| 00:01.49 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:02.48 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:03.07 | `Sauron | husimon: yes |
| 00:03.40 | dewd | I've been looking at my billsec problem and can't figure out what the hell is wrong i've updated to 1.4.18 but the problem persists, it appears to be internal (not in cdr driver) as showing the contents of $CDR(billsec) already gives strange output this is after a few secs of calling time (mostly bad words) |
| 00:03.55 | dewd | Executing [h@klanten-menu1:2] NoOp("CAPI/ISDN1#02/7959xxxx-0", "67") in new stack |
| 00:04.46 | dewd | how can after a few f*** the value of ${CDR(billsec)} already be 67 seconds ? |
| 00:05.54 | dewd | it is more like 6 seconds but how can I trust this, ${CDR(billsec)} is always the same as ${CDR(duration)] |
| 00:22.16 | husimon | is it not possible to reload the skinny.conf in asterisk? |
| 00:24.46 | husimon | is there a way to see what asterisk thinks it has in a config file via the cli? |
| 00:24.54 | husimon | I think it's not reloading one of hem |
| 00:25.22 | drmessano | asterisk -rx reload |
| 00:25.49 | husimon | i'm using reload in the cli |
| 00:25.53 | husimon | [Feb 27 14:25:31] WARNING[19400]: loader.c:484 ast_unload_resource: Soft unload failed, 'chan_skinny' has use count 11 |
| 00:26.08 | husimon | that's when I do a module unload |
| 00:27.16 | drmessano | Maybe the Cisco phones have the Never_Die flag set |
| 00:27.21 | drmessano | ~sike |
| 00:27.22 | jbot | [Sike!] The preceeding statement was intented to be humourous, and may have involved false statements used in a humorous manner, or an exaggeration of a real life situation with humourous intent, but in no way should be taken as pure fact or not considered questionable at best. |
| 00:27.30 | drmessano | I dunno |
| 00:30.27 | husimon | hmm |
| 00:30.36 | husimon | guess I can totally reload asterisk |
| 00:30.45 | husimon | no idea why that stupid skinny.conf isn't reloading |
| 00:32.08 | husimon | ok that's odd |
| 00:32.11 | husimon | i just did a realod |
| 00:32.13 | husimon | reload |
| 00:32.17 | husimon | via the cli |
| 00:32.29 | husimon | and it isn't even parsing the skinny.conf |
| 00:32.40 | husimon | hmm i wonder if I took chan_skinny out of my modules.conf |
| 00:33.02 | husimon | nope |
| 00:33.08 | husimon | any idea why it wouldn't reload that config? |
| 00:34.12 | husimon | is there such a thing as asterisk safe restart? like it will restart when there are no channels in use? |
| 00:34.40 | [hC] | yeah, restart when convenient |
| 00:35.08 | husimon | is that conrestart? |
| 00:35.15 | husimon | condrestart |
| 00:35.47 | tzafrir_home | husimon, you can do that via the manager interface. But I don't know of any way of using the cli |
| 00:35.47 | [hC] | ? |
| 00:35.58 | husimon | [hc] nm |
| 00:36.04 | husimon | hmm |
| 00:36.17 | husimon | i've never used the manager interface |
| 00:37.20 | [hC] | oh i was talking about the cli. |
| 00:37.59 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.106.109) |
| 00:38.43 | husimon | [hc] ah thanks, found it. |
| 00:38.53 | *** join/#asterisk stybba (n=stybba@190.10.0.136) |
| 00:38.59 | stybba | hi |
| 00:39.10 | husimon | hi |
| 00:41.22 | husimon | nice it works now |
| 00:43.15 | stybba | sorry for my bad english... |
| 00:43.23 | stybba | can someone explain me the right way to recompile zaptel? |
| 00:43.58 | riddlebox | stybba, I believe it is in a readme file |
| 00:45.21 | *** join/#asterisk seanbright (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
| 00:48.14 | *** join/#asterisk RoyK (n=roy@ip-21-60-149-91.dialup.ice.no) |
| 00:49.08 | *** part/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 00:51.31 | *** join/#asterisk FuturePrimitive (n=stephenm@75.144.231.33) |
| 00:52.59 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 00:53.15 | FuturePrimitive | Hello All. I have setup an OpenSER in front of my PBX to handle a TCP>UDP conversion to Asterisk. Running version 1.4.17, and the call comes to Asterisk just fine, but it keeps saying it is from an Unknown peer and just sends it to the from-sip-external context. |
| 00:53.57 | FuturePrimitive | I know it must be something simple, but how do I get it to recognize the traffic as a peer? I have tried setting up a trunk to the OpenSER box, and a few other manual tweaks, but nothing seems to work. |
| 00:55.39 | FuturePrimitive | It is recognizing the SIP traffic as SIP/+19165551212@SACOCS002.mcc.corp-b770051b0 which is the PBX that is passing the traffic to OpenSER. |
| 00:58.53 | *** join/#asterisk hterag (n=chatzill@210.18.209.85) |
| 00:59.27 | hterag | G'day all, does anyone know if there is a way to get CLI comands to stop truncating? |
| 00:59.31 | hterag | e.g. |
| 00:59.40 | hterag | sip show subscriptions |
| 01:00.26 | hterag | keeps truncating the context and I need to see all of it |
| 01:02.53 | riddlebox | wohoo just found an article on how to do SLA, got a customer who will be happy to have that |
| 01:05.02 | drmessano | ZOMGGGGGGGGGGGG |
| 01:05.14 | drmessano | FreeBSD 7.0 is RELEASE |
| 01:05.18 | drmessano | 1991 is here! |
| 01:05.19 | *** join/#asterisk Jocko (i=daemon@c-67-168-165-185.hsd1.wa.comcast.net) |
| 01:05.30 | Qwell | eww |
| 01:05.51 | JunK-Y | wow, ive downloaded 6.3 2 days ago, since it was still RC |
| 01:06.11 | Qwell | JunK-Y: yeah, I downloaded it about a week ago |
| 01:08.07 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 01:15.17 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 01:18.01 | Mavvie | just moved another 500 lines of my dialplan into an autogenerated file. |
| 01:18.02 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.213) |
| 01:18.44 | Mavvie | 4000 lines in total: 750 lines of macros and 250 lines of hand-written extensions. the rest is all automatically generated. |
| 01:19.18 | J4k3 | damn |
| 01:19.20 | J4k3 | thats painful |
| 01:19.28 | Mavvie | that will teach them to attract new users: they now have to do the provisioning themselves :-) |
| 01:19.37 | J4k3 | haha |
| 01:20.29 | *** join/#asterisk ccesario_ (n=Taikonau@189-47-253-18.dsl.telesp.net.br) |
| 01:20.34 | bkruse | [hC]: We are using your macro-trunkdial fix now, thank you so much! |
| 01:20.40 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 01:20.58 | [hC] | bkruse: oh sweet! :) I actually thought about testing that again, cause it might not actually be doing things right. |
| 01:21.08 | bkruse | It worked for us because the macro works if NULL is based (or a completely empty string, in which case LEN returns non zero, which made it seem like it was working) |
| 01:21.24 | bkruse | [hC]: Its now on the aa50 (appliance), BE-B and BE-C |
| 01:21.25 | bkruse | ty ty ty |
| 01:21.30 | bkruse | jbot: [hC]++ |
| 01:21.41 | [hC] | bkruse: I needed to test that below "n" digits, it would return false... i wasnt sure if it did properly after trying last time. |
| 01:21.55 | bkruse | [hC]: Worked great for me |
| 01:22.06 | [hC] | oh ok good :) |
| 01:22.06 | russellb | bkruse: are you here? |
| 01:22.14 | [hC] | i must have been tired when i was dicking with it last :) |
| 01:22.17 | russellb | here as in the office .. |
| 01:22.29 | bkruse | russellb: yep :D |
| 01:22.39 | russellb | bkruse: head on over here when you have a sec please |
| 01:22.39 | bkruse | russellb: hear my music? |
| 01:22.44 | russellb | yes, i feel it |
| 01:22.49 | bkruse | Yes. |
| 01:25.34 | drmessano | I know how this movie is gonna end |
| 01:25.36 | *** join/#asterisk PepOSX (n=angeldav@190.72.146.61) |
| 01:25.56 | drmessano | Welcome to "Asterisk After Dark" |
| 01:28.49 | *** join/#asterisk angryuser (n=nononon@df01t2-212-194-106-235.d4.club-internet.fr) |
| 01:28.58 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:34.38 | DarkRift | Shouldn't the peer here be darkrift instead of darkrift-081ab940 since darkrift-081ab940 is the channel name and not the peer name ? |
| 01:34.39 | DarkRift | [Feb 27 19:40:18] DEBUG[28622]: chan_sip.c:15806 sip_devicestate: Checking device state for peer darkrift-081ab940 |
| 01:35.06 | DarkRift | Cause that always result in a warning when I have my peers loaded in a realtime config |
| 01:35.09 | DarkRift | [Feb 27 19:40:18] DEBUG[28622]: res_config_pgsql.c:170 realtime_pgsql: 1Postgresql RealTime: Result=0x81c40f8 Query: SELECT * FROM sip_conf WHERE name = 'darkrift-081ab940' AND host = 'dynamic' |
| 01:35.15 | DarkRift | [Feb 27 19:40:18] WARNING[28622]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table sip_conf. |
| 01:37.59 | djs | Hey all. Reading and trying to learn about asterisk, but for now, I am not doing anything with pstn lines, just sip (I guess). Most of the howtos I have read are using zapata.conf, but if I understand correctly, I don't need to, right? |
| 01:40.04 | DarkRift | djs, right |
| 01:40.58 | djs | Thank you. Could you recommend a guide not using zapata, or do I just ignore zapata, and do the rest? |
| 01:41.55 | DarkRift | I'd just ignore it, I'm not a guru either |
| 01:43.18 | djs | Gotcha. Hoping to eventually rid myself of the little ATA box, and have more options and voip phones, too. |
| 01:43.34 | FuturePrimitive | Hey everyone. Is there an easy way to strip the + sign in Asterisk? The + sign is being passed in with the number from OpenSER and then the trunk is adding another + to it, so it is going out the trunk as ++number. |
| 01:43.53 | plik | djs: you'll need zapata - or at least ztdummy (for timing) if you want to use the meet-me conference application... but safe to ignore nor now ... have fun learning |
| 01:44.09 | plik | s/nor/for/ |
| 01:44.33 | plik | no, jbot - thats not what I meant |
| 01:44.36 | djs | lmao |
| 01:44.45 | plik | dumb bot |
| 01:45.11 | plik | s'pose I should have been more specific :/ |
| 01:46.21 | djs | plik - Thank you. Any recommendations for a good guide? You never know if you are getting a bad one or not... |
| 01:46.33 | plik | ~book |
| 01:46.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:46.41 | plik | is pretty awesome |
| 01:47.01 | djs | Nice. Thanks |
| 01:47.12 | plik | the wiki at voip-info.org is useful, but there's a fair amount of wrongness there too |
| 01:48.09 | *** join/#asterisk BeeBuu (n=beebuu@219.132.190.199) |
| 01:49.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5761966dd953c68a) |
| 01:50.39 | djs | I started reading it, but didn't do well with it for some reason. |
| 01:51.14 | plik | it takes a little to get into, but well worth the effort, really |
| 01:52.11 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 01:52.29 | *** join/#asterisk kiscokid (n=ron@38.104.140.82) |
| 01:52.30 | Yourname`` | So, callfiles are lagging on the server. And manager dialing is obviously not a good idea. |
| 01:52.56 | angryuser | callfiles lagging, why ? |
| 01:53.29 | angryuser | i dont have any lag |
| 01:54.05 | plik | nor me, but I don't use callfiles to place that many calls as to notice really |
| 01:55.03 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 01:55.49 | kiscokid | anyone know why calls would suddenly hangup prematurely when calling someone else's voice menu system? |
| 01:55.50 | angryuser | the only thing to take attention with callfiles, it's the best to crete them locally, and then copy to callback, coz asterisk grabs them too fast |
| 01:57.23 | plik | kiscokid: pressing *0 or whatever is defined in your features.conf to disconnect maybe? otherwise no idea |
| 01:58.11 | kiscokid | not touching the dial |
| 01:58.24 | plik | angryuser: I found it's better to mv call files as asterisk can sometimes see partial files when copy is used |
| 01:58.46 | kiscokid | how can I tell if I hungup or the other side hungup? |
| 01:59.05 | angryuser | mv, move? |
| 01:59.14 | plik | angryuser: yes |
| 02:00.21 | angryuser | i create them locally on windows, and move them too, i al using a little internal app, it's like a phonebook |
| 02:00.25 | ManxPower | angryuser: create them somewhere else ON THE SAME PARTITION, then mv them to the spool directory |
| 02:00.49 | plik | yeah, what ManxPower said :) |
| 02:01.01 | angryuser | <ManxPower> moving them on network never had problems |
| 02:02.09 | angryuser | <ManxPower> if one day problems appear, i will change maybe, but not it"s working |
| 02:02.37 | ManxPower | angryuser: if you don't do it the way I said, you can have asterisk try to use a partially created file |
| 02:02.48 | ManxPower | if that is acceptable to you, then don't worry about it. |
| 02:04.07 | angryuser | if one day callback become unstable, i will take your advise in consideration |
| 02:04.37 | Yourname`` | ok |
| 02:04.44 | [hC] | any of you guys done rtp load testing using SIPp? |
| 02:05.05 | *** join/#asterisk Sweet-P (n=me@ip68-111-191-246.sd.sd.cox.net) |
| 02:05.10 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 02:05.11 | [hC] | oops |
| 02:05.25 | Yourname`` | It still lags on picking up the callfile especially when there's some load on the box. |
| 02:06.00 | Yourname`` | Meanwhile, how can we connect an extension to an extension? |
| 02:06.15 | angryuser | huh ? |
| 02:07.07 | Yourname`` | Sorry, I have a dumbass asking me these questions on the phone, and I'm typing it. |
| 02:07.13 | Yourname`` | Makes no sense. Anyhoo. |
| 02:07.21 | Yourname`` | Back to my callfiles stuff.. so what could be the dealio. |
| 02:08.02 | angryuser | Try to do as <ManxPower> said |
| 02:08.45 | *** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
| 02:09.12 | angryuser | create to another dir, located at the same partition, write a script or whatever, which moves the .call file to callback |
| 02:12.36 | Sweet-P | alright, I've got the asterisknow thing installed and I can make calls from my softphone to a cellular but no sounds can be heard on the cell but the other way works fine. what file can I check to make sure I have all of the right ports forwarded? |
| 02:12.56 | Sweet-P | I also cannot call into my asterisk, it say unavailable |
| 02:13.23 | husimon | has anyone ever had speaker phones have worse echo canceling then normal phones |
| 02:14.10 | *** join/#asterisk anthm (n=anthm@68-31-195-6.area4.spcsdns.net) |
| 02:14.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:14.32 | J4k3 | wow, the px10000g looks like a mighty nice asterisk box ;) |
| 02:14.50 | J4k3 | well, a native usb port would really round it out |
| 02:15.40 | J4k3 | oh, and via was too dense to run it off a single psu rail, dammit. |
| 02:16.01 | J4k3 | nevermind, the via-suck never ends. |
| 02:19.22 | Sweet-P | when I call from my cell to my asterisk, I get a "Failed to authenticate user "WIRELESS CALLER" |
| 02:19.52 | J4k3 | Sweet-P: asterisknow leads to anger. |
| 02:20.00 | husimon | anger leads to hate |
| 02:20.03 | Sweet-P | i'm starting to understands that |
| 02:20.06 | J4k3 | trixbox leads to a working system with a microscope in your backside |
| 02:20.09 | husimon | hate leads to the dark side |
| 02:20.15 | husimon | TRIXBOX |
| 02:20.18 | J4k3 | and switchvox just makes me ill |
| 02:20.42 | Sweet-P | i've been trying all of the "easy" methods for asterisk...i think I'm going crazy with them |
| 02:20.59 | J4k3 | yeah, you're making work for yourself |
| 02:21.01 | husimon | you are usually better off spending the week and a half learning the cli |
| 02:21.04 | husimon | and raw config files |
| 02:21.11 | J4k3 | its really kinda sad, since theres no real reason why there isn't a good optimized asterisk distro out there |
| 02:21.14 | Sweet-P | yeah, I think you're right |
| 02:21.49 | J4k3 | trixbox, imho, was heading in a decent direction then they lost direction and tried to jump in your wallet. |
| 02:21.50 | husimon | you can probably implement a subset of features via gui |
| 02:21.53 | husimon | and only have those and it works |
| 02:21.56 | husimon | but if you go outside that box |
| 02:21.59 | husimon | poof you're screwed |
| 02:22.38 | *** join/#asterisk reallost1 (n=reallost@72.169.24.231) |
| 02:23.20 | J4k3 | husimon: well, for me personally, the limits of trixbox were quite acceptable to me |
| 02:23.20 | J4k3 | its just the lack of updates and the newer versions being shit are making me 'learn correctly' |
| 02:23.20 | J4k3 | now I wish I would have started that way |
| 02:23.40 | Yourname`` | Are callfiles going to be phased out? |
| 02:24.48 | Yourname`` | And 1.6 is like a devel tree or like a release tree? |
| 02:25.34 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 02:25.41 | *** join/#asterisk deltaray2 (n=deltaray@adsl-76-248-67-30.dsl.bltnin.sbcglobal.net) |
| 02:26.02 | angryuser | 1.6 beta |
| 02:26.11 | Sweet-P | ok, i give up....what's the best os to run asterisk on? centos? |
| 02:26.13 | deltaray2 | Should I use a surge protector on my phone lines going into an FXO card? If so, what kind. Just the normal kind or is there something specific? |
| 02:26.31 | angryuser | <Sweet-P> i like debian |
| 02:26.44 | Sweet-P | i'm an ubuntu/debian fan as well |
| 02:26.48 | angryuser | <Sweet-P> debian likes asterisk |
| 02:27.58 | angryuser | <deltaray2> yes recommended |
| 02:29.33 | husimon | Sweet-P, honestly just choose the linux distro you know how to use best |
| 02:30.54 | angryuser | debian with minimum install it like 10-20 min till you done compiling/installing |
| 02:31.30 | angryuser | without time for install debian |
| 02:32.21 | reallost1 | Anyone here an expert on sound quality for converting MP3 files for playback on asterisk? |
| 02:32.30 | Sweet-P | i'm most familiar with ubuntu these days |
| 02:32.35 | angryuser | 8000hz mono |
| 02:32.58 | reallost1 | WAV or Ulaw, whats the new native format? |
| 02:33.02 | angryuser | <reallost1> ^ |
| 02:34.01 | angryuser | it is my mp3 answer |
| 02:34.22 | reallost1 | 8k mono mp3? |
| 02:34.34 | deltaray2 | angryuser: Do plain old telephone surge protectors I'd buy in say K-mart affect the voice quality or should I look for a high quality one? |
| 02:34.35 | angryuser | use sound forge, wav work fine also |
| 02:34.46 | angryuser | sam format |
| 02:34.49 | angryuser | same* |
| 02:36.20 | angryuser | <deltaray2> cant say, i live in europe, i am using the 'not the first price' but not high end either, my sound is not affected |
| 02:36.22 | reallost1 | I'm running 1.4.18 and trying to get the best sound quality for playing songs. |
| 02:36.32 | drmessano | Uh |
| 02:36.46 | drmessano | Who gives a crap about MOH Quality |
| 02:37.18 | reallost1 | I do. |
| 02:37.35 | drmessano | Anything more than 8k mono is LOST |
| 02:37.46 | angryuser | <reallost1> nobody will ever head the diff on the phones, aboce 8k useless |
| 02:37.56 | angryuser | above* |
| 02:38.05 | reallost1 | I understand that 8k is how it is played. |
| 02:38.31 | drmessano | No |
| 02:38.39 | angryuser | hear* hate my keyboard at night |
| 02:38.39 | reallost1 | So I'm trying to figure out what will give the best sound quality for the least CPU cycles. |
| 02:38.43 | drmessano | It's more than that |
| 02:39.00 | drmessano | You're still dealing with POTS |
| 02:39.11 | drmessano | It has it's limits |
| 02:39.26 | drmessano | So higher quality than 8k mono is wasted |
| 02:40.16 | reallost1 | drmessano, right. I've heard some horrible sounding MOH and some decent sounding MOH, I'm trying to get mine to sound as good as possible. |
| 02:41.08 | drmessano | So downsample it with a decent editor and don't let Asterisk do it |
| 02:41.35 | plik | you'd be better off with an announcement that says "please download our high quality hold music from www.reallost1squalitychoonz.com" ;) |
| 02:41.41 | angryuser | <reallost1> yes of course if you record your moh from cheap micro, it wikk be horrible whatever format and/or bitrate |
| 02:42.16 | angryuser | will* |
| 02:42.41 | angryuser | have you herd about a good voice generator ? |
| 02:42.46 | angryuser | heard? |
| 02:42.52 | angryuser | online? |
| 02:43.00 | plik | cepstral ? |
| 02:43.04 | reallost1 | drmessano, what do you recommend for downsampling mp3s in batches? sox? lame? something else? |
| 02:43.32 | drmessano | I've only messed with tools on windows |
| 02:43.46 | drmessano | I use Adobe Audition |
| 02:43.58 | angryuser | soundforge here |
| 02:44.07 | drmessano | Ugh |
| 02:44.08 | reallost1 | hmm.. I need a way to automate this... |
| 02:44.17 | J4k3 | bleh, ok... my inner sql geek says 'fuck these conf files' |
| 02:44.57 | reallost1 | J4k3, realtime config is your friend. ;-) |
| 02:44.58 | angryuser | <reallost1> http://www.dbpoweramp.com/dmc.htm |
| 02:45.00 | plik | reallost1: sox will do the trick if not too many at once... you may have to play with settings - trial & error, then batch them when youve found what works |
| 02:45.45 | *** join/#asterisk jameswf-home (n=james@mobile-166-214-039-059.mycingular.net) |
| 02:45.52 | reallost1 | plik, I've been playing around with sox, and the voice quality has just been so so. I |
| 02:45.57 | angryuser | convert one with pro soft, copy params,>>mass convert |
| 02:46.06 | reallost1 | plik, I've been playing around with sox, and the voice quality has just been so so. I'm not sure I'm sending it to the right format. |
| 02:46.22 | plik | hmmm try angryusers suggestion then |
| 02:46.46 | reallost1 | What do you suggest is the proper final format? wav, ulaw, something else? |
| 02:46.50 | J4k3 | jbot: dbpoweramp is http://www.dbpoweramp.com/dmc.htm |
| 02:46.50 | jbot | okay, J4k3 |
| 02:47.14 | plik | an 8k wav sould be fine |
| 02:47.14 | J4k3 | jameswf-home: it could be worse.... you could be in atlanta outside the airport. |
| 02:47.20 | angryuser | wav |
| 02:47.37 | jameswf-home | yeah its 3o degrees outside and 90% locals :)) |
| 02:48.25 | jameswf-home | bring on SC :)) |
| 02:48.40 | reallost1 | thanks |
| 02:49.01 | angryuser | by @All have fun |
| 02:49.10 | angryuser | bye ;) |
| 02:49.16 | plik | au revoir angryuser |
| 02:49.27 | drmessano | Welcome to Georgia |
| 02:49.48 | drmessano | Would offer you a peach, but they're all effin frozen |
| 02:50.03 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 02:50.42 | pigpen | Since I am loosing my mind, I thought I would spread the Joy. |
| 02:50.59 | pigpen | I have a very simple outbound dialplan, calls go out G0. |
| 02:51.03 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:51.39 | jameswf-home | Itell you what |
| 02:51.41 | pigpen | I dial a simple 7 digit number begins, 537xxxx, works. (has for 15 years for that matter, long before asterisk) |
| 02:51.59 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:51.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:52.04 | pigpen | Then, I dial 249xxx, fails, either fast busy, or "must dial a 1 or 0" |
| 02:52.05 | jameswf-home | ha to fight for a spot to connect, power is a luxery |
| 02:52.25 | J4k3 | jameswf-home: carry SLAs in your carry-on |
| 02:52.30 | pigpen | I then dial another local exchange, 755xxxx, works fine. |
| 02:52.31 | J4k3 | and, always, keep Intel Inside (tm) |
| 02:52.55 | pigpen | So just to be sure, I dial a 4th one, 248xxxx, works fine as well. |
| 02:53.16 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
| 02:53.22 | pigpen | All 4 are using the same dialplan. One would expect it to be the telco. Nope. I hook a flatline phone on the line, it works fine. |
| 02:53.41 | pigpen | Any ideas? (Other than the fact I am loosing my mind) |
| 02:53.47 | pigpen | Asterisk 1.4.18 |
| 02:54.24 | russellb | pigpen: zaptel analog FXO dialing out problems? |
| 02:54.52 | jameswf-home | the d00d at the gate says c16 my ticket says c7 and the internet says c1 wtf |
| 02:55.31 | J4k3 | I say.. Cessna 182 |
| 02:55.37 | drmessano | YOU SANK MY BATTLESHIP |
| 02:55.38 | pigpen | russellb, yeah. |
| 02:55.41 | russellb | pigpen: if so, what version of zaptel? |
| 02:56.00 | pigpen | 1.4.8 |
| 02:56.11 | jameswf-home | boing 717 just as bad |
| 02:56.20 | J4k3 | wtf |
| 02:56.23 | jameswf-home | boeing even |
| 02:56.26 | russellb | pigpen: ok, update to zaptel 1.4 from svn, and i'm pretty confident that it will be fixed |
| 02:56.26 | J4k3 | got insurance? |
| 02:56.27 | J4k3 | haha |
| 02:56.28 | pigpen | In fact, you know, I moved to 1.4.18 and 1.4.8 recently....I just remembered. |
| 02:56.40 | russellb | pigpen: there is a known issue in 1.4.8 and 1.4.9 with that that was just fixed last night |
| 02:56.42 | jameswf-home | its only an hour hop |
| 02:56.53 | russellb | yep, it's a zaptel bug introduced in 1.4.8 |
| 02:57.00 | J4k3 | jameswf-home: 35 minutes taxi, 25 minutes in air |
| 02:57.02 | J4k3 | P |
| 02:57.03 | J4k3 | :P |
| 02:57.09 | J4k3 | its like houston/dallas flights here... total waste of time |
| 02:57.12 | pigpen | russellb, thanks...just wanted to make sure I wasn't needing to see a Psych doctor. |
| 02:57.20 | J4k3 | you can damn near drive it faster because of the taxi time and constant delays |
| 02:57.25 | pigpen | russellb, thanks again. |
| 02:57.26 | russellb | pigpen: nope ... it was just hard to find because it only happened for some people, and not all of the time |
| 02:57.30 | jameswf-home | 5 minutes to altitude 2 minutes at altitude and desent |
| 02:57.37 | russellb | pigpen: turned out to be a bizarre uninitialized memory issue |
| 02:57.38 | tzanger | jameswf-home: heh |
| 02:57.48 | pigpen | new zap released soon I am sure. |
| 02:57.51 | *** join/#asterisk supa_disko (n=bleh@secure27.lnk.telstra.net) |
| 02:58.03 | russellb | pigpen: yeah, probably tomorrow |
| 02:58.03 | jameswf-home | yeah between flight and layover I could have.... options were 2hours or 4 hours in ATL i chose 2 |
| 02:58.17 | pigpen | I won't go into my all-page issue, I think it was resolved today as well. |
| 02:58.24 | pigpen | I hope. |
| 02:58.28 | drmessano | jameswf-home: Could be worse.. Try an AT7 |
| 02:58.36 | pigpen | If not, I am pulling in all the debug crap tongiht. |
| 02:58.39 | russellb | pigpen: a crash? |
| 02:58.42 | drmessano | I flew one from AGS to ATL |
| 02:58.44 | pigpen | yeah... |
| 02:58.47 | russellb | pigpen: i saw a commit for that today ... that was a hairy issue, too |
| 02:58.59 | pigpen | I can do 100 pages across 90 phones....works fine. |
| 02:59.05 | pigpen | then out of the blue, it pukes. |
| 02:59.11 | jameswf-home | drmessano: get in your car start driving lets see who shows up first |
| 02:59.18 | russellb | pigpen: the joy of race conditions |
| 02:59.38 | drmessano | jameswf-home: I was sitting on this turboprop, squeezed in like sardines, and I started singing "La Bamba" |
| 02:59.42 | drmessano | Comedy Gold |
| 02:59.51 | pigpen | yeah, it has been hitting me for awhile, but we thought it was more of a max command length issue. |
| 02:59.58 | pigpen | 256 vs. max of 8000 |
| 03:00.17 | jameswf-home | i am afraid if the next confrence is in vegas we will be on the boses personal plane |
| 03:00.19 | pigpen | but kinda hard to find the issue when it does it once every 3 weeks. |
| 03:00.39 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 03:00.40 | J4k3 | see |
| 03:00.43 | drmessano | jameswf-home: As long as he's not a kennedy, you're set |
| 03:00.54 | russellb | pigpen: well thanks for your help in getting resolved! |
| 03:00.57 | J4k3 | I'd rather fly private any day |
| 03:01.12 | fujin | anyone here had any experience with redhat-cluster-suite and Asterisk? |
| 03:01.21 | jameswf-home | by bosses plane I dont bean corprate jet I mean sesna |
| 03:01.23 | jameswf-home | :) |
| 03:01.26 | fujin | and I'm also looking for some comments on ODBC Voicemail - is it stable enough for production |
| 03:01.27 | russellb | fujin: which part of it? |
| 03:01.29 | Yourname`` | Is Asterisk getting rid of callfiles anytime soon? |
| 03:01.29 | jameswf-home | *mean |
| 03:01.31 | J4k3 | too many drunk, coked up, middle-of-a-divorce, wife-just-cheated-on-him pilots on commercial flights. |
| 03:01.34 | Yourname`` | yo Fu |
| 03:01.36 | Yourname`` | fujin: |
| 03:01.37 | russellb | and tons of people use ODBC storage |
| 03:01.43 | fujin | russellb: I want to setup clustering between my two - replacing heartbeatv1 |
| 03:01.51 | fujin | two asterisk servers* |
| 03:01.56 | fujin | Yourname``: howdyhowdy |
| 03:02.02 | jameswf-home | J4k3: anything I should look for to find said women and do they put out |
| 03:02.10 | pigpen | russellb, I wish I could say I helped, I just have been ruling out other crap that is in my control, then decided to start debugging the crashes... |
| 03:02.19 | pigpen | I have had the issue since 1.4.13 |
| 03:02.22 | J4k3 | jameswf-home: great for women, awful for pilots. |
| 03:02.24 | jameswf-home | oh crap read that wrong |
| 03:02.30 | pigpen | but we will see., thanks again. |
| 03:02.37 | russellb | pigpen: reporting it is helpful |
| 03:02.44 | Yourname`` | howdyhowdy |
| 03:02.45 | *** part/#asterisk deltaray2 (n=deltaray@adsl-76-248-67-30.dsl.bltnin.sbcglobal.net) |
| 03:02.49 | Yourname`` | You disappear every other day! |
| 03:02.53 | pigpen | yeah, but I want to make sure it wasn't a "me" thing. |
| 03:02.53 | fujin | indeed |
| 03:02.59 | fujin | Yourname``: system has been stable for a long time |
| 03:03.06 | fujin | got a request for some upgrades ;] |
| 03:03.07 | jameswf-home | lmao at pilots flying 10 miles past hawaii island |
| 03:03.12 | fujin | russellb: so, have you? |
| 03:03.22 | fujin | r-c-s for service failover between 2+nodes |
| 03:03.39 | russellb | fujin: i haven't used all of it, no ... i just try to pay attention to what people are using |
| 03:03.39 | fujin | either using an APC PDU or Vmware ESX for fencing |
| 03:03.44 | fujin | ah, I see. |
| 03:03.49 | fujin | I'll have to do it the hard way then |
| 03:03.50 | russellb | i know heartbeat is a popular tool for HA ... |
| 03:03.55 | fujin | yea |
| 03:04.05 | russellb | on the development side, i have been playing with integrating some services from OpenAIS |
| 03:04.16 | russellb | I have it working for distributed events among a local cluster |
| 03:04.17 | fujin | I've been having issues with it lately, whether it's due to misconfiguration or the package just being terrible I don't know |
| 03:04.24 | Yourname`` | lol fujin nice |
| 03:04.30 | russellb | to provide a high performance method of distributing MWI, and also powering distributed device state |
| 03:04.37 | fujin | heh, that's cool. |
| 03:04.58 | russellb | so hopefully i can keep things moving with that ... |
| 03:05.09 | russellb | but i have heard good results from people using heartbeat |
| 03:05.22 | fujin | it's just kind of messy how it works, I keep getting split-brain |
| 03:05.26 | jameswf-home | <PROTECTED> |
| 03:05.43 | fujin | like the primary will loose ping count to one of the upstream checks momentarily or something and then the other will fail in to master |
| 03:05.49 | fujin | overzealous |
| 03:05.49 | drmessano | LOL |
| 03:05.52 | jameswf-home | they need to import some mexicans |
| 03:05.54 | J4k3 | southwestern food... is this like tex-mex? |
| 03:06.05 | drmessano | jameswf-home: We have lots of mexicans here |
| 03:06.14 | jameswf-home | none of them cook? |
| 03:06.21 | russellb | you guys are a bit too off topic ... |
| 03:06.31 | russellb | i mean ... a bout of sillyness here and there is one thing ... but ... |
| 03:06.51 | russellb | i know, i'm no fun. |
| 03:07.06 | russellb | but it's irritating when i'm trying to have a serious conversation |
| 03:07.40 | fujin | ;\ |
| 03:07.43 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 03:07.51 | fujin | ho-hum. |
| 03:08.00 | nvrpunk | what does "[Feb 28 01:19:58] NOTICE[1749]: chan_iax2.c:7344 socket_process: Rejected connect attempt from 63.211.239.14, who was trying to reach '2055337323@'" mean? |
| 03:08.26 | russellb | that asterisk reject an IAX2 call |
| 03:08.33 | russellb | most likely due to authentication. |
| 03:08.36 | J4k3 | theres always ##asterisk |
| 03:08.38 | [hC] | I wish someone had documented a way to test call load with SIPp, this is confusing as hell. |
| 03:09.47 | russellb | [hC]: this might help .... http://svn.digium.com/view/asterisk/team/murf/bug11210/doc/chan_sip-perf-testing.txt?view=markup |
| 03:09.49 | fujin | it's easy |
| 03:09.51 | fujin | read manpage |
| 03:10.01 | [hC] | russellb: savior. |
| 03:10.04 | fujin | you just pcap an SIP&RTP conversation |
| 03:10.46 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 03:10.57 | [hC] | Well yes ive got that, but from what i understand the length of the rtp stream matters because you must tell sipp to wait.. then the values sipp was reporting seemed to be totally different than what was in progress according to asterisk |
| 03:10.59 | [hC] | i'll play some more |
| 03:11.08 | fujin | hmm |
| 03:11.10 | fujin | who wrote that |
| 03:11.11 | fujin | that's dumb |
| 03:11.14 | fujin | that just initiates sip |
| 03:11.23 | fujin | SIPp needs to bring up RTP for a true representation (imo) |
| 03:11.41 | fujin | where the hell is * going to connect RTp to |
| 03:11.55 | [hC] | yeah. also doesnt it matter that in your pcap, the ip's have to match for your sipp test? |
| 03:12.01 | ManxPower | fujin: Playback |
| 03:12.17 | *** join/#asterisk BeeBuu (n=beebuu@219.132.190.199) |
| 03:13.12 | J4k3 | woo, asterisk realtime seems to be what I'm looking for |
| 03:13.20 | J4k3 | yay, I hate text config files |
| 03:13.25 | J4k3 | they make me wanna tear out my eyeballs |
| 03:14.04 | [hC] | I've got sipp to actually connect and exchange rtp. What I cant guarantee is if its doing it 'properly' (2 way audio, proper pause time) |
| 03:14.28 | [hC] | im really just trying to test how many calls this connection can handle before the internet connection becomes the bottleneck. |
| 03:14.50 | [hC] | and not just presuming that i can calculate bandwidth. I'm stress testing the cpu of the routers, modems, etc. |
| 03:15.07 | [hC] | because i have a theory that the cpu will screw up calls before i hit bandwidth saturation |
| 03:18.03 | fujin | [hC]: generall yyeah, but only if you're transcoding |
| 03:18.11 | fujin | if the endpoints are native cpu usage will be minimal |
| 03:18.21 | fujin | (load on my system is <5% cpu during workhours) |
| 03:18.34 | husimon | so has anyone had problems with conference phones having more echo? |
| 03:18.45 | [hC] | fujin: im not concerned about system cpu usage, im concerned about the dsl modem thats feeding it. |
| 03:19.02 | jameswf-home | I was running 96 channels on OSLEC got my load average to 33.... then there was a flash of light..... |
| 03:19.22 | J4k3 | !! |
| 03:19.38 | jameswf-home | correction trying 96 got 28 |
| 03:19.42 | reallost1 | russellb, I'm doing lots of streaming MOH and having trouble with music caching... |
| 03:20.00 | jameswf-home | We have been beeting up an everex it hates us |
| 03:20.36 | reallost1 | russelb, I read your [moh_hack], but I was wondering if there was a better way yet? |
| 03:20.51 | jameswf-home | bah on carbon neutral |
| 03:20.52 | JT | jameswf-home: a flash of light, like an ex,plosion? |
| 03:21.18 | russellb | reallost1: ah, read that from google? |
| 03:21.24 | drmessano | J4k3: Have you seen the ham rigs with bluetooth on them? |
| 03:21.29 | russellb | reallost1: hmm ... |
| 03:21.34 | russellb | reallost1: that's the best solution right now, yeah |
| 03:21.36 | J4k3 | drmessano: no, but thats an awesome idea. |
| 03:21.56 | reallost1 | russellb, I'm testing this out on 1.6.0b and using realtime MOH |
| 03:21.57 | husimon | drmessano, do you have any conference phones? |
| 03:22.02 | husimon | i'm curious about echo |
| 03:22.09 | drmessano | They have several.. for handsfree + VOX while mobile.. Was thinking about one of those with chan_mobile |
| 03:22.15 | drmessano | husimon: I do not |
| 03:22.20 | husimon | kk |
| 03:22.21 | J4k3 | if PCs weren't so damned noisy RF-wise, we'd all be using ham rigs based on PCI cards and external tx/rx amps. |
| 03:22.40 | jameswf-home | mmm boarding soon ttyl |
| 03:22.46 | reallost1 | russellb, I've got like 100 MOH streams, so the hack isn't really the best option... |
| 03:22.48 | drmessano | cya jameswf-home |
| 03:23.01 | husimon | Qwell, By the way the 7936 works with the normal 79xx config |
| 03:23.03 | husimon | not 79x1 |
| 03:23.10 | reallost1 | russellb, unless I missed something in how its implemented. |
| 03:23.25 | WilliamK | J4k3, you can't get the noise level down on the PC by using a dc power supply? |
| 03:23.52 | J4k3 | WilliamK: maybe, there are lots of switching regulators in the system... mobo/video/etc. |
| 03:24.08 | drmessano | J4k3: http://www.universal-radio.com/catalog/fm_txvrs/0813.html |
| 03:24.25 | russellb | reallost1: no, it's not the best option. |
| 03:24.26 | drmessano | This looks like the answer to Asterisk + Radio |
| 03:24.29 | WilliamK | yeah but if you're using a controlled environment for the dc supply, my thought would be that it would be more stable |
| 03:24.37 | WilliamK | vs shifting from ac to dc |
| 03:24.44 | drmessano | Of course, that's $460 worth of radio |
| 03:25.04 | russellb | reallost1: either the tool you're using to bring in the stream needs to throw away the audio when it's not being read, or asterisk needs to be modified to be able to constantly service a stream, even with no callers |
| 03:25.23 | drmessano | Hmmm $280 with coupons |
| 03:25.31 | J4k3 | thats really not too bad |
| 03:25.39 | J4k3 | well, you need the add-ons |
| 03:25.42 | J4k3 | to make it bluetooth |
| 03:26.04 | drmessano | Yes |
| 03:26.18 | drmessano | $75 |
| 03:26.20 | reallost1 | russellb, can I PM you more on this topic? |
| 03:26.43 | J4k3 | I wish I could get two bluetooth earpieces to work peer-to-peer, and cover about 300-400' range |
| 03:26.47 | J4k3 | that'd be really nice |
| 03:27.02 | J4k3 | beat the shit out of the crap walkietalkies we use for tower climbing |
| 03:27.37 | russellb | reallost1: sorry ... |
| 03:27.47 | russellb | reallost1: i'm doing a few other things right now |
| 03:28.54 | reallost1 | russellb, Do you think the custom programming on asterisk will be the right solution? I'm sure my boss will spring for it. |
| 03:29.22 | drmessano | J4k3: I'm beginning to think that a Bluetooth interface for a radio with chan_mobile is the answer to radio interfacing |
| 03:29.28 | russellb | reallost1: yeah, i don't see a way around it, really |
| 03:30.05 | reallost1 | k, I'll contact Digium tomorrow about getting it written. |
| 03:37.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:39.57 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-06d68911cb01f61a) |
| 03:40.11 | *** join/#asterisk eatmypiano (n=eatmypia@CPE00195b4be0d6-CM001ac3152abe.cpe.net.cable.rogers.com) |
| 03:43.55 | J4k3 | drmessano: hmm good point. that'd be f'n sexy. |
| 03:47.47 | *** join/#asterisk deltaray2 (n=deltaray@adsl-76-248-67-30.dsl.bltnin.sbcglobal.net) |
| 03:48.06 | deltaray2 | I think I have my FXO card working. Is there a way I can test it other than ztcfg -vvv? |
| 03:48.20 | deltaray2 | I have a POTS line plugged into the first channel. |
| 03:51.03 | ecto | try dialing out of it via asterisk |
| 03:51.18 | ecto | or do you not have asterisk installed yet? |
| 03:52.25 | deltaray2 | Yes, I have it installed. I guess what I'm asking is, what is the command to dial out from the asterisk command line. Like if I wanted to call my cell phone. |
| 03:52.37 | ecto | deltaray2: make a call file |
| 03:52.58 | ecto | then copy the call file to /var/spool/asterisk/outgoing |
| 03:53.10 | deltaray2 | ecto: sorry, how do I do that. You can just point me to a website if you want. I still don't have my bearings with asterisk. |
| 03:53.48 | ecto | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 03:54.11 | ecto | deltaray2: can I pm so I don't flood the channel? |
| 03:54.33 | deltaray2 | ecto: Well, I think that's enough to get my started, thanks. |
| 03:55.11 | ecto | basically, to dial out, you just need a Channel, and Application, and a Data line... |
| 03:57.44 | ecto | deltaray2: I usually set Channel to be Zap/X/XXXXXXXXXXX (X is a number) |
| 03:57.49 | [hC] | dialing SIP now instead of IAX (previously Dial(IAX2/user@peer/exten)..) how do i pass the user into the sip dial string? user@peer/exten is not cutting it. and if i leave user@ out, it keeps trying to flop back to 's' |
| 03:59.15 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 04:00.41 | ecto | [hC]: s on the remote side? |
| 04:01.11 | [hC] | yeah. I just added fromuser= to the dialing side, and that took care of it. Dunno if thats right or not. |
| 04:02.37 | [hC] | well. that worked once. |
| 04:03.46 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 04:04.55 | ecto | Normally it's SIP/<peer>/<exten>@<context... |
| 04:05.11 | [hC] | yep |
| 04:05.25 | ecto | I guess you may need to register to <peer> to not have to use a username... |
| 04:05.33 | [hC] | no im not registering |
| 04:05.36 | [hC] | its in the sip digest |
| 04:05.36 | ManxPower | actually SIP does not support @context |
| 04:05.59 | [hC] | I think it might have something to do with once it calls over to the other box, THEN it gets confused and thinks the user should be 's' |
| 04:06.17 | ecto | so it's a configuration issue at the peer |
| 04:06.26 | [hC] | sip show peers shows name/username as "testbox/testbox" until i call the peer once, then its "testbox/s" |
| 04:06.39 | [hC] | lame. |
| 04:08.14 | [hC] | argh, this whole time ive been dicking with pcap files for RTP... sipp supports rtp echo now. |
| 04:08.33 | *** part/#asterisk jeffj (n=jeff@jeffjirsa.net) |
| 04:08.58 | Qwell | now? it has for a while.. |
| 04:09.06 | Qwell | over a year |
| 04:09.13 | [hC] | yeah my bad, it was asterisk 1.2's issue where it wouldnt send rtp if it didnt receive rtp first. |
| 04:09.20 | Qwell | ahh, yeah |
| 04:10.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:12.30 | husimon | Qwell, oh did you catch my earlier message? |
| 04:12.36 | Qwell | yeah |
| 04:12.38 | Qwell | weird |
| 04:12.39 | husimon | kk |
| 04:12.51 | Qwell | 7936 is the same line as the newer ones |
| 04:12.58 | Qwell | that's Cisco for you though |
| 04:12.59 | husimon | 7937 is the new one eh |
| 04:13.03 | Qwell | what? |
| 04:13.10 | Qwell | they made another one? |
| 04:13.16 | Qwell | I bet that is HD |
| 04:13.26 | husimon | woo woo hd audio.... |
| 04:13.32 | husimon | omg hd video conferencing... |
| 04:13.40 | Qwell | video? |
| 04:13.44 | Qwell | err, huh? |
| 04:13.46 | husimon | polycomd has it |
| 04:13.49 | husimon | polycom |
| 04:14.00 | denon | polycom is doing hd video? |
| 04:14.02 | husimon | i was just making a remark about hd enabled audio/video |
| 04:14.03 | husimon | yep |
| 04:14.03 | Qwell | oh, that crazy half million dollar system? |
| 04:14.07 | husimon | nope |
| 04:14.10 | Qwell | or whatever the hell it costs, heh |
| 04:14.11 | husimon | their vsx 9000 series |
| 04:14.22 | husimon | sec finding price |
| 04:14.30 | Qwell | it's like > $100k though, isn't it? |
| 04:15.00 | Qwell | holy crap, the 7937 is nice |
| 04:15.05 | Qwell | http://www.smithonvoip.com/voip-hardware/cisco-cp-7937g-conference-phone-review/ |
| 04:15.31 | husimon | wow they even have a new one |
| 04:15.35 | husimon | rpx |
| 04:15.36 | Qwell | eh? |
| 04:15.40 | husimon | just realized |
| 04:15.42 | Qwell | polycom? |
| 04:15.59 | denon | Qwell: heh, that looks like a calculator duct taped to a conf phone |
| 04:16.02 | Qwell | sweet, 7937 is skinny |
| 04:16.04 | husimon | wow check that shit out |
| 04:16.04 | husimon | RPXâ„¢ HD 400 Series |
| 04:16.07 | husimon | sorry wron glink |
| 04:16.14 | husimon | http://www.polycom.com/usa/en/products/video/visual_communication/vc2_video.html |
| 04:16.27 | Qwell | $1250 for it...huh |
| 04:16.45 | J4k3 | f that, buy a 2mp webcam and a nice DTR laptop to plug it into. |
| 04:17.57 | [hC] | When am i going to see some of the nice perks of digium and polycom working together so closely come to fruition? :) |
| 04:18.13 | [hC] | Id love a videoconf product that worked with asterisk via sip, especially with other devices |
| 04:18.20 | denon | you already are .. polycom does sip |
| 04:18.22 | denon | unlike cisco |
| 04:18.22 | denon | :) |
| 04:18.27 | Qwell | heh, vsx8000 can do skinny |
| 04:18.50 | denon | was mocking that cisco doing sccp-only actualy |
| 04:18.51 | husimon | qwell can't it do sip too? |
| 04:18.52 | denon | actually |
| 04:18.55 | Qwell | dunno |
| 04:19.04 | [hC] | will polycom's sip video conferencing devices work with asterisk? if they have to transcode, rather? say i want one endpoint to be my soft phone.. |
| 04:19.11 | husimon | yeah it can |
| 04:19.22 | husimon | asterisk doesn't really understand video yet right? |
| 04:19.33 | Qwell | it can do video passthrough |
| 04:19.52 | [hC] | so i guess its just a question of wether or not polycom uses a normal codec which other devices use as well. |
| 04:19.55 | denon | you wondering if you can MeetMe video or something/ |
| 04:19.56 | denon | ? |
| 04:20.17 | [hC] | nah, i just want an affordable video conf solution i can sell to people |
| 04:20.22 | husimon | Qwell, so I have a vsx 8000. I wanted to use the soundstation 1000 it has connected to it for a conference phone |
| 04:20.24 | [hC] | and not have to sell them two polycom units specifically |
| 04:20.35 | husimon | wonder if it can do both a h323 call + use it as a conference phone |
| 04:20.36 | husimon | probably not |
| 04:20.44 | [hC] | I want to be able to tell them that they could have say, a boardroom and someone on a laptop |
| 04:21.30 | husimon | [hC], i can just imagine the other ramifications of employees in the bathroom with high res video equipment. |
| 04:22.03 | [hC] | hahaha |
| 04:22.14 | [hC] | so see here's my problem |
| 04:22.21 | [hC] | sipp says: 50 current calls |
| 04:22.23 | [hC] | and i hear no problem |
| 04:22.31 | [hC] | asterisk's show channels says 39 |
| 04:22.49 | [hC] | 10 seconds later, audio is screwed |
| 04:23.34 | [hC] | for f sakes, heh... now sipp says 4 and show channels says 15. |
| 04:23.35 | husimon | qwell, have you had more problems with echo when using conference phones? |
| 04:23.38 | WilliamK | next question... for just you or all channels? |
| 04:23.51 | Qwell | nope, only ever used polycom conf phones |
| 04:24.05 | husimon | how about in general |
| 04:24.18 | husimon | wow that 7937 looks sexy |
| 04:25.12 | WilliamK | Husimon: you need a new gf or something :) |
| 04:25.24 | husimon | oh fuck cisco |
| 04:25.25 | husimon | it's sccp |
| 04:25.33 | [hC] | sccp only yeah? |
| 04:25.36 | [hC] | not surprising |
| 04:25.47 | [hC] | oh qwellllllllll..... :P |
| 04:26.02 | husimon | greedy proprietary fucks |
| 04:26.06 | Qwell | buy me one, I'll get it working :p |
| 04:26.32 | husimon | augh |
| 04:26.34 | Qwell | I really want a 7985 though... that'll likely never happen |
| 04:26.35 | husimon | laugh |
| 04:26.41 | husimon | ebay one |
| 04:26.42 | husimon | hehe |
| 04:26.53 | Qwell | and a 7921, or 7922 when it comes out (I want color, damnit) |
| 04:27.14 | husimon | heh |
| 04:27.21 | husimon | wireless sucks in my building |
| 04:27.24 | husimon | i don't even care to tryone of those |
| 04:28.29 | Qwell | oh, 7921 is color |
| 04:29.38 | husimon | i hate how bad my cell phone converage is near my office |
| 04:29.44 | husimon | i swear you go 30 feet down the hall, it works |
| 04:30.45 | husimon | is it possible to buy a 3g or edge repeater? |
| 04:31.19 | x86 | you can get a GSM bridge that works with asterisk :) |
| 04:31.27 | x86 | make calls out asterisk with your GSM phone |
| 04:31.35 | x86 | I think they make them with CDMA too |
| 04:32.33 | husimon | that doesn't help my cell phone from not sucking |
| 04:33.05 | husimon | unless that can recieve and send data |
| 04:33.30 | *** join/#asterisk reallost1 (n=reallost@72.169.24.231) |
| 04:33.43 | Qwell | woah, woah, woah |
| 04:33.47 | Qwell | 7921 runs Linux |
| 04:34.00 | husimon | jewix by cisco? |
| 04:34.04 | J4k3 | leenooks |
| 04:34.31 | x86 | husimon: 3G rides GSM... with the right bridge I'm sure you could do EDGE across it |
| 04:34.34 | husimon | where do you see this qwell? |
| 04:34.36 | Qwell | hmm, I need to add tls support to chan_skinny apparently |
| 04:34.39 | Qwell | http://www.cisco.com/en/US/products/ps7071/index.html |
| 04:34.53 | J4k3 | husimon: there are some cdma and gsm 'repeater amps' on the market. some are shit, some really work well |
| 04:35.01 | Qwell | mmm, 7975 is out. that's hot |
| 04:35.02 | husimon | that page doesn't say linux |
| 04:35.07 | deltaray2 | ecto: Well, I couldn't get it to call, but I could call it and it said 'goodbye', so I know the hardware is working. :-) |
| 04:35.11 | J4k3 | if you've got gsm and your signal sucks, I suggest trying a cdma phone :) |
| 04:35.20 | deltaray2 | That's good because I had bought a few other cheap x100p cards that didn't work. |
| 04:35.20 | J4k3 | gsm wasn't designed for indoor use |
| 04:35.26 | deltaray2 | This one is a Rhino. :-D |
| 04:35.28 | Qwell | husimon: the video does |
| 04:35.32 | husimon | oh |
| 04:35.45 | husimon | J4k3, yeah gl with that |
| 04:35.48 | husimon | no one uses cdma anymore |
| 04:36.01 | J4k3 | uhm |
| 04:36.07 | J4k3 | verizon and sprint do in the USA |
| 04:36.08 | Qwell | yes! |
| 04:36.09 | J4k3 | and a few others |
| 04:36.09 | J4k3 | alltel |
| 04:36.11 | J4k3 | etc. |
| 04:36.19 | Qwell | 7975 supports g722 for HD, and...iLBC! |
| 04:36.36 | husimon | Qwell, you better be making bank to need a toy like that ;P |
| 04:36.59 | Qwell | 7945 |
| 04:37.08 | Qwell | I'll settle for that |
| 04:37.58 | Qwell | I need to put all these new IDs into Asterisk... |
| 04:38.17 | husimon | lol |
| 04:38.25 | husimon | wtf do you need gigabit on a voip phone for |
| 04:38.40 | Qwell | switch port |
| 04:38.46 | husimon | guess so |
| 04:38.52 | Qwell | only reason |
| 04:38.52 | husimon | to inline into your computer port |
| 04:38.56 | Qwell | well |
| 04:38.59 | Qwell | it is Cisco |
| 04:39.01 | husimon | i have all my phones on a different vlan so it's easy |
| 04:39.06 | husimon | but we have extra ports wired |
| 04:39.07 | Qwell | they probably want to sell more gbit switches, heh |
| 04:39.28 | husimon | are there any cool color applications you can actually add to those? |
| 04:39.42 | husimon | maybe they added a decent frigging directory services |
| 04:39.46 | husimon | instead of limiting it to 32 entries |
| 04:39.48 | husimon | *bleh* |
| 04:39.51 | Qwell | no idea |
| 04:40.09 | Qwell | I don't think it was ever limited to 32...you just had to page |
| 04:40.21 | husimon | it won't let me make more then one page |
| 04:40.44 | husimon | unless there is some xml directive that says page 1, page 2 |
| 04:40.56 | husimon | went through all the trouble of writing a perl script to parse our web page to create a directory of our 60 or so people and then it wouldn't work on the phone. |
| 04:41.12 | Qwell | no, it's just next page, and then it passes in a page thing for the html |
| 04:41.15 | Qwell | well, xml |
| 04:41.37 | husimon | well it's broken for my 7940 then |
| 04:41.47 | husimon | just stops at 32, unless i'm missing something. |
| 04:45.15 | Qwell | one of the entries has to be a next page entry |
| 04:45.19 | Qwell | or something |
| 04:45.31 | Qwell | it's been a while since I've done anything with the xml directory |
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| 04:53.48 | JT | i'm going to ask a silly question |
| 04:54.02 | JT | but is anyone aware of if a cisco 7910 can do SIP? |
| 04:54.07 | JT | or H.323? |
| 04:54.13 | *** join/#asterisk kiscokid (n=kiscokid@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:54.23 | JT | i know it's a pretty old phone :) |
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| 04:55.07 | kiscokid | I asked this question a few hours ago, so please excuse me if you heard it before... |
| 04:56.28 | kiscokid | All of a sudden when I call a certain toll free number and go through their voice menu the line hangs up after a certain amount of time |
| 04:57.29 | kiscokid | while running through the voice menu I have to enter the following dtmf: 1, 1, 1, 2613 |
| 04:57.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:57.36 | kiscokid | then it hangs up |
| 04:58.41 | [hC] | hmm... this asterisk box doesnt seem to be outputting anything RTP wise during this sipp test. |
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| 05:54.56 | drmessano | Hmm |
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| 06:02.35 | scooby2 | whats the best 2 port t1 card to use that is pci? |
| 06:04.04 | [hC] | either a sangoma a102d or a digium te220b |
| 06:04.23 | [hC] | pardon me digium te212p |
| 06:06.21 | scooby2 | hrm |
| 06:06.54 | *** join/#asterisk jameswf-home (n=james@mobile-166-214-210-137.mycingular.net) |
| 06:08.21 | scooby2 | I have the digium te212p. my issues must be software or server hardware based then |
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| 06:10.38 | jameswf-home | gong |
| 06:11.41 | *** join/#asterisk shtoom (n=godson@59.93.127.192) |
| 06:13.12 | [hC] | scooby2: whats your issue? |
| 06:18.43 | *** join/#asterisk _omer (i=omer@lhr-mp-dig-p11-203.brain.net.pk) |
| 06:19.17 | _omer | how would I know the reason behind the last crash of my asterisk? |
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| 06:25.02 | scooby2 | [hC]: seems like dead lock with 1.2.26.2. When the system gets busy it stops taking calls until a restart now. |
| 06:25.16 | [hC] | huh. strange. |
| 06:26.15 | scooby2 | under 1.4 it kernel panics Centos when the system gets busy |
| 06:26.17 | scooby2 | its odd |
| 06:28.32 | scooby2 | guess i should go back to this old sangoma card |
| 06:29.07 | [hC] | id start by checking that you arent sharing an interrupt |
| 06:29.13 | [hC] | then you're running the latest zaptel |
| 06:29.21 | [hC] | if all else fails, digium does support the cards. |
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| 06:32.02 | _omer | how to know about asterisk crash? my asterisk crashes daily in day time.....is there any log that could help me ? |
| 06:32.08 | scooby2 | 225: 0 369899580 IO-APIC-level wct2xxp |
| 06:32.32 | scooby2 | _omer: turn up debugging |
| 06:32.53 | scooby2 | edit /etc/asterisk/logger.conf |
| 06:33.17 | _omer | do I need to reload after making changes in logger.conf? |
| 06:33.40 | scooby2 | find the line that says: messages => notice,warning,error |
| 06:33.52 | scooby2 | change to: messages => notice,warning,error,debug,verbose |
| 06:33.56 | scooby2 | yes, then reload |
| 06:34.09 | scooby2 | and check /var/log/asterisk/messages next time it crashes |
| 06:34.25 | _omer | alright....thanks |
| 06:36.28 | scooby2 | np |
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| 06:50.04 | drmessano | Anyone using the new IPKALL IAX feature? |
| 06:55.16 | reallost1 | dr messano, I had to have them switch me back to SIP |
| 06:59.36 | drmessano | It's user selectable now |
| 07:01.41 | drmessano | Hmmm |
| 07:02.03 | drmessano | "No Authority Found" |
| 07:06.03 | jameswf-home | you will respect my authoritah |
| 07:08.17 | Supaplex | yes we respect your bad breath, shut your trap already |
| 07:08.23 | Supaplex | :) |
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| 07:10.17 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:16.47 | drmessano | What is "NO AUTHORITY FOUND" with cause code 50 |
| 07:16.57 | drmessano | For an IAX call |
| 07:18.12 | awk | lol |
| 07:18.18 | awk | asterisk 1.4.18 is so buggy its scary... |
| 07:18.37 | drmessano | Why is that a bug? |
| 07:18.55 | awk | where? |
| 07:19.00 | awk | im talking about it being a piece of shit |
| 07:19.04 | awk | I have a site with 40k calls a day |
| 07:19.10 | awk | upgraded them, show channels, errr dead lock |
| 07:19.12 | awk | nice! |
| 07:19.19 | awk | rolled back down to 17 |
| 07:19.37 | awk | but have no io threads for iax channels.. fix iax in 18 but fuck up zap channels |
| 07:21.59 | drmessano | Well, you "lol"'ed at my statement and followed with "asterisk 1.4.18 is so buggy its scary...", and therefore I assumed the two were related |
| 07:22.06 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:22.18 | awk | well no i just find 1.4.X such a joke... |
| 07:22.28 | awk | not 1 version i have used that has not been totally %*@(#%@*%#@ |
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| 07:31.09 | drmessano | I got IAX working with IPKALL.. sweet |
| 07:32.37 | reallost1 | drmessano, the problem I had with IPKALL and IAX was them losing my registration |
| 07:32.49 | reallost1 | So I couldn't get calls reliably. |
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| 07:33.57 | drmessano | No registration here |
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| 07:47.42 | Navion | "service announcement codes"? |
| 07:49.42 | BeeBuu | anyone know how many users can be in A conferences? |
| 07:55.42 | Navion | I can't find a limit but from a computational perspective it's a pretty hard thing to do. You have to decode every speech input then add then scale the distribute to the listeners. But probabily for well manered conferences i be you could have 8 or 10 people without too much trouble. |
| 07:56.19 | Navion | If there was a finite (small number) limit I would have thought it would be pretty clearly spelled out. |
| 07:56.21 | BeeBuu | how about if i want meet over 100 guys? |
| 07:56.39 | Navion | If only one tlks at a time... ;) |
| 07:57.02 | BeeBuu | mute many.. |
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| 07:57.58 | Navion | Just need a vox fuction that only letst the loudest speaker send. |
| 07:58.17 | Navion | fuction == function |
| 08:01.01 | drmessano | There is no limit |
| 08:01.08 | drmessano | It depends on hardware |
| 08:01.23 | drmessano | When it starts sounding like crap, you've hit the limit |
| 08:01.32 | BeeBuu | drmessano: sure? |
| 08:01.46 | drmessano | You asked and I answered |
| 08:02.48 | BeeBuu | drmessano: as you know,can i connect conferences on two asterisk server? |
| 08:03.37 | drmessano | It's not a feature of asterisk, but you can make it happen with some fancy dialplan work |
| 08:04.33 | BeeBuu | per conference got 50 users, so connect them |
| 08:05.04 | BeeBuu | 2 conference rooms got 100 users meeting together? |
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| 08:05.59 | Chris-NB | hi |
| 08:06.33 | Chris-NB | if I have a sip parameter set twice (nat for example) in the sip config section. which parameter is used? the first or the last? |
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| 08:10.07 | drmessano | ~yay |
| 08:10.08 | jbot | Yipee! |
| 08:13.52 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 08:14.25 | Chris-NB | if I have a sip parameter set twice (nat for example) in the sip config section. which parameter is used? the first or the last? <- anyone knows that? |
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| 08:16.51 | BeeBuu | drmessano: what's your yay mean? |
| 08:17.00 | BeeBuu | am i right? |
| 08:17.12 | BeeBuu | or incorrect? |
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| 08:53.24 | dandre | Hello, |
| 08:54.28 | dandre | How can I write option string in modprobe.conf to put my tdm800 in french mode? |
| 08:57.07 | tzafrir | options wctdm24xxp opermode=fr |
| 08:57.15 | tzafrir | If you actually need it |
| 08:57.40 | tzafrir | It is also generally recommended to use a file under /etc/modprobe.d on newer distros |
| 08:58.47 | dandre | ok tzafrir |
| 08:59.14 | tzafrir | dandre, also: http://zaptel.tzafrir.org.il/#_module_parameters |
| 08:59.25 | tzafrir | Need to add opermode there, I guess |
| 08:59.40 | dandre | my false ring detection of yesterday hasn't been solved by the batdebounce patch :-( |
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| 09:17.02 | dandre | tzafrir I have done echo -n 1000 > /sys/.../ringdebounce and that solved my ring problem. |
| 09:17.02 | dandre | Does this 'high" value have any drawback. |
| 09:21.33 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 09:22.40 | Jason99 | hello, i just upgraded two servers from 1.2 to 1.4.18 and when i have canreinvite=yes, i often get 482 "Loop Detected"..changing canreinvite=no fixes the problem, but i don't want the RTP going through both servers... any ideas? |
| 09:23.17 | Jason99 | when I do a SIP trace is appears that it thinks there is a loop when the invite and reinvite packets are sent within a short period of time.. |
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| 09:30.16 | Jason99 | oej: you around? |
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| 09:43.29 | patrick-- | Hey everyone. can someone tell me how i can make MISDN use CLIR? |
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| 09:52.53 | patrick-- | xten => _*0#.,1,SetCallerPres(prohib_not_screened) |
| 09:52.53 | patrick-- | exten => _*0#.,n,Dial(mISDN/1/${EXTEN:4}) |
| 09:52.59 | patrick-- | why wont this work? |
| 09:53.10 | patrick-- | (ofc there is an e in front of the first line ;) ) |
| 09:54.02 | *** join/#asterisk GMitre (n=nda@189.24.242.235) |
| 09:55.22 | GMitre | Can some one help-me ? I compiled and configured all in Asterisk but wen i try to log in with my ATA Linksys PAP2T-NA, i receive this error in the debug server: [0]SIP:ICMP Error -1 (4276b33c:5060, 5) ... But this only show in this specific server. |
| 09:56.04 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
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| 10:01.30 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 10:01.56 | madduck | can anyone imagine why my softphone does not ring back? it seems that asterisk tells it that it will ring back while it's making a call, but there's nothing coming out the speakers... |
| 10:04.34 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:04.58 | pif | hi, can I connect an analog modem to a ATA box on the same LAN as asterisk? |
| 10:05.15 | pif | whith reasonable hope of it working |
| 10:05.48 | madduck | with reasonable hope, yes. |
| 10:06.31 | pif | the asterisk box is on a telco T2, can I get 14.4K on the modem? |
| 10:06.48 | madduck | is there a pingback service, like a tets asterisk that you can call, and one you can tell to call you for testing? |
| 10:07.07 | madduck | pif: not guaranteed, but it should work. faxes might be hard though because of latencies |
| 10:07.46 | pif | would a sipura 3000 be a good choice ? |
| 10:08.04 | pif | or are there better boxes out there? |
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| 10:11.55 | sbingner | that would probably be fine... just make sure you do ULAW |
| 10:12.02 | sbingner | or ALAW |
| 10:12.10 | pif | ok, thx |
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| 10:17.26 | yangvnc | What usually causes the other side to become deaf, i cant hear no voice from them...http://openpaste.org/en/5306/ |
| 10:17.50 | defswork | yangvnc: old age seems to be what's affecting me |
| 10:18.34 | yangvnc | haha |
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| 10:30.32 | dandre | tzafrir, will genzaptelconf correctly update zaptel conf file if -M option isn't present? |
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| 10:55.17 | awk | hmm, cdr_custom is that relevant to inbound and outbound |
| 10:55.36 | awk | how can I customize inbound to have certain vars set and outbound certain vars? |
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| 11:13.04 | clickonce | Hello folks |
| 11:13.56 | clickonce | It seems as if the remote end of the call (PSTN) does not receive any DTMF tones. I see in the log that Asterisk does receive them from my handset, but the remote side doesn't do what it should. |
| 11:14.26 | clickonce | "Press 2 for..." doesn't work, it did yesterday. I've tried with all sorts of settings on the dtmfmode variable. |
| 11:26.59 | dandre | How can I get custom cdr fields with mysql backend? |
| 11:28.50 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 11:29.00 | joelsolanki | buyed 6 licenses for g729 |
| 11:29.11 | joelsolanki | we have 2 pstn lines connected to rhino 4port card. it is fxo card. |
| 11:29.21 | joelsolanki | now we have made local calls to go thru pstn lines and long distance calls thru voip provider. |
| 11:29.31 | joelsolanki | i have xlite with g729 & g711 |
| 11:29.41 | joelsolanki | <PROTECTED> |
| 11:29.58 | joelsolanki | and same with long distance. if i dial long distance by selecting both g729 & g711 then calls doesnt go but then i select only g729 then it works |
| 11:30.10 | joelsolanki | how can we solve this issue. i dont know why transcoding is not being done |
| 11:30.20 | joelsolanki | any hints plz ? |
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| 11:49.42 | badcfe | how do i debug dtmf on cli when its coming as rfc2833 |
| 11:49.43 | badcfe | ? |
| 11:50.08 | badcfe | how do i chack that its really coming as rfc2833 and not inband |
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| 12:05.57 | slavon_net | hello all |
| 12:06.43 | slavon_net | what is it? ERROR[23239]: chan_zap.c:9980 zt_pri_error: !! Got a UA, but i'm in state 7 |
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| 12:07.32 | SteveTotaro | is that a bri? |
| 12:08.56 | slavon_net | pri |
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| 12:46.41 | PodMan99a | hey all .. can my asterisk over sip send SMS TXT to my mobile via the smsq command? |
| 12:47.15 | *** join/#asterisk lnx (n=lnx@183-83-56.ip.adsl.hu) |
| 12:47.18 | lnx | hi all |
| 12:53.59 | lnx | there is an extension in a local machine 0003*002; i'm trying call that ~# asterisk -r -x 'Dial 0003*002' ; and * says No such extension '0003*002' in context 'local' ; can you help me please to use call command correctly? |
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| 12:57.09 | dandre | How can I get custom cdr fields with mysql backend? |
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| 13:13.37 | d00gster | guys, I have an astersik server sitting behind a linksys NAT router> got ports 5060 and 10000-20000 inbound forwarded to the astersik server |
| 13:14.10 | d00gster | on the other side of the city, I have a client behind a nat router too trying to connect |
| 13:14.29 | d00gster | sip works and I can dial out but rtp (media) isn't going through |
| 13:14.53 | d00gster | what did I do wrong? are the rtp ports wrong? |
| 13:15.06 | d00gster | I read the how-to but fail to see what's wrong. |
| 13:15.34 | d00gster | I also have nat=yes on both my upstream and extension profiles |
| 13:15.46 | d00gster | also noreinvite |
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| 13:22.39 | Havokmon | My 9133is are going 'No Service' randomly. If I put one in my server room, it works fine for hours. Could it be a cabling issue that affects UDP, but not the PCs connect through the aastras? |
| 13:23.16 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 13:23.21 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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| 13:24.45 | Havokmon | My 9133is are going 'No Service' randomly. If I put one in my server room, it seems to work fine for hours. Could it be a cabling issue that affects UDP, but not the PCs connect through the aastras? |
| 13:25.00 | Havokmon | Is there a way to get Asterisk to use tcp? |
| 13:25.57 | *** join/#asterisk chronos (n=fuckyou@189.7.84.40) |
| 13:26.26 | Havokmon | Hmm I suppose voip doesn't really lend itself to tcp. |
| 13:28.13 | [TK]D-Fender | Havokmon: AMI works great over TCP! |
| 13:28.40 | Havokmon | Message Indicator? :) |
| 13:29.16 | Havokmon | I don't want to run cable just to test a theory :) |
| 13:29.31 | nixguy | i dont belive that's your problem |
| 13:30.37 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 13:30.53 | Havokmon | I'm grasping at straws.. I've reconfigd the phones from scratch to even without xml button configs, just sip info and a speed dial (and what setup-aastra does). |
| 13:32.24 | Havokmon | I pulled the PCs off the aastra hub. All my aastras will go 'No Service' at some point, but it comes back in a couple seconds |
| 13:33.09 | Havokmon | Yesterday I pulled one off a user's desk and put it right in front of me, plugged directly into the POE switch, and it didn't do it for 2 hours |
| 13:33.25 | chronos | here I have standard to dial like 0 + DDD (2 digits) + phone (8 digits). I'm in 1.2, I can use something like _0ZXX{0,7} to get numbers with minus than 8 digits on phone part? |
| 13:33.30 | Havokmon | So far that's the only change that's been successful, even if it's possible coincidental |
| 13:35.15 | [TK]D-Fender | chronos: You'd have to to 8 extens because of the variable length, or you could do 1 unlimited variable length for which you start with a length check. That won't 404 on an invalid one however |
| 13:35.49 | chronos | eh, I make 8 extens to do :) |
| 13:36.00 | chronos | okay, thx [TK]D-Fender |
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| 13:48.44 | fors1 | have a problem. In iax.conf I have peers (type friend). Both peers authenticates to the right "account", but when I receive a call from the first peer in iax.conf, asterisk thinks it comes from the last peer specified in iax.conf. I have added deny=0.0.0.0 / allow=X.X.X.X to both peers, without success |
| 13:50.22 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 13:51.42 | [TK]D-Fender | fors1: use "host=" |
| 13:52.05 | ruied | what do I need to set in asterisk compilation to have mp3 musiconhold? do I need anything special? |
| 13:52.06 | [TK]D-Fender | fors1: and make sure they each have a separate context specified |
| 13:52.21 | [TK]D-Fender | ruied: you need to install asterisk-addons |
| 13:52.45 | fors1 | [TK]D-Fender: the context thing may be the problem, why can't they have the same context specified? I want to send the incoming calls to the same context eventually |
| 13:53.18 | [TK]D-Fender | fors1: just set the host so it lands on the right entry |
| 13:53.40 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 13:54.16 | fors1 | well, i tried that, setting host= to IP of the peer, but then it wouldn't authenticate. the error message said that "host is not dynamic" |
| 13:54.35 | fors1 | (now I have host=dynamic) |
| 13:54.46 | ruied | [TK]D-Fender, SO, I have to compile asterisk, than get the addons and compile it? |
| 13:55.04 | [TK]D-Fender | fors1: If the host is fixed, you don't NEED to register |
| 13:55.15 | [TK]D-Fender | ruied: Yes |
| 13:55.45 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:55.45 | ruied | [TK]D-Fender, ok, going to try... thanks! |
| 13:56.07 | fors1 | damn, didn't think of that. heh. thanks, i'll see if it helps :) |
| 13:57.04 | dandre | How can I get custom cdr fields with mysql backend? |
| 13:57.22 | *** join/#asterisk quigon (n=matias@32.59.64.129) |
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| 13:58.35 | RoyK | someone in here asked a day or two ago how one could start MoH and then record both the music and the caller, just in case they were singing along.. I found out |
| 13:58.44 | *** join/#asterisk beasty_ (n=jdecoste@about/apple/macbook/beasty) |
| 13:59.05 | beasty_ | anyone any idea what can cause this ? |
| 13:59.07 | beasty_ | app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/jdecoste-081d3928 compatible with IAX2/ |
| 14:00.22 | [TK]D-Fender | RoyK: load chan_karoake.so |
| 14:01.10 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 14:01.13 | [TK]D-Fender | beasty_: Th isn't showing us the error, only the result |
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| 14:01.28 | [TK]D-Fender | that* |
| 14:01.51 | Katty | hai |
| 14:02.32 | Daejeo | Got SIP response 503 "Service Unavailable" back from . can anyone elaborate tthe possible causes ?? |
| 14:02.42 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:02.54 | awk | can anyone answer this.. when asterisk is not started with ANSI colors.. how can you do a reattach with say asterisk -rn to display it? |
| 14:02.57 | docelmo | mew! |
| 14:03.00 | awk | or does it have to be started with color support |
| 14:03.09 | docelmo | MEW! MEW! MEW! |
| 14:03.17 | Daejeo | MAOO |
| 14:03.28 | fors1 | [TK]D-Fender: didn't help much. I have peer1 at 10.30.0.250, and peer2 at 10.40.0.5.. when a call comes from peer1, it still says -- Accepting AUTHENTICATED call from 10.40.0.5: |
| 14:03.57 | docelmo | awk you have to start it from the initial point. I dont believe you can start the CLI with colors |
| 14:04.24 | Daejeo | Got SIP response 503 "Service Unavailable" back from . can anyone elaborate about the possible causes ?? |
| 14:04.44 | docelmo | Daejeo what were you trying to do when you got that message? |
| 14:04.47 | Katty | docelmo: herro. |
| 14:04.59 | docelmo | So whats cooking K long time |
| 14:05.13 | awk | docelmo thought so thanks.. |
| 14:05.31 | Daejeo | docelmo: making a outbound call thru my ITSP |
| 14:05.39 | Katty | docelmo: baroo? |
| 14:05.41 | docelmo | awk if you use safe_asterisk to start asterisk then you will have colors.. |
| 14:05.44 | docelmo | eh? |
| 14:06.03 | stansmith | XML::Parser |
| 14:06.11 | Daejeo | docelmo: any idea? |
| 14:06.15 | docelmo | Daejeo well there are MANY things that can cause a 500 error.. Did your ITSP return the 500? |
| 14:06.37 | Daejeo | only 503 first time |
| 14:06.41 | Daejeo | it was ok before |
| 14:07.33 | docelmo | Maybe you sent them something they didnt like. Which typically there are errors for that sort of thing but asterisk kinda cludges everything into one and you get an error that makes no sense,.. Who's your ITSP? |
| 14:07.35 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 14:07.51 | Daejeo | sipplan.com |
| 14:08.13 | Daejeo | as I said |
| 14:08.27 | docelmo | I would say go talk to them.. They probably can help you. Also at the same token SIP traces provide wonderful information the asterisk CLI doesnt |
| 14:09.49 | Daejeo | alright |
| 14:09.53 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-15-77.lns10.syd7.internode.on.net) |
| 14:11.22 | [TK]D-Fender | Katty: O HAI |
| 14:11.25 | stansmith | the #perl IRC channel sucks |
| 14:11.49 | docelmo | Perl sucks in general |
| 14:12.03 | docelmo | gimme some good old fashioned PHP |
| 14:12.09 | Katty | less sucking more positiviness. |
| 14:12.20 | stansmith | docelmo: im writing a perl script right now, for asterisk |
| 14:12.21 | docelmo | but SUCKING is good |
| 14:12.24 | [TK]D-Fender | .... |
| 14:12.28 | stansmith | it is requesting a php page |
| 14:12.32 | Katty | docelmo: Get Out. |
| 14:12.37 | stansmith | ^ |
| 14:12.38 | docelmo | why not use phpagi and just code it out |
| 14:12.44 | docelmo | Katty byte me |
| 14:12.50 | stansmith | php and asterisk on 2 diff servers |
| 14:12.57 | Katty | docelmo: no thanks. i already brushed my teeth this morning. |
| 14:12.59 | stansmith | perl is the glue that holds it all together! |
| 14:13.17 | stansmith | and a lil bit of xml |
| 14:13.18 | docelmo | eww.. stansmith why not do a fast agi to call to the server running the php |
| 14:13.30 | stansmith | needs dynamic text to be generated |
| 14:13.30 | docelmo | which I must say PHPAGI supports.. |
| 14:13.50 | docelmo | Which Im sure php can do.. :) |
| 14:13.56 | stansmith | dynamic text needs to be returned to the asterisk server |
| 14:14.10 | stansmith | using the AGI class with perl seemed like a good fit |
| 14:14.19 | stansmith | perl can parse like its no ones business |
| 14:14.25 | docelmo | ya thats simple enough.. You could use the PHPAGI class for php and do it also |
| 14:14.34 | docelmo | I know cause Im one of the dev's working on it. |
| 14:14.35 | *** join/#asterisk linmax (n=maxi@2002:5b08:5732:b388:2c0:9fff:fe29:99a7) |
| 14:14.35 | stansmith | coulda woulda shoulda |
| 14:14.44 | stansmith | you are a dev working on phpagi? |
| 14:14.48 | docelmo | yes |
| 14:15.04 | Katty | my brain's all foggy *cry* |
| 14:15.06 | stansmith | to be honest, its what i looked at first, my companies core app is one in php, but my boss shunned the idea |
| 14:15.13 | Daejeo | stansmith: if you use PHP , do u face any problem? |
| 14:15.13 | dandre | as AppendCDRUserField is deprecated, How can write it using CRD function? |
| 14:15.23 | stansmith | Daejeo: if i use phpagi? |
| 14:15.42 | Daejeo | yeak |
| 14:15.53 | docelmo | have been for awhile now.. Im working on the next release.. BUNCHES of changes going to be made based on how asterisk is evolving |
| 14:15.57 | stansmith | none really, except that apache and asterisk are on 2 diff boxes |
| 14:16.02 | docelmo | dandre CDR(userfield) |
| 14:16.06 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:16.13 | Daejeo | stansmith: man it does not matter |
| 14:16.21 | Daejeo | u can call external server |
| 14:16.25 | docelmo | stansmith you can use fastagi w/ phpagi to talk to any server |
| 14:16.33 | stansmith | OK LETS ALL YELL AT ME |
| 14:16.39 | dandre | ok for setcdruserfield but for appeluserfeild? |
| 14:16.42 | stansmith | I WENT WITH PERL, CALM DOWN, IT WILL BE OK |
| 14:16.45 | Katty | SUGGESTION OF INSANITY |
| 14:16.54 | Katty | SUGGESTION OF NO MORE CAPS |
| 14:17.00 | stansmith | sorry |
| 14:17.10 | Katty | APPOLOGY ACCEPTED |
| 14:17.22 | stansmith | but yea, im still interested in phpagi, ill prob play around with it at my house |
| 14:17.29 | [TK]D-Fender | oK FiNe No MoRe aLl cApS ThEn! |
| 14:17.33 | dandre | I had write ,Set(CDR(userfield)=${CDR(userfield)}"data") but I am getting an error |
| 14:17.35 | lirakis | any one know what resporged means |
| 14:17.36 | Katty | [TK]D-Fender: i am going to shoot you. |
| 14:17.37 | lirakis | ? |
| 14:17.44 | Daejeo | stansmith: are you familiar with PHP script? |
| 14:17.45 | ruied | I'm compiling asterisk-addons. It's reporting the following error: "format_mp3.c:99: error: ‘struct ast_filestream’ has no member named ‘_private’ format_mp3.c: In function ‘mp3_close’" |
| 14:17.59 | stansmith | Daejeo: im familar with php |
| 14:18.02 | ruied | what could be the problem? |
| 14:18.09 | Daejeo | then go for it |
| 14:18.14 | [TK]D-Fender | lirakis: resporg = org that handles a toll-free # |
| 14:18.23 | stansmith | yea it seemed natural to use phpagi, but my boss said no |
| 14:18.34 | lirakis | [TK]D-Fender: ahh.. responsible organization ? |
| 14:18.35 | Daejeo | ask him why no? |
| 14:18.40 | [TK]D-Fender | lirakis: yup |
| 14:18.41 | *** join/#asterisk kannan (n=kann@123.201.60.116) |
| 14:18.46 | lirakis | [TK]D-Fender: cool thanks |
| 14:18.49 | stansmith | am i allowed to do that? he is the check writer lol |
| 14:19.22 | lirakis | stansmith: what are you trying to do exactly? |
| 14:19.31 | docelmo | stansmith If I was your boss my only requirement is it does X how you get there doesnt matter |
| 14:19.31 | stansmith | trying to do? |
| 14:19.36 | lirakis | stansmith: i mean .. if your using agi .. why does he care if its php or perl or whatever |
| 14:19.37 | docelmo | stansmith does he know how to program? |
| 14:19.42 | stansmith | yea |
| 14:19.49 | docelmo | then tell him to do it. |
| 14:19.50 | docelmo | :) |
| 14:19.52 | stansmith | lol |
| 14:20.02 | Daejeo | then may be your boss does not know php |
| 14:20.10 | lirakis | Daejeo ++ |
| 14:20.18 | stansmith | he coded the entire app in php |
| 14:20.20 | Katty | why do we have to argue about who knows what |
| 14:20.31 | Katty | can't simple questions have simple answers :< |
| 14:20.33 | docelmo | awww Katty feels left out |
| 14:20.36 | lirakis | stansmith: so .. wtf does he NOT want you to use php ? |
| 14:20.44 | coppice | stansmith: so, he can't program :-) |
| 14:20.49 | Katty | docelmo: i think sleepy and grumpy are the words you're looking for. |
| 14:20.57 | stansmith | you guys just hate perl |
| 14:21.02 | Daejeo | then maybe he wanted to learn perl from u |
| 14:21.06 | Katty | stansmith: i agree. |
| 14:21.09 | docelmo | I do and have no problems saying it |
| 14:21.12 | Duplicator | Query: Where can I find which phones support paging? |
| 14:21.14 | Katty | stansmith: i think there's a perfectly good perl answer for you. |
| 14:21.21 | Katty | Duplicator: i know polycom can. |
| 14:21.22 | stansmith | i already have the answer |
| 14:21.32 | Katty | Duplicator: there's a doc on voip info about paging, and how to do it with various phones. that might help. |
| 14:21.36 | stansmith | everything is all done, proof of concept and everything, im just tightening the system up now |
| 14:21.40 | lirakis | stansmith: ... (shrug) .. not sure what the question ever was |
| 14:21.41 | Duplicator | Katty: thx |
| 14:21.46 | Katty | jbot: paging? |
| 14:21.58 | Katty | jbot: intercom? |
| 14:22.00 | stansmith | lirakis: there wasnt ever a question, i was just getting attacked for not using phpagi lol |
| 14:22.01 | Katty | jbot: you suck. |
| 14:22.01 | jbot | and very well I might add |
| 14:22.08 | lirakis | stansmith: oh i see |
| 14:22.13 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 14:23.35 | Daejeo | extremely slow execution times |
| 14:24.15 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 14:24.20 | Katty | stansmith: been there :/ |
| 14:24.25 | Katty | stansmith: people are kinda upity sometimes. |
| 14:24.30 | Daejeo | anyone victim of vtwhite.com? |
| 14:25.23 | ruied | Having problems compiling asterisk-addons, error in mp3. what could be wrong? http://paste.uni.cc/18406 |
| 14:27.15 | tzafrir | I figure you just have to avoid the question mark |
| 14:27.20 | tzafrir | ~paging |
| 14:27.20 | jbot | extra, extra, read all about it, paging is the action of copying memory pages in or out of the swap. |
| 14:27.30 | tzafrir | how useful |
| 14:27.39 | tzanger | haha |
| 14:27.58 | tzanger | jbot, no, paging is the action of flipping through a newspaper, magazine or book. |
| 14:27.58 | jbot | tzanger: okay |
| 14:28.11 | stansmith | ~paging |
| 14:28.12 | jbot | paging is, like, the action of flipping through a newspaper, magazine or book. |
| 14:28.16 | stansmith | thats cool |
| 14:29.01 | tzanger | jbot, paging is also something a modern operating system does when it swaps applications out to disk in order to make room for an application that wants to run. |
| 14:29.02 | jbot | tzanger: okay |
| 14:31.06 | tzafrir | ruied, on what distro? |
| 14:31.32 | tzafrir | asterisk-addons should include mp3lib |
| 14:32.26 | plik | jbot paging is also with regard to telephony, the ability to make instantaneous announcements over the speaker of 1 or phones in a group |
| 14:32.27 | jbot | plik: okay |
| 14:32.29 | tzafrir | ruied, what version of asterisk? What version of asterisk-addond? |
| 14:33.27 | plik | s/1\ or/1\ or\ more/ |
| 14:34.28 | kannan | hello all, any recommendations for service provider to send SMS from my * to mobiles (in usa)? or is it better to invest in a GSM gateway? |
| 14:35.23 | *** part/#asterisk linmax (n=maxi@2002:5b08:5732:b388:2c0:9fff:fe29:99a7) |
| 14:36.37 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:38.10 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 14:38.34 | yang | Are there any special parameters required to load rtp.conf ? |
| 14:38.48 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 14:39.47 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.111.94) |
| 14:39.48 | yang | And, heh most important - If I have in rtp.conf ports 10000-10500 do i need to define these RTP ports also on my VOIP phones ? |
| 14:43.24 | yang | becouse my phone has LOCAL RTP PORT = 5004 |
| 14:43.35 | yang | and on asterisk i have rtp.conf 10000-10500 |
| 14:45.48 | Havokmon | kannan: most mobiles will accept emails directly. For example tmobile is 1111234567@tmomail.com I send notifications to myself this way all the time. |
| 14:45.58 | styelz | yang: thats fine |
| 14:46.26 | yang | Suddenly the sound when i dial in the number becomes distorted, i wonder what could be breaking it? |
| 14:46.41 | styelz | your phone uses 5004 for rtp, asterisk uses 10000-10500 for rtp. |
| 14:47.02 | Havokmon | kannan: http://tinywords.com/mobile.html |
| 14:47.05 | yang | styelz: so my phone doesnt need to "know" about the asterisk rtp ports? |
| 14:47.11 | kannan | Havokmon -> thats kool, thx, i will try . |
| 14:47.11 | styelz | as it serves multiple calls. it needs more than 1 rtp unline your phone |
| 14:47.34 | styelz | yang: no. it will find out |
| 14:47.50 | yang | styelz: and is a special parameter somewhere to enable rtp.conf load? |
| 14:48.09 | Havokmon | whoops tmomail is .net not .com |
| 14:48.19 | kannan | thx |
| 14:48.21 | kannan | :) |
| 14:49.50 | styelz | yang: dunno. none that i know of |
| 14:50.21 | yang | also this error comes unexpected [Feb 28 15:49:56] WARNING[19068]: chan_mgcp.c:4201 reload_config: Unable to get our IP address, MGCP disabled |
| 14:52.44 | *** join/#asterisk mitcheloc (n=mitchel@216.207.245.1) |
| 14:55.20 | kannan | hmm, there is no app_sms itself in my 1.2.24 version * |
| 14:56.17 | *** join/#asterisk adjohn (n=adjohn@p8135-ipad502marunouchi.tokyo.ocn.ne.jp) |
| 14:56.58 | styelz | nothing when you type... module show like sms |
| 14:57.05 | kannan | actually i have the app_sms.c, app_sms.o and app_sms.so in the asterisk |
| 14:57.50 | kannan | yes thx , that show it |
| 14:57.56 | kannan | in the *CLI |
| 14:58.02 | styelz | must be loaded then |
| 14:58.30 | kannan | do we have to use the .call file, or can we test from the CLI? |
| 14:58.36 | kannan | or dialplan? |
| 14:58.47 | styelz | dont know |
| 14:58.51 | styelz | never used it |
| 14:59.30 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 14:59.40 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:59.47 | *** join/#asterisk mltlnx (n=mltlnx@user-12lcfks.cable.mindspring.com) |
| 15:00.02 | styelz | check out http://www.voip-info.org/wiki-Asterisk+cmd+Sms |
| 15:01.08 | kannan | stylez thx , now thats where i am at , |
| 15:02.50 | d00gster | guys, I have an astersik server sitting behind a linksys NAT router> got ports 5060 and 10000-20000 inbound forwarded to the astersik server on the other side of the city, I have a client behind a nat router too trying to connect sip works and I can dial out but rtp (media) isn't going through |
| 15:02.50 | d00gster | what did I do wrong? are the rtp ports wrong? I read the how-to but fail to see what's wrong. I also have nat=yes on both my upstream and extension profiles. also noreinvite |
| 15:03.33 | dandre | I have in my dialplan this: |
| 15:03.33 | dandre | [foo] |
| 15:03.48 | dandre | sorry! |
| 15:04.06 | pigpen | d00gster, double nat won't work. |
| 15:04.25 | Havokmon | d00gster: Did you add the internIP and localnat on asterisk in your sip.conf ? |
| 15:04.35 | d00gster | yes |
| 15:04.37 | pigpen | stick the * box outside your linksys, or setup a VPN.....or use IAX for that matter. |
| 15:04.51 | pigpen | It will never work the way you have it with SIP. |
| 15:04.56 | Havokmon | d00gster: are the * and phone behind the same nat device? |
| 15:05.05 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-164-21.dsl.teksavvy.com) |
| 15:05.12 | d00gster | no |
| 15:05.25 | Havokmon | pigpen: works for me. |
| 15:05.29 | d00gster | Havokmon: no |
| 15:05.45 | pigpen | Havokmon, not using polycom right? |
| 15:05.51 | dandre | I have in my dialplan this: |
| 15:05.51 | dandre | [foo] |
| 15:05.51 | dandre | include => bar |
| 15:05.51 | dandre | [bar] |
| 15:05.51 | dandre | exten = s,1,Macro(some-macro) |
| 15:05.52 | dandre | how can I from some-macro get the name of bar context (and not foo context) ? |
| 15:06.13 | d00gster | Havokmon, what did you do for it to work? |
| 15:06.20 | Havokmon | d00gster: That's all I've ever done, you usually lose rtp when * doesn't have the nat details in sip.conf |
| 15:06.24 | tzafrir | dandre, core show application macro |
| 15:06.26 | *** join/#asterisk alexurc (n=Administ@mail.rjtcpp.com) |
| 15:06.32 | tzafrir | ORIG_CONTEXT, IIRC |
| 15:06.40 | Havokmon | pigpen: no. aastra, xlite, kphone.. |
| 15:07.05 | pigpen | right..some of these support stun and handle it better, but still have their moments. |
| 15:07.07 | dandre | Ok I had tried MACRO_CONTEXT |
| 15:07.11 | tzafrir | dandre, also generally for a paste longer than 3 lines, you better use a pastebin |
| 15:07.17 | d00gster | havokmon: externalip=x.x.x.x you mean |
| 15:07.26 | *** join/#asterisk mltlnx (n=mltlnx@user-12lcfks.cable.mindspring.com) |
| 15:07.35 | Havokmon | d00gster: sip.conf for nat: externip=sancitionedip localnet=192.168.0.0/24 (your internal network) nat=yes |
| 15:07.42 | tzafrir | dandre, MACRO_CONTEXT, right |
| 15:07.42 | pigpen | d00gster, what sip client are you using? |
| 15:08.02 | d00gster | pigpen: x10 |
| 15:08.09 | dandre | but this gives foo instead of bar |
| 15:08.30 | dandre | this was not a paste but a description |
| 15:08.36 | ruied | tzafrir, sorry, I was out, I'm using asterisk 1.4 |
| 15:09.18 | pigpen | d00gster, most of my experience has been with Polycom's for corporations. I never had the time for phones that wouldn't be acceptable for a CEO making $300K a year. |
| 15:09.36 | JenniferAkemi | morning |
| 15:09.42 | *** join/#asterisk zagaya (n=almeli@LPointe-a-Pitre-151-14-9.w81-248.abo.wanadoo.fr) |
| 15:10.01 | tzafrir | dandre, it was not a rule, but a rule of thumb and an advice. I'm not an admin in this channel :-) |
| 15:10.10 | d00gster | pigpen: good for you |
| 15:10.32 | dandre | ok tzafrir ;-) |
| 15:11.09 | pigpen | d00gster, yeah, good for me. Good luck on getting that to work, you'll need it. |
| 15:14.08 | lnx | can i tur off to use /dev/dsp in case of call with >dial from CLI ? |
| 15:15.11 | d00gster | pigpen: thanks |
| 15:16.03 | styelz | kannan: check out http://www.voip-info.org/wiki/view/chan_mobile too |
| 15:16.36 | tzafrir | ruied, what version exactly of asterisk? What version exactly of addons do you try to build? |
| 15:16.39 | kannan | hmm, ok now the smsq cmd work from the linux prompt |
| 15:16.43 | tzafrir | on what linux distro? |
| 15:17.01 | styelz | cool |
| 15:17.24 | kannan | it places a call thru /var/spool.asterisk/outgoing , i didnt write an extension for the default , now asterisk wont stop attempting to send the SMS |
| 15:17.31 | kannan | lol |
| 15:17.39 | styelz | haha |
| 15:17.45 | wmaulik | can anyone link me the page to download the pdf file for "Asterisk the future of telephony" 2nd edition |
| 15:18.02 | nixguy | im planning to use the asterisk app disa, so users cann use the asterisk to call to other countries from the pstn |
| 15:18.11 | nixguy | whats the difference of using the auth in disa |
| 15:18.11 | ruied | tzafrir, Asterisk SVN-branch-1.4-r77490; Addons: 1.4.6 |
| 15:18.16 | nixguy | compared to Authenticate |
| 15:18.20 | styelz | ~book |
| 15:18.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:18.30 | kannan | aha , it stopped after X number of re-tries |
| 15:18.39 | styelz | cool |
| 15:19.10 | kannan | :(, it says wont work thru a VOIP line using compressed codec, need a PSTN line that supprots text messaging |
| 15:19.21 | kannan | brb |
| 15:19.45 | kannan | stylez :hey the second link is really KEWL, cannot wait to try it |
| 15:19.56 | styelz | yea. same |
| 15:20.20 | wmaulik | thanks |
| 15:20.36 | styelz | np |
| 15:20.51 | *** join/#asterisk jomarie (n=asterisk@158.59.192.218) |
| 15:21.33 | drfreeze | Anyone here use NEC's SoftPhone on one of their thin clients? |
| 15:24.39 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 15:24.52 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 15:26.33 | [TK]D-Fender | nixguy: Functionally little... due to taht they should yank out the auth code entirely and leave it up to the dialplan. |
| 15:30.14 | *** join/#asterisk GrumpManAtWork (n=meanderi@pool-72-78-182-204.phlapa.east.verizon.net) |
| 15:31.10 | nixguy | [TK]D-Fender: k thanx ill use Auth then, dont wanna get phreaked :) |
| 15:32.37 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:32.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:38.54 | kannan | Havokman : what is the SMSC number for an USA provider? |
| 15:39.36 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:40.37 | Havokmon | kannan: I don't know. I was giving you an alternative to SMS by using email. |
| 15:42.14 | kannan | Havokman : thanks |
| 15:45.36 | pif | tzafrir: hi, you here? |
| 15:46.44 | *** join/#asterisk CVirus (n=GoD@82.201.174.81) |
| 15:48.59 | scooby2 | <PROTECTED> |
| 15:49.44 | pif | anyone know a quickfix to build zaptel with kernel 2.6.24 ? |
| 15:49.48 | *** join/#asterisk SpaceBass (n=SP@pool-72-84-196-212.rcmdva.fios.verizon.net) |
| 15:49.50 | SpaceBass | hey folks |
| 15:50.25 | SpaceBass | I have a problem that has persisted over plane asterisk, trixbox, elastix, etc .... when two people speak at the same time it gets very distorted and sounds like cymbals clashing |
| 15:50.34 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:50.36 | SpaceBass | this is over a VoIP trunk (iax or sip_ |
| 15:50.40 | nixguy | join a band! |
| 15:50.51 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 15:50.51 | SpaceBass | ha! |
| 15:50.54 | SpaceBass | Almost could |
| 15:55.29 | *** join/#asterisk highvoltz (i=rogers@bling.bling.org) |
| 15:55.46 | highvoltz | been having a weird problem, where incoming calls are showing as 0000000000 and you cant talk or hear the person on the other end. Any idea what could be the problem? Asterisk box to the internet through a SIP account. Box has an external IP (NO NAT) |
| 15:57.55 | stansmith | SpaceBass: are you related to anyone named becca? |
| 15:58.14 | kannan | This link says , text SMS to USA and Canada from * systems is notably absent, so they provide a gateway --- http://www.bayhamsystems.com/asterisk.html |
| 15:58.44 | SpaceBass | stansmith, hummmm not that I can think of right off the bat...why? |
| 16:01.11 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 16:01.59 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:01.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:02.06 | pigpen | any way to have an iax trunk place the call regardless of what it think it's state is? ie: I have a DS3 with a trunk, now and then, it says latency jumps to 220ms and incoming calls will not be placed. |
| 16:02.19 | tzafrir_laptop | pif, use latest zaptel |
| 16:02.30 | pigpen | if the DS3 actually does hit 220ms, it recovers almost immediatly. |
| 16:03.03 | denon | pigpen: you could set the qualify much higher |
| 16:03.04 | denon | in iax.conf |
| 16:03.12 | denon | say, qualify=3000 |
| 16:03.20 | denon | if it's over 3k, you probably really don't want to use it |
| 16:03.29 | pigpen | then it would poll less often, right? |
| 16:03.33 | tzafrir | pif, latest zaptel . Either 1.4 or 1.2 |
| 16:03.51 | denon | pigpen: well, it would wait longer before thinking it's unavailable |
| 16:04.04 | pif | tzafrir: brb |
| 16:04.07 | denon | like a ping timeout |
| 16:04.16 | pigpen | ah, so setting it lower actually makes it more sensitive. |
| 16:04.34 | stansmith | SpaceBass: is bass your last name? |
| 16:04.37 | tzafrir | pif, what version do you have? |
| 16:04.51 | denon | pigpen: more or less |
| 16:04.59 | pigpen | Dam, I had it backwards.... |
| 16:05.04 | pigpen | tks. |
| 16:05.09 | pif | tzafrir: I'll be right back, phone |
| 16:05.09 | SpaceBass | stansmith, no ... the nick is Space Bass (rhymes with face) ...and very different from my real name |
| 16:05.48 | stansmith | o...my real name isnt stan or smith |
| 16:05.49 | denon | pigpen: a quick googles yields these, might be helpful to you: |
| 16:05.50 | denon | http://www.voip-info.org/wiki/view/Asterisk+sip+qualify |
| 16:06.03 | pigpen | yeah, I am reading the IAX ver of it now. |
| 16:06.31 | denon | and http://www.voip-info.org/wiki/view/Asterisk+iax+qualify |
| 16:06.35 | denon | nod |
| 16:07.07 | denon | you an also just turn off qualify |
| 16:07.08 | pigpen | Sad thing is I have read some of this before...but not "How Qualify Works".... RTFM right? |
| 16:07.38 | pigpen | So essentially, place the call, I don't care what is going on.... |
| 16:07.59 | denon | well, you should care somehow |
| 16:08.02 | denon | if it really is down hard |
| 16:08.11 | denon | you should find an alternate path |
| 16:08.22 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 16:08.37 | denon | but yeah, I heard ya, you can just shut it down |
| 16:08.57 | cesar_CR | somebody that can help me with iax2 cause I can not get 2 * to connect to each other.. ?? |
| 16:09.01 | pigpen | Yeah, actually this is a DS3 & DS1 doing EBGP. |
| 16:09.01 | lnx | what is the best solution to auto dialing a number, if ringing hangup :) else do something ? |
| 16:09.09 | pigpen | denon, tks. |
| 16:09.12 | denon | np |
| 16:09.31 | Duplicator | Can anyone tell me the status of the 'parked call bug' where the parked ext. is not announced? |
| 16:09.51 | Katty | what's everyone's opinon of ShoreTel? |
| 16:10.01 | Duplicator | FWIW, I'm on 1.4.17-1 |
| 16:10.42 | lnx | i think AGI perl and extensions in my case |
| 16:12.50 | *** part/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 16:17.00 | *** part/#asterisk alexurc (n=Administ@mail.rjtcpp.com) |
| 16:18.53 | pif | tzafrir: hi again, I'm using debian/sid packages |
| 16:19.20 | pif | 1:1.4.8~dfsg-1 |
| 16:22.13 | pif | tzafrir: and I get ""/usr/src/modules/zaptel/Makefile". Fix it to use EXTRA_CFLAGS. Stop." |
| 16:24.09 | *** part/#asterisk SpaceBass (n=SP@pool-72-84-196-212.rcmdva.fios.verizon.net) |
| 16:28.44 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 16:30.07 | x86 | Where are changelogs stored for Asterisk? |
| 16:30.28 | x86 | I'm running 1.4.12.1, and not really having any problems with it, but I want to see what's changed between that and 1.4.18 |
| 16:33.46 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 16:35.13 | flujan | hi all. I need to purchase a board to do the g729 compresion on a hardware. Does the T400C supports it? |
| 16:35.30 | flujan | Digium is not activating my codecs so I go straight forward for the board. |
| 16:39.04 | mvanbaak | x86: did you try the CHANGES file ? |
| 16:42.10 | x86 | mvanbaak: nope |
| 16:42.25 | x86 | can I check that out online without doanloading the new tarball? |
| 16:42.32 | *** join/#asterisk steliosk (n=Stelios@athedsl-284196.home.otenet.gr) |
| 16:44.02 | anonymouz666 | flujan: Vc adora esquecer nosso canal em pt_BR |
| 16:44.10 | x86 | asterisk.org points me to the SVN commit log for changes, which is cryptic and doesn't give a concise view of the changes (unless, of course, I was heavily involved in Asterisk development) |
| 16:47.20 | Katty | i have hiccups! :< |
| 16:47.43 | Qwell | flujan: TC400 does G.729, yes |
| 16:48.33 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:48.36 | yang | Could someone advise me the issue about zaptel channels...In my case I have only defined channels 4&5 in zapata.conf, does it mean that if the call is coming to the first channel that I cannot hear it (this is what happens) - here is the config http://openpaste.org/en/5310/ |
| 16:49.38 | RoyK | http://karlsbakk.net/callweaver/karaoke.php |
| 16:49.48 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 16:49.48 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 16:50.13 | flujan | Qwell: how many simultaneos calls does it support? and I can stack two of them on a server? |
| 16:50.31 | flujan | Qwell: I am just asking before buy it... The page with the product have not much information... |
| 16:50.35 | Qwell | flujan: I think...120? and as far as I know, you can. |
| 16:50.42 | Qwell | call sales and ask them :) |
| 16:50.46 | flujan | Qwell: and asterisk automatically uses it? |
| 16:50.48 | Qwell | they'll have firmer numbers |
| 16:51.12 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:51.25 | flujan | firmer numbers? |
| 16:51.35 | mvanbaak | x86: http://svn.digium.com/view/asterisk/branches/1.4/CHANGES?view=markup |
| 16:51.35 | flujan | Qwell: ah, ok... |
| 16:51.40 | flujan | thanks for the help. :D |
| 16:51.42 | Qwell | I'm not sales, so I don't really know for sure. |
| 16:51.44 | Qwell | they'll be sure |
| 16:51.48 | exvito | hi all.. this is a bit OT: anyone knows how to disable/hide the "factory reset" menu option on the linksys SPA phones ? |
| 16:56.13 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 16:56.22 | nixguy | are there any nice digital receptionist gui thingys for asterisk? |
| 16:56.47 | exvito | (sorry, lost my connection -- repeat) a bit OT: anyone knows how to disable/hide the "factory reset" menu option on the linksys SPA phones ? |
| 17:02.03 | iCEBrkr | [TK]D-Fender: You around? |
| 17:03.33 | ber___ | heh cyberdyne |
| 17:03.58 | ber___ | wasnt that part of cyberdrive? |
| 17:04.03 | iCEBrkr | Interesting, I didn't think my domain resolved. |
| 17:04.15 | iCEBrkr | oh you're on cow. |
| 17:04.26 | ber___ | heh POW |
| 17:04.43 | ber___ | i thought i recognized your nickname from efnet |
| 17:05.00 | iCEBrkr | Yeah |
| 17:05.05 | iCEBrkr | So you know chris? |
| 17:05.10 | iCEBrkr | and probably Bruce |
| 17:05.26 | iCEBrkr | Cyberdyne = 216 BBS back in the day :P |
| 17:05.50 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 17:06.03 | tdonahue-laptop | hi all |
| 17:06.27 | tdonahue-laptop | i'm getting an error when i get calls from my carrier |
| 17:06.30 | tdonahue-laptop | Found no matching peer or user for '207.138.151.33:5060' |
| 17:06.58 | tdonahue-laptop | even though there is a sip account (friend) setup with that host |
| 17:07.12 | tdonahue-laptop | how does one resolve this error? |
| 17:07.21 | iCEBrkr | I need a Polycom person :P |
| 17:07.43 | iCEBrkr | Wondering if there's a way to configure vlan settings via the XML file |
| 17:08.39 | tdonahue-laptop | iCeBrkr, i don't remember there being a way, but haven't used the latest firmware... it can be set by CDP though |
| 17:09.29 | iCEBrkr | tdonahue-laptop: Hrrm, ok.. Cuz at the moment, we have to touch each phone and manually set this up. |
| 17:09.49 | tdonahue-laptop | no cisceze switches then? |
| 17:09.57 | ber___ | yeah i know chris and bruce. |
| 17:10.28 | iCEBrkr | ber___: cool, cool. |
| 17:10.49 | iCEBrkr | tdonahue-laptop: It's client location, it could be anything. |
| 17:10.57 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 17:11.32 | tdonahue-laptop | ahhh... check their admin manuals, they have all the manuals on their site |
| 17:11.33 | stansmith | ~timeout |
| 17:11.34 | jbot | ACTION sullenly sits in the timeout chair in the corner for 5 minutes. |
| 17:14.01 | nixguy | what do you guys think of freepbx vs asterisk? |
| 17:14.32 | stansmith | ~bugs |
| 17:14.35 | iCEBrkr | woot. |
| 17:14.35 | iCEBrkr | device.set = 1 |
| 17:14.35 | iCEBrkr | device.net.vlanID = |
| 17:14.35 | iCEBrkr | device.net.vlanID.set = 1 |
| 17:15.15 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:16.59 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 17:17.04 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 17:17.35 | tdonahue-laptop | no one has any idea how to fix the 'No matching peer or user for' error? |
| 17:18.20 | pigpen | iCEBrkr, i had the understanding that the polycom's will get it's vlanid from dhcp. |
| 17:18.53 | cesar_CR | hello I get this msg when try to call another asterisk via iax trunk Unable to create channel of type 'IAX2' (cause 66 - Channel not implemented) |
| 17:19.39 | *** join/#asterisk mrdigital (n=rob@65-78-101-179.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 17:20.41 | pigpen | cesar_CR, not absolutly sure, but do a iax2 show peers, if it borks, then IAX probably is not active in the current running thread. Maybe it had an issue. Stop asterisk and start back up. |
| 17:20.49 | iCEBrkr | pigpen: Yea, but it needs to be on the vlan to get dhcp :P |
| 17:20.50 | *** join/#asterisk hfb (n=hfb@pool-72-67-142-193.lsanca.dsl-w.verizon.net) |
| 17:20.51 | [TK]D-Fender | cesar_CR: means chan_iax2.so is not loaded. Go try to laod it |
| 17:20.51 | pigpen | I have had IAX bork after minor mods. |
| 17:21.23 | pigpen | iCEBrkr, yeah...that's what a correctly switch is for. |
| 17:21.29 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 17:21.36 | pigpen | geesh..I wish I could type. |
| 17:21.41 | iCEBrkr | haha |
| 17:21.42 | pigpen | iCEBrkr, yeah...that's what a correctly configured switch is for. |
| 17:22.47 | cesar_CR | [TK]D-Fender, pigpen it say 0 IAX peers in bouth servers, is iax not loaded ? and how do I load it ? |
| 17:22.48 | pigpen | Also, you could to a lease for the phone, identifying the vlan id static. |
| 17:23.27 | iCEBrkr | ber___: sorry |
| 17:23.29 | iCEBrkr | lol |
| 17:23.30 | pigpen | 0 iax peers? that is why no channel is avail. Peer is not being recognized. |
| 17:23.44 | [TK]D-Fender | cesar_CR: module load chan_iax2.so |
| 17:23.45 | *** join/#asterisk draygon (n=draygon@216.52.176.254) |
| 17:24.16 | iCEBrkr | pigpen: well switch configured vlan is a Cisco thing, no? |
| 17:25.03 | pigpen | if it is a cisco switch, yeah... |
| 17:25.24 | cesar_CR | pigpen, [TK]D-Fender wow, got a lot of syntax error on my file... so I'll keep traying |
| 17:30.29 | *** part/#asterisk dbtid (i=m1bmcfmr@cpe-65-24-173-154.columbus.res.rr.com) |
| 17:31.32 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 17:33.22 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 17:36.03 | drmessano | Jeez |
| 17:36.46 | drmessano | Google is giving grandcentral numbers to all the homeless folks in SF.. Because, you know, you can dial into your voicemail from a tree stump |
| 17:36.58 | *** join/#asterisk nirz (n=nir@77.125.24.166) |
| 17:40.36 | drmessano | I cant ever get enough of how Google helps out |
| 17:40.52 | drmessano | "Free GMAIL invites for the homeless" |
| 17:40.57 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:41.07 | drmessano | "Free Google Apps accounts for the homeless" |
| 17:41.08 | stansmith | they need a myspace too |
| 17:41.19 | Qwell | drmessano: google domains? |
| 17:41.37 | drmessano | "Free .hobo domains for the homeless" |
| 17:41.41 | iCEBrkr | HoboSpace? |
| 17:41.44 | stansmith | lol |
| 17:42.18 | drmessano | They honestly have to be doing this as a joke |
| 17:42.30 | Qwell | why? |
| 17:42.48 | Qwell | surely there are payphones in SF. It's not like an 800 call costs money |
| 17:43.10 | drmessano | I doubt they're tollfree numbers |
| 17:43.19 | drmessano | They're Grandcentral accounts.. |
| 17:43.20 | Qwell | no, the number to check voicemail |
| 17:44.28 | drmessano | Hmm |
| 17:44.56 | drmessano | I didnt see anything about an 800 number to check VM... it looked as though they simply gave them grandcentral accounts |
| 17:46.20 | drmessano | Which is about as good as Digium giving out Asterisk CDs to the homeless |
| 17:46.33 | Qwell | drmessano: that's a bad idea? |
| 17:46.41 | Qwell | I better call our marketing folks, and tell them to wait |
| 17:46.49 | drmessano | HA! |
| 17:46.59 | drmessano | I can see it now |
| 17:47.23 | drmessano | "Digium gives out free PBX software to homeless" |
| 17:47.51 | drmessano | That would be Digiums last stand |
| 17:48.11 | drmessano | Because someone would try to eat one, choke, and then the lawyers would step in |
| 17:48.33 | [TK]D-Fender | stansmith: They're not much bigger than 2 meters/.... |
| 17:48.35 | drmessano | and Jackie "No Toes" Johnson would own the company |
| 17:48.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:49.00 | stansmith | [TK]D-Fender: LOL |
| 17:49.49 | drmessano | Now... |
| 17:50.20 | Kobaz | hmm |
| 17:50.26 | Kobaz | i can't get originate to work |
| 17:50.29 | Kobaz | originate Zap/g1/18005558355 2324@internal |
| 17:51.05 | drmessano | An Asterisk Appliance plugged crudely wired into a lamp post in golden gate park would be a Flickr moment |
| 17:51.56 | drmessano | Put a Wireless adapter on there and glue a Polycom to the side |
| 17:52.04 | drmessano | and a big note that says "Do not Eat" |
| 17:52.44 | [TK]D-Fender | Kobaz: helps when you use all the rigth keywords.... |
| 17:52.47 | [TK]D-Fender | right* |
| 17:53.01 | [TK]D-Fender | Kobaz: Usage2: originate <tech/data> extension [exten@][context] |
| 17:57.15 | *** join/#asterisk markh1158 (n=markh@1016.dca.net) |
| 18:00.35 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:00.35 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:00.57 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 18:01.18 | madduck | so now that i have asterisk running on my home gateway, i'd really rather put voicemail stuff on a proper server |
| 18:01.32 | madduck | is it okay/common to put asterisk on a server in a data centre? |
| 18:01.39 | madduck | or is this what openser is for? |
| 18:02.02 | Kobaz | yeah even with extension it's not doing anything |
| 18:02.03 | Kobaz | hmm |
| 18:02.26 | Kobaz | [TK]D-Fender: apparently i was reading an old doc that didn't use extension |
| 18:03.25 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:05.11 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 18:05.16 | TJNII | Hmmm... That's an interesting idea, hosting voicemail on another server. I wonder what the best way to do it is, a IAX trunk calling voicemailmain on the remote server comes to mind, but it seems a trifle hackish. I wonder, does * support that natively somehow? |
| 18:05.40 | madduck | i thought voicemail is just recording to wav and mailing off |
| 18:06.10 | TJNII | With wrappers and menus, yes. |
| 18:06.10 | Kobaz | TJNII: could nfs mount |
| 18:06.18 | TJNII | True |
| 18:06.24 | TJNII | Share /var/spool/asterisk |
| 18:07.39 | *** join/#asterisk lyroy (n=lyroy@bas1-montreal02-1096727705.dsl.bell.ca) |
| 18:09.30 | lyroy | Does someone can tell me how can I do something like this: I have a voip provider to do my calls but when a user register to my server and try to call another user on my server... I want to be able to andle this call (my asterisk server) instead of going out on the pstn of my provider? |
| 18:11.24 | TJNII | You give each user an extension |
| 18:11.29 | TJNII | ~book |
| 18:11.29 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:11.40 | TJNII | Read up on the dialplan. |
| 18:12.16 | lyroy | thx |
| 18:13.29 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 18:15.27 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:19.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:22.10 | TJNII | Phoo.. Lot of 7 Polycom Soundpoint 500s. Let's see how high this one gets. |
| 18:22.20 | TJNII | Last lot of 8 I found went for over $800 |
| 18:23.32 | *** join/#asterisk ManxPower (n=manxpowe@207.191.118.2) |
| 18:25.56 | TJNII | Anyone played with a SNOM 220? Any good? |
| 18:25.58 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 18:26.00 | Nivex | A friend of mine has an IP501 collecting dust for over a year. I've sent another friend to extract it from his posession. |
| 18:26.25 | ^shark_ | hi guys -- i have a 7912g Cisco ip phone that is locked in reboot, how can i solve this issue? |
| 18:26.52 | TJNII | Nivex: Ooh, yea. Get it. |
| 18:27.26 | Nivex | I'm in NC and it's in OH. I'm hoping it makes its way to my parents' place. |
| 18:27.44 | Nivex | <3 VPN :) |
| 18:27.55 | markh1158 | ^shark_: you should be able to fix it by trying to force it to load a specific image via the TFTP config files |
| 18:28.09 | markh1158 | is it running SIP? |
| 18:28.37 | ^shark_ | markh1158: how is this possible, yes it was running sip |
| 18:28.49 | markh1158 | sometimes it's tricky with those phones and you have try to force different images |
| 18:29.25 | markh1158 | on the TFTP server you need create a file called OS7912.TXT that contains only the load information |
| 18:29.36 | markh1158 | i'm sorry |
| 18:29.41 | markh1158 | OS79XX.TXT |
| 18:30.11 | markh1158 | then also create a SIP<mac-of-phone>.cnf |
| 18:30.19 | ^shark_ | markh1158: which load information do you mean ? |
| 18:30.37 | markh1158 | you should be able to get some of the specific file format and layouts from asteriskguru |
| 18:30.48 | markh1158 | one second |
| 18:30.53 | ^shark_ | ok |
| 18:30.56 | markh1158 | I'll get a specific format |
| 18:31.04 | ^shark_ | okay |
| 18:31.43 | markh1158 | on my tftp server my OS79XX.TXT file has just this in it: P0S3-06-0-00 |
| 18:32.07 | markh1158 | I also have two files: P0S3-06-0-00.bin and P0S3-06-0-00.sbn |
| 18:32.16 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 18:32.16 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:32.19 | markh1158 | you should be able to download some of these sip images from ciscos site |
| 18:32.30 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 18:33.16 | ^shark_ | markh1158: i only have CP7912080001SIP060412A.ZIP |
| 18:33.26 | ^shark_ | markh1158: can it be any helpful? |
| 18:33.46 | markh1158 | hmmm |
| 18:33.53 | markh1158 | not familiar with that file |
| 18:34.34 | *** join/#asterisk plaxxx (n=no@CPE00195e554efc-CM000a73a8677f.cpe.net.cable.rogers.com) |
| 18:34.47 | ^shark_ | markh1158: its the firmware from cisco |
| 18:34.59 | ^shark_ | markh1158: luckly someone managed to download it for me |
| 18:36.38 | markh1158 | the file names I have are usually the images I download from cisco are usually <image-name>.zip |
| 18:37.17 | ^shark_ | cfgfmt.linux, prserv.exe, bmp2logo.exe, cfgfmt.sun, gkdefault.cfg, prserv.linux, sip_ptag.dat, cfgfmt.exe, CP7912080001SIP060412A.sbin, gkdefault.txt,prserv.sun, and sipexample.txt |
| 18:37.25 | ^shark_ | do you recognise any..? |
| 18:38.21 | ^shark_ | i have this >> CP7912080001SIP060412A.ZIP which contains all the above files |
| 18:38.22 | markh1158 | that must be the call manager zip file |
| 18:40.15 | markh1158 | cisco has a public site with SIP images |
| 18:40.16 | markh1158 | ftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/ |
| 18:40.21 | markh1158 | oops |
| 18:40.25 | markh1158 | thats not the right phone |
| 18:42.25 | juanjoc | Has anybody experienced RTP skewed timestamps with Asterisk 1.4 recently? |
| 18:42.47 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 18:43.35 | stansmith | does anyone know how to make my cisco phone blink its LED twice whenever someone on the other line says my name? |
| 18:44.08 | ManxPower | stansmith: that feature will be added to Asterisk 1.7 |
| 18:44.14 | stansmith | kthx |
| 18:44.29 | ^shark_ | markh1158: are you looking for the right URL for me? |
| 18:44.38 | markh1158 | yeah |
| 18:44.39 | markh1158 | sorry |
| 18:44.50 | markh1158 | got bombarded with people asking questions at work |
| 18:44.52 | markh1158 | :) |
| 18:45.06 | ^shark_ | markh1158: let me keep waiting then ;) |
| 18:45.19 | ^shark_ | markh1158: as i also look around |
| 18:46.30 | markh1158 | ok |
| 18:46.38 | markh1158 | I have a 7912 here so I am looking at it |
| 18:46.55 | markh1158 | of course it has sccp but I will switch it to sip since I like that better |
| 18:47.26 | ^shark_ | markh1158: okay my phones have CP7912080001SIP060412A firmware |
| 18:47.58 | markh1158 | thats what I am looking for cause I usually deal with 7960 and 7940 |
| 18:49.55 | ^shark_ | okay |
| 18:51.59 | ^shark_ | markh1158: you can also look in here >> http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml |
| 18:52.24 | markh1158 | I have access to all the firmware |
| 18:52.54 | markh1158 | I actually just downloaded a version of the 7912 firmware |
| 18:55.02 | markh1158 | have you already followed the prcedure in the Readme.txt file for the tftp upgrade? |
| 18:55.25 | *** join/#asterisk galeras (n=Martin@201.244.193.198) |
| 18:55.41 | markh1158 | also, have you done a tcpdump on your tftp server while the phone is booting |
| 18:55.54 | markh1158 | look for anything coming into tftp from that phone |
| 18:55.58 | markh1158 | and what it is requesting |
| 18:57.37 | ^shark_ | markh1158: i wanted to force this phone to boot the firmware image b'se from my first trial it was locked in reboot |
| 18:59.05 | galeras | Dear Sirs, Any idea why a TDM2451E works fine on a generic computer but doesn't generate dialtone in an IBM X3500 server? |
| 18:59.27 | galeras | power cord is working on the server |
| 18:59.54 | ^shark_ | markh1158: i found this >> http://www.xs4all.nl/~graver1/cisco/ |
| 18:59.57 | Katty | dadaladeda |
| 19:00.06 | Katty | jbot: tell me about voice recognition |
| 19:00.16 | *** join/#asterisk josephp (n=none@shell4.sea5.speakeasy.net) |
| 19:00.24 | ^shark_ | markh1158: one of the files you asked me for.. but i am not sure its for the 7912g phone |
| 19:01.12 | Katty | o hai |
| 19:01.15 | Katty | i upgraded your pbx |
| 19:01.17 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 19:01.24 | josephp | Okay I'm trying to record inbound AND outbound calls on a queue, I figured out AgentMonitorOutoging but I can't seem to figure out where to put the commands for the incoming recording |
| 19:02.03 | josephp | because its recording on a queue rather than an individual extension |
| 19:02.28 | josephp | the outgoing I put in a different context and set it up so their queue lines on their phones use that context |
| 19:02.57 | josephp | that way if they dial a call normally its not recorded but if they hit that line for the queue then dial, it does get recorded |
| 19:03.06 | josephp | any thoughts on the inbound recording though? |
| 19:03.12 | genz | Here's an odd pickle... Zaptel 1.4.7, Asterisk 1.4.15, Digium TE210P card with just one T1 line, since we moved into our new office we have to restart zaptel about every other day. Somedays its twice, somedays not. zap shows normal, pri debug is silent, but when you dial out, it is just pure quiet. Any thoughts? |
| 19:06.26 | ManxPower | genz: what carrier is the PRI with? |
| 19:06.56 | ManxPower | So, just unplugging the PRI for 5 mins then plugging it back in does NOT fix the problem? |
| 19:07.13 | genz | ManxPower: Qwest. They claim their analytics are fine. I bought a T1-cleaner that I can have sit between me and the hub and that ddn't help. |
| 19:07.55 | genz | I haven't tried that. Obviously not a long term solution, what would that point to the problem being? (We're an outbound call center, the wolves get hungry without phones for more than a minute) |
| 19:07.57 | ManxPower | genz: a few switches out there don't like Asteirsk's restarting of the B-channels, other switches require the restarting of channels |
| 19:08.49 | ManxPower | there's an option to change the reset interval |
| 19:09.17 | ^shark_ | bbl |
| 19:09.21 | stansmith | pace |
| 19:09.29 | ManxPower | The thing is to find out if it's a Zaptel/Asterisk problem or a carrier switch issue. Unplugging the line makes the telco switch reset the line. |
| 19:11.08 | genz | Ok, so I'll leave it out for 5 minutes. If it comes back, I call the Telco and complan about the switch requiring chanel restarts - right? |
| 19:11.42 | ManxPower | no. If it comes back change the resetinterval in zapata.conf to some huge number like 99999 |
| 19:12.01 | ManxPower | then restart asterisk and zaptel (or reboot) |
| 19:12.55 | genz | Right now, I don't have resetinterval defined at all in zapata. |
| 19:13.00 | markh1158 | ^shark_: it looks like the files are different for the 7912 but it seems like they request them similar to the 7960 and 40s |
| 19:13.08 | genz | So I'm defaulting to 1 minute, I'll change it to 'never' |
| 19:13.32 | ManxPower | "never" must be new in 1.4 |
| 19:14.23 | genz | I'm readng that from voip-info's site, I'll look at the source quick to see if its valid. |
| 19:14.53 | ManxPower | look at the sample config file included in Asterisk |
| 19:14.56 | genz | Its listed in the sample zapata.conf that comes with 1.4 |
| 19:14.59 | ManxPower | the wiki is full of wrong information |
| 19:15.24 | genz | Its listed in the sample zapata.conf that comes with 1.4 |
| 19:15.32 | genz | Sorry, didn't mean to double post that. |
| 19:15.54 | genz | Apparently 'never' and -1 are valid, so says chan_zap.c |
| 19:16.23 | galeras | Any idea why a TDM2451E card works fine on a generic computer but doesn't generate dialtone in an IBM X3500 server? |
| 19:17.12 | Pagautas | hi |
| 19:17.15 | Qwell | galeras: got power connected? |
| 19:17.24 | Pagautas | anybody can help with vierling gsm gateway? |
| 19:17.30 | genz | ManxPower: want to know the results on if it does/doesn't happen again? |
| 19:18.07 | Qwell | or, more importantly, are you sure that the power that is connected actually works? |
| 19:18.07 | ManxPower | genz: not really 9-) |
| 19:18.07 | galeras | Qwuell: yep, and is working because a cd unit operate properly with same cable. |
| 19:18.07 | genz | ManxPower: Well then thanks for your help! I love this channel. |
| 19:18.20 | ManxPower | feel free to show your appreciation by sending a paypal to eric@fnords.org |
| 19:18.22 | ManxPower | ~manxpower |
| 19:18.23 | jbot | it has been said that manxpower is Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. |
| 19:18.37 | galeras | :D |
| 19:19.33 | bkruse | ManxPower: are you cisco certified? |
| 19:19.48 | galeras | I'ts sure I will show my appreciation if i can get that that card generating dialtone! |
| 19:19.59 | bkruse | I was going to do some CCN* certs back in the day |
| 19:21.25 | *** join/#asterisk mltlnx (n=mltlnx@pool-70-23-254-167.ny325.east.verizon.net) |
| 19:23.14 | ManxPower | bkruse: no. Polycom certified, however. |
| 19:23.21 | stansmith | does anyone here use rapidsvn? |
| 19:23.30 | ManxPower | I keep meaning to do some cisco certs, but other things keep getting in the way |
| 19:23.58 | bkruse | ManxPower: coolness |
| 19:24.08 | bkruse | ManxPower: Ya, they are time-consuming |
| 19:24.13 | ManxPower | Most of the cisco certs seem to require things like SNA, token ring, and other useless stuff |
| 19:24.37 | stansmith | ManxPower: just like the tests i had in school |
| 19:24.50 | ManxPower | If you are running token ring then I just don't want you to be a client |
| 19:25.35 | *** join/#asterisk Daejeo (n=chatzill@211.177.189.102) |
| 19:26.01 | stansmith | my professor used to talk about token ring and i was like "wtf? i want some ethernet!" |
| 19:26.57 | ManxPower | bkruse: my experience with Cisco is 262x, 364x, 17xx routers, 55xx switches, access lists, OSPF, T-1 and Frac T-1 interfaces |
| 19:27.02 | *** part/#asterisk mltlnx (n=mltlnx@pool-70-23-254-167.ny325.east.verizon.net) |
| 19:27.16 | ManxPower | as well as GigE fiber |
| 19:28.31 | bkruse | ManxPower: It is all about history. The CCNA is focused on networking topology (half) and then the cisco OS (other half) and being familiar with both |
| 19:29.53 | Daejeo | vtwhite.com does not allow me to use a non-us based ip address. is there any way i could use a us-based proxy for my pbx? |
| 19:30.17 | *** part/#asterisk galeras (n=Martin@201.244.193.198) |
| 19:31.11 | ManxPower | Daejeo: the solution is to use a different provider |
| 19:31.44 | Daejeo | ManxPower: But they already took my money |
| 19:31.47 | ManxPower | bkruse: I've been trying to diagnose high router load recently |
| 19:31.52 | ManxPower | Daejeo: then get a refund |
| 19:32.07 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 19:32.31 | ManxPower | There are a zillion VoIP ITSPs out there, don't give money to the bad ones. |
| 19:33.03 | Daejeo | right. i try to get my money back |
| 19:36.06 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-106-235.d4.club-internet.fr) |
| 19:41.45 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 19:44.20 | TJNII | Daejeo: You could try a proxy, but that isn't a good solution as the hop to the proxy will add lag to the conversation. |
| 19:44.24 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 19:45.16 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:45.38 | kannan | anyone?, will the jughanns [http://www.junghanns.net/en/duo_gsm_pci_produkt.html] GSM card work for USA SIM cards? |
| 19:47.11 | TJNII | GSM trunk? |
| 19:47.16 | TJNII | Looks expensive. |
| 19:47.43 | kannan | is there a less expensive way ? |
| 19:47.59 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 19:48.31 | kannan | i saw on the Bayhamsystems site that USA and Canada do not have a SMS Center for text msg from Asterisk |
| 19:48.46 | TJNII | I don't know what you need is. I guess if you need it to be mobile, than that is probably a good option. |
| 19:48.55 | kannan | it is to have a SMS application |
| 19:49.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:49.05 | TJNII | You should probably call and ask your carrier if it will work. |
| 19:49.13 | kannan | on mobiles only |
| 19:49.15 | ManxPower | kannan: you need to call them to ask if the card supports the USA/CANADA GSM bands (which are different from the rest of the world) |
| 19:49.50 | kannan | ManxPower : thx, do you know a GSM PCI card that works in the USa? |
| 19:49.59 | ManxPower | kannan: I take it you can't use the carrier provided SMS gateways and you can't use one of the GSM SMS service providers? |
| 19:50.12 | ManxPower | kannan: I strongly doubt there is ANY c |
| 19:50.21 | ManxPower | USA GSM cards |
| 19:50.32 | kannan | oh ok |
| 19:51.21 | ManxPower | and yes, any GSM SIM card should work in any GSM device, HOWEVER, different locations use different frequencies and different devices support different frequencies |
| 19:51.49 | ManxPower | you want to look for tri-band or quad band devices. |
| 19:52.04 | ManxPower | the triband may not work in your location or it may work in your location |
| 19:52.05 | kannan | is the mobiles in USA mostly GSM or CDMA? |
| 19:52.15 | TJNII | GSM, I believe |
| 19:52.20 | stansmith | GotoIf( $["NULL ${my_var}" = "NULL" |
| 19:52.22 | stansmith | oops |
| 19:52.36 | ManxPower | kannan: all mobiles in the USA are GSM or CDMA, with a few old TDMA networks that are quickly being phased out |
| 19:52.42 | stansmith | GotoIf( $["NULL ${my_var}" = "NULL"]? 2:1 ) |
| 19:52.50 | ManxPower | stansmith: That is correct |
| 19:52.51 | stansmith | ^ best way to check for null values in gotoif ? |
| 19:53.06 | stansmith | isnull was giving me headache |
| 19:53.07 | [TK]D-Fender | stansmith: wasteful. |
| 19:53.23 | ManxPower | stansmith: I believe there is a function to do that, but I just use $[X${MYVAR} = "X"] |
| 19:53.39 | [TK]D-Fender | stansmith: GotoIf($["${my_var}" = ""]?2:1) |
| 19:53.51 | [TK]D-Fender | ManxPower: Close..... |
| 19:53.53 | stansmith | [TK]D-Fender: ahh, yea for some reason i thought you couldnt do that |
| 19:53.56 | ManxPower | Ah! Yes, that should work in at least 1.4, maybe 1.2 |
| 19:54.03 | ManxPower | stansmith: you could not do that in older Asterisk releases |
| 19:54.05 | [TK]D-Fender | ManxPower: Both |
| 19:54.13 | ManxPower | [TK]D-Fender: must be a 1.0ism |
| 19:54.31 | stansmith | cuz i mean... "${my var}" = "" .. makes too much sense :-) |
| 19:54.32 | TJNII | kannan: You coule email them, ask for the model number of the Motorola GSM module they are using, and look up the datasheet on that. |
| 19:54.37 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 19:54.39 | *** join/#asterisk SteveTotaro (n=root@wsip-98-172-19-245.dc.dc.cox.net) |
| 19:55.11 | SteveTotaro | hey i am having issues with a quad bri card and a sangoma quad port FXO |
| 19:55.16 | SteveTotaro | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 19:55.21 | SteveTotaro | what does that mean |
| 19:55.45 | kannan | TNJII : right , will do that |
| 19:55.48 | ManxPower | SteveTotaro: it means the drivers are not loaded or your config file is not correct. |
| 19:56.25 | SteveTotaro | i figured that much |
| 19:56.41 | SteveTotaro | cat /proc/zaptel/* gives me a kernel panic |
| 19:57.19 | Qwell | SteveTotaro: don't do that then |
| 19:57.22 | stansmith | lol |
| 19:57.25 | SteveTotaro | lol |
| 19:57.26 | *** join/#asterisk luisetelondon (n=luis@82-69-88-130.dsl.in-addr.zen.co.uk) |
| 19:57.28 | ManxPower | SteveTotaro: perhaps you should try getting ONE of them working |
| 19:57.39 | SteveTotaro | one works fine |
| 19:57.48 | luisetelondon | <PROTECTED> |
| 19:57.49 | *** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
| 19:57.51 | Qwell | TJNII: could you do my house next? |
| 19:58.00 | luisetelondon | <PROTECTED> |
| 19:58.00 | SteveTotaro | Did you forget that FXS interfaces are configured with FXO signalling |
| 19:58.00 | SteveTotaro | and that FXO interfaces use FXS signalling? |
| 19:58.07 | Qwell | I have...one jack. |
| 19:58.13 | luisetelondon | anyone has suffered it before? Asterisk 1.4 |
| 19:58.14 | TJNII | Qwell: $20/hr + materials + travel\ |
| 19:58.17 | Qwell | 0 ethernet jacks.. sad |
| 19:58.18 | TJNII | :P |
| 19:58.23 | Qwell | TJNII: You in Huntsville? :p |
| 19:58.29 | TJNII | Iowa |
| 19:58.36 | stansmith | ha |
| 19:58.38 | TJNII | Like I said, + travel |
| 19:58.49 | St1ckm4n | Hi, I'm hoping someone who has experience setting up queues w/o agentcallback login might be able to shed some light on how to keep all the existing functionality |
| 19:58.56 | SteveTotaro | ~pb |
| 19:58.57 | 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 |
| 20:00.41 | *** join/#asterisk r0d3nt (n=astrutt@backup.realprohosting.com) |
| 20:03.14 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: no sharks in this channel |
| 20:04.08 | ManxPower | luisetelondon: what is a "cos"? |
| 20:04.22 | russellb | cosine, probably |
| 20:04.28 | SteveTotaro | http://pastebin.ca/922410 |
| 20:04.53 | luisetelondon | sorry, a "cosine", I mean...I loooong beep |
| 20:05.11 | SteveTotaro | please if you can give any suggestion on this, i would be indebted |
| 20:05.26 | ManxPower | SteveTotaro: I already gave you a suggestion |
| 20:05.39 | russellb | ManxPower: are you trying to murder me? |
| 20:05.42 | russellb | or get me drunk ... |
| 20:05.42 | SteveTotaro | and i told you i can get one or the other working |
| 20:05.48 | SteveTotaro | but not both |
| 20:06.06 | ManxPower | SteveTotaro: Hint: channels are allocated based on the order of the driver loadind. |
| 20:06.27 | ManxPower | russellb: No, I'm showing you what "probably" can do |
| 20:06.31 | SteveTotaro | yes i know that |
| 20:06.43 | russellb | ManxPower: ah :-D |
| 20:07.02 | russellb | well, asterisk probably works. :-p |
| 20:07.08 | SteveTotaro | so i load wanrouter/zaptel, then i load qozap |
| 20:07.21 | ManxPower | russellb: and no, you are too valuable to murder -- and hiding the body is just SUCH a hassle. |
| 20:07.29 | russellb | <3 |
| 20:10.36 | Qwell | russellb: nice topic :p |
| 20:10.45 | Qwell | or did somebody else add that? |
| 20:10.55 | russellb | i added it on request |
| 20:10.58 | Qwell | :D |
| 20:11.01 | Qwell | ~shark |
| 20:11.01 | jbot | There are NO sharks in #Asterisk. We want to be perfectly clear on that. |
| 20:11.02 | russellb | i joined #asterisk the other night, and there was a debate going on |
| 20:11.02 | ManxPower | the no sharks part? |
| 20:11.18 | russellb | yeah, some people were saying OMG SHARK, and then others were all like, OMG THERE ARE NO SHARKS |
| 20:11.21 | russellb | so i clarified |
| 20:11.38 | russellb | it was madness |
| 20:11.42 | ManxPower | russellb: thank you! the debate would have torn the channel to pieces. |
| 20:11.48 | russellb | i know! |
| 20:12.10 | Qwell | who's coming with me?! |
| 20:12.11 | josephp | Can someone give me the mixmonitor command(s) to add into this sequence to record calls inbound to 1017? |
| 20:12.15 | josephp | <PROTECTED> |
| 20:12.17 | josephp | <PROTECTED> |
| 20:12.19 | josephp | <PROTECTED> |
| 20:12.28 | josephp | (1017 goes to my desk, my home and my softphone) |
| 20:12.36 | SteveTotaro | no help on my issue.... |
| 20:14.51 | josephp | or we could both get ignored =) |
| 20:17.00 | [TK]D-Fender | josephp: You can't jsut make a "specific" priority 2 like that. |
| 20:17.09 | Qwell | [TK]D-Fender: yes you can |
| 20:17.12 | josephp | yes I can |
| 20:17.16 | josephp | that works fine |
| 20:17.21 | *** join/#asterisk mildk (n=mil@duke.code3.dk) |
| 20:17.22 | Qwell | [TK]D-Fender: welcome to 1.4 |
| 20:17.31 | [TK]D-Fender | josephp: not in relation to those 2 other extens... |
| 20:17.37 | Qwell | [TK]D-Fender: sure you can |
| 20:17.45 | Qwell | it checks most specific at every step |
| 20:18.02 | [TK]D-Fender | Qwell: I'm looking in relation to those 2 other extens... |
| 20:18.09 | [TK]D-Fender | Qwell: They will have nothing to do with each other |
| 20:18.12 | Qwell | so? |
| 20:18.16 | josephp | <PROTECTED> |
| 20:18.18 | josephp | there |
| 20:18.22 | josephp | thats what catches the other 2 |
| 20:18.35 | josephp | err |
| 20:18.41 | josephp | sorry thats a ,2, |
| 20:18.42 | [TK]D-Fender | josephp : you have no exten => _10XX,2 there |
| 20:18.48 | Qwell | [TK]D-Fender: you don't need to |
| 20:19.04 | [TK]D-Fender | Qwell: ..... you can't jsut skip priorities either... |
| 20:19.06 | yang | !sipnat |
| 20:19.10 | yang | -sipnat |
| 20:19.11 | Qwell | he's not skipping priorities |
| 20:19.14 | Qwell | he's got a 2 |
| 20:19.18 | josephp | <PROTECTED> |
| 20:19.18 | josephp | <PROTECTED> |
| 20:19.18 | josephp | <PROTECTED> |
| 20:19.18 | josephp | <PROTECTED> |
| 20:19.19 | josephp | there |
| 20:19.28 | Qwell | now, if something he calls ISN'T 1017, there would be problems |
| 20:19.29 | yang | !natsip |
| 20:19.46 | josephp | what I need is the ability to record anything that came into 1017 |
| 20:19.47 | [TK]D-Fender | josephp: and exten => 1017,2, will not get called by itself like that |
| 20:20.00 | Qwell | but, realistically, you could put the 2 and 3 at n, and it would also work |
| 20:20.01 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 20:20.06 | Qwell | _10XX,1,blah |
| 20:20.10 | Qwell | 1017,n,blah |
| 20:20.12 | Qwell | _10XX,n,blah |
| 20:20.20 | yang | anyone knows the link for sipnat |
| 20:20.26 | josephp | I tried that and using ,n, it never continues |
| 20:20.26 | Qwell | ~sipnat |
| 20:20.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:20.50 | Qwell | okay, so maybe n wouldn't work, thinking about it |
| 20:21.01 | josephp | anyways thats not the problem I'm having |
| 20:21.01 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 20:21.08 | josephp | what I need is to record inbound calls |
| 20:21.09 | josephp | =) |
| 20:21.24 | josephp | I've read all the various iterations on voip-info but nothing seems to be very helpful |
| 20:21.40 | josephp | probably because different things have been deprecated along the way and there is no clear "this is what works NOW" |
| 20:22.20 | josephp | I got outbound call recording to work using AgentMonitorOutgoing |
| 20:22.24 | [TK]D-Fender | josephp: "core show application monitor" |
| 20:24.00 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:24.08 | josephp | do those commands go in the area that I've been talking about? |
| 20:24.45 | josephp | i.e. |
| 20:24.51 | josephp | <PROTECTED> |
| 20:24.51 | josephp | <PROTECTED> |
| 20:24.51 | josephp | <PROTECTED> |
| 20:24.51 | josephp | <PROTECTED> |
| 20:24.51 | josephp | <PROTECTED> |
| 20:24.53 | josephp | <PROTECTED> |
| 20:25.17 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-66.static.insightbb.com) |
| 20:26.05 | [TK]D-Fender | josephp: please don't spam like that again, use a PASTEBIN |
| 20:26.06 | [TK]D-Fender | ~pb |
| 20:26.07 | 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 |
| 20:26.44 | [TK]D-Fender | josephp: And I've told you a few times you can't jsut shove that 1017 2-4 stuff in there in the middle like it can flow different conditionally. |
| 20:28.07 | josephp | okay... |
| 20:28.39 | josephp | http://pastebin.com/d565be094 |
| 20:28.47 | josephp | moved 4 down below the other 3 |
| 20:29.22 | [TK]D-Fender | josephp: the POSITION does not matter, you have no priority *1* for 1017. |
| 20:29.34 | husimon | [TK]D-Fender, hehe |
| 20:29.36 | [TK]D-Fender | josephp: Never inter-mix these things. |
| 20:29.37 | josephp | the _10XX,1, catches it |
| 20:29.41 | husimon | no it doesn't |
| 20:29.44 | [TK]D-Fender | josephp: No it doesn't |
| 20:29.53 | husimon | just put it in front |
| 20:29.56 | husimon | and put a 1 priority |
| 20:29.59 | *** join/#asterisk chigital (n=chigital@p4FE5D336.dip.t-dialin.net) |
| 20:30.09 | [hC] | Hrm. I think he's right? |
| 20:30.19 | josephp | yeah I'm right.. heh |
| 20:30.25 | josephp | what I pasted originally works fine |
| 20:30.26 | [hC] | I have some extens like _5XX,1, and 505,2, |
| 20:30.39 | etfonhomey | [TK]D-Fender, when looking at the list of features on the asterisk.org website, it lists "Graphical Call Manager". Is that talking about astman? |
| 20:30.39 | husimon | it's probably best to not do it like that for clarity |
| 20:30.41 | husimon | if that's the case |
| 20:30.50 | [hC] | Well, clarity != it works |
| 20:30.51 | [hC] | :) |
| 20:31.36 | josephp | well I have a BUNCH of lines similar to 1017 who ring multiple extensions |
| 20:31.55 | josephp | if I had to have a 1 line for each of those as well, and a 3 line for voicemail that would get excessive |
| 20:32.05 | josephp | so the pri 1 entry catches all of them |
| 20:32.22 | [TK]D-Fender | josephp: it will not continue from a differently prefixed exten... |
| 20:32.24 | josephp | then any extensiosn that dont have their own pri 2 entry get caught by the _10XX,2, |
| 20:32.45 | [TK]D-Fender | etfonhomey: no idea |
| 20:32.46 | josephp | then _10XX,3, sends anything to voicemail |
| 20:33.04 | [TK]D-Fender | josephp: Go show me a call that works this way |
| 20:33.29 | josephp | I'd rather not argue this point since it's not what is broken =) |
| 20:33.53 | [TK]D-Fender | josephp: Well I already told you the app to use to record your calls. |
| 20:34.14 | josephp | and what I'm asking is if that is where it belongs |
| 20:34.33 | josephp | every other example I've found seems to have it set up to record ALL calls |
| 20:34.41 | St1ckm4n | IMHO the new alternative to agentcallbacklogin sucks |
| 20:35.12 | josephp | St1ckm4n: agreed... I was looking at that today |
| 20:35.31 | [TK]D-Fender | josephp: You call "Monitor" before your dial statement just like we have for the past 5+ years |
| 20:36.01 | josephp | ahh thats where I had it wrong then, I was calling it after |
| 20:36.06 | *** part/#asterisk corramor (n=corramor@216.207.245.1) |
| 20:36.25 | iCEBrkr | How difficult would it be to setup a extension where you prefix the number with an 8 to enable recording? |
| 20:36.28 | iCEBrkr | CAKE |
| 20:36.29 | [TK]D-Fender | josephp: thats like interrogating your witness AFTER you execute him. |
| 20:36.30 | iCEBrkr | geesh |
| 20:36.44 | clickonce | I GOT IT! |
| 20:37.14 | husimon | [TK]D-Fender, can't he just separate out the 1017 call and monitor that by itself instead of trying to throw it in the match all pattern? |
| 20:37.24 | [TK]D-Fender | josephp: and I'd still love to see how exten => 1017,2,Dial(SIP/1017&SIP/1092&SIP/1089,20) ever gets called... |
| 20:37.27 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:37.47 | iCEBrkr | That's a bit recursive. |
| 20:37.56 | [TK]D-Fender | husimon: 1017 doesn't have a priority 1, so its not going anywhere. |
| 20:38.08 | josephp | _10XX,1, catches it |
| 20:38.22 | husimon | [TK]D-Fender, yeah I mean add a 1 priority |
| 20:38.36 | [TK]D-Fender | josephp: SHOW ME. This ougghte be good.. |
| 20:38.48 | iCEBrkr | josephp: yeah, pastebin.ca it |
| 20:38.49 | iCEBrkr | :) |
| 20:39.08 | husimon | so something calls 1017, then it rings itself? |
| 20:39.08 | lyroy | I have a problem with cdr mysql thaht I can't solve... I always get Failed to insert into database: (1062) Duplicate entry '' for key 1 ... I modify the Makefile and the cdr_addon_mysql.so in the asterisk-addons source dir and recompile... still the same problem... does someone know whats the problem? |
| 20:39.11 | josephp | what do you want to see, CLI output of a call coming in? |
| 20:39.12 | husimon | thats pretty odd |
| 20:39.18 | [TK]D-Fender | josephp: Yup |
| 20:39.23 | clickonce | I found this on Wikipedia about comfort noise: "The speech may sound choppy and even be hard to understand". This is what people talking to me experience. So, the problem is CN. When I start Asterisk it says that my ITSP has CN enabled and that it should be turned off since Asterisk doesn't fully support it. Is there any way to fix this without turning it on their side? (Since I can't) |
| 20:39.26 | josephp | thtas gonna be really difficult during the day |
| 20:39.31 | [TK]D-Fender | josephp: that'll trgger that 1017 exten match... |
| 20:39.33 | josephp | too many calls coming and going |
| 20:39.42 | iCEBrkr | lyroy: Modify the sql table |
| 20:39.57 | husimon | josephp, just grep the log for you calling the 1017 extension |
| 20:40.08 | lyroy | iCEBrkr> I have the uniqueid field already set in mysql table |
| 20:40.09 | josephp | ah HAH.. ok I got test.wav now in my monitor directory |
| 20:40.17 | josephp | moved the mixmontior before |
| 20:40.52 | josephp | what /var/log/asterisk/messages ? |
| 20:41.02 | husimon | josephp, so explain why you redial the 1017 extension after it is already being called? |
| 20:41.02 | josephp | it doesnt seem to be as verbose as the CLI output |
| 20:41.13 | husimon | that's cause you need to at verbose in your log conf |
| 20:41.17 | husimon | at=add |
| 20:41.35 | josephp | husimon: because its not been dialed yet, the actual SIP phone |
| 20:42.01 | husimon | messages => notice,warning,error,verbose |
| 20:42.17 | husimon | josephp, ah nm i was missing that part |
| 20:42.52 | husimon | so you put the mixmonitor at step 2 and it works now? |
| 20:43.04 | josephp | yep |
| 20:43.08 | josephp | one sec I'll pb it |
| 20:43.23 | [TK]D-Fender | husimon: He doesn't |
| 20:43.46 | josephp | http://pastebin.com/d37a34141 |
| 20:43.52 | josephp | thats what I have now and it works |
| 20:44.48 | husimon | I don't see why you don't just split that off you are only saving yourself one line in the config |
| 20:45.02 | [TK]D-Fender | josephp: Show us the call... |
| 20:45.19 | josephp | yeah I'm workin on the call part... |
| 20:45.44 | husimon | or are you going to put all the numbers in there and record each to a different wav |
| 20:45.46 | josephp | <PROTECTED> |
| 20:46.07 | josephp | I'm only recording one extension |
| 20:46.16 | husimon | josephp, you need to add messages => notice,warning,error,verbose to your logger.conf |
| 20:46.22 | josephp | yeah I did |
| 20:46.33 | ManxPower | then you issued a "logger reload", right? |
| 20:46.41 | josephp | I ussed a reload |
| 20:46.53 | josephp | doesnt that get everything? |
| 20:47.03 | ManxPower | it should |
| 20:47.21 | ManxPower | perhaps you are out of disk space, or the log files are owned by a user other than the user Asterisk is running as. |
| 20:47.26 | josephp | ok got it |
| 20:47.29 | josephp | no it didnt reload everything |
| 20:47.31 | josephp | now its there |
| 20:47.35 | josephp | one sec pastebinning it |
| 20:48.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:50.13 | josephp | http://pastebin.com/dea5221 |
| 20:51.05 | [TK]D-Fender | josephp: O RLY? |
| 20:51.11 | josephp | also added another mobile extension line in there as a _10XX,2, |
| 20:51.19 | [TK]D-Fender | josephp: And where do we see "-- Executing [1017@Company:1] SIPAddHeader("Zap/1-1", "Alert-Info: Peterson") in new stack" in http://pastebin.com/d565be094 ? |
| 20:51.57 | husimon | omg what did I eat last night |
| 20:52.00 | [TK]D-Fender | josephp: exten => _10XX,1,SIPAddHeader(Alert-Info: Internal) clearly isn't whats getting called per your pastebin |
| 20:52.04 | husimon | almost had to leave my office |
| 20:52.23 | josephp | bah missed that line in the earlier pb.. its exten => 1017,1,SIPAddHeader(Alert-Info: Peterson) |
| 20:52.30 | josephp | I wanted to change my ringtone +P |
| 20:52.31 | josephp | =p |
| 20:53.02 | [TK]D-Fender | josephp: Which confirms that 1017 an _10XX ' s execution have NOTHING to do with each other. |
| 20:53.15 | josephp | *sigh* one sec let me remove that and do it again |
| 20:53.27 | [TK]D-Fender | josephp: You should dientangle your extens before you confuse yourself any further. |
| 20:53.35 | [TK]D-Fender | disentangle* |
| 20:53.58 | iCEBrkr | haha |
| 20:54.16 | clickonce | Hmm, Bug 5374 seems to address this. |
| 20:54.18 | [TK]D-Fender | iCEBrkr: they never listen ..... *sigh* |
| 20:54.34 | iCEBrkr | huh? what'd you say? |
| 20:54.53 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-66.static.insightbb.com) |
| 20:55.04 | josephp | ewwww mixmonitor is continuing through even if it goes to voicemail, that kinda sucks.. |
| 20:55.13 | [TK]D-Fender | josephp: Shouldn't... |
| 20:55.42 | *** join/#asterisk chigital (n=chigital@p4FE5F7AF.dip.t-dialin.net) |
| 20:55.42 | etfonhomey | [TK]D-Fender, I noticed you weren't on here for a few days. Did you go on vacation/ |
| 20:55.44 | etfonhomey | ? |
| 20:55.54 | clickonce | For a production system, what version of Asterisk would you recommend? |
| 20:55.58 | [TK]D-Fender | etfonhomey: Yup, 5 days |
| 20:56.10 | [TK]D-Fender | clickonce: 1.4.lastest release |
| 20:56.13 | [TK]D-Fender | (full) |
| 20:56.19 | etfonhomey | [TK]D-Fender, where'd you go? |
| 20:56.21 | clickonce | Okay, that's what I got now. |
| 20:56.30 | [TK]D-Fender | etfonhomey: Toronto to visit family |
| 20:56.34 | [TK]D-Fender | (and watcha show) |
| 20:57.33 | etfonhomey | [TK]D-Fender, back to the "graphical call manager". I lost my connection before (if) you answered. Any idea whatsoever what that could mean? |
| 20:58.15 | etfonhomey | Or does anyone else? |
| 20:58.52 | [TK]D-Fender | not sure here... |
| 21:00.22 | josephp | Okay here is the latest (and more complete) context from extensions.conf http://pastebin.com/d3b4493b5 |
| 21:00.25 | josephp | and here is the latest output http://pastebin.com/d48301fbe |
| 21:01.27 | josephp | you'll see why I dont want priority 1 and 2 entries for everyone, as well as the voicemail line |
| 21:02.20 | josephp | II just wish the line 22 for voicemail could be ,n, rather than ,4, but if I do that it doesnt continue to voicemail |
| 21:02.29 | ManxPower | josephp: you seem to be under the naive impression that you can have a simple dialplan. You can't. |
| 21:02.48 | ManxPower | josephp: you have some OTHER issue. |
| 21:03.14 | ManxPower | Dialplans are complex ugly beasts -- the most you can hope for is that they don't maul you to death. |
| 21:03.45 | ManxPower | josephp: you realize that "n" is translated to load time into "the previous priority number + 1 |
| 21:03.47 | clickonce | What about comfort noise? Can it eat me? :P |
| 21:03.48 | josephp | ManxPower: I dont have ANY issue .. as you can see from the latest output, calling to the DID that forwards to 1017 executes _10XX,1 and _10XX,2 then it hits the specific 1017,3 1017,4 and 1017,5 |
| 21:03.55 | ManxPower | you can see this by doing a "show dialplan" |
| 21:04.14 | ManxPower | josephp: you don't want it to do that. |
| 21:04.22 | ManxPower | you want to to stay in the same extension pattern |
| 21:04.37 | ManxPower | oh, you used the word "forwards" incorrectly, nevermind. |
| 21:04.55 | josephp | s/forwards/calls/ |
| 21:04.58 | josephp | whatever |
| 21:05.07 | josephp | yeah that |
| 21:05.15 | ManxPower | you are looking for "calling to the DID 1017 MATCHES _10XX,1 ...." |
| 21:05.22 | sp00k3y | hi, can someone explain to me how I navigate the directory in asterisk? |
| 21:05.27 | [TK]D-Fender | Still doesn't match... |
| 21:05.32 | [TK]D-Fender | josephp: ^^ |
| 21:05.46 | [TK]D-Fender | josephp: And you keep showing broken little mismatched bits.... |
| 21:05.46 | ManxPower | sp00k3y: Um, follow the directions the directory application tells you |
| 21:05.47 | josephp | [TK]D-Fender: how doesnt it match? |
| 21:06.33 | josephp | ManxPower: so in your scenario I would have an explicit priorty 1 for 1002, 1003, 1004, 1005, 1007, 1009, 1010, 1016 etc etc etc |
| 21:06.34 | [TK]D-Fender | josephp: What in here ( http://pastebin.com/d3b4493b5 ) will cause "exten => 1010,3,Dial(SIP/1010&SIP/1099,20)" to get called? |
| 21:06.52 | ManxPower | sp00k3y: you should remember that the Directory application takes it's information from voicemail.conf and the name recordings from the user's mailbox |
| 21:06.55 | josephp | ManxPower: and an explicit priorty 4 to go to voicemail for all those extensons as well? |
| 21:07.00 | josephp | [TK]D-Fender: one sec looking |
| 21:07.12 | ManxPower | josephp: I would use a macro, but yes. |
| 21:07.22 | ManxPower | or I would use a pattern match |
| 21:07.47 | ManxPower | What I would NOT do is mix patterns and non-patterns |
| 21:08.04 | josephp | <PROTECTED> |
| 21:08.11 | josephp | [TK]D-Fender: ^^ |
| 21:08.38 | josephp | ManxPower: what about exten => _10XX,1, is NOT a pattern match? |
| 21:08.40 | SteveTotaro | Feb 28 16:08:14 ERROR[3169]: chan_zap.c:7445 mkintf: Signalling requested on channel 1 is PRI Signalling but line is in Unknown signalling 61695 signalling |
| 21:08.46 | sp00k3y | right i get that part, but when it promts you to enter the first 3 letters of the person you want to search for |
| 21:08.57 | sp00k3y | i enter them in and it always says invalid |
| 21:08.57 | ManxPower | josephp: that is a pattern match. Don't mix patterns and non patterns |
| 21:09.03 | [TK]D-Fender | josephp: Go show me this working per your PB... |
| 21:09.22 | ManxPower | sp00k3y: it does not say that. listen more carefully for the two words you are mussing |
| 21:09.23 | SteveTotaro | chan_zap.c:7445 mkintf: Signalling |
| 21:09.23 | josephp | why not? it saves me a dozen or more lines of code in this section alone |
| 21:09.27 | [TK]D-Fender | ManxPower>josephp: that is a pattern match. Don't mix patterns and non patterns <-- what I've been trying to get across for the past while now. |
| 21:09.29 | josephp | ManxPower: ^^ |
| 21:09.40 | SteveTotaro | signaling is misspelled |
| 21:09.56 | ManxPower | josephp: and it will cause you hours and hours of work down the road, also issues are virtually impossible to diagnose doing it your way. |
| 21:09.56 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 21:10.13 | ManxPower | But if you want to do more work, then more power to you, but nobody here is going to help you do it that way |
| 21:10.16 | [TK]D-Fender | josephp: You're showing broken little bits to us... that line was not in your PB, nor is there anything proper for it to land on. |
| 21:11.03 | ManxPower | [TK]D-Fender: I suggest you just give up and stop helping him. It's obvious he either does not know enough to assist in his own troubleshooting, or doesn't want to do it the way we both know is correct. |
| 21:11.09 | josephp | [TK]D-Fender: the question I came for has been anwered, quite a while ago, I didn't want or need to paste the entire extensions.conf file |
| 21:11.11 | ManxPower | Why stress yourself and stress him. |
| 21:11.12 | sp00k3y | "Welcome to the phonebook, please enter the first three letters of the entry you are looking for" |
| 21:11.26 | [TK]D-Fender | ManxPower: *sigh* |
| 21:11.40 | josephp | ManxPower: I'm not troubleshooting anything! nothing is broken! |
| 21:11.41 | josephp | lol |
| 21:11.51 | ManxPower | sp00k3y: WEIRD. Mine says "the first three letters of the person's FIRST name" or "LAST NAME" depending on how you run app_directory |
| 21:11.53 | SteveTotaro | well troubleshoot my problem then |
| 21:11.56 | SteveTotaro | it is broken |
| 21:12.04 | josephp | ManxPower: [TK]D-Fender wanted to know how/why my config worke |
| 21:12.07 | josephp | worked* |
| 21:12.29 | [TK]D-Fender | josephp: well you came in showing us mixed up stuff that would not work as shown. |
| 21:12.37 | [TK]D-Fender | josephp: And then dugg it in FURTHER |
| 21:12.42 | ManxPower | sp00k3y: you must be running a different version of Asterisk than me, as mine starts out saying "welcome to the directory" |
| 21:12.44 | sp00k3y | but am i correct in saying that the entry it wants is the name listed for each voicemailbox entry in voicemail.conf? |
| 21:12.55 | SteveTotaro | what's the point? |
| 21:13.01 | sp00k3y | im running asterisk 1.4 with freePBX on top |
| 21:13.04 | ManxPower | sp00k3y: app directory gets ALL it's info from voicemail.conf |
| 21:13.08 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 21:13.11 | ManxPower | sp00k3y: oh. |
| 21:13.11 | [TK]D-Fender | josephp: And are compounding our belief that your system is either just broken, or your understanding of its execution. |
| 21:13.13 | ManxPower | ~freepbx |
| 21:13.14 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:13.34 | SteveTotaro | manx is an asterisk god |
| 21:13.36 | ManxPower | so you are not running the same Asterisk as I am. |
| 21:13.40 | josephp | [TK]D-Fender: have your concerns been addressed? we've established that I'm simply using pattern and static in a way y'all don't feel is appropriate? |
| 21:13.46 | SteveTotaro | better listen or be turned to a pillar of salt |
| 21:13.51 | x86 | SteveTotaro: rofl |
| 21:13.57 | sp00k3y | apparently not ;P |
| 21:14.22 | ManxPower | josephp: [TK]D-Fender wants to help people. Sometimes he does not understand that some people do not want help. |
| 21:14.23 | [TK]D-Fender | josephp: not "inappropriate", but rather "non-functional". |
| 21:14.44 | x86 | i know this is a bit off-topic, but yall ever consolidate a bunch of T1's at a site on to a channelized DS3 from the CO to the company, then mux it back down to individual T1's? |
| 21:14.47 | [TK]D-Fender | josephp: Indeed this doesn't seem to be reaching a positive conclusion. |
| 21:14.52 | iCEBrkr | SO is there any type of 'official' e911 logo? |
| 21:15.01 | SteveTotaro | and it is God's (Manx) responsibility to tell people when they should give help or not |
| 21:15.04 | iCEBrkr | All I found was the triangle with 911 across it |
| 21:15.05 | ManxPower | sp00k3y: "show application directory" will show you which option to use for "search by first name or search by last name" |
| 21:15.10 | x86 | SteveTotaro: haha |
| 21:15.18 | SteveTotaro | x86 yes |
| 21:15.28 | Blackthorn | Hi, how can i see the current version of zaptel drivers that I am running? |
| 21:15.34 | SteveTotaro | adtran mx2800 m13 a great unit |
| 21:15.37 | x86 | SteveTotaro: I'm looking at mux options... what did you use? |
| 21:15.37 | ManxPower | Blackthorn: it's in the dmesg info |
| 21:15.38 | josephp | [TK]D-Fender: I'm amused because my phone system is fully functional. I will wisp you a pb with the entire extensions.conf if it would satisfy your curiosity =) |
| 21:15.41 | cesar_CR | I got this warning when loading iax ... Unable to support trunking on peer 'office' without zaptel timing |
| 21:15.43 | x86 | that's what i was looking at on ebay |
| 21:15.50 | cesar_CR | what does it mean ? |
| 21:15.51 | x86 | SteveTotaro: they're relatively inexpensive.... |
| 21:15.58 | SteveTotaro | you cant go wrong, it has build in failover for everything |
| 21:16.01 | x86 | SteveTotaro: do the patch panels on those things work decent enough? |
| 21:16.05 | sp00k3y | ok lemme try that |
| 21:16.25 | [TK]D-Fender | josephp: well you kept showing us little bits that by themselves never added up to anything functional. In the end if it works the way you want, hey, whatever right... |
| 21:16.35 | josephp | [TK]D-Fender: and for the record, I didn't originally write it, I just build off of what existed when I started here.. I do intend to rewrite the entire thing |
| 21:16.37 | ManxPower | cesar_CR: IAX2 trunking reduces bandwidth usage by putting audio from more than 1 call into one packet. It only saves you bandwidth if you have 3 or more calls going between the same two servers |
| 21:16.45 | [TK]D-Fender | cesar_CR: Means you can't do "trunk=yes" without a valid zaptel timer |
| 21:16.50 | x86 | SteveTotaro: so i bring DS3/coax into the back of that thing, and AMP60x2 it to the patch panel/break-out box? no need for anything else? |
| 21:17.06 | SteveTotaro | two coax for t3 |
| 21:17.22 | x86 | right, one for receive and one for send... I know the concept... I meant a pair ;) |
| 21:17.23 | SteveTotaro | nope that should do it |
| 21:17.23 | josephp | One other behavior that I find odd, unsure if this is standard or not, but with call queues, when I start up the asterisk service I have to log in every agent, is there a way to have them all automatically logged in at startup? |
| 21:17.38 | josephp | again, its how it was when I started working here |
| 21:17.38 | x86 | SteveTotaro: why would you need two controller cards in the thing? for redundancy only? |
| 21:17.41 | ManxPower | x86: make sure you know how the telco will hand off the line to you. Most do coax, but I suspect some do fiber |
| 21:17.45 | [TK]D-Fender | josephp: Considered using static agents? |
| 21:17.48 | x86 | SteveTotaro: or will one controller card only handle half a DS3? |
| 21:18.02 | SteveTotaro | yes, adtran makes killer stuff |
| 21:18.03 | cesar_CR | ok [TK]D-Fender do I have to enable it? is something missing ? |
| 21:18.06 | x86 | ManxPower: it's FTTP, then they convert it to coax |
| 21:18.13 | [TK]D-Fender | cesar_CR: Zaptel <- |
| 21:18.14 | SteveTotaro | who wants their t3 to go down?!? |
| 21:18.24 | x86 | SteveTotaro: I DO I DO!!!!!1!eleven |
| 21:18.29 | josephp | [TK]D-Fender: looking that up |
| 21:18.33 | [TK]D-Fender | cesar_CR: You need a zaptel set up and ready before compiling * |
| 21:18.53 | husimon | how good is the software echo cancellation in asterisk? |
| 21:19.13 | ManxPower | cesar_CR: how many calls do you expect to have going between the two asterisk servers |
| 21:19.17 | ManxPower | husimon: utterly horrid |
| 21:19.21 | husimon | eww |
| 21:19.27 | Blackthorn | so whats dsmeg? :P |
| 21:19.32 | cesar_CR | [TK]D-Fender, I followed the installation from the book... |
| 21:19.34 | SteveTotaro | just go into the phone closet and pull the fiber |
| 21:19.47 | stansmith | ~dmesg |
| 21:19.47 | jbot | it has been said that dmesg is a busybox option too |
| 21:19.47 | ManxPower | husimon: it has improved somewhat over the past couple of years, but we almost always use hardware EC |
| 21:19.49 | [TK]D-Fender | cesar_CR: Well you clearly don't have a zaptel timing source set up. |
| 21:19.52 | SteveTotaro | forget the t3, you will take out the whole building or block |
| 21:19.57 | ManxPower | Blackthorn: type dmesg in a shell prompt |
| 21:19.58 | husimon | ManxPower, yeah my hardware ec doesn't really work as well as I'd like |
| 21:20.05 | stansmith | dmesg is your friend Blackthorn |
| 21:20.15 | SteveTotaro | octastic |
| 21:20.21 | ManxPower | husimon: just what EC hardware do you have? |
| 21:20.25 | husimon | ManxPower, stupid redfone phonebridge2-ec |
| 21:20.33 | cesar_CR | ManxPower, its for testing righ now... I only want to pass one call trought the *s |
| 21:20.38 | husimon | ManxPower, it's tdmoe so it can't use the dynamic echo learning stuff as far as I know |
| 21:20.41 | ManxPower | husimon: what does it do? |
| 21:20.42 | Blackthorn | ohhh got it! |
| 21:20.53 | cesar_CR | [TK]D-Fender, ok so I will compile again... |
| 21:20.54 | husimon | ManxPower, it just seems like with conference phones they tend to cut in and out |
| 21:20.59 | husimon | ManxPower, maybe that's not ec at all |
| 21:21.01 | ManxPower | cesar_CR: If you enable trunking with only 1 call you will use MORE bandwidth |
| 21:21.16 | husimon | ManxPower, If the remote party is talking and i'm talking it cuts out |
| 21:21.30 | [TK]D-Fender | ManxPower: I never saw anything to corroborate this.... |
| 21:21.37 | stansmith | http://en.wikipedia.org/wiki/Dmesg <--- is the CSS on this page screwed up for anyone else? |
| 21:21.41 | ManxPower | husimon: Huh? EC has to happen where the call us converted to/from voip |
| 21:21.44 | [TK]D-Fender | husimon: Sounds like a duplex issue |
| 21:22.02 | ManxPower | husimon: that's pretty common with speaker phones |
| 21:22.16 | ManxPower | We use Tellabs EC cards on T-1s for EC |
| 21:22.24 | cesar_CR | ManxPower, ok got it and [TK]D-Fender told me wath was the reason of the warning thanks... |
| 21:22.25 | ManxPower | Tellabs is what the telcos use for EC |
| 21:22.31 | Duplicator | Don't tell me the redfone stuff doesn't work! :) |
| 21:22.40 | [TK]D-Fender | Duplicator: Redfone = bleh |
| 21:22.42 | Blackthorn | found the info that i was needing, just wanted to see if my zaptel install i just did took. and it did. |
| 21:23.01 | Duplicator | [TK]D-Fender: we've got one in the box - should I send it back before we use it? |
| 21:23.41 | JenniferAkemi | stansmith: loads fine for me |
| 21:23.49 | madduck | why would asterisk tell my client that it will send the ring back tone and then not send anything? |
| 21:24.08 | Blackthorn | I am using five sipura 2002 boxes, and when setup and installed the ring sounds like a normal phone but after a few weeks every one of them changes ring to like a single then three short rings. Idea's what to look at? |
| 21:24.43 | Duplicator | No redfone = real kink in my new colo plans |
| 21:24.44 | husimon | ManxPower, i see, it's not always that it cuts out |
| 21:24.45 | ManxPower | Blackthorn: I suggest you look at the SIPuras |
| 21:24.47 | husimon | just sometimes |
| 21:24.50 | JenniferAkemi | tellabs is what we use |
| 21:25.04 | SteveTotaro | 3com kicks asterisk's ass |
| 21:25.22 | Duplicator | ooohhh, 3com. My fav. |
| 21:25.28 | SteveTotaro | pbxnsip is awesome too |
| 21:25.58 | JenniferAkemi | tellabs are really cheap on ebay now |
| 21:26.14 | husimon | Duplicator, it works fine. I guess my speakerphone issue is not ec |
| 21:26.15 | ManxPower | they are VERY cheap on ebay, it's just a matter of finding the right stuff |
| 21:26.46 | ManxPower | husimon: speakerphone would pretty much invalidate any answer you give us. DOES it happen WITHOUT using the speakerphone? |
| 21:26.50 | Duplicator | husimon: I haven't seen anyone actually using Redfones yet |
| 21:27.02 | husimon | Duplicator, I have it working in a HA setup right now |
| 21:27.16 | Duplicator | husimon: That's what I want. With 2 PRI's |
| 21:27.30 | husimon | ManxPower, it mostly does. If the remote extension has a huge amount of delay i sometimes catch bits of my own voice coming back. |
| 21:27.44 | Blackthorn | manx: yes I agree too. i feel it's a sipura thing. Though I also had thoughts it could be a voice mail issue? like a ring that says wiating to pickup? or a distintive ring issue maybe? |
| 21:27.55 | husimon | ManxPower, but it's like if I raise my voice a lot, then talk the next word after the loud word will be heard for a brief moment |
| 21:28.05 | husimon | ManxPower, probably it trying to adjust dynamically |
| 21:28.25 | iCEBrkr | Blackthorn: If I recall correctly, the sipuras have a WMI ring by default, which is annoying. |
| 21:28.26 | ManxPower | Blackthorn: even then it would be a SIPUra config option. |
| 21:28.42 | husimon | Duplicator, You need to add a special version of zaptel, but it works after that |
| 21:28.49 | Blackthorn | manix: ice: thanks i'll go read up on it |
| 21:29.08 | iCEBrkr | Don't see how that'd make the phones ring differently. |
| 21:29.21 | Duplicator | husimon, are you on 1.14? |
| 21:29.30 | husimon | i'm using a 1.4 based system |
| 21:29.35 | husimon | it's actually ABE 1.4 |
| 21:29.36 | ManxPower | iCEBrkr: the sipras can give a special ring if there is voiceamil |
| 21:29.52 | iCEBrkr | hrrm |
| 21:29.52 | Duplicator | husimon, er, 1.4.16 |
| 21:29.56 | ManxPower | husimon: since you are on ABE there is very little we can do to help |
| 21:30.09 | iCEBrkr | I've only had mine do a short ring when there's new VM. |
| 21:30.10 | Blackthorn | i'm not using any voice mail on my * box its not even configured actually. But I can take a new unit out of the box, set it up, dial it sounds fine |
| 21:30.18 | iCEBrkr | and the blippy dialtone when you pick up |
| 21:30.29 | Blackthorn | 2-3 weeks later it gives a goofy ring and people think it's broken :p |
| 21:30.44 | husimon | ManxPower, yeah I was just curious how the software ec does, if it sucks i'll just stick with my hw ec |
| 21:30.49 | Blackthorn | it's a normal long ring then three short rings |
| 21:30.50 | St1ckm4n | has anyone setup a call center using the dynamic agent option instead of agentcallbacklogin() ? |
| 21:31.56 | husimon | Duplicator, btw I do get some HDLC warnings with the redfone but it doesn't seem to be causing any problems |
| 21:32.09 | ManxPower | husimon: I have NO idea if zaptel EC works on TDMoE links. I know it does not support non-zaptel ports |
| 21:32.09 | husimon | but I have pretty low call volume, only 55 total users. |
| 21:32.13 | ManxPower | (i.e. voip) |
| 21:32.28 | husimon | ManxPower, yeah I don't think the learning part can work |
| 21:32.46 | husimon | ManxPower, I haven't tested it yet though so no idea. |
| 21:32.57 | *** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 21:33.01 | ManxPower | husimon: So what idiot bought the redfone anyway? |
| 21:33.38 | BrianR___ | ManxPower: There's no echo on a pure tdm audio path, so I'd suspect no. |
| 21:33.53 | ManxPower | BrianR___: I would suspect that too. |
| 21:33.54 | husimon | ManxPower, someone wanted an HA setup that didn't require you to swap the cable |
| 21:33.57 | husimon | we only have one pri |
| 21:34.12 | Duplicator | ManxPower: This idiot bought one, but is now wondering if he should return it for some PCI PRI cards... |
| 21:34.16 | husimon | it does work nicely in that sense |
| 21:34.18 | ManxPower | husimon: Ah. It sucks to be you. |
| 21:34.43 | *** join/#asterisk jmls (n=jmls@mail.tessera.co.uk) |
| 21:34.47 | jmls | evening all |
| 21:34.54 | ManxPower | somehow I suspect you have MANY single points of failure |
| 21:34.55 | husimon | ManxPower, Do you know of other ways to do a HA setup without manually switching the pri if you only have one pri? |
| 21:34.58 | BrianR___ | Oh.. Speaking of the tellabs stuff - I bought some cards and a chassis to frontend my old digium T1 cards, but wound up using the sangoma card with the built in canceller instead... |
| 21:35.01 | husimon | i guess some bigger tdmoe |
| 21:35.06 | husimon | like a mediant |
| 21:35.12 | ManxPower | husimon: I don't know of a way to do ANY real HA setup with Asterisk. |
| 21:35.16 | jmls | does anyone know why sipp is saying that all my placed calls failed (even though I know they didnt) ? |
| 21:35.27 | ManxPower | BrianR___: want to sell them? |
| 21:35.29 | BrianR___ | husimon: There's a few purpose-built boxes for that purpose. |
| 21:35.35 | jmls | sipp -sn uac -s 444678 -mp 23787 -l 25 -d 300000 -t un 127.0.0.1 -m 1000 -rp 2s -r 5 |
| 21:35.49 | husimon | ManxPower, I'm using heartbeat, redfone, and drbd shared disk |
| 21:35.52 | BrianR___ | ManxPower: company will prolly put them on ebay eventually... |
| 21:35.54 | *** join/#asterisk ruied (n=ruied@bl10-127-164.dsl.telepac.pt) |
| 21:36.01 | husimon | works great for a 2 node setup |
| 21:36.08 | bkruse | jmls: sipp is seriously awesome. |
| 21:36.16 | bkruse | redfone = $$$ |
| 21:36.20 | ManxPower | husimon: with redundant switches, redundant PRIs with diverse physical routes into the building? |
| 21:36.34 | ManxPower | redundant redfone boxes? |
| 21:36.37 | jmls | yeah it's great. Just wish I knew why it thinks it's failing |
| 21:36.39 | BrianR___ | I just got a second PRI for our asterisk setup here. One lands on one box, and one lands on the other. |
| 21:36.43 | ManxPower | That still won't keep calls up if one box fails. |
| 21:36.49 | BrianR___ | Will get dropped calls if one box falls over though. |
| 21:36.57 | x86 | SteveTotaro: http://cgi.ebay.com/Adtran-MX2800-Router-Patch-Panel-1200290L1-1200291L1_W0QQitemZ300197283571QQihZ020QQcategoryZ3706QQrdZ1QQssPageNameZWD1VQQ_trksidZp1638.m118.l1247QQcmdZViewItem |
| 21:36.59 | bkruse | ManxPower: You then have to have redundant redundancy |
| 21:37.04 | husimon | ManxPower, yeah i get the point but if that much shit fails then oh well |
| 21:37.21 | ManxPower | husimon: at that point moving a cable seems like a bargain |
| 21:37.29 | x86 | SteveTotaro: that look decent / have a decent price? |
| 21:37.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:37.35 | bkruse | because state is not shared across all of them, sip phoens will still lose registration even if you do a quickfailover (via roundrobin dns or similar) |
| 21:37.41 | husimon | ManxPower, It's more the ability to take down a box remotely and upgrade one of the pair |
| 21:37.46 | husimon | then switch it back and it works fine |
| 21:37.53 | bkruse | It does help recovery time to minimal, but does not take 100% failing recovery out of the picture |
| 21:37.56 | BrianR___ | You do need to play some games with your SIP registry or have phones register with both PBX's... |
| 21:38.07 | bkruse | husimon: That is an expensive upgrade |
| 21:38.14 | BrianR___ | Or crank the SIP re-register time way down. |
| 21:38.20 | husimon | bkruse, I don't lose sip registration |
| 21:38.24 | bkruse | BrianR___: More traffic |
| 21:38.32 | *** join/#asterisk jacco (n=root@unaffiliated/jacco) |
| 21:38.36 | bkruse | husimon: You do not lose registration when a box goes down, and your other one takes over? |
| 21:38.38 | ManxPower | husimon: even when the redfone box dies? |
| 21:38.55 | jacco | Hey, I have two modems connected but one seems not to be working. |
| 21:38.57 | bkruse | How does the asterisk box (there) know that the particular phone is already registered to it? (are the registry events shared also?) |
| 21:39.03 | bkruse | oh, redfone :] |
| 21:39.05 | jacco | <PROTECTED> |
| 21:39.06 | jacco | *.. |
| 21:39.07 | BrianR___ | But there's lots of cheap ways to ensure a single machine failure doesn't result in "no phones" in a typical office setup. It's not like they should be falling over often anyway. |
| 21:39.15 | bkruse | jacco: ztcfg -vv output? |
| 21:39.16 | husimon | bkruse, I mean obviously calls in progress go down |
| 21:39.18 | ManxPower | jacco: We don't support modems |
| 21:39.20 | bkruse | husimon: right |
| 21:39.25 | jacco | ManxPower: who does? |
| 21:39.35 | ManxPower | We don't. |
| 21:39.42 | ManxPower | or at least most of us here in this channel |
| 21:39.44 | x86 | SteveTotaro: http://cgi.ebay.com/ADTRAN-MX2800-with-Patch-Panel-and-Cables-AC-POWER_W0QQitemZ290210446223QQihZ019QQcategoryZ51271QQssPageNameZWDVWQQrdZ1QQcmdZViewItem <-- $100 cheaper |
| 21:39.49 | jacco | ManxPower: don't most PBXs use modems? |
| 21:39.50 | bkruse | husimon: I meant registration. How does box #2 (if asterisk) know that phones a,b,c,d,e,f,g already registered to "asterisk" and expects calls from their DID or similar to go to them? |
| 21:39.51 | BrianR___ | bkruse: Doesn't really matter for enterprise applications. Maybe matters if you're a TSP, but not for the rest of us. |
| 21:40.23 | husimon | bkruse, Im trying to remember how that worked out. |
| 21:40.28 | ManxPower | jacco: not in the way I am assuming you want to use them as I have to pretty much just make up your setup out of thin air. Can you guess why? |
| 21:40.33 | bkruse | BrianR___: I completely agree. But not one can fully boast 100% HA until you have that. |
| 21:40.46 | jacco | bkruse: it shows 2 channels. |
| 21:40.49 | Kobaz | i've heard of one company that can keep a call alive if a box goes down |
| 21:40.50 | etfonhomey | Anyone here use the Asterisk Appliance in a production situation? |
| 21:40.52 | bkruse | BrianR___: Especially with low re-registry time, it is probably invisible to the user, agreed |
| 21:40.57 | jacco | ManxPower: yeah yeah my box is an old P2 with two modems serving to connect to the main phone lines. |
| 21:41.00 | Kobaz | i forgot the name, but they claim they can do it |
| 21:41.05 | jacco | It redirects them to like 6 VoIP phones |
| 21:41.16 | BrianR___ | bkruse: Even the big iron PBX's still drop calls if the T1 linecard fails... Gotta pick your battles. |
| 21:41.21 | husimon | you better have a really good reason to need a call to stay active for how much they will cost |
| 21:41.22 | ManxPower | jacco: A modem does data. A modem does not do voice. |
| 21:41.27 | ManxPower | perhaps you are thinking of something else. |
| 21:41.32 | jacco | ManxPower: it is a special modem that does voice. |
| 21:41.43 | ManxPower | jacco: ASTERISK DOES NOT SUPPORT VOICEMODEMS |
| 21:41.56 | jacco | ManxPower: so how is my setup running exactly? |
| 21:41.57 | jacco | thin air? |
| 21:42.03 | BrianR___ | Uh.. There's a few winmodems that _are_ supported as fxo/fxs linecards... |
| 21:42.09 | fujin | that's a dumb idea |
| 21:42.15 | husimon | ManxPower, can't you use an old modem working on that really low end digium driver? |
| 21:42.20 | ManxPower | Well, you may or may not have a Digium card. I asssume you are using something like a modem, connected to a serial port. |
| 21:42.29 | ManxPower | husimon: no. |
| 21:42.31 | BrianR___ | also, quite a few isdn modems (well... BRI linecards) are also supported. |
| 21:42.34 | jacco | ManxPower: it is installed in the computer on a PCI slot. |
| 21:42.48 | husimon | ManxPower, I'm pretty sure you can use that driver with a normal modem and it can work as an fxo |
| 21:42.53 | husimon | certain chipsets |
| 21:42.55 | husimon | but it sucks |
| 21:42.59 | ManxPower | BrianR___: actually ONLY FXO and ONLY specfic chipset. |
| 21:43.01 | Kobaz | teh sux |
| 21:43.01 | jacco | The signal is pretty clear. |
| 21:43.03 | bkruse | BrianR___: Completely agree, mine is more of a complaint about "that is not REAL FULL high availibility" |
| 21:43.06 | jacco | And whatever! The system works. |
| 21:43.18 | ManxPower | Perhaps you mean "X100P clone" |
| 21:43.21 | ManxPower | that would be accurate. |
| 21:43.42 | ManxPower | As far as asterisk and zaptel are concerned it's an X100P card, not a modem. |
| 21:43.43 | BrianR___ | Like the X100P is essentially a winmodem... |
| 21:43.46 | bkruse | ManxPower: Right |
| 21:43.56 | jacco | ManxPower: what do most PBXs use then to hook to the phone lines? |
| 21:44.03 | bkruse | jacco: An analog card |
| 21:44.06 | ManxPower | jacco: they use special cards |
| 21:44.12 | bkruse | a real pbx device |
| 21:44.20 | BrianR___ | bbiab |
| 21:44.25 | jacco | This is a really hacked pbx device, ManxPower. |
| 21:44.30 | bkruse | jacco: http://store.digium.com/products.php?category_id=1 |
| 21:44.38 | ManxPower | jacco: Best of luck. |
| 21:44.38 | jacco | bkruse: no web browser, sorry. |
| 21:44.44 | bkruse | jacco: wow. links? |
| 21:44.55 | bkruse | That would be useless, just saying, you could have one :] |
| 21:44.55 | jacco | bkruse: mayyybe. The package manager is dead though. |
| 21:45.03 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:45.05 | *** join/#asterisk _kb1_kanobe (n=jsmith@bdr2.fieldrd.scrd.ca) |
| 21:45.10 | jacco | bkruse: is there any way to find out what device is installed? |
| 21:45.12 | bkruse | oh :/ that is a list of analog cards that work with asterisk, supported and backed by digium |
| 21:45.23 | bkruse | jacco: lspci and lspci -vv maybe a startup driver message in dmesg |
| 21:45.53 | jacco | bkruse: I just get "Individual Computers - Jens Schoenfield Intel 537" |
| 21:46.23 | jacco | *Schoenfeld |
| 21:47.18 | ManxPower | jacco: what specific problem are you having? |
| 21:47.19 | SteveTotaro | can anyone help with bri |
| 21:47.25 | SteveTotaro | i am STUCK |
| 21:47.38 | bkruse | SteveTotaro: b410p? |
| 21:47.46 | jacco | ManxPower: one of these "modems" works but the other one doesn't. |
| 21:47.49 | jacco | Thus, we have only one line. |
| 21:47.50 | bkruse | ManxPower: I believe the problem is that he does not have a card |
| 21:48.02 | jacco | No no no, I can use the pbx... people make calls to it every day. |
| 21:48.14 | bkruse | jacco: oh, interesting |
| 21:48.17 | ManxPower | jacco: if lspci does not list both cards, then the MOTHERBOARD does not see the other card, nothing WE can do about that |
| 21:48.20 | SteveTotaro | jung(whatever) |
| 21:48.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:48.41 | bkruse | jacco: ls /dev/zap |
| 21:48.47 | x86 | OH NOES TEH MOFFERBOARDZ! |
| 21:48.52 | ManxPower | bkruse: he has two cards, he believes are X100P clones. I suspect one of the cards is not actually X100P compatable |
| 21:48.54 | bkruse | x86: lol |
| 21:49.00 | SteveTotaro | <PROTECTED> |
| 21:49.02 | bkruse | ManxPower: I bet your right |
| 21:49.03 | x86 | ;) |
| 21:49.03 | jacco | ManxPower: lspci lists both cards |
| 21:49.10 | bkruse | SteveTotaro: b410p ? what bri card? |
| 21:49.15 | bkruse | jacco: ls /dev/zap ! |
| 21:49.22 | ManxPower | jacco: does it list them the same? |
| 21:49.29 | jacco | bkruse: lots of numbers |
| 21:49.31 | jacco | ManxPower: yes. |
| 21:49.32 | SteveTotaro | signalling = bri_cpe_ptmp |
| 21:49.39 | bkruse | jacco: lots? You should have 2. |
| 21:49.51 | bkruse | and ctl, channgel, psuedo, timer |
| 21:49.54 | jacco | bkruse: there's like 1 through 247 |
| 21:49.57 | jacco | and then those 4 |
| 21:50.05 | ManxPower | bkruse: perhaps you mean /proc/zap ? |
| 21:50.11 | ManxPower | or /proc/zaptel |
| 21:50.16 | jacco | <PROTECTED> |
| 21:50.26 | bkruse | ManxPower: Well, yes, but /dev/zap should have the actual channel numbers loaded from the card |
| 21:50.35 | SteveTotaro | it is a four port Junghanns |
| 21:50.47 | bkruse | like, for my tdm400p is it 1 2 3 4 channel ctl pseudo timer |
| 21:50.56 | jacco | bkruse: there are 1 through 249 and channel ctl pseudo timer |
| 21:51.41 | ManxPower | bkruse: I'll bet he's not running udev or whatever the replacement for static device inodes are |
| 21:51.48 | jacco | I'm running hotplug |
| 21:51.52 | jacco | So ManxPower is right. |
| 21:51.56 | *** part/#asterisk ManxPower (n=manxpowe@207.191.118.2) |
| 21:52.05 | *** part/#asterisk jmls (n=jmls@mail.tessera.co.uk) |
| 21:53.08 | jacco | bkruse: one of the modems (1 of 2) has two phone ports on it. |
| 21:53.10 | bkruse | Geez, what flavor? are you running and how old? |
| 21:53.15 | bkruse | gotcha |
| 21:53.17 | jacco | bkruse: it's slackware from '05. |
| 21:53.31 | jacco | I didn't build this thing... my dad just called me in to fix it. :| |
| 21:54.21 | bkruse | Weak. |
| 21:54.23 | jacco | bkruse: I tried plugging both phone cords into that one modem but it went directly to a dead line when I tried to call. |
| 21:54.31 | bkruse | jacco: I am really not sure then, so wierd. |
| 21:54.54 | jacco | Yeah.. I was thinking about reinstalling but I really don't know enough about Asterix to rebuild the setup |
| 21:56.46 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:57.06 | jacco | bkruse: you think I should try instead to get that one modem with two ports to work? |
| 21:57.17 | madduck | why would asterisk send 183 and 180 in response to an INVITE, but then not send a ring back tone? |
| 22:01.12 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 22:01.46 | stansmith | is it possible to capture "#" in a variable using Read() ? |
| 22:01.56 | stansmith | as in, check for "#" vs just "" |
| 22:02.19 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 22:02.33 | grandpapadot | Given the choice between GSM and iLBC, which codec would yeild the best quality for the bandwidth? |
| 22:03.53 | stansmith | jacco was in IRC as root user? |
| 22:04.17 | *** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 22:06.47 | J4k3 | the name 'root' doesn't really mean much |
| 22:06.55 | grandpapadot | Unless you got a big one ... |
| 22:06.56 | J4k3 | I mean, I could go rename 'root' to whatever, and make a regular user named 'root' |
| 22:07.11 | J4k3 | a few badly coded apps might vomit |
| 22:07.12 | J4k3 | but thats it |
| 22:08.01 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 22:08.04 | grandpapadot | Given the choice between GSM and iLBC, which codec would yeild the best quality for the bandwidth? |
| 22:08.09 | nvrpunk | hello |
| 22:08.39 | tzanger | wtf |
| 22:08.54 | tzanger | mulaw is inverted beofre transmission, but alaw has only even bits inverted?? |
| 22:12.35 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:14.13 | Katty | jbot: hudlite? |
| 22:14.18 | Katty | jbot: HUD? |
| 22:14.26 | Katty | jbot: you're just not fun! |
| 22:15.21 | Katty | how does one get hudlite without trixbox |
| 22:17.29 | [TK]D-Fender | Katty, http://www.hudlite.org/ |
| 22:18.22 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 22:18.24 | Katty | [TK]D-Fender: it just redirects me to trixbox.org |
| 22:19.30 | Katty | hrmm |
| 22:19.32 | Katty | nevermind |
| 22:19.38 | Katty | i think i found the page that 404ed me earlier |
| 22:19.42 | Qwell | eww, hud |
| 22:19.48 | Katty | Qwell: i'm curious about it. |
| 22:19.51 | Katty | Qwell: i think i shall play with it. |
| 22:19.57 | Katty | tho, really, i'm not sure what to do with an RPM |
| 22:20.01 | Qwell | Katty: for some stupid reason, it requires an ircd |
| 22:20.02 | Katty | is it like a tar? |
| 22:21.34 | fujin | hudlite so stupid |
| 22:21.44 | fujin | it made me want to cry when I realised it uses an ircd for some sort of bogus IPC |
| 22:22.32 | Katty | what is it with people who just love going $object is stupid all the time. |
| 22:22.40 | nvrpunk | does anyone know any really good commercial billing systems for asterisk? |
| 22:22.40 | Katty | whatever happened to adventure and testing with things |
| 22:22.41 | fujin | uh |
| 22:22.44 | fujin | because it's stupid |
| 22:22.47 | nvrpunk | outside of a2billing, astbill |
| 22:22.47 | fujin | testing stupid things === stupid |
| 22:22.50 | Katty | fujin: no, you're attitude is :P |
| 22:22.54 | fujin | no, seriously |
| 22:22.57 | fujin | why would you test somethign so stupid? |
| 22:22.59 | Katty | let me put it this way |
| 22:23.02 | fujin | better off beating your head into the wall |
| 22:23.03 | Katty | i dont' care wha tyour opinion is |
| 22:23.07 | fujin | actually, do that, and do me a favour |
| 22:23.08 | Katty | i'm going to get my own opinion |
| 22:23.08 | fujin | and the world |
| 22:23.21 | Katty | and you're not really helping me develop my opinion by just spewing negativity everywhere |
| 22:24.07 | fujin | ok buddy |
| 22:24.18 | Katty | thanks. |
| 22:24.19 | nvrpunk | would you both stop quarelling? |
| 22:24.23 | nvrpunk | please |
| 22:24.52 | grandpapadot | ilbc rocks |
| 22:25.06 | nvrpunk | jbuffer=yes in sip.conf and jitterbuffer=yes in iax.conf |
| 22:25.11 | nvrpunk | are those interchangable? |
| 22:25.21 | nvrpunk | ie jbuffer or jitterbuffer |
| 22:28.03 | SteveTotaro | tzafrir i need you |
| 22:30.28 | lyroy | Does anybody know if there a site where I can find well build extensions.conf examples? |
| 22:35.03 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 22:35.04 | grandpapadot | Ok, ilbc just solved 99% of my problems with call quality, echo, and DTMF. Is there something wrong with it? Nobdy ever talkes about it. |
| 22:35.11 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 22:35.18 | obnauticus | Would anyone here know where this error is comming from |
| 22:35.19 | obnauticus | [Feb 28 17:31:52] WARNING[15731]: app_dial.c:1163 dial_exec_full: Skipping dialing interface 'SIP/memali/00903333333333' again since it has already been dialed |
| 22:35.53 | JT | obnauticus: it sounds like you are trying to dial the same extension twice in the one dial line |
| 22:36.29 | obnauticus | heh |
| 22:36.37 | obnauticus | it's being executed by an agi script |
| 22:37.39 | obnauticus | JT: http://pastebin.ca/922650 |
| 22:38.01 | genz | My TE210P keeps needing a restart. 3 times today I've had to restart zaptel. We just lose the ability to place outgoing/receive incoming calls. 1.4 branches of Zaptel and Asterisk (built from release in December 07, can't remember exact number). Any ideas on how I can proceed about fixing it? |
| 22:38.43 | genz | I've modified resetinterval, it was at the default (60 seconds) and it crashes even faster when its to never. |
| 22:39.53 | *** join/#asterisk anthm][ (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:40.57 | obnauticus | JT what does this mean: |
| 22:40.58 | obnauticus | everything is fine except Got SIP response 482 "Loop Detected" back from 67.205.74.168 |
| 22:41.23 | *** join/#asterisk IOscanner (n=IOscanne@76.187.195.124) |
| 22:43.46 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 22:49.07 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@adsl-99-162-117-1.dsl.austtx.sbcglobal.net) |
| 22:53.55 | eric2 | anyone have routing issues when you're with CLEC A and you try to call a number that is with CLEC B? |
| 22:54.07 | CoffeeIV_ | I have an asterisk/iaxmodem/hylafax setup. About a week ago I quit receiving faxes -- looks like incoming faxes get as far as dialing the iaxmodems, then asterisk goes to "congestion" and nothing ever appears in the iaxmodem / hylafax logs. Iax show peers shows the iaxmodems as registered. Any suggestions for how to debug this ? |
| 22:55.04 | eric2 | CoffeeIV_ - I've been trying to get faxing going for the last month, who are you using for your lines? |
| 22:55.29 | CoffeeIV_ | it's a T1/PRI at texlink in Austin, TX |
| 22:55.52 | CoffeeIV_ | the only problems I have ever had are my own configuration issues |
| 22:57.01 | eric2 | what codec or faxing technology are you using? Was faxing reliable? |
| 22:57.26 | CoffeeIV_ | I have also done faxing over some SIP providers, with various degrees of success, never really good though -- I used older versions of asteirsk and the RxFax/TxFax way for those |
| 22:57.31 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 22:57.49 | eric2 | hmm, have you looked at t.38? |
| 22:58.10 | CoffeeIV_ | just read docs on it, haven't tried to set t.38 up |
| 22:58.12 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:58.23 | MatBoy | what are the best billing solutions for asterisk ? asterisk2billing is one opensource, I also found another one, have to check the name |
| 22:59.05 | CoffeeIV_ | This iaxmodem / asterisk problem is a killer though -- I can't figure out how to get more debuging info out, and I don't think I changed anything |
| 22:59.20 | eric2 | I was going to make my own billing product, but I'll check out asterisk2billing |
| 22:59.36 | CoffeeIV_ | I can send faxes out through hylafax -> iaxmodem -> asterisk, so I don't think it is a registration problem |
| 22:59.41 | eric2 | so all you have installed is iaxmodem and hylafax? |
| 22:59.52 | eric2 | I toyed with those 2 but never succeeded |
| 23:00.29 | CoffeeIV_ | right now I have the T1/PRI/zaptel -> asterisk -> iaxmodem -> hylafax set up, and it worked great until this current problem |
| 23:01.22 | eric2 | zaptel is using some hardware card right? |
| 23:01.26 | CoffeeIV_ | yes |
| 23:01.29 | scooby2 | what would cause the first few seconds of Playback(prompts/disclaimer) to be cut off? putting a Wait before it does not change anything. |
| 23:01.33 | MatBoy | eric2, problem is resellers in a2bllling... but there seems to be something going on that part |
| 23:02.18 | eric2 | I don't think I'll use that billing product, gonna roll my own |
| 23:02.27 | eric2 | looks like it has great features though |
| 23:02.31 | clyrrad | Any of you guys know of a softphone that can be deployed and provisioned remotely? |
| 23:02.35 | eric2 | but I'm anti php |
| 23:02.53 | MatBoy | eric2, <eric2> I was going to make my own billing product, but I'll check out asterisk2billing |
| 23:03.01 | MatBoy | I don;t follow you |
| 23:03.29 | eric2 | that product is made with the php programming language |
| 23:03.31 | eric2 | I hate php |
| 23:03.50 | eric2 | well, am just against it so I won't be installing it on my machine |
| 23:04.03 | CoffeeIV_ | Is there a way to increase the amount of log infomation that comes from iaxmodem ? |
| 23:04.16 | MatBoy | eric2, I think you have a difficult life than :) |
| 23:04.26 | eric2 | not sure CoffeeIV_ - never used it |
| 23:04.44 | eric2 | MatBoy - perhaps, I prefer j2ee stuff |
| 23:06.21 | sp00k3y | does anyone know of any test prep courses or something like that for the dCAP? |
| 23:06.25 | sp00k3y | other than the bootcamp |
| 23:08.10 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:08.10 | *** join/#asterisk anthm][ (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:08.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:08.17 | MatBoy | eric2, that is nice too, as long as it's no perl ;) |
| 23:09.38 | *** join/#asterisk redax (i=redax@r6.hu) |
| 23:09.41 | redax | good evening |
| 23:09.56 | husimon | eric2 why do you hate php? |
| 23:10.14 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-3-183.dsl.pltn13.pacbell.net) |
| 23:10.53 | redax | can you help me? I'm looking for a ISDN interface for asterisk, I need 16BRI ports ;-) |
| 23:10.55 | husimon | oh no wonder, ou're a j2ee person |
| 23:11.26 | redax | where should I start? 2x Xorcom Astribank with 8BRI ? |
| 23:12.03 | redax | anyway, what's your opinion about Xorcom HW? |
| 23:13.33 | husimon | friggin users complaining about no documentation, oh i'll get them good. making 4'x4' posters and going to paste them on the walls |
| 23:13.35 | husimon | EAT IT |
| 23:16.17 | *** part/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 23:16.41 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 23:17.09 | phix | Why does coffee smell better than it tastes? |
| 23:17.20 | fujin | it's awesome like that |
| 23:17.20 | *** part/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 23:17.22 | J4k3 | I dunno, but that always pisses me off |
| 23:20.36 | CoffeeIV_ | is there any reason OTHER than the type=friend/peer/user thing that an IAX2 extension would only work in one direction, making outgoing calls but not receiving incoming ? |
| 23:22.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:23.08 | *** join/#asterisk d00gster (n=doughant@bas1-toronto44-1177789678.dsl.bell.ca) |
| 23:27.10 | pkunkra | U.S. is 120 volt power, correct. Something is wrong with my math. servers are "supposed" to be using 7 amps of power but are only used 3.5. |
| 23:27.18 | pkunkra | used -> using |
| 23:28.56 | J4k3 | usa is anywhere from 110 to 130v |
| 23:29.00 | J4k3 | 120v typical |
| 23:29.10 | J4k3 | but, whats written on the PSU has nothing to do with what their actual *required* output is |
| 23:29.33 | J4k3 | most psus are: entirely underrated on the output side, and list their input current needs based on inrush current |
| 23:29.53 | J4k3 | most systems: don't need more than about 150W *absolute maximum* sustained, once the drives are spun up |
| 23:29.56 | JT | pkunkra: what do you mean "supposed" to be using 7A? |
| 23:30.01 | J4k3 | err |
| 23:30.04 | J4k3 | entirely overrated |
| 23:30.18 | J4k3 | 120*7 = a lot of damned watts |
| 23:30.19 | JT | i think pkunkra must be a little insane |
| 23:30.24 | pkunkra | the vendor's specs have them listed as 7amps. |
| 23:30.25 | JT | his power bill would be huge |
| 23:30.26 | pkunkra | let me show you |
| 23:30.27 | J4k3 | yeah |
| 23:30.30 | JT | pkunkra: MAX POWER DRAW |
| 23:30.31 | JT | jesus |
| 23:30.38 | JT | as if it will draw that unless it needs to |
| 23:30.40 | J4k3 | max, at full output, at lowest efficiency |
| 23:30.51 | J4k3 | 3.5A is rather high, in reality |
| 23:30.54 | J4k3 | they must be heavy servers |
| 23:30.58 | JT | that would be stupid if it normally pulled that much power |
| 23:31.03 | J4k3 | multiple processors, multiple fast drives, etc. |
| 23:31.05 | pkunkra | http://h18004.www1.hp.com/products/quickspecs/12509_na/12509_na.html#Power%20Specifications |
| 23:31.10 | JT | pkunkra: please read up on switchmode power supplies |
| 23:31.14 | JT | pkunkra: ... |
| 23:31.21 | J4k3 | dude |
| 23:31.25 | J4k3 | if that thing actually drew 7A sustained |
| 23:31.30 | J4k3 | it'd glow a dull red |
| 23:31.31 | J4k3 | due to the heat |
| 23:31.36 | pkunkra | i know very little about power |
| 23:31.38 | JT | i can't believe people think their PSUs are meant to draw that much |
| 23:31.38 | ruied | I've heard that there is some kind of patent related to mobile PBX, is that true? |
| 23:32.17 | JT | pkunkra: it will only draw the load it has on the DC side + inefficiency overhead |
| 23:32.39 | J4k3 | these psus appear to be 77-81% efficient |
| 23:32.44 | J4k3 | fairly good psus |
| 23:33.06 | JT | alright ;) |
| 23:33.26 | pkunkra | i don't really have an accurate way of measuring power consumption. have to rely on the datacenter. i have about 8 servers per 20 amp circuit. it averages about 12amps. I keep them at about 60% consumption as much as possible. |
| 23:33.46 | pkunkra | so. each server should be using about 1.5 amps then. |
| 23:34.04 | pkunkra | but i'm looking at these specs and scratching my head |
| 23:34.08 | JT | pkunkra: buy a clamp meter |
| 23:35.26 | pkunkra | the ones that measure via electro/magnetic/whatever around the cord. they give an accurate reading even though they don't touch the metal? |
| 23:35.42 | JT | what do you mean by keep them at 60%, you try to pump up power usage? |
| 23:35.44 | JT | yes |
| 23:35.52 | JT | electricians everywhere use them |
| 23:35.58 | JT | some can measure thousands of amps |
| 23:36.20 | pkunkra | jt, i keep the circuit loaded at about 60% of the 20amp max it was rated for |
| 23:36.32 | JT | pkunkra: intentionally? |
| 23:36.41 | pkunkra | so i shoot to keep it at 12amps usuage. |
| 23:36.44 | pkunkra | usage* |
| 23:37.38 | JT | why |
| 23:37.45 | pkunkra | yes. i don't have a reliable figure as to the max draw they could do. i had one circuit trip at what appeared to be 16amps |
| 23:38.00 | pkunkra | what do you recommend i keep them at? |
| 23:38.33 | JT | as low as you can :) |
| 23:38.40 | pkunkra | i figure about 0.5 amps buffer per server is a good number |
| 23:38.55 | pkunkra | there are 8 servers per circuit |
| 23:39.39 | JT | 0.5A @ 120V seems ambitious unless they're lower power usage units |
| 23:39.48 | JT | that's only around 60W |
| 23:42.17 | pkunkra | it appears to work well. haven't tripped any breaks with those numbers |
| 23:42.36 | pkunkra | mind you the circuit is kept at 12A out of 20A |
| 23:42.47 | pkunkra | so that "should" be 8A of buffer |
| 23:42.48 | pkunkra | total |
| 23:43.17 | pkunkra | unless i reboot all eight servers at the same time, i doubt it would get up to 20A |
| 23:43.41 | mvanbaak | aaaaaaaaaaaah |
| 23:43.42 | pkunkra | JT, so when you say "as low as you can", what do you consider to be low? |
| 23:43.54 | CoffeeIV_ | Is there a really simple way to use Dial() to dial several iax extensions at once, going to the first that picks up ? |
| 23:44.00 | JT | pkunkra: that depends on the server specs i guess |
| 23:44.04 | JT | 150W is low for a server |
| 23:44.06 | mvanbaak | the joys of being at a hosting company that allows you to use 2x32A for free |
| 23:44.17 | pkunkra | ?!! |
| 23:44.17 | JT | mvanbaak: @ what voltage? |
| 23:44.27 | pkunkra | mvanbaak, where do you host? :-) |
| 23:44.29 | mvanbaak | JT: 230V |
| 23:44.36 | JT | that's a lot of power |
| 23:44.39 | mvanbaak | pkunkra: http://www.bit.nl |
| 23:44.42 | pkunkra | ah, that would explain it. |
| 23:44.55 | JT | do they have the cooling to backup 64A of power draw? |
| 23:44.58 | pkunkra | ah. Netherlands... wrong side of the pond for me. |
| 23:44.59 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:45.18 | Nugget | ik ben een vliegende koe! |
| 23:45.28 | mvanbaak | JT: they allow a max of 8x128A on 320V |
| 23:45.39 | JT | 320v? |
| 23:45.41 | mvanbaak | on a 52u rack case |
| 23:45.51 | mvanbaak | yeah |
| 23:45.59 | JT | i thought you said 230V |
| 23:46.03 | *** join/#asterisk nauCe (n=nauce@ip24-255-116-169.dc.dc.cox.net) |
| 23:46.08 | JT | do they have the cooling capacity? |
| 23:46.19 | JT | if all their customers started pulling that power? |
| 23:46.25 | mvanbaak | JT: default is 2x32A on 230 |
| 23:46.42 | mvanbaak | but they do provide power for blade servers that need high voltage |
| 23:46.44 | JT | for every amp drawn by computer hardware, it needs an amp of redundant cooling |
| 23:46.49 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 23:47.45 | mvanbaak | JT: their new housing (where our servers are) is build for 8X128A@320V |
| 23:47.57 | JT | that's crazy |
| 23:48.01 | JT | what is redundancy like? |
| 23:48.02 | mvanbaak | JT: uhhuh |
| 23:48.29 | mvanbaak | JT: they have 3 UPS systems. each build to handle the full load |
| 23:49.02 | mvanbaak | and they have 2 power feeds from the power company. one on the south entry one on the north entry |
| 23:49.27 | JT | mvanbaak: battery or rotary ups? |
| 23:49.30 | mvanbaak | and they have 30 days of diesel to handle the load |
| 23:49.31 | JT | what about generators? |
| 23:49.37 | JT | 30 days is not bad :) |
| 23:49.49 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 23:50.10 | mvanbaak | they test their UPS and generator setup every week |
| 23:50.21 | mvanbaak | we run on diesel power every friday |
| 23:50.23 | JT | are they battery or rotary upses? |
| 23:50.29 | mvanbaak | battery |
| 23:50.51 | mvanbaak | they have a room that is 5x8 meters full of batteries |
| 23:51.08 | mvanbaak | you really dont want to be there |
| 23:51.08 | JT | figures ;) |
| 23:51.23 | mvanbaak | I've been there once |
| 23:51.30 | JT | rotary UPSes are a much nicer solution, especially for bigger datacentres |
| 23:51.59 | mvanbaak | the signs on the door tell you they dont want you to be in there for more than 2 minutes without special clothing/equipment |
| 23:52.01 | pkunkra | why don't you want to be in a room full of that many batteries. |
| 23:52.04 | pkunkra | ? |
| 23:52.09 | JT | pkunkra: acid gas |
| 23:52.16 | JT | fumes from charging |
| 23:52.34 | mvanbaak | acid gas, current, magnatic fields etc |
| 23:52.46 | mvanbaak | and the hum is,... LOAD |
| 23:52.50 | JT | mvanbaak: how many racks fit in the datacentre? |
| 23:52.50 | mvanbaak | LOUD |
| 23:53.13 | mvanbaak | JT: 1250 |
| 23:54.44 | *** join/#asterisk stubert (n=stu@techtools.actusa.net) |
| 23:54.49 | mvanbaak | JT: and in 3 months it was 80% full |
| 23:54.50 | pkunkra | JT, yuck. yeah. i wouldn't want to be in there either |
| 23:55.16 | mvanbaak | we are short in colocation in .nl |
| 23:55.18 | JT | mvanbaak: how much does it cost for a rack? |
| 23:55.27 | JT | mvanbaak: everywhere's short in co-location i think |
| 23:55.46 | mvanbaak | JT: we pay 2500 euro/month for a full 52u rack with 100mbit dark fiber |
| 23:55.57 | stubert | Is it possible to use a softlink for a VM box to have 2 extensions share the same vm using different passwords? |
| 23:56.01 | pkunkra | mvanbaak. i hate you |
| 23:56.04 | pkunkra | that's very good. |
| 23:56.29 | mvanbaak | it is |
| 23:56.38 | JT | mvanbaak: dark fibre? |
| 23:56.40 | JT | to where |
| 23:56.47 | *** join/#asterisk RoyK (n=roy@ip-183-50-149-91.dialup.ice.no) |
| 23:56.56 | mvanbaak | JT: AMS-IX |
| 23:57.29 | stubert | mvanbaak: was that "it is" for me or JT? |
| 23:58.31 | JT | mvanbaak: so it's not quite dark |
| 23:58.41 | mvanbaak | 00:56 < pkunkra> that's very good. |
| 23:58.41 | mvanbaak | 00:56 < mvanbaak> it is |
| 23:58.53 | JT | mvanbaak: do you have to pay for traffic usage on top? |
| 23:59.16 | mvanbaak | JT: the first 100 mbit/s are for free |
| 23:59.31 | JT | free traffic to anywhere? |
| 23:59.35 | mvanbaak | yup |
| 23:59.47 | JT | pity i don't speech dutch i guess :) |