| 00:04.16 | pabelanger-lap | bougyman: What version of Asterisk has you tested with? |
| 00:12.02 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 00:12.29 | *** part/#asterisk thansen (~thansen@67.199.179.78) |
| 00:12.56 | Beltechs | 1.6 |
| 00:13.53 | Micc_ | how stable is 1.8? |
| 00:18.18 | Micc_ | I'd love to start playing with it, but I don't have a free server at the moment. |
| 00:18.18 | Micc_ | I want to play with SRTP. |
| 00:22.01 | *** join/#asterisk Mhaddog (~Mhaddog@173-133-0-97.pools.spcsdns.net) |
| 00:28.13 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 00:31.42 | *** join/#asterisk xpot-mobile (~james@c-98-202-72-167.hsd1.ut.comcast.net) |
| 00:43.09 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 00:50.36 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 00:53.19 | *** join/#asterisk [Outcast] (~anonymous@24-183-177-242.dhcp.oxfr.ma.charter.com) |
| 01:23.48 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 01:36.57 | *** join/#asterisk adolfomaltez (~taro@190.87.103.192) |
| 01:40.08 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 01:41.34 | *** join/#asterisk shapr (~shapr@c-76-29-246-212.hsd1.al.comcast.net) |
| 02:09.53 | *** join/#asterisk shido6 (~shido6@c-24-4-217-126.hsd1.ca.comcast.net) |
| 02:23.34 | *** join/#asterisk OrNix (~ornix@host89-251-107-50.hnet.ru) |
| 02:46.16 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:46.34 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:48.15 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:48.35 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 03:07.26 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 03:08.28 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:08.28 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 03:11.06 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 03:13.07 | leifmadsen | Curious if someone is good with math here. I want to graph some general numbers to see the numbers of calls I'd have over a period of time if I had a value of Calls Per Second (CPS), Length of each call (in seconds), along with a total duration (in seconds). |
| 03:15.28 | *** join/#asterisk LemensTS (~LemensTS@70.238.154.222) |
| 03:16.56 | LemensTS | am i safe to just remove voicemails in /voicemail/default/users/INBOX/.txt & .WAV & .wav ? or is there also data stored in a db somewhere? |
| 03:28.52 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:29.17 | golikwid|mac | cant you just use the ari to remove them |
| 03:42.25 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 03:42.48 | LemensTS | golikwild: you mean ami? |
| 03:45.12 | LemensTS | i dont see a ami voicemail deletion cmd. I wrote my own voicemail portal like FPBX has but mine is for Asterisk 1.6...I was just going to use PHP unlink to remove the voicemail files when the user wants to remove, didnt know if there would be a trail left in a database or something |
| 03:49.00 | *** join/#asterisk philipp64|laptop (~chatzilla@71-38-140-211.ptld.qwest.net) |
| 03:53.37 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 03:59.51 | gamedna | has anyone tried asterisk on amazon EC2? |
| 04:01.34 | JerJer | gamedna: it works. expensive imho |
| 04:01.56 | gamedna | what about their IP assignments? |
| 04:02.08 | gamedna | i hear that they are dynamic ,and problematic for sip connecitons |
| 04:08.39 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 04:08.54 | LemensTS | Apparently asterisk just looks at the txt files in the voicemail directory because i deleted the files and now it shows only 2 messages in the cli for that user instead of 3... |
| 04:11.35 | JerJer | gamedna: they are dynamic in the sense of you will get a new ip for each instance, but you keep that ip for as long as your instance is running |
| 04:12.06 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.55) |
| 04:12.28 | gamedna | JerJer: any other providers that you would recommend? |
| 04:12.37 | gamedna | similar to EC2 |
| 04:12.43 | JerJer | i would be biased |
| 04:12.54 | gamedna | who do you work for? |
| 04:12.58 | gamedna | ;) |
| 04:15.13 | JerJer | its my company, which is not quite ready to discuss, yet |
| 04:16.43 | gamedna | ah |
| 04:17.29 | *** join/#asterisk bsdfreak (~alex@ninja.paranode.net) |
| 04:17.32 | xuser | leifmadsen: define a call |
| 04:18.15 | xuser | gamedna: check out rackspacecloud |
| 04:19.13 | gamedna | xuser: do you use it? |
| 04:27.16 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 04:27.32 | *** part/#asterisk LemensTS (~LemensTS@70.238.154.222) |
| 04:58.11 | *** join/#asterisk hemantvoip (~hemant.vo@122.163.9.90) |
| 05:01.05 | hemantvoip | hi all |
| 05:01.51 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:08.39 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:25.08 | *** join/#asterisk b14ck (~b14ck@64.206.146.2) |
| 05:35.47 | Micc_ | gamedna, where are you located? |
| 05:40.13 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 05:40.20 | gamedna | Micc_: why do you ask? |
| 05:40.31 | gamedna | im in TX, USA |
| 05:41.42 | Micc_ | Trying to think of hosted asterisk companies that might be in the area. |
| 05:41.51 | gamedna | ah, thanks |
| 05:42.10 | Micc_ | vitelity might have something like that, but I don't think its scalable like EC2, but I don't think you would want to do that even if you could. |
| 05:42.28 | Micc_ | an infinitely scalable asterisk box would have deminishing returns. |
| 05:42.42 | gamedna | true |
| 05:42.53 | gamedna | but finitely scaleable may have some advantages |
| 05:43.09 | Micc_ | TX is a good location, its a major hub. |
| 05:43.20 | gamedna | yea, too bad im in san antonio and not dallas |
| 05:43.40 | gamedna | but that is only a few hrs away |
| 05:43.53 | Micc_ | yeah, you could still host in dallas and be good probably. |
| 05:44.01 | gamedna | yeap |
| 05:44.15 | Micc_ | What exactly are you looking to do with it that you think you need such scalability? |
| 05:44.24 | gamedna | mostly experiment |
| 05:44.53 | gamedna | but im looking at hosted voice apps |
| 05:45.01 | gamedna | not pbx stuff |
| 05:46.04 | gamedna | another option is to just buy a few servers in key datacenters and run Vsphere / Esxi |
| 05:46.19 | gamedna | Xen, or something else |
| 05:46.30 | gamedna | i am just tired of dealing w/ hardware. |
| 05:46.43 | Micc_ | depending on what your doing, virtualization can be a problem for time sensative data. |
| 05:47.02 | gamedna | i have been using VMware in one form or another since its inception |
| 05:47.18 | gamedna | you are right though |
| 05:48.27 | gamedna | right now im facinated with virtualized asterisk |
| 05:48.44 | gamedna | and what i can actually do with it |
| 05:48.57 | gamedna | from and application standpoint |
| 05:50.07 | shamelessn00b | we are running asterisk primarily for IVR |
| 05:50.43 | shamelessn00b | its quite easily managable really |
| 05:51.08 | Micc_ | shamelessn00b, virtualized? |
| 05:51.11 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 05:52.11 | Micc_ | Most of the time we found virtualized asterisk was fine, but there were some weird things that happened every once in a while that we couldn't explain. Haven't had any of those issues since moving to a dedicated server. |
| 05:52.38 | Micc_ | We tried two different hosting companies with different virtualization platforms, both with the same results. |
| 05:52.50 | Micc_ | one was xen and one was virtuoso I think. |
| 05:54.36 | gamedna | ever use ESXi? |
| 05:54.45 | gamedna | i have found it to be very stable |
| 05:55.43 | Micc_ | No, but I have read about it. Thats the vmware commercial product, right? |
| 05:55.45 | gamedna | shamelessn00b: are you running it virtualized? what provider do you use? |
| 05:56.00 | gamedna | yes, but you can use it for free |
| 05:56.24 | Micc_ | Is it a trial? I think I remember, its the management peice that costs money. |
| 05:56.30 | gamedna | http://www.vmware.com/products/vsphere-hypervisor/index.html |
| 05:56.32 | gamedna | no |
| 05:57.11 | gamedna | its an ISO that you install on hardware, and it completely reformats and takes over as a baremetal hypervisor |
| 05:57.25 | gamedna | you then need to use their VirtualCenter to manage the VM's running |
| 05:57.38 | gamedna | the trick is that you are only allowed to connect to one ESXi instance at a time |
| 05:57.46 | gamedna | and cant manage the whole bunch from a central location |
| 05:57.54 | gamedna | that is what you need to pay for, the VSphere |
| 05:58.08 | gamedna | you can run 10 ESXi servers, but you will need to manage them seperately |
| 05:58.16 | gamedna | logout of one, and login to the next |
| 05:58.16 | gamedna | etc |
| 05:58.37 | gamedna | but... there are API's that you can use to manage the machines from the command line |
| 05:58.48 | gamedna | you can write your own scripts. |
| 05:58.54 | ChannelZ | BORED |
| 05:59.06 | gamedna | w/o vsphere you loose the ability to migrate VMS's automagically, etc... |
| 05:59.13 | gamedna | but in general i have not needed that |
| 05:59.29 | gamedna | <<< Juggles for ChannelZ |
| 06:00.09 | gamedna | ChannelZ: better? |
| 06:02.15 | Micc_ | gamedna, right thats what I call the management piece. |
| 06:02.56 | Micc_ | I did look at that before, and I really liked the looks of it, but I would want to buy that management piece so I can migrate vms and stuff. |
| 06:03.00 | gamedna | IMHO, if you are managing < 20 ESXi boxes.. you dont really need vsphere |
| 06:03.30 | Micc_ | yeah I should try it first I guess. |
| 06:03.32 | gamedna | Micc_: you can do it, but you need to write a few scripts... not that hard acutally |
| 06:03.49 | gamedna | i have 4 ESXi boxes here in my sandbox at home |
| 06:04.07 | gamedna | some powerful, some not |
| 06:04.12 | gamedna | i move VM's all the time |
| 06:04.18 | gamedna | just have to do it from linux |
| 06:04.20 | Micc_ | I might have to try that on our next machine. |
| 06:04.24 | gamedna | another tip... |
| 06:04.30 | gamedna | if you decide to go ESXI |
| 06:04.39 | gamedna | always have at least ONE windows XP or Win7 VM |
| 06:04.44 | Micc_ | I'd like to run a couple linux and one windows server. |
| 06:04.49 | gamedna | with the management software on there |
| 06:05.17 | gamedna | (i think linux works too) |
| 06:05.26 | Micc_ | can I run the management software on windows server 2k8? |
| 06:05.29 | gamedna | yes |
| 06:05.34 | gamedna | i believe so |
| 06:05.48 | gamedna | the only drawback is that you cant manage that particular VM |
| 06:05.50 | Micc_ | can the management OS be running in VM too? |
| 06:05.57 | gamedna | yes... |
| 06:06.21 | Micc_ | can't manage it, what do you mean? |
| 06:06.23 | gamedna | i have a 512MB / 8GB VM that runs XP for the management OS |
| 06:06.40 | gamedna | well, cant really shutdown an OS that you are running on |
| 06:06.41 | gamedna | right? |
| 06:06.47 | drmessano | So, I can run the VM management in a VM that I manage with the VM management? |
| 06:06.52 | drmessano | FAR TEH HELL OUT |
| 06:06.59 | gamedna | ;) |
| 06:07.14 | gamedna | i do it all the time |
| 06:07.25 | gamedna | Since i am on MAC, i have no other choice |
| 06:07.36 | gamedna | cant run the tools on mac yet, and the web stuff just blows IMHO |
| 06:07.36 | drmessano | I hear that all the time |
| 06:07.48 | Micc_ | I thought you didn't need to run an OS to host the other OSs? |
| 06:07.49 | drmessano | "I am on a MAC, I have no other choice" |
| 06:08.05 | gamedna | drmessano: dont reboot, be-root. |
| 06:08.11 | drmessano | lol |
| 06:08.28 | gamedna | Micc_: you dont... |
| 06:08.39 | gamedna | you can install ESXi |
| 06:08.43 | gamedna | on 10 machines |
| 06:08.45 | *** join/#asterisk nova911 (~Adium@115.118.151.80) |
| 06:08.49 | Micc_ | but you have to run the management software at all times to keep the other OSs up? |
| 06:08.55 | gamedna | and run the management software on a seperate computer |
| 06:09.00 | gamedna | no |
| 06:09.05 | gamedna | not at all |
| 06:09.19 | gamedna | im just giving you the tip for convenience |
| 06:09.36 | gamedna | having a small XP vm with the tools on there makes it super easy |
| 06:09.41 | Micc_ | but you can't shut it down, why? |
| 06:09.49 | gamedna | you cant shut it down from ITSELF |
| 06:10.01 | gamedna | if you are logged into that VM, you cant shut yourself down |
| 06:10.02 | drmessano | An Egg can't lay a chicken |
| 06:10.05 | Micc_ | oh, you mean the management tools allows you to shut down the other VMs. |
| 06:10.09 | gamedna | from the MGT console |
| 06:10.12 | gamedna | right |
| 06:10.39 | Micc_ | ok, got it. I would want my VMs always up though probably, but I get it now. |
| 06:10.50 | gamedna | here is an example |
| 06:10.57 | gamedna | say you got new hardware and you wanted to migrate everything |
| 06:10.58 | Micc_ | So the management tools are free too? |
| 06:11.09 | drmessano | .... |
| 06:11.20 | gamedna | if you were logged into that VM MGT OS, you could do all the other OS's except the MGT OS |
| 06:11.27 | gamedna | all you do then, is logout |
| 06:11.30 | gamedna | and use it on another desktop |
| 06:11.37 | gamedna | yes, mgt tools are free |
| 06:11.41 | gamedna | it comes w/ it |
| 06:11.58 | gamedna | you can do iSCSI |
| 06:12.03 | gamedna | NAS |
| 06:12.03 | gamedna | etc |
| 06:12.05 | gamedna | all works |
| 06:12.19 | drmessano | What about iNAS ? |
| 06:12.28 | Micc_ | Thats some good shit. |
| 06:12.30 | gamedna | nope, but it has SNAS |
| 06:12.37 | gamedna | and PIZAZ |
| 06:12.42 | drmessano | What about iCUP? |
| 06:12.51 | gamedna | pervert |
| 06:12.54 | gamedna | j/k |
| 06:13.12 | Micc_ | I might use that for one of my consulting gigs. They need some in house servers for a 30 person office. I think one server with 3 or 4 VMs might do the trick. |
| 06:13.28 | drmessano | You're the one who stated "I scuzzy" |
| 06:13.45 | gamedna | yes, got me again, i did. |
| 06:13.48 | Kyosh | sup game |
| 06:13.55 | drmessano | Sup Yoshi |
| 06:14.09 | gamedna | Micc_: one word of caution, be sure to read the hardware requirements.. ESXi is slightly finiky about what hardware it runs on |
| 06:14.11 | Kyosh | peers at the doc |
| 06:14.21 | drmessano | Slightly? |
| 06:14.25 | gamedna | supports all major "server" class machines, but white boxes are hit / miss |
| 06:14.38 | Micc_ | gamedna, I was thinking a supermicro box. |
| 06:14.42 | gamedna | there is a website that is dedicated to listing all the white box machines that esxi supports |
| 06:14.46 | gamedna | should be fine w/ supermicro |
| 06:14.53 | Micc_ | dual quad core or dual dual core. |
| 06:14.57 | gamedna | i have an older Dell SC400 P4 box, and it runs fine there |
| 06:14.57 | drmessano | ESXi bitches unless you have a box with hardware virtualization and 2GB RAM minimum |
| 06:15.10 | gamedna | correct |
| 06:15.15 | drmessano | I know it is |
| 06:15.22 | Micc_ | of course. |
| 06:15.22 | Kyosh | he is the doc |
| 06:15.34 | gamedna | but 2gb is not enough |
| 06:15.38 | gamedna | really go for 4 or 8gb of ram |
| 06:15.39 | gamedna | min |
| 06:15.44 | Micc_ | Its gotta be a decently new proc with hyper v support. |
| 06:15.57 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 06:16.10 | Micc_ | ram is cheap. |
| 06:16.11 | drmessano | Windows XP with Virtual PC running 512MB RAM should be enough for anyone |
| 06:16.13 | gamedna | look i run on a P4-2.4ghz w/ 4gb of ram, single core |
| 06:16.18 | gamedna | and i have 6 Vms on there |
| 06:16.20 | gamedna | and they all run fine |
| 06:16.26 | Kyosh | 640kb ram should be enough for anyone |
| 06:16.43 | gamedna | hahaha |
| 06:16.59 | Micc_ | lets use 7 bit bytes while we're at it |
| 06:17.19 | Kyosh | are we using modems again? |
| 06:17.26 | drmessano | gamedna: Isn't that all relative? I mean, my NT Workstation domain of 75 workstation VMs and 1 NT Domain Controller runs fine with 4GB RAM, but you know |
| 06:17.38 | gamedna | 7 bit bytes + 1 crumb |
| 06:17.59 | gamedna | drmessano: of course... |
| 06:18.14 | drmessano | You only need 11MB RAM to install NT4 |
| 06:18.36 | gamedna | drmessano: you could run 100 freeDOS vms too |
| 06:19.01 | drmessano | I prefer PC-DOS, TYVFM |
| 06:19.17 | drmessano | I'll pay for my DOS, thank you |
| 06:19.20 | Kyosh | i dont wanna work |
| 06:19.21 | Micc_ | Dr. DOS |
| 06:19.23 | gamedna | my point is that im running Win2k8 (32bit) server, 2 modern linux distros, freeNAS, TRIXBOX, and something else |
| 06:19.26 | gamedna | and it runs fine |
| 06:19.28 | Kyosh | i just wanna bang on my drum all day |
| 06:19.34 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 06:19.34 | drmessano | Trixbox? |
| 06:19.40 | drmessano | gets the hounds |
| 06:20.03 | Micc_ | oh no, you can't swear in here, gamedna. |
| 06:20.08 | Kyosh | gamedna, what kinda server and how much ram? |
| 06:20.10 | drmessano | I couldn't hear you over the sound of me racking my shotgun |
| 06:20.23 | gamedna | Dell Poweredge SC400 |
| 06:20.23 | ChannelZ | It's The PBX For Whores |
| 06:20.36 | drmessano | "Tricks Box" |
| 06:20.39 | gamedna | ITS the PBX for people who like to easily provision 60 IP phones in under 30 mins |
| 06:20.47 | Kyosh | yup |
| 06:20.52 | ChannelZ | I can do that |
| 06:20.53 | drmessano | Friends don't let friends turn Trix |
| 06:21.35 | gamedna | Kyosh: Dell Poweredge SC400, P4, 2.4ghz, 4GB ram, 2x500GB - RAID 1 |
| 06:21.37 | Micc_ | gamedna, then when you have a problem with one of those phones, you'll never know how to fix it because you don't know what it did. |
| 06:21.43 | drmessano | Trixbox is the PBX for people who think Kerry Garrison is a swell guy, and don't care if Fonality knows what he ate for dinner |
| 06:21.59 | Micc_ | gamedna, so you saved yourself a little time, but you lost the war. |
| 06:22.00 | gamedna | Micc_: i am familiar w/ the dialplan |
| 06:22.12 | Kyosh | gamedna, yea that should work fine |
| 06:22.15 | gamedna | been running trixbox since 2008 |
| 06:22.23 | gamedna | no problems thus far |
| 06:22.28 | drmessano | You poor bastard |
| 06:22.30 | gamedna | but i only use it for what its indented for |
| 06:22.40 | gamedna | IVR, Voicemail, and Extensions |
| 06:22.50 | gamedna | for everything else i use asterisk |
| 06:22.51 | drmessano | No gaming? |
| 06:23.02 | gamedna | ;) |
| 06:23.04 | drmessano | Trixbox is my fav first person shooter |
| 06:23.08 | gamedna | hahaha |
| 06:23.10 | Micc_ | haha |
| 06:23.15 | gamedna | i thought it would be your favorite target. |
| 06:23.19 | Micc_ | Thats a good one, I'll have to remember that. |
| 06:23.22 | ChannelZ | Shooting hookers is so 2009 |
| 06:23.37 | drmessano | ChannelZ: Nothing wrong with a dead hooker |
| 06:23.39 | gamedna | ChannelZ: cant wait till its retro... |
| 06:24.17 | ChannelZ | One of the few funny SNL skits I can remember.. a gameshow called "Old French Whore" |
| 06:24.31 | drmessano | ChannelZ: If you party a whole weekend and don't end up with at least one dead hooker in the back of someone's car, you may as well go hang out at the retirement home and buy a minivan |
| 06:24.46 | ChannelZ | *buzzes in* "Uhh, I think my whore is dead." |
| 06:25.50 | *** join/#asterisk shido6 (~shido6@c-24-4-217-126.hsd1.ca.comcast.net) |
| 06:25.59 | ChannelZ | Oh looks! http://www.myvideo.de/watch/5620408/old_french_whore |
| 06:26.24 | drmessano | Hell, Vegas was so hard up during the recession that if LVPD caught you leaving town without a dead hooker to take home, they would give you one as a souvenir |
| 06:27.07 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 06:27.11 | ChannelZ | sweet I know where I'm going for the weekend |
| 06:30.12 | drmessano | Hmmmm |
| 06:30.35 | drmessano | I think I just got figure out how i am gonna make my first billion |
| 06:31.30 | drmessano | Well, typing a coherent sentence, first off |
| 06:32.02 | drmessano | hosted whores.. cloudwhore.com |
| 06:32.14 | Kyosh | i love cludwhores |
| 06:32.15 | drmessano | Who wants to write an API? |
| 06:32.25 | gamedna | me me me me! |
| 06:32.45 | gamedna | hahah |
| 06:32.53 | drmessano | Trick is gonna be sanitizing inputs and outputs versus overall function |
| 06:33.43 | drmessano | I would hate to kill an instance due to a senseless buffer overflow |
| 06:33.46 | gamedna | iWhores |
| 06:34.07 | ChannelZ | we already have those, walk into any Apple store |
| 06:34.08 | gamedna | hosted-whores.com |
| 06:34.13 | drmessano | lol |
| 06:34.18 | Kyosh | cloudwhores are better |
| 06:34.35 | gamedna | yea but hosted-whores are more reliable and perform better |
| 06:34.38 | ChannelZ | Yes but iWhores cost more and come with a sticker |
| 06:35.09 | Micc_ | it says I can evaluate vsphere for 60 days. |
| 06:35.15 | Micc_ | I thought it was free? |
| 06:35.21 | drmessano | vsphere isn't |
| 06:35.22 | gamedna | vsphere is the management stuff |
| 06:35.29 | Kyosh | yea but cloudwhores are distributed for your satisfaction |
| 06:35.50 | gamedna | personally i like timeshare-whores |
| 06:36.26 | gamedna | https://www.vmware.com/tryvmware/index.php?p=free-esxi&lp=1 |
| 06:36.35 | gamedna | YOUR FREE VSPHERE HYPERVISOR REGISTRATION INCLUDES ACCESS TO |
| 06:36.35 | gamedna | <PROTECTED> |
| 06:36.36 | gamedna | <PROTECTED> |
| 06:36.36 | gamedna | <PROTECTED> |
| 06:37.07 | Micc_ | ok, I guess I signed up for a vsphere evaluation. |
| 06:37.08 | gamedna | when you register you can request as many keys as you want |
| 06:37.11 | shamelessn00b | Micc_: no not virtualized, but we are running a cluster |
| 06:37.19 | shamelessn00b | SS7 |
| 06:37.19 | Micc_ | I just need vsphere hypervisoresxi. |
| 06:37.29 | gamedna | https://www.vmware.com/tryvmware/?p=esxi&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official |
| 06:37.33 | gamedna | that is for 32 bit hardware |
| 06:37.56 | gamedna | ESXi 3.5 (32bit) or ESXi 4.1 for 64bit |
| 06:38.06 | *** part/#asterisk adolfomaltez (~taro@190.87.103.192) |
| 06:38.12 | gamedna | right... |
| 06:38.36 | gamedna | 3.5 does not give you access to GO |
| 06:42.33 | gamedna | Micc_: remember esxi install will re-format your entire machine |
| 06:43.09 | Micc_ | thats fine |
| 06:43.30 | Micc_ | I'm only going to test when i get a new machine |
| 06:43.33 | gamedna | k |
| 06:43.41 | gamedna | seen it happen to many people |
| 06:43.52 | Micc_ | good to know |
| 06:44.25 | Micc_ | I've got an old server I could try it on, but I'd have to image the drive first. |
| 06:44.33 | *** join/#asterisk arossouw (~arossouw@dsl-146-51-03.telkomadsl.co.za) |
| 06:44.36 | Micc_ | I need the windows server 2003 installatino thats on it. |
| 06:44.49 | Micc_ | I wouldn't mind turning it into a vm if theres an easy way to do that. |
| 06:44.59 | *** part/#asterisk arossouw (~arossouw@dsl-146-51-03.telkomadsl.co.za) |
| 06:45.05 | shamelessn00b | How popular is SS7 in USA btw? |
| 06:45.12 | Micc_ | I dunno if that server supports a hypver visor though |
| 06:45.31 | Micc_ | shamelessn00b, dunno, I've never used it. |
| 06:45.32 | gamedna | what machine? |
| 06:45.41 | Micc_ | its an older super micro. |
| 06:46.12 | *** join/#asterisk arossouw (~arossouw@dsl-146-51-03.telkomadsl.co.za) |
| 06:46.45 | arossouw | hi, i am thinking of writing a script to monitor line status of isdn, is there a way to get a return code from asterisk to indicate line ok |
| 06:46.47 | gamedna | got a #? |
| 06:47.30 | Micc_ | intel xeon 2.4 ghz. 1gb ram. |
| 06:47.39 | gamedna | should work |
| 06:47.57 | gamedna | will run only 1 vm |
| 06:48.06 | gamedna | w/ only 1GB but good for testing |
| 06:48.34 | Micc_ | I could probably get more memory for it pretty easily. |
| 06:48.45 | Kyosh | shamelessn00b: its popular |
| 06:48.54 | Micc_ | is there an easy way to turn the current OS into a vm? |
| 06:49.56 | shamelessn00b | Kyosh: where you from? |
| 06:51.06 | arossouw | found something http://www.voip-info.org/wiki/view/Asterisk+monitoring |
| 06:51.35 | Kyosh | new york |
| 06:51.49 | gamedna | Micc_: yes |
| 06:51.53 | gamedna | vmotion i think |
| 06:51.54 | gamedna | hold on |
| 06:52.12 | shamelessn00b | I'm using cacti graphs to monitor asterisk |
| 06:52.14 | gamedna | http://www.vmware.com/products/vmotion/ |
| 06:52.57 | gamedna | wait |
| 06:52.59 | gamedna | that is not it |
| 06:54.02 | gamedna | http://www.vmware.com/products/converter/ |
| 06:54.04 | gamedna | there we go |
| 06:59.15 | Micc_ | gamedna, thats perfect. |
| 06:59.20 | *** join/#asterisk iscsi (~light@78.108.73.46) |
| 06:59.26 | Micc_ | I'll have to play with that this weekend. |
| 06:59.36 | gamedna | w/ 4.1 you can use VMWARE Go |
| 06:59.46 | Micc_ | If I can find more memory for that machine, i'll be in business. It looks like its dual proc too. |
| 06:59.46 | gamedna | and that will help you provision and do the P2V conversion too |
| 07:00.08 | shamelessn00b | but I'm looking for something that would let me monitor call quality on sip trunks |
| 07:00.22 | shamelessn00b | not wireshark alone |
| 07:00.32 | Micc_ | shamelessn00b, sip show channelstats |
| 07:00.43 | Micc_ | it can help, although sometimes its not very helpful. |
| 07:00.54 | Micc_ | I find it doesn't report correct stats on some connections. |
| 07:01.27 | shido6 | shamelessn00b: u need something that will monitor MOS scores from the phones ( if they are linksys or spa or cisco you're good ..) |
| 07:01.42 | shamelessn00b | yeah exactly |
| 07:01.44 | shido6 | I was looking at VQadmin from Device Expert pplk |
| 07:01.55 | shamelessn00b | MOS scores on a trunk, all active calls |
| 07:02.15 | shido6 | VQManager |
| 07:02.28 | shamelessn00b | I use this program called AQUA to get MOS scores on SIP streams, but I have to extract the RTP data using wireshark |
| 07:02.46 | *** part/#asterisk arossouw (~arossouw@dsl-146-51-03.telkomadsl.co.za) |
| 07:03.07 | shido6 | http://www.manageengine.com/products/vqmanager/ |
| 07:03.09 | Micc_ | I don't think xeon's are 64bit, at least the older ones anyways. |
| 07:03.29 | shido6 | im sorry , http://demo.vqmanager.com/VoIPMain.cc |
| 07:03.38 | shamelessn00b | http://www.ntop.org/OpenSourceVoipMonitoring.pdf |
| 07:03.57 | shamelessn00b | I have yet to figure this one out though |
| 07:04.28 | shido6 | if u need help |
| 07:04.43 | *** join/#asterisk qvsqvs (~anonymous@mail.logical.co.za) |
| 07:04.49 | shido6 | ask |
| 07:05.09 | shido6 | you never know whos listening |
| 07:09.11 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 07:14.07 | goddva | shido6: Does the SPA series have the calculation of MOS scores built in? >> if they are linksys or spa or cisco you're good . |
| 07:19.13 | Micc_ | dnag, my xeon is a prestonia, only 32 bit. So I can't use esxi 4.1 |
| 07:20.12 | Micc_ | dnag/dang |
| 07:20.26 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 07:21.42 | gamedna | its ok, 3.5 is good |
| 07:24.45 | Micc_ | ok, I think I can only get 1gb sticks for it though, so I'll only be able to do 4gb. |
| 07:24.55 | Micc_ | But that should be great for running 4 vms. |
| 07:24.58 | gamedna | should be good for a test system |
| 07:25.58 | Micc_ | cool. time for bed. Good night all. |
| 07:26.58 | gamedna | Micc_: Nite! |
| 07:27.11 | shamelessn00b | gnite |
| 07:39.53 | *** join/#asterisk iscsi (~light@78.108.73.46) |
| 07:44.35 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:49.56 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 07:55.39 | *** join/#asterisk qvsqvs (~anonymous@mail.logical.co.za) |
| 08:01.00 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-rhitpgsanzahljcs) |
| 08:01.17 | henk | morning, i am still having an issue with custom sounds. where do i put them so they are found? i was told asterisk would look in 'astvarlibdir' as defined in asterisk.conf, so i put them in /var/lib/asterisk/sounds/ but it doesn't seem asterisk finds them there... how could i debug that further? |
| 08:01.49 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 08:04.47 | ChannelZ | what version of asterisk? |
| 08:05.19 | henk | Asterisk 1.6.2.9-1 |
| 08:05.47 | ChannelZ | Asterisk uses the language layout by default, so /var/lib/asterisk/sounds/en if you're setup as english |
| 08:05.51 | henk | http://pastie.org/1087779 asterisks error messages. |
| 08:06.18 | ChannelZ | Don't give the extension in the filename |
| 08:06.21 | henk | i 'gave up' and tried an absolute path. no luck either. |
| 08:06.27 | henk | afair i tried, let me try again. |
| 08:06.30 | ChannelZ | If you have "fooyou.ulaw" you would Playback(fooyou) |
| 08:07.10 | ChannelZ | make sure whatever user asterisk runs as has read permission to the files as well |
| 08:07.36 | henk | oy, you're right, must have overlooked that the first time! without .wav it gives a more meaningful error: http://pastie.org/1087784 |
| 08:07.49 | henk | i'll work on that and see what goes. thanks :) |
| 08:08.22 | ChannelZ | for wav they must be 16-bit 8khz mono |
| 08:08.57 | henk | damn, got the 16bit and the 8khz but not mono, let's see, audacity should be able to do that for me... |
| 08:09.00 | henk | thanks again :) |
| 08:13.55 | ChannelZ | sure good luck |
| 08:15.37 | WIMPy | henk: Make sure not to add any tags to the file. |
| 08:21.44 | tzafrir_laptop | henk, it's not astvarlibdir. it's astdatadir |
| 08:29.09 | henk | WIMPy: good to know, thanks! |
| 08:29.44 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:30.15 | henk | tzafrir_laptop: oh ok... are you sure? sorry to ask that, i'm pretty sure you know what you're talking about but the same counts for [TK]Defender afaik and he's the one who told me... |
| 08:33.26 | tzafrir_laptop | just looked again in main/asterisk.c (look for "astdatadir" and "astvarlibdir" , with quotes) |
| 08:34.00 | tzafrir_laptop | Also note that in latest debs /usr/share/asterisk/sounds/en is a symlink |
| 08:34.43 | tzafrir_laptop | If you want to override that, maybe place "custom" directly under /usr/share/asterisk/sounds |
| 08:37.46 | *** join/#asterisk ajm0716 (~ajm0716@c-76-97-74-68.hsd1.ga.comcast.net) |
| 08:42.17 | henk | tzafrir_laptop: ok, cool, thank you very much, sounds good :) |
| 08:43.20 | henk | tzafrir_laptop: oh, i guess that's even better: /usr/share/asterisk/sounds/custom -> ../../../local/share/asterisk/sounds |
| 08:43.38 | tzafrir_laptop | yes |
| 08:46.57 | *** join/#asterisk ruyo (~psantos@a83-132-248-161.cpe.netcabo.pt) |
| 08:51.43 | *** join/#asterisk Beltechs (~Beltechs@208.127.3.20) |
| 08:52.24 | henk | ok, i saved the recorded file in audacity as a microsoft wave file with 16bit, 8khz, mono. it plays fine in audacity and sox, but asterisk plays it at double speed. also it says "Playing 'custom/I7_Geschaeftszeiten.slin' (language 'en')" although it's .wav. what is wrong? |
| 08:54.56 | ChannelZ | that's normal (the .slin part) |
| 08:55.12 | ChannelZ | the double speed, not so much |
| 08:55.28 | Beltechs | hello Im trying to debug an extension I used sip debug and have posted my findings. Can someone please help decipher? I see something about sip:Unknown@..... http://pastebin.com/MmX7T82a |
| 08:55.55 | tzafrir_laptop | henk, does asterisk play other files well? |
| 08:56.07 | ChannelZ | henk: so what does it say if in a shell you do "file I7_Geschaeftszeiten.wav" |
| 08:56.13 | henk | so i'm almost good now :-/ is there a definitive reference on what formats, bitrates, whatever asterisk actually _likes_ in soundfiles? |
| 08:56.57 | tzafrir_laptop | WAV is a rather simple container format. Look at the wikipedia entry for ".wav" . |
| 08:57.25 | ChannelZ | Beltechs: looks like your Asterisk is just having problems reaching the peer.. firewall perhaps |
| 08:57.52 | henk | tzafrir_laptop: good question. no. i tried the 'your-msg-has-been-saved.gsm' that comes with asterisk in debian. same issue... |
| 08:57.55 | tzafrir_laptop | Asterisk likes a single one - "telephony" - 16 bits per sample, 8000 samples per second, mono |
| 08:58.30 | ChannelZ | the gsm plays fast? |
| 08:58.30 | tzafrir_laptop | I suppose this is a timing issue. Try: timing test |
| 08:58.58 | henk | http://pastie.org/1087868 |
| 08:59.12 | ChannelZ | hmm |
| 08:59.35 | ChannelZ | what codec is the channel you're testing on using? |
| 08:59.40 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:59.44 | henk | tried several times, same result each time. |
| 09:00.32 | henk | ChannelZ: uhm, i'm not sure how to tell... is that it: Using SIP RTP CoS mark 5 |
| 09:00.40 | Beltechs | 1 out of 4 extensions is working |
| 09:01.09 | henk | ChannelZ: ah no, you mean sip show channels? then it's 0x8 (alaw) |
| 09:01.47 | ChannelZ | Beltechs: Your asterisk is unable to talk to 208.127.3.20 on port 9448... or it's replies are not making it back. I don't know why that is, but it's generally a firewall issue on either (or both) sides |
| 09:02.02 | tzafrir_laptop | henk, interesting. I thought timerfd was working well |
| 09:02.37 | tzafrir_laptop | Any chance you could try to use res_timing_pthread instead ? |
| 09:02.37 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 09:02.58 | henk | tzafrir_laptop, ChannelZ: i'm not sure if that matters, but regarding timing it might: the server asterisk is running on is a xen guest. |
| 09:03.17 | henk | tzafrir_laptop: yeah, sure, the system is not really productive anyway, yet. |
| 09:04.50 | henk | tzafrir_laptop: i guess i have to noload the timerfd module and load the pthread module in modules.conf? anything else? |
| 09:04.54 | ChannelZ | You say it plays twice the speed, is it pitched up as well? |
| 09:04.59 | tzafrir_laptop | yes |
| 09:05.14 | tzafrir_laptop | THough IIRC noloading timerfd should do it on its own |
| 09:05.20 | henk | ChannelZ: no, sounds like it just skips every second 'frame' or whatever. |
| 09:05.32 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:05.43 | ChannelZ | ok... so yeah it seems like a timing issue |
| 09:06.04 | gamedna | nite all |
| 09:06.11 | ChannelZ | though interesting your 'timer test' numbers seemed fine |
| 09:06.27 | *** part/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 09:06.55 | ChannelZ | but perhaps the vm is bending the rules of time |
| 09:07.22 | *** join/#asterisk Tim_Toady (~moi@77.49.3.102.dsl.dyn.forthnet.gr) |
| 09:07.31 | henk | okaayyyy... great. now the stock gsm files plays just fine and my own file sounds 4x faster than expected... |
| 09:07.42 | henk | ChannelZ: yeah, i guess xen vms tend to do that sometimes... |
| 09:07.46 | ChannelZ | sweet |
| 09:07.55 | ChannelZ | You'll be able to get through so many more voicemails this way |
| 09:09.04 | henk | lol |
| 09:09.38 | henk | too bad it's our "We are currently not available. You can reach us Monday to friday..." announcement when no one's here... |
| 09:11.06 | henk | timing test with pthread: http://pastie.org/1087883 |
| 09:11.19 | henk | sometimes it's 1004 milliseconds |
| 09:15.14 | henk | i'll try other formats of wav that audacity supports... |
| 09:15.49 | ChannelZ | still it's not twice as fast. Does it feel like a second between when it says "Using the 'xxxx' timingmodule for this test..." and "It has been 1000 milliseconds..." |
| 09:16.47 | henk | ChannelZ: yes. |
| 09:16.59 | *** join/#asterisk modsaid (~modsaid@82.201.210.162) |
| 09:17.09 | modsaid | greetings everyone |
| 09:17.20 | modsaid | has anyone tried using the jack_hook before ? |
| 09:17.22 | ChannelZ | hmm that makes no sense |
| 09:17.22 | henk | too bad the command's not blocking, i could use 'time asterisk -rx "timing test"' to check exactly... |
| 09:17.37 | ChannelZ | are you testing on a softphone? |
| 09:18.29 | henk | ChannelZ: setup is a bit special: i have a cisco 7905 series here, connected to a cisco C1700 call manager which asterisk registers to via sip. |
| 09:18.54 | ChannelZ | hmm a few variables |
| 09:19.27 | ChannelZ | I don't suppose your * is reachable via SIP on a public IP I could call to test |
| 09:19.47 | henk | it isn't... |
| 09:20.04 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 09:20.13 | *** part/#asterisk lauris (~la@unaffiliated/lauris) |
| 09:20.15 | henk | ok, tried the gsm6.10 format audacity knows, that's even worse... lot of errors. |
| 09:20.23 | ChannelZ | well perhaps try a softphone on your workstation just to narrow it down and make sure it's not something odd happening with your other setup |
| 09:20.28 | henk | asterisk tries to deduce the file format from the extension right? |
| 09:21.21 | ChannelZ | Sort of; It tries to find the best possible file to use based on the channel codec, and looks for the file with various extensions as a result |
| 09:21.44 | henk | ok, what channel codec(s) should i use/allow? |
| 09:21.48 | ChannelZ | it assumes the extension actually matches the content though |
| 09:22.14 | ChannelZ | Depends on your needs, but ulaw/alaw at least |
| 09:22.50 | ChannelZ | * will transcode when it can if for instance the channel is gsm but the only available format is wav or ulaw or whatnot |
| 09:23.54 | ChannelZ | while a call is going you can do 'sip show channels' and it should list the codec being used |
| 09:25.31 | henk | ok, i'm on ulaw, was on alaw, both have problems... |
| 09:26.44 | ChannelZ | well try a softphone just to rule out your cisco setup |
| 09:26.48 | henk | ok, next wavefile format... |
| 09:26.57 | henk | hm, yeah, good idea too... |
| 09:27.03 | ChannelZ | bed time for me, good luck |
| 09:27.10 | henk | thanks, good night! |
| 09:28.09 | modsaid | anyone tried jack_hook before? |
| 09:33.29 | henk | omg, it really plays almost flawless via a softphone :-/ |
| 09:35.26 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 09:36.25 | henk | argh :( wrong again, must be something else... i guess i'll just take real hardware and try with that... |
| 09:36.42 | henk | that eliminates one possible cause of problems... |
| 09:37.01 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 09:40.50 | *** join/#asterisk RZer0 (~RZer0@ra.enta.net) |
| 09:45.37 | *** join/#asterisk tomarch (~tom@fbx1.reseau-concept.net) |
| 09:46.31 | *** join/#asterisk hrhrhr (~c1@213.1.224.2) |
| 09:51.34 | *** join/#asterisk lukhas (~lucas@bearstech/lukhas) |
| 09:51.44 | lukhas | hello |
| 09:52.39 | lukhas | I'm having trouble to set up transfers to internal extensions (SIP phones) for incoming IAX calls |
| 09:53.03 | lukhas | transfer from SIP to SIP is working fine |
| 09:53.34 | *** join/#asterisk pinoyskull (~pinoyskul@72.26.105.43) |
| 10:02.11 | ruyo | lukhas, is DTMF working in incomming IAX calls? |
| 10:03.43 | lukhas | how can I test that? Whenever I press keys, I hear the DTMF tones in the other phone, in both directions |
| 10:04.04 | lukhas | (even if I press # before) |
| 10:05.38 | lukhas | I have dtmfmode=rfc2833 in iax.conf |
| 10:06.48 | ruyo | Is the IAX account a phone? |
| 10:07.10 | lukhas | nope, it's a VoIP provider |
| 10:07.48 | lukhas | "regular telephone" -> VoIP provider <- iax trunk -> our asterisk -> SIP phones |
| 10:07.51 | ruyo | Ok. So, when you receive a call from your provider, you can't transfer it to another phone? |
| 10:07.54 | lukhas | yes |
| 10:08.00 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 10:08.11 | ruyo | Are you allowing transfers on those calls? |
| 10:08.22 | lukhas | I thought so |
| 10:08.44 | lukhas | there is "transfer=yes" in iax.conf |
| 10:08.56 | lukhas | but obviously that's not enough :) |
| 10:09.40 | ruyo | Try using the Dial app like: exten => <exten>,<prio>,Dial(SIP/<exten>,,t) |
| 10:10.04 | ruyo | The 't' argument allows the callee to transfer calls. |
| 10:10.07 | lukhas | ah |
| 10:10.15 | ruyo | And the 'T' allows the caller. |
| 10:10.47 | lukhas | indeed, no sign of 't' in my dialplan, damn |
| 10:12.19 | *** join/#asterisk AlienPenguin (~my@79.171.63.250) |
| 10:13.16 | AlienPenguin | hi, if i n my dialplan i have: exten => 911,1,Transfer(SIP/120@192.168.23.2:5080/j) then every call should go to that specific sip address? |
| 10:13.50 | sawgood | exten => 19164892236,1,Dial(SIP/4104) |
| 10:13.50 | sawgood | exten => 19164892225,n,Dial(SIP/4104) |
| 10:14.11 | sawgood | hi if I want specific CID to match, is this the right dialplan statement? |
| 10:14.44 | *** join/#asterisk ickmund (~magnus@cli-5b7e85d4.bcn.adamo.es) |
| 10:14.59 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 10:15.34 | ruyo | AlienPenguin, I think only SIP calls get transfered like that. |
| 10:16.10 | AlienPenguin | ruyo, and if 120 is a registered user? i just want the call to go to the direct IP address and not be handled by asterisk |
| 10:16.30 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 10:18.29 | lukhas | ruyo: 't' in the Dial application did the trick, thanks! |
| 10:18.33 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 10:18.37 | ruyo | AlienPenguin, I think that if the user accepts unauthenticated calls it should work. I'm just guessing, though. I don't really know how Transfer works. |
| 10:18.37 | sawgood | What I am facing is this (the Asterisk 1.6.2.10 box) receives a call from the ITSP (it is processed correctly) ... Now, the Asterisk box has a 2nd DID, but it is not processing the call correctly when it is received (it is following the rules for the 1st DID) |
| 10:19.02 | ruyo | lukhas, np. |
| 10:19.12 | doolittlework | hi there i have a wierd implementation of asterisk i must do, i need to link it to our legacy pbx, we want to record our cusomer service line, I managed to get this working i have made some custom filenames for the recordings and setup a samba share |
| 10:20.21 | doolittlework | my problem is that i wan t to reference the call whiles on the line with the customer, is there a way to display something like a unique id in the xlite screen for every call, and then link that into my recoding file name |
| 10:21.28 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 10:21.44 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 10:21.57 | Kalidarn | hey does anyone know how to set openmode on a tdm400p? |
| 10:22.26 | doolittlework | Kalidarn: what do you mean by open |
| 10:23.00 | WIMPy | doolittlework: Generate an ID and put it into CALLERID(name)? |
| 10:23.03 | Kalidarn | it's saying http://www.voipuser.org/forum_topic_2743.html |
| 10:23.08 | Kalidarn | for example |
| 10:23.16 | Kalidarn | options wctdm opermode=UK |
| 10:23.23 | Kalidarn | sorry my blindness i meant to say opermode |
| 10:23.23 | ruyo | sawgood, what is the extension string you're using to handle the call? |
| 10:23.43 | Kalidarn | course that only will work with linux kernel modules |
| 10:23.47 | Kalidarn | i'm using freebsd |
| 10:24.16 | Kalidarn | hint.wctdm.0.opermode="AUSTRALIA" put in /boot/device.hints should work |
| 10:24.42 | Kalidarn | last time i had to modify the source |
| 10:24.47 | Kalidarn | but ill try this :P |
| 10:25.07 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 10:25.23 | AlienPenguin | ruyo, well i do remember it used to work with asterisk 1.4.x but now i get a different behaviour with 1.6.2.10 |
| 10:25.41 | Kalidarn | anyways doolittlework (normally i'd stick around but i have to reboot this box to test it so ill be back in a bit). |
| 10:26.58 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 10:31.23 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 10:31.38 | Kalidarn | mmm doolittlework seems that doesn't work. :( |
| 10:31.45 | Kalidarn | guess i'lll have to modify the source |
| 10:32.36 | *** join/#asterisk jrz- (~jrz@a190165.upc-a.chello.nl) |
| 10:35.15 | ruyo | kaldemar, to set opermode try using "modprobe wctdm opermode=AUSTRALIA" |
| 10:35.37 | ruyo | *Kalidarn |
| 10:38.20 | sawgood | how does one tell exactly what Asterisk is doing with an incoming SIP call when it arrives at the box (I have sip set debug on) |
| 10:40.04 | Kalidarn | ruyo: yeah that's for linux. |
| 10:40.09 | Kalidarn | because it's linux module. |
| 10:40.51 | *** join/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 10:41.08 | Kalidarn | what i've found is it doesn't actually work when setting hint.wctdm.0.opermode="AUSTRALIA in /boot/device.hints |
| 10:41.27 | Kalidarn | so i had to edit line 300 of wctdm.c |
| 10:41.51 | Kalidarn | and change static char opermode[128] = "FCC" |
| 10:41.59 | Kalidarn | to static char opermode[128] = "AUSTRALIA" |
| 10:43.19 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.194) |
| 10:44.33 | geemee | Hi all. When having 2 asterisk boxes linked together I can dial another extensions from one side to the other side OK however I cannot route calls to another box when using IVR. I presume this is since 1 box isnt aware of the other extensions. Is there a way around this? |
| 10:47.17 | DennisG | geemee: you can use DUNDi for it |
| 10:47.33 | *** join/#asterisk m0t3jl (~petr.mote@213.29.237.1) |
| 10:47.48 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 10:48.18 | Kalidarn | i'm back |
| 10:48.43 | doolittlework | if i record files in mp3 can asterisk play them back to me or do i need to add additional apps to asterisk |
| 10:49.30 | Kalidarn | hmm, yeah kernel panic but yeah |
| 10:49.34 | Kalidarn | now it says AUSTRALIA mode |
| 10:49.40 | geemee | DennisG: I will look into Dundi. However the other box is very old and reluctant to upgrade will this be a problem? |
| 10:49.44 | m0t3jl | Hi, would someone mind explaining me the equipment I need to buy when I want to connect Asterisk PC to some 4 state lines and about 15 office phones? I'm not really sure that I understand the logic involved, what is it with the cards and modules and stuff? ;) Thanks a lot |
| 10:50.17 | Kalidarn | that said freebsd probably hasn't got the newest zaptel driver |
| 10:50.18 | m0t3jl | I have a working test environment with just pure VoIP-enabled Asterisk. |
| 10:51.07 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 10:53.25 | *** join/#asterisk fors1 (~forsen@195.189.143.116) |
| 10:56.19 | SiNGLer | m0t3jl: you need to connect lines and phones to PC. you can use internal cards, or converters (gateways) to SIP, and connect via ethernet to server |
| 10:57.48 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 10:58.23 | m0t3jl | SiNGLer, that much I know, but what are the advantages/disadvantages of using for example internal card instead of a SIP gateway? |
| 10:58.57 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 10:59.04 | m0t3jl | SiNGLer, I need fax support, so I probably need an internal card so faxes are not digitalized and dedigitalized ;) |
| 10:59.30 | UQlev | m0t3jl: internal card for desktops only, and it is powered from PC |
| 10:59.36 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 10:59.36 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 11:00.00 | SiNGLer | I personally think that is available, internal card should be used |
| 11:00.18 | SiNGLer | saves power, space, better quality |
| 11:00.22 | UQlev | m0t3jl: it worth to have traditional telephone network apart from asterisk, imho |
| 11:00.25 | *** part/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 11:01.23 | m0t3jl | UQlev, not sure what you mean ;) |
| 11:01.51 | UQlev | m0t3jl: I mean if your asteris server is down, no communications? |
| 11:01.53 | m0t3jl | SiNGLer, I personally think that too. It seems to me that it should even be a bit cheaper than external gateways |
| 11:02.21 | m0t3jl | UQlev, What do you propose? ;) |
| 11:02.30 | *** join/#asterisk markitoxs (~miranda83@lumison-gw.dub.ftuk.net) |
| 11:02.33 | markitoxs | hello |
| 11:02.36 | UQlev | m0t3jl: see above |
| 11:03.01 | SiNGLer | if server is down, you can use mobile phone :P |
| 11:03.05 | markitoxs | how can i set the rtp TOS flag? |
| 11:03.43 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 11:03.52 | SiNGLer | backup telephony can be implemented but it depends if it pays off. |
| 11:04.31 | markitoxs | tos_audio=EF ? |
| 11:04.36 | UQlev | flat rate for pstn is not that high |
| 11:05.16 | m0t3jl | SiNGLer, that's my concern, we have lots of mobile phones, so it's more likely Asterisk will be the backup in case mobile phones are down :D |
| 11:05.56 | SiNGLer | UQlev: but you will not install analog and VoIP phones on every worldspace |
| 11:06.14 | SiNGLer | and mobile rates aren't high too |
| 11:06.18 | UQlev | m0t3jl: main reason of VoIP to reduce call charges using termination providers |
| 11:07.40 | m0t3jl | UQlev, what I'm trying to do here is to replace our old analogue PBX capable of the same stuff with added features like VoIP (so our customers who have VoIP can call us free of charge), voice menus, queues, etc |
| 11:08.13 | m0t3jl | oh half the sentence I wrote just disappeared ;) ... analogue PBX with Asterisk PBX capable of ... |
| 11:08.22 | UQlev | m0t3jl: afaik in Praha call rates are not the same as in USA ;P |
| 11:09.10 | *** join/#asterisk zorp75ck (~zorp75ck@pool-71-162-38-96.altnpa.east.verizon.net) |
| 11:09.21 | m0t3jl | UQlev, actually I'm not in Prague, I live a bit to the east of Prague ;) |
| 11:09.28 | OlafsenM | guys, help |
| 11:09.30 | OlafsenM | "chan_dahdi.c: !! Got reject for frame , but we have nothing -- resetting! " |
| 11:09.44 | OlafsenM | what's happening? |
| 11:09.50 | UQlev | m0t3jl: are there cheaper telephones? :) |
| 11:09.56 | doolittlework | WIMPy: you still here? |
| 11:11.23 | *** part/#asterisk lukhas (~lucas@bearstech/lukhas) |
| 11:11.23 | doolittlework | WIMPy: thank you callerid(name) works like a charm |
| 11:11.45 | m0t3jl | But back to my original question. Suppose I want to replace our old analogue PBX with Asterisk without changing much of the stuff the users are used to. I think I'll go with the internal cards and I won't think about failover. I figured out that when I want to buy such cards I don't only need to buy the card itself, but also some modules, is that true? |
| 11:11.51 | m0t3jl | UQlev, I guess not |
| 11:12.15 | m0t3jl | UQlev, I usually take USA as an example of a cheaper country as far as communications are concerned ;) |
| 11:12.42 | doolittlework | m0t3jl: why dont you use the legasy as an fxs, isdn gateway? |
| 11:13.01 | SiNGLer | m0t3jl: user will use analog or VoIP phones? |
| 11:13.58 | SiNGLer | and your state lines are PRI or BRI, I guess BRI, because you have only 15 users |
| 11:14.30 | doolittlework | if a customer does not want to upgrade to asterisk on the fly because of cost, it is always better to link it to the old system then fade it out as money becomes available |
| 11:15.27 | UQlev | m0t3jl: voice menues, queues and other crap are just fassion, it is not helpful for business |
| 11:15.52 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 11:15.53 | doolittlework | I have successfully integrated asterisk to work with Philips, siemens, avaya, samsung, plessey, nortel, ericson, keeping the feutures of the legacy plus all the bonus uf having asterisk |
| 11:16.25 | UQlev | doolittlework: asterisk server itself is very little investment |
| 11:16.41 | doolittlework | if i dial from a softphone is there a way to change what is on the screen, while in a call? |
| 11:16.50 | sawgood | doolittlework: what type of IAD do you use to support the legacy key system? |
| 11:17.00 | m0t3jl | SiNGLer, they use analog and I don't think we'll be switching to VoIP any time soon ;) |
| 11:17.06 | doolittlework | UQlev: the cost of the phones is what scares the customers off |
| 11:17.22 | m0t3jl | SiNGLer, PRI is ISDN? If yes, then we have BRI ;) |
| 11:18.05 | SiNGLer | PRI and BRI are basically ISDNs. PRI is E1 in Europe (30 channels), BRI - 2 channels |
| 11:18.44 | doolittlework | sawgood: use pri, bri to link to system |
| 11:18.53 | m0t3jl | SiNGLer, oh, then I think it's BRI, we have 4 lines (cables) as far as I know |
| 11:19.08 | UQlev | m0t3jl: changing traditional telephone to VoIP phone will not give your users any additional comfort or benefits |
| 11:19.14 | m0t3jl | SiNGLer, and you can only have one call at a time (using one of the lines). |
| 11:19.16 | SiNGLer | I would use card to connect state lines to server, use analog card for fax (or use analog line, if possible), connect users via analog-sip gateway |
| 11:19.23 | m0t3jl | UQlev, I am not doing that |
| 11:19.56 | SiNGLer | m0t3jl: wait a minute, one call per line? maybe it is analog line? |
| 11:20.09 | SiNGLer | can you connect ordinary phone to that line? |
| 11:20.10 | m0t3jl | UQlev, I am replacing the old malfunctioning analogue PBX with Asterisk. |
| 11:20.12 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 11:20.13 | UQlev | m0t3jl: Install pilot asterisk server with softphones only let users see what is this |
| 11:20.16 | m0t3jl | SiNGLer, it is ;) |
| 11:20.31 | SiNGLer | oh |
| 11:20.32 | SiNGLer | :) |
| 11:21.23 | m0t3jl | UQlev, gosh, I am talking to a wall :D I've already said that earlier. I have a working Asterisk server with VoIP accounts. Now I want to connect it to our phone network and slowly replace the old analogue PBX ;) |
| 11:21.46 | SiNGLer | then you'll need FXO and FXS cards/modules to connect lines and users. |
| 11:22.08 | doolittlework | m0t3jl: do you need help doing this |
| 11:22.38 | m0t3jl | SiNGLer, that I know, but I can't figure out which cards work well with Asterisk and if I just buy a card and that's it or if I do have to buy also those modules they talk about ;) |
| 11:23.34 | m0t3jl | doolittlework, I am trying to find out which hardware should I buy to be able to connect my Asterisk to our phone network ;) |
| 11:24.30 | SiNGLer | I can say only about Sangoma cards, because I work with them (but never used FXO/FXS cards, only digital). You need a card, and module to that card, depends what you need FXO, or FXS modules. one module can support a few lines, don't remember correctly. |
| 11:25.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:25.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:26.57 | m0t3jl | SiNGLer, so when I buy a card with let's say 8 phone lines, I have to buy 8 modules for that card? |
| 11:27.20 | leifmadsen | yes |
| 11:27.37 | leifmadsen | 1 module = 1 port |
| 11:27.39 | SiNGLer | not exactly |
| 11:27.43 | m0t3jl | :( |
| 11:28.01 | SiNGLer | example is Sangoma's BRI card (A500). One module is 2 lines |
| 11:28.02 | doolittlework | m0t3jl: if you are using 2 bri(4 lines) the b410p is the best 4 port bri, you can use to from telco to asterisk and 2 to link to your old pbx |
| 11:28.05 | m0t3jl | I would never thought there's enough room for that on one card :D |
| 11:28.52 | m0t3jl | doolittlework, the old PBX will be burned at least ;) It's malfunctioning :D |
| 11:30.29 | doolittlework | m0t3jl: never kill old technology, if it was not for the old we would not have the new |
| 11:30.49 | *** part/#asterisk markitoxs (~miranda83@lumison-gw.dub.ftuk.net) |
| 11:31.21 | m0t3jl | doolittlework, I don't have anything against the old, but it's malfunctioning. I got hit by a lightning bolt ;) |
| 11:31.27 | m0t3jl | It ;) |
| 11:31.41 | doolittlework | so where is the guru, i need to send a ref number to a phone whiles in a call, pref display it on the screen of eyebeam if possible |
| 11:31.58 | *** join/#asterisk bklang (~bklang@tesla.alkaloid.net) |
| 11:32.01 | doolittlework | m0t3jl: are you ok??? |
| 11:32.32 | doolittlework | natures death row |
| 11:32.43 | m0t3jl | doolittlework, it was the PBX that got hit, not me, though ;) |
| 11:32.57 | m0t3jl | doolittlework, so I am fine, thanks for asking ;) |
| 11:33.20 | UQlev | m0t3jl: do you think Asteris will protect itself from surges? |
| 11:34.04 | m0t3jl | UQlev, I don't |
| 11:34.32 | SiNGLer | m0t3jl: take a look at sangoma's A400 and/or A200. One module is 2 lines. you'll need 2 modules for analog and 8 modules for users. |
| 11:34.42 | m0t3jl | UQlev, it's just that it's impossible to repair the old PBX, so we're switching to Asterisk |
| 11:35.10 | m0t3jl | SiNGLer, so there's not place for 8 modules, but only 4 ;) |
| 11:35.37 | SiNGLer | m0t3jl: you need expansion card, see pictures |
| 11:35.59 | UQlev | m0t3jl: those zap cards will cost you more than local PBX |
| 11:36.15 | m0t3jl | SiNGLer, so there's a baseboard, to which I connect daughter cards, and then I have the modules? ;) |
| 11:36.56 | SiNGLer | where is base card, to which you connect modules, if you need more, you connect daughter card. It's cheaper than buying additional card |
| 11:37.20 | SiNGLer | and cards are synced this way |
| 11:37.39 | m0t3jl | SiNGLer, I was thinking that way |
| 11:38.02 | m0t3jl | SiNGLer, so it is more stable when I use daughter cards instead of using more basecards? |
| 11:38.12 | SiNGLer | yes |
| 11:38.13 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 11:38.49 | SiNGLer | also you can go with SIP gw. buy one (4 lines) for state lines. and one bigger for users. connect fax directly to line for better quality, of live with little degraded faxs |
| 11:39.29 | m0t3jl | SiNGLer, they are not little degraded, by the way, from like 15 tries, I got one received :D |
| 11:39.30 | SiNGLer | I worked with Audiocodes devices, Can't say about others |
| 11:39.41 | SiNGLer | It depends on config |
| 11:39.56 | SiNGLer | I managed to get good results |
| 11:40.44 | SiNGLer | but maybe your faxes had higher speed, then where would be more degradation |
| 11:40.55 | SiNGLer | also you can terminate fax on server |
| 11:42.17 | doolittlework | what asterisk application contols what is displayed on the eyebeam screen |
| 11:43.56 | m0t3jl | SiNGLer, I tried that as well, but it did not work either. There was some error that Asterisk is not able to recognize the image data ;) |
| 11:44.19 | SiNGLer | I use iaxmodem+hylafax |
| 11:44.48 | SiNGLer | from asterisk dial iaxmodem. Hylafax picks it up and proccesses |
| 11:44.49 | m0t3jl | And you terminate faxes on server? |
| 11:44.59 | SiNGLer | yes |
| 11:45.14 | m0t3jl | And then you send them via e-mail? |
| 11:45.18 | SiNGLer | yes |
| 11:45.53 | WIMPy | BTW: There is no support for G4 fax in any of the solutions, is there? |
| 11:46.57 | m0t3jl | WIMPy, it is when you connect the fax directly to a state line :D |
| 11:47.19 | SiNGLer | and fax machine supports it :) |
| 11:47.24 | WIMPy | That even works without Asterisk. I know :-) |
| 11:47.41 | SiNGLer | yes, a failover! :) |
| 11:47.45 | m0t3jl | SiNGLer, do you have any experience with Sangona A400E? |
| 11:48.38 | SiNGLer | nop, but I guess it wouldn't be difficult |
| 11:49.34 | m0t3jl | SiNGLer, it does not have any ports directly on it, instead it has some sort of LPT port-like connector, do I get a special box with phone line connectors with that? |
| 11:50.46 | SiNGLer | 12 ports would fit on card. you get cable, and you connect to phones via cross (don't know if it is correct name in english) |
| 11:50.57 | SiNGLer | *would not |
| 11:51.48 | SiNGLer | I guess legacy PBX was connected similar way |
| 11:52.19 | SiNGLer | I googled image: http://www.hyperline.com/img/sharedimg/cross/krn-plint-03.jpg |
| 11:52.36 | SiNGLer | something like that |
| 11:52.42 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 11:52.47 | m0t3jl | SiNGLer, what the bloody hell is that ugly-looking thing? ;) |
| 11:53.08 | m0t3jl | SiNGLer,oh |
| 11:53.27 | m0t3jl | SiNGLer, I don't use connectors but directly wire the phone cable to that, am I right? |
| 11:53.30 | SiNGLer | And I saw at sangomas site they say, that can provide Y cable to connect 2 cards to one to connect to similar things |
| 11:53.52 | SiNGLer | yes, ex lower wire - phone, upper - card |
| 11:54.01 | doolittlework | WIMPy: thanks for the help on ethe incoming ref number using callerid(name) is there a way to send sip info messages to eyebeam softhone whilis in a call |
| 11:54.19 | SiNGLer | and you pair it at digits. first wire on left of "1", second on right |
| 11:55.12 | m0t3jl | SiNGLer, I think I'll have some telco guy do that wiring for me ;) |
| 11:55.17 | SiNGLer | :) |
| 11:55.50 | SiNGLer | and using that thing is easier to manage everything, if needed you can tap into line, or insert a fuse |
| 11:57.24 | WIMPy | doolittlework: I don't think so. You might be able to use system(sipsak...). But I can't help on that one. |
| 11:58.05 | m0t3jl | SiNGLer, nice |
| 11:58.48 | doolittlework | ta thx for the direction |
| 11:59.46 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 12:05.56 | *** join/#asterisk modsaid (~modsaid@196.218.219.102) |
| 12:07.26 | *** join/#asterisk gnusiva (~gnusiva@77.31.72.100) |
| 12:08.17 | gnusiva | hi all, is it possible to connect a gsm modem to asterisk instead of pstn line? |
| 12:10.09 | WIMPy | Not the calssic 'modem' things AFAIK, but the are cahnnels for USB sticks and Phones via BT. Or you can use SIP-GSM gateways. |
| 12:12.50 | WIMPy | finally got a phone connected via dahdi on 1.8. *fanfare* |
| 12:14.21 | hrhrhr | gratz |
| 12:14.24 | hrhrhr | what card? |
| 12:14.48 | hrhrhr | im gonna have to move our pri card over to dahdi at some point |
| 12:15.17 | russellb | also check out chan_datacard - http://forge.asterisk.org/gf/project/chan_datacard/ |
| 12:15.50 | WIMPy | Junghanns |
| 12:19.49 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:21.19 | m0t3jl | SiNGLer, damn. I was just told that one of our lines is actually an ISDN ... |
| 12:22.02 | m0t3jl | SiNGLer, so there's actually 3 analogue lines and one PRI (it has 4 channels). |
| 12:22.38 | WIMPy | m0t3jl: Sounds a little unlikely. Are you sure? |
| 12:22.41 | *** join/#asterisk joobie (~joobie@CPE-121-219-40-191.lnse1.lon.bigpond.net.au) |
| 12:22.55 | m0t3jl | WIMPy, what is unlikely? |
| 12:23.21 | WIMPy | A PRI with only 4 channels. You'd use 2 BRIs for that. |
| 12:23.39 | m0t3jl | WIMPy, the ISDN line is not currently connected to the PBX, it has one phone of its own. It's sort of detached office, but in the future we would like to connect it to Asterisk ;) |
| 12:24.12 | WIMPy | If it has a phone connected it's definitely not PRI. |
| 12:24.13 | hrhrhr | can you even use 4 channels from pri |
| 12:24.18 | m0t3jl | WIMPy, oh, then it's probably 2 BRIs. I know there are four numbers on the line, but we only use one. |
| 12:24.29 | WIMPy | hrhrhr: In theory... |
| 12:24.31 | hrhrhr | i don't think you can get provisioned on less than 6/8 around here |
| 12:24.54 | joobie | guys |
| 12:25.00 | joobie | anyone know much about isdn redundancy? |
| 12:25.10 | joobie | i pay my provider for a PRI30 |
| 12:25.14 | WIMPy | I think the concept of fractional PRIs was ditched here some years back. |
| 12:25.15 | joobie | if they drop |
| 12:25.17 | SiNGLer | m0t3jl: you can have many numbers on one BRI/PRI line |
| 12:25.22 | joobie | how can i ensure my ISDN doesnt with them? |
| 12:25.55 | WIMPy | joobie: What's your question? |
| 12:26.14 | SiNGLer | m0t3jl: I think you should find out exactly how many which types you do have, then plan accordingly |
| 12:26.44 | joobie | WIMPy, is there a way to ensure ISDN redunancy in the event that the provider you use for ISDN drops? |
| 12:26.53 | WIMPy | Or even better, try to tidy up in that process. |
| 12:26.54 | joobie | is there like some sort of failover technology for ISDN that can be used?? |
| 12:27.02 | WIMPy | joobie: Get a 2nd line. |
| 12:27.24 | joobie | WIMPy, how will that help with routing of the numbers that are assigned to the ISDN trunk with Telco#1 when they go down? |
| 12:27.30 | SiNGLer | m0t3jl: you can check your bills or use analog phone to check which lines will work with it |
| 12:27.32 | OlafsenM | "chan_dahdi.c: !! Got reject for frame 119, but we have nothing -- resetting! " |
| 12:27.36 | WIMPy | joobie: Or forward clls in case of failure. |
| 12:27.37 | OlafsenM | what's wrong |
| 12:27.37 | OlafsenM | ? |
| 12:27.41 | OlafsenM | libpri 1.4.10.2 |
| 12:27.47 | joobie | WIMPy, telco is DOWN |
| 12:27.50 | joobie | aka they cant forward |
| 12:27.53 | joobie | their shit is off the air |
| 12:28.12 | joobie | is there some failure mechanism that can be used in ISDN to revert the numbes to another trunk.. something in the industry as a standard for this? |
| 12:28.17 | WIMPy | joobie: If your telco goes down, your screwd, unconditionally. |
| 12:28.22 | joobie | if they are down, i cant ring them to ask them to forward the numbers |
| 12:28.33 | joobie | u serious? there's NO failover mechanism in ISDN? |
| 12:28.40 | joobie | for example in IP, there is BGP |
| 12:28.43 | WIMPy | There is |
| 12:28.49 | joobie | wher eyou can setup a BGP relationshp with another isp |
| 12:28.50 | m0t3jl | SiNGLer, that's how I found out about the ISDN line ;) Totally it's 5 FXO, 4 EuroISDN numbers and 16 office phones... |
| 12:28.55 | WIMPy | But there is not fail over for an entire telco. |
| 12:28.55 | joobie | and then route the same ip to their equipment |
| 12:29.14 | joobie | WIMPy, not an entire telco, just my ISDN circuit with the numbers that are associated to it |
| 12:29.36 | WIMPy | joobie: Then it's back to my first answer: Get a 2nd line. |
| 12:29.47 | SiNGLer | m0t3jl: how many ISDN lines do you have? 4 numbers can be on one or 2 lines |
| 12:30.00 | joobie | WIMPy, then it's back to my 2nd comment - the telco is down and they cant forward when they are down |
| 12:30.32 | *** join/#asterisk michael-i (~michael-i@141.41.40.118) |
| 12:30.36 | m0t3jl | SiNGLer, by the looks of the bill it seems like one line |
| 12:30.53 | WIMPy | joobie: Now what is your question? Is it about a line being down or a telco being down? |
| 12:31.11 | joobie | WIMPy, i have numbers assigned to my ISDN from my telco |
| 12:31.21 | joobie | if my telco goes down, how can i make those numbers accessible |
| 12:31.29 | joobie | .. if they cant forward because their systmes are down |
| 12:31.41 | joobie | .. with IP, if this happened, BGP would handle this and route the same IP over a different route |
| 12:31.57 | joobie | my question is, is there s similar thing i can get the telco to setup which will be able to route the numbers over a different trunk |
| 12:32.05 | joobie | is there a "standard" to handle this type of siutaiton |
| 12:32.07 | WIMPy | You get a 2nd line with the same numbers. |
| 12:32.24 | joobie | and you can do that with 2 providers? |
| 12:32.29 | WIMPy | And yes you can also use forwarding, as I already wrote. |
| 12:32.29 | joobie | how so .. what technology is it called? |
| 12:32.41 | joobie | forwarding i cannot do if the telco is down |
| 12:32.50 | joobie | ive had this situation where my telco had a complete outage in their NOC |
| 12:32.57 | WIMPy | Forwarding would work with a different telco. |
| 12:33.00 | joobie | they couldnt forward numbers, so the numbers just went to a busy tone |
| 12:33.19 | WIMPy | You do the forwarding before the line goes down. |
| 12:33.24 | joobie | but if telco#1 went down, whom owns the numbers, forwarding wouldnt wokr?!? |
| 12:33.27 | m0t3jl | SiNGLer, do you know if it's possible to have that numbers transfered? Like moving all the FXO numbers to ISDN or vice versa? |
| 12:33.41 | joobie | i'm talking about unscheduled outages |
| 12:33.52 | m0t3jl | SiNGLer, I don't quiet like the idea of using two technologies... |
| 12:33.59 | joobie | dood |
| 12:34.02 | WIMPy | m0t3jl: That would sound like a sensible idea. |
| 12:34.03 | joobie | is there something like BGP |
| 12:34.07 | joobie | for ISDN indials? |
| 12:34.09 | WIMPy | joobie: YES! |
| 12:34.12 | joobie | it sounds like there isnt from what you're saying |
| 12:34.16 | joobie | forwarding is NOT BGP liek |
| 12:34.23 | joobie | if the telco #1 drops, so does the forwarding |
| 12:34.26 | joobie | what dont you get about that? |
| 12:34.33 | WIMPy | You setup forwarding in case of failure, just like in case of busy. |
| 12:34.47 | joobie | what is that called |
| 12:34.51 | joobie | "forwarding in csae of failure" |
| 12:34.55 | WIMPy | You still don't seem to know what your question is. |
| 12:34.57 | joobie | is there a proper industyr term for this? |
| 12:35.18 | joobie | i know my question, you don't understand |
| 12:35.25 | joobie | you're going aroudn in circles |
| 12:35.26 | WIMPy | Yes, but it doesn't spring to mind ATM. |
| 12:35.27 | SiNGLer | m0t3jl: not sure how it is in your country, but it is possible |
| 12:35.30 | joobie | and im getting pissed |
| 12:35.37 | joobie | fuk |
| 12:35.39 | joobie | that term |
| 12:35.40 | joobie | is what i need |
| 12:35.55 | WIMPy | Ask someone else, please. |
| 12:36.33 | joobie | WIMPy, my penis is hard |
| 12:36.55 | WIMPy | Great. Try youporn, then. |
| 12:37.40 | m0t3jl | SiNGLer, damn the fool that invented this system :D |
| 12:38.03 | SiNGLer | blame the progress :P |
| 12:38.05 | m0t3jl | SiNGLer, why would he buy an ISDN line if he had already 5 FXOs ;) |
| 12:38.41 | SiNGLer | to use more numbers on one line? to save on wires? because ISDN = 2 analogs |
| 12:38.45 | WIMPy | m0t3jl: Better ask the other way round. Then the answer is probably: Because of the old PBX. |
| 12:38.50 | SiNGLer | maybe it was cheper |
| 12:38.52 | SiNGLer | *cheaper |
| 12:38.56 | m0t3jl | SiNGLer, I know, but we presently still use only one of the numbers :D |
| 12:39.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:39.51 | m0t3jl | SiNGLer, I'm now looking on some of the prices they have for ISDN cards. Do I get it right that on one of the ports I get two numbers? |
| 12:40.00 | joobie | hey WIMPy that's a good site |
| 12:40.33 | WIMPy | m0t3jl: Usually you can get up to 10 or alternatively a DDI block. |
| 12:40.41 | SiNGLer | m0t3jl: what exactly do you look at? one ISDN line can have 2 calls at one time. Numbers can be much more |
| 12:41.23 | [TK]D-Fender | BRI = 2 channels. |
| 12:41.29 | WIMPy | And that's two active calls, BTW. |
| 12:42.04 | m0t3jl | WIMPy, oh, so it's like that. Maybe we have one line with four numbers, but only 2 channels. |
| 12:42.36 | WIMPy | Absolutely possible. |
| 12:42.54 | m0t3jl | But when there are four ports on an ISDN card, I could be having 8 channels, therefore 8 simultaneous calls, right? |
| 12:43.12 | WIMPy | right |
| 12:43.31 | m0t3jl | That's what I call clever ;) |
| 12:43.33 | SiNGLer | depends on card, sangoma's A500 have Y cable, su you can connect 2 BRI's to one port |
| 12:44.02 | SiNGLer | A500 use jack with 8 pins, BRI need only 4 :) |
| 12:44.25 | SiNGLer | I mean socket RJ-11 - 4 pins, RJ-45 - 8 |
| 12:44.26 | WIMPy | I'd prefer to call that two ports on one socket. |
| 12:44.37 | SiNGLer | agreed |
| 12:46.30 | m0t3jl | By the looks of it I'd be more happier if they could take the one number we use from the ISDN and make it an analogue FXO ;) |
| 12:46.57 | m0t3jl | Cause the cards are cheaper and I could use the same card and daughter card I would use for the office phones... |
| 12:47.38 | WIMPy | Err, no. BRI cards cost next to nothing. And you have a lot less trouble. |
| 12:47.40 | SiNGLer | But then you will not be able to receive multiple calls on one number |
| 12:48.02 | m0t3jl | WIMPy, next to nothing? |
| 12:48.05 | WIMPy | And are much more flexible, right. |
| 12:48.05 | SiNGLer | maybe it was a reason for ISDN? |
| 12:48.23 | m0t3jl | WIMPy, here the A400E costs 8K CZK, the A500 costs 10K CZK ;) |
| 12:48.38 | WIMPy | m0t3jl: <30 EUR in the next consumer electronics store. |
| 12:49.13 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 12:49.13 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:50.23 | m0t3jl | WIMPy, are we living in the same world? 30 EUR is like 762 CZK... I don't think I can get an ISDN card for that kind of money here... |
| 12:51.07 | WIMPy | I'm pretty sure, you can. Just go for a normal one, not one that has voice applications printed on the package. |
| 12:51.22 | m0t3jl | WIMPy, what is a normal one? |
| 12:51.50 | m0t3jl | SiNGLer, how much is that E500 you were talking about worth? |
| 12:51.50 | WIMPy | There are tons of el-cheapo ones, like, e.g. Longshine. |
| 12:51.53 | SiNGLer | does voipango.de sell cheap ISDN cards? |
| 12:52.05 | SiNGLer | m0t3jl: A500? |
| 12:52.12 | WIMPy | I don't think they sell the cheap ones. |
| 12:52.28 | m0t3jl | SaiSoma, the ISDN one |
| 12:52.32 | *** part/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 12:52.52 | m0t3jl | SaiSoma, ups, sorry about that |
| 12:52.56 | m0t3jl | SiNGLer, the ISDN one |
| 12:52.57 | SaiSoma | m0t3jl: np:) |
| 12:53.17 | SiNGLer | check voipango.de there prices are fair |
| 12:53.55 | WIMPy | If you need a multiport one, check the Junghanns.net ones. |
| 12:54.01 | SiNGLer | not sure from where we get them, afaik there are sangoma's distributor in Poland, but I personally don't buy them |
| 12:54.19 | SiNGLer | my boss does :) |
| 12:55.03 | m0t3jl | SiNGLer, lol, the A500BRM would cost something about 6K CZK (converted from EUR) there ;) |
| 12:55.28 | SiNGLer | :) |
| 12:56.45 | m0t3jl | THat's like 150 EUR difference, man ;) |
| 12:57.43 | SiNGLer | maybe that other is with echo canceller? |
| 12:58.14 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:58.41 | m0t3jl | SiNGLer, echo canceller costs 150 EUR? |
| 12:59.04 | SiNGLer | at voipango.de version with hardware ec is 483,46 euro |
| 12:59.19 | WIMPy | Probably not. They are really expensive. But can be very helpfull as well. |
| 13:00.58 | joobie | anyone want to setup a trunk to AU with me |
| 13:01.03 | joobie | i got free local calls |
| 13:01.17 | joobie | need a UK / US trunk preferably |
| 13:02.40 | m0t3jl | joobie, AU is Australia? |
| 13:02.45 | joobie | ya |
| 13:02.56 | m0t3jl | jonmasters, pitty, we don't call there much ... |
| 13:03.39 | m0t3jl | SiNGLer, if that version had ec it would be a lot cheaper than the voipandgo.de, which I doubt... |
| 13:04.04 | SiNGLer | m0t3jl: does it have D at the end of model? |
| 13:04.15 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:04.36 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:05.21 | m0t3jl | SiNGLer, it's just A500 |
| 13:05.31 | SiNGLer | oh |
| 13:05.45 | m0t3jl | SiNGLer, why does it have 3 ports, when the modules are 2 port? ;) |
| 13:06.24 | SiNGLer | because there are cables which split, and you can connect 2 port on one socket |
| 13:06.42 | SiNGLer | we discussed this earlier |
| 13:07.05 | m0t3jl | SiNGLer, I know you talked about it, but I did not realize it |
| 13:07.09 | SiNGLer | sangoma has rj45, which is 8 wires. rj11 is 4 wires |
| 13:07.34 | SiNGLer | so first 4 wires are for first isdn, later 4 for second |
| 13:08.14 | m0t3jl | So one port on that card is actually two lines? |
| 13:08.23 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 13:08.27 | SiNGLer | yes |
| 13:09.02 | SiNGLer | I have to go afk, will be back in 30min |
| 13:09.12 | m0t3jl | SiNGLer, no problem, thanks for the advice, though |
| 13:10.03 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:11.22 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:19.50 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 13:21.02 | *** part/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:24.37 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:24.52 | *** part/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:29.08 | ruyo | Aynone installed bristuff on a Debian Lenny? |
| 13:29.27 | ruyo | Thinking on trying that instead of mISDN... |
| 13:30.15 | sawgood | I have a situation/need to solve a concern for an Asterisk 1.6.x box: I have a group of 4 SIP phones ... which all need to 'act' as one single extension (not as a ring group) ... is this even possible? |
| 13:30.57 | [TK]D-Fender | sawgood: What is the functional difference? |
| 13:31.31 | ruyo | Since a SIP account is associated to an IP address I don't think it's possible to use the same account on 4. |
| 13:31.34 | sawgood | might be symantics ... but ... call one number (four phones ring) because any work at any of the four phones all do the same job |
| 13:31.57 | sawgood | any worker I mean at any of the 4 phones all do the same job |
| 13:32.02 | [TK]D-Fender | sawgood: How does a "ring group" (shit term) not do that? |
| 13:32.14 | ruyo | 601,1,Dial(SIP/601&SIP/602&SIP/603) doesn't do the trick? |
| 13:32.30 | sawgood | well, if a VM is left for a ring group can it be left on all four phones at once? |
| 13:32.35 | [TK]D-Fender | sawgood: "make 4 phones ring where only 1 might be expected for a similar action" <- sounds the same to me |
| 13:33.07 | [TK]D-Fender | sawgood: You don't leave VM on a PHONE. VM is in ASTERISK, and * can inform whatever phones about whatever mailboxes you want |
| 13:33.24 | ruyo | You can use the same mailbox=xxx@xxx for them. |
| 13:33.30 | sawgood | good point ... |
| 13:34.02 | sawgood | well, this leads me to another sub-variable ... which I'll leave off for the moment ... |
| 13:34.16 | ruyo | CallerID? =P |
| 13:34.26 | sawgood | I cannot recall exactly why I felt like a ring group would be different then calling four phones at once |
| 13:34.29 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:34.36 | sawgood | But, now I'll go back to the LAB and set it up .. |
| 13:35.11 | sawgood | The client wants the four phones in the 'staff A room' to all ring anytime someone needs help from the 'staff a people' ... |
| 13:35.33 | sawgood | sometimes, one phone will ring and the worker is not at their desk, but the other 3 staff members are (but their phone did not ring) |
| 13:35.55 | sawgood | I call it lazy worker not picking up a known call for them, but they say, well I did not hear my phone ring |
| 13:36.11 | jamko | Then create an extension that they can dial when they want all phones to ring. |
| 13:36.55 | ruyo | Yes, like and extension that represents the whole room. |
| 13:36.59 | ruyo | Or a queue. |
| 13:37.18 | sawgood | So, let me ask this sub-variable ... (if the four phones are extensions 101,102,103, and 104) and I have a ring group of x400 (which calls all four phones) |
| 13:37.27 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 13:37.44 | sawgood | How can I 'tell' Asterisk to deliver the voicemail left for any missed call to x400 to go to 101-104? |
| 13:37.49 | [TK]D-Fender | sawgood: Phones are not extensions |
| 13:37.57 | [TK]D-Fender | sawgood: And AGAIN, there is no DELIVER |
| 13:37.58 | jamko | sip.conf mailbox= |
| 13:38.10 | jamko | voicemail.conf |
| 13:38.17 | [TK]D-Fender | sawgood: Devices are INFORMED about VM. Everything else is dialplan. |
| 13:38.26 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:38.50 | sawgood | [TK]D-Fender: perfect answer ... let me try to re-word this |
| 13:39.20 | [TK]D-Fender | sawgood: And if you want a single VM to be saved as SEPARATE COPIES into multiple mailboxes, that is another matter |
| 13:39.36 | jamko | sip.conf peer definitions, tell asterisk which phones to inform of voicemails in specific mailboxes. One peer / friend can have as many mailboxes assigned to it as you want. |
| 13:39.38 | sawgood | Imgine a SIP phone with a MWI light ... it goes red when a VM is 'ready' |
| 13:39.46 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 13:39.56 | [TK]D-Fender | sawgood: mailbox=101@default |
| 13:40.00 | [TK]D-Fender | sawgood: mailbox=400@default |
| 13:40.02 | [TK]D-Fender | DONE |
| 13:40.07 | [TK]D-Fender | WATCH TWO BOXES |
| 13:40.20 | sawgood | The light only goes RED if a VM is for x101 for example ... how do I make it go RED for x400? |
| 13:40.32 | sawgood | Its really that easy? |
| 13:40.36 | jamko | yes |
| 13:40.39 | ruyo | mailbox=x400@default |
| 13:40.47 | jamko | lol .. that was easy. |
| 13:40.51 | ruyo | In all the phones. |
| 13:41.02 | sawgood | Generally, the web GUI for the phones only support one entry for VM |
| 13:41.04 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 13:41.06 | rocksfrow | hey guys, http://www.mentby.com/chris-miller-3/asterisk-gplonly-dependency-in-asterisk-addons-rpm.html |
| 13:41.15 | rocksfrow | this guy complaining about the issue, but doesn't bother letting others know how to fix it |
| 13:41.21 | rocksfrow | how can I get YUM to force install res_fax updates? |
| 13:41.26 | [TK]D-Fender | sawgood: FUCK THE FUCKING GUI |
| 13:41.39 | rocksfrow | please help |
| 13:41.39 | [TK]D-Fender | sawgood: You should know better than even refer to it here |
| 13:41.43 | SiNGLer | m0t3jl: I have 20min before I go home, if you need, I can answer your questions |
| 13:41.57 | rocksfrow | Error: asterisk14-res_fax_digium conflicts with asterisk14-addons-core |
| 13:42.03 | ruyo | You can make the phone call a VoiceMailMain extension and _then_ you log in with whatever account you want. |
| 13:42.08 | jamko | Yea don't use the phone gui for anything more than you have to. |
| 13:42.19 | sawgood | I was talking about the setup of the individual phone and its process to register |
| 13:42.22 | [TK]D-Fender | sawgood: and You have a general entry block where you can enter other parms for your "extensions" and can add it THERE |
| 13:42.42 | sawgood | you are right, sir ... sorry to bother you .. I guess its too early in the AM |
| 13:42.56 | jamko | rocksfrow: why don't you just uninstall and recompile... |
| 13:43.02 | rocksfrow | i've done this previously but forget how to get YUM to force install this package, or do I need to use rpm directly |
| 13:43.06 | rocksfrow | i'm installing via repos, jamko |
| 13:43.11 | rocksfrow | and because it's a live system, lol. |
| 13:43.28 | rocksfrow | research shows it's simply a license issue and it IS safe to force install |
| 13:43.36 | jamko | rocksfrow: never update live system, or it may not be live anymore., |
| 13:43.37 | rocksfrow | i just can't find how to force install |
| 13:43.40 | rocksfrow | .... |
| 13:44.00 | rocksfrow | well, same goes for uninstalling/recompiling |
| 13:44.05 | rocksfrow | does anybody know how to force rpm install of this package? |
| 13:44.23 | *** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-196-191.pools.arcor-ip.net) |
| 13:44.28 | [TK]D-Fender | rocksfrow: man rpm <------- |
| 13:44.31 | jamko | point taken, yes, but in that scenario you should put up a backup box. |
| 13:44.43 | rocksfrow | jamko, i'm updating the backup box first |
| 13:44.46 | rocksfrow | then if all goes ok, updating the live box |
| 13:44.51 | rocksfrow | i was lieing when i said its live |
| 13:44.52 | rocksfrow | lol. |
| 13:44.55 | [TK]D-Fender | rocksfrow: Glorius list of little 1-char codes that like DO STUFF and such awaits you... |
| 13:45.00 | jamko | lol |
| 13:45.10 | rocksfrow | [TK]D-Fender, lol... you answered my question by simply saying rpm |
| 13:45.17 | rocksfrow | i just wasn't sure if YUM could do it directly |
| 13:45.42 | jamko | make /uninstall |
| 13:45.49 | jamko | : ) |
| 13:45.52 | rocksfrow | im not installing from source bro |
| 13:45.56 | rocksfrow | that's not hellpful |
| 13:45.56 | [TK]D-Fender | rocksfrow: man yum <--- didn't look there either I bet |
| 13:46.04 | rocksfrow | [TK]D-Fender, i did so |
| 13:46.07 | rocksfrow | have you" |
| 13:46.08 | rocksfrow | lol |
| 13:46.13 | rocksfrow | you see a force tag? |
| 13:46.15 | rocksfrow | doubt it. |
| 13:46.16 | [TK]D-Fender | rocksfrow: If it ain't there... it doesn't exist |
| 13:46.25 | *** join/#asterisk _structz (~structz@gandalf.ai.unesp.br) |
| 13:46.25 | rocksfrow | ok |
| 13:47.10 | WIMPy | ruyo: Bristuff still exists? |
| 13:47.11 | sawgood | How about this situation (I hope I get it right) ... a manager tells his assistant (I'll be in a meeting for 30 minutes) catch all my calls (without the manager having to use DND or call forwarding) |
| 13:47.20 | *** part/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:47.22 | sawgood | This is more common then you might think |
| 13:47.24 | _structz | Hello All, I'm having a problem with WaitExten() When the context is executed from a sip number waitexten works properly, when is a incoming call from E1 the waitexten goes to timeout even if the numbers are typed |
| 13:47.50 | [TK]D-Fender | [09:47]<sawgood>How about this situation (I hope I get it right) ... a manager tells his assistant (I'll be in a meeting for 30 minutes) catch all my calls (without the manager having to use DND or call forwarding) <-- FreePBX owns your ass.... stop asking in here. |
| 13:47.53 | jamko | dtmf |
| 13:48.58 | sawgood | 100% correct (again) |
| 13:49.06 | sawgood | today might be a rough day for me |
| 13:49.12 | [TK]D-Fender | sawgood: And you're depicting it like the server and/or phones are supposed to be PSYCHIC. "Hi I don't want to take any action, but I expect magically automatic results". |
| 13:49.24 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 13:49.42 | _structz | jamko, dtmfmode is set to rfc2883 |
| 13:49.45 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 13:49.50 | jamko | sawgood: you need to have asterisk in a db, and write a gui for users to control the dialplan. Short of that there is nothing you can do, unless of course you push "DND".. |
| 13:50.37 | ruyo | WIMPy, I don't know.. |
| 13:50.39 | [TK]D-Fender | jamko: He's already running FreePBX. |
| 13:50.52 | ruyo | Seems it doesn't anymore. |
| 13:50.54 | [TK]D-Fender | JamPerhaps you've missed the big print... |
| 13:51.01 | jamko | yes perhaps. |
| 13:51.06 | rocksfrow | yay. |
| 13:51.16 | rocksfrow | i used yumdownloader to download the package file, then rpm -ivh --nodeps |
| 13:51.29 | rocksfrow | thanks, i guess smartass answers are better than no answers :) |
| 13:52.03 | *** join/#asterisk cherva (~cherva@93.152.158.160) |
| 13:52.09 | *** join/#asterisk LemensTS (~LemensTS@70.238.154.222) |
| 13:52.12 | jamko | structz: Do you have ,T,t at the end of the dial command? |
| 13:52.29 | [TK]D-Fender | rocksfrow: It helps when you read the little 1-lett instructions it HANDS you on demand... |
| 13:52.37 | LemensTS | Can anyone look up who owns a mobile phone number? Someone keeps calling me and annoying me... |
| 13:52.50 | [TK]D-Fender | [09:49]<_structz>jamko, dtmfmode is set to rfc2883 <--- I hope NOT |
| 13:52.53 | rocksfrow | [TK]D-Fender, yumdownloader was the trick... not rpm |
| 13:52.59 | _structz | jamko, There is no Dial command at all.... the call incomes to a certain number. and goes to WaitExten and even if the number dialed goes to timeout |
| 13:53.07 | rocksfrow | im very aware of rpm and it's flags, i should couldn't figure out how to get the damn package file from repo |
| 13:53.10 | [TK]D-Fender | rocksfrow: How you get the RPM isn't important... |
| 13:53.11 | WIMPy | Looks like Asterisk can't hadle a phone disapperaing during a call. Wonder if that also happens on PRI. |
| 13:53.19 | rocksfrow | [TK]D-Fender, i wasn't to get it using the repos |
| 13:53.21 | rocksfrow | wanted* |
| 13:53.24 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 13:53.27 | [TK]D-Fender | rocksfrow: and you can surf it with a browser just fine... |
| 13:53.27 | _structz | [TK]D-Fender, ?? |
| 13:54.00 | rocksfrow | [TK]D-Fender, do you know if these same GPL issues exists with 1.6? |
| 13:54.16 | [TK]D-Fender | rocksfrow: Which? |
| 13:54.19 | rocksfrow | i plan to setup a new box on 1.6 soon, and hoping these annoyances disappear |
| 13:54.38 | _structz | jamko, When the calls is from another sip number(on the same server) works properly, but when is a incoming call from E1 goes to Timeout |
| 13:54.48 | rocksfrow | ...the GPL issue I was just speaking of, with res_fax_digium and addons |
| 13:55.11 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:55.26 | *** part/#asterisk LemensTS (~LemensTS@70.238.154.222) |
| 13:55.31 | cherva | can someone explain to me why when I dial a trunk named "out_9" in the logs I see "SIP/foo......" when sip foo is another number, but the extensions specified for the DID of out_9 are ringing........... in short everything works but in the logs there is different "SIP/<trunk_name>" is shown... |
| 13:55.39 | [TK]D-Fender | rocksfrow: Ok, I only just glanced at that... Never touched before... |
| 13:55.43 | [TK]D-Fender | rocksfrow: and I don't do RPM |
| 13:55.53 | *** part/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 13:56.11 | _structz | anyone ? can help me out? |
| 13:56.13 | rocksfrow | [TK]D-Fender, ok |
| 13:56.49 | [TK]D-Fender | cherva: Where do you get the idea that "out_9" is a "trunk"? |
| 13:57.10 | jamko | sorry,,, I am looking at freepbx real quick.. Is there a gui that works well for stupid end-users? |
| 13:57.17 | [TK]D-Fender | cherva: and what you dial is a TECH/PEER/NUMBER. I'm sure the "peer" sure isn't "out_9" |
| 13:57.19 | cherva | [TK]D-Fender, my bad I just used that name for the example .......... |
| 13:57.46 | jamko | structz.. stand by |
| 13:58.05 | [TK]D-Fender | cherva: Maybe you should show something real for us to look at. |
| 13:58.42 | cherva | [TK]D-Fender, what do you need ? |
| 13:59.32 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 13:59.39 | [TK]D-Fender | cherva: For you to SHOW us the problem. |
| 14:03.08 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 14:03.12 | *** join/#asterisk deonv (~Adium@pixfirewall.itn.com.na) |
| 14:03.18 | wcselby | o/ |
| 14:04.00 | _structz | anyone?! |
| 14:04.04 | *** part/#asterisk deonv (~Adium@pixfirewall.itn.com.na) |
| 14:04.10 | _structz | wcselby, hi :D |
| 14:04.15 | [TK]D-Fender | _structz: You are showing us NOTHING |
| 14:04.43 | WIMPy | Interesting. I can only get the wcb4xxp module to work when loading (and unloading) hfcmulti first. |
| 14:05.08 | cherva | [TK]D-Fender, ok I have 2 trunks, named Viki and ZzZ, when I call Viki in the logs I get this http://pastebin.org/475547 .. why there is SIP/ZzZ not SIP/Viki there ? except this everything is ok the extensions specified for Trunk Viki are ringing when I call this trunk......... |
| 14:05.32 | _structz | [TK]D-Fender, I have a number(6298) on the incoming context when the call is placed from another sip number(on the same server, and different servers) works properly, but when is a incoming call from E1 goes to Timeout (sip numbers on a server E1 on another, comunication between=SIP) |
| 14:05.52 | [TK]D-Fender | cherva: because that is the peer it matched |
| 14:06.34 | [TK]D-Fender | cherva: Which is probably because you set "insecure=port,invite" for them and it picks the FIRST because it doesn't need to CHALLENGE them |
| 14:06.51 | ruyo | WIMPy, wcb4xxp is the new module to use BRI with DAHDI? |
| 14:07.05 | [TK]D-Fender | _structz: Don't give us a story, show us the FAILED CALL AND CONFIGS |
| 14:07.46 | cherva | [TK]D-Fender, is this a problem ? and what should be the insecure part to show me the right trunk names ? |
| 14:07.53 | WIMPy | ruyo: Yes. Well, not that new. |
| 14:08.21 | ruyo | WIMPy, yeah, as new as dahdi. How is it working out for you? |
| 14:08.48 | ruyo | Because mISDN is giving me an headache. |
| 14:09.02 | ruyo | I can't get regular PTP to function correctly. |
| 14:09.32 | WIMPy | ruyo: Well with a patch from issue 17694 you can get hfc4/8s cards to work in NT mode. |
| 14:10.13 | WIMPy | ruyo: Forget mISDN, at least the old one. It has been abandoned more than two years ago. |
| 14:10.23 | [TK]D-Fender | cherva: because it ISN'T checking it matches the FIRST peer that has the same IP <- |
| 14:10.27 | WIMPy | At the moment I'd still recomment mISDN2 with LCR. |
| 14:10.50 | ruyo | WIMPy, yeah, I'm trying to get that one up now. |
| 14:11.12 | ruyo | I beleive I need to compile a newer kernel though. |
| 14:11.25 | cherva | [TK]D-Fender, so there is no problem ? |
| 14:11.28 | ruyo | (Debian Lenny) |
| 14:11.30 | WIMPy | With dahdi I have two issues ATM: The first that I need to use mISDN befor dahdi will recognize the card and the other that it won;t end a call if the line goes down. |
| 14:11.47 | _structz | [TK]D-Fender, http://pastebin.com/3YgneKKS |
| 14:12.20 | WIMPy | ruyo: Use a current kernel and use a git version of LCR. |
| 14:12.24 | ruyo | WIMPy, maybe mISDN is blocking the resources, no? |
| 14:12.51 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:13.03 | WIMPy | ruyo: No it seems neccessary to put the card into a state for dahdi to make use of it. |
| 14:13.33 | anonymouz666 | JerJer: how was the T.38 1.6 test? |
| 14:14.08 | [TK]D-Fender | cherva: that is how it works. You want it to do something else, make sure they auth instead of "insecure". If you can't then change what else you log to separate them easier. Or update your CDR's after the fact. Or whatever |
| 14:14.09 | malcolmd | <JerJer> -- Channel 'SIP/FlowRoute-00000011' FAX session '6' is complete, result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 2, resolution: '204x196', transfer rate: '14400', remoteSID: '' |
| 14:14.09 | malcolmd | <JerJer> so far so good :) |
| 14:14.10 | malcolmd | <JerJer> added the redundancy and maxdatagram |
| 14:14.13 | malcolmd | anonymouz666: ^ |
| 14:14.39 | anonymouz666 | nice |
| 14:14.41 | ruyo | WIMPy, if you don't have mISDN installed, do you have the same problem? |
| 14:14.59 | malcolmd | had to edit t38pt_udptl in sip.conf for redundancy and maxdatagram options |
| 14:15.00 | cherva | [TK]D-Fender, thanks |
| 14:15.12 | anonymouz666 | was using the spandsp then |
| 14:15.15 | anonymouz666 | no FFA |
| 14:15.18 | WIMPy | ruyo: I _NEED_ misdn to be able to use dahdi. |
| 14:15.26 | malcolmd | you were? jerjer's test was using ffa |
| 14:15.35 | [TK]D-Fender | _structz: I only see a single exten they ACN dial in there (6298), Also as this is a SIP call I don't see what mode is being negotiated, nor what mode you set |
| 14:15.40 | anonymouz666 | ah |
| 14:15.45 | malcolmd | i don't know if he retested using spandsp later |
| 14:15.51 | tzafrir_laptop | WIMPy, huh? |
| 14:16.12 | ruyo | WIMPy, ah, thought dahdi could take care of everything |
| 14:16.36 | tzafrir_laptop | It does |
| 14:16.37 | [TK]D-Fender | CAN* |
| 14:16.40 | WIMPy | tzafrir_laptop: the wcb4xxp module won't recognize the card unless I modprobe hfcmulti;rmmod hfcmulti first. |
| 14:16.52 | WIMPy | ruyo: That's the idea. |
| 14:17.10 | tzafrir_laptop | WIMPy, that's a bug in the module. What card is it, exactly? |
| 14:17.23 | _structz | [TK]D-Fender, the person will dial some number that will match _6[4-6]XX its a little above that live.. i didn't putted on the pastebin |
| 14:17.31 | _structz | live=line* |
| 14:17.41 | tzafrir_laptop | WIMPy, and also: are you sure it's not simply an issue of hfcmulti getting there first? |
| 14:17.48 | tzafrir_laptop | (not blacklisted) |
| 14:17.54 | *** join/#asterisk Futnet_Jkenney (~jkenney@c-76-20-171-4.hsd1.mi.comcast.net) |
| 14:18.00 | Futnet_Jkenney | Good morning everyone |
| 14:18.04 | WIMPy | tzafrir_laptop: Not sure. Tehy are all sold as Junghanns, but I don't know what's original and what isn't. |
| 14:18.24 | [TK]D-Fender | _structz: So far I don't see anything USEFUL for this call. Nothing that proves DTMF modes, etc, and you showed me a crap looking dialplan sample. |
| 14:18.30 | WIMPy | tzafrir_laptop: No modules get auto-loaded. |
| 14:18.51 | Futnet_Jkenney | i have a problem i am attempting to figure out SLA, However i want the feature to be as a receptionist where you can see if the person at (ie extension 201) is on the phone by turning the line button red on the polycom side car |
| 14:18.52 | WIMPy | I just rebooted several times to confirm that behaviour. |
| 14:18.59 | Futnet_Jkenney | can someone point me to a good tutorial |
| 14:19.26 | [TK]D-Fender | Futnet_Jkenney: that isn't "SLA", that is "presence" |
| 14:19.40 | jamko | buddy watch |
| 14:19.46 | Futnet_Jkenney | Thanks TK |
| 14:19.52 | [TK]D-Fender | Futnet_Jkenney: just look up "polycom presence" on the WIKI and read up on "dialplan hints" while you're at it |
| 14:19.54 | ruyo | Do you have libpri, WIMPy? |
| 14:19.55 | [TK]D-Fender | ~wikis |
| 14:19.55 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:20.00 | _structz | [TK]D-Fender, what u mean? |
| 14:20.13 | Futnet_Jkenney | Yes but if i am running out of mysql can i still run hints? |
| 14:20.16 | WIMPy | ruyo: Sure. |
| 14:20.44 | [TK]D-Fender | [10:20]<_structz>[TK]D-Fender, what u mean? <- Ok, you don't seem to be getting it. HOW ME THE FUCKING SIP CONFIGS AND CALL WITH SIP DEBUG. |
| 14:20.58 | [TK]D-Fender | Futnet_Jkenney: I don't believe so. |
| 14:21.12 | ruyo | WIMPy, that was my last idea, sorry.. |
| 14:21.12 | Futnet_Jkenney | sigh |
| 14:21.15 | [TK]D-Fender | Futnet_Jkenney: Unless ther has been a very recent change this has not been possible to date |
| 14:21.23 | _structz | [TK]D-Fender, http://pastebin.com/2cXfgkiJ full context :D |
| 14:21.35 | WIMPy | tzafrir_laptop: It identifies as "Cologne Chip Designs GmbH Device b562" |
| 14:21.37 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:21.45 | Futnet_Jkenney | ok |
| 14:21.47 | Futnet_Jkenney | Thanks TK |
| 14:22.27 | tzafrir_laptop | WIMPy, have you tried Junghanns' latest drivers? http://junghanns.net/downloads/jnet-dahdi-drivers-1.0.2.tar.gz |
| 14:25.28 | anonymouz666 | anyone in here already access a serial modem (v92) through E1/T1 using DAHDI? It seems to connect but doesn't work any changes. |
| 14:25.49 | *** join/#asterisk mifadir (~ifadir@dynamic.casap1-190-30-137-41.wanamaroc.com) |
| 14:25.57 | mifadir | Hi everybody |
| 14:25.57 | anonymouz666 | PSTN -> E1/T1 -> Asterisk -> E1/T1 -> Siemens -> Modem |
| 14:25.59 | WIMPy | tzafrir_laptop: Nope. I just tried latest dahdi with the te_ne_override patch. Will take a look at that version. |
| 14:26.05 | mifadir | any one try asterisk with ZRTP |
| 14:26.15 | mifadir | ? |
| 14:28.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:28.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:28.38 | tzafrir_laptop | WIMPy, does latest dahdi get rid of the need for mISDN? |
| 14:29.50 | WIMPy | tzafrir_laptop: That was on 2.3.0.1. |
| 14:31.21 | *** join/#asterisk b14ck (~b14ck@cpe-static-irontonconf-rtr-f0.iro.ptd.net) |
| 14:32.08 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 14:32.24 | Kobaz | hmm |
| 14:34.27 | Kobaz | it looks like the behavior of chanisavail changed between 1.6.0 and 1.6.2... it seems to always return an available channel when checking local channels (Even if the exten doesn't exist) |
| 14:35.27 | *** join/#asterisk moy (~moy@bas1-toronto47-1177731847.dsl.bell.ca) |
| 14:35.57 | _structz | [TK]D-Fender, http://pastebin.com/4fC0CqcR < sip debug on the sip numbers server. http://pastebin.com/mV4SkNAA sip debug on E1 server |
| 14:36.07 | WIMPy | qozap doesn't want to build. |
| 14:37.58 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:39.29 | *** join/#asterisk simond (~simon@syria.uc.org) |
| 14:39.55 | simond | Is there some way to adjust voicemail volume when using IMAP as the store? |
| 14:39.57 | hrhrhr | what has happened to 'sip show peers' |
| 14:40.03 | hrhrhr | what's the new cmd fs |
| 14:40.17 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:40.35 | WIMPy | hrhrhr: Better ask why you don't have it. |
| 14:41.03 | hrhrhr | why dont i have it :( |
| 14:42.42 | henk | when i have an entry in sip.conf with 'context=callman' shouldn't every call passed to asterisk from that peer/friend look for extensions in that context? |
| 14:43.35 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:43.42 | WIMPy | tzafrir_laptop: qozap misses function set_current_state and TASK_UNINTERRUPTIBLE. I guess it needs an older version of something (as usual). |
| 14:44.42 | [TK]D-Fender | _structz: From 1 * to another? |
| 14:44.57 | [TK]D-Fender | _structz: Where do I see the the FAR end GOT any DTMF properly? |
| 14:45.00 | henk | the calls just land in default context and can't find the extension :-/ |
| 14:45.26 | [TK]D-Fender | henk: Who said it MATCHED that peer on incoming? Perhaps you should loko at the CALL |
| 14:45.35 | WIMPy | hrhrhr: module show like sip |
| 14:46.45 | henk | [TK]D-Fender: i'm not sure if i can follow you, but i have only 2 friends configured and none of them have 'context=default'. how do i look at a call? |
| 14:47.34 | hrhrhr | why does everyone feel compelled to answer in riddles |
| 14:47.54 | hrhrhr | sip show peers = 1.6 command has changed to...? :) |
| 14:48.16 | _structz | [TK]D-Fender, forget it! |
| 14:48.27 | _structz | I'll figure it out here.. tkz anyway |
| 14:48.42 | WIMPy | hrhrhr: It's still the same. You seem to lack sip support. |
| 14:49.10 | hrhrhr | how can this be possible with an out of the box install of *now |
| 14:49.46 | hrhrhr | i realise this may not be a question for this channel tho, so thanks for your reply |
| 14:49.59 | WIMPy | hrhrhr: Possibly because it's not configured? I have no idea, what *nows defaults are. |
| 14:50.13 | hrhrhr | i could be going out on a limb here |
| 14:50.18 | hrhrhr | but i kinda assumed sip would be default |
| 14:50.30 | Qwell | click the big orange button and restart asterisk |
| 14:50.31 | hrhrhr | that wildly exotic protocol... sip |
| 14:50.33 | henk | how do i place calls coming from a certain peer/friend in a certain context? |
| 14:51.11 | hrhrhr | there are no changes pending Qwell |
| 14:51.20 | Qwell | click it anyways |
| 14:51.30 | hrhrhr | there's no orange button tho |
| 14:51.37 | Qwell | then restart asterisk |
| 14:52.02 | hrhrhr | best way to do so from *now? |
| 14:52.13 | henk | 'sigh' i'll just use [default] then it seems... |
| 14:52.37 | *** join/#asterisk adyn (~adyn@unaffiliated/adyn) |
| 14:54.29 | jamko | In case anyone is banging their head against the wall with a problem related to a T.38 reinvite from sip cluster, Digium just acknowledged by bug report: https://issues.asterisk.org/view.php?id=17842 |
| 14:54.53 | *** part/#asterisk simond (~simon@syria.uc.org) |
| 14:56.05 | [TK]D-Fender | _stuYou'll probably get ti once you actually start looking at the big picture |
| 14:56.20 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:56.25 | [TK]D-Fender | henk: * CLI + SIP DEBUG |
| 14:56.39 | henk | [TK]D-Fender: and look out for what? |
| 14:56.46 | henk | i was that far already... |
| 14:56.51 | [TK]D-Fender | henk: just LOOK. Use your eyes. |
| 14:57.11 | [TK]D-Fender | henk: and if you don't see anything that is tipping you off then show US |
| 14:58.40 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 14:59.04 | henk | yeah, tomorrow, thanks for the 'help' though... |
| 15:00.25 | hrhrhr | Qwell: that sorted it, n1 |
| 15:01.17 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 15:01.56 | [TK]D-Fender | Nobody looks in here anymore. You'd all be selected as jurors and the accused murderer would get off because you collectively refused to actually look at the videotape that clearly shows him doing it. |
| 15:02.07 | [TK]D-Fender | I wonder why some people even bother. |
| 15:13.31 | Naikrovek | [TK]D-Fender: because to some people, computers are magic, and we're just supposed to know. we're expected to have seen the problem before and to remember what the solution was. |
| 15:13.46 | Naikrovek | and to give up the solution freely |
| 15:14.08 | [TK]D-Fender | Naikrovek: "remember" the problem? We never get to SEE it. |
| 15:14.19 | Naikrovek | part of the magic thing |
| 15:14.23 | [TK]D-Fender | Naikrovek: "Hi my car doesn't work. WHY!?!?!?!?" |
| 15:14.27 | Naikrovek | there's no logic in magic |
| 15:14.39 | Naikrovek | to some people, cars, computers, televisions, are all magic |
| 15:14.39 | anonymouz666 | [TK]D-Fender: try to put gasoline |
| 15:17.40 | Futnet_Jkenney | cars aren't magic? |
| 15:17.47 | [TK]D-Fender | Naikrovek: Of of my most hated words : programagically. |
| 15:17.48 | Naikrovek | definitely not |
| 15:17.55 | Naikrovek | yeah |
| 15:18.08 | Naikrovek | automagically programagically, high on my hate list as well |
| 15:18.32 | [TK]D-Fender | Naikrovek: Used by some dumb-fucks at my head office who wish to engender the thought that actual programming even for minor shit is VOODOO. |
| 15:18.55 | [TK]D-Fender | And this is NOT by anyone in IT. |
| 15:18.56 | Naikrovek | or that it's easy to fix any problem with magic |
| 15:19.52 | [TK]D-Fender | Naikrovek: It is... because if it were difficult... then it wouldn't be magic, and it'd look a lot more like "Real Work" (tm) |
| 15:19.56 | Futnet_Jkenney | I believe i have figured out my presence issue. Thanks to you guys and the folks over at Digium |
| 15:20.37 | Futnet_Jkenney | I big thanks to TK in pointing me in the correct direction |
| 15:20.54 | Naikrovek | [TK]D-Fender: yes, and their request to add simple functionality to existing apps & services requires real money and time to accomplish. hogwash! it sounds simple, so it is simple! |
| 15:21.35 | Naikrovek | there needs to be some public shaming for people who make decisions but don't have to deal with the consequences themselves, in the business world |
| 15:21.36 | Futnet_Jkenney | Some people live in a harry potter world when it comes to IT and development |
| 15:21.43 | Naikrovek | Futnet_Jkenney: some? most. |
| 15:22.02 | Futnet_Jkenney | I was trying to be kind |
| 15:22.05 | Naikrovek | heh |
| 15:22.07 | Naikrovek | i know |
| 15:22.10 | Naikrovek | just annoying |
| 15:22.11 | Naikrovek | not you |
| 15:22.24 | Naikrovek | the people who assume i can fix things in a few moments |
| 15:22.27 | Futnet_Jkenney | That is why my support extension plays the harry potter theme as the ringing |
| 15:22.29 | Futnet_Jkenney | :) |
| 15:22.40 | Naikrovek | that's a damned good idea right there |
| 15:22.41 | Futnet_Jkenney | or sometimes the old skewl bat man theme |
| 15:22.55 | Naikrovek | mine plays zelda stuff but it's getting old |
| 15:23.16 | Futnet_Jkenney | I have Super Mario playing when you call my house |
| 15:23.58 | Futnet_Jkenney | as the ring back |
| 15:24.08 | Naikrovek | that's cool |
| 15:24.17 | Futnet_Jkenney | I found a site that has all the audio from all the games |
| 15:24.20 | Futnet_Jkenney | for download |
| 15:26.21 | Naikrovek | ooh i need to set up some overwatch (from half-life 2) sounds on my phone |
| 15:26.24 | WIMPy | Wow. The ISDN part seems to be seriousely b0rked :-( |
| 15:26.28 | Naikrovek | for when certain people call |
| 15:27.07 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.66) |
| 15:27.28 | *** join/#asterisk flambo (~luke@mobileinternet2.o2.ie) |
| 15:28.49 | flambo | hey there folks.. i've been asked to look at a project for dial-in data connections, an order system receives dial-up connections from shops that order stock, it's to leverage modem based order system that's already around the country here, so we're looking to setup an FRI ISDN connection into a datacenter,, we have quotes from some folks, but what kind of ISDN card do i need for the server? |
| 15:29.12 | flambo | one person suggested a BRI card from Digium, which sounds sensible, but at $550 it seems very high end, |
| 15:29.33 | flambo | if i'm supporting a multi channel FRI ISDN line, is something that expensive required? |
| 15:29.45 | flambo | the card i was pointed to was the Digium te122 |
| 15:29.56 | Qwell | what is FRI? |
| 15:30.50 | flambo | maybe it's a UK/ european standard,,, PRI or a FRI ISDN line. is 32 and 16 channel ISDN |
| 15:30.53 | flambo | repectively. |
| 15:31.33 | flambo | so we're just going with the 16 channel line, as we don't imagine receiving more than 8 calls at once (i'm hedging a little in assuming that 1 call requires 2 channels, just in case; ) |
| 15:31.44 | WIMPy | flambo: For data connections I'd look for an old NAS. |
| 15:31.56 | flambo | an old NAS? |
| 15:32.05 | Qwell | If you're doing data stuff, Asterisk is really the wrong thing to use.. |
| 15:32.39 | WIMPy | flambo: Network Access Server |
| 15:32.45 | doolittlework | hi there is there a way to assign a unique identifier to a recording filename |
| 15:32.56 | doolittlework | never mind dum question |
| 15:33.06 | [TK]D-Fender | doolittlework: You pick the name. Increment a number or something |
| 15:33.11 | flambo | the situation is that shops around the contry use a stock dispensary computer installed at each location that dials up with a modem to wholesalers, in this case my client, we just have to plug in a server, how does a NAS help here? |
| 15:33.39 | flambo | WIMPy: oh, not network attached storage? |
| 15:34.18 | WIMPy | flambo: Like Cisco AS or Lucent Max |
| 15:34.31 | flambo | we're just going to put some python code on the back of the dial-up data connections that receives the order transmissions and stuffs the decoded orders into mysql |
| 15:34.36 | doolittlework | [TK]D-Fender: was hoping u show up, do yo know of a way where i can send text to the screen of eyebeam whiles on a call or just before a call starts? |
| 15:35.02 | WIMPy | flambo: Pre IP? |
| 15:35.15 | [TK]D-Fender | doolittlework: Maybe you should read their admin guide or something <- |
| 15:35.16 | flambo | Pre IP? |
| 15:35.33 | doolittlework | ta thanx that a start |
| 15:35.47 | WIMPy | flambo: From times before the invention of IP? |
| 15:36.09 | flambo | WIMPy: you mean using some kind of serial protocol or something, i honestly don't know. |
| 15:36.48 | flambo | WIMPy: in this case we're just leasing a server from a hsoting company and sticking in the ISDN PCI card,, UPC are connecting the ISDN line into the Data Centre, |
| 15:37.04 | WIMPy | flambo: That could make a big difference. Other question is if they use modem or ISDN? |
| 15:37.25 | flambo | it's all been talked through, just wondering if that card is a good choice,, and what kind of system we can use to connect of the back such as asterisk? or should i look into some kind of modem library? |
| 15:37.47 | flambo | something that provides an interface to python for the ip stack on the back of the modem or something. |
| 15:37.52 | WIMPy | Depends on your needs. |
| 15:38.04 | WIMPy | So is it IP? |
| 15:38.18 | flambo | ok, so you're saying i need to know more about the type of dial-up connection,, what are the possabilities? i think it is IP |
| 15:38.57 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 15:39.21 | WIMPy | IP or not and what kind of connection, i.e. modem or ISDN and in that case what protocoll. |
| 15:41.07 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:50.09 | golikwid|mac | anyone know where i can sell a used fonality pbx |
| 15:50.17 | Qwell | good luck with that. |
| 15:50.24 | malcolmd | heh |
| 15:50.25 | [TK]D-Fender | golikwid|mac: Craigslist |
| 15:51.32 | *** join/#asterisk flambo (~luke@mobileinternet2.o2.ie) |
| 15:57.29 | *** join/#asterisk Tim_Toady (~moi@188.4.51.212.dsl.dyn.forthnet.gr) |
| 15:57.56 | telnettech | golikwid|mac donate it to the local tech school so that they can reuse the hardware for learning purposes....Haha!! |
| 15:59.34 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 16:01.13 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 16:03.17 | flambo | WIMPy: sorry was on phone, (and i'm teathered so it cuts my connection while i'm talking) |
| 16:03.43 | flambo | WIMPy: you were saying regardless of IP or not, i need to know what kind of connection, modem or ISDN and in that case what protocoll? |
| 16:04.08 | drmessano | A used Fonality PBX? |
| 16:04.25 | drmessano | Didn't we have the conversation about *dead hookers* last night? |
| 16:04.28 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 16:04.31 | flambo | well, the client locations are using modems to dial up over regular POTS lines. |
| 16:04.42 | *** join/#asterisk BANSAL (~bansal@117.199.116.192) |
| 16:04.45 | *** part/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 16:04.58 | flambo | kind of like old dial-in banking telephone connections were done, i just don't know what to look for in terms of determining the call connection types. |
| 16:05.05 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 16:05.24 | flambo | and haing a multi channel ISDN line on our server, will that work? |
| 16:06.04 | flambo | If we wanted to accept calls from this software and capture the connection types, if we just hooked up a modem to a regular telephone line and tried to figure out the protocol that way, what would i need? |
| 16:06.18 | flambo | I think at this point, i'm confused as to what hardware to commit to. |
| 16:09.45 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 16:14.20 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:14.50 | Qwell | flambo: Why would you use a PBX for this? |
| 16:15.40 | WIMPy | flambo: Ok, if you need modem connectivity, a ISDN card won't get you very far. |
| 16:16.38 | WIMPy | So a NAS migt be a better idea, but they are best at IP. For anything else you'd need something rather special. |
| 16:18.03 | *** join/#asterisk BANSAL (~bansal@117.199.119.245) |
| 16:20.23 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:20.26 | drmessano | Why wouldn't an ISDN card work? |
| 16:21.24 | drmessano | Sounds like he needs a simple PPP connection from the remote clients, which would work fine over a PRI |
| 16:22.18 | *** join/#asterisk grolloj (~chatzilla@h-68-166-73-162.nycmny83.static.covad.net) |
| 16:24.09 | doolittlework | can one write date to mysql using the application mYSQL? |
| 16:24.32 | Qwell | doolittlework: That is the idea. |
| 16:26.45 | doolittlework | how would one do that @Qwell exten => _X.,1,MYSQL(Connect test localhost testdb 123456 refgen) this connects so can one just add this exten => _X.,n,MYSQL(INSERT INTO table 1 ??????? |
| 16:26.53 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 16:28.31 | kalimc | I have a basic setup for incoming calls, when I dial my DID (444-555-6666) I see on my asterisk that the EXTEN value is 3701, I have nothing like that setup, however my actual number is XXX-XXX-3701. Where should I start looking for why the exten is only a 4 digit value? |
| 16:28.53 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:29.26 | *** join/#asterisk flambo (~luke@mobileinternet2.o2.ie) |
| 16:30.48 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:34.17 | telnettech | kalimc.....sound like that is the DNIS that telco is providing to you when your DID number is called |
| 16:35.13 | kalimc | I see, I will check that. Thanks |
| 16:35.38 | telnettech | kalimc......you can either setup in your dialplan to dial your internal extension when the DID is called or call your Telco and tell them what you want to come in as DNIS |
| 16:36.21 | kalimc | Ok, this is all coming back to me, I will call the telco, and get this taken care of. |
| 16:37.09 | JerJer | doolittlework: if you expect something like that to scale, I would use FastAGI |
| 16:37.11 | JerJer | but that's me |
| 16:40.53 | WIMPy | flambo: Ok, if you need modem connectivity, a ISDN card won't get you very far. |
| 16:40.58 | WIMPy | So a NAS migt be a better idea, but they are best at IP. For anything else you'd need something rather special. |
| 16:41.18 | anonymouz666 | WIMPy: I need it and I am using an ISDN card |
| 16:41.24 | anonymouz666 | and I am also in trouble |
| 16:42.12 | WIMPy | anonymouz666: What do you need? |
| 16:42.34 | anonymouz666 | there are basic 2 cases. |
| 16:42.42 | anonymouz666 | make the ContactID protocol to work |
| 16:43.06 | anonymouz666 | and then make a modem v92 to work passing-through an Asterisk box integrated with Siemens |
| 16:43.53 | WIMPy | Passthru is something entirely different. |
| 16:44.30 | anonymouz666 | is it? I don't understand modems |
| 16:44.36 | anonymouz666 | but still doesn't work |
| 16:45.10 | WIMPy | You can't get the modem to connect via Asterisk? |
| 16:45.21 | anonymouz666 | it connects and it stops. |
| 16:45.43 | WIMPy | What's your setup like? |
| 16:45.45 | anonymouz666 | you can't issue any settings |
| 16:46.14 | anonymouz666 | PSTN -> E1 -> Asterisk -> E1 -> Siemens PBX -> modem |
| 16:46.15 | WIMPy | ? |
| 16:47.10 | WIMPy | Hmm. I can't see why Asterisk should do anything bad in there. What happens exactely? |
| 16:48.10 | anonymouz666 | I think the problem is due Echo Cancellation in DAHDI. |
| 16:48.48 | flambo | WIMPy: ok so i'm confused, we will have an ISDN line going to our server in the data Centre, the digium te122 card is in the server and can receive incoming calls from modems in other locations? is this not the case? |
| 16:49.17 | anonymouz666 | right now, I am looking for a way to port the DAHDI to disable the echocan from DIALPLAN |
| 16:49.21 | anonymouz666 | and test the whole thing again |
| 16:49.34 | WIMPy | anonymouz666: That's quite possible. Have you tried to disable it? |
| 16:50.05 | anonymouz666 | WIMPy: If I disable what happens to the calls? :) |
| 16:50.09 | flambo | WIMPy: I also realise this is an asterisk chat room, so if you don't know or are feeling this is too off topic i also understand :) |
| 16:50.10 | WIMPy | flambo: It can receive calls, yes. But it cannot terminate them. It's not a modem. |
| 16:50.58 | flambo | WIMPy: terminate you mean be the end point of a telephone call? |
| 16:51.49 | flambo | WIMPy: i think this is where my understanding lacks,, i'm just a part time code monkey and project manager. |
| 16:51.53 | WIMPy | flambo: You need a modem on your end. Or some software modem, but I don't know what the capabilities of iaxmodem are. Mind you fax is a lot easier than normal modem. |
| 16:52.02 | WIMPy | flambo: Yes. |
| 16:55.17 | flambo | WIMPy: well, the te122 card is PCI card in the server, and then we would use a software library that implements the modem stack or to control the calls by code ourselves? does this cover what you're meaning here? |
| 16:56.15 | flambo | WIMPy: you're making me concerned now that our idea of getting an ISDN Line is pointless? |
| 16:57.05 | Kobaz | isdn = i still don't need |
| 16:58.00 | WIMPy | That software is some heavy dsp stuff! |
| 16:58.19 | WIMPy | An ISDN line is always good, but you need the right equipment. |
| 16:58.23 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 16:58.39 | WIMPy | That's why I suggested to look out for an old NAS. |
| 16:59.01 | WIMPy | That will have the neccessary modems in hardware. |
| 16:59.11 | [TK]D-Fender | ZapRAS <------------- |
| 16:59.43 | flambo | WIMPy: ok, so what kind of servers, Cisco AS and lucent Max you mentioned. |
| 17:00.14 | WIMPy | [TK]D-Fender: He needs modem |
| 17:00.19 | flambo | any model names i can watch out for? |
| 17:00.31 | [TK]D-Fender | WIMPy: Look closer |
| 17:00.58 | WIMPy | [TK]D-Fender: Is there modem emulation in zapras now? |
| 17:02.02 | WIMPy | flambo: Another thing: Do have any idea on how to communicate with the callers at all? |
| 17:02.27 | flambo | WIMPy: not sure what you mean? |
| 17:03.12 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.222.111) |
| 17:03.12 | WIMPy | flambo: I guess you want more than just a connect, i.e. transfer some sort of information. |
| 17:03.58 | WIMPy | Info on dahdiras still says "no modem emulation included". |
| 17:04.12 | flambo | WIMPy: ahh, i havej a file specification,, i'll be a bit transparent here,, it's for pharasuiticle orders. |
| 17:04.55 | WIMPy | That's a much later stage. How will that file be transferred? |
| 17:04.59 | flambo | WIMPy: there's an order message definition already in place, that's why these dial-up systems are in place, it's a system that's in operation for about 15 years, the client wants to take advantage of the fact that pharmacists in our country are already used to the dispensary system |
| 17:05.32 | flambo | so they have a definition document covering the messages formats. |
| 17:06.29 | WIMPy | Ok, so if you have that, you should know whether it's running on IP. |
| 17:07.00 | flambo | WIMPy: well no, they dont' discuss the connecitno medium in the documentation. |
| 17:07.08 | flambo | just the message structure. |
| 17:07.46 | WIMPy | Well, you need to find out, how to get these messages. |
| 17:10.50 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:10.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:13.09 | flambo | WIMPy: ok, thank you for all your feedback.. i'll do some googleing and reading and see where that gets me,, i'll be able to ask more informed questions next time :) |
| 17:13.41 | flambo | right, gotta disconnect,, my phone is teathered, my one internet connection right now. |
| 17:13.41 | flambo | :) |
| 17:13.44 | flambo | thanks for everything : ) |
| 17:17.14 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 17:22.11 | *** join/#asterisk bkruse (~bkruse@76.73.216.184) |
| 17:22.11 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 17:33.52 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 17:35.32 | doolittlework | hi there i am no mysql expert, but i can not figure out how this works, or why this ---> system(mysql -u refgen -h 127.0.0.1 -e "INSERT INTO callinfo VALUES ('', '${CALLTIME}', '${NUMBERDIALED}','${REFERENCE')" --password=123456)<-------does not work |
| 17:36.04 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 17:36.16 | leifmadsen | doolittlework: first thing is you'r emissing a } on ${REFERENCE |
| 17:36.44 | leifmadsen | does it work from the console? |
| 17:38.12 | [TK]D-Fender | doolittlework: And you aren't referencing the DATABASE to use either |
| 17:38.27 | [TK]D-Fender | doolittlework: And the reason you aren't doing this via MySQL() is ... ? |
| 17:41.31 | b14ck | doolittlework, there's a MYSQL application you can use in your dial plan (it can be installed via asterisk-addons). It'll simplify what you're doing, and give you verbose error messages if your credentials are wrong and stuff like that. That approach would be much better. |
| 17:43.49 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 17:44.00 | tzanger | hello everyone |
| 17:44.08 | tzanger | been a long time |
| 17:44.34 | tzanger | anyone here (besides me) who has some lower-level POTS experience? |
| 17:45.03 | tzanger | working on a new FXO interface, and the people who designed it are getting all kinds of echo even with MG2 |
| 17:45.10 | doolittlework | ive tries MYSQL() but they are rattling my brain with the \ for spaces ,,,,,, and '''''' |
| 17:45.19 | WIMPy | POTS as in the historic stuff? ;-) |
| 17:45.37 | tzanger | I'm convinced that they've got their DAA configuration or part-68 set up incorrectly since dialing the local Milliwatt number gets them a -12dBm tone |
| 17:45.41 | tzanger | WIMPy: exactly :-) |
| 17:46.00 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 17:46.13 | tzanger | It's been a while since I've had to play with FXO, but I am pretty sure that the acceptable path loss is about 3-4dB, is it not? |
| 17:47.05 | Naikrovek | uh oh. noticed i've been chewing on a bit of Cat5e cable sheath for a while. PVC? Chlorine bad... |
| 17:47.36 | tzanger | Naikrovek: han a little lead steadies the nerves |
| 17:48.01 | tzanger | I used to hold the solder in my mouth when working on circuits... quick and dirty "third hand" |
| 17:48.21 | Naikrovek | ew |
| 17:49.48 | Naikrovek | are there any games that are better than Half-Life 2? Gebus this game is wonderful |
| 17:52.10 | doolittlework | Other characters that need to be escaped are quotes (\' and \"), commas (\,), backtick (\`), and backslash (\\). so this is so confusing First i do the connect hing using MYSQL() then the following--> INSERT/ INTO/ callinfo/ VALUES/ (/'/'/,/ /'${WHATEVER}/'/,/ /'${WHENEVER}/') ?????????????????????????????????????????????????? |
| 17:52.22 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 17:52.36 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
| 17:52.54 | Qwell | what? |
| 17:53.02 | doolittlework | lol |
| 17:53.11 | doolittlework | ok i lost it |
| 17:53.20 | Naikrovek | your example shows / but you need to \ |
| 17:53.36 | Naikrovek | my stupid technical mind noticed that first |
| 17:53.45 | Qwell | and spaces aren't in that list... |
| 17:53.56 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 17:53.57 | doolittlework | yes i know i saw i reversed it but stil no entry in database |
| 17:54.03 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 17:54.26 | doolittlework | spaces are on the web page forgot to include |
| 17:55.02 | Qwell | what web page? |
| 17:55.16 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 17:56.14 | doolittlework | http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL |
| 17:57.01 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 17:57.48 | *** join/#asterisk jhirley (~jhirley@c-75-74-13-194.hsd1.fl.comcast.net) |
| 17:58.45 | tzanger | anyone have the "rule of thumb" for acceptable path loss on POTS lines? is it 3-4dB? I'm seeing some things online that are giving 5.5dB as acceptable but htat seems awfully high |
| 18:00.42 | tzanger | also seeing "line" specs at -8.5dB(!) and "trunk" specs at -5.5dB, but this seems to be an article from the 80s |
| 18:02.49 | Naikrovek | 3.5db is 50% loss I believe |
| 18:03.03 | Naikrovek | 5 seems high if that's correct |
| 18:05.19 | WIMPy | Unfortunaletly there are different types of db, but either 6db or 12db makes a factor of two. |
| 18:05.33 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 18:05.45 | Naikrovek | i may be thinking of wireless db or cable or something |
| 18:05.51 | Naikrovek | but i remember reading it somewhere... |
| 18:06.37 | [TK]D-Fender | 10db = double |
| 18:06.47 | Naikrovek | or half |
| 18:06.59 | Naikrovek | in the case of -10db, yes? |
| 18:07.18 | [TK]D-Fender | is only learning while shopping for amplifier cabinets... |
| 18:07.47 | Naikrovek | see sound is different yes |
| 18:07.48 | Naikrovek | yet |
| 18:08.12 | Naikrovek | sound, radio, and waveguide are all different i think |
| 18:09.26 | WIMPy | Err, right it is 3db or 6db for a factor of 2. |
| 18:13.38 | *** join/#asterisk rezzen (~rezzen@nat/transgaming/x-kezrdrmxpnozpqdv) |
| 18:16.27 | jdoe | looking at http://www.voip-info.org/wiki/view/Asterisk+non-root -- why does /usr/lib/asterisk need to be writable? Is there anything there other than modules? |
| 18:18.06 | Naikrovek | if only there were an online ... encyclopedia (i guess) with some ability to search... maybe in the future this will happen and i'll be able to learn about decibels. in the meantime I'll eat my yogurt and wish for a better day. |
| 18:18.49 | Naikrovek | http://www.savagechickens.com/images/chickenhallucination.jpg |
| 18:20.16 | *** join/#asterisk deonv (~adium@pppoe-whk-127.cust.na.afrisp.net) |
| 18:24.29 | *** join/#asterisk Micc__ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 18:29.00 | Qwell | jdoe: Have you tried doing `ls /usr/lib/asterisk/` ? |
| 18:29.29 | Qwell | Naikrovek: don't be silly. go to a library like the rest of us. |
| 18:32.02 | Slugs_ | i got a ticket that somebody has hijacked the pbx and is making outbound calls from a particular #, if i know the number can I stop this? |
| 18:32.31 | Qwell | Slugs_: or you could change your passwords and fix your dialplan |
| 18:33.09 | bougyman | Slugs_: do you have a B2BUA extension available public or a DIDS extension? |
| 18:33.16 | jdoe | Qwell: I don't have a stock install so I'm not sure offhand what that dir in a distro package is in an install from source. |
| 18:33.18 | bougyman | anything that would let them relay? |
| 18:33.24 | bougyman | if not, you've got a bad dialplan, yeah. |
| 18:33.47 | Slugs_ | so the only way to make unauthorizesd outbound calls is if somebody had our passwords? |
| 18:34.14 | bougyman | ERROR[1721]: chan_sip.c:15385 sipsock_read: We could NOT get the channel lock for SIP/1002-085cd050! |
| 18:34.16 | Qwell | Slugs_: or if your dialplan went against what the README-SERIOUSLY.txt file says. |
| 18:34.19 | bougyman | is that always a deadlock? |
| 18:34.27 | ChannelZ | Your dialplan could be sufficiently bad as to allow someone to call IN to you and then dial back out |
| 18:34.28 | tzanger | Naikrovek: you're talking about power levels vs voltage levels I think |
| 18:34.41 | tzanger | [TK]D-Fender: will know |
| 18:34.43 | tzanger | he knows everything |
| 18:34.46 | *** join/#asterisk RoyK (~roy@cFDB1BF51.dhcp.bluecom.no) |
| 18:34.50 | Naikrovek | according to wikipedia, i'm talking about power vs field |
| 18:35.05 | Naikrovek | but they're both logarithmic |
| 18:35.08 | tzanger | [TK]D-Fender: what's the rule of thumb for acceptable path loss on a typical POTS line? is it as high as 5.5dB? |
| 18:35.08 | Slugs_ | ok one sec. |
| 18:35.13 | tzanger | well decibels are logarithmic |
| 18:35.47 | Naikrovek | a) power or intensity, b) amplitude |
| 18:35.49 | Qwell | tzanger: I'd bet tzafrir_laptop knows |
| 18:35.51 | Naikrovek | different measurements |
| 18:37.10 | Slugs_ | ~pb |
| 18:37.11 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:37.36 | bougyman | hrm... no one has seen "ERROR[1721]: chan_sip.c:15385 sipsock_read: We could NOT get the channel lock for SIP/1002-085cd050!" behavior before? |
| 18:37.43 | bougyman | i have this on both of the * boxes I have left. |
| 18:38.13 | tzafrir_laptop | Qwell, I'm not sure what the question is, but dB is basically log of power |
| 18:38.15 | tzanger | tzafrir_laptop: ping? Do you know what the rule of thumb for acceptable path loss on a typical POTS line is? |
| 18:38.35 | Naikrovek | tzafrir_laptop: OR ampliyude |
| 18:38.54 | Naikrovek | amplitude |
| 18:39.15 | tzanger | got a friend with crazy-ass echo using MG2, and he's blaming MG2 for his woes, but I always found MG2 to be pretty good, at least of FXS. calling the local milliwatt number says -12dB and that seems awfully damn low to me |
| 18:39.37 | anonymouz666 | MG2 is good, OSLEC is better |
| 18:39.38 | tzanger | I'd be happy with -3 to -4dB but -12? Stuff I'm seeing on the intertubes is saying -5.5 and even -8.5dB as what the telco will call acceptable |
| 18:39.50 | tzanger | anonymouz666: yeah OSLEC's next |
| 18:39.52 | tzafrir_laptop | tzanger, is that really FXS there? |
| 18:39.53 | Qwell | tzanger: try without MG2? :p |
| 18:40.01 | tzanger | tzafrir_laptop: no, FXO in this case |
| 18:40.07 | tzanger | silabs 3018+3050 DAA |
| 18:40.26 | Slugs_ | here is the pb of my dialplan |
| 18:40.29 | Slugs_ | http://pastebin.com/1A935rpM |
| 18:40.31 | tzanger | well if the rx audio really is -12dB he's got a problem with his part 68 |
| 18:40.50 | Slugs_ | any obviously issues allowing this 'hijacking' |
| 18:42.22 | Qwell | Slugs_: what about the #include'd files? |
| 18:42.37 | Katty | stretches |
| 18:43.08 | Slugs_ | Qwell: ill paste them as well |
| 18:43.10 | Qwell | Slugs_: and how are they getting in to the PBX? |
| 18:43.44 | tzanger | Katty: it's a little late to be getting out of bed, dear. |
| 18:44.09 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-137.dhcp.embarqhsd.net) |
| 18:45.42 | Naikrovek | kats stretch all day long |
| 18:45.50 | Qwell | they also sleep all day long |
| 18:46.05 | Naikrovek | yes |
| 18:46.08 | Naikrovek | a Kat can do both |
| 18:46.58 | tzanger | tzafrir_laptop: any ideas? |
| 18:47.53 | tzafrir_laptop | tzanger, for starters, is it indeed mg2's fault? If you disable it, it's gone? |
| 18:49.51 | [TK]D-Fender | [14:35]<tzanger>[TK]D-Fender: what's the rule of thumb for acceptable path loss on a typical POTS line? is it as high as 5.5dB? <- not a clue :) |
| 18:51.16 | Naikrovek | doesn't someone, ANYONE, have an uncle or something that works at a phone company that can answer this |
| 18:51.24 | TheDavidFactor | is there a way to know in the dialplan what local IP the call came in on? I thought some the SIP* functions might have it, but didn't find anything. Using asterisk 1.6.2.x |
| 18:51.56 | [TK]D-Fender | TheDavidFactor: They do. |
| 18:52.29 | tzanger | tzafrir_laptop: no, it's not that it's MG2's fault, it's that MG2 isn't handling it |
| 18:52.49 | tzanger | but again, -12dB when calling the local mW number tells me he has a problem in his part68 interface |
| 18:54.02 | tzanger | tzafrir_laptop: I'm positive that MG2 will work just fine, but they like to balme software so I'm making them inject a 0dBm signal into the tip+ring with a simulated phone line and measure the waveform at the DAA input pins, and then measure what is sampled |
| 18:54.09 | tzanger | something's not right and I"m positive it's in their design |
| 18:54.24 | TheDavidFactor | is it one of the sip headers? SIPCHANINFO has two IPs but they're both for the peer/client |
| 18:57.15 | *** join/#asterisk kpettit (~keith@99-172-37-26.lightspeed.tblltx.sbcglobal.net) |
| 18:58.33 | TheDavidFactor | the SIP header "TO" gives me what I need, thanks! |
| 18:58.51 | Qwell | TheDavidFactor: you're welcome! |
| 18:59.10 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn219.78-98-89.t-com.sk) |
| 18:59.15 | tzanger | man I feel like such an old fogie.. I still have my 1.4.x servers and 1.8 is in the works |
| 18:59.20 | tzanger | hell some of my contracts are still on 1.2.x |
| 18:59.36 | Qwell | Slugs_: ? |
| 19:01.34 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 19:03.57 | *** part/#asterisk kpettit (~keith@99-172-37-26.lightspeed.tblltx.sbcglobal.net) |
| 19:04.58 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 19:05.34 | *** join/#asterisk BezNalogov (~arjan@62.88.25.30) |
| 19:05.44 | BezNalogov | Hello people. |
| 19:05.57 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 19:06.22 | Kalidarn | hi, i'm trying tor un fxotune -i 4 but it causes my system to kernel panic, I'm running on freeebsd 8.1 |
| 19:07.07 | *** join/#asterisk deonv (~adium@pppoe-whk-127.cust.na.afrisp.net) |
| 19:07.18 | Kalidarn | it used to happen more before i made sure my TDM400P card was sharing an IRQ, its now on its own IRQ but it still happens. |
| 19:07.36 | BezNalogov | I used asterisk 1.4 before and in there I made an IVR based on opening hours. Now I migrated to asterisk 1.6 and this menu doesn't work anymore. I get this error: [Aug 12 21:06:29] WARNING[12194]: pbx.c:4349 __ast_pbx_run: Channel 'IAX2/000E3000448C-613' sent into invalid extension 's' in context 'tmp-mainmenu-be', but no invalid handler. The menu can be viewed here: http://www.pastebin.ca/1915935. What is wrong with my menu and how can I get it working |
| 19:07.36 | BezNalogov | under asterisk 1.6? |
| 19:07.55 | Kalidarn | fxotune.conf's last entry is 210=5,255,252,0,2,254,0,255,255 would this infer it got up to 210 before /dev/zap/210 crashing? |
| 19:08.24 | *** join/#asterisk alex_voip (~alex@201.161.45.81) |
| 19:08.51 | Kalidarn | infact all the entries from 1-210 have 5,255,252,0,2,254,0,255,255 as the recorded numbers, could i perhaps make up 211-252 from that? |
| 19:11.16 | henk | [TK]D-Fender: sorry for reacting a little harsh before, i was annoyed from work and assumed my question was clear and easy to answer without having a specific problem. looks like i was wrong. |
| 19:12.00 | alex_voip | hello trying to iax trunk a 1.4.32 box with 1.6.2.11 and the caller on 1.4 box gets hungup but i can see the iax dialog continuing between the 2 boxes and the 1.6 box running through the priorities...probably something simple but i'm just not seeing it |
| 19:15.48 | *** join/#asterisk simplydrew (~simplydre@198.7.249.227) |
| 19:18.30 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 19:19.15 | Kalidarn | oh i'm using asterisk 1.4. |
| 19:19.53 | Kalidarn | :( can't do fxotune -s get, open error on /dev/zap/1: Device not configured |
| 19:21.37 | BezNalogov | I used asterisk 1.4 before and in there I made an IVR based on opening hours. Now I migrated to asterisk 1.6 and this menu doesn't work anymore. I get this error: [Aug 12 21:06:29] WARNING[12194]: pbx.c:4349 __ast_pbx_run: Channel 'IAX2/000E3000448C-613' sent into invalid extension 's' in context 'tmp-mainmenu-be', but no invalid handler. The menu can be viewed here: http://www.pastebin.ca/1915935. What is wrong with my menu and how can I get it working |
| 19:21.37 | BezNalogov | under asterisk 1.6? |
| 19:21.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:24.02 | alex_voip | BezNalogov, tmp-mainmenu-be only has includes there is no s extension in it and that is where the call is being sent |
| 19:24.10 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 19:24.14 | Kalidarn | grrr :( kernel panic |
| 19:24.35 | *** join/#asterisk titter (~titter@c-98-208-152-139.hsd1.fl.comcast.net) |
| 19:24.48 | BezNalogov | alex_voip, This menu did work perfectly under asterisk 1.4. Now under 1.6 I get this error |
| 19:25.39 | BezNalogov | Normally the includes will include a s extension |
| 19:28.56 | [TK]D-Fender | All of the includes are CONDITIONAL... perhaps you should actually look athte current CONDITIONS. |
| 19:28.59 | titter | Just curious how people are handling receptionist phones, are you using the call parking feature, or using a queue? |
| 19:29.01 | [TK]D-Fender | like DATE & TIME |
| 19:30.20 | BezNalogov | I found the problem. in Asterisk 1.6 the | sign has been replaced with a comma (,)...... nasty.... |
| 19:30.31 | BezNalogov | I changed the | to comma's now and it works |
| 19:31.36 | [TK]D-Fender | \o/ |
| 19:31.53 | alex_voip | yeah sorry just went through changing my dialplan for that same reason and i just ignored it in yours :P |
| 19:33.37 | alex_voip | now if i could just figure out why 1.6 talks to itself.... |
| 19:33.46 | [TK]D-Fender | .... |
| 19:33.49 | [TK]D-Fender | pardon? |
| 19:33.49 | *** join/#asterisk mroe (~anon__@unaffiliated/roe) |
| 19:34.02 | BezNalogov | No problem, I was also already messing with it for 1.5 hours while the solution was actually so simple... |
| 19:34.05 | BezNalogov | Can happen |
| 19:34.08 | mroe | does anyone know of a way to test faxing if all you have available is the one fax machine you would like to test? |
| 19:34.29 | mroe | like is there some service that will send a test fax to a number? |
| 19:35.30 | titter | http://faxzero.com/ will send a free fax in about 15 minutes or so |
| 19:35.58 | Qwell | mroe: hp has a service. some 800 number |
| 19:37.05 | titter | mroe: the HP one you send a fax from your machine to their toll free, and they send one back. |
| 19:37.18 | mroe | titter, that is handy |
| 19:37.20 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 19:38.03 | Kalidarn | hmm so when i do fxotune -s why does it tell me "open error on /dev/zap/1: Device not configured" |
| 19:38.21 | Kalidarn | i have my fxotune.conf file 1-252 entries in /etc. |
| 19:38.29 | *** join/#asterisk Alagar (~Administr@122.164.36.142) |
| 19:38.33 | Kalidarn | the asterisk service is stopped. |
| 19:39.03 | Kobaz | axeterisk |
| 19:39.04 | Kalidarn | I was referring to http://www.voip-info.org/wiki/view/Asterisk+fxotune which seemed pretty straight forward |
| 19:46.45 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 19:49.55 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 19:58.30 | jamko | Anyone know of a good wholesale origination and termination provider, that supports T.38 traffic?? |
| 19:58.45 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 19:58.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:00.33 | *** join/#asterisk rizwank (~rizwank@173.153.22.138) |
| 20:01.06 | rizwank | Hi there! Does anyone have any recommendations on the best way to prepare music for MusiconHold or MP3Playback so that it sounds decent on a G711 connection... or (gasp!) a G729 connection... |
| 20:01.52 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 20:02.59 | jamko | music on hold sucks on compressed channels. G711 should be decent right out of the box. |
| 20:03.36 | rizwank | What can do I do the music as a pre-process to help.... make a mono-mix, make it 8bit, reduce the freq range...? |
| 20:03.44 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:04.22 | [TK]D-Fender | rizwank: G.729 was not MADE for "music" frequencies |
| 20:04.40 | RoyK | neither was g711 |
| 20:04.44 | rizwank | yeah. Sorry, I meant, on G711. =) |
| 20:04.47 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:04.48 | [TK]D-Fender | rizwank: It tends to make it sound shitty. |
| 20:05.02 | [TK]D-Fender | rizwank: You shouldn't have to do anything for G.711 |
| 20:05.04 | rizwank | I totally get that. |
| 20:05.16 | rizwank | I'm just trying to make it sound *less* shitty. (Working within constraints.) |
| 20:05.54 | RoyK | rizwank: transcode it to 8bit 8kbps, that's what g711 is |
| 20:06.27 | [TK]D-Fender | Which will sound totally differnt than * transoding it like it already does... |
| 20:06.34 | [TK]D-Fender | CRAZY PEOPLE |
| 20:06.51 | Kobaz | for shizzle |
| 20:09.47 | *** join/#asterisk bkruse (~bkruse@76.73.216.184) |
| 20:09.47 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:10.23 | RoyK | [TK]D-Fender: it'll save some cpu, and other software have better ways of transcoding music to alaw than the ones in asterisk |
| 20:10.53 | [TK]D-Fender | ok/fine/sure...... |
| 20:11.02 | *** join/#asterisk Beltechs (~Beltechs@rrcs-76-79-247-40.west.biz.rr.com) |
| 20:11.15 | war9407 | anyone know why when I use iAX2 dialer for the iphone, it sounds great but if I setup a Monitor() for it, it sounds like crap, but if I disconect from the iphone and it only plays the other person talking on the other side, it sounds fine? |
| 20:11.31 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 20:11.33 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-137.dhcp.embarqhsd.net) |
| 20:11.53 | t_dot_zilla | any body ever use voipmonitor.org ? |
| 20:12.06 | *** join/#asterisk flambo (~luke@62.40.36.13) |
| 20:15.57 | *** join/#asterisk bkruse (~bkruse@76.73.216.184) |
| 20:15.57 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:17.38 | [TK]D-Fender | war9407: My guess being that Monitor gets the raw end of the jitter-buffer |
| 20:18.31 | mroe | talking to telco representatives that don't understand the service they are providing is extremely frustrating |
| 20:18.50 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 20:19.08 | RoyK | are there any plans on opening asterisk as in pure GPL, or does Digium still stick to this quasi-openness? |
| 20:19.25 | RoyK | there's a lot of code out there that could be used to fix things........ |
| 20:19.36 | russellb | RoyK: welcome back, troll |
| 20:19.41 | [TK]D-Fender | RoyK: * is released as GPL |
| 20:20.35 | RoyK | hehe |
| 20:20.36 | RoyK | [TK]D-Fender: I know, but no GPL code gets into * without disclaiming the code...... |
| 20:20.36 | leifmadsen | that's totally true |
| 20:20.48 | leifmadsen | and it will not change in any foreseeable future |
| 20:21.04 | RoyK | I feared taht |
| 20:21.06 | RoyK | that |
| 20:21.28 | RoyK | if the Digium folks opened up, it'd be the best PBX in history |
| 20:21.36 | russellb | it already is, kthx |
| 20:21.38 | Qwell | Go ask the GNU folks why all of their software requires similar licensing. |
| 20:21.49 | russellb | Qwell: even more so, they require copyright assignment |
| 20:21.53 | [TK]D-Fender | RoyK: Just because its released as GPL doesn't mean the main branch has to accept your patches and be redistributed as such. |
| 20:21.57 | *** mode/#asterisk [+b *!*roy@cFDB1BF51.dhcp.bluecom.no] by leifmadsen |
| 20:21.57 | *** kick/#asterisk [RoyK!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (RoyK) |
| 20:22.08 | russellb | burn |
| 20:22.08 | seanbright | well... |
| 20:22.08 | Qwell | He's welcome to help fork it again. It worked out so well last time. |
| 20:22.12 | leifmadsen | this was going nowhere fast |
| 20:22.20 | seanbright | he's stating opinion |
| 20:22.21 | [TK]D-Fender | RoyK: Doesn't make it any less GPL. It just means they own your submission for the right to re-use intenally for whatever closed offshoot they feel like |
| 20:22.31 | seanbright | nothing he said was inflammatory |
| 20:22.44 | [TK]D-Fender | Qwell: Which fork? |
| 20:22.53 | russellb | seanbright: true |
| 20:23.00 | seanbright | kickbanning was a bit much |
| 20:23.03 | leifmadsen | inflammable means flammable?! |
| 20:23.03 | russellb | though he does have a history of being a serious troll |
| 20:23.13 | thehar | trolls about |
| 20:23.28 | seanbright | he's infamous... you know... more than famous |
| 20:24.18 | russellb | who wants to get me a soda |
| 20:24.22 | russellb | i will totally pay you 15 cents |
| 20:24.44 | thehar | hand deliver the 15 cents USD and I'll get you a soda. |
| 20:25.10 | russellb | i'll hand deliver a swift kick in the balls |
| 20:25.11 | russellb | how's that! |
| 20:25.20 | Qwell | You kick with your hands? |
| 20:25.21 | russellb | i lie. i just didn't have a good comeback. |
| 20:25.22 | Qwell | FREAK. |
| 20:25.29 | thehar | hey now |
| 20:29.57 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:30.00 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 20:30.36 | csnook | I feel like I'm missing something obvious, but I can't find any actual packages on packages.asterisk.org |
| 20:31.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 20:31.26 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 20:31.35 | csnook | so, um, where do I download it? |
| 20:34.16 | Qwell | csnook: are you on CentOS 5? |
| 20:34.27 | csnook | yup |
| 20:34.42 | Qwell | http://packages.asterisk.org/centos/5/current/i386/RPMS/asterisknow-version-1.7.0-1_centos5.noarch.rpm |
| 20:34.46 | Qwell | install that, then you can use yum |
| 20:35.09 | csnook | thanks |
| 20:35.27 | mmlj4 | I get them: http://packages.asterisk.org/centos/5/current/i386/RPMS/ |
| 20:35.38 | csnook | I guess "tested" is dead? |
| 20:36.13 | csnook | btw, the asterisk.org install instructions need updating |
| 20:36.29 | Qwell | howso? |
| 20:36.40 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:37.01 | csnook | they enable tested by default, not current |
| 20:37.07 | csnook | and tested is empty |
| 20:37.15 | Qwell | no it doesn't |
| 20:38.08 | csnook | okay, the ones at /downloads/yum are right |
| 20:38.13 | *** join/#asterisk deonv (~adium@pppoe-whk-127.cust.na.afrisp.net) |
| 20:38.19 | Qwell | what are you referring to? |
| 20:38.33 | mmlj4 | we're just full of incorrect info today |
| 20:39.28 | csnook | nevermind, I had a dyslexic moment |
| 20:39.54 | csnook | thanks for setting me straight |
| 20:40.22 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 20:40.44 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:41.23 | titter | Is there a way to ring members in a queue in the order they are listed in the queues.conf, and also if they are on a call to not ring their phone, but skip to the next member? round robin sort of does this, but if the second member is avilable and they answered last ... the next call will go to member number three, even if the second member is off the call. |
| 20:44.02 | mmlj4 | I could do that with a AGI script, dunno if * supports it out of the box |
| 20:44.52 | xheliox | titter: Weights and ringinuse=no |
| 20:45.02 | xheliox | with call limit (presuming you're using SIP) |
| 20:45.58 | xheliox | Sorry, not weights.. penalties -- same difference though. |
| 20:46.39 | xheliox | agents with the higher penalty will be tried last. |
| 20:46.41 | csnook | do I need the centos-digium repo if I'm only doing VoIP? |
| 20:47.25 | xheliox | You don't "need" it regardless of what you're using Asterisk for. |
| 20:47.34 | csnook | figured |
| 20:47.42 | titter | I tried both, and it didn't work ... I have the strat set to ring only, and assigned the penalties as 1-4 ... but if I call the queue as the second caller it rings penality 1 again |
| 20:47.54 | csnook | I just thought it might be some sort of addons for digium hardware |
| 20:48.07 | xheliox | titter: I suspect you don't have call limit set properly in sip.conf then. |
| 20:48.17 | csnook | which, given that my server is 1200 miles away, doesn't apply to me |
| 20:48.29 | Qwell | csnook: Digium commercial software. FaxForAsterisk, etc |
| 20:48.34 | csnook | gotcha |
| 20:49.18 | csnook | all I'm doing is interrupting people's dinners |
| 20:49.35 | xheliox | titter: Are you using chan_agent? |
| 20:49.58 | xheliox | or just SIP/agent_id ? |
| 20:50.40 | titter | xheliox: no, basically I am trying to simulate using the four line apperance for the Polycom with a queue for a receptionist. So I have 4 sip accounts registered to the phone as labeled them as line 1-4. I will check the call-limit ... didn't think of that |
| 20:51.03 | xheliox | for ringinuse=no to work properly, you have to have call-limit set. |
| 20:53.10 | titter | seems to work |
| 20:53.23 | titter | xheliox: thanks |
| 20:54.09 | xheliox | excellent.. np |
| 20:54.48 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 20:55.59 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 21:00.16 | [TK]D-Fender | checkout time, BBIAB |
| 21:10.35 | *** join/#asterisk mtryfoss (~chatzilla@80.239.93.22) |
| 21:11.21 | mtryfoss | anyone experienced one-way audio on a dahdi-channel ? |
| 21:12.41 | ChannelZ | Yeah I called this mute guy once |
| 21:12.53 | mtryfoss | hah. funny :) |
| 21:13.00 | tzafrir | mtryfoss, this is a call from DAHDI to? What DAHDI exactly? |
| 21:13.32 | mtryfoss | comes in through a dahdi-channel, and is bridged with iax2 to another server |
| 21:13.42 | mtryfoss | tried a mixmonitor on the dahdi, and no sound |
| 21:14.10 | ChannelZ | which way |
| 21:14.30 | mtryfoss | caller han hear my customer, but not opposite |
| 21:14.36 | mtryfoss | inbound call |
| 21:15.50 | mtryfoss | the funny thing is.. we have 22 E1's with a lot of traffic, and only one customer complaints |
| 21:16.10 | ChannelZ | maybe their phone is busted |
| 21:16.40 | mtryfoss | this only happens in about 1-2% of their calls |
| 21:17.01 | ChannelZ | you said it goes IAX2 to another server, then where does it go? |
| 21:17.07 | mtryfoss | as I wrote.. tried mixmonitor on the gateway, and it's true |
| 21:17.28 | mtryfoss | the first thing that happens when the call arrives is the monitor, and there's silence |
| 21:18.16 | titter | I have had it happen but it was dahdi -> iax -> sip -> pbx with down syndrome aka shoretel -> sysadmin with an iq of 6 configured IIS -> mpls -> remote office |
| 21:18.26 | mtryfoss | I'm wondering if those redirected calls sometimes sends some crazy signals which makes the channel not get completely up |
| 21:18.59 | csnook | mtryfoss, my old workplace used some voip service, and my boss would sometimes not be able to hear me when he called my cell |
| 21:19.05 | csnook | vonage, that's what it was |
| 21:19.13 | csnook | see if they're on vonage |
| 21:19.24 | mtryfoss | no voip involved here |
| 21:19.29 | mtryfoss | only isdn |
| 21:19.38 | csnook | I mean the customer |
| 21:19.42 | csnook | the customer is on isdn? |
| 21:19.47 | mtryfoss | yes |
| 21:19.52 | csnook | weird |
| 21:19.56 | csnook | shrugs |
| 21:22.30 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 21:27.18 | *** join/#asterisk beardy_ (~beardy@unaffiliated/beardy) |
| 21:27.35 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 21:29.00 | boodu | re |
| 21:31.35 | *** join/#asterisk b14ck (~b14ck@64.206.146.2) |
| 21:32.47 | *** join/#asterisk {Repelex} (~{Repelex}@201-71-129-252-arpa.vsp.com.br) |
| 21:33.45 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 21:34.04 | {Repelex} | hi... asterisk and java have a good integration ? |
| 21:34.56 | *** join/#asterisk digilink (~digilink@tn-76-5-159-171.sta.embarqhsd.net) |
| 21:35.03 | *** join/#asterisk aaronyy (~aaronyy@ipv6.aaronyy.com) |
| 21:35.03 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 21:35.29 | *** join/#asterisk aaronyy (~aaronyy@ipv6.aaronyy.com) |
| 21:35.29 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 21:40.23 | *** part/#asterisk {Repelex} (~{Repelex}@201-71-129-252-arpa.vsp.com.br) |
| 21:42.17 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:42.17 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta3 (2010/08/10), 1.6.2.11 (2010/08/10), 1.4.35 (2010/08/10), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 21:42.20 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 21:42.49 | csnook | Anyone know of a good doc for installing asterisk on centos without having the ability to install kernel modules? I'm just trying to do VoIP stuff, so I don't need the hardware support, but the package deps want to mess with my kernel. |
| 21:42.49 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 21:44.02 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 21:44.11 | *** join/#asterisk exothermc (~miles@174.127.153.10) |
| 21:44.52 | Micc_ | BezNalogov, did you solve your 1.6 include problem? |
| 21:45.13 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 21:45.17 | Micc_ | I think he left. |
| 21:46.09 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 21:46.40 | timholum | hello, I have a question, I have in my extenstions.conf exten => s,n,Dial(SIP/208&SIP/207&SIP/200&SIP/201&SIP/205|20|tT) is there a way to run a script based on which one picks up? |
| 21:47.02 | timholum | I would like to have a log of which tech picks up the phone |
| 21:47.18 | mmlj4 | you just want a log? |
| 21:47.30 | mmlj4 | that's logged already |
| 21:47.46 | timholum | and to be able to have my hang up script upload the recording and tag it apropriatly |
| 21:47.57 | mmlj4 | that's harder |
| 21:48.33 | mmlj4 | I haven't done anything like that... perhaps someone else knows? |
| 21:48.59 | timholum | there is there any enviromental variable that tells what extention picked up? |
| 21:49.10 | timholum | is there not there is there :) |
| 21:49.29 | mmlj4 | probably so |
| 21:51.55 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 21:53.05 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 21:53.06 | *** mode/#asterisk [+o file] by ChanServ |
| 21:53.41 | ChannelZ | csnook: build it yourself from the source |
| 21:53.51 | leifmadsen | timholum: try doing a DumpChan() in the 'h' extension and see if anything exists there to tell you who picked it up |
| 21:53.57 | csnook | ChannelZ, that's what I was afraid of |
| 21:54.11 | leifmadsen | it's not hard |
| 21:54.23 | leifmadsen | there's even a free book... |
| 21:54.37 | Qwell | leifmadsen: I read that it was written by nubs. |
| 21:55.23 | leifmadsen | Qwell: well it's free; what do you expect?! |
| 21:55.30 | Qwell | Asterisk is also free! |
| 21:55.34 | Qwell | are you suggesting... |
| 21:55.43 | leifmadsen | I might be |
| 21:55.55 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 21:55.56 | leifmadsen | ok, I am done -- time to go drink wine |
| 21:59.13 | ChannelZ | csnook: the package shouldn't require any kernel modules unless they built DAHDI into the same package. If it's separate, just don't install DAHDI |
| 21:59.33 | timholum | I have exten => s,h,DumpChan() and nothing happend when I hung up? |
| 21:59.46 | timholum | I have the console open, should i print stuff to the screen? |
| 22:00.30 | JerJer | no its h,1,blah |
| 22:00.32 | Qwell | csnook: it is separate |
| 22:00.47 | leifmadsen | timholum: you used 'h' as a priority, not an extension |
| 22:00.52 | JerJer | h is a special extension |
| 22:00.54 | timholum | ahh, ok :) |
| 22:01.00 | leifmadsen | *facepalm* |
| 22:01.12 | JerJer | here's your sign :) |
| 22:01.14 | csnook | I tried to install asterisk16 and it wanted to mess with my kernel |
| 22:01.31 | csnook | which, honestly, I'd be fine with, if I had the technical capability to do so |
| 22:01.32 | JerJer | csnook: must be some lame package then |
| 22:01.37 | Qwell | csnook: "asterisk16" is a meta package |
| 22:01.54 | Qwell | it just has deps on "standard" things. you can install just the -core package and it'll work fine. |
| 22:01.55 | csnook | Qwell, yeah, I figured that, digging into it now to see what I really need |
| 22:02.02 | csnook | ah |
| 22:02.05 | csnook | will do |
| 22:02.21 | timholum | DIALEDPEERNUMBER is what I am looking for thanks for the help :) |
| 22:03.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:07.29 | jamko | Anyone know of a good wholesale origination and termination provider, that supports T.38 traffic?? |
| 22:07.57 | mmlj4 | it's a free edition of an o'reilly book... those aren't written by newbies... if you want that, use apress |
| 22:08.29 | exothermc | jamko: national or regional? |
| 22:08.45 | jamko | US 48 for now |
| 22:09.07 | exothermc | jamko: There are a few that do term, but I'm not aware of anyone doing full 48. |
| 22:09.18 | exothermc | for origination that is. |
| 22:09.42 | jamko | Well I would settle for the origination not being full 48. |
| 22:10.00 | exothermc | jamko: Qwest does toll free origination with t.38, but that is about as good as it gets. |
| 22:10.16 | exothermc | 360 Networks has a fairly good foot print that does t.38 |
| 22:10.46 | exothermc | paetec or point 1 may also, but I'm not sure about their t.38. |
| 22:11.14 | exothermc | most of those have a west coast focus though. |
| 22:11.32 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 22:12.03 | exothermc | actually broadvox does a full deck with t.38 I believe. Their prices are so high though I wouldn't consider them wholesale even though they claim it. |
| 22:12.15 | jamko | Thanks! I use gafachi for the T.38 termination, but they do not offer local DIDs for Origination, and their TF termination is a "best effort" scenario, and they do not care much if your calls don't place. |
| 22:12.31 | jamko | on the TF that is. |
| 22:12.56 | exothermc | that is surprising. Typically providers love TF term, and make sure it works. |
| 22:13.10 | exothermc | If you have enough TF term, you can get paid for it. |
| 22:13.48 | jamko | Exactly what I said when I called today.. Supposedly they are "disussing" it, but for now don't care for it. |
| 22:13.54 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 22:14.04 | nix8n82 | what is TF? |
| 22:14.07 | jamko | I know Simwood out of the UK loves TF traffic because of wht he gets paid for it. |
| 22:14.23 | jamko | nix8n82: Toll Free |
| 22:14.29 | nix8n82 | thanks |
| 22:14.35 | exothermc | jamko: I don't think gafachi are that customer centered. |
| 22:14.47 | jamko | lol .. yea.. |
| 22:15.05 | exothermc | I wouldn't have even thought of them as a wholesale provider actually. |
| 22:15.26 | jamko | Unfortunately Simwood has cid issues on termination, and doesn't support T.38. |
| 22:15.46 | exothermc | jamko: sounds sketchy at best. |
| 22:15.55 | jamko | yea |
| 22:16.06 | jamko | Thanks again, I will check out 360 for sure |
| 22:16.33 | exothermc | jamko: Ya they are a largish, org, so nothing moves fast, but it is rock solid. |
| 22:16.46 | jamko | That |
| 22:16.53 | exothermc | jamko: I have had the odd routing issue with them though, but on the whole been satisfied. |
| 22:17.04 | exothermc | no more issues than we have had with VZB or Qwest though. |
| 22:17.15 | jamko | exothermc: gafachi or simwood? |
| 22:17.20 | exothermc | 360 |
| 22:17.28 | jamko | ahh |
| 22:17.38 | jamko | what type of odd routing issues? |
| 22:17.43 | exothermc | haven't used simwood, and bailed on gafachi years ago (which means nothing now) |
| 22:18.18 | exothermc | jamko: Just standard stuff, getting 503s when it is a valid DID, and once a 404 on a valid did |
| 22:18.44 | jamko | yea typical.. |
| 22:18.44 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 22:18.53 | exothermc | jamko: How many minutes a month of US term do you have? |
| 22:19.14 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 22:19.23 | jamko | Very minimal right now.. Maybe 20,000 at best |
| 22:19.32 | exothermc | jamko: lol ok. |
| 22:19.43 | exothermc | Ya I don't think you are looking for a wholesale provider then. |
| 22:19.55 | exothermc | minimum commits are going to start at $1k |
| 22:20.43 | exothermc | Just get a high rate reliable provider that you can send traffic to and forget about it then focus on growing the business instead of starting off chasing rock bottom rates. |
| 22:22.07 | jamko | Yea.. that's what I'm trying to do. Perfect the infrastructure, build the biz, and then go real wholesale. Right now I do fine at .009/minute |
| 22:23.09 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 22:23.46 | jamko | Shit I just got T.38 working properly yesterday on termination.. but did get my first Digium bug acknowledged: https://issues.asterisk.org/view.php?id=17842 |
| 22:24.22 | exothermc | jamko: If you can get me $0.01/min for the full US I'll send you 500,000 minutes per day to start. |
| 22:24.37 | Micc_ | We've been using Vitelity for years, pretty solid service for TF. |
| 22:24.52 | Micc_ | And they have failover DIDs you can set for each TF DID. |
| 22:25.09 | Micc_ | vitelity is 1.2 cents a minute I think. |
| 22:26.02 | exothermc | or even 48 state us. |
| 22:26.14 | jamko | Yea I have heard Vitality is pretty solid. I use Voip.MS for the fail over DIDs. Their management gui is real nice. Do you have an e-mail address? I or my partner can send you some information. |
| 22:27.00 | kfife | .01 to the 'full us' (not including rural clecs :-) ) |
| 22:27.39 | Micc_ | jamko, you said you got t.38 working properly yesterday, so did you find a work around for your re-invite bug? |
| 22:27.41 | xheliox | I've only been using VoIP.ms for a few months, but I'm highly impressed. |
| 22:27.42 | kfife | VITEL is nice but every now and then the two-bit carrier they farm out to messes up my outbound callerid. |
| 22:27.57 | kfife | Very infrequent, but it happens. |
| 22:28.02 | Micc_ | jamko, I've been playing with T.38 for almost a year now. It drives me crazy some times. |
| 22:28.41 | Micc_ | kfife, has that happened recently? I think they may have changed providers. |
| 22:28.51 | jamko | MIcc_ : yes I had to turn off t.38 reinvite from the provider side, and let my ATA initiate the T.38 re-invite. It seems a T.38 re-invite from a SIP cluster was causing my problem with Asterisk. |
| 22:28.54 | *** join/#asterisk jmacz (~jmacz@186.29.155.200) |
| 22:28.58 | kfife | Micc_: Hmmm. I want to say within the last two months. |
| 22:29.33 | Micc_ | jamko, so set canreinvite=false on the sip peer that is the provider? |
| 22:29.36 | kfife | Micc_: It's quite hard to tell right? You would only notice if your called party says: "WTF. why does your callerid say X" |
| 22:29.46 | exothermc | kfife: I still can work with that. |
| 22:30.09 | Micc_ | kfife, I think I remember seeing that sometimes. |
| 22:30.18 | kfife | exothermc: Micc_: Still they're great. Best in show says me. |
| 22:30.40 | kfife | My favorit thing about Vitelity is that they can set OUTBOUND CNAM. |
| 22:30.41 | Micc_ | kfife, I think they've changed how they do callerid too, I can't send them 3 digit callerid anymore, but I used to be able to. |
| 22:30.42 | exothermc | kfife: who? |
| 22:30.53 | jamko | Micc: no set canreinvite=yes across the boardm, and tell your provider NOT to issue a reinvite on T.38... Let you sip peer do the initial re-invite. |
| 22:31.15 | exothermc | kfife: outbound cnam isn't technically possible. |
| 22:31.45 | kfife | Be careful what you assert :-). In other words, called party sees "Karl Fife" or "YourCompanyName" instead of "Chicago, IL" |
| 22:31.57 | Micc_ | exothermc, yeah I was just going to say that too. |
| 22:31.59 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 22:32.05 | kfife | exothermc: CNAM is alive and well |
| 22:32.12 | jamko | Level 3 constantly swaps caller ids downstream on term providers. It sucs. |
| 22:32.22 | exothermc | kfife: Oh yI thought you said "send" It can't be sent, but you can set it, correct. |
| 22:32.49 | Micc_ | kfife, that all depends on the cnam lookup provider they are using though. They can't change them all on the fly. |
| 22:33.11 | kfife | You send the number, the terminating telco dips against the LIDB entry managed by the originating telco. Terminating telco pays, originating telco gets paid. |
| 22:33.26 | exothermc | Micc_: Right you can't change it on a per call basis, but it is near realtime. |
| 22:33.26 | joobie | hey guys.. anyone got an idea on how to use the queue system in asterisk, but make the queue member a PSTN telephone number (that's seperate to the asterisk box) |
| 22:33.28 | jamko | kfife: let me elaborate, I have canreinvite=yes in sip.conf for all peers, and then I actually go to my providers website and turn off T.38 reinvite. |
| 22:33.48 | Micc_ | exothermc, right and with vitelity it costs 10$ I think or 5$. |
| 22:34.10 | kfife | Micc_ if the CNAM source is some half-assed google lookup, then yes, it depends. If the CNAM source is the LIDB, then no. |
| 22:34.15 | exothermc | Micc_: To set names on DIDs? that doesn't make sense. |
| 22:34.22 | Micc_ | kfife, exothermc, I would think that being able to set the cnam on a did you order is a requirement that all providers should be able to do. |
| 22:34.45 | exothermc | Micc_: mostly because they would be stupid not to, yes. |
| 22:35.24 | Micc_ | Is there a way to interface with the LIDB directly? |
| 22:35.49 | exothermc | Micc_: You can, but easier to use an aggregator like SNET or TNSi |
| 22:36.17 | exothermc | Micc_: then you can dip directly using SIP, SS7 or SIGTRAN |
| 22:36.21 | kfife | LIDB is tightly controlled by the ILECS and CLECS because it's a valuable asset. Vitel has figured out a way to legally obligate the 'keepers' of the LIDB to populate with their records. |
| 22:37.07 | kfife | When you dip the 'real' LIDB it costs money every time, and you pay about 2¢ per dip, and you sign an agreement not to cache. |
| 22:37.33 | exothermc | kfife: "obligate" is wrong, if you actually own DIDs/TNs you are required to have access to set them. and by own I mean there is no one above you. |
| 22:38.04 | exothermc | and if you are paying $0.02/dip you are getting bent over and..... |
| 22:38.07 | Micc_ | which means you would have to be a clec to own the dids. |
| 22:38.16 | kfife | exothermc: I believe you are wrong. |
| 22:38.23 | *** join/#asterisk flapjacks (~flapjacks@wsip-72-214-208-206.ph.ph.cox.net) |
| 22:39.07 | Micc_ | Last time I asked vitelity about their clec status, they said they were not a clec. |
| 22:39.14 | Micc_ | and they didn't need to be. |
| 22:39.17 | kfife | exothermc: I'd be very interested to know a provider who will do 'real' CNAM dips against the LIDB for less than 2¢ with less than a $1000/month committment. |
| 22:39.31 | Micc_ | but I would think they must have another entity that is that they control or work closely with. |
| 22:39.39 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 22:39.42 | exothermc | kfife: Our rate with TNSi is about $0.0043 per dip |
| 22:39.52 | exothermc | kfife: No commit. |
| 22:40.01 | flapjacks | Im tring to pass a did from my zaptel card to my legacy PBX so it now how to route the call it will ring the legacy PBX but does not pass the did what im i missing |
| 22:40.19 | kfife | exothermc: How do you connect? |
| 22:40.50 | exothermc | kfife: IP |
| 22:41.04 | kfife | exothermc: how much money do you spend with them in a month? |
| 22:41.19 | Micc_ | flapjacks, what does your dial command look like? |
| 22:41.29 | exothermc | kfife: actually just looked at the bill it is $0.004 /dip if it is on net and $0.0055 if it is off net. |
| 22:41.39 | flapjacks | dial(ZAP/g0/didnumer) |
| 22:41.54 | exothermc | kfife: we spend $23k with them a month now, but started out with next to 0. |
| 22:42.03 | flapjacks | one i pick up the phone i hear dtmf tones like its trying to dial the number |
| 22:42.22 | exothermc | but we also do our LRN dips with them, and SS7 for our TDM stuff with them. |
| 22:42.49 | exothermc | but those were also more recent additions. |
| 22:43.06 | kfife | exothermc: Is this their 'home-brewed' data source or is it ultimately the same source that Verizon dips? |
| 22:43.18 | Micc_ | flapjacks what kind of pbx is the legacy pbx and how is it connected to asterisk? through the PSTN? |
| 22:43.25 | kfife | exothermc: I hope you're right by the way. |
| 22:43.30 | exothermc | they may have a commit now, but I would be surprised. $0.02 has an extreme amount of gravy built in. |
| 22:44.05 | exothermc | kfife: It is LIDB |
| 22:44.49 | kfife | exothermc: Not bad. I know a lot of CNAM providers were recently 'chased' (read threatened) out of buisness by upstream providers. |
| 22:44.52 | Micc_ | exothermc, what is LRN? |
| 22:45.04 | exothermc | kfife: The LRN stuff is sourced from telcordia I think but that is the only DB that you are actually allowed to buy and cache. |
| 22:45.15 | flapjacks | its a samsung OfficeServ500 connect to my rinoT1 card which is acting like the PSTN for the PBX I can make calls from the samsung out the rhinoT1 and out sip trunk provider |
| 22:45.36 | exothermc | Micc_: If a number is ported from one carrier to another the LRN dip will show what carrier it is actually pointed at now. |
| 22:46.18 | flapjacks | Micc_: I can recive calls to on the samsung they just ring the operator group because no did was seen |
| 22:46.25 | exothermc | kfife: Ya not to many upstreams allow you to resell access to the DB. |
| 22:46.30 | exothermc | for obvious reasons. |
| 22:48.04 | kfife | exothermc: I wonder if TNSi is in violation of their TOS. The compensation system is a bit messed up these days, but it was my understanding that the originating LEC was compensated right around one cent for the dip. It may depend on who you are too--in other words I'm sure AT&T doesn't pay Verizon 1¢--but I |
| 22:48.05 | exothermc | kfife: Just to put the cnam stuff in perspective for you. $0.002 / dip would be a high rate for payout, much less a sell rate. |
| 22:48.28 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 22:48.36 | exothermc | kfife: ya no $0.01 is really high. |
| 22:48.38 | kfife | I'd bet some rural lec might a lot more. |
| 22:49.00 | exothermc | rates are all individually negotiated between carriers. |
| 22:49.12 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 22:49.16 | exothermc | Well you see it isn't that simple. |
| 22:49.41 | exothermc | you can't just charge "more" since you have to have an aggreement to do so. and getting a big boy to agree to $0.01 won't happen. |
| 22:50.01 | kfife | I'm happy to use TNSi. I havent' shopped it in years. It is (was) obviously a racket, and maybe they've broken the BS in the supply chain. It was bound to happen. |
| 22:50.20 | exothermc | plus as a small fish trying to get agreement with hundreds of carriers..... |
| 22:51.31 | exothermc | That is the reason TNSi exists. You give them the keys for your cnam data, then they pay you $0.00143 or around there depending on the source carrier of each dip, and they take a cut and make all the magic happen. |
| 22:51.54 | exothermc | They already interface with all the carriers at some level. |
| 22:52.12 | exothermc | I think it is them and SNET who are basically in the game anymore. |
| 22:52.26 | exothermc | and SNET got bought out by ATT(and company) |
| 22:52.45 | kfife | beautiful. I always thought it was a matter of time before somebody was able to do that without pissing off the incumbents!! |
| 22:52.53 | kfife | exothermc: ^^^^ |
| 22:52.56 | kfife | Thanks for the tip! |
| 22:53.16 | exothermc | but ya if you have someone selling you at $0.02 they are either taking a big cut or getting screwed themselves. |
| 22:53.43 | kfife | I've talked to the proprietor. He's getting screwed. |
| 22:54.00 | kfife | only slightle less than I am :-) |
| 22:54.04 | exothermc | get a TNSi connection and sell to him. |
| 22:54.21 | exothermc | of course getting them to give you legal rights to do so will be a miracle. |
| 22:54.44 | exothermc | but you don't have to tell them, and they should never know. |
| 22:55.10 | kfife | exothermc: LOL |
| 22:55.29 | Qwell | who actually owns/runs ILDB? |
| 22:55.34 | Qwell | LIDB* |
| 22:55.54 | kfife | Qwell: good question |
| 22:56.08 | exothermc | a great racket is get your own numbers, then you are paid for the dips, then you just place outbound calls to a source that will dip them, and never connect the calls. Kinda like printing money. |
| 22:56.36 | exothermc | Qwell: It is kinda like who owns/runs DNS? |
| 22:56.54 | Qwell | fair enough |
| 22:57.26 | exothermc | Qwell: managed by Telcordia I believe, either that or NEUSTAR. |
| 22:59.14 | exothermc | Qwell: but ya being able to access the LIDB I think is kinda like being a root registrar, you need to be certified that you know what you are doing etc. |
| 23:04.12 | Micc_ | exothermc, Maybe thats why I get all these calls to all of my customers from DOCTORS NETWORK, but they never connect. |
| 23:06.18 | Micc_ | They call someone on my network almost every day, somtimes many different customers in a day. Theres no way they know all these different customers and their DIDs in different rate centers. Its not like they are all grouped together or something. |
| 23:06.34 | Micc_ | It really makes me wonder wtf they are doing. |
| 23:07.03 | Micc_ | And what really ticks me off is sometimes it makes aastra phones ring forever until each phone is picked up. |
| 23:07.54 | Micc_ | which reminds me, I need to call aastra about that. Theres got to be something they can do in the mean time until asterisk can fx it the right way. |
| 23:13.40 | exothermc | Micc_: Ya that isn't it, because if someone picks up then their will be a duration no matter how small and whoever would just be going backwards. |
| 23:14.00 | kfife | Micc_: can't you just have asterisk ring them for x seconds then fall through to the next priority (for example answer, and play them monkeys, or "Hang the hell up" message :-) |
| 23:15.14 | Micc_ | kfife, yeah but the did is always different. not always, but it seems to be a pretty big range. |
| 23:15.37 | kfife | exothermc: Do you know the mechanism by which companies like YouMail (and MNO's |
| 23:15.37 | kfife | using their own voicemail system) are able to redirect ALL calls from a ALL |
| 23:15.37 | kfife | subscribers to *just one* voicemail DID, yet determine WHICH subscriber did |
| 23:15.37 | kfife | the redirection? |
| 23:17.08 | kfife | Micc_: Do you have hundreds of DID's or are you sending just ANY sip call to your phone? |
| 23:17.21 | exothermc | kfife: I'm not sure I get what you are saying, but if you mean the "new VM provider" be able to tell which voicemailbox and original calling party? |
| 23:17.52 | *** join/#asterisk roe (~roe___@unaffiliated/roe) |
| 23:18.33 | kfife | For example T-Mobile sends ALL unanswered calls from all subscribers to ONE voicemail system access DID number. |
| 23:19.06 | exothermc | kfife: Ok ya that is just do to the fact that SS7 has more information built in than SIP does. |
| 23:19.30 | exothermc | SS7 have information about DNIS and ANI, but also billing DNIS and billing ANI. |
| 23:19.40 | flapjacks | would it be the pbx or zaptel config issue that is wiating to send the DID number until after the call is picked up on the zaptel channel |
| 23:20.02 | Micc_ | kfife, we have hundreds of customers, so we have hundreds of DIDs. |
| 23:20.24 | kfife | exothermc: I'm talking ISDN PRI. |
| 23:20.33 | kfife | exothermc: I had always assumed this was simply done using RDNIS. In other words, the original calling party's CallerID is passed with the redirected call, (and I assumed) the redirecting subscriber's number was passed via RDNIS--thusly the voicemail system knows to place the call into the voicemail account belonging to the RDNIS value. |
| 23:20.59 | *** join/#asterisk ManxPower (~manxpower@user-24-236-87-78.knology.net) |
| 23:21.41 | kfife | Is there a 'normal' way to do this (such as a QSIG call transfer message), or is it more likely a home-spun and carrier-specific message, such as a Q.931 facility message. |
| 23:22.03 | exothermc | kfife: I'm not sure you have much control over what is put on the PRI. I know those things are set, but I would have to look at libpri to see what is exposed. |
| 23:23.12 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 23:23.27 | kfife | exothermc: I'll just have to debug the PRI channel and redirect the calls to myself so I can see what's visible. Basically I want to redirect calls to Youmail for their transcription service without using DropBox because doing so obliterates the original CallerID |
| 23:23.32 | exothermc | kfife: I'm out of my water on this one, but I know on ss7 you have access to 4 number fields. |
| 23:24.32 | kfife | exothermc: I see. So big boys like MNO's have better-than-PRI type connections enabling them to do telephone voo-doo that may be outside of scope for a lowly PRI group. |
| 23:25.07 | exothermc | kfife: you can just do the transcription via wav file and http service, probably going to get better results. |
| 23:25.16 | ManxPower | I just arrived. What specifically do you need to do? |
| 23:25.40 | kfife | exothermc: you mean for example via the YouMail API? |
| 23:25.50 | exothermc | kfife: Correct, but with that "may be" caveat. I'm pretty new to PRIs. |
| 23:26.16 | exothermc | kfife: I didn't know youmail allowed wav file transcription. |
| 23:26.44 | kfife | exothermc: The idea is to give smartphone users access to all features of the well-implemented youmail apps--including from our terrestrial lines. |
| 23:26.48 | Micc_ | I'm lookin at the cdrs for all my customers and this DOCTORS GROUP or DOCTORS NETWORK comes up with disposition answered but generally always about 3-6 seconds in billsec. The callerID numbers they call from are varied but looks like within the same block maybe. |
| 23:26.53 | exothermc | kfife: I'm saying their are services out there that will allow you to send them a wav file via http API call and get back text. |
| 23:27.03 | ManxPower | It is trivial to set your own callerid on a PRI (or SIP) if your carrier allows you to. (many do) |
| 23:27.23 | ManxPower | Specifically the Caller*ID NUMBER. Name can also be sent to the telco, but they will always ignore it. |
| 23:27.36 | Slugs_ | Question about asterisk call queues, if all agents are busy in a queue, it says 'your call cannot be connected at this time' instead of going to the mailbox that was setup. |
| 23:27.44 | exothermc | ManxPower: Ya you are missing the point. |
| 23:27.57 | ManxPower | exothermc, I did ask for details. Nobody provided any. |
| 23:28.03 | exothermc | ManxPower: if you pass just the CLI then the service won't know which VM box to attach it too. |
| 23:28.16 | kfife | exothermc: Such a service may not have the well-implemented smartphone apps. YouMail does transcription via DropBox (email), but like I said they fu¢k up the callerid. |
| 23:28.50 | ManxPower | exothermc, So you want to pass DNID to the destination telephone number? |
| 23:28.52 | exothermc | kfife: You sure that you are passing out the correct caller ID? |
| 23:29.09 | ManxPower | Do you need to know the callerid info of the caller at all? |
| 23:29.11 | kfife | exothermc: I'll look into their API, but redirecting the call with their RDNIS like VOO-DOO is the simplest, most robust solution. |
| 23:29.21 | exothermc | ManxPower: Well there is the key, you need to pass both the DNIS and the billing DNIS. |
| 23:29.45 | ManxPower | exothermc, that does make it more complicated 8-) |
| 23:30.09 | exothermc | ManxPower: They use one to see what number is actually sending to them (for the correct voicemail box) then they use the other to put on the mail to say who the call actually originated from. |
| 23:30.18 | kfife | exothermc: Yes. We've redirected the calls to ourselves, and it's showing up right. My bet is on QSIG redirect |
| 23:30.53 | ManxPower | exothermc, *nod* Maybe 2BCT is something that should be investigated. |
| 23:32.43 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 23:33.07 | mmlj4 | what up, ManxPower? |
| 23:33.43 | kfife | Love the new 1.8 chanspy: S: Stop when no more channels are left to spy on. |
| 23:34.36 | Micc_ | kfife, http://www.voip-info.org/wiki/view/RDNIS talks about RGN and using it to know which voicemail box to use but says it only works on a PRI. |
| 23:35.01 | Micc_ | kfife, how stable is 1.8? |
| 23:35.02 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 23:35.14 | *** join/#asterisk twisted (~twisted@c-71-207-231-57.hsd1.al.comcast.net) |
| 23:35.14 | *** mode/#asterisk [+o twisted] by ChanServ |
| 23:36.49 | Slugs_ | Question about asterisk call queues, if all agents are busy in a queue, it says 'your call cannot be connected at this time' instead of going to the mailbox that was setup. |
| 23:39.23 | Slugs_ | hardwire: you there :) |
| 23:39.38 | hardwire | Slugs_: no .. I died |
| 23:39.57 | hardwire | check your context and queue flags. |
| 23:40.10 | hardwire | make sure you have your priority numbers in order or 'n' |
| 23:40.20 | Slugs_ | sorry to hear that, and ok |
| 23:40.21 | kfife | Micc_: Right. RDNIS is not specified is SIP. (BUt it is in SIP-T) |
| 23:40.25 | hardwire | make sure the PBX at your office isn't saying that vs asterisk |
| 23:40.27 | hardwire | turn on verbose. |
| 23:40.38 | Slugs_ | k |
| 23:40.42 | kfife | Micc_: Beta-3 Not going to risk my job over it :-) |
| 23:40.51 | kfife | Micc_: not on a production system |
| 23:42.38 | Micc_ | right, I just remember 1.6.0 was super unstable even after final release, it wasn't till about 1.6.0.6 I think it started to get better. |
| 23:43.58 | Micc_ | It was about 6-8 months after first release that it was stable enough to run in production. |
| 23:44.58 | leifmadsen | you can't run any commercial modules at this point either, so you're pretty much limited to development servers at this point unless you don't use G729, Skype, etc... |
| 23:47.11 | kfife | leifmadsen: Thanks for the tip the other day about 1.8 chanspy |
| 23:47.19 | leifmadsen | kfife: hope it was useful |
| 23:47.21 | kfife | Is there a backport to 1.6.x |
| 23:47.43 | kfife | leifmadsen: ^^ ? |
| 23:47.54 | leifmadsen | have you created it? :) |
| 23:48.10 | leifmadsen | it's probably not that difficult unless it uses something crazy |
| 23:48.21 | leifmadsen | just do a diff between the files and find the feature you want |
| 23:48.25 | leifmadsen | might not be very difficult |
| 23:48.27 | kfife | leifmadsen: lol. I'm not technically there yet. One day I will be. |
| 23:48.44 | kfife | leifmadsen: good idea. |
| 23:48.51 | Micc_ | I've never used chanspy. |
| 23:49.19 | Micc_ | It sounds like a lot of fun, but just never needed it. |
| 23:49.47 | kfife | leifmadsen: If there's no crazy changes, would it be as simple as copying the app_chanspy.c from the 1.8 source tree to the 1.6.2.x tree? |
| 23:49.56 | leifmadsen | kfife: it might be :) |
| 23:50.48 | kfife | leifmadsen: Please evaluate this statement: "If it works for a handful of test calls, it probably wont cause asterisk to segfault. " :-) |
| 23:51.10 | leifmadsen | YMMV :) |
| 23:51.38 | kfife | 1 = Strongly disagree, 10 = Strongly agree, 11 = this is my favorite guitar--it's so special that you shouldn't even look at it. |
| 23:52.11 | kfife | It's ONE louder |
| 23:52.27 | leifmadsen | [1..11] |
| 23:52.46 | leifmadsen | exten => start,1,Set(Result=${RAND(1,10)}) |
| 23:53.03 | kfife | leifmadsen: :-) |
| 23:54.17 | [TK]D-Fender | kfife: http://xkcd.com/670/ |
| 23:55.47 | kfife | [TK]D-Fender: :-) |
| 23:57.30 | jamko | is there a cli command to show the t.38 codec when it is in use?.. ie sip show channels (which does not show t.38) |
