| 00:00.06 | Kobaz | all of a suddon musiconhold would bomb out... or call parking would crash in a +.1 release |
| 00:00.08 | hipitihop | Kobaz, what framework are you using to automate the unit tests ? |
| 00:00.09 | leifmadsen | Get_The_Fish: that's the purpose of the testing frameworks :) |
| 00:00.10 | Get_The_Fish | I am currently playing Russian Roulette to some degree |
| 00:00.31 | Kobaz | leifmadsen: aqua? |
| 00:00.36 | leifmadsen | Kobaz: ya, it's slightly better now that we're doing releases every month instead of 2-4 months like before |
| 00:00.39 | leifmadsen | Kobaz: yep that's the one |
| 00:00.46 | Kobaz | hipitihop: Perl Tests::More, along with thousands of lines of custom modules |
| 00:00.57 | Kobaz | we just started using Test::Deep as well |
| 00:00.57 | leifmadsen | but a testing framework is 100% necessary to keep the quality going up |
| 00:01.09 | Kobaz | you know sqlite |
| 00:01.19 | Kobaz | the guy has over 2 million unit tests |
| 00:01.44 | Kobaz | i haven;t used aqua yet, but it's on my list of more stuff to test |
| 00:02.00 | leifmadsen | Kobaz: I will have to ping you this week so we can start getting your tests into the external testing framework so you don't have to manage it yourself and you'll know the branches are always able to pass your tests |
| 00:02.08 | leifmadsen | Kobaz: ya, same here -- on the list.... :) |
| 00:02.26 | WindBack | I have the following network schema: |ATA SPA2102| <--> |Asterisk| <--> |SIP voip provider|. I want to connect a fax machine in my ATA. My VoIP provider works with T.38 protocol. My ATA supports this protocol. I have my sip.conf file configured with: t38pt_udptl=yes. When I sent a fax from the PSTN i get the following message in the CLI: "WARNING[15287]: udptl.c:1054 ast_udptl_write: (SIP/bdi137): UDPTL asked to send 6 bytes of IF |
| 00:02.26 | WindBack | <PROTECTED> |
| 00:02.28 | WindBack | any idea? or some documentation I can read? |
| 00:02.32 | hipitihop | what is there a relationship to Asterisk and Perl |
| 00:02.43 | Kobaz | hipitihop: perl is SO AWESOME for rapid development |
| 00:03.11 | Kobaz | leifmadsen: sounds good |
| 00:03.29 | Kobaz | leifmadsen: the local channel thing was killing me for a while... i was stuck on older versions and needed fixes from new ones |
| 00:03.56 | leifmadsen | Kobaz: gotcha -- hopefully the latest RCs have fixed that |
| 00:04.06 | leifmadsen | ok, off to finish helping clean instead of working on the computer |
| 00:04.09 | Kobaz | yeah i saw an update to my ticket i'll haver to check out the patch |
| 00:04.10 | Kobaz | k |
| 00:06.14 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 00:06.31 | Kobaz | testing is so amazing |
| 00:06.34 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 00:06.41 | Kobaz | you never quite realize how many bugs you have until you start writing tests |
| 00:08.53 | hipitihop | Kobaz, indeed, in true TDD you write tests before you write real code |
| 00:09.36 | *** join/#asterisk TelTek (~overlord@cpe-76-188-232-92.neo.res.rr.com) |
| 00:09.39 | Kobaz | yeap |
| 00:10.40 | coppice | you don't usually realise who poorly you understand a problem until you start to think up test cases |
| 00:10.47 | Kobaz | hah |
| 00:11.12 | hipitihop | I'd like to see more OSS projects use unitesting frameworks so that end user can easily and optionally load the unit test package and be comfortable that there have been no regressions |
| 00:11.30 | coppice | then you must hate Asterisk |
| 00:12.47 | Kobaz | hipitihop: 'no regressions' is not quite the proper term |
| 00:13.16 | Kobaz | if all the tests pass. you know that all the things that were tested... behave as the tests expect |
| 00:13.24 | Kobaz | but that's all you know |
| 00:13.48 | Kobaz | there's probably 2834908234098234 tests that aren't in there that should be... and there might be bugs in the tests themselves |
| 00:13.56 | hipitihop | Kobaz, well assuming the functionality has been previously well covered by a unit test |
| 00:14.24 | Kobaz | hehe |
| 00:15.22 | hipitihop | and coverage as you know is a another issue .. depending on the language you are using |
| 00:16.06 | hipitihop | but hey, I suspect we have strayed ot here :-) |
| 00:17.22 | *** join/#asterisk voipmonk (~shido6@m155e36d0.tmodns.net) |
| 00:18.01 | hipitihop | but on the point of coverage, ideally your unit testing framework actually allows you to see which functions/methods in your code have been exercised and those that have not |
| 00:19.04 | Kobaz | and then you need to write tests for your unit testing framework |
| 00:19.06 | *** join/#asterisk nightrid3r (kvirc@41.214.142.170) |
| 00:19.41 | Arsenick | Hi all! Is it possible to make asterisk log his cdr call live ? I mean not only when the call is ended ? We want to build live statistic on the calls and it's not so accurante if we got all the information when the call is ended.. |
| 00:19.57 | hipitihop | well that is a given... the unit testing framework should be backed up by its own full suite of unit tests written in itself. |
| 00:20.20 | hipitihop | as is the case with most of the xUnit variants |
| 00:22.33 | ChannelZ | Arsenick: answered last night |
| 00:25.19 | TJNII | Oh snap. |
| 00:26.25 | hipitihop | I know in Dial() I can specify multiple phones, what is not clear is the optional timeout value ... I assume one can only specify one timeout which applies to all phones. |
| 00:26.50 | Arsenick | ChannelZ, yeah I know but don't take it personal, I just wanted to see what others could think about it :) |
| 00:27.24 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:27.40 | TJNII | hipitihop: Yes, that is correct. The timeout applies to the dial, and since you are using one dial for multiple phones, it applies to all the phones. |
| 00:30.10 | Katty | hiphop hippotamus |
| 00:30.28 | hipitihop | Katty :-) |
| 00:34.14 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-161-207.dsl.stlsmo.sbcglobal.net) |
| 00:37.37 | LemensTS | . |
| 00:39.48 | hipitihop | I guess this is a question like "how long is a piece of string" but what is considered a reasonable/standard timeout before sending to IVR or voicemail |
| 00:40.41 | ChannelZ | I think I use 15 seconds |
| 00:40.50 | ChannelZ | it's like 5-ish rings |
| 00:41.29 | ChannelZ | I just looked, it's 20 |
| 00:42.06 | hipitihop | ChannelZ, ok will exoeriment with that... had 10 and seems a little short for someone to get to the phone |
| 00:44.08 | TJNII | I use 30 seconds. I felt anything shorter and callers may not fully appriciate my MOH. |
| 00:44.55 | hipitihop | now I have a |ATA LinksysPAP2t| <--> |Asterisk| <--> |SIP voip provider| .. and in my incoming context I dial the ATA with a timeout ... when I call in from my mobile and wait for the timeout, I start to hear the voicemail announcement but the handset on the ATA still rings a couple of times before it stops.. is that normal, is there some way to control that ? |
| 00:45.25 | Katty | i feel cranky. |
| 00:45.25 | TJNII | Voicemail in Asterisk or on your ITSP? |
| 00:46.04 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:46.27 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 00:46.29 | hipitihop | TJNII, * voicemail .... my diaplan has exten => _X.,1,Dial(SIP/LinksysPAP,15) and then exten => _X.,n,VoiceMail(... and then Hangup() |
| 00:47.16 | hipitihop | TJNII, should I instead, have an answer, a welcome and then the voicemail perhaps. |
| 00:48.05 | TJNII | Your dialplan is fine. There is a propogation delay between Asterisk and the ATA, but that should be very small. One ring, if that. |
| 00:49.48 | hipitihop | TJNII, ok, although that could be disconcerting when you hear the phone and you are russhing to get to it, you still hear it ring and you go answer but already lost it into voicemail |
| 00:50.24 | TJNII | Understandable. |
| 00:50.51 | hipitihop | but that is not really * fault either .. in teh hands of the ATA and handset I guess. |
| 00:55.54 | TelTek | Question for the group: Does broadvoice send CallerName with their CallerID? |
| 00:56.21 | TJNII | Yes |
| 00:56.44 | TJNII | Though for most cell carriers I've noticed the name is just the city and state the phone bills to. |
| 00:57.11 | TelTek | Thanx, I guess I'm just misconfigured to handle them. All I get is the number twice. |
| 00:57.49 | TJNII | It worked for me out of the box, so I don't know of any of the gotchas you might hit. I never had to look into it. |
| 00:59.10 | Katty | any of you folks have a website you know of that sells gaming apparels. |
| 00:59.18 | Katty | specifically xbox 360 games |
| 00:59.58 | hipitihop | I asked this yesterday but I don't I got a response from anyone... is it possible for asterisk to query the voip provider's voicemail ? |
| 01:00.21 | TelTek | I would tell you, but then you'd never answer any of my questions again. |
| 01:00.40 | hipitihop | cringes at his terrible grammar |
| 01:01.57 | hipitihop | TelTek, I'm guessing that is a no then ;-) |
| 01:02.19 | TelTek | ya, my grmmr iz bad, butt not az bad as sum i've seen. |
| 01:03.22 | hipitihop | so if my asterisk system was offline or my Internet connection was down, calls would end up in voip provider's voicemail and no real indication that there has been a failure |
| 01:03.32 | TJNII | hipitihop: It should be possible, but I have no idea how. |
| 01:05.21 | TJNII | wants a blade server with a EVA SAN configured for failover, but doesn't have a spare mil sitting around to buy it. |
| 01:09.59 | *** join/#asterisk medicineman (~medicinem@70.94.45.233) |
| 01:10.09 | TJNII | That's the problem working in the enterprise lab. "Wow! This thing is neat. I want one! $20,000 for the chassis alone? Awwww...." |
| 01:12.15 | hipitihop | well it's good to see the developers like monty python but what odes this mean just cam eup in console: "asterisk.c:3054 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisib |
| 01:12.15 | hipitihop | le!! THIS is an EX-CANARY. (Reducing priority)" |
| 01:15.36 | TelTek | What can * do, to fail to pass along the CallerName? |
| 01:17.30 | TJNII | I've seen caller name come in as the number before, albiet rarely. Have you tried another phone? |
| 01:17.58 | TelTek | It does this on my Cisco 7940 and my xlite softphones. |
| 01:18.33 | TJNII | I mean phone that is calling your system, not your voip phones. |
| 01:18.34 | *** part/#asterisk medicineman (~medicinem@70.94.45.233) |
| 01:19.39 | TelTek | It does this on all my calls. I get the phone number listed twice. Once in the number location and once in the name location. |
| 01:25.44 | hipitihop | is there a way for me to see caller id of incoming calls in the console ? |
| 01:29.09 | TelTek | probably in debug mode |
| 01:31.15 | hipitihop | can I add a Verbose() into my dialplan, and how would I list the callerid ? |
| 01:32.51 | TelTek | I don't know. I was asking the callerID questions. I'm not familiar with how the CalledID data is passed from one point to another. |
| 01:38.59 | *** join/#asterisk TelTek (~overlord@cpe-76-188-232-92.neo.res.rr.com) |
| 01:38.59 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 01:38.59 | *** join/#asterisk Katty (~User@adsl-209-30-145-230.dsl.stlsmo.swbell.net) |
| 01:38.59 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 01:38.59 | *** join/#asterisk skymeyer (~skymeyer@91.183.54.9) |
| 01:38.59 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 01:38.59 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 01:38.59 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 01:39.00 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 01:39.00 | *** mode/#asterisk [+o angler] by farmer.freenode.net |
| 01:42.44 | *** join/#asterisk nosbig (~nosbig@cpe-75-179-4-171.neo.res.rr.com) |
| 01:44.05 | nosbig | I haven't worked with MeetMe in a long time, and I am having trouble with DAHDI configuration... Is there a good resource for understanding DAHDI and its configuration? Last time I worked with conferencing, Zaptel was king... |
| 01:44.30 | manxpower | nosbig: the readme included with dahdi? |
| 01:49.03 | nosbig | Thanks... I am playing around using the Debian distro packages and am finding that the dahdi-linux package really contains almost nothing... |
| 01:49.14 | nosbig | I am building the actual modules now... |
| 02:00.41 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 02:02.57 | neurosys | Ok Lemme know if i understand correctly. SIP os supposed to handle the init of teh call, etc.. but the RTP is supposed to be directly between endpoints? |
| 02:04.36 | TelTek | SIP is for initiating the communications session. RTP is for the raw communications channel data. |
| 02:05.39 | neurosys | TelTek: Ok. Doesnt SIP pass the two endpoint the releveant info to directly pass the RTP traffic to each other directly? Or am I mistaken? |
| 02:08.29 | neurosys | canreinvite |
| 02:08.31 | neurosys | :P |
| 02:08.31 | TelTek | something like that |
| 02:09.09 | neurosys | canreinvite=no forces the traffic to go thru the asterisk box. |
| 02:09.14 | neurosys | ok... i get it :P |
| 02:10.04 | TelTek | canreinvite=no forces the asterisk to relay. it helps with firewalls |
| 02:12.08 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 02:12.15 | *** join/#asterisk tivo1 (~user1@66-169-98-50.dhcp.ftwo.tx.charter.com) |
| 02:14.35 | tivo1 | Evening all, Im knew to asterisk.. as far as finally trying to get it setup... I have a s100-fx which i am trying to set up (just did a install using asterisknow, but when i pick up the phone all i get is a busy signal, so i must be doing something wrong... anyone avail to help me out? |
| 02:16.07 | nosbig | Imagine that, once the module is built and installed, MeetMe works! Thanks! |
| 02:16.56 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 02:44.17 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 03:08.25 | nosbig | Is there a way to configure Asterisk to record each caller's audio in a conference call to a separate track, in the same way that you can with a standard call? |
| 03:14.05 | *** join/#asterisk timeshell (~chatzilla@206.248.136.108) |
| 03:15.00 | *** join/#asterisk UserReg_CL (~COB@pc-51-211-30-200.cm.vtr.net) |
| 03:15.23 | UserReg_CL | hi, helpme... |
| 03:17.21 | dlynes | I've experienced an issue with 1.6.1.14 whereby 28.5 minutes into the call for certain subscribers, the call turns into 'turkey gobble' as one person so eloquently put it...basically...probably the underwater garbled conversation that's so often common with voip calls |
| 03:17.50 | *** join/#asterisk timeshell (~chatzilla@206.248.136.108) |
| 03:18.07 | dlynes | Has anyone experienced this issue? It's a common occurrence with this one particular sip peer, and the person calling them usually hangs up on them and calls back when it happens |
| 03:18.37 | carrar | 'turkey gobble' sounds like a QoS issue |
| 03:18.45 | carrar | BW issues |
| 03:18.49 | dlynes | bw? |
| 03:18.52 | dlynes | oh...nvm |
| 03:18.59 | carrar | or not enough CPU on the * box |
| 03:18.59 | dlynes | Yeah...that's what I was thinking, too |
| 03:19.28 | dlynes | Asterisk box: 19:19:11 up 464 days, 1:37, 1 user, load average: 0.24, 0.25, 0.21 |
| 03:19.45 | carrar | cpu is fine |
| 03:20.15 | carrar | congestion on that users link |
| 03:20.29 | dlynes | I'm actually thinking about rebooting the asterisk box when I get the clear to do so, so that i can reset the packet counts on netstat -s |
| 03:20.31 | carrar | OpenSSH 5.4 released |
| 03:20.54 | dlynes | carrar, openssh 5.4 released is related to what? |
| 03:21.25 | dlynes | or is that just an automated channel speak? |
| 03:21.56 | carrar | not related to your issue |
| 03:22.06 | carrar | that was just a general annoucement |
| 03:22.09 | dlynes | ah |
| 03:23.36 | dlynes | hrm...well, I guess I'll see if I can figure out some code to spit out how many active channels they have every time they get a call, or make a call |
| 03:23.50 | dlynes | Apparently this issue only affects calls coming into them...not calls going out |
| 03:24.46 | carrar | look at core show function GROUP |
| 03:25.29 | carrar | (counting calls) |
| 03:25.56 | dlynes | carrar, that helps process both incoming and outgoing? |
| 03:26.08 | carrar | You can do it however you like, yes |
| 03:26.14 | dlynes | Ok, thanks |
| 03:26.39 | carrar | could have a incoming category and a outgoing one |
| 03:27.32 | dlynes | Ok, and when the call is hung up on, or hung up, the group count is automatically decremented? I don't need to write additional dialplan code to handle that? |
| 03:27.39 | carrar | yes |
| 03:27.44 | dlynes | Ok, thanks |
| 03:36.15 | dlynes | carrar, so any idea what the 'category' is that's associated with groups? |
| 03:36.30 | dlynes | nvm... |
| 03:36.53 | dlynes | I meant what's the difference between category and groupname? |
| 03:43.57 | dlynes | carrar, hrm...doesn't seem to be working...you invoke it using 'Set(GROUP=groupname)', right? |
| 03:45.22 | dlynes | carrar, it's not showing up in group_count (unless it requires the other end to pick up the call), and it doesn't show up using 'group show channels' at the asterisk cli, either |
| 03:47.27 | carrar | try something like Set(GROUP(in)=${ACCOUNTCODE}) |
| 03:49.43 | carrar | then look at GROUP_COUNT(${ACCOUNTCODE}) |
| 03:49.47 | dlynes | Yeah...got it sorted out now |
| 03:50.07 | dlynes | Set(GROUP(category)=groupname) |
| 03:50.17 | dlynes | It's working just fine now...thanks |
| 03:50.54 | carrar | then for outbound calls do something like Set(GROUP(out)=${ACCOUNTCODE}) |
| 03:51.33 | dlynes | carrar, yeah...in this case, I'm doing Set(GROUP(business)=customer) for both outbound and inbound |
| 03:51.50 | dlynes | carrar, because I want to see how many simultaneous calls they're doing |
| 03:51.52 | carrar | go nuts |
| 04:06.18 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 04:06.22 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 04:10.21 | *** join/#asterisk simplydrew_ (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 04:14.25 | tivo1 | any idea what an fxs would get a busy tone? |
| 04:16.16 | Kobaz | it's busy? |
| 04:17.37 | tivo1 | new setup |
| 04:17.43 | tivo1 | thats as far as i am getting |
| 04:18.12 | Kobaz | did you set it up correctly in the configs |
| 04:18.16 | tivo1 | asterisknow with a x100-fx setup to some hand help phones |
| 04:18.28 | Kobaz | pastebin your dahdi/system.conf and chan_dahdi.conf |
| 04:18.29 | Kobaz | oh |
| 04:18.32 | Kobaz | ~asterisknow |
| 04:18.33 | infobot | methinks asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 04:19.37 | tivo1 | interesting, i thought the only difference was the freepbx web interface |
| 04:19.53 | TJNII | The web interface adds thousands of lines of config |
| 04:20.02 | TJNII | And there is a problem in one of them. |
| 04:20.18 | TJNII | We have no idea where, and have little motivation to slog through them. |
| 04:21.56 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qpwiuworwbhzteju) |
| 04:22.38 | tivo1 | well that sux. why have it on the website then? |
| 04:23.04 | TJNII | Some people like it. |
| 04:23.36 | *** join/#asterisk terracon (~greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:23.38 | TJNII | You can always try #asterisknow |
| 04:24.14 | tivo1 | oh i thought you were saying that there was a problem with it over all |
| 04:24.54 | TJNII | No, no. I'm just saying this channel is generally for pure asterisk. To get GUI support you need to ask people who know it. |
| 04:25.27 | TJNII | Sometimes people will help with the GUIs here, but most times it is best to just follow the support channels for them. |
| 04:26.52 | tivo1 | got cha |
| 04:27.17 | tivo1 | i think part of the progblem may be the chan_dahdi.conf |
| 04:27.36 | tivo1 | i dont mind doing things command line.. just thought this would be easier, as there are LOTS of configs |
| 04:27.54 | tivo1 | the chan_dahdi.conf is emply |
| 04:27.57 | tivo1 | er empty |
| 05:18.34 | ChannelZ | part of the problem is once you do it their way, you pretty much have to keep doing it their way or else the GUI will either overwrite or otherwise trash your configs the next time you make a change through the GUI |
| 05:19.29 | *** join/#asterisk ellisdee (~ellisdee@cosmic.sized.penisinyourface.com) |
| 05:21.03 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 05:33.01 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 05:37.37 | Kobaz | tivo1: well that's your problem |
| 05:37.38 | Kobaz | anyways |
| 05:37.39 | Kobaz | nap time |
| 05:42.14 | *** join/#asterisk Kumbang (~dsp@rusnas.paume.itb.ac.id) |
| 06:08.38 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 06:17.14 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 06:34.46 | *** join/#asterisk sourcode (~code@ppp-58-8-125-5.revip2.asianet.co.th) |
| 06:35.09 | *** join/#asterisk sourcode (~code@ppp-58-8-125-5.revip2.asianet.co.th) |
| 06:35.39 | *** join/#asterisk sourcode (~code@ppp-58-8-125-5.revip2.asianet.co.th) |
| 06:37.03 | *** join/#asterisk nullslash (~Nulll@pdpc/supporter/student/nullslash) |
| 06:38.52 | nullslash | Hello, I'm looking for any hardware that help me to connect my regular phone to asterisk server. For example, Ethernet to RJ11 |
| 06:41.25 | carrar | http://shop.ebay.com/i.html?_nkw=FXS |
| 06:41.26 | [TK]D-Fender | nullslash: Linksys PAP2T-NA |
| 06:41.46 | *** join/#asterisk simplydrew (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 06:42.37 | nullslash | [TK]D-Fender, Thanks ^_^ |
| 06:43.00 | nullslash | carrar, Thank you too ^_^ |
| 06:43.03 | carrar | well |
| 06:43.08 | carrar | the 1979 Harley-Davidson Other FXS Low Rider won't help |
| 06:43.09 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 06:43.14 | carrar | but then you won't be on the phone either |
| 06:44.14 | nullslash | carrar, How about Linksys PAP2T-NA ? |
| 06:44.39 | nullslash | I want to cut my telephone billing by building phone server in my home |
| 06:44.53 | nullslash | bills* |
| 06:47.08 | carrar | they work |
| 06:52.09 | *** join/#asterisk Tim_Toady (~moi@188.4.62.188.dsl.dyn.forthnet.gr) |
| 06:55.50 | nullslash | carrar, Do I need an asterisk server if I have Linksys PAP2T-NA ? |
| 06:56.13 | nullslash | can I connect to my provider with out having asterisk server ? |
| 07:00.02 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 07:01.33 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 07:01.55 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 07:02.33 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 07:04.02 | *** join/#asterisk oej_ (~olle@ns.webway.se) |
| 07:04.31 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-ovodejmshomaxokz) |
| 07:04.55 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:13.57 | *** join/#asterisk Polysics (~luca@host39-66-dynamic.30-79-r.retail.telecomitalia.it) |
| 07:14.01 | Ad-Hoc | hi |
| 07:14.04 | Polysics | hello |
| 07:14.33 | nullslash | hi |
| 07:14.44 | Polysics | is it possible to have this: user is already in a call, someone else calls user, caller is put on hold with a message then connected after user closes current call? |
| 07:14.54 | Polysics | something like a single user uque |
| 07:14.57 | Polysics | *queue |
| 07:22.41 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 07:22.58 | *** join/#asterisk ktwilight[m] (~ktwilight@168.209-201-80.adsl-dyn.isp.belgacom.be) |
| 07:38.53 | Polysics | too early in the morning :-) |
| 07:45.20 | *** join/#asterisk ktwilight[m] (~ktwilight@206.193-201-80.adsl-dyn.isp.belgacom.be) |
| 07:46.53 | *** join/#asterisk Hexie (~shane@dsl-244-161-206.telkomadsl.co.za) |
| 07:47.51 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:02.37 | Hexie | Hi guys, |
| 08:04.39 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 08:09.25 | hipitihop | I have just received two registration attempts tha tlook like chan_sip.c:21005 handle_request_register: Registration from '"131383857"<sip:131383857@203.xxx.xxx.xxx>' failed for '204.236.164.208' - No matching peer found |
| 08:10.21 | Hexie | you'll need to add that "peer" to your dial plan |
| 08:11.03 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:11.40 | Hexie | hey guys, hoping someone could assist here - i have a current PBX system (siemens) and i am trying to check the phone "setups" - anyone know if they have a "default" IP address for me to access? |
| 08:11.49 | hipitihop | and another, strieght after for "135093537" .. in both cases from 204.236.164.208 which reverse dns lookup shows as amazonaws... is this someone attempting to register with my asterisk server ? |
| 08:11.51 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 08:12.40 | Hexie | yip and because you dont have them allowed (in your setup) they cant register - have you tried locating that IP address /browsing to it / etc? |
| 08:12.54 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:13.19 | hipitihop | no trying now. |
| 08:13.34 | hipitihop | is this something suspicious ? |
| 08:13.55 | Hexie | or someone doing something that they not sure how to do :) |
| 08:14.27 | hipitihop | hmm, although I can't ping it either |
| 08:15.45 | hipitihop | do those sip numbers look like anything familiar e.g. US trunk calls or some such |
| 08:17.24 | hipitihop | and it is something running in amazon's cloud by the look of it |
| 08:18.31 | Hexie | us-west-1.compute.amazonaws.com is where it seems to be coming from |
| 08:23.20 | *** join/#asterisk ktwilight[m] (~ktwilight@206.193-201-80.adsl-dyn.isp.belgacom.be) |
| 08:26.21 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 08:26.49 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 08:31.33 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:31.44 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 08:34.52 | *** join/#asterisk sulex (~sulex@88-149-154-95.static.ngi.it) |
| 08:35.29 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 08:38.02 | *** join/#asterisk sourcode_ (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 08:38.26 | *** join/#asterisk sourcode_ (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 08:45.24 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:52.31 | *** join/#asterisk Diffen2 (~diffen2@c-2875e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 08:52.43 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 08:53.30 | shamelessn00b | Hi anyone using dicea/netfors ss7 library |
| 08:53.31 | shamelessn00b | ?? |
| 08:56.37 | *** join/#asterisk hluesea (hluesea@213.144.126.108) |
| 08:57.15 | *** part/#asterisk centoslinux (~linuxcent@rhelbox.uio.no) |
| 08:58.39 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 09:00.47 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 09:05.19 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 09:09.07 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:12.03 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 09:39.23 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 09:41.51 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:42.35 | *** join/#asterisk fofware (~chatzilla@190.30.113.46) |
| 09:44.41 | *** join/#asterisk Z_God (~julius@wlan230197.mobiel.utwente.nl) |
| 09:46.49 | *** join/#asterisk Da-Geek (~Da-Geek@62.189.160.33) |
| 09:52.55 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 10:00.24 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 10:02.53 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 10:05.15 | *** join/#asterisk mike8 (~mike@c14.audioline.ba.cust.gts.sk) |
| 10:06.44 | *** join/#asterisk smooth_penguin (~smoove@59.96.228.179) |
| 10:07.07 | *** part/#asterisk arossouw (~arossouw@dsl-146-50-11.telkomadsl.co.za) |
| 10:07.07 | gego | hi there, can anybody give me some hint how to build dahdi with vzaphfc driver on debian (custom 2.6.31.12)? |
| 10:08.22 | *** join/#asterisk Pitel (~pitel@ip-89-103-6-29.karneval.cz) |
| 10:11.19 | Pitel | http://pitel.pastebin.com/KvtnsMiH any idea why ast_string_field_build() does not use string from ctx->event->organizer? |
| 10:12.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:23.28 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 10:30.24 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:32.19 | Polysics | anyone is writing AGI stuff with Ruby, please? |
| 10:32.41 | Polysics | i need to replace Adhearsion, as I did not know that it was not compatible with 16. :-/ |
| 10:33.01 | Polysics | and have quite a bit of non-framework stuff (DB access, PDF invoices) already working |
| 10:33.19 | Polysics | what I am most interested in is actually replacing the AMI part |
| 10:38.29 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 10:54.51 | *** join/#asterisk seba (~seba@p57BDB3A1.dip.t-dialin.net) |
| 10:54.54 | seba | hi |
| 11:01.39 | *** join/#asterisk hluesea (hluesea@213.144.126.108) |
| 11:06.25 | shamelessn00b | hello, anyone with some experience on dicea/netfors chan_ss7 and asterisk? |
| 11:08.27 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 11:26.24 | *** join/#asterisk nullslash (~Nulll@pdpc/supporter/student/nullslash) |
| 11:31.03 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 11:31.29 | Micc | anyone know why a new install of asterisk 1.6.1.17 would not have dialplan reload? |
| 11:34.14 | Micc | I think I figured it out. invalid include file in extensions.conf |
| 11:38.06 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 11:43.23 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 11:44.17 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 11:51.59 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 11:59.48 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 12:02.34 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 12:03.47 | sun28 | moin \o/ |
| 12:12.35 | Gido-E | There is stil no 1.4.29.2? |
| 12:20.59 | *** join/#asterisk e3eli3h (~Chris@95.211.84.170) |
| 12:24.15 | tzafrir | shamelessn00b, a more specific question might actually yield a useful answer |
| 12:26.08 | *** join/#asterisk voipmonk (~shido6@dsl-67-55-18-24.acanac.net) |
| 12:26.20 | shamelessn00b | tzafrir: hi, heres the situation, I have got 2 signalling links coming from A, 1 each on host A and host B, host A and B have a single point code |
| 12:26.36 | shamelessn00b | I want to make the setup failsafe |
| 12:26.49 | shamelessn00b | both A and B have 4 PRIs |
| 12:27.11 | shamelessn00b | like if A goes down I still have 4 PRIs usable |
| 12:27.27 | shamelessn00b | how can I achieve this |
| 12:29.36 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 12:32.03 | shamelessn00b | tzafrir: there? |
| 12:32.30 | shamelessn00b | using mtp3 I was not able to figure how to enable failsafe |
| 12:35.50 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 12:37.24 | Micc | is dahdi suppose to show the d channel on dahdi show channels? |
| 12:37.50 | tzafrir | Micc, no |
| 12:38.08 | *** join/#asterisk Akiraa (~Akira@92.81.196.178) |
| 12:38.21 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 12:39.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:39.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:42.48 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:43.17 | *** join/#asterisk jkroon (~jkroon@dsl-244-3-142.telkomadsl.co.za) |
| 12:44.17 | Micc | I'm getting an error when dahdi loads that pri_cpe is an unknown signalling type |
| 12:45.08 | shamelessn00b | can you share your system.conf file |
| 12:45.10 | shamelessn00b | ? |
| 12:45.36 | Micc | yeah, hold on though. I might have it. |
| 12:46.14 | Micc | this is the error. chan_dahdi.c:14679 process_dahdi: Unknown signalling method 'pri_cpe' at line 275. |
| 12:47.07 | jkroon | ok guys, i've got a weird one, what Telkom calls an ISDN2a box, so might be something standard, might be not. Traditional analog PABX plugged into port A on the unit, B410P into port B, then I get OK when both is plugged in, when I unplug the PABX, I get RED on asterisk. |
| 12:47.21 | jkroon | any ideas? clues? i think wiring but I have no idea how to confirm. |
| 12:47.25 | Micc | is there an easy way to put system.conf file on pastebin? |
| 12:48.23 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 12:49.27 | jkroon | Micc, copy&paste? |
| 12:49.49 | Micc | Ok, here it is. http://pastebin.com/J6fRu5j2 |
| 12:52.19 | Micc | do I need libpri to get signalling method pri_cpe? |
| 12:53.26 | shamelessn00b | yes |
| 12:53.48 | upb | hmm what options are used to calculate AST_BUILDOPT_SUM in buildopts.h ? |
| 12:54.16 | Micc | ok, so I didn't have libpri installed, but I think I need to rebuild dahdi after installing libpri then. |
| 12:54.53 | Micc | So, thats got to be it then, I just need it to see libpri |
| 12:55.12 | shamelessn00b | which cards are you using |
| 12:55.40 | Micc | wcte11xp |
| 12:57.40 | shamelessn00b | I've worked with sangoma mostly |
| 12:57.42 | Micc | would I need to rebuild asterisk too? |
| 12:57.50 | shamelessn00b | but system.conf wont be any different |
| 12:58.02 | shamelessn00b | no, no need to rebuild |
| 12:59.07 | Micc | I'm still getting that pri_cpe error |
| 13:00.47 | Micc | just read someone else with same problem had to rebuild asterisk |
| 13:00.49 | shamelessn00b | whats your setup like |
| 13:01.07 | shamelessn00b | are you trying to connect port 1 to port 2 |
| 13:01.09 | shamelessn00b | ? |
| 13:01.18 | Micc | no. |
| 13:01.31 | shamelessn00b | then? |
| 13:01.53 | *** join/#asterisk ManxPower-work (~manxpower@11.sub-75-216-117.myvzw.com) |
| 13:01.54 | Micc | PRI to integra. 23 bchans and 1 d chan on 24 |
| 13:02.15 | shamelessn00b | ok wait, lemme write you a file |
| 13:03.22 | shamelessn00b | how many ports do you have in the card |
| 13:03.28 | Micc | just one |
| 13:03.34 | shamelessn00b | ok |
| 13:03.58 | shamelessn00b | is this the master port |
| 13:04.00 | shamelessn00b | ? |
| 13:04.10 | Micc | I guess |
| 13:04.15 | Micc | theres only one. |
| 13:04.23 | ManxPower-work | Yjere can only be one! |
| 13:04.29 | ManxPower-work | There can only be one! |
| 13:04.53 | Micc | and only one card. |
| 13:05.05 | Micc | so does that mean its the master? |
| 13:05.25 | *** join/#asterisk pentanol (~pentanol@77-35-10-222.pppoe.primorye.net.ru) |
| 13:05.28 | ManxPower-work | Micc, the 2nd field of the span= line. |
| 13:05.29 | Micc | chan_dahdi.so needs to reference libpri so it needs rebuilt. |
| 13:05.51 | ManxPower-work | Micc, then rebuild it. |
| 13:06.03 | Micc | I am. |
| 13:06.08 | Micc | span=1,1 |
| 13:06.18 | *** join/#asterisk steliosk (~Stelios@ipa107.2.tellas.gr) |
| 13:06.34 | shamelessn00b | no but are you getting the signalling from far end |
| 13:06.45 | shamelessn00b | or are you the master for the far end |
| 13:07.10 | Micc | I don't know, but we have used pri_cpe for years with no problem. |
| 13:07.27 | Micc | I'm guessing we get signaling from far end. |
| 13:07.29 | pentanol | hi there, can I found hidden callerid fron sip\iax? |
| 13:07.40 | shamelessn00b | http://pastebin.com/FZ02Z6fk |
| 13:08.03 | ManxPower-work | Micc, PRI to telco == cpi_cpe, pri to PBX == pri_net, but you would get an error saying that |
| 13:08.15 | ManxPower-work | Micc, what is your specific issue? |
| 13:09.04 | Micc | pri_cpe unknown signalling method. |
| 13:09.35 | Micc | which I'm guessing is because of libpri. so I should have that fixed in a minute. |
| 13:10.20 | *** join/#asterisk aandrade (~aandrade@189.58.7.58.dynamic.adsl.gvt.net.br) |
| 13:12.27 | *** join/#asterisk ManxPower (~manxpower@11.sub-75-216-117.myvzw.com) |
| 13:12.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:13.11 | ManxPower | Micc, Come back when you are done rebuilding Asterisk |
| 13:19.53 | *** join/#asterisk Skeeter- (~Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 13:20.42 | Micc | ok, that fixed it. |
| 13:20.53 | Micc | Its all good now. I need to go to bed. |
| 13:26.07 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 13:40.54 | *** join/#asterisk ManxPower-work (~manxpower@11.sub-75-216-117.myvzw.com) |
| 13:42.32 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 13:45.45 | *** part/#asterisk pentanol (~pentanol@77-35-10-222.pppoe.primorye.net.ru) |
| 13:48.23 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-tzuchrwdrwiysfnj) |
| 13:51.20 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 13:52.03 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 13:58.07 | *** part/#asterisk e3eli3h (~Chris@95.211.84.170) |
| 14:00.05 | *** part/#asterisk ManxPower-work (~manxpower@11.sub-75-216-117.myvzw.com) |
| 14:00.10 | *** join/#asterisk ManxPower-work (~manxpower@11.sub-75-216-117.myvzw.com) |
| 14:02.03 | *** join/#asterisk WinZ (~winz@82.146.61.218) |
| 14:03.37 | *** part/#asterisk WinZ (~winz@82.146.61.218) |
| 14:06.42 | upb | hmmm, what do i need to do, besides selecting 'app meetme' in make menuselect, for the line |
| 14:06.58 | upb | '#define AST_MODULE_APP_MEETME 1' to appear in asterisk/buildopts.h |
| 14:06.58 | upb | ? |
| 14:07.29 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:07.42 | [TK]D-Fender | upb: I'd say something like fulfilling its dependencies... |
| 14:07.59 | upb | app_meetme.so was built |
| 14:12.22 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 14:12.38 | *** join/#asterisk ManxPower-work (~manxpower@11.sub-75-216-117.myvzw.com) |
| 14:13.21 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 14:14.01 | *** join/#asterisk Skeeter- (~Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 14:14.11 | Pitel | http://pitel.pastebin.com/ksA5k8Cy any idea what migh be wrong? |
| 14:14.31 | Katty | GOOD MORNING |
| 14:14.58 | upb | which file does menuselect save the selections info to ? |
| 14:15.54 | ManxPower-work | Pitel, You might have better luck in #asterisk-dev |
| 14:16.08 | Pitel | thx |
| 14:16.32 | *** part/#asterisk Pitel (~pitel@ip-89-103-6-29.karneval.cz) |
| 14:17.28 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:21.47 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:c97:55aa:a6d0:dcfd) |
| 14:22.53 | upb | [TK]D-Fender: could you have a look ? http://pastebin.com/G9ENJxyh |
| 14:24.03 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:24.42 | *** join/#asterisk Keal (~chiamuff@unaffiliated/jargon) |
| 14:24.55 | Keal | What is the source code written in? |
| 14:25.12 | tzafrir | vim |
| 14:25.26 | Keal | What language. |
| 14:25.39 | tzafrir | C, mostly |
| 14:25.56 | Keal | C or C++? |
| 14:26.58 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 14:28.39 | Keal | Who is that guy that helps with Asterisk for donations? |
| 14:28.41 | upb | is there some option to compile apps statically into asterisk, and those appear in buildopts.h ? |
| 14:29.14 | ManxPower-work | upb, not sure what you are doing with buildopts.h, but why don't you just use the menuselect framework like everything else does? |
| 14:29.51 | Keal | voipmonk? |
| 14:30.04 | voipmonk | yes? |
| 14:30.25 | upb | ManxPower-work: i need to generate the same buildopts.h as packaged asterisk |
| 14:30.28 | upb | so i can load my module |
| 14:31.19 | upb | because there is some 'security' check based on options which results in a md5 checksum compiled into asterisk binary and modules and when those differ, the module is ignored |
| 14:32.07 | ManxPower-work | upb, have you read /path/to/src/asterisk/menuselect/README ? |
| 14:32.08 | Katty | file: ski ba bop ba dop bop |
| 14:32.16 | file | BAZINGA! |
| 14:32.18 | nullslash | Hello, I'm looking for SIP provider. Any recommendations ? |
| 14:32.30 | ManxPower-work | upb, you're not going to get much help with packaging in this channel |
| 14:33.11 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 14:33.13 | Katty | file: it's a scoobie oobie doobie scoobie doobie melody |
| 14:33.30 | upb | ManxPower-work: i dont need to package asterisk, i need to compile a module that LOADS into packaged asterisk |
| 14:33.35 | file | offers Katty some strawberry yogurt |
| 14:33.49 | ManxPower-work | upb, have you read /path/to/src/asterisk/menuselect/README ? |
| 14:33.54 | upb | no, let me read it |
| 14:34.30 | neurosys | Oh great and powerful ones. The n00b is stuck once again. I keep getting a congestion error on outbound trunks and cant figure out why. I dont see anything that helps in the debug. WOuld someone be so kind as to double check me? |
| 14:34.31 | neurosys | http://pastebin.ca/1828531 |
| 14:35.12 | upb | ok, read :) |
| 14:35.48 | nullslash | I'm look for anyone that has a cheep SIP accounts. |
| 14:36.31 | nullslash | There is a company that provide account for 11 dollar a month |
| 14:37.09 | ruyo | How can I know if a patch was included in Asterisk and, if it was, in which version? |
| 14:42.30 | *** join/#asterisk McBoingbo (~Galabaga@mail.hrsg.ca) |
| 14:43.21 | ManxPower-work | ruyo, Um, the Changelog included with every version of Asterisk and on downloads.digium.com ? |
| 14:43.45 | ManxPower-work | NEXT! |
| 14:44.10 | McBoingbo | what is usually the problem when error "Got SIP response 603 "Decline" back from 192.168.128.xxx" comes up? Is this typically a provider problem? |
| 14:44.26 | ruyo | Ok, thanks ManxPower-work. |
| 14:44.50 | ManxPower-work | McBoingbo, your provider is rejecting the call. |
| 14:45.22 | McBoingbo | is there typically some config mismatch that I need to tweak to resolve this? |
| 14:45.26 | ManxPower-work | Assuming your provider is at 192.168.128.xxx, which seems unlikely. |
| 14:45.29 | McBoingbo | or is it more like your ISP going down |
| 14:45.44 | ManxPower-work | McBoingbo, There is no "typical" for this. |
| 14:46.27 | McBoingbo | sorry, I am trying to determine if the problem is on my end |
| 14:46.35 | ManxPower-work | McBoingbo, You sent a request to 192.168.128.xxx. 192.168.128.xxx declined to process it. You do the math. |
| 14:46.52 | Katty | file: mmmmnothanks. |
| 14:46.58 | Katty | file: i have a granola bar |
| 14:47.13 | Katty | file: thank you for reminding me about it |
| 14:47.20 | file | Katty: <3 |
| 14:47.48 | ManxPower-work | McBoingbo, Chances are the problem is on your end. Bad username, bad passeword, account doesn't have enough money in it, provider is going out of business, I'm sure I could than of 20 more |
| 14:48.01 | McBoingbo | heh, ok |
| 14:48.23 | ManxPower-work | paste JUST the dial line as it appears on the console when you do a call |
| 14:48.46 | ManxPower-work | If you're doing something obviously stupid we should see it in that one line paste. |
| 14:48.54 | McBoingbo | k |
| 14:48.58 | Katty | file: it does not taste like smores :< |
| 14:49.07 | file | Katty: unsurprising. |
| 14:49.13 | Katty | file: my granola bar must have gotten into the wrong package somehow |
| 14:49.34 | Katty | file: tastes like too much sugar and not enough oatmeal |
| 14:49.43 | *** join/#asterisk ktwilight[m] (~ktwilight@206.193-201-80.adsl-dyn.isp.belgacom.be) |
| 14:50.29 | McBoingbo | ManxPower-work: Executing Dial("SIP/248-b6704ce8", "SIP/613xxxxxxx@sip.babytel.ca|300|r") in new stack |
| 14:50.57 | ManxPower-work | McBoingbo, What version of Asterisk are you using? |
| 14:50.58 | Katty | McBoingbo: no one here really cares who you're dialing |
| 14:51.00 | Katty | McBoingbo: just FYI |
| 14:51.12 | McBoingbo | I know, habit Katty :) |
| 14:51.14 | Katty | k |
| 14:51.26 | ManxPower-work | McBoingbo, does your provider require a leading 1? |
| 14:51.35 | McBoingbo | I am embarressed to say which version |
| 14:51.38 | *** join/#asterisk Jhirley (~Jhirley@adsl-3-64-160.mia.bellsouth.net) |
| 14:51.45 | ManxPower-work | McBoingbo, also, don't dial by hostname, dial by sip.conf entry |
| 14:51.51 | ManxPower-work | that is likely your issue |
| 14:51.57 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 14:53.39 | Katty | superbeef: so are you like superlean |
| 14:53.44 | Katty | superbeef: or superjuicy |
| 14:54.01 | superbeef | ground chuck with a cape |
| 14:54.07 | superbeef | meat that saves |
| 14:54.13 | Katty | what are you saving? |
| 14:54.23 | Katty | cause ground chuck is kinda...well... |
| 14:54.41 | neurosys | getting a congestion error on my outbound trunk only. can anyone lend a hand? http://pastebin.ca/1828531 |
| 14:54.43 | Katty | not all that healthy for you (= |
| 14:54.49 | coppice | Katty supercollesteric |
| 14:55.09 | superbeef | I save deer burgers from falling apart |
| 14:55.16 | Katty | ohisee. k |
| 14:55.23 | McBoingbo | ManxPower-work: Sorry still learning Asterisk, what do you mean by do not dial by hostname, I assume you mean "SIP/248-b6704ce8" and to use the contexts within sip.conf instead? |
| 14:55.52 | ManxPower-work | no, I mean sip.babytel.ca |
| 14:55.53 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:55.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:56.02 | ManxPower-work | McBoingbo, you should read the Asterisk Book. |
| 14:56.06 | ManxPower-work | ~answers |
| 14:56.07 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 14:56.12 | Katty | good morning mister madsen |
| 14:56.19 | *** join/#asterisk nullslash (~Nulll@pdpc/supporter/student/nullslash) |
| 14:56.24 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 14:56.27 | Kobaz | teh yawns |
| 14:56.33 | Katty | hi Kobaz |
| 14:56.42 | Kobaz | yello |
| 14:56.47 | puzzled | hi |
| 14:57.02 | Katty | hi puzzled :> |
| 14:57.38 | McBoingbo | my Asterisk is version 1.2.12.1 btw lol |
| 14:57.54 | Katty | ha |
| 14:57.55 | ManxPower-work | McBoingbo, then there is no hope for you. |
| 14:58.01 | McBoingbo | wtf hehehe |
| 14:58.06 | Katty | that was like 5 years ago |
| 14:58.25 | ManxPower-work | almost nobody here has even ever used 1.2.x |
| 14:58.27 | McBoingbo | yes, I admin here, most systems were seriously outdated |
| 14:58.41 | Kobaz | i used 1.2 |
| 14:58.51 | McBoingbo | I am working on getting our phone system updated |
| 14:58.54 | ManxPower-work | McBoingbo, don't worry, when your server gets hacked and you get a $10,000 phone bill, I'm sure you'll finally upgrade. |
| 14:59.15 | McBoingbo | eeek, 1.2 exploitable? |
| 14:59.20 | superbeef | McBoingbo: I feel your pain... all the Boxes were 1.2.wayold when I was handed hte phones |
| 14:59.33 | ManxPower-work | McBoingbo, 1.2 has not had security updates for years. |
| 14:59.36 | Katty | McBoingbo: get upgrading. |
| 14:59.39 | McBoingbo | I dont make the decisions, but I can knudge |
| 14:59.39 | Katty | McBoingbo: stat (= |
| 14:59.48 | McBoingbo | Manx is hacking meeeeeeeee |
| 14:59.54 | puzzled | anyone know if 1.6.1.x supports tls/srtp with self signed certs? |
| 15:00.03 | Katty | it supports your mom |
| 15:00.06 | Katty | OH SNAP |
| 15:00.10 | McBoingbo | lawl |
| 15:00.16 | ManxPower-work | McBoingbo, Also most documentation you'll find will be fore 1.4\ |
| 15:01.00 | nullslash | Hello, I |
| 15:01.47 | nullslash | Hello, I'm looking for any company that sells SIP account. Could you suggest any? |
| 15:01.52 | Kobaz | ~itsp |
| 15:01.53 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 15:02.00 | Kobaz | i win |
| 15:02.06 | Katty | HE"S TOO FAST FOR YOU OLD MAN |
| 15:02.17 | tuxx- | :D |
| 15:02.33 | Kobaz | ~itsplist-us |
| 15:02.34 | infobot | rumour has it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 15:02.38 | superbeef | Katty: i think your overly nutritious breakfast gave you too much morning energy |
| 15:02.52 | Katty | no |
| 15:02.56 | Katty | i'm always like this |
| 15:03.04 | superbeef | lol |
| 15:03.27 | McBoingbo | ManxPower will I be able to make the change you spoke of earlier with Asterisk 1.2? |
| 15:03.59 | *** join/#asterisk orioni (~chatzilla@95.107.233.98) |
| 15:04.16 | orioni | is there a module to connect asterisk to my skype account so my outgoing calls are made from skype ? |
| 15:04.51 | ManxPower-work | ~skype |
| 15:04.52 | infobot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 15:05.37 | orioni | ~SkypeForAsterisk |
| 15:05.38 | infobot | rumour has it, skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
| 15:06.17 | Kobaz | i need another monitor |
| 15:06.28 | Kobaz | two's just not enough |
| 15:06.33 | Katty | Kobaz: i feel your pain |
| 15:06.44 | Katty | Kobaz: i too, could use a 3rd monitor |
| 15:06.56 | Katty | Kobaz: instead, i'm just using an additional laptop |
| 15:07.47 | Kobaz | heh |
| 15:07.56 | Kobaz | x2vnc is quite cool |
| 15:08.10 | Kobaz | i used that for a while... i don't usuually bring a laptop to the office these days though |
| 15:09.23 | McBoingbo | ManxPower-work: Also when I pasted that DIAL line, it was to my cellphone, when dialing an internal extension it does not append that host info, thats only for external calls |
| 15:09.55 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 15:13.23 | [TK]D-Fender | McThat dial looks like you're calling with NO AUTH and not using a proper peer entr |
| 15:13.26 | [TK]D-Fender | y |
| 15:15.04 | Skeeter- | ManxPower-work, just got real time weather to work with polycoms |
| 15:16.26 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:16.31 | Katty | cute |
| 15:18.10 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:23.44 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:23.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:24.22 | adnc | when i start asterisk with -cvvvv i get a colorfull console where attaching with -rvvvv doesnt. is there a way to get such coloring when attaching to asterisk with -r option? |
| 15:25.42 | Kobaz | i think 1.6.2 supports attaching and getting colors |
| 15:26.04 | *** join/#asterisk af_ (~getsmart@88-149-240-251.dynamic.ngi.it) |
| 15:26.20 | ManxPower-work | Kobaz, and 1.6.1 and 6.2.0 and 1.4, and 1.2, and 1.0, and 0.65 |
| 15:26.36 | Kobaz | oh hmm, never noticed |
| 15:27.13 | adnc | ManxPower-work, how can i activate this? |
| 15:27.17 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:27.21 | ManxPower-work | adnc, it happens automatically |
| 15:27.35 | adnc | well, here it doesnt. only when doing -c |
| 15:27.43 | adnc | i do have 1.4 |
| 15:28.11 | ManxPower-work | adnc, then you have some issue somehwere. |
| 15:28.22 | sbrath | I have a queue configures with ringinuse=no, and joinempty=strict .... Is InUse still allowed to get a call with strict? |
| 15:28.34 | adnc | strange that it does it when called with -c and doesnt when calling with -r |
| 15:29.05 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:29.05 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:30.06 | Katty | hi Deeewayne |
| 15:30.13 | leifmadsen | adnc: what distro? |
| 15:30.23 | adnc | leifmadsen, debian lenny |
| 15:30.55 | leifmadsen | adnc: I think I remember there being some change that went in recently to fix that on Debian/Ubuntu in the init scripts... do you start with /etc/init.d/asterisk start ? |
| 15:31.00 | ManxPower-work | adnc, did you build Asterisk from source. |
| 15:31.08 | adnc | leifmadsen, yes i do |
| 15:31.12 | Katty | hmm |
| 15:31.13 | leifmadsen | ManxPower-work: I've seen that before too on all my Ubuntu installs |
| 15:31.16 | adnc | ManxPower-work, no, i use the debian package |
| 15:31.33 | ManxPower-work | adnc, then ask the package maintainer |
| 15:31.34 | Katty | leifmadsen: how is 1.6.2.5 doing? |
| 15:31.34 | nullslash | Unlimited USA Plan: $17.95 mo. Hmm, is that the cheapest thing I can get ? |
| 15:31.38 | Katty | leifmadsen: IS IT SAFE |
| 15:31.45 | leifmadsen | adnc: try checking out the latest asterisk from branch (1.6.0, 1.6.1, or 1.6.2, whichever you're using), and try running "make config" to see if that fixes it. Then restarts with the init script |
| 15:31.53 | ManxPower-work | nullslash, I pay about $2/month for my calls |
| 15:31.57 | Deeewayne | Katty, morning! |
| 15:32.02 | leifmadsen | Katty: that is not a question that can be answered -- does it pass your regression testing? |
| 15:32.03 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:32.06 | nullslash | ManxPower-work, What is your provider ? |
| 15:32.11 | adnc | leifmadsen, you are suggesting to compile it myself? |
| 15:32.18 | leifmadsen | adnc: that's not what I said |
| 15:32.18 | ManxPower-work | nullslash, Vitelity. |
| 15:32.34 | adnc | leifmadsen, where should i make config then? |
| 15:32.39 | leifmadsen | adnc: I said get the latest source, and run 'make config' (which will install the latest init scripts) |
| 15:32.46 | adnc | ahh |
| 15:32.47 | leifmadsen | in the source directory you check out |
| 15:32.47 | adnc | i see |
| 15:32.48 | Katty | leifmadsen: hmm. well i haven't done any 1.6.x testing yet |
| 15:32.56 | Katty | leifmadsen: so i guess we will find out! |
| 15:32.58 | adnc | leifmadsen, thanks, i will try |
| 15:33.00 | leifmadsen | Katty: I can't answer your question -- it works for lots of people :) |
| 15:33.08 | leifmadsen | Katty: I have it in production on a couple of servers |
| 15:33.08 | Katty | leifmadsen: WELL IT BETTER WORK |
| 15:33.12 | Katty | leifmadsen: OR ELSE I WILL FIND YOU |
| 15:33.14 | nullslash | ManxPower-work, Do you have e911 ? |
| 15:33.18 | leifmadsen | can't be held liable for anything |
| 15:33.20 | adnc | Katty, ping! |
| 15:33.22 | ManxPower-work | nullslash, yes. |
| 15:33.24 | Katty | adnc: pong |
| 15:33.28 | adnc | :) |
| 15:33.36 | ManxPower-work | But I do not have an unlimited account. |
| 15:33.38 | Katty | leifmadsen: that's okay. i'll still blame you ;) |
| 15:33.40 | Katty | leifmadsen: <3 |
| 15:33.54 | Katty | so what are the rest of you crazy folks using |
| 15:34.06 | nullslash | ManxPower-work, Is that unlimited plan? |
| 15:34.09 | Katty | 1.4 still? |
| 15:34.26 | ManxPower-work | <ManxPower-work> But I do not have an unlimited account. |
| 15:34.31 | leifmadsen | I do new installs with 1.6.2 currently, and I don't upgrade existing systems that are working fine |
| 15:34.35 | leifmadsen | that is my rule |
| 15:34.44 | Katty | kay |
| 15:34.49 | Katty | well this is a shiny new box. |
| 15:34.49 | nullslash | ManxPower-work, I didn't see that |
| 15:35.01 | leifmadsen | Katty: then I suggest trying out 1.6.2 :) |
| 15:35.02 | Katty | so we're gonna see how she drives. |
| 15:35.07 | Katty | vroomvroom |
| 15:35.08 | leifmadsen | Katty: 1.6.2.6-rc2 is out |
| 15:35.23 | leifmadsen | has a bunch of fixes since 1.6.2.5 (which is actually 1.6.2.2 with security fixes) |
| 15:35.29 | ManxPower-work | nullslash, 1) there is no such thing as an unlimited account. All providers have usage caps 2) unless you do a lot of calling, "unlimited" will be more expensive for you than per/min accounts |
| 15:35.31 | leifmadsen | so I'd suggest playing with that |
| 15:35.41 | leifmadsen | ManxPower-work: amen |
| 15:35.44 | Katty | checks releases directory |
| 15:36.03 | leifmadsen | ManxPower-work: I have an account at 1.1c per minute, and $50 gets me like 4 months of talk time |
| 15:36.08 | Katty | yay, there it is |
| 15:36.14 | leifmadsen | Katty: \o/ |
| 15:36.24 | ManxPower-work | leifmadsen, I have very low calling volume on my personal account |
| 15:36.27 | leifmadsen | ManxPower-work: and I use my phone a lot |
| 15:36.37 | leifmadsen | ManxPower-work: ya, I have 3 DIDs on that account, plus talk time |
| 15:36.52 | leifmadsen | personal, business, and personal (London, ON) for local inbound for the parents |
| 15:38.36 | Katty | leifmadsen: dahdi 2.2.1? |
| 15:38.53 | Katty | leifmadsen: or the rc2 one? |
| 15:39.18 | leifmadsen | Katty: not sure... just use the latest DAHDI I guess (full release) unless you want to help test |
| 15:39.27 | leifmadsen | if it's a test box, I just say use the latest of everything |
| 15:39.38 | Katty | well it'll be in production eventually |
| 15:41.42 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:41.51 | tivo1 | anyone want to try and help me out... did an install yesterday... was able to get a sip softphone to dial the echo test, but am not able to get my fxs to work (i only get a buys signal when i pick the phone up) i beleive the s100-fx is causeing the busy signal. |
| 15:42.08 | [TK]D-Fender | tivo1: ... #freepbx <-------- |
| 15:42.11 | [TK]D-Fender | ~freepbx |
| 15:42.12 | 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 |
| 15:42.38 | leifmadsen | Katty: might as well test on the RCs then since they will likely be out by the time you go to production (unless its going in today :)) |
| 15:44.05 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.12.6) |
| 15:44.35 | Katty | looks at apt |
| 15:44.41 | Katty | newp, not today i don't think |
| 15:44.55 | Katty | not unless our internets suddenly speed up |
| 15:44.58 | Katty | and everyone gets off youtube |
| 15:45.19 | ManxPower-work | Katty, block youtube 8-) |
| 15:45.25 | Katty | screw a bunch of that |
| 15:45.38 | leifmadsen | no way, make youtube faster so people can watch their videos and get back to work! |
| 15:45.54 | Katty | my day is filled with cute animal videos off youtube |
| 15:46.00 | *** join/#asterisk Kheru (~rah@212.80.198.1) |
| 15:46.03 | Katty | i'd lose my sanity without it :P |
| 15:46.05 | ManxPower-work | *gasp* Really? |
| 15:46.17 | Katty | your sarcasm is delicious |
| 15:47.02 | Katty | so anyone using any fun asterisk manager stuffs i dont' knwo about yet? like isymphony or asterisk-stat |
| 15:47.21 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:50.51 | tzafrir | Is the infobot factoid dictionary channel-specific? |
| 15:51.12 | tzafrir | If not, I wonder where else "FreePBX is unable to be supported" |
| 15:51.21 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:52.03 | leifmadsen | heh |
| 15:52.31 | leifmadsen | * [infobot] #asterisk-doc #asterisk-dev #asterisk-bugs #asterisk |
| 15:52.35 | leifmadsen | just asterisk channels it seems |
| 15:52.54 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 15:55.07 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:55.51 | adnc | https://issues.asterisk.org/9048 why isnt this working? |
| 15:56.53 | elred_ | 404 |
| 15:56.58 | *** join/#asterisk moy (~chatzilla@74.12.129.100) |
| 15:57.56 | *** join/#asterisk smanek (~smanek@rrcs-24-213-164-136.nyc.biz.rr.com) |
| 15:58.20 | Katty | adnc: just to piss you off |
| 15:58.22 | Katty | adnc: is it working? |
| 15:58.52 | adnc | they must have changed the bug tracking software |
| 15:59.20 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 15:59.40 | adnc | Katty, yes it is! but tell me why this happens: Mar 8 15:26:08 pbx asterisk[6945]: rc_avpair_new: unknown attribute 1490026597 |
| 15:59.42 | Katty | adnc: i guess that's also a possibility |
| 15:59.56 | Katty | adnc: because something isn't setup right. |
| 16:00.12 | adnc | Katty, and what do you think isn't setup right? |
| 16:01.10 | Katty | adnc: i would guess perhaps your rtp.conf |
| 16:01.15 | Katty | adnc: and port filtering |
| 16:01.19 | adnc | http://www.voip-info.org/boards/index.php?t=17493 |
| 16:01.25 | Katty | adnc: but that's like asking me why my car isn't running right |
| 16:01.27 | Kobaz | installs freebsd |
| 16:02.10 | Katty | watches the channel start an OS war. |
| 16:02.19 | Kobaz | heh |
| 16:02.29 | Kobaz | no wars intended... i love my teh linuxes |
| 16:02.31 | Katty | Kobaz: i have never used FreeBSD |
| 16:02.35 | Kobaz | i just wanna use zfs |
| 16:02.38 | Katty | Kobaz: but the lil demon is cute |
| 16:02.46 | Kobaz | i installed opensolaris but it doesn't support my sata controller i wanna use |
| 16:02.56 | Katty | :< |
| 16:02.58 | Katty | how sad. |
| 16:03.00 | Kobaz | i've used freebsd back in the day... version 3ish |
| 16:03.08 | Katty | REDHAT6 |
| 16:03.09 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 16:03.12 | Kobaz | haha |
| 16:03.15 | Katty | that's where i started. |
| 16:03.20 | Kobaz | i used redhat 4 5 and 6 |
| 16:03.24 | Katty | hottt |
| 16:03.26 | Kobaz | upgrades were always so painful |
| 16:03.31 | Kobaz | so i switched to debian |
| 16:03.35 | Katty | for awhile, i used Mandrake |
| 16:03.50 | Kobaz | i started with slackware 96 |
| 16:04.02 | Katty | mandrake was good for people who had no clue |
| 16:04.10 | adnc | WARNING[12890]: chan_iax2.c:11161 load_module: Unable to open IAX timing interface: No such file or directory |
| 16:04.16 | adnc | what is a timing interface? |
| 16:04.20 | Katty | facepalms |
| 16:04.27 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:04.29 | Katty | oh boy.... |
| 16:06.26 | adnc | and which file is it looking for |
| 16:06.40 | adnc | Katty, ??? |
| 16:06.48 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 16:06.50 | Katty | it's looking for your mom. |
| 16:07.05 | Tim_Toady | http://www.fallen-legion.eu/news/data/upimages/DoubleFacePalm.jpg |
| 16:07.09 | Katty | adnc: if you don't know what timing is for, you need to go read some more. |
| 16:07.35 | Katty | Tim_Toady: that's hott. |
| 16:07.45 | adnc | Katty, the problem is not the timing interface, the problem is the file or directory it is looking for |
| 16:07.56 | [TK]D-Fender | adnc: DAHDI timing source <- |
| 16:07.59 | adnc | Katty, didn't your mom tell this |
| 16:08.08 | ManxPower-work | adnc, no, the problem is the timing interface. Why are you using IAX trunking anyway? |
| 16:08.33 | adnc | ManxPower-work, i'm not using it, i'm just switching of all i do not need and preloaded |
| 16:08.50 | adnc | from the modules.conf |
| 16:08.56 | ManxPower-work | adnc, that message has nothing to do with that you are claiming. |
| 16:09.08 | adnc | ManxPower-work, what has it todo than? |
| 16:09.14 | Kobaz | okay |
| 16:09.34 | ManxPower-work | adnc, that message means you enabled trunk=yes in iax.conf |
| 16:10.11 | adnc | ManxPower-work, in iax.conf there is nothing enabled |
| 16:10.37 | ManxPower-work | adnc, that's not what the error message is indicating |
| 16:10.39 | Katty | adnc: k. |
| 16:11.05 | Katty | file: hungry :< |
| 16:11.21 | Katty | file: granola bar was not enough :< |
| 16:11.39 | leifmadsen | Katty: ya, my yogurt wasn't enough |
| 16:11.46 | Katty | leifmadsen: let's go get brunch. |
| 16:11.52 | leifmadsen | Katty: yay! |
| 16:11.59 | Katty | leifmadsen: WAFFLES |
| 16:12.06 | adnc | Katty, k. |
| 16:12.10 | leifmadsen | oh gawd I wish I had waffles |
| 16:12.22 | leifmadsen | then I could put peanut butter and syrup on them |
| 16:12.26 | Katty | eww. |
| 16:12.29 | leifmadsen | WHAT?! |
| 16:12.30 | Katty | actually. |
| 16:12.33 | leifmadsen | soooooooooooooo good |
| 16:12.34 | Katty | that does sound kinda good. |
| 16:12.37 | Katty | but! |
| 16:12.43 | Katty | can you imagine the fat content? >.< |
| 16:12.45 | leifmadsen | only if you like peanut butter / won't die from it |
| 16:12.57 | leifmadsen | Katty: we're already having waffles... so I presumed we could ignore that part :) |
| 16:12.58 | Katty | it'd be like 300% DV |
| 16:13.06 | Katty | don't be redonkulus. |
| 16:14.14 | *** join/#asterisk hurdman (~ngeek@ys.antredugeek.fr) |
| 16:14.16 | hurdman | hi |
| 16:14.21 | Katty | ohai |
| 16:14.24 | adnc | ManxPower-work, there is no trunk=yes in iax.conf |
| 16:14.49 | hurdman | i'm looking for something about the Dial(SS7/number:redirecting-number) command into chan_ss7 module |
| 16:14.52 | [TK]D-Fender | adnc: Feel free to ignore the message then |
| 16:15.30 | hurdman | i try to make a transfer without a dial, is-it possible with ss7 ? |
| 16:16.05 | ManxPower-work | adnc, I wish you the BEST of luck. |
| 16:16.10 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 16:16.19 | adnc | ManxPower-work, whats wrong? |
| 16:16.41 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 16:16.44 | Katty | he has hit the end of his rope. |
| 16:17.11 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.18 | *** join/#asterisk sourcode (~code@ppp-58-8-119-121.revip2.asianet.co.th) |
| 16:17.37 | ManxPower-work | sourcode, try using pastebin.ca instead of flooding the channel |
| 16:17.44 | ManxPower-work | ~pb |
| 16:17.44 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:18.09 | *** join/#asterisk Polysics (~Luca@host39-66-dynamic.30-79-r.retail.telecomitalia.it) |
| 16:18.26 | Polysics | ok, no sound on a normal SIP call, and no errors in the console |
| 16:18.29 | Polysics | any ideas? |
| 16:18.36 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 16:18.37 | Kobaz | rtp debug |
| 16:18.41 | ManxPower-work | Polysics, NAT and/or firewall |
| 16:18.44 | ChannelZ | Firewalls or incompatible codecs on both sides |
| 16:18.58 | Polysics | from what i can read, the codecs are ok |
| 16:19.04 | ManxPower-work | ChannelZ, that should cause the call to drop |
| 16:19.11 | Polysics | the clients are properly negotiating ulaw-alaw |
| 16:19.12 | Katty | Polysics: then check your firewall logs. |
| 16:19.16 | Kobaz | not necessarily |
| 16:19.28 | Kobaz | it could be allowing sip but not rtp |
| 16:19.33 | Kobaz | or not forwarding it properly |
| 16:19.41 | sourcode | ManxPower-work: Oh, i'm so sorry :) |
| 16:19.52 | Katty | i wonder if AT&T blocks rtp ports like they do 25 |
| 16:20.24 | Polysics | clients inside that network have used SIP so far |
| 16:21.18 | Katty | it's not polite to ask for advice and then ignore it. |
| 16:22.36 | ManxPower-work | Polysics, Is English not your primary language? |
| 16:22.47 | coppice | Katty: they could block the SIP port, that's well defined, but how could they block RTP? It can use any port |
| 16:22.50 | Polysics | i can't check the firewall logs right now |
| 16:23.04 | Polysics | so i was trying to bring what i have to the table |
| 16:23.06 | ManxPower-work | Polysics, come back when you can. |
| 16:23.11 | Katty | coppice: well i meant the 10k to 30k range. |
| 16:23.16 | coppice | or rather how could they block RTP by port? |
| 16:23.19 | Katty | coppice: but you are right. that would be a lot to block |
| 16:23.31 | coppice | RTP is not confined to that range |
| 16:23.36 | leifmadsen | or any range :) |
| 16:23.39 | Katty | coppice: no, it's not. but that's what i meant. |
| 16:23.43 | Polysics | ManxPower-work, could the fact that a user on both networks can use the classic "echo" app? |
| 16:23.45 | Katty | coppice: :P |
| 16:23.50 | Polysics | or is a call entirely something else? |
| 16:23.56 | coppice | they are more likely to block by inspecting the packet contents |
| 16:24.16 | Kobaz | hmm |
| 16:24.22 | Katty | coppice: probably. |
| 16:25.13 | coppice | Deep packet inspection - studying the full philosophical meaning is each packet |
| 16:25.36 | Katty | ^- what 5 year olds do to ketchup packets. |
| 16:26.16 | coppice | I thought they just tore off the corner and put them under a cushion |
| 16:26.26 | Katty | that's what you think. |
| 16:27.08 | Katty | next you're going to tell me mice dont' rule the world. |
| 16:28.00 | Kobaz | FreeBSD stan.home 8.0-RELEASE FreeBSD 8.0-RELEASE #0: Sat Nov 21 15:02:08 UTC 2009 |
| 16:28.03 | Kobaz | sexy |
| 16:28.10 | Katty | 08? |
| 16:28.17 | Katty | dude you're living in the past |
| 16:28.31 | Kobaz | ? |
| 16:28.54 | Kobaz | it's the newest release |
| 16:29.54 | *** join/#asterisk SystemT` (~omer@212.143.125.250) |
| 16:30.08 | SystemT` | Hey all |
| 16:30.39 | Tim_Toady | 'Sat Nov 21 15:02:08' is not the system time, its the compile date of the kernel |
| 16:31.07 | SystemT` | hey i have problem on Sip - incoming call |
| 16:31.43 | SystemT` | ? |
| 16:31.54 | ManxPower-work | ~ask |
| 16:31.55 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:32.14 | ManxPower-work | SystemT`, Your question was none of those things |
| 16:33.10 | Katty | considers making all questions on topic |
| 16:33.17 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:33.17 | Katty | decides that's a silly idea. |
| 16:33.30 | Katty | jaytee: oh, i've been meaning to pester you |
| 16:33.32 | Kobaz | all questions must be in the form of an answer |
| 16:33.35 | Katty | jaytee: lemme know when you are around. |
| 16:33.53 | jaytee | what? |
| 16:33.56 | Katty | -> |
| 16:34.11 | SystemT` | ~passting |
| 16:34.27 | SystemT` | ~pastebin |
| 16:34.28 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:38.01 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:38.48 | tivo1 | people are not very active in asterisknow and freepbx, If i start over does anyone have a good guide to follow? also what OS is most recomended.. |
| 16:39.32 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 16:40.02 | Tim_Toady | if u talking about asterisk now then its centos, if u talk in general whatever distro u feel more comfortable with |
| 16:40.26 | Tim_Toady | as for guide, try freebpx docomentation |
| 16:40.42 | ManxPower-work | There really isn't a "guide" Each Asterisk install is special. We don't know yet if that's special as in a "snowflake" or special as in "special olympics". |
| 16:40.57 | ManxPower-work | You should, however, read the Asterisk book. |
| 16:40.58 | ManxPower-work | ~book |
| 16:40.59 | 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 |
| 16:41.27 | tivo1 | ....If i reinstall something.. and need some help, I dont want to come back and have people say go to asterisknow or go to freepbx |
| 16:41.32 | [TK]D-Fender | ManxPower-work: You are not a beautiful and unique snowflake. You are the same decaying organic matter as everyone else. |
| 16:41.42 | tivo1 | have the book, not much help when freepbx renames things |
| 16:41.47 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:41.58 | tivo1 | then tells you not to edit files |
| 16:42.06 | ManxPower-work | tivo1, correct. Now you understand why we don't support FreePBx here. |
| 16:42.13 | ManxPower-work | If you want FreePBX help then find it somewhere else. |
| 16:42.26 | tivo1 | tis what im saying.. im about to give up on that... |
| 16:42.44 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 16:42.49 | ManxPower-work | Right. so now that you've given up on that, re-read the book. |
| 16:42.55 | tivo1 | and wanted to know if i start over.. what i should use (in order to have the most amount of help) abail |
| 16:42.58 | tivo1 | er availablre |
| 16:43.01 | tivo1 | sp |
| 16:43.04 | idespinner | is there a recommended GUI for asterisk at this point? |
| 16:43.11 | tivo1 | lo |
| 16:43.13 | ManxPower-work | idespinner, all GUIs suck. |
| 16:43.14 | [TK]D-Fender | idespinner: recommended for what? |
| 16:43.21 | idespinner | recommended for asterisk |
| 16:43.28 | idespinner | (other than switchvox of course!) |
| 16:43.32 | neurosys | ManxPower-work: :( Not always |
| 16:43.58 | idespinner | ManxPower-work, the Switchvox GUI is a very good one although its commercial |
| 16:44.08 | [TK]D-Fender | idespinner: Every GUI has its own strenths.... "for asterisk" is not a useful answer |
| 16:44.11 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 16:44.16 | ManxPower-work | idespinner, how long does it take for you to add an extension? |
| 16:44.24 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 16:44.39 | ManxPower-work | for me it's about 5 seconds for just the extension, maybe 20 seconds for extension and phone. |
| 16:44.58 | ManxPower-work | now, provosion 20 phones your way and my way. Yes. All guis sug. |
| 16:45.03 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 16:45.04 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:45.04 | ManxPower-work | suck |
| 16:45.06 | idespinner | ManxPower-work, you do have a point but there are still some stengths to be had with a GUI displaying extensions |
| 16:45.27 | idespinner | like when you have >25, editing a file can become a little more tedious |
| 16:45.35 | idespinner | >25 extensions |
| 16:46.24 | ManxPower-work | idespinner, If you mean for admins that are too stupid or lazy to learn Asterisk or do you mean users. I can see how a GUI for users miht be useful. |
| 16:46.55 | ManxPower-work | idespinner, um, what makes you think I'm editing a file. you forget it's easy to script stuff when not using a GUI |
| 16:48.37 | idespinner | ManxPower-work, I would more refer to admins that are to stupid to learn SSH and VI or NANO or whatever txt editor |
| 16:48.48 | idespinner | also yes, an end user GUI |
| 16:48.51 | idespinner | that aswell |
| 16:49.06 | outtolunc | go joe <G> |
| 16:49.32 | idespinner | but a script is an excellent idea, it would be nice is asterisk came with it |
| 16:49.34 | ManxPower-work | goes back to hating PHP sessions |
| 16:50.09 | [TK]D-Fender | idespinner: hard for * to "come with it" when ever install is unique <- |
| 16:50.16 | [TK]D-Fender | idespinner: You don't seem to get it |
| 16:50.25 | [TK]D-Fender | every* |
| 16:50.34 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:51.16 | Kobaz | ManxPower-work: php sessions are aweful |
| 16:51.34 | ManxPower-work | Kobaz, I can't even get them to work. |
| 16:51.48 | Kobaz | you need to do session_start() |
| 16:51.55 | ManxPower-work | Kobaz, correct. |
| 16:52.15 | Kobaz | the problem with php sessions is there's a giant mutex that locks each users own session data |
| 16:52.49 | ManxPower-work | Kobaz, and yet, if I don't use them, there's no way to tap into the FreePBX authentication framework. |
| 16:52.52 | Kobaz | and it's active for the life of the page request (or until you destroy the session) |
| 16:53.06 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-kyyexnzwrqprepqk) |
| 16:53.11 | Kobaz | ManxPower-work: check the temp dirs... make sure php can write to where it's supposed to store the sessions |
| 16:53.35 | ManxPower-work | Kobaz, Um, it's working just fine for every other page on this FreePBX box. |
| 16:54.13 | Kobaz | so what's the issue |
| 16:55.21 | ManxPower-work | Kobaz, I do a session_start() in my script an have none of the session info set by freePBX |
| 16:56.00 | Kobaz | you'll need to run whatever function the freepbx libs provide for getting the session data |
| 16:56.13 | Kobaz | just doing session_start() isn't going to get you anything other what's already in your session |
| 16:56.18 | ManxPower-work | Kobaz, I thought that's what session_start is for. |
| 16:56.23 | Kobaz | no |
| 16:56.27 | ManxPower-work | right. that's what I'm tRYING to get. |
| 16:56.55 | Kobaz | session_start() will lock the session mutex for the session id, start a new session if one doesn't exist, otherwise read the stored session data from the session files |
| 16:56.58 | Kobaz | and that's it |
| 16:57.02 | ManxPower-work | I hit the login page, it authenticates me, sets some session variables, then I load my page and the session vars are not there. |
| 16:57.28 | ManxPower-work | the FreePBX login page, so I doubt that is broken. |
| 16:57.57 | ManxPower-work | (and before anyone complains, we put up with offtopic rodent, pet, dog, and bad hair day topics) |
| 16:58.04 | Kobaz | so log in to freepbx.. stuff a print_r $_SESSION at the end of the login code... just for sanity |
| 16:58.40 | Kobaz | in your own script, dump $_SESSION after session_start... make sure you have the right session id |
| 16:58.56 | Kobaz | you may need to change the cookie variable or whatever... if freepbx does it's own thing |
| 16:59.30 | Kobaz | if you dont have the session variables you should... high probability is that your own script is starting a brand new session |
| 16:59.33 | ManxPower-work | Kobaz, Array ( [srce] => 4408 ) |
| 17:00.12 | ManxPower-work | Kobaz, As I understood it, the session should be set automatically |
| 17:00.24 | Kobaz | er, it's been a few months away from php... there's a function somewhere to get the session id |
| 17:00.43 | Kobaz | sessions will only start automatically if they are configured to do so in php.ini |
| 17:01.01 | ManxPower-work | Kobaz, I'll look it up. I guess I should not have expected FreePBX to use the perfectly good authentication framework the web server comes with. |
| 17:02.11 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:04.10 | *** part/#asterisk ManxPower-work (~manxpower@11.sub-75-216-117.myvzw.com) |
| 17:04.14 | *** join/#asterisk soman (~somnath@dsl-jklbrasgw1-fe19fb00-113.dhcp.inet.fi) |
| 17:07.41 | *** join/#asterisk MAbbas (~abbas4u@203.215.177.194) |
| 17:07.54 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:08.30 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:09.37 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 17:14.31 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 17:15.50 | *** join/#asterisk SystemT` (~omer@212.143.125.250) |
| 17:16.33 | *** join/#asterisk michael-i (~michael-i@p3EE29C6D.dip0.t-ipconnect.de) |
| 17:30.28 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 17:30.31 | wcselby | o/ |
| 17:33.52 | tivo1 | heh.. got it... thanks to a friend pointing me to the right log stuffs... chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.1.145 in the calltokenoptional list or setting user 1001 requirecalltoken=no |
| 17:34.16 | tivo1 | i can now call from my softphone to the s100 iax analog |
| 17:34.46 | *** join/#asterisk DMeloUK (~DominicMe@64.129.95.226) |
| 17:35.17 | *** join/#asterisk Tim_Toady (~moi@188.4.62.188.dsl.dyn.forthnet.gr) |
| 17:40.20 | *** join/#asterisk bsaxon_ (~bsaxon@12.68.234.174) |
| 17:41.27 | *** join/#asterisk iksik (tx@2002:5e17:a1a0::1) |
| 17:49.51 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 17:51.56 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:53.26 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:00.38 | Katty | shivers |
| 18:01.49 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:05.22 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 18:07.05 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 18:07.06 | Katty | hi wcselby :> |
| 18:07.15 | Katty | wcselby: how do you pronounce wcselby? |
| 18:07.23 | Qwell | Katty: just like it's spelled |
| 18:07.38 | Katty | Qwell: i wasn't talking to you mister smarty pants. |
| 18:07.45 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 18:07.47 | wcselby | w c selby |
| 18:08.10 | Katty | i think we should pronounces it wesselby |
| 18:08.22 | wcselby | first initial, middle initial, last name |
| 18:08.25 | Katty | wesselbee |
| 18:08.36 | Katty | mr selby? |
| 18:08.45 | wcselby | I'm highly creative, I know |
| 18:08.48 | Katty | hmm |
| 18:08.51 | Katty | sell bee? |
| 18:09.18 | wcselby | yes |
| 18:09.20 | Katty | k |
| 18:09.28 | Katty | hugs Qwell |
| 18:09.37 | Katty | Qwell: you were RIGHT |
| 18:09.41 | Qwell | see |
| 18:09.45 | Qwell | don't question the Qwell |
| 18:09.48 | Katty | k |
| 18:10.00 | wcselby | most people like to insert an "h" into my last name, i.e - "Shelby". But there's no H. |
| 18:10.22 | [TK]D-Fender | wcselby: the "h" is silent....like the "p" in swimming ;) |
| 18:10.29 | wcselby | lol |
| 18:11.10 | leifmadsen | "Welcome to our ool. Notice there is no "P" in it. Please keep it that way." |
| 18:11.58 | Katty | wcselby: i kept saying wesselbee in my head |
| 18:12.06 | Katty | wcselby: nuclear wessels. |
| 18:12.23 | [TK]D-Fender | Katty: Sure thing Chekov |
| 18:12.26 | Katty | :P |
| 18:12.32 | Katty | yay someone got the reference |
| 18:14.23 | wcselby | lol |
| 18:14.33 | wcselby | sorry at a client, only half paying attention to this window |
| 18:14.40 | *** join/#asterisk jpcansa (~jpbenavid@201.196.95.200) |
| 18:14.52 | wcselby | running 105 ldap queries to discover where certain faxes get emailed |
| 18:14.54 | wcselby | always fun |
| 18:15.00 | *** part/#asterisk DMeloUK (~DominicMe@64.129.95.226) |
| 18:15.08 | wcselby | sound like a good time for some lunch |
| 18:15.09 | wcselby | bbl |
| 18:15.54 | jpcansa | hi, any idea why my linksys spa962 is rejecting anonymous calls from my E1? got this message: Got SIP response 406 "Not Acceptable" back from xxx.xxx.xxx.xxx |
| 18:16.47 | Katty | KBAI |
| 18:16.48 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:21.09 | [TK]D-Fender | jpcansa: perhaps you should look at a CALL. |
| 18:22.46 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:24.39 | jpcansa | [TK]D-Fender: look at a call? |
| 18:28.02 | [TK]D-Fender | jpcansa: whats the point of looking at a response when you aren't looking at what it is a response TO? |
| 18:36.22 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:36.54 | jpcansa | [TK]D-Fender, i´m not english native and sometimes i dont get what you´re trying to tell in order to help |
| 18:39.15 | jpcansa | I´m looking at the calls, when i redirect calls to a spa962, anonymous calls gets rejected, but if i redirect incoming calls to a spa942, everything goes through |
| 18:39.38 | [TK]D-Fender | jpcansa: Redirect? How? With what? |
| 18:39.47 | [TK]D-Fender | jpcansa: FROM what? |
| 18:39.52 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:40.12 | jpcansa | redirect i mean DIAL from the dial plan |
| 18:40.40 | [TK]D-Fender | jpcansa: where is the ENTIRE call for us to look at? |
| 18:41.59 | leifmadsen | [TK]D-Fender: simmer down or you'll be put on timeout. Emphasis on words like you're using add absolutely nothing to the conversation. |
| 18:42.00 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:53.55 | idespinner | jpcansa, could you elaborate anonymous calls vs incoming calls? |
| 18:54.14 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 18:54.51 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:01.37 | *** join/#asterisk Z_God (~julius@130.89.234.30) |
| 19:05.00 | gr0mit | jpcansa, are you able to paste a sip trace for us to look at? there are soo many reasons why a call could be rejected |
| 19:06.08 | jpcansa | idespinner, yes, what i mean with "anonymous calls" is incoming calls with no caller ID, the rest of the incoming calls shows the phone number on the caller ID |
| 19:06.53 | idespinner | jpcansa, the only diference is callerid? thats odd |
| 19:07.33 | idespinner | you could add an if/then in the dialplan to change the callerid to something else if its 'anonymous' as a workaround |
| 19:07.51 | idespinner | i'm not familiar with the cisco phones too much unfortunatley |
| 19:09.37 | jpcansa | thanks anyway |
| 19:12.43 | LemensTS | . |
| 19:17.09 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 19:17.45 | [TK]D-Fender | LemensTS: I see your point |
| 19:17.55 | shamelessn00b | hi, I'm configuring a basic ss7 link on two ports of my sangoma card using a cross cable but I keep on getting MTP2 timer T2 timeout (failed to receive 'O', 'N', or 'E' after sending 'O'), initial alignment failed on link 'l1' |
| 19:18.11 | shamelessn00b | using chan_ss7 |
| 19:18.13 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:19.51 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 19:19.56 | LemensTS | :) |
| 19:21.55 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:22.44 | *** join/#asterisk Alagar (~Administr@122.164.37.13) |
| 19:23.03 | LemensTS | shamelessn00b: is it voice crossover cable and not data crossover cable? |
| 19:23.09 | Katty | hrmm |
| 19:23.24 | Katty | anyone know off the top of their head which runlevel the GUI is traditionally started in? |
| 19:23.26 | LemensTS | Katty: 66 rocks |
| 19:23.29 | Katty | or have i been smoking again |
| 19:24.24 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 19:26.07 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 19:30.05 | [TK]D-Fender | Katty: 5 |
| 19:31.13 | *** join/#asterisk caraconan (~caraconan@27.63.18.95.dynamic.jazztel.es) |
| 19:31.56 | caraconan | Hi. How can I get the 'keys' pressed (once the call has started) for the caller? Wich variables? |
| 19:32.43 | [TK]D-Fender | carrar: there is no "collective log" |
| 19:32.49 | [TK]D-Fender | caraconan: there is no "collective log" |
| 19:33.12 | caraconan | Sorry, I don't understand |
| 19:33.20 | [TK]D-Fender | caraconan: You can put some measure of logging in the dialplan yourself wherever you'd like |
| 19:33.32 | [TK]D-Fender | caraconan: There is no logging <- |
| 19:33.55 | caraconan | I only want to: 1) listen a menu 'press 1 to talk with 1001,... 2) wait 3) redirect to the extension that has marked the caller |
| 19:34.14 | [TK]D-Fender | caraconan: then that is a basic IVR |
| 19:34.41 | [TK]D-Fender | caraconan: "core show application waitexten" , "core show function TIMEOUT" |
| 19:34.44 | caraconan | ok, I will google. Thanks a lot |
| 19:35.26 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.165.106.dsl.dyn.forthnet.gr) |
| 19:38.05 | tivo1 | a friend and I are now trying to set up a iax2 trunk between us, from him I am seeing in /asterisk/full No registration for peer |
| 19:38.28 | tivo1 | can someone point me in the right direction? |
| 19:40.16 | caraconan | I define a 's' extension. Wich number should I dial to listen that menu? |
| 19:40.20 | wcselby | mmmmmm |
| 19:41.16 | *** join/#asterisk jo3sm1th (TalkToMe@m83-185-38-90.cust.tele2.se) |
| 19:41.33 | jo3sm1th | How can you select all text in Putty I want to save a .conf file for records but it doesnt allow me to drag with mouse |
| 19:42.45 | wcselby | just highlighting with the mouse |
| 19:42.53 | wcselby | or you can enable logging in putty |
| 19:42.59 | wcselby | that's what I prefer to do |
| 19:43.37 | wcselby | highlight with the mouse one screen, paste to a text file, move to the next screen, highlight the screen, paste to the text file, etc. |
| 19:43.43 | wcselby | is another option |
| 19:44.06 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:44.29 | *** join/#asterisk trentcreek (~kvirc@red1.cs.panam.edu) |
| 19:46.25 | *** join/#asterisk dzup (dzup@support.team.at.shellium.org) |
| 19:46.33 | *** join/#asterisk Warp4 (~rwolfe@cpe-69-204-101-172.buffalo.res.rr.com) |
| 19:47.33 | shamelessn00b | LemensTS: yes |
| 19:47.37 | shamelessn00b | voice crossover |
| 19:47.58 | shamelessn00b | I've been using it for testing previously configured E1 links |
| 19:49.15 | trentcreek | Kobaz: I was reading up on your info. I see there are Asterisk proxys available. What ablut using that instead? One that looks promising is AsManProxy. I understand FOP has one, but the docs are very limited. |
| 19:49.17 | [TK]D-Fender | joEver consider actually copying the files off.... |
| 19:50.21 | *** join/#asterisk fofware (~chatzilla@190.30.113.46) |
| 19:58.36 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 20:11.13 | Katty | i have hiccups )=< |
| 20:12.20 | *** join/#asterisk dobry (~d@95.111.7.95) |
| 20:14.52 | *** join/#asterisk pyite_ (~dschreibe@unaffiliated/pyite) |
| 20:20.10 | jaytee | Katty, try swallowing a teaspoon of sugar. I add a little lemon juice to it. |
| 20:20.32 | Katty | they're gone now (= |
| 20:21.01 | Katty | what all is in extra sounds 1.5.10 |
| 20:21.04 | Katty | er 1.4.10 |
| 20:21.09 | Katty | anything useful? |
| 20:21.27 | Katty | ARE THERE MORE MONKIES?! |
| 20:21.41 | russellb | download it and diff against the old sounds.txt |
| 20:21.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:21.49 | jaytee | there's a couple funny ones in there |
| 20:22.12 | Katty | k, i'll get it |
| 20:22.25 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 20:22.46 | jaytee | there's an "all your base belong to us" one |
| 20:23.22 | jaytee | blue-eyed-polar-bear.ulaw? |
| 20:25.11 | Katty | oh noes connection timed out :< |
| 20:26.08 | bn-7bc | 'Does enyone know if Asterisk 1.6.x supports ipv6? (Have tried google but it gives conflikting answers) |
| 20:26.44 | caraconan | Hi. There's a way to, using variables, dial the extension? http://pastebin.com/SJXscQfF doesn't works |
| 20:26.50 | tzafrir | bn-7bc, no |
| 20:27.18 | tzafrir | there's a separate branch with support for that, but I have no idea of any immediate plan to merge it |
| 20:27.27 | Katty | jaytee: is the package just a bunch of audio files? |
| 20:27.35 | Katty | jaytee: or does it have to make something or other |
| 20:27.57 | Tim_Toady | i think ipv6 is on the todo list for 1.8 |
| 20:28.20 | jaytee | no, it's just audio files. the extras files I've downloaded in the past were available as separate tarballs with either ulaw, wav or gsm formats |
| 20:28.23 | Katty | jaytee: ehnn i'm sure there's a readme |
| 20:28.36 | Katty | jaytee: k |
| 20:28.55 | bn-7bc | ok is there any public updated todolist/ptogress on 1.8 |
| 20:28.59 | Tim_Toady | bn-7bc http://www.asteriskv6.org/ |
| 20:29.08 | bn-7bc | hanks for the info |
| 20:34.01 | shamelessn00b | can anyone help me out inbasic ss7 configuration |
| 20:34.03 | shamelessn00b | ? |
| 20:35.04 | Katty | i just learned what a salivary stone is. |
| 20:35.07 | Katty | be afraid. be very afraid. |
| 20:39.40 | *** join/#asterisk sulex (~sulex@host-78-14-168-79.cust-adsl.tiscali.it) |
| 20:40.23 | *** join/#asterisk simplydrew_ (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 20:46.52 | Deeewayne | Katty, I'm afraid |
| 20:47.29 | Deeewayne | I don't want anything obstructing my Wharton's Duct |
| 20:47.57 | Qwell | hey, hey, hey, keep it PG-13 |
| 20:48.01 | Qwell | :p |
| 20:48.05 | Katty | giggles |
| 20:48.09 | jaytee | hehehe |
| 20:48.10 | *** join/#asterisk TimeRider (~steve@5ac318a4.bb.sky.com) |
| 20:48.13 | Katty | i'll PG your 13 in a minute! |
| 20:48.29 | Qwell | O.O |
| 20:48.35 | russellb | O.o |
| 20:48.38 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 20:49.37 | *** join/#asterisk GameGamer43 (~GameGamer@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 20:50.32 | Deeewayne | russellb, you have a lazy eye |
| 20:50.38 | russellb | ha |
| 20:50.43 | russellb | stop making fun of me :-( |
| 20:50.50 | Deeewayne | sorry :-( |
| 20:50.57 | Katty | so i'm doing my wanpipe setup, and it's asking me about signalling type. do i want zaptel/dadhi PRI CPE or Sangoma SMG PRI CPE |
| 20:51.06 | russellb | frowns some more |
| 20:51.22 | Katty | turn that frown upside down cupcake. |
| 20:51.59 | Qwell | Do you frost the bottom of an upside down cupcake? |
| 20:52.16 | Katty | i suppose you could if you wanted to. |
| 20:52.26 | Katty | i don't think there's any sort of law against it |
| 20:54.57 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:381a:3c4f:2ac0:762d) |
| 20:54.58 | cusco | hi |
| 20:55.09 | cusco | Im looking into hidding my caller ID |
| 20:55.30 | cusco | when doing PRI calls |
| 20:55.32 | cusco | over dahdi |
| 20:57.26 | *** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl) |
| 20:57.36 | spenguin[work] | hey Katty :> |
| 20:57.40 | niekvlessert | hello, what if keepstats=yes in queues.conf doesn't work? |
| 20:57.40 | *** join/#asterisk ManxPower-work (~manxpower@11.sub-75-216-117.myvzw.com) |
| 20:57.56 | Katty | cusco: hiding it? |
| 20:58.02 | Katty | cusco: you mean like pretending you're dell? |
| 20:58.05 | Katty | hugs spenguin[work] |
| 20:58.22 | spenguin[work] | :> |
| 20:58.30 | Katty | spenguin[work]: HERROES DER PENGUIN |
| 20:58.36 | Katty | i must listen to smooth operator now |
| 20:58.39 | *** join/#asterisk smike (~mike.alic@74.255.249.66) |
| 20:58.43 | spenguin[work] | lol |
| 20:59.22 | cusco | Katty: lol, no. I mean like in gsm I would use #31# and *31# |
| 20:59.36 | Katty | that does not parse. |
| 20:59.39 | Katty | please try again. |
| 21:00.02 | cusco | so it shows a withold number |
| 21:00.09 | cusco | or private number |
| 21:00.26 | cusco | I tried hidecallerid=yes in chan_dadi-conf |
| 21:00.28 | Katty | cusco: interesting. |
| 21:00.30 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-wugnlwlfhjnjwfsd) |
| 21:00.58 | spenguin[work] | hows the critters Katty |
| 21:01.08 | spenguin[work] | and the critter link pls k thx |
| 21:01.16 | spenguin[work] | itll keep me awake |
| 21:01.23 | Deeewayne | ~crittercam |
| 21:01.24 | infobot | i heard crittercam is Katty's live broadcast of The Nut House @ http://ustre.am/8H5d and The Fuzzy Ferret Flat @ http://ustre.am/bEBU |
| 21:01.33 | spenguin[work] | thx |
| 21:01.40 | Katty | i think they're offline at the moment |
| 21:01.52 | spenguin[work] | how could you do that to em! |
| 21:02.57 | spenguin[work] | you promised them fame. |
| 21:03.03 | cusco | would anybody point me in the right direction? |
| 21:03.27 | Katty | spenguin[work]: :P |
| 21:04.59 | cusco | telco says they have no prefixes to hide the callerid, but if I do it at our telephone central, theirs should interpret it |
| 21:05.02 | cusco | :/ |
| 21:05.38 | smike | Anyone know how to get the Polycom 650 to contact boot server when using dhcp? It only works with a static ip. Seems like a timeout issue. |
| 21:06.00 | Naikrovek | yes |
| 21:06.08 | Naikrovek | what kind of dhcp server are you using |
| 21:06.14 | Naikrovek | linux or windows or other |
| 21:06.51 | smike | Windows |
| 21:07.14 | Naikrovek | smike: set dhcp option 66 to point to the boot server |
| 21:07.31 | Naikrovek | smike: like "ftp://user:pass@host" |
| 21:12.14 | ManxPower-work | simply setting it to the ip of the boot server will default to FTP w/ the default user/pass |
| 21:12.49 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 21:14.51 | shamelessn00b | trying to configure ss7 for the last 7 hours |
| 21:15.00 | shamelessn00b | getting hopeless now |
| 21:15.03 | shamelessn00b | :( |
| 21:15.08 | hluesea | hello i have a problem on asterisk cdr. Cdr is not update the incoming calls. I looked into cdr.conf and cdr_mysql.conf's but i can't find any solutions |
| 21:15.21 | shamelessn00b | Full dahdi input buffer detected |
| 21:15.30 | hluesea | have anybody know the updated to incoming calls to cdr db ? |
| 21:15.40 | shamelessn00b | Empty Dahdi output buffer detected |
| 21:15.52 | shamelessn00b | I get a horde of these messages |
| 21:16.02 | shamelessn00b | thats all that I get |
| 21:16.55 | ManxPower-work | shamelessn00b, almost nobody here uses SS7. It it usually only available outside of USA/Canada |
| 21:17.44 | shamelessn00b | :( |
| 21:17.59 | Naikrovek | yeah why are you playing with it |
| 21:18.04 | Naikrovek | out of curiosity |
| 21:18.40 | *** join/#asterisk miamiseb (~a@208.76.35.132) |
| 21:19.02 | ManxPower-work | shamelessn00b, where are you located? |
| 21:19.04 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:19.14 | shamelessn00b | pakistan |
| 21:19.39 | ManxPower-work | shamelessn00b, You are SURE you have an SS7 connection and not just a standard EuroISDN PRI? |
| 21:20.04 | ManxPower-work | (I understand you are not in EuroLand, but the standard is used outside of Europe) |
| 21:20.07 | miamiseb | Anyone used app_broadcast successfully before? Just want some comment on the stability before I go loading modules |
| 21:20.18 | shamelessn00b | pretty much, I'm trying to configure a PRI card BOTH as a master and a slave (seperate ports ofc) |
| 21:20.34 | smike | Thanks Naikrovek, it worked! |
| 21:20.36 | ManxPower-work | shamelessn00b, You do not have a "PRI card" |
| 21:21.11 | shamelessn00b | I meant E1 |
| 21:21.33 | smike | One more question. Anyone know what this means when running the dadhi_hardware command: Unrecognized garbage ' Master ' in WCT1/0/1 |
| 21:22.09 | Naikrovek | smike: no sweat |
| 21:23.06 | smike | So just ignore? |
| 21:24.02 | Naikrovek | oh i was saying "no sweat" to your thank you |
| 21:24.06 | Naikrovek | i don't know DAHDI |
| 21:25.03 | miamiseb | as far as popups, right now I'm trying to get asterisk desktop manager, is that the best solution or is there something else you guys might have had success with. As an aside, its important for me to pass the DIALED number in the url, not the CALLING number (caller id and name) |
| 21:25.10 | cusco | maybe dahdi thinks your card is garbage heh |
| 21:25.15 | hluesea | any one know the cdr incoming fail ? :) |
| 21:26.12 | Katty | maybe dahdi is out for LUNCH |
| 21:26.33 | cusco | miamiseb: callerid can show you the dialed number |
| 21:26.39 | cusco | I think it is callerid(dnid) |
| 21:27.22 | cusco | or can't you just use ${EXTEN} ? |
| 21:34.29 | cusco | and... while Im not finding out how to hide our PRI number identification |
| 21:34.45 | cusco | how can I tell a call to go out trough some other number.. |
| 21:34.55 | cusco | I mean, e have 100 numbers from telco |
| 21:35.05 | cusco | but our calls always go trough the first one |
| 21:35.22 | cusco | I would like to be able to make some calls trough other numbers |
| 21:35.35 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.4) |
| 21:37.08 | *** join/#asterisk Jaelae (~Jaelae@4133247a.cst.lightpath.net) |
| 21:37.37 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 21:38.20 | Jaelae | asterisk would essentialy replace whatever software system i have managing my phone system correct? call routing / assigning jacks to numbers, etc. ? |
| 21:39.11 | Qwell | Jaelae: pretty much |
| 21:39.18 | miamiseb | cusco: forcing callerid? why would you want to though? |
| 21:40.03 | miamiseb | asterisk desktop manager just crashed, and only launched the popups 2 out of 3 times... |
| 21:40.04 | miamiseb | bah |
| 21:40.04 | bjhaid | i am a complete newbie, i am creating my new .conf files, do i save them in the etc folder on my ubuntu machine, and if so how do i get that done |
| 21:40.25 | Jaelae | Qwell - thanks just clarifying. we have an avaya system and trying to see what is involved and feature benefits of Asterisk compared to the cost to upgrade parts of this software |
| 21:41.27 | miamiseb | cusco: and yeah, I have it under MACRO_EXTEN, but not all the popup software allows you to popup whatever variable you want, most have only the predefined uniqueid, and calling name and number |
| 21:42.04 | miamiseb | asterisk desktop manager added CUSTOM, and I set the correct variable in my dialplan, but I need something reliable, not the crashing and non poping up I've seen with ADM |
| 21:42.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:42.17 | miamiseb | I tried vtiger too, but by the time the popup arrives the call is at voicemail |
| 21:43.50 | miamiseb | I'm hoping I don't have to get unlazy and just write something that interfaces with the AGI myself |
| 21:44.18 | *** join/#asterisk DocAwesome (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:44.18 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 21:44.54 | miamiseb | Guess I'll try jabber first, since that modules is already built |
| 21:45.25 | *** join/#asterisk fink (~guest@static-162-84-93-164.fred.east.verizon.net) |
| 21:46.31 | *** join/#asterisk |AnToS| (~31749@93-44-103-101.ip96.fastwebnet.it) |
| 21:54.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:55.22 | Naikrovek | ~thebook |
| 21:55.23 | infobot | thebook is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 21:55.26 | Naikrovek | bjhaid: ^^^^^^^^^^^^^ |
| 21:55.35 | *** join/#asterisk DigitalFlux (~DigitalFl@unaffiliated/digitalflux) |
| 21:56.09 | DigitalFlux | Guys |
| 21:56.15 | DigitalFlux | i have 2 asterisk boxes |
| 21:56.24 | DigitalFlux | one of them has the PRI interface |
| 21:56.29 | niekvlessert | hey guys, keepstats=yes does not exist yet in 1.4.26.2, how can i make sure my stats are saved when reload? I can't I guess? |
| 21:56.56 | bjhaid | Naikrovek, i have the book, do not know what to do after editing the extensions.conf file, because it doesnt permit me to save on my ubuntu machine into the etc/asterisk folder |
| 21:57.00 | DigitalFlux | i need any incoming call that is caught by the 1st box connected to the PRI to get routed over a SIP trunk to the 2nd asterisk box |
| 21:57.03 | ManxPower-work | niekvlessert, 1.4.x NEVER gets new features |
| 21:57.10 | DigitalFlux | how can i do that ? |
| 21:57.23 | fink | sorry if this is a stupid question: if i want to use an aastra 620D DECT with my asterisk install, i need to buy a dect base station as well? |
| 21:57.35 | niekvlessert | ManxPower-work: i know that, so I guess this info can't be saved? |
| 21:58.00 | ManxPower-work | niekvlessert, no idea. I was only commenting on your use of "yet" |
| 21:58.22 | spenguin[work] | are there any voip certs, any of you recommend? |
| 21:58.33 | *** join/#asterisk Jhirley_ (~Jhirley@adsl-161-67-247.mia.bellsouth.net) |
| 21:58.37 | spenguin[work] | also asterisk should come up with some voip certs :p |
| 21:58.42 | [TK]D-Fender | fink: Yes |
| 21:58.48 | Qwell | voip certs? |
| 21:59.04 | niekvlessert | well that sucks :( |
| 21:59.05 | spenguin[work] | Qwell: well yeah just for the paper work |
| 21:59.17 | niekvlessert | is there an option to remove several things from a reload command? |
| 21:59.29 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 21:59.34 | fink | [TK]D-Fender: cool thanks. i guess what i m looking for is a wireless handset which can communicate directly with the asterisk server |
| 21:59.36 | niekvlessert | so that everything gets reloaded and some stuff doesnt |
| 21:59.41 | fink | am i looking for wifi phone? |
| 22:00.07 | spenguin[work] | if its a wifi phone it better have awesome battery life |
| 22:00.18 | [TK]D-Fender | fink: DECT w/ SIP base is preferable |
| 22:00.21 | Qwell | fink: go with DECT |
| 22:00.52 | fink | ok, thanks |
| 22:01.10 | fink | are there any IAX DECT phones? |
| 22:02.07 | [TK]D-Fender | fink: Nope |
| 22:02.13 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 22:03.11 | fink | [TK]D-Fender: thanks |
| 22:04.36 | DigitalFlux | Can any body help me with this situation guys ? |
| 22:05.02 | niekvlessert | reset queue is not in 1.4.... too bad |
| 22:05.22 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 22:05.36 | niekvlessert | DigitalFlux: is the call on the first box yet? |
| 22:06.03 | DigitalFlux | niekvlessert: yes it arrives at the first box normally |
| 22:06.20 | niekvlessert | DigitalFlux: and then what are you trying? |
| 22:06.22 | [TK]D-Fender | DigitalFlux: Just like you would any otehr ITSP |
| 22:06.23 | DigitalFlux | niekvlessert: i need to forward it over a trunk to the 2nd box |
| 22:06.37 | niekvlessert | i get that... but what are you trying to get it there? |
| 22:06.41 | [TK]D-Fender | DigitalFlux: Peers on each side. Dial it over |
| 22:08.55 | DigitalFlux | [TK]D-Fender: OK, can you give me a hint on how to do that ? |
| 22:08.59 | niekvlessert | DigitalFlux: Dial(SIP/123@host) |
| 22:09.35 | niekvlessert | http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ |
| 22:10.27 | DigitalFlux | niekvlessert: OK, now i get it, will try it now |
| 22:10.31 | DigitalFlux | Thanks guys |
| 22:11.07 | cusco | miamiseb> cusco: forcing callerid? why would you want to though? --- Well... say We have 100 numbers from telco.. some of them are different services.. We would like to use the same number that the costumer used to call him back |
| 22:11.13 | cusco | so... |
| 22:11.32 | cusco | how would I say that an outbound PRI call would take some other number? |
| 22:11.50 | cusco | setting callerID doesn't seem to sufice |
| 22:12.45 | Kobaz | http://www.wonderhowto.com/wonderment/build-tiny-living-room-in-pc-0113462/ |
| 22:13.26 | adnc | when i use such a set in a macro: exten => s,1,set(CALLFILENAME=${STRFTIME(${EPOCH},GMT+1,%d%m%Y-%H%M)}.${CALLERID(num)}.${MACRO_EXTEN}) the variable ${MACRO_EXTEN} or even ${EXTEN} gets the value s, not the extension-number, what am i doing wrong? |
| 22:14.15 | Kobaz | pokes Katty |
| 22:14.16 | [TK]D-Fender | cusco: It does. Feel free to show us your failed attempt so we can tell you where it went wrong |
| 22:14.37 | [TK]D-Fender | adnc: it IS "s", just like it says |
| 22:14.46 | Katty | squeaks |
| 22:14.52 | Kobaz | ^^ url |
| 22:14.58 | Katty | goes up |
| 22:15.07 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 22:15.14 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.4) |
| 22:15.21 | Katty | :>>>>>>>>>>>>> |
| 22:15.22 | adnc | [TK]D-Fender, i understand, so it is not possible |
| 22:15.24 | Katty | i'm movin in! |
| 22:15.28 | Kobaz | hrh |
| 22:15.30 | Kobaz | heh |
| 22:15.49 | Kobaz | it even has a gumball machine |
| 22:15.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:16.59 | niekvlessert | cusco: did you look in chan_dahdi.conf? |
| 22:17.02 | [TK]D-Fender | adnc: You should be looking to pass a better number and looking at where those vars you were using even come from. |
| 22:17.34 | niekvlessert | cusco: usecallingpres, usecallerid etc |
| 22:17.37 | raden_work | morning Katty :) |
| 22:18.05 | Katty | herroes der raden |
| 22:18.07 | Katty | hugs raden |
| 22:21.01 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 22:22.40 | *** join/#asterisk youngproguru (~youngprog@smtp.deltasoniccarwash.com) |
| 22:24.36 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 22:24.50 | rocksfrow | hey guys, looking for some recommendations on phones for a calling center |
| 22:24.55 | rocksfrow | also, headsets |
| 22:24.57 | Kobaz | polycom |
| 22:25.01 | beek | polycom |
| 22:25.03 | rocksfrow | does anybody have experience? |
| 22:25.05 | Kobaz | plantronics |
| 22:25.10 | beek | plantronics |
| 22:25.13 | Kobaz | haha |
| 22:25.13 | rocksfrow | yes, polycom has been my favorite so far |
| 22:25.15 | *** join/#asterisk simplydrew_ (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 22:25.24 | beek | You mean someone else makes phones? |
| 22:25.26 | rocksfrow | lol..very original beek! |
| 22:25.32 | rocksfrow | haha.. |
| 22:25.46 | beek | rocksfrow: seriously, polycom. |
| 22:25.47 | [TK]D-Fender | rocksfrow: with Plantronics M22 amps, and in my case H261N Polaris quick-connect headsets |
| 22:26.00 | beek | Hello [TK]D-Fender. |
| 22:26.04 | beek | hugs Katty |
| 22:26.15 | Katty | yay it's BEEK! |
| 22:26.18 | rocksfrow | [TK]D-Fender, those wireless? |
| 22:26.18 | Katty | hugs beekster |
| 22:26.22 | rocksfrow | (the headsets) |
| 22:26.32 | [TK]D-Fender | rocksfrow: nope |
| 22:26.41 | rocksfrow | any favorite stores? |
| 22:26.49 | rocksfrow | voipsupply, voiplink, etc..? |
| 22:27.11 | Katty | telephonydepot |
| 22:27.12 | rocksfrow | [TK]D-Fender, do you find reps getting irritated with the cords? |
| 22:27.22 | Katty | DOT COM |
| 22:28.07 | Katty | my extensions.conf is messy |
| 22:28.11 | Katty | vows to clean it up a bit |
| 22:29.25 | rocksfrow | beek, any polycoms in particular? |
| 22:29.41 | rocksfrow | i'm looking for something to replace the four-line PoE grandstreams i have |
| 22:29.41 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 22:29.43 | murdock_ut | Howdy all. |
| 22:32.33 | [TK]D-Fender | rocksfrow: Nope... audio quality = king |
| 22:32.56 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 22:33.09 | *** join/#asterisk youngproguru (~youngprog@smtp.deltasoniccarwash.com) |
| 22:33.38 | bjhaid | i am a newbie, trying to create my extensions.conf file on my ubuntu machine, but it would not allow me save into the etc/asterisk folder can anyone please help me out on what to do |
| 22:33.57 | upb | su - |
| 22:33.59 | rocksfrow | sudo |
| 22:34.00 | rocksfrow | not su- |
| 22:34.02 | rocksfrow | he's on ubuntu |
| 22:34.02 | carrar | contact your system admin |
| 22:34.04 | rocksfrow | you cant' switch to root |
| 22:34.05 | upb | ah yes :) |
| 22:34.06 | rocksfrow | you need to use sudo |
| 22:34.21 | beek | rocksfrow: Sure you can: sudo su - |
| 22:34.21 | murdock_ut | Question, after I park a call is normal for the dial plan to continue execution from where the dial command was given that called the device to begin with? |
| 22:34.24 | rocksfrow | sudo vi extensions.cnf |
| 22:34.27 | rocksfrow | beek, no you can't |
| 22:34.33 | rocksfrow | beek, not unless you override the unknown root password |
| 22:34.39 | rocksfrow | by default, the root pass is secret |
| 22:34.45 | beek | Au contraire... I do it frequently. |
| 22:34.48 | rocksfrow | you do everything via sudo |
| 22:34.54 | rocksfrow | ..... |
| 22:34.55 | beek | And I have NOT changed the root password. |
| 22:35.03 | rocksfrow | whaaat? |
| 22:35.04 | *** join/#asterisk youngproguru (~youngprog@smtp.deltasoniccarwash.com) |
| 22:35.05 | miamiseb | beek: am man who likes a full login shell, I like that |
| 22:35.06 | rocksfrow | what version of ubuntu? |
| 22:35.08 | rocksfrow | standard install? |
| 22:35.17 | miamiseb | damn people with just sudo su that never change directory and proccess profiles |
| 22:35.28 | beek | miamiseb: 'sudo' reminds me too much of "simon says". |
| 22:35.28 | miamiseb | s/am/a |
| 22:35.29 | rocksfrow | i know by default ubuntu uses sudo..and does not allow root access |
| 22:35.39 | bjhaid | 9.10 |
| 22:36.09 | rocksfrow | perhaps beek can assist you in configuring root access |
| 22:36.53 | bjhaid | i have configured root, but root access is strictly to terminal, cant save on gedit |
| 22:37.02 | jaytee | sudo su elevates the current user, if they are a member of the sudoers group to root level priviledges |
| 22:37.03 | beek | rocksfrow: Really... all I ever do on Ubuntu is 'sudo su -' and I get a root shell. |
| 22:37.03 | miamiseb | bjhaid: and what does "sudo su -" do for you? |
| 22:37.05 | jaytee | same thing |
| 22:37.08 | miamiseb | ooo |
| 22:37.17 | LemensTS | . |
| 22:37.29 | beek | except that root's profile is read... path, etc. |
| 22:37.35 | bjhaid | i am on windows right now, so cant check |
| 22:37.37 | miamiseb | bjhaid: launch an xterm and sudo su -, then use cli editor, or launch gedit elevated |
| 22:37.45 | LemensTS | Any reason i could call from my sip phone to my cell phone and get CID, but when i call someone with att land line i dont? |
| 22:37.52 | miamiseb | beek: lmao, I know, I just commented on it doing those things, I was asking if it gave him errors |
| 22:37.53 | miamiseb | lol |
| 22:37.53 | [TK]D-Fender | bjhaid: sudo gedit <--- it ain't Raw-Cat Sigh Hence |
| 22:37.58 | rocksfrow | or you could just: 'sudo vi extensions.conf' |
| 22:38.05 | rocksfrow | sorry, sudo gedit extensions.conf |
| 22:38.26 | LemensTS | I mean call someone from my sip phone to att land line they get unkown cid from me... |
| 22:38.28 | bjhaid | miamiseb: how do i lauch xterm and use cli editor |
| 22:38.33 | rocksfrow | theres no reason to switch to root if you just need to edit a file, thats whats sudo is for :-p |
| 22:39.15 | beek | unless you get tired of typing sudo for every f***king thing you want to do. |
| 22:39.21 | rocksfrow | haha |
| 22:39.24 | rocksfrow | beek, agreed. |
| 22:39.38 | miamiseb | yeah, the principle of least privilege is great, for other people, I want less barriers to my rm -rf / |
| 22:40.03 | miamiseb | honestly though, on most system I only run one command as a NON-uid 0 user, and thats sudo su - |
| 22:40.25 | beek | Which is why I prefer learning to READ MY COMMAND LINE before hitting enter instead of getting muscle memory typing sudo as a prefix to each command. |
| 22:40.44 | beek | miamiseb: I'm with you. |
| 22:40.56 | rocksfrow | beek, i guess it depends on how many people you're giving access to |
| 22:41.11 | miamiseb | bjhaid: your terminals should be in your utilities menu under gnome |
| 22:41.28 | beek | rocksfrow: Now if you're talking a mutli-user system with other people having sub-admin rights, that's a horse of another color. |
| 22:41.31 | bjhaid | thanks everyone, would see how well this would help |
| 22:41.34 | rocksfrow | its applications -> accessories -> terminal |
| 22:42.22 | miamiseb | a horst colored selinux? |
| 22:42.33 | miamiseb | s/horst/horse |
| 22:42.35 | rocksfrow | do you guys know...the only diff. between the polycom 550 and 560 is the gigE? |
| 22:42.35 | *** join/#asterisk JackKingOff (hector@dipsy.tch.org) |
| 22:43.05 | jaytee | what an awesome nick! |
| 22:43.09 | rocksfrow | isn't gigE a waste? how can you even benefit off the extra bandwidth? |
| 22:43.20 | *** join/#asterisk netpro25_ (~mmanning@64-238-176-105.ksg.apt.gru.net) |
| 22:43.20 | miamiseb | rocksfrow: if you've got a computer chained... |
| 22:43.32 | rocksfrow | miamiseb, ohhh, duh..didn't think about that |
| 22:43.42 | JackKingOff | thx jaytee |
| 22:45.57 | [TK]D-Fender | sofa king hill airy us |
| 22:46.05 | miamiseb | we todd did |
| 22:47.26 | miamiseb | I have a feeling I'm just going to end up writing an agi interface for screen popping, what a damn pain |
| 22:47.29 | rocksfrow | lol |
| 22:50.15 | miamiseb | trying FOP as a last ditch effort |
| 22:50.27 | *** join/#asterisk ibercom (~5f100c74@gateway/web/freenode/x-moldlfzykrnoinuk) |
| 22:50.38 | jaytee | I'm a Dapper Dan man, don't give me no FOP |
| 22:52.00 | beek | jaytee: still starting next week? |
| 22:52.10 | jaytee | beek, afaik |
| 22:52.16 | beek | Cool? What's the gig? |
| 22:52.29 | jaytee | a mix of IT support and coding |
| 22:52.47 | beek | Any Asterisk? |
| 22:53.05 | ibercom | Hello, anybody know how shutdown a primary E1 port ? I don´t want restart asterisk or dahdi, Is it possible ? |
| 22:53.32 | jaytee | not to start but my friend and I have talked about branching out the core business to include some telephony integration |
| 22:53.38 | Kobaz | i wish polycom made a cheaper phone with more buttons |
| 22:54.12 | jaytee | wish in one hand, shit in the other....see which fills up first |
| 22:54.25 | beek | jaytee: :D |
| 22:54.37 | Kobaz | ibercom: edit dahdi/system.conf and take out the configs for the port you want to drop... dahdi_cfg -vv (you may drop current calls though) |
| 22:56.59 | ibercom | No problem, I want down one E1 port, is it similar to remove the wire ? |
| 22:57.08 | Kobaz | yeah |
| 22:57.11 | Kobaz | similar |
| 22:57.35 | Kobaz | not exactly the same... your low level drivers will still provide clocking and etc if it's say, a sangoma board |
| 22:57.46 | Kobaz | but you will drop the d channel |
| 22:59.33 | ibercom | I think that it is enough. I have a digium board (4 PRI). |
| 23:00.50 | ibercom | If the config don't change, dahdi_cfg doesn't nothing. Ok ? |
| 23:01.11 | Kobaz | i don't remember exactly... but yeah i think so |
| 23:01.27 | ibercom | Thank you. |
| 23:05.03 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 23:08.55 | *** join/#asterisk Jhirley (~Jhirley@adsl-3-198-154.mia.bellsouth.net) |
| 23:11.40 | *** join/#asterisk pentanol (~Unknown@91.195.60.231) |
| 23:17.29 | *** join/#asterisk nix8n82 (~nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 23:26.27 | hipitihop | does asterisk have a dbus interface ? |
| 23:28.06 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 23:29.21 | russellb | no |
| 23:31.10 | tivo1 | Registration of 'fnc-user' rejected: 'Registration Refused' from: x.x.x.x |
| 23:31.12 | murdock_ut | russellb: Isn't the Dial application supposed to end all dialplan execution if the call is answered? |
| 23:31.17 | tivo1 | someone know what im doing wrong here? |
| 23:32.39 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 23:35.05 | kfife | Question: If I am IAX trunking between 2 Asterisk instances, and connecting to SIP endpoints on BOTH ends of the call, can I let the ENDPOINTS do the jitterbuffering, or must the iax-trunk do the jitterbuffering? |
| 23:35.45 | kfife | Certainly the sip endpoints would do their OWN jitterbuffering no matter what I do, but can I eliminate the IAX jitterbuffer to decrease latency? |
| 23:36.38 | *** join/#asterisk Jhirley_ (~Jhirley@adsl-161-69-109.mia.bellsouth.net) |
| 23:37.09 | beek | kfife: Wouldn't the jitter cause problems on the conversion from IAX to SIP? |
| 23:38.54 | kfife | beek: that's kinda gets right to the heart of the question. If it's just passing packets 'blindly' the it doesn't matter if they go through the asterisk box out-of-order. The local endpoint's jitterbuffer can re-order them. |
| 23:39.30 | kfife | OTOH, if they're being repackaged somehow by asterisk, then naturally iax would need its own buffer. |
| 23:39.43 | kfife | 'they' = the voice frames |
| 23:40.07 | beek | Hmmm... well, the packets would need to be demultiplexed from the IAX stream and placed into individual SIP channels. |
| 23:40.48 | *** join/#asterisk Get_The_Fish (~IceChat7@c-24-8-50-199.hsd1.co.comcast.net) |
| 23:41.53 | beek | kfife: Interesting question. |
| 23:43.37 | Get_The_Fish | hello bug marshalls- wanted to check with you guys on attended xfers in 1.6.2.5... having a problem with the "C" leg of the call being connected but the "b" party still hearing ringing, then we get an AT timeout exceeded in the console |
| 23:44.01 | Get_The_Fish | I am pretty sure that this is/has been addressed, but I cant seem to find the issue in Mantis |
| 23:55.42 | *** join/#asterisk pyite_ (~dschreibe@unaffiliated/pyite) |
| 23:55.57 | *** join/#asterisk nightrid3r (kvirc@41.214.152.132) |
