| 00:02.39 | *** join/#asterisk pjm (~pjm@uhfsatcom.plus.com) |
| 00:10.06 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-vpxbhfgzunsilund) |
| 00:10.37 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-wqxhplpozuywaxvp) |
| 00:10.44 | *** join/#asterisk DanQuinney (sid18169@gateway/web/irccloud.com/x-qlnhdedkxntmdgkh) |
| 00:11.17 | *** join/#asterisk foamz (sid25727@gateway/web/irccloud.com/x-pnawispzhlfyxypg) |
| 00:11.51 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-xhxlgbiwaxxstbqc) |
| 00:13.10 | *** join/#asterisk dan_j (sid21651@gateway/web/irccloud.com/x-kljelnldwcfqbzsm) |
| 00:16.25 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 00:16.38 | *** join/#asterisk DanQuinney (sid18169@gateway/web/irccloud.com/x-yzbgbmgpvyijfopp) |
| 00:16.40 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-dpumowaufxukfccn) |
| 00:33.45 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 00:54.14 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 00:59.29 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:00.29 | *** join/#asterisk unforgiven512 (~unforgive@freebsd-dev.unforgivendevelopment.com) |
| 02:08.15 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 02:23.49 | *** join/#asterisk boris_t (~boris_t@212.119.243.183) |
| 02:31.30 | *** join/#asterisk ks3 (~ks3@cpe-24-209-22-231.cinci.res.rr.com) |
| 03:37.59 | drmessano | uh |
| 03:38.21 | WIMPy | That sounds serious. |
| 04:34.04 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 04:57.07 | *** join/#asterisk tzafrir (~tzafrir@91.183.149.90) |
| 05:41.39 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 05:43.46 | *** join/#asterisk Gugge_ (gugge@92.246.2.105) |
| 05:44.31 | *** join/#asterisk stephanj (stephan@opennic/stephan) |
| 05:48.28 | *** join/#asterisk x5eb (~seb@seb-hpws2.w1.tele.crt1.net) |
| 05:48.34 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@echo.freevision.sk) |
| 05:49.45 | *** join/#asterisk Happzz_ (void@unaffiliated/ducch) |
| 05:50.17 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 05:51.58 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 05:52.38 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 05:56.29 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 05:56.59 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 05:59.50 | *** join/#asterisk picard276 (79f49be2@gateway/web/freenode/ip.121.244.155.226) |
| 06:00.07 | picard276 | hey guys i was wondering how do i set a custom context for sip SIMPLE messages in chan_pjsip |
| 06:00.43 | picard276 | in the normal chan_sip you can do accept_outofcall_message=yes outofcall_message_context=astsms |
| 06:00.56 | *** join/#asterisk AviiNL (~AviiNL@h178213.upc-h.chello.nl) |
| 06:01.19 | picard276 | that send all sip SIMPLE messages to that custom context... i just don't know how to do that for chan_pjsip .. any ideas? |
| 06:02.48 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 06:08.42 | *** join/#asterisk goddva (~goddva@77.40.154.242) |
| 06:09.09 | picard276 | any ideas ? |
| 06:14.01 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 06:17.23 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 06:24.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 06:24.20 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 06:52.18 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 06:53.38 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:54.11 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 07:23.27 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 07:52.22 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 07:59.29 | *** join/#asterisk pawiecki (~pawiecki@217.97.180.1) |
| 08:04.14 | *** join/#asterisk hehol (~Adium@ip-178-203-97-214.hsi10.unitymediagroup.de) |
| 08:11.51 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 08:13.04 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 08:23.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:24.57 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 08:28.06 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:54.06 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 09:41.12 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 09:52.07 | *** join/#asterisk ruied (~ruied@a79-169-114-18.cpe.netcabo.pt) |
| 10:03.13 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 10:11.19 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 10:31.25 | *** join/#asterisk sepikas_antanas (~sepikas_a@91.216.163.18) |
| 10:36.53 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 10:37.10 | sepikas_antanas | Hi, I am looking for someone who could give me basic information about using asterisk, anyone that could help me? |
| 10:41.56 | *** join/#asterisk matt_ (~matt@ccpc-buzzer.bath.ac.uk) |
| 10:42.20 | *** join/#asterisk sepikas_antanas (5bd8a312@gateway/web/cgi-irc/kiwiirc.com/ip.91.216.163.18) |
| 10:43.18 | sepikas_antanas | Hi, anyone that could give me some information about asterisk? |
| 10:43.59 | MaliutaLap | ~book |
| 10:43.59 | 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 |
| 10:44.47 | sepikas_antanas | I was hoping at first somebody could just tell where to look directly |
| 10:45.26 | sepikas_antanas | I have external software that I need in someay to change data between asterisk |
| 10:45.46 | wyoung | sepikas_antanas: yes, I like it when I get people to do things for me for nothing |
| 10:46.17 | wyoung | sepikas_antanas: or, you can do some reading like the rest of us |
| 10:46.20 | sepikas_antanas | well im not expecting anybody to do anything for me, just to point me in the right direction |
| 10:46.27 | wyoung | sepikas_antanas: or transfer me some money |
| 10:46.44 | wyoung | sepikas_antanas: read and ask questions |
| 10:46.49 | wyoung | if you dont understand something |
| 10:47.14 | wyoung | asterisk.org, google.com, //www.asteriskdocs.org/, etc.. |
| 10:47.45 | wyoung | sorry for being blunt / to the point but you are not the first person to ask that |
| 10:48.01 | wyoung | I asked that once and I receved teh same treatment so you will be no exception :) |
| 10:48.58 | wyoung | sup? |
| 10:53.14 | *** join/#asterisk Bordr (~Bordr@23-24-156-49-static.hfc.comcastbusiness.net) |
| 11:08.11 | *** join/#asterisk C1ph3r5 (~C1ph3r@179.177.130.198.dynamic.adsl.gvt.net.br) |
| 11:31.48 | *** join/#asterisk keithf (~keithf@ool-2f151dc2.static.optonline.net) |
| 11:33.12 | *** join/#asterisk ruied (~ruied@94.62.44.125) |
| 11:34.17 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 11:35.17 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 11:35.17 | *** mode/#asterisk [+o file] by ChanServ |
| 11:56.39 | *** join/#asterisk gerhard7 (~gerhard7@77.172.82.111) |
| 12:03.16 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:03.50 | marceloamorim | guys, how could I send a variable when I use app Dial? |
| 12:04.03 | Kunsi | "send"? |
| 12:04.47 | *** join/#asterisk goddva (~goddva@77.40.154.242) |
| 12:04.48 | marceloamorim | maybe send isn't the correct word, I'm dialing asterisk to another asterisk but I didn't get the callerid from original channel |
| 12:08.14 | marceloamorim | so I will send the callerid as "${ARG1} if I could =) |
| 12:25.52 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 12:31.34 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 12:39.50 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 12:55.39 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:59.29 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:05.37 | *** join/#asterisk mub (~jub@107.1.220.2) |
| 13:06.54 | mub | What is the highest level of verbosity in the Asterisk CLI? Every time I end up doing asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 13:07.27 | mub | I grow tired of holding 'v' down |
| 13:19.25 | wdoekes | depends on the asterisk version. they tried to limit it to 5 |
| 13:19.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:19.57 | mub | Right now mine is set to 1410065408 |
| 13:20.01 | mub | Big difference from 1410065407 |
| 13:22.40 | wdoekes | in ast13 there are 27 ast_verb messages with 5 or higher, total 1324 |
| 13:23.50 | wdoekes | debug goes up to level 5 and only 16 messages with higher level |
| 13:32.59 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 13:41.34 | *** join/#asterisk tompaw (U2FsdGVkX1@tompaw.xxx) |
| 13:47.52 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 13:53.05 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 14:26.28 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 14:28.37 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 14:31.04 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 14:47.59 | *** join/#asterisk mjordan (mjordan@nat/digium/x-geajvtowzndkwsek) |
| 14:47.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:49.05 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:51.56 | *** join/#asterisk C1ph3r5 (~C1ph3r@177.96.244.213) |
| 15:04.07 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 15:11.24 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 15:26.18 | *** join/#asterisk kharwell (kharwell@nat/digium/x-adhfwctqzjicnesz) |
| 15:29.17 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 15:38.32 | *** join/#asterisk acidoo (~textual@modemcable002.114-70-69.static.videotron.ca) |
| 15:41.59 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 15:41.59 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:47.41 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-mwppbphqadfcsepn) |
| 15:49.52 | *** join/#asterisk aness (~aness@cm-84.215.107.62.getinternet.no) |
| 15:50.36 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-tnlgxguqnvfnrhbf) |
| 15:52.53 | *** join/#asterisk togtog (tog@tog.net) |
| 15:55.15 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 15:56.53 | *** join/#asterisk resist0r (~resist0r@unaffiliated/mrresist0r) |
| 16:27.10 | c0ldg0ld | seem to be having an issue with latest versions of dahdi compiling completely. I'll go through the same steps I have for years but at the end of it all "service dahdi start" yields a dahdi: unrecognized service |
| 16:28.22 | [TK]D-Fender | That's just an init script issue at worst |
| 16:28.33 | c0ldg0ld | anyone else seen this? I've tried running make config again and I've found an older tarball, compiled that, then compiled the new one and I'm good to go on the new version but while I've been experimenting with kickstart installs I've had a lot of experience lately with fresh installs |
| 16:28.48 | [TK]D-Fender | check to make sure you've executed the command that actually copies it into place |
| 16:28.51 | c0ldg0ld | that's what I thought as I cna't see any errors in the actual compile |
| 16:28.55 | [TK]D-Fender | and do it manually if you have to |
| 16:29.05 | [TK]D-Fender | this has nothing to do with compiling it |
| 16:29.09 | c0ldg0ld | thx |
| 16:29.16 | [TK]D-Fender | its a raw copy |
| 16:30.28 | [TK]D-Fender | sh"make config" should do it IIRC |
| 16:30.49 | [TK]D-Fender | Back up any configs you have just in case |
| 16:30.58 | c0ldg0ld | fresh system so no worries there |
| 16:31.04 | c0ldg0ld | will do some testing. thanks |
| 16:40.06 | c0ldg0ld | make -C tools config |
| 16:40.08 | c0ldg0ld | make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.11.0+2.11.0/tools' |
| 16:40.10 | c0ldg0ld | make[1]: Nothing to be done for `config'. |
| 16:40.20 | c0ldg0ld | that looks like it's just for the "tools" portion of the complete install |
| 16:40.39 | [TK]D-Fender | I said just "make config" |
| 16:40.42 | [TK]D-Fender | nothing extra |
| 16:41.03 | c0ldg0ld | the command I typed was "make config" nothing extra. This is output from the command |
| 16:41.36 | c0ldg0ld | that's all I've ever done to setup init scripts for both dahdi and Asterisk was a make config after compiling |
| 16:42.29 | [TK]D-Fender | c0ldg0ldmake -C tools config <- this looked like you did otherwisse |
| 16:42.43 | [TK]D-Fender | Check the docs again for what's required at that step |
| 16:42.44 | c0ldg0ld | yes I know but that was part of the script. not something I typed in |
| 16:42.55 | c0ldg0ld | I have. it says the same thing you said |
| 16:43.43 | c0ldg0ld | http://pastebin.com/emqMBdFT |
| 16:43.57 | c0ldg0ld | you can see the only input I did at the top and then what I pasted in here at the bottom |
| 16:45.53 | c0ldg0ld | that part may not actually mean anything at all... it might just mean that there's no "config" script for the dahdi tools portion of the "complete" package which would be understandable. I wouldn't think any of those tools would need to start at boot |
| 16:47.58 | [TK]D-Fender | make[1]: Nothing to be done for `config'. |
| 16:48.00 | [TK]D-Fender | hrm |
| 16:48.21 | [TK]D-Fender | file: / mjordan : any idea on this for him? |
| 16:48.32 | c0ldg0ld | If you are installing a Digium Digital Card, download and install the latest version of libpri after installing DAHDI. libpri is available for download from: http://downloads.digium.com/pub/telephony/libpri |
| 16:48.54 | c0ldg0ld | from the kb, I'd always been told to install libpri first, would this make a diff? I wouldn't think so... |
| 16:48.57 | file | I... know nothing of that world... |
| 16:49.21 | rmudgett | Install DAHDI then libpri then asterisk |
| 16:49.28 | c0ldg0ld | 10-4 |
| 16:49.47 | WIMPy | The order used not to matter. At some point it started to do. |
| 16:50.03 | c0ldg0ld | this dahdi issue, I've not ever seen and I've been compiling this stuff since 1.4 |
| 16:50.34 | c0ldg0ld | I originally assumed (Since I was trying to use CentOS 7) that is was a "not ready for systemd" type thing |
| 16:50.42 | [TK]D-Fender | c0ldg0ld ... has dahdi issues.... |
| 16:50.57 | c0ldg0ld | I've since gone back to 6.x and issue remains with the most recent dahdi |
| 16:51.04 | [TK]D-Fender | Check the WIIKI for install instructions and commentary |
| 16:53.32 | mub | We should all have a dahdi_test contest |
| 16:53.49 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 16:54.11 | c0ldg0ld | Yep, have read all 14 lines in the WIKI about building and installing dahdi. Had skimmed it at first but just noticed the order thing that I hadn't before |
| 16:54.30 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 16:56.36 | mub | c0ldg0ld: oh, are you trying to 'make config' in the dahdi source folder and it's complaining that there are no init scripts? |
| 16:56.52 | c0ldg0ld | that's what appears to be happening |
| 16:57.03 | c0ldg0ld | yes I am trying to make config from the src dir |
| 16:57.04 | mub | I've gotten a full vicidial cluster running on centos 7 and I also ran into that |
| 16:57.14 | mub | Let me figure out what I did |
| 16:57.58 | c0ldg0ld | I've gotten it working several times on 7 before but it seems like every new install on 7 I run into new errors either with Ast or one of the support programs I used so I decided to stick with 6.x for now. |
| 17:00.13 | c0ldg0ld | getting dahdi working has been so trivial in the past that the last few systems I rebuilt I didn't even realize I had an issue until I had everything configured and I tried to start making calls |
| 17:01.48 | *** join/#asterisk jhester (jhester@nat/digium/x-oigbhrmaxarifqfh) |
| 17:02.52 | mub | c0ldg0ld: see if this works: cp /etc/dahdi/system.conf.sample /etc/dahdi/system.conf && cp /path/to/dahdi-linux-complete/tools/dahdi.init /etc/rc.d/init.d/dahdi |
| 17:03.05 | mub | then systemctl enable dahdi && systemctl start dahdi |
| 17:03.10 | mub | then dahdi_test |
| 17:03.46 | mub | I have no idea what I did and I didn't write documentation and I'm terrible at my job |
| 17:04.42 | mub | I'm assuming the make install succeeded in your dahdi src folder, just not the make config |
| 17:05.57 | c0ldg0ld | ah, yep that at least attempts to start it up. Now it's whining about not finding the libtonezone.so.2 lib so lemme look back at the install and see why that failed to copy |
| 17:06.22 | c0ldg0ld | I already had a system.conf so I just copied the init script |
| 17:06.38 | c0ldg0ld | and obviously service dahdi start instead of systemctl since I'm back on 6 for this one |
| 17:07.50 | c0ldg0ld | actually running ldconfig fixed the library issue |
| 17:09.00 | WIMPy | What installs libraries without running ldconfig? |
| 17:09.15 | c0ldg0ld | yeah that was it... I just didn't know where the stupid init script was in the source to copy it manually. Thanks for the help. Would like to figure out why it didn't do it in the make config script still |
| 17:09.45 | c0ldg0ld | WIMPy, some issue with the dahdi tarball I've been pulling down over the last week or so |
| 17:23.02 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1:248e:fef6:1aa6:7034) |
| 17:24.06 | *** join/#asterisk jasonwert (~quassel@75-134-81-98.static.aldl.mi.charter.com) |
| 17:29.34 | WIMPy | Speaking of dahdi... What are dahdi-tools looking for to find out if dahdi-linux has been installed? Looks like I'm missing something. |
| 17:33.23 | c0ldg0ld | I don't know. Is there a way to download the packages by themselves anymore? I haven't seen anything but a complete package for a long time but it would be a lot easier to troublshoot if they were apart |
| 17:33.55 | WIMPy | Last time they were available seperately. |
| 17:34.27 | WIMPy | Point is that I'm cross comiling dahdi-linux together with linux. I'm trying on a Raspberry Pi... |
| 17:34.44 | c0ldg0ld | ah |
| 17:35.16 | c0ldg0ld | it's been quite a while since I did that. I remember running into a lot of issues and eventually got it workin but this is back around the time of the original pi's |
| 17:35.41 | c0ldg0ld | and come to think of it, I didn't bother with dahdi because I couldn't get an interface card hooked up to rpi! |
| 17:35.51 | WIMPy | So at leas I know it must be possible somehow. |
| 17:36.24 | WIMPy | Well, I give it a try. |
| 17:36.34 | c0ldg0ld | gl |
| 17:36.51 | WIMPy | Would be a goo use for that Astribank. |
| 17:36.55 | WIMPy | good |
| 17:37.01 | c0ldg0ld | That's a good point |
| 17:37.41 | c0ldg0ld | and I have one of those sitting around because I could never seem to get it stable enough (and ran out of time bringing up that system) |
| 17:38.06 | c0ldg0ld | Are they the only ones still doing dahdi over uSB? |
| 17:38.17 | WIMPy | So you have a reason to try as well :-) |
| 17:38.31 | c0ldg0ld | reason, yep. Time... nope |
| 17:38.36 | WIMPy | Sangoma seem to have some USB stuff as well. |
| 17:39.40 | WIMPy | And tehre are off couse those bog standard USB ISDN dongles. But they just work. No time required for them. |
| 17:40.01 | c0ldg0ld | I have 16 systems ranging from centOS 5.4 and asterisk 1.6.x to centos6.5 and asterisk 11 that I wanna get all standardized on one set of hardware and all up to the same code versions. This stuff has been built from 2009 till now so you can imagine I learned a few things along the way that didn't always get pushed out to the old boxes |
| 17:40.37 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 17:40.47 | c0ldg0ld | havent had any exp with those dongles yet |
| 17:42.34 | WIMPy | "unknown type name 'u32'" samells like a bigger one. |
| 17:42.54 | *** join/#asterisk theron (~theron@2620:10d:c091:200::f:ac2c) |
| 18:16.22 | WIMPy | All of a sudden I'm getting "WARNING[6048]: db.c:332 ast_db_put: Couldn't execute statment: SQL logic error or missing database". Does anyone have an idea what's going on there? 'database show' looks normal. |
| 18:25.29 | *** join/#asterisk nerdcore (~mike@nerdcore.net) |
| 18:26.18 | nerdcore | I'm having some reliability issues contacting out IAX2 server from our office network. I'm curious if I should use iptables "--set-tos" value of Minimize-Delay or Maximize-Reliability or if this option is even expected to have some benefit? |
| 18:26.38 | nerdcore | I just set up wondershaper today so am hopeful it has a positive impact on the situation |
| 18:27.17 | WIMPy | Don't know what that is, but tc on outgoing traffic surely helps. |
| 18:27.25 | nerdcore | it seems our staff fail to receive a lot of calls (answer and no one is there) while the caller is sent directly to voicemail; This problem is very sporradic it seems :( |
| 18:27.27 | WIMPy | Unless it's happening somewhere else. |
| 18:27.42 | nerdcore | WIMPy: what's tc? |
| 18:28.01 | WIMPy | Traffic Control. |
| 18:29.51 | nerdcore | I just setup wondershaper today which has already improved my overall ping times, so hoping that alleviates the issues. |
| 18:30.26 | nerdcore | any general advice on troubleshooting this kind of issue, where empty calls are being received and callers are sent directly to voicemail? |
| 18:30.50 | nerdcore | like, the phone rings, but the caller is sent to voicemail instead. seems very odd to me and not sure how to fix this |
| 18:31.06 | WIMPy | Smells like packet loss. |
| 18:31.25 | WIMPy | VoIP requires a perfect network connection. |
| 18:33.49 | nerdcore | stupid UDP hehe ;) |
| 18:33.51 | WIMPy | Does anyone have an idea how to fix that database issue? |
| 18:33.55 | nerdcore | I get it, though. |
| 18:40.08 | [TK]D-Fender | WIMPy: check your astdb file |
| 18:40.28 | [TK]D-Fender | WIMPy: backup any keys you need and if you delete it * will regenerate one. |
| 18:40.36 | [TK]D-Fender | WIMPy: Check perms on it, etc first |
| 18:42.06 | WIMPy | Nothing has changed. It happened out of thin air. |
| 18:42.32 | WIMPy | The database is readable and the disk is not full :-) |
| 18:43.13 | WIMPy | is currently trying to figure how to restore the contents to a fresh db. |
| 18:44.12 | *** join/#asterisk asterite (asterite@aeshna.de) |
| 18:44.48 | *** part/#asterisk asterite (asterite@aeshna.de) |
| 18:46.05 | *** join/#asterisk waj (~waj@ec2-54-164-240-122.compute-1.amazonaws.com) |
| 18:49.44 | waj | Quick question: does Asterisk have support for SRV records in pjsip? |
| 18:58.40 | file | 13 has support for SRV records, master (what will become 14) has superior support which includes NAPTR and failover |
| 18:58.44 | WIMPy | still wonders what happened there... |
| 19:00.06 | waj | file: I'm using 13, but I can see it's not requesting SRV records. Just looking for A and AAAA instead. |
| 19:00.35 | file | what is the URI you are dialing? |
| 19:02.28 | waj | No... I'm just setting up a sip trunk. Here is my config: https://gist.github.com/waj/27c901d99c4a334a5641 |
| 19:03.17 | waj | And when I run `pjsip show identify callcentric` I can see it has the A records, but not the SRV ones |
| 19:04.00 | waj | And that seems to lead to random incoming call rejections when the call comes from some of the servers listed as SRV but not A |
| 19:04.46 | file | the type=identify section does not use SRV at this time |
| 19:05.44 | file | you can potentially use the line functionality though, there's a blog post at https://www.asterisk-blog.com/ about it |
| 19:05.55 | file | https://www.asterisk-blog.com/2016/01/27/the-pjsip-outbound-registration-line-option/ for the direct link |
| 19:07.10 | waj | file: Ok, I'll take a look. Thanks for your time! :) |
| 19:13.06 | *** join/#asterisk mjordan (mjordan@nat/digium/x-rsstpdgwevddskef) |
| 19:13.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:19.38 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 19:32.15 | *** join/#asterisk matt_ (~matt@ccpc-buzzer.bath.ac.uk) |
| 19:40.41 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-deodypxdqghwqlku) |
| 19:40.41 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:41.40 | klow | Hey guys - anyone around? Need some help. Still working on just getting 2 soft phones working on asterisk 13.7 with pjsip |
| 19:42.00 | klow | I have I think both configurations identical - but calling each other both fails, for different reasons |
| 19:42.19 | klow | I'm getting 488 not acceptable here and call immediately fails calling one direction: |
| 19:42.19 | klow | [Feb 1 11:32:54] DEBUG[9956] pjsip: sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=1 (rdata0x7fcd000045b8) |
| 19:42.19 | klow | [Feb 1 11:32:54] VERBOSE[9956] res_pjsip_logger.c: <--- Received SIP request (991 bytes) from TCP:73.53.31.109:51614 ---> |
| 19:42.19 | klow | INVITE sip:6000@dev1.haloprivacy.com SIP/2.0 |
| 19:42.20 | klow | Via: SIP/2.0/TCP 10.128.30.241:50847;branch=z9hG4bK-524287-1---a814fe4b616c2060;rport |
| 19:42.24 | klow | Max-Forwards: 70 |
| 19:42.26 | klow | Contact: <sip:6001@73.53.31.109:51014;transport=tcp;rinstance=63fac2d5fea4e955> |
| 19:42.28 | klow | To: <sip:6000@dev1.haloprivacy.com> |
| 19:42.30 | klow | From: "Kevin"<sip:6001@dev1.haloprivacy.com>;tag=696c2a5f |
| 19:42.32 | klow | Call-ID: 78939MjlmZGUxOGI0MDBlNzA5NjUyYWY3MjdmNDY3OWEwZTU |
| 19:42.34 | klow | CSeq: 1 INVITE |
| 19:42.35 | WIMPy | ~pb |
| 19:42.35 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:42.36 | klow | Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE, OPTIONS |
| 19:42.38 | klow | Content-Type: application/sdp |
| 19:42.40 | klow | Supported: replaces |
| 19:42.42 | klow | User-Agent: Bria 4 release 4.3.1 stamp 78939 |
| 19:42.44 | klow | Content-Length: 384 |
| 19:42.46 | klow | v=0 |
| 19:42.48 | klow | o=- 1454355128190927 1 IN IP4 10.128.30.241 |
| 19:42.50 | klow | s=Bria 4 release 4.3.1 stamp 78939 |
| 19:42.54 | klow | c=IN IP4 10.128.30.241 |
| 19:42.56 | klow | t=0 0 |
| 19:42.58 | klow | m=audio 63988 RTP/AVP 9 8 18 120 0 122 101 |
| 19:42.59 | [TK]D-Fender | file: HAELP |
| 19:43.00 | klow | a=rtpmap:18 G729/8000 |
| 19:43.02 | klow | a=fmtp:18 annexb=yes |
| 19:43.04 | klow | a=rtpmap:120 opus/48000/2 |
| 19:43.06 | klow | a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 |
| 19:43.08 | klow | a=rtpmap:122 SILK/16000 |
| 19:43.10 | klow | a=rtpmap:101 telephone-event/8000 |
| 19:43.12 | klow | a=fmtp:101 0-15 |
| 19:43.14 | klow | a=sendrecv |
| 19:43.16 | klow | [Feb 1 11:32:54] DEBUG[9956] netsock2.c: Splitting '73.53.31.109:51614' into... |
| 19:43.18 | klow | [Feb 1 11:32:54] DEBUG[9956] netsock2.c: ...host '73.53.31.109' and port '51614'. |
| 19:43.20 | *** kick/#asterisk [klow!~file@asterisk/developer-and-muffin-lover/file] by file (klow) |
| 19:43.35 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 19:43.37 | klow | [Feb 1 11:32:54] DEBUG[19366] pjsip: endpoint .Response msg 401/INVITE/cseq=1 (tdta0x7fcce0006db0) created |
| 19:43.40 | klow | [Feb 1 11:32:54] DEBUG[19366] netsock2.c: Splitting '10.50.55.10' into... |
| 19:43.42 | klow | [Feb 1 11:32:54] DEBUG[19366] netsock2.c: ...host '10.50.55.10' and port ''. |
| 19:43.44 | klow | [Feb 1 11:32:54] DEBUG[19366] netsock2.c: Splitting '73.53.31.109' into... |
| 19:43.46 | file | klow, please stop flooding |
| 19:43.46 | klow | [Feb 1 11:32:54] DEBUG[19366] netsock2.c: ...host '73.53.31.109' and port ''. |
| 19:43.48 | klow | [Feb 1 11:32:54] DEBUG[19366] netsock2.c: Splitting '10.50.55.10:5060' into... |
| 19:43.50 | *** kick/#asterisk [klow!~file@asterisk/developer-and-muffin-lover/file] by file (klow) |
| 19:43.54 | [TK]D-Fender | occasionally misses being able to take care of those things himself... |
| 19:44.01 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 19:44.03 | klow | SIP/2.0 401 Unauthorized |
| 19:44.05 | klow | Via: SIP/2.0/TCP 10.128.30.241:50847;rport=51614;received=73.53.31.109;branch=z9hG4bK-524287-1---a814fe4b616c2060 |
| 19:44.10 | klow | Call-ID: 78939MjlmZGUxOGI0MDBlNzA5NjUyYWY3MjdmNDY3OWEwZTU |
| 19:44.12 | klow | From: "Kevin" <sip:6001@dev1.haloprivacy.com>;tag=696c2a5f |
| 19:44.14 | klow | To: <sip:6000@dev1.haloprivacy.com>;tag=z9hG4bK-524287-1---a814fe4b616c2060 |
| 19:44.16 | klow | CSeq: 1 INVITE |
| 19:44.17 | *** mode/#asterisk [+b *!*klong@*.hsd1.wa.comcast.net] by file |
| 19:44.35 | file | klow, I will unban you in a moment |
| 19:45.10 | file | klow, in the mean time place your logging in a pastebin such as http://www.pastebin.net/ |
| 19:45.18 | *** mode/#asterisk [-b *!*klong@*.hsd1.wa.comcast.net] by file |
| 19:45.19 | klow | Max-Forwards: 70 |
| 19:45.23 | klow | Contact: <sip:6001@73.53.31.109:51014;transport=tcp;rinstance=63fac2d5fea4e955> |
| 19:45.25 | klow | To: <sip:6000@dev1.haloprivacy.com> |
| 19:45.26 | *** mode/#asterisk [+b *!*klong@*.hsd1.wa.comcast.net] by file |
| 19:45.34 | file | well this is a bit crazy |
| 19:46.55 | file | do we think it has stopped yet? |
| 19:47.07 | [TK]D-Fender | in the past minute? doube it |
| 19:47.44 | *** mode/#asterisk [-b *!*klong@*.hsd1.wa.comcast.net] by file |
| 19:49.49 | *** join/#asterisk ruied (~ruied@bl7-223-226.dsl.telepac.pt) |
| 19:52.02 | *** join/#asterisk klow (~klong@73.53.31.109) |
| 19:52.08 | klow | doh. sorry. |
| 19:54.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:08.25 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:16.32 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 20:16.51 | klow | bad luck w/ IRC today gah |
| 20:19.11 | klow | OK .. http://pastebin.com/pkuRx8sp |
| 20:19.28 | klow | anyone mind having a look at this call fail? I only have 2 soft phones, pjsip .. |
| 20:20.37 | klow | I put my extensions.con and pjsip.conf here http://pastebin.com/3MVNcBqp |
| 20:42.31 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-dwtosfxbpxpbkkgd) |
| 20:42.31 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:07.05 | *** join/#asterisk Bordr (~Bordr@23-24-156-49-static.hfc.comcastbusiness.net) |
| 21:19.59 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 21:20.36 | klow | Can anyone recommend someone for paid hourly asterisk configuration support , someone who is online and can work with me at a reasonable hourly rate? |
| 21:21.23 | klow | I contacted Digium and was confused about their support offerings, sounds like they offer paid "support" for core modules, but not for configuration help of any kind, which I don't understand exactly |
| 21:21.35 | klow | they referred me to a local integrator in seattle area, who didn't respond to my emails |
| 21:21.45 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:21.45 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 21:22.27 | klow | I am a fairly competent linux sysadmin engineer but I need to speed up some of this asterisk related research .. taking days to get simple configurations working i need some help .. |
| 21:22.31 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-xdgegigvqwznfdit) |
| 21:22.31 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:25.14 | mub | klow: Well, no, but I can look at your call logs. |
| 21:25.29 | klow | ;) |
| 21:25.41 | klow | I'm just trying to get 2 soft phones able to call each other , have been stuck here for about a week. |
| 21:25.50 | klow | different issues when i use TLS vs TCP it seems |
| 21:26.02 | klow | I can't tell if its a nat issue, codec issue, dial plan, pjsip.conf issue .. |
| 21:27.12 | mub | klow: hmm.. lines 152, 171, 181, I've never seen those messages before |
| 21:28.34 | klow | i can't tell if that coming from the endpoint which is being called or from asterisk saying "no" |
| 21:28.58 | mub | klow: Does your carrier conf have 'insecure=port,invite' in there somewhere? |
| 21:29.13 | mub | Let me see your carrier conf (with sensitive data *****'d out) |
| 21:29.21 | klow | i don't have a carrier conf |
| 21:29.26 | mub | Oh it's internal |
| 21:29.34 | *** join/#asterisk Fanch (~fanch@linuxquimper/fanch) |
| 21:29.53 | WIMPy | klow: For debugging SIP traffic, try sngrep. Makes tings easier to read. |
| 21:30.31 | klow | yes, I've only just begun a ground-up SIP configuration pjsip, asterisk 13.7 |
| 21:30.54 | klow | internal calling is my only concern at the moment. just 2 soft phones , Bria for mac and bria for iPhone |
| 21:31.08 | klow | sngrep ok |
| 21:31.21 | mub | klow: And they're both extension 6000? |
| 21:31.31 | mub | Oh my bad, I can't read |
| 21:34.40 | *** join/#asterisk Guest41 (~textual@modemcable002.114-70-69.static.videotron.ca) |
| 21:35.04 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 21:35.18 | klow | did I miss anything ? can't maintain a connection to freenode every |
| 21:35.20 | klow | ever* |
| 21:35.40 | mub | klow: nope |
| 21:35.51 | mub | I'm stumped |
| 21:37.40 | mub | [Feb 1 11:32:54] VERBOSE[19366] res_pjsip_logger.c: <--- Transmitting SIP response (551 bytes) to TCP:73.53.31.109:51614 ---> |
| 21:37.42 | mub | SIP/2.0 401 Unauthorized |
| 21:38.31 | Get_The_Fish | Hello all, I am struggling with Polycom subscriptions and message waiting with PJSIP. I am getting an error message saying that it cant find the AOR of the server. https://dpaste.de/v3oT |
| 21:40.14 | klow | mub : ya so INVITE, unauthorized - another INVITE - 488 not acceptable |
| 21:40.19 | klow | but no obvious reason why |
| 21:40.29 | klow | I see this: [Feb 1 11:32:54] DEBUG[19366] res_pjsip_endpoint_identifier_ip.c: No identify sections to match against |
| 21:40.29 | klow | <PROTECTED> |
| 21:40.41 | klow | but but then this : res_pjsip_endpoint_identifier_user.c: Retrieved endpoint 6001 , so maybe that isn't the problem |
| 21:41.17 | Get_The_Fish | klow: pastebin the whole thing for me please |
| 21:41.42 | klow | http://pastebin.com/pkuRx8sp |
| 21:42.10 | klow | heres my relevant, very very simple conf files http://pastebin.com/3MVNcBqp |
| 21:44.01 | Get_The_Fish | klow: OK, so I am missing where this INVITE is coming from, this 73.53.31.109. Is this an inbound call from an ITSP? |
| 21:44.39 | klow | nope I just have 2 soft phones and I'm sitting here at home |
| 21:44.44 | klow | on on my iPhone and one on my macbook |
| 21:44.50 | klow | and my server is at my data center |
| 21:45.08 | klow | I've only "configured" two extensions, context=internal |
| 21:45.12 | klow | nothing outside yet |
| 21:45.41 | Get_The_Fish | klow: Also, NAT. It makes babies cry. So that is the issue. Disable STUN, TURN, ICE or whatever NAT traversal methods that may be enabled in Bria. |
| 21:46.27 | klow | Indeed NAT sucks. Unfortunately my task here is to set up a PBX for people roaming around god knows where with Bria on their iPhones |
| 21:46.36 | klow | and I have to sort it eventually to deal with NAT as best as humanly possible |
| 21:46.57 | WIMPy | likes NAT |
| 21:47.08 | Get_The_Fish | klow: So since you are on the same LAN, we shouldnt be seeing IP addresses like we see on line 3 |
| 21:48.40 | klow | sorry I should clarify - my 2 phones are here on my LAN at home, my server is collocated . |
| 21:49.03 | klow | making it more confusing - my home router is VPN split tunnel, so I can also connect to the LAN ip of the PBX if needed |
| 21:49.04 | Get_The_Fish | klow: yeah ok that changes things quite a bit! |
| 21:49.16 | klow | and I have been switching back and forth in my testing so even I don't know what I did in that call heh |
| 21:50.11 | Get_The_Fish | klow: well that is going to be important. This is most likely going to come down to a NAT/configuration issue. |
| 21:50.59 | klow | OK I'm going to get back on connection from outside with no VPN 1 sec |
| 21:51.18 | Get_The_Fish | klow: and do me a favor and turn the debugger off |
| 21:52.57 | *** join/#asterisk Fanch (~fanch@linuxquimper/fanch) |
| 21:53.18 | klow | core set debug off? |
| 21:53.25 | klow | core set verbose .. ? |
| 21:53.37 | Get_The_Fish | core set debug off |
| 21:53.49 | Get_The_Fish | core set verbose 4 |
| 21:56.39 | Get_The_Fish | Hello all, I am struggling with Polycom subscriptions and message waiting with PJSIP. I am getting an error message saying that it cant find the AOR of the server. https://dpaste.de/v3oT |
| 21:59.49 | *** join/#asterisk captain118 (~Adium@43.sub-70-193-96.myvzw.com) |
| 22:00.59 | *** join/#asterisk pankid (~quassel@mail.nschoonover.com) |
| 22:04.34 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 22:30.47 | *** part/#asterisk nerdcore (~mike@nerdcore.net) |
| 22:44.55 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:51.02 | *** join/#asterisk djiboutiii (~chrisp@50.125.99.242) |
| 22:51.19 | djiboutiii | Hello - I had a question about RTP packets. |
| 22:52.05 | djiboutiii | I have an asterisk server. When I call it, listen to the menu options (RTP working at this point), and then I dial an extension. The extension is simply forwarding the call to a cell phone. |
| 22:52.15 | djiboutiii | However, there is no audio when the call is done. |
| 22:52.22 | djiboutiii | call is answered* |
| 22:52.34 | djiboutiii | There is no NAT here, it's a public facing machine. |
| 22:52.46 | djiboutiii | Nothing else has changed, this worked for 3 years. |
| 22:52.57 | djiboutiii | Only package updates have taken place |
| 22:53.23 | djiboutiii | With rtp debug on, I can see packets being sent from asterisk to two IP's |
| 22:53.29 | djiboutiii | There are no packets received. |
| 22:53.47 | djiboutiii | Does that seem odd to have rtp packets sending to two ip's? |
| 22:54.04 | newtonr | Check the SDP we sent to the far end to see where we ask them to send audio back |
| 22:54.12 | newtonr | Maybe you are looking at RTP for two different calls |
| 22:54.18 | djiboutiii | Hey, sorry. Lacking in my terminology. What's SDP? |
| 22:55.01 | djiboutiii | And that's a good point about 2 calls, though this is a closed test so it would only be a symptom of my issue |
| 22:56.32 | djiboutiii | newtonr: If you don't mind, could you tell me how to check SDP? |
| 22:57.26 | newtonr | Nope, that sorta goes beyond the discussion here - I'll find an article or guide that explains it though. One minute |
| 22:57.45 | newtonr | https://supportforums.cisco.com/document/113271/understanding-sip-traces |
| 22:58.35 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:58.47 | djiboutiii | That's appropriate. ty |
| 22:59.59 | newtonr | djiboutiii, Mostly I don't have the time. If I did I'd be glad to dive into SIP and SDP. :) Pay particular attention to reading about the "c=" parameter within the SDP and how that relates to where the audio streams end up getting sent |
| 23:01.35 | djiboutiii | newtonr: Oh i wouldn't expect you to dedicate much time. This is perfect, I can hopefully narrow down my issue or at least ask a more targeted question. |
| 23:01.44 | newtonr | djiboutiii, in addition this article goes into RTP stream analysis with wireshark as well http://www.linuxjournal.com/article/9398 |
| 23:01.59 | djiboutiii | Nice - that's what I've been staring at |
| 23:02.17 | newtonr | Yeah if you go that route you'll make a lot of progress |
| 23:03.42 | newtonr | In the meantime if you pastebin logs of the call (https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information) then someone may be able to look through that and help |
| 23:04.51 | djiboutiii | Sounds good. I'll take a crack at it first, I have some good logs (using that link) already so this first link you sent is really informative. |
| 23:04.56 | djiboutiii | And ty again. |
| 23:09.31 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 23:16.50 | *** join/#asterisk miltux (~miltux@d8D876C5A.access.telenet.be) |
| 23:17.07 | djiboutiii | newtonr: Still researching/reading but I did want to answer your original question. Seems like this might be a smoking gun: |
| 23:17.10 | djiboutiii | <PROTECTED> |
| 23:17.28 | djiboutiii | Processing session-level SDP o=Sonus_UAC 11028 31584 IN IP4 152.184.3.143... OK. |
| 23:17.28 | djiboutiii | [Feb 1 14:29:45] DEBUG[5050][C-00000002] chan_sip.c: Processing session-level SDP s=SIP Media Capabilities... UNSUPPORTED OR FAILED. |
| 23:29.51 | djiboutiii | Well, I'm stuck again aside from those SDP Unsupported/Failed logs. |
| 23:30.13 | djiboutiii | Here's a full debug log from a test call I made a moment ago. http://pastebin.com/G5cAe1pG |
| 23:32.04 | newtonr | Pretty normal to see some UNSUPPORTED or FAILED lines in there in the DEBUG |
| 23:32.21 | newtonr | Unsure about in this particular case |
| 23:32.21 | djiboutiii | ah |
| 23:32.33 | newtonr | You'll see a lot of stuff in the "DEBUG" lines that looks scary |
| 23:33.36 | djiboutiii | True, a lot of that in there. |
| 23:35.10 | newtonr | Looks like Asterisk is sending RTP to 199.199.12.54 and receiving RTP from that address |
| 23:36.17 | djiboutiii | Interesting... and odd. |
| 23:38.20 | djiboutiii | Seems like it communicates with that IP until the forwarded call is dialed and answered. Then talks with that IP again after the call ends |
| 23:57.40 | *** join/#asterisk italorossi (~Adium@179.156.47.253) |
| 23:58.43 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |