02:43.32 | *** join/#asterisk infobot (~infobot@rikers.org) |
02:43.32 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-rc2 (2011/11/15), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
02:43.39 | p3nguin | One way you could do it is to use an alternate extension to reach your phone, which sets the CALLERID(num) before it dials your phone. |
02:44.01 | [TK]D-Fender | j-fish: ... |
02:44.02 | [TK]D-Fender | ~book |
02:44.03 | 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 |
02:44.06 | [TK]D-Fender | ^^^ |
02:44.10 | j-fish | thanks:) |
02:44.35 | p3nguin | Instead of 600@phones, perhaps a600@phones. Then extension a600 sets any callerid info you want before it dials you. |
02:44.45 | p3nguin | That's just one idea. |
02:45.24 | SeRi | infobot: YAY!!!! |
02:45.43 | [TK]D-Fender | dijib: What do you want the CID to be? |
02:46.05 | SeRi | ok p3nguin all set. reverted back and done with my cow build..... taking a brake :) going to eat.... |
02:46.25 | p3nguin | When Originate() runs and calls his phone, it's just asterisk calling... without the real caller's info. |
02:46.43 | p3nguin | I'd imagine he just wants the real info to show up. |
02:46.51 | dijib | yes i do. |
02:46.57 | p3nguin | But I don't know how Originate will do it on its own. |
02:47.05 | p3nguin | So I'd use the alternate extension. |
02:47.24 | dijib | huh alternate extension? |
02:47.27 | [TK]D-Fender | p3nguin: If it shows up with no callerid, what do you imagine every other call he uses that peer for does? Just takes device level CID.. |
02:47.43 | p3nguin | dijib: You should have been paying attention. I'm not typing it again. |
02:47.49 | [TK]D-Fender | p3nguin: Or he's setting it in the dialplan the same way can't can't stretch that logic to this process on his own. |
02:48.11 | p3nguin | Asterisk originates a call to his phone. There is no callerid info at that point. |
02:48.17 | dijib | ok i saw it a600 |
02:48.41 | p3nguin | When Asterisk originates a call to my phone, it says External Call. |
02:48.43 | dijib | also if originate fails i need to voicemail |
02:48.56 | dijib | mines all anon |
02:49.23 | [TK]D-Fender | dijib: If originate fails... who's going to leave a VM? It didn't connect. There is nobody. |
02:49.41 | p3nguin | If you use the alt extension, you can take the caller's info from the original call and set it between the Originate() and the Dial(). |
02:49.41 | dijib | gotof |
02:50.25 | p3nguin | Originate requires your channel to be answered before it will call the other number. |
02:50.45 | p3nguin | If your channel does not answer, there is no call to the other phone, and there is no voicemail. |
02:51.05 | p3nguin | But........ |
02:51.14 | [TK]D-Fender | His Originate is backwards. |
02:51.31 | p3nguin | That's where I was going. |
02:51.34 | [TK]D-Fender | But I figured I'd leave him to beat that around a while first |
02:52.10 | p3nguin | If you reverse it, then it will call the other person first, and when that phone answers, it'll call you. |
02:52.32 | [TK]D-Fender | dijib: You Originate to your target then dump them into your dialplan. Instead you are Originating to * dialplan where you intend to answer and then trigger the callout. The Originate will always be "successful" (or fail depending on how tragically you coded it). |
02:53.31 | p3nguin | Originate(Local/${DB(callback/${CALLERID(num)})}@phones,exten,phones,600) |
02:54.54 | [TK]D-Fender | Should specify a priority in there... |
02:55.04 | p3nguin | If none, assume 1. |
02:55.10 | [TK]D-Fender | EW |
02:55.20 | p3nguin | Extensions start at 1. |
02:55.31 | p3nguin | Seems safe/sane to me. |
02:55.39 | [TK]D-Fender | ~assume |
02:55.39 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
02:55.51 | p3nguin | Alternatively, you may specify a priority if you want. |
02:56.25 | Sedorox | Anyone happen to know if it's possible to stop a Polycom IP650 from flashing the BLF light when a phone associated with that line is not registerd? (I'm not sure if this would fall on the Polycom side, or the Asterisk/Switchvox side) |
02:56.50 | p3nguin | Having the originate reversed is my fault. I have a tendency to call my side first when originating so that I hear the ringing while calling the other person. I don't like sending a call out, get an answer, and then having ringing. |
02:57.01 | p3nguin | If someone called my phone and it started ringing, I'd hang up. |
02:57.13 | p3nguin | So that's why I did the originate the way I did. |
02:57.36 | p3nguin | It's an easy change, which I already listed, to turn it around. |
02:58.43 | [TK]D-Fender | Sedorox: Change the indication pattern in your provisioning |
02:59.04 | p3nguin | I'm going to go have some tomato soup and grilled (baked) cheese sandwich, so I'll be back soon. |
03:00.16 | [TK]D-Fender | p3nguin: he's doing an automated callback to the caller. 2 ideas depending on the goal. If he wants to just let the caller call in free as inbound, then caller first, then internal. If he wants an agent to call him back and know that the agent is there then the other idea works. |
03:00.21 | Sedorox | I don't think I can change that manually (at least easily), although I'm running Switchvox 5.1.2, which is handing the phone provisioning |
03:00.25 | dijib | thanks p3nguin that was very helpful |
03:00.30 | dijib | and [TK]D-Fender |
03:00.38 | dijib | im going to switch the context now. |
03:00.39 | [TK]D-Fender | Sedorox: That's what you've got to do. |
03:05.17 | Sedorox | wonder if that is dug in the GUI somewhere.. I wouldn't have a problem manually editing the stuff, I just want to make sure if there were any changes done on the system, that it carried over |
03:07.27 | *** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
03:08.13 | [TK]D-Fender | Sedorox: I can't imagine any PBX GUI would cover anything like this. You are cheating standard indications... |
03:08.36 | Sedorox | good point :) |
03:09.56 | Sedorox | for this setup, the majority of extensions aren't going to be a problem (hardphones in the office), however they have about 5 right now that will most likely be softphones, and not always registered |
03:10.11 | Sedorox | so lines flashing on the sidecars I'm sure will get annoying |
03:10.13 | Sedorox | but we'll see |
03:12.44 | [TK]D-Fender | Sedorox: I manually do my receptionist's directory and was negligent for months (as in lots). We had a lot of turnover in that time and people changing ext #'s, Her phone was a Christmas Tree before I got off my ass to fix it up right... |
03:14.01 | Sedorox | lol, nice |
03:14.43 | [TK]D-Fender | 3 loaded sidecars |
03:15.31 | Sedorox | ouch |
03:15.37 | Sedorox | this is only one luckily |
03:16.42 | [TK]D-Fender | Sedorox: Do you have proper filesystem access on your box and to that folder? |
03:17.59 | Sedorox | mm most likely |
03:18.13 | dijib | this callback is ready for production. shall i pastebin? or you guys dont care. |
03:18.16 | dijib | ? |
03:18.31 | [TK]D-Fender | Sedorox: then you could simply override them in the <mac>-phone.cfg |
03:18.51 | [TK]D-Fender | dijib: We already know how to do it... just tell us if you need help on failure :) |
03:27.24 | dijib | its working solidactually. |
03:27.30 | dijib | like this |
03:27.33 | dijib | now to populate. |
03:27.44 | dijib | which will save me like x7 the cost. |
03:27.50 | dijib | of the toll free for known callers |
03:29.08 | dijib | ok ive got one for you, how do you dial an outside did and then send an extension argument for that did's pbx? |
03:31.01 | [TK]D-Fender | dijib: "core show application dial" or on answer do it in your dialplan you toss them into |
03:31.11 | [TK]D-Fender | "core show application senddtmf" |
03:43.18 | dijib | that sounds more like it |
03:48.13 | SeRi | bbl finishing kids bday stuff (Angry Birds Theme) |
03:48.24 | *** join/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com) |
03:48.48 | Flyingbull | Good evening everyone:) |
03:50.21 | *** part/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com) |
03:50.42 | *** join/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com) |
03:51.23 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
03:51.38 | Flyingbull | Hi there, I was wondering if someone can help me figure out an error I'm getting with dahdi -- during the install it gives me an odd error, and I can't seem to figure out how to get around it. |
03:52.12 | ChannelZ | like what error |
03:53.42 | Flyingbull | One moment, my screen crashed over there. I'll give you the exact error: |
03:54.54 | Flyingbull | You do not appear to have the sources for the 2.6.18-028stab091.2 kernel installed. |
03:56.47 | Flyingbull | I thought about going and getting the sources Centos.org, but I wanted to verify that it wasn't something else entirely. I did read something about grabbing the source code that there was a patch for this problem -- but I can't seem to find a place to find that source code. |
03:57.17 | [TK]D-Fender | Flyingbull: Seems to say pretty clearly that you are missing your kernel source & headers |
03:57.39 | [TK]D-Fender | Flyingbull: yum install kernel-headers |
03:57.45 | Flyingbull | Nope, I thought that was the problem and I went and checked, I had the kernel source and headers. |
03:58.16 | [TK]D-Fender | Flyingbull: re-run ./configure |
03:58.45 | [TK]D-Fender | Flyingbull: IIRC that is also sometimes given if you are missing ncurses-devel, etc |
03:58.52 | [TK]D-Fender | check the full dependency list for * |
04:01.08 | Flyingbull | Ok, I had done that, but I'll go through it again, to make sure I got everything. brb |
04:07.27 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
04:14.54 | Flyingbull | Ok, what I have to wonder, does FreePBX even need the dahdi part for a pure voip network? |
04:15.10 | ChannelZ | only for MeetMe |
04:15.49 | ChannelZ | though I beg you not to go down the FreePBX road |
04:17.20 | *** join/#asterisk gajini (~root@61.12.17.170) |
04:17.33 | Flyingbull | What is that? I'm setting up a "simple" IVR with it, so I was thinking it would work better. I don't think I can do Queues with Asterisk -- with annoucnements every 30 seconds saying where you are in the queue.. |
04:17.41 | Flyingbull | what s/b Why. |
04:18.07 | Flyingbull | Not by tomorrow morning anyway ;) |
04:20.41 | ChannelZ | That's how it starts, then you want to do something cool and you're stuck trying to figure out why FreePBX keeps erasing your configs |
04:23.14 | Flyingbull | Well that is danger when using frameworks. You become an expert of the framework not the underlining technology. While I've taken the time to read about Asterisk, time is of the essence at the moment. |
04:23.40 | Flyingbull | make menuselect |
04:23.54 | Flyingbull | wront window -- damn multiple monitors screw me up once in a while. |
04:24.22 | ChannelZ | Maybe you should just download AsteriskNOW |
04:25.17 | Flyingbull | Using a remote server, or I would have. |
04:27.35 | [TK]D-Fender | ChannelZThat's how it starts, then you want to do something cool and you're stuck trying to figure out why FreePBX keeps erasing your configs <-- jumping the gun on this... |
04:28.11 | [TK]D-Fender | Flyingbull: I don't think I can do Queues with Asterisk -- with annoucnements every 30 seconds saying where you are in the queue.. <- sure you can |
04:28.44 | p3nguin | Maybe he can't. |
04:29.06 | p3nguin | Asterisk can with ease, but that can't speak to his capacity. |
04:29.27 | Flyingbull | I can't at the moment, I've only been playing with Asterisk for a few weeks overall really. |
04:29.29 | [TK]D-Fender | can != going to on his own |
04:29.54 | p3nguin | The sample queues.conf has pretty good commenting. |
04:30.01 | [TK]D-Fender | != guaranteed immediate |
04:33.24 | Flyingbull | Ok, well it seems to me that would require a good knowlege of the dial plan as well, because I have the tree to consider: if it is during a certain time, then announce that the queue is closed. |
04:34.13 | [TK]D-Fender | Flyingbull: All easy to do. |
04:34.27 | [TK]D-Fender | Yourself or using FreePBX |
04:36.06 | Flyingbull | Interesting. Well I don't like Framework from the perspective that they do their own stuff. I guess I forego some sleep and look at that, the first question I need to figure out, do you have a sample of where I can look at an IVR for Asterisk? It seems like google insists on sending me to the paid for advertising sites lately. |
04:36.41 | [TK]D-Fender | Flyingbull: it's in the book |
04:36.44 | [TK]D-Fender | ~book |
04:36.44 | 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 |
04:37.10 | [TK]D-Fender | Flyingbull: Dump call in context. Call WaitExten. Have options they can dial in the context./ |
04:37.53 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-bccupyjiphguomaq) |
04:38.27 | Flyingbull | Well the plan is that they have the option to join the queue, leave a message, check store hours or hangup. I mean it is pretty simple. There will be 15 operators in the queue. |
04:39.44 | Flyingbull | The next headache is setting up a predictive dialer. |
04:43.20 | [TK]D-Fender | Flyingbull: Decidedly less volunteers for that one... |
04:43.56 | ChannelZ | have fun with the treefrog |
04:44.24 | [TK]D-Fender | Flyingbull: as for queue options... exiting to leave a message is certainly doable. checking store hours you'd have to rejoin the queue and lose your place, so better to ask before going to queue. Also.. if you don't let them in outside of hours... you shouldn't need to mention the hours. |
04:46.08 | *** join/#asterisk radic (~radic@dslb-178-002-225-173.pools.arcor-ip.net) |
04:48.10 | p3nguin | During business hours, I don't offer playback of hours. When the call comes in, you have a chance to enter a person's extension or pressing a single digit for the directory. If you choose to not ignore those, you head off to the queue, where phones start ringing. |
04:48.39 | p3nguin | if you choose to ignore, rather |
04:49.16 | p3nguin | Outside of hours, you there is a menu choice for playback of hours. |
04:49.21 | Flyingbull | [TK]D-Fender Thanks for the book, already figured out part of the problem with Dahdl. Well the idea was that the store hours they can hit 1 to speak with an agent, 2 for store hours and directions. The deal is, it is for a pharmacy, and sometimes people just want to know they can come in to pick up their perscriptions. otherwise they need to talk to someone to renew it. |
04:49.57 | [TK]D-Fender | Flyingbull: then make the hours an IVR option before hitting the queue (which is a separate options |
04:50.22 | Flyingbull | 1 >> to speak with an agent is the queue. |
04:54.24 | *** join/#asterisk troyt (~troyt@c-24-10-222-127.hsd1.ut.comcast.net) |
04:55.48 | [TK]D-Fender | Flyingbull: So "Hours" is not a queue exit option, only "Leave a VM" |
04:56.44 | *** join/#asterisk dijib (~root@bas10-kitchener06-1279681924.dsl.bell.ca) |
05:09.18 | *** join/#asterisk razu_ (~razu@195.222.7.35) |
05:19.29 | connex | [TK]D-Fender, sorry, had to get some sleep in. The tech support is helpful as my Spanish. |
05:20.48 | *** join/#asterisk irroot (~gregory@197.172.59.133) |
05:26.03 | jercos | nomnomnomasterisk |
05:33.53 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
05:46.26 | SeRi | p3nguin: cower -d b43-firmware |
05:46.27 | SeRi | cd b43-firmware |
05:46.27 | SeRi | makepkg |
05:46.27 | SeRi | su - root |
05:46.31 | SeRi | pacman -U /home/builds/b43-downloads/b43-firmware-5.10.56.27.3-2-i686.pkg.tar.xz |
05:46.53 | SeRi | exit |
05:47.02 | SeRi | p3nguin: and now we are up and running :) |
05:48.15 | p3nguin | Easy? |
05:49.38 | SeRi | yes sr! |
05:50.14 | p3nguin | If you'll configure sudo for pacman, you'll be able to skip three of those steps. |
05:50.38 | p3nguin | You'll run makepkg -i and it will do the rest for you. |
05:51.18 | SeRi | nice |
05:51.34 | Flyingbull | Corrrect -- if someone hits 3 for store hours, they get returned to the main root, afterwards. |
05:51.46 | p3nguin | or you can use packer and skip like four or five steps. |
05:51.48 | Flyingbull | Anywy, thanks for your help, I'm going to see if I can implement it. |
05:51.53 | Flyingbull | later. |
05:51.59 | *** part/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com) |
05:53.56 | p3nguin | I think I've decided that I probably shouldn't skip a second night of sleep in a row, so I'm heading off in a few minutes. |
05:54.47 | SeRi | p3nguin: head to bead bro. have a good night sleep |
05:55.13 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
06:01.25 | SeRi | p3nguin: I am out for teh night as well. I am connected to the wirless with my laptop... Now all of my devices are officially arch :) |
06:01.40 | SeRi | Sleep and have a g/n |
06:01.56 | SeRi | g/n #asterisk ! |
06:09.46 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
06:12.53 | *** join/#asterisk coppice (~chatzilla@m121-202-59-61.smartone.com) |
06:18.12 | dijib | g/n SeRi |
06:23.17 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
06:33.53 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
06:41.54 | *** join/#asterisk irroot (~gregory@197.168.83.75) |
06:52.15 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
06:54.38 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
06:55.47 | *** join/#asterisk BuenGenio (~Gene@203.145.92.172) |
06:57.02 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
06:57.47 | dym | mornings |
07:15.34 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
07:27.16 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
07:28.54 | *** join/#asterisk irroot (~gregory@196-210-140-50.dynamic.isadsl.co.za) |
07:30.43 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
07:35.03 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
07:35.07 | gavimobile | is it a bad idea to build a newer version of say libri when an older version is already installed? |
07:47.40 | *** join/#asterisk Takapa (vegard@svanberg.no) |
07:52.21 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
07:53.36 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
07:59.31 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
07:59.49 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
08:00.41 | *** join/#asterisk stix (~stix@193.89.191.209) |
08:01.34 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
08:02.57 | *** join/#asterisk ChannelZ (channelz@burner.com) |
08:04.35 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
08:14.19 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
08:15.45 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
08:16.49 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
08:22.13 | ollii | gavimobile: sounds like an update |
08:22.34 | ollii | if you install a newer version of libpri you might need to recompile * and dahdi |
08:36.59 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
08:41.09 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:1566:5343:ebbc:4a96) |
08:42.15 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
08:42.20 | gavimobile | ollii: thanks |
08:42.25 | gavimobile | that's what I did! |
08:42.28 | *** join/#asterisk Nasga (~Nasga@110.113.117.78.rev.sfr.net) |
08:46.22 | dym | Any way I can prevent this? ERROR[23893] res_jabber.c: PubSub Server error, 503 |
08:46.37 | dym | occours quite frequently (1.8.7.1) |
08:47.04 | *** join/#asterisk irroot (~gregory@196-210-140-50.dynamic.isadsl.co.za) |
08:47.22 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
08:58.18 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
08:59.58 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
09:04.15 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
09:10.28 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
09:10.46 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
09:14.15 | pa | question: is it possible/easy to configure asterisk such that it creates a sort of "com" device, in order to let legacy fax software work with the pbx as it was an old landline? |
09:14.21 | pa | or an old modem? |
09:16.26 | bulkorok | try iaxmodem |
09:16.39 | pa | ah thanks! |
09:16.48 | ollii | http://www.voip-info.org/wiki/view/Asterisk+IAXmodem |
09:16.56 | bulkorok | :) |
09:18.01 | ollii | could someone tell me about queue holdtime ? after reading * src code it seems like an avg value...is that value persistent or is it reseted after a restart of * ? |
09:18.06 | ollii | asterisk 1.4 and 1.8 |
09:22.12 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
09:23.47 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
09:27.55 | *** join/#asterisk jetlag (jetlag@pool-71-168-248-89.cmdnnj.east.verizon.net) |
09:32.42 | *** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk) |
09:35.31 | *** join/#asterisk turtlefence (~turtlefen@CPE-144-132-156-210.eypg1.ken.bigpond.net.au) |
09:38.39 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
09:40.20 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
09:40.54 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
09:42.01 | verywiseman | in "call files" that is using for automatic dialout , can i put many extensions with suitable priority for each ? |
09:42.44 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
09:43.36 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
09:51.30 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
09:54.35 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
09:58.58 | *** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk) |
09:59.58 | *** join/#asterisk jkroon (~jkroon@196.25.195.42) |
10:00.40 | jkroon | hi guys, not sure if i'm just missing something or what is going on, but is it possible to get dynamic (realtime) sip peers from automatically re-apearing after an asterisk restart (probably a crash)? |
10:02.45 | bulkorok | what do you mean? Do the clients should register after crash? |
10:03.09 | jkroon | i have a peer, it registers so now I can send calls to it. |
10:03.26 | jkroon | asterisk -rx "core restart now" ... now the peer is no longer registered to asterisk |
10:03.34 | jkroon | and as a result I can't send calls to it. |
10:04.24 | jkroon | heck, sip reload is sufficient. under normal operation this won't happen, however, crashes do (unfortunately) happen and I'd like to provision for them. |
10:04.34 | bulkorok | the client can not know this. so you have to tell the client to register all 2 minutes or so. |
10:04.39 | jkroon | https://issues.asterisk.org/jira/browse/ASTERISK-6591 (they guy's tone is very harsh, but same problem) |
10:05.36 | jkroon | bulkorok, i don't want the client to have to reregister, i'd like to simply have asterisk store it's realtime peers as part of some database (either the rt database it got loaded from, or the astdb) and automatically reload them as part of any reload (permitting they haven't expired yet) |
10:06.40 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
10:07.45 | *** join/#asterisk turtlefence (~turtlefen@CPE-144-132-156-210.eypg1.ken.bigpond.net.au) |
10:07.48 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
10:08.13 | bulkorok | jkroon: I know that in extconfig you can configure the table "sipregs" where the reg-informations will be stored. maybe this can help you |
10:08.31 | jkroon | sipregs? i'll def take a look at that thanks! |
10:08.39 | jkroon | sounds like what i need. |
10:08.47 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
10:13.04 | *** part/#asterisk gajini (~root@61.12.17.170) |
10:14.54 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
10:20.13 | *** join/#asterisk tomer67 (~tomer@95.143.243.2) |
10:20.30 | tomer67 | hi all |
10:21.07 | tomer67 | I want to create some rule or macro, which will be perform conference handling |
10:21.37 | tomer67 | I want to have only one number, 5000 |
10:22.22 | tomer67 | when I call to it, I should type PIN and after that accoridng with my PIN should be transfer to properly conf room |
10:22.28 | tomer67 | is it possible? |
10:22.49 | tomer67 | one number multiple conf room ? |
10:25.53 | dym | tomer67: yes, of course. you just start the "meetme" application and then define conference rooms in the config |
10:26.21 | dym | check out meetme.conf in /etc/asterisk |
10:26.38 | tomer67 | dym: but it isn't only one to one? |
10:26.49 | tomer67 | one exten one conf room |
10:26.51 | dym | tomer67: depends on your dialplan |
10:27.03 | dym | you can explicitly set Meetme(1000) for one conference room |
10:27.19 | dym | or you can just MeetMe to have the caller insert a number |
10:27.45 | dym | Well MeetMe() (: |
10:28.03 | dym | But the rooms have to be defined on meetme.conf |
10:28.23 | tomer67 | brb |
10:29.54 | dym | tomer67: http://www.voip-info.org/wiki/view/Asterisk+config+meetme.conf |
10:34.35 | *** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com) |
10:37.26 | *** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com) |
10:38.51 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
10:41.22 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
10:45.05 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
10:45.34 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
10:48.06 | *** join/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au) |
10:53.53 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
11:02.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
11:14.08 | *** join/#asterisk Lovelu (BanglaCafe@202.126.124.58) |
11:14.08 | Lovelu | [ GrEEtiNgS EvERyOnE ] |
11:15.13 | Lovelu | hi |
11:15.32 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
11:16.00 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
11:18.45 | *** join/#asterisk fromol (~n1x@mail.orient-logic.com) |
11:19.02 | fromol | hi guys , anyone is experienced in dlink fxo fxs configuration? |
11:19.03 | dym | Lovelu: lovely. Hi. |
11:19.13 | fromol | i have one simple problem but can't understand |
11:19.32 | dym | ~ask |
11:19.32 | 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. |
11:19.59 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
11:20.14 | fromol | okeya , |
11:20.23 | fromol | i have problem about outgoing calls on fxo |
11:20.42 | fromol | when im ringing to number call is going different number and added additional number |
11:20.54 | fromol | but where im cant see |
11:23.43 | Lovelu | can any one help me out transcoding to g729 to g23 |
11:25.45 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
11:26.16 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
11:26.43 | *** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
11:27.21 | Lovelu | hmmm |
11:27.22 | defswork | I have a very strange problem - got a site on 1.6.2.20 with 20+ extens and every now and then all the extens show no service and asterisk thinks they are "NOT IN USE" - the networking is fine as * shows pings times still and I can ping etc.. |
11:27.47 | defswork | I stop/restart * every night just to see if it was a resource issue but it has had no effect |
11:28.52 | defswork | I'm wondering if its a handset issue - they are all aastras - the same model |
11:29.29 | *** join/#asterisk dom| (~domi@mail.tas.de) |
11:29.58 | Lovelu | hmmmm |
11:30.38 | tomer67 | dym: thank you |
11:30.44 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
11:31.56 | tomer67 | but in my configuration I have configured me room from 9100-9200 |
11:32.46 | tomer67 | and when I call on external number, for example 123800 |
11:33.07 | tomer67 | after type of PIN it should be transfered to properly meeting room |
11:33.17 | tomer67 | so call to 123800 |
11:33.31 | tomer67 | pin 5555 ->room 9100 |
11:33.41 | tomer67 | pin 6666 -> romm 9150 |
11:33.43 | tomer67 | etc. |
11:33.59 | tomer67 | I don't know if it is clear for you |
11:39.30 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
11:42.00 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
11:42.44 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
11:48.38 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
11:50.55 | *** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com) |
11:54.01 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
11:57.07 | *** join/#asterisk AmirBehzad (~behzad@86.57.4.5) |
12:04.02 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
12:12.33 | *** part/#asterisk AmirBehzad (~behzad@86.57.4.5) |
13:10.21 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
13:19.24 | *** join/#asterisk lcat (~lcat@187.45.254.174) |
13:20.06 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
13:24.56 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:1566:5343:ebbc:4a96) |
13:26.20 | *** join/#asterisk james_zhu (~Administr@113.91.166.60) |
13:26.56 | *** join/#asterisk bintut (~bintut@111.65.29.43) |
13:27.06 | james_zhu | hello |
13:27.24 | james_zhu | i have a problem with digium 2 port E1 |
13:27.25 | james_zhu | <PROTECTED> |
13:27.34 | james_zhu | anyone what is the problem? |
13:30.54 | *** join/#asterisk FainaUkraina (~Gene@cm61-15-218-59.hkcable.com.hk) |
13:32.09 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
13:35.24 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
13:37.51 | james_zhu | 328]: l4isup.c:1559 t22_timeout: T22 timeout (No "circuit group reset acknowledge" from peer) CIC=1. |
13:37.51 | james_zhu | [Dec 5 21:37:05] NOTICE[23328]: l4isup.c:1559 t22_timeout: T22 timeout (No "circuit group reset acknowledge" from peer) CIC=17. |
13:39.32 | james_zhu | 328]: l4isup.c:1559 t22_timeout: T22 timeout (No "circuit group reset acknowledge" from peer) CIC=1. |
13:39.33 | james_zhu | ===anyone knows that? give me help |
13:41.47 | gordonjcp | is there a GUI for asterisk that will show pretty diagrams of call states as you dial up extensions and stuff? |
13:42.00 | gordonjcp | it doesn't have to be particularly practical or useful |
13:42.04 | WIMPy | james_zhu: Maybe you should tell us what kind of line you've got. |
13:42.08 | gordonjcp | it just has to impress a PHB |
13:43.04 | WIMPy | gordonjcp: Like FOP or astman? |
13:43.24 | gordonjcp | WIMPy: possibly, let me google |
13:44.15 | oej | Asterisk is 12 years today! Happy Birthday! |
13:44.49 | gordonjcp | WIMPy: that sounds like it'll do |
13:45.19 | gordonjcp | WIMPy: I need to do a swift bit of baffling with bull organic fertiliser product |
13:45.20 | WIMPy | Pretty old and still so young. |
13:49.58 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-fpkvlndilsxkledc) |
13:54.48 | *** join/#asterisk irroot (~gregory@196-210-140-50.dynamic.isadsl.co.za) |
13:56.36 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
13:59.24 | jkroon | hi guys, I've asked this before but seemingly can't find it now. COUNTRYCODE=27, so on an outbound call to 2712... I want to replace the leading 27 with a single 0, however, this needs to work generically, so the COUNTRYCODE needs to remain in a variable. |
14:00.30 | jkroon | in dialplan (if COUNTRYCODE was static) I could do exten => +27!,1,Goto(0${EXTEN:2},1) but alas, as soon as the 27 is stored in a variable it's a tad trickier. |
14:00.45 | *** part/#asterisk james_zhu (~Administr@113.91.166.60) |
14:00.46 | WIMPy | No problem as long as it is a global variable defined in your dialplan. |
14:01.21 | WIMPy | You can do exten => +${COUNTRYCODE}. |
14:01.34 | jkroon | it is, so I can actually do exten => _${COUNTRYCODE}!,1,... awesome, let me test! |
14:01.57 | WIMPy | Missed the _, sorry. |
14:02.16 | jkroon | s/+/_ :p |
14:02.34 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
14:02.49 | WIMPy | You had +27 in your example, so I intended to write _+... |
14:03.23 | jkroon | hehe, and I intended to write _27 to begin with, not +27 (those are stripped off long before) |
14:03.56 | WIMPy | And how do you know if it's an international number then? |
14:04.54 | *** join/#asterisk as001 (~uros@82.117.198.142) |
14:05.16 | *** join/#asterisk serafie (~erin@nat/digium/x-ylqjachnxsmvczle) |
14:05.38 | as001 | Hello is it possible to configure to playback sound message to client which is in Queue during his conversation to Agent ? |
14:07.22 | jkroon | WIMPy, i always deal with the calls internally in full international format |
14:07.29 | [TK]D-Fender | as001, Not though any normal process. You'd have to brigde in a local channel to use chanspy, etc to do this |
14:07.37 | jkroon | so if I get 0XYZ I replace the leading 0 with COUNTRYCODE |
14:07.51 | jkroon | if I get 00Z or +Z I just strip off the leading 00 or + |
14:07.57 | jkroon | so far it works quite well. |
14:08.10 | jkroon | and I don't have to deal with NANP |
14:09.56 | as001 | Can I redirect or transfer that call to some new extension where I will have just Playback and then to return that call to same agent who transfered call ? |
14:10.33 | [TK]D-Fender | as001, Depends how you dial the agent |
14:10.56 | as001 | agent is sitting in queue after AgentLogin |
14:12.00 | [TK]D-Fender | as001, Membermacro |
14:12.34 | as001 | ok thanks |
14:12.52 | *** part/#asterisk as001 (~uros@82.117.198.142) |
14:20.31 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
14:23.42 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
14:26.34 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
14:32.22 | *** join/#asterisk mzahariev (adminimini@piem-nafta-v.unixsol.org) |
14:35.40 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-ukknxitskcjokjzn) |
14:35.44 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-ukknxitskcjokjzn) |
14:38.00 | *** join/#asterisk jastrup (jastrup@login.konstant.no) |
14:38.41 | jastrup | Do any of you have any recommendations regarding a switchboard application for asterisk? |
14:39.42 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
14:40.59 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il) |
14:42.01 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
14:42.40 | [TK]D-Fender | jastrup, FOP / FOP2 , astassistant |
14:45.24 | davlefou | hi, |
14:45.52 | davlefou | is it possible to show an history of the communication? |
14:46.55 | WIMPy | davlefou: Ask the NSA. |
14:49.19 | dym | or KGB |
14:49.45 | WIMPy | Do they still exist? |
14:49.57 | [TK]D-Fender | davlefou, what kind of history? For what communication? |
14:50.25 | davlefou | [TK]D-Fender: yes! |
14:50.27 | chuckf | WIMPy: by that name, I don't think so |
14:50.34 | [TK]D-Fender | davlefou, EXACTLY |
14:50.53 | WIMPy | But for telecommunication, Mossad is definitely 1st place. |
14:50.55 | dym | WIMPy: course they do |
14:51.02 | dym | you |
14:51.10 | dym | you're so uninformed! |
14:51.15 | *** join/#asterisk ZogG_n900 (~michael@213.8.57.217) |
14:51.17 | ZogG_n900 | hello |
14:51.31 | dym | Hi ZogG_n900 |
14:51.38 | dym | whats cracking? |
14:52.00 | dym | (except from your telephony) |
14:52.01 | ZogG_n900 | i have a lot of warrning kinda "NOTICE[4363]: chan_sip.c:9510 check_auth: Correct auth, but based on stale nonce received from ...." |
14:52.23 | dym | i suggest removing asterisk |
14:52.30 | dym | would certainly get rid of that problem |
14:52.31 | ZogG_n900 | dym, i hope for better future let's say like that |
14:52.41 | ZogG_n900 | dym, yeah i know =) |
14:53.02 | WIMPy | dym: Are you a medical doctor? |
14:53.12 | ZogG_n900 | i googled the problem, it said that it's releated to snom certain setting while i have audio codec |
14:53.23 | davlefou | [TK]D-Fender: i want to looks about historique about the techincal message. |
14:53.23 | ZogG_n900 | WIMPy hope not |
14:53.23 | dym | WIMPy: Philosophy |
14:53.30 | ZogG_n900 | no head no pain =) |
14:53.48 | WIMPy | Yes :-( |
14:53.56 | dym | davlefou: /var/log/asterisk ? |
14:53.58 | [TK]D-Fender | davlefou, That does not describe anything specific. |
14:54.05 | WIMPy | davlefou: You need to be more specific if you hope for any sensible answer. |
14:54.32 | [TK]D-Fender | HOW CAN I EVERYTHING!?!? :S |
14:54.43 | WIMPy | Just do it! |
14:55.05 | davlefou | ok, i install and i want to know if my call had works well with asterisk! I start to looks log! |
14:58.11 | dym | WIMPy: works for nike! |
14:58.19 | dym | KNEWIT |
15:05.10 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
15:05.31 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
15:08.35 | *** join/#asterisk jboy1010 (~jhash1010@bas1-montreal19-1177815033.dsl.bell.ca) |
15:09.49 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
15:09.58 | jboy1010 | urgent: Need some help fixing what i beleive is a firewall problem with my asterisk server. Can't hear anything on any side of calls after a call is connected, willing to pay through paypal for help |
15:13.00 | jboy1010 | anyone here? |
15:14.48 | jkroon | jboy1010, tcpdump is your friend. |
15:19.46 | jboy1010 | i'm really new at this, i'll try looking into tcpdump, thanks |
15:22.04 | WIMPy | Has anyone ever seen the screening indicator "user provided, veryfied and faild" being used anywhere? If so, in what situation? |
15:22.17 | Qwell | ~nat |
15:22.17 | infobot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
15:22.19 | Qwell | jboy1010: ^^^ |
15:22.41 | Qwell | Feel free to donate to the FSF once you've fixed your problem with that info. |
15:24.49 | jboy1010 | well its weird because everything was working great until i tried to install openfire onto my asterisk box |
15:25.00 | jboy1010 | (elastix pbx) |
15:25.22 | jboy1010 | so i had to enable port 9090 and ever since then calls can connect but there is no voice transfer happening |
15:25.31 | jboy1010 | Nat is enabled in config files |
15:25.43 | Qwell | "enable" how? |
15:26.11 | jboy1010 | nat=yes in trunk settings |
15:26.18 | Qwell | no, the port 9090 |
15:26.26 | jboy1010 | oh |
15:26.40 | jboy1010 | i ssh'ed into my box |
15:27.00 | jboy1010 | and used "system-config-securitylevel-tui" |
15:27.20 | Qwell | and in the process probably blocked your NAT ports. |
15:27.53 | jboy1010 | i did not change any setting that where already there, all i did was add a new setting for 9090 port forward |
15:28.07 | Qwell | 9090 falls in the default NAT range |
15:29.41 | *** join/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net) |
15:29.46 | jboy1010 | i wasn't able to access the openfire installation dialogue before what i did, so after forwarding port 9090 openfire was ok but my calls arent working |
15:30.06 | jboy1010 | this is in the elastix gui |
15:30.09 | Qwell | yes, because adding 9090 killed your NAT ports |
15:30.51 | jboy1010 | well i tried removing the same rule i added but it didn't help.. |
15:31.21 | Qwell | no, it wouldn't. Since it overlaps, I'm betting it removed any other conflicting (read: NAT ports) rule. |
15:32.04 | jboy1010 | so i have to reset my firewall rules? |
15:32.08 | Qwell | maybe |
15:32.09 | leifmadsen | ...and this is why we test on development servers and write deployment procedures |
15:32.42 | *** join/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net) |
15:33.23 | jboy1010 | anyone willing to help? its pretty urgent, i can pay through paypal |
15:33.40 | Qwell | pastebin your iptables rules |
15:34.49 | jboy1010 | sure |
15:35.22 | *** part/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net) |
15:35.22 | [TK]D-Fender | Qwell, what range is 9090 interfere with? Stock sample RTP is 10k-20K, and its nowhere near SIP... |
15:35.24 | *** join/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net) |
15:36.15 | Qwell | [TK]D-Fender: my coffee is still too hot. |
15:36.19 | Qwell | my point stands though |
15:36.52 | *** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net) |
15:36.58 | tuxxie | if a call is transfered does the ${UNIQUEID} get reassigned? |
15:37.10 | jastrup | [TK]D-Fender: FOP2 is not an alternative, astassistan might be, but is there any others? Maybe enterprise ones? |
15:37.53 | [TK]D-Fender | jastrup, HUD / HUDlite |
15:39.49 | jastrup | [TK]D-Fender: thanks |
15:42.09 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
15:42.30 | jboy1010 | Qwell, http://pastebin.com/YRRpRhsg |
15:42.57 | Qwell | it's not a firewall issue then. check your router. |
15:43.24 | Qwell | Your default policy is accept. |
15:43.35 | Qwell | also, you have your NAT ports using tcp, which is wrong |
15:45.53 | [TK]D-Fender | s/NAT/RTP |
15:46.09 | Qwell | COFFEE IS TOO DAMN HOT |
15:46.19 | Qwell | more caffeine needed, stat. |
15:46.33 | jboy1010 | i don't get it.... these firewall settings worked perfect before. I didnt change a thing, except for the openfire isntall. |
15:46.45 | Qwell | shrugs |
15:47.10 | jboy1010 | 100$ paypal? |
15:47.17 | WIMPy | Don't use shiney tools that do what they want without you knowing. |
15:47.23 | Qwell | jboy1010: See above, re: FSF |
15:47.41 | Qwell | or I can PM you a link for DWB |
15:48.11 | jboy1010 | sure |
15:48.26 | [TK]D-Fender | Qwell, http://www.quickmeme.com/meme/35f1p8/ |
15:48.33 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
15:48.39 | Qwell | [TK]D-Fender: heh, nice |
15:51.49 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
15:53.26 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
15:55.51 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
15:55.54 | wcselby | o/ |
15:57.38 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
15:58.06 | akrohn | does anyone know how compatible 1.8 and freebsd are these days? |
15:59.43 | pabelanger | akrohn: should work, we use it for testing our remote bamboo build agents |
16:03.14 | carrar | not the BAMBOO BUILDS!!! |
16:05.27 | wcselby | anyone know the default max size of a CNAM record? |
16:05.55 | *** join/#asterisk ik_5 (~ik@109.226.17.43) |
16:06.00 | ik_5 | hello |
16:06.11 | akrohn | excellent. thanks pabelanger |
16:06.52 | ik_5 | i have an incoming call from a SIP trunk that arrive with the proper caller id, but Asterisk sends invite back with From: "Anonymous" <sip:Anonymous@anonymous.invalid> . how can i figure out why ? |
16:07.18 | Qwell | wcselby: 255 |
16:07.26 | pabelanger | carrar: vs? |
16:07.32 | Qwell | I'm wrong. 255 * 63? |
16:08.08 | wcselby | i was trying to figure out max char that is retrieved from a CNAM dip, but thanks :) |
16:08.19 | Qwell | that's a completely different question |
16:09.08 | carrar | VS ALIEN! |
16:09.11 | Qwell | The theoretical limit for CNAM is bigger than valid response limits. |
16:09.15 | r0m|u | waz up wcselby. God is cold! |
16:09.15 | [TK]D-Fender | ik_5, sHOW US |
16:09.40 | [TK]D-Fender | dern capz... |
16:09.41 | wcselby | Qwell obviously |
16:09.42 | Qwell | wcselby: It would be invalid to return something that isn't a valid record (A, MX, etc) |
16:09.44 | wcselby | r0m|u sup |
16:09.52 | wcselby | Qwell wait what? |
16:10.01 | wcselby | CNAM, not CNAME |
16:10.03 | Qwell | CNAM is a pointer. It has to return another record. |
16:10.07 | r0m|u | wcselby: nothing much man just cold... :( |
16:10.13 | wcselby | callerID name |
16:10.18 | wcselby | afk, stupid meeting |
16:10.20 | Qwell | umm |
16:10.25 | Qwell | don't use the word record then. :P |
16:10.51 | ik_5 | [TK]D-Fender, show what exactly ? |
16:11.08 | defswork | willy ? |
16:11.54 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
16:14.30 | [TK]D-Fender | ik_5, These calls so we can compare |
16:14.35 | [TK]D-Fender | ik_5, * CLI w/ SIP DEBUG |
16:14.37 | [TK]D-Fender | ~pb |
16:14.38 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
16:14.39 | [TK]D-Fender | ^^^ |
16:15.57 | *** join/#asterisk oej (~olle@87.96.134.129) |
16:17.12 | *** join/#asterisk ChannelZ (channelz@burner.com) |
16:18.31 | *** join/#asterisk jetlag (jetlag@pool-71-168-248-89.cmdnnj.east.verizon.net) |
16:19.17 | [TK]D-Fender | bai bai |
16:20.16 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
16:20.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
16:23.30 | *** part/#asterisk mzahariev (adminimini@piem-nafta-v.unixsol.org) |
16:24.45 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
16:26.12 | *** join/#asterisk mandla (~quassel@168.167.180.161) |
16:28.58 | *** join/#asterisk navaismo (~navaismo@189.230.118.2) |
16:40.00 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
16:44.21 | *** join/#asterisk irroot (~gregory@41.51.173.137) |
16:44.38 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
16:47.02 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
16:47.13 | IsUp | hello |
16:51.15 | *** join/#asterisk oej (~olle@87.96.134.129) |
16:53.49 | r0m|u | quiet morning it seems. |
16:55.24 | leifmadsen | apparently all the asterisk systems are running fine |
16:55.54 | r0m|u | hehehe lol :) |
16:55.58 | r0m|u | indeed |
16:58.26 | WIMPy | Or everyone os busy fixing them. |
16:59.02 | WIMPy | is |
17:12.22 | *** join/#asterisk eZz (~ez@178.137.178.86) |
17:12.42 | eZz | hi |
17:12.44 | *** part/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net) |
17:19.25 | eZz | guys I have a problem with originate from AMI to SIP channels. |
17:19.31 | eZz | When I'm sending a request to Local channel, with some context and exten and Dial to SIP - it's fine (I see ... Executing [s@dialout:1] ...) |
17:19.39 | eZz | But when I'm using SIP channel, I'm getting ... Executing [@dialout:1] ... |
17:19.48 | eZz | I don't understand, is it a bug or I |
17:19.58 | eZz | I'm wrong |
17:20.19 | WIMPy | Show us what you send. |
17:20.40 | eZz | do you mean Channel,Context,Exten in AMI request ? |
17:21.07 | eZz | one second please, will publish on pastebin |
17:21.12 | WIMPy | The whole request |
17:21.25 | eZz | sure, one moment pls |
17:23.35 | *** join/#asterisk darkskiez_ (~dz@cpc4-broo7-2-0-cust167.14-2.cable.virginmedia.com) |
17:32.27 | eZz | WIMPy: http://pastebin.com/FkCg7kkV |
17:32.32 | eZz | here is a link |
17:35.43 | WIMPy | doesn't spot anything obvious. |
17:36.54 | ZogG_n900 | <PROTECTED> |
17:37.06 | ZogG_n900 | what is this error about i get a lot of output |
17:37.14 | ZogG_n900 | any ideas tips? |
17:37.18 | ZogG_n900 | audio-codes phones |
17:37.21 | oej | You're device is trying to authenticate based on an old challenge-response |
17:37.29 | WIMPy | But it looks like it execues , even if it doesn;t say so. |
17:37.34 | oej | It needs to read the current auth message and respond |
17:38.07 | ZogG_n900 | as i get every second from all exts |
17:38.28 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
17:38.34 | eZz | WIMPy: even I tried to use _. to catch everything and DumpChan and I did not found an extension too |
17:38.43 | ZogG_n900 | may it ruin the quality as it uses traffic and actually it's not good? |
17:38.55 | Qwell | eZz: _. is 1 or more characters. |
17:39.10 | ZogG_n900 | oej should i update fw or is there way to fix it? |
17:39.17 | eZz | yes I know, it should be s |
17:39.31 | WIMPy | eZz: But you second log also shows s being executed ifven if it says [@.. in the log. |
17:40.44 | Qwell | eZz: Are you sure that the entire originate action is being received by Asterisk? |
17:40.46 | eZz | yes but it is a strange... |
17:40.56 | eZz | Qwell: yes |
17:41.18 | eZz | btw that was working on 1.6.some |
17:41.45 | eZz | now I'm using 1.8.7.1 |
17:43.24 | Qwell | Are you willing/able to test earlier versions of 1.8, to see if you can figure out where it started happening? |
17:43.25 | oej | I don't know if there's an update of your phones. It's the phone firmware that retries an old authentication nonce |
17:43.43 | Qwell | it might also be useful to try the latest 1.8.8.0 RC |
17:44.40 | eZz | I will be able to test it but some later. Now I have to use Local trick to localize a problem for now |
17:45.11 | eZz | also I will test IAX the same way |
17:45.22 | WIMPy | eZz: Why do you think it does't work? From your logs both attemts look successfull. |
17:46.33 | eZz | WIMPy: I don't like to see that it's working 'somehow'. It should work as designed but not somehow |
17:47.33 | eZz | there are 2 cases: 1) I'm wrong or doing some incorrect ways, 2) it's a bug |
17:47.41 | WIMPy | I don;t see any issue except for an inaccuracy in the console output. |
17:47.43 | eZz | nothing else |
17:48.19 | eZz | ok, I will do another test, one sec pls |
17:48.49 | *** join/#asterisk dijib (~root@bas10-kitchener06-1279681924.dsl.bell.ca) |
17:48.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
17:49.01 | *** join/#asterisk honree (~honree@net2.icemans.co.uk) |
17:49.02 | honree | hi |
17:49.20 | honree | is it possible to change the callerid for console orginated calls? |
17:49.39 | honree | it seems i can change the name but not number (which shows as unavailable) |
17:50.03 | honree | i do this : |
17:50.05 | honree | exten => 229,1,Set(CALLERID(name)="TEST NAME") |
17:50.05 | honree | exten => 229,2,Set(CALLERID(num)="1234") |
17:50.22 | honree | test name appears on the phone's display below 'UNAVAILABLE' |
17:50.49 | honree | if i call from a sip fone it displays the callers's number (222 for example) |
17:50.50 | [TK]D-Fender | honree, You are never supposed to put quotes for those |
17:50.55 | eZz | WIMPy: ok, I did another test. I changed 'Extension: s' to 'Extension: 9529'. It should hit '_X.' isn't it ? |
17:51.02 | eZz | but it's not |
17:51.05 | wcselby | i hate stupid meetings |
17:51.09 | honree | ok let me try |
17:51.11 | [TK]D-Fender | eZz, No |
17:51.17 | wcselby | this one is still going on |
17:51.29 | WIMPy | honree: What TK said, and you might have to set CALLERID(pres)=allowed. |
17:51.36 | [TK]D-Fender | eZz, Well not in that context. show a complete sample please |
17:52.06 | eZz | Executing [9529@dialout:1] NoOp("Local/9529@from-dialers-8afe;1", "Incoming call to 9529") in new stack |
17:52.06 | honree | unquoted 1234 works :D |
17:52.12 | eZz | why it's hits on Local ? |
17:52.18 | honree | is possibel to send alpha as well as num in that field? |
17:52.18 | eZz | [TK]D-Fender: it's to you |
17:52.36 | eZz | in this case: |
17:52.36 | eZz | exten => _X.,1,NoOp(Incoming call to ${EXTEN}) |
17:52.41 | WIMPy | honree: yes |
17:52.42 | [TK]D-Fender | eZz, You are not showing us the whole picture and I don't trust the little dialplan segment youa re showing us |
17:53.01 | honree | doesnt work :( |
17:53.13 | [TK]D-Fender | honree, Fix it right and show us |
17:53.19 | honree | unquoted numeric works |
17:53.28 | eZz | what whole picture ? |
17:53.31 | [TK]D-Fender | honree, Good, that is what you should be doing |
17:53.38 | eZz | ok it seems need to search myself |
17:53.42 | honree | what? |
17:53.52 | [TK]D-Fender | eZz,It's clearly executing things you haven't shown us, in odd contexts. Dump larger portions |
17:54.11 | [TK]D-Fender | nvm |
17:54.13 | [TK]D-Fender | scratch that |
17:54.22 | [TK]D-Fender | eZz, Ok, what wrong with it now? |
17:54.41 | WIMPy | I fail to see anything going wrong so far. |
17:54.55 | honree | wimpy : was your earlier 'yes' directed at my clid issue? |
17:55.03 | WIMPy | honree: yes |
17:55.17 | honree | ive tried text in the num field and it doesnt seem to work |
17:55.38 | [TK]D-Fender | honree, And no reason to imagine why it should. It says number, so give it a number. |
17:55.48 | WIMPy | honree: Must be your phone then. |
17:55.57 | honree | i wasnt imagining - i was responding to what wimpy said |
17:56.19 | honree | tkd-fender can you see what wimpy is typing? |
17:56.53 | honree | wimpy it's the gigaset :} |
17:57.04 | WIMPy | Well, as "number" in SIP usually means account name, non-numeric characters aren't uncommon there. |
17:57.17 | WIMPy | Then the gigaset doesn't like it. |
17:58.24 | honree | okey |
18:12.45 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
18:14.41 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
18:18.41 | eZz | [TK]D-Fender: ok, I published another one strange log. See it: http://pastebin.com/gTBEAEtz |
18:18.51 | eZz | my brain is going crazy |
18:19.10 | eZz | I'm sure it's my hands but I haven't a clue where is an error |
18:20.48 | eZz | this is a full dump, not a chunks |
18:21.01 | *** join/#asterisk gpearson (~gspearson@fw1.niesc.k12.in.us) |
18:22.37 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il) |
18:23.59 | eZz | ok nevermind, will figure out myself... |
18:24.21 | *** join/#asterisk autofsck (~que@unaffiliated/autofsckk) |
18:24.25 | [TK]D-Fender | eZz, I see a lot of missing parms in your AMI calls, no context, no priority, etc... Yuo are filtering things and I'd jsut as soon see your actual code that calls it |
18:25.24 | eZz | [TK]D-Fender: I did not listed them. The full params was listed here: //pastebin.com/FkCg7kkV |
18:25.41 | eZz | oops, http is missing, http://pastebin.com/FkCg7kkV |
18:26.13 | eZz | [TK]D-Fender: on that link you can see all the params that was not listed |
18:28.12 | [TK]D-Fender | eZz, Show me your actual code that is generating the last call |
18:28.18 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-vmryrbkmhaycokpz) |
18:30.13 | eZz | [TK]D-Fender: why you need this code ? the code is written in python, is working on a production the last 2 years on 1.6 and so on... Ok, I will paste a code |
18:31.02 | [TK]D-Fender | eZz, Also when in doubt do a tcpdump on manager. We need to see this raw. Hand-made reconstructions often fail to represent what's really happening. |
18:31.14 | [TK]D-Fender | I can accpt just a TCP dump rather than raw code |
18:31.20 | [TK]D-Fender | at least that's what * really sees |
18:33.07 | eZz | [TK]D-Fender: http://pastebin.com/dQTyS5dm |
18:33.48 | eZz | note: varList is empty so nothing is adding there |
18:34.54 | [TK]D-Fender | eZz, Ok, new attempt with TCPDUMP would do it... |
18:36.20 | eZz | ok I think it's just a waste a time... Will figure out myself but some later... [TK]D-Fender, WIMPy, anyways thanks guys for help |
18:36.32 | [TK]D-Fender | ok... |
18:37.07 | eZz | the problem is somewhere on the surface but my tired brain can't find it... Just need to rest and find it |
18:38.52 | [TK]D-Fender | Rest is always a good idea... |
18:39.05 | eZz | yeah |
18:39.16 | [TK]D-Fender | If you're too bombed out you'll burn yourself even further while missing tings |
18:42.25 | r0m|u | I learn that the hard way :/ |
18:44.10 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
19:05.11 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
19:07.41 | gpearson | On a brand new * install 1.8.7.1, is it common to have to press # to dial an extension? If I dont then it takes 10 seconds to dial an extension, if I do then the extension rings instant. Where should I look as I am learning on the Fly. |
19:08.22 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
19:09.24 | navaismo | gpearson: timeout in the phones dialplan |
19:10.18 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-fxzwnhvrjemdovvo) |
19:14.11 | Qwell | p3nguin: You're slacking. |
19:14.45 | *** join/#asterisk pietro (~pietro@88-149-227-4.dynamic.ngi.it) |
19:20.18 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
19:22.18 | *** join/#asterisk ASUChander (~asuchande@fl-71-52-2-90.dhcp.embarqhsd.net) |
19:22.52 | *** join/#asterisk pdtpatrick (~pdtpdt@12.249.4.226) |
19:23.04 | vader-- | whats the average time you guys have seen from verizon for porting a number? |
19:23.04 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
19:23.18 | pdtpatrick | Question .. lets say ur PRI goes down or ur service provider d/c your account. Can Voicemail still work? |
19:23.31 | ASUChander | Hello all. I'm trying to port a number away from a VoIP provider (FlowRoute), but the CLEC is telling me that the number (which is an NC 919 number) is outside of the "NC voice footprint" so they have to charge an additional $25/month for service for the ported number - are they just trying to bill me extra? |
19:23.37 | pdtpatrick | for instance can another server connect to the voicemail app using IAX or SIP ? |
19:24.00 | ASUChander | FlowRoute has told me they support porting the number "away" from flowroute.. |
19:30.41 | gordonjcp | what's an "ur PRI"? |
19:31.08 | Qwell | I imagine it's like a sub-par US PRI. |
19:31.15 | Qwell | us-- pri |
19:32.15 | gordonjcp | is it like a primordial PRI? |
19:32.38 | gordonjcp | urstoff, primordial matter, that which classical matter is made of |
19:32.42 | vader-- | ASUChander hehe I am actually moving a customer over to flowroute |
19:32.55 | ASUChander | flowroute is great |
19:33.05 | vader-- | how come you are moving away? |
19:33.08 | ASUChander | Been with them for awhile, but I have a customer that wants to move away |
19:33.11 | ASUChander | It's not me... |
19:33.25 | ASUChander | But they're screaming at me because of this $25/month fee that TWC wants them to pay |
19:34.05 | vader-- | im actually tying to port two of my buddy's numbers over from verizon... trying to figure out how long that is going to take |
19:35.16 | pdtpatrick | gordonjcp, i was talking about my landline provider |
19:36.10 | leifmadsen | Qwell: I asked him to turn off that auto-join-announcement when cbwest joins |
19:36.23 | Qwell | leifmadsen: boo :p |
19:36.27 | leifmadsen | cbwest: are you a bot, or do you actually talk? |
19:36.59 | gordonjcp | pdtpatrick: oh, okay, I just wasn't sure if an "ur PRI" was some groovy new presentation I hadn't played with yet |
19:37.03 | Qwell | I've never seen it talk. It disconnects a lot, which makes sense being on Ciscos network. |
19:37.16 | leifmadsen | Qwell: heyo! |
19:37.35 | gordonjcp | Qwell: his switch power supply is probably on fire |
19:37.37 | gordonjcp | again |
19:38.10 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
19:40.12 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-ivpegsbsjzlqdoys) |
19:43.22 | *** join/#asterisk linuxplatform (~centoslin@88.87.48.115) |
19:44.22 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il) |
20:07.29 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
20:07.48 | *** join/#asterisk andygraybeal (~andy@h28.215.22.98.dynamic.ip.windstream.net) |
20:08.31 | andygraybeal | what is a good voip provider, for personal use.. mainly just to learn how to run asterisk and have a voip provider? |
20:09.18 | [TK]D-Fender | andygraybeal, To learn, absolutely anything will do and doesn't even have to hit the PSTN |
20:09.40 | [TK]D-Fender | andygraybeal, SIP is SIP. The fact it might hit the PSTN doesn't change the quality or range of the test |
20:09.46 | andygraybeal | ok |
20:09.52 | andygraybeal | thanks [TK]D-Fender |
20:10.19 | [TK]D-Fender | andygraybeal, Ekiga.net for in/out testing. ipkall for a free DID in a few zones in washington, etc |
20:10.41 | andygraybeal | nice |
20:12.19 | wcselby | I use flowroute, they're nice, cheap, prepaid |
20:12.29 | wcselby | only need like 20 to open an account with them I think |
20:12.40 | wcselby | and the monthly cost of a did is like 1.49 I think |
20:12.55 | wcselby | i always end up throwing 20 buck sinto the account every 2-3 months |
20:13.01 | wcselby | and that's for my home number / business line |
20:13.23 | wcselby | sorry, all of that was directed towards andygraybeal |
20:18.25 | *** join/#asterisk heffer (~felix@fedora/heffer) |
20:21.11 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
20:21.23 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-nztnxcpgnkxysahk) |
20:21.52 | andygraybeal | wcselby: ah thanks man |
20:22.29 | andygraybeal | wcselby: i grew up with some selby's here in southeastern ohio ;) |
20:22.44 | andygraybeal | i'll look up flowroute |
20:25.12 | *** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net) |
20:31.15 | *** join/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au) |
20:43.17 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il) |
20:43.35 | vader-- | andygraybeal i like flowroute, they give you 25 cents to test your system with... Good rates and yuo can be setup with a DID in like 5 mintues |
20:49.31 | *** join/#asterisk fprior (c8317ffd@gateway/web/freenode/ip.200.49.127.253) |
20:55.56 | andygraybeal | vader--: nice i booked marked it - i will keep it in mind. thank you guys. |
20:56.18 | andygraybeal | who spells bookmarked like that? |
20:56.21 | andygraybeal | anyway thanks again. |
20:58.49 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
20:58.52 | DelphiWorld | 'Lo all |
20:58.57 | DelphiWorld | do asterisk support the AMR codec ? |
21:03.22 | fprior | hi all, what about MixMonitor, is the correct way to record all calls ? Is mandatory use StopMixMonitor() for each call ? |
21:04.53 | r0m|u | p3nguin: ping |
21:07.07 | DelphiWorld | ack r0m|u Dynamic fake firewall:P |
21:07.58 | *** part/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
21:12.22 | leifmadsen | I wish delphiworld would actually help himself once in a while and actually look |
21:16.18 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
21:16.20 | *** join/#asterisk ariel_ (u3533@pdpc/supporter/active/abatista) |
21:16.42 | ariel_ | Hello everyone |
21:17.40 | ariel_ | Is there a way from the dial plan to setup an extension that lets you reload asterisk or the queues? Like doing an asterisk -rx "dialplan reload" |
21:18.23 | p3nguin | Yes. Create a new extension that runs that command via System(). |
21:19.25 | p3nguin | leifmadsen: He's blind, so looking might be rather difficult. |
21:19.55 | ariel_ | I was hoping not to do it via a system() call but directly from the dial plan |
21:20.00 | leifmadsen | p3nguin: perhaps "look" was the wrong word; I meant investigate |
21:20.16 | p3nguin | ariel_: Do it from the dial plan. Use System(). |
21:20.26 | leifmadsen | ariel_: only via AGI(), SHELL(), System() etc. |
21:20.38 | leifmadsen | there is no DIALPLAN_RELOAD() function |
21:20.53 | p3nguin | They are dial plan apps, so choose one and use it. |
21:21.03 | ariel_ | OK, t/y I am already doing it via a system() call. |
21:21.25 | p3nguin | If you're already doing it in the dial plan, what's the problem? |
21:21.40 | ariel_ | did not want to put that much load on it. |
21:22.00 | p3nguin | qwell: leifmadsen had me turn off the announcement for cbwest. |
21:22.17 | Qwell | p3nguin: I saw that. What a nub. |
21:22.37 | p3nguin | ariel_: Are you sure you understand what you're saying? |
21:22.45 | ariel_ | yes |
21:22.51 | p3nguin | I have my doubts. |
21:23.15 | ariel_ | System(/usr/sbin/asterisk -rx reload) |
21:23.23 | p3nguin | You asked how to do it via dial plan, and you've been given three apps to do it. Yet you still keep asking. |
21:23.34 | ariel_ | is what is there now, just don't need to reload everything cutting it down to just the dial plan right n ow |
21:23.36 | p3nguin | System(asterisk -rx "dialplan reload") |
21:24.07 | ariel_ | p3nguin: yes, thank you. |
21:29.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
21:31.08 | *** join/#asterisk singler (~singler@84.15.129.49) |
21:32.24 | r0m|u | p3nguin: you in? |
21:32.33 | p3nguin | sure |
21:33.14 | r0m|u | I have the script. I am going to pb. |
21:33.51 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-ljtcqggohjnuepvx) |
21:34.07 | p3nguin | is leery of scripture. |
21:41.21 | r0m|u | p3nguin: http://pastebin.com/kjyutmWB |
21:41.59 | r0m|u | that will emil. But you can take out the email portion and have it echo under on your bash... |
21:42.16 | r0m|u | I have it set on my bash to just display fs space. and the rest to be emailed |
21:42.54 | r0m|u | the email portion is set on a cron |
21:43.01 | Maliuta | I fail to see how that script is asterisk related |
21:43.28 | Qwell | throws a paper airplane at cbwest |
21:43.54 | r0m|u | Maliuta: It was not directed to you. It was to p3nguin. We had a conversation about this script :) |
21:45.30 | *** join/#asterisk turtlefence (~tsmart@110.76.135.10) |
21:46.46 | p3nguin | r0m|u: Oh, that script. I couldn't figure out what script you were talking about. |
21:47.20 | leifmadsen | eyes the ban button on cbwest |
21:47.23 | r0m|u | p3nguin: :) Yes that script :P |
21:47.34 | Qwell | leifmadsen: according to infobot, he's talked. once. |
21:47.40 | leifmadsen | how many years ago? |
21:47.47 | Qwell | 3 weeks ago |
21:47.53 | Qwell | wants to use Asterisk 10 packages O.o |
21:48.05 | leifmadsen | Qwell: I asked for it in years |
21:48.19 | r0m|u | Is a disguised! |
21:48.19 | leifmadsen | (0.057496 years is the answer) |
21:48.33 | Qwell | ...308 |
21:48.35 | *** join/#asterisk thebitguru (~Adium@50.93.209.154) |
21:48.46 | Qwell | wait, where'd your 4 come from? |
21:48.46 | p3nguin | 0.057692308 years |
21:48.54 | Qwell | silly Canadian years. |
21:48.54 | leifmadsen | The Google is fun |
21:49.02 | leifmadsen | I typed in: 3 week / 1 year |
21:49.10 | leifmadsen | (3 weeks) / (1 year) = 0.0574960946 |
21:49.14 | Qwell | 3 weeks in years |
21:49.15 | Qwell | boom |
21:49.18 | Qwell | l2google, sir :p |
21:49.26 | Qwell | 3 weeks = 0.0574960946 years |
21:49.40 | p3nguin | I used 3weeks/52weeks |
21:49.44 | leifmadsen | p3nguin: ah |
21:49.49 | Qwell | p3nguin: same, at first |
21:49.52 | leifmadsen | I wonder if it takes in account for a leap year |
21:49.58 | Qwell | it would |
21:50.01 | leifmadsen | Qwell: lies |
21:50.06 | Qwell | 1 year = 52.177457 weeks |
21:50.19 | Qwell | 1 year = 365.242199 days |
21:50.26 | thebitguru | Hi, I have recently started having a problem with my PIAF install where I can't hear the ring tone when calling out, and no audio with outbound calls, but inbound calls seem to work OK. I am thinking that this is probably related to firewall, but I am not sure what it might be. Any ideas? I have ports 5060 TCP/UDP and 10k-20k UDP open |
21:50.26 | Qwell | it knows leap second even |
21:52.42 | leifmadsen | huh, asterisk does not like loading modules with: <description></description> or <description />, but using <description><para /></description> is fine |
21:52.54 | leifmadsen | dtd must not be quite right |
21:53.00 | Qwell | leifmadsen: it dislikes the empty description |
21:53.16 | Qwell | I remember hitting that a while back |
21:53.20 | leifmadsen | Qwell: yes, I understand that :) |
21:53.37 | Qwell | the <para/> makes it not empty. |
21:53.39 | leifmadsen | I'm trying to think where the best spot is to "fix" that |
21:53.46 | leifmadsen | Qwell: I also understand that, which is why I put in <para /> |
21:53.51 | Qwell | lamesauce |
21:53.53 | leifmadsen | but I don't think that is the right fix |
21:54.03 | Qwell | Putting a real description would be a good start. ;p |
21:54.10 | leifmadsen | I don't have time for that right now :) |
21:54.23 | leifmadsen | oh well, it's just a warning really |
21:54.28 | leifmadsen | well, not even a logged notice |
21:54.39 | leifmadsen | just console noise on start up |
21:56.55 | Qwell | main/pbx.c, probably remove this: |
21:56.59 | Qwell | <PROTECTED> |
21:57.09 | Qwell | I totally lied. |
21:57.45 | Qwell | moves on |
21:58.53 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
21:59.00 | *** part/#asterisk turtlefence (~tsmart@110.76.135.10) |
21:59.18 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
21:59.28 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
21:59.42 | dijib | anybody have a working wakeup call script running? |
21:59.58 | p3nguin | fprior: I don't know if anyone answered you or not, but when the call ends, MixMonitor() also ends. You only need to stop MixMonitor() manually if you need to stop recording while dialplan progresses. Some people might use a dial plan that starts MixMonitor when a call comes in and record the entire call as it goes to a queue and whatnot, but if the caller enters a personal extension the record would stop. That's just an ... |
22:00.04 | p3nguin | ... example, but there are other cases where someone would need to stop the recording when doing different things. |
22:04.16 | r0m|u | dijib: make one up. |
22:04.29 | r0m|u | shouldnt be that hard :) |
22:05.24 | r0m|u | dijib: the link to leifs book that p3nguin gave you has examples of wake up calls |
22:05.47 | r0m|u | time to go, |
22:05.55 | r0m|u | cya in a bit! |
22:06.22 | p3nguin | gives an odd look at "inbound call: +905548142836 <905548142836>" |
22:07.13 | p3nguin | +9 ? That's a new one. |
22:07.50 | _Corey_ | I think it's turkey |
22:08.00 | p3nguin | +90 Turkey |
22:08.06 | p3nguin | I think you might be right. |
22:10.49 | honree | is it possible to make a console call to a sip extension, and hang up the call as soon as the called party picks up? |
22:11.15 | honree | i spose i could do it by playing a short announcement... |
22:11.26 | p3nguin | You can make the call to a SIP *phone* and then make it hang up. |
22:11.47 | honree | yea, whatd i say? ;) |
22:11.57 | p3nguin | You said SIP extension. |
22:12.10 | p3nguin | channel originate SIP/whoever application Hangup |
22:12.12 | p3nguin | Try that. |
22:12.36 | honree | oo |
22:12.50 | honree | hang on then... |
22:13.33 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
22:15.08 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
22:15.36 | honree | mmmm that doesnt quite do what i want.... ive got an entry in extensions.conf that is like a dummy number that allows me to chnage the clid |
22:15.55 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
22:15.58 | p3nguin | How do you want to incorporate that into this call? |
22:16.04 | honree | the purpose of all this is to use a sip fone as a programmable alarm |
22:16.21 | p3nguin | Does that extension end up Dial()ing the phone? |
22:16.31 | honree | yus |
22:16.38 | p3nguin | Okay, that's easy, then... |
22:16.51 | p3nguin | channel originate Local/123@context application Hangup |
22:17.00 | p3nguin | where 123 is the extension in context |
22:17.09 | honree | ok just a mo |
22:17.38 | p3nguin | As soon as there is an answer, Hangup will run. |
22:17.48 | honree | awesome |
22:17.52 | honree | works a treat :D |
22:18.02 | honree | thanks :) |
22:18.30 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-157-152.chyn.qwest.net) |
22:18.36 | p3nguin | You could also make it play back a sound file. |
22:18.54 | p3nguin | channel originate Local/123@context application Playback your-announcement |
22:19.10 | p3nguin | channel originate Local/123@context application Playback silence/1&your-announcement |
22:19.21 | honree | cool |
22:20.02 | p3nguin | Once the Playback() ends, the call ends. |
22:20.25 | honree | right |
22:20.37 | p3nguin | If you need more elaborate things to happen, you can create another extension to do things... |
22:20.45 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
22:20.51 | p3nguin | channel originate Local/123@context extension 5555@alarms |
22:21.22 | p3nguin | When the phone picks up, extension 5555 runs. |
22:21.23 | honree | so once the called party picks up they get connected to 5555... |
22:21.27 | honree | right |
22:21.31 | honree | very cool |
22:21.46 | p3nguin | 5555 could do Playback(), or anything, really. Even Dial() another phone if you wanted. |
22:22.18 | honree | i have quite an old version of asterisk i probably should update... |
22:22.30 | honree | 1.4.4 |
22:22.34 | leifmadsen | woh |
22:22.37 | honree | heh |
22:22.46 | p3nguin | It could even be used for allowing the called party to press a key to talk to a person, or press another key to do something else. |
22:23.18 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fpkvlndilsxkledc) |
22:25.11 | pabelanger | leifmadsen: I did not see that coming |
22:25.23 | leifmadsen | pabelanger: I almost typed that too :) |
22:27.12 | *** join/#asterisk darkskiez_ (~dz@cpc4-broo7-2-0-cust167.14-2.cable.virginmedia.com) |
22:32.48 | Netgeeks | pfaw, 1.4.4 is relatively new. I still have a 0.95 asterisk running |
22:33.11 | Qwell | pfft, versioned releases? |
22:33.24 | Netgeeks | ^^ |
22:34.20 | honree | o |
22:34.58 | Netgeeks | i should have added a ;) at the end of that just to be sure no one took me serious |
22:35.24 | honree | ive had it installed for ages but only used it with a pretty basic config - couple of sip fones and a sip pstn gateway |
22:36.32 | Netgeeks | at least you didn't say 'I've got a fairly old trixbox install'..... |
22:36.53 | honree | heh |
22:37.25 | Netgeeks | the only response I have to that line is, 'Where do I send the flowers'? |
22:37.32 | Qwell | Netgeeks: The heads on stakes outside deter those people now |
22:37.42 | Netgeeks | lol |
22:38.15 | Netgeeks | i got a call recently from a guy who wanted me to fix his asterisk install, he was trying to do some fancy stuff like chanspy and such and it wasn't working right |
22:38.21 | honree | pushes his trixbox under the desk with his toe and whistles... |
22:39.07 | Netgeeks | I get in there to find out he had installed trixbox to get asterisk, then deleted all the trixbox config garbage and wrote his own dialplan.... he said that was easier than just installing a clean unix repo and then asterisk... I cried |
22:43.22 | fprior | hi all: howto manage asterisk failover where telefony is core business, like call center, but when finance don't permit a second Asterisk server for HA nor precious motherboard for using with VMware Esxi (for example) ? |
22:44.07 | Qwell | You can't failover without a box to failover...to. That doesn't even make sense. |
22:44.48 | Qwell | If your box dies, it's dead. That's it. |
22:45.05 | [TK]D-Fender | fprior: NOW is the point where telephony becomes "faith based", because you'd better PRAY it doesn't go down, but prayer is really all you've got left :P |
22:45.40 | [TK]D-Fender | s/but/because/ |
22:50.33 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
22:50.35 | fprior | yeah, I understand .....are you faithful or do you have implemented any solution of failover/HA ? Which is the scenario most used in * environment: HW virtualization or cluster ? |
22:52.38 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
22:54.54 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
22:55.31 | Netgeeks | if you don't have any telephony hardware involved, i.e. pri cards, etc., your options are many |
22:56.41 | Netgeeks | srv records and secondary registration/proxie servers in SIP UA configs will get you quite far, you can get even farther with a little more effort.... |
22:56.52 | *** join/#asterisk master_of_master (~master_of@p57B540F4.dip.t-dialin.net) |
22:58.43 | Netgeeks | I've not played with VMWare's cluster function in a while, and back when I was playing with it, asterisk didn't get along vmware in general too well. I've heard that has changed, and I've seen some asterisk systems on vmware platforms running with no issue. |
23:02.20 | fprior | Netgeeks: thanks. what about emergency: manuals don't explain how to manage this. do you give root password to IT Admin of your customer or they must waiting for you ? |
23:04.23 | Netgeeks | my experience with giving root password access to customers is not a pleasant one. Most treat it with the respect that they should, but some..... |
23:05.32 | Netgeeks | If it were me, I would try and design it so that a single failure doesn't create an outage, and that gives you leeway to fix the failure and bring the until back up as a backup |
23:09.42 | *** join/#asterisk WebSprocket (~user@dsl82-163-49-147.as15444.net) |
23:10.38 | *** join/#asterisk jpsharp (~jsharp@74-95-145-82-Naples.hfc.comcastbusiness.net) |
23:10.38 | WebSprocket | Hey guys, just needing a little advise, i have a sangoma card attached to a POTS line how in asterisk do i make our voice louder for the person we are calling, |
23:10.50 | WebSprocket | We can hear them fine, but they cannot hear us. |
23:11.11 | fprior | Netgeeks: yes, but depends on which is failure. Isn't the same an HW failure without remote access than a problem with Dialplan. |
23:13.37 | jpsharp | WebSprocket: Increase TX gain in your config file? |
23:13.50 | Netgeeks | fprior: well, a problem with the dialplan shouldn't be an emergency, that kind of problem should be debugged before you put the system into production. The kind of issues you need to deal with in production will be failures that cause the underlying hardware/os environment to 'go away' and network failures. |
23:14.46 | WebSprocket | jpsharp Thanks was confirm it was TX not RX, what is an advised value to start off with to see if it helps. |
23:16.00 | jpsharp | 6 is a good start. |
23:16.17 | jpsharp | I believe that will apply 6db of gain. |
23:17.09 | fprior | Netgeeks: I'm so sorry, I need to go. Tomorrow we can continue the conversation, if you agree |
23:17.56 | Netgeeks | fprior: unfortunately I don't often monitor this channel fprior, I just happened to have it open today while working on something, I'll try and have it open tomorrow |
23:18.08 | Netgeeks | You might do better by asking this question on the mail list |
23:19.20 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-xjaapnzolluuodaz) |
23:19.51 | fprior | Netgeeks: I will do this. thanks |
23:30.00 | *** join/#asterisk grantm (~grant@68.142.138.4) |
23:30.10 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
23:50.43 | *** part/#asterisk pietro (~pietro@88-149-227-4.dynamic.ngi.it) |
23:58.56 | *** join/#asterisk mducharme-work1 (~nothing@206.188.121.4) |
23:59.29 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-157-152.chyn.qwest.net) |