| 00:07.37 | *** join/#asterisk eric2 (n=nobody@69.60.247.142) |
| 00:07.50 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 00:09.08 | edibrac | when you've bricked a Cisco 7940 does that mean there's absolutely nothing showing up on the screen? or is it a certain error message like "unprovisioned"? |
| 00:09.50 | *** join/#asterisk docelmo (n=vircuser@pool-151-199-175-104.lyn.east.verizon.net) |
| 00:10.06 | wonderworld | hey, i am trying to build chan_mobile on a debian stable box. make menuconfig doesn't let me choose chan_mobile, probably because something is wrong with my bluetooth installation. i apt-get'ed bluetooth and bluez-* . the bluez version should work, according to the chan_mobile website. am i missing something? |
| 00:17.36 | edibrac | are cisco 79xx phones considered the cream of the crop? |
| 00:17.42 | edibrac | by today's standards |
| 00:20.26 | *** join/#asterisk carpenike (n=ryan@c-98-218-125-247.hsd1.md.comcast.net) |
| 00:20.38 | carpenike | hi can anyone help me get IMAP_TK installed on my system? |
| 00:20.45 | carpenike | I have uw-imap installed |
| 00:20.47 | carpenike | with Kerberos support. |
| 00:23.58 | *** join/#asterisk dgoner (n=david@mx1.repairpc.net) |
| 00:24.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:27.38 | _ShrikE | edibrac: IMHO you can do better than cisco |
| 00:28.25 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 00:28.46 | edibrac | the speakerphone is pretty good. I really dislike the way you need to configure them throught tftp |
| 00:29.08 | eppigy | lol |
| 00:29.10 | edibrac | i can figure it out, just it seems unecessary |
| 00:29.25 | eppigy | not when you have 300 phones |
| 00:29.30 | eppigy | you have to configure identicaly |
| 00:29.34 | eppigy | and make changes on |
| 00:31.43 | edibrac | ok tftp isn't the problem, it's the whole maze of ways you do the firmware upgrade that seems unecessarily complex. |
| 00:33.55 | *** join/#asterisk docelm0 (n=vircuser@pool-151-199-179-251.lyn.east.verizon.net) |
| 00:36.09 | edibrac | XMLDefault.cnf.xml and OS79xx.txt seem redundant |
| 00:36.36 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-db8ffe712794bbc2) |
| 00:41.40 | Qwell | edibrac: to answer your original question - no. not by any stretch of the imagination |
| 00:42.02 | Qwell | and as for the speakerphone being good...well, it *IS* just a Polycom |
| 00:42.35 | Qwell | oh Polycom.. Polycom, Polycom, Polycom. |
| 00:42.45 | Qwell | polycom.com doesn't work, but www.polycom.com does. |
| 00:43.05 | keith4_ | good for them |
| 00:43.11 | keith4_ | i wish more people would stick to that |
| 00:43.30 | Qwell | keith4_: uhh, no |
| 00:44.29 | Qwell | edibrac: Look familiar? http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_ip6000.html |
| 00:46.43 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 00:46.47 | ruben23 | hi |
| 00:48.09 | JAMMAN2110 | I have a USB FM Sender (glorified USB soundcard) anyone ever used one with asterisk in any way? |
| 00:48.27 | Qwell | JAMMAN2110: does it work with alsa? |
| 00:48.37 | JAMMAN2110 | I dont even know what alsa is.. |
| 00:48.48 | Qwell | that's your first step in configuring it |
| 00:49.10 | JAMMAN2110 | *googles* cool :) |
| 00:49.16 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:49.31 | Qwell | after that, it'll work just like any other alsa device |
| 00:49.56 | JAMMAN2110 | "All USB devices that are standards compliant will work." |
| 00:50.14 | Qwell | with is few |
| 00:50.19 | Qwell | which* |
| 00:50.49 | JAMMAN2110 | Yes |
| 00:50.50 | JAMMAN2110 | True |
| 00:51.04 | JAMMAN2110 | Might have to try it in a virtual machine |
| 00:51.52 | ruben23 | after restarting asterisk cannot connect to remote does (/var/run/asterisk.ctl, exist)..what is the command to correct that...? |
| 00:51.58 | keith4_ | make sure you use a virtualization method that lets you pass USB devices to VMs |
| 00:57.16 | JAMMAN2110 | keith4_, of course :) |
| 00:57.45 | carrar | JAMMAN2110, there are laws regarding replaying radio stations as "on-hold' music |
| 00:57.59 | JAMMAN2110 | Its an FM transmitter |
| 00:58.06 | JAMMAN2110 | Not reciever |
| 00:58.07 | carrar | ah |
| 00:58.28 | carrar | gonna air peoples calls? :) |
| 00:59.27 | JAMMAN2110 | No |
| 00:59.29 | JAMMAN2110 | That would be silly |
| 01:00.01 | carrar | unless it's a pager thing |
| 01:00.07 | JAMMAN2110 | Just a thought for a cheap way to implement an overhead intercom |
| 01:00.11 | carrar | yeah |
| 01:00.17 | JAMMAN2110 | :) |
| 01:00.49 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 01:01.32 | dalbaech | ruben23: it might be a silly question, but is asterisk running? |
| 01:01.50 | dalbaech | and can the user you're logged in as see/access the socket file? |
| 01:01.52 | dalbaech | and what distro? |
| 01:04.56 | ruben23 | dalbaech: centos |
| 01:06.33 | dalbaech | ruben23: ok; can your current user see the socket file"? |
| 01:07.29 | ruben23 | dalbaech:yes.. |
| 01:08.57 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-94be20e2b604064b) |
| 01:09.18 | dalbaech | ruben23: is it owned by the user you are, or by asterisk? |
| 01:09.22 | dalbaech | what are the permissions on the socket file? |
| 01:11.18 | dalbaech | here's a quick test... can you run rasterisk as root? |
| 01:11.30 | ruben23 | dalbaech:ill cehck..actually its running..but when i restart it did not run... |
| 01:11.36 | ruben23 | yes im running on root |
| 01:11.58 | dalbaech | ok; if you're root, it should be working regardless of ownership. |
| 01:12.31 | dalbaech | well, in /etc/asterisk/asterisk.conf does the run directory match the directory that the ctl file is in? |
| 01:13.41 | ruben23 | yes..it does.. |
| 01:14.06 | dalbaech | then i'm at a loss. |
| 01:14.37 | ruben23 | dalbaech:ill tried a hard restart |
| 01:21.06 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 01:26.57 | *** join/#asterisk mrsci (n=mrsci@ppp-70-251-250-110.dsl.rcsntx.swbell.net) |
| 01:38.47 | *** join/#asterisk cguerrero (n=cguerrer@200.34.66.137) |
| 01:40.14 | cguerrero | has any one make h232 work with asterisk 1.4.21.1? |
| 01:41.41 | cguerrero | i have no audio when I make a call and after that asterisk crashed |
| 01:45.08 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 01:47.00 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 02:05.59 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 02:10.31 | *** part/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 02:17.15 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 02:19.04 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 02:32.04 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 02:44.11 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:45.48 | *** join/#asterisk StanManCan (n=stan_man@S010600195b3059b4.gv.shawcable.net) |
| 02:46.15 | StanManCan | Is it possibe to make it so if you dial extension 100 it rings an ipphone, and if nobody picks up it forward syou to another number ? |
| 02:46.48 | eric256 | StanManCan: follow me? |
| 02:47.12 | StanManCan | eric256: is that an application in asteisk ? |
| 02:48.06 | carrar | StanManCan, you can cascade calls or dial multiple numbers at the same time or delayed |
| 02:48.19 | carrar | pretty much anything you can think of |
| 02:48.37 | carrar | SKY *IS* the limit |
| 02:48.48 | carrar | Unless you have a ROCKET |
| 02:48.58 | eric256 | StanManCan: dunno its built in to asterisk |
| 02:49.01 | eric256 | err trixbox |
| 02:49.02 | carrar | Then space really is your final Frontier |
| 02:49.34 | StanManCan | carrar: Yea i've been meaning to look into how to make all extensions ring when an incoming call comes |
| 02:49.42 | carrar | Dial() |
| 02:49.55 | StanManCan | carrar: just don't specify an extension to dial ? |
| 02:49.57 | *** join/#asterisk JJ2110 (n=James@219-89-96-244.jetstart.xtra.co.nz) |
| 02:50.03 | carrar | Dial(SIP/100) |
| 02:50.12 | StanManCan | yea but that will only dial that one device |
| 02:50.15 | carrar | yes |
| 02:50.22 | carrar | Dial(SIP/100&SIP/101) |
| 02:50.29 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:50.31 | StanManCan | what happens if you got 30 devices ? |
| 02:50.39 | StanManCan | same thing ? |
| 02:50.42 | carrar | You put 30 in there |
| 02:50.45 | StanManCan | lol |
| 02:50.46 | StanManCan | kk |
| 02:50.53 | carrar | though there are nicer ways to do that |
| 02:51.03 | carrar | but thats the easy way |
| 02:51.16 | carrar | not necessary the most nicest |
| 02:51.51 | carrar | You can dial multiple Local extension |
| 02:52.00 | carrar | then each local dials like 10 extenions |
| 02:52.10 | carrar | after testing them 1st to make sure they are registered |
| 02:52.25 | carrar | and any delays you want to add in |
| 02:53.21 | StanManCan | and is there a certain application i would use to dial a second number if the first one doesn't pick up? |
| 02:53.28 | StanManCan | or is it stil dial |
| 02:53.35 | carrar | the most basic is |
| 02:53.45 | carrar | Dial(SIP/100,10) |
| 02:53.47 | carrar | Dial(SIP/101,10) |
| 02:53.49 | carrar | Dial(SIP/102,10) |
| 02:53.52 | carrar | something like that |
| 02:53.57 | carrar | dials 100 for 10 seconds |
| 02:54.01 | carrar | then rolls over to 101 |
| 02:54.01 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:54.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:54.03 | carrar | etc... |
| 02:54.06 | StanManCan | ah ha |
| 02:54.07 | StanManCan | thank you |
| 02:54.13 | StanManCan | so i'll be going |
| 02:54.15 | carrar | heh |
| 02:54.18 | carrar | ok then! |
| 02:54.52 | hardwire | bows to the digium-open-source-team-lead |
| 02:55.08 | StanManCan | Dial(IAX2/100,10) |
| 02:55.09 | StanManCan | Goto(outbound,100,1) |
| 02:55.09 | thehar | lol |
| 02:55.14 | thehar | hi russellb |
| 02:55.16 | StanManCan | and then make extension 100 in outbound dial my cell phone |
| 02:55.28 | russellb | waves to thehar |
| 02:55.29 | carrar | I would dial my cell from the dialplan |
| 02:55.33 | carrar | after a delay |
| 02:55.54 | carrar | but that works also |
| 02:56.01 | StanManCan | it's still in the dial plan though isn't it? just in a differnet context |
| 02:56.08 | carrar | sure |
| 02:56.10 | thehar | don't test against an iphone.. they take eons to connect |
| 02:56.25 | StanManCan | lol carrar, what would you do ? |
| 02:56.36 | thehar | WWJD |
| 02:56.39 | thehar | lawlz |
| 02:56.51 | carrar | I dial both the desk and cell at the same time, but put a wait 8 seconds before actually dialing the cell |
| 02:57.30 | carrar | giving the desk time to answer befor annouying the person with a cell ring also |
| 02:57.38 | StanManCan | hmm |
| 02:57.48 | carrar | and also allowing both to ring uninterrupted |
| 02:57.56 | StanManCan | ohhh |
| 02:57.57 | StanManCan | good call |
| 02:58.20 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:58.26 | StanManCan | but my incoming context points to my internal, which would then be dialing the extensionand hen i'd need to point to my outgoing context to dial out |
| 02:58.34 | StanManCan | how would i dial out from my internal context? |
| 02:58.40 | StanManCan | without including the outbound |
| 02:58.43 | carrar | let me create a exmaple for you |
| 03:03.19 | carrar | StanManCan, something like this perhaps http://www.osburn.com/example |
| 03:03.30 | carrar | would need to adjust ring times etc.. |
| 03:08.19 | carrar | http://www.uberwoo.com/index.php/Extension_dialing |
| 03:08.21 | carrar | there |
| 03:08.24 | carrar | thats sexier now |
| 03:08.25 | carrar | haha |
| 03:08.35 | StanManCan | lol |
| 03:09.04 | carrar | granted you also need to make sure you are monitoring sip phones to make the ChanIsAvail work properly |
| 03:09.34 | StanManCan | I use IAX :) |
| 03:09.44 | carrar | s/SIP/IAX2/g |
| 03:10.13 | StanManCan | you need to use chanavail for all of them ? |
| 03:10.38 | carrar | If you don't want too those annouying error messages in your CLI |
| 03:10.43 | carrar | ^see |
| 03:10.52 | carrar | and why try to ring something if it's not there |
| 03:11.12 | *** join/#asterisk nOgAnOo (i=Gizmo@network184-253.wctc.net) |
| 03:12.12 | hardwire | lalala |
| 03:12.36 | StanManCan | oh weird I never knew that... I just called the same extension from 2 phone sand one got hung up on |
| 03:12.38 | StanManCan | interesting..! |
| 03:13.44 | eric256 | okay i have a sip phone that registers properly, rings and then once answered it hangs up immediatly (its an X-Lite softphone) |
| 03:13.46 | eric256 | any ideas? |
| 03:13.53 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 03:14.29 | carrar | sure it's hanging up or just no audio? |
| 03:14.41 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.65) |
| 03:14.41 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 03:14.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 03:15.05 | eric256 | pretty sure it hangs up, and if she calls out it works fine |
| 03:15.09 | Micc | does the polycom ip 320 support call appearances? |
| 03:15.14 | StanManCan | How do you reload a conf from rasterisk again :S |
| 03:15.20 | carrar | Codec missmatch? |
| 03:15.29 | carrar | dialpla reload |
| 03:15.32 | carrar | dialplan reload |
| 03:15.45 | StanManCan | and that reloads the extensions.conf and iax.conf |
| 03:15.45 | StanManCan | ? |
| 03:15.47 | carrar | (to reload extensions.conf) |
| 03:15.53 | StanManCan | ah thanks |
| 03:16.13 | carrar | iax2 reload |
| 03:16.15 | eric256 | she also gets audio from the server if she logs in or out etc |
| 03:16.15 | carrar | for iax stuff |
| 03:16.56 | carrar | What does the console say? |
| 03:17.34 | carrar | might binpaste your sip.conf and extension.conf |
| 03:17.38 | carrar | ~binpaste |
| 03:17.42 | stabler | eric256, is it xlite to xlite |
| 03:17.47 | stabler | or |
| 03:17.51 | stabler | a hardphone to xlite |
| 03:18.37 | eric256 | hardphone to xlite and xlite to xlite |
| 03:18.57 | eric256 | ~binpaste |
| 03:19.06 | eric256 | hmm where can i paste it? |
| 03:19.18 | carrar | binpaste.ca |
| 03:19.34 | carrar | http://www.binpaste.com/ |
| 03:19.43 | stabler | www.pastebin.com |
| 03:19.50 | carrar | that works too |
| 03:20.00 | carrar | any nat involved? |
| 03:20.39 | carrar | ~pastebin |
| 03:20.40 | 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 |
| 03:20.42 | carrar | heh |
| 03:20.52 | eric256 | http://www.binpaste.com/v.php?id=p7qvy seems to be the relevant portion |
| 03:20.59 | stabler | nat seems to fail sauce alot of things.. |
| 03:21.49 | eric256 | behinds a linksys router |
| 03:22.05 | carrar | eric256, you work for Mike? |
| 03:22.29 | eric256 | never heard of Mike |
| 03:22.39 | carrar | ah cortland is a company here in Seattle |
| 03:22.41 | carrar | just curious |
| 03:22.56 | harry_v | carrar ever work with call files before? |
| 03:23.14 | carrar | yeah |
| 03:23.23 | stabler | eric256.. thats way over my head wish i could help |
| 03:23.24 | carrar | eric256, thats not enough info, sorry |
| 03:23.24 | stabler | lol |
| 03:23.54 | carrar | try reducing it to it's simplest form |
| 03:23.57 | carrar | toss the agi |
| 03:24.05 | harry_v | what is the correct syntax for adding on more then one sip number in Channell: in the call file? tried a few different combinations like that of Dial in extentions and did not dial the second number. |
| 03:24.24 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 03:24.43 | StanManCan | ohh god carrar your awesome |
| 03:24.46 | StanManCan | it works :) |
| 03:25.02 | stabler | woot success |
| 03:25.17 | carrar | harry_v, you are just joining the outside 'call file' with a internal dialplan, say a extension right? |
| 03:25.27 | carrar | StanManCan, thanks, pay it forward! |
| 03:25.29 | *** join/#asterisk chikkis (n=chikkis@121.243.138.136) |
| 03:25.34 | harry_v | well the typical syntax is SIP/200 |
| 03:25.44 | chikkis | hello everyone |
| 03:25.45 | StanManCan | carrar: i will! :) |
| 03:25.50 | chikkis | i am new here |
| 03:25.56 | StanManCan | carrar: once i'm more familiar at least, ;) |
| 03:25.59 | harry_v | adding on &SIP/201 does not dial the next number |
| 03:26.12 | carrar | not in a callfile |
| 03:26.13 | carrar | no |
| 03:26.39 | chikkis | wow this channel is alive |
| 03:26.42 | *** join/#asterisk etherealite_ (n=evan@adsl-75-35-110-11.dsl.pltn13.sbcglobal.net) |
| 03:27.02 | Talkradio | first time for everything ;) |
| 03:27.02 | stabler | chikkis, did you expect a dead channel? |
| 03:27.07 | carrar | harry_v, your destination I think only takes one argument |
| 03:27.14 | carrar | 'Channel:' |
| 03:27.30 | harry_v | so Channel: is just pointing to a extention or context? |
| 03:27.44 | carrar | once thats connected you can hit multiple extensions on the 'extension & priotry' dial part |
| 03:28.13 | carrar | Channel is pointing to something like Zap/ |
| 03:28.15 | carrar | or SIP/ |
| 03:28.21 | chikkis | no but the channels were so dead |
| 03:28.25 | chikkis | like |
| 03:28.38 | chikkis | "i see dead people" |
| 03:28.51 | harry_v | well I know it can go to individual SIP or ZAP channels |
| 03:28.51 | chikkis | other channels |
| 03:28.59 | carrar | Lets see what you are doing |
| 03:29.03 | carrar | ~pastebin |
| 03:29.04 | 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 |
| 03:29.45 | harry_v | but what about context such as [page-out] exten => 200,1,Dial(SIP/200&SIP/201 |
| 03:29.46 | harry_v | ) |
| 03:30.10 | carrar | yeah? |
| 03:30.19 | chikkis | like work on astersik box |
| 03:30.20 | harry_v | actually it would be something like 300,1, |
| 03:30.39 | chikkis | i am kind not so much of linux kind of guy |
| 03:30.52 | chikkis | i mean i dont know much of linux |
| 03:30.53 | carrar | Thats kind of odd |
| 03:31.04 | carrar | But Kind of ok |
| 03:31.22 | StanManCan | Carrar: the first time i called it works. now when I try to call my number and dial the extension i just made (500) or the extension that just goes tto the desk phone, it says it's not available and please try again |
| 03:31.32 | stabler | chikkis, you should get familar with linux before playing with asterisks |
| 03:31.49 | carrar | what does iax2 show peers say? |
| 03:32.00 | chikkis | i have had my hands dirty on liunx before |
| 03:32.02 | harry_v | actually, can channel: in callfile.call dial a virtual channel to do the multiple sip calls? |
| 03:32.03 | carrar | try removing the Channel checking |
| 03:32.16 | chikkis | but i am rookee |
| 03:32.17 | StanManCan | yea it's not in there |
| 03:32.51 | carrar | harry, you could do the multi channels once it's connected using the 'context/extension/priority' part of the call file, not in the 'Channel' part |
| 03:33.00 | stabler | chikkis, what are you wanting to know? did you have a question? |
| 03:33.12 | chikkis | yeah |
| 03:33.33 | StanManCan | carrar: http://pastebin.com/d7748722d |
| 03:33.50 | chikkis | i think it will be too much to ask compared what knowledge i have on asterisk and linux |
| 03:34.01 | carrar | your second line is wrong |
| 03:34.08 | StanManCan | exten 200 still works (goes to brad2) but 100 (goes to brad) and 500 (goes to brad then cell) don't |
| 03:34.11 | carrar | priority needs to be 1 for the 1st line |
| 03:34.37 | StanManCan | duh, not sure where that came from |
| 03:34.37 | StanManCan | lol |
| 03:34.41 | StanManCan | my bad :S |
| 03:34.46 | chikkis | but however this is what i want to know |
| 03:35.33 | chikkis | i want to configure the High availbility / redundency on the asterisk boxs |
| 03:35.36 | StanManCan | out of curisotiy on exten 500 whats with the ",Hangup()" and 502,Hangup() |
| 03:35.48 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 03:35.51 | carrar | well thats was part of the ChanIsAvil |
| 03:36.04 | carrar | jump 101 priorities if not registered |
| 03:36.06 | carrar | err |
| 03:36.08 | brunner | is there a module I need to load in order to use a wave file as music on hold? |
| 03:36.10 | carrar | reachable I mean |
| 03:36.20 | StanManCan | ohhhh |
| 03:36.24 | stabler | chikkis, have you downloaded "Asterisk: The Future of Telephony" |
| 03:36.26 | stabler | the book |
| 03:36.27 | StanManCan | thank you |
| 03:36.28 | StanManCan | lol |
| 03:36.30 | eric256 | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0"...that shows up in the xlite log, does that mean anything? |
| 03:36.38 | carrar | ~book |
| 03:36.39 | jbot | from memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 03:36.49 | chikkis | yeah i do have it with me |
| 03:36.55 | harry_v | carrar simular to this http://www.pastebin.ca/1335639 |
| 03:37.25 | carrar | pageout isn't a channel |
| 03:37.32 | carrar | thats a context right? |
| 03:37.39 | carrar | or extension |
| 03:37.44 | harry_v | yes context |
| 03:37.53 | carrar | Wants a channel |
| 03:38.01 | harry_v | like 300 |
| 03:38.09 | carrar | like Zap/ |
| 03:38.12 | stabler | chikkis, high avalibility/redundancy to created like any other server on a network |
| 03:38.12 | carrar | or SIP/ |
| 03:38.15 | harry_v | k |
| 03:38.22 | harry_v | so it ignores context |
| 03:38.23 | stabler | *is created |
| 03:38.25 | chikkis | by teaming |
| 03:38.35 | chikkis | the nic |
| 03:38.43 | *** join/#asterisk N|ght (n=N_ght@adsl-76-209-55-213.dsl.emhril.sbcglobal.net) |
| 03:38.57 | chikkis | but what about the active call status |
| 03:39.04 | brunner | I have a wave file in my moh directory, but when execute musiconhold(), I only hear silence. Here's my musiconhold.conf and extensions.conf: http://pastebin.com/m222e1935 |
| 03:39.09 | stabler | chikkis, backup trunks, back up power, etc.. |
| 03:39.18 | carrar | harry_v, connext is a error for that field |
| 03:39.21 | chikkis | or rather active calls when failiure happens |
| 03:39.30 | harry_v | k |
| 03:39.36 | carrar | context != channel |
| 03:39.39 | chikkis | hmmm |
| 03:39.57 | carrar | brunner, wrong format? |
| 03:40.18 | carrar | file youraudiofilehere.wav |
| 03:40.19 | brunner | carrar: I used the exact sox parameters from the asterisk book |
| 03:40.20 | harry_v | so 300/SIP |
| 03:40.21 | carrar | what does that say? |
| 03:40.32 | carrar | -r 8000 ? |
| 03:40.35 | brunner | does anyone have a file that *should* work with asterisk |
| 03:40.37 | brunner | carrar: yes |
| 03:40.45 | carrar | Harry: SIP/300 |
| 03:40.55 | harry_v | yea other way around |
| 03:40.58 | harry_v | :) |
| 03:41.02 | brunner | -r 8000 -c 1 -s -w moh1.wav resample -ql |
| 03:41.17 | carrar | brunner, do: file youraudiofilehere.wav |
| 03:41.29 | chikkis | wow i love this channel |
| 03:41.33 | brunner | carrar: where? command line? |
| 03:41.36 | carrar | yeah |
| 03:41.42 | *** join/#asterisk nbags (n=nbags@60-241-170-44.static.tpgi.com.au) |
| 03:41.45 | carrar | unix cli |
| 03:42.19 | harry_v | going to give this a test. |
| 03:42.20 | brunner | carrar: chris@thinkpad:/var/lib/asterisk/moh$ file moh1.wav |
| 03:42.20 | brunner | moh1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 03:42.21 | harry_v | :) |
| 03:42.31 | carrar | yeah thats good |
| 03:42.34 | chikkis | @stabler may dum thing to ask |
| 03:42.34 | carrar | bad file then perhaps? |
| 03:42.53 | chikkis | but if you have two servers |
| 03:43.04 | StanManCan | Just out of curiosity, is this normal? |
| 03:43.04 | StanManCan | http://pastebin.com/d6fa7ed71 |
| 03:43.09 | chikkis | how do you take care of active calls |
| 03:43.13 | carrar | brunner, you also have to "restart now" for Asterisk to pick that up |
| 03:43.24 | brunner | carrar: plays fine in totem |
| 03:43.26 | brunner | ah, okay |
| 03:43.49 | harry_v | brb, see if this works. |
| 03:44.00 | stabler | chikkis, I dont think you can incorperate that kind of redundancy for active calls.. but im not sure... hopefully someone more knowledgable will chime in |
| 03:44.25 | chikkis | hmm |
| 03:44.52 | nbags | hi. i've got a weird problem. i have an analogue line connected to a ZAP channel, but I have call-divert enabled on that line so that incoming calls actually come via a SIP channel, not a ZAP channel. The problem is that Optus (in australia) indicate that a incoming call is being diverted by ringing the line for just a split second. asterisk doesn't realise that the ringing has stopped and processes this as an incoming call. Does anyone kn |
| 03:45.03 | brunner | carrar: still no luck. call 253-242-8652 to test. |
| 03:45.14 | carrar | 253! |
| 03:45.14 | chikkis | cause i have work on servers from avaya (the communication manager) |
| 03:45.16 | carrar | local! |
| 03:45.19 | carrar | <- Bellevue |
| 03:45.29 | brunner | carrar: <- IPKall |
| 03:45.30 | brunner | =] |
| 03:45.33 | carrar | I'll let you call |
| 03:45.43 | chikkis | and i think the implement some called as plat |
| 03:45.52 | chikkis | "plat" |
| 03:46.03 | carrar | brunner, might check the location of the file |
| 03:46.11 | carrar | and the config for moh |
| 03:46.32 | brunner | carrar: http://pastebin.com/m222e1935 |
| 03:46.37 | stabler | chikkis, Ive never had hands on with a live avaya system |
| 03:46.50 | chikkis | 8 years on them |
| 03:47.02 | stabler | ive only had schooling |
| 03:47.09 | chikkis | cool |
| 03:47.10 | brunner | carrar: the location matches what's in musiconhold.conf, I promise |
| 03:47.14 | stabler | actually in a class im currently taking |
| 03:47.20 | chikkis | some on cisco and nortel also |
| 03:47.24 | carrar | heh |
| 03:47.32 | chikkis | okay |
| 03:47.45 | chikkis | i worked with avaya for 5 years |
| 03:48.01 | carrar | any other files in that direcory? |
| 03:48.10 | chikkis | pretty short time huh |
| 03:48.10 | stabler | yea.. it looks like a decent system.. ive only played with simulations |
| 03:48.25 | chikkis | simulators |
| 03:48.30 | chikkis | which one??? |
| 03:48.36 | *** join/#asterisk etherealite (n=evan@adsl-75-35-76-215.dsl.pltn13.sbcglobal.net) |
| 03:48.38 | brunner | carrar: no |
| 03:48.45 | carrar | brunner, what does the console say? |
| 03:48.52 | carrar | lesseee that |
| 03:48.58 | thehar | seeeeee |
| 03:49.02 | brunner | carrar: nothing. how can I make it more verbose? |
| 03:49.06 | stabler | chikkis, citrix |
| 03:49.12 | carrar | show me anyways |
| 03:49.16 | thehar | i wanna avaya |
| 03:49.17 | thehar | :( |
| 03:49.32 | brunner | Connected to Asterisk 1.4.21.2~dfsg-1ubuntu3 currently running on thinkpad (pid = 10825) |
| 03:49.32 | brunner | thinkpad*CLI> |
| 03:49.34 | chikkis | citrix |
| 03:49.36 | carrar | no |
| 03:49.40 | carrar | in binaste |
| 03:49.41 | carrar | oh |
| 03:49.46 | carrar | set verbose 8 |
| 03:49.47 | brunner | lol, that's all there is |
| 03:49.50 | carrar | then capture that |
| 03:49.51 | chikkis | things are still not clear to me |
| 03:49.57 | stabler | bleh... im way more interested in open source administration |
| 03:50.05 | chikkis | as far i know there were no simulators |
| 03:50.16 | chikkis | me |
| 03:50.20 | chikkis | me too |
| 03:50.22 | chikkis | now |
| 03:50.36 | chikkis | atleast |
| 03:50.42 | chikkis | haha :p |
| 03:50.43 | stabler | uhh.. its not a simulator owned by avaya.. its some third party simulation used by DeVry |
| 03:50.54 | thehar | DeVry? |
| 03:50.56 | brunner | carrar: http://pastebin.com/m203702be |
| 03:50.58 | thehar | hahhHAHAHhahahahahaha |
| 03:51.05 | chikkis | can you give me some link |
| 03:51.06 | stabler | DeVry University |
| 03:51.10 | thehar | snorts |
| 03:51.16 | thehar | "University" |
| 03:51.17 | thehar | hahahahaha |
| 03:51.27 | carrar | moh module is not loaded |
| 03:51.28 | chikkis | i know there is one simulator they are working on now |
| 03:51.31 | stabler | i take it you arnt a fan |
| 03:51.40 | thehar | uhm no. |
| 03:51.56 | thehar | DeVry is as much a "university" as ITT. |
| 03:52.30 | stabler | not so much.. theyre accredited |
| 03:52.33 | brunner | carrar: but I have this in my modules.conf: load => res_musiconhold.so |
| 03:52.48 | carrar | brunner, try "module load func_moh.so" |
| 03:52.58 | thehar | i'd rather go to a community college |
| 03:53.06 | carrar | I think thats the right module |
| 03:53.09 | brunner | [Feb 12 21:53:01] WARNING[10864]: loader.c:655 load_resource: Module 'func_moh.so' already exists. |
| 03:53.10 | chikkis | could you give us the exact link |
| 03:53.12 | stabler | have you attended at DeVry before? |
| 03:53.20 | chikkis | nope |
| 03:53.29 | thehar | no i went to Berekely |
| 03:53.34 | carrar | is res_musiconhold.so loaded? |
| 03:53.47 | stabler | chikkis are you refering to a link to citrix |
| 03:53.50 | brunner | carrar: I guess it thinks so. How can I tell? |
| 03:53.58 | carrar | same command |
| 03:54.19 | stabler | thehar, i currently enjoy the classes |
| 03:54.23 | brunner | carrar: same message |
| 03:54.45 | stabler | thehar, i have had very few issues with there teaching practices |
| 03:54.52 | chikkis | DeVry simulators |
| 03:55.04 | stabler | citrix is a paid service |
| 03:55.17 | thehar | i'm sorry i just don't have a great image of those fast easy technical schools |
| 03:55.27 | chikkis | well i only attened high school |
| 03:55.30 | chikkis | only |
| 03:55.36 | carrar | brunner, try MusicOnHold(default) |
| 03:56.04 | stabler | http://www.citrix.com/lang/English/home.asp |
| 03:56.08 | carrar | (in your dialplan) |
| 03:56.35 | brunner | trying |
| 03:57.09 | brunner | <PROTECTED> |
| 03:57.16 | stabler | only reason devry is "fast" is because they attend year round |
| 03:57.28 | chikkis | here is the link of avaya simulators |
| 03:57.29 | chikkis | http://support.avaya.com/japple/css/japple?PAGE=Document&temp.productID=235561&temp.bucketID=108020&temp.documentID=284104&temp.selectedRelease=235562 |
| 03:57.41 | chikkis | it is called aes-cm |
| 03:57.45 | stabler | chikkis, citrix is just an application delivery service |
| 03:58.00 | chikkis | versioned as 3.0 |
| 03:58.17 | chikkis | hmm |
| 03:58.28 | *** part/#asterisk nbags (n=nbags@60-241-170-44.static.tpgi.com.au) |
| 03:58.39 | *** join/#asterisk yoanis (n=fred@200.55.139.218) |
| 03:58.42 | stabler | i wish i could get into the open source field... |
| 03:58.44 | yoanis | hi there |
| 03:58.52 | stabler | just cant get my foot in the door |
| 03:59.00 | yoanis | i i need some help figuring out why voicemail are not been recorded |
| 03:59.07 | stabler | *open source administration |
| 03:59.50 | carrar | brunner, what cards do you have in your asterisk box? |
| 04:00.15 | stabler | no one wants to take in an unexperienced admin/tech |
| 04:00.16 | *** join/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 04:00.23 | brunner | carrar: none |
| 04:00.33 | carrar | using usb for timing? |
| 04:01.03 | brunner | carrar: whatever that module is that the book told me to use when I don't have a digium pci card |
| 04:01.19 | brunner | carrar: I'm trying to develop a dialplan on my laptop, lol |
| 04:02.07 | chikkis | me too |
| 04:02.44 | chikkis | for me it more i am scared to enter into it |
| 04:02.44 | stabler | hehe |
| 04:02.59 | stabler | i have alittle fear |
| 04:03.21 | chikkis | that it will be shot-in-my-foot |
| 04:03.30 | chikkis | hahaha |
| 04:03.39 | stabler | but im sure i would get over it as i posses alot of knowledge im just unsure of myself alot |
| 04:04.21 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net) |
| 04:04.21 | brunner | carrar: for use on a co-located box |
| 04:04.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:04.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:04.30 | chikkis | but i am sure i can beat you on being unsure on myself |
| 04:04.34 | brunner | carrar: any ideas? |
| 04:04.57 | chikkis | i have iso image of avaya cm with me |
| 04:05.05 | chikkis | right now |
| 04:05.24 | chikkis | and i am trying to emulate it on the vmware |
| 04:05.25 | carrar | brunner, I remember something about needing a timing source |
| 04:05.26 | chikkis | but |
| 04:05.36 | carrar | but it's been a while as all my boxes have t1 cards |
| 04:05.47 | chikkis | i am getting stuck with File system failures |
| 04:05.58 | chikkis | MBR failures |
| 04:06.07 | chikkis | it sucks big time |
| 04:06.17 | stabler | :/ |
| 04:06.19 | brunner | carrar: the book seems to think it will work with the timing provided by that kernel module |
| 04:06.32 | carrar | show application MusicOnHold |
| 04:06.34 | carrar | that works? |
| 04:06.42 | chikkis | if any one wanna have look at it let me now cause i can show you the installation proccess |
| 04:06.58 | chikkis | congrats carrar |
| 04:07.19 | eric256 | fyi i setup the stun server to point to a real stun server and it fixed my one way audio issues, musta been some kind of NAT issue |
| 04:07.21 | brunner | carrar: yes, that works |
| 04:07.39 | carrar | brunner, I am sure it's something obvious I am just missing |
| 04:08.00 | stabler | darn nat |
| 04:08.02 | chikkis | i can give access to my system to have frist hand assult on self pride kicked to dirt |
| 04:08.30 | stabler | hehe |
| 04:08.37 | stabler | what are you trying to install? |
| 04:08.57 | brunner | carrar: =/ |
| 04:09.12 | chikkis | i wanna to work more on some gateways and stuff like that |
| 04:09.13 | brunner | pages Corydon76-dig |
| 04:09.45 | chikkis | and try integrate with other biggee like cisco and nortel |
| 04:10.27 | chikkis | i have complete lab of cisco cm on my system |
| 04:10.54 | chikkis | eric did have nat on both ends |
| 04:10.59 | stabler | and you want to interface asterisk with cm? |
| 04:11.45 | chikkis | yeah that is also one of the plans i wanna try |
| 04:12.28 | chikkis | well have done interface with trixbox once before |
| 04:13.00 | stabler | i plan to play with trixbox soon |
| 04:13.12 | carrar | brunner, ensure you having a timing source |
| 04:13.13 | eric256 | chikkis: nope, just nat on one end |
| 04:13.13 | chikkis | it is cake walk |
| 04:13.19 | chikkis | hmmm |
| 04:13.20 | brunner | carrar: how can I check that? |
| 04:13.33 | carrar | well it was like ztdummy in the older version |
| 04:13.54 | brunner | uhg, gotta run. taxi waiting. bbl |
| 04:13.59 | chikkis | @eric if you have nat on both the ends nat fails what what i have seen |
| 04:14.04 | brunner | carrar: thank you so much for your help! |
| 04:14.09 | carrar | come back |
| 04:14.13 | carrar | when you are done |
| 04:14.15 | chikkis | your link keeps flaping |
| 04:14.17 | brunner | will do |
| 04:14.17 | carrar | I'll find it |
| 04:15.02 | chikkis | okay here is one of the most major project i want to try |
| 04:15.23 | chikkis | take a router simulator of cisco |
| 04:15.36 | chikkis | like dynamips / gns3 |
| 04:15.47 | *** part/#asterisk yoanis (n=fred@200.55.139.218) |
| 04:15.48 | chikkis | and add telephony part to it |
| 04:16.27 | chikkis | so that i can simulate the funcational aspect of telephoney on the simulator |
| 04:16.32 | carrar | chikkis, hi |
| 04:16.43 | carrar | I missed your message :) |
| 04:17.16 | stabler | chikkis, sounds interesting |
| 04:17.37 | chikkis | and i thinking of doing this buy taking the TAPI code and |
| 04:17.41 | chikkis | filling in |
| 04:17.56 | chikkis | by then i will be dead |
| 04:17.58 | chikkis | hahaha |
| 04:18.03 | stabler | lol |
| 04:18.20 | *** join/#asterisk yoanis (n=fred@200.55.139.218) |
| 04:18.42 | chikkis | nothing carrar me just being stupid |
| 04:18.43 | yoanis | is there a way to send 'voicemail' from command line |
| 04:18.44 | yoanis | ? |
| 04:18.58 | chikkis | and nominating my self to darwin awards |
| 04:19.01 | chikkis | :P |
| 04:19.09 | yoanis | like reading from a wav file and forwarding it to a voicemail |
| 04:19.58 | chikkis | so @stabler |
| 04:20.31 | chikkis | i am still try to figure the citrix simulator part ' |
| 04:20.41 | chikkis | looks very interesting |
| 04:23.09 | stabler | i basicly just log into the citrix server |
| 04:23.17 | chikkis | oh okay |
| 04:23.24 | chikkis | got it |
| 04:23.25 | stabler | and it is a software based simulation |
| 04:23.45 | stabler | im sure devry pays a pretty penny for it |
| 04:24.09 | chikkis | when you say software simulation |
| 04:24.13 | chikkis | what do you mean |
| 04:24.32 | chikkis | you mean like you do "remote desktop" |
| 04:24.46 | drmessano | Using "Citrix" doesnt make anything a friggin "Simulator" |
| 04:25.04 | chikkis | beacause to critix is used for that |
| 04:26.40 | chikkis | @drmessano i am little confused here |
| 04:26.57 | drmessano | Me too |
| 04:28.25 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 04:28.26 | chikkis | DDC |
| 04:28.37 | chikkis | is it remote desktop |
| 04:29.06 | *** part/#asterisk eric256 (n=Administ@229.sub-70-215-60.myvzw.com) |
| 04:32.45 | stabler | uhh its basicly.. i think its called "ip office manager" or something like that |
| 04:33.34 | chikkis | you acctuly download "ip office" image for free |
| 04:33.43 | chikkis | let me give you guys a link |
| 04:33.43 | stabler | yup |
| 04:33.51 | stabler | probably |
| 04:33.57 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:34.01 | chikkis | ip office soho sytem |
| 04:34.34 | stabler | i dont know why we work through citrix |
| 04:35.21 | chikkis | a question here |
| 04:35.24 | *** join/#asterisk jablko (n=jablko@gallery/soc/jablko) |
| 04:35.42 | chikkis | do you get the login prompt directly |
| 04:36.03 | chikkis | or you login to some windows/linux system |
| 04:36.04 | stabler | i think so |
| 04:36.10 | stabler | login directly |
| 04:36.25 | stabler | but im pretty sure it starts a vm |
| 04:36.31 | chikkis | and the you use Avaya site administration tool |
| 04:36.39 | chikkis | hmmm |
| 04:37.08 | stabler | no |
| 04:37.15 | stabler | just ip office manager |
| 04:37.16 | chikkis | did have look at the lin whihc i gave for "AES-CM" avaya simulator |
| 04:37.24 | chikkis | ok |
| 04:37.42 | stabler | i was alittle disappointed that we didnt go more in depth |
| 04:38.26 | chikkis | i have seen enterpise installation and managed them right from citi gruops |
| 04:38.36 | chikkis | to JPMC's |
| 04:38.38 | stabler | do you currently work as an avaya engineer/admin |
| 04:38.50 | *** join/#asterisk Gopaul (n=chatzill@59.97.121.76) |
| 04:39.07 | chikkis | more on voice consulatant |
| 04:39.16 | stabler | oh |
| 04:39.24 | chikkis | cause i moved on to management levels |
| 04:39.25 | chikkis | <PROTECTED> |
| 04:39.35 | chikkis | 15 yers on this work |
| 04:39.43 | jablko | the first line of my dialplan needs to check an expression, and exit if it is true |
| 04:39.47 | chikkis | i am from india by the way |
| 04:39.48 | jablko | is there a convention for this? |
| 04:40.03 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net) |
| 04:40.18 | stabler | wow india |
| 04:40.32 | stabler | interesting |
| 04:40.43 | chikkis | @drmessano DDC |
| 04:40.58 | jablko | basically i do not want asterisk to pickup if the callerid is null |
| 04:41.38 | chikkis | what it is saved a file on desktop |
| 04:41.59 | jablko | i am toying with s,1,GotoIf(${ISNULL(CALLERID())}?something) |
| 04:42.08 | chikkis | still a blank file |
| 04:42.13 | jablko | but i am not sure what something i should use |
| 04:42.28 | chikkis | yep |
| 04:42.40 | chikkis | @stabler where you from? |
| 04:43.00 | stabler | US |
| 04:43.11 | chikkis | cool |
| 04:43.21 | stabler | not really.. lol |
| 04:43.41 | chikkis | hmmm |
| 04:44.07 | stabler | its ok |
| 04:44.13 | stabler | i suppose |
| 04:45.57 | chikkis | "land of oppertiunity " |
| 04:47.36 | stabler | yea.. im a poor SOB.. lol |
| 04:48.05 | stabler | scraping along barely affording life |
| 04:50.02 | chikkis | so i am dude |
| 04:50.53 | chikkis | but i am sure richness is sickness |
| 04:51.25 | stabler | i dont want to be rich by any means.. i just want to comfortably afford life |
| 04:51.28 | jaytee | jablko, something would be the named priority that you Goto IF the condition resolves true otherwise it will simply go to the next priority in that context |
| 04:51.56 | chikkis | http://support.avaya.com/japple/css/japple?PAGE=Document&temp.productID=235561&temp.bucketID=108020&temp.documentID=284104&temp.selectedRelease=235562 |
| 04:52.10 | chikkis | here is the link for avaya simulator |
| 04:52.28 | chikkis | for enterprise system |
| 04:52.31 | jablko | jaytee: what named priority do folks usually use if they want to exit from the dialplan? |
| 04:53.26 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 04:53.44 | jaytee | jabklo, something is just a word. you could have 9999,n(something),Hangup() if you wanted. |
| 04:56.08 | jablko | jaytee: there is no common practice - like "i" for invalid, "t" for timeout... |
| 04:56.23 | jaytee | labeled priorities allow you to jump to that priority by using the label name of the priority. Earlier versions of Asterisk didn't have the n priority and had to be in numeric sequence so if you modified or added lines to a context you had to renumber all the priorities. |
| 04:59.10 | jaytee | i and t are special extensions in Asterisk. |
| 04:59.47 | jaytee | http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions |
| 05:01.11 | jablko | there is no "exit" extension? something i could put in GotoIf() to simply exit from the dialplan? |
| 05:01.30 | jablko | or an alternative application which would exit from the dialplan when an expression is true? |
| 05:02.27 | jaytee | jablko, what about h? |
| 05:03.08 | jaytee | or just add a line at the end of the context, exten => "extensionnumber",n(something),Hangup() |
| 05:03.20 | chikkis | here is the link to download "IP office" |
| 05:03.22 | chikkis | http://www.tek-tips.com/viewthread.cfm?qid=1512189&page=1 |
| 05:03.36 | chikkis | there are some files |
| 05:03.43 | jablko | jaytee: hey, thanks - there is an "h" standard extension |
| 05:03.44 | chikkis | and an iso image |
| 05:03.48 | jablko | did not realize |
| 05:07.21 | chikkis | anyone wanna have look at the installer |
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| 05:40.19 | tamseel | hi all |
| 05:41.00 | tamseel | i want to update the asterisk but i have g729 on the server is this update will create some problem to g729 installed on the server? |
| 05:44.01 | carrar | maybe |
| 05:47.09 | carrar | did you read the readme file? |
| 05:47.10 | carrar | http://downloads.digium.com/pub/telephony/codec_g729/README |
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| 06:24.43 | xacatecas | hi, is it possible to configure an IAX2/SIP user to be allowed from a set of IPs? Eg, 192.168.0.1 or 192.168.0.15 or 192.168.0.78 (as an example)? |
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| 06:27.24 | kaldemar | xacatecas: yes, take a look at the permit parameter |
| 06:28.17 | xacatecas | ok. that solves that. is it possible to construct a peer with multiple host= lines such that it'll load balance the calls to those destinations? |
| 06:28.36 | xacatecas | or do I need to create multiple peers and then do some special handling in the dialplan? |
| 06:29.29 | [TK]D-Fender | xacatecas: No |
| 06:29.43 | [TK]D-Fender | xacatecas: All dialplan |
| 06:30.09 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 06:30.42 | xacatecas | [TK]D-Fender, i faintly recall there being some function that can tell me how many calls is currently going out over a specific peer definition - is this the case and can you maybe recall the function name off the top of your head? |
| 06:30.43 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 06:31.23 | [TK]D-Fender | xacatecas: Not peer specific, but dialplan control specific "core show functions like GROUP" |
| 06:32.29 | *** join/#asterisk medjr (n=medjr@41.224.105.66) |
| 06:32.30 | xacatecas | put each "peer" in it's own group and then do GROUP_COUNT ? |
| 06:32.47 | xacatecas | actually no, each channel ... |
| 06:41.20 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 06:41.34 | [TK]D-Fender | checkout time, later all |
| 06:50.05 | aiksa[LV] | exit |
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| 06:57.14 | yoanis | hello |
| 06:57.42 | yoanis | i'm running asterisk 1.4.21.2 and when a message is recorded file permissions are set to 0006 |
| 06:58.16 | yoanis | which causes a conflict because the file is owned by the asterisk user |
| 06:58.40 | yoanis | and then the voicemail app is unable to read recordings |
| 06:58.52 | yoanis | is this a bug? |
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| 08:04.46 | Signius | I have compiled and installed the zaptel drivers with the following: make clean * ./configure * make * make install * make config and then configured zaptel.conf but for some reason i dont have /etc/asterisk/zapata.conf |
| 08:05.16 | Signius | This is my first time trying to setup and install asterisk so if i have done something dumb i dont know what it is |
| 08:08.06 | *** join/#asterisk unasi7 (n=unasi7@84-75-21-204.dclient.hispeed.ch) |
| 08:08.30 | ultrav1olet | why doesn't exten => 1000,1,VoicemailMain(${CALLERID(num)},s) work as expected? |
| 08:09.23 | ultrav1olet | I want anyone who calls 1000 to be able to work with his/her mailbox immediatly without entering number and password ... but asterisk still asks for a number and password |
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| 08:17.24 | tzafrir_laptop | Signius, it is now called chan_dahdi.conf |
| 08:17.32 | tzafrir_laptop | What version of Asterisk do you have? |
| 08:17.40 | kaldemar | Signius: zapata.conf is (or was) a part of asterisk. it got removed from the source package in favor of chan_dahdi.conf. |
| 08:18.20 | *** join/#asterisk und3r (n=und3r@213-140-22-122.ip.fastwebnet.it) |
| 08:18.24 | und3r | hello |
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| 08:18.37 | Signius | tzafrir_laptop: I have asterisk-1.4.23.1 |
| 08:19.01 | tzafrir_laptop | The file in samples is chan_dahdi.conf . |
| 08:19.13 | tzafrir_laptop | In your case, just rename it to zapata.conf |
| 08:19.13 | und3r | is there anyone who can help me with a problem with dahdi pri on E1? |
| 08:19.47 | tzafrir_laptop | Though if this is a new installation, DAHDI instead of Zaptel is something to consider |
| 08:19.53 | ultrav1olet | what about my question? ;) |
| 08:20.12 | Signius | tzafrir_laptop: Thank you ......Should i have gone for a more recent Asterisk release ? I went with 1.4 coz it said current |
| 08:20.22 | tzafrir_laptop | und3r, maybe. Depending on the details :-) |
| 08:21.21 | tzafrir_laptop | ultrav1olet, what does happen? |
| 08:21.40 | tzafrir_laptop | When a call comes in, what do you see in the CLI trace? |
| 08:22.02 | ultrav1olet | Now I see I'm wrong ;) |
| 08:22.25 | kaldemar | ultrav1olet: and show voicemail.conf |
| 08:22.42 | ultrav1olet | <PROTECTED> |
| 08:23.04 | ultrav1olet | is there a variable to extracts user's mailbox number from iax2.conf? |
| 08:24.44 | ultrav1olet | and one more question: what is the point of mailbox in iax2.conf user configuration? |
| 08:24.48 | tzafrir_laptop | CALLERID(num) gives you that name? |
| 08:24.55 | kaldemar | quite strange to feed a string to CALLERID(num). |
| 08:24.58 | ultrav1olet | tzafrir_laptop: yes, it should |
| 08:25.13 | kaldemar | num is for numbers, name for names. |
| 08:25.41 | ultrav1olet | My iax2.conf lacks any number except mailbox |
| 08:25.44 | tzafrir_laptop | And 'ultraviolet' is a name, rather than a number |
| 08:25.48 | ultrav1olet | I see |
| 08:26.13 | ultrav1olet | so, how can I resolve this conundrum? |
| 08:26.14 | kaldemar | what is iax2.conf? |
| 08:26.24 | ultrav1olet | I meant iax.conf ;) |
| 08:26.55 | kaldemar | do you have ultraviolet => ... in voicemail.conf under [default]? show some facts so we don't have to guess what's wrong. |
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| 08:29.10 | kaldemar | ultrav1olet: show something relevant to your problem. |
| 08:30.15 | ultrav1olet | http://pastebin.ca/1335831 |
| 08:31.45 | ultrav1olet | That way it doesn't work - it still asks voicemail number and password |
| 08:33.23 | kaldemar | and you have tested it with that particular user? |
| 08:34.44 | ultrav1olet | yes |
| 08:35.39 | kaldemar | in stead of yes, you should be showing an output of a call... |
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| 08:36.28 | ultrav1olet | http://pastebin.ca/1335832 |
| 08:36.52 | kaldemar | your callerid parameters are wrong |
| 08:37.05 | ultrav1olet | I see that |
| 08:37.12 | kaldemar | it's "name" <number>, NOT "number" <name>. |
| 08:37.58 | kaldemar | you're trying to access a mailbox called "birdie", but you don't have one since the mailboxes are named with numbers. fix the caller id's. |
| 08:38.00 | ultrav1olet | Is my iax.conf is also wrong? |
| 08:38.22 | ultrav1olet | I see, wait a minute |
| 08:38.27 | kaldemar | your iax.conf is the ONLY thing that is wrong. |
| 08:38.40 | ultrav1olet | I got that |
| 08:39.22 | ultrav1olet | so, callerid must be something like "Peter Pan" <NNN>, right? |
| 08:39.54 | kaldemar | yes |
| 08:40.26 | *** join/#asterisk mvanbaak (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 08:40.51 | ultrav1olet | thanks! |
| 08:41.14 | *** join/#asterisk mvanbaak (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 08:42.27 | und3r | is there who anyone can help me? i've a problem to reach all numbers start with 199 from a pri E1 with DAHDI... |
| 08:42.40 | und3r | -- Executing [s@macro-call-milano:2] Dial("SIP/449-b68a17a8", "DAHDI/g1/199309241|48|tT") in new stack |
| 08:42.43 | und3r | -- Requested transfer capability: 0x00 - SPEECH |
| 08:42.45 | und3r | -- Called g1/199309241 |
| 08:42.48 | und3r | -- DAHDI/4-1 is proceeding passing it to SIP/449-b68a17a8 |
| 08:42.50 | und3r | -- DAHDI/4-1 is making progress passing it to SIP/449-b68a17a8 |
| 08:42.55 | und3r | -- Channel 0/4, span 1 got hangup request, cause 1 |
| 08:42.56 | kaldemar | ~pb |
| 08:42.57 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 08:42.58 | und3r | -- Hungup 'DAHDI/4-1' |
| 08:43.00 | und3r | == Everyone is busy/congested at this time (1:0/0/1) |
| 08:43.05 | und3r | ops sorry :) |
| 08:45.13 | kaldemar | und3r: find out what numbers the other end accepts and wants to route your call. cause 1 means unallocated number. |
| 08:50.05 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
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| 08:54.51 | und3r | kaldemar: there's a FAQ about the "causes" that describe each error number? |
| 08:57.36 | kaldemar | und3r: http://www.google.fi/search?q=q.931+cause+codes :) |
| 08:59.54 | und3r | kaldemar: tnx! :) |
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| 09:04.47 | und3r | kaldemar: the strange thing is that the customer say that he was able to call that number with his old PBX... |
| 09:05.20 | und3r | i think there's someting that i've wrong in configuration.. |
| 09:06.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:06.20 | und3r | or is possible that there are some tuning for numbers that have an IVR responder? |
| 09:06.33 | mort_gib | Tuning?? |
| 09:07.47 | und3r | sed -e s/tunign/particular configuration parameters/ :) |
| 09:08.20 | mort_gib | Why would you need that??? |
| 09:08.33 | kaldemar | Nugget: customers are known to say strange things. you can't know what the old PBX has done to the number before sending it forward. |
| 09:09.48 | brunner | how do I determine what my installation is using as a timing source? |
| 09:10.05 | und3r | kaldemar: right. but in that case i don't know how can i solve my problem :/ |
| 09:12.02 | kaldemar | und3r: contact who ever is controlling the other end and ask how they route numbers. you might need to do some number translation, e.g. prefix the number with something. |
| 09:13.12 | brunner | ztdummy is an adequate timing source for music on hold, correct? |
| 09:14.40 | und3r | kaldemar: i'll try, thank you very muxh! |
| 09:14.48 | unasi7 | simple question: when i place a register in sip.conf, in which context will asterisk try to find my extension? Now all register go to the same context? how can i define the context? |
| 09:15.58 | *** join/#asterisk jermey_g (n=me@static-213-115-44-229.sme.bredbandsbolaget.se) |
| 09:16.21 | jermey_g | is this still a problem in asterisk http://archives.free.net.ph/message/20080618.080019.66a0b6cd.en.html#asterisk-users |
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| 09:21.06 | unasi7 | again: in which context asterisk search for the extension defined in a sip.conf register? |
| 09:23.09 | jermey_g | unasi7:who is again? |
| 09:23.11 | kaldemar | unasi7: be more specific. do you mean a register statement? |
| 09:23.15 | jermey_g | giigles |
| 09:23.42 | jermey_g | probably he means the /incoming at the end of register |
| 09:23.48 | kaldemar | jermey_g: is that really a problem or just an attempt to use a function in a way that it doesn't work? |
| 09:24.35 | jermey_g | it seems to be a problem because option G is never suppose to hangup any party |
| 09:24.38 | jermey_g | but it does in 1.6 |
| 09:25.11 | kaldemar | of course it is supposed to if the dialplan makes it hang up. |
| 09:25.21 | jermey_g | dialplan doesn't make it hangup |
| 09:25.49 | jermey_g | read second last box on this http://voip-info.org/wiki/index.php?page_id=71&tk=6fe53416e9de28d4be98&comments_page=1 |
| 09:26.12 | kaldemar | a dialplan doesn't need an explicit Hangup for a hangup. |
| 09:26.13 | jermey_g | thats a very old paste though and the problem is of a different nature though |
| 09:26.20 | jermey_g | nopes |
| 09:26.58 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:27.34 | jermey_g | in my case, Dial(A,,G(4)); at 4, just store callerid in a variable, at 5,again store callerid in a different variable. thats it. the call hungup for no reason. the callerid did get stored but it executed 4,5 twice |
| 09:27.35 | kaldemar | what are you trying to achieve? |
| 09:27.40 | unasi7 | kaldemar, in sip.conf i have in [general] 3 registers. One sould end up at [default] extension 1670 another should end up in [anothercontext] 1671. But all registers end up now in the same context. Know what i mean? |
| 09:28.02 | jermey_g | kaldemar:actually it was not the callerid, it was the channel name i needed. |
| 09:28.19 | jermey_g | channel towards a party, store in a different variable |
| 09:28.30 | jermey_g | channel towards b-party store in a different variable |
| 09:28.40 | jermey_g | G has some issue i think |
| 09:28.41 | kaldemar | unasi7: they are just registers to let the other side know where you are. to handle calls in different contexts, use peer contexts. |
| 09:29.36 | jermey_g | kaldemar:how long have you worked with *, since which version |
| 09:29.42 | kaldemar | jermey_g: where what happens after 5 in your dialplan? where and how are you trying to store the caller id? |
| 09:29.48 | unasi7 | kaldemar, okay.. will google peer contexts |
| 09:29.51 | unasi7 | kaldemar, thx |
| 09:29.55 | kaldemar | jermey_g: since version 0.7.something |
| 09:30.08 | jermey_g | kaldemar:wait a sec |
| 09:31.25 | xacatecas | ok, a funny question, normally if I do a ring group I basically do Dial(SIP/1&SIP/2&SIP/3,${timeout}) ... now I can easily add SIP/prov/extnumber in there too, however, generally when I dial extnumber it goes through a hunt sequence of sorts to detemine which outbound "trunk" to use ... is it possible to perform this hunting via some other mechanism whilst the local SIP/* accounts are ringing and if that channel answers before any of |
| 09:31.28 | xacatecas | <PROTECTED> |
| 09:31.30 | xacatecas | that's too long ... |
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| 09:34.05 | *** join/#asterisk [psy] (n=psy0rz@lounge.datux.nl) |
| 09:34.15 | kaldemar | xacatecas: you could use a Local/exten@context to do additional decicions in the dialplan while the other peers are getting called. was that what you meant? |
| 09:34.53 | xacatecas | I'm not familiar with Local/. where can I find more info? |
| 09:34.57 | Gido-E | we just recorded our regression test of our Asterisk server: http://video.google.nl/videoplay?docid=3226117917363953075&hl=nl |
| 09:35.20 | xacatecas | does this imply I can do something like Dial(SIP/1&SIP/2&Local/${extnumber}@from-internal,${timeout}) ? |
| 09:35.41 | jermey_g | kaldemar:it was basically like this, Dial(SIP/someuser,,G(5)); At 5,store the channel name in db/f/1 for one party; At 6, store the channel name of other party in db/f/2 |
| 09:36.40 | *** join/#asterisk kjs (i=kjs@fedora/kjs) |
| 09:36.50 | jermey_g | but in the cli, the G option jumped to 5 and then 6. but it then again goes to 5 and 6. so 5 and 6 were exected twice in a row. |
| 09:37.11 | kaldemar | xacatecas: Local creates a pseudo channel which sort of dials the given extension in your dialplan. http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels |
| 09:37.11 | jermey_g | Gido-E:what did you use |
| 09:37.29 | kjs | Hi guys, I have setup a call queue, atm it plays hold music when people are in the queue, I want it to just ring like a normal phone would, is there a sound file i can use for this? |
| 09:38.15 | kaldemar | xacatecas: and yes, you can do that. |
| 09:38.31 | *** join/#asterisk freckle (n=chatzill@195.74.96.118) |
| 09:38.42 | Gido-E | jermey_g it is not just one tool. |
| 09:38.47 | xacatecas | kaldemar, looks exactly like what i want ... only one risk - that of hitting voicemail too early :( |
| 09:38.55 | xacatecas | at least, that I can think of. |
| 09:39.14 | kaldemar | xacatecas: make a new context that doesn't hit voicemail. |
| 09:39.56 | Gido-E | we will boost the regressen test to a 10K useraccount etc... test. Lately a lot of crashes in production environments. :-( |
| 09:41.11 | kaldemar | jermey_g: still i'd like to see the actual dialplan and know what you expect it to do. |
| 09:41.24 | xacatecas | kaldemar, can't be controlled if you dial outbound to things like cell phones I think ? |
| 09:42.04 | kaldemar | xacatecas: sure, that's always a risk when dialing outside. |
| 09:42.19 | xacatecas | for local extensions i'll always just use the direct account for this specific application. |
| 09:42.41 | xacatecas | ok well, it's good to know about Local/ but I'm guessing that I'm going to use it sparingly. |
| 09:42.56 | xacatecas | i don't quite follow the difference between /n and without it though. |
| 09:43.34 | kjs | anyone know of a sound file for a phone ringing noise? |
| 09:44.15 | [psy] | jermey_g various tools and a lot of scripting (pjsua as sipclient) |
| 09:44.54 | [psy] | we found a reproducable segfault in meetme, in asterisk 1.4.23.1 |
| 09:45.09 | kjs | inbound ringing sound |
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| 09:53.38 | xacatecas | can anybody think of a better name for the trixbox daynight mode feature thing? the name is crap imho as it's not very generic. |
| 09:54.46 | xacatecas | and no, i'm not working on trixbox ... looking to get my last install migrated away from it. |
| 09:55.16 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 09:56.00 | *** part/#asterisk ultrav1olet (n=telnet@94.180.49.133) |
| 09:57.19 | *** join/#asterisk Thorn (n=Thorn@unaffiliated/thorn) |
| 10:16.11 | jermey_g | Gido-E:which toolS :) |
| 10:16.20 | jermey_g | i used sipp and winsip |
| 10:17.37 | Gido-E | Fri 13-Feb-09 10:44 < [psy]> jermey_g various tools and a lot of scripting (pjsua as sipclient) |
| 10:19.13 | *** join/#asterisk medjr (n=medjr@41.226.178.114) |
| 10:19.27 | [psy] | jermey_g we used pjsua (gido and i) |
| 10:19.40 | [psy] | however its a bitch to wrap shell script around |
| 10:20.22 | [psy] | i'll have a look at sipp too |
| 10:21.13 | [psy] | i wrote the tests in such a way that it should be easy to use another sipclient |
| 10:32.41 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 10:33.23 | jermey_g | [psy]:lemme know of other various tools you used. cuz i did this testing for a big telco and sipp and winsip worked. sipp is best at inter-operability while winsip rocks when it comes stress testing. winsip is commercial. |
| 10:33.56 | [psy] | there is also sipsak for stresstesting i believe |
| 10:34.01 | [psy] | but it cant make actual calls |
| 10:39.38 | *** part/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
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| 10:39.50 | *** part/#asterisk elsego (n=jc@unaffiliated/elsego) |
| 10:42.03 | jermey_g | i used sipsak but its very primitive compared to sipp |
| 10:42.39 | jermey_g | [psy]:what results did you get. thruput? |
| 10:43.11 | [psy] | ah k |
| 10:43.19 | [psy] | we didn't stresstest and performance test yet |
| 10:43.42 | [psy] | we have 2 tests now: one that uses webinject to test and configure via asterisk_gui |
| 10:43.56 | [psy] | and one that tests if the configuration actually works using pjsua |
| 10:44.36 | [psy] | we will start on the thrird test soon, that will test performance and stress |
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| 11:04.16 | freckle | does anyone know of a way to stream live a phone call made from a SIP client on the web? |
| 11:11.22 | *** join/#asterisk HeMan (n=jimmy@193.12.106.19) [NETSPLIT VICTIM] |
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| 11:25.10 | joat | freckle, does it pass thru an * box that you manage? |
| 11:25.26 | freckle | joat: yes |
| 11:25.38 | joat | search voip-info for Ices |
| 11:25.47 | freckle | joat: ok thanks |
| 11:25.52 | joat | * allows you to stream calls to Icecast |
| 11:26.23 | joat | give a yell if you have issues |
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| 11:30.34 | *** part/#asterisk Hackwar2 (n=Hackwar2@p54BB5B57.dip.t-dialin.net) |
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| 11:32.53 | *** join/#asterisk enriq (n=evetere@33-138-235-201.fibertel.com.ar) |
| 11:34.01 | enriq | hello |
| 11:34.19 | *** join/#asterisk TheIceMan (n=theicema@86.122.46.21) |
| 11:34.57 | TheIceMan | CLI> dahdi show status |
| 11:34.58 | TheIceMan | No such command 'dahdi show status' (type 'help dahdi show' for other possible commands) |
| 11:35.04 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:35.05 | TheIceMan | i get this error in 1.6 |
| 11:35.17 | TheIceMan | no dadhi ? |
| 11:35.33 | kaldemar | no dahdi. |
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| 12:23.34 | chikkis | hello everyone |
| 12:31.35 | *** join/#asterisk M07w (n=M07w@216.234.116.122) |
| 12:39.45 | *** join/#asterisk sheri_rao (n=sheri_ra@115.186.130.188) |
| 12:40.13 | sheri_rao | can anyone send me test call |
| 12:40.49 | sheri_rao | Dovid, can u help me please |
| 12:45.18 | sheri_rao | can someone help? |
| 12:45.32 | sheri_rao | ./j #trixbox |
| 12:46.35 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 12:46.42 | Zeeek | morning all |
| 12:46.49 | Zeeek | or evening as the case may be |
| 12:55.28 | *** join/#asterisk ta^3 (n=tacvbo@189.146.172.59) |
| 12:57.47 | path_ | I have one FXO port, it is possible to connect a fax and use it as a extension? |
| 12:58.07 | coppice | what if its afternoon (though in practice its evening anywhere that matters)? |
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| 13:16.46 | *** join/#asterisk carranca (n=carranca@pampero.itba.edu.ar) |
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| 13:19.01 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 13:24.17 | carranca | Hi, I have a question about an error a coworker is having with an Asterisk I am administrating. He is getting an error 488 "Not acceptable here". He is sending an invite with an empty media for doing late sdp negotiation. Does someone knows what this could be? |
| 13:24.29 | carranca | Does Asterisk supports late sdp negotiation? |
| 13:25.01 | [TK]D-Fender | carranca: Probably not. |
| 13:25.22 | xacatecas | is there any text-to-speech engines available in asterisk? |
| 13:25.28 | xacatecas | and are they any good? |
| 13:25.32 | [TK]D-Fender | carranca: Because the 488 tells right up it doesn't like the selection. pastebin the * side SIP debug |
| 13:25.34 | [TK]D-Fender | ~pb |
| 13:25.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 13:25.47 | [TK]D-Fender | xacatecas: Cepstral is pretty good. |
| 13:26.08 | xacatecas | [TK]D-Fender, 3rd party tool? |
| 13:26.26 | coppice | some people think he is |
| 13:26.27 | [TK]D-Fender | xacatecas: Always... Digium doesn't make one. |
| 13:26.48 | [TK]D-Fender | coppice: This is a 2 horse race, whuddya talkin' bout?! |
| 13:26.49 | [TK]D-Fender | :) |
| 13:26.50 | xacatecas | cool. |
| 13:26.55 | anonymouz666 | coppice: a local vendor is offering AGC on their E1 channels |
| 13:27.12 | [TK]D-Fender | coppice: Good one by the way ;) |
| 13:27.27 | [TK]D-Fender | anonymouz666: Oh no.. now you're rally gonna get it! |
| 13:27.31 | [TK]D-Fender | really even! |
| 13:27.38 | coppice | anonymouz666: a local vendor is offering crack. you want some? |
| 13:27.45 | anonymouz666 | haha |
| 13:29.40 | *** join/#asterisk SparFux (n=raoul@e182022124.adsl.alicedsl.de) |
| 13:31.03 | SparFux | Hi all! Before buying even more crap, I would like to ask here first. What do you think about this hifi headset to do online telephony.? http://www.hama.de/portal/searchSelectedProduct*NO/articleId*142170/action*2563/searchMode*1/bySearch*bsh-240 |
| 13:32.38 | coppice | SparFux: it seems to lack a mic |
| 13:33.45 | [TK]D-Fender | SparFux: Has nothing to do with anything except bluetooth |
| 13:33.46 | SparFux | coppice: the description says, it has a mic. Besides, isn't a headset something with earphones and mic? |
| 13:34.34 | coppice | SparFux: my daughter's hair band is set on her head. terminology is flexible :-) |
| 13:34.45 | M07w | does asterisk directly run the phones, or control the server that runs the phones? |
| 13:34.48 | SparFux | Fender: so it should work and would be quite cool for using my software telephones. and I can even listen to music with that one. |
| 13:35.13 | SparFux | M07w: it controls the server which the phones are connected to. |
| 13:35.14 | [TK]D-Fender | SparFux: If yoursystem supports BT and softphones suck. |
| 13:35.47 | M07w | do you know of an opensource phone server for hipath 4000? |
| 13:36.06 | SparFux | Fender: I can even use it with my mobile and I can call my asterisk box via mobile for free. |
| 13:36.11 | [TK]D-Fender | coppice: My hair band opens for Spinal Tap in 2 months :p |
| 13:36.46 | coppice | does that hair band go to 11? |
| 13:36.49 | [TK]D-Fender | SparFux: Stop treating it like its something special. ITS A FUCKING BLUETOOTH HEADSET. |
| 13:37.05 | [TK]D-Fender | coppice: Everything louder than everything else - Meat Loaf |
| 13:37.19 | xacatecas | [TK]D-Fender, thanks. that cepstral does indeed sound pretty impressive. |
| 13:37.19 | carranca | [TK]D-Fender, sry for the delay, here is the pastebin http://pastebin.com/m42c02494 |
| 13:37.26 | SparFux | Fender: I have to admit that I had problems finding a headset which would give me a mic and stereo hifi sound. |
| 13:37.31 | xacatecas | i think that is some of the best I've heard in a LONG while. |
| 13:38.13 | carranca | the scenario im using is describe as the flow 4 in the rfc3725 |
| 13:38.17 | [TK]D-Fender | carranca: [Feb 13 11:34:18] WARNING[4251]: chan_sip.c:5108 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 172.16.97.57') <-- no late media neg allowed here |
| 13:38.40 | [TK]D-Fender | SparFux: And are you 100% sure your other devices SUPORT these features? |
| 13:39.05 | SparFux | Fender: Isn't it like hands-free-profile and then it should support it? |
| 13:39.22 | [TK]D-Fender | carranca: Asterisk..... where RFC meets "no fucking comment" :) |
| 13:40.05 | carranca | jajaja |
| 13:40.29 | carranca | is there a page where it says which RFC features/scenarios are supported? |
| 13:41.15 | Zeeek | Has anyone here written an application for Polycom? If not have you tried their samples? TKDfender? |
| 13:42.30 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:42.33 | Zeeek | JaJa |
| 13:42.53 | Zeeek | love Spinal Tap |
| 13:43.05 | Zeeek | almost ready to watch it again for the fifth time |
| 13:43.15 | Zeeek | but we digress.... Polycom apps? |
| 13:44.11 | [TK]D-Fender | carranca: Thats the best part... NO :| |
| 13:44.53 | [TK]D-Fender | Zeeek: Apps? I've don't XHTLM browser stuff, but The new stuff is more than that, isn't it? |
| 13:44.58 | SparFux | What is it with software phones, that sucks so much? |
| 13:45.15 | Zeeek | TKD I can't figure out how they work! |
| 13:45.40 | Zeeek | I can see them on a browser, I can see them on the microbrowser, but they don't actually work once I log in |
| 13:45.52 | Zeeek | However, I have a radical app wirkong on my own server |
| 13:46.07 | Zeeek | It shows the latest asterisk versions :) |
| 13:46.11 | Zeeek | exciting, no? |
| 13:46.13 | coppice | SparFux: developing softphones is like being a politician. the kind that *want* to do it, really shouldn't |
| 13:46.24 | Zeeek | coppice: LOL |
| 13:46.53 | Zeeek | is happy cause I found the PC --> Sony TV cable |
| 13:47.12 | SparFux | coppice: Yes, only people who don't understand how to use the ALSA API seem to actually develop softphones. |
| 13:47.33 | [TK]D-Fender | Zeeek: Yeah, I'm reading the new integration guide... they did add quite a bit including soft-key control |
| 13:47.57 | coppice | I thought it was people who lack a grasp of the flow of time that develop softphones |
| 13:48.10 | *** join/#asterisk gremly (n=gremly@190.144.69.234) |
| 13:48.30 | SparFux | coppice: hm... |
| 13:48.59 | coppice | or maybe its just people who actually stutter in real life |
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| 13:50.14 | SparFux | All I can say is that it is brainwashed to have a fast personal computer up and running all the time and connected to the internet and the pstn and then buy a 100$ hardware phones with additional electrical power consumption and use this device to do the phoning. |
| 13:50.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:51.06 | Zeeek | TK where is this, do you have a lonk? ARe you referring to those Java apps thazt work only on WIndows? |
| 13:51.14 | Zeeek | s/lonk/link/ |
| 13:51.49 | [TK]D-Fender | Zeeek: No, they seem to have beefed up the base XHTML we've used before.... maybe there is other stuff too... but I' haven't looked into more yet |
| 13:52.35 | Zeeek | maybe I don't have the most recent version, either |
| 13:52.38 | [TK]D-Fender | Zeeek: the guide is on their site : Web_Application_Developers_Guide_SIP_3_1.pdf |
| 13:53.36 | Zeeek | I'm looking at SampleAPps 220 somthing |
| 13:54.55 | Zeeek | http://downloads.polycom.com/voice/voip/spip/Sample_LicenseAgreementForDevelopmentPurposes.htm |
| 13:55.41 | Zeeek | this is where you see how useless the search is on those sites |
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| 14:00.02 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:06.07 | Zeeek | ok |
| 14:08.24 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
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| 14:09.46 | Katty | morning (= |
| 14:10.07 | laggo | i've recorded a call with MixMonitor(), but every program i use to try and play back the .wav seems to play it back too fast (im guessing because of the 8000 hz sample rate). is there a special program or conversion i should use? |
| 14:10.25 | Katty | how is everyone? |
| 14:10.40 | Zeeek | {{{{{ Katty }}}}} |
| 14:10.54 | Katty | hugs Zeeek |
| 14:11.07 | _ShrikE | Good morning Katty |
| 14:11.08 | Zeeek | falls down the stairs |
| 14:11.18 | Katty | :< |
| 14:11.25 | Katty | hugs _ShrikE |
| 14:19.08 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:19.38 | *** join/#asterisk medjr (n=root@41.226.178.114) |
| 14:20.32 | medjr | when i type asterisk -r in the shell i cannot connect to asterisk and i get the following error message : "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 14:21.35 | medjr | kifech na3mel rani faddit |
| 14:21.46 | [TK]D-Fender | medjr: Either the user you are running as doesn't ahve the rights to see the .ctl file, your installation is screwed up, or most likely : Asterisk isn't RUNNING |
| 14:22.15 | medjr | asterisk is running |
| 14:22.53 | medjr | i typed /etc/init.d/asterisk restart |
| 14:23.12 | medjr | root@med-desktop:~# /etc/init.d/asterisk restart |
| 14:23.13 | medjr | Stopping Asterisk PBX: asterisk. |
| 14:23.13 | medjr | Starting Asterisk PBX: asterisk. |
| 14:23.44 | medjr | but still the same error message |
| 14:24.07 | medjr | the user is "root" |
| 14:24.26 | medjr | so i guess he does have every possible right, true ? |
| 14:25.30 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:28.02 | [TK]D-Fender | medjr: "ps -A|grep asterisk |
| 14:28.18 | [TK]D-Fender | medjr: a stupid init script doesn't prove much to me. |
| 14:28.33 | medjr | ok |
| 14:28.52 | [TK]D-Fender | medjr: It could be crashing in a loop for all you know |
| 14:29.36 | medjr | ps -A doesnt have any asterisk in its output |
| 14:29.46 | medjr | asterisk not running |
| 14:29.55 | medjr | :( |
| 14:30.36 | [TK]D-Fender | medjr: so run it MANUALLY and look at what happens. asterisk -gvvvvvvvvvvvvvc |
| 14:30.51 | medjr | ok |
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| 14:51.00 | brunner | When I try to play music on hold, I get the following message in my console: -- Music class default requested but no musiconhold loaded. |
| 14:51.28 | brunner | even though the moh module is set to load in modules.conf |
| 14:53.24 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096591554.dsl.bell.ca) |
| 14:57.15 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:57.40 | *** join/#asterisk mphill_ (n=mphill@204.14.193.163) |
| 14:58.10 | Whitor | Hola. Lurk mode enabled. |
| 14:59.30 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:00.18 | flujan | hello guys, it is possible to customize the service unavailable message on asterisk for congestion problems? |
| 15:00.24 | flujan | here is the why http://forums.counterpath.com/viewtopic.php?t=13411&highlight=503. |
| 15:00.29 | flujan | damn softphone do not work... |
| 15:00.53 | flujan | i am looking for a alternative to change asterisk without messing up the chan_sip. |
| 15:06.33 | jer | any ideas how i can go about troubleshooting a bad file descriptor error returned from sip_xmit ? |
| 15:06.59 | jer | (there's no firewall running on ths box) |
| 15:08.30 | *** join/#asterisk jad_jay (n=chatzill@public.axolys.fr) |
| 15:08.32 | jer | (1.4.23.1) |
| 15:08.40 | jad_jay | hi all |
| 15:08.53 | jad_jay | i'm in trouble with asterisk package on lenny |
| 15:08.54 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 15:09.04 | jad_jay | i can't switch the voice to french |
| 15:09.54 | jad_jay | i installed the prompr-fr I change every occurence of language in conf files, i restarted and reload but => english voice ... |
| 15:10.01 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 15:10.25 | jad_jay | thus the prompt-fr have demo voice |
| 15:10.37 | *** join/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 15:10.37 | *** mode/#asterisk [+o Mog] by ChanServ |
| 15:11.00 | jer | oh nm, in my sip.conf i was binding to the old ip |
| 15:11.11 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:11.45 | orn | brunner: Did you install the moh packages? |
| 15:11.47 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 15:11.50 | orn | brunner: Do you have the actual audio files? |
| 15:12.10 | brunner | orn: yes, I have audio files, created using the sox parameters in the book |
| 15:12.11 | Dr-Linux|home | i' using asterisk 1.4.22 and i'm having this problem: http://www.syednetworks.com/asteriskforums?forumaction=showposts&forum=5&thread=303&start=0 |
| 15:12.21 | Dr-Linux|home | is it a bug in new version? |
| 15:13.23 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:13.40 | brunner | orn: I'm not sure what moh packages you're referring to, but the module seems to exist already |
| 15:13.40 | jad_jay | is there anybody who use the debian lenny packages |
| 15:14.22 | orn | brunner: I was talking about the actual songs that come with it, but since you made your own it doesn't apply. |
| 15:14.58 | orn | brunner: When are you playing this? Are you doing it in a meetme conference or just using the application MusicOnHold? |
| 15:15.06 | jad_jay | i could i tell asterisk to use the voice in /usr/share/asterisk/sounds/fr |
| 15:15.27 | brunner | orn: http://pastebin.com/m222e1935 |
| 15:15.51 | brunner | orn: MusicOnHold |
| 15:16.06 | orn | also put up your musiconhold.conf |
| 15:16.14 | orn | oh sorry, you did |
| 15:16.23 | brunner | =] |
| 15:16.52 | orn | ls -l /var/lib/asterisk/moh ? |
| 15:16.53 | orn | :D |
| 15:17.47 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 15:18.08 | brunner | -rw-r--r-- 1 chris chris 7615262 2009-02-12 14:06 moh1.wav |
| 15:19.11 | dalbaech | hey guys... |
| 15:19.15 | dalbaech | anyone using the "new" queues? |
| 15:19.25 | *** join/#asterisk medjr (n=medjr@41.226.178.114) |
| 15:19.41 | dalbaech | 'wrandom' isn't a valid strategy for queue |
| 15:19.45 | dalbaech | any idea? |
| 15:19.46 | orn | does it make difference if you do MusicOnHold(default) ? |
| 15:20.13 | medjr | [TK]D-Fender i found the problem dud |
| 15:20.15 | medjr | [TK]D-Fender i found the problem dude* |
| 15:21.46 | *** join/#asterisk beherit (n=albert@netsys.bts.corp.amdatex.net) |
| 15:22.37 | beherit | I have two * and I want to have a conference room that both user in the two * can meet. what do i need to do? |
| 15:23.14 | orn | set up a conference room on either one of them and create a trunk between the *'s |
| 15:23.42 | Dr-Linux|home | any clue on my question? |
| 15:24.09 | orn | brunner: also, what does "moh show files" show you? |
| 15:24.19 | asteriskmonkey | anyone know what cause a � to be stuffed infront of callerid name? |
| 15:24.28 | brunner | orn: nothing |
| 15:24.29 | beherit | orn, I already done that, users on both * can connect to the conference room but they can't hear each other |
| 15:24.56 | brunner | hmm.... that's an interesting sign |
| 15:25.50 | brunner | "moh show classes" returns nothing, either |
| 15:26.06 | orn | brunner: then run "moh reload" and re-run the previous command |
| 15:26.37 | orn | beherit: is either one of them behind nat? also you might want to try adding "canreinvite = no" to the SIP trunks in sip.conf |
| 15:26.48 | jad_jay | hé i don't know what i did but all my conf files disappeared and ther is only .conf.orig files |
| 15:27.05 | orn | jad_jay: most likely you deleted *.conf? |
| 15:27.14 | jad_jay | never did that |
| 15:27.27 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 15:27.56 | jad_jay | well could you explain me for the language |
| 15:28.48 | orn | jad_jay: They don't just vanish on their own.. |
| 15:28.53 | orn | jad_jay: What language? |
| 15:29.09 | [TK]D-Fender | medjr: Good |
| 15:29.11 | jad_jay | french with prompt-fr of lenny |
| 15:29.27 | brunner | orn: IT WORKS!!!!!!! |
| 15:29.30 | brunner | hurray! thank you! |
| 15:29.35 | orn | no problem :-) |
| 15:30.04 | medjr | [TK]D-Fender : i changed the bindaddress in manager.conf |
| 15:30.37 | brunner | orn: if I use a stream instead of static files, will it create several instances of the stream for each caller, or will it use one stream for everyone? |
| 15:30.41 | jad_jay | orn: well could it be that destar do this trick |
| 15:31.16 | orn | brunner: I'm not sure. I've never done it, but I think it will create one stream per user |
| 15:31.38 | brunner | hmm.. that will eat gobs a bandwidth. is there any way to prevent that? |
| 15:31.39 | orn | brunner: If you find out, please let me know :-) |
| 15:31.44 | brunner | orn: will do |
| 15:32.04 | brunner | bets Corydon76-dig would know |
| 15:32.14 | orn | brunner: Maybe some sort of a wrapper... a process running outside of * that fetches the stream remotely and then allows local connections to itself? |
| 15:32.29 | brunner | yeah, that would work |
| 15:32.51 | brunner | still wasteful of resources, but not as bad as downloading the same stream 30 times |
| 15:33.29 | orn | yeah, i've never really found out whether asterisk can multicast, so to speak, a stream but often wondered |
| 15:33.55 | asteriskmonkey | is 1.6.05 stable? for production |
| 15:34.19 | wonderworld | asteriskmonkey: i am using it and it hasn't crashed in a month |
| 15:34.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:34.42 | wonderworld | dahdi sometimes doesn't bring up the card though |
| 15:34.57 | wonderworld | but we don't reboot often, so it's not really a problem |
| 15:35.03 | asteriskmonkey | wonderful, any issue compiling sangoma drivers with the new dahdi/ast 1.6? |
| 15:35.17 | wonderworld | no idea, have a digium card |
| 15:35.37 | asteriskmonkey | i have loads of digium cards :) just the box im upgrading dosnt boo |
| 15:35.48 | *** join/#asterisk nOgAnOo (n=noganoo@network184-253.wctc.net) |
| 15:36.52 | wonderworld | it *seems* to be very stable for me. but it probably depends on what you want to do with it |
| 15:38.06 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:38.09 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:38.40 | Mog | mmmhm |
| 15:38.59 | Mog | i run 1.4 / 1.6 in my production box |
| 15:39.56 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:40.13 | brunner | orn: have you ever tried streaming music on hold? |
| 15:40.24 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 15:42.15 | brunner | it works!! |
| 15:43.02 | beherit | orn, they are in local network so no nat. |
| 15:43.10 | wonderworld | i'd say most of the stuff in asterisk runs surprisingly well |
| 15:43.58 | guax | wonderworld, i dont think it should be different |
| 15:44.20 | guax | well, it should be better in some cases |
| 15:45.44 | wonderworld | i think in general SIP is a big problem. a pitty that it became the "free industry standard". all that NAT'ing makes things ugly. |
| 15:46.03 | wonderworld | but that's not asterisks fault.... |
| 15:46.08 | beherit | <PROTECTED> |
| 15:46.31 | orn | beherit: Do they hear the music on hold? |
| 15:46.39 | orn | brunner: no |
| 15:46.59 | orn | brunner: what works? the streaming moh? |
| 15:47.38 | wonderworld | beherit: can they hear eachother when they call eachother? |
| 15:50.27 | *** join/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 15:51.19 | *** join/#asterisk coriolanus (n=coriolan@c-76-16-20-61.hsd1.il.comcast.net) |
| 15:51.26 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.9.59) |
| 15:51.34 | tAnkOSX | Anyone suggestions for a wireless SIP phone? I have a Siemens Gigaset S675IP but I do not like the build quality, settings and webinterface. |
| 15:51.50 | tAnkOSX | Bit dissapointed... |
| 15:52.45 | brunner | orn: using a shoutcast stream as moh |
| 15:52.49 | *** join/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 15:52.55 | *** part/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 15:54.34 | brunner | now it's time to get call queuing working =] |
| 15:54.42 | orn | you should make two calls and see whether the shoutcast server registers another connection :) |
| 15:56.23 | *** join/#asterisk seanmh (n=johndoe@abq-216-31-109-157.dsl.zianet.com) |
| 15:56.44 | brunner | yeah, I'm about to borrow a bunch of phones from a friend of mine that owns a taxi company |
| 15:57.07 | brunner | orn: but if you want to call it when I do to test with me, we can |
| 15:57.35 | brunner | s/when/with |
| 15:57.45 | brunner | err, nm |
| 15:58.10 | *** join/#asterisk grabes (n=gaving@209.183.177.102) |
| 15:58.29 | jplank | brunner: that sounds interesting, I'd love to hear how that turns out |
| 15:59.10 | brunner | jplank: I'll let you know what I learn |
| 15:59.15 | brunner | does anyone here have experience with call queuing? |
| 15:59.38 | orn | i'm sure a lot of people do :) |
| 16:00.21 | brunner | yeah, stupid question |
| 16:00.31 | brunner | I wish there was more info about it in the asterisk book |
| 16:01.07 | mort_gib | brunner: Sure what's up?? |
| 16:01.46 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 16:02.06 | grabes | Can someone take a look at this SIP debug for me. I have a meditrix ATA behind NAT that fails whenever an incoming fax comes in, but works fine when a call is made outbound, or if I take NAT out of the equation |
| 16:02.08 | grabes | http://www.pastebin.ca/1336149 |
| 16:02.08 | brunner | mort_gib: do you know of a good resource for learning how to set up call queues? I only have one phone number, and I'd like for agents to be able to login through the same number that everyone else is using |
| 16:02.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:02.30 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:02.40 | mort_gib | brunner: Eh, it wont work like that |
| 16:02.51 | brunner | no? |
| 16:02.53 | mort_gib | You need different numbers |
| 16:03.12 | mort_gib | Like 1234 (Reception girl) |
| 16:03.44 | mort_gib | 56789 queue of Indians supporting hardware |
| 16:03.45 | brunner | aww, really? there's no workaround or function I can redirect someone to so they can log in as an agent? |
| 16:03.48 | jameswf | drmessano: |
| 16:03.56 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-63ad80f12bc333b2) |
| 16:03.56 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:04.19 | mort_gib | brunner: Sure, you can give callers a choice between the nice reception and a queue of hapless.... |
| 16:04.25 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:05.14 | [TK]D-Fender | mort_gib: ping :) |
| 16:05.18 | jameswf | jbot is now randomly chatty? |
| 16:06.15 | Zeeek | ok, so messing with the "new" Polycom 3.1 app document |
| 16:06.19 | mort_gib | [TK]D-Fender: 150 bytes from mort_gib |
| 16:06.24 | mort_gib | :-) |
| 16:06.26 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 16:06.45 | Zeeek | still nothing that works, but more fun |
| 16:06.56 | orn | brunner: What do you mean log in through the same number everyone else is using? Do you mean they call the same phone number the customers do, or do you mean that there is only one number for the PBX? |
| 16:07.10 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 16:07.26 | *** join/#asterisk [gnubie] (n=[gnubie]@119.56.59.7) |
| 16:07.44 | [gnubie] | waves |
| 16:07.47 | Zeeek | in around an hour we'll be partying with VoIP Users on #voip-users-conference and talking via g722 see the IRC channel for how to dial in |
| 16:08.07 | [gnubie] | is there a binary .deb asterisk-1.4.23.1 for debian etch somewhere? |
| 16:08.15 | Zeeek | It's the big Friday the 13th bash with Allison Smith |
| 16:08.42 | jameswf | Zeeek: word of the day Paraskavedekatriaphobic |
| 16:09.00 | brunner | orn, mort_gib: I have two people on the line, and only one instance of the stream app is running |
| 16:09.15 | orn | brunner: Brilliant... thanks :) |
| 16:09.26 | Zeeek | jameswf: how do I react? |
| 16:09.50 | brunner | orn: np |
| 16:09.52 | Zeeek | ah |
| 16:09.59 | Zeeek | fear is not irrational |
| 16:10.15 | brunner | orn: btw, asterisk seems to run the stream 24/7 once it starts |
| 16:10.16 | coppice | only G.722? that's soooo last year. what about some serious wideband? |
| 16:10.17 | Zeeek | fear can be used to create party atmosphere |
| 16:10.36 | Zeeek | coppice: sure, bring it on and the bandwidth to carry it! |
| 16:11.06 | orn | brunner: Good to know... thanks :) |
| 16:11.07 | coppice | ultrawideband works great at bit rates a lot lower than G.722 |
| 16:11.15 | Zeeek | I do not allow any codecs on my system that might interfere with the smooth downloading of multiple pr0n streams |
| 16:11.24 | orn | brunner: but back to the queue question -- can you clarify a bit better what you mean? |
| 16:12.28 | Zeeek | Orange is really pissing me off with their daily service message spam |
| 16:12.34 | brunner | orn: sure. the PBX current has only one phone number. I want call screeners to be able to call in remotely and log in as agents and start taking calls from the queue |
| 16:13.14 | orn | ahh i see... |
| 16:13.30 | brunner | is that possible? |
| 16:13.32 | orn | so they would not be sip devices registered on the asterisk? |
| 16:14.43 | coppice | G.722 is a great illustration of what's screwed up about patents and codecs. nobody would pay the slightest attention to G.722 is its patents hadn't run out |
| 16:15.31 | rene- | when will g729 run out? |
| 16:15.38 | rene- | g729 patents |
| 16:16.10 | coppice | another 10 years or so |
| 16:16.18 | rene- | hmm |
| 16:16.55 | orn | brunner: They would not be SIP devices registered on the Asterisk box? |
| 16:17.00 | coppice | and by then I hope G.729 will have no real significance. what's more important for the future is patents on things like G.729.1, AMR-WB, AAC LD, etc |
| 16:19.29 | jad_jay | orn: what is the file where i tell asterisk to use only files in fr dir for voices? |
| 16:20.39 | *** join/#asterisk jsolis (n=Jimmy@190.41.153.85) |
| 16:22.26 | *** join/#asterisk jpmcallister (n=jpmcalli@kapla.escelsa.com.br) |
| 16:23.47 | grabes | Anyone help me out with a SIP debug? |
| 16:24.19 | orn | jad_jay: try setting language = fr in sip.conf |
| 16:24.33 | jad_jay | i did it |
| 16:24.37 | orn | jad_jay: But I'm not sure... I've never used it. Try using voip-info.org |
| 16:24.42 | jad_jay | then reload then nothing |
| 16:25.03 | jad_jay | i'm on it until this morning |
| 16:25.10 | jsolis | Anyone help me if i can change the varaiable savecallsin in agents.conf per queue |
| 16:25.43 | [TK]D-Fender | jsolis: Clearly not. |
| 16:25.54 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 16:25.58 | orn | grabes: Have you tried to use nat=yes ? |
| 16:26.07 | brunner | orn: there's only phone number, no SIP phones. I can get more phone numbers, but the box will be co-located |
| 16:26.28 | grabes | That is the setting in the sip.conf now, and the device itself is using STUN |
| 16:26.28 | brunner | orn: the one phone number is connected by SIP |
| 16:26.36 | brunner | orn: that is, asterisk is acting as a SIP client |
| 16:26.58 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:27.06 | orn | brunner: There are two things I can think of you could do... 1. You could put up dialplan rules to match the incoming numbers -- if it is the known number of an agent it will put them to the agentlogin script |
| 16:27.27 | *** join/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 16:27.32 | brunner | folks, there is a problem with the shoutcast stream. it seems to work, but it also seems to pause when nobody is listening, so the method I'm using now is no good for playing a live radio broadcast while people are on hold, as it lags |
| 16:27.46 | orn | 2. You could use an un-announced DTMF digit (like # or * or something) to redirect them to the agent login script |
| 16:28.03 | brunner | orn: #2 is what I had planned |
| 16:28.19 | orn | or option 3... use a script on a webpage to make them type in their phone number and click login or something |
| 16:28.48 | brunner | orn: I can't do outbound very easily. ideally, it would all work through the inbound numbers. |
| 16:29.16 | brunner | orn: if I do option 2, what should my agents.conf look like? |
| 16:30.52 | orn | same as regular |
| 16:30.57 | brunner | okay |
| 16:31.15 | orn | the only thing that woudl be different would be your extensions.conf, where you make the IVR redirect them to the agentlogin() |
| 16:32.29 | brunner | okay, thanks |
| 16:33.49 | brunner | orn: once an agent accepts a call from a remote phone, how can he or she transfer the call to a different extension? |
| 16:36.11 | brunner | orn: where should I start setting this up? agents.conf? queues.conf? |
| 16:36.14 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:36.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:36.17 | sheri_rao | can anyone send me test call |
| 16:37.02 | brunner | sheri_rao: sure... what number? |
| 16:37.05 | orn | brunner: You need to setup your functions.conf to allow transfers... then the agent could press a DTMF sequence to transfer... |
| 16:37.14 | orn | brunner: I'd start in queues.conf |
| 16:37.21 | brunner | orn: thanks |
| 16:38.15 | sheri_rao | brunner, wait let me do some settings |
| 16:38.34 | brunner | k |
| 16:39.18 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 16:39.37 | CrashSys | Anyone ever had any luck changing PRI protocol without restarting zap/asterisk? |
| 16:40.14 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:42.10 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:47.37 | sheri_rao | brunner, can u send me call |
| 16:48.36 | kaldemar | sheri_rao: why on earth are you not making the call yourself? a bit easier than asking here every day. |
| 16:49.31 | sheri_rao | kaldemar, i want to do from external .outside my country |
| 16:49.59 | Zeeek | Partying with VoIP Users on #voip-users-conference and talking via g722 see the IRC channel for how to dial in |
| 16:50.11 | sheri_rao | kaldemar, tu chootia hai |
| 16:50.33 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 16:51.19 | kaldemar | sheri_rao: come again? |
| 16:52.53 | orn | sheri_rao: Why does that matter? If you are terminating a SIP call, asterisk doesn't care where it comes from.. |
| 16:53.44 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:53.57 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 16:53.57 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:54.04 | *** join/#asterisk km- (n=pgrace@vsix.me) |
| 16:54.13 | km- | does anyone here have experience with using sipp with pcap audio |
| 16:55.38 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 16:56.04 | *** join/#asterisk jeffp81 (n=jeff@aegis1.lextech.com) |
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| 16:56.33 | agx | hi, any idea why with notifyringing=no in sip.conf the BLFs on grandstream phones still blinks while on Snom they looks like busy ? |
| 16:56.41 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 16:57.37 | jeffp81 | Is there an easy way to pump arbitrary audio data onto one end-point of an Asterisk SIP connection. Or libraries to do so? This is for an situation where the audio device on one end is not SIP enabled and will have to manually have data gathered and sent across the pipe. |
| 16:58.19 | Zeeek | Allison Smith live now if you want to meet her and say hi: http://tr.im/voip to join the call |
| 16:58.27 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:58.55 | carrar | jeffp81, if it's not SIP enabled, how are you gonna do it over SIP? |
| 16:59.09 | *** join/#asterisk [netman] (n=netman@96.Red-83-45-36.dynamicIP.rima-tde.net) |
| 17:00.01 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 17:00.23 | jeffp81 | carrar: I assume I'll have to create the connection in software. I should have mentioned this will interface through a PC |
| 17:00.44 | carrar | using the SIP protocol? |
| 17:01.07 | jeffp81 | If you have other suggestions I am very open to hearing them |
| 17:01.11 | carrar | could just get a softphone to do auto answer |
| 17:01.31 | carrar | whats the goal? |
| 17:01.35 | carrar | paging? |
| 17:01.36 | jeffp81 | This will be a large distributed system with asterisk residing on a central server |
| 17:01.41 | jeffp81 | 2-way audio |
| 17:01.44 | jeffp81 | is the goal |
| 17:01.51 | carrar | how about a phone? |
| 17:02.00 | carrar | auto answer to speakerphone |
| 17:02.03 | jeffp81 | One endpoint will be a phone |
| 17:02.04 | jpcansa | whats the best way to cancel vm feature for everyone in my * ?? |
| 17:02.25 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:02.37 | carrar | remove the config jpcansa |
| 17:02.46 | jeffp81 | The other endpoint will be a device that can produce digital audio, but not designed for SIP specifically |
| 17:02.56 | carrar | or remove the voicemail command from the dialplan |
| 17:03.06 | jpcansa | carrar: the config in every ext? |
| 17:03.32 | carrar | You did say cancel |
| 17:03.52 | carrar | I assume cancel == remove |
| 17:04.00 | jpcansa | yes |
| 17:04.22 | carrar | use sed to remove it all with 1 line |
| 17:04.34 | *** join/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 17:04.59 | jeffp81 | carrar: I'm assuming that this is a non-standard requirement and there is not a lot of available resources to assist me? |
| 17:05.07 | Dr-Linux|home | How long i can set CallerID? |
| 17:05.25 | Dr-Linux|home | 15 digits according to E.164? |
| 17:05.34 | Dr-Linux|home | please suggest |
| 17:05.50 | carrar | jeffp81, "non-standard" is relative :) |
| 17:06.17 | jeffp81 | carrar: But sounds like no one here has done it before :( |
| 17:06.41 | carrar | Dr, Set(CALLERID(number)=8675309) |
| 17:06.57 | carrar | jeffp81, you said remove |
| 17:07.01 | carrar | so remove it from your config |
| 17:07.07 | Dr-Linux|home | carrar: you didn't understand my question i guess |
| 17:07.14 | carrar | not seeing why that doesn't make sense |
| 17:07.25 | jpcansa | carrar: can i remove deactivate it from voicemail.conf? |
| 17:07.48 | carrar | You can, but then you have all those commands that will errror |
| 17:07.57 | Dr-Linux|home | carranca: will it work? Set(CALLERID(number)=30 digit here) |
| 17:08.04 | carrar | and your dial by directory won't work if used |
| 17:08.25 | carrar | Dr-Linux|home, yes |
| 17:08.27 | carrar | try it |
| 17:08.54 | Dr-Linux|home | carrar: is it not E.164 format? |
| 17:09.03 | Dr-Linux|home | carrar: what's the limit? |
| 17:14.33 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:16.59 | *** part/#asterisk km- (n=pgrace@vsix.me) |
| 17:20.23 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 17:20.32 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 17:20.43 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 17:23.55 | brunner | is there any way to get asterisk to kill the custom moh process when nobody is listening? |
| 17:24.09 | brunner | when nobody is on hold, I mean |
| 17:24.14 | carrar | you install the dummy dadhi driver brunner? |
| 17:24.21 | carrar | pretty sure that was your issue |
| 17:25.06 | brunner | carrar: after sleeping, I realized musiconhold.conf wasn't readable to asterisk. doh. sorry about the trouble. |
| 17:25.19 | *** join/#asterisk jeffp81 (n=jeff@aegis1.lextech.com) |
| 17:25.20 | carrar | worked after that? |
| 17:25.27 | brunner | yep, works great now ==] |
| 17:25.29 | carrar | k |
| 17:25.49 | brunner | except that it builds up this giant buffer when nobody is listening |
| 17:25.53 | brunner | I'm using a shoutcast stream |
| 17:25.59 | brunner | a custom process |
| 17:26.21 | brunner | and when I call, listen, hang up, wait 30 minutes, call back, it picks up right where it left off |
| 17:26.26 | brunner | I need it to be live |
| 17:26.40 | carrar | could pipe it to the sound card |
| 17:26.48 | carrar | so something is reading from it |
| 17:26.57 | brunner | I mean, I just need it to cut off the process when nobody is listening |
| 17:27.03 | carrar | I think there are some examples of using streams |
| 17:27.08 | carrar | if you look around |
| 17:27.35 | brunner | there are on voip-info.org, but they don't talk about how to get asterisk to stop using the custom moh process when somebody hangs up |
| 17:28.42 | Corydon76-dig | brunner: it does not work that way |
| 17:28.50 | Corydon76-dig | brunner: but we accept patches |
| 17:28.51 | carrar | use a outside process to stream it to your audio card |
| 17:29.06 | carrar | and then just tape into that channel when you need it |
| 17:30.08 | carrar | I have a box that is slightly different then that, they have a remote audio streaming device that i put in the linein jack and it plays that whenever someone is on hold |
| 17:30.11 | sheri_rao | one end having CISCO gateway 3745 & other is Asterisk , Protocol SIP, COdec g729. no reason , it should not work |
| 17:30.13 | brunner | carrar: that's a good idea, but I'm not sure that the server this will be running on will have a sound card |
| 17:30.19 | brunner | but I guess they all do these days |
| 17:30.29 | carrar | put one it |
| 17:30.32 | *** part/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 17:30.35 | carrar | they are cheap |
| 17:30.49 | carrar | or there might be some app out there to do something like that |
| 17:30.50 | brunner | carrar: I don't have physical access to the box |
| 17:31.24 | brunner | Corydon76-dig: what would it cost to hire someone to write such a patch? |
| 17:32.09 | carrar | bruner, http://www.ctunion.com/node/228 |
| 17:32.10 | sheri_rao | anybody has has used CISCO 3745 gateway with asterisk? |
| 17:33.21 | brunner | carrar: that's what I'm already doing |
| 17:33.24 | carrar | ah |
| 17:34.43 | carrar | keep searching, you'll figure it out :) |
| 17:35.08 | Corydon76-dig | brunner: if you're willing to test it, won't cost anything. |
| 17:35.14 | brunner | I guess I could use an external program to monitor the number of people on hold and kill the process when it reaches zero |
| 17:35.35 | brunner | Corydon76-dig: I'll test it |
| 17:36.25 | Corydon76-dig | brunner: give me some time to write it |
| 17:36.44 | brunner | Corydon76-dig: you are my hero. |
| 17:44.28 | *** join/#asterisk jeffp81 (n=jeff@aegis1.lextech.com) |
| 17:48.51 | ruben23 | does this cahnnel support asterisk:AGI...scripting..? |
| 17:49.55 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) |
| 17:50.03 | SparFux | ruben: I bet yes. |
| 17:51.40 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.112.83) |
| 17:51.53 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 17:57.50 | blackest_mamba | I'm just getting started. I'm pretty proficient with FreeBSD, but haven't touched Linux in a long time - not sure I care to. Is there a particular OS that is more friendly to Asterisk over another? |
| 17:58.35 | *** join/#asterisk esperegu (n=esperegu@145.116.15.244) |
| 17:58.54 | esperegu | how to see which codecs are available in asterisk? |
| 18:02.02 | kaldemar | esperegu: core show codecs in asterisk's cli |
| 18:03.00 | esperegu | kaldemar: but it saids: It does not indicate anything about your configuration. |
| 18:03.27 | esperegu | kaldemar: I thought that that meant that asterisk could use them. not that they are available? |
| 18:03.32 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 18:03.49 | esperegu | I want to know which ones I can enable. |
| 18:04.05 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-2c99d10d39851bad) |
| 18:04.16 | esperegu | kaldemar: or does it print only the ones that are on the system? |
| 18:05.17 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-fdbb1c822c8295d0) |
| 18:05.57 | kaldemar | esperegu: ah, core show translation will show translation times for the codecs that you have in use. |
| 18:06.04 | *** join/#asterisk sack (n=sack@50.Red-88-24-156.staticIP.rima-tde.net) |
| 18:06.24 | kaldemar | esperegu: show codecs also prints such codecs that are not in use. |
| 18:06.39 | esperegu | kaldemar: but they are all on the system? |
| 18:06.43 | *** join/#asterisk jeffp81 (n=jeff@aegis1.lextech.com) |
| 18:07.04 | jeffp81 | Does anyone here have experience with AGI? |
| 18:07.22 | kaldemar | esperegu: no, not necessarily. |
| 18:07.28 | kim0 | Hi any idea why 'ztcfg -vv' results in ==> "1 channels to configure" |
| 18:07.32 | esperegu | kaldemar: how can I check that? |
| 18:07.38 | esperegu | kaldemar: which are available? |
| 18:07.53 | Corydon76-dig | brunner: what version are you on? |
| 18:08.19 | kaldemar | esperegu: another way would be to check which codec modules you have in the modules directory, /usr/lib/asterisk/modules by default. they're named codec_xxx.so. |
| 18:08.55 | kaldemar | or module show like codec in cli. |
| 18:09.06 | kaldemar | timtowtdi |
| 18:13.05 | esperegu | grrr. |
| 18:13.09 | esperegu | I keep getting SIP/2.0 488 Not acceptable here |
| 18:13.25 | esperegu | that is codec issue if I'm not mistaken? |
| 18:15.32 | kaldemar | likely so. have you configured codecs with allow and disallow parameters in sip.conf? |
| 18:15.52 | *** join/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 18:15.52 | Corydon76-dig | brunner: it may actually work with the latest version |
| 18:16.24 | Corydon76-dig | brunner: (1.4 SVN) |
| 18:16.24 | esperegu | kaldemar: in freepbx |
| 18:16.42 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 18:17.07 | kaldemar | esperegu: well that's a whole another story. have you asked in #freepbx? |
| 18:17.10 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 18:17.47 | esperegu | It now answers |
| 18:17.54 | esperegu | but I don't hear anything. |
| 18:18.22 | esperegu | kaldemar: it's even worse.... it is http://linuxmce.org ;-) |
| 18:18.37 | kaldemar | most likely a nat issue or still something with codecs. |
| 18:19.20 | kaldemar | i doubt it could be any worse. :) |
| 18:20.17 | kaldemar | i suggest you ask in #freepbx, people here don't generally use it nor like to debug it. |
| 18:23.53 | *** join/#asterisk SkramX (i=mark@phalse.2600.COM) |
| 18:24.53 | *** join/#asterisk _darkKnight_ (n=kvirc@189.59.228.170) |
| 18:26.37 | *** join/#asterisk ingenius (n=alektro@host169.190-30-123.telecom.net.ar) |
| 18:28.23 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 18:28.49 | brunner | Corydon76-dig: 1.4.21.2 |
| 18:29.08 | `paul | if i want to allow/receive calls from a certain ip ill do type=friend on sip.conf right? |
| 18:29.22 | *** join/#asterisk jupeterson (n=John@c-24-126-160-141.hsd1.ga.comcast.net) |
| 18:29.46 | jupeterson | does anyone know why asterisk leaves files open? |
| 18:30.08 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 18:30.09 | jupeterson | I have a large number of recorded IVR files they seem to stay open forever |
| 18:35.14 | [TK]D-Fender | `paul: "type=peer" for almost all entries |
| 18:35.32 | [TK]D-Fender | `paul: "type=friend" = hardly used since 1.4 |
| 18:36.13 | jupeterson | does anyone know why asterisk leaves files open? |
| 18:36.27 | jupeterson | <PROTECTED> |
| 18:36.29 | Corydon76-dig | brunner: revision 166262 should fix it for you |
| 18:40.28 | Katty | humm |
| 18:40.46 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 18:41.56 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 18:41.56 | [TK]D-Fender | *ouch* |
| 18:41.56 | *** join/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 18:41.56 | [TK]D-Fender | kick WinXP in the nads |
| 18:42.34 | CrashSys | Careful, windows XP might go Dr. Watson on you |
| 18:45.24 | bmoraca | freakin idiots... |
| 18:46.00 | bmoraca | somebody designed a network of web servers running all off of a 10mbit port on a router with all the servers in a different subnet, thus having to constantly pass through the router even though they're on the same switch |
| 18:46.01 | bmoraca | arg |
| 18:46.06 | bmoraca | and they didn't even do THAT right |
| 18:46.13 | bmoraca | didn't even fuck up correctly |
| 18:48.33 | Katty | i have this weird dread feeling |
| 18:49.52 | Nugget | huggles Katty |
| 18:50.06 | Katty | hugs Nugget |
| 18:50.26 | Katty | why don't girls come with readme-emotions.pdf |
| 18:50.32 | *** join/#asterisk yoanis (n=fred@200.55.139.218) |
| 18:51.11 | Corydon76-dig | Katty: the manual is out of date as soon as it's published, of course |
| 18:51.17 | Katty | sighs |
| 18:51.22 | Qwell | published? it's out of date before it's written |
| 18:51.27 | [TK]D-Fender | Katty: And then it gets read out of context. |
| 18:51.57 | CrashSys | vi readme-girls-emotions.txt |
| 18:51.59 | Katty | i'm just tired of having these weird emotions and moods and not knowing where it's coming from! |
| 18:51.59 | CrashSys | segfault |
| 18:52.03 | [TK]D-Fender | Katty: And then you get the Uncut, Director's Cut, Unrated, and Indecipherable releases out out on simulcast a week after |
| 18:52.46 | Corydon76-dig | I think women should date other women for a time... and let them figure out how men ever put up with it |
| 18:53.13 | Corydon76-dig | Men are FAR easier to date |
| 18:53.25 | Katty | i don't know that i could date myself. |
| 18:53.50 | CrashSys | My father is a landlord and he's gotten to where he wont rent to lesbians if he can help it... they always fight and destroy the place in the process... |
| 18:54.46 | Corydon76-dig | CrashSys: clearly, your father should rent only to gay men |
| 18:55.01 | CrashSys | Gay men, on the other hand, usually end up fixing the place up, pay their rent on time, and are happy to not get hassled. |
| 18:55.01 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 18:55.48 | CrashSys | He also hates renting to "mechanics" too... they usually leave 2-3 cars in the yard and 1 or 2 cars worth of parts in the house |
| 18:55.56 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/Staff/Sargun) |
| 18:55.56 | CrashSys | not to mention the oil stains |
| 18:55.57 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 19:10.11 | Micc | How can I prevent someone from accidently transfering someone to their own voicemail? |
| 19:10.40 | Micc | I want it in the context so they can get to it with a button on the phone. |
| 19:11.06 | Micc | But if they hit transfer then accidently hit the voicemail button, the person is traferred into their voicemail without needing a password. |
| 19:11.19 | Micc | I suppose I could always require the password. |
| 19:11.37 | manxpower | that happens when you don't use passwords |
| 19:11.53 | [TK]D-Fender | Micc: only 1 kind of transfer is detectable. Go read channelvariables.txt to find out which and select based on that. |
| 19:12.10 | [TK]D-Fender | Micc: and that IS a bad idea. |
| 19:12.25 | [TK]D-Fender | Micc: Coworkers listening in on their VM's |
| 19:12.30 | manxpower | cd /tftpboot |
| 19:16.20 | Katty | puts imperial march on new phone |
| 19:16.34 | [TK]D-Fender | DarthCheney.mp3! |
| 19:17.28 | CrashSys | DarthBobo! |
| 19:18.02 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 19:19.30 | rue_mohr | ok, I have an interetsing problem, the test line keeps ringing (dahdi origin) and when they pick up nobody is there, a test calling in with a phone worked fine, in the logs I see the test channel (on the dahdi card) keeps going in and out of red alarm between ring detects. any idea whats going on? |
| 19:19.43 | rue_mohr | I dont know where to find out what sends a dahdi channel into red alarm |
| 19:20.38 | *** join/#asterisk Khratos (n=khratos@190.166.103.112) |
| 19:20.42 | rue_mohr | to me it seems that red alarm shoud be when it dosn't see battery on the line |
| 19:20.58 | CrashSys | mahmi sent dahdi into an alarm cause she caught him messing around with a PRI! |
| 19:21.18 | rue_mohr | ok, I'm going to assume a) nobody knows anything about this |
| 19:21.49 | rue_mohr | b) that the dahdi drivers are junk, totally non-production ready, and I need to switch over to zaptel drivers if I can find them |
| 19:22.04 | rue_mohr | can anyone argue with that using facts? |
| 19:22.14 | *** join/#asterisk FarrisG (n=FarrisG@h-69-3-161-203.dllatx37.covad.net) |
| 19:22.17 | CrashSys | Pfft, facts... |
| 19:22.55 | CrashSys | a red alarm is usually loss of connectivity (layer-1 or layer-2) |
| 19:23.02 | CrashSys | check yer cabling, provider, smartjack, etc etc... |
| 19:23.11 | rue_mohr | its a pci card, and its a pots line |
| 19:23.15 | CrashSys | T1/POTS configuration |
| 19:23.21 | rue_mohr | the pci card is a tdm800P |
| 19:23.39 | rue_mohr | to me, on a pots line, red alarm means no battery |
| 19:23.49 | FarrisG | I've got a hosted asterisk environment, and several sites with multiple phones. Ever since we switched some of our sites to a new ISP, ALL of our Cisco phones are having intermittent issues with not receiving calls. All other SIP phones (Polycom, Grandstream, X-Lite) are working fine. Any idea where to start investigating? |
| 19:24.11 | FarrisG | My first instinct is some kind of keepalive/timeout issue |
| 19:24.19 | rue_mohr | does your new isp be port filtering? |
| 19:24.22 | CrashSys | did you plug in the power plug on the card? did you set fxo/fxs correctly in zapata.conf/zaptel.conf? |
| 19:24.24 | rue_mohr | you couldbe right |
| 19:24.37 | rue_mohr | CrashSys, yes, its all configured right |
| 19:24.50 | rue_mohr | and the aux power is in |
| 19:25.17 | CrashSys | Just for fun, have you tried reversing the FXO/FXS setting? |
| 19:25.19 | rue_mohr | the card works, the two analog sets and the pots lines work |
| 19:25.29 | rue_mohr | we can make calls out on it and get them in |
| 19:25.34 | CrashSys | How can it work if it has a red alarm? |
| 19:25.49 | bmoraca | FarrisG: the only issues I've had with Cisco phones have been related to NAT settings...Cisco phones are much more picky about them than Polycoms i've noticed |
| 19:25.50 | FarrisG | rue_mohr: No, we handle all the port filtering ourselves, and I've verified all the proper ports are open on both ends |
| 19:25.56 | rue_mohr | IT SEEEMS to go out of red alarm when there isa call, then go back into it after |
| 19:26.33 | rue_mohr | it must be the dahdi drivers |
| 19:26.36 | FarrisG | bmoraca: Makes sense. Do you have any details or remember any of the parameters on the phones that I should check out? |
| 19:26.40 | rue_mohr | I shoudl switch to zaptel |
| 19:26.55 | bmoraca | FarrisG: are you configuring them via TFTP? |
| 19:27.00 | FarrisG | bmoraca: Yes |
| 19:27.45 | bmoraca | FarrisG: and you said that they work intermittantly? |
| 19:28.58 | rue_mohr | the other thing I dont understand is that there is a few rings of delay between the analog set on the pots line and when asterisk rings the digital sets |
| 19:29.03 | FarrisG | bmoraca: Correct. They can ALWAYS dial out, but will only ring about 50% of the time |
| 19:29.27 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 19:30.32 | ruben23 | hi |
| 19:30.36 | bmoraca | FarrisG: that's not a NAT issue then. a NAT issue would keep them from registering correctly. check your TFTP config file to make sure nat_enable is set to 1 and in your sip.conf that NAT=yes, but this likely isn't the issue |
| 19:30.50 | bmoraca | FarrisG: what kind of router are they using? |
| 19:35.12 | Khratos | [TK]D-Fender, may I ask you something? |
| 19:35.28 | rue_mohr | does anyone know what the ring delay from a tdm800P pots line to the sip phone should be ? |
| 19:35.50 | bmoraca | rue_mohr: i've seen it anywhere from 2 to 5 seconds |
| 19:36.25 | Signius | Is dahdi-linux a directly replacement for zaptel ? |
| 19:36.44 | StanManCan | What are some cool things to setup in asterisk ? |
| 19:37.31 | StanManCan | Signius: backup your configs before installing dahdi, it's personally given me alot of grief.... like 3 formats before giving up grief |
| 19:37.33 | StanManCan | ! |
| 19:37.51 | bmoraca | rue_mohr: one of the detriments to using analog...which sucks |
| 19:37.58 | StanManCan | Signius: but also keep in mind it's likely that could of been a PEBKAC |
| 19:38.04 | Signius | Its a brand new test install i am only just learning so the test machine can be wiped as many times as needed |
| 19:38.06 | FarrisG | bmoraca: on the host side it's all cisco. on the user agent sites we have various routers. Edgemarc at two, cisco at another, and just a linksys at one small one |
| 19:38.45 | bmoraca | FarrisG: and which ones are giving issues with Cisco phones? all of them? |
| 19:39.22 | Signius | StanManCan: I had an issue with my first attempt with using Asterisk 1.6.2-current with Zaptel-1.4 bacuase i knew nothing about the dahdi stuff |
| 19:39.56 | [TK]D-Fender | Signius: * 1.6.x knows nothing of Zaptel |
| 19:40.06 | Signius | StanManCan: So for my second attempt i am going to try and do it with all the latest versions ? but i didnt know if still needed zaptel or just use dahdi instead |
| 19:40.29 | FarrisG | bmoraca: issues at all sites |
| 19:40.32 | Signius | [TK]D-Fender: Thats exactly problem i didnt know that |
| 19:41.28 | bmoraca | FarrisG: what does sip show peers and call logs say when you attempt a call to an affected phone? |
| 19:43.14 | Khratos | people, If exten [A] calls exten [B] , on the same Asterisk box, AMI ExtensionState should return the status number corresponiding to 'InUse' for both of them, right ? |
| 19:43.22 | ruben23 | i installed asterisk-perl...and test if its installed...i make a test script then run it..got this error:http://pastebin.com/m2480274 and this is my script: http://pastebin.com/m15775bb4 any ideas. |
| 19:44.25 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-9deed7a600296ef5) |
| 19:44.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:45.40 | FarrisG | bmoraca: show peers usually shows them registered, sometimes doesn't. Call log shows unanswered or unreachable |
| 19:46.59 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 19:47.19 | zeeesh | getting error at console: Remote host can't match request CANCEL to call? |
| 19:48.00 | bmoraca | FarrisG: I'd wager it's a problem with the customer premises equipment. is SPI turned off on them? is fixup turned on for the Cisco? |
| 19:49.11 | [TK]D-Fender | ruben23: Can't locate object method "new" via package "Asterik::AGI" (perhaps you forgot to load "Asterik::AGI"?) at ./test.pl line 4. <--- learn how to spell |
| 19:49.37 | Kobaz | haha |
| 19:50.39 | [TK]D-Fender | Kobaz: You know what the best part about must of the advice I hand out here is? |
| 19:51.07 | [TK]D-Fender | s/must/most/ |
| 19:52.45 | Khratos | people, If exten [A] calls exten [B] , on the same Asterisk box, AMI ExtensionState should return the status number corresponiding to 'InUse' for both of them, right ? |
| 19:52.49 | ruben23 | [TK]D-Fender::-D thanks |
| 19:53.17 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:53.47 | [TK]D-Fender | Khratos: that is a dangerously worded question. |
| 19:54.08 | *** join/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 19:54.19 | *** part/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 19:55.28 | Khratos | mmm, ok. I will try to make it simpler |
| 19:57.04 | Beave | anyone have any idea what's up with nufone? |
| 19:57.23 | Khratos | Exten A calls Exten B on the same Astersisk box, B answers. ExtensionState to A, and ExtensionState to B, should return the same state number, right? |
| 19:57.53 | [TK]D-Fender | Khratos: STILL dangerously worded. SHOW US |
| 19:58.06 | Khratos | Ok. |
| 19:58.52 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 20:00.28 | Khratos | A bridge call beetwen 123 and 221 : http://khratos.pastebin.com/m2ecfb79b |
| 20:00.29 | zeeesh | error, Remote host can't match request CANCEL to call? |
| 20:00.45 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 20:01.12 | Khratos | It should indicate the same status on the AMI response, is that correct? |
| 20:01.34 | Khratos | Or only for the extension that received the call |
| 20:02.22 | Khratos | I as because the 'ExtensionState' command shows the 'InUse' status number for the extension that received the call |
| 20:02.41 | Khratos | The extension that initiated the call still appears as 'iddle' |
| 20:03.25 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:03.53 | SuPrSluG | what would cause one way audio when paging. regular calls work. any ideas/ |
| 20:03.56 | SuPrSluG | ? |
| 20:04.30 | *** join/#asterisk mvanbaak (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:04.45 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:05.33 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:07.25 | [TK]D-Fender | Khratos: that is anything but complete. Try again and include EVERYTHING that is going on and being checked |
| 20:09.04 | Khratos | Look at a complete session through Telent to AMI interface: http://pastebin.com/m7ac2b89d |
| 20:09.24 | Khratos | In that moment, there was a briged call from 140 to 122 |
| 20:10.28 | path_ | DigitTimeout was replaced on newer versions ? |
| 20:10.35 | path_ | can't find it on core show applications |
| 20:10.58 | [TK]D-Fender | path_: ANCIENT. Was deprecated in 1.2 |
| 20:11.10 | [TK]D-Fender | path_: read your upgrade docs. |
| 20:11.25 | path_ | I only need to new for what was replaced |
| 20:11.29 | path_ | s/new/know |
| 20:11.54 | [TK]D-Fender | path_: "core show cuntion TIMEOUT" |
| 20:12.03 | [TK]D-Fender | path_: "core show funtion TIMEOUT" |
| 20:12.11 | CrashSys | cuntion...... |
| 20:12.15 | CrashSys | trademarks |
| 20:12.24 | path_ | haha |
| 20:12.27 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 20:12.29 | path_ | thanks [TK]D-Fender |
| 20:12.35 | [TK]D-Fender | Khratos: Try again and include a complete channel dump from CLI and "core show hints, dialplan, SIP configs, etc |
| 20:16.41 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 20:17.09 | SQLDarkly | >AGI Tx >> 510 Invalid or unknown command |
| 20:17.18 | SQLDarkly | oops sorry for double |
| 20:17.38 | SQLDarkly | The console is spitting this out when I call my AGI script |
| 20:17.39 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 20:17.47 | SQLDarkly | what the heck does that mean |
| 20:17.53 | *** join/#asterisk mighty-d (i=500@190.29.5.207) |
| 20:18.01 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:18.08 | mighty-d | how many octastic softecho license would i need for 4 FXO? |
| 20:18.15 | mighty-d | is it a license for each line? |
| 20:18.20 | [TK]D-Fender | SQLDarkly: Means you sent something that was not a recognized AGI comamnd |
| 20:18.22 | bkw_ | mighty-d: well Hi to you too |
| 20:18.25 | bkw_ | :P |
| 20:18.31 | bkw_ | mighty-d: I suspect its one per line |
| 20:18.47 | SQLDarkly | D-Fender. Is there any way I can get a more verbose error? |
| 20:19.03 | [TK]D-Fender | SQLDarkly: Do you get other AGI debugin that call? |
| 20:19.25 | SQLDarkly | I do |
| 20:19.55 | SQLDarkly | brb going to run the AGI on teh *nix shell. Maybe that will provide an error I can troubleshoot |
| 20:20.27 | `paul | if a number is dialed and it enters the queue how come it does not appear on CDR? waht appears is the callerid of the person and the agent extension... but not the number dialed :( |
| 20:21.36 | `paul | oh nevermind its in the database :D |
| 20:21.47 | [TK]D-Fender | `paul: Queue's generate EXTRA CDR's fromt he call prior to hitting "queue" |
| 20:22.16 | mighty-d | bkw_ sorry for beeing rude... :( and thanks a lot btw |
| 20:22.18 | mighty-d | !! |
| 20:32.54 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 20:35.31 | bkw_ | mighty-d: Its ok |
| 20:40.22 | *** part/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
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| 20:40.57 | *** join/#asterisk cp5 (n=samy@72.37.252.50) |
| 20:40.59 | cp5 | hola |
| 20:41.28 | *** part/#asterisk LapTop006 (n=laptop00@gemini.chriskaine.com.au) |
| 20:41.32 | cp5 | how can i identify which thread crashed in gdb from a core? i've enabled DEBUG_THREADS and DONT_OPTIMIZE |
| 20:43.43 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 20:44.27 | *** join/#asterisk punter (n=punter@athedsl-136469.home.otenet.gr) |
| 20:44.32 | punter | Hi all |
| 20:45.35 | carrar | hi!! |
| 20:45.55 | carrar | Hows the land of gods? |
| 20:47.28 | punter | :-) |
| 20:48.12 | *** part/#asterisk punter (n=punter@athedsl-136469.home.otenet.gr) |
| 20:49.51 | *** part/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
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| 20:59.52 | SQLDarkly | solved it.... sweet ;) the linux CLI revealed more info so I was able to diagnose and correct the problem in my AGI script |
| 21:03.27 | *** part/#asterisk [psy] (n=psy0rz@lounge.datux.nl) |
| 21:04.09 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 21:06.59 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 21:08.27 | *** join/#asterisk sacitec (n=tobi@189.129.83.110) |
| 21:08.41 | *** join/#asterisk Deeewayne1 (n=dwayne@nat/digium/x-322345ce1c9c37fe) |
| 21:08.41 | *** mode/#asterisk [+o Deeewayne1] by ChanServ |
| 21:08.49 | sacitec | hello everyone, does asterisk 1.4.22 comes with polarity reversal for zap ? |
| 21:09.49 | jameswf | sacitec: rhino cards do reverse polarity in all versions... |
| 21:09.53 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-544e5e482f41ace9) |
| 21:09.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:10.24 | sacitec | i'm working with sangoma (A200) any clue about them ? |
| 21:11.05 | [TK]D-Fender | sacitec: Same |
| 21:11.45 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-af8b72bb3ca4b2ab) |
| 21:11.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:11.58 | sacitec | thanks :) |
| 21:12.04 | jameswf | sacitec: revp is usualy done at driver leve afaik |
| 21:12.56 | sacitec | but, i i still have the issue, i'm able to work with 'hanguponpolarityswitch' parameter in zapata.conf ? |
| 21:21.50 | *** join/#asterisk martyn-dev (n=admin@190.24.134.154) |
| 21:22.12 | martyn-dev | Hi, somebody here used JAGIServer ? |
| 21:22.17 | martyn-dev | I'm trying to use it and i get a message ("510 Invalid or unknown command") from res_agi.c, but i dont know why JAGIserver send or detect this message . Do you know some about it ? |
| 21:22.24 | martyn-dev | I'm seeng that in JAGIClient.java the the function readLine() try detect some from asterisk-agi but res_agi.c send to JAGIClient this message. Do you know why ? |
| 21:27.28 | [TK]D-Fender | checkout time, BBL |
| 21:31.11 | pfn | [FAX ERROR] code: 13 Unexpected message received |
| 21:31.15 | pfn | wtf does that mean... |
| 21:31.32 | pfn | 13 Far end cannot receive at the size of image |
| 21:31.34 | pfn | that doesn't sound right |
| 21:32.23 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
| 21:32.46 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 21:34.49 | pfn | kicks rxfax |
| 21:35.40 | jameswf | ~pickles |
| 21:35.41 | jbot | pickles are very hard to digest |
| 21:50.21 | cp5 | i'm able to crash asterisk...when i load the core into gdb, is the current thread always the thread that crashed? |
| 21:52.50 | edoceo | Is there a command in the Asterisk CLI to batch delete everything in a VM box? |
| 21:54.27 | *** join/#asterisk lilkid (n=chatzill@87-194-38-230.bethere.co.uk) |
| 21:55.17 | lilkid | May I ask for help installing a2billing (on *) in this channel? |
| 21:55.56 | bombaclat667 | If I want to change the port asterisk listens to for incoming iax/iax2 connections, do I need to edit the aix.conf, aix2.h in the sources or both? |
| 21:57.28 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 21:59.15 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:00.59 | pfn | hmm, how do I check return code in apps? |
| 22:01.25 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 22:01.50 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:01.50 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:01.58 | lmadsen | FYI: Asterisk 1.6.0.6-rc1 has been released! |
| 22:02.40 | rob0 | Just in time for 1234567890! |
| 22:03.57 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 22:09.31 | path_ | indeed! |
| 22:09.33 | path_ | :D |
| 22:09.59 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:10.50 | icebrew54 | jbot: in Capitalist America, we socialize our Banks....and BOTS! |
| 22:11.52 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 22:13.12 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 22:14.11 | stabler | lol |
| 22:14.35 | *** join/#asterisk `g0rt (n=jrandom@or.vr.lt) |
| 22:14.37 | `g0rt | hai |
| 22:15.32 | `g0rt | anyone could explain or point me to a url which explain the concept of "sip trunk" ? |
| 22:15.37 | `g0rt | ( if it even exists ) |
| 22:16.08 | SkramX | google? |
| 22:16.24 | rob0 | ~siptrunk |
| 22:16.24 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 22:16.37 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 22:16.40 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 22:16.41 | SkramX | http://www.siptrunk.org/whatissiptrunking.php perhaps |
| 22:19.32 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 22:19.58 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:19.59 | path_ | `g0rt, you mean voip/sip trunks maybe |
| 22:20.52 | `g0rt | path_: maybe, i do not know so much, it started with sending faxes over voip infrastructure |
| 22:20.59 | `g0rt | several docs point me to "sip trunk" |
| 22:21.02 | `g0rt | i was "lol wat" |
| 22:21.42 | path_ | it meant normally your voip provider |
| 22:22.27 | `g0rt | ok |
| 22:22.45 | *** join/#asterisk ingenius (n=alektro@host169.190-30-123.telecom.net.ar) |
| 22:22.49 | *** join/#asterisk boghog (n=boghog@infinidim.aphax.nl) |
| 22:22.49 | boghog | hi |
| 22:30.00 | *** join/#asterisk Sparks001 (n=postmast@87-194-163-16.bethere.co.uk) |
| 22:31.38 | *** part/#asterisk Sparks001 (n=postmast@87-194-163-16.bethere.co.uk) |
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| 22:49.39 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 22:49.53 | freezey | can you downgrade from SIP8.2 to SIP6.3 on asterisk 7940G phones |
| 22:49.55 | freezey | ? |
| 22:51.26 | freezey | i mean cisco asterisk=cisco |
| 22:52.56 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
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| 23:06.59 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 23:10.17 | *** part/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 23:10.27 | pfn | freddyk, why would you want to downgrade? |
| 23:12.04 | pfn | calling it sip trunk is confusing in the face of iax trunking... I guess it's not a term coined by the asterisk crowd |
| 23:13.41 | carrar | *** Signoff: freezey () |
| 23:15.10 | jameswf | Only 16 minutes and 26 seconds until the Epoch Time is 1234567890! (Friday, February 13th 2009, 23:31:30 UTC) |
| 23:20.05 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:21.17 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 23:21.29 | *** join/#asterisk cp6 (n=samy@72.37.252.206) |
| 23:22.05 | cp6 | i've sent ~20k calls into asterisk 1.6.0, but there are no active calls right now. asterisk res is at 330mb, is there a mem leak? |
| 23:23.49 | yoanis | jameswf: watch -n1 date +%s |
| 23:23.51 | yoanis | :D |
| 23:29.00 | drmessano | http://coolepochcountdown.com/ |
| 23:29.38 | *** join/#asterisk docelmo (n=vircuser@pool-141-152-199-2.lyn.east.verizon.net) |
| 23:30.03 | jameswf | <PROTECTED> |
| 23:30.08 | Qwell | yoanis: ha, I just looked at the manpage for watch... |
| 23:30.16 | Qwell | <PROTECTED> |
| 23:30.16 | Qwell | <PROTECTED> |
| 23:30.16 | Qwell | <PROTECTED> |
| 23:30.28 | drmessano | 30!!!!! |
| 23:30.34 | yoanis | lol |
| 23:30.59 | drmessano | Happy 1234567890!!!!! |
| 23:31.03 | Qwell | better ntpdate now before the ntp servers all die |
| 23:31.06 | Qwell | drmessano: slow. |
| 23:31.23 | drmessano | SHUDDUP |
| 23:32.24 | Micc | cp6, how many peers do you have? |
| 23:32.43 | jameswf | WE ARE STILL ALIVE.... guy across from me playing celebrate good times.... |
| 23:33.35 | cp6 | Micc, about 50 |
| 23:33.46 | cp6 | 2 PRIs, < 50 peers |
| 23:33.51 | Corydon76-dig | Are you sure you're alive? |
| 23:34.02 | Nugget | come on! (let's celebrate) |
| 23:34.14 | Corydon76-dig | You may have entered into a dreamlike state just now. |
| 23:34.15 | drmessano | Oh crap |
| 23:34.20 | Micc | cp6, what is your average mem usage? |
| 23:34.20 | drmessano | All my calls just dropped |
| 23:34.22 | Corydon76-dig | Prove that you're alive. |
| 23:34.29 | cp6 | Micc, asterisk's? |
| 23:34.33 | drmessano | Y1234567890 BUG! |
| 23:34.36 | Micc | cp6, yeah. |
| 23:34.39 | Qwell | cp6: When'd you get an upgrade? |
| 23:34.52 | cp6 | qwell, i'm just playing with it |
| 23:35.09 | drmessano | Qwell, I got a dozen virtual PRIs now |
| 23:35.25 | cp6 | Micc, well i'm not sure...it was at 100mb earlier, then was at 200mb for a while, then at 330 for a while -- it won't go down from 330mb though and there are no calls going through at all |
| 23:35.27 | Qwell | err...is +r broken? |
| 23:35.47 | drmessano | +R? |
| 23:35.48 | jameswf | drmessano: you twitter? |
| 23:35.52 | Micc | cp6, if you place a call is everything still working? |
| 23:35.54 | drmessano | jameswf: Yeah |
| 23:36.00 | jameswf | url? |
| 23:36.07 | drmessano | <-- drmessano |
| 23:36.08 | andresmujica1 | hi all, i'm working with an asterisk HA, but i'm having some trouble with IAX2 using the virtual ip or 0.0.0.0 as bind address... |
| 23:36.14 | cp6 | Micc, yeah, everything works fine. the problem is the memory usage seems very high for a system with 0 calls |
| 23:36.25 | cp6 | my concern is it will grow during high call usage and not free |
| 23:36.36 | Micc | cp6, its probably fine. do you have history turned on? |
| 23:36.39 | cp6 | i'm not sure if this is a memory leak or not |
| 23:36.55 | Qwell | cp6: I mean your nick |
| 23:36.57 | cp6 | Micc, what do you mean history? i'm logging up to verbose |
| 23:37.02 | cp6 | Micc, haha! |
| 23:37.04 | cp6 | er |
| 23:37.08 | cp6 | Qwell, haha! cp5 timed out |
| 23:37.14 | cp5 | changed internet |
| 23:37.47 | *** join/#asterisk nOgAnOo (i=Gizmo@network184-253.wctc.net) |
| 23:38.00 | Micc | cp6, sip history, it keeps track of sip debug info in memory I assume of calls. |
| 23:38.11 | Micc | cp6, so even if nothing is happening it could have all that history still there. |
| 23:38.57 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 23:39.19 | Qwell | if the world explodes, I apologize in advance. I have no idea what this does. |
| 23:39.20 | cp5 | Micc, ahh |
| 23:39.29 | cp5 | Micc, i'll check that |
| 23:40.21 | cp5 | Micc, yeah |
| 23:40.32 | cp5 | er, sorry, ignore last message |
| 23:40.56 | Micc | cp5, how long has it been running? |
| 23:41.03 | cp5 | few hours tops |
| 23:43.55 | cp5 | i'll turn sip history off and see how it goes |
| 23:46.32 | cp5 | very weird, call won't enter a queue until queues.conf has a new mtime since load and a reload occurs. it assumes the queue is empty if the queue only points to Local/ channels |
| 23:46.48 | cp5 | i have to: touch queues.conf + asterisk -rx reload every time i start asterisk for queues to work |
| 23:46.53 | cp5 | in latest 1.6.0 |
| 23:47.24 | cp5 | seems like it doesn't set member->status until reload i think |
| 23:48.06 | jameswf | damnit Error calling settimeofday({-59357590,0}): Invalid argument |
| 23:48.49 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 23:49.25 | Qwell | jameswf: wanna be a test dummy for something for me? |
| 23:49.52 | jameswf | depends will my wife be mad? |
| 23:49.57 | Qwell | maybe |
| 23:50.27 | Qwell | rejoin... |
| 23:50.28 | *** mode/#asterisk [+b jameswf!*@unaffiliated/jameswf-home!#lolnub] by Qwell |
| 23:50.29 | *** kick/#asterisk [jameswf!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 23:50.49 | Qwell | counts to 10 |
| 23:51.04 | *** mode/#asterisk [-b jameswf!*@unaffiliated/jameswf-home!#lolnub] by Qwell |
| 23:51.26 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 23:51.35 | Qwell | <3 |
| 23:52.23 | Qwell | I'm going to have a lot of fun with that. |
| 23:55.56 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 23:56.31 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:57.11 | Micc | cp5, thats a little worrisome if queuesdon't remember their status. |
| 23:57.30 | Micc | I'm feeling less tempted to upgrade to 1.6 |
| 23:57.41 | cp5 | well this is using Local/ channels only |
| 23:57.44 | cp5 | this bug it seems |
| 23:57.59 | Micc | cp5, I won't have local channels, so hopefuly I'll be ok. |
| 23:58.07 | cp5 | hope so! |
| 23:58.11 | Micc | hows the mem with history off? |
| 23:59.49 | cp5 | well i just produced another crash and am tracking it down |
| 23:59.52 | cp5 | heh |