| 00:05.03 | godmachine-x6 | == Spawn extension (phones, 501, 1) exited non-zero on 'SIP/laptop-0000006a' |
| 00:05.09 | godmachine-x6 | every time at 36 seconds |
| 00:05.40 | godmachine-x6 | but only when the call is from A to B and never when B to A sip.conf users reflect same settings for all the users, so i doubt the problem is in the sip.conf |
| 00:14.05 | *** join/#asterisk jasoncarter (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 00:14.17 | godmachine-x6 | -- (10 headers 0 lines) --- |
| 00:14.17 | godmachine-x6 | Sending to 127.0.0.1:5061 (no NAT) |
| 00:14.18 | godmachine-x6 | Scheduling destruction of SIP dialog '78c2eabb7511aecf318636576c3a34f6@127.0.0.1:5060' in 32000 ms (Method: BYE) |
| 00:14.31 | godmachine-x6 | thats the problem whatever that means |
| 00:14.53 | godmachine-x6 | because in 3 seconds after that message comes through the call gets dropped |
| 00:18.42 | jasoncarter | If MyServer1 has a Trunk connected to SipServer1 and the DID for SipServer1 receives a call, is SipServer1 creating a connection to MyServer1 or does SipServer1 use the already established connection by MyServer1? My goal is to drop all traffic to port 5060 from outside of this network by iptables |
| 00:19.32 | jasoncarter | But I'm worried if I do, calls won't come through. I don't have a development server to work with |
| 00:30.11 | jeremy_g | godmachine-x6:u didnt send any traces. blink is one of my fav. clients, |
| 00:30.24 | p3nguin | I doubt there's a problem with blink. |
| 00:30.56 | p3nguin | The problem happens when calling from blink TO a softphone ON THE ASTERISK SYSTEM. |
| 00:31.04 | p3nguin | I believe that's where the problem lies. |
| 00:31.11 | jeremy_g | jasoncarter:what do you mean when you say m1 has a trunk connected to s1 in terms of asterisk config. |
| 00:31.46 | jasoncarter | Sorry. m1 has Asterisk, s1 is e.g. sipgate |
| 00:32.14 | jeremy_g | jasoncarter:how is trunk configured in asterisk |
| 00:34.59 | jeremy_g | jasoncarter:have u defined sipgate as the peer in asterisk, if so then sipgate on receiving call on its did wills end to asterisk which will hit the dialplan somewhere. the communciation would be on port 5060 depending on how u configured. |
| 00:34.59 | jeremy_g | jasoncarter:dont forget to exclude the rtp ports fromt he firewall. see /etc/asterisk/rtp.conf |
| 00:34.59 | jeremy_g | jasoncarter:i gotta sleep now. |
| 00:34.59 | jeremy_g | bye |
| 00:39.11 | *** join/#asterisk jasoncarter (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 00:40.27 | jasoncarter | jeremy_g thank you man. i couldn't find any articles on this. I blocked 5060 UDP port and no calls came through and then I read what you wrote. I'm going to get settings for a trunk 2 ticks please |
| 00:46.18 | Kobaz | do de do |
| 00:55.25 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:55.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.4 (2011/06/28), 1.6.2.19 (2011/06/29), 1.4.42 (2011/06/29), *-Addons 1.6.2.4, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 01:13.09 | *** join/#asterisk iulius (~iulius@adsl-74-160-133-160.asm.bellsouth.net) |
| 01:14.00 | *** part/#asterisk iulius (~iulius@adsl-74-160-133-160.asm.bellsouth.net) |
| 01:17.48 | *** join/#asterisk dfamorato (~dfamorato@c-75-74-235-36.hsd1.fl.comcast.net) |
| 01:28.09 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 02:04.44 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 02:46.16 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 02:46.29 | Bipul | is there any one |
| 02:46.30 | Bipul | ? |
| 02:49.29 | p3nguin | ~ask |
| 02:49.29 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:51.36 | Bipul | p3nguin, i have problem in configuring :( |
| 02:52.00 | p3nguin | And I have a problem with people not asking anything. |
| 02:52.14 | Bipul | p3nguin, i will ask :| |
| 02:54.56 | Bipul | You have any Guidelines for confguring |
| 02:55.01 | Bipul | for Beginners |
| 02:55.08 | p3nguin | ~book |
| 02:55.08 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 03:01.06 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 03:02.05 | *** join/#asterisk coppice (~coppice@m121-202-47-88.smartone-vodafone.com) |
| 03:04.25 | tyman | Does anyone know the polycom xml parameter for showing a single line across two buttons? |
| 03:04.48 | tyman | I've done this in the past and I can't seem to find it again? |
| 03:04.55 | tyman | s/?/./ |
| 03:11.11 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 03:13.39 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 03:15.45 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
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| 04:13.36 | pa | hello: i have a problem |
| 04:13.45 | pa | i tried to upgrade my machine to ubuntu natty |
| 04:13.56 | pa | now when i try to load asterisk, it stops here: [Jul 11 06:13:09] NOTICE[3454]: loader.c:1064 load_modules: 198 modules will be loaded. |
| 04:14.06 | pa | and the cpu stays at 100% |
| 04:20.25 | ChannelZ | did you build * from source? |
| 04:24.17 | pa | no, i used the packages |
| 04:24.37 | pa | now i tried to reinstall it, and remove my old asterisk configuration, and it starts properly |
| 04:25.00 | pa | so i assume there is something wrong in my modules.conf/extensions.conf/zapata.conf |
| 04:39.35 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:03.23 | *** join/#asterisk Jeffy_ (~Jeffy@c211-30-56-77.frank3.vic.optusnet.com.au) |
| 05:03.41 | Jeffy_ | Hello. I'm hoping you can help me, as your guide didn't catch this one (I know it cant catch all errors) |
| 05:03.52 | Jeffy_ | lock.o: In function `__ast_cond_wait': |
| 05:03.53 | Jeffy_ | /usr/local/src/asterisk-1.8.4.4/utils/lock.c:558: undefined reference to `ast_bt_get_addresses' |
| 05:04.47 | *** join/#asterisk Syrex (~syrex@dsl-146-6-57.telkomadsl.co.za) |
| 05:24.27 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 05:25.46 | pa | i think i understand what i have to do.. |
| 05:25.53 | pa | i have to upgrade from zaptel to dahdi |
| 05:25.54 | pa | sigh |
| 05:28.10 | ChannelZ | yeah.. that transition happened like 2 years ago |
| 05:28.26 | ChannelZ | What on earth did you upgrade your Ubtuntu/Asterisk versions *from*? |
| 05:28.46 | *** join/#asterisk Ehsanfzali (~chatzilla@84.47.229.157) |
| 05:30.15 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 05:30.43 | Ehsanfzali | Hi, i'm new to asterisk can anyone help me to install a TE121 and config it using asteriskNow with asterisk 1.8 and asterisk gui? |
| 05:31.16 | joako | Ehsanfzali, How are you using Asterisk GUI on 1.8? |
| 05:31.45 | Ehsanfzali | when I search the internet they all talking about zaptel but I couln't find it in my asterisknow should I install it or its not required in 1.8? |
| 05:32.22 | Ehsanfzali | I just burned the latest ISO and selected asterisk 1.8 with asterisk gui in boot menu |
| 05:33.08 | Ehsanfzali | When the system boots it says AsterisksNOW 1.7.1 |
| 05:36.12 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 05:38.54 | Jeffy_ | Can anyone help with my question? |
| 05:38.57 | Jeffy_ | lock.o: In function `__ast_cond_wait': |
| 05:38.59 | Jeffy_ | /usr/local/src/asterisk-1.8.4.4/utils/lock.c:558: undefined reference to `ast_bt_get_addresses' |
| 05:39.04 | Jeffy_ | when trying to "make" asterisk |
| 05:39.57 | irroot | Jeffy you got a clean build ?? run configure / make menuconfig |
| 05:40.11 | Jeffy_ | yeah |
| 05:40.14 | Jeffy_ | did both of those irroot |
| 05:40.43 | Jeffy_ | but i'll try a distclean just to be safe again. |
| 05:41.00 | *** join/#asterisk jkroon (~jkroon@41.49.11.128) |
| 05:41.40 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:42.27 | *** join/#asterisk nickmarino (~anonymous@97.104.216.76) |
| 05:42.36 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:44.04 | Ehsanfzali | joako is it OK? |
| 05:44.44 | nickmarino | Hi there...don't know if theres anyone out there who can help me, but I'm trying to set up a digium T1 card and need some guidance. |
| 05:45.19 | vikapi | nickmarino, good.. |
| 05:46.42 | nickmarino | In a nutshell, I have Libpri and Dahdi complete installed and asterisk compiled, but not sure what to do next. The manual on digium's website does not go any further than that |
| 05:48.11 | vikapi | nickmarino, hope u ve maintaind the order of installtion of dahdi first,then libpri, then asterisk.. |
| 05:48.18 | irroot | nickmarino look at the sample config in /etc/dahdi |
| 05:49.10 | nickmarino | Hmm, I think I installed libpri -> dahdi -> asterisk, thats what was in the t1 card manual. |
| 05:50.17 | nickmarino | I have system.conf in that folder, should there be more? |
| 05:50.35 | irroot | perfect nickmarino need to configure it |
| 05:51.03 | irroot | read that file |
| 05:51.14 | vikapi | nickmarino, `dahdi_gencfg modules` will auto configure the /etc/dahdi/system.conf file i guess.. |
| 05:51.17 | irroot | for guidelines |
| 05:51.22 | joako | Ehsanfzali, Sure. But zaptel name was changed to: DAHDI |
| 05:51.32 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 05:52.08 | joako | Ehsanfzali, And I asked about the GUI because the official Digium Asterisk GUI was supported in Asterisk 1.4, it could work with Asterisk 1.6, but certainly not with 1.8. |
| 05:52.27 | Ehsanfzali | uh OK and the commands also has changed? for example zttool? is it dahdi_tool? |
| 05:52.40 | joako | Ehsanfzali, To start off I don't recommend the GUI. Learn the system first by setting up everything manually. Later if you want to use a GUI, sure try it out. |
| 05:53.18 | joako | Ehsanfzali, Correct. If you are looking at voip-info.org site, be aware that much of the information is outdated, especially if you are using Asterisk 1.8 |
| 05:53.18 | nickmarino | vikapi: when I try to run that command it says command not found, did I miss a step during install? |
| 05:54.17 | Ehsanfzali | how can I check the my box version? |
| 05:54.51 | vikapi | nickmarino, oops my bad..it was just `dahdi_genconf` |
| 05:55.23 | Ehsanfzali | core show version : Asterisk 1.6.2.11 built by root @ localhost.localdomain on a x86_64 running Linux on 2010-08-24 20:45:59 UTC |
| 05:55.37 | Ehsanfzali | you are right sorry it's 1.6 not 1.8 |
| 05:55.41 | joako | Ehsanfzali, asterisk version: rasterisk -x "core show version" |
| 05:56.05 | nickmarino | vikapi: Thanks, that ran. Should there be any output or will it just write a new conf? |
| 05:56.20 | joako | Ehsanfzali, and you have the GUI that looks like this? http://www.asteriskguru.com/tutorials/asterisk_gui_image274491.jpg |
| 05:56.37 | Ehsanfzali | yes |
| 05:57.00 | joako | Ehsanfzali, In theory you can just login and it should detect the card and set it up |
| 05:57.33 | Ehsanfzali | yes joako, it does but on my PBX the sync-err light is red |
| 05:57.33 | joako | But realisticly I always encounter a small bug or two and I need to edit the GUI files for it to work properly, but that would be sort of hard for someone just getting started with asterisk to figure it out |
| 05:57.54 | Ehsanfzali | and I recieve this warning in CLI WARNING[6031]: chan_dahdi.c:4160 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 05:57.57 | joako | Did you set one for master timing and one for slave? |
| 05:58.21 | joako | Did you setup the PRI for PRI... or channelized T1? Some PBX don't even support PRI |
| 05:59.11 | vikapi | nickmarino, it won't give any output. jus see if /etc/dahdi/system.conf file is edited and then run `lsdahdi`, `dahdi_cfg -vvvv`, `dahdi_scan`and pastebin output.. |
| 05:59.53 | Ehsanfzali | my PBX is KX-TDA100 and I installed kx-TDA0290 E1 PRI30 card on it |
| 06:01.03 | Ehsanfzali | on PBX CRC4 is disabled, Port Type is QSIG-Master and Network Type is Euro ISDN |
| 06:01.23 | nickmarino | vikapi, everything looks good there, no need to pastebin. How can I check it is being recognized within asterisk though? |
| 06:02.08 | Ehsanfzali | of course I'm not sure about this settings should I use QSIG-Master or Slave on my PBX? |
| 06:02.10 | vikapi | nickmarino, open asterisk console, then give the CLI commmand, `dahdi show channels`..wat output does it give. |
| 06:02.31 | joako | Ehsanfzali, You need to make sure all your settings match, except that in asterisk you will use: timing: slave & pri_cpe |
| 06:02.56 | nickmarino | vikapi, No such command 'dahdi show channels' |
| 06:02.57 | joako | You could also reverse it and set asterisk to be pri_net & timing master: but then you need to reverse it in the PBX |
| 06:05.43 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:06.18 | vikapi | nickmarino, u need to configure chan_dahdi.conf in /etc/asterisk then restart asterisk and try tat command.. |
| 06:11.44 | nickmarino | vikapi: there was no chan_dahdi.conf but there was a chan_dahdi.conf.template and a dahdi-channels.conf |
| 06:12.25 | nickmarino | vikapi: I took the .template off and included dahdi-channels.conf and restarted asterisk but still the same problem |
| 06:14.01 | vikapi | nickmarino, it should be, dahdi_channels.conf..did u do make samples after asterisk compilation..?? |
| 06:14.29 | *** join/#asterisk Ehsanfzali (~chatzilla@84.47.229.157) |
| 06:15.02 | nickmarino | no, I did not make samples |
| 06:15.53 | nickmarino | vikapi: here is a pastebin of my configs http://pastebin.com/wJmK2Yyg |
| 06:15.56 | Ehsanfzali | joako sorry I had network problem |
| 06:15.57 | Ehsanfzali | joako, to set timing to slave Sync/Clock Source should be 0 or 1? |
| 06:17.02 | vikapi | nickmarino, ok..uncomment the 22nd line in your post.. |
| 06:17.10 | nickmarino | ehsanfzali it looks like you're in the same boat as me. Asterisk seems pretty easy to learn but this Dadhi stuff seems poorly documented |
| 06:17.52 | joako | Ehsanfzali, I'm not sure... it should be well documented. In /etc/dahdi/system.conf I believe |
| 06:18.13 | Ehsanfzali | yes nickmarino |
| 06:18.27 | joako | span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>,yellow |
| 06:18.57 | joako | http://www.voip-info.org/wiki/view/system.conf The default file will document all the possible options |
| 06:19.58 | Ehsanfzali | joako when I have change the dahdi/system.conf what command should I run to apply configuration? |
| 06:20.17 | Ehsanfzali | in dahdi restart in CLI is enough? |
| 06:20.53 | joako | Try: dahdi_cfg -vvvv |
| 06:21.00 | WIMPy | No. You Need to start dahdi_cfg. |
| 06:21.12 | WIMPy | Maybe you need to dahdi_cfg -s first. |
| 06:21.14 | joako | I think you need to do that when asterisk is not running... dahdi is the driver, and asterisk talks to dahdi when it's setup |
| 06:22.20 | *** join/#asterisk hrolf (~hrolf@static-host202-61-49-9.link.net.pk) |
| 06:23.31 | nickmarino | vikapi: uncommented and restarted, still says command not found |
| 06:23.55 | nickmarino | tried some of the suggestions from WIMPy and jaoko too |
| 06:24.27 | WIMPy | What command? |
| 06:24.56 | irroot | the busy it has got me |
| 06:25.19 | WIMPy | Hi irroot |
| 06:25.25 | joako | Anyone knows how to format a polycom phone without booting it? |
| 06:25.31 | irroot | lo there wimpy .... |
| 06:26.00 | WIMPy | joako: Use a hammer or an oven? That should give it a new format. |
| 06:27.05 | *** join/#asterisk Leddy (leddy@emailing8-cardata.com) |
| 06:27.31 | joako | WIMPy, I replaced 1 phone with another my mv macaddress-phone.cfg to macaddress-phone.cfg. So I want to format the old one to make sure it doesn't use its config and try to connect to the server again |
| 06:28.34 | WIMPy | There must be a "reset to factory defaults" entry in the menu for sure? |
| 06:28.45 | Ehsanfzali | joako I have set my Panasonic PBX to QSIG-Slave and configured system.conf the way you said but I still have the same problem .... what about framing/coding ccs/hdb3 is OK? |
| 06:29.00 | joako | WIMPy, Sure is. But I need to plug it into the network and let it fully boot to format the phone. |
| 06:29.42 | joako | Ehsanfzali, Everything else must match between the two. And At this point you should be also looking at your chan_dahdi.conf to make sure your channels and d-channel match |
| 06:31.52 | *** join/#asterisk gravin (~gravin@175.139.239.3) |
| 06:33.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 06:33.38 | Ehsanfzali | I have checked chan_dahdi.conf almost everything is commented in it! |
| 06:34.22 | Ehsanfzali | how should I config it? |
| 06:35.12 | WIMPy | You need at least 'signalling' and 'switchtype' before 'cahnnels'. |
| 06:36.41 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:36.45 | schmidts | good morning |
| 06:36.53 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-xajytagiilqwmyvp) |
| 06:37.36 | irroot | morning schmidts |
| 06:40.18 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 06:40.31 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:42.02 | *** join/#asterisk Ehsanfzali (~chatzilla@84.47.229.157) |
| 06:42.53 | Ehsanfzali | joako any guide to config chand_dahdi.conf? |
| 06:44.57 | tzafrir_laptop | Ehsanfzali, start with running dahdi_genconf and then: echo '#include dahdi-channels' >>/etc/asterisk/chan_dahdi.conf |
| 06:45.02 | ChannelZ | dahdi_genconf should help |
| 06:45.13 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:46.01 | joako | Ehsanfzali, not trying to ignore you... I need to get going, sorry. |
| 06:47.16 | Ehsanfzali | no problem joako, thanks a lot for you help |
| 06:52.36 | Ehsanfzali | tzafrir_laptp the file created name is dahdi-channels.conf .... should I use echo '#include dahdi-channels.conf' >>/etc/asterisk/chan_dahdi.conf instead or it appned .conf itself?! |
| 06:53.50 | WIMPy | That's just a question of your personal taste. |
| 07:00.03 | *** join/#asterisk tyman_ (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 07:09.19 | nickmarino | vikapi: don't know if you're still there, but i think I got it working now, thanks |
| 07:10.09 | nickmarino | vikapi: I realized I made an error compiling dahdi so I recompiled that and asterisk and now dahdi show channels shows 23 channels in service |
| 07:10.40 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:12.07 | Jeffy_ | thank you :) it's compiling |
| 07:12.09 | Jeffy_ | good night all |
| 07:14.47 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 07:17.29 | Ehsanfzali | anyone know what else should I check? |
| 07:17.47 | Ehsanfzali | I still receive this warning WARNING[4125]: chan_dahdi.c:4160 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 07:19.17 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 07:20.12 | Ehsanfzali | dahdi-channels.conf: http://pastebin.com/Ak6HnN1V & dahdi/system.conf :http://pastebin.com/FtjcSkN9 |
| 07:22.13 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 07:28.57 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 07:29.16 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:30.14 | kaldemar | Ehsanfzali: if your panasonic PBX is configured to use QSIG, asterisk must too. and your asterisk is configured as CPE, which means that the panasonic PBX would have to be the network side, which PAPBX's usually are not. |
| 07:30.19 | *** join/#asterisk tyman (~tyler@173-167-242-146-ip-static.hfc.comcastbusiness.net) |
| 07:31.17 | kaldemar | Ehsanfzali: so, to use qsig, you need switchtype=qsig instead of switchtype=euroisdn. |
| 07:33.05 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 07:33.37 | tzafrir_laptop | kaldemar, but then again, qsig and euroisdn are similar enough at layer2, right? |
| 07:33.49 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:34.06 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 07:34.09 | tzafrir_laptop | So if he gets "no D-channels available", my guess is that this is not the problem |
| 07:34.13 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:34.26 | tzafrir_laptop | Ehsanfzali, what's the output of lsdahdi ? |
| 07:34.26 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 07:34.31 | E-bola | Helloe all |
| 07:35.15 | E-bola | I'm looking for a cdr/statistics module that will let me show users call statistics via a webpage but divided by context's. So a login is limited to only seeing stats from his own context |
| 07:35.28 | E-bola | Have anybody ever seen or heard of such a thing? |
| 07:35.51 | tzafrir_laptop | set some field in the CDR in the context? |
| 07:36.22 | Ehsanfzali | its http://pastebin.com/pc2bzFZG |
| 07:36.22 | tzafrir_laptop | or to ${CONTEXT} |
| 07:36.40 | E-bola | It should be easy to differentiate tzafrir_laptop, but i need a webpart that will let me do authentication/presentation based on it |
| 07:36.52 | tzafrir_laptop | Ehsanfzali, it's RED. It's in a red alarm |
| 07:37.14 | tzafrir_laptop | It basically means that you don't have basic (layer 1) connectivity to the other side |
| 07:37.19 | tzafrir_laptop | No bits are flowwing |
| 07:37.22 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:37.26 | tzafrir_laptop | Make that RED go away first |
| 07:37.44 | E-bola | Ahh cdr-stats apears to have gotten ACL functionality in latest version, ile start by checking that out |
| 07:37.57 | tzafrir_laptop | For starters: are both sides E1? |
| 07:38.12 | Ehsanfzali | tzafrir_laptop so what should I do? what could be the problem? |
| 07:38.32 | Ehsanfzali | yes they are both E1 |
| 07:38.53 | tzafrir_laptop | Is the remote side configured as "network"? |
| 07:39.05 | tzafrir_laptop | Do you use a crossed E1 cable between them? |
| 07:39.20 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 07:40.59 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:41.28 | Ehsanfzali | remote side is Panasonic KX-TDA100 and I can configure the Port Type to one of the followings CO/Extension/QSIG-Master/QSIG-Slave, its now QSIG-slave |
| 07:41.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 07:42.07 | Ehsanfzali | no cable is not cross its straight Cat5 network cable |
| 07:43.41 | Ehsanfzali | and thios is output of dahdi_scan http://pastebin.com/u3K0CRhc |
| 07:44.19 | Ehsanfzali | tzafrir_laptop do I using wrong cable?! |
| 07:47.11 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 07:47.12 | tzafrir_laptop | Ehsanfzali, I suppose you need a crossed (E1/T1, not Ethernet!) cable |
| 07:47.20 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:88dd:acee:be20:8102) |
| 07:47.57 | *** part/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:48.15 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:48.54 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 07:55.57 | Ehsanfzali | oh does the cable or sockets diffrent comlpetley? |
| 07:56.33 | Ehsanfzali | 'cause I have connected my card with it's tester with this cable and the lights goes green |
| 07:57.00 | Ehsanfzali | do I need a cable like http://www.chebucto.ns.ca/Chebucto/Technical/Manuals/Max/max6000/gs/cables.htm#21066? |
| 07:58.29 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:00.34 | tzafrir_laptop | Ehsanfzali, the "T1/PRI crossover cable" from there, yes |
| 08:03.07 | Ehsanfzali | could I create one with cat5 cable and RJ45 socket? or I need to Use RJ48 and different cable type? |
| 08:03.45 | Ehsanfzali | what is the different of rj45 and rj48? |
| 08:06.47 | Ehsanfzali | tzafrir_laptop sorry for my irrelevant questions |
| 08:07.17 | tzafrir_laptop | Ehsanfzali, that's RJ45, yes |
| 08:07.40 | tzafrir_laptop | That's what your card has, right? |
| 08:07.51 | Ehsanfzali | yes |
| 08:07.57 | tzafrir_laptop | is not well familiar with cabling |
| 08:12.19 | coppice | rj45 vs rj48 == same physical plug. different pin usage |
| 08:20.18 | *** join/#asterisk StaRetji (~BigAll@80.93.240.172) |
| 08:20.40 | StaRetji | Hello there good people, so good always willing to help :P |
| 08:22.55 | E-bola | Quick question: Do anybody know whats causing this warning upon starting asterisk 1.6.2.9: utils.c: trying to reset empty pool |
| 08:29.16 | StaRetji | I have 5678 sip account and caller id, I call some DID number which is redirected to my mobile phone. So, naturally, I call from 5678 that DID in goes trough but once when asterisk has to redirect to my mobile it fails due to authentication problem caused by 33 country prefix added to my caller id. It is 335678, this prefix is added by sip provider I guess. Any ideas to force sip provider to send clean caller id 5678? Thx |
| 08:31.51 | StaRetji | can I do something to remove 33 from caller id? |
| 08:33.16 | Ehsanfzali | tzafrir_laptop I have changed the cable but nothing change its' still RED |
| 08:33.22 | Ehsanfzali | what else should I check? |
| 08:36.58 | kaldemar | StaRetji: caller id should not cause authentication issues. how about showing a CLI output of a call? |
| 08:37.43 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 08:38.55 | StaRetji | hi kaldemar, I will post it on pastebin in a minute, thx |
| 08:42.04 | *** join/#asterisk Dovid (~Dovid@ool-4355c88f.dyn.optonline.net) |
| 08:43.33 | StaRetji | kaldemar: here it is http://pastebin.com/TBmjA2aa |
| 08:44.24 | StaRetji | you can see that first call is okay, caller id is 5551 |
| 08:44.29 | StaRetji | and second call is 335551 |
| 08:49.53 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 08:56.23 | Ehsanfzali | tzafrir_laptop, sorry I used wrong cable its OK, problem solved thanks alot |
| 08:58.14 | kaldemar | StaRetji: you're the one who controls what goes out of your asterisk box. your paste is somewhat unclear and an agi script won't make it easier to interpret. something in your box seems to add the 33 however, find it and remove it. |
| 08:58.51 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 09:05.16 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 09:06.21 | StaRetji | kaldemar: ok, thx, it's appreciated |
| 09:08.05 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 09:09.18 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:10.05 | puzzled | morning |
| 09:10.17 | Bipul | http://pastebin.com/myZZcHDp why i am getting error |
| 09:12.07 | Gugge | Unable to open pid file '/var/run/asterisk/asterisk.pid': Permission denied |
| 09:16.19 | Bipul | oki i have to be a root user |
| 09:16.39 | Bipul | No such command 'clear' (type 'core show help clear' for other possible commands) |
| 09:16.40 | Bipul | bipul-desktop*CLI> sip show |
| 09:16.40 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 09:17.01 | Bipul | No such command 'clear' (type 'core show help clear' for other possible commands) ? why it says this |
| 09:19.20 | Gugge | no idea |
| 09:20.14 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 09:29.37 | Faustov | when I run "asterisk -x command" I get Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. Is there any way I could get commands passed to asterisk despite the service running? |
| 09:32.53 | *** join/#asterisk SunTsu_ (miyamoto@unaffiliated/suntsu) |
| 09:33.05 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 09:33.53 | *** join/#asterisk heffer_ (~felix@fedora/heffer) |
| 09:36.02 | puzzled | Faustov: if you look at the output of asterisk -h you will notice it says: "-x <cmd> Execute command <cmd> (only valid with -r)" |
| 09:36.47 | Faustov | puzzled: weird, I was trying with -xR... |
| 09:37.03 | puzzled | Faustov: how about try with asterisk -r -x ... |
| 09:37.09 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 09:37.17 | Faustov | puzzled: that worked |
| 09:37.28 | Faustov | puzzled: but not completely |
| 09:37.35 | Faustov | w8 |
| 09:38.54 | Faustov | puzzled: any idea why it replies "no such command channel" for asterisk -rx channel originate... |
| 09:39.24 | *** join/#asterisk sonstwo (~garland@unaffiliated/ffs) |
| 09:39.28 | puzzled | no time. google and read the manual/wiki |
| 09:40.41 | *** join/#asterisk Bidik (~bidik@li267-109.members.linode.com) |
| 09:41.02 | *** join/#asterisk wonderworld (~ww@port-92-201-126-230.dynamic.qsc.de) |
| 09:44.10 | *** join/#asterisk Sylnai (~andi@andimiller.net) |
| 09:44.53 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 09:44.53 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 09:45.51 | *** join/#asterisk jks (jks@193.189.93.254) |
| 09:46.23 | Faustov | ok sorted, I needed more quotes |
| 09:46.38 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 09:47.09 | *** join/#asterisk coppice (~coppice@m121-202-47-88.smartone-vodafone.com) |
| 09:47.27 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 09:48.13 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:48.15 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:48.40 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 09:48.48 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 09:49.21 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 09:49.57 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 09:56.55 | *** join/#asterisk SunTsu_ (miyamoto@unaffiliated/suntsu) |
| 09:57.03 | *** join/#asterisk jeremy_g (~jeremey_g@gw-sthlm01.rebtel.com) |
| 10:04.42 | *** join/#asterisk Sylnai (~andi@andimiller.net) |
| 10:28.16 | Bipul | any one some free softphones for linux user |
| 10:30.24 | irroot | blink |
| 10:41.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:45.41 | aberrios | Faustov, asterisk -rx 'command' |
| 10:46.37 | aberrios | Bipul, Twinkle |
| 10:47.05 | Bipul | ok |
| 10:48.40 | Faustov | aberrios: indeed, quotations needed & need to be escaped if it's passed further... |
| 10:48.45 | Faustov | thanks |
| 10:49.27 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-131-153.twcny.res.rr.com) |
| 10:52.31 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 10:56.36 | *** join/#asterisk ik_5 (~ik@194.90.222.218) |
| 10:56.40 | ik_5 | hello |
| 10:57.02 | ik_5 | how can I get the sip response code itself ? |
| 10:58.32 | *** join/#asterisk dobby156 (~joe@79.135.102.10) |
| 10:58.54 | dobby156 | How do you find out when a sip peer last registered? Thanks |
| 11:01.36 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 11:05.08 | ik_5 | dobby156, sip show peers |
| 11:06.22 | dobby156 | jks: that show the currently registerd peers |
| 11:07.03 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-219.telkomadsl.co.za) |
| 11:07.11 | dobby156 | I wish to know if possible, when a Peer (which maybe unregistered) last registered |
| 11:07.15 | dobby156 | thanks |
| 11:07.16 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 11:09.23 | E-bola | cdr_addon_mysql.c:537 my_load_module: Unable to query table description!! Logging disabled |
| 11:09.32 | E-bola | this seems to be triggered by having a tablename with a "-" in it |
| 11:09.40 | E-bola | Do anybody know if thats a known bug? |
| 11:11.17 | E-bola | Hmm using ` instaid of " around the tablename did the trick |
| 11:11.50 | jkroon | hi guys - how does one go about tracking a memory corruption bug in asterisk? |
| 11:12.15 | jkroon | I've tracked a structure where the string "parkedcalls" gets written into the memory, even though the memory is allocated to a different object... |
| 11:14.19 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 11:15.03 | jacc0 | hi all |
| 11:20.19 | jkroon | E-bola, yes, - usually interprets as a - sign, so any table tables with - in them has to be between `` |
| 11:23.05 | *** join/#asterisk StaRetji (~BigAll@80.93.240.172) |
| 11:23.09 | StaRetji | folks, I would appreciate help with setting extension for recording sounds. Standard script with extension 205 seems not to work on Asterisk 1.6 while on 1.4 works fine. |
| 11:23.35 | StaRetji | I googled and found Example 4 (by Leif Madsen using Asterisk 1.6 Syntax) |
| 11:24.00 | StaRetji | but it wont work by simply copy paste off course |
| 11:24.18 | StaRetji | I can't get to understand where to input extension number, such as 205 |
| 11:24.31 | StaRetji | thx in advance |
| 11:25.24 | *** join/#asterisk gravin (~gravin@175.139.239.3) |
| 11:30.29 | E-bola | jkroon: i just always asumed i could use " as well, guess not hehe |
| 11:30.44 | E-bola | took me the better part of an hour to figure out why it wasnt working :( |
| 11:31.18 | jkroon | E-bola, it's partially a bug in *, but primarily a misunderstanding of SQL on your side. |
| 11:32.05 | jkroon | asterisk does a straight copy of the string you supply into the query, without trying to escape, without adding `` or anything else. Adding `` automatically to table and column names automatically might be a good idea though. |
| 11:32.36 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 11:33.12 | E-bola | jkroon: Yep completely see it that way too, i guess i counted on asterisk to fix minor syntax stuff like that for me :) |
| 11:33.20 | E-bola | Im not that used to handling sql syntax |
| 11:33.43 | jkroon | how's your C? |
| 11:33.52 | E-bola | Close to non existing hehe |
| 11:34.26 | E-bola | I've only had to deal with perl and php lately and a little tcl |
| 11:34.39 | E-bola | although i did take java and c++ classes a looong time ago hehe |
| 11:35.19 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:35.27 | jkroon | your opinion, which is more likely: memory allocator bug in libc and/or STL or a memory corruption bug in * ? |
| 11:36.27 | E-bola | I'd asume libc and STL is more polished, so i'd say *. But then again what do I know :) |
| 11:41.36 | jacc0 | I'd say asterisk 2 |
| 11:41.46 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 11:42.22 | E-bola | Can anybody tell me what the disposition field in the cdr data is? |
| 11:45.01 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 11:49.47 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 11:50.40 | *** join/#asterisk elfelvin (~elfelvin@87-194-69-88.bethere.co.uk) |
| 11:50.53 | Gugge | ${CDR(disposition)} = status of the call (ANSWERED, BUSY, NO ANSWER) |
| 11:50.56 | Gugge | http://www.asteriskguru.com/tutorials/cdr_custom_conf.html |
| 11:57.55 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 11:59.55 | E-bola | Hmmmm mine. lists disposition = torture |
| 11:59.58 | E-bola | sort of weird.... |
| 12:02.29 | *** join/#asterisk iMelnik (~melnik@217.147.17.74) |
| 12:02.30 | E-bola | Ahh its because the values apparently was changed |
| 12:15.30 | felimwhiteley | hi folks, anyone ever successfully built 1.8(.2.3) with the res_ais module? I've followed the guide but I keep getting "Invalid IPC Credentials".. there's not too many things in the howto so I can't see where I'd have made the mistake |
| 12:15.52 | felimwhiteley | trying with 1.8.4.4 now.. |
| 12:16.09 | *** join/#asterisk kamh (~kamh@static-78-9-97-126.ssp.dialog.net.pl) |
| 12:16.21 | *** part/#asterisk kamh (~kamh@static-78-9-97-126.ssp.dialog.net.pl) |
| 12:16.31 | Bipul | can any one tell me how to config spi.conf |
| 12:16.51 | Bipul | it's totally diffrent from what is writeen over in wiki |
| 12:19.17 | E-bola | just read the file? |
| 12:20.11 | jkroon | WARNING[2773]: utils.c:1538 __ast_string_field_init: trying to reset empty pool <-- receiving lots and lots of these during * startup - are they problematic or can I safely ignore them? |
| 12:20.12 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 12:20.42 | felimwhiteley | damn same error again |
| 12:20.43 | felimwhiteley | ERROR[19134]: ais/clm.c:140 ast_ais_clm_load_module: Could not initialize cluster membership service: Unknown |
| 12:21.28 | jkroon | felimwhiteley, never used ais - very interested though. |
| 12:23.37 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 12:28.16 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:28.43 | ik_5 | What variable/function holds this infomation: SIP/out-000001b6 ? |
| 12:33.39 | *** join/#asterisk ezano (~ezano@sto93-2-82-228-142-248.fbx.proxad.net) |
| 12:33.46 | ezano | hi o/ |
| 12:34.36 | ezano | need help someone ? |
| 12:34.46 | ezano | sorry my english is very limited |
| 12:35.21 | *** join/#asterisk coppice (~coppice@m121-202-47-88.smartone-vodafone.com) |
| 12:36.20 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 12:37.12 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:38.28 | jacc0 | @ezano: I guess you are looking for ${CHANNEL} |
| 12:39.12 | jacc0 | you can use it to bridge() with the active channel or to softhangup() |
| 12:40.03 | jacc0 | sorry, this was for you ; ik_5 |
| 12:40.17 | jacc0 | ~ask |
| 12:40.17 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:43.06 | jkroon | rofl @ against our will |
| 12:44.12 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:44.41 | ik_5 | jacc0, thanks, but ${CHANNEL} provide me the "dial" channel name, and not the peer name :( I need to know at the end the SIP return code (200, 480 etc...) and for using hash I need the peername |
| 12:45.16 | *** join/#asterisk rabby (~rabby@mnch-4d0db280.pool.mediaWays.net) |
| 12:45.20 | *** join/#asterisk nightrid3r (nightrid3r@91.176.246.46) |
| 12:46.00 | rabby | how to use the result from a System(...) command for a GoToIf line in extensions.conf? |
| 12:46.18 | ezano | I have an asterisk interface with a2billing, when I want to call a peers, I have two solutions, 401 first this is the digest auth if I give the nice credentials I have a 403 forbidden and If I give wrong credentials too |
| 12:46.41 | Gugge | rabby: what result? |
| 12:46.54 | ezano | dialplan is nice ans sip.conf too |
| 12:47.05 | rabby | Gugge: the output should be the result and can be 1 or 0 |
| 12:47.21 | Gugge | ${SYSTEMSTATUS} ? |
| 12:47.36 | Gugge | it can be "FAILURE" or "SUCCESS" |
| 12:47.44 | rabby | Gugge: System(/bin/application do something) |
| 12:47.44 | ezano | on asterisk: sip show peers // says me I'm connecting7 |
| 12:47.53 | Gugge | if you want to read the output of a command you dont want to use System() |
| 12:49.17 | Gugge | rabby: System() can only return success or failure, maybe you want ${SHELL()} ? |
| 12:49.23 | jacc0 | s/nice/good |
| 12:50.00 | ezano | hop mistake, its a 503 error message, not 403. |
| 12:50.42 | jacc0 | 503 = service unavailable |
| 12:51.00 | ezano | yes I know |
| 12:51.11 | ezano | It's write on my twinkle |
| 12:51.18 | jacc0 | what kind of peer is it? an asterisk pbx? |
| 12:51.35 | ezano | asterisk console says me: error on transmission |
| 12:51.37 | rabby | Gugge: seems like my asterisk does not know this |
| 12:51.55 | Gugge | rabby: what asterisk version? |
| 12:51.57 | ezano | yes I have two asterisk serv |
| 12:52.16 | ezano | retrans_pkt: Retransmission timeout reached on transmission etc ... |
| 12:52.30 | rabby | 1.6 |
| 12:52.41 | Gugge | "core show function SHELL" works on my 1.6.2.something |
| 12:53.08 | hrolf | I'm sending a custom SIP Header, how can I extract it in extensions.conf ? |
| 12:53.13 | rabby | Gugge, you are right. i'll re-check the command in the config... |
| 12:53.55 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:54.03 | felimwhiteley | jkroon: hmm sorry I replied in wrong channel! :) |
| 12:54.04 | felimwhiteley | jkroon: yeah trying to build a test config to see can I get message waiting and device state.. so far nothing has worked.. tried a few versions. I get corosync working as per https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+AIS |
| 12:55.53 | *** join/#asterisk valera (~valera@88.211.48.174) |
| 12:57.38 | valera | hello, guys, I am seeing rather strange picture in the logs (asterisk-1.8.2) sometimes - first digit strips off successfully with ${EXTEN:1} and sometimes it does not - of course when it does not strip - call fails - any pointers (it worked reliably with asterisk 1.6.x) ? |
| 12:57.51 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 12:59.44 | jkroon | felimwhiteley, what I know of AIS is ... uhrm, well, rudementary to say the least. |
| 13:00.14 | jkroon | ideally I'd like to have 2 core servers that knows everything about everybody, and then remote servers that only tracks a subset each. |
| 13:01.01 | felimwhiteley | well if they are ont he same fast local network then AIS should in theory be fine.. otherwise if it's slower links then XMPP |
| 13:01.46 | jkroon | XMPP ? |
| 13:02.35 | jkroon | yea, the core nodes should be on relatively fast links (<5ms round-trip, with at least 1Mbps of BW), and the "remote" nodes sits on slower (512Kbps) links at latences between 10 and 50ms. |
| 13:03.06 | jkroon | does your cluster at least link? |
| 13:03.38 | felimwhiteley | aye XMPP, pu-sub mode can be used on slower links |
| 13:03.50 | felimwhiteley | yeah the cluster seems to go ready on it's won |
| 13:04.00 | *** join/#asterisk suma (~pongada@c-98-245-176-77.hsd1.co.comcast.net) |
| 13:04.08 | felimwhiteley | but once I try to fire up asterisk I can see IPC comminication error in the logs |
| 13:04.15 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 13:04.16 | felimwhiteley | and asterisk jsut won't load the module |
| 13:04.30 | felimwhiteley | ERROR[19134]: ais/clm.c:140 ast_ais_clm_load_module: Could not initialize cluster membership service: Unknown |
| 13:06.25 | jacc0 | damn, astrisk keeps crashing on me : |
| 13:06.27 | jacc0 | WARNING[11011]: channel.c:2801 ast_hangup: Hard hangup called by thread -1253889136 on Local/13171@dialer-bd05;2, while fd is blocked by thread -1250448496 in procedure ast_waitfor_nandfds! Expect a failure |
| 13:07.53 | valera | any reason why Dial(SIP/${EXTEN:1}@sipprovider,60) - could sometimes Dial as ${EXTEN}@sipprovider without removing first digit ? |
| 13:09.10 | *** join/#asterisk ickmund (~ickmund@91.126.133.242) |
| 13:10.37 | jacc0 | @valera: maybe it works for local numbers and fails for non-local numers? |
| 13:11.16 | jacc0 | or maybe it fails on a + |
| 13:11.33 | valera | jacc0: well, it fails for the same local number calling to the same destination - sometimes call is passing through fine so 9XXXXX - cuts 9 properly |
| 13:11.36 | valera | sometimes it does not |
| 13:11.46 | valera | all happends after upgrade, trying to figure out why |
| 13:12.23 | jacc0 | I have no explenation for that |
| 13:12.56 | valera | jacc0: no worries :) thats usual when dealing with asterisk :) |
| 13:13.43 | *** join/#asterisk m_tadeu (~m_tadeu@static-b5-252-50.telepac.pt) |
| 13:13.48 | Katty | goooooooooooood morning beautifuls! |
| 13:13.53 | *** join/#asterisk prologic (~prologic@unaffiliated/prologic) |
| 13:13.54 | jacc0 | :) |
| 13:13.55 | irroot | mawnin katty |
| 13:13.57 | jacc0 | good afternoon |
| 13:13.59 | Katty | heat index of 113F today! i hope you have air conditioning! |
| 13:14.07 | m_tadeu | g'morning |
| 13:14.17 | jacc0 | it's not that warm over here |
| 13:14.28 | prologic | Is there any way I can debug why my LinkSys SPA 2102 isn't registering with my PBX (asterisk) ? |
| 13:14.43 | prologic | It's NATed if that helps |
| 13:14.58 | irroot | has the winter chills 14c |
| 13:15.02 | jacc0 | @prologic: tshark -R sip -V |
| 13:15.21 | prologic | is that a cli tool ? |
| 13:15.30 | m_tadeu | I'm getting lots of warnings in asterisk log saying " WARNING[21739] xmldoc.c: Couldn't find application" or " WARNING[21739] xmldoc.c: Couldn't find function"...can't google anything useful |
| 13:15.31 | jacc0 | nope its text mode wireshark |
| 13:15.32 | prologic | I can't exactly use wireshark on my desktop at home atm :/ |
| 13:15.40 | prologic | ahh sweet |
| 13:15.44 | prologic | I'll try that :) thanks! |
| 13:16.20 | Katty | irroot: nice. |
| 13:16.26 | Katty | irroot: i'm looking forward to fall ^_^ |
| 13:16.48 | irroot | where you falling *duck* |
| 13:17.02 | prologic | Also does anyone know much about PBX in a Flash ? |
| 13:17.18 | Katty | irroot: well FIRST there is autumn camping. |
| 13:17.32 | Katty | irroot: pretty leaves, lots of friends, cool weather, and SMORES |
| 13:17.49 | irroot | back on the sugar :P |
| 13:18.05 | Katty | then there is halloween |
| 13:18.08 | Katty | thanksgiving |
| 13:18.10 | Katty | christmas |
| 13:18.13 | Katty | new years |
| 13:18.29 | Katty | and all the fun indoor stuff that we've been putting off all summer |
| 13:18.30 | *** join/#asterisk engrxyz (~fgdfgfdg@212.23.51.7) |
| 13:18.41 | Katty | theatre, bowling, museums ^_^ |
| 13:18.44 | Katty | boingboing |
| 13:19.26 | Katty | ohh and the japanese festival in st. louis |
| 13:19.28 | Katty | and the ZOO! |
| 13:19.53 | irroot | hehe looks like you got it planned :P |
| 13:20.25 | Katty | oh idk about planned. |
| 13:21.54 | ssureshot | I previously used "VoiceMail(u151@office)" for voicemail,, If I'm reading right has the correct syntax changed to "VoiceMail(151@office,u)"? |
| 13:22.27 | *** part/#asterisk prologic (~prologic@unaffiliated/prologic) |
| 13:25.08 | E-bola | Anybody here uses cdr-stats? |
| 13:27.07 | Katty | yes. |
| 13:27.30 | E-bola | Is it normal for the graphs to be empty for the first couple of days? |
| 13:27.52 | E-bola | I can see all the cdr entries in search cdr, but more or less all the other tabs are blank |
| 13:30.03 | Katty | hmm |
| 13:30.11 | Katty | are you getting any errors about java? |
| 13:31.13 | E-bola | nope no errors |
| 13:31.44 | Katty | are you getting any data when you do a basic search? |
| 13:31.58 | *** join/#asterisk prologic (~prologic@unaffiliated/prologic) |
| 13:32.11 | E-bola | yes under search cdr it apears fine |
| 13:35.44 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 13:36.11 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 13:37.32 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-eknjvnglytpnkyui) |
| 13:37.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:38.53 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:38.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:43.19 | m_tadeu | I'm getting lots of warnings in asterisk log saying " WARNING[21739] xmldoc.c: Couldn't find application" or " WARNING[21739] xmldoc.c: Couldn't find function"...can't google anything useful...what is wrong with this? |
| 13:43.41 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 13:45.41 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 13:46.03 | prologic | thansk guys - got my pbx working :) |
| 13:48.33 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:50.05 | pabelanger | m_tadeu: do you have libxml2-dev installed before you compiled asterisk? |
| 13:52.51 | m_tadeu | pabelanger, I didn't compile...It was installed from packages, using the packages.asterisk.org repository |
| 13:53.28 | pabelanger | m_tadeu: which OS? |
| 13:53.40 | m_tadeu | pabelanger, ubuntu |
| 13:53.48 | pabelanger | Hmm |
| 13:54.27 | pabelanger | lucid, maverick, natty? |
| 13:56.08 | m_tadeu | pabelanger, lucid |
| 13:57.35 | jacc0 | you have doxygen installed? |
| 13:58.04 | jacc0 | I guess the warnings are about generating progdocs |
| 13:58.09 | russellb | nope |
| 13:58.33 | russellb | sounds like it failed to load the XML file that has the application, function, etc. docs in it |
| 13:58.48 | *** join/#asterisk [Outcast] (~anonymous@westford-nat.juniper.net) |
| 13:59.02 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:00.17 | jacc0 | asterisk install uses doxygen to generate the XML; or am I mistaking? |
| 14:00.46 | pabelanger | jacc0: no, xmldoc are just stripped from source |
| 14:00.59 | pabelanger | m_tadeu: create a new issue on JIRA, I'll look at it today |
| 14:01.02 | pabelanger | see if I can reproduce |
| 14:01.25 | pabelanger | m_tadeu: *CLI> core show version |
| 14:01.43 | jacc0 | make progdocs requires doxygen to generate the help |
| 14:02.33 | jacc0 | isn't that the help info asterisk shows in cli? |
| 14:02.47 | m_tadeu | pabelanger, Asterisk 1.8.4.1-1digium1~lucid built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-05-23 21:10:48 UTC |
| 14:03.01 | russellb | jacc0: no |
| 14:03.14 | jacc0 | okay, then what is it for? |
| 14:03.19 | russellb | that's C API documentation |
| 14:04.17 | russellb | http://www.asterisk.org/doxygen/trunk/index.html |
| 14:05.30 | *** join/#asterisk bchia (~Adium@nat/digium/x-pwfediiwnecnehio) |
| 14:07.09 | jacc0 | ty |
| 14:07.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:07.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:07.55 | schmidts | btw russellb it would be very nice if doxygen will also work without the index.html ;) |
| 14:08.45 | russellb | marketing runs www.asterisk.org |
| 14:08.47 | russellb | complain to them |
| 14:08.52 | schmidts | :D |
| 14:09.35 | russellb | actually, apparently they don't want feedback - http://www.asterisk.org/feedback |
| 14:09.38 | russellb | there's no form there, heh |
| 14:11.16 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 14:11.46 | *** part/#asterisk iMelnik (~melnik@217.147.17.74) |
| 14:11.59 | m_tadeu | pabelanger, I'll create the issue a bit latter....confirm me the place to do it...is it http://www.atlassian.com/software/jira? |
| 14:12.30 | pabelanger | m_tadeu: https://issues.asterisk.org/jira/ |
| 14:13.26 | m_tadeu | pabelanger, ok...thanx |
| 14:15.40 | *** join/#asterisk last1 (~last1@modemcable238.94-200-24.mc.videotron.ca) |
| 14:16.16 | *** part/#asterisk prologic (~prologic@unaffiliated/prologic) |
| 14:16.40 | last1 | has the asterisk-addons package on debian been migrated to asterisk-mysql ? |
| 14:16.45 | schmidts | russellb i have sent a mail through the contact page maybe it reach the right person ;) |
| 14:20.17 | m_tadeu | pabelanger, ok I creted the issue....thanx again |
| 14:27.00 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 14:27.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:27.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:34.00 | last1 | anyone alive ? |
| 14:38.13 | E-bola | You can asume everyone is generally not dead when they are on the internet... |
| 14:38.43 | irroot | T Virus ?? |
| 14:39.14 | *** join/#asterisk albertoandrade (~albertoan@187.88.248.135) |
| 14:39.28 | last1 | they could be dead |
| 14:39.35 | last1 | but their computers still online! |
| 14:39.42 | last1 | that's why I was asking for human confirmation :) |
| 14:42.00 | last1 | so, does anybody know what happened to the asterisk-addons package in debian ? |
| 14:42.05 | *** join/#asterisk binbash_ (~peter@insley.demon.nl) |
| 14:46.34 | Chainsaw | last1: If the main Asterisk package was (finally) upgraded to 1.8; the -addons would disappear. |
| 14:46.40 | Chainsaw | last1: Because 1.8 has those extras bundled in. |
| 14:47.36 | E-bola | I dont think tzafrir_laptop changed it to 1.8 in stable yet |
| 14:47.53 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:47.59 | E-bola | Nope he luckily did not |
| 14:48.37 | tzafrir_laptop | It's in testing, though. So it should eventually be backported |
| 14:49.11 | E-bola | tzafrir_laptop: How's the timer modules behaving in ur package? |
| 14:49.28 | tzafrir_laptop | I normally use the dahdi one... |
| 14:49.39 | E-bola | Do you have dahdi hardware? |
| 14:51.53 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:53.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:53.35 | *** join/#asterisk dfamorato (~dfamorato@2001:470:5:630:cabc:c8ff:fe9b:26b7) |
| 14:57.59 | *** join/#asterisk Mateuss (~Mateus@62.85.93.44) |
| 14:58.46 | Mateuss | hallo everybody... ive got a problem with configure SRTP in Asterisk... |
| 14:58.49 | Mateuss | checking for the ability of -lsrtp to be linked in a shared object... no |
| 14:59.09 | Mateuss | tryed ./configure CFLAGS=-fPIC --prefix=/usr and without prefix |
| 14:59.13 | Mateuss | the same. |
| 14:59.31 | Mateuss | lsrtp can not be linked in a shared object. |
| 14:59.43 | Mateuss | Did someone expierienced such a issue? |
| 15:02.01 | last1 | chainsaw: I'm actually using asterisk 1.4 |
| 15:02.16 | last1 | but with apt-cache search I can't find asterisk-addons |
| 15:03.17 | Mateuss | i think that TLS/SRTP is one of the greatest features...but its pain in the ass to get it working :/ |
| 15:03.27 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:03.44 | Chainsaw | Mateuss: Please blog or otherwise document a working deployment at your earliest convenience. |
| 15:03.59 | Chainsaw | Mateuss: I'm not going to bother until I've heard from at least 3 people that have it operational. |
| 15:04.36 | Mateuss | thats PIAF what i am using. With Asterisk 1.8.4.4 |
| 15:04.47 | Mateuss | i have dowloaded that one ftp://ftp.owlriver.com/pub/local/ORC...4-1orc.src.rpm |
| 15:04.56 | Mateuss | installed source package |
| 15:05.03 | last1 | in here for example: http://packages.debian.org/search?keywords=asterisk |
| 15:05.08 | last1 | I can't see any addons package |
| 15:06.08 | Mateuss | extracted, ./configure srtp make make install / and then reconfigured asterisk and there is my error with libsrtp :/ |
| 15:14.19 | *** join/#asterisk pbati (~pbati@189.58.102.230) |
| 15:14.38 | pbati | olaaa , alguem fala portugues? |
| 15:16.13 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:16.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:17.11 | pbati | Hello! Anyone know the versions that accept the asterisk skypeforasterisk? |
| 15:21.07 | Mateuss | you mean that skype for asterisk for a 66$ for a single channel? |
| 15:21.12 | ChannelZ | pretty much any |
| 15:21.28 | Mateuss | like i know you can integrate it to any version |
| 15:21.37 | Mateuss | oh... allready answered :) |
| 15:22.59 | pbati | I want to include in version 1.8? |
| 15:23.07 | ChannelZ | http://downloads.digium.com/pub/telephony/skypeforasterisk/ |
| 15:24.34 | pbati | I am not able to explain, I already bought the site of Digium, my doubt is if I can install the Asterisk 1.8 as soon as the textbooks say so version 1.6 |
| 15:24.34 | pbati | Novo! Clique nas palavras acima para ver traduções alternativas. Dispensar |
| 15:24.34 | pbati | 0. |
| 15:24.35 | pbati | 0. |
| 15:24.51 | ChannelZ | uhm... |
| 15:25.22 | ChannelZ | See the link. Download the appropriate version based on your major version of Asterisk |
| 15:25.39 | last1 | so in Debian, is asterisk-addons actually split into: asterisk-mp3, asterisk-mysql and asterisk-ooh323c |
| 15:25.51 | pbati | ok |
| 15:25.53 | pbati | thanks |
| 15:26.42 | *** join/#asterisk irroot (~irroot@41.52.219.172) |
| 15:26.55 | Mateuss | great... SRTP on Ubuntu+Asterisk 1.8.2.2 is working just great, but PBXinFlash with 1.8 not working... |
| 15:29.27 | m_tadeu | I'm trying to set a new user in sip.conf, but it's set as unreachable. I don't know what is wrong with it, how can I check what is going on? |
| 15:31.59 | ChannelZ | is it a host=dynamic peer? |
| 15:33.16 | ChannelZ | Or is it otherwise registering? NAT involved? |
| 15:34.49 | m_tadeu | the peer is not dynamic...nat is involved...that peer is another asterisk server |
| 15:35.18 | m_tadeu | I'm using it as a gateway to landline phones |
| 15:36.04 | m_tadeu | I have a registration string and in the console says it's registered |
| 15:36.54 | *** join/#asterisk brainiac (~brainiac@c-98-193-140-192.hsd1.tn.comcast.net) |
| 15:37.04 | m_tadeu | when I call my DID I get " chan_sip.c:21512 handle_request_invite: Sending fake auth rejection for device <sip:+351xxxxxxxxx@213.13.89.67;user=phone>;tag=001b0f49000011d6" |
| 15:39.12 | *** join/#asterisk dfamorato (~dfamorato@2001:470:5:630:cabc:c8ff:fe9b:26b7) |
| 15:55.05 | *** join/#asterisk hfb (~hfb@pool-96-247-108-172.lsanca.dsl-w.verizon.net) |
| 15:57.20 | godmachine-x6 | how do i fix my dialplan to dial a regular number when the user provides the + their self.. for example the sip phone user has contacts saved with +1(areacode)number what do i need to specify in extensions.conf to recognize the + when the user dials it like that? |
| 16:03.28 | *** part/#asterisk pbati (~pbati@189.58.102.230) |
| 16:04.29 | *** join/#asterisk lcat (~lcat@187.45.253.242) |
| 16:09.49 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 16:11.16 | p3nguin | godmachine-x6: The plus is not supposed to be dialed. Remove it from any contacts. |
| 16:11.43 | godmachine-x6 | i know its not suppose to but my google voice saves a lot of contacts including the thing |
| 16:11.56 | godmachine-x6 | would [\+] work ? |
| 16:12.16 | p3nguin | The Google Voice contacts don't interact with Asterisk's dial plan. |
| 16:12.26 | godmachine-x6 | no i know they dont |
| 16:12.34 | godmachine-x6 | but with blink my soft phone i use on the laptop |
| 16:12.40 | godmachine-x6 | i can sync my google voice contacts |
| 16:12.53 | godmachine-x6 | and it stores a lot of the numbers as +1 |
| 16:12.54 | p3nguin | And it tries to dial the + from the contact info? |
| 16:12.58 | godmachine-x6 | yes |
| 16:13.07 | godmachine-x6 | so i want something in the dialplan to fix that |
| 16:13.10 | godmachine-x6 | and allow it to work |
| 16:13.31 | _Corey_ | godmachine-x6: Just add a dial plan for +... exten => _+1.,1,Whatever |
| 16:14.24 | p3nguin | Try something like _+1NXXNXXXXXX,1,Dial() |
| 16:14.26 | godmachine-x6 | i tried that |
| 16:14.34 | p3nguin | Make sure you take out the extra + in the Dial() string. |
| 16:14.34 | godmachine-x6 | not allowed it says |
| 16:14.40 | godmachine-x6 | i did |
| 16:14.59 | p3nguin | Did you try _[+]1NXXNXXXXXX ? |
| 16:15.01 | godmachine-x6 | exten => _+1NXXNXXXXXX,n,Dial(gtalk/asterisk/${EXTEN}@voice.google.com) << doesnt work |
| 16:15.08 | godmachine-x6 | yep i tried that as well |
| 16:15.18 | godmachine-x6 | and im reloading the dialplan each time |
| 16:16.41 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 16:17.02 | atheos | godmachine-x6, have you tried two underscores? __NXXNXXXX |
| 16:17.11 | godmachine-x6 | nope |
| 16:17.14 | godmachine-x6 | let me try that |
| 16:17.27 | atheos | hmm, on second thought, I think that probably won't work |
| 16:17.46 | godmachine-x6 | what about [_] ? |
| 16:18.28 | atheos | maybe _.NXXNXXXX |
| 16:19.26 | godmachine-x6 | nope |
| 16:19.55 | godmachine-x6 | guess ill just have to go through all my gv contacts and edit out the + in them |
| 16:21.41 | p3nguin | I've never ever considered accepting a + in a dialed number, so that's what I would do. |
| 16:21.44 | *** part/#asterisk vikapi (~vikapi@124.125.34.134) |
| 16:21.47 | *** join/#asterisk binbash_ (~peter@insley.demon.nl) |
| 16:22.51 | _Corey_ | A couple of my carriers insist on e164 with the +... I've never had an issue matching it |
| 16:23.09 | atheos | I have a bill collector that sends a + inside of their callerid (numeric portion), and I've not found a way to filter them in my blackhole macro. different, but similar issue. |
| 16:24.47 | p3nguin | _corey_: How do you dial the + on your phones? |
| 16:25.17 | godmachine-x6 | yea _Corey_ |
| 16:25.19 | _Corey_ | We don't, the carrier originates calls with +1NPANXXxxxx |
| 16:25.54 | godmachine-x6 | i have a feeling that when i take all these +'s out of my contacts gv isn't going to tie the names to the numbers correctly when i text/call |
| 16:26.04 | godmachine-x6 | more so on the receiving end.. |
| 16:27.14 | _Corey_ | Doesn't matter really, just match whatever you're getting in the dial plan and clean it up |
| 16:27.54 | p3nguin | atheos: That would be easy. GotoIf($["${CALLERID(num):0:1}" = "+"]?true:false) |
| 16:28.50 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 16:28.54 | wcselby | o/ |
| 16:29.07 | *** join/#asterisk coppice (~chatzilla@116.92.39.52) |
| 16:29.15 | atheos | penguin - any idea why this wouldn't match in a switch statement? '+18664319119' |
| 16:29.37 | godmachine-x6 | i took the leading +1 out of all my contacts |
| 16:29.40 | godmachine-x6 | we'll see how that works |
| 16:29.55 | p3nguin | I have no idea why a plus won't match in exten or in switch, but apparently it doesn't. |
| 16:29.57 | wcselby | atheos- it depends on the switch statement |
| 16:30.06 | wcselby | you need to show it to us first |
| 16:30.11 | godmachine-x6 | p3nguin:: someone pointed out the reason my calls were dropping when trying to dial the SIP phone user located on the same box as the asterisk server was simply because of that |
| 16:30.19 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 16:30.21 | p3nguin | Yeah, that's what I told you. |
| 16:30.29 | atheos | wcselby - it's matched against ${CALLERID(num)} - everything else works. |
| 16:30.49 | atheos | it's not an issue I'm working on anymore, just thought about it when godmachine-x6 described his matching issue |
| 16:30.51 | godmachine-x6 | looks like there would be a way to make it work.. it works fine for exactly 36 seconds then asterisk sends a (BYE) |
| 16:31.37 | wcselby | I've matched extens with the _+X. or _+1X. pattern before.....? |
| 16:31.50 | wcselby | but I guess I'm jumping into the middle of a conversation here, sorry |
| 16:31.52 | p3nguin | Are you asking us if you've matched that? |
| 16:32.15 | _Corey_ | wcselby: Yeah, same here.. :) Works fine for me the same way. |
| 16:32.43 | wcselby | no p3nguin I was saying that I've done it before, and I was questioning why others were having trouble doing the same. But I didn't bother spelling it all out, because I realized I had jumped into what seemed to be the middle of a conversation |
| 16:32.55 | godmachine-x6 | so you can dial from a sip phone the number +11231231234 and it will allow it? |
| 16:33.03 | p3nguin | I'll get you up to speed: godmachine-x6 has indicated that he is unable to match a dialed phone number with + on the front of the exten. |
| 16:33.36 | wcselby | godmachine-x6- I don't normally dial the + from the phone, but I can handle it with the correct pattern matching inside the dialplan (the patterns I indicated just a short while ago) |
| 16:33.43 | p3nguin | Some SIP phones, even if you can actually enter the + on the number to dial, will not actually send the + in the SIP traffic. |
| 16:34.07 | godmachine-x6 | blink and linphone are both trying to send it as a part of the dialed number |
| 16:34.16 | godmachine-x6 | and _+ doesn't work for me |
| 16:34.29 | wcselby | so, if for instance, the callerid on the phone said +17133437300, and I clicked the call back button, I could actually dial that out. But my phone apparently supports sending the + in the SIP traffic. |
| 16:35.01 | wcselby | godmachine-x6- show us the CLI output of a failed call |
| 16:35.10 | p3nguin | I'm going to need to see both dial plan snippets and verbose log of a call that either does or does not work. |
| 16:35.18 | wcselby | and the matching dialplan code from extensions.conf |
| 16:35.19 | _Corey_ | I suspect it's more of a problem with the phone dialing something else not matching the pattern |
| 16:35.23 | godmachine-x6 | Call from 'hxnpbgcl' to extension '+18889799949' rejected because extension not found in context 'phones'. |
| 16:35.40 | wcselby | so now you need to show us the context phones from your extensions.conf |
| 16:35.57 | godmachine-x6 | exten => _+XNXXNXXXXXX,n,Dial(gtalk/asterisk/${EXTEN}@voice.google.com) |
| 16:36.04 | p3nguin | That doesn't match. |
| 16:36.13 | p3nguin | Try it on the CLI, too: dialplan show +18889799949@phones |
| 16:36.38 | godmachine-x6 | There is no existence of +18889799949@phones extension |
| 16:36.45 | p3nguin | well, maybe it matches... I may have misread. |
| 16:36.47 | wcselby | and godmachine-x6 - show us the entire phones context |
| 16:36.51 | wcselby | please pastebin it |
| 16:36.57 | godmachine-x6 | i will im not a flooder |
| 16:36.58 | godmachine-x6 | haha |
| 16:37.18 | p3nguin | I'm going to test + in my dial plan just to see what happens. |
| 16:37.56 | godmachine-x6 | http://pastebin.com/ULBdiGvw |
| 16:38.24 | wcselby | you skipped the 1 |
| 16:38.26 | _Corey_ | uh, yeah |
| 16:38.37 | wcselby | the 1 priority |
| 16:38.37 | _Corey_ | you need a 1 |
| 16:38.37 | godmachine-x6 | no i used X instead of 1 |
| 16:38.42 | _Corey_ | ,n, should be ,1, |
| 16:38.42 | wcselby | on your exten => _+XNXXNXXXXXX,n,Dial(gtalk/asterisk/${EXTEN}@voice.google.com) |
| 16:38.49 | wcselby | it should be exten => _+XNXXNXXXXXX,1,Dial(gtalk/asterisk/${EXTEN}@voice.google.com) |
| 16:38.54 | godmachine-x6 | oh |
| 16:39.04 | p3nguin | All extensions have to begin with priority 1. |
| 16:39.10 | _Corey_ | :) |
| 16:39.10 | godmachine-x6 | what will that change? |
| 16:39.11 | p3nguin | n means next. |
| 16:39.15 | godmachine-x6 | hmm |
| 16:39.19 | nightrid3r | n=next but 1st need to be 1 |
| 16:39.22 | godmachine-x6 | all of mine had 1 before |
| 16:39.26 | wcselby | if you have an n without a 1 it will never match |
| 16:39.26 | godmachine-x6 | i mean n |
| 16:39.38 | p3nguin | You took out what used to be pri 1. |
| 16:40.17 | godmachine-x6 | thats working now |
| 16:40.21 | godmachine-x6 | haha |
| 16:40.22 | wcselby | you also need to add the 1 priority oin the next line |
| 16:40.26 | p3nguin | It's common for me to always use priority 1 for a NoOp(), and then never touch it again. That way I always use n on every other line that I will ever ever ever edit. |
| 16:40.30 | godmachine-x6 | i made them all 1 |
| 16:40.36 | wcselby | exten => _1NXXNXXXXXX,n,Dial(gtalk/asterisk/+${EXTEN}@voice.google.com) should be exten => _1NXXNXXXXXX,1,Dial(gtalk/asterisk/+${EXTEN}@voice.google.com) |
| 16:41.00 | wcselby | p3nguin- I do the same thing but with a Verbose() and some detail about what's going on |
| 16:41.20 | wcselby | I mean about what I'ma bout to do in the code. |
| 16:41.21 | p3nguin | If I have a Verbose() on it, then I skip the NoOp(). |
| 16:41.39 | p3nguin | I use NoOp() on the ones that I don't require verbosity on. |
| 16:41.43 | wcselby | gotcha |
| 16:41.49 | godmachine-x6 | im learning here |
| 16:41.51 | wcselby | which is what NoOp was designed for. :) |
| 16:41.54 | godmachine-x6 | it workes now though |
| 16:41.59 | wcselby | nice to see someone using it correctly |
| 16:42.07 | wcselby | godmachine-x6- heh, yeah, it should :) |
| 16:43.02 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 16:44.13 | godmachine-x6 | so basically everything in an outgoing dialplan needs to be set to priority 1 and use n for doing things with incoming calls. unless i have some kind of thing i need to do after dialing out with a call. right? |
| 16:44.23 | wcselby | um |
| 16:44.38 | wcselby | every new exten in any context needs to start with a priority 1 |
| 16:44.50 | p3nguin | I got tired of having to change 1/n and n/1 when I was editing extensions, so I constructively eliminated it. |
| 16:45.06 | wcselby | p3nguin- :) |
| 16:45.13 | atheos | or just switch to AEL and be done with 1/n altogether |
| 16:45.43 | p3nguin | AEL doesn't have the same flexibility as the standard .conf, as far as I know. |
| 16:45.57 | wcselby | i just plain don't know AEL and have never taken the time to learn it |
| 16:46.30 | Mateuss | well... my SRTP looks like working now. But i have a question... to use TLS we need to put cert files to softphone... doeas anyone done this with Bria for Android? |
| 16:46.34 | p3nguin | godmachine-x6: Every single extension, regardless of which direction it makes a call go, must start with priority 1. That's where it begins. |
| 16:46.35 | atheos | I've had great results with AEL. Of course, I do a lot with AGI, so I may not need the flexibility that the standard .conf offers. |
| 16:47.49 | wcselby | i guess that's why I never learned AEL - I do what I need in regular .conf, and if I need anything extra I write a perl AGI |
| 16:48.15 | *** join/#asterisk byronsmith (~smithers@84.12.253.146) |
| 16:48.21 | wcselby | again - because I know perl and the Asterisk::AGI module |
| 16:49.05 | wcselby | if someone else finds that their way works better for them, more power to 'em. |
| 16:52.06 | wcselby | which directory does mixmonitor record to by default? i forget off the top of my head |
| 16:52.34 | p3nguin | /var/spool/asterisk/monitor/ |
| 16:52.42 | wcselby | thx |
| 16:53.47 | *** join/#asterisk godmachine-x6 (~godmachin@h92.171.140.67.dynamic.ip.windstream.net) |
| 16:54.04 | godmachine-x6 | i got disconnected did you guys see what i said about my CID? |
| 16:54.08 | p3nguin | no |
| 16:54.12 | p3nguin | godmachine-x6: Also, I would not use _+XNXXNXXXXXX, but _+1NXXNXXXXXX instead. |
| 16:54.19 | godmachine-x6 | yea i changed it |
| 16:54.27 | godmachine-x6 | <PROTECTED> |
| 16:54.53 | wcselby | there's a trick to that on the asterisk wiki |
| 16:54.56 | p3nguin | Okay, that's a pretty simple fix. Just a sec. |
| 16:55.16 | godmachine-x6 | i'll need it to remove that +1 in the CID as well |
| 16:55.20 | p3nguin | In 1.4 format, ExecIf($["${CALLERID(num):0:2}" = "+1"],Set,CALLERID(num)=${CALLERID(num):2}); |
| 16:55.38 | p3nguin | In 1.8 format, the Set will be slightly different. |
| 16:55.49 | godmachine-x6 | yea im using 1.8 |
| 16:55.54 | wcselby | http://pastebin.com/9azpj7di |
| 16:55.58 | p3nguin | like ExecIf($["${CALLERID(num):0:2}" = "+1"],Set(CALLERID(num)=${CALLERID(num):2}(); |
| 16:56.02 | wcselby | that's from the asterisk wiki about google voice integration |
| 16:56.50 | p3nguin | Oh, that's a completely different problem than I just solved. |
| 16:56.53 | wcselby | and here's the original page - https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 16:57.01 | p3nguin | punches himself in the ear |
| 16:57.02 | wcselby | yeah, p3nguin removed the +1 |
| 16:57.18 | wcselby | this cleans up the whole CID |
| 16:57.51 | p3nguin | I promise that I will try to pay better attention. |
| 16:57.59 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 16:58.01 | godmachine-x6 | hmm |
| 16:58.03 | godmachine-x6 | im confused lol |
| 16:58.21 | wcselby | go the the second link I provided |
| 16:58.30 | wcselby | go about 2/5 of the way down, the green box |
| 16:58.44 | wcselby | the second green box |
| 16:58.56 | godmachine-x6 | yea i see it |
| 16:59.04 | godmachine-x6 | but doesn't look like it will strip the +1 out too |
| 16:59.14 | wcselby | follow the example there (you can use your own variable names) |
| 16:59.26 | wcselby | if it doesn't, just add p3nguin's example before the last line of my example |
| 16:59.27 | p3nguin | Oh, I failed when I tried to alter for 1.8. I'll redo that. |
| 16:59.33 | p3nguin | like ExecIf($["${CALLERID(num):0:2}" = "+1"],Set(CALLERID(num)=${CALLERID(num):2})); |
| 16:59.51 | p3nguin | typoed the ) the first time |
| 17:00.20 | godmachine-x6 | ok and that needs to go in my dial plan before it tries to dial my SIP phone? |
| 17:00.32 | wcselby | before it tries to dial any phone |
| 17:00.37 | wcselby | on an inbound call from google voice |
| 17:01.11 | p3nguin | I would put it where ever wcselby said to put it. I didn't look at the wiki page, but I bet he told you the correct placement. |
| 17:01.12 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 17:01.19 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:02.00 | p3nguin | after all the other CID modification stuff, and before a Dial() seems good to me. |
| 17:03.14 | godmachine-x6 | exten => s,n,SendDTMF(1) why do i need this if i don't use call screening? |
| 17:03.14 | godmachine-x6 | i don't need either of these if im not using google voices call screening do i? : |
| 17:03.15 | godmachine-x6 | exten => s,n,Wait(2) |
| 17:03.15 | godmachine-x6 | exten => s,n,SendDTMF(1) |
| 17:03.16 | godmachine-x6 | thats going to wait 2 seconds and dial a 1 and then connect to my sip phone right? |
| 17:03.32 | p3nguin | You cannot turn off call screening when using the Chat method. |
| 17:03.40 | wcselby | google voice requires you to send the 1, even if you don't use the call screening |
| 17:03.53 | p3nguin | Even if you set it to Off, it is still going to be On. |
| 17:04.02 | godmachine-x6 | oh i see |
| 17:04.09 | godmachine-x6 | so then ill just have to manually press one when i get the call |
| 17:04.34 | p3nguin | If you don't do the Answer(), Wait(), and SendDTMF() you will. |
| 17:04.35 | godmachine-x6 | let me see if i got this right then |
| 17:04.43 | p3nguin | If you leave them, Asterisk presses 1 for you. |
| 17:04.45 | wcselby | or add D(:1) to your dial statement when you call your SIP phone |
| 17:04.51 | godmachine-x6 | yea i understand now.. didn't know the chat used the screening method |
| 17:04.56 | wcselby | which will also do it for you |
| 17:05.08 | godmachine-x6 | ill just leave it the way it is |
| 17:05.11 | p3nguin | oh, that's another interesting way to overcome the screening "bug." |
| 17:06.26 | godmachine-x6 | http://pastebin.com/wR0LAAiv |
| 17:06.35 | wcselby | p3nguin- plus, using this method allows you to use google voice's voicemail instead of asterisk's |
| 17:06.48 | p3nguin | Yeah, I was just thinking about that. |
| 17:06.51 | wcselby | since you won't actually send the 1 until after the call is answered |
| 17:06.58 | wcselby | erm, picked up by the SIP phone |
| 17:07.04 | godmachine-x6 | i see |
| 17:07.06 | godmachine-x6 | interesting |
| 17:07.08 | p3nguin | It would also allow me to use both a forwarding number AND the chat method to Asterisk. |
| 17:07.22 | wcselby | godmachine-x6- that's not going to do it |
| 17:07.26 | p3nguin | Right now, the cell phone will ring once before Asterisk answers and sends a 1. |
| 17:07.27 | wcselby | that will just strip the +1 |
| 17:07.45 | p3nguin | Now I have some dial plan editing to do. |
| 17:08.00 | godmachine-x6 | so i need to remove those two extens and do something like :exten => s,n,Dial(SIP/lance&SIP/laptop&SIP/ata,20)D(:1) ? |
| 17:08.19 | wcselby | godmachine-x6- here's what you need - http://pastebin.com/FrYb820z |
| 17:08.21 | godmachine-x6 | or add a new exten=> s,n,D(:1) ? |
| 17:08.25 | p3nguin | Do you want Google Voice to handle voice mail for unanswered calls? |
| 17:08.32 | godmachine-x6 | yes that would be great |
| 17:09.04 | wcselby | godmachine-x6- the D(:1) is an option to the Dial() command |
| 17:09.32 | p3nguin | In that case, I'd remove the Answer(), the Wait(), and the SendDTMF() in favor of wcselby's great suggestion of using the D() option in the Dial(). |
| 17:09.37 | wcselby | yeah |
| 17:09.49 | wcselby | here you go - http://pastebin.com/9NVfdMU8 |
| 17:10.14 | wcselby | heh |
| 17:10.23 | wcselby | crap i forgot to take off the Answer() on priority 1 |
| 17:10.27 | wcselby | so just take that off |
| 17:10.33 | wcselby | change the second line to 1 |
| 17:10.34 | wcselby | and go from there |
| 17:10.38 | p3nguin | The next line was already 1 anyway. :) |
| 17:11.01 | godmachine-x6 | why dont i need the Answer() ? |
| 17:11.10 | godmachine-x6 | asterisk don't pick it up before the sip does? |
| 17:11.14 | p3nguin | right |
| 17:11.19 | godmachine-x6 | nice |
| 17:11.43 | wcselby | sorry, i'm dividing attention between chat and my 2 year old's desire to watch veggie tales on netflix |
| 17:11.44 | p3nguin | The Answer() was used to get Asterisk to answer the line, then it would be presented with the call screen, which required pressing 1. |
| 17:12.11 | p3nguin | So it waited 2 seconds, and then pressed 1 for you. And THEN it sent the call to the SIP phones. |
| 17:12.42 | p3nguin | http://pastebin.com/xf96ztCE |
| 17:13.12 | godmachine-x6 | http://pastebin.com/AU2nExFT |
| 17:13.17 | godmachine-x6 | thats what i have |
| 17:13.22 | Mateuss | reboot |
| 17:13.22 | p3nguin | I hope I got that right. I'm a 1.4 guy, so some of my 1.8 syntax is not up to par. |
| 17:13.27 | p3nguin | REBOOT?! |
| 17:13.31 | p3nguin | Are you a Windows admin, too? |
| 17:14.08 | godmachine-x6 | i saved that dial plan |
| 17:14.16 | wcselby | lol. i think he was saying that HE was the one rebooting |
| 17:14.23 | godmachine-x6 | so now if i don't answer my sip phone, my calls will show in google voice as missed calls and not received |
| 17:14.27 | godmachine-x6 | right? |
| 17:14.36 | wcselby | godmachine-x6- correct |
| 17:14.44 | p3nguin | I'm not sure what he meant, but it's annoying how all these "Windows admins" always feel the need to reboot shit. |
| 17:14.46 | wcselby | and the caller should be presented with google voicemail |
| 17:14.51 | godmachine-x6 | and they will be able to just use the voicemail i already have setup |
| 17:14.59 | godmachine-x6 | very nice |
| 17:15.08 | wcselby | p3nguin- well, he disconnected from chat three lines after he said the word REBOOT |
| 17:15.14 | wcselby | oh, that was you |
| 17:15.16 | godmachine-x6 | and i can add me an extension like dialing 9 to dial my own number to get to my voice mail |
| 17:15.16 | wcselby | but yeah |
| 17:15.23 | godmachine-x6 | and never have to mess with asterisk voicemail |
| 17:15.37 | p3nguin | 9 kind of sucks, but yes it would work. |
| 17:15.45 | valera | p3nguin: hahaha, dont be stupid - when it comes to asterisk + dahdi - everyones need to reboot - simple insert of plug into wrong socket - bahm - you superstable linux server went down |
| 17:15.52 | p3nguin | I prefer *86 (*VM). |
| 17:15.56 | godmachine-x6 | well i just used that as an example |
| 17:15.57 | godmachine-x6 | lol |
| 17:16.27 | wcselby | holy crap p3nguin i never made that connection with *86 and *VM |
| 17:16.32 | godmachine-x6 | p3nguin:: i hope your string works for 1.8 |
| 17:16.33 | wcselby | slaps forehead |
| 17:16.38 | p3nguin | ;) |
| 17:16.40 | wcselby | it's so obvious now |
| 17:16.46 | wcselby | lol |
| 17:16.52 | godmachine-x6 | i reloaded my dial plan |
| 17:16.58 | godmachine-x6 | now i just have to wait on a call to see how it works |
| 17:17.03 | p3nguin | I still laugh at people who say they need to set up a dial plan so that they dial 9 "to get an outside line." |
| 17:17.12 | wcselby | lol yeah |
| 17:17.48 | *** join/#asterisk babilen (~babilen@unaffiliated/babilen) |
| 17:17.53 | *** part/#asterisk babilen (~babilen@unaffiliated/babilen) |
| 17:18.14 | p3nguin | Obviously some people require such silliness, such as when integrating with a legacy PBX or KSU, but with pure Asterisk it just doesn't make any sense. |
| 17:20.03 | *** join/#asterisk tamiel (~tamiel@ip-202.net-89-2-114.rev.numericable.fr) |
| 17:21.37 | godmachine-x6 | p3nguin:: care to make a test call so i can try your CID string? i wont answer and we'll see if it goes to my VM |
| 17:21.54 | p3nguin | sure |
| 17:22.32 | *** join/#asterisk PopAlex (~chatzilla@92.86.97.241) |
| 17:22.35 | godmachine-x6 | 931 413 9270 |
| 17:23.10 | p3nguin | Five ringy dingies, then GVvm. |
| 17:23.17 | godmachine-x6 | sip didn't ring though |
| 17:23.19 | godmachine-x6 | look at this |
| 17:23.37 | godmachine-x6 | eprecated syntax found. Please upgrade to using ExecIf(<expr>?Set(CALLERID(num)=4019034562)((null))) |
| 17:24.14 | p3nguin | Change the , to a ? in that line. |
| 17:24.40 | p3nguin | like ExecIf($["${CALLERID(num):0:2}" = "+1"]?Set(CALLERID(num)=${CALLERID(num):2})); |
| 17:25.02 | godmachine-x6 | done |
| 17:25.10 | p3nguin | 1.4 ExecIf($["${CALLERID(num):0:2}" = "+1"],Set,CALLERID(num)=${CALLERID(num):2}); |
| 17:25.20 | p3nguin | You can see the very subtle difference. |
| 17:25.47 | godmachine-x6 | yea i just wonder why they change things like that. was there some kind of improvement in the new design? |
| 17:26.00 | godmachine-x6 | saved it and reloaded the dial plan |
| 17:26.03 | p3nguin | The old design was somewhat nonsensical. |
| 17:26.10 | wcselby | they tried to move away from commas at one point |
| 17:26.20 | wcselby | well, that's not exactly true |
| 17:26.28 | wcselby | p3nguin's answer is more accurate |
| 17:26.34 | p3nguin | The new design follows Set()'s normal syntax. |
| 17:26.53 | godmachine-x6 | ok |
| 17:28.21 | godmachine-x6 | you can give it a shot again if you want p3nguin.. im hoping the 5 rings got sent to voicemail because asterisk rejected part of the plan but 5 rings is aweful short to go to the voicemail wonder if i will have time to answer most calls by the time they process using this method |
| 17:28.40 | p3nguin | ExecIf is the only one I have noticed so far that changed, but I'd imagine others such as GotoIf also got changed. |
| 17:29.17 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:30.07 | godmachine-x6 | ringing here |
| 17:30.20 | godmachine-x6 | i wont answer did it go to voicemail? |
| 17:30.23 | p3nguin | it did |
| 17:30.28 | godmachine-x6 | very nice |
| 17:30.39 | godmachine-x6 | CID worked right this time |
| 17:30.58 | godmachine-x6 | "4019034562" <sip:4019034562@127.0.0.1> |
| 17:32.42 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 17:33.03 | godmachine-x6 | now i guess the only thing i need to do is get my ATA and set it up to use the ATA account i made for it |
| 17:33.13 | godmachine-x6 | and hook it into the houses phone jacks |
| 17:33.22 | godmachine-x6 | and everything will be just like having a normal house phone :) |
| 17:33.37 | *** join/#asterisk avb (~avb@190.122.106.167) |
| 17:34.10 | godmachine-x6 | dont know how Call Waiting or 3-Way calling would work with GV.. for now im not worried about those though |
| 17:34.27 | wcselby | i wish google voice allowed me to port over my home number and business lines |
| 17:34.33 | p3nguin | Don't forget to disconnect the telco from your house at the NID. |
| 17:34.35 | wcselby | i've had those numbers for years |
| 17:34.48 | wcselby | and I don't want to have to teach new numbers to clients / family / friends |
| 17:35.00 | godmachine-x6 | p3nguin:: i don't have a telco phone line |
| 17:35.03 | p3nguin | They allow it, but they won't allow you to port them out later if you decide you don't want them on GV anymore. |
| 17:35.09 | avb | guys, im tired with my digium problem. :( After couple calls are passing fine with voice over digium quad t1 card, asterisk is losing an abbility to transfer audio. I cant hear even playback () audio |
| 17:35.18 | avb | probably anybody had such a problem? |
| 17:35.21 | godmachine-x6 | it was cut from the poll 3 yrs ago |
| 17:35.29 | godmachine-x6 | since then i've only used GV and cell |
| 17:35.31 | godmachine-x6 | lol |
| 17:35.47 | p3nguin | You should still remove the plug in the NID. |
| 17:35.51 | malcolmd | avb: please open a ticket with digium's support department directly: http://www.digium.com/support |
| 17:35.51 | wcselby | p3nguin- i thought it was just mobile numbers they were allowing to be ported in right now? |
| 17:36.17 | avb | malcolmd: ok, i will try |
| 17:36.19 | p3nguin | I don't know about that. You can buy a DID from an ITSP and port it in to Google. |
| 17:36.24 | avb | thanks for advice |
| 17:36.27 | godmachine-x6 | p3nguin:: i will do that in case someone was to try to hook up a telco line into my house |
| 17:36.28 | malcolmd | np |
| 17:36.28 | godmachine-x6 | lol |
| 17:36.58 | p3nguin | godmachine-x6: If you get voltage into the FXS port on the ATA, it could and probably will burn it out. |
| 17:37.18 | tyman | p3nguin: I've been happily using flowroute since your recommendation many months ago. They've had a few outages lately which make me want an secondary itsp configured. Who would you recommended that's comparable? |
| 17:37.21 | wcselby | p3nguin- hmmmmmm. both my numbers are currently sitting at flowroute. i ported my home number to them a couple years ago and bought a business line from them even before that. i wonder.......i'll have to dig aorund some later |
| 17:37.31 | p3nguin | tyman: VoIP.ms |
| 17:37.38 | godmachine-x6 | p3nguin:: i'll just cut the line from the box |
| 17:37.49 | godmachine-x6 | that way nobody will hook anything to it without me knowing first |
| 17:37.52 | p3nguin | Why? Just unplug it in the box. |
| 17:38.02 | tyman | p3nguin: thx |
| 17:38.52 | godmachine-x6 | be back in a few. going to fix some dinner |
| 17:38.59 | wcselby | tyman- i use vitelity as a secondary voip provider, they have a few extra bells and whistles and the cost isn't much higher |
| 17:39.22 | p3nguin | Since VoIP.ms resells Vitelity, and costs less, I see no reason to use Vitelity. |
| 17:39.33 | p3nguin | But that's just my opinion. |
| 17:39.41 | wcselby | well, there you go. i've not used or looked at VoIP.ms |
| 17:39.45 | wcselby | so I didn't know that's what they did |
| 17:39.59 | tyman | what's the best practice way to configure an itsp for redundancy? |
| 17:40.02 | p3nguin | I'm told that phone companies are required to leave the line connected to the house even in the absence of phone service. |
| 17:40.08 | tyman | any links? |
| 17:40.16 | *** join/#asterisk Mateuss (~Mateus@62.85.93.44) |
| 17:40.36 | p3nguin | What do you mean by configure _an_ ITSP for redundancy? |
| 17:40.57 | p3nguin | In the literal sense, you can't configure the ITSP. |
| 17:41.07 | tyman | i'd like to use, say, flowroute for all calls unless it goes down. |
| 17:41.14 | ssureshot | I'm trying to get MOH working with the cisco 7940 phone's hold button,,, works on asterisk 1.2 but not with asterisk 1.8,,, phone firmware is 7-04 |
| 17:41.16 | p3nguin | for outbound calls? |
| 17:41.29 | tyman | yes...sorry |
| 17:41.43 | ssureshot | moh works for meetme and such but when I press the hold button it;'s just silence |
| 17:41.50 | tyman | dont know of stateful tracking like we can with networking |
| 17:42.00 | p3nguin | I'd probably try sequencial Dial() commands. |
| 17:42.15 | p3nguin | sequential, rather |
| 17:42.44 | wcselby | tyman- just add a second Dial() string on your outbound extension that rings your second ITSP. if the first one fails, it will go through tot he second one. if the first one succeeds, it will hang up before it gets to the second one. |
| 17:42.47 | p3nguin | Dial(SIP/flowroute/${EXTEN}) |
| 17:42.52 | p3nguin | Dial(SIP/voipms/${EXTEN}) |
| 17:42.55 | tyman | ok...thought there might be a built-in way...that's what I was thinking but sounded a bit raw |
| 17:43.06 | p3nguin | That seems pretty built-in to me! |
| 17:43.08 | tyman | yes...ok |
| 17:43.11 | wcselby | ssureshot- what does the CLI say when you put the call on hold? |
| 17:43.19 | *** join/#asterisk jkroon (~jkroon@dsl-242-2-151.telkomadsl.co.za) |
| 17:43.30 | jkroon | does MeetMe _still_ depend on dahdi? |
| 17:43.38 | wcselby | jkroon- yes |
| 17:43.41 | jkroon | even though alternative timing methods are now available? |
| 17:43.47 | wcselby | it doesn't depend on it for timing |
| 17:43.51 | wcselby | it depends on it for mixing |
| 17:43.58 | jkroon | o.O |
| 17:44.01 | wcselby | ConfBridge doesn't depend on it |
| 17:44.15 | jkroon | ok, so I should switch to using ConfBridge? |
| 17:44.18 | wcselby | and apparently there's a new ConfBridge in trunk that's the bee's knees |
| 17:44.35 | wcselby | jkroon- only if you have some overriding need to not have DAHDI installed on the system |
| 17:44.40 | jkroon | just trying to evade random memory corruption at the moment. |
| 17:44.50 | wcselby | be sure to evaluate the differences |
| 17:44.55 | wcselby | it's not a drop-in replacement |
| 17:45.24 | p3nguin | ssureshot: Verify your moh works. There's no reason that I can see that moh won't work from your phone just because it's the newer asterisk version. |
| 17:45.35 | jkroon | wcselby, i really would prefer to avoid dahdi. |
| 17:46.23 | wcselby | jkroon- you can install dahdi without having any hardware installed and without using it as a timing source, even though it's anecdotally the best / most stable timing method currently |
| 17:46.37 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 17:46.43 | fullstop | Good afternoon! |
| 17:47.03 | wcselby | o/ fullstop |
| 17:47.10 | jkroon | hmm, no, i find that timerfd actually is more accurate and since I switched a lot of clients that previously complained about voice quality also stopped complaining. |
| 17:48.12 | wcselby | jkroon- i didn't say definitively the best, just anecdotally (I'm not sure I"m spelling that correctly). I've just heard reports that it seems to be better. now i've heard otherwise. :) |
| 17:48.47 | jkroon | clearly has a misunderstanding of the word anecdotally :p |
| 17:49.35 | wcselby | :) |
| 17:49.49 | ssureshot | p3nguin: If I use MusicOnHold(mp3) just after answer MOH works, but it goes straight to MOH,, If I use Set(CHANNEL(musicclass)=(mp3) nothing happens,,, when I press the hold button the cli doesn't act like it receives any input |
| 17:50.17 | fullstop | I don't get the allure of mp3 moh. |
| 17:50.19 | wcselby | ssureshot- you're expecting the hold to be heard on the caller's end, not the person with the 7940, correct? |
| 17:50.34 | jkroon | wcselby: It has been 1000 milliseconds, and we got 50 timer ticks (it never varies, dahdi i often had that it gives 1040 or even 1080 ms) |
| 17:51.04 | p3nguin | mp3 moh allows people to drop in mp3s they's acquired and not make them worry about converting to wav or something else. |
| 17:51.13 | jkroon | fullstop, me neither. but people seem to think that music has to be stored in mp3 :p |
| 17:51.31 | fullstop | p3nguin: Yeah, but that wastes resources more valuable thank disk space. |
| 17:51.32 | p3nguin | s/they's/they've/ |
| 17:51.54 | p3nguin | I personally use all kinds of moh. |
| 17:52.07 | p3nguin | wav, mp3, streaming mp3, streaming ogg |
| 17:52.08 | jkroon | just uses sox to convert to wav and ast file convert to convert to gsm + g729 and then i leave it at that. |
| 17:53.44 | wcselby | jkroon- i've always run res_timing_dahdi, and so far have been lucky I guess. I haven't run into those kinds of issues you mentioned. |
| 17:53.57 | Kobaz | aughh |
| 17:54.20 | jkroon | if i've got physical hardware i agree it's better, but dahdi_dummy vs timerfd ...timerfd any day. |
| 17:54.21 | wcselby | ssureshot- so, show us your musiconhold.conf file please. |
| 17:54.22 | Kobaz | not that i should wait till the last minute anyway, but i specicially remember the last day for the cheapo astricon passes to be july 11 |
| 17:54.54 | wcselby | Kobaz- if you call tmc directly (or wait for them to call you) you can often get a 10-15% discount, no matter how late you call |
| 17:55.04 | Kobaz | mm |
| 17:55.06 | wcselby | or well, I think that latest I called was within a month of the show, and still got the discount |
| 17:55.20 | Kobaz | i was waiting to hear back to see if i got a spot as a backup speaker |
| 17:55.30 | Kobaz | the date was definitly july 11 |
| 17:55.37 | Kobaz | but the price bumped |
| 17:55.46 | wcselby | where is it this year again? |
| 17:55.56 | Kobaz | near denver |
| 17:55.56 | wcselby | somewhere in colorado? |
| 17:57.18 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:57.54 | jkroon | thinks he collected more stack traces on asterisk today than in the three weeks before |
| 17:58.02 | jkroon | and not two of them are the same |
| 17:59.10 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 18:00.26 | jkroon | i'm somewhat suspect of the string_field api, but it seems there has been some work on it recently. |
| 18:00.29 | p3nguin | Well, pewp. I tried using D(:1) in the Dial() rather than using SendDTMF(1) before the Dial()... |
| 18:00.37 | p3nguin | It didn't work out. |
| 18:01.03 | wcselby | p3nguin- it works for me.... are you on 1.8? |
| 18:01.06 | wcselby | or still 1.4? |
| 18:01.22 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:01.29 | p3nguin | The call came in. I picked up the phone, heard "Call fr..." and then a break where I assume my phone sent the 1, then "To accept, press 1." |
| 18:01.42 | p3nguin | The verbose CLI shows -- Sending DTMF '1' to the calling party. |
| 18:01.55 | p3nguin | I guess it sends the tone too quickly. |
| 18:02.25 | p3nguin | The previous method answers the line, waits 2 whole seconds, then sends 1. |
| 18:02.38 | p3nguin | The new method sends quite soon after the line goes up. |
| 18:03.00 | godmachine-x6 | p3nguin:: so i am going to have to make a change to my dial plan ? |
| 18:03.02 | p3nguin | Could I add wait time in the D()? |
| 18:03.07 | wcselby | now _I_ have to try it out....lol. Let me fire up the test box I was using that for.... |
| 18:03.27 | p3nguin | I'll add some w in the D() to see if it helps. |
| 18:04.37 | ssureshot | p3nguin: moh works, I believe...... It isn't working with the hold button on the cisco 7940 phone,, that's where I believe the issue is... The phone doesn't seem to be sending the hold signal back to asterisk |
| 18:04.55 | wcselby | ssureshot- so, show us your musiconhold.conf file please. pastebin it |
| 18:06.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 18:06.24 | felimwhiteley | jkroon: ah hah... solved my issue.. turns out the documentation in book, asterisk wiki and everywhere else fails to mention corosync has a new security mecahnism |
| 18:06.24 | wcselby | hmmm |
| 18:06.28 | wcselby | i didn't have to send a 1 |
| 18:06.31 | wcselby | let me check my dialplan |
| 18:06.34 | *** part/#asterisk PopAlex (~chatzilla@92.86.97.241) |
| 18:06.39 | felimwhiteley | only disvcvoered it due to internet being out all afternoon :) |
| 18:06.45 | felimwhiteley | will write something up about it |
| 18:07.32 | p3nguin | ssureshot: Have you considered upgrading your SIP firmware to 8.11? |
| 18:08.00 | wcselby | p3nguin- here's what I have in my extensions.conf - http://pastebin.com/r1haCkCe |
| 18:08.06 | wcselby | and I wasn't required to dial a 1 |
| 18:08.13 | wcselby | after answering it, I was immeadiately connected |
| 18:08.18 | p3nguin | wcselby: Even using D(:wwww1), which I assume to be two seconds of wait time and then 1, it still does not work for me. |
| 18:08.36 | p3nguin | Are you using forwarding or gtalk method? |
| 18:08.46 | wcselby | this was on asterisk 1.8.4.2 and a polycom 550 |
| 18:08.52 | wcselby | gtalk |
| 18:09.05 | wcselby | I think.... |
| 18:09.10 | wcselby | yeah |
| 18:09.11 | wcselby | gtalk |
| 18:09.21 | p3nguin | When I answer the phone, there is a brief pause, then "Call from..." |
| 18:09.29 | wcselby | i don't get that at all |
| 18:09.29 | godmachine-x6 | 1.4 may not recognize that D() properly |
| 18:09.45 | p3nguin | D is certainly working. |
| 18:10.01 | p3nguin | <PROTECTED> |
| 18:10.43 | wcselby | i've got it set to forward calls to my google chat |
| 18:11.02 | wcselby | and i've even got call screening set to On |
| 18:11.05 | wcselby | in the gv settings |
| 18:11.09 | ssureshot | p3nguin: yes I have considered, and am still considering provided that is the issue,, I guess Ill have to put a phone under support to get the firmware I am running 7-04 right now |
| 18:11.26 | ssureshot | my musiconhold.conf is http://pastebin.com/RmR7JQ0R |
| 18:12.02 | p3nguin | ssureshot: You can always "find" the firmware online and avoid smartnet contracts. |
| 18:12.03 | ssureshot | moh plays when I enter the meetme app |
| 18:12.41 | wcselby | but you're not seeing anything in the CLI (up your verbosity to 10) about starting music on hold when you press the hold button? |
| 18:12.50 | ssureshot | right on,, friend |
| 18:12.53 | wcselby | that was for ssureshot ^^ |
| 18:13.36 | ssureshot | wcselby: correct nothing displays in the cli for the hold button |
| 18:13.52 | wcselby | grab a sip debug of the call |
| 18:19.12 | p3nguin | ssureshot: A google search for the file name will absolutely get you the firmware. The file name is P0S3-08-11-00.zip :) |
| 18:19.37 | wcselby | hey p3nguin |
| 18:19.43 | wcselby | i just tried it again and got that prompt |
| 18:19.44 | p3nguin | That's as much as I can do, short of giving it to you against the rules. |
| 18:20.08 | p3nguin | wcselby: I got the prompt several tries in a row. |
| 18:20.13 | ssureshot | p3nguin: ha, I hear ya.. |
| 18:20.27 | wcselby | i called right back and did not get the prompt again |
| 18:20.43 | ssureshot | wcselby: am I looking for anything specific in this sip debug log? |
| 18:21.44 | wcselby | ssureshot- i just tested it |
| 18:21.48 | p3nguin | I quit using SIP on my Cisco phones some time ago, so I can't test moh with SIP 7.4 easily. |
| 18:22.05 | wcselby | and my phone sent an INVITE back to asterisk the second I hit the hold button |
| 18:22.13 | wcselby | i'm using a polycom 550 for testing though |
| 18:22.24 | wcselby | the only cisco phone i have is a 7941 running the 8.5.2 firmware |
| 18:22.31 | _Corey_ | I can confirm 7.4 was buggy |
| 18:22.39 | _Corey_ | I wouldn't advise using it |
| 18:22.54 | p3nguin | I have a feeling he'll upgrade to 8.11 pretty soon. |
| 18:23.21 | _Corey_ | It's REALLY old |
| 18:23.24 | ssureshot | p3nguin: yes right now I assume,, this is a replacement upgrade so might as well bring it current |
| 18:24.05 | p3nguin | The reason I didn't suggest 8.12 to you is because of the caller ID bug in 8.12. |
| 18:24.36 | _Corey_ | Most of the Cisco phones I have around here still alive are on 8.8 |
| 18:24.42 | ssureshot | that was actually my next question... |
| 18:25.36 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 18:27.20 | wcselby | p3nguin- the deal was though, his 7940 would probably start playing hold music on his last asterisk install, just not with this new asterisk install, correct? |
| 18:27.44 | wcselby | now, that's not saying that asterisk responds different now than the old version does, and his old phone doesn't know how to handle this |
| 18:28.37 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:756a:18b4:df3a:9829) |
| 18:29.19 | ssureshot | wcselby: my old version is 1.2 going to 1.8 but the firmware stayed the same,, I assume that things have changed in the sip protocol that maybe caused this? |
| 18:29.29 | ssureshot | firmware for the phone |
| 18:29.39 | wcselby | ssureshot- which is why I suggested you grab a sip debug |
| 18:30.00 | wcselby | when I just tested doing exacly that, the phone sent an invite to asterisk, asterisk responded, and then put the call on hold |
| 18:30.03 | ssureshot | wcselby: I've got that you want a pastebin? |
| 18:30.14 | wcselby | or well, started playing music on hold |
| 18:30.17 | wcselby | oh, please share it |
| 18:30.25 | wcselby | i must have missed it earlier, sorry |
| 18:30.43 | ssureshot | np at all I just asked what to look for as it's all jibberish lol |
| 18:31.24 | wcselby | hey, totally unrelated to anything we're talking about here, but - YAY, RAIN! |
| 18:31.52 | p3nguin | I'd kind of like some rain here today. It's already 98 degrees outside. |
| 18:32.08 | p3nguin | ssureshot: Your next question was going to be why I didn't go to 8.12? |
| 18:32.23 | p3nguin | _corey_: Any known improvements between 8.8 and 8.11? |
| 18:32.36 | ssureshot | p3nguin: exaclty |
| 18:32.40 | _Corey_ | p3nguin: :) no idea |
| 18:33.54 | wcselby | ssureshot- do you have that pastebin for me to look at? |
| 18:34.19 | ssureshot | http://pastebin.com/bCBDruBE |
| 18:34.45 | ssureshot | wcselby: I pressed the hold button quite a few times in there |
| 18:39.17 | wcselby | ssureshot- that's how my sip debug reads too (without all the extra fluff) |
| 18:39.46 | wcselby | except that right at where, on your paste would be line 615, I have my system telling me it's started music on hold |
| 18:40.47 | wcselby | the sip packets are exactly the same |
| 18:40.53 | felimwhiteley | jkroon: in case you do every try the ais cluster :) https://felimwhiteley.wordpress.com/2011/07/11/asterisk-invalid-ipc-credentials-or-how-no-interwebs-saved-the-day/ |
| 18:41.00 | ssureshot | hmm,, wonder wehre the damage is then |
| 18:41.02 | wcselby | well, not exactly |
| 18:41.04 | wcselby | but close enough |
| 18:41.31 | wcselby | what's your CLI verbosity set to? |
| 18:41.55 | ssureshot | I had it set to something over 10 on that one... |
| 18:42.10 | ssureshot | actually it was 10 |
| 18:42.28 | wcselby | wait |
| 18:42.36 | wcselby | you're setting your musiconhold to something called random |
| 18:42.40 | wcselby | to a class called random |
| 18:43.57 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 18:44.16 | wcselby | please pastebin your extensions.conf that deals with a call going to this extension |
| 18:44.29 | ssureshot | yeah sorta,, I had changed all that and have restored,, let me just copy the mp3 class as the random class also |
| 18:45.22 | wcselby | but in the musiconhold.conf you pb'ed earlier, there was no random class |
| 18:46.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:47.10 | ssureshot | right, there is now and same thing,, I'm trying to put together the dial plan for this user and get that right up give me a few |
| 18:48.11 | p3nguin | ssureshot: Verbosity does not improve above level 4, so 10 or more than 10 will still show the maximum verbosity available. |
| 18:48.29 | ssureshot | ah good to know |
| 18:48.44 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 18:49.07 | p3nguin | I keep waiting for an easter egg to be put in at some greater random verbosity level. |
| 18:49.54 | leifmadsen | heh nope :) |
| 18:50.15 | leifmadsen | I think there is something in one of the modules at like verbose 7, or something |
| 18:50.28 | p3nguin | New in 1.8, I guess, huh? |
| 18:50.56 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 18:51.19 | Joe_CoT | any idea on why sound wouldn't work when using a queue? The person calling into the queue can't hear the agent. other calling works fine |
| 18:51.33 | p3nguin | Now I'm going to have to test every verbose level between 5 and 2147483647 to find it! |
| 18:51.47 | wcselby | lol @ p3nguin |
| 18:52.06 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-235.mobile.uci.edu) |
| 18:52.11 | wcselby | i thought verbosity ended at like 53565? |
| 18:52.25 | ssureshot | wcselby: http://pastebin.com/Lyn0SC1x .. |
| 18:52.27 | p3nguin | I'm also wondering if that max level is the limitation of a 32-bit system or if it would be far higher on a 64-bit. |
| 18:52.45 | p3nguin | s/or/and/ |
| 18:53.44 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 18:53.51 | wcselby | ssureshot- can you do moh show classes and moh show files also? |
| 18:53.54 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:54.12 | p3nguin | I moved to a 32-bit box as soon as it was ready for production. When I changed to a new box, it's still 32-bit. |
| 18:54.27 | *** join/#asterisk rabby (~rabby@mnch-4d0db280.pool.mediaWays.net) |
| 18:54.44 | p3nguin | Oh, I might have a 64-bit Asterisk to check it on. |
| 18:55.35 | ssureshot | wcselby: here ya go,, http://pastebin.com/yupkSWaH |
| 18:55.56 | p3nguin | Nope, all of them I have access to are i686. |
| 18:58.41 | wcselby | ssureshot- i'm not sure what to say.... |
| 18:59.00 | wcselby | ssureshot- everythign looks right |
| 18:59.52 | wcselby | ssureshot- you're running asterisk 1.8.4.4? |
| 19:00.44 | ssureshot | yes asterisk 1.8.4.4 |
| 19:00.55 | wcselby | let me update my system to that and test again |
| 19:01.18 | wcselby | while I do that, can you test with just the 'default' moh class? |
| 19:01.20 | ssureshot | I just upgraded the firmware let me test that and see before you go through all that trouble lol |
| 19:02.30 | ssureshot | I will also test with the default class |
| 19:03.39 | wcselby | heh, it's not trouble. it's just a test system |
| 19:08.01 | *** join/#asterisk cerberus_za (~coert@196-215-58-254.dynamic.isadsl.co.za) |
| 19:09.09 | *** join/#asterisk Welagor (~Balthazar@AAmiens-154-1-78-24.w86-208.abo.wanadoo.fr) |
| 19:11.10 | ssureshot | wcselby: well that settles it I guess, the firmware upgrade for he phones fixed the issue.. I |
| 19:11.22 | wcselby | ssureshot- well, good to know :) |
| 19:11.50 | ssureshot | thank you much for helping out,, |
| 19:12.05 | ssureshot | p3nguin: thanks to you also |
| 19:12.06 | wcselby | no worries, it was p3nguin and _Corey_ who really came up with the firmware idea. :) |
| 19:12.28 | ssureshot | yeah and thanks to _Corey_ also :) |
| 19:13.20 | ssureshot | I'm sure Ill have more questions now that I can move past that lol... I believe that I'm about there for the drop in replacement ythough |
| 19:18.14 | ssureshot | wcselby: one more question on moh,, Documentation says that "SetMusicOnHold()" is depriciated so should I be using this then ? Set(CHANNEL(musicclass)=(mp3) |
| 19:19.49 | godmachine-x6 | p3nguin:: did you figure out why your calls wasn't sending the 1 correctly? |
| 19:19.58 | wcselby | yeah, probably. i think you're parens are off....maybe Set(CHANNEL(musicclass)=mp3), I would think. |
| 19:21.03 | p3nguin | godmachine-x6: Nope. It simply does not work for me. I am able to consistently reproduce the failure. |
| 19:21.15 | godmachine-x6 | try to call me let me see if it works ok for me |
| 19:21.27 | godmachine-x6 | you wont be able to hear me say hello from this machine (no mic) |
| 19:21.32 | godmachine-x6 | but i can tell if it works or not |
| 19:21.39 | p3nguin | In a bit, I'll call you. Busy right this minute. |
| 19:21.57 | godmachine-x6 | ok no prob |
| 19:22.08 | p3nguin | If I don't do it after a while, remind me. |
| 19:22.23 | irroot | ~beer p3nguin |
| 19:22.23 | infobot | ACTION pours a pint of La Maudite for p3nguin |
| 19:22.25 | godmachine-x6 | maybe i can find someone else to |
| 19:22.43 | godmachine-x6 | i might forget myself until i actually need to receive a call on here lol |
| 19:22.54 | ssureshot | roger |
| 19:24.48 | wcselby | oh jeez |
| 19:25.02 | wcselby | a Mad Max reboot? |
| 19:25.02 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:26.15 | malcolmd | if michael bay's involved, i'm there!!! ;) |
| 19:27.37 | wcselby | http://www.imdb.com/title/tt1392190/ |
| 19:27.37 | wcselby | Director: |
| 19:27.37 | wcselby | George Miller |
| 19:30.12 | wcselby | I guess he was the directory of the original movies |
| 19:34.00 | ssureshot | I love Mad Max.... though remakes always ruin classics.. but as long as Charlize Theron is in it I'm in lol |
| 19:42.06 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 19:49.26 | *** join/#asterisk iscario (~Miranda@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 19:53.51 | iscario | Hello, |
| 19:53.51 | iscario | I have a problem with the application MixMonitor, it won't record and I can't find why... |
| 19:53.52 | iscario | Just to explain a bit the context : I'm doing some tests to provide users a feature which allow them to record the call. I have set up an asterisk server on an OpenBSD machine (probably the OS i will have to use, that is why...). By now , i'm just trying to use MixMonitor each time i call anyone else with the Dial application. |
| 19:53.52 | iscario | Could you please help me to understand what is wrong in my dialplan ? http://pastebin.com/9HR3uxVP |
| 19:53.52 | iscario | You'll notice i did several tests, and in the end I gave you the output when i tried to create a single file named "ok.wav" in the directory asterisk wanted (the default one for MixMonitor, ie /var/spool/asterisk/monitor/ in my case) |
| 19:53.52 | iscario | Here is the debug output: http://pastebin.com/C3YUrHRt |
| 19:53.52 | iscario | thank you for your help! |
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| 20:22.44 | wcselby | iscario- it says it's begun and ended the mixmonitor recording....is there nothing in the /var/spool/asterisk/monitor folder? |
| 20:23.41 | iscario | no wcselby |
| 20:24.19 | wcselby | there's nothing in there, or just not a filename named ok.wav? |
| 20:24.45 | iscario | find / -name "ok.wav" does not return anything |
| 20:25.51 | wcselby | try removing the b option from your mixmonitor command and try again |
| 20:26.46 | p3nguin | Yeah, if you never bridge the call, there is nothing to record. |
| 20:27.01 | iscario | I noticed that the recording work when i do not use the flag "b" (eg if I use the function Playback). In this case, I see in the log "MixMonitor closed stream ...." |
| 20:27.34 | iscario | But i need to use the flag "b" when i call the dial function... |
| 20:28.11 | wcselby | .... |
| 20:28.23 | Joe_CoT | any idea on why sound wouldn't work when using a queue? The person calling into the queue can't hear the agent. other calling works fine |
| 20:28.27 | wcselby | try removing the b option from your mixmonitor command and try the call again and see if anything is recorded |
| 20:28.56 | p3nguin | Unless you can come up with a very good reason that you MUST have the b option get rid of it. |
| 20:29.08 | wcselby | Joe_CoT- can the agent hear the caller? what exactly do you mean by "other calling works fine". Is this an internal call or a call from outside the system? |
| 20:29.14 | p3nguin | Use MixMonitor(), then Dial() |
| 20:29.20 | p3nguin | no b option to MixMonitor |
| 20:29.32 | p3nguin | It will record when the channel goes Up. |
| 20:30.15 | Joe_CoT | wcselby, I actually got it further down to only being an issue with the intercom. If the intercom calls a single person, works fine. If I call a queue from my phone, works fine. If the intercom calls the queue, the agent can hear, but the intercom can't hear the agent. |
| 20:30.34 | wcselby | what intercom? |
| 20:31.18 | iscario | oh, really ? so when is the case when am i supposed to call the b flag ? I maybe did not understand what a bridge is.... |
| 20:31.44 | p3nguin | Bridged is when two legs of the call are joined toghether. |
| 20:32.14 | p3nguin | Without the b option, the recording will start when the channel becomes state "Up" as opposed to waiting for two legs to get hooked together. |
| 20:32.43 | p3nguin | There's not a lot of difference in how the resulting recording will sound. |
| 20:32.59 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 20:33.52 | iscario | ok. Do you why it was working with the "b" flags set in the dialplan in 1.6.10.11 p3nguin ? (and now it does not...) |
| 20:33.59 | p3nguin | no clue |
| 20:34.23 | p3nguin | If you had given me a reason that you require the b option, I might have given more thought as to why it would be messed up now. |
| 20:35.49 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 20:36.42 | iscario | p3nguin: i just followed a tutorial, and considering it was working before, I thought it would work. So there is no reason to use the "b" flag if it work without. I'll try asap ;) Thx for your help |
| 20:37.01 | p3nguin | It was wcselby's idea to take out the b option. |
| 20:37.26 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:37.29 | p3nguin | I just had to expand on the idea. |
| 20:37.43 | wcselby | :) |
| 20:39.16 | iscario | thx wcselby ;) |
| 20:39.47 | iscario | i'll be back tomorrow if it does not work^^ |
| 20:40.33 | wcselby | o/ |
| 20:41.08 | Joe_CoT | wcselby, this intercom http://www.cyberdata.net/products/voip/digitalanalog/intercom/index.html |
| 20:41.39 | Joe_CoT | it's a pain in the ass, and I don't understand why a direct call is working, but a call in the queue is not. I did an rtp debug of the call in the queue, it's getting the sound packets |
| 20:42.41 | wcselby | why is an intercom calliing into a queue? you've given a lot of random details, but no complete picture |
| 20:43.23 | Joe_CoT | OK. We've got an intercom at the door. There's a button on the intercom. When you hit the button, it goes into a queue. That queue calls everyone in the office |
| 20:44.06 | Joe_CoT | Then whoever picks up the phone can talk to whoever's at the door, type in a code, and let them in |
| 20:45.09 | wcselby | And it goes into a queue instead of just ringing multiple sip phones with a big dial statement because.....? |
| 20:45.52 | Joe_CoT | well, separately I've been having problems on 1.8 where ringing everyone normally pegs asterisk at 200% cpu and the server dies. I'm not sure why. One thing at a time, though. |
| 20:46.03 | wcselby | hahaha |
| 20:46.05 | wcselby | gotcha |
| 20:46.29 | wcselby | can you provide a sip trace of the intercom call into the queue? and dialplan and queues.conf snippets? |
| 20:47.28 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:48.03 | Joe_CoT | heh. I've got the debug output from the console, I didn't do SIP debug. and if I give you snippets from the dialplan, that's probably about the point I'd get yelled at for using FreePBX |
| 20:49.12 | wcselby | lol |
| 20:49.50 | wcselby | freepbx makes things much more difficult to debug |
| 20:49.59 | wcselby | i guess let's start with the sip debug |
| 20:50.04 | wcselby | and go from there |
| 20:51.05 | Joe_CoT | ok, so by sip debug you mean the output from sip set debug, right? |
| 20:51.59 | wcselby | yeah, filtered by the intercom and whichever phone picks it up |
| 20:52.49 | Joe_CoT | ok, sec |
| 20:56.04 | iscario | well, while i'm here... Is it really hard to use SIP&Asterisk with NAT ? My ideal use would be : users with SIP softphones on a lan connected to Internet behind a NAT, and an Asterisk server connected directly to internet or behind a different NAT. |
| 20:56.21 | *** join/#asterisk TimeRider (steve@5ace69b0.bb.sky.com) |
| 20:56.38 | wcselby | iscario- i do that here at my house |
| 20:57.07 | wcselby | two phones, both behind the same NAT, connect to my asterisk server which is sitting on the internet without NAT |
| 20:57.15 | wcselby | ~sipnat |
| 20:57.15 | 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, canreinvite, externhost or externip, and localnet. |
| 20:57.29 | wcselby | there's some good resources for getting SIP and NAT and Asterisk to all play good together |
| 20:57.57 | p3nguin | SIP and NAT is quite easy to configure, and it will work without problems as long as your NAT device doesn't suck too badly. |
| 20:58.42 | Joe_CoT | wcselby, ok, let's try this: http://pastebin.com/qkB5eeyv |
| 21:00.47 | Joe_CoT | hmm, i see i got 1033, don't know if I got 3055's packets. Is there a way to set sip set debug peer to 2 different peers? |
| 21:01.45 | *** join/#asterisk wonderworld (~ww@port-92-201-126-230.dynamic.qsc.de) |
| 21:01.54 | p3nguin | Does anyone here happen to know how I can confirm that a volume with ext4 does or does not have journaling on it? |
| 21:02.44 | WIMPy | dumpe2fs? |
| 21:04.28 | p3nguin | The string "journal" does not appear in the output of dumpe2fs /dev/sda1, so is it safe to assume it does not have journaling enabled? |
| 21:05.10 | p3nguin | I remember having the intention of turning off journaling when I formatted it, but now I'd like to ensure that I did in fact disable the journal. |
| 21:05.17 | WIMPy | yes |
| 21:06.15 | WIMPy | has_journal should be under features, as well as several other Journal entries. |
| 21:06.51 | wcselby | Joe_CoT- i notice that the intercom is on a 192.168.x.x network, and the phone that answers is on a 10.10.x.x network......correct? |
| 21:07.50 | wcselby | Joe_CoT- do you ahve all your nat settings correct? |
| 21:07.56 | p3nguin | I ran the same command on a volume that is known to have a journal, and it has plenty of journal information... so I guess I got my answer. |
| 21:09.10 | Joe_CoT | wcselby, The intercom is 10.10.10.83, the phone is 10.10.10.46. 192.168.2.2 is the asterisk server. If the nats settings are broken, I don't know how it works for calls that aren't the queue |
| 21:09.43 | wcselby | oh well i probably just misread the sip debug |
| 21:09.48 | wcselby | which is possible |
| 21:09.55 | wcselby | freepbx debug / cli makes my head hurt |
| 21:10.19 | wcselby | so, have you tried creating a ring group (I think that's what fpbx calls it) instead of a queue? and is that what's taking up 200% cpu? |
| 21:10.49 | Joe_CoT | yes and yes. I started with a ring group |
| 21:10.55 | p3nguin | tune2fs -l /dev/sda1|grep features |
| 21:11.00 | p3nguin | tune2fs -l /dev/sdc3|grep features |
| 21:11.12 | p3nguin | One shows "has_journal" and the other does not. |
| 21:11.28 | p3nguin | Perfect. That's exactly what I needed to find out. |
| 21:11.49 | p3nguin | sda1 is a flash module, so I just wanted to double or triple check that journaling was turned off. |
| 21:12.04 | iscario | wcselby ; p3nguin : let's say i do not know how the nat for the softphone client is configured ( to explain the context, my idea would be to allow a technical people working for my company to go to a client company [a company which has a lan accessing Internet behind NAT ;) ] and to be able to call my Asterisk server with his sip softphone ). I guess it is not that easy because i do not know anything (before going there) |
| 21:12.49 | p3nguin | It is as easy as I already expressed. |
| 21:13.05 | p3nguin | Always always configure the NAT stuff in Asterisk for the phones. Never never configure the phones for NAT. |
| 21:13.05 | WIMPy | p3nguin: dumpe2fs -h does the same. Did you set noatime as well? |
| 21:13.25 | p3nguin | /dev/sda1 / ext4 rw,noatime,barrier=1,data=writeback 0 0 |
| 21:14.44 | wcselby | iscario- what p3nguin said is correct. take his statement to heart: <p3nguin> Always always configure the NAT stuff in Asterisk for the phones. Never never configure the phones for NAT. |
| 21:14.48 | WIMPy | Should be as far as you get with rw. |
| 21:15.44 | *** join/#asterisk TimeRider (steve@5ace69b0.bb.sky.com) |
| 21:16.00 | wcselby | and now, I must go eat. Joe_CoT I'm sorry, I'm not sure I can figure it out in this context. you may want to try #freepbx or whatever their support channel is. |
| 21:16.05 | wcselby | o/ |
| 21:16.10 | Joe_CoT | ok, thanks anyway |
| 21:18.25 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 21:18.52 | iscario | thanks p3nguin, i'll try then ; ) |
| 21:19.10 | p3nguin | I know we discussed this before, but I don't set it ro because I want to save recordings and, albeit minimal, save logs. What other options are there for saving files while keeping the flash module mounted ro? |
| 21:20.15 | WIMPy | Use two of them. Or one flash and one real disk. |
| 21:21.32 | WIMPy | If you use two flash devices you can at least ensure the system disk will work, while the data disc can fail. |
| 21:21.51 | p3nguin | It's a pretty small appliance, so there is no room for a second disk in addition to the primary DoM. I also don't really like the idea of an external hdd connected. |
| 21:22.22 | WIMPy | Network storage? |
| 21:24.10 | p3nguin | That's only slightly more possible than the best option of having an additional disk. |
| 21:24.46 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 21:24.51 | p3nguin | hammer |
| 21:24.58 | p3nguin | a really big hammer |
| 21:25.29 | Qwell | Katty: INSERT COIN |
| 21:26.44 | p3nguin | Filesystem state: not clean |
| 21:26.57 | p3nguin | I guess it needs a bath. |
| 21:35.45 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 21:36.58 | anonymouz666 | "insert coin" hehe |
| 21:37.12 | anonymouz666 | I remember my old times playing street fighter II |
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| 21:48.57 | *** part/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 21:51.37 | jkroon | Error loading module 'ÀLí': /usr/lib64/asterisk/modules/ÀLí.so: cannot open shared object file: No such file or directory <-- i'm guessing the fact that the module I requested (via cli) be loaded has a different name would indicate a memory corruption bug? |
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| 22:01.32 | Bob_Pierce | I'm running 1.8.4.3 - we're seeing issues where in the middle of some calls the caller from the pstn can't hear the caller attached to the Asterisk system - then after a few seconds of "can you hear me now", the PSTN side hears audio from the asterisk side again. The whole time, the asterisk side can hear the PSTN caller fine. We have a SIP trunk to a MetaSwitch which is our PSTN connection. We've been working for a while and haven't come up with mu |
| 22:02.27 | *** join/#asterisk rutski (~rutski@96.56.54.186) |
| 22:02.30 | rutski | hey all |
| 22:02.42 | rutski | I keep getting this when trying to dial out: http://codepad.org/UZWikcM2 |
| 22:02.58 | rutski | I'm not sure if I have the "DAHDI/G1" part right |
| 22:03.09 | rutski | but the phone lines definitely aren't busy/congested |
| 22:03.14 | rutski | so I don't quite know what to do here :-/ |
| 22:03.25 | Qwell | Are they configured in group 1? |
| 22:04.14 | rutski | I think so? http://codepad.org/8bWbOmGJ |
| 22:04.29 | rutski | oh, group 0 |
| 22:09.21 | Bob_Pierce | It seems the jitter and packet delay is mot predominately seen on calls where the caller enters one of our queues, listens to the MOH for a little while and then is connected to an agent. Here's the scenarios we've tested http://codepad.org/3b2tHGcS |
| 22:10.21 | Bob_Pierce | This is a major issue for us, but particularly frustrating to track down since it is not occurring on every call. |
| 22:11.28 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:11.36 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:13.53 | Bob_Pierce | It seems the problem with the dropped audio is on the audio path going from the Asterisk system to the MetaSwitch |
| 22:15.16 | Bob_Pierce | I understand it might be the end of the day for most people. Should I come back and ask this question again tomorrow? |
| 22:20.59 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 22:26.40 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 22:27.13 | jeffspeff | what is the name of the adapter that's similar to vonage and does rj-45 to rj-11 for SIP ? |
| 22:27.33 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 22:28.34 | _Corey_ | jeffspeff: Most common is Linksys PAP2-NA |
| 22:29.55 | jeffspeff | thanks, i remember something that looked more similiar to a magic jack / vonage type thing |
| 22:30.12 | _Corey_ | There are others, though that's the one I'd pick |
| 22:30.40 | jeffspeff | why's that? |
| 22:30.41 | _Corey_ | Vonage used to use them |
| 22:30.55 | _Corey_ | Cheap/easy/plentiful |
| 22:31.07 | _Corey_ | solid firmware |
| 22:31.34 | jeffspeff | ok, thanks |
| 22:32.34 | _Corey_ | no prob |
| 22:51.45 | *** join/#asterisk dfamorato (~dfamorato@2001:470:5:630:cabc:c8ff:fee3:9c4b) |
| 22:55.01 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 23:04.15 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 23:08.28 | *** join/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 23:10.26 | batfastad | Hi everyone. We have a hosted Asterisk/FreePBX system currently in an OpenVZ container. Virtualised performance seems great to us. Only a 15 extension system. I'm looking to cut costs and possibly set it up on as a Xen guest on our co-located server. |
| 23:10.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:10.35 | batfastad | Anyone have experiences with Asterisk/FreePBX under Xen? Using Xen's kernel-level virtualisation, which I believe is similar to OpenVZ. Would it work as well as on OpenVZ? No special telephony hardware/adaptors, a SIP provider converts from PSTN so it's all IP. |
| 23:13.20 | p3nguin | What is the assumed memory sized for the LOW_MEMORY compile flag? Is it a seriously low value that this flag is intended for? |
| 23:16.50 | *** join/#asterisk pabelanger (~pabelange@c-71-207-215-147.hsd1.al.comcast.net) |
| 23:16.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:26.55 | WIMPy | Oh, great. |
| 23:27.05 | WIMPy | Someone DOSed me by using IAX. |
| 23:27.32 | thehar | lol |
| 23:37.02 | WIMPy | Wouldn't it be great if Asterisk at least crashed instead of just sitting there doing nothing? |
| 23:37.23 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-231.ips.direcpath.com) |
| 23:42.39 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:46.53 | KavanS | WIMPy, that one line says it all lol |
| 23:56.47 | *** part/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 23:58.09 | p3nguin | How low should my system memory be before I need to worry about using the LOW_MEMORY compile flag? |
| 23:58.19 | *** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 23:58.24 | WIMPy | Interestingly it crashed when I did 'asterisk -rx "core show locks"'. Maybe I should put that in to a cron job? |
| 23:59.01 | p3nguin | Will it put any additional load on the system to do it every few minutes or even as often as every minute? |
| 23:59.09 | WIMPy | When your kernel invokes the oom-killer it's too late. |
| 23:59.48 | godmachine-x6 | p3nguin:: just whenever you get a chance try that test call for me. told me to remind ya |
| 23:59.57 | WIMPy | I don't think I could notice. |
