| 00:00.23 | DrDigital | 1.14. uhhh |
| 00:01.40 | DrDigital | Asterisk 1.4.26.1 built by root @ rpmbuild32.elastix.palosanto.com on a i686 running Linux on 2009-08-24 23:16:22 UTC |
| 00:02.52 | WiretapWork_ | 1.4.26.1 |
| 00:03.16 | WIMPy | Ok, so if it's due to a forged SIP packet, chances are, it has already been fixed :-) |
| 00:04.38 | DrDigital | can i upgrade asterisk easily? |
| 00:04.46 | DrDigital | 1 word |
| 00:04.49 | DrDigital | Elastix |
| 00:05.21 | WIMPy | NFI |
| 00:05.29 | DrDigital | if i have to |
| 00:05.40 | WIMPy | But maybe you can pinpoint the source with tcpdump. |
| 00:05.54 | DrDigital | i can start over, the biggest thing was my guy had a bitch of a time getting the hardware timer to work |
| 00:06.08 | WIMPy | Or find out what it actually is, you're receiving. But that's for the SIP experts. |
| 00:07.30 | DrDigital | i just see comcast and vitel.net |
| 00:08.31 | DrDigital | cvomcast is both companies isp's |
| 00:08.43 | DrDigital | including the server |
| 00:08.43 | DrDigital | and vitel.net im assuming is vitelity.com |
| 00:09.17 | DrDigital | 17:06:49.206769 IP 173-162-2-138-Stockton.hfc.comcastbusiness.net.11848 > 64.2.142.210.GIGe-net.vitel.net.10560: UDP, length 172 |
| 00:12.08 | *** join/#asterisk Gugge (~gugge@91.208.16.1) |
| 00:17.52 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:24.22 | jc319 | exten => _07[5-9]XXXXXXX.,n,GotoIf(${OutPeer}=${INVALID_OUTPEER}?cant-call:);; if OutPeer variable is set to INVALID_OUTPEER, Hangup() by going to 'cant-call' label - otherwise just continue with the next priority |
| 00:25.40 | jc319 | This does not compare as I intend. How can I compare if a variable (string text) is exactly what I expect? |
| 00:26.36 | jc319 | Basically, if ${OutPeer} is set to "INVALID_OUTPEER" I want to jump to Hangup() otherwise continue with the next priority |
| 00:27.48 | WiretapWork_ | jc319, you want gotoiff |
| 00:27.49 | WIMPy | strip the ${} from the literal string. It's only for variables. |
| 00:27.50 | WiretapWork_ | gotoif* |
| 00:28.12 | WIMPy | And maybe you want to use contexts insted? |
| 00:28.13 | WiretapWork_ | oh, I see you found it |
| 00:29.13 | WIMPy | goes to sleep |
| 00:29.44 | DrDigital | WIMPy, i changed the ip |
| 00:29.44 | DrDigital | and the load went away |
| 00:29.58 | jc319 | I don't know what I want, p3nguin helped me with this code, it works with OutPeer which is defined in every peer definition and routes outbound calls using correct ITSP sub-account (mandatory for pre-paid credits & CDR). everything 'works' so far, I am trying to improve it now so that I don't need to calll 00<country code> all the time |
| 00:29.59 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 00:33.52 | jc319 | WiretapWork_: OK I removed the ${}, so it's now: exten => _07[5-9]XXXXXXX.,n,GotoIf(${OutPeer}=INVALID_OUTPEER?cant-call:);; if OutPeer variable is set to INVALID_OUTPEER, Hangup() by going to 'cant-call' label - otherwise just continue with the next priority |
| 00:34.04 | jc319 | it still kills all calls |
| 00:34.04 | jc319 | <PROTECTED> |
| 00:34.04 | jc319 | <PROTECTED> |
| 00:34.28 | jc319 | <PROTECTED> |
| 00:34.28 | WiretapWork_ | it should execute the gotoif |
| 00:34.34 | WiretapWork_ | however it shouldn't evaluate to true |
| 00:49.31 | DrDigital | OMG |
| 00:49.34 | DrDigital | look at this load! |
| 00:49.35 | DrDigital | 2815 asterisk 15 0 142m 104m 8600 S 1.0 5.2 31:14.53 asterisk |
| 00:49.47 | DrDigital | 1.0% |
| 00:49.51 | DrDigital | like usually is |
| 00:50.04 | DrDigital | now .7 |
| 00:50.44 | DrDigital | I would really like to know what was causing it |
| 00:50.55 | DrDigital | guess if it makes it to the 24th it wont matter much |
| 00:52.38 | jc319 | WiretapWork_: Fixed it thanks. It was about [] and this worked: |
| 00:52.39 | jc319 | exten => _07[5-9]XXXXXXXX,n,GotoIf($[${OutPeer}=INVALID_OUTPEER]?cant-call:go-ahead-dial);; if OutPeer variable is set to INVALID_OUTPEER, Hangup() by going to 'cant-call' label - otherwise go-ahead-and-dial |
| 00:52.39 | jc319 | exten => _07[5-9]XXXXXXXX,n(go-ahead-and-dial),Dial(SIP/${OutPeer}/0044${EXTEN:1})); |
| 00:52.39 | jc319 | exten => _07[5-9]XXXXXXXX,n(cant-call),Hangup(); |
| 00:53.03 | WiretapWork_ | yep |
| 00:54.36 | *** part/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 00:55.03 | DrDigital | elastix shows 46.41% used of 2,657.91 MHz however |
| 00:55.26 | WiretapWork_ | ignore that |
| 00:55.37 | WiretapWork_ | any cpu counter that tries to measure in mhz should be hit |
| 00:56.30 | DrDigital | and people ask why i want to ditch elastix |
| 00:56.31 | DrDigital | i dont want to because im familure with it |
| 00:56.58 | DrDigital | i dont WANT to learn something new |
| 00:57.22 | WiretapWork_ | DrDigital, the freepbx netinstall distro will play nice for you, and is ast 1.8 |
| 00:57.31 | WiretapWork_ | I've just deployed one here |
| 00:57.32 | DrDigital | but i want something more reliable and easier to update |
| 00:57.42 | WiretapWork_ | and it will let you update :P |
| 00:57.51 | DrDigital | i was gonna play with asterisknow |
| 00:58.02 | WiretapWork_ | asterisknow is a bit older I think |
| 00:58.18 | DrDigital | I really wished i could have like a virtual systems |
| 00:58.25 | WiretapWork_ | http://downloads.freepbxdistro.org/ISO/FreePBX-Distro-Net-1.8.1.4.iso |
| 00:58.30 | DrDigital | so each company has there own area |
| 00:58.41 | WiretapWork_ | you'll want to look at BlueBox for that |
| 00:58.44 | WiretapWork_ | its FreeSwitch based |
| 00:58.56 | WiretapWork_ | I didn't think it was mature enough for real use though |
| 00:59.11 | DrDigital | yeah thats what everyone says |
| 00:59.19 | DrDigital | nothing for what i want exsist |
| 00:59.28 | DrDigital | that i may as well put 1 new system up for each company |
| 00:59.40 | DrDigital | then one gets effected they all dont |
| 00:59.54 | DrDigital | like the cms, vTiger |
| 00:59.57 | WiretapWork_ | DrDigital, atom based machines |
| 01:00.01 | DrDigital | it isnt made to run multiple companies |
| 01:00.20 | DrDigital | in elastix at least |
| 01:00.27 | DrDigital | i made a slight hack |
| 01:00.27 | DrDigital | i made it have a pop up list |
| 01:00.43 | WiretapWork_ | elastix is just freepbx + asterisk + a few little custom bits |
| 01:00.45 | DrDigital | you selected the company from the pop up list which just changed databases |
| 01:00.45 | DrDigital | why couldnt elastix or whatever do this |
| 01:00.50 | DrDigital | make multiple tables/databases |
| 01:00.56 | DrDigital | and let you change companies names |
| 01:01.50 | DrDigital | wehen you access pbx-1.domain.com it have a pop up list ORRR because pbx-1 is M&S and pbx-2 the sub domain tells the php script which database we are using today |
| 01:02.09 | DrDigital | assign a number to each company |
| 01:02.17 | WiretapWork_ | because its not designed for multitennant? |
| 01:02.55 | DrDigital | it wouldnt be hard to make them do it |
| 01:02.55 | DrDigital | i dont think |
| 01:02.55 | DrDigital | store each company in its own database |
| 01:03.36 | WiretapWork_ | do it then? |
| 01:03.40 | DrDigital | hardware timers would be needed for each atom system |
| 01:04.23 | DrDigital | dont get why you have to have a hardware timer, its just software o nthe hardware no? |
| 01:04.32 | WiretapWork_ | I don't use a hardware timer |
| 01:04.40 | WiretapWork_ | hardware timers are only needed for conferencing |
| 01:05.02 | WiretapWork_ | and even then, if your machine is decent you can usually get away without them |
| 01:05.02 | DrDigital | before adding it |
| 01:05.10 | DrDigital | when we got a lot of calls, the IVR played slow |
| 01:05.23 | WiretapWork_ | that's what I mean about nice enough machine |
| 01:05.32 | WiretapWork_ | your CPU load was too high for the timer to tick reliably |
| 01:05.41 | DrDigital | 2.66ghz p4 2gb ram |
| 01:05.48 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 01:05.48 | DrDigital | adding the hardware timer fixed it |
| 01:06.10 | WiretapWork_ | 2.66 P4 counts as 'not great' |
| 01:06.59 | DrDigital | i thought it was pretty good for the usage we have, not like we got a thousand calls at once, i normal every day use the peak would be like 14 |
| 01:07.19 | DrDigital | today was 8 calls |
| 01:07.37 | DrDigital | most are below 5 calls |
| 01:08.07 | DrDigital | is there a better way to see in the gui other then the graphic chart |
| 01:08.12 | DrDigital | like actual numbers area |
| 01:08.40 | DrDigital | so you think an atom processor is better then the 2.66ghz p4? |
| 01:09.10 | DrDigital | i wished you could buy a case and mount several motherboards into the case with atoms |
| 01:10.54 | WiretapWork_ | you can, some special cases exist IIRC |
| 01:11.55 | DrDigital | http://www.frys.com/product/6310331?site=sr:SEARCH:MAIN_RSLT_PG |
| 01:12.40 | DrDigital | or buy 4 of those? |
| 01:14.44 | DrDigital | im not seeing atom motherboards/processors |
| 01:14.45 | WiretapWork_ | I do like mini-itx machines for PBXs |
| 01:14.55 | DrDigital | i got a lot of space here |
| 01:15.00 | DrDigital | and a rack |
| 01:15.01 | WiretapWork_ | atom motherboards are all mini-itx and generally the processors are soldered on |
| 01:15.03 | WiretapWork_ | ah |
| 01:15.11 | DrDigital | i know |
| 01:15.20 | WiretapWork_ | 3 or 4 IBM x336 do ya |
| 01:15.27 | DrDigital | i figured it be listed in with the motherboards |
| 01:15.43 | DrDigital | finally found one |
| 01:15.46 | DrDigital | http://www.frys.com/product/6147609?site=sr:SEARCH:MAIN_RSLT_PG |
| 01:15.54 | WiretapWork_ | 336 are dual xeon, x64, handle up to 8GB of ram |
| 01:16.13 | WiretapWork_ | if you'd mentioned you had a rack, I'd have never specced atoms :P |
| 01:16.21 | WiretapWork_ | theyre really more for the space concious |
| 01:16.45 | DrDigital | i could put several 1u systems up |
| 01:16.59 | DrDigital | more or a price deal |
| 01:17.12 | WiretapWork_ | I would go with a bunch of 1Us personally |
| 01:17.19 | DrDigital | would need to try to keep cost below 400 or so |
| 01:17.25 | DrDigital | per system |
| 01:17.30 | WiretapWork_ | what, are x336 expensive the? |
| 01:17.35 | WiretapWork_ | there* |
| 01:17.49 | DrDigital | dont really need a hard drive |
| 01:17.49 | DrDigital | dont even know what x335 |
| 01:17.53 | WiretapWork_ | theyre about NZ$300 each for a decent spec |
| 01:17.54 | DrDigital | frys.com |
| 01:17.56 | WiretapWork_ | 335 = old |
| 01:17.56 | DrDigital | newegg.com |
| 01:17.59 | WiretapWork_ | 336 = newer |
| 01:18.02 | DrDigital | or amazon.com |
| 01:18.36 | WiretapWork_ | http://cgi.ebay.com/IBM-Server-8837-4AU-X336-Dual-3-6GHz-88374AU-Used-7-/190535256921?pt=COMP_EN_Servers&hash=item2c5cc94b59#ht_3851wt_905 |
| 01:18.56 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:19.01 | WiretapWork_ | available with both SAS/SATA and 3.5" SCA |
| 01:19.40 | DrDigital | okay... how much new? |
| 01:19.47 | WiretapWork_ | no longer available new |
| 01:19.55 | DrDigital | x336 ? |
| 01:19.56 | WiretapWork_ | a new 1U box of any sort will set you back at least 2K |
| 01:20.06 | WiretapWork_ | the x336 was discontinued in 2008 |
| 01:20.20 | WiretapWork_ | doesn't make it a crap box for the job |
| 01:20.40 | DrDigital | i see |
| 01:20.49 | DrDigital | i do have 2 1u's i dont even use |
| 01:20.53 | DrDigital | they got some xeon processor |
| 01:21.07 | WiretapWork_ | make sure theyre 800MHz FSB xeons or better |
| 01:21.10 | WiretapWork_ | or you can forget about them |
| 01:21.12 | DrDigital | one works fully, the other i dont have ram or a IDE hd for it |
| 01:21.19 | WiretapWork_ | ... okay |
| 01:21.25 | WiretapWork_ | if it uses IDE HDDs, forget about it |
| 01:21.37 | WiretapWork_ | IDE was only provided in very-low-end machines |
| 01:22.46 | DrDigital | i wonder how many of these this guy has |
| 01:22.58 | DrDigital | these are a few years old |
| 01:23.02 | WiretapWork_ | yep |
| 01:23.11 | WiretapWork_ | new enough to be 64bit, and have SATA support though :) |
| 01:23.23 | WiretapWork_ | trust me, people down here would kill to be able to get the SAS/SATA ones |
| 01:23.26 | WiretapWork_ | we only get the SCA ones |
| 01:23.32 | *** join/#asterisk MDesade (~desade666@ip24-251-93-137.ph.ph.cox.net) |
| 01:23.38 | DrDigital | im checking his store |
| 01:23.39 | MDesade | hello all |
| 01:40.29 | MDesade | i could use some help figuring out why after a new install on kubuntu 11.04-64bit: ./etc/init.d/asterisk start fails |
| 01:41.15 | MDesade | specifically after installing then after asterisk, and the dahdi driver is correct |
| 01:42.06 | WiretapWork_ | MDesade, did you compile asterisk for that arch? |
| 01:46.50 | MDesade | http://pastebin.com/s2Vfh2Kp |
| 01:46.59 | MDesade | i used the apt-get install |
| 01:47.10 | MDesade | asterisk sox curl |
| 01:47.12 | MDesade | etc |
| 01:47.26 | MDesade | so, no, i did not compile from source |
| 01:48.03 | MDesade | the pastebin link i posted is what i see when i execute "./asterisk debug -v |
| 01:48.48 | DrDigital | what sucks about macs |
| 01:48.48 | DrDigital | command w and command q are to close together |
| 01:48.53 | DrDigital | one closed the window, the other quits the app |
| 01:49.24 | DrDigital | all apps should have to confirm before quiting |
| 01:49.34 | WiretapWork_ | MDesade, make sure you're only loading one voicemail module |
| 01:50.18 | MDesade | ok, i see that, however i haven't configured anything yet, its a new install of freepbx that is generating that somehow |
| 01:50.50 | MDesade | prolly from some module |
| 01:51.24 | MDesade | after installing freepbx and logging in, i did a "refresh modules" and "download all" for the updates or add-ons |
| 01:51.30 | *** join/#asterisk alex5771_ (~alex@ool-1892e7b5.dyn.optonline.net) |
| 01:52.58 | MDesade | im not a total asterisk newb, ive been running trixbox for a few years, but this is the first time i have installed asterisk, then freepbx as packages, instead of some jumpstart type install (trixbox) |
| 01:53.48 | WiretapWork_ | MDesade, make sure you're only loading one voicemail module' |
| 01:53.52 | WiretapWork_ | make good and properly sure |
| 01:54.15 | MDesade | ok, lemme take a look at that section |
| 02:01.18 | MDesade | WiretapWork_ ok, i was looking through the freepbx modules, and unloaded 1 that pertained to "voicemail blasting" still no change |
| 02:01.31 | WiretapWork_ | MDesade, freepbx modules are NOT what I was talking about |
| 02:01.40 | WiretapWork_ | I was talking about asterisk modules, as this is #asterisk |
| 02:01.56 | MDesade | where can i find the .conf file, that asterisk is reading to load multiple Voicemail module |
| 02:15.54 | MDesade | WiretapWork i understand that this is not Freepbx chat, however freepbx is running and asterisk is not as it fails on startup, and i was asking why that is |
| 02:16.38 | MDesade | so, where is the conf file that asterisk is reading, that loads those 5 voicemail modules, so i can look at it using VI |
| 02:17.42 | WiretapWork_ | MDesade, I know, and I'm telling you to remove the excess modules from /var/lib/asterisk/modules |
| 02:17.52 | WiretapWork_ | the conf file is /etc/asterisk/modules.conf |
| 02:18.12 | WiretapWork_ | but you won't have any specifics in there |
| 02:18.27 | MDesade | here is the complete verbose output: |
| 02:18.46 | MDesade | http://pastebin.com/du7T6hkx |
| 02:18.57 | MDesade | ok, i will look at that |
| 02:19.45 | WiretapWork_ | there is voicemail with imap, voicemail with odbc and voicemail with file storage |
| 02:19.50 | WiretapWork_ | you probably only want the last one |
| 02:20.21 | MDesade | yeah, i "think" this is a problem with the odbc module |
| 02:20.48 | WiretapWork_ | remove it |
| 02:20.52 | WiretapWork_ | and the imap one |
| 02:20.55 | MDesade | ok, under /var/lib/asterisk there is no file or sub-dir for "modules |
| 02:20.56 | WiretapWork_ | since you likely want neither |
| 02:21.01 | MDesade | right |
| 02:21.05 | WiretapWork_ | try /usr/lib/asterisk/modules/ |
| 02:22.44 | MDesade | under /usr/lib/asterisk/modules, there is several hundred .SO files, no confs |
| 02:23.26 | WiretapWork_ | nope, no confs in there |
| 02:23.31 | WiretapWork_ | the .so files are the modules |
| 02:23.38 | WiretapWork_ | pastebin an ls |
| 02:23.46 | MDesade | ? |
| 02:24.01 | MDesade | the LS output? |
| 02:24.04 | WiretapWork_ | do 'ls /usr/lib/asterisk/modules' |
| 02:24.06 | WiretapWork_ | pastebin |
| 02:24.07 | WiretapWork_ | ? |
| 02:24.08 | WiretapWork_ | profit |
| 02:24.25 | MDesade | gotcha |
| 02:25.37 | MDesade | http://pastebin.com/MfnzHgWn |
| 02:27.30 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 02:27.47 | WiretapWork_ | MDesade, remove/move away app_voicemail_imap.so and app_voicemail_odbc.so |
| 02:27.51 | *** join/#asterisk sourcode (~code@ppp-58-8-43-133.revip2.asianet.co.th) |
| 02:28.14 | MDesade | ok... |
| 02:28.21 | WiretapWork_ | if you compile asterisk |
| 02:28.25 | WiretapWork_ | you only get one of those three |
| 02:28.34 | WiretapWork_ | if you yum install, or apt-get install, you get all three |
| 02:28.36 | WiretapWork_ | very annoying |
| 02:29.56 | MDesade | ok, done |
| 02:30.01 | MDesade | restart asterisk? |
| 02:30.03 | WiretapWork_ | now try |
| 02:30.04 | WiretapWork_ | yep |
| 02:30.19 | MDesade | haot damn |
| 02:30.27 | MDesade | er, hot damn, she flies now |
| 02:30.36 | MDesade | i knew it was something stupid |
| 02:30.51 | WiretapWork_ | not your fault |
| 02:30.54 | MDesade | hehe, cause its a new install, i haven't messed anything up by hand (yet) |
| 02:30.57 | WiretapWork_ | smack up your distro maintainer |
| 02:31.11 | WiretapWork_ | it had me stuck for freakin hours until I worked it out |
| 02:32.30 | MDesade | yeah, i asked some of the freepbx people the same question, and i got no answers... like i said, i installed asterisk, then freepbx, and hadn't configured anything beyond dahdi for my hardware |
| 02:32.58 | MDesade | which was reporting it was happy, sees my channels, my hardware... so? wtf would ./asterisk reload? |
| 02:33.07 | MDesade | anyway? thank you very much for the help! |
| 02:33.27 | WiretapWork_ | no problem |
| 02:33.39 | MDesade | now i gotta figure out where the FOP went, which IS a freepbx question |
| 02:33.52 | MDesade | very much appreciate the hlep |
| 02:33.54 | MDesade | er, help |
| 02:34.44 | WiretapWork_ | MDesade, amportal start_fop |
| 02:36.21 | MDesade | where is that? /etc/init.d? |
| 02:37.10 | WiretapWork_ | nope |
| 02:37.13 | WiretapWork_ | in the path |
| 02:37.16 | WiretapWork_ | part of freepbx |
| 02:38.46 | MDesade | nope... |
| 02:39.19 | MDesade | i looked under my source, not there... i think i have to DL it separately? it doesnt come with freepbx? |
| 02:39.30 | *** join/#asterisk thedavidfactor (~david@nc-71-52-232-56.dhcp.embarqhsd.net) |
| 02:40.26 | MDesade | i know the FOP is flaky, it was under trixbox also |
| 02:41.25 | thedavidfactor | anyone using chan_gtalk with asterisk 1.8.x? I'm trying to get it working having some issues |
| 02:43.52 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 02:51.43 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:52.37 | *** join/#asterisk rue_mohr (~rue@h24-207-19-104.cst.dccnet.com) |
| 02:54.35 | *** join/#asterisk emsLinux (~dave@190.251.101.201) |
| 02:55.30 | MDesade | hrmmm, installing FOP is not part of the freepbx install? |
| 02:56.47 | emsLinux | Night people, i got an issue with the hold option for call, everytime i try to use the hold option, the music sounds but can't get back to the call, if i press hold again the call hangs up, does anyone have an idea what is going on, cant find any info about it. |
| 02:56.55 | WiretapWork_ | MDesade, should be |
| 02:58.06 | russellb | thedavidfactor: i think google keeps changing the protocol ... |
| 02:58.38 | MDesade | weird |
| 02:58.55 | russellb | FreePBX and FOP are separate projects |
| 03:06.57 | thedavidfactor | russellb, you're up too late ;-) and I think I haven't gotten far enough to have protocol issues, I think I'm still missing some basic config stuff. I found a couple of guides online, but was hoping to talk to someone that had done it |
| 03:14.29 | emsLinux | can anyone help me with my notworking hold? |
| 03:16.18 | WiretapWork_ | emsLinux, your level of supplied information is insufficient |
| 03:22.43 | *** join/#asterisk radic (~radic@dslb-094-216-234-213.pools.arcor-ip.net) |
| 03:26.22 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 03:33.48 | thedavidfactor | night all |
| 03:34.36 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 03:39.00 | cj | I'm having some problems with res_crypto.so |
| 03:39.15 | cj | when I reload it, I don't see the rsa keys being loaded |
| 03:39.40 | cj | there's nothing in the asterisk/messages log, either |
| 03:47.09 | cj | http://paste2.org/p/1474350 |
| 03:48.09 | emsLinux | WiretapWork_ I know it is insufficient, but i have not idea what is happening, i dont even get an error from CLI, everytime i use the hold option can't go back to the call, it just hangup |
| 03:48.21 | cj | http://paste2.org/p/1474352 |
| 03:49.59 | cj | $ dpkg -l asterisk-1.8 | grep ast |
| 03:50.00 | cj | ii asterisk-1.8 1:1.8.3.3-1digium1~squeeze Open Source Private Branch Exchange (PBX) |
| 03:50.31 | WiretapWork_ | emsLinux, turn on sip debug |
| 03:51.26 | *** join/#asterisk Maxxed (~Maxxed@216.215.95.118) |
| 03:58.27 | cj | WiretapWork_: is there a way to turn up the debug for the res_crypto module? |
| 03:58.29 | emsLinux | WiretapWork_ gonna do it |
| 03:58.48 | WiretapWork_ | cj, no idea sorry |
| 03:58.54 | cj | I'm going to dig in to res_crypto.c in that case :) |
| 04:02.34 | emsLinux | WiretapWork_ sip debug command is not working in the CLI, how can i turn on the Sip Debug? |
| 04:02.53 | WiretapWork_ | emsLinux, sip set debug on |
| 04:09.29 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 04:15.21 | emsLinux | Hey WiretapWork_, here is the debug output, is quit large, couldn't find a problem yet http://pastebin.us/6979 |
| 04:16.33 | WiretapWork_ | emsLinux, what lines does it go on hold and what lines does it try to come off |
| 04:19.09 | emsLinux | WiretapWork_ line 1480 goes on hold, couldnt find the come off line yet |
| 04:23.31 | WiretapWork_ | emsLinux, I thought you were using a 7960 not a PAP2T |
| 04:27.14 | emsLinux | in fact, i'm using an iPod to call, and my friend is using Twinkle over Ubuntu Linux, the PAP2T is from another user |
| 04:28.54 | emsLinux | WiretapWork_ you made me think, thats why theres no error, the problem is the iPod App, it work when i use another device... |
| 04:29.07 | WiretapWork_ | lol |
| 04:29.33 | emsLinux | WiretapWork_ I'm so srry i made you waste your time, but thanks for your help, you are very kind |
| 04:29.47 | WiretapWork_ | no problemo |
| 04:30.27 | *** join/#asterisk devdvd (Jason@c-71-61-188-154.hsd1.wv.comcast.net) |
| 04:31.45 | devdvd | Hi, I can still confirm this bug report at https://issues.asterisk.org/view.php?id=13209 to still be an issue for me in 1.6.2.9. This is in regards to the DTMF "repeating/getting lost" issues when using SIP and rfc2833. Thoughts? |
| 04:33.18 | WiretapWork_ | devdvd, update your asterisk |
| 04:33.58 | *** join/#asterisk golikwid|mac (~chrislees@64.45.192.151) |
| 04:34.09 | DrDigital | i want to update my asterisk |
| 04:34.11 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 04:34.15 | DrDigital | breaks elastix |
| 04:34.25 | WiretapWork_ | lol |
| 04:34.28 | DrDigital | yum upgrade asterisk |
| 04:34.35 | DrDigital | or is it update |
| 04:34.37 | devdvd | WiretapWork_ I have tried it all the way up to the first version of 1.8, still have the issue |
| 04:34.53 | WiretapWork_ | devdvd, you're not using 7911s by any chance are you? |
| 04:34.57 | devdvd | no |
| 04:35.09 | WiretapWork_ | ah, cause they have a problem in their SIP firmware that causes that |
| 04:35.13 | DrDigital | so tomorrow ill have the money to buy at least 1 system |
| 04:35.21 | DrDigital | or go with a weaker system and get 2 |
| 04:35.47 | DrDigital | i could always continue using current server with reformat and reinstall of new system |
| 04:35.59 | DrDigital | put all the small guys on it |
| 04:36.14 | devdvd | hard phone is a polycon soundpoint 331, i have also had this issue with 3cx phone and x-lite |
| 04:36.31 | devdvd | s/polycon/polycom |
| 04:36.38 | DrDigital | ive had nothing but issues with x-lite and zoiper |
| 04:36.56 | DrDigital | one minute the softphones ring |
| 04:36.58 | DrDigital | another they wont |
| 04:37.00 | WiretapWork_ | DrDigital, I would really recommend something with at least two cores for even the small guys |
| 04:37.03 | DrDigital | another they dont want to register |
| 04:37.20 | WiretapWork_ | devdvd, have you forced jitterbuffer on by chance? |
| 04:37.32 | DrDigital | WiretapWork_, small guys i mean the 800 toll free numbers that forward to a single cell phone |
| 04:37.34 | devdvd | i thought that was an iax2 thing? |
| 04:37.39 | DrDigital | no SIP, IAX |
| 04:37.52 | WiretapWork_ | oh, right |
| 04:37.59 | WiretapWork_ | if you're using IAX you will need hardware timing |
| 04:38.03 | devdvd | im not |
| 04:38.05 | devdvd | sip only |
| 04:38.12 | WiretapWork_ | devdvd, was talking to DrDigital |
| 04:38.15 | devdvd | ah ok |
| 04:38.17 | devdvd | sry |
| 04:38.19 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 04:38.24 | WiretapWork_ | your issue unfortunately I don't know enough about |
| 04:38.26 | DrDigital | im not using IAX yet, but id like to |
| 04:38.43 | DrDigital | as i they say it uses less bandwidth server to server |
| 04:38.50 | DrDigital | with the small guys, id just use sip |
| 04:38.57 | DrDigital | they onlyt getting 1 call at a time |
| 04:39.16 | DrDigital | that usb hardware timer was like $50 |
| 04:39.24 | DrDigital | i spent $80 to have it over night shipped |
| 04:39.40 | DrDigital | and the girl in shipping department took it to fedex after work because they already picked up |
| 04:39.49 | DrDigital | i sent her a box of chocolates and flowers |
| 04:40.18 | WiretapWork_ | DrDigital, what type is it? |
| 04:41.19 | DrDigital | shit, i dont remember |
| 04:41.22 | DrDigital | it was a year ago? |
| 04:42.20 | DrDigital | help me find some and ill tell you which one |
| 04:43.14 | DrDigital | or some way to probe it on the server |
| 04:43.50 | *** join/#asterisk micols (~0x2AA7F64@rlogin.dk) |
| 04:43.53 | *** join/#asterisk tzanger (tzanger@mail.mixdown.ca) |
| 04:44.10 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 04:44.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:44.57 | WiretapWork_ | DrDigital, lsusb? |
| 04:45.33 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 04:47.45 | DrDigital | Bus 002 Device 002: ID 10c4:8460 Cygnal Integrated Products, Inc. |
| 04:48.18 | WiretapWork_ | cheers |
| 04:48.39 | WiretapWork_ | ah, its a sangoma |
| 04:48.42 | WiretapWork_ | no wonder it was a pain |
| 04:48.48 | WiretapWork_ | great hardware, but can be a bitch to config |
| 04:49.56 | DrDigital | http://wiki.sangoma.com/sangoma-wanpipe-voicetime |
| 04:49.58 | DrDigital | thats it |
| 04:50.34 | DrDigital | i knew ztdummy had to do with it |
| 04:50.39 | DrDigital | i remembered that part |
| 04:54.40 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 04:55.02 | *** join/#asterisk golikwid|mac (~chrislees@64.45.192.151) |
| 04:56.57 | DrDigital | <PROTECTED> |
| 04:56.57 | DrDigital | <PROTECTED> |
| 04:57.02 | DrDigital | look at that top load |
| 04:57.02 | DrDigital | :) |
| 04:57.08 | DrDigital | i dont see asterisk at all |
| 04:57.37 | DrDigital | ssh is so fast now too |
| 04:57.49 | DrDigital | and im only one 22/3mbit at homer |
| 04:57.54 | DrDigital | <PROTECTED> |
| 04:57.59 | tonsofpcs | rue_mohr: i figured out a way to test but it's through some poor sounding service. gonna try to get a phone # approved |
| 04:58.15 | DrDigital | before that was like 3.40 2.80 2.90 |
| 04:59.21 | tonsofpcs | <PROTECTED> |
| 04:59.33 | tonsofpcs | how do you get to >2?!??! |
| 04:59.49 | WiretapWork_ | tonsofpcs, by being SIPDoSed |
| 05:00.54 | rue_mohr | tonsofpcs, so no need for me to set soemthing up for you on my system then eh? ok |
| 05:01.49 | *** join/#asterisk liuyan (~lovetide@211.154.128.135) |
| 05:03.19 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 05:05.50 | *** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 05:06.13 | DrDigital | we assumed it was sipdosed, we just know changing the servers ip fixed the load issue |
| 05:06.28 | WiretapWork_ | yep |
| 05:06.35 | tonsofpcs | rue_mohr: nope, thanks though :) |
| 05:06.41 | WiretapWork_ | changed the IP before you did a sip set debug on :P |
| 05:06.42 | DrDigital | i also updated the dns server |
| 05:06.56 | WiretapWork_ | but I can't think of anything else that could do it |
| 05:06.58 | DrDigital | so if it was, and they attacked via domain they would have the new address |
| 05:07.11 | WiretapWork_ | it is unlikely it was attack by domain |
| 05:07.18 | DrDigital | i can change it back |
| 05:07.21 | WiretapWork_ | probably your friend who you got in trouble just firing off at your IP |
| 05:07.25 | DrDigital | everyone that uses the system is closed now |
| 05:07.33 | WiretapWork_ | nah, if it ain't broke don't fix it :P |
| 05:07.54 | WiretapWork_ | when you replace all the gear, put the system out as a honeypot box :P |
| 05:08.18 | DrDigital | i use to have a windows system sitting at my spair desk in my office |
| 05:08.23 | DrDigital | i had windows xp on it |
| 05:08.25 | DrDigital | VNC |
| 05:08.28 | DrDigital | server |
| 05:08.30 | WiretapWork_ | you did the old AVTest? |
| 05:08.31 | DrDigital | and steady state |
| 05:08.54 | DrDigital | i had an old CRT monitor that made a shit load of noise when it powered on |
| 05:09.01 | DrDigital | people would gain access to it |
| 05:09.06 | DrDigital | and id watch them use the system |
| 05:09.11 | WiretapWork_ | lol |
| 05:09.14 | DrDigital | installing scripts |
| 05:09.20 | DrDigital | doing this, doing that |
| 05:09.30 | DrDigital | i took one of the programable keys on the keyboard |
| 05:09.38 | DrDigital | made it load a graphic |
| 05:09.40 | DrDigital | FBI logo |
| 05:09.45 | WiretapWork_ | DAHAHAHAAH |
| 05:09.47 | DrDigital | with FBI terminal info |
| 05:09.51 | DrDigital | id hit that button |
| 05:09.52 | DrDigital | AND BAMB |
| 05:09.55 | DrDigital | they dropped |
| 05:10.10 | WiretapWork_ | righty |
| 05:10.12 | WiretapWork_ | hometime |
| 05:10.13 | WiretapWork_ | back in 30 |
| 05:10.17 | DrDigital | rebooted and back to normal |
| 05:10.24 | DrDigital | i think its video game time for me |
| 05:10.28 | DrDigital | StarCraft II |
| 05:10.55 | DrDigital | i was forced to leark asterisk CLI today |
| 05:10.56 | DrDigital | thanks |
| 05:11.00 | DrDigital | learn* |
| 05:17.56 | *** join/#asterisk jetlag (~jetlag@pool-71-168-192-91.cmdnnj.east.verizon.net) |
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| 06:18.03 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
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| 06:27.15 | DrDigital | Wiretap7, |
| 06:33.36 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
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| 06:38.05 | *** join/#asterisk Tim_Toady (~moi@79.103.30.231.dsl.dyn.forthnet.gr) |
| 06:39.02 | Wiretap7 | DrDigital, ? |
| 06:39.32 | DrDigital | what does COA mean |
| 06:39.56 | DrDigital | THERE IS NO OPERATING SYSTEM INSTALLED BECAUSE THE HARD DRIVES WERE WIPED AND THERE IS NO COA. |
| 06:40.15 | mandla | Morning |
| 06:40.33 | DrDigital | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=220797513883#ht_1132wt_1141 does this need rails? |
| 06:40.43 | *** join/#asterisk jetlag (~jetlag@pool-71-168-244-171.cmdnnj.east.verizon.net) |
| 06:53.05 | ChannelZ | DrDigital: Certificate of Authenticity? |
| 06:59.57 | DrDigital | so windows |
| 07:00.37 | DrDigital | its going to be an asterisk system so no biggy |
| 07:08.13 | *** join/#asterisk irroot (~irroot@41.49.195.99) |
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| 07:12.06 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:12.08 | schmidts | good morning |
| 07:15.35 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 07:15.38 | kleszcz | morning |
| 07:17.44 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
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| 07:49.42 | mandla | irroot, goeiemôre, did you get my email/ |
| 07:50.15 | irroot | mandla possibly dude was holiday yesterday and im out the office today ill look in a bit |
| 07:50.22 | irroot | you making some progress i hope |
| 07:56.28 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 07:58.09 | mandla | Just wanted to confirm something, i still cant dial outside. |
| 07:58.27 | mandla | Anyway um going for tea. |
| 07:58.34 | mandla | Il be back soon. |
| 07:59.30 | Ryuho68 | hello |
| 07:59.42 | Ryuho68 | is there any french peaople ? |
| 07:59.48 | Ryuho68 | people* |
| 08:04.40 | Ryuho68 | hum, does anyone can help me with a digium TDM card & trixbox config ? (sorry for my english, i am french) |
| 08:06.10 | WIMPy | There should be many people familar with Digium cards, but for help with trixbox, you better ask in #trixbox. And generally |
| 08:06.16 | WIMPy | ~ask |
| 08:06.17 | 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. |
| 08:07.38 | Ryuho68 | ok WIMPy, i go look on #trixbox thanks |
| 08:08.21 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 08:08.23 | jacc0 | hi all |
| 08:13.10 | *** join/#asterisk teathsch (~desktop@ip68-4-55-105.pv.oc.cox.net) |
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| 08:13.32 | Lantizia | tzafrir_laptop, don't suppose your about? |
| 08:13.50 | teathsch | did google voice do some weird crap with their signalling? i can't get inbound calls to work right half the time |
| 08:17.13 | jacc0 | what about the bugs in the old bugtracking system? do I have to repost them in the new bugtracking system? |
| 08:17.53 | irroot | jacc0 they have been copied and you can get to them still they read only on old system |
| 08:18.12 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
| 08:20.31 | jacc0 | ty |
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| 08:28.37 | AdvoWork | im doing sip set debug peer (my peer that im using for registering remotely(sip) from a company we get ddis from), thats showing loads of things like: http://pastebin.com/Mb7i5Ep2 but im struggling to work out what the problem could be, any ideas please? |
| 08:29.12 | AdvoWork | theyre saying they are receiving no packets from us at all, so its not even getting to them, i can ping their ip, i can resolve their name to ip, i can trace that to them |
| 08:32.52 | showme | is there a possiblity that your isp could be blocking it, or some filter on a firewall? |
| 08:33.19 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
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| 08:40.17 | AdvoWork | showme, it all goes through us, so it wouldnt be the isp if you get me, but ive opened the ports etc, it did originally work a week ago(but hasnt worked since). |
| 08:40.43 | AdvoWork | also, it keeps showing Reliably Transmitting (no NAT) to 78.46.43.9:5060: "no NAT" but ive got nat=yes&yes |
| 08:40.53 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:41.37 | AdvoWork | and also, i do similar with another company, and do sip show registry, and it shows them registered.. |
| 08:42.53 | AdvoWork | sip show registry for the one thats failing shows that its request sent, but they dont get it :/ |
| 08:52.56 | kaldemar | AdvoWork: sounds much like your issue is not related to asterisk, but networking in general. |
| 08:56.41 | kaldemar | AdvoWork: do you have nat=yes under [general]? external address seems to be configured ok because asterisk picks it to the messages. also, do you have localnet configured? |
| 08:57.30 | *** join/#asterisk vfabi (~fabi@host-static-188-237-240-227.moldtelecom.md) |
| 09:00.00 | *** join/#asterisk morbidwar (~ovidiu@81.196.150.82) |
| 09:00.12 | kaldemar | AdvoWork: do you see the registration packets going out of the asterisk box? "tcpdump -ni YOUR_INTERFACE_NAME host 78.46.43.9" |
| 09:07.42 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:12.43 | AdvoWork | kaldemar, these are my full settings: http://pastebin.com/a2yxBjbU |
| 09:12.54 | AdvoWork | just going to test that tcpdump |
| 09:14.03 | AdvoWork | and here are the packets from the dump: http://pastebin.com/fKB8C1rb |
| 09:14.08 | AdvoWork | what does that mean? |
| 09:16.09 | kaldemar | it means that the packets go out of your machine towards address 78.46.43.9 and port 5060. |
| 09:17.41 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 09:17.46 | kaldemar | which means that the problem is not in the asterisk box, but somewhere else in the path to 78.46.43.9. |
| 09:19.13 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 09:19.29 | kaldemar | AdvoWork: btw. your full settings pastebin makes no sense at all. |
| 09:20.37 | kaldemar | just a mixed pile of settings with a load of syntax errors. |
| 09:23.23 | AdvoWork | kaldemar, thats the settings that the mydivert.com give you to use |
| 09:23.51 | kaldemar | well they are a load of crap. |
| 09:24.02 | AdvoWork | so does that mean the problem is outside of my box between here and that ip, or could it be outside of asterisk and in my network some how? |
| 09:24.39 | kaldemar | outside your asterisk box. |
| 09:26.14 | kaldemar | if you're pasting configs, do it properly. don't cut any [context]'s and tell from which file they are from. |
| 09:31.39 | AdvoWork | kaldemar, thats literally what ive got, ive not removed anything though, they list it on their support page, im using trixbox and have pasted them into the PEER DETAILS, CONTEXT DETAILS etc on the trunk page |
| 09:32.20 | AdvoWork | any suggestions how i can debug further to work out whats going on? |
| 09:35.10 | *** join/#asterisk jkroon (~jkroon@dsl-242-11-09.telkomadsl.co.za) |
| 09:36.05 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 09:36.57 | AdvoWork | or, could it be due to the settings im sending its not working? but as said, it did originally work last week |
| 09:41.09 | AdvoWork | also, i have a general purpose machine that everything goes through, so ive done a dump on that listening for the asterisk server which has produced: http://pastebin.com/T5z2JmCh |
| 09:43.43 | kaldemar | AdvoWork: for further questions on asterisk configuration, go to #trixbox of #freepbx. |
| 09:44.21 | kaldemar | AdvoWork: there was no response messages in the tcpdump you pasted, you have a network problem. not an asterisk problem. |
| 09:45.02 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 09:45.23 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 09:46.07 | kaldemar | AdvoWork: define "everything goes through". |
| 09:48.47 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:50.17 | AdvoWork | kaldemar, well that machine controls dhcp and dns |
| 09:52.23 | AdvoWork | you know you said my settings were incorrect, which ones? i keep seeing: etransmitting #5 (no NAT) to 78.46.43.9:5060 but it differs, one time its no NAT, then its NAT etc |
| 10:00.56 | *** part/#asterisk lovetide (~lovetide@211.154.128.135) |
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| 10:47.57 | AdvoWork | argh, going to go insane |
| 10:49.25 | dym | Insanity is never a good choice. |
| 10:50.23 | WIMPy | But it's funny. |
| 10:53.00 | AdvoWork | i just cant work out what it is, it seems like its out of asterisk, but still on my network, yet i do the same for other companies(registering sip) and they work fine.. this company are saying theyre not seeing any packets, so im struggling to work out what to do next |
| 10:54.36 | AdvoWork | any ideas why it would be doing this? [Jun 17 11:53:41] WARNING[2048]: chan_sip.c:1950 retrans_pkt: Cancelling retransmit of OPTIONs (call id 124e0f0b2264ec9524c034a3549bdd8e@92.27.64.135) |
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| 11:12.41 | pabelanger | AdvoWork: Enable 'sip set debug on' and see what is happening. You can also go to your gateway and confirm the packets are being passed |
| 11:12.50 | pabelanger | ~collectdebug |
| 11:12.50 | infobot | from memory, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 11:12.57 | pabelanger | AdvoWork: ^pb |
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| 11:25.31 | AdvoWork | pabelanger, im just trying that now |
| 11:25.41 | AdvoWork | also, how come: Retransmitting #2 (NAT) to 78.46.43.9:5060: Retransmitting #3 (no NAT) to 78.46.43.9:5060 its doing one by Nat, one No nat? |
| 11:25.57 | AdvoWork | pabelanger, ive got the sip set debug on, struggling to see anything relevant |
| 11:26.02 | AdvoWork | shall i paste some of the output? |
| 11:26.25 | pabelanger | AdvoWork: yes. follow the steps on the wiki page |
| 11:34.09 | tzafrir_laptop | Lantizia, I'm here now |
| 11:36.10 | Lantizia | tzafrir_laptop, I've forgotten the question lol :P thanks for getting back to me though! |
| 11:40.41 | AdvoWork | pabelanger, ive followed the wiki, here are some logs: http://pastebin.com/taVwbFDP |
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| 11:44.15 | iulhk | hi using asterisk 1.6, how to get dtmf ? |
| 11:46.52 | AdvoWork | pabelanger, the problem is only related to mydivert.com(that trunk) |
| 11:47.03 | pabelanger | Contact: <sip:Unknown@MYEXTERNALIP> |
| 11:47.13 | pabelanger | Did you scrub the pb? |
| 11:47.45 | pabelanger | AdvoWork: ^ |
| 11:52.22 | atheos | iulhk - if you're trying to capture DTMF (if I read your question correctly) then: http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
| 11:53.34 | AdvoWork | pabelanger, pb? do you mean Unknown? the only thing ive changed was MYEXTERNALIP if you get me |
| 11:54.29 | pabelanger | AdvoWork: can you pastebin(pb) an unchanged debug log. Makes it hard to understand what is going on |
| 11:54.41 | AdvoWork | ok, sec |
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| 11:55.50 | AdvoWork | pabelanger, pm'd the link |
| 11:56.39 | AdvoWork | thats unchanged apart from the userid, a fair few request come from Unknown, but its specified and works for some requests.. |
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| 11:59.05 | billmania | I'm using the "leastrecent" strategy and "autopause" with asterisk 1.6.0.22. When an incoming call is offered to the least recently used agent and that agent ignores the call, the call is then offered to the second least recently used agent, but not long enough for the agent to answer. The result is that both agents are then auto-paused. |
| 11:59.26 | billmania | How do I increase the amount of time the call is offered to the second agent? |
| 11:59.51 | AdvoWork | brb |
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| 12:19.11 | dandre | Hello, |
| 12:19.51 | dandre | I am trying to use connectedline function with no success. |
| 12:20.58 | dandre | It is called in my dialplan and when I issue 'sip show history ...' I don't see any trace of the connectedline. |
| 12:21.20 | dandre | Is there some information on how to use this feature? |
| 12:24.04 | WIMPy | Maybe you should tell us what you try to do and what (doesn't) happen. |
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| 12:31.35 | dandre | in my dialplan, before the Dial(...) I have put Set(CONNECTEDLINE(name)=foo) |
| 12:35.36 | *** part/#asterisk hc (~hc@pdpc/supporter/active/hc-e) |
| 12:35.54 | WIMPy | Hmm. Won't Dial() overwrite that, if it can? |
| 12:36.44 | dandre | so How should I do? |
| 12:38.13 | WIMPy | If you have the callerID of the destination set, it will happen automatically. |
| 12:40.43 | dandre | the issue is that I have a shortcut system that define short numbers for frequently used destination numbers. When thos short numbers are dialed, I want to replace on the called device the real party information (name and number) |
| 12:42.14 | WIMPy | It's taken from the called peers callerID. |
| 12:42.54 | dandre | not in my test case |
| 12:43.16 | WIMPy | Worked for me out of the box. |
| 12:43.29 | WIMPy | Are you sure, your phone displays such information? |
| 12:43.57 | WIMPy | I don't thin there are that many, yet, that do. |
| 12:45.45 | dandre | I don't know |
| 12:46.00 | dandre | How can I check that? |
| 12:46.32 | WIMPy | Read the output of sip debug and see if you find soem p-asserted-identity there. |
| 12:46.52 | dandre | ok |
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| 12:49.21 | dandre | I don't have p-asserted-identity |
| 12:49.26 | dandre | nor p-asserted |
| 12:53.30 | WIMPy | I guess you have to wait for someone who knows more about that then. |
| 12:57.47 | Faustov | what's a decent IT recruitment agency in London, specializing in networks? can anyone recommend anything based on experience? |
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| 13:13.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:18.53 | Katty | hai |
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| 13:25.45 | chuckf | hi |
| 13:26.07 | billmania | Morning. |
| 13:26.24 | billmania | Which medical or dietary condition shall we discuss this morning? :-D |
| 13:28.48 | atheos | back to soft drinks billmania. How bad are the sugar free energy drinks? Inquiring minds want to know! |
| 13:29.22 | billmania | atheos: I am leery of all of the artificial sweeteners. |
| 13:29.41 | billmania | I have stopped using table sugar (sucrose) and militantly avoid high fructose corn syrup. |
| 13:30.02 | billmania | I've started baking and sweetening with agave syrup or brown rice syrup or maple syrup or molasses, depending upon the application. |
| 13:30.27 | atheos | I avoided sugar yesterday (based on what I was reading in the channel), and noticed that I felt full on less intake. that's a good change. |
| 13:32.11 | billmania | Be sure to account for (or allow for) the psychosomatic effect of any recent change to your diet. |
| 13:32.56 | billmania | I do have an actual asterisk question. I'm still trying to understand the "leastrecent" queue strategy with asterisk 1.6.0.22. |
| 13:33.01 | billmania | Anyone have any experience with that? |
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| 14:19.44 | billmania | How do I list the current configuration and parameters for a running module in asterisk 1.6.0.22? |
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| 14:35.33 | dacqueries | hi all does anyone know how to get music on hold mp3s to play in a particular order? |
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| 14:48.48 | leifmadsen | dacqueries: sort=alpha and name the files to appear linearly (a.wav, b.wav, etc...) |
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| 15:20.29 | dacqueries | leifmadsen: do you know how to reload the moh module? |
| 15:24.34 | jaytee | wow! if you type help at the CLI it shows you all kinds of cool stuff!!! like moh reload |
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| 15:33.48 | Katty | puts on a Free To Good Home sign. |
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| 15:39.30 | breardo | i'll take you in |
| 15:39.35 | breardo | BUT.. no pissing on the floor |
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| 15:42.21 | leifmadsen | dacqueries: yes I do |
| 15:46.40 | Faustov | Katty: mortgage? |
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| 16:00.40 | luckman212 | any polycom dudes in here? |
| 16:01.49 | luckman212 | any polycom'ers in here? |
| 16:03.12 | Kobaz | polycrums |
| 16:03.23 | leifmadsen | ~ask |
| 16:03.24 | 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. |
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| 16:09.09 | brianjarita | hey all ... when I try to build dahdi I get dahdi-linux-2.4.1.2/drivers/dahdi/dahdi_transcode.c:49: error: ‘SPIN_LOCK_UNLOCKED’ undeclared here (not in a function) ... how do I fix that? |
| 16:09.29 | brianjarita | I'm using kernel 2.6.39 |
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| 16:16.46 | brianjarita | nvm i solved it |
| 16:17.24 | brianjarita | dahdi-linux-2.4.1.2/drivers/dahdi/dahdi_transcode.c ... change static spinlock_t translock = SPIN_LOCK_UNLOCKED; to static DEFINE_SPINLOCK(translock); |
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| 16:41.06 | hardwire | so bria3 for linux is basically crippled. |
| 16:41.09 | hardwire | I feel kinda ripped off |
| 16:41.42 | paulc | how so? |
| 16:41.43 | linuxgecko | i just tried to google how to setup call-waiting for a line i have, and i don't see anything so far.. can someone point me in the right direction? |
| 16:41.46 | Qwell | hardwire: "handicapped" is the P.C. term. |
| 16:41.54 | Qwell | or, "handicapable" |
| 16:41.55 | hardwire | paulc: missing xmpp support. Can't select an outbound line. |
| 16:42.10 | hardwire | completely stupid audio device selection |
| 16:42.49 | paulc | hardwire: xmpp wouldn't bother me much but not being able to select the line/registration to use for an outbound call? That seems kinda dumb.. nothing you can change in configuration etc? (I haven't played with Bria) |
| 16:42.56 | DrDigital | Wiretap7, |
| 16:43.02 | linuxgecko | hardwire: they have a pay version that doesn't duplicate windows-version capability? |
| 16:43.06 | *** join/#asterisk gruvfunk (~chatzilla@cpe-68-172-221-157.hvc.res.rr.com) |
| 16:43.15 | hardwire | linuxgecko: are you being sarcastic? |
| 16:44.15 | *** join/#asterisk wonderworld (~ww@port-92-201-16-166.dynamic.qsc.de) |
| 16:44.51 | linuxgecko | hardwire: no. i think it's stupid for a company to release a linux version that's not as capable as the windows version ( which most companys put the greater effort into). esppecially if you have to pay for it. free version, i can see it being "acceptible" to leave broken. |
| 16:45.15 | hardwire | linuxgecko: I'm learning as we speak.. but it appears the windows version shares most of this dumbness |
| 16:45.26 | hardwire | you select the outbound line using a dialplan. |
| 16:46.19 | gruvfunk | greetz |
| 16:46.30 | hardwire | #2777 would select your second account |
| 16:46.33 | hardwire | and dial 777 |
| 16:46.39 | gruvfunk | anyone here in Australia at the moment, willing to run a test call to a Toll Free number for me? |
| 16:46.42 | linuxgecko | hardwire: that's how I'd do it anyway. select lines on the server, or via exten "encoding" ( dial *x for out on line one, and *y for out on line 2) |
| 16:46.55 | hardwire | linuxgecko: I'd click on it. |
| 16:47.00 | hardwire | as well as show a status of it. |
| 16:47.38 | linuxgecko | ok |
| 16:47.41 | hardwire | cause I don't really want to prepend a bunch of #x to contact entries :) |
| 16:47.51 | hardwire | click on line first.. select contact.. huzzah! |
| 16:48.02 | linuxgecko | yeah, i hear you |
| 16:48.06 | hardwire | right click on contact.. set line.. huzzah! |
| 16:48.19 | hardwire | meh. |
| 16:48.24 | hardwire | we'll see how it ends up working out |
| 16:48.25 | hardwire | :) |
| 16:48.44 | citywok | g' mornin' |
| 16:49.00 | linuxgecko | I'm trying to find a way to engineer a project for a friend who runs an office of agents. but i can't find the call waiting docs for asterisk.. yes, i know the phone has to support it. |
| 16:49.27 | citywok | linuxgecko: you mean call waiting as in a second call being able to come in? |
| 16:49.42 | citywok | linuxgecko: there are no docs, b/c the phone simply needs to accept a second simultaneous call. |
| 16:50.02 | linuxgecko | citywok: yeah.. ... |
| 16:50.43 | citywok | if you have your call-limit set to 1 or whatever in * that would prevent it from sending the call to the phone, but that's about all i can think of in sip land. |
| 16:50.45 | gruvfunk | linuxgecko: and your voip trunk should have at least 2 channels in order to do call waiting |
| 16:50.59 | linuxgecko | citywok: ok, so as long as the phone supports accepting the second call being Dial()'d to it, it works automagically??:) |
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| 16:51.11 | citywok | linuxgecko: that's the idea |
| 16:51.33 | gruvfunk | linuxgecko: are you using xlite softphone to test? it has built in capability to take 2 calls |
| 16:52.05 | linuxgecko | citywok: ...... how do i setup 2 channels for my google-voice? i only have one account. |
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| 16:53.34 | linuxgecko | gruvfunk: I'm using ekiga and sipdroid ATM for my personal phone, but I'm trying to set this up as easy and scalable as i can, for a variable number of agents, also, using my own line as a guinea pig :) |
| 16:56.21 | gruvfunk | linuxgecko: I just tested my google voice setup and it automagically supports call waiting |
| 16:56.51 | gruvfunk | (2 channels are very common for most DID's) |
| 16:58.21 | gruvfunk | anyone here in Australia at the moment? and still awake? |
| 16:58.57 | linuxgecko | s/anyone here/ anyome ELSE here/ i assume :) |
| 17:00.03 | gruvfunk | no I'm in the US |
| 17:00.25 | gruvfunk | I need an Australian friendly to run a telephone test (at no cost to them) |
| 17:00.43 | citywok | gruvfunk: what do you need, i'm not in the U.S. but will spend 10 cents calling you :) |
| 17:00.57 | citywok | s/U.S./australia/ |
| 17:01.35 | gruvfunk | I need somebody who can dial Australian Toll Free numbers - I ordered one from voip.ms (several actually, all tested working but this one) - I can't seem to dial this number |
| 17:01.58 | gruvfunk | several = 1 toll free DID in each country, all working except Australia |
| 17:03.57 | gruvfunk | tried from my att mobile, from my verizon pstn, and also from a callcentric voip trunk known to work internationally - mixed results |
| 17:06.00 | gruvfunk | either I get a busy/congested dial tone (from Verizon), or I get an Australian accent " this number is not connected, please check the number before trying again" from AT&T mobile, and I get continuous ring when calling from callcentric |
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| 17:17.26 | e7e5 | Hey! Can anybody help me with "DEBUG[3817] cdr_radius.c: Unable to create RADIUS record. CDR not recorded!" ? |
| 17:17.26 | e7e5 | I want use accounting. |
| 17:17.26 | e7e5 | Asterisk 1.6.2.13. |
| 17:17.26 | e7e5 | FreeRADIUS 1.1.3 and radiusclient-ng 0.5.6 from yum. |
| 17:17.26 | e7e5 | CentOS 5.6. |
| 17:18.39 | paulc | gruvfunk: if you can wait a few hours, I can get my buddy in Australia to make a call for you |
| 17:18.53 | gruvfunk | thanks paulc |
| 17:19.19 | gruvfunk | i'm guessing this Australian Toll Free is limited to only being called from within Australia |
| 17:23.08 | gruvfunk | yep, voip.ms just confirmed it.. |
| 17:23.27 | gruvfunk | worst part is, we need to issue this number out to our customers, but can't issue it without knowing the line is quality assured |
| 17:23.40 | gruvfunk | and voip.ms can't send a test call to the line either, strange |
| 17:24.20 | gruvfunk | paulc: so yes, if you can get a pal to run a test, please send me a PM for the number - it's just a simple IVR at the moment ,hear it and hang up |
| 17:26.20 | paulc | gruvfunk: Sure - PM'd you.. buddy should be online in a few hours. Got a meeting to go to - back in 30 or less |
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| 17:58.25 | thedavidfactor | I'm playing around with chan_gtalk on asterisk 1.8 branch. jabber show connections is showing my google account, but when I attempt to dial out I get "chan_gtalk.c:1863 gtalk_request: Could not find recipient" I'm not seeing any packets getting sent over the wire so I'm assuming I'm missing some part of the config can anyone point me in the right direction? google hasn't been much help |
| 18:00.20 | leifmadsen | thedavidfactor: there should be stuff on the asterisk wiki... |
| 18:00.28 | leifmadsen | I know malcolm did a lot of documenting |
| 18:03.27 | Kobaz | wiki wiki |
| 18:06.27 | *** join/#asterisk suzie_needs_help (48edd5a2@gateway/web/freenode/ip.72.237.213.162) |
| 18:06.33 | suzie_needs_help | hi asterisk |
| 18:07.01 | suzie_needs_help | is there anyway for a voicemail message to automatically be distributed among other mailboxes once it has been left in it? |
| 18:07.24 | WIMPy | No. |
| 18:07.29 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 18:07.31 | WIMPy | But you can share mailboxes, off course. |
| 18:08.31 | suzie_needs_help | what about copying the message to other mailboxes using ${VM_MESSAGEFILE} ? |
| 18:09.30 | suzie_needs_help | or is there a way to send a "soundfile" to a mailboxe's INBOX ? |
| 18:09.36 | WIMPy | Sure, but you need to enable the old polling thing to make MWI work then. |
| 18:09.51 | suzie_needs_help | not interested in MWI |
| 18:10.17 | thedavidfactor | leifmadsen, thanks is the wiki spidered by google? |
| 18:10.20 | suzie_needs_help | WIMPy, how does one enable "POLLING" ? |
| 18:10.24 | WIMPy | In that case a simple copy should do, I think. |
| 18:10.50 | leifmadsen | thedavidfactor: not too sure |
| 18:10.55 | WIMPy | I think it's a parameter in sip.conf. |
| 18:10.58 | suzie_needs_help | WIMPy: I'm using voicemail ODBC though |
| 18:11.08 | michael-i | I'm trying to timestamp some recordings with the current call time. Using ANSWEREDTIME and DIALEDTIME returned nothing.That is still the way to accomplish this, right? |
| 18:11.19 | WIMPy | Ok, I've got no clue then. |
| 18:12.16 | suzie_needs_help | does polling work with voicemail ODBC ? |
| 18:12.58 | WIMPy | It you're not interested in MWI, you don;t have to care. |
| 18:13.04 | pabelanger | michael-i: ${EPOCH} |
| 18:14.15 | suzie_needs_help | WIMPy: i'm interested using the notify in voicemail.conf though, it needs to work |
| 18:14.21 | michael-i | pabelanger: that doesn't really accomplish my goal though. I want to eventually be able to lay these out on a timeline, so I need elapsed call seconds. (or to record an additional epoch at call begin) |
| 18:15.39 | pabelanger | michael-i: so you want both the start and stop time in the filename? |
| 18:16.12 | michael-i | pabelanger: just start, I can compute stop from the recording itself |
| 18:16.49 | *** join/#asterisk bchia (~Adium@nat/digium/x-witsanuzamhlvnvl) |
| 18:16.57 | michael-i | (but a relative start, not the absolute epoch start since this can change depending on timezone, ntp adjusts, etc...) |
| 18:18.03 | pabelanger | michael-i: well, you could use it with ${STRFTIME} |
| 18:18.10 | pabelanger | and format it however you like |
| 18:18.10 | *** join/#asterisk bchia (~Adium@nat/digium/x-hrgqtqagligskfdt) |
| 18:18.53 | Katty | i don't spose theres anyone in the st. louis area |
| 18:19.29 | e7e5 | can somebody tell me, how to troubleshoot radiusclient-ng? |
| 18:19.45 | michael-i | pabelanger: true again...I was just looking for something as simple as "callseconds" which would stamp "15" for a recording that started at the 15 second mark |
| 18:22.07 | suzie_needs_help | is there anyway to send a sound recording to a voicemailbox's inbox? |
| 18:23.21 | pabelanger | michael-i: Don't think it exists on the current channel running, but you have the write solution. Store the EPOCH once your channel starts, then subtract the current EPOCH when start recording; there is your "callseconds" |
| 18:23.28 | pabelanger | wow, fail |
| 18:23.31 | pabelanger | s/write/right |
| 18:24.15 | michael-i | pabelanger: this seems to be my only way forward...short of rewriting app_dial to constantly update ANSWEREDTIME |
| 18:25.29 | WIMPy | Where does 'core show channels verbose' get the information from? Does it also subtract from now? |
| 18:26.09 | pabelanger | Ya, ANSWEREDTIME and DIALED time only get set once Dial() has returned, on the originating channel. I assume you are trying to access the values on the dialled channel |
| 18:26.50 | michael-i | either one...but was going to use MASTER_CHANNEL to keep things consistent with the cdr |
| 18:26.54 | *** join/#asterisk lupestro (~chatzilla@c-75-68-77-80.hsd1.nh.comcast.net) |
| 18:26.57 | DrDigital | WIMPy, you see how i 'fixed' my issue yesterday? |
| 18:27.14 | suzie_needs_help | is there anyway to cancel leaving a message in a voicemail? |
| 18:27.22 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 18:27.23 | WIMPy | DrDigital: No. Did you find the cause? |
| 18:27.26 | DrDigital | no |
| 18:27.33 | DrDigital | I changed the IP of the server |
| 18:27.34 | suzie_needs_help | or a key to verify you want to send the voicemail? |
| 18:27.45 | DrDigital | and the load went away as SOON as i service network restart |
| 18:28.00 | DrDigital | it went from 99.9% to not even registering on top |
| 18:28.05 | WIMPy | DrDigital: So you didn't find out, what was going on? |
| 18:28.23 | DrDigital | someone had to been slamming asterisk with that IP |
| 18:28.39 | DrDigital | system has 10 calls right now it says |
| 18:28.45 | DrDigital | and asterisk still isnt showing in top |
| 18:28.56 | WIMPy | Someone might do it again. |
| 18:28.59 | DrDigital | top and init are at the top for loads |
| 18:29.00 | pabelanger | michael-i: Another option might be to use CEL, it may have more detailed information |
| 18:29.19 | DrDigital | yeah Im trying to order some HP xeon G5's today |
| 18:30.02 | WIMPy | DrDigital: You won't fix that with hardware. |
| 18:30.06 | michael-i | pabelanger: the more I dig into different things I'll need to access in the future...it's looking like the way to go :) |
| 18:30.08 | DrDigital | going to put them on the rack and asterisknow and put each company on there own box with the ones whos spending over $200 a month everyone else is usually less then $50 they will go onto a single system |
| 18:30.21 | DrDigital | no, it will be new software |
| 18:30.30 | DrDigital | to me the box was hacked/comprimised |
| 18:30.41 | DrDigital | changing the ip they lost there way to the server |
| 18:30.56 | DrDigital | its only 1 ip up, but lost still |
| 18:31.06 | DrDigital | and i changed the root password and the web gui |
| 18:31.36 | DrDigital | the idea with new yard ware is if 1 system does get hacked everyone isnt effected |
| 18:31.37 | WIMPy | Usually a owned box connects to the outside, polling for commands. |
| 18:32.05 | DrDigital | its been up for 18 hours i believe no load yet |
| 18:32.23 | DrDigital | i can keep changing the IP each time it happens till the new systems arrive |
| 18:32.32 | WIMPy | It might be better to find out, how to prevent that from happening again. |
| 18:32.56 | DrDigital | we spent awhile trying to figure it out |
| 18:33.03 | DrDigital | i was up till 2am working on it |
| 18:33.08 | DrDigital | and you left around 6pm my time |
| 18:33.26 | DrDigital | i had the issue patched right when you left |
| 18:33.38 | DrDigital | i said patched because i know its not really fixed |
| 18:33.54 | WIMPy | Hmm. |
| 18:34.58 | lupestro | Does an integration question around settings using a SIP toolkit from code belong here or in -dev? |
| 18:35.19 | lupestro | 'Cuz its about the sip.conf but its also around the right SIP calls. |
| 18:36.17 | lupestro | Trying to SUBSCRIBE but getting auth errors - supplying credentials isn't helping. |
| 18:37.43 | lupestro | Not sure if subscribers need to register and provide a known SIP FROM address. |
| 18:38.23 | lupestro | Don't want to bend-over the settings to get past this |
| 18:38.48 | WIMPy | lupestro: I think that depends on your configuration. If you allow guests ans also set a subscribecontext for them, II'd expect it to work without. |
| 18:39.16 | lupestro | Yeah, but that would be the ben-dover option, wouldn't it :) |
| 18:39.40 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 18:39.47 | WIMPy | You started to question if credentials are neccessary :-) |
| 18:40.18 | lupestro | So if I get the 401 and supply valid credentials for a user, even if I'm letting my toolkit generate the FROM with my IP, should that work? |
| 18:40.18 | *** join/#asterisk bchia (~Adium@nat/digium/x-wzbothacifvvohtn) |
| 18:41.05 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:41.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:41.06 | lupestro | Or do the user/pass need to match the FROM value? I started out with a user and ended up making it a friend but I'm trying to avoid the peer side. |
| 18:41.07 | WIMPy | I think so. But I have to add that I'm not the SIP expert. |
| 18:41.44 | WIMPy | I think type=user needs the corresponding from:. |
| 18:42.09 | timholum1 | My asterisk server just froze on me, when I restarted it everything came up fine, but when I looked in the log, I noticed about 1500 lines like [Jun 17 13:16:15] WARNING[25839] chan_sip.c: sip_xmit of 0x57781c0 (len 646) to 68.117.120.138:47166 returned -2: Success all just befor it froze and all within a minute |
| 18:42.51 | gruvfunk | can somebody enlighten me on Asterisk redundancy - High Availability options, Active-Active vs. Active-Passive? |
| 18:43.05 | lupestro | WIMPy: Which means the corresponding from: will need a REGISTER to know what it means, huh? |
| 18:43.17 | Qwell | gruvfunk: Sure! Get a ticket to Astricon, and see my talk in October. You can wait, right? |
| 18:43.51 | WIMPy | lupestro: No, a register is something else. |
| 18:44.11 | breardo | anyone know any good resources for troubleshooting CRC4 errors on a PRI? |
| 18:44.20 | WIMPy | Registering is only done to tell Asterisk, where to send calls to dynamic peers. |
| 18:44.27 | breardo | "dahdhi show status" is showing CRC4 errors on my PRI span.. |
| 18:44.32 | WIMPy | breardo: CRC 60 |
| 18:44.41 | gruvfunk | lol Qwell |
| 18:44.47 | WIMPy | doh |
| 18:44.50 | lupestro | Ah, good. |
| 18:45.14 | breardo | what do you mean WIMPy? |
| 18:45.23 | *** part/#asterisk ketema (~ketema@ketema.net) |
| 18:45.25 | WIMPy | CRC indusries cantact cleaner 60 that is, not related to cyclic redundancy checks. |
| 18:45.35 | *** join/#asterisk Insonic (~kvirc@ip-178-203-122-117.unitymediagroup.de) |
| 18:45.41 | breardo | oh :) |
| 18:46.22 | gruvfunk | Qwell: Link me up to some reads? |
| 18:46.39 | Qwell | gruvfunk: http://www.linux-ha.org/wiki/Pacemaker |
| 18:47.00 | WIMPy | breardo: Or your box isn't able to process the read IRQs. |
| 18:47.23 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
| 18:47.32 | *** join/#asterisk scalex000 (~chatzilla@186.6.178.17) |
| 18:47.48 | Qwell | gruvfunk: I would be interested in hearing what types of things you'd want to do (for my talk). |
| 18:47.54 | scalex000 | hi Guys, I need to unistall some modules how to do it |
| 18:47.55 | scalex000 | :P |
| 18:48.17 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 18:48.20 | _Corey_ | Qwell: Which is your talk? |
| 18:48.41 | Qwell | _Corey_: Asterisk redundancy: Handling the inevitable |
| 18:48.49 | gruvfunk | Qwell: Looking to support a new "provider" who insists on redundancy.. serious about the quality of service and uptime to his clients |
| 18:49.13 | gruvfunk | i'm sure this topic has been covered before, but I'm new to |
| 18:50.01 | michael-i | scalex000 : quick&brutal, rm -f /path/to/your/module.so; asterisk-rx "core restart now" |
| 18:50.12 | _Corey_ | Looks like a good program for Astricon |
| 18:50.22 | Qwell | _Corey_: I would hope so |
| 18:51.25 | gruvfunk | Qwell: ahh, so it's not Asterisk HA.. it's really OS HA? |
| 18:51.49 | Qwell | gruvfunk: there are lots of ways to do it |
| 18:51.53 | scalex000 | and that it |
| 18:51.55 | scalex000 | :D |
| 18:51.57 | scalex000 | lo |
| 18:51.59 | scalex000 | lol |
| 18:53.47 | carrar | redundancy? |
| 18:53.55 | carrar | phhhhbt |
| 18:54.31 | citywok | Qwell: i signed us up for astricon... but james never got his receipt and they only charged my CC for one... lol, fail. |
| 18:55.30 | leifmadsen | Qwell has nothing to do with that |
| 18:55.37 | Qwell | yes he does |
| 18:55.45 | Qwell | PM me your CC, and I'll charge it for another. |
| 18:55.46 | citywok | i figured as much, but a couple lines up he said go to astricon :P |
| 18:55.56 | citywok | just saying we tried, they wouldn't let us! :P |
| 18:55.58 | leifmadsen | yes but he doesn't charge the CC :) |
| 18:56.05 | Qwell | leifmadsen: no but I could! |
| 18:56.05 | leifmadsen | maybe you aren't wanted? |
| 18:56.08 | leifmadsen | Qwell: totally could |
| 18:56.09 | citywok | Qwell: sure, it's 41475551212121212 |
| 18:56.10 | Qwell | he might not get a ticket, but... |
| 18:56.33 | citywok | haha but what? |
| 18:56.40 | Qwell | but I'd have moneys? |
| 18:57.07 | citywok | haha, fair 'nuf |
| 18:57.11 | _Corey_ | lol |
| 18:57.36 | lupestro | WIMPy: I must be getting further - I've gone from a 401 to a 404 now :) |
| 18:58.35 | lupestro | i.e. now instead of getting 401, supplying creds, getting another 401, I'm getting 401, supplying creds, getting 404. |
| 18:58.44 | citywok | leifmadsen: in all seriousness who do i bug that it failed? lol |
| 18:59.20 | leifmadsen | someone in marketing I'm sure |
| 18:59.26 | WIMPy | lupestro: At this point, the console might give you a hint. |
| 18:59.31 | leifmadsen | or maybe BMJ? |
| 18:59.49 | _Corey_ | Lisa King I think |
| 18:59.59 | leifmadsen | ya probably LK |
| 19:00.01 | _Corey_ | she fixed me up last year when I was doing some registrations |
| 19:00.16 | lupestro | WIMPy: Yeah, I was just headed there... |
| 19:00.28 | citywok | leifmadsen: yea i don't know who those people are :) |
| 19:00.29 | Qwell | I should just start emailing Lisa with all my problems |
| 19:00.38 | leifmadsen | shrugs |
| 19:00.45 | lupestro | Just the usual received SIP subscribed for peer without mailbox: fifi" |
| 19:01.23 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:01.35 | citywok | i'll just reply to the expo@tmcnet email addy |
| 19:01.43 | lupestro | s/subscribed/subscribe/ |
| 19:01.44 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 19:02.05 | citywok | maybe that'll work, lol |
| 19:03.44 | lupestro | WIMPy: I suppose I can get more detailed console output. Probably time to dig into starfish book agn. |
| 19:05.59 | lupestro | WIMPy: Can I crank the verbosity level without hitting asterisk.conf and cycling the server? |
| 19:06.26 | WIMPy | lupestro: core set [verbose|debug] <level> |
| 19:17.00 | lupestro | WIMPy: Did sip set debug on - helped some - It appears it is looking for fifi in the "subscribers" context where my "from" user lives. |
| 19:18.25 | lupestro | but the phone I'm trying to subscribe to presence on is set up in the LocalSets context. |
| 19:20.26 | *** join/#asterisk logicwrath_work (~no@mail.vistitude.com) |
| 19:21.59 | *** join/#asterisk mickecarlsson (~Micke@h95n1c1o1101.bredband.skanova.com) |
| 19:23.28 | lupestro | Back to the books... |
| 19:30.07 | ChannelZ | Books are for squars! |
| 19:30.20 | ChannelZ | s/squars/squares |
| 19:30.37 | ChannelZ | blew that one. |
| 19:32.23 | lupestro | ChannelZ: Ah but such a lovely starfish :) |
| 19:33.16 | ChannelZ | Is the new one the same cover? |
| 19:33.58 | lupestro | ChannelZ, s'pose so... didn't have the old one... |
| 19:34.17 | ChannelZ | must be then |
| 19:34.21 | leifmadsen | yes same cover |
| 19:34.30 | leifmadsen | but has a different corner that says 3rd edition :) |
| 19:34.37 | ChannelZ | I guess it's just a new edition so yeah |
| 19:34.37 | lupestro | ChannelZ: Leif aughta know :) |
| 19:34.51 | leifmadsen | 1st and 2nd editions have mirrored covers |
| 19:35.07 | leifmadsen | (flipped horizontally) |
| 19:35.20 | lupestro | kseritsA? |
| 19:35.23 | ChannelZ | Hmm. I can't remember which I have.. 2nd I think (1.4?) |
| 19:35.35 | leifmadsen | ya 2nd is 1.4 |
| 19:35.37 | leifmadsen | 1st is 1.2 |
| 19:35.43 | leifmadsen | 3rd is 1.8 |
| 19:35.52 | leifmadsen | (basically all the LTS releases) |
| 19:36.17 | ChannelZ | Oh, there's a Cookbook too. |
| 19:39.36 | lupestro | Does SIP subscribe require any particular .conf file setup? It looked like it would be on by default...and it didn't seem like something relevant to dialplan... |
| 19:39.55 | suzie_needs_help | is creating distrubtion capabilites in app_voicemail.c |
| 19:40.10 | WIMPy | sip.conf and hits in extensions.conf. |
| 19:40.48 | lupestro | Ah, it does hit extensions.conf! OK, commencing digging - any hints appreciated... |
| 19:41.08 | WIMPy | s/hits/hints/ |
| 19:41.12 | WIMPy | sorry. |
| 19:42.13 | *** part/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 19:42.43 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 19:42.55 | lupestro | WIMPy: Ah - ok. I'll step away from the keyboard for a bit and really read the chapter covering hints, etc. Many thanks. |
| 19:46.07 | *** join/#asterisk coppice (~chatzilla@210.17.219.183) |
| 19:46.07 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 19:47.32 | *** join/#asterisk Cadey (~x@host81-135-124-78.range81-135.btcentralplus.com) |
| 19:47.41 | Cadey | anyone in here c# devs by any chance? |
| 19:50.00 | gruvfunk | Anyone in here use Ubuntu repos? When shall we see a 1.8.4.2 ? |
| 19:51.05 | Qwell | real soon now |
| 19:51.13 | gruvfunk | I see lots of packages updating to 1.8.4, but "core show version" stlil says 1.8.3.3 |
| 19:51.23 | gruvfunk | cool deal |
| 19:51.38 | *** join/#asterisk luckman212_ (~irc@2001:470:1f07:1225:c99a:b4c:1414:52e) |
| 19:51.46 | Qwell | pabelanger: ^^ |
| 19:52.09 | pabelanger | ~asterisk-packages |
| 19:52.09 | infobot | somebody said asterisk-packages was Asterisk is available for automated binary installation using APT for Debian and Ubuntu or YUM utility on CentOS 5 Linux and for RedHat Enterprise Linux 5: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 19:52.19 | pabelanger | gruvfunk: ^ did you follow that? |
| 19:53.15 | gruvfunk | pabelanger: yes that's how I installed several systems |
| 19:53.51 | pabelanger | gruvfunk: did you restart asterisk? |
| 19:53.57 | pabelanger | Hmm, wait |
| 19:54.20 | gruvfunk | I have yes. I did not install the proposed/optional branch |
| 19:54.47 | pabelanger | pb the output of *CLI> core show version |
| 19:56.56 | cj | pabelanger: have you tried using res_crypto.so on these .deb packages? |
| 19:58.37 | pabelanger | cj: using it how? |
| 19:58.58 | gruvfunk | Asterisk 1.8.3.3-1digium1~lucid built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-04-22 00:43:36 UTC |
| 20:00.00 | cj | it doesn't seem to be loading the keypairs I've got in /var/lib/asterisk/keys |
| 20:01.14 | cj | pabelanger: also, did the deb-src repository get pulled, too? I was going to apt-get source asterisk-1.8 and build with debugging symbols so I could step through res_crypto.so |
| 20:03.10 | pabelanger | gruvfunk: $ dpkg -l | grep asterisk |
| 20:03.48 | pabelanger | cj: no, it should be there |
| 20:04.44 | gruvfunk | pabelanger: http://pastebin.com/sQMu51Fj |
| 20:04.52 | pabelanger | cj: DEB_BUILD_OPTIONS="debug" apt-get -b source asterisk-1.8 |
| 20:05.30 | cj | $ grep asterisk /etc/apt/sources.list |
| 20:05.30 | cj | deb http://packages.asterisk.org/deb squeeze main |
| 20:05.30 | cj | deb-src http://packages.asterisk.org/deb squeeze main |
| 20:05.34 | cj | E: Unable to find a source package for asterisk-1.8 |
| 20:05.54 | *** join/#asterisk irroot (~irroot@41.125.132.126) |
| 20:06.20 | pabelanger | gruvfunk: Hmm, looks like a result of some changes I made. You may need to back up your configs and purge asterisk, then install |
| 20:06.27 | pabelanger | cj: DEB_BUILD_OPTIONS="debug" apt-get -b source asterisk |
| 20:06.33 | pabelanger | drop -1.8 |
| 20:06.39 | cj | alright. thanks. |
| 20:06.45 | cj | cool. looks good. |
| 20:07.04 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:07.09 | pabelanger | cj: but will fix it |
| 20:07.10 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
| 20:07.10 | irroot | evening folks from the bush just got back from mid winter open veichle 2hr game drive through the pilanesberg was awesome |
| 20:07.52 | gruvfunk | pabelanger: so a fresh install should lay down 1.8.4.2 ? |
| 20:08.12 | pabelanger | gruvfunk: do you care about your existing configs? |
| 20:08.34 | pabelanger | If not... |
| 20:08.44 | pabelanger | $ sudo apt-get purge asterisk* |
| 20:08.53 | pabelanger | $ apt-get update |
| 20:09.06 | pabelanger | $ sudo apt-get install asterisk |
| 20:10.06 | gruvfunk | of course I do :) |
| 20:10.14 | cj | hurm. debug uses both -g and -O2... is this known to work? |
| 20:10.32 | pabelanger | gruvfunk: then copy your /etc/asterisk dir somewhere safe |
| 20:11.23 | gruvfunk | ya, in addition to custom sound files, yada yada |
| 20:11.32 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-154-252.dsl.stlsmo.sbcglobal.net) |
| 20:11.48 | pabelanger | cj: unknown honestly, I just added debug support recently. So, there maybe some issues with compiler flags that need to be removed |
| 20:12.40 | cj | ok. I'll let you know if I run into weird optimization problems |
| 20:14.44 | cj | pabelanger: btw, /etc/init.d/asterisk-1.8 restart fails if a client console is connected and paused. should I file this with reportbug? |
| 20:15.23 | cj | and by paused, I mean sent to terminal background with SIGSTOP/^z |
| 20:15.56 | pabelanger | cj: not reportbug, but you can use JIRA |
| 20:16.04 | cj | sounds good. thanks. |
| 20:34.24 | DrDigital | I cant seem to find the default login/pass for an Aastra 6739i |
| 20:35.21 | DrDigital | eh just found it |
| 20:35.27 | DrDigital | admin 22222 |
| 20:36.00 | DrDigital | anyone here happen to be familure with this phone? |
| 20:37.32 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 20:38.29 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 20:47.35 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 20:53.17 | cj | pabelanger: do you have a link to the debian package component? |
| 20:53.32 | cj | can't find it, but I've become unfamiliar with JIRA since last I used it |
| 20:54.43 | cj | ah. maybe filed under AsteriskNOW? |
| 20:55.17 | pabelanger | cj: AsteriskNOW is fine |
| 20:57.24 | lupestro | I think I gotta check my facts here - SIP subscriptions - URL and To: should both be the URL of the phone whose dialogs or presence I'm watching? (e.g. cortland@mydomain.org) |
| 20:57.42 | lupestro | Or should it be some extension hint or something? |
| 20:58.13 | lupestro | A little confused as to what Asterisk is looking for. |
| 20:58.46 | lupestro | "Looking for fifi in LocalSets (domain denis.lupestro.net)" |
| 20:58.55 | lupestro | "SIP/2.0 404 Not Found" |
| 21:05.16 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:05.44 | Wiretap7 | DrDigital, I'm not generaly awake at 4am on a saturday :P |
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| 21:26.09 | scalex000 | how to create a configuration to ring back a call when ext noanswer |
| 21:26.20 | scalex000 | to the origin ext |
| 21:34.34 | doolittlework | scalex000: catch the callerid then pass call back to user |
| 21:35.03 | russellb | scalex000: http://ofps.oreilly.com/titles/9781449303822/c02-CallControl_id302603.html#c02-CallControl_id379499 |
| 21:41.08 | scalex000 | thank you |
| 21:41.18 | scalex000 | I need callback too |
| 21:41.26 | scalex000 | but I mean i another system |
| 21:41.45 | scalex000 | when u call an extention but he user not reach after some ring |
| 21:41.51 | scalex000 | the call ring back |
| 21:42.45 | DrDigital | i tried to find out what time zone you are in |
| 21:42.45 | WIMPy | You mean when doing a blind transfer? |
| 21:43.45 | WIMPy | ctcp time? |
| 21:45.08 | russellb | scalex000: that's what the example does that i showed you |
| 21:45.48 | DrDigital | how do you upgrade the firmware on a PAP2T |
| 21:46.35 | DrDigital | i got 3.1.15 as the firmware and they are at like 5.1.6 |
| 21:46.46 | scalex000 | ok |
| 21:46.56 | scalex000 | russellb, thank you |
| 21:50.13 | scalex000 | russellb, let me ask you something, I have a nortel connect with my asterisk pbx, in my country to dial a mobile phone u need to dial 1 first and the number |
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| 21:51.02 | scalex000 | russellb, for some reason when dial throught asterisk I can hear the message from nortel, its only ring and ring |
| 21:52.36 | scalex000 | russellb, I connect both using SIP protocol |
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| 22:30.46 | scalex000 | asterisk 1.6.2.... its compatible with 1.6.0.28 |
| 22:30.48 | scalex000 | ? |
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| 23:11.54 | jc319 | Hey just adding voicemail feature (and send-to-email works, cool!) I'm using this code from the free book: |
| 23:11.54 | jc319 | exten => 101,1,Dial(${JOHN},10) |
| 23:11.54 | jc319 | exten => 101,n,VoiceMail(101@default,u) |
| 23:12.37 | jc319 | when user calls on cisco phone, the setting *86 from SIP<MAC>.cnf works and it reachs voicemail prompt. However needs to enter mailbox ID, is there no way to bypass this? |
| 23:13.02 | jc319 | whatever extension it calls from (101@default) should be the mailbox we will be entering, so why does it prompt? |
| 23:15.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:16.20 | paulc | jc319: do a "core show application voicemailmain" - it will show you how to pass in the mailbox number, and skip the password. |
| 23:16.45 | paulc | jc319: You can do some clever stuff with "if mailbox exists", based on the caller ID, to drop them straight into their mailbox if they have one, or prompt for one if not. |
| 23:17.55 | jc319 | OK sorry I got confused, added (hardcoded number 201@default to extension 201) but missed adding ${EXTEN} to *86 exten's definition |
| 23:19.15 | jc319 | paulc: Thanks for replying. I am testing the plain mailbox now, there's no doubt it will work but still want to see first, then I will try this script: http://bernaerts.dyndns.org/linux/179-asterisk-voicemail-mp3 it seems to improve default voicemailbox-to-email feature a bit. |
| 23:20.09 | paulc | ah.. that's cool.. hadn't seen that page before - thanks for that :) |
| 23:20.34 | jc319 | Also I have got this code from book which seems to be a more efficient way of adding voicemail to user extensions because it looks more tidy. http://pastebin.com/Fbg5Hu0a |
| 23:20.58 | jc319 | Only 1 line per user rather than minimum 2. |
| 23:52.09 | cj | protip: cert dir is /usr/share/asterisk/keys on debian, not /var/lib/asterik/keys |