| 01:25.27 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 02:12.41 | GGD | eveing all |
| 02:12.44 | GGD | evening |
| 02:20.36 | *** join/#asterisk roswell (roswell@62.69.14.137) |
| 02:23.52 | roswell | hello. a question: what are syntaxic limitations on context identifiers? |
| 02:24.43 | roswell | i.e., length, allowed characters, etc? |
| 02:25.41 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:25.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:41.49 | carrar | roswell: |
| 02:41.52 | carrar | ~book |
| 02:41.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:51.15 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:51.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 07:19.02 | *** join/#asterisk datarecal (~data@loxely.illusivecreations.com) |
| 07:19.07 | datarecal | evnin |
| 07:19.18 | datarecal | are their any good dialer addons asterisk |
| 07:21.40 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:27.11 | ChannelZ | what, spammer dialers? |
| 07:27.55 | datarecal | no, i have a list of clients that we need to contact for sales |
| 07:28.15 | datarecal | but I guess yes spammer dialers GNUdialer is all I have found so far |
| 07:33.35 | ChannelZ | I'd say most probably write their own since the integration is specific to their own setup as well |
| 07:38.30 | datarecal | can you reccomend a good starting point ? |
| 07:44.54 | ChannelZ | For the Asterisk half you could interface via AMI to place calls |
| 07:45.34 | ChannelZ | What you do to talk to your database or give the 'agents' control over who they are calling etc. is entirely up to you |
| 07:48.56 | datarecal | nothing you can think of that is premade though |
| 07:55.39 | beaver_rrr | hello. I have many lines of log filled with "[Jul 30 00:20:25] NOTICE[5275] chan_sip.c: Call from '' (192.168.1.2:20123) to extension '810972598148355' rejected because extension not found in context 'default'." and i suppose its wrong thing as 192.168.1.2 is my gateway. |
| 07:56.56 | beaver_rrr | i think that if i block incoming connections from gw, my sip trunks behind it will not be able to reach my asterisk |
| 08:02.35 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 08:02.41 | linocisco | hi all |
| 08:03.03 | linocisco | hi all |
| 08:03.52 | linocisco | how to install asterisk on WRT160N without changing or wiring or tweaking hardwares inside? |
| 08:04.40 | linocisco | hello |
| 08:05.47 | linocisco | how to install asterisk on WRT160N without changing or wiring or tweaking hardwares inside? |
| 08:06.26 | beaver_rrr | linocisco: i suppose you have to install alternate firmware on it, read this: http://www.dd-wrt.com/wiki/index.php/Linksys_WRT150N_%26_WRT160N |
| 08:07.59 | linocisco | beaver_rrr, so is that asterisk firmware builtin? |
| 08:08.19 | beaver_rrr | sure it is not |
| 08:09.00 | beaver_rrr | you have to install alternate firmware that allows to install third-party software and install asterisk all by yourself |
| 08:17.38 | linocisco | beaver_rrr, how to install asterisk after that? |
| 08:18.22 | beaver_rrr | i dunno, read dd-wrt documentation, something about installation of 3rd party software |
| 08:19.00 | mariusno | http://www.dd-wrt.com/wiki/index.php/Asterisk |
| 08:19.03 | beaver_rrr | but i suppose that with such skills in information search you'll have poor luck with that :( |
| 08:22.46 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 08:29.09 | linocisco | beaver_rrr, sorry Mine are WRT110 and WRT54G2 V1 |
| 08:29.29 | linocisco | beaver_rrr, not WRT160N. sorry for my confused reading |
| 08:29.43 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 08:31.15 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 08:40.53 | *** join/#asterisk li3p (~li3p@cc-nsk.ru) |
| 08:41.12 | li3p | hi |
| 08:41.42 | li3p | who speek is russian? |
| 08:42.18 | linocisco | li3p, just type in russian, translate.google.com will help |
| 08:42.35 | li3p | haha |
| 08:44.18 | *** join/#asterisk henningb (~henning@118.175.66.17) |
| 08:45.10 | beaver_rrr | li3p: i do speak russiam, pm me |
| 08:45.19 | beaver_rrr | oh |
| 08:47.54 | linocisco | how to install asterisk on WRT110 without changing or wiring or tweaking hardwares inside? |
| 08:49.33 | *** join/#asterisk _zoom_ (~Eissa@196.1.219.122) |
| 08:49.43 | _zoom_ | hello, |
| 08:49.51 | _zoom_ | am looking for sugarcrm connector/ |
| 08:49.54 | *** join/#asterisk dandre (~daniel@81.56.122.47) |
| 08:50.01 | Roelt | linocisco, You can't. wrt110 is not supported. |
| 08:50.22 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:50.34 | linocisco | Roelt, how about WRT54G2 V1 ? |
| 08:51.46 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 08:52.03 | Roelt | linocisco, no. not enough memory |
| 08:52.24 | ChrisInSydney | Try a WNR3500Lv1 |
| 08:52.39 | ChrisInSydney | 64 RAM and a USB2 for disk |
| 08:52.57 | ChrisInSydney | either OpenWRT or D-WRT |
| 08:53.13 | ChrisInSydney | s/D-WRT/DD-WRT/ |
| 08:54.31 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 08:55.34 | linocisco | I have only WRT110 and WRT54G2 V1 |
| 08:57.57 | Roelt | well, you can't install asterisk on those boxes, you should get a faster router or something else |
| 08:59.03 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 08:59.56 | li3p | who had the joy of setting up a DAHDI TE820 |
| 08:59.59 | li3p | ? |
| 09:00.25 | li3p | in slackware) |
| 09:03.48 | Roelt | ChrisInSydney, thanks for that netgear mentoin. Nice router and not that expensive. |
| 09:08.26 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:08.30 | linocisco | Roelt, how on Linksys EA4500? |
| 09:08.58 | linocisco | Roelt, I would like to know how and where to check supported devices on sites rather than openwrt |
| 09:11.36 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:13.14 | Roelt | linocisco, you only ahve 2 options. openwrt of dd-wrt |
| 09:13.39 | Roelt | linocisco, check those sites for support on your router and make sure you have a few mb flash, or an USB port for storage |
| 09:13.45 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:20.46 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:28.09 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:28.41 | linocisco | Roelt, Linksys EA4500 has USBport but according to http://www.dd-wrt.com/wiki/index.php/Known_incompatible_devices, it is also listed as unsupported device |
| 09:30.21 | Roelt | linocisco, in that case you can't use it. the normal operating system doesn't allow thirdparty software |
| 09:40.25 | tzafrir_laptop | li3p, not that I had the joy, but any problems? |
| 09:41.33 | tzafrir_laptop | 64MB should probably be good enough for a humble Asterisk install (1-2 channels and such) |
| 09:42.04 | li3p | tzafrir_laptop >> there is a problem with setting the alarm, the provider says that our frames do not go |
| 09:42.27 | ChrisInSydney | EA4500 is no good as its a Marvell chipset = no OpeWRT support, which limits your options |
| 09:42.52 | ChrisInSydney | WNR3500Lv2 has limited builds at the moment with DD-WRT |
| 09:42.58 | tzafrir_laptop | li3p, I suppose there's nothing slackware (what version?) specific about it |
| 09:43.03 | linocisco | Roelt, here is the link for WRT54G2 V1. http://www.dd-wrt.com/wiki/index.php/Linksys_WRT54G2_v1.0. what does that mean?? |
| 09:43.06 | ChrisInSydney | Cisco E3000 / E4200v1 work well |
| 09:43.16 | li3p | 13.37 |
| 09:43.21 | ChrisInSydney | have to run |
| 09:43.26 | ChrisInSydney | good luck |
| 09:43.31 | tzafrir_laptop | Please pastebin your /etc/dahdi/system.conf |
| 09:43.53 | li3p | ok |
| 09:44.09 | tzafrir_laptop | Also post the output of lsdahdi |
| 09:44.42 | tzafrir_laptop | or at least: cat /proc/dahdi/1 |
| 09:48.28 | Roelt | linocisco, I've told you that already. |
| 09:49.01 | li3p | send in private |
| 09:51.10 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-dqfqvgiokghaglsu) |
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| 10:06.50 | tzafrir_laptop | li3p, I don't see any alarm on the spans. BTW: any reason not to post the link on this channel? |
| 10:17.48 | henningb | How do you change CID on a E1 dahdi connection? |
| 10:22.38 | li3p | henningb, in dialplan SET(Callerid(num)=you number) |
| 10:23.01 | henningb | Thank you |
| 10:24.27 | *** join/#asterisk wonderworld (~ww@dsdf-4db55fd5.pool.mediaWays.net) |
| 10:29.28 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:34.36 | linocisco | Hi all, I have a computer on which i will install asterisk .and then I will connect this to wifi router. so I think I dont need to choose any specific router, right? |
| 10:35.59 | li3p | what settings should be for the signaling in the TE820 mode user at the provider network is established |
| 10:36.05 | li3p | ? |
| 10:37.23 | WIMPy | cannot parse this question, but I do miss information anyway. |
| 10:37.37 | li3p | pri_cpe? |
| 10:38.24 | Roelt | linocisco, no, any router will do |
| 10:46.51 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 11:03.03 | li3p | yohoho |
| 11:04.13 | li3p | always update chan_dahdi when install new hardware |
| 11:09.47 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-syvdcugtszceetkq) |
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| 11:56.49 | v0lZy | lo |
| 11:57.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 12:22.45 | *** join/#asterisk bipul (~root@unaffiliated/bipul/x-4918593) |
| 12:24.45 | bipul | is there any link for beginners ? |
| 12:25.00 | WIMPy | ~book |
| 12:25.00 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:25.05 | WIMPy | Try that. |
| 12:26.28 | bipul | p3nguin, hi :) |
| 12:29.50 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-7-26.se.biz.rr.com) |
| 12:30.07 | bipul | i am looking e-books which are free |
| 12:33.37 | bipul | i want to contribute to asterisk as i am open source lover. this is my blog www.vipullinux.wordpress.com so please help me :) in learning. |
| 12:35.53 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 12:35.53 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 12:36.14 | acidfoo | bipul, you can read that book online for free |
| 12:36.36 | bipul | acidfoo, thank's |
| 12:37.08 | acidfoo | bipul, http://ofps.oreilly.com/titles/9780596517342/# |
| 12:43.02 | *** join/#asterisk Nivex (~kjotte@atlantis.home.nivex.net) |
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| 13:03.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:06.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:07.16 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 13:07.22 | mcolombo | hi all |
| 13:07.41 | WIMPy | lo you |
| 13:17.10 | *** join/#asterisk timahvo1 (~rogue@41.215.82.210) |
| 13:18.52 | bipul | http://vipullinux.wordpress.com/2012/07/30/introduction-to-asterisk-project/ |
| 13:26.21 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-7-26.se.biz.rr.com) |
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| 13:32.44 | mcolombo | anyone say if it possible to remove media attribute from sdp? |
| 13:33.20 | mcolombo | i need to remove this line : Media Attribute (a): rtpmap:8 PCMA/8000 |
| 13:33.46 | mcolombo | thanks |
| 13:34.58 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
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| 13:38.01 | pabelanger | mcolombo: disallow=ulaw |
| 13:38.06 | pabelanger | I think |
| 13:38.38 | drmessano | disallow=alaw |
| 13:44.13 | mcolombo | great! Thanks you |
| 13:47.20 | *** part/#asterisk Nebukadneza (~quassel@2a01:238:43b3:8200::3) |
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| 13:57.31 | Katty | morning |
| 13:59.39 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
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| 14:06.23 | *** join/#asterisk _zoom_ (~Eissa@196.1.219.122) |
| 14:06.42 | _zoom_ | hello, anyone works with yaai the sugarcrm connector? |
| 14:06.49 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:06.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:07.47 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
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| 14:14.35 | *** join/#asterisk fubada (~fubada@ool-4573155d.dyn.optonline.net) |
| 14:14.43 | fubada | hi |
| 14:15.05 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 14:16.06 | fubada | can someone tell me if theres an app for this: a call is placed to a support number, the call is auto parked/queued, an employee is then notified on their cellphone that there is a call waiting in the pbx and has an option to be auto connected |
| 14:17.09 | [TK]D-Fender | Originate <- AMI / CLI / dialplan app. Take your pick |
| 14:17.29 | [TK]D-Fender | Call files also |
| 14:26.05 | fubada | TK: is there such an app out of then box? |
| 14:26.09 | fubada | s/then/the |
| 14:26.20 | fubada | would I need to use a call queue? |
| 14:27.55 | Gugge | fubada: put the mobile phone in a queue, then it is notified when there is a caller (the mobile rings), and the user has an option to be auto connectd (he can pick up the phone) |
| 14:42.15 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:52.18 | [TK]D-Fender | fubada, Or something else |
| 14:52.55 | [TK]D-Fender | This is the dialplan. You do what yuo want. What you have requested is several different actions all acting in unison. This is PROGRAMMING, and what Asterisk is for. This is your job to create. |
| 14:53.37 | [TK]D-Fender | fubada, You could also use a queue and call out that was as gugge described as well |
| 14:57.50 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
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| 15:21.34 | Katty | scoooby dooo where are you?! |
| 15:22.13 | [TK]D-Fender | Rut Roh! |
| 15:25.21 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 15:26.10 | DrDamnit | When migrating between versions of asterisk (say 1.8.8.1 to 1.8.14.1) is it possible to copy the outputted file from make menuselect to the source code directory of the new versions so that I'll get the exact same compile with the new version as I did with the old? |
| 15:26.32 | *** join/#asterisk last1 (~last1@modemcable153.206-57-74.mc.videotron.ca) |
| 15:29.29 | last1 | what's the difference between core show channels and sip show channels ? |
| 15:30.21 | dax_roc | One will show all, and sip will show only sip ? |
| 15:30.44 | last1 | that's what I thought so too, but sip show channels shows a lot more than core (?) |
| 15:30.53 | last1 | sip will show like 140 and core show channels only 30 |
| 15:30.58 | last1 | is that even possible ? |
| 15:31.04 | kaldemar | "core show channels" shows asterisk channels, "sip show channels" shows SIP dialogs. they are not the same thing. |
| 15:31.36 | last1 | is it possible that sip dialogs don't end and just stack up ? |
| 15:32.00 | last1 | I have a whole bunch with format (nothing) and last message RX: Bye |
| 15:32.04 | kaldemar | SIP dialogs are not necessarily related to calls. |
| 15:32.11 | dax_roc | Can anyone point me in the right direction , I'm trying to log the Callerid to mysql, with the following http://pastebin.ca/2175884 |
| 15:32.52 | last1 | ok, I'm going to back up a bit. my asterisk crashes every once in a while and all I could find is that the number of sip channels keeps on growing |
| 15:33.05 | last1 | when it reaches about 130/140, asterisk restarts |
| 15:33.24 | last1 | the majority of the channels show what I said above: format: nothing, last message RX: Bye |
| 15:33.29 | kaldemar | for example a registration creates a dialog that is listed by "sip show channels". |
| 15:34.37 | kaldemar | if the last message is BYE, then they are likely to be call-related. |
| 15:35.05 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:35.08 | last1 | and are they supposed to linger in there ? can they cause asterisk to crash ? |
| 15:35.28 | last1 | this is on Asterisk 1.8.7.0 by the way |
| 15:35.41 | fubada | Thanks Gugge and TK |
| 15:36.16 | fubada | TK: i was just looking for a direction to minimize custom coding...a queue sounds like it will do half of what i need |
| 15:36.28 | kaldemar | last1: they're supposed to disappear. there can always be a bug in the version you're using. you should produce a backtrace upon a crash. |
| 15:37.13 | kaldemar | last1: which version are you using? |
| 15:37.22 | kaldemar | last1: oh, you said it already. |
| 15:38.14 | last1 | yes |
| 15:38.20 | last1 | I've tried 1.4, 1.6 and 1.8 |
| 15:38.47 | citywok | dax_roc: is the number field a varchar or int? callerid(all) would be something like 1115551212 <barack obama> |
| 15:42.54 | dax_roc | citywok: It's a varchar in the db. |
| 15:43.52 | dax_roc | citywok: callerid(num) would probably be more appropriate |
| 15:44.08 | citywok | gotcha. turn on debugging and watch the debug log and see what it has to say. |
| 15:44.34 | dax_roc | still learning, How do I do that again. |
| 15:45.31 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 15:46.54 | dax_roc | I'm watching the output on the asterisk daemon, It's Giving me a warning on [2012-07-30 15:42:25] WARNING[17119]: pbx.c:5397 __ast_pbx_run: Channel 'SIP/skype-00000067' sent into invalid extension '<external-phone#>' in context 'commitCallback', but no invalid handler |
| 15:46.56 | [TK]D-Fender | last1, Those are entire branches and you haven't listed an actual "problem" yet |
| 15:47.20 | last1 | D-Fender: i'm pretty sure it's this issue here |
| 15:47.21 | last1 | https://issues.asterisk.org/jira/browse/ASTERISK-19455 |
| 15:47.26 | last1 | apparently fixed in asterisk 10 |
| 15:47.49 | [TK]D-Fender | dax_roc, means what it says. It's looking for match in that context, doesn't find one, and you don't have an "i" exten to handle the exception. |
| 15:48.38 | [TK]D-Fender | last1, Well you're 7 releases out of date in your own branch |
| 15:49.00 | [TK]D-Fender | last1, 1st step before even asking anything else is to bring yourself up to current |
| 15:49.22 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
| 16:04.54 | dax_roc | Do I need to load the mysql app before use ? |
| 16:12.10 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 16:15.04 | pzn | is it possible to add a custom audio codec to asterisk? I intend to add a new codec, and "provide" a function for asterisk convert it to g711. is this too ambitious to be done? (motivation: implement in the equipment an uncompressed multichannel audio output, one of the channels take audio, others take ambient noises; then the codec-"converter" in asterisk would apply some algoritm to filter noise and echo and produce an g711 audio) |
| 16:17.40 | fubada | !docs |
| 16:17.47 | fubada | what is the url for the asterisk book |
| 16:18.04 | Qwell | ~book |
| 16:18.05 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:18.10 | fubada | thanks |
| 16:18.34 | mjordan | pzn: yes, its possible. Yes, its ambitious. |
| 16:18.36 | Qwell | pzn: sure, why not? |
| 16:19.32 | dax_roc | The book is fairly good ! Well writen. |
| 16:21.01 | Katty | your mom is fairly good. |
| 16:21.48 | Qwell | Katty: So's your FACE. |
| 16:22.03 | Qwell | whatever that means. |
| 16:22.18 | dax_roc | more like a barrel of hammers... |
| 16:29.13 | fubada | is |
| 16:29.24 | fubada | asteriskdocs.org down |
| 16:29.34 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 16:30.32 | *** join/#asterisk wonderworld (~ww@dsdf-4db55fd5.pool.mediaWays.net) |
| 16:32.03 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 16:34.08 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:37.06 | *** join/#asterisk bipul (~root@unaffiliated/bipul/x-4918593) |
| 16:39.47 | citywok | fubada: works for me |
| 16:48.40 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
| 16:58.41 | fubada | citywok: thanks its back, hows citysushi? |
| 16:59.41 | fubada | anyone familiar with asterisk queues can help :) I need to add a cellphone as the only agent/member |
| 17:00.11 | fubada | and ideally setup a rotation schedule, where cellphoneA and B rotate every week as agents |
| 17:03.08 | citywok | fubada: use local channels. local/cellphone@Provider |
| 17:03.16 | citywok | local/cellphone@outbound-context that is |
| 17:03.44 | fubada | as agent? |
| 17:04.27 | [TK]D-Fender | as a MEMBER |
| 17:06.48 | citywok | fubada: yes, as a dynamic member. you can use a cronjob to add/remove your 2 cellphone numbres weekly and alternate |
| 17:06.59 | fubada | nice :) |
| 17:07.09 | fubada | thanks guys |
| 17:09.08 | [TK]D-Fender | Add 1 static memeber and use an AstDB value to change the actual target |
| 17:09.40 | [TK]D-Fender | Better than having a screw-up leave extra members potentially logged in |
| 17:09.53 | fubada | thansk, ill have to look into asdb |
| 17:10.25 | [TK]D-Fender | "core show function DB" |
| 17:10.27 | [TK]D-Fender | ^ |
| 17:11.40 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:13.56 | citywok | TK's way is probbly better |
| 17:14.14 | citywok | TK can you do that? as a member local/${DB(USER/KEY)}@outbound-context |
| 17:14.30 | [TK]D-Fender | No, you make the membe fixed... waht it DOES is dynamic |
| 17:15.21 | [TK]D-Fender | local/dialtheguy@somecontext |
| 17:15.24 | fubada | you guys sticking around for a few right ? :) |
| 17:15.27 | fubada | im not there yet |
| 17:16.09 | [TK]D-Fender | exten => dialtheguy,1,Dial(SIP/provider/${DB(whoto/dial)},30,OPTIONSTOCONFIRM) |
| 17:16.54 | citywok | gotcha, i like that approach more than switching members, it's a little more elegant |
| 17:19.32 | fubada | if the cellphones voicemail system answers |
| 17:19.59 | fubada | nvm, i need to make my members press a digit to accept the queue call |
| 17:20.12 | fubada | otherwise their cellphone vm will play if they dont answer |
| 17:22.58 | malcolmd | digium phones users: there's a new dpma release, a new phone firmware release, and a new release of certified asterisk. |
| 17:25.02 | *** join/#asterisk GGD (~deberle@pool-173-72-206-210.clppva.fios.verizon.net) |
| 17:32.46 | citywok | fubada: yep, and make it short otherwise they will end up with a voicemail message. |
| 17:39.06 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:39.32 | _Corey_ | malcolmd: coolio |
| 17:39.44 | malcolmd | *finally* :D |
| 17:43.38 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 17:44.15 | Qwell | ! |
| 17:44.15 | fubada | citywok: is there any way thatyou know of to require queue members to press a digit before accepting the next person |
| 17:44.47 | malcolmd | Qwell: ;) |
| 17:45.06 | citywok | fubada: you would have to do it in the dialplan. if they don't enter the digit then call the other person. |
| 17:45.17 | malcolmd | that's not a dig at Qwell |
| 17:45.35 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:45.47 | Qwell | what's not, now? |
| 17:46.08 | malcolmd | oh, thought you were thinking the "*finally*" was a dig; it wasn't |
| 17:46.20 | Qwell | oh, nope |
| 17:48.40 | [TK]D-Fender | <[TK]D-Fender> exten => dialtheguy,1,Dial(SIP/provider/${DB(whoto/dial)},30,OPTIONSTOCONFIRM) <----------- |
| 17:48.46 | [TK]D-Fender | fubada, "core show application dial" |
| 17:48.56 | fubada | thanks tk |
| 17:49.22 | fubada | im workign on my queue.conf member string now |
| 17:51.29 | DrDamnit | Call to extension 8569 rejected because it is not found in the context... but it IS in the context, and dialplan show proves it. What's wrong here? http://pastebin.ca/2175916 |
| 17:55.41 | [TK]D-Fender | DrDamnit, PB another call with full SIP debug along with your actuall dialplan and SIP config for that pepr |
| 18:02.52 | DrDamnit | [TK]D-Fender: http://173.165.161.166/debug.log |
| 18:04.43 | DrDamnit | [TK]D-Fender: http://pastebin.ca/2175920 |
| 18:15.11 | DrDamnit | [TK]D-Fender: http://pastebin.ca/2175922 |
| 18:15.11 | DrDamnit | [TK]Defender: Those posts (in order) are the requested debug.log created with asterisk -rvvvvvvvvvvvv | tee /var/log/debug.log, the sip config, and the actual dialplan for that context. |
| 18:15.11 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:15.11 | fubada | Tk: im trying to set a db key/valueusing "database put supportadmin "2031234567", but i guess it also needs a "family"? |
| 18:15.11 | fubada | what is this 'family'? |
| 18:15.11 | leifmadsen | fubada: family key value |
| 18:15.11 | fubada | does that need to already exist? |
| 18:15.11 | leifmadsen | fubada: database put admins supportadmin 2031234567 |
| 18:15.11 | leifmadsen | no |
| 18:15.11 | fubada | ok |
| 18:15.11 | leifmadsen | you just need to specify it |
| 18:15.11 | leifmadsen | everything requires a family/key |
| 18:15.49 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 18:15.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.6.1 (2012/07/16), 1.8.14.1 (2012/07/16), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:16.21 | *** join/#asterisk navaismo (~navaismo@189.191.27.69) |
| 18:16.31 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:18.44 | [TK]D-Fender | DrDamnit, [Jul 30 13:49:19] NOTICE[797]: chan_sip.c:22147 handle_request_invite: Call from 'iitc-xlite' (173.165.161.161:35440) to extension '8569' rejected because extension not found in context 'iitc_local_stations'. |
| 18:18.49 | [TK]D-Fender | DRyOUR FIRST pb |
| 18:18.56 | [TK]D-Fender | DrDamnit, Your 1st PB |
| 18:19.09 | [TK]D-Fender | DrDamnit, in your new log I do not see it failing on that exten |
| 18:19.43 | datarecal | are their any good dialer addons asterisk |
| 18:19.45 | DrDamnit | I'll capture another. I may have been to quick (trying to keep extraneous stuff out of the log to make it easier for you to read)... |
| 18:20.08 | [TK]D-Fender | Looking for s in iitc_local_stations (domain 8569) |
| 18:20.17 | [TK]D-Fender | I do see this broken use of the DOMAIN |
| 18:20.19 | fubada | [TK]D-Fender: Im trying to figure out what should take place of OPTIONSTOCONFIRM in your line above, Im looking through "core show application dial" for the asnwer but Im kind of confused |
| 18:20.23 | [TK]D-Fender | but it is looking for "s" |
| 18:20.34 | [TK]D-Fender | fubada, MACRO <- |
| 18:20.37 | Katty | it is lookin gfor your mom. |
| 18:20.37 | fubada | I see WaitExten as a possible useful thing |
| 18:20.58 | carrar | MOM! |
| 18:21.31 | Katty | carrar: ohia |
| 18:21.37 | Katty | carrar: did you see the coffee cheesecake? |
| 18:21.51 | Qwell | Katty: cheesecake is totally shippable, right? |
| 18:22.02 | leifmadsen | Qwell: yes |
| 18:22.03 | carrar | no I didn;t |
| 18:22.06 | Qwell | (I'm totally joking) |
| 18:22.14 | Katty | sad day. |
| 18:22.17 | Katty | luckily, it's easy to make! |
| 18:22.22 | leifmadsen | Qwell: shipping yes; editable upon reception? no. |
| 18:22.33 | Katty | http://42ndrecipestreet.blogspot.com/2012/07/no-bake-coffee-cheesecake.html |
| 18:22.39 | Qwell | leifmadsen: meh, microwave will soften it up a bit |
| 18:22.59 | carrar | mmm cheese |
| 18:23.15 | carrar | You can make cheesecake in a rice maker BTW |
| 18:23.24 | Katty | oh probably |
| 18:23.26 | leifmadsen | o.O |
| 18:23.31 | Katty | and a crockpot, too. but this is the easy cheesecake |
| 18:23.40 | Katty | real cheesecake takes me 6 hours in the oven |
| 18:23.42 | carrar | heh |
| 18:23.46 | Katty | and i was just feeling to Meh for that. |
| 18:23.47 | Qwell | I make bathtub cheesecake |
| 18:23.52 | carrar | and has a bazillion cals |
| 18:23.53 | leifmadsen | it's easier to make it by purchasing a box of cheesecake |
| 18:23.55 | Katty | wassat. |
| 18:24.08 | leifmadsen | read 'cals' at 'cats' |
| 18:24.10 | carrar | costco cheesecake! |
| 18:24.10 | Katty | leifmadsen: but you can't purchase andys mint cheesecake with girl scout cookei crust |
| 18:24.22 | leifmadsen | Katty: made with real girl scouts? |
| 18:24.30 | carrar | cheesecake with CATS! |
| 18:24.31 | carrar | YUM |
| 18:24.41 | Katty | leifmadsen: http://42ndrecipestreet.blogspot.com/2012/02/andes-mint-cheesecake.html |
| 18:24.47 | Katty | leifmadsen: yes. with totally, real, girlscots. |
| 18:24.50 | carrar | cats with laser eyes |
| 18:24.55 | leifmadsen | stop it, I'm going to end up making cheesecake |
| 18:25.06 | Katty | nom nom cheesecake nomnom! |
| 18:25.18 | Katty | we can talk about red velvet white chocolate chip cookies instead. |
| 18:25.22 | vlad_starkov | Hello everybody! I'd be grateful if someone recommend me IRC channel where can I ask a question about sales managers salary in VOIP industry. Thank you. |
| 18:25.35 | Katty | vlad_starkov: eleventy billion dollars. |
| 18:25.39 | Katty | vlad_starkov: an hour. |
| 18:25.42 | vlad_starkov | Katty: -) |
| 18:25.47 | carrar | http://www.thecheesecakefactory.com/menu/cheesecake/godiva_chocolate_cheesecake |
| 18:25.50 | carrar | thats YUM |
| 18:25.55 | Katty | looks |
| 18:26.12 | Katty | hmm |
| 18:26.15 | Katty | i think i could make that |
| 18:26.16 | Qwell | VoIP sales manager? What does that even mean? |
| 18:26.24 | Katty | they manage...sales. |
| 18:26.26 | Katty | voip sales. |
| 18:26.29 | leifmadsen | sales of voip. |
| 18:26.32 | Katty | yes, that. |
| 18:26.34 | leifmadsen | moar voipz! |
| 18:26.42 | Katty | wait but they /manage/ the sales of voip. |
| 18:26.49 | Katty | so you basically just stand around looking, right? |
| 18:26.49 | carrar | vlad_starkov, whats the question |
| 18:26.52 | Katty | and totally not doing anything |
| 18:26.52 | carrar | we have all your answers |
| 18:26.54 | vlad_starkov | I mean the person who make direct sales PBX services to the customer. |
| 18:26.59 | Qwell | Katty: That's basically what all sales managers do, so yes |
| 18:27.03 | fubada | in soviet russia, voip sells you! |
| 18:27.04 | leifmadsen | Qwell: :) |
| 18:27.09 | Katty | vlad_starkov: comission! |
| 18:27.14 | [TK]D-Fender | vlad_starkov, the same as any other teclo guy |
| 18:27.14 | leifmadsen | Qwell: they manage the people doing the real work |
| 18:27.17 | [TK]D-Fender | telco* |
| 18:27.27 | Katty | people doing real work don't need to be managed |
| 18:27.32 | Katty | they need to GTFO so i can do my job |
| 18:27.38 | Qwell | I would say at least $60/hour. |
| 18:27.39 | carrar | heh |
| 18:27.46 | [TK]D-Fender | In Soviet Russia work manages YOU |
| 18:27.54 | Katty | work can go jump. |
| 18:28.00 | Katty | i'm too busy for this Work stuff. |
| 18:28.03 | vlad_starkov | I'm preparing new PBX service and I do some calculation about how much should I pay to sale manager |
| 18:28.23 | carrar | depends if they sell anything |
| 18:28.47 | fubada | [TK]D-Fender: did you suggest I use a macro at the end of that dial plan string to require a digit press before a call in queue is answered? |
| 18:28.50 | vlad_starkov | [TK]D-Fender: oh yeah, that's it! )) |
| 18:28.55 | fubada | and the macro is where id do that validation? |
| 18:29.17 | Katty | carrar: but they don't sell. |
| 18:29.21 | Katty | carrar: they /manage/ the sell |
| 18:29.23 | leifmadsen | s/macro/gosub |
| 18:29.26 | _Corey_ | vlad_starkov: Generally, account execs will have a smaller base salary and a larger commission-oriented revenue stream... sales managers, because they're managing and not selling per se, have higher base salaries |
| 18:29.26 | Katty | carrar: of voips. |
| 18:29.27 | [TK]D-Fender | fubada, Yes |
| 18:30.06 | _Corey_ | vlad_starkov: It absolutely depends on where you're located and what kind of customers you're selling to, as well as the overall size of the sales team |
| 18:30.10 | carrar | sounds like a fluff position |
| 18:30.31 | vlad_starkov | carrar: Say he or she sell 5 contracts per day. Each contract makes profit of 100$ per month. |
| 18:31.10 | Katty | carrar: i want a fluff position |
| 18:31.18 | carrar | Guess you should hire a buisness school graduate :) |
| 18:31.23 | Katty | carrar: darn all that pesky work that gets in the way of knitting! |
| 18:31.24 | carrar | heh |
| 18:31.26 | _Corey_ | vlad_starkov: Sounds like you're talking about a salesperson, not a sales manager ... ? |
| 18:31.38 | fubada | [TK]D-Fender: i dont haz the skills to code up a macro like that in asterisk |
| 18:31.45 | leifmadsen | moar skillz! |
| 18:32.06 | carrar | Katty, are you knitting cheesecake? |
| 18:32.10 | Katty | no coffee, no worky |
| 18:32.20 | Katty | no. i'm still working on the tardis. |
| 18:32.22 | leifmadsen | snack and coffee time |
| 18:32.36 | Qwell | leifmadsen: coffee cheesecake == have both at once! |
| 18:32.48 | Katty | i also have one heck of a headache, for some reason |
| 18:32.57 | vlad_starkov | _Corey_: ok, I'm sorry for that. You're right. I need a salesperson. Actually in Russian we call that "sales manager". |
| 18:33.23 | [TK]D-Fender | fubada, this is a MACRO. This is basic dialplan. If you're in a hurry go hire a consultant. |
| 18:33.37 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:33.37 | DrDamnit | TK[D]-Fender: It appears that the softphone was buggy. I switched to Zoiper, ran the call, and it works just fine. Doesn't make sense to me... Does that make sense to you? |
| 18:34.04 | [TK]D-Fender | DrDamnit, I already told you it was passing a funny DOMAIN... |
| 18:34.15 | vlad_starkov | leifmadsen: thanks for such a book you wrote! -) |
| 18:34.16 | _Corey_ | vlad_starkov: OK, so generally you pay them a base salary that's pretty modest and a percentage of their sales/profits depending on how much detail you want to share with them about their costs |
| 18:34.18 | [TK]D-Fender | and the only thing I saw was towards "s" |
| 18:34.30 | [TK]D-Fender | DrDamnit, Show me something new if there's something to see |
| 18:35.06 | fubada | [TK]D-Fender: not in a hurry, its jsut that * is like 1% of my job duties and i always try to cut corners on it |
| 18:35.22 | fubada | the learning curve is steep for macros |
| 18:35.36 | DrDamnit | [TK]D-Fender: Ahhh... yes... there we go. I don't think so. Now the remote box is rejecting it because of a "no authority found" error, but that's progress over what was going on before. I wasn't even getting that far before. |
| 18:36.28 | vlad_starkov | _Corey_: Yeah, I understand that. I'm just wondering, how much people pay to salesperson in cloud VOIP/PBX industry in different areas. |
| 18:37.32 | carrar | vlad_starkov, do it startup style |
| 18:37.38 | carrar | vlad_starkov, offer them stock instead of money! |
| 18:37.46 | carrar | heh |
| 18:37.47 | [TK]D-Fender | DrDamnit, well that's IAX auth issues, not SIP |
| 18:38.13 | vlad_starkov | _Corey_: for example in Russia it's ok for salesperson to have a salary about 2000-2500$ per month. |
| 18:38.39 | fubada | where in russia? |
| 18:38.45 | vlad_starkov | Moscow |
| 18:38.50 | fubada | because im from moscow and thats moscow rate |
| 18:39.01 | vlad_starkov | Or St.Petersburg |
| 18:39.01 | carrar | Isn't Mosvow a expensive place to live? |
| 18:39.04 | carrar | Moscow |
| 18:39.06 | _Corey_ | vlad_starkov: I was over there about 13 years ago... sounds like things have improved :) |
| 18:39.08 | DrDamnit | [TK]D-Fender: Yes. The original problem was with a SIP softphone while testing the dialplan to route a call over IAX to a remote box. It was telling me the extension did not exist, when, in fact, it did. That problem appears to be resolved, so now I am on to fixing the IAX auth issues. |
| 18:39.10 | [TK]D-Fender | fubada, our coding all of this for you so you can "cut corners" isn't support ; it's our doing your whole job for you. Don't expect much by way of volunteers |
| 18:39.10 | Katty | moocow, Russia. |
| 18:39.19 | carrar | MOOCOW! |
| 18:39.19 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:39.27 | vlad_starkov | -) |
| 18:39.42 | [TK]D-Fender | DrDamnit, I presume you meant that it hit that DIAL in there to go OUT IAX2... which should mean that the call does make it in... |
| 18:39.49 | fubada | [TK]D-Fender: im figuring it out though, you cant say that the * documentation is straight forward on all thiss tuff |
| 18:40.00 | fubada | theres a lot of concepts to grasp before one "just gets it" |
| 18:40.04 | [TK]D-Fender | fubada, I can't? |
| 18:40.31 | [TK]D-Fender | fubada, If you're starting from ZERO, yes... but what you asked implied having a running system processing calls already |
| 18:40.35 | vlad_starkov | _Corey_: it's unreasonably expensive place, actually |
| 18:40.41 | fubada | its by far one of the most difficult open source projects ive had to work with |
| 18:41.08 | DrDamnit | [TK]D-Fender. Yes. The DIAL app was failing when coming from the SIP softphone. When I switched to an IAX softphone, it works, and goes out the IAX2 trunk to the remote box, which is now rejecting it for auth reasons. |
| 18:41.19 | Katty | carrar: everyone seems cranky today |
| 18:41.33 | carrar | yeah whats up witdat? |
| 18:41.34 | fubada | vlad_starkov: ill be returning to moscow for the first time since 1996 this August |
| 18:41.38 | fubada | excited |
| 18:41.39 | _Corey_ | vlad_starkov: It sounds like the average salaries have some up a bit though... when I was there I met people who were making $300/mo |
| 18:41.40 | Katty | carrar: no idea :< |
| 18:41.48 | vlad_starkov | _Corey_: do you know some statistics about how many new registration per month cloud PBX service could have? Say, ringcentral.com? |
| 18:41.51 | carrar | It's happy Monday! |
| 18:41.55 | [TK]D-Fender | DrDamnit, You are very vague between your description of a SIP vs IAX call "IN" vs the fact I saw dialplan that was supposed to dial OUT. |
| 18:42.03 | Katty | distributes rainbows and sunshine. |
| 18:42.15 | carrar | gay rainbows! |
| 18:42.16 | _Corey_ | vlad_starkov: Sorry, you'd have to ask them :) |
| 18:42.20 | vlad_starkov | _Corey_: it was 2001-2003 I think |
| 18:42.23 | [TK]D-Fender | Katty, I'm so happy I do that on the toilet ;) |
| 18:42.25 | carrar | because rainbows are HAPPY! |
| 18:42.35 | Katty | and PRETTY! |
| 18:42.38 | carrar | yes |
| 18:42.44 | carrar | and COLORFULL! |
| 18:42.54 | carrar | and mathmatically correct |
| 18:43.03 | Katty | yes, they are all sorts of prismy. |
| 18:43.18 | vlad_starkov | fubada: are you russian? |
| 18:43.23 | fubada | aga |
| 18:43.33 | fubada | i grew up in ramenki and then moved to the states as a teenager |
| 18:43.47 | vlad_starkov | _Corey_: ok, thank you for your time anyway. |
| 18:44.01 | DrDamnit | [TK]D-Fender, agreed. My descriptions are not great. SIP Softphone DIAL(IAX2/iitc/8569) = fail. IAX Softphone DIAL(IAX2/iitc/8569) = win for dialplan, fail for call setup. |
| 18:44.38 | _Corey_ | vlad_starkov: No problem... i may not have answered your commission question directly. 10% is average on sales, 30% on profit |
| 18:44.39 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:44.56 | DrDamnit | [TK]D-Fender, so my original problem appears to have vanished. The auth issue I can handle. (This box is missing openssl and the remote box requires md5 / rsa authentication, etc...) |
| 18:45.38 | vlad_starkov | fubada: nice choice. It's absolutely clear for huge amount of 20-25 y.o. people, that it's better to move somewhere from Ru in nearest 1-2 years. |
| 18:45.55 | vlad_starkov | _Corey_: thanks for that |
| 18:46.13 | fubada | vlad_starkov: i moved in 1996 to chicago, i am now 28 and i want to go visit/move back if i enjoy it |
| 18:46.24 | fubada | not sure how much IT work there is msk |
| 18:46.36 | carrar | move to Hawaii and live in a grass hut on the beach! |
| 18:46.43 | _Corey_ | I want to do another vacation there... i'm sure it's changed a lot since I was there in 1999 |
| 18:46.52 | vlad_starkov | fubada: it's ok to have a business in russia and live somewhere else. London is popular place :) |
| 18:47.09 | fubada | bad teeth and poor diets |
| 18:47.31 | _Corey_ | I drank more vodka on that one trip than I believe I have in all the time since ;) |
| 18:47.45 | navaismo | wants to leave Mexico and loves vodka |
| 18:48.16 | vlad_starkov | fubada: we have no professionals. If you're one of that kind you'll be fine. |
| 18:48.33 | vlad_starkov | carrar: dream...) |
| 18:48.35 | carrar | we're all one of a kind |
| 18:48.36 | fubada | vlad_starkov: I understand that most pay is in cash/tax free, i like that |
| 18:49.24 | Katty | yes, yes we are. |
| 18:50.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-pzezhyuledwjikuc) |
| 18:51.10 | vlad_starkov | _Corey_: You should come and see. Talk with people, and probably you'll understand better about what's going on in russia last 12 years... |
| 18:52.56 | fubada | vlad_starkov: ill come and see this august the 23 :) just booked nonstop JFK-SVO for 850$ with Delta |
| 18:53.04 | fubada | a good deal for August |
| 18:53.13 | _Corey_ | Yeah, I'll definitely go back. it was one of the best places I've visited |
| 18:54.28 | vlad_starkov | fubada: Delta is much better than Aeroflot. |
| 18:54.45 | fubada | yea aeroflot was more $ anyways |
| 18:55.37 | vlad_starkov | fubada: horoshego poleta :-) |
| 18:55.44 | fubada | spasiba :p |
| 18:57.13 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 18:58.45 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 19:01.42 | bipul | My blog dedicated to Asterisk Project :) www.vipullinux.wordpress.com |
| 19:02.59 | engrxyz | fuck the french |
| 19:03.34 | bipul | engrxyz, french ? |
| 19:03.34 | [TK]D-Fender | bipul, This is How a real IP-PBX (IP04 IP-PBX) Looks like. :) <--- INCORRECT |
| 19:03.42 | bipul | really? |
| 19:03.47 | [TK]D-Fender | bipul, that clearly has ANALOG INTERFACES ON IT |
| 19:03.58 | bipul | [TK]D-Fender, can you help me in that picture . |
| 19:04.05 | [TK]D-Fender | bipul, You seem to have completely missed the "big print" |
| 19:04.41 | [TK]D-Fender | bipul, What is there to "help"? That embedded PBX you show a picture of has ANALOG PHONE LINES plugged ito it. |
| 19:04.42 | bipul | [TK]D-Fender, sory i don't get you. |
| 19:05.00 | [TK]D-Fender | bipul, Like telling me you have a car with no wheels on it ... and I SEE THE WHEELS |
| 19:05.40 | bipul | ok i got it :p |
| 19:05.57 | [TK]D-Fender | bipul, If you can't understand what I'm saying then you clearly don't understand what yuo put in your own blog post. I'd quickly believe this if you had basically jsut stolen someone elses whriting wholesale... |
| 19:06.20 | [TK]D-Fender | dangit... writing getting very sloppy today... |
| 19:07.19 | bipul | well i read articles and post my view in it :) by taking help of that article. |
| 19:07.53 | [TK]D-Fender | "Telecommunications companies that choose to ignore Asterisk do so at their peril i.e Serious and immediate danger." <- I don't see telco's in a flying panic or peril yet. This is very exaggerated |
| 19:08.16 | [TK]D-Fender | "This is because Asterisk is the ultimate hacker’s PBX" <- debatable |
| 19:09.03 | [TK]D-Fender | "hosted conferencing"? what is "hosted" conferencing? Or "non-hosted" for that matter? |
| 19:10.16 | engrxyz | lochte is a joke |
| 19:10.23 | engrxyz | bring back michael phelps |
| 19:10.32 | leifmadsen | michael phelps got lazy |
| 19:10.52 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 19:10.57 | mjordan | I was going to say, he is there... |
| 19:10.58 | xoveruk | hi |
| 19:10.59 | [TK]D-Fender | engrxyz, They can't .. he's still on display at Sea World :p |
| 19:11.43 | xoveruk | I understand that with SIP audio channels will communicate between endpoints, i.e. if I have two remote extensions connecting to a central server the server is skipped out which will reduce latency, can I do this with IAX2? |
| 19:12.18 | [TK]D-Fender | bipul, "Inside a PBX, you only need to dial three-digit or four-digit numbers to make a call to another phone in the network" <- Noone said what format your extensions have to be in. |
| 19:12.29 | bipul | [TK]D-Fender, hosted confrencing meansVideoconferencing , face to face |
| 19:13.01 | bipul | [TK]D-Fender, in my college i saw they using 3 to 4 number to connect other department. |
| 19:13.19 | bipul | and i am beginner :) so don't pull my leg i am not legend like you |
| 19:13.25 | [TK]D-Fender | bipul, In whose dictionary? Also, Asterisk does not do "video conferencing". There is no video MUX. there is no multiple views. * has AT BEST "follow the speaker" |
| 19:14.04 | [TK]D-Fender | bipul, my issue is the claims you're putting out which is showing that your content is by-and-large plagiarized. |
| 19:14.24 | [TK]D-Fender | bipul, And not even actually understood by you. |
| 19:14.24 | leifmadsen | xoveruk: transfer=yes |
| 19:15.45 | bipul | [TK]D-Fender, i saw on e-books about that |
| 19:15.46 | bipul | ok [TK]D-Fender thank you let me correct my self |
| 19:17.36 | bipul | Next time i am going to post anything about asterisk , i must show you the data :) ,and you have to make me correct. if some whear i found wrong :) |
| 19:18.53 | [TK]D-Fender | bipul, That was jsut a quick glance I evaluated. Your posts packed with exaggerations, generalizations, and some bits that are just flat-out wrong. |
| 19:18.53 | engrxyz | michael is much better than lochte |
| 19:19.01 | engrxyz | i am dissapointed with locte |
| 19:19.15 | engrxyz | i think lochte swims backward |
| 19:20.18 | xoveruk | leifmadsen: thank you |
| 19:20.52 | leifmadsen | engrxyz: #olympics please |
| 19:22.45 | bipul | [TK]D-Fender, can you sugesst me some e-books whear i can learn real asterisk knowleg. |
| 19:23.19 | Katty | maybe a guide to basic spelling, too, fender. |
| 19:23.31 | Katty | :P :P |
| 19:23.33 | Katty | bipul: <3 |
| 19:24.09 | leifmadsen | bipolar: www.asteriskdocs.org |
| 19:25.08 | navaismo | or the book |
| 19:25.57 | navaismo | ~book |
| 19:25.57 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:26.37 | [TK]D-Fender | bipul, Actually USE *. You're hoping to steal and recycle someone else's knowledge and don't have any experience to back it up apparently. |
| 19:27.08 | [TK]D-Fender | When trying to teach something to someone it really helps to actually know the subject matter. |
| 19:27.14 | bchia | This will actually get to ATDG as the 3rd hit: https://www.google.com/search?sugexp=chrome,mod=4&sourceid=chrome&ie=UTF-8&q=e-books+whear+i+can+learn+real+asterisk+knowleg |
| 19:27.59 | bipul | no it's not like that, i learn and put my stuff in my blog . simple |
| 19:28.17 | bchia | Fender has a great point. There is no substitute for hand-ons experience. |
| 19:28.54 | navaismo | or http://lmgtfy.com/?q=how+to+learn+asterisk |
| 19:36.46 | Katty | bipul: please don't send me private messages. if you have questions please ask them in here where you can recieve answers. |
| 19:37.04 | Qwell | sends Katty a message |
| 19:37.05 | *** join/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 19:37.46 | Katty | Qwell: you don't count. |
| 19:37.54 | Qwell | story of my life |
| 19:37.56 | Katty | Qwell: you're all like...1...2...5 |
| 19:38.12 | Katty | Qwell: chin up buttercup! |
| 19:38.19 | Qwell | mmm, buttercups |
| 19:39.14 | DrDamnit | Are there plans to include video in Asterisk in the future? |
| 19:39.21 | Qwell | Who says there isn't video now? |
| 19:41.14 | CrossWired | Question: I am using Asterisk.Net to Originate outbound reminder calls, but I am getting a Timeout on the SendAction. I am using Originate Async, my initial context immediately instantiates a Dial function, with a timeout of 120 seconds. I can't quite figure out where I should be investigating for the Timeout |
| 19:41.17 | Katty | your mom. |
| 19:41.40 | CrossWired | Any suggestions? |
| 19:41.49 | Katty | checks CAH deck |
| 19:42.02 | Katty | CrossWired: Jon Wilkes Booth |
| 19:42.18 | CrossWired | Katty: of course, why didn't I think of that |
| 19:42.32 | Katty | *hee* |
| 19:43.05 | CrossWired | Katty: could you be more specific as to what JWB is doing to the interwebs of my box? I know he's a jokester and all but ... : |
| 19:43.07 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.0 (2012/07/30), 1.8.15.0 (2012/07/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:43.12 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:5119:cf47:b87d:607c) |
| 19:43.24 | Katty | checks deck to see what JWB is doing in CrossWired's computer |
| 19:43.41 | Katty | CrossWired: Friendly Fire. |
| 19:44.09 | Z_God | any CAPI experts around? |
| 19:44.37 | Z_God | I'm trying to dial to an analog phone which is presented to asterisk through a CAPI controller, but I have no clue how to make it ring |
| 19:44.49 | CrossWired | Katty: Que? |
| 19:45.08 | Katty | if you don't liek the answer i can refresh you a nother one. |
| 19:45.27 | WIMPy | Z_God: You don't. You just place calls. |
| 19:45.44 | Katty | how to make a CAPI controller ring... *refresh* Crumpets with the queen. |
| 19:45.58 | Katty | crumpets fix everything! |
| 19:46.02 | Z_God | WIMPy: yep, but I cannot set up an MSN in my analog (or DECT) device |
| 19:46.18 | Z_God | so I guess there are some MSNs inherent to the controller |
| 19:46.25 | WIMPy | Z_God: Tgen place a global call. |
| 19:46.26 | bobb_WU | i'm having an issue with one of my voicemail servers. it answers normally (as seen in the CLI) but it is not transmitting sound for a period of time. it seems consistent (around 10 seconds). the problem occurs between the mitel SIP controller and the voicemail, but i do not have any trouble between asterisk nodes and the VM server. any troubleshooting steps to look at here? |
| 19:46.42 | WIMPy | Z_God: Only the device knows. |
| 19:46.44 | Z_God | hmm, maybe I can at least try that indeed... |
| 19:48.32 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:49.37 | Katty | for sale, 1 uncomfortable headache, for cheaps. will deliver. |
| 19:50.14 | Z_God | WIMPy: do you know what I need to specify as the destination then? |
| 19:51.42 | WIMPy | Z_God: nothing |
| 19:52.34 | Z_God | Couldn't call contr5 |
| 19:52.55 | Z_God | I have this Dial(CAPI/contr5) |
| 19:53.35 | Z_God | doesn't work with an extra / either |
| 19:53.56 | WIMPy | Maybe the trailing / is important? |
| 19:54.11 | WIMPy | Hmm, well. |
| 19:54.51 | WIMPy | Don't know if chan_capi will do global calls correctly. |
| 19:55.08 | WIMPy | Is it alive, BTW? |
| 19:55.14 | Z_God | I was also thinking of dtracing the controller when the normal software is using it |
| 19:55.19 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 19:55.22 | Z_God | yes, the other way around works fine |
| 19:55.32 | Z_God | I have an ISDN phone connected to controller 3 |
| 19:55.41 | Z_God | and I can dial to it from the phone on controller 5 |
| 19:56.01 | Z_God | but for this 5th controller, I have to specify 'immediate=yes' for it to work |
| 19:56.14 | WIMPy | No, chan_capi. There wasn;t much activity lately. |
| 19:56.34 | WIMPy | immediate=yes usually makes sense. |
| 19:56.39 | Z_God | yeah, it seems it's quite old |
| 19:56.45 | Z_God | there is also a rewrite |
| 19:57.01 | WIMPy | Oh. Where? |
| 19:57.06 | Z_God | sec |
| 19:57.21 | Z_God | http://www.selasky.org/hans_petter/capi4pbx/index.html |
| 19:58.33 | Z_God | I'm using the older plain 1.1.5 release now |
| 19:58.42 | WIMPy | Thanks. Need to take a look at that. |
| 19:58.48 | Z_God | no problem :) |
| 19:59.01 | Z_God | you have a lot of experience with different CAPI controllers? |
| 19:59.18 | WIMPy | But that only works up to Asterisk 1.6. |
| 19:59.25 | WIMPy | No |
| 19:59.41 | Z_God | the regular one works with 1.8 and 10 as well? |
| 19:59.50 | Z_God | I happen to be using 1.6 as well now |
| 19:59.54 | WIMPy | I just try to keep an overview of possibilities to connect Asterisk to the ISDN. |
| 20:00.20 | Z_God | are there many options for that or does it depends on the hardware? |
| 20:00.23 | WIMPy | The SVN works with 1.8, but not 10. Or that's my last state. |
| 20:00.26 | Z_God | depend* |
| 20:00.28 | Z_God | ok |
| 20:00.44 | WIMPy | But I wouldn't use anything <1.8. |
| 20:01.15 | Z_God | I'm not aware of all the improvements, but I'm just a basic user |
| 20:01.24 | Z_God | my current box is running 1.2 |
| 20:01.41 | Z_God | and its ISDN controller is an ISA card ;) |
| 20:02.27 | WIMPy | ISA? How do you connect that? |
| 20:02.46 | Z_God | the box is an old pentium 200MHz system |
| 20:02.57 | Z_God | I installed it many years ago, it still works |
| 20:03.05 | Z_God | asterisk runs fine on it |
| 20:03.06 | *** join/#asterisk gusto (~gusto@ppp-83-171-171-64.dynamic.mnet-online.de) |
| 20:03.15 | WIMPy | misdn1? |
| 20:03.30 | Z_God | no, I had to install freebsd |
| 20:03.41 | Z_God | with linux I could only use chan_modem |
| 20:03.45 | Z_God | with the hisax driver |
| 20:03.51 | WIMPy | Oh, sounds interesting. |
| 20:04.11 | WIMPy | Not ideal, but probably not a restriction when using 1.2. |
| 20:04.14 | Z_God | I installed freebsd, because it had CAPI support for this hardware (the other alternative was windows, but that didn't work so well) |
| 20:05.22 | Z_God | but this controller is only passive, I have to explicitly dial into it |
| 20:05.55 | Z_God | now I have a fritzbox from the fiber ISP, which has an active controller |
| 20:06.04 | WIMPy | active is pretty senseless anyway. |
| 20:06.36 | Z_God | I mean it supports nt_mode |
| 20:06.56 | WIMPy | That has nothing to do with active/passive. |
| 20:07.02 | Z_God | yeah sorry, my mistake |
| 20:07.14 | CrossWired | Originate with Async set to true, theoretically should not timeout, correct? |
| 20:07.16 | Z_God | I am not very familiar with that |
| 20:08.18 | Z_God | this box can run asterisk itself (mips linux) and offers 5 capi controllers |
| 20:08.54 | Z_God | nr. 3 has all the internal ISDN stuff and just works |
| 20:09.03 | WIMPy | I tried that once. But I only succeeded with the two S0 ports. |
| 20:09.05 | Z_God | nr. 5 has all the analog and DECT phones |
| 20:09.18 | Z_God | same here so far |
| 20:09.35 | Z_God | except calls from analog & DECT, those work, just need to fix to somehow |
| 20:09.58 | WIMPy | I suspected that the FXO/FXS and DECT are each one controller. |
| 20:10.48 | Z_God | controller 4 has the analog line out I read online, it also only has one B channel, so that seems right |
| 20:11.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-jiukotauxogsvybt) |
| 20:11.26 | Z_God | the first two map to the external S0 port I understand |
| 20:12.18 | WIMPy | I didn't look closer as chan_capi looked rather dead :-( |
| 20:12.48 | Z_God | would there be a way to access this hardware with another channel? (other than sip+the fritzbox software) |
| 20:13.25 | WIMPy | AFAIK the AVM hardware is not documented anywhere. |
| 20:14.23 | WIMPy | So, CAPI is the only sensible way to access it. |
| 20:15.04 | Z_God | yep... guess so |
| 20:15.37 | Z_God | I just wonder whether indeed the fritzbox has some hardcoded MSN assigned to the analog & DECT phones |
| 20:15.49 | Z_God | and if there's a way for me to find that out |
| 20:16.06 | Z_God | my dtrace debugging on another box didn't reveal so much at first sight, but I'm not sure what to look for either |
| 20:21.50 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 20:22.23 | EmleyMoor | Is there any way to see the IP an inbound SIP call is from? |
| 20:23.32 | WIMPy | ${CHANNEL(peerip)}) |
| 20:23.52 | EmleyMoor | WIMPy: I'll try that - thanks... |
| 20:32.41 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:33.23 | [TK]D-Fender | checkout time, later all |
| 20:36.36 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 20:41.38 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 20:55.34 | *** join/#asterisk millsu2 (~brad@mail.serverplus.com) |
| 20:55.41 | *** join/#asterisk gusto (~gusto@ppp-83-171-179-197.dynamic.mnet-online.de) |
| 20:56.28 | millsu2 | is it possible to force a channel to hangup if "hangup request" isn't working? |
| 20:57.39 | millsu2 | When I run the cmd 'hangup request SIP/100-0000cb56' it says Requested Hangup on channel 'SIP/100-0000cb56', but the channel never hangs up. |
| 21:08.19 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ejbujddamirrniuo) |
| 21:14.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:32.44 | ectospasm | millsu2: try soft hangup |
| 21:33.40 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-aoxuvtdypzoxpwjl) |
| 21:39.00 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 21:44.20 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 21:47.45 | *** join/#asterisk danfromuk (~IceChat77@2.27.2.78) |
| 21:56.27 | millsu2 | it's asterisk 1.8 and soft hangup says it's an invalid command |
| 21:57.25 | [TK]D-Fender | channel request hangup <- |
| 21:59.41 | fubada | TK |
| 22:00.03 | fubada | i have a dynamic member in my queue.conf and a AstDB key/val for it |
| 22:00.11 | fubada | but it doesnt get parsed on startup |
| 22:00.20 | fubada | howver, of I add it in the cli, things work |
| 22:01.07 | fubada | <PROTECTED> |
| 22:01.14 | fubada | thats works in the cli ^ |
| 22:01.52 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-64-42.pppoe.mtu-net.ru) |
| 22:03.47 | [TK]D-Fender | fubada: and I told you to add a FIXED member |
| 22:04.35 | [TK]D-Fender | Please pastebin everything you've done so I can make sense of these bits and pieces you're dropping here now |
| 22:06.28 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 22:11.43 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:13.12 | fubada | TK :) |
| 22:13.14 | fubada | coming up |
| 22:13.47 | fubada | http://pastie.org/private/2r6lspx3xj31iczkn3fzq queues.cong <- |
| 22:14.41 | fubada | http://pastie.org/private/z8m60efwijerkliiiggzg <- xtensions |
| 22:15.41 | fubada | and then in the cli: database put admins telsupport "2038071234" |
| 22:16.51 | fubada | seems to work, my cellphne rings when someone dials "7001", but not until I do " queue add member local/telsupport@outgoing to telsupport" |
| 22:17.06 | fubada | the member line in queues.conf is ignored |
| 22:30.01 | [TK]D-Fender | fubada: telsupport ... is in [outgoing], not [queues] |
| 22:32.23 | fubada | oh shit right, i had that switched recently |
| 22:32.42 | fubada | sorry you're right, still, on restart it does not gettingadded as a member |
| 22:33.03 | [TK]D-Fender | you also have no = on the member line |
| 22:33.12 | [TK]D-Fender | You seem to leave out syntax all over the place |
| 22:33.27 | [TK]D-Fender | heads out for a few hours |
| 22:33.30 | fubada | thanks TK |
| 22:35.33 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:40.46 | fubada | its working |
| 22:40.46 | fubada | thanks |
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| 22:51.20 | *** join/#asterisk anaxagoras (~gregor@p4FC7212E.dip.t-dialin.net) |
| 22:51.24 | anaxagoras | hello everybody |
| 22:51.48 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 22:52.20 | *** join/#asterisk powerunits (b6b1a980@gateway/web/freenode/ip.182.177.169.128) |
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| 22:52.36 | powerunits | dear team mates,, how is every one |
| 22:52.54 | powerunits | i have just configured voicemail to email setup |
| 22:53.17 | anaxagoras | I have recently installed asterisk on my tp-link mr3020 with running openwrt. I configured asterisk with my sipgate and google-talk accounts. I configured two users and try to authenticate but: The connections fail because they cannot register. what can be the problem? |
| 22:53.32 | powerunits | when i set it up it was working fine .. |
| 22:53.37 | powerunits | email was going through |
| 22:53.44 | powerunits | but after 2 hours i test again |
| 22:53.52 | powerunits | email is not going out... |
| 22:54.19 | powerunits | please could any one guide me |
| 22:54.24 | powerunits | where the issue can be |
| 22:56.41 | powerunits | pastebin is not working |
| 22:56.57 | powerunits | is there any other tool so i can show you my configration |
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| 23:00.47 | powerunits | any one one this room ? |
| 23:00.55 | Nivex | no |
| 23:01.24 | xSmurf | hey guys, I wanna install asterisk in a xen domU and from what I gather this requires some patching of the timing modules |
| 23:03.25 | xSmurf | now, there seems to be some confusing between dahdi and zaptel stuff |
| 23:03.38 | xSmurf | there doesn't seem to be any zaptel pkgs (in debian sqeeze) |
| 23:03.55 | xSmurf | and there doesn't seem to be any reference to USE_RTC or ZAPTEL_RATE in the dahdi source |
| 23:05.03 | xSmurf | (1.6) |
| 23:13.52 | xSmurf | is USE_RTC completely gone or something? I don't find it even in the repo |
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| 23:26.41 | shadowapex | Hi, I'm using Asterisk 1.8.12.0 (Elastix) and am currently running into a problem where Asterisk is sending the RTP audio to the wrong server. I currently have a trunk configured to go to an OpenSIPS server which acts as a signaling server and have a separate server that handles media. Right now when an inbound call comes from OpenSIPS, Asterisk is sending the RTP audio to the signaling server instead of the media server. In |
| 23:28.19 | shadowapex | I thought that Asterisk might be thinking that there is a NAT, so I set "nat=no" in the trunk configuration, but the problem is still persisting. I confirmed with a packet capture that Asterisk is sending the RTP audio to the signaling server instead of the media server specified in the SDP of the invite. Any thoughts? |
| 23:32.14 | gusto | hey ppl |
| 23:32.41 | gusto | well, someone talking about NAT<- i had same issue right now |
| 23:33.58 | gusto | i have a peer that is behind nat, and it works when he transfers on the same port in/out |
| 23:35.17 | gusto | but sometimes it happens that he comes from a port that is different form that port where he was saying that he will come from |
| 23:36.27 | gusto | a good idea would be a dealy, so that asterisk would wait till the first packets arrive and then send the packets to that port instead of sending packets to some port and getting the recieving from another port in the meantime |
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| 23:37.10 | gusto | it's probable that provides who do carrier nat - mobile broadband operators - do drop NATed port redirections very soon |
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