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| 00:36.16 | gusto | p3nguin: i am talking about that driller in a bin |
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| 00:48.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:55.02 | Sean-Der | Does anyone else use Debian as their host for Asterisk? |
| 01:55.21 | Sean-Der | I just pulled it on Debian and their asterisk-config package and don't know if I really need all of this stuff |
| 01:55.27 | WIMPy | Start a limesurvey. |
| 01:55.53 | Sean-Der | I feel overwhelmed by all the skeleton configs, or is this really all needed? |
| 01:56.11 | Sean-Der | More configs than apache 0_0 |
| 01:56.13 | WIMPy | You ounly need the configs for the parts you want to use. |
| 01:57.37 | WIMPy | That's not hard. |
| 01:57.58 | Sean-Der | should I just move everything into another folder for now and move as I need |
| 01:58.18 | Sean-Der | WIMPy: I just don't want a bunch of stuff enabled that I don't need |
| 01:58.37 | WIMPy | The best way is not to load it in the first place. |
| 01:58.58 | WIMPy | Some things won;t load without configuration. Others use defaults. |
| 01:59.15 | WIMPy | Or you can not even build them, off course. |
| 02:00.08 | Sean-Der | Ok I will, I just need the wiki to get a better grounding |
| 02:00.30 | WIMPy | Try the |
| 02:00.34 | WIMPy | ~book |
| 02:00.34 | 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 |
| 02:03.43 | Sean-Der | Ok thanks WIMPy ! |
| 02:15.52 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
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| 02:19.54 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:43.18 | gusto | i do not understand this dnsmgr.conf thing. do i need some module for it, or is it always there and i can just create a dnsmgr.conf file and say [general] \ enable=yes there and everything is OK then? |
| 02:44.57 | p3nguin | Check the sample file. |
| 02:46.04 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:48.23 | gusto | what sample file? |
| 02:48.56 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 03:00.11 | slav3_kitten | ok, what's a low cost server that will happily run 20 sip channels with very little if any transcoding? |
| 03:19.49 | *** join/#asterisk mintos (~mvaliyav@114.143.160.125) |
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| 05:44.37 | elliot98 | can one set t38pt_udptl to "1" instead of "yes"? |
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| 05:47.35 | *** join/#asterisk Lepon (Lepon@60-241-109-208.static.tpgi.com.au) |
| 05:48.57 | Lepon | question for anyone with any idea. My asterisk server has stopped generating ring tones for out going calls. We originally had Dial(blah,rgx) and it has been working fine for years. last week it just suddenly stopped outputting any ring tone (i can't see anything that has changed to cause it) |
| 05:49.21 | Lepon | I change it to Dial(blah,,Rgx) and it seem to work again but now its stopped |
| 05:49.59 | Lepon | doesn't matter what options I put or don't put in the dial command I get no ring tone when the other end is ringing (calling out over a voip provider to external number) |
| 05:50.24 | Lepon | any ideas would be much apperciated |
| 05:56.02 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
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| 06:04.48 | Alphacube | Mornin peeps |
| 06:05.46 | Dennisvj | morning |
| 06:10.36 | Jo_Bell | When asterisk works as a B2BUA (between 2 SIP clients), once the session is established, how is the media data flow? Is it from client 1 to Asterisk to Client 2 and then from Client 1 to Asterisk to Client 1? |
| 06:15.25 | Lepon | question for anyone with any idea. My asterisk server has stopped generating ring tones for out going calls. We originally had Dial(blah,rgx) and it has been working fine for years. last week it just suddenly stopped outputting any ring tone (i can't see anything that has changed to cause it) |
| 06:15.25 | Lepon | I change it to Dial(blah,,Rgx) and it seem to work again but now its stopped. doesn't matter what options I put or don't put in the dial command I get no ring tone when the other end is ringing (calling out over a voip provider to external number) |
| 06:15.26 | Lepon | Any ideas would be much apperciated |
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| 06:45.31 | *** join/#asterisk Samuel_CTX (~svanderp@62-213-207-59.colo.housingcenter.be) |
| 06:51.23 | bulkorok | hi |
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| 07:28.02 | Jo_Bell | Hi I have a query - in the case of same network and remote networks, what is the difference of media data flow? In the case of same network, will data flow through Asterisk also? Or will the clients interact between themselves? |
| 07:28.33 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 07:35.25 | bulkorok | Jo_Bell: you can define that in sip.conf with directmedia=yes/no |
| 07:35.41 | bulkorok | usually all traffic runs through asterisk |
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| 07:41.08 | oyugik | Hey anyone used asterisk and quexs before? |
| 07:45.24 | *** join/#asterisk Alex_Bkash (b4eaf0d5@gateway/web/freenode/ip.180.234.240.213) |
| 07:46.15 | Alex_Bkash | I need to use asterisk on a mini linux distro |
| 07:46.32 | Alex_Bkash | which distro will be better |
| 07:46.38 | Alex_Bkash | any suggestion plz |
| 07:54.04 | Samuel_CTX | Debian? Or is that to big? |
| 07:57.00 | Dennisvj | Depends, debian is easy to maintain. Any flavour is sufficient tho. |
| 07:58.18 | oej | Morning folks! |
| 07:58.37 | oej | Look at astlinux. It's made for just that use. |
| 07:58.40 | *** join/#asterisk RobZombie (cce48afa@gateway/web/freenode/ip.204.228.138.250) |
| 08:00.50 | RobZombie | Anyone around that could help out with a few asterisknow questions from a noob? |
| 08:01.25 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 08:01.44 | Alex_Bkash | okk |
| 08:01.47 | Alex_Bkash | i'll try |
| 08:01.54 | Alex_Bkash | astlinux |
| 08:02.32 | *** join/#asterisk rolandow (~rolandow@546BB29B.cm-12-4c.dynamic.ziggo.nl) |
| 08:03.16 | rolandow | hi guys.. anybody awake yet? :) |
| 08:03.22 | RobZombie | Okay, I got a new machine today, installed the AsteriskNOW current stable. its all up and working etc. Got the gui going and the SHH. So I'm kind of stuck here. I firstly want to use, the built in apache server thats running to host my own person site and I'm unsure how to set that up. |
| 08:03.27 | rolandow | i wonder if there's a good article around the net about debugging with wireshark |
| 08:03.39 | rolandow | or any other packet sniffer |
| 08:04.24 | RobZombie | Not a domain for my gui, I want to use it to host my site on my local machine. I have the router setup to dedicate 75/75 bandwidth for it, and I have a 250/250m connection, full fiber. So I shouldnt have issues. |
| 08:04.31 | rolandow | RobZombie: i think you want to google for virtualhosts |
| 08:05.03 | rolandow | although i don't know asterisknow all that well :) |
| 08:05.09 | *** join/#asterisk areq (~areq@pld-linux/areq) |
| 08:05.17 | RobZombie | Why though, Apache is there and installed and working, just dont now how to configure a domain with it. |
| 08:05.25 | rolandow | but if it's running apache, couldn't you just add a virtual host? |
| 08:06.30 | RobZombie | I mean with apache it should be the same for any provider, just configure it and the configure the registar and should be all gravy? I just dont know how to do it. |
| 08:07.05 | areq | hi |
| 08:07.12 | RobZombie | hello |
| 08:07.16 | Alphacube | Hello |
| 08:07.32 | areq | i'm tying to run g722 |
| 08:07.41 | areq | cisco spa 303 |
| 08:07.43 | rolandow | i don't know what distro asterisknow is based on |
| 08:08.00 | RobZombie | So is this the best room to speak out AsteriskNOW in considering its just asterisk with freepbx gui right? |
| 08:08.03 | areq | <PROTECTED> |
| 08:08.03 | areq | <PROTECTED> |
| 08:08.03 | areq | <PROTECTED> |
| 08:08.03 | areq | <PROTECTED> |
| 08:08.03 | areq | <PROTECTED> |
| 08:08.08 | rolandow | but you will probably find config files in /etc/apache2 (debian ubuntu) or /etc/httpd for i think redhat |
| 08:08.09 | RobZombie | I'm taking a shotat that comment, I'm unsure. |
| 08:08.31 | RobZombie | It's curent stable release of centos as the os |
| 08:08.35 | areq | spa303(g722)-sip-asterisk-sip-spa303(g722) |
| 08:08.43 | rolandow | i think centos has /etc/httpd or something |
| 08:08.55 | areq | why is Transcode to alaw and slin ? |
| 08:08.59 | rolandow | you should probably just google for "apache virtualhost centos howto" |
| 08:09.21 | RobZombie | Okay I;ll give that a shot, thanx for the input mate. |
| 08:09.21 | rolandow | you will find config examples then |
| 08:09.22 | RobZombie | cheers |
| 08:09.43 | rolandow | and there's probably something default in there already .. so combined with that you should be able to construct the config file |
| 08:10.00 | rolandow | good luck! |
| 08:10.24 | RobZombie | thanx mate |
| 08:10.41 | areq | any idea how avoid transcode ? |
| 08:11.32 | rolandow | areq: i think if both parties support g722, asterisk will not transcode |
| 08:12.11 | areq | rolandow: both parties support |
| 08:13.48 | areq | but core show channel show that asterisk transcode ;( |
| 08:14.25 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:14.34 | rolandow | hmm that's too bad .. |
| 08:14.38 | rolandow | don't know then :( |
| 08:17.25 | Dennisvj | areq: try forcing the right codec in sip.conf by disallowing all and enable only g722: http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 08:18.20 | Dennisvj | Don't use it myself, but it seems there are different versions of g722, or so it seems as you look on the wiki link. Perhaps anyone can shed some light here about the g722 codec. |
| 08:19.54 | areq | in sip.conf I disable all and allow only g722 and alaw |
| 08:20.32 | areq | nat all my phones can use g722, and on ISDN E1 I habe alaw |
| 08:21.29 | areq | my idea is: betwean spa303 via SIP use g722 without transcode, other connection alaw without transcode |
| 08:21.48 | Dennisvj | ah I see your point. |
| 08:22.22 | areq | on spa spa303: prefered codes is g722, second alaw |
| 08:27.10 | Dennisvj | My experience is that asterisk cannot handle that. your SIP device always goes for the preffered codec |
| 08:27.24 | Dennisvj | I have a similar issue with alaw and another codec |
| 08:28.10 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:31.03 | bulkorok | how do I can decode captured g729!? |
| 08:34.31 | Dennisvj | areq: See ${SIP_CODEC} http://www.voip-info.org/wiki/view/Asterisk+variables In the dialplan one should be able to arrange this. I have to try this myself tomorrow :) |
| 08:38.10 | Dennisvj | bulkorok: on Windows I used this some time ago: http://www.voiceage.com/openinit_g729.php There are some wireshark options, google around a bit |
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| 08:56.57 | areq | 8 |
| 08:56.58 | areq | [Jul 6 10:55:53] NOTICE[531]: chan_sip.c:6455 try_suggested_sip_codec: Changing codec to 'g722' for this call because of ${SIP_CODEC} variable |
| 08:57.09 | areq | and still transcode ;( |
| 09:04.25 | *** join/#asterisk philfine (~tino@a95-92-251-229.cpe.netcabo.pt) |
| 09:04.34 | philfine | Hello everyone |
| 09:07.26 | philfine | As I read DUNDi is intended to distribute number recognition and eventually forward calls through SIP instead of traditional lines. Although I don't see any procedure for connection to any public dundi enabled asterisk box, but only few tutorial experiments with several "in house" ones. |
| 09:07.40 | philfine | Did I miss understood its goal |
| 09:08.41 | philfine | I though I could announce my own traditional line numbers through DUNDi and eventually someone also connected to DUNDi would be able to connect using my traditional numbers, but through SIP |
| 09:12.31 | rolandow | philfine: that's how i understand it as well |
| 09:13.11 | rolandow | i think you can query a sip account based on the number, and then just dial that sip account |
| 09:17.18 | rolandow | hm... i have an issue with two dect phones that have one way audio.. with other phones it's okay.. problem is with the two dect sets. |
| 09:17.22 | rolandow | any idea's? |
| 09:18.57 | philfine | No idea sorry |
| 09:19.24 | philfine | Are you connected to a DUNDi network ? |
| 09:20.43 | rolandow | nope.. but i read about it in the book :) |
| 09:20.59 | rolandow | and figured i may check it out one day .. i think it's also less secure |
| 09:21.09 | rolandow | because you'd have to allow anyone to connect to your box |
| 09:21.46 | rolandow | do i didn't want to struggle with security too much, i just firewalled the ports and opened them up for only my users. |
| 09:32.15 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 09:36.24 | philfine | rolandow: Which book ? |
| 09:39.37 | *** join/#asterisk like_a_horse (~like_a_ho@firect.saao.ac.za) |
| 09:43.19 | rolandow | ~book |
| 09:43.19 | 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 |
| 09:43.30 | rolandow | that book .. |
| 09:43.41 | rolandow | i must say i read that before i discovered this channel :) |
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| 10:41.27 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:42.36 | philfine | Up until now it seems to me DUNDi is defined to be shared between service providers and not consumers |
| 10:43.01 | philfine | There is actual DUNDi open server |
| 10:43.09 | philfine | …no... |
| 11:02.11 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 11:02.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:18.05 | *** join/#asterisk RZero (~RZero@gateway-hq-uk.oxygen8.com) |
| 11:20.27 | RZero | Can any tell me if it is possible to have meetmecount count the number of users when some one hangs up in a conf call |
| 11:20.29 | RZero | '? |
| 11:36.34 | *** join/#asterisk gusto (~gusto@DSL01.83.171.176.215.ip-pool.NEFkom.net) |
| 11:36.44 | gusto | that dnsmgr seems to work btw. |
| 11:37.10 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.169.248.dsl.dyn.forthnet.gr) |
| 11:57.21 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-78-180.bois.qwest.net) |
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| 12:19.40 | rolandow | [TK]D-Fender: are you on the mailing list of asterisk as well? |
| 12:20.00 | rolandow | [TK]D-Fender: since you seem to know much about asterisk and answers a lot of questions |
| 12:21.15 | [TK]D-Fender | rolandow, Nope. I've just been in here for the past 8 years |
| 12:21.27 | *** join/#asterisk Neptu (~Neptu@adsl196-248-165-217-196.adsl196-14.iam.net.ma) |
| 12:21.54 | rolandow | hm .. ok .. |
| 12:22.19 | rolandow | because i think irc isn't really suitable for comprehensive questions |
| 12:22.32 | rolandow | one would be flooding the channel :) |
| 12:22.53 | [TK]D-Fender | rolandow, I would beg to differ. |
| 12:22.59 | WIMPy | That's what pastebins are for. |
| 12:23.06 | [TK]D-Fender | yup |
| 12:23.15 | rolandow | also for large questions? |
| 12:23.23 | [TK]D-Fender | yup |
| 12:24.42 | rolandow | could an attachment be used as well? a little drawing would be nice. |
| 12:25.08 | *** join/#asterisk wonderworld (~ww@dsdf-4db5d187.pool.mediaWays.net) |
| 12:25.22 | rolandow | anyways.. i'll write down my story and see what comes of it.. |
| 12:25.24 | [TK]D-Fender | tinypic.com / imagebin.ca / hostityourselfyoulazyass.whatever :) |
| 12:26.05 | [TK]D-Fender | <PROTECTED> |
| 12:26.12 | rolandow | yes.. but then they're seperated |
| 12:26.17 | rolandow | hehe |
| 12:26.21 | rolandow | you'll see :) |
| 12:30.32 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 12:31.09 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
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| 12:39.53 | rolandow | http://pastebin.com/xNjhkDdX |
| 12:40.14 | rolandow | there's my story .. sort of drama |
| 12:40.19 | rolandow | maybe comedy for the experts ;) |
| 12:41.20 | [TK]D-Fender | Handsets don't use "ports", they speak DECT. Its the BASE that speaks |
| 12:41.57 | *** join/#asterisk af_ (~getsmart@88.149.212.175) |
| 12:42.06 | [TK]D-Fender | Sounds like a basic NAT setup you need to make |
| 12:42.07 | rolandow | yes.. that's true |
| 12:42.15 | [TK]D-Fender | ~sipnat |
| 12:42.15 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 12:42.17 | [TK]D-Fender | ^^ |
| 12:42.34 | [TK]D-Fender | and swap "canreinvite=no" for "directmedia=no" for 1.6+ |
| 12:42.54 | [TK]D-Fender | And that was entirely too full of fluff :) |
| 12:43.09 | rolandow | what was? my story? |
| 12:43.17 | rolandow | but if it's NAT: howcome the tiptel works? |
| 12:43.23 | rolandow | same route, same connections .. |
| 12:43.44 | [TK]D-Fender | rolandow, Becuase it is smarter and know the WAN IP, etc most likely |
| 12:44.25 | [TK]D-Fender | rolandow, * side should be port forwarded, no others. |
| 12:44.54 | [TK]D-Fender | And set "type=friend" for those multi-device ones that are remote behind the same router |
| 12:46.09 | rolandow | * side should be port forwarded: i opened the 5060 and rdp ports on the asterisk public ip. |
| 12:46.32 | rolandow | is that what you mean? i don't need to port forward for every SIP account/device behind * |
| 12:46.56 | rolandow | yes, they are type=friend |
| 12:47.13 | [TK]D-Fender | ok, check the rest from the guide |
| 12:47.23 | rolandow | yes i will study that first |
| 12:47.53 | rolandow | just curious: if tiptel would be that smart, would it "detect" the ip address by checken whatismyip.org or something similar?? or does it get that info from * ? |
| 12:48.24 | rolandow | because the public ip of the tiptels is different than the public ip * is on .. |
| 12:48.48 | rolandow | so unless * tells it's public ip somewhere in the sip, there's no way the tiptel can know imho |
| 12:48.58 | [TK]D-Fender | It would be getting it itself. Go make sure all the other settings are right, and then pastebin a failed call with SIP DEBUG enabled. |
| 12:52.24 | rolandow | my * is not behind nat, should i still put directmedia=no in the general section? |
| 12:52.35 | [TK]D-Fender | yes. |
| 12:52.38 | rolandow | ok |
| 12:52.47 | [TK]D-Fender | So * has a public IP right on the box itself? |
| 12:52.54 | rolandow | yes |
| 12:53.16 | [TK]D-Fender | then your devices should all be "nat=yes" , "directmedia=no" |
| 12:53.47 | rolandow | yes .. i had that.. |
| 12:54.13 | [TK]D-Fender | rolandow, pastebin a complete call attempt with SIP DEBUG enabled |
| 12:54.27 | rolandow | well i'm still reading the guide you gave me |
| 12:54.59 | rolandow | but in [general] i put nat=no, because asterisk box has public ip .. on all devices i put nat=yes.. correct? |
| 12:56.26 | [TK]D-Fender | Seither way should do it. |
| 12:56.27 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:56.27 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:56.53 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 12:56.54 | [TK]D-Fender | rolandow, What is the remote router model? |
| 12:57.46 | rolandow | not sure .. i think it's all netgear |
| 12:58.03 | rolandow | i'd have to check with out sysop but he's not in now |
| 12:58.17 | rolandow | you think it has sip_alg ? |
| 12:58.20 | *** part/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 12:58.56 | rolandow | i'll check with our sysop when he's back |
| 12:59.49 | [TK]D-Fender | Could be. Make sure that's disabled first |
| 13:00.05 | [TK]D-Fender | Also some routers just mangle their NAT implementation... |
| 13:00.17 | rolandow | how can i dump sip debug to a specific file? |
| 13:00.31 | treborsux | <PROTECTED> |
| 13:00.31 | treborsux | [08:57] <treborsux> ON all my systems it does this on and off both my pri and dahdi systems |
| 13:00.31 | treborsux | [08:58] <treborsux> line goes to ring group and i get one way communication with the answerer hearing a ringing |
| 13:00.31 | treborsux | [08:58] <treborsux> the call 6062262231 is the call this occured on |
| 13:00.31 | treborsux | [08:59] <treborsux> Last question drmessano asked before i had to pick up my kids was is itjust 510? The answer is no any extension in the ring group that incoming rute goes to has it happen but just sometimes |
| 13:00.31 | treborsux | [09:00] <treborsux> Any ideas? |
| 13:00.44 | rolandow | ~sipdebug |
| 13:01.26 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:01.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:02.27 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bzprrflmzunejyad) |
| 13:02.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:05.24 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:06.57 | rolandow | [TK]D-Fender: http://pastebin.com/W3Y83a1F |
| 13:12.28 | RZero | <PROTECTED> |
| 13:12.45 | [TK]D-Fender | rolandow, I don't see the OUTWARD call's SIP debug |
| 13:13.30 | [TK]D-Fender | RZero, No, it counts the number of users when you CALL IT |
| 13:14.20 | treborsux | Fender will you look at that for me? I had to leave before drmessano could last night? |
| 13:14.50 | RZero | yeah thought as much being trying ways around it :) |
| 13:14.55 | RZero | to find * |
| 13:15.38 | [TK]D-Fender | RZero, your concept of when things are getting called seems skewed |
| 13:16.37 | [TK]D-Fender | RZero, When you call that app you get the count right then and there |
| 13:17.07 | RZero | I want a way to show how many people are in conf call not as some one enters |
| 13:18.12 | rolandow | [TK]D-Fender: what do you mean with outward? |
| 13:18.23 | rolandow | [TK]D-Fender: it's an extension on a remote location |
| 13:18.26 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:18.31 | [TK]D-Fender | rolandow, Executing [9030@StumpelZwaag:6] Dial("SIP/Adriaan_Snoek-000005dc", "SIP/Middenwaard_ALL,,tTkhH") in new stack |
| 13:19.14 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:19.20 | [TK]D-Fender | gah |
| 13:19.24 | rolandow | [TK]D-Fender: yes, middenwaard is our extenal location |
| 13:19.40 | [TK]D-Fender | and I only see the SIP debug from the internal side of that call |
| 13:19.51 | [TK]D-Fender | Not the debug for what is going to middenwaard |
| 13:20.10 | [TK]D-Fender | You are looking at HALF the picture |
| 13:20.19 | rolandow | ah ok .. i thought it would capture both sides |
| 13:20.29 | rolandow | ok i will turn on sip debug without ip filter then |
| 13:22.08 | [TK]D-Fender | treborsux, I see TWO calls with that # |
| 13:22.36 | [TK]D-Fender | treborsux, One INBOUND from PRI that gets to 510 via a ring-group, and ANOTHER, from the phone itself (510) TO that external number |
| 13:23.13 | *** join/#asterisk gtTuna (~TunaWork@38.101.60.42) |
| 13:23.14 | [TK]D-Fender | [Jul 5 15:38:30] VERBOSE[3045] sig_pri.c: -- Accepting call from '6062262231' to '1511' on channel 0/1, span 1 |
| 13:23.48 | [TK]D-Fender | <--- SIP read from UDP:10.26.80.212:5060 ---> INVITE sip:6062262231@10.26.80.201:5060 SIP/2.0 From: <sip:510@10.26.80.212>;tag=CBE0064E-727A1899 |
| 13:24.05 | [TK]D-Fender | [Jul 5 15:38:46] VERBOSE[3078] chan_sip.c: |
| 13:24.15 | [TK]D-Fender | 16 seconds later a call out. |
| 13:24.16 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 13:26.55 | [TK]D-Fender | treborsux, [Jul 5 15:38:47] VERBOSE[3078] chan_sip.c: Retransmitting #3 (NAT) to 10.26.77.201:5060: <- I'm alwso wondering why all of your PRIVATE IP devices are configured as though they were behind NAT, and that you are still getting timeouts requiring retransmissions all the time. |
| 13:27.45 | [TK]D-Fender | treborsux, [Jul 5 15:38:47] VERBOSE[3078] chan_sip.c: <--- SIP read from UDP:10.26.80.212:5060 ---> INVITE sip:6062262231@10.26.80.201:5060 SIP/2.0 From: <sip:510@10.26.80.212>;tag=CBE0064E-727A1899 |
| 13:28.13 | [TK]D-Fender | treborsux, ONE SECOND LATER. ANOTHER CALL |
| 13:28.30 | [TK]D-Fender | treborsux, What kind of foolish forwarding are you doing on that phone? |
| 13:28.44 | drmessano | I saw that too... |
| 13:29.11 | drmessano | There was something else I caught that looked like the device had a timeout |
| 13:29.12 | [TK]D-Fender | The derp is strong with this one... |
| 13:33.02 | rolandow | [TK]D-Fender: http://pastebin.com/r7xwFjxj |
| 13:33.07 | drmessano | Im gonna guess a firewall between the phones and PBX, and something incredibly reliable and awesome too, like a Sonicwall |
| 13:34.20 | [TK]D-Fender | drmessano, I've never had issues with mine... Disable ALG and it works just fine |
| 13:34.50 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:38.11 | drmessano | I managed a couple dozen TZ-170's a few years back, and they were hell. SIP wasn't the only problem. They desperated needed a nightly reboot option |
| 13:38.30 | [TK]D-Fender | rolandow, Ok, I'd check the remote firewall. |
| 13:38.38 | [TK]D-Fender | rolandow, The rest looks fairly OK |
| 13:38.44 | gtTuna | can someone help me with a Asterisk -> PBX SIP trunk issue? I'm trying to pass ~30 different DIDs over this SIP trunk, but it appears when it goes to send out, the SIP URI is of the following format: Unknown@SERVER_IP |
| 13:39.00 | [TK]D-Fender | gtTuna, Show us |
| 13:39.03 | [TK]D-Fender | ~pb |
| 13:39.03 | 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. |
| 13:39.04 | [TK]D-Fender | ^^^^^^^^^ |
| 13:39.09 | gtTuna | kk, sec |
| 13:39.21 | gtTuna | gotta sanitize it a bit, lol |
| 13:39.33 | [TK]D-Fender | DON'T |
| 13:39.50 | [TK]D-Fender | There should not be a pass in there for this as it is... |
| 13:40.04 | gtTuna | well, just IP addresses, but |
| 13:40.04 | [TK]D-Fender | And I'm not going to waste time on filtered debug. |
| 13:40.09 | [TK]D-Fender | Leave them |
| 13:40.14 | rolandow | [TK]D-Fender: spoke to the sysadmin, we have a netgear srxn3205 |
| 13:40.44 | rolandow | [TK]D-Fender: but i shouldn't open up any ports in the firewall right?? it wouldn't be a problem though, since i have only one base station on the other end. |
| 13:40.48 | [TK]D-Fender | rolandow, I know there haev been some Netgear, and a pile of D-Link's lower models that were broken as far as NAT goes... but check for ALG's, etc on it first |
| 13:41.12 | [TK]D-Fender | rolandow, Next step is do a more massive port forward on the remote side and test |
| 13:41.16 | [TK]D-Fender | After that |
| 13:41.59 | rolandow | [TK]D-Fender: yes, in our office we even had a netgear with sip_alg enabled, but not configurable yet.. we had to firmware upgrade it :( |
| 13:41.59 | gtTuna | http://pastebin.com/fwMnPGbT |
| 13:42.38 | rolandow | [TK]D-Fender: and of course lost all our settings.. didn't make good friends with my collegue |
| 13:42.40 | [TK]D-Fender | gtTuna, that is SIP OPTIONS, not a CALL |
| 13:42.51 | [TK]D-Fender | gtTuna, that is completely unimportant |
| 13:44.40 | rolandow | [TK]D-Fender: but if it's a router issue on the other end, it's still possible that my tiptel works just fine? did you see that the siemens isn't that smart? |
| 13:45.22 | [TK]D-Fender | rolandow, Not really... focus on the trouble one... remember its the router first in that chain... the local one to * is... well local... shouldn't matter |
| 13:46.31 | *** join/#asterisk ctaloi (~Chris@nat-66-218-1-130.usadatanet.com) |
| 13:46.44 | gtTuna | [TK]D-Fender, not sure if this is any better http://pastebin.com/F39ewajB |
| 13:47.25 | [TK]D-Fender | gtTuna, ASTERISK SIP DEBUG from CLI. a complete call. |
| 13:49.10 | [TK]D-Fender | INVITE sip:8594146383@216.29.249.130 SIP/2.0 SIP/2.0 404 Not Found User-Agent: LG-Ericsson iPECS-LIK 600 E.5Db <-- |
| 13:49.19 | [TK]D-Fender | They don't seem to like the number being passed |
| 13:49.42 | [TK]D-Fender | It hits 100 Trying which means the auth should ahve cleared so it appears the receiver doesn't like the number |
| 14:06.19 | rolandow | thanks again for your help [TK]D-Fender .. have a nice weekend! |
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| 15:18.42 | wcselby | o/ |
| 15:20.15 | wcselby | have an issue loading res_jabber.so at asterisk start - it doesn't seem to autoload with all of my other modules. If I specifically add a "preload => res_jabber.so" to my modules.conf, it still doesn't load it. If i add "load => res_jabber.so" it doesn't load. Running asterisk 1.8.13.1 downloaded and compiled last night. I am able to manually load res_jabber.so after the system has started just fine. It's just not loading it a |
| 15:20.15 | wcselby | utomatically or obeying the "preload / load" commands |
| 15:21.02 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:21.29 | wcselby | any thoughts, ideas, things for me to check? |
| 15:22.27 | wdoekes | start with -c and look for errors/warnings? |
| 15:22.39 | wdoekes | add -v's if needed |
| 15:22.53 | wcselby | that's actually what I"ve already done |
| 15:22.56 | wcselby | that's how I'm loading |
| 15:23.00 | wcselby | http://pastebin.com/d0wKFvwG |
| 15:23.06 | wcselby | the output |
| 15:23.13 | wcselby | of asterisk -c |
| 15:25.12 | wcselby | i get the erros on lines 58-61 because res_jabber.so isn't loading |
| 15:29.19 | leifmadsen | I suspect res_jabber.so isn't loading due to another module dependency not loading prior to it |
| 15:29.56 | leifmadsen | so you need to find out what is loading after res_jabber to cause it to not load and use a load => for that module prior to loading res_jabber |
| 15:30.56 | wcselby | shouldn't i get some error if I were to preload => res_jabber.so, if that were the case? |
| 15:31.08 | leifmadsen | it appears you aren't? |
| 15:31.12 | wcselby | no |
| 15:31.13 | leifmadsen | perhaps you should.... |
| 15:31.14 | leifmadsen | shrugs |
| 15:31.17 | wcselby | heh |
| 15:31.34 | wcselby | i know everything works fine on another machine running 1.8.13.0 |
| 15:31.44 | wcselby | i'm going to try loading that on this same machine and see if I still see the error |
| 15:31.54 | wcselby | that way I can tell if it's an error specific to my machine, or perhaps a bug with the build? |
| 15:32.00 | leifmadsen | you could check to see if it's a regression by testing the same configuration on the same machien with 1.8.13.0 vs 1.8.13.1 |
| 15:33.12 | wcselby | trying that now |
| 15:33.22 | wcselby | while that builds, apparenlty I have some honey-do's to take care of |
| 15:33.27 | wcselby | i'll be back around in a few minutes |
| 15:36.24 | Katty | cries |
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| 15:37.53 | leifmadsen | Katty: omg!? |
| 15:38.00 | leifmadsen | Katty: can you help me fix my system?! |
| 15:38.00 | Katty | today is too much )= |
| 15:38.04 | leifmadsen | it is broked! |
| 15:38.05 | Katty | YES |
| 15:38.08 | Katty | pours vodka on it |
| 15:38.13 | leifmadsen | omg what a great idea |
| 15:38.21 | leifmadsen | can we pour the vodka into a watermelon instead? |
| 15:38.30 | Katty | yes, if we also do a pineapple |
| 15:38.40 | leifmadsen | done |
| 15:39.31 | Katty | what do you spose grilled pineapple would taste like, if you soaked it in vodka? |
| 15:40.06 | _Corey_ | I think you'd burn off the alcohol that way |
| 15:40.17 | Katty | yeah you're probably right |
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| 15:42.45 | wcselby | :/ |
| 15:43.01 | wcselby | i'm having the same issue on 1.8.13.0 on this system as well, so it's something local |
| 15:44.52 | leifmadsen | look at the diff between the other machine I guess |
| 15:44.58 | leifmadsen | what modules are installed, etc |
| 15:45.07 | leifmadsen | suggests using something like chef or puppet for installing multiple systems |
| 15:45.27 | leifmadsen | the kickstand project has a lot of puppet manifests for installing asterisk systems |
| 15:45.49 | wcselby | well, these are basically two test systems that I use for playing around with new features |
| 15:46.04 | wcselby | so I'm not too concerned about automation |
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| 15:50.27 | wcselby | oh |
| 15:50.28 | wcselby | heh |
| 15:50.30 | wcselby | nevermind |
| 15:50.33 | wcselby | i see what happened |
| 15:50.38 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:50.40 | wcselby | it's working properly now |
| 15:51.06 | leifmadsen | for future reference, what was the issue? (since this room is logged) |
| 15:51.41 | wcselby | it helps if you remove the noload => res_jabber.so line from the bottom of the modules.conf file that I had added last night while testing something else... |
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| 15:52.40 | wcselby | slaps forehead and makes the homer "DOH" sound |
| 15:53.17 | wcselby | i probably shoudn't work on these kinds of things at 2 in the morning |
| 15:53.33 | wcselby | :) |
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| 16:01.16 | leifmadsen | wcselby: heh :) |
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| 16:02.00 | Weezey | anyone know where I can find SIP firmware for a Cisco 7970? |
| 16:02.16 | wcselby | from cisco |
| 16:02.28 | Weezey | other than from Cisco? |
| 16:02.34 | leifmadsen | use the power of google |
| 16:02.49 | leifmadsen | people here won't tell you how to obtain illegal software |
| 16:03.06 | leifmadsen | s/illegal software/software illegally/ |
| 16:03.18 | slav3_kitten | leifmadsen, it's not software |
| 16:03.24 | slav3_kitten | it's firmware, huge difference |
| 16:03.46 | wcselby | but not the whole legality part ofit |
| 16:04.01 | slav3_kitten | cisco being firmware nazis pisses me off too, you already have the equipment you should be able to get the firmware for the hardware you already own |
| 16:04.19 | leifmadsen | firmware and software are vastly different? |
| 16:04.22 | leifmadsen | suspects not |
| 16:04.44 | _Corey_ | leifmadsen: more tensile strength |
| 16:04.51 | slav3_kitten | you don't see d-link not offering updated firmware for their high end switches. they figure you have the switch you need the firmware |
| 16:04.59 | wcselby | i was able to pickup a cisco smartnet contract for like 8 bucks, that enabled me to legally download the firmware |
| 16:05.10 | leifmadsen | ya I had one of those a few years ago as well |
| 16:05.34 | wcselby | mine shoudl technically still be valid, however I don't use any cisco phones anymore |
| 16:05.35 | slav3_kitten | leifmadsen, firmware runs inside an embedded device and controls things way more low level |
| 16:05.45 | leifmadsen | ok... |
| 16:05.55 | leifmadsen | I understand what firmware is |
| 16:06.10 | leifmadsen | it's still a programmed application |
| 16:06.11 | slav3_kitten | wcselby, were and how did you get a smartnet contract for 8 bucks |
| 16:06.20 | wcselby | cdw ? |
| 16:06.26 | wcselby | let me find the link |
| 16:06.45 | slav3_kitten | leifmadsen, except it's not. you'd not say the linux kernel is an application. applications run on top of the kernel |
| 16:06.45 | _Corey_ | $8 sounds right |
| 16:06.48 | slav3_kitten | same with firmware |
| 16:07.10 | slav3_kitten | firmware is a kernel, applications run on top of the firmware |
| 16:07.22 | _Corey_ | slav3_kitten: The linux kernel is software, as is firmware, etc. |
| 16:07.27 | wcselby | here is the one i used, the price has gone up slightly to 11 bucks now |
| 16:07.28 | wcselby | http://www.cdw.com/shop/products/Cisco-SMARTnet-extended-service-agreement/993513.aspx?enkwrd=ALLPROD%3aCON-SNT-CP7941 |
| 16:07.31 | wcselby | that's for a 7941 |
| 16:07.50 | wcselby | the hardest part was dealing with the licensing people at cdw |
| 16:08.00 | slav3_kitten | well you need that for every device you have right? |
| 16:08.00 | wcselby | to prove i had an actual cisco 7941 |
| 16:08.12 | wcselby | you get one license, you donwload the firmware |
| 16:08.45 | wcselby | now, the whole legality issues revolving around cisco 79xx phones is another topic entirely |
| 16:08.54 | wcselby | i think Qwell has dealt with it pretty in depth |
| 16:09.26 | slav3_kitten | i mean i have a 7911, 7960, 7940, 7925, 2600xm, 2900xl, 3500xl by cisco. so i'd need 7 11 dollar smartnet agreements right? |
| 16:09.44 | wcselby | uh....probably? |
| 16:09.53 | wcselby | i know when I first got my smart net, I could download firmware for just about anything |
| 16:10.01 | Weezey | yeah, me too |
| 16:10.04 | wcselby | but when I logged in like 6 months ago, I was restricted to just a few things |
| 16:10.22 | Weezey | unforatunately I wasn't forward thinking enough to get 7970 firmware |
| 16:10.24 | slav3_kitten | time to email cdw |
| 16:10.36 | Weezey | calling my rep now. |
| 16:11.46 | slav3_kitten | thankfully when i got my 79xx phones i got firmware CDs with them, but i'd really like to get the latest image for my 2621xm |
| 16:12.01 | slav3_kitten | you know, on account of it being my router and all |
| 16:12.14 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:14.38 | wcselby | routers and switches I think have more expensive smartnet costs, but it's been a while since I've checked |
| 16:15.18 | slav3_kitten | dear chromium, why were you using 4 gb of saw and 1gb of ram.... |
| 16:15.21 | wcselby | speaking of cisco....have you seen the crap they're pulling with their home routers now? |
| 16:15.40 | slav3_kitten | you mean the linksys routers they slapped their name on? |
| 16:15.54 | dijib | whats up in here today boys? |
| 16:16.01 | dijib | what crap? |
| 16:16.18 | wcselby | yeah, although they've been cisco branded long enough now they're not really linksys anymore |
| 16:17.03 | slav3_kitten | so anyhow, what are they pulling? |
| 16:17.33 | wcselby | http://www.neowin.net/news/cisco-locks-users-out-of-their-routers-requires-invasive-cloud-service |
| 16:17.41 | wcselby | i was looking for a better article, I think there may be one on ars |
| 16:17.45 | *** join/#asterisk ctaloi (~ctaloi@50-56-202-179.static.cloud-ips.com) |
| 16:17.45 | drmessano | It's actually very funny |
| 16:18.05 | drmessano | One day you can update your router, the next day, locked out |
| 16:18.09 | drmessano | Sign up or DIE |
| 16:18.18 | wcselby | http://arstechnica.com/gadgets/2012/07/cisco-backpedals-after-uproar-drops-cloud-from-default-router-setting/ |
| 16:18.51 | drmessano | Its sad now how services are forced on consumers now |
| 16:18.52 | wcselby | looks like they realized they made a mistake and will have to wait a bit before they force it back on users :) |
| 16:19.08 | drmessano | 1. Sell a product. Market it as cheap and easy to set up |
| 16:19.13 | drmessano | 2. Wait 6 months |
| 16:19.27 | drmessano | 3. Force every lucky owner to sign up for some crap |
| 16:19.28 | wcselby | by the way, with the smartnet contract I linked earlier, I'm able to get to the 7970 sip download |
| 16:19.34 | drmessano | 4. ???? |
| 16:19.36 | drmessano | 5. Profit |
| 16:21.07 | wcselby | with cisco, 4. seemed to be "collect the browsing habits of all of our users, and then sell that info to the highest bidder(s) |
| 16:21.08 | wcselby | " |
| 16:21.26 | drmessano | That wasn't the motivation |
| 16:23.00 | slav3_kitten | speaking of forced services |
| 16:23.18 | drmessano | Cisco is really pushing cloud services because they are seriously behind. This extends to both the consumer and professional markets. They know that nobody is going to just go out and install their cloud services, so they are forcing it on the unwilling. The more users they have on the consumer side, the more they can push the professional side. Also, the upsell for more storage, etc |
| 16:23.24 | slav3_kitten | i love how google forced everyone into that social networking thing only to be sued to hell an back |
| 16:23.40 | *** join/#asterisk ctaloi (~ctaloi@50-56-202-179.static.cloud-ips.com) |
| 16:23.56 | slav3_kitten | google buzz |
| 16:24.18 | drmessano | Wasn't that like 20 years ago? |
| 16:24.36 | slav3_kitten | it was a while.. |
| 16:24.45 | slav3_kitten | i'm old :( |
| 16:24.57 | drmessano | Nobody complained when it was Google+.. Guess they got over it |
| 16:25.49 | slav3_kitten | google+ allowed you to sign up for it |
| 16:26.04 | slav3_kitten | not automatically posting all your personal information out there for everyone to see |
| 16:26.55 | wcselby | afk |
| 16:28.23 | drmessano | I think that's an exaggeration. Status updates and the gtalk buddy list. |
| 16:38.07 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-127.mobile.uci.edu) |
| 16:49.20 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 16:50.16 | JerJer | Anyone from Digium here? There is no tarball (or hash sigs) for asterisk-1.8.14.0-rc1 on downloads.asterisk.org |
| 16:51.11 | Qwell | huh |
| 16:51.23 | Qwell | JerJer: thanks - I'll see if I can get it going |
| 16:51.28 | JerJer | woot |
| 16:51.47 | Qwell | oh, it should be rc2 |
| 16:57.50 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:57.51 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:57.53 | Qwell | JerJer: It looks like the mirroring stopped mid-commit. I'm trying to poke somebody. You should be seeing rc2 tarballs there. |
| 17:00.17 | JerJer | kewl |
| 17:00.30 | Qwell | worst case scenario, you can always grab it from svn |
| 17:00.43 | Qwell | http://svn.asterisk.org/svn/asterisk/tags/1.8.14.0-rc2/ |
| 17:00.49 | JerJer | yeah - i was just in there grabbing a new version and noticed the anomaly |
| 17:00.56 | Qwell | figured so |
| 17:24.38 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:25.31 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:28.48 | *** join/#asterisk newtonr (~newtonr@173-25-121-44.client.mchsi.com) |
| 17:32.25 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:32.33 | dijib | does anybody know of a google docs alternative that can be run on a private server? |
| 17:33.47 | treborsux | I dont have forwarding of any kind I know of. She called back because they hung up becasue she just heard ringing. |
| 17:34.56 | treborsux | I have dhscp assign addresses when you say like behind nat because I have gateways on the phones? |
| 17:35.46 | treborsux | I have dhscp assign addresses when you say like behind nat because I have gateways on the phones? |
| 17:35.49 | *** part/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 17:37.58 | Katty | pants |
| 17:38.05 | Katty | lays on cold floor |
| 17:38.07 | Qwell | shirts |
| 17:38.25 | Katty | how hot is it out east Qwell |
| 17:38.43 | Katty | it's umm... |
| 17:38.45 | Qwell | <whalebeef> Huntsville International, Alabama. (45m 40s ago): 93.2°F / 34.0°C Partly Cloudy. Humidity: 44%. |
| 17:39.15 | Katty | oh you suck. |
| 17:39.21 | Katty | it's 102 with heat index of 107 here |
| 17:39.30 | Katty | 29% humidity, and the pressure is steady |
| 17:39.36 | Katty | so no signs of clouds coming in )= |
| 17:40.00 | Katty | shakes fist |
| 17:40.51 | Qwell | I'd prefer that over what happened on the way home yesterday. |
| 17:41.13 | Qwell | My car had a nice golf ball shower, while traffic was stopped. |
| 17:41.23 | Katty | oh dear :< |
| 17:41.28 | Qwell | and not like 1 here and there |
| 17:41.36 | Katty | we had hail a few days ago...but it wasn't /golfball/ sized. |
| 17:41.50 | Katty | did it do much damage? |
| 17:41.50 | Qwell | I was getting hit with several per second |
| 17:41.58 | Qwell | doesn't look like it.. I noticed one small dent, but I think that was already there. |
| 17:42.07 | Katty | phew |
| 17:42.29 | Qwell | That's what I get for leaving early. |
| 17:42.57 | Katty | oh don't say that. |
| 17:43.12 | slav3_kitten | Qwell, you part of the huntsville hackerspace? |
| 17:43.25 | Qwell | slav3_kitten: no, but a bunch of Digium folks are |
| 17:43.52 | slav3_kitten | yea that's what i've heard. i've been meaning to make my way down there one day |
| 17:44.23 | *** join/#asterisk ctaloi (~ctaloi@50-56-202-179.static.cloud-ips.com) |
| 17:44.43 | Qwell | slav3_kitten: they have open house on Tuesdays, and I think they have an event coming up soon here |
| 17:45.05 | Qwell | mmm, tomorrow, actually |
| 17:45.23 | Qwell | https://256.makerslocal.org/2012/06/15/retro-gaming-computing-night-2012/ |
| 17:45.40 | *** join/#asterisk nykac (~nykac@unaffiliated/nykac) |
| 17:45.46 | slav3_kitten | yea i've not got the money to drive down there right now, tis like a 4 or 6 hour drive |
| 17:47.49 | *** part/#asterisk nykac (~nykac@unaffiliated/nykac) |
| 17:54.45 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:58.16 | zoell | is there any way to get how many minutes a queue member in a call? |
| 17:58.23 | zoell | real time |
| 18:03.34 | [TK]D-Fender | zoell, AMI/CLI channel dump |
| 18:08.31 | zoell | thx gonna check that via AMI |
| 18:13.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:13.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:20.54 | *** join/#asterisk green91 (~green91@unaffiliated/green91) |
| 18:23.15 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 18:26.49 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:37.53 | *** join/#asterisk Ralcala (~operador@189.235.207.44) |
| 18:49.41 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:52.52 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 18:56.15 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 18:57.16 | Ralcala | Hi fellas |
| 18:57.30 | Ralcala | i really need help from you |
| 18:57.43 | Ralcala | i can registger to my sip provider |
| 18:58.04 | Ralcala | they said no need to password to register |
| 18:58.27 | Ralcala | software asterisk 1.8.11 |
| 19:00.22 | Ralcala | http://pastebin.com/zr5FcHds |
| 19:01.21 | Ralcala | this is part of my cli sip debug log |
| 19:02.18 | nny | odd, any reason why neither asterisk core sound or additional contains digit files (1,2,3,etc) on 1.8.11? |
| 19:02.39 | Qwell | nny: Because you're looking in the wrong place. |
| 19:02.44 | nny | Qwell: thanks |
| 19:03.11 | Qwell | Playback(digits/1) |
| 19:03.32 | nny | Qwell: gotcha, thanks! |
| 19:03.48 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 19:03.54 | Ralcala | hi fellas i can register to my sip provider |
| 19:04.07 | Ralcala | i mean i cant |
| 19:04.22 | Ralcala | can anyone here help me please? |
| 19:04.37 | Qwell | Ralcala: Why can't you? |
| 19:04.42 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:04.54 | [TK]D-Fender | Ralcala, You stopped the pastebin and we don't see any response. |
| 19:05.09 | Ralcala | let me check |
| 19:05.33 | Ralcala | no i dont stop |
| 19:05.39 | [TK]D-Fender | Ralcala, You also haven't described your server's working environment so we can't do more than wildly guess what else you may have done that is interfering with it. |
| 19:05.48 | Ralcala | ok |
| 19:06.06 | Ralcala | you got my cli sip debug log? |
| 19:06.24 | Ralcala | i mean you see on pastebin? |
| 19:06.37 | Ralcala | i gonna put my sip conf? |
| 19:07.28 | [TK]D-Fender | your PB had 1 packet going out. No answer. We don't know your firewalls, or your routing. If the IP passed is legit. nothing. |
| 19:07.44 | Qwell | Why is there an fe80 IPv6 address in here? O.o |
| 19:09.32 | Ralcala | ok |
| 19:09.36 | Ralcala | i got no fw |
| 19:09.55 | Ralcala | its a ethernet cable directly from the provider |
| 19:10.08 | Ralcala | cause they are my neighboors |
| 19:17.18 | Ralcala | this is my sip conf |
| 19:17.20 | Ralcala | http://pastebin.com/RQcgm4aL |
| 19:17.46 | Ralcala | need more? |
| 19:19.51 | Ralcala | [TK]D-Fender are you there pal? |
| 19:22.10 | Ralcala | hey QWell? |
| 19:22.12 | Ralcala | none |
| 19:23.36 | Qwell | I can't help you if you ignore my questions. |
| 19:23.51 | Ralcala | ok what you need pal? |
| 19:24.07 | Qwell | How is this failing? You didn't put that part in your log. |
| 19:24.35 | Ralcala | ok |
| 19:25.12 | Ralcala | falling? |
| 19:25.27 | Ralcala | i dont understand sorry can you explain me? |
| 19:28.20 | jpsharp | Isn't it failing with a 400 "Bad Request" error message? |
| 19:28.42 | Qwell | I don't know - I've not seen a log with that as a response to a REGISTER. |
| 19:32.30 | Ralcala | http://pastebin.com/MYcQJMWS |
| 19:32.35 | Ralcala | there you go |
| 19:32.43 | Ralcala | all my sip debug log |
| 19:32.57 | Ralcala | i hope you can help me |
| 19:33.42 | Ralcala | please remeber they dont need password for registration |
| 19:33.50 | Ralcala | just username |
| 19:33.52 | Ralcala | really odd |
| 19:34.20 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:34.38 | Qwell | Warning: 399 SoftX3000-MSG "Error ID:[259] LiteFME Tg register ,get Wll param error " |
| 19:34.40 | Qwell | wtf is that supposed to mean? |
| 19:35.30 | Ralcala | i dont know pal |
| 19:35.35 | Qwell | so ask your provider |
| 19:36.08 | Ralcala | they dont help me |
| 19:36.15 | Qwell | Then fire them and get a new one. |
| 19:36.36 | Ralcala | mmmm |
| 19:37.11 | Ralcala | you check my sip conf with register string? |
| 19:37.18 | Qwell | No. |
| 19:37.30 | Qwell | Your provider needs to tell you why there is an error. Asterisk is not at fauly. |
| 19:37.32 | Qwell | fault* |
| 19:38.14 | Ralcala | please check my registration string maybe syntax are wrong |
| 19:38.35 | Qwell | Nobody here can help you, until your provider says what the problem is. |
| 19:38.57 | Ralcala | ok i got another PBX working |
| 19:39.08 | Ralcala | with the same line but is no asterisk |
| 19:39.47 | Ralcala | you know chinese equipment with web gui |
| 19:40.42 | Ralcala | proxy only registration and just username with the number of phone |
| 19:40.59 | Ralcala | not big deal but its working |
| 19:41.26 | *** join/#asterisk twanny796 (~twanny@46.11.2.155) |
| 19:41.43 | Ralcala | can you point to step by step guide from another sip providers? |
| 19:41.59 | Qwell | ~book |
| 19:42.00 | 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 |
| 19:43.24 | Ralcala | mmmm |
| 19:43.42 | Ralcala | thanks anyway |
| 19:50.19 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:14.06 | *** join/#asterisk blindhog (~josh@adsl-75-62-197-176.dsl.rcsntx.sbcglobal.net) |
| 20:20.24 | nny | anyone have a suggestion for a good way to use System to check if a file exists? |
| 20:20.36 | nny | assuming that's the best method, that is |
| 20:21.16 | nny | would touch trigger the APPERROR return code if the file is missing? |
| 20:21.53 | Qwell | core show function SHELL |
| 20:22.36 | nny | what about STAT? |
| 20:22.39 | nny | http://www.voip-info.org/wiki/view/Asterisk+func+stat |
| 20:22.56 | Qwell | sure |
| 20:23.06 | nny | just learend about it heh, \o/ |
| 20:23.13 | nny | thanks though, shell seems useful too |
| 20:23.40 | [TK]D-Fender | checkout time, BBL |
| 20:30.37 | *** join/#asterisk catphish (~charlie@2001:9d8:2005:12::3) |
| 20:31.09 | catphish | does asterisk support any particular scripting languages that can be called from a dialplan? |
| 20:31.25 | Qwell | any |
| 20:31.31 | Qwell | ~agi |
| 20:31.31 | infobot | [~agi] AGI is the Asterisk Gateway Interface. Similar to CGI for web applications, AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), PERL (astperl?), and other languages. See also: https://wiki.asterisk.org/wiki/display/AST/Application_AGI or http://www.voip-info.org/wiki-Asterisk+AGI |
| 20:31.55 | catphish | thanks :) |
| 20:34.11 | catphish | is it simply the execution of a script (with an external interpreter) that returns specific data? |
| 20:34.39 | catphish | oh yeah, it can be any binary or script |
| 20:35.48 | catphish | i see there are also modules for php, perl, python that can avoid loading an external interpreter |
| 20:37.25 | catphish | AGI is awesome :) |
| 20:41.10 | *** join/#asterisk philfine (~tino@a95-92-251-229.cpe.netcabo.pt) |
| 20:41.48 | philfine | Hii everyone |
| 20:43.53 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 20:44.13 | cj | my extensions.conf looks like it was written by a fifth grader... |
| 20:45.30 | *** join/#asterisk citrusfizz (~IRC@c-50-137-218-206.hsd1.mn.comcast.net) |
| 20:46.47 | citrusfizz | possible to send fax to asterisk over SIP trunk or do i need an analog line for my asterisk box? |
| 20:48.22 | *** join/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 20:48.46 | treborsux | http://pastebin.com/8dUJ42jq is that right for the phone not to dial when picked up |
| 20:50.28 | catphish | citrusfizz: you can use fax over sip, the best way is if the remote end supports T38 |
| 20:51.30 | citrusfizz | catphish: i will have to ask my provider, i'm trying to implement something so a user can accept a fax on a certain DID and asterisk just emails it to them. |
| 20:52.24 | catphish | citrusfizz: it's definitely possible, we use it :) |
| 20:52.56 | catphish | check http://www.asterisk.org/docs/asterisk/1.6.2/applications/receivefax |
| 20:53.03 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
| 20:53.26 | catphish | i always ensure i have the spandsp library installed |
| 20:53.59 | vader-- | hola |
| 20:55.56 | nny | is there something like "record now" or "begin recording" that doesn't add temp or unavailable etc (or the word record by itself?) Trying to basically have "begin recording after the tone" or something to that effect but in Allison's voice. |
| 20:56.21 | catphish | i apologise if i'm misunderstanding the nature of channels here, but is it possible for asterisk to initiate a call (not to a local channel but for example to an extension at a remote sip server) and put the called party into a dialplan |
| 20:57.41 | Qwell | nny: http://store.digium.com/productview.php?category_id=8&product_code=8IVRPROMPT |
| 20:57.46 | catphish | actually i am being stupid |
| 20:58.04 | ChannelZ | Originate.... ? |
| 20:58.09 | catphish | originate accepts a complete channel which can include a number |
| 20:58.16 | nny | Qwell: thanks |
| 20:58.22 | catphish | yeah i can use originate with a sip channel and a number to dial |
| 20:59.06 | catphish | i've only ever used originate with a sip 'phone' and not specified an extension as part of the channel specificaion |
| 20:59.48 | vader-- | Any of you guys experiences with Polycom phones and digitmaps? I have two phone types, the Polycom 670 and the Polycom 335. I am trying to use feature codes on them but the 335 doesn't seem to like the feature code. When i enter *XX that works fine but when i enter *XXX, *XXXX, *XXXXX the Polycom 335 says "Enter more digits" on the screen after *XX |
| 20:59.52 | catphish | then again, i'm struggling to think of many legitimate uses for it now |
| 20:59.59 | Qwell | vader--: Ask your question in one place. |
| 21:06.16 | *** join/#asterisk Alex_Bkash (b4ea72b2@gateway/web/freenode/ip.180.234.114.178) |
| 21:06.33 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:09.28 | cj | hey, Qwell, can you recommend something to read to get familiar with the dialplan macros, best practices for building contexts, etc? |
| 21:19.53 | *** join/#asterisk KNERD (~KNERD@cpe-72-178-173-150.rgv.res.rr.com) |
| 21:24.47 | *** join/#asterisk TimeRider (~steve@027bde0c.bb.sky.com) |
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| 21:42.45 | Sean-Der | Using tcpdump I am able to see traffic on port 5060 that I make with netcat, but using Linphone I am not able to get any sort of activity or authentiaction |
| 21:43.10 | Sean-Der | I have 5060 and my RTP ports opened... is there something I am missing? |
| 21:43.29 | Sean-Der | Or maybe I am using my softphone wrong |
| 21:43.30 | KNERD | hyeah..missing asterisk logs of it trying to athenicate |
| 21:44.04 | *** join/#asterisk bmg505 (~leon@196-209-101-109.dynamic.isadsl.co.za) |
| 21:44.17 | Sean-Der | tcpdump isn't showing any activity :/ I can only set it off with netcat |
| 21:44.39 | KNERD | i wrote Asterisk.. |
| 21:45.24 | *** join/#asterisk Freeaqingme_ (~Freeaqing@91.214.168.110) |
| 21:45.48 | Sean-Der | KNERD: ok I don't mean any respect just trying to figure this out |
| 21:46.10 | sarsaeol | hopefully you dont mean any disrepect* :) |
| 21:46.33 | sarsaeol | disrespect* |
| 21:46.35 | Sean-Der | oops bad typo :( |
| 21:48.36 | KNERD | well let us help you |
| 21:48.44 | KNERD | you are showing us nothing.. |
| 21:49.36 | Sean-Der | I have the logging level at the highest verbosity, I have nothing to show though. |
| 21:50.02 | Sean-Der | The only thing I have is the fact that my softphone isn't making any activity in tcpdump |
| 21:50.17 | Sean-Der | which makes me think its a network level issue, but I really don't know |
| 21:51.25 | *** join/#asterisk navaismo (~navaismo@189.144.118.69) |
| 21:52.56 | navaismo | Hello, im testing the asterisk-cert last version, And im find that it does not write the CALLERID to the DB, i see in the debug this INSERT INTO cdr (`calldate`,`start`,`answer`,`end`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`uniqueid`) |
| 21:53.17 | navaismo | to add the clid value I need to edit the source code? |
| 21:59.20 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bzprrflmzunejyad) |
| 22:00.16 | KNERD | Sean-Der: what about the softephone log? |
| 22:02.37 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 22:09.11 | nny | http://pastebin.com/4SwuJ03h why does asterisk think this sound file doesn't exist when I can see it does? The perms are all asterisk/asterisk and asterisk created the file in the previous step |
| 22:11.20 | nny | hmm file /var/lib/asterisk/sounds/campaignrecordings/100/1/1.ulaw = /var/lib/asterisk/sounds/campaignrecordings/100/1/1.ulaw: data |
| 22:11.32 | nny | maybe i recorded it wrong via Record |
| 22:11.51 | nny | nm normal files say data too |
| 22:16.08 | navaismo | nevermind I have enable wrong alias y cdr_mysql.conf |
| 22:22.01 | WIMPy | cj: Macros will be deprecated. Try to use Gosub. |
| 22:23.01 | nny | WIMPy: any suggestion on why http://pastebin.com/4SwuJ03h happened? |
| 22:26.00 | WIMPy | you must not specify an extension on the filename. |
| 22:30.15 | *** part/#asterisk newtonr (~newtonr@173-25-121-44.client.mchsi.com) |
| 22:32.26 | nny | WIMPy: derp, thanks |
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| 22:35.18 | sarsaeol | can anyone tell me where a 'register string' for trunks would be placed in config files? |
| 22:38.11 | WIMPy | The good old register goes under [general], but you can use callbackextension in the peer definition as well. |
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| 22:51.54 | PaybackTony | K guys, serious question, don't kick me for it. What do you think is better for a high-volume use case (think call centers). Asterisk or FreeSwitch? I've only used asterisk before. |
| 22:52.44 | PaybackTony | Of course this would be for anyone with experience. |
| 22:53.18 | navaismo | define high-volume |
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| 22:57.25 | Jasnejac | PaybackTony: sip calls or pstn calls? |
| 22:57.56 | PaybackTony | ptsn |
| 22:58.08 | PaybackTony | 2m+ minutes a month |
| 22:58.30 | PaybackTony | for call centers that do inbound customer service and outbound (predictive dialing) sales |
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| 22:59.30 | Jasnejac | well, therefore my gut feeling would be to look at where they were designed from. asterisk has a pstn background, freeswitch a sip bacjground. asteriks is an evolution, freeswitch a pure software inplmentation |
| 22:59.51 | Jasnejac | with better typing. ahem |
| 23:00.00 | *** join/#asterisk sonstwo (~garland@unaffiliated/ffs) |
| 23:00.13 | WIMPy | Where do you see a PSTN background in Asterisk? |
| 23:00.30 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 23:00.35 | WIMPy | That's exactely what I've always been missing. |
| 23:00.35 | Jasnejac | digium is a pstn interface company |
| 23:01.13 | WIMPy | Yes, interfaces do exist. |
| 23:01.45 | Jasnejac | I'm a software engineer that sort of fell into telecoms by accident. It's different. Very different. Freeswitch is the better design IMHO but asterisk has the track record |
| 23:03.01 | WIMPy | When I first came across Asterisk about 10 years ago, there was no way to connect it to the PSTN. |
| 23:03.56 | Jasnejac | I don't think that's really the point. Asterisk is driven by a company that lives for the pston interfaces; it's their bread and butter or has been |
| 23:04.00 | jpsharp | That's about when I came across it and Mark was using the Tormenta 1 ISA card plus the Winmodems as FXO ports. |
| 23:04.38 | Jasnejac | sangoma support freeswitch of course |
| 23:04.45 | jpsharp | And some other card (I can't remember the name) for FXS ports. |
| 23:05.04 | WIMPy | As I said: The hardware exists. The support has been growing very slowly. |
| 23:06.56 | jpsharp | I'm glad it has evolved past the Tormenta 1. That card was a bear. |
| 23:08.37 | Jasnejac | a slightly different question - does anyone actually use ael or lua? I use ael but I don't really understand why the lua interface is there and there has to be a reason |
| 23:09.00 | WIMPy | Hardware has never been an issue/ There have always been lots of companies doing interfaces. |
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