| 00:15.29 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:42.23 | rue_house | the aastra console can connect to remote aastra machines right? |
| 00:42.29 | rue_house | many questions about the aastra 160 pro yet? |
| 00:42.30 | Linuturk | so, if I understand this right: http://www.8774e4voip.com/GXW4024_p/grandstream-gxw4024.htm << that will enable me to replace analog channelbanks with a device that connects to asterisk over SIP? |
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| 01:06.12 | raden_work | bla |
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| 03:35.06 | *** join/#asterisk jhardy (~jveezee@S010600265a138609.vf.shawcable.net) |
| 03:35.17 | jhardy | anyone using a UTstarcom f1000? |
| 03:35.37 | WIMPy | ~ask |
| 03:35.38 | 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. |
| 03:36.27 | jhardy | I am having trouble connecting to my access point with anything but open authentication. ie wep or WPA-PSK does not work |
| 03:36.59 | jhardy | looking at my AP's real time logs, it says failed authentication.. but all my other devices are working OK |
| 03:37.04 | WIMPy | It will only do WPA, not WPA 2 |
| 03:37.18 | jhardy | i realize that.. my AP is setup for WPA version 1 / tkip |
| 03:37.41 | jhardy | if i put my ap into no authentication it works fine. |
| 03:39.18 | WIMPy | It seems to dislike certain traffic, but I don't know what exactely. |
| 03:39.52 | WIMPy | That is, it sometimes doesn't coexist very well with other clients. |
| 03:40.00 | jhardy | hmm. I am going to try another AP i guess.. i am currently using a cisco 1131 .. although it is a G AP , it is backwards compatible .. |
| 03:40.08 | jhardy | oh |
| 03:40.10 | jhardy | hmm |
| 03:40.24 | jhardy | what a POS |
| 03:40.54 | jhardy | now I need a drink.. I am upset. |
| 03:41.02 | jhardy | :) |
| 03:41.08 | joobie | you guys hear |
| 03:41.12 | joobie | Egypt shut down the Internet |
| 03:41.14 | joobie | for all of Egypt |
| 03:41.15 | joobie | :P |
| 03:41.42 | WIMPy | Well, most of the wifi phones seem to be raterh cheap. |
| 03:44.10 | jhardy | wimpy: it looks that way.. think i might give this mobile_chan / bluetooth thing a shot. Have you tried it out? |
| 03:44.23 | WIMPy | No |
| 03:44.41 | jhardy | looks interesting |
| 03:44.56 | WIMPy | If you want something reliable for a known place, go DECT. |
| 03:45.09 | WIMPy | Or rather CAT-iq. |
| 03:45.29 | jhardy | i would then need a ATA or FXS? |
| 03:45.44 | coppice | the wideband DECT IP phones are a good choice |
| 03:45.56 | joobie | http://6.mshcdn.com/wp-content/uploads/2011/01/egypt_graphic.jpg |
| 03:45.56 | WIMPy | Or ISDN or SIP. |
| 03:46.02 | jhardy | any particular model i can read up on? |
| 03:46.35 | WIMPy | But my experience with the SIP DECT from Siemens was rather painful as well. |
| 03:48.44 | jhardy | i have a picture of a big internet on/off switch |
| 03:48.50 | jhardy | lol |
| 03:55.19 | *** join/#asterisk Miller357 (453df542@gateway/web/freenode/ip.69.61.245.66) |
| 03:55.35 | Miller357 | hello |
| 03:56.34 | Miller357 | 23 years late to IRC, is anyone out there? |
| 03:56.53 | Miller357 | radio check ? |
| 03:58.03 | WIMPy | Nope. All moved to Skype. |
| 03:58.19 | jhardy | Egypt shut down IRC |
| 03:58.35 | p3nguin | 220+ disconnected BNCs |
| 04:00.28 | Miller357 | Anyone running into problems with voicemailmain, can't forward messages and the file sequencing problems? |
| 04:10.06 | *** join/#asterisk Miller357 (453df542@gateway/web/freenode/ip.69.61.245.66) |
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| 04:10.33 | Miller357 | lanning, I'm about 23 years late to IRC.... can I get a radio check? |
| 04:11.02 | lanning | 5 by 9 |
| 04:11.21 | Miller357 | Thanks |
| 04:14.47 | Miller357 | anyone running 1.6.2.15 with voicemail problems? can't forward messages, message file sequence issues as well |
| 04:15.23 | WIMPy | Miller357: Take a look at http://issuses.asterisk.org/ |
| 04:15.36 | p3nguin | issuses? |
| 04:16.03 | WIMPy | s/uses/ues/ |
| 04:16.26 | Miller357 | Yea, I have... I still a little new to the issues tracking stuff. They have a few closed issues out there for these problems. Can't tell if they are for 1.6 or 1.8 |
| 04:16.56 | Miller357 | Monitoring https://issues.asterisk.org/view.php?id=18685#bugnotes |
| 04:17.40 | p3nguin | If it is for 1.8, it shouldn't say it's for 1.6.2.something |
| 04:17.57 | russellb | Miller357, sounds familiar. Try updating. |
| 04:18.03 | russellb | Might as well go with 1.6.2.17-rc2. |
| 04:18.16 | Miller357 | where does it tell me the version? |
| 04:18.24 | russellb | hm? |
| 04:19.48 | p3nguin | Product Version 00-THIS IS NOT USED <---- this? |
| 04:19.58 | p3nguin | No clue what _that_ means. |
| 04:20.57 | ChannelZ | Maybe that it's not used. |
| 04:21.14 | Miller357 | I'm confused about it. If you look through the notes you see aragon reporting that he is not using trunk but 1.4? |
| 04:21.21 | russellb | i'm not looking at the bug |
| 04:21.29 | russellb | i'm just saying that i know there has been work in that area lately |
| 04:21.38 | russellb | so you should update since 1.6.2.15 is a couple updates behind |
| 04:21.54 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 04:22.14 | russellb | is the manager of the full time asterisk dev team |
| 04:22.21 | Miller357 | tried 1.6.2.16 and same results. |
| 04:22.27 | russellb | try 1.6.2.17-rc2 |
| 04:22.33 | Miller357 | will do |
| 04:23.33 | Miller357 | forgive me if I'm not saying this right but I checkout the latest trunk. Voice exited after password if the mailbox had a message. I assume this is work on the sequencing problem. |
| 04:24.03 | russellb | $ svn log --limit=3 http://svn.asterisk.org/svn/asterisk/branches/1.6.2/apps/app_voicemail.c |
| 04:24.38 | Miller357 | sorry, still new to the linux world.... what does that mean/do? |
| 04:25.04 | Miller357 | Does that get the latest source for that module? |
| 04:25.10 | russellb | it gives you the commit messages from the last 3 changes to the app_voicemail source code |
| 04:25.24 | russellb | 2 of the 3 last changes are related to the issues you're talking about |
| 04:25.28 | russellb | just sharing info |
| 04:25.54 | russellb | it shows you what was changed, who changed it, when, their explanation of what/why they changed |
| 04:27.37 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 04:27.44 | Miller357 | no problem.... I would like to learn more. Been running asterisk (trixbox) since 2007, latest install have been asterisk + freepbx |
| 04:27.58 | russellb | cool. you should give AsteriskNOW a try. |
| 04:28.19 | Miller357 | Yes... getting there |
| 04:28.22 | p3nguin | without FreePBX |
| 04:28.53 | russellb | i'm headed out ... good luck |
| 04:29.16 | Miller357 | That was a big help. |
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| 04:54.32 | Miller357 | quit |
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| 07:53.41 | ChannelZ | Wow. Friday is a right bore around here. |
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| 08:11.05 | wdoekes2 | morning |
| 08:18.06 | ChannelZ | indeed |
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| 08:57.34 | rue_house | aastralink pro 160 anyone? |
| 09:10.08 | drmessano | No thanks |
| 09:14.26 | rue_house | heh |
| 09:14.44 | rue_house | I'm looking to learn mroe about the network services offered |
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| 11:09.15 | saxa | hello, I have * 1.8.2.3 running now, but first thing I noted is that when I start the asterisk -rvvvvv command there is no longer the few welcome lines with version and other info. |
| 11:09.27 | saxa | is this a new behaivor ? |
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| 11:17.22 | *** join/#asterisk BlackBishop (dexter@89.43.120.25) |
| 11:17.39 | BlackBishop | I got -- User entered '0760123123' ( into ${digito} ) . Why do I get -- Executing [123@users:3] Set("SIP/500-00000000", "thing= ") in new stack having this: exten => 123,n,Set(thing=${IF($[ ${digito}:0:4 = "0760"]?"339837")} ) |
| 11:17.55 | BlackBishop | Basicaly I'm trying to check if the user entered the first 4 digits "0760" to set thing to 339837 |
| 11:22.11 | BlackBishop | yey, made it ! :) |
| 11:22.36 | BlackBishop | exten => 123,n,Set(thing=${IF($["${digito:0:4}"="0760"]?"339837")}) |
| 11:36.46 | saxa | another question is, to enable the fax in 1.8.2.3 which dependencies I need to have to compile it in ? |
| 11:39.00 | Tim_Toady | saxa u need spandsp so you can compile res_fax and res_fax_spandsp |
| 11:39.53 | Tim_Toady | its a good idea to download and install the lastest version of spandsp, dont use the one ur distro comes with |
| 11:40.18 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 11:41.55 | saxa | Tim_Toady: ok, thx, will do that |
| 11:42.35 | saxa | Tim_Toady: actually I'm even not sure, slackware is shippind that |
| 11:43.04 | Tim_Toady | i dont think so :P |
| 11:43.34 | Tim_Toady | btw there is a slackbuild scirpt somewhere for asterisk, it might help you |
| 11:43.48 | Tim_Toady | i ve seen it on the lists a few days ago |
| 11:45.33 | saxa | so is it good to go with the 0.0.6pre18 ? |
| 11:45.52 | Tim_Toady | yes |
| 11:46.03 | saxa | Tim_Toady: I have just readopted the old 1.6.1.2 slackbuild |
| 11:46.15 | saxa | since there is a small error in it |
| 11:46.26 | saxa | anyway I already got it compilled and fixed |
| 11:46.47 | saxa | are you the maintainer for the slackware rc.slackware.asterisk file ? |
| 11:47.08 | Tim_Toady | no |
| 11:47.12 | saxa | in slackware it seems asterisk is now using /var/run/asterisk dir |
| 11:47.42 | saxa | so this needs to be corrected since the rc.slackware scripts expects the pid file in /var/run/asterisk.pid |
| 11:47.58 | saxa | and therefore doesnt stop the servic3e |
| 11:48.46 | Tozz_ | fuck me in the beard. Slackware still exists? :) |
| 11:48.53 | saxa | yeah :) |
| 11:48.57 | Tim_Toady | so as a true slackware user edit the file in ed and fix it :P |
| 11:48.59 | Tozz_ | wow ;) |
| 11:49.05 | saxa | i did that |
| 11:49.08 | Tim_Toady | :> |
| 11:49.12 | saxa | lol |
| 11:49.32 | saxa | but would be nicer to have it fixed in the source of the problem :D |
| 11:49.47 | saxa | will send a patch to the ML |
| 11:59.00 | Corydon76-home | Patches are to be sent to the issue tracker, not the ML |
| 11:59.36 | Corydon76-home | and I would recommend changing the Slackware init script, not the Asterisk build system |
| 12:03.19 | saxa | Corydon76-home: ok, the script is in contrib dir |
| 12:03.24 | saxa | in asterisk |
| 12:31.11 | StaRetji | WIMPy: just inform that I was able successfully install dahdi, if I do not make with ousfxs drivers, which I unfortunately need :/ So, there is a problem with those drivers which I don't know how to solve. |
| 12:33.05 | StaRetji | if anyone with this kind of knowledge is willing to help, I would appreciated it and will repay somehow :) |
| 12:42.49 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:42.56 | AliRezaTaleghani | hi |
| 12:43.50 | AliRezaTaleghani | i need to change the place of a caller, whom is in a queue! right now |
| 12:44.31 | AliRezaTaleghani | i mean, for example, there is queue which have 10 caller in waiting stage right now |
| 12:44.54 | AliRezaTaleghani | one of the is caller no:7 and she is VIP! so i need to change her place to no:2 |
| 12:45.25 | AliRezaTaleghani | of course via AMI commands.... |
| 12:45.33 | AliRezaTaleghani | is there any? |
| 12:52.07 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
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| 13:58.49 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 14:41.49 | doolittlework | hi there, havent worked with asterisk for a while, my box was running 100% but then nature took its cource and wiped it out, my question is B410p card is detected by the dahdi driver, can one use this driver or is it best to install the misdn driver? |
| 14:42.11 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 14:42.31 | doolittlework | anyone active? |
| 14:44.17 | neurosys | is there a proper way to upgrade via source or do i just need to DL the latest tarball and upgrade from that? |
| 14:44.22 | manji | doolittlework, why don't you simply try it |
| 14:44.32 | manji | doolittlework, do some tests |
| 14:44.47 | manji | and then trey the misdn driver as well, if any |
| 14:48.37 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 14:48.51 | doolittlework | manji: building box at home isdn lines at office just reading up on it, i know i used the misdn on the forst install but that was on asterisk 1.4 and erlier dahdi, just wanted to know if they updated the dahdi drivewrs to now work with b410P card |
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| 14:56.19 | manji | doolittlework, |
| 14:56.20 | manji | http://www.voip-info.org/wiki/view/B410PInstallationInstructions |
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| 14:59.18 | *** join/#asterisk degli (~degli@host114-128-dynamic.16-87-r.retail.telecomitalia.it) |
| 14:59.19 | degli | hi all |
| 15:13.59 | saxa | hi |
| 15:15.05 | saxa | ok, again me, is there a good how-to or some instructions for * 1.8 fax setup ? |
| 15:18.12 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 16:01.13 | blee | So i just updated a polycom from 2.1.1 to 3.2.4, the time on the OSD is no longer in the middle |
| 16:01.25 | blee | and I cannot seem to find a pertinent setting to center it, anyone have this issue? |
| 16:07.11 | leifmadsen | I don't worry about where the time is on the phone as long as it's correct :) |
| 16:10.52 | doolittlework | thx manji all is working |
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| 16:13.25 | blee | leifmadsen: lol I would tend to agree, but my end users like to complain |
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| 17:06.00 | *** join/#asterisk Thorn (~Thorn@unaffiliated/thorn) |
| 17:06.18 | oblom | hi. i am using usernames instead of the extension numbers in setup. is is possible to supply it somehow to voicemail when it asks for extension number ? |
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| 17:19.13 | leifmadsen | oblom: yes, create a relationship in AstDB between the username and extension number, then look it up prior to calling Voicemail() |
| 17:19.21 | leifmadsen | ~newbook |
| 17:19.21 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 17:19.48 | leifmadsen | oblom: check out the Database Integration chapter on hot-desking for an idea. Just substitute the ODBC lookups for AstDB lookups |
| 17:20.06 | leifmadsen | oblom: alternatively just do setvar=extension=101 on the sip peer |
| 17:20.28 | leifmadsen | (although that makes the username/extension numbers static and not dynamic like it would be in a database) |
| 17:23.05 | oblom | leifmadsen: and there is no way somehow via dtmf to enter actual username ? |
| 17:23.17 | leifmadsen | oblom: how would you do that? |
| 17:23.49 | leifmadsen | you could program something that allowed them to type in 555-33-444-333 for LEIF but how effin' annoying |
| 17:24.00 | leifmadsen | it's not practical |
| 17:24.11 | oblom | leifmadsen: yeah. it's kinda what i meant :) |
| 17:24.14 | *** join/#asterisk corretico (~laguilar@201.201.44.82) |
| 17:24.15 | leifmadsen | yes, don't do that |
| 17:24.22 | leifmadsen | your users will want to kill you |
| 17:24.24 | oblom | won't |
| 17:24.35 | leifmadsen | do it the way I described as that is much more efficient and flexible |
| 17:24.44 | oblom | thank you |
| 17:25.16 | leifmadsen | you could of course: setvar=extension=55562377773366 for [lmadsen] |
| 17:25.33 | leifmadsen | I would never recommend that though |
| 17:28.17 | leifmadsen | runs off for lunch then a 2 hour drive to see family |
| 17:28.31 | *** join/#asterisk xSanon (Sanon@189.24.238.139) |
| 17:36.36 | Thorn | hello |
| 17:38.17 | Thorn | I'm getting weird behavior with music on hold. files play in random order even when random=yes is not set (if it's not set, the order is the same every time but still not the alphabetical filename order) |
| 17:38.51 | Thorn | moh show files shows files in random irder, too, and the order seems to change every time asterisk is restarted |
| 17:39.11 | Thorn | version is 1.4.21.2 |
| 17:40.04 | Thorn | I've read somewhere there're some moh bugs in this version, maybe I need to upgrade? |
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| 18:29.11 | _zoom_ | hello, anyone tested yealink phone t22 ? |
| 18:29.52 | _zoom_ | when I call yealink, it replies with busy, any idea? |
| 18:36.17 | raden_work | how do i make asterisk start on machine boot ? |
| 18:37.01 | ChannelZ | depends on your machine |
| 18:37.24 | ChannelZ | there are some init scripts in the contrib director or somewhere.. |
| 18:39.18 | raden_work | hmmm |
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| 20:01.13 | MikeH | Hi guys. I've just built a machine with an Atcom AX4G and AEX410 (1 FXO 1FXS), and was originally hoping to use askozia, but it seems it supports neither of my cards. |
| 20:01.35 | MikeH | Is there something similar and Asterisk based available that is likely to have better product support? |
| 20:01.55 | p3nguin | Digium cards would be a good bet. |
| 20:02.12 | MikeH | THe AEX410 is Digium.... |
| 20:02.24 | p3nguin | Then it should be fine with Asterisk. |
| 20:02.24 | MikeH | I'm talking about software though, I do not wish to replace the cards. |
| 20:02.55 | MikeH | Yes, I understand this :) I'm looking for something similar to Askozia - ie something a little more simple. |
| 20:03.13 | p3nguin | What is an Askozia? |
| 20:03.46 | MikeH | Asterisk based linux distribution basically. |
| 20:03.58 | MikeH | with a web based configuration interface. |
| 20:04.03 | p3nguin | You mean a Linux-based Asterisk distribution? |
| 20:04.36 | p3nguin | You could use AsteriskNOW if you are incapable of installing Asterisk on any other Linux distro. |
| 20:05.22 | MikeH | It isn't a case of inability to install - it is simplicity and easy configuration I'm looking for. |
| 20:05.36 | p3nguin | AsteriskNOW is your answer. |
| 20:05.55 | MikeH | Which distro is it based on? |
| 20:06.01 | chiwawa_42 | anyone willing to help us out on getting a software modem implementation running ? We'd need devs/hackers :s |
| 20:06.02 | p3nguin | It is built on CentOS. |
| 20:06.26 | MikeH | Ok, what makes it better/easier? |
| 20:06.37 | p3nguin | You'll also have the choice of no GUI, Asterisk GUI, or FreePBX. |
| 20:06.54 | MikeH | oh, there is a GUI. This is good. |
| 20:06.57 | p3nguin | For one, it Just Works. That's apparently what you are looking for. |
| 20:07.09 | p3nguin | It is Digium supported. |
| 20:07.10 | WIMPy | chiwawa_42: What do you try to do / in what way? And what do you need? |
| 20:07.34 | MikeH | Ok. Is there any loss of functionality by going AsteriskNOW over installing Asterisk on CentOS? |
| 20:07.36 | p3nguin | It uses official Asterisk packages, created by a Digium member. |
| 20:08.08 | p3nguin | It's exactly the same, really. If you install CentOS and then install the AsteriskNOW packages, you'll end up with the same functionality. |
| 20:08.42 | MikeH | Seems pointless me installing CentOS first then if I can do it all in one hit. Thanks for your help :) |
| 20:08.47 | p3nguin | It's just easier to install the AsteriskNOW system. From boot to making phone calls, it took me 15 minutes. |
| 20:08.53 | p3nguin | boot of the CD, that is. |
| 20:08.57 | MikeH | WOw |
| 20:09.23 | MikeH | I have something a little complicated to setup - should be fun for my first time with Asterisk. |
| 20:09.25 | MikeH | :D |
| 20:10.17 | p3nguin | I did a very basic setup because I was only interested in AsteriskNOW (since I had never used it before). It was a very nice experience. |
| 20:11.59 | *** join/#asterisk MikeH (~mike@86.63.17.141) |
| 20:13.12 | p3nguin | There's also an AsteriskNOW quick start guide to help if there are any questions. |
| 20:13.32 | MikeH | Thanks. |
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| 20:14.37 | MikeH | Hopefully I'll manage OK. I need to have analogue line for incoming calls, a selection menu, voicemail, VoIP phone and fax detection which should go off to fax machine via FXS. |
| 20:15.10 | MikeH | Oh, and to make matters complicated, calls need to be routed via a GSM card to a mobile as well as the VoIP phone |
| 20:15.29 | MikeH | All standard(ish) stuff in terms of dialplans etc. though, right? |
| 20:19.11 | IsUp | you should expand "etc" ... |
| 20:22.49 | MikeH | I don't really know what etc. - I'm planning on figuring out what needs setting up as I go along :P |
| 20:24.24 | p3nguin | It sounds pretty normal to me. |
| 20:25.14 | MikeH | great - I thought it would all be pretty standard stuff, it is a releif to hear it from someone else though :P |
| 20:30.01 | chiwawa_42 | WIMPy: we want to terminate dialup calls from v.34 modems on an IPBX |
| 20:30.14 | chiwawa_42 | without hardware modems on the provider side |
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| 20:33.47 | WIMPy | chiwawa_42: And how do you plan to connect or integrate with Asterisk? |
| 20:34.00 | chiwawa_42 | WIMPy: IAXmodem, i guess |
| 20:34.11 | chiwawa_42 | spandsp only miss v34 support |
| 20:35.04 | WIMPy | Ah, so it's about adding V.34 to iaxmodem? |
| 20:35.18 | chiwawa_42 | yes |
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| 21:57.42 | Finkregh | any idea if it is possible to setup an ppp-dialin-server /w sip-backend? as routing calls to an 'normal' modem... |
| 22:02.24 | fenrus | this should be doable |
| 22:02.38 | fenrus | though ive never seen any data about it.. :D |
| 22:06.17 | Finkregh | hmm, ppl have quite a bunch of unused sip-accounts lying around... would be quite a help for situations as in egypt... |
| 22:07.16 | raden_work | there any way to play files in a directory in order ? |
| 22:19.38 | ChannelZ | for MOH you mean? |
| 22:21.00 | Thorn | it doesn't seem to work for moh: http://lists.digium.com/pipermail/asterisk-users/2010-April/247377.html |
| 22:21.16 | Thorn | I have this problem too |
| 22:24.16 | ChannelZ | hmm seems to work here. I don't think alpha sort appeared until 1.6.? though |
| 22:25.06 | Thorn | what's the expected behavior without alpha, i.e. mode=files; directory=... ? |
| 22:25.39 | Thorn | in my case asterisk seems to load moh files in random order on startup |
| 22:26.03 | Thorn | which can be seen with moh show files |
| 22:26.30 | Thorn | then if there's no random=yes it always plays them in the same order (until next restart that is) |
| 22:26.43 | ChannelZ | with nothing I believe it just comes in at whatever order the filesystem happens give asterisk |
| 22:28.45 | Thorn | so which version should I upgrade to? I currently have 1.4.21 |
| 22:30.04 | ChannelZ | well if you're going to go to the trouble you might as well try out 1.8 |
| 22:31.51 | Thorn | that would require me to switch from zaptel to dahdi? |
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| 22:33.05 | ChannelZ | yah but it's basically the same, that's not a hard switch |
| 22:33.58 | Thorn | ok, thank you. the only thing left is to not bring my call center down in the process :) |
| 22:34.48 | ChannelZ | Yah that probably wouldn't be good. |
| 22:35.38 | ChannelZ | Some config directives have changed so it'd probably be a good idea to do the upgrade on a test machine and get as much of it working as you can |
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| 22:38.01 | fbc_ | Is there any way to disable people from choosing to be UNAVAILABLE? The sip software allows people to choose to be unavailable. |
| 22:41.41 | drmessano | Find an app that allows provisioning, like something from counterpath |
| 22:42.28 | drmessano | Otherwise, they are always going to be able to pwn the app locally |
| 22:43.23 | fbc_ | Kinda stinks. I figured there would be a way to tell the server not to allow that extension to go unavailable. |
| 22:43.51 | fbc_ | Some kinds of AWAYRING(yes/no) parameter. |
| 22:44.08 | fbc_ | *ALWAYSRING(YES/NO) |
| 22:44.12 | drmessano | It's all about what the client reports back |
| 22:44.36 | drmessano | Just like AutoAnswer is useless if the client has it turned off |
| 22:44.41 | drmessano | I can send the headers all day long |
| 22:44.51 | fbc_ | right. but ultimately there server can ignore the clients request to go unavailable. |
| 22:45.22 | fbc_ | right? |
| 22:45.49 | drmessano | The client doesn't have to "request" anything.. if I set DND on the extension, it's DND |
| 22:46.13 | drmessano | My phone is now DND and there's nothing you can do about it.. send me calls all day long, my CLIENT is DND |
| 22:46.50 | fbc_ | got it... hmmm. guess I'll lok into provisioning. |
| 22:46.57 | Thorn | fbc_: what if your people simply shut their softphone down? |
| 22:47.06 | Thorn | you can't really disallow that |
| 22:47.38 | fbc_ | Thorn, but I can easy tell that they aren;t doing their job. |
| 22:47.45 | Thorn | at least by technical means |
| 22:48.39 | fbc_ | The problem is the client comes configured by default to go unavail if the computer sleeps, or if manually selected to. So people can chat all day long it they wanted to. |
| 22:49.05 | fbc_ | I do no work. |
| 22:49.16 | fbc_ | *and do no work |
| 22:49.24 | fbc_ | I work, unfortunately. |
| 22:50.39 | fbc_ | hmm, well, thanks for the input, Thorn and drmessano . I'm gonna look for a provisioning solution. |
| 22:50.46 | Thorn | probably the best solution is to implement monitoring and look at it regularly, so youo can see how many calls are answered, how many aren't and who is (not) answering when |
| 22:51.26 | Thorn | this is what I've done anyway |
| 22:53.19 | drmessano | or tell them "Look, at the end of the month, we fire everyone here. But you have a full month to make me not fire you. WOW me" |
| 22:53.23 | ChannelZ | yah.. even if it rings they can turn the volume down and still not do any work |
| 22:53.32 | ChannelZ | You have a personel problem, not a technological one :) |
| 22:54.09 | Thorn | technology can help but ultimately it's a management problem |
| 22:54.42 | drmessano | Yep, Nihilists, man |
| 22:54.51 | ChannelZ | electrodes on chairs might be an option |
| 22:55.52 | drmessano | Nihilism in the workplace is best solved with pain* |
| 22:55.54 | drmessano | . |
| 22:56.05 | drmessano | *as permitted by HR policies and OSHA regulations |
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| 23:48.33 | MikeH | Guys, I've installed AsteriskNOW with AsteriskGUI, except when I setup the initial firewall config on first boot, it would seem I was supposed to manually allow the asterisk port |
| 23:48.38 | MikeH | is there a way to get back to this config? |
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