| 00:00.06 | Qwell | s/minute/second/ |
| 00:00.27 | Qwell | s/at this.*/for this call, up until right now/ |
| 00:02.30 | jblack | I'm sure they don't get a whole lot of donations |
| 00:03.49 | ManxPower | cappiz: you don't have an entry for that sip username. |
| 00:04.04 | cappiz | hum :) |
| 00:04.20 | ManxPower | so you don't have a [SIP username from provider] section of sip.conf |
| 00:06.28 | ManxPower | I don't know why it would play ss-noservice unless you have that set up or are using a GUI. |
| 00:06.36 | *** part/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 00:06.39 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 00:07.35 | apocn | Hello, is anyone experienced with queuemetrics? |
| 00:10.38 | apocn | Im using the queuemetrics queueDial.agi for monitoring outbound calls using: "exten => xxx,1,DeadAGI(queueDial.agi|Number|DialString|QueueName|Agent)", but when I make the call it doesnt work (in the debugging console I see that its executed but jumps to the next step (hangup))... any help? |
| 00:12.21 | apocn | it says: Launched AGI Script /var/lib/asterisk/agi-bin/queueDial.agi and then: AGI Script queueDial.agi completed, returning 0 |
| 00:13.56 | *** join/#asterisk AndyGraybeal (n=andy@node178.34.251.72.1dial.com) |
| 00:14.21 | gene2 | i'm new to configuring asterisks, i played around with it about 3 years ago and still running that old version, I decided to try out asterisk-gui and something is off, each time i add a provider asterisk-gui writes bunch of empty pages to to the end of extensions.conf file and never saves the provider anywhere |
| 00:14.53 | gene2 | i also see this in the console "fd == -1 in astman_append, should not happen" |
| 00:15.57 | gene2 | any clue anyone? |
| 00:17.22 | gene2 | actually it is writing empty pages to probably more config files, i just noticed users.conf is the same way |
| 00:22.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:24.06 | *** join/#asterisk [Outcast] (n=outcast@203-114-166-26.eth.sta.inspire.net.nz) |
| 00:25.38 | jblack | lmadson: that would be nice. |
| 00:25.46 | *** part/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 00:25.56 | jblack | There's a small one already running, managed from chicago, that I'm in. |
| 00:26.16 | lmadsen | ya, I've just never heard of anyone really doing it anymore |
| 00:26.41 | lmadsen | I've got a script that Ed Guy III wrote that allows people to sign up online that I might modify to put the data into a DB or something |
| 00:26.46 | jblack | There's still a couple commies in the world. ;) |
| 00:26.53 | *** join/#asterisk Twotone (n=Sp00gE@24-158-189-53.dhcp.jcsn.tn.charter.com) |
| 00:27.07 | lmadsen | heh |
| 00:27.15 | lmadsen | I'm running off to the gym for a bit... back latah |
| 00:27.24 | Twotone | Is there a known DMZ issue on the linksys WRT54G that will not let a SIP phone work properly? |
| 00:28.00 | Twotone | externally |
| 00:28.19 | jblack | twotone: No idea. sorry |
| 00:29.22 | lmadsen | you should not need to change anything... |
| 00:29.36 | lmadsen | or is Asterisk behind the router? (I think I misread) |
| 00:29.38 | [Outcast] | Twotone: no it should work fine as you have set externip and localnet setting |
| 00:29.52 | lmadsen | what [Outcast] said |
| 00:29.55 | Twotone | lol |
| 00:30.12 | Twotone | TY for the help |
| 00:30.16 | [Outcast] | np |
| 00:30.31 | [Outcast] | Twotone let me know if it does not work |
| 00:30.33 | Twotone | I've set all that up. It's either the router that the phone is behind or the phone itself |
| 00:30.37 | cappiz | lmadsen: does this look OK http://pastebin.com/d2b876106 ? |
| 00:30.57 | [Outcast] | have you set nat=yes for the extension? |
| 00:31.02 | Twotone | yeah |
| 00:31.17 | Twotone | I set it up identical to another external extension that I have which works |
| 00:31.47 | [Outcast] | so your setup is asterisk(DMZ)--->Router--->Modem--->internet--->modem--->router--->phone ? |
| 00:32.13 | jblack | cappiz: Looking |
| 00:32.28 | Twotone | asterisk >> Router >> Modem >> Internet >> Modem >> router >> Phone(DMZ) |
| 00:32.34 | Twotone | DMZ'ing the phones |
| 00:32.38 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 00:32.43 | jblack | cappiz: This isn't comprehensible. Please paste the entire file. |
| 00:33.02 | cappiz | the sip_additional? |
| 00:33.22 | *** part/#asterisk rEph (n=robf@24.214.206.254) |
| 00:33.47 | [Outcast] | Twotone do you have control of both routers? |
| 00:34.05 | Twotone | Yes but not at the moment |
| 00:34.17 | jblack | What is sip_additional ? |
| 00:34.29 | jblack | >> ManxPowerHexDump: sip_nat.conf and sip_additional.conf are trixbox/freepbx stuff |
| 00:34.43 | [Outcast] | need to have asterisk in the dmz of the router, the phone should not matter if it is the DMZ or not |
| 00:34.46 | jblack | If that's the case, you need to try #freepbx or #trixbox instead. |
| 00:35.04 | jblack | [outcast]: Mostly. It depends upon whether enable_redirect is enabled. |
| 00:35.31 | [Outcast] | reinvitiing or redirect should be turned off |
| 00:35.39 | jblack | Paron, I meant reinvite. |
| 00:36.02 | Twotone | [outcast]: I'm just confused as to why one phone works behind a dlink but another doesn't behind a linksys :| |
| 00:36.04 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 00:36.34 | phix | hey |
| 00:36.39 | [Outcast] | i have used linksys for all my SOHO install and they work great |
| 00:36.46 | phix | http://rafb.net/p/6iVycT49.html |
| 00:37.06 | [Outcast] | Twotone, you have most likely missed something very small |
| 00:37.33 | cappiz | jblack: http://pastebin.com/d160c5bc0 |
| 00:38.03 | Twotone | [outcast]: This could be the case but myself and another have checked it. I'm not 100% knowledgable as to how to set asterisk up yet. It was already set up before I started messing with it. |
| 00:38.03 | jblack | cappiz: Are you running trixbox or freepbx? |
| 00:38.05 | phix | This works but is it the best way to do it? |
| 00:38.14 | phix | can any one see any issues I have have with this ? |
| 00:38.14 | cappiz | trix |
| 00:38.26 | jblack | cappiz: THen I can't help you. Try #trixbox |
| 00:38.26 | cappiz | with freepbx:P |
| 00:38.31 | jblack | It's in the /topic |
| 00:38.43 | phix | also, when I call an asterisk box on landline it is always engaged, but show channels says there is nothing happening |
| 00:38.44 | Twotone | [outcast]: What configs would you need to look at in order to tell? |
| 00:39.02 | cappiz | yeah right |
| 00:39.08 | [Outcast] | sip.conf |
| 00:39.31 | jblack | cappiz: That file you pasted looks nothing like I'm accustomed to. I can only give you advice that won't work. What's the point of that? |
| 00:39.35 | phix | hey |
| 00:40.14 | cappiz | so the asterisk connfig doesnt look like them same? |
| 00:40.55 | jblack | No. That's why people running trixbox are told to join #trixbox. |
| 00:41.15 | cappiz | k |
| 00:41.30 | Twotone | [outcast]: The whole file or just the ext part I'm trying to setup? |
| 00:41.47 | Shaun2222 | lmadsen: around still |
| 00:42.15 | Twotone | [18:27] <@lmadsen> I'm running off to the gym for a bit... back latah |
| 00:42.25 | *** join/#asterisk obnauticus (n=obnautic@c-24-22-14-101.hsd1.wa.comcast.net) |
| 00:43.34 | [Outcast] | give me the whole file, just be sure to comment or change passwords. |
| 00:45.15 | Twotone | know how to copy the whole file with putty? |
| 00:46.57 | [Outcast] | cat the file, copy from buffewr |
| 00:47.52 | [Outcast] | i have a better idea, just let me see your general settings for now. |
| 00:48.00 | Twotone | ok |
| 00:48.07 | *** join/#asterisk _wishbone (n=wishbone@189.70.22.252) |
| 00:48.41 | Twotone | Can I paste it to you instead of using pastebin? |
| 00:48.56 | Twotone | in PM of course |
| 00:51.08 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177887020.dsl.bell.ca) |
| 00:51.21 | [Outcast] | PM it I guess |
| 00:51.32 | Twotone | http://pastebin.com/d17cb865b there's the top of sip.conf |
| 00:52.35 | *** join/#asterisk Agrajag- (n=filip@c211-30-185-177.artrmn2.nsw.optusnet.com.au) |
| 00:53.04 | Twotone | I've got the 3 external extensions on another link. Let me know when you're ready for it. |
| 00:55.17 | [Outcast] | op i have two line to you sip.conf got check them out. |
| 00:55.46 | [Outcast] | man i can't type to day |
| 00:56.12 | [Outcast] | i have added two lines to you sip.conf add them and make sure asterisk is in the DMZ |
| 00:58.22 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 00:58.45 | Twotone | [Outcast]: Would it show up in the same link on pastebin because I'm not see any differences |
| 00:59.09 | Twotone | oh... followups n/m |
| 00:59.38 | Twotone | [Outcast]: Should any of the external extensions be working without this? |
| 01:01.59 | [Outcast] | they may |
| 01:02.03 | jblack | Heh. I can't park a musiconhold extension. Bugged local config, or * feature, you decide. =) |
| 01:02.43 | Twotone | 2 Of the external extensions work at the moment |
| 01:02.56 | Twotone | I'll make those changes tomorrow when I get into work and let you know if it works |
| 01:03.15 | [Outcast] | send me an email |
| 01:03.40 | Twotone | alright |
| 01:04.34 | Twotone | ty for the help :) |
| 01:05.00 | *** join/#asterisk tasterisk (n=keyser30@rrcs-24-73-85-186.se.biz.rr.com) |
| 01:06.44 | phix | hmmmm |
| 01:06.53 | phix | so no suggestions? |
| 01:09.36 | *** part/#asterisk Docfxit (n=Docfxit@ip-64-32-143-214.lax.megapath.net) |
| 01:12.07 | *** join/#asterisk AndyGraybeal (n=andy@node178.34.251.72.1dial.com) |
| 01:15.16 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a3e294b6bc0d4535) |
| 01:17.16 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 01:18.43 | jblack | Has anyone gotten around to doing traffic shaping on linux-2.6? |
| 01:20.42 | *** join/#asterisk ManxPower (n=manxpowe@69.2.85.41) |
| 01:25.47 | tzanger | uh, what do you mean? |
| 01:25.53 | tzanger | I've been running my tc script for years |
| 01:27.51 | Nugget | I use altq in openbsd. Nothing else comes close. |
| 01:29.09 | fujin | traffic shaping? no, not here |
| 01:29.14 | fujin | we just use diffserv/ToS |
| 01:30.08 | lmadsen | I do QoS by over provisioning my circuits :) |
| 01:30.20 | *** join/#asterisk Victor_Yure (n=Victor_Y@201.9.1.95) |
| 01:31.20 | tzanger | :-) |
| 01:31.31 | AndyGraybeal | i just called my computer from an old phone !!! |
| 01:31.33 | AndyGraybeal | awesome |
| 01:31.38 | AndyGraybeal | so awesmoe |
| 01:32.14 | AndyGraybeal | http://forums.digium.com/viewtopic.php?p=62233&sid=286b4a92783464f99987c7a8bbe3b504 <--- this was the only thing that helped me |
| 01:32.20 | AndyGraybeal | no other site i've found worked |
| 01:32.23 | AndyGraybeal | for me atleast |
| 01:32.25 | lmadsen | I want a red rotary phone for my desk |
| 01:32.31 | AndyGraybeal | lmadsen: that is awesome |
| 01:33.19 | lmadsen | ok, off to the basement locker to grab a bricked Mitel 5220 to see if I can bring it back to life... |
| 01:33.27 | lmadsen | (no pun intended) |
| 01:34.11 | tzanger | I am gonna get that thinkgeek bluetooth retro handset |
| 01:35.29 | *** join/#asterisk ZX81_ (n=ZX81@202.49.106.158) |
| 01:38.52 | JayTee52 | lmadsen, check this out: http://www.sparkfun.com/commerce/product_info.php?products_id=287 |
| 01:39.14 | ManxPower | Those SPA31xx's are miserable to configure |
| 01:41.11 | tzanger | http://www.thinkgeek.com/gadgets/cellphone/8928/ |
| 01:42.04 | lmadsen | JayTee52: holy crap -- I'd never pay $250 for a rotary phone :) |
| 01:42.17 | JayTee52 | me neither but it's a CELL phone |
| 01:42.19 | lmadsen | oh -- it has different parts :) |
| 01:42.27 | lmadsen | I didn't read the desc first :) |
| 01:42.34 | *** join/#asterisk FrigidZephyr (n=FrigidZe@24.96.131.66) |
| 01:42.57 | JayTee52 | but I thought it was interesting because it's 1) rotary and 2) red |
| 01:42.59 | drmessano | HAHAH |
| 01:43.03 | drmessano | Now THAT is a phone |
| 01:44.04 | trippss | so any interesting side effects that can happen if you have host=<someip> and host=dynamic in the same context? |
| 01:44.05 | JayTee52 | I would love a standard analog red rotary phone |
| 01:45.23 | trippss | besides registration issues - would this possibly effect media streams, call quality, etc., in any way? |
| 01:45.34 | tzanger | http://www.thinkgeek.com/gadgets/cellphone/9c9d/ |
| 01:45.45 | tzanger | if I liked big digital watches I'd so get that |
| 01:46.37 | *** join/#asterisk RoyK (n=roy@91.149.11.40) |
| 01:49.44 | JayTee52 | lmadsen, I found another red rotary that's just a standard 500 set which is "new-never used" in cherry red for $125 |
| 01:53.26 | JayTee52 | ooooh! Presidential Deep Red, only $85 bucks |
| 01:53.58 | JayTee52 | that's the one I want! "Get me the Kremlin, ASAP!" |
| 01:56.41 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 01:58.24 | jblack | Ok wondershaper. Hope you help |
| 01:59.48 | *** part/#asterisk mgaal (n=Mike@c-24-5-165-3.hsd1.ca.comcast.net) |
| 02:11.52 | ManxPower | I should install one at a customer |
| 02:13.49 | *** join/#asterisk Abydos313 (n=abydos31@adsl-76-214-25-242.dsl.lsan03.sbcglobal.net) |
| 02:17.57 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:19.34 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:25.03 | lmadsen | tzanger: ping? |
| 02:25.09 | tzanger | yeah |
| 02:26.19 | JayTee52 | lmadsen, red rotary phones on ebay! One is 20 bucks! |
| 02:26.46 | jblack | Yay wondershaper. |
| 02:28.22 | jblack | JayTee52: A couple days ago, we all went on a big search for red batphones. |
| 02:28.49 | JayTee52 | lol |
| 02:29.03 | jblack | we did find them |
| 02:29.04 | JayTee52 | I'm tempted to buy one for my offce |
| 02:29.32 | jblack | Shrug. It's cute, but of limited use. |
| 02:29.37 | JayTee52 | "I'll be right there, Commissioner Gordon!" |
| 02:30.29 | JayTee52 | "Yob tvoyu mat, Vladimir!" |
| 02:32.02 | mmlj4 | "go *** your mother"? |
| 02:32.08 | JayTee52 | when I was at NORAD we had a red phone like that but it didn't have a rotary dial. You just picked it up, it was a true hotline phone. |
| 02:32.23 | JayTee52 | mmlj4, ah, you speak russian? |
| 02:32.37 | mmlj4 | i read books, comrade |
| 02:32.47 | JayTee52 | reading is good! |
| 02:33.20 | JayTee52 | personally I can't think of anything else I'd say to Putin if I had him on the hotline |
| 02:33.26 | mmlj4 | hehe |
| 02:33.29 | nhuisman_work | anyone seen this kind of problem? [root@hilo tftpboot]# /usr/sbin/safe_asterisk: line 62: 22155 Segmentation fault (core dumped) ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} |
| 02:33.37 | nhuisman_work | over and over when a phone is trying to register |
| 02:34.22 | jblack | JayTee52: ayup. I'm planing on getting a few of these next month for doorbells: http://www.scdlink.com/Details.cfm?ProdID=2789&category=23&cf=fr |
| 02:35.33 | JayTee52 | those look cool |
| 02:36.23 | jblack | Before too long, I'll need another spa-8k |
| 02:36.48 | nhuisman_work | has anyone had a phone registering crash asterisK? |
| 02:37.01 | JayTee52 | not me |
| 02:37.43 | *** join/#asterisk ukine (n=Dan@115-29.200-68.tampabay.res.rr.com) |
| 02:37.49 | lmadsen | nhuisman_work: no, but if you have a backtrace you can open a bug |
| 02:38.03 | JayTee52 | nhuisman_work, just one particular phone or any phone? |
| 02:38.10 | nhuisman_work | a cisco using a sccp image |
| 02:38.33 | nhuisman_work | setting up for the first time |
| 02:38.42 | nhuisman_work | maybe I should try some sip soft phones first. |
| 02:39.25 | JayTee52 | yeah, see if SIP works plain vanilla. that message you posted, the tftpboot part is kinda curious |
| 02:39.38 | nhuisman_work | that was just part of my shell |
| 02:40.49 | ukine | speaking of phones :]. for receiving calls behind a router all i need to forward is 5060 UDP to the SIP device or machine with softphone right? |
| 02:41.12 | [TK]D-Fender | ukine, No, read this now : |
| 02:41.13 | [TK]D-Fender | ~sipnat |
| 02:41.14 | 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 |
| 02:41.16 | [TK]D-Fender | ^^^^^^ |
| 02:41.29 | JayTee52 | I've never used Cisco phones, I just use a couple soft phones, 6 Grandstream GT-2000s and a couple HandyTone 286's for fax machines. |
| 02:42.30 | ukine | ty [TK]D-Fender, will do |
| 02:45.33 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 02:47.24 | *** join/#asterisk adjohn (n=adjohn@p1089-ipad76marunouchi.tokyo.ocn.ne.jp) |
| 02:48.39 | adjohn | Hello all, I am having an issue with a very basic asterisk setup. On localhost, a softphone will connect with no problems. But, when I try to connect one externally, the asterisk console appears that the client connected and does its thing, but the client never acknowledges that a connection was made and eventually times out. This happens on several clients that work fine on other servers. Any ideas where I can look? |
| 02:48.53 | adjohn | I also disabled my iptables just to make sure it wasn't a firewall issue. |
| 02:49.36 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 02:49.57 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 02:51.28 | Agrajag- | g'day. my company is looking at depolying asterisk. we have two phone lines and need 3 real phones (will have softphones too though), and i'm trying to figure out what hardware is best to buy. is getting a TDM800P with 3 fxs modules and 2 fxo modules the best/cheapest way to go? |
| 02:51.57 | nhuisman_work | how do I turn the verbosity in the asterisk cli up? |
| 02:52.06 | nhuisman_work | nm |
| 02:52.27 | nhuisman_work | what's the max level? |
| 02:52.43 | adjohn | 9 |
| 02:52.55 | nhuisman_work | it doesn't go up to 11? |
| 02:52.57 | nhuisman_work | ;) |
| 02:53.05 | adjohn | hehe |
| 02:53.09 | ukine | listen to that sustain.. |
| 02:53.29 | nhuisman_work | *waits to watch asterisk dump* |
| 02:55.24 | nhuisman_work | mmm sweet |
| 02:55.52 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 02:56.04 | nhuisman_work | check out that |
| 02:56.05 | nhuisman_work | http://pastebin.com/d422f3919 |
| 02:56.50 | tasterisk | Anyone been able to run Asterisk 1.4 or Amazon EC2? |
| 02:57.05 | tasterisk | Meant on Amazon EC2? |
| 02:57.18 | nhuisman_work | like what, use it as a storage device? |
| 02:58.44 | *** join/#asterisk _ShrikE_ (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:58.54 | adjohn | tasterisk, there is no reason why you couldn't do it, but you would have to pay to keep it up 24/7, which isn't that cheap. |
| 02:58.55 | nhuisman_work | anyone got any ideas on that error in pastebin? |
| 02:59.12 | nhuisman_work | whereas you can get other services which are unlimited and much cheaper |
| 02:59.28 | jblack | If my kid's friends didn't call, I could set Dennis Leary as my hold music. |
| 02:59.35 | jblack | note to self: Put kid up for adoption |
| 02:59.44 | nhuisman_work | jblack |
| 02:59.46 | nhuisman_work | just add a menu |
| 02:59.49 | nhuisman_work | press 2 for the good shit |
| 03:00.21 | jblack | Hmm. I would set different MOH classes for me and her |
| 03:01.13 | [TK]D-Fender | jblack, "well I'm just a regular Joe with a regular job........." ;) |
| 03:01.27 | tasterisk | Thanks adjohn, I'm looking to do it for short periods of time to manage load (if possible). I need a setup for sending outbound calls. |
| 03:01.50 | adjohn | tasterisk, that would be a good option then actually. the startup time is only 1-2 minutes |
| 03:02.04 | nhuisman_work | 3 foot hardon with a cheese burger at the end! |
| 03:02.08 | adjohn | as long as you don't need any hardware with it ;) |
| 03:02.24 | tasterisk | nhuisman_work -- no using the virtual computer services Amazon provides. |
| 03:03.33 | tasterisk | Thanks adjohn, when I get more familar with Asterisk then I'll play around with trying to create an Amazon AMI... I'm a total newbie. |
| 03:03.51 | [TK]D-Fender | jblack, real admins use Slayer or Stryper for MoH ;) |
| 03:04.02 | jblack | I do have Macabre |
| 03:04.12 | jblack | I figured it would be an un-understandable mess. |
| 03:04.13 | [TK]D-Fender | jblack, that'll do in a pinch |
| 03:04.19 | nhuisman_work | oh wait, lol. I bet this phone is on sip and i'm registering it with skinny |
| 03:04.24 | adjohn | tasterisk, me too.. I am having problems even getting an external sip agent to connect. :( |
| 03:04.45 | nhuisman_work | nm it's sccp |
| 03:05.33 | jblack | Hmm. Is there a way to reload moh? |
| 03:06.04 | [TK]D-Fender | jblack, "module reload res_musiconhold.so" |
| 03:06.06 | jblack | I tried "moh reload" and "module reload res_musiconhold.so" |
| 03:06.17 | esaym | what is the default user name in trixbox? My friend installed it via hitting the enter button rapidly and now it is setup and he doens't know the login name |
| 03:06.32 | [TK]D-Fender | esaym, Trixbox is NOT supported here. |
| 03:06.42 | [TK]D-Fender | esaym, Please ask in their support channel |
| 03:08.00 | jblack | darn. |
| 03:08.27 | esaym | thank you, i figured it out |
| 03:08.50 | jblack | I forgot that format_mp3 doesn't care much for id3 tags |
| 03:09.09 | ManxPower | in 1.2 it is reload res_musiconhold.so |
| 03:09.11 | [TK]D-Fender | jblack, correct, no ID3, no VBR |
| 03:09.22 | jblack | Yeah. It's well documented. |
| 03:09.47 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583804.dsl.bell.ca) |
| 03:09.47 | ukine | so trying to use id3'd mp3s won't work? |
| 03:14.26 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:14.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:15.22 | tasterisk | adjohn, I was able to get connected using Broadvoice (making inbound and outbound calls) with a Worldconnect ATA I bought from craigslist. |
| 03:16.07 | jblack | Dead milken is PERFECTG |
| 03:16.45 | adjohn | I tried on a couple different sip soft clients like x-lite, and one of my own I made.. but it is strange because the client acts like it never connects. but the server will process the call |
| 03:17.21 | adjohn | on localhost things are perfect, but when I try to call from a different machine it does that.. |
| 03:17.30 | adjohn | no firewall or anything |
| 03:19.16 | adjohn | i don't know if I missed anything in the config or not, i only made a new account in sip.conf and some demo in extensions.conf.. is there something else i need to edit? |
| 03:19.17 | jblack | Who could "Left handed eskimo midget albino" offend? |
| 03:19.30 | ukine | AHEM. |
| 03:20.24 | ukine | :] |
| 03:20.32 | [TK]D-Fender | adjohn, did I work with you on this previously? |
| 03:20.36 | tasterisk | adjohn I was having a similar issue. Try turning to logging on on x-lite (or I think you can use a sip proxy to get logging information, too). Then have the sip proxy connect your proxy (again I'm new at this, it's just a thought). |
| 03:21.21 | adjohn | [TK]D-Fender, yes but I have tried with other clients as well and the same problem. |
| 03:21.59 | adjohn | tasterisk, will give that a look thanks |
| 03:22.20 | *** join/#asterisk AndyGraybeal (n=andy@node178.34.251.72.1dial.com) |
| 03:23.52 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 03:24.12 | jblack | I know! Warcraft II battle music on loop! |
| 03:24.34 | jblack | Perhaps with some william shatner mixed in |
| 03:25.12 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 03:25.22 | *** part/#asterisk RoyK (n=roy@91.149.11.40) |
| 03:27.18 | jblack | Yeah. Definitely. All William shatner songs, all the time. |
| 03:28.00 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 03:28.06 | adjohn | http://pastebin.ca/858083 is the log output |
| 03:28.30 | drmessano | All. William. Shatner. Songs. |
| 03:28.48 | Nugget | Spock singing about Bilbo Baggins is worse. |
| 03:30.15 | jblack | That's. Right. |
| 03:30.22 | nhuisman_work | you know |
| 03:30.27 | nhuisman_work | there is one william shatner song I like |
| 03:30.29 | nhuisman_work | amazingl |
| 03:30.32 | jblack | No, but Kirk knew. |
| 03:30.34 | nhuisman_work | amazingly. |
| 03:30.52 | ZX81 | man we've had like $3000 of paypal fraud topups today! |
| 03:32.18 | *** join/#asterisk ukine (i=ukine@115-29.200-68.tampabay.res.rr.com) |
| 03:32.43 | Nugget | dang |
| 03:37.00 | ZX81 | ~adn rocks :) |
| 03:37.09 | ZX81 | ~adn |
| 03:37.09 | jbot | from memory, adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 03:37.18 | jblack | JayTee lies. They don't even let you keep crappy quit messages. |
| 03:37.19 | ZX81 | jbot: yep you got it |
| 03:37.37 | ZX81 | jblack, lol |
| 03:39.56 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:40.37 | jblack | Heh. I have Hayseed Dixie. |
| 03:40.43 | jblack | Know who those guys are? |
| 03:41.18 | jblack | They took ACDC's best hits, and redid it in hillbilly bluegrass. |
| 03:41.34 | jblack | It's some of the worst stuff I have ever, ever heard. |
| 03:41.58 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 03:43.29 | nhuisman_work | is it ok to not have a username and password in sip.conf and SIP0XXXXXXXXX.cnf.xml for cisco phones? |
| 03:43.33 | nhuisman_work | or must it use a password |
| 03:44.49 | [TK]D-Fender | nhuisman_work, Sure... if you want to allow completely un-authed calls to do whatever they want on your system.... |
| 03:45.17 | nhuisman_work | k i was just testing stuff and didn't want to have to bother at first. |
| 03:45.26 | nhuisman_work | i'll just throw it in there now for safety |
| 03:45.48 | nhuisman_work | i guess that means someone could steal a mac address and take an extension? |
| 03:46.27 | [TK]D-Fender | nhuisman_work, who cares about MAC? You have no USER ACCOUNTS! |
| 03:46.40 | [TK]D-Fender | nhuisman_work, they can throw calls at you from anywhere |
| 03:46.51 | nhuisman_work | I'm just trying to understand at what level the user accounts get checked |
| 03:47.12 | [TK]D-Fender | nhuisman_work, And of course you wouldn't be able to have * call any of your phones either |
| 03:49.07 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 03:51.54 | nhuisman_work | i don't understand why these phones sit there and say configuring vlan for over a minute |
| 03:51.57 | phix | hey |
| 03:52.29 | phix | to register to more than one SIP provider I just have multiple register => lines in [general]? |
| 03:53.16 | *** join/#asterisk mihinomenest (i=Obqe@66.255.220.17) |
| 03:53.28 | [TK]D-Fender | phix, Well if you use 1 for 1... do the math.. |
| 03:53.56 | mihinomenest | I had this crazy idea today. |
| 03:54.04 | mihinomenest | anyone know where I can find a GSM gateway? |
| 03:54.16 | nhuisman_work | i thought that would be neat too |
| 03:54.17 | [TK]D-Fender | mihinomenest, www.google.com |
| 03:54.17 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 03:54.21 | mihinomenest | (GSM to SIP) |
| 03:54.26 | nhuisman_work | from what I saw there aren't many |
| 03:54.33 | mihinomenest | [TK]D-Fender I tried that. |
| 03:54.48 | nhuisman_work | course i was looking for cell phone -> bluetooth -> asterisk |
| 03:55.06 | mihinomenest | cheapest I found was $159.99. I was hoping that someone might have one that isn't well-known. |
| 03:55.31 | mihinomenest | nhuisman_work: I remember there being one that was cellphone -> usb data cable -> SIP gateway |
| 03:57.21 | mihinomenest | if I can get something like that, then I can buy a $10 pre-paid phone and use that. |
| 03:57.42 | riddlebox | if you configure zaptel.conf and zapata.conf for a fxo card then run ztcfg -vv, it will say fxs_ks is configured right? |
| 03:58.04 | mihinomenest | I think the problem with blootooth is the part where asterisk has to dial a number. |
| 04:00.41 | nhuisman_work | i'd consider running asterisk at home if I could get it to talk to my cell |
| 04:01.49 | Abydos313 | can you have the pbx dial your cell as an extention |
| 04:02.07 | mihinomenest | that'd be the idea. |
| 04:02.25 | mihinomenest | get the GSM gateway connected to the same cell carrier as your cellphone. |
| 04:02.42 | mihinomenest | that way, it's likely free to call your cell from the gateway, or the gateway from your cell. |
| 04:03.55 | phix | [TK]D-Fender: I was confirming |
| 04:04.02 | phix | [TK]D-Fender: that is what i thought too, ok great |
| 04:04.32 | phix | [TK]D-Fender: also, insecure=very == insecure=invite,port ? |
| 04:04.32 | mihinomenest | then, use asterisk to route calls out through a cheap SIP "line", or in via the same SIP connection to your cellphone. |
| 04:06.02 | nhuisman_work | WHEEE the phone finally registered |
| 04:06.07 | nhuisman_work | *dances around* |
| 04:06.19 | nhuisman_work | so much stupid tftp trickery to get cisco phones to work |
| 04:06.43 | [TK]D-Fender | nhuisman_work, Aren't you glad you insist on buying them! :) |
| 04:06.58 | nhuisman_work | wasn't me |
| 04:07.01 | nhuisman_work | these were here before my time |
| 04:07.12 | nhuisman_work | i'm in the process of dumping ccm and replacing it with asterisk |
| 04:07.14 | mihinomenest | I'd perfer a cisco to my grandstreams. |
| 04:07.16 | nhuisman_work | woot :) |
| 04:07.39 | nhuisman_work | i think it's just that the ciscos are a little bit of a pain to provision and setup. |
| 04:07.43 | nhuisman_work | from what i've read |
| 04:08.32 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-2767bc6d55090089) |
| 04:08.44 | riddlebox | I have configured a tdm card with a fxo card, when I run ztcfg -vv it reports 1 fxs configured, and I have my dialplan configured and it doesnt see a call come in or when I try to call out it says channel type registered for Zap? |
| 04:12.47 | drmessano | Wow |
| 04:13.06 | drmessano | Chris Lymans slam of Digium is interesting |
| 04:13.52 | outtolunc | eh? url? |
| 04:13.52 | tzanger | ? |
| 04:14.49 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 04:15.18 | drmessano | hang on |
| 04:15.25 | drmessano | http://www.trixbox.com/about-us/blog/open-source-closed-minds |
| 04:15.35 | drmessano | Biggest pile of self service BS I have seen from Fonality |
| 04:15.41 | AndyGraybeal | hahah.. i got it to call the phone connected to the fxs!! |
| 04:15.43 | AndyGraybeal | <-- god |
| 04:15.44 | riddlebox | what do you guys think of asterisknow? |
| 04:16.13 | [TK]D-Fender | riddlebox, bleh |
| 04:16.33 | *** join/#asterisk Abydos313 (n=abydos31@adsl-76-214-25-242.dsl.lsan03.sbcglobal.net) |
| 04:16.40 | riddlebox | [TK]D-Fender, I like to edit conf files.. |
| 04:17.02 | [TK]D-Fender | riddlebox, pastebin everything for your TDM problem. |
| 04:17.27 | [TK]D-Fender | riddlebox, whats the point of asking about stuff you aren't showing us? |
| 04:17.45 | drmessano | Hes a "riddle", duh |
| 04:17.53 | [TK]D-Fender | riddlebox, and ztcfg and your dialplan don't mean squat if zapata isn't right. |
| 04:18.17 | tzanger | holy fuck |
| 04:18.31 | [TK]D-Fender | riddlebox, and I'm betting your statement of "channel type registered for Zap" isn't quite what the message said |
| 04:18.31 | riddlebox | [TK]D-Fender, I have followed the ATFOT book |
| 04:18.35 | tzanger | the man can string more words together and still have no meaningful thought than anyone I know |
| 04:18.41 | drmessano | lol |
| 04:18.50 | [TK]D-Fender | riddlebox, thanks... more non-info. PASTEBIN <--- |
| 04:18.54 | tzanger | "wah, digium bought my competitor not me" crap |
| 04:19.12 | drmessano | He can also justify his actions based on his opinions of other peoples motives more than anyone I know |
| 04:19.24 | riddlebox | [TK]D-Fender, I am trying to get it all together |
| 04:19.39 | [TK]D-Fender | tzanger, who? |
| 04:19.41 | tzanger | I don't know, I fail to see how ABE is a problem |
| 04:19.45 | tzanger | http://www.trixbox.com/about-us/blog/open-source-closed-minds |
| 04:19.48 | tzanger | from drmessano |
| 04:21.30 | drmessano | Fact is, people are passionate about ****box and all he has done since buying it has been to bastardize and attempt to monetize it.. the community has spoken out, and hes whining about it.. more and more |
| 04:21.41 | *** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com) |
| 04:22.27 | drmessano | Digium can sell any incarnation of asterisk because digium never said "We will never attempt to monetize Asterisk.. it and every variation will be free!!!" |
| 04:22.29 | neoalex | hi, how can I make asterisk hangup when the other caller hangs up? |
| 04:22.58 | [TK]D-Fender | neoalex, thats pretty much automatic. Details would be helpful... |
| 04:23.03 | drmessano | Effing hippocrit |
| 04:23.38 | neoalex | pastebining what happens right now |
| 04:24.29 | riddlebox | [TK]D-Fender, http://pastebin.ca/858141 |
| 04:24.34 | neoalex | here's what happens: http://pastebin.com/m263bf2dd |
| 04:24.53 | neoalex | [TK]D-Fender: ^^^ |
| 04:25.29 | [TK]D-Fender | riddlebox, Jan 16 22:19:39] WARNING[5398]: channel.c:3281 ast_request: No channel type registered for 'Zap' |
| 04:25.37 | neoalex | now afer that the problem is 2100 the extension which answered the call is still on |
| 04:25.48 | [TK]D-Fender | riddlebox, looking like you compiled Zaptel AFTER Asterisk and chan_zap was never built |
| 04:26.04 | [TK]D-Fender | riddlebox, go try "load chan_zap.so" |
| 04:26.20 | riddlebox | [TK]D-Fender, I could have swore I compiled zaptel first before I did anything else |
| 04:26.29 | [TK]D-Fender | riddlebox, because * will not have built support if you didn't do zaptel first |
| 04:26.51 | [TK]D-Fender | riddlebox, keep swearing... it identifies the quality of your character immediately :) |
| 04:27.11 | drmessano | Time for the DR to put his two cents in |
| 04:27.16 | riddlebox | [TK]D-Fender, hrmm I says it could not be loaded |
| 04:27.39 | [TK]D-Fender | riddlebox, because its not there. No go flush your * source folder, re-extract and recompile. |
| 04:28.00 | riddlebox | alrighty I will do it |
| 04:28.39 | neoalex | so... any thoughts on this: http://pastebin.com/m263bf2dd |
| 04:29.00 | neoalex | line stays open after caller hung up |
| 04:29.15 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 04:30.05 | riddlebox | [TK]D-Fender, do you have any other issues with the way the system is configured? |
| 04:30.11 | [TK]D-Fender | riddlebox, and a freebie : "callerid = 6182246161" should be "callerid=asreceived" |
| 04:30.29 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:30.57 | riddlebox | [TK]D-Fender, ok, I wil change it |
| 04:31.20 | [TK]D-Fender | riddlebox, exten => _1NXXNXXXXXX,2,Congestion() <- yuck. I advise you do that for 5s then hangup. that'll jsut sit around |
| 04:31.50 | riddlebox | [TK]D-Fender, that came from broadvoice's webpage on how to setup asterisk to work with broadvoice |
| 04:32.05 | [TK]D-Fender | riddlebox, And of course you should make an abstract Zaptel dial macro and shick up the redundancy in your outbound context |
| 04:32.19 | nhuisman_work | [TK]D-Fender, do you have any idea why a phone registering would crash * |
| 04:32.23 | [TK]D-Fender | riddlebox, Funny... there isn't a single BV related bit in there.. |
| 04:32.28 | riddlebox | ok, I will read more on macros |
| 04:32.57 | nhuisman_work | or maybe better, what logs to look at to find out. |
| 04:32.58 | riddlebox | [TK]D-Fender, just my dialing stuff, I used BV until I moved here, then I went with a pots line |
| 04:32.59 | [TK]D-Fender | riddlebox, and BV is not a standard for configuring * more than what should be done to connect to their service, and prep for incoming.outgoing calls. The rest of your system is yours. |
| 04:33.27 | [TK]D-Fender | nhuisman_work, Using chan_skinny, right? |
| 04:33.35 | nhuisman_work | yessir |
| 04:33.41 | [TK]D-Fender | nhuisman_work, theres the reason. |
| 04:33.48 | [TK]D-Fender | nhuisman_work, always been unstable. |
| 04:34.12 | nhuisman_work | any way to find out what I have wrong with my options that could be crashing it? |
| 04:34.34 | Agrajag- | g'day. my company is looking at depolying asterisk. we have two phone lines and need 3 real phones (will have softphones too though), and i'm trying to figure out what hardware is best to buy. is getting a TDM800P with 3 fxs modules and 2 fxo modules the best/cheapest way to go? |
| 04:36.27 | MrTelephone | try a sangoma car |
| 04:36.27 | MrTelephone | d |
| 04:36.28 | mosty | Agrajag-, FXS modules on pci cards suck, i recommend either a sip phone, or some ATA's |
| 04:36.50 | Agrajag- | why do they suck? |
| 04:36.54 | mosty | and go for a sangoma card with one fxo module (it has two ports) |
| 04:37.02 | [TK]D-Fender | nhuisman_work, not a clue... |
| 04:37.10 | nhuisman_work | [TK]D-Fender, k thanks. |
| 04:37.31 | neoalex | [TK]D-Fender: any idea on why my calls don't get hung up when they should? |
| 04:37.38 | neoalex | here's the pastebin link again: http://pastebin.com/m263bf2dd |
| 04:37.50 | MrTelephone | neoalex, packet loss? |
| 04:37.51 | mosty | Agrajag-, fxs ports with zaptel are just more work to setup and keep working within acceptable limits for echo in my experience |
| 04:38.02 | Agrajag- | oh ok |
| 04:38.20 | neoalex | not a chance... there's about 3 feet of cable between the asterisk and 2100 |
| 04:38.29 | [TK]D-Fender | neoalex, you showed me as little as humanly possible. You should include sip debug, I should see the DIAL that originated the calls, and the sip.conf entries |
| 04:38.48 | MrTelephone | neo, it says extied on non-zero.. that mean its hung up? |
| 04:39.13 | neoalex | yes... that's what it says when I hagup |
| 04:39.22 | [TK]D-Fender | Agrajag-, I'm with mosty on this 100% |
| 04:39.23 | neoalex | should mention 2100 is a PAP2T |
| 04:39.30 | MrTelephone | fender i get a kick how you spent an extra 3 seconds typing "as humanly" |
| 04:39.32 | MrTelephone | hahah |
| 04:39.58 | neoalex | let me get more details though |
| 04:40.03 | [TK]D-Fender | MrTelephone, Sorry.. I know a lot of better trained monkeys.... |
| 04:40.09 | nhuisman_work | any idea what this error is : handle_request_invite: Failed to authenticate user "1000" <sip:@xxx.xxx.xxx.xxx..... |
| 04:40.16 | Agrajag- | [TK]D-Fender: ok - i'm looking at http://www.sangoma.com/main/products/hardware/cards and can't really see which card i'd need? |
| 04:40.27 | nhuisman_work | i saw something about adding insecure=very in sip.conf to fix it |
| 04:40.37 | MrTelephone | nhuisman, add the user 1000 to sip.conf |
| 04:40.44 | [TK]D-Fender | nhuisman_work, means first you want to send un-auhed calls, then you fail to set up auto properly. PASTEBIN is your friend <- |
| 04:40.56 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 04:41.18 | [TK]D-Fender | nhuisman_work, insecure=bad. Your config=bad. Now lets look at your setup and see what we can do... |
| 04:41.24 | MrTelephone | nhuisman_work, try adding [1000] username=1000 host=xxx.xxx.xxx.xxx insecure=very |
| 04:41.40 | [TK]D-Fender | nhuisman_work, skip that, and show both parts of what youve got NOW. |
| 04:41.48 | nhuisman_work | yeah kk, sec. |
| 04:41.51 | MrTelephone | uninstall asterisk an goto bed <--- my new favorite solution |
| 04:42.02 | neoalex | [TK]D-Fender: do you also need the sip.conf settings for the extension? |
| 04:42.06 | neoalex | [2100] |
| 04:42.19 | [TK]D-Fender | MrTelephone, And don't forget to pick up your parking validation pass on the way out! ;) |
| 04:42.40 | [TK]D-Fender | neoalex, Didn't I just say that explicitly? |
| 04:42.57 | *** join/#asterisk RoyK (n=roy@91.149.11.40) |
| 04:43.10 | MrTelephone | have you watched that parking wars show yet? |
| 04:43.22 | mosty | Agrajag-, a200d |
| 04:43.38 | Agrajag- | mosty: cheers |
| 04:44.52 | *** join/#asterisk AndyGraybeal_ (n=andy@node178.34.251.72.1dial.com) |
| 04:46.14 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 04:46.59 | Agrajag- | mosty: so you suggested that with 2 fxo ports - would it be cheaper to buy sip phones or some kind of device which has (at least) 3 fxs ports? |
| 04:47.07 | MrTelephone | i had to remove nonce checking for my clients |
| 04:47.18 | *** part/#asterisk RoyK (n=roy@91.149.11.40) |
| 04:47.50 | mosty | Agrajag-, fxo ports are for connecting to analogue phone lines. you can't replace that with a sip phone |
| 04:49.19 | neoalex | [TK]D-Fender: here is all the info you requested: http://pastebin.com/m2ebe106f |
| 04:49.38 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 04:49.42 | mosty | Agrajag-, my preference for phones is sip phones, you get more features than from an ATA. however 2 port ATA's are fairly cheap if you want to use analogue phones |
| 04:50.24 | [TK]D-Fender | neoalex, "r" in dial = BAD. Bypasses standard call-progress |
| 04:53.41 | neoalex | [TK]D-Fender: still does the same thing even without the r |
| 04:54.05 | [TK]D-Fender | neoalex, so you hangup the PAP2 and the caller stays around? |
| 04:54.36 | neoalex | no... the other way arround... hung up the caller and the pap2 stays on |
| 04:55.39 | [TK]D-Fender | neoalex, what do you see on "show channels"? |
| 04:55.45 | [hC] | fender |
| 04:55.53 | [hC] | have you seen the polycom 3.0 release notes yet? |
| 04:56.00 | [hC] | <- has a huge boner for the new software release |
| 04:56.02 | [TK]D-Fender | [hC], nope... cheking now |
| 04:56.06 | neoalex | [TK]D-Fender: during the call, right? |
| 04:56.08 | [hC] | LDAP support baby. |
| 04:56.17 | [hC] | And hook switch headset support, so no lifters needed anymore. |
| 04:56.18 | [TK]D-Fender | [hC], yummeh |
| 04:56.45 | [TK]D-Fender | [hC], answer triggered how? |
| 04:57.25 | neoalex | ServerAlex*CLI> sip show channels |
| 04:57.26 | neoalex | Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message |
| 04:57.28 | neoalex | 192.168.1.5 2100 788e3f68661 00102/00000 ulaw No Tx: ACK |
| 04:57.30 | neoalex | 216.246.73.186 1917512497 2f60bdeb2ee 00101/00102 ulaw No Rx: ACK |
| 04:57.31 | neoalex | 2 active SIP channels |
| 04:57.46 | [hC] | [TK]D-Fender: Jabra has a protocol spec apparently for taking the phone offhook with a cable from the headset base to the phone |
| 04:57.54 | [hC] | they added VQMon too |
| 04:58.12 | [hC] | it integrates with active directory out of the box too |
| 04:58.22 | [TK]D-Fender | neoalex, I said "show channels" not "sip show channels" |
| 04:58.30 | *** join/#asterisk DrVince_ (n=x@modemcable082.136-56-74.mc.videotron.ca) |
| 04:58.34 | DrVince_ | Hi |
| 04:58.58 | mosty | [hC], LDAP eh? nice |
| 04:59.30 | neoalex | ServerAlex*CLI> core show channels |
| 04:59.32 | neoalex | Channel Location State Application(Data) |
| 04:59.33 | neoalex | SIP/2100-0078bc30 (None) Up Bridged Call(SIP/19175124979-0 |
| 04:59.35 | neoalex | SIP/19175124979-0078 8100@incoming:2 Up Dial(SIP/alex&SIP/2100&SIP/Ale |
| 04:59.36 | neoalex | 2 active channels |
| 04:59.38 | neoalex | 1 active call |
| 05:00.03 | mosty | neoalex, it's probably better if you use pastebin.com instead of pasting here |
| 05:00.08 | [TK]D-Fender | neoalex, please don't spam it. I needed half of that tops. |
| 05:00.37 | neoalex | noted |
| 05:00.45 | [TK]D-Fender | neoalex, doesn't look like your phone has hung up\ |
| 05:00.50 | *** part/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 05:01.03 | DrVince_ | I'm running a fresh install on ubuntu from ports and when I try to start asterisk with "asterisk -vvvgc" it core dumps after "res_config_odbc.so => (ODBC Configuration)". Where could be the problem? |
| 05:01.07 | neoalex | this is during the call |
| 05:01.10 | AndyGraybeal_ | is there a better softphone for linuxen than xlite? |
| 05:01.16 | neoalex | after hungup they are all 0 |
| 05:01.28 | [TK]D-Fender | neoalex, "show channels concise" please. |
| 05:01.39 | [TK]D-Fender | AndyGraybeal_, Ekiga |
| 05:01.40 | mosty | AndyGraybeal_, twinkle? |
| 05:01.43 | neoalex | 0 active channels |
| 05:01.45 | neoalex | 0 active calls |
| 05:01.50 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 05:01.56 | [TK]D-Fender | neoalex, go check the other now |
| 05:01.57 | AndyGraybeal_ | awesome thank you [TK]D-Fender and mosty |
| 05:02.05 | [TK]D-Fender | neoalex, these should not be in disagreement |
| 05:02.23 | neoalex | concise doesn't show anything |
| 05:02.57 | [TK]D-Fender | neoalex, pastebin the CLI attempts of each consecutively. |
| 05:03.24 | neoalex | both during a call and after the call? |
| 05:03.31 | [TK]D-Fender | neoalex, sure. |
| 05:03.37 | neoalex | k |
| 05:03.46 | [hC] | [TK]D-Fender: I only found the sip 3.0 firmware when clicking a link in the downloads area called 'sip downloads matrix' |
| 05:03.52 | [hC] | it references bootrom 4.1.0 but i cant find it yet. |
| 05:04.05 | [TK]D-Fender | [hC], I should have them tomorrow. |
| 05:04.08 | DrVince_ | "res_odbc.c:511 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified" What is that about? |
| 05:04.50 | [TK]D-Fender | [hC], I jsut picked up the admin guide, but didn't find the RN yet |
| 05:05.15 | DrVince_ | Doesn't ODBC require no external aid |
| 05:05.20 | [TK]D-Fender | DrVince_, means you don't have ODBC setup properly and are missing a DSN |
| 05:05.23 | [hC] | [TK]D-Fender: http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_sip_rel_3_0_0.pdf |
| 05:06.26 | neoalex | [TK]D-Fender: http://pastebin.com/mf63df8c |
| 05:07.51 | [TK]D-Fender | [hC], CRAP.... licensing. |
| 05:08.48 | [TK]D-Fender | neoalex, well the last looks like the calls cleared fine |
| 05:09.17 | neoalex | still the analog phone which is connected to 2100 (2100 being a PAP2T-NA ATA) stays on |
| 05:10.27 | [hC] | [TK]D-Fender: the partner license stuff? |
| 05:10.37 | [hC] | [TK]D-Fender: I get the impression that its just an agreement, not a charge. |
| 05:10.40 | jblack | Well, this I didn't expect. Very few things actually encode usably to 8k gsm. |
| 05:11.33 | [TK]D-Fender | neoalex, what do you mean "on"? |
| 05:11.50 | [TK]D-Fender | neoalex, the PAP can't tell your phone to hang up... its an ata and jsut supplies battery |
| 05:11.53 | jblack | Do you mean that the phone doesn't automatically hang up? |
| 05:12.09 | jblack | [TK]D-Fender: I imagine he's used to cell phones. ;) |
| 05:12.14 | neoalex | really... shouldn' |
| 05:12.28 | neoalex | t it cut the power on the line for example? |
| 05:12.44 | [TK]D-Fender | neoalex, No. So everything fine and you are just oblivious to the world of analog |
| 05:13.06 | jblack | neoalex: It's probably been awile since you've used a traditional phone? It's typical for desk phones to not hang up until they are actaully hung back up. |
| 05:13.06 | [TK]D-Fender | neoalex, grab a phoen anywhere and sit around... do you see the telco "cutting you off"? |
| 05:13.41 | neoalex | you're right it has been a while since I've used a traditional phone |
| 05:13.53 | *** join/#asterisk tuxd00d (n=Tuxd00d@128.187.129.147) |
| 05:14.42 | jblack | Don't feel bad. The exact same thing happened to me for 2 days. =) |
| 05:14.49 | neoalex | I guess that's why a traditional phone can be "off the hook" |
| 05:14.56 | jblack | yup. exactly |
| 05:15.48 | neoalex | ok, well thank you, and sorry for wasting your time [TK]D-Fender |
| 05:15.56 | nhuisman_work | [TK]D-Fender, do the usernames and passwords of sip.conf and the SIPXXXXXXXXXXXX.cnf have some rules? ie no extended chars, only numbers? |
| 05:16.13 | nhuisman_work | I changed the username to the extension # and changed the pass to not have a $ in it and now the phone registers |
| 05:16.21 | [TK]D-Fender | nhuisman_work, you telling me that you've been getting "creative" while trying to get things to "work"? |
| 05:16.54 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:16.58 | nhuisman_work | no, I'm saying I had username of "abcdabcd" and a password of "blah$blah" and that didn't work |
| 05:17.05 | nhuisman_work | and when I went to 1000 for username and testpass |
| 05:17.06 | [TK]D-Fender | nhuisman_work, I'll take that as a "yes" |
| 05:17.06 | nhuisman_work | it works |
| 05:17.23 | [hC] | Hmm... where the heck do i get bootrom 4.1.0 |
| 05:17.27 | [hC] | the 4.0 download link isnt even working. |
| 05:17.28 | DrVince_ | Which file is the ODBC setup? |
| 05:17.36 | nhuisman_work | username has to equal extension? |
| 05:17.46 | [TK]D-Fender | DrVince_, : |
| 05:17.48 | [TK]D-Fender | ~book |
| 05:17.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 05:18.04 | [TK]D-Fender | nhuisman_work, extension has nothing to do with getting your phone to register |
| 05:18.10 | *** join/#asterisk andrewn (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 05:18.48 | nhuisman_work | yes I know. I'm just wondering if you know why it wouldn't register and then with a more simple username and pass it did. |
| 05:20.36 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 05:20.39 | [TK]D-Fender | nhuisman_work, well... I never saw anything so I can't tell if they matched, and can't comment on what chars might be illegal. |
| 05:21.18 | nhuisman_work | the username was voiphilo, and the password was voip$pass |
| 05:21.21 | nhuisman_work | maybe the $ did it. |
| 05:21.44 | [TK]D-Fender | nhuisman_work, Too late, and little bits at a time don't tell me much.. |
| 05:21.52 | [TK]D-Fender | nhuisman_work, but glad its working now at least |
| 05:22.06 | nhuisman_work | yeah me too, i think I'll change the pass to have a $ in it just to find out. |
| 05:22.30 | mosty | [hC], bootrom 4.0.0 is there |
| 05:22.38 | [TK]D-Fender | mosty, thats not the new one |
| 05:22.44 | [TK]D-Fender | most 4.1.0 <_ |
| 05:23.18 | mosty | i thought the admin guide mentioned 4.0.1 as a minimum version for some model, maybe i misread |
| 05:24.02 | *** join/#asterisk Sargun (n=Sargun@atarack/staff/sargun) |
| 05:24.22 | Sargun | Does anyone know of any B2B SIP/IAX2 Providers which send ANI information? |
| 05:25.15 | jblack | I don't. |
| 05:25.22 | [TK]D-Fender | wow.. SIP 3.0 admin guide implies support for GBE |
| 05:25.52 | mosty | sargun: any major telco should if you're legally allowed i think |
| 05:25.58 | jblack | GBE? Gigabit ethernet? |
| 05:26.42 | [TK]D-Fender | jblack, yup |
| 05:27.07 | [TK]D-Fender | DAMN, they still haven't restricted directory entries to specific registrations. DUMB |
| 05:27.11 | Sargun | mosty, Most major telcos do it over ISDN/T1+PRI |
| 05:27.13 | jblack | Bulky protocols? |
| 05:27.25 | Sargun | mosty, Currently I'm dealing with Teliax. |
| 05:28.02 | nhuisman_work | [TK]D-Fender, for your information $ are not allowed in passwords |
| 05:28.04 | mosty | Sargun, oh you want a telco that will give you voip? |
| 05:28.06 | nhuisman_work | registration fails |
| 05:28.12 | mosty | major telco |
| 05:28.19 | Sargun | mosty, yeah |
| 05:28.20 | [TK]D-Fender | nhuisman_work, not allowed by which side <- |
| 05:28.25 | Sargun | jblack, I know. |
| 05:28.32 | nhuisman_work | in sip.conf and SIPXXXXXXXXX.cnf |
| 05:28.38 | Sargun | jblack, I tried another one, uh, astricon |
| 05:28.38 | Sargun | no. |
| 05:28.40 | [TK]D-Fender | nhuisman_work, this is where you should show your configs and SIP debug to back it |
| 05:28.44 | Sargun | ugh, 800 number provider |
| 05:28.56 | [TK]D-Fender | nhuisman_work, Doesn't say who can't handle it... is it * or Cisco <- |
| 05:28.56 | nhuisman_work | k, how do I get sip debug? |
| 05:28.59 | jblack | Yeah, thus me saying I don't know of any IAX2/SIP providers that provide ANI. |
| 05:29.19 | [TK]D-Fender | nhuisman_work, have a soft phone reg to that account with a $ |
| 05:29.20 | Sargun | jblack, mosty, My Telco provider is AT&T/SBC/"The New AT&T", and their cost for ISDN is not nice. |
| 05:29.42 | nhuisman_work | k, i'll do some more testing. Gotta go eat. Thanks for your help [TK]D-Fender |
| 05:29.45 | Sargun | $36.99*2 for a vanilla ISDN + Dialtone is more. |
| 05:29.54 | nhuisman_work | I'll let you konw. |
| 05:29.55 | nhuisman_work | know |
| 05:29.56 | jblack | Since when does AT&T, Verizon, Pacific Bell, Sprint, MCI, ... any major phone company provide ANI over IAX2/SIP? |
| 05:29.58 | Sargun | and if you want PRI you need to double that. |
| 05:30.08 | Sargun | jblack, No, I'm not saying over IAX2/SIP |
| 05:30.15 | Sargun | jblack, uh, Level3 does. |
| 05:30.26 | Sargun | But AT&T's policies suck |
| 05:30.37 | jblack | IAX2/SIP was part of his qualification. ;) |
| 05:31.03 | Sargun | Unless you want to bring me over copper from like 500 miles (Qwest) |
| 05:31.34 | mosty | Sargun, why do you need ANI, where you can't afford PRI? |
| 05:32.01 | Sargun | mosty, Uh, because I need it for 10 lines minimum simultaneously... |
| 05:32.36 | Sargun | Which gets expensive with AT&T |
| 05:32.44 | Sargun | especially for what I'm doing. |
| 05:32.47 | jblack | Split a T1. |
| 05:33.06 | Sargun | jblack, that's $500 + Termination/origination. |
| 05:33.21 | Sargun | I wish Phone systems were as cheap/simple as Layer 3 networking. :-/ |
| 05:33.24 | jblack | Yeah, that sounds about right. |
| 05:33.41 | Sargun | hm. |
| 05:33.45 | Sargun | Which is more. |
| 05:33.48 | *** join/#asterisk dalbaech (n=Tassach@c-98-201-18-122.hsd1.tx.comcast.net) |
| 05:33.51 | Sargun | because AT&T has a large minimum |
| 05:34.11 | jblack | You may have multiple local carriers. |
| 05:34.15 | dalbaech | Hey, does anyone know of a provider that actually captures the ANI of calls and susbstitutes it in the place of caller id or adds it in the headers? |
| 05:34.17 | jblack | There are two here. |
| 05:35.08 | jblack | dalbaech: Simply speaking, not cheaply, no. |
| 05:35.46 | DrVince_ | It turns out it was one of the extra pkg that was causing harm. I didn't had to config ODBC. |
| 05:35.51 | dalbaech | well, it doesn't matter if it's an toll free provider or not (as that requires ANI for the call to be billed properly) |
| 05:36.11 | dalbaech | I don't really care about "cheaply"... |
| 05:36.28 | jblack | dalbaech: Ok, then look at getting ISDN or a partial T1. |
| 05:36.36 | dalbaech | That requires ANI to be sent? |
| 05:37.04 | dalbaech | or they always send it? |
| 05:38.19 | dalbaech | n/m |
| 05:38.21 | dalbaech | *reading* |
| 05:38.26 | jblack | as I understand things, to get ANI, you'll need to get something equivilant. a T1 or PRI, possibly isdn. there aren't any commonly known SIP/IAX2 providers that supply ANI that I'm aware of. |
| 05:42.47 | *** join/#asterisk cell76 (n=a@c-75-70-151-4.hsd1.co.comcast.net) |
| 05:52.51 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 05:52.56 | [hC] | ooh, hints in the polycom config file that a color phone is coming: |
| 05:52.57 | [hC] | bitmaps bg.hiRes.color.bm.1.name="Leaf.bmp" |
| 05:53.31 | [TK]D-Fender | [hC], thats something I guess. |
| 05:53.53 | [TK]D-Fender | [hC], GB, color, LDAP... things are moving forwards.... but still poorly handling contacts. |
| 05:54.07 | [hC] | [TK]D-Fender: in what way? |
| 05:54.14 | [hC] | in the -directory.xml files? |
| 05:54.39 | [TK]D-Fender | [hC], can't target them to a specific reg, same with presence aspect, can't do in-call DTFM... all the cool stuff Aastra lets you do. |
| 05:54.55 | [TK]D-Fender | [hC], incomplete presence indications |
| 05:55.38 | [hC] | [TK]D-Fender: nod. by incomplete you mean only showing free/busy, not DND, offhook, etc? |
| 05:57.35 | *** join/#asterisk AndyGraybeal (n=andy@node113.34.251.72.1dial.com) |
| 05:58.03 | [TK]D-Fender | [hC], ringing" is the real diff. off-hook isn't a real state, DND is more like how your phone will hande a call, not an indication per-se |
| 05:58.13 | [TK]D-Fender | [hC], but all the rest stands |
| 05:58.21 | [TK]D-Fender | [hC], and its really not hard at all. |
| 05:58.47 | [hC] | [TK]D-Fender: yeah aastra just shows availalbe/not available/ringing |
| 05:59.02 | [TK]D-Fender | [hC], which is fine. |
| 05:59.25 | [TK]D-Fender | [hC], ringing is key to making directed pickup more functional. |
| 06:00.12 | [TK]D-Fender | [hC], state-based multi-presence/speed-dials... is that so much to ask? |
| 06:00.14 | [hC] | [TK]D-Fender: indeed. I'm still trying to get around a problem at one customer site where when using side cars with 20+ hints, the phone will slow down/reboot once a week or so for no reason. |
| 06:00.21 | [hC] | [TK]D-Fender: good luck getting polycom's help with it though... heh |
| 06:00.42 | [TK]D-Fender | [hC], never had a lock like that on mine... I have 2.5 loaded sidecars |
| 06:01.13 | [TK]D-Fender | [hC], At least.... not with anyones awareness :) |
| 06:01.16 | [hC] | [TK]D-Fender: i have a 1.6.7 phone that doesnt screw up, then this one that does has gone from 1.6.8->2.0.2 |
| 06:01.35 | [TK]D-Fender | [hC], its probably wisened up and is doing it behind my back :) |
| 06:01.52 | [TK]D-Fender | [hC], I'm on 2.1.2 at the office now. |
| 06:02.32 | [TK]D-Fender | [hC], didn't bother with 2.2.0 and wills ee about 3.0.0. My office is GUI'd and I'm going to see about making sure my vendor will support 3.0.0 in all its glory. |
| 06:02.59 | [hC] | [TK]D-Fender: im going to try to upgrade the problematic set to 3.0.0 and see if it goes away, if it doesnt, ill see about going down to 1.6.7 |
| 06:03.08 | [TK]D-Fender | [hC], after that I'll divest myself of hand-configuring them |
| 06:03.17 | [hC] | [TK]D-Fender: who does your GUI? |
| 06:03.26 | [TK]D-Fender | [hC], I'd say baby-step it to 2.2.0 first |
| 06:03.31 | [TK]D-Fender | [hC], ScopServ |
| 06:03.39 | [hC] | [TK]D-Fender: how do you like it? |
| 06:03.47 | [TK]D-Fender | [hC], I was one of their North American pilot customers. |
| 06:03.58 | [TK]D-Fender | [hC], as GUI's go its really good. |
| 06:04.07 | *** join/#asterisk Prayer (n=Administ@c-71-225-221-149.hsd1.pa.comcast.net) |
| 06:04.42 | *** join/#asterisk Patrickz_ (n=patrickz@61-90-163-149.static.asianet.co.th) |
| 06:04.50 | [TK]D-Fender | [hC], I helped a bit with their Polycom handling, and would like to collaborate some more for 3.0+ and multi-lingual support for prompts etc |
| 06:05.04 | Patrickz_ | hello all |
| 06:05.15 | AndyGraybeal | downloading svn asterisk from trunk so i can use russellb's asterisk-jack :) |
| 06:05.23 | [hC] | [TK]D-Fender: is it in the style of trixbox where you install it just on a single box and thats that, or does it have a distributed control mechanism for loading up multiple servers? |
| 06:05.58 | [TK]D-Fender | [hC], Not sure really... Feels single-box-y |
| 06:06.06 | [TK]D-Fender | [hC], is big-league through |
| 06:06.58 | mosty | 2.2.0 takes a lot longer to boot than 2.0.0 for me |
| 06:07.38 | [TK]D-Fender | mosty, make sure to use the segmented LD's and not the composite and you will see it start faster than either. |
| 06:10.29 | [TK]D-Fender | mosty, my IP 501 with single takes 1:40 with the full image. split is faster. |
| 06:11.00 | mosty | i should time it |
| 06:11.45 | [TK]D-Fender | ok, well its bed time. back tomorrow... |
| 06:11.46 | [TK]D-Fender | later all |
| 06:13.17 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 06:15.18 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 06:15.57 | Sargun | uh |
| 06:24.10 | Sargun | jblack, it isn't normal for SIP providers to do ANI |
| 06:24.24 | Sargun | jblack, Teliax buys from L3 right? |
| 06:25.39 | Sargun | jblack, that's true. |
| 06:26.45 | Sargun | jblack, did you request Teliax to do it? |
| 06:29.21 | AndyGraybeal | is anyone familiar with a sip phone that supports JACK? |
| 06:36.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:38.54 | mosty | AndyGraybeal, what are you trying to do? |
| 06:42.05 | AndyGraybeal | mosty, well... i run jack, so i was hoping there was a softphone that supported jack |
| 06:42.16 | [hC] | Is there a way to reboot a polycom phone from the web interface? |
| 06:42.18 | AndyGraybeal | but maybe i don't know what i'm talking about and all i need is ALSA |
| 06:42.36 | [hC] | or by sending a particular packet to it? (aside from SIP NOTIFY) |
| 06:42.45 | mosty | AndyGraybeal, i have a feeling jack has an alsa interface |
| 06:43.05 | AndyGraybeal | it does |
| 06:43.13 | mosty | [hC], just hit any submit button without changing options, it should reboot |
| 06:43.21 | [hC] | ok. |
| 06:43.26 | [hC] | i'll give that a shot. thanks! |
| 06:43.56 | mosty | AndyGraybeal, then the phone shouldn't need to support JACK directly |
| 06:45.07 | AndyGraybeal | mosty: then i don't understand how to hear a phone that tries to connct to ALSA....... i understand it if they connect to JACK though |
| 06:46.18 | mosty | you lost me there. what? |
| 06:47.00 | AndyGraybeal | yea, me too sorry, i don't know how to get the phone to talk to alsa or anything; i just know jack |
| 06:47.17 | mosty | which softphone? |
| 06:47.30 | AndyGraybeal | twinkle for now |
| 06:47.58 | AndyGraybeal | it looks like ekiga might have a jack interface, but it's 13MB and i'm on dialup, plus i'm downloading the svn asterisk trunk |
| 06:48.41 | mosty | so in the audio setup for twinkle you see no audio devices? |
| 06:48.51 | AndyGraybeal | i do, hold on i just crashed twinkle |
| 06:49.26 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 06:49.40 | AndyGraybeal | thank you for t he help so far mosty |
| 06:50.56 | mosty | afaik, JACK is a server that makes use of alsa/oss/whatever sound drivers |
| 06:51.07 | mosty | so you should be able to use the sound drivers directly if you need to |
| 06:52.15 | AndyGraybeal | it's got: ALSA: default, ALSA plughw:0.0: Hammerfall Digiface, ALSA: other device, OSS: /dev/dsp (says it's busy), and OSS: other device |
| 06:52.56 | AndyGraybeal | well if i use JACK which i would like to run all the time, i don't know how to get something to talk to something else if it doesn't run in jack! |
| 06:53.15 | AndyGraybeal | i need to route stuff to the sound card, and i do that with jack |
| 06:54.20 | AndyGraybeal | i'll mess around with it |
| 06:55.35 | mosty | doesn't your soundcard do hardware mixing? |
| 06:56.01 | AndyGraybeal | i don't understand what that means; i have a software mixing console for it though |
| 06:56.19 | AndyGraybeal | i guess that means it runs the hardware on the soundcard |
| 06:56.56 | AndyGraybeal | i guess jack just makes it easier |
| 06:57.18 | AndyGraybeal | no worries though, after i get this asterisk-jack thing working i'll be good |
| 07:07.43 | AndyGraybeal | so far twinkle seems nicer than xlite |
| 07:07.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:07.55 | AndyGraybeal | xlite is weird, and seemed to stop working after a while |
| 07:08.05 | mosty | xlite is old and buggy |
| 07:08.11 | mosty | full of memory leaks |
| 07:09.30 | AndyGraybeal | think a lot of my problems getting asterisk to work today was mostly xlite just not responding after a while |
| 07:09.40 | AndyGraybeal | but i thought it was asterisk :) |
| 07:14.04 | *** join/#asterisk d-tech (n=d-dtech@72.245.233.107) |
| 07:14.57 | AndyGraybeal | when choosing a voip service, are you limited by companies in your area? how does that work? |
| 07:15.14 | AndyGraybeal | er.. i mean -- are you limited *to* companies in your area |
| 07:15.22 | mosty | no |
| 07:15.27 | AndyGraybeal | limited to having to pick from companies only in my area |
| 07:15.32 | AndyGraybeal | thanks mosty |
| 07:15.37 | AndyGraybeal | do you have a voip service? |
| 07:15.46 | mosty | there's plenty, look on the wiki |
| 07:15.58 | AndyGraybeal | rock on |
| 07:21.17 | AndyGraybeal | yea, i don't understand how to do that |
| 07:21.30 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:21.32 | AndyGraybeal | er gah i was scrolled up |
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| 07:32.09 | AndyGraybeal | i get an error when i try to do this: svn co http://svn.digium.com/svn/asterisk/team/russell/jack asterisk-jack ; it says: svn: REPORT request failed on '/svn/asterisk/!svn/bc/90912/team/russell/jack' and also says: svn: '/svn/asterisk/!svn/bc/90912/team/russell/jack' path not found |
| 07:32.20 | AndyGraybeal | any ideas on what i might be doing wrong? |
| 07:37.48 | *** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.ca.comcast.net) |
| 07:38.07 | tzafrir_home | AndyGraybeal, http://svn.digium.com/svn/asterisk/team/russell/jack gives a "not found" messages |
| 07:38.59 | tzafrir_home | http://svn.digium.com/svn/asterisk/tags/ - I see we're up to 1.4.15 . almost there? |
| 07:39.00 | AndyGraybeal | yea, i'm poking around, but i don't understand it |
| 07:40.03 | AndyGraybeal | he musta taken it offline |
| 07:41.15 | AndyGraybeal | i guess now is a good time to go to sleep and see if russell is here in the morning |
| 07:43.39 | jblack | Does anyone else here use sipphone? |
| 07:44.49 | d-tech | <--- x-lite, zoiper and cipc |
| 07:45.18 | jblack | I mean the sip aspect, not the software aspect. |
| 07:45.30 | jblack | sipphone has conference rooms |
| 07:45.42 | d-tech | i use SIP softphone, yes! |
| 07:46.12 | jblack | Let me rephrase. Do you have a sipphone account, and are configured to dial their 222 area code? |
| 07:47.09 | d-tech | oh ... sorry no ... FWD only and routed out my FXS |
| 07:47.11 | *** join/#asterisk Al_WinKiller (i=Alex_Win@83.139.12.190) |
| 07:47.38 | jblack | Do you want to set up a sipphone account and we can try the conference channels out? |
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| 07:48.27 | d-tech | sure, gimme a couple clicks ... brb |
| 07:58.05 | jblack | heh. tellme is cool |
| 07:58.11 | z80asm | Does anyone in here develop thier own asterisk agi's in perl? |
| 07:58.25 | jblack | I've fiddled with existing one. |
| 08:00.10 | tzafrir_home | z80asm, maybe, Ask a specific question |
| 08:00.46 | z80asm | Would this work to allow more then 10+ dtmf tones |
| 08:00.48 | z80asm | $spoof_number = ''; |
| 08:00.48 | z80asm | for ($i = 1; $i <= @digits; ++$i) { |
| 08:00.49 | z80asm | for ($i = 10; $i >= @digits; ++$i) { |
| 08:01.51 | tzafrir_home | z80asm, what type of channel do you use? |
| 08:02.05 | jblack | Either you're relying on some really funky side effects, or you're new to perl. |
| 08:02.08 | z80asm | I'll give you the pastebin |
| 08:02.16 | z80asm | new |
| 08:02.25 | z80asm | http://rafb.net/p/I1NovQ76.html |
| 08:02.32 | z80asm | can I use two for statements? |
| 08:02.44 | z80asm | or shoud I use an If then? |
| 08:03.14 | jblack | What are you trying to do, exactly? In english |
| 08:03.30 | z80asm | allow 10+ digits to be enterd |
| 08:03.48 | jblack | Ok, So, read ten digits and set $call_number to them? |
| 08:03.52 | z80asm | if you enter 10 it will still work if you enter more then 10 it will still work. |
| 08:04.04 | z80asm | but no less then 10 |
| 08:04.18 | tzafrir_home | you can use two for{} loops. But you better close the second one |
| 08:04.20 | jblack | Ok. Now you're making sense. Hold. |
| 08:04.29 | tzafrir_home | You must close it, actually, |
| 08:04.43 | z80asm | But I want you guys to help me figure it out on my own not to tell me it. |
| 08:04.50 | tzafrir_home | And do use decent indentation. In vim: hit == |
| 08:04.56 | tzafrir_home | or =% |
| 08:05.02 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:05.15 | tzafrir_home | This would have exposed that error |
| 08:06.37 | jblack | You're aware of Asterisk::AGI, right? |
| 08:06.45 | tzafrir_home | Also: you use the same varualy in the pair of nested loops. Not nice |
| 08:07.03 | z80asm | it works for 10 |
| 08:07.07 | tzafrir_home | s/varualy/variable/ |
| 08:07.14 | z80asm | but I need to figure out how to do more then 10 .. |
| 08:07.27 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 08:07.34 | tzafrir_home | "more than 10" what? |
| 08:07.45 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 08:07.49 | tzafrir_home | dialing a number longer than 10 digits? |
| 08:07.49 | jblack | z80asm: There's the Asterisk::AGI module for perl, which automates most of this work for you. |
| 08:08.17 | tzafrir_home | z80asm, and it's not like Asterisk::AGI is not part of CPAN and included in some Linux distros |
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| 08:10.03 | jblack | Anyways, I'll make you a new get_digits |
| 08:14.18 | jblack | This untested code should be close to what you want: http://rafb.net/p/jcQPRI36.html |
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| 08:14.41 | ZX81 | hi, anyone know about [Jan 17 21:03:59] clamping target from 54 to 2 |
| 08:14.48 | ZX81 | generated by the jitterbuffer |
| 08:14.54 | ZX81 | prints millions of them |
| 08:15.11 | jblack | There are bugs in it. line 7 should have a my, line 8, // isn't a proper perl comment. |
| 08:15.12 | ZX81 | always from 54 to 2 |
| 08:15.18 | jblack | etcetra, etcetra. |
| 08:15.56 | jblack | And the while is backwards. |
| 08:16.12 | jblack | No, the while is fine |
| 08:16.23 | jblack | though, I think perl uses length, not strlen... =) |
| 08:17.48 | ZX81 | heh I fixed it meh |
| 08:18.54 | jblack | Why does it seem like most free sip providers are either crappy coders, marginally insane, or both? |
| 08:19.44 | jblack | Take sipphone for example. Their first conference line (the PARTY LINE!!) is loaded with jungle music. Nobody can hear anyone else because of the screaming monkeys, tigers and jungle music beat. |
| 08:19.59 | creativx | blame the drugs they are on |
| 08:20.43 | jblack | Let's stage an intervention, get these guys off of LSD, and get them some ritalin. At least until they straighten their services out. |
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| 08:29.54 | jblack | holy wow. Look at the peering list at sipbroker.com |
| 08:29.55 | Al_WinKiller | hi ppl, i got question, I can't impement radius support with asterisk,, but,, is there any way to install postgresql support for asterisk ? I mean to store usernames and passwords for asterisk users in pgsql db ? |
| 08:32.54 | Alexandre_fr | hey |
| 08:33.26 | Alexandre_fr | I would like to test my b410p card without using my PSTN line |
| 08:33.36 | Alexandre_fr | How can i do that ? |
| 08:35.07 | mosty | what are you trying to test? |
| 08:39.20 | mosty | Al_WinKiller, yes- asterisk realtime config |
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| 09:00.30 | *** join/#asterisk kodomo (n=memyself@emu.net.informatik.tu-muenchen.de) |
| 09:01.05 | kodomo | hi folks |
| 09:02.09 | kodomo | out of curiosity: is anyone planning or working on changing the SRV lookup routine to consider more than one SRV entry? |
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| 09:04.02 | jblack | kokomo: For E164? I hope so. |
| 09:04.31 | jblack | I know people have managed to manually hack up multiple lookups, but dont' remember specifically how they did it |
| 09:05.36 | kodomo | jblack: hm - I'm asking because looking up only one entry seriously breaks things for us ;) |
| 09:06.13 | jblack | enum, I meant. |
| 09:06.18 | jblack | Check this page out: http://messinet.com/node/190 |
| 09:06.55 | kodomo | Firewall issues forced my university to have an asterisk running on two interfaces - one just accepting connections from the inside, the other just from the outside of university firewall |
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| 09:07.34 | jblack | You're not in CT, are you? |
| 09:08.13 | jblack | Nope. Germany |
| 09:08.31 | kodomo | It's been tried to solve this by having multiple SRV entries - but asterisk only looks up one... and quite often the wrong one... leading to non-reachability of the number and seemingly to an endless loop printing out debug messages ;) |
| 09:08.33 | jblack | The Dundi peer that I peer with has connections with a number of german dundi peers. You guys are on the ball. |
| 09:08.55 | jblack | kodomo: Look at the page I gave you a link to. It has a enum macro that does multiple lookups, and tries them in order. |
| 09:08.56 | kodomo | jblack: actually I'm speaking of university of zuerich ;) |
| 09:09.21 | kodomo | jblack: *reading* |
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| 09:13.48 | kodomo | jblack: hm - not sure whether it's solving the issue - the problem really is with the SRV lookup, which I believe to be handled somewhere in the ENUM lookup (out of reach for people working with extension.conf macros) |
| 09:14.14 | jblack | That's why he's dumping into an array |
| 09:14.28 | kodomo | but it's only one part for me being able to reach UZH and another for UZH to be reachable by arbitrary *-servers ;) |
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| 09:14.39 | jblack | He's grabbing _ALL_ enum records, then 1 by one, checking for validity |
| 09:15.41 | kodomo | hm - I'll try it... |
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| 09:17.34 | kodomo | jblack: ok - I'm quite sure that this won't work, as there are only (exactly) 2 ENUM entries... both going to ifi.uzh.ch (one for iax2, one for sip) |
| 09:19.16 | kodomo | but ifi.uzh.ch is going to be resolved to different results by SRV entries... which are chosen randomly, as far as I can see... so you'd basically have to run the ENUM lookup (stuffing results into the array) several times, until you're satisafied that you probably have all records |
| 09:19.41 | jblack | <PROTECTED> |
| 09:20.26 | jblack | I got nothing else for ya. If that's not what you need, then good luck |
| 09:20.58 | kodomo | jblack: correct me, but it's not what he does... he makes the lookup, which returns the number of results of one query, then he requests each of those result entries |
| 09:22.32 | jblack | Honestly, I don't know exactly how it works. I'm telling you what he told me on the phone. :) |
| 09:23.12 | kodomo | jblack: appreciated :) thx anyhow - but I really was interested, whether there were plans on changing this in * itself (as the comment I read w.r.t. this issue was something like 'we don't care about load balancing'... but in this case the multiple entries are not used for load balancing, but failover... |
| 09:24.00 | jblack | I'd say in line 2 he creates an array, then he's requesting offsets 1 by 1. |
| 09:24.12 | jblack | ,c, looks like count to me, then ${i} is his iterator |
| 09:24.28 | kodomo | so - if the multiple SRV entries are not looked up and tried, the result's the unreachability of destination, which would otherwise be reachable |
| 09:24.48 | jblack | You see the loop, right? |
| 09:25.02 | jblack | Particularly exten => s,n(increment),Set(i=${MATH(${i}+1,i)}) |
| 09:25.19 | jblack | Then he goes back to next, with i one larger, and on the next line, he again does: |
| 09:25.23 | jblack | exten => s,n,Set(uri=${ENUMLOOKUP(${enum-num},ALL,,${i},${enum-domain})}) |
| 09:25.33 | kodomo | jblack: I do - but look at the second and third line of macro-enum |
| 09:25.54 | jblack | Yeah, it looks to me like he's making an empty array called i. |
| 09:26.06 | kodomo | there, apparently, he receives the number of results of the query... |
| 09:26.16 | jblack | with the count result from ENUMLOOKUP defining the size of the array. |
| 09:26.33 | kodomo | and then he iterates through the results |
| 09:26.54 | kodomo | so - if you have an ENUM entry with let's say 5 entries, |
| 09:27.13 | kodomo | he gets the information about 5 results, and then proceeds be requesting each of them |
| 09:27.23 | jblack | that's how it looks to me, yes. |
| 09:27.51 | kodomo | problem in the SRV case: one single result is resolved by SRV to several others - nondeterministically |
| 09:28.08 | jblack | Well, look at the Set(uri=) line.... |
| 09:28.20 | jblack | See how he uses ${i} ? I think he's specifing which srv record he wants. |
| 09:28.44 | jblack | ENUMLOOKUP(number[|Method-type[|options[|record#[|zone-suffix]]]]) |
| 09:29.31 | jblack | I imagine if you don't give record#, it's nondeterministic. |
| 09:29.50 | kodomo | jblack: I concur - but in this case 'Entry#1 <=> Entry#1' does not hold |
| 09:30.03 | jblack | are you sure? |
| 09:30.35 | jblack | I know if no record# is given, a random one is chosen. I don't know that's the case if a record number is. |
| 09:31.06 | kodomo | quite - what I read is that * only considers the first SRV entry it encounters, when expanding ENUM entries... and I don't see that handled on the Macro level |
| 09:31.34 | jblack | I don't know what you want from me. I'm really getting frustrated though. |
| 09:31.54 | jblack | You're asking me to explain a macro that I already told you that I don't understand. All that I can really tell you is he tells me this works. |
| 09:31.55 | kodomo | so - ENUM entry #1 looks differently depending on which SRV entry happened to be the first one |
| 09:32.12 | kodomo | jblack: sorry - I didn't intend to frustrate you |
| 09:32.16 | jblack | Either it does, or it doesn't. Try it, or don't. There's nothing more that I can add. |
| 09:32.17 | *** join/#asterisk ifnotwhynot (n=davidh@c1-124-9.rrba.isadsl.co.za) |
| 09:32.37 | kodomo | I also didn't intend to solve an issue with my own system |
| 09:32.56 | jblack | That enum is a mess is a very popular discussion topic. =) |
| 09:33.06 | kodomo | but I wanted to know whether this SRV lookup issue was currently being addressed or ignored |
| 09:33.10 | kodomo | ;) |
| 09:33.24 | jblack | I don' know. |
| 09:33.40 | ifnotwhynot | hi ther anyone know if there is a sip phone application for ones mobile phone that can link to asterisk? |
| 09:34.00 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 09:34.06 | kodomo | symbian based nokia phones can |
| 09:34.30 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 09:34.42 | jblack | Perhaps you'd be willing to fix it? |
| 09:35.02 | kodomo | (just google for nokia and asterisk and you will find howtos) |
| 09:35.18 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:35.53 | ifnotwhynot | thx kodoma |
| 09:36.13 | kodomo | jblack: actually considering it - but that's not necessary if someone's already working on it - and I'm not sure if and when I'll/I'd find the time to work on this :-/ |
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| 09:37.33 | ifnotwhynot | i have i jasjam imate running windows mobile 5 think i wil get it to work using asterisk? |
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| 10:41.10 | Olobola | what do charge hourly to write dialplans? |
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| 10:46.29 | jblack | damnit! |
| 10:50.51 | Alexandre_fr | I have a question I want tu put my te220B on e1 mode |
| 10:51.22 | Alexandre_fr | I put the jumper on the pin 1 2 3 and 4 |
| 10:51.53 | Alexandre_fr | and it's still in T1 mode |
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| 10:53.02 | JT | Alexandre_fr: and did you configure zaptel.conf appropriately for an E1? |
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| 10:53.26 | phix | hey |
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| 11:02.08 | *** join/#asterisk alinux-lb22 (n=asdf@mail.splendor.net) |
| 11:02.31 | alinux-lb22 | hi all is it legal to create VOIP appliance that contains astersik without telling the client that it is based on asterisk ? |
| 11:03.26 | dennis- | hmm, what could be the problem if the caller can hear me, but i can't hear him/her? i am on a sip softphone connected to an asterisk pbx |
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| 11:07.57 | tzafrir | alinux-lb22, you must notify the clients of their right according to the GPL - either provide a download of the source code or provide a written offer to provide it |
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| 11:15.13 | alinux-lb22 | do I have to mention that it is asterisk |
| 11:15.25 | alinux-lb22 | or can I provide the license without naming the program |
| 11:15.37 | alinux-lb22 | I know this is a delicate matter |
| 11:15.37 | alinux-lb22 | that is why I am asking |
| 11:18.29 | kodomo | alinux-lb22: not really sure... but... when asked... you have to provide the source code... so what's the gain anyway? ;) |
| 11:19.04 | alinux-lb22 | kodomo, some stupid thing my boss asked about..I am against it but I need to gather some facts |
| 11:19.52 | kodomo | alinux-lb22: did you already read the GPL? It's actually quite short and simple... |
| 11:20.06 | mosty | you're allowed to rename software under the gpl i believe |
| 11:20.53 | tzafrir | In fact, the Digium trademark usage guidelines practically require you to rename it |
| 11:22.13 | kodomo | alinux-lb22: the basic elements are that a) you have to allow customers access to the source code and b) you have to explicitly inform them about their rights to that respect |
| 11:24.06 | kodomo | I don't think you have to name the name of the original program (actually I believe there's been a dispute over one of the X11 implementations because they added a restriction that they had to be mentioned in derivatives, which clashed with the GPL |
| 11:24.43 | kodomo | but: you have to inform the customers of their rights to get the source code and provide it... so anyone who bothers will find out... |
| 11:25.31 | kodomo | so I really don't see a gain in not telling them that it's asterisk based... |
| 11:29.24 | mosty | tzafrir, i have an issue with a sangoma card on an x86_64 machine with 4G of ram (I compiled with the 64 bit/4G ram flag), using the latest wanpipe and zaptel 1.4.7.something, is it worth trying 1.4.8 before submitting a bug report to sangoma? |
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| 11:45.59 | tzafrir | mosty, nothing I can think of |
| 11:46.52 | phix | hi |
| 11:47.02 | phix | I am having issues registering two SIP providers |
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| 11:47.16 | phix | Both say they are registered but the 2nd one isn't |
| 11:47.19 | phix | any ideaS/ |
| 11:48.19 | mosty | phix, how do you know it isn't registered? |
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| 12:32.26 | z80asm | I keep getting this error |
| 12:32.30 | z80asm | an 17 07:31:09 NOTICE[22224]: pbx_dundi.c:451 reset_global_eid: No ethernet interface found for seeding global EID You will have to set it manually. |
| 12:32.31 | z80asm | Jan 17 07:31:09 ERROR[22224]: pbx_dundi.c:4771 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use |
| 12:34.42 | z80asm | nvm |
| 12:35.23 | *** join/#asterisk RoyK (n=roy@91.149.13.189) [NETSPLIT VICTIM] |
| 12:35.23 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 12:36.06 | *** join/#asterisk [Gandhi] (n=hellboy@208.50.70.1) |
| 12:36.43 | *** join/#asterisk allankardec (n=root@201.50.80.173) |
| 12:41.34 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.64.193) |
| 12:44.22 | *** join/#asterisk pablus (n=pablus@test.conama.cl) |
| 12:46.32 | *** join/#asterisk itguru (n=gabriel@82.108.189.18) |
| 12:46.34 | *** join/#asterisk oej (n=olle@213.115.215.130) |
| 12:48.11 | itguru | I've got an AsteriskNOW installation, no errors on install, now I got the hard task of getting my handsets to work. I've been told that my handsets use something called skinny? |
| 12:49.09 | pablus | hi!, i have a doubt about asterisk, we have in our organization a new PBX, our question is it's possible to use asterisk in front of our legacy pbx without paid more $$ for to use 8 line from legacy PBX? |
| 12:49.16 | mosty | we don't really deal with asterisknow in this channel, see the topic. but the quick answer would be upgrade your phones to use sip if at all possible |
| 12:49.40 | itguru | mosty: It's not possible to upgrade them :( |
| 12:49.56 | mosty | pablus, depends what your legaxy pbx does and how you want to integrate it. it will definitely take time and effort to do |
| 12:50.59 | pablus | mosty: we want to use for make a call to other lines (inside our organization) |
| 12:52.05 | mosty | pablus, it really comes down to what your legacy pbx supports |
| 12:55.21 | defswork | pablus: bite the bullet and ditch the entire system |
| 12:57.29 | tzafrir | pablus, if the 8 lines are analog it is possible, but probably will not be completely transparent to users |
| 13:02.06 | pablus | Hmmm, for the other users... I could to use a software telephones... ;-) |
| 13:04.11 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:04.17 | defswork | pablus: I've only been using/installing/recommending asterisk for 10 months or so and I wouldn't touch a "traditional" PBX now |
| 13:04.46 | *** join/#asterisk mkl1525 (n=qwertz@tdsl.de-p-fw1.cp-net.de) |
| 13:05.00 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 13:05.43 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 13:05.50 | jblack | ~dundi |
| 13:05.51 | jbot | well, dundi is http://www.dundi.com |
| 13:05.57 | mkl1525 | Hi, in * 1.4 there's the AGENT() function to get agent names is there a way to get something similar in 1.2? |
| 13:06.04 | jblack | Shame, jbot. You should be smarter than that. |
| 13:06.54 | JT | tzafrir: no reason why it's not possible if the lines are BRI or PRI, either |
| 13:08.42 | tzafrir | JT, with ISDN you have much nicer signalling |
| 13:09.54 | tzafrir | So you don't need to answer a call to know what it is about. Or generally - delay it a few seconds |
| 13:11.15 | RoyK | if using hwec cards from digium or sangoma - how does zaptel know it has hwec? |
| 13:11.31 | JT | tzafrir: i realise |
| 13:12.02 | tzafrir | RoyK, each span has an optional span method for echo cancelling |
| 13:12.15 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:12.25 | tzafrir | If the span does not provide it (that pointer is NULL) , the generic Zaptel one is used |
| 13:13.01 | *** join/#asterisk [gnubie] (n=[gnubie]@cm58.gamma178.maxonline.com.sg) |
| 13:13.06 | jblack | ~dundi |
| 13:13.06 | jbot | from memory, dundi is at http://www.dundi.com. DUNDi, an optional Asterisk component, provides routes to PSTNs between peers on the same DUNDi network. |
| 13:14.03 | JT | that doesn't really say what dundi is though |
| 13:14.09 | JT | that's certainly a use of it |
| 13:15.29 | tzanger | I call my kids a dundi when they pull a boner |
| 13:15.37 | JT | hah |
| 13:15.38 | RoyK | tzafrir: how can I query zaptel to see if hwec is in use? |
| 13:15.39 | jblack | This is a little better. |
| 13:15.41 | jblack | ~dundi |
| 13:15.42 | jbot | it has been said that dundi is at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 13:16.24 | jblack | It's a step up from "dundi is at http://www.dundi.com" |
| 13:16.48 | tzanger | heh |
| 13:16.53 | tzafrir | RoyK, I don't know of a way |
| 13:19.48 | *** join/#asterisk ManxPower (n=manxpowe@69.2.85.41) |
| 13:20.03 | RoyK | tzafrir: do you know where I can find this in the code? it'd be nice to see. I have quite high load on a system, in zaptel asa reported by oprofile, but I don't have zaptel symbols for oprofile and I don't know where I can find them, so adding debug lines might help |
| 13:20.47 | tzafrir | RoyK, you should be able to point oprofile to extra modules |
| 13:21.04 | tzafrir | From the modules themselves? |
| 13:21.21 | RoyK | tzafrir: do you know how I can do that? |
| 13:21.41 | tzanger | how many zaptel lines? |
| 13:21.46 | tzanger | RoyK: you look at zaptel when it loads I believe |
| 13:21.55 | tzanger | it emits the echo canceller in dmesg |
| 13:22.01 | tzanger | not sure about hpec though |
| 13:24.10 | JT | hwec you mean? |
| 13:31.14 | tzanger | oh hwec... that should come up automatically when the card with the hardware ec is loaded |
| 13:31.22 | tzanger | i.e. dmesg should say something about it when the module loads |
| 13:31.31 | tzanger | From: "0522240105" <sip:01@192.168.1.15;user=phone>;tag=e4762b74-685800 |
| 13:31.42 | tzanger | asterisk would pull the CID from that, wouldn't it? i..e "0522240105" ? |
| 13:32.15 | RoyK | tzafrir: won't that be the cid name? and 01 the cid num? |
| 13:32.29 | tzanger | that's kind of what I'm thinking |
| 13:32.35 | *** join/#asterisk ming_zym (n=ming_zym@124.14.234.212) |
| 13:39.05 | itguru | Linksys PAP2, has anyone connected one of these devices to an Asterisk box before? |
| 13:41.06 | jblack | yup. all the time |
| 13:41.48 | jblack | heh. Jim Cramer blew another gasket. |
| 13:43.13 | mosty | itguru, yes of course |
| 13:43.41 | RoyK | anyone that knows how I can have oprofile read the zaptel symbols? |
| 13:47.26 | *** join/#asterisk RoyK (n=roy@91.149.13.189) [NETSPLIT VICTIM] |
| 13:47.26 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 13:49.01 | *** join/#asterisk Victor_Yure (n=Victor_Y@postfix.tradein.com.br) |
| 13:50.31 | *** join/#asterisk Twotone1 (n=Sp00gE@24-158-189-53.dhcp.jcsn.tn.charter.com) |
| 13:53.27 | *** join/#asterisk my007ms (i=master@botmaster.x86.be) |
| 14:01.23 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 14:09.12 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:10.27 | *** join/#asterisk mkl1525 (n=qwertz@tdsl.de-p-fw1.cp-net.de) [NETSPLIT VICTIM] |
| 14:11.27 | cappiz | when i get a: SIP/2.0 404 Not found e |
| 14:11.28 | mkl1525 | Hi, (* 1.2) is there a way to get the longest wait time of all callers in the queue using AMI or events? |
| 14:11.36 | cappiz | -e* What might tha be caused by? |
| 14:12.12 | [TK]D-Fender | cappiz: How about showing us the COMPLETE SIP debug for that call attempt |
| 14:12.20 | [TK]D-Fender | cappiz: PASTEBIN is your friend |
| 14:12.22 | [TK]D-Fender | ~pb |
| 14:12.23 | 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 |
| 14:12.55 | [TK]D-Fender | mkl1525: yes, "show queue [queuename]" |
| 14:13.00 | cappiz | [TK]D-Fender, its a register attepmt :P |
| 14:13.04 | cappiz | ill pastebin |
| 14:13.09 | [TK]D-Fender | cappiz: PASTEBIN |
| 14:14.07 | *** join/#asterisk ormog (n=talrasha@p508E2DC9.dip0.t-ipconnect.de) |
| 14:15.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:15.05 | cappiz | [TK]D-Fender, http://pastebin.com/d7b1f52c8 |
| 14:15.49 | [TK]D-Fender | cappiz: "username" did not match anyone in sip.conf. Easy as that |
| 14:16.26 | cappiz | at cbktele.com's sip.conf? |
| 14:16.52 | [TK]D-Fender | cappiz: Yes |
| 14:17.16 | cappiz | [TK]D-Fender, ok - they told me they activated it, bah.... |
| 14:17.34 | [TK]D-Fender | cappiz: maybe your register statement is bad. |
| 14:17.42 | *** join/#asterisk allankardec (n=root@201.50.80.173) |
| 14:17.54 | [TK]D-Fender | cappiz: or it could be the account isn't there in any way. |
| 14:18.11 | cappiz | [TK]D-Fender, i tried username:pass@sip.cbktele.com/username and username:pass@sip.cbktele.com |
| 14:18.37 | [TK]D-Fender | cappiz: the latter is proper way |
| 14:19.02 | cappiz | ok, i have one provider that needs that /username |
| 14:19.07 | [TK]D-Fender | cappiz: actually... highlighting threw me off, EITHER is fine. The latter specifies a return estension. |
| 14:19.34 | [TK]D-Fender | cappiz: no, the "/whatever" on the end is jsut a return extension and not tecnically necessary |
| 14:19.54 | cappiz | ok :) |
| 14:21.07 | *** join/#asterisk grayhame (n=IceChat7@ns2.dalcon.com) |
| 14:24.45 | *** join/#asterisk nighty^ (n=nighty@p7068-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 14:25.16 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 14:26.05 | ^shark_ | hi guys i am running 1.4.17 and i am ooking for a good tutorial on music on hold |
| 14:26.08 | ^shark_ | any ideas |
| 14:26.46 | ^shark_ | looking* |
| 14:28.17 | [TK]D-Fender | ^shark_: ... |
| 14:28.19 | [TK]D-Fender | ~book |
| 14:28.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 14:28.21 | [TK]D-Fender | ^^^^^^^^^^ |
| 14:29.44 | my007ms | hello any guru there |
| 14:29.56 | my007ms | my asterisk not load chan_sip.so |
| 14:30.00 | my007ms | is was work fine |
| 14:30.08 | my007ms | and stop work |
| 14:30.09 | [TK]D-Fender | my007ms: You have a soft-phone running on your server? |
| 14:30.10 | my007ms | now |
| 14:30.19 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:30.19 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:30.26 | ^shark_ | [TK]D-Fender: thanks |
| 14:30.32 | my007ms | no in my server |
| 14:30.36 | my007ms | but in other PC yes |
| 14:30.57 | [TK]D-Fender | my007ms: what happens when you try "load chan_sip.so">? |
| 14:31.35 | my007ms | it run the command then asterisk CLI hung |
| 14:32.03 | my007ms | but sip not load |
| 14:32.06 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 14:32.27 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 14:32.40 | [TK]D-Fender | my007ms: if it seems to hang that can happen on things like DNS resolution problems on registers, etc... go check that DNS is working normally. If it seems to, then start commenting out your REGISTER statements one by one. |
| 14:32.56 | *** part/#asterisk ming_zym (n=ming_zym@124.14.234.212) |
| 14:34.11 | my007ms | [TK]D-Fender, sorry where is REGISTER statements |
| 14:34.12 | my007ms | ? |
| 14:34.21 | [TK]D-Fender | my007ms: ..... SIP.CONF |
| 14:34.37 | my007ms | DNS work fine |
| 14:34.58 | fiXXXerMet | In extensions.conf, I have MeetMe(555,cros) but when I place a call to the conference, the output shows only the 'r' option. I am not told how many users are in the room, and there is no menu presented. |
| 14:35.22 | lmadsen | uhhh... did you 'dialplan reload' ? |
| 14:35.40 | lmadsen | verify with 'dialplan show my_context_with_meetme' |
| 14:35.56 | fiXXXerMet | Odd, now it is announcing how many are in the room, but i am not getting a menu. |
| 14:37.20 | my007ms | [TK]D-Fender, where i can find full debug info |
| 14:37.24 | my007ms | no much info in full |
| 14:38.00 | [TK]D-Fender | my007ms: I jsut told you to start by commenting out your registers, one at a time. Try this first, applying after each. |
| 14:38.59 | [TK]D-Fender | fiXXXerMet: so you press * and nothing happens? |
| 14:39.40 | fiXXXerMet | Correct. |
| 14:39.54 | *** join/#asterisk saftsack (n=saftsack@p4FC764C2.dip.t-dialin.net) |
| 14:40.16 | [TK]D-Fender | fiXXXerMet: Is there anyone else in the conference? |
| 14:40.24 | fiXXXerMet | No. |
| 14:40.45 | [TK]D-Fender | fiXXXerMet: jsut a thought that it might not give a menu if there wasn't anything practical you could do... |
| 14:40.48 | fiXXXerMet | Works when there is another person. |
| 14:40.55 | fiXXXerMet | :) thanks |
| 14:40.59 | [TK]D-Fender | fiXXXerMet: or perhaps your DTMF is simply not functional at all |
| 14:41.13 | [TK]D-Fender | fiXXXerMet: Feel free to slap yourself now :) |
| 14:41.20 | *** join/#asterisk ming_zym (n=ming_zym@124.14.234.212) |
| 14:41.27 | fiXXXerMet | I do it enough already. |
| 14:41.49 | [TK]D-Fender | fiXXXerMet: then this one should blend in nicely. Don't forget the brass knuckles :0 |
| 14:42.00 | fiXXXerMet | :-/ |
| 14:42.10 | *** join/#asterisk oej (n=olle@213.115.215.130) |
| 14:42.49 | my007ms | [TK]D-Fender, i comment them all |
| 14:42.53 | my007ms | still not work |
| 14:43.07 | [TK]D-Fender | my007ms: Well you're going to have to pastebin something for us... |
| 14:43.25 | my007ms | log ?? |
| 14:44.00 | [TK]D-Fender | my007ms: log, CLI output, configs. CLI upon STARTING *, dns verification tests, network connectivity tests, etc |
| 14:44.55 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1088827265.dsl.bell.ca) |
| 14:45.33 | my007ms | and what is dns verification tests |
| 14:45.47 | my007ms | i try to resolve www.yahoo.com and it's work :) |
| 14:46.41 | ^shark_ | [TK]D-Fender: do i need to download asterisk-addons for my 1.4.17 music on hold functionality? |
| 14:47.21 | [TK]D-Fender | ^shark_: if you plan on using MP3's and Native MoH, yes |
| 14:47.55 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 14:48.31 | ^shark_ | [TK]D-Fender: ok and Native is a replacement for mpg123 right? |
| 14:48.48 | [TK]D-Fender | ^shark_: aN OPTION, YES |
| 14:49.00 | ^shark_ | [TK]D-Fender: thanks |
| 14:49.37 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583870.dsl.bell.ca) |
| 14:50.18 | *** join/#asterisk ifnotwhynot (n=davidh@196.211.34.2) |
| 14:51.02 | ifnotwhynot | does anyone know of better softphones that sjphone for conecting pda's to asterisk? |
| 14:51.15 | ifnotwhynot | batling with voice quality |
| 14:51.19 | ^shark_ | [TK]D-Fender: then later i just edit my musiconhold.conf file right? |
| 14:51.43 | ifnotwhynot | hi [TK]D-Fender |
| 14:51.58 | [TK]D-Fender | ^shark_: yup |
| 14:52.28 | tzafrir | But why would you use mp3 for native music on hold? That's a waste of CPU cycles for nothing |
| 14:52.34 | ifnotwhynot | hows it ringing, when will you be up for that South african safari sonsored by myselve? |
| 14:53.14 | [TK]D-Fender | tzafrir : because people often have cycles to spare and don't care to bother converting file formats. |
| 14:53.18 | ifnotwhynot | hows it ringing, when will you be up for that South african safari sonsored by myselve? [TK]D-Fender??? |
| 14:53.30 | tzafrir | Just use wavs. It's not even a waste of disk space, normally |
| 14:53.47 | mosty | is it possible to tell a polycom phone to load a specific config file via the web interface? i have some remote phones that (obviously) won't get my dhcp settings, and therefore won't download my config files |
| 14:53.55 | tzafrir | Because the mp3s you have are typically of higher bit rate |
| 14:55.16 | ^shark_ | tzafrir: wavs is an option for mpg123? is it an application too and where can i find the information to install it on etch |
| 14:55.47 | [TK]D-Fender | mosty: no need for DHCP... just point them to a remote provisioning server |
| 14:55.59 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:56.14 | [TK]D-Fender | ^shark_: Native MoH can play anything you have a "format_XXX.so" for |
| 14:56.31 | [TK]D-Fender | ^shark_: "core show modules like format" |
| 14:56.34 | my007ms | [TK]D-Fender, :( |
| 14:56.44 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:56.46 | mosty | [TK]D-Fender, where is that in the web interface? |
| 14:57.04 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 14:57.05 | [TK]D-Fender | mosty: If you even TOUCH the web interface you should be dragged out and SHOT :p |
| 14:57.13 | [TK]D-Fender | mosty: in the BootROM <- |
| 14:57.28 | [TK]D-Fender | mosty: reboot the phone,enter the BR menu and enter your server IP. |
| 14:57.50 | mosty | [TK]D-Fender: ack. i provisioned these phones with dhcp/http, and now some of them have been taken offsite, i'm trying to change the settings of the offsite phones |
| 14:58.12 | [TK]D-Fender | mosty: have someone go over, pump in your extern IP for provisioning and be done with it. |
| 14:58.18 | Toerkeium | does anyone knows if a company gives free DID for mexico, brazil and spain? |
| 14:58.25 | [TK]D-Fender | mosty: could have been done in the span of this request :) |
| 14:58.55 | mosty | [TK]D-Fender, these particular phones are three time zones west of here ;) i'll write a doc for the local tech |
| 14:59.08 | [TK]D-Fender | mosty: good idea :) |
| 14:59.45 | tzafrir | ^shark_, wavs are not an option for mpg123 . But you can use sox instead |
| 15:00.05 | RoyK | is zaptel multithreaded, or is just one thick rope? |
| 15:00.34 | tzafrir | The "non-native" music on hold interface is basically a pipe through which a program sends slin data |
| 15:01.02 | cesar_CR | hi guys, can I install the digium cards in a Netra 20 server ? |
| 15:01.35 | tzafrir | Zaptel is kernel. Asterisk reads it from one side. drivers push data on the other side |
| 15:01.51 | tzafrir | Drivers push data from interrupts |
| 15:02.24 | RoyK | tzafrir: sure, but it looks like all the time zaptel spends, is spent on cpu 0, and the linux kernel normally scales well across cpus |
| 15:02.24 | cesar_CR | I'm planing to install linux over it... |
| 15:02.38 | tzafrir | Asterisk itself is multithreaded, basically - a thread per active channel. And one monitor thread for the rest of them |
| 15:03.03 | RoyK | I just don't understand why cpu0 is overloaded while the others are more or less idle |
| 15:03.10 | tzafrir | RoyK, try sending the interrupts elsewhere |
| 15:03.34 | tzafrir | Interrupts from the same device tend to stick to one CPU |
| 15:04.41 | RoyK | hm. there was an interrupt balancer daemon somewhere, wasn't it? |
| 15:05.40 | tzafrir | Not really sure. But maybe just sending the interrupts to a different CPU will help balance things |
| 15:05.53 | RoyK | apt-get install irqbalance |
| 15:05.58 | tzafrir | It probably helps to keep the handling of the interrupt local to a specific CPU |
| 15:06.06 | mosty | Toerkeium, unlikely, since DID's cost the telco actual money |
| 15:06.52 | *** join/#asterisk AndyGraybeal (n=andy@node49.32.251.72.1dial.com) |
| 15:07.26 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/staff/sargun) |
| 15:08.21 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 15:08.22 | Toerkeium | mosty: I've found for US, UK, IT.. was hoping a few more countries :) |
| 15:08.41 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:08.48 | RoyK | Toerkeium: we have for .no |
| 15:08.48 | xbmodder_ | Is there a way I can get a dump of all the variables in the asterisk dial plan? |
| 15:08.52 | xbmodder_ | Like as I'm dialing |
| 15:08.55 | bsdwarrior | im running 1.2.14 and im having trouble getting periodic_announce to work with realtime queues. |
| 15:09.07 | xbmodder_ | Is there a UNIX eq to "export" |
| 15:09.09 | [TK]D-Fender | xbmodder_: You've got the source... |
| 15:09.10 | tzafrir | xbmodder_, channel variables or global variables? |
| 15:09.21 | xbmodder_ | [TK]D-Fender, haha |
| 15:09.23 | xbmodder_ | tzafrir, both. |
| 15:09.26 | tzafrir | For globals: you can do that in the CLI |
| 15:09.35 | Toerkeium | RoyK_ what's .no? |
| 15:09.38 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:09.49 | dandre | Hello , |
| 15:10.05 | Qwell | Just say no to .no |
| 15:10.17 | RoyK | Toerkeium: norway |
| 15:10.25 | tzafrir | no-way |
| 15:10.39 | Toerkeium | RoyK_: what's the company? |
| 15:10.47 | RoyK | fortel.no |
| 15:11.00 | tzafrir | xbmodder_, for local vars: I think that you get htat by running an AGI script |
| 15:11.02 | dandre | I am trying to setup a connection to a sip proxy from my asterisk and I can't get calls from that proxy. here is the sip debug: |
| 15:11.04 | dandre | http://pastebin.ca/858520 |
| 15:11.04 | Toerkeium | RoyK_: thank you |
| 15:11.10 | RoyK | Toerkeium: np |
| 15:11.20 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:11.26 | *** join/#asterisk AndyGraybeal (n=andy@node49.32.251.72.1dial.com) |
| 15:11.33 | xbmodder_ | Like ANI, CID, CID[name] |
| 15:11.37 | dandre | I don't understand why my call is rejected |
| 15:11.48 | bsdwarrior | is anyone else using periodic_announce ? |
| 15:13.41 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 15:14.00 | *** join/#asterisk luke-jr (n=luke-jr@wsip-70-167-147-10.om.om.cox.net) |
| 15:14.05 | luke-jr | Anyone know who Mark Turner is? |
| 15:14.34 | mort_gib | Hi |
| 15:14.40 | mort_gib | Hardware question.... |
| 15:14.57 | xbmodder_ | Does asterisk support ANI even? |
| 15:14.57 | [TK]D-Fender | bsdwarrior: And continuing from yesterday, you have still shown us absolutely nothing of value for us to help you with. |
| 15:15.10 | [TK]D-Fender | xbmodder_: "show function CALLERID" |
| 15:15.36 | [TK]D-Fender | dandre: I'm betting you ITSP doesn't want to be challenged when sending you calls. |
| 15:15.38 | mort_gib | I'm building a 20 user system, and the we have found out the client can do with 3 X ISDN2 (bri) lines |
| 15:15.41 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:15.56 | [TK]D-Fender | dandre: put "insecure=port,invite" into their sip.conf entry. |
| 15:16.06 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:16.10 | mort_gib | I got hold of a Sangoma A500 PCI Express version |
| 15:16.44 | xbmodder_ | [TK]D-Fender, So, like "Noop(${CALLERID(ANI)}) |
| 15:16.52 | [TK]D-Fender | xbmodder_: go try.. |
| 15:16.54 | xbmodder_ | Would noop with the ANI as the argument? |
| 15:17.20 | mort_gib | But I can't fit it in the PCI Express slot, I comes with a little "tab" that prevents the card from going all the way down in the slot. -Had that issue before?? |
| 15:17.27 | my007ms | [TK]D-Fender, what is this site i can use to copy and past |
| 15:17.32 | dandre | [TK]D-Fender: ok thanks it works |
| 15:17.38 | [TK]D-Fender | ~pastebin |
| 15:17.39 | 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 |
| 15:17.41 | mosty | mort_gib, are you sure you have a pci-express slot? |
| 15:17.41 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:17.47 | [TK]D-Fender | dandre: You're welcome. |
| 15:18.10 | mort_gib | mosty: Eh, pretty sure! The server don't come with anything else.... |
| 15:18.41 | mosty | mort_gib, can you get a photo of it or something? |
| 15:19.03 | mort_gib | Difficult right now (of course) sigh :-( |
| 15:19.26 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 15:19.43 | my007ms | http://pastebin.ca/858531 this log i get when i run asterisk with -vvvvgc |
| 15:19.57 | ifnotwhynot | [TK]D-Fender]: does anyone know how to setup asterisk and ESCsoftphone for wifi. |
| 15:20.00 | mosty | mort_gib, is the "tab" on the slot or on the card? |
| 15:20.24 | *** join/#asterisk Dovid (n=Dovid@bzq-79-183-138-50.red.bezeqint.net) |
| 15:20.38 | Dovid | hi. anyone here set up ooh323 with cisco ? |
| 15:20.39 | xbmodder_ | Hmph. |
| 15:21.04 | xbmodder_ | H.323 |
| 15:21.04 | xbmodder_ | H.264 |
| 15:21.19 | mort_gib | The "tab" is on the card, the PCIe version has two short tabs with connectors on, they fit into the PCIe slot fine, but towards the end of the cabinet the A500 has another tab without connectors half height of the PCIe tabs |
| 15:21.24 | [TK]D-Fender | my007ms: that doesn't show anything useful. There is no reference to chan_sip.so, and no crash. |
| 15:21.26 | Dovid | xbmodder_:Some cisco box's work with asterisk and some dont |
| 15:21.39 | [TK]D-Fender | mort_gib: just call Sangoma up. |
| 15:21.40 | Dovid | cant seem to figure out why. using h323 for termination only |
| 15:21.45 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 15:21.49 | mort_gib | -Yeah?? |
| 15:22.00 | xbmodder_ | Dovid, Odd. |
| 15:22.02 | mosty | mort_gib, you lost me sorry |
| 15:22.11 | xbmodder_ | Dovid, I love my polycom. |
| 15:22.19 | xbmodder_ | am I going to San Jose station yet? |
| 15:22.32 | mort_gib | mosty -So they actually pick up the phone! |
| 15:23.02 | my007ms | [TK]D-Fender, yes i search full log also and find nothing so asterisk stop b4 it come to chan_sip.so but where and why ? |
| 15:23.12 | *** join/#asterisk andresmujica (n=andresmu@190.24.108.35) |
| 15:23.19 | ^shark_ | i cant find anything about Native as software for music on hold |
| 15:23.22 | ^shark_ | on etch |
| 15:23.26 | Dovid | xbmodder_: It's wierd. I have not used h323 much so I am not good at trouble shooting it |
| 15:23.34 | [TK]D-Fender | my007ms: So what tells you that chan_sip.so is even responsible for any of your problems? |
| 15:23.39 | xbmodder_ | Dovid, you bought a bunch of Cisco IP Phones? |
| 15:23.39 | Dovid | trying to figure out where I am going wrong. |
| 15:23.43 | Dovid | nnno |
| 15:23.53 | Dovid | xbmodder_: ay i PM ? |
| 15:23.55 | my007ms | [TK]D-Fender, my problem that chan_sip.so not load |
| 15:23.58 | Dovid | h323 gateway |
| 15:24.02 | [TK]D-Fender | ^shark_: it doesn't have anything to do with external packages or distro related matters <- |
| 15:24.03 | my007ms | coz i can not use any sip command |
| 15:24.15 | [TK]D-Fender | ^shark_: Native = "mode=files" <--- |
| 15:24.15 | my007ms | and all phone not working |
| 15:24.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:24.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:24.38 | [TK]D-Fender | my007ms: Well you'd better show us something useful... your output isn't complete |
| 15:24.53 | my007ms | [TK]D-Fender, like ? |
| 15:24.59 | xbmodder_ | I'm in the last run. Must not disconnect. |
| 15:25.26 | *** join/#asterisk zpertee (n=chatzill@130.101.68.101) |
| 15:26.02 | [TK]D-Fender | my007ms: We don't see the start, nor the end of your attempt to start * manually from CLI. |
| 15:26.58 | my007ms | ok i will use asterisk -vvvdgc to start asterisk is that oki or need any extra option |
| 15:27.21 | [TK]D-Fender | my007ms: 10 x "v" please just because. |
| 15:29.29 | ^shark_ | !chanspy |
| 15:29.45 | my007ms | this first 200 line http://pastebin.ca/858538 |
| 15:31.31 | dandre | now all seems to work in both directions but I don't have audio. I must connect to internet thru a nat router and I have set localnet, externip correctly in my sip.conf |
| 15:31.51 | dandre | I also have pu nat=yes |
| 15:31.58 | dandre | what should I do? |
| 15:32.56 | [TK]D-Fender | dandre: You should pastebin your sip.conf because you shouldn't expect for a second that we trust that you did this properly :) |
| 15:33.27 | [TK]D-Fender | dandre: If you wish to save incriminating yourself, give this a read again : |
| 15:33.29 | [TK]D-Fender | ~sipnat |
| 15:33.30 | 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 |
| 15:33.37 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:34.59 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:35.13 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:35.56 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 15:36.38 | xbmodder_ | eryone's first vi session. ^C^C^X^X^X^XquitqQ!qdammit[esc]qwertyuiopasdfghjkl;:xwhat |
| 15:36.41 | my007ms | [TK]D-Fender, did u check mine ? |
| 15:36.54 | dandre | [TK]D-Fender: Our Asterisk server will also have to have ports 5060 (UDP), and the port range specified in “rtp.conf” (typically 10000-20000 UDP) forwarded to it |
| 15:36.54 | dandre | does that mean that I must doport forwarding on the router to the asterisk box for these ports? |
| 15:37.40 | [TK]D-Fender | my007ms: I don't see a failure in there... |
| 15:37.46 | hi365 | ix the iaxtel site down? |
| 15:38.27 | my007ms | [TK]D-Fender, i have no failure in all log :( chan_sip.so not work in silent |
| 15:38.46 | hi365 | nevermind my confusion |
| 15:38.57 | [TK]D-Fender | my007ms: You show partial output that shows us nothing. What do you expect here? |
| 15:39.33 | [TK]D-Fender | dande :Let me know if the big print isn't quite big enough :) |
| 15:39.54 | ^shark_ | [TK]D-Fender: have u heard of chanspy? |
| 15:39.56 | [TK]D-Fender | xbmodder_: Sounds familiar... |
| 15:40.04 | [TK]D-Fender | ^shark_: Yup. |
| 15:41.02 | ^shark_ | [TK]D-Fender: this is a patch to integrate with asterisk for music on hold right? |
| 15:41.23 | mosty | ^shark_, nope |
| 15:41.31 | [TK]D-Fender | ^shark_: no, it is a completely seperate * dialplan application that has nothing to do with it. |
| 15:41.37 | my007ms | [TK]D-Fender, the output is very bog over my buffer |
| 15:41.46 | [TK]D-Fender | my007ms: Get a bigger buffer |
| 15:41.50 | my007ms | so i can not copy the wall output |
| 15:42.10 | ^shark_ | [TK]D-Fender: i have totally failed to work on Native |
| 15:42.42 | ^shark_ | !Native |
| 15:42.50 | [TK]D-Fender | ^shark_: And the amount of help you are receiving is in direct proportion to the amount and quality of information you have provided. |
| 15:43.05 | [TK]D-Fender | ^shark_ / my007ms : thus there is balance. |
| 15:44.03 | ^shark_ | [TK]D-Fender: how do i work on Native in etch? |
| 15:44.10 | ^shark_ | [TK]D-Fender: i want to use it |
| 15:44.40 | [TK]D-Fender | my007ms: Oh and..... Parsing '/etc/asterisk/extensions_trixbox.conf': Found <------------ yo are in the wrong channel. Go read the channel /topic |
| 15:44.43 | *** join/#asterisk sargun_n810 (n=user@atarack/staff/sargun) |
| 15:45.18 | [TK]D-Fender | ^shark_: go look at your MoH file, and their associated config. |
| 15:46.04 | ^shark_ | [TK]D-Fender: ok |
| 15:48.40 | eric_hill | Would someone please fax me a beer? |
| 15:49.32 | eric_hill | <<--- id10t... *whoops* -- sorry, wrong channel. |
| 15:50.20 | [TK]D-Fender | eric_hill: 1 step down, 11 to go.... |
| 15:50.36 | eric_hill | AA is for quitters. |
| 15:51.06 | ^shark_ | [TK]D-Fender: i have checked this out music is playing but there is an echo in the sound, i dont even know what is the default music player |
| 15:51.14 | [TK]D-Fender | eric_hill: yes, thats sorta the point... |
| 15:51.23 | ^shark_ | [TK]D-Fender: i am using etch and i have not installed any music player |
| 15:51.33 | eric_hill | People ask me if I have a drinking problem. I say "No, I don't have a problem. It's easy to get alcohol..." |
| 15:52.05 | [TK]D-Fender | ^shark_: And you've shown me nothing (still), and haven't described anything about your scenario except "ETCH,ETCH,ETCH,ETCH,ETCH,ETCH,ETCH,ETCH,ETCH,ETCH" |
| 15:53.07 | [TK]D-Fender | ^shark_: Your distro does not even MATTER |
| 15:53.44 | ^shark_ | [TK]D-Fender: sorry about that but is it okey for asterisk to have echos in the the music? what can i do to change this? |
| 15:54.06 | [TK]D-Fender | ^shark_: Echo is a problem with your ENDPOINT and has nothing to do with MoH. |
| 15:54.45 | *** join/#asterisk andrewn (n=andrew@76.191.151.50) |
| 15:54.51 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 15:54.52 | ^shark_ | [TK]D-Fender: ok cos i realised this when i tried to move the headset around it played the echos in the musc |
| 15:54.56 | ^shark_ | music* |
| 15:55.44 | my007ms | [TK]D-Fender, in the end i am good in asterisk but use freepbx to make things easy |
| 15:55.59 | my007ms | [TK]D-Fender, problem stop when i add noload chan_iax.so |
| 15:56.15 | my007ms | so the probelm was in chan_iax.so |
| 15:56.31 | my007ms | i still don't know what is iax probelm |
| 15:56.34 | [TK]D-Fender | my007ms: Trixbox is NOT supported here. Please go to their channel. |
| 15:56.49 | my007ms | i am not search Trixbox issue |
| 15:58.03 | [TK]D-Fender | my007ms: You have no clue what you're doing, you aren't supposed to be messing around in those configs manually. this is your issue configuring it. |
| 15:58.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:58.52 | [TK]D-Fender | my007ms: it sets up your whole environment hence you are not in control and changes you do will likely get wiped out by the first change you commit via any of their interfaces. |
| 15:59.21 | my007ms | [TK]D-Fender, i know exact what i do i am not this mouse click :) |
| 15:59.27 | my007ms | i add noload => chan_iax2.so in module.conf |
| 15:59.37 | my007ms | to stop chan_iax2.so |
| 15:59.55 | my007ms | coz i was see asterisk stop every time in this module |
| 16:02.46 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 16:04.09 | bsdwarrior | when someone is transfered to a queue, how can I get a digit that they press while in the queue? |
| 16:04.33 | bsdwarrior | I.E. if they press 1 while in queue I want to Read(DIGIT||1) |
| 16:05.57 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 16:06.19 | [TK]D-Fender | bsdwarrior: Go read the sample queues.conf |
| 16:07.44 | [T]ank | i have set up an asterisk realtime database and have an extension working from it. however i get an error everytime i dial it. The phone rings, but the cli gives an error: http://pastebin.ca/858590 |
| 16:07.47 | [T]ank | any ideas? |
| 16:08.25 | [T]ank | if i monkey with the database connection settings i can get a failed to open database instead of failed to query. so i am pretty sure my settings for the database are accurate |
| 16:08.54 | ifnotwhynot | anyone here familiar with ESCsoftphone config for asterisk? |
| 16:10.58 | *** join/#asterisk c4t3l (n=c4t3l@74.95.210.124) |
| 16:11.06 | c4t3l | greetings all |
| 16:11.40 | itguru | Has anyone here had experiene of adding the Linksys PAP2 to thier asterisk box? |
| 16:11.41 | bsdwarrior | tkd fender can you be a bit more specific |
| 16:11.58 | *** join/#asterisk fukz (n=basti@p5B060544.dip.t-dialin.net) |
| 16:12.48 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:13.14 | [TK]D-Fender | bsdwarrior: How much more specific can I be? * came with sample configs that showcase the majority of features. GO RREAD THEM. |
| 16:13.35 | *** join/#asterisk docelmo (i=vircuser@c-68-32-135-157.hsd1.de.comcast.net) |
| 16:16.17 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 16:18.46 | [T]ank | I am looking for a way to have my phones go to a different server if asterisk is in shutdown mode. For example if for any reason I want to restart asterisk gracefully so that calls are not dropped on a restart, I want all new calls to go to a new server, how can that be done? Right now they just get a busy signal till asterisk finishes the shutdown. |
| 16:18.49 | [T]ank | is this even possible? |
| 16:18.57 | [T]ank | i am using the polycom 301 and 501 |
| 16:19.08 | [T]ank | and have the primary and secondary servers in the sip.cfg |
| 16:19.12 | [TK]D-Fender | [T]ank: setup server2 in their provisioning |
| 16:19.21 | [T]ank | so if i stop asterisk all together it works, |
| 16:19.39 | [TK]D-Fender | [T]ank: you should do that in the <reg key under "phoneXXXXXX.cfg" |
| 16:19.40 | [T]ank | however so long as asterisk is shutting down, it still tries to go to the primary server |
| 16:19.52 | [T]ank | let me try there... thanks |
| 16:20.09 | [TK]D-Fender | [T]ank: "shutting down" != "shut down" |
| 16:21.15 | [T]ank | right... shutting down means that I have issued the stop gracefully. there are still a number of calls going on that server. I want all new calls to go to secondary server |
| 16:21.51 | [TK]D-Fender | [T]ank: then you are up a creek |
| 16:22.14 | *** part/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 16:25.34 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:27.41 | *** join/#asterisk uluatu (n=deg@200.186.47.58) |
| 16:28.01 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:28.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:28.45 | *** join/#asterisk hanchi (n=chatzill@24.182.209.194) |
| 16:31.36 | itguru | One of my extensions call make calls, but it also seems to be permenetly engaged. |
| 16:32.19 | itguru | *can |
| 16:32.28 | itguru | I can't call it |
| 16:33.12 | *** join/#asterisk docelmo (n=vircuser@c-68-32-135-157.hsd1.de.comcast.net) |
| 16:41.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:41.37 | ifnotwhynot | i have never setup a iax phone before could someone please point where i can find some relavent of basic iax.conf samples, i tried gogling it but can't seem to get this phone to work any help welcome please |
| 16:41.51 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 16:41.54 | iq | Hi |
| 16:41.59 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:42.00 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 16:42.29 | grandpapadot | Hi all. Anyone using Cisco's 8.8 firmware for the 7940? Any issues? Will it allow conference calls with g729? (8.2 apparently will not) |
| 16:45.19 | yang | Could someone help me around my config, I made it preety much by the instructions from the website and when i try to call in the DID number which I ordered I get a busy tone - here is the conf http://openpaste.org/en/4718/ |
| 16:47.50 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 16:48.09 | nephfl | hello, im having some trouble with figuring out some phone wiring |
| 16:48.12 | iq | Hi |
| 16:49.42 | tzafrir | ifnotwhynot, iax.conf.sample ? |
| 16:51.12 | AndyGraybeal | russellb: are you here? |
| 16:51.37 | [TK]D-Fender | itguru: PASTEBIN comprehensive backup of your failure (CLI with debug) |
| 16:52.02 | yang | hi there tzafrir |
| 16:52.36 | AndyGraybeal | russellb: svn co http://svn.digium.com/svn/asterisk/team/russell/jack asterisk-jack <--- this didn't work when i tried it last night, did you move asterisk-jack? |
| 16:53.10 | Qwell | AndyGraybeal: it's in trunk now |
| 16:53.29 | AndyGraybeal | aaah i didn't know |
| 16:53.47 | yang | btw. my working phone extension is 600 .. I don't know if those 123 & 456 are needed in my config |
| 16:53.48 | tzafrir | yang, hi |
| 16:54.13 | AndyGraybeal | Qwell: awesome, so if i did this last night: "svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk" i have asterisk-jack already? |
| 16:54.16 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:54.26 | bsdwarrior | can someone give me an idea as to why this doesnt work? http://pastebin.com/d3b7c5b1d |
| 16:54.32 | Qwell | AndyGraybeal: no, update again this morning. you would have caught it mid-sync |
| 16:54.46 | AndyGraybeal | Qwell: alright thank you |
| 16:54.53 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:55.04 | russellb | AndyGraybeal: what Qwell said. |
| 16:55.23 | AndyGraybeal | what is the update comman? svn update ? |
| 16:55.27 | Qwell | yes |
| 16:55.38 | Qwell | though one should usually do a make update instead |
| 16:55.39 | AndyGraybeal | rad thanks Qwell and russellb |
| 16:55.54 | AndyGraybeal | ah make update, i had no idea |
| 16:55.58 | AndyGraybeal | thank you qwell |
| 16:56.15 | AndyGraybeal | russellb: do i select 'asterisk-jack' in the make menuselect menu system? |
| 16:56.47 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 16:59.21 | AndyGraybeal | make updating right now :) |
| 16:59.52 | russellb | AndyGraybeal: as long as the configure script finds libjack and the jack headers, it will build it |
| 17:00.00 | russellb | but you can check menuselect to ensure that app_jack is turned on |
| 17:00.06 | russellb | if it has XXX, that means configure didn't find the stuff |
| 17:00.08 | grandpapadot | Hey Qwell, you guys still using SCCP with your Cisco's? |
| 17:00.10 | AndyGraybeal | aah okay very fun |
| 17:00.16 | Qwell | grandpapadot: chan_skinny |
| 17:00.44 | grandpapadot | Anyone venture into the Cisco SIP 8.8 territory yet? |
| 17:00.49 | Qwell | nope |
| 17:00.57 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 17:01.59 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 17:02.58 | hanchi | any suggestions on a brand and model of ata, that is very stable and handles in band dtmf well, i've tried iaxy s101 and grand stream ht502 |
| 17:03.59 | AndyGraybeal | russellb: it looks like i don't have something called 'libjack' but i do have something called: 'libjackasyn' that probably isn't the same, correct? |
| 17:04.14 | russellb | right, it's not the same |
| 17:04.15 | russellb | what distro? |
| 17:04.19 | AndyGraybeal | opensuse |
| 17:04.24 | [TK]D-Fender | bsdwarrior: because you'll only land on that exten is you are KICKED OUT of the queue. |
| 17:04.27 | russellb | ugh, theni have no idea |
| 17:04.37 | AndyGraybeal | okay thank you, that's fine i'll poke around |
| 17:04.37 | [TK]D-Fender | bsdwarrior: And you have not shown your queue CONFIGS at all. |
| 17:04.42 | russellb | it's literally libjack ... |
| 17:04.50 | russellb | might be packaged as jackd ... |
| 17:05.52 | AndyGraybeal | ah yea, just under "jack" is where it is |
| 17:05.57 | Qwell | silly suse |
| 17:06.37 | AndyGraybeal | what distro do you use russellb? |
| 17:06.45 | yang | I am just having problem with this voip Provider, the other one works well for me... |
| 17:06.49 | russellb | AndyGraybeal: ubuntu |
| 17:06.59 | AndyGraybeal | cool |
| 17:07.10 | Qwell | AndyGraybeal: pretty much all of the devs at Digium use some form of Debian.. |
| 17:07.25 | AndyGraybeal | okay, thank you Qwell |
| 17:07.48 | Qwell | you know...I'm a little surprised, actually. (mini-rant coming) |
| 17:08.18 | Qwell | before I worked here, I tried Debian once. I hated it, because of dselect... that was a LONG time ago.. I never looked at it again because I could never get past that little thing |
| 17:09.22 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-222-225.dsl.irvnca.pacbell.net) |
| 17:10.32 | bsdwarrior | How do I set a timeout so that if a user doesnt press anything I can run a goto ? |
| 17:10.41 | AndyGraybeal | i did a make update, and it stopped with saying: svn: Failed to add file 'include/asterisk/version.h' : object ofthe same name already exists |
| 17:10.48 | AndyGraybeal | does that error matter? |
| 17:11.00 | *** join/#asterisk gardo (n=gardo@121.97.142.167) |
| 17:11.03 | Qwell | rm include/asterisk/version.h, then go again |
| 17:11.07 | AndyGraybeal | rad thanks |
| 17:12.04 | bsdwarrior | ReponseTimeout ? |
| 17:12.30 | brodiem | Does anyone know if the new app_queue commit for state_interface requires the use of using "/n" or without on local channels to work? |
| 17:12.40 | Qwell | putnopvut: ^^ |
| 17:12.56 | grandpapadot | ooooooh, Cisco SIP 8.8 has some new stuff ... |
| 17:14.13 | brodiem | more specifically, I found out that switching to not using /n (in order for atxfer to free up the channel) resulted in none of the recordings working I guess because of the bridge difference, and want to be able to use atxfer + recordings |
| 17:15.24 | tzafrir | Qwell, dselect? Don't use it. Just use aptitude instead |
| 17:15.30 | *** join/#asterisk CVirus (n=GoD@196.205.193.51) |
| 17:15.55 | yang | Could someone help me around my config, I made it preety much by the instructions from the website and when i try to call in the DID number which I ordered I get a busy tone - here is the conf http://openpaste.org/en/4718/ , my working phone extension is 600 |
| 17:16.00 | Qwell | tzafrir: yeah, now you tell me |
| 17:16.09 | Qwell | tzafrir: this was during install, and it was the "recommended" option |
| 17:16.17 | Qwell | again, it was a long long time ago... |
| 17:16.54 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:17.04 | putnopvut | brodiem: the state_interface change was designed so that you would not have to use /n. |
| 17:17.14 | Kobaz | what's a good sip bulk call generator app |
| 17:18.53 | brodiem | putnopvut: suppose it makes sense since the state is monitored correctly with /n in the first place.. using Monitor though fails to create any recordings without /n =/ |
| 17:19.27 | AndyGraybeal | must be "app_jack", it it's selected in the menu :) |
| 17:19.27 | putnopvut | brodiem: I believe there is a bug open regarding that right now. Let me find the issue number. |
| 17:19.28 | brodiem | I think because it's monitoring the wrong bridge |
| 17:19.31 | brodiem | cool |
| 17:20.10 | putnopvut | brodiem: http://bugs.digium.com/view.php?id=11741 |
| 17:20.19 | brodiem | haven't tried MixMonitor but in order to not use up any g729 lics I had to stick with Monitor |
| 17:20.21 | putnopvut | Is that what's happening? |
| 17:20.49 | putnopvut | (I know you said atxfer, but it probably would have the same effect) |
| 17:21.37 | brodiem | putnopvut: actually it's _all_ recordings from a queue that it's happening with |
| 17:21.58 | brodiem | putnopvut: it creates an audio file, but is only about ~300b each time with no audio in it of course |
| 17:22.13 | putnopvut | Even when no transfer is involved? |
| 17:22.19 | brodiem | yeah |
| 17:22.36 | *** join/#asterisk jdunck (n=jdunck@74.7.153.189) |
| 17:22.38 | brodiem | putnopvut: that's using Monitor() and MONITOR_OPTIONS=b |
| 17:22.45 | jdunck | anyone have a pointer to a sipconnect howto? |
| 17:23.07 | *** join/#asterisk tuxd00d (n=Tuxd00d@128.187.129.147) |
| 17:23.16 | [TK]D-Fender | bsdwarrior: Your timeout question has nothing to do with DTMF queue options... |
| 17:23.19 | putnopvut | brodiem: All right. And this problem started showing itself after you changed over to the state_interface-related change? |
| 17:23.52 | brodiem | putnopvut: what I do is record each stream as g729, then I have a wrapper around soxmix to use CLI file convert to create a WAV, then call the regular soxmix to produce the resulting merged Wav |
| 17:24.39 | putnopvut | I'll set up a test with a queue involving monitoring and see if the problem happens for me too. |
| 17:24.47 | brodiem | putnopvut: I haven't tried state_interface yet actually, I'm using 1.4.14 right now. I just found out this was happening when I needed to pull a recording and found they were all just empty audio files |
| 17:24.54 | brodiem | cool, thanks |
| 17:25.07 | putnopvut | Oh, so this has nothing to do with the state_interface stuff? |
| 17:25.26 | brodiem | putnopvut: lol no sorry |
| 17:26.06 | putnopvut | Okay, so it's just a problem with Monitoring using local channels then. |
| 17:26.15 | brodiem | I was just asking if I were to switch to state_interface if it may correct the problem of atxfer using /n so that I could use recordings again |
| 17:26.37 | putnopvut | brodiem: my suggestion would be to try with a newer 1.4 first to see if it's been fixed since then. |
| 17:26.56 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:26.57 | putnopvut | Then, if it works there, try moving to the state_interface implementation. If it's not working there, file a bug. |
| 17:27.03 | brodiem | yeah, I'll do that |
| 17:27.05 | brodiem | thanks |
| 17:27.39 | brodiem | I'll try mixmonitor first also, just avoided it due to g729 usage |
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| 17:27.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:28.47 | ifnotwhynot | i have never setup a iax phone before could someone please point where i can find some relavent of basic iax.conf samples, i tried gogling it but can't seem to get this phone to work any help welcome please |
| 17:29.22 | *** join/#asterisk sudhir492 (n=sudhir@adsl-154-183-50.mco.bellsouth.net) |
| 17:29.25 | sudhir492 | hi all |
| 17:30.06 | *** part/#asterisk magumbade (n=magumbad@p5497EF11.dip.t-dialin.net) |
| 17:31.00 | puzzled | hi |
| 17:31.07 | puzzled | tzafrir: ping |
| 17:31.35 | *** join/#asterisk uluatu (n=deg@200.186.47.58) |
| 17:32.08 | tzafrir | pong |
| 17:32.42 | puzzled | tzafrir: hi, have you by any chance built zaptel-1.4.8 with the latest oslec? |
| 17:34.03 | tzafrir | puzzled, not yet. |
| 17:34.36 | puzzled | tzafrir: compilation fails due to some changes (I think) in zaptel and my non-C-foo isn't helping fixing it |
| 17:34.42 | tzafrir | puzzled, tzanger reported an oops with latest oslec due to locking changes |
| 17:34.48 | puzzled | ugh |
| 17:34.53 | tzafrir | any trace? |
| 17:34.54 | puzzled | better stay away from it then |
| 17:35.15 | tzafrir | anyway, can you pastebinb a trace? |
| 17:35.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 17:35.53 | puzzled | tzafrir: http://pastebin.ca/858709 |
| 17:36.53 | puzzled | tzafrir: echo_can_create takes 3 args while oslec_echo_can_create is defined to have only 2 |
| 17:38.10 | *** join/#asterisk CrashSys (n=kumba@t1.databalance.com) |
| 17:38.18 | CrashSys | Anyone remember what PRI code 90 meant? |
| 17:39.25 | puzzled | CrashSys: non existant CUG (dunno what that is). see here: http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 17:40.56 | grandpapadot | In Cisco 7940 configs, does anyone have experience with the "preferred_codec" setting? I have it set to g729a, but when attempting a conference call using the phones conference button when in a call, asterisk just says no codecs as the phone is trying to use g711. |
| 17:41.05 | grandpapadot | Both 8.2 and 8.8 SIP firmware |
| 17:41.12 | alrs | CrashSys: the magic google term for PRI errors is "ISDN cause codes' |
| 17:41.23 | CrashSys | Ahhhh |
| 17:41.24 | grandpapadot | BTW: 8.8 has a broken NAT stack like the 7941's. |
| 17:41.46 | Qwell | grandpapadot: and like every other version... |
| 17:41.52 | puzzled | lovely. seems Cisco always breaks stuff with newer versions |
| 17:42.04 | grandpapadot | 8.2 seems to work great w/NAT but that's the only luck I've had |
| 17:42.12 | Qwell | puzzled: no, they just convince people that the problem didn't exist in old version ;) |
| 17:42.30 | puzzled | heheh |
| 17:45.08 | nephfl | i have a very silly question, how do i connect multiple drops to one connection on a 66 panel? Normally I can just run extra leads from the nut connecting the incoming line, but in this case it is only a pair coming in landed to a 66 panel |
| 17:45.11 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:45.17 | CrashSys | Maybe it's cause i'm not setting the callerid from the dialing station to the PSTN # |
| 17:46.04 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:49.54 | nephfl | anyone? |
| 17:52.03 | RoyK | someone |
| 17:53.37 | Havokmon | uhh there are two sets of two pins .. just run wire down the 2nd set, and use the first to connect back to your device. |
| 17:53.50 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 17:54.20 | nephfl | i need to connect 4 extensions to 1 line...so i have one pair that needs to connect to 4 pairs... |
| 17:54.34 | AndyGraybeal | twice in a row so far with asterisk trunk.. i do the 'make clean', 'make update', './configure', 'make menuselect', 'make install' and it fails at the downloading of the asterisk-core-sounds-en-gsm-1.4.8.tar.gz, it says "gzip: stdin: unexpected end of file" and tar goes on to say more or less the same thing; any help? it's happened twice... the first time it happened, i figured it was just a downloading problem, but i don't know what to |
| 17:54.34 | AndyGraybeal | think now, any suggestions would be helpful. |
| 17:54.45 | Havokmon | nephfl: analog - right? |
| 17:54.45 | nephfl | and there doesnt seem to be a distribution block that will let me patch 4 to 1 |
| 17:54.48 | nephfl | yes |
| 17:55.13 | nephfl | but i might just not know how to bridge it correctly..im very new to analog phone wiring |
| 17:55.19 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 17:55.29 | Havokmon | you have a punch tool? |
| 17:56.04 | nephfl | there are two 66 blocks and a 66 surge protector block...i have 2 lines coming into the surge protector 66 block and out to another 66 block...but cant you only connect 66 blocks 1to1 |
| 17:56.07 | nephfl | yes |
| 17:56.20 | Havokmon | the 66 block is the distribution block. the pins are connected side by side, but underneath the plastic |
| 17:56.32 | Havokmon | if you want to connect multiple pins together |
| 17:56.36 | Havokmon | just wire them together |
| 17:57.07 | nephfl | so i can punch down multiple connections on one spot? |
| 17:57.31 | tzafrir | puzzled, right. echo canceller options |
| 17:57.33 | Havokmon | well - you could - but just use more of the pins. |
| 17:57.55 | outtolunc | never punch down more than 1 wire per clip |
| 17:57.56 | Havokmon | you have 4 pins in a row |
| 17:58.02 | outtolunc | sheesh |
| 17:58.14 | Havokmon | the left middle is connected to the left outside, right middle right outside |
| 17:58.37 | Havokmon | you run your 'drop' wire (extensions) to the outside pins |
| 17:58.51 | Havokmon | then connect your multiples with the inside pins - going downward |
| 17:59.05 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:59.27 | Havokmon | outtolunc: lol, yeah you could just twist the wires together too :) |
| 17:59.39 | nephfl | im going to look at a photo and try to make sense of what you are saying, because i thought they were only connected 1 to 1, I feel slow because I don't know this stuff..lol |
| 17:59.41 | outtolunc | hehe |
| 18:00.00 | Havokmon | nephfl: I don't think anyone learned how to punch down blocks on irc |
| 18:00.09 | Havokmon | it's more of a watch and do it thing |
| 18:01.01 | outtolunc | i just see it too often, and the second wire is ready to fall out of the clip (or has, hense the to me) |
| 18:01.09 | AndyGraybeal | does someone think the sound file in trunk right now could be corrupt, or there is something wrong with what i'm doing? |
| 18:01.51 | file | AndyGraybeal: it's not, the download was stopped before it was completed... and there is no check to make sure it is the complete file... delete the file and it'll redownload and go fine |
| 18:02.17 | AndyGraybeal | it's happened twice that's the only reason i was asking. i'll try again |
| 18:02.20 | AndyGraybeal | thank you |
| 18:04.54 | AndyGraybeal | dialup shouldn't be a problem right? |
| 18:06.00 | nephfl | http://www.homephonewiring.com/add-line3.html |
| 18:06.16 | nephfl | that seems to be saying to punch 2 wires to 1 pin to jumper it to the next connection |
| 18:06.28 | nephfl | is that right? |
| 18:06.37 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 18:08.21 | outtolunc | no, in that diagram each 'pair' of pins (2 left, and 2 right) are physically connected to eachother |
| 18:08.35 | *** part/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 18:09.00 | outtolunc | so punching down on the LL and LR connect to eachother... then punchdown on RL and RR is together |
| 18:09.26 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 18:09.39 | outtolunc | if you wanted LL LR AND RL RR to ALL be connected you need a bridging clip connecting LR and RL |
| 18:09.44 | nephfl | so is he using a punchdown tool that doesnt automatically cut to pat the line to the next pin? |
| 18:09.51 | nephfl | patch not pat |
| 18:10.38 | outtolunc | not in that example as it is continuing to another pin set 'daisy chaining' |
| 18:11.14 | CrashSys | in order to dial-out a PRI, do I just issue the Dial cmd? or do I need to tell the PRI what DID to dial-out on? |
| 18:11.17 | nephfl | i see...i was assuming it was 2 connections because my punchdown tool has a cutting edge...so i should just use one that doesnt and that solves all my problems |
| 18:11.26 | nephfl | i could just daisy chain them |
| 18:12.45 | nephfl | is that technically the correct way to do it? |
| 18:13.32 | outtolunc | not in my book <G> |
| 18:13.41 | nephfl | how would you do it? |
| 18:13.51 | nephfl | put in a distrabution block? |
| 18:14.01 | outtolunc | i'd rather mount a wall mount, and wire to it and 'screw down' multiple pairs |
| 18:14.30 | outtolunc | then it is obvious to the next installer what you are doing |
| 18:15.13 | nephfl | I see...well, it looks like the reason i am dealing with this is that originally each connection had its own line and now i am switching so to only 2 lines...so ill probably go ahead and do that |
| 18:19.39 | AndyGraybeal | file: rad it downloaded! |
| 18:19.45 | AndyGraybeal | file: thank you :) |
| 18:20.06 | sudhir492 | how to setup userfield in CDR? |
| 18:20.42 | sudhir492 | I tried Set(CDR(userfield)=foo) but it does not work. I am using 1.2.12.1 |
| 18:24.46 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:31.01 | aydiosmio | I'm executing a Dial() with m() option, m() executes an AGI. How do I set a variable from this AGI in the calling context? inheritance doesn't seemt o want to cross between them, and seeting a global variable would clobber simultaneous calls |
| 18:31.35 | [TK]D-Fender | aydiosmio: use AstDB |
| 18:31.37 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 18:31.56 | aydiosmio | word. |
| 18:32.08 | aydiosmio | i.e. thank you |
| 18:32.17 | [TK]D-Fender | aydiosmio: pass the originting UNIQUEID as a parm to the macro and use that as a DB key |
| 18:32.25 | [TK]D-Fender | (part of) |
| 18:33.11 | [TK]D-Fender | aydiosmio: I might suggest adding an "expiration" check as well so you can do cleanups of dead keys. |
| 18:34.14 | aydiosmio | yeah sounds good |
| 18:37.25 | aydiosmio | [TK]D-Fender: is it possible to change the DIALSTATUS for the Dial from within the m()? If the call answered but the channels not bridged I'd like the Dial to return a different status |
| 18:37.49 | [TK]D-Fender | aydiosmio: based on your macro exit code it hink it might change the status |
| 18:37.59 | aydiosmio | I'll check on that |
| 18:38.06 | *** join/#asterisk rdsousa (n=chatzill@213-205-87-88.net.novis.pt) |
| 18:38.07 | [TK]D-Fender | aydiosmio: But for sure its read-only |
| 18:39.01 | aydiosmio | ok |
| 18:39.24 | rdsousa | hello i need a solution for a SBC |
| 18:39.58 | rdsousa | is there any way to make a SBC with asterisk? |
| 18:40.57 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:40.57 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:40.59 | [TK]D-Fender | rdsousa: No. * is a B2BUA |
| 18:41.57 | rdsousa | in your opinion what's the best solution? |
| 18:42.47 | [TK]D-Fender | rdsousa: www.google.com <- |
| 18:43.15 | rdsousa | lol |
| 18:43.17 | rdsousa | ok thanks |
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| 18:43.24 | *** part/#asterisk Duke_Fluke (n=duke@S01060050046c6c84.ed.shawcable.net) |
| 18:53.05 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 18:57.32 | *** join/#asterisk Victor_Yure (n=Victor_Y@postfix.tradein.com.br) |
| 18:57.38 | grandpapadot | Is there a way to hangup a channel from the asterisk 1.2 CLI? I have a bunch of 'hung' calls for some reason that all happened at the same time today. |
| 18:57.56 | grandpapadot | soft hangup? |
| 18:58.29 | RoyK | try soft hangup |
| 18:58.38 | RoyK | if that fails, killall -9 asterisk :P |
| 18:58.41 | grandpapadot | Got it, thanks RoyK. |
| 18:58.44 | grandpapadot | It works. |
| 18:59.07 | RoyK | but you'd better report bugs like that, if it is indeed a bug |
| 18:59.11 | RoyK | channels shouldn't hang |
| 19:01.34 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:02.24 | JunK-Y | no reports like that for 1.2 |
| 19:02.41 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 19:03.27 | aydiosmio | alright so, the Dial's m() status doesn't affect the DIALSTATUS, but the macro does not continue if the dialed channel is hung up, so I'll work from there. |
| 19:04.18 | *** join/#asterisk AndyGraybeal (n=andy@node49.32.251.72.1dial.com) |
| 19:04.31 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
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| 19:07.43 | hesco | I'm using an iax2 account from diamondcard.us for my outgoing. My tests are working fine. Now that I'm thinking of deployment in production, I'm going to need to accomodate multiple simultaneous outgoing phone lines. Can anyone advise how to do that, please? Can I set up multiple iax2 accounts in iax2.conf? How do I have asterisk manage the choice of an appropriate line for the next outgoing call? Is there a way to monitor whether a |
| 19:07.43 | hesco | still busy or is now available for the next call? |
| 19:08.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:08.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:09.30 | keith4 | hesco: you just use a call group, and then there are different "algorithms" for choosing an outgoing line |
| 19:10.17 | keith4 | er, trunk group |
| 19:11.17 | hesco | where are trunk groups documented? I didn't see anything about that in the pdf I read, or perhaps I missed that. |
| 19:12.43 | drmessano-LT | Anyone know of any good trade publications that cover VoIP ... and not just "M$ VoIP", but feature Asterisk solutions? |
| 19:12.51 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 19:13.40 | puzzled | drmessano: VON Magazine |
| 19:13.54 | drmessano-LT | Is it paper? |
| 19:15.23 | RoyK | what's paper? |
| 19:16.00 | hesco | On page 82, it says: The [trunkgroups] section is for NFAS and GR-303 connections,and it won’t be discussed |
| 19:16.00 | hesco | in this book. If you require this type of functionality,see the zapata.conf.sample |
| 19:16.00 | hesco | file for more information. |
| 19:16.06 | hesco | . |
| 19:16.22 | keith4 | hesco: IAX supports trunking though, so it should be able to support multiple voice streams |
| 19:16.32 | keith4 | i think that depends on your provider |
| 19:16.57 | [TK]D-Fender | keith4: IAX trunking has nothing to do with NFAS |
| 19:17.06 | keith4 | NFAS? |
| 19:17.18 | [TK]D-Fender | keith4: What he was asking about |
| 19:17.23 | JunK-Y | drmessano-LT: yes von mag is paper. |
| 19:17.26 | hesco | NFAS was mentioned in the pdf documentation from OReilly's |
| 19:17.46 | hesco | But that reference was to zapata.conf. |
| 19:17.48 | [TK]D-Fender | keith4: NFAS is using a single D-Cahn on PRI to cover multiple PRI interfaces. |
| 19:18.05 | hesco | I'm using iax2, not a hardware connection to tpc |
| 19:18.16 | [TK]D-Fender | hesco: as well it should. this is between zaptel.conf and zapata.conf |
| 19:18.35 | keith4 | [TK]D-Fender: i'm reading about NFAS... but i think he's strictly talking about iax |
| 19:18.37 | [TK]D-Fender | hesco: well then stop looking at NFAS if you aren't looking at PRI |
| 19:18.54 | [TK]D-Fender | hesco: So what are you actually trying to do then? |
| 19:19.24 | hesco | I was looking at trunk group and the only reference I found to it was in the zapata.conf chapter. |
| 19:19.25 | keith4 | i think he wants the IAX equivalent of a zap group |
| 19:19.36 | hesco | I'm using an iax2 account from diamondcard.us for my outgoing. My tests are working fine. Now that I'm thinking of deployment in production, I'm going to need to accomodate multiple simultaneous outgoing phone lines. Can anyone advise how to do that, please? Can I set up multiple iax2 accounts in iax2.conf? How do I have asterisk manage the choice of an appropriate line for the next outgoing call? Is there a way to monitor whether a |
| 19:19.36 | hesco | still busy or is now available for the next call? |
| 19:19.51 | keith4 | iax just kinda does that by itself, i thought |
| 19:20.36 | keith4 | hesco: i don't know anything about diamondcard.us, but i believe the number of simultaneous calls you can have is determined by your provider |
| 19:20.49 | hesco | when I initiate a second call, while the first is still going, I get a recorded message saying that account is already in use, my call file is disposed of and I'm not sure which number did not get called. |
| 19:21.01 | [TK]D-Fender | hesco: there is no such thing as "lines" in VoIP |
| 19:21.24 | [TK]D-Fender | hesco: If you want multiple simultaneous channells, thats just something you pay for from your provider and requires NO conficgurations |
| 19:21.34 | keith4 | hesco: http://wiki.diamondcard.us/podwiki?page=SimCalls |
| 19:21.39 | puzzled | drmessano-LT: yes it's a magazine |
| 19:22.16 | keith4 | hesco: did you even look at their FAQ page? |
| 19:22.27 | [TK]D-Fender | puzzled: Those bastards found me and have been spamming me on e-mail AND snail mail... |
| 19:22.45 | puzzled | [TK]D-Fender: same here. the amount of stuff they send out is pretty amazing |
| 19:22.49 | puzzled | but I like the magazine |
| 19:23.14 | [TK]D-Fender | puzzled: it isn't "news" if its on paper :p |
| 19:23.23 | puzzled | [TK]D-Fender: and it must cost them a bundle to send all that paper from the US to .nl |
| 19:24.25 | hesco | keith4: I had read through their materials, but apparently had missed this. Thanks for the lead. I'm trying that now. |
| 19:25.47 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 19:25.50 | drmessano-LT | lol |
| 19:25.57 | drmessano-LT | Ok, thanks I am gonna sign up |
| 19:26.02 | Cherebrum | Ha ha ha! this reminds me of Asterisk! http://img209.imageshack.us/img209/5781/deadlocknajkcomafarialibh3.jpg |
| 19:26.03 | drmessano-LT | and FWIW, they all do |
| 19:26.12 | drmessano-LT | Network World sends me 5 emails a day |
| 19:26.53 | CrashSys | Anyone got any suggestions what is popping this up: "Unable to handle return result on switchtype 1!"? |
| 19:27.08 | drmessano-LT | Cherebrum: Is that a re-enactment of transcoding on a P-66 with 32MB Ram? |
| 19:27.42 | CrashSys | Looks like asterisk locks |
| 19:27.54 | Cherebrum | I think it's just normal locking operation |
| 19:30.47 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 19:32.24 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 19:45.28 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:45.55 | Yourname`` | Holy crap confusion! |
| 19:46.03 | Yourname`` | Termination = Outbound |
| 19:46.08 | Yourname`` | Origination = Inbound |
| 19:46.10 | Yourname`` | Correcto? |
| 19:46.34 | [TK]D-Fender | Yourname``: yes |
| 19:46.52 | Yourname`` | Well, hello there [TK]D-Fender |
| 19:46.59 | Yourname`` | Great, thank you. |
| 19:48.16 | *** join/#asterisk `paul (n=aldee@125.252.68.68) |
| 19:51.29 | `paul | what seems to be the problem when sometimes the connection from my soft phone (ekiga) to my server times out sometimes... and when i register again it fails |
| 19:54.11 | Uni | [1;2C/4 |
| 19:54.15 | Uni | w00t! |
| 19:55.32 | drmessano-LT | You might have a bad OC30 |
| 19:57.11 | brodiem | putnopvut: if you're there, FYI that recording problem I mentioned is isolated to Monitor(), MixMonitor() works correctly (but replicated the same Monitor prob on a separate installation) |
| 19:57.30 | `paul | OC30? |
| 19:59.31 | putnopvut | brodiem: interesting. Which version were you testing with? |
| 19:59.49 | brodiem | putnopvut: I was using 1.4.14, but also replicated on a new install of 1.4.17 |
| 20:00.25 | AndyGraybeal | okay, so i grabbed trunk from svn last night, and did a 'make update' today... so far everything appears to configure and compile fine, but when i run 'asterisk -cvvv' it runs down a bit and then segfaults; any ideas on how i can get out of this mess i made? |
| 20:00.29 | AdamWest | brodiem: is there already an open bug? It's better for comments to go into the bug tracker so they don't get lost or forgotten |
| 20:00.39 | AndyGraybeal | i don't think i did anything wrong, but i'm not sure how to troubleshoot either |
| 20:00.42 | putnopvut | brodiem: Sounds like it's a current bug then. I'd file a bug. |
| 20:01.00 | brodiem | AdamWest: I have no idea, lol, I'll see if I can find anything |
| 20:01.06 | AdamWest | AndyGraybeal: back out to a different version that works |
| 20:01.13 | AdamWest | brodiem: sounds good -- bugs.digium.com fyi |
| 20:01.24 | brodiem | yep |
| 20:01.32 | AdamWest | and now I'm off to get ready for dinner |
| 20:02.17 | AndyGraybeal | <PROTECTED> |
| 20:02.25 | AndyGraybeal | AdamWest: any other things i can try? |
| 20:03.34 | AdamWest | AndyGraybeal: try: make distclean && ./configure && make install |
| 20:03.48 | AndyGraybeal | rad thank you AdamWest |
| 20:03.48 | AdamWest | <-- lmadsen :) |
| 20:03.52 | AdamWest | gone for dinner! |
| 20:03.53 | AndyGraybeal | aah mr lief |
| 20:04.05 | AndyGraybeal | is that batman? |
| 20:04.07 | AndyGraybeal | adam west? |
| 20:04.27 | AdamWest | wait.... Mr. Leif... did you call me that when we met? Were you in one of the training classes I was at? |
| 20:04.38 | AndyGraybeal | oh no no no |
| 20:04.45 | putnopvut | AdamWest: you're lmadsen? |
| 20:04.54 | AdamWest | oh ok :) I only know one person who called me that before, and it sounded familiar :) |
| 20:05.03 | AdamWest | putnopvut: only sometimes |
| 20:05.14 | AndyGraybeal | i'm some southeastern ohio hick that's trying to figure this crap out |
| 20:05.44 | putnopvut | AdamWest: that's very confusing :) |
| 20:07.38 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 20:07.47 | javar | hello |
| 20:08.06 | *** join/#asterisk ik_5 (n=ik@85.64.203.142.dynamic.barak-online.net) |
| 20:08.08 | AdamWest | putnopvut: that's life :) |
| 20:08.23 | drmessano-LT | BATMAN! |
| 20:08.29 | javar | somebody works with asterisk 1.4.x and fax? |
| 20:11.21 | javar | :( |
| 20:11.30 | lotho | javar: yes, but at the moment only to receive faxes |
| 20:11.40 | ik_5 | hello, I have a wierd problem with manager... i set up a user that seems to be ok, but when i try to login, it returned Auhentication Failed, what am I missing ? (http://pastebin.com/d3564cbfa) |
| 20:11.48 | javar | lotho: yes i need the same |
| 20:12.06 | javar | i found a tutorial to install spanDSP.. |
| 20:12.14 | lotho | then rxfax is what your are searching |
| 20:12.21 | javar | yes |
| 20:12.47 | javar | the problem is that rxfax.c does not in the spanDSP site |
| 20:13.00 | javar | you know if that is not necessary? |
| 20:13.49 | *** join/#asterisk Y0da^ (n=Bunny@70.159.118.70) |
| 20:13.52 | lotho | you need the rxfax application |
| 20:13.56 | *** join/#asterisk Grnd-Wire (n=grundofw@65.101.128.1) |
| 20:13.56 | lotho | https://sourceforge.net/projects/agx-ast-addons/ |
| 20:14.07 | lotho | there you can get it ;) |
| 20:14.11 | Grnd-Wire | greetings all! Does anyone have any experience with the Asterisk Appliance? |
| 20:14.24 | lotho | and spandsp ofcourse |
| 20:15.04 | javar | ok |
| 20:15.31 | javar | i compiled spanDSP, now... Now how compile the rxfax app? |
| 20:15.34 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 20:16.07 | bsdwarrior | is it possible to add a caller back to a queue with the same prority ? |
| 20:16.27 | lotho | you need to download the agx-ast-addons package and compile it |
| 20:16.49 | lotho | then you have rxfax and txfax |
| 20:17.19 | brodiem | putnopvut: ahh are you serious: "Removed the monitor-join option. If one wishes to mix audio, they should instead use monitor-type=mixmonitor" |
| 20:17.28 | Ritzerisk | would i have to create some sort of script if i needed to input 6 digits via dtmf and then use those 6 digits to save as the filename... like it sends 432322 at 430pm.tif but the person that is faxing faxes to the numberPPPPP432322# |
| 20:18.19 | javar | lotho, i should do that before compile asterisk, right? |
| 20:18.25 | putnopvut | brodiem: yeah, you're not the first person to complain. That decision could be reversed. |
| 20:18.25 | Ritzerisk | where P is the pause to wait for the ivr prompt them forward it to like a hunt group with my iaxmodem |
| 20:19.38 | putnopvut | brodiem: when you have the problem with calls not getting recorded, are you using /n on the local channel? |
| 20:20.20 | brodiem | putnopvut: lol, the reason I use it is because I use g729 end-to-end. Using MixMonitor causes licenses to be consumed (I guess in converting to slin?), but with Monitor I can record each stream separately w/o utilizing licenses. Then, monitor-join calls soxmix (which I have as a wrapper around it to convert the streams to Wav before mixing so that licenses are consumed only during the period of conversion) |
| 20:20.30 | bsdwarrior | is there any way to put a user back into a queue in the same spot as they were ? |
| 20:20.42 | *** join/#asterisk mkl1525 (n=qwertz@212.204.47.147) |
| 20:20.57 | brodiem | putnopvut: Using /n, recordings are done correctly, but removing /n causes it to fail (it creates the audio file but is empty) |
| 20:21.43 | putnopvut | Hmmm, I just ran a test with the latest SVN checkout of 1.4 and used a non-/n local channel for a queue member and the recording was fine. |
| 20:21.53 | putnopvut | Let me try with 1.4.17. It may have been fixed since then. |
| 20:22.15 | brodiem | putnopvut: using monitor-type=monitor or mixmonitor? |
| 20:22.27 | putnopvut | monitor. |
| 20:22.43 | putnopvut | It produced two files that I played back and they sounded fine. |
| 20:23.23 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-76-246-191-68.dsl.lsan03.sbcglobal.net) |
| 20:23.23 | brodiem | hm |
| 20:24.28 | putnopvut | I'll try with 1.4.17. |
| 20:24.38 | Kobaz | [Jan 17 10:25:43] WARNING[3751]: chan_sip.c:16948 reload_config: Unable to get own IP address, SIP disabled |
| 20:24.42 | Kobaz | how would i fix that? |
| 20:24.53 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 20:25.00 | javar | lotho: still there? |
| 20:25.49 | lotho | javar: yes, sorry |
| 20:26.00 | javar | lotho, i should do that before compile asterisk, right? |
| 20:26.22 | lotho | no, do it after you have compiled asterisk |
| 20:26.33 | javar | Ah |
| 20:26.45 | javar | ok, thanks |
| 20:26.47 | lotho | i do it after and it works ;) |
| 20:26.56 | javar | i'll try it |
| 20:27.30 | ik_5 | any idea why I get on telnet: Authentication Failed over http://pastebin.com/d3564cbfa ? |
| 20:27.30 | Nugget | telnet is eeeeeeevil! |
| 20:28.07 | Grnd-Wire | Does anyone have any experience with the Asterisk Appliance? |
| 20:28.41 | putnopvut | ik_5: do you have enabled=yes in the general section of manager.conf? |
| 20:28.55 | lotho | Nugget: telnet is great, when you are the listener ;) |
| 20:28.56 | ik_5 | putnopvut, yes i do |
| 20:29.04 | putnopvut | Ah, nevermind then. |
| 20:30.43 | brodiem | putnopvut: would you be able to show me the queue definition you used and I'll use the same? |
| 20:31.35 | putnopvut | Okay hold on, I'll pastebin it. |
| 20:32.23 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 20:32.26 | brodiem | thanks |
| 20:34.04 | putnopvut | brodiem: http://pastebin.ca/858960 |
| 20:34.15 | brodiem | thanks |
| 20:34.34 | putnopvut | Your members and queue name will probably be different, but aside from that you could probably plug that in and see how it works. |
| 20:34.35 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 20:35.23 | Kobaz | apparently there's a bug in asterisk, if you have no gateway defined, sip is broken |
| 20:35.59 | mkl1525 | Hi, I'd like to give our queue agents an voice menu after the caller has finished. Is there any way to do this? h extension seems not to work |
| 20:40.19 | Nugget | lotho: sure, but with telnet *everyone* is listening. ;) |
| 20:40.28 | grayhame | has anyone had any issues with Monitor in/out files not getting mixed at the end of a call? |
| 20:40.47 | CrashSys | What's the variable that has the peername that's dialing? IE sip/101 calls sip/102, what variable gives me sip/101? |
| 20:40.57 | CrashSys | ${DIALEDPEER}? |
| 20:42.53 | *** join/#asterisk awormus (n=Aaron@193.138.164.61) |
| 20:43.08 | *** join/#asterisk greekguy8888 (n=alex@c-76-118-201-12.hsd1.ma.comcast.net) |
| 20:43.24 | greekguy8888 | does anyone use realtime voicemail with mysql? |
| 20:43.31 | CrashSys | course, if my call is originating then callerid(num) would give me the peer name... |
| 20:43.35 | nhuisman_work | could you guys take a look at http://pastebin.com/m77874226 and tel me why my phone won't register? |
| 20:43.44 | _ShrikE | greekguy8888: yes |
| 20:44.20 | greekguy8888 | shrike, do u have issues with the directory feature? mine doesn't pickup any users in the realtime vm db |
| 20:44.42 | greekguy8888 | i see it check voicemail.conf when directory is accessed |
| 20:44.56 | nhuisman_work | i keep getting |
| 20:44.56 | nhuisman_work | Jan 17 15:36:05 NOTICE[6553]: chan_sip.c:11379 handle_request_register: Registration from '<sip:1000@128.171.77.21>' failed for '128.171.77.50' - Wrong password |
| 20:44.57 | nhuisman_work | errors |
| 20:44.57 | greekguy8888 | but no matter how many correct combos i enter, it says noone in the directory |
| 20:46.16 | _ShrikE | greekguy8888: what version of asterisk are you using, I believe that has worked for quite some time. |
| 20:46.30 | nhuisman_work | anyone? |
| 20:47.35 | greekguy8888 | <PROTECTED> |
| 20:47.54 | _ShrikE | look at bug 2475 |
| 20:49.08 | nhuisman_work | ok seriously wtf |
| 20:49.11 | *** join/#asterisk Schreiber1337 (n=cee4b465@gateway/web/cgi-irc/ircatwork.com/x-8149e3c84e44e3b6) |
| 20:49.17 | nhuisman_work | i set the password on my sip.conf to hilophones |
| 20:49.18 | nhuisman_work | no dice |
| 20:49.20 | nhuisman_work | i set it to testpass |
| 20:49.21 | nhuisman_work | dice |
| 20:49.24 | nhuisman_work | how in the hell |
| 20:49.43 | nhuisman_work | how do I do a sip trace? |
| 20:50.02 | awormus | I am setting up a PBX in our office, we are having a 3 mbit Integrated PRI coming in with a cisco QoS switch at the end of it. I don't understand why I can't just take any box and install Asterisk in it and then plug that into the switch. |
| 20:50.20 | awormus | does plugging into a switch through a T1 card give you better reliability |
| 20:50.28 | awormus | isn't that what the QoS switch is for? |
| 20:50.53 | Schreiber1337 | I have a question about SIP extension names in Asterisk 1.4.13.... |
| 20:50.55 | awormus | the PRI card in the PBX means another $1200 on the price |
| 20:51.09 | eric_hill | awormus: Is the Cisco terminating the PRI? |
| 20:51.14 | awormus | eric_hill, yes |
| 20:51.15 | alrs | awormus: is this an XO flex t1? |
| 20:51.30 | eric_hill | awormus: At 3MB, it's probably two mft-t1 cards, right? |
| 20:51.46 | eric_hill | awormus: Next, does the Cisco have DSP modules? |
| 20:51.48 | awormus | alrs, not sure - it is probably the 2 t1s, as I know the switch has the 2 cards |
| 20:52.16 | eric_hill | awormus: Lastly, you need the IOS-VOICE software bundle (which you probably have) on the Cisco |
| 20:52.35 | greekguy8888 | shrike saw that and seems ok, i did the unload/load and no errors |
| 20:52.35 | awormus | eric_hill, not sure about the DSP modules, we haven't settled on the actual hardware that we will use. They want to put an IAD on the end, but I don't think that is nessesary since we're not goign to do any analogue |
| 20:52.47 | CrashSys | Hmm... ${CALLERID(num):4} returns null... that's interesting... |
| 20:52.56 | eric_hill | awormus: You'll need to set up VoIP targets for outbound SIP calls, and VoIP peers for inbound calls. |
| 20:53.10 | bsdwarrior | can someone help me with this. http://pastebin.com/d44980257 |
| 20:53.23 | eric_hill | awormus: *if* you're not using the Cisco for anything else, just pick up a dual port Digium T1 PRI card. |
| 20:53.29 | bsdwarrior | When I press any digit nothing happens, it doesnt get past the waitexten(5) |
| 20:54.18 | awormus | eric_hill, the cisco will deliver our internet as well and it will allocate the internet based on the number of phone lines in use |
| 20:54.36 | nhuisman_work | something tells me you have to restart asterisk after changing sip.conf |
| 20:54.39 | nhuisman_work | boy that was dumb |
| 20:54.58 | eric_hill | awormus: The Cisco is good and all, but just adds complexity to the situation. |
| 20:54.58 | nhuisman_work | no wonder it wouldn't take any other passwords, it still had testpass loaded from the first time I used it. |
| 20:55.14 | eric_hill | awormus: Ah - that's a different story. |
| 20:55.27 | *** join/#asterisk atamurad (n=chatzill@dialin-ppp-89.telecom.tm) |
| 20:55.38 | eric_hill | awormus: Using a bonded PRI link with drop and insert... yes, you want a Cisco ;) |
| 20:56.39 | awormus | eric_hill, OK - most of that went over my head but I will digest :) can you recommend a good book which will get me up to speed on Asterisk / VoIP and general concepts? |
| 20:56.48 | awormus | hardware etc. |
| 20:57.29 | atamurad | hi guys. when i press 1419 on the phone, get_data (phpagi) returns some digits repeated, like 111444411999. how can i fix it? |
| 20:57.32 | eric_hill | awormus: I haven't found a good book... Just realize that the Cisco is simply going to be a SIP target for placing calls out to the CO |
| 20:57.45 | [TK]D-Fender | bsdwarrior: taht is not how to allow DTMF to exit a queue and go somewhere else. |
| 20:57.58 | eric_hill | awormus: The Cisco will accept an incoming call and direct it over to the Asterisk box as an inbound SIP call. |
| 20:58.41 | Grnd-Wire | [TK]D-Fender: good afternoon! You don't know anything about the Asterisk Appliance do you? |
| 20:59.01 | [TK]D-Fender | Grnd-Wire: Yes, its a dead end wimpy device I'll have nothing to do with! :p |
| 20:59.09 | CrashSys | CISCO = Can I Still Call Out! |
| 20:59.24 | [TK]D-Fender | Grnd-Wire: And make sure to mount it port-side down otherwise the heat'll choke it out ;) |
| 20:59.42 | Grnd-Wire | [TK]D-Fender: hmm - I figured you'd say that.. but it IS an appliance.. I wish it wasn't so much money just to buy one to trial.. |
| 20:59.48 | awormus | eric_hill, so (and excuse my ignorance) if I have a PBX that runs asterisk and I use an online SIP based service - the service will basically play the same role as the cisco |
| 21:00.12 | Grnd-Wire | [TK]D-Fender; HAHA! I saw that in the manual, but they didn't make it clear why.. It's thermal radiation eh? That's funny.. |
| 21:00.48 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net) |
| 21:01.16 | Schreiber1337 | I have a question about SIP extension names in Asterisk 1.4.13.... |
| 21:01.22 | nny_1 | for the feeble minded (me) what does allowguest in sip.conf actually allow? It reads like "guest sip user" which I think I am reading wrong |
| 21:01.34 | Schreiber1337 | The last system that I setup was running Asterisk 1.2.16... I used to be able to setup a 4 line phone by making line1 3000, line2 3000b, line3 3000c, line4 3000c... |
| 21:01.47 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:01.49 | Schreiber1337 | but in 1.4.13 I don't seem to be able to connect to 3000b-c |
| 21:03.31 | Schreiber1337 | Is anyone else lable extensions this way? |
| 21:05.26 | eric_hill | awormus: Correct. A online SIP service is simply playing the role of a SIP->Telco gateway, just as the Cisco will do. |
| 21:06.20 | *** part/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 21:06.55 | awormus | eric_hill, great - thanks again |
| 21:07.30 | CrashSys | Anyone know a good international dial pattern? _011. ? |
| 21:07.40 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 21:12.23 | AlexTO | hi everyone, some one can tell me how can i fix it? http://pastebin.ca/858994 Thanks |
| 21:12.29 | [TK]D-Fender | CrashSys: sure |
| 21:13.02 | [TK]D-Fender | alex : You are missing a dialplan hint for presence. Your phone is looking for one that doesn't exist. |
| 21:13.21 | [TK]D-Fender | AlexTO: Also, you are running * GUI and it is not supported here, please ask in their channel |
| 21:13.46 | AlexTO | nobody answer there :-S |
| 21:13.55 | *** part/#asterisk greekguy8888 (n=alex@c-76-118-201-12.hsd1.ma.comcast.net) |
| 21:14.44 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 21:14.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:15.07 | jburbage | Is there any way to configure a queue to check GROUP_COUNT before dialing an agent, while still having the agent logged in on their actual interface (e.g. SIP/123 and not Local/123@agents)? |
| 21:15.08 | [TK]D-Fender | AlexTO: Well i've told you what it means. Crack open the book if you think you're going to try and mess with it by hand |
| 21:15.29 | grayhame | has anyone had any issues with Monitor in/out files not getting mixed at the end of a call? |
| 21:17.34 | Schreiber1337 | [TK]D-Fender: Can you help with my extension problem? |
| 21:18.12 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 21:18.14 | [TK]D-Fender | Schreiber1337: Why would you setup each key to a different registration? And what model? |
| 21:19.26 | [TK]D-Fender | nny_1: it allows non-authed calls to fall the the context specified under [general] |
| 21:19.43 | Schreiber1337 | [TK]D-Fender: I'm looking at using all 4 lines in each phone (Linksys SPA942) and that's just how I have always configured them... |
| 21:20.03 | [TK]D-Fender | Schreiber1337: a waste. Reg once and use all 4 keys for it. |
| 21:20.06 | nny_1 | [TK]D-Fender: hmm seems no would be a good option by default for that |
| 21:20.19 | [TK]D-Fender | Schreiber1337: you don't need multiple identities... that creates a management nightmare. |
| 21:20.39 | [TK]D-Fender | nny_1: Depends where. I have "yes" because I allow direct URI dialing. |
| 21:20.53 | Schreiber1337 | [TK]D-Fender: So I would register each line as the same extension? |
| 21:21.10 | [TK]D-Fender | Schreiber1337: No. You reg only the FIRST and tell it to use all 4 keys <- |
| 21:21.10 | bsdwarrior | is there any way to put a user back in the queue in the same spot ? |
| 21:21.48 | [TK]D-Fender | bsdwarrior: No. |
| 21:22.16 | Schreiber1337 | [TK]D-Fender: I guess I don't know what you are refering to as "keys" or how to configure them? |
| 21:22.16 | bsdwarrior | tkd-fender im chasing my tail here |
| 21:22.39 | Schreiber1337 | [TK]D-Fender: Could you provide an example? |
| 21:22.43 | [TK]D-Fender | bsdwarrior: Correct.... |
| 21:23.00 | jburbage | Schreiber1337: he's talking about your 4 lines. Most phones have a key (button) for each line. |
| 21:23.21 | [TK]D-Fender | Schreiber1337: your 4 buttons on the SPA are not "lines", they are just "keys" and the represent a distribution of identites based on your config |
| 21:23.36 | [TK]D-Fender | Schreiber1337: You can say use all 4 for *!* registration in the first reg |
| 21:23.46 | [TK]D-Fender | *1* |
| 21:25.16 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 21:25.32 | esaym | I keep getting a busy tone trying to end a meetme room. meetme.conf =" [rooms]\conf => 555" and extensions.conf= "[rooms]\ exten => 555,1,MeetMe(555)" any clue? |
| 21:25.57 | esaym | opps, not "end a meetme room" but "enter a meetme room" |
| 21:26.00 | [TK]D-Fender | esaym: yeah... go look in CLI to see whats happening. |
| 21:26.43 | nny_1 | [TK]D-Fender: thanks btw |
| 21:26.47 | [TK]D-Fender | ok, I'm off, BBIAB |
| 21:26.53 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:27.14 | javar | lotho, still there? |
| 21:27.21 | esaym | cli with verbose 9 shows nothing |
| 21:29.48 | deeperror | Need some help tracking down an issue. For some reason core show channels after 3-5 days uptime on the server will show thousands of channels so many in fact that it never really stops to show a total. Any way to find more core related info or setup logging on that? my current logs don't show much and I'm unable to really reproduce just seems to happen ever 3-4 days. That would also be about 20-40,000 calls in those days |
| 21:30.22 | grayhame | when using the Monitor application, what would keep the in and out files from being mixed together after the call? |
| 21:30.25 | deeperror | sip show channels however will return a farily accurate count |
| 21:31.17 | nhuisman_work | does anyone know much about cisco phones and their tlv files (CTLSEPXXXXXXXXXXX.tlv and CTLfile.tlv) and how to setup the certificates? |
| 21:31.27 | nhuisman_work | my phone is displaying File Auth Fail : CTLFile.tlv |
| 21:31.52 | *** join/#asterisk MaliutaWrk (n=nikolai@119.11.100.210) |
| 21:36.30 | *** join/#asterisk AndyGraybeal_ (n=andy@node49.32.251.72.1dial.com) |
| 21:36.52 | esaym | do I have to have that ztdummy timer installed to use meetme in 1.2-26? |
| 21:40.41 | tzafrir_home | esaym, actually meetme uses Zaptel for the mixing itself |
| 21:41.06 | *** join/#asterisk `paul (n=aldee@125.252.68.68) |
| 21:41.33 | tzafrir_home | So yes - if you don't have a hardware zaptel device, you need ztdummy |
| 21:41.38 | `paul | what seems to be the problem when the sip phone seems to time out often |
| 21:41.58 | drmessano-LT | Did someone suggest using an X100p for timing in lieu of ztdummy? |
| 21:44.38 | CrashSys | I've ran into problems with X100p's crashing servers... even the 'good' ones from x100.com |
| 21:44.56 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:45.15 | esaym | ok I am on to something |
| 21:45.18 | J4k3 | the whole x100 situation is pathetic |
| 21:45.34 | esaym | in cli "show applications" does not show meetme. How do I install meetme? |
| 21:45.45 | J4k3 | you've obviously got a huge market for single POTS termination devices, and all anyone can come up with to fill the niche is a goddamn shitty modem from 1999. |
| 21:46.01 | lotho | javar: yes |
| 21:46.12 | javar | hi lotho |
| 21:46.13 | CrashSys | A200? |
| 21:46.19 | javar | i have an error: |
| 21:46.22 | drmessano-LT | lol |
| 21:46.25 | javar | CMake Error: MISSING HEADER: glibconfig.h |
| 21:46.27 | drmessano-LT | Ok |
| 21:46.29 | J4k3 | quite simply, there needs to be a PCI card, under $60 USD, that can do this just fine. |
| 21:46.39 | J4k3 | no modem h4x, no nonsense. |
| 21:46.40 | tzanger | J4k3: do what just fine |
| 21:46.47 | nhuisman_work | man cisco phones are cryptic in their tftp stuff :P |
| 21:46.48 | javar | can you help me? |
| 21:46.50 | J4k3 | tzanger: FXS |
| 21:46.52 | tzanger | actually no, single or dual port termination is the home of the ATA |
| 21:46.59 | J4k3 | ATAs blow |
| 21:46.59 | tzanger | I used to agree that you want it in a PC |
| 21:47.04 | tzanger | but I've decided that's wrong |
| 21:47.10 | tzanger | get an ATA that can run linux/asterisk and that's it |
| 21:47.13 | J4k3 | they're on the same level of sucking as using an ATA for extensions. |
| 21:47.24 | drmessano-LT | Maybe I should rephrase.... Heard someone mention using an X100P for timing, forget the shitty line performance.. any truth to it? |
| 21:47.26 | tzanger | nah |
| 21:47.34 | tzanger | I did like the TDM400 but it's getting long int he tooth |
| 21:47.45 | J4k3 | tzanger: shitty solution to a simple problem. I've yet to see an ATA that works as well as an x100 :| |
| 21:47.49 | tzanger | I don't want a PC for asterisk+firewall, PC for office, PC for everything |
| 21:47.51 | J4k3 | and the x100 is godawfulbad. |
| 21:47.53 | *** join/#asterisk BiGrAr (n=bigrar@208.178.99.170) |
| 21:48.21 | tzanger | I want a little box that's got a DSL modem, wireless interface, an fxo, two or three FXS and a couple ethernet ports... that's it |
| 21:48.31 | lotho | javar: try searching this file and tell me where it is on your system |
| 21:48.31 | tzanger | it's called an appliance, almost. |
| 21:48.32 | nhuisman_work | like the asterisk appliance? |
| 21:48.35 | J4k3 | so lightning can hit your phone line and blow the whole investment? |
| 21:48.35 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:48.41 | tzanger | nhuisman_work: no, not quite bu tyeah that's the idea |
| 21:48.41 | BiGrAr | any advice on deploying phones out over the internet behind god knows what sort of soho devices that will register back via sip to an asterisk server w/ a public ip? |
| 21:48.51 | J4k3 | sorry, I'll take a modular device, thx |
| 21:48.56 | nhuisman_work | i dunno i think you need a wifi router + the appliance |
| 21:49.02 | javar | I check the Cmake file and it said: FIND_PATH(SPANDSPCONF_INCLUDE_DIR glibconfig.h /usr/include/glib-2.0 /usr/lib/glib-2.0/include) |
| 21:49.02 | tzanger | BiGrAr: polycom. I wub 'em |
| 21:49.07 | nhuisman_work | tzanger, btw you can run asterisk on linksys routers |
| 21:49.10 | tzanger | J4k3: I used to think that way |
| 21:49.13 | tzanger | nhuisman_work: I'm aware of that |
| 21:49.16 | tzanger | no FXS/FXO ports |
| 21:49.16 | BiGrAr | polycom what? phones? |
| 21:49.21 | nhuisman_work | and you dont' want atas? |
| 21:49.21 | tzanger | BiGrAr: correct |
| 21:49.24 | javar | But i don't have that dir |
| 21:49.28 | DaveCanoe | When I use DISA() normally, the 'h' extension is called when it hangs up. However, when the called is initiated by asterisk itself (using /var/spool/asterisk/outgoing), the 'h' member of the context is _not_ called when the call (in DISA) hangs up. |
| 21:49.39 | J4k3 | tzanger: *shrug* some of us are actual users, and not just consultants throwing shit into a shop and walking away |
| 21:49.39 | lotho | javar: yes right, i had the same error |
| 21:49.52 | tzanger | J4k3: I'm not a consultant throwing shit in a shop and wlaking away |
| 21:49.53 | javar | what can i do? |
| 21:50.09 | J4k3 | tzanger: if I was consulting for end users, I'd agree... |
| 21:50.14 | J4k3 | less crap for the lusers to screw up |
| 21:50.19 | tzanger | for decent rollouts (PRI+) you want a PC. However for SOHO, a PC is overkill, overmaintenance, noisy and a pain in the ass |
| 21:50.21 | BiGrAr | tzanger, are there nat settings in them for RTP ports or something? do they always behave well behind bs soho equipment? |
| 21:50.30 | lotho | javar: do you have the file on your system? |
| 21:50.31 | deeperror | For some reason core show channels after 3-5 days uptime on the server will show thousands of channels so many in fact that it never really stops to show a total. Any way to find more core related info or setup logging on that? my current logs don't show much and I'm unable to really reproduce just seems to happen ever 3-4 days. And last time this occured a file ast-ami-TpaOO6 appeared in tmp...any clues!? |
| 21:50.46 | lotho | javar: perhaps in a different directory |
| 21:50.46 | javar | lotho, let me check |
| 21:50.46 | J4k3 | tzanger: so instead you buy something thats slightly quieter for twice the price and 1/10th the horsepower? |
| 21:50.51 | drmessano-LT | I think a modem should be a modem.. Its also a big FUSE, and easier to replace than a PBX |
| 21:50.52 | nhuisman_work | you could use a redfone gateway with an appliance |
| 21:50.57 | nhuisman_work | then you don't need a pc for the pri |
| 21:50.58 | tzanger | BiGrAr: They've taken pretty much every kind of nat I've thrown at 'em without ANY adjustment. I think some here though have run into specific shitty NAT routers it can't stand, though |
| 21:51.11 | J4k3 | tzanger: embedded systems with any real horsepower cost significantly more than a modern celeron+mobo+512mb ram+cf adapter+cf card. |
| 21:51.31 | tzanger | J4k3: explain to me the obsession with horesepower on a device you CANNOT use it on -- you want your asterisk PC ot do your file serving too? Or your CAD? Explain to me why I want ot throw a 90W processor on it that I CANNOT use?! |
| 21:51.32 | BiGrAr | i have a hard time not recommending vpn for this stuff.... it is just expensive |
| 21:51.42 | J4k3 | tzanger: ever transcoded? |
| 21:51.45 | lirakis | later all |
| 21:52.01 | nhuisman_work | seems like if you have a pri you have at least 50 users |
| 21:52.04 | nhuisman_work | or more |
| 21:52.05 | fujin | yuck, transcoding is a bad idea anywhere you are. |
| 21:52.08 | J4k3 | tzanger: you've got these people that are like "whee I can run asterisk on my linkydink"... try having a couple folks leaving voicemail at the same time :P |
| 21:52.08 | tzanger | J4k3: you won't be oding much of that, and I can throw a blackfin BF537 DSP at you for 1/5 the price and get MORE transcodes out of it |
| 21:52.08 | fujin | embedded or not |
| 21:52.09 | javar | lotho: /usr/share/doc/glibc-2.5 |
| 21:52.12 | *** part/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 21:52.22 | tzanger | J4k3: you are confusing things to prove your point. |
| 21:52.33 | nhuisman_work | tzanger, you can always buy a really small pc |
| 21:52.35 | tzanger | J4k3: 1) asterisk needs to run on its own, or Bad Things will happen. |
| 21:52.35 | J4k3 | tzanger: ok, and would asterisk support that configuration? |
| 21:52.40 | BiGrAr | it all comes down to concurrent calls and voicemail accesses |
| 21:52.41 | lotho | javar: make a symlink |
| 21:52.45 | nhuisman_work | tzanger, with really low voltage and shit |
| 21:52.49 | nhuisman_work | nano-atx mobo |
| 21:52.49 | BiGrAr | just like any other phone system |
| 21:52.54 | nhuisman_work | fanless |
| 21:52.56 | tzanger | 2) Asterisk needs to be on hardware that is specc'd for its intended application and number of simultaneous calls |
| 21:52.58 | nhuisman_work | etc etc |
| 21:53.00 | javar | lotho: how ? |
| 21:53.01 | J4k3 | tzanger: asterisk, technically, should be its own OS from end to end. |
| 21:53.09 | tzanger | haha |
| 21:53.09 | J4k3 | linux/bsd doesn't offer the time management requirements for a decent pbx. |
| 21:53.49 | J4k3 | its designed to serve warez and pr0n, and let geeky developers run 16 processor servers. |
| 21:53.50 | javar | lotho: But the file glibconfig.h, i don't have it |
| 21:54.13 | J4k3 | err, 16 processor servers as desktops |
| 21:55.17 | lotho | javar: is the package glib installed? |
| 21:55.36 | tzanger | small PC running compact flash is an option for sure |
| 21:55.39 | esaym | is zaptel 1.2 for asterisk 1.2 and zaptel 1.4 for asterisk 1.4? Or can zaptel 1.4 be used with asterisk 1.2? |
| 21:55.40 | J4k3 | tzanger: ever considered running asterisk under windows? replace windows with linux... thats how I feel :P |
| 21:55.45 | Havokmon | J4k3: Do you really need decent time management when horsepower is so high now? |
| 21:55.47 | javar | lotho: i'm checking that, let me one moment |
| 21:55.56 | mocker | J4k3: How'd you know about all my warez and pr0n servers???? |
| 21:55.59 | [TK]D-Fender | esaym, You need matching versions |
| 21:56.18 | J4k3 | Havokmon: yeah, because a pbx is a very realtimeish thing. you fail to deliver a packet to a phone for a couple dozen milliseconds, and you might as well not send it at all. |
| 21:56.38 | mocker | ohwait, you are being serious? |
| 21:57.07 | Havokmon | J4k3: Sure - but we're only talking 64kb packets, your PC isn't going to be the issue. |
| 21:57.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:57.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:58.20 | Havokmon | err |
| 21:58.28 | Havokmon | you know what I mean :P |
| 21:58.38 | Havokmon | Damnit. Just run OS/2 and be happy |
| 21:58.46 | *** join/#asterisk marlow (n=marlow@sleipner.sca.airwire.ie) |
| 21:58.47 | nhuisman_work | *vomit* |
| 21:58.51 | J4k3 | Havokmon: well, it doesn't matter what the packet size is if the kernel is, say, wrapped up around shoving some data to a hard drive at the time. |
| 21:58.55 | CrashSys | Yay OS/2! |
| 21:59.31 | J4k3 | smp helps, but x86 smp is painfully bad |
| 22:00.05 | javar | lotho: glib.i386 1:1.2.10-20.el5 installed |
| 22:00.08 | Havokmon | J4k3: No, that's actually Ok. I have a test box with a single bad HD... I had about 10 agents signed in (it was an emergency), and when the HD started to thrash all you had was "Youre-re-re-re Secondndndn in liiiiiinenene" ;) No dropped calls though ;) |
| 22:00.11 | mocker | J4k3: Then go pay $30000 for a PBX. :) |
| 22:00.13 | nhuisman_work | anyone using 7940s or other cisco phones with skinny? If so may I take a look at your conf |
| 22:00.18 | DaveCanoe | opterons are significantly better at cache sync than intel's. |
| 22:00.53 | javar | lotho:/usr/lib/glib/include/glibconfig.h |
| 22:00.53 | J4k3 | DaveCanoe: AMD's cpus are great.. too bad they only sell $300-desktop-quality chipsets for them these days. |
| 22:00.56 | DaveCanoe | Anyways... Seems like a bug. the 'h' hangup extension isn't called from the callback context. |
| 22:00.57 | mocker | J4k3: I even have an old Avaya Definity I can sell you. |
| 22:01.20 | DaveCanoe | get a sun MB if you want a good opteron. |
| 22:01.23 | lotho | javar: you found it :) |
| 22:01.28 | javar | yeah |
| 22:01.29 | J4k3 | DaveCanoe: hmm good call. |
| 22:01.31 | javar | :) |
| 22:01.36 | nhuisman_work | anyone? |
| 22:01.40 | DaveCanoe | ~$1500-ish for the base model. |
| 22:01.46 | jblack | I laid down for a nap at 9am. I didn't wake up until 4:30pm |
| 22:01.48 | javar | now? i'll need edit the CMake fike? |
| 22:01.54 | javar | * file |
| 22:01.59 | DaveCanoe | and they include all the drive sleds now (so you can by reasonably priced drives) |
| 22:01.59 | CrashSys | I'll stick with my M2N32's for my AMD's... |
| 22:02.19 | J4k3 | thats not too bad. I've been buying g33-based boards for c2d's lately for pc-servers. |
| 22:02.31 | *** join/#asterisk Stefan1979 (n=stan@4204ds2-vby.0.fullrate.dk) |
| 22:02.45 | J4k3 | its cheap and consistant but not exactly high performance. |
| 22:02.51 | lotho | javar: i would do a symlink to the place where cmake is looking for the file |
| 22:03.02 | lotho | or edit the cmake file your choice |
| 22:03.17 | javar | lotho: how can i do that? |
| 22:03.24 | lotho | the symlink? |
| 22:03.33 | javar | yeah |
| 22:04.04 | J4k3 | maybe I'd like linux's time management more if I dumped the crappy IDE drives I'm using. ;) |
| 22:04.33 | J4k3 | IDE = Inadiquate Disks (for) Enterprise |
| 22:04.38 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:04.55 | nhuisman_work | get some seagate sata ES class drives |
| 22:04.57 | CrashSys | Linux has more issues with SATA... like the infamous soft-reset issues |
| 22:04.58 | DaveCanoe | anyways... what's the deal with callback's not calling the hangup 'h' extension? |
| 22:05.16 | DaveCanoe | FreeBSD + ZFS :). |
| 22:05.25 | J4k3 | well, SATA suffers from most of the same problems as IDE, except DMA commands |
| 22:05.55 | nhuisman_work | i guess if you have money get sas |
| 22:05.57 | lotho | javar: ln -s /usr/lib/glib/include/glibconfig.h /usr/include/glibconfig.h |
| 22:05.59 | J4k3 | fully coprocessed SATA stuff is cute til it bombs out (I had a dell cerc card here, omfg it was junk and died very quickly) |
| 22:06.04 | lotho | this should do it |
| 22:06.09 | javar | lotho: :) |
| 22:06.23 | nhuisman_work | dell cerc cards are the biggest pieces of shit ever |
| 22:06.25 | J4k3 | SCSI is so darned old, I hate the phyical wiring issues. |
| 22:06.25 | nhuisman_work | get areca |
| 22:06.28 | J4k3 | yes they were |
| 22:06.31 | marlow | depends |
| 22:06.37 | marlow | nhuisman_work : depends |
| 22:06.44 | nhuisman_work | on what if you get lucky? |
| 22:06.46 | marlow | nhuisman_work :if they are SATA or PATA :) |
| 22:06.49 | J4k3 | haha, we're making new words for the IDE acronym in the office now... IDE = It Doesn't Evolve |
| 22:06.52 | J4k3 | ;) |
| 22:06.53 | nhuisman_work | they are both crap :P |
| 22:06.58 | javar | lotho: same problem |
| 22:07.00 | marlow | nhuisman_work : PATA is AMI/LSI base |
| 22:07.04 | CrashSys | I got me an old P2-300a overclocked to 450 on my 440bx is md in raid 1!!! |
| 22:07.06 | javar | lotho:CMake Error: MISSING LIBRARY: glib-2.0 |
| 22:07.09 | marlow | nhuisman_work : SATA is Adaptec base |
| 22:07.09 | [hC] | J4k3: it'll die eventually |
| 22:07.21 | nhuisman_work | i'm not so sure lsi is that spectactular either |
| 22:07.34 | marlow | nhuisman_work : the LSI has shit performance |
| 22:07.35 | [TK]D-Fender | IDE + SATA = PITA |
| 22:07.36 | DaveCanoe | is there a workaround for this 'h' extension problem? |
| 22:07.47 | marlow | nhuisman_work : the adaptec SATA controller never fails |
| 22:07.50 | lotho | javar: is glib2 installed? |
| 22:07.52 | nhuisman_work | it's still slow as hell |
| 22:07.54 | nhuisman_work | the adaptec |
| 22:08.03 | marlow | depends on what you want it for |
| 22:08.04 | javar | lotho: let me check |
| 22:08.25 | [hC] | anyone familiar with the hylafax+iaxmodem combo? specifically where the limitation with only being able to send @ 9600bps comes from |
| 22:08.25 | marlow | nhuisman_work : adaptec is twice as quick as the LSI ones :) |
| 22:08.29 | nhuisman_work | plus have fun finding a nice linux interface so email you when the cards die |
| 22:08.34 | javar | lotho: no, but i'll install it now |
| 22:08.43 | nhuisman_work | oh yeah you can use dells bullshit java program |
| 22:08.56 | nhuisman_work | rather when drives die. |
| 22:08.58 | javar | lotho: glib2.i386 2.12.3-2.fc6 installed |
| 22:09.04 | [TK]D-Fender | [hC], Because any kind of faxing through * is a miracle as it is. |
| 22:09.29 | [hC] | [TK]D-Fender: apparently its actually spandsp's fault. my faxing solution is solid as hell aside from being only 9600 baud |
| 22:09.29 | lotho | javar: and? |
| 22:09.36 | [hC] | [TK]D-Fender: 100% success rate.. but this is email to fax.. |
| 22:09.44 | javar | lotho: i dunno :( |
| 22:09.44 | [hC] | and vice versa. |
| 22:09.53 | [hC] | [TK]D-Fender: get the new firmware yet from polycom? |
| 22:10.06 | [TK]D-Fender | [hC], You know what happens onces there's just a little jitter.... thats why the rate is forced low |
| 22:10.14 | [TK]D-Fender | [hC], No... I really did hope to... |
| 22:10.54 | [hC] | [TK]D-Fender: let me know if you want it. |
| 22:11.06 | [TK]D-Fender | [hC], Of course I do. You got them today? |
| 22:11.13 | DaveCanoe | Has anyone even run across the problem where callback calls don't call the 'h' hangup extension? |
| 22:11.16 | [hC] | [TK]D-Fender: I got it last night. |
| 22:11.25 | [hC] | [TK]D-Fender: sec, i'll get you a URL. |
| 22:12.25 | nhuisman_work | bleh fuck chan_skinny |
| 22:12.49 | lotho | javar: is the "missing glib2"-error already there? |
| 22:13.07 | [TK]D-Fender | nhuisman_work, s/chan_skinny/cisco/ |
| 22:13.14 | javar | lotho: yes |
| 22:13.15 | nhuisman_work | yeah both of them |
| 22:13.17 | nhuisman_work | can die on fire |
| 22:13.23 | nhuisman_work | won't stop crashing my asterisk |
| 22:13.44 | nhuisman_work | i bet 1.4 latest doesn't crash too |
| 22:13.55 | nhuisman_work | abe... sigh. I think I might just install 1.4 |
| 22:16.08 | drmessano-LT | ~cisco |
| 22:16.09 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 22:16.09 | esaym | anyone have any tips on getting my voip number in my name? Right now if I call a time warner digital phone it will show the name as being some weird company. Arn't there websites where you can register your phone number and name for caller id looks ups? |
| 22:16.29 | drmessano-LT | Cisco makes a good router that somewhat load balances half the time |
| 22:17.20 | drmessano-LT | Their algorithm is Line1, Line2, Line1, Line2, Drop |
| 22:17.29 | drmessano-LT | YAY |
| 22:19.52 | Shaun2222 | where's lmadsen |
| 22:20.46 | Shaun2222 | he told me to switch from macro to gosub and the dial() had a U option for gosub but i dont see it in trunk |
| 22:22.46 | Shaun2222 | putnopvut: i'm running trunk, i didnt see it when doing that |
| 22:23.00 | putnopvut | It's listed between the T and w options. |
| 22:23.16 | putnopvut | Sure you're running trunk? |
| 22:23.48 | Shaun2222 | Asterisk SVN-trunk-r60662 built by root @ pbx1.irv.xxxxxxxxxxx.com on a x86_64 running Linux on 2008-01-16 23:17:48 UTC |
| 22:24.09 | Qwell | update |
| 22:24.14 | Shaun2222 | what |
| 22:24.18 | Qwell | asterisk |
| 22:24.22 | Shaun2222 | i just downloaded that yesturday |
| 22:24.23 | putnopvut | Yeah, 60662 is really old. |
| 22:24.28 | Qwell | svn up |
| 22:24.32 | Shaun2222 | lol, did i find a old trunk? |
| 22:24.39 | nhuisman_work | hey, btw should digium cards blink red lights when there are no pris plugged in? |
| 22:24.47 | Qwell | Shaun2222: no, there were issues with svn earlier. |
| 22:24.48 | putnopvut | Shaun2222: yeah, the svn public mirror was being rebuilt the last couple of days. |
| 22:25.03 | putnopvut | It finished earlier today, so if you svn up, you'll be current. |
| 22:25.10 | Shaun2222 | make clean is broken |
| 22:25.16 | putnopvut | ... |
| 22:25.36 | Shaun2222 | maybe it's because i just 'svn up' |
| 22:25.42 | Shaun2222 | svn: Failed to add file 'include/asterisk/version.h': object of the same name already exists |
| 22:25.47 | Shaun2222 | should i be worried about that |
| 22:25.48 | Qwell | rm include/asterisk/versionh |
| 22:25.48 | Qwell | svn up |
| 22:25.52 | Qwell | version.h too |
| 22:26.15 | nhuisman_work | I just booted the server and the cards lights are blinking red |
| 22:26.19 | Shaun2222 | fuck it, rm -rf *;svn up |
| 22:26.48 | Shaun2222 | ok, looks good, let me build it real quick |
| 22:27.06 | brodiem | putnopvut: hey your queue def worked fine in creating recordings |
| 22:27.38 | brodiem | putnopvut: in comparison, it will only create the recordings after adding memberdelay to the queue def! |
| 22:28.26 | putnopvut | Whoa, that's really weird. |
| 22:28.41 | brodiem | yeah, just to be sure let me just remove it from your queue def leaving everything else as is |
| 22:29.00 | putnopvut | Yeah, I'm going to look in the code and see if I can find a correlation. |
| 22:31.23 | putnopvut | Nothing obvious. |
| 22:31.58 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 22:32.58 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:34.07 | jblack | Who are the dundi code gods? |
| 22:34.26 | brodiem | putnopvut: I may have jumped the gun on that sorry, =/ let me keep trying the differences |
| 22:36.43 | putnopvut | Yeah, I just did a quick test and all seems okay. |
| 22:38.57 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 22:39.11 | javar | lotho: still there? |
| 22:40.15 | lotho | javar: yes |
| 22:40.29 | javar | lotho:[100%] Built target test_spandsp |
| 22:40.57 | javar | but i don't see the app's on /usr/lib/asterisk/modules/ |
| 22:42.17 | lotho | you did "make install", right? |
| 22:42.34 | javar | lotho: ooops |
| 22:42.56 | javar | lotho: i just did ./build.sh |
| 22:43.10 | Shaun2222 | for codecs right now i'm only loading gsm ilbc and ulaw... any others any of you would recommend? |
| 22:43.11 | *** part/#asterisk marlow (n=marlow@sleipner.sca.airwire.ie) |
| 22:43.45 | javar | lotho: i did make install, and i see the app's now!!! |
| 22:43.52 | lotho | :) |
| 22:43.54 | brodiem | putnopvut: this is the damn weirdest thing... if I add memberdelay to the queue def, I get a recording. If I remove memberdelay, it keeps recording. If I rename the queue def/dialplan ext (leaving memberdelay out), it doesn't work again until I add memberdelay. WTF?? lol |
| 22:44.24 | javar | lotho: now? |
| 22:45.03 | lotho | javar: you have to write app_rxfax.so in your modules.conf |
| 22:45.23 | lotho | and then extend your dialplan |
| 22:45.23 | putnopvut | brodiem: since you're using 1.4.17, I can understand why removing the memberdelay would keep allowing you to record (there was a bug where removing memberdelay didn't actually remove it) |
| 22:45.38 | putnopvut | It's been fixed since 1.4.17 was released. |
| 22:45.47 | javar | lotho: ah ok, let me try it |
| 22:46.53 | putnopvut | brodiem: so it sounds like the memberdelay is in some way affecting the recording...I don't know if it's the delay itself or the memberdelay option. |
| 22:46.56 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 22:47.47 | javar | lotho: loader.c:363 load_dynamic_module: Error loading module 'app_rxfax.so': libspandsp.so.0: cannot open shared object file: No such file or directory |
| 22:48.22 | brodiem | putnopvut: ahhh so I'm not completely crazy lol |
| 22:48.56 | putnopvut | Right. I'm trying a test now with no memberdelay or announce file in place. |
| 22:49.04 | [TK]D-Fender | brodiem, No... you're probably still completely crazy.... you just happen to be right as well ;) |
| 22:49.06 | brodiem | ahaha |
| 22:49.16 | brodiem | good call |
| 22:49.46 | putnopvut | As soon as the #$(# phone registers... |
| 22:50.34 | putnopvut | ah ha! |
| 22:51.08 | brodiem | confirmed? |
| 22:51.11 | putnopvut | So, my preliminary analysis is that you either need an announce file, reportholdtime on, or a memberdelay in order for monitor to record local channels correctly. I just got 4 byte recordings. |
| 22:51.19 | putnopvut | They're empty. |
| 22:51.35 | putnopvut | So, now to figure out WHY. |
| 22:51.50 | javar | lotho: any idea? |
| 22:51.52 | brodiem | putnopvut: I am using announce-holdtime=once |
| 22:52.05 | lotho | javar: would be to easy if it worked out of the box ;) |
| 22:52.11 | putnopvut | No not that. reportholdtime. The one that tells the member the holdtime for the caller they're answering. |
| 22:52.20 | brodiem | ahh my bad |
| 22:52.27 | lotho | javar: spandsp is installed? |
| 22:52.40 | javar | lotho: sure |
| 22:53.09 | javar | lotho: spandsp-0.0.4pre16 |
| 22:54.15 | lotho | javar: ok, then look in /usr/lib/ if there is something named libspandsp |
| 22:54.26 | javar | ok |
| 22:55.35 | javar | lotho: no |
| 22:56.48 | javar | lotho: /usr/local/lib/libspandsp.so.0 |
| 22:56.49 | lotho | then ssearch for libspandsp on your system |
| 22:56.52 | lotho | ah |
| 22:57.11 | javar | i need a smylink? |
| 22:57.17 | lotho | yes |
| 22:57.23 | javar | which? |
| 22:57.32 | javar | O:) |
| 22:57.50 | lotho | ln -s /usr/local/lib/libspandsp.so.0 /usr/lib/libspandsp.so.0 |
| 22:58.08 | lotho | you should look at the man page for ln ;) |
| 22:58.22 | javar | yeah :P |
| 22:59.12 | brodiem | putnopvut: you're right, reportholdtime=yes works also |
| 22:59.30 | javar | lotho: show applications : RxFAX: Receive a FAX to a file |
| 22:59.35 | javar | lotho: great!!! |
| 22:59.47 | lotho | :) |
| 23:00.06 | javar | lotho: many thanks |
| 23:00.06 | putnopvut | brodiem: Okay, so now to figure out why that works. This is odd indeed. I'm going to make sure it also works for a channel which is not local. |
| 23:00.14 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net) |
| 23:01.33 | lotho | javar: was a pleasure ;) |
| 23:01.48 | putnopvut | brodiem: works fine with a SIP channel instead of local. I think I may know what's up. |
| 23:02.10 | brodiem | awesome |
| 23:03.08 | javar | lotho: :) |
| 23:03.28 | nhuisman_work | does asterisknow only detect analog cards? or can it work with digium digital cards too? |
| 23:04.24 | Shaun2222 | <PROTECTED> |
| 23:05.54 | putnopvut | Shaun2222: zaptel trunk is not recommended as it is highly broken atm. |
| 23:06.26 | Shaun2222 | rebuilding asterisk from trunk broke my chan_zap.so. |
| 23:06.55 | Shaun2222 | [Jan 17 15:06:46] ERROR[11740]: chan_zap.c:13494 process_zap: Unknown signalling method 'pri_cpe' |
| 23:07.00 | putnopvut | brodiem: apparently what the problem is is that the Monitor is monitoring the Local channel, and then the call is bridged. When the call gets bridged, the Local channel masquerades into the SIP (or Zap or whatever) channel that answers, and then is hung up. When it hangs up, Monitor thinks the call is over, thus making no files. |
| 23:07.12 | Shaun2222 | i got that before, i cant remember what i did to fix it, think it was the build order |
| 23:07.52 | putnopvut | Shaun2222: yeah, it thinks you don't have libpri installed. |
| 23:08.01 | Shaun2222 | ah, was lib pri. |
| 23:08.15 | brodiem | putnopvut: so the delay allows it to finish the bridge before monitor starts? |
| 23:08.37 | putnopvut | Correct, therefore the monitor actually is recording the SIP (or Zap or whatever) channel. |
| 23:08.45 | brodiem | cool |
| 23:08.54 | putnopvut | Well, actually the bridge doesn't happen first, but the masquerade happens. |
| 23:09.02 | kyron | is the extension 700 somehow reserved in * (or 70) ?? |
| 23:10.00 | kyron | garh! |
| 23:10.11 | kyron | this is a phone, not a car.. damned...sorry about that |
| 23:10.16 | brodiem | now if only someone would backport state_interface to 1.4... ;) |
| 23:10.18 | brodiem | lol |
| 23:10.42 | russellb | someone already did |
| 23:10.48 | brodiem | oh really |
| 23:10.49 | russellb | and posted a link to the -dev list |
| 23:11.15 | brodiem | nice |
| 23:11.49 | Shaun2222 | bah wtf am i doing wrong... putnopvut: i rebuild libpri then asterisk... still borked. |
| 23:12.03 | brodiem | well then it's been a productive day for me over all, lol |
| 23:12.11 | putnopvut | Shaun2222: same problem as before? |
| 23:12.17 | Shaun2222 | ya |
| 23:12.22 | Shaun2222 | the one i pasted above |
| 23:12.30 | putnopvut | I think you need to rerun the configure script in Asterisk. |
| 23:12.46 | putnopvut | That's what detects the installation of libpri. |
| 23:13.41 | Shaun2222 | with a sangoma card do i even need the zaptel source? |
| 23:14.02 | putnopvut | Shaun2222: I'm no authority on Sangoma, but I am pretty sure you do. |
| 23:14.32 | tzafrir_home | Shaun2222, Sangoma PRI? |
| 23:14.48 | Shaun2222 | ya A101D i think |
| 23:14.55 | tzafrir_home | its setup scripts generally likes to patch the Zaptel source |
| 23:15.28 | Shaun2222 | i dont remember giving it the path to the source but it may have found it on it's own since i build in /usr/src |
| 23:17.21 | Shaun2222 | rebuilt libpri then zaptel then asteris |
| 23:17.25 | Shaun2222 | same problem |
| 23:19.10 | JT | Shaun2222: you need wanpipe + zaptel + libpri + asterisk |
| 23:22.05 | brodiem | russellb: you weren't talking about func_devstate were you? |
| 23:23.22 | putnopvut | brodiem: no, he means the queue_state branch. I'll provide a link |
| 23:23.56 | brodiem | appreciate it |
| 23:24.15 | putnopvut | brodiem: http://lists.digium.com/pipermail/asterisk-dev/2008-January/031545.html |
| 23:24.21 | Shaun2222 | so much for ./Setup upgrade with sangoma... whats the point it asks all the same questions... |
| 23:24.36 | brodiem | cool thanks |
| 23:29.50 | *** part/#asterisk RoyK (n=roy@91.149.13.189) |
| 23:30.51 | putnopvut | brodiem: FYI, that problem with monitoring and local channels is not a queue-specific thing. I just ran a test where I directly dialed a local channel and the same behavior occurred. |
| 23:31.18 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-76-246-191-68.dsl.lsan03.sbcglobal.net) |
| 23:32.50 | Shaun2222 | JT: still getting that error... |
| 23:32.50 | Shaun2222 | [Jan 17 15:32:32] ERROR[12829]: chan_zap.c:13494 process_zap: Unknown signalling method 'pri_cpe' |
| 23:32.56 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-76-246-191-68.dsl.lsan03.sbcglobal.net) |
| 23:33.02 | jblack | Hello, hello, hello |
| 23:34.56 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 23:36.50 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:39.09 | *** join/#asterisk metfan2007 (n=metfan20@fw.grupositel.com.mx) |
| 23:40.00 | metfan2007 | Hi all, I'm trying to use H323, witch driver do you recomend?? H323? OH323? or OOH323??? I'm using OOH323 (asterisk-addons) but I have some problems |
| 23:40.45 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 23:41.01 | Shaun2222 | errrrrrrrrr |
| 23:42.14 | *** join/#asterisk Patrickz_ (n=patrickz@ppp-124-121-61-164.revip2.asianet.co.th) |
| 23:42.43 | tzafrir_home | oh323 is unmaintained. |
| 23:43.03 | tzafrir_home | Rumour has it that h323 is in the best shape right now |
| 23:43.18 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:43.35 | JT | chan_woomera is meant to be the most stable |
| 23:43.37 | nhuisman_work | so apparently going from asterisk 1.2 to 1.4 fixes the problem of sccp crashing asterisk |
| 23:43.40 | nhuisman_work | rather skinny |
| 23:43.53 | JT | rather skinny? |
| 23:44.00 | nhuisman_work | skinny rather then sccp |
| 23:44.07 | JT | same thing. |
| 23:44.07 | nhuisman_work | chan_skinny |
| 23:44.10 | JT | ah |
| 23:44.10 | nhuisman_work | instead of chan_sccp |
| 23:44.22 | nhuisman_work | stupid ABE |
| 23:44.25 | nhuisman_work | better come out with 1.4 soon |
| 23:44.30 | JT | chan_skinny speaks sccp though :) |
| 23:44.34 | nhuisman_work | yeah :) |
| 23:44.35 | JT | why bother with ABE? |
| 23:44.46 | nhuisman_work | my boss wanted support. |
| 23:45.28 | JT | i'd argue the support is worse |
| 23:45.33 | JT | less help from the community |
| 23:45.50 | tzafrir_home | JTm you provide better support? ;-) |
| 23:46.20 | JT | well many people will be able to readily help out with standard asterisk |
| 23:46.23 | denon | Ive gotta wonder, people actually use asterisk sccp in production? |
| 23:46.40 | denon | I mean, not to diss the efforts or anything .. |
| 23:47.26 | nhuisman_work | the reason i am going to use sccp initially is that if If asterisk has major issues during the first bit of rollout then I can swap back to call manager |
| 23:47.26 | nhuisman_work | if I upgrade all the phones to sip i can't go back |
| 23:47.26 | nhuisman_work | once I see it working I'll one by one upgrade the phones to sip |
| 23:47.30 | denon | why one by one? |
| 23:47.39 | denon | it's easy to mass deploy SIP, then mass deploy sccp back |
| 23:47.51 | nhuisman_work | maybe not one by one |
| 23:47.53 | denon | like 3 lines on your tftp server, and script out a mass rebote |
| 23:47.57 | nhuisman_work | but at least a few at a time to make sure things are ok |
| 23:47.58 | denon | reboot |
| 23:48.10 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:48.17 | denon | spose, though I think the proper approach would be to test with SIP in a lab |
| 23:48.22 | denon | then move into prod, knowing it'll probably work |
| 23:48.28 | nhuisman_work | i'm about to test going back to sccp |
| 23:48.34 | nhuisman_work | our ccm is so old |
| 23:48.38 | nhuisman_work | and it's running in backup mode |
| 23:48.40 | nhuisman_work | read only |
| 23:48.45 | JT | nhuisman_work: why can't the phones go back to sccp? |
| 23:48.56 | denon | somehow I dont think you'll be happy with asterisk using only sccp |
| 23:49.04 | nhuisman_work | i'll explain |
| 23:49.08 | denon | then again, if you're used to call mangler, anything's an improvement |
| 23:49.56 | nhuisman_work | the phones currently run a really old version of sccp, like version 3. Once you upgrade past sip or sscp 5 you are in universal application loader mode and can't downgrade below 5.0. I need to test whether I can use the later versions of sccp with our ccm |
| 23:50.09 | nhuisman_work | i might need to go edit all the SEPXXXXXXXXXXX.cnf.xml files on the ccm tftp server |
| 23:50.16 | nhuisman_work | to change the image to a newer version |
| 23:50.17 | denon | you could load an old sip |
| 23:50.22 | *** join/#asterisk Mmurdock (n=vnjyjta@211.sub-72-121-29.myvzw.com) |
| 23:50.26 | denon | pre-universal loader sips work fine with asterisk |
| 23:50.35 | nhuisman_work | sip 4.4 or whatever? |
| 23:50.38 | denon | yeah |
| 23:50.42 | denon | I dont remember the "good" versions |
| 23:50.46 | denon | but I think the wiki details em |
| 23:50.57 | nhuisman_work | i dunno if I remember but I think long ago i had issues with 4.x |
| 23:51.05 | nhuisman_work | phones rebooting if you pressed too many keys |
| 23:51.16 | denon | I was thinking 3.something was extremely stable for us |
| 23:51.20 | denon | there were a couple flakey builds .. |
| 23:51.31 | denon | but we ran tons of 7960s on asterisk pre-universal boot loader |
| 23:51.36 | nhuisman_work | hmm |
| 23:51.41 | denon | in fact, the universal loader "feels" recent to me |
| 23:51.42 | nhuisman_work | might be worth a shot |
| 23:51.49 | *** join/#asterisk RoyK (n=roy@91.149.13.189) |
| 23:51.53 | denon | it'd be much easier to move back and forth that way |
| 23:51.59 | denon | and the upgrade happens faster if you need to roll back in a hurry |
| 23:52.07 | denon | sip->sccp or vice versa |
| 23:52.33 | nhuisman_work | i'll test a phone with sip->sccp and in reverse |
| 23:52.36 | *** part/#asterisk RoyK (n=roy@91.149.13.189) |
| 23:52.36 | denon | though I guess i'd be surprised if your phone didn't work on an old Call Mangler with a new sccp anyway .. but yeah |
| 23:52.40 | nhuisman_work | to make sure it will still work with ccm |
| 23:52.53 | denon | check out the wiki, Im pretty sure it details the versions |
| 23:53.01 | nhuisman_work | it doesn't say much about 4.x |
| 23:53.16 | denon | perhaps an older rev of the page |
| 23:53.16 | TJNII | I was wondering today ... How could I pick a ringing phone from another phone? Let's say SIP/1216 is ringing and I want to take it on SIP/1214. How could I make an extension which stops 1216 from ringing and sends it to 1214? |
| 23:54.03 | Shaun2222 | err |
| 23:54.07 | Shaun2222 | this is pissing me off now... wtf |
| 23:54.19 | Mmurdock | Lookup pickup groups |
| 23:54.20 | denon | nhuisman_work: we ran tons of 3.01-4 |
| 23:54.59 | nhuisman_work | what distro of linux you running |
| 23:55.06 | denon | uh, where? |
| 23:55.06 | nhuisman_work | now I need to pick one since I can't use ABE for now |
| 23:55.12 | nhuisman_work | for your asterisk boxen |
| 23:55.14 | denon | we run lots of different * boxes |
| 23:55.18 | nhuisman_work | i was thinking fedora |
| 23:55.19 | denon | I personally tend to run a lot of debian |
| 23:55.24 | denon | but it doesnt really matter |
| 23:55.26 | nhuisman_work | yeah |
| 23:55.29 | denon | whatever you're most comfy with |
| 23:55.29 | nhuisman_work | what ever works eh |
| 23:55.35 | denon | debian's a pretty good choice for average |
| 23:55.46 | denon | some people go nuts and build their own asterisk dist |
| 23:55.47 | nhuisman_work | do you install with debian packages or from source |
| 23:55.48 | nhuisman_work | i guess from source |
| 23:55.54 | denon | definitely from svn source |
| 23:56.01 | denon | using svn makes it much easier to update later |
| 23:56.07 | denon | not trunk -- just checking out the tags |
| 23:56.14 | denon | tarballs are wasteful imho |
| 23:56.29 | nhuisman_work | so you just svn import and then go into a tag and build it? |
| 23:56.31 | denon | though I know a lot of effort goes into the packages |
| 23:56.40 | denon | I'm sure they're fine as well |
| 23:56.54 | denon | svn checkout http://svn.asterisk... |
| 23:57.04 | Shaun2222 | JT: i'm still getting that error, any idea wth is going on. |
| 23:57.06 | denon | then svn sw /whatevetag later |
| 23:57.11 | *** join/#asterisk anthm (n=anthm@70-8-116-145.area4.spcsdns.net) |
| 23:57.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:58.28 | TJNII | Mmurdock: Thanks for pointing me in the right direction, I found the pickup command which is exactly what I want. |
| 23:58.38 | Qwell | nhuisman_work: see msg |
| 23:59.04 | JT | Shaun2222: i'm not sure, how did you set it up? |
| 23:59.55 | Shaun2222 | well it was working with 1.4.17 and libpri 1.4.3 |
