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| 00:04.33 | generalhan_ | i was hoping he was ... lol. but i didnt want to ping him, just incase someone else happen to have the same issues i am ! |
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| 00:08.21 | WIMPy | What does "ooh323_onReceiveSetup: Unacceptable ip" mean? The IP listed looks perfectely valid to me. |
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| 00:18.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:24.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.6.0 (2012/07/10), 1.8.14.0 (2012/07/10), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 02:03.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 03:07.18 | dijib | p3nguin: did you recieve that package yet? |
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| 03:13.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 03:50.00 | luckyaba | Curious if anyone has seen issues with the kernel timer causing packet loss |
| 03:50.04 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 04:06.45 | sruffell | luckyaba: talking about res_timining_timerfd? |
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| 05:09.26 | deadcow777 | Hi, does anybody know a good sip trunk provider in India ? |
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| 05:42.16 | din3sh | PRI got event: HDLC Abort (6) on D-channel of span |
| 05:42.35 | tzafrir_laptop | generalhan, still here? |
| 05:44.28 | din3sh | PRI got event: HDLC Abort (6) on D-channel of span:---> what could be the reasons for this |
| 05:47.35 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 05:49.24 | deadcow777 | CRC errors or IRQ interrupts |
| 05:50.40 | deadcow777 | check dahdi status |
| 05:52.22 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:05.51 | tzafrir_laptop | din3sh, bad line? |
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| 06:17.34 | deadcow777 | <PROTECTED> |
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| 06:25.55 | din3sh | tzafrir_laptop: calls are coming in and out correctly |
| 06:25.55 | din3sh | no call drops have been reported yet |
| 06:26.48 | tzafrir_laptop | din3sh, that error means that there were some errors in the D-channel |
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| 06:43.01 | *** join/#asterisk _omer (~omer@116.71.186.211) |
| 06:43.03 | _omer | hello |
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| 07:07.51 | *** join/#asterisk k1ng_ (~k1ng@unaffiliated/k1ng) |
| 07:07.54 | k1ng_ | hi |
| 07:08.15 | k1ng_ | can anyone tell me what should i use for web interface? |
| 07:08.35 | _omer | freepbx |
| 07:09.56 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:10.07 | k1ng_ | _omer, thank you, thats what i thought :) |
| 07:10.20 | _omer | ;) |
| 07:18.01 | k1ng_ | Error: 'deb-src http://packages.asterisk.org/deb oneiric-proposed main' invalid |
| 07:25.21 | k1ng_ | _omer, can you help me install asterisk? |
| 07:26.31 | k1ng_ | sudo: dahdi_genconf: command not found |
| 07:27.22 | _omer | you just need to install asterisk ? |
| 07:27.28 | _omer | which Distro ? |
| 07:27.32 | k1ng_ | ubuntu |
| 07:27.33 | k1ng_ | yes |
| 07:27.40 | k1ng_ | ubuntu 11.4 |
| 07:29.15 | k1ng_ | Digium Asterisk Hardware Device Interface |
| 07:29.19 | k1ng_ | do i need a device for that? |
| 07:29.42 | _omer | do you plan to use a device with asterisk ? |
| 07:33.40 | k1ng_ | its running yes |
| 07:33.42 | k1ng_ | yes |
| 07:34.27 | _omer | Is it a digium device? |
| 07:34.38 | _omer | or sip device ? |
| 07:37.23 | k1ng_ | yes |
| 07:37.26 | k1ng_ | digium |
| 07:37.34 | k1ng_ | _omer, its digium |
| 07:38.05 | k1ng_ | someone said there is a package called dahdi_dummy |
| 07:38.28 | _omer | yes. in this case you need to install some extra packages as well. |
| 07:38.40 | k1ng_ | is there any howto? |
| 07:39.50 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 07:40.10 | slav3_kitten | why the hell am i getting sub 1m/s on speed test at 4am |
| 07:40.17 | ChannelZ | dahdi_dummy is old |
| 07:41.31 | k1ng_ | ChannelZ, what should i use? |
| 07:42.41 | ChannelZ | What version of Asterisk? What is this 'device' you have? |
| 07:43.16 | k1ng_ | 1.8 |
| 07:43.20 | k1ng_ | Digium |
| 07:44.09 | ChannelZ | Digium is a company. Is it a TDM card of some sort? |
| 07:44.47 | k1ng_ | the server does not have any card |
| 07:45.26 | ChannelZ | so what is it you have? Are you talking about one of their phones? |
| 07:45.33 | k1ng_ | yes |
| 07:45.58 | ChannelZ | ok.. well you don't really need DAHDI at all then |
| 07:46.23 | ChannelZ | If you want conferencing, use ConfBridge instead of MeetMe |
| 07:46.54 | k1ng_ | ok :) |
| 07:46.55 | ChannelZ | Or you can just build/install DAHDI, but you don't have to worry about 'dummy' specifically anymore |
| 07:47.07 | k1ng_ | ah ok |
| 07:47.08 | k1ng_ | thank you |
| 07:50.11 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1001) |
| 07:50.42 | k1ng_ | ChannelZ, i am extremely sorry but no hardware based phone, its just softphone :) |
| 07:52.40 | ChannelZ | Yah. DAHDI is primarily hardware drivers for TDM cards, but older versions/components of asterisk require TDM as a timing source. dahdi_dummy was a 'virtual device' to provide the timing on systems with no TDM hardware installed |
| 07:53.29 | ChannelZ | MeetMe for instance depended on DAHDI for timing and mixing audio |
| 07:55.04 | ChannelZ | ConfBridge is the replacement for MeetMe which doesn't require DAHDI to do the work, but the capabilities of ConfBridge in 1.8 I don't think were that developed compared to MeetMe, but they've been greatly expanded in Asterisk 10. |
| 07:56.52 | k1ng_ | i am using Asterisk 1.8 |
| 07:56.53 | k1ng_ | :) |
| 07:56.55 | salz212 | does confBridge sends event when the caller speaks (in case mute is enabled) ? |
| 07:59.13 | ChannelZ | well it can do noise gating though I'm not sure if it sends any sort of AGI event if that's what you're meaning |
| 08:00.25 | ChannelZ | s/AGI/AMI/ |
| 08:00.48 | ChannelZ | actually looking at Asterisk 10 anyway, it appears it can: talk_detection_events=yes |
| 08:06.15 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 08:14.36 | *** join/#asterisk Madkiss (~madkiss@freenode/staff-emeritus/madkiss) |
| 08:14.37 | Madkiss | hello there. |
| 08:14.40 | Madkiss | I get |
| 08:14.45 | Madkiss | was not compiled with the same compile-time options |
| 08:14.47 | Madkiss | when trying to load a module. |
| 08:14.50 | Madkiss | can I force it into asterisk anyway? |
| 08:15.47 | ectospasm | Madkiss: can't you recompile and reinstall Asterisk? That'd probably be a LOT simpler... |
| 08:16.58 | Madkiss | This is some elastix system, actually. I got the SRPM for the asterisk version that is running there and rebuilt it. And then I took the modules from the compiled source. |
| 08:17.10 | fling | hello! |
| 08:17.39 | Madkiss | ectospasm: the only difference is that the latter was built with --with-odbc and the now-installed version was not. |
| 08:18.57 | k1ng_ | what is the website for ConfBridge? |
| 08:19.19 | ectospasm | Madkiss: it doesn't matter what the difference is |
| 08:19.35 | ChannelZ | it's an Asterisk module that comes with it, it's not a separate product or anytthing |
| 08:19.35 | ectospasm | Madkiss: you need to recompile and reinstall Asterisk |
| 08:20.12 | ectospasm | we don't typically support Elastix here |
| 08:20.37 | Madkiss | This is asterisk 1.8.7.0. Nothing elastix-specific there, the packages are from centos 5.something, I think. |
| 08:20.57 | Madkiss | Neither has that anything to do with the system asterisk is running on. There's gotta be a way to force Asterisk into loading modules. |
| 08:21.13 | ectospasm | you run the risk of breaking Asterisk when you do that |
| 08:21.21 | ectospasm | A simple flag could mean an ABI change |
| 08:21.54 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:26.09 | ectospasm | ...so when the module is loaded you get a segfault |
| 08:26.31 | ectospasm | ...or worse, it loads OK but when some function provided by the module gets called, it segfaults |
| 08:26.47 | Madkiss | Isn't segfaulting an asterisk core feature anyway? |
| 08:26.56 | Madkiss | Seriously, this installation has mISDN loaded. |
| 08:26.56 | ectospasm | har har har |
| 08:27.04 | ChannelZ | I don't think mine ever has. |
| 08:27.16 | Madkiss | Chances are that the kernel will crash long before asterisk will have the opportunity to do so. |
| 08:27.35 | ectospasm | it all depends |
| 08:27.57 | Madkiss | In terms of mISDN, it doesn't. |
| 08:28.05 | Madkiss | mISDN is just broken, and that's it |
| 08:29.27 | ectospasm | Digium hasn't supported mISDN for their BRI adapters in quite some time, so... |
| 08:35.14 | k1ng_ | how do i install ConfBridge? |
| 08:37.06 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:40.28 | k1ng_ | someone please help me with ConfBridge |
| 08:42.10 | fling | What does this mean? > == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/101-00000005' status is 'CONGESTION' |
| 08:42.41 | kaldemar | fling: it means a call failed. see the previous lines and sip debug to see what happens and why. |
| 08:43.02 | fling | kaldemar: I can't understand anything it this debug :p |
| 08:43.52 | kaldemar | ~pb |
| 08:43.52 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 08:45.43 | fling | kaldemar: /etc/asterisk/sip.conf > http://bpaste.net/show/34614/ ; /etc/asterisk/extensions.conf > http://bpaste.net/show/34615/ |
| 08:46.52 | kaldemar | fling: and the whole CLI output of a call. |
| 08:47.06 | fling | kaldemar: with debug enabled? |
| 08:47.54 | kaldemar | "sip set debug on", yes |
| 08:49.04 | fling | kaldemar: I will comment register then, because a lot of spam from register |
| 08:50.09 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:51.19 | fling | kaldemar: http://dpaste.com/769744/ |
| 08:52.43 | kaldemar | fling: SIP/2.0 404 Not Found <--- the destination does not know the dialed number |
| 08:53.00 | kaldemar | fling: Reason: Q.850;cause=3;text="No route to destination" |
| 08:53.35 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 08:55.27 | *** join/#asterisk devyll (~devyll@hw.net.hostway.ro) |
| 08:56.34 | *** join/#asterisk PBXman (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 08:56.39 | PBXman | hello asterisk people |
| 08:58.41 | fling | hello, PBXman |
| 08:58.43 | devyll | hello guys. what whould be the recommended method for identifieng the caller id for a give extension. I'm working at a interface, using AMI, and wanted to know your opinion |
| 08:58.55 | devyll | using show channels means a lot of parsing to be done it seems |
| 08:59.02 | devyll | I was wondering of something more to the point |
| 09:05.07 | beaver_rrr | can ChanSpy app affect spied channel in any way? I have a user who tells me that his tries to spy on some channel end with call interruption |
| 09:05.37 | *** join/#asterisk Petchaw (~Pascal@c-174-48-158-108.hsd1.fl.comcast.net) |
| 09:07.16 | fling | kaldemar: what about this? > http://dpaste.com/769749/ < the same? because the number exists |
| 09:08.22 | *** part/#asterisk Petchaw (~Pascal@c-174-48-158-108.hsd1.fl.comcast.net) |
| 09:12.17 | fling | kaldemar: SIP/2.0 404 Not Found Reason: Q.850;cause=3;text="No route to destination" |
| 09:12.25 | fling | kaldemar: what am I doing wrong? |
| 09:15.21 | *** join/#asterisk hehol (~Adium@ip-178-202-243-147.unitymediagroup.de) |
| 09:25.36 | fling | kaldemar: something is wrong with incoming calls too > http://dpaste.com/769756/ |
| 09:26.25 | fling | kaldemar: is it something wrong with my /etc/asterisk/extensions.conf? > http://bpaste.net/show/34626/ |
| 09:30.38 | kaldemar | fling: you're dialing invalid numbers |
| 09:31.14 | fling | kaldemar: last paste is incoming call |
| 09:31.32 | fling | kaldemar: from the number I tried to call |
| 09:32.18 | fling | kaldemar: If numbers are right then the problem is at ISP? |
| 09:34.17 | kaldemar | the call from metro did not authenticate. change [metro] from friend to peer and add insecure=port,invite if they are not supposed to authenticate to you when calls come in. |
| 09:35.03 | kaldemar | nothing wrong with your dialplan really, you just need to dial a valid number. ask your provider what they expect. |
| 09:35.41 | fling | kaldemar: thanks, now incoming works |
| 09:36.55 | fling | kaldemar: so my problem is in this line? > exten => _XXXXXX,1,Dial(SIP/${EXTEN}@metro,,Tt) |
| 09:38.25 | kaldemar | fling: yes |
| 09:47.14 | curfont | any way to Dial more than one phone at the same time in the Dial Plan? |
| 09:47.21 | curfont | in extensions? |
| 09:47.40 | curfont | For example if I want the incoming call to ring on a couple of phones at the same time |
| 09:56.47 | kaldemar | curfont: Dial(tech/...&tech/...) |
| 09:56.54 | kaldemar | first to answer gets the call |
| 09:56.58 | curfont | oh, & is the seperator |
| 09:57.04 | curfont | thanks |
| 09:57.28 | curfont | my other question i asked earlier, when doing IAX2 extensions, in some cases it needs the password and user in the dialplan and others it doesnt |
| 09:57.29 | curfont | why? |
| 09:57.50 | curfont | IAX2/user:pass@host/number <- it needs it like this sometimes |
| 09:57.59 | curfont | but I usually do IAX2/host/number |
| 10:00.15 | kaldemar | taht is a syntax difference. IAX2/user:pass@host/number dials host directly with given credentials, IAX2/host/number dials using a configured peer ([host] in iax.conf). |
| 10:01.57 | *** join/#asterisk k1ng_ (~k1ng@unaffiliated/k1ng) |
| 10:02.16 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 10:02.18 | Ice_Strike | Hi |
| 10:03.46 | Ice_Strike | Why I can't hear the call: exten => 222,2,Extenspy(classification@callrecord,w) |
| 10:04.01 | Ice_Strike | I want to listen: |
| 10:04.02 | Ice_Strike | [classification] |
| 10:04.03 | Ice_Strike | exten => callrecord,1,Answer() |
| 10:06.21 | kaldemar | Ice_Strike: ExtenSpy(exten[@context][,options]) |
| 10:07.36 | Ice_Strike | oops other way way |
| 10:07.39 | Ice_Strike | around |
| 10:08.02 | curfont | kaldemar: aha, I think I must have my config wrong in that case, where should the "register" statements go? I always thought [general] |
| 10:09.01 | kaldemar | curfont: yes, they go under [general]. |
| 10:09.36 | kaldemar | Ice_Strike: what else does exten callrecord have besides Answer? |
| 10:15.05 | Ice_Strike | kaldemar Call get answered via AMI |
| 10:16.09 | kaldemar | Ice_Strike: show it. |
| 10:21.10 | Ice_Strike | kaldemar http://pastebin.com/hsCGmzJj |
| 10:25.20 | kaldemar | Ice_Strike: there's nothing to spy on. the call will just run through the extension and hang up. |
| 10:29.04 | Ice_Strike | kaldemar ? |
| 10:29.15 | kaldemar | unless the AGI does something that keeps the call up. show a CLI output of a call. |
| 10:29.27 | Ice_Strike | You see the wait() |
| 10:29.31 | *** join/#asterisk VSpike (~johncc@host217-41-3-16.in-addr.btopenworld.com) |
| 10:29.48 | Ice_Strike | its get recording in 6 seconds |
| 10:30.34 | VSpike | If I have a setup that uses SIP on an internal LAN, uses IAX trunk to reach another site, and uses a DAHDi trunk for external POTS routing, do I need NAT enabled in asterisk? |
| 10:30.37 | VSpike | I think probably not |
| 10:31.26 | VSpike | I think if I'm not sending SIP through a router I don't need NAT, but can anyone confirm? |
| 10:32.09 | kaldemar | there is no nat config on the IAX side. |
| 10:33.47 | VSpike | Right..I'm thinking about the SIP settings |
| 10:35.34 | kaldemar | that's an affirmative. |
| 10:35.35 | VSpike | Thanks |
| 10:49.20 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 10:49.24 | v0lZy | Hello |
| 10:49.45 | v0lZy | I'm editing my call flows and i have this issue where i cant pickup a call |
| 10:50.08 | v0lZy | for example.. my public number that ends in 16 is bridged to 316 |
| 10:51.05 | v0lZy | 316 then attepts to call internal number 16... and if it doesnt answer in 15 seconds, it forwards the call to internal number 00 ... now 01 sits at the other end of the room and should be able to picukp with *800 |
| 10:51.18 | v0lZy | but 01 just gets disconnected and can't pickup... |
| 10:51.37 | v0lZy | What would be the reason for this? |
| 10:51.41 | kaldemar | bridged? |
| 10:52.22 | Ice_Strike | kaldemar I dont understand why I can't hear the call and its not recording into a file as well? |
| 10:52.39 | v0lZy | kaldemar: what do u mean? |
| 10:52.54 | kaldemar | v0lZy: how does the call come in? what is 316, an extension in your dialplan? what does it dial? what is 00? what is 01? how are they configured? |
| 10:53.19 | v0lZy | 00 is a phone... as is 01 as is 16 |
| 10:53.31 | kaldemar | Ice_Strike: where's the CLI output? |
| 10:53.38 | v0lZy | 316 is an application that mediates between the public number and the internal |
| 10:54.08 | kaldemar | there are many types of phones. be more specific. |
| 10:54.18 | kaldemar | and what do you mean by "bridged"? |
| 10:54.24 | v0lZy | the idea is that when somone rings my external number, asterisk rings the application which then rings number 16... and if 16 isnt picking up, it rings 00.... and 00 is in the same room as 01... and 01 should be able to pick up |
| 10:54.43 | kaldemar | also, what "application"? |
| 10:55.08 | kaldemar | also "room" means nothing. |
| 10:58.09 | v0lZy | let me pastebin |
| 10:58.39 | v0lZy | http://pastebin.com/YfRUh6ri |
| 10:59.32 | v0lZy | but in this case |
| 10:59.37 | v0lZy | the actors are 16 and 44 |
| 11:00.11 | Ice_Strike | kaldemar http://pastebin.com/SWGk0ah8 |
| 11:00.51 | v0lZy | the overall idea is that i have an application on the 316 number. When it receives a call it rings 16, and if 16 isnt picking up within 10 seconds, it rings 44 .... now some other phone wants to pick up from 44, but it cant |
| 11:01.26 | kaldemar | v0lZy: and how are they configured? and the device that is trying to pick the call up? |
| 11:01.39 | kaldemar | v0lZy: and why have you put an Answer on the first line? |
| 11:01.59 | v0lZy | thought that was mandatory |
| 11:02.37 | kaldemar | v0lZy: an application is something like GotoIf or Dial. [CALLFLOW-15021801744ffe94234c160] is a context. CALLFLOW-START is an extension. SIP/16 is a device. |
| 11:03.20 | kaldemar | on the contrary, answering a call like that when nothing really has answered just screws up real call progress and causes you headaches. |
| 11:04.31 | v0lZy | ok ... in that case |
| 11:05.03 | v0lZy | in the CALLFLOW context, i have many callflows defined |
| 11:05.47 | v0lZy | callflow-start is an extension (916), and its supposed to ring SIP/16, and after 10 seconds, if SIP/16 doesnt pick up, ring SIP/44 ... so far so good, i have this working |
| 11:06.20 | v0lZy | but my problem is that once SIP/44 is ringing, no other SIP device can pick up the call from SIP/44 |
| 11:06.30 | kaldemar | i asked you for something. |
| 11:06.55 | v0lZy | I removed the answer at the start now, still the same problem |
| 11:08.01 | v0lZy | I dont think i understood the question then :( |
| 11:09.00 | kaldemar | v0lZy: < kaldemar> v0lZy: and how are they configured? and the device that is trying to pick the call up? |
| 11:09.47 | v0lZy | well.. they are registered to the PBX ... they are configured to pick up with *8<sip/num they want to pick up> |
| 11:09.53 | kaldemar | Ice_Strike: no pickup attempt there, only a broken AGI. |
| 11:09.56 | v0lZy | and picking up works otherwise |
| 11:10.19 | v0lZy | its just in this call flow application scenario, where *8 44 doesnt pick up from SIP/44 |
| 11:10.27 | kaldemar | v0lZy: HOW are they configured? |
| 11:10.34 | kaldemar | v0lZy: show the configs. |
| 11:10.43 | v0lZy | oh, ok |
| 11:11.05 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 11:11.51 | v0lZy | what part interests u? |
| 11:11.52 | v0lZy | outgoing? |
| 11:12.33 | kaldemar | v0lZy: the devices. |
| 11:13.34 | v0lZy | http://pastebin.com/FKy4uDFm |
| 11:13.42 | v0lZy | well like this |
| 11:13.58 | kaldemar | that's dialplan. sip.conf. |
| 11:15.34 | v0lZy | ok.. sip.conf, got it, moment |
| 11:17.09 | v0lZy | http://pastebin.com/yTx3gs9k |
| 11:17.23 | Ice_Strike | kaldemar but pickup is done by AMI ? |
| 11:18.24 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 11:18.39 | Ice_Strike | whats broken AGI? |
| 11:18.50 | kaldemar | v0lZy: no pickup or callgroups in the config. and SIP/16 is not what you were trying to pick up. |
| 11:19.08 | kaldemar | Ice_Strike: [Jul 12 11:57:05] ERROR[20235]: utils.c:1130 ast_carefulwrite: write() returned error: Broken pipe |
| 11:19.12 | v0lZy | picking up kaldemar sip/44 has same configuration, except every 16 is a 44 |
| 11:19.36 | v0lZy | and as far as pickup groups go.. .i tried with those, have no efect on the whole thing |
| 11:19.41 | Ice_Strike | kaldemar According to AMI - call has been answered though? |
| 11:20.16 | kaldemar | v0lZy: good luck. |
| 11:20.37 | v0lZy | well i can easily create a pickup group |
| 11:20.41 | v0lZy | holdon |
| 11:20.55 | kaldemar | Ice_Strike: i have no idea. i haven't seen any output from your AMI. |
| 11:22.46 | Ice_Strike | kaldemar it say this in AMI output: |
| 11:22.47 | Ice_Strike | Response: Success |
| 11:22.47 | Ice_Strike | Message: Originate successfully queued |
| 11:23.18 | v0lZy | kaldemar: the thing is that if i call 44 and pick up with 16, i dont have an issue |
| 11:23.22 | kaldemar | Ice_Strike: that says nothing about any answers. |
| 11:23.33 | v0lZy | kaldemar: but if i call 16 and try to pick up with 44, i cant |
| 11:23.38 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 11:24.29 | v0lZy | kaldemar: the only difference in configuration is this callflow application i made |
| 11:24.36 | kaldemar | v0lZy: show what you have configured or stop wasting my time. |
| 11:25.22 | v0lZy | kaldemar: sorry for all the handholding, i don tmean to waste your time. I'm using a gui thing and finding my way around is a bit difficult in the configs, im doing my best |
| 11:27.12 | *** join/#asterisk Neptu (~Neptu@41.248.231.2) |
| 11:27.19 | v0lZy | kaldemar: i've added pickupgroup=1 to both SIP/16 and SIP/44 |
| 11:27.59 | kaldemar | v0lZy: you must also configure callgroup with the same value. |
| 11:28.08 | v0lZy | kaldemar: yeah, i have |
| 11:28.18 | kaldemar | show it. |
| 11:28.20 | v0lZy | i have callgroup=1 and pickupgroup=1 under both |
| 11:28.49 | kaldemar | and show CLI output of a call with a pickup attempt. |
| 11:29.33 | v0lZy | http://pastebin.com/SEnATFjY |
| 11:30.49 | *** part/#asterisk PBXman (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 11:32.38 | v0lZy | is there a way to filter just some traffic and numbers? |
| 11:32.41 | v0lZy | (active system) |
| 11:32.51 | kaldemar | no. |
| 11:33.18 | v0lZy | so |
| 11:33.21 | v0lZy | core set verbose 9 |
| 11:34.53 | v0lZy | can i do |
| 11:35.04 | v0lZy | core set verbose 9 > test |
| 11:35.41 | kaldemar | no. but you can do "asterisk -vvvvvvvvvr | tee /tmp/asterisk_output.log" |
| 11:36.01 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 11:36.42 | v0lZy | I think i got somethingk |
| 11:37.52 | v0lZy | http://pastebin.com/uBtyAkAs |
| 11:38.26 | v0lZy | Jul 12 13:35:59 asterisk[1507]: VERBOSE[8853]: app_dial.c:1434 in wait_for_answer: -- Nobody picked up in 15000 ms |
| 11:38.26 | v0lZy | Jul 12 13:35:59 asterisk[1507]: VERBOSE[8853]: pbx.c:4074 in pbx_extension_helper: -- Executing [CALLFLOW-START@CALLFLOW-15021801744ffe94234c160:13] Dial("SIP/SIP-PROVIDER-2238567204f4621c81c410-0000207c", "SIP/44,,go") in new stack |
| 11:38.43 | v0lZy | after that i try to pickup SIP/44 with SIP/16 |
| 11:41.36 | kaldemar | so, you're using app Pickup like Pickup("SIP/16-00002080", "44@PICKUPMARK") |
| 11:41.51 | kaldemar | so callgroup and pickupgroup are not involved. |
| 11:43.09 | kaldemar | you conveniently left out the parts that might have told if you set such a mark. |
| 11:43.27 | kaldemar | read what "core show application Pickup" says about PICKUPMARK |
| 11:44.49 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-7-139.bois.qwest.net) |
| 11:45.16 | v0lZy | [*844@SIP-PHONE-16738116994f2a6c5a41c20:1] Pickup("SIP/16-00002080", "44@PICKUPMARK") in new stack |
| 11:45.16 | v0lZy | Jul 12 13:36:04 asterisk[1507]: NOTICE[8894]: app_directed_pickup.c:313 in pickup_exec: No target channel found for 44. |
| 11:45.16 | v0lZy | No target chanel found for 44 |
| 11:45.20 | v0lZy | is this a pickup directed pickup thing? |
| 11:45.53 | v0lZy | thats all i captured |
| 11:45.55 | v0lZy | its in the paste |
| 11:46.00 | v0lZy | what i pasted here is just what caught my eye |
| 11:46.04 | v0lZy | there should be more between these 2 |
| 11:46.54 | v0lZy | hm |
| 11:47.13 | v0lZy | i guess i need to ... add something to my application so that it will register a picupmark or something? |
| 11:47.50 | *** join/#asterisk deadcow777 (~deadcow@102.130.233.220.static.exetel.com.au) |
| 11:48.09 | kaldemar | the dialing channel should have Set(PICKUPMARK=44) somewhere in the dialplan for you to be able to pick it up using Pickup(44@PICKUPMARK) |
| 11:49.02 | v0lZy | i have an option to input an asterisk command |
| 11:49.04 | v0lZy | can i use |
| 11:49.29 | v0lZy | core Set(PICKUPMARK=44) or something? |
| 11:49.36 | v0lZy | before passing on to 44* |
| 11:50.17 | kaldemar | go ask the askozia folks how to modify your dialplan. |
| 11:51.20 | v0lZy | i have an option to add that in |
| 11:51.24 | v0lZy | gonna try |
| 11:51.26 | v0lZy | sec |
| 11:52.06 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 11:52.18 | v0lZy | didnt help |
| 11:55.51 | v0lZy | Jul 12 13:53:52 asterisk[1507]: VERBOSE[10668]: pbx.c:4074 in pbx_extension_helper: -- Executing [CALLFLOW-START@CALLFLOW-15021801744ffe94234c160:13] Set("SIP/SIP-PROVIDER-2238567204f4621c81c410-000020a3", "PICKUPMARK=44") in new stack |
| 11:55.54 | v0lZy | I see this in the log |
| 11:56.20 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 11:57.00 | v0lZy | Jul 12 13:53:52 asterisk[1507]: VERBOSE[10668]: pbx.c:4074 in pbx_extension_helper: -- Executing [CALLFLOW-START@CALLFLOW-15021801744ffe94234c160:14] Dial("SIP/SIP-PROVIDER-2238567204f4621c81c410-000020a3", "SIP/44,,go") in new stack |
| 11:58.14 | v0lZy | Jul 12 13:54:02 asterisk[1507]: VERBOSE[10695]: pbx.c:4074 in pbx_extension_helper: -- Executing [*844@SIP-PHONE-16738116994f2a6c5a41c20:1] Pickup("SIP/16-000020a8", "44@PICKUPMARK") in new stack |
| 11:58.14 | v0lZy | Jul 12 13:54:02 asterisk[1507]: NOTICE[10695]: app_directed_pickup.c:313 in pickup_exec: No target channel found for 44. |
| 11:58.33 | v0lZy | PICKUPMARK=44 ... that would be 44@44 ? |
| 11:59.15 | *** join/#asterisk rolandow (~rolandow@546BB29B.cm-12-4c.dynamic.ziggo.nl) |
| 12:00.52 | v0lZy | or global local var thing? |
| 12:01.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:02.56 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:03.29 | v0lZy | ah |
| 12:03.32 | v0lZy | __PICKUP ! |
| 12:03.37 | v0lZy | that does the trick i think |
| 12:04.06 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
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| 12:06.13 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:06.21 | v0lZy | ok |
| 12:06.27 | v0lZy | just 1 little detail to solve |
| 12:06.28 | rolandow | can i manually poke a sip account in the asterisk cli? |
| 12:07.15 | [TK]D-Fender | no |
| 12:07.20 | acidfoo | what you mean by "poke" |
| 12:08.19 | rolandow | acidfoo: to see if the account reacts, so it registers... i see calls from a sip account that is "unreachable" in sip peers. |
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| 12:11.00 | v0lZy | is there a command to send a BLF to all subscribers? |
| 12:12.46 | [TK]D-Fender | no |
| 12:13.29 | [TK]D-Fender | You can create a custom one that isn't attached to another device and toggle that, but you can't force any other kind of update |
| 12:14.18 | v0lZy | [TK]D-Fender: on my phones, i just set BLF numbers for other phones and they flash and everything is working ok |
| 12:14.30 | v0lZy | But in my situation, i pass an incoming call through another application |
| 12:14.42 | v0lZy | and in that scenario, it doesnt blink an incoming notification. |
| 12:14.50 | [TK]D-Fender | v0lZy, There is no options to push "refresh" data |
| 12:15.53 | v0lZy | how does BLF work then? |
| 12:15.57 | v0lZy | i see some sip notifys |
| 12:16.44 | v0lZy | this thing: Jul 12 13:54:02 asterisk[1507]: VERBOSE[1529]: chan_sip.c:13868 in cb_extensionstate: == Extension Changed 16[SIP-PHONE-8251182284f2b8e82b5bdf] new state Idle for Notify User 61 |
| 12:17.09 | [TK]D-Fender | v0lZy, When a watched device does something * send out the NOTIFY's. You can't hit "refresh" hoever. There is no option |
| 12:18.12 | rolandow | hm .. i am having one way audio problems when i dial to a sip account that can be answered by several handsets on a dect station ... when i call to a sip account that has only one handset configured on that dect staion, everything is ok .. :S |
| 12:18.37 | v0lZy | so theres no way to send out the NOTIFY so that 16 would blink when 916 is called?* |
| 12:18.56 | [TK]D-Fender | v0lZy, * sends them out to subscribers AUTOMATICALLY |
| 12:19.16 | v0lZy | ah |
| 12:19.23 | v0lZy | seems the phones BLF was fxed.. |
| 12:19.27 | v0lZy | had to reboot the phone |
| 12:19.29 | v0lZy | now it works |
| 12:19.33 | v0lZy | i now only have 1 more issue |
| 12:19.41 | v0lZy | when i hangup |
| 12:19.46 | v0lZy | the call still goes forward |
| 12:20.55 | v0lZy | i think i need to put something inbetween these 2 lines |
| 12:20.59 | v0lZy | exten => CALLFLOW-START,n(module-9),Dial(SIP/16,15,go) |
| 12:21.01 | v0lZy | exten => CALLFLOW-START,n(module-12),Dial(SIP/44,,go) |
| 12:21.22 | v0lZy | because right now, if 16 picks up and then hangsup, the call is not finished but continues to ring on 44 |
| 12:21.24 | [TK]D-Fender | v0lZy, I can't imagine why you would put "g" there |
| 12:21.27 | [TK]D-Fender | that is NOT normal |
| 12:21.42 | v0lZy | erm, i didnt do it by hand |
| 12:21.46 | v0lZy | its an interface thing |
| 12:21.49 | [TK]D-Fender | If the callee hangups it it DOES go on.. because you told it to |
| 12:21.53 | v0lZy | what does g do? |
| 12:22.02 | [TK]D-Fender | "core show application dial" <--------- |
| 12:23.01 | v0lZy | i see |
| 12:23.07 | v0lZy | proceed with the dialplan even if the caller hangsup |
| 12:23.31 | v0lZy | doest seem to make any difference though |
| 12:23.34 | [TK]D-Fender | ... |
| 12:23.35 | v0lZy | i call in, i hang up, thats that. |
| 12:23.47 | v0lZy | it doesnt go on ringing. |
| 12:24.11 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 12:24.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:24.12 | [TK]D-Fender | that is another matter and isn't the fault of your dialplan unless it's processing in "h" |
| 12:24.26 | [TK]D-Fender | And you should be looking at the actual call with IP debug to see what's really happening |
| 12:24.33 | v0lZy | oh |
| 12:24.36 | v0lZy | DESTINATION CHANNEL! |
| 12:24.49 | v0lZy | that explains it i think |
| 12:24.50 | v0lZy | hm... |
| 12:25.04 | v0lZy | so i remove the g... |
| 12:25.07 | v0lZy | and this should be ok |
| 12:29.20 | rolandow | would it be possible to use my asterisk server as gateway server as well? or would that confuse SIP just more? |
| 12:30.23 | v0lZy | ok, removed the g, lets see if it helps |
| 12:30.44 | v0lZy | nope |
| 12:30.46 | v0lZy | it doesnt help. |
| 12:30.59 | v0lZy | it still goes forward [TK]D-Fender |
| 12:31.15 | [TK]D-Fender | rolandow, * is NOT a "sip server" "proxy", or "gateway" |
| 12:31.51 | v0lZy | ah |
| 12:31.53 | [TK]D-Fender | ~b2bua |
| 12:31.53 | infobot | extra, extra, read all about it, b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 12:31.55 | v0lZy | just needed to reload |
| 12:32.48 | bulkorok | v0lZy: you have to 'dialplan reload' everytime if you change sth in extensions.conf |
| 12:33.26 | v0lZy | thanks bulkorok ill keep it in mind |
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| 12:41.04 | rolandow | [TK]D-Fender: i know .. but i can still use the same server to masquerate |
| 12:42.07 | Jo_Bell | Today I heard about SIP aggregators which can provide telephony service via calling SIP URIs. Can any of you suggest trusted names from your data? What is the advantage of using SIP aggregators in place of ITSPs? |
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| 13:40.22 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 13:42.23 | beek | waves to Katty |
| 13:42.32 | leifmadsen | Katty: use your boat! |
| 13:42.46 | beek | hello leif |
| 13:42.57 | leifmadsen | ohai |
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| 13:46.39 | KaiserSuse | Hi there! Does anyone knows if Ubuntu binaries from asterisk.org will be provided in the future? I mean, they we're merrily being updated (thanks a lot guys!) but some time in the past there have been no updates... |
| 13:48.36 | rolandow | Katty: what is asterisk? |
| 13:54.44 | Nivex | KaiserSuse: I asked about an update to the Debian packages on packages.asterisk.org yesterday. They said it's being worked on but no ETA. |
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| 13:57.19 | KaiserSuse | Nivex: Thanks a lot, mate! |
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| 14:08.49 | Faustov | Anyone got a reasonable VoIP provider they could recommend for Mumbai, without some insane routing? |
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| 14:19.10 | fubada | hi are there any upgrade steps necessary to go from 10.5.2->10.6? |
| 14:24.09 | *** join/#asterisk wonderworld (~ww@dsdf-4db55bf6.pool.mediaWays.net) |
| 14:26.01 | drmessano | Yeah, you need to install it |
| 14:28.56 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:28.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:30.07 | *** join/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 14:31.19 | leifmadsen | fubada: check UPGRADE.txt if in doubt |
| 14:31.32 | leifmadsen | usually there isn't much to be done upgrade wise in minor version releases |
| 14:44.12 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 14:46.42 | slav3_kitten | test? |
| 14:46.55 | rolandow | what are you testing? |
| 14:49.19 | slav3_kitten | that i'm still connected |
| 14:49.35 | slav3_kitten | i was doing some stupid shit to my switch remotely |
| 14:51.48 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 14:51.53 | Hive | Is there a way to fire a script once a queue member answers a call from a queue? |
| 14:52.15 | *** join/#asterisk aberrios (~aberrios@77.107.183.144) |
| 14:53.52 | WIMPy | What does "ooh323_onReceiveSetup: Unacceptable ip" mean? The IP listed looks perfectely valid to me. |
| 15:01.02 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 15:01.38 | *** join/#asterisk Neptu (~Neptu@41.251.141.156) |
| 15:07.30 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 15:09.39 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:30.04 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:30.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:35.32 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 15:36.22 | sereal-work | Hello, Is it the job of the handset to manage caller id, or is it asterisk's job to pass that information onto the handset - I am currently having a problem where people are getting their extension number show up as the incoming call - not the number of the calling party. |
| 15:37.58 | leifmadsen | sereal-work: well it kind of depends who is controlling it, because it could be both |
| 15:38.08 | leifmadsen | typically you want asterisk to control it though so you know what things should be set to |
| 15:38.32 | sereal-work | Yeah, thats what I thought. I'm more so wondering if there is something wonky going on with the hardphones - they are aastra ones. |
| 15:38.59 | sereal-work | Because I tried with my softphone and it appears to be passing the calling number. |
| 15:40.45 | [TK]D-Fender | * takes CID from the calling device and may override it based the device config in * and/or the dialplan. When sending that call out Dial() it may take into account setting from the outbound peer as well |
| 15:48.10 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:57.11 | Katty | waves to beek |
| 15:57.22 | Katty | leifmadsen: BOAT |
| 15:57.30 | Katty | i need more coffee :< |
| 15:58.03 | Katty | 3 days off soda now. i wasn't /quite/ as zombied this morning. |
| 15:58.20 | Katty | soda is evil. |
| 16:01.30 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 16:05.54 | *** join/#asterisk roflcopter0 (roflcopter@D97B72E9.cm-3-4b.dynamic.ziggo.nl) |
| 16:06.23 | roflcopter0 | Anyone here? |
| 16:06.36 | pabelanger | nope |
| 16:06.40 | pabelanger | and before you ask |
| 16:06.41 | roflcopter0 | ah |
| 16:06.42 | pabelanger | I am a bot |
| 16:06.50 | roflcopter0 | great |
| 16:07.10 | pabelanger | unknown command: please try ~info for more |
| 16:09.09 | roflcopter0 | Anyway, for you non-bots or trolls, I've been trying to hook up my webhost with my own server (hosting on a Oracle VM), however, I can't connect. While my IP is accessable, it's just not corresponding with my fsockopen() PHP function. |
| 16:09.37 | roflcopter0 | CLI message: " == Connect attempt from '10.0.0.2' unable to authenticate" |
| 16:09.41 | [TK]D-Fender | roflcopter0, And why does #asterisk care about this? |
| 16:09.52 | [TK]D-Fender | Ah, AMI |
| 16:09.56 | [TK]D-Fender | finally a related piece |
| 16:09.59 | roflcopter0 | AMI indeed. |
| 16:10.23 | [TK]D-Fender | go look at the attempt and show the config that should permit this |
| 16:10.37 | roflcopter0 | However, disregard " == Connect attempt from '10.0.0.2' unable to authenticate", that's what my CLI shows when I access my IP. |
| 16:12.51 | [TK]D-Fender | roflcopter0, if we disregard that ... what HAVE you told us? |
| 16:13.13 | Qwell | [TK]D-Fender: That he has network issues? |
| 16:13.40 | [TK]D-Fender | No... we'd need an error or something to corroborate a claim that something is "wrong" |
| 16:13.59 | [TK]D-Fender | We see nothing to correlate |
| 16:13.59 | roflcopter0 | That's the thing though, the 5038 port is forwarded etc. |
| 16:14.03 | vastina | having issues setting up a basic CO from my digium TE122 |
| 16:14.28 | [TK]D-Fender | roflcopter0, Show us firewall dumps, TCP dump, etc, and realize that * is not responsible for ANY of that |
| 16:14.29 | vastina | LED is still red, and I can't seem to get it to emulate a PRI and documentation online is being a bit of a hassle |
| 16:14.43 | roflcopter0 | One sec. |
| 16:17.38 | vastina | can anyone point me to documentation that would help me set up a north american 24 8 zero bipolar north american CO online? not the Asterisk 3rd edition book by the gifted lief |
| 16:18.14 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:18.15 | vastina | s/lief/leif/ |
| 16:18.23 | vastina | dmaned bot |
| 16:19.31 | leifmadsen | just use s/foo/bar (without the trailing /) to make the bot not fix |
| 16:19.32 | [TK]D-Fender | vastina, plenty of stuff googlable. Or you could start by showing us what you've done and we can see if you're only a small step or two from gtting it in the right state |
| 16:20.56 | vastina | leifmadsen: thanks, usually get chided for improper syntax in other channels I frequent |
| 16:22.29 | Nugget | [TK]D-Fender: I'm trying to ssh in to my asterisk box and it's asking for a passphrase. What should I do next? Am I supposed to type something? |
| 16:22.58 | [TK]D-Fender | Yes, SSH is too advanced for yuo. You should use ... telnet ;) |
| 16:23.02 | Nugget | hee |
| 16:23.07 | *** part/#asterisk KaiserSuse (c837b812@gateway/web/freenode/ip.200.55.184.18) |
| 16:23.19 | [TK]D-Fender | </comicrelief> |
| 16:23.42 | *** join/#asterisk vfabi (~fabi@host-static-37-75-105-231.moldtelecom.md) |
| 16:24.23 | WIMPy | suggests rsh |
| 16:27.19 | slav3_kitten | head desks repeatedly as he shells out for a 2.4ghz spectrum analyzer |
| 16:27.20 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:27.56 | bmoraca_work | when using asterisk 10, I should be able to use sip fax detection with g729 as my codec, correct? and receivefax to receive those faxes using t38, correct? |
| 16:30.55 | [TK]D-Fender | G.711 or DOA |
| 16:32.04 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 16:33.53 | bmoraca_work | hrm |
| 16:34.21 | bmoraca_work | isn't asterisk 10 supposed to be able to function as a t.38 gateway? |
| 16:34.54 | anonymouz666 | yes |
| 16:35.13 | bmoraca_work | ohh |
| 16:35.18 | bmoraca_work | but what's not working is the fax detection |
| 16:35.20 | bmoraca_work | i see |
| 16:35.21 | bmoraca_work | hrm |
| 16:35.42 | Qwell | Fax detection works fine, when there is audio to be detected. |
| 16:35.52 | bmoraca_work | even over g729? |
| 16:36.03 | bmoraca_work | cause for whatever reason, it's not detecting it |
| 16:36.08 | Qwell | G.729 does not count as audio. |
| 16:36.27 | bmoraca_work | so we're back to no fax detection of g729 then |
| 16:36.38 | Qwell | Nobody every said otherwise. |
| 16:36.42 | Qwell | s/every/ever/ |
| 16:37.09 | bmoraca_work | well that was my initial question...your first comment was ambiguous in that context. i was merely clarifying |
| 16:37.30 | slav3_kitten | who the hell is here |
| 16:37.33 | slav3_kitten | brb |
| 16:38.16 | *** join/#asterisk celord (~celord@201.192.210.69) |
| 16:41.11 | Igneous | I wonder if infobot supports grabs |
| 16:41.19 | Igneous | s/.*if (.*) supports/$1 butt/ |
| 16:41.30 | Igneous | neg :< |
| 16:44.29 | wdoekes | it doesn't support regex at all, afaik |
| 16:44.36 | wdoekes | s/n.t// |
| 16:45.02 | Igneous | lol, so just really basic gsubs :/ |
| 16:47.21 | *** join/#asterisk Vasya (~ks@80.80.210.41.ip.tps.uz) |
| 16:48.15 | Vasya | People hi! |
| 16:58.55 | *** join/#asterisk dorphalsig (be93997b@gateway/web/freenode/ip.190.147.153.123) |
| 16:59.05 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:00.38 | dorphalsig | hi |
| 17:01.36 | dorphalsig | I am trying to install a TDM400 with * 1.8.13, however I am not seeing the kernel dahdi modules loaded when I do dmesg |
| 17:02.03 | Qwell | Did you install them? Did you load them? |
| 17:04.07 | *** join/#asterisk mazpe (~lesterm@li315-205.members.linode.com) |
| 17:05.10 | Katty | OH WHERE IS MY HAIRBRUSH |
| 17:05.18 | mazpe | any ideas why /etc/init.d/asterisk start wont create a /var/run/asterisk/asterisk.ctl but as root if i do an asterisk -cvvvv it will start asterisk and create the .ctl file |
| 17:05.50 | *** join/#asterisk dorphalsig_ (be93997b@gateway/web/freenode/ip.190.147.153.123) |
| 17:06.05 | Qwell | mazpe: You need to run the init script as root. |
| 17:06.13 | mazpe | Qwell: i did |
| 17:06.18 | dorphalsig_ | Hello |
| 17:06.31 | dorphalsig_ | sorry, this thing just shut down |
| 17:06.36 | dorphalsig_ | ao anyway... |
| 17:06.42 | mazpe | Qwell: even if i reboot the machine... I can see asterisk in ps aux | grep asterisk and even .pid file is created, but no .ctl |
| 17:06.54 | dorphalsig_ | I am trying to setup a tdm400 card with dahdi |
| 17:07.12 | Katty | hugs Qwell |
| 17:07.13 | dorphalsig_ | but I am not even seeing the dahdi kernel module loaded |
| 17:07.24 | Qwell | Katty: I know where your brush is! |
| 17:07.50 | Katty | me too. it's in my car. |
| 17:07.57 | Qwell | Precisely where you left it. |
| 17:08.03 | Katty | mhmm |
| 17:08.36 | Katty | so you jumped out of a plane, eh? |
| 17:08.47 | Qwell | Katty: yar |
| 17:08.48 | Katty | brave soul. |
| 17:08.54 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 17:08.57 | Katty | not even i would do that |
| 17:08.59 | Qwell | I was more pushed than anything, but that's okay. |
| 17:09.12 | Katty | scary |
| 17:10.30 | *** join/#asterisk rgagnon (~rgagnon@99-185-128-25.lightspeed.austtx.sbcglobal.net) |
| 17:11.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:11.19 | rgagnon | I've been googling for a day and a half and thought I might ask here... Anyone familiar with t38 gateway configuration for asterisk and chan_sip ? |
| 17:12.05 | Qwell | https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway |
| 17:12.19 | rgagnon | read that, but its failing with 488's all the time |
| 17:13.00 | rgagnon | were on 10.5.2, and set FAXOPT(gateway-yes,15) in extensions.conf... res_fax.conf is setup, but everything hits a 488 after 5 seconds |
| 17:13.12 | rgagnon | oops..."gateway=yes" |
| 17:13.21 | Katty | gives rgagnon a fisher price fax machine! |
| 17:14.17 | rgagnon | From code, it looks like the other end sends us the T38 reinvite, asterisk sends the 100 trying back, and then just waits 5 seconds, and sends a 488 Not acceptable |
| 17:14.27 | Qwell | rgagnon: What, exactly, does your function call look like? |
| 17:15.15 | rgagnon | exten => _X!,n,Set(FAXOPT(gateway)=yes,30) |
| 17:17.53 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:21.31 | rgagnon | with t38 faxing, is it a MUST to have a "fax" extension, or just optional if you want to change the dialplan flow? |
| 17:22.12 | rgagnon | I ask because we do NOT have a "fax" extension. We just let our normal dialplan handle routing as it would do normally, but allow for t38 to go through |
| 17:23.55 | dorphalsig_ | Hi. I am running debian squeezy 64 bit and I am trying to get dahdi running, however it complains and says it could not find dahdi module |
| 17:24.04 | dorphalsig_ | can anybody give me some light? |
| 17:24.34 | dorphalsig | has already looked @ the wiki, and about 80% of the first pages in google |
| 17:26.35 | generalhan | tzafrir_laptop: you around ? |
| 17:26.59 | tzafrir_laptop | generalhan, yes |
| 17:27.31 | generalhan | wow! i figured missing you last night by 8 hours meant that it would be difficult to get a hold of you |
| 17:30.51 | generalhan | I finally got the USB firmware loaded on my XR0079, but i cant seem to get dahdi_genconf to pull the modules that i have in there... i was hoping you could give me some pointers when you have a free minute or two |
| 17:32.33 | generalhan | tzafrir_laptop: i am reading through your documentation, and one thing i keep coming back to is "wait until the Astribank has a product ID of 11x2" which i am not seeing, or not understanding -> if that has anything to do with my issue |
| 17:33.21 | dorphalsig | Hi. I am running debian squeezy 64 bit and I am trying to get dahdi running, however it complains and says it could not find dahdi module. From what I understand its something about the module version not matching the kernel version |
| 17:34.55 | [TK]D-Fender | If you compiled from source then your kernel has changed since you did so and the versions don't match. You'll need to recompile |
| 17:35.07 | tzafrir_laptop | generalhan, what version of DAHDI? Does it have an echo canceller module? |
| 17:35.59 | tzafrir_laptop | dorphalsig, If you did build it but suspect a version mismatch: uname -r; find /lib/modules -name dahdi.ko |
| 17:36.26 | generalhan | tzafrir_laptop: DAHDI is 2.6.1 ... and i thought all the units had HWEC (so im not sure, i just assumed it did) |
| 17:36.33 | dorphalsig | installed the deb package |
| 17:36.51 | tzafrir_laptop | dorphalsig, also dahdi-source? If so: m-a a-i dahdi |
| 17:37.23 | dorphalsig | so I should compile it? |
| 17:37.40 | tzafrir_laptop | dorphalsig, that what m-a should do, yes |
| 17:38.01 | tzafrir_laptop | generalhan, what's the output of: dahdi_hardware -v |
| 17:38.21 | generalhan | tzafrir_laptop: usb:002/006 xpp_usb- e4e4:1161 Astribank-modular USB-firmware |
| 17:40.11 | tzafrir_laptop | Any errors in /var/log/messages ? |
| 17:40.29 | *** join/#asterisk Johnny- (~John@23-24-48-118-static.hfc.comcastbusiness.net) |
| 17:41.27 | generalhan | tzafrir_laptop: nope |
| 17:42.08 | tzafrir_laptop | What error do you get when you run: /usr/share/dahdi/xpp_fxloader load |
| 17:43.40 | generalhan | lol, i was just typing to you that i *do* get an error running load! |
| 17:43.44 | generalhan | http://pastebin.com/bh5YTbvh |
| 17:44.13 | dorphalsig | tzafrir_laptop: same story, downloaded a bunch of stuff and then FATAL: Module dahdi not found. |
| 17:44.44 | tzafrir_laptop | dorphalsig, is the package dahdi-source installed? |
| 17:45.12 | dorphalsig | it wasnt :$ |
| 17:45.46 | *** join/#asterisk k1ng__ (~k1ng@unaffiliated/k1ng) |
| 17:46.49 | tzafrir_laptop | generalhan, any chance you don't have a USB-2 host controller? |
| 17:47.11 | dorphalsig | tzafrir_laptop: and... guess what? It didnt work! =/ |
| 17:47.13 | tzafrir_laptop | grep hci /proc/interrupts # should show ehci |
| 17:47.23 | tzafrir_laptop | dorphalsig, error message, please |
| 17:47.37 | dorphalsig | Loading DAHDI hardware modules: FATAL: Module dahdi not found. |
| 17:47.42 | dorphalsig | That's all I get |
| 17:47.55 | generalhan | tzafrir_laptop: 23: 4656 0 0 0 IO-APIC-fasteoi ehci_hcd:usb2 |
| 17:48.08 | dorphalsig | Error: missing /dev/dahdi! |
| 17:48.30 | tzafrir_laptop | dorphalsig, did you run: m-a a-i dahdi #? |
| 17:48.42 | tzafrir_laptop | generalhan, ok. |
| 17:48.49 | dorphalsig | yes, and it tries to download t he binary |
| 17:48.55 | dorphalsig | I have the source installed and all |
| 17:49.05 | dorphalsig | but it just keeps downloading the damned binary |
| 17:49.09 | UnixDev | i have some questions about how to configure asterisk with multiple parking lots… does each one need its own context in features.conf? and if so, where can I read detailed docs on this? |
| 17:49.18 | tzafrir_laptop | dorphalsig, what damaged binary? |
| 17:49.45 | dorphalsig | not damaged, damned |
| 17:49.55 | dorphalsig | which is close, but worse |
| 17:50.48 | tzafrir_laptop | dorphalsig, it explains why it doesn't work: it's damned |
| 17:51.13 | tzafrir_laptop | But please give an error message. Pastebin is your friend |
| 17:52.14 | tzafrir_laptop | generalhan, Another sanity check: was /usr/sbin/astribank_hexload built when you built dahdi 2.6.1 ? |
| 17:52.36 | tzafrir_laptop | Any chance it is an older one? |
| 17:52.55 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-80-180-30.dhcp.gldl.ca.charter.com) |
| 17:53.18 | generalhan | tzafrir_laptop: astribank_hexload is there. |
| 17:54.27 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-085.mobile.uci.edu) |
| 17:55.31 | generalhan | tzafrir_laptop: this is a brand new machine asterisk/dahdi are the only things on it. so everything that is there was put there by the versions i installed |
| 17:56.38 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:56.53 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:56.53 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:57.00 | tzafrir_laptop | The error does not look like a simple hardware issue, but still it's worth a shot: try disconnecting reconnecting the USB cable |
| 17:57.07 | tzafrir_laptop | Maybe it's badly connected |
| 17:57.15 | dorphalsig | tzafrir_laptop: http://pastebin.com/b3yGy6KW |
| 17:57.53 | generalhan | tzafrir_laptop: tried that first. moved it to each of the 3 USB ports. last night, at my wits end, i even rebooted in between each move |
| 17:59.50 | tzafrir_laptop | dorphalsig, hmm... I fied this one a while ago. Anyway: edit /etc/init.d/dahdi , add a 'exit 0' somewhere near the top, run 'dpkg --configure dahdi' and once this is done , remove the line 'exit 0' |
| 18:00.45 | *** join/#asterisk jsjc (~Adium@138.Red-2-136-107.dynamicIP.rima-tde.net) |
| 18:07.48 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 18:08.21 | dorphalsig | tzafrir_laptop: ok it configured, but I still have the same problem... |
| 18:08.48 | tzafrir_laptop | What problem? |
| 18:09.13 | dorphalsig | <PROTECTED> |
| 18:10.47 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 18:12.45 | FinboySlick | In a (fairly old) asterisk dialplan. Can extensions that begin with * have more than two digits? I'm trying to make a rule for *4XX and the I get an invalid extension for *42 (jumps to invalid before reaching the last digit). |
| 18:13.17 | Qwell | FinboySlick: What does the *4XX extension look like? |
| 18:13.26 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-80-180-123.dhcp.gldl.ca.charter.com) |
| 18:13.48 | FinboySlick | Qwell: For the time being it's just a noop() |
| 18:13.55 | FinboySlick | Qwell: I'm just testing it out. |
| 18:13.57 | Qwell | show me |
| 18:14.36 | FinboySlick | exten => *4XX,1,noop() |
| 18:14.47 | Qwell | So your phone has an X key? |
| 18:15.31 | FinboySlick | Qwell: I forgot the underscore, didn't I? |
| 18:15.37 | Qwell | Did you? |
| 18:15.57 | FinboySlick | Let's find out. |
| 18:18.43 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-mkzygkhejlvhivfd) |
| 18:23.27 | leifmadsen | FinboySlick: you did |
| 18:23.32 | leifmadsen | _ == pattern match |
| 18:23.40 | leifmadsen | otherwise *4XX would match literally |
| 18:24.21 | FinboySlick | leifmadsen: Yup. |
| 18:24.28 | Qwell | leifmadsen: oh, sure, spoil the surprise. |
| 18:25.17 | leifmadsen | :) |
| 18:25.52 | FinboySlick | leifmadsen: Don't encourage my weaknesses, you'll make him mad. |
| 18:25.55 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:26.21 | leifmadsen | that's ok, I don't mind when I see Qwell beat people down |
| 18:27.16 | FinboySlick | I remember a somewhat longer nick with a similar personality a while back. Did he change it? |
| 18:28.34 | FinboySlick | Ah, no, the one I was thinking of was [TK]D-Fender. |
| 18:32.36 | [TK]D-Fender | I wouldn't qualify that as a "beat down" at all. More than a few levels short of deserving that title |
| 18:33.19 | [TK]D-Fender | but nice to know where personalized standards fall in the meantime of course. |
| 18:33.20 | FinboySlick | [TK]D-Fender: Well, I mostly remember you as helpful. |
| 18:33.44 | *** join/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 18:33.47 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-80-180-123.dhcp.gldl.ca.charter.com) |
| 18:34.14 | [TK]D-Fender | FinboySlick, I'm sure you feel at most a little silly for the error you now picked up on. Life goes on |
| 18:34.33 | FinboySlick | I remember an asterisk speech at some large open source conference where the speaker invited people to join the project on IRC, with a big disclaimer that some people are mean but that they all mean well. |
| 18:34.44 | FinboySlick | [TK]D-Fender: Of course. And amused. |
| 18:43.36 | drmessano | http://en.wikipedia.org/wiki/Online_disinhibition_effect <---- Goes for all things IRC.. Whether it's a big open source project or support for my script that just writes "poop" to /dev/ttyS0 |
| 18:46.15 | jgowdy | I like the penny arcade version better |
| 18:46.23 | Qwell | wait, was I just called mean? |
| 18:46.57 | drmessano | Well good for you.. nobody told you to look at the link |
| 18:47.00 | jgowdy | http://www.penny-arcade.com/comic/2004/03/19 |
| 18:47.37 | FinboySlick | Qwell: Nah. I think you were okay... In fact, your approach is a lot better than hand-holding in most cases. |
| 18:48.36 | mazpe | any thoughts on what will cause (as root or after a reboot) /etc/init.d/asterisk start not to create a /var/run/asterisk/asterisk.ctl but if i do an asterisk -cvvvv it will start asterisk and create the .ctl file |
| 18:50.34 | drmessano | mazpe: probably something not satisified in the safe_asterisk script, which is what is called in the init script |
| 18:50.46 | drmessano | Try running safe_asterisk |
| 18:53.10 | mazpe | safe_asterisk runs just fine... with a .ctl |
| 18:53.39 | mazpe | but /etc/init.d/asterisk start wont create the .ctl |
| 18:53.47 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 18:53.47 | mazpe | this is the second server i install where this happens. |
| 18:54.07 | FinboySlick | Qwell: ... which reminds me to thank you. |
| 18:54.27 | mazpe | asterisk 1.8.4 on centos 6.2 |
| 18:55.19 | *** part/#asterisk FinboySlick (~shark@74.117.40.10) |
| 18:58.43 | drmessano | 1.8.4? |
| 19:03.28 | treborsux | I have pri system and I need to breakout a couple fxs lines for fax lines what device do you guys recomend? I have a generic fxs-04a and it does not work seems real junky. |
| 19:04.55 | treborsux | Any way to program these 14 vonage linksys things i have around? |
| 19:04.58 | *** join/#asterisk vfabi (~fabi@host-static-37-75-83-144.moldtelecom.md) |
| 19:06.02 | dorphalsig | Hi. I am running debian squeezy 64 bit and I am trying to get dahdi running, however it complains and says it could not find dahdi module. From what I understand its something about the module version not matching the kernel version. I've already tried getting the source of dahdi and m-a a-i dahdi and it wont work |
| 19:06.30 | Qwell | Haven't you asked this question like 7 times now? |
| 19:07.04 | mazpe | drmessano: 14 |
| 19:07.11 | dorphalsig | Qwell: Yeah, and have got no results yet. This is called persevering |
| 19:07.13 | mazpe | drmessano: 1.8.14 |
| 19:07.25 | jpsharp | treborsux: Some of the vonage boxes are unlocked/unlockable. Some of them are not. |
| 19:07.38 | dorphalsig | Qwell: maybe you could help me get dahdi running? |
| 19:07.57 | dorphalsig | Its just a simple idea. I have a TDM400 and I need it running :P |
| 19:08.27 | treborsux | I have chased several geese so far with this. So I am going to try to start from the top. This is what happens: A call comes to an extension (this may be extension to extension, pri or dahdi) the extension picks up the call and they hear ringing. The caller can hear the other end but untill it is put on hold and resumed all that is heard on the side called is ringing. This seems to happen randomly. Also somet |
| 19:08.37 | treborsux | Thats my question that seems to have no answer |
| 19:09.05 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 19:09.53 | Qwell | treborsux: Show us the XML file. |
| 19:09.58 | treborsux | drmessano i guess the only answer is that it is elastix yet i cant find it happening to anyone else |
| 19:10.42 | Qwell | dorphalsig: If the packages don't work, don't use them. Install from source. |
| 19:11.35 | dorphalsig | Qwell: Now that's wisdom! :P |
| 19:11.41 | *** part/#asterisk rgagnon (~rgagnon@99-185-128-25.lightspeed.austtx.sbcglobal.net) |
| 19:12.04 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 19:12.17 | dorphalsig | downloads the sources |
| 19:12.55 | treborsux | http://pastebin.com/sK4PT3M9 xml fer me phone |
| 19:14.09 | *** join/#asterisk luckman212 (~irc@2001:470:8abb:0:c9e0:2297:73ad:9050) |
| 19:19.20 | *** part/#asterisk Vasya (~ks@80.80.210.41.ip.tps.uz) |
| 19:21.49 | treborsux | any ideas? |
| 19:26.07 | dorphalsig | Qwell: I just found out the issue. You need to install dahdi-modules |
| 19:26.07 | treborsux | qwell any ideas. I may have to go use a diffrent distro |
| 19:26.09 | dorphalsig | not just dahdi, or dahdi-linux or asterisk-dahdi |
| 19:26.22 | [TK]D-Fender | treborsux, Distro has nothing to do with this |
| 19:26.26 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 19:26.47 | [TK]D-Fender | terYour information is weak, proves nothing about local overrides on the phone itself and you're in the WRONG CHANNEL |
| 19:26.53 | [TK]D-Fender | treborsux, ^ |
| 19:28.29 | treborsux | I was asking for general help |
| 19:28.39 | treborsux | any ideas |
| 19:30.36 | [TK]D-Fender | treborsux, I just gave you some |
| 19:30.50 | [TK]D-Fender | And take this to #freepbx where it belongs |
| 19:31.41 | carrar | Install Windows Server 8 |
| 19:31.43 | carrar | and be done with it |
| 19:38.12 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.40.134) |
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| 20:11.06 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-xdlwfducmqzbnkrq) |
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| 20:21.06 | navaismo | Hi, where are the sounds "vm-youhaveno" and "digits/1M" in asterisk 1.8.11-cert2, also I downloaded the core sounds package but these sounds not exist |
| 20:23.59 | Qwell | What language? |
| 20:24.12 | Qwell | Those do not exist in English. |
| 20:24.49 | navaismo | yep English |
| 20:24.56 | *** join/#asterisk gbit (~no@unaffiliated/gbit) |
| 20:25.00 | Qwell | app_voicemail does not use those prompts for English. |
| 20:25.58 | navaismo | hmm weird this is the log http://pastebin.com/FxBER2xr |
| 20:26.12 | Qwell | es is not English. |
| 20:26.19 | navaismo | dammit |
| 20:26.19 | gbit | Hello, I'm having problems to enable echo training on dahdi. I have 3 asterisk servers, same instalation, only one server does not work. The error is: dahdi_train_ec: Unable to request echo training on channel 1: Invalid argument . When cat /dev/dahdi/1 I see no echo enabled too. |
| 20:26.22 | navaismo | im blind |
| 20:26.34 | navaismo | facepalms |
| 20:26.41 | navaismo | thanks Qwell |
| 20:27.01 | sruffell | gbit: Do you have echocanceler set in /etc/dahdi/system.conf ? |
| 20:27.12 | gbit | sruffell, yes I have |
| 20:27.19 | gbit | mg2,1-4 |
| 20:27.40 | sruffell | is dahdi_echocan_mg2 loaded in the kernel? |
| 20:27.46 | sruffell | on the server that is failing? |
| 20:28.39 | gbit | sruffell, actualy I have dahdi_echocan_hpec loaded, but when hpec was enabled on system.conf I got the same error. |
| 20:29.12 | sruffell | Do you see that the license was activated in the kernel log? |
| 20:29.36 | sruffell | and if hpec is loaded, you should have hpec in /etc/dahd/system.conf, right? |
| 20:30.29 | gbit | right, let me check the kernel log, thanks |
| 20:30.46 | sruffell | np |
| 20:31.12 | gbit | kernel: hpec_license_check: License granted for 4 channels |
| 20:32.15 | sruffell | What is on the EC: line in /proc/dahdi/1 for channel 1? |
| 20:33.01 | gbit | sruffell, I guess that is the problem, I dont see EC on it. 1 WCTDM/0/0 FXSKS (In use) |
| 20:33.03 | sruffell | i.e. when I have MG2 defined in /etc/dahdi/system.conf for the channel, I see: (EC: MG2 - INACTIVE) |
| 20:33.24 | sruffell | any error when you ran dahdi_cfg? |
| 20:33.33 | gbit | let me try out |
| 20:33.57 | gbit | nope |
| 20:34.21 | sruffell | When you run "dahdi_cfg -vvf" Do you see "Setting echocan for channel 1 to hpec"? |
| 20:35.04 | gbit | no, only Echo Canceller(s): HWEC, HPEC |
| 20:35.37 | sruffell | and you have a line that says "echocanceller=hpec,1" in your /etc/dahdi/system.conf |
| 20:35.39 | sruffell | ? |
| 20:35.45 | mazpe | any recommendations for asterisk to rival Inter-tel Attendand Console ? |
| 20:36.03 | gbit | yes = echocanceller=hpec,1-4 |
| 20:36.20 | mazpe | http://tinyurl.com/5na2m4 |
| 20:37.14 | sruffell | gbit and only one of your servers has this problem? |
| 20:37.44 | gbit | sruffell, yes! I did the same instalation on they |
| 20:38.53 | sruffell | can you pastebin your /etc/dahdi/system.conf and the output from dahdi_cfg -vvf ? |
| 20:39.05 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:39.05 | gbit | sure, one moment please |
| 20:41.30 | gbit | sruffell, http://pastebin.com/g9XHcsFE |
| 20:41.55 | sruffell | ahh..no channels are defined. |
| 20:41.59 | sruffell | in /etc/dahdi/system.conf |
| 20:42.53 | mazpe | I guess that asterisk Flash Operator Panel |
| 20:43.54 | sruffell | gbit: you could load the driver, the run dahdi_genconf system, then edit change mg2 to hpec in /etc/dahdi/system.conf that dahdi_genconf generated. |
| 20:44.06 | sruffell | edit or change…pick one :) |
| 20:47.19 | gbit | sruffell, totaly my bad, thanks alot for helping! I'm testing now |
| 20:47.26 | sruffell | np |
| 20:49.00 | gbit | sruffell, it works! thanks again, what ec should be the best one? |
| 20:49.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:49.58 | sruffell | if you have HPEC licenses…those will most likely be the best for you to use. |
| 20:50.23 | gbit | allright |
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| 21:13.26 | xTina | Hey, I've got a question about tracking the busy state of queue members ... |
| 21:13.31 | xTina | My queue members and queues are configured in realtime. My extensions are looked up dynamically via ODBC in the dialplan. An extension is calling multiple SIP channels. I'm looking for a way for queues to be aware of the (combined) busy state of a member's SIP channels (similar as in hints). Does anyone know how that could be accomplished? |
| 21:15.58 | xTina | I have seen that it's possible for the state_interface column of the queue_member table to actually use an extension's hint, but I have no idea how I could create hints for extensions and associated SIP channels that are looked up dynamically from a database, not hard-coded in the dialplan. |
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| 23:23.43 | mcf3782 | I've started getting a lot of these messages: |
| 23:23.46 | mcf3782 | [Jul 12 19:06:17] NOTICE[1507] res_jabber.c: Got presence packet from xdivajessz2@aol.com, someone not in our roster!!!! |
| 23:24.13 | *** join/#asterisk Sean-Der (~sean@cpe-68-175-54-64.nyc.res.rr.com) |
| 23:24.44 | Sean-Der | I am running Asterisk 1.8 on Debian and not having any luck getting it to work. |
| 23:24.47 | Sean-Der | http://pastebin.com/jfysEuSC |
| 23:24.50 | mcf3782 | Any way to block/filter this account so that I don't see traffic from them? It's probably a spammer that has my gmail address in their address book. |
| 23:24.53 | Sean-Der | Would some one mind dumping theire config |
| 23:25.14 | Sean-Der | fail2ban-client -d | grep asterisk |
| 23:25.20 | Sean-Der | I can't seem to get mine to work right :/ |
| 23:26.27 | Sean-Der | Or any FreePBX users mind dumping yours. I think you guys have it out of the box |
| 23:41.01 | navaismo | Sean-Der, a spanish link but i think with google translate can see it http://asteriskmx.com/2011/07/protege-tu-asterisk-de-ataques-usando-fail2ban/ |
| 23:46.02 | mcf3782 | Sean-Der: https://dl.dropbox.com/u/20463157/fail2ban.tar.gz |
| 23:46.35 | mcf3782 | That's the stock fail2ban from my lab FreePBX 2.9.0 box |
| 23:53.19 | Sean-Der | Thanks guys! |
| 23:53.25 | mcf3782 | np |
| 23:53.44 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:53.44 | *** mode/#asterisk [+o sruffell] by ChanServ |