| 00:13.29 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 00:13.52 | *** join/#asterisk GameGamer43|Mac (~GameGamer@65.27.76.78) |
| 00:14.35 | WIMPy | Not much luck at all. No audio, except for dialtone/busy and no forther calls without reconnect. |
| 00:15.42 | WIMPy | chan_skinny used to barf on the 2nd call, but at least one was ok. |
| 00:17.18 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.168.251.dsl.dyn.forthnet.gr) |
| 00:19.33 | p3nguin | I use chan_sccp every day, so I know it's good. |
| 00:21.12 | WIMPy | There don't seem to much debug options. |
| 00:21.57 | WIMPy | But probably they both just don't like historic phones. It should be "fully supported" tho. |
| 00:22.20 | p3nguin | What phones? |
| 00:22.24 | WIMPy | It's an old 30vip. |
| 00:23.17 | p3nguin | Debug level goes from 1 - 10, so maybe 10 would be helpful. |
| 00:26.40 | WIMPy | It says "Open receive channel with format G.711 u-law"... but no network activity. |
| 00:27.28 | WIMPy | Then after some time the phone reboots. |
| 00:53.57 | p3nguin | I don't really know... I use Cisco phones, which work pretty good with SCCP (not surprisingly). |
| 00:56.50 | WIMPy | I try to look into chan_skinny. That used to work before. But I'm not sure when. It might have been with 1.4. |
| 00:59.56 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 01:03.03 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 01:11.25 | p3nguin | I use 1.4, and chan_skinny sucks. |
| 01:12.15 | carrar | 0.1.0 or bust! |
| 01:13.06 | WIMPy | I have always been told to use SIP software with Cisco phones. But I only own that dinosaur, so I can't say much about it myself. |
| 01:13.42 | WIMPy | Anyway I have been able to place and receive calls with chan_skinny ... some time. |
| 01:14.00 | p3nguin | I started out using SIP on my 7940 and it works, but it lacks features. |
| 01:14.25 | p3nguin | I tested SCCP on the phone along with chan_skinny, and it barely worked at all. |
| 01:14.47 | WIMPy | I'm missing quite some features with SIP, but that's another story. |
| 01:16.38 | p3nguin | I moved back to SIP for a quite some time. Then someone recently talked me into trying chan_sccp-b, and I've been using it since. |
| 01:17.10 | p3nguin | It has nice features, so I don't see any good reason not to use it. |
| 01:17.39 | carrar | What SCCP version you running? |
| 01:19.33 | WIMPy | sccp-b? I tried SCCPv2. |
| 01:20.29 | drmessano | sccp-b works. Period. The others are all buggy |
| 01:21.16 | WIMPy | There are too many of them. Didn't realize that was a different verion again. |
| 01:23.11 | drmessano | Yep, there are. Sadly, the implementation in Asterisk isn't the best, so you're going to continue to see other attempts at it... (Like Fax and conferencing had been in the past) |
| 01:41.42 | WIMPy | Now the console shows OFFHOOK, DIALLING, INVALIDNUMBER as soon as I lift the receiver and DOWN when I place it again. It's getting worse and worse. |
| 01:43.10 | p3nguin | I've installed chan-sccp-b-svn 1246. |
| 01:44.39 | p3nguin | There are newer versions, but for some reason I ended up with 1246 when I checked out. |
| 01:45.16 | WIMPy | 3.0-RC1 |
| 01:45.40 | WIMPy | And it still doesn't support reload. |
| 01:46.18 | WIMPy | This is really not worth it. |
| 01:50.44 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 01:51.04 | mattwj2002 | hi guys |
| 01:51.17 | mattwj2002 | I need help with sip not asterisk |
| 01:51.25 | mattwj2002 | is there a good channel for that? |
| 01:52.15 | mattwj2002 | I am trying to get google voice to connect to gizmo5 to connect to my cell phone |
| 01:52.20 | p3nguin | Worth it? It should take you like a minute to do svn checkout and install the driver. |
| 01:52.35 | p3nguin | If a minute of your time isn't worth it, then do without SCCP. |
| 01:53.14 | WIMPy | I've tried enough versions and spent far too much time on the different configurations. |
| 01:54.24 | mattwj2002 | I can make outgoing sip calls |
| 01:54.25 | WIMPy | That's just a historic piece of a telephone anyway. |
| 01:54.40 | mattwj2002 | but not incoming....it doesn't work on wifi or 3G |
| 01:54.58 | *** join/#asterisk Circlefusion (~circlefus@74-130-62-234.dhcp.insightbb.com) |
| 01:55.25 | p3nguin | mattwj2002: Did you configure your gizmo account to forward calls to your cell phone number? |
| 01:56.00 | mattwj2002 | no |
| 01:56.17 | mattwj2002 | that would cost me money |
| 01:56.29 | mattwj2002 | I have an android sip client on my phone |
| 01:56.30 | p3nguin | mattwj2002: Did you configure your google voice account to forward calls to your gizmo sip number? |
| 01:57.17 | mattwj2002 | hmmm |
| 01:57.33 | mattwj2002 | oh I think I found the problem |
| 01:57.36 | mattwj2002 | one moment |
| 01:57.38 | p3nguin | If you are going to use a sip client on the phone, you'll have to have a defined sip uri that you can gizmo will be able to send calls. |
| 01:59.17 | p3nguin | I use gizmo to send calls to an asterisk system, so I have it forward all calls to mygizmonumber@myhostname, where the call is processed by the PBX. |
| 01:59.57 | mattwj2002 | oh |
| 02:00.01 | mattwj2002 | I bet that is what is wrong |
| 02:00.02 | mattwj2002 | :) |
| 02:01.45 | mattwj2002 | so where do I forward it to? |
| 02:03.04 | mattwj2002 | I think I got it p3nguin |
| 02:03.05 | mattwj2002 | :) |
| 02:04.48 | mattwj2002 | it works! |
| 02:04.55 | mattwj2002 | I had the dang number forwarded |
| 02:04.56 | mattwj2002 | :) |
| 02:05.17 | mattwj2002 | it was forward to a non-existing asterisk box |
| 02:08.26 | *** join/#asterisk guilhermebr (~Guilherme@201.47.183.137.dynamic.adsl.gvt.net.br) |
| 02:10.35 | WIMPy | Jup, missing forwarding indications is one of the things SIP is missing. |
| 02:32.47 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 02:55.44 | pwell | anyone know what system is the old school party lines use? Defcon for example and other from back in the day. It was always the same setup so it must have been either one guy or propriatary equiptment. "Welcome to the Board.." "You are in the Lobby" "Room 3" |
| 02:57.17 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 03:09.08 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 03:21.09 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 03:33.32 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
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| 04:25.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
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| 04:48.32 | daemon | hey guys just checking something |
| 04:48.55 | daemon | if I have asterisk running on my headgateway so it has 3 private interfaces xl[0-2] and one public vr0 |
| 04:49.04 | daemon | public r0 connects to random sip trunks etc |
| 04:49.31 | daemon | all my lan clients connect to asterisk via xl[0-2] 10.0.0.0/16 |
| 04:49.38 | daemon | I should have NAT disabled |
| 04:49.42 | daemon | NAT 'never' |
| 04:49.48 | daemon | as it should not need it? |
| 04:58.45 | daemon | ah well seems to have worked |
| 04:59.05 | daemon | just a side note, I have been asking in here for the last two days if anyone knew about any weird problems where randomly first calls of the day |
| 04:59.09 | daemon | the line would just drop |
| 04:59.18 | daemon | it seems forcing nat to 'never' mode fixed it |
| 05:08.26 | *** join/#asterisk Jomu (~Jomu@188.124.200.2) |
| 05:18.50 | *** join/#asterisk Jumpie (~khja@108.sub-75-243-24.myvzw.com) |
| 05:24.27 | *** join/#asterisk Jomu (~Jomu@188.124.200.2) |
| 05:52.28 | *** join/#asterisk pepselap (~pepse@ip68-109-163-65.ph.ph.cox.net) |
| 05:53.21 | pepselap | Anyone in Europe by chance using pbxes.org? |
| 05:53.40 | pepselap | Or can anyone suggest some free SIP providers in Europe? |
| 05:58.26 | *** join/#asterisk mrbnet (~ryanbantz@c-75-73-142-28.hsd1.mn.comcast.net) |
| 06:00.11 | mrbnet | I have 5 Polycom 331 which are downloading configs from the server and registering to asterisk. There is no ring tine when someone calls it. If I go into the ring settings it freezes and reboots. What am I missing? |
| 06:00.38 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 06:01.08 | mattwj2002 | boy sip over 3G is nice |
| 06:01.09 | mattwj2002 | :) |
| 06:03.01 | fenrus | mrbnet, is there some kind of tune available ? |
| 06:03.14 | fenrus | sounds like its some faulty tunes/path to tunes |
| 06:03.37 | fenrus | i have no clue on how theese phones work, it's just a guess |
| 06:10.09 | mrbnet | fenrus: I am looking for the same info, not sure. I think there should be some default rings in the phone |
| 06:10.40 | x-demon | does asterisk supports sip via tcp? |
| 06:10.54 | *** join/#asterisk raj-darkmystery (~suraj@122.169.5.85) |
| 06:12.58 | raj-darkmystery | hi friends need a li'l help with asterisk |
| 06:13.01 | fenrus | mrbnet, is it possible to give the phone a syslogserver to syslog to, and see if it gives you any hints there? |
| 06:13.10 | raj-darkmystery | rejected because extension not found. |
| 06:13.20 | pepselap | mattwj2002: when i register directly to my * from any mobile device, I miss a lot of incoming calls |
| 06:13.29 | pepselap | mattwj2002: For some reason, pbxes.org works every time |
| 06:13.50 | mattwj2002 | weird |
| 06:14.01 | fenrus | x-demon, dont think that asterisk support sip/tcp or sip/tcp-ssl yet |
| 06:14.10 | ChannelZ | raj-darkmystery: I think its pretty much telling you what the problem is... |
| 06:14.12 | mattwj2002 | what happens when you register direct with your sip provider? |
| 06:14.25 | mattwj2002 | *directly |
| 06:15.06 | ChannelZ | pepselap: is the peer set to qualify in asterisk? |
| 06:15.09 | raj-darkmystery | ChannelZ, yes but i am pretty new to asterisk and the thing i am trying to do is 'm trying to use my office asterisk server to call from home |
| 06:15.37 | pepselap | ChannelZ: qualify=yes |
| 06:15.48 | raj-darkmystery | ChannelZ, hope you are getting what 'm trying to say :( |
| 06:15.59 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 06:16.00 | pepselap | mattwj2002: i dunno, actually. i don't do that often :) |
| 06:16.10 | ChannelZ | raj-darkmystery: well we're not going to be able to tell you any more than Asterisk is. Whatever extension you're dialing doesn't exist, possibly because of being in the wrong context. |
| 06:16.21 | pepselap | mattwj2002: I will try doing that some time and testing a bunch of calls to it |
| 06:16.29 | ChannelZ | needs food |
| 06:16.57 | Kyosh | f00d |
| 06:16.58 | pepselap | but, just fyi, if you ever have that problem, use pbxes.org as a go-between |
| 06:17.03 | raj-darkmystery | ohhk ChannelZ ... i'll try to solve it by myself then.. thanks for your help |
| 06:17.43 | pepselap | noone can recommend a free European SIP service? |
| 06:18.33 | x-demon | pepselap, voip.ms. using them around ~2 mnth |
| 06:18.49 | x-demon | pepselap, pbxes sux :( |
| 06:19.04 | pepselap | x-demon: Where are you at? |
| 06:19.28 | x-demon | pepselap, russia, moscow, using their london server |
| 06:19.50 | x-demon | btw, i also used them in UAE |
| 06:19.54 | pepselap | Thanks. I was wondering how pbxes was from Europe area :) |
| 06:20.01 | pepselap | I will try voip.ms |
| 06:20.21 | pepselap | x-demon: It gives you a SIP URI, yes? |
| 06:20.51 | mattwj2002 | callwithus |
| 06:20.56 | mattwj2002 | works well with 3G |
| 06:20.57 | mattwj2002 | :) |
| 06:21.10 | x-demon | mattwj2002, yes, callwithus is also ok - using them as fallback |
| 06:21.13 | mattwj2002 | actually I think it works better over 3G than over comcast |
| 06:21.51 | pepselap | My parents are going on some kind of tour through a bunch of European countries.. |
| 06:22.00 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 06:22.22 | x-demon | fenrus, i saw in changelog "* Added TCP and TLS support for SIP." |
| 06:22.23 | pepselap | I'm going to try to set up some voip on my mom's iPhone |
| 06:22.27 | x-demon | for 1.6 |
| 06:22.30 | mattwj2002 | I can't get over the quality of sip on my phone |
| 06:22.41 | pepselap | probably will only be able to use WiFi.. can't find a decent international data deal |
| 06:22.43 | mattwj2002 | I have a Google Nexus one |
| 06:22.56 | x-demon | mattwj2002, which app do you use for sip? |
| 06:23.02 | pepselap | mattwj2002: Which app are you using? I'm using SipDroid |
| 06:23.03 | mattwj2002 | sipdroyd |
| 06:23.17 | mattwj2002 | *Sipdroid I guess |
| 06:23.19 | x-demon | sipdroid supports only inband dtmf :( |
| 06:23.45 | pepselap | yeah non working dtmf sux |
| 06:23.52 | mattwj2002 | indeed |
| 06:24.00 | x-demon | i love csipsimple |
| 06:24.02 | mattwj2002 | I can't dial! |
| 06:24.07 | mattwj2002 | :-s |
| 06:24.10 | x-demon | too bad it FC on my moto droid |
| 06:24.13 | mattwj2002 | csipsimple? |
| 06:24.14 | verywiseman | how can i make trunk btw asterisk and call manager express 7.1? |
| 06:24.17 | fenrus | x-demon, interesting |
| 06:24.27 | x-demon | mattwj2002, yes, very new client, in alpha stage |
| 06:25.15 | mattwj2002 | nice |
| 06:25.34 | pepselap | searching for it now |
| 06:25.55 | mattwj2002 | it is on the market |
| 06:26.09 | x-demon | yeah |
| 06:26.19 | x-demon | but please note that this is alpha |
| 06:26.31 | x-demon | i reported few bugs to developer, very fast response btw |
| 06:26.57 | *** join/#asterisk Junior (~Juni@unaffiliated/junior) |
| 06:27.00 | Junior | yello :) |
| 06:27.24 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 06:27.34 | x-demon | looks like it have problems with dns, so i can't use it, because my pbx is on another server :( |
| 06:32.17 | x-demon | Maximum retries exceeded on transmission 221999062580@192.168.1.12 for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt. |
| 06:32.21 | x-demon | arrrgh |
| 06:32.26 | x-demon | how i can fix that? |
| 06:35.50 | pepselap | hm, looks like voip.ms charges to have a SIP URI. Does that sound right? |
| 06:35.58 | pepselap | are you using voip.ms for outgoing calls? |
| 06:39.56 | FutureWeb | People, My ISP has VoIP but when it arrvies at my modem, the modem changes that service into an analogue output, does anyone have an idea how I can tap in and get that direct VoIP service to goto my asterisk machine please ? |
| 06:40.43 | *** part/#asterisk jerome` (~jerome@atom.dedwen.info) |
| 06:42.51 | mattwj2002 | icsiphone isn't working correctly |
| 06:42.55 | mattwj2002 | oops |
| 06:43.03 | mattwj2002 | csiphone isn't working correctly |
| 06:44.15 | mattwj2002 | I am not getting any audio |
| 06:44.16 | mattwj2002 | :( |
| 06:46.20 | FutureWeb | People, My ISP has VoIP but when it arrvies at my modem, the modem changes that service into an analogue output, does anyone have an idea how I can tap in and get that direct VoIP service to goto my asterisk machine please ? << anyone got an idea ? |
| 06:51.38 | fenrus | ask the isp for the sip account settings |
| 06:51.52 | fenrus | probably they wont give it to you, but it might be worth a try |
| 06:54.45 | p3nguin | From my understanding, most cable ISPs which provide phone service don't use VoIP at all. |
| 06:54.50 | x-demon | pepselap, only for ougoing |
| 06:54.54 | x-demon | *outgoing |
| 06:55.14 | pepselap | damn. :) i'm looking for a place that will give me a free sip URI that i can register to in Europe :) |
| 06:55.18 | x-demon | for sip url i use personal pbx and i have short me@lex.gs sip uri :) |
| 06:55.49 | x-demon | pepselap, well, voxalot is okay, but they give only numerical ids |
| 06:55.56 | x-demon | something like 253252@voxalot.com |
| 06:55.58 | p3nguin | YOu don't need a "place" to give you a sip uri. Set up a sip phone and attach it to the internet. |
| 06:56.08 | pepselap | doesn't matter what it is, as long as it works well in Europe and is free |
| 06:56.28 | pepselap | p3nguin: this is for a mobile client that is going to be traveling through a bunch of countries |
| 06:56.35 | x-demon | pepselap, why not just buy a VDS in europe and configure asterisk? |
| 06:56.40 | p3nguin | I guess ekiga might be useful. |
| 06:56.58 | FutureWeb | well I did ask my ISP for the SIP information, and they had no idea what it means, so let alone give it to me, the reps are idiots :/ |
| 06:57.05 | pepselap | x-demon: that involves spending money :) .. buuut, a friend of mine does have a shell in UK |
| 06:57.35 | x-demon | i think asterisk should be started from root |
| 06:57.38 | FutureWeb | if you want a shell I can give you one in Europe aswell lol |
| 06:57.52 | x-demon | but since it doesnt use any privileged ports/functions... |
| 06:58.58 | pepselap | FutureWeb: please :) |
| 06:58.59 | x-demon | oh, btw. Does asterisk support multi-domain hosting? |
| 06:59.23 | pepselap | i wonder what i have to do with asterisk to make all incoming calls come through |
| 07:00.02 | mattwj2002 | linphone is available for android! |
| 07:00.03 | mattwj2002 | :) |
| 07:00.16 | x-demon | mattwj2002, does not work for me |
| 07:00.29 | x-demon | registered, but i can't call |
| 07:00.29 | mattwj2002 | I'll try it |
| 07:00.33 | pepselap | csipsimple just decided to FC every time now |
| 07:00.34 | mattwj2002 | hmm |
| 07:00.40 | mattwj2002 | fc? |
| 07:00.44 | pepselap | force close |
| 07:00.44 | x-demon | force close |
| 07:00.49 | x-demon | pepselap, which phone? |
| 07:00.52 | pepselap | g1 |
| 07:00.54 | mattwj2002 | yeah I uninstalled it |
| 07:01.25 | x-demon | i'll uninstalled it, but waiting for update |
| 07:01.48 | x-demon | promising... sipdroid becoming more and more "pbxes"-dependent |
| 07:02.18 | x-demon | and that dtmf problem makes sipdroid almost useless |
| 07:03.03 | pepselap | indeed. which one you using? |
| 07:03.31 | pepselap | there was another one i saw, but was a little funky. had a picture of a dude with sunglasses for the answer slider |
| 07:03.45 | FutureWeb | lol |
| 07:05.09 | ChannelZ | Website fail: "... To make changes to these services, please call null." |
| 07:06.15 | FutureWeb | anyone knows any site that offers cheap/free Maltese DID numbers ? |
| 07:15.50 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 07:33.11 | FutureWeb | I have a Thomson(ST546 v6) Modem/router, for DSL in it has an analogue phone line and outputs via Ethernet plugs, any ideas if I can use this to convert an analogue phoneline into VoIP (Meaning I input the analogue line trough the DSL plug |
| 07:33.14 | FutureWeb | ? |
| 07:33.27 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 07:44.23 | Jumpie | you don't "convert an analog phone line into voip" |
| 07:44.29 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 07:45.01 | Jumpie | i assume you are talking about some kind of FXO setup? |
| 07:46.21 | pepselap | FutureWeb: No. |
| 07:46.28 | *** join/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
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| 08:47.58 | *** join/#asterisk guven (~guven@217.131.136.178) |
| 08:48.31 | guven | Hey everyone |
| 08:54.33 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 08:56.21 | guven | does asterisk support referring to outbound ? I mean referring through a proxy,is it possible or can i just refer to someone without going to proxy server |
| 08:56.27 | guven | any idea ? |
| 09:00.16 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 09:06.48 | guven | does anyone even talk about anything here,or is everyone sleeping? It's not about my question's not getting answered.I'm just curious.. |
| 09:07.28 | FutureWeb | Jumpie: I need it to so I can connect a regular phoneline to my asterisk machine |
| 09:22.00 | mattwj2002 | hi guys |
| 09:23.19 | guven | hey |
| 09:28.05 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
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| 09:47.36 | pepselap | app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 09:47.39 | pepselap | what the poo? |
| 09:48.00 | ChannelZ | you probably have the wrong device name or something |
| 09:48.29 | *** join/#asterisk BANSAL (~bansal@117.199.124.89) |
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| 10:26.31 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.250.210.adsl.dynamic.totbb.net) |
| 10:27.04 | BugKhaM | do we need to have SELINUX=disabled for running asterisk? |
| 10:34.24 | tzafrir_laptop | BugKhaM, for Asterisk? I don't think so. For FreePBX: I suppose so |
| 10:36.42 | *** join/#asterisk mloe (~casper@217.79.80.213) |
| 10:36.43 | mloe | How to become a VoIP provider who provides only outbound calls? What are the system requirements? What network provider I need to contact? |
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| 10:52.33 | guven | does asterisk support referring to outbound ? I mean referring through a proxy,is it possible or can i just refer to someone without going to proxy server |
| 10:58.28 | tzafrir_laptop | guven, what do you mean by "referring"? |
| 10:58.42 | guven | let's say |
| 10:59.02 | guven | user A and B are talking to each other and they are in the same network |
| 10:59.38 | guven | B decides to refer A to user C, so that phone of C will ring and A-C will start talking |
| 10:59.57 | guven | but user C is in another VPN not in the same network as A and B |
| 11:00.45 | guven | did i able to explain it ? o_o |
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| 11:10.59 | guven | Didn't it make any sense tzafrir ? |
| 11:11.32 | sudhir492 | After a few minutes, Polycom 650 stops responding |
| 11:11.50 | sudhir492 | Does anyone here have a suggestion? |
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| 12:58.55 | ivanvujisic | I did setup ISDN PRI (OpenVox D115E) with DAHDI. Inbound/outbound calls work. How can I set MSN? |
| 13:01.38 | ivanvujisic | anybody there? |
| 13:06.29 | tzafrir_laptop | ivanvujisic, an MSN (DID) is simply an extension number |
| 13:06.46 | tzafrir_laptop | exten => 1234567,1,DoWhatever() |
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| 13:11.22 | Faithful | anyone who can help me with connecting an SPA3000 |
| 13:11.35 | Faithful | I think it is an SPA3000 problem though... when you try to connect to the VoIP to PSTN interface it always says it is busy. |
| 13:12.37 | fenrus | what are you connecting |
| 13:12.38 | fenrus | and how |
| 13:14.10 | fenrus | FXS or FXO ? |
| 13:14.53 | ivanvujisic | tzafrir_laptop, I want all 31 lines (E1) to ring when calling same phone number |
| 13:15.14 | tzafrir_laptop | ivanvujisic, for starters, you only have 30 lines |
| 13:16.06 | tzafrir_laptop | anyway, just send them all to the context that handles those extension numbers (the MSNs) |
| 13:16.31 | ivanvujisic | ok, 30 lines, just give me idea where to search, chan_dahdi or somewhere else? |
| 13:16.33 | Faithful | fenrus, FX0 interface |
| 13:18.38 | ivanvujisic | I want all 30 lines to simultanously ring when calling same phone number |
| 13:19.23 | Faithful | ivanvujisic, just create a ring group |
| 13:19.43 | fenrus | Faithful, you connected the "pstn-cable" to line1 ? |
| 13:19.53 | Faithful | No |
| 13:20.01 | Faithful | to the PSTN line |
| 13:20.09 | Faithful | line1 is FXS |
| 13:20.22 | fenrus | hm, guess i was looking at some other version |
| 13:21.03 | fenrus | but your're sure that its the fxo port |
| 13:21.03 | Faithful | 100% |
| 13:21.03 | Faithful | because I have a phone connected to the fxs port. |
| 13:21.13 | Faithful | it is actually the SIP interface |
| 13:21.19 | fenrus | Faithful, check http://www.tux89.com/wp-content/uploads/2009/10/SipuraSPAAdminGuidev2.0.10.1.pdf 4.12 and 4.13 |
| 13:21.50 | fenrus | hm, and you're sure that the pinout of the connection is correct ? |
| 13:23.23 | ivanvujisic | ring group is for 30 phones to ring simultanously, but how can I arrange all 30 lines to ring if calling same phone number? |
| 13:24.05 | Faithful | ivanvujisic, so your incoming number is the same? |
| 13:24.32 | ivanvujisic | yes, that's what I'm trying to set |
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| 13:53.49 | jrhicks | I'm looking for the ability to centralize calls, assess the call, log the call in an issue tracker, transfer the call to a qualified agent, and have the agent follow-up to the issue tracker .... I'm looking to support about 1000 potential callers, a dozen agents, and a few operators ... should I be looking at switchvox or something like adhearsion |
| 13:53.51 | tzafrir_laptop | ivanvujisic, you have E1 PRI connected to your telephony provider? |
| 13:54.02 | tzafrir_laptop | Or to some channel bank with phones? |
| 13:54.40 | tzafrir_laptop | jrhicks, Asterisk's CDR logging can go to some external DB server. IIRC also to Radius |
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| 13:55.44 | jrhicks | tzafrir_laptop, implimentation speed is the most critical component, I'm trying to set something up for managing an evacuation of the gulf for some of the oil spill responders in case of a hurricane |
| 13:57.09 | tzafrir_laptop | sorry, I misinterepreted "log calls" |
| 13:57.50 | tzafrir_laptop | search keyword: you're looking for a call center |
| 13:58.31 | jrhicks | tzafrir_laptop, thanks - any additional advise, or referral to call center expert would be appreciated |
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| 14:00.27 | ivanvujisic | tzafrir_laptop, E1 PRI connected to my telephony provider, but I don't know if the Telco have to set MSNs or me? |
| 14:01.17 | jrhicks | tzafrir_laptop, based on your advise I searched Google for "Hosted Call Center", I will be looking into "contactual" |
| 14:01.22 | WIMPy | ivanvujisic: PRIs have DDI, no MSNs. |
| 14:01.22 | tzafrir_laptop | which lines do you want to ring when a call comes in? |
| 14:02.08 | ivanvujisic | let's say one phone number for all lines |
| 14:04.23 | ivanvujisic | when I do multiple calls to same phone number it should ring on two extensions |
| 14:08.28 | ivanvujisic | I know how to put it ringing on two extensions, but I'm in doubt how to push PRI to assign all 30 lines to one phone number |
| 14:08.44 | WIMPy | ivanvujisic: Could it be, you're confusing incomming calls and what YOU do with them? |
| 14:09.33 | WIMPy | ivanvujisic: The PRI has phone numbers, not the channels. |
| 14:09.50 | WIMPy | Channels are used dynamically. |
| 14:10.07 | ivanvujisic | ok, I want to assign all 30 lines to one phone number |
| 14:10.35 | WIMPy | Not 30 lines. One line, 30 channels. And channels don't have numbers. |
| 14:11.03 | ivanvujisic | if I call same phone number multiple times simultanously it must open 30 channels on asterisk, right? |
| 14:11.31 | WIMPy | No. It will open one channel per call. No matter what number. |
| 14:12.03 | WIMPy | There is no relationship between channels and numbers, what so ever. |
| 14:12.13 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:12.44 | ivanvujisic | ok, but how can I make all 30 channels ring on same phone number? |
| 14:13.18 | WIMPy | Read again. |
| 14:13.45 | dohd | your pri channels 'enter' in a certain context, in that context you have a dialplan that says what to do |
| 14:13.55 | WIMPy | Yu just configure the one phone number you want to do what you want. You don't care about channels. |
| 14:14.12 | dohd | so that's where you route them to other channels like your internal phones |
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| 14:15.46 | ivanvujisic | if I take 5 phones and dial the very same phone number I want to see 5 calls in asterisk console, then I can route it with ease |
| 14:16.21 | ivanvujisic | one word - MSN |
| 14:18.12 | WIMPy | Lets try this: A call is a call, no mater what number was dialled (as lon as it belongs to that line). The dialled number is just an attribute to that call that is interpreted by your dialplan. |
| 14:18.31 | WIMPy | And I very much doubt, you've got MSNs on a PRI. |
| 14:18.54 | ivanvujisic | I'm in doubt if the Telco must set MSNs? |
| 14:19.11 | ivanvujisic | that's what I'm asking |
| 14:22.39 | WIMPy | Your Telco surely gave you a set of numbers. Now it's up to you, what you whant to do with them. |
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| 14:24.14 | ivanvujisic | ok, just confirm me that I can not set MSN is asterisk config files, but the Telco must set MSNs on his side? |
| 14:24.28 | WIMPy | Both |
| 14:25.21 | WIMPy | When you ordered your PRI you must have got a set of numbers. That's the set you can work with in your dialplan. |
| 14:25.42 | ivanvujisic | but Telco have to set MSNs first? |
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| 14:26.25 | WIMPy | They do, when they give you a line, but MSNs are for ptmp BRIs only. |
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| 14:27.00 | WIMPy | On a PRI you get a DDI block, like 123456[000-599]. |
| 14:27.56 | ivanvujisic | ok, then how can I set PRI to accept 30 calls to the same phone number simultaneously? |
| 14:28.00 | WIMPy | Check your paperwork, if in doubt. |
| 14:28.35 | WIMPy | You still don't it will happen all by itself. |
| 14:28.50 | dohd | it kinda depends on what you call 'accept 30 calls' |
| 14:28.56 | dohd | they will arrive at your asterisk box |
| 14:29.14 | dohd | the 30 channels will all be active and banging your asterisk to be dealth with |
| 14:29.29 | dohd | depending on what you do with it, phones might start ringing |
| 14:29.51 | dohd | if you route it to 1 phone with only 1 line, probably the rest will get busy signals |
| 14:30.12 | ivanvujisic | now I have 30 different phone numbers, but I want only one phone number which will accecpt all calls |
| 14:30.18 | dohd | if you route it to a group number, agent queue whatever, each incoming call will be dealt with accordingly |
| 14:30.28 | dohd | you can't |
| 14:30.31 | dohd | they are 30 phonenumbers |
| 14:30.46 | dohd | I think you are mixing up all kinds of things with approximately the same name |
| 14:30.55 | ivanvujisic | and it will accept 30 calls to the very same phone number? |
| 14:31.11 | dohd | what do you call "30 different phone numbers" ? numbers one can dial on the public net to reach you? 30 numbers on your internal phone system? |
| 14:31.26 | WIMPy | >>There is no relationship between channels and numbers, what so ever. |
| 14:31.27 | dohd | and what do you call 'one phone number'? an external public number? or your internal lines? |
| 14:31.31 | *** join/#asterisk devdvd (~myemail@173-31-149-231.client.mchsi.com) |
| 14:31.48 | ivanvujisic | yes , external public number |
| 14:31.52 | dohd | when you start using clear descriptions, you might start understanding the answers :-) |
| 14:31.59 | WIMPy | You can have up to 30 active calls. No matter which of your numbers have been called. |
| 14:32.01 | dohd | what external? the 30? or the 1? |
| 14:32.18 | ivanvujisic | 1 external number |
| 14:32.45 | ivanvujisic | 30 simultanous calls |
| 14:32.52 | ivanvujisic | to one external public number |
| 14:32.53 | dohd | so 30 people calling that 1 number will all reach your asterisk box |
| 14:32.57 | dohd | ok. great. |
| 14:33.01 | ivanvujisic | yes |
| 14:33.03 | dohd | now what's your problem? |
| 14:33.30 | x-demon | hey guys, can someone help me test my sip calling? |
| 14:33.36 | x-demon | just drop call to sip url |
| 14:33.41 | ivanvujisic | I know how to route it in asterisk, but how to join all 30 external public number to one |
| 14:34.01 | dohd | what do you mean "join all 30 external public number to one" |
| 14:34.07 | dohd | you just said you had only 1 external number |
| 14:34.22 | WIMPy | ivanvujisic: For the last time: You don't. It happens automatically. |
| 14:34.38 | ivanvujisic | I said I want to join all 30 external public number to one |
| 14:34.45 | dohd | yeah, I can read |
| 14:34.49 | dohd | but what does it mean |
| 14:34.53 | dohd | 16:32 < ivanvujisic> 1 external number |
| 14:34.58 | dohd | you said 1 external number |
| 14:35.03 | dohd | and 30 calls |
| 14:35.09 | WIMPy | No, you don't, you kust use one and don't care about the rest. |
| 14:35.13 | dohd | but then you started about 'joining public numbers' |
| 14:35.31 | dohd | as far as asterisk is concerned, you will get an incoming call on any of the pri channels |
| 14:35.31 | ivanvujisic | I want it to be 1 external number, but now it 30 |
| 14:35.45 | dohd | so you have 30 public numbers, not 1? |
| 14:35.51 | ivanvujisic | yes |
| 14:35.53 | dohd | sure you are not confused with '30 channels'? |
| 14:36.00 | ivanvujisic | no |
| 14:36.08 | dohd | and what should happen if someone dials number 3 instead of number 1? |
| 14:36.10 | dohd | drop? |
| 14:36.12 | dohd | or accept? |
| 14:36.20 | WIMPy | ivanvujisic: Please scroll up, read again and think about it. Repeating it all over and over again doesn't seem to make much sense. |
| 14:36.40 | dohd | what you do in your dialplan is your business, you said you could do the routing in asteirsk |
| 14:36.53 | dohd | you just seem to fail to see that for asterisk it's just an incoming call |
| 14:37.02 | dohd | there is no "incoming call on number X", just "incoming call" |
| 14:37.17 | dohd | and in your dialplan you can use the attributes for various routing decisions |
| 14:37.29 | dohd | attributes like "the number that is dialled, the person calling you", etc |
| 14:37.40 | dohd | but still: it's only "an incoming call on your PRI channels" |
| 14:37.42 | ivanvujisic | moment |
| 14:38.22 | dohd | <- goes on with other work :-) |
| 14:39.15 | devdvd | Hi All, I just wanted to give everyone a heads up. I was running 1.6.2.9 on my dev box which is debian 5 fully patched and other than the standard OS stuff, asterisk is the only thing running (no apache, no php, etc). Not sure how yet but someone managed to gain shell access and was running a program called scan-ssh (not sure if the program they were running is part of the compromise or just something arbritary they installed). I am st |
| 14:39.22 | devdvd | I will update you once I find out more |
| 14:40.43 | x-demon | devdvd, i encountered the same problem |
| 14:40.54 | x-demon | debian 5.0... but. No asterisk on that server |
| 14:41.01 | devdvd | hmmm |
| 14:41.02 | x-demon | and yes, ssh-scan. |
| 14:41.16 | devdvd | ah thats not gonna be pretty |
| 14:41.32 | fenrus | reinstall the machine |
| 14:41.33 | x-demon | maybe debian problem? |
| 14:41.40 | fenrus | set secure passwords |
| 14:42.00 | x-demon | fenrus, i used pubkeys-only auth with disabled root login |
| 14:42.09 | fenrus | x-demon, all the way from install ? |
| 14:42.13 | fenrus | scanssh is a software that's available in the repo |
| 14:42.24 | x-demon | fenrus, yes |
| 14:42.28 | fenrus | have there been any other logins to the machine ? |
| 14:42.43 | x-demon | if they was, logs is clean. |
| 14:42.49 | x-demon | so i can't say. |
| 14:42.52 | fenrus | okay |
| 14:43.07 | fenrus | i never encounterd scanssh on any of my debian machines |
| 14:43.15 | fenrus | have been running since potatoe |
| 14:44.08 | devdvd | this is the first one ive seen and i work for a company that has over 1000 of them...so to think theres a flaw in debian that could allow this is concerning to say the least |
| 14:44.25 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 14:44.47 | fenrus | could you post your dpkg -l somewhere ? |
| 14:44.57 | fenrus | and a ps -ef |
| 14:45.27 | x-demon | fenrus, that server has been blocked after ssh-scan and i decided to switch to another hoster |
| 14:45.29 | x-demon | so - no |
| 14:45.54 | fenrus | it's a shame |
| 14:47.11 | fenrus | was zabbix installed? |
| 14:47.32 | devdvd | no |
| 14:47.56 | devdvd | the box is off right now |
| 14:48.06 | devdvd | i shut it down till i can get aroudn to looking at it today |
| 14:48.22 | devdvd | fenrus: do you know of a vulnerability like this in zabbix? |
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| 14:48.59 | fenrus | devdvd, no - i've read about users using some kind of guide setting some useless password to the zabbix user |
| 14:49.13 | fenrus | and idiots reading the guide doing the same thing |
| 14:49.38 | devdvd | ah |
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| 15:05.57 | tzafrir_laptop | devdvd, what services do you have listening, besides Asterisk? |
| 15:06.06 | tzafrir_laptop | Only ssh? |
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| 15:49.12 | x-demon | can anyone drop me sip call? |
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| 16:22.09 | p3nguin | x-demon: I would have, but I never saw the URI. |
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| 16:46.24 | x-demon | p3nguin, me@lex.gs |
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| 17:04.34 | x-demon | p3nguin, saw your call in logs, it says registration failed |
| 17:04.57 | x-demon | what i should fix? :) |
| 17:05.04 | x-demon | i have allowguests=yes |
| 17:05.20 | x-demon | *allowguest |
| 17:07.26 | [TK]D-Fender | x-demon: BAD |
| 17:07.40 | [TK]D-Fender | x-demon: And that has nothing to do with a "guest" REGISTERING |
| 17:08.05 | x-demon | [TK]D-Fender, well i'm still newbie... |
| 17:08.18 | x-demon | so, how i can fix that? |
| 17:09.24 | x-demon | oh yes, now i see that this is for _placing calls_ |
| 17:09.37 | x-demon | dunno how someone can plase a call without registering, btw |
| 17:10.55 | [TK]D-Fender | x-demon: allowguest=yes and point them to a context that lets you do what you'd trust any complete stranger to dial |
| 17:11.41 | x-demon | [TK]D-Fender, [default] context does not contain any dialing rules |
| 17:12.26 | x-demon | okay, now i know some more, but my question is still open |
| 17:12.36 | x-demon | i can't find any information related sip2sip :( |
| 17:14.46 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:15.17 | [TK]D-Fender | x-demon: There is no more information. That's all you need. Now open your eyes and LOOK AT THE CALL. |
| 17:15.40 | x-demon | i see that peer can't authenticate on my server. |
| 17:16.09 | [TK]D-Fender | x-demon: Know what I see? NOTHING |
| 17:16.13 | [TK]D-Fender | PASTEBIN |
| 17:16.15 | [TK]D-Fender | ~pb |
| 17:16.16 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:17.03 | p3nguin | You saw my call in the logs, huh? It says registration failed, huh? That's pretty fucked up considering I didn't call you and I didn't try to register on your system as a peer. |
| 17:17.19 | x-demon | p3nguin, strange. I paste the log then. |
| 17:17.52 | [TK]D-Fender | x-demon: Screw logs. Pastebin a CALL attempt from * CLI with SIP DEBUG enabled <- |
| 17:17.58 | p3nguin | Even if I had tried to call you, I certainly wouldn't have had any reason to try registering. |
| 17:18.13 | x-demon | so i need some to call my sip url |
| 17:18.35 | [TK]D-Fender | p3nguin: You saying you didn't try to call him? |
| 17:18.38 | p3nguin | I was going to earlier when I had time, but you hadn't provided the uri. |
| 17:18.42 | p3nguin | I didn't. |
| 17:18.51 | x-demon | p3nguin, strange. i saw attempts to call, btw |
| 17:19.24 | p3nguin | You also said you saw attempts to register, which has fuck-all to do with calling you. |
| 17:19.25 | x-demon | wait. that must be e164.org. but that's doesn't change anything |
| 17:20.11 | x-demon | okay, here is short log when i call from another provider |
| 17:20.16 | x-demon | [Jun 27 18:19:42] NOTICE[18214]: chan_sip.c:20073 handle_request_invite: Failed to authenticate device <sip:me@lex.gs;transport=UDP>;tag=71985b60 |
| 17:21.07 | x-demon | and with debug, one second... |
| 17:21.07 | *** join/#asterisk valajbeg (~hamo@b199c55.pptp-gw50.cable-internet.GlobalNET.ba) |
| 17:21.38 | [TK]D-Fender | x-demon: I see no configs, and no SIP DEBUG |
| 17:21.42 | [TK]D-Fender | x-demon: You are wasting time |
| 17:21.56 | [TK]D-Fender | x-demon: Do you have a SIP peer named "me" by any chance? |
| 17:22.02 | x-demon | [TK]D-Fender, one second, i uploading them right now |
| 17:22.20 | x-demon | [TK]D-Fender, yes, and i use that peer for outbound calls |
| 17:23.19 | x-demon | http://pastebin.com/amds29iQ - sip debug |
| 17:23.27 | [TK]D-Fender | x-demon: then you shouldn't be receiving calls claiming to be from "me" that aren't from a device YOU set up |
| 17:23.51 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 17:23.56 | x-demon | about config, which of them? sip.conf? |
| 17:24.52 | [TK]D-Fender | x-demon: Who's Zoiper is that? |
| 17:25.10 | x-demon | [TK]D-Fender, name of a sip softphone |
| 17:25.50 | [TK]D-Fender | x-demon: Who's Zoiper is that? |
| 17:25.54 | p3nguin | [tk]d-fender meant: x-demon: Whose Zoiper is that? |
| 17:26.14 | [TK]D-Fender | p3nguin: Score one to the Grammar Nazi.. I'm barely awake |
| 17:26.20 | x-demon | oh, zoiper connected to a sipnet.ru sip provider |
| 17:26.32 | x-demon | from my notebook |
| 17:26.39 | x-demon | if i correctly understand that question |
| 17:26.47 | [TK]D-Fender | x-demon: Who is that fucking PERSON that is running the God-damn softphone?!?! |
| 17:26.51 | x-demon | me |
| 17:27.02 | [TK]D-Fender | FINALLY! |
| 17:27.05 | p3nguin | yay! |
| 17:27.15 | x-demon | yeah, yeah, sorry. I'm bad at english. |
| 17:27.29 | p3nguin | I peed in your pool! Yay! |
| 17:27.34 | [TK]D-Fender | x-demon: Fine so YOU can't et up your soft-phone right. Now what does this have to do with allowing unauthenticated calls from OTHER people? |
| 17:27.41 | Godfather_ | I'm recording this, its so funny |
| 17:27.43 | x-demon | p3nguin, why the hell? :D |
| 17:28.08 | x-demon | uhh, i'm completely failed |
| 17:28.46 | [TK]D-Fender | x-demon: You are talking about one thing and showing us something completely different |
| 17:29.45 | x-demon | my fault. |
| 17:30.04 | p3nguin | allowguest is for anonymous calls. Anonymous calls are calls from user agents which are not defined as peers/users/friends. |
| 17:31.28 | p3nguin | If I were to place a call to you, and you didn't have a peer definition for me, it would be an anonymous call. |
| 17:31.45 | p3nguin | If you have allowguest=no, my call to you would be rejected. |
| 17:32.17 | p3nguin | If you have allowguest=yes, my call to you would be sent into the context you have set in the general section of sip.conf. |
| 17:33.23 | x-demon | okay, thanks. But allowguest was set to yes, and calls got rejected |
| 17:33.23 | p3nguin | At that point, the call would try to match an extension. If there is a match, the call would do whatever the extension says to do; if there is no match, the call would fail with no valid extension. |
| 17:33.56 | x-demon | but error was not related to dialplans... |
| 17:34.10 | x-demon | can it be softphone fault? |
| 17:34.18 | p3nguin | I haven't seen the debug nor the configuration. |
| 17:34.20 | [TK]D-Fender | [13:33]<x-demon>okay, thanks. But allowguest was set to yes, and calls got rejected <- because the call is ID's as someone who IS a user on your system <- |
| 17:34.31 | p3nguin | I mean I didn't look at it if you've pasted it. |
| 17:34.42 | [TK]D-Fender | x-demon: allowguest allows unauthed calls from people it DOESN'T know about |
| 17:34.51 | [TK]D-Fender | x-demon: x-demon it knows "me". |
| 17:34.55 | [TK]D-Fender | facepalms |
| 17:35.22 | x-demon | so just one question... why the hell it calls as me@, since i call from another provider... |
| 17:35.28 | x-demon | p3nguin, http://pastebin.com/amds29iQ |
| 17:35.34 | [TK]D-Fender | x-demon: [me] is aproper user on your system and can be allowed to do things that OTHER RANDOM PEOPLE cannot. It HAS to be authed |
| 17:35.49 | [TK]D-Fender | x-demon: You put the fucking name in there! |
| 17:35.58 | [TK]D-Fender | x-demon: YOU called it "me". |
| 17:36.01 | x-demon | yes. |
| 17:36.10 | p3nguin | Just because the user has not authed does not make it anonymous. |
| 17:36.28 | p3nguin | The system has the peer name defined, so it is not anonymous. |
| 17:36.34 | x-demon | so i need numerical user id |
| 17:36.37 | p3nguin | no |
| 17:36.42 | p3nguin | You need to configure things better. |
| 17:36.43 | x-demon | argh. |
| 17:36.44 | [TK]D-Fender | x-demon: From: <sip:me@lex.gs;transport=UDP>;tag=56a4aa04 <------- YOU put the name. Therefore this isn't a RANDOM person. It is going to get authed against [me] and it is FAILING |
| 17:37.10 | [TK]D-Fender | x-demon: If you think you are testing an random un-authed call you are WRONG |
| 17:37.13 | x-demon | WTF |
| 17:37.16 | x-demon | From: |
| 17:37.28 | [TK]D-Fender | x-demon: change the fucking name so it doesn't match [me] |
| 17:37.33 | p3nguin | Maybe you can rename your testing phone from 'me' to 'notme' |
| 17:37.45 | [TK]D-Fender | x-demon: How many more times do we have to say it? |
| 17:37.49 | [TK]D-Fender | CHANGE THE FUCKING NAME |
| 17:38.20 | x-demon | second |
| 17:38.31 | Kyosh | how about instead of calling it "me@lex.gs" call it "?totalfukintard@lex.gs" is what TKD is suggesting |
| 17:38.52 | p3nguin | or anything other than "me" would be okay, too. |
| 17:39.06 | Kyosh | i kinda like "totalfukintardo" |
| 17:39.10 | [TK]D-Fender | p3nguin: ANTHING OTHER THAN "ME" THAT IS NOT another USER HE STARTED TO SET UP. |
| 17:39.23 | x-demon | ok ok i understand |
| 17:39.42 | [TK]D-Fender | [Jun 27 18:20:56] Found peer 'me' for 'me' from 79.139.139.29:5060 <---NOT FROM A STRANGER |
| 17:40.20 | p3nguin | I guess another way would be to comment out all the peers you tried to create. |
| 17:40.44 | p3nguin | a.k.a. start over! |
| 17:40.46 | [TK]D-Fender | p3nguin: No.. having a clue about who you are claiming to be when calling is enough |
| 17:40.58 | [TK]D-Fender | p3nguin: 1 field. all it takes |
| 17:41.10 | [TK]D-Fender | p3nguin: Changeable in one of 2 easy places. |
| 17:41.18 | [TK]D-Fender | p3nguin: in fact 1 CHARACTER is all it takes |
| 17:41.50 | x-demon | lol |
| 17:41.52 | x-demon | handle_request_invite: Call from '' to extension 'lex.gs' rejected because extension not found in context 'default'. |
| 17:42.07 | x-demon | at least something changed... |
| 17:42.11 | p3nguin | NOW you're in the default context. |
| 17:43.18 | p3nguin | Why you're dialing "lex.gs" doesn't make sense to me, but at least you've made a change that did something. |
| 17:43.30 | x-demon | i dialing sip:me@lex.gs |
| 17:43.36 | [TK]D-Fender | x-demon: that is not a proper URI |
| 17:43.50 | [TK]D-Fender | x-demon: just put a NUMBER there |
| 17:44.14 | [TK]D-Fender | x-demon: Zoiper was not made for sending un-authed calls normally |
| 17:44.23 | [TK]D-Fender | x-demon: You don't jsut shove a URI there |
| 17:44.38 | [TK]D-Fender | x-demon: it passes through the target server for the account you defiend |
| 17:44.41 | [TK]D-Fender | defined* |
| 17:45.16 | x-demon | uh. outgoing and incoming calls to/from numbers works flawlessly for me. |
| 17:45.42 | x-demon | at least for week |
| 17:46.00 | [TK]D-Fender | x-demon: Now what the hell are you talking about? |
| 17:47.46 | x-demon | [TK]D-Fender, i'm talking about sip2sip calls, made with sip urls like user@domain.tld |
| 17:49.07 | [TK]D-Fender | x-demon: Zoiper was not built for that <- |
| 17:49.07 | x-demon | i can use sip calls from like echo@blyon.com |
| 17:49.30 | x-demon | [TK]D-Fender, can you recommend linux client with that feature? |
| 17:50.07 | [TK]D-Fender | x-demon: Actually, it IS calling like that... however it is passing a URL to *. *'s dialplan does not process URI's normally. You have to do a LOT of processing work. |
| 17:50.25 | [TK]D-Fender | x-demon: Because right now you seem to be trying to use it as a proxy. It is NOT a proxy |
| 17:51.04 | WIMPy | IIRC it works with zoiper when you create an empty account. |
| 17:51.44 | [TK]D-Fender | x-demon: besides, are you trying to call your own * server? |
| 17:52.41 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 17:54.24 | x-demon | yes |
| 17:54.49 | *** join/#asterisk Knightfal (~j@75.142.144.171) |
| 17:59.08 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 17:59.29 | [TK]D-Fender | x-demon: then stop putting it in URI format. You don't feed an entire URI to * |
| 17:59.52 | [TK]D-Fender | x-demon: jsut pass it the EXTENSION to dial |
| 18:00.25 | *** join/#asterisk path (~path@gateway/shell/bshellz.net/x-xzcjzcwbktgktcwj) |
| 18:00.33 | path | hello guys :) |
| 18:08.57 | carrar | HARRO |
| 18:10.09 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 18:10.45 | x-demon | what if i calling from another sip server? i know about extensions for local calls |
| 18:10.45 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 18:15.04 | *** join/#asterisk bio-tty (~c@142.6.34.95.customer.cdi.no) |
| 18:18.35 | [TK]D-Fender | x-demon: the extension to dial is not a COMPLETE URI |
| 18:19.14 | x-demon | yes i know it |
| 18:19.33 | [TK]D-Fender | x-demon: WTF are you trying to actually accomplish? |
| 18:19.54 | x-demon | inbound calls to sip uri me@lex.gs |
| 18:19.59 | x-demon | so i can receive them |
| 18:20.04 | x-demon | for example, for e164 |
| 18:20.18 | x-demon | or for DIDs which provides only sip url forwarding |
| 18:20.40 | p3nguin | That's basic SIP delivery. What's the problem? |
| 18:20.47 | *** join/#asterisk jblack (~jblack@pool-71-173-1-106.sctnpa.east.verizon.net) |
| 18:21.27 | pepselap | i'm having an issue where a call comes in for just a second and then it hangs up :/ |
| 18:21.44 | x-demon | p3nguin, i can't receive calls, at least from e164 |
| 18:21.46 | x-demon | it says 404 |
| 18:21.48 | [TK]D-Fender | x-demon: Well you are doing it wrong with zoiper |
| 18:22.01 | jblack | pepselap: a firewall can cause that sort of problem. |
| 18:22.02 | x-demon | [TK]D-Fender, already understood |
| 18:22.03 | [TK]D-Fender | x-demon: you filled in the server in your account entry. |
| 18:22.12 | [TK]D-Fender | x-demon: So jsut put the NUMEBR TO DIAL in the damn "to dial" box |
| 18:22.32 | x-demon | as i already said, number calling works perfectly for me |
| 18:22.37 | [TK]D-Fender | [14:21]<x-demon>p3nguin, i can't receive calls, at least from e164 <- when the hell are you going to SHOW US THIS? |
| 18:22.47 | pepselap | jblack: the * i'm registering to isn't behind a fw.. altho i do have my incoming RTP ports forwarded to -my- * server, but that shouldn't make a diff |
| 18:22.55 | [TK]D-Fender | x-demon: You keep showing us something COMPLETELY FUCKING DIFFERENT. |
| 18:23.00 | pepselap | i can dial out through the one i'm registering to and back into mine just fine |
| 18:23.38 | x-demon | e164 simply dials sip url for validation, isn't it? |
| 18:26.23 | [TK]D-Fender | x-demon: Again wasting time. SHOW US THE PROBLEM. |
| 18:26.37 | [TK]D-Fender | x-demon: There IS no validation |
| 18:27.10 | x-demon | ok, that's what i reached after reconfiguring asterisk |
| 18:27.18 | x-demon | i almost fixed problem, but |
| 18:27.21 | x-demon | handle_request_invite: Call from '' to extension 'me' rejected because extension not found in context 'default'. |
| 18:27.30 | x-demon | exten => me,1,Macro(doDialExten,me) |
| 18:27.46 | [TK]D-Fender | x-demon: So where the hell do I SEE you having an extension "me" in your dialplan? |
| 18:28.06 | [TK]D-Fender | x-demon: that one line isn't enough here, and I don't see the complete failed attempt and complete configs |
| 18:29.58 | x-demon | http://www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/ |
| 18:30.11 | x-demon | i read that article and tried to configure it |
| 18:31.44 | [TK]D-Fender | x-demon: where is YOUR code and YOUR failed called for us to debug? |
| 18:31.56 | [TK]D-Fender | \x-Stop showing us other worthless bullshit |
| 18:31.56 | x-demon | uploading it right now |
| 18:32.17 | p3nguin | Another diversion? |
| 18:33.07 | p3nguin | You've been going at this for hours already. I'd think by now you'd want to get it solved and move on. |
| 18:34.29 | x-demon | [TK]D-Fender, which of configs? extensions only? |
| 18:36.17 | [TK]D-Fender | x-demon: sip & extensions AND th failed call to look at |
| 18:36.45 | pepselap | when do you want to use qualify=yes and qualify=no? |
| 18:37.34 | pepselap | someone implied that when using a mobile client you would want to use one over the other |
| 18:38.04 | p3nguin | You can use qualify=yes when the phone is behind NAT or when you're trying to speed up the failover to another system. |
| 18:38.45 | p3nguin | I would probably use qualify=yes for a mobile client so the server knows when the client has disappeared. |
| 18:38.51 | x-demon | [TK]D-Fender, http://pastebin.com/CgUBGqED - here is everything that you asked. |
| 18:38.57 | p3nguin | Otherwise it will keep trying to send calls to it when it is gone. |
| 18:39.11 | path | [TK]D-Fender: full logs stores also where the traffic is coming? |
| 18:39.12 | pepselap | hm. still can't figure out why it won't ring my extension |
| 18:39.29 | p3nguin | You're doing it wrong, since extensions don't ring. |
| 18:40.01 | path | I need to know traffic activity coming from a SIP account |
| 18:40.09 | p3nguin | Provide configs and evidence of a failed call. |
| 18:40.32 | pepselap | wow. |
| 18:40.50 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 18:42.11 | p3nguin | There's no extension 'me' in the default context. |
| 18:42.39 | p3nguin | And you've incorrectly placed a 'me' extension under [general]. |
| 18:42.51 | x-demon | well yes, how i can set non-digits extensions? |
| 18:43.00 | x-demon | nor _me nor me doesnt work :( |
| 18:43.00 | [TK]D-Fender | x-demon: you put it in the wrong CONTEXT |
| 18:43.02 | [TK]D-Fender | x-demon: you put it in the wrong CONTEXT |
| 18:43.03 | p3nguin | by typing them, I guess. |
| 18:43.04 | [TK]D-Fender | x-demon: you put it in the wrong CONTEXT |
| 18:43.06 | [TK]D-Fender | x-demon: you put it in the wrong CONTEXT |
| 18:43.08 | [TK]D-Fender | x-demon: you put it in the wrong CONTEXT |
| 18:43.11 | x-demon | ooops |
| 18:43.14 | [TK]D-Fender | [14:42]<p3nguin>And you've incorrectly placed a 'me' extension under [general]. <--------------------------- |
| 18:43.41 | p3nguin | Do you see [default] in extensions.conf? |
| 18:43.47 | x-demon | damn :D |
| 18:44.00 | p3nguin | I don't. |
| 18:44.07 | p3nguin | But you've sent a call to it. |
| 18:45.11 | x-demon | yes. Now i see |
| 18:45.36 | p3nguin | You should have [general] followed by general settings, then [globals] followed by global settings, then [default] where you define extensions which are to be used by phones with context=default. |
| 18:45.55 | pepselap | what context should register lines be in? default? |
| 18:46.00 | p3nguin | nope |
| 18:46.03 | pepselap | err general, not default |
| 18:46.09 | pepselap | general? |
| 18:46.18 | p3nguin | register statements go in sip.conf. |
| 18:46.30 | p3nguin | contexts are dialplan. |
| 18:47.00 | pepselap | what do you call the thing that says [general] in sip.conf? |
| 18:47.02 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-71-250.home.otenet.gr) |
| 18:47.20 | pepselap | is that a context? |
| 18:47.22 | [TK]D-Fender | pepselap: REGISTER statements belong after everything in [general] and BEFORE the first named section |
| 18:47.32 | [TK]D-Fender | pepselap: a "section" |
| 18:47.32 | p3nguin | In sip.conf, register statements go in [general]. |
| 18:47.35 | pepselap | D-Fender: That's what I was looking for, thanks |
| 18:51.42 | pepselap | hmph. i think my sip.conf is fine, it's gotta be my extensions.conf that's causing it to hang up right away. |
| 18:52.17 | *** join/#asterisk n3hxs (~HAMming@63.164.47.229) |
| 18:53.00 | p3nguin | I haven't seen the configs nor the failed call, so I won't even try to guess. |
| 18:54.15 | x-demon | [TK]D-Fender, p3nguin, thanks for listening to my absurd |
| 18:54.34 | x-demon | srsly, if noy you two, i never discovered mistyped context :( |
| 18:54.38 | x-demon | *not |
| 18:54.49 | pepselap | assuming an extension of 101, where does the "exten => 101,1,Dial(SIP/101)" line go in extensions.conf? under [default]? |
| 18:55.00 | p3nguin | I hope you didn't change context=default to context=general. :/ |
| 18:55.16 | pepselap | who? |
| 18:55.24 | p3nguin | x-demon: ^^ |
| 18:55.36 | p3nguin | pepselap: exten 101 goes in whichever context you want to contain 101. |
| 18:55.39 | x-demon | p3nguin, no of course, i renamed context in extensions.conf |
| 18:55.48 | p3nguin | x-demon: That's even worse. |
| 18:55.55 | p3nguin | x-demon: <p3nguin> You should have [general] followed by general settings, then [globals] followed by global settings, then [default] where you define extensions which are to be used by phones with context=default. |
| 18:56.25 | p3nguin | That's a minimum of three contexts. |
| 18:57.43 | p3nguin | If you changed [general] to [default], you just broke your extensions.conf. |
| 18:58.00 | p3nguin | maybe you need to study The Book more. |
| 18:58.03 | p3nguin | ~book |
| 18:58.04 | 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 |
| 18:58.28 | pepselap | p3nguin: is that sip.conf you're talking about that needs general, globals, and default? |
| 18:58.40 | p3nguin | no, extensions.conf |
| 18:58.40 | pepselap | or extensions.conf |
| 18:58.41 | pepselap | ok |
| 18:58.49 | [TK]D-Fender | [14:54]<pepselap>assuming an extension of 101, where does the "exten => 101,1,Dial(SIP/101)" line go in extensions.conf? under [default]? <-- it should be in the place the call is LOOKING for it in |
| 18:59.10 | [TK]D-Fender | You should never, EVER have a context named [default] |
| 18:59.23 | [TK]D-Fender | pick any other name but that |
| 18:59.26 | p3nguin | unless you want to, of course. |
| 18:59.39 | [TK]D-Fender | p3nguin: No, there are failover reasons not to |
| 18:59.45 | pepselap | so if extension 101 in sip.conf says context=default, then it'd look for that "exten => 101" line under [default] |
| 18:59.45 | p3nguin | I have one called default, and that's where anonymous calls are processed. |
| 19:00.17 | [TK]D-Fender | p3nguin: it is a hard-coded one used by call processing where intended targets aren't found. Ugly security risk |
| 19:00.18 | p3nguin | pepselap: That's correct, it would look there. |
| 19:00.53 | [TK]D-Fender | p3nguin: You want misc, then make one called that. but other apps and dialplan failures can't land stuff there unintentionally and fuck shit shit up. |
| 19:01.02 | [TK]D-Fender | p3nguin: NEVER use [default] |
| 19:01.10 | pepselap | what would be the recommended followup to that? "exten => 101,2,HangUp" or something? |
| 19:01.15 | [TK]D-Fender | s/can't/can |
| 19:01.26 | p3nguin | So [default] should remain empty, or remove it completely? |
| 19:01.58 | [TK]D-Fender | p3nguin: Burn it. Burn it with fire |
| 19:05.34 | x-demon | p3nguin, i set [common] context as default in sip.conf |
| 19:05.45 | x-demon | and yes, i slowly reading that book |
| 19:06.51 | x-demon | but i'll fix that asap. |
| 19:10.22 | pepselap | on a separate note, any idea why calls coming into an extension show that extension's number in callerid? |
| 19:10.38 | pepselap | it shows the correct callerid name, but the callerid num is the extension |
| 19:11.04 | pepselap | it's not because i have callerid=blah <number> under the extension, is it? |
| 19:11.29 | p3nguin | So confusing! |
| 19:12.03 | p3nguin | under the extension? No. In the peer definition? Absolutely. |
| 19:12.29 | pepselap | so callerid= isn't for outgoing calls? |
| 19:12.35 | p3nguin | it's for calls from that peer. |
| 19:12.54 | [TK]D-Fender | pepselap: Did you do something ridiculous like set "fromuser=thedevicethatthispeerisfor"? |
| 19:12.59 | [TK]D-Fender | pepselap: that would do it <- |
| 19:13.00 | *** join/#asterisk xuser_ (~xuser@unaffiliated/xuser) |
| 19:13.10 | p3nguin | If SIP/John makes a call to anyone and there is no other caller id information, it would look at [John]'s callerid= line. |
| 19:13.12 | pepselap | d-fender: That may be it |
| 19:13.22 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 19:14.32 | pepselap | hm, commenting out just that made that extension not ring anymore |
| 19:15.14 | pepselap | hm. weird. |
| 19:15.51 | pepselap | yeh, commenting out the callerid= line makes no diff. |
| 19:16.29 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 19:20.01 | *** part/#asterisk iamthelostboy (~nathan@210.48.114.74) |
| 19:20.22 | pepselap | D-fender: at least now i have a better idea of what that fromuser does :D |
| 19:26.13 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 19:33.37 | pepselap | guh. it says "-- SIP/101-0000005e is ringing", but it just hangs up right away :/ |
| 19:36.53 | [TK]D-Fender | yup.. people clearly jsut don't want help... |
| 19:36.57 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 19:37.19 | pepselap | stinkin people. |
| 19:43.30 | pepselap | hm. "Got SIP response 486 "Busy Here"" |
| 19:43.42 | pepselap | maybe it's he client |
| 19:46.06 | *** join/#asterisk cdahmedeh (~cdahmedeh@62.68.65.243) |
| 19:48.19 | *** join/#asterisk cdahmedeh (~cdahmedeh@62.68.65.243) |
| 19:49.07 | *** join/#asterisk pervy_sage (~patrick@unaffiliated/svminvictvs/x-938456) |
| 19:49.09 | pervy_sage | Heya |
| 19:49.24 | pervy_sage | I'm looking for an example of how to forward calls during certain times of the day. |
| 19:50.21 | pervy_sage | I've been poking around but having a hard time finding the righ texample |
| 19:50.27 | pervy_sage | I know it's possible to do somehow. |
| 19:56.44 | [TK]D-Fender | pervy_sage: "core show application GotoIfTime" |
| 19:58.20 | pervy_sage | Ah ha |
| 19:58.30 | pervy_sage | [TK]D-Fender: Thanks. |
| 19:59.34 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 20:13.25 | pepselap | it -was- my damn client. |
| 20:13.29 | ivanvujisic | how can I set outbound callerid for extenstion on asterisk PRI ISDN? |
| 20:14.13 | pabelanger-lap | ivanvujisic: core show function CALLERID |
| 20:14.51 | ChannelZ | Set(CALLERID(num)=1234567890) |
| 20:15.06 | ivanvujisic | I know, but I'm in doubt will that work for outboind call on PRI ISDN E1? |
| 20:15.43 | ChannelZ | Why don't you try it and see what happens? |
| 20:16.14 | ivanvujisic | ok, I will, sorry |
| 20:16.34 | pabelanger-lap | ivanvujisic: Then talk to your telco. |
| 20:25.08 | ChannelZ | or maybe your ISP.. |
| 20:25.53 | [TK]D-Fender | cause yeah.. ISP' shave a lot to do with CID issues on a PRI... |
| 20:26.00 | [TK]D-Fender | </sarcasm> |
| 20:26.07 | ChannelZ | <-- ivanvujisic has quit (Ping timeout: 252 seconds) |
| 20:27.21 | [TK]D-Fender | ChannelZ: its FreeNode. Not him |
| 20:31.28 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 20:43.45 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:43.59 | jblack | There's been serious talk in congress to ratchet down on callerid |
| 20:46.46 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 20:47.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 20:47.31 | WIMPy | jblack: What does that mean? |
| 20:48.48 | *** join/#asterisk Z_God (~julius@wlan225062.mobiel.utwente.nl) |
| 20:49.43 | pepselap | yeh they're trying to make anything that changes CID illegal |
| 20:52.01 | Gugge | so basically they are gonna make it illegal for the telco to change the (fake) CID the customer sets? :P |
| 20:53.58 | WIMPy | Cool. So at the moment it is possible to make anonymous calls in the US? |
| 20:54.53 | WIMPy | Thew country that forces telcos on the rest of the world to even store the coulour of the underwaer, peple were wearing when placing a phone call? |
| 20:56.33 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 21:02.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:11.33 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:23.56 | Godfather_ | i've set up a mobile with my bluetooth, but when i do mobile show devices it says "Connected: no" |
| 21:24.05 | Godfather_ | here is my config, http://pastebin.com/Jn0g7PiA |
| 21:24.41 | Godfather_ | and curiosly, if i do mobile search entering asterisk with -r it returns nothing, i've to open it stopping asterisk and then asterisk -cvvv |
| 21:32.24 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:38.02 | Alton35 | chan_mobile? |
| 21:38.09 | *** join/#asterisk DelphiWorld (~Delphi@41.200.23.11) |
| 21:38.11 | DelphiWorld | hi |
| 21:38.18 | DelphiWorld | right that H.323 don't do rtp? |
| 21:38.20 | Alton35 | I did the bluetooth thing with that before and it worked ok, a little bit of a trick to get connected though. |
| 21:42.15 | *** part/#asterisk DelphiWorld (~Delphi@41.200.23.11) |
| 21:48.14 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 21:50.20 | doolittlework | hi there i having some trouble of getting data from cdr_custom into my management system |
| 21:50.47 | doolittlework | i have it setu p to update the master.csv file to a samba share |
| 21:50.50 | Godfather_ | Alton35, can you help me? |
| 21:51.51 | Alton35 | golly, it's been some time, I do believe that I have samples saved |
| 21:52.00 | Alton35 | can we work on it tonight? in a few hours? |
| 21:52.12 | Godfather_ | Alton35, ok, no prob |
| 21:52.31 | Alton35 | I hope I can help you. We'll see. |
| 21:52.42 | doolittlework | this shared file gets converted to a text .log file and gets accessed by the management system via the network share, the problem is that all the records gets writen to one line so the management system is going o0 i do not comply |
| 21:53.07 | Godfather_ | Alton35, sure. |
| 21:53.55 | doolittlework | please help me is there a way to convert the .csv to a text file so that it won't confuse the management system? |
| 21:54.44 | Gugge | doolittlework: a .csv file _is_ a text file |
| 21:54.46 | [TK]D-Fender | doolittlework: .. it IS text |
| 21:55.14 | [TK]D-Fender | CRAZY PEOPLE |
| 21:55.35 | [TK]D-Fender | suspects a glaringly obvious suspect... |
| 21:55.55 | doolittlework | [TK]D-Fender: thats waht i been reading but why would the management add it all to one line |
| 21:56.45 | Gugge | doolittlework: ask the people who made the braindead management system |
| 21:56.46 | doolittlework | if i create a text file in notepad with the same data and then load it to the management system it does not join the strings |
| 21:57.26 | doolittlework | how can it be the management system for it works if i uload a normal text file |
| 21:58.23 | Gugge | the linebreak in your normal text file is not the same as in the other normal text file (the .csv) |
| 21:58.30 | Gugge | is my guess |
| 21:58.46 | Gugge | ask the management system people |
| 21:58.47 | Gugge | they know |
| 21:58.54 | Gugge | or they should get another job :P |
| 21:59.15 | doolittlework | Gugge: are you looking for a new job? |
| 21:59.27 | doolittlework | i ahve an opening if you help me solve this |
| 21:59.36 | Gugge | nope, i have more than enough |
| 21:59.43 | doolittlework | lol |
| 22:00.03 | Gugge | go read about newline difference in unix and dos |
| 22:00.35 | Gugge | and just so you know, this has nothing to do with asterisk :) |
| 22:01.21 | doolittlework | i think cdr is part of asterisk |
| 22:01.28 | Gugge | yes |
| 22:01.37 | Gugge | but your broken import system is not |
| 22:01.50 | Gugge | the cdr file is a perfectly normal text file with unix linebreak |
| 22:06.23 | *** join/#asterisk Kevin` (~kevin@rrcs-67-52-47-69.west.biz.rr.com) |
| 22:06.46 | [TK]D-Fender | [18:00]<Gugge>go read about newline difference in unix and dos <--- |
| 22:06.53 | [TK]D-Fender | LF not CRLF |
| 22:06.55 | Kevin` | how do you test 911 service |
| 22:07.01 | [TK]D-Fender | Kevin`: Dial it |
| 22:07.11 | Kevin` | and? will they complain? |
| 22:07.21 | doolittlework | lol |
| 22:07.24 | [TK]D-Fender | Kevin`: Odds are if you do it once and right, no. |
| 22:07.33 | doolittlework | no help for the short sighted |
| 22:07.44 | Kevin` | right? |
| 22:08.18 | [TK]D-Fender | Kevin`: Well if you spam them with calls they will get pissed fast |
| 22:08.34 | Kevin` | [TK]D-Fender: have you ever done this? |
| 22:09.01 | [TK]D-Fender | Kevin`: called 911? Yes |
| 22:09.12 | Kevin` | what did you say |
| 22:11.16 | Kevin` | http://www.trixbox.org/forums/trixbox-forums/help/how-test-911-through-pstn-works-dont-want-meet-unhappy-cops-again |
| 22:11.19 | Kevin` | :( |
| 22:12.33 | Chainsaw | Kevin`: Read: talked gibberish at operator. |
| 22:12.58 | Kevin` | anyone who has actually done this please? I don't want cops complaining and fining me |
| 22:13.46 | WIMPy | Kevin`: Call their normal office phone and ask them how you do it. |
| 22:16.23 | [TK]D-Fender | Kevin`: It's been a while. What are you dialing out of? |
| 22:16.40 | Kevin` | voip provider |
| 22:16.47 | [TK]D-Fender | Kevin`: And that isn't testing a "service" that is using a service to call a specific number |
| 22:16.59 | [TK]D-Fender | Kevin`: Can you dial other #'s through them? |
| 22:17.03 | Kevin` | yes |
| 22:17.21 | [TK]D-Fender | Kevin`: What does your provider say about dialing 911 as 911 through them> |
| 22:17.50 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 22:18.04 | Kevin` | i'd have to call them. they recently added some kind of e911 stuff but don't explain if anything special is required for it. their normal routing examples don't cover it |
| 22:19.56 | [TK]D-Fender | Kevin`: Perhaps you should actually talk to your provider |
| 22:20.42 | Kevin` | yeah, I will. I will still need to test my own rules at least once though |
| 22:24.41 | doolittlework | Gugge: thank you for not helping but the remarks and the pointer about linebreaks help me out,, alls goood working so i guess my job is safe thx to u,,,, thank u dude |
| 22:30.27 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:31.19 | *** join/#asterisk nightwalk (~nightwalk@daimon.vixel.org) |
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| 22:48.20 | Godfather_ | Alton35, are you busy now? |
| 22:48.59 | Godfather_ | i feel sleepy, maybe tomorrow we can have a look to that |
| 22:49.43 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:01.11 | *** join/#asterisk Arsenick (~rp@modemcable022.82-21-96.mc.videotron.ca) |
| 23:02.56 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:09.24 | *** join/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 23:13.20 | *** join/#asterisk Da-Geek (~Da-Geek@87-194-2-213.bethere.co.uk) |
| 23:13.40 | mattwj2002 | hi guys |
| 23:14.00 | mattwj2002 | anyone have any luck with sipdroid and pbxes.org? |
| 23:28.59 | jblack | The dorkbot pcb fab is due tomorrow morning, if anyone's making pcbs. |
| 23:30.45 | jblack | sorry. wrong channel |
| 23:31.02 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 23:36.38 | [TK]D-Fender | mattwj2002: both work |
| 23:47.33 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 23:47.33 | mattwj2002 | [TK]D-Fender: can you help me? |
| 23:47.33 | boodu | hello |
| 23:47.33 | mattwj2002 | I am getting a 404 |
| 23:47.50 | mattwj2002 | the username and password should be what I login into pbx.org right? |
| 23:47.59 | mattwj2002 | *pbxes.org right? |
| 23:48.35 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 23:50.32 | *** join/#asterisk misc-- (~misc@202-154-80-42.people.net.au) |
| 23:51.27 | [TK]D-Fender | mattwj2002: 404 can be a response to several actions. |
| 23:51.36 | [TK]D-Fender | mattwj2002: Depends WHICH |
| 23:51.45 | mattwj2002 | what does your username have to be? |
| 23:51.56 | [TK]D-Fender | mattwj2002: either way, both clearly work. Either your routes aren't set up right or your device is authing as the wrong user |
| 23:52.07 | mattwj2002 | okay |
| 23:52.14 | [TK]D-Fender | [19:51]<mattwj2002>what does your username have to be? <-- that SAME on both sides |
| 23:52.49 | mattwj2002 | so if my login is username? |
| 23:52.57 | mattwj2002 | and my extension is 1000 |
| 23:53.19 | mattwj2002 | should my authorization username be username-1000? |
| 23:53.20 | [TK]D-Fender | mattwj2002: Go LOK AT THE CALL |
| 23:53.40 | [TK]D-Fender | mattwj2002: No clue where you get this composite name bit from... |
| 23:54.02 | mattwj2002 | several websites |
| 23:55.47 | [TK]D-Fender | GARBAGE |
| 23:55.54 | [TK]D-Fender | mattwj2002: Go look at the call |
| 23:57.50 | mattwj2002 | status? |
| 23:58.10 | *** join/#asterisk s14ck (~s14ck@190.72.27.63) |
| 23:58.55 | [TK]D-Fender | mattwj2002: Go look at the call |
| 23:59.12 | mattwj2002 | how? |
| 23:59.29 | mattwj2002 | there is nothing under call monitor |
| 23:59.31 | [TK]D-Fender | mattwj2002: If you have to ask then you're in the wrong channel. |
| 23:59.33 | mattwj2002 | if that is what you mean |