| 00:02.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:05.41 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 00:06.05 | *** join/#asterisk jsjc (~Adium@100.Red-79-147-248.dynamicIP.rima-tde.net) |
| 00:17.04 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:17.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:30.56 | *** join/#asterisk Platinum (~Platinum@72.42.68.245) |
| 00:31.07 | *** join/#asterisk shawny_ (~shawny@CPE0023eb226093-CM00159a0978a0.cpe.net.cable.rogers.com) |
| 00:31.26 | Platinum | does anyone have experience with asterisk as voicemail for definity g3si |
| 00:34.27 | *** join/#asterisk italorossi (~textual@187.61.182.39) |
| 00:35.16 | jpsharp | I've not done it, but supposedly it can be done by using the SMDI functionality of Asterisk. |
| 00:38.01 | shawny_ | everytime I load asterisk I have no ability to use SIP commands. when I hit ? I get a bunch of stuff, but nothing SIP related. I changed modules.conf to require chan_sip.so to load, and asterisk does not load (because it cannot load SIP) I thought the asterisk would just come with sip support already in there.....am I doing something wrong in the install? |
| 00:39.20 | mjordan | shawny_: you most likely don't have all of the dependencies for chan_sip loaded. When you run the menuselect tool (make menuselect), can you enable chan_sip? If you cannot, what dependencies does it say you need? |
| 00:40.13 | jpsharp | Probably built it on a debian or ubuntu box and didn't have libssssl install. |
| 00:40.17 | jpsharp | er, libssl. |
| 00:41.46 | shawny_ | it tells me it depends on chan_local(M), res_crypto(M) are those from libssl? because I distinctly remember not installing that.... |
| 00:42.09 | Platinum | jpsharp have you heard if a few analog linst between the definity pbx and the asterisk... I have seen people talking about isdn trunks between the devices but no regular analog lines. |
| 00:42.31 | shawny_ | and yes, I'm running ubuntu. |
| 00:42.59 | jpsharp | install libssl, libssl-dev, rerun ./configure. That should solve your issue. |
| 00:43.39 | jpsharp | Platinum: I don't know the details. I've just heard them grumble about SMDI. I don't think they were using full ISDN trunks, though. |
| 00:45.46 | Platinum | Ok, thanks. I will have to look up SMDI info. |
| 00:56.13 | *** part/#asterisk deo_ (~dnepangue@112.198.90.243) |
| 01:06.00 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 01:18.41 | shawny_ | jpsharp: THANK YOU! It worked like a charm! funny how things work when you do them right!!! |
| 01:36.47 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:36.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:52.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:59.13 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
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| 15:20.39 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:20.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.8.0 (2012/09/13), 1.8.16.0 (2012/09/13), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:20.45 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:21.35 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 15:23.38 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 15:27.52 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-249.west.biz.rr.com) |
| 15:27.59 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.94.107.dsl.dyn.forthnet.gr) |
| 15:28.29 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:30.54 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:39.03 | ke-esc | It turned out to be stupid- the config files were root:root and not asterisk:asterisk. I fixed that and reloaded it- now it appears to be fine |
| 15:39.13 | *** part/#asterisk scgm11_ (~Sebastian@r186-50-84-36.dialup.adsl.anteldata.net.uy) |
| 15:44.27 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:47.55 | tompaw | Pizza & Coke = Heaven |
| 15:48.32 | pabelanger | Hookers & Blow = Fun |
| 15:48.59 | tompaw | Pizza & Coke & Topless Beach Bar & LTE = Costa del Sol :D |
| 15:52.06 | _Corey_ | writes down Costa Del Sol... :) |
| 15:53.40 | tompaw | (: |
| 15:58.47 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:14.40 | Katty | hai. |
| 16:15.04 | Qwell | glomps Katty |
| 16:15.11 | Katty | hello mister Qwell |
| 16:15.13 | Katty | hugs Qwell |
| 16:15.34 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:16.17 | Katty | Qwell: i was thinking of getting my PhD in theoretical quantum physics |
| 16:16.39 | Qwell | k |
| 16:16.39 | Katty | Qwell: but then i couldn't figure out what kind of job i'd really have. |
| 16:16.57 | Katty | Qwell: other than teach, which would be horrid. |
| 16:17.38 | Qwell | you could like fire lasers at people |
| 16:17.51 | Katty | so you mean be a lab rat. |
| 16:18.07 | Qwell | sure |
| 16:18.12 | Qwell | or a "researcher" |
| 16:18.13 | Katty | meh. |
| 16:18.17 | Katty | exactly, a lab rat. |
| 16:19.33 | Katty | i can see the panel reviewing my dissertation now... |
| 16:20.57 | Katty | 5 minutes in, their eyes glaze over. |
| 16:21.01 | Katty | 10 minutes later, asleep on desk. |
| 16:21.09 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:21.13 | Katty | hi tony |
| 16:22.11 | [TK]D-Fender | Qwell, No, Leonard is the one who shoots lasers at things. Sheldon is the theoretical quantum physicist ;) </bbt> |
| 16:22.32 | Qwell | I bet drmessano does both. He's a doctor. |
| 16:22.57 | drmessano | lol |
| 16:23.13 | *** join/#asterisk cjk (~cjk@vodsl-9215.vo.lu) |
| 16:23.23 | *** join/#asterisk the_5th_wheel (~edd@41-132-4-180.dsl.mweb.co.za) |
| 16:23.25 | drmessano | I'm not a doctor, but I will be glad to have a look |
| 16:24.41 | the_5th_wheel | Howdy all. I have a bit of a problem, which I think is just me misunderstanding something. My one provider requires me to send all traffic to them via a sip proxy, but the TO header must be a different device. I thought I would use outboundproxy for this, but it seems I am mistaken. Can anyone shed some light? |
| 16:24.59 | Katty | maybe i should go wtih a PhD in chemistry. |
| 16:25.17 | [TK]D-Fender | Qwell, http://www.youtube.com/watch?v=FO0kRE5OTZI |
| 16:25.32 | cjk | hello, I am loosing one ore more digit when sending dtmfs from one asterisk to another via SIP. I sniffed the traffic and the dtmf is sent and received on the network level. I have the same problem with voxbone.com where i receive incoming calls from them. So i am sure its a configuraiton issue on my side. Is anyone willing to help me solve the issue? As I need fast help, I pay 80$ per hour |
| 16:25.51 | Qwell | cjk: Using rfc2833? |
| 16:26.09 | Qwell | cjk: what version of Asterisk? |
| 16:26.15 | [TK]D-Fender | the_5th_wheel, that sounds correct. |
| 16:26.24 | cjk | Qwell, rfc2833 asterisk 1.8 |
| 16:26.31 | Qwell | cjk: specifically, what version? |
| 16:27.01 | cjk | Qwell, I have it on all versions, but here it is 1.8.9.0 |
| 16:27.14 | cjk | in sip.conf I have dtmfmode=rfc2833 |
| 16:27.44 | the_5th_wheel | @[TK]D-Fender: The host does not accept sip on to the host my traffic is destined for directly from me. Does that still sound right? |
| 16:28.05 | Qwell | cjk: and you've actually tried the latest version? |
| 16:28.07 | the_5th_wheel | @[TK]D-Fender: I guess basically I need to be able to change the To header |
| 16:28.13 | cjk | Qwell, yes, really |
| 16:28.16 | [TK]D-Fender | the_5th_wheel, I'm not seeing packets & configs..... |
| 16:28.20 | cjk | Qwell, I had it in 1.4 as well |
| 16:28.53 | the_5th_wheel | @[TK]D-Fender: One moment |
| 16:29.43 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:30.02 | *** join/#asterisk bchia (~Adium@nat/digium/x-ouctanyqchbvbgag) |
| 16:31.00 | the_5th_wheel | @[TK]D-Fender: http://pastebin.com/nWdGua5U <- Sip config |
| 16:31.33 | Katty | Qwell: i should just learn to speak 5 languages. |
| 16:31.38 | Katty | Qwell: and have a hospital hire me as a translator |
| 16:31.46 | Qwell | Katty: meh |
| 16:31.55 | *** join/#asterisk adeel|work (~adeel@216.183.80.220) |
| 16:32.03 | Katty | Qwell: i could go into forensic science |
| 16:32.05 | the_5th_wheel | @[TK]D-Fender: I believe all sip traffic requests to this host needs to go via the outboundproxy=. But it tries to send it directly to host= |
| 16:32.14 | Qwell | Katty: You could be Dexter! |
| 16:32.38 | Katty | Qwell: not so much. not sure i want to tinker with dead bodies either. |
| 16:32.49 | *** join/#asterisk grEvenX (~even@46.249.249.64) |
| 16:33.25 | Katty | Qwell: criminology would be fun tho, me thinks |
| 16:34.36 | the_5th_wheel | <PROTECTED> |
| 16:34.59 | Katty | Qwell: except the dead bodies and all. |
| 16:35.05 | Katty | Qwell: and possible pissing off of serial killers. |
| 16:36.58 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 16:50.31 | adeel|work | are there any high level devel documentation on the CALLERPRES() function? |
| 16:50.47 | leifmadsen | core show function CALLERPRES ? |
| 16:50.51 | Qwell | devel docs, no, probably not |
| 16:50.57 | adeel|work | that just provides a basic summary output |
| 16:51.02 | leifmadsen | oh I missed the 'devel' part |
| 16:52.35 | adeel|work | i'm getting an RPID of Remote-Party-ID: <sip:647523XXXX@193.46.30.28>;party=calling;privacy=off;screen=yes;screen-ind=3;npi=1;ton=2 from my provider, yet asterisk is spitting out: Remote-Party-ID: <sip:647523XXXX@193.46.30.28>;party=calling;privacy=off;screen=yes;screen=no....which seems a little off |
| 16:53.01 | adeel|work | and the provider is set to trust rpid |
| 16:59.02 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:03.41 | Qwell | that does seem a little off |
| 17:04.24 | Qwell | The parser is actually checking for "screen=", and not just "screen" |
| 17:07.34 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 17:13.50 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 17:16.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:21.51 | adeel|work | sorry, the 2nd leg RPID header is actually: Remote-Party-ID: <sip:647523XXXX@193.46.30.28>;party=calling;privacy=off;screen=no .... |
| 17:22.45 | adeel|work | is there a method to manipulate the RPID that * is sending out? specifically, based upon the npi/ton value of the A-leg, i need to prefix the user field with a + or 011 |
| 17:25.38 | mjordan | adeel|work: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
| 17:26.15 | adeel|work | mjordan, yeah i'm reading that now....however, that doesn't specify if that affects the RPID header on the B-leg...i'm assuming it does (and am about to test it) |
| 17:26.36 | mjordan | it does |
| 17:26.43 | mjordan | if you've set the information on the appropriate channel |
| 17:27.21 | adeel|work | i noticed that the CALLERID() function has a 'ton' datatype...any idea if that matches the ISDN specs? |
| 17:29.13 | mjordan | ton is an alias for num-plan |
| 17:29.54 | mjordan | without digging into it, I'm not sure whether or not how it will be used throughout the code. |
| 17:30.39 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:30.39 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:31.05 | adeel|work | well, not technically....ton is a secondary descriptor for the NPI, since the NPI can be unknown (0), E.164 Public (1) or Private (9) |
| 17:31.09 | adeel|work | the ton then subclassifies |
| 17:45.18 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:48.44 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 18:03.52 | *** join/#asterisk WindBack (~quassel@190.123.122.18) |
| 18:04.22 | WindBack | Is there anya way to change voicemail password from the dialpan? |
| 18:05.29 | pabelanger | voicemail() |
| 18:07.13 | citywok | I have had several crashes lately, finally caught it with debugging turned on. Does this look like something to anybody? http://pastebin.com/25QTy4WJ |
| 18:08.19 | cmendes0101 | citywok: Is that all the debug that came out of the crash? |
| 18:08.42 | citywok | cmendes0101: no, but that's the snippit related to it. the gdb output is thousands of lines of that same output. |
| 18:08.58 | citywok | err i meant with asterisk -g, not debugging. that is the gdb trace of the core dump. |
| 18:09.57 | citywok | Right before asterisk dies it spits out 700 of these: [2012-09-24 15:04:06] ERROR[23926] astobj2.c: refcount -477 on object 0xaeb36048 |
| 18:10.34 | cmendes0101 | Hmm ok, I don't really see anything that points to what the problem is but this is probably over my head lol |
| 18:11.16 | mjordan | citywok: what version? |
| 18:11.26 | citywok | started on 1.8.13 so i updated to 1.8.16 |
| 18:11.33 | citywok | and it happened twice yesterday |
| 18:11.40 | mjordan | citywok: looks like a ref counting problem |
| 18:11.42 | citywok | happens right when a call gets originated (via AMI) |
| 18:12.21 | mjordan | citywok: you may want to enable ref count debugging in chan_sip: https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging |
| 18:12.24 | citywok | each time it's crashed a call was originated, the 700 refcounts get spit out, and asterisk dumps. we make 5000 calls the same way each day, and it has been happening just about once every other day (with a nightly restart) |
| 18:12.44 | mjordan | well, its definitely a problem. |
| 18:14.30 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 18:14.30 | citywok | mjordan: okay, enabled & recompiling. I won't be able to load that until after production tonight, or if it crashes between now and then. :p |
| 18:15.03 | citywok | mjordan: if i have another crash, should i post the /tmp/refs log along with my backtrace? |
| 18:15.42 | mjordan | yes |
| 18:15.53 | mjordan | please open an issue in JIRA when you have the ref debug log and the backtrace |
| 18:16.39 | citywok | is the entire backtrace needed or is it okay if i clip it? It's many thousands of lines of the same refcount errors. |
| 18:16.59 | mjordan | the whole backtrace is fine |
| 18:17.05 | mjordan | just attach as a text file |
| 18:17.18 | mjordan | way down in the stack trace it might provide who started the whole crazy chain to nowhere |
| 18:17.18 | citywok | okay, good to know |
| 18:17.30 | citywok | haha |
| 18:17.56 | citywok | i'm not sure what has changed, we haven't really made changes to our PBX in a while and this went from never happening to several times in the last 2 weeks, seemingly out of nowhere. |
| 18:18.22 | citywok | at least if i had an idea i could undo whatever was changed. oh well :-\ |
| 18:19.19 | *** join/#asterisk jsjc (~Adium@164.148.78.188.dynamic.jazztel.es) |
| 18:22.27 | WindBack | pabelanger: i didn't find a way to change the voicemail password from the voicemail application (inside dialplan). Are you sure that it is possible? |
| 18:31.43 | *** join/#asterisk nisam (nisam@115.242.85.73) |
| 18:32.14 | nisam | hi friends |
| 18:32.23 | nisam | i require one help |
| 18:34.28 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:34.29 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:34.46 | Qwell | nisam: just one help? |
| 18:35.06 | nisam | ;) |
| 18:35.14 | nisam | i new new to asterisk.. |
| 18:36.54 | nisam | Qwell: i installed asterisk and working fine. i have created sip trunk to provider. when i am calling from sip phone i am getting forbiddon error..why..? |
| 18:37.24 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:38.07 | nisam | @Qwell...do i require any SIP proxy server |
| 18:38.17 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a1d92.pool.mediaWays.net) |
| 18:38.21 | Qwell | no |
| 18:40.29 | nisam | provider given one user name..when i am calling from client phone to outside which user name will reach on provider side |
| 18:41.53 | nisam | ...? |
| 18:42.36 | drmessano | The one you set up on the peer |
| 18:44.59 | *** join/#asterisk Deeewayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 18:44.59 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:47.40 | nisam | means it is not my client user name..correct..? |
| 18:48.12 | Qwell | ~book |
| 18:48.13 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:48.18 | Qwell | nisam: You need to read that. |
| 18:49.38 | Kobaz | does IFTIME hack off the current minutes for comparison? |
| 18:49.57 | Kobaz | Set(during_hours=${IFTIME(17:00-19:00?1:0)}); [2012-09-19 19:00:14] VERBOSE[18906] pbx.c: [2012-09-19 19:00:14] -- <SIP/pbx-sip-t1-gw1-00001ae4> Executing [1@ivrHardSurfaceNight:1] Set(during_hours=1) |
| 18:50.02 | Kobaz | that looks like it should be false to me |
| 18:50.20 | Kobaz | er, current seconds rather |
| 18:51.06 | Kobaz | i would think current time of 19:00:04 should not fall into 17:00-19:00 |
| 18:51.08 | nisam | no no.. i am little bit confused when iam read about SIP proxy |
| 18:51.24 | Kobaz | i wonder if IFTIME takes seconds |
| 18:51.25 | nisam | i am a sarter |
| 18:54.05 | Kobaz | looks like seconds gets hacked off |
| 18:54.14 | Kobaz | Set(foo=${IFTIME(0:00-14:53:00?1:0)}); |
| 18:54.16 | Kobaz | is still true |
| 18:54.24 | Kobaz | even if the current time is 14:53:30 |
| 18:54.54 | drmessano | nisam: Asterisk is a B2BUA. Your connection from your phone to your PBX and from your PBX to your provider are different. Different peers. The BOOK will show you how all this works |
| 18:55.04 | Qwell | Kobaz: Your syntax is...broken |
| 18:55.12 | Kobaz | borken? |
| 18:55.34 | Qwell | oh, god, that's terrible. |
| 18:55.38 | Qwell | Who wrote that function? |
| 18:55.41 | Qwell | glares at them |
| 18:55.44 | Kobaz | i don't know |
| 18:55.54 | Kobaz | but i looked at the code a few times |
| 18:56.00 | Qwell | no, you're right :( |
| 18:56.03 | Kobaz | half the stuff you send to IFTIME doesn't work |
| 18:56.13 | Kobaz | like if you try and do sat|sun for days |
| 18:56.20 | Kobaz | it thinks sun is invalid |
| 18:56.40 | Kobaz | sat by itself works, sun by itself works |
| 18:56.43 | Kobaz | but sat|sun = fail |
| 18:57.53 | Kobaz | so i think i have to to 16:59 instead of till 17:00, due to the the lack of support for seconds |
| 19:07.33 | Kobaz | struct ast_timing doesn't have a field for seconds |
| 19:08.35 | nisam | drmessano: thanks for your explanation. i got some points |
| 19:09.18 | Kobaz | i->minmask[st_h * 2 + (st_m >= 30 ? 1 : 0)] |= (1 << (st_m % 30)); |
| 19:09.20 | Kobaz | i love this code |
| 19:23.40 | *** join/#asterisk angryuser_laptop (~angryuser@85-171-142-218.rev.numericable.fr) |
| 19:35.11 | *** join/#asterisk angryuser_laptop (~angryuser@85-171-142-218.rev.numericable.fr) |
| 19:44.30 | kresp0 | hi all, |
| 19:44.30 | kresp0 | I've several asterisks using DUNDi. It works, except if I do a lookup for a number that is on the same host. why? how should I fix it? |
| 19:44.52 | Katty | hi kids. |
| 19:45.05 | kresp0 | hi :) |
| 19:45.37 | Katty | Qwell: i have a theory. |
| 19:47.12 | Katty | Qwell: if you could design a program, based on a set of constant variables... |
| 19:47.44 | Katty | Qwell: such as...basic chemical reactions... |
| 19:48.06 | Katty | Qwell: rust = iron plus 2 oxygen molecules... |
| 19:48.25 | Katty | Qwell: rust then becmoing ${newvariable} |
| 19:48.56 | Katty | Qwell: you could technically write a program to come up a way to reverse engineer any possible set of complex reactions |
| 19:49.14 | Katty | Qwell: and then from that list of possible actions, likely pick the most plausible |
| 19:50.05 | Katty | Qwell: it would require a huge database with all the reactions, possible catalysts... |
| 19:50.38 | Katty | Qwell: ions, compounds...all those fancy special metals in the middle of the periodic table |
| 19:51.41 | Qwell | uh huh |
| 19:52.21 | Katty | Qwell: well anyone, say pennacillin |
| 19:52.31 | Qwell | I would guess such a thing would already exist. |
| 19:52.37 | Katty | Qwell: it molecularly binds to the bacteria around the cell, changing its physical composition |
| 19:53.10 | Katty | Qwell: but given a set of variables, a program could run through allt he statistics and present the most plausible ways of 'curing' an infection |
| 19:53.15 | Katty | Qwell: based on it's molecular makeup |
| 19:53.49 | Katty | Qwell: maybe. |
| 19:53.54 | Katty | Qwell: if not, someone should write it |
| 19:54.12 | Qwell | see Folding@Home |
| 19:55.09 | *** join/#asterisk Luke-Jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 19:55.11 | Katty | that's hot. |
| 19:56.51 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 19:57.10 | carrar | no db can hold chaos |
| 19:57.47 | Katty | it's not chaos. |
| 19:57.51 | Katty | it's just a lot of variables. |
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| 20:17.33 | scgm11 | anyone knows how to fix safari on mac for issues.asterisk.org |
| 20:17.43 | scgm11 | because I can´t access through there |
| 20:18.17 | scgm11 | says that the server did not accept the certificate |
| 20:19.26 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 20:19.29 | sruffell | hmm....confirmed....strange |
| 20:19.32 | sruffell | newtonr: ^^^ |
| 20:21.39 | *** join/#asterisk scgm11_ (~Sebastian@r186-50-71-64.dialup.adsl.anteldata.net.uy) |
| 20:22.19 | jeffspeff | scgm11, try another browser? |
| 20:22.49 | pabelanger | sruffell, that is a know issue |
| 20:23.02 | pabelanger | I though kpfleming fixed it though |
| 20:23.02 | scgm11_ | yes other browsers work |
| 20:23.03 | sruffell | pabelanger: orly? :/ ok.... |
| 20:23.19 | sruffell | I didn't think any cert was required to just connect...only for certain accounts to login. |
| 20:23.25 | pabelanger | right |
| 20:23.41 | pabelanger | something in safari does not read the apache2 configuration correctly |
| 20:24.03 | scgm11 | try to find on google this seems to be an issue for a long time now |
| 20:24.21 | scgm11 | but no solution yet |
| 20:24.40 | pabelanger | http://lists.digium.com/pipermail/asterisk-dev/2011-July/049996.html |
| 20:24.46 | pabelanger | scgm11, sruffell ^ |
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| 20:25.07 | sruffell | nods |
| 20:25.11 | sruffell | thanks |
| 20:26.16 | pabelanger | I think JIRA was upgrade the other week by serafie. Might be worth checking the configuration again and see if the issue is back |
| 20:26.22 | pabelanger | could be wrong |
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| 21:16.39 | fubada | hi. i have a sweet asteirsk10 config working in a single office. so sweet that unfortunately "they" want another asteirsk box in a remote office. the two are bridged via vpn, and they want to dial extensions as if they are local |
| 21:16.45 | fubada | is this possible? |
| 21:16.58 | Qwell | sure. IAX2 switch |
| 21:17.10 | fubada | why iax2 |
| 21:17.20 | fubada | cant i peer my asterisks boxes with sip |
| 21:17.22 | aster1sk | Much simpler set up. |
| 21:17.31 | aster1sk | IAX also only needs one port forwarded. |
| 21:17.35 | Qwell | Because an IAX2 switch does exactly what you want. |
| 21:17.40 | fubada | okay, but does that require me to switch to iax2 for my phones also? |
| 21:17.49 | aster1sk | Nope. |
| 21:17.53 | Qwell | Asterisk is a B2BUA... |
| 21:18.03 | fubada | anyone got a doc handy? |
| 21:18.03 | Qwell | one side doesn't care what the other side is |
| 21:18.07 | Qwell | ~book |
| 21:18.07 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:18.13 | Qwell | leifmadsen: You do document that, right? |
| 21:18.27 | leifmadsen | eh? |
| 21:18.33 | Qwell | IAX2 switches |
| 21:18.36 | aster1sk | Whoa leifmadsen haven't seen you in a while. it's http://geekhut.org |
| 21:18.39 | leifmadsen | we don't document switches |
| 21:18.43 | Qwell | lame |
| 21:18.55 | fubada | is this a hardware device or? |
| 21:18.55 | leifmadsen | doesn't believe in switches |
| 21:19.27 | leifmadsen | aster1sk: oh snap :) |
| 21:19.44 | aster1sk | We'll be at Gracies King / Duncan if you care to come by (round 6) |
| 21:20.15 | leifmadsen | aster1sk: heh I'm in Caledon now :) |
| 21:20.33 | aster1sk | Ahh too bad, welp I should pipe down as to not flood with personal things. |
| 21:20.58 | leifmadsen | heh |
| 21:22.07 | tompaw | hates Avaya. |
| 21:22.44 | aster1sk | Leavinrg now. |
| 21:23.10 | tompaw | Why make an IP phone that requires an httpd to set up? :/ |
| 21:24.53 | [TK]D-Fender | Because it's proprietary and you're not even supposed to know that's how it works..... |
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| 22:24.41 | WIMPy | Qwell: Switches seem to be almost non existant. The sectret features for a lucky few. |
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| 23:54.38 | skroot | Once asterisk is running (1.8 on Ubuntu) how can I verify my gtalk connection is connected? |
| 23:54.53 | [TK]D-Fender | use it |
| 23:55.12 | skroot | heh |
| 23:55.12 | skroot | well |
| 23:55.23 | skroot | i dont have the voip device hooked up or a phone yet |
| 23:55.30 | skroot | i didnt know if i could at least verify im that far though |
| 23:55.37 | [TK]D-Fender | So have * do something with the call |
| 23:55.49 | [TK]D-Fender | Everything short of using = lie |
| 23:56.01 | skroot | heh |
| 23:56.07 | skroot | alright, i gotta wait for my wife to get homwe |
| 23:56.10 | skroot | my cell is my gv # |