| 00:00.07 | jr_ewing | yes |
| 00:00.24 | fiber0pti | What is auto fallthrough |
| 00:00.25 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 00:01.11 | jr_ewing | try to forward tc^port 5060 and udp range (find it in \etc\asterisk\rtp.conf) but you will still have probleme with sip if you have Nat at home |
| 00:01.44 | The_X | I already forward 5060 on the linksys to my 7960 |
| 00:01.52 | The_X | and a bunch of other ports |
| 00:02.57 | The_X | the rtp ports are forwarded too |
| 00:03.03 | The_X | this sucks :) |
| 00:03.07 | Qwell | The_X: udp? |
| 00:03.07 | jr_ewing | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:03.44 | jr_ewing | try first with a dmz on your linksys |
| 00:03.45 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 00:04.29 | The_X | yeah |
| 00:05.58 | Tecky` | I'm presuming w/ one analog phone line from verizon i want a FXS card ?!? |
| 00:06.01 | Tecky` | is there one i should look at |
| 00:06.18 | Qwell | phone line = fxo, phone = fxs |
| 00:06.20 | Tecky` | one in peticular ?!? |
| 00:06.20 | Qwell | so, no |
| 00:07.18 | Tecky` | so optimally one w/ a fxo and one w/ fxs (too loop back to a switch and go to the outlets in my home ?!? |
| 00:07.27 | Qwell | sure |
| 00:08.15 | Tecky` | whats a decent price for one of those cards? seeing 300+ on ebay |
| 00:09.36 | Qwell | Tecky`: http://store.digium.com/products.php?category_id=17 |
| 00:10.01 | The_X | Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk |
| 00:10.04 | The_X | that's my setup |
| 00:10.09 | The_X | I did put nat=yes and qualify=yes |
| 00:10.14 | The_X | fwding ports |
| 00:10.20 | The_X | I guess there's nothing we can do about it :( |
| 00:11.00 | *** join/#asterisk imran (n=codentes@68.206.53.81) |
| 00:12.10 | Qwell | Tecky`: Something like this would do the trick. http://www.voipsupply.com/product_info.php?manufacturers_id=13&products_id=295&osCsid=14fc24533eecc93b268f033a89e090ab |
| 00:12.41 | jr_ewing | The_X the only thing i can tell you is to use a public stun , it works in your case( asterisk public, phone behiond nat) but i don"t know if you can set Stun Server into 7960 |
| 00:13.30 | The_X | I'll try :) |
| 00:13.32 | The_X | thanks again |
| 00:15.25 | Tecky` | hmm |
| 00:15.37 | *** join/#asterisk SERGEUS|W (n=SERGEUS@ippe-245.ippe.ru) |
| 00:16.00 | The_X | can't do stun on 7960 |
| 00:16.00 | The_X | dammit |
| 00:16.25 | Tecky` | does that card support iax2 ? |
| 00:16.36 | Qwell | Tecky`: no, it isn't a voip card |
| 00:16.50 | Qwell | and it doesn't need to support it |
| 00:17.39 | *** join/#asterisk muzzz_ (n=chatzill@218.111.66.117) |
| 00:17.43 | razu | does anyone have experience with Teles iswitch hardware ? |
| 00:17.55 | jr_ewing | Tecky before lose money with asterisk in hardware |
| 00:18.25 | jr_ewing | try to get familiar with asterisk with free voip provider like voipbuster |
| 00:18.39 | Tecky` | i've used voip before / vonage etc... |
| 00:19.24 | *** join/#asterisk muzzz_ (n=chatzill@218.111.66.117) |
| 00:20.49 | nroej | can anyone assist me testing my fresh sip setup? |
| 00:21.11 | nroej | want to see if my srv and naptr records are ok |
| 00:21.19 | nroej | cant test it myself right now |
| 00:21.55 | *** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 00:26.45 | *** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com) |
| 00:27.01 | Seedy | How it going people... |
| 00:27.11 | nroej | great |
| 00:27.46 | nroej | world domination is near |
| 00:27.54 | nroej | ;-) |
| 00:30.40 | *** join/#asterisk oceanlan|dstn|di (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 00:32.13 | Seedy | Ha |
| 00:32.37 | Seedy | I'm very new to all this voip stuff... So i've got some questions |
| 00:32.57 | Seedy | Is it possible to use asterisk with my vonage softphone number? |
| 00:33.50 | nurfe | goonight legion |
| 00:34.19 | Qwell | Seedy: It's against the TOS, and people have been slammed for it |
| 00:34.29 | Seedy | Hmmm |
| 00:35.51 | Seedy | I'm trying to set up a very simple asterisk set-up to accept incoming calls. What would a good (Cheap) provider for that be? |
| 00:35.55 | [av]bani | because asterisk is only used by EVIL HAX0RZ |
| 00:36.05 | Qwell | Seedy: how many minutes per month? |
| 00:36.44 | Seedy | for now, its just for fun. So under 200 for sure |
| 00:37.15 | Qwell | then $40/month is far too much to pay |
| 00:37.20 | [av]bani | i've had good luck so far with junction networks |
| 00:37.22 | Qwell | just get a per minute provider, like asterlink |
| 00:37.29 | Qwell | or nufone, or whatever |
| 00:38.18 | Krill | why is it against the TOS? |
| 00:38.24 | Seedy | Thanks guys... I'll look into those |
| 00:38.48 | [av]bani | Krill: you can only use vonage authorized endpoints with vonage service, for whatever reasons they decide. |
| 00:38.52 | Qwell | Krill: because the softphone account is for a softphone |
| 00:39.08 | Krill | ahh fair enough |
| 00:39.16 | [av]bani | completely arbitrary decision like any thing large corporations make :) |
| 00:39.20 | Krill | i dont think i can get vonage here anyway |
| 00:39.31 | [av]bani | they're scared you'll actually do something useful with their service |
| 00:39.52 | Krill | haha |
| 00:41.31 | [av]bani | teliax is ok |
| 00:47.51 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 00:50.49 | Seedy | [av]bani: So, junction networks will allow me to receive incoming calls via asterisks? |
| 00:52.39 | *** join/#asterisk Hott-The-Moople (n=icechat5@24-52-139-116.agstme.adelphia.net) |
| 00:53.23 | *** join/#asterisk brian__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 00:55.49 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 00:58.56 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:00.39 | Seedy | I have to choice between Session protocol (SIP vs. IAX). I don't plan on hooking up any hard phones to astrisks so is one better than the other |
| 01:01.16 | X-Rob | Seedy, if you're connecting two asterisk boxes together, use IAX |
| 01:01.57 | *** join/#asterisk Donkcyde (i=jk@packetsexchange.net) |
| 01:02.01 | Seedy | X-rob: I'm connecting outside lines (Regular phone connections) to an asterisk box. So SIP would be better for that? |
| 01:02.12 | *** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 01:02.35 | X-Rob | you need some way of converting those lines into VoIP. |
| 01:02.46 | X-Rob | If you're plugging them directly into the asterisk box with a TDM card, then you don't care. |
| 01:08.22 | razu | can i send more information (than src and dst callerid) to a E1 port ? |
| 01:09.07 | X-Rob | ...like what? Weather? CPU Temperature? |
| 01:09.15 | razu | well |
| 01:09.15 | X-Rob | (the non-arse answer is 'no') |
| 01:09.22 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:09.49 | razu | i'd need to send smthing like the cdr info to other system |
| 01:10.26 | kink0 | razu: http://www.freesoft.org/CIE/Topics/126.htm |
| 01:10.58 | kink0 | yes , you can send more infor to an E1 port. Are you speaking about q.931 ? |
| 01:11.16 | razu | i think so |
| 01:11.24 | razu | i'm using Teles iSwitch |
| 01:11.32 | razu | and their idb |
| 01:11.45 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 01:11.52 | kink0 | razu: ccs, hdb3 ? |
| 01:11.58 | razu | but its logging only src and dst callerids ... |
| 01:12.13 | razu | but i need more info about each call going thrue it |
| 01:12.43 | kink0 | well, that may be the firmware, but q931 requires more info while dialoging with your E1 |
| 01:12.53 | kink0 | i.e. the channel id, and so. |
| 01:13.16 | razu | ok |
| 01:13.32 | razu | so teoretically ... i can send lots of info to teles |
| 01:13.44 | razu | thats good :) |
| 01:14.52 | kink0 | yes, but other aspect is that teles logs it in cdr's |
| 01:15.37 | [av]bani | Seedy: yes, JN will route calls to you |
| 01:15.37 | razu | yea |
| 01:15.44 | razu | kink0 : thx :) |
| 01:16.24 | *** join/#asterisk Skumling (n=skumling@fw.sg12.dk) |
| 01:26.21 | *** join/#asterisk xtr-II (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 01:38.34 | *** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com) |
| 01:39.13 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-253.msy.bellsouth.net) |
| 01:40.50 | Seedy | in extionsions.conf what is the difference between exten => 12129130352 and exten => _12129130352 |
| 01:41.27 | razu | if you are using X |
| 01:41.32 | razu | then you need _ |
| 01:41.39 | kink0 | Seedy, _ means starting with 1 |
| 01:42.08 | *** join/#asterisk pr0m (n=t849779@24-75-196-70.chvlva.adelphia.net) |
| 01:42.24 | Seedy | kink0: How is that used? Or what is it used for? |
| 01:42.27 | *** join/#asterisk nahirean (n=Amorith@c-68-36-161-8.hsd1.nj.comcast.net) |
| 01:42.55 | kink0 | Seedy, to setup prefixes |
| 01:43.21 | _DAW-LAPTOP | Seedy _ means pattern match |
| 01:43.35 | nahirean | Hey folks, I am working with a Sipura 2000 and I've got it registered to Asterisk and I can make outgoing calls with no issue.. I've forgotten how to ring the adapter from extensions.conf when you receive an incoming call (lost my old configs due to hard disk crash).. does anyone have a resource or a few tips on how to point the incoming call to the sip device? |
| 01:44.00 | *** join/#asterisk _8ball (n=spunk@206.163.81.30) |
| 01:44.41 | nroej | arrr chan_capi sucks |
| 01:44.56 | nroej | or i am too stupid to set it up |
| 01:46.39 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 01:47.32 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 01:47.44 | austinnichols101 | nroej: isn't that accums razor :) |
| 01:47.52 | *** join/#asterisk _8ball (n=spunk@206.163.81.30) |
| 01:49.43 | nroej | wtf ist accums razor? |
| 01:49.48 | *** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net) |
| 01:52.09 | austinnichols101 | all things being equal, simplest solution usually is the correct one |
| 01:52.18 | austinnichols101 | paraphrasing |
| 01:53.50 | nroej | okay |
| 01:54.23 | jebba | nahirean, Dial(SIP/sipura) |
| 01:54.35 | jebba | e.g. if you have a [sipura] section in sip.conf |
| 01:56.12 | nahirean | Jebba, thanks for your reply, I figured it out.. i was being a douche. In my incoming context, I didnt have NXXNXXXXXX, I was pointing it to the name of the Sipura context.. ie: exten => <sip context> .. sigh.. |
| 01:56.22 | nroej | [101981.104191] kcapi: card 1 "fcpci-e000-10" ready. |
| 01:56.47 | nroej | looks good but wont work |
| 01:57.00 | nahirean | now if I can remember how to record messages I'll be in business ;) |
| 02:00.04 | *** part/#asterisk kink0 (n=k@62.37.205.161) |
| 02:00.32 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 02:00.40 | *** join/#asterisk WillSip (i=WillSip@200.119.223.246) |
| 02:00.47 | WillSip | alguien conoce un numero de subscripcion en Red Hat |
| 02:01.08 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
| 02:01.11 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 02:02.04 | nahirean | have a good one folks |
| 02:04.16 | WillSip | somebody subscription number for red hat 3 enterprise |
| 02:18.40 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:19.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 02:25.11 | *** join/#asterisk AbbieHoff (n=ahoffman@conr-adsl-209-169-118-15.consolidated.net) |
| 02:28.38 | AbbieHoff | commen |
| 02:28.39 | AbbieHoff | t |
| 02:30.53 | *** part/#asterisk AbbieHoff (n=ahoffman@conr-adsl-209-169-118-15.consolidated.net) |
| 02:33.48 | Trazz | anyone here familiar with the x100p card? |
| 02:34.14 | Trazz | also can you bring vonage in to * ? |
| 02:38.41 | *** join/#asterisk J_- (n=raz@55-pool1.ras14.floca.alerondial.net) |
| 02:39.32 | Qwell | Trazz: their crap, and no, it's against the TOS...they will shut you off |
| 02:39.37 | Qwell | they're crap* |
| 02:39.57 | Trazz | heheeh ok |
| 02:39.59 | J_- | h |
| 02:40.01 | J_- | hy all lol |
| 02:40.58 | J_- | hya Qwell |
| 02:41.35 | pauldy | you can do it but they only give you 500 minutes per month and if you go voer it is like 5 cents a minute |
| 02:41.42 | pauldy | total crapola |
| 02:43.09 | Trazz | whats the difference if you use the ata or * why is not unlimited? |
| 02:43.54 | pauldy | the difference is vontage sucks balls |
| 02:43.58 | *** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 02:44.00 | tzafrir_laptop | Trazz, in some places x100ps are horrible. In some places they may work resonably |
| 02:44.18 | tzafrir_laptop | Not recommended for professional setup |
| 02:44.36 | tzafrir_laptop | But can be handy for testing |
| 02:44.48 | Trazz | what should i get for production then? |
| 02:44.49 | J_- | 5 cents a min? thats a rip |
| 02:44.57 | Trazz | i only need 1 or 2 lines max as pots |
| 02:44.57 | pauldy | there is the potential of people abusing the service with asterisk and setting up pass through for their friends and family etc... with a closed box it is much mroe secure |
| 02:45.44 | tzafrir_laptop | Trazz, either an FXO ATA or a Digium card (TDM400P with one FXO module) |
| 02:46.00 | pauldy | traz depends on how many phones you are hooking up and what the REN requirements fo the phones are |
| 02:46.32 | Trazz | we are using cisco 7940 or 7960 phones or eyebeam softphone |
| 02:46.51 | Trazz | and then use broadvoice for some lines and need to use a line from pots side |
| 02:47.00 | pauldy | ahh |
| 03:03.39 | Aughey | Trazz: I'm have a X100P to experiment with, and it's crap. The echo, sound quality, and volume is horrible. |
| 03:04.19 | Trazz | :( |
| 03:05.40 | *** join/#asterisk jef_ (i=fischer@p548445A3.dip.t-dialin.net) |
| 03:05.54 | argentas_ | okay, i've got a bit of an issue with asterisks CDRs, i've got a service which answers an incoming call, play some prompts and generally interacts with the caller for a while, then makes an outbound call. I'd like the duration of the outbound call (from answer to hangup) to be logged in addition to the duration of the inbound call - is there any easy way to do this? |
| 03:08.41 | Trazz | Aughey, check my pm |
| 03:18.49 | *** join/#asterisk FastJack_ (i=fastjack@p5091EE7D.dip.t-dialin.net) |
| 03:19.22 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 03:23.23 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 03:28.16 | tzafrir_laptop | Aughey, where are you from? |
| 03:36.37 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 03:42.38 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 03:43.09 | SkramX | anyone done SMS messaging? |
| 03:49.29 | *** join/#asterisk bmg505 (n=leon@dsl-146-58-116.telkomadsl.co.za) |
| 03:50.57 | *** join/#asterisk dalfry (n=vaibhav@santa.vsharma.net) |
| 03:59.48 | *** join/#asterisk Cool_One (n=bclinton@adsl-69-155-12-51.dsl.ltrkar.swbell.net) |
| 04:01.06 | Cool_One | who is the asterisk super man in here |
| 04:01.32 | BlueDevi1 | super man is sleeping now |
| 04:01.46 | X-Rob | I have a 'W' on my chest. for 'Weenie' |
| 04:03.46 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 04:03.56 | *** join/#asterisk tronix (n=dsf@mappy.catbert.org) |
| 04:04.09 | tronix | is there a way to reboot a cisco 7960g phone by pressing the buttons? |
| 04:05.19 | Qwell | tronix: sip or sccp? |
| 04:05.29 | tronix | right now, sccp. i'm trying to set up sip |
| 04:05.39 | Qwell | hit settings, then **#** |
| 04:05.41 | tronix | (working on getting the upgrade) |
| 04:05.46 | tronix | ahh! thanks! |
| 04:05.58 | Qwell | then once you get sip, *+6+settings |
| 04:06.07 | Qwell | (like ctrl-alt-del) |
| 04:06.08 | tronix | ahh no wonder that didn't work. :) |
| 04:06.19 | tronix | (didn't realize that was sip firmware specific...figures.) |
| 04:06.31 | Cool_One | can ayone tell me what this means |
| 04:06.34 | Cool_One | <PROTECTED> |
| 04:06.34 | Cool_One | asterisk: relocation error: /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_cust_config_register |
| 04:06.34 | Cool_One | [root@pbx root]# Ouch ... error while writing audio data: : Broken pipe |
| 04:08.53 | Cool_One | must of been a stumper |
| 04:12.27 | *** join/#asterisk coppice (n=chatzill@69.204.17.210.dyn.pacific.net.hk) |
| 04:12.47 | tronix | yep. |
| 04:12.50 | tronix | big stumper. |
| 04:13.09 | tronix | I think it means you've got asterisk 1.2 |
| 04:13.18 | tronix | but odbc module compiled against asterisk 1.0 |
| 04:14.14 | Cool_One | jmm |
| 04:14.17 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 04:14.22 | tronix | cause that symbol exists in 1.0 but not in 1.2 |
| 04:14.32 | Cool_One | so it would be a perl problem? |
| 04:14.39 | tronix | module probably needs recompile |
| 04:14.44 | tronix | to match asterisk 1.2 setup |
| 04:14.53 | tronix | for res_config_odbc.so |
| 04:14.57 | X-Rob | are you using odbc? |
| 04:15.00 | Cool_One | yes |
| 04:15.06 | X-Rob | recompile the module then. |
| 04:15.15 | Cool_One | I have a fresh install of asterisk 1.2 |
| 04:15.22 | J_- | odbc just sucks |
| 04:15.22 | Cool_One | newest active perl |
| 04:15.29 | J_- | threading issues |
| 04:15.36 | Cool_One | and newest mysql |
| 04:15.58 | J_- | works fine |
| 04:16.11 | J_- | and its easy to cluster |
| 04:16.22 | Cool_One | I was trying to load a package called astgui |
| 04:16.36 | Cool_One | system was working till then |
| 04:17.02 | Cool_One | this is my 3rd week on asterisk |
| 04:17.21 | Cool_One | also 3rd week on linux platform |
| 04:17.51 | Cool_One | I kinda thought this would be a fun project |
| 04:18.12 | X-Rob | it is. |
| 04:18.13 | Cool_One | I have figured out real quick that I am way over my head |
| 04:18.16 | X-Rob | but odbc is _not_ fun. |
| 04:18.20 | Cool_One | haha |
| 04:18.38 | Cool_One | ODBC in windows servers... i can hadle |
| 04:18.50 | Cool_One | but this linux is messing with my head |
| 04:18.57 | X-Rob | do why are you using odbc with asterisk? |
| 04:19.13 | X-Rob | that's not something I'd expect a 3-week-user to be doing |
| 04:19.34 | Cool_One | required to install this gui manager for asterisk |
| 04:19.41 | J_- | its something to be expected from someone coming from a windows background |
| 04:19.52 | Cool_One | I had my pbx up and going in about 8 hours of console time |
| 04:20.22 | X-Rob | Hurm, doen't seem like _asterisk_ needs odbc for that. |
| 04:20.23 | nroej | can someone help me with chan capi on ubuntu it isnt working.. just inbound not outbound |
| 04:20.26 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 04:20.39 | nroej | always get: == Everyone is busy/congested at this time |
| 04:20.41 | Cool_One | I have a good background in windows server and novell servers |
| 04:20.54 | X-Rob | mv /usr/lib/asterisk/modules/res_config_odbc.so /tmp/res_config_odbc.so |
| 04:20.54 | *** join/#asterisk opsys (n=opsys@68-235-141-52.miamfl.adelphia.net) |
| 04:21.15 | J_- | yea I can tell because in windows odbc is the main choice for db conectivity |
| 04:21.47 | Cool_One | :) |
| 04:21.53 | Cool_One | what would you recomend for a gui interface with asterisk |
| 04:21.59 | J_- | lpic-1/lpic-2/+cisco/mcse/mcsa/mcdba too |
| 04:22.09 | X-Rob | Cool_One, AMP |
| 04:22.24 | J_- | I love windows its allways getting broken and active directory sucks, but it pays well too :) |
| 04:22.27 | X-Rob | http://www.coalescentsystems.ca/ |
| 04:22.35 | X-Rob | active directory is good. |
| 04:22.49 | X-Rob | It's people who fuck with it that don't know what they're doing that causes trouble. |
| 04:22.54 | opsys | Does anyone know how to get the channel you are bridged to? |
| 04:22.56 | X-Rob | it's just ldap and kerberos, nothing new. |
| 04:23.03 | J_- | yea, I fix there stuff all the time |
| 04:23.18 | X-Rob | nroej, look further up the logs in /var/log/asterisk/full |
| 04:23.25 | X-Rob | opsys, 'sip show channels'?? |
| 04:23.54 | nroej | X-Rob: okay |
| 04:24.48 | opsys | I need to get the bridged chan from the dialplan, I tried useing the ast_bridged_channel function but I get invalid pointers |
| 04:25.35 | nroej | X-Rob: ahh forgot msn=14 in the config |
| 04:26.03 | X-Rob | opsys, hurm. call an AGI that does a sip show channels and figures it out? There's no easy way, you'd probably want to explain exactly what you're doing and why you want to do it on -users |
| 04:26.47 | nroej | hmm hmm |
| 04:26.57 | opsys | is there a pastebi avail that I can show you my diff to ChanisAvail?? I figured that would be the best place. |
| 04:27.03 | Darwin35 | damn blender does not work on ia64 yet |
| 04:27.23 | Darwin35 | wrong window |
| 04:27.50 | X-Rob | I'm not a coder |
| 04:28.53 | opsys | X-Rob: I can do the lookup via an AGI but I didn't want to take the perfomrance hit. Time to go back into the C books! |
| 04:32.06 | J_- | opsys: thats the correct way to do it in c |
| 04:35.02 | *** join/#asterisk opsys (n=opsys@68-235-141-52.miamfl.adelphia.net) |
| 04:36.15 | opsys | <PROTECTED> |
| 04:39.26 | *** join/#asterisk ckram (n=mcravey@cpe-70-112-233-182.austin.res.rr.com) |
| 04:40.44 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 04:41.16 | tronix | Qwell: rock on -- I've got the 7960G up to SIP 7.4 firmware. thanks!!! (tip was a big help) |
| 04:41.31 | tronix | I've stayed away from 7.5 due to reports of */7960 interaction issues |
| 04:41.39 | tronix | but man, this looks pretty good. |
| 04:42.05 | opsys | tronix: hows the sound quality on the 7.4 load? |
| 04:42.22 | tronix | opsys: ahh, well, that's a good question. :-) (I'm deaf) |
| 04:42.29 | tronix | I'm setting up * mostly to deal with my TDD and fax |
| 04:42.31 | [av]bani | o_O |
| 04:42.33 | tronix | hahaha |
| 04:42.38 | [av]bani | lol |
| 04:42.41 | tronix | I know. I find it pretty funny too :) |
| 04:43.07 | tronix | tho I do want to set up a phone (the reason why I got the 7960G) |
| 04:43.14 | [av]bani | thats about the most unusual setting for * i've heard of so far :) |
| 04:43.14 | tronix | is cause sometimes hearing people comes over |
| 04:43.17 | tronix | hahaha |
| 04:43.34 | [av]bani | so err. why a 7960g? |
| 04:43.43 | tronix | had lots of them at work. |
| 04:43.47 | opsys | I guess I'l have to wait until you can ask someone.. |
| 04:43.58 | tronix | opsys: once I find out, I'll be sure to let you know. |
| 04:44.25 | Qwell | tronix: hack up sphinx and the XML stuff...make it spit out the text of a conversation :p |
| 04:44.26 | [av]bani | oh, you stole one ;) |
| 04:44.28 | opsys | tronix: have you had sucess with tdd and faxing? |
| 04:44.29 | tronix | [av]bani: it's also easier to "train" people like parents, contractors working here, etc. on using hardphones |
| 04:45.03 | tronix | [av]bani: nah. boss: "we have too many. help yourself" -- that's the nice part of working for a telecom ;) |
| 04:45.26 | Qwell | tronix: got a link to a job application? heh |
| 04:45.50 | nroej | hmm not working at all |
| 04:45.56 | tronix | Qwell: hmmmm. I'm thinking about seeing what it takes to do a TDD-type softphone. think i'm gonna have to dive into baudot and other interesting/hairy old stuff |
| 04:46.15 | Qwell | tronix: talk to SarahEmm |
| 04:46.16 | tronix | Qwell: no hires yet. but if something comes up, I'll mention it here or your way |
| 04:46.25 | [av]bani | tronix: ooo, can i have some :) |
| 04:46.25 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 04:46.26 | Qwell | she's done a bit of work with TDD stuff |
| 04:46.36 | tronix | [av]bani: :-) |
| 04:46.42 | tronix | Qwell: oh? very nice! thanks. will do! |
| 04:46.45 | [av]bani | let me help you with your "problem" |
| 04:46.49 | tronix | hahahaha |
| 04:46.56 | coppice | tronix: * is supposed to support TDD, but SarahEmm says its very quirky |
| 04:47.39 | tronix | understandable. |
| 04:47.44 | tronix | probably not one of the more exercised code paths |
| 04:47.55 | [av]bani | coppice: * is very quirky ;) |
| 04:48.00 | Qwell | tronix: you two are the only two I've heard of |
| 04:48.17 | coppice | its one of the oldest, though. it goes right back to the original zapata code in 1999 |
| 04:48.36 | tronix | wow, neat. |
| 04:48.37 | *** join/#asterisk iq (n=iq@71-214-3-164.omah.qwest.net) |
| 04:48.54 | coppice | I have solid TDD support, but its not properly integrated into * right now |
| 04:49.05 | tronix | hm, interesting. |
| 04:49.30 | nroej | no voice maybe my testsetup is to complicated |
| 04:49.52 | opsys | there was some fixes to the zatel code to 'fix' faxes on the tdm240 |
| 04:50.00 | opsys | sorry TDM2400 |
| 04:51.22 | nroej | i go gsm->sip->iax2->iax2->capi |
| 04:55.30 | nroej | but it seems like chan_capi doesnt get it when i pick up the call |
| 04:56.10 | *** join/#asterisk Kari1 (n=Karim@243.209-89-66-0.interbaun.com) |
| 04:58.06 | Kari1 | hello |
| 04:58.51 | *** part/#asterisk Kari1 (n=Karim@243.209-89-66-0.interbaun.com) |
| 05:06.59 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 05:17.09 | *** part/#asterisk ckram (n=mcravey@cpe-70-112-233-182.austin.res.rr.com) |
| 05:55.49 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 05:55.52 | l-fy | hello |
| 05:56.30 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 05:57.36 | *** join/#asterisk YARICK (n=spiderma@pool-71-255-197-13.bltmmd.east.verizon.net) |
| 06:02.47 | *** join/#asterisk brian__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 06:04.08 | *** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net) |
| 06:06.20 | Qwell | bkw__: eww |
| 06:07.04 | [av]bani | o_O |
| 06:07.19 | Feral_Kid | Ok, this is probably a lame question. I am connected to a provider through my Sipura 2000... Incoming and outgoing work fine. When I try to access the same provider under *, although the console shows the call is running, I get no sound. |
| 06:07.32 | [av]bani | Feral_Kid: upgrade to 1.2.3 |
| 06:08.59 | Feral_Kid | But the Sipura and the * are plugged into the same router... Incidentally, IAX calls on the * run great, but nothing on the SIP side... |
| 06:09.21 | Feral_Kid | [av]bani> Yes, that has already been done... |
| 06:10.13 | [av]bani | sounds like nat |
| 06:11.07 | trixter | the rtp bug that was introduced doesnt bridge 2 channels, it does let you call an application though |
| 06:11.13 | trixter | this manifested itself on Jan 25 |
| 06:11.32 | Feral_Kid | What would be the difference between the SIPURA and *, the SIPURA is not suffering from that problem, and keep in mind that both the SIPURA and * are on the same router... |
| 06:11.36 | Qwell | trixter: nice hat |
| 06:11.44 | trixter | did you see the lights? |
| 06:11.48 | Qwell | I saw the lights |
| 06:11.53 | trixter | it was off most of the time but the balls light up |
| 06:12.01 | Qwell | trixter doesn't know who I was. :D |
| 06:12.05 | trixter | no one put together 'trixter' and that hat |
| 06:12.27 | trixter | yeah I wasnt there to hide |
| 06:13.04 | Qwell | trixter: we were in the same place at the same time on multiple occasions...mostly outside |
| 06:13.06 | trixter | btw if the rtp bug isnt enough reason, http://www.trxtel.com/crashterisk.c lets you spoof IPs and segfault asterisk remotely for most versions in use |
| 06:13.36 | trixter | 1.2.3 has the patch applied |
| 06:13.44 | [av]bani | Feral_Kid: the sipura does nat pretty seamlessly, * requires a bit of spanking |
| 06:14.19 | [av]bani | tronix: O RLY? |
| 06:14.30 | tronix | hm? |
| 06:14.39 | trixter | its a 1 line fix, dont hangup a channel that doesnt exist.. |
| 06:14.47 | Feral_Kid | [av]bani> Any references of what I need to spank to get this working... Anything on voip-info or elsewhere? |
| 06:14.50 | trixter | I think he used tab completion and your nick is alphabetically before mine |
| 06:14.54 | tronix | ahhh :) |
| 06:15.03 | trixter | Feral_Kid: externip & localnet in sip.conf |
| 06:15.09 | trixter | then turn nat=yes in your client definition |
| 06:15.33 | [av]bani | someone should add stun client support to * :) |
| 06:15.39 | Feral_Kid | trixter> Both of those are defined in sip.conf |
| 06:15.44 | trixter | I have a stun server |
| 06:15.54 | trixter | it really shouldnt be part of asterisk becuase there is no real reason for it to be |
| 06:16.03 | [av]bani | tronix: except in cases like feral_kid's |
| 06:16.05 | trixter | just get a stun server from elsewhere and install it where you want |
| 06:16.08 | [av]bani | where * is a client sitting behind nat |
| 06:16.26 | trixter | my asterisk sits behind nat at home nad I have no problems |
| 06:16.36 | trixter | after setting those two correctly, and nat=yes for the phone definition |
| 06:16.37 | [av]bani | yeah, because you config it manually... |
| 06:16.53 | [av]bani | stun lets it be automagic |
| 06:16.56 | trixter | for GUIs I have to say the mac cocoa stuff is slick |
| 06:17.07 | trixter | there is a reason that people using that GUI config dont ask questions about it |
| 06:17.07 | [av]bani | cocoa... ugh |
| 06:17.11 | Feral_Kid | Actually, I am dealing with double NAT issues |
| 06:17.20 | [av]bani | double nat, welcome to hell |
| 06:17.31 | trixter | my friend did a double nat and didnt have problems |
| 06:17.38 | X-Rob | iax? |
| 06:17.40 | trixter | well initially he did because he set externip but forgot localnet |
| 06:17.45 | trixter | sip |
| 06:17.49 | X-Rob | lucky. |
| 06:17.54 | trixter | its not that hard |
| 06:17.57 | [av]bani | see, stun lets * figure that all out automagic and stuff |
| 06:18.07 | Qwell | trixter: Who was that guy you were with most of the time? Anybody on irc? |
| 06:18.08 | trixter | ahh you want a stun client |
| 06:18.17 | [av]bani | so when you got a * box you tote from site to site, it all automagically configures |
| 06:18.17 | trixter | that would make sense, triggered by some timeout to recheck |
| 06:18.21 | [av]bani | without having to touch a conf file |
| 06:18.32 | trixter | Qwell: the guy that did the double nat stuff without a problem :) he isnt on this network though |
| 06:18.36 | Qwell | ahh |
| 06:18.41 | trixter | he works for me |
| 06:18.46 | [av]bani | tronix: yes, i want * to be able to be a stun client |
| 06:19.00 | trixter | try tri<tab> :P |
| 06:19.16 | [av]bani | :P |
| 06:21.12 | trixter | personally I think a stun client in asterisk wont happen anytime soon |
| 06:21.27 | trixter | as long as sip is intentionally crippled people can argue that something else is better and sip should be avoided |
| 06:21.37 | trixter | dundi v enum is done the same way as sip v iax |
| 06:21.56 | [av]bani | iax!11!!!oneone |
| 06:22.01 | trixter | cripple the standard protocol slightly so that the non-standard one (ie the one that can change at a moments notice) appears better |
| 06:22.16 | [av]bani | well, you could make an external stun support for *, it would be kludgy though |
| 06:22.24 | trixter | but that is my opinion, not backed up by anything other than the fact that sip and enum are intentionally crippled :P |
| 06:22.34 | [av]bani | make it poke a stun server, pick up the settings, ands tuff them in the asterisk conf files |
| 06:22.35 | trixter | unless someone wants to admit to poor coding ability, anyone? anyone? |
| 06:22.48 | trixter | you have to periodically renew that |
| 06:23.12 | trixter | incase your dynamic IP changes - which is you are using NAT there is a better chance of that happening than if you arnet |
| 06:24.05 | justinu | how is sip intentionally crippled? |
| 06:24.12 | X-Rob | ooh, ooh. |
| 06:24.15 | X-Rob | I admit it. My code sucks. |
| 06:24.19 | trixter | silence supression |
| 06:24.21 | trixter | CNG |
| 06:24.22 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 06:24.32 | trixter | its lack of stun for IP resolution |
| 06:24.41 | justinu | oh, you mean crippled in ast? |
| 06:24.46 | trixter | yes |
| 06:24.59 | trixter | so iax seems better in asterisk.. and dundi can publish info but enum cant |
| 06:25.02 | trixter | so dundi seems better |
| 06:25.13 | justinu | sip works pretty well for me in ast |
| 06:25.17 | trixter | little trhings like that, why I dont think a stun client will be approved any time soon for asterisk sip |
| 06:25.21 | justinu | we don't use things like CNG, VAD |
| 06:25.33 | trixter | I personally would like to save on bandwidth |
| 06:25.37 | justinu | and static routable IPs |
| 06:25.47 | trixter | when you have 2 people talking typically one listens one talks, so VAD == 50% bandwidth savings |
| 06:26.00 | trixter | in a conference its 1/N where N is the number of people in the conf |
| 06:26.06 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 06:26.09 | justinu | our upstream provider doesn't support stuff like that either.... |
| 06:26.28 | justinu | g711u or g729 are the choices |
| 06:26.51 | trixter | CNG & VAD arent in the codec |
| 06:26.53 | trixter | or shouldnt be |
| 06:26.58 | trixter | oh no jitter buffer in sip either |
| 06:27.11 | trixter | it was embedded into the iax channel driver instead of generic to the system for a post codec filter |
| 06:27.28 | justinu | i wonder why |
| 06:27.53 | trixter | I personally think it was becuase its an effort to get people to use the digium only standard of iax - rather than something like sip that has RFCs |
| 06:28.04 | trixter | unless someone wants to take credit for bad code ... anyone? anyone? |
| 06:28.17 | Math` | lol |
| 06:28.25 | justinu | i dunno, i think they put it in iax because of the fact that asterisk originally was setup to do iax to TDM conversion |
| 06:28.35 | justinu | i could be totally wrong tho |
| 06:28.41 | trixter | so you think it was a lazy bad coder? |
| 06:28.43 | trixter | well that could be |
| 06:28.59 | trixter | instead of doing it right and making it generic for any protocol tie it to one specifically ... |
| 06:28.59 | [av]bani | tronix: jitter buffer "coming soon" ! |
| 06:29.04 | trixter | heh |
| 06:29.11 | [av]bani | to a pbx near you |
| 06:29.20 | [av]bani | or you can be brave and play with the patches |
| 06:29.45 | justinu | zoa supposedly did a generic jitter buffer |
| 06:30.01 | *** join/#asterisk dmz (n=dmz@209.133.52.162) |
| 06:30.06 | dmz | any fwd users here? |
| 06:30.10 | Math` | yeah |
| 06:30.11 | trixter | it is easier to do it as a channel driver thing becuase you need the timestamps, but it doesnt have to be that way, so its either someone being lazy and not thinking about the real problem, or someone doing it intentionally, either way that doesnt say good things |
| 06:30.23 | dmz | Math, was that yeah for me? |
| 06:30.27 | Math` | yeah |
| 06:30.29 | Math` | :P |
| 06:30.31 | trixter | asterisk is gpl and my religion bars me from contributing to a gpl product |
| 06:30.38 | Math` | lol |
| 06:30.45 | trixter | my patches only goto my customers (who so far havent asked for any of them they dont care) |
| 06:30.47 | dmz | cool...hey is it working for you? i was playing with some settings and now I only get a message saying fwd is only for members |
| 06:30.50 | Math` | what's your religion's license? :P |
| 06:31.04 | dmz | i can't seem to call 888### or 411 anymore |
| 06:31.08 | justinu | trixter: you have some good points |
| 06:31.34 | trixter | I get no rtp on FWD, just tried their echo test |
| 06:31.45 | dmz | rtp? |
| 06:31.46 | X-Rob | it hates you. |
| 06:31.47 | Math` | dmz: 612 works here |
| 06:31.47 | trixter | it worked last week (my system isnt the one with the problem) |
| 06:31.52 | dmz | let me try that |
| 06:32.16 | *** join/#asterisk flashnet (n=flashnet@213.83.63.227) |
| 06:32.22 | trixter | they return congestion on a tollfree |
| 06:32.29 | trixter | so they may be having issues, that one I remember last week |
| 06:33.00 | dmz | that's what i must be seeing, 612 works for me |
| 06:33.19 | dmz | bad error message though, it says it's only available for registered members instead of saying all lines busy |
| 06:33.47 | dmz | cool, at least i know it's not me :) now to get this iax2 switch config working properly |
| 06:35.03 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 06:37.18 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:37.44 | dmz | is there any "easy" way to create/manage meetme conferences other than editing the meetme.conf file? |
| 06:38.04 | harryvv | via a web page interface? |
| 06:38.14 | mdave | ok, heres a Q. if I have a BV account, and I have a SPA directly registering with them, they provide 'three way' and 'conference' calling. Is this handled by the spa making two calls, or are the calls joined at BV? |
| 06:38.38 | mdave | if the latter, how can I access bv's three way and/or conference function from * ? |
| 06:38.46 | Math` | mdave: spa |
| 06:38.49 | mdave | instead of having * make two seperate 'calls' to... |
| 06:38.51 | mdave | really? |
| 06:38.58 | Math` | usually yeah |
| 06:39.00 | mdave | hrm |
| 06:39.10 | Assid | bah,.. i gotta write a php code to understand the envelopes of voicemail |
| 06:39.12 | mdave | the spa doesnt send the 'flash' to bv, and bv gives 2nd tone? |
| 06:39.22 | Math` | some providers might be able to accept sip reinvites for call transfer tho |
| 06:39.37 | Math` | mdave: no the SPA juste originates another call and mixes down their audio |
| 06:39.44 | justinu | mdave: you can configure your spa to use a conference uri, but it's probably not doing that |
| 06:39.54 | Math` | there's no "dial-tone" for sip providers :) |
| 06:40.02 | mdave | seems like the audio quality is much better when I use 3-way when the spa is direct to bv than when I join two channels thru * |
| 06:40.49 | mdave | I was sort of hoping the 'call transfer' concept was being handled at bv's end |
| 06:40.59 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 06:41.14 | Math` | mdave: that might be true if they supporte ReInvites |
| 06:41.18 | mdave | eg, if I had one call one, made a second via 3-way, then conferenced them, and hung up, |
| 06:41.24 | mdave | then the spa was no longer in the loop |
| 06:41.45 | mdave | if they do, how can I make it work that way with * ? |
| 06:42.32 | mdave | so if there are two calls to bv, and I conference them, and then drop out leaving bv to handle the audio between them |
| 06:45.44 | mdave | well I gather thats easier said than done then |
| 06:46.15 | mdave | maybe i'll live without it.. anyway, its off to bed for me |
| 07:07.42 | *** part/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 07:15.15 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.135.125.telnor.net) |
| 07:16.26 | [av]bani | weird... asterisk is hung |
| 07:16.27 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.254.84.45.Dial1.Chicago1.Level3.net) |
| 07:16.37 | *** join/#asterisk Dandan (i=dandan@ellie.pacanka.com) |
| 07:18.33 | [av]bani | hanging on dns lookup for a sip peer... |
| 07:18.39 | [av]bani | nice |
| 07:19.04 | *** join/#asterisk fugitivo (n=ajf@201.255.176.5) |
| 07:20.15 | [av]bani | wow... nasty |
| 07:20.25 | [av]bani | if a peer dns lookup fails, asterisk goes all kinds of wonky |
| 07:28.22 | X-Rob | yep. |
| 07:28.26 | X-Rob | known problem |
| 07:28.33 | X-Rob | async dns is something that they're going to fix 'one day' |
| 07:28.41 | X-Rob | simple answer: use IP addresses. |
| 07:31.16 | [av]bani | roll the clock back to 1983 :)) |
| 07:31.49 | *** join/#asterisk Assid (n=assid@59.183.31.43) |
| 07:32.04 | Assid | hea |
| 07:32.10 | Assid | umm.. |
| 07:32.13 | Assid | for voicemail |
| 07:32.31 | Assid | if i set the timezone to America/New_York instead of eastern.. would it work? |
| 07:35.17 | *** join/#asterisk Netslayer (n=chris@c-24-126-202-231.hsd1.ca.comcast.net) |
| 07:36.51 | Netslayer | what benefit would i have doing voicePulse Voip -> Asterisks server -> my phones vs voicePlus Voip -> call box converter -> my phones? |
| 07:37.09 | Netslayer | basically using asterisk over their converter solution to regular phone lines |
| 07:40.04 | mattwj2005 | I am too as far as that goes |
| 07:40.05 | mattwj2005 | :P |
| 07:41.29 | Netslayer | i'm trying to basically find out 1. who i should use for voip service 2. if i should go with an asterisks server and 3. if i should buy voip phones or buy regular phones |
| 07:41.51 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 07:41.53 | [av]bani | i think an octothorpe server might be better |
| 07:42.14 | harryvv | I need to chat with Ariel so far he is uusing a wholsale provider for his clients and I think the provider is keeping up and providing good service. |
| 07:42.34 | *** join/#asterisk DrPHP (n=assid@203.115.64.12) |
| 07:43.01 | mattwj2005 | personally I like an ATA....for regular phones....it makes cordless so easy |
| 07:43.10 | Netslayer | ata? |
| 07:43.24 | trixter | analog telep[hone adapter |
| 07:43.33 | Netslayer | auh |
| 07:43.43 | trixter | it has an fxs port and typically an ethernet port, just plug it into your network and your telephone and you are good to go |
| 07:44.03 | Netslayer | so if i want asterisk + regular phones i'd buy a compatible PCI card to do that? |
| 07:44.15 | Netslayer | or if i want to use the voip providers hardware i go that route? |
| 07:44.31 | harryvv | to bad there wasnt such thing as a wireliss voip phone with ethernet capability. That way if there is no wireless conectivity at say a office you could just plug it in. |
| 07:44.34 | trixter | pci card or ata |
| 07:44.43 | trixter | typically atas are cheaper $30-50 on average |
| 07:45.14 | harryvv | or, mabey carry a wireless wifi base. |
| 07:45.23 | mattwj2005 | yeah I found my off of ebay for $30 |
| 07:45.25 | Netslayer | are there any benefits really to going with voip phones that plug into cat 5? |
| 07:45.33 | mattwj2005 | it supports two lines |
| 07:45.49 | harryvv | net, compared to what? |
| 07:45.52 | Netslayer | ata |
| 07:46.00 | trixter | I got the dlink AT&T callvantage one for $30, reflashed to sip (it comes mgcp) and it has 2 fxs 1 fxo |
| 07:46.21 | trixter | I am not overly impressed with it but for generic home usage it should be fine |
| 07:46.24 | harryvv | netslayer, a ipphone is just a digital phone with the ata built into it. |
| 07:46.29 | Netslayer | auh |
| 07:46.46 | harryvv | if you want to simplify it. |
| 07:46.47 | Netslayer | so where does asterisk fit into this? |
| 07:46.58 | trixter | well its simplier than that, becuase of the lack of real fxs/fxo signalling but meh close enough :P |
| 07:47.07 | *** join/#asterisk Astar (n=astar@ANantes-154-1-18-6.w81-53.abo.wanadoo.fr) |
| 07:47.09 | mattwj2005 | I haven't played around with them much....but I am guessing you get better sound quality |
| 07:47.19 | mattwj2005 | anyone? |
| 07:47.23 | harryvv | if you travel alot a ATA or simple voip phone would work but you need to also carry a analog phone with the ata. |
| 07:47.53 | Netslayer | this will be my primary phone in my new place |
| 07:47.58 | trixter | a lot of wifi phones wont let you auth with a webbrowser so those arnet good for travel in a lot of circumstances |
| 07:47.59 | Netslayer | no travelling or minimal |
| 07:48.31 | Netslayer | cost wise i'd probably prefer using an ATA. unless I can find a cordless phone (expandable set) that starts at like 150 |
| 07:48.37 | harryvv | a wifi voip phone is good because of the convinence |
| 07:48.57 | Netslayer | ok so what kind of wifi voip phones are available? |
| 07:49.17 | harryvv | google it |
| 07:49.25 | harryvv | im going down stairs see ya |
| 07:49.46 | Netslayer | ic lots here http://www.voipsupply.com/index.php?cPath=95_115 |
| 07:52.29 | Netslayer | I can split an ATA RJ11 cable to as many phones as I want on the same line right? ie it has the power output to handle a few rooms of phones |
| 07:56.14 | mattwj2005 | any ideas anyone? |
| 07:59.08 | mattwj2005 | when I tested my setup in lab....I only had one port per phone......I have no idea what type of signal lost you might have |
| 08:01.02 | argos73 | don't suppose anyone's using a multitech MT5600ZDX hanging off an asterisk box? damn thing doesn't want to recognize caller id info. (MT5600BL on a different port does work) |
| 08:02.19 | I-MOD | Netslayer: depends on the REN value the ata puts out and what REN each phone takes |
| 08:06.19 | I-MOD | http://www.digium.com/downloads/product_sheets/IAXy.pdf |
| 08:06.30 | I-MOD | REN of 5 at 1500 ft |
| 08:06.57 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 08:06.58 | I-MOD | the ren of all attached phones cant add up to more than 5 |
| 08:07.54 | Netslayer | auh thx |
| 08:08.22 | konfuzed | Supaplex: hm |
| 08:08.24 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:08.46 | Supaplex | you think? |
| 08:11.46 | *** join/#asterisk tzafrir_laptop (n=tzafrir@82.166.242.248) |
| 08:15.39 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.254.84.45.Dial1.Chicago1.Level3.net) |
| 08:17.39 | *** join/#asterisk lorinc (n=ang@caracas-3662.adsl.interware.hu) |
| 08:23.05 | *** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net) |
| 08:24.38 | Feral_Kid | Is there anyway to make use of stun with asterisk to get me past this sip issue? |
| 08:41.31 | [av]bani | asterisk cant act as a stun client yet |
| 08:48.25 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 08:53.01 | *** join/#asterisk oatis (n=oatis@c-67-172-176-64.hsd1.ca.comcast.net) |
| 08:54.12 | *** join/#asterisk dalbjerg (n=dalbjerg@host095a.malmohus16.se) |
| 09:03.13 | *** join/#asterisk elcuco (n=diego@82.166.242.248) |
| 09:03.46 | elcuco | tzafrir_laptop, ping |
| 09:11.39 | *** part/#asterisk joshua_ (i=joshua@cl-5.chi-01.us.sixxs.net) |
| 09:21.15 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 09:24.21 | *** join/#asterisk ToTo (n=ToTo@host191-142.pool874.interbusiness.it) |
| 09:33.07 | *** join/#asterisk jorge_ (n=jorge@84.77.52.120) |
| 09:33.19 | jorge_ | hi all |
| 09:33.28 | *** join/#asterisk oatis (n=oatis@c-67-172-176-64.hsd1.ca.comcast.net) |
| 09:33.51 | jorge_ | I was trying to apply the spanish line chan_zap patch to asterisk 1.2.3 |
| 09:34.00 | jorge_ | but I'm getting trouble |
| 09:34.07 | oatis | does anyone know of a site that explains setting up xten softphones to asterisks? so I can forward extentions to them etc... |
| 09:34.28 | jorge_ | does anybody know if that patch code is applied to 1.2.3 code? |
| 09:35.34 | oatis | Im not too clear on how to setup the username/passwords that get applied to the xten settings |
| 09:35.57 | *** join/#asterisk coppice (n=chatzill@236.155.17.210.dyn.pacific.net.hk) |
| 09:38.04 | newl | oatis: http://www.voip-info.org/wiki/view/xten should get you started. |
| 09:44.17 | *** join/#asterisk CyberScripter (n=Arme@213.150.165.190) |
| 09:46.42 | *** join/#asterisk BBytes (n=dw@Toronto-HSE-ppp3710682.sympatico.ca) |
| 09:54.50 | hugo-v6 | gd morning |
| 09:59.12 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:02.00 | BBytes | mornin, hugo-v6 |
| 10:03.11 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 10:11.57 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 10:13.46 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-15-223.37-151.net24.it) |
| 10:14.08 | *** join/#asterisk jluk (n=jon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk) |
| 10:14.59 | oatis | how do I change what the caller id reads when we dial out from the system to like a cell or landline? |
| 10:20.51 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:24.49 | [av]bani | how are you dialing out? |
| 10:27.32 | coppice | decadic pulsing on the string between the cans |
| 10:28.42 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 10:43.23 | wasim_ | its going to be fun, the entire test series depends on how well the indian batsmen handle our pace attack |
| 10:48.46 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 10:53.16 | *** join/#asterisk d-tech (n=dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 10:53.49 | wasim_ | oooh ... laxman and dravid opening |
| 10:59.38 | coppice | wasim: you're going to attack their pacemakers? :-\ |
| 10:59.47 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:01.43 | *** join/#asterisk pauly (n=pauly@84.133.233.220.exetel.com.au) |
| 11:02.01 | pauly | hello need some help with my asterisk server when i dial ext 1235 it give me 1234's mailbox |
| 11:02.14 | pauly | any idea why that might happen people can call my servers box but cannot call another client? |
| 11:02.57 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:03.01 | PakiPenguin | hello everyone |
| 11:05.28 | pauly | hello |
| 11:07.44 | pauly | .server irc.efnet.net |
| 11:12.02 | wasim_ | the question remains, has sehwag sat enough time out to come into bat |
| 11:12.24 | wasim_ | apparently so ... |
| 11:14.14 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 11:15.47 | *** join/#asterisk MatsK (n=mk@6.80-203-84.nextgentel.com) |
| 11:16.06 | [av]bani | hmm |
| 11:19.16 | *** join/#asterisk gambolputty3 (n=gambolpu@cblmdm72-240-116-131.buckeyecom.net) |
| 11:20.20 | RoyK | wasim_: ding |
| 11:20.30 | wasim_ | oh well, sehwag didn't end up mattering too much |
| 11:20.46 | wasim_ | NOW, will the lil' master show us magic ... |
| 11:21.04 | PakiPenguin | shewag bye bye |
| 11:23.14 | [av]bani | is it possible to use zaptel driver's echo canceller as an AEC for non digium hardware? |
| 11:23.54 | wasim_ | ooh ... ouch |
| 11:27.34 | coppice | [av]bani: since it isn't very good, why would you want to? :-\ |
| 11:27.52 | [av]bani | because the spa3000's is worse? |
| 11:28.13 | [av]bani | mg2 is supposed to be ok |
| 11:28.20 | coppice | [av]bani: you certainly can't use it for something like an spa3000 |
| 11:28.29 | [av]bani | because |
| 11:29.13 | coppice | because you will be at the other end of a VoIP pipe from the interface. EC requires a very tight constant length loop |
| 11:30.52 | [av]bani | hm.. so this 128ms, 256ms means nothing? |
| 11:30.57 | [av]bani | and sliding windows |
| 11:31.35 | coppice | how do sliding windows come into this? |
| 11:32.50 | X-Rob | because you'll have to slide open the window to yell at your neighbour when the phone doesn't work? |
| 11:33.47 | coppice | I thought he was referring to microsoft's market share |
| 11:35.31 | X-Rob | speaking of EC, lastest stuff in truck works very very well. |
| 11:35.48 | X-Rob | trunk even |
| 11:36.20 | coppice | X-Rob there are various stories about that. how well it works depends a lot on what you do. its still actually a very crude canceller, easily upset |
| 11:36.51 | coppice | unless miracles occured in the last couple of weeks :-) |
| 11:37.02 | X-Rob | Oh yes, definately. But it works, 99% of the time, which is far better than the 30% we were getting. |
| 11:37.17 | X-Rob | and you can confuse it easily |
| 11:38.22 | coppice | i think its time we started on a profoundly useful kernel EC module, not specifically tied to zaptel |
| 11:38.36 | *** join/#asterisk _upsite (n=upsite@wls.swh.uni-halle.de) |
| 11:42.55 | *** join/#asterisk thosa (n=thosa@p54878371.dip0.t-ipconnect.de) |
| 11:47.16 | RoyK | anyone here that knows a good sip load balancer without the nat problems that ser introduces? |
| 11:51.24 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 11:51.38 | RoyK | hm..... |
| 11:51.48 | RoyK | humtitum.... |
| 11:52.02 | [av]bani | hm, cisco's EC are any good on their VIC-blabla-FXO cards? |
| 11:53.24 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 11:55.00 | X-Rob | coppice, when you say 'we', I hope you're not including me with it - unless I can write the docco for it 8) |
| 11:55.08 | [av]bani | cisco says G.165 32ms ... |
| 12:00.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:01.07 | puzzled | morning |
| 12:01.51 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 12:03.02 | RoyK | morning, puz |
| 12:03.12 | RoyK | i wonder.... |
| 12:03.24 | RoyK | how many calls can an asterisk box handle? |
| 12:04.54 | *** join/#asterisk BugKham (n=lamer@125.24.2.72) |
| 12:04.57 | X-Rob | not fucking 5000. |
| 12:05.27 | BugKham | anyone using Soyo G668? |
| 12:05.38 | coppice | why not? if its all SIP and the audio bypasses he box? |
| 12:05.59 | X-Rob | because the internal bandwidth of the machine is going to be insufficient to switch that much IO |
| 12:06.37 | [av]bani | BugKham: it's a p168 clone |
| 12:06.58 | coppice | I said "and the audio bypasses the box" |
| 12:07.06 | X-Rob | Oh |
| 12:07.08 | X-Rob | I didn't see that |
| 12:07.09 | X-Rob | don't mind me. |
| 12:07.35 | [av]bani | how many packets/sec would 5000 calls be? |
| 12:07.54 | BugKham | [av]bani: can't dial a 4-digit extension |
| 12:08.09 | BugKham | [av]bani: u know where to change? |
| 12:08.20 | [av]bani | BugKham: probably dialplan in the phone |
| 12:08.57 | BugKham | [av]bani: k, will have a look again? |
| 12:09.00 | RoyK | [av]bani: 20ms per packet means 50 packets per second each direction |
| 12:09.03 | coppice | [av]bani: for 20ms packets it would be 5000*50*2 = 500,000 |
| 12:09.13 | [av]bani | http://www.voip-info.org/wiki/view/Asterisk+phone+Soyo+G668 |
| 12:10.05 | [av]bani | and those packets sizes are... |
| 12:10.13 | coppice | the Soyo phone is made by one of the chinese PA168 makers. can't remember which one, though |
| 12:10.37 | [av]bani | i suspect you'd need gbe backbone for 5000 calls alone ... |
| 12:11.32 | X-Rob | [av]bani, I posted this to -users a couple of hours ago: |
| 12:11.32 | X-Rob | To handle 5000 calls coming in over a PRI, you’d need 210 or so T1s or 170 E1’s. |
| 12:11.33 | X-Rob | All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec) |
| 12:11.56 | X-Rob | If you do, honestly, need to handle 5k calls, you’d probably have to have a bank of Cisco 5850s doing the termination – With a max of 5 DS3 (1 DS3 = 28 T1’s) into each one, you’ll need 8, or probably 9 as you’d want to have one as a hot spare. |
| 12:12.11 | X-Rob | Each of those DS3’s would go into some beefy switching fabric (note, that each one is producting 225mbit) |
| 12:12.22 | [av]bani | hm, grandstream calls it 'early dial', snom calls it 'overlap dialing' |
| 12:12.25 | [av]bani | which is the correct term? |
| 12:12.33 | X-Rob | overlap dialing. |
| 12:13.12 | [av]bani | asterisk sure is noisy when snom tries to overlap dial |
| 12:13.16 | [av]bani | lots of errors and warnings |
| 12:13.31 | coppice | the technical term is "premature ejaculation of dialed digits" |
| 12:13.40 | X-Rob | hehe |
| 12:14.56 | [av]bani | finding it hard to get the 'dial 9 for outside line' pbx-ish thing going with overlap dialing |
| 12:15.57 | *** join/#asterisk folsson_ (n=filip@h147n1fls32o985.telia.com) |
| 12:16.11 | [av]bani | sipura doesnt seem to have the feature at all |
| 12:16.14 | [av]bani | ... |
| 12:18.59 | RoyK | is there something that should limit the number of calls apart from cpu load? |
| 12:19.02 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:19.04 | wasim_ | concurrent setup or concurrent? |
| 12:19.14 | RoyK | concurrent |
| 12:19.17 | [av]bani | RoyK: mem bandwidth |
| 12:20.24 | RoyK | [av]bani: that shouldn't be too important,should it? it's not like it's that much data..... |
| 12:20.47 | [av]bani | it could be a bottleneck apart from cpu load |
| 12:25.42 | wasim_ | i'm a happy man at 25 calls per second setup and 250 concurrent |
| 12:26.14 | wasim_ | tdm(iax|sip) |
| 12:26.55 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 12:27.07 | frenzy | hello |
| 12:27.42 | *** join/#asterisk pb__ (n=pb@2002:5612:a976:1:a00:1fff:fe06:93c) |
| 12:28.08 | frenzy | I'm configuring my grandstream device, it asks me iLBC payload between (96 and 127) |
| 12:28.31 | frenzy | which value is best ? |
| 12:29.43 | wasim_ | if there was a best, they wouldn't give you a choice, they'd just stick that in and be done with it |
| 12:29.50 | RoyK | ops |
| 12:29.50 | RoyK | Jan 29 12:29:44 ERROR[17067]: rtp.c:947 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files |
| 12:29.50 | RoyK | Jan 29 12:29:44 WARNING[17067]: chan_sip.c:3083 sip_alloc: Unable to create RTP audio session: Too many open files |
| 12:30.00 | [av]bani | :) |
| 12:30.28 | frenzy | wasim_: I meant compatibility wise with asterisk |
| 12:30.31 | coppice | RoyK: you need to up the kernel file limit |
| 12:31.51 | RoyK | coppice: yeah, and user (ulimit) |
| 12:32.21 | coppice | ulimit is usually set to unlimited |
| 12:36.06 | RoyK | 4663 active channels |
| 12:36.06 | RoyK | 2346 active calls |
| 12:36.16 | RoyK | coppice: no, not on linux systems. the normal is 1024 |
| 12:36.33 | RoyK | which is of course far too low... |
| 12:36.43 | wasim_ | RoyK: iax-iax, same box or two different boxes, and where are teh far ends? |
| 12:39.29 | *** join/#asterisk HaKim (i=_Yesim-@62.162.14.46) |
| 12:42.54 | [av]bani | you need to up the kernel limit |
| 12:43.09 | [av]bani | er :) |
| 12:44.00 | [av]bani | fs.file-max = 16384 |
| 12:44.03 | [av]bani | or something like that |
| 12:44.34 | [av]bani | hmm, its mem dependent now... |
| 12:44.39 | [av]bani | $ cat /proc/sys/fs/file-max |
| 12:44.39 | [av]bani | 152930 |
| 12:47.09 | *** join/#asterisk bbrdrgz (n=alex@p54B01C5B.dip0.t-ipconnect.de) |
| 12:47.22 | *** join/#asterisk ibob63 (n=hp@bb-87-82-26-136.ukonline.co.uk) |
| 12:49.17 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 12:57.18 | RoyK | [av]bani: i'm aware of it...... |
| 12:57.25 | RoyK | :) |
| 12:57.52 | RoyK | [av]bani: i've written a new safe_asterisk script (or rewritten it) to do that on startup. it's in trunk now |
| 12:59.12 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:00.07 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
| 13:00.31 | RoyK | this is quite interessting..... |
| 13:00.35 | RoyK | interesssssssssssting |
| 13:01.15 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 13:03.19 | wasim_ | <PROTECTED> |
| 13:03.20 | RoyK | on a single xeon 3.0 with ht and some 150 concurrent calls, asterisk makes the server eat some 30% cpu |
| 13:03.26 | RoyK | mainly system time |
| 13:03.36 | wasim_ | codec? |
| 13:03.49 | RoyK | just alaw |
| 13:04.19 | RoyK | i dial into one box, which dials another over sip, which dials back with sip, ping pong to a total 300 channels |
| 13:04.49 | RoyK | but _system_ time? |
| 13:08.31 | *** join/#asterisk cianhughes (n=cian@87.192.36.98) |
| 13:09.49 | *** join/#asterisk [ToTo] (n=ToTo@host198-163.pool872.interbusiness.it) |
| 13:11.52 | RoyK | wtf |
| 13:11.58 | RoyK | kernel is using all the time in get_offset_pmtmr |
| 13:27.49 | *** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com) |
| 13:32.48 | *** join/#asterisk _cleric_ (n=dacleric@84-245-189-173.fra.bpool.celox.de) |
| 13:37.05 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 13:44.51 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 14:14.08 | RoyK | hm |
| 14:14.12 | RoyK | quiet |
| 14:15.14 | BBytes | too quiet |
| 14:16.25 | *** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com) |
| 14:22.30 | thazza | Its mostly quite at this time. |
| 14:22.46 | thazza | even quiet |
| 14:22.56 | Seedy | Anyone use RAGI with asterisk? |
| 14:25.31 | RoyK | wtf is ragi? |
| 14:25.35 | RoyK | ~ragi |
| 14:25.47 | Seedy | It is a ruby interface to asterisk |
| 14:25.58 | Seedy | It would be really cool if I could get it to work |
| 14:27.30 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:28.39 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 14:30.24 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 14:37.45 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 14:42.08 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 14:42.09 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 14:46.50 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 14:54.17 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 15:02.16 | *** join/#asterisk Tili (i=Tili@203.101.164.10) |
| 15:06.26 | RoyK | anyone around? |
| 15:06.43 | coppice | no |
| 15:07.58 | *** join/#asterisk WezzeyA (n=weezey@206.210.109.233) |
| 15:08.12 | WezzeyA | asterisk dumped a core this morning. |
| 15:08.41 | RoyK | testing asterisk, one box gets a call from pstn, forwards it to another box over SIP, that box forwards it back on SIP, and this is done 500 times, so a total of 1000 calls are generated. the call is then answered with app_echo. on one server, the one answering the call, the linux system load goes way up. on the other, cpu load is still < 5% |
| 15:08.55 | RoyK | WezzeyA: backtrace it :) |
| 15:09.21 | WezzeyA | RoyK: I just need the core file to do that, right? |
| 15:09.33 | RoyK | yes |
| 15:09.40 | RoyK | and the asterisk binary, of course |
| 15:09.46 | RoyK | gdb asterisk core.xxx |
| 15:09.48 | RoyK | bt |
| 15:10.03 | WezzeyA | now to find that core file. |
| 15:10.34 | RoyK | find / -name core\* -type f :) |
| 15:10.50 | WezzeyA | I found it. |
| 15:12.13 | WezzeyA | http://pastebin.ca/39010 |
| 15:13.41 | WezzeyA | Running: SVN-trunk-r8643M |
| 15:15.40 | WezzeyA | http://bugs.digium.com/view.php?id=6319 |
| 15:15.47 | WezzeyA | already reported. |
| 15:16.40 | RoyK | WezzeyA: is this with a recent 1.2? |
| 15:17.10 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 15:20.24 | *** join/#asterisk copantl (n=copantl@205.240.200.102) |
| 15:23.37 | *** join/#asterisk areski (n=areski@134.Red-88-5-211.staticIP.rima-tde.net) |
| 15:29.48 | bmg505 | hoe ban ons daai guid |
| 15:29.56 | bmg505 | soz wrong channel |
| 15:34.24 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:40.20 | jeffik | <PROTECTED> |
| 15:41.56 | *** part/#asterisk gambolputty3 (n=gambolpu@cblmdm72-240-116-131.buckeyecom.net) |
| 15:43.07 | *** join/#asterisk _Martin_ (n=martin@fast.tomato.it) |
| 15:43.10 | _Martin_ | hi all |
| 15:43.21 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 15:47.39 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
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| 15:49.12 | *** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com) |
| 15:49.17 | *** part/#asterisk aaaa (n=gambolpu@cblmdm72-240-116-131.buckeyecom.net) |
| 15:51.06 | pifiu | morning everyone |
| 15:51.32 | *** part/#asterisk _Martin_ (n=martin@fast.tomato.it) |
| 15:53.04 | *** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 15:55.22 | Assid | hrmm.. when you hear a message.. does it always go to the Old directory? |
| 15:55.40 | Assid | or does it work like imap and change some flag or somethuing but remain in inbox? |
| 15:56.03 | Assid | err.. thats regarding voicemail |
| 15:59.07 | *** join/#asterisk Alric (n=nbowyer@ppp-db.1stel.com) |
| 16:00.04 | *** join/#asterisk _Sam-- (n=sam@dca.kneedraggers.com) |
| 16:00.15 | *** join/#asterisk ToTo (n=ToTo@host137-131.pool872.interbusiness.it) |
| 16:00.58 | _Sam-- | anyone using a mini-itx motherboard with asterisk? |
| 16:01.59 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:02.11 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 16:02.22 | l-fy | hello |
| 16:02.39 | WeezeyD | Assid: always jumps into Old if you've heard it. |
| 16:02.51 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 16:03.58 | Assid | k |
| 16:08.28 | *** join/#asterisk ComPuTeR (n=DobERman@88.224.162.77) |
| 16:11.19 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 16:12.07 | markit | hi :) I don't remember how to prevent 2 sip phones to do a direct bridge with the dial command (want asterisk to stay in the middle), anyone can help? asterisk 1.2.2 |
| 16:15.09 | nroej | markit: canreinvte=no |
| 16:15.16 | _Sam-- | first thing to do is to get off 1.2.2 |
| 16:15.21 | _Sam-- | check topic |
| 16:15.41 | ManxPower | 1.2.2 is massibly broken. |
| 16:15.46 | markit | nroej: ah, a sip flag, I guess |
| 16:16.03 | nroej | markit: sip.conf |
| 16:16.09 | markit | thanks, I've seen, the truth is that I'm using 1.2.x svn, so it's updated, thanks a lot |
| 16:16.28 | nroej | but its not canreinvte its canreinvite ... |
| 16:17.52 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 16:18.09 | markit | even with that at =no, I have in the CLI this: " -- Attempting native bridge of SIP/bt101-17a7 and SIP/bt102-e366" |
| 16:18.38 | markit | (canreinvite is spelt ok since was already in the sip.conf, just commented) |
| 16:19.32 | *** join/#asterisk razu (n=razu@217-159-187-162-dsl.prn.estpak.ee) |
| 16:20.25 | tzafrir_laptop | massively broken? |
| 16:20.42 | tzafrir_laptop | depends on when you try... |
| 16:21.26 | markit | damn, it ignores the atxfer *2, sigh |
| 16:23.36 | wunderkin | you cant use dtmf when you have reinvite on right? |
| 16:25.06 | markit | wunderkin: I've canreinvite=no, even if in the cli seems to try to do a native bridge |
| 16:25.21 | wunderkin | doesnt matter what it says on the console |
| 16:25.33 | wunderkin | i thought that was removed, maybe only in head |
| 16:26.12 | markit | ok, so the native bridge shoud be disabled (yes, I've restarted the asterisk server after editing sip.conf) |
| 16:26.25 | wunderkin | you dont have to restart |
| 16:26.41 | markit | wunderkin: well, it does not hurt ;) |
| 16:26.59 | wunderkin | sip reload or reload chan_sip.so should have worked |
| 16:27.02 | markit | do you know a way to see if really the native bridging is disabled, from the CLI? |
| 16:28.13 | wunderkin | native bridge isnt the same as a reinvite i dont think |
| 16:28.49 | markit | ok, maybe I've problems with music on hold |
| 16:28.55 | markit | let me investigate further |
| 16:29.21 | markit | first time I make it work after long long time, updated the config files but a lot of things to check :) |
| 16:29.46 | markit | wunderkin: thanks a lot :) |
| 16:29.49 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 16:30.10 | markit | btw, no more asterisk news here: http://www.sineapps.com/news.php |
| 16:30.21 | markit | sigh... anyone know what happend to it? |
| 16:30.52 | Math` | domain not renewed? |
| 16:30.53 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 16:30.55 | markit | maybe he just forgot to pay the revenue |
| 16:30.57 | markit | yes |
| 16:31.02 | *** join/#asterisk paulo_Jr (i=bootylic@210.1.82.19) |
| 16:31.27 | markit | is it an accident, or the guy that wrote those usefull news has no interest in it anymore? maybe he is here in the channel... |
| 16:31.43 | wunderkin | math is right |
| 16:32.01 | wunderkin | i wonder how much nsi gauges you for it |
| 16:35.32 | *** join/#asterisk scud (n=scud@12-214-190-139.client.mchsi.com) |
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| 16:40.36 | *** join/#asterisk jmcc (n=jcorgan@64-142-68-61.dsl.static.sonic.net) |
| 16:41.25 | jmcc | hope this is an easy one, but can't seem to find the answer: how to make a zap fxo channel *not* answer on incoming ring? |
| 16:42.13 | I-MOD | just dont call Answer() |
| 16:44.18 | jmcc | I-MOD: I see...so if I want to make an FXO channel outbound only, don't call Answer() in the dialplan context for it |
| 16:44.35 | jmcc | better yet, can I just give zapata.conf an invalid context ? |
| 16:44.46 | I-MOD | maybe.... |
| 16:45.01 | jmcc | easy to try. thanks |
| 16:45.06 | I-MOD | or a blank context |
| 16:45.39 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 16:45.44 | *** part/#asterisk jmcc (n=jcorgan@64-142-68-61.dsl.static.sonic.net) |
| 16:46.35 | ctooley | Anyone know how to get the "Accounts" tab to show up in Windows Messenger to be able to configure it to talk to Asterisk? |
| 16:49.25 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 16:49.40 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 16:49.41 | ctooley | Aha, required an _older_ version of 4.7 |
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| 16:53.55 | *** part/#asterisk jwest- (n=jwest@217.164.240.177) |
| 16:56.16 | *** join/#asterisk phpboy (i=shane@dsl-165-109-237.telkomadsl.co.za) |
| 16:56.18 | phpboy | hey all |
| 16:56.23 | I-MOD | howdy |
| 16:56.34 | ctooley | hello |
| 16:56.49 | phpboy | I'm having trouble dialing out to pstn with my extention... this is the error I get:- |
| 16:57.07 | phpboy | Jan 29 18:55:58 WARNING[7114]: chan_modem_i4l.c:608 i4l_dial: Outgoing MSN 9780 not allowed (see outgoingmsn=,*, in modem.conf) |
| 16:57.38 | *** join/#asterisk mastix (n=mastix@ip-85-160-10-55.eurotel.cz) |
| 16:57.44 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 16:57.54 | mastix | hi everybody |
| 16:58.37 | mastix | i have a question does anybody tried to use g.729 or g.723 codec on asterisk |
| 16:58.41 | mastix | ? |
| 16:59.14 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 17:00.16 | *** join/#asterisk apardo (n=apardo@87.218.44.151) |
| 17:00.59 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 17:02.06 | The_X | hi folks |
| 17:02.27 | mastix | hi there |
| 17:02.49 | The_X | I have a patton acting as a gateway for the asterisk |
| 17:03.06 | The_X | does the sip phone have to be able to talk to the gateway to work outbound? |
| 17:03.12 | The_X | or asterisk routes everything |
| 17:03.34 | The_X | ie my asterisk is on a public IP and the gateway is on a private subnet |
| 17:03.34 | mastix | it is up to you |
| 17:03.43 | The_X | how do you configure it? |
| 17:03.57 | mastix | hmm |
| 17:04.05 | mastix | its litle bit complicated |
| 17:04.24 | mastix | but it is better if you route the call through asterik |
| 17:04.26 | nassy | does anyone know of a web site that compares various PBX's to asterisk? |
| 17:05.08 | mastix | the_x:if you make the routing on asterisk you have to make nat just for asterisk not for each phone |
| 17:06.10 | The_X | I have a 7960 at home |
| 17:06.13 | The_X | asterisk at work |
| 17:06.20 | The_X | and all the phones at work use that asterisk box |
| 17:06.28 | The_X | when I talk from home to the 7960s at work, it works fine |
| 17:06.34 | The_X | but if I call my cell from home |
| 17:06.46 | The_X | cell -> works |
| 17:06.52 | The_X | but voice from 7960 -> cell doesn't |
| 17:07.06 | *** join/#asterisk JohnnyG (n=email@70.114.247.89) |
| 17:07.09 | The_X | so I'm wondering if the problem is not the patton being unreachable from home |
| 17:07.35 | *** join/#asterisk ___root__ (i=crapmars@209.167.68.254) |
| 17:09.11 | *** join/#asterisk SGM (n=stoyan@213.91.216.130) |
| 17:09.27 | mastix | the patton is also on private ip address at work? |
| 17:09.30 | *** join/#asterisk dalbjerg (n=dalbjerg@host095a.malmohus16.se) |
| 17:09.34 | The_X | yes |
| 17:09.44 | The_X | asterisk eth0 = public |
| 17:09.47 | The_X | eth1= private |
| 17:09.51 | The_X | for talk with patton |
| 17:09.55 | JohnnyG | Hello all, I'm currently deciding between forking over ~50k to Fonality.com for a PBX system or building one myself. I've found some amazing open source apps, asterisk@home, astlinux - what is the tool I should look closely at if I want to run a phone system over 6 states and 120 people? |
| 17:11.19 | dpryo | JohnnyG: Debian + asterisk is a win :) |
| 17:12.31 | *** join/#asterisk saw (n=saw@skywalker.patronas.de) |
| 17:12.39 | saw | moin gents. |
| 17:13.05 | saw | is there already bristuff out for 1.2.3? ... the patches for 1.2.2 fail on 3 hunks. |
| 17:13.11 | JohnnyG | dpryo: I've seen Zaptel by Digitum mentioned as the hardware of choice, since Digitum is the main developer for asterick (at least, thats what I gathered from the reading) |
| 17:13.46 | JohnnyG | dpryo: how do their cards stack up in terms of price, functionality and ease of install |
| 17:13.46 | dpryo | JohnnyG: Sangoma also produces good boards |
| 17:14.31 | dpryo | It's very cheap if you configure stuff your self. |
| 17:15.28 | dpryo | And the documentation is out there, easy to find |
| 17:15.29 | SkramX | ctooley: Hey man. |
| 17:15.43 | *** join/#asterisk BugKham (n=lamer@125.24.31.22) |
| 17:15.48 | SkramX | We spoke a while back about /possible/ internship, etc. |
| 17:15.50 | tzafrir_laptop | saw, use 1.2.2 and the i patch. It includes the critical fix of 1.2.3 |
| 17:16.12 | saw | ah, cool. |
| 17:16.14 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 17:16.18 | JohnnyG | dpryo: right now, each of the 6 offices has analog lines and old phones - are these old phones upgradable to VOIP or is that a pipe dream? how do phones work? is there a "phone PCI" slot? |
| 17:16.45 | saw | at least its less work than fiddle with the patches stuff and so. thanks, mate. |
| 17:17.10 | BugKham | can we connect two asterisk servers with 2 E100Ps? |
| 17:17.59 | BugKham | don't quite get it when reading from the wiki |
| 17:17.59 | wunderkin | BugKham, yeah you can |
| 17:18.12 | The_X | if I have a sip phone on a public ip address talking to an asterisk server on a public address to phones behind the asterisk on private addresses, can the sip phones talk to each other? |
| 17:18.17 | dpryo | JohnnyG: There is hardware for connecting analog pots-phones to asterisk. I think sangoma got it |
| 17:18.19 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:18.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:18.29 | phpboy | I'm having trouble dialing out to pstn with my extention... this is the error I get:- |
| 17:18.34 | phpboy | Jan 29 18:55:58 WARNING[7114]: chan_modem_i4l.c:608 i4l_dial: Outgoing MSN 9780 not allowed (see outgoingmsn=,*, in modem.conf) |
| 17:18.34 | dpryo | JohnnyG: or you could get sip-adapters for each phone. |
| 17:18.35 | BugKham | wunderkin: what media do I use? a cross over cable? |
| 17:18.38 | I-MOD | digium tdm2400p or tdm400p |
| 17:18.39 | *** part/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:18.51 | I-MOD | for the analog -> asterisk |
| 17:18.52 | wunderkin | BugKham, a t1 crossover |
| 17:19.03 | JohnnyG | dpryo: I hear PRI and i hear SIP, whats the difference? |
| 17:19.11 | I-MOD | s/t1/e1 |
| 17:19.20 | I-MOD | nvm |
| 17:19.26 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 17:19.31 | dpryo | JohnnyG: PRI is t1/e1 trunks, SIP is a networking protocol |
| 17:19.32 | phpboy | JohnJacob: SIP == sip protocal PRI is an ISDN type |
| 17:19.32 | I-MOD | JohnnyG: completely different |
| 17:19.55 | phpboy | JohnnyG: dirrectly they have nothing to do with one another |
| 17:19.56 | BugKham | wunderkin: hmm, u mean just like conecting two computers on LAN, right? |
| 17:20.01 | JohnnyG | dpryo: what protocol does SIP oppose or replace |
| 17:20.25 | I-MOD | SIP is a voip protoco; |
| 17:20.27 | I-MOD | l |
| 17:20.29 | dpryo | JohnnyG: It's a protocol that sets up connection between pbx's or voip-phones over TCP/IP |
| 17:20.33 | dpryo | JohnnyG: or udp/ip |
| 17:20.36 | phpboy | JohnnyG: None of the above... it's the 'telephone' protocal of VOIP |
| 17:20.38 | wunderkin | BugKham, not sure how you mean.. but a t1 crossover is not the same as an ethernet crossover, google for it... the cable will run from card to card |
| 17:20.46 | phpboy | guys... my outgoingmsn |
| 17:20.50 | phpboy | anyone go any ideas? |
| 17:21.32 | wunderkin | BugKham, also you can use voip.. iax,sip,etc |
| 17:21.51 | I-MOD | phpboy: you want to connect msn messenger to *? |
| 17:22.10 | BugKham | wunderkin: k, i will need an E1 crossover then |
| 17:22.18 | dpryo | Or perhaps set the isdn msn? |
| 17:22.25 | wunderkin | BugKham, yes but they are the same |
| 17:22.38 | wunderkin | i would think |
| 17:22.40 | dpryo | (Though it's possible to connect messenger to asterisk, via SIP) |
| 17:23.31 | I-MOD | uggg....acronyms getting jumbled in head |
| 17:23.32 | wunderkin | yes it is the same |
| 17:24.36 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 17:24.48 | *** join/#asterisk roulduke_ (i=8lmlq569@p508D313B.dip0.t-ipconnect.de) |
| 17:24.54 | BugKham | wunderkin: they are made from CAT 5 cable with RJ45, I guess |
| 17:25.15 | I-MOD | yeah |
| 17:25.25 | I-MOD | 1->4,2->5 |
| 17:26.16 | phpboy | hmmm... this is stange |
| 17:26.34 | phpboy | strange... even if I remove my modem.conf file... it still dials through ISDN |
| 17:26.40 | phpboy | weird |
| 17:27.13 | *** join/#asterisk crapmars (i=crapmars@209.167.68.254) |
| 17:27.23 | phpboy | dpryo: how do I reload modem.conf... I'm obviously not doing it correctly |
| 17:27.29 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 17:27.44 | tronix | can Asterisk do transcoding? e.g. IAX2 soft phone -> Asterisk -> Cisco 7960G SIP hardphone? |
| 17:27.58 | I-MOD | yes |
| 17:28.08 | tronix | hm. ok. I must be goofing my IAX2 setup somewhere. :) thanks! |
| 17:28.15 | tronix | (I've got all of my SIP hard/soft phones working great.) |
| 17:28.34 | websae | where is the best place to purchase SIP phones and equipment---for the best deals? |
| 17:28.56 | [TK]D-Fender | websae : Depends on what products specifically, and where you live. |
| 17:29.03 | tronix | depends. I've seen a few telecom dealers with huge batches of stuff to unload, NIB and all |
| 17:29.09 | dpryo | phpboy: 'reload' in cli should reload it |
| 17:29.14 | websae | Wisconsin---looking for sip phones |
| 17:29.15 | phpboy | hmm |
| 17:29.16 | [TK]D-Fender | websae : Some dealers are better for certain models than others. |
| 17:29.24 | phpboy | I removed the modem.conf file but it still works |
| 17:29.31 | phpboy | wtf? :/ |
| 17:29.39 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 17:29.42 | The_X | how can you get a phone on a public IP to talk to a phone on a private ip behind asterisk |
| 17:29.51 | websae | how about sipura/cisco |
| 17:30.05 | The_X | ie 7960 (public) -> asterisk(public) -> 7960(private) |
| 17:30.27 | The_X | asterisk server has 2 interfaces |
| 17:30.44 | JohnnyG | dypro: so I buy a server, I buy PRI lines, i install debian and asterisks, I buy a PRI supporting card and phones that support VOIP using the SIP protocol |
| 17:30.49 | tronix | websae: I can't do PMs at moment (no pw) but let me look up where I got mine |
| 17:31.08 | websae | tronix: would greatly appreciate that |
| 17:31.58 | dpryo | JohnnyG: Yes, however, you don't need 6 different PRI lines. If the offices has a good internet connection you could route all calls to one main office. |
| 17:32.22 | phpboy | dpryo: I restarted asterisk and it seemed to reload the modem.conf file |
| 17:32.42 | phpboy | but even if I set the out going MSN... ie the last 4 digits of the number |
| 17:32.50 | tzafrir_laptop | phpboy, noload chan_modem.so ? |
| 17:33.07 | phpboy | it still doesn't ring with the correct extention |
| 17:33.28 | tzafrir_laptop | oops, sorry. People actually still use it |
| 17:33.35 | phpboy | tzafrir_laptop: nope... it does work to the PSTN |
| 17:33.47 | phpboy | just doesn't dial through the correct extentio |
| 17:33.50 | phpboy | extention even |
| 17:33.55 | JohnnyG | dpryo: we are out of Houston, Tx and Cbeyond seems like a good VOIP provider. We'd like to run everything out of there. If an office in New Orleans has a broadband connection and 3 phones - how do I get those phones to use our Houston office to call out? |
| 17:34.00 | tzafrir_laptop | phpboy, analog? |
| 17:34.27 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 17:34.36 | tronix | websae: I bought my new 7960G from optimumdata.com via eBay but looks like they only deal with Cisco gear. I saw another voip hw provider somewhere but can't remember, alas. |
| 17:34.42 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 17:34.43 | dpryo | JohnnyG: Either get the SIP-phones to connect to the Houston-asterisk, or set up an asterisk in new orleans too, and connect the two asterisks via IAX2 |
| 17:35.17 | tronix | (for folks following in channel, yes, I have a borrowed work 7960G that's hooked into the corporate CCM via VPN *plus* my own 7960G -> my * server) |
| 17:35.31 | phpboy | tzafrir_home: ISDN |
| 17:36.02 | phpboy | it works from the outside... if I dial a certain extention it goes through to the right extention... but if I dial out it dials through the main number :< |
| 17:36.07 | phpboy | not the MSN I specify :/ |
| 17:36.36 | JohnnyG | dpryo: how does a SIP enabled phone get told to connect to our asterisk server in houston? do you configure it to say "connect to this static IP with this password/key?" |
| 17:37.13 | dpryo | JohnnyG: Yeah, you define extension and ip-addresses |
| 17:37.28 | dpryo | JohnnyG: You probably want a VPN-connection. |
| 17:37.34 | tronix | if it's something like the 7960G, you define SIP proxy server IPs in the phone's config, then let the SIP proxy server (asterisk) sort out routing |
| 17:37.40 | *** join/#asterisk TiDO (i=CaKo@p548888B8.dip0.t-ipconnect.de) |
| 17:38.00 | tronix | ditto with user/pass for * in the phone's config too |
| 17:38.54 | phpboy | tzafrir_laptop: what do u think? |
| 17:39.31 | JohnnyG | dpryo: how much call traffic can a low end broadband connection handle without quality problems? New Orleans broadband is understandably recovering |
| 17:40.25 | dpryo | JohnnyG: One single call needs at least 64kbits (depending on the codec used of course) |
| 17:40.42 | websae | use g.729 |
| 17:40.51 | websae | 20kbps |
| 17:40.52 | dpryo | yeah, g729 uses around 8k? I think |
| 17:40.58 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 17:41.01 | slan | What 3 letters would I type to lookup the directory entry for "Larry Office"? Tried 527, 633, 634 and the extension # 120. |
| 17:41.43 | websae | how do you get voicemail setup with g.729 codec without having to by the ports to do so? |
| 17:41.45 | JohnnyG | dpryo: so codecs basically let you choose how much quality you want - some are for 64k and some only take 8k and you trade size for quality? |
| 17:41.45 | websae | anyone know? |
| 17:41.46 | markit | language=it in extensiosn.conf seems have no effects... has the internationalization structure changed since 1.0? is having sounds/it ok still? |
| 17:42.21 | dpryo | JohnnyG: Well, g.729 uses little bandwidth, however it got very good quality too |
| 17:42.30 | markit | well, CLI states "(language 'en')" at every message, so seems it does not understand the language setting |
| 17:42.33 | JohnnyG | no kidding... |
| 17:42.49 | tzafrir_laptop | markit, hi just got your email |
| 17:43.13 | markit | tzafrir_laptop: are you debian manteiner? :) |
| 17:43.15 | tzafrir_laptop | markit, no, the structure hasn't |
| 17:43.23 | dpryo | brb, the cat is attacking the tarantella.. argh |
| 17:43.26 | tzafrir_laptop | markit, sort of. I help there |
| 17:44.01 | tzafrir_laptop | I believe that the layout in the SVN has changed a bit, but they'll restore it for tarballs. |
| 17:44.17 | phpboy | :/ |
| 17:44.32 | *** join/#asterisk websae_ (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 17:45.00 | tzafrir_laptop | markit, maybe the language was set explicitly to "en"? either in the channel definitions or in the dialplan |
| 17:45.14 | markit | tzafrir_laptop: well, CLI tells that the language selected is "en", even if I've language=it in [general]... this worked fine with 1.0.x (I'm restarting using asterisk since octoper 2004...) |
| 17:45.38 | websae_ | hrm |
| 17:45.43 | websae_ | anyone using the g.729 codec |
| 17:45.49 | websae_ | i have issues getting vmail setup |
| 17:45.52 | tzafrir_laptop | markit, The syntax of setting variables might have changed a little. |
| 17:46.55 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 17:48.05 | JohnnyG | drpyo: thank you very much for taking the time to answer my questions. :) |
| 17:48.13 | markit | tzafrir_laptop: I had nothing in sip.conf, and language=it in extensions.conf, and ignored it |
| 17:48.14 | slan | Is there a CLI command to force Asterisk to re-read its configuration files? |
| 17:48.24 | I-MOD | reload |
| 17:48.25 | markit | tzafrir_laptop: setting language=en in sip.conf solved |
| 17:48.39 | markit | tzafrir_laptop: is it a bug or just my ignorance? |
| 17:48.50 | slan | I-MOD: Thanks I'll try that now. |
| 17:51.26 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:52.02 | tzafrir_laptop | markit, I never saw the order in which those definitions should be applied documented anywhere |
| 17:52.06 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl29.dialup.mindspring.com) |
| 17:52.32 | tzafrir_laptop | hmm, misread |
| 17:53.45 | tzafrir_laptop | I'm not sure languagge=it in extensions.conf should apply. |
| 17:53.50 | tzafrir_laptop | Should it? |
| 17:53.55 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 17:54.20 | Connor | Is their any major routing issues past few days?? I've been having a hard time pulling a few major sites.. |
| 17:56.31 | slan | I-MOD: Tried changing the HOSTNAME in sysconfig/network, then asterisk.reload but HOSTNAME did not change. There is no reload, just asterisk.reload in A@Home |
| 17:56.58 | slan | Hate doing shutdown -r now all the time |
| 17:57.15 | markit | tzafrir_laptop: I'm reading documentation, probably should not... no error is raisen, though, but it does not work! |
| 17:57.16 | I-MOD | just do a stop now in asterisk cli |
| 17:57.23 | I-MOD | and start it up again |
| 17:57.39 | *** join/#asterisk YARICK (n=spiderma@38.118.54.14) |
| 17:57.53 | slan | I-MOD: What commands to 'stop' and 'start'? |
| 17:58.05 | *** join/#asterisk _cleric_ (n=dacleric@84-245-189-173.fra.bpool.celox.de) |
| 17:58.14 | I-MOD | can you get to the asterisk cli? |
| 17:58.18 | I-MOD | type stop now |
| 17:58.53 | slan | I-MOD: Yes I do cli. I'm at another machine. Just a min. |
| 17:59.13 | slan | I-MOD: Did you mean type 'stop' or 'stop now'? |
| 17:59.20 | I-MOD | "stop now" |
| 17:59.25 | slan | I-MOD: |
| 17:59.34 | slan | I-MOD: Roger thanks. |
| 18:00.41 | markit | tzafrir_laptop: I'd better upload a new readme.pdf for the italian set and specify this (my mistake) |
| 18:00.45 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 18:01.22 | slan | I-MOD: There is no 'stop' in which and no variation of stop in locate. Maybe AAH is different. |
| 18:01.26 | tzafrir_laptop | markit, any chance you could have a simple plain-text README? I don't always have a PDF reader when in the shell |
| 18:02.10 | markit | tzafrir_laptop: it's a text with a lot of formatting, prepared with OpenOffice, it's 99% italian |
| 18:02.12 | znoG | does anyone use distinctive ring detection with a Zaptel FXO card? |
| 18:02.35 | I-MOD | slan: not just different, retarded too |
| 18:02.42 | tzafrir_laptop | znoG, I used it a bit. read the docs. |
| 18:02.57 | tzafrir_laptop | znoG, anyway, I'm going soon |
| 18:03.08 | znoG | tzafrir_laptop: have read' the docs, did everything the docs said to do (which isn't much, simply usedistinctiveringdetection=yes and setting the dring patterns) |
| 18:03.19 | I-MOD | kill -9 `pidof asterisk` |
| 18:03.30 | tzafrir_laptop | znoG, It was nice, bt didn't work in 100% of the cases. Ad anyway, it was a x100p card |
| 18:03.38 | znoG | the main problem is the distinctive ring patterns are always 0,0,0 .. and asterisk never really waits those 2 seconds before answering to detect the pattern |
| 18:03.53 | slan | I-MOD: Is that dangerous? Will it reboot or just re-read the configurations? |
| 18:04.00 | tzafrir_laptop | znoG, did you set a separate context for each type? |
| 18:04.23 | tzafrir_laptop | znoG, also, some verbosity would help |
| 18:04.31 | slan | I-MOD: Maybe it's _me_ thats retarded <g> |
| 18:04.38 | znoG | tzafrir_laptop: i started asterisk with -vvvvvvvvvvvc ... |
| 18:04.50 | znoG | tzafrir_laptop: yep, i set a different context for each dring patter |
| 18:04.50 | znoG | n |
| 18:05.08 | tzafrir_laptop | so maybe the ring is not close enough to your patterns? |
| 18:05.10 | znoG | tzafrir_laptop: but, as i said, the main problem is that Asterisk doesn't wait 2 seconds to detect the ringing pattern, no idea why |
| 18:05.27 | tzafrir_laptop | znoG, also: did you restart asterisk after changes to zapata.conf? |
| 18:05.29 | tzafrir_laptop | GTG |
| 18:05.44 | znoG | yeah did that many times |
| 18:05.58 | znoG | ok thanks tzafrir if you're on later, and you feel like helping out, i'd love a look at your conf files |
| 18:08.15 | kuku5 | anyone reselling level1 termination ? |
| 18:09.51 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 18:10.29 | wunderkin | lol you mean level3? |
| 18:12.08 | JohnnyG | what can asterisk do that asterick@home can not |
| 18:12.26 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.135.125.telnor.net) |
| 18:14.44 | dogtanian | JohnnyG: nothing much |
| 18:15.48 | JohnnyG | if I ran astericks@home for 120 users, would that be foolish? |
| 18:15.59 | dogtanian | hmm |
| 18:16.06 | dogtanian | i'm not sure about that one tbh |
| 18:19.03 | RoyK | anyone that knows how to simulate, say, 20k SIP clients? |
| 18:29.21 | markit | tzafrir_laptop: new files updated and uploaded (time: 19:17) |
| 18:33.29 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 18:37.22 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 18:38.14 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 18:38.17 | jhiver | hi all |
| 18:38.51 | *** part/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 18:38.53 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 18:39.20 | *** join/#asterisk kippi1 (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com) |
| 18:39.22 | kippi1 | hey |
| 18:39.33 | jhiver | I was wondering how to manage prefix translation in asterisk |
| 18:39.42 | jhiver | you know, like I have a [world] context |
| 18:39.52 | jhiver | with numbers in international format, so for instance |
| 18:39.52 | kippi1 | whats the call stats program called? that gives you all your call info in a web page? |
| 18:40.10 | jhiver | _33X. => Dial (...) ; provider for france |
| 18:40.34 | jhiver | and I want in another context to be able to dial 0X. instead of 33X. |
| 18:40.47 | jhiver | how would you handle this in the dialplan? |
| 18:44.06 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 18:47.07 | kippi1 | is there away on queues to be able to put a lan line or mobile number in there? |
| 18:47.12 | Qwell | JohnnyG: yes, it would be very foolish |
| 18:47.25 | Qwell | kippi1: sure |
| 18:47.46 | Alric | kippil: Yeah, thats possible in a few ways. |
| 18:47.48 | Qwell | just put in the same thing you would for Dial() |
| 18:47.53 | kippi1 | in agents put the number in there? |
| 18:52.26 | kuku5 | wunderkin: level3 :) |
| 18:52.30 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 18:53.23 | *** join/#asterisk Drew___ (n=foo@zux221-065-169.adsl.green.ch) |
| 18:55.28 | *** join/#asterisk hickins (n=dtg19@213.186.161.29) |
| 18:56.48 | kippi1 | how would you add a agent on the cmd? |
| 18:58.10 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:59.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:59.38 | *** join/#asterisk arbius (n=arbius@c-67-173-45-34.hsd1.il.comcast.net) |
| 19:04.32 | hickins | can anyone tell me how can I detect hangups? i need to execute code when hangup |
| 19:04.45 | wunderkin | in the queues.conf put member => SIP/peer/number maybe? i would use agentcallbacklogin though, because otherwise you would get multiple calls at once |
| 19:06.28 | kippi1 | when I try and login it asks me for a new extension |
| 19:06.35 | hickins | we use zap only |
| 19:07.25 | wunderkin | hickins, h exten |
| 19:07.53 | wunderkin | kippi1, exten => 201,1,AgentCallbackLogin(1001||1000@autodial-outagent) |
| 19:08.30 | *** part/#asterisk saw (n=saw@skywalker.patronas.de) |
| 19:08.59 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 19:10.00 | kippi1 | hmm still asking for my new extension |
| 19:10.19 | wunderkin | then you arent supplying a valid exten/context |
| 19:11.07 | hickins | thanks |
| 19:11.11 | kippi1 | hmm |
| 19:11.24 | *** join/#asterisk trixter_ (n=trixter@65.172.209.246) |
| 19:11.57 | wunderkin | [autodial-outagent] exten => 1000,1,Dial(... |
| 19:14.30 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 19:14.31 | *** join/#asterisk backblue (n=moo@87-196-9-78.net.novis.pt) |
| 19:15.30 | kippi1 | wunderkin: can i pm you? |
| 19:15.33 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 19:16.51 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 19:17.50 | trixter | if I believed in symbolism I 3would have to comment on the fact there are 420 users in here.. doh! someone just left |
| 19:21.10 | Drew___ | i have a grandstream gxp2k phone with the newest beta firmware that supports "asterisk BLF" for the quickdial buttons/LEDs - i would like to use one of the LEDs with the dialplan - is there a way to manipulate the LEDs using the dialplan? OR how do i spoof the status uf a extension so that the phone activates the LED?? |
| 19:23.27 | Aughey | Drew: If you find out the answer to this question, I'd be interested too. I just got a gxp-2000 to test and that would be a nice feature to have |
| 19:23.52 | Aughey | I'd like to light up those buttons if someone is transferred to a park extension |
| 19:24.29 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 19:24.41 | trixter | I woiuld like to make them blink in series so its like a cylon eye |
| 19:24.59 | xachen | :p |
| 19:25.09 | tronix | what port was iax2 on? was it 4569 or was it 5060? |
| 19:25.14 | tronix | 5060's sip and 4569's iax? |
| 19:25.16 | Qwell | 4569 udp |
| 19:25.17 | trixter | 5060 is sip |
| 19:25.17 | Qwell | yes |
| 19:25.20 | tronix | sweet. thanks |
| 19:25.21 | Aughey | Draw: have you looked at http://www.jackenhack.com/blog/archives/2005/11/22/setting-up-subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/ |
| 19:25.40 | tronix | i've also finally gotten my iaxy to work and 7960g and some other stuff... getting there. |
| 19:25.56 | Drew___ | yes aughey - that works - but i dont need the led to show the stutus of a _real_ extension |
| 19:26.05 | luke-jr_ | Jan 29 19:21:32 [kernel] asterisk[11234]: segfault at 00000000000000f8 rip 0000000000415320 rsp 00000000409fdc60 error 6 |
| 19:26.05 | luke-jr_ | :( |
| 19:26.15 | Drew___ | i what it to show some kind of info - so i need to spoof the status of a extension |
| 19:26.15 | *** join/#asterisk Silivrenion (n=admin@unaffiliated/silivrenion) |
| 19:26.15 | trixter | that is no good |
| 19:26.22 | Qwell | Aughey: there are so many things wrong with that URL, I don't even know where to start |
| 19:27.01 | *** join/#asterisk simong (n=simong@c-4680e455.68-0008-74657210.cust.bredbandsbolaget.se) |
| 19:27.14 | trixter | luke-jr_: what were you doing when it segfaulted? |
| 19:27.16 | Aughey | Qwell: with the text, or the url itself |
| 19:27.19 | Qwell | Aughey: the url |
| 19:27.57 | Drew___ | hm.... this might work: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate - but what is this bristuff stuff?? |
| 19:28.00 | Aughey | I'm just the messenger |
| 19:28.02 | Silivrenion | i saw a video about asterisk with someone using a wireless phone thing that connected to wireless access points to communicate with the home line of the asterisk box. I know i'll need that sip box to interface the phoneline to ethernet, but where can I find the actual wireless phone device? |
| 19:28.28 | Math` | a wifi videophone? |
| 19:28.30 | Drew___ | silvi - look in the voip wiki under hardware phones ;-) |
| 19:28.40 | Silivrenion | Math` :: not videophone... |
| 19:28.46 | Qwell | Math`: when you need pr0n on the go |
| 19:28.56 | Silivrenion | Drew___ :: do you have a link? |
| 19:28.57 | Drew___ | lol |
| 19:28.58 | Math` | Qwell: heh, I wanted to make smth up with my tv tuner card |
| 19:29.17 | Math` | "Please dial the channel you wish to watch followed by the pound key" |
| 19:29.40 | Nivex | dude! That is such a good idea! |
| 19:29.54 | Math` | lol |
| 19:29.55 | Nivex | or since I don't watch much TV, tie it in to my media player |
| 19:30.16 | Math` | TV over IP aint new |
| 19:30.16 | Silivrenion | no, i just want to interface a wifi phone with my server, so i need to find out the hardware costs to do this |
| 19:30.24 | Drew___ | http://www.voip-info.org/wiki/view/VOIP+Phones#WLANorWiFiPhones |
| 19:30.29 | Nivex | 4 for prev track, 5 for pause/play, 6 for next track |
| 19:30.29 | Math` | there's some on voipsupply |
| 19:30.34 | Silivrenion | thanks |
| 19:30.36 | Math` | http://www.voipsupply.com/index.php?cPath=270_279 |
| 19:30.44 | Qwell | Nivex: there is an AGI jukebox |
| 19:30.55 | Drew___ | any ideas on that bristuff ting an the LEDs? |
| 19:30.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net) |
| 19:31.12 | Math` | Silivrenion: but... motorola released a dual wifi+gsm phone |
| 19:31.20 | Drew___ | math - thats overkill |
| 19:31.38 | trixter | redundant wifi! |
| 19:31.56 | trixter | that way if the RF signal is too weak on one you cna use the backup! yeah that will work |
| 19:32.07 | Nivex | seamless hopping... you get the connection on the 2nd wifi before you release the first |
| 19:32.31 | *** join/#asterisk cpm (n=Chip@border1.avitecture.net) |
| 19:32.38 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 19:32.55 | luke-jr_ | trixter: nothing |
| 19:33.01 | luke-jr_ | Jan 29 19:26:46 [kernel] asterisk[2862]: segfault at 00000000000000f8 rip 0000000000415320 rsp 00000000409fdc60 error 6 |
| 19:33.01 | luke-jr_ | just happened again |
| 19:33.07 | luke-jr_ | I wasn't even home the first time |
| 19:33.25 | Silivrenion | badassaur!!! http://www.satougaki.com/badasssaur.GIF |
| 19:33.26 | luke-jr_ | or maybe I just got home |
| 19:33.36 | trixter | I dunno, sounds like asterisk is b0rked |
| 19:34.01 | luke-jr_ | :/ |
| 19:34.05 | luke-jr_ | it's been working fine for months |
| 19:34.15 | Nivex | Qwell: link? Can't seem to find on google or voip-info |
| 19:34.17 | trixter | it shouldnt segfault for no reason |
| 19:34.28 | Qwell | Nivex: it's on the bug tracker |
| 19:34.36 | luke-jr_ | well, it seems to be :( |
| 19:35.11 | trixter | looking at the logs a ton of people downloaded crashterisk.c last night, that will cause that |
| 19:35.24 | trixter | http://www.trxtel.com/crashterisk.c |
| 19:35.26 | luke-jr_ | crashterisk.c? |
| 19:35.30 | trixter | maybe someone is picking on you |
| 19:35.33 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 19:36.02 | luke-jr_ | hrm |
| 19:36.19 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 19:36.28 | luke-jr_ | probably |
| 19:36.48 | tzafrir_laptop | trixter, heck, simply add Segfault to the dialplan and dial there |
| 19:37.01 | luke-jr_ | Jan 29 19:21:32 [asterisk] NOTICE[11088]: chan_sip.c:7527 in handle_request: Client '65.172.209.246' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead_ |
| 19:37.04 | trixter | that isnt as remote as crashterisk :P |
| 19:37.04 | Drew___ | any ideas on how i can get the gxp2k to _look_ different dependant on a status in the dialplan?? - i.e. agent logged on or DND - are there any commands to change i.e. backlighting, LEDs, text on screen??? |
| 19:37.43 | luke-jr_ | trixter: maybe I should fight back? ;) |
| 19:38.09 | trixter | luke-jr_: that is what it will do once a newer version is done, what bothers me is that is my IP.. since I specifically put IP spoofing in crashterisk that makes me wonder ... |
| 19:38.23 | luke-jr_ | O.o |
| 19:38.32 | Silivrenion | it seems that a lot of these phones are pretty expensive ($200 - $500).:( |
| 19:38.49 | trixter | its a udp packet its *trivial* to spoof, I just dont like the fact that someone is spoofing my IP, unless you believe that I would use a tool that has spoofing and infact requires a hostname to send from when run, but put my own in |
| 19:39.04 | ManxPower | Silivrenion, Um, that's the going price for a prepaid cell phone. |
| 19:39.12 | ManxPower | it's not expensive for VoIP phones |
| 19:39.38 | Silivrenion | # Hitachi Cable: WirelessIP5000 WiFi SIP phone, about $250+ USD, available $319 from VoIP Etc. |
| 19:39.43 | Silivrenion | thats a VOIP phone |
| 19:39.43 | luke-jr_ | trixter: didn't notice the spoofing stuff till now |
| 19:39.55 | trixter | yeah it requires the source hostname when run |
| 19:39.59 | luke-jr_ | trixter: I guess Ethereal can't tell me anything? |
| 19:40.02 | ManxPower | Ahrimanes, WiFi SIP phones. most people seem to think they suck |
| 19:40.11 | ManxPower | ohiFi SIP phones. most people seem to think they suck |
| 19:40.14 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
| 19:40.23 | trixter | it can only tell you if its that program or not, the id is fixed, the packet is fixed, etc |
| 19:40.54 | luke-jr_ | :/ |
| 19:40.57 | Silivrenion | a phone that costs $200 to interface into whatever wireless networks are nearby to make calls on your house line... |
| 19:41.05 | Silivrenion | sounds iffy |
| 19:41.33 | ManxPower | Silivrenion, I agree. |
| 19:42.16 | Silivrenion | if there are any less expensive phones, it might be a sensible thing |
| 19:42.20 | luke-jr_ | If it moves between access points, that'd be nice |
| 19:42.29 | ManxPower | Silivrenion, get a wired VoIP phone. |
| 19:42.31 | Silivrenion | they do, don't they? |
| 19:42.45 | tronix | gnophone's IAX only, not IAX2, right? |
| 19:42.48 | Qwell | not really |
| 19:42.49 | luke-jr_ | is that possible? ;) |
| 19:42.53 | *** kick/#asterisk [ComPuTeR!i=denon@synapse.subneural.net] by denon (denon) |
| 19:42.53 | *** mode/#asterisk [+b *!*@88.224.162.77] by denon |
| 19:42.59 | denon | (spammer) |
| 19:43.06 | ManxPower | denon, thanks |
| 19:43.19 | denon | np |
| 19:43.30 | Qwell | luke-jr_: this was talked about at ETel...none of the current phones can really switch APs properly (and quickly) |
| 19:43.34 | denon | sorry I didnt see it sooner, wasnt around .. but someone msg'd me |
| 19:43.45 | *** part/#asterisk ibob63 (n=hp@bb-87-82-26-136.ukonline.co.uk) |
| 19:43.53 | Silivrenion | i saw something that makes sense in having a VoIP cell phone that connects back home and interfaces with your home phone line |
| 19:43.58 | luke-jr_ | Qwell: I expect it would need IPv6? |
| 19:44.08 | tronix | guess gnophone's IAX only. that'd explain my issues. ;) (meaning, I really do have a fully working * setup now) |
| 19:44.11 | Qwell | no, it has nothing to do with IP |
| 19:44.21 | Qwell | tronix: try iaxcomm or idefisk |
| 19:44.26 | tronix | np. thanks! |
| 19:44.27 | luke-jr_ | can IAX2/SIP/RTP handle changing IPs? |
| 19:44.46 | Qwell | luke-jr_: You'd get the same IP from two APs on the same LAN |
| 19:44.50 | Qwell | That isn't the issue |
| 19:45.05 | luke-jr_ | I'm not referring to the same LAN... I just assumed that would work |
| 19:45.11 | Qwell | nope |
| 19:45.14 | luke-jr_ | I'm talking about moving from random open APs |
| 19:45.22 | Qwell | no, that won't ever work |
| 19:45.28 | luke-jr_ | ...w/o IPv6 |
| 19:45.36 | ManxPower | I'll stick to stuff that is likely to work, thankyouverymuch |
| 19:45.39 | trixter | voice is less tolerant of delays, most wireless devices take a while to switch, you dont notice it as much on data but with voice where you are streaming often without buffers its a problem |
| 19:46.30 | luke-jr_ | why switch? why not associate w/ all and just use one? |
| 19:46.33 | trixter | random open APs will work with a tunnel so you have a fixed IP or some weird hack to the sip stack to deal with different IPs but odds are the time it takes to do dhcp, set up routing, etc you will lose the call |
| 19:47.00 | Silivrenion | alright, so make calls economically within access points |
| 19:47.29 | luke-jr_ | trixter: that's why you do DHCP on the new AP before you drop the old AP |
| 19:47.49 | luke-jr_ | don't drop the old AP until you begin getting packets from the new one |
| 19:48.18 | denon | easier said than done, when its a crappy network, and everything running the same channels, but not much physical overlap, etc |
| 19:48.31 | luke-jr_ | might need to fix the OS to handle multiple AP association, but single AP association is a bug anyway, IMO |
| 19:48.46 | Silivrenion | what if the ap's use different channels |
| 19:48.56 | luke-jr_ | use both channels at once |
| 19:48.57 | denon | luke-jr_: I dont think most subscription-based networks will like you associating with lots of APs |
| 19:48.58 | Silivrenion | most AP's come on channel 6, but my AP uses channel 8 |
| 19:49.03 | trixter | microsoft has a program where you even get code, cli program that lets you associate with multiple wifi nodes at the same time off one physical card |
| 19:49.12 | trixter | can be AP, ad-hoc or a mix of both |
| 19:49.17 | luke-jr_ | denon: then be smart and only do 1 at a time, and 2 when switching |
| 19:49.28 | denon | even when switching.. |
| 19:49.37 | luke-jr_ | trixter: any idea to do that w/ Linux? |
| 19:49.37 | denon | doesnt matter, I'll never use it :) |
| 19:49.38 | Qwell | If you can switch APs in under 20ms...yeah |
| 19:49.45 | Qwell | that won't happen for some time |
| 19:49.55 | luke-jr_ | Qwell: read what I said? ;) |
| 19:50.13 | luke-jr_ | Qwell: maintain both AP connections until the switch is complete |
| 19:50.26 | trixter | this tool works in windows, dunno if its NDIS so NDIS wrapper could be used, I really havent looked at it |
| 19:50.31 | Qwell | so, you'd need two in the phone, which means battery life will be crap/// |
| 19:50.39 | Qwell | no thanks |
| 19:50.47 | Silivrenion | well either way, are there any affordably priced wifi VoIP phones? |
| 19:50.52 | luke-jr_ | Qwell: no, use one hw to do both APs |
| 19:51.05 | Qwell | luke-jr_: You'd have to switch back and forth MANY MANY times |
| 19:51.11 | Qwell | it doesn't "use both at once" |
| 19:51.13 | trixter | luke-jr_: the problem is you increase the noise floor when you Tx on an adjecent freq, so it can be hard to actually see the other AP until you are so far away that you dont have connectivity |
| 19:51.18 | luke-jr_ | Qwell: why not? |
| 19:51.23 | Qwell | because it doesn't work that way |
| 19:51.26 | znoG | weeeeird, "invalid transfer information" in asterisk console then...... asterisk dies |
| 19:52.09 | Qwell | luke-jr_: it may appear to use both at once, but no, it most certainly does not |
| 19:52.18 | luke-jr_ | trixter: you won't be Tx all the time, just enough for audio... |
| 19:52.20 | ManxPower | znoG, what version? |
| 19:52.21 | Drew___ | anybody know why i only geht hear audio while i am talking while using GSM (G711a is ok) with a GXP2k ?? |
| 19:52.26 | De_Mon | I'm runnin' asterisk 1.2.1 as non-root on debian & the whole system stops responding while running music-on-hold |
| 19:52.27 | Qwell | and when you have something that works very poorly with latency...it won't work |
| 19:52.51 | ManxPower | Drew___, what codecs does the GXP2K support? |
| 19:52.58 | trixter | luke-jr_: the way you Tx, the way the radio Rx, with only one tranceiver its a bit of a problem ... |
| 19:53.09 | Drew___ | i guess it should support gsm |
| 19:53.15 | trixter | and with two different radios you can desensitize them by having em Tx next to each other |
| 19:53.19 | Qwell | Drew___: don't guess...know |
| 19:53.20 | ManxPower | De_Mon, since asterisk not running as root is not allowed to increase it's priority. |
| 19:53.34 | Drew___ | hang on |
| 19:53.34 | ManxPower | see :man nice" |
| 19:54.32 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 19:54.44 | Qwell | jtodd: afternoon |
| 19:55.00 | Drew___ | according to the voip wiki the GXP2k supports GSM - the strange thing is i get audio, but only while i am talking myself |
| 19:55.15 | Drew___ | as soon there is silence on the mic the audio is stopped |
| 19:55.50 | Qwell | Drew___: turn off VAD |
| 19:55.54 | De_Mon | ManxPower that would be a reason it SHOULDN't be doing what it's doing in my book |
| 19:56.30 | De_Mon | prior to the unresponsive system I've got 94% idle cpu |
| 19:56.39 | Drew___ | VAD?? |
| 19:56.47 | znoG | ManxPower: Connected to Asterisk 1.2.1 currently running on ares (pid = 15620) |
| 19:57.05 | Qwell | ~vad |
| 19:57.07 | jbot | well, vad is Voice Activity Detection |
| 19:57.07 | De_Mon | nothing except a bunch of deadlock warnings in the core debug logs |
| 19:57.28 | Qwell | Drew___: aka silence suppression |
| 19:57.30 | Qwell | turn it off on the phone |
| 19:58.19 | Drew___ | silence suppression = off |
| 19:58.37 | Drew___ | correction: silence suppression = no |
| 19:58.59 | ManxPower | Drew___, classic symptom of VAD being enabled |
| 19:59.22 | RoyK | jbot: no, vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 19:59.23 | jbot | okay, RoyK |
| 19:59.24 | Drew___ | it doesnt explain why it only happens with GSM |
| 19:59.28 | RoyK | ~vad |
| 19:59.30 | jbot | i heard vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 19:59.41 | Qwell | RoyK: better.. |
| 19:59.49 | RoyK | :) |
| 20:00.01 | RoyK | ~rfc3389 |
| 20:00.25 | h3x | dammit |
| 20:00.31 | Drew___ | ~rfc2833 |
| 20:00.32 | h3x | when is somebody gonna implement vad on asterisk |
| 20:00.40 | Qwell | h3x: dunno, when? |
| 20:00.46 | Drew___ | rfc2833 is the dtmf thing |
| 20:01.01 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 20:01.22 | Drew___ | maybe its a bug in the BETA firmware... |
| 20:01.24 | RoyK | jbot: rfc3389 is a standard for silence suppression and comfort noise. Asteris does not support this, so please turn off silence suppression or voice activity detection on the client |
| 20:01.26 | jbot | okay, RoyK |
| 20:01.31 | RoyK | ~rfc3389 |
| 20:01.32 | jbot | from memory, rfc3389 is a standard for silence suppression and comfort noise. Asteris does not support this, so please turn off silence suppression or voice activity detection on the client |
| 20:01.43 | RoyK | thanks, jbot |
| 20:02.23 | RoyK | ~lart himself |
| 20:02.35 | *** part/#asterisk Silivrenion (n=admin@unaffiliated/silivrenion) |
| 20:07.38 | *** join/#asterisk WillSip (i=WillSip@200.119.223.246) |
| 20:11.06 | *** join/#asterisk nahirean (n=Amorith@c-68-36-161-8.hsd1.nj.comcast.net) |
| 20:12.04 | znoG | ManxPower: any particular problem with my ver of * ? |
| 20:12.17 | nahirean | anyone familiar with spiura 2002's-3000's? Know a easy dial plan to pass *everything* dialed to Asterisk? |
| 20:12.17 | ManxPower | h3x, Once asterisk does not depend on the incoming stream to time the outgoing stream, then VAD could be done. |
| 20:12.46 | ManxPower | znoG, not that I know of. If it still happens with 1.2.3 then file a bug. Asterisk should never segfault. |
| 20:13.11 | ManxPower | nahirean, Impossible to do on a digit by digit basis. |
| 20:13.28 | znoG | ManxPower: ok thanks, i'll upgrade |
| 20:13.29 | nahirean | Manx, well I have something similar to *xxx |
| 20:13.34 | ManxPower | you can do it with dialplan and timeouts on the SIPUra, but that introduces delay in dialing. |
| 20:13.34 | nahirean | and it's not passing the command to my pbx |
| 20:15.21 | ManxPower | What I do is put the correct patterns for most types of calls on the sipura so they don't need a timeout, then out a catch all on the sipura with a timeout. |
| 20:15.21 | ManxPower | nahirean, then you have an error on the dialplan config on the SIPura. |
| 20:15.32 | nahirean | I most likely do. Let's say all I wanted to do is pass "*" commands to asterisk through the dial plan, it would read something like this, correct?: (*x|*xx|*xxx|*xxxx|) ? |
| 20:16.00 | Qwell | nahirean: That would wait until * and 4 digits were dialed |
| 20:16.15 | Qwell | it'll keep trying until a match is found, or the timeout happens |
| 20:16.37 | tronix | i can call outbound from idefisk (great tip, thanks!) just fine to my 7960 |
| 20:16.43 | tronix | but my 7960 can't call idefisk |
| 20:16.49 | tronix | error message in console is 'no route to host' |
| 20:16.52 | tronix | extensions.conf has: |
| 20:16.55 | tronix | exten => 6976,1,Dial(IAX2/idefisk,20) |
| 20:16.58 | tronix | reloaded and all |
| 20:17.06 | tronix | am I missing something really obvious? |
| 20:17.08 | nahirean | So you're saying it's not passing the data to the pbx? Or? Obviously I am not very familiar with the Sipura dial plan syntax.. |
| 20:17.11 | Qwell | tronix: is it registered? |
| 20:17.16 | tronix | hmm. |
| 20:17.21 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-3.claranet.co.uk) |
| 20:17.22 | Qwell | nahirean: yes, it won't pass it until there is a match |
| 20:17.37 | Qwell | (or there is a timeout) |
| 20:17.39 | nahirean | so if I had exten => *335, etc wouldn't that be a match? |
| 20:17.45 | h3x | ManxPower: with that in mind how would asterisk renegotiate with the endpoints to turn vad on |
| 20:17.49 | WeezeyD | I have two Cisco config questions |
| 20:17.54 | h3x | if it did theroetically support it |
| 20:17.55 | tronix | qwell: should show up in 'iax2 show registry'? if so, doesn't. i'll poke further at that. |
| 20:17.58 | Qwell | nahirean: no, because the SPA will wait until you do *xxxx |
| 20:18.13 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-3.claranet.co.uk) |
| 20:18.17 | nahirean | I see.. |
| 20:18.19 | h3x | er nevermind |
| 20:18.23 | h3x | i read what you said wrong |
| 20:18.40 | h3x | i get it |
| 20:18.41 | Qwell | h3x: VAD support is a SIP header |
| 20:18.43 | tronix | maybe it's a NAT issue for my idefisk problem. hmm. |
| 20:18.57 | WeezeyD | 1) When * dies, my cisco phones don't re-register. They can make calls once it comes back online, but they can't receive without the re-registration. How do I change that? |
| 20:19.12 | Qwell | Weezey: lower the register timeout |
| 20:19.17 | nahirean | is there a resource I can check to view the dial plan syntax of this device? |
| 20:19.24 | nahirean | I've been googling for quite some time |
| 20:19.25 | WeezeyD | 2) is there any way to disable the Missed Calls on the Cisco 7940? |
| 20:19.35 | WeezeyD | Qwell: in * or the Cisco config? |
| 20:19.46 | Qwell | the cisco config |
| 20:20.35 | WeezeyD | Qwell: timer_registration_expires ? |
| 20:20.39 | Qwell | sure |
| 20:20.44 | tronix | Qwell: ah, I see. 'inappropriate auth received; registration refused'. that's a starting point. thanks :) |
| 20:21.39 | nahirean | currently, my configuration is 6+the 11 digits to dial out, and I want to make some * extensions.. can anyone suggest a dialing plan to do that |
| 20:21.40 | WeezeyD | Qwell: it's set to 3600, but it won't re-register if it hasn't seen * for a while. |
| 20:23.20 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.gen.twtelecom.net) |
| 20:23.31 | kuku5 | Qwell: maybe you know someone that does... |
| 20:23.40 | Qwell | kuku5: That does what? |
| 20:23.54 | Qwell | I know a lot of people that do a lot of things |
| 20:23.56 | fiber0pti | Does anyone know where I could find an example of a findme macro that would hold variables for everyone in the office and they can set their cell phone number as well as toggle the find me capabilities on and off? |
| 20:24.17 | Qwell | fiber0pti: there is something on the bug tracker for that |
| 20:24.50 | fiber0pti | Qwell, working? |
| 20:24.57 | Qwell | should work |
| 20:25.15 | kuku5 | Qwell: Im looking to purchase incoming level3 minutes - anyone ? |
| 20:25.22 | Qwell | kuku5: call level3 |
| 20:25.24 | kuku5 | or any other quality |
| 20:25.32 | kuku5 | I dont do 40k$ a month yet |
| 20:25.34 | Qwell | You do realize how much money they want you to spend though, right? |
| 20:25.38 | Qwell | Then you can't do level3... |
| 20:25.49 | Qwell | find another provider |
| 20:25.52 | kuku5 | But I heard people get in groups |
| 20:25.55 | kuku5 | and split it |
| 20:26.42 | kuku5 | Qwell: can you recommend a diffent provider - been using voicepulse but dtmf doesnt always go through correctly. |
| 20:26.57 | fiber0pti | Qwell, do you have to login in order to see it? |
| 20:27.02 | Qwell | fiber0pti: shouldn't |
| 20:27.11 | Qwell | kuku5: try asterlink or nufone |
| 20:27.28 | kuku5 | Didnt nufone die for a few hours a couple of days ago / |
| 20:28.09 | benjk | yes they died right in the midst of ETel |
| 20:28.11 | nahirean | Er.. even with (*xxx) in the dial plan it still rings fast busy and doesnt sent the data to the pbx |
| 20:28.11 | Qwell | yes |
| 20:28.27 | benjk | I had to reconfigure my demo because of it |
| 20:28.40 | nahirean | send* |
| 20:29.16 | tronix | Qwell: figured out... needed auth=md5 in iax.conf. further now, I get incoming call notifications now |
| 20:29.26 | kuku5 | Qwell: I dont see asterlink offering origination |
| 20:29.34 | fiber0pti | Qwell, Since that is an app, does it have to be compiled with asterisk? |
| 20:29.35 | Qwell | benjk: You had to reconfigure your zeroconf demo? |
| 20:29.45 | Qwell | kuku5: they do |
| 20:29.47 | Qwell | fiber0pti: yes |
| 20:30.20 | Qwell | benjk: or was that somebody else? |
| 20:30.42 | WillSip | hi |
| 20:30.59 | WillSip | whats is channels Red Hat |
| 20:31.58 | benjk | yeah, for the call I made during the demo |
| 20:32.09 | benjk | it was supposed to go via NuFone |
| 20:32.12 | trixter | benjk: check your messages :P |
| 20:32.25 | trixter | that network was hosed |
| 20:32.41 | kuku5 | Qwell: nufone is expensive - 2 cents per minute on origination |
| 20:32.55 | tronix | sweet. I've got idefisk working now. looks sharp. my entire * setup is finally done :) now onto to tweaking the dial plan. thanks for the pointers! |
| 20:33.05 | benjk | but so far quality was good to justfy paying a little more |
| 20:33.12 | *** join/#asterisk hickins (n=dtg19@213.186.161.29) |
| 20:35.39 | Qwell | kuku5: That's hardly expensive |
| 20:37.14 | kuku5 | 1.5 is much better |
| 20:37.21 | Drew___ | any ideas on how i can get the gxp2k to _look_ different dependant on a status in the dialplan?? - i.e. agent logged on or DND - are there any commands to change i.e. backlighting, LEDs or a text on the screen??? |
| 20:37.23 | Qwell | no, 1.5 is much cheaper |
| 20:37.43 | Qwell | in other words, less of your calls will actually...you know...work |
| 20:38.06 | *** join/#asterisk areski (n=areski@58.Red-83-55-101.dynamicIP.rima-tde.net) |
| 20:38.28 | kuku5 | hm |
| 20:38.55 | Qwell | Drew___: on an $80 phone? Think again |
| 20:39.05 | *** join/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 20:39.31 | *** join/#asterisk tainted- (n=identd@adsl-71-129-32-116.dsl.irvnca.pacbell.net) |
| 20:39.44 | tainted- | ~seen docelmo |
| 20:39.46 | jbot | docelmo <n=docelmo@55-65.126-70.tampabay.res.rr.com> was last seen on IRC in channel #asterisk, 4d 19h 39m 45s ago, saying: '1300 I paid 1800 a year ago for it.'. |
| 20:39.56 | tainted- | ~seen docelm0 |
| 20:39.58 | jbot | docelm0 <n=docelmo@66.239.192.34.ptr.us.xo.net> was last seen on IRC in channel #asterisk, 4d 2h 54m 25s ago, saying: 'depends on how you look at it..'. |
| 20:39.58 | Drew___ | right... i think i need cisco phones... ^^ |
| 20:40.00 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi25.dialup.mindspring.com) |
| 20:40.21 | Drew___ | anybody got any money for me to buy some real phones?? :) |
| 20:40.37 | Qwell | Drew___: sure, let me just paypal that on over |
| 20:41.05 | Drew___ | :D |
| 20:42.28 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 20:42.36 | Drew___ | since i dont have a credit card i dont have a paypal acount - but i can give you my IBAN ;) |
| 20:42.55 | kuku5 | Qwell: are there any instrctions for * setup on nufone? wall I see is payments. |
| 20:44.29 | WillSip | whats is channels Red Hat |
| 20:44.30 | tronix | they mail it when you sign up |
| 20:44.37 | tronix | (re: nufone setup for * ) |
| 20:44.39 | JohnnyG | what is the best way to handle a lot of old POTS analog phones and a Astericks PBX? Can they be made to work together cheaply? |
| 20:44.48 | fiber0pti | I'm trying to set global vars but I'm not even seening them being set in the CLI |
| 20:45.04 | Aughey | JohnnyG: How many old POTS phones? |
| 20:46.03 | *** part/#asterisk arbius (n=arbius@c-67-173-45-34.hsd1.il.comcast.net) |
| 20:46.08 | tronix | with the IAXy, what does blinking orange LED mean? |
| 20:46.10 | kuku5 | tronix: I dont remeber getting one - just looked fr it and nothing |
| 20:46.20 | tronix | kuku5: i got it... lemme dig up |
| 20:46.47 | cpm | well, it doesn't mean anything good |
| 20:46.53 | tronix | :) |
| 20:47.14 | cpm | I've never seen blinking orange, (thankfully) |
| 20:47.17 | tronix | kuku5: i've found it. if you've got an email address, I can send a copy |
| 20:47.31 | tronix | cpm: iaxy actually works ok for calls, so I'm curious what blinking orange means. weird. |
| 20:48.00 | tronix | when I have calls going on, it's solid |
| 20:48.04 | tronix | otherwise, it just blinks. |
| 20:48.05 | Drew___ | isnt it in the user manual of IAXy? |
| 20:48.20 | tronix | I looked there, don't remember seeing it there |
| 20:48.56 | tronix | nope, not in the user manual. |
| 20:49.05 | tronix | they don't actually state what the indicators mean. :) |
| 20:49.13 | tronix | blue=reg, figured out |
| 20:49.21 | tronix | solid orange=call established, figured out |
| 20:49.27 | tronix | blinking orange=good question |
| 20:49.36 | tronix | (since there's no call attempt in progress) |
| 20:49.40 | tronix | not really a huge deal. |
| 20:49.43 | tronix | just a curiosity. |
| 20:50.27 | cpm | my orange blinks now and again, when the phone is idle, but not regularly |
| 20:50.33 | tronix | I do know that no blue or orange LEDs at all = check power ;) |
| 20:50.42 | tronix | cpm: ahh! could be it. thanks |
| 20:51.39 | JohnnyG | Aughey: 100 |
| 20:52.42 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.47.133.Dial1.Chicago1.Level3.net) |
| 20:52.53 | cpm | Okay, just broke an IAXy out of the wrapper, powered up (with phone and ethernet) and I have a blinking orange. I guess that implies not provisioned |
| 20:53.37 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:53.56 | Aughey | JohnnyG: I'd get a couple of systems configured with 24 port FXS cards. |
| 20:55.43 | fiber0pti | I have a bunch of variables like: cell0686, cell0687 which hold the cell phone numbers of employees. How can I utilize that variable without explicitly calling it, like this: cell${EXTEN} (which doesn't work). Is there someone to eval cell${EXTEN} so that it would return the varable for ${cell0686}, ${cell0687} depending on the extension dialed? |
| 20:56.16 | JohnnyG | Aughey: I've seen these "POTS Banks" that seem to hook into the astericks server, but they only have like 8 ports each. Is it cheaper to upgrade all phones to VOIP phones? |
| 20:56.19 | iCEBrkr | $[cell{$EXTEN}] |
| 20:56.42 | Aughey | JohnnyG: Depends if you have the capability to run Ethernet to everywhere you want a phone |
| 20:56.59 | xachen | this cell phone is costing me too much :( |
| 20:57.14 | fiber0pti | iCEBrkr: hrm.. doesn't work, on the CLI it returns "cell{$EXTEN}" |
| 20:57.17 | Aughey | JohnnyG: A card like this http://www.voipsupply.com/product_info.php?products_id=1158 will give you 24 ports |
| 20:57.35 | Aughey | A couple of cards and/or machines will give you enough lines |
| 20:57.39 | iCEBrkr | Set(var=$[cell{$EXTEN}]) |
| 20:57.56 | BBytes | Heya! I am having a problem dialing on my zaptel channel. It seems the wrong number is dialed. The problem is intermitent, as very rarely it works, but usually it just calls up a poor person's landline. I've tried delaying with w's. Any leads? I suspect it might be the hardware. |
| 20:58.03 | Aughey | I don't know if multiple 24 port cards in the same machine might overwhelm a single PCI bus |
| 20:58.10 | JohnnyG | Aughey: each location that comprises the 100 phones is broadband enabled, but those 100 POTS phones are spread over 6 states |
| 20:58.14 | fiber0pti | iCEBrkr: but I'm setting them at the begining of the dial plan. How can I set all of them then use them later? |
| 20:58.51 | JohnnyG | Digium, Digium, Digium - they own this market :) |
| 20:58.52 | iCEBrkr | fiber0pti: Come up with a clever macro? |
| 20:59.22 | tronix | cpm: ahh, guess that makes sense. thanks :) |
| 20:59.48 | Aughey | JohnnyG: Well you'd have several options. At each site, outfit a box with FXS card(s) to support all the available phones. Then link them all together over the broadband connections |
| 21:00.00 | *** join/#asterisk calvinhp (n=calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 21:00.17 | Aughey | it'd be an interesting project |
| 21:00.28 | tronix | kuku5: thou hast new mail |
| 21:01.12 | JohnnyG | Aughey: it seems like I can do what you suggest or I could send VOIP phones to all locations and have them connect to one central box |
| 21:01.38 | JohnnyG | ballpark, which one would be the cheaper way to go? |
| 21:01.56 | Drew___ | JohnnyG - do your locations already have PBXs? |
| 21:03.06 | JohnnyG | Drew: from what I understand, which is little, they have analog lines coming out from the wall into which phones are plugged. If there is a box powering these lines, it is hidden from site and not administered by us |
| 21:03.50 | De_Mon | anyone have 1.2.3 debs? |
| 21:04.22 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:04.41 | *** join/#asterisk areski (n=areski@58.Red-83-55-101.dynamicIP.rima-tde.net) |
| 21:05.10 | Drew___ | johnnyG - a important aspect is if you need the funcionality of new phones or if the old POTS ones will do for what you need them for |
| 21:05.50 | Drew___ | i.e. if you need multiple lines, caller id display, phonebooks, conferencing, call transfer etc. built into the phone |
| 21:06.34 | JohnnyG | Drew: POTS phones can do what? receive and transfer calls? |
| 21:06.39 | bkw_ | OMG its Qwell |
| 21:06.54 | justinu | like totally |
| 21:07.47 | Drew___ | transfer would work with * and some DTMF codes/extensions |
| 21:07.56 | [TK]D-Fender | JohnnyG : On analog you can all sort : transfers (blind & consultative), hold, 3-wy conferencing, call-waiting for multiple lines (2), etc. Problem is having to power the phone at the location. |
| 21:08.05 | fiber0pti | iCEBrkr: Think that EVAL could be used? |
| 21:08.11 | WeezeyD | in the CDR, what's FAILED mean exactly? The call went through just fine and it gathered billseconds correctly, so shouldn't it say Answered? |
| 21:08.24 | [TK]D-Fender | Drew___ : Not through DTMF.... use an ATA with hook-flash access to SIP funcionality. |
| 21:08.40 | Drew___ | yes dfender - but its a question of userfriendly interface |
| 21:09.00 | Drew___ | fancy new overkill phones or old/ugly POTS ;) |
| 21:09.02 | iCEBrkr | fiber0pti: It's possbile. |
| 21:09.04 | [TK]D-Fender | Drew___ : What interface? Setting u the ATA or its use once in service? |
| 21:09.13 | Drew___ | sure |
| 21:09.23 | iCEBrkr | fiber0pti: I'm just guessing here, I haven't really dug into trying what you're attempting to get working |
| 21:09.28 | fiber0pti | iCEBrkr: I have eval successfully setting a var to the name of the var I want to use but am unable to actually get the contents of that var returned |
| 21:09.43 | [TK]D-Fender | Drew___ : that a eXclusive OR (XOR) |
| 21:09.45 | [av]bani | ... |
| 21:09.45 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
| 21:10.04 | [TK]D-Fender | [av]bani : So hows the testing going? |
| 21:11.15 | Drew___ | i guess it is XOR |
| 21:11.46 | fiber0pti | [TK]D-Defender: maybe you can help me out. I have a bunch of variables cell(extension) (like: cell0686) but I want to use them dynamically depending on which extension was dialed. I'm currently trying to use eval to set a variable to the name of the variable i want to use (i.e. cell0686) but can't get the contents of cell0686 to return. Any ideas? |
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| 21:12.20 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi5a.dialup.mindspring.com) |
| 21:12.58 | Drew___ | dfender - is there any possibility get a extension to simulate being busy? |
| 21:13.11 | [TK]D-Fender | fiber0pti : Give me a specifcs case sample |
| 21:13.23 | JohnnyG | thanks Drew and Aughey for your time and answers |
| 21:13.28 | JohnnyG | I'm off to ride bikes with the gf :) |
| 21:13.34 | [TK]D-Fender | Drew___ : what does SIMULTAING busy imply? |
| 21:13.47 | Drew___ | johnny - hf |
| 21:13.59 | fiber0pti | D-fender: someone dials extension 0686 and then asterisk will dial their cell phone from variable cell0686. Same with cell0687, etc... |
| 21:14.29 | [TK]D-Fender | fiber0pti : SCrew variables.. they die at the end of calls. Use the ASTDB |
| 21:14.41 | fiber0pti | hmm.. ok.. |
| 21:15.06 | [TK]D-Fender | fiber0pti : I had 2 major implementation of forwarding based on ASTD stuff... thats the way to go. |
| 21:15.44 | [TK]D-Fender | fiber0pti : Do you want a disgustingly big sample? |
| 21:15.52 | fiber0pti | D-fender: sure |
| 21:15.56 | Drew___ | i just loaded the new beta firmware on the GXP2k phone i have - it supports BLF status lights - i would like to implement a autoattend on/off feature - to show the status of this setting i would like to use a BLF LED - they are configured to show the status of a extension - so i need a extension to simulate being busy for the LED to light up |
| 21:16.10 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 21:16.40 | fiber0pti | D-Fender: so once I get the values stored in the db, how do I pull them out dynamically? |
| 21:16.45 | Drew___ | maybe _simulating_ is the wrong word... i just want to use the LED ;) |
| 21:16.46 | [TK]D-Fender | fiber0pti : http://pastebin.com/529320 |
| 21:17.01 | [TK]D-Fender | fiber0pti : look at my sample |
| 21:17.04 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 21:17.29 | [TK]D-Fender | fiber0pti : The STDEXTEN pulls values from ASTDB that are set in the macro's and contexts below it. |
| 21:17.34 | [av]bani | [TK]D-Fender: i have sipura/snom working fully now, just need to add polycom |
| 21:18.09 | [av]bani | [TK]D-Fender: btw, i found why the spa3k does that weird volume wavering thing |
| 21:18.56 | [TK]D-Fender | Drew___ : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate |
| 21:19.10 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 21:19.17 | [TK]D-Fender | [av]bani : Yay.. enlightnement... so why does it do that? |
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| 21:20.26 | [av]bani | [TK]D-Fender: in pstn line turn 'echo suppress enable' off. it's sipura's baseball bat method of dealing with echo |
| 21:20.28 | [TK]D-Fender | [av]bani : Oh and the GF flipping out over a bad day trying to place a call had me tear down my * install here down to just the 2 phones on my desk which are no longer hooked to our main line. Looks like I'll just be running another number for it. |
| 21:20.48 | [av]bani | [TK]D-Fender: they cut incoming audio by 12db while you speak when that's enabled |
| 21:20.52 | [TK]D-Fender | [av]bani : So the price is getting more echo? |
| 21:20.54 | *** join/#asterisk nahirean (n=Amorith@c-68-36-161-8.hsd1.nj.comcast.net) |
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| 21:21.12 | [TK]D-Fender | [av]bani : sick form of cheating... |
| 21:21.16 | nahirean | Hello folks |
| 21:21.50 | [av]bani | [TK]D-Fender: yeah, the reason it wavers while you hear remote ringing is because the spa3k hears background noise in your handset and keeps kicking in the 12db muffler. |
| 21:22.18 | [av]bani | [TK]D-Fender: their echo canceller is pretty poor, so that's their method of dealing with it -- bashing it with a baseball bat |
| 21:22.45 | litecode | can the voicemail "sound" files be stored in a mysql database? |
| 21:22.47 | [TK]D-Fender | [av]bani : delicate.... I like it! Then again for $100 getting 1 FXS, 1 FXO is pretty good.. shouldn't complain.... |
| 21:23.09 | [TK]D-Fender | [av]bani : If I wanted "real" PSTN stuff I'd go with an A200 :) |
| 21:23.40 | SibRphrek | how do internatinoal calls work with asterisk? |
| 21:23.43 | [av]bani | [TK]D-Fender: i disabled it and now calls are constant volume, no wavering. you can _just_ hear some echo, but i was able to hear a little echo on PRI lines too so i'm not too disappointed |
| 21:24.36 | mattwj2005 | so what do you guys like for voip service providers? |
| 21:25.26 | [av]bani | [TK]D-Fender: i've been pondering a mediatrix or cisco fxo gateway |
| 21:26.01 | nahirean | Is zaptel an absolute requirement for using Background and Playback or can it be avoided without having to purchase a card? =] |
| 21:26.15 | AndyCap | nahirean: ztdummy module for timing? |
| 21:26.30 | mattwj2005 | this would be a residental setup |
| 21:26.47 | nahirean | Well I just wanted to use the Background/Playback commands.. they aren't functioning and I assumed it was due to the lack of ztdummy |
| 21:27.15 | nahirean | The console is complaining about a file not found, but i've double checked and triple checked, and it is in the directory I am specifying |
| 21:27.54 | mattwj2005 | I was thinking of broadvoice byod for incoming and voip jet for out going......but would it better to have broadvoice unlimited.....or is there a better solution out there? |
| 21:28.20 | [TK]D-Fender | I don't hear echo on my PRI lines :D |
| 21:28.21 | [av]bani | [TK]D-Fender: so i take it no more fxo @ home? |
| 21:28.27 | *** join/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
| 21:28.45 | [TK]D-Fender | [av]bani : Cisco or Mediatrix? What port density do you need? |
| 21:28.50 | [av]bani | 4 |
| 21:29.07 | [TK]D-Fender | [av]bani : Nope. Now I'm running my * at hom off my work PRI alone instead of PRI + 1 :) |
| 21:29.08 | [av]bani | i'm just supposing cisco and mediatrix have better echo cancellers |
| 21:29.29 | [TK]D-Fender | [av]bani : Screw Cisco/Mediatrix. Get an A200 :) |
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| 21:29.34 | [av]bani | a200? |
| 21:29.49 | [TK]D-Fender | nahirean : Don't need Zaptel to playback sounds.... |
| 21:30.01 | [TK]D-Fender | [av]bani : Sangoma's new analog card. |
| 21:30.10 | [av]bani | oh, the $1600 one |
| 21:30.25 | nahirean | TK: Odd.. I am using Background(custom/<filename>) and it's saying the file doesn't exist, however it's in the directory |
| 21:30.28 | [TK]D-Fender | [av No, FAR cheaper.. for 4 ports... |
| 21:30.43 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:30.52 | nahirean | I've also tried the full path |
| 21:30.52 | [TK]D-Fender | nahirean : If it says its not there, then you named something wrong. It doesn't lie.... |
| 21:31.08 | [TK]D-Fender | nahirean : make sure there is a proper EXTENSION on it. |
| 21:31.15 | [av]bani | [TK]D-Fender: still, an external fxo lets us separate the pbx from the phoneroom |
| 21:31.21 | [TK]D-Fender | if its a GSM file it'd better end with ".gsm" |
| 21:31.23 | nahirean | there is no extension, i wasnt aware that one was mandated |
| 21:31.48 | [TK]D-Fender | [av]bani : There is taht... if its a deciding factor, sure go with Mediatrix or someone like that |
| 21:31.53 | [av]bani | [TK]D-Fender: and if the pbx pc dies, we can still place calls direct from the sip phones to the fxo's |
| 21:32.04 | ManxPower | nahirean, the actual filename should have an extension. when you call it in the dialplan (playback, backghround, etc) it should NOT be specified with a filename |
| 21:32.12 | [av]bani | [TK]D-Fender: http://www.voicetronix.com/vpb4_v4pci.htm <- look reasonable? |
| 21:32.36 | nahirean | ManxPower, I see, thank you |
| 21:32.39 | [av]bani | [TK]D-Fender: with the a200 you need to buy an echo canceller separately. else you need to use zaptel's i guess, which is crap |
| 21:32.40 | nahirean | TK: Thank you as well |
| 21:32.57 | [TK]D-Fender | [av]bani : Yeah external gateways are nifty ideas. they can be good, but read the reviews on them really well first. Some are shittier than you might suspect. |
| 21:33.42 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 21:33.47 | [TK]D-Fender | [av]bani : The Voicetronixs stuff is old news and shouldn't be touched with a 10' pole. its only barely supported in * as it is. Don't paint yourself into a corner... A200 is a considerably better choice for most applications... |
| 21:34.01 | [av]bani | [TK]D-Fender: a200 isnt out yet... |
| 21:34.18 | [av]bani | 'coming may 2006' or something |
| 21:34.37 | *** join/#asterisk sdgusler-M33 (i=WebChat@i.think.napoleon.dynamiteblows.com) |
| 21:34.46 | [TK]D-Fender | [av]bani : is now :)_ |
| 21:34.48 | [av]bani | kind of hard to buy a card which isnt shipping :) |
| 21:35.21 | ManxPower | and stupid to buy before you hear about it from people that were in a hurry and bought it as soon as it shipped. |
| 21:35.22 | [TK]D-Fender | Please Note: VOIPSupply is currently taking pre-orders for the new Sangoma A200 Remora series cards. These are anticipated to ship around 01/27/06 |
| 21:35.45 | [TK]D-Fender | Looks like last Friday to me..... |
| 21:35.48 | [av]bani | so i'm curious how you know its any good if it doesnt exist yet :) |
| 21:35.56 | justinu | did anything actually ship? |
| 21:36.08 | SibRphrek | [TK]D-Fender: hey man - question for you about international calls...are there special setups needed for calling cards? |
| 21:36.31 | [TK]D-Fender | [av]bani : they exist and should now be shipping. as for the quality, its pretty established as to how they designed it and from my experience with their A104d I'd say it'll be Godly... |
| 21:36.45 | [TK]D-Fender | SibRphrek ..... huh?! |
| 21:37.00 | SibRphrek | calling cards in asterisk - any sort of special setup? |
| 21:37.17 | [TK]D-Fender | SibRphrek :Depends what you mean... using, and treating * as a CC center... |
| 21:37.25 | [TK]D-Fender | s/and/or |
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| 21:38.00 | [av]bani | [TK]D-Fender: the 104d is pri though... this is sangoma's first fxo/fxs ... |
| 21:38.00 | SibRphrek | [TK]D-Fender: no - i wanna have clients use calling cards to call international....does asterisk need a special setup for that? |
| 21:38.14 | Aughey | anything special I need to turn on to get MWI to work? It's not working with my GXP-2000 |
| 21:38.42 | hardwire | voip-info.org |
| 21:38.42 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:38.51 | areski | SibRphrek, u can have a look at a2billing |
| 21:38.59 | [TK]D-Fender | [av]bani : the A200 is build on the same base card as their PRI's and use the same DSP for EC. Its just that they break it out to analog in their sandwich cards. So basically its a PRI + Channelbank in a card w/ec, etc.... |
| 21:39.00 | areski | SibRphrek, or astcc |
| 21:39.03 | ManxPower | in the sip.conf entry for that phone (friend or peer) put mailbox=nailboxnum@mailboxcontext |
| 21:39.25 | [av]bani | [TK]D-Fender: i'm wondering if the cisco EC is any good. i'm guessing its ok |
| 21:39.28 | Aughey | never mind, I got it |
| 21:39.32 | [TK]D-Fender | [av]bani : thats the AFT base showing through.... |
| 21:39.33 | Aughey | dumb mistake |
| 21:39.53 | [TK]D-Fender | [av]bani : Could be.. I never really heard anything about them though. The name gives some hope though... |
| 21:39.57 | SibRphrek | areski: isn't astcc part of the addons? i coulda sworn i installed those but i don't see astcc anywhere |
| 21:40.10 | [av]bani | [TK]D-Fender: as in 'likely not entirely shit' ? |
| 21:40.16 | areski | SibRphrek, Yes it s |
| 21:40.26 | [TK]D-Fender | [av]bani : A distinct possibility! |
| 21:40.30 | areski | SibRphrek, u can catch it frm svn |
| 21:40.42 | SibRphrek | areski: huh? |
| 21:40.46 | [av]bani | [TK]D-Fender: one can snag used cisco gateways and fxo's from ebay pretty cheaply... |
| 21:40.48 | areski | SibRphrek, I am the creator of a2billing |
| 21:40.52 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 21:41.01 | SibRphrek | areski: and what is a2billing? |
| 21:41.08 | areski | SibRphrek, if u want to look areski.net/a2billing |
| 21:41.17 | areski | SibRphrek, it s a callingcard platform :) |
| 21:41.29 | areski | SibRphrek, it wasnt u was looking for |
| 21:41.31 | [TK]D-Fender | [av]bani : Confirm support & licensing and it might be a good deal. If you're becoming the test victim, let us know how it turns out :) |
| 21:41.49 | SibRphrek | i am looking for something to allow calilng card calls |
| 21:41.50 | areski | SibRphrek, there is some others too that u can find via voip-info.org |
| 21:41.52 | [av]bani | [TK]D-Fender: you can 'find' ios like you can 'find' polycom firmware though :) |
| 21:41.56 | SibRphrek | so people can call internatinoal |
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| 21:42.14 | z-killz | moin |
| 21:42.15 | areski | SibRphrek, so u need termination with a voip provider |
| 21:42.17 | SibRphrek | so a2billing is like a calling card in itself |
| 21:42.24 | [av]bani | the devices i'm talking about have been EOLd which is why they go cheaply.. they should still work though |
| 21:42.44 | [av]bani | still, there's the yuckiness of interfacing with analogue PSTN at all... :( |
| 21:42.57 | [TK]D-Fender | [av]bani : for 4 port w/EC an A200 costs the same as the Mediatrix w/4 ports. However the A200 scales better from there, and I know the EC on it is great |
| 21:43.00 | areski | SibRphrek, it s a software that will make your life easier if u want to sell callingcard, manage termination et.c.. |
| 21:43.01 | SibRphrek | areski: i see you have your report's being exported in .csv - i'd like to talk to you about that privatly please |
| 21:43.18 | [av]bani | [TK]D-Fender: you can buy mediatrix off ebay used for $160-$200 :) |
| 21:43.24 | areski | SibRphrek, no prob |
| 21:44.00 | [TK]D-Fender | [av]bani : You don't ahve to "find" Polycom firmware, just ask an authorized reseller. Its not like they are obliged to sell you anything to give it away :) |
| 21:44.16 | [TK]D-Fender | [av]bani : Well used is used.... plenty of bargains to be had..... |
| 21:44.32 | cpm | [TK]D-Fender, please give me firmware. |
| 21:44.37 | cpm | :) |
| 21:44.54 | [av]bani | [TK]D-Fender: some have outright refused to give polycom 'unless you are a customer'. i did find one who did hand it out though |
| 21:45.56 | [av]bani | you have spa941's at home right? |
| 21:46.13 | ManxPower | one might assume, that if you go even a little ways up the polycom sales food chain, you'll find an authorized dealer and get get it with some work. |
| 21:46.23 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 21:46.30 | [TK]D-Fender | [av]bani : Yup, a 941 & 600 on my desk |
| 21:46.42 | ManxPower | the company we buy polycoms from is becoming an authorized reseller because of us. |
| 21:47.09 | *** join/#asterisk wiredBOT (n=d2D@adsl-80-40-78.sdf.bellsouth.net) |
| 21:47.11 | [av]bani | ManxPower: it's still a retarded policy. polycom isnt selling firmware like cisco does |
| 21:47.12 | [TK]D-Fender | ManxPower : I'm going for my cert and am starting a VoIP consultancy soon. |
| 21:47.16 | cpm | Since polycom sells at Office Depot, or Staples or whatever, they really should source-support their products, as they retail though 'big box' retailers. |
| 21:47.23 | justinu | what cert? |
| 21:47.31 | wiredBOT | hello |
| 21:47.35 | ManxPower | [av]bani, You are correct. It's one we are willing to live with because of the phones. |
| 21:48.04 | [TK]D-Fender | ManxPower : Amen! Our shit is WAY better than their crap! |
| 21:48.04 | [av]bani | i'm still waiting for polycom to put a backlight on the fuckers |
| 21:48.15 | [TK]D-Fender | [av]bani : You and the rest of the world... |
| 21:48.26 | [av]bani | someone needs to cluebat them |
| 21:48.43 | ManxPower | Polycom has a few issues/annoyances, but they are still better than some others. |
| 21:48.55 | Nivex | cisco has no backlight, and they're doing ok |
| 21:49.00 | cpm | heh |
| 21:49.01 | ManxPower | Cisco and SNOM are the only ones I know that come close to a Polycom. |
| 21:49.06 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfi5a.dialup.mindspring.com) |
| 21:49.20 | ManxPower | and each have their issues. |
| 21:49.32 | [av]bani | ManxPower: as does polycom. 3 minute boot :/ |
| 21:50.06 | ManxPower | [av]bani, That depends on a few things. |
| 21:50.24 | ManxPower | Polycom's issues were ones we could live with, Cisco's and SNOM's we were not. |
| 21:50.36 | [av]bani | what snom issues did you have? |
| 21:50.48 | cpm | ManxPower, You mean a product that works vs one that might work? |
| 21:51.03 | SibRphrek | areski: you get my /msg? |
| 21:51.13 | areski | SibRphrek, yes |
| 21:51.27 | areski | SibRphrek, and u my answers ? |
| 21:51.29 | areski | :) |
| 21:51.31 | SibRphrek | no |
| 21:51.33 | SibRphrek | no answers |
| 21:52.47 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:52.52 | p0g0_ | Hi I'm trying to route two PSTN POTS lines connected to Zaptel FXO X100P's (clones) to L1/L2 on unlocked Sipura SPA-2002s via asterisk. Can anyone point me to example working asterisk configuration files and dialing rules for such a layout (vanilla dialing in the U.S.)? |
| 21:53.16 | [TK]D-Fender | ok, GTG for now.... back later |
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| 22:04.33 | exonic2 | Hey there fellas |
| 22:05.18 | exonic2 | I am attempting to implement time based call forwarding, it's not too difficult since i'm using realtime xtensions, but saving the CDR's for both the inbound call and the forwarded call is proving to be quite difficult |
| 22:05.26 | *** join/#asterisk jr_ewing (n=jeanmaro@d83-179-134-77.cust.tele2.fr) |
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| 22:06.01 | exonic2 | If I use the forkCDR() application, that will save the inbound, but as far as the outbound goes i'm not sure what do to |
| 22:06.17 | jr_ewing | hi all |
| 22:06.38 | exonic2 | Hey there |
| 22:06.49 | exonic2 | not very active today |
| 22:06.52 | jr_ewing | Someone hass already installed Nufone h323 on fedora core 4 ? |
| 22:06.59 | justinu | exonic2: asterisk cdrs pretty much suck |
| 22:07.26 | exonic2 | justinu, Yes. so I've found. |
| 22:07.40 | exonic2 | justinu, How does one recommend doing it though? |
| 22:08.02 | justinu | good question |
| 22:08.11 | justinu | i'd like the answer to that myself |
| 22:08.17 | exonic2 | :) |
| 22:09.00 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:10.21 | *** part/#asterisk jr_ewing (n=jeanmaro@d83-179-134-77.cust.tele2.fr) |
| 22:13.36 | nahirean | hey folks, how would you make exten => 1,2,Read([var]) assign this variable an extension as well? I am trying to use the record command to then do: Record(custom/${PHRASEID}) but asterisk errors saying there's no extension specified for audio file |
| 22:13.57 | nahirean | would it be :gsm? |
| 22:15.36 | nahirean | yep =] |
| 22:16.43 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:18.08 | *** join/#asterisk mononukeleosis (n=d2D@12-202-40-92.client.insightBB.com) |
| 22:18.51 | tronix | justinu: curious -- how do the CDRs suck? |
| 22:19.03 | justinu | let me count the ways |
| 22:19.07 | tronix | uh oh :) |
| 22:19.07 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-53-16.zoominternet.net) |
| 22:19.30 | justinu | 1) if you change the outgoing caller id, there's no native way to preserve the inbound |
| 22:20.57 | mononukeleosis | what are some good providers that allow setting outbound cid |
| 22:21.13 | tronix | justinu: interesting. good to know, thanks. I'm just now starting to look at the CDR side of things |
| 22:21.40 | justinu | 2) you can only see the last thing a call did in the dialplan |
| 22:22.32 | mononukeleosis | nufone ,voipjet, iax.cc etc,, |
| 22:22.48 | tronix | ahh. (dialplan) |
| 22:23.16 | SwK | mononukeleosis, asterlink |
| 22:23.46 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 22:23.48 | SwK | tronix, if you want 2 CDRs per call and them to be what you really want, better get to coding |
| 22:23.54 | mononukeleosis | thanks |
| 22:24.39 | tronix | Swk: :) |
| 22:25.02 | mononukeleosis | im trying to compile a list |
| 22:26.27 | mononukeleosis | not to feed the skiddies |
| 22:28.28 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:30.36 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 22:33.00 | tronix | what's the best way to see what codecs are loaded in *? e.g. I installed g729a but don't know how to determine if it's loaded? |
| 22:33.06 | *** part/#asterisk mononukeleosis (n=d2D@12-202-40-92.client.insightBB.com) |
| 22:33.15 | tronix | I'd have thought 'show codecs' but that's not quite it. |
| 22:33.17 | franck | tronix: show convertion I think |
| 22:33.22 | justinu | show translation |
| 22:33.31 | tronix | ahh! thanks |
| 22:33.35 | franck | ^^^ thats the one! |
| 22:33.53 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:34.27 | tronix | :) |
| 22:36.27 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:36.27 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:38.20 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 22:38.26 | jhiver | Hiya all |
| 22:38.33 | z-killz | lo |
| 22:38.50 | franck | for pickup group I can put any number I like, I don't have to be careful with extensions numbers? |
| 22:39.04 | jhiver | has anybody tried running Asterisk inside virtualization software such as VMWare ESX server or Xen? Any experience to share? |
| 22:41.06 | jhiver | I guess either a) everybody's dead or asleep b) nobody tried :) |
| 22:41.35 | iDunno | there's a possibility of c) |
| 22:41.41 | iDunno | they're reading things elsewhere. |
| 22:42.00 | jhiver | yeah :) |
| 22:42.12 | jhiver | I thought that was part of a) asleep :) |
| 22:42.26 | gaupe | jhiver: just trying :) |
| 22:42.40 | jhiver | gaupe? how is it coming? |
| 22:43.14 | gaupe | just got xen3 installed, don't know how long I'm working with it tonight |
| 22:43.32 | gaupe | but I do not have high hopes, asterisk is quite timing sensitive |
| 22:45.11 | tronix | jhiver: folks have run * on UML |
| 22:45.23 | tronix | which works unless host is under load |
| 22:45.31 | tronix | not recommended, though, since timing sensitive |
| 22:47.05 | tronix | I'm honestly surprised anybody on a machine with ~20-30 running UML instances got it to work somewhat reliably at all ;) |
| 22:47.30 | trixter | I do that for breakfast |
| 22:48.34 | tronix | it's one thing if the timing source was a hardware clock, or was in a real time OS instance, or a single user workstation, but something like 20-30 instances... |
| 22:49.07 | tronix | whereas any temporary cpu hog in any one of the instance is enough to call calls to fall apart due to timing |
| 22:49.14 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 22:51.00 | jhiver | tronix, that's interesting |
| 22:51.11 | jhiver | well user mode linux is very slow isn't it |
| 22:51.57 | tronix | xen was benchmarked to be 3x faster on the same hw |
| 22:51.59 | tronix | (fwiw) |
| 22:52.03 | jhiver | Xen is supposed to do isolation as well |
| 22:52.25 | jhiver | so if one virtual instance goes mad it ought not to crash all the rest... but I'm not sure about this |
| 22:52.37 | tronix | not sure it provides good real-time guarantees |
| 22:52.50 | jhiver | I guess I have to roll up my sleeves and give it a try... not for tonight though :) |
| 22:52.54 | tronix | :) |
| 22:53.26 | jhiver | it will be interesting to play some MOH while trashing the disk in another Xen instance and doing some computational expensive stuff in another |
| 22:53.34 | tronix | indeed |
| 22:54.18 | jhiver | ideally I would like to combine SER + Asterisk (for LCR) + Asterisk (for IVR) + MySQL, all in one box |
| 22:54.47 | jhiver | I've actually ordered the hardware so I'll have to make it work or go with VMWare I guess :/ |
| 22:55.19 | jhiver | but VMWare costs like as much as the server itself it's really stupid :) |
| 22:56.01 | tronix | if it comes to the worst, hack in some hard real-time guarantees for the xen clock. ;) |
| 22:56.23 | *** join/#asterisk rene- (i=rene@201.144.60.114) |
| 22:56.51 | jhiver | I think this is outside the scope of my immediate skills :) |
| 22:57.11 | jhiver | I would need like really big sleeves... |
| 22:57.22 | jhiver | that would be a lot of rolling up to do :) |
| 22:57.27 | z-killz | hm highlight |
| 22:59.01 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:00.24 | jhiver | ok, found this: |
| 23:00.28 | jhiver | http://wiki.xensource.com/xenwiki/Scheduling |
| 23:00.42 | jhiver | looks like there are many schedulers available for Xen |
| 23:00.55 | jhiver | some of them being real-time... looks good |
| 23:02.32 | jhiver | ok... off to bed. Speak to you later! cya |
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| 23:07.08 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
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| 23:22.33 | *** join/#asterisk Drew___ (n=foo@zux221-065-169.adsl.green.ch) |
| 23:23.24 | Drew___ | i registerd my number at E164.org but it seems as if * cant find my record |
| 23:23.45 | Drew___ | enumlookup always fails |
| 23:23.48 | Drew___ | any ideas? |
| 23:24.03 | *** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net) |
| 23:24.45 | *** join/#asterisk draga (i=draga@host209-235.pool8253.interbusiness.it) |
| 23:25.03 | draga | hello everybody, I'm trying to use voipstunt on my asterisk for calling |
| 23:25.07 | draga | I get this message: |
| 23:25.15 | draga | *CLI> Jan 30 00:23:53 NOTICE[24091]: chan_sip.c:1985 auto_congest: Auto-congesting SIP/voipstunt-1bf9 |
| 23:25.20 | draga | what does it mean? |
| 23:27.40 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 23:28.05 | *** join/#asterisk JCC_ (n=chatzill@207.41.92.131) |
| 23:39.48 | Drew___ | what does a ENUM DNS entry look like? |
| 23:41.11 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 23:41.31 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:41.32 | trixter | http://www.itu.int/osg/spu/enum/ |
| 23:42.01 | trixter | http://www.ietf.org/rfc/rfc2916.txt it has examples |
| 23:42.15 | tronix | for a real world one: |
| 23:42.30 | tronix | host -t any 4.7.2.5.3.2.7.2.9.6.9.4.e164.arpa |
| 23:42.53 | tronix | 4.7.2.5.3.2.7.2.9.6.9.4.e164.arpa NAPTR 100 10 "u" "e2u+sip" |
| 23:43.00 | tronix | "!^.*$!sip:274@denic.de!" . |
| 23:43.01 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:43.17 | tronix | that's from the phone number used in DENIC's ENUM announcement. :) |
| 23:43.45 | tronix | (serving Germany / Deutschland, that is) |
| 23:45.23 | tronix | also another nice place with various ENUM-related info: |
| 23:45.25 | tronix | http://www.denic.de/en/enum/index.html |
| 23:46.59 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:47.48 | Drew___ | at least the lookup succeeds with that DENIC test number |
| 23:48.07 | Drew___ | i registerd my number with e164.org but it doesnt seam to work |
| 23:48.12 | tronix | hmm. |
| 23:48.26 | Drew___ | it gives me a DNS parse error |
| 23:49.30 | Drew___ | +41 32 510 6147 - 7.4.1.6.0.1.5.2.3.1.4.e164.org |
| 23:50.41 | tronix | works ok for me, btw |
| 23:50.43 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 23:50.52 | tronix | <PROTECTED> |
| 23:50.52 | tronix | 7.4.1.6.0.1.5.2.3.1.4.e164.org has NAPTR record 100 10 "u" "E2U+SIP" "!^\\+41325106147$!sip:41325106147@adrianraez.dyndns.org!" . |
| 23:50.58 | tronix | dns-wise, at least. |
| 23:51.33 | Drew___ | strange |
| 23:51.53 | tronix | you using older 'host' utility or something? I'm using BIND 9 tools |
| 23:51.53 | Drew___ | do i need to wait for the DNS records of my ISP to update? |
| 23:52.00 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:52.04 | tronix | how long ago was it put in? |
| 23:52.37 | Drew___ | 45min ago |
| 23:52.43 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 23:52.49 | tronix | ahh yeah, I'd guess probably needs a little more time |
| 23:52.49 | *** join/#asterisk watchy (n=watchy@70.238.56.18) |
| 23:52.57 | tronix | most places usually has a min of 1h |
| 23:53.00 | watchy | how do you reset a 7960g without unpluging? |
| 23:53.06 | Qwell | watchy: sip or sccp? |
| 23:53.08 | watchy | sip |
| 23:53.14 | Qwell | *+6+settings |
| 23:53.17 | Qwell | ctrl-alt-del style |
| 23:53.32 | watchy | i had been doing it like 400 times last week |
| 23:53.35 | watchy | but forgot what it was |
| 23:53.39 | Qwell | :p |
| 23:53.39 | tronix | :) |
| 23:53.43 | Drew___ | right - ill try again over the hour - maybe my ISP will have crond it by then... |
| 23:53.50 | watchy | what do you push on reboot to reset defaults? |
| 23:54.11 | Qwell | You can do that from the settings menu |
| 23:54.21 | Qwell | now, if you mean factory reset, that's a different story |
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| 23:56.00 | *** join/#asterisk ibob63 (n=hp@bb-87-82-26-136.ukonline.co.uk) |
| 23:58.02 | *** join/#asterisk sk4tedad (n=chatzill@206-248-153-29.dsl.teksavvy.com) |
| 23:58.02 | ibob63 | I am thinking of using AMP for my asterisk GUI. Does anyone have opinions of these software? |
| 23:58.14 | ManxPower | ~amp |
| 23:58.16 | jbot | i heard amp is NOT supported here! people using it should join #amportal |
