| 00:00.03 | leifmadsen | pdtpatrick1: then no, the functionality doesn't exist |
| 00:00.10 | leifmadsen | you'd need to write a script to trigger via AMI |
| 00:00.26 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-132-115.chyn.qwest.net) |
| 00:00.28 | *** join/#asterisk corretico (~luis@200.12.40.18) |
| 00:00.52 | pdtpatrick1 | oh i c |
| 00:01.01 | pdtpatrick1 | okay thanks will do that then |
| 00:02.18 | pdtpatrick1 | is this ur blog ? |
| 00:02.20 | pdtpatrick1 | http://leifmadsen.wordpress.com/2009/07/17/howto-getting-jabberxmpp-notifications-from-your-pbx/ |
| 00:02.28 | leifmadsen | yes it is |
| 00:02.43 | pdtpatrick1 | haha cool stuff. |
| 00:02.53 | leifmadsen | :) |
| 00:04.15 | pdtpatrick1 | mine => thinkfirstblinksecond.com |
| 00:04.24 | pdtpatrick1 | anyway .. the script .. that would have to be agi ? |
| 00:04.40 | leifmadsen | no, AGI is triggered from the dialplan |
| 00:04.45 | leifmadsen | AMI is triggered from an external script |
| 00:05.13 | pdtpatrick1 | okay.. going to read up on it |
| 00:07.13 | pdtpatrick1 | but in the dialplan is where jabber is checking for the user's status and setting 1 = online etc. |
| 00:07.26 | pdtpatrick1 | right? so shouldn't then the script run from there? |
| 00:07.41 | leifmadsen | pdtpatrick1: you wanted to dial an extension through jabber |
| 00:07.51 | leifmadsen | which to me, means you want to initiate the connection via jabber |
| 00:08.08 | leifmadsen | which means, you need to actually trigger that from an external script that triggers the call via the AMI |
| 00:08.28 | pdtpatrick1 | oh sorry .. i guess i thought of another question and never asked it |
| 00:08.33 | leifmadsen | scripts talk to AMI, Asterisk executes scripts via AGI |
| 00:08.36 | pdtpatrick1 | I'm not trying to set the status of the user |
| 00:08.47 | leifmadsen | I'm not talking about setting the status of the user at all |
| 00:08.52 | pdtpatrick1 | so for instance, I call someone - if they pick up, i would like to set their status to say "On the phone" |
| 00:09.00 | pdtpatrick1 | right i just realized that now |
| 00:09.01 | leifmadsen | <pdtpatrick1> Question .. does anyone know of a bot that will permit you to dial an extension through jabber? |
| 00:09.05 | leifmadsen | this is what I'm try to answer |
| 00:09.24 | leifmadsen | I suggest asking the question you want answered :) |
| 00:09.27 | pdtpatrick1 | yeah sorry about that .. i was researching that and thought of another question but never asked it and assumed we were talking about that |
| 00:10.03 | pdtpatrick1 | Question - what's a good way to have asterisk set the user's status in Jabber to on the phone when they answer a call |
| 00:10.25 | leifmadsen | I think there is a JABBER_STATUS() dialplan function |
| 00:11.54 | pdtpatrick1 | isn't that just for retrieving the status.. at least that's what i got when i read and tested it |
| 00:12.02 | leifmadsen | it may not set the status |
| 00:12.24 | leifmadsen | if you need to set it programmatically from another script outside of Asterisk, then yes, AGI is what you want to use |
| 00:14.48 | pdtpatrick1 | do u have an ami example on ur blog or something i can follow? |
| 00:15.10 | leifmadsen | I do not -- I've never set that up before |
| 00:15.13 | leifmadsen | sorry |
| 00:15.23 | leifmadsen | AMI: script --> Asterisk |
| 00:15.31 | leifmadsen | AGI: Asterisk --> dialplan --> script |
| 00:15.53 | beta2k | Anyone around willing to look at my polycom configs or share theirs? I'm not sure what I'm missing reading the docs... |
| 00:16.21 | Naikrovek | how are you configuring them? tftp ftp |
| 00:16.58 | beta2k | tftp |
| 00:17.07 | beta2k | Got them to download the new FW and bootrom |
| 00:17.18 | beta2k | but I'm missing something when I'm setting up the configs |
| 00:19.27 | beta2k | Just poking around on the phone, looks like I'm missing the auth settings |
| 00:21.11 | Naikrovek | maybe. in one of my phone configs there's reg.1.x which has the name and the extension and the label and the password and blah blah blah |
| 00:21.55 | beta2k | Yeah that's all there |
| 00:22.24 | beta2k | But when I head down further to authentication I get the previous settings from flash |
| 00:24.08 | beta2k | Hrm, reg1.auth.userId="520" is in there so it should be right.... |
| 00:33.08 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 00:33.08 | *** join/#asterisk dre (dre@gateway/shell/xzibition.com/x-aagulhyylwfnqgjb) |
| 00:33.08 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 00:37.03 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:43.31 | beta2k | Maybe there's a problem with the new bootrom/app and asterisk... |
| 00:43.52 | beta2k | I turned on sip debugging and see it failing to register with bad auth |
| 00:44.05 | leifmadsen | what version? |
| 00:44.18 | beta2k | the userid is right, realm ok |
| 00:44.19 | leifmadsen | because I'm using the latest one (that I'm aware of) on about 50+ phones |
| 00:44.31 | beta2k | 3.2.5 |
| 00:44.32 | leifmadsen | does it try to register? if not, then the problem is your Polycom configuration |
| 00:44.44 | p3nguin | You have the "address" and the "auth ID" both set to the name of the device as configured in sip.conf? |
| 00:44.47 | beta2k | It tries to register, but fails with bad auth |
| 00:45.03 | leifmadsen | then you have a configuration problem still |
| 00:45.21 | leifmadsen | the authentication you're using isn't matching the peer that is matched by asterisk |
| 00:45.25 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:52.58 | beta2k | Here's my phone config file |
| 00:53.00 | beta2k | http://paste.pocoo.org/show/479562/ |
| 00:53.57 | beta2k | and sip.conf override, http://paste.pocoo.org/show/479564/ |
| 00:54.29 | beta2k | I can't see what I'm missing :) |
| 00:58.04 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 01:05.24 | Naikrovek | i'll take a squiz |
| 01:05.29 | Naikrovek | a look |
| 01:05.52 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:06.09 | Naikrovek | what's [mac].cfg look like |
| 01:06.20 | Naikrovek | beta2k: ping |
| 01:07.36 | Naikrovek | don't modify your sip.cfg, ever. |
| 01:08.00 | Naikrovek | revert all files to default. create a [mac].cfg file for a phone. 0004f22abcdef.cfg or whatever |
| 01:09.05 | Naikrovek | it will look like: ah crap i don't have my copies locally. it'll tell the phone(s) what to look for as far as other files go. it'll tell the phone where to log, where to find its directory, and where the override folder is. |
| 01:09.17 | Naikrovek | grr let me find an example |
| 01:11.11 | Naikrovek | i had all these on pastebin at one point, but i lost the bookmarks. |
| 01:12.27 | Naikrovek | afk a moment |
| 01:17.01 | *** join/#asterisk micols (~0x2AA7F64@2001:16d8:dd54::2000) |
| 01:33.39 | Naikrovek | argh why does that bulletin from polycom say NOT to use [mac].cfg files, but then the admin guide says TO use them... |
| 01:33.57 | Naikrovek | I'll buy a technical proofreader for $50, Pat. |
| 01:34.46 | Naikrovek | alright well i'm going to use the admin guide as my polycom config bible. i'm ignoring that bulletin. |
| 01:35.09 | Naikrovek | remove all your changes from all files. unzip the firmware again, overwriting sip.cfg and phone1.cfg. |
| 01:37.33 | tonsofpcs | anyone know much about h323? |
| 01:38.43 | Naikrovek | beta2k: create a .cfg for each phone, according to its mac address. here's an example for a phone with mac 0004f22abcdef: http://paste.pocoo.org/show/479588/ |
| 01:39.30 | p3nguin | I don't know if he's even here anymore. |
| 01:44.32 | Naikrovek | beta2k: then create phone1_0004f22abcdef.cfg, as referenced in the 0004f22abcdef.cfg file. it will look like this: http://paste.pocoo.org/show/479591/ fill in the appropriate details. |
| 01:45.14 | Naikrovek | you'll need another file for the server information, timezone (if not sent via DHCP) and such. I don't have my own example of that handy atm. Ask me tomorrow and I'll send it to you. |
| 02:02.15 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 02:02.21 | *** join/#asterisk saliak_ (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 02:05.13 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:14.03 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-132-115.chyn.qwest.net) |
| 02:15.21 | *** join/#asterisk jeffgus (~jeffgus@2001:470:f2eb:1::4) |
| 02:21.05 | *** join/#asterisk shido6 (~shido6@206.222.178.23) |
| 02:29.15 | Kalavera | hello I am trying to configure an asterisk with a common fxo line |
| 02:29.38 | Kalavera | but not sure how to proceed about trunks and DIDs |
| 02:37.36 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 02:39.15 | *** join/#asterisk fluppie (~fluppie@546BA16F.cm-12-4c.dynamic.ziggo.nl) |
| 02:39.18 | beta2k | Naikrovek: Ok, I'll try that :) I'd been ignoring all the tutorials saying to do that because the polycom docs said not to do it like that :) |
| 02:39.42 | Naikrovek | well the admin guide says do to that, and that's how I do it. |
| 02:39.47 | Naikrovek | and it works very well |
| 02:39.56 | Naikrovek | i have 120 phones all configured that way |
| 02:40.33 | Kalavera | Naikrovek: I do have some troubles making asterisk to work out |
| 02:41.53 | Kalavera | 1.- card's let is on, 2.- I can not see the channels by typing dahdi show channels |
| 02:42.09 | Naikrovek | i don't know anything about DAHDI devices |
| 02:42.14 | Kalavera | but I can see information by typing cat /proc/dahdi/1 |
| 02:42.38 | Naikrovek | there's a fella in #freepbx, [tk]d-fender, who will be able to tell you how to solve this. |
| 02:42.56 | Kalavera | ok thank you |
| 02:43.14 | Naikrovek | Night time in the US is a slow time in this channel |
| 02:43.38 | Naikrovek | if you ask again in 12 hours you'll get your answers |
| 02:51.45 | beta2k | Cripes, night time in the US is when I'm getting going on this stuff |
| 02:53.12 | *** part/#asterisk fluppie (~fluppie@546BA16F.cm-12-4c.dynamic.ziggo.nl) |
| 02:54.13 | Kalavera | beta2k: lol |
| 02:54.31 | Kalavera | beta2k: me too |
| 02:56.09 | *** join/#asterisk bradleyprice (~bradleypr@75-121-65-143.dyn.centurytel.net) |
| 02:56.43 | Kalavera | I have troubles trying to make outgoing calls |
| 03:01.05 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 03:07.26 | ChannelZ | In what way |
| 03:08.17 | beta2k | Yay it works :) Thanks Naikrovek |
| 03:08.44 | beta2k | Kalavera: Intermittent? |
| 03:23.57 | Kalavera | it says that all the lines are busy , seems becaused some missconfiguration but it is my first day if not the first hours trying to configure asterisk |
| 03:24.00 | Kalavera | with freepbx |
| 03:24.26 | ChannelZ | ugh |
| 03:24.33 | ChannelZ | can't help you much with freepbx |
| 03:24.40 | p3nguin | ~freepbx |
| 03:24.41 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:24.45 | p3nguin | ~book |
| 03:24.46 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 03:24.57 | Kalavera | anyways I don't think FreePBX touches dahdi files |
| 03:25.07 | ChannelZ | it touches everything |
| 03:25.14 | ChannelZ | it's like cancer |
| 03:25.24 | Kalavera | lol |
| 03:25.32 | ChannelZ | What is the FXO? |
| 03:25.50 | Kalavera | Span 1: WCTDM/0 "Wildcard AEX410" (MASTER) |
| 03:25.53 | Kalavera | four ports |
| 03:26.13 | Kalavera | another thing is that I am not sure how to identify which one is port one |
| 03:26.39 | Kalavera | except that unplugging and plugging the wire in both ends |
| 03:28.04 | ChannelZ | Do you have a softphone or something else setup too? |
| 03:28.19 | Kalavera | mmm I do have a Grandstream IP phone |
| 03:28.24 | Kalavera | and a Snom 300 |
| 03:28.31 | ChannelZ | working? |
| 03:28.36 | Kalavera | I can make calls from Grandstream to the snom |
| 03:28.49 | Kalavera | but can not make calls from Snom to Grandstream |
| 03:28.56 | Kalavera | it alBUT it did |
| 03:29.20 | Kalavera | must be something that I moved in the phone config |
| 03:29.33 | Kalavera | and what doesn't totally works is outgoing calls |
| 03:36.53 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:58.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:21.08 | *** join/#asterisk irroot (~irroot@41.53.66.3) |
| 04:29.38 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 04:48.40 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 05:14.40 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 05:25.01 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:32.19 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 05:32.21 | schmidts | good morning |
| 05:34.36 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 05:44.39 | ChannelZ | fnord! |
| 05:46.41 | *** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net) |
| 06:15.35 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:21.26 | *** join/#asterisk irroot (~irroot@197.169.192.33) |
| 06:37.08 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:44.39 | *** join/#asterisk atan (atan@unaffiliated/atan) |
| 06:46.49 | *** join/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 06:53.39 | *** join/#asterisk ocx (4d2adb52@gateway/web/freenode/ip.77.42.219.82) |
| 06:57.30 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 07:16.22 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:31.37 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:32.08 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:39.01 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 07:44.53 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 07:49.38 | *** join/#asterisk docid (~PISSS@69.17.142.138) |
| 07:49.38 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 07:49.38 | *** join/#asterisk Polis_ttt (~lasse@irc.mussla.se) |
| 07:49.38 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 07:49.38 | *** join/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net) |
| 07:49.38 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 07:49.38 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 07:49.38 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 07:49.38 | *** join/#asterisk CheBuzz (~CheBuzz@76.164.192.124) |
| 07:49.38 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 07:49.38 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 07:49.38 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 07:49.38 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 07:49.38 | *** join/#asterisk mtbf (~ewilded@195.117.148.48) |
| 07:49.38 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 07:49.38 | *** mode/#asterisk [+o pabelanger] by niven.freenode.net |
| 07:51.34 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 07:51.34 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 07:51.34 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 07:51.34 | *** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net) |
| 07:51.34 | *** join/#asterisk psilikon (~joel@rrcs-24-73-84-158.se.biz.rr.com) |
| 07:51.34 | *** join/#asterisk aglenday (~Impatient@59.167.161.74) |
| 07:51.34 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 07:51.34 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 07:51.34 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 07:51.34 | *** join/#asterisk beta2k (~Beta2K@d24-36-128-84.home1.cgocable.net) |
| 07:51.34 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 07:51.34 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 07:51.55 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 07:51.55 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 07:51.55 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 07:51.55 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 07:51.55 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 07:51.55 | *** join/#asterisk 16WAAETMD (~w7@freenode/staff/wikimedia.werdan7) |
| 07:51.55 | *** join/#asterisk master_of_master (~master_of@p57B5410F.dip.t-dialin.net) |
| 07:51.55 | *** join/#asterisk dr_ (~dr@83.166.214.174) |
| 07:51.55 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 07:51.55 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:51.55 | *** join/#asterisk piney0 (~piney@pool-70-111-58-192.nwrk.east.verizon.net) |
| 07:51.55 | *** join/#asterisk pdtpatrick1 (~pdtpdt@mainstwan.farheap.com) |
| 07:51.56 | *** join/#asterisk scatterp_ (~scatterp@02dfe731.bb.sky.com) |
| 07:51.56 | *** join/#asterisk Merlin (merlin@evendata.net) |
| 07:51.56 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 07:51.56 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 07:51.56 | *** join/#asterisk hardwire (~spencersr@12.17.188.86) |
| 07:51.56 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 07:51.56 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 07:51.56 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 07:51.56 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 07:51.59 | *** join/#asterisk nmjnb (~nmjnb@c-567e72d5.026-18-73746f23.cust.bredbandsbolaget.se) |
| 07:52.12 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 07:52.13 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 07:52.13 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 07:52.13 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 07:52.13 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 07:52.13 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 07:52.13 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 07:52.13 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 07:52.32 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 07:52.32 | *** join/#asterisk dr0ck (~dr0ck@nat/digium/x-uxvfrprthsebyzwt) |
| 07:52.32 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 07:52.32 | *** join/#asterisk dmz (~dmz@64.203.235.49.dyn-cm-pool-34.pool.hargray.net) |
| 07:52.32 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 07:52.32 | *** join/#asterisk atheos (~atheos@208.119.68.16) |
| 07:52.32 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 07:52.32 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 07:52.32 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 07:52.32 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 07:52.32 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 07:52.32 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 07:52.32 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 07:52.32 | *** join/#asterisk deever (~deever@static.172.68.46.78.clients.your-server.de) |
| 07:52.32 | *** mode/#asterisk [+o malcolmd] by niven.freenode.net |
| 07:52.45 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 07:52.45 | *** join/#asterisk dre (dre@gateway/shell/xzibition.com/x-aagulhyylwfnqgjb) |
| 07:52.45 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 07:54.07 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 07:54.45 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 07:54.45 | *** join/#asterisk nightwalk (~null@daimon.vixel.org) |
| 07:54.45 | *** join/#asterisk pushpop (~pushpop@pool-173-77-211-176.nycmny.fios.verizon.net) |
| 07:54.45 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.229.82.dsl.dyn.forthnet.gr) |
| 07:54.45 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 07:54.45 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-nzekxlawjquavfeg) |
| 07:54.45 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 07:54.45 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 07:54.45 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:54.45 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 07:54.46 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:54.46 | *** join/#asterisk tuxx- (~tuxx@2a02:2308::1c1:1337:1337) |
| 07:54.46 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 07:54.46 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:54.46 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 07:54.46 | *** mode/#asterisk [+o Qwell] by niven.freenode.net |
| 07:54.49 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 07:55.11 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:56.36 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 07:56.59 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 07:59.08 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es) |
| 08:02.01 | ruben23 | hi guys any idea or help i have a tmpfs on my asterisk server set liek this http://pastebin.com/fM3SK4AJ <------------but problme its getting full and cause an error on my recordings, any idea how to resolve the issue geting its full always and i always reboot to clear it out |
| 08:03.12 | kaldemar | ruben23: move the files elsewhere when it gets full |
| 08:04.50 | ruben23 | kaldemar:how to do that..? i mean last time its not getting full at all now its always coming up full |
| 08:04.58 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 08:05.39 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 08:06.44 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:07.37 | kaldemar | ruben23: last time? make something that checks the available space on the tmpfs and moves files out of it if it's getting full. |
| 08:08.18 | irroot | ruben23 you using tmpfs mount ?? that is a bad idea that is machine ram |
| 08:08.23 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:08.30 | irroot | oej o/ |
| 08:10.59 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:11.34 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 08:15.43 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 08:17.45 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 08:20.30 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:21.11 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:22.34 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 08:34.47 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 08:38.22 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 08:48.44 | *** join/#asterisk jkroon (~jkroon@dsl-242-10-94.telkomadsl.co.za) |
| 08:51.25 | jkroon | hi guys, when I've got a fromdomain= set up in a SIP peer, but if I channel originate SIP/peer/number application Echo asterisk still sends anonymous.invalid as the domain - is this intended behaviour or a bug? |
| 08:51.42 | jkroon | also, what's the difference between defaultuser and fromuser? |
| 08:55.47 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 08:58.15 | jkroon | nm @ defaultuser q - that's used in the absence of authuser / fromuser (according to a quick perusal of the code) |
| 09:01.35 | *** join/#asterisk beersingh (~tirveni@115.240.98.13) |
| 09:01.37 | irroot | jkroon there is some other stuff to be added |
| 09:01.54 | jkroon | ?? |
| 09:02.28 | *** join/#asterisk Ramaskez (~Ramaskez@ip-87-86-154-70.easynet.co.uk) |
| 09:02.34 | irroot | sendrpid=yes |
| 09:02.35 | irroot | trustrpid=yes |
| 09:04.41 | Ramaskez | Hi All, I have an asterisk 1.6.2.19 box that periodically stops accepting new calls and ramps the CPU up to 100% on all CPU's. Asterisk is still running and dosent crash however wont accept any new calls and wont hangup any old calls. If I do asterisk -rx "core show channels" then I get a list of channels but the command never completes and just halts. Any ideas how I will go about debuging this? |
| 09:09.25 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:10.36 | stix | is it the asterisk-process which uses 100% of the CPU? |
| 09:10.56 | jkroon | irroot, i don't even know what rpid is ... never needed to set that before. |
| 09:11.08 | Chainsaw | Morning jkroon. |
| 09:11.36 | jkroon | Ramaskez, a few bugs there that triggers that. I've filed bugs on issues.asterisk.org - you're welcome to trawl them and custom patch. |
| 09:11.41 | irroot | jkroon ^^^ got it know it works as daughter loves prank calling ala whackhead from auntie ... [fakecli] |
| 09:11.51 | jkroon | morning Chainsaw - don't usually see you here :) |
| 09:12.17 | Ramaskez | Thanks jkroon. Ill look in to it |
| 09:14.39 | jkroon | personally i just gave up on the 1.6.2.X branch - I can recommend moving to >1.8.5.0 - mostly it's a trivial migration. |
| 09:15.48 | jkroon | only one serious bug in 1.8.5.0 onwards that I'm aware of and that is an issue if a sip reload is still running whilst sip peers is executing. |
| 09:16.24 | jkroon | Chainsaw will likely correct me as he maintains the patches being merged on gentoo - I just send him the patches after submitting to issues.asterisk.org. |
| 09:16.25 | irroot | id say get 1.8.7.0-rc2 |
| 09:17.02 | jkroon | Chainsaw, perhaps we need to consider a 1.8.7.0-r2 for gentoo? i value irroot's opinion about as high as you do mine. |
| 09:17.11 | irroot | the fix for the timerfd hangup is in and some others |
| 09:17.19 | Chainsaw | jkroon: If you want that, I can get you that tomorrow. |
| 09:17.20 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 09:17.29 | Chainsaw | jkroon: On a train towards London right now, my connectivity isn't really up to CVS work. |
| 09:17.30 | jkroon | wasn't the timerfd issue fix reverted again? |
| 09:17.43 | jkroon | due to performance degradion in some or other unrelated subsystem? |
| 09:17.50 | irroot | that was 1.8.6 |
| 09:17.51 | jkroon | i won't touch it before sat anyway. |
| 09:18.01 | jkroon | irroot, cool beans - that is indeed VERY good news. |
| 09:18.09 | Chainsaw | jkroon: In that case, just send me a reminder e-mail to the linx domain please. |
| 09:18.18 | Chainsaw | jkroon: I won't be working this Friday, but should have time for it tomorrow. |
| 09:18.52 | irroot | there does not look to be anything in the pipe line for -rc3 but its early this could be 1.8.7 |
| 09:21.46 | irroot | jkroon RB1310 you commiting it ?? |
| 09:25.38 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 09:25.53 | jacc0 | good morning all :) |
| 09:29.05 | irroot | jacc0 you is what i was thinking what happening with RB1310 |
| 09:31.45 | jacc0 | what is with RB1310? |
| 09:32.33 | jacc0 | will it be intergrated? |
| 09:32.48 | jacc0 | in asterisk 10 perhaps? |
| 09:34.20 | jacc0 | Do I have to add anything ? |
| 09:36.49 | irroot | jacc0 for 10/trunk |
| 09:37.29 | *** join/#asterisk din3sh (29d4d17e@gateway/web/freenode/ip.41.212.209.126) |
| 09:37.52 | din3sh | hi all |
| 09:38.23 | din3sh | stupid question, how do I download a specitic trunk version? |
| 09:38.31 | din3sh | revision sorry* |
| 09:50.21 | *** join/#asterisk MariusAgon (~MariusAgo@84.15.44.250) |
| 09:52.54 | *** join/#asterisk ocx (c2ed8e14@gateway/web/freenode/ip.194.237.142.20) |
| 09:53.02 | irroot | din3sh "svn co -r .... |
| 09:55.16 | din3sh | -r is the revision number? |
| 09:59.30 | din3sh | thnx |
| 10:00.28 | *** join/#asterisk lister_ (~lister@scarda1.lnk.telstra.net) |
| 10:01.14 | irroot | din3sh yip pleasure |
| 10:21.46 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 10:35.44 | *** join/#asterisk bmg505 (~leon@196-209-44-105.dynamic.isadsl.co.za) |
| 10:38.12 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:38.41 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 10:43.12 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:56.22 | *** join/#asterisk kchehab (~kchehab@77.42.241.66) |
| 10:56.39 | kchehab | hi |
| 10:57.14 | kchehab | where i can ask a question relation to a2billing call cut for the last call duration |
| 10:57.26 | kchehab | related* |
| 11:05.36 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 11:10.54 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 11:17.07 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 11:35.17 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 11:36.02 | *** join/#asterisk ocx (4d2adb52@gateway/web/freenode/ip.77.42.219.82) |
| 11:43.16 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 11:44.17 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:51.09 | *** join/#asterisk neurosys (~neurosys@adsl-068-157-087-124.sip.bct.bellsouth.net) |
| 11:51.39 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 12:02.44 | *** join/#asterisk aglenday (~Impatient@59.167.161.74) |
| 12:07.14 | jacc0 | !ask |
| 12:07.18 | jacc0 | ãsk |
| 12:07.22 | jacc0 | ~ask |
| 12:07.22 | 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. |
| 12:14.44 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 12:20.52 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 12:20.58 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 12:22.03 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:23.11 | jacc0 | any change asterisk 1.8.7 is going to be released somewhere today? |
| 12:26.04 | jacc0 | *chance |
| 12:28.54 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 12:40.34 | *** join/#asterisk orioni (~chatzilla@46.183.121.39) |
| 12:40.43 | orioni | how do i log on the cdr the IP of an extensions ? |
| 12:41.21 | leifmadsen | orioni: if you're using cdr_adaptive_odbc, just create a new field in the database and write to it using the CDR() function |
| 12:41.33 | *** join/#asterisk bchia (~Adium@nat/digium/x-lhfucnzgfzaqpqpu) |
| 12:41.52 | orioni | im using mysql as a backend , but cant find the variable that i have to use to get the IP |
| 12:42.07 | leifmadsen | I don't understand the question |
| 12:42.14 | leifmadsen | the IP of what, and from where, and when? |
| 12:42.19 | orioni | cant find how to get the IP of the extension |
| 12:42.29 | orioni | the IP of the user |
| 12:42.33 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:42.33 | orioni | or the ATA devices |
| 12:42.34 | leifmadsen | try SIPPEER() function if it is a SIP end point |
| 12:43.36 | orioni | like this exten => 123,1,Set(sip_ip=${SIPPEER(2001,ip)}) |
| 12:43.40 | leifmadsen | sure |
| 12:43.53 | orioni | and then to insert on the mysql as a userfield ? |
| 12:43.57 | leifmadsen | sure |
| 12:44.02 | orioni | thanks man |
| 12:44.11 | orioni | last question |
| 12:44.29 | leifmadsen | no, it won't be |
| 12:44.30 | leifmadsen | :) |
| 12:44.42 | orioni | what is the diference of SIP/exten@provider vs SIP/provider/exten |
| 12:45.29 | orioni | :) |
| 12:46.22 | *** join/#asterisk d3wayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 12:48.37 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 12:49.01 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 12:49.05 | *** join/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 12:49.47 | NourSs | Hi, i sell 2 digiums cards ( TE 122 and TDM 411 ), private me for more information |
| 12:57.52 | *** join/#asterisk IsUp (5b8e8e8f@gateway/web/freenode/ip.91.142.142.143) |
| 12:59.16 | IsUp | hello |
| 12:59.41 | IsUp | I am getting an error when i make an outbound call "app_dial.c:1298 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown)" |
| 13:00.11 | IsUp | I have 2 GSM gateways connected to my PBX via PRI, and their loads are high |
| 13:00.37 | IsUp | I mean i have very busy traffic on that PRI lines. It says "PRI Flags: Resetting" when i do 'zap show channel XX' |
| 13:01.36 | IsUp | Same as here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg95362.html and http://lists.digium.com/pipermail/asterisk-users/2007-July/192052.html |
| 13:01.51 | IsUp | No solution so far... any ideas? |
| 13:02.35 | WIMPy | What version are you on? zap was replaced by dahdi years ago. |
| 13:02.49 | kaldemar | more than 3,5 years ago. |
| 13:02.51 | WIMPy | Is that gateway acting as NT? |
| 13:04.05 | IsUp | Its DAHDI but i am using as "Zap" because of my codes and database data etc... |
| 13:04.40 | WIMPy | 3.5 years ago, channel reallocation might not have worked at all with libpri. It's still not water tight, as far as I know. |
| 13:04.53 | IsUp | I have dahdichanname = no under my asterisk.conf |
| 13:05.18 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 13:05.23 | *** join/#asterisk serafie (~erin@nat/digium/x-ytbkpihjwahpgyyo) |
| 13:05.28 | Dovid | what is this error ? (I already looked on Google): Prodding channel 'Local/2@enswitch-call-exten-437b;2' failed |
| 13:05.36 | IsUp | WIMPy: whats is NT? do you mean Master or Slave Timer? |
| 13:06.41 | WIMPy | One side of the link hast to be in NT and the other end in TE mode. Who is acting as the network? |
| 13:06.47 | IsUp | Also i am using Sangoma A108 and Digium TDM2400P on same server |
| 13:06.51 | IsUp | Ah okay let me see |
| 13:08.20 | IsUp | It's switchtype=euroisdn and signalling=pri_cpe in zapata.conf, also span=3,0,0,ccs,hdb3,crc4,yellow in dahdi/system.conf |
| 13:08.37 | IsUp | i'll check TE/NT in 2 mins, i have to connect to GSM gateway |
| 13:08.42 | jacc0 | @leifmadsen: guess that was an answer to my question :) |
| 13:09.12 | WIMPy | Right. So they're both trying to allocate channels. |
| 13:09.33 | leifmadsen | I don't even know what the question was :) |
| 13:10.02 | WIMPy | Find out in what order the gateway allocates channels and configure chan_dahdi to use the opposite strategy (if there is one). |
| 13:10.29 | IsUp | WIMPy: It's "NT" on my GSM gateway. |
| 13:10.31 | WIMPy | That should minimise the issue. But a real fix doesn't exist, I think. |
| 13:10.54 | IsUp | WIMPy: I think you are not talking about hunting policy, right? because i am sending calls with exact port numbers, not with group option |
| 13:11.02 | WIMPy | Or if you can reverse roles, that might help/ |
| 13:11.04 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 13:11.10 | devil_evoxxx | hi all :) |
| 13:11.16 | WIMPy | Yes, I am. |
| 13:11.38 | jacc0 | @orion: in your first example asterisk will do a dns look up for 'provider' and send the call to the result ip |
| 13:11.54 | IsUp | WIMPy: I am sending calls like ZAP/XX/<number>, i mean my scripts are setting port when dialing, i am not using g1 or g2, grouping |
| 13:12.17 | kaldemar | IsUp: cpe should take timing from the line, i.e. your span should be defined as span=3,1,... |
| 13:12.41 | jacc0 | @orion: in your second example the ip of `provider` has to be in the sip.conf,user.conf or iax.conf (or in the database) |
| 13:12.41 | IsUp | also my GSM gateway has "Master" timing, is that right? |
| 13:12.45 | WIMPy | If you're specifying the channel yourself, you probably won;t know if it is in use. But you should get another error then. |
| 13:13.14 | WIMPy | Any way, specifying channels doesn't make much sense usually. |
| 13:13.14 | jacc0 | any chance asterisk 1.8.7 is going to be released somewhere today? that was the question |
| 13:13.32 | jacc0 | I'm off |
| 13:13.35 | jacc0 | bye all |
| 13:13.42 | *** part/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 13:13.43 | WIMPy | Yes, the NT should be master. |
| 13:13.45 | kaldemar | IsUp: even more reason to have span=3,1,... if the gateway is providing timing. |
| 13:13.52 | IsUp | WIMPy: I have to specify channel. Because i have an IVR system and its picking ports with some checks |
| 13:14.30 | IsUp | kaldemar: I have 2 GSM gateways, one of them is MASTER and other one is SLAVE |
| 13:14.33 | WIMPy | Are you using the gateway one way only? |
| 13:14.38 | *** join/#asterisk josta (~josta@pdpc/supporter/active/josta) |
| 13:14.48 | IsUp | WIMPy: Yes, but i have inbound calls, sometimes |
| 13:15.07 | WIMPy | That's gettig interesting. Why are they configured differently? |
| 13:15.07 | IsUp | Let me paste my configuration to somewhere |
| 13:17.09 | WIMPy | I guess that thing leads to a question I wondered about before: Can you have multiple timing sources per card? |
| 13:17.57 | IsUp | WIMPy: http://pastebin.com/gzEk3mdL |
| 13:18.00 | WIMPy | I can't find the answer from the sample configs. |
| 13:18.09 | NourSs | <PROTECTED> |
| 13:18.53 | IsUp | WIMPy: I have all PRI connections on my Sangoma 108, (8 port) and i have 1 Digium TDM2400P for FXO/FXS calls. |
| 13:19.05 | WIMPy | ok |
| 13:19.46 | WIMPy | The aswer might give different results for Digium or Sangoma hardware, I guess. |
| 13:19.58 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 13:20.19 | Katty | hello my asterisk does not work at all how to fix plz???? |
| 13:20.41 | WIMPy | But as kaldemar said: You should not try to provide timing on any of those interfaces. |
| 13:21.00 | WIMPy | That should usually cause trouble. |
| 13:21.04 | IsUp | kaldemar: May you check my configuration http://pastebin.com/gzEk3mdL |
| 13:21.49 | IsUp | WIMPy: Should i upgrade to 1.8? I am using 1.4.31 at the moment |
| 13:22.33 | WIMPy | I would, but it won't make a difference as far as timing goes. |
| 13:23.00 | kaldemar | IsUp: change timing setting for all spans. |
| 13:23.00 | WIMPy | But there have been lots of fixes since then. |
| 13:23.41 | IsUp | WIMPy: understood |
| 13:23.53 | IsUp | kaldemar: Should i change any settings on GSM gateway? May you help? |
| 13:24.09 | IsUp | kaldemar: or should i set SLAVE on both GSM gateways? |
| 13:24.18 | WIMPy | No, dahdi/system.conf |
| 13:24.28 | kaldemar | IsUp: leave the gateways be. |
| 13:24.46 | WIMPy | You could do that. But you still need to chane it for your telco. |
| 13:25.12 | WIMPy | Don't you have any issues on that line? |
| 13:25.31 | IsUp | WIMPy: Telco PRI is working fine. Just GSM gateways are problematic |
| 13:25.40 | *** join/#asterisk jkroon (~jkroon@dsl-242-10-94.telkomadsl.co.za) |
| 13:26.04 | IsUp | WIMPy: Probably 20 outbound calls per minute on my GSM gateways |
| 13:26.26 | WIMPy | My only guess would be that maybe Sangoma ignores the timing configuration. It shouldn't work very well with that configuration. |
| 13:26.29 | IsUp | WIMPy: and after some load, some ports are going to "Resetting" state as i said |
| 13:26.35 | jkroon | hi guys, when doing channel originate Local/123@context extension 321@forwardchannel - is there any way to get channel variables into the Local/123@context channel? |
| 13:27.09 | IsUp | WIMPy: It's possible, Sangoma has TE_CLOCK, TDMV_DCHAN, TDMV_SPAN settings on configuration |
| 13:27.25 | IsUp | WIMPy: Basicly, wanrouter (sangoma software) controls the DAHDI, its patching DAHDI i think |
| 13:27.36 | WIMPy | We'd need a full trace to be able to find out exactely what's happening. |
| 13:28.19 | *** join/#asterisk Aurs (~ove@nidaros.netcore.no) |
| 13:28.39 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 13:28.48 | IsUp | WIMPy: ill find a way to reproduce this error. and ill debug span, but its a production server and i cant shutdown it now |
| 13:28.53 | WIMPy | Well, that timing configuration isn't really needed in theory. |
| 13:29.42 | IsUp | Oh i found a stucked port now, ill try to place a call and debug it |
| 13:30.34 | WIMPy | Please use intense debug. |
| 13:32.41 | IsUp | Well, on problematic port, theres no any debug output, but if call via a working port, its giving debug output |
| 13:33.27 | IsUp | ZAP/33 is problematic one, if i place a call, its just giving: Unable to create channel of type 'ZAP' (cause 0 - Unknown, but if i place a call via ZAP/45 or any other working port, its giving debug output |
| 13:33.48 | WIMPy | So it's f..ed up inside chan_dahdi. |
| 13:34.10 | IsUp | Yeah, i think so |
| 13:34.53 | WIMPy | Looks like a good candidate for an up to date version. |
| 13:36.35 | NourSs | <PROTECTED> |
| 13:36.47 | IsUp | Yeah but the problem is i have an IVR software, which running over AGI. there are some changes in Asterisk 1.8 and its not worrking well. I have to re-code this IVR software. |
| 13:37.03 | WIMPy | NourSs: You don't need to tell that every 10 minutes. |
| 13:37.59 | NourSs | WIMPy: Ok :-) |
| 13:38.14 | IsUp | WIMPy: Maybe i can set resetinterval=180 or something. |
| 13:38.42 | IsUp | WIMPy. Because port is working after some time. |
| 13:38.54 | IsUp | WIMPy: without a restart or anything |
| 13:39.12 | WIMPy | It's worth a try. |
| 13:39.49 | IsUp | WIMPy: And should i change anything in system.conf for timings? |
| 13:41.24 | WIMPy | Yes, you should set all ports to slave. But if it works otherwise, I guess that part of the configuration is ignored. You should have issues with the telco line otherwise. |
| 13:41.55 | WIMPy | NourSs: Ah, 2nd try after 0 bids? |
| 13:42.34 | NourSs | WIMPy: Yes, lower price now ;-) |
| 13:42.40 | *** join/#asterisk bchia (~Adium@user-24-236-94-155.knology.net) |
| 13:43.05 | orioni | what is the diference of SIP/exten@provider vs SIP/provider/exten |
| 13:43.14 | NourSs | If anyone buy my two card i offer an Polycom IP 331 |
| 13:44.41 | WIMPy | NourSs: The price for the E1 is still a lot higher than what I payed for a 2xE1. But at least below a 4xE1. |
| 13:45.12 | IsUp | WIMpy and kaldemar, thanks for the help |
| 13:45.21 | NourSs | Now i sell it for 100$ |
| 13:45.24 | IsUp | WIMPy: I just talk to Sangoma support. They'll check my configurations |
| 13:45.25 | NourSs | earch card |
| 13:45.36 | WIMPy | And paypal only is a serious restriction. |
| 13:45.41 | IsUp | WIMPy: And i'll be back later. Thanks again. |
| 13:45.59 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 13:46.03 | NourSs | WIMPy: Paypal, a restriction ? |
| 13:47.22 | WIMPy | It's good if you want to sell to US or UK, but the popularity in the rest of europe is limited. |
| 13:49.54 | WIMPy | They just produced seriousely bad pess again last week. |
| 13:52.12 | SunTsu | NourSs: paypal really tried hard to be ignored by most people politically aware |
| 13:52.50 | NourSs | I want to sell in US or UK, i'm french.. ;-) |
| 13:53.16 | Katty | bloody french!! *shakes fist* |
| 13:53.17 | *** join/#asterisk cerberus_za (~coert@8ta-151-78-57.telkomadsl.co.za) |
| 13:53.24 | Katty | *hee* <3 the french |
| 13:54.43 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 13:56.16 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:56.45 | WIMPy | I used paypal in the very beginning, but their T&C have become unacceptable many years ago. And parts of them are even illegal here. But that doesn't stop them from enforcing them. |
| 13:57.08 | Naikrovek | WIMPy: where is "here" |
| 13:57.38 | WIMPy | EU, especially DE. |
| 13:58.01 | *** join/#asterisk modexi (~modexi@adams.osre.org) |
| 13:58.10 | WIMPy | Probably other countries likewise. |
| 13:59.04 | SunTsu | WIMPy: I used them for some time, being annoyed of their T&C, when they totally went over the top by locking down wikileaks and Wau-Holland-Stiftung accounts - while at the same time allowing "evil" organizations to still use them |
| 13:59.36 | SunTsu | never used them since, never will again |
| 13:59.45 | WIMPy | neither |
| 14:00.47 | Katty | WIMPy: you gonna join the xmas card exchange this year (sorry if i already asked my memory is crap!) |
| 14:01.02 | Katty | hugs Naikrovek |
| 14:01.11 | Naikrovek | reciprocates. |
| 14:01.15 | Naikrovek | how ya doing |
| 14:01.24 | WIMPy | No, you didn't ask, but no, I can't stand paper. |
| 14:01.24 | Katty | good...good. working on caffeine levels. |
| 14:01.28 | Katty | how're you dear? how's the family? |
| 14:01.33 | Naikrovek | all good |
| 14:01.46 | Naikrovek | bought my wife an android tablet for her birthday. she literally did a backflip |
| 14:02.33 | SunTsu | Naikrovek: hitting the tablet with her feet while doing so? ;) |
| 14:02.46 | Naikrovek | SunTsu: no, thankfully |
| 14:02.50 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 14:03.03 | Katty | Naikrovek: aww, that was very nice |
| 14:03.13 | SunTsu | Naikrovek: what did you get her? |
| 14:03.17 | Naikrovek | yeah, now i really want one. going to ask my employer to get me one. |
| 14:03.28 | Naikrovek | SunTsu: samsung galaxy tab 10.1 |
| 14:03.31 | Katty | Naikrovek: are you joining the xmas card exchange this year? |
| 14:03.35 | Naikrovek | it's an awesome device |
| 14:03.43 | Naikrovek | Katty: i dunno |
| 14:04.06 | SunTsu | Naikrovek: nice device, too nice as far as apple is concerned |
| 14:04.14 | Naikrovek | apple can shove it |
| 14:04.21 | Naikrovek | i hate apple |
| 14:04.41 | Katty | Naikrovek: well if you want to, let me know (= |
| 14:04.48 | Naikrovek | Katty: okay :) |
| 14:04.50 | Katty | Naikrovek: the list is already up and a handful of people are on it |
| 14:05.45 | *** join/#asterisk master_of_master (~master_of@p57B53978.dip.t-dialin.net) |
| 14:06.53 | Katty | goes back to the tardis knitting project. |
| 14:07.04 | jkroon | irroot, are there backports for T.30 <-> T.38 yet? (to 1.8) |
| 14:12.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:12.36 | irroot | jkroon i have a backport for it in my branches and patch |
| 14:13.09 | irroot | http://svnview.digium.com/svn/asterisk/team/irroot/patches/ |
| 14:13.17 | irroot | you can pick it up there |
| 14:13.25 | jkroon | already busy downloading :) |
| 14:14.06 | anonymouz666 | irroot: this patch is generated automatically ? |
| 14:14.26 | irroot | i do it manually periodicallly |
| 14:14.50 | irroot | once 10 is out ill stop updating it |
| 14:14.52 | anonymouz666 | I see, there's a change in 10, you backport to you 1.8 branch, and then generate the patch |
| 14:15.26 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 14:15.38 | irroot | something like that the initial work was done for 1.8 and merged with 10 but now its 10->1.8 |
| 14:15.40 | jkroon | irroot, which of those patches exactly do I need? |
| 14:16.06 | irroot | http://svnview.digium.com/svn/asterisk/team/irroot/patches/t38gateway-1.8.patch |
| 14:16.17 | jkroon | got it thanks. |
| 14:16.29 | jkroon | eventually realized the "description" is simply the last changelog entry :p |
| 14:17.39 | NourSs | Hi, i sell 2 digiums cards ( TE 122 and TDM 411 ), private me for more information |
| 14:18.11 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 14:18.11 | jkroon | NourSs, you deliver to ZA ? |
| 14:18.33 | NourSs | jkroon: possible |
| 14:21.23 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 14:32.27 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:33.33 | p3nguin | Ah, so I'm back to THAT problem again... when I hang up my phone after having received a call, asterisk crashes. Well done, chan_sccp-b. Well done. |
| 14:36.12 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 14:38.43 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 14:40.16 | *** join/#asterisk psilikon (~joel@rrcs-24-73-84-158.se.biz.rr.com) |
| 14:43.09 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 14:48.44 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 14:49.43 | eject_ck | How can I make asterisk's console output colorful for "SIP DEBUG" ? |
| 14:52.48 | leifmadsen | eject_ck: do you get colour at all right now? |
| 14:53.24 | *** join/#asterisk felimwhiteley_ (~quassel@46.7.101.58) |
| 14:53.48 | *** part/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 14:54.32 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:56.29 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 14:56.54 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:57.09 | *** join/#asterisk neurosys (~neurosys@50.20.67.41) |
| 14:57.10 | Katty | helllllllloooo nurse. |
| 14:57.39 | Naikrovek | leifmadsen: doest thou have a moment? |
| 14:58.34 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 14:58.36 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 15:00.48 | eject_ck | leifmadsen: yes, I have some colored messages like verbose level or when I have calls |
| 15:01.24 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:01.24 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 15:01.27 | eject_ck | want to colorize output during debug session to make it much easier to find errors and etc. |
| 15:02.58 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:04.36 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:07.04 | leifmadsen | eject_ck: you'll need to patch Asterisk then |
| 15:07.40 | p3nguin | Put some ointment on it first. |
| 15:08.24 | *** join/#asterisk ph_tamu (a55b7056@gateway/web/freenode/ip.165.91.112.86) |
| 15:08.48 | Naikrovek | and if you don't stop picking, it'll never heal |
| 15:09.02 | ph_tamu | hello everyone |
| 15:09.10 | Naikrovek | heya |
| 15:13.42 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 15:13.46 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:16.02 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 15:17.35 | Katty | is someone hurt? |
| 15:17.49 | Katty | has pro-nurturing skillzors. |
| 15:17.52 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 15:17.58 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 15:19.14 | Faustov | just nod if you can hear me |
| 15:21.39 | Katty | love that song |
| 15:21.47 | SeRi | guys I am running asterisk 1.8.4.4 on a alix box. The alix box has no audio engine so I keep getting the following: Sep 20 19:24:02 pbx local0.warn asterisk[2318]: WARNING[4263]: chan_oss.c:487 in setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 15:21.57 | Katty | wonders what an alix box is |
| 15:22.07 | SeRi | how can I disable it |
| 15:22.22 | SeRi | Alix its an embedded motherboard maker |
| 15:22.31 | file | tackle hugs Katty |
| 15:22.32 | Katty | ahh |
| 15:22.34 | SeRi | http://www.pcengines.ch/alix2d3.htm |
| 15:22.36 | Katty | hugs file |
| 15:22.42 | Katty | file: are you joining the xmas card exchange this year? |
| 15:22.43 | p3nguin | seri: I suppose you could set a noload for chan_oss.so in modules.conf. |
| 15:22.55 | file | Katty, Christmas doesn't exist! LA LA LA LA LA |
| 15:23.06 | SeRi | p3nguin, ok I thought so... I just didnt know if it was safe. |
| 15:23.15 | SeRi | Thanks for the confirmation |
| 15:23.28 | p3nguin | seri: You could module unload chan_oss.so for now to test it. |
| 15:23.38 | SeRi | just did :P |
| 15:23.42 | SeRi | all ok... |
| 15:24.16 | SeRi | though there would be no sound processing in the cli I guess |
| 15:24.24 | *** join/#asterisk bchia (~Adium@user-24-236-94-155.knology.net) |
| 15:24.37 | SeRi | thats where i though it could harm the system but guess not :) |
| 15:24.37 | Katty | file: sure it does!! |
| 15:24.47 | Katty | file: if you don't want to join the list that's fine, but i sitll want to send you a card. |
| 15:24.54 | Katty | file: will you /query me with your address? |
| 15:25.00 | file | sure! |
| 15:25.07 | Katty | ty |
| 15:25.18 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:26.01 | p3nguin | seri: There is a note in modules.conf about loading one of chan_oss, chan_alsa, or chan_console. Not sure what that's all about. |
| 15:27.23 | WIMPy | You can't have more than one console, AFAIK. |
| 15:27.36 | p3nguin | But do you have to have one? |
| 15:27.42 | WIMPy | no |
| 15:27.52 | p3nguin | What does it actually do? |
| 15:28.23 | WIMPy | Make your sound card a phone that's controlled via *cli. |
| 15:28.52 | WIMPy | Can be quite handy for overhead paging. |
| 15:30.51 | Katty | hugs ChannelZ |
| 15:33.49 | SeRi | WIMPy, so if I dont have a sound processor than I am sh* out luck? |
| 15:34.05 | catphish | can anyone suggest why changing CALLERID(all) does not caused the callerid number to be changed in the CDR? |
| 15:34.37 | WIMPy | SeRi: Only if you need one. |
| 15:34.51 | WIMPy | Asterisk doesn't. |
| 15:35.23 | SeRi | mhhhh well not really I guess :) ah key word "asterisk doesn't" |
| 15:35.29 | SeRi | cool thanks |
| 15:36.30 | ph_tamu | can anyone assist me with echo problem on an analog card? |
| 15:38.15 | navaismo | ph_tamu try with fxotune if you use dahdi |
| 15:41.09 | ph_tamu | navaismo: thx for reply. i've ran fxotune. followed several instructions online. additional info: asterisk 1.62, TDM800P, oslec, followed instruction to adjust rx/tx gains. |
| 15:41.36 | ph_tamu | the echo is intermittent, even on the same call. |
| 15:42.22 | navaismo | maybe its time to consider buy a HPEC license |
| 15:43.27 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 15:44.06 | Katty | what do i want for lunch |
| 15:44.11 | ph_tamu | navaismo: well, what's got me wondering is why the card worked fine before. we had it in an older machine the had a failure and moved it to newer box and decided to go from v1.4 to v1.62 |
| 15:44.35 | catphish | strange, i can't seem to change the callerid saved in the CDR at all |
| 15:44.54 | catphish | (the number part anyway) |
| 15:46.49 | catphish | ah - https://issues.asterisk.org/view.php?id=15613 |
| 15:47.17 | KavanS | anyone aware of any chrome/chromium plugins with click-2-dial functionality w/asterisk? |
| 15:48.35 | catphish | you have to use callerid(ani) :) |
| 15:48.37 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 15:50.47 | navaismo | ph_tamu debug echo it difficult |
| 15:51.11 | navaismo | the usage of cpu and ram when you hear the echo its normal? |
| 15:51.13 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:53.46 | ph_tamu | navaismo: yes, it's normal. theres not indication of high load or mem usage |
| 15:59.36 | ph_tamu | navaismo: I should mention that our sip user is only one that experience the echo issue; not heard on other end (thru pstn). |
| 16:00.20 | navaismo | this sip user use an ATA? |
| 16:02.13 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 16:02.16 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:02.16 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:02.16 | ph_tamu | navaismo: no, all are softphone users. |
| 16:02.36 | p3nguin | It's 5 o'clock somewhere. |
| 16:02.41 | navaismo | maybe the echo came from the headset |
| 16:02.57 | atan | Can you factory reset a polycom IP300 without a tftp server? |
| 16:03.16 | Naikrovek | yes. format the phone. that's as close as you'll get with or without a tftp server, I think. |
| 16:03.28 | atan | Naikrovek, where does that option hide? :-) |
| 16:03.34 | Naikrovek | what model phone |
| 16:03.38 | atan | IP300 |
| 16:03.49 | Naikrovek | i dont' have one, but it's in the admin menu somewhere |
| 16:04.57 | *** join/#asterisk libryder (~david@stg.maculon.com) |
| 16:05.00 | ph_tamu | navaismo: i've considered that but... some users are using different model/style headsets and issue didn't exist b4. |
| 16:05.04 | libryder | helllo |
| 16:05.34 | ph_tamu | navaismo: i'll add that i've tested iax2 softphone client and that didn't help either. |
| 16:05.50 | leifmadsen | p3nguin: great point |
| 16:06.09 | p3nguin | I do what I can. |
| 16:08.58 | p3nguin | What's the term for a system where you call and record a message, which then calls a list of phone numbers to play an important announcement? Schools often do it for bad weather closings or early dismissals which weren't previously scheduled. |
| 16:11.13 | Naikrovek | dunno the term |
| 16:11.29 | Naikrovek | not wardialer, not voicemail blasting something inbetween |
| 16:13.00 | p3nguin | How does voicemail blasting work and what's the reason to do it? |
| 16:13.33 | Naikrovek | well i use it for phone system changes, to tell everyone that there is a new general purpose teleconf room at XXX or whatever |
| 16:14.00 | Naikrovek | you record a message, and it just puts a copy of it in everyone's voicemail box |
| 16:14.51 | chuckf | p3nguin: robodialing? |
| 16:15.15 | *** join/#asterisk shtoom (~shtoom@59.93.122.140) |
| 16:15.32 | p3nguin | Okay, yeah that won't work for this application. The numbers are going to be home, work, and/or mobile phones rather than on an internal system. |
| 16:15.37 | Naikrovek | chuckf: that's the same as wardialing I think. what he's looking for is some phone subscription service |
| 16:15.50 | Naikrovek | i don't know the term. |
| 16:16.00 | ph_tamu | p3nguin: i thought that was just robocalling... "voicemail blasting" and "wardialing"... hmm... learned something new. |
| 16:16.05 | p3nguin | Right now, the lady has to call each number on the list and speak the message to either a person or to voicemail. |
| 16:16.14 | Naikrovek | ew |
| 16:16.23 | p3nguin | It needs to be automated. |
| 16:16.27 | Naikrovek | most definitely |
| 16:16.37 | Naikrovek | my daughter's school uses some software to do it |
| 16:16.52 | Naikrovek | it sends me an email that takes me to a site where i listen to the message |
| 16:16.53 | p3nguin | I can do it easily with Asterisk, but I don't know the term for it. |
| 16:17.16 | *** join/#asterisk imox (~imox@p4FC5C77A.dip0.t-ipconnect.de) |
| 16:17.48 | navaismo | ph_tamu same extension in other PC? |
| 16:18.20 | p3nguin | I'll have her call a specific number, enter her PIN, record the message, optionally listen to it, then activate the system to call all the people concerned. |
| 16:18.30 | chuckf | Naikrovek: I thought that robodialing was what it was commonly called when the calls are targeted. Wardialing is an old school term for calling random numbers automaticlly |
| 16:18.47 | Naikrovek | chuckf: ah maybe |
| 16:18.51 | Naikrovek | you may be right |
| 16:18.53 | libryder | how can i troubleshoot this? |
| 16:18.53 | libryder | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? (it does)) |
| 16:19.01 | ph_tamu | navaismo: each softphone user has their own extension. [or maybe i didn't understand the question.] |
| 16:19.03 | p3nguin | War dialing had a very specific purpose for calling the random numbers, though. |
| 16:19.28 | chuckf | from wikipedia: Robocall is a term for an automated phone call that uses both a computerized autodialer and a computer-delivered pre-recorded message. The implication is that a "robocall" resembles a telephone call from a robot. Robocalls are often associated with political and telemarketing phone campaigns, but can also be used for public-service or emergency announcements. |
| 16:19.41 | p3nguin | I would classify any automated calling from a list as robo-dialing. |
| 16:20.13 | p3nguin | But more specific applications using a robo-dial technique should have more specific names. |
| 16:20.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:20.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:20.39 | *** mode/#asterisk [-b *!*chatzilla@216.191.106.*] by Qwell |
| 16:21.14 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:21.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:21.44 | p3nguin | Wow, the ban is lifted. |
| 16:21.48 | navaismo | amm, you can try change that extension to another PC and see if the echo persist if not its something in the original PC |
| 16:22.54 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 16:23.26 | p3nguin | But the +q remains. |
| 16:23.33 | *** join/#asterisk irroot (~irroot@41.51.142.97) |
| 16:23.50 | *** mode/#asterisk [-q *!*chatzilla@216.191.106.*] by leifmadsen |
| 16:23.56 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 16:23.57 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:24.05 | p3nguin | Still there. |
| 16:24.08 | leifmadsen | huh... |
| 16:24.23 | p3nguin | hisnick!*@* |
| 16:24.43 | *** mode/#asterisk [-q [TK]D-Fender!*@*] by leifmadsen |
| 16:25.08 | atan | is going to take these IP300 phones and throw them across the parking lot |
| 16:25.25 | p3nguin | I'll give you my shipping address. |
| 16:25.27 | Qwell | p3nguin: where did you see the +q set? |
| 16:25.30 | chuckf | is going to stand on the other side and catch them |
| 16:25.36 | p3nguin | /mode +q |
| 16:25.37 | Qwell | it doesn't show up in /bans |
| 16:25.38 | Qwell | ahh |
| 16:25.43 | atan | All of them are preloaded with settings for another provider and won't boot to DHCP :X |
| 16:25.56 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:26.23 | atan | The admin login works fine though :-) I just want them to get on the network so I can use the web interface to change the SIP details but *no* that's too much to ask of these darn things. |
| 16:34.12 | ph_tamu | navaismo: thanks for running through troubleshooting options with me. |
| 16:34.35 | navaismo | no problem |
| 16:35.48 | atan | thinks there will be $20 phones going on eBay very soon |
| 16:35.58 | atan | None of them will use DHCP to save their life :-( |
| 16:36.12 | p3nguin | Does it have a powwer supply with it? |
| 16:36.40 | atan | All of them have power, lol :-( I had really hoped to use them but meh they hate life right now |
| 16:37.03 | p3nguin | Got PayPal? |
| 16:37.19 | atan | should, for his own sake, get back to you on that one |
| 16:37.28 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:37.39 | atan | There's 7 of them right now =\ I have 8, one does DHCP fine the other 7 hate life. |
| 16:37.57 | p3nguin | I guess you'll be keeping the one that is working. |
| 16:38.44 | atan | I'm sure it's just some setting I walked past somewhere. It must be. But I've set it all up the best I can tell and it just says "Failed to get boot paramaters via DHCP." |
| 16:38.57 | p3nguin | If you're getting rid of them, I'd take one just so to mess with in an attempt to overcome the problem you're having. |
| 16:39.23 | Faustov | Hi, I get this: == Executing [/usr/bin/sox 2011-09-21-17:36.wav 2011-09-21-17:36.mp3] in the CLI, however the mp3 does not get created. Manually it works - where can I see what fails? |
| 16:39.23 | WIMPy | Any VLAN stuff going on? |
| 16:40.03 | atan | WIMPy, it is indeed plugged in to a new router... interesting question. Will move it over to another router and see what happens and report back :) |
| 16:40.12 | libryder | http://www.amazon.com/Avoid-Huge-Ships-John-Trimmer/dp/0870334336/ref=sr_1_1?ie=UTF8&qid=1316623056&sr=8-1 |
| 16:40.35 | atan | Some router they gave me to use with the new fiber install =\ not sure how it's setup right now |
| 16:40.55 | atan | Be back in a bit. Interesting to see what happens. |
| 16:41.27 | irroot | atan is it insured ?? petrol and match ?? |
| 16:41.44 | Faustov | nevermind it was missing the paths |
| 16:42.12 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 16:46.23 | *** join/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
| 16:46.40 | atan | Well I'll be. Thank you WIMPy. Using the other router resolves this issue. However, now I wonder what on earth is not setup in the current router... hmmm... ! |
| 16:47.06 | Micc | can I ignore the comfort noise messages when its talking about two asterisk servers? Because neither supports comfort noise it really shouldn't be complaining about it. |
| 16:47.58 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 16:50.35 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 16:50.53 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:03.30 | pdtpatrick1 | Question .. has anyone implemented something like this with Jabber? |
| 17:03.30 | pdtpatrick1 | http://www.igniterealtime.org/projects/openfire/plugins/asterisk-im/readme.html |
| 17:06.39 | Naikrovek | if you count their spark client, yes |
| 17:06.55 | Naikrovek | the spark client talks to asterisk though, not the other way around |
| 17:15.06 | *** join/#asterisk ocx (5ebb3951@gateway/web/freenode/ip.94.187.57.81) |
| 17:15.34 | leifmadsen | sounds like something I said last night :)_ |
| 17:22.07 | tzanger | pdtpatrick1: what's that for, presence information for extensiosn? |
| 17:22.10 | tzanger | er extensions? |
| 17:23.00 | *** part/#asterisk libryder (~david@stg.maculon.com) |
| 17:23.48 | *** join/#asterisk _pll (c8250919@gateway/web/freenode/ip.200.37.9.25) |
| 17:24.26 | _pll | Hi, a quick question. Is there a way to leave confbridge conference? |
| 17:24.48 | _pll | application features don't seem to work there. |
| 17:25.55 | Naikrovek | hang up? |
| 17:26.10 | p3nguin | Too easy! |
| 17:26.23 | _pll | I am trying to implement n-way conference http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 17:26.28 | _pll | using app_confbridge |
| 17:26.50 | _pll | because there is no dahdi card in this server. |
| 17:27.19 | p3nguin | You don't need a card to use MeetMe. |
| 17:27.21 | _pll | It works fine until I have to invite a 4th person because features don't work inside. |
| 17:27.38 | p3nguin | Cards are for analog connectivity. |
| 17:27.50 | _pll | Meetme uses timer from dahdi |
| 17:28.00 | p3nguin | Oh yeah? |
| 17:28.12 | p3nguin | You must think I'm new here. |
| 17:30.05 | _pll | how reliable is to use meetme with no card timer. |
| 17:30.12 | p3nguin | very |
| 17:30.42 | p3nguin | I've been using MeetMe for years using the timer provided by Dahdi. |
| 17:32.29 | p3nguin | Don't take this the wrong way; I'm not trying to discourage you from using ConfBridge, but I do want you to know that MeetMe works just fine and it does not require a card (because cards are for analog connectivity). |
| 17:32.38 | *** join/#asterisk bobg (~bobg@ool-4576d9c2.dyn.optonline.net) |
| 17:32.49 | Qwell | p3nguin: Have you looked at confbridge lately? |
| 17:32.55 | p3nguin | Negative. |
| 17:32.58 | pdtpatrick1 | tzanger, presence information .. so what i am trying to do is when the user picks up the phone - i want their jabber client to change to on the phone or busy |
| 17:33.00 | Qwell | Look in 10-beta |
| 17:33.07 | p3nguin | I've heard it's pretty nice, though. |
| 17:33.12 | pdtpatrick1 | currently i only have it go to VM if busy or away |
| 17:33.14 | pdtpatrick1 | and ring when online |
| 17:33.19 | tzanger | pdtpatrick1: gotcha |
| 17:33.25 | pdtpatrick1 | any idea? |
| 17:33.27 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-132-115.chyn.qwest.net) |
| 17:33.29 | tzanger | I'm just reading about openfire now. never heard of it before |
| 17:33.37 | p3nguin | I use openfire. |
| 17:33.44 | p3nguin | Easy deployment. |
| 17:35.19 | tzanger | needing to use java is kind of putting me off of it |
| 17:35.36 | tzanger | p3nguin: how do you use openfire, what is your use case? |
| 17:35.40 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:36.07 | p3nguin | I use it for a basic XMPP messaging platform. |
| 17:36.29 | p3nguin | It's for internal IM service and I also use it with asterisk for call notification. |
| 17:36.31 | ph_tamu | would cpu affinity help reduce echo issue? |
| 17:36.35 | bobg | I have a snom 821 SIP phone connecting to Asterisk 1.6.2.5 via NAT with qualify=yes. Status shows "UNREACHABLE" yet wireshark shows me that the phones "Status: 200 OK" message is reaching the Asterisk box. Any ideas on what to check next? |
| 17:36.50 | pdtpatrick1 | p3nguin, what other plugins do u use? |
| 17:36.56 | pdtpatrick1 | besides call notification |
| 17:37.00 | pdtpatrick1 | i already have that working |
| 17:37.19 | p3nguin | IM is all I use it for. |
| 17:37.39 | pdtpatrick1 | no presence detection or nifty tricks? |
| 17:37.45 | p3nguin | Asterisk is configured as a component, and I use JabberSend() in dial plan. |
| 17:37.57 | pdtpatrick1 | right got that working |
| 17:38.10 | p3nguin | I didn't know I needed anything else. |
| 17:38.37 | pdtpatrick1 | but it would be super cool if someone was say online, ur phone rings and then soon as u pick up, your jabber client goes to DnD so all subsequent calls are either parked or forwarded or sent to voicemail |
| 17:38.59 | pdtpatrick1 | i also saw something where u can actually interact with the ami .. and type something like foward + extension |
| 17:39.06 | pdtpatrick1 | and it would forward to that person |
| 17:39.22 | pdtpatrick1 | that's a totally different level but now i have to battle with getting this api to work |
| 17:39.31 | p3nguin | Why would I rely on jabber for that stuff? Asterisk does that fine all by itself. |
| 17:39.58 | _pll | Damn, beta 10 confbridge looks so sweet. |
| 17:40.38 | Qwell | p3nguin: ^ I told you so. |
| 17:41.33 | Qwell | It's cool though. You can keep using meetme, while us big kids video conference in hyper wideband. |
| 17:41.56 | Qwell | The_Boy_Wonder: PS, we'll need the term "hyper wideband" some day, for really reals. |
| 17:42.28 | p3nguin | I *JUST* moved to 1.8. I won't be using 10 for years! |
| 17:43.20 | Qwell | You're a whole 184kHz behind |
| 17:43.41 | p3nguin | I don't have any video phones, anyway. |
| 17:44.09 | Qwell | But your conference calls will sound like they're on...phones...gross. |
| 17:44.17 | p3nguin | haha |
| 17:44.59 | Qwell | We rock out at higher-than-CD-quality in ours. |
| 17:45.13 | The_Boy_Wonder | 192khz ftw |
| 17:45.23 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 17:45.29 | Qwell | The_Boy_Wonder: we really need to get some hardware that supports that >.> |
| 17:45.40 | p3nguin | I'd probably have to upgrade the hardware that Asterisk is on. |
| 17:45.46 | Micc | What are some good video phones that work with that? |
| 17:45.47 | bobg | is there a conference app for * that lets you create and manage temporary conf rooms for meetings? |
| 17:45.54 | Qwell | Micc: the polycom ones |
| 17:46.08 | Micc | aren't those like starting at $1500? |
| 17:46.16 | Qwell | got me |
| 17:46.23 | Qwell | jitsi is supposedly a good softphone |
| 17:46.35 | p3nguin | Buy a dozen of them so everyone can have fun in meetings. |
| 17:46.38 | The_Boy_Wonder | yeah, jitsi worked best with confbridge and video |
| 17:46.41 | Micc | I'll have to find a big customer that wants some polycom video phones so I can play with em. |
| 17:47.06 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:47.26 | pdtpatrick1 | Question .. someone please educate me.. does Meetme depend on dahdi and if so does that mean without a PRI (if one is using sip trunking) .. you cannot conferece/bridge calls? |
| 17:48.01 | _pll | I am interested in the answer too. |
| 17:48.18 | Qwell | The_Boy_Wonder: We need these. http://www.polycom.com/products/voice/accessories/soundstation_vtx1000_subwoofer.html |
| 17:48.21 | Micc | I bridge/conference calls all the time without a pri card. |
| 17:48.51 | pdtpatrick1 | Micc, are u using sip-trunks ? |
| 17:48.58 | atan | Anyone happen to know of a toll-free number setup to play hold music I can play with to test the value route on voip.ms? Want to see if it can hold a call for 4+ hours. |
| 17:48.59 | The_Boy_Wonder | wow, a sub for a speaker phone |
| 17:49.06 | atan | Or perhaps someone just knows if it works... that's cool too. |
| 17:50.49 | pdtpatrick1 | Micc, any response? |
| 17:51.06 | Qwell | atan: Call my electric company. They'll keep you on hold for 4 hours. |
| 17:51.28 | pdtpatrick1 | haha |
| 17:52.19 | SeRi | lmao |
| 17:52.37 | Micc | pdtpatrick1, yes only use sip trunks |
| 17:52.40 | bobg | people who call me to sell me things I don't want are often put on hold for 4 hours |
| 17:53.14 | bobg | My on hold message then tells them to contact their doctor |
| 17:53.18 | pdtpatrick1 | Micc, looks like u still have to use a dummy zaptel driver for the conferencing to work .. can u verify this? |
| 17:54.57 | bobg | i assume that you need at least a dummy zaptel/DHADI driver for a lot of things in * to work |
| 17:55.27 | Micc | pdtpatrick1, I suppose thats possible. I don't think I'm building dahdi anything on my new machines, so I doubt it requires a dummy driver thats in the dahdi package. |
| 17:55.45 | Micc | pdtpatrick1, I haven't build zaptel or dahdi in years now. |
| 17:56.59 | pdtpatrick1 | what version of asterisk are you on ? |
| 17:57.11 | Micc | 1.6.2.17.3 |
| 17:57.37 | pdtpatrick1 | i c |
| 17:57.56 | pdtpatrick1 | well i'll soon find out :( .. couple more days will be off PRI and we'll see. |
| 17:58.18 | pdtpatrick1 | Another question .. has another successfully integrated their PBX with exchange calendaring ? |
| 17:58.27 | _pll | meet me requires dahdi to be loaded in the kernel. |
| 17:58.36 | Micc | PRIs are a hassle IMO, compared to sip trunks from a good provider. |
| 17:59.00 | p3nguin | _pll: Good thing it has a module for that. |
| 17:59.19 | _pll | hmm? |
| 17:59.23 | Micc | _pll: yes, lsmod does show dahdi_dummy. |
| 17:59.29 | jaytee | pdtpatrick1, I integrated an Asterisk PBX with Exchange 2007 Unified Messaging as the voicemail system. |
| 17:59.57 | jaytee | and it could access the user's Outlook calendar |
| 18:00.39 | pdtpatrick1 | what's the benefit you would think of having such? does it send reminders to your phones? maybe allow people to book meetings etc? |
| 18:00.54 | p3nguin | bobg: My problem is that they often call me toll-free, so I want to get them off the line well before the 4 hour mark. |
| 18:01.00 | jaytee | but Exchange UM, OCS and now Lync only talk SIP TCP, they won't talk SIP UDP. |
| 18:01.24 | pdtpatrick1 | awww |
| 18:01.36 | SeRi | somehting funny is happening when I recive calls.... when the call comes in my phones ring very weird.... Its like they short ring pause ring pause full ring pause... Any ideas what cuases this behavior? |
| 18:01.50 | SeRi | sometimes I answer and the other side still rining |
| 18:01.57 | SeRi | wierd.... |
| 18:02.41 | Micc | what is the advantage of hooking up asterisk to exchange? |
| 18:02.49 | jaytee | pdtpatrick1, no it won't send messages to phones, it will provide autoattendant voice recognition for a Directory of employees, will read your email over the phone to you, let you check meetings, appointments or cancel/reschedule them over the phone. Plus it will do that easily in multiple languages. |
| 18:03.54 | jaytee | the price of course for adding those kinds of bells and whistles is that you have to buy Exchange Enterprise CAL licenses for every Unified Messaging client. |
| 18:04.45 | jaytee | pdtpatrick1, Asterisk 1.6.2.x and above will talk SIP TCP to Exchange |
| 18:05.00 | *** part/#asterisk irroot (~irroot@41.51.142.97) |
| 18:05.06 | jaytee | I've had it working on 1.6.1.12 back a few years ago. |
| 18:05.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:05.44 | jaytee | prior to that I had to use SipXecs as a sip proxy doing UDP/TCP transforms |
| 18:05.54 | jaytee | when running * 1.4 |
| 18:07.25 | _pll | hmm, can't load app_meetme |
| 18:07.45 | jaytee | do you have DAHDI installed and loaded? |
| 18:07.50 | _pll | Does it depend in anything else? |
| 18:08.18 | _pll | yes, I compiled latest dahdi then recompiled asterisk, started dahdi and restarted asterisk |
| 18:08.37 | p3nguin | What exactly do you mean by "started dahdi" there? |
| 18:09.03 | _pll | service dahdi start |
| 18:09.09 | _pll | loads dahdi modules? |
| 18:09.15 | _pll | lsmod shows dahdi |
| 18:09.15 | p3nguin | That's not needed. |
| 18:09.17 | p3nguin | You load the dahdi module in the kernel, and load the chan_dahdi module in Asterisk. |
| 18:09.28 | _pll | chan_dahdi required? |
| 18:09.30 | _pll | hmmm |
| 18:09.54 | p3nguin | If you want the pseudo channel, which provides timing, you'll want chan_dahdi to be loaded. |
| 18:10.22 | Micc | lsmod should show dahdi_dummy |
| 18:10.33 | p3nguin | No it shouldn't. |
| 18:10.37 | *** join/#asterisk bob_kelso (~john@nat/digium/x-cqyqecgedmbafywj) |
| 18:10.57 | Micc | p3nguin, mine does. actually it shows both dahdi and dahdi_dummy |
| 18:11.05 | p3nguin | dahdi_dummy has been gone for some time; the dummy is now part of the regular dahdi module. |
| 18:11.11 | Micc | I thought he was doing it without the real hardware. |
| 18:11.15 | p3nguin | Which means you need to upgrade. |
| 18:11.17 | Micc | oh |
| 18:11.21 | Micc | good to know. |
| 18:13.20 | jaytee | yeah, on 1.6.2.12 running lsmod I get dahdi_transcode, dahdi and dahdi_voicebus but don't see a dahdi_dummy even though I know that * uses the dummy module. |
| 18:14.53 | Qwell | dahdi_dummy is dead |
| 18:15.02 | Qwell | It's part of dahdi now |
| 18:15.44 | _pll | hmm, can't load chan_dahdi either. |
| 18:16.10 | Qwell | Did you re-run configure after installing dahdi? |
| 18:16.16 | _pll | yes |
| 18:16.36 | _pll | because the module appears as chan_dahdi.so app_meetme.so |
| 18:16.39 | p3nguin | That all seems like a lot of work. I've never had to do anything with dahdi other than install it and load it. |
| 18:16.54 | _pll | I am using autoload = no |
| 18:17.01 | _pll | Is there anything else I need to load? a res_ ??? |
| 18:17.03 | SeRi | any body know how fix ring issues? my phones are some what ringing all broken... like short ring pause full ring... etc... |
| 18:17.06 | SeRi | any ideas? |
| 18:17.27 | _pll | res_timing_dahdi fails too |
| 18:19.36 | _pll | in fact, I can't load anything. Something went wrong in the compile/install. |
| 18:23.18 | _pll | blep, freepbx rewrote modules.conf |
| 18:23.28 | leifmadsen | shocking |
| 18:23.41 | _pll | module was already autoloaded |
| 18:23.55 | leifmadsen | with freepbx you can pretty always assume it is going to rewrite all the .conf files |
| 18:24.15 | anonymouz666 | chmod -w *.conf |
| 18:24.17 | anonymouz666 | :P |
| 18:25.29 | *** join/#asterisk singler (~singler@c.wapgw.bi.lt) |
| 18:26.49 | ocx | queue pause is not available for asterisk 1.4 ? |
| 18:28.52 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 18:28.57 | raden | hugs Katty |
| 18:29.59 | Naikrovek | Katty ignores raden |
| 18:30.01 | Naikrovek | lulz |
| 18:30.08 | raden | lol |
| 18:30.10 | raden | whaz up bro |
| 18:30.24 | p3nguin | She probably really did have to put him on ignore. |
| 18:30.31 | *** part/#asterisk dr0ck (~dr0ck@nat/digium/x-uxvfrprthsebyzwt) |
| 18:31.26 | _pll | You don't need chan_dahdi for app_meetme |
| 18:31.47 | p3nguin | If you want the pseudo channel, how will you get it? |
| 18:33.40 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 18:35.30 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 18:35.33 | leifmadsen | _pll: why do you think that? (you do) |
| 18:36.33 | p3nguin | Maybe he thinks that because app_meetme.so will load without having chan_dahdi.so loaded. |
| 18:37.31 | p3nguin | I'd be interested to see meetme work without it, though. |
| 18:38.08 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 18:40.15 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 18:40.25 | [sr] | yellow |
| 18:41.17 | _pll | leifmadsen: Because I haven't loaded chan_dahdi and it works. |
| 18:42.25 | _pll | p3nguin: Test it, it works. |
| 18:47.56 | *** join/#asterisk bmint (~bmint@h174.92.190.173.static.ip.windstream.net) |
| 18:49.03 | SeRi | is dtmfmode=rfc2833 need it when using a PAPT2 for sip with asterisk? |
| 18:49.55 | p3nguin | I'd certainly use that mode unless I was forced to use another for an extremely good reason. |
| 18:51.23 | SeRi | p3nguin, only reason I ask is because my phones are all strugling to ring and sometimes when I pick up the call the other side stays ringing even though I answerd... I thought that had something to do with it |
| 18:51.41 | p3nguin | That's not DTMF. |
| 18:53.01 | SeRi | p3nguin, any clues on where I should be looking at? the logs show everything to be ok |
| 18:58.00 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 19:04.32 | bmint | Recently updated from Asterisk 1.6 to 1.8. Now when I try to park a call by transferring to 500 (our parkext) it does not playback the park position. It shows that is is playing in the CLI but the parker does not hear it. |
| 19:05.16 | p3nguin | Are you parking with features? |
| 19:05.20 | bmint | Yes |
| 19:06.27 | bmint | Actually I am transferring to ext 500. |
| 19:06.37 | bmint | We have had issues with one touch parking. |
| 19:07.02 | psilikon | What is the preferred method for clustering Asterisk servers together for the largest aggregate number of concurrent channels? |
| 19:09.41 | Naikrovek | raden: playing with windows 8, trying to design a new phone system in my head, working on an important internal project, designing a new virtualization system, trying not to make fun of my receptionist's laugh, thinking of other things to add to this sentence. |
| 19:13.33 | Chainsaw | sets a loop of the receptionists laugh as hold music on the phone system in Naikrovek's head |
| 19:15.18 | Nugget | Corporate Accounts Payable, Nina Speak, Just a Moment |
| 19:15.25 | *** join/#asterisk libryder (~david@stg.maculon.com) |
| 19:18.49 | Naikrovek | Nugget: hah |
| 19:18.56 | Naikrovek | you're a power-lurker dude |
| 19:19.11 | Naikrovek | you were in #slashdot also, the 13-14 years ago when i met you there |
| 19:19.34 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-171.mobile.uci.edu) |
| 19:19.37 | Naikrovek | where the hell is drdink anyway |
| 19:19.57 | Naikrovek | drwiii |
| 19:20.00 | Naikrovek | all the oldies |
| 19:20.54 | *** part/#asterisk libryder (~david@stg.maculon.com) |
| 19:21.32 | Nugget | he's around, I stalk him on facebook |
| 19:21.42 | Nugget | cowboyneal is the one who fell off the face of the earth |
| 19:21.55 | Naikrovek | ah yeah wow |
| 19:22.52 | Naikrovek | drdink was a fun guy |
| 19:23.06 | Nugget | http://macnugget.org/stuff/res0/ |
| 19:24.09 | Naikrovek | what are those |
| 19:24.14 | Naikrovek | mp3s, but whose |
| 19:24.22 | Naikrovek | yours, but who wrote them/performed them? will listen later |
| 19:24.32 | Nugget | res0 |
| 19:24.38 | jaytee | telnet |
| 19:24.38 | Nugget | telnet is eeeeeeevil! |
| 19:24.42 | jaytee | LOL |
| 19:24.51 | Naikrovek | lol |
| 19:25.00 | tzanger | telnet is AWRSOME |
| 19:25.07 | Naikrovek | shyeah, if you're in 1995 |
| 19:25.13 | tzanger | oh wait, that's a misspelling. ARSEOME. |
| 19:25.31 | Micc | telnet is still good to test protocols that aren't encrypted. |
| 19:25.33 | Naikrovek | when i worked for verio, we did all of our cisco router work via telnet |
| 19:25.36 | tzanger | yep I still use it |
| 19:25.43 | jaytee | for a second I thought it was an accent like Gary Oldman had in The Fifth Element |
| 19:25.52 | Nugget | ha ARSEOME |
| 19:29.26 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 19:32.17 | ocx | is there any way of getting device states for analog phones? |
| 19:33.17 | ocx | device state information for analog |
| 19:34.00 | *** join/#asterisk Fritz09 (~Adium@pop1-2551.catv.wtnet.de) |
| 19:34.42 | Micc | ocx, an analog phone connected to an ATA? |
| 19:34.53 | ocx | fxs port |
| 19:35.06 | Micc | no idea about that. don't use fxs ports. |
| 19:38.56 | Naikrovek | analog hardware is a pita, ime |
| 19:42.26 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 19:46.06 | *** join/#asterisk Syrex (~syrex@dsl-146-17-198.telkomadsl.co.za) |
| 19:49.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:53.38 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 19:57.32 | *** join/#asterisk Assuero (~Assuero@187.114.181.21) |
| 20:03.41 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:09.26 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:11.07 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:13.10 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:30.37 | *** join/#asterisk haon_OS (~nseis@74.63.151.37) |
| 20:31.02 | Micc | is there any way to stop these comfort noise errors between two asterisk servers? |
| 20:31.16 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:31.44 | p3nguin | Does it say that Asterisk does not support comfort noise? |
| 20:31.50 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.229.82) |
| 20:32.04 | malcolmd | asterisk a > asterisk b shouldn't result in comfort noise errors...asterisk doesn't support comfort noise and shouldn't try to invite to another server with it or generate it |
| 20:32.42 | p3nguin | Do you see the message on both systems? |
| 20:34.36 | jaytee | asterisk would not be the source of comfort noise but a phone connected to asterisk a might be producing it. |
| 20:35.34 | jaytee | look in the phone configs for options like Silence Suppression or Comfort Noise and if the phones have it disable the option. |
| 20:35.36 | malcolmd | yup, but in that case, the second asterisk server shouldn't be generating the error because the first server shouldn't have created an invite to the second with cng enabled. so, really, just find the endpoint and turn off cng |
| 20:35.46 | p3nguin | I have another theory as well, but until he answers me, I won't be able to apply my thoughts. |
| 20:36.37 | *** join/#asterisk sunfone (~jeff@184-202-223-43.pools.spcsdns.net) |
| 20:40.04 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 20:40.04 | *** join/#asterisk dre (dre@gateway/shell/xzibition.com/x-aagulhyylwfnqgjb) |
| 20:40.04 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 20:54.56 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:03.35 | *** join/#asterisk neurosys (~neurosys@69.198.214.225) |
| 21:04.07 | devil_evoxxx | hi all |
| 21:04.17 | devil_evoxxx | someone here use Quescom Gw? |
| 21:05.41 | *** part/#asterisk bob_kelso (~john@nat/digium/x-cqyqecgedmbafywj) |
| 21:07.47 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:10.15 | *** join/#asterisk xpot-mobile (~james@155-99-214-52.uconnect.utah.edu) |
| 21:10.54 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:14.25 | Micc | Yes I see them on both systems, and they are both the same version of asterisk. 1.6.2.17.3 |
| 21:16.20 | Micc | its not generated by the device. I only get the error between servers. |
| 21:16.55 | leifmadsen | asterisk does not generate CNG |
| 21:16.59 | leifmadsen | and it will not offer it |
| 21:17.07 | leifmadsen | it *has* to come from the end points outside of asterisk |
| 21:18.13 | Micc | I don't see it with just provider -> asterisk -> end point |
| 21:18.26 | Micc | I only see error when it is provider -> asterisk -> asterisk -> end point |
| 21:19.17 | Micc | I'm not saying it is doing CNG, but maybe its detecting it incorrectly. |
| 21:21.13 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-230-101.pools.arcor-ip.net) |
| 21:26.50 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:36.27 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 21:36.27 | *** join/#asterisk dre (dre@gateway/shell/xzibition.com/x-aagulhyylwfnqgjb) |
| 21:36.27 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 21:44.56 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-132-115.chyn.qwest.net) |
| 21:58.40 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 22:01.32 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 22:01.51 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 22:03.27 | *** join/#asterisk neurosys (~neurosys@69.198.214.225) |
| 22:07.23 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:11.11 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 22:11.41 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:27.58 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:33.18 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 22:34.12 | *** part/#asterisk haon_OS (~nseis@74.63.151.37) |
| 22:36.01 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-171.mobile.uci.edu) |
| 22:39.30 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 22:55.39 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 22:56.13 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:08.08 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:13.53 | *** join/#asterisk Eitan (~Eitan@cpe-76-90-28-67.socal.res.rr.com) |
| 23:15.17 | Eitan | hey guys... running asterisk 1.4.41 - i am haveing a lot of strange behavior with my queues.... nothing static... just problems come up randomly and disapear the same... |
| 23:15.48 | Eitan | phones will do half rings, and then transfer to other reps.... reps logged into multiple queues arent working properly |
| 23:16.05 | Eitan | i know its vague... but anybody deal with this kind of issue ever? |
| 23:16.13 | p3nguin | You're using chan_agent? |
| 23:16.44 | Eitan | curve ball... freepbx |
| 23:16.50 | p3nguin | ~freepbx |
| 23:16.50 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 23:16.55 | Eitan | lol |
| 23:16.56 | p3nguin | Thank you. Come again. |
| 23:17.22 | Eitan | thanks |
| 23:25.15 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-171.mobile.uci.edu) |
| 23:37.47 | *** join/#asterisk treborsux (~treborsux@74-93-169-85-Jacksonville.hfc.comcastbusiness.net) |
| 23:37.53 | *** join/#asterisk mocker (~mocker@99-155-190-107.lightspeed.mssnks.sbcglobal.net) |
| 23:38.03 | treborsux | I have tftp running phones with bootrom 4+ load and work fine |
| 23:38.17 | treborsux | i have some that have 3.x and the sip app fails |
| 23:38.51 | treborsux | i copied the boot rom zip for 4.3.1 into tftp directory |
| 23:39.09 | treborsux | what do i have to do to tell phones to upgrade boot roms? |
| 23:41.10 | treborsux | anyone here? |
| 23:41.52 | pabelanger | ~patience |
| 23:41.52 | infobot | somebody said patience was a Godly attribute, or the solution for most things. |
| 23:42.08 | SeRi | ^^nice! |
| 23:42.51 | treborsux | :< |
| 23:42.56 | treborsux | you havent met my boos |
| 23:43.18 | treborsux | sip app dies with config error |
| 23:43.36 | treborsux | how do i tell phone to upgrade bootrom that is in tfftp |
| 23:43.52 | treborsux | boos=boss whoopie |
| 23:43.56 | *** join/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net) |
| 23:44.56 | pabelanger | Well, you'll have to wait until somebody with polycom phones jumps in to help. This is a channel for asterisk after all |
| 23:46.31 | Naikrovek | if the phone sees the bootrom files, it'll upgrade them on the next boot |
| 23:47.03 | SeRi | Does anybody know what is the default user and password for the polycom phones or at least how to reset it to factory? |
| 23:47.54 | Naikrovek | default user/pass is Polycom/456 |
| 23:48.01 | Naikrovek | just 456 if you're on the phone menus |
| 23:48.09 | Naikrovek | note the capital P |
| 23:52.51 | treborsux | anyone know how to upgade bootrom on a polycom? |
| 23:53.18 | Naikrovek | yes, i already said it. dump the bootrom files the same place the firmware files are, and reboot the phone |
| 23:53.35 | Naikrovek | provided the phone can communicate with the (t)ftp server, the phone will see the files and upgrade itself |
| 23:53.37 | Naikrovek | also: |
| 23:53.43 | Naikrovek | ADMIN GUIDE HAS THIS ALL IN IT. |
| 23:53.51 | Naikrovek | download it, read it, be enlightened. |
| 23:58.23 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-230-101.pools.arcor-ip.net) |