| 00:03.09 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.21 | *** join/#asterisk jakent (~john@c-98-233-13-157.hsd1.va.comcast.net) |
| 00:04.12 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.35.193) |
| 00:15.50 | dlynes_laptop | the nickname going around the industry for the new ipad, is the new 'Maxipad' |
| 00:16.08 | *** part/#asterisk DOP251 (~dmitrypan@host86-144-192-234.range86-144.btcentralplus.com) |
| 00:18.43 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 00:19.02 | kfife | Anybody here using SFA? |
| 00:19.28 | coppice | I think there are more people here doing SFA |
| 00:19.55 | coppice | (SFA was a strange choice of name) |
| 00:20.08 | kfife | Trying to use an existing account as the main account but it appears not to be allowed |
| 00:20.39 | kfife | In other words, skype user CompanyINC can't use SFA |
| 00:21.02 | kfife | Instead Skype USERID CompanyINC can create sub-accounts |
| 00:21.10 | kfife | and THOSE accounts can use SFA> |
| 00:21.37 | kfife | But HELLO! what if somebody calls Skype user CompanyINC? |
| 00:21.41 | kfife | I |
| 00:22.05 | kfife | 've gotta have skype running? I've gotta use forarding? Isn't that the whole point of SFA? |
| 00:22.56 | kfife | There MUST be a workaround somehow -- forwarding etc |
| 00:22.59 | kfife | Ideas? |
| 00:23.57 | dlynes_laptop | kfife, there's sfa users on here...but not many of them |
| 00:24.06 | dlynes_laptop | kfife, that being said, sfa is one of their most popular products |
| 00:24.22 | kfife | Thanks for your help yesterday: |
| 00:24.24 | kfife | the answers: |
| 00:24.33 | dlynes_laptop | kfife, have you tried asking on asterisk-users regarding sfa? |
| 00:24.35 | kfife | in 1.6.2 it's channel request hangup |
| 00:24.45 | kfife | and it's now core show help |
| 00:24.50 | kfife | not 'help' |
| 00:24.55 | dlynes_laptop | kfife, ah...good to know about the 'channel request hangup' |
| 00:25.06 | dlynes_laptop | kfife, i know...i mentioned core show help to you last night |
| 00:25.35 | kfife | My question is how the fü¢k is anybody supposed to know to look for help when buried if HELP is buried. |
| 00:25.53 | kfife | right. It's goofy though. you type 'help' |
| 00:26.01 | dlynes_laptop | kfife, no idea....but maybe russellb might know |
| 00:26.01 | kfife | and it says tye "core show help help" |
| 00:26.19 | dlynes_laptop | laughs. |
| 00:27.00 | dlynes_laptop | or actually leifmadsen might be a better person...I think he's the main documenter for asterisk |
| 00:27.03 | kfife | if you DO type core show help help as instructed it should laugh at you |
| 00:27.32 | kfife | They should actually code a special routine that outputs HaHaHa |
| 00:27.35 | JayTee52 | he's the documenteur extrordinaire |
| 00:27.39 | leifmadsen | :D |
| 00:27.51 | kfife | I TOTALLY understand moving commands around to be conceptually clean but HELP should be on the first level even if by aliasing |
| 00:27.57 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:28.58 | random_mike | turn debug up? |
| 00:29.06 | random_mike | erm |
| 00:30.01 | dlynes_laptop | random_mike, core debug help set on? |
| 00:30.15 | random_mike | that wasnt a quetsion |
| 00:30.25 | random_mike | i just accidently hit enter and there it was D: |
| 00:30.28 | dlynes_laptop | ah |
| 00:30.43 | dlynes_laptop | i guess it was one of those random thoughts :) |
| 00:30.50 | random_mike | lol yea |
| 00:30.56 | random_mike | actually u may be able to assist |
| 00:31.10 | random_mike | im looking at initiating calls from a website |
| 00:31.21 | random_mike | ie have asterisk call a phone number, via a website |
| 00:33.18 | random_mike | AJAM? |
| 00:34.14 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 00:35.25 | leifmadsen | kfife: the default cli_aliases.conf file should alias 'help' (and a couple others) for you automatically |
| 00:36.39 | leifmadsen | kfife: i.e. the [friendly] template is loaded automatically by default |
| 00:37.07 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 00:39.19 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 00:41.27 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 01:18.55 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:18.55 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 01:21.34 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 01:23.05 | *** join/#asterisk pawz (~pawz@ppp118-208-187-211.lns20.bne4.internode.on.net) |
| 01:27.02 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:27.02 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 01:33.07 | *** join/#asterisk DarkRift (~dark@bas1-sthubert21-1096612042.dsl.bell.ca) |
| 01:34.01 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 01:43.31 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:43.31 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 01:43.39 | carrar | anchovy muffins? |
| 01:44.36 | af_ | uhm, almost all |
| 01:44.44 | af_ | anchovy? |
| 01:44.47 | af_ | let me see |
| 01:46.39 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:46.39 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 01:47.03 | *** join/#asterisk pawz (~pawz@210.56.95.203) |
| 01:48.04 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 01:51.59 | *** join/#asterisk fuxu2 (iconicflux@www.kevinlynn.com) |
| 01:55.44 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:55.44 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 02:09.35 | *** join/#asterisk infobot (ibot@rikers.org) |
| 02:09.35 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 02:12.00 | *** join/#asterisk infobot (ibot@rikers.org) |
| 02:12.00 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 02:17.14 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 02:17.26 | doolittlework | help iax is killing me |
| 02:17.41 | voipmonk | love the name |
| 02:17.48 | voipmonk | whats happening doolittlework ? |
| 02:17.49 | Pan3D | death by protocol |
| 02:18.05 | doolittlework | : No authority found |
| 02:18.05 | doolittlework | but username an passwor matches |
| 02:19.08 | voipmonk | excellent |
| 02:19.12 | voipmonk | mind if I take a look? |
| 02:19.32 | doolittlework | no static ip's |
| 02:19.38 | doolittlework | on lan |
| 02:24.11 | doolittlework | http://pastebin.com/m6d862cb8 |
| 02:25.29 | fuxu2 | sup all |
| 02:26.14 | doolittlework | nothing much but iax pain in * |
| 02:29.33 | fuxu2 | which paper would you be more interested in: "Asterisk Security: Multiple Levels of Fail" that goes into a look at why some of asterisk's security issues exist, how prevalent they are, and what to do about them; or, "A Walk on the Dark Side" -- A look at why so many attacks on asterisk happen, what methods are used to break in, breakdowns of successful attacks, and how some vulnerabilities exist that the attackers haven't even seen yet. |
| 02:37.29 | BlueDay | The second |
| 02:38.00 | BlueDay | Or you could make it one paper. |
| 02:47.24 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 02:50.22 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 02:51.06 | *** part/#asterisk lanning (~lanning@208.87.235.224) |
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| 03:04.26 | *** join/#asterisk trnzmeta (~bleh@secure27.lnk.telstra.net) |
| 03:10.54 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 03:35.14 | florz | fuxu2: the former would provide more insight than "the code is affected by pretty much any programming and design stupidity every invented"? |
| 03:35.21 | florz | -y |
| 03:36.59 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:37.29 | fuxu2 | florz: yeah it wouldn't be totally negative |
| 03:42.39 | florz | fuxu2: But, like, would you expect it to provide any practical benefits to current or potential users of asterisk? What I mean is: I would think that it's pretty easy to find out that asterisk's code is crap, but fixing it seems almost impossible to me, in particular as the community doesn't seem to care particularly ... |
| 03:48.41 | *** join/#asterisk girlny (~girlny@CPE00195b4be142-CM001a668ec076.cpe.net.cable.rogers.com) |
| 03:49.30 | voipmonk | nope |
| 03:49.34 | voipmonk | so fix it |
| 03:50.44 | jaytee | asterisk's code is crap? wow! that's news to me. I never get the memos. |
| 03:52.06 | florz | Memos? You should read the code ;-) |
| 03:52.29 | jaytee | hmmmm, lets see, my max uptime before rebooting server to do an image backup was 143 days, total dropped calls during that period: 4 due to a momentary outage caused by my idiot coworker assigning the same IP address to a network printer as our asterisk server. |
| 03:53.31 | jaytee | but then I don't use the internet for calls, I use PRI |
| 03:53.34 | girlny | hello i need to know how hevy is the trasnscodinding betewn G.711 A-Law and G.711 µ-Law will i consume alot of cpu and will i losse alot of quality |
| 03:53.36 | girlny | ? |
| 03:55.07 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:57.18 | *** part/#asterisk dseeb_ (~Adium@CPE-124-177-30-127.lns2.dav.bigpond.net.au) |
| 03:57.45 | p3nguin | jaytee: Is there a counter for dropped calls, or were you keeping a running total? |
| 03:58.11 | jaytee | p3nguin, I was keeping a running total |
| 03:58.37 | florz | jaytee: Yeah, apparently there are enough people using asterisk that most of the bugs that lead to immediate crashes have been fixed or worked around. But that doesn't change much about the overall quality of the code - and 143 days of uptime really isn't all that much anyhow that you could consider software "stable" because of. A linux kernel with some 3 years or more of uptime maybe is ... |
| 03:58.42 | p3nguin | I don't know that I ever dropped any calls, so that's too bad there isn't a built-in counter. |
| 04:00.14 | florz | plus, obviously, it wouldn't make much sense to use a buggy piece of software's counter of its own failures to measure its reliability |
| 04:00.22 | jaytee | all the dropped calls had more to do with the telco than my asterisk server except for the IP conflict that dropped a 1 in progress and denied any other calls. the others I suspect were random cell phone calls losing signal. |
| 04:01.22 | jaytee | florz, if you don't like asterisk this much then might I ask, "What in bloody blue blazes are you doing here?" |
| 04:02.08 | florz | jaytee: I do chat here, in case you didn't notice. Plus, I didn't say I "don't like asterisk". I just said that its code sucks. |
| 04:02.12 | jaytee | pimping for FreeSWITCH? |
| 04:03.00 | florz | jaytee: but that's essentially what I meant by "in particular as the community doesn't seem to care particularly" |
| 04:03.05 | jaytee | florz, have you contributed any patches or code to asterisk? |
| 04:04.07 | p3nguin | oohh oh |
| 04:04.09 | p3nguin | me me me |
| 04:04.11 | p3nguin | I know this one |
| 04:04.21 | random_mike | is anyone able to advise how to change the From: header when asterisk makes an outbound call? At the moment I see: From: "$CallerID" <sip:asterisk@xxx.xxx.xxx.xxx> |
| 04:04.24 | random_mike | $callerID I can alter |
| 04:04.35 | random_mike | but the sip:asterisk@ip I'd like to chang etoo |
| 04:04.57 | random_mike | im using AMI to make outbound calls |
| 04:05.01 | jaytee | random_mike, I think fromuser= in sip.conf |
| 04:05.19 | random_mike | thanks |
| 04:05.25 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 04:05.29 | florz | jaytee: No. What does that change about my argument? |
| 04:05.48 | voipmonk | username in the device or the fromuser in sip.conf |
| 04:06.08 | jaytee | florz, movie critics criticize movies but they never make movies of their own. I don't particularly care much for their opinions either |
| 04:08.31 | florz | jaytee: Given that you don't care you are writing a lot. Plus, not caring doesn't change reality much with stuff that's not about taste. |
| 04:09.10 | jaytee | florz, then go write a voip system of your own from scratch. |
| 04:09.17 | florz | jaytee: why? |
| 04:09.46 | jaytee | so that you might have a better appreciation of what other people have accomplished and contributed to |
| 04:10.12 | florz | jaytee: better appreciation would make all the bugs go away? |
| 04:10.32 | *** join/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 04:10.39 | jaytee | florz, no but all software has bugs. name one program on linux, windows, OSX or Unix that doesn't |
| 04:10.55 | LemensTS | hey im trying to add *67 feature. _X., Does not catch *, how do i catch it? |
| 04:10.55 | rossand | Got my snom370 working. Under the SIP tab for a given identity, there's a Server Type Support drop down. It *must* be Asterisk. Otherwise you get a Received response: "Forbidden" error. |
| 04:11.20 | rossand | dialing out was working. This was just for dialing in. |
| 04:11.42 | jaytee | shit, Windows XP is the Alsace Lorraine Swiss cheese of operating systems when it comes to bugs and security holes and it's pretty much ubiquitous and made Microsoft a shitton of money. |
| 04:12.59 | florz | jaytee: So because everyone has bugs, many bugs are good? I am of a slightly different opinion on that matter, I guess. |
| 04:13.43 | jaytee | florz, where in any of my statements did I say bugs were good? |
| 04:13.49 | jaytee | where? |
| 04:14.23 | voipmonk | LemensTS: exten => *67,1,blah |
| 04:14.52 | florz | jaytee: Well, if you didn't it was a non sequitur instead: 05:10 < jaytee> florz, no but all software has bugs. name one program on linux, windows, OSX or Unix that doesn't |
| 04:15.28 | LemensTS | voipmonk: so only way is *67X.,1,blah ? |
| 04:15.34 | jaytee | florz, stating a fact is one thing but you're making an implication that has no validity based on what I said |
| 04:16.10 | florz | jaytee: for, the fact that most software does have bugs isn't much of an argument against stating the fact that those bugs are there and are bad, unless you consider bugs a good thing |
| 04:16.58 | jaytee | you're reasoning is flawed |
| 04:17.10 | florz | No, I am reasoning is not flawed. |
| 04:18.21 | jaytee | yep, I made a grammatical error. should have been "your reasoning is flawed" |
| 04:18.58 | florz | At least, you still didn't point out either why my statement that asterisk is full of bugs is false or why I should not mention it. |
| 04:19.20 | jaytee | and for the record, I don't consider bugs a good thing. but they are inevitable in all large software projects. |
| 04:19.56 | florz | Yeah, but the levels in asterisk ought to be avoidable, I think. |
| 04:20.23 | jaytee | and yes, asterisk has bugs. i suspect that 10 years from now if asterisk is still around it will still have bugs, albiet different bugs than the ones that exist today. |
| 04:21.05 | p3nguin | lemensts: Hard-code it. |
| 04:21.16 | p3nguin | Oh, voipmonk already said that. |
| 04:21.21 | jaytee | florz, well if you feel so strongly about the number of bugs in asterisk then pick one and figure out a fix to contribute. it's a much more productive use of time than bitching about it in a support chat channel. |
| 04:23.06 | p3nguin | lemensts: Also note that what voipmonk told you was not what you subsequently questioned. exten => *67,1,blah is not the same as *67X.,1,blah |
| 04:25.14 | florz | jaytee: well, (a) fuxu2 was talking about writing a paper on the topic, and I thought that maybe writing the right paper could be a productive thing, so possibly worth some discussion and (b) the major problems are the architectural/structural deficiencies that are close to unfixable without some major effort ... |
| 04:26.25 | florz | jaytee: Take this simple example: How would you limit a SIP phone to calling a specific variable-length range with a specific prefix on the PSTN? |
| 04:28.26 | jaytee | not sure I understand your question, can you give me a specific example? |
| 04:30.03 | voipmonk | with pattern matching |
| 04:30.34 | voipmonk | and call limits either in sip.conf or in the dialplan using a gotoif and astdb to count calls |
| 04:30.38 | jaytee | that's how I manage outbound calls to the PSTN |
| 04:30.55 | florz | jaytee: Well, you have a SIP phone and a PSTN connection of some kind connected to your asterisk. Now, you want the telephone to be able to call exactly those numbers starting with, say, 0800, on the PSTN. But the 0800 range on the PSTN is variable length. |
| 04:31.26 | ChannelZ | exten => _0800X. |
| 04:31.29 | jaytee | florz, what country do you live in? |
| 04:32.16 | florz | jaytee: .de - I am not exactly sure whether +49-800 is variable length, but most of the numbering plan is. |
| 04:32.18 | jaytee | ChannelZ just answered your question I think if I understood your question correctly |
| 04:33.17 | florz | jaytee/ChannelZ: Well, that's not exactly a complete answer, but I guess a ,1,Dial(Foo/${EXTEN}) was implied? |
| 04:33.21 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 04:33.52 | jaytee | florz, well here in the US we have 7 digit local numbers and 10 digit long distance number and we prefix all internation calls with 011 so it's pretty much a no brainer to setup pattern matches to handle any of those but with a specific prefix and a variable length number the example that ChannelZ gave would work. |
| 04:35.41 | florz | jaytee: Well, see, it would not. You can dial ampersands and slashes and whatnot on SIP phones ... this is commonly known as "in band signalling" or maybe "format string" or even a "code injection" vulnerability. And this kind of violation of abstraction boundaries you can find almost everywhere in asterisk. |
| 04:35.44 | jaytee | florz, then _0800X.,1,Dial(Foo/0800${EXTEN}) |
| 04:36.31 | *** join/#asterisk Buklov (~buklov@213.138.71.254) |
| 04:36.45 | jaytee | ooops, that should be _0800X.,1,Dial(Foo/0800${EXTEN:4}) |
| 04:36.46 | p3nguin | That's going to create 08000800... |
| 04:36.51 | p3nguin | yeah, better. |
| 04:37.15 | florz | yeah, either one doesn't enforce any limit, so doesn't really matter ;-) |
| 04:38.05 | jaytee | so when was the last time you dialed a cell phone with someone's number that had an ampersand in it? |
| 04:38.06 | *** join/#asterisk RobH_ (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 04:38.11 | p3nguin | lol |
| 04:38.35 | florz | jaytee: you seriously don't see the seucrity problems in what I described? |
| 04:38.38 | p3nguin | Three amphersands and a duck. |
| 04:39.33 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 04:41.23 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ttkhxxczmyimvocs) |
| 04:41.32 | jaytee | florz, so if I had an asterisk server on my local network and connected to the PSTN via T1 PRI and blocked all SIP traffic at my firewall then you'd still be able to hack me? |
| 04:42.20 | florz | jaytee: what's your point? |
| 04:42.33 | p3nguin | It's a security problem just having the thing powered on! |
| 04:43.21 | jaytee | runs off in a panic flailing his arms and screaming "we're doomed! we're doomed!" |
| 04:43.49 | voipmonk | the sky is falling, why on earth arent we all just hacked , right now? |
| 04:45.15 | p3nguin | And please call me. My number is in hieroglyphics, though: owl, arm, canoe, lion, eel, lion. |
| 04:45.21 | jaytee | LOL |
| 04:45.24 | p3nguin | I assume your SIP phone can dial that. |
| 04:45.52 | florz | I have the slight impression that you really don't understand ... |
| 04:46.31 | jaytee | they should do an Internet Hitler routine on YouTube with him raving about Asterisk security. The recent iPad one was hilarious |
| 04:46.54 | florz | You claimed that the dialplan snippet above would limit the SIP phone to calling 0800* |
| 04:47.28 | florz | But if you dial 0800&Foo/123456, you can essentially dial anywhere you want. |
| 04:47.40 | jaytee | it would dial any number with an 0800 prefix and any variable length number after that. |
| 04:48.07 | florz | (including other contexts that you normally don't have access to via a local channel and such) |
| 04:48.42 | jaytee | florz,so you're trying to limit the types of calls internal users can dial outbound on their sip phones? am I getting this correctly? |
| 04:49.22 | girlny | hello i need to know how hevy is the trasnscodinding betewn G.711 A-Law and G.711 µ-Law will i consume alot of cpu and will i losse alot of quality |
| 04:50.08 | jaytee | girlny, minimal load and you should have no noticeable loss of quality |
| 04:50.38 | p3nguin | If your dialplan matched _0800X. with a command of Dial(SIP/{$EXTEN}), and you decide to dial 0800&Foo/123456... the command would be Dial(SIP/0800&Foo/123456)... but if Foo/123456 is not a valid tech/resource in that context, what is going to happen? |
| 04:51.10 | florz | jaytee: No, I didn't say anything about "internal". And I am also not interested in fighting about whether this specific setup is pointless from your point of view or whether Foo channels don't exist in your setup, as that doesn't have any bearing on the fundamental problem. |
| 04:51.23 | florz | p3nguin: see above |
| 04:51.29 | girlny | ok thanks |
| 04:51.46 | p3nguin | You'll have to direct me a little better than "above" |
| 04:52.00 | florz | p3nguin: And I am also not interested in fighting about whether this specific setup is pointless from your point of view or whether Foo channels don't exist in your setup, as that doesn't have any bearing on the fundamental problem. |
| 04:52.15 | jaytee | florz, perhaps you could explain in simple terms that someone with my obviously limited intelligence can understand just WTF the fundamental problem is? |
| 04:52.22 | p3nguin | I wasn't arguing it. |
| 04:52.43 | p3nguin | I'm interested in the possible outcomes. |
| 04:52.44 | voipmonk | lol |
| 04:53.36 | florz | p3nguin: ok, let me point it out explicitly: "Foo" in this context was, as usual, a meta-syntactic variable. |
| 04:54.23 | jaytee | ooooh, he's using big words now! |
| 04:54.25 | p3nguin | Sure, we could say it was Local instead. |
| 04:54.56 | jaytee | meta-syntactic, that sounds cool. I'm gonna go google that! |
| 04:56.28 | p3nguin | So can you dial an @ in your phone? |
| 04:56.38 | p3nguin | 'Cause that's going to be needed for a local channel. |
| 04:56.54 | p3nguin | to get out of the context, anyway. |
| 04:57.23 | jaytee | what's the DTMF frequencies for an @? |
| 04:58.23 | florz | jaytee: The fundamental problem is the violation of abstraction boundaries. Or, more specifically in this case, the violation of the boundary between what you could call "code" and "data". What you expect to be a phone number probably should not be interpreted by an interpreter that allows more operations than addressing within a numbering plan. |
| 04:59.28 | florz | p3nguin: Well, I certainly can dial an @, yeah. But indeed I haven't tried the Local channel variant of the attack, so no clue whether asterisk does decode it "correctly" |
| 05:00.21 | jaytee | they must have some weird phones in Germany. There's no @ on my Polycom 330. Maybe the 331 comes with an @ key |
| 05:00.22 | p3nguin | Are you merely placing these ASCII characters into the dial line of a softphone and pressing Dial? |
| 05:01.40 | p3nguin | If the PBX will not interpret the characters and an associated DTMF, is this really even an issue at all? |
| 05:02.14 | jaytee | p3nguin, that's what I've been trying to figure out for about a half hour now :-) |
| 05:02.26 | florz | p3nguin: nah, specifically I am using a snom370 - and IIRC it should be possible to enter it directly in URI encoded form |
| 05:02.29 | p3nguin | Sorry, I came late to the game. |
| 05:03.20 | *** part/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 05:04.03 | florz | plus, after all, there are many scenarios where you don't even have any control over the "phone" being used, so the problem would even be a problem if you couldn't dial such stuff with any existing telephones |
| 05:04.15 | *** join/#asterisk aeperezt (~aperez@cpe-00064f60e848.cpe.cableonda.net) |
| 05:04.57 | jaytee | yep, face it, we're doomed! |
| 05:04.57 | p3nguin | Sure this sounds like a theoretical problem, but if it isn't literally something that can be achieved, it can hardly be classified as a design flaw. |
| 05:05.04 | florz | Anyhow, the part about dialing into a different number range on a Zap channel (where channel names are particularly easy to guess) does work with the snom370 |
| 05:07.03 | *** join/#asterisk joobie (~joobz@mx01.anric.com.au) |
| 05:07.27 | joobie | guys how can i increase the volume level of a sip user? when they talk it's really low.. when they listen to others it's loud.. so trying to just get the speaking volume to increase |
| 05:07.54 | florz | and, in a way, this "well, it's just a theoretical problem" kind of argument is also one of the reasons why I think that there isn't much of a chance of fixing the problems |
| 05:07.54 | jaytee | boost the mic volume |
| 05:08.41 | joobie | how jaytee ? got a polycom 320 |
| 05:09.22 | p3nguin | Maybe the other end needs to turn up his volume. |
| 05:10.40 | joobie | ahh found the config optiuon for the 320 |
| 05:10.43 | joobie | but can this be done in sip.conmf |
| 05:10.52 | p3nguin | no |
| 05:11.04 | jaytee | no, not in 1.4 at least |
| 05:11.45 | jaytee | I think there might be a way to increase tx gain for SIP phones in one of the 1.6 branches but not certain |
| 05:12.42 | florz | But yeah, quite possible that the bugs in the SIP parser in this case prevent you from getting into different contexts via a local channel. But to be clear: As far as SIP is concerned, dialing an @ in the local part is possible, so the security in this case depends on the SIP parser being broken ... I am not really sure that that's the kind of stability I'd like to rely on. |
| 05:14.12 | jaytee | florz, then maybe Cisco Call Manager is right for you! |
| 05:14.27 | florz | jaytee: Why do you think that? |
| 05:15.03 | jaytee | because you just said you're not sure that that's the kind of stability you'd like to rely on |
| 05:15.31 | jaytee | you could always ask for your money back |
| 05:16.09 | florz | jaytee: And you think that that would be the kind of stability I'd like to rely on? |
| 05:16.29 | florz | jaytee: Do you like to rely on bugs for your security? |
| 05:16.30 | jaytee | Cisco? lots of people do |
| 05:16.44 | jaytee | I don't rely on bugs for security |
| 05:17.06 | florz | Yeah, and lots of people do use asterisk, so? |
| 05:17.07 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 05:17.17 | jaytee | I just try to make things as secure as possible within the limits of the software |
| 05:17.46 | joobie | ahh k |
| 05:17.50 | jaytee | so should everyone running Windows stop using it since it has so many potential exploits? |
| 05:17.50 | joobie | thanks jaytee |
| 05:18.46 | florz | jaytee: How do you know that you don't rely on bugs for security? |
| 05:19.23 | jaytee | florz, you're tilting at windmills and pissing into the wind. give it a rest |
| 05:20.03 | florz | jaytee: I don't know what "everyone running Windows" should do, that's not my expertise. |
| 05:21.56 | jaytee | florz, well perhaps you should write your own secure voip software that meets your expectations. you're from Germany, right? they have a reputation for cleverness. they managed to fuck up Europe and most of the rest of the world twice in one century. I'm sure you can come up with something better than Asterisk. |
| 05:23.01 | *** join/#asterisk kerx (~Peter@76-240-161-68.lightspeed.irvnca.sbcglobal.net) |
| 05:23.26 | florz | jaytee: Well, at least so far you have been doing well in proving my assumptions correct, so it's not completely pointless. Also, why should I give it a rest? |
| 05:23.56 | kerx | Hi all, I'm trying to setup a web based sip phone. Any suggestions on how I should tackle this? Has anyone tried tackling a problem like this and can give suggestions, I would really appreciate it. |
| 05:24.38 | jaytee | florz, don't give it a rest then.....just "Talk to the hand!" cuz I ain't listening any longer. |
| 05:24.40 | florz | jaytee: Why should I write my own secure VoIP software? Yes, I am. Dunno whether they have. *lol*. |
| 05:25.36 | p3nguin | <PROTECTED> |
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| 05:28.42 | jaytee | time to go read and get some sleep, peas out, mashed potatoes in |
| 05:29.00 | *** join/#asterisk nightrid3r (kvirc@41.214.133.18) |
| 05:29.15 | p3nguin | This again? |
| 05:30.45 | *** join/#asterisk nightrid3r (~kvirc@41.214.133.18) |
| 05:31.44 | kerx | Any one can provide suggestions for creating a SIP phone on Windows platform? Something that is integrated with a browser is what I'm aiming for. |
| 05:37.28 | fuxu2 | kerx: learn visual c++? |
| 05:38.01 | kerx | fuxu2, Yes, but what's going to do the audio processing? I'm hoping to see if anyone knows of a library maybe I can use (like an API or something) ? |
| 05:38.33 | fuxu2 | kerx: port the libs? |
| 05:38.40 | kerx | fuxu2, port the libs from where? |
| 05:39.08 | [TK]D-Fender | kerx: http://www.google.ca/#hl=en&source=hp&q=visual+C%2B%2B+SIP+library&btnG=Google+Search&meta=&aq=f&oq=visual+C%2B%2B+SIP+library&fp=7564381002806979 |
| 05:40.24 | kerx | Hey TK |
| 05:40.42 | kerx | I assume you've never used any library, otherwise you'd specifically suggest one. Is my assumption correct? |
| 05:41.10 | fuxu2 | fender: did I hear that you've gone to using FOP? |
| 05:41.17 | fuxu2 | wrong chan |
| 05:41.24 | kerx | what's FOP ? |
| 05:41.42 | kerx | eew, FOP... |
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| 07:01.17 | Madkiss | ahum. when updating from asterisk 1.4 to asterisk 1.6 ... |
| 07:01.30 | Madkiss | whatr has changed withj regards to the IAX2 channel? |
| 07:01.38 | Madkiss | [Feb 2 07:59:36] WARNING[1549]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 07:02.15 | ChannelZ | Whatever you're dialing doesn't exist |
| 07:02.38 | ChannelZ | Either not registered, or wrong IP, or... |
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| 07:03.23 | *** join/#asterisk Ahmadi (~4e26ab14@gateway/web/freenode/x-kcmzvybprfnjzofz) |
| 07:03.35 | Ahmadi | Hi |
| 07:03.39 | Ahmadi | I have following error: |
| 07:03.41 | Ahmadi | Call from '' to extension '202' rejected because extension not found. |
| 07:03.45 | Ahmadi | Can anyone help me? |
| 07:03.58 | jeff_phillips | Ahmadi: create an extension 202 |
| 07:04.11 | Ahmadi | i did it before |
| 07:04.17 | Ahmadi | but also same error |
| 07:05.21 | Ahmadi | when my extension want register itself, i have following error: |
| 07:05.24 | Ahmadi | Registration from 'Ahmadi <sip:202@78.38.X.x>' failed for '78.38.x.x2' - No matching peer found |
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| 07:07.24 | ChannelZ | you have a big mess |
| 07:08.02 | ChannelZ | start by pastebining your sip.conf |
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| 07:08.40 | Ahmadi | Where can i found "sip.conf" ? |
| 07:09.34 | ChannelZ | Let me guess, you're using freepbx |
| 07:10.26 | Ahmadi | yes |
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| 07:10.43 | Ahmadi | ;) |
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| 07:11.17 | Ahmadi | please help me more |
| 07:11.41 | ChannelZ | I can't, I dont' use freepbx and don't know it's craziness |
| 07:12.12 | ChannelZ | All I can offer is this: An extension IS NOT the same thing as a SIP device name. They _can_ be the same but aren't always. |
| 07:12.19 | jeff_phillips | no, it's elastix |
| 07:12.43 | BeeBuu | hi,all |
| 07:12.49 | ChannelZ | Your SIP device registers with * using a username and password (secret). That 'username' is usually the device name. Which again is not the same thing as the extension |
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| 07:13.25 | Madkiss | ChannelZ: hm. |
| 07:13.34 | BeeBuu | i wnat to run SS7, what packages need to install? |
| 07:13.44 | BeeBuu | libss7? |
| 07:13.46 | ChannelZ | So you configure a sip device, perhaps called 'MyPhone', with the password 'barf'. You configure the device to register with * as 'MyPhone' with the password 'Barf'. You configure an extension, let's pick 555, which dials the device... SIP/MyPhone |
| 07:14.21 | ChannelZ | How any of that relates to freepbx... *shrug* Go ask #freepbx |
| 07:14.35 | Ahmadi | ok, |
| 07:14.47 | Ahmadi | What is mean of "No matching peer found" |
| 07:14.58 | Ahmadi | Because my extensions can not register and throw "No matching peer found" error |
| 07:15.45 | ChannelZ | Your phone is logging in as a sip peer that doesn't exist. |
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| 07:33.07 | Madkiss | ChannelZ: you were indeed right, thank you :) |
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| 08:11.24 | jsohler | My asterisk PHP app was working, but they changed something on the server... and now when there are are a series of play() commands, or a series of play()s followed by a read()... The caller only hears the last one in the sequence. |
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| 08:16.46 | jsohler | Example... play("greeting"); play("announce"); $result = read($message_id, 5000, 1); .... the caller would only hear the message referenced by $message_id. |
| 08:19.59 | Diffen2 | morning. does anyone here knows if its possible to send queue calls to a cellphone? like if the extension are using followme to a cellphone or so. |
| 08:20.38 | jeff_phillips | yes |
| 08:21.35 | fiddur | just add e.g. SIP/mysipprovider/cellphonenumber as a queue member |
| 08:22.57 | Diffen2 | ahhh ok |
| 08:22.58 | Diffen2 | cool |
| 08:23.06 | Diffen2 | i will try it out. thank you! |
| 08:24.21 | jsohler | Any help is greatly appreciated. It happened before, and they were ableto "undo" whatever was done on the server... but this time they don't even know what happened, even though I can tell them a 10 minute window when the change happened. |
| 08:25.17 | fiddur | Diffen2: My experience with sending away calls from queues though, is that you usually need to change the timeouts, since there'll be a delay... |
| 08:26.17 | fiddur | jsohler: It sounds like the php-script isn't waiting for the application to finish... |
| 08:27.02 | fiddur | jsohler: or, that the play-function isn't executing anything |
| 08:27.17 | fiddur | jsohler: You should be able to see some output in asterisks logs, about what commands AGI is executing. |
| 08:27.51 | Diffen2 | fiddur you mean delay when the call is connected to the cellphone? |
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| 08:28.30 | jsohler | I will look at the trace again |
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| 08:29.52 | fiddur | Diffen2: Yes... depending on the provider off course, but it sould be slower than calling a local phone :) |
| 08:30.30 | Diffen2 | yes :) |
| 08:31.40 | fiddur | Diffen2: For me, there's ben over 15 seconds delay a short period... but we're changing sip-provider in a few days now :) |
| 08:33.49 | Diffen2 | ahh ok fiddur. my experience is like 4-6 seconds before the cellphone starts to ring. |
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| 08:37.15 | jsohler | Fiddur: I looked at the trace and it shows as playing all the files specified, but the caller only hears the last one. I even changed the code to play the same file twice in a row, but as mentioned only the last one is heard by the caller, so it only played once. |
| 08:37.21 | Diffen2 | fiddur hmm do i just add member => cellphonenumber in the queue conf? |
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| 08:41.58 | fiddur | jsohler: In asterisks log, is there the correct delay between the plays, or are they run directly after eachother, same second? |
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| 08:42.59 | fiddur | Diffen2: A queue member is an interface... so the full interface id would be SIP/provider/cellnr if you use SIP |
| 08:43.33 | jsohler | They play one after another, because one of them in the middle of the sequence has a variable length. This same code worked before the change on the server, and had worked for months. |
| 08:44.17 | jsohler | I'm looking for something that could have changed with the behavior of "root" that would impact the app... or a change in PHP or Asterisk settings that would impact the app. |
| 08:44.57 | jsohler | This happened about a year ago, and they were unable to "undo" what had been done... but they didn't document the solution. |
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| 08:48.11 | Diffen2 | fiddur thanks again |
| 08:49.05 | fiddur | np |
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| 09:09.28 | jsohler | Any ideas? |
| 09:10.12 | ChannelZ | What does the console say |
| 09:10.32 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 09:11.51 | jsohler | ChannelZ: no console... but a did an asterisk -r, and all the play commands look just fine... but the caller only hears the last one. |
| 09:12.23 | ChannelZ | Do the other ones whiz by, or does the caller hear silence for the others? |
| 09:16.26 | fish9370 | check microphone.. )) |
| 09:17.28 | jsohler | ChannelZ: They just whiz by as if the play never happened, but the trace shows them. If you were following this from the start, it is being caused by some change on the server... No code needed to be changed to fix it when it happened a year ago. |
| 09:18.49 | jsohler | ChannelZ: So its as if the subsequent plays are interupting the previous ones, but it happens so quick that the caller only hears the last one. |
| 09:19.17 | ChannelZ | well it sounds like the playback commands are asyncronous and stopping one another |
| 09:19.27 | ChannelZ | you're using 'exec playback' or something? |
| 09:20.07 | jsohler | Example... play("greeting"); play("announce"); $result = read($message_id, 5000, 1); .... the caller would only hear the message referenced by $message_id. |
| 09:20.29 | jsohler | This is a PHP app |
| 09:20.47 | kaldemar | jsohler: feel free to show a cli output of a call |
| 09:22.03 | jsohler | kaldemar: It will take a bit for me to get this, as I don't have putty on this computer. |
| 09:23.21 | jsohler | kaldemar: However, the cli looks normal as far as I can see. |
| 09:23.22 | ChannelZ | you must be using a set of AGI functions, what is play() doing? |
| 09:24.23 | jsohler | ChannelZ: play just does a playback of a gsm file to the caller. |
| 09:24.43 | ChannelZ | sighs |
| 09:25.11 | ChannelZ | yes but is it using 'exec playback', or 'control_stream_file', or 'stream_file'... |
| 09:25.12 | kaldemar | jsohler: you won't get help if you don't show anything concrete. |
| 09:25.25 | ChannelZ | What version of * is this? |
| 09:26.04 | jsohler | ChannelZ: 1.4.18.1 |
| 09:26.55 | jsohler | kaldemar: Yes, I will show the cli output... I just have to get it. |
| 09:28.27 | Diffen2 | fiddur sorry that i havent told you the result with the queue, but it works really fine :) thanks for your help :D |
| 09:28.31 | ChannelZ | well if I had to guess you're getting file not founds or something |
| 09:28.39 | jsohler | I was hoping this was going to be a common, and therefore simple problem where someone said... Oh, that's happening because of "this"... I'm a real newbie as far as Asterisk goes. |
| 09:29.13 | ChannelZ | or the permissions on the sound files changed such that * can't read them |
| 09:29.16 | jsohler | ChannelZ: I can get any of the files in the sequence to play, IF they are the last one in the sequence... So it's not a file not found. |
| 09:31.23 | *** join/#asterisk pietro1 (~pietro@88.149.164.32) |
| 09:31.33 | pietro1 | hi |
| 09:31.56 | pietro1 | where i can find zaphfc modules ? |
| 09:32.29 | pietro1 | i have a Cologne HFC-PCI pci card |
| 09:36.34 | kaldemar | pietro1: zaphfc comes with bristuff. |
| 09:41.14 | Madkiss | how do I see a functions help text? |
| 09:41.36 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 09:41.47 | ChannelZ | core show function WHATEVER |
| 09:42.11 | ChannelZ | they are case sensitive |
| 09:43.58 | pietro1 | kaldemar: ok |
| 09:48.22 | ChannelZ | bed time |
| 09:48.36 | jsohler | I have my trace... |
| 09:49.08 | ChannelZ | do pastebin hurry |
| 09:49.44 | jsohler | Connected to Asterisk 1.4.18.1 currently running on host (pid = 18989) |
| 09:49.44 | jsohler | Verbosity is at least 3 |
| 09:49.44 | jsohler | <PROTECTED> |
| 09:49.44 | jsohler | <PROTECTED> |
| 09:49.44 | jsohler | <PROTECTED> |
| 09:49.44 | jsohler | <PROTECTED> |
| 09:49.44 | jsohler | <PROTECTED> |
| 09:49.45 | jsohler | <PROTECTED> |
| 09:49.45 | jsohler | <PROTECTED> |
| 09:49.46 | jsohler | <PROTECTED> |
| 09:49.46 | jsohler | <PROTECTED> |
| 09:49.47 | jsohler | <PROTECTED> |
| 09:49.47 | jsohler | <PROTECTED> |
| 09:49.48 | jsohler | <PROTECTED> |
| 09:49.57 | ChannelZ | PASTEBIN damnit |
| 09:50.03 | ChannelZ | pastebin.ca |
| 09:50.50 | jsohler | sorry... I don't know what you meant by pastbin.ca |
| 09:51.04 | ChannelZ | http://pastebin.ca |
| 09:51.12 | ChannelZ | don't flood the channel with configs and crap |
| 09:51.29 | ChannelZ | suffice to say what you did flood with doesn't really show anything interesting |
| 09:51.32 | jsohler | so sorry about that... I am very new the IRC :( |
| 09:51.52 | jsohler | Right... It all looks fine |
| 09:52.13 | jsohler | but the only thing the caller hears is "donation-out3" |
| 09:52.54 | jsohler | I can switch the positions of "donation-out3" and "lotw612", and they will only hear "lotw612" |
| 09:53.36 | ChannelZ | don't you mean in1 |
| 09:54.22 | jsohler | donation-out3 is last... Whatever is last is the only thing the caller will hear. |
| 09:54.24 | ChannelZ | anyways no other playbacks are showing, so it's probably something busted with the AGI |
| 09:54.39 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 09:54.40 | ChannelZ | well according to your flood it was donation-in1 |
| 09:54.57 | jsohler | donation-in1 is first |
| 09:55.32 | jsohler | then lotw612, sil500, and donation-out3 |
| 09:55.52 | ChannelZ | re: according to your flood the only thing played was in1 |
| 09:56.09 | ChannelZ | so perhaps you have bigger problems like looking at the wrong PHP script? |
| 09:56.35 | ChannelZ | if you post it fast I will give it a glance but it's way past my bed time |
| 09:57.14 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 09:57.38 | jsohler | The problem isn't in the code. I'm looking for something that could have changed with the behavior of "root" that would impact the app... or a change in PHP or Asterisk settings that would impact the app. |
| 09:57.49 | ChannelZ | great have fun with that |
| 09:58.55 | ChannelZ | I've asked enough questions you won't answer. Goodnight |
| 09:59.23 | jsohler | Ok, I thought I was answering everyting asked. |
| 10:00.19 | jsohler | I just know its not the code because this happened before, and they "undid" whatever they had done with the server, and it went back to working fine... and has worked fine for months. |
| 10:00.27 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 10:01.43 | benngard | did notice some strange results when running: Action: Originate\r\nChannel: SIP/0317998975\r\nApplication: Dial\r\nData: OOH323/003190510@Avaya\r\nTimeout: 15000\r\nCallerID: Magnus Benngard<0317998975>\r\n\r\n, can any one se any error on that line? |
| 10:02.20 | jsohler | Also... Please tell me a question that I didn't answer. I'm a bit confused. |
| 10:02.52 | benngard | the cdr.dst is empty on that call but the call "works" |
| 10:03.26 | jsohler | ChannelZ: Anyway... thx and sleep well |
| 10:06.54 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 10:09.08 | *** part/#asterisk BeeBuu (~beebuu@121.9.232.26) |
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| 10:42.31 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 10:42.36 | E-bola | Hello |
| 10:43.21 | zafar_ | hi |
| 10:44.02 | E-bola | Do anybody have any information on using asterisk as a multi tenant platform (having multiple companies use the same asterisk server) |
| 10:44.28 | zafar_ | i ve just installed asterisk 1.6 on centos 5.4 i m getting this error |
| 10:44.30 | zafar_ | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? |
| 10:44.31 | E-bola | I thought it was simply giving each company its own context but it seems to have a couple of issues, so im wondiering if there is a prefered method or any examples etc |
| 10:44.51 | E-bola | im looking for a way to do this with a "raw" asterisk and not using any of the GUI overlays |
| 10:45.38 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 10:46.49 | zafar_ | i can run asterisk after asterisk -& but when i stop it i cant run it agian, the file asterisk.ctl files was not created so i created a blank asterisk.ctl with touch |
| 10:47.06 | zafar_ | any help? |
| 10:48.56 | zafar_ | plz guys help |
| 10:49.04 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 10:54.44 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 11:00.41 | kaldemar | zafar_: use an init script to start asterisk. if you compiled from source, you can install one with "make config". |
| 11:01.00 | kaldemar | zafar_: don't create the .ctl file manually. |
| 11:01.29 | *** join/#asterisk Heretic (~fallen@dsl-246-86-195.telkomadsl.co.za) |
| 11:01.39 | kaldemar | you see "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?" when trying to attach asterisk's CLI with asterisk -r when asterisk is not running. |
| 11:02.09 | Heretic | lo all |
| 11:02.31 | zafar_ | i followed instructions here : http://www.selbytech.com/2010/01/how-to-setup-asterisk-1-6-2-on-centos-5-4/ |
| 11:04.25 | zafar_ | i cant find the asterisk user in /etc/passwd file even :( |
| 11:04.55 | zafar_ | do i need to re-install it? |
| 11:06.04 | zafar_ | one more thing dahdi is not installed on this machine, does it make any difference? |
| 11:06.12 | kaldemar | if you followed those instructions, you shouldn't find asterisk user in /etc/passwd. |
| 11:06.36 | kaldemar | you don't need to re-install asterisk. just start it with the init script, "/etc/init.d/asterisk start" |
| 11:06.41 | *** join/#asterisk albertoandrade (~albertoan@189.58.28.177.dynamic.adsl.gvt.net.br) |
| 11:06.46 | kaldemar | then you can attach to the CLI with "asterisk -r" |
| 11:07.30 | kaldemar | zafar_: dahdi is needed for e.g. MeetMe conferences and IAX2 trunking, you won't need it for basic usage. |
| 11:07.50 | zafar_ | i need iax2 tunks |
| 11:08.45 | zafar_ | when i tried installing dahdi i got error : You do not appear to have the sources for the 2.6.18-164.10.1.el5PAE kernel installed. |
| 11:08.51 | kaldemar | trunking is a special case of IAX2 usage to decrease bandwidth usage. |
| 11:09.28 | zafar_ | i ll need to connect this box to other boxes through iax2 |
| 11:09.40 | kaldemar | you won't need DAHDI for that. |
| 11:10.17 | zafar_ | ok |
| 11:18.48 | zafar_ | Kaldemar you rock :) |
| 11:22.23 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 11:22.39 | kaldemar | zafar_: no problem |
| 11:23.13 | zafar_ | Kaldemar asterisk is good now, would u recomment a link to follow for the rest of the setup, like extension, routs, trunks etc |
| 11:24.22 | kaldemar | ~book |
| 11:24.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:24.27 | kaldemar | that's a good one. |
| 11:27.37 | zafar_ | hmmm that 604 pages book need a week to study that :) |
| 11:27.46 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 11:29.43 | kaldemar | you don't have to read it all, just the relevant part. see dialplans and connecting asterisk boxes. configuring a working box won't happen in a day, unless you're experienced. |
| 11:30.16 | zafar_ | u r right |
| 11:31.00 | zafar_ | i have used piaf but first time using raw asterisk |
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| 12:52.37 | *** join/#asterisk HenrikJott (~info@d83-183-134-141.cust.tele2.se) |
| 12:53.56 | HenrikJott | Hi all! I want to completely turn off music on hold for all my queues, what´s the best way to do that? |
| 12:54.14 | Gido-E | just no sound? |
| 12:54.24 | Gido-E | you can remove the audio files |
| 12:55.35 | fiddur | HenrikJott: Set i.e. musicclass = none in queues.conf |
| 12:55.48 | c0rnoTa | 'r' -- ring instead of playing MOH. Periodic Announcements are still made, if applicable. |
| 12:55.58 | c0rnoTa | option for Queue |
| 12:57.04 | benngard | for example: Queue(0317998989,rt) |
| 12:57.45 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 12:57.46 | HenrikJott | fiddur: alright! thanks, i will try that! |
| 12:58.42 | Kobaz | anyone aware of issues happening with very complex dialplan affecting audio processing? |
| 12:58.57 | HenrikJott | c0rnoTa: okay, but my agents already have a connection so i want the call to be connected immediatly, will it work with the -r anyway? |
| 13:00.21 | *** join/#asterisk pawz (~pawz@210.56.91.72) |
| 13:01.55 | *** join/#asterisk Yedidya (~chatzilla@host86-142-22-34.range86-142.btcentralplus.com) |
| 13:02.35 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-rdvbcemwilnhpdpk) |
| 13:03.12 | c0rnoTa | never tried queueing with connected offhook agents and immediatly connections with callers :) i'm always using ringing type (incoming call on agent). that option should switch off the music and play dial tones for callers. |
| 13:04.08 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 13:04.24 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 13:04.50 | E-bola | Queue(Reception,r,16) |
| 13:04.54 | c0rnoTa | by the way, HenrikJott, i think switching off the music and immediatly connection to agent are not wired |
| 13:05.03 | E-bola | is that not valid syntax for a 16 timeout queue in 1.6? |
| 13:05.13 | E-bola | it seems as if asterisk is not registering that i want 16 seconds timeout |
| 13:05.18 | E-bola | it just keeps ringing int he queue |
| 13:05.42 | HenrikJott | c0rnoTa: ahh, okay.. i´m building a dialer for a telemarketing company to the incoming calls to the queue are actually outgoing and answered calls. But we experience a short delay between the customers answer ang the agents connection and i noticed that asterisk tried to play some musiconhold and i just wanted to turn that off to minimize the delay |
| 13:05.47 | c0rnoTa | E-bola: Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,gosub][,rule]): |
| 13:05.58 | E-bola | yes ive read the voip entry |
| 13:06.06 | E-bola | do i need to insert all the ,'s ? |
| 13:07.00 | c0rnoTa | E-bola: Queue(reception,r,,,16) should work |
| 13:07.15 | E-bola | thanks, ile try it |
| 13:07.26 | HenrikJott | c0rnola: right now my queue-line is Queue(beep_connection||/home/asterisk/agent_connection.php) , i´m using mysql realtime |
| 13:08.00 | HenrikJott | sorry, missed a bit: Queue(${queue}|||beep_connection||/home/asterisk/agent_connection.php) |
| 13:09.10 | c0rnoTa | try Queue(${queue}|r||beep_connection||/home/asterisk/agent_connection.php) |
| 13:09.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:10.04 | E-bola | c0rnoTa: thanks works like a charm |
| 13:10.42 | HenrikJott | c0rnola: i will, brb |
| 13:11.35 | c0rnoTa | E-bole, glad to read it :) |
| 13:12.08 | *** join/#asterisk fiddur (~fiddur@192.121.104.121) |
| 13:15.06 | HenrikJott | c0rnola: i have one more question, i found another option for queue: "d : data-quality (modem) call (minimum delay)." do you know what this means? |
| 13:15.18 | *** join/#asterisk TheDavidFactor (~chatzilla@fw1.safedataisp.net) |
| 13:16.08 | c0rnoTa | no, sorry |
| 13:17.54 | HenrikJott | c0rnola: okay, thanks anyway.. i will try it and get back to you with results =) |
| 13:18.27 | c0rnoTa | :) thanks, i'll be waiting for results |
| 13:18.56 | Heretic | www.thatsnotsexy.com |
| 13:18.57 | Heretic | :(((( |
| 13:19.47 | *** join/#asterisk Skeeter- (Skeeter@190-141.cgocable.ca) |
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| 13:27.57 | Kobaz | do de do |
| 13:28.10 | Kobaz | so noone's had experience with audio issues when running complex dialplan? |
| 13:28.52 | *** join/#asterisk garymc (~garymc@81.138.225.161) |
| 13:29.32 | *** join/#asterisk fiddur (~fiddur@192.121.104.121) |
| 13:30.08 | c0rnoTa | complex? |
| 13:30.34 | Kobaz | yeah like, many local channels running in tight loops |
| 13:31.33 | Kobaz | when there's 5 incoming calls, there's probably 60 channels total, many of then looping local channels |
| 13:32.02 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-gqwxxwtqkgcqyjlo) |
| 13:32.02 | Kobaz | i get random audio frame loss |
| 13:32.13 | Kobaz | in between asterisk and local lan polycom phones |
| 13:32.37 | c0rnoTa | wow. No, it's too hard to imagine ^) not to transmit frames |
| 13:32.45 | c0rnoTa | what's about LA? |
| 13:32.45 | Kobaz | the call recordings sound perfect.. originally I thought it was the t1... because previously I was seeing audio loss on the recordings from the t1 side |
| 13:33.09 | garymc | [TK]D-Fender : Ive installed my A200 connected the sync cable to my A101D, configured wanpipe. (I think correctly) now im wondering how i can test to see if its installed correctly |
| 13:34.06 | c0rnoTa | Kobaz: what's about load average on your system? |
| 13:34.25 | Kobaz | almost nothing |
| 13:34.31 | Kobaz | 0.43, 0.26, 0.16 |
| 13:34.34 | c0rnoTa | Kobaz: direct calls are sounds clear? |
| 13:34.45 | Kobaz | station to station sound fine |
| 13:35.27 | Kobaz | i rewrote all my dialplan in agi, so that should reduce processing on the asterisk side |
| 13:35.29 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:35.33 | Kobaz | we'll have to see how that works |
| 13:36.01 | Kobaz | which, i've been wanting to do that anyway.. there's so much crap in the dialplan now that asterisk printing every line, makes debugging incredibly difficult |
| 13:36.29 | garymc | :( |
| 13:36.35 | Kobaz | 450 lines of dialplan to process a single call |
| 13:36.38 | Kobaz | heh |
| 13:36.46 | Kobaz | well actually, it was more dialplan |
| 13:36.49 | Kobaz | it's not 450 lines of perl |
| 13:36.53 | Kobaz | s/not/now |
| 13:37.20 | Kobaz | it was like 800 lines of dialplan |
| 13:37.43 | c0rnoTa | :) yeah, it should works. AGI id good application |
| 13:37.46 | Kobaz | well, not even dialplan actually... haha... i twas 800 lines of AEL, which is probably like 1200 lines of dialplan |
| 13:37.52 | c0rnoTa | s/is/id |
| 13:38.09 | c0rnoTa | ^)) |
| 13:38.28 | [TK]D-Fender | garymc: USE IT |
| 13:38.33 | c0rnoTa | ~s/^/: |
| 13:38.38 | c0rnoTa | ~help |
| 13:38.47 | Kobaz | [TK]D-Fender: i'm losing audio frames in between asterisk and local lan sip phones (polycom) |
| 13:39.08 | garymc | [TK]D-Fender : I cant get it to work. In freepbx channel now trying to find out if ive setup something wrong in there :S |
| 13:39.09 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 13:39.27 | [TK]D-Fender | garymc: And the detail of your troubles is amazing as always |
| 13:40.07 | Kobaz | i assume that's also for me too |
| 13:40.11 | *** join/#asterisk grayhame (~miller@74-94-250-169-Nashville.hfc.comcastbusiness.net) |
| 13:40.31 | Kobaz | i really don't have much detail... the call recordings sound fine on the asterisk side |
| 13:40.36 | garymc | [TK]D-Fender : sorry :( |
| 13:40.39 | Kobaz | i wonder if i can get the polycoms to record audio |
| 13:41.20 | Kobaz | or replicate the problem with a sip phone that has built in recording |
| 13:41.28 | Kobaz | soft phone of some sort |
| 13:43.33 | c0rnoTa | Kobaz: one day i meet a problem like yours: record was fine, but sound on peer side was... not chopped but polluted by other sounds. It sounds like asterisk's files, but on my station it wasn't played. |
| 13:44.17 | Kobaz | there isn't other asterisk sounds playing to the phones |
| 13:44.31 | Kobaz | it could be an internal sound in the phone, or something, playing at the same time someone is speaking |
| 13:44.50 | Kobaz | but this is like, a dropout, a loss of audio |
| 13:46.19 | c0rnoTa | Kobaz: as result my customer gives me back station because i' hadn't any proofs. But some times ago he told me, that it was a NAT issue. I don't know how, but gateway mixed UDP traffic from different station for a short time :) |
| 13:46.47 | c0rnoTa | that's why my recordings was clear |
| 13:47.01 | Kobaz | interesting |
| 13:47.13 | c0rnoTa | Kobaz: have you tried different codecs? |
| 13:47.17 | Kobaz | i haven't |
| 13:47.19 | Kobaz | it's all ulaw |
| 13:48.05 | c0rnoTa | ulaw UDP flow as like DoS attack on different switches |
| 13:48.24 | Kobaz | heh |
| 13:48.33 | c0rnoTa | if your AGI script wouldn't help you, try to change codec |
| 13:48.33 | Kobaz | the weird thing is it happens very randomly |
| 13:48.50 | Kobaz | it could be one call, it could be 6 calls |
| 13:49.02 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
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| 13:50.51 | c0rnoTa | looks like issue wired with workload on hardware |
| 13:51.40 | c0rnoTa | server, switches, phones, LAN bridging on phones |
| 13:52.14 | *** join/#asterisk pawz (~pawz@210.56.91.72) |
| 13:52.20 | Kobaz | could be |
| 13:52.38 | Kobaz | i just wanna make sure it's not the asterisk side |
| 13:53.05 | c0rnoTa | right way. and then try to use more compressed codec in future. |
| 13:53.09 | Kobaz | i need to eliminate potential sources of problems so i can have a smaller range of items to work with |
| 13:54.37 | c0rnoTa | it's world wide debugging route: to reduce of affected area |
| 13:54.42 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 13:55.08 | *** join/#asterisk hluesea (hluesea@213.144.126.109) |
| 13:55.50 | Kobaz | heh i know |
| 13:57.13 | c0rnoTa | that's why we are part of "world wide" :) |
| 14:06.49 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
| 14:13.07 | Katty | coughs up a lung |
| 14:13.49 | coppice | its OK. you have a spare |
| 14:14.00 | jaytee | put it in the feeder, maybe the squirrels will eat it |
| 14:14.54 | Katty | that made me giggle |
| 14:14.59 | Katty | which is very rare this early in the morning! |
| 14:15.02 | Katty | jaytee <3 |
| 14:15.05 | jaytee | :-) |
| 14:15.48 | Katty | i think i have my FPS thing sorted out. |
| 14:15.59 | jaytee | FPS? |
| 14:16.04 | Katty | frames per second |
| 14:16.05 | jaytee | frames per second? |
| 14:16.08 | jaytee | aha! |
| 14:16.18 | Katty | i wasn't allowing enough bandwidth to properly stream the video. |
| 14:16.24 | Katty | especially at the resolution i was wanting to stream it at |
| 14:16.45 | Katty | so now i've kicked it down to 640x480 @ 10fps, 300KBps stream speed |
| 14:17.03 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 14:17.03 | Katty | the second issue is the wireless is integrated directly into the dsl modem/router device. |
| 14:17.03 | jaytee | 10fps is kinda 'herky-jerky" |
| 14:17.09 | [TK]D-Fender | Katty: SD, ewwwwwww!!!!! |
| 14:17.17 | atis_work | huh, when i played CS i used the term FPM |
| 14:17.20 | Katty | and i think the sheer quantity of data is overloading the router. |
| 14:17.22 | atis_work | as Frags-per-minute |
| 14:17.36 | atis_work | per second it was to few.. |
| 14:17.37 | Katty | so i'm going to go get a wireless access point and disable the wireless on the router |
| 14:17.37 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-feosrtfdfpeekifj) |
| 14:17.45 | coppice | that is the sad thing about ADSL upstream speed. a 1920x1080p web cam would get you nowhere |
| 14:17.49 | Katty | jaytee: i know 10fps isn't the greatest around... |
| 14:18.06 | jaytee | the old AT&T vidphone in the 90's did like 11fps |
| 14:18.07 | Katty | jaytee: but if i go much higher, our upload speeds won't be able to accomodate |
| 14:18.28 | Katty | right now i'm comparing the logitech c600 and the pro 9000 |
| 14:18.30 | [TK]D-Fender | Katty: Upgrade! |
| 14:18.38 | Katty | [TK]D-Fender: i plan on it, but not yet (= |
| 14:18.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:18.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:18.55 | Katty | which one do you guys think looks better? |
| 14:18.57 | Katty | infobot: crittercam |
| 14:18.58 | infobot | crittercam is, like, Katty's broadcast of The Nut House Critter Cam @ http://ustre.am/8H5d and The Nut House Bird Bath @ http://ustre.am/bEBU |
| 14:19.06 | Katty | BB is running the c600, CC is running the 9000 |
| 14:19.06 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 14:19.07 | atis_work | uh, poor ADSL users.. |
| 14:19.13 | jaytee | my Windows 7 Enterprise ISO is finished burning....time to go fuck....er....upgrade someone. |
| 14:19.20 | atis_work | I have 100mbps ethernet at home for just 30$ month |
| 14:19.24 | Katty | and yes, i do know the BB is slightly crooked :P |
| 14:19.31 | jaytee | atis_work, where do you live? |
| 14:19.40 | atis_work | Eastern europe, Latvia |
| 14:19.50 | Katty | well we're paying about 60 a month for 712 upload speed |
| 14:20.04 | Katty | obviously my download speed is a lot better htan 712 |
| 14:20.09 | [TK]D-Fender | atis_work: DOOM!!!!!!!!! |
| 14:20.16 | jaytee | wow, that's pretty good speed. Is that 30 dollars latvian money or US$. |
| 14:20.27 | atis_work | us$ |
| 14:20.41 | Katty | i'll take two (= |
| 14:20.46 | atis_work | heh :) |
| 14:20.46 | jaytee | geez, I get 4mpbs for 42 bucks a month from Comcrap |
| 14:20.53 | jaytee | er.....Comcast |
| 14:21.03 | Katty | is that a cable connection? |
| 14:21.04 | coppice | there are people offering 100M for about $12 here now, but not where I live :-( |
| 14:21.05 | atis_work | internet is way cheaper in europe.. especially small countries :) |
| 14:21.06 | Katty | ours is through AT&T |
| 14:21.21 | Katty | atis_work: i couldn't agree more. |
| 14:21.40 | atis_work | coppice: just walk around building and collect some 10 interested people :) |
| 14:22.11 | Katty | jaytee: just out of curiosity, how do you feel about the FPS on CC or BB? |
| 14:22.18 | Katty | jaytee: does it feel "herky jerky" to you? |
| 14:23.09 | coppice | atis_work: we need to wait until hutchison finish laying new cables through the town. for some reason they seem to have stopped a couple of hundred m from us |
| 14:23.15 | bXi | hmm is there something in 2.6.20-vs2.2.0-gentoo that could block asterisk calls ? |
| 14:23.23 | Katty | blacklist |
| 14:23.34 | Katty | but that's more like rerouting calls |
| 14:23.52 | Katty | but you can reroute them to non existance! |
| 14:25.18 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:25.36 | *** join/#asterisk mechbangirc (~mechbangi@115.167.1.226) |
| 14:25.49 | bXi | oh wait that was actually the wrong question to the wrong channel :p |
| 14:26.37 | Katty | well fine |
| 14:26.38 | Katty | be that |
| 14:26.39 | Katty | way |
| 14:26.45 | Katty | DON"T TAKE MY ADVICE, SEE IF I CARE |
| 14:26.46 | Katty | sobs |
| 14:26.53 | bXi | *pats Katty on the back* |
| 14:26.56 | bXi | thing is |
| 14:26.58 | Katty | k, all better. |
| 14:27.05 | Zap-W | is the concept AOR record present in asterisk? |
| 14:27.05 | mechbangirc | hi i am supposed to reject calls with ISDN Q.391 #34 when my gateway capacity is reached. I have an asterisk server how can I do this with sip, any idea? |
| 14:27.10 | bXi | got a vserver running asterisk but i can't get 2 softphones to call eachother |
| 14:27.26 | Katty | that is most unfortunate. |
| 14:27.44 | bXi | i have twinkle which seems to authenticate properly |
| 14:28.00 | Katty | can't say i'm familiar with twinkle...i use zoiper. |
| 14:28.06 | bXi | and "Pocket Talk Business" for windows mobile phones that can register but times out |
| 14:28.25 | Katty | smells spiciously of natting |
| 14:28.50 | kaldemar | mechbangirc: when is your gateway capacity reached? when you know that in dialplan, you can just Hangup(34). |
| 14:29.19 | Katty | Hangup, little suzie, hangup |
| 14:29.37 | bXi | when i try to call extension 101 with twinkle (which is 100) i get a 404 Not Found |
| 14:29.43 | mechbangirc | kaldemar: sip wont send 34, it I think converts it to 503 |
| 14:29.52 | bXi | i've made 2 simple extensions using freepbx |
| 14:30.08 | Katty | oh...freepbx. |
| 14:30.14 | Katty | isee. |
| 14:30.22 | Katty | well that's an easy fix. |
| 14:30.30 | bXi | rm -rf freepbx ? |
| 14:30.31 | Katty | go to the nearest Home store and buy a sledge hammer. |
| 14:30.51 | Katty | apply sledge hammer to server, as hard as possible, until broken into eleventy billion pieces. |
| 14:30.58 | Katty | problem has been neutralized. |
| 14:31.00 | kaldemar | mechbangirc: what do you have connected to your asterisk box and where is the Q.931 cause 34 supposed to end up to? |
| 14:31.20 | bXi | so i take it that freepbx is a very bad choice |
| 14:31.43 | *** join/#asterisk crazeekennee (~crazeeken@static-72-87-91-10.prvdri.fios.verizon.net) |
| 14:31.48 | bXi | what about asterisk gui then? |
| 14:31.50 | *** join/#asterisk x86 (~x86@p3m/member/x86) |
| 14:31.58 | mechbangirc | kaldemar: i have a gsm gateway attached via sip interface and the other leg is internet via sip |
| 14:32.42 | mechbangirc | kaldemar: i know when the capacity is reached but there is no signal 34 for sip as far i know |
| 14:32.48 | crazeekennee | Hello all. I currently have a Samsung OfficeServ system in my office that we don't like. Does anyone know if we can use Asterisk with the Samsung phones ? |
| 14:32.52 | jaytee | Katty, sorry I had to step away for a bit. CC or BB? |
| 14:33.27 | StuZZZs | bXi: see /etc/asterisk/* |
| 14:33.28 | Katty | jaytee: short for crittercam or birdbath |
| 14:33.41 | jaytee | ah, so you have two separate cams now? |
| 14:34.02 | [TK]D-Fender | bXi: Asterisk GUI is not being maintained currently |
| 14:34.04 | Katty | nods |
| 14:34.10 | [TK]D-Fender | bXi: and no GUI is supported here |
| 14:34.16 | kaldemar | mechbangirc: you'll have to live with what you can send by SIP and how other leg interprets it. |
| 14:34.27 | bXi | okay so i'll have to go the manual route again then |
| 14:34.39 | [TK]D-Fender | crazeekennee: If they are proprietary digital sets, then no. |
| 14:35.47 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 14:36.03 | mechbangirc | kaldemar: i captured few packets. packets sent with hangup(34) are automatically converted to SIP:503. I dont know about the other leg interpretation yet. I have to ask them about it. |
| 14:36.57 | ManxPower-work | There is an RFC for Q.931<->SIP cause code mappings |
| 14:37.10 | ManxPower-work | (Asterisk users the same hangupcauses as Q.931, even for IAX) |
| 14:38.47 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:39.11 | mechbangirc | ManxPower-work: so how would you make a difference between congestion(capacity full) and some other server side problem as in 5XX codes |
| 14:39.39 | mechbangirc | ManxPower-work: and how can you measure ASR in this way? |
| 14:41.02 | voipmonk | oh lord |
| 14:41.40 | voipmonk | mechbangirc: if one person doesnt pick up , doesn't that kill your ASR ? |
| 14:42.00 | kaldemar | mechbangirc: it sends different 5XX messages depending on the issue. |
| 14:43.13 | kaldemar | mechbangirc: afaik, asterisk even adds SIP headers for the cause and the cause code. |
| 14:44.00 | mechbangirc | kaldemar: cause 34 says "no circuit available" which maps to 503 "service unavailable" and so are the ISDN causes: 38, 41, 42 ,47 converted to 503 |
| 14:44.00 | *** join/#asterisk gr0mit (~tim@78.149.128.243) |
| 14:45.40 | kaldemar | mechbangirc: the other end has to look deeper into the message, not just the message number. |
| 14:45.48 | mechbangirc | kaldemar: Asterisk sending this status line: SIP/2.0 503 Service Unavailable |
| 14:46.14 | mechbangirc | kaldemar: which field you are referring to? |
| 14:46.33 | Katty | fields of gold. |
| 14:47.03 | mechbangirc | voipmonk: ASR is never 100%. however this issue is decreasing it further |
| 14:47.05 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 14:48.01 | kaldemar | mechbangirc: X-Asterisk-HangupCause and X-Asterisk-HangupCauseCode are added if the cause is known, at least in "newer" versions. i took a peek at 1.6.2 source. it's not a standard solution, but the info is there. |
| 14:48.04 | Katty | so many critters this morning... |
| 14:48.17 | Katty | 6 squirrels are up and about |
| 14:48.35 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 14:48.35 | mechbangirc | kaldemar: thanx i will look into it. my laptop battery is dying see ya |
| 14:49.21 | bXi | okay got a custom sip.conf now |
| 14:50.18 | bXi | http://bluepunk.pastebin.com/m6b03d21d |
| 14:50.26 | bXi | this is giving me the same results as freepbx |
| 14:50.44 | [TK]D-Fender | bXi: because its a DIALPLAN ERROR |
| 14:51.21 | *** join/#asterisk came0 (~came0@rrcs-71-42-53-182.se.biz.rr.com) |
| 14:51.32 | [TK]D-Fender | bXi: *extension not found* |
| 14:51.45 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 14:52.50 | ManxPower-work | If just stopped talking to GUI users maybe they will go away. |
| 14:52.56 | ManxPower-work | Perhaps I need more coffee. |
| 14:53.03 | dlynes_laptop | hehehe |
| 14:53.13 | ManxPower-work | If people just stopped talking to GUI users maybe they will go away. |
| 14:53.23 | ManxPower-work | It's not like we can help them anyway. |
| 14:53.32 | Katty | oh, dlynes_laptop |
| 14:53.35 | Katty | just the person i'm looking for |
| 14:53.36 | dlynes_laptop | hi katty |
| 14:53.47 | dlynes_laptop | have you had your cup of joe yet this morning? |
| 14:53.52 | dlynes_laptop | it seems Manx hasn't :) |
| 14:53.57 | ManxPower-work | working on it |
| 14:55.04 | bXi | okay right i've changed my extensions.conf as well to reflect the sip accounts |
| 14:55.47 | ManxPower-work | bXi: you put the extension in the [sip] section of extensions.conf, right? |
| 14:55.54 | *** join/#asterisk moy (~moy@74.12.123.169) |
| 14:56.47 | bXi | yes |
| 14:57.09 | [TK]D-Fender | ManxPower-work: .... be nice |
| 14:57.16 | E-bola | [TK]D-Fender: any chance u kept the thing u helped me with about a week ago, about how to set an outgoing callerid in sip.conf on the account and then use it as a variable when dialing out? |
| 14:57.31 | *** part/#asterisk zafar_ (~IceChat7@82.137.239.121) |
| 14:57.48 | ManxPower-work | y don't u type you? |
| 14:58.02 | Katty | two more letters is too much work |
| 14:58.05 | [TK]D-Fender | E-bola: SetVar=outgoingcid=2125551234 |
| 14:58.06 | Katty | donchaknow |
| 14:58.13 | ManxPower-work | Katty: obviously. |
| 14:58.56 | ManxPower-work | E-bola: This isn't rocket science. You set the callerid info in sip.conf callerid= line and you access it as CALLERID(num) and CALLERID(name) in the dialplan. |
| 14:59.08 | E-bola | [TK]D-Fender: cool thanks |
| 14:59.15 | Katty | it really is rocket science |
| 14:59.18 | Katty | is disguise. |
| 14:59.20 | E-bola | ManxPower-work: No need to re-invent the wheel... |
| 14:59.21 | Katty | s/is/in/ |
| 14:59.55 | [TK]D-Fender | E-bola: .... not what he needed. |
| 14:59.59 | [TK]D-Fender | ManxPower-work: rather |
| 15:00.01 | [TK]D-Fender | gah |
| 15:00.49 | E-bola | hmm debian doesnt like upgrading asterisk from lenny to squeeze, bummer. |
| 15:02.00 | *** join/#asterisk mechbangirc (~mechbangi@115.167.1.226) |
| 15:03.27 | Kobaz | you should always build asterisk from source |
| 15:04.36 | E-bola | umm why? |
| 15:04.59 | Katty | why not |
| 15:05.02 | Kobaz | because it's one of those things that you want to "just work(tm)" if you upgrade other parts of the system |
| 15:05.13 | Katty | anything worth doing, is worth doing from scratch! |
| 15:05.18 | E-bola | because its easier and simpler to use ur dists package manager |
| 15:05.25 | Kobaz | E-bola: and more likly to break |
| 15:05.35 | Katty | yeah and i guess microwaved burritos is easier and simpler too |
| 15:05.39 | E-bola | umm why? Unless you install all requirements from scratch as well |
| 15:05.41 | Katty | but would you rather have a burrito, or a steak? |
| 15:05.43 | E-bola | they might break when you upgrade them |
| 15:05.56 | E-bola | i dont have time to make steak |
| 15:06.01 | E-bola | so ile take the microwave solutionm thank you :) |
| 15:06.05 | Kobaz | E-bola: as long as you keep required dependencies around, asterisk will be unaffected |
| 15:06.53 | Kobaz | E-bola: it's more about control, really |
| 15:07.05 | Kobaz | i want to control what happens when asterisk gets upgraded |
| 15:07.08 | E-bola | sure, but i dont need that control at all |
| 15:07.14 | Kobaz | i don't want the package manager to do it for me, since my systems are too important |
| 15:07.23 | E-bola | i prefer to let somebody handle it for me when possible |
| 15:07.37 | Kobaz | since what the package manager thought was best, might not, and probably will not be the best in my case |
| 15:07.54 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 15:09.47 | Katty | E-bola: okay, pick the microwaved burrito, but don't come complaining to use when you get the runs! |
| 15:10.11 | Katty | E-bola: you want quality, make it from scratch |
| 15:10.20 | Katty | E-bola: if you want quick and easy, you will pay for it in the long run. |
| 15:10.43 | E-bola | I'd complain to the debian guys, no worries |
| 15:10.56 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 15:11.00 | guax | ~book |
| 15:11.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:11.34 | ManxPower-work | Asterisk is (almost) the only software I compile from source. |
| 15:11.44 | Katty | http://i.imgur.com/PZEBE.jpg <- oh hey, i found ManxPower-work |
| 15:12.09 | ManxPower-work | I'm not even going to look. I'm sure it's a picture of a squirrel |
| 15:12.22 | ManxPower-work | or a dog, I guess. |
| 15:12.44 | Katty | no. |
| 15:12.47 | Katty | it's off reddit. |
| 15:12.59 | Katty | but you don't have to look, it's okay. i won't be offended. |
| 15:13.12 | ManxPower-work | What is reddit? |
| 15:13.26 | Katty | it's like digg |
| 15:13.43 | ManxPower-work | I've heard of digg. Don't know what it is. |
| 15:13.51 | ManxPower-work | I have little use for social networking sites. |
| 15:14.01 | Katty | people link news articles, pictures, movies etc. |
| 15:14.15 | Katty | they then get upvoted or downvoted, based on popularity by the subscribers of the website |
| 15:14.16 | ManxPower-work | I already talk to my friends and the rest of the world can go to hell. |
| 15:14.32 | bXi | which configuration files can affect my dialplan? i'm still getting the 404 |
| 15:14.33 | guax | Katty, LOL, i opened the link =P |
| 15:14.34 | ManxPower-work | Katty: If that's what makes you happy. |
| 15:14.41 | ManxPower-work | bXi: all of them. |
| 15:14.55 | Katty | guax: i think it applies to ManxPower-work very well, don't you think? ;) |
| 15:15.05 | kaldemar | bXi: sip.conf and extensions.conf |
| 15:15.13 | ManxPower-work | bXi: A better question will yield better answers. |
| 15:15.16 | [TK]D-Fender | bXi: Dialplan = Extensions.CONF |
| 15:15.18 | guax | not fallowing the discussion, just a addicted link clicker =P |
| 15:15.31 | Kobaz | hmmm |
| 15:15.42 | guax | following*? |
| 15:15.43 | *** join/#asterisk albertoandrade (~albertoan@189.58.28.177.dynamic.adsl.gvt.net.br) |
| 15:15.45 | *** join/#asterisk Ahmadi (~4e26ab14@gateway/web/freenode/x-poljhxkpkpnomsha) |
| 15:15.48 | Kobaz | how do i check if the channel i'm bridged too hung up in agi |
| 15:15.52 | Ahmadi | Hi |
| 15:16.03 | Kobaz | if i spawn off a local channel into an agi script with an infinite loop, it'll keep running |
| 15:16.10 | Ahmadi | my PBX dont detect keypress of my phones |
| 15:16.11 | Kobaz | even if the other side hung up... asterisk doesn't kill the agi leg |
| 15:16.19 | Ahmadi | Please help me |
| 15:16.20 | bXi | i've changed my extensions.conf to a very basic sample with 2 extensions calling my devices |
| 15:16.24 | Katty | mah pbx don't worrkkkk |
| 15:16.41 | Katty | don't be hatin yo |
| 15:16.56 | ManxPower-work | bXi: context= in sip.conf tells asterisk what context the call will land in when a call comes in from that device. |
| 15:17.06 | ManxPower-work | (what context in extensions.conf, of course) |
| 15:17.22 | ManxPower-work | bXi: pastebin your sip.conf, extensions.conf and the debug you have of the failed sip call. |
| 15:17.24 | kaldemar | Ahmadi: change dtmfmode |
| 15:17.25 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 15:17.39 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.195) |
| 15:17.47 | Ahmadi | kaldemar, Where is it? |
| 15:17.50 | Katty | hello mister smooth operator |
| 15:17.59 | ManxPower-work | Kobaz: that is correct. Your agi needs to handle the signals sent to it by Asterisk. This has been the case since Asterisk 1.2 |
| 15:18.01 | ks3 | Is there an application similar to Read that will allow the caller to enter # ? |
| 15:18.10 | smooth_penguin | hello Katty :> |
| 15:18.20 | smooth_penguin | I didnt see the woodpecker yesterday |
| 15:18.20 | kaldemar | Ahmadi: what protocol do your phones use? |
| 15:18.22 | Chainsaw | ks3: Is there anything about Read that makes it unsuitable for your purpose? |
| 15:18.31 | Katty | smooth_penguin: well, the FPS was pretty messed up yesterday |
| 15:18.33 | Ahmadi | DTMF |
| 15:18.33 | Kobaz | ManxPower-work: what signal is it? |
| 15:18.35 | [TK]D-Fender | ks3: Make a small IVR to read in digit-by-digit however you want |
| 15:18.38 | *** join/#asterisk davevg-btwtech (~davevg__@75.97.64.33.res-cmts.senj.ptd.net) |
| 15:18.44 | Katty | smooth_penguin: i can getyou a big photo of bubbles tho, if you'd like |
| 15:18.46 | ks3 | Chainsaw: It seems to use # as the end-of-input marker |
| 15:18.51 | ManxPower-work | Kobaz: every single example of using asterisk-perl should have sample code for you. |
| 15:18.52 | smooth_penguin | yeah, cool! |
| 15:18.56 | Katty | k |
| 15:19.02 | Kobaz | ManxPower-work: generally my scripts end by themselves, but this one has an infinite loop |
| 15:19.06 | Kobaz | ManxPower-work: hmm, oh okay |
| 15:19.33 | Ahmadi | kaldemar,DTMF |
| 15:19.34 | bXi | http://bluepunk.pastebin.com/m667ab6e4 this is the extensions.conf http://bluepunk.pastebin.com/m6b03d21d sip.conf and i'm quite unsure how to get the debug info |
| 15:19.49 | ManxPower-work | bXi: however you are seeing the 404 error on the Asterisk console |
| 15:20.03 | bXi | thats just my sip client telling me 404 Not Found |
| 15:20.04 | ks3 | [TK]D-Fender: You mean just have an extension like _[0-9#*] or similar that updates a variable? |
| 15:20.08 | kaldemar | Ahmadi: analog phones? SIP? something else? |
| 15:20.26 | [TK]D-Fender | ks3: Yes. |
| 15:20.26 | bXi | only thing i get on asterisk console is Using SIP RTP CoS mark 5 (i am using a verbosity of 95 at this moment) |
| 15:20.27 | Kobaz | ManxPower-work: well, this is fastagi |
| 15:20.34 | ManxPower-work | bXi: what number are you dialing? |
| 15:20.40 | Kobaz | ManxPower-work: do i don't get signals from asterisk |
| 15:20.40 | bXi | 101 from 100 |
| 15:20.42 | ManxPower-work | Kobaz: thank you for wasing my time. |
| 15:20.45 | Ahmadi | kaldemar, my phone is SIP Client, my gateway is Cisco Access Server, im using Asterisk as PBX |
| 15:20.52 | ks3 | [TK]D-Fender: That was my "last resort" options, just wasn't sure if there was anything I was missing |
| 15:20.53 | Kobaz | ManxPower-work: heh sorry |
| 15:21.04 | [TK]D-Fender | ks3: Only a little motivation :) |
| 15:21.18 | bXi | i've also tried calling 100 from 100 which does work |
| 15:21.28 | ManxPower-work | bXi: Ah! Your extensions.conf is not valid. I don't see a [general] and I don't see a [globals] |
| 15:21.29 | Kobaz | ManxPower-work: so how would i check if my bridged channel is gone? i think i'll need to use a GROUP() |
| 15:21.47 | bXi | hmmm i must've missed those from my example |
| 15:21.48 | ManxPower-work | Kobaz: I have no fucking clue or I would have answered you the first 10 times you asked. |
| 15:21.55 | ks3 | [TK]D-Fender: I'm sure my boss will provide proper motivation if I don't get it done soon :D |
| 15:22.00 | kaldemar | Ahmadi: dtmfmode is a parameter in sip.conf. change it to match what your phone and the cisco server know. |
| 15:22.02 | ManxPower-work | bXi: the less accurate your information the more useless it is. |
| 15:22.05 | Kobaz | ManxPower-work: hehe, fine fine, i'll figure it out :) |
| 15:22.15 | [TK]D-Fender | ManxPower-work: Not both necessary |
| 15:22.24 | Kobaz | ManxPower-work: i only asked twice total |
| 15:22.30 | [TK]D-Fender | bXi: go enable SIP DEBUG from * CLI and show us the failed call |
| 15:22.43 | *** join/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 15:23.02 | LemensTS | anyone know xlite syntax to allow *69 to dial? ive read their manual can't get it to work |
| 15:23.19 | Ahmadi | kaldemar, Where can i found the file, im newbie |
| 15:24.17 | kaldemar | Ahmadi: by default it is in /etc/asterisk, but since i scent a GUI is involved, use it to change the mode. |
| 15:24.32 | bXi | ManxPower-work: so basicly i need to fix those 2 sections you just mentioned |
| 15:24.47 | [TK]D-Fender | LemensTS: So * doesn't get the call at all? |
| 15:25.06 | [TK]D-Fender | bXi: No, you need to enable SIP DEBUG from * CLI like I told you and show us the actual ERROR |
| 15:25.23 | bXi | so what do i type into the CLI? |
| 15:25.33 | bXi | just "SIP DEBUG from *" |
| 15:25.59 | kaldemar | bXi: sip set debug on |
| 15:26.43 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-zinngvmqqpsbqblh) |
| 15:27.30 | [TK]D-Fender | bXi: "sip set debug" |
| 15:27.43 | LemensTS | TKD-Fender: nope it says "Call failed: Address Incomplete" on xlite, doesn't make it to asterisk. I actually put just *69 in it, and tried /*69 \*69 |
| 15:28.12 | ManxPower-work | LemensTS: you don't have *69 in your dialplan, |
| 15:29.03 | [TK]D-Fender | LemensTS: that sounds like a server response which should mean * got a request. |
| 15:29.12 | [TK]D-Fender | LemensTS: You say nothing appears in SIP DEBUG? |
| 15:29.19 | bXi | gimme a moment to copy paste this into pastebin |
| 15:29.41 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:30.13 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:30.16 | LemensTS | ManxPower: doh forgot to reload asterisk to make changes go into effect. I woulda thought it would of said something in the cli at least, i got verbose up decently high. Guess i should have done sip debug like TKD-Fender said. Thanks |
| 15:31.04 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 15:31.15 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 15:31.26 | bXi | http://bluepunk.pastebin.com/m7dafb6a9 the sip debug log |
| 15:31.46 | ManxPower-work | LemensTS: now you understand why we need sip debug |
| 15:32.29 | ManxPower-work | bXi: put allowguest=no in sip.conf and fix the configuration of your phone to send the username Asterisk expects |
| 15:32.37 | *** part/#asterisk ks3 (~ks3@74.203.195.1) |
| 15:32.48 | Ahmadi | kaldemar, the file have only some #include , What must i do on it? |
| 15:33.02 | ManxPower-work | bXi: NEXT time TELL us you are using NAT!!!!!!!! |
| 15:33.22 | ManxPower-work | I give up. |
| 15:33.26 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 15:34.30 | Katty | don't worry, he'll be back later |
| 15:34.54 | [TK]D-Fender | bXi: http://bluepunk.pastebin.com/m6b03d21d <--- where is the 100 & 101 STUFF WE KEEP SEEING? |
| 15:34.55 | bXi | i'm actually suprised that theres NAT invloved |
| 15:35.06 | [TK]D-Fender | bXi: yOU ARE GETTING AUTH ERRORS everywhERE IN THERE |
| 15:35.19 | [TK]D-Fender | bXi: tHERE ISN'T... HE'S JUST BURNING OUT AGAIN |
| 15:35.46 | *** join/#asterisk DrAk0 (~luisjosed@nelug/coreteam/luisjose) |
| 15:40.57 | *** join/#asterisk DrAkO (~luisjosed@nelug/coreteam/luisjose) |
| 15:41.38 | dlynes_laptop | [TK]D-Fender, my god...you've degenerated into aol-speak :((( |
| 15:41.58 | [TK]D-Fender | dlynes_laptop: No, just complete caps fail |
| 15:42.41 | E-bola | Now you have to buy cookies for everyone! |
| 15:43.20 | [TK]D-Fender | spins up "Nookie" by Limp Bizkit, just for E-bola |
| 15:43.45 | dlynes_laptop | damn...haven't heard of that band in a long time |
| 15:44.56 | Ahmadi | kaldemar, my Cisco setting is "dtmf-relay h245-alphanumeric" , What must use is Asterisk !? |
| 15:46.25 | kaldemar | Ahmadi: how are you connecting to the cisco device? with what protocol? |
| 15:46.36 | Ahmadi | SIP |
| 15:46.41 | Ahmadi | SIP/UDP |
| 15:46.59 | bXi | [TK]D-Fender: oops that sip.conf is actually the template i used for my sip.conf |
| 15:47.31 | [TK]D-Fender | bXi: Your auth is busted everywhere, and you've shown us stuff that doesn't add up. You aren't doing anyone any favours here.. |
| 15:47.35 | bXi | http://bluepunk.pastebin.com/m1c1a5dc6 this is the correct one |
| 15:49.02 | [TK]D-Fender | bXi: Contact: <sip:asterisk@192.168.101.250> |
| 15:49.15 | [TK]D-Fender | bXi: I can also see you have not set your server up properly to work from behind NAT |
| 15:49.18 | kaldemar | Ahmadi: rfc2833, info, shortinfo or inband. |
| 15:49.33 | [TK]D-Fender | bXi: Either way your device's aren't authing |
| 15:50.09 | [TK]D-Fender | bXi: You have not specified codecs, prevented reinvites, or much else... |
| 15:50.29 | [TK]D-Fender | bXi: Your entire setup is largely messed up |
| 15:50.30 | *** part/#asterisk icyValk77 (~icyValk77@cl-670.lon-02.gb.sixxs.net) |
| 15:50.31 | Ahmadi | kaldemar, Im using that, but dont work |
| 15:50.57 | Ahmadi | kaldemar, What do you think, in my extension option, dtmf=rfc2833. its default value of it |
| 15:51.17 | [TK]D-Fender | Ahmadi: Test with another SIP client |
| 15:52.14 | Ahmadi | [TK]D-Fender, ok, please wait a moment |
| 15:52.54 | kaldemar | Ahmadi: they have to be the same between cisco and asterisk and asterisk and the SIP client. |
| 15:53.40 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:53.53 | E-bola | hmm after upgrading to 1.6 from 1.4 i got an issue with my sip->pstn provider |
| 15:54.06 | E-bola | it seems like its now matching the first peer in sip.conf that matches the host ip |
| 15:54.20 | E-bola | i got several accounts from the provider each with a different DID, but all coming from the same ip address |
| 15:54.24 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:54.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:54.25 | E-bola | how do i make asterisk match the right one? |
| 15:56.06 | p3nguin | extensions! |
| 15:56.50 | E-bola | http://pastebin.com/d6e391df |
| 15:56.58 | E-bola | well its all about the accounts as u can see on pastebin |
| 15:57.06 | E-bola | it basically matches the wrong peer |
| 15:57.26 | p3nguin | Failed to authenticate device "43332730" |
| 15:57.39 | [TK]D-Fender | E-bola: "insecure-no" |
| 15:57.44 | [TK]D-Fender | E-bola: "insecure=no" |
| 15:57.58 | E-bola | alright |
| 15:59.12 | E-bola | mm nope |
| 15:59.14 | E-bola | doesnt change a thing |
| 15:59.46 | E-bola | if its not obvious there is a out-telsome-39163337 which it should be matching instaid of out-telsome-39163330 |
| 15:59.53 | p3nguin | Did you remember to sip reload? |
| 16:00.09 | E-bola | ya i even restarted |
| 16:00.52 | [TK]D-Fender | e-me-I'm not seeing configs.. |
| 16:01.01 | [TK]D-Fender | E-bola: I'm not seeing configs.. |
| 16:02.26 | E-bola | http://pastebin.com/d5b52b807 |
| 16:02.28 | E-bola | thats a sip debug |
| 16:02.30 | E-bola | configs coming up |
| 16:03.36 | E-bola | http://pastebin.com/d6c017a9c |
| 16:04.41 | E-bola | i dont get whats changed between 1.4 and 1.6 to cause this... |
| 16:05.06 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:05.33 | p3nguin | Is => even valid in sip definitions? |
| 16:05.55 | bXi | hm |
| 16:06.07 | bXi | i suspect that this windows mobile software doesnt play nice with asterisk at all |
| 16:06.22 | E-bola | p3nguin: works fine on all my servers... |
| 16:06.32 | [TK]D-Fender | E-bola: type=user |
| 16:06.33 | p3nguin | That's too bad. |
| 16:08.28 | E-bola | user did the trick |
| 16:09.12 | p3nguin | [tk]d-fender: If I use type=user (on 1.4) for my ITSP that sends calls INBOUND, it fails. With it set to type=friend, everything is fine. And no, I don't have proof handy, just believe me that it's what happens. |
| 16:09.46 | [TK]D-Fender | E-bola: You're welcome |
| 16:10.27 | E-bola | hmmm |
| 16:11.41 | E-bola | [TK]D-Fender: would changing it from friend (which i originally had type to) tyo user make any difference when i call out like: Dial(SIP/${DialedNumber}@out-telsome-39163337) |
| 16:11.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:11.55 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.172.179) |
| 16:12.09 | [TK]D-Fender | E-bola: Yes, you'll need another peer for it |
| 16:12.12 | superbeef | exten => _X.,1,Set(CALLERID(num)=${IF([$(LEN(${CALLERID(num)}) < 7]?51${CALLERID(num)}:${CALLERID(num)}) |
| 16:12.12 | superbeef | Trying to prepend "51" if the length of hte callerid is less than 7 digits..... somehow my syntax is not delivering. Anything obvious? |
| 16:12.49 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 16:13.04 | *** join/#asterisk albertoandrade (~albertoan@189.58.28.177.dynamic.adsl.gvt.net.br) |
| 16:13.05 | E-bola | [TK]D-Fender: hmm so i now gotta create dual entries in sip.conf for 1 account at my providers? One for incomign and one for outgoing? |
| 16:13.08 | [TK]D-Fender | superbeef: (LEN <--- wrong brace |
| 16:13.17 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 16:13.26 | [TK]D-Fender | E-bola: Yup |
| 16:13.34 | superbeef | [TK]D-Fender: thanks lemme try again |
| 16:14.09 | stones_ | Anyone here use a2billing? |
| 16:14.15 | E-bola | [TK]D-Fender>: shouldt that be solved by settign type=friend? acording to voip-info its: either "friend" (peer+user), "peer" or "user" |
| 16:14.22 | E-bola | so friend should be both peer and user.... |
| 16:14.40 | stones_ | please? someone heh |
| 16:14.42 | [TK]D-Fender | E-bola: Friend = DEAD |
| 16:14.42 | p3nguin | In 1.6, friend has changed. |
| 16:14.49 | E-bola | ohhh |
| 16:14.50 | Ad-Hoc | hi |
| 16:14.55 | E-bola | bummer makes my sip.conf much larger |
| 16:17.37 | p3nguin | So what's the reason that type=user puts my ITSP into the default context instead of the one configured? When type=friend, the context is obeyed. |
| 16:17.56 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:18.08 | *** join/#asterisk minotaur01 (~minotaur0@24.215.3.50) |
| 16:19.17 | dlynes_laptop | [TK]D-Fender, how did 'friend' change in 1.6? |
| 16:20.22 | p3nguin | With type=peer, it goes to the right context, though. I thought type=user was for calls going inward to asterisk. |
| 16:20.57 | dlynes_laptop | p3nguin, peer is for incoming; user is for outgoing |
| 16:21.12 | p3nguin | Well damn. |
| 16:21.36 | E-bola | missed the memo on that one |
| 16:21.42 | E-bola | it explains alot though :) |
| 16:23.15 | *** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru) |
| 16:23.46 | [TK]D-Fender | [11:20]<dlynes_laptop>p3nguin, peer is for incoming; user is for outgoing <- backwards |
| 16:23.53 | E-bola | ya i was just about to say lol |
| 16:24.06 | E-bola | i hope, peer is for outgoing since thats what im changing for 20 accounts... |
| 16:24.06 | [TK]D-Fender | And if there is no user taking precedence, peer is ALSo incoming |
| 16:24.08 | bmoraca_work | peers can be matched for either. |
| 16:24.20 | E-bola | if its either, why should u have both? |
| 16:24.45 | p3nguin | There are plans to completely remove type= for sip in the future. |
| 16:25.47 | bmoraca_work | E-bola: you don't need both if you have a register |
| 16:26.13 | E-bola | ahh, maybe thats why i've chosen not to use register lines |
| 16:26.14 | p3nguin | I can't remember where I read the changes to "friend" |
| 16:26.24 | E-bola | my provider just sends my calls directly to my static ip |
| 16:26.58 | bmoraca_work | E-bola: that's why you need both, then. unless you disable the requirement of SIP to authenticate (not advised) |
| 16:27.19 | E-bola | hmm [TK]D-Fender: now ive created a user and peer for each account, but now it just matches the first peer again |
| 16:28.17 | stones_ | anyone here use a2billing... ? |
| 16:28.29 | E-bola | http://pastebin.com/d1b9cc27a |
| 16:28.51 | [TK]D-Fender | stones_: please find a SPECIFIC question to aks already. This is not an a2billing support channel |
| 16:29.00 | E-bola | its like any peer with a matching host ip is being matched regardless of what number/username its for |
| 16:29.09 | dlynes_laptop | [TK]D-Fender, well, i suppose it depends on your perspective |
| 16:29.12 | stones_ | Im having trouble tieing asterisk and a2 together.. something is wrong somewhere I dont know what |
| 16:29.19 | stones_ | My call is being rejected due to no extensions. |
| 16:29.22 | p3nguin | e-bola: type=user matches on username. |
| 16:29.32 | p3nguin | e-bola: type=user never matches on IP address. |
| 16:29.36 | dlynes_laptop | [TK]D-Fender, i was thinking from the asterisk/phone perspective....asterisk is the peer, the phone is the user |
| 16:29.41 | [TK]D-Fender | stones_: Then you have failed to grasp even basic dialplan... time to sit down with THE BOOK... |
| 16:29.42 | [TK]D-Fender | ~book |
| 16:29.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:29.56 | E-bola | p3nguin: does peer only match on ip? |
| 16:30.02 | E-bola | because that seems to be what im seeing |
| 16:30.24 | p3nguin | e-bola: for inbound or outbound? They are different. |
| 16:30.29 | E-bola | inbound |
| 16:30.37 | E-bola | atm my issue is inbound |
| 16:30.47 | stones_ | im going everything through a2billing. |
| 16:30.49 | p3nguin | e-bola: For inbound, peer matches on IP and port. |
| 16:31.08 | E-bola | p3nguin: how am i supposed to have multiple incoming acounts/DID's from the same IP then? |
| 16:31.14 | stones_ | what is a good book to get? |
| 16:31.42 | [TK]D-Fender | stones_: that's what you WANT to do perhaps, but you've set your dialplan or devices up wrong. |
| 16:31.57 | [TK]D-Fender | stones_: I just linked it to you. Please pay attention |
| 16:32.06 | dlynes_laptop | p3nguin, so what's changed with respect to the behaviour of 'type=friend' in 1.6? |
| 16:32.23 | *** join/#asterisk zorton (~zorton@unaffiliated/zorton) |
| 16:34.09 | zorton | i'm thinking of putting asterisk into my soekris box. It seems like I should have hardware echo on my card but the $600 price tag for a 1 FXO 3 FXS PCI card is putting me off somewhat. Are there good alternatives or will the software echo canceler work okay? |
| 16:35.14 | dlynes_laptop | zorton, for four analog ports, if you're in the middle of the city, you probably won't have a problem with the software echo canceller |
| 16:35.16 | p3nguin | dlynes_laptop: I wish I remembered where I read it so I could read it again rather than guessing. |
| 16:35.45 | dlynes_laptop | zorton, however, if you're living somewhere with notoriously bad lines (even with the telco), you're going to want a hardware echo canceller |
| 16:35.55 | zorton | dlynes_laptop: i'm far enough out that my DSL service is only 320kb/sec down |
| 16:36.09 | E-bola | Just so i understand correctly: If you have peers defined for outgoing calls, you can not make them NOT match incoming calls? |
| 16:36.10 | dlynes_laptop | zorton, and in some areas, even a hardware echo canceller helps, but doesn't completely solve the problem |
| 16:36.27 | *** join/#asterisk MaineITGuy (~MaineITGu@216.204.70.166) |
| 16:36.32 | ChannelZ | I've read that it's "going to change" but never that it has |
| 16:36.33 | dlynes_laptop | zorton, you probably want a hwec, then |
| 16:36.39 | zorton | sounds lke it |
| 16:36.52 | zorton | so then the next question is the price tag of the card |
| 16:36.53 | zorton | brb |
| 16:37.04 | p3nguin | e-bola: I guess you have to use type=user for multiple accounts with inbound calls, since those match first on username. |
| 16:37.26 | E-bola | > brb |
| 16:37.26 | E-bola | <p |
| 16:37.30 | E-bola | woops sorry |
| 16:37.38 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.203.235) |
| 16:37.43 | E-bola | p3nguin: but didnt we just conclude i cant call out via type=user ? |
| 16:37.44 | p3nguin | e-bola: If you register, it matches the From username. |
| 16:37.46 | dlynes_laptop | zorton, it might be worth gambling 50 or 60 bucks on a linksys 3102 to see if it provides sufficient quality for you, too |
| 16:37.52 | E-bola | p3nguin: I dont register... |
| 16:38.09 | dlynes_laptop | zorton, it has an fxo port and has onboard echo canceller |
| 16:41.40 | [TK]D-Fender | zorton: What kind of use is this for? Also I never recommend PCI FXS |
| 16:42.44 | HenrikJott | Hi all! i keep getting the errormessage in my * messages-log: WARNING[10221] channel.c: No translator path exists for channel type Agent (native -1) to 0. Any ideas what this can be? |
| 16:43.01 | E-bola | Where can u read how type= works in sip.conf? |
| 16:43.12 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:43.13 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:46.27 | E-bola | hmm i found the solution somewhat |
| 16:46.31 | E-bola | insecure= |
| 16:46.33 | E-bola | woops |
| 16:46.38 | zorton | aiming for 1 FXO port and 3 FXS ports for home use |
| 16:46.41 | E-bola | insecure=port,invite |
| 16:46.54 | [TK]D-Fender | zorton: Do you have call-waiting on that line? |
| 16:47.05 | zorton | yep |
| 16:47.09 | zorton | and DSL |
| 16:47.19 | zorton | right now my router and AP are eatting up a total of 8 watts |
| 16:47.22 | p3nguin | e-bola: Do you know what those do? |
| 16:47.32 | [TK]D-Fender | zorton: Ok, thats tricky. You will want a PCi solution for the FXO, but go with ATA's for FXS if you have to. |
| 16:47.33 | E-bola | [TK]D-Fender: If i have insecure=port,invite and a user and a peer for every account, wont that only allow incoming calls from my provider? |
| 16:47.38 | zorton | which i'm really happy about so I thought it would be nice to integrate PBX functions into it |
| 16:47.49 | E-bola | p3nguin: it allows everything from the ip, without authenticating if im understanding it correctly |
| 16:47.52 | E-bola | ? |
| 16:48.25 | p3nguin | insecure=port allow matching of the peer by IP address without matching the port number |
| 16:48.42 | [TK]D-Fender | zorton: If you don't care about CW then you can go with another solution. |
| 16:49.05 | [TK]D-Fender | zorton: For cheap home use you can forgo the hardware echo canceller and work out the SW ones |
| 16:49.06 | zorton | how much do you figure on the cost of a card that can handle the call waiting |
| 16:49.07 | *** join/#asterisk helloritesh (~hellorite@121.243.172.188) |
| 16:49.10 | E-bola | p3nguin: yes and insecure=invite ; Do not require authentication of incoming INVITEs |
| 16:49.22 | p3nguin | insecure=invite don't require auth ... right. :) |
| 16:49.23 | E-bola | so this solves my problem, without opening me up to the world? |
| 16:49.34 | *** join/#asterisk crazeekennee (~crazeeken@static-72-87-91-10.prvdri.fios.verizon.net) |
| 16:49.43 | E-bola | i read it as: allow everything from the ip defined in host= |
| 16:49.50 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 16:50.08 | mcrownover | good morning |
| 16:51.16 | E-bola | p3nguin: is that ur understanding as well? |
| 16:51.43 | p3nguin | not entirely, no. |
| 16:52.29 | E-bola | mm well how do u understand it then? |
| 16:53.08 | p3nguin | I take it at face value: disregard the port number and don't make the peer AUTH for INVITE |
| 16:53.36 | E-bola | well thats equally fine |
| 16:53.55 | E-bola | since i trust my provider, it still only allows his ip, and it means i can route the calls correctly |
| 16:54.09 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:54.18 | mcrownover | is anyone here familiar with the recording features of asterisk? |
| 16:54.22 | p3nguin | As long as you have specified the host (or created ACLs), it's probably safe. |
| 16:54.44 | p3nguin | mcrownover: Which "recording features" are you talking about? |
| 16:55.24 | mcrownover | well it might be a freepbx feature - but setting up an extension to record on demand or all the time |
| 16:55.25 | E-bola | p3nguin: yep all my entries specify the host |
| 16:57.39 | [TK]D-Fender | mcrownover: Correct... it is a FreePBX feature.... |
| 16:57.42 | [TK]D-Fender | ~freepbx |
| 16:57.43 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 16:57.44 | [TK]D-Fender | ^^^^^^^ |
| 16:57.57 | mcrownover | actually it works fine - but i am trying to figure out why it seperates the audio from the recording - ie one parties entire stream is played first then the other party next |
| 16:58.13 | mcrownover | cool - i will ask over in freepbx |
| 16:58.22 | mcrownover | didnt mean to offend if i did |
| 16:58.48 | raden_work | am i correct in presuming tar does not follow symbolic links ? |
| 16:59.32 | p3nguin | raden_work: It can. |
| 17:00.04 | raden_work | but by default ? |
| 17:01.34 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 17:01.39 | p3nguin | raden_work: Do you want the symlink to be included, or do you want the file the symlink points to? |
| 17:01.56 | raden_work | the files the sym link point to |
| 17:02.06 | p3nguin | Try the -h option in your create command. |
| 17:02.14 | raden_work | thanks |
| 17:02.19 | raden_work | was just readin man page :D |
| 17:02.51 | garymc | so here i am, bought the A200 to combine with my A101D and turns out the A200 needs a power supply :( gotta find a way for my HP proliant DL380 G3 server to do this as sangoma say you cant use an external one? |
| 17:02.52 | p3nguin | tar chzvvf archive.tar.gz listoffiles |
| 17:03.21 | garymc | otherwise ive kissed £200 |
| 17:03.27 | garymc | or need a new server? |
| 17:03.28 | *** part/#asterisk crazeekennee (~crazeeken@static-72-87-91-10.prvdri.fios.verizon.net) |
| 17:03.53 | Chainsaw | garymc: Have you talked to HP? |
| 17:04.23 | Chainsaw | garymc: They might be able to offer a suitable cable. (I know the server type, there are no molex plugs available by default) |
| 17:04.32 | garymc | yes, they say they dont support sangoma or digium or anyone other than HP |
| 17:04.40 | garymc | I told them it was a HP issue the say no |
| 17:04.42 | Chainsaw | garymc: Cute. |
| 17:04.42 | MaineITGuy | Greetings. The boss has asked me to look at Asterisk. But, in order to even consider it, we need to be able to connect to a ShoreTel system at another office, and use 4 digit extensions to reach anyone in either system. Is this possible? |
| 17:05.01 | Chainsaw | MaineITGuy: Details for the "ShoreTel" system are required. |
| 17:05.11 | Chainsaw | MaineITGuy: How do you plan on connecting to it? An IP interface? What protocol? |
| 17:05.17 | coppice | garymc: you have a server with no power plugs at all? |
| 17:05.20 | *** part/#asterisk pietro1 (~pietro@88.149.164.32) |
| 17:05.26 | Chainsaw | coppice: Yes, DL-series don't have embedded molex plugs. |
| 17:05.42 | Chainsaw | coppice: The drives are in 2.5" bays and the DVD is a slimline with a 50-pin plug. |
| 17:05.50 | MaineITGuy | Those details are up in the air, and would be dealt with after finding out if it's even possible. I'll come back with more info about the Shoretel system. |
| 17:06.02 | Chainsaw | MaineITGuy: Can't answer your question without those details, sorry. |
| 17:06.06 | coppice | and no SATA plugs? |
| 17:06.19 | Chainsaw | coppice: No, just SAS. They're embedded in the 2.5" bays. |
| 17:06.54 | Chainsaw | MaineITGuy: If your ShoreTel speaks SIP or standard ISDN signalling, there's hope. |
| 17:06.55 | coppice | providing no power at all is a bit limiting. sounds very 1990's Compaq |
| 17:07.20 | Chainsaw | coppice: It's not limited me before. It's just FXS ports that want it. |
| 17:07.47 | coppice | a lot of specialist embedded stuff needs a power plug of some kind |
| 17:08.14 | coppice | just like the graphics cards do |
| 17:09.12 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 17:10.22 | Chainsaw | garymc: Are you in warranty? You can probably lift the voltages you need from the 24-pin ATX power plug. |
| 17:11.52 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:12.33 | *** part/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 17:16.23 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:20.03 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 17:29.47 | [TK]D-Fender | garymc: Make your own cable |
| 17:30.19 | coppice | you can buy cables to break out the power quite cheaply |
| 17:30.59 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:32.05 | Chainsaw | coppice: Is there some 24-pin sandwich connector that could sit between mainboard & ATX plug? |
| 17:32.30 | *** join/#asterisk albertoandrade (~albertoan@189.58.28.177.dynamic.adsl.gvt.net.br) |
| 17:32.47 | coppice | you can get short 24 pin plug to 24 pin socket leads which break out power |
| 17:40.05 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:40.46 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:41.47 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 17:44.15 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:44.16 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 17:45.32 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 17:46.36 | *** join/#asterisk Yedidya (~chatzilla@host86-142-22-34.range86-142.btcentralplus.com) |
| 17:50.18 | angryuser | I have a zap channel stuck in reserved state, OffHook http://www.pastebin.ca/1775924 i have reset/rebooted, but after first call its stucks, any ideas ? |
| 17:50.27 | angryuser | its a sangoma fxs port |
| 17:51.00 | angryuser | Tryed kewlstart and loopstart also |
| 17:51.44 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 17:52.20 | *** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:52.29 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 18:01.52 | *** join/#asterisk Victor_Yure_ (~victor@unaffiliated/victoryure/x-837844) |
| 18:03.46 | *** join/#asterisk coppice (~chatzilla@172.169.232.220.dyn.pacific.net.hk) |
| 18:05.35 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:08.16 | *** join/#asterisk ktwilight (~keliew@24.0-241-81.adsl-dyn.isp.belgacom.be) |
| 18:08.54 | *** join/#asterisk imcdona (~t@173.160.189.65) |
| 18:11.37 | HenrikJott | * keeps telling me: pbx.c:1564 ast_func_write: Function CDR not registered. I have recently slimmed down asterisk but i guess i need something more to make cdr work. Any ideas? |
| 18:18.25 | *** join/#asterisk nickakavic (~sat4240@cnsa328-pc3.soft.ad.mtu.edu) |
| 18:19.03 | stones_ | ok a2billing sucks. heh |
| 18:19.19 | *** join/#asterisk imcdona (~t@173.160.189.65) |
| 18:19.19 | [TK]D-Fender | stones_: How so? |
| 18:19.37 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 18:20.19 | stones_ | I cant even add a card. |
| 18:20.40 | stones_ | TK, right now im trying to atleast get incoming calls to work |
| 18:20.47 | stones_ | completely new, but ising the gui in a2billng |
| 18:20.56 | stones_ | do you mind helping me get a start |
| 18:21.06 | stones_ | Ill give you a server or something, I run a datacenter |
| 18:21.11 | stones_ | but know nothing about voip :) |
| 18:21.15 | [TK]D-Fender | stones_: No need. |
| 18:21.50 | nickakavic | dahdi says no card? |
| 18:21.53 | [TK]D-Fender | stones_: What GUI? |
| 18:22.24 | stones_ | Well a2billings Web interface |
| 18:22.35 | stones_ | im adding stuff like the call rates etc |
| 18:22.43 | stones_ | should I see the a2billing conf files populate? |
| 18:22.47 | stones_ | because they are emptying |
| 18:22.52 | stones_ | wehn I try t call the number I get: |
| 18:23.52 | ariel_ | angryuser: your stuck port, have you diconnected the wire and restarted the system, after that does it still show reserved? |
| 18:24.00 | stones_ | [Feb 2 11:53:03] NOTICE[31345]: chan_sip.c:19098 handle_request_invite: Call from '' to extension '17182893151' rejected because extension not found. |
| 18:24.04 | stones_ | crap |
| 18:24.07 | stones_ | im sorry |
| 18:24.15 | stones_ | that was an accident. |
| 18:24.29 | angryuser | ariel_, yes |
| 18:24.38 | angryuser | ariel_, sorry no xD |
| 18:24.56 | angryuser | but after first call it get reserved |
| 18:25.24 | paulc | stones_ what contexts do you have in extensions.conf, and what's your general "context=" line from sip.conf? |
| 18:25.34 | nickakavic | stones_: that means you have no dialplan set up probably |
| 18:26.25 | ariel_ | angryuser: have you switched your tip and ring around? |
| 18:26.31 | stones_ | im using a2billing on top of asterisk |
| 18:26.43 | nickakavic | stones_: you should ditch the a2billing and just get a handle on asterisk first |
| 18:26.47 | [TK]D-Fender | stones_: still jsut a dialplan issue.. has nothing to do with biling processing. |
| 18:27.03 | nickakavic | there are a ton of great beginners tutorials online |
| 18:27.31 | stones_ | doesnt a2billing create everything for me? |
| 18:27.42 | *** part/#asterisk possy (~nnnpossy@smtp.theinternet.de) |
| 18:27.48 | nickakavic | apparently not |
| 18:27.55 | nickakavic | http://www.asteriskguru.com/ |
| 18:28.07 | nickakavic | good links there |
| 18:28.13 | [TK]D-Fender | stones_: Doesn't create ANY of it for you. |
| 18:28.24 | [TK]D-Fender | stones_: All it has is a single AGI for you to call. |
| 18:28.43 | stones_ | Ok so what are the a2billing additional confs for? |
| 18:28.46 | [TK]D-Fender | stones_: It isn't a complete PBX builder. |
| 18:29.00 | [TK]D-Fender | stones_: for the bits of your system involved with calling out on account |
| 18:29.05 | hackeron | hey, I'm seeing this in syslog: asterisk[2923]: rc_avpair_new: unknown attribute 1490026597 -- what does it mean? |
| 18:29.11 | *** join/#asterisk imcdona (~t@173.160.189.65) |
| 18:29.24 | stones_ | could you tell me what id put in the sip.conf pls |
| 18:30.10 | angryuser | ariel_, nope, you think its the fax with pins inverted ? |
| 18:30.21 | QubeZ | the qualify=ms|yes|no (milliseconds is only to determine when the client side is dead right in asterisk 1.4?). I cannot change this frequency? |
| 18:30.25 | angryuser | (its fax behind) |
| 18:30.30 | stones_ | i just created customers |
| 18:30.36 | stones_ | and i see the info in a2billing's sip.conf |
| 18:30.36 | QubeZ | i wanted to change the frequency at which the SIP OPTIONS messages are sent |
| 18:30.53 | Katty | ohhh so full |
| 18:31.00 | Katty | sprawls |
| 18:31.08 | Katty | wtb lounge pants |
| 18:31.09 | Katty | and pillow |
| 18:31.10 | jaytee | me too! |
| 18:31.16 | nickakavic | stones_: it would be very beneficial to uninstall that a2billing and start with asterisk from scratch. it's not as simple as it looks. |
| 18:31.32 | stones_ | Ok. |
| 18:31.35 | voipmonk | :) |
| 18:31.36 | jaytee | went to TGI Fridays and had prime rib stroganoff and some tostada nachos |
| 18:31.39 | stones_ | so |
| 18:31.43 | Katty | oh man |
| 18:31.46 | Katty | that sounds delicious |
| 18:31.49 | jaytee | it was |
| 18:31.53 | stones_ | i mean, do i just simply add something to the conf or what? |
| 18:31.53 | nickakavic | find a good tutorial online |
| 18:31.54 | Katty | i had nachos at a local mexican place |
| 18:31.55 | stones_ | what am i missing? |
| 18:31.58 | stones_ | i looked |
| 18:32.09 | jaytee | now I want to put my head down on the desk and take a nap before fingerpainting |
| 18:32.10 | stones_ | i cant find anything with a2billing and asterisk on configuring |
| 18:32.15 | jaytee | but this isn't kindergarten |
| 18:32.18 | stones_ | setup yes, but not step by step configuring |
| 18:32.24 | Katty | sure it is |
| 18:32.28 | Katty | you just gotta make believe |
| 18:32.28 | nickakavic | hold on |
| 18:32.30 | nickakavic | ill find you one |
| 18:33.00 | jaytee | those were difficult years for me. i was one of the "weird" kids that ate the paste |
| 18:33.10 | Katty | ahhh |
| 18:33.20 | Katty | kintergarden was a bit scary for me too |
| 18:33.23 | Katty | there were screaming monsters |
| 18:33.28 | Katty | who fought over things |
| 18:33.29 | jaytee | junior high was scarier |
| 18:33.32 | Katty | and threw things when they didn't get their way |
| 18:33.40 | nickakavic | http://www.linuxformat.co.uk/wiki/index.php/Asterisk_tutorial |
| 18:33.54 | ariel_ | angryuser: inverted pins on fax? It just might be the wire you have connected between them |
| 18:34.07 | nickakavic | thats a good one, but you'll want to google more about sip and extensions conf files to really get what you're doing |
| 18:34.14 | nickakavic | i gotta go stones, good luck |
| 18:34.21 | stones_ | blah |
| 18:34.22 | jaytee | I'd grown into a 98 pound weakling with an armload of books and seemed to draw bullies like a black hole sucking in matter around it. |
| 18:34.24 | stones_ | whats a good book |
| 18:34.30 | stones_ | i can pick up from barns and nobles guys? |
| 18:34.48 | jaytee | Asterisk: The Future of Telephony |
| 18:34.58 | jaytee | if you like print copies |
| 18:35.00 | stones_ | Ok, I will read. |
| 18:35.04 | stones_ | :D |
| 18:35.05 | jaytee | ~book |
| 18:35.06 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:35.18 | stones_ | Im going to go drive over there now |
| 18:35.26 | stones_ | So pretty much, you have to master asterisk |
| 18:35.30 | jaytee | you can download the pdf for free but I bought two print copies as it makes reading easier |
| 18:35.31 | stones_ | before using a2billing? |
| 18:35.52 | stones_ | I thought A2billing was like the cPanel of Asterisk |
| 18:35.59 | stones_ | where it just maeks life easier, heh |
| 18:36.04 | Katty | heh |
| 18:36.17 | Katty | since when do you get a bowl of chocolate covered strawberries wihtout paying a price? |
| 18:36.39 | stones_ | lol |
| 18:36.41 | voipmonk | no its not the cpanel of asterisk |
| 18:37.02 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:37.05 | stones_ | well, im going to get my ass up and go buy the book |
| 18:37.11 | voipmonk | its a body kit and maybe a nice exhaust set |
| 18:37.15 | stones_ | im good with linux and ccna certified |
| 18:37.20 | stones_ | so i dont think im that far off :D |
| 18:37.23 | voipmonk | but u still need to the car and know how to drive |
| 18:37.24 | voipmonk | :) |
| 18:37.27 | jaytee | then you should be able to ramp up quickly |
| 18:37.31 | Katty | a Body Kit(tm)? |
| 18:37.39 | Katty | i assume that has something to do with vehicles. |
| 18:37.48 | stones_ | yes. |
| 18:37.50 | Katty | k |
| 18:37.53 | stones_ | ;) |
| 18:37.58 | stones_ | Oh btw |
| 18:38.02 | stones_ | VOIP for dummies |
| 18:38.05 | stones_ | I saw that there as well |
| 18:38.08 | stones_ | worth it or no? |
| 18:38.16 | [TK]D-Fender | stones_: Just read THE BOOK |
| 18:38.18 | [TK]D-Fender | ~book |
| 18:38.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:38.32 | [TK]D-Fender | stones_: [13:35]<stones_>I thought A2billing was like the cPanel of Asterisk <- and NO, it isn't |
| 18:38.37 | jaytee | stones_, might look for SIP Demystified. a good book but B&N probably doesn't stock it. I had to get mine from Amazon |
| 18:38.43 | Katty | fender would you like half a tummy ache |
| 18:38.52 | stones_ | Ok. |
| 18:38.53 | [TK]D-Fender | stones_: It is 1 tiny script for your * config to call. |
| 18:38.54 | stones_ | Ill go check |
| 18:39.10 | stones_ | tk gotcha |
| 18:39.15 | stones_ | it all fucking makes sense now |
| 18:39.15 | [TK]D-Fender | stones_: the rest of its PIN management, reporting, etc has nothing to do with * |
| 18:39.26 | stones_ | i can see clearly now |
| 18:39.28 | stones_ | the rain is gone. |
| 18:39.46 | [TK]D-Fender | collects the last of it in a bucket and holds stones_ 's head under |
| 18:39.49 | Katty | can't we use more colorful words. |
| 18:39.51 | jaytee | bright, bright, bright, sunshiny day |
| 18:39.55 | stones_ | haha |
| 18:40.00 | seanbright | hmm? |
| 18:40.01 | Katty | like snickerdoodles |
| 18:40.04 | Katty | i am such a fan of that word. |
| 18:40.06 | Katty | also seanbright |
| 18:40.08 | Katty | also a fan of him |
| 18:40.19 | seanbright | word. |
| 18:40.22 | jaytee | snickerdoodles is a good word but has few appropriate uses |
| 18:40.39 | Katty | you can't inappropriately use snickerdoodles |
| 18:40.44 | jaytee | yes you can |
| 18:40.50 | Katty | shut the snickerdoodles up |
| 18:40.54 | Katty | see, not inappropriate |
| 18:40.59 | jaytee | I stand corrected :-) |
| 18:41.00 | *** join/#asterisk Skeeter- (Skeeter@190-141.cgocable.ca) |
| 18:41.02 | Katty | get the snickerdoodles out of my way |
| 18:41.06 | jaytee | actually, I think I'll sit |
| 18:41.10 | Katty | what the snickerdoodles are you doing |
| 18:41.29 | jaytee | that's way better than frak or frell |
| 18:41.34 | [TK]D-Fender | LIES |
| 18:41.37 | jaytee | gonna have to start using that |
| 18:41.46 | Katty | lies, my snickerdoodles! |
| 18:42.35 | *** join/#asterisk cweagans (~cweagans@66-193-105-126.static.twtelecom.net) |
| 18:42.50 | [TK]D-Fender | Katty: It's so obnoxiously a "cutesy" word that it incites projectile vomit at the thought of its ever increasing utterance. |
| 18:42.51 | Katty | jaytee: what's your favorite dessert |
| 18:43.12 | [TK]D-Fender | wretches some more... |
| 18:43.13 | Katty | [TK]D-Fender: well, snickerdoodles to you too, sir. |
| 18:43.27 | Katty | ;P |
| 18:43.32 | beek | guesses snickerdoodles is jaytee's favorite dessert. |
| 18:43.36 | beek | Hello gang. |
| 18:43.40 | Katty | yay beek |
| 18:43.42 | Katty | hugs on beek |
| 18:43.47 | cweagans | My boss just ordered a set of 5 Cisco 7940 phones and is expecting me to come up with a way to make them work. Is Asterisk a good choice for this? If so, is there a guide that I can read or something? We don't have any hardware other than the phones and cat5. I'm literally starting from scratch |
| 18:44.00 | jaytee | warmed razzleberry pie with vanilla ice cream and a dash of Saigon cinnamon on the ice cream |
| 18:44.09 | Nugget | cweagans: is it too late to cancel the order? |
| 18:44.23 | Katty | cweagans: i'm sure your boss had something in mind when he bought the phones. |
| 18:44.23 | [TK]D-Fender | cweagans: that's like buying specialized tires and then wondering what car to use them on... |
| 18:44.26 | Nugget | Cisco phones are a nightmare of pain, horror, and hemorrhoids. |
| 18:44.26 | Katty | cweagans: what does he want |
| 18:44.37 | Nugget | and the 7940 is the second worst cisco phone you can use. |
| 18:44.41 | Katty | jaytee: is there a name this particular dish has? |
| 18:44.42 | cweagans | lol, yes, too late to cancel the order. The phones are being delivered tomorrow |
| 18:44.43 | jaytee | the last part being the main component |
| 18:44.51 | voipmonk | it wont be all that bad |
| 18:44.52 | [TK]D-Fender | Nugget: No... there are several worst ones.. |
| 18:44.54 | voipmonk | you will learn a lot |
| 18:44.55 | Katty | jaytee: or just razzleberry pie al mode? |
| 18:45.01 | [TK]D-Fender | worse* |
| 18:45.12 | Katty | cweagans: so what did your boss have in mind when he ordered the phones |
| 18:45.13 | [TK]D-Fender | Katty: "a la" |
| 18:45.16 | Nugget | Step one: you'll need to buy a service contract from cisco for the phones in order to get the firmware. |
| 18:45.20 | cweagans | so, basically, we have a phone number that's going to our new office. I guess we just want extensions for each person |
| 18:45.23 | jaytee | Katty, yes it's called razzleberry pie with vanilla ice cream but the a la mode would work too |
| 18:45.28 | QubeZ | jaytee im still having issues even after removing the Answer section. Not sure what to do next. It says "No Fax Reply" on the other real fax machine I'm trying to fax from to the ATA. |
| 18:45.33 | Katty | jaytee: hmm |
| 18:45.42 | Katty | jaytee: i think i'd just stick with Razzleberries on this one then |
| 18:46.08 | Katty | "who the razzleberries cares" |
| 18:46.19 | QubeZ | is there any way to change the frequency of the qualify OPTIONS command in Asterisk 1.4? |
| 18:46.27 | Katty | who the razzleberry cares |
| 18:46.30 | jaytee | QubeZ, don't know what to tell ya then. Have you tried another model of fax machine on the other end? some fax machines just plain don't like t38 passthrough to a fax on an ATA |
| 18:46.38 | [TK]D-Fender | QubeZ: vi chan_sip.c |
| 18:46.50 | Nugget | cweagans: http://www.voip-info.org/wiki/view/Cisco |
| 18:46.57 | jaytee | Katty, Marie Calendar's in the frozen dessert section. |
| 18:47.08 | Katty | googles |
| 18:47.27 | coppice | QubeZ: the T.38 support in many ATAs is extremely buggy. which ATA are you using |
| 18:47.29 | Nugget | cweagans: the phones won't be functional for you until you've got the service contract from cisco and downloaded the firmware. that should take a couple weeks minimum to resolve |
| 18:47.32 | QubeZ | [TK]D-Fender so edit source code, recompile... argh :) |
| 18:47.42 | QubeZ | coppice Linksys SPA3102 |
| 18:47.57 | jaytee | they bought out a restaurant bakery where i lived in southern Oregon and branded the pie with their chain name. |
| 18:48.02 | coppice | how long is the lead from the fax machine to the 3102? |
| 18:48.08 | Katty | ohhh wow |
| 18:48.15 | jaytee | place was originally called the Wild Plum |
| 18:48.17 | Katty | this ingredient list is very...very promising |
| 18:48.20 | QubeZ | coppice cable line is typical 6ft phone cable |
| 18:48.31 | jaytee | had two places, Medford, OR where I lived and Redding, CA |
| 18:48.39 | Katty | well cept for the hydrogenated soybean oil |
| 18:48.43 | Katty | that screams transfat |
| 18:48.44 | cweagans | Nugget: there's no way around that? is there no open source firmware or something? |
| 18:48.49 | Nugget | cweagans: correct |
| 18:48.56 | cweagans | that freakin' blows. |
| 18:48.58 | cweagans | =P |
| 18:49.01 | jaytee | once it changed hands from the family business the place kinda went downhill but the frozen pies are still good |
| 18:49.02 | Katty | jaytee: do you think it's more raspberry or blackberry tasting? |
| 18:49.02 | coppice | QubeZ: do you have a much longer lead you can try? I don't know what happens, but the 3102 is *really* quirky with short cables |
| 18:49.04 | Nugget | yes it does. is it too late to return the phones? |
| 18:49.04 | Katty | jaytee: sweet or tart |
| 18:49.10 | QubeZ | coppice i can try |
| 18:49.14 | QubeZ | but outbound works |
| 18:49.16 | jaytee | Katty, tart with more raspberry |
| 18:49.23 | Katty | wouldn't like it then :< |
| 18:49.29 | Katty | bummer. |
| 18:49.34 | Katty | can you recommend any other flavor? |
| 18:49.36 | coppice | the LinkSys T.38 is not amongst the best |
| 18:49.37 | jaytee | and the original was raspberry, blackberry and boysenberry |
| 18:50.02 | Katty | checks to see if local schnucks carries it |
| 18:50.09 | jaytee | katty, it's not REAL tart, just a hint but still sweet enough to please most folks |
| 18:50.29 | cweagans | Nugget: hm. I'm not sure. Is there a better set of phones that you'd recommend? |
| 18:50.39 | cweagans | (preferably not super expensive) |
| 18:50.42 | Katty | jaytee: what's the sweetest one you've had? |
| 18:50.45 | Nugget | cweagans: polycom |
| 18:51.03 | Nugget | I have an office full of cisco phones and I enthusiastically endorse polycoms :) |
| 18:51.03 | Naikrovek | i second that |
| 18:51.03 | Katty | jaytee: ah HA! local schnucks carries it |
| 18:51.17 | Katty | jaytee: which one shall i add to the grocery list? |
| 18:51.41 | cweagans | Nugget, Nailkrovek: Any particular model? |
| 18:52.06 | Katty | mmm chocolate satin pie |
| 18:52.10 | Katty | pamples things |
| 18:53.19 | *** join/#asterisk fofware (~chatzilla@186.124.184.65) |
| 18:54.40 | *** join/#asterisk comradeb14ck (~comradeb1@72.37.252.50) |
| 18:54.45 | Naikrovek | cweagans: IP335, maybe IP450 |
| 18:54.51 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-042-222-050.hsi3.kabel-badenwuerttemberg.de) |
| 18:55.21 | Naikrovek | cweagans: i bought a slew of ip321s and they're great too |
| 18:55.32 | Naikrovek | depends on your budget, but all phones they make are good |
| 18:57.35 | Katty | ! |
| 18:57.37 | Katty | it's Naikrovek!!! |
| 18:57.43 | Katty | who hasn't been here for THREE WHOLE DAYS |
| 18:57.45 | Katty | hugs Naikrovek |
| 18:57.50 | Naikrovek | where! |
| 18:57.55 | Naikrovek | i want an autograph! |
| 18:57.56 | Naikrovek | oh |
| 18:58.49 | *** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net) |
| 18:59.04 | Qwell | Naikrovek: pfft, nobody wants your autograph |
| 18:59.18 | Qwell | notes he has 8x10" glossies for sale in the back |
| 18:59.31 | Naikrovek | well i sent checks once that weren't signed and i got complaints |
| 18:59.36 | rnp | Who here does software and development work with asterisk and custom CRM's and can walk a newb through the process of getting everything setup. Will pay |
| 18:59.55 | Naikrovek | rnp: ask voipmonk |
| 19:00.03 | Naikrovek | rnp: tell him i sent you if you do |
| 19:00.26 | Katty | not me. |
| 19:00.28 | Katty | but! |
| 19:00.34 | Katty | if you need a custom bird seed mix |
| 19:00.36 | Katty | i'm your girl |
| 19:00.40 | Katty | i mean |
| 19:00.44 | Katty | 40 year old guy in the basement |
| 19:00.51 | *** join/#asterisk nny (~scott@64.203.239.83) |
| 19:01.05 | Nugget | heh |
| 19:01.06 | nny | so.. NI1 or NI2 for PRIs? |
| 19:01.09 | Nugget | huggles Katty |
| 19:01.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:01.14 | nny | iirc NI2 but want to confirm |
| 19:01.23 | Qwell | Katty: I don't live in my parents basement. They live in *MY* attic! |
| 19:01.35 | Katty | hi Nugget |
| 19:01.35 | Kobaz | hah |
| 19:01.37 | Katty | hugs Nugget |
| 19:01.44 | Naikrovek | Qwell: yeah. at some point you get to say that you don't live with them, you get to say you take care of them |
| 19:01.46 | Katty | Nugget: telnet |
| 19:01.46 | Nugget | telnet is eeeeeeevil! |
| 19:01.48 | Katty | FIRST |
| 19:01.51 | Katty | :>>>>>> |
| 19:01.52 | Naikrovek | lol! |
| 19:02.15 | Katty | black: hai der |
| 19:02.21 | black | yo |
| 19:02.24 | black | sup ^^ |
| 19:02.26 | Naikrovek | Qwell: at that point, one goes from highest on loser scale to highest on humanitarian scale |
| 19:02.31 | black | my last week at fonality |
| 19:02.34 | black | feeling a bit sad |
| 19:02.35 | Naikrovek | really |
| 19:02.41 | Qwell | black: last week? |
| 19:02.43 | black | yea |
| 19:02.49 | Qwell | details |
| 19:02.57 | Naikrovek | yes |
| 19:02.57 | black | Got a new job. |
| 19:02.58 | Naikrovek | deets |
| 19:03.01 | Naikrovek | where |
| 19:03.07 | black | And my side business is making a lot of money now too. |
| 19:03.14 | black | RCG communications |
| 19:03.24 | Nugget | hah |
| 19:03.26 | Naikrovek | never heard of them. are they imaginary |
| 19:03.30 | black | nope |
| 19:03.35 | Naikrovek | awesome |
| 19:03.43 | black | They do free telephony services. |
| 19:03.52 | Naikrovek | free, eh |
| 19:03.54 | black | yea |
| 19:04.04 | Katty | i can haz free phone? |
| 19:04.09 | Qwell | Why you leaving? |
| 19:04.19 | black | Well this new place is paying me like 5x what I make @ fonality. |
| 19:04.24 | Naikrovek | really |
| 19:04.26 | black | And I can also work from home 100% of the time. |
| 19:04.27 | Naikrovek | hrm. |
| 19:04.29 | Naikrovek | 5x is huge |
| 19:05.00 | Katty | where is fonality based out of? |
| 19:05.07 | black | Los Angeles, CA |
| 19:05.08 | Qwell | Katty: L.A. |
| 19:05.11 | Katty | ahh |
| 19:05.14 | black | I live here, but traffic is horrible. |
| 19:05.14 | [TK]D-Fender | Katty: Hell, 7th circle |
| 19:05.27 | Qwell | black: sounds like a scam to me, but hey. gl :p |
| 19:05.30 | [TK]D-Fender | channels Dante |
| 19:05.30 | Katty | [TK]D-Fender: you are just full of it today |
| 19:05.35 | black | Qwell, it isn't, but thanks ^^ |
| 19:05.46 | Katty | black: are you gonna have to move? |
| 19:05.48 | black | Nope.l |
| 19:06.01 | Naikrovek | telecommuting is nice |
| 19:06.03 | black | I'm getting a new place though. |
| 19:06.06 | Naikrovek | i did that at my last job, 100% of the time |
| 19:06.10 | Naikrovek | was saweet |
| 19:06.12 | Katty | black: oh? |
| 19:06.23 | black | Yea, been doing apartment shopping th epast few weeks. |
| 19:06.28 | Katty | oh. |
| 19:06.30 | Katty | apartments. |
| 19:06.31 | Katty | blah |
| 19:06.32 | black | word |
| 19:06.39 | black | I have no need for a house :x |
| 19:06.43 | black | Its just me + fiancee =/ |
| 19:06.51 | Katty | that's enough reason to get a house |
| 19:07.02 | black | We both want to move aroudn though =p |
| 19:07.05 | black | Not get stuck in 1 place |
| 19:07.06 | Katty | want a better reason? |
| 19:07.08 | Katty | get a dog |
| 19:07.13 | black | we have a dog :D |
| 19:07.17 | Katty | orly |
| 19:07.18 | Katty | what kind |
| 19:07.23 | black | 2 doberman pinscers |
| 19:07.25 | Katty | ! |
| 19:07.27 | black | and a chihuaha |
| 19:07.28 | Katty | :>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> |
| 19:07.32 | Katty | yay dobies |
| 19:07.40 | black | thats one of my dog's names =-p |
| 19:07.41 | black | doby |
| 19:07.42 | black | and jeny |
| 19:07.43 | black | ^^ |
| 19:07.47 | black | and scribbles is the chihuaha |
| 19:07.50 | Katty | k wait |
| 19:07.51 | Katty | 2 dobermans |
| 19:07.54 | Katty | and no house? |
| 19:08.02 | black | Well atm I live in a house. |
| 19:08.06 | Katty | those poor pupscicles |
| 19:08.16 | black | My parents own a home in LA, but don't live here. |
| 19:08.19 | black | So I live in this house. |
| 19:08.33 | Katty | and you're going to take those 3 babies to an apartment? |
| 19:08.38 | black | But I'm looking for an apartment / townhome to rent around here in the suburbs somewhere. |
| 19:08.44 | black | Well the ones I've been looking at are really big. |
| 19:08.44 | Katty | what a terrible parent :P |
| 19:08.46 | Naikrovek | what's wrong with the house |
| 19:08.58 | black | Well Im' getting married on april 3, and dno't want to live in my parents house. |
| 19:09.00 | black | Would be kinda lame. |
| 19:09.01 | Naikrovek | my mothers fam is from monroe, la |
| 19:09.07 | Naikrovek | nto if they don't live in it |
| 19:09.11 | black | Ya but still =/ |
| 19:09.14 | Katty | pay them rent! |
| 19:09.18 | black | Na. |
| 19:09.24 | Katty | THINK OF THE CHILDREN |
| 19:09.26 | black | Dude the places I'm looking at are really nice. |
| 19:09.30 | black | They have yards and stuff |
| 19:09.34 | Katty | k |
| 19:09.38 | Katty | yard is an acceptable alternative |
| 19:09.43 | black | This past weekend I was out in Palmdale lokoing at these townhomes. |
| 19:09.52 | Katty | well see a townhouse isn't really an apartment |
| 19:09.55 | black | They are 2 levels (upstairs and downstairs), 3 bedrooms, 2 baths, and a yard. |
| 19:10.06 | black | All for like $1,200 a month |
| 19:10.12 | black | They were really nice. |
| 19:10.13 | Katty | not too shabby |
| 19:10.19 | black | Oh and garage too |
| 19:10.27 | Katty | we looked at townhouses for awhile |
| 19:10.31 | black | But its nice because palmdale is really uot of the way |
| 19:10.35 | Katty | but i didn't wanna get one |
| 19:10.35 | black | It is a lot more open. |
| 19:10.39 | black | Not like downtown LA |
| 19:10.40 | Katty | cause it was a newish area |
| 19:10.40 | black | =D |
| 19:10.41 | Katty | with no trees |
| 19:10.48 | Katty | and no trees, means no wildlife |
| 19:10.52 | Katty | and no wildlife, means...DOOM |
| 19:11.01 | black | palmdale is kinda like the desert |
| 19:11.09 | black | its near la (30 minute drive) |
| 19:11.19 | black | but really open and still developing |
| 19:11.26 | Katty | see i wouldn't wanna move there |
| 19:11.32 | Katty | if there's not HUGE trees...i ain't goin |
| 19:12.00 | black | they have these things called joshua trees |
| 19:12.03 | black | they are weird looking trees |
| 19:12.16 | black | http://images.google.com/images?client=firefox-a&rls=org.mozilla:en-US:official&channel=s&hl=en&source=hp&q=joshua%20trees&um=1&ie=UTF-8&sa=N&tab=wi |
| 19:12.43 | Katty | wow that does look weird. |
| 19:12.51 | black | ya |
| 19:13.03 | black | The other nice thing though, is the place I was looking at has a dogpark. |
| 19:13.14 | black | And when I went to check it out this past saturday there were a lot of owners there with their dogs. |
| 19:13.17 | black | It was a pretty nice place. |
| 19:13.58 | Katty | http://farm3.static.flickr.com/2563/3945377204_801d0112ec_b.jpg <- trees |
| 19:14.22 | black | heh |
| 19:14.48 | Katty | unless a deer is gonna walk into my back yard, i don't wanna move there |
| 19:15.04 | Katty | it's too urban where i am now |
| 19:17.19 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 19:23.15 | *** join/#asterisk voxter (~voxter@s64-180-60-188.bc.hsia.telus.net) |
| 19:23.29 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 19:24.27 | voxter | Has anyone experienced this with 1.6.2.1: I park a call, no problem.. Pick it up, no problem. Once i've picked it up, I can no longer hit "#" to transfer, which i was able to do before parking it the first time. |
| 19:24.57 | Katty | can't say i've had that problem. |
| 19:25.19 | [TK]D-Fender | voxter: When you pick it up are you using some sort of option that says you SHOULD be able to do that? |
| 19:25.39 | [TK]D-Fender | voxter: because only DIAL and QUEUE have those. |
| 19:25.44 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:26.16 | voxter | [TK]D-Fender: previously, I have been able to do that. For example my park button on these phones is a speed dial set to #70# |
| 19:26.23 | voxter | I am normally able to park, pick up, park, pick up, multiple times. |
| 19:27.02 | leifmadsen | voxter: !!! |
| 19:27.27 | leifmadsen | voxter: I think that is an open issue you're experiencing |
| 19:27.40 | voxter | Hmm. Shit! |
| 19:27.43 | leifmadsen | I've seen something like that recently -- there are a few transfer issues open right now |
| 19:28.04 | *** join/#asterisk Igramul (~black@p5497ACC7.dip.t-dialin.net) |
| 19:28.08 | Katty | voxter: you mean snickerdoodles. |
| 19:28.13 | voxter | I was just about to light this up for a new restaurant and this is definitely a show stopper... I'm using the multi parking lots feature in 1.6.2.1 |
| 19:28.14 | Katty | voxter: HintHintHint |
| 19:28.16 | leifmadsen | I think one of them being specifically to do with parking multiple times |
| 19:28.26 | leifmadsen | ya, that sounds very familiar |
| 19:28.38 | [TK]D-Fender | voxter: Feel free to bypass that by calling ParkedCall() via a local channel w/ opts |
| 19:28.56 | *** join/#asterisk oej_ (~olle@ns.webway.se) |
| 19:29.56 | voxter | [TK]D-Fender: ie Dial(Local/newexten@context,Tt) and newexten = Parkedcall(x) ? |
| 19:30.36 | Katty | hugs eppigy |
| 19:31.20 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 19:31.33 | Katty | hi puzzled |
| 19:31.36 | leifmadsen | voxter: i can't find an issue that exactly says what you're experiencing, but some close ones |
| 19:31.46 | leifmadsen | 16701 perhaps |
| 19:31.50 | *** join/#asterisk Circlefusion (~circlefus@96-28-114-10.dhcp.insightbb.com) |
| 19:32.11 | voxter | Hmm. That local channel for parkedcall() that i just tried didnt help anything. |
| 19:32.26 | voxter | oh wait, wrong context. |
| 19:32.27 | leifmadsen | or 16657, 16563, 16630 |
| 19:32.46 | Katty | why |
| 19:32.48 | Katty | mister madsen |
| 19:32.59 | Katty | i dont' think we've hugged yet! |
| 19:33.02 | Katty | hugs leifmadsen |
| 19:33.41 | puzzled | hi Katty |
| 19:33.57 | leifmadsen | Katty: I don't think so either! |
| 19:34.01 | leifmadsen | hugs Katty |
| 19:36.20 | *** join/#asterisk praet (~praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 19:38.40 | *** join/#asterisk sw (~sw@unaffiliated/sw) |
| 19:39.19 | sw | hi.. is it possible to set the output of a linux command as value of an asterisk variable ? |
| 19:40.07 | Katty | hmm. well i've piped variables TO linux with System() |
| 19:40.12 | *** join/#asterisk fofware (~chatzilla@186.124.184.65) |
| 19:40.17 | Qwell | func_shell in 1.6+ |
| 19:40.30 | sw | i'm on 1.4 :( |
| 19:40.37 | Qwell | then nope |
| 19:41.00 | Katty | what about a call file |
| 19:41.07 | Katty | a call file is just text |
| 19:41.17 | Katty | you might could write the contents of a call file |
| 19:41.21 | Katty | and then execute the call file |
| 19:41.33 | black | shameless plug: if you are using callfiles, check out my website: http://pycall.org/ |
| 19:42.16 | sw | well actually i'm trying to implement a wake up application under 1.4 |
| 19:42.30 | Katty | well you could just execute a call file with cron |
| 19:42.37 | kaldemar | sw: app ReadFile might do the job |
| 19:43.21 | kaldemar | sw: in case it's acceptable to write the command output to a file and then read it from dialplan. |
| 19:43.32 | black | sw, use call files |
| 19:43.32 | p3nguin | I use wakeup.php and it works pretty alright. |
| 19:43.41 | black | sw, you can ues the SetVar callfile directive |
| 19:43.47 | black | to pass shell info to the asterisk dialplan |
| 19:43.55 | black | and you can have it ran at a pre-determined time |
| 19:43.55 | p3nguin | exten => *55,1,AGI(wakeup.php) |
| 19:44.09 | sw | I already have something working with call files... the challenge is I want to call WAKE-HHMM to set the alarm |
| 19:44.29 | Katty | do what? |
| 19:44.33 | sw | the challenge for this system is to figure out when it's the last day of the month |
| 19:44.53 | Katty | 12 ifthen statements |
| 19:44.54 | black | sw, with callfiles you can set the access time of the file |
| 19:44.57 | Katty | if nothing else |
| 19:44.57 | black | to whatever date / time you want |
| 19:45.07 | black | like here's the high-level logic |
| 19:45.14 | Katty | or you could just set up 12 individual cronjobs |
| 19:45.31 | black | if (user wants wakeup call) { set_callfile_access_time = (what_time()?) } |
| 19:45.33 | Katty | whatever's easier for you |
| 19:45.53 | Katty | Read() |
| 19:46.06 | Katty | System(echo a bunch of stuff blahblahblah ${VARIABLE} blahblahblahblah) |
| 19:46.08 | voxter | This is very bizarre.. If a call is NOT parked, and i dial 71, it uses the dial plan entry ive created to use a local channel to call ParkedCall() |
| 19:46.18 | voxter | if a call IS parked, it skips my dialplan and executes ParkedCall on its own |
| 19:46.30 | Katty | cronjob copy foo.txt from here to there |
| 19:46.49 | Katty | System(variables and stuff | foo.txt) |
| 19:46.53 | Katty | etc. |
| 19:47.16 | Katty | there's probably an easier way |
| 19:47.33 | sw | i'll have a look at wakeup.php first.. this seems to do what I want the app to do |
| 19:47.54 | Katty | if you're good at php it'll be easy breezy beautiful |
| 19:48.20 | Katty | hmm. both my video streams went offline )= |
| 19:50.36 | p3nguin | voxter: Don't include => parkedcalls or use a different exten for ParkedCall(). |
| 19:52.16 | nny | stupid question, I am trying to theory craft a queue system for a new client. Right now they just manually pick who gets the calls by sacrificing a chicken while playing the kazoo. |
| 19:52.33 | nny | Idea is Q1 + Q2. if everyone in Q1 is busy (on phone) than ring 1 in Q2 |
| 19:52.47 | nny | and when a call comes in it rings one member of Q1 first |
| 19:52.49 | nny | rather than all |
| 19:52.55 | p3nguin | Cascading queues, penalties, rrmemory for the strategy. |
| 19:52.56 | black | alright |
| 19:52.59 | black | now i really am going to lunch |
| 19:53.00 | black | woot |
| 19:53.04 | black | be back in a bit |
| 19:53.26 | nny | p3nguin: i am worried about people getting up though and feel log in/out would be too hard for them |
| 19:53.33 | nny | p3nguin: would DND send it down the line? |
| 19:53.35 | p3nguin | Or did you want it to ring everyone? |
| 19:53.40 | nny | (ring next person in q) |
| 19:54.28 | nny | i'd rather it ring one person, and if they don't answer in 8 seconds or so escalate. I can see how that would work, but thinking if DND would allow them to take themselves out of queue easily |
| 19:54.42 | nny | i can probably write it into the dial context |
| 19:54.45 | Katty | DND will help |
| 19:54.57 | p3nguin | if you have a DND, the queue should see that phone as unavailable. If the phone is unavailable, it will try another phone in the queue if the strategy is a rotating one, or it would ring all the others if you have a ringall strategy. |
| 19:54.58 | Katty | what your wanting seems fairly straight forward |
| 19:55.02 | Katty | we have something similiar here |
| 19:55.10 | Katty | but we have about 5 people in each Ring Group |
| 19:55.11 | nny | oh cool so I don't need to make that magic, awesome |
| 19:55.30 | Katty | you could check the status of the phone |
| 19:55.31 | nny | yeah Katty that' about what I am looking at, 5 people that always answer the phone, 6 or so that answer wheni t's extra busy |
| 19:55.36 | Katty | so it wouldn't even ring, if they are on the phone |
| 19:55.43 | nny | yeah that's the other goal |
| 19:55.44 | p3nguin | You can use penalties for escalation to another level within the same queue. |
| 19:55.48 | leifmadsen | nny: there is a new patch that might do what you want with penalties |
| 19:55.50 | nny | if they are on a call then don't ring them |
| 19:56.02 | nny | leifmadsen: 1.6 need the patch? |
| 19:56.05 | nny | needs* |
| 19:56.08 | p3nguin | nny: ringinuse=no |
| 19:56.15 | nny | ahh |
| 19:56.17 | nny | very cool |
| 19:56.20 | Katty | (= |
| 19:56.25 | leifmadsen | nny: it's a new feature -- not ringing the phones when they are on the phone is something entirely different |
| 19:56.29 | nny | first time seriously playing with queues, seems pretty well thought out |
| 19:56.39 | Katty | leifmadsen: that does seem very handy |
| 19:56.42 | nny | leifmadsen: you mean penalties for not answering etc? |
| 19:56.49 | nny | leifmadsen: I read about that |
| 19:56.51 | voxter | Crap, using a local channel to retrieve a parking slot from one of the multi-parking lots doesnt work |
| 19:56.52 | Katty | leifmadsen: but they wouldn't want that here |
| 19:56.54 | leifmadsen | M16722 |
| 19:56.56 | MuffinMan | [ready for testing] [Asterisk] Applications/app_queue 0016722: [patch] app_queue: Give members a penalty time for not answering reported by haakon https://issues.asterisk.org/view.php?id=16722 |
| 19:57.07 | Katty | leifmadsen: our receiptionist is sadly expected to put the current client on hold to answer the next call |
| 19:57.13 | Katty | leifmadsen: i don't approve, but that's how they want it :< |
| 19:57.19 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:57.21 | *** join/#asterisk squeeb (~squeeb@cpc3-staf2-0-0-cust8.brhm.cable.ntl.com) |
| 19:57.25 | leifmadsen | Katty: so old school :) |
| 19:57.33 | squeeb | Can anyone direct me to the support channel for the chan_skype module? |
| 19:57.34 | p3nguin | nny: http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 19:57.41 | Katty | it's southern missouri--what can i say? |
| 19:57.41 | Qwell | squeeb: call Digium |
| 19:57.52 | p3nguin | nny: The Penalties section can help you understand how it could help you. |
| 19:57.54 | squeeb | :/ |
| 19:57.58 | squeeb | ok |
| 19:58.07 | Katty | squeeb: perk up buttercup |
| 19:58.14 | Katty | squeeb: you get to talk to a real person! |
| 19:58.16 | squeeb | I'll try |
| 19:58.21 | Katty | squeeb: isn't that exciting |
| 19:58.23 | squeeb | yea but I'm hopeless on the phone |
| 19:58.27 | squeeb | which is ironic |
| 19:58.28 | squeeb | really |
| 19:58.28 | squeeb | :P |
| 19:58.41 | Katty | not really |
| 19:59.00 | *** join/#asterisk toppe (~root@cs181001225.pp.htv.fi) |
| 19:59.02 | voxter | Is it also a bug, or should you not be able to dial Local/71@parkedcalls for example? (parkedcalls being a dynamic context that res_features populates) |
| 19:59.04 | Katty | just try to be nice! |
| 19:59.11 | squeeb | I'm always nice |
| 19:59.34 | leifmadsen | voxter: I don't think you can dial that directly like that |
| 19:59.58 | squeeb | before I call though, has anyone come across "asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: ast_str_case_hash" before |
| 19:59.58 | voxter | leifmadsen: hmm. That kinda sucks... since 71 is a valid extension inside context 'parkedcalls', youd think it would work. |
| 19:59.58 | p3nguin | nny: The "Note" in the section is especially important to understand. |
| 20:00.27 | voxter | leifmadsen: I was trying to use a Local channel and specify Tt as dial options to get my transfers back... More interesting to what I said before, if i let a parked call 'time out' and it rings back to me, I CAN transfer again. |
| 20:00.35 | voxter | leifmadsen: just not if i retrieve it using ParkedCall() |
| 20:00.50 | Katty | oh boy, willie nelson is coming to missouri |
| 20:00.59 | Katty | i'm sure that's the hilight of someone's afternoon |
| 20:01.07 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 20:01.11 | p3nguin | Nellie Wilson |
| 20:01.40 | Naikrovek | spoonerism |
| 20:01.48 | Naikrovek | ... or something lik eit |
| 20:02.44 | Katty | i don't get it |
| 20:02.45 | Nugget | related: http://macnugget.org/creative/bandnames |
| 20:02.49 | Katty | _ <- |
| 20:02.56 | Katty | - <- your comment |
| 20:03.16 | Katty | _ <- me - <- your comment |
| 20:03.29 | p3nguin | naikrovek: You are correct, it's a spoonerism. |
| 20:03.53 | Katty | is that somehow related to forkerism? |
| 20:04.57 | Naikrovek | no. |
| 20:05.50 | p3nguin | It's kinda like getting your mords wixed. |
| 20:06.23 | *** join/#asterisk toppe (~chris@cs181001225.pp.htv.fi) |
| 20:06.39 | Katty | well that's kinda like forkerism |
| 20:06.53 | Katty | forkerism is when you get the syllables snickerdoodled up |
| 20:07.16 | p3nguin | lysdexia? |
| 20:07.20 | Katty | no |
| 20:07.22 | Katty | like uhh |
| 20:07.22 | ChannelZ | I thought it was when I jammed a fork in someone's hand for touching my pizza. |
| 20:07.22 | Naikrovek | i think for a spoonerism, swapping the first letters of two words which go together in modern vernacular, they must create two different words that also go together in modern vernacular. |
| 20:07.26 | Katty | duck and doochess of windsor |
| 20:07.31 | Naikrovek | i don't have an example, but i bet wikipedia does |
| 20:07.41 | p3nguin | oh |
| 20:07.42 | Katty | or hypodeemic nerdle |
| 20:07.58 | Naikrovek | i'm wrong too |
| 20:08.00 | Katty | i do it all the time |
| 20:08.01 | Naikrovek | here's an example |
| 20:08.02 | Naikrovek | "Is it kisstomary to cuss the bride?" (customary to kiss) |
| 20:08.10 | Katty | yeah, that's forkerism |
| 20:08.42 | Naikrovek | http://en.wikipedia.org/wiki/Spoonerism |
| 20:08.43 | p3nguin | http://dictionary.reference.com/browse/spoonerism |
| 20:08.48 | Naikrovek | jinx |
| 20:08.53 | Katty | ya'll need your own brainwave |
| 20:08.57 | p3nguin | "a blushing crow for a crushing blow" |
| 20:09.27 | p3nguin | forkerism - no dictionary results |
| 20:10.03 | Katty | look up douglas hofstadter |
| 20:10.10 | Naikrovek | "forkerism" is a Kattyism |
| 20:10.45 | *** join/#asterisk rizwank (~rizwank@76.89.131.47) |
| 20:10.59 | squeeb | urrh |
| 20:11.01 | p3nguin | Spoonerism is apparently just for the beginning of the words, which was what I had done with Nellie Wilson. |
| 20:11.05 | rizwank | When running virtualized servers, is CentOS's 1000Hz kernel timer a benefit or hinderance? |
| 20:11.07 | squeeb | wrong version of Asterisk for skype chan :( |
| 20:11.16 | squeeb | I wish debian would update their apt-source more often |
| 20:11.40 | p3nguin | What version do you have and what version do you want? |
| 20:11.43 | Naikrovek | rizwank: you need to be able to separate out 20 or 30ms voice packets very reliably |
| 20:11.45 | p3nguin | s/want/need/ |
| 20:11.52 | Naikrovek | rizwank: seems that a 1000hz timer could do that |
| 20:12.00 | Katty | rizwank: prepare yourself for headaches |
| 20:12.04 | Naikrovek | rizwank: but i recommend testing |
| 20:12.06 | Katty | rizwank: and arm yourself with tylenol |
| 20:12.13 | Katty | rizwank: virtualizing asterisk can be a nightmare |
| 20:12.27 | Naikrovek | rizwank: i have a couple of phone servers running on Hyper-V and they run great, but they run on modern hardware |
| 20:12.45 | squeeb | I have a couple of 1.4.21 setups on Vmware server |
| 20:12.47 | squeeb | that's ok |
| 20:12.49 | rizwank | Katty - I've got a massive 16 core server, so there's lots of ticks available. But I keep hearing its a nightmare and while I build out the infratstrcuture I'm trying to get it right. |
| 20:13.06 | Katty | rizwank: i wish you the very best of luck in your endeavors. |
| 20:13.07 | rizwank | LEave it at 1000Hz timer then --- and is the centos kernel sufficient? It's not tickless... |
| 20:13.08 | Katty | endevors |
| 20:13.11 | Katty | endeavors? |
| 20:13.13 | p3nguin | squeeb: And you've already done apt-get update? |
| 20:13.17 | *** join/#asterisk Igramul (~black@p5497D637.dip.t-dialin.net) |
| 20:13.19 | Katty | ....something like that... |
| 20:13.20 | Naikrovek | rizwank: i have a dual quad core ibm bladeserver blade, and i have two asterisk implementations on there for testing, and they both work perfectly |
| 20:13.21 | squeeb | p3nguin: yea |
| 20:13.26 | squeeb | well, aptitude update |
| 20:13.29 | squeeb | there's nothing newer |
| 20:13.29 | rizwank | which OS? |
| 20:13.31 | squeeb | than .41 |
| 20:13.34 | squeeb | I mean .21 |
| 20:13.42 | Katty | DO YOU REALLY MEAN IT |
| 20:13.45 | p3nguin | squeeb: That's pretty old, so I would expect they should have updated. Maybe not, though. |
| 20:13.50 | Naikrovek | rizwank: ubuntu and AsteriskNOW (CentOS) |
| 20:13.58 | Katty | i miss dav.e |
| 20:14.02 | Katty | eppigy: i miss you. |
| 20:14.05 | Katty | eppigy: :< |
| 20:14.07 | Katty | eppigy: /tear |
| 20:14.29 | Katty | eppigy: no one has spouted nonsense like churros in days |
| 20:14.45 | Katty | file: not even you've made a peep :<<< |
| 20:14.51 | squeeb | p3nguin: yea, I guess that's why they call it stable |
| 20:15.01 | file | ZOMBIES ARE EATING MY BRAINS |
| 20:15.05 | Katty | :>>>>>>>>>>>>> |
| 20:15.14 | rizwank | hmmm |
| 20:15.16 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 20:15.16 | Katty | hugs on file |
| 20:15.22 | file | tickles Katty |
| 20:15.42 | p3nguin | squeeb: Can you use the testing repos to get a newer version? |
| 20:15.47 | p3nguin | testing is still stable software. |
| 20:16.11 | squeeb | maybe |
| 20:16.18 | Katty | asterisk? stable? wait, what?? |
| 20:16.33 | squeeb | I've just compiled 1.4.29 |
| 20:16.35 | p3nguin | squeeb: Alternatively, build your own from source and use checkinstall to roll it into a deb package. |
| 20:16.39 | squeeb | yea |
| 20:16.53 | squeeb | although I went for --prefix=/otp |
| 20:16.55 | squeeb | woah |
| 20:16.56 | squeeb | opt |
| 20:17.10 | *** join/#asterisk xpot-mobile (~xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 20:17.44 | p3nguin | I prefer to set prefix and sysconfdir. |
| 20:18.18 | p3nguin | Otherwise, you stand the chance of things going to /usr/local/ when packaged software doesn't necessarily need to be there. |
| 20:19.27 | Naikrovek | i'm a little surprised GNU Stow never took off |
| 20:19.46 | Naikrovek | you set up a directory, where you could install and compile stuff to |
| 20:20.03 | Naikrovek | then you would use Stow to move the files to /usr/bin/whatever or wherever they needed to go |
| 20:20.22 | Naikrovek | later, if you wanted to remove that software, you could unstow it, and it would cleanly remove the compiiled software |
| 20:20.51 | p3nguin | Someone trying to reinvent the wheel, or just make the current wheel a little better? |
| 20:21.05 | Katty | just watched mr roger speak for about 5 minutes to prpesident nixon |
| 20:21.09 | Naikrovek | configure scripts know if you have stow installed and would work with it, doing things a bit differently if stow were installed |
| 20:21.39 | p3nguin | Sounds similar to checkinstall, but with system integration rather than a standalone app. |
| 20:21.42 | Naikrovek | the purpose being clean uninstallation without package management, in the cases where packages weren't kept up to date or available or whatever |
| 20:21.47 | Katty | 5 minutes, and 20 million for public broadcast. |
| 20:22.12 | squeeb | well |
| 20:22.16 | squeeb | it's all running in /opt now |
| 20:22.19 | squeeb | now to recompile skype |
| 20:22.26 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 20:22.31 | voxter | leifmadsen: https://issues.asterisk.org/view.php?id=16757 |
| 20:22.52 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 20:23.03 | garymc | Hey guys anyone think my card could be faulty A200? We got a power supply to it we checked the power voltage to each pin and they are all correct. we have even plugged in the card and tested the power going to the card. It shows the card is getting power. The var/log/messages is showing the card is not getting power and have we remebered to plug the power cable in? |
| 20:23.31 | garymc | Sangoma say they have never had a card with this fault????? |
| 20:24.16 | Katty | well, garymc |
| 20:24.21 | Katty | i can suggest one of two things. |
| 20:24.36 | garymc | ok |
| 20:24.50 | Katty | the first would be to call sangoma, and have them remote into your box and have a poke around at the equipment |
| 20:24.57 | garymc | done that |
| 20:25.03 | Katty | the second would be to simply buy another a200 card and swap your cards around |
| 20:25.12 | Katty | if the card is at fault, then rma it |
| 20:25.17 | garymc | not doing that, they said they will send a new one |
| 20:25.19 | Katty | if it's not the card, simply return the new one that you bought |
| 20:25.37 | p3nguin | Should a sip phone be configured as a friend or a peer? The phone should be able to make and receive calls. |
| 20:25.38 | garymc | rma? |
| 20:25.39 | Katty | well if they are sending you a new one, simply swap it around |
| 20:25.47 | Katty | err Return Merchandise Authorization |
| 20:25.51 | Katty | for warranty purposes |
| 20:26.00 | garymc | ok |
| 20:26.10 | garymc | nightmare |
| 20:26.13 | Katty | Sangoma has always treated me very well |
| 20:26.27 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:26.35 | Katty | if there was so much of a hint of their hardware being at fault, they were the first to help get it replaced |
| 20:26.53 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 20:27.22 | Katty | p3nguin: it should be setup the same as the rest of your registrations |
| 20:27.32 | Katty | p3nguin: canreinvite=no would probably be extremely helpful |
| 20:27.44 | p3nguin | But I reinvite, so that's out. |
| 20:27.55 | Katty | k |
| 20:28.16 | Katty | p3nguin: obviously keep the extension numbers and passwords different |
| 20:28.48 | Katty | p3nguin: but you already knew that |
| 20:29.57 | p3nguin | If I configure phones as peers, I need to set insecure=very. If I make them friends, the username is used. Friend seems to make more sense, but I wanted someone else's opinion. |
| 20:32.05 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:34.52 | [TK]D-Fender | p3nguin: umm.. that's just wrong... |
| 20:34.54 | squeeb | AAAAAAARHG |
| 20:34.55 | squeeb | same issue |
| 20:35.02 | squeeb | even with a newer version of asterisk |
| 20:35.12 | squeeb | undefined symbol: ast_str_case_hash :/ |
| 20:35.28 | p3nguin | Let me rephrase: It is my opinion that phones should be configured as type=friends (especially if they are configured as host=dynamic), but I wanted the opinion of another because I have been reconsidering. |
| 20:36.22 | p3nguin | [tk]d-fender: Which part do you think is wrong? |
| 20:38.08 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 20:38.26 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 20:38.33 | [TK]D-Fender | p3nguin: You never set insecure for a phone, and you always use peer |
| 20:39.23 | p3nguin | Alright, then in order to change from friend to peer, some other setting will need to be used. Keep in mind that the phone is portable, so host must be set to dynamic. |
| 20:39.26 | *** join/#asterisk mlr78 (~mratliff@74.5.197.138) |
| 20:41.02 | p3nguin | [tk]d-fender: Can you provide a sip definition of your own system where the phone is a peer with host=dynamic? I'm obviously missing something important. |
| 20:41.23 | mlr78 | I'm having trouble provision an SPA942 phone....I"m using dhcp option 66. It can access the path fine, but doesn't seem to like the file. should I be using $MA.xml or the fullmac.xml? Also, is it a best practice to give the phones a dhcp reservation? |
| 20:41.30 | [TK]D-Fender | p3nguin: host=dynamic is fine |
| 20:41.58 | p3nguin | With a dynamic host, it is difficult for peer to match IP/port without an insecure setting being added as well. |
| 20:42.51 | [TK]D-Fender | p3nguin: the point is you DON'T match an IP. |
| 20:43.05 | [TK]D-Fender | p3nguin: and you Do auth all calls |
| 20:43.06 | Igramul | Hi, how can I put a newline into a string in extension.conf? I'd like to send a multiline message via JabberSend. |
| 20:43.14 | p3nguin | hence using "type=friend" |
| 20:43.27 | [TK]D-Fender | p3nguin: No. |
| 20:43.37 | p3nguin | friend lets me use a username, peer matches by IP/port. |
| 20:44.11 | p3nguin | This is what Olle says, anyway. |
| 20:44.12 | Katty | Igramul: i would assume you pipe something into System() which executes a console command to pipe info into a console based jabberclient |
| 20:44.39 | rossand | I'm looking for information on how to start/stop recording a call in progress. Read Monitor(), Record() but that's not quite what I'm looking for. Could someone please point me in the right direction. Thanks. |
| 20:45.36 | [TK]D-Fender | rossand: Who would be triggering it? |
| 20:45.44 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 20:45.59 | carrar | footpedal |
| 20:46.17 | carrar | with a Y connector, one to asterisk one to the guitar |
| 20:46.18 | rossand | [TK]D-Fender: the person calling out (with an asterisk sip account). |
| 20:46.34 | [TK]D-Fender | rossand: "core show application dial" <--------- |
| 20:47.00 | [TK]D-Fender | carrar: jsut bought a new (used0 guitar last night. Last one for a long wihle hoepully |
| 20:47.13 | carrar | PICS!! |
| 20:47.30 | [TK]D-Fender | carrar: none handy, and hard come-by online.... |
| 20:47.37 | [TK]D-Fender | carrar: its a bit of a custom job on top... |
| 20:48.18 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 20:48.25 | Katty | [TK]D-Fender: is it plasticy and say fisher price on the side? |
| 20:49.06 | Katty | or maybe it's a gibson 6 string double cutaway (= |
| 20:49.13 | p3nguin | [tk]d-fender: So can you provide one of your own sip definitions for a sip phone from your system where type=peer and host=dynamic? If I'm missing something or I'm wrong, I want to know -- no sense in continuing to use type=friend if it's the wrong way. |
| 20:49.55 | [TK]D-Fender | carrar: swamp-ash neck-through design, 24 fret, Floyd Rose trem, Grover tuners, DiMarzio PAF PRO humbucker at the neck (w/ single-coil tap), DiMarzio Super Distortion humbucker at the bridge. |
| 20:50.09 | rossand | [TK]D-Fender: option W or X. Thanks! That's exactly what I was looking for. |
| 20:50.10 | [TK]D-Fender | p3nguin: jsut f-ing change it to peer already. |
| 20:50.15 | [TK]D-Fender | :p |
| 20:50.32 | p3nguin | Yeah, that doesn't work. Already explained all that. |
| 20:50.37 | Katty | bleh |
| 20:50.48 | [TK]D-Fender | Katty: I dislike Gibsons...... and Fenders (Never had anything to do with my nick). |
| 20:50.48 | Katty | and here i thought it was going to be somethin pretty |
| 20:50.57 | [TK]D-Fender | p3nguin: yeah... it does.. |
| 20:51.04 | [TK]D-Fender | p3nguin: All of mine are like that. |
| 20:51.09 | [TK]D-Fender | p3nguin: ditto for clients |
| 20:54.43 | stones_ | wow |
| 20:54.49 | stones_ | that was a big waste of time |
| 20:54.54 | stones_ | they dont have any asterisk books |
| 20:54.58 | stones_ | went to two locations |
| 20:55.17 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.16.63) |
| 20:56.21 | rossand | real0m2.550s |
| 20:56.48 | rossand | I can try my other machine with the latest and created code. One sec. |
| 20:57.02 | Katty | [TK]D-Fender: http://userserve-ak.last.fm/serve/_/8061187/David+Gilmour.jpg <- now that is a piece of artwork |
| 20:57.04 | leifmadsen | voxter: thanks! |
| 20:57.12 | Katty | [TK]D-Fender: and i ain't talkin about the guitar ;P |
| 20:57.19 | *** join/#asterisk voipmonk (~shido6@bas10-ottawa23-1177951905.dsl.bell.ca) |
| 20:57.59 | stones_ | Katty, I was thinking while I was driving |
| 20:58.08 | stones_ | Basically, he said a2billing is a small script |
| 20:58.19 | stones_ | So I basically have to make asterisk use the a2billing-did |
| 20:58.23 | Katty | thinking and driving at the same time? that sounds very dangerious. |
| 20:58.28 | stones_ | and then a2billing takes it from there? |
| 20:58.40 | Katty | how should i know |
| 20:58.45 | stones_ | dangerious? not too sure... dangerous.. maybe :) |
| 20:58.46 | Katty | why are you asking me again? |
| 20:58.56 | stones_ | I thought you were involved in the conversation |
| 20:58.57 | Katty | i must have missed something |
| 20:59.01 | Katty | oh. |
| 20:59.08 | Katty | i probably made a wisecrack |
| 20:59.24 | stones_ | well, you know what not to do next time |
| 20:59.31 | stones_ | *wink* |
| 21:00.04 | Katty | i provide sarcastic remarks indiscriminately |
| 21:00.18 | Katty | it's my voluntary contribution to the channel |
| 21:00.27 | Chainsaw | Really livens up the place you know? |
| 21:00.53 | Chainsaw | It's all doom & gloom about broken phone systems otherwise. That wouldn't be nice. |
| 21:01.26 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 21:01.45 | voipmonk | muahahahhahahhaa |
| 21:02.22 | Katty | ^- father gone psychotic. |
| 21:03.06 | p3nguin | [tk]d-fender: If I set the phone as type=peer, calls from the phone do not go into the context configured in the sip definition because the peer match does not occur. Setting insecure=port will allow the phone/peer to match and obey the context. |
| 21:03.31 | [TK]D-Fender | p3nguin: ...... no.... |
| 21:03.31 | *** join/#asterisk Frost7217 (~Frost7217@pool-173-57-96-245.dllstx.fios.verizon.net) |
| 21:03.41 | Frost7217 | Using the Asterix PBX, there is a database that stores the DID to the extension for each line for calling 911. Is it possible to dynamically insert rows into this db frequently? We want to insert when a device calls into the server, and remove after. |
| 21:03.48 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:03.49 | p3nguin | Yeah, that's what's happening. I'm not saying what it should be, I'm saying what it is actually doing. |
| 21:04.04 | [TK]D-Fender | p3nguin: I'm not seeing anything... |
| 21:04.08 | p3nguin | If I dial 2000 from my phone, it is looking for 2000 in default, because the peer did not match. |
| 21:04.39 | p3nguin | If I set insecure=port, now it looks for 2000 in the appropriate context which has been configured in the peer definition. |
| 21:04.48 | [TK]D-Fender | Frost7217: a DID is an INCOMING number. What you are calling an EXTENSION is actually a DEVICE, and call history is stored in CDR <- |
| 21:04.55 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 21:05.11 | Katty | hi tzafrir |
| 21:05.35 | [TK]D-Fender | p3nguin: PASteBIN |
| 21:05.52 | p3nguin | Personally I would rather use type=friend to match on usernames rather than set insecure=port on phones. |
| 21:06.07 | [TK]D-Fender | p3nguin: Every single phone on every single system I've worked on is type=peer |
| 21:06.29 | p3nguin | But what OTHER settings are you using? That's the part you keep leaving out of the puzzle. |
| 21:06.31 | [TK]D-Fender | p3nguin: its even required for presence |
| 21:06.38 | [TK]D-Fender | p3nguin: No other part. |
| 21:07.07 | [TK]D-Fender | p3nguin: FFS. [myphone] type=peer host=dynamic context=someplaceunique |
| 21:07.36 | *** join/#asterisk fiftycal (~fiftycal@cpe-76-174-211-243.socal.res.rr.com) |
| 21:07.37 | p3nguin | I do that, and there is failure. I'm going to show you the sip debug in just one moment. |
| 21:08.31 | Frost7217 | Is E911 commonly supported by hosted asterisk solutions, if so, are there any recommended solutions? |
| 21:10.22 | Katty | sighs |
| 21:10.29 | *** part/#asterisk voipmonk (~shido6@bas10-ottawa23-1177951905.dsl.bell.ca) |
| 21:10.32 | *** join/#asterisk voipmonk (~shido6@bas10-ottawa23-1177951905.dsl.bell.ca) |
| 21:10.44 | Katty | i'm going to refrain from a sarcastic comment this time. |
| 21:16.31 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754708.dsl.bell.ca) |
| 21:17.33 | *** join/#asterisk tlarsen (~tlarsen@adsl-074-237-182-025.sip.hsv.bellsouth.net) |
| 21:17.45 | tlarsen | russellb: Are you around? |
| 21:17.46 | voipmonk | there's an appstore game called "Boody" |
| 21:18.13 | tlarsen | Does anyone at Digium on this channel know if Russell ever got my last email about having lunch and handing out more Android phones? |
| 21:18.21 | voipmonk | I can see it now - kids telling their parents their laying with boody on the phone |
| 21:18.33 | voipmonk | Im playing boody on the itouch, mom |
| 21:19.44 | tlarsen | voipmonk: At least the game isn't named "with my boody". |
| 21:20.35 | tlarsen | (yes, that comment was sophomoric, sorry) |
| 21:20.49 | *** join/#asterisk Wildy (~simba@wn1nat23.beelinegprs.ru) |
| 21:24.40 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 21:25.47 | SuPrSluG | using the asterisk manager how can i get the output of the DIALSTATUS to see if the call was sucessful? |
| 21:28.02 | leifmadsen | tlarsen: only Russell would know the answer to that :) |
| 21:28.09 | leifmadsen | tlarsen: I haven't hacked his email yet |
| 21:28.21 | tlarsen | leifmadsen: Unless he was asking the team about my invitation. |
| 21:28.35 | leifmadsen | I didn't see anything like that |
| 21:28.43 | tlarsen | leifmadsen: (and hadn't gotten around to replying because there was no consensus, etc.) |
| 21:28.53 | leifmadsen | tlarsen: I'd say send him another email :) |
| 21:28.57 | leifmadsen | he likely just got busy |
| 21:29.54 | p3nguin | [tk]d-fender: http://pastebin.com/d2bdeb55f |
| 21:30.29 | p3nguin | [tk]d-fender: type=peer, nat=no, host=dynamic ... |
| 21:31.24 | *** join/#asterisk pawz (~pawz@210.56.91.237) |
| 21:32.02 | [TK]D-Fender | p3nguin: [201](defaults) <-- [rob] |
| 21:32.34 | [TK]D-Fender | p3nguin: [thisisyourusernamedon'tgogettingcreativeonme] |
| 21:32.51 | kfife | I believed that the message "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible." ...could safely be ignored. However a Digium SFA tech told me that "for performance reasons I ought ot contact my ITSP & have them turn it off." I can't think of how it could possibly matter. SANITY CHECK. It TRULY doesn't matter does it? Is there ANY benefit spening even one microsecond of time sc |
| 21:32.52 | kfife | rewing around trying to get my ITSP to do something different? |
| 21:32.58 | [TK]D-Fender | p3nguin: when you split section headers vs username fields like that you cause problems. |
| 21:33.22 | p3nguin | But the phone was assinged 201 for the name. And I even tried using username=201 to keep it consistent. |
| 21:33.38 | *** join/#asterisk mog (~mog@c-71-228-185-24.hsd1.al.comcast.net) |
| 21:33.38 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:33.54 | [TK]D-Fender | p3nguin: WTF is "username=rob" then? |
| 21:34.02 | *** part/#asterisk superbeef (~lanej@74.84.194.4) |
| 21:34.09 | [TK]D-Fender | p3nguin: you aren't consistent right now. |
| 21:34.16 | QubeZ | If I am having no obvious network issues (zenoss verified) around 1130am but in one section of the building, all phones are cutting out with customer on the line (3-4 secs) but do not drop. Happens every hour after 11-ish am. Any ideas on how to start troubleshooting this? |
| 21:34.17 | p3nguin | I obviously changed it when I was trying to get it to do something different. |
| 21:34.24 | p3nguin | I'll change it back. |
| 21:35.36 | p3nguin | [tk]d-fender: The fact of the matter is that even with username=201 (which is the same as the device's name), the results are the same. |
| 21:36.19 | [TK]D-Fender | p3nguin: When I see something consistent I'll give you some credence. Until then I've got years with every system I've ever deployed standing against it. |
| 21:38.32 | p3nguin | Way to avoid that one. |
| 21:39.22 | p3nguin | The fact is that it's the same result regardless of what the username= is set to. username doesn't even appear in the sip debug at all, so how is it even relevant at this point? |
| 21:40.18 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 21:41.06 | stones_ | voipmink... |
| 21:41.17 | stones_ | Could you guys look @ this please http://pastebin.com/m2453b26b |
| 21:42.22 | p3nguin | Not only that, but the reason username is not used is because I am making an inbound call as type=peer. The username is not even checked in that case. |
| 21:43.39 | black | back |
| 21:43.47 | black | nice, relaxing lunch ;) |
| 21:44.23 | stones_ | Could you guys look @ this please http://pastebin.com/m2453b26b |
| 21:45.38 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 21:49.51 | *** join/#asterisk AndyGraybeal (~AndyGray@128-177-27-78.ip.openhosting.com) |
| 21:49.56 | AndyGraybeal | does asterisk work with postgresql ? |
| 21:50.02 | AndyGraybeal | or is there a recommended db? |
| 21:50.10 | leifmadsen | AndyGraybeal: works with many types -- use ODBC |
| 21:50.33 | leifmadsen | although there are native pgsql connectors |
| 21:50.38 | AndyGraybeal | leifmadsen: okay thank you, we are just starting out - should we use 1.4 or 1.6? |
| 21:50.53 | leifmadsen | AndyGraybeal: depends what you need feature wise, and how testing works in your environment :) |
| 21:51.01 | leifmadsen | AndyGraybeal: however, I always start with 1.6.2 now |
| 21:51.11 | leifmadsen | take that as you will |
| 21:51.18 | AndyGraybeal | thank you leifmadsen |
| 21:51.54 | AndyGraybeal | now, since I don't undrestand a whole lot, i know that ODBC means like object database connector.. but other than that -- i don't know - is it like... some postgresql thing like php-pgsql ? |
| 21:52.22 | nny | leifmadsen: is it safe to say 1.6.2 is stable enough for production? |
| 21:52.37 | *** part/#asterisk tlarsen (~tlarsen@adsl-074-237-182-025.sip.hsv.bellsouth.net) |
| 21:52.44 | leifmadsen | AndyGraybeal: no, it's an abstraction layer so Asterisk can talk to ODBC, and ODBC can talk to whatever drivers you have installed -- so you can change databases if you needed to with few changes (if any) on asterisk |
| 21:53.14 | AndyGraybeal | thank you leifmadsen |
| 21:53.19 | leifmadsen | nny: I've installed 1.6.2 at a customer and I don't get calls from them with issues for over a month, and they are running something pre-1.6.2.0, so take that as you will |
| 21:53.42 | leifmadsen | probably something circa 1.6.2.0-rc2 or -rc3 |
| 21:53.54 | *** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net) |
| 21:53.55 | nny | leifmadsen: thanks I will try it myself, I have our own system running 1.6.2 with no issues, just wondering if there are any game breaking issues |
| 21:54.15 | *** join/#asterisk catojo (~catojo@189.24.132.231) |
| 21:54.37 | leifmadsen | nny: nothing is recommended for production without testing -- if you don't have issues, it is good for production -- if you have issues, it isn't. That's the jist of it for all versions of Asterisk. |
| 21:54.50 | nny | yeah |
| 21:55.11 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 21:55.27 | leifmadsen | I never have a belief that there is any point in time where I can recommend any version of asterisk over any other version of asterisk in a general way without specific information about the implementation |
| 21:56.06 | leifmadsen | so really asking a question like, "is X.Y.Z stable enough for production" is an unanswerable question -- some people run trunk in production happily, but they are able to support it |
| 21:57.00 | nny | my main reason for using 1.6 right now is specifcally calling asterisk -rx "channel originate SIP/${WHO}@vitel-outbound application Bridge ${BRIDGEPEER}" |
| 21:57.17 | nny | or "originate: |
| 21:57.17 | p3nguin | Okay, so maybe the entire issue is exactly that -- the username is not being used when it should be. |
| 21:57.54 | p3nguin | Another phone of another model and another sip image can make calls, but my phone cannot. The search continues. |
| 21:58.02 | leifmadsen | nny: right -- the main reason for any version upgrade or implementation of one version over another is features |
| 21:58.03 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 21:58.42 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 21:58.43 | leifmadsen | when a bug is fixed, it's applied to all applicable branches, which means no branch should *really* be significantly more "stable" (whatever that means) than any other |
| 21:58.56 | leifmadsen | steps down off his soapbox |
| 22:00.03 | *** join/#asterisk c4t3l (~c4t3l@c-76-31-57-251.hsd1.tx.comcast.net) |
| 22:00.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:00.37 | *** join/#asterisk Circlefusion (~circlefus@74-132-112-4.dhcp.insightbb.com) |
| 22:00.49 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 22:03.59 | *** join/#asterisk Circlefusion (~circlefus@74-132-112-4.dhcp.insightbb.com) |
| 22:06.15 | Katty | hmm. i feel doomy |
| 22:07.10 | leifmadsen | DOOM |
| 22:07.25 | nny | and now we sing the doom song.. doom doom doom |
| 22:08.32 | p3nguin | Okay, so where does the proxy username belong in a Cisco 7940G? |
| 22:09.08 | p3nguin | I expected linex_name to take care of that. |
| 22:09.09 | *** part/#asterisk nny (~scott@64.203.239.83) |
| 22:09.34 | c4t3l | hello world |
| 22:09.36 | Katty | no i'm serious |
| 22:09.38 | Katty | i feel doomy |
| 22:09.40 | Katty | dreadful |
| 22:09.59 | c4t3l | Katty: no! Stop it! Feel better? |
| 22:10.04 | *** join/#asterisk grantm (~grant@68.142.138.4) |
| 22:10.12 | Katty | that would require identifying the source of doom. |
| 22:12.34 | *** join/#asterisk cweagans (~cweagans@66-193-105-126.static.twtelecom.net) |
| 22:19.41 | *** join/#asterisk fofware (~chatzilla@186.124.184.65) |
| 22:22.53 | *** join/#asterisk fofware (~chatzilla@186.124.184.65) |
| 22:29.22 | *** join/#asterisk fofware (~chatzilla@186.124.184.65) |
| 22:29.30 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.2 (2010/02/02), Asterisk 1.6.1.14 (2010/02/02), 1.6.0.22 (2010/02/02), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels |
| 22:30.43 | bmoraca_work | dag nubbit...another new update |
| 22:31.04 | leifmadsen | Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 have been released to deal with a remote crash vulnerability in T.38. For more information please see http://www.asterisk.org/node/49894 |
| 22:32.50 | p3nguin | Related channels ... |
| 22:32.52 | p3nguin | none? |
| 22:33.16 | *** join/#asterisk titter (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 22:33.47 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.2 (2010/02/02), Asterisk 1.6.1.14 (2010/02/02), 1.6.0.22 (2010/02/02), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: |
| 22:33.50 | leifmadsen | huh |
| 22:33.54 | titter | Can someone explain why the CLI closes after executing this dialplan? The dialplan works, and I get the fax to my e-mail ... but I get some T30 errors, and the CLI cloeses at the end. |
| 22:33.56 | leifmadsen | my xchat doesn't allow me to enter a topic that long |
| 22:33.57 | titter | http://pastebin.com/d7e72ab4b |
| 22:34.04 | eppigy | Katty: <333 |
| 22:34.10 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.2 (2010/02/02), Asterisk 1.6.1.14 (2010/02/02), 1.6.0.22 (2010/02/02), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- |
| 22:34.20 | leifmadsen | [TK]D-Fender: can add them back in if he wishes :) |
| 22:34.21 | bmoraca_work | titter: maybe the same issue that caused the new patch? |
| 22:34.25 | leifmadsen | no one reads the topic anymore :) |
| 22:34.36 | bmoraca_work | titter: per leifmadsen, http://www.asterisk.org/node/49894 |
| 22:34.44 | titter | LOL |
| 22:34.58 | titter | see, I don't idle for a few days |
| 22:35.28 | *** join/#asterisk orangepower (~jon@jonsaves.xen.prgmr.com) |
| 22:35.41 | orangepower | anyone ever setup a polycom VX 1500? |
| 22:35.56 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 22:36.59 | *** join/#asterisk jakent (~john@c-98-233-13-157.hsd1.va.comcast.net) |
| 22:37.49 | *** join/#asterisk aidinb (~Aidin@h-67-103-140-66.lsanca54.static.covad.net) |
| 22:39.12 | ruben23 | hi is it possible to do the vocie recording on asterisk be done on tamdrive- tmpfs rather on HDD storage for fast access..anyone done this.. |
| 22:40.09 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 22:40.11 | *** join/#asterisk niekvlessert (~niekniekn@5ED25657.cable.ziggo.nl) |
| 22:40.18 | niekvlessert | hello! |
| 22:40.36 | niekvlessert | I've got a problem with CID and I'm willing to pay $100 for a solution |
| 22:40.52 | niekvlessert | including the changes in the code ofcourse. ;) |
| 22:41.26 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:41.28 | orangepower | i'm configuring a VX1500.. but everytime i save settings, it reboots and the settings are the old ones.. |
| 22:41.34 | niekvlessert | http://www.freepbx.org/v2/ticket/2563 |
| 22:41.50 | niekvlessert | this says a lot about the problem... |
| 22:42.17 | niekvlessert | I know Asterisk is not a proxy, but I still need a fix |
| 22:43.24 | niekvlessert | It is possible for a proxy or call server to update the Caller ID information that displays on the phone during a call, by modifying the SIP Contact header in the 200 OK message and/or in a re-INVITE message. The phone displays the updated name and number information contained within the Contact header. |
| 22:43.59 | niekvlessert | sounds a little stupid if I have to put opensips in the middle to fix things like this.... |
| 22:44.21 | niekvlessert | I only need it for SIP, not for Zaptel or stuff like that |
| 22:44.54 | niekvlessert | ok $150. :) |
| 22:44.58 | niekvlessert | Anyone? |
| 22:45.15 | p3nguin | leifmadsen: TOPICLEN=390 ... you currently have 371. It's not your client limiting it. |
| 22:45.35 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:46.21 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
| 22:46.32 | joesuffceren | any recommendations on a wireless SIP endpoint or thoughts on wireless SIP endpoint VS FXS and regular cordless phone? |
| 22:48.19 | black | the sip phones usually look cooler |
| 22:48.22 | black | and work better |
| 22:48.33 | black | + you get extra functionality that you may not get with your fxs phones |
| 22:48.46 | black | for a recommendation, I'm not really sure, I don't have any. But I hear the SNOM m6 is good. |
| 22:48.46 | p3nguin | like transfers |
| 22:48.49 | p3nguin | and hold |
| 22:48.59 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 22:49.04 | p3nguin | Siemens has some that people like. |
| 22:49.53 | titter | http://pastebin.com/d529c31a |
| 22:49.56 | titter | Got this error now |
| 22:50.01 | titter | Fax still worked |
| 22:51.32 | niekvlessert | joesuffceren: why not use a Nokia E52 for example |
| 22:51.53 | niekvlessert | it has a SIP client |
| 22:52.10 | niekvlessert | and e-mail support. ;) |
| 22:53.56 | niekvlessert | there's so much 200 OK going on in chap_sip.c... anyone can give me some guidelines how I add SIP headers in chan_sip.c? |
| 22:53.59 | p3nguin | http://www.buy.com/retail/product.asp?sku=212634728&listingid=54137057 |
| 22:54.02 | joesuffceren | niekvlessert: thanks for the recommendation. I'll check it out |
| 22:54.08 | p3nguin | Siemens Gigaset IP Phone |
| 22:54.20 | joesuffceren | black: I'm a big SNOM fan, so I'll check theirs out |
| 22:54.46 | joesuffceren | p3nguin: I'll check siemens out, too. Thanks a bunch! |
| 22:55.23 | p3nguin | This Siemens has lots of calling features. |
| 22:55.39 | niekvlessert | <PROTECTED> |
| 22:55.39 | niekvlessert | <PROTECTED> |
| 22:55.40 | niekvlessert | <PROTECTED> |
| 22:55.40 | niekvlessert | <PROTECTED> |
| 22:55.53 | p3nguin | Try not to flood. |
| 22:55.57 | niekvlessert | excuse me |
| 22:56.16 | *** join/#asterisk steliosk (~Stelios@ipa107.2.tellas.gr) |
| 22:56.25 | niekvlessert | am I close? :) |
| 22:58.54 | niekvlessert | can I maybe do things with the application SIPAddHeader? |
| 23:00.24 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:00.33 | niekvlessert | I have located the actual transfer code |
| 23:00.37 | niekvlessert | in chap_sip.c |
| 23:01.09 | niekvlessert | transmit_notify_with_sipfrag |
| 23:01.17 | niekvlessert | maybe that's something? |
| 23:04.45 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 23:09.50 | niekvlessert | well it seems to be add_header... |
| 23:10.28 | *** join/#asterisk voxter (~voxter@76.77.69.66) |
| 23:10.50 | niekvlessert | A dials B, B can see on the display that A is calling (name + number), A only sees the number dialed |
| 23:11.03 | niekvlessert | what do I add where to display the name on phone A? |
| 23:12.09 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 23:12.36 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 23:13.43 | *** join/#asterisk voipmonk (~shido6@67.204.40.42) |
| 23:15.08 | *** join/#asterisk diatonic (~diatonic@mail.clearwater-research.com) |
| 23:23.44 | p3nguin | niekvlessert: callerid in sip.conf |
| 23:24.26 | niekvlessert | p3nguin: this changes the callee's display, not the caller's display,right? |
| 23:24.32 | p3nguin | no |
| 23:24.47 | p3nguin | Why would the callerid of a peer change what the recipient sees? |
| 23:25.22 | p3nguin | Or maybe I interpreted your question incorrectly... |
| 23:25.30 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 23:25.39 | p3nguin | The callerid setting changes the outgoing callerid for the peer in which you are configuring it. |
| 23:26.14 | niekvlessert | I know |
| 23:26.25 | niekvlessert | the thing is |
| 23:26.27 | niekvlessert | A calls B |
| 23:26.41 | niekvlessert | On B's phone the name A is shown and the number |
| 23:26.57 | p3nguin | Okay, so A's callerid is good. |
| 23:27.04 | niekvlessert | Yes |
| 23:27.09 | p3nguin | Change B's callerid setting. |
| 23:27.24 | niekvlessert | But on A's phone you won't see B's name, only the number |
| 23:27.28 | p3nguin | Change B's callerid setting. |
| 23:27.46 | niekvlessert | it's fine |
| 23:28.01 | niekvlessert | because when I call A from phone B, A will display the name B and his number |
| 23:28.02 | p3nguin | Great, then the problem is nonexistent. |
| 23:28.04 | *** join/#asterisk CrashSys (Kumba@azrael.crashsys.com) |
| 23:28.35 | niekvlessert | You get it now? |
| 23:28.36 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 23:28.46 | p3nguin | First you say B's name/number doesn't show up correctly, then you say it does. Which is it? |
| 23:28.47 | CrashSys | Is there a command in 1.4 that will show me if a SIP friend has active channels or not? (basically if that ext is in use) |
| 23:28.57 | p3nguin | sip show channels |
| 23:29.10 | niekvlessert | read what i say plz... |
| 23:29.12 | p3nguin | crashsys: and extensions cannot be in use |
| 23:29.23 | p3nguin | crashsys: devices can be in use. |
| 23:29.40 | niekvlessert | problem = A doesn't display B's name on the phone when A initiates the call |
| 23:29.49 | p3nguin | Why would it? |
| 23:30.00 | niekvlessert | because callerid has been set correctly |
| 23:30.04 | p3nguin | no |
| 23:30.06 | CrashSys | I am looking for a command that I can use from extensions.conf |
| 23:30.08 | p3nguin | That's now what callerid does. |
| 23:30.21 | niekvlessert | ok great |
| 23:30.23 | p3nguin | s/now/not/ |
| 23:30.25 | niekvlessert | what does that then |
| 23:30.35 | orangepower | ok.. so I have a polyco vx 1500... all the settings seem correct, but the lines say "not registered" and freepbx doesn't list it as connected.. any ideas? |
| 23:30.36 | p3nguin | phone book on the phone |
| 23:30.47 | niekvlessert | yeah but sip can also do it |
| 23:30.59 | niekvlessert | but asterisk doesn't |
| 23:31.01 | CrashSys | Hmmm, I guess I can try to use $(channel_status) |
| 23:31.03 | p3nguin | CallerID is the data SENT when a phone calls another phone. |
| 23:31.09 | niekvlessert | It is possible for a proxy or call server to update the Caller ID information that displays on the phone during a call, by modifying the SIP Contact header in the 200 OK message and/or in a re-INVITE message. |
| 23:31.38 | p3nguin | WHen you make a call, you do not receive callerid by placing the call. |
| 23:31.48 | niekvlessert | Asterisk is not a proxy but it is a call server right? |
| 23:32.01 | p3nguin | right |
| 23:32.15 | niekvlessert | yeah I know that... but It's been done with Asterisk before... I've seen patched Asterisk's |
| 23:32.15 | p3nguin | It is a call control device -- a PBX. |
| 23:32.50 | niekvlessert | So the thing is... I want to update the caller ID information that displays on the phone during a call |
| 23:32.54 | niekvlessert | that's what I want |
| 23:33.03 | p3nguin | That's not what callerID is. |
| 23:33.04 | niekvlessert | and if someone can do that I'm willing to pay it |
| 23:33.16 | CrashSys | Asterisk is a B2BUA. You can freely change anything that goes through it. It is not a proxy. |
| 23:33.23 | p3nguin | You do not receive caller ID by placing a call. |
| 23:33.31 | niekvlessert | I know that callerid doesn't do it, but I need something it does |
| 23:33.39 | niekvlessert | CrashSys: thanks, I know that |
| 23:33.39 | CrashSys | niekvlessert: So you send a call to asterisk, you want to change the CID, then send it elsewhere? |
| 23:34.20 | niekvlessert | No, I want do update the display from the phone CALLING |
| 23:34.36 | p3nguin | crashsys: He thinks that when you place a call to some arbitrary phone number that your phone should display the name of the person you are calling. |
| 23:34.36 | CrashSys | So modify CID Name? |
| 23:34.36 | niekvlessert | and not from some local phonebook but by SIP |
| 23:34.50 | Qwell | niekvlessert: That would be connected line appearance. :D |
| 23:35.03 | Qwell | That feature is (semi?) available in trunk |
| 23:35.19 | niekvlessert | Qwell: and where would this be? |
| 23:35.25 | CrashSys | Just use set(CALLERID(name) = Extra-Stuff${CALLERID(name)}) |
| 23:35.31 | p3nguin | You have to subscribe to that device's hint, right? |
| 23:35.36 | titter | Still got the same fax error on 6.1.14, backed off to 6.1.9 and it's stable again |
| 23:35.46 | Qwell | niekvlessert: I think it was benngard that set that up recently. Maybe he can shed some light on it. |
| 23:36.00 | p3nguin | crashsys: No, he's incorrectly using the term "caller id" here. |
| 23:36.07 | CrashSys | ok |
| 23:36.10 | niekvlessert | i heard asterisk 1.6.3 could do it |
| 23:36.15 | p3nguin | crashsys: He thinks that when you place a call to some arbitrary phone number that your phone should display the name of the person you are calling. |
| 23:36.30 | CrashSys | Ok, gotcha |
| 23:36.34 | niekvlessert | p3nguin: I agree, sorry when i said it wrong |
| 23:36.38 | CrashSys | Yeah, set-up phone books on the phone then |
| 23:36.41 | p3nguin | Which is not what CallerID does. |
| 23:36.42 | niekvlessert | i'm dutch you know :) |
| 23:36.50 | Qwell | niekvlessert: there is no such thing as 1.6.3 |
| 23:36.56 | niekvlessert | i have +500 extensions |
| 23:36.58 | niekvlessert | so no phonebook |
| 23:37.15 | niekvlessert | Qwell: too bad |
| 23:37.15 | p3nguin | 500 extensions, or 500 end devices? |
| 23:37.26 | niekvlessert | +500 extensions, +500 devices |
| 23:37.29 | p3nguin | Learn the terminology, please. |
| 23:37.33 | Qwell | That feature will eventually make its way into 1.8 |
| 23:37.58 | niekvlessert | Qwell: and what about some 'ugly' hacking? |
| 23:38.12 | niekvlessert | this is just one of the things I need |
| 23:38.13 | p3nguin | I have a shared directory available to all phones via http. If I can do it, you can do it for your 500 devices. |
| 23:38.15 | Qwell | As I recall, it's an extremely large patch. |
| 23:38.45 | CrashSys | I would just set-up either an HTTP directory for the phone (hope you are using the same phone brand) or a provisioned ext directory |
| 23:38.46 | niekvlessert | Yeah, but I only need it for SIP |
| 23:39.09 | niekvlessert | p3nguin: ok, nice, that would solve that thing |
| 23:39.14 | niekvlessert | another situation now |
| 23:39.16 | niekvlessert | same problem |
| 23:39.17 | CrashSys | I still find it interesting that the person does not know who they are dialing, but just randomnly is dialing a number |
| 23:39.19 | niekvlessert | call coming in |
| 23:39.28 | niekvlessert | i do a *8 directed call pickup |
| 23:39.34 | CrashSys | Eww |
| 23:39.40 | CrashSys | sorry, continue |
| 23:39.40 | niekvlessert | phone displays *8, not the number from the person i picked up |
| 23:39.57 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:39.59 | carrar | CrashSys, you don't use the randomdial() function? |
| 23:40.14 | CrashSys | No, but I do love me some widgetfubar() function! |
| 23:40.16 | niekvlessert | CrashSys: I agree.... customers... argl :) |
| 23:40.18 | Qwell | or Party A calls Party B. Party B transfers Party A to Party C. Party A has no way of knowing who he is talking to. |
| 23:40.22 | p3nguin | Once again, caller ID is not sent TO YOU when you dial a number. |
| 23:40.26 | niekvlessert | Qwell: yes |
| 23:40.36 | niekvlessert | that's the stuff I'm having trouble with |
| 23:40.37 | Qwell | That is the use-case for connected line stuff |
| 23:40.40 | niekvlessert | GREAT |
| 23:40.47 | niekvlessert | now we're getting some where |
| 23:40.55 | CrashSys | Qwell: That's a training issue. Blind vs Attended transfer |
| 23:40.56 | niekvlessert | so anyone can provide some ugly patches? :) |
| 23:41.17 | niekvlessert | Yeah, but overhere in Europe, everyone does Attended transfer |
| 23:41.20 | Qwell | CrashSys: either way, it doesn't matter. I don't answer my phone with "Hello, Qwell speaking". Do you? |
| 23:41.21 | orangepower | anyone in san francisco want to make $200 by debugging and setting up 1 IP phone? freepbx is installed, running, extensions are in.. small issues.. |
| 23:41.28 | p3nguin | niekvlessert: tell them not to |
| 23:41.46 | CrashSys | Yes, I pick up the phone and say 'Hello this is James' |
| 23:41.49 | Qwell | (I actually do, but that's beside the point) |
| 23:41.55 | niekvlessert | What about a receptionist... |
| 23:42.03 | niekvlessert | she should be able to tell people who's coming in |
| 23:42.10 | niekvlessert | no blind transfer |
| 23:42.12 | CrashSys | As far as a receptionist goes get a hot one |
| 23:42.23 | p3nguin | If I answer a ringing phone, I identify myself by name. |
| 23:42.24 | niekvlessert | lo |
| 23:42.28 | niekvlessert | lol |
| 23:42.48 | CrashSys | Well if she doesn't do a blind transfer then she's doing an attended one, which means she says "Joe is on the phone for you, would you like it?" |
| 23:42.49 | niekvlessert | well nevermind how the world should be, opensips for example does all this stuff |
| 23:42.53 | CrashSys | before hitting the transfer button |
| 23:42.54 | p3nguin | niekvlessert: That's the point of caller id on the receptionist's phone. |
| 23:42.55 | [TK]D-Fender | [18:40]<p3nguin>Once again, caller ID is not sent TO YOU when you dial a number. <-- CPID |
| 23:43.08 | niekvlessert | crashsys: agree |
| 23:43.31 | Qwell | CrashSys: This is for the person on the *other* end. The one on hold while that conversation takes place. |
| 23:43.49 | CrashSys | I fail to see the relevancy in that |
| 23:44.00 | Qwell | Party A has no clue who they are talking to. |
| 23:44.01 | niekvlessert | crashsys describes the problem perfectly... |
| 23:44.18 | CrashSys | If my receptionist is going to transfer me calls without telling me who is on the other end then I need to fire her and find a new one |
| 23:44.19 | niekvlessert | let me try to rephrase |
| 23:44.27 | CrashSys | It's a training issue |
| 23:44.42 | Qwell | CrashSys: no, you are the transfer destination. The problem is the person *being* transfered. |
| 23:45.10 | niekvlessert | Ok let's stay at the directed call pickup thing |
| 23:45.15 | niekvlessert | that's a more obvious problem i guess |
| 23:45.21 | niekvlessert | ? |
| 23:45.25 | Qwell | Or if Party Bs phone is forwarded to Party C. Party A has absolutely no way of knowing. |
| 23:45.37 | niekvlessert | Qwell: agree |
| 23:45.41 | niekvlessert | look at this page |
| 23:45.45 | Qwell | That is the problem he's trying to solve. |
| 23:45.46 | niekvlessert | http://www.freepbx.org/v2/ticket/2563 |
| 23:45.50 | CrashSys | Ehh, I don't see this as much of a problem |
| 23:45.59 | p3nguin | your CID should show you the information for the person who is calling you -- the person who will be on the phone when you pick it up and say Hi. |
| 23:46.06 | CrashSys | So basically what we are trying to prevent is Party A from asking who Party C is |
| 23:46.16 | Qwell | CrashSys: correct |
| 23:46.41 | CrashSys | Yeah, thats almost a useless function |
| 23:46.55 | niekvlessert | P3nguin: agree |
| 23:47.00 | niekvlessert | and that's not always true |
| 23:47.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:47.17 | Qwell | CrashSys: it's a rather common requirement |
| 23:47.24 | *** join/#asterisk fofware (~chatzilla@186.124.184.65) |
| 23:47.27 | niekvlessert | Qwell: Agree (!) |
| 23:47.49 | niekvlessert | so back to the main question |
| 23:47.52 | niekvlessert | having read that page |
| 23:48.19 | niekvlessert | can someone hack some of this contact header stuff in Asterisk to solve my problems?? :) |
| 23:48.21 | orangepower | anyone in san francisco? |
| 23:48.22 | p3nguin | If Joe calls your receptionist, she sees his CID on her phone's display. She she blind transfers to you, YOU will see his CID on your phone's display. If she does an attended transfer, she is calling you, so HER CID will show up on your phone and SHE can tell you that Joe is holding for you, and then she can press the transfer button again. What's the problem? |
| 23:49.00 | Qwell | p3nguin: Joe's POV |
| 23:49.04 | CrashSys | p3nguin: The problem is not outside callers but when calling internally from ext to ext |
| 23:49.10 | p3nguin | Joe knows he called your office. |
| 23:49.18 | CrashSys | It's a useless function as soon as the call is external to your PBX |
| 23:49.20 | Qwell | Joe doesn't know that Susie transfered him to Bobby |
| 23:49.26 | p3nguin | If Joe doesn't know that he called your office, he has problems. |
| 23:49.31 | niekvlessert | p3nguin: there's always a workaround, but it's ugly |
| 23:49.37 | niekvlessert | so people want it fixed |
| 23:49.53 | niekvlessert | and not a ridiculous question |
| 23:50.19 | niekvlessert | what if Joe wants to write down the number from the person calling? |
| 23:50.22 | niekvlessert | he has to ask |
| 23:50.22 | p3nguin | If Joe called your office and Susie answered the call, Joe's phone's CID isn't going to say that Susie called HIM, because she didn't. |
| 23:50.37 | niekvlessert | SORRY not Joe |
| 23:50.44 | niekvlessert | i mean me in the example :) |
| 23:50.51 | CrashSys | p3nguin: The scenario only works when you have all the extensions, including joe, on the same PBX |
| 23:50.52 | niekvlessert | again: |
| 23:51.16 | p3nguin | You can retain the CID information through a semi-attended transfer. |
| 23:51.17 | Qwell | CrashSys: That actually isn't the case |
| 23:51.31 | Qwell | That information can be transmitted via ISDN |
| 23:51.40 | niekvlessert | It's simple: a phone should always display the most correct information about who you are with on the phone |
| 23:51.41 | p3nguin | YOu can make it so that when Susie transfers to you, Joe's CID shows up on your phone. |
| 23:52.04 | niekvlessert | By blind transfer yes.... |
| 23:52.08 | p3nguin | nope |
| 23:52.11 | p3nguin | Read it again. |
| 23:52.17 | p3nguin | I don't agree with it, but it can be done. |
| 23:52.21 | niekvlessert | who's Susie. ;) |
| 23:52.31 | p3nguin | qwell's receptionist |
| 23:52.38 | niekvlessert | what's a semi attended transfer |
| 23:52.49 | CrashSys | Qwell: Asterisk doesn't handle extra/extended ISDN elements too well. It usually tends to treat anything extra as an error and hang up channels. Unless this has changed in 1.6 |
| 23:53.10 | niekvlessert | CrashSys: I only need it for SIP |
| 23:53.10 | p3nguin | She takes the call, she hits Transfer, dials your number, you answer, she tells you that Joe is calling, she presses Transfer again |
| 23:53.24 | p3nguin | That's an attended (semi-attended) transfer. |
| 23:53.35 | niekvlessert | Yeah, but than Susie's number will be on my phone |
| 23:53.39 | p3nguin | no |
| 23:53.40 | niekvlessert | 100% sure |
| 23:53.40 | Qwell | That would be an attended transfer.. |
| 23:53.42 | p3nguin | (1751.15) <p3nguin> You can retain the CID information through a semi-attended transfer. |
| 23:53.48 | p3nguin | (1751.41) <p3nguin> YOu can make it so that when Susie transfers to you, Joe's CID shows up on your phone. |
| 23:53.58 | niekvlessert | I guess that depends on the type of phone... |
| 23:54.19 | niekvlessert | Aastra 675xi does not do it the way you just typed |
| 23:54.26 | [TK]D-Fender | niekvlessert: Bioe |
| 23:54.27 | niekvlessert | it's 2 conversations |
| 23:54.41 | niekvlessert | bridged together |
| 23:54.48 | niekvlessert | D-Fender: ? |
| 23:54.52 | [TK]D-Fender | niekvlessert: ... |
| 23:54.58 | [TK]D-Fender | nope |
| 23:55.02 | [TK]D-Fender | lol @ shift |
| 23:55.09 | p3nguin | Look at the Dial() options. Particularly, the 'o' option. |
| 23:55.21 | [TK]D-Fender | had fun trying to figure out what he just typed |
| 23:55.25 | niekvlessert | o - Specify that the CallerID that was present on the *calling* channel |
| 23:55.25 | niekvlessert | <PROTECTED> |
| 23:55.26 | niekvlessert | <PROTECTED> |
| 23:55.49 | [TK]D-Fender | nieklFor what you'd want is an attended transfer with a CPID update on rebridge |
| 23:56.11 | p3nguin | Which version is that available in? |
| 23:56.14 | niekvlessert | CPID = Called Party ID? |
| 23:56.23 | niekvlessert | 1.4.26.2 |
| 23:56.32 | niekvlessert | in BriStuff |
| 23:56.42 | [TK]D-Fender | niekvlessert: Correct... something for which there is a patch for, but not implemented in this manner, nor merged into the main tree |
| 23:56.44 | [TK]D-Fender | ~cpid |
| 23:56.45 | infobot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 23:57.05 | niekvlessert | Great: I know about the patches |
| 23:57.13 | niekvlessert | but I'm not the best C hacker |
| 23:57.21 | CrashSys | Then you don't get it |
| 23:57.26 | niekvlessert | and patches tend to not work on other Asterisk versions |
| 23:57.28 | [TK]D-Fender | niekvlessert: From a practical stand-point I understand your goal, and there is no patch for so far |
| 23:57.33 | niekvlessert | I want pay for it! :) |
| 23:57.33 | p3nguin | If it is a correct patch, there is no hacking involved. |
| 23:57.37 | CrashSys | if you don't apply the patches, you don't get the functionality |
| 23:57.46 | p3nguin | You patch the code then compile. |
| 23:57.51 | [TK]D-Fender | niekvlessert: the trick is finding which kind of fruit to contribute to ;) |
| 23:58.42 | niekvlessert | p3nguin: a patch for asterisk 1.4.21 won't work on 1.4.26.. so I have to patch 1.4.21 and the rewrite the patch for 1.4.26... |
| 23:58.56 | niekvlessert | I actually called Junghanns, he was willing to do it |
| 23:59.03 | p3nguin | 1.4.21 is the latest version that the patch is available for? |
| 23:59.04 | niekvlessert | but it didn't work on 1.4.21 either |
| 23:59.11 | niekvlessert | the last version i could find |
| 23:59.20 | CrashSys | Use 1.4.21.2 |
| 23:59.26 | [TK]D-Fender | niekvlessert: I did say that the patch didn't do EXACTLY what you wanted.... |
| 23:59.30 | CrashSys | 1.4.21 = funky |
| 23:59.34 | niekvlessert | [TK]D - Fender: how do you mean? |
| 23:59.44 | [TK]D-Fender | niekvlessert: just that is offers the framework you could use as the basis of your next patch |
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