| 00:04.24 | manxpower | Um, where are all the idiots? |
| 00:04.45 | leifmadsen | points up :) |
| 00:05.09 | manxpower | 90 mins and not even a FreePBX question or a question easily answered by reading the docs. |
| 00:05.29 | carrar | My Asterisk browser is not working |
| 00:05.35 | carrar | pls fix thanks |
| 00:06.20 | carrar | and how to I make these soundcards function like a ATA |
| 00:07.06 | carrar | I have solder |
| 00:11.46 | ChannelZ | Halp me, when I dial my peer my ITSP rings and sounds like it's congested, what button do I click?????!??!!11! |
| 00:12.03 | p3nguin | only if you put out |
| 00:12.25 | *** join/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 00:14.44 | manxpower | I should have kept quiet. |
| 00:22.24 | carrar | Thats against IRC regulations |
| 00:26.41 | hardwire | fun cop |
| 00:27.24 | manxpower | OT: http://imagebin.ca/view/yrYGGLGJ.html |
| 00:28.11 | manxpower | If Katty can do it.... |
| 00:30.52 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 00:31.16 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:39.30 | *** part/#asterisk tvaughn (tyler@66.55.71.171) |
| 00:59.58 | *** join/#asterisk diegomad (mad@186.81.147.229) |
| 01:10.15 | *** join/#asterisk Faithful (~Faithful@119.12.29.20) |
| 01:10.50 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 01:12.01 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 01:16.46 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 01:19.13 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 01:22.36 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 01:23.41 | *** join/#asterisk Jed (~Jed@ool-4574c525.dyn.optonline.net) |
| 01:23.56 | Jed | hello |
| 01:24.17 | PeterHup | Hi Jed |
| 01:24.35 | Jed | Hey, I'm trying to figure out a way to disable call waiting in asterisk |
| 01:24.51 | Jed | I know in trixbox *71 works, but on asterisk it does not. |
| 01:25.04 | Jed | Do I need to enable something for that? |
| 01:26.01 | *** part/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 01:27.09 | Jed | I've tried a few work arounds, like setting call limits to 1, but then was unable to transfer calls. |
| 01:30.33 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 01:30.38 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 01:31.24 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:32.45 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 01:44.03 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:44.44 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:51.28 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:52.26 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:07.49 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 02:09.34 | *** join/#asterisk bmg505 (~leon@196-209-71-68-rndf-esr-3.dynamic.isadsl.co.za) |
| 02:12.31 | *** join/#asterisk CatLynx (~booga@173-11-77-182-SFBA.hfc.comcastbusiness.net) |
| 02:13.46 | CatLynx | Anyone know how the mwi works on the Analog adaptor for the FXS port? |
| 02:14.12 | CatLynx | as in, what does the FXS port do to trigger the WMI on the phone? is it like CID? |
| 02:15.53 | CatLynx | wonders if everyone is alseep now. |
| 02:16.21 | CatLynx | still finding new issues with his TDM410 card |
| 02:22.21 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 02:23.51 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 02:23.58 | *** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 02:33.07 | *** join/#asterisk maxagaz (~maxagaz@222.128.36.151) |
| 02:34.35 | manxpower | CatLynx: Yes |
| 02:34.53 | manxpower | Assuming you are not using some bizzare hotel type phone with hotel type mwi |
| 02:35.27 | manxpower | CatLynx: I would have to look up the specs for VMWI |
| 02:35.34 | manxpower | that's what you want, right? |
| 02:36.07 | manxpower | Jed: correct. *71 is trixbox. |
| 02:36.19 | manxpower | Set it up however you want in Asterisk. |
| 02:36.49 | Jed | Whats the syntax for it? |
| 02:37.01 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:37.17 | manxpower | Jed: however you write it. |
| 02:37.36 | manxpower | Jed: do you have Asterisk? |
| 02:37.39 | Jed | yes |
| 02:37.45 | manxpower | Good. Now. What kind of phone do you have? |
| 02:37.45 | Jed | 1.4.29 iirc |
| 02:37.53 | Jed | cisco 7940 |
| 02:38.13 | manxpower | Good. Now disable the call waiting on the phone or write a complex set of dialplan macros to do it for you. |
| 02:38.28 | manxpower | Perhaps you should go back to FreePBX/Trixbox? |
| 02:38.36 | Jed | Well, i tried from the phone, but it doesn't actually seem to disable anything |
| 02:38.51 | manxpower | Jed: YOU HAVE TO WRITE THE CODE TO DISABLE CALL WAITING IN THE ASTERISK DIALPLAN |
| 02:38.58 | manxpower | or you can edit the phone config files and disable it there. |
| 02:39.37 | spenguin[work] | hands manxpower a drink |
| 02:39.55 | Jed | manxpower: Yes, I know I have to write out the dial plan but I'm not really sure on how to do it |
| 02:40.08 | manxpower | Maybe not all that complex, actually. Use AstDB to store the CW state, then use ChanIsAvail to determine of the channel is in use, then determine what to do with the call. |
| 02:40.23 | manxpower | Jed: Right. We are not here to teach you dialplan stuff. You should start with reading the Asterisk book |
| 02:40.24 | manxpower | ~book |
| 02:40.25 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:40.26 | manxpower | ~toolkit |
| 02:40.27 | infobot | hmm... toolkit is Remember Asterisk isn't really a PBX. Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 02:41.14 | manxpower | look at "core show application chanisavail" |
| 02:41.20 | jaytee | a toolkit that's missing a 6mm hex wrench |
| 02:41.37 | manxpower | jaytee: naw, a toolkit missing some poor programmer to write it for you |
| 02:41.50 | Jed | I was actually looking into devstate |
| 02:42.13 | manxpower | With devstate I think you need to set up asterisk to track device states. MUCH more complex |
| 02:42.49 | Jed | Ok, I think you've pointed me in the right direction with chanisavail, which is what I was trying to ask I guess |
| 02:42.51 | Jed | Thanks |
| 02:42.56 | Jed | I will look into it |
| 02:43.38 | manxpower | Jed: no, you need to read the Asterisk book |
| 02:44.24 | Jed | I've started to |
| 02:47.46 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 02:53.26 | manxpower | Jed: you should do "core show applications" |
| 02:55.07 | Jed | ok |
| 02:56.07 | Jed | I've written basic dial plans before and some interactive auto attendants, nothing that complex |
| 02:57.18 | manxpower | really, it is better to leave it disabled in the phone itself (I don't recall the Cisco provisioning stuff) |
| 02:57.36 | manxpower | let the 2nd call roll over to the 2nd line if you must, but leave call waiting off. |
| 02:58.17 | Jed | Well, the issue seems to be the call waiting beep is cutting out part of the conversation |
| 02:58.37 | Jed | I don't if its the phone or asterisk thats causing this |
| 03:00.34 | manxpower | it is the phone |
| 03:00.50 | Jed | Is this a known cisco phone issue? |
| 03:01.09 | manxpower | this is an issue with all phones I'm aware of, including POTS service from the telephone company |
| 03:02.02 | manxpower | I never could understand "call waiting" being a feature of a multiline phone |
| 03:03.12 | patrb | grrr...asterisk 1.6.1.18 cdr_mysql doesnt seem to be working correctly...the cdr_addon_mydql module is loaded and the config is correct....but under 'registered backends' no mysql |
| 03:03.27 | patrb | can anyone give me some suggestions? I have it working in 1.4 |
| 03:04.20 | patrb | the cdr_mysql module was compiled with asterisk-addons-1.6.12 |
| 03:04.24 | patrb | err 1.6.1.2 |
| 03:05.39 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 03:12.34 | CatLynx | is back |
| 03:13.05 | CatLynx | Hi Max, so I am testing a TDM410 with a AT&T coredless phone and the WMI not working for that phone on the FXS port. |
| 03:13.24 | manxpower | CatLynx: when you ho off hook do you get a stutter dialtone? |
| 03:13.30 | manxpower | s/ho/go |
| 03:13.56 | CatLynx | I take the cordless phone and plug it in to a SPA-3102 the wmi works |
| 03:13.59 | manxpower | MWI is the stutter dialtone. VMWI is a blinking light on the phone. |
| 03:14.17 | manxpower | I'm waiting for the answer to my question |
| 03:14.20 | CatLynx | I get a stutter dialtone on the TDM410 |
| 03:14.39 | CatLynx | I plug the panasonic phone in to the TDM410 the WMI works |
| 03:15.01 | CatLynx | but not on the AT&T phone, and if I use the SPA-3102 both phone works for the WMI |
| 03:15.07 | *** join/#asterisk badweather (~brentw@modemcable176.244-81-70.mc.videotron.ca) |
| 03:15.23 | manxpower | So when you plug the AT&T phone into the TDM card you do NOT get a stutter tone? |
| 03:15.32 | CatLynx | it gets stutter tone |
| 03:15.39 | manxpower | the stutter tone is "MWI" |
| 03:15.40 | CatLynx | but no WMI lite |
| 03:15.56 | CatLynx | so it could be a audio level issue? |
| 03:15.58 | badweather | Is there an ENUM number that can be used to test an outbound trunk? Just set it up on my trixbox but I wonder if the numbers I'm trying are no longer functional |
| 03:16.00 | manxpower | CatLynx: check the settings for the AT&T phone. |
| 03:16.09 | manxpower | it works with one phone and not the other. |
| 03:16.32 | manxpower | The Uniden phone I used to have had an option to enable/disable VISUAL MWI |
| 03:16.36 | CatLynx | max: the at&t phone works on the SPA-3102 adatpor along with the pansonic phone |
| 03:16.44 | CatLynx | err |
| 03:17.13 | manxpower | CatLynx: then you have an issue not related to Asterisk's MWI. |
| 03:17.48 | CatLynx | manx: I was wondering if its a audio level issue |
| 03:18.42 | CatLynx | manx: so when the phone is onhook the FXS port sends a stutter tone when it needs to trigger the WMI lite? |
| 03:19.40 | CatLynx | I was thinking if the audio level as at a level that it would work on one brand phone and not the other |
| 03:20.23 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:21.17 | CatLynx | what chan_dahdi.conf setting would adjust the WMI tone level? |
| 03:22.03 | *** join/#asterisk bmg505 (~leon@196-209-71-68-rndf-esr-3.dynamic.isadsl.co.za) |
| 03:22.10 | CatLynx | this is asterisk 1.6.2.6 dahdi 2.2.1.1 |
| 03:22.10 | ChannelZ | don't think there is one but there is an overall txgain and rxgain in the dahdi config |
| 03:22.56 | ChannelZ | actually I guess it is in chan_dahdi |
| 03:23.30 | CatLynx | Chan: I try messing with that one and had no luck I try txgain from +10 to -10 in 5db steps and had no luck :( |
| 03:25.19 | voxter | Anyone familiar with the MACRO_RESULT variable changing behavior in 1.4? I was using it in a call screening macro that seems to have broken. |
| 03:26.17 | manxpower | voxter: it should be documented in UPGRADE*.txt |
| 03:31.02 | CatLynx | wonder if my problem is the stutter timing |
| 03:35.47 | CatLynx | hmmm |
| 03:36.10 | CatLynx | im reading there is 2 kind of MWI from the telco Stutter Tone and FSK |
| 03:36.25 | CatLynx | I need to take a butt set and see which one is the SPA adaptor is doing |
| 03:36.32 | CatLynx | and compair it with the asterisk |
| 04:03.52 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 04:06.03 | *** join/#asterisk Faithful (~Faithful@121.91.118.214) |
| 04:08.35 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 04:16.02 | ChannelZ | snickers |
| 04:16.33 | hardwire | butterfingers |
| 04:18.21 | hardwire | sigh. Codec negotiation makes me insane. |
| 04:18.27 | hardwire | insane.. got no brain. |
| 04:22.16 | badweather | What would cause enum outbound to always give a congestion message? |
| 04:32.12 | ChannelZ | cold a flu season |
| 04:36.33 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.59) |
| 04:38.15 | badweather | harr. ;) |
| 04:39.36 | badweather | I can see that it's resolving ok, but doesnt' want to connect. Is it possible those in routes are not set up on the other end? |
| 04:46.55 | *** part/#asterisk badweather (~brentw@modemcable176.244-81-70.mc.videotron.ca) |
| 04:48.12 | CatLynx | manxpower: I listen to the TDM410 for the MWI on the FXS port and its using FSK |
| 04:48.30 | CatLynx | manxpower: the SPA is using FSK as well |
| 04:49.16 | CatLynx | ponders why the MWI not working on the TDM410 but it works on the SPA |
| 04:49.47 | *** join/#asterisk rdircio (~admin@201.137.45.224) |
| 04:50.25 | p3nguin | ~answers |
| 04:50.26 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 04:51.28 | CatLynx | I could not find any info on setting the MWI levels |
| 04:52.22 | ChannelZ | I sure hope I did my taxes right |
| 04:52.48 | ChannelZ | fucking Rube Goldberg tax system |
| 04:56.01 | p3nguin | http://www.youtube.com/watch?v=qI3IHahHQIg |
| 05:01.04 | hardwire | ~questions |
| 05:01.05 | infobot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
| 05:07.15 | *** join/#asterisk kartik (~koolkarti@117.199.116.113) |
| 05:12.18 | CatLynx | ok I see something diffrent on my MWI issue the SPA is doing RP pulse before sending FSK tone |
| 05:12.32 | CatLynx | the TDM410 is not doing that its just sending FSK |
| 05:14.20 | CatLynx | im going to add cidstart=polarity and see if that works for MWI also |
| 05:17.03 | CatLynx | sighs |
| 05:19.29 | *** join/#asterisk dangerkoffe (~dangerkof@93.90.47.30) |
| 05:21.12 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 05:29.57 | *** join/#asterisk synch (~d2d46c84@gateway/web/freenode/x-ogprgdiezweabyhs) |
| 05:36.14 | *** join/#asterisk CatLynx (~booga@173-11-77-182-SFBA.hfc.comcastbusiness.net) |
| 05:38.44 | *** join/#asterisk kartik (~koolkarti@117.199.127.217) |
| 05:39.49 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 05:39.49 | *** join/#asterisk ivaxer (~ivaxer@dev.ptr.sgu.ru) |
| 05:39.49 | *** join/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl) |
| 05:39.49 | *** join/#asterisk hurdman (~ngeek@ys.antredugeek.fr) |
| 05:43.32 | *** join/#asterisk b14ck (~b14ck@cpe-76-95-129-196.socal.res.rr.com) |
| 05:44.56 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 05:51.14 | CatLynx | [2010-04-14 22:50:42] WARNING[11772]: chan_dahdi.c:9167 mwi_send_process_buffer: MWI FSK Send Write failed: Resource temporarily unavailable |
| 05:51.21 | CatLynx | scratches head and ponders |
| 05:58.32 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 05:59.08 | *** join/#asterisk dangerkoffe (~dangerkof@93.90.47.30) |
| 06:06.44 | *** join/#asterisk Tulga (~chatzilla@203.91.113.10) |
| 06:07.11 | Tulga | how to limit incoming call 1 for sip agents? I'm using 1.6.2 |
| 06:07.42 | CatLynx | call-limit = 1 |
| 06:07.48 | Tulga | in sip.conf? |
| 06:08.04 | Tulga | it limit all call limit or each SIP agent limit? |
| 06:08.08 | Jed | doesnt that limit all calls? |
| 06:08.10 | CatLynx | depends where you want it |
| 06:08.11 | Jed | inbound and outbound? |
| 06:08.21 | CatLynx | it limits both direections |
| 06:08.30 | CatLynx | you can do it global or per user |
| 06:08.38 | CatLynx | oh wait |
| 06:08.41 | CatLynx | per user |
| 06:10.05 | *** part/#asterisk synch (~d2d46c84@gateway/web/freenode/x-ogprgdiezweabyhs) |
| 06:10.57 | Tulga | I put call-limit=1 in general block. but my sip members still have calling |
| 06:11.04 | Tulga | I did sip reload |
| 06:11.29 | CatLynx | put it in where the user config is |
| 06:11.36 | *** join/#asterisk ktwilight[m] (~ktwilight@91.180.37.231) |
| 06:12.10 | *** join/#asterisk kartik (~koolkarti@117.207.86.117) |
| 06:18.11 | Tulga | ok it works |
| 06:18.14 | Tulga | thank you CatLynx |
| 06:30.16 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-xmrmeeggtkrbxnhs) |
| 06:34.24 | *** join/#asterisk Faithful (~Faithful@119.12.29.20) |
| 06:48.55 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 06:51.39 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 06:52.00 | kruemeltee | hello again :-) |
| 06:59.06 | *** join/#asterisk khussein78 (~khussein7@188.225.192.238) |
| 06:59.10 | khussein78 | hi |
| 07:00.00 | *** join/#asterisk Faithful (~Faithful@121.91.118.214) |
| 07:00.01 | khussein78 | i am configuring iax2 trunk with voip provider in my country, i opened port 4569 from and to their server on my juniper SGG5 |
| 07:00.18 | khussein78 | i can see peer is connected but i cannot register to their server |
| 07:00.43 | khussein78 | in asterisk logs i cannot find any thing helpful about why registration is not established |
| 07:00.48 | khussein78 | any help here |
| 07:03.26 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 07:03.28 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 07:04.24 | khussein78 | how can i debug iax registration ? |
| 07:19.43 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 07:20.48 | *** join/#asterisk kartik (~koolkarti@117.207.82.145) |
| 07:21.44 | *** join/#asterisk J4zen (~j4zen@95-36-106-201.dsl.alice.nl) |
| 07:22.21 | J4zen | Hi there, im about to setup a STUN server for a client. What package would you recommend? It'll be used for Asterisk/SIP routing. |
| 07:25.56 | ChannelZ | Hmm never had to set one up |
| 07:26.09 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 07:35.47 | ChannelZ | khussein78: turn in iax debugging, turn up the console verbose a little, it should tell you if it's timing out or rejecting or what |
| 07:36.23 | khussein78 | ChannelZ, i run iax2 trunk debug |
| 07:37.00 | khussein78 | ChannelZ, or i should run another command ? |
| 07:45.48 | ChannelZ | iax2 set debug on |
| 07:47.40 | ChannelZ | core set verbose 5 |
| 07:48.08 | *** join/#asterisk Kumbang (~kumbang@167.205.24.69) |
| 07:53.08 | khussein78 | only thing i saw about their IP is doing dnsmgr_lookup for '1.2.3.4' |
| 07:54.25 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 07:54.49 | ChannelZ | what does your 'iax2 show registry' say? |
| 07:55.47 | khussein78 | empty |
| 07:55.52 | khussein78 | i see this in log |
| 07:55.53 | khussein78 | No IAX provisioning configuration found, IAX provisioning disabled |
| 07:56.09 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 07:59.08 | ChannelZ | ok so if you think you're trying to register with someone else but iax2 show registry shows nothing, you probably have jacked up your iax.conf |
| 08:00.25 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-vigalhehieqfxqdk) |
| 08:02.16 | *** join/#asterisk AlHafoudh (~alhafoudh@195.46.69.4) |
| 08:02.17 | AlHafoudh | hi guys |
| 08:04.04 | AlHafoudh | when i have h323 on local network and SIP provider and I have faststart=0 in h323.conf, i cannot hear ringback while calling to SIP, no control tones, just disconnection if destination is unreachable or immediately call connection, during call building i hear silence, is there any parameter that does that? from my cisco colleague I got that cisco has "tone ringback alert-no-pi" parameter on dialpeer |
| 08:06.13 | ChannelZ | you lost me at 'hi guys' |
| 08:06.44 | AlHafoudh | :) |
| 08:07.04 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 08:10.11 | AlHafoudh | anyone? |
| 08:12.07 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:17.13 | *** join/#asterisk kartik (~koolkarti@117.207.82.145) |
| 08:23.08 | *** join/#asterisk kartik (~koolkarti@117.207.82.145) |
| 08:25.19 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 08:25.34 | *** join/#asterisk Faithful (~Faithful@119.12.29.20) |
| 08:29.35 | *** join/#asterisk pentanol (~pentanol@77.35.49.164) |
| 08:29.37 | khussein78 | ChannelZ, sorry, but what do you mean by jacked up iax.conf |
| 08:29.51 | *** join/#asterisk Tim_Toady (~moi@188.4.0.11.dsl.dyn.forthnet.gr) |
| 08:34.57 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 08:35.07 | *** join/#asterisk chendy_ (~chatzilla@204.152.211.137) |
| 08:36.38 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 08:36.43 | *** join/#asterisk kartik (~koolkarti@117.199.117.116) |
| 08:59.47 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:01.52 | khussein78 | i added #include iax_registrations.conf before #include iax_additional.conf |
| 09:03.06 | khussein78 | i got this |
| 09:03.08 | khussein78 | WARNING[26135] pbx.c: Context 'from-trunk-iax2-6656399200' tries to include nonexistent context 'from-trunk-iax2-6656399200-custom' |
| 09:07.44 | *** join/#asterisk emaia (~emaia@adonis.iportalmais.pt) |
| 09:08.12 | emaia | hello |
| 09:08.49 | emaia | i need help with JTAPI. anyone can help? thanks |
| 09:13.46 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 09:17.51 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 09:18.03 | *** join/#asterisk Z_God (~julius@wlan234052.mobiel.utwente.nl) |
| 09:24.36 | AlHafoudh | please anyone? i cannot hear ringback when calling from h323 phone to sip provider |
| 09:25.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:30.30 | *** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru) |
| 09:38.44 | *** join/#asterisk maxagaz (~maxagaz@222.128.36.151) |
| 09:47.25 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 09:50.51 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 09:59.41 | *** join/#asterisk Akiraaa (~Akira@92.81.196.47) |
| 10:01.01 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-lgbrjulmjmzbtlzc) |
| 10:07.12 | *** join/#asterisk chendy_ (~chatzilla@204.152.211.137) |
| 10:13.49 | manxpower | khussein78: FreePBX/Trixbox is not supported here |
| 10:13.51 | manxpower | ~freepbx |
| 10:13.52 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 10:18.52 | Wildy | Ok, I've a question about Local channels. We're using 1.6.0.22 and have noticed that the a Local channel might become hung and the call would get written to the CDR much later than it should. Any advice? |
| 10:19.58 | Wildy | I'll try to arrange an update tonight and see what would happen. But I'd like to know if this is a known situation (possibly version-unrelated?) |
| 10:27.11 | manxpower | Wildy: if it is then the fix should be listed in the changelog |
| 10:28.07 | Wildy | ok, i'll look it up on bugs |
| 10:28.32 | Wildy | all in all, freepbx drives me mad, and we're using it (with some mods) on a live call center |
| 10:30.29 | kruemeltee | can anybody tell my what's happening, if I do a : Goto(context,007,1) and within the context the is no extension with "007", but another one with pattern matching (_X.) ... will * enter this extension instead? |
| 10:30.41 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 10:31.24 | manxpower | yes |
| 10:32.39 | kruemeltee | okay ... puuh ... :-) |
| 10:36.17 | Mark22 | I currently use DTMF=rfc2833 and that works great when we call to most external numbers, but for some numbers it doesn't work. What could I use so we can use a call menu with anyone we can call? |
| 10:37.00 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:44.21 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 10:44.27 | Mark22 | setting dtmfmode=info did fix it |
| 10:44.48 | shamelessn00b | hi, I needed help regarding integrating my asterisk PBX with coolswitchIP |
| 10:56.03 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 11:06.09 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.60) |
| 11:16.07 | *** join/#asterisk davido1 (~davido1@p54B0A607.dip0.t-ipconnect.de) |
| 11:16.08 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 11:17.19 | *** join/#asterisk joobie (~joobie@CPE-124-180-152-53.lnse4.lon.bigpond.net.au) |
| 11:30.13 | AlHafoudh | when i have h323 on local network and SIP provider and I have faststart=0 in h323.conf, i cannot hear ringback while calling to SIP, no control tones, just disconnection if destination is unreachable or immediately call connection, during call building i hear silence, is there any parameter that does that? progressinband=never did not work in sip.conf |
| 11:31.24 | Akiraaa | If anyone is using Skype for Asterisk, do you have some general impressions or recommendations? Are there quirks and caveats? |
| 11:32.04 | Chainsaw | Akiraaa: Skype for Asterisk will not allow you to use a Skype subscription, only SkypeOut credits. |
| 11:32.09 | shamelessn00b | how can I create a SIP trunk between asterisk and some other SIP switch |
| 11:39.17 | *** part/#asterisk moos3 (~rgenthner@rrcs-24-39-23-74.nys.biz.rr.com) |
| 11:39.18 | Akiraaa | Chainsaw: from the digium description Key Features: "Make Skype to Skype calls " |
| 11:39.34 | Akiraaa | Chainsaw: http://store.digium.com/productview.php?product_code=1SFA0001 |
| 11:39.43 | Chainsaw | Akiraaa: Yes, you will be able to call Skype users for free. |
| 11:39.54 | Chainsaw | Akiraaa: If you have SkypeOut credit in the account, you will be able to make outbound calls to the PSTN. |
| 11:40.10 | Chainsaw | Akiraaa: If you have a Skype *subscription* that allows you to call out to the PSTN, it will *not* work. |
| 11:40.33 | Akiraaa | Chainsaw: Ah, I see. Thanks for the clarification! |
| 11:43.14 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 11:47.16 | *** join/#asterisk Faithful (~Faithful@119.12.29.20) |
| 11:53.56 | *** join/#asterisk cjk (~cjk@80.90.44.29) |
| 11:54.13 | cjk | hi, is there a way in asterisk to launch a call, wait for the first ring and hangup? |
| 12:14.26 | *** join/#asterisk zarnick (~zarnick@unaffiliated/zarnick) |
| 12:15.19 | *** join/#asterisk coppice (~chatzilla@202.64.176.93) |
| 12:15.23 | zarnick | guys, I'm having a serious problem with Asterisk (someone is using it to make international calls :O) can someone help me block this? |
| 12:16.54 | Chainsaw | zarnick: Make sure you are using contexts correctly. |
| 12:17.04 | zarnick | I can see on the logs conections from SIP/<IP>, which should be wrong, every connection should be made from SIP/<extensions> |
| 12:17.05 | Chainsaw | zarnick: I should not be able to dial out on the PSTN if I'm not a local user. |
| 12:17.27 | Chainsaw | zarnick: So local SIP users & remote SIP users should *never* be in the same context. |
| 12:17.46 | WIMPy | Certainly not guests. |
| 12:17.55 | zarnick | Chainsaw: yeap, that's the point, however, I wasn't the one who installed this server, and know squat about Asterisk, how can I check this? |
| 12:18.13 | Chainsaw | zarnick: There are context settings in sip.conf |
| 12:18.41 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:18.47 | Chainsaw | zarnick: Have a look, see where your authenticated users are ending up. |
| 12:20.12 | zarnick | Hum, I'll take a look, any specific option that I should look for? |
| 12:20.20 | Chainsaw | zarnick: context= |
| 12:22.09 | zarnick | what does the context= means? |
| 12:22.41 | zarnick | I can see some stuff |
| 12:22.45 | Chainsaw | zarnick: It sets the context that the SIP account is in. Please go read up on the use of contexts in Asterisk. |
| 12:22.54 | Chainsaw | zarnick: It is the only way you can secure this installation. |
| 12:24.49 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.220.217) |
| 12:25.22 | zarnick | I will, I'm just afraid to actually lock on everyone...since this is in production and I don't have a test environment :$ |
| 12:26.45 | Chainsaw | zarnick: That's not a workable situation. Clone what you have and fiddle with the test box until you get it right. |
| 12:27.23 | Chainsaw | zarnick: Or turn the production system off until you can secure it. It sounds like a disaster waiting to happen. |
| 12:27.36 | zarnick | Chainsaw: you got that right |
| 12:28.16 | zarnick | another question, is there any way to record the international calls that are being generated? |
| 12:28.32 | Chainsaw | zarnick: Yes, you need to set Asterisk to full logging. What version is it? |
| 12:29.27 | zarnick | One sec |
| 12:29.52 | zarnick | 1.4.22-3 |
| 12:30.00 | Chainsaw | asterisk -r |
| 12:30.01 | Chainsaw | core set debug 10 |
| 12:30.03 | Chainsaw | core set verbose 10 |
| 12:30.27 | Chainsaw | core show channels <- That will show you calls in progress. |
| 12:30.52 | Chainsaw | Further calls will show on that console, keep it open. |
| 12:31.40 | zarnick | allright...and how do I filter only calls on a specific trunk? |
| 12:32.00 | zarnick | route...sorry |
| 12:32.07 | Chainsaw | zarnick: If you see a channel you like, core show channel X |
| 12:32.23 | Chainsaw | zarnick: There is tab completion in this console, you will find that handy. |
| 12:32.47 | zarnick | thanks...and to record the call? |
| 12:33.11 | Chainsaw | I don't monitor on my system, you'd have to ask others. |
| 12:33.58 | zarnick | thanks |
| 12:34.09 | zarnick | anyone knows how to record a incoming call? |
| 12:34.50 | *** join/#asterisk TommyBotten (tommy@broken.pipe.no) |
| 12:34.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:35.57 | zarnick | Chainsaw: isn't there an option to block SIP connections from specific IPS? for now I could use this until we have a full test server |
| 12:36.13 | Chainsaw | zarnick: That would be a firewall. Don't overthink it :) |
| 12:37.05 | zarnick | hum... |
| 12:37.34 | Chainsaw | There may already be a firewall in the path in your network. If so, blacklist the problematic IPs there. |
| 12:37.50 | Chainsaw | Failing that, you may configuring a firewall easier than familiarising yourself with a complete Asterisk configuration. |
| 12:38.01 | Chainsaw | Especially if time is of the essence. |
| 12:38.13 | zarnick | damn...having to become a asterisk admin without knowing how to be, and on such a stress situation...is hard...hehehe |
| 12:38.22 | zarnick | the firewall team has already being warned... |
| 12:38.51 | Chainsaw | Don't warn them. Tell them what to do. "Blacklist IP addresses 1, 2 & 3 and report to me when you're done." |
| 12:38.56 | TommyBotten | I'm using *1.6.0.22, and a queue with three static agents. When agent 1 has answered an incoming queue call and 2 and 3 are idle - And two additinal calls comes into the queue. The result is that agent 1 can never get call 2, even after he is finished call 1. Call 3 however is directed at agent 1. |
| 12:39.19 | TommyBotten | I have set autofill to yes. |
| 12:39.21 | Chainsaw | zarnick: You've clearly been put in charge of this situation. So take charge and neutralise the threat. |
| 12:39.41 | zarnick | on the contexts, there's an option called "outbound-allroutes" that's disabled on the context for international calls, what this option means? |
| 12:40.04 | zarnick | Chainsaw: yeap, I think they already blocked (I get no more international calls from those ips) |
| 12:40.18 | Chainsaw | zarnick: I can't decode a dial plan based on off-hand observations. You'd have to pastebin me the whole thing. |
| 12:40.30 | Chainsaw | zarnick: The dial plan lives in /etc/asterisk/extensions.conf in most cases. |
| 12:41.40 | zarnick | Chainsaw: any thing I should clear of the extensions.conf file before pasting it? |
| 12:42.10 | Chainsaw | zarnick: Check it over for passwords, internal IP addresses, hostnames that you wouldn't want a random guy on the street to know, etc. |
| 12:42.54 | zarnick | yeah...checking it...the problem I see is that it has a lot of includes...so it would end up being a gigantic paste |
| 12:42.59 | zarnick | (BTW: This is TrixBox) |
| 12:43.18 | Chainsaw | I can't help you with that. |
| 12:43.27 | Chainsaw | You really should have mentioned that a good half hour ago. |
| 12:43.43 | zarnick | sorry... |
| 12:43.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:43.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:45.21 | Chainsaw | zarnick: There is a #trixbox channel, I'd suggest asking your question there. |
| 12:45.30 | Chainsaw | Morning Leif. |
| 12:45.36 | leifmadsen | morning |
| 12:46.35 | *** join/#asterisk ickmund (~magnus@cli-5b7ee15c.bcn.adamo.es) |
| 12:46.37 | zarnick | Chainsaw: I will, thanks anyway ;) |
| 12:55.48 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 12:55.54 | *** join/#asterisk guyvdb_ (~guy@dsl-240-172-102.telkomadsl.co.za) |
| 12:55.57 | eject_ck | Hi all |
| 12:56.57 | *** join/#asterisk mdg (~mdg@unaffiliated/mgroman) |
| 12:57.09 | guyvdb_ | Hi, I am looking to start using the AMI. Is the file astman.js in static-http the implementation of AJAM? And is this calling AMI to execute? Finally is asterisk-gui written using AJAM and AMI? |
| 12:57.59 | eject_ck | I need to send PDF file from Linux machine to fax SIP chan |
| 12:58.18 | mdg | Hi |
| 12:58.29 | eject_ck | I have SIP connection to provider and want to send pdf to fax machine |
| 12:58.42 | [TK]D-Fender | eject_ck: Unless you're running T.38 on it your odds are low |
| 12:58.47 | eject_ck | I have it |
| 12:58.51 | eject_ck | t.38 |
| 12:59.11 | eject_ck | [TK]D-Fender: what the next ? |
| 12:59.14 | [TK]D-Fender | eject_ck: Go lset up your peer accordingly and "core show applications like fax" at * CLI for the apps. |
| 12:59.31 | [TK]D-Fender | eject_ck: You'll ahve to convert it to TIFF first |
| 12:59.34 | *** join/#asterisk _gm (~quassel@203.215.176.22) |
| 12:59.39 | [TK]D-Fender | eject_ck: plenty of CLI tools for that |
| 12:59.45 | eject_ck | ok |
| 12:59.45 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 12:59.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:59.57 | eject_ck | I need send faxes from asterisk server |
| 13:00.42 | *** join/#asterisk russo (~russo@p5799E9EB.dip.t-dialin.net) |
| 13:00.44 | [TK]D-Fender | eject_ck: Yes... we got that already... |
| 13:00.54 | [TK]D-Fender | eject_ck: So go look at the apps that do this |
| 13:01.17 | [TK]D-Fender | eject_ck: "core show applications like fax" |
| 13:01.21 | eject_ck | [TK]D-Fender: s04*CLI> core show applications like fax |
| 13:01.21 | eject_ck | <PROTECTED> |
| 13:01.21 | eject_ck | <PROTECTED> |
| 13:01.40 | [TK]D-Fender | eject_ck: then you are clearly missing the SpanDSP libs reuired for them to have been built |
| 13:01.52 | eject_ck | I'm on ubuntu |
| 13:02.01 | [TK]D-Fender | eject_ck: Go install the pre-reqs |
| 13:02.37 | eject_ck | libspandsp1 |
| 13:03.31 | [TK]D-Fender | eject_ck: Then rebuild * |
| 13:03.45 | russo | hey guys, i'm trying to setup asterisk to run as a sip proxy to my sip providers (i want to be able to use multiple providers for example), in any case i noticed that asterisk is listening on a bunch of ports with protocols i don't need... i.e. dundi, anyway where can i diable all these daemons? (all i really need iirc is sip, right? http://pastie.org/921147 <- netstat, to show you whats running) |
| 13:04.04 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:04.05 | eject_ck | [TK]D-Fender: pre-compiled binaries not work ? |
| 13:04.21 | [TK]D-Fender | eject_ck: If you don't have the .so for those apps... clearly NOT |
| 13:04.40 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:04.45 | [TK]D-Fender | russo: modules.conf <- |
| 13:04.52 | [TK]D-Fender | russnoload the ones you don't need |
| 13:04.59 | [TK]D-Fender | russo: noload the ones you don't need |
| 13:05.13 | russo | [TK]D-Fender: thats where i looked first too, but i didn't find dundi in there for example |
| 13:05.25 | russo | this is on debian lenny |
| 13:05.29 | russo | its the stable asterisk |
| 13:05.31 | [TK]D-Fender | russlook in your modules folder.. it will be rather obvious |
| 13:05.39 | russo | ah okay |
| 13:07.30 | *** join/#asterisk rdircio (~admin@201.137.45.224) |
| 13:09.02 | russo | [TK]D-Fender: where is the modules folder by default? |
| 13:09.20 | russo | ah found it |
| 13:09.21 | russo | :) |
| 13:09.28 | [TK]D-Fender | russo: depends on your OS. common is /var/lib/asterisk/modules |
| 13:09.30 | russo | find / | grep asterisk | grep modules ;) |
| 13:10.15 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 13:10.27 | pabelanger | anybody using an Aastra 6731i? |
| 13:13.54 | [TK]D-Fender | pabelanger: Got a more specific question about them? |
| 13:14.18 | eject_ck | [TK]D-Fender: I found asterisk_faxreceive in packages. WIll it work? |
| 13:14.34 | [TK]D-Fender | eject_ck: Go try |
| 13:14.48 | eject_ck | what .so I need ? |
| 13:15.16 | eject_ck | core show applications like fax return 0 as well :) |
| 13:15.17 | [TK]D-Fender | eject_ck: just install it and see if the previous command shows you an app. |
| 13:15.28 | [TK]D-Fender | eject_ck: then its no good |
| 13:15.35 | [TK]D-Fender | BRB |
| 13:15.41 | Chainsaw | <PROTECTED> |
| 13:15.45 | eject_ck | there is list of files |
| 13:15.46 | eject_ck | http://packages.ubuntu.com/jaunty/amd64/asterisk-app-fax/filelist |
| 13:16.23 | mdg | thats mighty jaunty of you |
| 13:18.02 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:18.38 | eject_ck | ok |
| 13:18.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:18.46 | eject_ck | I loaded teh modules manually |
| 13:19.09 | eject_ck | [TK]D-Fender: I have rxFAX and TXFax applications |
| 13:19.24 | [TK]D-Fender | ejctGood. there you go. |
| 13:19.33 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:19.38 | eject_ck | [TK]D-Fender: what the next ? |
| 13:19.50 | [TK]D-Fender | eject_ck: USE them |
| 13:19.56 | eject_ck | [TK]D-Fender: how :)? |
| 13:20.17 | [TK]D-Fender | eject_ck: ever heard of dialplan? Have * call out and call that app |
| 13:20.21 | eject_ck | sure |
| 13:20.24 | eject_ck | ok |
| 13:20.34 | [TK]D-Fender | eject_ck: "AMI originate" "call files", etc. Take your pick |
| 13:21.16 | eject_ck | So I need send .TIFF files using console |
| 13:21.24 | eject_ck | is it possible ? |
| 13:21.40 | russo | hey thanks again [TK]D-Fender! |
| 13:21.59 | russo | i just like to audit my logs :P |
| 13:22.11 | mdg | So.. Adhearsion is a Ruby DSL for creating dialplans? |
| 13:22.17 | russo | and i would be going wtf at every audit if i saw asterisk using stuff that i don't actually use ;) |
| 13:22.27 | [TK]D-Fender | russo: Its good to disable channel drivers and listening daemons you have no need of for security reasons alone |
| 13:22.56 | russo | that too, i mean i would have them firewalled off... but disabling is still better |
| 13:23.06 | [TK]D-Fender | eject_ck: Have * call out. dump call into dialplan. Call the app, passing it a TIFF. The End |
| 13:23.22 | guyvdb_ | He do i get help on AMI commands? For instance if I type "manager show command login" it does not show me the parameters to pass. Where would i find them? |
| 13:23.27 | guyvdb_ | he=how |
| 13:23.43 | [TK]D-Fender | russIndeed. More stable as less to load, and You do't have to worry about holes, when you don't mount the wall. |
| 13:24.26 | *** join/#asterisk ecolitan (~aaron@li57-124.members.linode.com) |
| 13:28.49 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-adbossrnlwpbgspg) |
| 13:29.06 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:29.12 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 13:29.58 | *** join/#asterisk clive- (ident@dsl-243-93-255.telkomadsl.co.za) |
| 13:30.34 | clive- | here is a beginners question, what is RTP format 101 referring to?...or which codec is that ? |
| 13:30.55 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 13:31.51 | [TK]D-Fender | cliG.722 I believe |
| 13:32.01 | [TK]D-Fender | clive-: iG.722 I believe |
| 13:32.49 | clive- | TK, thanks.... |
| 13:33.52 | c0rnoTa | Hello all |
| 13:34.18 | c0rnoTa | i'm getting libpri.so.1.4 segfault sometimes |
| 13:34.38 | c0rnoTa | using libpri-1.4.10.2 |
| 13:36.50 | c0rnoTa | can anyone say that there is no problem in latest SVN version? :) |
| 13:38.34 | leifmadsen | c0rnoTa: no |
| 13:38.45 | leifmadsen | c0rnoTa: i'm sure there is at least 1 issue that has no yet been found |
| 13:42.19 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:44.42 | hurdman | is there someone here who can set CALLERID with a T2 NGN ? i try and try again and i can set it , only show or mask :'( |
| 13:45.00 | hurdman | s/can/can't |
| 13:45.05 | [TK]D-Fender | c0rnoTa: Given that its about 4 months old now... why isn't everyone else having your problems with it? Perhaps you should look at the bigger scope of your scenario. |
| 13:48.25 | Chainsaw | Right, so what format of certificate would Asterisk like? The .crt followed by the CA and then the key? |
| 13:48.27 | Chainsaw | Or just the key? |
| 13:48.42 | Chainsaw | <PROTECTED> |
| 13:50.30 | *** join/#asterisk ktwilight[m] (~ktwilight@91.180.37.231) |
| 13:51.19 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:51.47 | *** join/#asterisk cyyaw (~cyyaw@190.43.168.28) |
| 13:51.55 | Chainsaw | (It's not a self-signed certificate either, so why this is so problematic... I don't quite understand) |
| 13:52.12 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:52.37 | cyyaw | hi, why allow = alaw, ulaw is incorrect , ' ulaw' no codec |
| 13:53.15 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 13:53.26 | c0rnoTa | [TK]D-Fender: thanks for your advice. I'm looking in bugtracker and see that somebody had segfaults wired with libpri-1.4.10.2 So, i'll dig deeper and in bigger scope for some more info about my segfault. May be, someone already solved it. Anyway, thanks. |
| 13:56.08 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:57.01 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:58.57 | *** join/#asterisk korcan (~kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net) |
| 14:00.27 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:02.28 | *** join/#asterisk LND (~LND@94.197.105.144.threembb.co.uk) |
| 14:05.05 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 14:08.43 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 14:12.49 | *** join/#asterisk Brooklyn (miracle@the.city.that.never-sleeps.us) |
| 14:17.01 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:18.42 | *** join/#asterisk TimeRider (~steve@82.132.136.201) |
| 14:20.38 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 14:20.43 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:20.47 | Katty | ohai |
| 14:22.09 | Chainsaw | ltns Katty |
| 14:22.50 | eject_ck | [TK]D-Fender: can you look at my extension for sending faxes ? |
| 14:22.54 | eject_ck | exten => _X.,1,Set(CALLERID(numm)=12345)}) |
| 14:22.54 | eject_ck | exten => _X.,2,Dial(SIP/trunk-sip/${EXTEN:1},20,rt) |
| 14:22.54 | eject_ck | exten => _X.,3,txfax(/tmp/file.tiff) |
| 14:23.13 | [TK]D-Fender | eject_ck: No. |
| 14:23.28 | [TK]D-Fender | eject_ck: I told you what you need to do to have CALL OUT |
| 14:23.52 | [TK]D-Fender | eject_ck: "call file", "AMI Originate", "CLIE originate". Go read |
| 14:24.50 | hurdman | eject_ck: you have to mm in your CALLERID(num |
| 14:25.10 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:25.55 | eject_ck | ah |
| 14:25.56 | eject_ck | ok |
| 14:26.03 | [TK]D-Fender | hurdman: Irrelevant. This solution does not work. |
| 14:26.19 | luke-jr | peers |
| 14:26.20 | [TK]D-Fender | hurdman: CID won't be an issue when the fax will never get sent. |
| 14:26.47 | *** part/#asterisk hc_e (~hc@pdpc/supporter/active/hc-e) |
| 14:26.49 | luke-jr | won't that call the target, and if that fails, transmit a fax to the person who dialed? |
| 14:27.08 | [TK]D-Fender | luke-jr: Something like that... but is not at all what he wants to do. |
| 14:27.29 | *** join/#asterisk hohum (dcorbe@apollo.corbe.net) |
| 14:32.50 | kruemeltee | says goodbye :-) |
| 14:35.28 | *** join/#asterisk CraigW76 (~techcaw@addr33.mimc.com) |
| 14:35.37 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:36.44 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:36.49 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:38.39 | WIMPy | I'm still cowfused regarding dahdi grouping. I have a 'line' comming on two PRIs. Using one D-Channel seems to work incl. fail-over. |
| 14:39.18 | *** part/#asterisk cm_ (~chris@datura-v2.ielf.org) |
| 14:39.39 | WIMPy | However I cannot dial out using dahdi/g1/, setting an explicit channel is ok. What are the relations between spanmal and group settings? |
| 14:40.44 | WIMPy | I found different examples. Is the third parameter of spanmal the logical link within the group or should it be globally unique? |
| 14:42.55 | [TK]D-Fender | WIMPy: PASTEBIN <- |
| 14:43.10 | [TK]D-Fender | WIMPy: Show us your configs and your failed dialout attempt |
| 14:43.36 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:43.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:45.01 | Katty | well i did my good deed for the day. |
| 14:45.11 | Katty | one of my co-workers had to bring her son in because the lady that watches him has strep |
| 14:45.21 | Katty | so i donated my laptop and netflix account so he could watch spongebob |
| 14:46.22 | WIMPy | Wait. Looks like I broke it further now. |
| 14:47.58 | [TK]D-Fender | Katty: No good deed goes unpunished |
| 14:48.09 | Naikrovek | Katty: you may wanna listen in on spongebob; it's pretty good at times |
| 14:48.34 | Kobaz | this is strange |
| 14:49.14 | Kobaz | http://pastebin.ca/1861889 i park someone... and then my 'h' exten runs... but it doesn't run correctly... it does a gosub to prio 1, but prio 2 starts executing |
| 14:49.28 | Kobaz | Gosub("Parked/SIP/240-000031d6<ZOMBIE>", "dialOut,h,1") |
| 14:49.34 | Kobaz | that should go to priority 1 you would think |
| 14:49.56 | leifmadsen | o.O |
| 14:50.10 | Kobaz | leifmadsen: my thoughts too |
| 14:50.27 | leifmadsen | I'm not sure you should be doing a GoSub() on the 'h' exten |
| 14:50.31 | Kobaz | leifmadsen: the bugs just seem to find me... |
| 14:50.34 | leifmadsen | the channel is dead |
| 14:50.43 | Kobaz | hmm |
| 14:50.51 | leifmadsen | you can't just loop back up to the top and start a new call |
| 14:51.02 | Kobaz | well it's running a hangup handler |
| 14:51.22 | Katty | Naikrovek: naw, i got work to do |
| 14:51.27 | Katty | Naikrovek: but his mom's sittin with him |
| 14:51.30 | leifmadsen | I usually do it by writing the hangup handler in a new context, then include => hangup_handler |
| 14:51.42 | Kobaz | hmm |
| 14:51.44 | leifmadsen | [hangup_handler] |
| 14:51.46 | Kobaz | i've never done it that way before |
| 14:51.48 | leifmadsen | exten => h,1,DoStuff() |
| 14:51.53 | beek | Katty: Spongebob rocks! |
| 14:51.59 | leifmadsen | I've never done it the way you're doing it :D |
| 14:52.02 | Kobaz | i always just gosub |
| 14:52.02 | Kobaz | heh |
| 14:52.10 | Kobaz | but this gosub isn't like the others |
| 14:52.15 | Kobaz | it's starting on line 2, and not line 1 |
| 14:52.15 | leifmadsen | there are 100s of ways to shoot yourself in the foot! |
| 14:52.25 | leifmadsen | hrmmm |
| 14:52.26 | leifmadsen | interesting |
| 14:52.32 | leifmadsen | can I see the whole console trace? |
| 14:52.36 | Kobaz | so, some important variable init isn't being done |
| 14:52.37 | Kobaz | sure |
| 14:52.40 | leifmadsen | you park someone, then it starts at h,2 ? |
| 14:52.47 | *** join/#asterisk tuxxie (~tuxxie@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 14:53.07 | Kobaz | http://pastebin.ca/1861893 |
| 14:53.41 | Kobaz | there should be a way to not specify a dial device with ParkAndAnnounce(), that bugs me... but that's another ossie |
| 14:53.44 | Kobaz | issue |
| 14:54.15 | leifmadsen | Kobaz: huh... that is very strange |
| 14:54.20 | Kobaz | when the caller gets parked, it runs the 'h' for some really weird reason |
| 14:54.34 | Kobaz | at the end of that log... the caller is spinning in park |
| 14:54.39 | Kobaz | and did not hang up |
| 14:54.50 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 14:54.53 | tuxxie | if i have a large voip network using with a router running QOS and only phones are on my swicthes do i need to enable qos on the layer 2 switches? It seems to me there would be no reason to set priority. Correct? |
| 14:54.53 | leifmadsen | the zombie channel is executing 'h' it would seem |
| 14:55.11 | leifmadsen | tuxxie: I've never set QoS to begin with, and my calls work just fine :D |
| 14:55.16 | Kobaz | yeah |
| 14:55.32 | leifmadsen | Kobaz: that does indeed look strange though... its like Asterisk is executing h,1 in one context, then falling to the next priority somewhere else |
| 14:55.36 | Kobaz | yeah |
| 14:55.40 | Kobaz | exactly |
| 14:55.46 | leifmadsen | what happens if you change the extension name? |
| 14:55.53 | Kobaz | change what to what? |
| 14:55.58 | leifmadsen | like GoSub(dialOut,hangup,1) |
| 14:56.02 | Kobaz | i can try |
| 14:56.13 | leifmadsen | more a curiosity than anything |
| 14:56.20 | Kobaz | rewrites the extensions view |
| 14:56.26 | leifmadsen | I wonder if it would go to hangup,2 |
| 14:56.31 | Kobaz | yeah i have no idea |
| 14:56.35 | leifmadsen | we'll find out |
| 14:56.43 | leifmadsen | that might help to narrow down what is actually happening |
| 14:57.47 | tuxxie | we are having some phone issues with pore phone quiltiy. I am using a edgemarc router for my for sip trunking and I see pore MOS scores in the edgemarc's logs. |
| 14:58.18 | Chainsaw | tuxxie: You probably mean "poor" instead of "pore". |
| 14:59.05 | tuxxie | We run around 70 concurrent calls and have 10Mbs detitacted to sip traffic |
| 14:59.09 | tuxxie | :/ sorry |
| 14:59.11 | p3nguin | better than pour, I guess. |
| 14:59.28 | Chainsaw | tuxxie: What are these "MOS scores" that you are speaking of? |
| 14:59.44 | coppice | a rolling log gathers no MOS |
| 14:59.56 | leifmadsen | *crickets* |
| 15:00.06 | Kobaz | <PROTECTED> |
| 15:00.09 | Kobaz | <PROTECTED> |
| 15:00.12 | Kobaz | tuxxie: ? |
| 15:00.29 | leifmadsen | Kobaz: huh! |
| 15:00.30 | leifmadsen | crazy |
| 15:00.32 | Kobaz | yeah |
| 15:00.40 | Kobaz | it's not supposed to do that |
| 15:00.43 | leifmadsen | I have no idea why it does that... |
| 15:00.44 | leifmadsen | agreed |
| 15:00.47 | leifmadsen | I'd file an issue |
| 15:01.00 | Kobaz | tuxxie: what are you inviting me to? |
| 15:01.01 | leifmadsen | you find some weird issues that I have no idea how they haven't been run into before |
| 15:01.09 | Kobaz | haha |
| 15:01.17 | Kobaz | leifmadsen: i know... every week i find something new |
| 15:01.27 | Kobaz | leifmadsen: i'm on a roll |
| 15:01.27 | leifmadsen | Kobaz: you're me about 2 years ago :) |
| 15:01.31 | Kobaz | heh |
| 15:01.35 | p3nguin | A good beta tester you are. |
| 15:01.35 | leifmadsen | I was the master bug finder back in the day |
| 15:01.38 | leifmadsen | now look at me! |
| 15:01.41 | leifmadsen | :D |
| 15:02.27 | Kobaz | heh |
| 15:02.45 | Kobaz | so if i put a noop on line 1 |
| 15:02.53 | Kobaz | it'll run the way it's supposed to |
| 15:03.36 | Kobaz | leifmadsen: am i the only one writing non-normal dialplan? |
| 15:03.38 | *** part/#asterisk tuxxie (~tuxxie@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 15:03.42 | leifmadsen | Kobaz: apparently ;) |
| 15:03.49 | Kobaz | leifmadsen: if all i do is dials and includes, everything is fine |
| 15:04.04 | Kobaz | as soon as i do something new, i find 23472389723497 bugs |
| 15:04.35 | Kobaz | oh yeah... i have a local channel race condition you can add to the documentation |
| 15:04.48 | leifmadsen | Kobaz: I've written some pretty complex dialplans in the past... so not sure why you're having so many issues |
| 15:04.53 | Kobaz | heh |
| 15:05.09 | leifmadsen | Kobaz: I have a system in production with like 1500 lines of dialplan |
| 15:05.13 | Kobaz | well |
| 15:05.29 | Kobaz | i have agi scripts (including libs) that are 3-5000 lines |
| 15:05.39 | leifmadsen | ya, I don't do much AGI stuff |
| 15:05.41 | Kobaz | it's not actually the dialplan complexity... you can get as complex as you want |
| 15:05.54 | Kobaz | but... it's like unexpected behaviors from asterisk that get me |
| 15:06.00 | leifmadsen | are you using realtime dialplans? |
| 15:06.01 | Kobaz | dialplan itself is fine for the most part |
| 15:06.14 | Kobaz | static-realtime and ael, and agi |
| 15:06.31 | Kobaz | but the static realtime is theoretically the same as writing extensions.conf directly |
| 15:06.49 | leifmadsen | interesting... I don't use AEL... so maybe the way its compiling back into dialplan is causing the issues? |
| 15:06.57 | Kobaz | i dont think so |
| 15:07.02 | Kobaz | it looks fine in the dialplan show output |
| 15:07.06 | leifmadsen | I remember trying AEL for about 20 mins once, and stopped when I ran into issues for something very simple |
| 15:07.11 | leifmadsen | but that was a LONG time ago |
| 15:07.11 | Kobaz | it's the executor that's skipping a line |
| 15:07.18 | leifmadsen | true |
| 15:07.22 | Kobaz | at least that's what it looks like |
| 15:07.23 | leifmadsen | strange... never seen that |
| 15:07.26 | leifmadsen | agreed |
| 15:07.29 | Kobaz | i mean, 1, just doesn't rnu |
| 15:07.32 | Kobaz | run... blarg |
| 15:07.44 | *** join/#asterisk omni__ (freeman@bb116-14-106-7.singnet.com.sg) |
| 15:07.49 | leifmadsen | not sure how we'd debug that.... but it should probably be filed as an issue |
| 15:07.53 | Kobaz | yeah |
| 15:07.54 | Kobaz | working on it |
| 15:08.01 | leifmadsen | coolio |
| 15:08.19 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:08.22 | hardwire | coolio eh |
| 15:08.36 | mdg | best rapper of all time |
| 15:08.46 | hardwire | http://goodjobbb.files.wordpress.com/2009/03/coolio_large.jpg |
| 15:09.22 | Kobaz | it takes so long to make a good bug report |
| 15:09.54 | Kobaz | leifmadsen: in may i'm actually hiring a guy whose job will partly be debuging/filing asterisk bugs |
| 15:10.09 | leifmadsen | Kobaz: yay! saves you some time I'm sure :) |
| 15:10.13 | Kobaz | it just sucks up so much of my time |
| 15:10.13 | Kobaz | yeah |
| 15:10.22 | hardwire | heh |
| 15:10.38 | omni__ | hi guys i need some help. i have been getting this while trying to dial out handle_request_invite: Call from '' to extension '81223544' rejected because extension not found |
| 15:10.39 | leifmadsen | sounds like a good job for someone. Be sure to introduce me to them so I know how it is when (s)he comes around |
| 15:10.44 | Kobaz | k |
| 15:10.56 | leifmadsen | omni__: sounds like a missing extension in the context that it is being looked up in |
| 15:11.06 | hardwire | omni__: your phone doesn't appear to have a peer associated with it |
| 15:11.10 | Kobaz | and i need to finish my group vars stuff one of these days, so it makes it into 1.8 |
| 15:11.23 | Kobaz | and the other 3 patches on the back burner too |
| 15:11.32 | leifmadsen | omni__: look at which context the peer is matching on with 'sip set debug peer <foo>' and then see what context it wants to match on, and then run "dialplan show 81223544@context-matching-on" |
| 15:11.45 | omni__ | ok let me try |
| 15:11.52 | leifmadsen | omni__: also make sure your phone or whatever is actually authenticating |
| 15:12.00 | omni__ | yah it is authenticated |
| 15:12.03 | leifmadsen | all those issues can be easily debugged with 'sip set debug' |
| 15:12.31 | Kobaz | sip set debug and a fancy comb |
| 15:13.32 | WIMPy | Ok, now I got it as far as I can dial out using a group, but it will only use the second line. |
| 15:13.41 | WIMPy | http://wimpy.yeti.dk/pastebin.txt |
| 15:13.46 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:13.49 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 15:14.17 | WIMPy | That's the box using old zaptel. |
| 15:14.27 | [TK]D-Fender | WIMPy: Where is the call? |
| 15:15.44 | *** part/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 15:15.56 | WIMPy | [TK]D-Fender: As I said, it works with group now, but it never uses the first line, only the second. |
| 15:16.09 | [TK]D-Fender | WIMPy: .... |
| 15:16.18 | *** join/#asterisk b14ck (~b14ck@cpe-76-95-129-196.socal.res.rr.com) |
| 15:16.32 | WIMPy | So unless I pull the second line, everything is fine. |
| 15:16.44 | WIMPy | But that's obviousely not he ide. |
| 15:16.46 | WIMPy | a |
| 15:18.53 | WIMPy | It only cycles channels 32-62, but it should be all 1-62. |
| 15:19.15 | WIMPy | (except 16 and 47, off course) |
| 15:20.09 | omni__ | There is no existence of 81223544@hkg-rshkg extension |
| 15:20.11 | omni__ | it shows this |
| 15:21.27 | omni__ | [Apr 15 23:16:01] NOTICE[7513]: chan_sip.c:15124 handle_request_invite: Call from '' to extension '81223544' rejected because extension not found.Scheduling destruction of SIP dialog '4a64d0ae106356c811449bcb02a1f4ac@ipaddress' in 32000 ms (Method: INVITE) -- and this.. |
| 15:21.29 | *** join/#asterisk Z_God (~julius@wlan229147.mobiel.utwente.nl) |
| 15:23.14 | *** join/#asterisk _omer (~omer@119.152.107.206) |
| 15:23.18 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 15:23.29 | Kobaz | leifmadsen: what module should this be filed under... what's the module for the dialplan executor |
| 15:23.40 | leifmadsen | Kobaz: try pbx_config |
| 15:23.44 | Kobaz | k |
| 15:23.53 | Kobaz | there it is |
| 15:24.25 | _omer | my asterisk 1.4.29 seems not getting details from say.conf ... I have commented out almost everything but saynumber() still works the same...I am using mode=new under [general] in say.conf |
| 15:24.35 | _omer | is there anything that I have missed? |
| 15:27.53 | Kobaz | what the bloody hell |
| 15:27.58 | Kobaz | leifmadsen: it has to do with park_timeout |
| 15:28.11 | Kobaz | leifmadsen: if you take out the park_timeout option... it works fine |
| 15:28.13 | leifmadsen | Kobaz: good ol' parking... |
| 15:28.27 | leifmadsen | that sounds wrong in so many ways |
| 15:28.39 | Kobaz | heh |
| 15:28.44 | leifmadsen | well, at least you can provide some additional information about how to reproduce it I think |
| 15:28.47 | Kobaz | yeah |
| 15:32.44 | WIMPy | [TK]D-Fender: No hint, what I'm doing wrong? |
| 15:33.02 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 15:33.02 | p3nguin | <PROTECTED> |
| 15:33.32 | WIMPy | He, I already said, that calls are working twice. |
| 15:34.05 | hardwire | .. |
| 15:34.05 | WIMPy | It's just that I can't group two spans togeter. r1 will only ever use the second. |
| 15:34.34 | WIMPy | Incomming calls are comming on both. |
| 15:34.50 | hardwire | so you have group=1 before span 1 and group=1 before span 2? |
| 15:34.53 | WIMPy | It's just that group thing I don't get working. |
| 15:35.11 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.77) |
| 15:35.34 | WIMPy | Before? Ok, that's a hint. |
| 15:36.04 | hardwire | you have to define a group before you say channels => xx-yy |
| 15:36.06 | *** join/#asterisk Tim_Toady (~moi@188.4.0.11.dsl.dyn.forthnet.gr) |
| 15:36.33 | hardwire | and it's inherited.. so if you are defining span 1 and 2 as long as group is set before span 1 and not set again.. span 2 will be in that group |
| 15:36.34 | omni__ | is it possible for a sip to dial to another sip and the sip dial out again? |
| 15:36.43 | hardwire | omni__: sure |
| 15:36.51 | WIMPy | so channel should always be the last entry? |
| 15:36.55 | hardwire | WIMPy: yes |
| 15:37.06 | WIMPy | ok |
| 15:37.13 | omni__ | as in the 1st sip will dial to the 2nd sip to make the 2nd sip to dial out |
| 15:37.14 | hardwire | otherwise there is no way to know what is specific to those channels |
| 15:37.24 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:37.38 | hardwire | omni__: this dialing that you speak of is not difficult |
| 15:37.39 | _omer | my asterisk 1.4.29 seems not getting details from say.conf ... I have commented out almost everything but saynumber() still works the same...I am using mode=new under [general] in say.conf |
| 15:38.08 | hardwire | _omer: pete and repete were in a boat.. pete jumped out.. who's left? |
| 15:38.18 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:38.29 | _omer | hmmm boat ;) |
| 15:38.34 | hardwire | fail |
| 15:38.34 | omni__ | thanks hardwire i tried to do this but i kept having extension rejected because is not found |
| 15:38.48 | _omer | F is still a grade |
| 15:38.50 | [TK]D-Fender | omni__: Go create one to match it then |
| 15:38.52 | p3nguin | Put the calls into a context where the extension exists. |
| 15:38.53 | hardwire | omni__: I'm not convinced that your sip phone is recognized as a peer |
| 15:38.58 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.20.207) |
| 15:39.08 | WIMPy | hardwire: Thanks. That was it. |
| 15:39.12 | WIMPy | Now it's all working. |
| 15:39.22 | hardwire | WIMPy: that'll be $5 |
| 15:39.26 | omni__ | i've set the sip type to be a peer |
| 15:39.36 | omni__ | it doesnt mean it'll be a peer? |
| 15:39.49 | *** join/#asterisk badweather (~brentw@modemcable176.244-81-70.mc.videotron.ca) |
| 15:40.01 | hardwire | if you do that then host has to be defined |
| 15:40.06 | hardwire | and correct |
| 15:40.28 | hardwire | typically you define a friend for a phone, not a peer, then the phone registers to asterisk and you can see the registration happen and succeed |
| 15:40.41 | hardwire | as well as see lots of information when using sip show peer xyz |
| 15:40.53 | WIMPy | Ok, thinking back it seems obvious. An explicit hint somewhere might be a good idea, tho. |
| 15:40.58 | hardwire | vital information like the UserAgent and the IP it registered from |
| 15:41.11 | hardwire | if this isn't what you'er doing.. then use a peer and make sure host is set correctly |
| 15:41.14 | p3nguin | Some people seem to think phones should be peers. |
| 15:41.32 | hardwire | p3nguin: it works fine as long as everything is static |
| 15:41.43 | omni__ | ok...the 1st sip in the 1st server shows that it is connected to the 2nd server |
| 15:42.10 | omni__ | but on the 2nd server, it shows unknown |
| 15:42.22 | hardwire | WIMPy: I believe the explicit hint is encoded into the brains of people that don't believe text files are magickal :) |
| 15:42.40 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:43.08 | WIMPy | No good for the blind. Unfortunaletly sometimes I'm amongst them. |
| 15:43.56 | hardwire | There are a few peeps in here that are blind afaik |
| 15:46.14 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:48.10 | *** join/#asterisk jsjc (~chatzilla@115.131.200.205) |
| 15:48.51 | jsjc | hello i am gavin some issues setting queues and looks like after joining the queues hangs up straight away (well I actually dont know if even joins it...) how coudl I debug queues? |
| 15:50.22 | _omer | any help on say.conf ? my asterisk 1.4.29 isn't getting details from it ... |
| 15:50.26 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 15:50.43 | badweather | What is a typical reason for getting a congestion message. I have it on 2 peers that I can't figure out what's wrong(ENUM, and Voxalot). I would like to get ENUM working most all |
| 15:51.01 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:51.10 | leifmadsen | badweather: more information needed |
| 15:52.20 | *** part/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 15:53.29 | badweather | leifmadsen: for my ENUM I can see it resolve the right domain and then it tries to made the contact with number@domain.com(whatever it may be). But it just says all circuits are busy |
| 15:53.44 | Naikrovek | that's it |
| 15:53.49 | Naikrovek | Katty: what's the number again |
| 15:53.53 | Naikrovek | for the immigration people |
| 15:53.58 | p3nguin | lol |
| 15:54.03 | p3nguin | Fed up, huh? |
| 15:54.05 | leifmadsen | badweather: again, more information required. Try pastebin'ing the sip trace somewhere |
| 15:54.15 | Naikrovek | so sick of my employer hiring idiots that need visas instead of qualified americans who are unemployed |
| 15:54.16 | leifmadsen | maybe the other end isn't answering? |
| 15:54.28 | leifmadsen | hides from Naikrovek |
| 15:54.39 | Naikrovek | canadians dont' bother me |
| 15:54.44 | leifmadsen | w00t! |
| 15:54.50 | Naikrovek | you're also not a complete moron |
| 15:54.53 | Naikrovek | not even a slight moron |
| 15:54.56 | leifmadsen | not completely |
| 15:55.04 | leifmadsen | there is light! |
| 15:55.16 | Naikrovek | these jackholes we're hiring are borderline retarded |
| 15:55.17 | leifmadsen | I'm slightly moronic at times |
| 15:55.32 | Naikrovek | yes, slightly, at times. these people are completely moronic all the time |
| 15:55.47 | Naikrovek | and they've spent $30k in visa fees in the past month |
| 15:55.59 | Naikrovek | while I go without tape backup or even switches capable of VLANs |
| 15:56.19 | badweather | leifmadsen: http://pastebin.com/b6RP1Duc |
| 15:56.47 | mdg | tape backup ?! |
| 15:56.54 | leifmadsen | badweather: you don't show me at all what I requested :) |
| 15:57.02 | leifmadsen | yep, you're making a call... yep, it's getting rejected... |
| 15:57.12 | leifmadsen | I have no idea why because you don't have any SIP trace output on the outgoing leg |
| 15:57.38 | leifmadsen | badweather: you're also passing back a pipe that I don't think you want to pass back |
| 15:57.41 | leifmadsen | that is likely causing the problem |
| 15:57.44 | Naikrovek | mdg: i can't back up anything and i have terabytes of stuff that need backed up daily |
| 15:57.50 | leifmadsen | <PROTECTED> |
| 15:57.50 | Qwell | leifmadsen: stop passing the pipe |
| 15:57.56 | leifmadsen | Qwell: beat you :) |
| 15:58.00 | Naikrovek | mdg: if too many disks fail, we're up a creek without a paddle |
| 15:58.14 | Qwell | leifmadsen: you misunderstood sir |
| 15:58.15 | leifmadsen | Naikrovek: at that point, you quit :) |
| 15:58.15 | Naikrovek | anyway |
| 15:58.22 | Naikrovek | leifmadsen: quit and do what |
| 15:58.25 | Naikrovek | there are no jobs here |
| 15:58.29 | leifmadsen | Qwell: I guess so? |
| 15:58.34 | Qwell | nevermind |
| 15:58.40 | Naikrovek | and probably thousands of unemployed nerds seeking work |
| 15:59.02 | leifmadsen | possibly |
| 15:59.03 | p3nguin | which they won't hire at that place. |
| 15:59.09 | leifmadsen | start a new business on company time :) |
| 15:59.16 | badweather | Ok I see that extra pipe is there. how do I enable the sip trace on the output? |
| 15:59.27 | leifmadsen | badweather: the trace is unnecessary -- the pipe is effing you up |
| 15:59.36 | leifmadsen | there isn't even an INVITE going out |
| 15:59.42 | anonymouz666 | to use sip realtime friends I just need to use both users+peers (extconfig) and insert into db as type 'friend'? |
| 15:59.42 | leifmadsen | fix your script |
| 15:59.45 | Qwell | leifmadsen: that sounds awesome without context |
| 15:59.48 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:59.54 | Qwell | ie: <leifmadsen> the pipe is effing you up |
| 15:59.56 | leifmadsen | Qwell: weeee! |
| 16:00.03 | leifmadsen | Qwell: perhaps I was being clever |
| 16:00.06 | Qwell | perhaps |
| 16:00.13 | badweather | leifmadsen: Ok, I'm using trixbox so I'm just using their settings. Will have to see where that might be occuring |
| 16:00.26 | leifmadsen | its in the enumlookup.agi |
| 16:00.29 | leifmadsen | fix the script |
| 16:00.35 | leifmadsen | another happy trixbox user |
| 16:00.42 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 16:00.44 | DelphiWorld | hi all! |
| 16:00.48 | DelphiWorld | error compiling dahdi: |
| 16:00.49 | DelphiWorld | You do not appear to have the sources for the 2.6.28-18-server kernel installed. |
| 16:00.53 | DelphiWorld | ubuntu server |
| 16:00.57 | leifmadsen | I need one of those cartoon graphics of a kid peeing on a trixbox logo |
| 16:01.02 | Qwell | DelphiWorld: So install the sources |
| 16:01.20 | leifmadsen | DelphiWorld: apt-get install linux-headers-$(uname -r) |
| 16:01.31 | p3nguin | sources, not headers. |
| 16:01.34 | *** join/#asterisk abatista (~chatzilla@63.214.236.169) |
| 16:01.42 | leifmadsen | if it's for DAHDI, the headers are all that are necessary |
| 16:01.48 | DelphiWorld | Qwell: how to please |
| 16:01.54 | leifmadsen | DelphiWorld: see my command above |
| 16:01.58 | omni__ | anyone attending asterisk conference |
| 16:01.59 | DelphiWorld | leifmadsen: k |
| 16:02.02 | leifmadsen | p3nguin: the text is misleading |
| 16:02.07 | leifmadsen | omni__: which one? |
| 16:02.12 | omni__ | in may |
| 16:02.21 | leifmadsen | omni__: which one? |
| 16:02.22 | p3nguin | If they meant headers and said sources, then I agree. |
| 16:02.22 | omni__ | in kuala lumpur malaysia |
| 16:02.39 | omni__ | http://www.asterconference.com/ |
| 16:02.40 | leifmadsen | I got invited to speak at it, but I can't attend because flights are $2000 |
| 16:02.53 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 16:03.01 | DelphiWorld | TheDavidFactor: TDF! |
| 16:03.08 | omni__ | i see |
| 16:03.21 | coppice | leifmadsen: don't be silly. flights to KL are very cheap |
| 16:03.31 | leifmadsen | coppice: from HK they are :) |
| 16:03.37 | leifmadsen | it's the flight to HK that is expensive I guess ;) |
| 16:03.47 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:04.33 | DelphiWorld | leifmadsen: only this? is installed |
| 16:04.51 | leifmadsen | DelphiWorld: what are you trying to install? |
| 16:04.55 | DelphiWorld | leifmadsen: compiling;) |
| 16:05.02 | leifmadsen | obviousl |
| 16:05.04 | DelphiWorld | leifmadsen: dahdi complete |
| 16:05.14 | leifmadsen | I've only needed the headers for that... unless they changed something |
| 16:05.22 | leifmadsen | you have the headers for your currently running kernel? |
| 16:05.30 | leifmadsen | try changing "header" to "source" I guess |
| 16:05.53 | DelphiWorld | leifmadsen: is installing i think |
| 16:06.40 | DelphiWorld | leifmadsen: after make/make install i do dahdicfg? |
| 16:06.53 | leifmadsen | I don't know... I don't do hardware very much |
| 16:07.00 | p3nguin | hardwire: I decided since you suggested that my phones (which are configured as peers with dynamic host) needed to have a static host setting to work, I would set host=<phone's IP address> just to see what happens... |
| 16:07.02 | leifmadsen | for dahdi, it's just 'make all' I thought |
| 16:07.46 | p3nguin | hardwire: chan_sip.c:9374 register_verify: Peer 'somename' is trying to register, but not configured as host=dynamic |
| 16:07.56 | hardwire | fix it! |
| 16:08.03 | p3nguin | hardwire: chan_sip seems to want it to be dynamic. |
| 16:08.23 | leifmadsen | the peer can't register if host= is set to dynamic |
| 16:08.31 | hardwire | p3nguin: I'm not really sure what you're trying to accomplish |
| 16:08.34 | hardwire | at all |
| 16:08.38 | hardwire | even if it's just beefing with me |
| 16:09.03 | hardwire | you can make all your phones peers.. just don't have them register.. |
| 16:09.07 | hardwire | that part seemed obvious |
| 16:09.10 | p3nguin | I had it dynamic, but changed it upon your suggestion that peers need to be static. It it now static and this message is generated, ending in failure. |
| 16:09.25 | p3nguin | it is, rather |
| 16:09.30 | hardwire | just shakes his head |
| 16:10.01 | p3nguin | Either I missed something, or the suggestion that peers need to be static was not complete. |
| 16:10.17 | DelphiWorld | thanks p3nguin, le! |
| 16:10.22 | DelphiWorld | thanks p3nguin, leifmadsen! |
| 16:10.40 | hardwire | p3nguin: this was all about another user right? not you're specific scenario? |
| 16:10.41 | hardwire | your |
| 16:10.42 | hardwire | haha |
| 16:10.46 | hardwire | I know you hate that.. sorry |
| 16:11.51 | hardwire | either way. I have several phone devices configured as peers for testing reasons.. and if omni__ had a good reason for configuring it that way then it shouldn't get in his way |
| 16:12.06 | hardwire | many things won't work for him.. but calls will |
| 16:13.02 | hardwire | as long as he has it configured right on both the phone and the asterisk box |
| 16:13.03 | p3nguin | hardwire: I simply took your word that peers should be set static and not dynamic. I changed a phone's peer definition accordingly, and it ended with failure as listed above. I'm just trying to understand your statement that peers need to be static. |
| 16:13.18 | hardwire | p3nguin: go take a walk off a short iceberg? |
| 16:13.41 | p3nguin | How is that productive? |
| 16:14.17 | hardwire | Why should it be? |
| 16:14.30 | hardwire | you like your little battles in here.. dunno why. |
| 16:14.38 | p3nguin | Or is this your way of saying you're offended? |
| 16:15.19 | hardwire | I am actually offended that you would just think that would do what you want.. as experienced as you are.. then you bring it back up to me as if it should have just because I said so. |
| 16:15.27 | hardwire | There's more too it and I honestly think you knew that. |
| 16:16.11 | hardwire | You're just being a stinker.. I have 0 idea why. |
| 16:16.39 | p3nguin | No, you're wrong about that. I was making an honest effort to follow your suggestion, and now you're being an ass about it. |
| 16:17.47 | p3nguin | If there was more to it, you could have said that instead of taking offense unnecessarily. |
| 16:21.27 | hardwire | Ni. |
| 16:21.29 | bmoraca_work | drama in the asterisk channelz! |
| 16:22.12 | hardwire | This is all stemming from past conversations. |
| 16:22.34 | DelphiWorld | how do i configure systel.conf for dahdi |
| 16:22.39 | DelphiWorld | using my te120? |
| 16:22.39 | hardwire | Meh.. I got upset and I shouldn't have.. I just didn't want to outright say "B.S." |
| 16:22.55 | DelphiWorld | leifmadsen: or p3 Acrony config? |
| 16:24.03 | p3nguin | <@leifmadsen> the peer can't register if host= is set to dynamic <-- chan_sip said "Peer is trying to register, but not configured as host=dynamic" ... meaning that it can only register if it IS set to dynamic. |
| 16:24.32 | p3nguin | These things don't add up, and I would like to know why. |
| 16:24.50 | hardwire | I wasn't concerned with what leif said. |
| 16:25.10 | hardwire | I just assumed, poorly, that people knew peers don't/can't register and that host would have to be set to a static ip/dns |
| 16:25.23 | hardwire | infact.. i said it like 3 times. |
| 16:25.30 | hardwire | I really don't want to go back and look |
| 16:25.56 | DelphiWorld | tzafrir_laptop: any clue? |
| 16:26.24 | tzafrir_laptop | DelphiWorld, I normally just generate it with dahdi_genconf |
| 16:26.35 | tzafrir_laptop | Is it ISDN? |
| 16:26.42 | tzafrir_laptop | E1, I suppose |
| 16:27.06 | DelphiWorld | tzafrir_laptop: yes, a pri E1 |
| 16:27.15 | p3nguin | And that's what doesn't add up for me. My phones (type=peer) do register and use host=dynamic. If this is a sensitive area for some reason, just forget the whole thing. |
| 16:27.53 | tzafrir_laptop | DelphiWorld, if the card is set as E1, dahdi_genconf should generate a proper configuration for it |
| 16:27.55 | *** join/#asterisk MAbbas (~abbas@203.215.177.194) |
| 16:28.05 | tzafrir_laptop | it's default is for CPE ("TE") |
| 16:28.07 | bmoraca_work | you don't need to register to a peer when it is configured with a static IP (asterisk already knows where to send packets) |
| 16:28.50 | MAbbas | Hi All, anybody know where dialplan "Log()" goes to? |
| 16:28.56 | p3nguin | Hmm... I'm not registering TO a peer... the phones are peers and register TO asterisk. Perhaps that was the confusion that led here? |
| 16:29.01 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 16:29.16 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:29.38 | bmoraca_work | p3nguin: the phones are trying to register to a PEER configured in asterisk. if the PEER configured in asterisk is set with a static host, it does not need a device to register to it. |
| 16:29.54 | DelphiWorld | tzafrir_laptop: no in dahdi_channel.conf but in /etc/dahdi/system.conf |
| 16:30.05 | bmoraca_work | p3nguin: the only point of registering to a SIP proxy is to let that SIP proxy know where that named peer resides. setting a static host accomplishes the same thing. |
| 16:30.21 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 16:30.22 | *** join/#asterisk Tim_Toady (~moi@62.1.243.39.dsl.dyn.forthnet.gr) |
| 16:30.23 | p3nguin | bmoraca_work: I agree. |
| 16:30.36 | bmoraca_work | p3nguin: right, so what's the point? you're trying to do something that makes no sense. |
| 16:31.02 | bmoraca_work | perhaps asterisk's handling of such case could be a little cleaner, but the fact is that trying to register to a static peer is pointless. |
| 16:31.20 | bmoraca_work | p3nguin: turn "Register with proxy" off on your phone and it will work just the same. |
| 16:31.33 | bmoraca_work | (the same as if it was registering with a dynamic peer, that is) |
| 16:32.50 | hardwire | anybody know a US ITSP that lets me set isup-oli? |
| 16:32.59 | hardwire | needs to pass ani 70 |
| 16:34.10 | tzafrir_laptop | DelphiWorld, by default it creates /etc/dahdi/system.conf and /etc/asterisk-dahdi-channels.conf |
| 16:35.32 | p3nguin | bmoraca_work: I have my phones' peer definitions set to type=peer and host=dynamic. The phones register; they work. The point was that hardwire said during another conversation that peers are to be configured statically. I changed my entries to use the IP addresses rather than being dynamic, but asterisk then complains that the peer is trying to register but is not set to dynamic. |
| 16:36.15 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:36.40 | bmoraca_work | p3nguin: so you fucked up a config ideology that hardwire mentioned offhand in a previous conversation and you're starting a fight over it? good lord, man, get a job. |
| 16:37.41 | hardwire | actually.. I put up the fight :) |
| 16:37.52 | bmoraca_work | oh, heh |
| 16:38.27 | p3nguin | bmoraca_work: I then provide this information in an attempt to understand the original claim that peers are not to be configured dynamically, like I had mine configured. I took it to mean that I had configured mine incorrectly. I didn't want to fight over anything -- just needed to comprehend the statement. |
| 16:38.54 | hardwire | p3nguin: any idea why you are using type=peer for phones? |
| 16:39.09 | hardwire | sorry.. for phones that register. |
| 16:39.35 | p3nguin | hardwire: Absolutely. [tk]d-fender told me that phones should be peers, never friends (except with few exceptions). |
| 16:39.37 | bmoraca_work | p3nguin: like i said, it's a config ideology. some people prefer "friend" for phones and "peer" for trunk connections. the bottom line is that there isn't really any difference between the two anymore. |
| 16:40.25 | hardwire | there is |
| 16:40.31 | hardwire | friends match auth names |
| 16:40.59 | hardwire | peers match IP |
| 16:41.10 | hardwire | this is based off of my own experience |
| 16:41.27 | hardwire | I had trouble using authenticated connections between machines when using peers |
| 16:41.49 | hardwire | mostly because I needed several authenticated connections between two machines and they needed to be distinctly identified |
| 16:41.53 | bmoraca_work | hardwire: match_auth_username=yes ftw |
| 16:42.13 | hardwire | bmoraca_work: that doesn't enable peers to match auth, afaik |
| 16:42.19 | bmoraca_work | yes it does |
| 16:42.26 | hardwire | interesting. |
| 16:42.37 | hardwire | I thought it just changed how the username was gleaned. |
| 16:42.39 | bmoraca_work | i ran in to the same problem as you (single machine, multiple peer auths) and that fixed all of my issues |
| 16:43.11 | hardwire | yeh.. i read it wrong |
| 16:43.17 | hardwire | that would have been useful |
| 16:43.21 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.220.217) |
| 16:43.48 | hardwire | actually.. I was using both sip and IAX.. it wouldn't have helped with IAX |
| 16:44.05 | hardwire | either way.. I landed on friends as being a good option.. with static defaultip |
| 16:44.26 | hardwire | and rsa auth.. since that seemed to be easy enough |
| 16:47.11 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 16:47.38 | Kobaz | so umm |
| 16:47.40 | Kobaz | Call Center Tie T1 (DS1 circuit 01C20) is setup for protocol a ( AT&T custom), b8zs / esf. |
| 16:47.46 | Kobaz | no wonder we were having problems |
| 16:47.54 | Kobaz | what the hell is protocol a |
| 16:49.08 | hardwire | would the polar opposite of a be z or m? |
| 16:50.21 | Kobaz | so... we were set up as NI2 with a sangoma card |
| 16:50.25 | Kobaz | it's amazing it worked at all |
| 16:51.14 | MAbbas | how do I print debug messages in asterisk dialplan? |
| 16:51.42 | hardwire | bmoraca_work: match_auth_username comment in sip.conf wouldn't have lead me to that idea... crazy.. I'm going to look at the changelog real quick |
| 16:52.18 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 16:52.38 | hardwire | hmm |
| 16:52.57 | hardwire | I wonder if your fixup is inadvertant or not. |
| 16:54.27 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 16:54.39 | DelphiWorld | where dahdicfg is located? |
| 16:54.40 | bmoraca_work | hardwire: it tells asterisk (at least, this is how i read it) to match the peer name specified in the SIP packet, rather than the IP address in the From: field |
| 16:54.58 | bmoraca_work | hardwire: which would be consistent with individual peers not being able to be told apart when registered from the same IP address |
| 16:56.06 | bmoraca_work | hardwire: the hardware i was using was an Adtran TA908e and what was happening was that each of my calls were not coming through as their independent registrations, but rather as the lowest numbered peer name (consistent with asterisk's peer matching when only IPs are available) |
| 16:56.08 | hardwire | ok. I was thinking differently.. that it would simply use the username in the digest rather than the user in the From .. since user is in there as well. |
| 16:56.10 | bmoraca_work | using that setting fixed it |
| 16:56.47 | bmoraca_work | hardwire: i don't believe asterisk uses the username in From, because that could technically be anything |
| 16:56.47 | hardwire | and that doing so inadvertantly resolved a problem you were having. |
| 16:57.04 | hardwire | bmoraca_work: this is a question for the great digifolk |
| 16:57.09 | DelphiWorld | any dahdi guy |
| 16:57.29 | MAbbas | [TK]D-Fender: any idea, how do I log messeages to logfiles in asterisk. I have tried dialplan application "log()". But I am unable to find my logged messages |
| 16:58.00 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:2c4e:5af5:41da:8e9b) |
| 16:58.23 | bmoraca_work | hardwire: my experience is that From: is very unreliable for anything except IP address. having it match the username in the digest instead of the IP address gives me the ability to match based on peer name instead of IP address, which is what I wanted. type=friend may do the same thing without needing the extra setting. |
| 16:59.21 | bmoraca_work | for instance, my AS5400 gives the source ANI of the calling party in the From: header. not useful for any kind of authentication. the TA908e does pretty much the same thing. |
| 16:59.52 | hardwire | with those types of situations you would typically use IP auth and a peer type |
| 16:59.57 | hardwire | yeh |
| 17:00.20 | *** join/#asterisk edwin_quijada (~macaruchi@200.26.172.50) |
| 17:00.24 | hardwire | I understand how this resolves the problem you were having now that I know the ANI was changing |
| 17:00.39 | hardwire | it's all clicking now |
| 17:00.44 | edwin_quijada | Somebody can put me in the rigth direction to get FastAGI works in Windows? |
| 17:00.48 | hardwire | but typically in a phone situation.. the ani never changes |
| 17:00.55 | bmoraca_work | right |
| 17:01.03 | bmoraca_work | and i usually have my phones as type=friend |
| 17:01.11 | [TK]D-Fender | MAbbas: Show me a call using it, and show me what logs you're looking in. |
| 17:01.14 | hardwire | I like that they added this option.. it's better IMHO to depend on the digest section |
| 17:01.15 | Katty | ohai |
| 17:01.52 | Katty | what's the good word, gents. |
| 17:02.37 | [TK]D-Fender | edwin_quijada: Go write an app in a windows hosted language. The End. |
| 17:02.52 | bmoraca_work | hardwire: indeed. was tearing my hair out for the better part of an hour before i figured that issue out. |
| 17:03.38 | DelphiWorld | i have: /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting |
| 17:03.41 | bmoraca_work | what was weirdest is that in all of my testing, i never ran into the issue |
| 17:03.53 | DelphiWorld | Katty: ;) |
| 17:04.05 | edwin_quijada | [TK]D-Fender: but who respond to that program? |
| 17:04.35 | [TK]D-Fender | edwin_quijada: Huh? It LISTENS for connections on the FastAGI socket. |
| 17:04.54 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.140.212.dsl.dyn.forthnet.gr) |
| 17:07.51 | *** join/#asterisk Tim_Toady (~moi@77.49.29.230.dsl.dyn.forthnet.gr) |
| 17:08.18 | Ad-Hoc | hoi |
| 17:10.23 | MAbbas | is agi_uniqueid and call Id the same? |
| 17:13.07 | _omer | my asterisk 1.4.29 seems not getting details from say.conf ... I have commented out almost everything but saynumber() still works the same...I am using mode=new under [general] in say.conf |
| 17:17.22 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 17:18.27 | hackeron | hey, I'm experiencing heavy echo even though I have a hardware echo canceller. I see "Booting VPMADT032" and "VPM present and operational (Firmware version 120)" in dmesg but I'm still experiencing heavy echo - any suggestions? |
| 17:18.29 | bmoraca_work | what's a good text-to-speech (free) engine for asterisk 1.6.2? |
| 17:18.55 | hackeron | it's a digium wctdm24xxp card with a hardware echo canceller |
| 17:19.12 | [TK]D-Fender | bmoraca_work: Nothing "good". Festival is pretty much it. |
| 17:19.20 | bmoraca_work | great |
| 17:20.04 | hackeron | I have no echo canceller set in /etc/dahdi/system.conf and echocancel = yes in chan_dahdi.conf |
| 17:20.07 | hackeron | is that right? |
| 17:20.15 | Kobaz | bmoraca_work: cepstral is not too expensive |
| 17:21.47 | bmoraca_work | lol...festival is in the CentOS default repos |
| 17:22.04 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 17:26.36 | hackeron | anyone? |
| 17:27.27 | hardwire | bueller? |
| 17:27.30 | tzafrir_laptop | hackeron, no. You need to set echocancel for every channel |
| 17:28.20 | hackeron | tzafrir_laptop: I have echocancel = yes |
| 17:28.21 | hackeron | channel = 1-4 |
| 17:28.24 | tzafrir_laptop | hackeron, hmm... you use a hardware ec... |
| 17:28.33 | hackeron | tzafrir_laptop: yeh, I do |
| 17:28.38 | hackeron | tzafrir_laptop: a VPMADT032 |
| 17:34.22 | hardwire | did you say the echo cancel was disabled in dahdi/system.conf? |
| 17:36.39 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 17:37.25 | DelphiWorld | root@freswitch:/usr/src/freeswitch# lsmod | grep dahdi |
| 17:37.25 | DelphiWorld | dahdi_voicebus 51648 1 wcte12xp |
| 17:37.25 | DelphiWorld | dahdi 211080 5 wcte11xp,wcte12xp,dahdi_voicebus |
| 17:37.25 | DelphiWorld | crc_ccitt 10112 1 dahdi |
| 17:37.25 | DelphiWorld | root@freswitch:/usr/src/freeswitch# |
| 17:38.39 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 17:46.32 | *** join/#asterisk vgster (~vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:51.41 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 17:51.50 | DelphiWorld | while runing /etc/init.d/dahdi start: |
| 17:51.51 | DelphiWorld | Loading DAHDI hardware modules: |
| 17:51.51 | DelphiWorld | wcte12xp: done wcte11xp: done |
| 17:51.51 | DelphiWorld | Running dahdi_cfg: . |
| 17:51.57 | DelphiWorld | Loading DAHDI hardware modules: |
| 17:51.57 | DelphiWorld | wcte12xp: done wcte11xp: done |
| 17:51.57 | DelphiWorld | ./usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting |
| 17:51.57 | DelphiWorld | Running dahdi_cfg: . |
| 17:52.16 | DelphiWorld | euro ISDN |
| 17:54.32 | bmoraca_work | you can't use festival without a soundcart? |
| 17:54.36 | bmoraca_work | card* |
| 17:57.00 | hardwire | bmoraca_work: false |
| 17:57.12 | bmoraca_work | it's telling me it can't open /dev/dsp |
| 17:57.28 | hardwire | are you telling it to write to a file? |
| 17:57.53 | bmoraca_work | i can use it from within asterisk without errors, but i get no audio |
| 17:58.04 | hardwire | :( |
| 17:58.13 | bmoraca_work | when i run the "festival" program itself and use (SayText "hello world"), it tells me it can't open /dev/dsp |
| 17:58.22 | *** join/#asterisk wunderkin (~kbockman@pool-71-106-236-25.lsanca.dsl-w.verizon.net) |
| 17:58.27 | hardwire | cause by default it attempts to use the sound card |
| 17:58.36 | hardwire | it can output to a pipe, file, or dsp |
| 18:01.11 | hardwire | bmoraca_work: check out the text2wave program that is insalled with festival |
| 18:01.46 | bmoraca_work | i want to use festival from within asterisk...not sure if text2wave is going to be appropriate |
| 18:01.55 | tzafrir_laptop | DelphiWorld, a symptom of missing libusb-devel at build time |
| 18:02.06 | tzafrir_laptop | if you don't have an astribank, it should be harmless |
| 18:02.36 | hardwire | bmoraca_work: are you issuing an Answer() first? |
| 18:02.39 | p3nguin | bmoraca_work: You just need the proper festivalrc and it'll work fine. |
| 18:04.18 | hardwire | the festival app should, if it hasn't already, implement disk caching. |
| 18:05.06 | bmoraca_work | p3nguin: i don't have the "aplay" application that is referenced by those festivalrc posts...so, i'm not sure how well that's going to help |
| 18:05.20 | DelphiWorld | tzafrir_laptop: span=1,1,0,cas,hdb3 |
| 18:05.26 | DelphiWorld | tzafrir_laptop: what is this line? |
| 18:05.30 | p3nguin | bmoraca_work: aplay is part of alsa. |
| 18:05.45 | p3nguin | YOu can have alsa without a sound card. |
| 18:05.49 | tzafrir_laptop | DelphiWorld, cas? That's odd. Id should be ccs |
| 18:05.54 | bmoraca_work | ahh, there it is. had to updatedb before it showed up |
| 18:06.01 | DelphiWorld | tzafrir_laptop: yes i know |
| 18:06.04 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 18:06.04 | tzafrir_laptop | span=1,1,0,ccs,hdb3,crc4 |
| 18:06.13 | DelphiWorld | tzafrir_laptop: span=1,1,0,ccs,hdb3,crc4 that is corect? |
| 18:06.19 | tzafrir_laptop | any chance you try to configure it as R2 ? |
| 18:06.26 | bmoraca_work | does that have to be in festivalrc or can i put it in festival.scm? |
| 18:06.27 | p3nguin | Actually, aplay is part of alsa-utils, but meh. |
| 18:06.36 | tzafrir_laptop | yes, you last line is correct |
| 18:06.45 | DelphiWorld | tzafrir_laptop: no, no in this server but in another server and i remember this line |
| 18:06.51 | *** part/#asterisk badweather (~brentw@modemcable176.244-81-70.mc.videotron.ca) |
| 18:07.22 | DelphiWorld | tzafrir_laptop: but anyway the dahdi device is unable to operate |
| 18:07.42 | tzafrir_laptop | DelphiWorld, what error do you get? |
| 18:08.20 | DelphiWorld | tzafrir_laptop: chan 1/2/3/4 not found... |
| 18:09.14 | p3nguin | I suppose it wouldn't hurt to make the changes to festival.scm, but I typically throw the stuff into my .festivalrc without too much trouble. |
| 18:09.19 | tzafrir_laptop | DelphiWorld, do you actually have /dev/dahdi/1 ? |
| 18:09.52 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 18:10.31 | DelphiWorld | tzafrir_laptop: yeah, i have from 1 to 31 |
| 18:10.54 | bmoraca_work | great |
| 18:10.56 | bmoraca_work | more errors |
| 18:11.07 | bmoraca_work | alsa complaining about not being able to find a soundcard |
| 18:11.21 | DelphiWorld | tzafrir_laptop: span=1,1,0,ccs,hdb3,crc4 |
| 18:11.26 | DelphiWorld | tzafrir_laptop: bchan=1-15,17-31 |
| 18:11.28 | Naikrovek | alsa and pulse are reasons why linux still sucks on the desktop. my god those are both abysmal |
| 18:11.31 | DelphiWorld | tzafrir_laptop: dchan=16 |
| 18:11.36 | tzafrir_laptop | DelphiWorld, please pastebin the output of lsdahdi |
| 18:12.16 | Naikrovek | most of the time, they work. when they don't... well best of luck to ya, fella |
| 18:12.29 | DelphiWorld | tzafrir_laptop: i can't read it all i need to elarge my putty window but i can't |
| 18:12.35 | DelphiWorld | tzafrir_laptop: may you ssh? |
| 18:14.07 | DelphiWorld | please anyone fix my euroISDN problem with dahdi |
| 18:16.26 | *** join/#asterisk QubeZ (~qube@64.128.254.34) |
| 18:16.31 | QubeZ | hello all |
| 18:17.10 | QubeZ | i have 2 queues assigned to a user with penalty null and penalty 10, i want to add a third with higher priority... do I need to change the null to something like 9 then add the new queue as 8 or can I use negative numbers (-1)? |
| 18:18.31 | p3nguin | I just realized that I have an onboard sound card, so maybe that's why I didn't have too much trouble with festival. The sound card isn't being used for sound, but it is present. |
| 18:19.32 | leifmadsen | shouldn't matter |
| 18:19.41 | *** join/#asterisk Wildy (~simba@194.186.220.116) |
| 18:19.44 | leifmadsen | sound card isn't used to create the audio |
| 18:20.58 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 18:21.43 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 18:21.49 | p3nguin | Just a thought... albeit not a reasonable one. |
| 18:22.22 | bmoraca_work | more work than it's worth at this point |
| 18:22.53 | DelphiWorld | tzafrir_laptop: in /dev/dahdi: http://asterisk.pastebin.com/qW9Yk1FP |
| 18:24.52 | p3nguin | Here's my festival.scm: http://pastebin.com/gdFRYjg4 |
| 18:27.37 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:28.11 | *** join/#asterisk moy (~chatzilla@74.12.121.97) |
| 18:29.47 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 18:31.19 | bent_screwdriver | anyone ever submitted a feature request to polycom before? |
| 18:31.27 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 18:33.04 | Naikrovek | i havent' |
| 18:33.25 | Naikrovek | submit one for me would ya? iax2 support would be nice |
| 18:33.34 | *** join/#asterisk jmkgreen (~jmkgreen@82-71-41-166.dsl.in-addr.zen.co.uk) |
| 18:34.11 | *** join/#asterisk Wildy (~simba@194.186.220.116) |
| 18:35.06 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.77) |
| 18:35.08 | bent_screwdriver | i submitted a couple. if they're responsive i'll let the group know...likeley not. |
| 18:35.24 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:35.39 | jmkgreen | i've upgraded from 1.4 to 1.6 and now the AJAM interface is giving me 404 not found. I'm not sure what I'm missing..? |
| 18:35.52 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:38.08 | jmkgreen | and soon as I do that, i fix the damned thing |
| 18:40.42 | bent_screwdriver | isn't there some program that will write a rule to iptables if it sees too many failed sip login attempts in the logs? |
| 18:41.16 | [TK]D-Fender | bent_screwdriver: fail2ban <- |
| 18:41.33 | bent_screwdriver | ahhh...that rings a bell....work well? |
| 18:41.42 | *** join/#asterisk aidanna (~aidanna@67.211.23.182) |
| 18:41.47 | [TK]D-Fender | bent_screwdriver: apparently |
| 18:41.59 | bent_screwdriver | [TK]D-Fender: thx! |
| 18:42.41 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 18:43.27 | Knightfal | Hey guys any one notice in 1.4.30 that queues.conf "persistentmembers = yes" is not loading agents from astDB |
| 18:43.39 | Knightfal | on restart :) |
| 18:51.31 | bent_screwdriver | anyone have the backlight on the Polycom 650's fail? they're always on and I'm worried they'll fail after a while. any way to put the light on standby? |
| 18:51.53 | Naikrovek | bent_screwdriver: LEDs don't usually fail for years unless they're overdriven |
| 18:51.58 | Naikrovek | which could very well be the case |
| 18:52.30 | Naikrovek | a slightly overdriven LED will only last, say, a year. an LED running on the proper current and voltage has an MTBF somewhere in the decades |
| 18:52.45 | Naikrovek | but it should not stay on |
| 18:52.50 | Naikrovek | upgrade firmware if you can |
| 18:52.58 | Naikrovek | may have been fixed already |
| 18:53.18 | bent_screwdriver | i'm on the latest 3.2, well as of a couple weeks ago... |
| 18:54.13 | bent_screwdriver | do you have ones that turn off when not in use? |
| 18:54.15 | ChannelZ | They don't have a 'screen saver' mode that turns it off after a time? Perhaps the config got set incredibly high |
| 18:54.46 | bent_screwdriver | i'll look thorugh the config/admin manual to see if i see anything. thx. |
| 18:55.37 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:56.39 | mdg | what a guy |
| 18:57.10 | leifmadsen | Naikrovek: there is an option for that |
| 18:57.20 | leifmadsen | errrr.... |
| 18:57.24 | Naikrovek | heh |
| 18:57.27 | leifmadsen | bent_screwdriver: there is an option for that to turn them off on idle |
| 18:57.27 | Naikrovek | none of my phones light up |
| 18:57.55 | bent_screwdriver | @leifmadsen: thx. looking throgh the manual now to find the setting |
| 18:58.09 | leifmadsen | <PROTECTED> |
| 18:58.15 | leifmadsen | in <user_preferences |
| 18:58.17 | leifmadsen | of sip.cfg |
| 18:58.33 | leifmadsen | about line 182 of my configuration |
| 18:58.50 | bent_screwdriver | @leifmadsen: perfect! found what you're talking about. Thanks! |
| 18:59.04 | leifmadsen | set to 0 and the backlight will turn off on idle |
| 19:01.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:06.35 | bent_screwdriver | @leifmadsen: that setting is working great. thats for the help. |
| 19:06.41 | leifmadsen | np! |
| 19:06.55 | leifmadsen | I just modified that the other day myself since I got my first backlit phone :) |
| 19:07.00 | leifmadsen | G.722 sounds amazing btw |
| 19:07.52 | hardwire | it sort of sounds like G.729 when you say it out loud.. except for the nine part |
| 19:08.07 | Qwell | hardwire: and the a |
| 19:08.10 | leifmadsen | NEIN! |
| 19:08.10 | bmoraca_work | stupid as5400 |
| 19:08.27 | hardwire | Qwell: oh yeh |
| 19:08.32 | hardwire | freaking annex |
| 19:08.51 | hardwire | Qwell: maybe it's just a canadian codec? |
| 19:08.55 | hardwire | g.729a |
| 19:10.14 | bmoraca_work | wtb my as5400 to hunt the way it's supposed to! |
| 19:13.33 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.35.181) |
| 19:17.43 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 19:18.16 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.17 | leifmadsen | The second round of release candidates for Asterisk 1.6.0.27, 1.6.1.19, and 1.6.2.7 are now available! http://www.asterisk.org/node/49928 |
| 19:18.43 | leifmadsen | Asterisk-Addons releases 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1 are now available! http://www.asterisk.org/node/49929 |
| 19:20.59 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 19:22.09 | *** join/#asterisk xpot-mobile (~james@66.60.101.91) |
| 19:32.35 | edwin_quijada | somebody has used fastagi with asterisk? |
| 19:34.53 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 19:35.51 | *** join/#asterisk aschneiderg (~aschneide@190.26.40.249) |
| 19:36.04 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 19:37.45 | [TK]D-Fender | edwin_quijada: Are yougoing to come up with a different question? This has been answered a dozen times now. |
| 19:39.21 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 19:43.15 | *** join/#asterisk RobH_ (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 19:44.29 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 19:47.55 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:47.55 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:48.32 | *** join/#asterisk jasonwert-work (~jasonwert@adsl-99-27-170-70.dsl.klmzmi.sbcglobal.net) |
| 19:54.31 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
| 19:55.14 | *** join/#asterisk smooth_penguin (~smoove@59.95.36.3) |
| 19:55.33 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:00.33 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 20:00.37 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:2c4e:5af5:41da:8e9b) |
| 20:21.30 | *** join/#asterisk Joel (~jjshoe@unaffiliated/joel) |
| 20:21.41 | Joel | anyone happen to know of a softphone that allows you to add custom sip headers? |
| 20:25.04 | Knightfal | Not really. What are you trying to accomplish? |
| 20:25.33 | Joel | I'm trying to add a custom sip header to a phone call :) |
| 20:25.38 | Joel | I'll just bounce it through a relay. |
| 20:25.43 | Joel | Was just hoping to save some setup |
| 20:28.14 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 20:29.44 | [TK]D-Fender | checkout time, BBIAB |
| 20:32.21 | *** part/#asterisk Joel (~jjshoe@unaffiliated/joel) |
| 20:34.40 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 20:46.13 | *** join/#asterisk CatLynx (~Booga@173-11-77-182-SFBA.hfc.comcastbusiness.net) |
| 20:47.31 | CatLynx | gave up on trying to get MWI working on the AT&T cordless phone |
| 20:49.30 | Knightfal | Hey guys any one notice in 1.4.30 that queues.conf "persistentmembers = yes" is not loading agents from astDB |
| 20:59.32 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 21:00.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:01.42 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 21:04.04 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 21:06.25 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 21:13.42 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 21:15.53 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 21:18.17 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 21:18.27 | *** join/#asterisk rare1980_ (~rare1980@115.186.10.40) |
| 21:19.29 | *** part/#asterisk TommyBotten (tommy@broken.pipe.no) |
| 21:28.49 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 21:29.25 | *** join/#asterisk Alagar (~Administr@122.164.33.136) |
| 21:37.28 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 21:38.36 | Micc_ | I'm trying to build 1.6.2.6, but I keep getting this libxml2 dev package needs to be installed when I run ./configure. But it is installed. xml2-config exists and apt-get install libxml2-dev says its already installed. |
| 21:41.02 | Corydon76-dig | Micc_: platform? |
| 21:41.13 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 21:41.27 | Micc_ | ubuntu server 9.10 |
| 21:42.20 | Corydon76-dig | Micc_: what's the actual error in config.log? |
| 21:43.48 | *** join/#asterisk citrus2 (~citrus2@mail.serviceobjects.com) |
| 21:44.39 | *** join/#asterisk lernest (~ircap@c-76-101-181-248.hsd1.fl.comcast.net) |
| 21:44.49 | citrus2 | i'm looking for a solution for sip failover, to auto detect if one provider is down to use another. anyone have any ideas? |
| 21:44.54 | citrus2 | outbound only |
| 21:45.35 | Corydon76-dig | Turn on qualify |
| 21:46.00 | bmoraca_work | citrus2: qualify + a second Dial command right after the first...if the first isn't available, the second will execute |
| 21:46.22 | *** join/#asterisk ecolitan (~aaron@li57-124.members.linode.com) |
| 21:48.52 | citrus2 | interesting |
| 21:48.55 | citrus2 | thanks i will look into that |
| 21:49.37 | Kobaz | http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.27-rc1 404 not found |
| 21:49.40 | Kobaz | :( |
| 21:50.04 | Kobaz | oh, it's rc2 |
| 21:50.11 | Kobaz | the link is wrong on the release page |
| 21:51.40 | lernest | im doing a fresh installation and after doing ./configure I get this |
| 21:51.40 | lernest | lernest ¦ The configure script must be executed before running 'make'. |
| 21:51.40 | lernest | lernest ¦ **** Please run "./configure". |
| 21:52.07 | Qwell | lernest: then you didn't run it, or didn't run it properly |
| 21:52.22 | lernest | [Qwell] I did |
| 21:52.22 | Qwell | or it wasn't successful, and there would have been an error |
| 21:53.17 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 21:53.41 | lernest | [Qwell] I get this at the end |
| 21:53.48 | lernest | configure: error: *** termcap support not found (on modern systems, this typically means the ncurses development package is missing) |
| 21:54.00 | Qwell | well, there you go |
| 21:54.32 | lernest | [Qwell] what shoul I do |
| 21:55.58 | Qwell | typically means the ncurses development package is missing |
| 21:59.28 | lernest | [Qwell] ok thanks |
| 22:02.30 | leifmadsen | Kobaz: my bad! fixed on asterisk.org (per #asterisk-dev) |
| 22:02.45 | Kobaz | heh |
| 22:04.46 | yonahw | just installed asterisk 1.4.30 dialplan reload --> no such command 'dialplan reload' |
| 22:04.55 | yonahw | has this been removed? |
| 22:05.00 | Kobaz | yonahw: module load pbx_config.so |
| 22:05.04 | Kobaz | no, not from 1.4 |
| 22:05.31 | yonahw | Kobaz: thank you |
| 22:05.49 | Kobaz | yonahw: that works? edit /etc/asterisk/modules.conf... and add load => pbx_config.so |
| 22:06.12 | yonahw | yes thank you |
| 22:06.26 | Kobaz | module load order is not guaranteed |
| 22:06.47 | Kobaz | restart and make sure it loads up... you may need to tweak the orer |
| 22:06.49 | Kobaz | order |
| 22:07.15 | yonahw | i didnt make the samples |
| 22:07.23 | yonahw | and dont yet have a modules.conf |
| 22:07.32 | Kobaz | ah |
| 22:07.39 | yonahw | havent touched asterisk in a few years and figured id be better off being explicit |
| 22:07.55 | yonahw | so as i find things i am missing i will add them in |
| 22:08.08 | yonahw | of course modules.conf was bound to be an early issue |
| 22:08.25 | Kobaz | well, you're going to be missing a whole lot without the default modules.conf |
| 22:08.30 | Kobaz | basically, you'll be missing everything |
| 22:08.43 | yonahw | indeed |
| 22:08.46 | Kobaz | it's much better to start off with the defaults and then remove things you dont need... there's lots of modules |
| 22:09.35 | yonahw | i hear where you are coming from but i find it easier to learn this way |
| 22:09.46 | yonahw | if everything is there how do i know what i do or dont need |
| 22:10.14 | Kobaz | i guess |
| 22:10.36 | yonahw | anyhow thanks |
| 22:10.43 | Kobaz | np |
| 22:16.01 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 22:38.41 | *** join/#asterisk infobot (ibot@rikers.org) |
| 22:38.41 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 22:39.45 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 22:40.39 | *** join/#asterisk jsjc (~chatzilla@115.131.203.5) |
| 22:41.22 | jsjc | I have some issues with queues I dont know why when get sto the queue the call hangs up... I wonder why and I would like to debug it but dont know how I can approach that... any help? |
| 22:44.56 | jsjc | It is strange.. all goes good I can sign up into the queue, but is all when it comes to the queue that call gets terminated |
| 22:49.03 | Kobaz | if you had an understandable question, it would probably help |
| 22:49.27 | Kobaz | "I dont know why when get sto the queue the call hangs up" <--- what does that mean? |
| 22:52.53 | *** join/#asterisk ecolitan (~aaron@li57-124.members.linode.com) |
| 22:56.02 | bmoraca_work | wonder if anyone knows this offhand before i configure it in a lab environment: if I have two Asterisk boxes reading the same realtime SIP table and a peer registers with one and then registers with the other without the first registration failing, will DUNDI update to the newest box or will it advertise out both boxes? |
| 22:59.59 | p3nguin | jsjc: core set verbose 10 |
| 23:00.21 | p3nguin | jsjc: Make a call, allow it to reach the queue where the hangup occurs. |
| 23:00.37 | jsjc | ok lets check! |
| 23:00.43 | jsjc | i had vervose to 5 so needs to be 10! |
| 23:00.47 | jsjc | hehe |
| 23:00.55 | p3nguin | jsjc: probably not |
| 23:01.04 | jsjc | verbose ME!!! |
| 23:01.11 | p3nguin | jsjc: But you do need to look at and interpret the output. |
| 23:01.28 | *** join/#asterisk Faithful (~Faithful@121.91.185.231) |
| 23:03.07 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:03.18 | jsjc | <PROTECTED> |
| 23:03.31 | jsjc | thats it... |
| 23:03.37 | p3nguin | The only thing? |
| 23:03.40 | jsjc | cannot interpret that.. |
| 23:03.45 | jsjc | becasue sounds just hang up |
| 23:03.53 | p3nguin | The phone is a SIP phone? |
| 23:04.55 | jsjc | yes where i am calling from is SIP |
| 23:05.01 | p3nguin | sip set debug on |
| 23:05.11 | p3nguin | Make the call again. |
| 23:05.26 | jsjc | i will pastebin it |
| 23:05.36 | p3nguin | Great idea! |
| 23:05.44 | jsjc | hehe |
| 23:06.21 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:08.24 | jsjc | http://pastebin.com/4KPygPnj |
| 23:09.53 | Kobaz | verbosity levels suck |
| 23:09.58 | Kobaz | verbosity tokens ftw |
| 23:11.59 | p3nguin | I don't see anything jumping out saying "here's what went wrong." |
| 23:16.18 | jsjc | p3nguin: i will recheck extensions and queues conf see what its is right but makes it jump |
| 23:22.14 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:28.24 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 23:30.49 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:38.56 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 23:39.15 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 23:39.50 | *** join/#asterisk mr_ian_ (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 23:49.54 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:54.41 | *** join/#asterisk rdircio (~admin@201.137.45.224) |
| 23:59.15 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
