| 00:02.15 | tcastleman | JT, ok broken it now (well it's broken itself) |
| 00:02.34 | tcastleman | JT, just adding the line to the dialplan for the test |
| 00:04.10 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-145-247.ny325.east.verizon.net) |
| 00:04.20 | *** join/#asterisk The_DoC^ (n=me@pool-71-244-151-88.chi.dsl-w.verizon.net) |
| 00:06.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:08.39 | k-man_ | jt, just out of interest, do you run asterisk on a linux box? |
| 00:08.47 | JT | yes |
| 00:09.04 | tcastleman | JT, still no sound |
| 00:09.15 | k-man_ | jt, which kernel version do you use? |
| 00:09.37 | JT | k-man_: a few, usually whatever is the most recent 2.6 at the time |
| 00:10.30 | k-man_ | jt, that jitter problem i have been experiencing may be an issue to do with the kernel version, could I trouble you to tell me a specific version you use that you know works? |
| 00:10.50 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 00:10.59 | JT | 2.6.18 2.6.20.1 |
| 00:11.18 | k-man_ | jt, ah, thanks |
| 00:13.52 | tcastleman | Executing Wait("<trunkname>", "2") in new stack |
| 00:13.53 | tcastleman | Executing Playback("<trunkname>", "custom/CallNotWanted") in new stack |
| 00:13.55 | tcastleman | Playing 'custom/CallNotWanted' (language 'en') |
| 00:14.21 | JT | did you Answer before the Wait? |
| 00:15.33 | tcastleman | JT, I created an test extension as follows: |
| 00:16.06 | tcastleman | [test-custom] |
| 00:16.08 | tcastleman | exten => s,1,Wait(2) |
| 00:16.09 | tcastleman | exten => s,2,Playback,custom/CallNotWanted |
| 00:16.11 | tcastleman | exten => s,3,Hangup |
| 00:16.23 | JT | Answer it first |
| 00:16.41 | JT | you should always Answer calls unless you are aiming for a special behaviour and you know what you are doing |
| 00:17.58 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:19.26 | tcastleman | JT, I'm not sure we're on the same page. I have created a dummy extension which I dial from my SIP extension which just plays that test CallNotWanted sound |
| 00:20.02 | JT | i know |
| 00:20.08 | JT | Answer the call already. :) |
| 00:20.27 | tcastleman | JT, I am not sure what you mean about Answer |
| 00:20.31 | Dovid | what cash call ? |
| 00:20.32 | Dovid | money ? |
| 00:20.51 | Dovid | where ? |
| 00:20.51 | Dovid | ;) |
| 00:20.57 | JT | tcastleman: in the cli: show application answer |
| 00:21.12 | JT | or search the wiki for asterisk cmd answer |
| 00:23.26 | Dovid | or JT |
| 00:23.29 | Dovid | ~RTFM |
| 00:23.30 | jbot | rumour has it, rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 00:24.10 | necromcr | :-| |
| 00:24.23 | JT | Dovid: i don't think that was necessary |
| 00:24.31 | necromcr | please direct me to sip client questions channel |
| 00:24.43 | infinity1 | Juggie: i added SIG{HUP} = IGNORE to my script, but for some reason its still being killed |
| 00:25.09 | infinity1 | Juggie: does it have something to do with streaming ? |
| 00:27.16 | tcastleman | JT, I'm not sure this is the root of the problem. Although I understand your point trying to preach best practices, I primarily use freepbx and this test extension I have created is as per your instructions. Asterisk answers it, waits, then plays the sound. Although I cannot hear it. |
| 00:27.59 | *** join/#asterisk ta^3 (n=tacvbo@189.146.198.59) |
| 00:29.13 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
| 00:29.13 | JT | tcastleman: did you add Answer as priority 1 in the test extension? |
| 00:29.55 | tcastleman | JT, yes it looks like this: |
| 00:29.57 | tcastleman | [test-custom] |
| 00:29.58 | tcastleman | exten => s,1,Answer() |
| 00:30.00 | tcastleman | exten => s,2,Wait(2) |
| 00:30.01 | tcastleman | exten => s,3,Playback,custom/CallNotWanted |
| 00:30.03 | tcastleman | exten => s,4,Hangup |
| 00:30.07 | JT | ok |
| 00:30.11 | JT | if that had no effect |
| 00:30.19 | JT | then it's not the problem i was thinking of |
| 00:30.26 | JT | that is solved with Answer |
| 00:30.29 | JT | and Wait(2) |
| 00:31.02 | tcastleman | JT, nope. Still can't hear anything |
| 00:31.07 | JT | hrm |
| 00:31.22 | JT | is your zap timing ok? doubt that'd effect sip even if it wasn't |
| 00:32.50 | tcastleman | JT, well.. the TDM880P has 8 FXS ports, which are used as 8 zap extensions setup in zapata. Zaptel is installed and running no problem. |
| 00:33.08 | JT | zttest tests zap accuracy |
| 00:36.26 | tcastleman | JT, ok running zttest now. |
| 00:39.18 | tcastleman | JT, 15 tests, average 99.999186% |
| 00:40.13 | JT | ok |
| 00:40.24 | *** join/#asterisk [hC] (n=hardcore@66.119.172.82) |
| 00:40.53 | infinity1 | my script AGI script terminates even though I have it ignoring SIGHUP and SIGINT ... whats going on here? |
| 00:41.10 | JunK-Y | infinity1: u need DeadAGI |
| 00:41.21 | infinity1 | JunK-Y: i tried calling it with deadagi as welL. |
| 00:41.24 | infinity1 | i'm perplexed! |
| 00:41.49 | infinity1 | the script seems to work okay as long as the person doesn't have up on the script. |
| 00:42.08 | tcastleman | JT, incidently, the TDM880 has power issues. None of the ZAP extension devices connected to it phycially ring.. although on the asterisk console they appear to ring normally. We think the power unit on it has gone and I am organising an RMA. Could this affect, at random, whether the sound asterisk outputs is audible do you think? |
| 00:42.35 | JunK-Y | doesnt have up? |
| 00:43.18 | infinity1 | er hang up |
| 00:43.19 | infinity1 | hehe |
| 00:43.20 | tcastleman | JT, when I say "we" I refer to myself and the guys a Digium |
| 00:43.34 | JT | tcastleman: is the aux power connector connected to the TDM800P? |
| 00:43.36 | JunK-Y | what about agi debug? |
| 00:43.47 | infinity1 | hmmm .. you know. I think my script might actually be working ..i just can't tell because it finishes so quickly because it doesn't stream |
| 00:43.51 | infinity1 | i "THINK" |
| 00:44.10 | tcastleman | JT, yes. And that is not faulty as we have swapped it with a CDROM drive which powers up np. |
| 00:44.11 | infinity1 | after the caller hangs up, agi debug doesn't have output |
| 00:44.30 | JunK-Y | show me CLI output with agi debug in |
| 00:44.41 | JT | tcastleman: ok |
| 00:45.55 | infinity1 | JunK-Y: thanks :) ..so my script is working. |
| 00:46.15 | tcastleman | JT, so that's the only thing I can think that's wrong with our setup. The misdn and asterisk itself have been setup from a fresh install of the latest sources. Freepbx is the latest version, and this doesn't seem to be a dialplan issue anyway. |
| 00:46.30 | infinity1 | JunK-Y: now i just have to make sure it executes even if the phone isn't answered (i'm using .call files) |
| 00:46.34 | *** join/#asterisk ltd (n=z@202-161-20-161.dyn.iinet.net.au) |
| 00:46.46 | JT | tcastleman: what kernel version? |
| 00:47.17 | tcastleman | 2.6.18-4-686 #1 SMP |
| 00:47.21 | tcastleman | Debian Etch |
| 00:47.29 | JT | ok |
| 00:48.24 | rad07 | I installed asterisk 1.4.2. Any web interface for it. Is freePBX and Asterisk GUI compatible with this version |
| 00:49.01 | tcastleman | rad07: the current stable version of Freepbx is not compatible with 1.4.X |
| 00:49.07 | rad07 | Did Asterisk GUI come with Asterisk installation. Somebody installed Asterisk and it is running. How to get Web GUI |
| 00:49.07 | infinity1 | okay. i'm running an agi script, and i need it to execute even if the person doesn't answer the phone. |
| 00:49.34 | tcastleman | rad07: compiled from sources - no. |
| 00:50.10 | tcastleman | rad07: www.freepbx.org see the #freepbx channel. |
| 00:50.10 | JT | no |
| 00:50.13 | infinity1 | JunK-Y: any ideas? :) |
| 00:50.14 | rad07 | tcastleman: I am just testing and becoming familiar with it. Any unstable version compatible with Asterisk |
| 00:50.15 | JT | tcastleman: he is not asking about freepbx |
| 00:50.30 | JT | rad07: asterisk gui can be added to asterisk |
| 00:51.05 | JunK-Y | infinity1: where the output i requested? |
| 00:51.21 | infinity1 | comming |
| 00:51.21 | rad07 | Is freePBX richer then Asterisk GUI? |
| 00:51.27 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 00:51.35 | JT | it's measier |
| 00:51.40 | tcastleman | JT, in the context of the precise questions I was answering I was correct :P |
| 00:51.44 | JT | and even less supported here |
| 00:51.46 | infinity1 | JunK-Y: http://pastebin.ca/411439 |
| 00:52.06 | rad07 | So at this point can I put a front end to A. 1.4.2 |
| 00:52.10 | JT | this channel is only for supporting asterisk with no gui though |
| 00:52.19 | JT | ~trixbox |
| 00:52.29 | jbot | extra, extra, read all about it, trixbox is junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 00:52.39 | infinity1 | JunK-Y: so basically, it says it is never answered, so my agi didn't run. if it doesn't run, its not going to continue. |
| 00:52.50 | JunK-Y | infinity1: and with the part the agi is executed? |
| 00:52.56 | infinity1 | JunK-Y: the call file? |
| 00:53.22 | cr4z3d | when putting someone on hold through my sip phone (hold button) is there a way to edit what happens |
| 00:53.30 | tcastleman | JT, anyway do you think our dodgy 880P could be at fault? |
| 00:53.37 | JunK-Y | ur agi is never executed, right? |
| 00:53.41 | infinity1 | yes |
| 00:54.06 | JunK-Y | so how do ya want it to continue after the channel hangups if isnt executed? |
| 00:54.14 | JT | tcastleman: i doubt it could affect everything, including sip |
| 00:55.05 | tcastleman | JT, or many an alsa or sound issue? |
| 00:55.07 | infinity1 | JunK-Y: i need to script to execute because in the agi, it marks a file that the call was unsuccessful and a cron job then gets the information and creates a new call file |
| 00:55.14 | rad07 | jbot: Are you saying that freePBX is tribox's thing? What should I use then? Do they make Web interface that is compatible with latest Asterisk? I didn't install Samples. I need to create some intial configuration. This is my 2 day with Linux? Can somebody throw a few commands in a private chate to get me started? I would prefere GUI based configuration at this point. |
| 00:55.23 | tcastleman | JT, but it's the randomness of it which is bizzare. |
| 00:55.25 | infinity1 | JunK-Y: er ..i need the script to execute because ... |
| 00:55.50 | JunK-Y | then , start ur agi as a deadagi and ya will be fine. |
| 00:55.56 | JT | tcastleman: do you even have also? |
| 00:56.04 | JT | rad07: no point talking to a bot |
| 00:56.10 | infinity1 | JunK-Y: notice in the pastebin it is started with deadagi :) |
| 00:56.12 | tcastleman | rad: lol |
| 00:56.12 | JT | rad07: trixbox is freepbx + centos |
| 00:56.42 | JT | rad07: |
| 00:56.42 | JunK-Y | infinity1: no, agi isnt started in that pastebin. |
| 00:56.47 | JT | ~thebook |
| 00:56.49 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:56.50 | JT | ~thewiki |
| 00:56.51 | jbot | from memory, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 00:56.51 | tcastleman | JT: you mean alsa.. it's built into the kernel isn't it? |
| 00:56.53 | rad07 | soory guys. I am even new to IRC. How do you recognize bot? by name? |
| 00:56.57 | JT | and you can still 'make samples' |
| 00:56.59 | infinity1 | JunK-Y: Attempting call on SIP/xten for application DeadAGI(one.agi) (Retry 1) |
| 00:57.10 | JunK-Y | thats not |
| 00:57.11 | JT | rad07: name and the fact i used a command before it said anything |
| 00:57.18 | JunK-Y | <PROTECTED> |
| 00:57.26 | JunK-Y | thats an agi forking. |
| 00:57.50 | infinity1 | JunK-Y: well, how am i suppose to get it to say that? if the call isn't answered, the script doesn't run. |
| 00:58.10 | JunK-Y | i cant help ya here. |
| 00:58.15 | infinity1 | hrmz |
| 00:58.36 | infinity1 | maybe i have to initiate the call from within an agi script somehow |
| 00:58.38 | infinity1 | i duno |
| 00:58.51 | infinity1 | seemed easy to just create a call file in the outgoing directory |
| 00:59.05 | cr4z3d | when i put someone on hold how do i execute something like Playback(your-call-is-important) |
| 00:59.11 | cr4z3d | and then start music on hold |
| 00:59.30 | JunK-Y | cr4z3d: use the ACD with queues. |
| 00:59.54 | cr4z3d | there's no like simple way to do it? |
| 00:59.57 | rad07 | JT: Is making samples a way to get me started whether I have or don't have Web interface for Asterisk? |
| 01:00.01 | cr4z3d | i really don't need queues |
| 01:00.08 | tcastleman | rad07: yes |
| 01:00.09 | infinity1 | cr4z3d: make the hold music the "your call is importatn" |
| 01:00.23 | JT | rad07: reading the book is good too |
| 01:00.45 | infinity1 | JunK-Y: any more ideas? this damn agi stuff is SO close to being done. this last part is kicking my a** |
| 01:01.00 | cr4z3d | infinity1, i want that to play first and then go into a random music file so i'm guessing putting that in my hold music wouldn't work |
| 01:01.11 | JunK-Y | brb in 30 min. |
| 01:01.18 | tcastleman | JT, I guess we'll wait on getting the RMA for the 880P.. do you think downgrading to sarge might help? |
| 01:01.35 | infinity1 | cr4z3d: well, you could edit your hold music and past clips of "you're important" in them. |
| 01:01.36 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:01.48 | infinity1 | cr4z3d: thats the only way i can think of |
| 01:01.56 | JT | tcastleman: doubt it |
| 01:01.57 | infinity1 | other than ACD |
| 01:02.15 | cr4z3d | infinity1, eh that's not a bad idea but for now i guess i do'nt really need it |
| 01:02.50 | tcastleman | JT, ok well thanks for your help. |
| 01:03.27 | cr4z3d | i was jst thining there would be something similar to the "h" extension |
| 01:05.23 | infinity1 | cr4z3d: hangup? |
| 01:05.44 | cr4z3d | infinity1, yeah but something like that for hold |
| 01:06.19 | infinity1 | i might have to use that to execute an AGI even if the person being called doesn't answer. |
| 01:06.22 | infinity1 | dunno |
| 01:07.34 | infinity1 | though... i don't know how to pass a variable between the call file and the first and second agi executions |
| 01:11.09 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
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| 01:26.17 | rad07 | JT: Is making samples the starting point when configuring Asterisk for the first time? Once I do it what is next? Do I have any Front End to Asterisk? What be some GUI that are compatible with A. ver 1.4.2? I have Linksys SPA-3102 I could run Starter SPA3102-Asterisk Setup web page to configure it, but I don't know where to start with configuring Asterisk without Web based interface (that I saw in one of Tribox freePBX webast). Can any |
| 01:28.32 | mmartinn | hey people; what's up? |
| 01:30.03 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
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| 01:30.30 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:31.09 | cr4z3d | she's a brick... house |
| 01:32.07 | cr4z3d | she's mighty mighty |
| 01:34.12 | mmartinn | Quiet tonight ;) |
| 01:36.50 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
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| 01:47.18 | mmartinn | Anyone alive here who's on the Digium asterisk dev or user list? |
| 01:47.42 | JunK-Y | probably a lot of ppl. |
| 01:47.44 | Qwell | a ton of people I'm sure |
| 01:48.05 | mmartinn | Has it been down lately? |
| 01:48.14 | mmartinn | I tried to subscribe like 2 days ago, twice |
| 01:48.30 | mmartinn | Wondered if it's been dead lately like voip-info or something |
| 01:48.51 | Qwell | been working fine |
| 01:48.51 | Qwell | check your junk filter |
| 01:49.10 | mmartinn | My junk is scored but not filtered |
| 01:49.17 | mmartinn | Weird... |
| 01:49.42 | mmartinn | http://lists.digium.com/mailman/listinfo/asterisk-dev/ the right place? |
| 01:50.00 | Qwell | yes |
| 01:50.20 | mmartinn | I get email just fine these last few days, and I'm not filtering; this would be the third time I've filled it out... |
| 01:50.58 | mmartinn | I have a university address from where I work; I can't imagine it's on an RBL or something |
| 01:51.08 | mmartinn | I know they use greylisting... I suppose it could be that. |
| 01:54.13 | cr4z3d | i'm trying to setup different hold music's for different phones in my house, what's the best way to go about doing that? is there a way to set a different default hold music for each sip phone user/password? |
| 01:55.00 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 01:55.17 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 01:55.48 | mmartinn | cr4z3d: If you setup different hold music classes in musiconhold.conf, you can set them in a context as the default right when you Answer() that call in the dialplan. |
| 01:56.16 | anonymouz666 | is it possible to concat a var? Set(var=1) Next time I wanna add \,2 so var will have the value 1\,2 |
| 01:56.19 | mmartinn | I don't know if there's a quicker way using sip.conf, without a bunch of different contexts. You might be able to set it based on the extension all within a single context. |
| 01:56.53 | mmartinn | exten => 6000,2,MusicOnHold(${EXTEN}) or something |
| 01:57.26 | cr4z3d | hmm |
| 01:58.43 | mmartinn | This is definitely not the asterisk-dev list's fault; my server is hurting something in the process. |
| 02:01.51 | cr4z3d | mmartinn, looks like there's a "musicclass" options you can set in sip.conf nice |
| 02:02.09 | mmartinn | hmmm news to me; |
| 02:02.36 | cr4z3d | yeah there's also musiconhold but in the description it says same as musicclass |
| 02:07.08 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 02:12.58 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 02:12.58 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007), Zaptel 1.4.1 (Mar. 23, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
| 02:16.33 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 02:16.33 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007), Zaptel 1.4.1 (Mar. 23, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
| 02:18.12 | *** join/#asterisk armchang (n=iceman@202.84.117.68) |
| 02:20.32 | heison | anyone here using rx_fax with Asterisk 1.4.2? |
| 02:21.26 | armchang | hi, i was wondering if Asterisk could be used for commercial purpose? |
| 02:21.40 | mog | yes |
| 02:21.45 | Qwell | armchang: absolutely, as long as you follow the GPL |
| 02:22.24 | JT | it would depend what that commercial purpose is, but there are plenty of people using asterisk for commercial purposes |
| 02:23.12 | armchang | thank you Qwell sir. I'm planning to use it for country to country VOIP calls, is that ok? I haven't read the GPL but will read later. |
| 02:23.20 | Qwell | yes, definitely |
| 02:24.06 | Qwell | hey mog... |
| 02:24.14 | mog | hey Qwell |
| 02:24.17 | russellb | armchang: you have to send me $100, but then it's ok |
| 02:24.24 | mog | lol |
| 02:24.25 | Qwell | Somebody at Digigraph is gonna have to start smoking now that millson quit :P |
| 02:24.34 | anonymouz666 | Qwell: If I modified the source code of a GPL software should I provide the patch? |
| 02:24.36 | russellb | Qwell: you need to quit |
| 02:24.37 | Qwell | I can't be the only one! |
| 02:24.39 | Qwell | I do |
| 02:24.41 | mog | lol Qwell |
| 02:24.45 | ManxPower | looks like the HD died on one of my servers |
| 02:24.47 | mog | ill hang outside with ya |
| 02:24.47 | russellb | Qwell: I just had to leave the pool hall because the smoke was giving me a terrible headache |
| 02:24.52 | Qwell | :( |
| 02:24.57 | russellb | so now I'm at digigraph waiting for kevin to finish up :( |
| 02:25.21 | russellb | so yeah, quit |
| 02:25.26 | heison | armchang: what countries will you be sending calls across? |
| 02:25.30 | ManxPower | dog hates me |
| 02:25.56 | Qwell | russellb: how'd you guys do? |
| 02:26.34 | russellb | won 2 out of 3 so far |
| 02:26.38 | Qwell | sweet |
| 02:26.41 | russellb | i was the one that lost :( |
| 02:26.49 | Qwell | less sweet |
| 02:26.53 | russellb | it's all good. |
| 02:27.22 | russellb | now what should I do ... |
| 02:27.50 | aptura | russellb that is from cabonmonoxide |
| 02:27.56 | aptura | carbon mono |
| 02:28.02 | russellb | beautiful |
| 02:28.13 | aptura | yea cigs suck |
| 02:28.19 | aptura | I never smoked :) |
| 02:28.42 | russellb | i can't take the smoke filled bars ... i may have to quit after this session is over :-/ |
| 02:28.55 | Juggie | you smoke russell? |
| 02:28.58 | russellb | no |
| 02:29.04 | Juggie | oh quit bars? |
| 02:29.07 | russellb | quit playing in these pool leagues |
| 02:29.12 | Juggie | ohh... |
| 02:29.12 | aptura | my stepdad who smoked for years once went into respitory distress and now has emphisima..well partly does the surgon removed 1/2 his lungs. |
| 02:29.17 | russellb | because it requires me going out to bars a few times a week |
| 02:29.34 | Juggie | here in ontario, its illegal to smoke in public places. |
| 02:29.38 | Juggie | same with most provinces in canada. |
| 02:29.40 | russellb | oh god how i wish |
| 02:29.44 | Qwell | was in CA too |
| 02:30.01 | russellb | Qwell: then i wonder how the hookah bar was legal |
| 02:30.08 | russellb | Qwell: there was one right by the hotel, heh |
| 02:30.08 | Qwell | that's different...slightly |
| 02:30.18 | russellb | yeah, i enjoy how that smells :) |
| 02:30.24 | Qwell | special permits and such, I guess |
| 02:30.46 | russellb | well, now that I have taken this channel *completely* off topic ... |
| 02:30.59 | aptura | I am glad of the no smoking laws |
| 02:31.15 | aptura | mom did it for years untill she saw alot of her friends kick off. |
| 02:31.21 | aptura | :) |
| 02:31.38 | *** join/#asterisk bsd_tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 02:31.53 | JunK-Y | Juggie: who will be the next quebec prime minister? |
| 02:32.01 | aptura | its been a long time since doing a recompile of asterisk the kenel ect. f |
| 02:32.08 | mmartinn | It's not legal in Florida here, but it doesn't stop some people |
| 02:32.17 | aptura | moroney |
| 02:32.48 | aptura | mmartinn my dad walks like he is on mount everest |
| 02:33.01 | aptura | step dad that is. He cannot even push a lawn mower |
| 02:33.25 | aptura | Anyway back to the fun stuff. |
| 02:33.45 | mmartinn | there's bars here that are no smoking until "later" when no one is around |
| 02:33.49 | *** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net) |
| 02:34.43 | ManxPower | I don't go out to eat nearly as much since Louisiana did their no-smoking laws |
| 02:35.42 | aptura | ManxPower what part of La are you from |
| 02:35.45 | mmartinn | Even when I was a smoker, I didn't like it in my face; I don't mind if people do it away from up on my face though |
| 02:35.48 | mmartinn | =) |
| 02:36.02 | ManxPower | aptura: I go to New Orleans for work about every 4 - 6 weeks. I live near Birmingham AL |
| 02:36.11 | aptura | ahh |
| 02:36.30 | Strom_C | the only thing I don't like about bars which allow smoking is that when you get home, all your clothes smell awful |
| 02:36.59 | ManxPower | Strom_C: go to ones that don't allow it |
| 02:37.24 | Strom_C | ManxPower: I live in california ;) |
| 02:37.41 | ManxPower | Strom_C: so none of them allow it |
| 02:37.45 | Strom_C | exactly |
| 02:38.02 | Strom_C | it's one of those once-in-a-while things which happens when I'm travelling |
| 02:38.23 | *** join/#asterisk neoalex (n=neoalex@user-0ccenic.cable.mindspring.com) |
| 02:38.26 | Strom_C | ManxPower: well, "The People's Republic of California" is certainly a ha-ha-only-serious |
| 02:38.50 | JT | exposing all the smokers are we |
| 02:38.57 | JT | i think ManxPower has been sprung |
| 02:38.58 | neoalex | hi guys, any Romanian users in by any chance (I have a question about a Romanian provider so...) |
| 02:39.33 | JT | neoalex: some of the yate people are from romania |
| 02:39.43 | aptura | #yate |
| 02:39.44 | ManxPower | JT: California moved WAY down on my list of possible vacation places when I realized you can't smoke anywhere. |
| 02:39.56 | JT | ManxPower: heh |
| 02:39.59 | neoalex | yeah I know, but it's 5 AM in RO :D |
| 02:40.00 | JT | :) |
| 02:40.35 | JT | ManxPower: i love the erradication of smoking from places where i will be |
| 02:40.41 | JT | like in public |
| 02:40.46 | aptura | my preferances if I am to move clean air, LITTLE rain, small population, cheap realestate |
| 02:41.00 | JT | cheap PRIs |
| 02:41.01 | JT | DSL |
| 02:41.02 | JT | ;) |
| 02:41.35 | aptura | well that but if I can help it move. Vancouver is getting really expensive to live not to mention Canada taxes as a whole. |
| 02:41.51 | JT | sydney is really expensive |
| 02:41.55 | JT | property wise |
| 02:41.58 | cr4z3d | anyone good here with batch files? |
| 02:42.03 | Strom_C | well, one advantage of the people's republic of california is that telecom is cheap cheap cheap here |
| 02:42.15 | JT | batch files, aren't they off topic for #asterisk? ;) |
| 02:42.15 | ManxPower | aptura: there is not as much of a gap in USA taxes and Canada taxes as you might thing. |
| 02:42.25 | aptura | what does pri run there. |
| 02:42.45 | aptura | Manx try 45% tax bracket over 60k a year. |
| 02:42.46 | Strom_C | I think the basic circuit is like $300 a month or somesuch |
| 02:43.03 | aptura | 700 here for pri |
| 02:44.09 | aptura | 600 to 700 roughly. The 2010 olympics has been drawing speculative investors here for the last few years and driving anything near the venues though the roof. Top that with immigration and a booming economy. |
| 02:45.34 | Strom_C | *nod* |
| 02:45.34 | Strom_C | still, it's a very nice city |
| 02:45.36 | Strom_C | I was there in february |
| 02:45.42 | ManxPower | So when you go into the BIOS setup and you get funny blinking characters and you get DMA SEEK errors on boot, that's bad, right? |
| 02:45.54 | Strom_C | ManxPower: generally, yes |
| 02:46.07 | aptura | crap my zaptel compile failed |
| 02:46.30 | aptura | Strom where did you do the instruction |
| 02:46.32 | ManxPower | aptura: between federal, state, sales, and social security tax, you can easily pay 45% over 60k/year |
| 02:47.00 | ManxPower | don't forget county and city taxes and sales taxes too |
| 02:47.05 | aptura | ManxPower which is why washington was not a bad place to live. |
| 02:47.28 | ManxPower | looks like this PBX server is totally fried. |
| 02:47.29 | aptura | no state sales income tax and 7% sales. |
| 02:47.37 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 02:47.50 | aptura | build_tools/genudevrules > /etc/udev/rules.d/zaptel.rules |
| 02:47.50 | aptura | build_tools/genudevrules: line 1: udevinfo: command not found |
| 02:47.55 | aptura | my issue |
| 02:48.08 | aptura | plus lots of undefines |
| 02:49.12 | JT | hrm |
| 02:49.25 | JT | the olympics is coming to vancouver? |
| 02:49.57 | aptura | uhh yea :) dont you ever read the news? That anouncment was a couple years ago. |
| 02:49.59 | JunK-Y | yep, 2010 |
| 02:50.10 | JT | i must've totally missed it |
| 02:50.42 | aptura | I really dont care for it. IT is all hype and bad for the locals especially for those who want to buy a house. |
| 02:51.06 | JT | it will be a bit of a boost to the economy |
| 02:51.41 | aptura | With a large amound of the work force about to retire BC is in a job shortage. IT is even worse in Alberta. |
| 02:51.47 | ManxPower | JTI thinkk BC has other ways to boot it's economy |
| 02:52.04 | CunningPike | ManxPower: Agriculture? ;) |
| 02:52.09 | aptura | Lots of industrys doing well here. |
| 02:52.27 | ManxPower | CunningPike: something like that |
| 02:52.30 | aptura | CunningPike you own your own place? |
| 02:52.41 | CunningPike | aptura: Yup |
| 02:52.45 | aptura | nice |
| 02:52.55 | aptura | if its a house |
| 02:52.56 | aptura | :) |
| 02:53.41 | CunningPike | aptura: It is |
| 02:56.20 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 02:56.33 | aptura | :) |
| 02:57.04 | mcab | a house? in Vancouver? |
| 02:57.15 | mcab | you must've bought a few years ago, or won the lottery ;-) |
| 02:57.33 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 02:57.43 | *** join/#asterisk zimdog (n=zimdog@c-24-9-69-162.hsd1.co.comcast.net) |
| 02:57.55 | aptura | so what would cause this? I am getting alot of undefined's when doing make clean |
| 02:58.03 | InHisName | any Freedigits.com users active ? Need * cnx data |
| 02:58.26 | aptura | make install I mean |
| 02:58.38 | pigpen | [TK]D-Fender, fyi, the other day you noted that my zaptel issue in 1.4.2 was related to a menuselect issue...but it was actually an issue in zaptel sources.... |
| 02:59.10 | pigpen | it should be resolved in zap 1.4.2 per my business partner (gentoo kernel dev) |
| 02:59.18 | voltagex | JT: you still here? |
| 02:59.43 | JT | yes |
| 03:00.13 | voltagex | Koalavoip still isn't registering... I know it's up, and I can ping koalavoip.com.au |
| 03:00.45 | JT | shrug |
| 03:01.25 | voltagex | err, the register line does go in [general], right? |
| 03:03.14 | neoalex | ok this question is more suited here: can I place MGCP calls using asterisk? |
| 03:03.23 | *** join/#asterisk shadou (n=ajc@unaffiliated/dj-fu) |
| 03:05.12 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:06.12 | pigpen | neoalex, seems pretty shady... |
| 03:06.39 | neoalex | there's info on how to do it here: http://www.voip-info.org/wiki-Asterisk+MGCP+channels |
| 03:06.49 | neoalex | just curious if anyone has ever tried it |
| 03:06.55 | neoalex | and if it actually works |
| 03:07.15 | pigpen | yeah..never tried it. |
| 03:07.53 | mmartinn | nite nite |
| 03:07.55 | *** part/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 03:08.52 | *** join/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:09.13 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:11.11 | *** join/#asterisk robert497 (n=robert@24-155-83-174.ip.grandenetworks.net) |
| 03:14.51 | pigpen | anyone using eyeBeam for video? |
| 03:15.34 | pigpen | We use primarly Mac's at the desktop, and sometimes while remote, iChat will puke over various hotel's firewall's. |
| 03:15.49 | pigpen | Where as, over a VPN, ip to ip would work fine, but iChat does not support it. |
| 03:16.23 | neoalex | pigpen: yes I use eyeBeam for video |
| 03:16.43 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 03:17.07 | pifiu | anyone have experience setting up a linksys 921 behind a firewall? |
| 03:17.25 | pifiu | i seem to have all the STUN server settings right but it just wont register |
| 03:17.26 | pigpen | neoalex, how well does it do? |
| 03:17.42 | pigpen | I know much of video depends on the line quality... |
| 03:18.59 | neoalex | well... if you're going to be using it from a hotel... I don't know how well it would work |
| 03:19.20 | neoalex | but in order to get through hotel firewalls I recommend using a vpn |
| 03:19.33 | pigpen | yeah...iChat does ok..but when a hotel has a suckie firewall....iChat pukes. |
| 03:19.41 | pigpen | Yeah...definatly a vpn. |
| 03:19.50 | neoalex | openvpn works but I've had better performance with hamachi |
| 03:20.01 | neoalex | it's a breeze to install too |
| 03:20.20 | pigpen | I stayed at a hotel in Sanpoint, ID. I replaced their firewall for a week using Monowall on a Soekris.... |
| 03:20.29 | pigpen | The owner wanted to buy it off of me... |
| 03:21.07 | pigpen | Yeah..we have pretty established vpn's using vpntracker connecting to strongswan on various linux boxes. |
| 03:21.29 | pigpen | So I guess I could figure eyeBeam would be as good as iChat... |
| 03:21.37 | pigpen | (same codec) |
| 03:21.59 | neoalex | I imagine so.. haven't use iChat though |
| 03:22.19 | neoalex | and I haven't used eyeBeam on Mac either |
| 03:22.30 | neoalex | didn't even know they had a version for Macs |
| 03:22.50 | pigpen | I used it about a year ago...which was the only one that would work with my bluetooth headset decently. |
| 03:23.38 | pigpen | Yeah..Windows and Mac. |
| 03:27.03 | neoalex | do you have a video phone at home, or would you're daughter use eyeBeam as well |
| 03:27.20 | pigpen | yeah...My wife would be using a mac book pro. |
| 03:27.28 | pigpen | or my Quad at my desk. |
| 03:27.51 | neoalex | well then if it's just PC to PC you don't really need to pass it through asterisk |
| 03:27.57 | pigpen | yeah... |
| 03:28.06 | neoalex | you can use any other IM out there |
| 03:28.21 | pigpen | I was looking for a mac program just to ip to ip video... |
| 03:28.27 | pigpen | but iChat/iSight does not. |
| 03:28.49 | pigpen | and where I do -allot- of asterisk work...having a sip softphone around is not a bad idea... |
| 03:28.58 | pigpen | I already use the hell out of idefisk. |
| 03:29.12 | neoalex | true |
| 03:29.51 | neoalex | make sure you use g729 with eyebeam so at least the sound won't eat to much bandwidth |
| 03:29.52 | pigpen | this way maybe I'll ditch my Polycom 650 :) |
| 03:30.14 | JT | and sell it to JT for cheap |
| 03:30.27 | pigpen | yeah, I figure that or gsm if it is available. |
| 03:30.29 | neoalex | yeah... on ebay private sale, only me invited |
| 03:30.32 | neoalex | please :D |
| 03:30.39 | pigpen | haha. |
| 03:31.31 | neoalex | ooooh... don't worry JT he has enough of them I see :D |
| 03:31.34 | pigpen | I swear, my customers love them. |
| 03:31.49 | neoalex | why? |
| 03:31.55 | pigpen | backlit. |
| 03:32.04 | pigpen | and a little bit better sound... |
| 03:32.13 | pigpen | about $35 more than the 601's |
| 03:32.15 | neoalex | ummm... you install pbxes in night clubs :D |
| 03:32.40 | pigpen | We have allot of customers who are AV, programers, etc...who like to work in the dark. |
| 03:33.05 | neoalex | yeah... I have a backlit phone too... I don't know why I'm bitching :D |
| 03:33.40 | voltagex | http://pastebin.ca/411582 << anything look wrong with that iax.conf? the koalavoip entry is not registering. |
| 03:34.35 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 03:36.08 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 03:36.08 | neoalex | pigpen: I see those polycom have something called HDvoice |
| 03:36.12 | JT | neoalex: enough what? |
| 03:36.19 | JT | hdvoice is g.722 |
| 03:36.27 | pigpen | yeah..they are a fuller sounding call... |
| 03:36.39 | pigpen | deeper and more "rich" than the 601. |
| 03:36.40 | neoalex | hmmm... what's the bandwidth usage on that |
| 03:36.51 | pigpen | same..just better speakers. |
| 03:37.01 | JT | neoalex: i have enough what? |
| 03:37.21 | neoalex | not you... pigpen has enough phones for both of us |
| 03:37.27 | neoalex | nevermind... dumb joke anyway |
| 03:37.38 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:38.06 | pigpen | Well, I am a true nerd...I am running a 650, 601, 2 x 501's & a 301 in my house. |
| 03:38.06 | aptura | what is the typical compile time of * on a 2.8? |
| 03:38.23 | pigpen | but...this is my test bed... |
| 03:38.23 | SwK | NERD! |
| 03:38.48 | JT | but i have a channel bank! |
| 03:38.49 | JT | ! |
| 03:38.55 | aptura | have one next to your shower pigpen? |
| 03:38.57 | pigpen | That's me... 5' 1", weighs 102 lbs, bleach white.... |
| 03:39.08 | pigpen | No..toilet.. |
| 03:39.37 | neoalex | :)))) |
| 03:39.41 | pigpen | Currently running Asterisk 1.4.2, Zap 1.4.1 in Real Time with Postgres. |
| 03:40.08 | pigpen | Talk about a challenge. |
| 03:40.19 | *** join/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:40.50 | neoalex | I should get a few test phones as soon as my boss wants me to start working on his pbx |
| 03:40.57 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:41.28 | aptura | btw there are gsm cards on the market that will bridge cell phones yet will not interfear with the subscribers cell service? |
| 03:41.29 | pigpen | I know it has been helpful here...I am moving a customer to the same setup this week...but they only have 200 polycom phones... |
| 03:41.31 | neoalex | currently I have a WRTP54g with regular cordless phones pluged in |
| 03:41.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:42.01 | pigpen | neoalex, I won't tell you what I am running...you would be Jellous. |
| 03:42.46 | neoalex | no I wouldn't shoot |
| 03:43.30 | voltagex | what would cause asterisk to not even attempt to register an IAX trunk? |
| 03:49.25 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:49.33 | shmaltz | anybody here done any fax to email with asterisk? |
| 03:51.32 | pigpen | shmaltz, yes...(as I hide) |
| 03:51.43 | shmaltz | pigpen, what solution did you use? |
| 03:52.30 | pigpen | Comming from the telco: pri - Digium DS1 card- iaxmodem - hylafax - generate a pdf - deliver to email |
| 03:52.34 | *** join/#asterisk bmg505 (n=leon@c1-218-5.rndf.isadsl.co.za) |
| 03:55.46 | pigpen | The only issue I had was that every few days, the iaxmodem would loose it's registration...so I setup a nightly job to knock all the fax parts down and bring it all back up... |
| 03:55.54 | pigpen | maybe takes about 30 seconds.. |
| 03:57.21 | shmaltz | pigpen, is there any other solution? |
| 03:57.36 | aptura | shmaltz not yet. |
| 03:58.07 | shmaltz | aptura, pigpen, how long do you think that for someone as inteligent as myself it would take to setup the first time? |
| 03:59.32 | Qwell | DS1 card? |
| 03:59.53 | pigpen | it took me 6 hours to have all the bugs worked out... |
| 04:01.50 | pigpen | Good evening Qwell. |
| 04:02.31 | pigpen | and fyi, the fastest you will get with a PRI with the above noted config, is 14.4... |
| 04:02.39 | pigpen | otherwise, expect 9600 |
| 04:04.24 | infinity1 | I worte a call manager program, but I have no way to execute the AGI to process the next step if asterisk doesn't connect the call because if the call doesn't connect, the agi doesn't run. |
| 04:04.31 | infinity1 | Is there a work-around? |
| 04:05.00 | ManxPower | infinity1: use a timeout |
| 04:05.17 | infinity1 | ManxPower: can you explain? |
| 04:05.45 | pigpen | Set(timeout=x) , I think.... |
| 04:06.26 | infinity1 | i drop a call file into outgoing, the AGI executes only if the call is answered. I'm not sure how timeout fits. |
| 04:06.31 | pigpen | sorry: Set(TIMEOUT(response)=10) |
| 04:06.47 | pigpen | I think this is what he was referring to. |
| 04:08.17 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:08.19 | infinity1 | Hmmm... that only works in extensions file. I'm in AGI land. |
| 04:08.20 | *** join/#asterisk shmaltz (n=mybox@ool-4571eef9.dyn.optonline.net) |
| 04:11.19 | shmaltz | what is rxfax? |
| 04:13.34 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 04:17.01 | pigpen | it is a basic fax recieve. |
| 04:17.09 | aptura | ManxPower do you know what are the typical * compile times with varios intel and amd cpus? I have a fairly old system and its taking a good amount of time to compile |
| 04:17.17 | pigpen | With my setup I have 200 DID |
| 04:17.25 | aptura | need to speed it up a bit :) |
| 04:17.27 | pigpen | With my setup I have 200 DID's going to 200 different emails. |
| 04:17.34 | aptura | man |
| 04:17.50 | aptura | 200 through which wholesaler? |
| 04:18.01 | pigpen | Oh...with 12 IAX modems. |
| 04:18.07 | pigpen | inbound and outbound. |
| 04:18.22 | pigpen | yes...faxing from the desktop, windows and Mac clients. |
| 04:18.41 | pigpen | for that matter linux clients would be easy...just none at this time. |
| 04:19.55 | pigpen | this customer has 500 did's. |
| 04:19.58 | pigpen | I have only 400. :) |
| 04:20.45 | shmaltz | pigpen, and rxfax can't do that? |
| 04:20.52 | pigpen | They are an Episcopal School...ATT gives them whatever they want. |
| 04:21.02 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:21.31 | pigpen | it is a basic fax receive. I would have to write such a back end to it, it wasn't worth my time as hylafax is geared to the enterprise. |
| 04:22.10 | pigpen | don't get me wrong...hylafax's documentation sucks. |
| 04:23.36 | shmaltz | pigpen, how do you send using hylafax? |
| 04:24.48 | jql | heh. I mostly ignored the docs |
| 04:25.58 | pigpen | jql, yeah..I think I had my head up my ass a few times... |
| 04:26.12 | pigpen | shmaltz, hylafax clients. |
| 04:26.26 | shmaltz | pigpen, thanks |
| 04:26.37 | jql | it wasn't that bad skimming the source for what stuff did |
| 04:26.45 | jql | the variables and functions aren't badly named |
| 04:27.08 | pigpen | I think I got some doc's that were outdated. |
| 04:27.21 | pigpen | but it all makes sense after I got it going. |
| 04:28.16 | shmaltz | ever tried this: |
| 04:28.17 | shmaltz | http://asterfax.sourceforge.net/ |
| 04:28.44 | jql | sniffed at it. put it away wet |
| 04:30.59 | pigpen | java...yuck. |
| 04:31.01 | jql | spandsp is great stuff, but I enjoy having it outside my asterisk process |
| 04:33.26 | shmaltz | the server needs java? |
| 04:34.03 | pigpen | "For the technically minded AsterFax is written in pure Java and utilizes the Asterisk Manager API to submit jobs. " |
| 04:34.44 | pigpen | hylafax is solid. |
| 04:36.19 | CtRiX | which will be outdated soon ... |
| 04:36.34 | jql | I don't mind using the hylaforks |
| 04:36.58 | *** join/#asterisk grandy (n=chatzill@c-69-181-153-165.hsd1.ca.comcast.net) |
| 04:37.08 | CtRiX | hyla${whatever} will be outdated |
| 04:37.20 | pigpen | as long as something better comes along. |
| 04:37.24 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-145-247.ny325.east.verizon.net) |
| 04:37.34 | grandy | hello... just realized that my ubuntu asterisk package lacks ilbc... can anyone help me figure out how to install ilbc? I can't seem to figure out how to do this... it doesn't seem to be a debian/ubuntu package... |
| 04:37.44 | CtRiX | which is about 2 months, i think |
| 04:37.47 | jql | well, considering how fax technology is compatible with ye olde 2400bps, that doesn't seem like a frightening prospect |
| 04:37.57 | shmaltz | nice photo of the prince: |
| 04:37.58 | shmaltz | http://drudgereport.com/sun.jpg |
| 04:40.06 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 04:48.58 | grandy | hello... just realized that my ubuntu asterisk package lacks ilbc... can anyone help me figure out how to install ilbc? I can't seem to figure out how to do this... it doesn't seem to be a debian/ubuntu package... |
| 04:49.15 | Qwell | grandy: You don't need to repeat |
| 04:49.26 | Qwell | if there isn't a package available, you'll need to build it from source |
| 04:49.32 | grandy | thanks |
| 04:49.41 | *** join/#asterisk welby (i=welby@gateway/tor/x-6b97ad8337b0b0b5) |
| 04:54.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:55.56 | voltagex | is it possible for asterisk to use a fax/voice modem connected to a real telephone line? |
| 04:56.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:56.17 | aptura | no |
| 04:56.25 | voltagex | :( |
| 04:56.54 | aptura | digium was partly to be paid for there work buy one of there cards or go pure voip. |
| 04:57.04 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 04:57.44 | Strom_C | there is a really cheapo card you can get, but it sounds like sheer hell |
| 04:58.30 | voltagex | guess what Strom, I can call you on FWD now. all working |
| 04:58.46 | Strom_C | awesome. |
| 04:58.57 | voltagex | what was your number again? |
| 05:00.24 | Qwell | Strom_C: what does sheer hell sound like, exactly? |
| 05:00.40 | Strom_C | qwell: the x100p clone card, obviously ;) |
| 05:00.44 | Qwell | ;) |
| 05:00.44 | Strom_C | voltagex: 95690 |
| 05:01.14 | lokkju_wrk | I'm looking for someone that has an active vonage motorola vt1005v ATA - I just need you to either run a tftp command, or to give me the MAC address of your unit - I am attempting to provide some unlock methods for it, and to do so I need to get a new formware image. someone want to help me? |
| 05:01.30 | Qwell | lokkju_wrk: Your unit doesn't have a mac address? |
| 05:01.44 | lokkju_wrk | Qwell, has to be an *active* unit |
| 05:01.53 | lokkju_wrk | Qwell, as in, have service |
| 05:01.59 | Qwell | why would anybody here have Vonage? |
| 05:02.04 | Qwell | I mean...really |
| 05:02.08 | lokkju_wrk | Qwell, otherwise, vonage does not serve up a firmware image from their servers |
| 05:02.39 | lokkju_wrk | Qwell, there is always the outside chance... someone who is switching away from vonage, or something |
| 05:03.06 | Qwell | might as well ask people in #windows for their kernel config |
| 05:03.11 | lokkju_wrk | rofl |
| 05:03.15 | aptura | :) |
| 05:03.23 | voltagex | just plugging a real phone into my ata |
| 05:03.52 | aptura | vonage noc is located in wv? |
| 05:04.31 | denon | aptura: oh? I thought it was in india |
| 05:04.37 | aptura | LOL |
| 05:04.37 | jql | s/''/ |
| 05:05.41 | Qwell | bed time |
| 05:05.57 | denon | g'nite Qwell |
| 05:09.14 | voltagex | i think i bought the worst phone evar |
| 05:09.26 | denon | so return it |
| 05:09.32 | MooingLemur | there are lots of bad voip phones :P |
| 05:09.42 | voltagex | nah, real phone for ata |
| 05:12.00 | aptura | this compile is taking all night |
| 05:14.54 | voltagex | is asterisk meant to be providing a dialtone by default? |
| 05:18.57 | *** join/#asterisk patrickv0x (n=patrick@64.235.249.83) |
| 05:19.00 | patrickv0x | does anyone know if there is a sugarcrm+asterisk integration out there ? |
| 05:25.04 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 05:25.28 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:25.39 | yidiyuehan | hi everyone, how i can retrieve the caller id from PSTN properly? |
| 05:25.45 | voltagex | <Strom_C> voltagex: 95690 << getting 404 on that number from fwd |
| 05:26.13 | CtRiX | patrickv0x, trixbox |
| 05:26.40 | yidiyuehan | dialparties.agi: Caller ID name is 'zap1' number is '4521', everytime i call the server PSTN line, the caller id number will only show my zap extension number instead of my cell phone number. |
| 05:26.57 | voltagex | no, it's voip |
| 05:26.59 | voltagex | oops |
| 05:27.17 | voltagex | someone thought fwd was dialup |
| 05:27.23 | lokkju_wrk | patrickv0x, trixbox |
| 05:27.35 | lokkju_wrk | oops, sorry CtRiX |
| 05:29.08 | yidiyuehan | anyone knows the problem? |
| 05:32.38 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 05:33.36 | ManxPower | ~feepbx |
| 05:33.44 | ManxPower | ~freepbx |
| 05:33.51 | jbot | it has been said that freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 05:35.15 | ManxPower | ~amp |
| 05:35.17 | jbot | extra, extra, read all about it, amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 05:35.23 | ManxPower | ~openpbx |
| 05:35.24 | jbot | openpbx is probably something that started off as an asterisk fork, but is more of a rewrite of the internals and all good old GPL instead of the split licence stuff in asterisk. see http://openpbx.org/ for more info, or join #openpbx |
| 05:36.20 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 05:37.08 | ManxPower | yidiyuehan: listen to jbot, he knows what he is talking about. |
| 05:37.39 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 05:38.04 | yidiyuehan | ok, thanks bro. |
| 05:38.48 | yidiyuehan | in fact i asked them for this but it seems everyone is busy this time and i posted in the forum,unluckily no one answers me. ;-( |
| 05:39.43 | ManxPower | yidiyuehan: we really can't support those things here. |
| 05:45.35 | yidiyuehan | well, ok. i fully understand. |
| 05:45.52 | yidiyuehan | thanks although bro ;-) |
| 05:49.50 | *** join/#asterisk alomelo (n=wyginwys@85.105.204.40) |
| 05:58.46 | voltagex | how can I convert dialplan instructions for trixbox to something I can use with asterisk |
| 06:00.33 | voltagex | no see my provider thinks everyone uses trixbox |
| 06:00.57 | sbingner | either way they're the ones who would know how to get a real config out of your trixbox mess |
| 06:01.23 | voltagex | my trixbox has been wiped |
| 06:01.28 | voltagex | asterisk only on this box now |
| 06:01.35 | pigpen | what were they smoking when they wrote that...? |
| 06:02.00 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 06:03.06 | voltagex | well I need a dialplan for 02 and 03xxxxxxxx |
| 06:03.35 | sbingner | voltagex, have you looked at extensions.conf.sample? |
| 06:03.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:04.14 | asteriskguy | also try to look into users.conf file voltagex |
| 06:04.32 | asteriskguy | and also sip-additional.conf |
| 06:04.49 | asteriskguy | i think trixbox puts users in those files also |
| 06:05.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:05.27 | voltagex | i have NO trixbox stuff on this machine now |
| 06:05.33 | voltagex | it was format/reinstall |
| 06:06.12 | asteriskguy | then what do you mean by convert dialplan instructions for trixbox to *? |
| 06:06.35 | voltagex | the instructions listed on my providers page are trixbox |
| 06:06.48 | pigpen | you mean a translator? |
| 06:06.56 | voltagex | sort of |
| 06:07.10 | asteriskguy | oohh....you mean like how to use freepbx or amp and configure for users and stuff? |
| 06:07.18 | pigpen | start from scratch, you will be better off. |
| 06:07.20 | asteriskguy | you can install freepbx on your asterisk box |
| 06:07.25 | voltagex | ffs |
| 06:07.41 | asteriskguy | but Digium does not encourage using freepbx either |
| 06:07.48 | voltagex | I only have trixbox instructions for dialplans, I need * |
| 06:07.59 | asteriskguy | look up the book |
| 06:08.12 | asteriskguy | Asterisk - The future of telephony |
| 06:08.23 | voltagex | https://members.koalatelecom.com.au/node/75 |
| 06:08.35 | voltagex | yes yes I have that. overly complicated |
| 06:08.53 | asteriskguy | I know, there's another book called switching to voip |
| 06:09.04 | asteriskguy | there's also two more books coming out on asterisk pretty soon |
| 06:09.09 | pigpen | ugh. |
| 06:09.17 | asteriskguy | checkout amazon for Asterisk Cookbook |
| 06:09.34 | asteriskguy | I think it will help you more because it's more up to date |
| 06:09.48 | *** join/#asterisk manopulus (n=manopulu@213.197.181.154) |
| 06:10.11 | sbingner | voltagex, if you're too lazy to learn how to make your dialplan you'll have to use a gui |
| 06:10.35 | voltagex | trying to find the info I need |
| 06:10.36 | pigpen | voltagex, they are giving you directions to setup the dialplan in freepbx, which can run on any asterisk box. |
| 06:10.41 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-153.mtco.com) |
| 06:10.45 | asteriskguy | voltagex - Also checkout voip-info.org, astrecipes.net, asterikast.com |
| 06:10.49 | voltagex | I don't want freepbx |
| 06:10.54 | pigpen | which I assure you...they were smoking something when they wrote it. |
| 06:10.55 | sbingner | voltagex, almost everything you need to know is shown in extensions.conf.sample |
| 06:11.31 | sbingner | and the other .sample files |
| 06:11.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:11.40 | asteriskguy | well sbingner, the sample files help some what but if he doesn't have the basic of * down then it could add confusion |
| 06:11.51 | asteriskguy | I also found the * for dummies really good |
| 06:12.05 | sbingner | asteriskguy, I guess -- I just looked at the .samples and went from there |
| 06:12.13 | asteriskguy | also, asteriskNOW is somewhat good if you use it for a small company |
| 06:12.45 | asteriskguy | but really, if you want to learn asterisk, go with command line and learn the configs by hand |
| 06:12.56 | asteriskguy | it'll help with troubleshooting later |
| 06:13.08 | asteriskguy | imho |
| 06:13.43 | asteriskguy | basically it's a lot of reading |
| 06:13.51 | asteriskguy | and trying things out |
| 06:14.11 | asteriskguy | it's good if you try them out wayyyy before you have to really implement them |
| 06:16.47 | asteriskguy | alright my * fellas. bed time....good night |
| 06:21.03 | grandy | hello... i'm compiling asterisk 1.4 and i can't figure out how to get configure to set my pidfile location |
| 06:21.25 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 06:22.47 | grandy | it looks like ASTVARRUN is set properly, but defaults.h has the wrong value and the compiled program has the wrong value... any ideas? |
| 06:25.26 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 06:27.13 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 06:36.45 | pfn | hmm, so how much work is it to migrate from asterisk 1.0 to 1.4, heh |
| 06:37.47 | infinity1 | hmm ..thats pretty old, prolly not too dificult though |
| 06:38.17 | pfn | yeah, I haven't been keeping up |
| 06:38.26 | pfn | ares*CLI> show version |
| 06:38.26 | pfn | Asterisk CVS-v1-0-12/03/05-13:01:54 built by pfnguyen@ares.gofti.com on a i686 running Linux |
| 06:38.41 | *** part/#asterisk CVirus (n=GoD@196.205.193.189) |
| 06:39.04 | *** part/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:43.45 | infinity1 | I worte a call manager program, but I have no way to execute the AGI to process the next step if asterisk doesn't connect the call because if the call doesn't connect, the agi doesn't run. |
| 06:43.49 | infinity1 | Is there a work-around? |
| 06:45.11 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 06:46.04 | brian | infernix: huh |
| 06:46.07 | brian | infinity1: huh |
| 06:47.18 | voltagex | brian: i think he means can you still run an agi if a call doesn't connect |
| 06:47.24 | brian | deadagi? |
| 06:47.41 | pfn | make the agi run first? |
| 06:47.52 | brian | does he want deadagi |
| 06:48.45 | infinity1 | basically, the agi script needs to run even if the person doesn't answer the phone |
| 06:48.53 | infinity1 | deadagi doesn't run if the phone isnt picked up |
| 06:48.54 | *** join/#asterisk vishmaker (n=vishmake@202.83.33.94) |
| 06:49.01 | brian | infinity1: Have the AGI call the person |
| 06:49.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:49.23 | infinity1 | brian: i was thinking that, but how do you make an agi call the person? i've tired that without success. |
| 06:49.33 | infinity1 | er tried. |
| 06:50.24 | brian | umm |
| 06:50.35 | brian | theres a way to execute the same commands |
| 06:50.41 | brian | that you would execute in extensions.conf |
| 06:51.33 | brian | Like I have something like this: agi.execute('MeetMe', roomNumber + '|1dqxX') |
| 06:51.44 | infinity1 | hmm |
| 06:52.02 | brian | it's the EXECUTE command |
| 06:52.07 | infinity1 | i know you can execute commands inside agi, but i can't get the agi to run on an unconnected call |
| 06:52.27 | infinity1 | where does agi.execute get put? in extensions or what? |
| 06:52.34 | brian | no |
| 06:52.40 | brian | thats from my code in Python |
| 06:52.58 | infinity1 | i'm using perm. hmm ..so you're executing another agi from an agi in python? |
| 06:52.59 | infinity1 | heh |
| 06:53.03 | infinity1 | er perl here |
| 06:53.30 | [hC] | damn you cisco 7940 why wont you factory reset!@# |
| 06:53.48 | infinity1 | i just found something! ...call files support a paramater called "ARCHIVE" ..that might be an answer |
| 06:53.59 | brian | infinity1 |
| 06:54.03 | brian | look at DeadAGI |
| 06:54.12 | infinity1 | brian: deadagi doesn't work either :) |
| 06:54.18 | brian | maybe FastAGI? |
| 06:54.28 | infinity1 | duno what fastagi is.h he |
| 06:58.09 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 06:58.25 | brian | persistent AGI |
| 06:58.39 | infinity1 | well, if i do call archiveing, it looks like i can figure out the success/failure of calls based on the files |
| 06:58.43 | infinity1 | persistenat agi? Whats that?? |
| 06:59.13 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 06:59.42 | infinity1 | brian: are you reading bugs threads full of crazy ideas? |
| 06:59.43 | infinity1 | hehe |
| 07:00.12 | brian | n o |
| 07:00.14 | brian | i use fastagi |
| 07:10.24 | *** join/#asterisk gordonjcp (n=gordonjc@cpc1-broo2-0-0-cust991.renf.cable.ntl.com) |
| 07:10.26 | infinity1 | so call files support archive in 1.4. hmm... looks like i have to upgrade! |
| 07:11.14 | *** join/#asterisk Turt|e (n=danny@0x55532532.adsl.cybercity.dk) |
| 07:17.50 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 07:19.08 | FlatFoot | morning all |
| 07:21.39 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 07:22.10 | voltagex | is it possible to have an IVR completely using Asterisk? I don't feel like recording, and it's nothing important so people can speak to Mr Roboto. |
| 07:25.32 | gordonjcp | if the samples are already there, yes |
| 07:25.50 | Strom_C | or you could use cepstral |
| 07:25.52 | gordonjcp | think you can use festival tts too |
| 07:27.48 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
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| 07:37.31 | *** join/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 07:37.38 | voltagex | err, flaky connection today |
| 07:37.50 | voltagex | Strom_C: did you get my message about your number? |
| 07:38.06 | voltagex | I can call the FWD test numbers just fine, can't call yours :/ |
| 07:39.37 | Strom_C | *shrug* worked last night |
| 07:39.45 | Strom_C | anyway, better to dial my box directly |
| 07:40.04 | voltagex | ok, I have to set up an extension or trunk for that? |
| 07:41.00 | Strom_C | yeah |
| 07:41.09 | Strom_C | set up an extension; you don't need an iax.conf entry |
| 07:41.10 | voltagex | trunk I'm guessing |
| 07:41.14 | voltagex | oh |
| 07:41.21 | Strom_C | look at stromcarlson.com for the address |
| 07:48.15 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
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| 08:18.39 | *** part/#asterisk testbaker (n=Cazybake@125.18.165.17) |
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| 08:32.23 | gordonjcp | what's the highest sample rate asterisk will deal with for conferences? |
| 08:32.24 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 08:33.59 | fbffff | that's a strange question.. |
| 08:34.11 | gordonjcp | what's strange about it? |
| 08:34.54 | fbffff | i'm not sure i can answer it, what do you mean my sample rate? are you looking for a specific codec. .are you trying to record it.. i mean.. it can accept any codec in that asterisk supports |
| 08:35.05 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 08:35.10 | voltagex | Strom_C: you still around? |
| 08:35.24 | gordonjcp | fbffff: what we want to do is use asterisk to set up a conference for recording a podcast |
| 08:35.45 | fbffff | it converts everything that comes in to 16bit signed linear mixes it and then sends it out to the peers in the format the calls are negoiated to be |
| 08:35.59 | fbffff | ah |
| 08:36.25 | gordonjcp | fbffff: yes, but at what sample rate? All the codecs I've tried seem to be kind of geared to 8kHz |
| 08:36.32 | gordonjcp | I'm keen to try using speex ultrawideband |
| 08:36.34 | *** join/#asterisk nothinman (n=martin@ast.advance-internet.com) |
| 08:36.46 | fbffff | ya, slin would be the best quality you could get |
| 08:37.08 | fbffff | just putting it in a raw wav file and not a gsm-wav file |
| 08:37.34 | gordonjcp | but the sample rate is too low |
| 08:38.02 | fbffff | its not 16khz with 44100samples per second? |
| 08:38.25 | gordonjcp | it doesn't seem to be internally |
| 08:38.36 | gordonjcp | maybe that's down to the client I'm using |
| 08:39.04 | fbffff | ah i see that format_wav only will write 8khz |
| 08:40.00 | fbffff | i don't think there is any method at the moment to write speex data to a file |
| 08:41.21 | gordonjcp | wav would be fine, if it was at a higher bitrate |
| 08:41.29 | fbffff | ya |
| 08:41.36 | gordonjcp | in fact, I could set up another client and have that dump the audio to a wav |
| 08:41.42 | fbffff | one sec im glancing at the code so i can respond with correct information ;) |
| 08:41.54 | gordonjcp | thanks ;-) |
| 08:43.23 | voltagex | anyone else using FWD right now? |
| 08:43.42 | *** join/#asterisk uwe (n=uwe1@dogbert.palnet.com) |
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| 09:05.18 | JT | gordonjcp: asterisk is 8kHz |
| 09:05.43 | Vec | I have a problem where the Dial cmd is (sometimes) still indicating rining to the person who dialed the call even after the call has been answered, it seems it only happens when dialing from a SIP phone to an IAX phone/trunk, any ideas ? |
| 09:06.09 | voltagex | JT, everything's (almost) working here! |
| 09:07.09 | *** join/#asterisk Fieldy (i=iWlHQKDR@gentoo/contributor/Fieldy) |
| 09:07.32 | gordonjcp | JT: 8kHz and nothing else? |
| 09:13.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:19.52 | gordonjcp | hello Frogzoo |
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| 09:44.00 | JT | gordonjcp: yes pretty much |
| 09:44.19 | gordonjcp | hrm |
| 09:44.40 | gordonjcp | any recommendations for something that will conference speex ultra-wideband? |
| 09:45.36 | JT | probably freeswitch or yate |
| 09:45.40 | JT | but freeswitch isn't a pbx |
| 09:45.50 | gordonjcp | that's ok, I don't need a pbx for this really |
| 09:45.51 | JT | so it will be a fair bit more effort to make it do it :) |
| 09:46.02 | JT | you can make a pbx with freeswitch... |
| 09:46.30 | gordonjcp | I'm not fussed about running more services |
| 09:46.38 | gordonjcp | asterisk does what it does, and does it well ;-) |
| 09:50.38 | *** join/#asterisk DoodleHu (n=doodle@52.9.ygomi.net) |
| 09:50.50 | DoodleHu | Hi! |
| 09:52.10 | DoodleHu | I have a small patch for the RTP code of Asterisk. Could someone please enlighten me who to contact with it? What is the "official" way for patches? |
| 09:52.35 | *** join/#asterisk michael-i (n=michael-@141.41.40.189) |
| 09:52.55 | JT | bugs.digium.com |
| 09:53.15 | DoodleHu | thanks! |
| 09:53.49 | *** join/#asterisk tmjb (n=tmjb@195.252.125.18) |
| 09:54.13 | *** join/#asterisk jubei_ (n=Stormtro@147.27.47.114) |
| 09:54.37 | jubei_ | guys I wanna dial the number ..say.. 30+${EXTEN} , how do I say that in the dialplan? |
| 09:54.48 | tmjb | is it posible to dialin into asterisk get console via analog TDM400 ? |
| 09:55.17 | michael-i | i just compiled 1.4.2 with apps, cdr, codecs, formats, funcs, pbx and res embedded into the core. now none of these things work! :) must I put something in modules.conf or is there a gotcha I missed? |
| 09:56.05 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 09:56.43 | *** join/#asterisk robwi (n=rowi@slave.rixport80.se) |
| 09:57.35 | JT | jubei_: explain? |
| 09:58.20 | jubei_ | eer.. exten => 300,1 Dial(SIP/30+${EXTEN}) ? |
| 09:58.38 | jubei_ | i wanna prefix a number before the thing that is to be dialed |
| 10:00.03 | JT | ok so what's wrong with exten => 300,1,Dial(SIP/<sip.conf entry>/30${EXTEN}) |
| 10:03.03 | *** join/#asterisk r0d3nt (n=RatMan@punk.valuetel.net) |
| 10:04.03 | *** join/#asterisk fbffff (n=fbffff@rrcs-24-153-196-185.sw.biz.rr.com) |
| 10:05.27 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 10:08.19 | jubei_ | JT, wouldit work with Dial(SIP/exten@sippeer) ? |
| 10:10.18 | JT | that's not the best practice way to dial |
| 10:10.18 | JT | unless your dialling without sip.conf entry, i have no idea why you'd use that form |
| 10:17.14 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 10:17.15 | jubei_ | JT: coz.. when I first configured asterisk that was the only way to dial :D |
| 10:17.33 | jubei_ | JT: thanks anyway, your way worked :D |
| 10:18.35 | tmjb | Is it posible to make one analog phone line on TDM400 work as a modem so i can use it as dial in line for administration of asterisk tnx :) |
| 10:19.36 | *** join/#asterisk lorinc (n=ang@pool-6233.adsl.interware.hu) |
| 10:32.14 | *** join/#asterisk advorak (n=advorak@c-24-7-54-97.hsd1.ca.comcast.net) |
| 10:33.36 | *** join/#asterisk shinux__ (n=shinux@196.220.24.41) |
| 10:35.28 | drako | morning |
| 10:35.47 | voltagex | evening |
| 10:37.09 | *** join/#asterisk shinux__ (n=shinux@196.220.24.41) |
| 10:39.35 | drako | voltagex, can I use zaptel 1.4 with asterisk 1.2? |
| 10:39.41 | voltagex | no idea |
| 10:39.44 | voltagex | I'm new to * |
| 10:46.26 | stoffell | drako, i'm sure that's not recommended ;) |
| 10:47.09 | drako | stoffell, are you sure? |
| 10:47.57 | stoffell | no other way to be sure then to test it... but I assume it's not the best thing to do .. |
| 10:48.29 | JT | drako: no, it can't be done. |
| 10:48.56 | drako | hmm |
| 10:49.20 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 10:49.30 | drako | The Server keep crashing (whole system lock up) when I activate echocanel on zaptel.conf |
| 10:49.36 | drako | I need a solution. |
| 10:50.14 | stoffell | upgrade to latest ? |
| 10:51.15 | drako | zaptel or asterisk? |
| 10:51.41 | JT | drako: using zaptel 1.4 with asterisk 1.2? |
| 10:51.42 | stoffell | both because they are related.. |
| 10:55.55 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 10:55.57 | zeeesh | hi |
| 10:56.04 | drako | JT, im using right now asterisk 1.2.14 with zaptel 1.2.16 |
| 10:59.11 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 11:01.03 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:01.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:06.09 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 11:06.19 | *** part/#asterisk DoodleHu (n=doodle@52.9.ygomi.net) |
| 11:06.39 | voltagex | how can I set an extension as unavailable so my phone doesn't wake me while I'm sleeping? |
| 11:08.02 | *** join/#asterisk friedrich| (n=friedric@e177249061.adsl.alicedsl.de) |
| 11:09.41 | creativx | do not disturb |
| 11:10.01 | creativx | ..depending on what phone you are calling |
| 11:10.23 | voltagex | creativx: there's no button on the phone to do that, so there's an asterisk setting somewhere? |
| 11:10.55 | creativx | not that i can think of |
| 11:11.05 | creativx | but you could always make something |
| 11:11.28 | creativx | set up an extension that sets a variable.. ${exten} is NA .. then before dial() check that var |
| 11:11.47 | creativx | and then you can call a number to set your status |
| 11:11.53 | creativx | probably other ways to do it too |
| 11:11.54 | creativx | :> |
| 11:12.42 | voltagex | hmm |
| 11:13.28 | creativx | did you get the idea |
| 11:13.49 | voltagex | yeah, looking into doing something with python that checks the time of day. |
| 11:18.53 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 11:18.57 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 11:19.13 | creativx | try *CLI> show application GotoIfTime |
| 11:19.27 | creativx | instead of reinventing the wheel |
| 11:19.27 | creativx | :> |
| 11:19.33 | skirmisha | guys |
| 11:19.50 | skirmisha | any idea how can i make asterisk to work as remote voicemail server |
| 11:22.11 | skirmisha | ? |
| 11:22.22 | JT | ???? |
| 11:23.25 | skirmisha | i want to send voicemail msg to remote asterisk server |
| 11:23.43 | skirmisha | so i need to configure local asterisk voicemail server to send msg to remote ast server |
| 11:23.43 | JT | yes i saw that already |
| 11:23.51 | JT | it has not disappeared from my screen |
| 11:24.01 | skirmisha | is there a patch or something like that |
| 11:24.18 | skirmisha | is it stable and easy to configure |
| 11:24.35 | skirmisha | this one |
| 11:24.35 | skirmisha | http://bugs.digium.com/view.php?id=4371 |
| 11:25.14 | skirmisha | or can i do it via dialplan |
| 11:25.31 | skirmisha | means astrisk to dial remote voicemail server |
| 11:26.47 | voltagex | when programming AGI with python, which module do you guys recommend? |
| 11:27.48 | skirmisha | JT any ideas |
| 11:28.06 | Gido-E | skirmisha it is possible with asterisk. |
| 11:28.15 | JT | dunno, you could probably just do something in the dialplan |
| 11:29.13 | skirmisha | Gido-E what is possible |
| 11:29.31 | skirmisha | with dialplan |
| 11:29.35 | JT | what you want to do |
| 11:29.35 | JT | gees |
| 11:29.41 | JT | stop being so dense |
| 11:30.15 | skirmisha | i want to have few asterisk servers but only one to serve all voicemail msg |
| 11:30.28 | JT | yes |
| 11:30.59 | skirmisha | so all others must send voicemail to the central voicemail server |
| 11:31.00 | skirmisha | can i use Voicemail with ip of remote server |
| 11:31.00 | skirmisha | in dialplan |
| 11:31.16 | JT | yes, but you will have to work out how |
| 11:31.41 | skirmisha | or i need a patch for that |
| 11:32.18 | JT | i don't think you need a patch |
| 11:32.20 | JT | best way to find out is try |
| 11:32.36 | JT | instead of wasting time asking in here repetatively |
| 11:32.43 | skirmisha | that's good |
| 11:32.54 | *** join/#asterisk saftsack (n=oliver@pd9e048b6.dip.t-dialin.net) |
| 11:33.12 | skirmisha | yes i know, just asking if someone already did that |
| 11:34.49 | Gido-E | skirmisha the question is verry global. |
| 11:39.36 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 11:41.19 | skirmisha | i think i found something |
| 11:41.26 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:44.10 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:44.31 | *** join/#asterisk shinux_ (n=shinux@196.220.28.199) |
| 11:52.35 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-190655.home.otenet.gr) |
| 11:56.39 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 11:56.57 | nasls_lsa | Beronet BN4S0 works fine with asterisk ? should I order it ? |
| 11:57.01 | nasls_lsa | anyone tried before ? |
| 12:04.10 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 12:07.54 | JT | nasls_lsa: i don't see why not |
| 12:07.59 | codefreeze | michael-i: I had somebody yesterday complaining about that... |
| 12:08.01 | JT | nasls_lsa: i prefer junghanns.net myself |
| 12:08.08 | JT | but very similar units |
| 12:08.12 | codefreeze | michael-i: Still having probs? |
| 12:08.51 | michael-i | codefreeze: yup :) I've set my experimenting back a bit and am now working with loadable modules again |
| 12:08.59 | nasls_lsa | thanks ;) |
| 12:09.18 | michael-i | codefreeze, really would like to get that working though since it saves me 400-500KB of precious space |
| 12:09.24 | JT | nasls_lsa: i also prefer the use of bristuff instead of misdn |
| 12:09.41 | nasls_lsa | what is bristuff ? |
| 12:10.11 | codefreeze | michael-i: What did it? Re-doing the menuselect stuff? |
| 12:10.12 | JT | a patchset that adds bri capabilities to asteris |
| 12:10.14 | *** join/#asterisk unice (n=tom_hens@port-83-236-223-18.static.qsc.de) |
| 12:10.16 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 12:10.17 | unice | hi |
| 12:10.36 | unice | which web frontend to asterisk has the coolest statistics? |
| 12:10.38 | michael-i | codefreeze: i just reset everything to loadable and it worked again of course |
| 12:10.53 | unice | by cool i mean best looking, most informative, professionally done |
| 12:10.57 | unice | :) |
| 12:11.11 | nasls_lsa | mmm .... I think after I get the card I'll ask you again for help ... of cource after I try my own :) |
| 12:11.43 | JT | nasls_lsa: junghanns wrote bristuff |
| 12:12.14 | nasls_lsa | I am from Greece , and here only that card I found for 4 ISDN lines :/ ... |
| 12:12.24 | JT | i see |
| 12:12.31 | JT | can't buy from overseas? |
| 12:13.35 | unice | octobri? |
| 12:13.35 | nasls_lsa | if one in a million have any problem , my cient wants in 24 hours his call center to works fine |
| 12:14.12 | JT | nasls_lsa: he wants the whole thing setup in 24 hours? that's ridiculous |
| 12:14.18 | JT | unice: what about it? |
| 12:15.03 | unice | well i just heard "junghanns" and "4 ISDN lines"... so i took a guess |
| 12:15.04 | nasls_lsa | no , se wants after I setup , and works fine |
| 12:15.13 | JT | octo means 8 |
| 12:15.27 | nasls_lsa | if ever has any problem ... even from thunder .. that I can replace any part in 24 hours |
| 12:15.48 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 12:16.04 | unice | the octobri connects 4 isdn lines, which gives you 8 channels |
| 12:16.08 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 12:16.10 | JT | unice: wrong |
| 12:16.20 | JT | it connects 8 isdn lines, which gives you 16 channels |
| 12:16.41 | unice | well, right for PRI |
| 12:16.54 | JT | "octo" "bri" - "eight BRIs" |
| 12:17.03 | JT | unice: right for PRI? huh |
| 12:18.27 | unice | well, we are running a octobri which connects to 2 PRI lines |
| 12:18.39 | JT | no, it doesn't |
| 12:18.46 | JT | now you're just making stuff up |
| 12:18.59 | JT | an octo bri has one purpose |
| 12:18.59 | *** join/#asterisk step_quasar (n=step_qua@250-171-114-200.fibertel.com.ar) |
| 12:19.02 | unice | err.. |
| 12:19.06 | JT | to connect EIGHT BRIs |
| 12:19.12 | unice | you are one of these experts, right? |
| 12:19.31 | JT | an expert that uses an octobri every day of the week, i guess you could say |
| 12:20.45 | unice | so do i :) |
| 12:20.53 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 12:21.24 | JT | well i say you're either mightily terminology confused |
| 12:21.29 | JT | or you're making stuff up |
| 12:21.33 | JT | so which is it? :) |
| 12:21.39 | unice | if you'd know a bit more 'bout isdn, you'd know that BRI is one possible type of isdn line |
| 12:21.42 | unice | not the only one |
| 12:22.06 | unice | you are free to use your octobri with 8 BRI |
| 12:22.10 | JT | i know that |
| 12:22.29 | JT | i know enough about isdn :) |
| 12:22.32 | unice | so as i said before, we are utilizing a octobri to connect to 2 PRI |
| 12:22.40 | JT | and i call bullshit |
| 12:22.48 | JT | because they're completely different L1 standards |
| 12:22.48 | unice | hehe |
| 12:22.52 | unice | you're a retard |
| 12:22.53 | JT | YOU CANNOT CONNECT THEM |
| 12:23.16 | JT | maybe you mean a DoubleE1 card? |
| 12:23.40 | JT | The octoBRI® 2.0 PCI ISDN brings powerful ISDN BRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. All 8 BRI ports can be configured for TE or NT mode individually by jumpers. This port configuration is detected by the driver automatically. |
| 12:23.55 | JT | weird it doesn't say PRI |
| 12:24.06 | dlynes_laptop | NO???!!!???!!! |
| 12:24.15 | JT | :( |
| 12:24.38 | dlynes_laptop | damn....I thought PRI and BRI were the same thing? :((( |
| 12:24.51 | JT | hehe |
| 12:24.52 | unice | well except a single letter |
| 12:24.57 | Makenshi | I got a pri ked up to an fxo here |
| 12:25.04 | dlynes_laptop | p, b, what's the difference? |
| 12:25.04 | Makenshi | *hooked |
| 12:25.10 | JT | and the fact that they're different L1 technologies |
| 12:25.28 | dlynes_laptop | Makenshi: you have a channel bank hooked up to a pri, and you have an fxo port on the channel bank |
| 12:25.32 | JT | a BRI is 2 * 64kbit/s B channels + 16kbi/s D channel |
| 12:25.36 | dlynes_laptop | Makenshi: you don't have an fxo hooked up to a pri :) |
| 12:25.38 | Makenshi | no i'm just being facetious |
| 12:25.47 | JT | Makenshi: i followed :) |
| 12:26.18 | JT | an E1 PRI is 30 * 64kbit/s B channels, + 1 64kbit/s D channel + 64kbit/s multiframe synch/alarms |
| 12:26.42 | JT | a T1 PRI is 23 * 64kbit/s C channels, + 1 64kbit/s D channel |
| 12:27.00 | JT | so please don't play the isdn brain game, unice :P |
| 12:27.06 | JT | to connect to a PRI or a BRI |
| 12:27.12 | JT | you need a totally different card |
| 12:27.19 | JT | i have not seen any combo cards yet |
| 12:27.25 | JT | there is no demand for them |
| 12:27.39 | JT | s/C channels/B channels/ |
| 12:27.59 | unice | ok i failed in terminology |
| 12:28.33 | JT | am i still a retard? :( |
| 12:29.07 | unice | we are connecting to BRI P2P (direct in dial) to a octobri |
| 12:29.22 | JT | right |
| 12:29.25 | unice | which makes 8 channels bound to 2 lines (which has two connections each) |
| 12:29.39 | unice | therefore, you are not a retard :) |
| 12:29.44 | JT | if it's only 1 BRI, it's 2 channels |
| 12:29.52 | JT | with 8 MSNs? |
| 12:30.02 | unice | no, 30 each |
| 12:30.11 | JT | ok |
| 12:30.31 | JT | not sure what you mean 8 channels bound to 2 lines |
| 12:31.33 | unice | well whats the international term for NTBA? |
| 12:31.45 | JT | NT1? |
| 12:32.29 | unice | we got four of em |
| 12:32.29 | JT | means you have 4 BRIs |
| 12:32.29 | unice | no |
| 12:32.29 | JT | 8 B channels |
| 12:32.29 | unice | yes |
| 12:32.29 | JT | which is 4 BRIs... |
| 12:32.29 | unice | ;) |
| 12:32.29 | unice | no |
| 12:32.45 | JT | yes, a BRI is one connection to the telco with 2 B channels and a D channel |
| 12:32.52 | JT | uses an NT1/NTBA |
| 12:33.00 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 12:33.13 | unice | as BRI doesnt allow direct inward dial to 30 msdn |
| 12:33.16 | unice | hehe, msn, even |
| 12:33.39 | JT | well it sounds like you have DIDs, so clearly it works |
| 12:34.07 | unice | hehe |
| 12:34.31 | unice | it's called "primaermultiplexanschluss" in german |
| 12:34.36 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 12:34.40 | unice | it's a crippled PRI, though |
| 12:34.49 | JT | i see |
| 12:34.51 | unice | 30 number did without the channels |
| 12:35.03 | JT | but it sounds like you are using BRIs at the L1 |
| 12:35.18 | unice | sort of :) |
| 12:35.34 | Ahrimanes | anyone noticed a break in subscriptions/hints between 1.2.14 and 1.2.17? when a sip phone is ringing in 1.2.14 show hints shows RINGING, while in 1.2.17 it shows InUse |
| 12:36.00 | JT | well yes, any MSNs or DIDs you have over the top of that are immaterial, they don't take away from it being a BRI |
| 12:37.24 | MrWup | i notice the tdb04b has a power socket |
| 12:37.28 | MrWup | does that need to be powered? |
| 12:37.31 | MrWup | or is the pwoer from PCI enough? |
| 12:37.37 | JT | if you have FXS ports |
| 12:37.45 | MrWup | mine is 4FXO |
| 12:37.52 | MrWup | no need for power? |
| 12:38.08 | unice | we recently purchased a ISDNguard from junghanns |
| 12:38.22 | JT | MrWup: nup |
| 12:38.26 | MrWup | thanks |
| 12:38.28 | MrWup | well |
| 12:38.31 | unice | works as expected so far |
| 12:38.33 | MrWup | got my asterisk installed |
| 12:38.33 | JT | is the isdnguard any good? |
| 12:38.37 | MrWup | got debian all hunky dorey |
| 12:38.40 | MrWup | got my card plugged in |
| 12:38.45 | unice | it's a bit expensive, but very well made |
| 12:38.48 | MrWup | now how on earth to make it answer calls |
| 12:38.52 | MrWup | the strife begins =] |
| 12:39.14 | unice | installation is quite easy if you manage to compile asterisk by hand |
| 12:39.31 | unice | current junghanns bristuff includes all whats needed |
| 12:40.29 | JT | MrWup: contexts... |
| 12:40.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:41.43 | drako | MrWup, a TDM400 with 4 FXO ports and Debian? |
| 12:42.55 | flenders | JT: still up? |
| 12:44.24 | JT | flenders: yep |
| 12:44.50 | flenders | how did it go? |
| 12:44.50 | JT | not bad |
| 12:45.18 | flenders | big headache? |
| 12:45.26 | JT | not too big |
| 12:45.32 | flenders | great mate! |
| 12:46.06 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:48.57 | unice | JT, btw, http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-d.tar.gz |
| 12:49.06 | JT | :) |
| 12:49.51 | unice | based on asterisk-1.2.17 now |
| 12:49.58 | JT | cool |
| 12:50.02 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 12:50.05 | unice | still needs patches on ubuntu |
| 12:50.59 | JT | howcome? |
| 12:52.13 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:52.20 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 12:53.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:55.36 | *** join/#asterisk Corydon76-home (i=eleven@pdpc/supporter/sustaining/Corydon76-home) |
| 12:55.36 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:01.25 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 13:02.17 | unice | zaptel compilation fails, it doesnt detect the kernel sources |
| 13:02.25 | unice | so one has to set KSRC by hand |
| 13:02.53 | unice | due to some unknown reason, compilation of asterisk/channels/chan_zap fails due to some libgsm woes |
| 13:03.10 | unice | so one has to disable inclusion of libgsm in asterisk' makefile |
| 13:03.47 | unice | well i mention unknown and reasons in the same sentence |
| 13:03.51 | unice | :) |
| 13:07.02 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:07.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:07.22 | *** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net) |
| 13:09.20 | Katty | morning lovables. |
| 13:09.45 | *** join/#asterisk awk (n=awk@dsl-242-123-186.telkomadsl.co.za) |
| 13:09.48 | awk | ~pb |
| 13:09.50 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 13:10.15 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 13:10.33 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:10.51 | awk | big heeeeeelp! |
| 13:10.52 | awk | http://channels.debian.net/paste/5839. |
| 13:10.58 | awk | anyone ever encountered this |
| 13:11.30 | Katty | big help? |
| 13:11.34 | Katty | no, no i've never seen that before. |
| 13:12.07 | unice | me neither |
| 13:12.12 | unice | so you are using a g729 codec? |
| 13:12.16 | _DAW | interesting |
| 13:12.24 | awk | that is the largest flood ever |
| 13:12.29 | awk | yes, using g729 |
| 13:12.40 | puzzled | unice: those bristuff patches are huge. did you test them already? |
| 13:13.04 | awk | ok, what about this? |
| 13:13.05 | awk | Mar 27 15:11:25 WARNING[31977]: file.c:512 ast_openstream_full: File 4000 does not exist in any format |
| 13:13.08 | creativx | sigh.. how can i do call pickups over the manager interface and also set the answer-after variable? |
| 13:13.08 | awk | Mar 27 15:11:25 WARNING[31977]: file.c:824 ast_streamfile: Unable to open 4000 (format ulaw): No such file or directory |
| 13:13.12 | awk | the system, seems to have gone to sh*** |
| 13:13.13 | unice | $ uname -a ; asterisk -V |
| 13:13.15 | unice | Linux voip-01 2.6.15-27-server #1 SMP Sat Sep 16 02:57:21 UTC 2006 i686 GNU/Linux |
| 13:13.16 | unice | Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1y-b |
| 13:13.35 | awk | ye, i'm not to sure about bristuff, have they even patched that sip vuln ,yety ? |
| 13:13.38 | awk | yet ? |
| 13:13.51 | unice | building 1y-d now |
| 13:13.53 | *** join/#asterisk Fieldy (i=ZQzhi85s@gentoo/contributor/Fieldy) |
| 13:14.06 | unice | well the vuln is related to asterisk, not bristuff |
| 13:14.10 | puzzled | awk: sip and bristuff are two different things |
| 13:14.40 | awk | puzzled: well, sip is part of asterisk |
| 13:14.44 | awk | and you using version 1.2.14 |
| 13:14.58 | awk | and the patch is part of 1.2.17 |
| 13:15.05 | awk | unless there is a patch for 1.2.14 |
| 13:15.10 | awk | err 1.2.16 |
| 13:15.14 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 13:21.48 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:21.52 | blitzrage | morning all! |
| 13:22.03 | Katty | hey blitz |
| 13:22.07 | blitzrage | anyone know of a dialplan function or application that'll check for the existance of a file on the hard drive? |
| 13:22.11 | blitzrage | Katty: mew! |
| 13:22.21 | Katty | blitzrage: you're never going to believe what happened. |
| 13:22.30 | Katty | blitzrage: there was a shooting, about a block or two from where i live |
| 13:22.30 | blitzrage | Katty: what's that?! |
| 13:22.34 | blitzrage | jeebuz! |
| 13:22.39 | blitzrage | Chicago? |
| 13:22.49 | blitzrage | that just happened to Kristian Kielhofner the other day... |
| 13:22.54 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 13:22.56 | Katty | no, i don't live in chicago |
| 13:23.28 | awk | bonjour Katty |
| 13:23.40 | Katty | hi awk |
| 13:24.51 | awk | Katty: any idea what this is about |
| 13:24.53 | awk | 15:13 < awk> Mar 27 15:11:25 WARNING[31977]: file.c:512 ast_openstream_full: File 4000 does not exist in any format |
| 13:24.57 | awk | 15:13 < awk> Mar 27 15:11:25 WARNING[31977]: file.c:824 ast_streamfile: Unable to open 4000 (format ulaw): No such file or directory |
| 13:25.04 | awk | dam, that was supposed to be 2 lines |
| 13:25.22 | blitzrage | it was 2 lines |
| 13:25.24 | Katty | awk: no |
| 13:25.37 | blitzrage | awk: it means /var/lib/asterisk/sounds/4000.ulaw doesn't exist |
| 13:25.52 | blitzrage | or in any format |
| 13:25.54 | *** join/#asterisk joeTSUNAMI (n=joeTSUNA@72.243.118.59) |
| 13:25.59 | blitzrage | the WARNING message is pretty obvious |
| 13:26.01 | P4C0 | hello guys, I'm having problems with codec g729, I just bougth a license for one channel, i have 3 local soft sip phones, (alaw codec) an asterisk server, and a link to my voip provider (g729) the problem is that when I try to call from one softphone outside, just when the other peer answers, the serve drop the call with ast_channel_make_compatible: No path to translate I have canreinvite=no not sure why is this happening or how can I debug it |
| 13:26.47 | Katty | i don't think people realize how hard it is to read runon sentances. |
| 13:27.06 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:27.20 | awk | thanks blitzrage |
| 13:27.26 | awk | so would 4000 be a default sound |
| 13:27.29 | JT | really i thought it was easy maybe it was not but anyway perhaps i should fill up the buffer long enough so that ircd cuts my sentence i mean who needs breivity anyway |
| 13:27.31 | P4C0 | Katty, sorry |
| 13:27.34 | awk | or would that be a sound that is required by some script that doesn';t exsist |
| 13:28.09 | JT | the enter key would be useful as well as . |
| 13:29.31 | Katty | my eyes just want to skip big paragraphs like that. |
| 13:29.44 | Katty | P4C0: i have no solution for you. i've not dealt with that before. |
| 13:31.30 | *** join/#asterisk egypcio (n=egypcio@asterisk.tradein.com.br) |
| 13:32.51 | awk | blitzrage: so would this mean |
| 13:32.51 | awk | Mar 27 15:32:24 WARNING[4830]: file.c:512 ast_openstream_full: File rands does not exist in any format |
| 13:33.04 | awk | that there is no audio file rands available ? |
| 13:33.07 | blitzrage | yes |
| 13:33.20 | awk | thanks |
| 13:33.22 | blitzrage | rands.gsm | rands.ulaw | rands.ilbc, etc... |
| 13:33.46 | awk | now, this is a strange one |
| 13:33.51 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:34.04 | awk | has asterisk changed so much since 1.2.early version to 1.2.17 |
| 13:34.05 | awk | as I get this |
| 13:34.05 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 13:34.07 | awk | Mar 27 15:32:47 WARNING[4815]: chan_sip.c:9840 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"0413732755" <sip:0413732755@196.15.190.75>;tag=as2d502d32' |
| 13:34.21 | awk | why is it using the number as the sip account details |
| 13:34.34 | awk | 04137*@ip ? |
| 13:35.25 | awk | has the way sip.conf changed alot ? |
| 13:35.31 | awk | or any idea how I can identify this ? |
| 13:36.27 | P4C0 | what's the way to debug codec problems? |
| 13:36.55 | *** join/#asterisk mmartinn (n=martins@128.227.123.22) |
| 13:38.37 | unice | we are still using 1.2.14 but asterisk is not getting any traffic forwarded |
| 13:39.40 | *** join/#asterisk joeTSUNAMI (n=joeTSUNA@72.243.118.59) |
| 13:42.00 | awk | could mine be due to insecure=port,invite ? |
| 13:42.04 | awk | should I try set that |
| 13:45.14 | awk | how do I set fromuser to allow anyuser ? |
| 13:47.17 | awk | how do I match any number? |
| 13:48.03 | m-00kie | _X* unless that's changed since i last used asterisk :P |
| 13:48.23 | cpm | by pressing the any key |
| 13:50.54 | *** join/#asterisk mkl1525 (n=qwertz@61.207.27.217.static.versanetonline.de) |
| 13:52.48 | mkl1525 | Hi, is there a way to run some agi script when a queue agent accepts a call? |
| 13:55.29 | *** join/#asterisk pifiu-laptop (n=someone@216.5.79.1) |
| 13:55.34 | pifiu-laptop | morning everyone |
| 13:57.31 | pifiu-laptop | if i have a certain extension/did on * that i want to particularly play different MOH than the rest of the phones in the setup, how can i specify to use a certain file? |
| 14:05.07 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 14:06.46 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:06.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:07.36 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:11.09 | P4C0 | is it normal that when I'm making a call, and it's ringing (haven't picked it up yet) the 'show g729' command shows: 0/1 encoders/decoders of 1 licensed channels are currently in use ? |
| 14:12.24 | pifiu-laptop | if i have a certain extension/did on * that i want to particularly play different MOH than the rest of the phones in the setup, how can i specify to use a certain file? |
| 14:12.26 | pifiu-laptop | anyone know? |
| 14:15.59 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 14:16.04 | menfin | -fr |
| 14:16.11 | menfin | excuse me |
| 14:16.14 | menfin | hi all |
| 14:18.32 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 14:18.33 | menfin | i'm trying ton make running an asterisk server, with a basic configuration and some ekiga client |
| 14:18.36 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 14:18.41 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 14:19.01 | menfin | but something may be wrong because an authentication problem is here |
| 14:19.51 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:20.02 | menfin | i'm trying to register to localhost (this computer is asterisk server and ekiga client) |
| 14:20.10 | genz | Is outgoing callerid possible? Like I dial a number and my phone resolves to the name of the number I'm dialing? |
| 14:20.26 | P4C0 | if I have only one channel with g729 in the link with my voip provider, is it possible to use another codec when the g729 is busy? |
| 14:20.59 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 14:20.59 | *** mode/#asterisk [+o angler] by ChanServ |
| 14:21.26 | Mercestes | p4c0: Sure. |
| 14:22.43 | menfin | someone have any idea ? |
| 14:23.38 | bulle | menfin: set verbosity high and turn on debugging on sip ( as i assume you are using sip with ekiga ) and see what the error is about |
| 14:25.23 | menfin | bulle, Transmitting (no NAT) to 192.168.1.117:5062: |
| 14:25.23 | menfin | SIP/2.0 401 Unauthorized |
| 14:25.40 | menfin | and in /var/log/asterisk/messages |
| 14:25.50 | menfin | some bad password |
| 14:26.11 | menfin | but my password is so simple that I can't have a mistake |
| 14:26.43 | menfin | I have made ten verification |
| 14:27.23 | genz | ~pastebin |
| 14:27.29 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 14:27.29 | bulle | menfin: well, i would say you still have |
| 14:27.35 | bulle | menfin: perhaps put the password in the wrong place in sip.conf |
| 14:27.49 | bulle | menfin: used a space in the wrong place, or other similar problem |
| 14:28.35 | menfin | when i type a sip show users command i can see the password |
| 14:28.44 | menfin | and is it the good password |
| 14:29.18 | bulle | dunno then |
| 14:29.21 | menfin | but in the sip debug log i can read "digest algorithm=md5" |
| 14:29.55 | menfin | is it possible that my ekiga client send password as plaintext or something like this ? |
| 14:30.06 | menfin | or ekiga send an md5 hash |
| 14:30.18 | bulle | i dont know what ekiga sends, you will have to check that yourself |
| 14:30.30 | menfin | right |
| 14:31.04 | menfin | what does "dunno" mean ? |
| 14:31.10 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 14:31.13 | menfin | (I'm french" |
| 14:31.15 | menfin | ) |
| 14:31.36 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 14:32.36 | bulle | menfin: "i dont know" |
| 14:33.01 | menfin | thanks |
| 14:33.03 | nettie | Hi guys, I just put an hfc isdn card in my asterisk server.. I only used it with sip and an external carrier, now I would like to handle my ISDN line as well. I noticed there're a couple of projects around, anyone know which one works better please? I'm actually runnign asterisk 1.2.12.1 |
| 14:37.18 | menfin | is there any way to test sip authentication in the cli interface ? |
| 14:37.27 | aydiosmio | anyone know why wav files for * are supposed to be 16 bit samples (and not 8 bit like G711 PCM)? |
| 14:38.12 | *** join/#asterisk SuD (n=Ask@30.red-213-227-60.user.auna.net) |
| 14:38.41 | SuD | hi, is it possible to perform a Playback() on "calleD" user instead of "calleR"? |
| 14:39.49 | P4C0 | Mercestes, how? just adding the codec to allow? or is there a way to set priority? |
| 14:40.43 | *** join/#asterisk kovger (n=kovger@217.27.219.154) |
| 14:40.45 | Mercestes | p4c0: Yes on both accounts |
| 14:40.54 | coppice | aydiosmio: why would storing the wave files in G.711 make more sense? |
| 14:42.49 | pifiu-laptop | whats a half decent cheap provider that can provide a did and no canceallation fees? I want one just for testing purposes, nothing reliable really |
| 14:42.56 | pifiu-laptop | i will only use it a month or less |
| 14:43.01 | P4C0 | Mercestes, it seems to just drop the call when there's not transations left |
| 14:43.41 | Mercestes | p4c0: Should be able to disallow=all allow=g729,ulaw and if there are no g729 codecs left fall back to g711. |
| 14:43.49 | Mercestes | p4c0: If not then I would prepare a bug report. |
| 14:43.59 | Mercestes | I woul dthink that this would be the preferred behavior. |
| 14:44.10 | *** join/#asterisk Fieldy (i=mW5LDfB6@gentoo/contributor/Fieldy) |
| 14:44.21 | aydiosmio | coppice: I wasn't suggesting that really, I'm asking why wav files should be stored in 16 bit format if none of the codecs support 16 bit samples (or do they?) |
| 14:44.46 | P4C0 | Mercestes, thanks, i'll try |
| 14:44.56 | coppice | aydiosmio: I don't know a codec supporting less than 12 bit samples (G.711 is 12 bit) |
| 14:45.16 | cpm | speex |
| 14:46.18 | aydiosmio | this document says g711 is 8bit/8khz |
| 14:46.25 | aydiosmio | hm |
| 14:46.45 | coppice | it is lossy compressed to 8 bits from 12 bits |
| 14:46.55 | aydiosmio | The µ-law and A-Law algorithms encode 14-bit and 13-bit signed linear PCM samples (respectively) to logarithmic 8-bit samples. |
| 14:47.21 | coppice | its more like 13 and 12 bits, rather than 14 and 13 |
| 14:47.48 | aydiosmio | add an extra bit for the sign? |
| 14:48.06 | aydiosmio | hm, okay, that makes more sense |
| 14:49.25 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 14:49.40 | kovger | Hello! I have a problem, i downgraded my (*) to 1.2.17 from 1.4 because i cant get app_rxfax compile under 1.4. Now everything compiled fine, but i cant make any call. Outbound calls get error: Unable to request channel Zap, inbound calls failed with my providers message, that out of service. With 1.4 i can call and receive call, so the line is ok. I checked it with zttool too and it says no error. I googled some, but no succes, can anyone point me in |
| 14:49.40 | kovger | the right direction for some reading about this? |
| 14:50.09 | kovger | (oh and sorry english is not my native language) |
| 14:51.25 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:51.50 | eald | did you downgraded zaptel too? |
| 14:52.30 | *** join/#asterisk tkowal (n=nospamto@74.93.82.14) |
| 14:52.32 | kovger | yes, and libpri too |
| 14:56.01 | SuD | i answer myself: I must dial with A(x): Play an announcement (x.gsm) to the called party. |
| 14:58.56 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 14:59.10 | nextime | hello. Is wcfxs module obsolete with zaptel 1.4.1? |
| 14:59.12 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 14:59.37 | nextime | wctdm.c:MODULE_ALIAS("wcfxs"); |
| 14:59.40 | nextime | yes. |
| 15:02.31 | Mercestes | kovger: can I suggest a clean install? |
| 15:03.49 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 15:04.12 | chefrs | Any idea why when Asterisk dials out, it *says* it is putting a 1 in front of the #, but the phone line will say "You must first dial a 1 |
| 15:05.32 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 15:06.06 | file | chefrs: analog FXO port? |
| 15:06.10 | chefrs | FXS |
| 15:06.13 | chefrs | SIP phones. |
| 15:06.23 | file | no, I mean where is the phone call going out |
| 15:06.28 | kovger | Mercestes: ok, i will try it. hope that helps :) |
| 15:06.36 | chefrs | SIP In, outbound Zap FXS trunk to PSTN |
| 15:06.38 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:06.41 | af_ | where could I find libnbs? |
| 15:07.13 | file | that's an FXO port... and it's because the line is being picked up so fast that dialtone is not present, so the first digit is chopped off |
| 15:07.23 | Mercestes | kovger: Do you rmake cleans and make sure it removes all the leftovers. I seem to recall a maek distclean somewhere *ponders* Maybe that was somewhre else. |
| 15:07.37 | file | adding w to the front of the number in your Dial line in extensions.conf will introduce a delay - you can experiment with how many you need to get it to recognize the 1 |
| 15:07.39 | Mercestes | kovger: then manually make sure all the left overs are gone and then do your install from there. |
| 15:08.21 | chefrs | file: Any idea how to fix it? |
| 15:08.24 | chefrs | Mmkay |
| 15:09.02 | Mercestes | Maybe I should get my soekris box to boot sucecssfully before I start dinking with it, huh? |
| 15:09.45 | Qwell[] | Mercestes: where's the fun in that? |
| 15:09.57 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 15:09.57 | Mercestes | I ran out of space on my device. :) |
| 15:10.15 | Mercestes | maybe emerge asterisk-sounds wasnn't the brighest of ideas |
| 15:10.30 | chefrs | file: I'm sorry but Im looking in my extensions.conf and don't see what to change. |
| 15:11.03 | menfin | bulle, ? |
| 15:11.06 | Mercestes | chefrs: Dial(zap/g1/wwwwwwww1234567|900) |
| 15:11.10 | file | chefrs: the Dial line that dials out of your FXO port, you need to put a w in front of the ${EXTEN} part (I'm guessing on what it looks like... without seeing your dialplan I can't say for sure) |
| 15:12.01 | kovger | Mercestes: ok, i did all the make cleans, but now i will check all the dirs to any left overs before install |
| 15:12.29 | Mercestes | kovger: Grep distclean |
| 15:13.12 | genz | Is outgoing callerid possible? Like I dial a number and my phone resolves to the name of the number I'm dialing? |
| 15:14.11 | mkl1525 | Hi, "sip show channels" shows all active sip phones. "zap show channels" shows only the incoming calls - is this correct or is there a bug in my config. If not how can I get the count of all used zap channels? |
| 15:14.34 | aydiosmio | genz: not on POTS phones |
| 15:14.56 | *** join/#asterisk synthetiq (n=syntheti@vonmail.vonworldwide.com) |
| 15:15.07 | genz | aydiosmio: But on SIP phones? |
| 15:15.40 | nettie | hey guys anyone know what's the difference from zaphfc and chan_capi please? |
| 15:17.34 | unice | they are diffrent |
| 15:17.37 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:17.39 | *** join/#asterisk doolph (n=doolph@200.105.35.219) |
| 15:18.26 | unice | http://www.voip-info.org/wiki-Asterisk+zaphfc |
| 15:18.30 | unice | http://www.melware.org/ChanCapi |
| 15:18.37 | unice | have a read |
| 15:19.20 | nettie | unice thanx I read that .. I also came accross this: http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ |
| 15:20.14 | unice | well it's zaphfc, too |
| 15:20.23 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:21.09 | *** join/#asterisk riverratt (n=root@adsl-153-115-165.tys.bellsouth.net) |
| 15:21.15 | nettie | do you have any experience with zaphfc? which one is supposed ot be more stable? I jsut tested th eone supplied with bristuff anf I had to powercycle the box twice, it was hanging when I tried to unlaod the module |
| 15:22.38 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:22.50 | brodiem | hey what's the best way to do this: My inbound SIP trunk passes DIDs as +1npanxxxxxx, but on the PBX side routes must be based on just npanxxxxxx. Should I just do a pattern match for +1npanxxxxxx, and then use Cut() to change ${EXTEN}? Or can I do this a cleaner way and just use a pattern match, and only pass everything after +1? |
| 15:25.05 | unice | nettie, depends |
| 15:25.29 | nettie | I'm trying to compile bristuff zaphfc doesnt even finish :( |
| 15:25.29 | unice | if you won't need any hardware supported by zaptel, go with chan_capi |
| 15:25.48 | unice | where do you get the bristuff from |
| 15:26.03 | nettie | junghanns.net |
| 15:26.09 | nettie | latest pre |
| 15:26.19 | nettie | is that correct? |
| 15:26.21 | unice | paste the filename plz |
| 15:26.34 | nettie | http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y.tar.gz |
| 15:26.50 | nettie | thanx for helping me btw |
| 15:27.08 | unice | try http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-d.tar.gz |
| 15:27.14 | P4C0 | how long does the asterisk support takes?? |
| 15:27.16 | nettie | ok |
| 15:27.25 | Qwell[] | P4C0: support for what? |
| 15:27.54 | P4C0 | Qwell[], the g729 codec license that I just bought and can't get working correctly |
| 15:28.16 | Qwell[] | Did you call them? |
| 15:28.22 | P4C0 | no... email |
| 15:28.25 | nettie | unice: no luck still hangs |
| 15:28.43 | nettie | unice: lemme see if I actually provided all the deps it needs |
| 15:28.48 | Qwell[] | P4C0: I don't know |
| 15:29.04 | unice | <PROTECTED> |
| 15:29.14 | nettie | unice: I'm not usign the scripts I moved all the needed items by hand |
| 15:29.28 | P4C0 | Qwell[], it's strange, I have a license for one channel, but it's like if the server (or remote) will try to create a new channel when I answer the phone... (i have canreinvite=no) |
| 15:30.03 | *** join/#asterisk jsmith (n=jsmith@000-193-159.area3.spcsdns.net) |
| 15:30.05 | brodiem | hey what's the best way to do this: My inbound SIP trunk passes DIDs as +1npanxxxxxx, but on the PBX side routes must be based on just npanxxxxxx. Should I just do a pattern match for +1npanxxxxxx, and then use Cut() to change ${EXTEN}? Or can I do this a cleaner way and just use a pattern match, and only pass everything after +1? |
| 15:30.13 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:30.38 | jsmith | brodiem: ${EXTEN:2} strips off the first two characters of ${EXTEN} |
| 15:30.50 | blitzrage | jsmith: omghi |
| 15:30.58 | jsmith | synthetiq: If all your connections are SIP, then you should really check out SIP proxies |
| 15:31.08 | jsmith | synthetiq: (Such as SER or OpenSER) |
| 15:31.15 | brodiem | jsmith lol that'll do it, thanks |
| 15:31.27 | jsmith | brodiem: No problem |
| 15:31.42 | jsmith | blitzrage: Funny seeing you here... |
| 15:31.51 | blitzrage | jsmith: you too |
| 15:32.12 | blitzrage | I only joined to move a non-dev discussion to more appropriate channels |
| 15:32.43 | jsmith | Me too... but I guess I can help answer other questions for a few minutes |
| 15:32.47 | aydiosmio | hm, I wonder if SER supports P-Asserted-Identity |
| 15:33.29 | jsmith | aydiosmio: Sure, you can have SER add it... |
| 15:33.46 | aydiosmio | heheh |
| 15:34.10 | m-00kie | =X |
| 15:34.35 | aydiosmio | MD5HASH("P-Asserted-Identity") |
| 15:35.13 | m-00kie | people still use md5?? :P |
| 15:36.06 | aydiosmio | nope -- no PAID in SER or OpenSER |
| 15:36.15 | chefrs | file: Thanks! It's all fixed now. |
| 15:36.21 | aydiosmio | same threads with Verizon Biz customers asking how to insert it |
| 15:36.34 | aydiosmio | m-00kie: sure, why not? |
| 15:37.09 | m-00kie | collisions.. and it's insecure |
| 15:37.17 | chefrs | Any idea why when someone dials out sometimes it rings once then it's just silence? |
| 15:37.37 | aydiosmio | m-00kie: that's an overstatement |
| 15:37.43 | m-00kie | how so? |
| 15:38.02 | P4C0 | does anyone if registering to a peer or making the call directly have any effect on code translation? |
| 15:38.16 | aydiosmio | in that what you said is an exaggeration of reality |
| 15:38.31 | m-00kie | i didnt ask for a definition of an overstatement. |
| 15:38.41 | Mercestes | So we have Jsmith wtih "SER/OpenSER has this." and it doesn't...m-ookie with "md5 sucks" and yesterday with the "digium is a crappy company because people in #asterisk are rude." ... |
| 15:38.42 | aydiosmio | it's not "insecure" and the collision rate is near insiginificat for most purposes |
| 15:38.52 | Mercestes | .....did we loose our troll shields? |
| 15:39.10 | aydiosmio | Mercestes: that Digium thing was a laugh riot |
| 15:39.13 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdkk.cable.mindspring.com) |
| 15:39.21 | m-00kie | md5 is insecure. everyone knows that |
| 15:39.30 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 15:39.41 | m-00kie | nobody uses it in encryption anymore, or any system that requires a level of security |
| 15:39.48 | Mercestes | m-00kie, Your statement is inherently falacitious, given my 7 year old doesn't even know what md5 is. |
| 15:39.58 | chefrs | lol |
| 15:39.58 | aydiosmio | yeah! what he said! |
| 15:40.11 | synthetiq | jsmith ... use one ser box to hold all my registrations? |
| 15:40.16 | blitzrage | md5 is used in SIP to generate the hash used for authentication, which contains the username|realm|secret|nonce |
| 15:40.29 | blitzrage | synthetiq: yes -- or a group of them |
| 15:40.40 | Mercestes | m-00kie, In fact, you seem to be iterating several statements of personal opinion, likely gleaned off some second-rate conspiracy site, and expressing it as known and accepted fact that only stupid people would not adopt, which is entirely not true, regardless of the validity ofyour opinions. |
| 15:40.42 | aydiosmio | I've read the new attacks on MD5, and I still use it for password storage in databases and creating identifiers |
| 15:40.51 | chefrs | Any idea why when someone dials out sometimes it rings once then it's just silence? The Zap channel says it's answering the outbound call and placing it but.. nothing. |
| 15:40.55 | P4C0 | i'm going crazy :( |
| 15:40.56 | chefrs | aydiosmio: Same here. |
| 15:41.16 | jsmith | Mercestes: I didn't say "SER/OpenSER has this"... I said SER/OpenSER *can add the header* |
| 15:41.16 | m-00kie | Mercestes - im just a programmer =/ i dont know anything. |
| 15:41.19 | Mercestes | m-00kie, So please, reanalyze what you consider fact and what you consider opinion and reassert yourself from a more appropriate stance. |
| 15:41.19 | blitzrage | chefrs: depends what the SIP trace (assuming you're suing SIP) says |
| 15:41.35 | chefrs | blitzrage: Yep. SIP. |
| 15:41.35 | blitzrage | jsmith: I guess this is why we don't hang out in here anymore :) |
| 15:41.45 | synthetiq | blitz, but if i want to handle the registrations in asterisk, not on SER.... |
| 15:41.50 | jsmith | Mercestes: I'm sorry that people in here seem rude to you, but that shouldn't reflect on Digium... most of the people in here *don't* work for Digium |
| 15:42.04 | aydiosmio | his whole arguement was silly |
| 15:42.05 | jsmith | synthetiq: You can setup multiple SER/OpenSER boxes to be redundant |
| 15:42.05 | synthetiq | (internal billing reasons) |
| 15:42.10 | aydiosmio | no need to defend against it |
| 15:42.10 | Mercestes | jsmith: yes, that would have been my stanec yesterday with the gentleman that had a problem with us. :) |
| 15:42.19 | Mercestes | blitzrage, Sorry to see you go. |
| 15:43.04 | chefrs | blitzrage: SIP trace? I imagine that's not the same as the Asterisk CLI with some verbosity? |
| 15:43.13 | blitzrage | chefrs: sip debug |
| 15:43.24 | chefrs | That's what I thought, just checking. |
| 15:43.25 | m-00kie | im not even sure he saw my comment ;-) probably too busy oogling at what he types, smug in wallowing in his own arrogance.. *chuckles* well im off to more challenging stuff brb |
| 15:43.27 | jsmith | chefrs: You can also use Wireshark or ngrep to capture a SIP trace |
| 15:43.41 | chefrs | jsmith: Out office is wired funny, doesn'y work. |
| 15:43.50 | Mercestes | oh burn.... |
| 15:43.56 | Mercestes | wallowing in my own arrogance. |
| 15:44.00 | Mercestes | I love it. |
| 15:44.00 | riverratt | hi all, anyone care to help a newb with tdm400p problem? I have 4 FXO modules, but can't seem to get it to dial out. I have a very simple dialplan to test: extension 800 just dials ZAP/1/(mycellphone#), But no ringing. Any suggestions? |
| 15:44.01 | chefrs | I'd have to move my connection into the main hub. |
| 15:44.33 | jsmith | riverratt: Does "zap show channels" in Asterisk show the channels? |
| 15:45.29 | riverratt | Chan Extension Context Language MusicOnHold |
| 15:45.29 | riverratt | <PROTECTED> |
| 15:45.30 | riverratt | <PROTECTED> |
| 15:45.46 | riverratt | just have channel 1 configured for testing |
| 15:46.09 | jsmith | riverratt: Did you try "Zap/1/ww(cellphone#) |
| 15:46.27 | jsmith | riverratt: Some times Asterisk needs to wait a second or so before dialing the number |
| 15:46.37 | chefrs | blitzrage: There is a ton of crap to try and watch debugging SIP =/ |
| 15:47.01 | Mercestes | chefrs, I agree with your statement. :) |
| 15:47.08 | chefrs | Haha. |
| 15:47.16 | Mercestes | chefrs: I tend to pipe the output into a file. |
| 15:47.20 | jsmith | chefrs: You can use the VoIP tools in Wireshark to capture the data for the call you need |
| 15:47.22 | chefrs | Good call, mate. |
| 15:47.23 | riverratt | same deal |
| 15:47.38 | chefrs | jsmith: Alright, I'll go switch some stuff on the network so I can watch better. |
| 15:47.53 | jsmith | chefrs: Why not just use tcpdump on the Asterisk box itself? |
| 15:48.06 | aydiosmio | the SIP stats in Wireshark are uber handy |
| 15:48.08 | chefrs | jsmith: < Linux noobie. I know enough to vi and follow instructions |
| 15:48.23 | aydiosmio | it'll diagram calls and whatnot |
| 15:48.24 | chefrs | Amazing this thing even got setup if you ask me. |
| 15:48.30 | Mercestes | chefrs, That's better than most. :) |
| 15:48.53 | Mercestes | and vi ftw. even if it is all written in klingon. |
| 15:48.53 | chefrs | Oh and host websites. I got a VPS I run but. Yep. That's it. Ill just go play with some cords. brb. |
| 15:49.02 | jsmith | chefrs: http://www.voip-info.org/wiki/view/How+To+Debug+and+Troubleshoot+VOIP |
| 15:49.20 | chefrs | jsmith: Thanks. |
| 15:50.03 | *** join/#asterisk bdheeman (i=bsd@122.162.1.233) |
| 15:50.13 | kovger | Mercestes: still the same error (Unable to create channel of type 'Zap' (cause 0 - Unknown) ) but once i got back cause 34 - Circuit/channel congestion, so now maybe i have some more clue |
| 15:50.13 | riverratt | still no call out, any other thoughts? |
| 15:51.51 | brodiem | is it possible to actually change ${EXTEN}? I need to strip digits from it, and the call is getting passed to an AGI script where it reads the extension var. |
| 15:52.11 | Mercestes | kovger: Did you update your config files or copy them? |
| 15:52.20 | brodiem | Set(EXTEN=${EXTEN:2}|g) appears to not actually change it |
| 15:52.55 | aydiosmio | brodiem: you're passing the number to an AGI, but want to digit strip in the dialplan? |
| 15:52.59 | aydiosmio | why not do it in the AGI? |
| 15:53.01 | *** join/#asterisk bawb2 (n=bawb2@129.237.2.66) |
| 15:53.25 | brodiem | aydiosmio I want to avoid it if possible, it's a proprietary set of AGIs and the "proper" place to make customizations is in the dial plan |
| 15:53.54 | jsmith | brodiem: I think you want Goto(${EXTEN:2}), as changing the ${EXTEN} variable won't work |
| 15:53.55 | kovger | Mercestes: updated except extensions.conf, but now i will double check all of them |
| 15:54.17 | jsmith | brodiem: Or, you can assign it to another variable: Set(MYEXTEN=${EXTEN;2}) |
| 15:54.25 | aydiosmio | what if a LD number isn't 1 prefixed? |
| 15:54.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:54.41 | Mercestes | kovger: Check zap show status and zap show channels. and what zttool says. |
| 15:54.42 | blitzrage | check ${LEN()} |
| 15:54.42 | brodiem | so far these options will require changing the AGI |
| 15:54.55 | jsmith | brodiem: Why? |
| 15:55.30 | brodiem | jsmith anything incoming hits _. and is passed to the AGI (basically a routing daemon) |
| 15:55.39 | jsmith | brodiem: _. is *bad* |
| 15:55.41 | brodiem | jsmith n/m I get it |
| 15:55.46 | jsmith | brodiem: You should at least use _X. |
| 15:56.09 | Mercestes | _. is very bad. Don't use it. |
| 15:56.14 | jsmith | brodiem: _. will match on all the special extensions, such as 'i' and 't' and 'h' |
| 15:56.32 | brodiem | jsmith the AGI checks for those |
| 15:56.50 | bkw_ | _. is NOT bad |
| 15:56.53 | bkw_ | and you can use it |
| 15:56.57 | bkw_ | if you understand things |
| 15:58.54 | jsmith | bkw_: That's just the thing... most people *don't* understand that... that's the only reason I said it's bad. |
| 15:59.23 | bkw_ | instead of telling people to now use it, someone should document its proper usage |
| 15:59.27 | bkw_ | I have in the past |
| 15:59.34 | bkw_ | on the mailing list if I recall |
| 16:00.27 | jsmith | bkw_: I have documented it... multiple times in multiple places, including in my book |
| 16:01.03 | bkw_ | jsmith, so true |
| 16:01.29 | jsmith | bkw_: If there's one person in this channel who truly cares about documentation, you're chattin' with him |
| 16:01.41 | bkw_ | jsmith, I know that ;) |
| 16:02.25 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:04.07 | *** join/#asterisk heison (n=heison@gw-yyz1.somanetworks.com) |
| 16:04.11 | MrWup | im an asterisk noob |
| 16:04.53 | MrWup | i got two local sip handsets defined |
| 16:04.55 | MrWup | as friends |
| 16:04.55 | Mercestes | MrWup: We all are. |
| 16:05.09 | MrWup | then in the default context... i got their dial extensions in |
| 16:05.17 | *** join/#asterisk Fieldy (i=hDshmeJT@gentoo/contributor/Fieldy) |
| 16:05.17 | MrWup | so when i do console dial 101, one of them rings |
| 16:05.23 | MrWup | which is all great |
| 16:05.39 | MrWup | but how can i make it so i pick up aastra101 and dial 102 and aastra102 rings? |
| 16:06.13 | Mercestes | MrWup, by repeating your process for 101 but using 102 in every place 101 occurse. |
| 16:06.27 | P4C0 | guys, does this make sence: |
| 16:06.31 | MrWup | i did |
| 16:06.37 | MrWup | i cna dial both 102 and 101 from the console |
| 16:06.41 | MrWup | but cant dial them from each other |
| 16:06.50 | MrWup | i pick up the handset and type 101 or 102 and nothing happens |
| 16:07.14 | Mercestes | MrWup: Change your dialplan in your phones to handle 3 digit extensions. |
| 16:07.26 | Mercestes | MrWup: "Digitmap" is the most likely term you are looking for. |
| 16:07.46 | *** part/#asterisk bdheeman (i=bsd@122.162.1.233) |
| 16:08.19 | kovger | Mercestes: evrything seems to ok, no alarm, zttool report back OK, the one suspicious thing is: zap show status: IRQ:0 is it ok? |
| 16:09.06 | Mercestes | kovger, give a cat /proc/interrupts a try. |
| 16:09.21 | P4C0 | I have a private network, 3 softphones 1 asterisk server, a voip provider, nat support, working fine with alaw codec, my provider support alaw and g729, so I bougth one g729 license, install it to my asterisk server, and I can't make outgoing calls... incoming calls works sometimes, the problems seems to be that asterisk try to create a new channel when the other peer answers the phone... since if I do a sip show channels while ringing I can see the local sof |
| 16:09.21 | P4C0 | tphone as alaw, and the voip link as g729, show g729 shows 0/1 when the other end answer the call, I got a "no path to translate..." do I need another license?? |
| 16:09.33 | Mercestes | p4c0 There is no way you type that fast. |
| 16:10.35 | P4C0 | i have canreinvite=no but maybe the nat is mixing the g729?? or the fact that my provider don't let me register to it... ? |
| 16:10.46 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:10.56 | jsmith | MrWup: It sounds like you don't have the context set correctly in SIP.conf for those devices... but that's just a guess |
| 16:11.30 | P4C0 | Mercestes, why? it was a long time ago since my previews message... |
| 16:11.47 | kovger | Mercestes: i think this is thet line that refers to my card: 20: 474094831 474097362 474095429 474097423 IO-APIC-fasteoi wcte11xp |
| 16:12.14 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 16:12.30 | Mercestes | p4c0: frequency isn't hte problem, that's just alot to paste at once. |
| 16:12.57 | unice | by |
| 16:13.00 | unice | e :) |
| 16:13.04 | Mercestes | p4c0: Didn't I tell you how to fix it and tell you to bugreport it if it didnt' work? you only need g729 licenses to transcode so if you only have one transcode from alaw to g729 you only need one license |
| 16:13.44 | Mercestes | kovger, you are correct. Don't think it's IRQs then. Hrm. No weird zap messages in CLI sitting at idle? |
| 16:14.29 | P4C0 | Mercestes, didn't work... I'm waiting for reply from digium support... I'm using asterisk 1.4.2 is that stable? |
| 16:14.43 | Mercestes | p4c0: Your guess is as good as mine. |
| 16:15.32 | Mercestes | p4c0: 1.2.something would be what I consider "stable" as far as production goes, and 1.4 would be what I would consider as experimental. |
| 16:16.04 | kovger | Mercestes, nothing so long, just the usual register messages from my IP phones |
| 16:16.08 | P4C0 | Mercestes, humm but I already register this codec... can I change versions to test? do I need to register it again? |
| 16:18.06 | Mercestes | p4c0: You'd have to call digium for that. |
| 16:18.17 | Mercestes | I avoid g729 with great abanadon. |
| 16:18.24 | Mercestes | if only I could spell. |
| 16:18.26 | Mercestes | abandon. |
| 16:18.34 | Mercestes | stupid english language.... |
| 16:18.34 | P4C0 | :) |
| 16:18.37 | blitzrage | g729 works fine |
| 16:18.48 | P4C0 | blitzrage, not for me... |
| 16:18.49 | Mercestes | troll |
| 16:18.49 | *** join/#asterisk tnt_ (n=tnt@2.89-244-81.adsl-dyn.isp.belgacom.be) |
| 16:18.55 | P4C0 | blitzrage, what asterisk version do you have? |
| 16:19.00 | blitzrage | 1.4 SVN |
| 16:19.00 | Mercestes | :P |
| 16:19.32 | tnt_ | Anyone knows how to convert a binary granstream provisionning file back to a text format ? |
| 16:19.49 | P4C0 | blitzrage, is there any changes between released 1.4 and svn related to translation or g729?? |
| 16:19.59 | Mercestes | tnt_ the *best* way is to take the grandstream in yoru left hand, carefully place it under yoru tire, and then back over it 4 or 5 times. |
| 16:20.15 | blitzrage | P4C0: not from what I've seen in the commit messages |
| 16:20.29 | blitzrage | Mercestes: I prefer at least 6 or 7 times |
| 16:20.35 | Mercestes | kovger: pastebin your configs and a copy of your CLI failed call |
| 16:21.00 | P4C0 | blitzrage, any suggestions? |
| 16:21.09 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 16:21.09 | tnt_ | Mercestes: right ... ;p |
| 16:21.16 | blitzrage | P4C0: not really... it "just works" for me |
| 16:21.17 | Mercestes | tnt_ honestly that's a grandstreap support question. No one here really likes grandstreams....well, no one with any respect in this channel. ;) |
| 16:22.00 | Mercestes | p4c0: I think blitzrage's success comes in having more than 1 $10 license to play with. And he likely has enough bandwidth to handle it. |
| 16:22.12 | tnt_ | Mercestes: lol ;) I didn't choose the phone, I'd like to use my own on the company net but I need the sip params and don't have them ... |
| 16:22.18 | blitzrage | Mercestes: yah... I'm the provider :) |
| 16:22.21 | Mercestes | and a descent enough system to handle any transcoding h emight be doing. |
| 16:22.34 | blitzrage | yah.... Dell 2950's will do that for ya |
| 16:22.50 | Mercestes | tnt_: I accept the parameters of your situation. |
| 16:22.54 | Dr-Linux | i've a question about Dynamic conference room. if i created a dynamic conference bridge and hangup the call, how long this conference bridge will be available? :S |
| 16:23.07 | P4C0 | so it's almost a reality that I need to buy another license... |
| 16:23.08 | blitzrage | Dr-Linux: until there are no more users in the conference |
| 16:23.33 | Mercestes | p4c0: It's a...do you really want to use G729? |
| 16:23.46 | Mercestes | p4c0: If you want to test it, yo uhave your 1 license, make a g729 call. |
| 16:24.15 | P4C0 | Mercestes, well, my upload link is 256kbps and my provider only support alaw and g729... so |
| 16:24.26 | P4C0 | Mercestes, humm you mean from the cli? |
| 16:24.26 | Dr-Linux | blitzrage: what about PIN? only the first user can setup the password? |
| 16:24.35 | Mercestes | p4c0: in your phone. |
| 16:24.56 | blitzrage | Dr-Linux: eh? |
| 16:25.04 | Mercestes | p4c0: yo uhave a phone that uses g729 don't you??? |
| 16:25.11 | P4C0 | Mercestes, doesn't work... when it rings it show everything ok... then when the other peer pick uyp the phone it dies |
| 16:25.13 | blitzrage | Dr-Linux: you should just try it |
| 16:25.27 | P4C0 | Mercestes, nop, only freedownloaded sip soft phones :p |
| 16:25.33 | Dr-Linux | blitzrage: hhm.. ok |
| 16:25.41 | Mercestes | ah |
| 16:25.42 | *** part/#asterisk jsmith (n=jsmith@000-193-159.area3.spcsdns.net) |
| 16:26.00 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:26.01 | Mercestes | so you wish to transcode from g729 to alaw? |
| 16:26.15 | P4C0 | g729 will be used for the link with my provider only |
| 16:26.51 | Mercestes | then your provider sip account shoul db eallow g729 and you rsip phone should be allow alaw. |
| 16:26.53 | P4C0 | Mercestes, yes... or gsm, I tried with gsm already, same problem |
| 16:27.05 | P4C0 | Mercestes, I have it like that |
| 16:27.22 | P4C0 | Mercestes, I can even sometimes get incoming calls ok ! |
| 16:27.43 | Mercestes | then something else is wrong. |
| 16:28.14 | P4C0 | Mercestes, I can't figure it what... since with alaw it works fine! |
| 16:28.26 | P4C0 | Mercestes, want my sip.conf ? |
| 16:31.05 | P4C0 | Mercestes, http://rafb.net/p/XmmXcs30.html |
| 16:35.41 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 16:36.21 | P4C0 | how can I see the current translations? |
| 16:36.33 | P4C0 | console dial seems to work :p |
| 16:37.09 | *** part/#asterisk riverratt (n=root@adsl-153-115-165.tys.bellsouth.net) |
| 16:37.11 | *** join/#asterisk SoftIce (n=awk@dsl-242-123-186.telkomadsl.co.za) |
| 16:37.24 | SoftIce | does anyone know of a free g729 softphone ? |
| 16:38.47 | coppice | there don't seem to be any. free and G.729 don't really go together |
| 16:40.00 | kovger | Mercestes, pastebin give back errors, so i uploaded the file here: http://hq.iec.hu/asterisk.error if anything missing plz say, and thanks for your help :) |
| 16:40.21 | Mercestes | p4c0: reading. |
| 16:41.07 | P4C0 | Mercestes, humm this is really strangee, I delete a couple of unused peers... and did a console dial... and it worked... after that it seems to work :S I have done 3 calls already... working |
| 16:43.11 | Mercestes | Nice. |
| 16:43.54 | Mercestes | is dial zap/1/###### a valid syntax? |
| 16:48.32 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:48.37 | *** join/#asterisk Ahrimanes (n=ma@x1-6-00-0a-e4-2e-90-43.k707.webspeed.dk) |
| 16:52.25 | MrWup | anyone familair with the astra 9331i? |
| 16:52.30 | MrWup | why does it take so long to dial? |
| 16:52.47 | MrWup | the only way i get immediate dial is by dialling and then pressing dial |
| 16:53.00 | MrWup | else i have to dial my number and then wait about 4 seconds for it to dial |
| 16:53.02 | Mercestes | MrWup: That is controlled by your digit map |
| 16:54.18 | MrWup | on the asterisk server? |
| 16:54.26 | Mercestes | on your phone. |
| 16:54.47 | MrWup | hm |
| 16:54.49 | MrWup | i have no clue |
| 16:54.54 | MrWup | i got aastra 9133is |
| 16:55.25 | *** join/#asterisk ping2921 (n=marc3234@206-248-178-169.dsl.teksavvy.com) |
| 16:55.32 | Mercestes | Right. |
| 16:56.44 | ping2921 | I have the moh start from zero whenever I have a queue announcement. I am using native moh. Is there a way to avoid this problem? |
| 17:01.14 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:04.35 | *** join/#asterisk CuriosCat (i=stian@64.135.54.157) |
| 17:04.38 | CuriosCat | Hi all |
| 17:05.10 | CuriosCat | I've got an application where I need a monitoring system to call people and tell them about alerts that come in. |
| 17:05.28 | CuriosCat | i believe I can do this using Asterisk, Festival and an FXO PCI card. |
| 17:05.36 | CuriosCat | has anyone done anything like this before? |
| 17:06.16 | CuriosCat | (it's not really a voip application, but the necessary components are there) |
| 17:07.40 | P4C0 | Mercestes, false alarm... my mistake... I had the codec in alaw :( but now I can acept incoming calls... but not outgoing... |
| 17:09.43 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:09.48 | syzygyBSD | morning all |
| 17:10.47 | CuriosCat | heya syzygyBSD |
| 17:11.11 | syzygyBSD | woohoo! I am not invisible! |
| 17:13.34 | *** join/#asterisk nutcase (n=nutcase@i59F565FC.versanet.de) |
| 17:13.51 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 17:14.08 | CuriosCat | heh, no. |
| 17:15.28 | [TK]D-Fender | Hey, I've got an Aastra 57i CT here and have a question to anyone using it or the 480i CT : I have line 9 set up for another reg for use by the cordless handset. How do I stop the base phone form reacting to calls to it? |
| 17:18.35 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 17:19.16 | *** join/#asterisk Fieldy (i=DrmhA9vm@gentoo/contributor/Fieldy) |
| 17:19.49 | angryuser | good day i have this strange problem, misdn ports go down after send or receive a call, aster 1.4.2 zap 1.4.1 does anyone know a solution? |
| 17:20.18 | angryuser | if i put the up manually, it work for the next call |
| 17:22.42 | Mercestes | [TK]D-Fender, Not sure you can even do that. |
| 17:24.45 | *** join/#asterisk Tido (n=rawr@c-24-30-19-33.hsd1.ga.comcast.net) |
| 17:27.59 | *** join/#asterisk c4t3l (n=c4t3l@cpe-72-181-205-77.houston.res.rr.com) |
| 17:28.55 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:30.34 | [TK]D-Fender | Mercestes: Thats always a possibility..... just loking for "best case". |
| 17:31.58 | Mercestes | [TK]D-Fender, Maybe you can set a "no ring" for that partiuclar line in the config file, for the base, with a ring for the headset. not sure how detailed the config gets on that. |
| 17:32.39 | [TK]D-Fender | Mercestes: I had thought about that, but the consequence is that I could lift the receiver for a call of my own without looking and answer the line if it appears onscreen. |
| 17:32.47 | [TK]D-Fender | Mercestes: Which would of course suck. |
| 17:32.53 | Mercestes | indeed. |
| 17:33.07 | Mercestes | I guess the end result is that yoru trying to save yourself from buying an extra phone? |
| 17:33.34 | [TK]D-Fender | Mercestes: I got this phone because it was a 2 in 1, and have takien it on in place of my IP 600 |
| 17:33.52 | Mercestes | ah. |
| 17:34.35 | ping2921 | anyone could help with the moh restart problem I'm having whenever there is queue announcement? |
| 17:34.37 | [TK]D-Fender | Mercestes: It has some advantages for me here, and I'd like to save someone else the adaptation from Polycom usabililty (still #1) |
| 17:35.04 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 17:35.07 | [TK]D-Fender | Mercestes: I DO like the backlight honestly, and having a pile of presence buttons, etc. |
| 17:35.18 | [TK]D-Fender | Mercestes: And it does look kinda sharp |
| 17:35.20 | Mercestes | lol |
| 17:35.30 | Mercestes | Sounds like it's worth the effort eh? |
| 17:36.02 | kavit | What does "channel.c: Didn't get a frame from channel: SIP/XXXX" in the asterisk logs indicate? |
| 17:36.04 | [TK]D-Fender | Mercestes: It should be... I know if I got it as the non-CT it'd be a moot point. |
| 17:36.35 | kavit | the channel.c stops bridging the call after that.....is that a codec issue? |
| 17:37.27 | eald | tzafrir, sorry to bother you, do you know which version of asterisk is going to be released with etch? 1.2.13? |
| 17:37.33 | kavit | or a bandwidth issue? well it cant really be bandwidth as this is on a LAN... i am using g.729 with 20 licenses on the machine |
| 17:37.46 | PupenoR | What method does Asterisk use to send voicemail emails? sendmail? direct SMTP connection? |
| 17:37.59 | JunK-Y | PupenoR: by default? sendmail |
| 17:38.06 | PupenoR | thanks. |
| 17:39.15 | Mercestes | [TK]D-Fender, Got a doc/config files? Maybe I acn poor through it later and pick out something useful. |
| 17:39.40 | angryuser | misdn+Asterisk 1.4 = hate |
| 17:39.44 | [TK]D-Fender | Mercestes: All on Aastra's site. I've already printed up the *600* page admin guide.... |
| 17:40.05 | [TK]D-Fender | Mercestes: And am working my way through it. Anyone who complains about Polycom should now be shot :) |
| 17:40.14 | syzygyBSD | lol |
| 17:40.21 | syzygyBSD | Polycoms are confusing... |
| 17:40.36 | [TK]D-Fender | syzygyBSD: "You Ain't Seen Nothing Yet" |
| 17:40.53 | syzygyBSD | oh I have, that is why I stick to things that are worth my time |
| 17:40.53 | [TK]D-Fender | syzygyBSD: I have *)* problems with Polycom... THIS on the other hand.... |
| 17:40.58 | [TK]D-Fender | *0* |
| 17:41.18 | Mercestes | rofl |
| 17:41.20 | syzygyBSD | I can never get my polycom to connect when asterisk isn't running... |
| 17:41.20 | [TK]D-Fender | Polycom is a real fast deal for me, right from scratch |
| 17:41.33 | [TK]D-Fender | syzygyBSD: ..... |
| 17:41.55 | syzygyBSD | laugh... it is funny |
| 17:43.24 | syzygyBSD | don't act like you haven't spent HOURS trying to figure out issues like that before |
| 17:43.51 | syzygyBSD | maybe not exactly that.. but oh.. cable isn't plugged in, or they were dialing the wrong number and said they wern't... |
| 17:44.09 | [TK]D-Fender | syzygyBSD: No, never so long, but of course those questions come up... |
| 17:44.22 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:44.38 | syzygyBSD | i have a hell of a time connecting my polycom through nat.. every time |
| 17:44.55 | *** join/#asterisk pagec (n=root@141.155.63.101) |
| 17:45.10 | syzygyBSD | I found the simple solution is to just connect to the local IP on the asterisk box |
| 17:46.00 | pagec | with zaptel 1.4 do I still need to do something like 'make linux26' or can i just do 'make' |
| 17:47.12 | syzygyBSD | pagec: try it? |
| 17:48.06 | pagec | it works, it also works for zaptel 1.2 but compiles for the 2.4 kernel by default, i was hoping someone know the definative answer |
| 17:49.01 | P4C0 | incoming calls works fine with g729, outgoing calls failed when the other end pick it up with ast_channel_make_compatible: No path to translate from... |
| 17:49.06 | P4C0 | any ideas? |
| 17:49.56 | *** join/#asterisk sarnold (n=sarnold@oftc/staff/sarnold) |
| 17:50.08 | *** part/#asterisk sarnold (n=sarnold@oftc/staff/sarnold) |
| 17:50.15 | [TK]D-Fender | P4C0: You were using Idefisk, correct? |
| 17:50.22 | *** join/#asterisk hackman2K7 (n=chatzill@67-43-245-129-cablemodem-roll.fidnet.com) |
| 17:50.33 | *** part/#asterisk hackman2K7 (n=chatzill@67-43-245-129-cablemodem-roll.fidnet.com) |
| 17:50.34 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 17:50.44 | P4C0 | [TK]D-Fender, I don't think so... I have no idea what Idefisk is... |
| 17:50.57 | cr4z3d | soft phone |
| 17:51.14 | [TK]D-Fender | P4C0: What phone? |
| 17:51.35 | P4C0 | 3 local SJphones using alaw, one link to my voip provider in g729 |
| 17:52.23 | [TK]D-Fender | P4C0: pastebin your sip.conf entries for them, and CLI output with SIP debug for the failed attempt |
| 17:52.59 | P4C0 | [TK]D-Fender, yes moment |
| 17:53.25 | P4C0 | [TK]D-Fender, this is my sip.conf: http://rafb.net/p/XmmXcs30.html |
| 17:55.29 | *** join/#asterisk CAPLOCK (n=CAPLOCK@host-115.kwic.com) |
| 17:56.22 | CAPLOCK | Hello, I am a sysadmin for a small office in Canada. We have an asterisk system which works just great. |
| 17:56.40 | CAPLOCK | The one problem we are having is with touch-tone identification. |
| 17:57.31 | CAPLOCK | For whatever reason, people dial 805 but are directed to extension 800 instead. |
| 17:57.45 | P4C0 | [TK]D-Fender, this is the cli output: http://rafb.net/p/tmBuSI74.htm |
| 17:58.28 | CAPLOCK | I have set dtmfmode to tfc2833, and it had no effect. |
| 18:00.04 | CAPLOCK | Does anyone have any thoughts? |
| 18:00.25 | JunK-Y | what does dialplan show 805@ur_context says? |
| 18:00.42 | [TK]D-Fender | P4C0: SIP/2.0 404 Not Found Line #386 |
| 18:02.12 | Fieldy | Hrm.. I am experimenting with Festival() according to the book. It's working for me but I get a small flood of warnings on the console, wondering if this is indicative of a problem: http://rafb.net/p/7ZJZI244.txt |
| 18:02.13 | [TK]D-Fender | P4C0: [Mar 27 12:54:16] WARNING[7079]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/1-08200650(8) to SIP/mysipprovider-out-081ee9b8(256) |
| 18:02.34 | [TK]D-Fender | P4C0: Line 525 <- I take it your licenses are all take un at this point. |
| 18:03.12 | P4C0 | [TK]D-Fender, i only have lincense for one channel |
| 18:03.58 | P4C0 | but I'm only using one... (i think)... for incoming calls it works fine |
| 18:04.18 | [TK]D-Fender | P4C0: Apparerntly NOT |
| 18:04.36 | [TK]D-Fender | P4C0: You are using it for incoming and outgoing. |
| 18:05.02 | P4C0 | [TK]D-Fender, humm but, if I have one phone with alaw and my voip provider with g729 it should be as one right? |
| 18:05.28 | P4C0 | [TK]D-Fender, yes... from my voip provider... since the sjphone doesn't support g729... |
| 18:06.00 | P4C0 | [TK]D-Fender, if I need 2 channels for a call, why the incoming call works |
| 18:06.03 | [TK]D-Fender | P4C0: No. If you place an outgoing channel, and one comes in, you run out of licenses |
| 18:06.10 | P4C0 | i have no idea about that line 386 tho |
| 18:06.29 | P4C0 | no no, only one outside call at a time |
| 18:07.28 | [TK]D-Fender | P4C0: 1 out + 1 in = 2 LICENCES |
| 18:07.38 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 18:07.49 | *** join/#asterisk nasls_lsa (n=chatzill@87.203.116.128) |
| 18:08.22 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 18:08.22 | P4C0 | [TK]D-Fender, ok, if x call me it works... if I want to call x i doesn't, and I'm not calling x while having a conversation (or trying to call, or getting a call from y) |
| 18:09.11 | ping2921 | there multiple methods, mpg123,native wav,mp3.... which is the recommended? |
| 18:09.21 | P4C0 | [TK]D-Fender, the strange is that I can receive calls... without problems |
| 18:09.47 | [TK]D-Fender | P4C0: Ok, I can't concentrate on this here and you have a lot of errors and what looks like multiple calls going on in there, maybe I'll be able to look at this later. |
| 18:10.02 | P4C0 | [TK]D-Fender, even so, if I set the allow=g729,alaw then when there's no more g729 license, it should try to use alaw right?? well it doesn't work either |
| 18:10.22 | [TK]D-Fender | P4C0: NO, that does not work that way. |
| 18:10.37 | P4C0 | [TK]D-Fender, ok, no problem, thanks |
| 18:10.48 | [TK]D-Fender | P4C0: It will find the first match between them REGARDLESS of if you have a licence to support it. Then it will FAIL |
| 18:11.42 | cr4z3d | hmm quick question why can't i add another sip account to my xlite? |
| 18:11.47 | Fieldy | [update with new and exciting high loads] Hrm.. I am experimenting with Festival() according to the book. It's working for me but I get a small flood of warnings on the console, and the system load skyrockets. any ideas? details: http://rafb.net/p/a5oFDC95.nln.html |
| 18:12.00 | JT | cr4z3d: because you dont give them enough dollars |
| 18:12.05 | [TK]D-Fender | cr4z3d: Because it is a limited free version. |
| 18:12.07 | cr4z3d | oh you have to pay for that |
| 18:12.11 | cr4z3d | damn |
| 18:12.58 | cr4z3d | is there any good completely free ones |
| 18:12.58 | P4C0 | [TK]D-Fender, ok, but the multiple calls is really strange... however sip show channels shows a a couple of them but with REGISTER and OPTION as last message... |
| 18:12.58 | *** join/#asterisk tmjb (n=tmjb@195.252.125.18) |
| 18:12.58 | [TK]D-Fender | P4C0: Ok, well maybe I'll have a chance to look at them later. |
| 18:13.11 | P4C0 | [TK]D-Fender, thanks |
| 18:13.36 | *** join/#asterisk raggi (n=ra66i@80.243.187.90) |
| 18:13.49 | P4C0 | [TK]D-Fender, just the las question... may it have something to do the fact that my voip provider doesn't let me register to him?? (ip based auth) ? |
| 18:14.09 | raggi | Does asterisk 1.4 support any basic H.323 gatekeeper functionality, or would I still have to use GnuGK for that? |
| 18:14.20 | P4C0 | not sure if this g729 licensing needs an active registration with the peer it tries to forward the call? |
| 18:23.35 | *** join/#asterisk dc3aes (n=matt@70.67.135.40) |
| 18:24.46 | [TK]D-Fender | P4C0: no, and no |
| 18:28.20 | P4C0 | humm but since incoming calls works fine, maybe it's the way I'm making the call? the rule is for outoing calls in extension.conf file is: http://rafb.net/p/68hMLl81.html |
| 18:31.16 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 18:31.29 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-146059.home.otenet.gr) |
| 18:32.32 | *** join/#asterisk jffmriii (n=jffmriii@c-24-1-191-136.hsd1.il.comcast.net) |
| 18:32.36 | Dr-Linux | any perl expert around? :P |
| 18:33.20 | jffmriii | <PROTECTED> |
| 18:33.24 | *** join/#asterisk _AxS_ (n=axs@gentoo/user/axs) |
| 18:33.30 | jffmriii | any thoughts |
| 18:33.57 | Qwell[] | ~trixbox |
| 18:34.09 | jbot | rumour has it, trixbox is junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 18:34.54 | jffmriii | thank you qwell |
| 18:35.57 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 18:39.09 | cr4z3d | what the hell is trixbox anyway |
| 18:39.37 | cr4z3d | oh and quick question how do i ring all of my SIP phones at once? |
| 18:39.53 | raggi | trixbox seems 'ok', the fonality HUD stuff is nice if you get it working. |
| 18:40.28 | jffmriii | how do you get it working |
| 18:40.33 | P4C0 | [TK]D-Fender, hey do you have a second? |
| 18:40.57 | raggi | jffmriii: fix the configs by hand with trixbox 2.0, as far as I remember, they were broken 'out the box' |
| 18:41.10 | raggi | Does asterisk 1.4 support any basic H.323 gatekeeper functionality, or would I still have to use GnuGK for that? |
| 18:41.20 | P4C0 | [TK]D-Fender, in line 3 of http://rafb.net/p/68hMLl81.html I add r option (to the Dial) and it seems to work now... :| |
| 18:42.29 | raggi | jffmriii: I dropped it as a potential solution because it was not complete, and if the configs are broken, I certainly can't trust the system build, which is a more complex task than config management |
| 18:45.06 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 18:45.10 | cr4z3d | how would i go about dialing all my SIP extensions? i know i can do them in order using priorities and i also know i can ring them using & inbetween but is there an easy way to just ring all SIP extens? |
| 18:46.31 | CuriosCat | Hrm. Digium discontinued the X100P? |
| 18:47.22 | Strom_C | CuriosCat: ages ago |
| 18:47.32 | Strom_C | like...what, three years ago or something? |
| 18:48.07 | CuriosCat | Heh |
| 18:48.13 | CuriosCat | what am I supposed to buy now? :P |
| 18:48.20 | Qwell[] | tdm400p |
| 18:48.24 | Strom_C | TDM400? TDM800? |
| 18:48.52 | CuriosCat | Hrm. Only 10x the price |
| 18:48.53 | CuriosCat | heh. |
| 18:49.17 | Strom_C | the official x100p was never all that inexpensive to begin with |
| 18:49.18 | CuriosCat | Guess I'll have to take my chances with a clone card. |
| 18:49.32 | CuriosCat | Strom_C: I figured about $100 (compared to $20 for a clone card), but not $400+ |
| 18:49.39 | Qwell[] | that's with 4 modules |
| 18:49.41 | Strom_C | $400 is the fully loaded card |
| 18:49.44 | CuriosCat | I only need one. |
| 18:49.45 | Fieldy | clones :/ |
| 18:49.49 | Qwell[] | You can buy it with only 1 |
| 18:49.58 | Strom_C | the TDM01B is, what, $120ish? |
| 18:49.59 | CuriosCat | I need to connect one POTS line to a server. |
| 18:50.18 | CuriosCat | the cheapest on their web site is the TDM11B at $240 |
| 18:50.58 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 18:51.02 | Strom_C | TDM01B - $149 at telephonydepot.com |
| 18:51.06 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:51.55 | CuriosCat | getting better. |
| 18:52.06 | Strom_C | the x100p clone card is absolute rubbish anyway |
| 18:52.28 | aydiosmio | but it's cheap! |
| 18:52.36 | CuriosCat | all I need to do is be able to call POTS lines and read back a recording |
| 18:52.37 | Qwell[] | ~ygwypf |
| 18:52.48 | jbot | somebody said ygwypf was You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 18:52.48 | Strom_C | it's cheap in every sense of the word :) |
| 18:52.49 | CuriosCat | basically, it's for our monitoring system to call the NOC guy on duty if something goes down. |
| 18:53.03 | CuriosCat | I can do that with a modem, but I thought it would be nicer if it can actually read the alert, rather than just call and be mute :P |
| 18:53.11 | Strom_C | CuriosCat: well if that's all you need, why not use an ITSP? |
| 18:53.20 | CuriosCat | ITSP? |
| 18:53.26 | aydiosmio | CuriosCat: just use a modem to outdial, and send some dtmf |
| 18:53.27 | Strom_C | internet telephony service provider |
| 18:53.40 | CuriosCat | aydiosmio: Humans aren't very good at interpreting DTMF |
| 18:53.55 | *** join/#asterisk NTJOCK (n=brian@txshirts.com) |
| 18:53.57 | aydiosmio | if you get a call ID from your notificaiton server and you hear dialing, you know there's a problem |
| 18:54.05 | NTJOCK | hello all |
| 18:54.08 | CuriosCat | Strom_C: I have my own VOIP system, but I'm deliberately avoiding it so that notifications can still go out if a component in the voip system breaks. |
| 18:54.26 | CuriosCat | aydiosmio: Sure, but if you can know WHAT the problem is, that's better. And niftier. |
| 18:54.28 | Strom_C | and what is this "voip system"? |
| 18:54.33 | CuriosCat | $149 is manageable |
| 18:54.35 | aydiosmio | CuriosCat: BUT EXPENSIVE |
| 18:54.39 | CuriosCat | I think I'll go with Strom_C's suggestion |
| 18:55.25 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 18:55.37 | CuriosCat | Strom_C: It's actually two systems. One is Cisco Call Mangler-based (for the office), the other is Broadsoft-based (not my choice, for customers). Each system has a Cisco AS5350 acting as a dedicated gateway between PSTN and VOIP |
| 18:55.50 | Strom_C | yeah, you dont need to touch that |
| 18:55.58 | Strom_C | just have the asterisk box talk to the ITSP directly |
| 18:56.07 | NTJOCK | I'm trying to get call parking working. I'm using the example in the wiki but * is ignoring the #790 (where i have CP set)... is there something special besides features.conf? |
| 18:57.03 | CuriosCat | I *could* do this through that, with just a sip stream. But if the alert is "your DS3 MUX just died", it won't get there :P |
| 18:57.03 | Strom_C | true |
| 18:57.03 | CuriosCat | the nice thing about POTS lines is, they are *damn* reliable. |
| 18:57.03 | Strom_C | yeah, for reliability, the POTS line is the way to go |
| 18:57.24 | CuriosCat | and I can't wait to exploit the prank potential here |
| 18:57.46 | CuriosCat | like when the system calls the chief noc technician and goes "The building is on fire." or "A train derailed and cut all the fiber." |
| 18:57.52 | CuriosCat | and then tell him it was a glitch in the system ;) |
| 18:58.06 | Strom_C | there's a fun way to lose your job :) |
| 18:58.12 | CuriosCat | Strom_C: I'm this guy's boss. :) |
| 18:58.15 | Strom_C | ah ok :) |
| 18:58.42 | NTJOCK | "Your daughters' Bf is calling" |
| 18:58.42 | NTJOCK | lol |
| 18:58.51 | CuriosCat | "Quit slacking and get back to work." |
| 18:59.01 | NTJOCK | "gambling.com collections dept" |
| 18:59.02 | NTJOCK | lol |
| 18:59.15 | NTJOCK | "internet porn callback service" |
| 18:59.36 | NTJOCK | "dial-a-date hotline" |
| 18:59.37 | NTJOCK | lol |
| 18:59.37 | CuriosCat | don't get to play with the caller ID on PSTN though |
| 18:59.47 | NTJOCK | the look on their face would be priceless |
| 19:00.29 | CuriosCat | and there are remote control cameras to capture it :) |
| 19:01.40 | NTJOCK | has anyone gotten call parking working that might give me a pointer on it? |
| 19:01.41 | NTJOCK | :) |
| 19:04.39 | anonymouz666 | SIP/2.0 482 Loop Detected.. just because I am calling using * from A to B and both are registered in the same machine. |
| 19:05.37 | aydiosmio | I've begun using SMS for all my notifications |
| 19:06.06 | *** join/#asterisk CuriosCat (i=stian@dynamic157.noc.host.net) |
| 19:07.19 | *** join/#asterisk fbcit (n=cnighswo@nc-71-0-121-24.sta.embarqhsd.net) |
| 19:07.55 | raggi | sorry, anyone in who has experience with h323 who might be able to advise me on a direction to proceed |
| 19:08.23 | aydiosmio | upward. |
| 19:08.49 | aydiosmio | and always twirlling, twirlling! |
| 19:10.36 | raggi | mmm, specifically, I have a radio network, which currently connects to asterisk over E1 |
| 19:10.51 | raggi | I've been requested to produce a 'backup circuit' capability using h323 |
| 19:11.25 | raggi | the network may have a gatekeeper, but at this point I'm not certain I can use it |
| 19:11.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:11.59 | raggi | I had a working config built a long time ago as a PoC, using gnugk, asterisk 1.2, on gentoo |
| 19:12.31 | raggi | but gentoo is really not suitable as an environment for that, and on top of that, the gnugk dependency is a PITA |
| 19:13.20 | aydiosmio | cmd Prefix and StripMSD are no longer on voip-info, were these once features? |
| 19:13.25 | fbcit | Qwell: I'm still having no sucess getting the buttons to work on a 30VIP; The * CLI reflects some button configuration when the phone registers, but not much other than the dialpad seems to work. Any help would be greatly appriciated as resources for this phone and * are slim. |
| 19:13.40 | raggi | so, I'm asking, is the gnugk definately required for this kind of setup (arbritrary incoming h323 calls from an h323 capable device) (may need radio network engineers to advise more, and am seeking this advice now) |
| 19:14.20 | aydiosmio | (what were they replaced with, looks like after 1.2.6?) |
| 19:14.21 | CuriosCat | OK, order placed for the TDM01B |
| 19:14.29 | CuriosCat | I'll be back in a few days for help to get it working, I'm sure :D |
| 19:18.26 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-64-115.dsl.peoril.sbcglobal.net) |
| 19:18.42 | MACscr | im having an issue making outgoing calls. Whats odd is that the issue just started |
| 19:18.44 | MACscr | http://pastebin.ca/412538 |
| 19:19.01 | MACscr | i can receive calls fine though |
| 19:19.08 | MACscr | and im not getting an error |
| 19:19.41 | MACscr | well, i guess i am after awhile. The phone just sits their and doesnt ring and eventually it says it cant communicate with the server, but if you look at the log, it obviously can |
| 19:22.02 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 19:22.08 | *** join/#asterisk shinux__ (n=shinux@196.220.24.193) |
| 19:22.40 | [[blah]asfd | i am getting a dependency error when trying to install a program to my centos 4.4 system. I am trying to install webmeetme. here is the output of the make command: http://pastebin.ca/412540. So i did a yum provides lmysqlclient.so. It found nothing. so I tried yum provides libmysqlclient.so. here is the output: http://pastebin.ca/412543 could anyone help me get past this issue? |
| 19:23.24 | *** part/#asterisk _AxS_ (n=axs@gentoo/user/axs) |
| 19:28.26 | *** join/#asterisk shinux__ (n=shinux@196.220.24.193) |
| 19:30.48 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 19:30.57 | cr4z3d | is there a way to dial all SIP accounts that are registered? typing Dial(SIP/user1 & SIP/user2.. ect) is kind of a pain |
| 19:31.31 | anonymouz666 | <PROTECTED> |
| 19:31.42 | cr4z3d | queue? |
| 19:31.44 | anonymouz666 | make a queue! |
| 19:31.51 | anonymouz666 | and put all peers inside |
| 19:32.47 | cr4z3d | ah ok but there's no way to just dial all the registered sip accounts? |
| 19:33.02 | tuan_modulis | sure there is |
| 19:33.04 | tuan_modulis | queue |
| 19:33.16 | tuan_modulis | ringall |
| 19:33.37 | cr4z3d | oh so i have to setup a queue to do that.. alright i guess i should google some info about queues |
| 19:34.10 | tuan_modulis | ~thewiki |
| 19:34.11 | jbot | thewiki is probably at http://www.voip-info.org/wiki-Asterisk |
| 19:34.17 | tuan_modulis | best place to start |
| 19:34.29 | cr4z3d | yeah haha usually that wiki is the first result from google |
| 19:34.42 | MACscr | im having issues calling toll free numbers, call just seems to sit there |
| 19:35.04 | MACscr | im not sure how my caller id could be set wrong though as it seems to be passed fine to my cell phone |
| 19:35.41 | MACscr | (caller id issue was somethign that found on google in reference to toll free number issues and *) |
| 19:36.43 | *** join/#asterisk psybach (i=nobody@foster.stonedcoder.org) |
| 19:37.03 | *** part/#asterisk psybach (i=nobody@foster.stonedcoder.org) |
| 19:40.39 | *** join/#asterisk vader-- (n=me@204.183.88.101) |
| 19:40.40 | vader-- | hola |
| 19:40.43 | vader-- | could anyone help me |
| 19:40.54 | vader-- | with some asterisk time issues |
| 19:40.57 | vader-- | due to DST |
| 19:41.09 | Qwell[] | vader--: That was over 2 weeks ago... |
| 19:42.19 | vader-- | im having issues with a gotoiftime function |
| 19:43.20 | Qwell[] | upgrade your timezone packages on your system |
| 19:43.26 | vader-- | i did |
| 19:43.34 | Qwell[] | What distro? |
| 19:43.49 | aydiosmio | what happens when Dial is Busy or times out? |
| 19:43.51 | cr4z3d | you shoudl move to arizona where they don't have dst |
| 19:44.04 | Qwell[] | cr4z3d: some places still do |
| 19:44.12 | cr4z3d | most places do |
| 19:44.20 | cr4z3d | at least in the US |
| 19:45.46 | fbcit | Qwell: I'm still having no sucess getting the buttons to work on a 30VIP; The * CLI reflects some button configuration when the phone registers, but not much other than the dialpad seems to work. Any help would be greatly appriciated as resources for this phone and * are slim. |
| 19:45.50 | *** part/#asterisk mmartinn (n=martins@128.227.123.22) |
| 19:46.04 | heison | I'm having trouble passing callerID to VoicemailMain, it used to work fine with 1.2.14 but stopped working after I have upgraded to 1.4.2: exten => 8599,1,VoicemailMain(${CALLERIDNUM}). When I tried to print the callerID exten => 8599,n,NoOp(${CALLERIDNUM}), I get an empty string on the console. If I hard code an account number when calling VoicemailMain (e.g. exten => 8599,n,VoicemailMain(1502)), it works as expected - only prompt for passwo |
| 19:46.13 | vader-- | debian 3.1 |
| 19:46.36 | Strom_C | heison: that's because CALLERIDNUM was deprecated as of 1.2; you were supposed to change to use CALLERID(num) |
| 19:46.41 | JunK-Y | heison: this is deprecated! |
| 19:46.55 | JunK-Y | ya ignored all warnings on 1.2, now times to pay :P |
| 19:47.07 | JunK-Y | ${CALLERID(num)} |
| 19:47.08 | SuD | hi, how do i install asterisk webmin module? is it really possible? |
| 19:47.15 | heison | Strom_C: i have tried that too, same thing: exten => 8599,n,NoOp(CALLERID(num)) gave me an empty string |
| 19:47.29 | Strom_C | heison: ${CALLERID(num)} |
| 19:47.41 | heison | ic, let me try again |
| 19:47.48 | Strom_C | instant win catsex |
| 19:47.52 | NTJOCK | Is anyone here familiar with call parking? |
| 19:48.01 | NTJOCK | I'm having some headaches getting it working per the wiki |
| 19:48.01 | JunK-Y | CALLERID(num) should display "CALLERID(num)" |
| 19:48.21 | aydiosmio | how would I dial another number if Dial times out in the dialplan? n + 101 or would I have to eval the DIALSTATUS in the next line? |
| 19:48.33 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 19:48.48 | JunK-Y | Gotoif the DIALSTATUS |
| 19:48.56 | aydiosmio | thx |
| 19:49.02 | heison | JunK-Y: yes, CALLERID(num) displayed "CALLERID(num)" |
| 19:49.19 | JunK-Y | so isnt empty, ya just forgot the $ before to call the parser. |
| 19:49.32 | JunK-Y | core show function CALLERID for more infos. |
| 19:50.22 | e-milio | make install |
| 19:50.33 | aydiosmio | make brownies, not installs |
| 19:50.41 | Strom_C | *** make ERROR: You're on IRC. |
| 19:50.58 | JunK-Y | Strom_C: nioyhgahahah |
| 19:51.17 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 19:51.18 | heison | JunK-Y: i'm still waiting for my wife to hit the message button on the phone :( |
| 19:51.19 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 19:51.38 | JunK-Y | heison: be patient, ya will wait all ur life for women :) |
| 19:51.53 | heison | JunK-Y: absolutely! |
| 19:52.05 | heison | has anyone got rx_fax working with 1.4.2? |
| 19:52.18 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:53.10 | cr4z3d | does asterisk do FoIP? |
| 19:53.19 | Strom_C | FoIP? |
| 19:53.22 | aydiosmio | lorf |
| 19:53.22 | Strom_C | Faces Over IP? |
| 19:53.23 | cr4z3d | fax over ip |
| 19:53.25 | mafkees | fax over ip |
| 19:53.32 | fordfrog | hi, what is the correct syntax for ex-girl extension when using realtime? I use called_number/calling_number (outbound call) but that gives me 404 when calling from caller. I use * 1.2.14. |
| 19:53.33 | Strom_C | Fudge over IP |
| 19:53.39 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 19:53.45 | mafkees | $foo over ip |
| 19:53.56 | fordfrog | if I remove /calling_number then it works |
| 19:54.02 | mafkees | fordfrog: that is not possible with realtime |
| 19:54.09 | quidpro | hmm... what's the best way to convert audio tracks to SLIN... have some prompts on CD I want to put on * |
| 19:54.12 | [[blah]asfd | I have never seen this before. When I try to dial an extension I get this in reply... can anyone tell me where I could begin to look to rectify this issue? Got SIP response 488 "Not Acceptable Here" back from 10.0.4.182 |
| 19:54.35 | mafkees | fordfrog: you can use the GotoIf dialplan application for that in realtime |
| 19:55.01 | heison | i thought i've seen rx_fax and T.38 somewhere but didn't go into details, i was actually referred to rx_fax on FXO |
| 19:55.19 | fordfrog | mafkees: ah, I was confused by one of comments in http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions where the person states it did not work by that time but it was processed like if /caller_number was not there. |
| 19:55.23 | mafkees | GotoIf($["${CALLERID(num)}" == "some.number"]?exgf|bla) |
| 19:55.35 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:56.09 | mafkees | that line is from memory, so best check it before putting it into production ;) |
| 19:57.07 | mafkees | [[blah]asfd: check sip registration etc |
| 19:57.27 | mafkees | quidpro: you tried sox ? |
| 19:57.50 | mafkees | one more beer and I'm off |
| 19:57.58 | cr4z3d | a beer it's only 1pm |
| 19:58.19 | fordfrog | mafkees: thx for help |
| 19:58.57 | mafkees | Tue Mar 27 21:58:56 CEST 2007 |
| 19:59.21 | mafkees | fordfrog: hope you get it working :) |
| 19:59.38 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:59.40 | fordfrog | mafkees: well, this makes things much more difficult :-) |
| 19:59.50 | cr4z3d | she's a maniac maniac and she's dancing like she's never danced before |
| 20:00.02 | quidpro | mafkees: not yet, I'm just not sure of the command line for SLIN w/sox |
| 20:00.11 | Strom_C | something something something on a saturday night |
| 20:00.37 | Strom_C | cr4z3d: congratulations, you've given me a new earworm |
| 20:00.44 | cr4z3d | haha |
| 20:01.02 | Strom_C | whether this is better or worse than "Get Outta My Dreams, Get Into My Car" remains to be seen. |
| 20:01.11 | cr4z3d | haha |
| 20:01.36 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 20:02.05 | Strom_C | ok, I think it's time to spin some 80s now :) |
| 20:02.30 | cr4z3d | haha yeah i've been filling my moh with 80s |
| 20:02.35 | cr4z3d | and they're all stuck in my head |
| 20:02.45 | Strom_C | i've got two crates of 12" singles |
| 20:02.53 | cr4z3d | haha |
| 20:03.04 | Strom_C | ooh, you know what I should do? |
| 20:03.07 | Strom_C | stream it also :) |
| 20:03.18 | cr4z3d | haha yes you should |
| 20:03.31 | Strom_C | one moment |
| 20:04.40 | Strom_C | http://www.stromcarlson.com:8000/stream.mp3 |
| 20:04.40 | Strom_C | there |
| 20:04.51 | Strom_C | i'll start playing records in a minute |
| 20:05.19 | cr4z3d | alright nice haha |
| 20:05.41 | aydiosmio | where's the meetme conf with the stream? |
| 20:05.48 | aydiosmio | huh? huh? |
| 20:05.51 | *** join/#asterisk kiko69 (n=keith@75.16.91.106) |
| 20:06.44 | Strom_C | you should be able to hear it... |
| 20:06.50 | cr4z3d | haha yeah i got it going |
| 20:06.54 | Strom_C | awesome |
| 20:07.13 | JunK-Y | aydiosmio: and the meetme conf streamed ? |
| 20:07.20 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 20:07.25 | mafkees | quidpro: sox -c 1 track01.mp3 -t raw -r 8000 -c 1 -s -w track01.raw |
| 20:08.21 | deeperror | quick q regarding sip debug info.....the output line for VIA: has the following Via: SIP/2.0/UDP 10.1.10.80:5060;branch=z9hG4bK109ec671;rport how would I change the 10.1.10.80 ip to another address? |
| 20:08.48 | deeperror | i'm getting the error that there was no reply to a critical packet |
| 20:08.53 | jans0n | anyone here using digisip as sip provider? |
| 20:09.07 | deeperror | and i think they are redirecting back to the 10 address and not the ISP address |
| 20:10.05 | Juggie | deeperror, sip.conf -> externip |
| 20:10.12 | mafkees | Strom_C: lol, nice stream |
| 20:10.54 | cr4z3d | haha yeah it's banging |
| 20:10.55 | mafkees | *piep* *piep* |
| 20:11.03 | deeperror | juggie: ahh in [general]!!! |
| 20:12.11 | mafkees | dude! get more upstream bandwidth ! |
| 20:12.20 | cr4z3d | why? |
| 20:12.29 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:12.29 | cr4z3d | it's comming perfect for me |
| 20:12.39 | mafkees | not here |
| 20:12.47 | *** join/#asterisk duki (n=duki@host-85-27-49-12.brutele.be) |
| 20:12.48 | mafkees | mostly perfect |
| 20:12.53 | cr4z3d | it sounds good enough to be cd quality |
| 20:12.55 | mafkees | but sometimes a small gap |
| 20:13.00 | mafkees | yeah |
| 20:13.04 | mafkees | the sound is great |
| 20:13.11 | cr4z3d | where is the server located? |
| 20:13.28 | mafkees | prolly interconnect issue indeed |
| 20:13.43 | cr4z3d | i'm in the US |
| 20:13.44 | mafkees | the things you say..... |
| 20:13.49 | mafkees | ur unbelievable ! |
| 20:13.57 | deeperror | juggie: added that in general and the ip is still in the same in debug info any other hints? |
| 20:13.58 | mafkees | I'm in .nl |
| 20:14.04 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 20:14.22 | cr4z3d | ah the netherlands |
| 20:14.26 | mafkees | uhhuh |
| 20:14.42 | cr4z3d | never been out of north america except jamaica |
| 20:14.46 | e-milio | heloo guys |
| 20:15.27 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 20:15.27 | mafkees | maybe iTunes is not buffering enough |
| 20:15.28 | mafkees | ;) |
| 20:15.47 | e-milio | on zttool -> ztdummy always says unconfigured. Where do you configure it ? |
| 20:15.51 | cr4z3d | i have no idea what i'm playing mine with haha i just opened the link in my browser on ubuntu and it played |
| 20:15.53 | mafkees | but I did not get VLC to stream the music to my airtunes device yet |
| 20:16.32 | mafkees | and I'm too lazy to get up and attach the audiocable to my ibook directly |
| 20:16.54 | cr4z3d | haha |
| 20:17.09 | cr4z3d | i just have 2 laptops on a desk and switch the speaker cable back and forth |
| 20:17.11 | JunK-Y | e-milio: /etc/zaptel.conf |
| 20:17.19 | deeperror | anyone know how i could set the ip address in the sip debug info here |
| 20:17.20 | deeperror | Via: SIP/2.0/UDP 10.1.10.80:5060;branch=z9hG4bK36c66458;rport |
| 20:17.23 | cr4z3d | my ghetto setup with a crappy small desk in college |
| 20:17.54 | mafkees | lol |
| 20:18.09 | mafkees | I have this nice airport express with audiocable running to my stereo |
| 20:18.22 | mafkees | iTunes is using the device as 'external speakers' |
| 20:18.26 | mafkees | works smooth |
| 20:18.47 | cr4z3d | yeah.. i don't have a real stereo i just have 5.1 logitechs on my media comp and 2.1 phillips in my room |
| 20:19.15 | mafkees | 5.1 digitally connected here |
| 20:19.28 | mafkees | crappy amp tho |
| 20:19.31 | *** join/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 20:19.41 | mafkees | cheap sony device |
| 20:19.44 | voltagex | hi, what's IAX2 hangup code 34? |
| 20:19.48 | cr4z3d | that's pretty cool.. it's funy though i have like a pretty sweet audio setup and then a crappy 22" sdtv from like 5 years ago |
| 20:19.56 | mafkees | still saving money to get myself a nice Harmen Kardon |
| 20:20.32 | *** join/#asterisk hedge77 (n=netwalke@209.42.192.202) |
| 20:20.42 | mafkees | ok, beer is finished |
| 20:20.56 | mafkees | time to finally finish 1984 by Orwell |
| 20:21.08 | mafkees | latero all |
| 20:21.10 | cr4z3d | later |
| 20:21.15 | mafkees | and thnx for the great music Strom_C |
| 20:21.30 | hedge77 | are dynamic queue members using a local channel still marked as invalid after restart or am I up against something else? |
| 20:22.31 | MACscr | anyone by chance know of a dialer for firefox where you can highlight a number and it will call that number and your extension? |
| 20:22.53 | cr4z3d | that would actually be pretty sweet |
| 20:22.56 | e-milio | JunK-Y: thanks |
| 20:23.04 | cr4z3d | or at least one that would integrate with like xlite |
| 20:23.20 | genz | Is outgoing callerid possible? Like I dial a number and my phone resolves to the name of the number I'm dialing? |
| 20:24.03 | deeperror | juggie: got it working...when using externip I also needed to have localnet defined as well...thanks for the point in the right direction! |
| 20:26.21 | Juggie | np |
| 20:29.31 | e-milio | JunK-Y: Any particular settin? I cant fine anything referencing ztdumy and google is not helping |
| 20:30.04 | JunK-Y | e-milio: configure ur zaptel.conf, has nothing to do with ztdummy at this point |
| 20:30.11 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:30.14 | JunK-Y | ztdummy will be always UNCONFIGURE anyways. |
| 20:31.19 | e-milio | mmmm... I followed the compilation/modprobs etc, i was thinking that at some point it will beecome"configured" |
| 20:32.00 | JunK-Y | nope |
| 20:32.20 | JunK-Y | when ya do: lsmod|grep zap, do ya see it loaded? |
| 20:32.58 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 20:33.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-97-204.red.bezeqint.net) |
| 20:33.47 | e-milio | it was loading, now it is not, checking why.. |
| 20:34.04 | e-milio | seems to take a little bit |
| 20:34.07 | e-milio | now it is up |
| 20:34.53 | JunK-Y | so ya should be able to use meetme now. |
| 20:36.02 | e-milio | app_meetme.c:1546 conf_run: Unable to write frame to channel: Success |
| 20:36.16 | e-milio | that message is apperaing hundreds of times |
| 20:38.45 | MACscr | found a pretty cool article on using ajax for caller id display on a webpage |
| 20:38.45 | MACscr | http://www.linuxjournal.com/article/9159 |
| 20:41.35 | MACscr | cr4z3d: that that FF extension http://www.snapanumber.com/ |
| 20:42.38 | cr4z3d | oh nice that sounds really good |
| 20:43.35 | MACscr | id love to find a GPLed one though |
| 20:44.46 | MACscr | cr4z3d : this is pretty cool too, not the same, but still cool |
| 20:44.46 | MACscr | http://adm.hamnett.org/ |
| 20:46.42 | hedge77 | are dynamic queue members using a local channel still marked as invalid after restart or am I up against something else? |
| 20:46.55 | hedge77 | are dynamic queue members using a local channel still marked as invalid after restart or am I up against something else? |
| 20:47.05 | cr4z3d | woah spammy |
| 20:47.07 | Strom_C | KICK PLEASE |
| 20:47.08 | hedge77 | aasf |
| 20:47.36 | JunK-Y | hedge77: just ause ya repeart like a child, no-ones will answer you. |
| 20:47.37 | hedge77 | woops |
| 20:47.45 | hedge77 | got rightclick happy and was scrolled back |
| 20:47.47 | hedge77 | sorry |
| 20:49.26 | anonymouz666 | Strom_C is cranky |
| 20:49.28 | anonymouz666 | :D |
| 20:49.45 | cr4z3d | Strom_C is laying down some 80s |
| 20:50.27 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 20:51.30 | hedge77 | I'm sorry :( |
| 20:52.37 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 20:52.45 | chefrs | Which conf file do I edit to add a Wait() before dialing out? |
| 20:53.01 | cr4z3d | chefrs, extensions.conf |
| 20:53.10 | *** join/#asterisk Fieldy (i=XAmXMldn@gentoo/contributor/Fieldy) |
| 20:53.12 | chefrs | I don't see the right spot =/ |
| 20:53.13 | Strom_C | yeah, im getting tired - this is the last song i think i'll do :) |
| 20:53.24 | cr4z3d | haha alright |
| 20:54.12 | Strom_C | been enjoying it? |
| 20:54.25 | *** join/#asterisk garreel (n=garreel@host143-34-dynamic.17-87-r.retail.telecomitalia.it) |
| 20:54.34 | cr4z3d | haha yeah me and my roommate have been working whilelistening to it |
| 20:54.42 | Strom_C | awesome :) |
| 20:55.50 | cr4z3d | damn it.. i just broke my desk chair.. off to ikea to grab a new one |
| 20:56.28 | chefrs | It should look like.. "Executing Dial("SIP/1000-b78032f0", "ZAP/6/13035555555|120|rwwwww") in new stack" right? |
| 20:56.45 | *** join/#asterisk Ast001 (n=uros@212-200-206-85.adsl.sezampro.yu) |
| 20:56.56 | Ast001 | good evening people |
| 20:57.12 | chefrs | hi. |
| 20:57.32 | Ast001 | i need help with few issues I have on asterisk |
| 20:57.39 | chefrs | You and me both. |
| 20:58.01 | *** join/#asterisk Fieldy (i=Gc8pnmoM@gentoo/contributor/Fieldy) |
| 20:58.25 | Ast001 | first when someone call me and dissconetct before answer command i get an error message said cdr didnt start warning and mysql didnt write error |
| 20:59.02 | Ast001 | 2 warnings about cdr |
| 20:59.10 | Ast001 | and 1 error on mysql |
| 20:59.29 | Ast001 | my dial plan looks like 1 noop 2 wait(1) 3 answer |
| 20:59.51 | Ast001 | 4 queue 5 hangup |
| 21:00.17 | Ast001 | i wonder if i delete wait step would i solve problem |
| 21:00.18 | e-milio | how can I check if fix for bug 4301 is incorporated on 1.2.14 ?? |
| 21:00.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:01.03 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 21:01.31 | jtexter3 | Anyone know if it's possible to send extra headers on an IAX call, similar to SIPAddHeader on SIP calls? |
| 21:02.35 | e-milio | top |
| 21:03.10 | hedge77 | chefrs: did you want it to wait when dialing everything (including other extensions) or just outbound calls? |
| 21:05.11 | chefrs | hedge77: Outbound. It's doing weird stuff like asking to dial a 1, when I see that it DOES dial a 1. |
| 21:05.21 | chefrs | I think its dialing too quickly before it gets the line. |
| 21:06.02 | hedge77 | you probably have a local context or something with _NXXNXXX or something like that right? |
| 21:06.17 | chefrs | Yeah. |
| 21:06.34 | Strom_C | chefrs: get rid of the r and w flags to Dial() |
| 21:06.47 | Strom_C | the W should go before the number if you want a pause |
| 21:06.54 | chefrs | Storm_C: Yeah. Caught that. |
| 21:07.01 | [jwb] | anyone had any problems implementing the DST fix for Polycom phones? My config matches what I've seen exactly, but my phones seem to refuse to change the time +1 hr |
| 21:07.02 | chefrs | I want it to grab the line, and then *wait* before dialing =/ |
| 21:07.04 | Strom_C | who the hell is Storm_C? |
| 21:07.13 | chefrs | err |
| 21:07.17 | chefrs | Right. Strom. |
| 21:07.46 | Strom_C | chefrs: you should have Dial(ZAP/6/www${EXTEN}) or whatever |
| 21:07.59 | [TK]D-Fender | [jwb], Show what you've got from that section of yoru config in a pastebin |
| 21:08.02 | [TK]D-Fender | ~pb |
| 21:08.03 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:08.21 | *** join/#asterisk Stridernzl (n=neville@125-239-175-26.jetstream.xtra.co.nz) |
| 21:08.34 | ping2921 | anyone has solution with moh restart from beginning whenever there is a queue announcement? |
| 21:08.35 | chefrs | Strom_C: I do. It's exten => s,n,Dial(${OUT_${DIAL_TRUNK}} yadda, so on and so forth |
| 21:08.49 | Strom_C | ok? |
| 21:08.50 | [jwb] | [TK]D-Fender: aight, standby |
| 21:08.55 | Strom_C | that's not what I told you to dial |
| 21:09.26 | chefrs | This is regular trunk dial from extensions.conf for macro-dialout-trunk. |
| 21:09.45 | Strom_C | chefrs: pastebin your extensions.conf |
| 21:09.58 | [jwb] | [TK]D-Fender: http://www.pastebin.ca/412661 |
| 21:10.01 | chefrs | http://channels.debian.net/paste/5842 |
| 21:10.24 | Strom_C | what the hell is all that? did you write those macros? |
| 21:10.29 | aydiosmio | haha |
| 21:10.32 | chefrs | ....maybe. |
| 21:10.33 | aydiosmio | that's freepbx |
| 21:10.35 | chefrs | Damnit. |
| 21:10.44 | Strom_C | LOSE |
| 21:10.45 | Strom_C | see topic plx |
| 21:10.49 | aydiosmio | only freepbx calls peers "trunks" |
| 21:10.51 | Strom_C | s/plx/plz/ |
| 21:10.58 | [TK]D-Fender | [jwb], Where are you located? |
| 21:11.18 | [jwb] | EST, NYC |
| 21:11.32 | Strom_C | EST? NYC doesn't observe daylight saving time? |
| 21:11.35 | Strom_C | wow. |
| 21:12.06 | *** join/#asterisk bkw__ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 21:12.10 | chefrs | Man, I can edit the conf files though. It'll override FreePBX. It wasn't my choice to have it installed but it is, and it works (most of the time) |
| 21:13.02 | aydiosmio | it's not required that you specify your daylight status when specifying a time zone |
| 21:13.07 | aydiosmio | you can figure that out for yourself |
| 21:13.11 | _DAW | FreePBX is shackles on the beauty of asterisk |
| 21:13.18 | aydiosmio | _DAW: hardly. |
| 21:13.23 | chefrs | _DAW: That may be your opinion. Right now I don't have a choice. |
| 21:13.23 | Strom_C | aydiosmio: well then it should be "Eastern Time" or "ET" |
| 21:13.41 | Strom_C | aydiosmio: because not every locality observes DST |
| 21:14.10 | hedge77 | timezones are dumb and we should just all run off gmt anyway |
| 21:14.19 | Qwell[] | gmt? why? |
| 21:14.26 | [TK]D-Fender | [jwb], looks legit. try changing your resynch to 1 Hr |
| 21:14.31 | [TK]D-Fender | [jwb], And reboot it |
| 21:14.55 | garreel | [Mar 27 23:14:02] WARNING[32766]: app_playback.c:437 playback_exec: ast_streamfile failed on Zap/1-1 for /var/lib/asterisk/moh/fpm-calm-river.wav |
| 21:15.24 | garreel | knows anyone why Playback won't play that file? |
| 21:15.39 | garreel | on moh it works |
| 21:16.07 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:16.19 | *** join/#asterisk ManOfMilk (n=garrett@71-210-15-25.eugn.qwest.net) |
| 21:17.13 | [jwb] | [TK]D-Fender: I've rebooted a few times, but I'll check the resync time |
| 21:17.38 | [jwb] | aydiosmio: thanks for the help :P |
| 21:17.59 | aydiosmio | I did what now? |
| 21:20.30 | [TK]D-Fender | garreel, pastebin the full CLI output of that call |
| 21:21.06 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 21:22.32 | *** join/#asterisk nikosd (n=nikosd@c-67-181-184-170.hsd1.ca.comcast.net) |
| 21:23.15 | *** part/#asterisk nikosd (n=nikosd@c-67-181-184-170.hsd1.ca.comcast.net) |
| 21:24.05 | garreel | http://channels.debian.net/paste/5844 |
| 21:24.41 | [TK]D-Fender | garreel, with VERBOSE please (10 would be nice). Although I'm pretty sure I know whats wrong |
| 21:25.06 | [TK]D-Fender | garreel, Do NOT specify file extensiosn with playback or background <- |
| 21:25.18 | garreel | asterisk -dvvvvvvvvvvr ? |
| 21:25.27 | [TK]D-Fender | garreel, See above |
| 21:25.38 | [TK]D-Fender | and you can always change verbose level at * CLI |
| 21:25.44 | garreel | ok |
| 21:26.47 | garreel | http://channels.debian.net/paste/5845 |
| 21:27.03 | garreel | doesn't show more than that |
| 21:27.23 | garreel | it says no such file... but the file is there |
| 21:27.24 | [TK]D-Fender | garreel, when I say "see above" I meant for my assessment as to what you did wrong. This more than proves it. Go fix them. |
| 21:27.47 | [TK]D-Fender | <[TK]D-Fender> garreel, Do NOT specify file extensiosn with playback or background <- |
| 21:27.52 | voltagex | if I want to code AGI in Python, which module do you guys suggest? There's at least 3 different python-asterisk bindings |
| 21:28.27 | garreel | exten => 933,1,Playback(vm-unknown-caller.gsm) |
| 21:28.27 | garreel | exten => 933,1,Playback(vm-unknown-caller.gsm) is this wrong? |
| 21:30.01 | [TK]D-Fender | garreel, remove the EXTENSION |
| 21:30.10 | chefrs | So noone can tell me how they would fix this for a non-FreePBX install? |
| 21:30.19 | garreel | do you mean to remove ".gsm"? |
| 21:30.22 | [TK]D-Fender | YES |
| 21:30.29 | garreel | :-) |
| 21:30.55 | [TK]D-Fender | chefrs, Whats the problem? |
| 21:30.58 | garreel | :-):-):-) |
| 21:31.02 | garreel | now it works :-) |
| 21:31.40 | chefrs | [TK]D-Fender: I keep getting "You must first dial a 1", but in CLI I show a 1 being dialed. The consensus was that it was dialing before the line was ready and I needed to insert a wait. |
| 21:32.10 | [TK]D-Fender | chefrs, Pastebin verbose 10 CLI output. |
| 21:32.13 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.70) |
| 21:32.14 | zeeesh | hi |
| 21:33.01 | chefrs | Oh and sometimes it also just sits there. Rings once (@ Asterisk box) and then just... dies. |
| 21:33.22 | Fieldy | I am experimenting with Festival() according to the book. It's working for me but I get a small flood of warnings on the console, and the system load skyrockets. any ideas? details: http://rafb.net/p/a5oFDC95.nln.html |
| 21:33.41 | chefrs | [TK]D-Fender: http://channels.debian.net/paste/5846 |
| 21:33.46 | ping2921 | I cannot find the queue periodic-annoucment field in queue table. ?? |
| 21:34.22 | [TK]D-Fender | - Executing Dial("SIP/1000-b7800470", "ZAP/6/13032742264|120|r") in new stack |
| 21:34.46 | [TK]D-Fender | chefrs, Remove the "indicate ringing" checkbox on this entry and add a "ww" prefix to the dial-out |
| 21:35.02 | chefrs | So "...|120|ww|" ? |
| 21:35.21 | [TK]D-Fender | chefrs, No, I said PREFIX. as in BEFORE the number |
| 21:35.50 | chefrs | Oh so "...ZAP/6/ww13035555555|120|" |
| 21:36.05 | [TK]D-Fender | chefrs, Is in the exact way Strom_C TOLD YOU TO. <Strom_C> chefrs: you should have Dial(ZAP/6/www${EXTEN}) or whatever |
| 21:36.53 | [TK]D-Fender | chefrs, I don't care what means you use to get the "ww" in there, just don't expect results until you do. |
| 21:37.15 | chefrs | Ive made the changes. |
| 21:42.07 | garreel | is Playback able to play mp3 files? |
| 21:43.05 | [TK]D-Fender | garreel, If you've installed asterisk-addons, yes |
| 21:43.10 | *** join/#asterisk oQPa (n=uawename@175.Red-83-50-32.dynamicIP.rima-tde.net) |
| 21:43.23 | garreel | ok |
| 21:43.24 | *** part/#asterisk oQPa (n=uawename@175.Red-83-50-32.dynamicIP.rima-tde.net) |
| 21:43.52 | *** join/#asterisk oQPa (n=uawename@175.Red-83-50-32.dynamicIP.rima-tde.net) |
| 21:44.30 | P4C0 | [TK]D-Fender, did you read my msg? sorry I was away (and ran out of log) |
| 21:45.21 | [TK]D-Fender | P4C0, no |
| 21:46.31 | *** join/#asterisk Fieldy (i=AVptAGQt@gentoo/contributor/Fieldy) |
| 21:46.32 | P4C0 | [TK]D-Fender, remember the problem? I add r option to the Dial command for calling outside in the externsions.conf and the problems seems to be fixed |
| 21:47.40 | P4C0 | [TK]D-Fender, now I can make calls and receive without problems, if I remove the r option I can't make calls... |
| 21:48.23 | P4C0 | [TK]D-Fender, i supposed it have something to do with the r option not sending data to the channel or something like that... does it makes any sence for you? |
| 21:48.48 | [TK]D-Fender | P4C0, No. "r" is evil and should have no impact on your ability to dial |
| 21:50.17 | *** part/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 21:50.34 | P4C0 | [TK]D-Fender, :) i have no idea then... but with r it works just fine :D I think other way in a small fraction of second ther're two channels or something... not sure |
| 21:51.09 | *** join/#asterisk xpato (n=pato@bart.it-linux.cl) |
| 21:51.30 | xpato | hi, how can i regex al alpha character? |
| 21:52.09 | P4C0 | xpato, :alpha: ? but I think that depends on what kind of regex are you using |
| 21:52.47 | aydiosmio | [a-zA-Z] is standard perl regexp |
| 21:52.58 | xpato | mmm, well im no sure if is alpha, i need to regex the + sign |
| 21:53.04 | aydiosmio | seems * has nothing that elaborate however. |
| 21:53.16 | aydiosmio | xpato: haha |
| 21:53.17 | P4C0 | {+a-zA-Z] then? |
| 21:53.36 | *** join/#asterisk JonathanW (i=funguise@c-24-10-32-18.hsd1.ca.comcast.net) |
| 21:53.40 | *** join/#asterisk kgx (n=karuna@60.234.196.160) |
| 21:54.03 | xpato | i need only the + sign, so [+] would work? |
| 21:55.11 | aydiosmio | again, it depeneds on what's paring the expression, what are you using it for? |
| 21:55.21 | aydiosmio | s/paring/parsing |
| 21:56.12 | JonathanW | two quick questions i'm getting an error and a notice warning timeout, but no rule 't' in context 'internal' and unable to create channel of type 'SIP' i'm using only SIP going from one extenction to another and it seams it cant find eachother always gives user unavailable never ring thrue tho |
| 21:57.06 | JonathanW | any clues? |
| 21:57.12 | [TK]D-Fender | JonathanW, pastebin the full CLI output of your call at verbose 10, along with your dialplan |
| 21:57.13 | [TK]D-Fender | ~pb |
| 21:57.25 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:57.28 | xpato | if i dial a number from my siphone, and there is no rule defined to call that kind of number, i get a busy tone, but i see nothing in asterisk (asterisk -r verbose 10) |
| 21:57.36 | xpato | how can i see this in asterisk? |
| 21:57.45 | MACscr | what file is the context [from-internal] using in? |
| 21:58.00 | aydiosmio | MACscr: freepbx? |
| 21:58.14 | MACscr | no, asterisk, the example is from trixbox though |
| 21:58.22 | aydiosmio | ? freepbx |
| 21:58.32 | MACscr | no, im not using freepbx either |
| 21:59.07 | aydiosmio | well asterisk only has extensions.conf for that |
| 21:59.11 | aydiosmio | so give thata whirl |
| 21:59.14 | MACscr | ok, thanks, thats what i was asking for |
| 22:01.38 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 22:03.42 | Dovid | evening all |
| 22:03.50 | JonathanW | id pastebin it but i got no way to copy paste on that box its debian console only |
| 22:04.25 | [TK]D-Fender | JonathanW, You can cut & paste witha mouse at console... |
| 22:04.31 | JonathanW | eg no mouse |
| 22:04.46 | anonymouz666 | I need to do the samething that background() do, but dynamic it's bad because background provide some facilities such as t, and i extensions |
| 22:05.00 | [TK]D-Fender | ok, gtg, later |
| 22:05.12 | *** join/#asterisk BruXo (n=celio@201.17.146.166) |
| 22:06.02 | BruXo | I need a help, do exist a syntax to group sip channels same zaptel with zap/g1 |
| 22:06.04 | BruXo | ?? |
| 22:06.38 | BruXo | I need this to made a truk with more with a sip channel |
| 22:06.51 | kovger | Mercestes, thanks for your help, finaly solwed the problem, my provider put the line in not working status for some reason, but i dialed them tomorrow, and they said its ok, now i dialled them again, and "put some presure on them", and finaly they find it on "level2" that my line is puted in not working status... sorry for all the trouble :( |
| 22:07.24 | Mercestes | NP. glad you got it figured out |
| 22:07.41 | Ast001 | can not paste inpastebin Query failure: Can't open file: 'pastebin.MYI'. (errno: 145) |
| 22:08.11 | [jwb] | uh oh |
| 22:08.14 | [jwb] | thats no good |
| 22:09.38 | Mercestes | rofl |
| 22:09.54 | Mercestes | oh how important can an MYI be anyways? |
| 22:10.56 | JonathanW | do i need to register a fwd number in iax for the extenctions to be able to dial eachother over the internet via sip? |
| 22:11.05 | JonathanW | even tho its sip only |
| 22:12.01 | voltagex | JonathanW: iax on FWD is flakey at the moment |
| 22:12.04 | voltagex | stick with sip |
| 22:12.54 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 22:14.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:14.27 | JonathanW | k |
| 22:14.36 | carrar | they do away with the dundi channel? |
| 22:15.30 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:19.12 | *** join/#asterisk nfi|ermes (n=ermsewrk@88-149-168-59.f5.ngi.it) |
| 22:20.43 | JonathanW | how do i create a channel? or is that automated via dial? |
| 22:22.17 | *** join/#asterisk jm|home (n=jm@zen.jamiem.com) |
| 22:22.17 | voltagex | err, I think so |
| 22:22.18 | voltagex | read the book, like everyone has been telling me to do |
| 22:22.44 | JonathanW | been reading it |
| 22:22.47 | voltagex | well a channel is automatically created when I use FWD |
| 22:22.49 | voltagex | so I'm assuming it will be for you, too |
| 22:22.50 | [jwb] | [TK]D-Fender: hrm, dropping the refresh time to 1800 hasn't fixed it.. unless I need to wait 1800 to see ;) |
| 22:22.54 | JonathanW | so same thought would be sip but it dont like to create sip type hrm back to pastebin |
| 22:23.12 | *** join/#asterisk jm|home (n=jm@zen.jamiem.com) |
| 22:23.48 | ManOfMilk | question, not to sound like a broken record, I only caught the last half of the strom_c/chefrs conversation.. |
| 22:24.25 | ManOfMilk | i have a SIP phone trying to dialoutbound on a Zaptel FXO card... |
| 22:24.39 | ManOfMilk | and I keep getting problems.. I put in wwww example: exten = _9XXXXXXX.,1,Macro(trunkdial,${trunk_1}/wwwww${EXTEN:1}) |
| 22:24.40 | voltagex | if I reload asterisk, will I lose any calls that are currently connected? |
| 22:24.58 | ManOfMilk | but I still can't get my SIP phone to dialoutbound, it just "clicks" at me |
| 22:26.17 | ManOfMilk | i have a feeling it's the waiting period, i'm just looking for confirmation about it. |
| 22:27.00 | *** join/#asterisk adamatomical (n=adamatom@24-116-67-252.cpe.cableone.net) |
| 22:27.14 | adamatomical | wow this place is packed |
| 22:27.18 | Fieldy | ;) |
| 22:27.44 | adamatomical | is this just a hangout and or a place for support questions? |
| 22:27.53 | ManOfMilk | depends on if they like you. lol |
| 22:28.00 | Fieldy | both really |
| 22:28.06 | adamatomical | i've come to realize that's the overarching theme on irc ;) |
| 22:30.28 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 22:30.33 | *** part/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 22:31.06 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 22:32.14 | Mercestes | voltagex, No |
| 22:33.54 | chrisknight | Does anyone have a link to a "how-to" for dial plans? |
| 22:34.05 | syzygyBSD | ~book |
| 22:34.08 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:35.09 | Ast001 | that book is not enough |
| 22:35.36 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 22:35.37 | Ast001 | I have it and still dont know why i have problems |
| 22:36.28 | Ast001 | more usful is asterisk guru and voip info org asterisk |
| 22:36.36 | Ast001 | are |
| 22:36.49 | adamatomical | I've tried those |
| 22:36.54 | Ast001 | good night |
| 22:36.58 | *** part/#asterisk Ast001 (n=uros@212-200-206-85.adsl.sezampro.yu) |
| 22:37.03 | adamatomical | I can't seem to find any resolution to my problems from that site |
| 22:37.17 | syzygyBSD | what problem |
| 22:37.18 | adamatomical | I have RTP audio, the call appears to be placed, but it never goes through |
| 22:37.29 | adamatomical | I'm attempting to make a call using .call files |
| 22:37.31 | adamatomical | http://harddrivecaddy.com/problem.txt |
| 22:38.31 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 22:38.35 | adamatomical | asterisk 1.2.16 off of debian 3.1 |
| 22:39.18 | adamatomical | [root@atomical ~]# iptables -L |
| 22:39.18 | adamatomical | Chain INPUT (policy ACCEPT) |
| 22:39.18 | adamatomical | target prot opt source destination |
| 22:39.18 | adamatomical | ACCEPT udp -- anywhere anywhere udp dpts:5004:5082 |
| 22:39.19 | adamatomical | ACCEPT udp -- anywhere anywhere udp dpt:4569 |
| 22:39.19 | adamatomical | ACCEPT udp -- anywhere anywhere udp dpt:5036 |
| 22:39.19 | voltagex | can I issue a bash command to reload asterisk? |
| 22:39.20 | adamatomical | ACCEPT udp -- anywhere anywhere udp dpts:10000:20000 |
| 22:39.22 | adamatomical | ACCEPT udp -- anywhere anywhere udp dpt:2727 |
| 22:39.24 | adamatomical | ACCEPT udp -- anywhere anywhere udp dpts:10000:40000 |
| 22:39.26 | adamatomical | Chain FORWARD (policy ACCEPT) |
| 22:39.28 | adamatomical | target prot opt source destination |
| 22:39.29 | *** kick/#asterisk [adamatomical!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (Qwell[]) |
| 22:39.40 | *** join/#asterisk adamatomical (n=adamatom@24-116-67-252.cpe.cableone.net) |
| 22:39.40 | Cybertoy | thanks |
| 22:39.52 | Qwell[] | voltagex: asterisk -rx "reload" |
| 22:40.38 | voltagex | Qwell[]: sometimes that works, sometimes it doesn't |
| 22:40.42 | anonymouz666 | Qwell: so what do you think about the server ? |
| 22:40.46 | syzygyBSD | adamatomical: I could be wrong, but it looks like it is expecting traffic on 5060 and is getting it on 17886 |
| 22:40.51 | voltagex | Qwell[]: as in, it only prints output from asterisk sometimes |
| 22:41.01 | anonymouz666 | Qwell: 2 running 2 ports instead of one running 4 full |
| 22:41.10 | Qwell[] | voltagex: whether it prints or not, it's reloading |
| 22:41.15 | voltagex | ok |
| 22:41.28 | JT | adamatomical: |
| 22:41.30 | JT | ~pb |
| 22:41.35 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 22:41.43 | voltagex | reloading won't disconnect current calls will it? |
| 22:41.47 | Qwell[] | voltagex: no |
| 22:42.02 | adamatomical | JT: sorry about that, I believe I've used that before but it seemed like a small amount of text |
| 22:42.30 | adamatomical | RTP audio doesn't go to the 5060 port does it, syz? |
| 22:42.54 | voltagex | "small amount" is <=3 lines |
| 22:43.34 | JonathanW | the only message i get now NOTICE[27794]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' |
| 22:43.48 | voltagex | JonathanW: sip debug |
| 22:44.02 | voltagex | JonathanW: there's a reason it can't create the channel |
| 22:44.06 | chrisknight | I'm looking more for dial plan examples, & what they mean... Specificly, blocking numbers... Can anyone help out? |
| 22:44.16 | JT | adamatomical: rtp traffic uses a port between udp 10000 and 20000 by default in Asterisk |
| 22:44.34 | adamatomical | that's what I thought |
| 22:44.48 | gambolputty | That's a lot of ports for only a handful of phone calls. |
| 22:45.06 | voltagex | on a softphone, you can specify the ports used |
| 22:45.21 | voltagex | JonathanW: once debug is turned on, make your call again |
| 22:46.11 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 22:46.48 | JT | gambolputty: so? you can narrow it in rtp.conf, it's no big deal anyway, is it? |
| 22:48.21 | adamatomical | There is some sort of control for RTP, correct? |
| 22:48.33 | MACscr | maybe its just me, but why are outgoing and incoming confs in different files? |
| 22:48.56 | MACscr | and why call it extensions.conf when its not really just about extensions |
| 22:49.35 | Qwell[] | MACscr: because people want it that way, and you can mix them, 2, because...yes it it |
| 22:49.41 | Qwell[] | is* |
| 22:49.50 | JT | adamatomical: i just mentioned rtp.conf... |
| 22:49.56 | adamatomical | i know |
| 22:50.01 | voltagex | JonathanW: ..... |
| 22:50.15 | adamatomical | is there a checksum that is sent back for rtp audio |
| 22:50.21 | *** part/#asterisk garreel (n=garreel@host143-34-dynamic.17-87-r.retail.telecomitalia.it) |
| 22:50.23 | adamatomical | the session is terminated right after i receive rtp audio |
| 22:50.32 | adamatomical | i'm not sure if that is indicative of my problem |
| 22:52.02 | JonathanW | well i can call myself but i can call any other ext's |
| 22:52.20 | voltagex | JonathanW: did you do sip debug? |
| 22:52.22 | voltagex | in the cli |
| 22:52.30 | JonathanW | i can call the server and have it play a file too that works fine but wont talk to eachother |
| 22:52.35 | JonathanW | ya it gives teh same error |
| 22:52.39 | voltagex | yes. what error |
| 22:52.46 | JT | does echo test work? |
| 22:52.59 | JonathanW | yep |
| 22:53.26 | voltagex | ah, maybe verbose would help |
| 22:53.50 | voltagex | set verbose 10 |
| 22:53.52 | JonathanW | asterisk -c verbose 10? |
| 22:54.02 | voltagex | try your call again, afterwards |
| 22:54.25 | voltagex | connect to the cli, type in set verbose 10, otherwise do asterisk -rvvvvvvvvvv from bash |
| 22:55.23 | JonathanW | no such command verbose 10 |
| 22:55.41 | JT | set verbose 10 |
| 22:55.45 | JT | please pay attention |
| 22:55.47 | JT | or read the book |
| 22:55.54 | JonathanW | ya |
| 22:55.56 | JonathanW | sorry |
| 22:56.04 | Sweeper | the book is political propaganda >.> |
| 22:56.08 | JonathanW | 22 hours at this now |
| 22:58.15 | MACscr | lol, looks like asterisknow doesnt like starting with blank conf files =P |
| 22:58.32 | syzygyBSD | is it bad I am modifying source without knowing what it does? |
| 22:58.34 | JonathanW | executing dial ("SIP/Nathan-2594', "SIT/Hammer||r" in new stack next line reads Notice[27935] pbx.c:1949 ast_pbx_run: timeout, but no rule 't' in context 'internal |
| 22:58.45 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 22:59.48 | voltagex | JonathanW: your extension is incorrect |
| 23:00.33 | voltagex | I'm guessung |
| 23:01.10 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 23:07.56 | voltagex | oi, JonathanW, I can't help you if you keep disappearing |
| 23:09.09 | *** join/#asterisk mrbnet (n=sureal@corpmail1.mrbnetworks.com) |
| 23:10.43 | mrbnet | how can I prevent someone from dialing somephonenumber@myipaddress and using my trunks. I have disabled annoymous sip calls. |
| 23:10.59 | Qwell[] | mrbnet: read the SECURITY file... |
| 23:11.02 | JT | then that should be the end of it |
| 23:11.05 | Qwell[] | the one you were told to read when you installed... |
| 23:11.22 | JT | if allowguest=no and there is no other way for them to get in |
| 23:11.47 | JT | using seperate contexts for different purposes is also a good idea :) |
| 23:11.48 | Qwell[] | there is also iax :p |
| 23:11.59 | JT | you can do guest iax |
| 23:12.28 | JonathanW | sorry voltagex takes time to copy from one termnial in another room to this screen in this room :-\ |
| 23:13.42 | mrbnet | Thank you for the help |
| 23:13.43 | JT | you're not copying it by hand are you? |
| 23:14.01 | JonathanW | for my extenction under incoming i have exten => 100,1,Dial(SIP/Nathan,10 under internal i have Exten => 100,1,Dial(sip/Nathan,,r) only refrences to ext 100 in extensions.conf i have |
| 23:14.12 | JonathanW | yeah by hand |
| 23:14.16 | JT | why?1 |
| 23:14.17 | JonathanW | no mouse |
| 23:14.21 | JT | err |
| 23:14.23 | JonathanW | on server |
| 23:14.35 | JT | just install sshd on the asterisk box |
| 23:14.39 | JT | and connect to it with ssh |
| 23:14.47 | JT | i assume your ircing computer has a mouse |
| 23:15.41 | JT | we cannot help you if you copy text by hand, as you are liable to make a transcription error |
| 23:16.20 | Qwell[] | liable? |
| 23:16.42 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 23:16.46 | JT | yes |
| 23:16.48 | *** join/#asterisk Tene (n=tene@poipu/supporter/slacker/tene) |
| 23:17.28 | Qwell[] | hmm, suppose so |
| 23:18.28 | JT | check the dictonary if you want ;) |
| 23:22.54 | MACscr | how do i find out exactly whate version of asterisk im running |
| 23:23.21 | JT | show version usually |
| 23:23.29 | MACscr | thanks |
| 23:23.40 | MACscr | doh, nope |
| 23:23.55 | JT | it's core show version if it's 1.4 |
| 23:24.23 | MACscr | hmm, nothing their either |
| 23:24.36 | JT | err ok |
| 23:24.56 | *** join/#asterisk ManxPower (n=manxpowe@196.sub-70-222-4.myvzw.com) |
| 23:25.03 | *** part/#asterisk ManxPower (n=manxpowe@196.sub-70-222-4.myvzw.com) |
| 23:25.31 | JonathanW | http://channels.debian.net/paste/5848. |
| 23:25.59 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 23:27.11 | JT | JonathanW: sounds like the device you are calling is busy or otherwise unavailable |
| 23:27.54 | JonathanW | however its not its says loged in ect.. all fine on both ends |
| 23:28.24 | JT | both ends can call playback fine? |
| 23:28.29 | JonathanW | yes |
| 23:28.40 | JT | and both ends can't call each other? |
| 23:28.46 | JonathanW | exactly |
| 23:29.03 | JT | pastebin your sip.conf, after masking the passwords |
| 23:29.08 | *** join/#asterisk ManxPower (n=manxpowe@196.sub-70-222-4.myvzw.com) |
| 23:29.11 | JonathanW | k |
| 23:29.24 | JT | also, what are the devices? |
| 23:29.49 | JonathanW | such as? |
| 23:29.59 | JT | the sip devices.... |
| 23:30.18 | JonathanW | each end is softphone x-lite no firewall or routers inbetween them and wan both are on static ips too |
| 23:30.29 | JT | ok |
| 23:31.06 | [jwb] | [TK]D-Fender: still here? |
| 23:32.57 | litage | why would ``asterisk -rx 'sip show peers' '' say that the remote sip server it's meant to register with is unreachable, but i can successfully register when using sipsak?: sudo /usr/bin/sipsak -U -a 'foobar' -s sip:foobar@sip.voxpak.com |
| 23:33.49 | JT | it would suggest that asterisk is unable to receive a reply from the OPTIONS SIP pickets it sends |
| 23:34.44 | voltagex | JonathanW: you can copy/paste sip.conf, don't have to copy it out by hand |
| 23:36.43 | litage | JT: why would asterisk be unable to receive OPTIONS packets? |
| 23:37.11 | JT | either the sip server is not sending them, or there is firewall or nat issues |
| 23:37.52 | JonathanW | i am copypasting it but its HUGE and my buffer on ssh isint big enought for it all still using the example config file |
| 23:37.54 | litage | JT: the sip server is definitely sending them, so it must be a firewall or NAT issue |
| 23:38.15 | litage | JonathanW: just make sure you paste to a pastebin site (eg: rafb.net/paste) rather than in the channel |
| 23:38.15 | JT | do a sip debug to see if you get anything |
| 23:38.51 | litage | JT: you mean run "sip debug" then "sip reload"? |
| 23:38.58 | JT | JonathanW: you might want to eventually cut down the sip.conf file |
| 23:39.03 | JT | or start from blank ;) |
| 23:39.10 | JT | litage: no, only sip debug |
| 23:39.16 | litage | k |
| 23:39.51 | voltagex | JT: doesn't he only need to paste the section for FWD? |
| 23:39.54 | *** join/#asterisk ctooley (n=ctooley@adsl-68-95-129-178.dsl.rcsntx.swbell.net) |
| 23:39.58 | voltagex | pastebin I mean |
| 23:40.12 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 23:40.18 | JT | voltagex: well if there's anything uncommented elsewhere, it could also be relevant |
| 23:40.22 | JonathanW | http://channels.debian.net/paste/5849 |
| 23:40.26 | JonathanW | i choped it down |
| 23:40.32 | ctooley | I have applied the asterisk.patch from the app_rx/txfax package. I have spandsp installed. I am trying to configure asterisk to use spandsp but I can't seem to make that work |
| 23:40.53 | voltagex | JonathanW: where's the FWD config? |
| 23:40.55 | JT | bindaddr-24.10.32.70 |
| 23:41.16 | JonathanW | ? |
| 23:41.28 | JT | servlookup=yes |
| 23:41.33 | JT | those are all invalid lines |
| 23:41.42 | voltagex | bindaddr=24.10.32.70 |
| 23:41.57 | voltagex | srvlookup=yes |
| 23:42.03 | voltagex | correct me if I'm wrong JT |
| 23:42.06 | voltagex | :P |
| 23:42.19 | JT | yeah |
| 23:42.21 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:42.26 | JT | also if the softphones are on the lan |
| 23:42.32 | JT | it's bad to bind to the public ip only |
| 23:42.33 | JonathanW | there all wan |
| 23:42.40 | JT | ah ok |
| 23:42.49 | JT | umm |
| 23:42.54 | JT | there's no usernames in the nathan and hammer entries |
| 23:43.02 | JonathanW | so remove the bind line and servlookup |
| 23:43.02 | JonathanW | ? |
| 23:43.24 | chrisknight | Can someone tell me how to block 1900 numbers? |
| 23:43.43 | JT | srv is largely irrelevant, bind shouldn't make a difference if that ip is on the same subnet |
| 23:44.04 | JT | you should add a username= to nathan and hammer |
| 23:45.38 | litage | JT: looks like you're right. with "sip debug" enabled, asterisk sends out plenty of OPTIONS packets, but no responses seem to come back |
| 23:45.54 | litage | do OPTIONS packets use a port other than 5060? |
| 23:46.16 | JT | no |
| 23:46.21 | *** join/#asterisk deeperror (n=deeperro@adsl-69-208-36-40.dsl.sfldmi.ameritech.net) |
| 23:47.25 | litage | JT: here're the details of one of the OPTIONS packets that asterisk spat out. is there anything "wrong" or unusual about it?: http://rafb.net/p/8iYKdJ34.html |
| 23:49.19 | litage | JT: might it be that the SIP server is trying to send the response back to asterisk's private IP address? |
| 23:49.33 | JT | litage: not that i can tell at a glance |
| 23:49.48 | JT | while that's possible, most ITSPs don't have retarded SIP servers anymore |
| 23:49.55 | JonathanW | http://channels.debian.net/paste/5850 |
| 23:50.04 | JT | can you run a packet sniffer at your gateway? |
| 23:50.16 | litage | JT: the sip server that asterisk is trying to register with is SER |
| 23:50.16 | voltagex | JT: FWD does that, so it's retarded |
| 23:50.28 | JonathanW | thats with out the bind and srv... verbose 10 |
| 23:50.28 | JonathanW | and username= |
| 23:50.48 | JT | litage: ser needs the nathelper modules setup and running i believe |
| 23:51.08 | litage | JT: yeah, we've got that going |
| 23:51.25 | Fieldy | could someone recommend some open source SIP soft phones? i've tried twinkle, it has a lot of severe problems. ekigia (or however it's spelt) is decent but prone to crashes and hangs left and right. x-lite is quite nice but isn't open. |
| 23:51.35 | JT | voltagex: there's a lot of people using FWD, there's probably a way to make it work |
| 23:51.42 | litage | JT: i have ip phones with private IP addresses on my local subnet registered successfully with the same SER server |
| 23:51.58 | litage | Fieldy: SJphone works quite well |
| 23:51.58 | JT | Fieldy: there's not much arround |
| 23:52.20 | Fieldy | JT: Yeah, i'm finding that out much to my own annoyance :) |
| 23:52.27 | litage | though i'm not sure if SJphone is OSS or just freeware |
| 23:52.33 | JT | softphones in general are crappy |
| 23:52.35 | Fieldy | i'll take a peek, thanks litage |
| 23:52.39 | litage | np |
| 23:52.40 | Fieldy | JT: i'll say... |
| 23:52.54 | JT | use a decent sip hardphone, the difference in audio is huge |
| 23:53.33 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 23:53.47 | Fieldy | yep. i'm aware of them but don't have any way to pay for them currently :) in the future |
| 23:54.19 | Fieldy | basically a hardphone is a standalone unit right? i mean all it needs is power and a net connection and it's set to go? |
| 23:54.25 | *** join/#asterisk hexa (n=hexa@modemcable189.156-200-24.mc.videotron.ca) |
| 23:54.38 | litage | Fieldy: yep |
| 23:54.43 | Fieldy | gotchya |
| 23:54.46 | JT | Fieldy: right, and some do PoE others use an AC adapter |
| 23:54.49 | litage | Fieldy: though the correct term is "IP phone" |
| 23:54.55 | Fieldy | mkay |
| 23:55.10 | JT | or sip phone |
| 23:55.15 | JT | unless it's sccp ;) |
| 23:55.22 | voltagex | crap, no dialtone anymore |
| 23:55.30 | voltagex | dialtones! come back!!!11one |
| 23:55.42 | Fieldy | an IP phone might be an alternative to what I was thinking (trying to be able to call to/from usa/philippines with real phones), and instead just get us each an IP phone. no DID crazyness then |
| 23:56.08 | Fieldy | s/with real phones/ with real phones + DID/ |
| 23:56.14 | JT | you'll still need a server if one of you is behind nat |
| 23:56.17 | Fieldy | hah cool |
| 23:56.28 | Fieldy | yep. i have the server up and running right now, works great |
| 23:56.37 | voltagex | cool |
| 23:56.47 | voltagex | s/cool/jbot sucks/ |
| 23:56.51 | Fieldy | heh. |
| 23:57.06 | Fieldy | peen peen |
| 23:57.11 | voltagex | I reckon IRC clients should have sed support |
| 23:57.12 | Fieldy | s/peen/splat/ |
| 23:57.16 | Fieldy | s/peen/splat/g |
| 23:57.19 | Fieldy | oh very smart |
| 23:57.24 | litage | JT: the asterisk box's gateway is an ADSL modem-router. i swapped it for a different modem-router, and asterisk was immediately able to register with its SIP server (SER), so i'm 99% sure the problem is a firewall or NAT issue on the router. however, i've re-checked the router's port forwarding config 10 times |
| 23:57.33 | voltagex | where the hell has my dialtone gone |
| 23:57.46 | JT | litage: what about qualify, that now works? |
| 23:58.10 | *** part/#asterisk deeperror (n=deeperro@adsl-69-208-36-40.dsl.sfldmi.ameritech.net) |
| 23:58.16 | litage | JT: qualify works only with the alternative router |
| 23:58.20 | Fieldy | hrm. so i'm able to call digiums PBX through my asterisk setup using an SIP softphone. what would I need to do to be able to accept SIP calls from outside my server/domain/realm/(?) ? |
| 23:58.29 | JT | litage: the one where everything works? |
| 23:58.55 | litage | JT: yes. if i change the router, qualify works |
| 23:59.00 | JT | right |
| 23:59.08 | JT | i really hate little hardware routers |
| 23:59.23 | JT | among other things, not being able to packet dump is ridiculous |
| 23:59.23 | Fieldy | the quality of the consumer level stuff has really plummeted lately |
| 23:59.34 | voltagex | SHIT |
| 23:59.36 | Fieldy | more and more of them are stunning examples of crap |
| 23:59.43 | voltagex | erased my iax and sip.conf |
| 23:59.48 | Fieldy | er uh oh |
| 23:59.54 | Fieldy | backups o_O ? |
