| 00:00.19 | grandpapadot | BINGO! #COLOR=yes |
| 00:00.23 | grandpapadot | MEGA thanks guys ... |
| 00:00.33 | spenguin[work] | is it fixed yet "[ |
| 00:00.35 | grandpapadot | No spikes, 0.0, all 4 cpus!!!! |
| 00:00.36 | spenguin[work] | :P |
| 00:00.42 | grandpapadot | WOOHOO!!! |
| 00:00.43 | spenguin[work] | interesting |
| 00:00.59 | Belgarath | color .... |
| 00:01.02 | Belgarath | cpu spiking |
| 00:01.06 | Belgarath | jesus ... |
| 00:01.23 | grandpapadot | lol |
| 00:01.24 | Belgarath | is it trying to paint each packet pink > |
| 00:01.29 | Belgarath | before sending it ? |
| 00:01.37 | grandpapadot | Man, I was going back compiling old version, all kinds of crazy sh*t |
| 00:01.46 | Belgarath | :) |
| 00:01.48 | grandpapadot | fucking ansi codes |
| 00:01.55 | grandpapadot | lol, hey big thanks guys, that was HUGE |
| 00:02.06 | Belgarath | and COLORful |
| 00:02.09 | *** join/#asterisk Yedidya (~Yedidya@94-192-216-27.zone6.bethere.co.uk) |
| 00:02.14 | TJNII | I think that is a known issue. I *think.* |
| 00:03.31 | Belgarath | TJNII: well i dodn;t know about it |
| 00:04.25 | grandpapadot | The default in the new debian init script is to have it enabled, I'm going to look in mantis and if it's not their submit a bug report because I'm sure this will get someone else. |
| 00:05.54 | Belgarath | :) |
| 00:05.56 | Belgarath | good |
| 00:05.59 | Belgarath | helpthe comunity |
| 00:06.41 | TJNII | Belgarath: Then you need to squat here more. :P |
| 00:07.06 | TJNII | squats here all the time... and is rarely on topic. |
| 00:08.40 | Belgarath | :) |
| 00:24.04 | jblack | Heh. Will Polgara be along presently? |
| 00:24.37 | Belgarath | don;t think so |
| 00:24.40 | Belgarath | it's too late |
| 00:31.01 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 00:39.13 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:42.36 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 00:45.24 | Naikrovek | where is this color=yes thing |
| 00:45.37 | Naikrovek | what in asterisk has a color |
| 00:46.50 | Belgarath | Naikrovek: some things if you run it in debug |
| 00:46.52 | Belgarath | :P |
| 00:46.56 | Naikrovek | aah |
| 00:47.28 | Belgarath | i bet it pays for the 75% of the cpu load |
| 00:56.18 | *** part/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 01:01.49 | ariel_ | color is ansi just displays on the cli, does not add much if any cpu load |
| 01:03.11 | Belgarath | :)) |
| 01:03.17 | Belgarath | well it does in this case:)) |
| 01:05.20 | Yedidya | Any clue on when 1.6.1/2 will be released as rpm? |
| 01:05.48 | Yedidya | Or will it be 1.8 next? |
| 01:10.46 | ChannelZ | It will be called AsteriskSquared |
| 01:12.35 | jblack | well, it requies 4 times as much effort to display as ascii . At least. ;) |
| 01:13.01 | jblack | I don't think that matters on actual computers |
| 01:14.56 | ChannelZ | besides who needs rpms |
| 01:15.52 | pentanol | hi, does anybody knows what's wrong in these selects? http://codepad.org/zReYhh4f |
| 01:16.15 | ChannelZ | Could be anything |
| 01:16.31 | ChannelZ | Do those columns exist? What database is it using? Does the user have proper privileges? |
| 01:16.34 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:16.37 | ChannelZ | It's a bit like asking how long is a piece of string |
| 01:17.02 | pentanol | yes I can exexutre this select from mysql client |
| 01:17.32 | ChannelZ | as the same user? Is the database name set properly in your config? |
| 01:18.34 | ChannelZ | Then perhaps the client library you've compiled against is jacked, or ODBC is busted, who knows. Turn on logging on the mysql side and see what the queries really look like from * (or if it's even logging in properly) |
| 01:19.39 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:22.31 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 01:23.21 | ChannelZ | tries to decide what to do for dinner |
| 01:26.44 | pentanol | ChannelZ you're australian? |
| 01:32.08 | *** join/#asterisk eppigy (~eppigy@c-69-180-16-188.hsd1.ga.comcast.net) |
| 01:32.13 | ChannelZ | uh, no |
| 01:35.49 | jaytee | hello Dave |
| 01:40.38 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 01:56.54 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:57.36 | *** join/#asterisk Slugs_ (~yeah@173-8-52-45-Jacksonville.hfc.comcastbusiness.net) |
| 01:58.54 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:02.30 | eppigy | hello |
| 02:09.45 | mjsilva | yes |
| 02:10.06 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 02:10.24 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 02:18.56 | ChannelZ | no |
| 02:20.09 | Slugs_ | maybe |
| 02:22.26 | ChannelZ | cusses screen |
| 02:40.14 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:42.19 | *** join/#asterisk CatLynx (sione@ocs.net) |
| 02:42.46 | CatLynx | anyone know if Intel ATOM 330 is decent enought for Asterisk? |
| 02:42.53 | CatLynx | to handle about 10 calls at the same time |
| 02:43.01 | synaptic2010 | no |
| 02:43.03 | [TK]D-Fender | CatLynx: Doing what? |
| 02:43.09 | synaptic2010 | ATOM is bad |
| 02:43.14 | [TK]D-Fender | Atom is fine |
| 02:43.15 | CatLynx | just SIP |
| 02:43.24 | [TK]D-Fender | CatLynx: Any recording? Transcoding? |
| 02:43.38 | CatLynx | maybe a few transcoding and voicemail |
| 02:44.00 | jaytee | ~cheap |
| 02:44.01 | infobot | [cheap] a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 02:44.10 | CatLynx | heh |
| 02:44.14 | [TK]D-Fender | CatLynx: probably more than fine |
| 02:44.37 | CatLynx | I have choice of Atom 330 or Intel Core2 Duo have both motherboards laying here :) |
| 02:44.47 | CatLynx | 330 uses very little power from the wall :) |
| 02:44.51 | jaytee | go with the Core2 Duo |
| 02:45.07 | CatLynx | its compaiing 30watts with 110watts :) |
| 02:45.57 | CatLynx | hard choice |
| 02:45.59 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 02:46.19 | [TK]D-Fender | CatLynx: You'll be fine |
| 02:46.23 | ChannelZ | maybe you should go with tin cans and string |
| 02:46.24 | CatLynx | if the 330 can handle what I need, then it saves money from drawing power form the power company :) |
| 02:46.56 | CatLynx | hmm wonder how tin cans and strings can do conferance ;) |
| 02:47.19 | jaytee | test all the features you want to use and make sure it can handle the load without unexpected load issues before you deploy it if you're setting it up for a customer. |
| 02:47.33 | ChannelZ | You un-twist the strands of the string to connect to multiple cans |
| 02:47.48 | CatLynx | its for myself, wanting to move my asterisk off my mail server on its own box |
| 02:48.10 | jaytee | yeah, it ought to do alright then |
| 02:48.34 | CatLynx | any sites put up a perfomance number of asterisk with Atom 330? |
| 02:49.08 | jaytee | not that I'm aware of of but you might Google Asterisk + Atom 330 |
| 02:51.36 | CatLynx | heh a friend was just saying I should stick with Core2 it would make asterisk upgrade much faster when i compile |
| 02:52.53 | ChannelZ | there's that |
| 02:53.17 | ChannelZ | my machine at home is a P3, it takes forever |
| 02:54.18 | CatLynx | guess I have to do my own testing and see how it works. |
| 02:55.36 | CatLynx | there is 2 people I know that uses this asterisk box using GSM codec |
| 02:55.56 | CatLynx | that might be some impact on the atom |
| 03:10.23 | CatLynx | ok atom is out, gsm to ulaw kills it |
| 03:10.36 | CatLynx | has very bad performance |
| 03:20.38 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:21.15 | synaptic2010 | "Newbie" seeking some support |
| 03:22.16 | synaptic2010 | I told you ATOM is bad, I did not even have to try it with asterisk to tell you |
| 03:23.26 | ChannelZ | well that's credible. |
| 03:23.52 | synaptic2010 | i tryed it with many other stuff |
| 03:24.23 | synaptic2010 | the CPU dies at 50% |
| 03:26.28 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 03:32.54 | *** join/#asterisk Taco24501 (~Taco24502@c-67-187-29-41.hsd1.va.comcast.net) |
| 03:38.46 | *** join/#asterisk b14ck (~b14ck@cpe-76-95-129-196.socal.res.rr.com) |
| 03:41.43 | ChannelZ | yawns |
| 03:43.05 | *** join/#asterisk coppice (~chatzilla@93.176.64.202.dyn.pacific.net.hk) |
| 03:46.44 | *** join/#asterisk pawz (~pawz@ppp118-208-157-82.lns20.bne1.internode.on.net) |
| 03:50.22 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 03:58.22 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 03:58.36 | *** join/#asterisk tecnico (~tecnico@75.76.169.148) |
| 04:02.58 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:13.55 | ChannelZ | jeebus, clamav takes forever and a day to compile now |
| 04:22.22 | hardwire | I use bins |
| 04:24.27 | TJNII | Heh, I just found a database dump of a customer database on a disk I found at a junk store. |
| 04:24.35 | TJNII | It is 20 years old, but still... |
| 04:25.05 | ChannelZ | guess security was just as bad then as it is now |
| 04:26.00 | TJNII | I guess |
| 04:45.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:46.10 | *** join/#asterisk smooth_penguin (~smoove@59.95.24.1) |
| 05:06.52 | *** join/#asterisk parag7732 (~PARAG@mail2.eastnets.com) |
| 05:08.27 | parag7732 | guys, do we have anything similar to t_relay_to("ipaddress"); function of OpenSer where i can forward my all REGISTER request to <ipaddress> |
| 05:16.22 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 05:25.18 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:37.25 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 05:49.24 | *** join/#asterisk dewinda (~sahan@123.231.102.240) |
| 05:49.46 | dewinda | hi all |
| 05:51.31 | dewinda | i have a cli issue with dahdi . Im getting No start bit found in fsk data |
| 05:52.06 | dewinda | im using TDM400 card and gsm modem |
| 05:52.44 | dewinda | usecallerid=yes |
| 05:52.45 | dewinda | cidstart=ring |
| 05:52.47 | dewinda | cidsignalling=bell |
| 05:52.48 | dewinda | callerid = asreceived (For incoming trunks) |
| 05:53.36 | dewinda | and tried different values for cid_rxgain |
| 05:54.32 | dewinda | [Apr 11 11:14:22] ERROR[9743]: callerid.c:562 callerid_feed: No start bit found in fsk data. |
| 05:54.33 | dewinda | [Apr 11 11:14:22] WARNING[9743]: chan_dahdi.c:8735 ss_thread: CallerID feed failed: Success |
| 05:54.35 | dewinda | [Apr 11 11:14:22] WARNING[9743]: chan_dahdi.c:8842 ss_thread: CallerID returned with error on channel 'DAHDI/1-1' |
| 06:01.12 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:05.36 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 06:11.58 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:14.53 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 06:19.44 | *** join/#asterisk talntid (talntid@173.160.174.126) |
| 06:40.03 | dewinda | can somebody help me on this cli issue |
| 06:49.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:50.20 | *** join/#asterisk hipitihop (~denis@203-221-91-83.dynamic.reverse.flexinet.com.au) |
| 07:00.30 | talntid | dewinda, ask :) |
| 07:19.00 | ChannelZ | Are you sure you have callerid? |
| 07:26.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:45.46 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 08:02.38 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:08.10 | *** join/#asterisk coppice (~chatzilla@202.64.176.93) |
| 08:18.44 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 08:25.04 | dewinda | sure |
| 08:25.13 | dewinda | sorry for the delay |
| 08:25.42 | dewinda | i tested it with cli phone |
| 08:28.47 | dewinda | it workis fine |
| 08:29.43 | *** join/#asterisk Wildy (~simba@217.197.248.94) |
| 08:38.38 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:50.28 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 09:11.21 | carrar | moof |
| 09:12.10 | *** join/#asterisk ivaxer (~ivaxer@dev.ptr.sgu.ru) |
| 09:14.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:17.29 | *** join/#asterisk pentanol (~pentanol@77.35.49.164) |
| 09:17.57 | ChannelZ | foom |
| 09:18.35 | ChannelZ | dewinda: well are you sure all your options are set right for your country/telco? Maybe they don't send CID until after the first ring, the format is different, etc. |
| 09:25.23 | *** join/#asterisk suma (~pongada@c-67-166-21-115.hsd1.co.comcast.net) |
| 09:25.26 | suma | hi |
| 09:25.38 | suma | may someone help to receive calls from localphone ? |
| 09:25.57 | suma | i can see my account is registered |
| 09:26.04 | suma | but not receiving calls |
| 09:27.46 | *** join/#asterisk MaliutaLap (~biteme@kiev.lusan.id.au) |
| 09:34.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:50.18 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:12.20 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 10:14.45 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 10:24.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:34.38 | talntid | No registration for peer '1000' |
| 10:35.07 | talntid | Trying to resolve this ^ just trying to register a softphone with a fresh install of * -- can someone assist? :) |
| 10:35.24 | ChannelZ | not with 0 information |
| 10:35.39 | talntid | right. what info would you like? sip.conf and extensions.conf? |
| 10:36.20 | ChannelZ | sip.conf to start and what is the complete error/warning in the console? When is it appearing? |
| 10:36.46 | talntid | http://pastebin.com/hX2FgBxj |
| 10:37.14 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-172-249.clienti.tiscali.it) |
| 10:38.13 | talntid | http://pastebin.com/szUEixwE |
| 10:38.18 | talntid | happens when i try to register the softphone |
| 10:39.22 | ChannelZ | you can remove 'username' for one |
| 10:39.27 | talntid | roger that |
| 10:39.50 | talntid | done. |
| 10:41.04 | ChannelZ | What softphone? |
| 10:41.26 | talntid | zoiper |
| 10:41.29 | talntid | same happens with x-lite |
| 10:41.37 | talntid | i have no softphone preference |
| 10:43.04 | *** join/#asterisk jsjc (~chatzilla@219-90-149-184.ip.adam.com.au) |
| 10:43.24 | jsjc | is there anywhere I can find a list with all the patterns and telephones explained?? |
| 10:43.37 | ChannelZ | talntid: did you 'sip reload' and try again? |
| 10:43.45 | jsjc | such us 3491XXXXXXX local spain |
| 10:43.49 | ChannelZ | jsjc: patterns and telephones? |
| 10:43.50 | talntid | yes, channelz, same error when i did sip reload and tried again |
| 10:43.54 | jsjc | 346XXXXXXXX spain mobile |
| 10:44.05 | jsjc | ChannelZ: how is that said in proper jargon? |
| 10:44.06 | ChannelZ | talntid: what does 'sip show peers' show |
| 10:44.09 | talntid | what is odd though, is that... i am doing these settings via SIP.. |
| 10:44.16 | talntid | the the error, is from chan_iax2.c |
| 10:44.29 | talntid | 2000/2000 (Unspecified) D 0 Unmonitored |
| 10:44.33 | jsjc | it is to configure the rounting of all my outgoing calls.... |
| 10:44.45 | jsjc | and now with this massive differneces in price from SIP to SIP provider |
| 10:44.48 | ChannelZ | talntid: Uhh how are you configuring Zoiper? Did you add a new SIP account or an IAX one? |
| 10:44.53 | jsjc | I wanted to use the best of a few of them. |
| 10:45.01 | talntid | added a SIP account |
| 10:45.06 | talntid | that was the first thing i re-checked ;) |
| 10:46.22 | talntid | feel free to register to my PBX using 2000/2000 --- address is "spokanepcrepair.com" |
| 10:46.30 | talntid | it doesn't *do* anything. it's just for my learning. |
| 10:46.39 | ChannelZ | talntid: well your sip show peers is not showing 1000 at all which means something isn't parsing right for your [1000] entry in sip.conf |
| 10:46.58 | ChannelZ | Make sure you don't have any strange characters in there that I'm not seeing in the paste |
| 10:47.01 | talntid | ChannelZ, I have 1000 and 2000 |
| 10:47.13 | talntid | i only pasted one, it says the same for 1000 - i didn't want to flood the channel |
| 10:49.04 | ChannelZ | I appear to have just registered as 2000 |
| 10:49.11 | talntid | yes, i see.... |
| 10:49.12 | talntid | hmmmm |
| 10:49.35 | talntid | using an asterisk 1.6 box? |
| 10:49.44 | ChannelZ | Yes. 1000 is working too |
| 10:50.11 | talntid | yes, i am looking at it. it is.... |
| 10:52.32 | ChannelZ | jsjc: I think you are talking about http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-5-SECT-3.html#asterisk-CHP-5-SECT-3.6 |
| 10:52.49 | ChannelZ | talntid: what are you entering into Zoiper? |
| 10:56.43 | talntid | ChannelZ: domain: spokanepcrepair.com | username: 1000 | password: 1000 |
| 10:57.12 | ChannelZ | Someone else is trying to register via IAX, your SIP config is fine (and functioning) |
| 10:57.23 | ChannelZ | Who is 173.160.174.126? |
| 10:57.48 | talntid | that's my IP |
| 10:58.03 | ChannelZ | So this * server is remote to you |
| 10:58.15 | talntid | yes |
| 10:58.20 | talntid | it's on my linode |
| 10:58.25 | ChannelZ | Then I'd say you have entered an IAX account under Zopier also that is still trying to register |
| 10:58.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:59.55 | talntid | http://spokanespeed.net/media/zoiper.png |
| 11:00.51 | ChannelZ | or something else is still running in the background |
| 11:01.06 | Mark22 | Hello, when I cal from 1011 to 1012 and pickup 1012 everything works fine, when I after that try to do an attended transfer from 1012 to 1010 it first starts to ring 1010 (as expected) and when I pickup 1010 I keep heering beeps at 1012 while 1011 is hearing a wait music (music on hold). Information about my dialplan http://yourpaste.net/5134/ (if you want more information, just ask), information from rasterisk (with core set verbose 10) http://yourpast |
| 11:01.08 | ChannelZ | Those messages shouldn't be coming from chan_iax2 |
| 11:02.41 | talntid | yeah, thats my thoughts. |
| 11:03.25 | ChannelZ | yet Zoiper isn't registering for you |
| 11:04.40 | talntid | trying x-lite |
| 11:04.44 | talntid | though i hate x-lite. |
| 11:05.17 | *** join/#asterisk ttwhy (~tekkno@p4FECF87C.dip.t-dialin.net) |
| 11:05.19 | Mark22 | talntid: do you have a better softphone for on windows? |
| 11:05.37 | talntid | no, i have not tried many |
| 11:06.32 | ChannelZ | hmm are you getting more reg errors on the console now? Trying to reg through Zoiper myself but not getting a response at all actually |
| 11:07.50 | talntid | no, i'm notg |
| 11:08.09 | talntid | i am getting nothing on console when trying with x-lite, zoiper, etc.. |
| 11:08.11 | ChannelZ | Mark22: You say when you do a transfer you are 'hearing beeps', as in ringing, or what sort of beeps? |
| 11:08.20 | Mark22 | as in ringing |
| 11:08.22 | ChannelZ | talntid: this is strange |
| 11:08.30 | talntid | indeed, ChannelZ |
| 11:08.35 | Mark22 | even when I answer the phone with number 1010 |
| 11:08.56 | ChannelZ | Mark22: And what do you hear when you pick up the phone you've transferred to? |
| 11:09.34 | *** join/#asterisk zetyui (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 11:09.39 | zetyui | hi there |
| 11:09.43 | talntid | ChannelZ, I am going to try to register using my Polycom. :) |
| 11:09.54 | ChannelZ | talntid: ok now I think your * either crashed or something has changed because I can't register with Asterisk now either |
| 11:10.04 | zetyui | i don't know if is it that good place to talk about callshop module for asterisk ? |
| 11:10.28 | talntid | I just tried registering using polycom, but i typo'd and made it say .1000 instead of just 1000 |
| 11:10.35 | talntid | changed the setting, now restarting polycom. |
| 11:10.43 | talntid | the console error'd on the .1000 |
| 11:11.08 | ChannelZ | I'm just getting timeouts now |
| 11:11.12 | talntid | odd. |
| 11:11.14 | talntid | i'll restart * |
| 11:12.03 | Mark22 | ChannelZ: I can hear what the person on phone 1012 says |
| 11:12.04 | zetyui | i don't know if is it that good place to talk about callshop module for asterisk ? |
| 11:12.48 | ChannelZ | Mark22: And you're using the xfer function of your phone? |
| 11:13.47 | ChannelZ | Mark22: Actually.. in your dialplan, you have _10XX doing Answer() and then Dial() -- remove the Answer step |
| 11:14.36 | Mark22 | no as xfer function on my phone doesn't exist, I use *2 (as mentioned in http://yourpaste.net/5135/ ) |
| 11:14.42 | Mark22 | should that fix it? |
| 11:15.05 | ChannelZ | What SIP phone are you using that has no transfer? |
| 11:15.48 | Mark22 | normall phone with a pap2t device between them and the asterisk server |
| 11:15.58 | ChannelZ | oh |
| 11:16.11 | Mark22 | and on the x-lite softphone it is disabled and I don't have polycoms here to test it with them |
| 11:18.06 | zetyui | yes or not merc |
| 11:20.34 | ChannelZ | Mark22: well remove the Answer and see if it makes any difference, I"m not sure it will but it's possible it's getting confused |
| 11:20.46 | Mark22 | ChannelZ: changing that to http://yourpaste.net/5136/ didn't change it |
| 11:21.06 | Mark22 | for 10XX it is now without the answer |
| 11:26.21 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 11:31.36 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 11:33.48 | ChannelZ | Mark22: not sure why you're hearing the transferer on the transferred-to phone, but the transferer still gets ringing |
| 11:35.05 | Mark22 | it is a known error for freepbx if I am correct, but we don't use that (I edit asterisk config files manually) |
| 11:46.23 | ariel_ | morning folks |
| 12:06.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:07.34 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 12:08.24 | dewinda | still no luck in getting caller id |
| 12:09.16 | dewinda | through dahdi can some one help me |
| 12:21.18 | *** join/#asterisk mhaddog (~mhaddog@83.33.30.137) |
| 12:21.26 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:37.24 | *** join/#asterisk TimeRider (~steve@5ad08e49.bb.sky.com) |
| 12:40.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:47.26 | *** join/#asterisk MaliutaLap (~biteme@kiev.lusan.id.au) |
| 12:47.43 | *** join/#asterisk Taco24501 (~Taco24502@c-67-187-29-41.hsd1.va.comcast.net) |
| 12:57.36 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 12:58.23 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 12:59.19 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 13:07.57 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 13:08.37 | adnc | hello, i'm trying to use a my own db field for sqlite with CDR(...) but it doesnt work. is it possible that this does not work with sqlite? |
| 13:13.20 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:20.30 | jsjc | I wonder the licenses for g729 channels... how they work. It says per fchannel so if I have Softphone -----G729---> Asterisk <-------G729--------VOIP provider. That counts as 2 channels? or just one? |
| 13:21.09 | jsjc | If I buy just 4 channels and they get in use and a 5th call happens.... it will jump to other codec if allowed? |
| 13:23.52 | *** join/#asterisk TJ^ (~TJ@193.47.83.49) |
| 13:48.37 | TJ^ | anyone know a good (free) way to implement some QoS for voip traffic without buying a managed router? |
| 13:48.41 | TJ^ | *switch |
| 13:52.44 | dlynes | TJ^, No free way...but it depends on what you mean by free |
| 13:52.54 | dlynes | TJ^, you can implement QoS using iptables |
| 13:52.58 | TJ^ | no hardware expense... |
| 13:53.02 | TJ^ | or software |
| 13:53.07 | TJ^ | just time and sweat |
| 13:53.09 | TJ^ | :P |
| 13:53.17 | dlynes | TJ^, then, no....because you still need hardware to run linux on |
| 13:53.32 | dlynes | TJ^, not to mention asterisk |
| 13:53.59 | talntid | smoothwall.org can, can't it? |
| 13:54.26 | dlynes | talntid, he's wanting something completely free...smoothwall's free, but it needs hardware to run on |
| 13:54.46 | talntid | maybe he has hardware laying around? *shrug* |
| 13:54.50 | talntid | hehe |
| 13:54.56 | talntid | all a technicality |
| 13:55.02 | TJ^ | well i mean not a new netgear managed switch for $100 |
| 13:55.15 | TJ^ | and yes hardware laying around |
| 13:55.18 | dlynes | TJ^, but a new $500 computer is fine? |
| 13:55.18 | talntid | TJ^, got an old POS computer laying around? |
| 13:55.29 | dlynes | TJ^, ah...well....that's why i asked what you meant by free |
| 13:55.35 | TJ^ | ok |
| 13:55.38 | TJ^ | free software :) |
| 13:55.45 | TJ^ | smoothwall looks good |
| 13:55.46 | talntid | TJ^, www.smoothwall.org -> go read. |
| 13:56.06 | dlynes | TJ^, smoothwall is just a distro with a gui interface for iptables |
| 13:56.12 | TJ^ | i see |
| 13:56.23 | dlynes | TJ^, You can also try looking at www.shorewall.org, if you'd prefer to use your own distro |
| 13:56.32 | dlynes | TJ^, It uses configuration files, that you hand edit |
| 13:56.57 | dlynes | TJ^, Most modern distros have it available as a pre-compiled package |
| 13:57.20 | TJ^ | as suppose both of those solutions would need 2nic's |
| 13:57.42 | talntid | dlynes, although I do know what smoothwall/shoreall are used for.... how are they implemented in a network? do they go in-line between the internet connection,and internal network? or can they go anywhere? |
| 13:58.43 | talntid | i have never had to use for one |
| 14:00.31 | talntid | yup. |
| 14:00.34 | talntid | TJ^: http://www.skullbox.net/smoothwall.php |
| 14:01.00 | dlynes | TJ^, correct...or 3 if you want a DMZ |
| 14:01.12 | TJ^ | yea i thought so |
| 14:01.24 | TJ^ | trying to get something that "slots" into my network |
| 14:01.35 | TJ^ | or for the business im working for atm |
| 14:01.55 | TJ^ | modem --> sonicwall --> unmanaged switch --> LAN |
| 14:02.40 | dlynes | talntid, they create what's known as the firewall...typically your proxy server can also run on this machine, as well as your mail server and web server...depends on what your security model is |
| 14:02.41 | TJ^ | and the more i look into this the more a managed switch makes sense to me... |
| 14:02.50 | dlynes | talntid, However, in a voip situation |
| 14:03.01 | talntid | I already have proxy server, DHCP, web, mail, etc.... |
| 14:03.18 | dlynes | talntid, you'll probably want three network cards....1 for asterisk, 1 for your regular network card, and 1 for the internet |
| 14:03.27 | talntid | i would not want it to take over that functionality.. :) |
| 14:03.35 | talntid | my * box goes out over PRI, so no worries |
| 14:03.47 | TJ^ | pri? |
| 14:03.49 | dlynes | talntid, or, instead of the extra card for voip, you could look at using vlans, but a separate physical adapter is more effective |
| 14:04.01 | talntid | TJ^, T1. |
| 14:04.06 | TJ^ | ah ok |
| 14:04.15 | talntid | primary rate interface |
| 14:04.27 | dlynes | TJ^, shorewall/smoothwall would replace your sonicwall |
| 14:04.33 | TJ^ | yea i know |
| 14:04.43 | dlynes | TJ^, the good thing about that |
| 14:04.46 | talntid | hmm, i feel like trying smoothwall out now :) |
| 14:04.48 | TJ^ | but that sonicwall is running a permanent VPN tunnel with 2 other locations |
| 14:04.48 | dlynes | TJ^, is no more buggy firmware |
| 14:05.06 | TJ^ | and the TZ 170 is working really well |
| 14:05.07 | dlynes | TJ^, and no more incompatibility with linux for vpns :) |
| 14:05.10 | TJ^ | don't want to touch it |
| 14:05.41 | TJ^ | brb |
| 14:05.52 | talntid | can someone here test my PBX from there... to see if you can register on my PBX from your location? the reason is because I am opening it up to telemarketing at home, and want to ensure it works so far... |
| 14:05.53 | dlynes | TJ^, ok, so if you already have a sonicwall....why are you asking about qos? |
| 14:06.00 | dlynes | TJ^, Can't sonicwall provide that? |
| 14:06.01 | talntid | i'm local to it, and it works for me.. |
| 14:07.16 | *** join/#asterisk memph (~memph@sd1438.sivit.org) |
| 14:08.18 | TJ^ | dlynes it cant |
| 14:08.40 | TJ^ | unless im being really retarted and didnt read the manual/router menus properly |
| 14:09.54 | TJ^ | yea i need the enhanced firmware |
| 14:10.15 | TJ^ | which will probably cost me more then i want to spend |
| 14:11.25 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:12.05 | jsjc | is there any way to set language of WHOLE asterisk to es instead of en. Not just channel by channell.... |
| 14:13.35 | Kobaz | jsjc: in sip.conf/iax.conf/etc in the [general] section... language=en |
| 14:13.38 | Kobaz | er |
| 14:13.39 | Kobaz | es |
| 14:13.47 | Kobaz | or whatever you want it to be |
| 14:14.06 | dlynes | TJ^, yeah...you probably need the enterprise firmware, rather than the standard firmware |
| 14:14.21 | dlynes | TJ^, sonicwall always knows how to squeeze every last penny out of you |
| 14:14.38 | Kobaz | heh |
| 14:14.49 | Kobaz | isn't sonicwall some gui on top of linux iptables |
| 14:14.52 | jsjc | Kobaz: for some bloody reason... voicemail still plays in english |
| 14:14.56 | dlynes | Kobaz, nah |
| 14:15.19 | TJ^ | i was thinking |
| 14:15.22 | dlynes | Kobaz, if it was, openvpn or strongswan would be compatible with it, and they're not |
| 14:15.28 | TJ^ | is there to initiate a call from a script? |
| 14:15.31 | TJ^ | say... |
| 14:15.41 | Kobaz | jsjc: i think you also need language=es in voicemail.conf |
| 14:15.41 | talntid | .. |
| 14:15.46 | talntid | look up call files, TJ^ |
| 14:15.48 | TJ^ | the backup server goes offline |
| 14:15.49 | talntid | .call files |
| 14:15.54 | Kobaz | jsjc: but i don't see a language= option in voicemail.conf samples |
| 14:16.04 | TJ^ | can i get the box to call me and tell me so using a pre-recorded message |
| 14:16.10 | TJ^ | call files? will do |
| 14:16.15 | jsjc | Kobaz: that is what i thought.. i give it a shot anyway. |
| 14:16.16 | Kobaz | jsjc: to answer your question.. there isn't a global for-everything language setting that i know of |
| 14:16.33 | jsjc | because when I do CLI> core show settings |
| 14:16.40 | jsjc | there is a setting is called Default Language |
| 14:16.44 | jsjc | and it is set to english of course |
| 14:16.49 | Kobaz | oh |
| 14:17.00 | Kobaz | try in the [general] of asterisk.conf |
| 14:17.19 | Kobaz | i really never played with language stuff, so that's a total guess |
| 14:17.51 | Kobaz | jsjc: you could look at the source code and see where it pulls that setting from |
| 14:18.45 | jsjc | will investigate and try to modify wiki with the where it is... |
| 14:18.50 | jsjc | because drives me insante |
| 14:19.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:19.49 | Kobaz | heh |
| 14:19.57 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 14:20.01 | Kobaz | good luck getting the wiki changed |
| 14:20.36 | WIMPy | it's neither language= nor defaultlanguage=, so I guess it's time for the source... |
| 14:23.31 | *** join/#asterisk fenrus (fenrus@oklart.com) |
| 14:23.33 | fenrus | o/ |
| 14:24.22 | WIMPy | Doesn't seem to be read from asterisk.conf. |
| 14:24.48 | synaptic2010 | Hello, I am looking for some consultants with VOIP experince |
| 14:26.19 | jaytee | well, people here usually have some experience with VOIP |
| 14:26.35 | synaptic2010 | not the onees I talked to |
| 14:26.36 | jaytee | and there's also #asterisk-consultants |
| 14:26.45 | jsjc | I cannot fin how... |
| 14:26.47 | synaptic2010 | ya that channel is emppty |
| 14:26.56 | synaptic2010 | I amm there for the past 12 hours |
| 14:27.04 | jaytee | what are you looking for? |
| 14:27.06 | jsjc | WIMPy: so it is jsut read from sip.conf? |
| 14:29.05 | synaptic2010 | I am seeking solutions for FAX over IP for 1, Whole Sale Implemintation over GUI for another, adjusting GUI so accounts are not randomely generated but admin creates it the way they like |
| 14:29.22 | WIMPy | jsjc: AFAICS it's only defined in ./include/asterisk.h as DEFAULT_LANGUAGE and not configurable. |
| 14:30.00 | jsjc | arghh |
| 14:30.10 | jsjc | lets recompile this stuff |
| 14:31.46 | WIMPy | Should be pretty easy to add a line to the config parser, if you want to. |
| 14:32.09 | jsjc | config parser? |
| 14:32.18 | jsjc | I just modify in asterisk.h and make && make install right? |
| 14:32.34 | jsjc | I am guessing... not an expert when it comes to this compiling things |
| 14:32.36 | WIMPy | ./main/asterisk.c |
| 14:32.39 | WIMPy | yes |
| 14:32.43 | jsjc | main saterisk? |
| 14:33.07 | WIMPy | You could make it runtime configurable there. |
| 14:33.08 | jsjc | I did here /usr/src/asterisk/include/asterisk.h |
| 14:33.29 | WIMPy | That's the one |
| 14:33.54 | jsjc | just changed the "en" for "es" and hopefully all sweetens up... |
| 14:36.58 | jsjc | ;) now says es |
| 14:37.00 | jsjc | lets test... |
| 14:38.30 | *** join/#asterisk smooth_penguin (~smoove@59.95.5.17) |
| 14:39.55 | jsjc | yeap! all good on the hood! |
| 14:41.34 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 14:53.23 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 14:59.11 | *** join/#asterisk rethus (~contio@p5087546D.dip.t-dialin.net) |
| 14:59.41 | rethus | hi @ all. |
| 15:00.00 | rethus | in my extensions.conf i have this line: exten => callin,3,NoOp(${SIPURI} ${SIPDOMAIN} ${SIPCALLID}) |
| 15:00.14 | rethus | but i got no output to cli |
| 15:00.15 | rethus | why |
| 15:01.00 | jaytee | rethus, that's not much to go on |
| 15:01.28 | jaytee | pastebin your extensions.conf and the output of the CLI for a failed call attempt |
| 15:02.00 | rethus | i only wan't to output some helping lines... how can i made this wit extensions.conf if not with Noop? |
| 15:02.55 | rethus | one moment... was my mistake, i forgot to reload the extensions |
| 15:06.38 | *** join/#asterisk mhaddog (~mhaddog@56.Red-79-152-193.dynamicIP.rima-tde.net) |
| 15:09.51 | tzafrir_laptop | WIMPy, what do you mean by "make it runtime configurable"? |
| 15:10.52 | tzafrir_laptop | It would still require some code changes. Not that difficult, but not a matter of simply editing asterisk.h |
| 15:13.51 | jsjc | I am trying to troubelshoot the voicemail and it pops gives me annoucement to leave a message adn just straight after the beep, hangs up.... why this could be? |
| 15:15.12 | WIMPy | tzafrir_laptop: No, but three lines in asterisk.c should do. |
| 15:15.56 | *** join/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-6-41.unitymediagroup.de) |
| 15:17.30 | *** join/#asterisk mhaddog (~mhaddog@56.Red-79-152-193.dynamicIP.rima-tde.net) |
| 15:17.56 | ArtemMakhutov | Hi, |
| 15:17.56 | ArtemMakhutov | where can I configure how asterisk should you the iLBC codec? |
| 15:17.56 | ArtemMakhutov | (13.3 Kbps with 30 ms frames or 15.2 Kbps with 20 ms frames) |
| 15:26.11 | jsjc | so how many channels for g729 needed for a call that goes trough asterisk but both ways with 729? |
| 15:26.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:26.42 | Researcher | is now away: You Can find me on Dalnet/Freenode/IrcNet/Efnet/GameSuge/Ircbox.com |
| 15:28.22 | Researcher | is now away: You Can find me on Dalnet/Freenode/IrcNet/Efnet/GameSuge/Ircbox.com |
| 15:28.42 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 15:35.04 | *** join/#asterisk mhaddog (~mhaddog@245.Red-88-25-254.staticIP.rima-tde.net) |
| 15:38.39 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 15:39.39 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 15:40.30 | *** join/#asterisk tecnico (~tecnico@75.76.169.148) |
| 15:43.17 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 15:57.35 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:58.38 | *** join/#asterisk maxime1986 (~maxime198@2001:5c0:1000:b::5bf7) |
| 15:58.53 | *** part/#asterisk maxime1986 (~maxime198@2001:5c0:1000:b::5bf7) |
| 16:03.45 | *** join/#asterisk devdvd (~twister19@173.155.51.32) |
| 16:04.59 | *** join/#asterisk HansF (~hans@78-22-102-245.access.telenet.be) |
| 16:07.48 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 16:12.34 | devdvd | hi all. I am looking to set up an asterisk system that will accept calls on an 800 number, then call external numbers that are on the pstn until one picks up (this is for a support queue system). What is the best voip provider to use for this task? |
| 16:13.51 | p3nguin | devdvd: That depends on how many calls you need to make and receive at any given moment. |
| 16:14.11 | *** part/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-6-41.unitymediagroup.de) |
| 16:15.10 | devdvd | well, given that one "call" will use 2 channels (one in off the 800 number and one out to the support tech) 20 at the absolute max, but im thinking more along the lines of 8-10 |
| 16:15.13 | rethus | is there a way to "PUSH" the info that a new caller dialed in into my php-application? |
| 16:15.25 | rethus | now i have a meta-refresch every 15 seconds |
| 16:15.41 | rethus | would be nice, if i have it in realtime |
| 16:17.20 | p3nguin | devdvd: Providers usually measure channels by inbound and outbound, not combined. So you might want 10 inbound and 10 outbound? |
| 16:17.35 | devdvd | p3nguin: at the max |
| 16:17.50 | devdvd | 10 in and 10 out yea |
| 16:18.37 | p3nguin | devdvd: Are you going to use all VoIP, or do you want to set up an analog line for connecting to the PSTN? |
| 16:19.03 | devdvd | voip |
| 16:21.26 | p3nguin | VoIP.ms offers unlimited channel DIDs, but I'm trying to determine if they limit outbound channels. |
| 16:22.04 | TJNII | rethus: You could probably stick an AGI in there to ping your app. I've never written an AGI that wasn't intended to interact with the caller, so I don't know how difficult it would be to integrate. |
| 16:22.18 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:c510:9858:15b2:4662) |
| 16:22.27 | p3nguin | VoIP.ms might not be the best ITSP for you as far as your toll-free number goes, though, because you'll pay 2.9 cents per minute on inbound calls. |
| 16:22.59 | TJNII | Yea, you'll need to find someone with a bulk rate on that 1800 number |
| 16:23.07 | p3nguin | devdvd: I think there are some providers out there that offer a fixed rate on unlimited inbound toll-free calling. |
| 16:23.10 | TJNII | They get pricy fast if your service is popular |
| 16:23.27 | p3nguin | Yep. That was my thought as well. |
| 16:24.08 | p3nguin | You don't even have to use the same company for inbound and outbound, though. |
| 16:24.51 | devdvd | ya that was another thought |
| 16:25.00 | TJNII | You can have multiple outbound carriers, too. (If, say, 5 of plan A is cheaper than one of plan B) |
| 16:25.55 | *** join/#asterisk dewinda (~sahan@123.231.102.240) |
| 16:26.25 | [TK]D-Fender | rethus: Have your PHP app connect via AMI and send out a UserEvent |
| 16:26.46 | [TK]D-Fender | rethus: hav your CALL issue the UserEvent so the PHP app which is sitting around listening can react to it |
| 16:30.05 | devdvd | p3nguin, TJNII: are there voip providers that I should absolutely stay away from? |
| 16:30.58 | devdvd | I have done quite a bit with this as a hobby (mostly used broadvoice). this is my first foray into doing it for something other than myself |
| 16:32.01 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:32.02 | p3nguin | devdvd: I don't have a list of bad ones, but some people might offer reasons that they've switched away from some company in favor of another. |
| 16:32.38 | devdvd | ok |
| 16:36.50 | devdvd | thanks for your help :) |
| 16:37.47 | *** join/#asterisk _omer (~omer@116.71.188.84) |
| 16:37.54 | _omer | hi |
| 16:38.28 | _omer | is there anyone who can provide me the correct syntax ....please check http://www.pastebin.org/147088 |
| 16:39.02 | Corydon76-dig | _omer: are you using 1.4 or 1.6? |
| 16:39.56 | _omer | 1.4 |
| 16:40.13 | Corydon76-dig | Okay, well, you're missing the $[ ] around the expression |
| 16:40.31 | _omer | I tried that too....but it didn't work...let me try again |
| 16:40.42 | p3nguin | GotoIf(condition?[labeliftrue]:[labeliffalse]) |
| 16:41.18 | *** join/#asterisk Faithful (~Faithful@119.12.159.34) |
| 16:41.18 | _omer | $[${CALLERID(num)}=947]|default|2|4 <-- here is what i have in realtime exensions "appdata" |
| 16:41.46 | Corydon76-dig | _omer: look at what p3nguin just said |
| 16:43.07 | _omer | hmmmmm |
| 16:43.36 | _omer | but syntax wont be same for realtime |
| 16:43.54 | Corydon76-dig | p3nguin: I may have written it, but I don't always remember it. Thank you. ;-) |
| 16:44.03 | Corydon76-dig | _omer: yes, it will |
| 16:44.44 | p3nguin | For what it's worth, I use 1.4, and GotoIf does accept "context,exten,priority" as the label, but I'm not using realtime. |
| 16:44.47 | *** join/#asterisk rare1980_ (~rare1980@115.186.0.163) |
| 16:45.12 | Corydon76-dig | p3nguin: true, but all you need in the label is to change the commas to pipes |
| 16:45.33 | Corydon76-dig | and then it'll work in realtime |
| 16:45.39 | p3nguin | Oh, interesting. |
| 16:46.51 | [TK]D-Fender | [12:41]<_omer>$[${CALLERID(num)}=947]|default|2|4 <-- here is what i have in realtime exensions "appdata" <--- where is the "?" that this app REQUIRES? |
| 16:47.35 | _omer | see! .. I am confused about syntax in realtime.. |
| 16:47.39 | p3nguin | Changing that first pipe to ? might solve his issue, eh? |
| 16:47.51 | _omer | alright |
| 16:47.53 | _omer | let me try |
| 16:48.01 | [TK]D-Fender | _omer: no.. this is the APP syntax. this is all part of the data and has nothing to do with being in realtime |
| 16:48.40 | _omer | p3nguin: u r right...changing first pipe to ? ... works |
| 16:48.45 | _omer | $[${CALLERID(num)}=947]?default|2|4 <--- works |
| 16:48.48 | _omer | in realtime |
| 16:48.52 | _omer | thanks all |
| 16:48.58 | p3nguin | woohoo! |
| 16:49.09 | _omer | :) bye |
| 16:52.01 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 16:53.05 | *** join/#asterisk Tim_Toady (~moi@77.49.52.245.dsl.dyn.forthnet.gr) |
| 16:55.16 | Corydon76-dig | Someday, when someone is inordinately bored, they should figure out what they can do with Asterisk when they eschew use of code written by each of the core developers |
| 16:56.08 | Mark22 | I asked before about this topic, but are there known issues with building asterisk from source and manually editing the config files to do an attended transfer (the person to transfer to can here the person that tries the transfer, but the person that tries to transfer the phone call can't here the person he wants to transfer the call to) |
| 16:56.28 | Mark22 | I use asterisk 1.6.2.1 (I know, I should update) |
| 16:56.36 | Mark22 | but I need to know if an update should fix this problem |
| 16:57.22 | [TK]D-Fender | Mark22: We see no circumstances of the actual failed attempt to judge |
| 16:57.41 | [TK]D-Fender | Mark22: Do yous ee a bug issued on the tracker for this? |
| 16:57.43 | Corydon76-dig | Mark22: we have not extensively analyzed the changes we've made to consider every bad possible situation in the code that would be rectified. We just fix issues and move on. |
| 16:59.01 | Corydon76-dig | So it's possible that your situation would be improved, even if there isn't an explicit issue with the same problem you're having. |
| 16:59.22 | [TK]D-Fender | Of course.. there is no proof this is abug so far. |
| 17:00.39 | Mark22 | I couldn't find a bug on the tracker about this problem to be honest (at least not something that is created after 2009 or for 1.6.x.x) |
| 17:00.50 | Corydon76-dig | Moreover, we do not have the resources to do that type of analysis, both because it's rather complex, and because we and most users would prefer if we spent our time on other issues |
| 17:00.53 | [TK]D-Fender | Mark22: Where's the failed call for us to look at? |
| 17:02.29 | Mark22 | http://yourpaste.net/5133/ http://yourpaste.net/5134/ from 1011 to 1012 and after that transfer from 1012 to 1010 (however if 1010 answers the phone 1012 didn't hear anything from 1010 (it keeps giving beeps), but 1010 can hear 1012 without any problem) |
| 17:03.31 | Mark22 | I use *2 to transfer the call (and did configure it in features.conf, information from features.conf can be found at http://yourpaste.net/5135/ ) |
| 17:03.33 | [TK]D-Fender | marthere is no call in there. |
| 17:03.38 | [TK]D-Fender | Mark22: there is no call in there. |
| 17:04.54 | Mark22 | http://yourpaste.net/5133/ << that is the log from the call (with verbose set to 10) |
| 17:05.41 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:06.17 | [TK]D-Fender | Mark22: look like 1012 is calling itself |
| 17:06.56 | [TK]D-Fender | Mark22: Also I don't know if the SIP side is fine as it is |
| 17:07.13 | [TK]D-Fender | Mark22: What phones are you using? |
| 17:08.23 | Mark22 | x-lite = 1010, 1011 and 1012 are connections on a linksys pap2t (sip-ata box) with "regular" phones connected to it |
| 17:08.49 | [TK]D-Fender | Mark22: Where are they located relative to *? |
| 17:09.20 | Mark22 | all phones are located in the same room (as this was a test) |
| 17:09.30 | Mark22 | or what do you mean? |
| 17:15.18 | *** join/#asterisk clyrrad (~IceChat7@CPE00270d2d7b09-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:21.34 | [TK]D-Fender | Mark22: Same lan... thats something.. |
| 17:21.47 | [TK]D-Fender | Mark22: [Apr 11 12:50:32] -- <SIP/1012-0000189b> Playing 'pbx-transfer.ulaw' (language 'en') <-- SIP/1012 is starting the transfer |
| 17:22.08 | [TK]D-Fender | Mark22: [Apr 11 12:50:35] -- Executing [1012@sip-internal:2] Dial("Local/1012@sip-internal-0ef1;2", "SIP/1012,,Tt") in new stack <- why is it calling ITSELF? |
| 17:24.33 | *** join/#asterisk dimitripietro (~gagnon.da@24-226-230-221.al.cgocable.ca) |
| 17:24.55 | Mark22 | trying again to look for what happens |
| 17:25.57 | dimitripietro | I need a windows software to generate popup for a CRM for a call center, anybody could help me ? |
| 17:26.20 | dimitripietro | I need to be able to set the url in the dialplan |
| 17:27.55 | Mark22 | http://yourpaste.net/5140/ << new call, but still with the same problem |
| 17:28.31 | Mark22 | and now it should give good data to check |
| 17:28.40 | Mark22 | with not to much noise I hope |
| 17:34.23 | [TK]D-Fender | Mark22: Same call with SIP DEBUG please |
| 17:35.09 | Mark22 | ok, however that will generate noise |
| 17:35.31 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 17:42.08 | Mark22 | http://yourpaste.net/5141/ |
| 17:43.19 | *** join/#asterisk theBruno (~theBruno@128-177-27-78.ip.openhosting.com) |
| 17:43.28 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 17:47.34 | [TK]D-Fender | Mark22: [Apr 11 19:35:49] VERBOSE[6048] chan_sip.c: [Apr 11 19:35:49] Reliably Transmitting (NAT) to 82.75.197.158:5060: <- 110 dosn't look like local LAN to me |
| 17:48.20 | Mark22 | the server isn't on the lan, that is in a datacenter |
| 17:48.25 | [TK]D-Fender | .... |
| 17:49.01 | [TK]D-Fender | [13:08]<[TK]D-Fender>Mark22: Where are they located relative to *? |
| 17:49.03 | [TK]D-Fender | [13:09]<Mark22>all phones are located in the same room (as this was a test) |
| 17:49.36 | [TK]D-Fender | Mark22: pastebin your sip.conf masking only passwords. |
| 17:53.57 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 17:54.42 | joako | Is there a way to determine in the dialplan if a call to a SIP device did not go through because the phone is in DND mode? |
| 17:56.22 | *** join/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-6-41.unitymediagroup.de) |
| 17:57.26 | Mark22 | http://yourpaste.net/5142/ (passwords are replaced with qwerty, that is not a real password) |
| 17:57.52 | ArtemMakhutov | Can someone help me building a asterisk 1.6.2.6 debian package? I am trying to build it with dpkg-buildpackage, but I am getting a undefined reference to `ast_log'. |
| 17:59.21 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 18:01.09 | *** join/#asterisk aschneiderg (~aschneide@190.24.67.105) |
| 18:02.37 | [TK]D-Fender | Mark22: REGISTER statements must occur after everything else in [general], not inbetween. Fix this, and remove all commented out lines |
| 18:04.24 | aschneiderg | Hello people. I'm doing a new install in a CentOS server. http://www.asterisk.org/downloads/yum. But, I'm getting this: http://pastebin.centos.org/32330 Any help will be appreciated. |
| 18:05.00 | aschneiderg | System is up to date. |
| 18:05.05 | Mark22 | [TK]D-Fender: didn't know that, fixing that now |
| 18:06.03 | [TK]D-Fender | aschneiderg: We can't even see what you ar calling |
| 18:08.34 | aschneiderg | [TK]D-Fender: I do not understand you. You can not see my pastebin? |
| 18:09.08 | [TK]D-Fender | asYou didn't include you call to YUM se we could even see what you asked it for <- |
| 18:09.21 | [TK]D-Fender | aschneiderg: You didn't include you call to YUM se we could even see what you asked it for <- |
| 18:09.33 | aschneiderg | hmm, sorry. |
| 18:09.44 | aschneiderg | "yum install asterisk16 asterisk16-configs asterisk16-voicemail dahdi-linux dahd" |
| 18:09.47 | aschneiderg | that's the line |
| 18:10.19 | aschneiderg | I tried one by one, but neither works |
| 18:10.29 | [TK]D-Fender | aschneiderg: ---> Package asterisk14-core.i386 0:1.4.30-2_centos5 set to be updated <-- UPDATED? Looks like it is installed. You'll have to remove all the old packages first |
| 18:10.37 | [TK]D-Fender | aschneiderg: Back up your configs, etc prior |
| 18:10.46 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 18:10.56 | aschneiderg | rpm -q says me not. |
| 18:11.20 | dewinda | I cant recive caller id to asterisk can someone help me on this issue http://yourpaste.net/5143/ |
| 18:12.40 | aschneiderg | [TK]D-Fender: wait, as it shows in http://www.asterisk.org/downloads/yum my centos-asterisk.repo is having both the tested and the current branch. |
| 18:12.47 | aschneiderg | May be is something there? |
| 18:12.59 | *** part/#asterisk rethus (~contio@p5087546D.dip.t-dialin.net) |
| 18:13.16 | [TK]D-Fender | aschneiderg: Go prove what * packages are already installed |
| 18:13.17 | Mark22 | [TK]D-Fender: http://yourpaste.net/5144/ << after changing it and removing the comments (but changing the position for register didn't fix my problem) |
| 18:13.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:14.06 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 18:15.01 | [TK]D-Fender | Mark22: Ok, don't see the issue yet. Upgrade immediately (plenty of reason to as it is) and retest |
| 18:16.01 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 18:16.16 | aschneiderg | <PROTECTED> |
| 18:16.40 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:17.58 | [TK]D-Fender | aschneiderg: yum clean all |
| 18:18.12 | [TK]D-Fender | aschneiderg: then install dahdi first, then asterisk16 |
| 18:18.45 | aschneiderg | ok, thanks. I will report results, sr. |
| 18:20.37 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 18:26.05 | b14ck | sup everyone |
| 18:30.11 | Naikrovek | hiya b14ck |
| 18:30.18 | Naikrovek | havent seen you in #trixbox for a while |
| 18:30.35 | b14ck | Oh shit. |
| 18:30.44 | b14ck | I just formatted, forgot to put #trixbox back into autojoin D: |
| 18:30.48 | Naikrovek | hehe |
| 18:31.01 | b14ck | fixed :) |
| 18:31.06 | Naikrovek | you don't work at fonality anymore, it's forgiven |
| 18:31.21 | b14ck | =0 |
| 18:31.24 | b14ck | *=p |
| 18:31.33 | b14ck | My new company is pretty fun so far. |
| 18:31.39 | b14ck | Been doing a lot of neat stuff. |
| 18:31.42 | Naikrovek | nice |
| 18:34.34 | aschneiderg | Nup, dahdi alone is not avaliable or aliased to any package. I installed dahdi-linux and asterisk16-dahdi but "yum install asterisk16" gives the same output. Searching for dahdi at the repos gives this: http://pastebin.centos.org/32333 |
| 18:37.48 | [TK]D-Fender | aschneiderg: asterisk16-dahdi seems to be what you want |
| 18:39.39 | aschneiderg | I tought that. Is installed. but, wait... that's asterisk! |
| 18:40.05 | aschneiderg | I ran asterisk -vvvgci and it puts it: Can't find indications config file indications.conf. |
| 18:40.18 | aschneiderg | that means is installed but not configured, right? |
| 18:44.25 | aschneiderg | nup, I need another stuff, sorry; first time on asterisk |
| 18:44.54 | [TK]D-Fender | aschneiderg: it means you have * installed already. You just said you didn't |
| 18:46.52 | aschneiderg | yup, I installed it. I did installed asterisk16configs and running asterisk -vvvgci takes me to a CLI. |
| 18:47.00 | aschneiderg | Man, that's nifty |
| 18:51.19 | aschneiderg | [TK]D-Fender: Thank you, very much for you time, pal. |
| 18:51.51 | aschneiderg | Do you guys know where should I go now? |
| 18:53.03 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:59.36 | *** join/#asterisk boch (~fran@190.51.69.138) |
| 19:00.08 | [TK]D-Fender | ~book |
| 19:00.09 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:00.11 | [TK]D-Fender | aschneiderg: ^^^^ |
| 19:00.45 | TJ^ | ok we all know how there are FXO cards... is there a device/card/adapter that allows you to use a mobile phone SIM card? |
| 19:01.27 | [TK]D-Fender | TJ^: many |
| 19:01.40 | TJ^ | cool |
| 19:01.40 | [TK]D-Fender | TJ^: Junhanns, VoiceBlue, etc make several |
| 19:01.55 | [TK]D-Fender | TJ^: there are many cell-phones you can use directly with * via Bluetooth |
| 19:02.08 | boch | Hi all, can you help me with a TDM400P FXO card? all my dials trough this card are being answered immediately after dial |
| 19:02.32 | [TK]D-Fender | boshthis is the standard for analog FXO channels. |
| 19:02.53 | [TK]D-Fender | boch: the alternative is using "callprogress=yes" which is sysnonymous with "disconnectmycallsatrandom=yes" |
| 19:04.49 | boch | [TK]D-Fender, thanks and where do i change that line ? |
| 19:05.08 | [TK]D-Fender | boch: in your channel definition |
| 19:05.51 | boch | do i have to reload something after mod? |
| 19:06.12 | [TK]D-Fender | boch: restart * or reload the channel driver |
| 19:06.23 | boch | [TK]D-Fender, okey thank you |
| 19:06.24 | aschneiderg | Good. see you in a while. |
| 19:07.30 | *** join/#asterisk kartik (~koolkarti@117.199.115.123) |
| 19:07.35 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:07.53 | *** join/#asterisk lcvleo (~chatzilla@189.60.207.46) |
| 19:08.48 | lcvleo | Hello everyone!!! ;) |
| 19:09.04 | lcvleo | I need some help from you all! |
| 19:09.06 | boch | [TK]D-Fender, i added that line to the four channels and the problem remains. My problem is when dialing out trough this ports |
| 19:09.30 | lcvleo | I use CentOS 5.4 64-bit + Asterisk 1.6.2.6 + FreePBX 2.7 |
| 19:09.52 | lcvleo | Caller Id stops to work when I place calls from my internal DAHDI extesions. |
| 19:10.36 | lcvleo | For example, when I call the Voicemail by *97, the internal DAHDI extension caller Id isn't identificated. |
| 19:10.56 | ChannelZ | Hurray for FreePBX! |
| 19:11.09 | *** join/#asterisk diegomad (mad@190.159.86.86) |
| 19:11.33 | lcvleo | When I place a call to an external number from an internal DAHDI and after that I take a look at the CDR, this don't shows who placed that call. |
| 19:11.57 | [TK]D-Fender | boch: And I see nothing that tells me you did it right |
| 19:12.15 | lcvleo | Someone has any ideia? |
| 19:12.19 | *** join/#asterisk rubbs (~rubbs@cpe-71-72-56-140.neo.res.rr.com) |
| 19:12.21 | boch | [TK]D-Fender, i added the same line to general and worked, thanks |
| 19:12.41 | [TK]D-Fender | boch: There is no [general] for your channel driver |
| 19:13.19 | boch | i see, now i understand the thing |
| 19:15.43 | clyrrad | Is there a v3 of THE BOOK that covers 1.6? |
| 19:15.54 | TJ^ | [TK]D-Fender gsm to SIP is expensive man |
| 19:16.19 | *** join/#asterisk lcvleo (~chatzilla@189.60.207.46) |
| 19:16.20 | [TK]D-Fender | TJ^: there are other makers, and as I said... you can use several cell phones directly |
| 19:16.26 | [TK]D-Fender | clyrrad: Not yes |
| 19:16.56 | clyrrad | thanks, do we know when to expect it? |
| 19:17.57 | leifmadsen | clyrrad: there won't be a book for 1.6.x |
| 19:18.04 | leifmadsen | clyrrad: next book will cover 1.8 |
| 19:18.13 | leifmadsen | don't expect to see it until about November of 2010 |
| 19:18.26 | clyrrad | thanks leifmadsen |
| 19:18.46 | clyrrad | I am also getting conflicting reports on if 1.6 is production ready or not, can anyone clairfy this? |
| 19:20.12 | dimitripietro | I'm using Asterisk 1.6.0.26 and 1.6.2.6 without any problem in multiple install |
| 19:20.22 | leifmadsen | that is not a question that has an answer |
| 19:20.42 | leifmadsen | "did the version of Asterisk you're testing pass all your tests? then perhaps it is ready" |
| 19:20.55 | leifmadsen | it depends what you're doing with it, etc... |
| 19:21.13 | leifmadsen | fwiw I have a pre-1.6.2.0 system in production |
| 19:21.36 | clyrrad | leifmadsen: well what i mean is, 1.4 is stable / production ready, can the same be said for 1.6 or is it still going thru the phases. I know I would need to test it and all that fun stuff, I am just curious if its safe enough for production enviorments yet |
| 19:21.56 | *** part/#asterisk mhaddog (~mhaddog@245.Red-88-25-254.staticIP.rima-tde.net) |
| 19:22.15 | lcvleo | anyone know how can I get NBS sftware to compile asterisk with support to chan_nbs ? |
| 19:22.16 | leifmadsen | first, 1.6 is ambiguous -- 1.6.0 vs 1.6.1 vs 1.6.2 are all separate branches. Second, it depends what you mean "production ready". How do you define that? |
| 19:23.16 | leifmadsen | clyrrad: like I said -- "is it production ready" is not a question that has an answer. There are 100s of different scenarios to determine what production ready means to you. |
| 19:23.32 | leifmadsen | It's like asking "how many calls can my sever handle?" |
| 19:23.43 | leifmadsen | the variables to determine the answer to that are vast |
| 19:23.49 | clyrrad | well by 1.6 I meant the latest release / most current one. I define Production Ready as - its safe enough to deploy for real every day phone service - whereby you can expect it to run with out many crashes or things "to be fixed" - recall when the 1.6 RC1 RC2 etc - it was said "dont run in prodcution" - so thats what I am getting at |
| 19:24.46 | leifmadsen | clyrrad: like I said -- 1.6. is ambiguous. The versions for 1.6.0.x vs. 1.6.1.x are the equivalent of 1.2 vs 1.4 |
| 19:25.03 | leifmadsen | the latest 1.6.x versions are 1.6.0.26, 1.6.1.18, and 1.6.2.6 |
| 19:25.20 | clyrrad | I see, did not know there was such a huge difference between the 1.6 versions |
| 19:25.27 | clyrrad | i've kept everything on 1.4 so far |
| 19:25.33 | leifmadsen | ~asterisk-versions |
| 19:25.33 | infobot | well, asterisk-versions is http://www.asterisk.org/asterisk-versions |
| 19:25.37 | leifmadsen | ~asterisk-versioning |
| 19:25.37 | infobot | hmm... asterisk-versioning is http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 19:25.52 | leifmadsen | those are links with information you should read then |
| 19:26.01 | clyrrad | perfect, thanks |
| 19:26.15 | TJ^ | [TK]D-Fender u used a normal mobile as a trunk? |
| 19:26.48 | leifmadsen | TJ^: you mean like with chan_mobile? |
| 19:26.56 | [TK]D-Fender | TJ^: personally no, but many have. |
| 19:27.06 | [TK]D-Fender | leifmadsen: yes, thats what I was suggesting for him |
| 19:27.06 | TJ^ | leifmadsen yeah |
| 19:27.07 | leifmadsen | I've tested it once :) |
| 19:27.11 | leifmadsen | it worked |
| 19:27.12 | TJ^ | was it good? |
| 19:27.28 | leifmadsen | define "good" |
| 19:27.40 | TJ^ | would u put it in a production environment? |
| 19:27.56 | leifmadsen | I don't understand how you would put that into "production" |
| 19:28.06 | leifmadsen | like... would I use it to run my business from? no. |
| 19:28.14 | leifmadsen | it's a cell phone connection via bluetooth... |
| 19:28.20 | leifmadsen | not exactly my definition of a reliable link |
| 19:28.38 | leifmadsen | did it work and is it a neat idea for a home PBX system? sure |
| 19:29.03 | TJ^ | cool |
| 19:29.05 | clyrrad | looks like it may be worth while to skip 1.6 completely and wait for 1.8 |
| 19:29.48 | leifmadsen | s/1.6/1.6.x/ |
| 19:30.06 | clyrrad | lol |
| 19:30.23 | clyrrad | bad habbit huh? |
| 19:30.28 | [TK]D-Fender | TJ^do you have noticable cost reason to look at that appropach? |
| 19:30.38 | [TK]D-Fender | approach* |
| 19:30.40 | leifmadsen | yes, you need to think of the various 1.6.x branches as separate versions |
| 19:30.46 | TJ^ | want to cut costs to mobile phones |
| 19:30.51 | clyrrad | leifmadsen: got it |
| 19:30.54 | [TK]D-Fender | tj^ go for it then. |
| 19:31.11 | leifmadsen | TJ^: so where you're from calling from a mobile phone to another mobile phone is cheaper than calling via other methods? |
| 19:31.21 | TJ^ | exactly |
| 19:31.27 | leifmadsen | interesting :) |
| 19:31.27 | clyrrad | leifmadsen: those links were helpful thanks again |
| 19:31.30 | leifmadsen | np |
| 19:37.56 | *** join/#asterisk Z_God (~julius@wlan228007.mobiel.utwente.nl) |
| 19:39.02 | TJ^ | ud think someone would have some sort of entry level SIM card to SIP adapter |
| 19:39.11 | TJ^ | something like the linksys PAPT adapters |
| 19:43.22 | *** join/#asterisk smooth_penguin (~smoove@59.95.5.17) |
| 19:48.48 | [TK]D-Fender | TJ^: not enough of a market to have much by way of "entry-level" vs anything else |
| 20:03.35 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 20:05.24 | *** part/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-6-41.unitymediagroup.de) |
| 20:05.29 | *** join/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-6-41.unitymediagroup.de) |
| 20:18.00 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 20:18.35 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:33.09 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 20:35.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:39.47 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:40.25 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 20:41.16 | hardwire | does the funky chicken |
| 20:42.03 | ChannelZ | fries the funky chicken |
| 20:42.10 | hardwire | sues |
| 20:46.00 | [TK]D-Fender | eats |
| 20:52.02 | *** join/#asterisk Skeeter- (~william@c75.154.173-9.clta.globetrotter.net) |
| 21:05.39 | *** join/#asterisk elzid (~IceChat7@host81-143-42-174.in-addr.btopenworld.com) |
| 21:05.58 | elzid | guys - anyone play with asterisk 1.6 and web-meetme 4? |
| 21:12.19 | [TK]D-Fender | elzid: 1.6.what? |
| 21:12.32 | [TK]D-Fender | elzid: Have you? |
| 21:13.45 | *** join/#asterisk TimeRider (steve@5ac7b39b.bb.sky.com) |
| 21:14.27 | hardwire | web-meetme 4 |
| 21:14.29 | hardwire | hmm |
| 21:14.58 | elzid | sorry guys - just getting version of asterisk - and yes, I have but have been hurdling many issues along the way |
| 21:15.11 | elzid | latest one is the recording feature |
| 21:15.12 | hardwire | another areski.net creation? |
| 21:15.50 | elzid | it doesn't record, complaining of format issues - it cant seem to find correct recording file format I presume |
| 21:16.29 | elzid | * version 1.6.2.5 |
| 21:17.13 | [TK]D-Fender | elzid: What do the authors say about support for that version? |
| 21:18.05 | elzid | not really sure - I think I got the impression of thumbs up from somewhere - Ill see if I can confirm |
| 21:18.40 | elzid | but I see a few people in the forums talking about 1.6 and web-meetme 4 so Im not alone |
| 21:19.59 | elzid | currently Im recording via mixmonitor but because thats the preceeding entry to meetme() in ext.conf, everything is recorded including prompt to enter pina dn pin entry attempts etc |
| 21:21.34 | elzid | [Apr 8 14:46:06] WARNING[18828]: file.c:1160 ast_writefile: No such format '' |
| 21:22.02 | [TK]D-Fender | elzid: Perhaps you should look at the actual application call VS its instructions |
| 21:22.17 | elzid | anyone know where the file format is read from? |
| 21:22.33 | elzid | <[TK]D-Fender>: sorry - what do you mean? |
| 21:22.51 | [TK]D-Fender | elzid: * CLI <- Go look at how it is being called, vs how * expects it to be called |
| 21:24.51 | elzid | <[TK]D-Fender> : this is what I see on the CLI at point of recording, but nothing happens: > Starting recording of MeetMe Conference 12990 into file ... |
| 21:25.02 | elzid | <[TK]D-Fender>: then I see the error printed above |
| 21:25.23 | [TK]D-Fender | elzid: And I see nothing. PASTEBIN is your friend. |
| 21:25.25 | [TK]D-Fender | ~pb |
| 21:25.26 | 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. |
| 21:25.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 21:26.20 | *** join/#asterisk farkus (~chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:27.19 | elzid | <[TK]D-Fender>: I've been through the file.c code briefly and it seems that the file format along with file name etc to be recorded into is sent to a function, but for some reason the format is missing - now I dunno where to look for format definition - only place I found was voicemail.conf but I think this is unrelated? |
| 21:27.29 | elzid | <[TK]D-Fender>: sorry, Ill check pastebin |
| 21:28.00 | [TK]D-Fender | elzid: Unrelated. Look at the CALL. |
| 21:28.34 | *** join/#asterisk TimeRider (steve@5ac7b39b.bb.sky.com) |
| 21:33.14 | elzid | <[TK]D-Fender>: Doesn't seem to give much info but here it is... http://pastebin.com/WV6ixgHP |
| 21:35.55 | [TK]D-Fender | elzid: And what does your room definition say? |
| 21:39.12 | elzid | exten => *****,1,Answer |
| 21:39.12 | elzid | exten => *****,2,Wait(1) |
| 21:39.12 | elzid | exten => *****,3,Set(MEETME_RECORDINGFILE=/var/spool/asterisk/meetme/conf_recording-${EXTEN:}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M)}-${CALLERID(num)}) |
| 21:39.12 | elzid | exten => *****,4,MixMonitor(${MEETME_RECORDINGFILE}.wav) |
| 21:39.12 | elzid | exten => *****,5,MeetMe() |
| 21:39.13 | elzid | exten => *****,6,Hangup |
| 21:39.22 | [TK]D-Fender | elzid: -- Executing [44*******@default:3] MeetMe("SIP/*******-00000002", "") in new stack <-- this is not saying that the conference is to be recorded |
| 21:39.38 | [TK]D-Fender | elzid: PASTEBIN that next time, do not flood in here |
| 21:39.44 | elzid | sorry |
| 21:40.12 | [TK]D-Fender | elzid: and that code does not match the CLI output you just showed me |
| 21:40.20 | *** join/#asterisk hipitihop (~denis@203-221-91-86.dynamic.reverse.flexinet.com.au) |
| 21:40.32 | elzid | the web-meetme app frontend gui is where the rooms are defined - they're stored in mysql db |
| 21:40.58 | elzid | odbc reads entries and validates room number with pins |
| 21:41.21 | [TK]D-Fender | elzid: that spammed code doesn't match the call. Why does it matter? |
| 21:42.21 | elzid | when the meetme app is called in step 5, the web-meetme code kicks in - let me get some more info into pastebin |
| 21:46.09 | elzid | extconfig.conf: I have "meetme => odbc,meetme,booking" |
| 21:48.03 | tvaughn | anyone know what I can open the asterisk sounds with? |
| 21:48.22 | elzid | I have odbc connection defined for meetme - and past/current/future conference room/pins are in the booking table |
| 21:49.07 | elzid | <[TK]D-Fender>: does this make sense? |
| 21:49.30 | [TK]D-Fender | elzid: You are setting ${MEETME_RECORDINGFILE} but not ${MEETME_RECORDINGFILE} |
| 21:49.45 | *** part/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-6-41.unitymediagroup.de) |
| 21:50.15 | [TK]D-Fender | elzid: Also the way you call meetme doesn't imply it will record normally anyway. I see a call to MixMonitor, but that isn't even executed (nor the preceeding bit). Sorry, the pieces don't add up |
| 21:50.17 | elzid | <[TK]D-Fender>: sorry? |
| 21:50.35 | elzid | ok - let me run again |
| 21:50.49 | [TK]D-Fender | ${MEETME_RECORDINGFORMAT} rather |
| 21:51.47 | elzid | yes - but the recording via mixmonitor works fine - it defaults to wav |
| 21:53.05 | elzid | but why doesn't it work for web-meetme built in record method, rather, it complains of format not found - so with my current setup I'd expect two files - 1 via mixmonitor, other via web-meetme recrding function - let me get new debug output to pastebin, I think the last pastebin must've been before I added mixmonitor! sorry |
| 21:54.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:57.50 | elzid | <[TK]D-Fender>: sorry - it was an old extract - here is new one: http://pastebin.com/08kPyW8e |
| 21:59.01 | [TK]D-Fender | [17:50]<[TK]D-Fender>${MEETME_RECORDINGFORMAT} rather <- set this |
| 21:59.59 | elzid | ok - I will try this now |
| 22:03.58 | elzid | <[TK]D-Fender>: Using similar syntax to my previous set, I set this MEETME_RECORDINGFORMAT=wav |
| 22:04.06 | elzid | <[TK]D-Fender>: but still got the error |
| 22:04.19 | elzid | <[TK]D-Fender>: does meetme.conf have a format switch/directive? |
| 22:04.58 | elzid | <[TK]D-Fender>: file.c is picking up format from somewhere, and I dunno where - seems not this variable |
| 22:05.01 | [TK]D-Fender | elzid: No. |
| 22:05.28 | [TK]D-Fender | elzid: Normally you should not be using mixmonitor for this either. There is already a flag to pass MeetMe for this |
| 22:05.55 | talntid | maybe i'm retarded... but any idea why this is giving me trouble? http://pastebin.com/KsG6pkes |
| 22:06.03 | elzid | <[TK]D-Fender>: yes - mixmonitor is just me trying to get conf recorded - it ideally shouldn't be there |
| 22:06.18 | elzid | <[TK]D-Fender>: perhaps meetme(,r) can be used somehow? |
| 22:06.22 | talntid | when dialing, it says the extension is not found - a test extension of "9999" below it plays weasels |
| 22:07.10 | talntid | and i could swear that last night, when i went to bed, it worked |
| 22:07.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:07.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:07.39 | [TK]D-Fender | elzid: Yes, following the apps instructions might help... |
| 22:08.19 | [TK]D-Fender | talntid: those are 3 random dialplan extensions. No mention of context, or where the call is actually looking. |
| 22:08.29 | elzid | <[TK]D-Fender>: I tried this earlier but it didnt succeed - maybe I got the syntax wrong - Ill try again |
| 22:08.44 | [TK]D-Fender | elzid: here you didn't even try it. |
| 22:09.39 | elzid | yes - I attempted previously (before resorting to mixmonitor) - will try again as Im not sure I got the syntax/flow right |
| 22:10.36 | elzid | I had this previously - does this look right to you? |
| 22:10.37 | elzid | ;exten => 44***,3(record),Set(MEETME_RECORDINGFILE=/var/spool/asterisk/meetme/conference_recording-${EPOCH}-${USER}) |
| 22:10.37 | elzid | ;exten => 44***,4,MeetMe(,r) |
| 22:10.37 | talntid | [TK]D-Fender, the sip phones are registered to the context "phones" - those are within that context. the sip phones can dial "9999" and get tt-weasels which is right below those dialplan extensions - the console says those extensions are not found, when I dial them, so I am thinking I have something wrong with them? They are just supposed to support outbound dialing |
| 22:10.41 | *** join/#asterisk pentanol (~Unknown@91.195.60.231) |
| 22:10.53 | elzid | ;exten => 44***,3(record),Set(MEETME_RECORDINGFILE=/var/spool/asterisk/meetme/conference_recording-${EPOCH}-${USER}) |
| 22:10.53 | elzid | ;exten => 44***,4,MeetMe(,r) |
| 22:10.55 | [TK]D-Fender | talntid: I see nothing to substantiate your problem. |
| 22:11.06 | [TK]D-Fender | elzid: STOP FLOODING. |
| 22:11.18 | talntid | roger that. me either - just figured i'd see. I'll keep troubleshooting :P |
| 22:11.20 | elzid | sorry - I thought it was 2 lines! |
| 22:11.47 | elzid | apologies |
| 22:16.02 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:18.20 | [TK]D-Fender | talntid: Also... you never even bothered to tell us what you dialed. |
| 22:18.25 | talntid | ok, actually it's odd... |
| 22:18.46 | talntid | that's actually a good question [TK]D-Fender. I dialed "509 850 1422" but also "850 1422" |
| 22:18.55 | talntid | if i dial that, it gives the error "no extension" |
| 22:19.07 | talntid | but if i dial "280 6106" it works, as well as any variation |
| 22:19.08 | [TK]D-Fender | talntid: Go read the PATTERN you are matching |
| 22:19.48 | [TK]D-Fender | talntid: Because it should fail |
| 22:20.26 | talntid | yeah, thanks - i actually didn't know that... x=0-9, z=1-2, n=2-9 |
| 22:20.33 | talntid | i'm a dummb |
| 22:20.36 | talntid | dummy :P |
| 22:21.55 | talntid | yep. works now |
| 22:23.03 | [TK]D-Fender | talntid: good |
| 22:23.24 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 22:38.19 | vlt | Hello. Any idea what leads to a "SIP/2.0 488 Not Acceptable Here" answer to a T.38 INVITE? Both endpoints (Grandstream ATA and Patton SmartNode) so support T.38. |
| 22:38.46 | vlt | (or rather _should_ support it) |
| 22:41.37 | [TK]D-Fender | vlt: we'd have to see the full comminucation |
| 22:46.45 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:48.39 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 22:59.24 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:03.11 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 23:05.36 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 23:08.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:08.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 23:10.20 | vlt | [TK]D-Fender: http://www.pastebin.org/147507 |
| 23:10.42 | vlt | That's what wireshark can tell me about the SIP communication |
| 23:11.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:17.45 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:40.42 | *** join/#asterisk aschneiderg (~aschneide@190.27.105.83) |
| 23:42.20 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:43.08 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 23:46.12 | *** join/#asterisk Faithful (~Faithful@121.91.178.93) |
| 23:48.48 | *** join/#asterisk devoid (~devnull@unaffiliated/devemo) |
| 23:51.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:52.50 | *** join/#asterisk variable_office (~variable_@10.61.pool.border1.ch1.iswan.net) |
| 23:53.02 | variable_office | is there a way to run a command on call end only? |
| 23:55.06 | hardwire | variable_office: the h extension |
| 23:55.13 | hardwire | or using the g flag in dial |
| 23:56.28 | hardwire | variable_office: roger? |
| 23:56.50 | variable_office | hardwire, thanks, I am going to try that right now |
| 23:57.03 | hardwire | as long as h is in the same context as the dial.. you're golden |
| 23:57.23 | hardwire | lots of stuff shows up after the h extension is called.. try out h,1,Dumpchan() before you get too far into it |
| 23:57.29 | hardwire | err.. DumpChan |
