I see the wifi from the La Quinta and the Starbucks" |
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| 05:43.24 | drmessano | "Oh.." |
| 05:43.30 | joako | How can I make voicemail play the name of the sender if the sender has a mailbox on the same system? |
| 05:44.32 | joako | Currently it plays only the number, but if I press 3 + "call the person who sent this message" it says "the number I have is" and plays the name, but not in the message envelope |
| 05:44.40 | jaytee | I'm very impressed with Starbucks, everyone I've been in throughout the US has the same overpriced bitter overroasted coffee and the same shitty wifi service. |
| 05:44.51 | jaytee | I respect consistency |
| 05:45.14 | [TK]D-Fender | joako: BElieve its based on CID and doesn't assume its from another VM user |
| 05:45.44 | [TK]D-Fender | jaytee: Thats SO for you. Sure they have standards... nobody said they were GOOD :p |
| 05:45.48 | [TK]D-Fender | ISO* |
| 05:45.53 | drmessano | jaytee: and the same assholes barreling out of there with 3 foot high cups of hot caffeine in one hand, clutching their blackberrys in the other hand, and steering their overpriced american cars with their belt buckles or knees |
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| 05:46.40 | jaytee | [TK]D-Fender, ok I'll admit I'm spoiled and a bit of a coffee snob but it's true! their coffee sucks |
| 05:46.48 | drmessano | I almost hit one other day.. sitting caddy corner turning the wrong direction into a turn lane, and blocking traffic looking the other way |
| 05:47.01 | [TK]D-Fender | jaytee: argv[-1] |
| 05:47.14 | joako | [TK]D-Fender: voicemail.conf says: cidinternalcontexts = default ; Internal Context for Name Playback |
| 05:47.22 | drmessano | Grantid, I love my blackberry.. but I CAN drive and type |
| 05:47.25 | drmessano | or so I think |
| 05:47.36 | [TK]D-Fender | joako: interesting... |
| 05:47.58 | drmessano | i cn tpehu and drvee at the smae tme! |
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| 05:48.03 | drmessano | lisers |
| 05:48.10 | jaytee | lol |
| 05:49.23 | [TK]D-Fender | loves SSH tunneling. |
| 05:49.53 | joako | [TK]D-Fender: Hmmm... it seems cidinternalcontexts just toggles it from saying "Message from extension" or "message from telephone number".. but why the hell does it say "the number I have is" and then play the recorded name?? |
| 05:50.16 | [TK]D-Fender | joako: Could be they started to actually implement it, but never finished... |
| 05:50.38 | [TK]D-Fender | joako: Should jsut play the "greet" recording IMO |
| 05:51.11 | joako | [TK]D-Fender: Happen to know a KDE SSH app? I loved SSH tunneling when I used Windows and PuTTy but in Linux it's a pain to do it in the CLI |
| 05:51.56 | drmessano | netcat |
| 05:51.57 | [TK]D-Fender | joako: I don't really use a GUI in Linux. It's mostly Putty from WinXP |
| 05:51.59 | drmessano | ftw |
| 05:52.51 | joako | I upgraded the hard drive in my main machine and never bothered to install Windows on it |
| 05:54.21 | jblack | whoah. That's the first time I've ever seen someone say putty is easier than ssh. |
| 05:54.51 | joako | I'm a sucker for pretty GUIs |
| 05:55.49 | drmessano | Is there a filter built into X-Chat that dings and flashes for the words "GUI" and "Windows" or is that custom? |
| 05:56.24 | jblack | I suppose one could make such a thing for most any irc client. |
| 05:57.29 | jblack | huh. the temperature has gone up 15 degrees since the sun set. |
| 05:57.32 | joako | drmessano: Konversation on KDE 3.5 has such an option |
| 06:03.57 | [TK]D-Fender | ok, bed time... later all |
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| 06:25.49 | jaytee | g'nite all |
| 06:35.58 | kb3ien | anyone got a better (more recent) guide than this http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
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| 06:38.16 | joako | kb3ien: see res_config_pgsql.c |
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| 07:19.48 | kb3ien | seems to be not liking me still... [Dec 10 02:36:47] WARNING[19040]: app_voicemail.c:3226 store_file: SQL Execute error [INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)] |
| 07:20.23 | kb3ien | all the manual tests show the table is there http://voip-info.kinghost.net/wiki/view/Asterisk+Documentation+1.4+voicemail_odbc_postgresql.html |
| 07:20.48 | kb3ien | kinda worried that the values are ALL ? is that a 'feature' ? |
| 07:21.15 | Corydon76-dig | kb3ien: they're called placeholders |
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| 07:21.47 | Corydon76-dig | kb3ien: and they're directly useful for not needing to escape field values |
| 07:21.59 | kb3ien | good, but why the execute error, then? |
| 07:22.39 | Corydon76-dig | Are all of the fields there? |
| 07:22.53 | Corydon76-dig | The flag field, in particular, is new |
| 07:23.00 | kb3ien | hrm, lets have a look |
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| 07:24.07 | kb3ien | no the field is not in that document, and thus not in my database. whats its type? |
| 07:24.24 | Corydon76-dig | char(3), I think |
| 07:26.08 | kb3ien | curiously that same document is in the docs/ dir of my repository, but that copy is also lacking in field... |
| 07:26.29 | Corydon76-dig | Oh, no, it's a char(6) |
| 07:27.44 | kb3ien | where is that recorded? |
| 07:27.56 | Corydon76-dig | In the source |
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| 07:28.27 | kb3ien | ah you found a char[6] in the src... okay. |
| 07:29.09 | Corydon76-dig | No, I found a 6 character string in the source |
| 07:30.20 | magic_hat | hey all. we've been having intermittent call quality problems for quite some time w/ our * system... recently upgraded to t1 internet, but problems are still continuing. I ran wireshark on a recent call and found a 3% packet loss, some out-of-sequence packets, and a max delta of 300 ms... I'm assuming this is enough to cause problems. Any idea where to go from here? |
| 07:30.27 | kb3ien | well i'm sure it helpd get me closer but something is still missing... the error hasnt changed. |
| 07:30.55 | Corydon76-dig | kb3ien: I'll look at it in the morning. There's a missing feature I need to add to voicemail. |
| 07:31.18 | kb3ien | is it a dedicated t1? |
| 07:31.50 | kb3ien | any easy way to convert it back to storing vm in the filesystem untill then? |
| 07:31.59 | magic_hat | kb3ien: yep. phone co. line straight into our router and * box. |
| 07:32.40 | kb3ien | t1 is clean? |
| 07:32.53 | kb3ien | no frameing errors etc? |
| 07:33.18 | magic_hat | kb3ien: not sure. we seem to be getting our provisioned speed. how would I check framing errors? |
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| 07:33.39 | kb3ien | magic_hat: the router will have to tell you. |
| 07:34.06 | magic_hat | ah, i'll look at that. assuming that's okay... do I just call teliax and bitch? |
| 07:34.24 | kb3ien | any way to have voicemails stored on the filesystem untill the database issues are fixed? |
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| 07:35.07 | kb3ien | magic_hat maybe the router is reordering packets because it thinks smaller packets go first? |
| 07:35.53 | kb3ien | magic_hat but 3% loss indicates some serious overloading of the pipe, which i assume you arnt doing... |
| 07:36.15 | magic_hat | nah, the only thing going thru the system at that point was one call and some casual websurfing. |
| 07:36.29 | joako | magic_hat: try another VoIP provider and see if the problem persists |
| 07:36.34 | kb3ien | does it still break when you just do the one call? |
| 07:37.09 | kb3ien | maybe the websurfing is breaking it, casually... :) |
| 07:37.18 | magic_hat | kb3ien: I'm just assuming casual websurfing -- there was one other user in the office and he's not a filesharer. lol |
| 07:37.45 | kb3ien | hehe. darnedest things happen without QoS... |
| 07:37.48 | joako | magic_hat: You can try sending your toll-free calls to carrierx.us via SIP |
| 07:38.01 | kb3ien | but yes its probably on the carrier end. |
| 07:38.26 | magic_hat | kb3ien: sorry, the voip provider or our t1 provider? |
| 07:38.32 | joako | voip provider |
| 07:38.47 | kb3ien | voip first :) |
| 07:38.51 | magic_hat | k. we like teliax, but man... i've been mucking around with this for six months. |
| 07:38.57 | joako | if you already tried another internet provider and the problem persists I would think its the voip provider |
| 07:39.15 | magic_hat | joako: yeah, that's some impeccable logic. |
| 07:39.41 | magic_hat | i think the reversed packets don't seem like a huge issue to my novice eyes -- there's 22 of them out of 2000 packets. |
| 07:39.55 | magic_hat | but the delta is the packet delay, right? |
| 07:40.07 | joako | you can try also http://www.flowroute.com/ and get some free credit.... |
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| 07:46.01 | kb3ien | okay seems i'm comitted untill i rebuid or figure out whats wrong... |
| 07:47.50 | *** join/#asterisk scardinal (n=supreme@90.184.100.170) |
| 07:48.57 | kb3ien | carrierx.us has very gimpy forms, are they new? |
| 07:49.51 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 07:50.50 | kb3ien | i'm keen to add them to the rotation and get some stats, but with all those red asterisks (things required) next to all those pulldown menus, im not filling out any forms just yet. |
| 07:52.11 | ultrav1olet | What could that mean "Got SIP response 400 "SIP Parser Error : Unexpected '@', line 3, column 48" back from my_sip_provider"? |
| 07:56.55 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 07:57.05 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 07:57.26 | ultrav1olet | Is there any asterisk channel where people can actually HELP? There are over two hundred people here and everyone keeps silent :-( |
| 08:00.37 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 08:00.39 | tengulre | ultrav1olet: maybe nobody known the answer. so keep silent. |
| 08:00.47 | drmessano | No kidding |
| 08:01.10 | edoceo | ultrav1olet: 400 level errors in SIP, like HTTP are client-request errors |
| 08:01.13 | drmessano | No answer in 5 mins... Guess the channel is dead |
| 08:01.38 | drmessano | It sounds like on line 3, column 48 you got an @ |
| 08:01.50 | edoceo | Exactly |
| 08:02.05 | edoceo | put some wireshark on that and get-to-tracing |
| 08:02.23 | ultrav1olet | My SIP provider doesn't like what I'm sending to it: From: "My name" <sip:username@sip.com@192.9.200.4>;tag=as350ee085 |
| 08:02.33 | drmessano | Well, there you go |
| 08:02.38 | edoceo | ultrav1olet: yep - that's what the error says |
| 08:02.48 | edoceo | If you send a bad request it will not work |
| 08:02.50 | drmessano | sip:username@sip.com@192.9.200.4 |
| 08:02.51 | ultrav1olet | How can I remove @192.9.200.4 part? |
| 08:02.52 | drmessano | Thats the problem |
| 08:02.55 | drmessano | Fix it |
| 08:03.04 | ultrav1olet | I have no idea how to get rid of it |
| 08:03.08 | edoceo | Your username is mal-formed |
| 08:03.12 | drmessano | Yep |
| 08:03.17 | edoceo | Edit the config of your SIP device |
| 08:03.40 | ultrav1olet | edoceo: do you mean [sip-out] in my sip.conf? |
| 08:03.50 | edoceo | pastebin that |
| 08:04.09 | ultrav1olet | hold on a second |
| 08:04.50 | ultrav1olet | http://pastebin.ca/1281453 |
| 08:05.53 | edoceo | your fromuser may just need to be internet203_1 |
| 08:06.01 | drmessano | fromuser is bad |
| 08:06.04 | drmessano | yep |
| 08:06.48 | drmessano | after the @ would be fromdomain |
| 08:06.52 | ultrav1olet | [2008-12-10 13:06:24] WARNING[21732]: chan_sip.c:12177 handle_response_invite: Received response: "Forbidden" from '"Artem" <sip:internet203_1@192.9.200.4>;tag=as35de7a85' |
| 08:07.10 | drmessano | then something else is hosed |
| 08:07.13 | ultrav1olet | with a plain "internet203_1" as a fromuser |
| 08:07.38 | drmessano | Ok, so your authentication is wrong |
| 08:07.51 | edoceo | Forbidden = not allowed (likely due to username/password mis-match) |
| 08:08.21 | ultrav1olet | but with username@host I don't get this error |
| 08:08.35 | drmessano | No, you dont.. you get that its broken |
| 08:08.47 | ultrav1olet | hm |
| 08:08.48 | drmessano | user@host is NOT acceptable |
| 08:09.48 | drmessano | try fromdomain=permngn.usi.ru |
| 08:10.46 | ultrav1olet | it works!!! |
| 08:11.06 | edoceo | champion! |
| 08:11.09 | ultrav1olet | drmessano: damn, every single SIP example in the internet lacks fromdomain option |
| 08:11.11 | drmessano | username @ fromdomain is constructed from those 2 parameters |
| 08:11.16 | ultrav1olet | thank you! |
| 08:11.16 | *** join/#asterisk xrmx__ (n=user@host38-179-dynamic.16-79-r.retail.telecomitalia.it) |
| 08:11.24 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 08:11.42 | edoceo | ultrav1olet: update the wiki on voip-info! |
| 08:12.01 | drmessano | sets the wiki on fire |
| 08:13.20 | ultrav1olet | edoceo: silly me! that option is there - I just omitted it. Sorry, guys! :-] |
| 08:13.54 | ultrav1olet | I just confused by host=sipserver.mysipprovider.com and fromdomain=fwd.pulver.com - in my situation they have to be the same! |
| 08:41.16 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 08:46.07 | *** join/#asterisk mikkel (n=mikkel@130.226.39.202) |
| 08:46.44 | *** join/#asterisk xacatecas (n=jkroon@41.26.86.220) |
| 08:47.55 | xacatecas | hi all, I get lots and lots of the following in my logs: WARNING[???] chan_iax2.c: Unable to cancel schedule ID ????. This is probably a bug (chan_iax2.c: iax2_sched_replace, line 1126). |
| 08:48.17 | xacatecas | any ideas what could be causing it? asterisk does eventually cause the system to become unresponsive and die. |
| 08:49.13 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 08:49.41 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:50.09 | xacatecas | i've seen some reports about re-injecting, however i'm not seeing the same ID multiple times. |
| 08:50.22 | xacatecas | this is with asterisk 1.6.0.2. |
| 08:54.25 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 08:55.49 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 08:56.22 | *** join/#asterisk sergey (n=Sergey@91.189.233.71) |
| 08:57.35 | sergey | Hi. Where look for describe fiels from meetme list XXX concise? |
| 09:01.32 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 09:01.45 | madduck | i have a client which insists on putting dots into phone numbers, |
| 09:01.59 | madduck | e.g. it tries to dial +41.800.123456@my.domain.tld |
| 09:02.04 | madduck | which asterisk refuses with 403 |
| 09:02.10 | madduck | is there a way to just filter out the dots? |
| 09:05.45 | madduck | actually asterisk does not refuse them, buyt the sip provider does |
| 09:05.55 | madduck | so I'd like to filter them out before passing them on. |
| 09:09.26 | madduck | hm, maybe FILTER(<allowed-chars>,<string>) |
| 09:11.10 | madduck | \o/ |
| 09:36.52 | dazoe | I'm wondering if there is a way to put music in meetme rooms, not just for single but constinly. |
| 09:43.55 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 09:47.07 | *** join/#asterisk Ast001 (n=uros@cable-89-216-154-3.dynamic.sbb.rs) |
| 09:48.28 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 10:02.13 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 10:06.04 | *** join/#asterisk atyuil (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 10:15.28 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 10:15.53 | *** join/#asterisk apollonx (n=admin@80.249.21.38.STATIC.FASTCOM.KZ) |
| 10:22.20 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 10:22.47 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 10:27.46 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 10:27.46 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-59.customers.d1-online.com) |
| 10:29.06 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 10:31.56 | *** part/#asterisk Ast001 (n=uros@cable-89-216-154-3.dynamic.sbb.rs) |
| 10:32.31 | *** join/#asterisk mikkel (n=mikkel@130.226.39.202) |
| 10:35.11 | xrmx__ | hi, i have an ancient asterisk 1.2.13 pbx that even if i set promiscredir=yes (tried both globally and to extension context) dial Local, any hint? |
| 10:35.24 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 10:37.30 | xrmx__ | oh and i have a2billing on top of asterisk |
| 10:43.09 | *** join/#asterisk timwilkes (n=timw@browse.net-serv.co.uk) |
| 10:50.34 | *** join/#asterisk propellerhead (n=yogurt2u@host89.190-136-111.telecom.net.ar) |
| 10:54.08 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 10:58.48 | *** join/#asterisk tiny (n=ivob@unaffiliated/tiny) |
| 11:01.29 | tiny | HI! I would like to run stable asterisk release. Should I pull from release branch or should I just pull the latest code? |
| 11:05.04 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 11:06.57 | *** part/#asterisk calmh (n=jb@acro.nym.se) |
| 11:12.24 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:16.05 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 11:23.15 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-59.customers.d1-online.com) |
| 11:25.39 | genin | anyone heard of trixbox? |
| 11:25.44 | genin | and is it any good? |
| 11:28.19 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 11:28.51 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
| 11:28.58 | cvnet | hello all |
| 11:29.04 | cvnet | anyone up ? |
| 11:30.25 | [netman] | ~trixbox |
| 11:30.26 | jbot | hmm... trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 11:31.40 | timwilkes | Any one know anything about creating modules for asterisk (in asterisk-addons) using libraries like gdbm, etc? I can't seem to pass the right arguments to compiler / linker. |
| 11:32.49 | cvnet | I really need some help here, We got a voip gateway which works fine in my friends system, when i try it on my system it works, but only one side can not hear the other side (calling from voip phone, voip phone cna hear hte other side, but other side can not hear voip phones voice) all sides agree on g729 codecs, what else coud it be? |
| 11:34.59 | timwilkes | Sounds like nat issues |
| 11:36.01 | cvnet | i tried both nat=yes, and nat=no |
| 11:36.03 | cvnet | what else could i do? |
| 11:36.39 | cvnet | asterisk server is in public ip not behind nat, but gateway is, however on my friends system which uses voipswitch it works fine |
| 11:37.48 | timwilkes | Tried canreinvite=no ? to force the audio through your * server? |
| 11:37.52 | cvnet | no |
| 11:37.58 | cvnet | I'll try it now |
| 11:39.34 | cvnet | my voip phone which is connect to *, should have the canreinvite=no correct? (which i make the calls from to a cell phoen) |
| 11:40.14 | timwilkes | You could put it on the sip trunk. |
| 11:40.59 | timwilkes | That way, internal calls will still go between each other and not via your * server. |
| 11:41.01 | cvnet | i mean if my userid for the voip phone is test <-- i put it there in sip.conf correct? |
| 11:41.38 | timwilkes | Yes, you set it per friend/peer |
| 11:42.40 | cvnet | ya tried, that, but nothing |
| 11:42.54 | timwilkes | You did do a sip reload, right? |
| 11:43.10 | *** join/#asterisk qdk (n=qdk@77.241.143.49.bredband.3.dk) |
| 11:43.16 | angryuser | hello, it is possible to show in phones LCD the total call waiting in Queue's ? and which phone support that ? |
| 11:43.31 | genin | we have an asterisk with a t1 card taking in a call but it only rings 2 twice and hangs up |
| 11:43.40 | genin | is there a timeout that i can set somewhere |
| 11:43.51 | angryuser | genin: pastebin the cli output |
| 11:44.07 | angryuser | genin: and also sip debug |
| 11:44.16 | genin | heh sorry but what is the pastebin site |
| 11:44.17 | genin | heh |
| 11:44.28 | angryuser | ~pastebin |
| 11:44.29 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 11:44.30 | cvnet | http://pastebin.com/m3535fe7e |
| 11:44.40 | genin | cool thnx |
| 11:44.46 | cvnet | yes i did reload |
| 11:44.49 | timwilkes | cvnet: try a tcpdump if you are still not getting the audio on your * server. |
| 11:45.32 | cvnet | btw it disconnects teh phone if i dont hangup after a min |
| 11:47.35 | cvnet | here is a debug |
| 11:47.36 | cvnet | http://pastebin.com/mb9281a5 |
| 11:50.23 | timwilkes | cvnet: can you remove the nat on the gateway? |
| 11:53.07 | cvnet | from actual voip gateway, or you mean from sip.conf? |
| 11:56.08 | cvnet | timwilkes: from actual voip gateway, or you mean from sip.conf? |
| 11:58.12 | timwilkes | cvnet: I mean, can you give the voip gateway a public ip, or at least a direct route with no nat from your * server? |
| 12:00.59 | cvnet | ok |
| 12:01.01 | cvnet | let me try that |
| 12:01.07 | cvnet | I have to get disconnected |
| 12:01.33 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 12:05.12 | jer_ | timwilkes, a vpn would do the trick |
| 12:06.54 | *** join/#asterisk kerx (n=prepro@adsl-69-105-21-113.dsl.irvnca.pacbell.net) |
| 12:07.11 | *** join/#asterisk telnettech (n=telnette@206.48.21.148) |
| 12:08.11 | telnettech | anyone familar with telecom standards in Aruba? |
| 12:09.07 | timwilkes | jer_: VPNs do sometimes bring their own baggage, like packet fragmentation, etc |
| 12:09.55 | genin | actually i get just this message which looks like the problem |
| 12:09.56 | genin | -- Channel 0/20, span 1 got hangup request, cause 102 |
| 12:09.57 | genin | [Dec 10 13:08:38] WARNING[22713]: app_dial.c:671 wait_for_answer: Unable to forward voice frame |
| 12:10.06 | genin | coming in from pstn |
| 12:10.17 | genin | it rings a few times and then plof |
| 12:10.26 | genin | but if i pick up right away i get comm |
| 12:11.43 | *** join/#asterisk stephank (n=urk@212.178.158.35) |
| 12:13.15 | timwilkes | genin: Hangup Cause 102: RECOVERY_ON_TIMER_EXPIRE |
| 12:13.31 | timwilkes | genin: What arguments are you passing dial? |
| 12:13.39 | genin | one sec |
| 12:13.54 | genin | [test_Massimo] |
| 12:13.54 | genin | ;exten => s,1,Dial(SIP/33650860646@rte_cg) |
| 12:13.54 | genin | exten => s,1,Dial(SIP/33493465574@rte_uh) |
| 12:13.54 | genin | exten => s,n,Hangup |
| 12:15.49 | genin | under another part of the extensions.conf it says |
| 12:15.51 | genin | exten => 1202,1,Goto(test_Massimo,s,1) |
| 12:15.52 | genin | exten => 7902,1,Goto(test_Massimo,s,1) |
| 12:16.19 | jer_ | timwilkes, indeed |
| 12:16.39 | jer_ | timwilkes, but one can compensate for that, by using a higher mtu on the vpn tunnel at both ends (if supported) |
| 12:16.51 | *** join/#asterisk galeras (n=galeras@Dynamic-IP-cr20011883249.cable.net.co) |
| 12:18.35 | *** join/#asterisk sosoriri (n=chatzill@222.47.180.130) |
| 12:20.07 | galeras | Please suggest me a good SIP provider to make calls to USA, Canada and Panama. |
| 12:21.49 | sosoriri | hi, everybody. i have some problem about asterisk. can i ask this question to you? |
| 12:22.03 | genin | timwilkes: any ideas? |
| 12:22.14 | angryuser | ~question |
| 12:22.14 | jbot | hmm... question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 12:22.23 | angryuser | ~questions |
| 12:22.24 | jbot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
| 12:22.38 | timwilkes | genin: pri debug |
| 12:22.38 | *** part/#asterisk ultrav1olet (n=telnet@89.20.117.109) |
| 12:24.05 | sosoriri | i use asterisk-1.4.18 now. |
| 12:24.15 | sosoriri | it's very useful for our lives. |
| 12:24.34 | *** join/#asterisk lbruno (n=me+irc@pa1-84-91-3-125.netvisao.pt) |
| 12:24.44 | *** join/#asterisk SibRphrek (n=SibRphre@cpe-67-243-43-136.nyc.res.rr.com) |
| 12:24.45 | sosoriri | but cpu usage was 99% for a long time. |
| 12:25.09 | lbruno | waves |
| 12:25.15 | genin | type pri debug at the cli? |
| 12:25.22 | timwilkes | genin: yeap |
| 12:25.22 | sosoriri | after that, asterisk was died. |
| 12:25.37 | genin | tecca03*CLI> pri debug |
| 12:25.38 | genin | No such command 'pri debug' |
| 12:25.50 | sosoriri | ??] |
| 12:26.00 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
| 12:26.26 | genin | pri debug span? |
| 12:26.29 | timwilkes | genin: Sorry, try pri debug span X |
| 12:26.34 | genin | heh |
| 12:26.35 | genin | cool thnx |
| 12:26.45 | cvnet | timwilkes: i put the gateway in public IP and it worked like a beauty |
| 12:27.03 | cvnet | timwilkes: however I need it to work behind a router (nat) |
| 12:27.17 | timwilkes | cvnet: NAT is your issue. jer_ did suggest using a VPN. |
| 12:27.43 | genin | what am i looking for in this debug |
| 12:27.44 | timwilkes | cvnet: Watch out for MTU issues, packet fragmentation, etc. |
| 12:27.46 | cvnet | waht you mean by VPN ? |
| 12:27.47 | genin | i see always |
| 12:27.53 | genin | -- Channel 0/22, span 1 got hangup request, cause 102 |
| 12:27.53 | genin | [Dec 10 13:27:14] WARNING[22747]: app_dial.c:671 wait_for_answer: Unable to forward voice frame |
| 12:27.53 | genin | <PROTECTED> |
| 12:28.16 | genin | <PROTECTED> |
| 12:28.17 | genin | <PROTECTED> |
| 12:28.37 | lbruno | distro for Asterisk? I was going to try trixbox, but will gladly accept recommendations. |
| 12:29.07 | *** join/#asterisk zeljkoMON (n=bum@cable-89-216-173-176.dynamic.sbb.rs) |
| 12:29.44 | timwilkes | genin: What version of * are you using? |
| 12:30.38 | timwilkes | genin: http://bugs.digium.com/view.php?id=9934 |
| 12:30.41 | zeljkoMON | any1 had probs with misdn not dialing on ports 2,3,4? |
| 12:31.20 | genin | 1.4.17 |
| 12:31.20 | timwilkes | cvnet: http://en.wikipedia.org/wiki/Vpn |
| 12:31.41 | zeljkoMON | im using dial(misdn/g:isdn.... |
| 12:32.15 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 12:33.27 | angryuser | genin: your destination channel hang's up so * is "Unable to forward voice frame" |
| 12:33.49 | genin | what could cause that? |
| 12:33.53 | galeras | no good sip providers to recommend? |
| 12:33.55 | genin | it is something in the dialplan? |
| 12:34.30 | timwilkes | genin: What do you see before the span 1 hangup ? |
| 12:34.39 | angryuser | i was out dining so i missed your pastebin's let me look for a min |
| 12:34.47 | *** part/#asterisk galeras (n=galeras@Dynamic-IP-cr20011883249.cable.net.co) |
| 12:35.23 | genin | < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: International network (7) |
| 12:35.23 | genin | < Ext: 1 Cause: Recover on timer expiry (102), class = Protocol Error (e.g. unknown message) (6) ] |
| 12:35.23 | genin | -- Processing IE 8 (cs0, Cause) |
| 12:35.25 | genin | q931.c:3568 q931_receive: call 1111 on channel 22 enters state 12 (Disconnect Indication) |
| 12:35.33 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:36.15 | timwilkes | genin: and just before that? In fact to do have the output for the entire call? |
| 12:36.38 | genin | < Protocol Discriminator: Q.931 (8) len=9 |
| 12:36.38 | genin | < Call Ref: len= 2 (reference 1111/0x457) (Originator) |
| 12:36.38 | genin | < Message type: DISCONNECT (69) |
| 12:36.39 | genin | < [08 02 87 e6]it |
| 12:36.42 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 12:36.51 | genin | yes but there are lots of customers passing stuff too |
| 12:36.59 | genin | so it is pretty crazy on the CLI |
| 12:37.12 | *** join/#asterisk propellerhead (n=yogurt2u@host89.190-136-111.telecom.net.ar) |
| 12:37.25 | angryuser | genin: "Protocol Error" are you sure your card is set as it should be ? |
| 12:37.34 | genin | the weird thing is |
| 12:37.39 | genin | it rings 3 times |
| 12:37.42 | genin | then hangsup |
| 12:37.49 | genin | but if i pick it up right away i have communication |
| 12:38.32 | genin | <PROTECTED> |
| 12:38.32 | genin | > Protocol Discriminator: Q.931 (8) len=9 |
| 12:38.32 | genin | > Call Ref: len= 2 (reference 1111/0x457) (Terminator) |
| 12:38.32 | genin | > Message type: PROGRESS (3) |
| 12:38.32 | genin | > [1e 02 81 88] |
| 12:38.33 | genin | > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) |
| 12:38.36 | genin | > Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] |
| 12:38.39 | genin | sorry for the chan spam |
| 12:42.03 | cvnet | oo my god |
| 12:42.06 | cvnet | got it figured out |
| 12:43.18 | cvnet | timwilkes: I was using ;exten => _X.,n,Dial(SIP/IP-Address/${EXTEN}) ; O changed that to _X.,n,Dial(SIP/Trunk-Name-From-Sip/${EXTEN}) and it works, |
| 12:43.37 | *** part/#asterisk lbruno (n=me+irc@pa1-84-91-3-125.netvisao.pt) |
| 12:44.04 | cvnet | timwilkes: I do appreicate your help, thanks a bunch, it took me 3 days to figure this out, but at the end its workin, I'm happy, THANKS ALOT |
| 12:47.22 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 12:52.43 | *** join/#asterisk galeras (n=galeras@Dynamic-IP-cr20011883249.cable.net.co) |
| 12:53.38 | zeljkoMON | anyone to help me with misdn? |
| 12:54.28 | galeras | Please, let me to know a reliable voip provider for calls from my asterisk to USA and LatinAmerica. |
| 12:58.42 | *** join/#asterisk sosoriri (n=chatzill@222.47.180.130) |
| 13:01.38 | *** join/#asterisk cosf (n=cosf@190.13.139.22) |
| 13:02.31 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 13:03.25 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
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| 13:05.59 | *** join/#asterisk zxd (n=zapw@213.31.43.2) |
| 13:06.00 | zxd | hi |
| 13:06.08 | gambler1 | Hi, does anyone know how to check "online" registered peers when using realtime? I seams that after session timer expires * does not update any column in db |
| 13:07.38 | etfonhomey | gambler1, you mean "sip show peers" from the CLI does not contain the correct information? |
| 13:08.20 | zxd | why asterisk does not mark dscp field of RTCP packets ? it only marks RTP media and the sip protocol |
| 13:12.13 | gambler1 | etfonhomey: yes, it because we are using dynamic relatime configuration |
| 13:12.48 | gambler1 | etfonhomey: and we have also enabled rtcachefriends to see at least something (if you know what I mean) |
| 13:13.39 | etfonhomey | gamber1, hmm, I haven't used * realtime yet. That just surprises me. |
| 13:14.09 | gambler1 | etfonhomey: well you can try and you will be suprised with the results :) |
| 13:14.21 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000374.dsl.bell.ca) |
| 13:14.31 | gambler1 | etfonhomey: in other words... you get something, you lose something |
| 13:15.03 | etfonhomey | gambler1, You've seen where other people report the same thing? |
| 13:15.14 | gambler1 | etfonhomey: no? |
| 13:16.12 | etfonhomey | gambler1, That's the first time I've heard about that "feature". |
| 13:17.53 | gambler1 | sippeers: about dynamic realtime? |
| 13:18.08 | gambler1 | etfonhomey: sorry, about dynamic realtime? |
| 13:19.28 | etfonhomey | gamber1, yes |
| 13:20.25 | gambler1 | etfonhomey: for the start you can read this book: http://downloads.oreilly.com/books/9780596510480.pdf |
| 13:20.56 | etfonhomey | gambler1, I know what it is and how it works. I've just never used it. |
| 13:21.35 | zxd | where do i configure how much speech to carry inside g792 packet |
| 13:22.19 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:23.38 | etfonhomey | gamber1, the "feature" I'm talking about is the incorrect info you mentioned when you do "sip show peers" |
| 13:24.52 | gambler1 | etfonhomey: that was example, my real question is how to find online peers within db? |
| 13:25.05 | anonymouz666 | zxd: read doc/rtp-packetization.txt |
| 13:25.26 | zxd | anonymouz666, how come asterisk dosen't mark RTCP packets with dscp values? |
| 13:26.58 | anonymouz666 | I don't know. Are you sure of that? |
| 13:27.16 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:28.25 | zxd | 100 percent sure |
| 13:28.35 | zxd | i saw with tcpdump |
| 13:28.51 | zxd | only rtp media the equal port number , and sip are marked |
| 13:29.05 | zxd | equal=even |
| 13:31.21 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 13:31.26 | *** join/#asterisk proppy (n=proppy@rosiers.mekensleep.com) |
| 13:31.37 | anonymouz666 | Why do you care about RTCP packets? |
| 13:31.57 | proppy | Hi, is there a way to change umask/permission of file recorded with MixMonitor ? |
| 13:32.26 | anonymouz666 | zxd: it's sent with intervals |
| 13:32.28 | anonymouz666 | not all the time |
| 13:32.32 | anonymouz666 | or at the end |
| 13:32.47 | zxd | anonymouz666, because it's part of signaling no ? |
| 13:33.01 | anonymouz666 | there's nothing to do with SIP signalling |
| 13:33.17 | anonymouz666 | it's stats about your RTP session |
| 13:33.32 | anonymouz666 | sent by the UAc |
| 13:33.35 | anonymouz666 | user agent client |
| 13:34.08 | *** join/#asterisk Jubei (n=chatzill@118x240x210x13.ap118.gyao.ne.jp) |
| 13:34.19 | Jubei | is the configuration language for asterisk stil the same as 1.2 ? |
| 13:35.06 | Jubei | also, is there any good GUI integration for asterisk? |
| 13:35.37 | angryuser | Jubei: what do you mean by good ? |
| 13:35.43 | *** join/#asterisk lilalinux (i=e-trolle@fellatio.deswahnsinns.de) |
| 13:36.04 | Jubei | angryuser: I mean one that ..eer... works and doesnt make a mess of extensions.conf when somebody wants to edit by hand etc |
| 13:36.25 | lilalinux | what was the magic number one has to dial to transfer the current call to another exten? |
| 13:36.40 | Jubei | angryuser: when I last checked asterisk 3-4 years back the gui's available werent very good and required a lot of work to integrate with asterisk |
| 13:37.08 | anonymouz666 | lilalinux: "feature show" |
| 13:37.11 | anonymouz666 | CLI |
| 13:37.21 | lilalinux | thxz |
| 13:37.28 | proppy | found it, should change umask in /etc/init.d/asterisk |
| 13:37.32 | angryuser | Jubei: the asterisknow has a nice gui which do not produce a mess and can be customized, forget about freepbx |
| 13:38.43 | angryuser | Jubei: druid is a db driven , but still you have agi in any case |
| 13:40.00 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 13:40.03 | zxd | anonymouz666, at it is important no ? |
| 13:40.07 | Jubei | angryuser: there is now db-based configuration for asterisk? ic. thanks ang |
| 13:40.08 | zxd | anonymouz666, this info rtcp |
| 13:40.31 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 13:41.16 | *** part/#asterisk sadleder (n=philipp@stgt-5d843fe2.pool.einsundeins.de) |
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| 13:41.50 | angryuser | Jubei: well druid still writes the configs but after it saves all the info in db, but it is not easy to customize that without passing some week's to learn how it is done, try asterisknow |
| 13:42.50 | *** part/#asterisk proppy (n=proppy@rosiers.mekensleep.com) |
| 13:43.08 | Jubei | angryuser: I'll try a different pbx because it seems to me asterisk is still in that same miserable state I left it a few years back |
| 13:45.06 | angryuser | Jubei: gui in general are not supposed to provide a high customisation, if you want to build something personal you have AMI or AGI or FAST agi or whatever you want |
| 13:45.12 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 13:45.34 | Jubei | angryuser: true |
| 13:46.05 | Jubei | angryuser: however, I have a feeling that asterisk's configuration could somehow be easier. |
| 13:47.08 | angryuser | Jubei: hm, maybe you should look into paid solutions like trixbox pro ? |
| 13:47.15 | angryuser | or else |
| 13:47.52 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:47.59 | Jubei | hmm... yes maybe something commercial could do the trick |
| 13:48.07 | angryuser | or hire someone to do the job it is easier ;) |
| 13:48.53 | Jubei | angryuser: I see. Thanks for your thoughts^^. |
| 13:49.10 | Jubei | for sharing* |
| 13:52.58 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
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| 14:00.29 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
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| 14:01.13 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 14:01.29 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
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| 14:24.08 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 14:24.41 | zeljkoMON | so anyone, misdn help? |
| 14:24.54 | *** join/#asterisk mintos (n=mvaliyav@203.153.39.18) |
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| 14:30.04 | *** part/#asterisk galeras (n=galeras@Dynamic-IP-cr20011883249.cable.net.co) |
| 14:30.38 | *** join/#asterisk Chesther (n=cam2@cam2-win.cit.cornell.edu) |
| 14:32.33 | cvnet | ~trunk gateway |
| 14:32.38 | cvnet | ~trunkgateway |
| 14:32.52 | cvnet | ~voiop |
| 14:33.15 | [TK]D-Fender | cvnet: Getting colder... |
| 14:34.26 | cvnet | lol |
| 14:35.08 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 14:35.11 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.2 (2008/12/02), 1.4.22 (2008/10/02), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0, dahdi-tools 2.1.0 (2008/12/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 14:35.55 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 14:35.57 | tiny | Hi, I've managed to login trough asterisk-gui but nothing really happens. I have a nonusable web page in front of me. What's with that? |
| 14:36.25 | ocnarf | can i force a peer to use g729? |
| 14:36.55 | ocnarf | in my general context, i have ulaw, alaw and g729. |
| 14:37.22 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 14:37.28 | [TK]D-Fender | ocnarf: You force a peer by setting it in the peer |
| 14:37.30 | *** part/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 14:37.52 | ocnarf | i did but it still says the incompatible codecs |
| 14:37.57 | ocnarf | any idea? |
| 14:38.29 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 14:39.00 | ocnarf | in my peer, i disallow all and allow g729 only then on my phone i force it to g729 |
| 14:39.07 | [TK]D-Fender | ocnarf: I suggest you pastebin the SIP debug of your call attempt and your sip.conf masking only psswords |
| 14:39.09 | [TK]D-Fender | ~pb |
| 14:39.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 14:39.11 | [TK]D-Fender | ^^^ |
| 14:39.27 | ocnarf | ok |
| 14:39.58 | cvnet | ~trunkgateway |
| 14:40.05 | cvnet | ~trunk gateway |
| 14:41.27 | ruben23 | hi..im planning to setup a PBX with my phone line & analog phones in asterisk...what ahrdware would ill be needing to do the job.. |
| 14:41.46 | [TK]D-Fender | cvnet: What are you looking for. Those bot-lets don't exist |
| 14:42.02 | [TK]D-Fender | ruben23: How many phones, how many lines? |
| 14:42.11 | ocnarf | D-Fender: Here.. http://pastebin.com/d11093494 |
| 14:42.35 | [TK]D-Fender | ocnarf: and the rest? |
| 14:43.30 | ocnarf | what else do u need? |
| 14:43.46 | ruben23 | two line 4 analog ohones |
| 14:43.48 | [TK]D-Fender | ocnarf: I told you to provider the complete SIP DEBUG CLI output for your failed attempt |
| 14:44.03 | ocnarf | ok |
| 14:44.05 | ocnarf | sorry |
| 14:44.29 | ruben23 | [TK]D-Fender:2 lines, 4 analog phones |
| 14:44.58 | [TK]D-Fender | ruben23: Any expansion expectations? |
| 14:46.04 | ruben23 | yes for expansion...im planning to deploy it @ office |
| 14:46.30 | [TK]D-Fender | ruben23: I'm talking about it growing past 2/4 |
| 14:47.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:47.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:48.03 | ruben23 | ok....no expansion expectations. |
| 14:48.37 | [TK]D-Fender | ruben23: Ok, tough part is finding something of quality for such a small requirement... |
| 14:49.07 | [TK]D-Fender | ruben23: That feels "economical |
| 14:49.13 | angryuser | intel mini itx atom board 60 € |
| 14:49.44 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 14:49.57 | ruben23 | [TK]D-Fender:what do i need to start it...? plan to setup voicemail |
| 14:50.15 | [TK]D-Fender | ruben23: Since this is for a business you should get something solid... |
| 14:51.02 | [TK]D-Fender | ruben23: here : http://www.telephonydepot.com/Catalog/Sangoma-B600/B600D-Analog-Voice-Card and 2 x http://www.telephonydepot.com/Catalog/Linksys-Analog-Adapters/Linksys-PAP2T-NA |
| 14:51.04 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:51.26 | [TK]D-Fender | ruben23: That'll laswt through adding lines & a fax for passthrough that probably WON'T turn to rat-shit |
| 14:51.48 | ocnarf | D-Fender: Here is my sip debug http://pastebin.com/d64b619ec |
| 14:52.27 | [TK]D-Fender | ocnarf: Capabilities: us - 0x100 (g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 14:52.47 | [TK]D-Fender | ocnarf: Your setup of the PAP2 configured *NO* codecs to be available. Fix your ATA |
| 14:52.47 | ocnarf | meaning? |
| 14:53.36 | zeljkoMON | i need misdn help |
| 14:54.08 | ruben23 | [TK]D-Fender:what features could i add up with asterisk to my PBX |
| 14:54.10 | coppice | [TK]D-Fender interesting that they are still launching PCI versions of these new cards |
| 14:54.30 | [TK]D-Fender | ruben23: Not sure what you mean. |
| 14:54.41 | [TK]D-Fender | coppice: is still the "norm" these days... |
| 14:54.56 | [TK]D-Fender | coppice: plenty of market left. |
| 14:55.02 | coppice | I don't see too many PCI slots on new boards |
| 14:55.13 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:55.17 | [TK]D-Fender | coppice: No, not many, and depends what kind of systems you look at as well. |
| 14:56.25 | ocnarf | i place g729 on preferred codec then "use preferred codec=yes" |
| 14:56.32 | coppice | the EC version of those cards is $180 more than the non-EC. Nice profit margin there :-) |
| 14:56.38 | ocnarf | anything more i should change? |
| 14:57.26 | ruben23 | [TK]D-Fender: what i mean is that, what features of asterisk i could add up to my plan PBX system that you could suggest..like voicemail |
| 14:57.35 | Mimmus | chan_sip of Asterisk 1.6 is always the same or was rewritten? |
| 14:57.36 | [TK]D-Fender | ocnarf: I suggest you look things over very carefully, read its manual, etc. The ATA was the one offering nothign |
| 14:57.54 | ocnarf | D-Fender: Thanks |
| 14:58.26 | [TK]D-Fender | ruben23: MeetMe conference rooms, fax -> e-mail, IVR's, TTS, ASR, etc |
| 14:58.44 | [TK]D-Fender | Mimmus: evolution, not rewrite |
| 14:59.37 | Mimmus | [TK]D-Fender: I remember that OEJ promised a chan_sip2 or similar... |
| 14:59.59 | [TK]D-Fender | Mimmus: "Promised", huh? |
| 15:00.04 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:01.25 | angryuser | coppice: some supermicro motherboards still have 6 pci port's or use 64 bit pci >32 bit converters, we are far away of pci death |
| 15:02.33 | Mimmus | http://bugs.digium.com/bug_view_page.php?bug_id=0000759 |
| 15:02.34 | coppice | angryuser: in the last round of motherboards PCI was actually making a comeback, but if you look at the current round there are very few left. The i7 generation is gonna practically wipe them out |
| 15:03.21 | [TK]D-Fender | Mimmus: Date Submitted 2004-01-07 12:06 , Last Update 2005-01-08 04:08 |
| 15:03.28 | Mimmus | Ooops! |
| 15:03.34 | rue_mohr | man my machine is gonna be to obsolete by the time I upgrade it |
| 15:04.12 | Mimmus | I'm not tuned in |
| 15:05.01 | zeljkoMON | i had hard time finding board with 3 pci slots |
| 15:05.33 | rue_mohr | looks like the problem is that manufacturers dont thin you need expansion slots anymore |
| 15:05.42 | rue_mohr | guess they think everythings built in |
| 15:05.54 | rue_mohr | this asus one I'm looking at has two |
| 15:06.04 | rue_mohr | a riser and 1 pcie |
| 15:06.14 | zeljkoMON | i got some gygabyte |
| 15:07.09 | rue_mohr | hmm wonder which caps go first ont his baord... |
| 15:07.32 | kerx | rtp.c:788 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. |
| 15:07.40 | kerx | anyone know what this is? |
| 15:07.58 | [TK]D-Fender | kerx: nothing hidden there. * does not support VAD/CNG |
| 15:08.03 | zeljkoMON | any1 had probs with misdn stop dialing on port numbers greater then 1? |
| 15:08.08 | kerx | what is VAD/CNG ? |
| 15:08.16 | [TK]D-Fender | ~vad |
| 15:08.17 | jbot | methinks vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 15:08.39 | kerx | i don't even know about it, let me Google it |
| 15:08.40 | kerx | thanks |
| 15:08.44 | Mimmus | a "medium" HP server has Six expansion slots (three PCI Express, three PCI-X) |
| 15:09.03 | coppice | no. VAD is Voice Activity Detection. Silence suppression is something completely different |
| 15:09.51 | [TK]D-Fender | coppice: Halves of a coin? |
| 15:09.57 | rue_mohr | hmm 4 pcie on this one... |
| 15:10.44 | coppice | [TK]D-Fender: nope. just different. silence suppression sucks. VAD is part of a good solution |
| 15:11.18 | [TK]D-Fender | coppice: Similar goal, opposite approach? |
| 15:11.48 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-4b4aab66d4f5e047) |
| 15:11.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:11.56 | coppice | nope. silence suppression is just useless. VAD + CNG is proper engineering |
| 15:12.27 | *** join/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 15:12.35 | [TK]D-Fender | coppice: I'll put aside some time to do some reading on this... |
| 15:12.43 | anonymouz666 | I think both sucks. VAD+CNG. |
| 15:12.48 | anonymouz666 | it never sounds good. |
| 15:13.10 | coppice | anonymouz666: you wouldn't even know you were using VAD + CNG |
| 15:13.16 | angryuser | coppice: they were saying this when core 2 duo went out, we will move to pci express eventually but not in near future, it will be faster in the mainstream and very slow in the *pro* |
| 15:14.02 | *** part/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 15:14.13 | anonymouz666 | coppice: there are two ways: just calling and sounds like walkie-talkie and doing a RTP dump. |
| 15:14.16 | anonymouz666 | :) |
| 15:14.49 | coppice | if it sounds like a walkie talkie you either have silence suppression or some other problem |
| 15:15.33 | anonymouz666 | coppice: what client did you test VAD+CNG? |
| 15:15.59 | coppice | Try a proper G.729AB implementation. The VAD in that works well |
| 15:16.41 | coppice | you loose more quality going from G.729 to G.729A than from going G.729 to G.729B |
| 15:17.47 | anonymouz666 | I can't even know if it's GSM or G729 :-) just hearing |
| 15:18.22 | *** join/#asterisk invalidrecord (n=fares@87.113.88.31.plusnet.pte-ag2.dyn.plus.net) |
| 15:19.04 | coppice | that's true. you can't tell the quality of the codec from listening to the result. you have no idea how good the original was. You need to compare |
| 15:19.18 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:20.34 | coppice | that's not quite true. there are specific artefacts in these codecs, which you tend to pick up when you work with them every day |
| 15:20.44 | *** join/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 15:23.49 | *** join/#asterisk docelmo (n=chatzill@206.248.239.194) |
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| 15:34.12 | bijit | can large asterisk log files make asterisk restart 2 -3 times a day? |
| 15:34.40 | glaz | I've seen this situation, rotate the logs once a while |
| 15:35.04 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-139.unitymediagroup.de) |
| 15:37.24 | wonderworld | hi. i am trying to connect two asterisks boxes via a SSH tunnel and IAX2. is that possible at all? i mapped a local port on box 1 with ssh to the iax-port on box 2, but box 1 never sees box 2 as channel. |
| 15:38.19 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-200.customers.d1-online.com) |
| 15:40.09 | bijit | glaz: has happend to you? |
| 15:41.33 | ruben23 | hi im setting up asterisk box as a PBX system @ home connecting my analog phone, can i setup my network layout like this INTERNET==>modem==>(eth0)Asterisk(eth1)==>switch==>ATA==>analog phone/phoneline. |
| 15:41.35 | glaz | bijit: no, someone I know tho, he crontabed asterisk -rx "logger rotate" every hours and it has fixed his problem. |
| 15:41.38 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 15:42.26 | bijit | glaz: good to know I have a log more than 4gb O.o |
| 15:42.41 | glaz | bijit: about time you rotate :) |
| 15:43.05 | glaz | you could create a script that would logger rotate and then bzip the rotated log file |
| 15:43.11 | bijit | glaz: definitely |
| 15:43.20 | glaz | and run this script every hours or so. |
| 15:43.31 | bijit | glaz: yeah I seen and example in wiki |
| 15:44.08 | glaz | ok cool. |
| 15:45.55 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 15:46.09 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 15:48.22 | mort_gib | Hey, I have a quick question, moh I get format_wav.c:148 check_header: Not in mono 2 when I try to play wav files, but the very same config works on another system... |
| 15:49.02 | mort_gib | So do I need additional packages in order to play wav as moh?? I though 1.4.X could do this natively... |
| 15:49.41 | bijit | http://www.voip-info.org/tiki-index.php?page=logrotate if I use this I will only keeps logs for a week right? |
| 15:52.45 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 15:54.25 | *** join/#asterisk c017 (n=c017@85.232.120.5) |
| 15:55.29 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:55.38 | jameswf | ping seanbright |
| 15:55.46 | seanbright | pong |
| 15:56.08 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:56.23 | jameswf | seanbright: you were getting hangup cause 99 at one point connecting to a nortel do you recall the fix? |
| 15:56.33 | seanbright | was i? |
| 15:56.47 | seanbright | i don't recall such a thing |
| 15:56.48 | jameswf | http://purl.rikers.org/%23asterisk/20080714.html.gz |
| 15:56.53 | seanbright | looking |
| 15:57.06 | seanbright | oh |
| 15:57.11 | seanbright | Set(CALLERID(name)=) |
| 15:57.17 | seanbright | for some reason that worked for me |
| 15:57.17 | seanbright | :) |
| 15:57.46 | dominic1 | I have a big problem. We are using misdn and zaptel cards and want to use did with a different length. So it's possible to have a number 123 and a number 1231. When I now dial the number 1231 with a analogue line it jumps into into 123 and is not using 1231 |
| 15:57.53 | jameswf | cool thx... someone is seeing it and I like i dunno..... then saw you on google and was like sweet.... |
| 15:58.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:59.43 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-197-154-rrdg-esr-2.dynamic.isadsl.co.za) |
| 15:59.48 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-197-154-rrdg-esr-2.dynamic.isadsl.co.za) |
| 16:00.53 | seanbright | jameswf: 22:00.22 |
| 16:00.56 | seanbright | in that log |
| 16:02.03 | *** join/#asterisk grx0 (n=dave@64-150-178-3.kansascity.abac.net) |
| 16:05.58 | [TK]D-Fender | wonderworld: "mapping port" and "channel" do not apply. there is no such thing as a continouos "like" |
| 16:06.00 | [TK]D-Fender | link* |
| 16:06.20 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 16:07.07 | [TK]D-Fender | wonderworld: Go watch for actually communication attempts when yuo place a call. |
| 16:11.59 | Whitor | Hi, I'm having difficulty routing a call through an intermediate asterisk box. I have two asterisks connected via iax2, (works fine) one of those boxes has an fxo card which connect to my company pbx (1xx extensions) When I call 1xx from the far asterisk box, the middle asterisk box answeres with the digital receptionist. How can I get it so that 1xx calls are handed directly over to the fxo card? (company pbx) |
| 16:12.22 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 16:12.42 | [TK]D-Fender | Whitthis is your dialplan. Go fix it |
| 16:12.45 | Whitor | if I place 1xx calls from sip phones connected to the system that has the fxo card in it ... it works fine |
| 16:13.06 | Whitor | [TK]D-Fender, heh. okie thanks |
| 16:13.33 | Whitor | dialplan on the remote system or the middle system? ... or both ? |
| 16:14.37 | ruben23 | [TK]D-Fender:can i do call transfer with my asterisk box using analog phones... |
| 16:14.55 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:14.58 | Whitor | you can with a sip gateway |
| 16:16.19 | ruben23 | Whitor:hows that possible? configuration on asterisk? |
| 16:17.01 | [TK]D-Fender | Whitor: you should already know what system's IVR is coming up. |
| 16:17.23 | Whitor | [TK]D-Fender, ok, thanks for hte clue |
| 16:17.33 | [TK]D-Fender | ruben23: Of course you can, and it depend on what they are plugged into |
| 16:17.46 | *** join/#asterisk mog (n=mog@nat/digium/x-603e8fe8cbd1d706) |
| 16:17.47 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:18.08 | *** join/#asterisk invalidrecord (n=fares@87.113.88.31.plusnet.pte-ag2.dyn.plus.net) |
| 16:18.18 | ruben23 | [TK]D-Fender:what you mean? can you explain further.. |
| 16:18.29 | carrar | ~centos |
| 16:18.29 | jbot | somebody said centos was an Enterprise-class Linux Distribution derived from sources freely provided to the public by a prominent North American Enterprise Linux vendor. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 16:18.58 | carrar | Is that issue with centos still around? |
| 16:19.05 | carrar | or is that fixed |
| 16:20.10 | ruben23 | [TK]D-Fender: can i pm...is it ok? |
| 16:20.46 | bijit | [TK]D-Fender: does asterisk -rx "logger rotate" rstart asterisk? |
| 16:22.27 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 16:22.43 | Whitor | is curious what "that issue" refers to |
| 16:23.09 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:23.40 | pif | hi, in 1.4 how do I ensure the original callerid is sent when transfering/forwarding a call? |
| 16:24.00 | SuPrSluG | anyone know of a multi tenant solution for ITSP's ? Something comparable to Enswitch. or using OpenSER |
| 16:24.19 | thehar | there was a digium announcement this morning |
| 16:24.22 | SuPrSluG | you can use setcallerid command |
| 16:24.54 | anonymouz666 | pif: the 'o' option. |
| 16:25.02 | pif | oh? thx |
| 16:25.03 | anonymouz666 | in Dial() |
| 16:25.28 | anonymouz666 | A -> B -> C C will got A number if B transfer |
| 16:25.36 | anonymouz666 | is that you want? |
| 16:26.04 | pif | ok, let's say I make an assisted transfer, initially C will get B and then A once the transfer is accepted? |
| 16:26.47 | anonymouz666 | no |
| 16:27.03 | pif | so it only works on blind transfers |
| 16:27.08 | anonymouz666 | no |
| 16:27.23 | anonymouz666 | if B transfer A to C |
| 16:27.25 | anonymouz666 | atxfer |
| 16:27.34 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 16:27.42 | anonymouz666 | with 'o' in Dial you will see A number. |
| 16:27.44 | anonymouz666 | not B |
| 16:28.10 | pif | with asterisk transfer only, not sip client transfer, right? |
| 16:28.49 | anonymouz666 | I think you are correct. |
| 16:29.16 | anonymouz666 | 'cause SIP CLIENT transfer will probably use REFER |
| 16:29.22 | [TK]D-Fender | ruben23: Analog phones dn't transfer calls. the INTERFACE you physically plug them into gives them certain functionality. |
| 16:29.28 | anonymouz666 | and atxfer uses chan_local to execute the dialplan logic. |
| 16:30.20 | pif | thanks, that helps |
| 16:30.21 | *** join/#asterisk nicox (n=nicox@dsl-82-106.utaonline.at) |
| 16:30.23 | nicox | Hi |
| 16:33.26 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:33.39 | nicox | any idea how it can happen after about 1 week running a asterisk 1.4.22 with about 100.000 calls it can no longer dial out through IAX channels with key authentication because every try would be rejected. After restarting the asterisk its working well again |
| 16:34.00 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:34.02 | [TK]D-Fender | anonymouz666: people using *=based DTMF transfers should be shot |
| 16:35.10 | *** join/#asterisk monstertruck (n=Tanenbau@68.240.116.95) |
| 16:35.18 | monstertruck | hey kids |
| 16:36.21 | Thorn | hello, is there any way to put queue member ID into MONITOR_FILENAME? I tried ^{BRIDGEPEER:4} but that variable doesn't seem to be set |
| 16:36.51 | [TK]D-Fender | Thorn: pastebin your attempt |
| 16:36.59 | [TK]D-Fender | Thorn: and the original code |
| 16:37.13 | nicox | any idea how it can happen after about 1 week running a asterisk 1.4.22 with about 100.000 calls it can no longer dial out through IAX channels with key authentication because every try would be rejected. After restarting the asterisk its working well again |
| 16:39.19 | Thorn | [TK]D-Fender: I tried exten => s,n,Set(MONITOR_FILENAME=...-^{BRIDGEPEER}-...) before Queue(), but ^{BRIDGEPEER} evaluates to an empty string in the actual file name |
| 16:41.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:43.38 | ruben23 | [TK]D-Fender: ah ok like...sipura 3000 as interface and asterisk would do the call tansfer process...,correct? |
| 16:45.42 | *** part/#asterisk nicox (n=nicox@dsl-82-106.utaonline.at) |
| 16:49.16 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:52.27 | [TK]D-Fender | ruben23: Go read the SPA's manual to see how to do trasnfers with it |
| 16:52.47 | [TK]D-Fender | Thorn: ^{BRIDGEPEER} $ <------- |
| 16:52.53 | [TK]D-Fender | Thorn: no "^" |
| 16:52.58 | [TK]D-Fender | not |
| 16:53.47 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:54.33 | Thorn | [TK]D-Fender: as in ${BRIDGEPEER}? but this is evaluated when call enters queue, not when agent picks up if I'm not mistaken |
| 16:57.30 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:59.18 | [TK]D-Fender | Thorn: ^{} <- this is not valid syntax |
| 16:59.47 | [TK]D-Fender | Thorn: and it does not evaluate the filename when you START monitoring. |
| 17:00.00 | [TK]D-Fender | Thorn: It evaluates the name the moment you set that var. |
| 17:03.14 | Thorn | still empty string |
| 17:03.39 | anonymouz666 | you probably want MEMBERINTERFACE var? |
| 17:05.27 | Thorn | anonymouz666: very likely, but can it be used in extensions.conf? |
| 17:06.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:06.16 | anonymouz666 | you must access via that one through AGI passing as arg to queue application |
| 17:07.05 | Thorn | that's what I'm reading too |
| 17:08.32 | Qwell | [TK]D-Fender: ^{} is special with queues. it's silly, but.. |
| 17:09.10 | [TK]D-Fender | Qwell: Doesn't matter AFAIK. that var is still set BEFORE the queue |
| 17:09.39 | Qwell | it's set on answer or something |
| 17:09.43 | Qwell | it's...magic |
| 17:09.48 | Thorn | found http://lists.digium.com/pipermail/asterisk-bugs/2007-September/003349.html on MEMBERINTERFACE - there's mention of ^{} there |
| 17:12.38 | [TK]D-Fender | Thorn: Would be nice for you to show that the patch was accepted in mainstram * and when. |
| 17:12.50 | [TK]D-Fender | Thorn: Because from what I read there I got the words "would be nice". |
| 17:13.13 | [TK]D-Fender | Thorn: I can tell you all sorts of things that "would be nice"... |
| 17:14.01 | Qwell | putnopvut: ^^ can you elaborate? |
| 17:14.35 | anonymouz666 | putnopvut is the app_queue master |
| 17:14.39 | anonymouz666 | he he |
| 17:14.49 | Thorn | nope, empty string |
| 17:14.53 | putnopvut | hold on just a sec... |
| 17:15.56 | ruben23 | Qwell:if i connect asterisk with analog phones...i need ATA...it would automatically convert my analog phone to SIP... |
| 17:17.17 | [TK]D-Fender | ruben23: Close enough... |
| 17:18.16 | putnopvut | Thorn: the ^{} syntax works only for the MONITOR_FILENAME variable. It's set once a member answers the phone. I need to read the scrollback a bit further to see exactly what your problem is. |
| 17:18.35 | Thorn | putnopvut: that's where I use it |
| 17:18.37 | iratik | Its been a long time since i set up a fresh box .... can someone remind me what are the typical causes of one way audio? Here is some config files data http://www.pastie.org/335895 |
| 17:18.39 | putnopvut | You should use ^{MEMBERINTERFACE} |
| 17:19.09 | putnopvut | And you need to have setinterfacevar=yes in queues.conf for your queue. |
| 17:19.34 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 17:19.36 | Thorn | putnopvut: I didn' t set that, will try again |
| 17:19.40 | putnopvut | all right |
| 17:20.14 | [TK]D-Fender | iratik: Remeber the LAST time we went through this for God know how long? |
| 17:20.25 | [TK]D-Fender | iratik: NAT ISSUES. |
| 17:20.27 | [TK]D-Fender | ~sipnat |
| 17:20.28 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:20.29 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:20.44 | [TK]D-Fender | iratik: And like last time, its all still there. So go read it all over again |
| 17:20.46 | iratik | I know its NAT issues... but the issue was so long ago and such a quick fix i don't remember what i did |
| 17:21.05 | [TK]D-Fender | iratik: pile'o'settings. Now get to it |
| 17:21.10 | iratik | thanks |
| 17:22.13 | ruben23 | [TK]D-Fender:hmmm...so my analog phone cannot stablished call without VOIP |
| 17:22.37 | Qwell | putnopvut: woot, ty |
| 17:22.56 | putnopvut | Qwell: for what? |
| 17:23.02 | putnopvut | oh for the queue help? |
| 17:24.56 | Thorn | success! thanks putnopvut |
| 17:25.16 | putnopvut | Not a prob! |
| 17:25.50 | Thorn | so the recipe is setinterfacevar=yes in queues.conf and ^{MEMBERINTERFACE:4} in Set(MONITOR_FILENAME) |
| 17:27.14 | [TK]D-Fender | ruben23: thats the whole point of the ATA. |
| 17:28.04 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 17:33.02 | ruben23 | [TK]D-Fender:ok but still i can use my telco line as normal.... |
| 17:37.24 | [TK]D-Fender | ruben23: meaning? |
| 17:37.41 | [TK]D-Fender | ruben23: * does not sit "transparently" in the middle of your setup |
| 17:37.49 | [TK]D-Fender | ruben23: Calls do not jsut right straight through |
| 17:41.09 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:42.09 | echinos | I guess it must be possible to check the caller ID before answering, correct? |
| 17:42.25 | echinos | I just want a yes/no, I'm just curious |
| 17:42.41 | Poincare | echinos: that is possible |
| 17:43.11 | Poincare | that's how I keep anonymous marketeers from bothering me |
| 17:46.40 | echinos | Yeah, I want a blacklist that can operate pre-answer so I don't get charged for any airtime |
| 17:49.09 | Thorn | is it possible for a Queue() to only answer the calling channel when an agent picks up (to reduce charges for callers)? |
| 17:50.42 | kb3ien | possible, some telcos seem to disconnect callers after too many rings... dunno why... |
| 17:50.46 | mmattice | anybody doing voice recognition systems with *? |
| 17:51.07 | kb3ien | no i hope to get there soon. |
| 17:51.16 | kb3ien | voicemessages is killing me today... |
| 17:52.26 | kb3ien | anyone got voicemail stored in odbc ? |
| 17:52.48 | mmattice | kb3ien: a blue box used to be able to be used to trick the telco into thinking the phone was still ringing but you could talk to the party on the other end. |
| 17:52.57 | mmattice | I doubt that still works, but the cutoff is still there. |
| 17:53.52 | kb3ien | i assumed it had to do with dropping stale calls, but that might have something to do with the introduction of said 'feature'. |
| 17:59.00 | jm|laptop | anyone managed to use DISA over chan_mobile ? |
| 17:59.37 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 18:01.15 | bijit | what is a good value for MWI? |
| 18:02.22 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:06.33 | *** join/#asterisk diter (n=martin@83.140.111.160.dyn.rp80.se) |
| 18:08.28 | diter | I have a ATA box vood 121. But it seems that it can not save my changes when using the web intetface. Someone have a clue ? |
| 18:08.40 | diter | Is it looked ? |
| 18:09.17 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 18:10.05 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 18:10.24 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 18:11.11 | SuPrSluG | any multi tenant solutions out there for asterisk. boss is looking at enswitch and would like something comparable |
| 18:11.42 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 18:13.57 | *** join/#asterisk Joe_CoT (n=Joe_CoT@ubuntu/member/joeterranova) |
| 18:15.10 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:19.02 | kb3ien | well swapped out asterisk for one that has no odbc, but now the voicemail.conf file, although read, is ignored. |
| 18:23.18 | *** join/#asterisk stephank (n=urk@2002:52c5:cf78:0:21c:c4ff:fece:ea94) |
| 18:23.50 | *** join/#asterisk metfan2007 (n=jc@201.103.90.143) |
| 18:25.11 | metfan2007 | Hi all, I have a question, I'm getting a lot of "Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)" messages while trying to use all my ISDN channels with a lot of traffic, Is that a message that the TELCO has problems with all the traffic? |
| 18:26.02 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 18:26.12 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 18:28.48 | *** join/#asterisk AliBB (n=ali@41.224.171.70) |
| 18:29.20 | Deeewayne | woot Mets |
| 18:29.58 | kb3ien | well that borked it nicely. now i'll have to figure out what files it changed in /etc/asterisk when it `make install'`` or just roll back all changes an manually install the asterisk binary.... |
| 18:30.11 | Deeewayne | metfan2007: do you have all the b-channels configured ? |
| 18:30.32 | metfan2007 | Deewayne: Yes, using Hardlc with sangoma cards |
| 18:32.05 | *** join/#asterisk ambriento (i=bd785c80@gateway/web/ajax/mibbit.com/x-c82187286f41f154) |
| 18:32.26 | kb3ien | yep that was the path of least resistance.... |
| 18:33.00 | kb3ien | all my features are back! |
| 18:33.17 | Deeewayne | metfan2007: how many channels is "a lot of traffic" ? |
| 18:33.28 | *** join/#asterisk telnettech (n=telnette@206.48.21.148) |
| 18:33.47 | tiny | "fd == -1 in astman_append, should not happen" Any ideas what's with this message showing up in CLI |
| 18:33.48 | tiny | ? |
| 18:33.50 | metfan2007 | Deewayne: 21 E1s |
| 18:34.46 | jasonwoot | 21? that's almost 22! |
| 18:34.57 | metfan2007 | jasonwoot: hehehehe |
| 18:35.11 | telnettech | guys and gals, have question.......if you already installed zaptel before libpri and then asterisk, can you go back and do the make install for the libpri and then zaptel to get zap channels to work |
| 18:35.11 | kb3ien | voicemail show users for shows the message i left. the MWI is baroque, and i cannot connect, but its 'progress' or maybe regress... anywho... |
| 18:37.51 | telnettech | anybody? |
| 18:39.38 | Deeewayne | telnettech: I always build libpri, then dahdi/zaptel, then Asterisk |
| 18:39.38 | Nugget | telnet is eeeeeeevil! |
| 18:39.56 | telnettech | I AM NOT NUGGET!!!! LOL |
| 18:40.20 | telnettech | im one of the nicest people around |
| 18:41.21 | Deeewayne | metfan2007: do you know sangoma is configured correctly ? have you ever used that many channels at once ? you can enable pri debug and pastebin, maybe someone will get a chance to look at it |
| 18:41.23 | *** join/#asterisk dthomas (n=darkness@linode.caliginous.net) |
| 18:42.50 | *** join/#asterisk vader-- (n=me@c-71-226-192-99.hsd1.nj.comcast.net) |
| 18:42.53 | vader-- | hello |
| 18:43.13 | Nugget | hee |
| 18:43.34 | vader-- | do you guys know if it's possible to remove the URL button on cisco 7940G phones? my users are constantly hitting it and can't figure out how to get it back |
| 18:43.47 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
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| 18:50.50 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 18:50.54 | dthomas | Hi. Can anyone give me a hint why Asterisk seems to be ignoring dahdichanname=no in asterisk.conf? |
| 18:51.08 | dthomas | It's almost like asterisk.conf isn't getting used. |
| 18:52.25 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 18:56.14 | *** join/#asterisk zzbender (n=zzbender@ool-4b7faae2.static.optonline.net) |
| 18:56.15 | *** join/#asterisk sprite-- (n=sprite@12.228.1.97) |
| 18:56.21 | sprite-- | Anyone here use Adhearsion? |
| 18:56.29 | sprite-- | http://rafb.net/p/LlTP8Y91.html getting that error |
| 18:57.21 | zzbender | Hi! |
| 18:57.31 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 18:57.51 | zzbender | I've got a question about PRI's and Asterisk. Anyone free to help? :) |
| 18:58.04 | seanbright | ~ask |
| 18:58.04 | jbot | it has been said that ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:59.03 | zzbender | I see |
| 18:59.05 | zzbender | Thanks. |
| 18:59.30 | seanbright | so... ask your question |
| 18:59.31 | seanbright | :) |
| 18:59.33 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 19:00.28 | invalidrecord | sprite--: try ahn start . in the app dir same error? |
| 19:00.29 | zzbender | I'm attempting to route calls to a PRI from some SIP Phones. I've been mostly using the GUI from digium. The goal is to be able to dial any of the current 8 lines on the PRI, or have them dial into the Asterisk box |
| 19:01.16 | zzbender | If I attempt to make a call from the PRI, I dont get any answer from the PRI on the asterisk side |
| 19:01.47 | *** join/#asterisk Jerjer[mobile] (n=PhatJ@wsip-70-182-253-52.ks.ks.cox.net) |
| 19:01.50 | zzbender | The PRI card is configued as a PRI-net, ISDN 2 using 8 channels |
| 19:01.53 | Jerjer[mobile] | grr - WARNING[3544]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol |
| 19:02.02 | Jerjer[mobile] | i wish shit would get fixed |
| 19:02.03 | zzbender | and my board on the side is configured for 5ESS |
| 19:02.23 | anonymouz666 | Jerjer[mobile]! |
| 19:02.35 | anonymouz666 | maybe the URI is fucked up? |
| 19:02.42 | anonymouz666 | do you have a dump of this SIP message? |
| 19:02.50 | anonymouz666 | R-URI* |
| 19:02.57 | Jerjer[mobile] | we are getting probed |
| 19:04.01 | zzbender | So, its not working. and I don't really know where to start toubleshooting it. All internal stuff seems to work fine ( SIP to SIP ) but I can't get anything in or out of the PRI |
| 19:04.21 | seanbright | zzbender: have you asked in #asteriskgui? |
| 19:04.31 | seanbright | zzbender: assuming you are doing all of this from within the GUI |
| 19:05.26 | zzbender | seanbright: I did not know there was an #asteriskgui channel. Would it be best if I went there? I dont mind editing files from a command line - this just seemed quicker. |
| 19:05.38 | Jerjer[mobile] | anonymouz666: lamers are purposely sending invalid SIP and IAX packets at us |
| 19:05.48 | *** join/#asterisk invalidrecord (n=fares@87.113.88.31.plusnet.pte-ag2.dyn.plus.net) |
| 19:05.50 | beek | zzbender: the T1 card in your Asterisk box should be set for PRI_CPE signalling. |
| 19:06.01 | Jerjer[mobile] | i can mitigate the SIP problems, but there is not a god damn thing I can do about IAX |
| 19:06.04 | seanbright | zzbender: well the problem is that any custom stuff you do in your confs will get blown away when you make changes again in the GUI |
| 19:06.22 | Jerjer[mobile] | yet Digium refuses to even acknowledge they have a problem |
| 19:07.19 | anonymouz666 | Jerjer[mobile]: try closing the IAX port :) |
| 19:07.32 | Jerjer[mobile] | it may come to that |
| 19:07.51 | zzbender | beek: I'll try that. I don't have many options on the Telephony server that I'm trying to connect to. ( Vocera ) If you've heard of them |
| 19:07.56 | Jerjer[mobile] | i know of at least two other VoIP providers that stopped offering IAX |
| 19:09.01 | beek | zzbender: I haven't. |
| 19:10.06 | beek | zzbender: Basically, if the port faces towards the PSTN then it's pri_cpe. If it faces away from the PSTN then its pri_net. |
| 19:10.38 | beek | zzbender: Also, make sure that it accepts timing from the telco. |
| 19:11.07 | zzbender | beek: its a VoWiFi voice recognition system ( Like StarTrek ) thats deployed in Healthcare |
| 19:11.16 | anonymouz666 | Jerjer[mobile]: write let's say a pike module for IAX ;) |
| 19:13.13 | zzbender | beek: I can set my other server to the following: NI1,DMS,5ESS,4ESS,NT1,CTR4,QTE,NE1, or QNT |
| 19:14.32 | Jerjer[mobile] | anonymouz666: we need something... i fear the protocol itself is the problem |
| 19:14.56 | Jerjer[mobile] | e.g. traffic amplification |
| 19:18.42 | jaytee | hehehe, http://wearehanging.files.wordpress.com/2008/04/mao_rtfm.jpg |
| 19:20.02 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:22.48 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 19:22.56 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 19:25.31 | *** join/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 19:26.44 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 19:27.37 | meuserj | ok.. I'm trying to port my 1.2 config to work in 1.4... I have two sip phones registered with the server and "sip show peers" shows them both as reachable, but when I try to call between the phones, asterisk claims that the extension doesn't exist. |
| 19:28.21 | kb3ien | seems that when i added additional bindings i broke mwi... http://pastebin.com/m67c3a9f4 |
| 19:29.34 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:29.36 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 19:31.30 | *** join/#asterisk dieguito84 (n=diego@host164-185-dynamic.13-79-r.retail.telecomitalia.it) |
| 19:31.41 | kb3ien | still polycoms seem the best thing going... |
| 19:35.24 | [TK]D-Fender | kb3ien: Dec 10 14:40:41] WARNING[30523]: pbx.c:3810 __ast_pbx_run: Channel 'SIP/0000540491-00b50308' sent into invalid extension '0000540491' in context 'outbound', but no invalid handler |
| 19:35.34 | [TK]D-Fender | kb3ien: this is a dialplan error. Go look at your dialplan |
| 19:36.25 | kb3ien | [TK]D-Fender what /should/ be dialed afaict is 7201 |
| 19:36.37 | [TK]D-Fender | kb3ien: rebooted your phone? |
| 19:36.43 | kb3ien | yes. |
| 19:37.44 | kb3ien | be nice if there was a syntax checker for that file. i'm going to merge the reg statements together see if that helps. |
| 19:38.15 | kb3ien | is it not normal to have one number to dial to check voicemail system wide? |
| 19:38.56 | jaytee | kb3ien, Gvim includes an Asterisk syntax checker. It isn't 100% but mostly works. |
| 19:38.57 | kb3ien | good to know. |
| 19:40.30 | kb3ien | is that not part of vim? |
| 19:40.44 | jaytee | don't think so |
| 19:40.58 | jaytee | Gvim is the gui version |
| 19:43.32 | kb3ien | a better than vim-xaw? |
| 19:43.53 | jaytee | never used that |
| 19:44.32 | kb3ien | i cant recall ever using anything but vi in so long... NANO last time i was working with thinclients... |
| 19:49.21 | [TK]D-Fender | kb3ien: make very sure your phone is actually picking up the changes |
| 19:54.24 | *** join/#asterisk reneger (n=reneger@dslb-088-078-120-087.pools.arcor-ip.net) |
| 19:59.07 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-197-154-rrdg-esr-2.dynamic.isadsl.co.za) |
| 20:01.14 | kb3ien | hihihi error 401 in apache logs! |
| 20:01.50 | vader-- | do you guys know if it's possible to remove the URL button on cisco 7940G phones? my users are constantly hitting it and can't figure out how to get it back |
| 20:02.22 | c017 | What would be the right configuration for just two x-lite clients using asterisk 1.4.11? I've already spent a day on this but no luck :X |
| 20:03.14 | c017 | Sorry for such a noob question. |
| 20:06.00 | kb3ien | sweet. its working. it asks for 'mailbox' everytime. |
| 20:06.03 | c017 | Ok it somehow magically worked! |
| 20:06.14 | kb3ien | and the mwi is broken but i can GET the messages. |
| 20:09.07 | ruben23 | hi what would be the best setup for an asterisk boxes.....used as gateway server or just a server connected to a LAN network.... |
| 20:10.27 | kb3ien | i guess it depends on what your tring to connect, right? |
| 20:11.07 | ruben23 | <PROTECTED> |
| 20:11.53 | kb3ien | i'll assume that inbound and outbound calls to the rest of the world matter, and put it where it can get a public IP to talk to your sip carriers. |
| 20:13.15 | ruben23 | kb3ien:ok so ill setup it as a gateway box..facing the public |
| 20:15.11 | ruben23 | kb3ien: ill setup it like this Internet==>(eth0)asterisk(eth1)==>switch==>ATA's==>POTS is this a good setup... |
| 20:17.12 | kb3ien | works for me. if your internet is reliable enough. putting the ast box offsite is good if not, that way at least ppl can leave you voicemail or you can ring your cellphone as a backup... |
| 20:17.35 | kb3ien | assuming by POTS you mean Plain Old Telephones. |
| 20:18.48 | ruben23 | <PROTECTED> |
| 20:20.27 | ruben23 | <PROTECTED> |
| 20:21.31 | kb3ien | ruben23: in what way handle the secuirty for my network? are you using NAT? |
| 20:21.45 | kb3ien | if so i'd make the nat a separte (virtual) box. |
| 20:22.03 | kb3ien | but the asterisk box should have some sort of port filtering i'd wager. |
| 20:22.24 | ruben23 | kb3ien:yeah....i should chenge my network setup that my asterisk be only part of my LAN... |
| 20:23.35 | ruben23 | kb3ien:how you mostly place your asterisk boxes on your network...? |
| 20:24.13 | kb3ien | i had ast 1.2 running nicely on a NAT box a few yeas ago. a 180 MHz 603e! wasnt pretty and when it died it took the whole house down. (it was a hobby box tho.) |
| 20:24.44 | kb3ien | i go for upstream where there is good cheap pipe, when i can. |
| 20:26.03 | kb3ien | i also dont use NAT, it takes long enough to make anything work without screwing with packets then seeing if the software can recoverd the original intent. That's surprisingly an uncommon position to take i hear. |
| 20:27.31 | ruben23 | <PROTECTED> |
| 20:28.11 | ruben23 | <PROTECTED> |
| 20:28.52 | kb3ien | right now the phones are on public IPs attached to an ADSL router. The asterisk box is sitting in a data center on an ethernet feed. |
| 20:29.54 | kb3ien | if i were doing this at home sans datacenter i'd put ever phone and the asterisk box on the same network such that the router is not involved in ATA->asterisk ethernetworking. |
| 20:30.28 | kb3ien | if i dont have enough IPs for every ATA, i'd have 2 network addresses on the asterisk box, one public and one rfc1918. |
| 20:30.54 | kb3ien | the phones would talk over the 1918 network, and the origination and termination to the internet would naturally use the public connexion. |
| 20:32.06 | kb3ien | when you resort to 1918 things aren't optimal, you might need to NAT or otherwise provide for DNS and even NTP. |
| 20:32.23 | *** part/#asterisk Joe_CoT (n=Joe_CoT@ubuntu/member/joeterranova) |
| 20:32.30 | *** join/#asterisk Joe_CoT (n=Joe_CoT@ubuntu/member/joeterranova) |
| 20:32.46 | Joe_CoT | is there a way to see what dtmf mode is being used on an active channel? |
| 20:33.04 | [TK]D-Fender | kb3ien: no need for * to have a public IP |
| 20:33.17 | ruben23 | <PROTECTED> |
| 20:33.40 | [TK]D-Fender | And phones should almost never need to bother with public IP's |
| 20:34.18 | ruben23 | <PROTECTED> |
| 20:34.54 | kb3ien | i dont ever see the need for anything not to have a public IP unless the network is so large that renting IP space is a problem. |
| 20:34.55 | sprite-- | sprite.rb:14: undefined method `proxy' for #<Object:0x7fe9edd6b3a0> (NoMethodError) |
| 20:35.08 | [TK]D-Fender | ruben23: I am saying that having * BEHIND a normal router is just fine |
| 20:35.34 | sprite-- | Trying use Adhearsion.proxy.call_into_context. I tried the fix posted on the Adhearsion site by mapping the stuff in events, no luck;. |
| 20:35.55 | *** join/#asterisk qdk (n=qdk@79.138.241.29.bredband.3.dk) |
| 20:36.01 | ruben23 | [TK]D-Fender:yeah the simplest kind of setup i imagine... |
| 20:36.06 | kb3ien | WARNING[30701]: app_voicemail.c:8432 vm_authenticate: Couldn't read username not sure what this is about. |
| 20:37.01 | [TK]D-Fender | kb3ien: No entry or bad DTMF mode |
| 20:37.08 | [TK]D-Fender | kb3ien: this is not a "mystery" |
| 20:39.43 | kb3ien | no entry for what? mailbox=56404@thisvmcontext is present in sip.conf |
| 20:40.03 | *** part/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 20:40.47 | *** join/#asterisk bijit (n=benji@200.122.158.243) |
| 20:42.12 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.4) |
| 20:42.14 | kb3ien | i dont want to dial my extension, i want to have it check /my/ voicemail by default. |
| 20:43.22 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 20:48.27 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:48.58 | *** join/#asterisk monstertruck (n=Tanenbau@70.2.188.111) |
| 20:49.05 | monstertruck | hi |
| 20:49.19 | monstertruck | is there a reliable way to get the ip address of an sip user? |
| 20:49.32 | monstertruck | header(via) is not working |
| 20:49.45 | kb3ien | i can force the mailbox with this: exten => 7201/0000540491,1,VoicemailMain(56404@mycontext) |
| 20:49.57 | seanbright | monstertruck: core show function SIPPEER |
| 20:49.59 | kb3ien | id doesnt fix the MWI led. |
| 20:52.19 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 20:57.32 | *** part/#asterisk korihor (n=korihor@201.210.239.172) |
| 20:58.58 | *** join/#asterisk eit (n=eit@64.122.178.15) |
| 21:00.01 | [TK]D-Fender | kb3ien: Show me the peer |
| 21:06.00 | *** join/#asterisk davidc (n=david@netman1.us.sargasso.net) |
| 21:07.13 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 21:07.39 | kb3ien | http://pastebin.com/d2ac1f474 |
| 21:08.30 | CGMChris | I use SIP VoIP (no Zapta), my router supports SIP prioritization. Whenever my network link is near capacity, my latency increases to above 150ms and I get echo. Aside from adding more bandwidth, is there a way to cancel the echo that occurs from the latency with an addon or asterisk setting? |
| 21:08.41 | *** part/#asterisk eit (n=eit@64.122.178.15) |
| 21:08.55 | kb3ien | should the context into which calls are made mycontext-dial and the context in voicemail.conf be the same? |
| 21:13.36 | *** join/#asterisk telecos (n=sergio@87.219.167.0) |
| 21:13.45 | kb3ien | sometimes one has to set the qos to use a number that is below the actual bandwidth by a few percent. |
| 21:13.58 | kb3ien | found that out the hard way. |
| 21:14.13 | kb3ien | still 150ms is a bit nuts. |
| 21:15.17 | CGMChris | sip show peers sits right at 55ms when my internet connection is idle. Is that too high? |
| 21:15.26 | Joe_CoT | is there a way to log when key dtmf tones are detected, or what dtmf mode is currently being used on a channel |
| 21:15.31 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 21:15.33 | CGMChris | that is, 55ms to my SIP provider |
| 21:15.47 | tzanger | got ya beat there... 3ms :-) |
| 21:16.12 | jasonwoot | Joe_CoT: I don't trust that figure on my system, it is always consistenly higher than reported by the network and OS level |
| 21:16.13 | kb3ien | my best is 5 ms. |
| 21:16.39 | CGMChris | Looks like my ping time to google is ~45ms, yahoo is ~70ms... so my sip provider is in the middle at 55. |
| 21:16.59 | CGMChris | so, I need to get it much much lower. |
| 21:17.57 | CGMChris | how do you get 3 and 5 ms? are you on an OC-XXX connection? |
| 21:18.05 | jasonwoot | I have found that, on my system, it's always inaccurate by about double |
| 21:18.27 | JonOnt | Hey guys, my incoming cid on my sip trunk comes in as +NNNXXXXXXX, im trying to strip the + from the number, this should do it (as long as its in the right context), right? "exten => _X.,1,Set(CALLERID(num)=${CALLERID(num):2:12})" |
| 21:18.55 | JonOnt | CGMChris, I get 4ms on my radio link |
| 21:19.18 | CGMChris | JonOnt: 4ms to your gateway or 4ms to your SIP provider? |
| 21:19.53 | JonOnt | CGMChris, gateway |
| 21:20.03 | CGMChris | JonOnt: I'm talking about to the SIP provider. |
| 21:21.05 | [TK]D-Fender | kb3ien: mailbox=56404 <-- does not include that context |
| 21:21.36 | [TK]D-Fender | JonOnt: exten => _X.,1,Set(CALLERID(num)=${CALLERID(num):1}) |
| 21:22.05 | JonOnt | [TK]D-Fender, hey TK, i was wondering if you might help me... lol, you rock |
| 21:23.20 | JonOnt | [TK]D-Fender, now, I have that in [from-trunk-sip-bandwith] in extentions_custom.conf, should that be in a diffrent file? |
| 21:23.40 | [TK]D-Fender | JonOnt: Its YOUR dialplan... you tell me. |
| 21:23.45 | [TK]D-Fender | already knows the answer |
| 21:23.57 | [TK]D-Fender | just goes through the motions anyway |
| 21:25.08 | JonOnt | [TK]D-Fender, well, thats the context for my incoming sip trunk, but is extentions_custom the right place, it doesnt seem to be working |
| 21:25.31 | [TK]D-Fender | JonOnt: the right place depends on where your PEER sends calls assuming it even hits a peer |
| 21:25.51 | JonOnt | [TK]D-Fender, that went over my head |
| 21:26.47 | [TK]D-Fender | JonOnt: the entry that shuold be used to auth & match incoming calls from a provider. |
| 21:26.59 | [TK]D-Fender | figures this confirms so much more. |
| 21:27.42 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-228-64.unitz.ca) |
| 21:28.12 | eric2 | with ip phones that have more than one line capability, is there a way to show when a line is in use by having the line light up? |
| 21:28.26 | eric2 | ie: office environment... |
| 21:29.36 | [TK]D-Fender | checkout time, BBIAB |
| 21:31.07 | *** part/#asterisk dthomas (n=darkness@linode.caliginous.net) |
| 21:31.39 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 21:31.51 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 21:32.35 | jasonwoot | "Polycom's CX family of products optimized for Microsoft Office Communications Server 2007. |
| 21:32.55 | jasonwoot | i hold that in high regard, like Vista certification |
| 21:33.21 | kb3ien | bakc |
| 21:33.23 | kb3ien | back |
| 21:38.30 | kb3ien | well if anyone can unravel the MWI on my polycoms, i'm out of ShinyNewDonkeys(tm) but i'll see what we have... |
| 21:40.36 | meuserj | ok.. I'm porting my 1.2 config to 1.4. It seems that it is attempting to load extensions.ael instead of extensions.conf. I put noload => pbx_ael.so and load => pbx_config.so into modules.conf, and it stopped loading the ael file, but it's still not loading the conf file. |
| 21:42.07 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 21:42.22 | *** join/#asterisk ESCulapio__ (n=ESCulapi@adsl-224-238.tricom.net) |
| 21:42.24 | *** part/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV) |
| 21:42.29 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 21:42.54 | ESCulapio__ | hi, Hell my please. |
| 21:43.04 | codefreeze-lap | meuserj: well, if you misnamed the file, it has bad permissions, or something like that, it might not load... |
| 21:43.05 | bkruse | lol |
| 21:43.24 | ESCulapio__ | I quireo set el sip codec for a call with ${SIP_CODEC} |
| 21:43.32 | ESCulapio__ | QUIEN ME PUEDE AYUDAR |
| 21:43.35 | ESCulapio__ | sorry |
| 21:45.42 | meuserj | codefreeze-lap: arg.. that should have been the first thing I checked... that was it 640 root:root... |
| 21:45.55 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:47.46 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 21:48.23 | bijit | can we speak spanish here? |
| 21:49.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:49.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:50.29 | *** join/#asterisk Cobra2599 (i=user@74.201.52.27) |
| 21:50.46 | Cobra2599 | Can anyone tell me why my IVR dont play the sound file? |
| 21:50.55 | Cobra2599 | i dont get any error messages |
| 21:51.13 | Cobra2599 | it acts like it is playing it but i cant hear anything |
| 21:51.39 | Joe_CoT | Cobra2599, do you see in the cli the call being made to play the file? If you do, either it can't find the file, it doesn't have permissions to play it, or it's in an unsupported format |
| 21:51.47 | *** join/#asterisk intrin (n=intrin@99-196-130-98.cust.wildblue.net) |
| 21:52.07 | Cobra2599 | yeah |
| 21:52.07 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:52.57 | Cobra2599 | <PROTECTED> |
| 21:53.04 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 21:53.06 | Cobra2599 | main is my file |
| 21:53.16 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 21:55.50 | intrin | whats the best flavor of linux to run asterisk on |
| 21:56.23 | jaytee | I like strawberry myself but some opt for vanilla or coffee |
| 21:56.29 | intrin | :p |
| 21:56.36 | intrin | distro! |
| 21:56.43 | jaytee | oh! distro :-) |
| 21:56.52 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 21:56.58 | jaytee | well, I prefer either RHEL 5 or CentOS 5 |
| 21:57.10 | jaytee | others like Debian |
| 21:57.16 | jaytee | whatever works for you |
| 21:57.27 | intrin | ill go with debian, i got a cd o f that some where :D |
| 21:57.33 | Joe_CoT | intrin, entirely arbitrary. I prefer running it on Ubuntu/Debian. Some of the gui software like FreePBX is written only with Redhat/CentOS in mind |
| 21:57.47 | intrin | ok cool |
| 21:57.47 | intrin | thanks |
| 21:57.51 | intrin | i got deb and ubun |
| 21:57.57 | intrin | gotta go install a door : |
| 21:57.58 | intrin | :( |
| 21:58.19 | cvnet | Funny, when i used my voip or direct DID to call my voice gateway which uses the gsm gateway it works fine, but when I use the A2Billing to make the same call, even if the other side picks the phone it still rings in calling part, any suggestions? |
| 21:59.29 | jaytee | cvnet, if you have the r option in your Dial command get rid of it? |
| 22:00.12 | cvnet | in the script u mean? |
| 22:00.17 | jaytee | yeah |
| 22:00.28 | cvnet | hum, never touched the script |
| 22:00.33 | cvnet | how does the option looks like? |
| 22:00.38 | jaytee | first time for everything |
| 22:00.41 | cvnet | dial(...., r)? |
| 22:01.28 | jaytee | the r option in Dial() provides ringback indication but it can often cause calls to get screwed up |
| 22:02.25 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 22:02.30 | cvnet | ok thanks, let me try taht |
| 22:06.49 | *** join/#asterisk clive- (i=ident@dsl-242-156-152.telkomadsl.co.za) |
| 22:07.02 | clive- | is there a Brad here from digium support? |
| 22:09.49 | *** part/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 22:10.24 | *** part/#asterisk Cobra2599 (i=user@74.201.52.27) |
| 22:10.35 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 22:10.46 | clive- | is anyone here from digium? |
| 22:13.23 | bkruse | nope |
| 22:13.48 | bkruse | clive-: I do know of who you are talking to, if you have a support contract of defective product, call em! |
| 22:16.59 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:17.05 | clive- | bkruse, thanks |
| 22:17.09 | bkruse | npnp |
| 22:17.14 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:20.40 | *** join/#asterisk DonX (n=alphasix@68-185-206-223.dhcp.dntn.tx.charter.com) |
| 22:20.44 | DonX | Hello, |
| 22:21.14 | DonX | Does anyone have a good howto page for installing app_*xfax on 1.4 ? |
| 22:21.33 | DonX | looks like the old pages aren't on soft-switch.org anymore |
| 22:23.55 | *** part/#asterisk clive- (i=ident@dsl-242-156-152.telkomadsl.co.za) |
| 22:27.16 | kb3ien | http://pastebin.com/d2ac1f474 gets me every feature i want out of the polycoms EXECPT the MWI LEDs and icons... |
| 22:27.30 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-3186893e7abbd0d2) |
| 22:27.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:27.50 | kb3ien | i changed the context on the user with a new messages from mycontext-dial to mycontext but to no avail. |
| 22:30.32 | [TK]D-Fender | kb3ien: vmexten=mycontext <- remove |
| 22:30.55 | [TK]D-Fender | kb3ien: mailbox=56404@mycontext <-- |
| 22:32.01 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 22:32.30 | kb3ien | [TK]D-Fender has solved it!!! |
| 22:32.47 | [TK]D-Fender | kb3ien: You're welcome |
| 22:33.05 | kb3ien | im oing to hangout w/ the family now. thanks aain! |
| 22:38.14 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 22:38.14 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 22:41.44 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:46.36 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:51.29 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 22:57.28 | *** join/#asterisk zamolxes (n=zamolxes@83.166.220.142) |
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| 23:00.27 | zamolxes | hey. can you recommend me a good VOIP reference? (not related to asterisk, more general stuff). It should cover terminology (like dialplan, extension etc), protocols (like sip, rtp), common setups & architectures. It should be useful for someone completely new to this domain, but otherwise decently skilled (unix, networking, programming etc). I need some material for me and my colleagues. :) |
| 23:00.33 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 23:01.37 | zamolxes | I did plenty of hacks with asterisk , got the job done, but I feel I need a better knowledge of the problem-domain, not just trial&error and copy-pasting bits of working config from docs |
| 23:04.24 | *** join/#asterisk Segnale007 (n=Pietro@host218-255-dynamic.8-79-r.retail.telecomitalia.it) |
| 23:11.42 | SkramX | zamolxes have you checked out the Asterisk Oreilly book? |
| 23:11.54 | SkramX | Im quite sure it covers VoIP basis before jumping into code |
| 23:11.54 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:11.55 | SkramX | ~book |
| 23:11.56 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:12.38 | zamolxes | SkramX: thank you. I guess I'll use wikipedia/voip-info for completion |
| 23:13.09 | SkramX | yeah I think that's a good idea |
| 23:14.15 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 23:14.52 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
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| 23:41.51 | JonOnt | Hey guys, whos awake? |
| 23:42.19 | Corydon76-dig | ~ask |
| 23:42.20 | jbot | methinks ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:44.52 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 23:46.26 | JonOnt | fair enough.. can any one take a look at this paste... i'm trying to strip the + from the incoming calls, doesnt seem to be working.. http://asterisk.pastebin.com/d5f54d5f |
| 23:47.42 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:48.40 | [TK]D-Fender | JonOnt: pastebin the failed call. |
| 23:51.45 | JonOnt | [TK]D-Fender, does this help? http://asterisk.pastebin.com/d68f8a88a |
| 23:52.01 | JonOnt | [TK]D-Fender, and the calls are still coming in, i just cant get rid of the darned + |
| 23:53.16 | [TK]D-Fender | JonOnt: The dialplan code you showed us is not even being USED |
| 23:54.14 | [TK]D-Fender | JonOnt: And FreePBX is not supported in here. if you want to hack your code into their processing, ask in their support channel as to if & how it might be doable. |
| 23:54.16 | [TK]D-Fender | ~freepbx |
| 23:54.17 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:55.19 | *** join/#asterisk gloin (i=me@unaffiliated/gloin) |
| 23:55.33 | cvnet | what does the option r in dial stands for? |
| 23:55.47 | [TK]D-Fender | cvnet: "core show application dial" |
| 23:55.48 | JonOnt | [TK]D-Fender, took that custom code right from thier site |
| 23:56.05 | [TK]D-Fender | JonOnt: Meaningless. it isn't getting executed. |
| 23:56.19 | SkramX | cvnet: http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 23:56.19 | cvnet | Funny, when i used my voip or direct DID to call my voice gateway which uses the gsm gateway it works fine, but when I use the A2Billing to make the same call, even if the other side picks the phone it still rings in calling part, any suggestions? |
| 23:56.19 | [TK]D-Fender | JonOnt: Go ask in their support channel about this. |
| 23:56.41 | [TK]D-Fender | cvnet: Stop using "r". "r" = EIVL |
| 23:56.41 | SkramX | sounds like an a2billing issue ;) |
| 23:56.49 | [TK]D-Fender | EVIL* |
| 23:57.15 | JonOnt | [TK]D-Fender, thanks man |
| 23:57.23 | cvnet | [TK]D-Fender they are not using the r, looked at the php script for last 1 hour (very busy script) couldnt find it there |
| 23:57.40 | [TK]D-Fender | cvnet: Go show us the call with the problem. |
| 23:57.55 | *** join/#asterisk zFinX^ (n=hejhopp@81-233-19-154-no30.tbcn.telia.com) |
| 23:57.56 | zFinX^ | http://www.sexyemilie.com/?id=519390 |
| 23:57.58 | zFinX^ | check it out :D |
| 23:57.59 | *** part/#asterisk zFinX^ (n=hejhopp@81-233-19-154-no30.tbcn.telia.com) |
| 23:57.59 | cvnet | ok one min |
| 23:58.57 | gloin | ok, that was random |
| 23:59.56 | SkramX | welcome to IRC |
