| 00:08.36 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:09.27 | neurosys | has anyone had experience with *1.6 and cbeyond communications? |
| 00:13.10 | russ | what was the thing where you could get the asterisk console to print help on a dialplan command? |
| 00:13.46 | russ | nm, core show application <app> |
| 00:16.25 | russ | 'r' doesn't give me ringing either |
| 00:20.13 | SiNGLer | russ: maybe your device has disabled ringer? |
| 00:21.39 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 00:24.28 | russ | ? |
| 00:24.37 | russ | I've called my DID from several devices |
| 00:24.56 | russ | for all of them, I receive total silence until I pick up my extension |
| 00:25.11 | SiNGLer | I mean end device, from which you pickup |
| 00:25.45 | russ | my end device is a dahdi card |
| 00:25.47 | *** join/#asterisk Bloudermilk (~Bloudermi@dsl081-234-075.lax1.dsl.speakeasy.net) |
| 00:25.56 | russ | there are 3 analog phones connected to it and I'm calling all three |
| 00:27.07 | SiNGLer | oh, can you connect sip phone/softphone, and check if it works? |
| 00:28.20 | russ | I don't have one |
| 00:28.39 | Bloudermilk | Hey guys, in asterisk 1.6.2.11, if I sent the HANGUP command over AGI, shouldn't I be able to keep an open TCP connection but hangup the channel? |
| 00:29.09 | Bloudermilk | For some reason my softphone is staying connected after HANGUP until I close the TCO connection |
| 00:29.11 | Bloudermilk | *TCP |
| 00:29.32 | Bloudermilk | http://pastie.org/1119259 |
| 00:29.42 | Bloudermilk | Those are the results from the logger |
| 00:29.55 | russ | when I call from my IP phone at work, I don't even get a comfort tone, its like the line is dead |
| 00:29.56 | Bloudermilk | "Unexpected: " just means I wasn't waiting on a response from a command |
| 00:44.11 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 00:53.22 | *** join/#asterisk csd-199 (~asdf@189.237.58.126) |
| 00:57.34 | csd-199 | Hi. I updated asterisk 1.6 using update in centos... was working fine, after the update is not working, shows a message: include => numberplan-custom-1 |
| 00:57.34 | csd-199 | no... sorry |
| 00:57.34 | csd-199 | Hi. I updated asterisk 1.6 using update in centos... was working fine, after the update is not working, shows a message: Call from '207rey' to extension '201' rejected because extension not found in context 'default'. |
| 00:57.34 | csd-199 | it was working fine, and I have the context default in sip.conf and extenstions.conf |
| 00:57.35 | csd-199 | what can be the problem? |
| 00:59.32 | *** join/#asterisk coppice (~chatzilla@140.145.17.210.dyn.pacific.net.hk) |
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| 01:35.08 | pabelanger | csd-199: *CLI> dialplan show 201@default |
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| 02:07.21 | csd-199 | does not show that extension, but it is in extensions.conf |
| 02:07.30 | csd-199 | may be it is reading other file |
| 02:09.41 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:12.24 | sawgood | I'm using Asterisk 1.4.33.1, and I was wondering what the CLI command would be to find out if my * box is 'registered' to the ITSP (no registration string is required by the ITSP) ... so sip show registry does not work |
| 02:13.05 | sawgood | I would like to know if after a reboot if the * box is registered at the ITSP |
| 02:15.52 | *** join/#asterisk theslob (kvirc@41.214.197.217) |
| 02:19.51 | *** join/#asterisk coppice (~chatzilla@m121-202-48-106.smartone-vodafone.com) |
| 02:20.17 | *** join/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 02:25.49 | sawgood | Is it possible to register more than one SIP account to the same provider using Asterisk? |
| 02:27.52 | *** join/#asterisk TimeRider (~steve@94.197.47.243.threembb.co.uk) |
| 02:28.48 | *** join/#asterisk TimeRider (~steve@94.197.47.243.threembb.co.uk) |
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| 02:45.22 | ChannelZ | yes |
| 02:46.12 | sawgood | Hi ChannelZ: how are you? |
| 02:46.24 | ChannelZ | lovely |
| 02:46.26 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:47.23 | sawgood | Well, can you tell me how I can register multiple trunks to the same ITSP on an Asterisk box? |
| 02:47.53 | sawgood | I have found a good article on the Internet which describes exactly what I want to do ... but no answer given for the solution |
| 02:48.58 | sawgood | http://pastebin.org/785575 |
| 02:49.13 | ChannelZ | Do you actually have multiple *accounts* or just multiple DIDs? |
| 02:49.21 | csd-199 | I found my problem... may be one of you know about it. The updated asterisk is not reading now the extensions.conf, it is reading extensions.ael instead |
| 02:49.30 | csd-199 | any of you know about it? |
| 02:49.47 | sawgood | I have both multiple accounts (username/passwords) are different and both have more than one DID |
| 02:49.54 | sawgood | I have incoming working 100% ... |
| 02:50.14 | sawgood | Outgoing calls always FAIL if more than one trunk is registered to the same provider |
| 02:50.54 | ChannelZ | csd-199: yeah. If you don't use it, just remove that config file |
| 02:50.54 | sawgood | I have no billing process setup on the * box, and this is a multi tenant operation so to speak |
| 02:51.15 | ChannelZ | show a failed call. |
| 02:54.19 | csd-199 | that is what I did... (well, renamed it). Is asterisk going to have new configuration structure or what? |
| 02:54.51 | ChannelZ | no |
| 02:55.07 | ChannelZ | it's just an alternate 'language' to write dialplans in if you want |
| 02:55.46 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 02:55.48 | sawgood | I do not understand this ... I simply added the 2nd account again ... did a sip/dialplan reload ... and both inbound and outbound calls are working now |
| 02:56.04 | sawgood | This can't be ... it failed over and over and over ... and now its working? |
| 02:57.03 | ChannelZ | We don't know why it failed, or what you're dialing, or how it's configured, etc...so I'm not sure what you want us to say |
| 02:57.42 | ChannelZ | you're either getting lucky now, or it was setup wrong the last time. |
| 02:58.25 | sawgood | Well ... in theory ... should multiple trunks (accounts) (to one single) ITSP IP address work? |
| 02:58.48 | ChannelZ | Again, yes |
| 02:58.49 | sawgood | This topic is documented pretty heavily on the Internet as not working under Astersk |
| 02:58.52 | *** join/#asterisk mdiehl (~quassel@173-10-242-193-Albuquerque.hfc.comcastbusiness.net) |
| 02:59.11 | *** part/#asterisk csd-199 (~asdf@189.237.58.126) |
| 03:00.39 | mdiehl | Hi all. |
| 03:02.01 | ChannelZ | Well, the internet is always right so you should just give up. It's clearly not possible. |
| 03:03.03 | mdiehl | Anyone have any experience with RealTime? |
| 03:04.05 | mdiehl | Just can't seem to RT Voicemail working |
| 03:05.01 | mdiehl | I've got the tables and modules configured. |
| 03:05.45 | mdiehl | I've added mailbox record to the database. |
| 03:06.13 | mdiehl | But when I try "voicemail show users for" and press tab, I don't see the new mailbox. |
| 03:06.17 | mdiehl | Any ideas? |
| 03:06.44 | mdiehl | Asterisk 1.6.2.9, btw. |
| 03:08.59 | ChannelZ | Sorry I don't use realtime. Does just 'voicemail show users' show nothing? |
| 03:12.09 | Kyosh | mdiehl: i do |
| 03:12.36 | Kyosh | "voicemail show users for" and press tab, I don't see the new mailbox. |
| 03:12.38 | Kyosh | and you wont |
| 03:12.52 | Kyosh | realtime cannot see what is not loaded/cached |
| 03:13.03 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 03:13.06 | Kyosh | many features/commands are disabled at the console in realtime |
| 03:13.22 | Kyosh | well, not disabled. |
| 03:13.26 | Kyosh | just dont work |
| 03:14.34 | Kyosh | like sip show peers, sip show users, sip show subscriptions |
| 03:14.48 | mdiehl | So, this isn't a valid way to test, then? |
| 03:14.53 | Kyosh | nopers |
| 03:15.00 | mdiehl | I don't have anything in voicemail.conf, yet. |
| 03:15.07 | Kyosh | dont matter |
| 03:15.13 | Kyosh | users are realtime, right? |
| 03:15.17 | mdiehl | Was hoping to put it ALL in the database. |
| 03:15.29 | mdiehl | Trying to get voicmail working, first. |
| 03:15.36 | *** join/#asterisk slobsor (kvirc@41.214.148.85) |
| 03:16.15 | Kyosh | first before realtime or.....? |
| 03:16.42 | mdiehl | Trying to get realtime voicemail working. |
| 03:16.56 | mdiehl | Before moving to rt sip. |
| 03:17.07 | mdiehl | My extensions are done via agi. |
| 03:17.19 | Kyosh | umm |
| 03:17.31 | mdiehl | Am I barking up the wrong tree? |
| 03:17.31 | Kyosh | you are going to use agi for users and not database? |
| 03:17.52 | mdiehl | extionsions = users? |
| 03:18.13 | Kyosh | yes |
| 03:18.18 | mdiehl | Then, yes. |
| 03:18.30 | mdiehl | The agi does cools things with the database, though. |
| 03:19.09 | Kyosh | so you're going to use AGI to access and query the database instead of going directly to the database? |
| 03:19.38 | mdiehl | Ya, because the logic is much easier to do in perl, than dialplan. |
| 03:19.50 | Kyosh | aye |
| 03:19.54 | Kyosh | or lemme say |
| 03:19.55 | Kyosh | oye |
| 03:20.07 | mdiehl | <grin> |
| 03:20.36 | Kyosh | i use realtime from mysql and the dialplan is so damn easy |
| 03:20.41 | mdiehl | So, the only way to TEST the rt voicemail is to actually USE it. |
| 03:20.42 | Kyosh | i cant even begin to tell you |
| 03:20.47 | Kyosh | yup |
| 03:21.13 | mdiehl | Then I guess I need to start configuring rt sip and put a phone on it. |
| 03:21.22 | Kyosh | yup |
| 03:21.25 | Kyosh | xlite will do |
| 03:21.31 | mdiehl | Will do, then. |
| 03:22.23 | mdiehl | Tell me about how you do dialplan in mysql. |
| 03:22.36 | Kyosh | wow where should i begin |
| 03:22.38 | mdiehl | Don't you have to put each LINE of the dialplan in a separate record. |
| 03:22.51 | Kyosh | voip-info.org has a good mysql table layout for extensions.conf realtime |
| 03:22.53 | mdiehl | ya, I knew that was a loaded question. |
| 03:23.01 | mdiehl | I've seen the sql scheme. |
| 03:23.05 | Kyosh | its not loaded, not at all |
| 03:23.16 | Kyosh | yes each line needs its own record |
| 03:23.25 | Kyosh | i wrote a webscript to manage the dialplans |
| 03:23.30 | Kyosh | pretty easy |
| 03:23.36 | mdiehl | Then inserting in the middle is tricky, right? |
| 03:23.40 | Kyosh | its just like asterisk understands it, step by step |
| 03:23.51 | Kyosh | nopes, just reorder the priorities |
| 03:24.05 | Kyosh | rather than using n for priority, i always use priority numbers |
| 03:24.20 | mdiehl | Then, you have to renumber after the insert. |
| 03:24.31 | Kyosh | so if i have 10 lines in a plan, just change the priorities to reorganize |
| 03:27.25 | Kyosh | the reason i like realtime is because i have about 50,000 dialplans, 2gb in my server and at 120 channels concurrent use, i'm only using 500MB with G711 and canreinvite=no |
| 03:27.43 | mdiehl | Guess I'm going to go off and try to get rt sip working. |
| 03:27.49 | mdiehl | You have 50K customers? |
| 03:28.07 | Kyosh | no, i have 50k dialplans |
| 03:28.16 | Kyosh | i have a few more customers than that |
| 03:28.23 | mdiehl | Wow1 |
| 03:28.52 | Kyosh | i wish it were more. i expected more by now |
| 03:29.13 | *** join/#asterisk GameGamer43|Mac (~GameGamer@65.27.76.78) |
| 03:29.32 | mdiehl | Where are you located, then? |
| 03:29.41 | Kyosh | nyc |
| 03:29.53 | mdiehl | NM. |
| 03:31.32 | Kyosh | how dee do da |
| 03:32.20 | Kyosh | you building a hosted pbx system? |
| 03:33.09 | mdiehl | Essentially, yes. |
| 03:33.15 | mdiehl | You? |
| 03:33.17 | Kyosh | startup biz? |
| 03:33.23 | mdiehl | Yup. |
| 03:36.35 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:55.20 | *** join/#asterisk russ (~russ@70.99.169.146) |
| 03:56.46 | russ | Is asterisk supposed to support early media for IAX2? |
| 04:04.31 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.229.54.adsl.dynamic.totbb.net) |
| 04:05.47 | BugKhaM | does sip.conf support authentication without username/password? |
| 04:15.43 | sawgood | I have three SIP trunks pointing to a single ITSP (each trunk has multiple DID numbers) ... outbound calling works 100% ... inbound calls fail to any DID 'except' for the 1st trunk ... |
| 04:15.46 | sawgood | http://pastebin.com/VcKmPWYW |
| 04:15.56 | sawgood | shows the failed incoming call ... |
| 04:16.56 | BugKhaM | or is there a way to access IP address variable in the dialplan |
| 04:17.11 | russ | ok, I can play early media with Progress(); Playback(tt-weasels,noanswer); |
| 04:17.20 | sawgood | can't you use (SIP/IP address)? |
| 04:17.22 | russ | but I can't figure out how to just send ringing audio |
| 04:20.22 | russ | Progress(); Dial(<exten>,,r); does not work |
| 04:26.53 | russ | Progress(); Dial(<exten>,,m); *does* work |
| 04:28.16 | ChannelZ | BugKhaM: yes, insecure=invite,port |
| 04:28.49 | ChannelZ | BugKhaM: but it has to match a peer *somehow* |
| 04:29.02 | russ | why would Dial(<exten>,,r); fail to send audio to the ringing party, but Dial(<exten>,,m) send audio ok? |
| 04:29.05 | ChannelZ | Unless you are talking about totally anonymous |
| 04:31.37 | ChannelZ | ARGH why must people RAR shit in a torrent |
| 04:31.51 | BugKhaM | ChannelZ, some of my clients want to use IP authentication instead of username/password |
| 04:31.59 | russ | And inside is a compressed executable installer |
| 04:32.23 | BugKhaM | ChannelZ, not for anonymous |
| 04:33.15 | ChannelZ | well there you go then |
| 04:39.01 | BugKhaM | ChannelZ, and where do I specify their IP addresses? |
| 04:41.32 | ChannelZ | host=x.x.x.x |
| 04:42.58 | BugKhaM | ChannelZ, OK, thanks |
| 04:45.26 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
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| 04:56.07 | p3nguin | bugkham: type=peer does not match on username, it matches on IP/port. You still need the secret, though. |
| 05:08.49 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:21.48 | akoma1s | how should I go about allowing anyone on the internet to call a specific extension on my box? |
| 05:36.57 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 05:38.19 | carrar | http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest |
| 05:40.36 | *** join/#asterisk HaMYaI (~BugKhaM@125.25.18.160.adsl.dynamic.totbb.net) |
| 05:42.15 | akoma1s | carrar: thank you! |
| 05:42.28 | *** join/#asterisk Bloudermilk (~Bloudermi@cpe-76-90-15-162.socal.res.rr.com) |
| 05:44.43 | HaMYaI | p3nguin: I'm using type=friend and it's for incoming sip calls, should it still be a problem? |
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| 05:46.26 | *** part/#asterisk GameGamer43|Mac (~GameGamer@65.27.76.78) |
| 05:52.40 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 05:54.15 | russ | is there a way to reject an IAX call without answering it? |
| 05:56.09 | *** join/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 05:56.35 | [TK]D-Fender | russ: Yes. Don't answer |
| 06:03.19 | russ | I have a rollover to my cellphone in case my pbx goes down |
| 06:04.09 | russ | hmm...I suppose I could play early audio until just before the rollover time out, then answer and hangup hoping that they hangup on their own before then |
| 06:04.26 | russ | ...course...if it is a call center, they never would hear the early audio |
| 06:04.42 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 06:05.46 | russ | btw,why would Dial(<exten>,,r) not send early media, but Dial(<exten>,,m) does? |
| 06:07.53 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 06:10.26 | *** join/#asterisk ankur_6997 (~Dev_1@122.177.234.204) |
| 06:10.42 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 06:12.54 | ankur_6997 | hi i am trying to use my mobile phone as FXO/FS port using a bluetooth adaptor but mobile search command give output as |
| 06:12.55 | ankur_6997 | mobile search |
| 06:12.55 | ankur_6997 | Address Name Usable Type Port |
| 06:12.55 | ankur_6997 | 7E:96:E8:9C:66:01 the sting operator No Headset 0 |
| 06:12.55 | ankur_6997 | 00:26:E2:E8:8E:DB LG GS290 No Headset 0 |
| 06:13.21 | ankur_6997 | so it takes my phone as Headset |
| 06:13.30 | ankur_6997 | what should i do ? |
| 06:15.42 | Bloudermilk | In Asterisk 1.6.2.11, after sending "HANGUP" over AGI, my soft phone is staying connected until I close the TCP socket. Isn't this incorrect behavior? |
| 06:17.49 | drclue | @Bloudermilk at the point your softphone is still connected , have you returned from your AGI to the dialplan? |
| 06:18.55 | Bloudermilk | I haven't |
| 06:19.11 | Bloudermilk | Returning would be closing the TCP socket. right? |
| 06:21.57 | drclue | I would think it should close the connection by that point :) The thought is that if the application (AGI) is still in play it *might* be holding the conversation open. These days I do almost all my work through the AMI so I'm a little out of practice with my AGI stuff. |
| 06:23.57 | drclue | Won't be out of practice for long though as I'm wiring my AGI and AMI tools together soon |
| 06:24.19 | Bloudermilk | What language do you write them in? |
| 06:26.39 | shamelessn00b | Anyone here wanna help me out why sip show channelstats is not working properly |
| 06:26.40 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 06:27.00 | drclue | Well actually while I write in a lot of languages , these particular AGI/AMI tools are written in PHP as persistent processes (servers). The whole mess acts as a middleware that allows me to write all my applications in HTML5 Javascript , with no polling. The latest tool I'm bringing out allows those browser applications to be deployed as desktop executable applications with zero modifications to the web browser code |
| 06:27.21 | shamelessn00b | It does not update the jitter/packet loss value even when there is significant degradation of audio quality on the trunk |
| 06:27.49 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 06:28.00 | shamelessn00b | either that or maybe you can point me to a different tool, to allow for SIP trunk monitoring |
| 06:28.34 | the_weard | we have a verry big fammily and house 20 fammily members and one uncapped router not a good idea the sharing is terrible is there a app i can install so i can capp my sisters and brothers? |
| 06:28.48 | the_weard | can some on plz point me in the right direction |
| 06:29.18 | shamelessn00b | the_weard: I use linux tool 'tc' for trafficshaping |
| 06:30.09 | drclue | @the_weard make friends with your router's web based interface. You can turn your siblings into 300 baud modems if you want |
| 06:30.16 | shamelessn00b | but you need to have a good networking background in order to understand the tool |
| 06:30.47 | Bloudermilk | drclue: sounds interesting. Are you using websockets? |
| 06:30.58 | Bloudermilk | For the browser tools, I mean |
| 06:31.15 | the_weard | yeah i use to use clear os or pfsence but i want a fisical cap like 2.5gig they are allowed to be used and after that total cuttoff cos they cost me R3500 this month with the usage and then they claim it wasnt them! |
| 06:31.49 | drclue | @Bloudermilk Yes , I use HTML5 WebSockets and my own home built WebSocket server http://asterclick.drclue.net |
| 06:32.16 | shamelessn00b | man tc |
| 06:32.30 | shamelessn00b | DROPPING |
| 06:32.31 | shamelessn00b | <PROTECTED> |
| 06:32.44 | Bloudermilk | Rad! I've been hacking around with that myself, but using a ruby server implementation |
| 06:33.19 | drclue | Of course because it uses HTML5 , I've also been developing a CEF tool that makes the HTML5 applications run as stand alone desktop applications in Windows and Linux without a need for a browser |
| 06:33.40 | Bloudermilk | CEF? |
| 06:33.51 | Bloudermilk | "Child Evangelism Fellowship"? |
| 06:33.55 | Bloudermilk | Haha... first result on google |
| 06:34.40 | drclue | CEF (Chrome Embedding Framework) = Stick Google Chrome browser engine in desktop application |
| 06:35.06 | Bloudermilk | Got it. Sounds like fun! |
| 06:35.53 | shamelessn00b | there is this other cool tool called trickle |
| 06:36.01 | shamelessn00b | but it only works in the user space |
| 06:37.29 | drclue | Right now , one can take an ordinary zip file representing the browser application's file tree , apply AES128 encryption and ship out ones web application as a stand alone executable. We have some minor little tweaks left to do , but the CEF thing (called WBEA) is aimed squarely at leveraging the AsterClick tool. |
| 06:38.44 | drclue | There is another fellow working on a droid wrapper and another fellow working on the iPhone thing, but the latter guys computer keeps rebootign when he grasps his keyboard in a comfortable way :) |
| 06:38.51 | Alton35 | That's good. Google knows how to do it more or less. |
| 06:40.01 | drclue | I like both Google and Firefox, but Firefox has been lagging behind a bit. Here is hoping FF4 pulls the fat out of the fire |
| 06:40.59 | drclue | So far my AsterClick / HTML5 applications run faster than any desktop applications I've seen doing the same taks |
| 06:42.35 | drclue | If one pressed for example the hold button on a hard phone , and watched the web display, it's impossible to tell the difference in time between the red led on the phone coming on and the reflection of same in the browser |
| 06:42.47 | ankur_6997 | now i have connected my phone with trixbox using bluetooth |
| 06:42.47 | ankur_6997 | i have added inbound routes using trixbox freepbx pannel and |
| 06:42.47 | ankur_6997 | when i call on my number it says the number you dialed is not in service |
| 06:42.47 | ankur_6997 | what should i do |
| 06:42.49 | ankur_6997 | ? |
| 06:44.10 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 06:44.13 | drclue | [has sudden Dick Van Dyke flashback] man is instructed to place phone in paper sack , walk out in the front yard and swing sack in circles overhead while screaming like a chicken. Not sure that advice would work in this situation though |
| 06:46.50 | ChannelZ | And once again this is not a FreePBX support channel |
| 06:46.52 | drclue | @Bloudermilk imagine this addEventListener("hangup", yourFunc, true) now imagine that works for every single Asterisk event. |
| 06:47.15 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 06:48.30 | Bloudermilk | @drclue brilliant |
| 06:48.33 | drclue | The biggest trick right now is working out the appropriate security strategies for exposed web applications |
| 06:49.37 | Bloudermilk | Yeah, definitely gotta lock that down |
| 06:49.49 | Bloudermilk | What ideas are you guys playing with at the moment? |
| 06:51.21 | drclue | Well , part of it is handled in the AsterClick.conf where one can specify users in much the same way one would in manager.conf , but on top of that is a location sensitive API key like Google API uses. |
| 06:53.09 | drclue | The hard part in this security design is going to be at the same time making it painless for the developer and avoiding various injection style attacks |
| 06:54.18 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 06:54.28 | Bloudermilk | injection style as in similar to mysql injection? |
| 06:54.28 | drclue | Even from the early screen shots of the AsterCLick HUD test application , you can see there is a lot of power to protect http://asterclick.drclue.net/snapshots/AsterClickHUD082510.png |
| 06:54.53 | drclue | Yes , exactly like what one might see in MySql |
| 06:55.07 | Bloudermilk | Can we not assume that one with a connection is "trusted"? |
| 06:55.35 | drclue | No , one cannot assume that one with a connection is trusted |
| 06:56.59 | Bloudermilk | Wouldn't the connection credentials only be accessible to trusted people? |
| 06:58.15 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:59.16 | drclue | First off lets consider that if you have unrestricted access to AsterClick , you can do anything you could do in the asterisk CLI |
| 07:02.05 | drclue | So of course I use user names , passwords , location based keys and the like to constrain the functionality output to a particular context (web page) but still , every possible command not excluded by those restrictions has parameters that those in a web browser are free to play with |
| 07:02.05 | *** join/#asterisk af_ (~getsmart@78.134.21.90) |
| 07:02.24 | drclue | This is where the injection risk is the greatest |
| 07:02.31 | Bloudermilk | I see |
| 07:02.51 | Bloudermilk | In any event I suppose it's always better to be more secure than less :) |
| 07:03.17 | akoma1s | hello again, I've enabled allowguest=yes in sip.conf, and when I try to call the asterisk box, if the username matches a SIP peer in asterisk, it gives me this: "Found peer 'antonis' for 'antonis' from 12.34.56.78:12345" |
| 07:03.33 | drclue | The trick will be in locking down those parameters in a way that is painless to the developer and discouraging to the script kiddie |
| 07:04.03 | akoma1s | how can I make it only match a sip peer if it has authenticated? |
| 07:05.02 | drclue | @akimal1s I send all my hackers to listen to some old hippie hold music forever |
| 07:05.29 | akoma1s | drclue: hahah |
| 07:05.33 | *** join/#asterisk tiny_D (~tiny_D@mail.snom.com) |
| 07:05.51 | akoma1s | well, in my case I *do* want to answer calls on a specific extension |
| 07:07.42 | drclue | @akoma1s I would suggest finding that old o'reilly asterisk PDF manual and curling up with it for a while |
| 07:07.54 | kaldemar | akoma1s: you can't. matching happens before authentication. unless you restrict your defined peers by their address. |
| 07:08.13 | ankur_6997 | my ivrs system work fi9 from a softphone but when calling from real phone it dos't recogonise the keys and just keep wating for user input |
| 07:08.21 | ankur_6997 | what should i do |
| 07:08.33 | ChannelZ | cry |
| 07:08.46 | ChannelZ | or hire an illegal to answer your phones |
| 07:09.03 | akoma1s | kaldemar: so, basically, if someone on the internet wants to call someone on my asterisk box, AND happens to have the same username, they simply can't? |
| 07:09.10 | ChannelZ | 'from a real phone'... via what? An ITSP? Analog DAHDI card? |
| 07:09.23 | *** join/#asterisk schmidts (d5ebd4c1@gateway/web/freenode/ip.213.235.212.193) |
| 07:09.26 | schmidts | morning |
| 07:09.48 | kaldemar | akoma1s: exactly. |
| 07:09.48 | drclue | in band , out of band RFC this thing or that, touch tones what a pain in the back. Get that o'reilly PDF manual and have a good read |
| 07:10.13 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 07:11.09 | drclue | Thats the hole thing about asterisk , there are so many answers that none of us can remember them. It's better to learn how to fish than to ask us if we remember where we put the fish |
| 07:12.58 | drclue | Really , while perhaps a little out of date this little file ought to help sort out most things http://asterclick.drclue.net/Assets/Docs/9780596510480.pdf |
| 07:13.17 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:15.17 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 07:16.03 | drclue | I would have to say that this single file has helped me more than every IRC conversation and every web page about Asterisk has added together |
| 07:16.06 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 07:16.10 | akoma1s | drclue: thank you, hopefully it's more helpful than the dozens of postings I found online so far |
| 07:17.22 | drclue | @akoma1s The online life of Asterisk documentation is schizophrenic at best. |
| 07:18.01 | akoma1s | indeed so |
| 07:21.42 | drclue | @akoma1s As you read through that file , it will take several attempts, as moments of brilliant insight will be washed away in a moment by the girth of the information like a see born wave attacks ones sand castle. None the less a few times through and the fog is guaranteed to lift , if but a bit at a time |
| 07:22.09 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 07:22.10 | *** join/#asterisk qvsqvs (~anonymous@mail.logical.co.za) |
| 07:27.30 | drclue | The really fun part for me is that Asterisk was written by a lot of different at differing times and with differing sets of development logic. This becomes really interesting when designing a tool that aims to make Asterisk commands and events appear to be a sane and consistent API, which is what I've been coding on day in and day out for almost a year |
| 07:30.09 | russ | drclue, I think part of it is that the developers know that removing old interfaces and configuration methods hurt users, so they leave everything in |
| 07:30.30 | russ | confuses new users, because there are several slightly different ways to do one thing |
| 07:31.35 | ChannelZ | damned if you do, damned if you don't. |
| 07:32.31 | ChannelZ | "Why do I need DAHDI for MeetMe, I can't get this shit to work!" "Use ConfBridge instead." "Why doesn't ConfBridge do *everything* MeetMe does!?" |
| 07:33.35 | ChannelZ | Most of the entire SIP protocol needs to be tossed out on its ass too, but people would get bitchy. |
| 07:37.04 | akoma1s | I can soon see Asterisk being relegated to the role of "DAHDI gateway" in my case |
| 07:38.35 | *** join/#asterisk goddva (~glarsen@77.40.154.242) |
| 07:38.51 | drclue | @russ I do understand the school of thought that would perhaps never want to retire an interface method. There are however many many many things that could be done with the Asterisk API that would break nothing and fix many things |
| 07:38.57 | akoma1s | MoH accepting only 8KHz input, channels created by Queue() not being able to access the original channel's variables (bit me hard in my last AGI script), and now random internet usernames clashing with local peers |
| 07:41.13 | ChannelZ | illegals work cheap but the only thing you'll get for MoH is "Feliz Navidad" over and over |
| 07:41.42 | akoma1s | lol |
| 07:43.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 07:43.38 | drclue | I for one have remapped all none event AMI utterances to appear as events anyways. I've taken every list producing source I come across and make sure it has a beginlist,listentry and listcomplete logic. All long returns are broken up into record orientated events. I have convinced the vast majority of CLI text blobs to render as structured data using rules that detect structure in freeform text and do not look for specific ques |
| 07:44.24 | *** join/#asterisk ahowlader (howlader@180.234.43.10) |
| 07:48.32 | drclue | I play most of my CD collection on MOH. Currently I'm making a little tool in my *spare* time to jack any audio source into MOH on the fly |
| 07:49.36 | drclue | I guess tool maker is really my major function in life |
| 07:51.20 | drclue | So if you could magically whip up any web application that exploited your Asterisk box , what would it be? |
| 07:52.24 | drclue | What skill sets do the average asterisk users have? |
| 07:52.56 | russ | I dunno, too many people build castles |
| 07:53.12 | russ | it ends up being difficult to re-use stuff |
| 07:53.25 | ChannelZ | I want it to play my Twitter feed in DTMF |
| 07:53.32 | russ | like freepbx |
| 07:54.45 | drclue | @russ Well, I build tools , not castles. I having worked with the FreePBX source code could not claim to be a big fan. I also don't like a tool that makes you choose training wheels that can't come off. |
| 07:55.28 | russ | drclue, I looked at their caller id superfecta stuff because I thought It'd be cool to integrate some of them |
| 07:55.37 | russ | but you have to pull in so much of their stack to get a small bit working |
| 07:56.25 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 07:56.48 | drclue | @russ If someone else was taking a serious effort at making tools and not castles , I would have not have written AsterClick |
| 08:00.55 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 08:01.06 | drclue | @russ The biggest deal with AsterClick is getting your PHP configured and the AsterClick server running. Once you have that , evey piece of Asterisk documentation out there is AsterClick documentation |
| 08:02.07 | russ | asterclick looks like it'd be fun to use |
| 08:03.03 | drclue | Once you have issued even a single AsterClick command , you could issue pretty much any possible AMI command without EVER seeing another piece of AsterClick Docs |
| 08:06.11 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:06.19 | drclue | Matter of fact after you issued your first AsterClick command , the Asterisk CLI itself would give you enough info to issue any other command |
| 08:08.15 | drclue | All results returned to AsterClick clients are nothing more in most cases than AMI results translated from their native HTTPish headers to XML. |
| 08:09.17 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 08:10.32 | drclue | The exceptions to the rule are really just limited to really bulky data outputs from Asterisk and unstructured free form text blobs like CLI commands. In those cases AsterClick applies structure and breaks the long results into XML event packets |
| 08:12.48 | drclue | Even when AsterClick applies structure to free form data , it is done in an automatic way based upon the data itself, so soon looking at the output of a CLI command , you'll be able to know what the XML will look like as the parsing is driven by the data itself and not any asterisk version specific rules |
| 08:13.29 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
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| 08:18.54 | shamelessn00b | so anyone who can help me on monitoring SIP trunk QoS with asterisk |
| 08:19.08 | shamelessn00b | and not that SIP show channelstats command |
| 08:19.17 | shamelessn00b | its broken |
| 08:19.50 | drclue | The biggest hurdles to most folks in using AsterClick involve System V messaging and shared memory , which are features built into PHP but require specific compile time flags to make available. I do have plans to make pseudo functions for those who can't figure out the three commands to turn them on in their installation, but right now refining the security is the biggest thing on my list |
| 08:22.33 | drclue | One thought I have had regarding the System V messaging and shared memory is to distribute AsterClick as a compiled executable instead of the PHP source , but we will have to see if the sponsrs are willing to pay for that effort |
| 08:24.13 | drclue | Making open source projects and paying ones rent in so doing is a tricky proposition |
| 08:29.05 | drclue | @shamelessn00b I suppose the first thing is to do the asterisk -rvvv thing and simply watch the show |
| 08:29.30 | *** join/#asterisk ruyo (~psantos@a81-84-220-57.cpe.netcabo.pt) |
| 08:30.48 | shamelessn00b | drclue: My SIP trunk is working fine, I just want to be able to tell if the network chokes or something happens to the trunk that the audio quality starts degrading, I should be able to raise an alarm |
| 08:32.49 | drclue | Well , unless your asterisk server is melting , it's mostly a matter of connectivity. |
| 08:34.29 | shamelessn00b | I know, its to do with connectivity |
| 08:34.47 | shamelessn00b | but I want to be able to tell, if the audio quality goes down and calls start to break |
| 08:35.14 | shamelessn00b | sip show channelstats shows you jitter and packet loss on active SIP calls |
| 08:35.31 | shamelessn00b | on your asterisk box, but the command isn't working as expected for me |
| 08:35.54 | drclue | I have multiple SIP born trunks and reliably report and react to their availability |
| 08:36.13 | shamelessn00b | availability is one thing, I'm talking about quality |
| 08:36.35 | shamelessn00b | like if your network is choked you make a call, you will experience stutter, garbled audio etc |
| 08:36.36 | drclue | Well , for me availability also includes useability |
| 08:36.52 | shamelessn00b | what do you use for that |
| 08:40.36 | drclue | I would be hard pressed to remember what I solved that particular issue with as I buried it all in the AsterClick server so I would not have to think about it anymore |
| 08:41.15 | shamelessn00b | '-' |
| 08:41.17 | shamelessn00b | lolk |
| 08:44.34 | drclue | It's exactly these types of time swallowing issues that inspired me to write a tool to bury those all too ripe smells of day to day pain in a middleware |
| 08:45.33 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.143.25.mtnl.net.in) |
| 08:48.01 | drclue | Besides , if my SIP trunks go down , it's mostly because someone is in a hurry to see a new movie before it hits the theater and the trunk is timing out reaching the provider |
| 08:49.26 | *** part/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 08:50.19 | drclue | One or two other times it has been a kid from Poland or something who thinks if they pound on my server enough they will get a free phone call. (Zero freebies so far) |
| 08:54.54 | *** join/#asterisk hrhrhr (~c1@213.1.224.2) |
| 08:56.39 | shamelessn00b | lol |
| 08:57.24 | shamelessn00b | btw whats your verdict on SIP v IAX |
| 08:57.37 | shamelessn00b | number of concurrent calls / trunk |
| 08:58.05 | *** join/#asterisk deonv (~adium@41.218.75.152) |
| 08:58.07 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 08:58.12 | drclue | Since I never seem to have a problem either way , I guess no verdict |
| 08:58.38 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 08:59.36 | shamelessn00b | IAX saves bandwidth |
| 08:59.43 | shamelessn00b | in messages etc |
| 09:01.05 | drclue | SIP providers are a dime a dozen and there is a lot of variability. They are really cheap and often FREE, so having a lot of them and coding good failovers should make such things invisible. |
| 09:01.32 | *** join/#asterisk F|shie (~chatzilla@58-65-172-114.nayatel.pk) |
| 09:02.07 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:02.48 | shamelessn00b | I'm not going to be doing this cuz I want to interconnect with SIP providers, I want to set up my own media gateway, isolate the IVR services from each other, so call from PSTN lands on 1 or 2 * boxes on E1 cards, and then they initiate a SIP call and forward it to other * boxes depending on what service was requested |
| 09:03.00 | shamelessn00b | damn, why did it bold the text T_T |
| 09:03.46 | *** join/#asterisk JCATS (~JCATS@80.78.74.210) |
| 09:05.15 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 09:06.09 | drclue | What do you mean by "media gateway" ? What is the experience to be had by the users? |
| 09:10.03 | *** join/#asterisk wimt (wimt@freenode/staff/wikipedia.wimt) |
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| 09:11.37 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 09:11.47 | drclue | Looks like after a month it is time for a re-boot folks, as the cursor has vanished. Back in a bit |
| 09:11.57 | *** part/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 09:21.56 | *** join/#asterisk af_ (~getsmart@78.134.21.90) |
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| 09:32.37 | drclue | Perhaps it all after a reboot lives mightily once more !! |
| 09:36.59 | drclue | How could I ever really complain? My Asterisk server, all it's phone lines,complete LAMP stack ,multi media center and everything running on a Linux PC the size of a packet of cigarettes, and I have to reboot once a month. Now really could someone please help me complain about that ?:) |
| 09:42.05 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.133.74.mtnl.net.in) |
| 09:43.10 | shamelessn00b | LOL |
| 09:45.50 | ankur_6997 | userinput is not recogonised by asterisk when calling from pstn line i have connected my * system with gsm mobile using bluetooth |
| 09:47.01 | *** join/#asterisk oelewapperke (wapper@85-158-215-1.powered-by.benesol.be) |
| 09:47.27 | oelewapperke | is there any way to have a AGI like control interface for SIP SUBSCRIBE type connections ? |
| 09:47.55 | oelewapperke | (basically a way to have a set of things to subscribe against, which I can programmatically change state for) |
| 09:48.01 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 09:48.47 | drclue | @oelewapperke Once you have achieved an AGI setting you can with the right effort do anything you dream up, but how you approach that dream determines your level of pain |
| 09:50.08 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 09:50.36 | drclue | Here is an open source link to some of my FastAGI code in this instance for PHP http://code.google.com/p/fastagi-php-drclue/ |
| 09:51.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:53.47 | oelewapperke | drclue: I know about AGI, but I haven't seen many commands having to do with sip subscribe |
| 09:55.56 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:56.05 | drclue | @oelewapperke Let's walk this dog back a few steps and get a better view. What do you expect the user to do , and what do you expect the basic outcome to be. |
| 09:57.48 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:58.40 | oelewapperke | I expect a phone to SUBSCRIBE to a number of addresses, then control from my application when the lights on the phone light up |
| 10:00.49 | oelewapperke | so that I can have arbitrary things for the user to subscribe to |
| 10:02.48 | drclue | Attempting to digest .... stand by .... |
| 10:03.40 | drclue | I assume we can say that the application in this instance is in the AGI context? |
| 10:04.30 | drclue | A phone subscribing to addresses could mean too many things , could you expound on that? |
| 10:05.09 | drclue | The lights on the phone , could you tell me a little more about what you mean there? |
| 10:06.22 | Chainsaw | drclue: It sounds like BLF control. |
| 10:06.27 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 10:06.35 | Chainsaw | drclue: I've implemented that with custom device states. |
| 10:07.36 | drclue | SNOM Phones ... digesting |
| 10:09.57 | drclue | Fairly device centric expression fainting to be http headers? |
| 10:12.02 | drclue | @chainsaw does this errr protocol have any traction, are their competing standards and what there is of these pseudo standards do you think they could be combined in perhaps XML or another structured medium? |
| 10:12.05 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 10:12.35 | Chainsaw | drclue: I've only done this on Polycom phones, which use SIP "buddies". |
| 10:12.55 | Chainsaw | drclue: Oddly it seems to require SIP over UDP to work (i.e. if I force full TCP, I lose that functionality entirely). |
| 10:15.34 | drclue | @chainsaw I could understand that. For me it really is a question of if the standard itself is worth perhaps spending some time on or more to the point if there are several and I could mitigate their differences or if there were one I could follow. I'm nearing the end of the basic development cycle for AsterClick and I'm always looking for the next thing to do |
| 10:16.42 | Chainsaw | drclue: More interested in having on-the-server DND working with Asterisk on a Polycom phone to be honest. |
| 10:17.29 | drclue | Well does DND work for other devices you have? |
| 10:19.03 | Chainsaw | drclue: It's a specific protocol that Asterisk does not currently implement. |
| 10:19.09 | drclue | I handle most of the basic phone functionalities in software , simply to ignore the phones at present. |
| 10:19.23 | Chainsaw | drclue: The other devices I have are unsophisticated, horrible Cisco 7960s. I'm amazed I manage to complete on calls on them sometimes. |
| 10:20.38 | drclue | Be it forwarding , following , DND or other common behaviors I implement them in my applications as I as of yet have not seen any solid footing in the marketplace |
| 10:21.18 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 10:23.30 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 10:23.57 | drclue | A little at a time as things will and have become more solid in this place or that, some quiet little class will bloom , settling ever further done the coding tree as habitat will promote. |
| 10:25.30 | drclue | [Ya , whatever Governor , can ya sport a ring?] |
| 10:31.39 | upb | anyone here using voiceglue in production? |
| 10:35.07 | drclue | @chainsaw My basic goal is make asterisk accessible and usable by the average coding punter with but a web browser , a text editor and the desire to do something they think is cool. |
| 10:36.14 | *** part/#asterisk ankur_6997 (~Dev_1@122.177.234.204) |
| 10:41.27 | joobie | drclue, sounds like you want #freepbx |
| 10:41.47 | joobie | the cli ownz .. there are already gay restrictive interfaces to asterisk |
| 10:44.46 | drclue | @joobie I've had to work with the source code of freePBX and it is an experiance I would much rather not have burned into my core |
| 10:45.23 | joobie | there are a few projects already with a gui front end to asterisk |
| 10:45.38 | joobie | save yourself some time and go join one of those |
| 10:45.43 | joobie | they are fairly developed |
| 10:46.03 | joobie | but i dont see the thing behind skinning the CLI when you can just learn the cli |
| 10:48.44 | drclue | Yes there are , but I've gone strictly HTML5. Not only is this where all browsers are going including them folks where it nothing but rains , but I've also designed a tool that allows that same HTML5 JavaScript code to run as a stand alone executable desktop application on both windows and Linux |
| 10:50.26 | drclue | There are already plans to hit the Droid and iWhatever realities |
| 10:50.54 | joobie | is HTML5 ratified? |
| 10:51.11 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.95.159) |
| 10:51.14 | FILLVAIO3 | Hi guys! |
| 10:52.06 | FILLVAIO3 | Does anybody know why this not working in asterisk 1.6? exten => s,n,Set(mcV0=sip:18005558355@tf.voipmich.com) | exten => s,n,Set(mcV1=$["${mcV0}" : "([a-zA-Z0-9]\+)\:"]) |
| 10:53.24 | drclue | @Joobie Who cares if it is ratified? I've been developing for the Internet since 1994 and so I make my own choices. I've examined both current and near term offerings from the various vendors and I'm covered. Even if they all lied to me , it matters not as I can deploy my HTML5 applications to all them platforms without waiting and it will work |
| 10:54.35 | FILLVAIO3 | How to use regular expression in asterisk? |
| 10:54.38 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 10:57.33 | joobie | drclue, about 5 years ago i was heavily into CSS |
| 10:57.47 | joobie | they had a new version in draft that everyone was talking about and had some neat features |
| 10:58.09 | joobie | it's still not formal nor is its browser support |
| 10:58.14 | ruyo | voip-info.org has discounts on their main page. :o |
| 11:02.19 | drclue | @joobie It is those things that vendors collectively adopt of the w3.org that we go by. It is the nature of the beast and that is how we code, by what they share in standards support, what we can normalize through coding practices, and lastly what we ignore for the former approaches might not be practical. This is how it was , how it is and how it will be. |
| 11:03.10 | *** join/#asterisk bodom (~quassel@2001:470:1f0b:ac0::2) |
| 11:03.38 | bodom | Hi there. May someone tell me what's wrong with this command? I'm trying to download asterisk-gui, but svn hangs: svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui |
| 11:04.25 | joobie | fuk man |
| 11:04.29 | joobie | when u type |
| 11:04.35 | joobie | i have to really think to understand what ur saying |
| 11:05.17 | joobie | bodom, maybe ur firewall is blocking it |
| 11:05.23 | drclue | No bazooka bubblegum answers here :) |
| 11:05.23 | joobie | tcpdump |
| 11:05.35 | joobie | i like bazooka bubblegum |
| 11:05.39 | joobie | less thought involved :P |
| 11:05.57 | bodom | joobie: lynks can access the URL |
| 11:06.01 | drclue | Bazooka bubblegum , good to eat not so to think |
| 11:07.56 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.133.74.mtnl.net.in) |
| 11:11.10 | joobie | what about tcpdump |
| 11:11.13 | joobie | and svn |
| 11:17.28 | bodom | joobie: svn shows command running for a while, then hanging |
| 11:20.04 | drclue | Was it a grand hanging or more of pedestrian fare? |
| 11:21.42 | *** part/#asterisk bodom (~quassel@2001:470:1f0b:ac0::2) |
| 11:22.22 | joobie | have a look at neon-debug-mask |
| 11:22.31 | joobie | oh he's gone.. meh |
| 11:22.46 | joobie | drclue, so what are the better features of HTML5? |
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| 11:27.34 | zamba | i've read quite a lot of news about asterisk servers getting hacked.. what do i need to do to stay safe? |
| 11:27.51 | zamba | SIP hacks, that is |
| 11:28.02 | zamba | fw the SIP port? or is it enough just to keep asterisk up to date? |
| 11:28.08 | kaldemar | zamba: start with this: http://blogs.digium.com/2009/03/28/sip-security/ |
| 11:28.12 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 11:33.07 | drclue | @joobie NO POLLING . All previous http methods have involved repeated requests to see if an event had occurred , like kids on a long road trip "Are we there yet?....Are we There yet?". At each ask , a new connection had to be made. With WebSockets the connection is both bi-directional and live!!! |
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| 11:35.09 | drclue | If your user presses the [hold] button on their hard phone , within "milliseconds" it can be reflected in every HTML5 application at that location or around the world, ate speeds near that on the hold lamp of the phone itself |
| 11:36.02 | drclue | You just cannot do that with pre HTML5 WebSockets browsers |
| 11:37.04 | zamba | how can i get asterisk to log to syslog? |
| 11:41.48 | drclue | Why do you want the material prmoted to syslog? |
| 11:42.02 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 11:46.04 | zamba | drclue: to include it into the security scheme of the server |
| 11:46.19 | zamba | kaldemar: but the basic method of fraud is trying to authenticate as a user? |
| 11:47.27 | kaldemar | zamba: that's probably the primary one. of course there are also known vulnerabilities in older versions. |
| 11:48.19 | drclue | I would at least myself not promote the log ,but rather deal with the issue at a local level |
| 11:49.12 | drclue | The more levels , the more options , the more you don't know to protect |
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| 12:08.12 | zamba | drclue: syslog-ng is the solution |
| 12:15.51 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:16.49 | drclue | Yes syslog-ng is a better tool in what it does. In the end of any tracing process I will probably end up monitoring items at a level benieth syslog-ng, so in most cases I would be better served to start there |
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| 13:21.15 | dbowlby | hello all |
| 13:21.31 | *** join/#asterisk Comete (~comete@91.88.228.81) |
| 13:21.37 | Comete | hi |
| 13:22.01 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 13:24.25 | Comete | if i have 2 or more sip providers, is it possible to put a priority on them for outgoing calls. For example, i want to use provider1 to call if no line is busy but if it is busy i want to use provider2 ? |
| 13:26.43 | Comete | if it is possible, could you show me an example ? thanks a lot |
| 13:27.06 | fauxalliance | can asterisk? yes, asterisk can... let me try to find a doc for you |
| 13:27.47 | fauxalliance | Comete, are you using any type of frontend or framwork? |
| 13:27.54 | fauxalliance | s/fram/frame |
| 13:28.39 | fauxalliance | ... |
| 13:28.47 | Comete | fauxalliance: no just asterisk |
| 13:32.12 | ruyo | Comete, have you read The Book? |
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| 13:33.50 | fauxalliance | http://www.voipuser.org/forum_topic_4413.html @ Comete |
| 13:33.58 | fauxalliance | 1book |
| 13:34.00 | fauxalliance | ~book |
| 13:34.00 | 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 |
| 13:34.21 | pabelanger | Comete: *CLI> core show function DIALGROUP |
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| 13:38.52 | Comete | thanks guys :) |
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| 13:41.21 | Katty | GOOD MORNING BEAUTIFULS |
| 13:41.47 | Comete | by the way is Asterisk 1.6.x a stable release ? i mean can be used in production ? |
| 13:42.01 | Katty | Comete: i use 1.6 in production perfectly fine. |
| 13:43.13 | Polysics | use 1.6.2, ti has more features |
| 13:44.13 | frigidzephyr | Comete: http://www.asterisk.org/asterisk-versions |
| 13:44.36 | drmessano | 1.6.x has only been out for 2 years, I would hope so |
| 13:44.46 | Katty | Qwell: so how's that nuclear fusion coming along. |
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| 13:46.24 | Zeeek | Start bot { Shill for VUC, Friday at 12 Noon EDT, http://vuc.me for connect info. Astricon DISCOUNT CODE when you listen} |
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| 14:01.42 | Katty | dear fox news. please shoot yourself. love, Katty. |
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| 14:05.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 14:08.58 | Polysics | hello everyone! it's me again, the "you are doing it wrong" guy :-) |
| 14:09.23 | Polysics | i have user A in a conference, alone. can i bridge him to user B that is in another call? |
| 14:09.28 | Polysics | taking A out of the conference |
| 14:09.51 | Polysics | second, sort of related, question is: can i turn a two person call in a conference? |
| 14:15.25 | drmessano | Yes |
| 14:17.10 | [TK]D-Fender | Polysics: Part 2 : AMI Redirect |
| 14:18.08 | [TK]D-Fender | Polysics: Part 1 : Hmmm... I think you could redirect him to an exten that calls Bridge() and that MIGHT cause an effective 3-way, but not one the PHONE can control. |
| 14:18.46 | [TK]D-Fender | Polysics: OR, you could just tos his call towards the phone and let them ANSWER to bridge them in on the phone, but that requires compliance on the user side. it isn't "Do-it" Grade |
| 14:19.25 | Polysics | [TK]D-Fender, the called party is already in the system, he has just pressed 1 t oaccept the calle |
| 14:19.27 | Polysics | ?call |
| 14:19.29 | Polysics | *call |
| 14:19.42 | Polysics | then again, Redirect looks like it might be useful for this |
| 14:19.58 | [TK]D-Fender | Polysics: "on the system"? PARDON? |
| 14:20.32 | Polysics | that's a sentence the wiki uses too, i mean, he is already connected to a call on the * server |
| 14:21.25 | Polysics | then again, i think i could try with Redirect, all i need is an extension that will do the bridging |
| 14:21.39 | [TK]D-Fender | Polysics: Which is what I just said for the 1st idea |
| 14:21.53 | Polysics | but won't that do the same thing as issuing a Bridge command directly from AMI? |
| 14:22.06 | Polysics | because the above right now doesn't work for me |
| 14:22.38 | Polysics | i mean, user A, which is in a ConfBridge as a means to keep him waiting, isn't taken out of the conference and into a call with B |
| 14:25.14 | *** join/#asterisk _structz (~structz@201-42-157-184.dsl.telesp.net.br) |
| 14:25.53 | Polysics | am i saying something that makes marginal sense? :-) |
| 14:25.58 | _structz | Hello all, having a problem with my asterisk version 1.6.2.11 everytime i get [Aug 27 11:19:28] WARNING[24664] pbx.c: Don't know what to do with 'SIP/6463-0000000 my asterisk crashes does anyone know how to prevent this? |
| 14:26.41 | [TK]D-Fender | [10:25]<Polysics>am i saying something that makes marginal sense? :-) <- sort of. ISH. maybe. |
| 14:26.48 | [TK]D-Fender | Polysics: SHOW US the faiure |
| 14:27.06 | Polysics | will try, although i do not see anything strange, tbh |
| 14:27.12 | Polysics | there are no errors |
| 14:27.29 | Polysics | let me double-check the script and re-test it |
| 14:30.51 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:32.57 | Marini | hi, can anybody tell me how to catch live on extensions.conf the number of actual calls on a trunk having trunk configured with call-limit? |
| 14:33.17 | *** join/#asterisk _structz (~structz@200-207-170-168.dial-up.telesp.net.br) |
| 14:33.21 | _structz | Hello all, having a problem with my asterisk version 1.6.2.11 everytime i get [Aug 27 11:19:28] WARNING[24664] pbx.c: Don't know what to do with 'SIP/6463-0000000 my asterisk crashes does anyone know how to prevent this? |
| 14:35.46 | Gugge | Marini: i would use GROUP and GROUP_COUNT |
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| 14:40.14 | Marini | Gugge: can you tell me an example or reference to look up use and functions for groups in asterisk, i want to be able to have the info of concurrent calls before i make a dial out |
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| 14:45.42 | [TK]D-Fender | Marini: "core show functions like GROUP" <- |
| 14:46.15 | Marini | thanks |
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| 14:58.25 | gandhijee | hey, if i want to use chan_mobile the old docs say i need to put some sd hcid.conf file, but the new bluez doesn't seem to have it |
| 14:58.36 | gandhijee | how do you set up the connection? |
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| 14:59.02 | Letoric | Can anybody advise me how to tell what timing source asterisk is using? I thought after 1.6.1 you didn't need to provide one, but I'm still getting distorted sound on dial by name directory |
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| 14:59.32 | _structz | Hello all, having a problem with my asterisk version 1.6.2.11 everytime i get [Aug 27 11:19:28] WARNING[24664] pbx.c: Don't know what to do with 'SIP/6463-0000000 my asterisk crashes does anyone know how to prevent this? |
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| 15:01.25 | [TK]D-Fender | Letoric: Playback of that isn't impated by a DAHDI timer |
| 15:01.33 | [TK]D-Fender | lerDescribe your server environment |
| 15:01.38 | [TK]D-Fender | Letoric: Describe your server environment |
| 15:02.07 | Letoric | D-Fender - it's a Sun X2100 M2 with 4GB RAM |
| 15:02.16 | *** join/#asterisk jo889 (d04149c9@gateway/web/freenode/ip.208.65.73.201) |
| 15:02.28 | Letoric | running asterisk on CentOS 5.5 I believe, but lemme double check |
| 15:02.37 | Letoric | it's using pthread for timing, I did figure that much out |
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| 15:03.29 | jo889 | hi all. i am using fastagi to initiate a dial on a T1 against a channel group g1. if the user doesn't answer, a hangup event is generated, but I can't seem to figure out how to corrolate that the hangup is on the same channel as my origination. |
| 15:03.44 | jo889 | since i am dialing using a group g1, i don't know which channel was actually assigned. |
| 15:03.52 | _structz | ? |
| 15:04.04 | jo889 | so when the hangup event comes, it states the channel, but i cannot know if it's the same one from my origination. |
| 15:04.20 | jo889 | just curious if people have any ideas |
| 15:04.24 | jo889 | i am doing a synchronous dial. |
| 15:05.04 | Letoric | D-Fender: CentOS 5.5 |
| 15:05.27 | [TK]D-Fender | Letoric: that shouldn't impact anything. Where is your client? Are your virtualized? what kind of connectivity? |
| 15:05.54 | Letoric | client is a sip phone, same physical network, same vlan, |
| 15:06.01 | Letoric | 100mb ethernet |
| 15:06.10 | Letoric | it also happens when I dial in from my cell |
| 15:06.17 | Letoric | it's playing ulaw sounds |
| 15:06.23 | [TK]D-Fender | jo889: Originate = Asynchronous AFAIK |
| 15:06.25 | Letoric | it used to happen with gsm sounds as well |
| 15:07.04 | [TK]D-Fender | Letoric: Only thing I can think so far is that you have an optimization issue of some sort... |
| 15:07.15 | Letoric | TBH, I'm too new to optimize ;) |
| 15:07.18 | jo889 | [TK]D-Fender: you can async=False, such that you can diffrentiate onDialFinished vs onDialFailure |
| 15:07.26 | Letoric | I ran the installs pretty simplistically |
| 15:07.40 | jo889 | but the hangup event is async and I just want to be able to match it. |
| 15:07.45 | Letoric | I did add the extra sounds and remove dahdi, since it isn't supposed to be needed |
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| 15:10.49 | kaldemar | Letoric: try using timerfd instead of pthread |
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| 15:11.47 | Letoric | is that a recompile? |
| 15:11.52 | Letoric | or is there an option somewhere I can change |
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| 15:21.35 | path | hey guys :) |
| 15:22.02 | path | If Im not mistaken, http://pastebin.com/PejBURDr DID_span_1 would be my context for incoming calls isnt it? |
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| 15:25.45 | [TK]D-Fender | path: Yes |
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| 15:42.36 | path | thanks for the reply [TK]D-Fender :) |
| 15:43.31 | BarthezZ | hmm lol fun challenge... an Asterisk 1.6 server, a polycom kirk wireless server 600v3, and a profoon (cheap ass) handset... Connected to the wireless server (DECT) the ringtone when called is continuos, connected to the normal (analogue) dect base, the ringtone stops normally (couple of seconds of sounds, couple silence, etc) |
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| 15:45.21 | Naikrovek | BarthezZ: the analog dect base rings then stops then rings then stops? |
| 15:45.48 | BarthezZ | yes, but the phone ofcourse keeps seeing an incoming call |
| 15:45.55 | Naikrovek | okay |
| 15:45.56 | BarthezZ | but there must be something different in the signalling |
| 15:45.59 | Naikrovek | this is normal i think |
| 15:46.09 | BarthezZ | well, it's purely the sound the dect handset makes when being called |
| 15:46.10 | Naikrovek | how do really really old phones work. they ring and they stop and they ring and they stop |
| 15:46.20 | Naikrovek | yes |
| 15:46.24 | EmleyMoor | Naikrovek: Ring cadences |
| 15:46.32 | Naikrovek | analog ring is a 48V or better voltage sent over the line |
| 15:46.37 | Naikrovek | it's periodic |
| 15:46.37 | BarthezZ | connected to the wireless server it just keeps making noise, but normally it's sounds, and a little halt in between |
| 15:46.54 | Naikrovek | or maybe it's 96v over a normal 48v pair or something |
| 15:47.03 | EmleyMoor | 85V AC |
| 15:47.23 | Naikrovek | right but analog phone hardware does this - the ring voltages is on, then off, then on, then off, a few seconds at a time |
| 15:47.32 | Naikrovek | it makes sense that an analog dect handset would replicate this |
| 15:47.42 | Naikrovek | when paired with an analog dect base |
| 15:47.42 | BarthezZ | well, that's the analogue part yes, but it seems to be in the dect part |
| 15:47.45 | EmleyMoor | Naikrovek: Yes - the line card does it |
| 15:48.02 | Naikrovek | digital lines however, they "ring" until someone answers |
| 15:48.26 | Naikrovek | there's no voltage change, just an io signal that says "time to ring" then another when it stops |
| 15:48.32 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 15:48.35 | EmleyMoor | Naikrovek: That's an instruction to the phone that makes it play its ringtone |
| 15:48.40 | Naikrovek | yes |
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| 15:49.12 | EmleyMoor | I use old phones - including one nearly 40 years old |
| 15:49.24 | BarthezZ | but why would the ringing be different between two dect base stations :x |
| 15:49.37 | Naikrovek | because the signal to ring is different between them |
| 15:50.02 | BarthezZ | dect != dect? :p |
| 15:50.10 | Naikrovek | analog != digital |
| 15:50.24 | Naikrovek | it's the base that matters here, because it's the base that receives the ring signalling |
| 15:50.30 | BarthezZ | but dect = digital :p |
| 15:50.37 | Naikrovek | digital between the base and the handset |
| 15:50.40 | Naikrovek | not beyond that |
| 15:50.58 | BarthezZ | and btw... that profoon handset is the only one where i've encountered this problem |
| 15:51.14 | EmleyMoor | Whether the base does anything special based on the cadence of the ring signal is up to it |
| 15:52.29 | BarthezZ | weird... |
| 15:52.40 | BarthezZ | but I don't think i can tell asterisk to hold the ringing once every couple of seconds |
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| 16:08.30 | _structz | Hello all, having a problem with my asterisk version 1.6.2.11 everytime i get [Aug 27 11:19:28] WARNING[24664] pbx.c: Don't know what to do with 'SIP/6463-0000000 my asterisk crashes does anyone know how to prevent this? |
| 16:12.05 | p3nguin | need more info |
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| 16:15.42 | _structz | what kind?! |
| 16:15.42 | _structz | what kind? |
| 16:15.59 | _structz | can't get more, don't know how to produce de error. it just happens :( |
| 16:16.05 | *** join/#asterisk Janos (~asd@190.10.52.113) |
| 16:17.56 | Janos | hey there, got a question, i have a queue with the rrmemory strategy that works mostly fine, but the problem is that when two calls enter the queue almost at the same time, one person gets both calls and this is undesireable, the way i would like it would be ... |
| 16:19.06 | Janos | that if a person already have a call, not to give him more calls unless everyone else has at least one call active, what would be the best approach to achieve this ? |
| 16:20.41 | Janos | i know i could limit the number of calls a sip channel can have, but i don't really want to limit this, all i want is the queue system to be "smart" enough to not give more than one call to one person unless everyone else has one, again, is this possible ? |
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| 16:44.16 | ChannelZ | Turn off call waiting on the phone is the only trick I know |
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| 16:49.48 | Janos | this is a phone feature right ? |
| 16:52.22 | ChannelZ | Yeah |
| 16:53.14 | ChannelZ | Most SIP phones can handle at least 2 calls internally. If this is turned on the phone will accept the second (or third, or fourth..) call from Asterisk |
| 16:54.15 | Janos | ok so this feature is pretty much that same a imposing a limit of 1 call on the sip channel, right ? using the call-limit that is |
| 16:54.50 | nny | had some opinions given on the matter, but more appreciated. How many calls do you think a server could record with dual quad core xeon 3.8 with 4 Gb of ram? This is assuming no codec conversion (g729) and recording to local SAS disk. |
| 16:55.09 | nny | and using the m flag to have asterisk mix the channels together |
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| 16:56.33 | Janos | ChannelZ, do you think i should consider reporting a bug to the rrmemory strategy, i mean it does work they way it's expected, except when two calls enter the queue system simultaneously |
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| 16:57.07 | ChannelZ | Well yes and no I suppose... The queue sends the device a call; It accepts. Someone else comes into the queue; Asterisk tries to send another call to the device, and the device refuses. |
| 16:57.46 | ChannelZ | That won't limit any outgoing calls the device could make (say calling someone, putting them on hold, calling someone else..) |
| 16:57.55 | nny | rather replace mixmonitor/ monitor with m flag, seems that reduces cpu load as well for my theoretical setup |
| 16:58.45 | ChannelZ | nny: If you're mixing audio streams you are doing codec conversions |
| 16:58.53 | nny | ChannelZ: tru enough |
| 16:59.07 | nny | ChannelZ: I guess for the sake of replaying them later I need to |
| 16:59.11 | Janos | well according to the rrmemory docs, it should send the call in a round-robin way, always remembering the last channel it sent a call to so it doesn't send two calls in a row to the same channel, what do you think ? |
| 16:59.43 | nny | ChannelZ: current concurrent call load is ~ 75 concurrent channels |
| 16:59.52 | nny | wow that's a mouthful lol |
| 17:00.22 | ChannelZ | How many devices are agents of the queue? How are you testing? It might be round-robining but other devices are busy or refusing calls or not connected... |
| 17:02.17 | Janos | ChannelZ, i tested when there was noone around, and if i call, let that call enter the queue system, and then make another call, it goes to another channel as expected, but if i place the two calls at the same/similar time and they enter the queue at the same/similar time, then one channel gets the two calls even if all other channels are available, i'm using the latest 1.4 asterisk, just in case |
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| 17:02.48 | ChannelZ | I'm not sure at what point the queue "advances the pointer" to the next agent; if only after they answer, or as soon as it's ringing, etc.. maybe there is a race condition happening in your case, I don't know. |
| 17:03.38 | Janos | ok, i'll post on the list and see if they have more info on how rrmemory behaves, thanks a lot for the info |
| 17:05.02 | nny | ChannelZ: I see mentions of using a RAM disk to offload the I/O bottleneck if needed |
| 17:07.39 | ChannelZ | I doubt you'll run out of disk IO before CPU, the datarate is pretty low |
| 17:08.43 | nny | ChannelZ: interesting, just looking at what other people have done. I would say the CPU is more than capable at least |
| 17:09.16 | ChannelZ | I dunno, haven't worked with g729 much so I don't know what the hit is |
| 17:09.18 | *** join/#asterisk ManxPower (~manxpower@136.sub-69-96-255.myvzw.com) |
| 17:10.08 | ManxPower | Does anyone know what the difference between the Manager events Newstate and ExtensionStatus are? The docs are not very clear. |
| 17:11.23 | ChannelZ | 'core show translation' will give you an idea of how long it takes to transcode 1 second of audio to and from g729, not including mixing.. you could make a ballpark extrapolation there |
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| 18:00.12 | Marini | hi, what problem is: channel.c:2522 __ast_read: Dropping incompatible voice frame on SIP/out_xxxxxx-000883d2 of format ulaw since our native format has changed... |
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| 18:00.26 | Marini | does it cause any problem on live calls? |
| 18:10.11 | Marini | <PROTECTED> |
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| 18:23.25 | ManxPower | Does anyone know what the difference between the Manager events Newstate and ExtensionStatus are? The docs are not very clear. |
| 18:26.12 | russellb | off of the top of my head, Newstate == a state change for a channel, ExtensionStatus == a state change for an extension, which is mapped to one or more channels (or other types of state providers) |
| 18:29.15 | ManxPower | russellb, Is there quite a bit of overlap? I'm trying to keep track of the extension (ringing, busy, in use, etc) |
| 18:29.28 | ManxPower | the standard stuff devstate provides |
| 18:29.57 | russellb | yes, if a hint is mapped to a single device, they will be almost identical |
| 18:30.57 | russellb | it sounds like you want ExtensionStatus |
| 18:31.08 | russellb | as NewState is for a single channel .... |
| 18:31.31 | russellb | so, you have channel states. one or more channels go into calculating the state of a device. one or more devices go into calculating the state of an extension. |
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| 18:57.46 | ManxPower | russellb, thanks for the info. That really helps. |
| 18:57.56 | ManxPower | you're right, extenstatus is what I want |
| 18:58.04 | russellb | np |
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| 19:52.53 | megalomano | hi , I have some doubts about the channels E1 in SIP mode ,, well ,,, I have a DID service that offers E1 channels over SIP ( not R2 ) , i want to if its possible to have these E1 ( 32 channels ) over SIP in asterisk |
| 19:54.18 | iratik | Maybe someone here can help me. This is regarding inbound calls to the Queue application in asterisk 1.4 as they pertain to the cdr log. Is there a variable that is set for an incoming call that i can use to look up all entries in the CDR database for a sepcific incoming call?. I have a uniqueid of the incoming call before the call hits the queue() app. Once inside the queueapp.. the uniqueid changes . Orderlystats uses the uniqueid after |
| 19:54.18 | iratik | queue is called. The popup url i pass to queue i can pass the uniqueid but it doesn't match up to what i'm getting in the orderlystats call log. I need to find all the uniqueid's that pertain to a specific call. CallerID will not work because it doesn't work with blocked/unknown callerids . I need some sort of key that is in common between all events pertaining to an incoming call instance in the CDR .. or if none exist, then a channel va |
| 19:54.19 | iratik | riable i can propagate to the DB.. ideas? |
| 19:55.37 | megalomano | i.e,,,, I have aswitch that receive the same service , and is capable to take E1 in SIP with 32 channels to distribute it with a IVR |
| 19:56.09 | moy_ | megalomano: well, it's just SIP, and Asterisk supports SIP, so there is no reason for that to not work |
| 19:59.19 | [TK]D-Fender | [15:52]<megalomano>hi , I have some doubts about the channels E1 in SIP mode ,, well ,,, I have a DID service that offers E1 channels over SIP ( not R2 ) , i want to if its possible to have these E1 ( 32 channels ) over SIP in asterisk <- SIP is SIP. E1 does not exist as far as anything else is considered at that point |
| 20:00.00 | [TK]D-Fender | iratik: UNIQUEID |
| 20:00.29 | iratik | The uniqueid changes within the queue app once the call is bridged with an agent |
| 20:00.38 | knightfal | It shouldnt |
| 20:00.48 | iratik | hmm |
| 20:01.00 | knightfal | Let me look at my stuff real quick |
| 20:01.01 | [TK]D-Fender | iratik: Setas a separate channel var. |
| 20:03.11 | knightfal | ya it stays the same |
| 20:03.34 | knightfal | we set the ID as a VAR for call recording before the call goes to app_queue |
| 20:03.47 | iratik | I think i figured a solution out.. I overrode the callerid Set(CALLERID(name)=${UNIQUEID}) before Queue() gets called. Now asteriskcdrdb shows the same callerid column value for both legs of the call (the queue leg and the agent leg). Orlderstats.orderlystats_call also absorbed the modified callerid variable and it shoes in the orderlystats_db |
| 20:04.47 | knightfal | iratik: It seems strange your having to do that. |
| 20:05.22 | iratik | Could it have anything to do with using agents.conf for the queues? |
| 20:05.26 | [TK]D-Fender | Nope, I did that about 5 years ago |
| 20:06.44 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:07.44 | *** join/#asterisk kb3ien (~kb3ien@ool-44c720b8.dyn.optonline.net) |
| 20:07.51 | carrar | moo |
| 20:07.54 | carrar | Did I miss anything? |
| 20:08.07 | carrar | Anyone build a ATA out of a sound card yet? |
| 20:08.21 | [TK]D-Fender | ~savemoney |
| 20:08.21 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 20:08.24 | [TK]D-Fender | \o/ |
| 20:08.26 | carrar | heh |
| 20:08.47 | kb3ien | in hell here... just installed asterisk 1.4.35 and have no func_devstate HELP |
| 20:08.47 | carrar | and build it in 30 mins or less!! |
| 20:09.00 | carrar | That is HELL |
| 20:09.28 | kb3ien | there is not, but never was an option for not building in in make menuconfig. should there be? |
| 20:09.40 | carrar | I'm only good for smartass comments kb3ien, otherwise I'd help |
| 20:09.45 | Qwell | kb3ien: Not in 1.4, no. |
| 20:09.56 | kb3ien | it should ALWAYS be in the main trunk then? |
| 20:10.17 | Qwell | depends on your definition of always |
| 20:11.56 | kb3ien | exten => _1700555777X,n,Set(DEVSTATE(Custom:${EXTEN})=INUSE); |
| 20:11.56 | kb3ien | should work, then? |
| 20:12.14 | Qwell | kb3ien: Not in 1.4, no. |
| 20:12.34 | kb3ien | okay. i used to in 1.4.25 lets try to figure out what changed. |
| 20:12.35 | [TK]D-Fender | [16:08]<kb3ien>in hell here... just installed asterisk 1.4.35 and have no func_devstate HELP <-- not part of 1.4 |
| 20:12.44 | [TK]D-Fender | ~devstate |
| 20:12.44 | infobot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/community/russell/asterisk-1.4/func_devstate-1.4 , or http://www.asterisk.org/node/48325 |
| 20:12.49 | [TK]D-Fender | BACKPORT |
| 20:12.55 | [TK]D-Fender | have to add it yourself |
| 20:12.57 | kb3ien | okay, si i must have installed a patch, and not documented correctly. |
| 20:13.02 | carrar | heh |
| 20:13.17 | carrar | thats the best kind of installation |
| 20:13.17 | kb3ien | googledocs when through about 5 months on not letting me (and about 100 others) edit ... |
| 20:13.19 | carrar | docless |
| 20:13.54 | kb3ien | first link broken, can we try again? |
| 20:14.13 | kb3ien | anyone? |
| 20:14.18 | *** join/#asterisk tiny_D (~tiny_D@e178226159.adsl.alicedsl.de) |
| 20:15.11 | kb3ien | got it from google. |
| 20:15.19 | kb3ien | http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c |
| 20:15.26 | kb3ien | update the bot, plz |
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| 20:16.10 | *** part/#asterisk ManxPower (~manxpower@136.sub-69-96-255.myvzw.com) |
| 20:19.34 | kb3ien | still not there after make clean... tryinf a biuld |
| 20:19.44 | russellb | infobot: no, devstate is <reply> func_devstate is a module included with Asterisk 1.6 and above. A 1.4 backport is available here: http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c |
| 20:19.45 | infobot | russellb: okay |
| 20:20.06 | Qwell | that branch includes a Makefile, doesn't it? |
| 20:20.11 | russellb | it does, yes |
| 20:20.12 | russellb | maybe |
| 20:20.15 | russellb | actually, no |
| 20:20.22 | russellb | I think it just says to drop it into your source tree to build it |
| 20:20.31 | russellb | i obviously have no idea. |
| 20:20.48 | Qwell | oh, it's a README, yeah |
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| 20:35.10 | megalomano | [TK]D-Fender: ok ,understand , maybe the question is not concrete , the follow descrption is about the switch that receive calls from PSTN " All calls coming from the specified Span and Channels will be processed by the Call Filter. Span/Channel filter combination is the only allowed means to catch calls coming from the PSTN." ----> asterisk is capable to make this? |
| 20:39.45 | *** join/#asterisk MagnusThundercoc (~Magnus@18.62.30.145) |
| 20:40.09 | MagnusThundercoc | i am having trouble getting incoming calls with xlite and a sip provider called talktt |
| 20:40.18 | MagnusThundercoc | but skype incoming works just fine |
| 20:40.24 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 20:40.37 | MagnusThundercoc | how might i configure xlite so that it is able to take incoming calls like skype? |
| 20:41.13 | MagnusThundercoc | behind a big network, but since skype works i presume it is possible for other sip to |
| 20:44.11 | zamba | does it make sense to still use OpenSER? |
| 20:44.52 | *** join/#asterisk bombaclat (bomba@173.178.187.141) |
| 20:44.54 | drmessano-lt | OpenSER is a very silly, silly application |
| 20:45.01 | drmessano-lt | With lots of silly, silly users |
| 20:45.08 | zamba | drmessano-lt: i agree |
| 20:45.30 | zamba | drmessano-lt: that's basically why i'm asking :) |
| 20:45.34 | drmessano-lt | Oh,I thought you paid for an argument |
| 20:45.48 | zamba | so i guess it has never made any sense to use openser? |
| 20:46.01 | drmessano-lt | For me, no.. It's VERY silly |
| 20:46.19 | drmessano-lt | Asterisk = A++++ |
| 20:46.25 | zamba | apart from the fact that it's much easier to add sip users dynamically through a database backend.. and it also has more flexible routing.. |
| 20:46.27 | drmessano-lt | OpenSER = VERY silly |
| 20:46.32 | bombaclat | Hi, I installed asterisk on CentOS using the guide from asterisk.org (http://www.asterisk.org/downloads/yum). It seems some files like /etc/init.d/dahdi or /etc/dahdi are missing. Is there something I missed? |
| 20:46.37 | zamba | like going through enum and what not |
| 20:46.47 | zamba | drmessano-lt: you're arguments aren't really very solid |
| 20:46.47 | drmessano-lt | Asterisk can route through ENUM |
| 20:46.49 | zamba | your* |
| 20:46.57 | zamba | "silly" and "A++++" :) |
| 20:47.11 | drmessano-lt | Ok, how about this.. |
| 20:47.23 | carrar | ok |
| 20:48.05 | carrar | zamba, a OpenSER like software is smart, if you have a application that needs it |
| 20:48.19 | carrar | otherwise it's a waste of time |
| 20:48.30 | zamba | carrar: what could be an application that needs it? |
| 20:48.30 | bmoraca_work | zamba: SER can be useful for abstracting the media path from the user (to provide redundancy, etc, more easily than you could with just asterisk). it can also separate signalling overhead from the servers that provide the fascilities |
| 20:48.33 | carrar | application == setup |
| 20:48.46 | zamba | carrar: ok |
| 20:48.51 | carrar | zamba, if you are servicing SIP out to the internet |
| 20:49.00 | zamba | carrar: that i am, actually |
| 20:49.03 | drmessano-lt | OpenSER is more of a SWITCH and designed to handle thousands of concurrent calls. Asterisk is more of a full featured telephony platform. OpenSER HAD been used in lesser applications when Asterisk was unable to handle large numbers of concurrent calls.. however, that has changed |
| 20:49.04 | carrar | or have SIP customers |
| 20:49.07 | MagnusThundercoc | where can i go to get help with sip issues? |
| 20:49.26 | carrar | but you will want to heavly research it |
| 20:49.43 | zamba | carrar: so it's not wise to have asterisk handling sip out to the internet? |
| 20:49.50 | carrar | zamba, why do you think you need it? |
| 20:49.51 | drmessano-lt | .... |
| 20:49.58 | zamba | i guess it's not OpenSER anymore, but forked into two different projects..? |
| 20:50.06 | carrar | zamba, Asterisk can hand out SIP just fine to the internet |
| 20:50.14 | zamba | 22:48:51 < carrar> zamba, if you are servicing SIP out to the internet |
| 20:50.28 | carrar | in mass Calls per second |
| 20:50.32 | zamba | implicitly that made me think that doing the same with asterisk was a bad idea |
| 20:50.32 | zamba | ok |
| 20:50.36 | carrar | like 50 calls per second |
| 20:50.39 | zamba | got it |
| 20:50.46 | carrar | or more |
| 20:50.52 | zamba | as in.. i understand.. not that i've got 50 calls per second :) |
| 20:51.22 | carrar | Thats just one example |
| 20:51.41 | zamba | but the routing looks to a bit more flexible, in the way that you can set up quite complex call patterns.. using m4 or whatever it's called |
| 20:52.00 | zamba | with if tests and so on |
| 20:52.16 | zamba | doing the same with asterisk seems like quite the hassle |
| 20:52.24 | carrar | doing what |
| 20:52.28 | zamba | but then again, i'm not that experienced with asterisk |
| 20:52.32 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:52.37 | carrar | What are you trying to offer that you think you need SER |
| 20:52.51 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 20:53.03 | carrar | SER can handle up to 900 calls per second |
| 20:53.07 | carrar | or more |
| 20:53.30 | carrar | Lets figure this out |
| 20:53.38 | carrar | LETS DO IT |
| 20:54.15 | *** join/#asterisk feeds (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 20:54.35 | MagnusThundercoc | where can i go to get help? |
| 20:54.40 | carrar | 911 |
| 20:54.48 | MagnusThundercoc | yeah i expected a stupid response but really |
| 20:55.03 | carrar | You can go to #freepbx for freepbx help |
| 20:55.26 | drmessano-lt | zamba: You have a lot to learn about Asterisk.. If you're talking about calls per second and Asterisk's suitability as a switch, that's one thing, but lets not confuse the feature set here. Asterisk is very capable of quite complex call routing |
| 20:56.02 | drmessano-lt | If you want someone to tell you OpenSER is better because you think the config is easier, you may want to try their IRC channel |
| 20:56.06 | carrar | zamba, start with Asterisk, if you think you need to add in SER you can |
| 20:56.28 | drmessano-lt | We're not going to talk you out of using Asterisk unless you want to scalability of a high order |
| 20:56.39 | drmessano-lt | -to |
| 20:56.41 | carrar | and less features |
| 20:56.51 | drmessano-lt | Indeed |
| 20:56.58 | carrar | SER != asterisk |
| 20:57.22 | *** part/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 20:57.37 | carrar | SER == rope with greese |
| 20:57.37 | drmessano-lt | SER = High scalable SIP PROXY... Asterisk = full featured telephony toolkit |
| 20:58.34 | moy_ | megalomano: I'd say still not concrete enough ... span/channel is a concept at E1 level, not SIP level, if you want to carry such information on SIP would have to be through extra headers in the SIP invite for example, and then catch them in Asterisk and route based on that info |
| 21:05.35 | carrar | *burp* |
| 21:08.24 | *** part/#asterisk MagnusThundercoc (~Magnus@18.62.30.145) |
| 21:11.22 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:12.33 | *** join/#asterisk seanjohn (~admin@173.50.101.14) |
| 21:12.33 | megalomano | moy_: I have a gateway that receive PSTN call ( toll freee ) and this gatway is in SIP mode , in the same # I can receive 25 calls at the same time ,, but ,aaahh!!!! may be is a service that iscapable to make span ( from the company that offers the number ) right? |
| 21:15.27 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 21:16.00 | moy_ | so what do you want asterisk to do? concretely |
| 21:16.16 | moy_ | receive what or send what? |
| 21:16.21 | *** join/#asterisk Goshen (~Goshen@c-174-52-7-122.hsd1.ut.comcast.net) |
| 21:16.52 | megalomano | receive multiple calls from de same # |
| 21:17.36 | Goshen | Is a Telrad Avanti phone an asterisk compatible phone, or is it proprietary? |
| 21:18.11 | moy_ | megalomano: yes it can do that |
| 21:19.41 | megalomano | excuse ,,,, multiple calls to the same # ,,, i.e. A,B,C dial XXXX at the same time and IVR plays a message ( this number is a local PSTN ) |
| 21:22.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:27.22 | seanjohn | I get: "Registration for 'provider.host.name' timed out, trying again (Attempt #2)" three or four times every day. any recommendations, as it is NOT my network (the old provider NEVER had this problem)? What would you recommend for maxexpirey, minexpiry, and defaultexpiry. The asterisk box's ip never changes and is never natted; based on this, what should "nat" equal - never, no, or update? |
| 21:28.40 | seanjohn | externhost=sip.sheltoncomputers.com externip=sip.sheltoncomputers.com domain=sip.sheltoncomputers.com,172.59.173.20 (internip) |
| 21:29.32 | [TK]D-Fender | seanjohn: externip has to be an IP, not a host |
| 21:30.45 | seanjohn | fender: the voipinfo.org says it can be a hostname; it is looked up by asterisk and inserted in sip messages as an ip |
| 21:31.40 | seanjohn | my main question is what the setting is for registration WAIT time upon a failed attempt |
| 21:32.12 | *** join/#asterisk fofware (~fabian@186.125.126.115) |
| 21:32.38 | seanjohn | externip = IP_Address or a hostname : Address that we're going to put in SIP messages if we're behind a NAT. I |
| 21:33.32 | seanjohn | i need it to keep retrying every 5 seconds no matter what as, when I do sip reload, it registers |
| 21:34.39 | [TK]D-Fender | seanjohn: If you could put a host then you wouldn't need EXTERNHOST |
| 21:35.37 | seanjohn | i'm not saying thats not true; i'm just saying what voipinfo says. Maybe you should edit it as it is an open wiki |
| 21:37.12 | seanjohn | my registration timeout is 5 seconds, 0 for registerattempts, 120 for minexpiry, 3600 for maxexpiry, and 260 for defaultexpiry. maybe the reason it isn't registering is because it is trying to reregister too early? |
| 21:38.57 | seanjohn | fender, expert, talk to me :) what should these register settings be optimally? |
| 21:40.39 | [TK]D-Fender | seanMaybe you should look at the sip debug |
| 21:41.03 | seanjohn | it will be hours before it fails to register again |
| 21:41.21 | seanjohn | i really think its them |
| 21:42.38 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 21:47.15 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 21:48.14 | seanjohn | Scheduling destruction of SIP dialog '3683beca657134e757b8c157543d9777@127.0.0.1' in 32000 ms |
| 21:48.25 | seanjohn | is it normal to have the localhost address? |
| 21:48.53 | [TK]D-Fender | seanMeaningles here |
| 21:49.08 | seanjohn | ? |
| 21:49.46 | seanjohn | it is a successful register attempt |
| 21:50.23 | seanjohn | Via: SIP/2.0/UDP 173.50.101.11:5060;branch=z9hG4bK283898c5;received=173.50.101.11;rport=1028 |
| 21:51.47 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 21:53.31 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 22:04.19 | seanjohn | fender, my provider supports static ip auth but this wouldn't allow me to do multiple backup accounts, like I currently have, would it? |
| 22:05.16 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:06.00 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 22:09.13 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 22:09.19 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 22:21.44 | ChannelZ | If you've got an Intel dual-core CPU at say 2GHz, isn't that 2GHz per core? |
| 22:22.14 | WIMPy | sure |
| 22:22.25 | ChannelZ | Does Linux not quite understand that? |
| 22:22.35 | BarthezZ | linux does perfectly |
| 22:22.43 | BarthezZ | mostly better than windows :p but that isn't a surprise |
| 22:23.15 | ChannelZ | well this system is a Core2 Duo at 2.93GHz but /proc/cpuinfo seems to show each as 1600.00 MHz |
| 22:23.35 | BarthezZ | speedstep/cool'n'quiet/whatever it's called these days? |
| 22:23.40 | [TK]D-Fender | ChannelZ: And what is that "system"? |
| 22:23.50 | WIMPy | cpufreq |
| 22:24.11 | frigidzephyr | yeah probably just scaling |
| 22:24.27 | ChannelZ | A generic server, lets see.. |
| 22:24.48 | WIMPy | And cpufreq shows wrong values for me as well. /proc/cpuinfo is correct, however. |
| 22:24.48 | ChannelZ | Ubuntu 10.04, kernel 2.6.32 |
| 22:28.59 | BarthezZ | turn off cpu power save stuff in your bios and it will run on the promised mhz all the time :p |
| 22:29.21 | ChannelZ | I've got a 2.2ghz AMD box benchmarking 'core show translation' twice as fast as a 2.93ghz Intel core2 |
| 22:29.21 | WIMPy | bios? |
| 22:29.38 | ChannelZ | I'll have to look in the bios later |
| 22:30.24 | ChannelZ | maybe something odd is turned on |
| 22:31.22 | BarthezZ | think so :p |
| 22:45.16 | *** join/#asterisk hfb (~hfb@96.247.66.242) |
| 22:46.33 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 22:49.09 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 22:56.12 | *** join/#asterisk md0 (~mdavid@c-98-222-7-129.hsd1.il.comcast.net) |
| 23:00.52 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 23:07.48 | *** join/#asterisk DelphiWorld (~Delphi@41.200.20.34) |
| 23:07.50 | DelphiWorld | hi |
| 23:07.57 | DelphiWorld | how do i can get asterisk french prompt? |
| 23:08.49 | BarthezZ | search or buy them |
| 23:09.23 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 23:11.03 | DelphiWorld | BarthezZ: wtf? if i found one why i ask here? |
| 23:13.37 | BarthezZ | I see 6 off them on the voip-info.org wiki |
| 23:15.12 | ChannelZ | There are stock french prompts |
| 23:15.30 | theslob | DelphiWorld: make menuselect and check the option for french voice promts |
| 23:15.30 | ChannelZ | They are on the FTP or do 'make menuconfig' in the source dir and select the appropriate set |
| 23:15.59 | theslob | menuconfig that is |
| 23:16.08 | DelphiWorld | theslob: i like to get it using wget, unable to read the make menu sory (gblind using tts) |
| 23:16.32 | DelphiWorld | ChannelZ: ftp path please? |
| 23:17.38 | ChannelZ | http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-fr-ulaw-current.tar.gz |
| 23:17.43 | ChannelZ | Shal I wipe your ass as well? |
| 23:17.59 | ChannelZ | I have a moist towlette somewhere |
| 23:22.16 | ChannelZ | sorry it's been a long day |
| 23:22.55 | ruben23 | hi guys..? how do we know asterisk have crashed and it reason why it did crashed, coz my setup every shift i once or twice reboot my asterisk it suddenly stop and goes out form asterisk CLI..any idea. |
| 23:24.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 23:24.58 | ChannelZ | Are you running with some verbosity turned on to see what it's doing at the time? |
| 23:25.18 | ChannelZ | Or anything in your syslogs about core dumps or any other clues |
| 23:25.22 | *** join/#asterisk garymc (~chatzilla@host86-181-172-161.range86-181.btcentralplus.com) |
| 23:26.41 | ChannelZ | Anyway DelphiWorld sorry for being an a-hole, it's time for me to home I think |
| 23:27.47 | ChannelZ | s/to /go / |
| 23:28.05 | ChannelZ | hell I screwed that up too. bleh |
| 23:28.27 | theslob | so its defenitly time to go home :) |
| 23:28.40 | ChannelZ | indeed |
| 23:28.47 | ChannelZ | bbl |
| 23:28.52 | theslob | cu |
| 23:28.56 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 23:29.32 | *** join/#asterisk carrar (~tim@2604:5000:11:1::3) |
| 23:30.12 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 23:37.42 | *** join/#asterisk Diffen2 (~diffen2@78-82-119-199.tn.glocalnet.net) |
| 23:39.42 | Diffen2 | Evning, is it possible to add a remote agent to a queue? so if you dial a number with you cellphone you get prompted that you are logged in to a queue and then you can receive queue calls in your cellphone and transfer the call with ##. |
| 23:56.53 | _structz | Hello all, having a problem with my asterisk version 1.6.2.11 everytime i get [Aug 27 11:19:28] WARNING[24664] pbx.c: Don't know what to do with 'SIP/6463-0000000 my asterisk crashes does anyone know how to prevent this? |
| 23:57.46 | bougyman | _structz: that's a feature |
| 23:57.58 | _structz | bougyman, oh nice |
| 23:58.11 | _structz | is there somehow "connect" to the crash? |
| 23:58.14 | _structz | there=that ? |
| 23:58.25 | _structz | cause that was the last message on the full log after the crash |
| 23:58.45 | bougyman | hard to say |
| 23:58.57 | bougyman | after 3 years of using asterisk, i consider it working an anomaly. |
| 23:59.04 | bougyman | and it crashing the normal specified behavior. |
| 23:59.11 | _structz | l0l |
| 23:59.36 | _structz | perhaps the lack of dahdi installed could be the crash problema? |
| 23:59.42 | bougyman | is that all of the warning? |
| 23:59.46 | bougyman | there should be more in the logs. |
| 23:59.58 | bougyman | if it's a dahdi malfunction dmesg may tell you. |
