| 00:00.22 | drclue | The behavior is 100% consistent |
| 00:00.29 | manxpower | If you're using a pre-built or GUI wrapper to Asterisk we really are not qualified to help. |
| 00:01.05 | manxpower | do you even know what version of Asterisk you are using now .vs. what version you were using? |
| 00:01.22 | bmoraca | That's just a copout because manxpower doesn't know what's wrong :P |
| 00:01.34 | bmoraca | just kidding, obviously :P |
| 00:01.37 | drclue | The FreePBX web GUI is something I can use or not use at will |
| 00:01.44 | manxpower | bmoraca: I suspect he upgraded to a version of Asterisk that has a bug. |
| 00:02.12 | bmoraca | manxpower: seems likely...agi has always seemed pretty hit-or-miss to me |
| 00:02.21 | LiNeTuX | current PIAF "stable" puts you on 1.4.21.2 |
| 00:02.28 | LiNeTuX | FWIW |
| 00:02.39 | drclue | I run 1.6 |
| 00:02.44 | manxpower | bmoraca: I found AGI to be pretty stable, but I've never found Dial via AGI to be a good thing. |
| 00:03.22 | bmoraca | drclue: doesn't PIAF use freepbx? if so, there's your problem...freepbx doesn't support 1.6 yet |
| 00:04.16 | drclue | BS , , my PIAF distro ISO came with 1,6 |
| 00:04.49 | bmoraca | must be magic...but i wasn't aware that they updated freepbx to work with 1.6 yet |
| 00:04.52 | manxpower | bmoraca: I've been forced to do some FreePBX stuff recently. It's not as distasteful if you don't actually have to manage it. |
| 00:05.34 | manxpower | I still think the GUI is horribly confusing. |
| 00:06.01 | bmoraca | manxpower: i use freepbx in my hosted asterisk deployments...i know what its limitations are and what i need to do to get around them, but the conveniences are just too great to be wothout |
| 00:06.39 | manxpower | The only reason I can see for it is if you want noob normal users to be able to manage it themselves. |
| 00:06.55 | manxpower | You're not a PBX admin if all you can do is click on web pages. |
| 00:07.14 | drclue | The GUI is fine. I mostly root around in the xxx_custom.conf includes , so I can use the GUI , or edit text files , whichever trips my trigger |
| 00:07.51 | bmoraca | manxpower: people (customers) like to see features, whether they intend to use them or not. they like to see that they can log in and read their voicemail on the web or set up find-me-follow-me by themselves or see that they can set up conference bridges, whether or not they ever actually intend to do any of it. showing them lines of code just doesn't have the same effect |
| 00:08.07 | LiNeTuX | drclue: the problem is that the files that FreePBX writes are not 'asterisk std' stuff... so folks in here might not know the specifics of each distro and the little 'bugs' each one attempts to 'fix'. |
| 00:08.15 | manxpower | bmoraca: I agree with that. n00b users that don't know anything find GUIs useful. |
| 00:08.59 | bmoraca | manxpower: i never let the customers log in to the admin guis...but freepbx has a (somewhat) neat user front-end that works pretty well. |
| 00:09.11 | manxpower | LiNeTuX: my problem with "debugging" with FreePBX is that a single simple call generates a zillion lines of CLI output, most of the important stuff is hidden in AGI scripts. A single call when trying to debug something should generate a couple of lines of CLI output at the most. |
| 00:09.47 | LiNeTuX | maxpower: tell me about it. just stay logged into the GUI and get annoyed while what you were looking for is now 100 lines above you. |
| 00:10.55 | drclue | Well, I'm a programmer of 30+ years, but a noob to asterisk (1 month). I've setup my sip trunk (sipgate) , worked my way through getting my WiFi phones working with NAT , doing all that iptable stuff, and even right now I can dial the extensions 2001 to 2000 and get perfect two way audio. The only thing that has stopped working correctly is the native bridging that occurs in FastAGI dials between these extensions which has gone |
| 00:11.15 | bmoraca | freepbx isn't terrible if you understand that you're in their little box. HOWEVER, their little box is a LOT bigger than some other GUIs' little boxes. anyone here ever looked at EvolutionPBX? |
| 00:11.19 | *** join/#asterisk engrxyz (n=sfgsfgs@92-237-248-183.cable.ubr07.basl.blueyonder.co.uk) |
| 00:12.55 | drclue | I let the FreePBX gui thing take care of the simple basic stuff. Each FreePBX generated file includes an include XXX_custom.conf entry to allow one to hack whatever else , so it's fine |
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| 00:13.57 | drclue | I suspect that it is simply something that occurred while I was trying to setup the Digium card that got plugged in today |
| 00:14.49 | manxpower | drclue: that does not address my main issue, which is debugging at the cli is next to impossible with GUIs |
| 00:15.16 | drclue | Perhaps I tripped over something in the zapata.conf while working out the setup for the card |
| 00:15.34 | LiNeTuX | drclue: I told you where the problem is. But feel free to try and fix it on your own. |
| 00:16.55 | drclue | LiNeTux : you said it is in the sip_nat.conf file, which has never had anything in it. |
| 00:17.21 | LiNeTuX | drclue: then go fix it and be done with your one-way audio issues. |
| 00:17.41 | drclue | What do you think I should be putting in that file? |
| 00:18.48 | LiNeTuX | drclue: I just sent it to you out of the kindness of my heart :) |
| 00:19.28 | ReDNeQ | why are there such large FULL.logs |
| 00:20.53 | LiNeTuX | wants a MiFi |
| 00:21.45 | manxpower | LiNeTuX: http://cradlepoint.com/ |
| 00:21.54 | manxpower | the original "mifi" |
| 00:23.06 | LiNeTuX | manxpower: I have something like that. It's a DLink (I think) that takes the PC-Card 3G from AT&T/Verizon. |
| 00:23.12 | manxpower | The Cradlepoint CTR350 I use is great. |
| 00:23.27 | manxpower | LiNeTuX: same thing. I feel the Cradlepoint is a better product, but they are similar |
| 00:23.46 | LiNeTuX | manxpower: I use it for testing SIP phones all the time. Sometimes the registration falls out, tho :) |
| 00:23.52 | coppice | a cradle with a point sounds dangerous |
| 00:24.18 | manxpower | coppice: it's for poking the carrier in the eye. |
| 00:24.23 | LiNeTuX | manxpower: I do have issues with TFTP with mine... ever use it with yours? Mine refuses to work with TFTP. |
| 00:25.13 | manxpower | LiNeTuX: nope. They are not perfect. For example the one I have won't let me portforward to a network not directly connected to the router, but is accessable by the router. (i.e. another gateway on the LAN side) |
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| 00:26.22 | drclue | manpower: Gosh LiNeTux was able to answer my question without even needing to consider the GUI |
| 00:26.31 | LiNeTuX | heh |
| 00:28.24 | drclue | I of course hate both the GUI and the .conf files equally , so I try and hide out in FastAGI/AMI where the air is much cleaner |
| 00:28.45 | manxpower | *shrug* Just hide out on a channel dedicated to the product you use. |
| 00:28.59 | drclue | Ya mean "asterisk"? |
| 00:29.13 | manxpower | I mean #PBXinaFlash or whatever their channel is. |
| 00:29.29 | drclue | FastAGI/AMI is a feature of Asterisk , not some distro or add on |
| 00:29.42 | manxpower | hanging out on #asterisk when using a PBX GUI is like hanging out on #DOS and asking Windows 98 questions. |
| 00:29.58 | drclue | Actually I never asked a FreePBX question |
| 00:30.18 | LiNeTuX | drclue: unfortunately your problem was with sip_nat.conf, not FastAGI |
| 00:30.19 | drclue | And the answer was not a FreePBX answer either |
| 00:30.22 | manxpower | drclue: Didn't you say the fix was in sip_nat.conf? |
| 00:30.41 | drclue | Yup , which is not a FreePBX thing AFAIK |
| 00:31.03 | *** join/#asterisk geneticx (n=geneticx@adsl-146-67-165.mia.bellsouth.net) |
| 00:31.03 | manxpower | it's certainly not an Asterisk thing |
| 00:31.21 | geneticx | hi everyone. |
| 00:31.22 | LiNeTuX | drclue: actually that is a FreePBX thing. |
| 00:32.01 | drclue | Oh well , I'm sure those lines could have been put in an Asterisk file somewhere too |
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| 00:35.03 | jaytee | I'm sure that you're sure but I'm unsure if anyone else is sure about what you're sure about. |
| 00:35.19 | drclue | This is part of why I'm running to FastAGI/AMI as fast as I can, so that whatever I make will work with Asterisk without regard to the distro |
| 00:41.38 | LiNeTuX | ha ha ... man accused of groping Minnie Mouse @ Disney ... http://www.wftv.com/news/20348258/detail.html |
| 00:43.59 | coppice | Minnie Mouse has breasts? Next we'll be finding she has a brain |
| 00:44.24 | LiNeTuX | I guess dude was trying to find out for himself. |
| 00:44.45 | coppice | Is there a get out clause for research? |
| 00:45.05 | LiNeTuX | heh. might be a good defense. |
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| 02:40.59 | geneticx | what's a good "BYOD" service provider that has competitive international calling rates? |
| 02:41.32 | geneticx | any feedback on voipvoip ? |
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| 02:54.10 | cryptanthus | Hi. Can someone point me in the direction of how to set up outbound calling in a dialplan that selects from a number of possible available lines. I have 4 lines that I want to use for outbound calling. I would like the system to check if a line is in use, if so, then check the next line. |
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| 03:15.53 | [TK]D-Fender | cryptanthus: what kind of "lines"? |
| 03:17.21 | ricko73 | cryptanthus: assuming analog lines, put the channels in a group and dial using Zap/G1 |
| 03:22.26 | ricko73 | [TK]D-Fender: do you know anything about the more recently added dial tone detection on analog lines? |
| 03:22.50 | cryptanthus | [TK]D-Fender: Hello. They are analog lines. I have 4 analog lines plugged into a Sangoma A200 pc-e card. |
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| 03:24.28 | [TK]D-Fender | cryptanthus: then as ricko73 suggested, set a group # in your channel definitions and dial the group |
| 03:25.10 | cryptanthus | [TK]D-Fender: I have the Asterisk - Future of Telephony book but I don't see an example of setting up a group. Can you point me somewhere to see an example. |
| 03:25.28 | [TK]D-Fender | cryptanthus: group=1 |
| 03:25.29 | ricko73 | cryptanthus: dialing Zap/g1 starts in top down order (1,2,3,4) while Zap/G1 starts from the highest channel number (4,3,2,1) |
| 03:26.19 | ricko73 | cryptanthus: are you using zaptel or dahdi? |
| 03:26.20 | [TK]D-Fender | cryptanthusAdd before your Chann => line and it will applt to them, then dial(dahdi/g1/12345......) |
| 03:26.31 | cryptanthus | ricko73: dahdi |
| 03:26.45 | [TK]D-Fender | add "group=1" |
| 03:27.04 | ricko73 | cryptanthus: then you need to edit /etc/asterisk/chan_dahdi.conf |
| 03:27.09 | [TK]D-Fender | Channel* |
| 03:28.44 | cryptanthus | ricko73, [TK]D-Fender: Hold on a sec please, I need to access the box asterisk is running on to see if I follow you. |
| 03:32.40 | cryptanthus | ricko73, [TK]D-Fender: Sorry about the delay are you guys still there. |
| 03:33.42 | cryptanthus | ricko73: Are we talking about editing chan_dahdi.conf? |
| 03:35.10 | [TK]D-Fender | Yes |
| 03:36.54 | cryptanthus | [TK]D-Fender: I'm looking at my chan_dahdi.conf file, at the top it says that it was auto generated by wancfg_dahdi do not hand edit. At the bottom there are four entries with a context=from-zaptel each one of these sections also has group=0. Does this mean then, that they are already in a group? |
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| 03:39.57 | cryptanthus | [TK]D-Fender: So if I follow you guys right, an local outbound call entry may be.... exten => _9NXXXXXX,1,Dial(Dahdi/g0/${EXTEN:1}) |
| 03:40.10 | [TK]D-Fender | cryptanthus: if that appears before a channel => line then yes |
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| 03:41.07 | ricko73 | cryptanthus: you should use Dahdi/G0 and not g0 |
| 03:41.22 | cryptanthus | [TK]D-Fender: Yes it does. |
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| 03:41.34 | ricko73 | you want outbound dialing to be reverse order of inbound dialing |
| 03:42.11 | cryptanthus | ricko73: That makes sense. Thanks guys. |
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| 04:13.50 | verywiseman | what is "usedistinctiveringdetection" in zapata.conf meaning? |
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| 04:25.53 | verywiseman | what is "usedistinctiveringdetection" in zapata.conf meaning? |
| 04:29.05 | [TK]D-Fender | exactly whaat it sounds like |
| 04:33.47 | [TK]D-Fender | brb |
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| 05:43.49 | ZX81 | hi all - anyone able to help with a system on which zttest is no longer working (i.e. no results) |
| 05:44.12 | ZX81 | The modules load fine, hpec loads fine, ztcfg -v works fine |
| 05:44.18 | ZX81 | zttest just shows nothing |
| 05:44.28 | ZX81 | I've seen it before but can't remember what I did to fix it |
| 05:44.35 | ZX81 | recompiled asterisk, zaptel |
| 05:45.16 | ZX81 | hmm no interrupts though |
| 05:45.18 | ZX81 | <PROTECTED> |
| 05:47.21 | ZX81 | maybe it was noapic or something I did last time |
| 05:49.17 | toasterisk | it shows no interrptus |
| 05:49.20 | toasterisk | is 0 |
| 05:49.54 | toasterisk | the driver is conflicted with something else |
| 05:50.18 | toasterisk | definatly, it can not be 0 |
| 05:50.19 | ZX81 | nothing on same interrupt |
| 05:50.21 | ZX81 | yeah |
| 05:50.28 | ZX81 | was working before restart though |
| 05:50.31 | ZX81 | for like 3 years |
| 05:50.32 | ZX81 | :) |
| 05:50.41 | toasterisk | are you running misdn? |
| 05:50.44 | ZX81 | nope |
| 05:50.53 | toasterisk | netject? |
| 05:51.00 | ZX81 | nope |
| 05:51.00 | ZX81 | :) |
| 05:51.26 | toasterisk | maybe there is a problem with your PCI |
| 05:51.44 | toasterisk | take it out and plug back |
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| 05:51.51 | ZX81 | would be quite amazing to happen just after a restart - more likely kernel was upgraded or something |
| 05:52.01 | ZX81 | can't take it out - is 500Km away :D |
| 05:52.41 | ZX81 | I did recompile zaptel in case kernel had changed |
| 05:52.59 | drmessano^ | remove the old kernel and old kernel source |
| 05:53.28 | ZX81 | in case it's compiling against wrong source? |
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| 05:55.27 | toasterisk | if you suspect the zaptel, you can try to recompile that. |
| 05:55.47 | toasterisk | yes, maybe it against the code |
| 05:55.49 | ZX81 | yep done - even did an svn up && make clean && make && make install && make config |
| 05:56.46 | ZX81 | thing is, if I replace wctdm with ztdummy I get results from zttest -v |
| 05:57.03 | ZX81 | which kinda makes me think something is either up with the wctdm driver or with the card |
| 05:57.14 | dandate2 | anyone offering voip colo, i already got all my own did |
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| 06:27.54 | psiforce | hi does anyone know how to get polycom vvx video phones working with asterisk 1.4 |
| 06:29.44 | kb3ien | i remember that video=yes must be decommented in sip.conf does that help? |
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| 06:32.45 | J4k3 | anyone here familiar with app_rpt? |
| 06:40.58 | psiforce | kb3ien: ya have that already but video calling still doesn;t work |
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| 06:44.21 | kb3ien | hrm, do both devices you are calling from and to, support the same codecs, are they enabled in sip.conf? (after that i'm afraid i'm not much use to you). |
| 06:45.43 | kaldemar | psiforce: what does not work? take a sip debug of a call and pastebin it. |
| 06:46.06 | psiforce | kb3ien: I have 2 polycom vvx 1500 |
| 06:46.22 | psiforce | both can call each other using direct ip dialing |
| 06:47.10 | psiforce | but when calling with asterisk in the middle only voice works and not video |
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| 06:51.41 | kb3ien | hrm, sounds like a codec issue. i'm not up on that phone. |
| 06:51.45 | kb3ien | sorry. |
| 06:53.23 | psiforce | ya I think its a bug with video codec negotiation under asterisk |
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| 07:06.20 | TimRiker | psiforce: can you try enabling all codecs in asterisk? or alternatively only enabling the one you expect to use? |
| 07:07.04 | psiforce | TimRiker: ya I tried disabling all and only enabling h264 and also enabling them all, still no joy |
| 07:08.02 | kaldemar | psiforce: if you really want help, show the sip debug and sip.conf |
| 07:08.54 | TimRiker | when the voice call is in progress, are the phones talking directly to each other? ie: do they handle stun, and/or other nat traversal? |
| 07:15.32 | *** part/#asterisk psiforce (n=mark@124-168-126-29.dyn.iinet.net.au) |
| 07:16.05 | *** join/#asterisk Shail9211 (n=IceChat7@220.225.96.24) |
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| 07:22.44 | Shail9211 | I've problem on trunk |
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| 07:23.01 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 07:42.02 | dandre | hello, |
| 07:42.37 | Shail9211 | Hi |
| 07:42.56 | Shail9211 | could u plz help me regarding trunk ?? |
| 07:42.56 | dandre | I am still trying to get callerid informations displayed on my analog phone connected to a tdm800 fxs port |
| 07:43.32 | dandre | describe your problem |
| 07:43.51 | Shail9211 | I have two asteriksnow boxes connected with IAX2 trunk. Configured by help of FreePBX |
| 07:44.04 | Shail9211 | Everything is working fine but CID is not |
| 07:44.13 | Shail9211 | the peer system does not show the CID, instad of CID it would showing IAX trunk ID |
| 07:44.42 | Shail9211 | the peer system does not show the CID |
| 07:44.58 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:49.50 | dandre | you'd better ask to #freepbx |
| 07:50.00 | dandre | I don't know it |
| 07:52.35 | Shail9211 | Thanks |
| 07:56.56 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
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| 08:08.48 | Faustov | doh |
| 08:08.52 | Faustov | I reported an issue to sangoma |
| 08:08.58 | Faustov | and all they want is root access now |
| 08:09.00 | Faustov | damn! |
| 08:14.20 | TSM | whats the sangoma issue? |
| 08:15.55 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
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| 08:26.31 | *** join/#asterisk ZX81 (n=ZX81@121-72-128-252.dsl.telstraclear.net) |
| 08:27.29 | ZX81 | hi all - anyone know why zaptel would stop getting interrupts after a restart? |
| 08:28.51 | ZX81 | likely that the machine has been updated, but even recompiling zaptel doesn't fix it |
| 08:29.10 | ZX81 | or even - can I keep the same HPEC license stuff with DAHDI? |
| 08:29.15 | *** part/#asterisk toasterisk (n=zhulizho@58.251.230.1) |
| 08:29.51 | ZX81 | :) |
| 08:30.24 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 08:46.40 | ZX81 | ok, I've managed to get dahdi installed - but still no interrupts |
| 08:49.08 | tzafrir_laptop | ZX81, what device? |
| 08:49.20 | ZX81 | tdm400 |
| 08:49.42 | tzafrir_laptop | restart of what, exactly? |
| 08:50.29 | TSM | if i have an incomming call and i blind transfer it to someone, that other person sees the incomming CID and not the CID of the person that transfered the call, this is all good, but if they do a normal transfer then hangup, it still shows the CID of the person that transfered the call, is there anyway to detect that they transfered then hungup then change the CID to the end user? |
| 08:51.04 | *** join/#asterisk Strogg (n=jean@unaffiliated/strogg) |
| 08:52.00 | ZX81 | tzafrir_laptop: restart of machine |
| 08:53.24 | tzafrir_laptop | The module found the card and reports channels? |
| 08:53.34 | Faustov | TSM: problem is that asterisk does not detect answer or hangup at all, despite setting koolstart |
| 08:54.29 | ZX81 | tzafrir_laptop: yah |
| 08:55.04 | tzafrir_laptop | That's odd |
| 08:55.12 | TSM | Faustov: i thought it did, the phone should send a BYE message for hangup |
| 08:55.47 | Faustov | TSM: i'm not quite sure what is missing, i did everything via wancfg_dahdi |
| 08:56.05 | Faustov | and it configured everything without warnings |
| 08:56.15 | Faustov | yet when someone calls in from the analog line into asterisk |
| 08:56.19 | ZX81 | ringing digium tech - see what I can do :) |
| 08:56.28 | Faustov | he gets indication as if no one answered |
| 08:56.43 | TSM | have you setup your inbound catchall route? |
| 08:57.09 | Faustov | while i can see that asterisk gets the call, starts playing the ivr messages |
| 08:57.24 | ZX81 | meh - no luck |
| 08:57.26 | Faustov | you mean the context to which it falls into? yeah, and I redirect that to an IVR |
| 08:57.56 | TSM | have you got asterisk -r -vvvvv output? |
| 08:58.48 | ZX81 | here's cat /proc/dahdi/1: |
| 08:58.49 | ZX81 | http://pastebin.com/m96e0e57 |
| 08:58.56 | ZX81 | all modules show up |
| 08:59.00 | ZX81 | hpec loads fine |
| 08:59.10 | ZX81 | no errors in dmesg/messages etc |
| 08:59.16 | ZX81 | asterisk won't start |
| 08:59.47 | ZX81 | <PROTECTED> |
| 09:00.03 | ZX81 | and box is 500Km away :) |
| 09:00.15 | TSM | ZX81: i guess this card did work orginaly? have you checked if dhadi is running? |
| 09:00.26 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 09:00.31 | ZX81 | :) yep worked about an hour ago |
| 09:00.37 | ZX81 | before restart |
| 09:00.55 | TSM | ZX81: have you tried another restart, or a cold restart? |
| 09:00.59 | ZX81 | yep |
| 09:01.02 | ZX81 | and recompile |
| 09:01.05 | ZX81 | and svn up |
| 09:01.10 | ZX81 | and kernel up |
| 09:01.14 | TSM | ZX81: you did a cold restart, machine off and then on |
| 09:01.21 | ZX81 | shutdown -r 0 |
| 09:01.26 | ZX81 | best I can do from 500k |
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| 09:02.07 | TSM | ZX81: i would try a cold restart, ive seen some cards not work after a warm restart, albit not TDM cards, usualy sound or nics |
| 09:02.20 | ZX81 | :) long drive |
| 09:02.47 | TSM | ZX81: invest in a APC remote rebooter or a server with ILO |
| 09:02.51 | TSM | :) |
| 09:02.54 | ZX81 | wish I could do shutdown -r 0 --after-wait-30 |
| 09:03.10 | ZX81 | yeah :) apcupsd --restart-machine-soon |
| 09:03.16 | ZX81 | :D |
| 09:03.47 | TSM | ZX81: possably, its a gamble though if your bios is not set to turn on via USB/Serial or loss of power |
| 09:03.59 | ZX81 | yeah |
| 09:04.26 | Faustov | TSM: yeah, usually verbosity 15 |
| 09:04.50 | Faustov | [Aug 11 10:56:46] WARNING[25666]: chan_dahdi.c:5116 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 4 |
| 09:04.55 | Faustov | this is the part i'm getting |
| 09:06.24 | TSM | FAustov: https://issues.asterisk.org/view.php?id=5304 |
| 09:07.07 | TSM | FAustov: https://issues.asterisk.org/view.php?id=136 |
| 09:07.59 | ZX81 | anyone seen: ACPI Exception (processor_core-0818): AE_NOT_FOUND, Processor Device is not present [20070126] in dmesg before? |
| 09:08.37 | TSM | FAustov: looks like the card is possably not setup correctly with line voltages and impedance etc... |
| 09:09.32 | tzafrir_laptop | Faustov, "State 6": Up (in the middle of a call) |
| 09:09.33 | Faustov | TSM: thx, reading through those |
| 09:09.51 | Faustov | tzafrir_laptop: what do you mean? |
| 09:10.34 | tzafrir_laptop | Some state machine (e.g. at the driver or in Asterisk) has gone wrong |
| 09:10.36 | BeeBuu | what are the steps of install asterisk 1.6? libpri-->dahdi-->asterisk,is it right? |
| 09:11.22 | Faustov | let me try the callprogress=no |
| 09:11.25 | ZX81 | hey tzafrir_laptop, 2.6.23.17-88 should be ok for kernel? |
| 09:11.37 | tzafrir_laptop | As a result Asterisk knows it is in a middle of a call, but at that time gets a ring for an incoming call |
| 09:12.02 | TSM | Faustov: the line has gone off hook for some reason but it does not know why, if you look at http://www.asterisk.org/doxygen/1.4/chan__dahdi_8c.html line 4544 it shows the code that gives that message |
| 09:12.14 | tzafrir_laptop | ZX81, I'm not familiar with it. I don't see why it shouldn't be |
| 09:12.23 | ZX81 | kk ty |
| 09:14.30 | ZX81 | interesting, if I use jiffies for current_clock_source rather than acpi_pm or tsc then "time sleep 1" never returns |
| 09:14.59 | ZX81 | tsc and acpi_pm are both fine |
| 09:15.29 | Faustov | TSM: ok, quite clear ast->state should be in state ring and it is not, but what conclusion can I get from there and what action? |
| 09:20.24 | ZX81 | yep, well that was stupid - now the box doesn't respond |
| 09:20.28 | Faustov | ok, the callprogress thingy is obviously for something else |
| 09:21.04 | TSM | Faustov: did this work before? or first time setup? |
| 09:21.17 | Faustov | no, first time |
| 09:31.33 | Faustov | ok, what was further mentioned in these bug reports |
| 09:31.37 | Faustov | busydetect no |
| 09:31.44 | Faustov | and combinations with callprogress=no |
| 09:31.47 | Faustov | didn't help |
| 09:32.46 | *** join/#asterisk arpu (n=arpu@62.178.159.144) |
| 09:35.46 | *** join/#asterisk jgoo (n=r3rman@athedsl-4536173.home.otenet.gr) |
| 09:36.37 | jgoo | I have an openvox 4 port isdn card - 3 isdn lines in - first one isdn terminal adapter wasn't having a link - now it does - how can I make the card 'refresh' or relink? I can dial the number, it rings, but the card no longer picks up (passes a call to asterisk) |
| 09:37.01 | jgoo | It was working, until yesterday, when the link light went out - and I'd like to fix it without restarting the PBX (this works, usually) |
| 09:40.09 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 09:45.22 | jgoo | I rebooted... it worked. I can't help think there is some awesome way of doing it that tastes like WTP |
| 09:57.34 | Faustov | tzafrir_laptop: after some reading i think my problem originates from different voltages than specified in protocols - but those are regulated by koolstart/groundstart etc, is this correct? |
| 09:57.43 | dandre | which is the best practice between NoOp(some debug info) and Verbose(3,some debug info) ? |
| 10:07.45 | TSM | Faustov: i think i said that earlier |
| 10:08.41 | *** join/#asterisk af_ (n=getsmart@88-149-230-210.dynamic.ngi.it) |
| 10:08.45 | TSM | Faustov: there are standard line voltage & ring voltage, if the ring voltage is too low it will think the phone is always off the hook |
| 10:09.00 | TSM | Faustov: if you get impedance wrong then you will get bad quality calls and echo |
| 10:09.26 | Faustov | TSM: ok I see, how can I control that? |
| 10:09.41 | TSM | Faustov: i duno how to setup analogue cards in asterisk, my only exp with thoes things was using ATAs |
| 10:09.47 | TSM | Faustov: what card is it? |
| 10:09.54 | Faustov | sangoma a200d |
| 10:12.10 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 10:12.37 | viraptor | has anyone here got experience with "avaya ip office"? |
| 10:15.28 | TSM | Faustov: did you remember to set your country in chan_dhadi.conf? |
| 10:16.33 | tzafrir_laptop | TSM, country in chan_dahdi.conf ? which specific setting? |
| 10:16.52 | TSM | loadzone |
| 10:17.21 | TSM | if the card does not know what country its in then it will use the wrong settings, thats what i was told |
| 10:17.39 | tzafrir_laptop | loadzone is /etc/dahdi/system.conf |
| 10:17.54 | TSM | nop its /etc/asterisk/chan_dhadi.conf for me |
| 10:19.42 | tzafrir_laptop | $ strings /usr/lib/asterisk/modules/chan_dahdi.so | grep loadzone |
| 10:19.42 | tzafrir_laptop | Unable to play dialtone on channel %d, do you have defaultzone and loadzone defined? |
| 10:20.09 | tzafrir_laptop | Nothing in the code to parse that keyword |
| 10:20.58 | tzafrir_laptop | Repeat the trick with dahdi_cfg |
| 10:21.44 | TSM | http://wiki.sangoma.com/wanpipe-linux-asterisk-debugging#analog_debugging |
| 10:22.09 | TSM | this is all good info as ile be getting a A200d soon but FXS ports |
| 10:25.50 | Faustov | TSM: first thing I tried |
| 10:26.56 | Faustov | got the zone set up to the same thing i got in indications.conf (pl) |
| 10:37.16 | *** join/#asterisk TommyBotten (n=tommy@217-14-12-26-dhcp-osl.bbse.no) |
| 10:37.29 | *** join/#asterisk Iain_ (n=IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 10:37.53 | Iain_ | Morning all |
| 10:39.54 | TommyBotten | Good morning |
| 10:39.59 | TommyBotten | Or afternoon for some of us :p |
| 10:40.24 | Iain_ | Lol fair enough |
| 10:40.44 | Iain_ | I have the following problem if you could help me please |
| 10:41.00 | Iain_ | I here no audio on one of sip phones when making an external call, the phone is in a remote location he has a router/firewall that appears to have all the correct ports open, the firewall our end also appears to have the correct ports open, the user can make and receive internal calls ok |
| 10:41.50 | Shail9211 | There might be natting issue |
| 10:42.04 | TommyBotten | Definetly sounds like a NAT issue. |
| 10:42.11 | Shail9211 | yep |
| 10:42.30 | Iain_ | that's my thoughts, but I'm unsure how to configure it |
| 10:43.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 10:44.25 | naif | Hi all |
| 10:44.31 | Shail9211 | sip is not NAT friendly proto, your application or architacture shd handle this |
| 10:45.59 | naif | I am using 2 mobile VoIP client on Symbian S60 with AMR-NB 4.75 and an Asterisk 1.6 server acting as a media relay (canreinvite=no). I am having problems as Asterisk, that should act as a simple media relay, does not accept AMR codec in negotiation. I would like to avoid to apply the AMR patch, as i don't need transcoding but simply have asterisk act with basic PBX features as a media relay. |
| 10:46.03 | naif | Does anyone have some ideas? |
| 10:47.40 | Shail9211 | Is your phone's gateway pointing to right gateway ?? |
| 10:56.19 | naif | no one there? |
| 10:57.56 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-23-32.w90-56.abo.wanadoo.fr) |
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| 10:59.13 | merlin8282 | Hi. I need to make an estimation on how many time I would need to set up an Asterisk in a company, and what hardware to use. |
| 10:59.37 | merlin8282 | For the hardware it's OK, it's only 10 persons in the company so the low-end PC is sufficient. |
| 10:59.54 | merlin8282 | But I don't know how many time it could take for setting it up. |
| 11:00.17 | merlin8282 | Can anyone help me ? |
| 11:00.21 | TommyBotten | merlin8282: Just focus the system on stability. Raid1, redundant UPS for instance. |
| 11:00.48 | TommyBotten | That very much depends. Is it internal only, or will you be calling the outside world?. If so, how? VoIP, E1/T1? .. Analog? |
| 11:00.59 | kaldemar | merlin8282: first define what you want asterisk to do for you |
| 11:01.05 | merlin8282 | Maybe analog or EuroISDN |
| 11:01.55 | merlin8282 | kaldemar: say for example a simple installation, only incoming/outgoing calls, transfers, voicemail. A basic system. |
| 11:02.55 | TommyBotten | An experienced asterisk technician could probably do that in 5-10 hours time. |
| 11:03.05 | merlin8282 | I also need advice on hardware for the line: actually i have played with junghanns QuadBRI ISDN and a TDM400P |
| 11:04.06 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 11:04.09 | merlin8282 | There may be better hardware, but I don't know which is good and which is not. |
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| 11:15.10 | shido6 | join #trixbox |
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| 11:17.56 | bbkt-trix | 5/part |
| 11:17.58 | *** part/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
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| 11:32.26 | SALstar | How I can call my local number using manager API? I have an application, which is working well, bug calling SIP/localnumber ends with "loop detected". |
| 11:33.18 | SALstar | When trying to call Local/localnumber, then I can call, but call is not accepted. I am trying to run SendFAX/ReceiveFAX this was (to send fax from one number to another). |
| 11:33.30 | SALstar | Faxing to non local numbers work well. |
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| 12:33.17 | dandre | I am still with my callerid display on an analog phone problem. I have tried almost all what is possible with the callerid parameters in zapata.conf and nothing help me. Where can I get tips for this? |
| 12:34.48 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
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| 12:37.01 | [TK]D-Fender | dandre: PASTEBIN <- |
| 12:37.07 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:37.52 | Faustov | I think I got a clue with my analog-asterisk problem. Looks like there is something wrong set up by the provider on the analog line (I need to send #21 before using the line every time) |
| 12:38.19 | [TK]D-Fender | Faustov: Sound like some sort of Centrex |
| 12:38.57 | Faustov | Centrex? never heard of |
| 12:38.59 | tzafrir_laptop | dandre, where is that line? What country? |
| 12:39.07 | Faustov | well, waiting for the provider to come back to me |
| 12:40.29 | dandre | tzafrir_laptop: I have no problem geting cid from my analog line (which is in france) but sending a callerid information to an analog phone connected to a fxs port of a tdm800 |
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| 12:48.14 | aurax | sup folks |
| 12:48.58 | aurax | can anyone help out? i'm trying to add prefix to incoming calls in IAX2 trunk, tried to customize my context but did no good. |
| 12:49.08 | dandre | here is my zapata.conf: http://pastebin.fr/5281 |
| 12:58.58 | manxpower | aurax: exten => _1NXXNXXXXXX,1,Goto(99${EXTEN},1) would add 99 to all incoming calls to whatever context that line is in. To do it ONLY to IAX2 calls would be a little more complicated. |
| 12:59.00 | *** join/#asterisk eliel (n=eliels@200.61.172.61) |
| 12:59.45 | eliel | hello, when trying to test skypeforasterisk i am getting a notice saying: "Found a total of 0 Skype For Asterisk licenses" |
| 13:00.26 | eliel | the license is in /var/lib/asterisk/licenses and expires 2009-08-31 |
| 13:01.04 | *** join/#asterisk hammerzone45 (n=hammer15@c-71-229-108-12.hsd1.fl.comcast.net) |
| 13:01.05 | aurax | manxpower, i should put context=from-something and push this line there, no ? |
| 13:01.45 | aurax | manxpower, my incoming numbers on the iax2 trunks are XXX so should i put XXX instead? |
| 13:02.00 | manxpower | aurax: yes |
| 13:02.06 | tzafrir_laptop | dandre, callerid=asreceived is not such a grand idea if you want to send callerid |
| 13:02.54 | tzafrir_laptop | dandre, try: callerid = Dan Dre <123> |
| 13:03.00 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 13:03.27 | aurax | alright, trying it right now :) |
| 13:04.00 | hammerzone45 | i have a problem with queues kicking our people automatically if they are BUSY can you please look at code http://pastebin.com/d5e486adc |
| 13:04.12 | aurax | manxpower: i was trying to use this command exten => _XXX,1,Set(CALLERID(number)=8${CALLERIDNUM}) |
| 13:04.17 | aurax | but it failed.. |
| 13:04.35 | dandre | tzafrir_laptop: just done but still doesn't work |
| 13:04.54 | aurax | which is actually better since all i have to do is rewrite the CID |
| 13:04.58 | [TK]D-Fender | aurax: ${CALLERIDNUM} does not exist in 1.4+ |
| 13:05.25 | [TK]D-Fender | auxPlease realize you are using the new function in this sample... only HALF of the time. |
| 13:05.27 | tzafrir_laptop | one thing to do would be to record the audio at the dahdi level with dahdi_monitor |
| 13:05.32 | [TK]D-Fender | aurax: Please realize you are using the new function in this sample... only HALF of the time. |
| 13:06.30 | aurax | so it should be: exten => _XXX,1,Set(CALLERID(number)=8(CALLERID(number)) ? |
| 13:07.21 | *** join/#asterisk shido641 (n=shido@mail.logical.co.za) |
| 13:07.25 | [TK]D-Fender | aurax: Close. Go read the instruction on how to reference variables & functions again |
| 13:07.54 | [TK]D-Fender | And I shouldn't say "new". 1.4 was introduced 3 YEARS ago |
| 13:07.57 | shido641 | hello everybody |
| 13:08.12 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:08.28 | hammerzone45 | any idea why queues kicks people out automatically when BUSY even if autologoffunavail = no in agents.conf |
| 13:09.23 | shido641 | can anyone help me with a problem with configuration files for dahdi?? |
| 13:10.08 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 13:10.27 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:10.48 | *** join/#asterisk [netman] (n=netman@65.Red-88-19-164.staticIP.rima-tde.net) |
| 13:11.39 | dandre | how can I get more debug informations? |
| 13:11.39 | *** part/#asterisk eliel (n=eliels@200.61.172.61) |
| 13:12.02 | hammerzone45 | dandre: core set debug XX |
| 13:12.08 | shido641 | anybody? |
| 13:12.24 | [TK]D-Fender | shido6Ask a specific question already... |
| 13:12.34 | [TK]D-Fender | shido6 : Nobody likes a fishing expedition. |
| 13:13.01 | dandre | hammerzone45: sorry for my callerid on my zap fxs port |
| 13:13.02 | shido641 | well ok heres the thing i need to configure a b410p card to accept incoming and outgoing calls |
| 13:13.54 | shido641 | how would i configure my system.conf chan_dahdi.conf extensions.conf and any other configuration files that need be configured |
| 13:15.30 | manxpower | shido641: For things like extensions.conf there is plenty of documentation for that. Read it. For the stuff like DAHDI people here might be able to help. |
| 13:15.49 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 13:16.04 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 13:16.23 | shido641 | im new to asterisk so i really dont know what to type in these files. I got asterisk, libpri, and dahdi running and i used a whole lot of tutorials i found on the net (which was really outdated) but anyways i need help with the dahdi configuration files |
| 13:16.36 | shido641 | for my card that is |
| 13:17.20 | hammerzone45 | anyone that masters queues here? i have a very specific question |
| 13:17.46 | shido6 | ask hammerzone45 |
| 13:18.17 | hammerzone45 | my queues are kicking out people imisiatelly after they log in .... |
| 13:19.06 | hammerzone45 | i have autologoffunavail = no in agennts.conf .... |
| 13:19.19 | hammerzone45 | and the CLI output is in here ... http://pastebin.com/d5e486adc |
| 13:19.50 | hammerzone45 | why a queue will auto log off an agent is status is BUSY? |
| 13:20.22 | shido641 | hammerzone45 could you help me out with this problem? |
| 13:20.28 | dandre | I have this when I do a core set debug channel Zap/2-1: |
| 13:20.29 | dandre | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/2-1] |
| 13:20.51 | hammerzone45 | shido641: no using dahdi here, still using zaptel, sorry |
| 13:21.14 | shido641 | argh ok thanx anyways ;) |
| 13:24.41 | shido641 | can anybody atleast point me to a page on the net that got instructions for dahdi configuration with a b410p card? |
| 13:25.32 | shido641 | coz i really cannot find ANY articles/tutorials on dahdi with b410p card for incoming and outgoing calls |
| 13:25.34 | [TK]D-Fender | hammerzone45: pastebin the ENTIRE configs |
| 13:25.54 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:26.25 | shido641 | AFK |
| 13:29.41 | *** join/#asterisk voipmonk (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 13:30.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:32.17 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-166-138.dsl.stlsmo.sbcglobal.net) |
| 13:32.39 | LemensTS | on 1.6, do i need to install zaptel/dahdi and load ztdummy if all im doing is itsp? |
| 13:32.48 | *** join/#asterisk moy (n=moy@74.12.123.137) |
| 13:32.49 | LemensTS | didnt know if it needed timing only with a card or not |
| 13:34.16 | [TK]D-Fender | LemensTS: Cards provide timing via DAHDI. Timing is required for specific things |
| 13:34.53 | LemensTS | i know it is for t1/e1 |
| 13:35.00 | LemensTS | didnt know what else |
| 13:35.41 | [TK]D-Fender | LemensTS: as a mixer source for IAX2 trunk mode and Meetmte |
| 13:36.16 | hammerzone45 | D-Fender: i do not have access to the entire configs, but i can certanly paste the parts that you are interested to see out of them, will that work? |
| 13:37.25 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 13:37.55 | [TK]D-Fender | hammerzone45: Why not? |
| 13:38.11 | *** join/#asterisk youngproguru (n=youngpro@74.10.229.45) |
| 13:39.11 | LemensTS | TKD-Fender: thanks, i remember the meetme thing uses it now. |
| 13:39.15 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-166-138.dsl.stlsmo.sbcglobal.net) |
| 13:42.29 | shido641 | anyone know any books i can get thats upto date on asterisk and dahdi? |
| 13:43.15 | [TK]D-Fender | shido6read the docs in the tarballs |
| 13:43.35 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:43.49 | tzafrir_laptop | moy, here? |
| 13:44.04 | moy | tzafrir_laptop: yep |
| 13:44.56 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:46.33 | shido641 | anybody can explain this? Whats a bchan and a dchan? What do they mean by this i see it in one configuration file....lol sorry im new to asterisk so its noob questions |
| 13:46.49 | Qwell | ~book |
| 13:46.50 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:46.52 | [TK]D-Fender | shido6: ISDN BRI = 2B+D |
| 13:47.11 | tzafrir_laptop | shido641, s/dchan/hardhdlc/ |
| 13:47.11 | [TK]D-Fender | shido6: E1 ISDN PRI = 30B+D |
| 13:47.43 | tzafrir_laptop | dahdi_genconf should generate a proper conf for you :-) |
| 13:48.16 | shido641 | ok thanx D-Fender but i didnt understand that lol |
| 13:48.40 | shido641 | thanx for books |
| 13:48.55 | *** join/#asterisk brad_mssw (n=brad@216.155.101.90) |
| 13:49.17 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 13:49.24 | [TK]D-Fender | shido641: The books should not have to tell you how your telephone line works... |
| 13:49.52 | [TK]D-Fender | shido641: no more than your car dealer should be responsible for teaching you how to drive... its an expected pre-requisite |
| 13:50.12 | shido641 | tzafrir_laptop you say it should generate a correct configuration file for me but how would it know i want to have both inbound/outbound calls through my b410p |
| 13:51.53 | shido641 | [TK]D-Fender: i see what you saying but i am totally new to this but where could i go to read more about this channel stuff and spans or isdn lines etc? |
| 13:52.16 | [TK]D-Fender | ~101 |
| 13:52.17 | infobot | rumour has it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 13:52.46 | shido641 | thanx :) |
| 13:52.54 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:56.29 | hammerzone45 | D-Fender i do not have direct access to the box right now, what part of the config files you need? I can ask for them ... |
| 13:59.53 | [TK]D-Fender | hammerzone45: Your queues and agents. COMPLTEE |
| 14:00.15 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:00.57 | *** topic/#asterisk by Corydon76-dig -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.4 (2009/08/11), 1.6.0.12 (2009/09/11), 1.4.26.1 (2009/08/11), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 14:03.16 | [TK]D-Fender | Corydon76-dig: Why skip 1.6.1.3? |
| 14:03.17 | [TK]D-Fender | Corydon76-dig: In fact they all seemed to skip |
| 14:03.52 | [TK]D-Fender | Corydon76-dig: Oh, and 1.6.0.12 comes bundled with res_fluxcapacitor.so, right? ;) |
| 14:04.16 | dandre | does sending cid to fxs port use alaw/ulaw? |
| 14:04.21 | Corydon76-dig | [TK]D-Fender: go read the release announcement |
| 14:04.28 | jaytee | the obvious deliberate sowing of confusion and addition of new undiscovered bugs in order to ensure job security |
| 14:04.38 | coppice | [Tk]D-Fender: that's just fiction. you need res_interocitor.so |
| 14:04.55 | [TK]D-Fender | Corydon76-dig: Not on the main page yet... |
| 14:05.29 | [TK]D-Fender | coppice: today's science fiction might very well be tomorrow's science fact. |
| 14:05.29 | Corydon76-dig | [TK]D-Fender: it most certainly is |
| 14:05.57 | [TK]D-Fender | Corydon76-dig: www.asterisk.org doesn't reflect this on my side... |
| 14:06.02 | coppice | [TK]D-Fender: and the Pope will be presenting Jesus on TV one day |
| 14:06.26 | tzafrir_laptop | dandre, it normally uses FSK or DTMFs |
| 14:06.35 | Corydon76-dig | [TK]D-Fender: shift-reload |
| 14:06.39 | [TK]D-Fender | coppice: Atheism : a non-prophet organization |
| 14:07.05 | dandre | ok |
| 14:07.05 | [TK]D-Fender | Corydon76-dig: Nope <- |
| 14:07.07 | Corydon76-dig | [TK]D-Fender: first story under "announcements" |
| 14:07.18 | tzafrir_laptop | dandre, but at that specific point, the audio is encoded as alaw / ulaw |
| 14:08.14 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:08.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:08.23 | [TK]D-Fender | Corydon76-dig: Still not there... |
| 14:08.30 | [TK]D-Fender | Corydon76-dig: Opened a new browser, etc |
| 14:08.44 | dandre | ok but I haven't seen where to set alaw/ulaw for zap |
| 14:11.04 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:11.32 | hammerzone45 | D-Fender: queues.conf and agents.conf complete --> http://pastebin.com/d740fabab |
| 14:11.53 | tzafrir_laptop | dandre, Unless you did something special, it's ulaw |
| 14:12.23 | tzafrir_laptop | And it's really nothing you should care about |
| 14:12.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:13.13 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 14:13.54 | dandre | yes it is ulaw |
| 14:14.29 | dandre | I am searching an searching why my cid isn't displayed on my analog phone |
| 14:15.23 | [TK]D-Fender | dandperhaps your zone info and CID standard aren't set right |
| 14:15.29 | tzafrir_laptop | nice. It seems recent trixbox asterisk 1.6.0 packages support BRI NT PtMP in chan_dahdi |
| 14:16.13 | [TK]D-Fender | hammerzone45: Ok, I don't see the problem yet... |
| 14:16.36 | tzafrir_laptop | That is, when you configure a span with "bri_net_ptmp" signalling you don't get the expected "sux" message. All's well |
| 14:17.02 | tzafrir_laptop | You get a fully-functioning BRI CPE PtMP span |
| 14:17.32 | hammerzone45 | D-Fender: me neither, everithing looks fince in the configs .... but agents are beign kicked out immidiately after login ... |
| 14:17.48 | dandre | [TK]D-Fender: the cid works fine on fxo ports |
| 14:18.10 | [TK]D-Fender | dandFunny, i didn't SEE any in your condifgs |
| 14:18.13 | hammerzone45 | D-Fender: Asterisk send the first call to the agent imidiatelly and state is reported as BUSY so Asterisk logs out agent automatically. |
| 14:19.02 | *** join/#asterisk Lyma (n=Lyma@unaffiliated/lyma) |
| 14:20.21 | dandre | [TK]D-Fender: my zapata.conf: http://pastebin.fr/5282 |
| 14:21.22 | [TK]D-Fender | dandre: cidsignalling=v23 <- try moving it after your first channel def |
| 14:24.29 | dandre | same problem |
| 14:25.35 | dandre | no display on my phone |
| 14:36.42 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:36.54 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 14:37.30 | Skarmeth | tzafrir_laptop, hey guy, can we talk in private about openr2 and debian pkg-voip? |
| 14:37.43 | tzafrir_laptop | Skarmeth, sure |
| 14:39.49 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:41.52 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 14:42.26 | Shail9211 | Hi |
| 14:42.45 | Shail9211 | I have two asteriksnow boxes connected with IAX2 trunk. Configured by help of FreePBX |
| 14:42.55 | Shail9211 | Everything is working fine but CID is not |
| 14:43.07 | Shail9211 | the peer system does not show the CID, instad of CID it is showing IAX trunk ID |
| 14:43.23 | Shail9211 | any suggestions |
| 14:43.52 | *** join/#asterisk oberon (n=oberon@89-138-172-78.bb.netvision.net.il) |
| 14:43.54 | oberon | hi |
| 14:44.05 | oberon | I installed asterisk as root under Linux |
| 14:44.17 | *** join/#asterisk datacompboy (n=datacomp@213.187.251.250) |
| 14:44.21 | [TK]D-Fender | Shail9211: don't stet callerid on the trunk |
| 14:44.24 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:44.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:44.25 | oberon | .. and I wanna run it as a non-root user |
| 14:44.41 | datacompboy | Hi! Anybody knows where to get DTMF sound files in slin format? |
| 14:44.57 | oberon | when I use the -U arg it tells me that it cant write a pid file in /var/run |
| 14:45.17 | jasonwoot | does anyone have any experience with tollfreeforwarding.com, positive or negative? |
| 14:45.17 | oberon | how do I control where it/whether it creates the pid file ? |
| 14:45.51 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 14:46.16 | *** part/#asterisk shido641 (n=shido@mail.logical.co.za) |
| 14:48.11 | [TK]D-Fender | oberon: asterisk.conf |
| 14:48.15 | [TK]D-Fender | oberon: ... |
| 14:48.20 | [TK]D-Fender | ~asterisk-non-root |
| 14:48.21 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 14:49.01 | [TK]D-Fender | datacompboy: Go generate your own... |
| 14:49.49 | oberon | [TK]D-Fender, I dont want to install it in a user's home dir |
| 14:49.55 | oberon | I already installed it as root |
| 14:50.06 | oberon | I just want to control where it creates the pid file |
| 14:50.09 | [TK]D-Fender | oberon: Who said home dir? I sure didn't |
| 14:50.29 | [TK]D-Fender | oberon: and * need write permissions to a hell of a lot more <- |
| 14:50.29 | oberon | thats what it talks about at chapter 13 in the book |
| 14:51.49 | [TK]D-Fender | oberon: and there is another LINK there |
| 14:52.17 | [TK]D-Fender | oberon: And not that the same methodologies can't be tweaked to apply elsewhere anyway |
| 14:53.32 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 14:53.57 | mweichert | when an incoming call comes in, is there a default context that the call is routed to? |
| 14:54.01 | oberon | hmm, the dir is specified at asterisk.conf |
| 14:54.19 | oberon | I would also make it possible to disable pid file creation |
| 14:54.25 | oberon | not needed it I dont fork |
| 14:54.37 | oberon | not needed if I dont fork |
| 14:56.26 | mweichert | ah, I'm an idiot... the channel defines the context |
| 14:57.31 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:58.41 | oberon | changing the dir in asterisk.conf doesnt work .. it is actualyl hard coded into the bin file: |
| 14:58.49 | oberon | # strings /usr/sbin/asterisk | grep asterisk.pid |
| 14:58.49 | oberon | /var/run/asterisk.pid |
| 14:58.59 | oberon | silly, huh ? |
| 14:59.34 | [TK]D-Fender | oberon: Pastebin your asterisk.conf |
| 14:59.43 | [TK]D-Fender | ~pb |
| 14:59.44 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:00.50 | Corydon76-dig | [TK]D-Fender: the announcement is now up on the site |
| 15:00.58 | oberon | http://pastebin.com/m5ca89dd8 |
| 15:01.05 | Corydon76-dig | apparently, there's a cache for people who aren't logged in |
| 15:01.29 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 15:01.38 | oberon | [TK]D-Fender, and I get: |
| 15:01.39 | [TK]D-Fender | feels like a second-rate citizen now... |
| 15:01.55 | oberon | # asterisk -U asterisk -G daemon -f |
| 15:02.01 | [TK]D-Fender | oberon: http://pastebin.com/m6d091145 |
| 15:02.02 | oberon | Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 15:02.02 | oberon | Unable to bind socket to /var/run/asterisk.ctl: Permission denied |
| 15:02.02 | oberon | [Aug 11 18:02:45] NOTICE[25368]: loader.c:869 load_modules: 1 modules will be loaded. |
| 15:02.32 | Nugget | heh, the 911 in that url triggered my context highlighting. |
| 15:02.51 | Nugget | wish I could do channel-specific keywords |
| 15:02.51 | carrar | You've been DENIED! |
| 15:03.35 | carrar | must |
| 15:03.35 | carrar | get |
| 15:03.37 | carrar | coffee |
| 15:03.44 | oberon | [TK]D-Fender, now it works |
| 15:03.52 | [TK]D-Fender | oberon: OMGZ |
| 15:04.04 | oberon | hmm, kinda non standard, but I should have read the fine print |
| 15:04.20 | oberon | "OMGZ" ? |
| 15:04.35 | oberon | "Oh My God a Zebra" ? |
| 15:05.41 | *** join/#asterisk af_ (n=getsmart@88-149-230-210.dynamic.ngi.it) |
| 15:05.43 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:06.08 | [TK]D-Fender | oberon: Wit a "Z" y0. |
| 15:06.15 | [TK]D-Fender | sheesh... |
| 15:07.05 | tzafrir_laptop | Oberon May Get a Zebra? |
| 15:07.22 | oberon | I like that one better |
| 15:08.34 | *** join/#asterisk voipmonk (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 15:09.16 | *** join/#asterisk s14ck (n=s14ck@190-76-84-86.dyn.movilnet.com.ve) |
| 15:09.20 | s14ck | hello! |
| 15:09.50 | s14ck | somebody have a dial plan where use ExternalIVR() |
| 15:10.13 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 15:14.08 | Qwell | s14ck: sure. try this |
| 15:14.13 | Qwell | s,1,ExternalIVR() |
| 15:14.19 | [TK]D-Fender | :D |
| 15:15.42 | s14ck | Qwell, I want to know if ExternalIVR() can exec agi's or bin apps in a local asterisk 1.6.2 |
| 15:16.28 | *** join/#asterisk af_ (n=getsmart@88-149-230-210.dynamic.ngi.it) |
| 15:17.20 | mweichert | when I receive an incoming call from a trunk, the context is from-internal - but my trunk definition looks like this: http://pastie.org/580085 |
| 15:18.29 | TSM | n3glv: you played much with HP server kit? |
| 15:19.04 | TSM | woops wrong place |
| 15:19.23 | *** join/#asterisk adamb0122 (n=Adam@140.239.216.61) |
| 15:20.55 | adamb0122 | So, I've got an interesting problem that i've never seen before, and i'd figure it'd run it past you guys before i do a System reboot in the middle of the day... |
| 15:21.01 | adamb0122 | ( http://nopaste.com/p/a9VIetaI1 ) |
| 15:21.49 | adamb0122 | I have a custom function for ChanSpy, it asks the extension, and then spys it. Mostly used by our managers & some trainers during a sales persons' frist few weeks, standard drill. |
| 15:22.30 | adamb0122 | Anyway, Today, for whatever reason, The system isn't asking "Please-enter-the" nor is it waiting for the READ line, it just right to user disconnected, and I have no idea why |
| 15:22.42 | adamb0122 | Haven't changed this code, or really anything on the phone system in months. |
| 15:23.34 | [netman] | hi, I got a lot of WARNING[14688]: rtp.c:891 ast_rtcp_read: RTCP Read too short, any suggestions, please? |
| 15:24.07 | s14ck | Qwell, do you know if ExternalIVR() can exec one AGI script? |
| 15:24.18 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
| 15:24.21 | Qwell | s14ck: They are completely different things... |
| 15:24.28 | Qwell | s14ck: That's like asking if your toaster can make coffee |
| 15:24.49 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:26.30 | carrar | someone should make a toaster/coffee maker |
| 15:26.47 | carrar | save on counter space |
| 15:26.52 | s14ck | :( |
| 15:26.55 | s14ck | well |
| 15:27.36 | s14ck | Qwell, but I can exec a bin apps with ExternalIVR(), right? |
| 15:27.38 | [TK]D-Fender | mweichert: Clearly the incoming call is not MATCHING your peer |
| 15:27.49 | dupe | i want a coffeepot integrated into a computer ;) like a real one! |
| 15:27.56 | Faustov | I didn't know rj11 is not a standard |
| 15:28.16 | Faustov | apparently the ports on Sangoma are 1mm smaller than the regular rj11 |
| 15:29.21 | mweichert | how do I turn off autofallthrough for a particular context? |
| 15:29.44 | [TK]D-Fender | dupe: http://farm3.static.flickr.com/2368/2422765061_d05c16d33a_o.jpg |
| 15:30.02 | carrar | aha |
| 15:30.02 | coppice | the sangoma cards don't have rj11 jacks. they use rj12 due to a cockup |
| 15:30.04 | carrar | heh |
| 15:31.40 | dupe | now.. thats totally badass :) i want one! |
| 15:32.21 | jaytee | Dial-A-Latte |
| 15:33.18 | [TK]D-Fender | DTMF : Damn This Moka's Fine! |
| 15:33.28 | Faustov | coppice: http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/A200_Analog_Voice_Card.htm |
| 15:33.40 | Faustov | coppice: http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/A200_Analog_Voice_Card.html |
| 15:33.44 | Faustov | ignore the first one |
| 15:33.50 | [TK]D-Fender | Faustov: it is RJ1. Its advertised as such and comes with patch cables |
| 15:33.56 | [TK]D-Fender | Faustov: So deal with it... |
| 15:33.56 | Faustov | they say they are using rj11 |
| 15:34.34 | [TK]D-Fender | adamb0122: What ver of *? |
| 15:34.40 | [TK]D-Fender | adamb0122: What ver of *? |
| 15:34.41 | coppice | maybe they've revised recent boards. they used to take RJ12 - the handset plug from a standard phone |
| 15:35.01 | Faustov | coppice: ok, that's exactly what I got here |
| 15:35.59 | [TK]D-Fender | Faustov: "Each Sangoma A200 Card is shipped with four 2 m cables terminating in a narrow RJ11/4 plug at one end and a telephone-standard RJ11/6 plug at the other." <--- |
| 15:36.36 | Faustov | [TK]D-Fender: trying to recall what I did with the box... :P |
| 15:36.54 | [TK]D-Fender | Faustov: "Shipped with standard RJ11-terminated cables." <- implies the other end "isnt't" |
| 15:37.07 | [TK]D-Fender | coppice: still do |
| 15:38.03 | [TK]D-Fender | s14ck: the docs show that ExternalIVR has NOTHING to do with calling dialplan apps. |
| 15:39.19 | s14ck | [TK]D-Fender, What are you talking about? |
| 15:40.14 | oberon | I've setup a SIP channel |
| 15:40.27 | [TK]D-Fender | [11:27]<s14ck>Qwell, but I can exec a bin apps with ExternalIVR(), right? |
| 15:40.32 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:40.43 | oberon | and I'm trying to call my cellular from asterisk using AMI |
| 15:40.51 | [TK]D-Fender | s14ck: Go read the docs it tells you to |
| 15:40.57 | oberon | so I authenticated and now I try this action: |
| 15:41.50 | s14ck | [TK]D-Fender, http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-B-357.html |
| 15:42.04 | oberon | http://pastebin.com/d769e5d4a |
| 15:42.36 | [TK]D-Fender | s14ck: and? What does that say about it relating to any other dialplan app call? Or AGI? |
| 15:43.13 | s14ck | [TK]D-Fender, once again, What are you talking about? |
| 15:43.29 | [TK]D-Fender | s14ck: You were asking about ExternalIVR & AGI. they have nothing to do with each other |
| 15:44.11 | s14ck | [TK]D-Fender, yes, i did think i can use agis with it, but is not work |
| 15:44.20 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 15:44.27 | [TK]D-Fender | oberon: Go enable SIP DEBUG at * CLI and see what its doing. |
| 15:44.53 | [TK]D-Fender | s14ck: That does not have anything to do with AGI. It does not call any kind of apps |
| 15:45.06 | [TK]D-Fender | (dialplan-wise) |
| 15:45.09 | oberon | what I get back is: http://pastebin.com/d496e6edc |
| 15:46.04 | s14ck | [TK]D-Fender, exten => 123,1,ExternalIVR(test_program,${MYARGUMENT}) |
| 15:46.40 | [TK]D-Fender | oberon: Cause-txt: User busy <- doesn't look like they accepted your call. And I said loko at SIP DEBUG |
| 15:47.02 | [TK]D-Fender | s14ck: and? |
| 15:47.55 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:48.48 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 15:49.31 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 15:50.36 | oberon | *CLI> sip debug |
| 15:50.36 | oberon | No such command 'sip debug' (type 'help sip debug' for other possible commands) |
| 15:51.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:51.18 | oberon | hmm "*CLI> sip set debug on" works |
| 15:55.39 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:56.40 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 15:57.33 | oberon | hmm, I see the sip header + variables |
| 15:57.49 | oberon | <PROTECTED> |
| 15:58.51 | [TK]D-Fender | oberon: in yuor peer entry |
| 16:00.09 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 16:00.49 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 16:00.49 | Katty | :> |
| 16:01.00 | oberon | my peer entry ? |
| 16:01.14 | oberon | I dont see any peer header in the debug output |
| 16:01.24 | [TK]D-Fender | oberon: what you set up in sip.conf and use as your "Channel:"..... |
| 16:01.25 | Katty | HELLO ALL YOU BEAUTIFUL PEOPLE |
| 16:01.32 | [TK]D-Fender | Katty: MEW! |
| 16:02.20 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:02.48 | oberon | in the sip.conf file I setup the username + password in fromuser + secret respectively |
| 16:02.59 | oberon | the debug shouldnt show them ? |
| 16:03.04 | Katty | [TK]D-Fender: i decided on doing a vpn tunnel. |
| 16:03.09 | Katty | [TK]D-Fender: instead of proxy, me thinks |
| 16:03.22 | Katty | [TK]D-Fender: cause i'm not sure what all you mean by peer and all that other fancy stuff. |
| 16:03.55 | [TK]D-Fender | Katty: ........ |
| 16:04.10 | [TK]D-Fender | Katty: And to think you've been here for YEARS |
| 16:04.16 | Katty | yes. |
| 16:04.17 | Katty | i knows. |
| 16:04.20 | Katty | but we don't do anything /fancy/ |
| 16:04.29 | Katty | just all lan stuff. so i've never had to do anything fancy :< |
| 16:04.33 | Katty | cries. |
| 16:04.37 | Katty | k, over it. |
| 16:05.02 | Katty | i bet eppigy would help me. |
| 16:05.12 | [TK]D-Fender | Katty: "nat=yes", "canreinvite=no", "qualify=yes" <- this ain't Raw-Cat Sigh Hence. |
| 16:05.21 | oberon | to make outgoing calls using the SIP channel do I need anything except type + host + fromuser + secret ? |
| 16:05.23 | Katty | [TK]D-Fender: yeah. but. |
| 16:05.27 | Katty | [TK]D-Fender: dynamic ip. |
| 16:05.30 | Katty | [TK]D-Fender: and firewall. |
| 16:05.32 | Katty | [TK]D-Fender: and yeah. |
| 16:05.51 | Katty | [TK]D-Fender: and i'm not opening the firewall to anyone on port 5060. tho i guess i could change the port number. |
| 16:06.01 | Katty | [TK]D-Fender: do you think bandwidth would hate me for changing the port number? |
| 16:06.14 | [TK]D-Fender | Katty: Doesn't matter (THEY reg to YOU), firewall (If theirs is really bad you are kinda screwed unless you can forward past it), and OK/FINE/SURE |
| 16:06.43 | Katty | [TK]D-Fender: i still don't understand how it doesn't matter. |
| 16:07.01 | Katty | [TK]D-Fender: my firewall refuses access on port 5060, unless it is explicitly allowed from another public, static, ip. |
| 16:07.13 | Katty | [TK]D-Fender: so something's going over my head on how i can use a dymanic. |
| 16:07.23 | Katty | [TK]D-Fender: cause i ain't removin that policy. |
| 16:07.26 | oberon | [TK]D-Fender, right now my sip.conf file looks like: http://pastebin.com/d6819bf53 |
| 16:07.31 | Katty | [TK]D-Fender: si answer is NO |
| 16:07.50 | [TK]D-Fender | Katty: Oh, YOUR firewall is garbage? then you're in real trouble |
| 16:08.05 | [TK]D-Fender | Katty: fix that, and follow the guide for the rest : |
| 16:08.07 | [TK]D-Fender | ~sipnat |
| 16:08.07 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:08.10 | Katty | sighs |
| 16:08.20 | Katty | i dont' want port 5060 open to whoever wants to connect to it |
| 16:08.36 | Katty | so how else do i tell the firewall it's okay, unless tell it that static ip is okay? |
| 16:08.42 | Katty | that's the part i don't get. |
| 16:08.51 | [TK]D-Fender | Katty: So proxy it or change *'s port |
| 16:09.07 | [TK]D-Fender | Katty: And I don't know how to configure your firewall. |
| 16:09.11 | Katty | i do. |
| 16:09.18 | Katty | i think the different port number might be neat. |
| 16:09.25 | Katty | port triggering, or whatnot |
| 16:09.43 | [TK]D-Fender | Katty: * isnt' that smart. It'll bind to one... |
| 16:10.04 | [TK]D-Fender | Katty: What FW are you using? |
| 16:10.07 | Katty | [TK]D-Fender: the firewall can take port 1234 and reroute it to port 5060 |
| 16:10.30 | Katty | [TK]D-Fender: and it won't leave 5060 open to random port scanner peoples wanting to use my server to call dominos |
| 16:10.42 | Katty | [TK]D-Fender: linksys rv082 |
| 16:11.06 | [TK]D-Fender | Katty: And why is your system so insecure that it accepts unauthed calls? |
| 16:11.22 | Katty | it doesn't. |
| 16:11.22 | [TK]D-Fender | Katty: Don't you guys have a linux/Asterisk person there? :p |
| 16:11.38 | Katty | no, sorry. |
| 16:11.39 | [TK]D-Fender | Katty: then unload res_paranoia.so :D |
| 16:11.40 | Katty | please come again. |
| 16:11.46 | Katty | NEVAR |
| 16:11.54 | Katty | it's a linksys rv082 :< |
| 16:12.02 | Katty | I WILL NOT STAND FOR THIS INSANITY |
| 16:12.20 | Katty | so either port triggering, or vpn tunnel. |
| 16:12.33 | Katty | vpn tunnel might cause network traffic proxying tho |
| 16:12.44 | Katty | and that'd be slowwwwwwwwww unless you had a way to split the tunnels., |
| 16:13.20 | [TK]D-Fender | Katty: And you'll have to prevent subnet overlap, etc.... |
| 16:13.25 | Katty | yesh. |
| 16:13.40 | ariel_ | vlan |
| 16:13.48 | Katty | i can do vpn tunnels. |
| 16:14.17 | ariel_ | vpn tunnel with the correct vlan setup will work |
| 16:16.32 | Katty | tests port triggering. |
| 16:16.56 | *** part/#asterisk SALstar (i=ondrejj@work.salstar.sk) |
| 16:17.17 | *** join/#asterisk jtodd (i=jhsyma60@ns.fox-den.com) |
| 16:17.17 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:19.49 | dandre | I have a musicclass parameter in my queue.conf. I this queue is reach from a local extension, say a sipphone, it's ok but if this queue is reach from my isnd line the music falls back to defaut. |
| 16:20.25 | Katty | :>>>>>>>>>>>>>>>>> |
| 16:22.24 | Katty | i need a random port number generator tool |
| 16:22.30 | Katty | Qwell: roll me a random 5 digit number |
| 16:22.42 | Qwell | 564 |
| 16:23.11 | Katty | :< |
| 16:23.13 | Katty | that's only 3 digits |
| 16:23.25 | Qwell | it's between 0 and 99999! |
| 16:23.27 | [TK]D-Fender | hacks Katty's shitty Linksys router for 3D20 + 5 MILLION DAMAGE!!! |
| 16:23.52 | Katty | dies. (4.9999999 million overkill) |
| 16:24.33 | [TK]D-Fender | \o/ -- Victory is mine! |
| 16:24.54 | ariel_ | does not care keeps listening to 3 doors down, Here without you..... |
| 16:25.49 | *** join/#asterisk RaDiC_rs (n=quassel@bchm-4d09036d.pool.mediaWays.net) |
| 16:27.04 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:27.20 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:27.32 | jameswf | mmorpg R-Tard |
| 16:28.38 | beek | Anyone had any success compiling 1.6.0.12 chan_sip.c? I'm getting "chan_sip.c:18669: error: expected expression before \u2018<<\u2019 token" |
| 16:28.47 | Qwell | beek: hold that thought |
| 16:29.02 | [TK]D-Fender | jameswf: Computers don't sue dice.... TABLE-TOP YOU NEWB KIDDIE BASTARD! |
| 16:29.09 | [TK]D-Fender | USE* |
| 16:29.28 | beek | Qwell: I've tried both 1.6.0.10 --> patch --> .12 and downloading the new tarball. Same problem. |
| 16:29.41 | Qwell | beek: yes, it's broken. hold on. |
| 16:29.51 | [TK]D-Fender | * 1.6.0.12.8.6.7.5.3.0.9!!! |
| 16:30.04 | beek | [TK]D-Fender: you missed the -2 at the end. |
| 16:30.18 | beek | [TK]D-Fender: and good day to you sir. |
| 16:30.44 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:31.01 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:31.30 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 16:31.49 | [TK]D-Fender | beek: DOH! |
| 16:32.40 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:33.08 | jameswf | table top? oh [TK]D-Fender what level dungeon master are you? |
| 16:33.52 | *** join/#asterisk IBC_Jkenney (n=jkenney@99.23.50.73) |
| 16:33.55 | [TK]D-Fender | jameswf: Who said D&D? You Elitist Newb Bastard :p |
| 16:34.04 | jameswf | make all your ports prime numbers |
| 16:34.13 | [TK]D-Fender | jameswf: I only did TT D&D once... didn't end pretty, but did at least end quick... |
| 16:34.23 | jameswf | lol |
| 16:34.23 | Katty | i think i'll just use a random number generator off google. |
| 16:34.29 | Katty | and then change the port triggers every month or so |
| 16:34.32 | dandre | how can I see weather musicclass is set by my misdn channel? |
| 16:34.44 | Katty | and maybe the password |
| 16:34.45 | *** join/#asterisk luminforce (n=luminbla@rrcs-71-42-115-245.sw.biz.rr.com) |
| 16:34.56 | jameswf | katty you can write one in 3 lines or so what language? |
| 16:34.58 | Katty | can asterisk make some sip.conf registrations LAN or WAN only? |
| 16:35.10 | [TK]D-Fender | Katty: By "trigger" do you just mean direct translation? |
| 16:35.28 | Katty | [TK]D-Fender: requests on port 1234 will make the internal port switch over to 5060 |
| 16:35.36 | beek | Time for lunch! |
| 16:35.40 | Katty | bye beek |
| 16:35.42 | [TK]D-Fender | Katty: create a port knocking scheme instead |
| 16:35.45 | beek | CU Katty |
| 16:35.47 | Katty | [TK]D-Fender: what's that? |
| 16:35.59 | [TK]D-Fender | Katty: Google-able :) |
| 16:36.17 | Katty | you're so helpful. |
| 16:37.23 | [TK]D-Fender | Katty:allows a client to hit a series of triggered ports in a spcific order to dynamically modify FW rules |
| 16:37.45 | [TK]D-Fender | Katty: Sort of a "secret knock" to let them in normal from their IP, etc |
| 16:41.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:42.01 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:44.29 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:45.25 | ariel_ | [TK]D-Fender: wow, I had not heard of that, but looks great t/y for the info, http://www.portknocking.org/ |
| 16:45.49 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 16:49.54 | jameswf | Katty: http://pastebin.com/m1cac6d7a << random 5 figit port in c |
| 16:50.04 | jameswf | s/figit/digit/ |
| 16:50.55 | tzafrir_laptop | jameswf, but it's not a random port number |
| 16:51.33 | Qwell | tzafrir_laptop: sure it is.. 6/10ths of the time |
| 16:52.00 | tzafrir_laptop | yeah, but why not just get a single integer in the range? |
| 16:52.18 | tzafrir_laptop | in fact, $RAND is a good start |
| 16:52.24 | tzafrir_laptop | happens to be in the range |
| 16:53.16 | jameswf | I lied change line 43 to v = (rand() % 5 ) + 1; this will pull it down |
| 16:53.22 | tzafrir_laptop | err... $RANDOM |
| 16:53.24 | jameswf | *42 |
| 16:53.41 | tzafrir_laptop | which is bashism |
| 16:53.53 | tzafrir_laptop | but may be good enough |
| 16:54.31 | jameswf | tzafrir_laptop: wins as he hates bash and suggested it anyway :) |
| 16:56.52 | coppice | if you want to run an early 70s shell like the bourne shell, why don't you run it on a 16 bit machine with 32k of RAM? |
| 16:58.44 | *** join/#asterisk saisoma (n=irchon@166.137.6.186) |
| 16:59.11 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 17:00.15 | tzafrir_laptop | coppice, bash is not bourne shell. |
| 17:00.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:00.22 | jameswf | http://pastebin.com/m70af27a9 <-- thisone actualy keeps within the valid port range... |
| 17:00.29 | tzafrir_laptop | it is way bloated than that |
| 17:00.58 | coppice | tzafrir_laptop: ya don't say :-) |
| 17:01.38 | tzafrir_laptop | However it is generally a good idea not to assume /bin/sh is bash |
| 17:01.48 | coppice | all the unix shells are erally mickey mouse. why doesn't some bite the bullet and actually produce something modern |
| 17:02.25 | tzafrir_laptop | e.g. dash (the default shell of Ubuntu, and now of Debian), busybox ash, and the bsd ksh |
| 17:02.30 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 17:03.06 | tzafrir_laptop | coppice, with a C-like syntax? |
| 17:03.31 | saisoma | ksh is supposedly more c like |
| 17:03.46 | tzafrir_laptop | saisoma, s/k/c/ |
| 17:03.55 | saisoma | i've used ksh. but since I don't know c. heh |
| 17:04.08 | coppice | maybe. maybe not. but something that feels like it wasn't written within the resource limitations of the Unix V7 on a PDP11 |
| 17:04.42 | tzafrir_laptop | decent tab completion? |
| 17:05.11 | coppice | well, that a command line editing issue, which isn't really a shell issue |
| 17:05.14 | tzafrir_laptop | try the tab completion of the command gpg |
| 17:05.40 | tzafrir_laptop | what are you missing? |
| 17:06.19 | SuPrSluG | coppice: what causes a t.38 PHASEESTATUS is 48 error message PHASESTRING is Disconnected after permitted retries |
| 17:06.25 | Nugget | if I was choosing a shell today I'd probably go with zsh, but inertia is a bitch. I'll use tcsh until I die. |
| 17:06.59 | coppice | SuPrSluG: well I expect that would be that it gave up after a number of retries |
| 17:07.00 | Nugget | one time I changed bovine's shell to emacs just as a joke and he liked it and left it that way. |
| 17:07.01 | SuPrSluG | coppice: i'm trying to fax from an audiocodes fxs with t.38 |
| 17:07.32 | *** join/#asterisk awkfu (n=awkfu@166.205.5.4) |
| 17:07.46 | SuPrSluG | using callweaver for fax |
| 17:08.36 | tzafrir_laptop | Nugget, Emacs as a terminal? |
| 17:08.56 | Nugget | no, as a shell. |
| 17:09.01 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 17:11.04 | *** join/#asterisk awkfu (n=awkfu@166.205.5.4) |
| 17:11.28 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 17:13.08 | *** join/#asterisk awkfu (n=awkfu@166.205.5.4) |
| 17:19.19 | mweichert | does * have any security restrictions or DoS detection? After initiating 30 concurrent calls from one extension within 5 seconds, my SIP phone looses it's connection to the * server temporarily |
| 17:20.36 | [TK]D-Fender | mweichert: No |
| 17:21.54 | *** join/#asterisk SBBsupport (n=chatzill@208.50.100.60) |
| 17:22.42 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 17:25.33 | dupe | mweichert: if your * server is outside the local network, its probably flood protection of some sort on the firewall/routers between them.. if its internal thats weird. |
| 17:25.52 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 17:26.08 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 17:26.53 | mweichert | dupe, no external - perhaps you're right |
| 17:28.34 | crazybyte | hi!. i saw that asterisk has an embedded http server with some management features, but when i try to enable it and connect to it asterisk dies with segmentation fault. is that a known bug and what kind of management access is allowed to the asterisk using it? thank you! |
| 17:29.08 | dustybin | do nearly all VOIP service providers allow free calls if you are connect to the internet using a softphone ? |
| 17:29.14 | dupe | mweichert: voip does a lot of Packet Per Second that a lot of firewalls "helpfully" classify as attacks |
| 17:29.29 | mog | crazybyte, that shouldn't happen |
| 17:29.31 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 17:29.35 | mog | crazybyte, which version of asterisk |
| 17:29.47 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:29.58 | crazybyte | latest stable |
| 17:30.26 | crazybyte | http://pastie.org/580235 |
| 17:31.02 | crazybyte | either i'm missing something in the configs or something is broken (bug) |
| 17:31.09 | jaytee | is it just me or do the two words "latest stable" kinda strike anyone else as oxymoronic? |
| 17:31.41 | eppigy | DONDE |
| 17:31.50 | jaytee | AQUI |
| 17:31.56 | dupe | dustybin: ddepends on the plans... nothing is free, however you can usually call for free to wherever they have access to, or to others using their voip service. but 100% free calling to anywhere in the world, well doesnt work that way really :P |
| 17:32.30 | *** join/#asterisk propellerhead (n=yogurt2u@200.117.115.9) |
| 17:33.06 | crazybyte | jaytee, in that case Asterisk 1.6.1.1 built by root @ yeti on a i686 running Linux on 2009-06-15 15:17:18 UTC |
| 17:34.00 | crazybyte | jaytee, i can be also an older version considered as stable that is why i said latest stable even if it's not entirely correct from several point of view |
| 17:34.08 | crazybyte | err i can / it can |
| 17:34.12 | jaytee | crazybyte, that may seem ok to you but I'd never take a yeti's word for anything. they are notorious liars. |
| 17:34.58 | dustybin | dupe: all my friends are connected to the internet, imagine i am using VOIP, could my friends use the internet to ring me for free, rather than using a normal phone? |
| 17:35.14 | dustybin | or does it not work like that |
| 17:35.51 | dustybin | maybe my friends could use a client to connect to my asterisk server? |
| 17:36.00 | dustybin | bypass the VOIP altogether? |
| 17:36.05 | dustybin | *provider |
| 17:36.08 | crazybyte | they could but you need to have a public ip (at least) |
| 17:36.50 | dustybin | of course i will need to pay to make calls, but for my friends to ring my VOIP over the internet, there must be a free way |
| 17:36.55 | dustybin | *me |
| 17:36.58 | *** join/#asterisk werdan7 (i=w7@freenode/staff/wikimedia.werdan7) |
| 17:37.02 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 17:37.34 | crazybyte | dustybin, you can setup an asterisk server an your friend can connect to it |
| 17:37.43 | dustybin | right i see |
| 17:37.54 | dustybin | i will need to research some asterisk clients |
| 17:38.12 | crazybyte | dustybin, if you're on linux i recommend twinkle |
| 17:38.21 | crazybyte | on windows i recommend xlite |
| 17:38.28 | dustybin | most of my friends are on winblowz |
| 17:38.35 | dustybin | right ok thanks!!! |
| 17:38.38 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 17:38.56 | dustybin | maybe my VOIP provider might have some special client software what does the same thing |
| 17:39.02 | crazybyte | dustybin, i tested them both when I started playing with asterisk |
| 17:39.08 | [TK]D-Fender | dustybin: No, and no need |
| 17:39.11 | dustybin | aye excellent |
| 17:39.21 | dustybin | this is _powerful_ stuff |
| 17:39.29 | [TK]D-Fender | ~x-lite |
| 17:39.32 | [TK]D-Fender | ~zoiper |
| 17:39.33 | infobot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 17:39.39 | [TK]D-Fender | ~xlite |
| 17:39.40 | infobot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 17:39.42 | [TK]D-Fender | ^^^^ |
| 17:39.43 | crazybyte | dustybin, you don't need a provider for your friends to connect to your server |
| 17:39.53 | dustybin | the word im looking for is 'softphone' |
| 17:40.02 | dustybin | learns 2 new words, endpoint and softphone |
| 17:40.03 | [TK]D-Fender | indeed |
| 17:40.05 | [TK]D-Fender | ~softphone |
| 17:40.06 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 17:40.08 | crazybyte | yeah but the xlite for linux is very old and very buggy (at least it was when i tried it) |
| 17:40.19 | dustybin | 'sips' some tea |
| 17:40.27 | [TK]D-Fender | dustybin: endpoint is a totally generic term. Softphone far less so |
| 17:40.39 | dustybin | ok |
| 17:40.56 | dustybin | is this xlite official website |
| 17:40.56 | dustybin | http://www.counterpath.com/ |
| 17:41.22 | [TK]D-Fender | dustybin: Yes, and pay attention to the links you were already handed above |
| 17:41.35 | dustybin | ok |
| 17:43.39 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 17:43.40 | dustybin | do some VOIP providers give you the option of having 2 numbers, i need one for personal use and one for my future business |
| 17:44.01 | dustybin | and i will need asterisk to manage them both accordingly |
| 17:44.22 | drclue | voip providers will do anything you pay them for |
| 17:44.27 | dustybin | ace |
| 17:44.52 | dustybin | i thought maybe you need 2x different IPs for each new number |
| 17:44.59 | dustybin | which i wont be able to provide |
| 17:45.13 | dustybin | i mean, a IP per number |
| 17:45.18 | drclue | Nope |
| 17:45.21 | dustybin | ace |
| 17:45.22 | crazybyte | no you don't |
| 17:45.35 | dustybin | feels double excited |
| 17:45.48 | drclue | Each phone number will have it's own login so to speak |
| 17:45.50 | *** join/#asterisk MrMeek (i=MrMeek@108-23.26-24.tampabay.res.rr.com) |
| 17:45.55 | dustybin | last night i had dreams of a polycom phone |
| 17:46.27 | MrMeek | rofl must have been exciting, dusty |
| 17:46.31 | eppigy | Are you flying that plane? |
| 17:46.38 | dustybin | im nearly there, i know what voip server i will be using, asterisk, i know what VOIP hardware i will use, polycom |
| 17:46.49 | dustybin | all i need to do know is research providers |
| 17:46.57 | dustybin | is there anything i should look out for during my research? |
| 17:47.12 | ariel_ | flying a plane where? |
| 17:47.18 | drclue | It depends on what you want to d owith th phone numbers |
| 17:47.24 | [TK]D-Fender | looks for tallk buildings... |
| 17:48.00 | dustybin | drclue: first i will need a VOIP number for pure personal use, so my friends can contact me |
| 17:48.21 | drclue | Some places will give you free phone numbers with free incoming calls (like sipgate.com) |
| 17:48.34 | [TK]D-Fender | dustybin: evaluate flat vs per-minute rates, simultaneous channels, # of DID's included or at what price each,e tc |
| 17:48.42 | ariel_ | ipkall |
| 17:48.47 | dustybin | ok thanks |
| 17:48.50 | [TK]D-Fender | An he's in the UK <- |
| 17:48.54 | dustybin | yar im UK |
| 17:49.41 | dustybin | i will be ringing mobile phones, i think that will be the most expensive charge |
| 17:49.45 | *** join/#asterisk x86 (n=porteb1@p3m/member/x86) |
| 17:49.48 | drclue | If your going to have a lot of phones answering the same number at the same time for different conversations , you'll want a number with multiple channels |
| 17:49.59 | dustybin | so far i found a VOIP provider offering 12p per min to ring most of the popular cellphone networks |
| 17:50.32 | dustybin | drclue: that might happens in the future, at the moment, the line will be very quiet |
| 17:51.11 | dustybin | rining other cellphone networks on my cellphone, costs are 50p - £1 + |
| 17:51.26 | drclue | A lot of times you'll find that different providers have plans that work better in one circumstance than another relating mostly to outbound calls , and there is no problem in having multiple vendors and routing your calls to the cheapest one |
| 17:51.43 | dustybin | ace |
| 17:52.09 | dustybin | do you guys remember your first home VOIP experience when a friend rang your phone? |
| 17:52.42 | dustybin | then you put him on hold whilst some of your favourite mp3s were playing? |
| 17:52.57 | dustybin | and recorded the whole conversation :D |
| 17:53.00 | drclue | Ya , it was a net2phone conversation and I still have that ATA , although I hardly ever use it |
| 17:53.06 | ariel_ | wow, my first voip/asterisk setup was for work, I did not use asterisk/voip for at least a year after at home. |
| 17:53.34 | *** join/#asterisk errotan (n=errotan@62.201.123.54) |
| 17:54.14 | *** join/#asterisk wtsexton (n=sexton@potatosalad.worldspice.net) |
| 17:54.42 | *** join/#asterisk hudony (n=chatzill@modemcable202.250-20-96.mc.videotron.ca) |
| 17:54.59 | drclue | Well, I've been coding for 30+years , but this is my first (almost second) month using asterisk. I started with a blank hard drive and now I have trunks , NAT, and yesterday plugged in a Digium card. It's been a real learning experience |
| 17:55.04 | hudony | Hi guys, I have a question regarding asterisk + nat |
| 17:55.41 | drclue | Well , ask us the question and will all try and give you the best dis-information possible |
| 17:55.46 | hudony | ok |
| 17:56.11 | jameswf | ~nat |
| 17:56.12 | infobot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 17:56.16 | jameswf | answered |
| 17:56.26 | wtsexton | I've got an interest problem where outbound calls when answered by the remote end may not hear audio for a few seconds however doing packet captures between the machine and the metaswitch the audio is there and when recording from the switch the audio is there. |
| 17:56.56 | drclue | Could be that echotraining setting |
| 17:57.45 | hudony | From home, I have set-up dd-wrt to act as a vpn client. I can now get my config file from the ftft server without any problem so the tunnel is working fine. However, I can't authentify to the asterisk server. When I do sip show peers, I get in Host : "(Unspecified)" and Nat is set to "N" |
| 17:58.05 | wtsexton | all calls are handed via sip and rtp, no analog |
| 17:58.07 | hudony | Weird since I have specify in sip.conf nat=yes and I've put my tunnel ip address as the externip |
| 17:58.57 | drclue | For NAT , I had to both forward a bunch of ports on my remote router , and add some entries to iptables on my asterisk server |
| 17:59.17 | [TK]D-Fender | CRAZY TALK |
| 17:59.23 | *** part/#asterisk kb3ien (n=kb3ien@ool-45766a2d.dyn.optonline.net) |
| 17:59.28 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 17:59.31 | *** join/#asterisk kb3ien (n=kb3ien@ool-45766a2d.dyn.optonline.net) |
| 17:59.39 | [TK]D-Fender | ~sipnat |
| 17:59.40 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:59.46 | [TK]D-Fender | ^^^^^ Read the guide people...... |
| 17:59.49 | kb3ien | anyone got a sample of the NANP touchtone in a friendly format? |
| 17:59.59 | drclue | I also had to add some stuff to the sip configuration too |
| 18:00.54 | drclue | Of course I work real hard to keep canreinvite=yes , whereas most people turn it off |
| 18:01.12 | hudony | ok, I'll have a look at these guides |
| 18:01.54 | wtsexton | http://pastebin.com/d45543ab6 a copy of the sip session from wireshark |
| 18:02.38 | [TK]D-Fender | wtsexton: PB SIP debug from * CLI <--- |
| 18:03.39 | [TK]D-Fender | wtsexton: The COMPLETE call attempt... |
| 18:04.00 | wtsexton | ouch, I have to 'desensitize' anything I post |
| 18:05.02 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 18:05.27 | [TK]D-Fender | wtsexton: Avoid. |
| 18:05.30 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 18:06.06 | [TK]D-Fender | wtsexton: generally attempts to "sanitize" PB's end up filtering the very crap that is screwing you over. |
| 18:06.48 | *** join/#asterisk jmworx (n=jeval@mail.octasic.com) |
| 18:06.53 | wtsexton | yea |
| 18:07.13 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:09.35 | wtsexton | unfortunately for me thats not an option |
| 18:11.02 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 18:11.10 | [TK]D-Fender | wtsexton: YMMV. PB what you've got. |
| 18:11.32 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:12.47 | kb3ien | whats the quickest way to disable # features on A sip connection? |
| 18:12.50 | *** join/#asterisk oftis (n=nicok@dslb-094-217-067-045.pools.arcor-ip.net) |
| 18:13.19 | [TK]D-Fender | kb3ien: like? |
| 18:14.38 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 18:14.51 | kb3ien | user makes a call into his bank's VXML system, enters his card number pushes # and then asterisk tries to transfer the call.... |
| 18:15.06 | [TK]D-Fender | kb3ien: SetVar in the peer entry and make the dialplan use that for options on the Dial(s) |
| 18:15.20 | [TK]D-Fender | kb3ien: And why are you using DTFM transfers in the first place? |
| 18:15.26 | kb3ien | can it be disabled with dial? i'm not using any additional arguments to dial. |
| 18:15.28 | [TK]D-Fender | DTMF* |
| 18:15.44 | kb3ien | i'm about to disable it unilatterally. |
| 18:15.56 | kb3ien | where can that best be done? features.conf ? |
| 18:15.57 | [TK]D-Fender | kb3ien: Then do it... its for the best... |
| 18:16.04 | [TK]D-Fender | kb3ien: Clearly. |
| 18:16.12 | kb3ien | can it be enabled on a per call basis? |
| 18:16.24 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 18:17.37 | kb3ien | my feature map has all those options commented out... |
| 18:17.58 | dustybin | do shoretel make good phones? |
| 18:18.01 | [TK]D-Fender | kb3ien: If its not just allowed by a dial parameter (I don't see how this is possible), then now... |
| 18:18.23 | [TK]D-Fender | dustybin: Avoid. they are rebranded Polycom's, may be stuck with mGCP firmware,e tc |
| 18:18.31 | dustybin | OHH |
| 18:18.37 | [TK]D-Fender | dustybin: Save yourself the trouble. What is your realistic call usage? |
| 18:18.48 | dustybin | not that much! |
| 18:19.02 | [TK]D-Fender | dustybin: Just for you at your desk? |
| 18:19.08 | [TK]D-Fender | (near your server) |
| 18:19.13 | dustybin | [TK]D-Fender: just home use |
| 18:19.15 | dustybin | nothing major |
| 18:19.22 | [TK]D-Fender | dustybin: Linksys SPA-942 <- |
| 18:19.23 | dustybin | yeah near server |
| 18:19.29 | dustybin | aye ok |
| 18:19.44 | dustybin | [TK]D-Fender: if you look at this list http://www.shoretel.com/products/ip_phones/ |
| 18:19.46 | kb3ien | how do i disable the features unilaterally then. |
| 18:19.54 | dustybin | what does it mean by '3' lines '6' lines etc |
| 18:20.00 | [TK]D-Fender | dustybin: Polycom pricing into the UK gets hit rather hard. Yes its a better phone, but there is a rgeater question of value here. |
| 18:20.09 | dustybin | thanks dude |
| 18:20.16 | [TK]D-Fender | dustybin: lines = #of separate identities it can have. |
| 18:20.22 | dustybin | [TK]D-Fender: polycom have some kind of deal with MS? why? |
| 18:20.32 | [TK]D-Fender | dustybin: MONEY of course |
| 18:20.33 | *** part/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 18:20.35 | dustybin | ok |
| 18:20.48 | dustybin | identities, what does that mean? |
| 18:21.30 | *** part/#asterisk giesen (i=giesen@dirtypackets.net) |
| 18:21.37 | dustybin | the phone will flash green on 1 line, flash red on another line? |
| 18:21.45 | [TK]D-Fender | dustybin: thats a pretty open statement. registering as a completely different device. as in you could have your phone register to 3 completely different PBX's |
| 18:21.53 | dustybin | OHHHHH |
| 18:21.55 | dustybin | i seeeeee |
| 18:21.58 | [TK]D-Fender | dustybin: Or to the same one as 3 different ID's |
| 18:22.03 | dustybin | that is far too powerful for my needs |
| 18:22.07 | dustybin | 1 is enough |
| 18:22.50 | [TK]D-Fender | dustybin: for that aspect yes. there may be a relationship with the number of simultaneous CALLS a phone can juggle with that in mind as well. |
| 18:22.52 | dustybin | i dont mind spending a little but more for a polycom, they look good |
| 18:23.23 | [TK]D-Fender | dustybin: you were looking at the 450 last, right? |
| 18:23.26 | dustybin | i will also keep this in mind |
| 18:23.27 | dustybin | http://www.linksys.com/servlet/Satellite?c=L_CASupport_C2&childpagename=US/Layout&cid=1169083356524&pagename=Linksys/Common/VisitorWrapper |
| 18:23.35 | dustybin | [TK]D-Fender: yes |
| 18:24.18 | [TK]D-Fender | dustybin: Go find a retailer that does each brand as agressively as they can and come back to us with links for opinions. |
| 18:24.41 | [TK]D-Fender | dustybin: the Linksys SPA-9XX series is about to be replaced by the SPA-5XX series as well. |
| 18:24.48 | dustybin | http://www.polycom.com/products/voice/desktop_solutions/soundpoint/desk_phones/soundpoint_ip450.html |
| 18:25.06 | dustybin | ok thanks dude! :-) |
| 18:26.09 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:27.26 | dupe | the polycom reboot dance of hell annoys me :| |
| 18:27.32 | kb3ien | doh! there was a dial(,gTM()) in there, now its just a dial(,${__dopts}M() |
| 18:27.40 | kb3ien | i'll enable it on calls that NEED it. |
| 18:27.51 | kb3ien | as they seem to be in the vast minority. |
| 18:27.58 | wtsexton | 450 is decent |
| 18:28.09 | [TK]D-Fender | dupe: ? |
| 18:28.24 | wtsexton | price isn't much different from the 550 |
| 18:28.34 | [TK]D-Fender | wtsexton: Is a fair bit.. |
| 18:29.15 | wtsexton | ~$20 |
| 18:29.38 | dupe | when network stuff changes, ive seen polcyom 300s 600s 500s just go wonky grabbing provisioning information and just hangs on network config... reboot it several times before it grabs the configs properly |
| 18:30.18 | wtsexton | yea, that happens sometimes |
| 18:30.20 | [TK]D-Fender | wtsexton: 450 = $187 USD, 550 = $220 |
| 18:30.23 | TSM2 | 330 & 550 are fine, mabey that was the old models |
| 18:30.32 | [TK]D-Fender | wtsexton: IP 550 = mal-placed product |
| 18:31.01 | TSM2 | the 550 should have had more buttons on the screen |
| 18:31.06 | [TK]D-Fender | TSM: they're all "fine", just some at a better price-point and usability scale than others |
| 18:31.34 | [TK]D-Fender | TSM: and if the 550 had more buttons it'd be a 650, which it was already too close to in features 7 price. |
| 18:31.43 | [TK]D-Fender | TSM: not enough differentiation. |
| 18:31.58 | dupe | polycom's speaker phones rock though |
| 18:32.08 | [TK]D-Fender | dupe: entirely true |
| 18:32.16 | wtsexton | yea thats why we use polycom |
| 18:33.36 | TSM2 | i use polys, i just think the problem with most ip phones is they have a lack of BLF fields, the yeylink (somthinglikethat) have a fair few |
| 18:34.23 | wtsexton | odd setup a outbound context using spool to test and I'm not having the outbound issue like I do if I call from a phone |
| 18:34.25 | bmoraca | the polycom expansion modules work pretty well for BLF if customers absolutely need it |
| 18:35.04 | wtsexton | never messed with that outside of a directors station |
| 18:35.04 | TSM2 | yeh true, but im talking about people that want 3-4 park buttons plus a few quick dial buttons for other users |
| 18:36.07 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 18:36.49 | TSM2 | the efk stuff only the polys is good, but it would be nice if * could report stuff back on the screen like what the number is of the call just been put on hold |
| 18:37.45 | bmoraca | i just tell people that if they want a key system, they shouldn't be asking me about it. although, i am looking at the adtran NetVanta 7100 which can emulate a key system very well |
| 18:39.23 | wtsexton | yea, get that all the time |
| 18:39.39 | TSM2 | yup a hard learning curve for me to teach people how to do things on the new system |
| 18:40.12 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:40.18 | [TK]D-Fender | yealink = WIKI spamming, design-knockoffs cheap-o stuff.... would never touch |
| 18:40.33 | wtsexton | hard, sometimes beating your head on the wall gets more done |
| 18:40.56 | TSM2 | yealink looks like simmens phones |
| 18:41.17 | [TK]D-Fender | Aastra soft-keys for BLF = Godly. Too bad I hate so much about the rest of the phones.... |
| 18:41.29 | TSM2 | so what do you like? |
| 18:41.41 | bmoraca | the desktop IP phones from yealink look like linksys SPA-9xx phones |
| 18:42.27 | [TK]D-Fender | TSM: Polycom wins on quality of build, stability, configurability (scope & methods), and in North America is par on price. Aastra's BLF & side-cars = sick though... |
| 18:42.57 | TSM2 | i get a good price on them, not sure how it compares to there |
| 18:43.08 | [TK]D-Fender | TSM: Linksys work friendlier from behind NAT and are easier to configure for the basics, but scrifice on audio quality, screen usability, tec |
| 18:43.10 | TSM2 | thats what im gona get |
| 18:43.12 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 18:43.35 | TSM2 | i had no probs on the polys behind nat, took a while to work out all the DHCP VLAN etc... |
| 18:44.02 | TSM2 | i do find that the sound on the polys is slightly missing some trebble, cant work out how to increase that |
| 18:44.04 | Katty | oh man |
| 18:44.06 | Katty | lunch was awesome |
| 18:44.08 | Katty | sprawls |
| 18:44.23 | [TK]D-Fender | TSM2: there are tone parms in sip.cfg |
| 18:44.38 | wtsexton | I've never put phones behind a nat, unless the pbx was also connected to the nat |
| 18:44.58 | [TK]D-Fender | wtsexton: In cases of remote agents, teleworkers, etc |
| 18:45.07 | TSM2 | [TK]D-Fender: yup i played with thoes, mabey i did not push the values enough |
| 18:45.08 | [TK]D-Fender | wtsexton: Rare here too, but something to consider |
| 18:45.19 | wtsexton | oh, I've done that and used vpn tunnels |
| 18:45.38 | [TK]D-Fender | wtsexton: can work, but more troubles. Depends on your infrastructure, etc |
| 18:45.50 | TSM2 | roll on ebil :) |
| 18:46.18 | bmoraca | biggest issue with phones behind NAT is making sure that the router is properly configured |
| 18:46.30 | bmoraca | and powerful enough to support the number of phones you have |
| 18:46.51 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:47.12 | bmoraca | 29 phones to a server on the other side of a NAT = too much |
| 18:47.49 | [TK]D-Fender | bmoraca: that's where local proxy = win :) |
| 18:47.59 | bmoraca | yes, most definitely |
| 18:48.05 | [TK]D-Fender | bmoraca: Not that it isn't a statement of "greater trouble" by itself... |
| 18:48.18 | bmoraca | bossman didn't want me to put in any extra CPE, though... |
| 18:48.47 | wtsexton | bossman doesn't have to go home and worry with getting calls either |
| 18:48.48 | [TK]D-Fender | bmoraca: Life sucks but rarely swallows... |
| 18:48.51 | bmoraca | so i ended up configuring an IPSEC VPN to the server. and then found out that PIX software 6.2 doesn't support SIP |
| 18:49.10 | [TK]D-Fender | PIX ? EWWWWWWWWW |
| 18:49.43 | beek | seconds that opinion |
| 18:49.48 | dustybin | i just read this amazon review of the oreilly asterisk book |
| 18:49.49 | dustybin | http://www.amazon.co.uk/gp/product/0596510489/ref=s9_simz_gw_s1_p14_i1?pf_rd_m=A3P5ROKL5A1OLE&pf_rd_s=center-1&pf_rd_r=0ZTZY9NACW9P25TYBS3T&pf_rd_t=101&pf_rd_p=467198433&pf_rd_i=468294 |
| 18:49.50 | wtsexton | I've had luck with adrans |
| 18:49.51 | bmoraca | it's a PIX 525, our service provider firewall. i upgraded it to 7.0, and will upgrade it further. |
| 18:49.59 | dustybin | it doesnt look good, anybody own that book? |
| 18:50.21 | voxter | anyone familiar with a small embedded voip gateway that may run asterisk, has one rj45 jack, one fxo port, and a db9 serial interface? small grey box with green lights on it |
| 18:50.25 | voxter | couldnt find any model number or vendor markings on it |
| 18:50.30 | wtsexton | thats the old book |
| 18:50.51 | dustybin | wtsexton: that was published 2007 |
| 18:50.54 | bmoraca | wtsexton: which Adtran models do you use? |
| 18:51.01 | bmoraca | ~book |
| 18:51.02 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:51.12 | wtsexton | ta900, netvanta 13xx |
| 18:51.34 | dustybin | oh ok thanks |
| 18:51.56 | bmoraca | the customer had a 3120. i really don't like how the adtrans handle natting SIP, though. |
| 18:52.32 | wtsexton | it has an sip alg, however I've had limited use with it |
| 18:52.34 | bmoraca | the TA900s are fantastic devices, though. i'm scheduled to install 2 of them for customers in the next 3-4 weeks. |
| 18:52.46 | [TK]D-Fender | dustybin: Book is good for background and is largely applicable |
| 18:52.57 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:53.10 | wtsexton | I've got 140~ Ta900s in the field |
| 18:53.19 | dustybin | [TK]D-Fender: i will buy it! |
| 18:53.23 | bmoraca | wtsexton: the ALG is garbage. the SIP transparent proxy is better, but depending on the model of router, doesn't support more than 10-12 phones reliably. |
| 18:53.50 | bmoraca | wtsexton: you use them for sip-to-PRI media gateways or just sip-to-analog? |
| 18:53.51 | wtsexton | never really used it, expect on as a test bench, with only one phone |
| 18:54.01 | wtsexton | both and a mix |
| 18:54.28 | [TK]D-Fender | dustybin: Or just DL for now, and then buy it out of appreciation afterwards :) |
| 18:54.40 | *** join/#asterisk ebil (n=ebil@ip70-174-136-104.dc.dc.cox.net) |
| 18:54.41 | dustybin | i prefer hard copy! |
| 18:54.47 | dustybin | so i can take it into the toilet |
| 18:55.04 | [TK]D-Fender | dustybin: taht's why God created HP LaserJet Printers :D |
| 18:55.12 | dustybin | yuk, messy paper |
| 18:55.25 | bmoraca | and adobe created the 2-on-1 page printing capability |
| 18:55.27 | wtsexton | yuk, poopy asterisk book :P |
| 18:56.15 | beek | [TK]D-Fender: God did not create HP LaserJet printers. If He did, they would have more superfluous parts and would break down far more frequently. |
| 18:56.37 | wtsexton | not sure who invented them but I HATE printers |
| 18:56.38 | bmoraca | wtsexton: who do you use as a distributor for your TA900s? and are you in the USA? |
| 18:56.59 | wtsexton | yes I'm in the USA, however I don't know who we get them from |
| 18:57.04 | bmoraca | ahh |
| 18:57.14 | wtsexton | I however will attempt to find out |
| 18:57.29 | bmoraca | supply seems sketchy from techdata and ingram micro, our two main suppliers |
| 18:57.52 | wtsexton | I tell you this, Adtran rocks, you can email/call their engineers for stupid questions |
| 18:58.04 | wtsexton | and believe me, I have lots of stupid questions |
| 18:58.21 | wtsexton | try to do that with Cisco |
| 18:58.34 | bmoraca | i have no complaints about the service i've received from them...they're even going to waive my ATSP cert class |
| 18:58.41 | bmoraca | er, waive the fee for it, anyway |
| 18:59.48 | *** join/#asterisk hetii (i=576333ac@gateway/web/freenode/x-2ada0144f303a96d) |
| 18:59.53 | hetii | hi |
| 19:00.01 | wtsexton | didn't even know about atsp, things to add to the list to research |
| 19:00.09 | [TK]D-Fender | beek: I'm still trying to set my LaserJets on "KILL" |
| 19:00.23 | bmoraca | required for Internetworking specialization in the Adtran partner program |
| 19:00.31 | beek | [TK]D-Fender: What part does the killing? The LASER or the JETS? |
| 19:00.32 | [TK]D-Fender | thinks "stun" is not sufficient |
| 19:00.34 | hetii | is it possible to connect sangoma a101 to normal isdn line 2b+d |
| 19:00.38 | [TK]D-Fender | beek: yES |
| 19:01.03 | TSM2 | hetii: no |
| 19:01.10 | TSM2 | hetii: its a pri card |
| 19:01.36 | hetii | ya, just wondering |
| 19:02.34 | hetii | i have one not used and normal ISDN line from my NT box |
| 19:03.52 | *** join/#asterisk jamicque (i=jamicque@jam.bema.one.pl) |
| 19:04.21 | TSM2 | shame |
| 19:04.51 | jamicque | hi @ll, anyone has succesfully lunch spandsp with asterisk 1.6.x and send and recieve fax via t38? |
| 19:05.34 | *** join/#asterisk justsomedood (n=somedood@mail.serverplus.com) |
| 19:05.49 | *** join/#asterisk RaDiC_rs (n=quassel@bchm-4d09036d.pool.mediaWays.net) |
| 19:06.06 | justsomedood | Does anyone know if I can find the stateinterface for a queue from the manager in 1.6? |
| 19:06.13 | justsomedood | *for a queue member |
| 19:07.58 | TSM2 | has anyone used diva BRI cards with * 1.4 ? |
| 19:13.44 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 19:14.08 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 19:15.00 | [TK]D-Fender | TSM2: http://www.voip-info.org/wiki/view/Asterisk+Eicon+Diva+CAPI+ISDN |
| 19:15.40 | TSM2 | [TK]D-Fender: yes i have read that, i want to know if anyone here had 1st hand experience of using it |
| 19:16.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:17.00 | hetii | did somebody of you have running asterisk on some routers ? |
| 19:17.26 | hetii | how its work on embended device ? voice quality is ok ? |
| 19:18.50 | wtsexton | like magic, any time a by pass the metaswitch and send calls to another asterisk box with pris all my problems go away |
| 19:21.35 | [TK]D-Fender | hetii: Most can handle only a few calls. |
| 19:21.45 | [TK]D-Fender | hetii: What are your needs? |
| 19:22.01 | hetii | how many sip user you have on it and with one router you use ? |
| 19:22.57 | hetii | i just plain to build asterisk for airOS system to handle few user, general for my family |
| 19:23.11 | IBC_Jkenney | I have a AEX2430E PCI express Card for my asterisk system it comes with a 12V power connector do I need to use it? |
| 19:23.37 | wtsexton | in all cases I've seen yes |
| 19:24.47 | wtsexton | yes on that card I'm certain you need to use it |
| 19:26.05 | TSM2 | problem ive seen with loads of rack mount servers is that they dont have any molex connectors inside, ive been trying to find suppliers of external power via a PCI bracket |
| 19:26.38 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:28.09 | bmoraca | IBC_Jkenney: you need to use it if you plan on using any FXS ports. for FXO ports, you do not need to use it. |
| 19:28.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:28.22 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 19:29.00 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 19:29.48 | bmoraca | TSM2: the power distribution areas of HP ProLiant servers are very well marked...a little solder and some electrical tape will get you a molex plug. i've done it before. i'd never do it to a customer's production server, but it works :P |
| 19:31.47 | IBC_Jkenney | bmoraca i am using it to provide Dialtone so i think i need it |
| 19:32.06 | wtsexton | yea you need it then |
| 19:32.08 | bmoraca | IBC_Jkenney: that would be FXS, and yes you would |
| 19:33.06 | IBC_Jkenney | does look like my dell 2950 has an additional 12V connector |
| 19:33.11 | wtsexton | I've had bad luck with the pci-e cards |
| 19:33.40 | *** join/#asterisk Tarantulafudge (n=Michael@ip70-178-64-168.ks.ks.cox.net) |
| 19:34.08 | wtsexton | out of six I've used, four have become unstable or lost a channel, may just be my luck |
| 19:34.58 | bmoraca | i just don't like them because i don't like dealing with analog circuits |
| 19:35.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:35.45 | TSM2 | bmoraca: bodge, naa im not invalidating warranty on brand new DL380 G5/G6 servers |
| 19:36.16 | wtsexton | oh I feel that, they day faxes die, I'll be a happy person |
| 19:36.18 | TSM2 | bmoraca: im very used to bodge work, but not doing it on 24x7 servers |
| 19:36.20 | bmoraca | TSM2: like I said...i'd never do it on a customer's machine. we have a G3 DL380 and it worked great |
| 19:37.03 | TSM2 | bmoraca: my bodges in my current job are limited to splitting 4 phone extentions over a single CAT5 and the like |
| 19:37.45 | bmoraca | TMS2: nothing wrong with that, as far as I'm concerned. 4 pairs is 4 pairs. |
| 19:38.06 | bmoraca | as long as everything is punched down properly and appropriately labeled, anyway |
| 19:38.07 | TSM2 | bmoraca: yup, its cheeper than pulling more cable |
| 19:38.11 | TSM2 | haaa |
| 19:38.52 | bmoraca | i know...i've never seen a clean 66 block in my life. they're ALWAYS a hodgepodge of crap |
| 19:39.01 | wtsexton | just don't try to do the reverse, ethernet doesn't like running over a non-twisted 25 pair cable |
| 19:39.01 | TSM2 | naa my bodge for the phones, is just zip tying 4 CAT5 modules together and at the cab i have a single patch cable with all the pairs split out to seperate cat5 |
| 19:39.25 | TSM2 | im UK based, we like to cat5 everything to panels and not to have phone & seperate data ports |
| 19:39.42 | TSM2 | i noticed they like that in the US, its like that in our US offices |
| 19:39.50 | beek | TSM: All of my wiring is done that way... CAT5 only. |
| 19:39.55 | beek | Err... now 6. |
| 19:39.58 | wtsexton | yea, we do that |
| 19:40.28 | bmoraca | we do CAT5 for phoen and CAT6 for data...not sure why...the $0.05/ft savings doesn't really seem worth it to me |
| 19:40.31 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 19:40.40 | bmoraca | not my decision, though |
| 19:40.41 | beek | Total waste of money. |
| 19:40.41 | TSM2 | its thinking about the future, in our NY office there are loads of cat5 ports, but only half of them are data ports |
| 19:41.23 | beek | The big $$$ is in pulling the cable. Labor is expensive. Copper, not so much. |
| 19:41.28 | TSM2 | CAT5e more than enough for 1G, doubt there will be much 10G to the desk for many many many years |
| 19:41.40 | TSM2 | server room, CAT6 |
| 19:42.05 | bmoraca | TSM2: nothing wrong with futureproofing. i wish more of the offices I came to were set up with CAT5 for their phone, rather than CAT3, as that would make it so much easier for me to install VOIP systems...not having to deal with passthrough on the phones and such |
| 19:42.08 | jaytee | what about CAT9? |
| 19:42.19 | *** join/#asterisk Faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net) |
| 19:42.21 | [TK]D-Fender | jaytee: Lost a few lives... |
| 19:42.32 | wtsexton | yea, I love when customers have cat5 for both |
| 19:42.33 | beek | jaytee: What's that? A run and a half of CAT6? |
| 19:42.37 | bmoraca | you can get 10G over copper with CAT6a |
| 19:42.37 | jaytee | [TK]D-Fender, YES!!! knew you'd pick up on that |
| 19:43.02 | TSM2 | CAT3, is that rated to 10M, cant remember |
| 19:43.09 | jaytee | TSM, yes |
| 19:43.10 | [TK]D-Fender | <- smarter than the average bear... |
| 19:43.18 | bmoraca | technically, but i wouldn't recommend it |
| 19:43.18 | wtsexton | I know people love to have one network for everything, but life is so much easier when you keep data and voice physically apart |
| 19:43.39 | Faiz | how about a sneakernet |
| 19:43.46 | bmoraca | wtsexton: or logically with VLANs...but most people don't want to pay for the managed switches to do that |
| 19:44.03 | jaytee | back in the early nineties I went to one customer's site and they were running IBM Baseband over standard telephone wire |
| 19:44.08 | TSM2 | yup, thats the way im doing it, i need 1G to much of the office computers, so unless i get expensive switches and phones i wont get passthru |
| 19:44.09 | jaytee | 1.5mbps |
| 19:44.14 | wtsexton | yea, thats normally how I do it |
| 19:44.52 | bmoraca | jaytee: speedy |
| 19:44.56 | TSM2 | we deal with images all day, the speed increase from going up from 100 was massive |
| 19:44.57 | wtsexton | with that, if adtran would switch to lldp instead of cdp, I'd be set |
| 19:45.56 | wtsexton | until then, I just manually set the ids on the phones, I know you can do it with dhcp |
| 19:46.42 | *** join/#asterisk cusco_ (n=tralala@2001:0:53aa:64c:18bc:29e8:2ac0:762d) |
| 19:46.46 | cusco_ | hi |
| 19:46.46 | *** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
| 19:46.53 | JerJer | mooo |
| 19:47.02 | cusco_ | we were using asterisk 1.6.1.1 and now using 1.6.1.2 |
| 19:47.15 | cusco_ | pri show is no longer available |
| 19:48.12 | TSM2 | :( boo hoo :) |
| 19:48.36 | wtsexton | one of these days I'll start using 1.6 |
| 19:48.45 | bmoraca | the Adtran 1335 is the most awesome network appliance ever...24pt PoE switch, wifi AP, T1 DSU, NAT router, and firewall all built in to one |
| 19:48.53 | wtsexton | yes |
| 19:48.55 | TSM2 | cusco: is dhadi running |
| 19:48.57 | wtsexton | thats what I use |
| 19:49.22 | wtsexton | *heart* 1335PoE |
| 19:49.33 | JerJer | Every time I have tried 1.6, i've been forced to go back to 1.4 :( |
| 19:49.56 | JerJer | 1.6 has some great new features and lots of error checking but i have always found a show stopper, thus far |
| 19:49.59 | wtsexton | how is the AP controller on the 1335? thats something I've never used |
| 19:50.29 | bmoraca | never used adtran's wireless stuff before... |
| 19:50.31 | TSM2 | that stuff looks nice, but itss too much to go wrong in one box, but then built to do the job |
| 19:51.15 | wtsexton | for what its worked good for us |
| 19:51.49 | bmoraca | TSM2: it's cheaper than Cisco and of nearly equal quality and featureset. plus it's lifetime warrantied. with most of the Adtran stuff, it's cheaper to keep an inhouse spare than it would be to buy the Cisco equivalent |
| 19:51.58 | JerJer | TSM2: maybe, but adtran is usually better than rock solid |
| 19:52.02 | TSM2 | biggest issue with wireless stuff outthere is that its so vaired, some works so slow, i moved to a top model netgear router and sent it back within 1 day because it was so so so bad, while the dlink stuff is fast |
| 19:52.22 | bmoraca | particularly with the IADs. you can almost buy 3 TA908s for what it costs for a single Cisco IAD |
| 19:52.59 | wtsexton | yea try and buy a cisco layer 3 switch with poe for what a 1335 costs lol |
| 19:53.15 | TSM2 | everyone has their own experiences that makes them like certian brands, personaly i like sonicwall stuff, no major issues and priced right, but one of the outside support techs we have in the US just hate them and is nearly screaming to me down the phone |
| 19:53.23 | bmoraca | well, to be fair, the 1335 isn't a fully multilayer switch |
| 19:53.47 | *** join/#asterisk brezular (n=brezular@adsl-dyn142.78-99-64.t-com.sk) |
| 19:53.52 | wtsexton | true |
| 19:54.32 | bmoraca | and Cisco 3550 PoEs are $255 on ebay right now :P |
| 19:54.36 | wtsexton | have you used the 1544P yet? |
| 19:55.01 | bmoraca | i haven't, no |
| 19:55.02 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:55.04 | wtsexton | yes but 3550 doesn't have a wan slot |
| 19:55.09 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/Staff/Sargun) |
| 19:55.24 | bmoraca | wow, that fucker's expensive |
| 19:56.07 | JerJer | anyone know if i can get the final CDR information via FastAGI ? |
| 19:56.40 | JerJer | guess i gotta test like ForkCDR (if that is still around) |
| 19:56.47 | wtsexton | I'd just like a gigabit version of the 1335 |
| 19:56.47 | bmoraca | i haven't been able to verify whether or not Adtran's swiches support port aggregation (Cisco's EtherFast)...i've received mixed information on that...the Adtran training stuff says yes, while several people I've spoken to say no... |
| 19:57.16 | wtsexton | never done it, however its something I could bench test when I have free time |
| 19:58.05 | JerJer | exten => h gets fired before the CDR is completed |
| 19:58.28 | wtsexton | one thing I noticed is the adtrans don't support per vlan spanning tree? |
| 19:59.13 | bmoraca | it doesn't look like they do |
| 19:59.45 | wtsexton | not really core or distribution switches anyways |
| 20:00.29 | bmoraca | no...they're pretty much all access layer switches. adtran doesn't really have any desire to get into network cores |
| 20:00.44 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:00.53 | *** join/#asterisk Micc (n=dotirc@174-21-26-36.tukw.qwest.net) |
| 20:01.21 | Micc | what is the cheapest card I can get with a timing source? |
| 20:01.22 | wtsexton | I found that out quickly when I had the priority set low on one and plugged it in |
| 20:01.23 | bmoraca | Cisco, Extreme, and Brocade pretty much have that eaten up |
| 20:02.15 | [TK]D-Fender | Micc: X100P |
| 20:03.00 | hardwire | wot? |
| 20:03.53 | Micc | TKD-Fender, I'm not seeing the X100P on digium's web site. |
| 20:04.01 | JerJer | heh |
| 20:04.04 | JerJer | you won't |
| 20:04.27 | [TK]D-Fender | Micc: Discontinued crap @ 15$ on ebay |
| 20:04.37 | JerJer | i'll sell ya one for $75 :) |
| 20:05.41 | JerJer | Micc: but why you think you need a timing source? |
| 20:06.13 | TSM2 | Sangoma UT50 VoiceTime USB Synch |
| 20:06.14 | Micc | JerJer, I have problems with using a local channel making calls from different types of phones. |
| 20:06.50 | Micc | from an aastra 6731i using g722 to an aastra 480i using ulaw, it doesn't sound right. all chopped up. |
| 20:06.50 | [TK]D-Fender | TSM2: Odds are no better than DAHDI_DUMMY.... |
| 20:06.59 | JerJer | unless something major has changed chan_local doesn't use any timing |
| 20:07.10 | JerJer | i haven't used any timing sources in years |
| 20:07.12 | [TK]D-Fender | indeed |
| 20:07.13 | TSM2 | [TK]D-Fender: that i dont know |
| 20:07.22 | JerJer | pretty much since app_conference came 'of age' |
| 20:07.26 | Micc | Maybe I have another problem then. |
| 20:07.51 | Micc | I'm not sure how I would even go about tracking it down. |
| 20:08.02 | JerJer | [TK]D-Fender: do you know the solution to my cdr / fastagi issue? hope you are not trying to be coy |
| 20:08.31 | [TK]D-Fender | JerJer: No idea... told you before I don't work with CDR processing. |
| 20:08.52 | JerJer | well you kept telling me to get more creative |
| 20:08.55 | *** join/#asterisk af_ (n=getsmart@88-149-230-54.dynamic.ngi.it) |
| 20:09.02 | JerJer | and i very much have |
| 20:09.09 | JerJer | leme bug those in -dev :) |
| 20:09.26 | tzafrir_laptop | JerJer, there's no also app_confbridge |
| 20:09.54 | tzafrir_laptop | also: meetme needs *mixing* from zaptel/dahdi . Not timing |
| 20:10.15 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 20:10.22 | JerJer | well yeah - this is irc not #nasa |
| 20:10.44 | JerJer | its still an interrupt that is needed |
| 20:10.57 | Micc | is this a good card http://www.x100p.com/products/FXO.php? |
| 20:11.07 | JerJer | i wouldn't buy it |
| 20:11.26 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:11.37 | JerJer | then again I have 20-30 X100Ps in storage here still :D |
| 20:11.58 | Micc | JerJer, how much you want for one? |
| 20:12.14 | Micc | this one doesn't look that bad. |
| 20:12.21 | Micc | and its only 30$ |
| 20:12.36 | Micc | are there dahdi drivers for the x100p? |
| 20:13.06 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 20:13.10 | jaytee | gets out his soldering gun and starts connecting the 48 Connexant modems for his channel bank. |
| 20:13.36 | JerJer | i'd want more than 30 bucks |
| 20:13.45 | jaytee | piker! |
| 20:13.52 | jaytee | highwayman! |
| 20:16.03 | JerJer | oddly enough I manage quite a few systems that still run app_meetme and need X100Ps, so i may keep these around for spares |
| 20:16.44 | jaytee | chinese clone knockoffs of the original X100P no doubt |
| 20:17.07 | JerJer | x100p.com - totally |
| 20:17.30 | jaytee | "Is that a real poncho or a Sears poncho?" "Look here, brother! Who you jiving with that Cosmic Debris?" |
| 20:17.36 | JerJer | mine - I acquired mine directly from Mark's source back even before digium existed :) |
| 20:18.47 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 20:20.18 | [TK]D-Fender | ~savemoney |
| 20:20.19 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 20:22.53 | jaytee | infamy, it'll get ya evertime :-) |
| 20:24.06 | drclue | If your wanting to save money , you could always try replacing your ISP with RFC1149 communications , it costs hardly chicken feed |
| 20:25.11 | jaytee | http://www.faqs.org/rfcs/rfc1149.html |
| 20:26.17 | *** join/#asterisk Guedes (n=guedes@unaffiliated/guedes) |
| 20:26.49 | drclue | :) |
| 20:27.02 | [TK]D-Fender | Checkout time, later all |
| 20:27.17 | Faiz | Question regarding inbound callers to making outbound calls: |
| 20:27.38 | Faiz | I'm using DISA to allow for a dialtone when a user calls my PBX |
| 20:27.51 | Faiz | when the user presses that extension, the dialtone is sound |
| 20:27.56 | jaytee | is this to reduce latency? :-) |
| 20:28.03 | Faiz | no no |
| 20:28.10 | jaytee | sorry, just kidding |
| 20:28.26 | Faiz | i'm wondering why I get a busy tone when I call my PBX and ask it to make an outbound call |
| 20:30.01 | jaytee | Faiz, first dialtone is for passcode, if it matches then you get system dialtone to dial out |
| 20:30.24 | Faiz | yes, i specified no passcode |
| 20:30.33 | Faiz | so as soon as i press the extension to initiate DISA, i get the dialtone |
| 20:30.45 | Faiz | but right after when i dial the 10 digit # to dialout to, i hear a busy tone |
| 20:30.48 | jaytee | no-passcode? or no-password? |
| 20:30.55 | Faiz | no-password |
| 20:31.00 | jaytee | good |
| 20:31.20 | *** join/#asterisk Defraz (n=T0tal@c72co-edge-router.fuzecore.com) |
| 20:31.43 | jaytee | Faiz, are you in the US? |
| 20:31.46 | Faiz | yes |
| 20:32.33 | jaytee | in the context you have the DISA app do you have a 10 digit pattern match for dialing your outbound trunk? |
| 20:32.45 | Faiz | yes, the extension looks as such: |
| 20:32.59 | jaytee | or in the context the DISA app points to? |
| 20:33.06 | Faiz | exten => 6,1,DISA(no-password,outbound,"OBC") |
| 20:33.15 | Faiz | where the outbound context satisfies all outbound calls with: |
| 20:33.28 | Faiz | _XXXXXXXXXX,1,Dial(trunk/exten) |
| 20:33.50 | JerJer | YAAY for russellb! endbeforehexten=yes |
| 20:34.00 | russellb | wins |
| 20:34.35 | jaytee | Faiz, what if you dial the same 10 digit number from an internal phone? |
| 20:34.48 | Faiz | it dials out perfectly fine |
| 20:34.48 | Faiz | heh |
| 20:34.56 | jaytee | same outbound context? |
| 20:34.59 | Faiz | yep. |
| 20:35.15 | *** join/#asterisk errotan (n=errotan@5403E51A.catv.pool.telekom.hu) |
| 20:35.24 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:35.28 | jaytee | Faiz, in most case i have to pass 11 digits |
| 20:35.36 | jaytee | with the first digit being a one |
| 20:35.46 | JerJer | russellb: being a minimalist (and very old skewl) I don't even run a cdr.conf on any of my systems |
| 20:36.01 | Faiz | ok, so _1XXXXXXXXXX in my outbound context? |
| 20:36.08 | Faiz | to specify "long ditance" calling? |
| 20:36.24 | Faiz | "long distance* " |
| 20:36.28 | JerJer | Faiz: not to confuse you, but the more correct answer is _1NXXNXXXXXX |
| 20:36.32 | JerJer | but otherwise, yes |
| 20:36.32 | jaytee | Faiz, is the outbound context in the DISA app the same outbound context that internal extensions use? |
| 20:36.58 | Faiz | yes, i have removed confusing by naming the outbound context uniquely |
| 20:37.04 | Faiz | thank you JerJer :p |
| 20:38.08 | Faiz | i'm guessing the DISA function |
| 20:38.11 | Faiz | as soon as the dialtone rings |
| 20:38.18 | Faiz | and you dial the outbound # |
| 20:38.26 | Faiz | it transfers you to that line? |
| 20:38.33 | Faiz | and all communication with the PBX is cut? |
| 20:39.18 | jaytee | no, you'd have a two leg call, the call coming in bridged to the outgoing call |
| 20:39.32 | *** join/#asterisk many (n=many@dslb-188-098-007-152.pools.arcor-ip.net) |
| 20:40.48 | jaytee | Faiz, read page 390 of the appendix in this. |
| 20:40.50 | jaytee | ~book |
| 20:40.51 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:41.53 | wtsexton | I need to read it myself, only used it as a index so far |
| 20:42.43 | jaytee | quittin time, back in a bit |
| 20:42.46 | Faiz | yes i've been reading heh |
| 20:42.49 | Faiz | thank you! |
| 20:42.59 | jaytee | Faiz, how many trunks do you have? |
| 20:43.33 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:43.51 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.150.253) |
| 20:46.33 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.4 (2009/08/11), 1.6.0.13 (2009/08/11), 1.4.26.1 (2009/08/11), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 20:46.57 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.4 (2009/08/11), 1.6.0.12 (2009/08/11), 1.4.26.1 (2009/08/11), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 20:46.59 | Qwell | n'yet |
| 20:47.15 | Faiz | @jaytee: sorry for the delayed response |
| 20:47.21 | Faiz | darn.. |
| 20:47.40 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 20:48.07 | beek | Qwell: ... I was just about to say... |
| 20:48.35 | beek | considers his pavlovian response to upgrade announcements. |
| 20:49.55 | [TK]D-Fender | Qwell:what just changed? |
| 20:50.06 | Qwell | [TK]D-Fender: release date was wrong on one |
| 20:50.48 | [TK]D-Fender | Qwell: Yeah, I commented on that as soon as Corydon76-dig changed it this morning |
| 20:55.53 | [TK]D-Fender | gives beek anotheer shock anyway... |
| 20:56.43 | beek | jumps |
| 20:57.01 | beek | then curses the bastard with the switch. |
| 20:57.19 | Katty | [TK]D-Fender: upnp |
| 20:57.23 | [TK]D-Fender | smirks |
| 20:57.31 | Faiz | I still can't seem to make outbound calls using DISA, is it related to the Trunk by any chance? |
| 20:57.34 | Katty | [TK]D-Fender: is what i decided on |
| 20:57.40 | Katty | [TK]D-Fender: and i'm changing the port number monthly |
| 20:57.46 | Katty | [TK]D-Fender: using a random number generator |
| 20:58.01 | Katty | [TK]D-Fender: with explicit denies on 5060, with the exception of bandwidth's two public, static, ip addresses. |
| 20:59.20 | [TK]D-Fender | Faiz: no |
| 20:59.47 | Faiz | i hear the dialtone (as i'm not specifyign a passcode or password) as soon as I press the extension, |
| 20:59.54 | [TK]D-Fender | Katty: I recoomen fail2ban <- |
| 20:59.56 | Faiz | and i reference my outbound context |
| 21:00.03 | Faiz | but when I dial the number, i get a busy signal |
| 21:00.40 | *** join/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 21:00.49 | *** part/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 21:00.56 | dustybin | is VOIP data encryted when it passes through networks? |
| 21:03.18 | _ShrikE | dustybin: It depends on what you mean by "VOIP data"... But usually no. |
| 21:03.50 | dustybin | _ShrikE: if my work uses VOIP, could i sniff everyones converstaions on the local network? |
| 21:04.38 | TSM2 | dustybin: yep its possable, dont ask me how though |
| 21:04.40 | _ShrikE | dustybin: If you can put yourself in a position to sniff the packets then yes... you can. |
| 21:04.47 | dustybin | ace :D |
| 21:05.00 | dustybin | looks forward to sniffing his boss tomorrow |
| 21:05.09 | TSM2 | dustybin: smells |
| 21:05.12 | _ShrikE | dustybin: Wireshark works nicely. Will even export the audio. |
| 21:05.20 | dustybin | double ace |
| 21:05.29 | _ShrikE | for sip at least. |
| 21:05.31 | [TK]D-Fender | dustybin: PERV |
| 21:05.59 | TSM2 | dustybin: if you can set port mirror on the VOIP server port then you should get most of it |
| 21:06.31 | dustybin | WOW :D |
| 21:06.40 | TSM2 | im not sure, if canreinvite is turned off, does all comms go though the server? or by default do phones communicate to eachother after the invite |
| 21:07.33 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.4 (2009/08/11), 1.6.0.13 (2009/08/11), 1.4.26.1 (2009/08/11), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 21:07.36 | Qwell | beek: psst |
| 21:08.21 | beek | smiles |
| 21:08.31 | dustybin | http://www.linuxjournal.com/article/9398 |
| 21:08.40 | beek | then frowns that the download failed. |
| 21:08.42 | Qwell | mirrors should be synced in like...I dunno. a minute and a half? |
| 21:09.04 | beek | drums fingers. |
| 21:09.09 | Qwell | :D |
| 21:09.20 | beek | starts drooling again (damned pavlovian conditioning) |
| 21:10.48 | beek | smiles again as the updated tarball starts coming down. Thanks Qwell! |
| 21:11.02 | Qwell | beek: thank Corydon76-dig |
| 21:11.08 | Qwell | he did the release |
| 21:11.10 | beek | Thanks Corydon76-dig |
| 21:12.06 | Corydon76-dig | Don't thank me; I'm the one who screwed it up in the first place |
| 21:13.39 | dustybin | looks forward to injecting audio into bosses phone calls |
| 21:13.43 | TSM2 | where can i find changelog for 1.4.26.1 |
| 21:13.51 | Qwell | TSM2: downloads.asterisk.org |
| 21:16.28 | dustybin | interesting... |
| 21:16.29 | dustybin | http://www.asteriskvoipnews.com/voip_security/interview_with_encryption_advocate_phil_zimmermann_regarding_voip.html |
| 21:16.56 | TSM2 | is there a release numbering scheme, im guessing the last bit .1 etc.. would be security stuff |
| 21:18.11 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 21:19.18 | [TK]D-Fender | bbrb |
| 21:19.54 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:20.09 | Katty | anyone know what bandwidth rtp range is off the top of their heads |
| 21:20.21 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 21:20.28 | TSM2 | wot |
| 21:20.43 | Katty | i see. |
| 21:20.47 | Katty | calls |
| 21:21.11 | rossand | dumb question - what's the proper way to add the dahdi and dahdi_dummy modules to load on boot? |
| 21:22.34 | justsomedood | there's usually a file in /etc that says which modules for the kernel to load |
| 21:23.26 | Katty | grooves to bandwidth's on hold musics. |
| 21:23.29 | justsomedood | <PROTECTED> |
| 21:24.08 | dustybin | zfone ftw |
| 21:24.17 | *** join/#asterisk cusco_ (n=tralala@2001:0:53aa:64c:cda:343e:2ac0:762d) |
| 21:24.25 | cusco_ | sorry but... how do I start dahdi ? |
| 21:24.25 | wtsexton | http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation#DAHDI |
| 21:24.35 | *** join/#asterisk engrxyz (n=sfgsfgs@92-237-248-183.cable.ubr07.basl.blueyonder.co.uk) |
| 21:24.36 | cusco_ | that |
| 21:24.37 | rossand | justsomedood: That's what I was used to as well. 2.6.x kernels use a new technique. |
| 21:24.38 | cusco_ | let me read |
| 21:24.49 | dustybin | anybody use this: |
| 21:24.49 | dustybin | http://zfoneproject.com/ |
| 21:26.16 | rossand | wtsexton: Thanks for the link. In case it was for my benefit - I've got all that. I need to have the dahdi module load on boot properly. |
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| 21:26.48 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:27.07 | wtsexton | which distro are you using? |
| 21:27.39 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 21:28.01 | rossand | wtsexton: FC 10 |
| 21:28.55 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 21:29.07 | wtsexton | chkconfig should enable it for load up on boot |
| 21:30.08 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 21:31.53 | *** join/#asterisk cusco_ (n=tralala@2001:0:53aa:64c:3891:f45f:2ac0:762d) |
| 21:32.01 | cusco_ | I don't understand, why asterisk cli does not respont to "pri show" |
| 21:32.30 | *** join/#asterisk lizone (n=zenst@user-0ccejib.cable.mindspring.com) |
| 21:32.37 | wtsexton | install libpri? |
| 21:32.44 | cusco_ | I think it is installed |
| 21:32.46 | cusco_ | hold |
| 21:32.53 | *** join/#asterisk RoPBX (n=nickserv@200.93.61.20) |
| 21:33.01 | RoPBX | Hi all |
| 21:33.31 | cusco_ | done |
| 21:33.59 | cusco_ | make install |
| 21:34.03 | cusco_ | what else? |
| 21:34.04 | RoPBX | please, some help... i'm having problems with dtmf tones rfc2833 using g729 when the call is been recorded. |
| 21:34.30 | wtsexton | not sure if you have to recompile asterisk after, you may want to do that |
| 21:34.33 | cusco_ | ah ok |
| 21:34.44 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 21:34.45 | wtsexton | rfc2833 is out of band |
| 21:34.45 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 21:34.54 | wtsexton | you won't hear the dtmf tones in the audio stream? |
| 21:35.00 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 21:35.53 | wtsexton | you'll see them in a packet sniff |
| 21:36.17 | RoPBX | ok, i don't want to hear the tones, but when i call to any place with an IVR that ask me to press 1 or 2 or any other key, the tones don't work |
| 21:36.39 | justsomedood | rossand: Is anything in /etc/rc.d/rc.modules ? |
| 21:36.50 | cusco_ | well I guess what he meant is that you should see if you hear the tones |
| 21:36.52 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 21:37.15 | justsomedood | never used FC 10, so i'm not sure |
| 21:37.59 | wtsexton | RoPBX, peering with sip or some other means? if sip make sure the provider has rfc2833 enabled for you |
| 21:38.54 | RoPBX | yes, provider is ok, if I make a direct call the tones are OK, but i'm using a PBX to make calls, and then the tones fail |
| 21:39.16 | RoPBX | if I insist and press like 10 times the key then it works, but that is not the idea |
| 21:39.24 | manxpower | RoPBX: Call someone that is using a NON-CELL and NON-VOIP phone. Press buttons, find out if they hear the tones clearly or not. Frequently DTMF issues with Asterisk are one of three things 1) Asterisk 1.2 had some pretty "interesting" DTMF interop issues with other equipment 2) On PRI or ZAP lines the default toneduration= is too short or 3) You are using SIP, not alaw or ulaw codec and using inband DTMF. |
| 21:39.51 | wtsexton | ^-- what they said |
| 21:40.30 | *** join/#asterisk cusco__ (n=tralala@2001:0:53aa:64c:453:e251:2ac0:762d) |
| 21:40.31 | wtsexton | out |
| 21:40.33 | cusco__ | sorry |
| 21:40.36 | cusco__ | nope, pri still won't work |
| 21:41.05 | cusco__ | how can I check what can I be missing |
| 21:41.48 | cusco__ | ok it works now |
| 21:41.50 | cusco__ | sorry |
| 21:41.55 | cusco__ | I needed to make samples |
| 21:41.56 | cusco__ | lol |
| 21:42.20 | cusco__ | I didn't think I needed conf files for it, we will use our previous confs... ok thanks |
| 21:43.50 | RoPBX | thanks manxpower. I making test via SIP with g729 and dtmf via rfc2833. If I use inband tones it works randomly |
| 21:44.04 | RoPBX | and Asterisk 1.4.21 |
| 21:44.34 | RoPBX | is there a way to change toneduration on SIP trunk? |
| 21:48.50 | bmoraca | hold the button down longer? |
| 21:50.17 | rossand | justsomedood: rc.modules does not exist on FC10 |
| 21:50.38 | Qwell | rossand: use the provided init script... |
| 21:51.03 | RoPBX | I can't tell users to hold the button longer.... |
| 21:51.17 | justsomedood | rossand: does the /etc/modprobe.d folder exist? |
| 21:51.30 | RoPBX | in fact, if i press the key 10 times it works |
| 21:51.50 | RoPBX | but its not a good solution for the problem |
| 21:52.58 | RoPBX | the problem is only if I'm recording the call, if I turn off recording the tones works fine |
| 21:54.31 | bmoraca | RoPBX: what if you revert back to alaw/ulaw instead of g729? |
| 21:55.16 | RoPBX | yes, but its better with g729 for quality |
| 21:55.29 | bmoraca | that's not what i asked |
| 21:55.36 | bmoraca | does it work when you use alaw/ulaw? |
| 21:55.58 | RoPBX | yes, i think so...let me try |
| 21:56.06 | drclue | ulaw sounds fine to me and it seems to cure a lot of asterisk hic-ups, at least in my limited experience |
| 21:58.02 | *** join/#asterisk davidandgoliath (n=David@S0106001d60d4e488.vn.shawcable.net) |
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| 22:00.30 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 22:04.21 | rossand | Qwell: yes. I tried adding "install dahdi /sbin/modprobe dahdi" to modprobe.conf.dist (with thoughts of making a new file after). That did not work unfortunately. |
| 22:04.21 | rossand | oops, I mean justsomedood |
| 22:04.30 | rossand | Qwell: as far as I can see, when compiling from source there is no init script. |
| 22:07.10 | *** join/#asterisk davidandgoliath (n=David@S0106001d60d4e488.vn.shawcable.net) |
| 22:10.23 | Qwell | rossand: it tells you how when you install DAHDI... make config |
| 22:12.08 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 22:13.03 | Kobaz | hmm |
| 22:13.08 | manxpower | RoPBX: I would expect inband and rfc2833 to work randomly. i.e. not work. If you are using Asterisk 1.4 or later and your RFC out of band DTMF messages are being sent to the carrier, then there is NOTHING YOU CAN DO ABOUT IT, kick your carrier or get a new carrier. |
| 22:13.14 | Kobaz | is there a dialplan app/function to set a channel variable on another channel |
| 22:13.17 | Kobaz | i can't seem to find one |
| 22:13.35 | *** join/#asterisk [netman] (n=netman@65.Red-88-19-164.staticIP.rima-tde.net) |
| 22:14.13 | RoPBX | manxpower, but if I call directly it works, it doesn't work using asterisk PBX and the call being recorded |
| 22:14.21 | manxpower | rossand: "make config" |
| 22:14.32 | manxpower | RoPBX: define "call directly" |
| 22:15.18 | manxpower | Kobaz: there is none that I'm aware of. |
| 22:15.28 | Kobaz | mmm |
| 22:15.48 | manxpower | you can make variables be inherited by child channels |
| 22:15.51 | RoPBX | i have a GS2020 phone, i put Voip username and password as a line, and I call using my Voip provider, and tones work fine. If i take that same username and password and use it as a SIP trunk in my asterisk PBX then the tones fail |
| 22:16.12 | Kobaz | manxpower: yeah, i need direct write access to another channel's variables |
| 22:16.14 | manxpower | RoPBX: I suspect the GS phone defaults to ulaw and inband. |
| 22:16.28 | manxpower | Kobaz: not going to happen with the existing apps. |
| 22:16.54 | Kobaz | manxpower: ami has a chanvar set on arbitrary channels |
| 22:17.03 | RoPBX | no, I can configure it with g729 and rfc2833 |
| 22:17.05 | manxpower | Kobaz: but you wanted an APP, right? |
| 22:17.13 | Kobaz | Setvar |
| 22:17.14 | Kobaz | yeah |
| 22:17.17 | Kobaz | app would be much easier |
| 22:17.33 | manxpower | RoPBX: get a SIP debug of a failed call, put it on pastebin and hope someone wants to help with the issue. |
| 22:17.40 | Kobaz | otherwise i kinda have to rework this whole bit |
| 22:17.50 | RoPBX | ok |
| 22:18.19 | manxpower | I hate SIP debug so much I only do it when paid large sums of untraceable cash. |
| 22:18.38 | justsomedood | rossand: well I'm of no help, good luck though :D |
| 22:19.04 | Kobaz | manxpower: there's a really cool utility, i forgot the name, but it takes a tcpdump output, and makes a html page with the sip progression |
| 22:20.04 | manxpower | RoPBX DTMF issues are some of the hardest ones to solve. |
| 22:20.33 | RoPBX | ouch! |
| 22:20.55 | manxpower | Qwell: does SIP debug still show RFC2833 DTMF, even though it's technically RTP and not SIP? |
| 22:21.07 | Kobaz | make sure local dtmf works.. (ie: call voicemail and log in)... |
| 22:21.28 | Qwell | manxpower: no, but a dtmf debug would (or rtp debug) |
| 22:21.43 | manxpower | RoPBX: see qwell's comment |
| 22:21.55 | Kobaz | and then call outside, and make sure your dtmf's are still going through... ie: call yourself and listen for some dtmfs |
| 22:22.01 | RoPBX | ok |
| 22:22.10 | manxpower | just don't call yourself on a cellphone |
| 22:22.42 | RoPBX | how can i get a dtmf debug??? |
| 22:22.48 | Kobaz | depending on the provider, sometimes rfc2833 doesn't work, and you need to use sip info. or in audio |
| 22:23.06 | manxpower | RoPBX: I imagine something along the line of "dtmf set debug on" |
| 22:23.15 | Kobaz | manxpower: there is no module called dtmf |
| 22:23.31 | Kobaz | the only dtmf debug is either sip debug, or listening to the call (ala chanspy or something similar) |
| 22:23.43 | RoPBX | the thing is that it only fails when the call is been recorded |
| 22:23.43 | manxpower | Kobaz: I'd have to dig up a 1.4.x server to find the exact output. rtp debug should do it as well |
| 22:23.57 | RoPBX | if I turn recording off the tones work fine |
| 22:24.11 | manxpower | RoPBX: so your comparison with the GS phone is not really valid. |
| 22:24.27 | Kobaz | rtp debug just dumps the packet source/dst's |
| 22:24.29 | rossand | justsomedood: thank you kindly just the same. I'll head over to #fedora to see if anyone there can advise. |
| 22:25.44 | manxpower | Kobaz: it should also dump DTMF as rfc2833 DTMF *IS* RTP. |
| 22:26.32 | Kobaz | if it had more output than source and dest ips, then yeah. it would probably output it |
| 22:26.36 | Kobaz | Sent RTP packet to 192.168.35.104:5004 (type 00, seq 012873, ts 17899024, len 000160) |
| 22:26.38 | Kobaz | Got RTP packet from 192.168.35.103:5006 (type 00, seq 006016, ts 1326556867, len 000160) |
| 22:26.42 | Kobaz | Got RTP packet from 192.168.35.104:5004 (type 00, seq 047119, ts 3221688273, len 000160) |
| 22:26.45 | Kobaz | you'll see stuff like that |
| 22:26.52 | Kobaz | not very useful for debugging dtmf, unless there's some option i don't see |
| 22:27.23 | manxpower | I'd have to look it up, but I suspect type would differ for DTMF. |
| 22:27.45 | manxpower | also the LEN would be much shorter |
| 22:27.50 | Kobaz | yeah true |
| 22:29.03 | Kobaz | Got RTP RFC2833 from 192.168.35.103:5004 (type 101, seq 057485, ts 3759483202, len 000004, mark 1, event 00000004, end 0, duration 00000) |
| 22:29.06 | Kobaz | okay |
| 22:29.06 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:29.08 | Kobaz | it does debug dtmf |
| 22:29.11 | Kobaz | heh, never noticed that |
| 22:29.27 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:29.43 | RoPBX | yes, manxpower, my test with GS2020 is not valid, but i did it to check the Voip provider |
| 22:30.21 | manxpower | RoPBX: *nod* A good thing to test. What version of Asterisk are you using? |
| 22:30.31 | RoPBX | 1.4.21 |
| 22:31.09 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:32.34 | manxpower | Try grabbing the latest. You'll need that if you end up filing a bug anyway. Just reinstall the new version over the old version just do NOT run "make samples", it will overwrite your existing configs. |
| 22:34.03 | exsync | did commands from 1.4 to 1.6 change a lot? like show queues, etc |
| 22:34.16 | exsync | or do i not have show queues because i'm using trixbox |
| 22:35.09 | manxpower | exsync: yes. see UPGRADE*.txt in the Asterisk source code. |
| 22:35.19 | exsync | ty |
| 22:35.42 | rossand | following up my question earlier about kernel modules on fedora. #fedora advised me - make a script that loads it under /etc/sysconfig/modules/asterisk.modules. In it, put the /sbin/modprobe commands |
| 22:35.49 | RoPBX | ok i'll try the latest |
| 22:36.06 | Qwell | rossand: ...install the init script |
| 22:36.06 | rossand | e.g /sbin/modprobe dahdi_dummy |
| 22:36.52 | rossand | Qwell: which would necessitate installing the package, which means voicemail won't work... I'll stick with the source instead. Thanks though. |
| 22:37.03 | Qwell | scroll up. read what I said. |
| 22:37.17 | exsync | anyone know of a 1.6 cheatsheet (commands) |
| 22:37.22 | exsync | ive only found 1.4 |
| 22:37.35 | Qwell | exsync: core show applications |
| 22:38.16 | *** join/#asterisk nixer (n=nixer@78.154.201.209) |
| 22:38.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:38.41 | manxpower | rossand: Dude! "make config" installs the init scripts. |
| 22:38.54 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-67-17.kgldga.dsl.dynamic.tds.net) |
| 22:39.06 | Qwell | manxpower: which I said about 40 minutes ago |
| 22:39.10 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 22:39.18 | manxpower | Qwell: I said it about 15 mins ago. |
| 22:41.04 | nixer | Hello. I have an Asterisk PBX setup at home and I was reading on SER. I still don't quite understand the difference between SER & *; is it that * can bridge between SIP & PSTN lines while SER is pure VoIP? |
| 22:41.17 | Qwell | nixer: SER is a proxy. That's it. |
| 22:41.43 | nixer | Qwell: I read on websites that it can act as a router, proxy & a SIP server. |
| 22:41.57 | justsomedood | exsync: commands for the asterisk cli? |
| 22:42.01 | nixer | "SIP Express Router (SER) is a high-performance, configurable, free SIP server licensed under the open-source GNU license ." |
| 22:42.24 | exsync | justsomedood yes.. like http://myspacefears.com/trixbox/ |
| 22:42.27 | exsync | but not 1.4 |
| 22:42.44 | Qwell | exsync: type 'help' |
| 22:42.52 | exsync | i'm aware of help |
| 22:42.58 | justsomedood | are you using tribox? |
| 22:43.09 | exsync | i prefer cheat sheets.. http://packetlife.net/cheatsheets/ ;) |
| 22:43.17 | exsync | they definitely help with legibility |
| 22:43.27 | manxpower | I prefer the built in cheatsheets |
| 22:43.32 | exsync | justsomedood i have various * systems, thats just what you find when you google "asterisk cheat sheet" |
| 22:43.51 | exsync | manxpower if asterisk commands were more descending, like IOS, i wouldn't mind |
| 22:43.58 | Qwell | exsync: none of the other stuff is relevant or changed |
| 22:44.08 | exsync | k |
| 22:44.10 | justsomedood | well, if you want the CLI commands 'help' will be up-to-date. The phone shortcuts on the tribox one arent' setup in the default asterisk config |
| 22:44.16 | exsync | i'd only tried a few and none worked |
| 22:44.24 | exsync | gotcha |
| 22:44.24 | manxpower | nixer: ALL SER does is route sip packets. It does not do conversion, voicemail, menus, sound, etc. |
| 22:44.48 | justsomedood | speaking of CLI commands, is anybody on SVN trunk? |
| 22:44.52 | nixer | "It can act as SIP (RFC 3261) registrar, proxy or redirect server. " -- http://www.iptel.org/ser |
| 22:44.54 | justsomedood | 'reload' doesn't work on it |
| 22:45.17 | nixer | manxpower: It does have a voicemail module. |
| 22:45.42 | manxpower | nixer: news to me. |
| 22:46.05 | manxpower | since SIP is not audio (audio is RTP), it must be more than just a SIP router then. |
| 22:46.09 | RoPBX | nixer read this http://www.voip-info.org/wiki/view/Asterisk+at+large |
| 22:47.05 | exsync | i decided to migrate a 500 agent 1000 extension phone system to clustered servers, what a rough task |
| 22:48.35 | *** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk) |
| 22:50.48 | exsync | for me at least |
| 22:51.12 | bmoraca | wow, that cheatsheet site is pretty slick |
| 22:51.32 | exsync | yea this guy stretch on freenode makes them |
| 22:51.40 | exsync | i love the bgp/eigrp sheets, but im a net admin |
| 22:51.47 | exsync | and the wireshark filter |
| 22:52.37 | *** part/#asterisk oftis (n=nicok@dslb-094-217-067-045.pools.arcor-ip.net) |
| 22:52.59 | bmoraca | ok, i take it back...most of them are pretty pointless...the BGP one was intersting though |
| 23:08.15 | *** part/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 23:08.49 | nixer | If I want to build a cluster, does it make sense (or does it hurt) to install SER on the same machine as asterisk (and make a cluster of that)? |
| 23:10.17 | bmoraca | use load balanced load balancers to load balance your asterisk servers |
| 23:10.32 | nixer | heh |
| 23:10.58 | rene- | bmoraca at what amount of calls per asterisk-box would you begin using load balancers and ser in front of asterisk |
| 23:11.11 | bmoraca | i have no idea |
| 23:11.37 | nixer | Each phone call uses about 33MHz, so it depends on your machines. |
| 23:11.45 | nixer | A dual core 3GHz gives you a total of 6GHz. |
| 23:12.04 | bmoraca | although some load balancers claim to be layer 7 aware...such as the Barracuda ones...i'd imagine that F5 load balancers would as well... |
| 23:12.41 | bmoraca | nixer: that's a little vague...what encryption and what kind of processors? a Core-based Xeon is much different than a netburst Xeon...33mhz does not always equal 33mhz |
| 23:12.47 | dupe | yeah but you lose some "speed" between the interconnects etc. 2x3 isnt "quite" as fast as 1x6 |
| 23:12.56 | dupe | but no 6ghz exists so its a moot point i suppose :P |
| 23:13.05 | nixer | 33MHz using ulaw with no encryption. |
| 23:13.18 | dupe | its not linear |
| 23:13.42 | bmoraca | i didn't mean to say encryption...i meant to say encoding...but the idea is the same |
| 23:13.43 | nixer | dupe: 6GHz does exist, if you overclock an AMD opteron using liquid nitrogen, like they did live in CES a year back I think. |
| 23:14.08 | nixer | transcoding consumes more than encryption. |
| 23:14.18 | nixer | Again, it's obvious that there's no transcoding. |
| 23:14.31 | nixer | I was just giving a basic number to calculate upon. |
| 23:14.38 | dupe | sure, but in that case nixer, your dual would be faster. theres more to a computer than just mhz speed |
| 23:15.05 | dupe | 6ghz of old tech vs 6ghz of new technology, new technology would win |
| 23:15.20 | nixer | I know. You're taking this way too far... :/ |
| 23:15.27 | dupe | but if you have some spare liquid nitro laying around i'll try it myself :P |
| 23:15.31 | bmoraca | my feeling is that if the load balancer appliance is perfectly layer 7 aware, it'd be easier to implement redundancy using that vs. trying to cluster asterisk servers. |
| 23:16.09 | LiNeTuX | froze a hard drive once with liquid nitrogen |
| 23:16.24 | bmoraca | did you shatter it with a hammer? |
| 23:16.29 | nixer | LiNeTuX: Why would you do that? |
| 23:16.33 | LiNeTuX | i wanted to, but it didn't go like in the movies |
| 23:16.47 | LiNeTuX | nixer: what else do you do with a busted hdd? keep it on the shelf? |
| 23:17.02 | LiNeTuX | freezing/destroying things is SOOOOO much more fun |
| 23:17.09 | nixer | LiNeTuX: Paperweight. |
| 23:17.18 | nixer | I have a few around :p |
| 23:17.20 | LiNeTuX | nixer: i don't have enough paper for my hdd's ;) |
| 23:17.42 | nixer | Some are used as foot rests or to raise my feet. |
| 23:17.42 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 23:18.34 | LiNeTuX | if you have a lot of time on your hands, you can take the platters out and put 'em in a slingshot to see if they'll embed themselves in various objects. usually they just shatter,tho. |
| 23:18.36 | nixer | So I gather from your responses is that there's nothing wrong with slapping SER on top of Asterisk? Cool! |
| 23:20.35 | nixer | LiNeTuX: Platters are magnets so you can actually take a bunch and walk around the city and slap each one on an ATM machine :p |
| 23:22.25 | LiNeTuX | nixer: actually most of them are glass with a magnetic coating |
| 23:23.03 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@rrcs-67-79-18-58.sw.biz.rr.com) |
| 23:25.20 | RoPBX | hey manxpower, it seems to work with Asterisk 1.4.26! |
| 23:28.03 | *** join/#asterisk xpot-mobile (n=james@173-14-225-193-Utah.hfc.comcastbusiness.net) |
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| 23:32.01 | *** join/#asterisk xpot-mobile (n=james@173-14-225-193-Utah.hfc.comcastbusiness.net) |
| 23:35.37 | *** join/#asterisk Micc (n=dotirc@174-21-26-36.tukw.qwest.net) |
| 23:43.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:53.21 | *** join/#asterisk awkfu (n=awkfu@66.162.90.57) |
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| 23:56.13 | *** join/#asterisk ingenius (n=alektro@host215.200-45-165.telecom.net.ar) |
| 23:56.37 | Tarantulafudge | so guys I'm getting some SIP Trunks and I just finished installing FreePBX, anyone got some advice for a newbie? |
| 23:58.33 | Tarantulafudge | it seems my future looks grim |
| 23:59.04 | Tarantulafudge | anyhow I'll be a regular of #asterisk as of now |
