| 00:03.47 | file | a VoIP provider. |
| 00:05.55 | Psil0Cybin | oh any is fine?! |
| 00:06.03 | Psil0Cybin | that is fantastic! |
| 00:10.27 | Psil0Cybin | so then in other words, I just popped in to say how much I love you guys! and open source software. |
| 00:12.36 | *** join/#asterisk Draecos (~Draecos@58-7-60-116.dyn.iinet.net.au) |
| 00:14.14 | *** join/#asterisk Psil0Cybin (~psil0cybi@unaffiliated/psil0cybin) |
| 00:25.29 | *** join/#asterisk makubi (~makubi@xdsl-87-78-23-168.netcologne.de) |
| 00:29.11 | *** join/#asterisk jsjc (~Adium@212.Red-83-59-178.dynamicIP.rima-tde.net) |
| 00:32.06 | *** join/#asterisk MaliutaLap (~Finux@eth637.qld.adsl.internode.on.net) |
| 00:39.30 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 00:54.04 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 00:54.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:55.54 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 00:56.13 | snadge | kinda weird.. i have an asterisk server thats not really doing anything at the moment, but I logged into it, and asterisk is using 100% cpu |
| 00:56.27 | snadge | log into the console.. set verbosity high, nothing happening there |
| 00:58.03 | snadge | clearly its a pebkac issue, i know this.. i just thought someone may have seen this and can go.. that's probably this |
| 00:58.54 | Kobaz | so weird |
| 00:59.33 | lnb_ | would log stating 603 declined mean the call is blocked ? |
| 00:59.45 | [TK]D-Fender | no |
| 00:59.46 | lnb_ | err the number of the caller |
| 00:59.47 | Kobaz | sooooo |
| 00:59.48 | lnb_ | no |
| 00:59.52 | lnb_ | ? |
| 00:59.57 | Kobaz | i have a really weird problem |
| 00:59.58 | lnb_ | what is it then/ |
| 01:00.16 | Kobaz | very randomly i get one way audio with an echo on sip calls |
| 01:00.37 | lnb_ | wikopedia states: The destination does not wish to participate in the call, or cannot do so, and additionally the client knows there are no alternative destinations (such as a voicemail server) willing to accept the call. |
| 01:00.40 | [TK]D-Fender | lnb_: 603 Declined without answer. That is not a refusal |
| 01:00.43 | Kobaz | i get an invite to my box from an itsp, asterisk calls my polycom... pick up the call... and theeeeeen echo |
| 01:00.51 | Kobaz | [TK]D-Fender: what's a good way to debug this? |
| 01:01.09 | [TK]D-Fender | Kobaz: echo = itsp fault |
| 01:01.10 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:01.16 | Kobaz | ah |
| 01:01.31 | Kobaz | rtp debug is showing audio flowing both ways |
| 01:02.07 | [TK]D-Fender | Kobaz: Unless yous psycho-pumped your gains on the unit (which means provisioning-overrides of gain, not LEVEL) and are generating Acoustic Echo |
| 01:02.19 | Kobaz | haven't touched the gains |
| 01:03.08 | lnb | i can call the DID from my sip phone but when one of my clients tries that same DID, cli show 603 declined |
| 01:03.12 | [TK]D-Fender | Kobaz: ITSP = suck |
| 01:03.40 | [TK]D-Fender | lnb: Same answer as always... show the full call full SIP debug. |
| 01:03.56 | lnb | it was hours ago |
| 01:04.38 | [TK]D-Fender | Don't ask for an autopsy without providing a body otherwise you're just wasting everyone's time |
| 01:05.29 | lnb | i dont mind searching and i've found it, but there is a ton of entries in /var/log/asterisk/full |
| 01:05.37 | Kobaz | [TK]D-Fender: unless you go direct to a nice tier1 |
| 01:05.45 | Kobaz | [TK]D-Fender: level3 and lightpath have excellent service |
| 01:05.59 | Kobaz | tier2/3 resellers do tend to suck, yeah |
| 01:06.12 | Kobaz | and this is a tier3 that this is happening on |
| 01:06.29 | Kobaz | nbs is good too |
| 01:07.33 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 01:08.14 | [TK]D-Fender | I've also just about never seen any provider throw a 603 at a caller. |
| 01:08.30 | Kobaz | first time for everything |
| 01:09.04 | [TK]D-Fender | Yeah well, there's always another half to the story we don't hear with the first-round explanation too. |
| 01:09.58 | Kobaz | why does that happen i wonder |
| 01:10.14 | Kobaz | i've never gotten anything like that with using sangoma cards or adtran gateways |
| 01:10.23 | Kobaz | the one way audio with echo |
| 01:10.32 | Kobaz | i get that quite a bit with itsp's though |
| 01:10.42 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 01:12.31 | [TK]D-Fender | Heading out for a while, BBL |
| 01:16.12 | *** join/#asterisk TSM (~the_softw@fw-lon2.wenn.com) |
| 01:21.59 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 01:30.54 | lnb | [TK]D-Fender: you still here? |
| 01:33.02 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 01:35.43 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 01:45.58 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 01:58.14 | *** join/#asterisk Fwny (~potato@192-0-250-120.cpe.teksavvy.com) |
| 02:04.26 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 02:04.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.243) |
| 02:07.22 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 02:16.14 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 02:22.30 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 02:23.53 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 02:27.35 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 02:30.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.215) |
| 02:32.22 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:40.13 | *** join/#asterisk D30_ (~deo@222.127.13.226) |
| 02:57.29 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 03:07.01 | *** join/#asterisk chare (322f51ac@gateway/web/freenode/ip.50.47.81.172) |
| 03:07.11 | chare | I need free voip |
| 03:07.22 | chare | what do we do when google removes chat support for google voice |
| 03:10.14 | lnb | [TK]D-Fender: you here? |
| 03:10.57 | lnb | pay? |
| 03:22.19 | chare | anyanoe here? |
| 03:26.03 | WIMPy | There are tons of free voip options. Stop whining. Or put on a skirt and go to the basement. |
| 03:26.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.176) |
| 03:26.23 | ChannelZ | too late |
| 03:32.10 | chare | WIMPy everytime you answer my question you don't help |
| 03:32.27 | WIMPy | Do you know googles other services? |
| 03:32.34 | WIMPy | Like their serach engine? |
| 03:32.39 | WIMPy | Use that. |
| 03:36.12 | chare | I need something that I can hook in the regular phone into |
| 03:39.53 | *** join/#asterisk Akuma (~Akuma@137.175.233.59) |
| 03:43.34 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 03:44.41 | *** join/#asterisk menstraighting (~menstraig@nc-76-5-178-34.dhcp.embarqhsd.net) |
| 03:44.41 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 03:45.46 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 03:48.14 | chare | so you guys have no clue |
| 03:50.52 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 03:51.34 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 03:53.17 | saint_ | hi - is there anything particular for asterisk to take in account dadhi ? I installed the drivers on the linux box, and my meetme is still now working. The logs shows "Unable to open DAHDI pseudo device" |
| 03:54.49 | Kobaz | saint_: either a) dahdi kernel module is not loaded b) asterisk does not have permissions to /dev/dahdi or c) dahdi asterisk module is not loaded |
| 03:56.21 | saint_ | Kobaz mmh.. i don t see any /dev/dahdi .. |
| 04:03.23 | saint_ | i did not use the dahdi from digium, but i did a yum install dahdi* instead. i guess this is no good..? |
| 04:04.22 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 04:04.55 | snadge | im trying to figure out why when this guy calls mobiles from his elastix box via iax.. the call quality is terribad.. quiet and distorted |
| 04:05.02 | snadge | but when the same guy rings land lines.. no problem |
| 04:05.29 | snadge | the thing is.. his calls are being converted to sip, and trunking via our sip gateway.. and calls to mobiles through that, sound fine |
| 04:05.54 | snadge | if it were a codec issue.. i would assume that all calls would sound equally bad |
| 04:06.33 | snadge | its times like this.. where I start to think i've signed up for the wrong job ;) |
| 04:10.50 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 04:14.27 | snadge | so i'll just try the latest alpha freepbx distro |
| 04:14.49 | *** join/#asterisk MaliutaLap (~Finux@eth637.qld.adsl.internode.on.net) |
| 04:14.54 | snadge | register to the same server with iax.. then place a call to a mobile.. if it sounds good, then i'll tell him his setup is fail |
| 04:16.39 | *** join/#asterisk menstraighting (~menstraig@nc-76-5-178-34.dhcp.embarqhsd.net) |
| 04:29.40 | saint_ | Kobaz i installed from the doc, I have codec_dahdi.so and chan_dahdi.so loaded in asterisk, but i still get Unable to open DAHDI pseudo device |
| 04:32.41 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 04:32.44 | Kobaz | kernel module |
| 04:32.47 | Kobaz | lsmod | grep dahdi |
| 04:33.06 | saint_ | yeah, it s not listed |
| 04:33.12 | Kobaz | did you download, build and install dahdi-complete ? |
| 04:33.33 | Kobaz | well if the kernel module isn't loaded, it's not going to work at all |
| 04:33.34 | saint_ | i followed both the wiki and asteriskdocs |
| 04:33.39 | saint_ | understood |
| 04:33.52 | Kobaz | apparently you didn't follow something, otherwise it would be working |
| 04:33.53 | Kobaz | heh |
| 04:34.03 | Kobaz | something is missing |
| 04:34.07 | saint_ | i downloaded 2.6.1+2.6.1 |
| 04:34.26 | Kobaz | getting the latest is generally a good idea |
| 04:34.28 | saint_ | i did follow the README (make all / install / config) with root |
| 04:34.46 | Kobaz | which is 2.8.0.1 |
| 04:35.01 | Kobaz | and did you /etc/init.d/dahdi start? |
| 04:35.25 | saint_ | ha.. no ... mmhh. |
| 04:35.46 | saint_ | yyeeaahh |
| 04:35.50 | saint_ | now i see the module. |
| 04:35.52 | saint_ | damn it |
| 04:36.03 | saint_ | and it works |
| 04:36.06 | saint_ | thanks a ton , lol |
| 04:36.11 | Kobaz | heh |
| 04:36.15 | Kobaz | told you something was missing |
| 04:36.23 | Kobaz | sleep time |
| 04:44.16 | saint_ | all this for paging.. i was expecting auto answer on digium phones. can they do that ? |
| 05:04.13 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 05:04.15 | *** part/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 05:06.44 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 05:09.31 | *** join/#asterisk sattellite (~sattellit@stl.bks.tv) |
| 05:45.44 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 05:51.13 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 06:05.45 | *** join/#asterisk gerhard7 (~gerhard7@77-172-86-84.ip.telfort.nl) |
| 06:10.22 | *** join/#asterisk Psil0Cybin (~psil0cybi@unaffiliated/psil0cybin) |
| 06:16.39 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 06:16.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 06:17.13 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-llqyclevixcbdwkt) |
| 06:27.46 | *** join/#asterisk D30_ (~deo@222.127.13.226) |
| 06:35.33 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 06:36.01 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 06:38.21 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 06:38.46 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 06:43.03 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 06:43.47 | hrolf | Hi I'm executing the Record() application, it seems to be working fine, (i.e. no errors or warnings on console) but there is no file generated. |
| 06:44.10 | hrolf | I checked in whole /var/lib/ |
| 06:46.09 | kaldemar | when are you checking for the file existence? |
| 06:46.53 | [TK]D-Fender | Show us the complete call |
| 06:47.08 | hrolf | kaldemar: After I drop the call. |
| 06:47.19 | hrolf | [TK]D-Fender: Let me post the complete log that appears in console. |
| 06:47.24 | kaldemar | option k |
| 06:48.29 | hrolf | kaldemar: Yes. Thanks I remember now. Let me try that. |
| 07:06.28 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 07:17.11 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 07:17.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:21.55 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 07:23.17 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:24.44 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:31.48 | hrolf | <PROTECTED> |
| 07:31.58 | hrolf | What does this mean ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ returning 4? |
| 07:36.56 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 07:49.14 | ChannelZ | Pretty sure that means either the script hung up or the channel did while it was running. |
| 08:00.41 | hrolf | ChannelZ: Any documentation where I can further read up on the "return values" ? |
| 08:04.25 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 08:10.42 | ChannelZ | for that, besides the source, I don't know. ${AGISTATUS} contains a string with the result, see 'core show application AGI' |
| 08:15.02 | ChannelZ | The return code listed on the console is -1 Failure, 0 Success, 1 Success (Fast AGI), 2 Success (ASync AGI), 3 Not Found (script not found), 4 Hangup. |
| 08:18.48 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:26.10 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 08:33.17 | *** join/#asterisk bipolar (~bipolar@24.229.96.76) |
| 08:36.52 | *** join/#asterisk D30_ (~deo@222.127.13.226) |
| 08:37.37 | *** part/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 08:39.10 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 08:39.44 | *** join/#asterisk michael_work (~michael@bzq-112-168-31-118.red.bezeqint.net) |
| 08:40.33 | michael_work | is there any tip when i use app and when function? |
| 08:41.17 | WIMPy | You use Functions as arguments to applications. |
| 08:41.33 | michael_work | the point is i want to develop something small |
| 08:41.43 | michael_work | and i wanted to go with fucntion |
| 08:42.05 | michael_work | but i'm not sure if it's better to return value or to set channel value in function |
| 08:42.28 | michael_work | so first would be proper function |
| 08:42.29 | WIMPy | Use a function if you want to return a value. |
| 08:42.38 | michael_work | while second looks more like app :( |
| 08:42.51 | WIMPy | That's your choice, what you like better. |
| 08:43.05 | michael_work | ok thanks |
| 08:43.25 | michael_work | function can get several arguments as well right?> |
| 08:43.53 | WIMPy | Right. |
| 08:44.02 | michael_work | thanks :) |
| 08:44.31 | michael_work | the last question is - how one speed up some attention to bug in JIRA. |
| 08:44.42 | michael_work | there is even one approved for a long time there |
| 08:44.43 | *** join/#asterisk bitwize1 (~S2TS@h87-96-213-2.dynamic.se.alltele.net) |
| 08:44.53 | WIMPy | Fix it yourself or bribe someone. |
| 08:45.10 | michael_work | and there is one reported by me 100% reproducable crashing asterisk |
| 08:45.41 | michael_work | i found the way to fix it but as i'm kind of n00b not sure if my fix can break other stuff |
| 08:46.02 | michael_work | bribe is nice option, can we use dogecoins :P |
| 08:46.31 | WIMPy | Attach it to the issue and hope someone comments on it or request you to put it up on reviewboard. |
| 08:47.38 | michael_work | done for a long time |
| 08:47.47 | michael_work | 2 devs were involved and disappeared |
| 08:48.19 | michael_work | https://issues.asterisk.org/jira/browse/ASTERISK-22920 |
| 08:48.27 | WIMPy | Try to ask again later in #asterisk-bugs |
| 08:49.11 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:52.55 | michael_work | oh there is one of those |
| 08:52.57 | michael_work | thanks |
| 08:57.13 | WIMPy | Oh cool. |
| 08:57.32 | WIMPy | You had the sam scary set of events that I had last weekend. |
| 08:57.48 | michael_work | ? |
| 08:58.19 | WIMPy | Some weeks ago I upgraded from 11.7-rc1 to 11.7 and found that Asterisk locks up if I try to transfer a call. |
| 08:58.35 | michael_work | it doesn't locks up it crashes :) |
| 08:58.45 | WIMPy | I didn;t have to to investigate that time so I just downgraded until I tried to dig in to it last WE. |
| 08:58.55 | michael_work | oh |
| 08:59.32 | WIMPy | I upgraded to 11.7 again and it loced up gain. Downgraded to 11.7-rc1 and it was fine again, Upgrade again lock up again. |
| 08:59.37 | WIMPy | 100% reproducible. |
| 08:59.50 | michael_work | yeah but bug is different |
| 09:00.06 | WIMPy | Then I turned up verbose an debug to gather more information and since then it has never locked up again. |
| 09:00.25 | WIMPy | I can restart or even reinstall. It doesn't happen any more. |
| 09:00.52 | WIMPy | So it suddenly went from 100% reproducible to not reproducible at all. |
| 09:01.07 | WIMPy | That scares me. |
| 09:01.22 | michael_work | oh |
| 09:01.34 | michael_work | in my option it's still 100% there |
| 09:01.41 | michael_work | i just know how to solve it |
| 09:01.48 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 09:01.49 | michael_work | it crashes on SRTP TLS |
| 09:01.58 | WIMPy | I just read that you have another installation where it doesn't happen. |
| 09:02.03 | michael_work | i tried TLS without SRTP and it doesn't and doesn;'t transfer |
| 09:02.11 | michael_work | i fixed that but small patch(last comnment) |
| 09:02.21 | michael_work | and it seems fixed crashes as well in SRTP :P |
| 09:03.36 | michael_work | the last comment |
| 09:04.09 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:05.17 | *** join/#asterisk vittorio88 (~vittorio8@37.119.228.10) |
| 09:05.42 | WIMPy | So you just commented out a few lines? |
| 09:05.44 | *** join/#asterisk oatha (~oatha@201.22.210.28.dynamic.adsl.gvt.net.br) |
| 09:06.52 | WIMPy | I suspect that might not be the correct solution, but that's a thing you could ask in #asterisk-dev as well. |
| 09:07.24 | WIMPy | Again, when the guys are awake. |
| 09:07.26 | vittorio88 | Hi guys! I'm trying to set up sip/tls on asterisk 11.5.1 and am having a hard time. Asterisk does not start listening on port 5061. Res_crypto.so is loaded. Any ideas? |
| 09:08.22 | WIMPy | vittorio88: Did you configure it? What's the output if you (re)load/configure chan_sip? |
| 09:08.54 | vittorio88 | There is no mention of tls or crypto |
| 09:08.59 | WIMPy | Also 'sip show settings' |
| 09:09.12 | vittorio88 | I have been reloading using service asterisk restart |
| 09:09.46 | vittorio88 | Yes it is enabled using tlsenable=yes |
| 09:10.14 | WIMPy | Does it show with 'sip show settings'? |
| 09:10.25 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:11.20 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 09:11.20 | *** join/#asterisk jsjc (~Adium@212.Red-83-59-178.dynamicIP.rima-tde.net) |
| 09:11.26 | vittorio88 | Yes. Transport =tls and it says listening on 0.0.0.0:5061 |
| 09:11.54 | WIMPy | And you say it doesn't? |
| 09:11.57 | vittorio88 | 5061 but netstat -tuna|grep 5061 turns up blank |
| 09:13.04 | WIMPy | Same machine? |
| 09:13.13 | vittorio88 | Yeah |
| 09:13.18 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 09:13.19 | WIMPy | That doesn't make any sense to me. |
| 09:13.33 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 09:13.38 | *** join/#asterisk D30_ (~deo@222.127.13.226) |
| 09:13.54 | vittorio88 | Tell me about it. |
| 09:14.18 | WIMPy | You can try to crank up verbose and debug and do a 'sip reload' and look for anything suspicious. |
| 09:14.18 | *** join/#asterisk jsjc (~Adium@212.Red-83-59-178.dynamicIP.rima-tde.net) |
| 09:14.30 | WIMPy | That's the only idea I have. |
| 09:15.18 | WIMPy | Or even unload an load chan_sip with verbose and debug turned on. |
| 09:15.34 | vittorio88 | Trying... |
| 09:16.39 | *** join/#asterisk JeffC_NN (~jcollell@mail.nownerd.com) |
| 09:17.14 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 09:17.16 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 09:19.18 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 09:21.02 | vittorio88 | Is there a way to crank verbosity other than asterisk -vvvvvvvr??? |
| 09:21.23 | WIMPy | core set verbose ... |
| 09:21.26 | vittorio88 | I have debug enabled in sip.conf |
| 09:21.33 | vittorio88 | Thanks |
| 09:21.43 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 09:21.48 | WIMPy | No, core set debug ..., not sip debug. |
| 09:22.35 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:31.33 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:33.45 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 09:40.39 | vittorio88 | Thanks wimpy for the help. I got an ssl error accessing ghe key file. However it only printed once and i can no longer get it to print. |
| 09:41.12 | vittorio88 | I dont know why but it is not a perm issue and the file exists. |
| 09:41.14 | tparcina | What happens if Asterisk is using external database for CDR records (either odbc or pgsql) and the database is down for some period of time? |
| 09:41.27 | vittorio88 | i will perform further tests and write back. |
| 09:44.13 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 09:48.31 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 09:50.51 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:57.06 | skrusty | morning |
| 10:07.42 | *** join/#asterisk pluesch0r (~pluesch0r@chello084112018119.27.11.vie.surfer.at) |
| 10:07.58 | *** join/#asterisk Moz (Moz@cpc1-yead1-0-0-cust45.7-1.cable.virginm.net) |
| 10:08.45 | pluesch0r | hi everybody. running asterisk 1.8.11 over here and need to log cdr records to sqlite3. i've found the config file for that in /etc/asterisk/cdr_sqlite3_custom.conf, but asterisk doesn't log anything to that file. sqlite3 is installed on that host. what could i be doing wrong? |
| 10:10.47 | pluesch0r | i've only got cdr_sqlite.so in the modules dir, there's no cdr_sqlite3_custom module. is that module a prerequisite? does that module even exist? |
| 10:13.52 | lnb | pluesch0r: dont know the answer to your question but just wondering why you want sqlite over mysql. |
| 10:14.27 | pluesch0r | lnb: because all i want are cdr records and i don't wan to run a db server for just that single purpose. |
| 10:14.51 | pluesch0r | lnb: because i don't want to deal with asterisk not re-connecting to the db server when the db server is restarted as it does with postgres. |
| 10:15.06 | pluesch0r | lnb: because i'm fed up with experimentes that break the delivery of cdr information to my ticket system. |
| 10:15.23 | pluesch0r | lnb: :) |
| 10:15.44 | lnb | since I started with voip, i have yet to see mysql or postres fail on any of our servers |
| 10:16.18 | pluesch0r | lnb: it's not failing. it's restarting because of the installation of a security update. |
| 10:16.19 | lnb | and since 2003 i have yet to see postgres die on our server configurator site |
| 10:16.37 | pluesch0r | lnb: i didn't say "die" or "fail" anywhere. |
| 10:16.46 | lnb | no yet didnt |
| 10:16.48 | pluesch0r | anyways. cdr logging to sqlite3 on asterisk 1.8 anybody? |
| 10:16.57 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 10:17.21 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 10:18.08 | lnb | try this -> http://www.voip-info.org/wiki/view/Asterisk+cdr+SQLite |
| 10:19.07 | pluesch0r | lnb: i appreciate you trying to help me but that wiki entry is meant for sqlite2 and i _really_ need information from somebody who knows how to log to sqlite3 on asterisk 1.8 |
| 10:19.59 | *** join/#asterisk DataWraith (~s1gny@188-194-90-27-dynip.superkabel.de) |
| 10:46.20 | *** join/#asterisk Zogot (~LeonProna@095-097-219-235.static.chello.nl) |
| 10:46.24 | Zogot | Ahoyhoy |
| 10:47.02 | *** join/#asterisk Cynagen (~cynagen@ip70-190-135-8.ph.ph.cox.net) |
| 11:06.26 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:06.40 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:08.03 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 11:14.45 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 11:18.49 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
| 11:42.57 | *** join/#asterisk bhuddah (~bhuddah@unaffiliated/bhuddah) |
| 11:42.59 | *** part/#asterisk bhuddah (~bhuddah@unaffiliated/bhuddah) |
| 11:46.17 | *** join/#asterisk tedstriker (~tedstrike@host-135-196-33-208.lines.viateldsl.com) |
| 11:48.45 | *** join/#asterisk kippi (c2325add@gateway/web/freenode/ip.194.50.90.221) |
| 11:48.48 | kippi | hey |
| 11:49.55 | kippi | I have a ISDN2 unit calling into my sip channel, this is then coming into asterisk, all I am getting is static, not getting any audio, the unit is saying it can't frame, any pointers? |
| 11:53.04 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 11:53.17 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3523:8a68:acec:efd5) |
| 11:54.58 | *** join/#asterisk Draecos (~Draecos@58-7-60-116.dyn.iinet.net.au) |
| 11:56.06 | Chainsaw | kippi: An "ISDN2 unit". You're going to have to be more specific. |
| 11:56.16 | Chainsaw | kippi: Is it an ISDN NT1 that you have connected to a BRI PCI card? |
| 11:56.29 | Chainsaw | kippi: Or is it a BRI to SIP gateway device? Does it have a brand? |
| 12:02.19 | tparcina | I have been searching for answer, but I can't find any. Question is, what happens if Asterisk is using external database for CDR records (either odbc or pgsql) and the database is down for some period of time? |
| 12:03.01 | Chainsaw | tparcina: Terrible things. Unspeakable evil. |
| 12:03.30 | tparcina | Chainsaw: Thank you. |
| 12:03.32 | Chainsaw | tparcina: Like the call completing but you having no record of it, and thus not billing for it. |
| 12:03.32 | tparcina | Chainsaw: Are you a writer? |
| 12:04.10 | Chainsaw | tparcina: Worse. I'm a middle manager. |
| 12:04.27 | tparcina | Chainsaw: And Asterisk won't keep data, so that he can sent it when connection is restored? |
| 12:04.28 | MaliutaLap | Chainsaw: begone evil creature :P |
| 12:04.47 | Chainsaw | tparcina: There is no buffering. If you cannot reliably run a database, do not use it for billing or other critical data. |
| 12:05.51 | *** join/#asterisk CeBe (~CeBe@port-92-206-64-217.dynamic.qsc.de) |
| 12:06.10 | tparcina | Chainsaw: Is there any documentation about this behavior (expect the source code :))? |
| 12:06.48 | Chainsaw | tparcina: Your conditions preclude me from answering you. |
| 12:07.17 | tparcina | Chainsaw: :) |
| 12:07.48 | tparcina | Chainsaw: And it there any truth in what the sipwiz has said - http://stackoverflow.com/questions/4026924/whats-the-best-architecture-for-cdr-integration-to-a-asterisk-based-application/4038546#4038546 |
| 12:08.22 | tparcina | Chainsaw: sipwiz: If there's a problem Asterisk will block while it keeps trying to connect to the database to write the CDR's. |
| 12:11.38 | Chainsaw | tparcina: Either it blocks completely and you have an outage because your database has an outage... |
| 12:11.52 | Chainsaw | tparcina: Or it does something else and you are placing calls that you cannot bill. |
| 12:12.14 | Chainsaw | tparcina: I cannot personally guarantee which of the two will occur, but either of them would fall under the banner of terrible things. |
| 12:13.14 | *** join/#asterisk BenC[UK] (~bcummins@213.137.20.152) |
| 12:13.18 | tparcina | Chainsaw: What does it depend will the Asterisk block, or just the calls won't be recorded in database? |
| 12:13.32 | BenC[UK] | Hi guys |
| 12:13.44 | BenC[UK] | I've got an asterisk server, which all of a sudden has start segfaulting |
| 12:13.49 | tparcina | BenC[UK]: Hi! |
| 12:17.29 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 12:18.36 | Chainsaw | tparcina: That is the million dollar question. |
| 12:21.53 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 12:21.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:39.57 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 12:46.00 | *** part/#asterisk Farkie (~Farkie@213.229.88.2) |
| 12:47.54 | *** join/#asterisk goddva (~glarsen@77.40.154.242) |
| 12:48.21 | goddva | extension files vs fastagi ? |
| 12:48.44 | goddva | should I build my new features in fastagi (java), or keep the extension files? |
| 12:50.32 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 12:56.24 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 12:59.05 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 13:04.28 | BenC[UK] | anyone got any suggestions on how to find out whats causing asterisk to seg fault on start pu? |
| 13:04.33 | BenC[UK] | this box has been running fine for months |
| 13:04.35 | *** join/#asterisk menstraighting (~menstraig@nc-76-5-178-34.dhcp.embarqhsd.net) |
| 13:04.39 | BenC[UK] | all of a sudden it dies and won't start |
| 13:04.42 | BenC[UK] | no other updates have been done |
| 13:04.51 | Chainsaw | BenC[UK]: You could build it with debugging support, so that you get a backtrace. |
| 13:05.07 | Chainsaw | BenC[UK]: If you're running selinux or gentoo hardened, remember to allow core dumps for the process. |
| 13:05.17 | BenC[UK] | I did that, but I don't seem to get anything more in the core dump |
| 13:06.28 | Chainsaw | BenC[UK]: Would this be an appropriate moment to point out that you haven't shown me the core dump yet? |
| 13:08.33 | BenC[UK] | http://pastebin.com/raw.php?i=9RVWbEUU |
| 13:08.42 | BenC[UK] | this is the GDB output for the core |
| 13:13.49 | BenC[UK] | was that any good chainsaw? or do you want the raw dump? |
| 13:15.18 | mjordan | That's not a full backtrace |
| 13:16.05 | *** part/#asterisk mjordan (~matt@75.76.55.191) |
| 13:16.06 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 13:16.17 | *** join/#asterisk makubi (~makubi@xdsl-213-196-217-41.netcologne.de) |
| 13:17.19 | BenC[UK] | ok, let me try sorting it again |
| 13:22.07 | BenC[UK] | hmm, nope I can't seem to get anymore than that |
| 13:22.16 | BenC[UK] | I've recompiled asterisk with the compiler flags for debug etc |
| 13:22.22 | BenC[UK] | is there something else I need to do |
| 13:24.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.215) |
| 13:25.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.215) |
| 13:27.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.215) |
| 13:32.15 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:39.14 | Tuju | crap, my server has run out space because of asterisk logs. |
| 13:39.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:39.41 | Tuju | is there an option to gzip files without creating another file next to it? |
| 13:43.28 | phpboy | what do you mean? |
| 13:44.14 | Tuju | something that it would start overwriting the same inodes on disk with compressed stuff. |
| 13:44.27 | Tuju | but i searched it and i think that's not possible. |
| 13:44.46 | Tuju | luckily i've some space on another filesystem where i can compress it. |
| 13:45.25 | Tuju | but this is a real problem, in another asterisk instnace i get - as we speak - some 5-10 registration attempts per second. |
| 13:46.04 | Tuju | i guess it would be good to have some dynamic iptables blocking and system to distribute those ip addresses. |
| 13:47.08 | wdoekes | Tuju: theoretically you could gzip a file inline, but the solution to your problem would be to never let the disk get that full in the first place |
| 13:47.28 | Tuju | wdoekes: i agree |
| 13:47.45 | wdoekes | did you configure logrotate? |
| 13:48.11 | Tuju | wdoekes: well now i did, but that system doesn't have the Grand Canyon for those log lines. |
| 13:48.59 | Tuju | if some asswipe runs that registration attempt hours per day, it's quite clear that it runs out-of-space. |
| 13:50.31 | Tuju | i just love the internet, it exposes whole world to you and likewise. |
| 13:52.30 | Tuju | huh, compressing those logs with gzip really helped a lot. |
| 13:52.40 | Tuju | i guess those repeat the same all over again. |
| 13:53.54 | Tuju | and once i get asterisk running again, big flood of that same crap again. |
| 13:54.36 | Tuju | 207.244.71.171 |
| 13:56.39 | *** part/#asterisk bitwize1 (~S2TS@h87-96-213-2.dynamic.se.alltele.net) |
| 13:58.34 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
| 13:59.53 | [TK]D-Fender | And why isn't he firewalled off? |
| 14:00.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.198) |
| 14:01.45 | lnb | [TK]D-Fender: the 603 declined was a problem with a rate card. it did not have the npanxx client was calling. So you were right in that it had nothing to do with call being rejected by the callee. |
| 14:02.42 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:03.11 | [TK]D-Fender | lnb: And as I wrote later... this is because you got it from YOUR server due to hitting the dialplan but not doing anything that would actually generate a status |
| 14:03.23 | [TK]D-Fender | lnb: So it completely fell through |
| 14:03.42 | [TK]D-Fender | lnb: Which was part of the comment about the other parts of the story we never get when you ask these things |
| 14:04.00 | [TK]D-Fender | lnb: Which is the fact you're talking to a second server. |
| 14:04.17 | [TK]D-Fender | lnb: Where everyone is going to be thinking this is to a provider |
| 14:05.03 | lnb | [TK]D-Fender: when I searched in search engine for the term, wikopedia said .... |
| 14:05.36 | lnb | http://en.wikipedia.org/wiki/List_of_SIP_response_codes |
| 14:06.13 | lnb | after reading that, i presumed the call failed because the calling DID was blocked |
| 14:06.32 | Katty | morning |
| 14:06.35 | [TK]D-Fender | You clearly weren't looking at your own server |
| 14:06.50 | lnb | i have the call so you can see... it took a while to figure out how to get the ^M out of the full file |
| 14:06.56 | [TK]D-Fender | Where you'd see it hit the dialplan look at stuff and not have done anything |
| 14:07.11 | [TK]D-Fender | Asterisk doesn't know shit about "blocked" |
| 14:07.18 | [TK]D-Fender | There is no "blocked" in Asterisk |
| 14:07.42 | lnb | [TK]D-Fender: what can i tell you other than look what wikopedia says in the url above |
| 14:07.58 | lnb | i assumed (wrongly) it was correct |
| 14:08.52 | [TK]D-Fender | You didn't look at your own server. There's just no excusing this... |
| 14:09.19 | lnb | [TK]D-Fender: http://pastebin.ca/2577674 |
| 14:10.13 | [TK]D-Fender | Where we can clearly see it accepting the call into the dialplan |
| 14:10.16 | [TK]D-Fender | And processing it |
| 14:10.35 | lnb | what line is that |
| 14:11.02 | [TK]D-Fender | [Jan 22 17:05:10] VERBOSE[2824] chan_sip.c: Looking for 16477104307 in a2billing (domain 107.6.51.151) |
| 14:11.04 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 14:11.10 | [TK]D-Fender | it is accepting into the dialplan |
| 14:11.22 | [TK]D-Fender | [Jan 22 17:05:10] VERBOSE[18187] pbx.c: -- Executing [16477104307@a2billing:1] DeadAGI("SIP/2965713865-000285d4", "a2billing.php,1") in new stack |
| 14:11.26 | [TK]D-Fender | and executing dialplan. |
| 14:11.49 | lnb | well perhaps it does, but then it goes looking for for the 'plan' the client is in, and in that plan there are 'rate' sheets |
| 14:12.15 | lnb | when I went into the a2b gui, can simulate calls |
| 14:12.39 | lnb | put in the phone number they were calling, used MY account # and it found the rate no problem |
| 14:12.56 | lnb | when I did same but used thier account number.... NO RATE FOUND! |
| 14:13.01 | lnb | it then would kill the call |
| 14:13.21 | [TK]D-Fender | Whatever problem A2B has isn't our concern here. It wasn't blocked so the rest is up to you and it |
| 14:13.30 | [TK]D-Fender | Asterisk took the call |
| 14:13.40 | [TK]D-Fender | And at least had dialplan to process it |
| 14:13.45 | lnb | yes |
| 14:13.48 | lnb | that is true |
| 14:14.22 | lnb | but you know telco well, if one doesnt tell the telco they want to call anywhere in the world, they don't open it up. |
| 14:14.45 | Tuju | [TK]D-Fender: i routed those pkgs to null. |
| 14:14.46 | lnb | my problem is/was it was missing the NPANXX |
| 14:15.03 | Tuju | now it wont fill firewall logs either. |
| 14:15.05 | lnb | in the cvs rate sheetp |
| 14:15.08 | lnb | sheet |
| 14:15.13 | Tuju | sheep |
| 14:15.44 | lnb | added it, uploaded it to a2b and then the client can make the calls to that number |
| 14:15.50 | lnb | and they did this morning |
| 14:15.54 | tparcina | So I have read that, when * is saving CDR's to external database (odbc module), if the database is unvailable, * won't memorize CDR data, and sometimes * can be unresponsive. |
| 14:16.11 | [TK]D-Fender | "The destination does not wish to participate in the call" <- indeed Asterisk didn't actually "participate. It was never answered, didn't general ANY status on its own or as a pass-through (busy, congestion, etc). That's what you get after * give them a 100 TRYING for "let me go to the dialplan and look what I can do for this" and doesn't actually DO anything that the other side can tell. |
| 14:16.57 | [TK]D-Fender | tparcina: You mean the DB can sometimes be unresponsive.... |
| 14:17.01 | tparcina | What does it depend on, will * "just" forget the CDR's, or he will stop be responsive as well? |
| 14:17.23 | lnb | i wish it had a more intuitive reply so one can more easlily find the issue |
| 14:17.38 | lnb | especially with north america numbers |
| 14:18.06 | tparcina | [TK]D-Fender: I have read that if connection to DB fails, that * will become unresponsive as well. Is this true or not? |
| 14:18.31 | [TK]D-Fender | tparcina: I'm curious... link it... |
| 14:18.43 | tparcina | [TK]D-Fender: Wait a min. |
| 14:18.58 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 14:19.42 | lnb | even in the log pasted, a2b should have some sort of code that it doesn't have the NPANXX |
| 14:19.47 | Greenlight | In my experience asterisk does not fail or crash if the CDR db goes offline |
| 14:19.59 | tparcina | [TK]D-Fender: http://stackoverflow.com/questions/4026924/whats-the-best-architecture-for-cdr-integration-to-a-asterisk-based-application/4038546#4038546 |
| 14:20.03 | tparcina | [TK]D-Fender: sipwiz: If there's a problem Asterisk will block while it keeps trying to connect to the database to write the CDR's. |
| 14:20.42 | [TK]D-Fender | tparcina: 2010.... |
| 14:21.06 | tparcina | [TK]D-Fender: But it's true that * won't memorize any CDR data? |
| 14:21.18 | Greenlight | Batch mode will |
| 14:21.28 | tparcina | [TK]D-Fender: You are trying to say that's not the issue anymore? |
| 14:21.57 | [TK]D-Fender | tparcina: If this was a problem in 2010 do you think we're here in 2014 and they HAVEN'T looked at this? |
| 14:22.03 | BenC[UK] | I've uploaded a better backtrace now.. |
| 14:22.04 | BenC[UK] | http://pastebin.com/raw.php?i=YeHSMY7M |
| 14:22.10 | BenC[UK] | last thing in there is XML |
| 14:23.37 | tparcina | [TK]D-Fender: I was searching for any information what would happen if * can't connect to external database, and I haven't find any page that says those two things: |
| 14:23.54 | tparcina | [TK]D-Fender: 1. Asterisk (phone calls) will work completely normaly. |
| 14:24.15 | Greenlight | It's very easy to test... |
| 14:24.25 | Greenlight | ... or even look at the code |
| 14:24.31 | [TK]D-Fender | tparcina: Do you know how many books aren't writting about things that don't matter? :) |
| 14:24.35 | [TK]D-Fender | written* |
| 14:24.54 | tparcina | [TK]D-Fender: 2. When database is available again, * will send all the data that wasn't delivered because the database was unavailable. |
| 14:24.57 | [TK]D-Fender | tparcina: Nobody is going to write NON-news |
| 14:25.09 | [TK]D-Fender | [09:21]GreenlightBatch mode will <- |
| 14:25.22 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 14:25.33 | Chainsaw | Greenlight: Careful, tparcina doesn't like that "look at the code" answer. I tried that earlier. |
| 14:25.49 | Greenlight | Heh :) |
| 14:26.06 | Greenlight | It's often the only sure fire way to see |
| 14:26.14 | Chainsaw | nods in agreement |
| 14:26.22 | tparcina | Chainsaw: Not every * user is C programmer. |
| 14:26.45 | Chainsaw | tparcina: You don't have to write C programs from scratch, you just have to read existing ones. |
| 14:27.09 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-pdrwwaibmfaceffo) |
| 14:27.15 | Greenlight | Quite. There's big difference between writing the code, and loosely following it, or even reading the comments... |
| 14:27.15 | *** join/#asterisk gerhard7 (~gerhard7@77-172-86-84.ip.telfort.nl) |
| 14:27.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.194) |
| 14:27.40 | tparcina | Chainsaw: I though that's enough to read documentation, not the source code. |
| 14:27.41 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/cdr.conf.sample?revision=371121&view=markup |
| 14:27.42 | Greenlight | I'd assume that even in batch mode there's an upper limit before you get out of memory |
| 14:27.51 | [TK]D-Fender | tparcina: what does this documentation say? |
| 14:27.55 | jaflong | Any tips on how to convert a standard h263/h264 format video file to asterisk compatible video format |
| 14:28.36 | [TK]D-Fender | jaflong: Those are * formats |
| 14:28.55 | tparcina | [TK]D-Fender: I'll read it and then I'll tell you. |
| 14:30.23 | tenspeed705 | Hey do we have anyone here from a carrier? |
| 14:30.54 | [TK]D-Fender | tenspeed705: ariel works on ships... not sure on the aircraft side though.... |
| 14:32.02 | jaflong | [TK]D-Fender: The files do not play over music on hold. |
| 14:32.17 | tenspeed705 | hehe. I meant a voip carrier/provider |
| 14:32.58 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 14:33.28 | [TK]D-Fender | tenspeed705: continue onward to your actual question before isolating your audience or apparent willingness to see what you get. |
| 14:34.41 | [TK]D-Fender | Any experts in X? Question that does not actually require expected and claimed level of expertise in X. |
| 14:35.08 | [TK]D-Fender | Meta-questions = built-in answer reduction. |
| 14:35.19 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:35.19 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:35.28 | [TK]D-Fender | not healthy |
| 14:35.37 | [TK]D-Fender | so counter-productive |
| 14:35.44 | [TK]D-Fender | much sad |
| 14:37.47 | tenspeed705 | we are having conversation in our office as to what extent of proof we need to dig up when customers dispute long distance calls. Also curious as to how out providers perpare or handle these claims. |
| 14:38.49 | [TK]D-Fender | They have their billing CDR's on their side, you have yours. They may have multiple levels of those logs as it passes through severl points in their archetecture. |
| 14:39.19 | [TK]D-Fender | Thta's what it's all about. Logs showing calls coming in, source IP's, account numbers, in the case of SIP, user-agent's etc... |
| 14:39.37 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 14:39.39 | [TK]D-Fender | Was X dialed? that's what logs are for. |
| 14:40.02 | [TK]D-Fender | Was it account X? Where was the call from? Log of registration status, etc |
| 14:42.33 | tenspeed705 | all good points, I suggested most of those as well. The Useragent is a good key point. Thanks |
| 14:43.51 | tenspeed705 | I think something worth looking in to as well, is policy in place to prevent calls being forwarded outside of local area |
| 14:44.13 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 14:44.14 | *** part/#asterisk leedm777 (~leedm777@nat/digium/x-pdrwwaibmfaceffo) |
| 14:45.43 | [TK]D-Fender | Optional services to limit call destinations are just that. |
| 14:46.08 | [TK]D-Fender | Also things like throttling, and duration safeguards |
| 14:51.42 | tparcina | [TK]D-Fender: It doesn't say much. |
| 14:52.16 | tparcina | [TK]D-Fender: It says that * can save x CDR records before trying to send them to the database. |
| 14:52.46 | [TK]D-Fender | tparcina: Keep reading |
| 14:53.06 | tparcina | [TK]D-Fender: Or, he can send every y minutes (whichever comes first). |
| 14:53.36 | tparcina | [TK]D-Fender: But the page doesn't say what would happen if database is unavailable. |
| 14:54.00 | [TK]D-Fender | tparcina: It does give you a pretty good idea. |
| 14:54.02 | [TK]D-Fender | tparcina: Keep reading |
| 14:54.06 | tparcina | [TK]D-Fender: I have read the whole page (carefully). |
| 14:54.52 | [TK]D-Fender | tparcina: Apparently not. |
| 14:55.25 | tparcina | [TK]D-Fender: Can you please tell me what have I missed? |
| 14:55.26 | [TK]D-Fender | tparcina: "If there are any problems, usually, the entry will silently ignored, and you get no output." <------------- |
| 14:57.11 | [TK]D-Fender | tparcina: I tells you clearly to expect no recovery unless the back-end config tells you otherwise |
| 14:57.14 | [TK]D-Fender | It* |
| 14:57.15 | tparcina | [TK]D-Fender: That part is about conf file, not about how the data is sent to the database. |
| 14:57.35 | [TK]D-Fender | tparcina: that is about the BACKEND... Databases ARE a backend |
| 14:58.14 | [TK]D-Fender | tparcina: So if you see nothing in the specific backend module for caching, recovery, etc, then you are told to expect none. |
| 14:59.30 | tparcina | [TK]D-Fender: Thank you for your time and willingness to help me understand this. |
| 15:00.06 | [TK]D-Fender | tparcina: If you are looking for a solution to possible issues on your side, use MULTIPLE backends and check them against each other |
| 15:01.02 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 15:02.13 | tparcina | [TK]D-Fender: Yes, I'll have to evaluate whole thing with my colleagues. |
| 15:02.28 | tparcina | [TK]D-Fender: Thank you once more. |
| 15:08.19 | *** join/#asterisk mjordan (~matt@nat/digium/x-wqbwlsuwciorwnso) |
| 15:08.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:09.12 | *** join/#asterisk Spengler1 (~spengler@dns1.aacc.edu) |
| 15:09.50 | Spengler1 | hi guys ; just wanted to say thanks for all of the help you guys have given in the past few months. i deployed my first asterisk server a few weeks ago and everything is working great. customer is very happy. |
| 15:10.45 | *** join/#asterisk serafie (~erin@nat/digium/x-gzkihddrlmqnazdz) |
| 15:12.01 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:18.55 | Chainsaw | Spengler1: Happy customers are paying customers. Excellent :) |
| 15:24.35 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:27.36 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.236) |
| 15:29.06 | navaismo | ~sla |
| 15:29.06 | infobot | well, sla is service level agreement, or shared line appearances |
| 15:40.26 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 15:42.22 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-oyxokxqyoolaosps) |
| 15:44.55 | *** join/#asterisk BCS-Satori (~BCS-Sator@2001:470:e169:860:6e3b:e5ff:fe38:da53) |
| 15:46.52 | kippi | I am using a sip gateway (voiptalk), I have a DDI number with them, I call in from my ISDN device to this number, this is forwarding it to my SIP trunk on asterisk, When I listen to the call I am just getting white noise and the ISDN unit that is pushing is saying it could not frame. Is this down to the SIP trunk or asterisk? |
| 15:48.03 | BCS-Satori | Morning All. I am trying to enable a transfer beep for an attended transfer letting the user know the transfer has been completed. In features.conf there are two lines that I uncommented "xfersound = beep" and "xferfailsound = beeperr". I reloaded asterisk as well as tried restarting the asterisk services but there still is no beep. Is there anything else that I must enable or should I file a bug report? |
| 15:48.15 | *** join/#asterisk niko1990 (~sunnyboy8@p5B31A676.dip0.t-ipconnect.de) |
| 15:49.50 | mirela666 | BCS-Satori: make sure beep.fomat is in en directory (or your lang dir if you change it) |
| 15:51.00 | BCS-Satori | mirela666: I do see /var/lib/asterisk/sounds/en/beeperr.gsm & /var/lib/asterisk/sounds/en/beep.gsm. Should I see asterisk console indicate is it doing the beep during the transfer? |
| 15:51.29 | mirela666 | yes i think so, just like the transfer.gsm |
| 15:52.07 | *** join/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 15:52.58 | [TK]D-Fender | kippi: You ISDN phone is complaining? that would be between your phone and the telco, no? |
| 15:54.00 | ipalmer | I'm trying to pass a comma seperated list as an argument to a macro but when I do it only sees the first item in the list, is it possible to do this? |
| 15:54.08 | mirela666 | BCS-Satori: yes, defenety you need to see beep.alaw , do you see Playing 'pbx-transfer.alaw' (language 'en') ? |
| 15:55.21 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 15:55.51 | mirela666 | ipalmer: are yu accepting all the in vars on macro? |
| 15:57.10 | niko1990 | Hello everyone :) |
| 15:58.56 | kippi | [TK]D-Fender: so it will be the way voiptalk are answering the call? |
| 15:58.59 | niko1990 | I have a question (I'm kinda new to asterisk and VOIP): Is it possible to setup asterisk to "listen" to a call for dtmf, and when it recognice a special dtmf for exampre #541 that it runs a script? |
| 15:59.12 | BCS-Satori | mirela666: I do not see it attempting to play any beep or pbx-transfer. I do not have alaw installed only gsm (does it need alaw?). I only the music on hold starting when they are being transferred and then the other caller being dialed along with the channel being bridged and the original caller leaving. I wonder if it is because they transfer calls using the page macro as they use their intercom |
| 15:59.32 | ipalmer | mirela666: yes I do a NoOp on the var before I pass it in as a macro with all the fields present, then a NoOp once inside in the macro but only the first field is there |
| 15:59.33 | mirela666 | niko1990: sure it is |
| 15:59.53 | [TK]D-Fender | kippi: Frames are between ISDN & your phone. that has nothing to do with who you call from what we can see |
| 16:00.20 | [TK]D-Fender | niko1990: "applicationmap" <- features.conf |
| 16:00.48 | mirela666 | BCS-Satori: hmm strange |
| 16:01.20 | mirela666 | ipalmer: I ment if you call Macro(var1, var2, var3) |
| 16:01.36 | niko1990 | mirela666 & [TK]D-Fender: Thank you both a lot! I will try to find what google could tell me about that =) |
| 16:01.40 | kippi | ok, so I need asterisk to answer the call and then I just want to be able to record the channel? how should I answer this? should it be going into a meet me? |
| 16:01.56 | mirela666 | ipalmer: you need to create macro test(in_var1, in)var2,in_var3)[] |
| 16:02.00 | [TK]D-Fender | [10:48]BCS-SatoriMorning All. I am trying to enable a transfer beep for an attended transfer letting the user know the transfer has been completed. <- which user? |
| 16:02.22 | *** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 16:03.35 | BCS-Satori | [TK]D-Fender: I want the user who is receiving the call from the receptionist to have a beep or an acknowledgment that the receptionist has completed the transfer and they are on the call with the person. |
| 16:03.35 | [TK]D-Fender | niko1990: Who knows what link yyou'll follow from "google" |
| 16:03.49 | [TK]D-Fender | niko1990: look at the sample configs from Asterisk, and the official WIKI & book |
| 16:03.53 | [TK]D-Fender | ~book |
| 16:03.53 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:03.55 | [TK]D-Fender | ~asteriskwiki |
| 16:03.55 | infobot | asteriskwiki is probably http://wiki.asterisk.org |
| 16:06.36 | niko1990 | [TK]D-Fender: Thanks again for your help =) |
| 16:06.47 | [TK]D-Fender | niko1990: You're welcome |
| 16:08.21 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:10.41 | *** join/#asterisk biox_ (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 16:12.57 | ipalmer | mirela666: sorry if being a bit dumb but could you explain what 'macro test(in_var1, in)var2,in_var3)[]' means I have 'macro(test,${var1},${var2}) where var2 is the csv list |
| 16:14.15 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:17.15 | *** join/#asterisk Ibrahim22 (~Ibrahim22@84.241.211.165) |
| 16:19.18 | Ibrahim22 | Hi, quick question, is it possible to use ARI to manipulate channels without having the user dial an extension first? Or do I need to use AMI to originate to my Stasis app and use it that way? I can create channels with the ARI, but manipulations gives me back an error of "Channel not in Stasis app" |
| 16:19.25 | JeffC_NN | ipalmer: Any chance you could change the delimiter? I'm guessing * is replacing ${var2} with its contents, which would look like separate parameters to the Macro. I wonder if macro(test,${var1},"${var2}") would fix anything? |
| 16:19.59 | pabelanger | Ibrahim22, If you don't create the channel is ARI, there is no way to control it. |
| 16:20.29 | pabelanger | Ibrahim22, what are you looking to do? |
| 16:20.52 | [TK]D-Fender | Ibrahim22: ARI is for creating apps that a channel is in. Not for manipulating another channel |
| 16:21.09 | Ibrahim22 | Ow okay |
| 16:21.49 | *** join/#asterisk Bearish (~Bearish@80.250.214.230) |
| 16:22.17 | Ibrahim22 | So it is not possible to create a channel through ari, then manipulate it |
| 16:23.39 | pabelanger | Ibrahim22, you 'manipulate it' using ARI. Again, what are you trying to do |
| 16:24.11 | ipalmer | JeffC_NN: I see what you're saying I'll give it a go |
| 16:25.14 | *** join/#asterisk julgr (~julgr@38.104.125.2) |
| 16:25.41 | Ibrahim22 | Let me try and explain it: I want a web interface where users can login as an agent, they get put into an agent pool using AgentLogin, then with ARI i can manipulate it so they can take calls via the web interface (pushing a button), and more but that is the basis im trying to do |
| 16:26.11 | Ibrahim22 | But as im reading your responses, i think ARI is for creating your own Dialplan Apps |
| 16:26.25 | Ibrahim22 | Am i right? |
| 16:27.07 | Ibrahim22 | I thougjt ARI was just an API to Asterisk itself, but oh well |
| 16:27.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.196) |
| 16:28.16 | pabelanger | Ibrahim22, Right, ARI will not interface with anything in the apps directory. It is meant to replace the apps directory. So, you would create an ARI application, and provide AgentLogin functionality. Then create a bridge to let your incoming callers wait in |
| 16:29.03 | pabelanger | Ibrahim22, Basically, if you want to use ARI with app_queue, it is not going to work. |
| 16:29.23 | Ibrahim22 | Too bad, so it's back to AMI for me |
| 16:29.42 | pabelanger | or wait until ARI queue is built |
| 16:30.10 | Ibrahim22 | Yeah |
| 16:30.15 | JeffC_NN | I've made my own queues with AGI and a db. |
| 16:30.37 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 16:30.37 | pabelanger | FWIW, my team is currently working on using ARI for a queue backend |
| 16:31.05 | pabelanger | hope to have something in 3months |
| 16:31.28 | Ibrahim22 | Is it going to be opensource/public, as in can i follow your developments? |
| 16:32.49 | pabelanger | Yup, apache2 |
| 16:32.53 | Ibrahim22 | I wonder why they put a limit on ARI's capabilities to just your own app |
| 16:33.23 | Ibrahim22 | It would have been so much more powerfull |
| 16:33.40 | pabelanger | Ibrahim22, http://payload.readthedocs.org/ is the documentation so far |
| 16:33.42 | Ibrahim22 | A great competitor to AMI |
| 16:33.47 | [TK]D-Fender | because manipulating other channels is what AMI is for |
| 16:33.58 | [TK]D-Fender | Why is Asterisk to COMPETE with itself? |
| 16:34.14 | pabelanger | We basically have the HTTP backend finished, we are still working on the actually voice integration now |
| 16:34.41 | pabelanger | Ibrahim22, Here is a overview of how things would work: http://wiki.kickstand-project.org/pabelanger?action=AttachFile&do=view&target=payload.png |
| 16:37.00 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:37.51 | Ibrahim22 | Because the AMI sucks, their way of generating actions is prehistoric, REST is so much more cleaner, i mean, using a special text format sent over a socket, and praying things won't fuck up |
| 16:38.25 | Ibrahim22 | pabelanger: i will definitely check it out |
| 16:38.36 | [TK]D-Fender | REST is linear in that way, because ARI is for controlling THAT call.. not a million other potential calls going through your system |
| 16:38.42 | Ibrahim22 | Maybe even contribute if my boss lets me |
| 16:38.43 | [TK]D-Fender | You're comparing apples & oranges |
| 16:38.51 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 16:39.44 | pabelanger | Ibrahim22, we are actually have a ARI development summit in March. We posted the info on the asterisk-app-dev mailing list: http://lists.digium.com/pipermail/asterisk-app-dev/2014-January/000312.html |
| 16:40.11 | pabelanger | I want the development for the queue app to be open, so people can contribute and help drive development |
| 16:40.27 | pabelanger | We have a general idea of how it will work, but always want more input |
| 16:41.17 | Ibrahim22 | I don't see how ari is different, i can originate calls, put them in bridges, destroy them, conference, hold, mute, etc. same thing AMI can, only AMI is not limited to your app. |
| 16:43.47 | Ibrahim22 | But i would love to talk some more about this, but I need to get back to work :) maybe ill talk to you all tomorrow :) |
| 16:43.59 | *** part/#asterisk Ibrahim22 (~Ibrahim22@84.241.211.165) |
| 16:45.23 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 16:45.28 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 16:45.54 | *** join/#asterisk jploh (~textual@49.144.21.129) |
| 16:45.55 | *** join/#asterisk biox (~jesseolso@173-160-122-45-Minnesota.hfc.comcastbusiness.net) |
| 16:49.08 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 16:55.08 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 16:57.24 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
| 17:07.54 | *** join/#asterisk SGjunior (607fde14@gateway/web/cgi-irc/kiwiirc.com/ip.96.127.222.20) |
| 17:10.22 | SGjunior | Trying to execute a script after a call is hanged up : I am using the "h" extension ofc and the actions are executed. Problem is that the ChannelVars are non-existant since it appears to be a new channel that runs through the "h" |
| 17:10.55 | SGjunior | It works when it is coming from a normal call, but not when the call is completed and from a queue |
| 17:11.45 | [TK]D-Fender | SGjunior: https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 17:12.46 | Katty | looks in |
| 17:13.24 | SGjunior | beautiful, thanks! didnt find those variables on voip-info so I didnt go looking for them :S |
| 17:14.36 | [TK]D-Fender | SGjunior: That old WIKI is not closer to being the last place to look |
| 17:16.55 | *** join/#asterisk dlynes_laptop (~dlynes@ip099.216.145.98.nas.net) |
| 17:19.11 | [TK]D-Fender | now* |
| 17:19.24 | *** part/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 17:25.31 | SGjunior | :S I used to go voip-info // sample files // manager command // (I have yet to upgrade to asterisk 11 ... stilll on 1.8) gonna do that before trying anything else |
| 17:27.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.213) |
| 17:28.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.213) |
| 17:31.49 | *** join/#asterisk u0m3 (~u0m3@92.80.105.67) |
| 17:49.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.239) |
| 17:54.06 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.132) |
| 18:00.15 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 18:01.48 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 18:12.48 | *** join/#asterisk yano (~yano@freenode/staff/yano) |
| 18:14.34 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 18:25.47 | *** join/#asterisk Bearish (~Bearish@84.23.61.81) |
| 18:29.15 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 18:29.20 | *** join/#asterisk Psil0Cybin (~psil0cybi@unaffiliated/psil0cybin) |
| 18:38.56 | *** join/#asterisk Pullphinger (~Pullphing@173-167-162-246-illinois.hfc.comcastbusiness.net) |
| 18:45.25 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:51.37 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.213) |
| 18:53.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.184) |
| 18:55.35 | ruben231 | hi guys i have a hosted asterisk server on a US datacenter- but im experiencing one way audio on a randomly manner on callers..any idea, no NAT involve on the server but on the SIP user they are from home connecting to the hosted asterisk server..any idea..? |
| 18:57.59 | mjordan | ruben231: one way audio is usually indicative of something directing the audio stream to the wrong location. Although it could also be gremlins eating your RTP. Hard to tell without a SIP trace. |
| 19:03.12 | WIMPy | res_gremlins.so |
| 19:05.32 | [TK]D-Fender | ruben231: Make sure your peers are right |
| 19:05.38 | [TK]D-Fender | ruben231: Look at the calls. |
| 19:09.49 | mjordan | WIMPy: we should write that. It should periodically just hangup active calls. |
| 19:10.17 | WIMPy | I think we already have enough random features without it. |
| 19:10.26 | [TK]D-Fender | mjordan: callprogress=yes <- we already have that in chan_dahdi ;) |
| 19:10.48 | WIMPy | [TK]D-Fender: Huw so? |
| 19:10.51 | WIMPy | How |
| 19:11.05 | [TK]D-Fender | WIMPy: analog channels.... |
| 19:11.20 | WIMPy | Oh, right. |
| 19:11.24 | [TK]D-Fender | Asterisk: Where failure is NOT an option (it comes bundled with the software) |
| 19:11.43 | WIMPy | Although session timers seem to do it quite effectively as well. |
| 19:11.56 | [TK]D-Fender | WIMPy: SUCKcess! |
| 19:14.43 | mjordan | hey, session timers at least is negotiated |
| 19:15.10 | WIMPy | Yes, but that doesn't mean they work reproducibly. |
| 19:15.41 | WIMPy | I have one peer where calls drop randomly when they expire. Sometimes it works, sometimes it doesn't. |
| 19:16.09 | mjordan | WIMPy: What version are you on? |
| 19:16.15 | WIMPy | I.e. sometimes calls are cut after 5 minutes, sometimes after 10 and sometimes they even work for 15 minutes. |
| 19:16.20 | WIMPy | 11.7 |
| 19:16.27 | mjordan | hm. Oh well. |
| 19:16.46 | mjordan | there was a lot of work done mid-stream 1.8 to fix a lot of odd behaviour in session timers. Generally, we negotiate it per the RFC correctly now |
| 19:17.41 | WIMPy | I have no idea what causes that random behaviour. That peer is a FritzBox with recent firmware. |
| 19:18.03 | WIMPy | Not the only FritzBox connecting to me. The others don't have that issue. |
| 19:18.27 | WIMPy | I have no idea where the difference is. |
| 19:19.36 | mjordan | tracing session timer weirdness is non-trivial as well, as who is the refresher is a rather confusing concept, and depends greatly as well on who initiated the dialog |
| 19:19.43 | mjordan | so you may be seeing it only inbound/outbound, etc. |
| 19:19.56 | WIMPy | Wait. Wrong. The box he's connecting to is still on 11.5.1. |
| 19:20.39 | WIMPy | I'm not really sure, but I thing it does indeed only happen inbound. |
| 19:20.44 | WIMPy | k |
| 19:29.25 | Katty | DOOR KNOBS |
| 19:30.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.184) |
| 19:30.23 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 19:31.16 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.133) |
| 19:43.44 | *** join/#asterisk pwerunits (b6b9f6a1@gateway/web/freenode/ip.182.185.246.161) |
| 19:44.25 | pwerunits | hey every one, |
| 19:44.33 | pwerunits | i got a issue on Asterisk 11.6.0-rc1 |
| 19:44.44 | pwerunits | on setting up nat=yes |
| 19:44.51 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 19:44.58 | pwerunits | i m getting warning msg on asterisk cli |
| 19:45.05 | pwerunits | sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead |
| 19:45.16 | pwerunits | how to fix it? please any help on this |
| 19:45.36 | WIMPy | Maybe read waht you just pasted. |
| 19:45.53 | pwerunits | lol ofcourse i did |
| 19:45.58 | pwerunits | i tried to set |
| 19:46.09 | pwerunits | nat=force_rport |
| 19:46.23 | pwerunits | but asteirskrealtime table is not |
| 19:46.30 | pwerunits | accepting this value |
| 19:46.33 | WIMPy | That's only half of what I just read. |
| 19:47.03 | pwerunits | you mean should i set nat=comedia |
| 19:47.03 | pwerunits | ? |
| 19:47.21 | WIMPy | No, both as it told you. |
| 19:47.41 | pwerunits | i have already tried that |
| 19:47.43 | WIMPy | And define "not accepting this value". |
| 19:47.49 | mjordan | pwerunits: if your realtime table is treating the column as an enum with only certain values, you'll need to update the schema |
| 19:48.16 | pwerunits | but asteriskrealtime is not accepting the values |
| 19:48.40 | WIMPy | Who? Where? When? |
| 19:49.42 | mjordan | How! |
| 19:50.03 | Katty | in the passive tone. |
| 19:50.04 | pwerunits | nat column is varchat |
| 19:50.05 | pwerunits | varchar |
| 19:50.16 | Katty | if you can put By Zombies next to it... |
| 19:50.31 | Katty | then it's in a passive tone. |
| 19:50.33 | Katty | let's try! |
| 19:50.46 | pwerunits | @mjordan: in realtime nat column is varchar |
| 19:51.09 | Katty | in realtime, nat column is varcharing By Zombies |
| 19:51.29 | mjordan | pwerunits: and how are you determining that? Are you looking at your database schema? If so, which database? What SQL command is returning the schema for the table? |
| 19:51.51 | pwerunits | i m using phpmyadmin |
| 19:52.01 | pwerunits | and it is showing column type |
| 19:53.06 | mjordan | k, phpmyadmin is not a database |
| 19:53.40 | pwerunits | phpmyadmin is connected with mysql database |
| 19:53.44 | mjordan | pwerunits: but a quick Google tells me that it administers MySQL |
| 19:54.15 | mjordan | yay. So, what is reporting the error message that 'comedia,force_rport' is not an acceptable value? |
| 19:54.22 | mjordan | is phpmyadmin reporting that? |
| 19:54.28 | mjordan | are you getting an error message on the Asterisk CLI? |
| 19:54.29 | pwerunits | let me tell you |
| 19:55.04 | *** join/#asterisk perezalvarezhi (~perezalva@187.175.255.65) |
| 19:55.55 | skrusty | evening |
| 19:56.02 | [TK]D-Fender | we haven't tried stopping you so far... |
| 19:56.26 | WIMPy | Nope. Still waiting. |
| 19:57.23 | perezalvarezhi | Hi everyone |
| 19:57.44 | pwerunits | https://issues.asterisk.org/jira/browse/ASTERISK-21225?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel |
| 19:57.51 | pwerunits | i have found a bug report |
| 19:57.58 | pwerunits | there is patch for this bug |
| 19:58.09 | pwerunits | how can i do the patch on asterisk? |
| 19:58.24 | WIMPy | Is that the answer to our question? |
| 19:58.36 | mjordan | that isn't your bug. |
| 19:58.51 | mjordan | and, as the issue shows, it is already fixed in 11.5.0+ |
| 19:59.49 | perezalvarezhi | Hi all, I will implement a PBX for my company, we have 12 locations some of them in the USA and some in Mexico. My question is if you recommend having a small server in each location(connecting via IAX) or having a single server for all the locations(Im also not sure if this is possible but it could save me a lot of $$ :D). |
| 19:59.59 | *** join/#asterisk mirela666 (~mirko.bra@95.180.108.153) |
| 20:00.28 | [TK]D-Fender | perezalvarezhi: You could have a single server if you wanted, but there are reasons to NOT do this as well |
| 20:00.50 | WIMPy | perezalvarezhi: Your choice. But the question how these locations will receive call might be important. |
| 20:01.15 | pwerunits | @mjordan: phpmyadmin give this error msg "Warning: #1265 Data truncated for column 'nat' at row 1 |
| 20:01.36 | perezalvarezhi | WIMPy: well maybe a combination of both, like having a good server for the USA and one for MX |
| 20:02.01 | mjordan | pwerunits: That bug had to do with the enumeration value being applied internally to Asterisk. There was no error message, and it had nothing to do with Asterisk RealTime - it just flag out didn't work. It would, in fact, have told you that it applied the setting, but it would have done nothing (and would have told you it did nothing if you asked it) |
| 20:02.13 | mjordan | pwerunits: Socratic method - what do you think that warning message means? |
| 20:02.29 | mjordan | pwerunits: it's telling you that the data you just put into the column 'nat' was truncated. Why might MySQL truncate the data? |
| 20:02.29 | perezalvarezhi | [TK]D-Fender: What kind of problems arise from having a single server ? |
| 20:02.50 | [TK]D-Fender | perezalvarezhi: That server has issues ALL your sites are dead |
| 20:02.53 | WIMPy | perezalvarezhi: No phones if your IP connection is down. |
| 20:03.08 | [TK]D-Fender | perezalvarezhi: One site has connectivity issues and they ahve no failover or ability to call internally. |
| 20:03.14 | WIMPy | But if you get your calls from an ITSP, that doesn;t really matter. |
| 20:04.18 | [TK]D-Fender | WIMPy: well that's another matter |
| 20:04.40 | perezalvarezhi | WIMPy: yeah I was thinking on a SIP Trunk for my calls |
| 20:05.38 | WIMPy | That makes a central server less of an (additional) issue then. |
| 20:06.05 | perezalvarezhi | WIMPy: cause if internet is down it will be down anyways right? |
| 20:06.24 | WIMPy | Yes. |
| 20:06.46 | WIMPy | Unless you need internal calls, as [TK]D-Fender mentioned. |
| 20:07.10 | perezalvarezhi | WIMPy: [TK]D-Fender god I have to think about this very carefully haha |
| 20:08.31 | perezalvarezhi | WIMPy: I wanted to completly take out my phone provider :S but I guess it is a good idea to conserve the traditional phone line with a voice gateway |
| 20:09.15 | WIMPy | I'd definitely use PCI interfaces instead of gateways. |
| 20:09.35 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:09.35 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:10.42 | perezalvarezhi | WIMPy:have you had bad experiences with those? well if you had a central server I would definetly go for PCI but my sites are no bigger than 4 lines in each place |
| 20:10.51 | *** join/#asterisk powerunits (272aa6ac@gateway/web/freenode/ip.39.42.166.172) |
| 20:11.19 | WIMPy | Only BRI/PRI |
| 20:12.48 | powerunits | hello . sorry my chat got disconnected |
| 20:13.46 | perezalvarezhi | WIMPy: well i will have to really think about this :D thank you very much to [TK]D-Fender too |
| 20:13.58 | perezalvarezhi | I wasnt sure if I could have a central server |
| 20:14.09 | perezalvarezhi | but now i have to think about my options :D thx |
| 20:14.22 | powerunits | please could some one tell me how can i install *.diff patch to asterisk? |
| 20:14.55 | WIMPy | Oh, and keep in mind that fax over IP can be a very interesting topic. |
| 20:15.07 | WIMPy | powerunits: man patch |
| 20:15.44 | WIMPy | powerunits: But did you read what mjordan told you? |
| 20:16.12 | powerunits | WIMPy: my chat got disconnected |
| 20:16.16 | powerunits | i didn't read it |
| 20:16.37 | tzafrir | powerunits, you apply them using patch |
| 20:16.54 | tzafrir | cat file.diff | patch -p1 --dry-run |
| 20:17.12 | tzafrir | (if that gives an error: try -p0 instead of -p1) |
| 20:17.23 | tzafrir | if you get no error, remove the --dry-run |
| 20:17.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:19.17 | mjordan | (02:02:01 PM) mjordan: pwerunits: That bug had to do with the enumeration value being applied internally to Asterisk. There was no error message, and it had nothing to do with Asterisk RealTime - it just flag out didn't work. It would, in fact, have told you that it applied the setting, but it would have done nothing (and would have told you it did nothing if you asked it) |
| 20:19.20 | mjordan | (02:02:13 PM) mjordan: pwerunits: Socratic method - what do you think that warning message means? |
| 20:19.23 | mjordan | (02:02:29 PM) mjordan: pwerunits: it's telling you that the data you just put into the column 'nat' was truncated. Why might MySQL truncate the data? |
| 20:20.52 | file | mjordan, because it's mean ;( |
| 20:21.10 | mjordan | that's what happens when you get bought up by a large, 3vil corporation |
| 20:21.21 | mjordan | you get all soulless and data length sensitive |
| 20:21.45 | mjordan | back in the day, you would have just taken my data and loved me for it. But now, no, you don't want that column length to be violated. |
| 20:22.02 | mjordan | I blame Oracle. |
| 20:26.14 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 20:30.36 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.223.130) |
| 20:33.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:33.28 | *** join/#asterisk Tagor (~Tagor@53567E1F.cm-6-7b.dynamic.ziggo.nl) |
| 20:33.40 | Tagor | How can I keep Dial() ringing even when the SIP device is offline? |
| 20:34.10 | [TK]D-Fender | Tagor: Dial something else along with that device that WILL keep on ringing |
| 20:35.50 | WIMPy | Like a local channel that Wait()s. |
| 20:36.08 | pabelanger | or us a queue |
| 20:36.11 | pabelanger | use* |
| 20:36.23 | Tagor | How did you know I'm was going to ask that? :P |
| 20:39.05 | *** join/#asterisk mrhank (~info@c83-250-229-157.bredband.comhem.se) |
| 20:39.08 | *** join/#asterisk Peppery (~textual@203.86.202.160) |
| 20:41.47 | mrhank | Hi! Im trying to register to a centrex pbx but the username contains a @, which causes asterisk to misinterpret the register string. Is it possible to register with a username like "user1@companyname" in asterisk? can i use quotes? Thankful for any help or tip! |
| 20:43.17 | Peppery | morning. anybody around to assist me in some issues i'm having with asterisk? I'm getting no audio on my phones, internal or external calls. my SIP debug: http://pastie.org/private/uuzm2odthe3ax12uehnaa |
| 20:45.24 | mrhank | Peppery: Im not an expert as such, but almost all audio problems are caused by firewall settings. Check your RTP ports and make sure your firewall is configured correctly. |
| 20:45.31 | *** join/#asterisk evilskater (~evilskate@38.99.136.114) |
| 20:46.06 | Peppery | mrhank: that's what I was thinking, but these phones are on my local LAN with no firewall |
| 20:46.27 | Peppery | that debug was an external one though |
| 20:46.56 | pabelanger | mrhank, look for domain support |
| 20:47.00 | pabelanger | in sip.conf |
| 20:47.25 | [TK]D-Fender | Peppery: <--- Transmitting (NAT) to 27.111.14.66:5060 ---> <- providers are almost never NAT'd. Fix your peer |
| 20:51.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.184) |
| 20:52.51 | Peppery | thanks [TK]D-Fender, doesn't seem to make a difference |
| 20:53.22 | [TK]D-Fender | pastebin your trunk config & phone setup masking on secrets |
| 20:53.25 | [TK]D-Fender | only* |
| 20:57.20 | Peppery | [TK]D-Fender: https://dl.dropboxusercontent.com/u/14947/sip.conf.txt |
| 20:59.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.184) |
| 20:59.35 | [TK]D-Fender | peppermake sure you also have "directmedia=no" for all your peers. |
| 21:00.24 | mrhank | @pabelanger: Okay, will i be able to register with the full username "user@domain" or should i just use "user"? I have enabled autodomain=true and domain=mydomain |
| 21:07.09 | *** join/#asterisk felipealmeida (~user@177.19.32.43) |
| 21:12.36 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:16.19 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:23.49 | *** join/#asterisk vittorio88 (~smuxi@net-188-216-87-216.cust.dsl.vodafone.it) |
| 21:32.55 | vittorio88 | Hi everbody! I am having trouble getting asterisk to start listening for sip/tls traffic on 5061. sip show settings says I'm listening for TLS on 0.0.0.0:5061, yet netstat -tuna | grep 5061 turns up blank. I also occasionally get 'SSL error loading cert file' even after having regenerated the certs. This is happening on a centos server and my ubuntu pc, running 11.5.1 and 1.8 respectively, so I'm guessing it may be a PBMK.... |
| 21:34.45 | [TK]D-Fender | Heading home, BBIAB |
| 21:43.40 | *** join/#asterisk Master_dot_csv (~Master_do@gw1.security.web.za) |
| 21:51.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.31) |
| 21:51.41 | *** join/#asterisk biox_ (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 21:54.47 | Chainsaw | vittorio88: Oh I've seen that. |
| 21:55.01 | Chainsaw | vittorio88: I take it your certificate isn't a self-signed one, but an actually valid, paid cert? |
| 22:03.39 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925269193.dsl.bell.ca) |
| 22:04.09 | hfp | Hi, I was wondering, the things you create with Asterisk, how are they called? VoIP apps? Telephony apps? Unified communication apps? |
| 22:05.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:05.26 | *** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 22:07.52 | Chainsaw | hfp: Telephone systems. |
| 22:08.11 | hfp | Chainsaw: Thanks |
| 22:08.12 | Chainsaw | hfp: That's what I call mine, anyway. I consider Asterisk to be a construction kit, that most people build telephone systems with. |
| 22:08.29 | Chainsaw | hfp: "PABX", if you like your office abbrevations. |
| 22:08.33 | hfp | I am trying ot come up with a term to put them under in my portfolio |
| 22:09.26 | Chainsaw | hfp: Bespoke VoIP telephone systems. |
| 22:09.34 | Chainsaw | hfp: That should score highly in CV buzzword bingo. |
| 22:09.54 | hfp | Haha |
| 22:29.48 | *** join/#asterisk Psil0Cybin (~psil0cybi@unaffiliated/psil0cybin) |
| 22:32.56 | vittorio88 | exit |
| 22:33.02 | vittorio88 | oops, sorry! |
| 22:35.12 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 22:36.30 | *** join/#asterisk mjordan (~matt@nat/digium/x-qkvpyjhtivugcwux) |
| 22:36.30 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:07.58 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 23:10.31 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 23:10.54 | *** join/#asterisk tedstriker (~tedstrike@host81-147-93-10.range81-147.btcentralplus.com) |
| 23:11.59 | *** join/#asterisk tedstriker (~tedstrike@host81-147-93-10.range81-147.btcentralplus.com) |
| 23:12.19 | madduck | Trying to connect to an *-server out there from behind a NAT (double NAT actually), we have an iphone, an android and an sflphone instance all able to make calls. But with a Gigaset S685IP, I can exchange SIP messages, but neither the server nor the router locally sees any type of RTP traffic. How would you debug this? |
| 23:12.44 | *** join/#asterisk Defraz (~Defraz@24-116-129-19.cpe.cableone.net) |
| 23:13.31 | madduck | it *has* worked before. I /just/ stopped working. Really strange and really annoying. |
| 23:16.11 | mjordan | madduck: usually when that happens, comedia isn't enabled |
| 23:16.33 | mjordan | or else the SDP in the SIP messages either from Asterisk or from the phone have the wrong IP address |
| 23:16.43 | mjordan | if none of that is the case, then look for a SIP ALG "helping" you somewhere in the middle |
| 23:17.42 | madduck | what is comedia, mjordan? |
| 23:18.11 | mjordan | ; nat = comedia ; Send media to the port Asterisk received it from regardless of where the SDP says to send it. |
| 23:19.11 | madduck | i have nat = yes, which includes comedia, no? |
| 23:19.31 | madduck | more importantly though: why would this only be a problem on one SIP client in the same network; three other sip clients work just fine |
| 23:19.42 | madduck | their sip.conf stanzas are identical except for the passwords |
| 23:19.44 | mjordan | depends. Do you have nat defined in the [general] section? |
| 23:19.46 | madduck | they inherit from the same stanza |
| 23:19.48 | mjordan | k |
| 23:19.53 | mjordan | then that probably isn't the case. |
| 23:20.04 | madduck | mjordan: I do not have nat enabled generally |
| 23:20.27 | mjordan | As for why one would have a problem and another would not: you'd have to look at the addresses in the messages |
| 23:20.37 | mjordan | I'd be guessing otherwise. |
| 23:21.18 | madduck | i am looking at the messages, but this stupid provider forcing NAT upon us is annoying |
| 23:21.34 | madduck | i.e. we are getting *one* RFC1918 address from them, which they NAT |
| 23:21.44 | madduck | and we have to NAT that at our perimeter, of course, too. |
| 23:21.56 | madduck | we'll change provider soon, but meanwhile… |
| 23:26.51 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.209) |
| 23:42.20 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 23:47.23 | *** part/#asterisk mjordan (~matt@nat/digium/x-qkvpyjhtivugcwux) |
| 23:48.10 | madduck | is this expected? |
| 23:48.26 | madduck | ← Transmitting (no NAT) to xxx.yy.zz.aaa:12345 → |
| 23:48.34 | madduck | why (no NAT) when there is NAT involved? |
| 23:49.40 | madduck | ah, it is if I set nat=comedia, but with nat=yes, it says "(NAT)" |
