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| 00:32.19 | cdahmedeh | hello, i would like to know if asterisk is for me, i'm looking for a simple server setup, basically run asterisk on it, and allow a few friends and family connect to it, and have voice and video conference |
| 00:32.43 | cdahmedeh | asterisk seems to be nice because i'd be able to control amount of traffic and bandwith usage |
| 00:45.40 | qxork | cdahmedeh: what type of linux / networking experience do you have? |
| 00:46.11 | cdahmedeh | i have 1.3 year linux experience, i can install arch linux and gentoo linux with ease |
| 00:46.30 | cdahmedeh | and i'm not scared of reading lots of documentation and learning |
| 00:47.00 | qxork | Asterisk for your points is great... you will run into NAT, and firewall issues generally with video, etc. |
| 00:47.22 | qxork | that being said, if you have decent experience and RTFM on SIP, RTP you will be fine |
| 00:47.34 | cdahmedeh | is it possible to have run on http mostly ? |
| 00:47.41 | qxork | the sip and rtp will be on any voip solution |
| 00:47.49 | qxork | voice and video will use udp |
| 00:48.03 | qxork | so would not be able to run on http |
| 00:48.09 | cdahmedeh | like i might in university wireless networks where i have access to http only |
| 00:48.24 | cdahmedeh | like port 80 basically |
| 00:48.26 | qxork | in that case, you'd probably want to to openvpn or something |
| 00:49.11 | WIMPy | I don't think you really want to send udp traffic over a tcp based vpn. |
| 00:49.37 | qxork | WIMPy... a few years ago, that was the case |
| 00:50.00 | cdahmedeh | like for example, how do skype and msn work on http ? is it some sort of gateway ? |
| 00:50.00 | qxork | today's systems are fine. I have many clients using openvpn without issue |
| 00:50.01 | WIMPy | He said wireless! |
| 00:51.15 | WIMPy | On a good link that may be fine, but I doubt you'd have much fun with it over wifi. |
| 00:51.50 | cdahmedeh | and asterisk works well with ekiga right ? |
| 00:51.52 | qxork | Yeah... on most wifi I don't think even straight would be anything to "call" home about |
| 00:51.55 | qxork | ;) |
| 00:52.22 | qxork | didn't see the wireless at first |
| 00:52.32 | WIMPy | Certainly not with tcp involved. |
| 00:52.46 | cdahmedeh | i think i'm going to setup a virtual machine, and setup a virtual network with limited connectivity, and see how it works |
| 00:52.55 | qxork | Even using low end codecs, I've never loved using wifi for voip unless totally inside the lan. |
| 00:53.14 | qxork | low end meaning low bandwidth |
| 00:53.29 | WIMPy | Well, if you can use normal udp, it works, but it's not optimal. |
| 00:55.24 | cdahmedeh | well i don't mind latency or anything like this |
| 00:55.30 | cdahmedeh | as long as i can control codec/bandwith |
| 00:55.34 | cdahmedeh | it's good |
| 00:56.27 | WIMPy | If you miss some packets on the wifi, you could easily end up with several seconds due to tcp. |
| 00:57.00 | cdahmedeh | that's quite interesting |
| 00:57.21 | cdahmedeh | i think this is going to deepen my knowledge about networking in general |
| 00:57.37 | cdahmedeh | like it feels very nice to setup my own linux server (haha laptop with busted screen) |
| 00:58.02 | cdahmedeh | ssh is very impressive stuff |
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| 01:06.35 | cdahmedeh | let's say i'm on dial-up somewhere, and i use really low quality codec, would it possible to get good audio ? |
| 01:07.13 | carrar | As good as quality as your codec and connectivity provide |
| 01:07.16 | WIMPy | If you define "low quality" as "good". |
| 01:07.37 | cdahmedeh | so it's all about the codec and bandwidth |
| 01:07.56 | WIMPy | And reliability. |
| 01:08.09 | carrar | and the benjamin |
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| 01:08.24 | antiwire | anyone use Zultys IP systems? |
| 01:08.27 | cdahmedeh | and of course, the phase of the moon, the position of the earth |
| 01:08.47 | carrar | Solar flares may effect performance also |
| 01:09.07 | Janos | hey there have a very simple conf here with automatic sip extension registration using regcontext=sip_autoreg and regexten=extnum, and then on the sip_autoreg context have this "exten => _XXXX,2,Dial(SIP/${EXTEN})" all that works great but ... |
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| 01:12.00 | Janos | i was wondering what would happen if my user name was different that my regexten, like for example "[janos] type=friend regexten=2000 etc..." if that was the case exten => _XXXX,2,Dial(SIP/${EXTEN})" would not work since 2000 would not really be a sip device and so i would not be able to place a call to it, if this is correct what would be the simplest way to overcome this ? |
| 01:12.46 | manxpower | Real PBXs seldom do something like ,Dial(SIP/${EXTEN}),Dial(SIP/${EXTEN}) |
| 01:13.40 | manxpower | I think you are confused. exten in this case would be whatever matched, if someone dialed 2000 then EXTEN would be 2000. What is the actual problem? |
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| 01:19.49 | Janos | yes that's why i wonder, on my current setup i create a sip device like this "[2000] type=friend regexten=2000 etc...", so this creates the SIP/2000 device via sip channel and also since i have regcontext=sip_autoreg adds and extension to this context like this "'2000' => 1. Noop(2000)" and then on the step 2 of this context i have this rule "'_XXXX' => 2. Dial(SIP/${EXTEN})", so far... |
| 01:19.51 | Janos | ...so good |
| 01:22.44 | Janos | but when i create my sip device like this "[janos] type=friend regexten=2000 etc..." i get this in dialplan show sip_autoreg, "'2000' => 1. Noop(janos)" and "'_XXXX' => 2. Dial(SIP/${EXTEN})" but in this case Dial(SIP/2000) which would be the expansion of that second rule would not work sin the real device name would be SIP/janos |
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| 01:25.44 | Janos | so what would be the way to go when we need to register the value of regexten as an extension that rings in SIP device with a different name than regexten ? |
| 01:26.34 | Janos | i know i can do it manually in extensions.conf but the idea would be to keep it dynamic |
| 01:28.22 | WIMPy | You could create a second entry in sip.conf, but I'm not so sure that's easier than a lin in extensions.conf. |
| 01:29.10 | WIMPy | Anyway. Bedtime. |
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| 01:32.58 | Janos | i can see asterisk does handle it neatly when it inserts it's own command "'2002' => 1. Noop(janos)" is there any way to tell asterisk to replace that Noop for Dial ? |
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| 03:08.46 | QbY | i've upgraded to 1.6 and believe we've found everything in the dialplan that needed changes; does anyone know of a tool/command that will parse the entire dialplan and find any issues |
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| 04:41.39 | VultureZ | Anyone ever seen where a provider requires a certian value to be sent over the PRI in order to utilize long distance? |
| 04:43.04 | TJNII | My 386 finally gave up the ghost. :( |
| 04:43.11 | florz | no, long distance usually should work without connection establishment |
| 04:44.07 | VultureZ | florz: thats what all my PRI installs have been in the past |
| 04:44.20 | VultureZ | and their second PRI works without a problem |
| 04:44.36 | VultureZ | tried 10 and 11 digits both turn up "LD Charges May Apply"... annoying |
| 04:44.53 | VultureZ | and I am passing valid CID so thats not the issue |
| 04:45.11 | VultureZ | is there any other values you can pass out over the D channel than CID? |
| 04:45.23 | florz | no |
| 04:45.36 | florz | with ISDN, you can not specify which destination to call |
| 04:46.24 | VultureZ | destination? |
| 04:46.44 | florz | the "other end", you know? |
| 04:47.20 | VultureZ | oh right, like the carrier |
| 04:47.57 | florz | .o() |
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| 04:57.35 | sawgood | ChannelZ: you around by chance? |
| 05:00.45 | sawgood | ExtenSpy(extension@context,options) |
| 05:01.08 | sawgood | If I wanted to put this in my * box ... what if I do not know the context (does it have to be used)? |
| 05:01.45 | sawgood | exten => 5302,1,ExtenSpy(302),w) |
| 05:01.48 | sawgood | would that statement work? |
| 05:02.13 | TJNII | I'm confident you need to know the context. |
| 05:02.27 | TJNII | Though I don't have experience with extenspy |
| 05:02.42 | sawgood | I had this working in the past, but I had to rebuild the box before I backed it up |
| 05:02.43 | TJNII | You should know the context of the extension you with to spy on. |
| 05:02.53 | TJNII | s/with/wish/ |
| 05:03.08 | sawgood | I remember when it did work ... all I had to do was enter the one line statement in one file |
| 05:03.46 | sawgood | any tips on how I can find tihs context information at? |
| 05:04.03 | sawgood | It must already exist, or I wouldn't have been able to make this work in the past |
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| 05:04.33 | TJNII | ...it should be the context you have the extension in within your dialplan. |
| 05:04.38 | sawgood | thank you |
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| 05:16.13 | ChannelZ | You can use ChanSpy to spy on channels in general |
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| 05:23.12 | sawgood | Hey ChannelZ: you helped me the other day with ExtenSpy |
| 05:23.17 | sawgood | do you remember by chance? |
| 05:23.36 | sawgood | I had to rebuild the box before backing up the config ... so I lost the syntax |
| 05:23.57 | sawgood | I think it was something like this ... |
| 05:23.57 | sawgood | exten => 5301,1,ExtenSpy(301@301,w) |
| 05:24.03 | ChannelZ | well the syntax pretty much is what it is.. |
| 05:24.14 | ChannelZ | It's not that, unless you have a context called "301" |
| 05:24.42 | sawgood | Well I have [301] with all the settings for extension 301 |
| 05:24.50 | sawgood | or is that something different? |
| 05:25.14 | ChannelZ | I'm assuming you mean in sip.conf |
| 05:25.24 | ChannelZ | In which case no that's just the name of the device. A device is NOT an extension |
| 05:25.29 | sawgood | yes [301] in sip.conf |
| 05:25.53 | sawgood | so, any tips on which file I should look for this context in? |
| 05:26.11 | ChannelZ | The context is in extensions.conf |
| 05:26.31 | sawgood | would it be the context for 301 only ... or for many extensions? |
| 05:26.36 | sawgood | I have the file open now |
| 05:27.47 | ChannelZ | It's for whatever extension you're spying on is. |
| 05:28.10 | ChannelZ | Chances are they're all in the same context in which case you can just leave it off and it will attempt to find the extension in the current context |
| 05:28.15 | sawgood | since this is a PIAF box, I think it would be in extensions_additional.conf |
| 05:28.29 | sawgood | I will try that |
| 05:29.31 | sawgood | exten => 5301,1,ExtenSpy(301,w) |
| 05:29.36 | sawgood | maybe like this? |
| 05:30.50 | sawgood | Did not work |
| 05:30.57 | sawgood | oh wait ... I need to do a reload |
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| 05:59.14 | sun28 | moin |
| 06:03.49 | ChannelZ | here comes the sun! |
| 06:04.12 | sun28 | sun28 please :b |
| 06:11.28 | sawgood | ChannelZ: any update for a tip to fix ExtenSpy? |
| 06:11.41 | sawgood | You helped me make it work under AN .. but I have an older PIAF box |
| 06:11.42 | ChannelZ | What's to fix? |
| 06:11.58 | sawgood | I lost the config you gave me because I had to reload the box |
| 06:12.04 | ChannelZ | ExtenSpy is kind of janky because AN EXTENSION ISN'T A DEVICE! |
| 06:12.15 | sawgood | It worked really well though last time |
| 06:12.15 | ChannelZ | ExtenSpy spies on an extension called |
| 06:12.48 | ChannelZ | If I pick up my phone and dial 3847, someone else has to ExtenSpy on 3847 |
| 06:13.19 | sawgood | ok |
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| 06:14.17 | sawgood | I have this: exten => 5302,1,ExtenSpy(302@snoop,w) |
| 06:14.24 | sawgood | I put this in extensions_custom.conf |
| 06:14.45 | sawgood | at the bottom of the file in a context I made called [snoop] |
| 06:14.49 | ChannelZ | RE: That will spy on someone who dialed 302 in the context 'snoop' |
| 06:15.08 | sawgood | last time, when we got this working the file I put the three line statements in was 100% empty |
| 06:15.24 | sawgood | Where should I put the syntax? |
| 06:16.51 | ChannelZ | First of all extensions_custom.conf is not a standard config file, it's something FreePBX creates and then includes from extensions.conf |
| 06:17.10 | ChannelZ | So since you've re-installed I have no idea what you have going |
| 06:18.10 | sawgood | Right ... before it was Asterisk 1.6.2.x on an AN1.5 build |
| 06:18.22 | sawgood | This is PIAF 1.4, with Asterisk 1.4.x |
| 06:18.47 | sawgood | Actually it was AN1.5 with Asterisk 1.6.0.26 |
| 06:19.25 | ChannelZ | Second, ExtenSpy does not probably do what you think it does, which is to spy on a DIALED extension |
| 06:19.34 | ChannelZ | What the heck is PIAF? |
| 06:20.02 | sawgood | PBX in a Flash version 1.4 |
| 06:20.20 | sawgood | I won't even try to geet it to load on this lab box ... |
| 06:20.34 | sawgood | I'll update the box to AN1.5 ... and put Asterisk 1.6.0.26 on it |
| 06:20.55 | sawgood | Then, I'll ask for your help again because this will be the same box you helped me with in the past |
| 06:22.02 | ChannelZ | As little as I know about FreePBX, I know even LESS about PBX In A Flash |
| 06:23.30 | ChannelZ | I'm really not sure why people want to put these GUIs and crap on top asterisk all the time |
| 06:24.38 | ChannelZ | apparently PIAF is using FreePBX again anyway |
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| 09:21.21 | carrar | moof |
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| 10:06.00 | samirddddddd | Hi |
| 10:06.58 | smrdoshi | Hello, is there any? I am facing an issue regarding DTMF, if someone can help me out |
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| 10:20.03 | tzafrir_laptop | smrdoshi, maybe if you'll be more specific |
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| 10:22.20 | smrdoshi | okay let me explain |
| 10:22.40 | smrdoshi | actually we have installed asterisk 1.4.18 |
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| 10:23.47 | smrdoshi | but in that when we make calls, some of the calls are having DTMF issue |
| 10:24.06 | smrdoshi | like some time asterisk is adding some extra digits |
| 10:24.15 | smrdoshi | and some time removing digits |
| 10:24.40 | smrdoshi | some time duplicating numbers |
| 10:24.42 | MAbbas | Hi, is there any known issue that asterisk AMI misseses some regirstered events? |
| 10:25.44 | WIMPy | MAbbas: I didn't miss any so far. That's all I can say. |
| 10:27.41 | smrdoshi | tzafrir_laptop: we are using get_data function of AGI script |
| 10:28.11 | tzafrir_laptop | smrdoshi, what type of channel is it? SIP? DAHDI? |
| 10:28.27 | smrdoshi | tzafrir_laptop:SIP |
| 10:28.50 | tzafrir_laptop | What dtmfmode do you use? |
| 10:28.58 | smrdoshi | tzafrir_laptop: auto |
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| 10:37.45 | MAbbas | <WIMPy>: Well on load asterisk starts missing events ... |
| 10:38.06 | MAbbas | <WIMPy>: BTW, on how much load you have on your system? |
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| 11:16.59 | WIMPy | MAbbas: Usually not much. |
| 11:17.11 | WIMPy | Hat kind of events? |
| 11:17.54 | WIMPy | s/Hat/What/ |
| 11:28.06 | MAbbas | <WIMPy> I am missing .. Link/Unlink events |
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| 12:54.33 | |amadeus| | good mornin evr1 |
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| 13:29.34 | Katty | morning |
| 13:29.42 | Katty | lots of squirrelies out today |
| 13:32.00 | carrar | Morning! |
| 13:32.20 | carrar | Are they in your kitchen? |
| 13:34.23 | |amadeus| | talking for nuts |
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| 14:04.41 | zomenox | hi, i have a dahdi question |
| 14:04.49 | zomenox | I have a tdm2400 |
| 14:04.57 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
| 14:05.04 | zomenox | I have it set up, i can call from the phones |
| 14:05.10 | zomenox | and calling to them works |
| 14:05.13 | zomenox | but they never ring |
| 14:05.22 | zomenox | the call connects when they are picked up |
| 14:05.30 | zomenox | just no ring |
| 14:05.43 | *** join/#asterisk fofware (~chatzilla@190.229.134.116) |
| 14:06.22 | zomenox | I used the r option in the dial command to try and force it, but to no avail |
| 14:08.10 | zomenox | the debug even says "DAHDI/21-1 is ringing" |
| 14:08.21 | zomenox | which would seem to be a bold face lie |
| 14:08.24 | carrar | You plug the power cable into the board? |
| 14:08.56 | zomenox | the old style drive power cable yes |
| 14:10.30 | zomenox | would that be the symptom of that cable not providing power? |
| 14:10.37 | carrar | yes |
| 14:10.55 | zomenox | perhaps i could power that machine down, and make sure that power cable is providing power |
| 14:11.03 | carrar | perhaps |
| 14:11.11 | zomenox | is off to steal a volt meter |
| 14:12.26 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 14:12.52 | pabelanger | Anybody using Eclipse for asterisk development? |
| 14:17.09 | tzafrir_laptop | zomenox, FXS (analog phone)? |
| 14:17.34 | zomenox | yes |
| 14:17.39 | tzafrir_laptop | are you sure you got the channel number right? (sanity question) |
| 14:17.49 | zomenox | and i got 12v out of the 12v and 5 out of the 5v |
| 14:18.13 | zomenox | and yes, because i can pick up the phone and the call completes |
| 14:18.24 | zomenox | i just dont get a ring to tell me i SHOULD pick up the phone |
| 14:19.47 | *** join/#asterisk momelod (~smelo@CPE00a065c98ce6-CM0012c91df0bc.cpe.net.cable.rogers.com) |
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| 14:19.51 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:20.01 | momelod | Greetings channel |
| 14:20.20 | momelod | is there a document someplace that lists all the voicemail feature codes? |
| 14:20.40 | momelod | feature list might be the wrong word, keypad shortcuts might be a better one |
| 14:20.51 | momelod | (prolly why my google searches arn't pulling up anything) |
| 14:21.24 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 14:21.37 | zomenox | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg09768.html |
| 14:22.05 | momelod | zomenox, TY |
| 14:23.54 | zomenox | just note that it is some draft |
| 14:24.03 | zomenox | not sure how accurate that is |
| 14:24.07 | zomenox | just a google result |
| 14:24.22 | zomenox | looks about correct though |
| 14:26.02 | *** join/#asterisk moy (~chatzilla@74.12.131.229) |
| 14:26.28 | zomenox | any other ideas on the no ring problem? |
| 14:28.14 | momelod | zomenox, run that by me again, i joined the channel after you posted your question |
| 14:28.39 | zomenox | Sure |
| 14:28.56 | zomenox | I have a tdm2400 |
| 14:29.21 | zomenox | I can send and recieved calls on the analog phones |
| 14:29.24 | *** part/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 14:29.30 | zomenox | But the do not ring |
| 14:30.06 | momelod | do u have the correct localization setup? ie, us for united states? |
| 14:30.55 | zomenox | <PROTECTED> |
| 14:30.59 | zomenox | chan_dahdi? |
| 14:31.09 | momelod | yeah |
| 14:31.12 | momelod | lemme check |
| 14:31.44 | momelod | /etc/dahdi/system.conf in my setup |
| 14:31.52 | zomenox | loadzone=us |
| 14:31.54 | momelod | loadzone=us |
| 14:32.08 | zomenox | defaultzone=us |
| 14:32.10 | momelod | what about your span line? |
| 14:32.19 | momelod | maybe u have to wrong signalling.. |
| 14:33.43 | zomenox | ar? |
| 14:34.20 | momelod | here is what i have: span=1,1,0,esf,b8zs |
| 14:34.24 | zomenox | fxsks right now |
| 14:34.26 | zomenox | oh |
| 14:34.28 | momelod | but that depends on your telco |
| 14:34.39 | momelod | call them up and ask them what should go there |
| 14:34.48 | zomenox | this is just a few phones on the card right now |
| 14:34.57 | zomenox | i haven't connected this box to anything else |
| 14:35.00 | momelod | i see |
| 14:35.02 | zomenox | starting small |
| 14:35.28 | momelod | hrmm.. not familiar w/ fxo myself |
| 14:35.34 | momelod | we use sip phones |
| 14:35.48 | zomenox | er |
| 14:35.56 | zomenox | fxo_ks, rather |
| 14:36.05 | zomenox | and thanks for the attempt anyway |
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| 14:38.50 | chazzam | wait, I think I may have missed something. Are you using a tdm2400 with a span line? |
| 14:39.39 | cdahmedeh | hello ! i am looking for the proper documentation to start a small video/voice over ip server, i don't need to use any phone lines, it's just for pc to pc. i have already setup and run asterisk on a debian server |
| 14:41.12 | zomenox | just a 4 port fxs |
| 14:41.22 | zomenox | er, to support fxo devices |
| 14:46.06 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.30.234) |
| 14:46.06 | TheDavidFactor | what's the process to upgrade DAHDI on a box? do I simply need to stop asterisk, install newer version of DAHDI, recompile asterisk, install asterisk and restart? |
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| 14:48.18 | chazzam | TheDavidFactor: well, if installing newer version of DAHDI in your statement includes stopping dahdi, then starting dahdi after running 'make install' and 'make config' then yes |
| 14:48.35 | chazzam | or if by restart you meant the whole system... |
| 14:48.37 | chazzam | doh |
| 14:49.39 | TheDavidFactor | yea I meant whole system, but I was wondering more about the asterisk process, do I need to recompile and reinstall asterisk too? |
| 14:50.14 | chazzam | I would consider it a good idea when upgrading dahdi |
| 14:50.31 | chazzam | it may not be strictly necessary in *every* case. but it is necessary in several cases |
| 14:50.47 | TheDavidFactor | do I need to rerun configure? |
| 14:51.25 | chazzam | I would say if you are going to both re-compiling, then it should be as close to new as possible, so 'make distclean' then do the whole shebang |
| 14:51.32 | chazzam | s/both/bother/ |
| 14:52.35 | TheDavidFactor | I'm working on a box I didn't install, is there anyway to tell what configure options (if any) were specified? |
| 14:53.21 | chazzam | I don't know actually =/ I've never had cause to ask that question before |
| 14:53.45 | TheDavidFactor | ok, well thanks for the help |
| 14:54.17 | chazzam | yup |
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| 15:08.14 | Katty | squirrely goes omnomnomnomnom |
| 15:08.59 | tzafrir_laptop | TheDavidFactor, basically: install a new version of dahdi, and then stop asterisk, restart dahdi , and start asterisk again |
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| 15:09.28 | tzafrir_laptop | (this is required because the asterisk process keeps the dahdi module busy) |
| 15:09.29 | Katty | Qwell: i joined Three Score |
| 15:09.40 | Katty | Qwell: supposedly a pretty big raiding guild |
| 15:10.59 | TheDavidFactor | tzafrir_laptop: so I shouldn't need to rebuild asterisk? I'm going from dahdi 2.2.0.2 to 2.2.1.1 |
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| 15:11.56 | tzafrir_laptop | TheDavidFactor, generally there should be no need. Sadly on the 2.1->2.2 change the interface of dahdi changed a little, requiring a rebuild |
| 15:12.14 | TheDavidFactor | ok, thx! |
| 15:14.15 | *** join/#asterisk jbeez (jbeez@pool-72-94-24-128.phlapa.fios.verizon.net) |
| 15:17.19 | TheDavidFactor | tzafrir_laptop: is there anyway from inside asterisk to confirm that I'm running the newer dahdi? |
| 15:18.05 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 15:18.27 | tzafrir_laptop | dahdi show version |
| 15:19.20 | TheDavidFactor | I don't have a dahdi show version, I'm running 1.4.24.1 |
| 15:19.31 | TheDavidFactor | *asterisk 1.4.24.1 |
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| 15:38.39 | jbeez | anyone in here use bigbluebutton? |
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| 15:44.46 | *** join/#asterisk KingDavidNYC (~Chris1232@rrcs-69-193-218-18.nyc.biz.rr.com) |
| 15:45.34 | KingDavidNYC | hello |
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| 15:46.35 | Katty | herroes |
| 15:46.41 | leifmadsen | yo |
| 15:46.48 | Katty | hugs leifmadsen |
| 15:46.49 | leifmadsen | zup? |
| 15:46.56 | Katty | nada, gonna go start spaghetti here in a min |
| 15:47.13 | leifmadsen | nice! I'm gonna warm up some lasagna |
| 15:47.20 | Katty | that sounds yummy too |
| 15:47.39 | Katty | did you make it or was it store bought? |
| 15:47.56 | Slugs_ | i vote store |
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| 15:51.00 | leifmadsen | Katty: store bought |
| 15:51.25 | Slugs_ | i win |
| 15:51.25 | Katty | well so much for making spaghetti |
| 15:51.31 | Katty | the beef is still frozen solid |
| 15:51.31 | leifmadsen | and now I'm sufficiently hungry so I'm going to go heat it up |
| 15:51.34 | leifmadsen | burn |
| 15:51.40 | leifmadsen | Katty: just cook it in a pan |
| 15:51.46 | Katty | no i mean it's like....SOLID |
| 15:51.52 | Katty | 3lbs |
| 15:51.52 | leifmadsen | ground beef, or ...? |
| 15:51.53 | Katty | solid |
| 15:51.55 | Katty | yes, ground beef |
| 15:52.07 | Katty | it's a big ole HUNK |
| 15:52.08 | leifmadsen | it'll cook in a pan |
| 15:52.16 | leifmadsen | just takes a little longer :) |
| 15:52.20 | Katty | meh |
| 15:52.26 | Katty | i'll just wait for it to thaw |
| 15:52.33 | Katty | afks |
| 15:52.37 | Slugs_ | run cold water over it |
| 15:52.43 | Slugs_ | in the sink |
| 15:52.51 | Slugs_ | not hot. |
| 15:53.07 | Slugs_ | im sure you knew that tho |
| 15:54.40 | KingDavidNYC | Hello guys, can anyone please tell me how do I get the value of a the other channel that is bridged to ${CHANNEL}? |
| 15:56.26 | chuckf | Katty: put it on a metal plate to help the thawing process |
| 15:56.48 | KingDavidNYC | Katty: microwave on defrost for 3 minutes |
| 15:56.51 | leifmadsen | I usually put it on a pan to thaw it out -- crazy how much faster it'll thaw on something metal |
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| 16:06.12 | QbY | PostgreSQL RealTime: Failed to query database. Check debug for more info. --- Exactly WHICH debug would this be referring to? I've checked the debug log---sip set debug on-- |
| 16:08.39 | QbY | Also, in 1.6 does anyone know of an issue with a command like "Dial(SIP/0006283E10ED&SIP/0016C77CB1B5)" - Both phones will ring, but when answered no audio |
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| 16:13.13 | KingDavidNYC | <PROTECTED> |
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| 16:20.47 | *** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-xrqlboaxtahcgwli) |
| 16:27.07 | aceio | hi |
| 16:27.40 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:27.50 | Naikrovek | hola |
| 16:28.28 | aceio | i having a strang problem |
| 16:28.30 | *** join/#asterisk toddejohnson (~toddejohn@ppp-70-226-210-72.dsl.spfdil.ameritech.net) |
| 16:28.38 | Naikrovek | your question: ask it. |
| 16:29.05 | aceio | my ip phone are not ringing the correct extentions |
| 16:29.25 | bmoraca_work | they're ringing whatever "extensions" you tell them to |
| 16:29.31 | Naikrovek | yes |
| 16:29.43 | Naikrovek | verify taht the phones are configured properly and that there's no mixups in the dialplan |
| 16:30.00 | aceio | i hav done that |
| 16:30.13 | aceio | sip show peers |
| 16:30.21 | Naikrovek | okay |
| 16:30.29 | aceio | has dup ip addr |
| 16:30.38 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:30.46 | Naikrovek | then you need to fix how your phones obtain IPs |
| 16:31.06 | Naikrovek | check for duelling DHCP servers or something |
| 16:31.35 | aceio | while the phone have have different address |
| 16:31.51 | Naikrovek | aceio: do you have qualify turned off? |
| 16:31.52 | ecrane | Naikrovek: is that anything like the dueling banjos? |
| 16:32.18 | aceio | on the phones |
| 16:32.31 | Naikrovek | aceio: in the extension config in asterisk |
| 16:32.36 | Naikrovek | look for qualify=no |
| 16:32.48 | aceio | okay |
| 16:34.02 | aceio | that entry is not in my config |
| 16:34.05 | Naikrovek | k |
| 16:34.16 | aceio | should i be adding it |
| 16:34.24 | Naikrovek | no, default is yes, and it should be yes |
| 16:34.34 | Naikrovek | so you can forget the qualify thing |
| 16:34.43 | aceio | okay |
| 16:34.59 | Naikrovek | i'm wondering why asterisk can't keep the IP phones straight |
| 16:35.09 | bmoraca_work | aceio: this sounds like a phone configuration issue. i.e. you told two phones to register to the same peer. |
| 16:35.22 | bmoraca_work | sip debug will illustrate that |
| 16:35.37 | bmoraca_work | console output of failed calls will confirm |
| 16:36.01 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 16:36.15 | *** part/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 16:36.26 | bmoraca_work | either way, sip debug will tell you what the phones think they're supposed to be doing, and what asterisk believes is happening. |
| 16:36.32 | aceio | okay let me get some output |
| 16:40.35 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:40.42 | aceio | this is my output http://pastebin.com/ep1kEFAv |
| 16:41.16 | aceio | so from ext 1260 i called 1261 and rang 1262 |
| 16:42.48 | aceio | this is also my sip show peers http://pastebin.com/xrGaXz05 |
| 16:44.32 | aceio | bmoraca_work: can you see what is worng |
| 16:45.02 | KingDavidNYC | hi, can anyone please tell me how do I get the value of a the other channel that is bridged to ${CHANNEL}? |
| 16:49.12 | qxork | KingDavidNYC: did you check the bridge peer value in core show channel X |
| 16:49.16 | *** join/#asterisk dymaxion_ (~dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 16:50.08 | KingDavidNYC | qxork: I mean I need to get the value from the dialplan |
| 16:51.10 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:51.20 | KingDavidNYC | qxork: one way I am trying is with the G() option of the dialplan, but asterisk terminates the call right after it branches to the G context |
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| 17:05.54 | Naikrovek | slow in here today |
| 17:05.59 | Naikrovek | katty isn't around to kickstart people |
| 17:06.04 | Naikrovek | like fonzie with the jukebox |
| 17:08.42 | chuckf | she punches people and they start talking? |
| 17:09.02 | Naikrovek | more or less, yes, but not with her fist, she punches people with words. |
| 17:09.11 | Naikrovek | in a nice way |
| 17:09.22 | Naikrovek | like by saying "hi" or ":)" or whatever |
| 17:09.47 | chuckf | :) |
| 17:09.51 | Naikrovek | hi |
| 17:10.04 | outtolunc | howdy |
| 17:10.07 | chuckf | that wasn't so hard was it? |
| 17:10.17 | chuckf | and you need her to do that for you? |
| 17:10.23 | Naikrovek | no, of course not |
| 17:10.35 | Naikrovek | it's just different when she's absent |
| 17:10.45 | Naikrovek | not better, not worse, just .... different |
| 17:11.54 | *** join/#asterisk DarthPointer (~no@82.218.68.216.DED-DSL.fuse.net) |
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| 17:44.58 | hardwire | hmm.. |
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| 17:49.59 | KingDavidNYC | Naikrovek: Hi, would you know how do I get the value of the bridged channel on the dialplan? |
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| 17:52.15 | Naikrovek | KingDavidNYC: sorry. dunno |
| 17:52.27 | *** join/#asterisk rubbs (~rubbs@cpe-71-72-56-140.neo.res.rr.com) |
| 17:53.23 | [TK]D-Fender | KingDavidNYC: what "value"? From which specific call? |
| 17:55.19 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 17:55.31 | rubbs | hello, I was here last week with some newbie questions, and I'm back with a few more. |
| 17:55.40 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:56.15 | hardwire | any direct method for getting CDR variables into the app_voicemail emailbody? |
| 17:56.28 | hardwire | I'm guessing I'm going to have to use a script. |
| 17:56.30 | rubbs | I was wondering what is needed (hardware/software) to get a callable number that comes into an asterisk system. |
| 17:57.05 | hardwire | !ITSP |
| 17:57.06 | hardwire | err |
| 17:57.10 | hardwire | ~ITSP |
| 17:57.11 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:57.18 | hardwire | those guys can provide inbound DID |
| 17:57.34 | Naikrovek | rubbs: no hardware required if you go with an ITSP |
| 17:57.36 | hardwire | rubbs: so your asterisk box just needs to be accessable from the Internet |
| 17:57.49 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 17:58.49 | rubbs | cool I'll look into that |
| 17:58.52 | rubbs | thank you. |
| 17:59.08 | rubbs | I'm good at reading up on things, i just don't know where to start sometimes |
| 18:00.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:03.48 | *** join/#asterisk adam_g (~adam@static-173-50-141-22.ptldor.fios.verizon.net) |
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| 18:10.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:10.47 | Kobaz | http://laughingsquid.com/ipad-will-it-blend/ |
| 18:11.42 | Naikrovek | of course it'll blend |
| 18:11.51 | Naikrovek | unless that is a mockery will-it-blend video |
| 18:12.42 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 18:13.17 | KingDavidNYC | [TK]D-Fender: Hi, I need the channel value of the called party right after the call was bridged, on the dial plan, while the call is in progress |
| 18:14.07 | Kobaz | heh |
| 18:18.14 | rubbs | ok, dumb question here, but when I look at a plan and it says it has up to 4 channels is that the number of simultanious calls it can handle at once? |
| 18:19.28 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:20.45 | *** join/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 18:23.37 | [TK]D-Fender | KingDavidNYC: which called party? |
| 18:23.38 | Naikrovek | rubbs: yes. that is the number of simultaneous calls into and out of the building that can happen at once |
| 18:23.49 | [TK]D-Fender | KingDavidNYC: You know AMI announces all of this stuff already right? |
| 18:23.49 | Naikrovek | rubbs: you can have LOTS more than that internally if you wish |
| 18:24.55 | rubbs | Naikrovek: yeah I figured you weren't limited for internal, (at least theoritically) |
| 18:25.27 | Naikrovek | rubbs: limited by phone server hardware and i suppose the bandwidth to the phone server |
| 18:25.34 | KingDavidNYC | [TK]D-Fender: party A calls party B on the dial plan. I can get the channel value of party A using ${CHANNEL}, then using DIAL() party A connects to party B, call is bridged, and now I need to know the value of the channel of party B |
| 18:25.36 | Naikrovek | but yeah |
| 18:26.15 | [TK]D-Fender | KingDavidNYC: "coer show channel X" will tell you things like that, and there are plenty of other AMI functions for this |
| 18:26.50 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 18:28.50 | KingDavidNYC | TK-Fender: yes, but isn't there a way within the dialplan? I am trying to use maybe the G() option of the dial(), which sends the call to another branch after connected, this allows me to get the second channels value!, but then, because the call is back in the dialplan, it terminates it |
| 18:29.29 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 18:29.47 | KingDavidNYC | TK-Fender: if you can suggest me what to do for the call not to terminate after it branches after the G option of the dial(), ot will be great |
| 18:30.17 | [TK]D-Fender | KingDavidNYC: "core show application dial" <- check this and the channelvariables doc to see what gets set if antyhign. Don't bet on it though |
| 18:31.37 | KingDavidNYC | TK-Fender: one way I can do it, is using the AMI save the second channel to a database and then retrieve after connected, but that will be so going around the world |
| 18:31.44 | *** part/#asterisk bsaxon (~bsaxon@mail.stonegaterealty.cc) |
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| 18:35.16 | *** join/#asterisk A500mg (~x@ACaen-156-1-77-50.w90-51.abo.wanadoo.fr) |
| 18:35.31 | A500mg | hellooo :) |
| 18:38.00 | chuckf | no one is here |
| 18:38.51 | leifmadsen | true |
| 18:40.05 | chuckf | well leifmadsen is here, but that's besides the point |
| 18:40.26 | chuckf | :D |
| 18:41.01 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 18:43.52 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 18:45.25 | rubberneck | Need some help with dial-plan syntax. I have something like this PATTERN,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN:-4},GatewayB) |
| 18:45.41 | rubberneck | n(GatewayB),Dial(SIP/${EXTEN:-4}@proxy-1) |
| 18:46.05 | A500mg | i'm here :) |
| 18:46.17 | rubberneck | Asterisk complains about the way im naming GatwayB. |
| 18:49.28 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:49.35 | Qwell | rubberneck: 1.6? I think it's complaining about your GotoIf syntax |
| 18:49.36 | rubberneck | is PATTERN,PRIORITY(NAME),APP not correct? |
| 18:50.12 | Qwell | Though it's entirely possible that GotoIf doesn't support priority labels. I'd be a little surprised if that were the case, but.. |
| 18:50.36 | TheDavidFactor | rubberneck: what version of asterisk? |
| 18:51.33 | A500mg | gotoif support priority label, i use it |
| 18:51.58 | rubberneck | Qwell: yea is says " Priority 'GatewayB' must be a number > 0, or valid label" |
| 18:52.24 | Qwell | rubberneck: pastebin the 2 lines in question |
| 18:52.26 | rubberneck | TheDavidFactor: 1.6.2.1 |
| 18:52.49 | Qwell | (the full lines) |
| 18:54.00 | rubberneck | http://pastebin.com/7gWbV7pq |
| 18:54.59 | Qwell | looks right to me |
| 18:55.26 | Qwell | show me the full error line? |
| 18:56.23 | A500mg | any idea of syntax for function CHANNELS() ? |
| 18:56.33 | A500mg | (the pattern) |
| 18:56.47 | QbY | i do believe realtime sucketh |
| 18:57.00 | DarthPointer | I have got asterisk throwing an error pbx.c:1555 func_args: Can't find trailing parenthesis for function 'LEN('? Any tips for tracking down specifically where it's hitting this problem? |
| 18:57.16 | A500mg | loss parenthesis ... |
| 18:57.20 | Qwell | DarthPointer: not without seeing the line |
| 18:57.22 | A500mg | show the entire line please |
| 18:57.43 | DarthPointer | [2010-04-05 14:56:42] WARNING[19237]: pbx.c:1555 func_args: Can't find trailing parenthesis for function 'LEN('? |
| 18:57.51 | A500mg | ...... |
| 18:57.56 | Qwell | facepalms |
| 18:57.58 | A500mg | the line in your extensions.conf !!! |
| 18:58.19 | rubberneck | Qwell: i found the problem, its becuase im trimming off the "+" int the gotoif line. |
| 18:58.32 | Qwell | rubberneck: ahh. subtle. |
| 18:58.55 | DarthPointer | sorry. htat's what I'd like to figure out; I have a similar line about 20 places in my config |
| 18:59.07 | rubberneck | Qwell: yeah, thanks for the help though. You caused me to take another look at that line. |
| 18:59.49 | DarthPointer | In 7 different Queue's, faxing config, etc. |
| 19:00.38 | DarthPointer | I do have freepbx 2.7.0.1 running as well |
| 19:01.55 | A500mg | you don't know which line is used at this moment ? |
| 19:01.59 | TheDavidFactor | DarthPointer: if you're doing your config through freepbx try #freepbx, if you're editing config files by hand carry on :-) |
| 19:02.03 | A500mg | we can help if you don't have the line |
| 19:02.35 | TheDavidFactor | is it something happens frequently? |
| 19:03.16 | TheDavidFactor | Do you CDRs or any other logs that might help you figure out which dialplan logic it was running through at the time? |
| 19:03.23 | DarthPointer | <shamefully>much of the config is done through freepbx</shamefully> however I'm trying to figure out where the error is coming from. I do have other logs |
| 19:03.45 | [TK]D-Fender | DarthPointer: Did you code any dialplan in this yourself? |
| 19:04.03 | DarthPointer | no. |
| 19:04.29 | DarthPointer | I understand this probalby a freepbx introduced dialplan problem |
| 19:05.09 | DarthPointer | what I"m looking for are some tips on tracking down where the problem is; for example, I used "core set debug 4 app_queue.c" to locate a bug I just filed w/ freepbx |
| 19:05.35 | DarthPointer | I can see the error @ the asterisk prompt; I'm just having trouble tracing it back to the dialplan |
| 19:06.00 | TheDavidFactor | well, you can increase your verbosity to see all the dialplan lines as they execute (try 3 or higher) |
| 19:06.20 | TheDavidFactor | maybe by seeing what executes before the error you can track it down |
| 19:08.24 | [TK]D-Fender | DarthPointer: if this isn't your code, take this to #freePBX. |
| 19:08.48 | [TK]D-Fender | DarthPointer: Anything you attempt to modify gets pretty much blown away unless you know what you're doing. |
| 19:10.21 | DarthPointer | thank you. Will do. I am aware of the overwrite. I just upgraded versions and am running down some errors that have appeared. |
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| 19:19.53 | adam_g | hi, im using a PHP script as an AGI script. the script quickly uses pnctl_fork() to fork a child and exit(0). i would have assumed execution goes back to dialplan after first exit, but asterisk seems to wait for child to exit before continuing. anyone have any tips? |
| 19:20.10 | Kobaz | don't fork |
| 19:20.58 | Kobaz | If you don't care about the output, use TrySystem |
| 19:21.06 | Kobaz | agi is going to be useless if you fork |
| 19:21.21 | aceio | i am try to set the ip 50 option an debians dhcp |
| 19:21.37 | aceio | and it's not working |
| 19:21.45 | Kobaz | why not just set it up correctly in the first place, and then it'll always be there |
| 19:21.49 | Kobaz | oh |
| 19:21.56 | Kobaz | you're not adam, never mind |
| 19:21.56 | Kobaz | heh |
| 19:22.00 | Kobaz | aceio: #debian |
| 19:23.21 | *** join/#asterisk Z_God (~julius@wlan232249.mobiel.utwente.nl) |
| 19:24.37 | adam_g | Kobaz, forking to call some external APIs that take 7-10s to complete and the result shouldnt have any effect on dialplan execution. i will look into TrySystem. |
| 19:24.55 | Kobaz | you definitly don't want agi then |
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| 19:43.48 | adam_g | Kobaz, TrySystem() is what i wanted, thanks |
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| 19:45.43 | Kobaz | no problem |
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| 20:16.38 | Fubard | for polycom phones: does sip.ld go into your ftp root folder, or is it retrieved through some other method? |
| 20:16.46 | Naikrovek | ftp root |
| 20:17.12 | Fubard | well i'm angry at the world then. i must be doing something else wrong. tyvm Naikrovek |
| 20:20.29 | Naikrovek | phones not pulling firmware? |
| 20:20.48 | Naikrovek | verify that the ftp server log shows the phone logging in |
| 20:20.57 | Naikrovek | and check to see which files are being requested by the phone |
| 20:22.41 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 20:22.43 | *** join/#asterisk adam_g (~adam@static-173-50-141-22.ptldor.fios.verizon.net) |
| 20:25.29 | Fubard | omfg i had the phone connecting to the wrong test server ffs! |
| 20:25.37 | Naikrovek | heh |
| 20:26.53 | rubbs | k, two more questions: 1) if a IP phone says "6 call presentation appearances" does that mean the phone can handle 6 channels at once? (if I got this down right, this doesn't have to match the same number of channels the server can handle) and 2) what phones can people recommend for about 20 phone office setup? |
| 20:27.20 | *** join/#asterisk rare1980_ (~rare1980@115.186.20.189) |
| 20:27.24 | rubbs | I've been told aastra is a good manufacturer |
| 20:27.47 | Naikrovek | rubbs: that just means there are 6 line appearances. that means the phone can have 6 different extensions. the number of calls it can handle simultaneously is probalbly 2x or 4x that |
| 20:27.59 | Naikrovek | rubbs: i use polycom IP 321s throughout my employer |
| 20:28.09 | Naikrovek | rubbs: $80 each |
| 20:28.18 | rubbs | anyone have any others? I'd appreciate model numbers/names too |
| 20:28.43 | Fubard | i can say cisco is bad and polycom is good. but i'm nubsauce to voip networks. |
| 20:28.53 | rubbs | have you had any issues with polycom support? I've been told they don't support asterisk systems |
| 20:29.04 | Naikrovek | avoid cisco and grandstream phones |
| 20:29.07 | Fubard | I know the PolyCom 501's are absolutely delicious, easy to setup. |
| 20:29.20 | Naikrovek | rubbs: they support asterisk, they even have documentation on how to use the phones with asterisk |
| 20:29.39 | Naikrovek | rubbs: but you can't contact polycom directly if you have a phone issue, you must go through a reseller |
| 20:29.58 | rubbs | Naikrovek: ah, now I c thanks |
| 20:30.07 | rubbs | what reseller did you go through? |
| 20:30.16 | Naikrovek | telephonydepot.com |
| 20:30.53 | rubbs | cool thanks |
| 20:31.09 | rubbs | have you had to get support at all for it? |
| 20:31.18 | Naikrovek | never need to contact them, ever |
| 20:31.28 | Naikrovek | have had 0 problems with the 80-ish phones |
| 20:31.58 | Naikrovek | IP 321s, 320s, 330s and 331s. |
| 20:32.18 | rubbs | wow, that's a pretty resounding recommendation then. |
| 20:32.30 | rubbs | we will take another look at polycoms then. |
| 20:32.37 | Naikrovek | you get what works for you |
| 20:32.46 | Fubard | rubbs: dunno if this will help or not but: if you go with polycoms avoid the web UI to setup the phones, using ftp will make you giggle like a school girl |
| 20:33.03 | Naikrovek | you compile some requirements, see which phones can meet those requirements, then ignore that data and get polycom |
| 20:33.13 | Fubard | lmao! |
| 20:33.15 | Naikrovek | yes, ftp setup is flawless |
| 20:33.17 | Naikrovek | well |
| 20:33.20 | Naikrovek | once you understand it |
| 20:33.36 | rubbs | haha |
| 20:33.43 | rubbs | ok, good to know. I'll keep the ftp in mind |
| 20:33.48 | Fubard | yea, its tricky at first to a new user but it begins to make good solid sense after a short time |
| 20:33.52 | rubbs | so aastra's probably not so good then. |
| 20:34.10 | Naikrovek | rubbs: i am expecting new phones in the next day. i don't even need to open the boxes to configure them, i just hand them to the user, let them plug them in |
| 20:34.18 | Naikrovek | in every case, the phone will configure itself and start working |
| 20:34.29 | Naikrovek | dhcp option 66 + ftp server |
| 20:34.51 | rubbs | cool |
| 20:34.56 | rubbs | definately will keep that in mind |
| 20:35.08 | Naikrovek | i have a script to generate the config files the phones need, and i've already configured the phone server for the new extensions |
| 20:35.12 | Fubard | as much as i hate to use MS, yes i do infact ~"get my kicks on option 66" |
| 20:35.49 | Naikrovek | so when the phones get plugged in, they use dhcp option 66 to get the ftp server address, user & pass, then log into the phone server, download new firmware, reboot, then download config files and start working |
| 20:35.52 | Naikrovek | it's nice |
| 20:36.22 | *** join/#asterisk Keyptor (~Keyptor@195.110.6.1) |
| 20:36.39 | *** part/#asterisk Keyptor (~Keyptor@195.110.6.1) |
| 20:36.47 | chazzam | I believe asterisk itself can handle provisioning polycom phones though |
| 20:36.58 | rubbs | so the boss is wondering (I should really just get him to log onto IRC) if there are any features we should really consider making sure we have. and if things like display screens on the phone really matter. |
| 20:37.01 | Naikrovek | asterisk can't generate the polycom config files or anything like that |
| 20:37.03 | chazzam | and it auto-generates the configuration files for them |
| 20:37.22 | chazzam | its all configured in phoneprov.conf |
| 20:37.34 | Naikrovek | rubbs: if you need a higher-resolution display, the 32x, 33x phones won't cut it. |
| 20:37.45 | Katty | hi |
| 20:37.50 | Naikrovek | katty where you been |
| 20:38.06 | Katty | workin n stuff |
| 20:38.32 | Naikrovek | chazzam: what asterisk version are you using? elastix or something? |
| 20:38.55 | chazzam | no, just regular open source |
| 20:39.03 | Naikrovek | i've never even heard of phoneprov.conf |
| 20:39.16 | chazzam | is it not in your /etc/asterisk directory? |
| 20:39.21 | chazzam | I know it is there in 1.4 and up |
| 20:39.25 | Naikrovek | no. |
| 20:40.22 | Naikrovek | ah i see it on my other server |
| 20:40.41 | Naikrovek | i wrote my own provisioning system for my phones, i see no reason to bother with this |
| 20:40.42 | Naikrovek | heh |
| 20:41.01 | chazzam | huh, well now that I look I can't seem to find it in 1.4, but it is in 1.6.X |
| 20:41.10 | beek | hugs Katty |
| 20:41.15 | chazzam | but yeah, if you wrote your own already, you wouldn't need it |
| 20:41.52 | [TK]D-Fender | [16:38]<Naikrovek>i've never even heard of phoneprov.conf <- its laughable |
| 20:41.54 | [TK]D-Fender | BBIAB |
| 20:43.51 | rubbs | thanks for all the help everyone btw. I'm going to try reading more from http://www.asteriskdocs.org/ |
| 20:44.32 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 20:44.33 | Fubard | hmmm, i keep getting an error when the phone starts up - error saving application APP_FILE_PATH, which is set to sip.ld ... |
| 20:44.50 | Katty | hugs beek |
| 20:45.32 | Naikrovek | Fubard: do you have the combined firmware or the split firmware |
| 20:45.45 | Naikrovek | rubbs: lots and lots of polycom expertise in here if you ever need help with them |
| 20:46.10 | Fubard | Naikrovek: it doesn't say split or combined, cuz its so ancient (3.1.1) |
| 20:46.18 | Naikrovek | 3.1.1 is not old |
| 20:46.26 | Naikrovek | 1.8 is ancient |
| 20:46.40 | Fubard | Naikrovek: old timey then? |
| 20:46.46 | Naikrovek | Fubard: are there a bunch of different .ld files, or just one sip.ld |
| 20:47.07 | Fubard | Naikrovek: only sip.ld, phone1.cfg, <MAC>.cfg |
| 20:47.23 | Naikrovek | Fubard: you have the combined firmware |
| 20:47.57 | Fubard | the zip had a ton of junk files in it, but i only took the sip.ld out |
| 20:47.58 | Naikrovek | get the split firmware, and just unzip it right into your ftp login's root directory |
| 20:48.09 | Naikrovek | Fubard: leave them all in there |
| 20:48.25 | Naikrovek | doesn't hurt anything, but you can remove something necessary by mistake |
| 20:48.41 | Fubard | Naikrovek: hmmmm ... sound reasoning. will do. |
| 20:49.12 | Fubard | i'm not seeing a 3.1.1 split, i'll just do 3.1.2 split and hope my phone is happy with it |
| 20:49.17 | *** join/#asterisk adam_g (~adam@173-12-184-89-oregon.hfc.comcastbusiness.net) |
| 20:49.28 | Naikrovek | what model phone |
| 20:49.36 | Fubard | 501 |
| 20:50.05 | Fubard | erm, polycom sp 501* |
| 20:50.08 | Naikrovek | why not get 3.1.6 |
| 20:50.19 | Naikrovek | oh wit |
| 20:50.20 | Naikrovek | wait |
| 20:50.27 | Naikrovek | http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip501.html |
| 20:50.32 | Naikrovek | you getting your firmware from there? |
| 20:50.43 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 20:50.53 | Fubard | mr. bossman has like 3.1.1.0137 on all his phones. trying to make the test network as similar to his setup as possible |
| 20:51.13 | Fubard | yessir, that exact same link |
| 20:51.18 | Naikrovek | okay. |
| 20:51.36 | Naikrovek | make sure your bootrom is up to date, then get whatever firmware you want |
| 20:51.40 | Naikrovek | but get the split one |
| 20:51.46 | Naikrovek | much less headache with the split |
| 20:52.41 | Fubard | weeeeeeelll. ok fine you twisted my arm. i'll indulge in a little new'ish'ness |
| 20:53.26 | Naikrovek | those minor revision numbers only contain bugfixes. if you're 3.1.x and he's 3.1.x functionally you're identical |
| 20:53.27 | Fubard | he's got 3.2.3.x bootROM on the live network |
| 20:53.34 | Naikrovek | now that's old |
| 20:53.38 | *** join/#asterisk KavanS (~KavanS@173-12-184-89-oregon.hfc.comcastbusiness.net) |
| 20:53.52 | Naikrovek | in order to use split firmware you need bootrom 4.0 or greater |
| 20:54.05 | Naikrovek | HIGHLY recommend you get latest bootrom available for your phone |
| 20:54.13 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 20:54.20 | Naikrovek | again, won't change the phone functionality, just lets you use split firmware |
| 20:54.52 | Naikrovek | polycom needs to send me a phone or two for all the recommending i'm doing |
| 20:54.54 | Naikrovek | ... |
| 20:55.18 | Naikrovek | couple 670s would do me fine |
| 20:55.23 | Fubard | lol, polycom's spokesperson |
| 20:55.51 | Naikrovek | i'm just a fan of their ip phones |
| 20:55.55 | Fubard | if you start a webpoll for who should be their spokesperson, i'd vote for you |
| 20:56.11 | Qwell | Fubard: You've clearly never met [TK]D-Fender |
| 20:56.12 | Naikrovek | well i had some grandstream in here and they were a nightmare compared to my polycoms |
| 20:56.32 | Naikrovek | yeah [tk]d-fender is also a big shill wannabe |
| 20:56.35 | Fubard | well then, perhaps you'll have to fight for my vote after all Naikrovek |
| 20:56.35 | Naikrovek | or something... |
| 20:56.39 | Naikrovek | heh |
| 20:56.44 | beek | Qwell: Be careful when you invoke the name of a deity. |
| 20:56.51 | Naikrovek | me and [tk] love our polycoms, there's no dooubt |
| 20:56.54 | Qwell | beek: Bacon. |
| 20:57.13 | Fubard | [tk]d-fender, peace be upon him. |
| 20:57.35 | Fubard | Bacon, peace be upon ... it? |
| 20:57.38 | beek | Wait guys... do you mean that there is another company other than Polycom that makes IP phones? |
| 20:57.44 | Naikrovek | Bacon, grease be upon it |
| 20:57.54 | Fubard | beek: yes, its call tin cans and a string |
| 20:57.55 | Naikrovek | beek: no. polycom > all |
| 20:58.04 | Fubard | called* even |
| 20:58.17 | beek | Naikrovek: That's what I've always thought. |
| 20:59.17 | Naikrovek | peace |
| 20:59.50 | *** join/#asterisk Whtsup (~sssi@115.167.107.71) |
| 21:00.11 | Whtsup | how can i loadbalance call b/w 2 sip accounts |
| 21:01.22 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:04.02 | ecrane | what is b/w? |
| 21:04.09 | Chainsaw | Bandwidth. |
| 21:04.25 | Chainsaw | (Or a failed attempt to shorten "between") |
| 21:04.36 | rubbs | Ok, so if I'm looking for a documentation resource for VoIP land newbs where would I go? I've found the asteriskdocs.org book, and that's helped a lot already, but I feel like I need to understand how everything fits together, for example pstn's and things. is there a good place to start with that stuff? |
| 21:04.45 | Fubard | i thought he meant 'but with" |
| 21:05.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:05.34 | Fubard | rubbs, check this out |
| 21:05.37 | Fubard | ~book |
| 21:05.38 | 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 |
| 21:05.52 | chazzam | he said he already found that... |
| 21:06.07 | Fubard | oh ... sorry was distracted while i read taht lol |
| 21:06.19 | chazzam | it sounds like he wants something basic about telephony in general without being specific to voip |
| 21:06.35 | chazzam | but I am not sure where to get that. I thought there was a brief chapter about it in the book |
| 21:07.05 | Fubard | tbh, the i got what i needed on that material from the preface and the first chapter or 2 in that asteriskdocs.org pdf |
| 21:07.09 | chazzam | Fubard: I think I read it three times before I caught all of what he said |
| 21:07.28 | rubbs | chazzam: yes, you are right |
| 21:07.30 | [TK]D-Fender | Since when was * specifically VoIP? |
| 21:08.08 | Qwell | [TK]D-Fender: Thursday |
| 21:08.14 | rubbs | I guess it isn't, but I'm looking into converting our business lines into VoIP |
| 21:08.27 | [TK]D-Fender | rubcertaainly no need to. I haven't |
| 21:08.31 | rubbs | I'm still very new and I hate to bother people here, so I'm tryign to get ideas on where to look |
| 21:08.35 | [TK]D-Fender | rubbs: certainly no need to. I haven't |
| 21:09.02 | [TK]D-Fender | rubbs: * processes calls. What kind of lines & tech you use is up to you |
| 21:09.23 | rubbs | ah, |
| 21:09.41 | rubbs | so you just convert to analog when you go "external" |
| 21:09.51 | [TK]D-Fender | rubbs: Want to use physical lines? Get an interface of some kind. |
| 21:10.12 | rubbs | well see that's the thing. right now in this building we use physical lines. but we are moving to a new building |
| 21:10.43 | rubbs | we were looking into foregoing the physical lines all together |
| 21:10.47 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com) |
| 21:11.19 | chazzam | mm, I guess one thing to keep in mind is emergency service then |
| 21:11.28 | [TK]D-Fender | rubbs: How do/will you have? |
| 21:11.35 | chazzam | not all VOIP providers do anything or offer anythingconcerning it |
| 21:11.47 | chazzam | like a guarantee |
| 21:12.03 | rubbs | chazzam: if we do go the all VoIP route we'd look into the emergency service thing |
| 21:12.45 | rubbs | we have 5 physical lines now |
| 21:13.02 | rubbs | we could get away with having 4 "external" channels/lines in the new place |
| 21:13.11 | rubbs | but somewhere around 5 is desirable at this point |
| 21:13.38 | Whtsup | how can i loadbalance call between 2 sip accounts |
| 21:13.47 | Whtsup | i have 2 sip account |
| 21:13.55 | Whtsup | and i want to make loadbalance |
| 21:14.08 | p3nguin | What would be the outcome of "loadbalance call between 2 sip accounts"? |
| 21:14.08 | Whtsup | one call will go from 1st sip another call will go from 2nd sip |
| 21:14.14 | p3nguin | Two people would share the call? |
| 21:14.51 | Whtsup | yah |
| 21:15.02 | Whtsup | i want to balance both account |
| 21:15.05 | p3nguin | That doesn't make sense to me. |
| 21:15.09 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:15.12 | [TK]D-Fender | Whtsup: Code the selection in your dialplan |
| 21:15.24 | Whtsup | what shall i code |
| 21:15.35 | p3nguin | If you want to balance incoming calls, queue them and use a strategy to keep the calls equal between members. |
| 21:15.38 | [TK]D-Fender | p3nguin: he just wants to alternate usage |
| 21:15.58 | rubbs | ah, ok nother (probably) stupid question. We have a few programmers who are off site. would we need to tie up an "external" call when they call in to our office, or is there a way for them to use an IP phone and call internally? |
| 21:16.43 | [TK]D-Fender | rubbs: Sure |
| 21:16.44 | chazzam | rubbs, ip phones can be remote either straight through the internet or over VPN |
| 21:16.53 | rubbs | awesome |
| 21:16.57 | [TK]D-Fender | rubbs: Call comes in to your server from anywhere however you want |
| 21:17.19 | [TK]D-Fender | rubbs: If they have SIP tech on thier side then it can jsut talk IP to your system direct over the internet/WAN/etc |
| 21:17.24 | rubbs | you guys are great, thanks for all your help. I'll try to read up more so I don't bother you guys so much |
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| 21:17.34 | *** join/#asterisk bmg505 (~leon@196-209-71-68-rndf-esr-3.dynamic.isadsl.co.za) |
| 21:17.35 | rubbs | cool |
| 21:17.59 | *** join/#asterisk arpu (~arpu@client054.9360.easyline.at) |
| 21:19.24 | Whtsup | can i paste my dialplan here |
| 21:20.19 | *** join/#asterisk arpu (~arpu@client054.9360.easyline.at) |
| 21:20.47 | [TK]D-Fender | Whtsup: No. Go set a flag using AstDB or a global variable and flip it between each call. Pick your destination accordingly |
| 21:21.01 | [TK]D-Fender | Whtsup: We told you all of this LAST WEEK |
| 21:22.34 | rubbs | Thanks for all your help folks |
| 21:22.46 | [TK]D-Fender | Whtsup: this is about 2 lines of basic dialplan.... |
| 21:23.03 | [TK]D-Fender | rubbs: hope to hear your project coming along. |
| 21:23.18 | rubbs | I'll post back with updates ;) |
| 21:24.06 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 21:27.48 | Whtsup | i m newbie |
| 21:27.52 | Whtsup | :s |
| 21:28.06 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 21:28.07 | Whtsup | i dont know much about dialplans how to use global variables |
| 21:28.58 | QbY | Other than Queues, and Dial(SIP/DEVICE&SIP/DEVICE&SIP/DEVICE&SIP/DEVICE) is there another way to ring a group of phones.... |
| 21:30.04 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 21:32.14 | [TK]D-Fender | Whtsup: Well the dialplan is 90% of Asterisk. You'd better get to it. I already gave you all the commands you'll need last week. |
| 21:32.49 | Fubard | Naikrovek: i know you're not here but you are a gawd |
| 21:33.30 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:36.17 | Whtsup | well you didnt gave me the commands |
| 21:36.28 | Whtsup | just told me use astdb or gv |
| 21:38.32 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 21:39.10 | [TK]D-Fender | Whtsup: gotif, set, DB() ,_ if you want something truly persistent. That is all |
| 21:40.11 | Whtsup | ok thanks |
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| 22:04.02 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:04.07 | miamiseb | Hey all, I've got a fax line that is coming in through a dahdi channel and I don't want* to anwser it, but I've got an inbound route that is matching it and I just want it to leave it alone, not anwser it, not send it to a ring group or ivr, just let the fax machine deal with it, any way to get that accomplished? |
| 22:05.32 | miamiseb | I'm not sure if what I would need is to simply do a long Wait (I think that might hang my line if the fax doesn't anwser or keep the line open after its done) or just somehow force it to never Anwser() or what |
| 22:06.13 | Slugs_ | . |
| 22:07.02 | manxpower | exten => whateveryouhave/thecidoffax,1,Wait(360) |
| 22:07.15 | manxpower | It's called the ex-girlfriend option |
| 22:07.22 | miamiseb | I tried setting it to non-existant context but it just plays goodbye and hangs up |
| 22:07.34 | manxpower | when you add a / after the extension, that is a cid match |
| 22:07.40 | miamiseb | Are we sure that won't keep the line up for 360? |
| 22:07.44 | manxpower | pattern matches would still have _ |
| 22:07.55 | manxpower | miamiseb: What type of connection to the telco do you have? |
| 22:08.01 | bmoraca_work | why is it plugged in to Asterisk if you never want to answer it? |
| 22:08.10 | miamiseb | because I want to use it to dial 911 outbound |
| 22:08.19 | bmoraca_work | ahh |
| 22:08.25 | manxpower | miamiseb: then why are you answering it for inbound? |
| 22:08.35 | miamiseb | that one dahdi channel is just a normal TDM line, the other lines are lines I'm providing via SIP |
| 22:08.38 | manxpower | This sounds a lot like a FreePBX question. |
| 22:08.45 | miamiseb | manxpower, I'm trying not to, it just does. |
| 22:09.01 | bmoraca_work | or some other GUI |
| 22:09.08 | manxpower | miamiseb: on SIP and PRI you can do something like Hangup(1) to return a disconnected number message to the caller |
| 22:09.17 | manxpower | miamiseb: Sorry, I can't help with GUI stuff |
| 22:09.29 | miamiseb | Right, but the problem is that the fax is on that line, and it needs to anwser to phone. |
| 22:09.46 | miamiseb | thats okay, I don't think it's gui related |
| 22:09.53 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 22:10.38 | bmoraca_work | miamiseb: of course it's GUI related. your GUI is inserting some default routing for unmatched contexts and extensions and thus messing you up. this is trivial if you know what you're doing. |
| 22:11.00 | miamiseb | nods. |
| 22:11.22 | miamiseb | your correct that its inserting a match for all unknown cids, but I can override anything I need to |
| 22:11.23 | manxpower | Why not ask on a channel where people actually use FreePBX? |
| 22:11.49 | bmoraca_work | miamiseb: unfortunately, no one here knows exactly what each and every GUI is doing behind the scenes and no one here really wants to go and dig through tons of source code to figure it out |
| 22:11.55 | miamiseb | I did actually, but if I can come up with an asterisk solution (apparently, override that context and never Anwser() ) then I figured id try to make it work |
| 22:12.18 | manxpower | .part #FreePBX |
| 22:12.20 | *** part/#asterisk manxpower (~ewieling@216.186.151.147) |
| 22:16.24 | mazpe | I have several DID that i route to my cell phone... any recommendations on how to identify which caller id is routing the call and still keeping the caller id? |
| 22:17.17 | mazpe | it will be great if i can append something to the name. But it seems that cellphones ignore the CALLERID(name)="stuff" |
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| 22:21.52 | *** join/#asterisk geneg1 (~gene@bas3-toronto01-1177779082.dsl.bell.ca) |
| 22:24.29 | miamiseb | Now that that is resolved, how can I make it detect weather there is a dial tone or not for proper fall through? |
| 22:24.57 | miamiseb | I've got an active call on dahdi/2 and it'll still dial the digits for 911 through that trunk rather than detecting an active call and falling through to backup trunk for e911 |
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| 22:41.56 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 22:59.12 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:00.52 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 23:02.58 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:04.01 | [TK]D-Fender | miamiseb: Stop plugging things inline with your dahdi channels then. |
| 23:05.22 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 23:05.36 | ruben23 | hi guys whats the port of rtp mdia..? |
| 23:06.03 | [TK]D-Fender | ruben23: Depends. |
| 23:06.05 | *** part/#asterisk adam_g (~adam@173-12-184-89-oregon.hfc.comcastbusiness.net) |
| 23:06.21 | ruben23 | common..? |
| 23:07.05 | [TK]D-Fender | ruben23: No such thing |
| 23:07.14 | [TK]D-Fender | ruben23: RTP is chosen from a RANGE. |
| 23:07.36 | [TK]D-Fender | ruben23: Every call runs on a unique port. |
| 23:09.01 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 23:10.55 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 23:15.14 | Fubard | ruben23: range is 10000-20000 by default |
| 23:16.16 | ruben23 | <PROTECTED> |
| 23:16.27 | Fubard | ruben23: rawr |
| 23:17.02 | p3nguin | According to rtp.conf, that isn't the default range. |
| 23:18.36 | Fubard | wasnt it? i must be going senile. sorry lol, my memory isn't so great |
| 23:18.53 | *** join/#asterisk Slugs_ (~yeah@c-76-97-217-69.hsd1.ga.comcast.net) |
| 23:22.26 | Fubard | ruben23: my bad, p3nguin is right. its 5000+; defined by the application managing the media; unspecified/default ports would be 5004 or 5005 |
| 23:23.31 | p3nguin | 5000-31000 |
| 23:24.03 | p3nguin | but 10000-20000 is a "normal" or "typical" range. |
| 23:24.29 | Fubard | i belong to a cult and we dont believe in numbers over 30999, so i wasn't able to define the max port as explicitly as you could |
| 23:24.53 | Fubard | non-believer! |
| 23:25.21 | Fubard | actually i just dont have the rtp.conf open lol |
| 23:27.53 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 23:28.04 | jaytee | everyone knows that there are no numbers above 27852, heretic!!! |
| 23:29.21 | bmoraca_work | i'm thinking about buying a Cisco 3725 so that I can do some CME implementations...see if i like it better than asterisk |
| 23:43.44 | Fubard | i get dialtone on this polycom501, but i'm almost certain that it doesn't have an extension registered to it. the display name is 'StandPoint IP' even though I've vigorously beat it with a newspaper telling it that its name is '1000' |
| 23:45.34 | Fubard | i've gotta <mac>-phone.cfg with <PHONE_OVERRIDES> <OVERRIDES reg.1.displayName="1000" /> |
| 23:46.14 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 23:46.26 | Fubard | eerm, that cfg file name is <mac>-registration.cfg* |
| 23:47.05 | Fubard | and i call to it within the main cfg, <mac>.cfg |
| 23:57.14 | *** part/#asterisk qxork (~qxork@ip70-171-7-177.ga.at.cox.net) |