| 00:11.53 | *** join/#asterisk jgowdy (~jgowdy@basharteg.com) |
| 00:13.02 | jgowdy | any idea when we can see the patches for issue 19303 make it into a release? they're still sitting in trunk and head of the 8 and 10 braches |
| 00:13.08 | jgowdy | https://issues.asterisk.org/jira/browse/ASTERISK-19303?page=com.atlassian.jira.plugin.ext.subversion%3Asubversion-commits-tabpanel#issue-tabs |
| 00:13.27 | jgowdy | https://code.asterisk.org/code/changelog/asterisk?cs=358116 |
| 00:19.29 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 00:37.11 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 00:42.01 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:02.45 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:02.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:35.13 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 02:05.26 | *** join/#asterisk caruizdiaz (~carlos@190.128.164.26) |
| 02:06.50 | caruizdiaz | Hello. Have anyone used chan_dongle successfully? |
| 02:09.10 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:09.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 02:10.45 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 02:17.35 | *** join/#asterisk dtrickie (47f18595@gateway/web/freenode/ip.71.241.133.149) |
| 02:17.58 | dtrickie | hey can anyone give me a hand really quick ? |
| 02:23.45 | caruizdiaz | everything looks dead here |
| 02:24.12 | *** join/#asterisk timahvo1 (~rogue@197.178.136.18) |
| 02:33.50 | [TK]D-Fender | 3 whole minutes |
| 02:34.01 | [TK]D-Fender | Now THAT's patience. |
| 02:47.48 | leifmadsen | ~ask |
| 02:47.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:54.51 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 03:12.35 | *** join/#asterisk luckman212_phone (~luckman21@2001:470:8abb:0:358d:eb41:5927:15c4) |
| 03:14.51 | *** join/#asterisk ppc (ppc@cloudy.cmang.org) |
| 03:15.06 | *** join/#asterisk luckman212_phone (~luckman21@2001:470:8abb:0:358d:eb41:5927:15c4) |
| 03:30.46 | *** join/#asterisk tac2 (~tac2@182.52.68.227) |
| 03:30.58 | *** part/#asterisk tac2 (~tac2@182.52.68.227) |
| 03:31.46 | *** join/#asterisk tac2 (~tac2@182.52.68.227) |
| 03:47.55 | ChannelZ | claps like a seal |
| 04:10.44 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 04:45.30 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 04:52.37 | *** join/#asterisk e_t_ (~et@2001:470:1f0f:e67:201:2eff:fe2b:386b) |
| 04:55.11 | *** join/#asterisk ks3_ (ks3@tatooine.digitallotus.com) |
| 04:58.29 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 05:03.21 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 05:12.04 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 05:26.04 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 05:38.21 | *** join/#asterisk Pizza401 (~Pizza401@unaffiliated/pizza401) |
| 05:48.40 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 05:59.56 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:07.34 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 06:08.19 | *** join/#asterisk din3sh (~din3sh@ADSL-TPLUS-111-34.telecomplus.net) |
| 06:10.02 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:26.29 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 06:34.40 | tac2 | I'm using the latest version stable of Asterisk 1.8 (1.8.11) as a ubuntu package and can't for the life of me get it to recognise the keys I've generated using astgenkey |
| 06:35.15 | tac2 | Does anyone have any advice? They're chowned to the asterisk user and I've placed them in /var/lib/asterisk/keys (I had to create the keys directory) |
| 06:45.21 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:01.27 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 07:07.13 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:23.44 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:25.10 | ChannelZ | Is that where your package is configured to look? (core show settings) |
| 07:28.41 | tac2 | ChannelZ: It is indeed however it seems the openssl library wasn't installed by apt-get so perhaps res_crypto isn't enabled in the ubuntu packages |
| 07:28.52 | tac2 | Am building from source instead to see if that sorts it |
| 07:29.56 | tac2 | That has sorted it so I'm guessing keys arn't possible in the ubuntu package release |
| 07:30.18 | tac2 | ChannelZ: Thanks for your tip tho :-) |
| 07:30.45 | ChannelZ | yay |
| 07:31.31 | *** join/#asterisk PBXman (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 07:36.35 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:fd7c:ef24:7f8f:bff9) |
| 07:39.17 | *** join/#asterisk jkroon (~jkroon@dsl-244-29-126.telkomadsl.co.za) |
| 07:39.27 | jkroon | hi guys, seeing loads and loads of [Apr 10 09:18:57] ERROR[11752] res_timing_timerfd.c: Call to timerfd_gettime() error: Bad file descriptor |
| 07:40.17 | jkroon | 2357050 lines in my log today alone! |
| 07:40.24 | ChannelZ | Hmm. |
| 07:41.49 | jkroon | asterisk 1.8.10.1 |
| 07:42.21 | ChannelZ | not sure why your timerfd would be failing but I guess you could switch and try pthread |
| 07:43.31 | jkroon | pthread ... much worse in terms of quality |
| 07:44.13 | jkroon | i suspect it'll be another double close bug somewhere in the code ... we've seen cases before where code does stuff like fclose(fp); close(fd); ... whereas the manpage on fclose indicates that fd already gets closed. |
| 07:44.28 | jkroon | and then inbetween those two calls a different thread end up re-using the fd. |
| 07:44.35 | ChannelZ | Does it go away if you restart and then start happening over time? |
| 07:44.49 | jkroon | uptime on that asterisk isn't even 24 hours yet. |
| 07:47.15 | jkroon | it went away now, without restarting |
| 07:47.57 | jkroon | so probably just a handful of calls experiencing this, but at 5000+ calls/day even a small fraction is a pain |
| 07:49.22 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:49.32 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:49.38 | ChannelZ | hmm. Running out of file descriptors? I have no idea. (does that still happen?) |
| 07:51.04 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 07:54.50 | jkroon | i hope not ... |
| 07:55.18 | jkroon | i fixed all the fd leaks that I could find and still have a script that runs hourly checking for excessive fd usage and it hasn't triggered yet. |
| 07:56.17 | jkroon | http://pastebin.com/ALUcuUpp |
| 07:56.37 | jkroon | small histogram showing frequency of the log lines per second. never seems to last for longer than a few seconds at a time. |
| 07:57.20 | *** join/#asterisk mtnbkr (~mtnbkr@75-150-91-17-NewEngland.hfc.comcastbusiness.net) |
| 08:25.01 | *** join/#asterisk petern_ (~petern@lachesis.fuzzle.org) |
| 08:29.18 | petern_ | hi |
| 08:29.25 | petern_ | i'm sure this should be easy to find, but i can't :( |
| 08:29.39 | petern_ | uk isdn, chan_dahdi, missing 0/00 prefix |
| 08:31.44 | WIMPy | set the *prefix in chan_dahdi.conf. |
| 08:31.55 | petern_ | they're set |
| 08:33.17 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:37.16 | petern_ | ah ha |
| 08:37.28 | petern_ | fixed it. it was the order lines in the configuration file |
| 08:38.27 | petern_ | thanks for the help. i think :) |
| 08:40.01 | WIMPy | Yes, that file being reversed to the others can be quite confusing. |
| 09:00.48 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 09:06.52 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:4545:a030:d005:4b96) |
| 09:18.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:21.36 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:21.45 | petern_ | had just upgraded the machine from 1.4 to 1.6, it was using chan_lcr before that which had its own hack for it |
| 09:23.09 | WIMPy | That aren't hacks but just the way things work. |
| 09:23.23 | WIMPy | But I wouldn;t recommand anything <1.8 for ISDN connections. |
| 09:23.25 | petern_ | well true |
| 09:23.37 | petern_ | heh |
| 09:23.44 | petern_ | running debian ;) |
| 09:24.45 | WIMPy | You can still build your own. I guess you had to do that with lcr anyway? |
| 09:24.55 | petern_ | only the module |
| 09:25.45 | petern_ | i prefer to keep packages as it's a pain to maintain otherwise |
| 09:26.25 | WIMPy | So you got both Asterisk and lcr as package, but had to build the module to link them yourself? |
| 09:27.11 | petern_ | no :) |
| 09:28.33 | WIMPy | Then I didn't understand. |
| 09:35.50 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:47.09 | jkroon | ChannelZ, eventually asterisk seemingly restarted by itself (didn't crash). logs: http://pastebin.com/LUi8PL1h ... also indicate that there is too many fd's being used. |
| 09:47.21 | jkroon | ulimit -n 4096 ... but it still seems to bomb out at around 1024 |
| 09:47.44 | jkroon | is there any way to from /proc/$(pidof asterisk) figure out what the ulimit values are set to? |
| 09:48.38 | jkroon | ah nm, we had a dude that managed to get a call to loop onto a bunch of Local/ channels, that would definitely explain that one. |
| 09:55.27 | *** join/#asterisk din3sh (~din3sh@41.212.202.60) |
| 09:57.54 | din3sh | my CEL table via odbc is somehow not logging the calls but "cel show status" :/ |
| 09:58.07 | din3sh | shows backed ok |
| 09:58.13 | din3sh | backend* |
| 10:08.42 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 10:21.39 | jkroon | ok, there is definitely something closing file descriptors that it shouldn't be closing on channel creation failure... |
| 10:21.45 | jkroon | will see if I can reproduce offline |
| 10:36.53 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 10:45.14 | din3sh | any idea why am having this trying to set up CEL table |
| 10:45.20 | din3sh | WARNING[2693]: cel_odbc.c:701 odbc_log: Column type -9 (field 'astrealtime:cel:peer') is unsupported at this time. |
| 10:49.57 | Faustov | ~book |
| 10:49.58 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:55.51 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 11:04.56 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 11:07.31 | *** join/#asterisk din3sh (~din3sh@41.212.202.185) |
| 11:07.38 | [ProB]CrazyMan | WIMPy: hi, I have another problem, from some callers the last digit is missing is it possible that I have to wait 1 second until I receive the total destination? but I get it wrong from LCR so could I set an delay in LCR ? |
| 11:10.03 | WIMPy | It sould end when the dialplan has a unique match. |
| 11:12.01 | jkroon | close(16) = -1 EBADF (Bad file descriptor) |
| 11:12.22 | jkroon | ok, given that I can get that from strace - how can I figure out WHERE that close() function is called from? |
| 11:28.09 | [ProB]CrazyMan | ok i will have an look into the dialplan |
| 11:37.00 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 11:37.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:38.46 | WIMPy | Fishy. Looks like my IAD maintains a cache of valid numbers. |
| 11:46.34 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:47.26 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 11:56.54 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-11-215.bois.qwest.net) |
| 11:58.19 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 11:58.23 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 12:00.29 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 12:06.39 | din3sh | WARNING[2693]: cel_odbc.c:701 odbc_log: Column type -9 (field 'astrealtime:cel:peer') is unsupported at this time. |
| 12:06.53 | din3sh | having this error with CEL logging , any idea why? |
| 12:14.01 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 12:14.46 | leifmadsen | din3sh: looks like a column is defined as a type that isn't supported |
| 12:15.04 | leifmadsen | which is pretty much what the warning says |
| 12:25.37 | din3sh | there's this warning for nealry all the columns |
| 12:25.38 | din3sh | :/ |
| 12:28.06 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 12:28.21 | *** join/#asterisk aruntomar (~arun@49.248.157.214) |
| 12:31.20 | din3sh | where can i find the proper table structure for mysql-odbc CEL backend? |
| 12:36.29 | *** part/#asterisk Bullmoose (~Bullmoose@71-33-11-215.bois.qwest.net) |
| 12:41.42 | jkroon | where do I report coding errors in codec_g729a.so (the digium proprietary one?) |
| 12:44.06 | jkroon | nm, digium.com ... support |
| 12:47.13 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 12:47.40 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:49.05 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:51.07 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:53.53 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:54.30 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:55.17 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 12:56.07 | leifmadsen | din3sh: while the mysql schema isn't provided, schemas for other DBs are provided here, which you can then easily create a schema for in mysql by defining it yourself: https://wiki.asterisk.org/wiki/display/AST/CEL+Storage+Backends |
| 12:57.43 | *** join/#asterisk ccesario_ (~ccesario@187.17.166.162) |
| 12:59.18 | WIMPy | Fishy². That cache survives a reboot. |
| 13:00.39 | din3sh | identical schema works on centos 5.7&ast1.8.9.3 |
| 13:00.56 | din3sh | but not with centos 6.2 & ast 1.8.11 |
| 13:01.13 | din3sh | scratching his head |
| 13:04.11 | jww_ | hello everyone. |
| 13:11.31 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:16.17 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-adgfdywadygzuryu) |
| 13:16.17 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:16.27 | *** join/#asterisk serafie (~erin@nat/digium/x-cxgfxffafmcadcup) |
| 13:21.27 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:26.32 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 13:38.13 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 14:01.49 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:07.30 | *** join/#asterisk blizzow (~jburns@67.50.165.60) |
| 14:21.58 | *** join/#asterisk albertoandrade (~albertoan@187.59.39.238) |
| 14:25.00 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:25.48 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:25.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:29.57 | *** join/#asterisk ccesario_ (~ccesario@189.29.37.189) |
| 14:31.31 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 14:38.47 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 14:42.41 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.13) |
| 14:47.42 | pabelanger | ~sipnat |
| 14:47.43 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 14:49.09 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:51.48 | [TK]D-Fender | I really should update that... |
| 14:51.51 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 14:55.32 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 14:58.11 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 14:59.34 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 15:04.37 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 15:06.03 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:12.12 | PBXman | is there a function to save a trace from extensions.conf? |
| 15:12.31 | pabelanger | trace? |
| 15:13.08 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:13.10 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 15:14.31 | PBXman | i'd like to save the whole siptrace to a log file but just at one specific point, any ideas? |
| 15:15.01 | [TK]D-Fender | PBXman, There is none |
| 15:15.02 | pabelanger | doesn't work like that |
| 15:17.08 | PBXman | what if I created a java agi and I called it from extensions.conf? could it work? could I retreive all sip headers? |
| 15:19.47 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:21.20 | pabelanger | PBXman: what are you trying to do |
| 15:21.31 | PBXman | if anybody is familiar with FS, i'm looking for something to this with Asterisk -> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info |
| 15:24.38 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 15:25.00 | PBXman | pabelanger: i just want to get all headers at one specific point in the dialplan |
| 15:25.13 | darkbasic | why don't "pri show span X" and "pri intense debug span X" work with asterisk 1.8? what should I type instead? |
| 15:25.14 | [TK]D-Fender | <PBXman> what if I created a java agi and I called it from extensions.conf? could it work? could I retreive all sip headers? <- You can only pull ones you know the names of to ask for. There is no "poll" |
| 15:25.27 | McBoingBo | I hear no ringing when I forward my phone to a cell, nature of the beast or is there something I need to tweak? |
| 15:25.52 | [TK]D-Fender | McBoingBo, Depends. Go prove that your call out is passing back progress in the first place |
| 15:26.13 | leifmadsen | it happens with mobile devices for some reason sometimes |
| 15:26.17 | PBXman | OK thank for your help |
| 15:26.41 | pabelanger | PBXman: core show application DumpChan |
| 15:26.46 | pabelanger | not sure what SIP headers will show up |
| 15:27.02 | [TK]D-Fender | pabelanger, Won't dump the SIP headers. You need to hit the function for that, and it's only by name... |
| 15:27.53 | [TK]D-Fender | pabelanger, Would be nice if there was a way to load up an array, etc |
| 15:28.36 | McBoingBo | [TK]D-Fender, this will be evident in the console? Works for local clients, but those who are remote in another state/province there is no ringing |
| 15:28.51 | [TK]D-Fender | mcIt'll be evident when you look at the actual call. |
| 15:28.56 | [TK]D-Fender | McBoingBo, It'll be evident when you look at the actual call. |
| 15:32.30 | *** join/#asterisk din3sh (din3sh@41.136.241.137) |
| 15:33.34 | *** join/#asterisk rampage73 (~rampage73@184.2.146.144) |
| 15:37.50 | McBoingBo | [TK]D-Fender, http://pastebin.com/5FYNxtyU |
| 15:37.54 | *** join/#asterisk adnc (~akif@unaffiliated/adnc) |
| 15:42.47 | [TK]D-Fender | McBoingBo, Is that really "looking at the call"? |
| 15:43.11 | McBoingBo | apparently not, I am going away now |
| 15:43.32 | [TK]D-Fender | McBoingBo, If you're placing a SIP call then that would imply needing SIP debug enabled. |
| 15:44.48 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 15:50.30 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 15:52.38 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 15:53.08 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:00.20 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:07.09 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:14.59 | *** join/#asterisk kessius (~cassio@189.123.213.38) |
| 16:18.03 | *** join/#asterisk din3sh (din3sh@41.136.247.71) |
| 16:29.24 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 16:29.24 | *** part/#asterisk rampage73 (~rampage73@184.2.146.144) |
| 16:32.08 | paolosupino | hi |
| 16:32.44 | paolosupino | I have the following sip.conf: http://pastebin.com/fS7SnXNk and I want to turn it into a decent one. Can anyone give pointers how to simplify it? |
| 16:33.40 | paolosupino | I already have that configuration in a separate file and include it in my sip.conf with #include and I already moved all the shared stuff into a template. Can anything else be done to simplify it? |
| 16:35.47 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 16:38.38 | leifmadsen | paolosupino: looks fine to me, but you could always use a template to limit the amount of duplication |
| 16:40.55 | [ProB]CrazyMan | WIMPy: i have the problem not with all callers, just a view that the last digit is missing. so i dont think its a problem from the dialplan |
| 16:41.09 | paolosupino | The paste bin is a copy from the SIP provider website… in mine I moved all the shared stuff into a template. mine is: http://pastebin.com/UmdBskFe |
| 16:41.27 | paolosupino | obviously the context has been changed in order to protect the innocent. |
| 16:41.52 | WIMPy | [ProB]CrazyMan: Any more information? |
| 16:42.09 | WIMPy | Or maybe it's the callers, just like I experienced? |
| 16:42.26 | [ProB]CrazyMan | I know there was an problem with gettig the callerid, so you have to wait one second ... could it be that there is an problem with LCR and callingpres |
| 16:42.50 | WIMPy | You never have to wait for anything. |
| 16:43.21 | [ProB]CrazyMan | no its not the caller ... I tested it with my girfriend ... and she was realy upsed when i told her she does not dial the complete number ... |
| 16:43.38 | WIMPy | Now is it the called or the calling party? |
| 16:44.42 | [ProB]CrazyMan | external person is calling to 72230 --> LCR catches only 7223 --> transmit to Asterisk ... and there is no extension for that |
| 16:44.43 | WIMPy | I tried earlier and after calling one number, I was unable to call any longer ones. That's obviousely a bug with Alice. |
| 16:45.40 | WIMPy | Show me the output of misdn_log. |
| 16:45.45 | [ProB]CrazyMan | no it doesnt depend on the lenght of the number .. e.g. 722328 gets to 72232 one digit more |
| 16:49.17 | [ProB]CrazyMan | http://pastebin.com/XurvCUS7 |
| 16:49.46 | [ProB]CrazyMan | the number should be 722328 |
| 16:50.19 | [ProB]CrazyMan | and the 8 is coming later as DTMF |
| 16:50.38 | WIMPy | Huh? |
| 16:51.16 | [ProB]CrazyMan | in line 16 |
| 16:51.48 | WIMPy | Doesn't look bad to me. |
| 16:52.16 | WIMPy | The interesting part is why it disconects after receiving the last digit. |
| 16:52.48 | [ProB]CrazyMan | because in asterisk I get only 72232 and asterisk does not have any extension with this number |
| 16:53.11 | WIMPy | Oh, wait... |
| 16:54.04 | WIMPy | No. I was just thinking that you might need a WaitExten, but you already have a partial extension. |
| 16:54.25 | [ProB]CrazyMan | ?? |
| 16:54.34 | [ProB]CrazyMan | in LCR ? |
| 16:54.45 | WIMPy | No in the dialplan. |
| 16:55.03 | [ProB]CrazyMan | in the asterisk dialplan i have wait(1) |
| 16:55.28 | WIMPy | Why do you want to wait? |
| 16:57.16 | [ProB]CrazyMan | it was somethimes a problem that not all calerid informations got at time whit bristuff so it was adviced to wait 1 second ... |
| 16:57.31 | [ProB]CrazyMan | but this is still 5 years ago ... maybe things changed ... |
| 16:58.09 | WIMPy | That doesn't make sense. The caller ID comes immediately. |
| 16:58.28 | WIMPy | On analog you may have to wait. |
| 16:59.49 | [ProB]CrazyMan | so what I know to this problem now ... i dont have the problem with my mobile phone .. i know that my girfriend has an isdn port but the phone itself is analog |
| 17:00.28 | WIMPy | Yes, but the information looks correctely recognized. |
| 17:00.39 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:01.13 | [ProB]CrazyMan | it comes as this in: Executing [72232@mISDN:1] Wait("lcr/202", "1") in new stack |
| 17:02.12 | WIMPy | Wait |
| 17:02.16 | WIMPy | oops |
| 17:02.24 | WIMPy | Wait? So that extension does exist? |
| 17:02.43 | [ProB]CrazyMan | yes |
| 17:03.26 | WIMPy | Hmm. Overlapping extension are usually not a good idea. But for me they actuially worked. |
| 17:03.55 | WIMPy | Ha, I see the issue. |
| 17:04.26 | WIMPy | Overlapping extensions only work in WaitExten, but not when directly hitting the dialplan. |
| 17:05.12 | WIMPy | That is: It there's a match, it won't wait any further. |
| 17:06.28 | WIMPy | I guess it's something that could be changed if neccessary. |
| 17:06.30 | [ProB]CrazyMan | hm |
| 17:08.04 | [ProB]CrazyMan | ok ... do i get the rest numbers transfered to asterisk extensions ? so does the extension change when a digit is comig, or do I have to detect this with dtmf ? |
| 17:08.48 | WIMPy | They will end up in the extension. |
| 17:09.09 | WIMPy | But only until an extension is found. |
| 17:09.39 | [ProB]CrazyMan | hm strange ok so I have to change the dialplan |
| 17:10.43 | WIMPy | I guess it might make sense to to check for possible further matches to support such configurations. |
| 17:11.07 | WIMPy | I think it was Switzerland where such configurations are usual. |
| 17:12.57 | WIMPy | Hmm. While on the same topic: Is it possible somehow to forward overlap digits between two Asterisk boxes? |
| 17:13.38 | WIMPy | I know chan_sip can't, but I had hoped chan_iax2 could. But it doesn;t seem to do so. |
| 17:17.03 | [ProB]CrazyMan | puh no idear, normaly you dont have overlap digits between to asterisk boxes does it not collect and then send ? |
| 17:17.12 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:17.31 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:18.17 | WIMPy | Yes, but I tried to dial out on a line connected to another box. |
| 17:18.55 | WIMPy | That would be handy to test such cases as te one you're on to. |
| 17:19.12 | WIMPy | And off course it would be much more user friendly. |
| 17:19.38 | [ProB]CrazyMan | why user friendly? |
| 17:19.57 | WIMPy | Because the user doesn't have to wait. |
| 17:20.05 | [ProB]CrazyMan | for what ? |
| 17:20.21 | WIMPy | For the call to get sent. |
| 17:20.44 | [ProB]CrazyMan | how long does this take ? 1 second ? |
| 17:20.57 | [ProB]CrazyMan | i dont think he will safe a lot time |
| 17:21.34 | [ProB]CrazyMan | but maybe its better for spit ? :/ |
| 17:21.41 | WIMPy | Not sure what the default is. IIRC 4s. |
| 17:22.05 | WIMPy | That will come with complete numbers, so no issue there. |
| 17:23.05 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:25.43 | jaytee | anyone here familiar with setting up a Cisco ASA 5505 with an Asterisk server on the inside network using a "SIP Trunk" (Flowroute account) or other ITSP? I've found several articles googling what I should need but I'm missing something vital and I'm stuck at this point. |
| 17:29.34 | *** join/#asterisk cbdev (~cbdev@2a01:4f8:121:4083:1333:3333:3333:3337) |
| 17:36.34 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:42.00 | *** part/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 17:43.21 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 17:43.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:45.57 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 17:46.34 | *** part/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 17:49.52 | *** join/#asterisk psilvao (~psilvao@ip-1-228-107-190.nextelmovil.cl) |
| 18:01.43 | *** join/#asterisk albertoandrade (~albertoan@187.112.128.103) |
| 18:09.48 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 18:13.06 | *** join/#asterisk albertoandrade (~albertoan@186.212.193.138) |
| 18:14.16 | *** join/#asterisk wonderworld (~ww@dsdf-4db51272.pool.mediaWays.net) |
| 18:20.54 | *** join/#asterisk timahvo1 (~rogue@41.81.23.104) |
| 18:24.16 | *** join/#asterisk albertoandrade (~albertoan@201.22.24.87.dynamic.adsl.gvt.net.br) |
| 18:25.41 | *** join/#asterisk vfabi (~fabi@178.76.80.139) |
| 18:28.17 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:34.17 | *** join/#asterisk albertoandrade (~albertoan@187.112.143.66) |
| 18:39.45 | *** join/#asterisk fireman_biff (~biff@65.48.222.203) |
| 18:45.49 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:46.28 | *** join/#asterisk oej_ (~olle@95.209.35.151.bredband.tre.se) |
| 18:50.09 | fireman_biff | Hi, I have 3 PBXs running Asterisk 1.6, lets call them A, B & C. Each one is a DUNDi peer with the other two to allow direct dialing of extensions. If the link between A and C drops, I want A to look up extensions on C via B, but that doesn't work. When A sends out the lookup request to B, it includes the EID for C (even though it sees C as unreachable and can't query it) so B never queries C. Is there a way to fix this? |
| 18:57.52 | Qwell | leifmadsen: ?! |
| 18:58.04 | leifmadsen | Qwell: ohai |
| 18:58.22 | Qwell | leifmadsen: you're probably one of a few people that could answer the above :p |
| 18:58.27 | leifmadsen | oic |
| 18:59.22 | *** join/#asterisk CoasterMaster (~xbmc@50-78-162-17-static.hfc.comcastbusiness.net) |
| 18:59.31 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 19:00.11 | leifmadsen | fireman_biff: well really it should be sending the request to both |
| 19:00.19 | leifmadsen | I'd have to see your configuration |
| 19:00.46 | fireman_biff | leifmadsen: it should send the request to both, but in this case one of the peers is marked as unreachable |
| 19:01.01 | leifmadsen | if you have full mesh, then when A sends a request, it should go to both B and C |
| 19:01.07 | leifmadsen | ok so... ? |
| 19:01.17 | fireman_biff | A shouldn't tell B not to query C if A sees C as unreachable, should it? |
| 19:01.25 | leifmadsen | they should be separate indepedent requests |
| 19:01.50 | leifmadsen | without seeing a configuration or debugging output it is impossible to say what it should do |
| 19:02.08 | fireman_biff | ok, but if A sees C as unreachable, will A query C anyway? |
| 19:02.15 | leifmadsen | yes |
| 19:02.21 | leifmadsen | well, that is unknown |
| 19:02.25 | leifmadsen | but it doesn't matter if it does |
| 19:02.25 | *** part/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 19:02.29 | leifmadsen | because it queried B |
| 19:02.40 | leifmadsen | and if B has a connection to C, it should ask C |
| 19:03.15 | leifmadsen | if it is unreachable, and A knows it can't reach it, it should really not attempt to ask it for info |
| 19:03.15 | fireman_biff | but A it telling B that it already queried C, or at least that's what I'm understanding from the debug output, I'll post it for you |
| 19:03.28 | leifmadsen | possible you found an implementation bug |
| 19:03.42 | leifmadsen | I'd suggest attempting to reproduce with 1.8 |
| 19:05.55 | *** join/#asterisk psilvao (~psilvao@ip-12-228-107-190.nextelmovil.cl) |
| 19:11.13 | fireman_biff | leifmadsen: These are the two parts of the debug output I'm looking at: http://pastebin.com/j9cxaCgw |
| 19:11.43 | fireman_biff | the EID ending with 35 is the PBX making the request, EIDs ending with 44, 34, and 5a were all unreachable from 35 at the time. This output is from another PBX, EID ending in 36 which was peered with EID 5a and could have done the lookup if it wasn't told not to. |
| 19:11.55 | fireman_biff | Am I at least understanding the output correctly? |
| 19:11.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:12.12 | leifmadsen | sorry, I can't spend a lot of time looking, but I'd prefer to see the configs to see if there is something obvious |
| 19:12.28 | leifmadsen | like I said, possible you found an implementation bug, or just a configuration issue |
| 19:13.10 | fireman_biff | alright, whats the best way to show all the config files? can i email them? |
| 19:13.24 | *** join/#asterisk macroevolve (~macroevol@c-98-234-125-202.hsd1.ca.comcast.net) |
| 19:13.36 | leifmadsen | pastebin with passwords masked |
| 19:13.42 | macroevolve | hi guys - was having a hard time finding a good patent attorney who focuses on telecom - does anyone happen to have any recommendations? |
| 19:13.45 | fireman_biff | k |
| 19:13.45 | leifmadsen | it's possible someone else can look at the configs and not just me |
| 19:14.53 | _Corey_ | macroevolve: You're not trying to patent "visual voicemail" are you...? ;) |
| 19:15.37 | macroevolve | no.. i was afraid my quesitons wouldn't bode too well in an irc room =) especially with how much people "love" patents |
| 19:16.06 | macroevolve | i'm looking for an uncle of mine who has been trying to get his patent through.. unfortuantely i was left with the responsibility with finding a good patent attorney (he is in China) |
| 19:16.15 | chuckf | macroevolve: well if you want to tell me your idea if its worth it I'll find a good patent attorney:) |
| 19:16.27 | fireman_biff | leifmadsen: is it ok if I PM you the pastebin links? |
| 19:16.49 | macroevolve | =) |
| 19:16.51 | leifmadsen | you can but I might not necessary look at them |
| 19:16.56 | leifmadsen | busy doing some other work right now |
| 19:17.31 | _Corey_ | macroevolve: Well, I haven't worked with this guy but I had a chat with him at one of the trade shows (ITEXPO I think): http://www.telecomlawyer.net/ |
| 19:17.46 | Qwell | yay for up and coming patent trolls! |
| 19:17.54 | Qwell | excuse me while I ignore you. |
| 19:18.01 | macroevolve | whoa his URL gives a pretty good indication of what he does |
| 19:18.05 | leifmadsen | Qwell: CLOUD LAWYER! |
| 19:18.06 | macroevolve | =) - thx |
| 19:18.30 | _Corey_ | My conversations with the guy had to do with CLEC issues (taxes, contracts, etc.) so I don't even know if he does patents |
| 19:18.41 | macroevolve | ok cool thanks! |
| 19:19.16 | _Corey_ | *Just for the love of god don't try to patent something everyone's already doing* |
| 19:20.08 | macroevolve | yeah i know.. i don't like patents either.. sounds weird coming from someone asking for a patent lawyer =) |
| 19:20.44 | _Corey_ | I have no problem with patents in theory... it's just these idiots who patent obvious stuff that everyone discovers independently anyway |
| 19:20.44 | macroevolve | i checked out that guys' website and will try and connect - thx, corey |
| 19:21.10 | macroevolve | yeah makes sense - i understand how that feels. |
| 19:21.43 | _Corey_ | good luck |
| 19:22.22 | macroevolve | thx! |
| 19:24.52 | _Corey_ | I have a feeling I'm going to come to regret doing that... |
| 19:36.18 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-253-067.sc.res.rr.com) |
| 19:51.46 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 19:54.46 | *** join/#asterisk RedShift (~RedShift@178-116-215-149.access.telenet.be) |
| 19:54.51 | RedShift | hi all |
| 19:55.17 | RedShift | why is asterisk sending a RST (not accepting a registration) when the phone is in another network? |
| 19:55.41 | *** join/#asterisk afink (~afink@mail.abcsignanddisplay.com) |
| 19:55.43 | RedShift | it's not denying it or anything via the SIP protocol, with a packet trace I can see it just sends a RST |
| 19:55.53 | [TK]D-Fender | RedShift, sHOW US YOUR PEER AND THE REG ATTEMPT MASKING ONLY THE SECRET |
| 19:55.54 | [TK]D-Fender | ~PB |
| 19:55.54 | 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.us , 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. |
| 19:55.56 | [TK]D-Fender | ^^^ |
| 19:55.57 | RedShift | (I'm using SIP over TCP, which means it's just terminating the connection) |
| 19:57.03 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:57.19 | RedShift | http://pastebin.com/VhCHkEX5 |
| 19:57.22 | RedShift | that's my sip.conf |
| 19:58.00 | RedShift | if you mean by reg attempt, you mean what's in the logs, well... asterisk logging nothing -_- |
| 19:58.07 | [TK]D-Fender | * CLI w/ SIP DEBUG enabled |
| 19:58.12 | [TK]D-Fender | ^ |
| 19:59.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:59.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:00.07 | RedShift | enabled debug & verbose in asterisk.conf, did sip reload, phone is restarting |
| 20:03.43 | [TK]D-Fender | SIP DEBUG |
| 20:03.52 | [TK]D-Fender | sip set debug on <-------- |
| 20:04.06 | RedShift | it's on |
| 20:04.11 | RedShift | not getting anything though |
| 20:04.41 | [TK]D-Fender | go verify that * has atually bound the port that the port is open, if anything is arriving on the interface at all, etc |
| 20:05.33 | RedShift | yes, the port is definitely open, stuff's arriving on the interface |
| 20:05.53 | RedShift | netstat says it's listening, there are no iptables rules, packets are being sent & received |
| 20:06.10 | RedShift | it's just that after the phone sends REGISTER, asterisk just sends a TCP RST back |
| 20:06.50 | [TK]D-Fender | RedShift, try transport=tcp for your peer |
| 20:07.15 | [TK]D-Fender | RedShift, Also , what ver of * is that running? |
| 20:08.00 | afink | hello everyone, I seem to be having a brain malfunction, I am having trouble configuring pri channels with a digium TE122p. I have edited, /etc/system/dahdi.conf and /etc/asterisk/chan_dahdi.cfg is there another one I am missing? |
| 20:08.39 | Qwell | .cfg? |
| 20:08.55 | [TK]D-Fender | afink, /etc/dahdi/system.conf |
| 20:09.03 | afink | oh kind of backwards on the first one /etc/dahdi/system.conf |
| 20:09.17 | [TK]D-Fender | afink, Those should do it |
| 20:13.37 | *** join/#asterisk albertoandrade (~albertoan@187.112.137.212) |
| 20:13.38 | RedShift | asterisk 1.8.4.4, 'sip show peer 200' says Allowd.Trsp: TCP,UDP; Prim.Transp.: TCP |
| 20:16.18 | Qwell | .4? |
| 20:16.48 | Qwell | only 11 months behind.. |
| 20:18.10 | RedShift | try something else? |
| 20:18.19 | Qwell | always try upgrading first |
| 20:29.28 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 20:30.25 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 20:31.18 | RedShift | it was some weird problem with that asterisk version |
| 20:31.33 | RedShift | dammit, lost two hours on this |
| 20:31.41 | Qwell | and this is why we always upgrade first |
| 20:33.11 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
| 20:35.06 | RedShift | oh well |
| 20:35.08 | RedShift | been through worse |
| 20:36.01 | *** join/#asterisk ircWALK (~FACEBOOK@unaffiliated/ircwalk) |
| 20:36.09 | *** join/#asterisk albertoandrade (~albertoan@186.212.34.214) |
| 20:36.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:36.42 | ircWALK | hi. |
| 20:36.44 | ircWALK | i need a relatively small device that can connect to a phone socket. it would also need a way of another telephone connecting to it. so people can pick up the phone, and use it regularly, even though the device is connected to the phone socket. it needs to be able to ring phone lines on time delay. before it attempts to make a phone call, it needs to check if somebody is on the phone. if |
| 20:36.45 | ircWALK | somebody is using the phone, it doesn't attempt to dial any phone. i also need it to have the ability to receive some sort of commands via the telephone. so if somebody rings the phone number and dials 91882, it allows the person to change the time at which phone calls are made, along with being able to change the phone number it is dialing. |
| 20:37.52 | Qwell | ircWALK: sounds like Asterisk to me |
| 20:38.28 | ircWALK | how much would it cost to make a gadget like that? |
| 20:39.03 | RedShift | what do you mean with "change the time at which phone calls are made"? |
| 20:39.07 | RedShift | you mean... alter reality? |
| 20:39.11 | ircWALK | lol |
| 20:40.17 | jaytee | what do you mean by "it needs to be able to ring phone lines on time delay"? |
| 20:40.17 | ircWALK | well lets say normally it makes a phone call at 9am every day. if the administrator dials the phone number where it is hosted, he can type a password like 91991. he can then change the scheduling so it dials numbers at 12pm instead |
| 20:40.40 | ircWALK | you connect the device to a phone line, and every 12hours or so it rings a phone number. without anybody having to do an ything |
| 20:40.59 | *** join/#asterisk timahvo1 (~rogue@41.90.126.2) |
| 20:41.01 | jaytee | aha! an autodialer with an custom scheduling function |
| 20:41.10 | ircWALK | yea |
| 20:41.49 | RedShift | oh, too bad, because bending space/time would have been much cooler |
| 20:42.32 | ircWALK | theres no "Q" on freenode |
| 20:42.50 | ircWALK | godlike alien from star trek |
| 20:43.09 | jaytee | yeah, asterisk with some a analog card with one fxo port and one fxs port should do the job plus whatever time it takes you to program the dialplan. |
| 20:43.29 | jaytee | s/some a/an |
| 20:43.33 | ircWALK | relatively simple project? |
| 20:44.18 | ircWALK | is programming the dialplan relatively easy? id be getting a freelancer for that |
| 20:45.13 | jaytee | if they understand Asterisk well then it would be relatively easy. |
| 20:46.33 | RedShift | sleepy time |
| 20:46.36 | RedShift | thanks for the help |
| 20:46.36 | RedShift | bye |
| 20:47.48 | ircWALK | how much do you think a freelancer would charge for making the dialplanner? |
| 20:48.02 | *** part/#asterisk RedShift (~RedShift@178-116-215-149.access.telenet.be) |
| 20:48.21 | ircWALK | and once that is done. how much would it cost to make this into an actual physical gadget? as small as possible |
| 20:48.45 | *** join/#asterisk chasing`Sol (~cS@197.134.15.161) |
| 20:49.01 | jaytee | rates vary. try looking or posting on www.asterisk.org. the site has a section for jobs. etc. where you can find Asterisk programmers who'll take contract gigs. |
| 20:49.27 | jaytee | small as possible? for mass production? |
| 20:49.30 | _Corey_ | ircWALK: I think the dial plan programming part will be minimal relative to the cost of manufacturing the physical gadget |
| 20:49.48 | jaytee | _Corey_, I agree. |
| 20:49.57 | ircWALK | yea of course |
| 20:50.06 | ircWALK | well, id want to make 10 or 20 at first |
| 20:50.35 | jaytee | how many calls would this device need to make concurrently? |
| 20:51.00 | ircWALK | just one |
| 20:51.07 | ircWALK | it will just connect to one phone line |
| 20:51.09 | jaytee | that helps |
| 20:51.31 | cbdev | could use a raspberry pi for the physical device |
| 20:51.38 | cbdev | though that would be massivley overpowered |
| 20:52.06 | ircWALK | that reminds me, what about the power supply? |
| 20:52.25 | ircWALK | connecting it to a plugpoint wouldn't be possible |
| 20:52.32 | ircWALK | it would need to run on batteries |
| 20:52.51 | jaytee | fuhgedaboudit! |
| 20:53.10 | ircWALK | argh |
| 20:53.10 | ircWALK | ok |
| 20:54.33 | mick_laptop | anyone know how I can change the music on hold? |
| 20:54.37 | ircWALK | why are batteries out of the question? |
| 20:54.56 | ircWALK | the batteries on my telephones in my house have lasted for /years/ |
| 20:55.20 | Qwell | ircWALK: Your phone isn't a computer. |
| 20:55.25 | pabelanger | http://en.wikipedia.org/wiki/SheevaPlug |
| 20:55.27 | mick_laptop | I swapped the file (same name and everything) for another wav, then I restarted asterisk - just thought that this might work, but it plays the default one instead :-/ |
| 20:55.46 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:55.49 | mick_laptop | Qwell: depends on the phone :P |
| 20:56.16 | mick_laptop | I consider smart phones computers |
| 20:56.19 | jaytee | even the a pico-ITX board with a custom fxo/fxs module would still consume at least 50w minimum and if it's running all the time you'd need a fairly good size battery and you'd still need to change or recharge it. |
| 20:56.38 | mick_laptop | If I can install linux on it, by golly it is a computer |
| 20:56.49 | Qwell | mick_laptop: So your cellphone battery lasts for years then? |
| 20:57.03 | mick_laptop | when it is plugged in, probably :P |
| 20:57.41 | ircWALK | but the phones i have in my house do so much complicated stuff. address book etc. the gadget i want to make seems really simple in comparision |
| 20:57.44 | mick_laptop | my HP batteries didn't last a week (on either HP laptop, I had 2) |
| 20:57.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:58.10 | mick_laptop | ircWALK: I came in late, what are you doing? |
| 20:58.16 | jaytee | my Kindle battery lasts weeks without a recharge if I'm not using the wireless to download books. The Kindle Fire isn't near as good at battery life but then it's more of a tablet than my regular Kindle is. |
| 20:59.05 | ircWALK | need a small autodialer that connects to phone line., with some really simple remote administration. a normal telephone needs to be able to connect to it aswell. so its like phone -> my gadget -> phone line. so phone can work, and my gadget can work |
| 20:59.07 | mick_laptop | jaytee: that is because it goes the sleep all the time and you don't notice that because it is an ebook reader |
| 20:59.58 | mick_laptop | why not just have an android app? |
| 21:00.07 | jaytee | mick_laptop, yeah. But he wants 24/7 computer on to do an autodialer function. |
| 21:00.17 | mick_laptop | VPS |
| 21:00.20 | mick_laptop | :P |
| 21:00.37 | ircWALK | well actually, i only need the computer on while its doing the autodialing. and it would only be used for a total of an hour or so per day. |
| 21:00.47 | ircWALK | the rest of the time it can be in some sort of low power standby mode |
| 21:00.49 | _Corey_ | ircWALK: Unless you have a prototype manufacturing budget, I suggest you find a creative engineering grad student... |
| 21:01.20 | _Corey_ | ircWALK: We can direct you to different solutions, but you're looking to build something that doesn't exist |
| 21:01.22 | mick_laptop | _Corey_: what if he is that creative grad student? |
| 21:01.28 | _Corey_ | Then he's SOL |
| 21:01.33 | mick_laptop | hahha |
| 21:01.37 | ircWALK | im not a creative grad student lol |
| 21:01.42 | ircWALK | id like to think im a creative undergrad student tho |
| 21:01.49 | ircWALK | but i do maths, not engeineering |
| 21:02.02 | Qwell | yeah, well, where's your maths now? |
| 21:02.11 | mick_laptop | lol |
| 21:02.45 | ircWALK | well, id like to know cost of product for 1 unit, 20units etc. not including programming costs of the dialplanner |
| 21:02.48 | mick_laptop | ircWALK: can you tell us what you are really doing w/ this? |
| 21:03.09 | mick_laptop | or is that super secret? |
| 21:03.15 | ircWALK | its a mystery project :] dont want to give away huge amounts of details unneccessarily |
| 21:03.35 | mick_laptop | usually the details give away the best answer |
| 21:04.17 | _Corey_ | ircWALK: Just about everything creative involving an analog line has already been done... ;) |
| 21:04.32 | ircWALK | mm. i definatley need this exact gadget made. there isn't really another solution imo |
| 21:04.34 | mick_laptop | since you get out of your line of thinking |
| 21:04.55 | mick_laptop | ircWALK: IYO is where the problems lie |
| 21:05.08 | mick_laptop | most likely |
| 21:05.47 | ircWALK | IYO <-- means...? |
| 21:06.10 | mick_laptop | the opposite of IMO |
| 21:06.23 | ircWALK | oh |
| 21:06.55 | mick_laptop | since you aren't describing the problem, you don't know if something exists that does what you want |
| 21:07.04 | mick_laptop | I learned this lesson the hard way |
| 21:07.11 | ircWALK | well, as much as i appreciate you trying to find another solution for me - its not neccessary atm. this is the gadget that needs to be made |
| 21:07.24 | mick_laptop | so you think |
| 21:07.34 | ircWALK | id just want a rough ballpark on how much it would cost, not including initial programming costs |
| 21:08.17 | ircWALK | ok, maybe it is /helpful/ to have another solution, but im not interested in any other soltuions at the moment. this is the solution i want to work on |
| 21:08.20 | mick_laptop | since the problem isn't defined, noone can answer you |
| 21:08.25 | _Corey_ | ircWALK: You really need manufacturing advice at this point, budget a couple thousand bucks for the Asterisk end of this and you'll probably be safe |
| 21:08.40 | Qwell | then a few tens of thousands (at least) for the hardware design |
| 21:08.48 | ircWALK | whattttttttttt |
| 21:09.11 | mick_laptop | ircWALK: you thought embedded hardware was cheap? |
| 21:09.13 | _Corey_ | um yeah |
| 21:09.43 | ircWALK | i had no idea about costs. i thought it could just be a computer like that sheevaplug, with some custom software |
| 21:09.48 | mick_laptop | again, if your problem was well defined, you probably wouldn't even need any custom hardware |
| 21:09.50 | FLeiXiuS | Whats the recommended way to record a call? |
| 21:10.03 | mick_laptop | FLeiXiuS: a single call? |
| 21:10.04 | Qwell | ircWALK: How could it be a sheevaplug, if your requirement says no power source other than batteries? |
| 21:10.10 | mick_laptop | FLeiXiuS: wireshark |
| 21:10.26 | ircWALK | which sits in standby mode 23hours per day, and switches to the mode neccessary for dialing the other 1hour/day |
| 21:10.32 | FLeiXiuS | mick_laptop, Yeah, no. Within the dial plan. |
| 21:10.49 | ircWALK | well it seems close - just batteries instead. but maybe thats hugely different |
| 21:10.50 | Qwell | ircWALK: are you aware of any off-the-shelf systems like that? |
| 21:10.54 | *** join/#asterisk mackhendricks (~mackhendr@c-69-246-107-12.hsd1.mi.comcast.net) |
| 21:11.02 | Qwell | no? so then you'll have to design one |
| 21:12.54 | mick_laptop | FLeiXiuS: http://www.fosslc.org/drupal/node/643 ? |
| 21:13.25 | FLeiXiuS | Let me be more specific - automatically record the call without DTMF |
| 21:13.43 | ircWALK | hmm. well seems like some people have messed with solarpowered sheevaplug |
| 21:14.12 | Qwell | ircWALK: alright, so use a sheevaplug - now, where are you getting the FXS interface? |
| 21:14.39 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:16.21 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 21:17.18 | ircWALK | i have no idea |
| 21:17.35 | ircWALK | having difficulty figuring out what it is. something that makes it recongised as a telephone? |
| 21:17.39 | Qwell | alright, so then you're planning on selling them, right? better see how much UL charges for certification |
| 21:18.17 | Qwell | oh, sucks, you can't get certification because your enclosure isn't right. have to design new plastics now |
| 21:18.31 | [TK]D-Fender | ircWALK: Device could cost somewhere in the ballpark of $100 on the outside, dev aside. |
| 21:18.37 | mick_laptop | ircWALK: you doing this for a school project or a business idea? |
| 21:18.58 | _Corey_ | That Sheevaplug doesn't seem to grounded... I don't think it's UL listed |
| 21:18.59 | mick_laptop | or just a hobby? |
| 21:19.14 | ircWALK | buisness idea |
| 21:19.21 | Qwell | _Corey_: there's actually lots more needed, since it's connecting to the PSTN now |
| 21:19.33 | Qwell | (hint: it's expensive) |
| 21:20.01 | _Corey_ | Yeah, no doubt... I don't think the thing would pass UL as is w/o the FXS component |
| 21:20.12 | ircWALK | hmm.. i might just scrap this idea for now |
| 21:20.14 | ircWALK | i got another idea though |
| 21:20.21 | ircWALK | seems overly complicated for me atm |
| 21:20.23 | _Corey_ | Qwell: I had my hands on one recently and it felt very cheap |
| 21:20.25 | Qwell | _Corey_: nah I bet it is |
| 21:20.54 | mick_laptop | ircWALK: because you haven't said what it should do (which is understandable because of your intention to market something) |
| 21:21.07 | _Corey_ | You guys have seen this right? http://www.plugpbx.org/ |
| 21:22.37 | ircWALK | anyway regarding the idea i just came up with. i dont know how lame it is |
| 21:23.03 | ircWALK | over here, afaik, and i haven't checked in a while, certain mobile phone contracts will allow you free calls to certain phone numbers you assign |
| 21:23.19 | _Corey_ | ircWALK: it all depends on what the problem is you're trying to solve |
| 21:23.25 | Qwell | it's been done |
| 21:23.44 | ircWALK | lets say that number is 12345. can you put some osftware on phone number 12345 so that the call is routed through it? |
| 21:24.12 | Qwell | yes, and it's been done |
| 21:24.14 | ircWALK | so you ring the number which is free from your mobile, and get the landline to make the calls for you |
| 21:24.16 | ircWALK | oh |
| 21:24.19 | ircWALK | i figured it would be |
| 21:24.34 | ircWALK | whats it called? |
| 21:24.59 | ircWALK | that would be pretty useful for some people i know. landline rates on mobile basically. aslong as you got a number you can ring for free on your mobile |
| 21:25.17 | Qwell | it's not called anything. it's a trivial concept |
| 21:25.37 | ircWALK | i mean the software that does it |
| 21:25.42 | Qwell | Asterisk? |
| 21:25.46 | mick_laptop | asterisk |
| 21:25.49 | ircWALK | lol |
| 21:25.58 | Qwell | it's like 2 lines of dialplan |
| 21:26.42 | ircWALK | has to be some way of promoting this to people? surely alot of people would find it useful? |
| 21:26.56 | ircWALK | they could ring 0870 numbers from their mobile for example for really cheap (landline rates) |
| 21:27.08 | ircWALK | or free, if they can ring their "best friend" for free all the time |
| 21:27.32 | *** join/#asterisk mackhendricks (~mackhendr@c-69-246-107-12.hsd1.mi.comcast.net) |
| 21:28.39 | Qwell | reiserfs: good thing I'm no longer married. I was almost worried for a second there. |
| 21:28.55 | mick_laptop | ? |
| 21:28.55 | cbdev | heh |
| 21:29.11 | [TK]D-Fender | ircWALK: You'd have to have TWO lines. One for the inbound call, one for the outbound |
| 21:29.28 | mick_laptop | not sure if inside joke... or a joke regarding Hans... |
| 21:29.46 | ircWALK | [TK]D-Fender: so are there loads of commercial products which allow you to route calls through your landline then? |
| 21:30.23 | ircWALK | so if you are outside on your mobile, and want to ring an 0870 number (buisness line which costs loads on mobile), you can ring your landline instead (alot cheaper on mobile), and have your landline ring the 0870 number? |
| 21:30.42 | [TK]D-Fender | ircWALK: Isn't this "market research" you're supposed to already have the answer for? |
| 21:30.59 | [TK]D-Fender | ircWALK: Have YOU ever seen one? |
| 21:31.16 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:31.21 | ircWALK | no i haven't seen one |
| 21:31.23 | _Corey_ | ircWALK: This is all basic toll bypass stuff |
| 21:31.25 | Qwell | I suspect that ircWALK has never heard of VoIP... |
| 21:31.36 | [TK]D-Fender | ircWALK: Do you think Joe User has 2 phonelines and just finds these things on shelves. No, Joe User is barely intelligent enough to place a single call on any kind of phone whatsoever |
| 21:31.46 | ircWALK | ive heard of it, but never really considered the uses of it |
| 21:32.08 | Qwell | then every idea you have is already wrong |
| 21:32.25 | _Corey_ | ircWALK: Example: If I'm abroad with a SIM card from some noname cell provider that charges a lot of outbound but offers free inbound, I will hop on a PC and have my asterisk server in the US call me and bridge the call to whomever I'm calling |
| 21:32.50 | ircWALK | joe user has 1 mobile phone, and 1 landline phone. and he often needs to ring 0870 numbers when outside. or people internationally. and only wants to pay landline rates |
| 21:32.55 | _Corey_ | It's like four lines w/PHP |
| 21:33.11 | Qwell | If he wants to pay landline rates, then he's paying about 1000x more than he needs to. |
| 21:33.17 | *** join/#asterisk hfb (~hfb@pool-96-229-38-102.lsanca.dsl-w.verizon.net) |
| 21:33.54 | Qwell | (that 1000x is probably not very far off, either...) |
| 21:35.24 | _Corey_ | ircWALK: Host this thing somewhere and pay a fraction... |
| 21:35.39 | _Corey_ | ircWALK: (Or check out Twilio or Voxeo) |
| 21:35.54 | _Corey_ | this has all been done a thousaand ways |
| 21:37.18 | [TK]D-Fender | ircWALK: takes 2 lines... |
| 21:37.20 | [TK]D-Fender | not 1 |
| 21:38.50 | ircWALK | ive forgotten how its even possible. sorry im new to all this stuff. if i ring my house line, the hosue line cant ring outbound, because its on the phone LOL |
| 21:38.59 | ircWALK | you need 2 landline numbers in the house right? |
| 21:40.47 | [TK]D-Fender | ircWALK: I think you need to start thinking... |
| 21:41.14 | mackhendricks | Hey, I'm looking for some part time Asterisk support engineers in the Eastern and Central Timezone. You have to have a flexible schedule and be able to work Thursday and Fridays from 9am-5pm. Take this short 10 minute quiz. If you pass, I will contact you: http://goflyball.com/wp/?page_id=425 |
| 21:43.40 | Qwell | 10 minutes. pfft. |
| 21:46.38 | Qwell | wonders if he got an A |
| 21:47.40 | ircWALK | is there anything like twilio which allows calls to premium rate numbers? |
| 21:50.06 | mackhendricks | You got an A |
| 21:50.45 | mackhendricks | Lot of people don't get the free form question, but you got it |
| 21:50.46 | Qwell | mackhendricks: but really though, you might have more luck on the asterisk-biz mailing list |
| 21:51.25 | Qwell | hey that sounds like extra credit to me then :p |
| 21:53.11 | [TK]D-Fender | Join Our Team! Fatal error: Maximum execution time of 30 seconds exceeded in D:\Hosting\2927455\html\wp\wp-content\plugins\wp-survey-and-quiz-tool\lib\Wpsqt\Tokens.php on line 121 |
| 21:53.21 | [TK]D-Fender | mackhendricks: Is fixing your webserver part of the test? ;) |
| 21:53.30 | Qwell | d'oh |
| 21:54.10 | mackhendricks | :-) |
| 21:54.24 | *** join/#asterisk timahvo1 (~rogue@197.177.112.130) |
| 21:54.28 | [TK]D-Fender | Or is pointing out that hosting on Windows is "less than brilliant a bonus question? |
| 21:54.50 | mackhendricks | darn...wordpress |
| 21:54.53 | mackhendricks | and windows |
| 21:55.58 | [TK]D-Fender | UR SOLUSHUN AM SUK |
| 21:56.29 | *** join/#asterisk Bullmoose (~Bullmoose@12.50.16.66) |
| 21:59.10 | [TK]D-Fender | Hardly a difficult test, and harly worth 3 minutes |
| 21:59.35 | [TK]D-Fender | Which is proportionate to the attention paid by his web-monkeys ;) |
| 22:02.20 | p3nguin | And there was an error on it, too. "SIP extension" <--- not a real thing. |
| 22:02.50 | p3nguin | Since that wasn't really pertinent to the question, I ignored it and answered the question that was asked. |
| 22:03.56 | p3nguin | I'm assuming we've all passed the test. |
| 22:05.32 | p3nguin | Now, what do we win? |
| 22:11.36 | p3nguin | Lots of extraneous apostrophes on that site, too. Someone needs to learn how to spell things. |
| 22:16.49 | mackhendricks | Thanks for those who took the quiz...I will reach out to the ones that provided real email addresses...thanks again |
| 22:17.17 | [TK]D-Fender | \o/ |
| 22:17.36 | p3nguin | My address may not look real, but I assure you that I will receive the mails sent to the address I provided. |
| 22:18.19 | p3nguin | Now go fix all those apostrophes. |
| 22:18.23 | [TK]D-Fender | I make no such promises ;) |
| 22:23.50 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-adgfdywadygzuryu) |
| 22:35.24 | *** join/#asterisk albertoandrade (~albertoan@186.206.18.74) |
| 22:37.17 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 22:40.25 | mick_laptop | mackhendricks: how many people took this test? 'm curious |
| 22:40.47 | mackhendricks | like 10 folks |
| 22:41.15 | mackhendricks | Average score 8 |
| 22:41.18 | mackhendricks | out of 10 |
| 22:41.26 | mick_laptop | any fails? |
| 22:41.48 | mackhendricks | Lowest is 60% |
| 22:42.14 | mick_laptop | thinks he should at least try it |
| 22:43.03 | mick_laptop | eh, I probably should study more (I'd like to do the dcap) |
| 22:43.21 | p3nguin | You don't need to study for his quiz. |
| 22:43.41 | afink | is there a variable for voicemail context? |
| 22:43.50 | p3nguin | It's not variable, no. |
| 22:44.18 | mick_laptop | did anyone here take the dcap? |
| 22:46.32 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:47.23 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 22:48.54 | *** part/#asterisk fireman_biff (~biff@65.48.222.203) |
| 22:53.07 | *** join/#asterisk bmg505 (~leon@196-209-123-205.dynamic.isadsl.co.za) |
| 23:15.37 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-qpinlvbnlaoqxwbp) |
| 23:33.33 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:40.00 | ectospasm | mick_laptop: I've taken (and passed) the dCAP, long ago. |
| 23:43.48 | *** join/#asterisk Foxi352 (~Foxi352@v-172-4.access.restena.lu) |
| 23:45.48 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 23:54.37 | WIMPy | Ah, the traffic is back. |