| 00:00.03 | kyron | PWahahahahahaha.... 726 16K <--- |
| 00:00.09 | kyron | husimon, ;) |
| 00:00.56 | lmadsen | DUNDi is used more internally for passing information between boxes you control than toll-bypass |
| 00:01.07 | lmadsen | for example, I use it when I build clusters |
| 00:01.18 | *** join/#asterisk real0ne (i=real0ne@adsl-110-185-192-81.marocconnect.net.ma) |
| 00:01.23 | AndyGraybeal_ | but the toll-bypass sounds very interesting :) |
| 00:01.39 | ZaVoid | what do you use it for lmadsen |
| 00:02.00 | lmadsen | ZaVoid: re-read what I just wrote |
| 00:02.01 | [TK]D-Fender | drmessano, You're confusing #dundi with #bdsm again.... |
| 00:02.07 | [TK]D-Fender | drmessano, silly rabbit... |
| 00:02.19 | ZaVoid | yeah i what info between your boxes in a cluster i mean |
| 00:02.26 | drmessano | [TK]D-Fender: I didn't tell you that you could SPEAK yet.. Back in your cage! |
| 00:02.40 | lmadsen | ZaVoid: where people are registered, what system is controlling a particular conference room or queue, etc... |
| 00:02.55 | ZaVoid | ahh i see |
| 00:03.29 | ZaVoid | features not routing configs.. gotcha |
| 00:03.29 | lmadsen | "Hey cluster, I have a call for lmadsen, who has him registered to them?" |
| 00:03.29 | lmadsen | then I route the call to the remote box that replies |
| 00:03.29 | kyron | Does anyone know if buying the 729 CODEC online is instantaneous or if I have to wait for a human to react to my request? |
| 00:03.31 | ZaVoid | i was gonna say.. i got all my carriers in my realtime db.. how i cluster my aterisk farm |
| 00:03.44 | kyron | lmadsen, that's cool |
| 00:03.47 | ZaVoid | i remember gettinga pdf with the license file kyron |
| 00:04.04 | hmmhesays | asterisk farm, that sounds scary |
| 00:04.10 | ZaVoid | lol |
| 00:04.18 | kyron | ZaVoid, as in, as soon as you bought the license it was sent to you via e-mail |
| 00:04.19 | ZaVoid | i call 12 boxes a farm :) or more like a zoo |
| 00:04.26 | kyron | The computers go Moooooooooooo |
| 00:04.29 | ZaVoid | honestly i don't remember that part kyron sorry |
| 00:04.39 | ZaVoid | its been a while since i bought g729 licenses |
| 00:04.49 | lmadsen | ZaVoid: you'll have to come to it360 in Toronto and listen to my talk on clustering I guess :) |
| 00:05.20 | kyron | lmadsen, when is that taking place? |
| 00:05.49 | lmadsen | kyron: www.it360.ca <-- April 7-9 |
| 00:06.10 | kyron | gerh...timing might be a wee bit off.. |
| 00:06.12 | ZaVoid | so the talk is on asterisk clustering? |
| 00:06.18 | lmadsen | my talk is, yes |
| 00:06.33 | ZaVoid | cool |
| 00:06.50 | ZaVoid | only clustering i worry about is the DB clustering.. i just let all the asterisk boxes pull from the DB cluster |
| 00:07.02 | kyron | lmadsen, so _you_ are one of the ones polluting my definition of Clustering! ...pfff (HPC :P) |
| 00:07.20 | lmadsen | kyron: quite :) what else do you want me to call it? :) |
| 00:08.22 | kyron | Uhhmmm...farming :P |
| 00:08.35 | plik | pharming |
| 00:09.03 | kyron | plik, h04k3r! Y0'r s0 1337 |
| 00:09.08 | kyron | :P |
| 00:09.11 | plik | he |
| 00:09.13 | plik | h |
| 00:09.26 | lmadsen | kyron: weird term... but I'll try and remember it :) |
| 00:09.45 | husimon | so I'm using the default call parking setup but can't figure out how to get it to announce what number it got parked on, what setting do I need to look at to change this? |
| 00:10.06 | lmadsen | although if I say I'm a consultant specializing in asterisk farming... I have a feeling I'm going to get some weird emails |
| 00:10.29 | ZaVoid | lol |
| 00:10.34 | kyron | lmadsen, get C0wagra Now! |
| 00:10.54 | kyron | lmadsen, load balancing |
| 00:11.02 | husimon | anyone? |
| 00:11.22 | kyron | lmadsen, well, that's the problem, it _is_ actually "clustering" even in the sense of distributed processing ;) |
| 00:11.42 | lmadsen | aye :) |
| 00:11.50 | lmadsen | plus it sounds so much cooler :) |
| 00:12.05 | ZaVoid | i just put a registration server in front of my asterisk boxes and load balance calls :) thats clustering too :) |
| 00:12.12 | drmessano | lmadsen, I wouldn't worry about it.. since when has the very specific use of industry terms ever been an issue for anyone? lol |
| 00:12.23 | lmadsen | oh I'm not too worried about it :) |
| 00:16.54 | husimon | anyone here using call parking? It says on the wiki that it should announce which extension the call got parked on but it's not doing anything. |
| 00:17.20 | plik | husimon: it just worked for me |
| 00:17.24 | ZaVoid | not me husimon sorry |
| 00:17.26 | husimon | hmm |
| 00:17.47 | hmmhesays | Zavoid who was that sms provider you use? |
| 00:18.03 | ZaVoid | its in pm |
| 00:18.28 | husimon | interesting it says it's playing the digits on the cli |
| 00:18.31 | husimon | maybe i'm doing it wrong |
| 00:18.36 | husimon | i'm doing a blind xfer to the call park numer |
| 00:19.00 | hmmhesays | thats right thanks ZaVoid |
| 00:19.07 | ZaVoid | np |
| 00:19.29 | kyron | lmadsen, yeahh....there _is_ a coolness factor...especially since HPC is becoming the hot cool topic :P |
| 00:19.34 | plik | husimon: you got K or k as an option at the end of the exten => line that you're using? |
| 00:19.45 | kyron | who's has * running off a PS3 here? |
| 00:19.54 | husimon | plik, it's just set to the default 700 |
| 00:19.58 | husimon | from features.conf |
| 00:20.19 | plik | yeah, bu tin extensions.conf... |
| 00:20.31 | plik | *but in |
| 00:20.44 | husimon | perhaps i'm missing something, the wiki said to just setup features.conf, then include the parkedcalls context |
| 00:21.05 | plik | soemthing like : exten => 333,1,Dial(SIP/333,20,TK) |
| 00:21.14 | plik | ----------------------------------------------------^ |
| 00:21.28 | husimon | i don't have a K in any of my extensions |
| 00:21.39 | plik | try that |
| 00:22.20 | husimon | well call parking does work |
| 00:22.39 | husimon | i'm not using the built-in transfer keys though i'm using a blindxfer button on the phone |
| 00:22.42 | husimon | that's probably why huh |
| 00:23.39 | plik | I use ## to transfer , features.conf is pretty much default I think, and likes as above in extensions.conf |
| 00:23.43 | plik | works for me |
| 00:23.56 | plik | s/likes/lines/ |
| 00:26.11 | [TK]D-Fender | husimon> i'm doing a blind xfer to the call park numer <- thats the problem |
| 00:26.20 | [TK]D-Fender | husimon, You should be doing an attended transfer |
| 00:26.47 | [TK]D-Fender | And you don't need a dial parameter for parking. |
| 00:27.02 | husimon | [TK]D-Fender, well I tried that too, |
| 00:27.09 | [TK]D-Fender | plik, and DTMF transfers = SUCK |
| 00:27.19 | husimon | [TK]D-Fender, i think part of the issue might be that i'm using the transfer button and blindxfer buttons on my phone |
| 00:27.26 | husimon | [TK]D-Fender, so i'm trying it without that |
| 00:27.39 | plik | [TK]D-Fender: yes, but ... |
| 00:27.58 | [TK]D-Fender | husimon, you include [parkedcalls] into your phones context and you simply transfer to 700 (or whatever you overrode it to) |
| 00:28.12 | plik | I guess the parameter must have been more outdated stuff on the wiki :/ |
| 00:28.25 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 00:28.46 | [TK]D-Fender | plik, More like "outright wrong" |
| 00:29.23 | plik | bah... so I can get rid of that then... what about the T & W for transfer & monitor? |
| 00:29.28 | plik | are they wrong too? |
| 00:29.57 | husimon | [TK]D-Fender, yeah did that. I'm just not hearing the digits |
| 00:30.28 | [TK]D-Fender | plik, no, not "wrong", just the wrong way to want to deal with transfers. "wW" for automon *IS* a legitamite use of features.conf |
| 00:30.45 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 00:30.46 | plik | ok thanks... |
| 00:30.48 | [TK]D-Fender | husimon, pastebin is your friend.... |
| 00:31.57 | husimon | [TK]D-Fender, k second let me get all the stuff and pastebin it |
| 00:32.28 | Putzz | what songoma card do you guys recommend for a single PRI? |
| 00:35.48 | lmadsen | Putzz: TE122 |
| 00:35.56 | lmadsen | which is actually a Digium card :) |
| 00:36.27 | husimon | [TK]D-Fender, http://pastebin.com/m56e572a, what other information do you need |
| 00:37.06 | [TK]D-Fender | Putzz, A101d |
| 00:37.33 | [TK]D-Fender | husimon, REAL extensions.conf, and comprehensive CLI output. |
| 00:38.17 | Putzz | thanks you sir |
| 00:38.55 | scooby2 | [TK]D-Fender: sorry i was just venting at the wonderful clueless tickets I get |
| 00:39.37 | scooby2 | now to figure out what this means: WARNING[3326] acl.c: Unable to lookup '' |
| 00:40.41 | lmadsen | scooby2: sounds like you have a deny= or permit= with no address in a file somewhere (probably sip.conf) |
| 00:40.48 | husimon | [TK]D-Fender, found the problem, i was using t instead of T which didn't let the blindxfer key to work. |
| 00:40.52 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-2f29ec8e0f4fa5c7) |
| 00:41.06 | x86 | what are ya makin? |
| 00:41.07 | husimon | [TK]D-Fender, apparently doing the xfer on the phone to the call park doesn't allow for the digits to be played back to the call |
| 00:41.28 | husimon | when I say xfer on the phone I mean using the gui menus on the phone instead of the xfer key # |
| 00:43.09 | scooby2 | lmadsen: thanks |
| 00:43.19 | jm|laptop | agi-test.agi: Failed to execute '/usr/local/share/asterisk/agi-bin/agi-test.agi': No such file or directory |
| 00:43.29 | jm|laptop | [root@voip /usr/local/share/asterisk/agi-bin]# ls -lha agi-test.agi |
| 00:43.30 | jm|laptop | -rwxr-xr-x 1 root wheel 1.7K Feb 4 21:56 agi-test.agi |
| 00:43.34 | jm|laptop | o.O |
| 00:44.46 | scooby2 | read the top line of it |
| 00:45.09 | jm|laptop | "No such file or directory" ? |
| 00:45.12 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 00:45.25 | scooby2 | whats the top line of agi-test.agi say? |
| 00:45.27 | [TK]D-Fender | husimon, What phone are you using? |
| 00:45.35 | scooby2 | head -1 /usr/local/share/asterisk/agi-bin/agi-test.agi |
| 00:46.00 | jm|laptop | [root@voip /tmp]# head -1 /usr/local/share/asterisk/agi-bin/agi-test.agi |
| 00:46.00 | jm|laptop | #!/usr/bin/perl |
| 00:46.00 | jm|laptop | [root@voip /tmp]# which perl |
| 00:46.00 | jm|laptop | [root@voip /tmp]# |
| 00:46.02 | jm|laptop | OMG |
| 00:46.05 | scooby2 | :) |
| 00:46.06 | mltlnx | hello, is there a way (without agi) to grab the channel names of a bridged call? |
| 00:46.06 | [TK]D-Fender | husimon, and the only reason not to hear the digits is becasue its a blind transfer. |
| 00:46.25 | jm|laptop | scooby2: thanks and sorry :( |
| 00:46.29 | scooby2 | np |
| 00:46.40 | husimon | [TK]D-Fender, actually it did playback the digits with the blindxfer |
| 00:46.52 | jm|laptop | :"> |
| 00:47.45 | [TK]D-Fender | husimon, its still BAD. ti will TRY, and FAIL |
| 00:48.04 | husimon | [TK]D-Fender, i heard it |
| 00:48.15 | [TK]D-Fender | husimon, If you do a blind transfer to parking, no, you won't |
| 00:48.21 | husimon | [TK]D-Fender, i just did it |
| 00:48.24 | husimon | blindxfer=# |
| 00:48.27 | husimon | i press #700 |
| 00:48.38 | husimon | it says "TRANSFER" "7" "0" "1" |
| 00:48.50 | [TK]D-Fender | husimon, Holy shit, why are you using DTMF transfers? |
| 00:49.16 | husimon | i'm not using K |
| 00:49.22 | husimon | i'm missing something here |
| 00:49.35 | [TK]D-Fender | husimon, Wrong answer, try again. |
| 00:50.31 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 00:51.54 | *** join/#asterisk chalco (n=chalco@about/networking/255.255.255.240/chalco) |
| 00:52.00 | husimon | ok so using # from features.conf is a DTMF transfer? |
| 00:52.05 | [TK]D-Fender | husimon, Yes |
| 00:52.15 | husimon | ok, so what is the other way to transfer? |
| 00:52.16 | chalco | hello |
| 00:52.16 | chalco | I currently have a trixbox server. If I wanted to consolidate to another server, I can just install asterisk and freepbx and have the same thing, right? I'm not using anything else trixbox provides |
| 00:52.31 | [TK]D-Fender | husimon, real transfer button on a decent phone. |
| 00:53.09 | husimon | [TK]D-Fender, that's the problem, I tried using the transfer buttons on my 7940 and in both cases (attended transfer and blind transfer) I hear no digits, yet on the CLI i see it is playing them. |
| 00:53.13 | [TK]D-Fender | chalco, Sort of answering your own question, aren't you? |
| 00:53.28 | chalco | [TK]D-Fender, just looking for validation :) |
| 00:53.32 | drmessano | That's an EXACT Paste of what he asked yesterday |
| 00:53.40 | husimon | chalco, you will need to figure out the differences between the gui's |
| 00:53.47 | chalco | drmessano, in another channel, where it was never answered |
| 00:53.48 | husimon | chalco, or write the dialplan yourself |
| 00:53.56 | [TK]D-Fender | chalco, even mentioning FreePBX in here is more likely to find you being lynched :) |
| 00:54.15 | drmessano | Personally, I love Trixbox |
| 00:54.18 | drmessano | It's so... green |
| 00:54.23 | mltlnx | Does FreePBX support SLA? Just kidding! |
| 00:54.52 | chalco | I'd stick with trixbox or a prebuilt thing, but I want to install it on another server I already have running |
| 00:55.06 | drmessano | Ahhh |
| 00:55.11 | [TK]D-Fender | chalco, Well if FreePBX is all you wanted, then by all means |
| 00:55.14 | fujin | <PROTECTED> |
| 00:55.22 | chalco | drmessano, perhaps you can answer my question next time, if you don't like copy/paste |
| 00:55.24 | drmessano | Nothing like throwing Asterisk and FreePBX on a webservermailservergameserver box |
| 00:55.28 | [TK]D-Fender | husimon, You are clearly doing something wrong... |
| 00:55.31 | drmessano | Ohhhhhhh |
| 00:55.31 | mltlnx | fujin: in what way? |
| 00:55.57 | drmessano | chalco, maybe you can stick around in a channel long enough for UNPAID VOLUNTEERS to answer you on THEIR time |
| 00:55.59 | drmessano | Ass |
| 00:56.03 | chalco | drmessano, we all have different needs |
| 00:56.11 | fujin | mltlnx: either with meetme or app_conference, I want to be able to create conferences on the fly and invite third parties into them |
| 00:56.11 | chalco | drmessano, I was there for a whole day |
| 00:56.16 | husimon | [TK]D-Fender, clearly, any idea where I should look to figure out what i'm doing wrong? this is a 7940 cisco phone. |
| 00:56.32 | drmessano | point? |
| 00:56.47 | [TK]D-Fender | husimon, I think you should actually show me the stuff I ask you for and answer my questions the first time... |
| 00:57.06 | chalco | drmessano, if I don't get an answer after a day, I'm likely to go elsewhere, aren't I? |
| 00:57.29 | [TK]D-Fender | drmessano, Ok cp'n.... you're goin' overboard... |
| 00:57.34 | [TK]D-Fender | cap'n |
| 00:57.57 | chalco | I help out in other channels, I know the score. |
| 00:57.58 | drmessano | Naah.. I said my piece.. :) |
| 00:58.10 | mltlnx | You can create conferences on the fly.....You can then transfer callers in to it. That said, you can use ChannelRedirect() to move caller into the conference room |
| 00:58.17 | kyron | FreePBX |
| 00:58.17 | [TK]D-Fender | drmessano, now rest in pieces :) |
| 00:58.20 | drmessano | lol |
| 00:58.38 | kyron | tehehehe |
| 00:58.59 | drmessano | [TK]D-Fender, I got an awesome "noob moment" for you... and it involved myself |
| 00:59.21 | kyron | drmessano, where you're the n00b? |
| 00:59.30 | drmessano | Set up a SPA-3102 at work.. Got it running, but not passing CID |
| 00:59.38 | fujin | mltlnx: how do you create meetme conferences on the fly? |
| 00:59.39 | drmessano | So, I banged my head a bit.. moved on |
| 00:59.52 | drmessano | Go back a few weeks later (today) and started working on it |
| 01:00.07 | husimon | [TK]D-Fender, here you go : http://pastebin.com/m39f11d60 |
| 01:00.11 | drmessano | Still not doing it.. I had the PSTN answer delay on 1 for some reason, but that wasn't it |
| 01:00.11 | jm|laptop | drmessano: have the same 'issue' |
| 01:00.19 | drmessano | So I am thinking.... thinking.... |
| 01:00.26 | jm|laptop | drmessano: for me: first time no CLID - ever time after CLID |
| 01:00.32 | drmessano | OH %$%$#$, I DONT HAVE CID ON THAT LINE |
| 01:00.38 | drmessano | No.. shit |
| 01:00.41 | jm|laptop | that doesn't help |
| 01:00.46 | mltlnx | You can use MeetMe with "d" option I believe |
| 01:01.15 | drmessano | I don't know why I thought I had it on there.. I manage the lines.. |
| 01:01.43 | husimon | [TK]D-Fender, by the way feel free to yell at me and tell me what is wrong with that extensions.conf so I can rework it. |
| 01:01.48 | drmessano | Typical engineer blinders.. Focus on one problem, ignore some glaring detail |
| 01:01.56 | mltlnx | fujin: give me a scenario with on the fly conferencing/ |
| 01:01.57 | drmessano | <---- n00b |
| 01:02.00 | kyron | drmessano, this is interesting, I'm about to maybe acquire one of those... and don't have CID on my line... |
| 01:02.02 | husimon | [TK]D-Fender, if you feel it is written in a bad way. |
| 01:02.36 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 01:02.48 | drmessano | I know the CID works on the 3102.. and I knew how to get it working.. I had it working at another location.. |
| 01:02.57 | drmessano | But damn.. I laughed for 5 minutes |
| 01:03.24 | kyron | drmessano, am-I to understand the SPA-3102 b0rks if it's configured for getting CID and the line doesn't give one? |
| 01:03.44 | drmessano | No, it doesn't bork.. but it doesnt pass it if it's not there ;) |
| 01:04.21 | kyron | re-read your initial "problem".... HAHAHAHAHAHAHAH |
| 01:04.30 | [TK]D-Fender | husimon, you still need to learn to make better use of macros and for your your phone internal ext's I'd put the comment at the ent of the line. and kill the blank lines between. |
| 01:04.51 | chalco | thank you, [TK]D-Fender, husimon. |
| 01:05.16 | mltlnx | fujin? |
| 01:05.17 | xp_prg | anyone use perl to interact with asterisk here? |
| 01:05.30 | [TK]D-Fender | drmessano, Wheneveer things go wrong, reach for the lowest level sanity test you can perform and work your way up. |
| 01:05.30 | husimon | [TK]D-Fender, I agree the about the blank lines. |
| 01:05.36 | *** part/#asterisk chalco (n=chalco@about/networking/255.255.255.240/chalco) |
| 01:05.51 | husimon | [TK]D-Fender, could you elaborate on the macros part? |
| 01:06.05 | real0ne | when i want to install zaptel |
| 01:06.10 | real0ne | i have this message |
| 01:06.12 | drmessano | Yep.. I felt so dumb.. Failed "Basic Troubleshooting 101".... |
| 01:06.26 | real0ne | No functioning zap hardware found in /proc/zaptel, loading ztdummy |
| 01:06.26 | real0ne | Running ztcfg: done. |
| 01:06.38 | [TK]D-Fender | husimon, your outbound dialing has a lot of repitiion. Sure you made ringing your phones better, but your OUTBOUND stuff is still very redundant |
| 01:06.58 | husimon | [TK]D-Fender, yeah I realized I could probably shorten that section to about 2 lines |
| 01:07.02 | husimon | one with a 9 and anything |
| 01:07.06 | husimon | and one without a 9 and anything |
| 01:07.23 | husimon | is that what you mean? |
| 01:07.35 | [TK]D-Fender | husimon, something like that. |
| 01:07.39 | husimon | in fact i guess screw it and ignore the 9 via ignorepat |
| 01:07.55 | [TK]D-Fender | husimon : you are using Zaptel FXS? |
| 01:08.40 | mltlnx | realone: and... |
| 01:09.22 | real0ne | i don't know mltlnx |
| 01:09.25 | husimon | [TK]D-Fender, do you mean what hardware am I using to access the pri? |
| 01:09.36 | real0ne | mltlnx is that not a problem? |
| 01:09.44 | [TK]D-Fender | husimon, no, ignorpat is only for zaptel FXS channels. |
| 01:09.49 | husimon | ohh |
| 01:10.07 | [TK]D-Fender | husimon, You are running on a small mountain of misconceptions about 8.... |
| 01:10.11 | [TK]D-Fender | * even |
| 01:10.19 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 01:10.28 | kyron | drmessano, heh, I have some wost ones... |
| 01:10.41 | mltlnx | no its not a problem, music on hold and conferencing need a zaptel timing device. If you do not have a zaptel device installed, then you can use ztdummy. |
| 01:10.41 | kyron | well..actually...comparable |
| 01:10.49 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:11.15 | drmessano | I'm trying to think of what my worst fail was |
| 01:11.34 | husimon | drmessano, imagine working on a server trying to figure out why all your applications are spitting out odd failure messages |
| 01:11.40 | real0ne | aha |
| 01:11.40 | real0ne | ok |
| 01:11.46 | husimon | then three days later finding out the disk space had ran out. |
| 01:11.49 | real0ne | i don't have anydevice |
| 01:11.51 | drmessano | HA |
| 01:12.05 | husimon | drmessano, most applications don't really display a useful error when that happens |
| 01:12.10 | Daviey | husimon: heheh |
| 01:12.10 | mltlnx | yeah, that is why ztdummy is getting loaded. |
| 01:12.13 | husimon | drmessano, they just go haywire |
| 01:12.17 | drmessano | I didn't go three days, but I ignored a disk space error for a few hours once |
| 01:12.26 | drmessano | yep |
| 01:12.53 | real0ne | i should install ztdummy |
| 01:12.54 | real0ne | ?? |
| 01:12.59 | husimon | The problem I had was there was no message that said there is no disk space |
| 01:13.05 | Daviey | I moved /var/mysql and /tmp to NFS once to circumvent low disk space, how nasty is that.. |
| 01:13.08 | husimon | just weird errors that didn't mean anything |
| 01:13.54 | drmessano | I tried once to combine my business and automation network, to keep us from using 2 drops and 2 NICs in machines that needed access to both |
| 01:14.06 | drmessano | They were 10M/b hubs.... |
| 01:14.15 | mltlnx | real0ne it sounds like it is loaded. What distro of Linux are you using? |
| 01:14.31 | real0ne | mltlnx debian |
| 01:15.07 | mltlnx | try lsmod | grep ztdummy |
| 01:15.13 | drmessano | That was 10 years ago... My first and last major FAIL with networking.. I decided after that I needed to "learn" it |
| 01:15.16 | mltlnx | you should see ztdummy is loaded |
| 01:15.45 | drmessano | Nothing like the sound of 4 Radio Stations going off the air.. |
| 01:16.34 | kyron | husimon, after a few times, you get used to type `df` |
| 01:16.42 | husimon | [TK]D-Fender, so based on those config files and cli output do you see any reason why I shouldn't hear the digits from the call parking? Or is it probably just a phone related issue. |
| 01:16.46 | husimon | kyron, yeah i do |
| 01:16.56 | husimon | now.... |
| 01:17.14 | kyron | drmessano, OUCH...you went big |
| 01:17.25 | real0ne | mltlnx wait plz |
| 01:17.32 | kyron | Like I always tell the students, you only learn from mistakes...so make em! |
| 01:18.16 | husimon | drmessano, lots of collisions eh. |
| 01:18.34 | kyron | bbl |
| 01:18.43 | drmessano | It took less than 45 seconds |
| 01:19.22 | drmessano | I went down the hall to check a machine.. and I didn't get inside their office doorway before the 120db alarms went off |
| 01:19.23 | husimon | [TK]D-Fender, btw yeah I mixed up the terms FXS and FXO a minute ago when you asked about zaptel FXS. |
| 01:19.42 | husimon | drmessano, what were the alarms for? |
| 01:20.07 | drmessano | The 4 radio stations I knocked off the air |
| 01:20.39 | husimon | oh |
| 01:20.40 | husimon | lol |
| 01:20.50 | drmessano | Two months on the job.. Figured "Eh, networking is easy" |
| 01:21.02 | drmessano | I was green as hell, cocky to go with it |
| 01:22.21 | [hC] | so, was there a big change in dtmf (primarily rfc2833 via SIP) in asterisk 1.4? |
| 01:23.20 | husimon | laugh yeah ignorepat=>9 just keeps the dial tone going, it doesn't remove it from the dial string.. |
| 01:23.21 | husimon | stupid me |
| 01:30.49 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:35.21 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:35.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:35.59 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 01:37.47 | *** join/#asterisk BeeBuu (n=beebuu@219.130.244.52) |
| 01:39.56 | husimon | is there any way to change the callerid of a transfered call? I want the callerid to be the person calling, not the phone it was forwarded from. |
| 01:40.35 | drmessano | Invalid access to mod_kfc |
| 01:40.38 | fujin | using featres.conf transferring or your phones transfer? |
| 01:40.38 | drmessano | COLONEL PANIC! |
| 01:41.15 | husimon | either i guess |
| 01:41.36 | husimon | i really meant to ask about forwarding not transferring, but both apply |
| 01:42.24 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 01:42.34 | fujin | husimon: well if you're using phones, I don't think there's a way to do it |
| 01:42.46 | fujin | apart from set callerid on your sip.conf stuff and don't override it in the dialplan at all |
| 01:43.07 | fujin | but if you're using asterisks transfers, you should be able to do it |
| 01:43.11 | husimon | fujin, if i'm doing a call forward via the dialplan, then what? |
| 01:43.26 | fujin | well, before the transfer (when the call comes in |
| 01:43.37 | fujin | Set(__CALLER=${CALLERID(all)}) |
| 01:43.47 | fujin | and then before the transfer, set callerid to ${CALLER} |
| 01:44.11 | lunaphyte | i've got some terminology questions - in sip.conf or iax.conf, are the object each entry defines called channels? |
| 01:44.18 | husimon | k trying it |
| 01:45.16 | husimon | fujin with SetCallerID(${CALLER}) right? |
| 01:45.27 | fujin | no, Set(CALLERID(all)=${CALLER}) |
| 01:45.29 | [hC] | bkruse: hey, you here still? |
| 01:45.37 | husimon | ah i must be looking at old syntax |
| 01:45.41 | fujin | yes, that's 1.2 |
| 01:45.43 | fujin | deprecated now |
| 01:45.49 | husimon | yep i see the new one now |
| 01:47.21 | kyron | drmessano, LOL |
| 01:47.58 | kyron | Q: anyone here used the IPP implementation of 729? |
| 01:48.17 | ZaVoid | ,Set(CCARD=${CALLERID(number)}) |
| 01:48.26 | ZaVoid | you mean the wierd license file one kyron ? |
| 01:49.14 | kyron | yeah, prolly the "this is how we implemented 729 but you shouldn't use it" ... |
| 01:49.36 | kyron | but I only wat to test and "play around" with my friend and out iax2 trunk ;) |
| 01:49.36 | ZaVoid | just but a real license file |
| 01:49.39 | ZaVoid | you'l be happy |
| 01:49.43 | ZaVoid | or get a device with g729 |
| 01:49.57 | kyron | my mediatrix and polycom both support 729 |
| 01:50.49 | ZaVoid | then you don't need a license |
| 01:50.51 | mihinomenest | ...or use 711u for your testing... |
| 01:50.59 | kyron | the way I understand the licensing is that I need 1 license per audio stream that needs to be converted from/to 729 |
| 01:51.15 | kyron | mihinomenest, the point is to test all other than 711u |
| 01:51.23 | kyron | bandwidth hog |
| 01:51.48 | mihinomenest | exactly, things break quicker when 711u is around. |
| 01:52.03 | ZaVoid | do you need to convert kyron? |
| 01:52.11 | ZaVoid | if both legs are g729 your fine |
| 01:52.13 | kyron | nohup_, I'm an agnostic |
| 01:52.31 | kyron | lol...I hit tab after no O_o |
| 01:53.02 | ZaVoid | no both legs are not g729? |
| 01:53.10 | kyron | ZaVoid, uhm...don't I need g729 for the following: 729 -- * -- iax2 -- * -- 729 ? |
| 01:53.22 | ZaVoid | no |
| 01:53.50 | jblack | [TK]D-Fender: Ping |
| 01:54.28 | ZaVoid | kyron: you would need it if your transcoding form say g711 to g729 |
| 01:55.11 | jblack | Anyone that's been following my mrdigital saga... http://pastebin.com/m34837a71 |
| 01:55.31 | drmessano | jblack: You know where I can get a good van? |
| 01:56.11 | jblack | Depends. What sort of price market are you in? |
| 01:57.41 | drmessano | $195,000 <>$205,000 |
| 01:57.55 | husimon | question: does anyone know if it is possible to get cisco 7940 phones to let you start typing phone numbers in without taking the phone off the hook? The old sccp firmware let you type in a number then press the speaker phone or pickup the phone and it would dial. |
| 01:58.02 | husimon | they are now on sip firmware |
| 01:58.03 | jblack | Sure. I can hook you up. I suppose you'll want a spare, in case the main one breaks? |
| 01:58.11 | jblack | Think of it as raid for wheels. |
| 01:58.16 | drmessano | HA |
| 01:58.17 | drmessano | Yes |
| 01:58.34 | drmessano | I wonder if I can get them with cash registers installed |
| 01:58.37 | drmessano | and fry baskets |
| 01:59.36 | jblack | heh. He got fired for making shitty fries. :) |
| 02:00.10 | drmessano | I call bullshit.. people get PROMOTED for making shitty fries |
| 02:00.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:02.05 | jblack | http://pastebin.com/m2b3d85bc |
| 02:03.03 | drmessano | HA |
| 02:03.04 | kyron | ZaVoid, oops, yeah, the details: 729 -- * -- iax2(711, since * doesn't support 729 ) -- * -- 729 |
| 02:03.29 | ZaVoid | huh? |
| 02:03.32 | ZaVoid | its passthrough its fine |
| 02:03.36 | ZaVoid | should be a tleast |
| 02:04.45 | kyron | so you only need the license if you need * to be the termination of the call (ie: conference, voicemail, any decoding...) |
| 02:06.02 | kyron | husimon, my cordless with an LCD does that... if CISCO can't do it with high-tech equipment like VoIP phones... |
| 02:06.59 | *** join/#asterisk joez212 (n=jhart@CPE001c101b40b5-CM0018c0d91624.cpe.net.cable.rogers.com) |
| 02:07.01 | joez212 | hey guys |
| 02:07.11 | joez212 | still having issues with setting up a sip softphone :( |
| 02:07.42 | joez212 | do you need to setup the ip address even if its alreaedy set via a dhcp server from the router? |
| 02:07.54 | husimon | kyron, i dunno they just changed their ui |
| 02:08.08 | husimon | btw I have an odd problem with one 7940 phone in my network. |
| 02:08.18 | husimon | it can place outbound calls, but not receive inbound |
| 02:08.20 | husimon | http://pastebin.com/m7a8eaa60 |
| 02:08.43 | husimon | It says it can't find a route to the sip user, but obviously it's there in sip peers |
| 02:09.27 | jblack | lol. http://pastebin.com/m699fc743 |
| 02:12.07 | husimon | what a moron |
| 02:12.50 | husimon | that's just about as good as "I'm a tool, ok cisco ream my ass for $100k and give me voip" |
| 02:13.19 | mihinomenest | that works really well for a lot of people btw. |
| 02:13.32 | mihinomenest | husimon: can I see your entire sip/extenions.conf? |
| 02:13.37 | husimon | mihinomenest, sure |
| 02:15.11 | [hC] | i can ping devices on this vlan ive assigned it to, but I cannot actually transmit data. |
| 02:15.15 | [hC] | how fun! |
| 02:15.37 | mihinomenest | hooray vlan! |
| 02:15.46 | *** join/#asterisk UnixDog (n=unixdog@ppp-71-129-91-93.dsl.irvnca.pacbell.net) |
| 02:16.56 | husimon | mihinomenest, http://pastebin.com/d4276427f |
| 02:17.42 | mihinomenest | which one's the phone that you're having problems with? 300 something? |
| 02:18.40 | joez212 | this is impossible |
| 02:18.41 | joez212 | damn |
| 02:19.53 | husimon | 300_2 |
| 02:20.09 | husimon | i have the DID 300 on two phones as two sip users, 300_1 and 300_2 |
| 02:20.10 | jblack | drmessano: http://pastebin.com/m541dc35c |
| 02:20.12 | husimon | 300_1 works fine |
| 02:21.58 | husimon | jblack how much you wanna be he's 14 |
| 02:21.58 | mihinomenest | husimon: ever think about using something like "exten => _37[0-9]" ? |
| 02:22.39 | mihinomenest | fwiw, I think there's something funny with the hardware that you've got. |
| 02:22.49 | husimon | do you mean the phone? |
| 02:23.05 | drmessano | ROFL |
| 02:23.17 | husimon | drmessano, i got bets on 14 years old. |
| 02:23.24 | mihinomenest | husimon: yes. |
| 02:23.33 | husimon | mihinomenest, i think i'll try and factory reset it |
| 02:23.40 | mihinomenest | 14 year olds can't get married or own homes. |
| 02:23.41 | husimon | mihinomenest, sick of banging my head against the wall about it. |
| 02:23.48 | husimon | mihinomenest, no shit but they can lie about it :P |
| 02:23.49 | mihinomenest | yeah. |
| 02:23.54 | mihinomenest | or replace it. |
| 02:24.12 | x86 | mihinomenest: 14 year olds can too be married |
| 02:24.16 | mihinomenest | true, and that doesn't even touch on the idea that he could be 35 with the mental maturity of a 14 year old. |
| 02:24.19 | x86 | if their parents sign consent waivers |
| 02:24.22 | husimon | 14 year olds need love too |
| 02:24.26 | mihinomenest | x86: only in arkansas. |
| 02:24.28 | husimon | especially girls..... |
| 02:24.30 | husimon | laugh |
| 02:24.31 | x86 | in any state |
| 02:24.37 | husimon | 15 in hawaii |
| 02:24.43 | x86 | any state |
| 02:24.45 | mihinomenest | (or the circus) |
| 02:24.55 | x86 | 15 is the legal age of consent for a minor to have consentual sex |
| 02:24.56 | mihinomenest | any state...with the consent of the parents. |
| 02:25.07 | husimon | mihinomenest, that _37[0-9] does what? |
| 02:25.15 | x86 | which is different from emancipation / minor marriage |
| 02:25.19 | Nugget | XP was released 10/2001 |
| 02:25.19 | Nugget | SP1 was released 9/2002 |
| 02:25.19 | Nugget | SP2 was released 8/2004 |
| 02:25.21 | Nugget | erp |
| 02:25.31 | Nugget | http://ageofconsent.com/ is what I meant to paste. |
| 02:25.51 | mihinomenest | husimon: in your extensions.conf, instead of writing a line for every extension, you can write one line that describes many extensions. |
| 02:25.53 | husimon | just in case you want to look up on the fly before you do her? |
| 02:25.55 | ZaVoid | anyone else see new pgsql released today? |
| 02:25.56 | drmessano | What does age of consent have to do with XP? |
| 02:25.59 | Nugget | http://ageofconsent.com/ageofconsent.htm in particular. |
| 02:26.11 | Nugget | nothing. the XP shit is just what was previously in my clipboard. |
| 02:26.16 | drmessano | LOL |
| 02:26.22 | mihinomenest | _37[0-9] would replace 370,371,372,373,etc. |
| 02:26.23 | jblack | drmessano: http://pastebin.com/m5a238677 |
| 02:26.26 | drmessano | Dude, gave you a chance for a punchline.. you blew it |
| 02:26.26 | Nugget | I must have fat-fingered command-c |
| 02:26.29 | mihinomenest | 1 line, ten extensions. |
| 02:26.32 | husimon | why the hell is the male and female age separate |
| 02:26.51 | lunaphyte | in sip.conf or iax.conf, are the objects each entry defines called channels? |
| 02:26.53 | husimon | mihinomenest, yeah but that removes the ability to customize dial flags |
| 02:27.01 | mihinomenest | sure. |
| 02:27.04 | drmessano | jblack: Im already up to that |
| 02:27.12 | jblack | You're up to 21:25? |
| 02:27.28 | mihinomenest | I just figured I'd suggest it. |
| 02:27.38 | jblack | I must be hitting the max for pastebin |
| 02:28.22 | russellb | jblack: i have only caught pieces of this story, but it's still hilarious |
| 02:28.34 | husimon | mihinomenest, yeah thanks |
| 02:28.34 | lunaphyte | can anyone enter text in the area code field on this page? http://www22.verizon.com/CallingAreas/RegionalTollMapLocator/Default.htm |
| 02:28.39 | jblack | Is there a way in irssi for me to save the contents of a chat/ |
| 02:28.55 | jblack | Not log from here forward, but to log what's already transpired? |
| 02:29.15 | x86 | jblack: hmm... dont think so |
| 02:29.21 | jblack | bother. |
| 02:29.24 | x86 | jblack: copy+paste+vi? |
| 02:29.31 | mihinomenest | you could build a time machine. |
| 02:30.01 | husimon | jblack is there more? |
| 02:30.12 | husimon | jblack less talky, more pasty! |
| 02:30.53 | ZaVoid | jblack: what version is irssi at now? |
| 02:30.54 | *** join/#asterisk angryuser (i=nononon@df01t2-212-195-107-139.d4.club-internet.fr) |
| 02:31.00 | ZaVoid | think last time i used that was like .081 or somthing |
| 02:31.23 | tzanger | whoo sittin in the aeroport :-) |
| 02:31.50 | jblack | Ok, Let me put this somewhere good. |
| 02:31.55 | jblack | I copied the whole thing into a text file |
| 02:32.44 | jblack | http://james.blackwell.cc/~jblack/MrDigital2 |
| 02:32.57 | jblack | The final chapter of the Mr Digital saga. |
| 02:33.31 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 02:33.44 | joez212 | i get an error when i start up asterisk |
| 02:34.11 | joez212 | chan_mcgp unable to load |
| 02:34.14 | joez212 | ip address nto found |
| 02:34.18 | joez212 | mcgp disabled |
| 02:34.27 | joez212 | could this explain why i am having troubles connecting? lol |
| 02:34.51 | plik | jblack: /LASTLOG -file ~/irc.log should do it |
| 02:35.06 | jblack | plik for one window? |
| 02:35.06 | plik | ZaVoid: 0.8.12 now :) |
| 02:35.24 | jblack | ahh /lastlog -window 5 ..... |
| 02:35.28 | jblack | plik: Thanks! |
| 02:35.30 | plik | as far as I know, for the window you're in... |
| 02:35.31 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:35.35 | plik | not tried it though |
| 02:35.43 | ZaVoid | maybe it was .61 |
| 02:35.51 | ZaVoid | it was like 6 years ago i used |
| 02:35.51 | ZaVoid | it |
| 02:36.14 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177611974.dsl.bell.ca) |
| 02:36.38 | jblack | Only three of you are going to read the conclusion of the MrDigital saga? You're missing out, because the opening chapters were a doozy. :) |
| 02:36.50 | plik | jblack: http://irssi.org/documentation/startup#c7 |
| 02:36.58 | jblack | i wish i had kept the first half of this story. |
| 02:37.54 | jblack | plik: Thanks. Wish I had known that 3 weeks ago. |
| 02:38.12 | joez212 | pbx-dundil also have an error about the lookup of my computer name |
| 02:38.58 | husimon | jblack reading now |
| 02:39.10 | husimon | jblack you could summarize for s |
| 02:39.11 | husimon | us |
| 02:39.21 | ZaVoid | lol |
| 02:39.26 | ZaVoid | i miss bitchx |
| 02:40.06 | plik | ZaVoid: Irssi 0.8 docs were written in 2000, so your were likely right the first time :) |
| 02:40.28 | ZaVoid | possible :) |
| 02:40.30 | husimon | jblack, what what a retard |
| 02:40.59 | ZaVoid | i just remember its windows were really annoying and bitchx and screen were much easier |
| 02:42.16 | husimon | hey jblack what server is that on? |
| 02:42.21 | husimon | i'll go offer to loan him money |
| 02:42.29 | jblack | He's here on freenode. |
| 02:42.50 | husimon | he log off? |
| 02:43.13 | drmessano | Im back |
| 02:43.13 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 02:43.14 | husimon | oh i misspelled |
| 02:43.26 | drmessano | Workin on my Resume' |
| 02:43.37 | drmessano | * Cooks Fry's |
| 02:43.50 | drmessano | * Under-court Cat5 installation |
| 02:44.13 | joez212 | would anyone be able to assist me? |
| 02:44.34 | joez212 | i just want to test out asterisk with a centos box |
| 02:44.41 | ZaVoid | i use centos! |
| 02:44.42 | joez212 | using x-lite software on my PC |
| 02:44.42 | ZaVoid | works great |
| 02:44.44 | lunaphyte | go for it. |
| 02:44.57 | joez212 | i was able to install and compile everything |
| 02:44.57 | husimon | joez212, do you need mgcp, if not just remove mgcp from your modules.conf |
| 02:45.17 | husimon | joez212, otherwise I think the issue is that the hostname of your box doesn't line up with you ip address and asterisk is complaining |
| 02:45.18 | joez212 | i have no idea if i just need mgcp for sip operations? |
| 02:45.20 | husimon | i had that same problem |
| 02:45.24 | husimon | joez212, you don't |
| 02:45.35 | joez212 | ok lemme take it out |
| 02:46.01 | husimon | add a "noload => chan_mgcp.so" in your modules.conf |
| 02:46.33 | mihinomenest | jblack: my home eq loan is killing me too. |
| 02:48.07 | joez212 | ok i stopped * gracefully |
| 02:48.10 | joez212 | how should i reload it? |
| 02:48.41 | mihinomenest | with the rc script? |
| 02:49.11 | joez212 | i actually set the ip of my * asterisk in my sip.conf file |
| 02:49.34 | joez212 | i am using the example directly from this website |
| 02:49.35 | joez212 | http://asteriskvoip.blogspot.com/2006/02/help-article-configuring-x-lite-for.html |
| 02:49.54 | joez212 | i replaced it with the dhcp ip address currently set by my router |
| 02:50.01 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 02:50.53 | joez212 | ok i just reloaded * with -vvvvvc |
| 02:50.55 | kyron | so which is the coolest Codec to use in an IAX trunk? |
| 02:50.57 | kyron | :P |
| 02:51.27 | joez212 | dundi.conf still shows an error with the host name |
| 02:51.35 | joez212 | but says its listening 0.0.0.0 ?? |
| 02:51.45 | ZaVoid | anything wrapped with GIPS KYRON |
| 02:51.59 | kyron | GIPS? |
| 02:52.33 | ZaVoid | http://www.gipscorp.com/default/overview.html |
| 02:52.41 | ZaVoid | its what all the cool kids are using these days :) |
| 02:52.49 | ZaVoid | yahoo, google, me :) |
| 02:52.59 | ZaVoid | those 2 are a bit cooler though i guess |
| 02:54.15 | joez212 | bah its taking too long to register |
| 02:54.18 | joez212 | :( |
| 02:54.32 | kyron | ZaVoid, LOL |
| 02:54.55 | joez212 | 408 again |
| 02:54.56 | joez212 | damn |
| 02:55.09 | joez212 | is there some other simple simple test to see if * is functioning? |
| 02:55.31 | drmessano | GIPS is so "last week", I am using GROPS now |
| 02:55.36 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177611974.dsl.bell.ca) |
| 02:56.03 | joez212 | i have a router |
| 02:56.09 | joez212 | do i need to add the iptables entries? |
| 02:56.16 | joez212 | the orielly book mentions them |
| 02:56.27 | joez212 | some parts of the book are not accurate |
| 02:57.12 | jblack | oh geesh. my check card caught on fire today. |
| 02:57.33 | joez212 | still won't register |
| 02:57.39 | joez212 | what could i be doing wrong? |
| 02:57.40 | husimon | my work cc got hacked off a website and used for wow registrations today |
| 02:57.43 | husimon | fun stuff |
| 02:57.49 | kyron | okok...let me restate that: whilst staying in the opensource realm, which codec is c00l to use for IAX, especially since IAX is probable the only one to support it for the moment (speex?) |
| 02:58.20 | mihinomenest | skinny. |
| 02:58.24 | mihinomenest | use skinny. |
| 02:58.31 | husimon | oh god please don't |
| 02:58.36 | kyron | CentOS 5 sucks, can't install icc 10.1.008 on it...pffff |
| 02:58.38 | joez212 | hmm |
| 02:58.49 | joez212 | i'm using 4.4 |
| 02:58.59 | kyron | O_o... skinny |
| 02:59.18 | joez212 | for some reason 0.0.0.0 doesnt work |
| 02:59.19 | joez212 | lol |
| 02:59.43 | joez212 | lol |
| 02:59.47 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 02:59.52 | joez212 | i get service unavailable when i use that address |
| 03:00.05 | joez212 | ipchains issue? |
| 03:00.22 | kyron | Polycom 601 any good? |
| 03:00.22 | mihinomenest | aren't IPChains/IPTables redundant? |
| 03:00.59 | lunaphyte | iptables supersedes ipchains. |
| 03:01.09 | *** join/#asterisk SomethingISOdd (n=TestMast@S010600a0d1757bfb.cg.shawcable.net) |
| 03:01.13 | joez212 | mihi, that's why i'm asking |
| 03:01.21 | joez212 | i just want to test out * |
| 03:01.25 | SomethingISOdd | hello all anyone here use chan_h323, as i am compiling i keep getting make[2]: *** [/root/openh323/lib/libh323_linux_x86_d.so.1.18.0] Error 1 |
| 03:01.27 | joez212 | it seems quite difficult to setup |
| 03:01.30 | SomethingISOdd | anyideas how to fix this pleas? |
| 03:01.45 | joez212 | SomethingISOdd: which flavour of linux are you running? |
| 03:01.50 | SomethingISOdd | Centos |
| 03:02.18 | joez212 | which one? |
| 03:02.26 | SomethingISOdd | Centos 4.6 |
| 03:02.39 | joez212 | 2.6 kernel? |
| 03:02.51 | SomethingISOdd | yes 2.6.9 |
| 03:03.09 | joez212 | is that when you do the compile on asterisk? |
| 03:03.25 | SomethingISOdd | thats when i do make on the h323 |
| 03:03.31 | SomethingISOdd | under channels. |
| 03:03.47 | joez212 | i believe that comes with * |
| 03:03.53 | SomethingISOdd | yes it does. |
| 03:03.54 | husimon | joez212, it's not a walk in the park, but if you read the book and play around for a day or so you can get it. |
| 03:04.22 | joez212 | husimon: i've been playing for 2 days, a bit disappointed that its not easy after 2 days? |
| 03:04.32 | mihinomenest | I got it in two and I'm a windows admin! |
| 03:04.34 | joe | SomethingISOdd: atrpm has packages iirc |
| 03:04.42 | husimon | joez212, I started 2 weeks ago and I still don't know what the hell i'm doing.... |
| 03:04.46 | joez212 | lol |
| 03:04.48 | SomethingISOdd | joe oh let me check |
| 03:05.46 | joez212 | something: the manuel i use said to install asterisk last, you install the other two firs |
| 03:05.57 | joez212 | i think its matters |
| 03:06.28 | SomethingISOdd | ya i did |
| 03:06.30 | joez212 | husimon: also many people in here are experts |
| 03:06.37 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 03:06.43 | joez212 | i am no expert |
| 03:07.02 | joez212 | if i put host=dynamic |
| 03:07.11 | joez212 | it should put the dhcp ip address from eth0 right? |
| 03:07.32 | [hC] | Qwell: Does skinny.conf not know about dtmfmode= anymore? |
| 03:07.44 | SomethingISOdd | host=dynamic under the user account will allow that phone to login from anywere as long as they are using a username/password |
| 03:07.51 | SomethingISOdd | that is assigned in the config file |
| 03:08.36 | joez212 | but i understand why my x-lite softphone refuses to register |
| 03:08.56 | *** join/#asterisk mbt (n=mbt@c-76-17-47-152.hsd1.ga.comcast.net) |
| 03:09.02 | joez212 | i'm sure the 5-6 sip.conf files have been fine |
| 03:09.12 | joez212 | i am using this one at the moment with no luck |
| 03:09.24 | joez212 | i dont think x-lite can see the * server |
| 03:09.42 | joez212 | http://asteriskvoip.blogspot.com/2006/02/help-article-configuring-x-lite-for.html |
| 03:09.50 | SomethingISOdd | are you sure you are confiruing x-lite correctly? |
| 03:10.08 | joez212 | something i would say about 90% certain |
| 03:10.13 | joez212 | but this version of x-lite is new |
| 03:10.19 | joez212 | and all screenshots utilize an old verison |
| 03:10.21 | joez212 | but still |
| 03:11.00 | SomethingISOdd | i havent used xlite in years. have you ever had anything registered to your * via sip |
| 03:11.19 | joez212 | something: no |
| 03:11.24 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 03:11.28 | SomethingISOdd | can you show me your bindaddr? |
| 03:11.43 | joez212 | its 192.168.1.109 |
| 03:11.58 | joez212 | sorry its .100 |
| 03:12.04 | mbt | Does anyone know if there are directories of PRI services somewhere? I am trying to figure out if I can find less expensive PRI services than AT&T in my area |
| 03:12.20 | SomethingISOdd | mbt message me.. |
| 03:12.24 | joez212 | mbt: no idea |
| 03:12.42 | husimon | so what are the best cheap ATAs? linksys SPA what? |
| 03:13.05 | joez212 | 3201 is what i have been told to get |
| 03:13.19 | mihinomenest | grandstream 386 is probably the "cheapest" |
| 03:13.45 | joez212 | i have a computer store that carries mine and its a 25 minute bus ride from my apartment |
| 03:13.51 | joez212 | the linksys atas |
| 03:14.20 | husimon | well i'd like something that is hassle free |
| 03:14.34 | husimon | anything under $100 is fine |
| 03:14.46 | husimon | as long as it works well and isn't a piece of crap |
| 03:14.55 | joez212 | the 3201 is under 80 dollars |
| 03:15.01 | *** join/#asterisk h3x (i=Hex@64.192.116.17) |
| 03:15.04 | joez212 | searching for the grandstream |
| 03:15.08 | h3x | i knew it |
| 03:15.10 | husimon | 63 |
| 03:15.14 | husimon | $63 for 3201 |
| 03:15.24 | h3x | that problem i was having with module-assistant in ubuntu server building zaptel was just ubuntu server edition |
| 03:15.35 | h3x | tried it on desktop and it works fine, which makes me believe theres something wrong with dependancies |
| 03:15.50 | joez212 | wow the grandstream is only 49 dollars |
| 03:16.03 | h3x | wow the grandstream sounds like a tin can |
| 03:16.03 | drmessano | Whats a 3201? |
| 03:16.55 | husimon | i think he means http://www.voipdw.com/Linksys-SPA3102-p/spa3102-na-vdw.htm?gclid=CMv7qN2KrJECFSZbiAodLidQYQ |
| 03:16.58 | husimon | 3102 |
| 03:17.09 | joez212 | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1146582257191&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 03:17.10 | husimon | which is what I meant too |
| 03:17.17 | joez212 | sorry guys lol |
| 03:17.24 | drmessano | For $63? |
| 03:17.25 | drmessano | Where |
| 03:17.31 | husimon | from the link i pasted |
| 03:17.41 | lunaphyte | yeah, i just bought one from them. |
| 03:17.43 | husimon | so what's the deal with asterisk and fax |
| 03:17.49 | lunaphyte | drmessano: the one you were helping me with. |
| 03:17.49 | husimon | does it work with any atas? |
| 03:17.52 | joez212 | h3x: have you tried linksys atas? |
| 03:18.04 | drmessano | Thats awesome |
| 03:18.10 | drmessano | Awesome price |
| 03:18.32 | drmessano | I paid $80 for mine lol |
| 03:18.46 | joez212 | my local ma/pop shop is higher cuz not many stores carry such exotic pieces of hardware |
| 03:18.51 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:19.04 | husimon | anyone have experience with the best way to do fax with asterisk? |
| 03:19.17 | joez212 | husimon: i think i read somewhere |
| 03:19.24 | joez212 | "dont do it" |
| 03:19.34 | drmessano | Yeah |
| 03:20.01 | husimon | that's kind of sad, I want it to work |
| 03:20.37 | *** join/#asterisk rcslex (n=lance@74-131-227-39.dhcp.insightbb.com) |
| 03:21.25 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 03:22.35 | h3x | yeah linksys atas are alright |
| 03:22.40 | lunaphyte | i have a somewhat long list of prefixes that i'm sending out a particular channel. how can i construct a list outside of my dialplan and reference it within? |
| 03:22.43 | h3x | i think overall snom phones are the best |
| 03:23.01 | h3x | theres still room for improvement, but at least it covers a full spectrum of features people usually want |
| 03:23.17 | h3x | polycoms look nice but they are the hardest to configure |
| 03:23.41 | husimon | so the 3102 is the standard working ata that people buy? |
| 03:23.50 | husimon | going to purchase about 7 if so |
| 03:23.55 | lunaphyte | seems to be. |
| 03:24.08 | lunaphyte | but i guess it depends on what your particular needs are. |
| 03:24.10 | h3x | uhm no its 1fxs 1fxo |
| 03:24.13 | h3x | are you sure you need a fxo port |
| 03:24.26 | rcslex | I have a lot of echo with the 3102s |
| 03:24.27 | h3x | i think you want the 2102 usually |
| 03:24.54 | drmessano | No |
| 03:25.07 | drmessano | The PAP2-T |
| 03:25.10 | drmessano | Thats the 2 port FXO |
| 03:25.14 | husimon | yeah this is a 1 port |
| 03:25.22 | drmessano | the 2102 is a router |
| 03:25.26 | drmessano | Well |
| 03:25.27 | ZaVoid | PAP2T IS 2 port fxs |
| 03:25.29 | drmessano | Router + ATA |
| 03:25.34 | drmessano | Sorry |
| 03:25.35 | drmessano | FXS |
| 03:25.39 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 03:25.39 | drmessano | I had FXO on the brain |
| 03:25.54 | husimon | fxs is the phone side right, fxo is the ptsn side? |
| 03:25.55 | ZaVoid | fxs=plug phone in fxo=plug phone line from pstn in :) |
| 03:25.56 | drmessano | PAP2-T = 2 port FXS |
| 03:26.01 | ZaVoid | right husimon |
| 03:26.04 | drmessano | yes |
| 03:26.14 | husimon | k just making sure :P |
| 03:26.16 | drmessano | SPA-2102 is a FXS + Router |
| 03:26.19 | ZaVoid | pap2t is cheap piece of junk |
| 03:26.21 | ZaVoid | for example |
| 03:26.26 | drmessano | SPA-3102 is a FXO + FXS + Router |
| 03:26.30 | drmessano | LOL |
| 03:26.34 | ZaVoid | it supports 1 line of g729 and 1 of g723 or 2 g723 |
| 03:26.34 | joez212 | now it says sip listening 0.0.0.0 on port |
| 03:26.35 | joez212 | woah |
| 03:26.37 | drmessano | PAP2's work very well |
| 03:26.47 | drmessano | PAP2T supports 2 lines of G729 |
| 03:26.57 | drmessano | PAP2 supports 1 line of G729 |
| 03:26.58 | ZaVoid | however if you have 1 session up and have advertised g729 in the SDP then G729 is not available on line2 even if line1 didn't use g729 |
| 03:27.09 | ZaVoid | nope both support only 1 line drmessano |
| 03:27.27 | drmessano | ..ok |
| 03:27.38 | husimon | so the spa 3102 can failback to to a normal phone line if the voip fails? |
| 03:27.50 | ZaVoid | 2102 can do two lines of g729 |
| 03:28.20 | lunaphyte | husimon: yes. |
| 03:28.27 | husimon | i guess i'll just never use that part |
| 03:28.38 | husimon | since i'm doing asterisk -> sip -> ata -> analog phone. |
| 03:28.41 | drmessano | The PAPT has 2 G729 licenses |
| 03:28.44 | husimon | that's correct right? |
| 03:28.48 | drmessano | Do a search of PAP2T and G729 |
| 03:28.52 | lunaphyte | don't waste your money if you don't need an fxo port. |
| 03:29.30 | husimon | ok so the lesser one is what model? 2102? |
| 03:29.35 | lunaphyte | pap2t |
| 03:29.48 | drmessano | 2102 is FXS |
| 03:29.53 | drmessano | 2 x FXS |
| 03:30.00 | drmessano | and a router too |
| 03:30.03 | husimon | ah |
| 03:30.11 | lunaphyte | both the 2102 and the pap2t are 2x fxs |
| 03:30.26 | plik | has anyone seen / got good docs on setting up the SPA 3102? ... mine should arrive tomorrow or Weds |
| 03:30.28 | drmessano | the pap2t isn't a router.. thats the difference |
| 03:30.46 | husimon | k yeah I don't need a router |
| 03:30.51 | drmessano | 3102 is a router, FXO and FXS |
| 03:30.54 | lunaphyte | plik: i was able to find (along with help here) what i needed. |
| 03:31.12 | husimon | the pap2t work well enough? |
| 03:31.15 | ZaVoid | it doesn't work drmessano |
| 03:31.17 | drmessano | The 3102 is a meant to be a branch office solution, but its also the cheapest way to get FXO on Asterisk |
| 03:31.26 | plik | and so you've wtirrten a nice concise how-to ? |
| 03:31.33 | ZaVoid | i got hundreds of them in use |
| 03:31.38 | lunaphyte | plik: yeah, in my head.. :p |
| 03:31.42 | husimon | ZaVoid, ah nice |
| 03:31.45 | plik | heh |
| 03:31.58 | *** join/#asterisk `Sean (i=Un1x@CPE001c351a764e-CM0014e8869416.cpe.net.cable.rogers.com) |
| 03:32.05 | husimon | drmessano, as opposed to a digium card eh |
| 03:32.25 | drmessano | yes, and as opposed to the piece of crap X100P |
| 03:32.31 | husimon | i'd be tempted to play with asterisk as a home line but I only use my cell |
| 03:32.42 | ZaVoid | husimon: not really |
| 03:32.51 | drmessano | It's effectively the best way to entry level an FXO |
| 03:32.52 | husimon | ZaVoid, not really as they suck? |
| 03:33.29 | ZaVoid | they are ok but they are cheap ata's |
| 03:33.39 | ZaVoid | granted thet are better then really cheap ata's from china.. but still |
| 03:33.42 | husimon | what type of problems have you had? |
| 03:33.54 | ZaVoid | mostly codec issues and general freezing of the units |
| 03:34.18 | husimon | see this why i'm asking, i'm willing to spend more money to get good units, i guess i'll just go with 3102 unless someone can suggest a better unit. |
| 03:35.40 | drmessano | husimon, I have several SPA-3102s in service and they work fine for me.. |
| 03:36.14 | drmessano | I've also got dozens of PAP2s out there, and they work fine too |
| 03:36.23 | drmessano | But, everyone has their experiences :) |
| 03:37.19 | lunaphyte | now, who can tell me how to avoid having a litany of extensions listed in extensions.conf to match all of the various prefixes that should go out a particular channel? |
| 03:37.40 | plik | how do you handle Transfer on an analogue phone plugges in to FXS? DTMF? ordoes the ATA understand flash-hook / recall ? |
| 03:38.18 | drmessano | flash-hook |
| 03:38.20 | ZaVoid | so drmessano you have a pap2(any variant) with 2 lines of g729 active at the same time when you go to the info page? |
| 03:38.28 | plik | cheers drmessano |
| 03:38.41 | drmessano | I only have standard PAP2's |
| 03:38.56 | [hC] | Qwell: alive? |
| 03:38.59 | drmessano | I take that back.. I have one V2 |
| 03:39.32 | h3x | i hate those locked V2's |
| 03:39.41 | h3x | such a pain |
| 03:39.46 | drmessano | I've always heard the PAP2Ts could handle two channels of G729.. even seen it said here a few times.. but if you say it doesn't work, then ok |
| 03:39.51 | drmessano | Yeah, the V2s such |
| 03:39.52 | ZaVoid | nope they don't |
| 03:39.52 | drmessano | Yeah, the V2s suck |
| 03:39.59 | ZaVoid | i've tested each variant |
| 03:40.22 | ZaVoid | whats crazy drmessano is that the SDP advertises the g729.. but it won't allow it to be used |
| 03:40.30 | h3x | hahahhahahah |
| 03:40.31 | h3x | nice |
| 03:41.01 | ZaVoid | yep quality |
| 03:41.20 | drmessano | So you'd rather use a grandstream? :) |
| 03:41.34 | h3x | use an old spa-2100 |
| 03:41.35 | h3x | they rock |
| 03:41.43 | h3x | t.38, 2x g.729 |
| 03:41.49 | h3x | router |
| 03:41.50 | ZaVoid | hehe i didn't say that drmessano |
| 03:41.59 | ZaVoid | actually i do like my gxp-2000 on my desk.. works fine |
| 03:42.05 | ZaVoid | cept for the speakerphone |
| 03:42.15 | drmessano | Other than the glaring G729 issue, i've had no issues with the PAP2s.. Running 3.1.6 firmware on the lot |
| 03:42.39 | drmessano | G711 works fine for me anyway :) |
| 03:44.19 | drmessano | oh |
| 03:44.41 | drmessano | ..and the CID on the SPA-3102 doesn't work with a shit if you don't have CallerID on the line :) |
| 03:45.10 | kyron | hehehehe |
| 03:45.40 | husimon | drmessano, laugh |
| 03:45.41 | drmessano | I'm going to fix that crap tomorrow |
| 03:45.49 | drmessano | First thing |
| 03:45.50 | husimon | i am thinking the 2102 because then I get two FXS |
| 03:45.58 | husimon | i need about 5 analog lines so... |
| 03:46.03 | drmessano | My business manager will complain.. but I have a reputation to rebuild |
| 03:46.16 | kyron | `core show translation` (<<--is it really re-calculated?) is interesting, basically telling me not to use more than 1 codec.. |
| 03:47.10 | drmessano | allow=gsm is all you need.. |
| 03:47.57 | ZaVoid | lol |
| 03:48.01 | ZaVoid | g.729 for life |
| 03:48.24 | [hC] | I use g729 on PAP2s? |
| 03:48.27 | [hC] | Whats the problem? |
| 03:48.38 | ZaVoid | none |
| 03:48.46 | ZaVoid | try to use 2 lines of g729 simultaneously though |
| 03:49.06 | [hC] | I presume a second call on call waiting counts? |
| 03:49.11 | drmessano | No |
| 03:49.11 | [hC] | I can do that. |
| 03:49.15 | drmessano | Both LINES |
| 03:49.27 | ZaVoid | line 1 and line 2 |
| 03:49.33 | [hC] | Oh, you mean physical line1 and 2. |
| 03:49.36 | ZaVoid | yes |
| 03:49.36 | drmessano | Both HOLES |
| 03:49.39 | drmessano | yeah |
| 03:49.50 | [hC] | Thats interesting. |
| 03:50.07 | [hC] | using 2 g729 licenses that way should count the same as 2 licenses being used on one active handset. |
| 03:50.12 | [hC] | You'd think, anyways. |
| 03:50.17 | kyron | well well...got meself a poly 601 now... |
| 03:51.21 | drmessano | I'm gonna get a bunch of Cisco softphones.. X-Lite is too cheapy for me |
| 03:51.24 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 03:51.27 | drmessano | j/k |
| 03:51.46 | [hC] | You worked last time i tried you, and now you refuse to dial |
| 03:52.21 | kyron | drmessano, LOL..one soft for another... tsk |
| 03:52.24 | ZaVoid | drmessano: this is form the pap2 admin guide |
| 03:52.26 | ZaVoid | A codec resource is considered as allocated if it has been included in the SDP codec list of an active call, even though it eventually may not be the one chosen for the connection. So, if the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G.729a resource is already allocated and since only one G.729a resource is allowed per |
| 03:52.59 | kyron | that "sucks" |
| 03:53.03 | ZaVoid | yes |
| 03:53.36 | drmessano | *shrug*.. thats cool... Just always heard the PAP2T allowed 2.. but I never used one, so have nothing to back it up :) |
| 03:53.37 | kyron | I have doubts on the validity of `core show translation` , the figures are too stable :P |
| 03:53.39 | drmessano | Not surprising |
| 03:53.39 | ZaVoid | so even advertising g729 disables g723 from being used on line 2 if its in use on line 1 |
| 03:54.07 | drmessano | I got enough PAP2s I unlocked that I never bought a T |
| 03:54.34 | drmessano | After my one try at T38 with a SPA3102, I decided I didn't need a T for anything anyway |
| 03:56.40 | husimon | i'm just gonna get a handful of pap2t |
| 03:56.47 | husimon | and throw more at the problem until it goes away! |
| 03:56.48 | husimon | hehe |
| 03:56.59 | husimon | screw you stupid wireless handset users! |
| 03:57.31 | drmessano | PAP2 + cordless phone = much love |
| 03:57.45 | drmessano | Screw those $300 wifi sip phones! |
| 03:57.55 | husimon | sip phone -> forward -> cell phone |
| 03:58.03 | drmessano | I got S-IP... Sorta IP |
| 03:58.17 | husimon | although I think I might get a nice headset so I can walk into the server room and talk |
| 03:58.17 | kyron | drmessano, yeah, Mediatrix 1104 + 4 cordless ;) |
| 03:58.20 | husimon | my cell phone cuts out there |
| 03:58.20 | jameswf-home | We use asstra +DEC |
| 03:59.30 | *** join/#asterisk techie (n=techie@adsl-76-214-31-194.dsl.lsan03.sbcglobal.net) |
| 03:59.31 | drmessano | I've become addicted to grandcentral |
| 03:59.40 | drmessano | I cant get my number to ring enough places |
| 03:59.44 | drmessano | 8 is NOT enough |
| 03:59.58 | husimon | do you really want to be bugged that much :P |
| 04:00.07 | drmessano | lol |
| 04:00.18 | drmessano | Well, I am going to give the number out at work |
| 04:00.28 | husimon | i wonder what frequency wireless headset is best for going through walls |
| 04:00.30 | husimon | 900mhz? |
| 04:00.33 | [hC] | why on earth would you do that |
| 04:00.41 | [hC] | thats the last people i want calling me on EVERY PHONE IMAGINABLE |
| 04:00.45 | drmessano | I sent myself an invite on our corporate email addy and got a "work only" number |
| 04:00.53 | J4k3 | husimon: 49 mhz |
| 04:00.57 | kyron | the higher the frequency, the lower the penetration (900MHz) |
| 04:01.05 | husimon | they make 49mhz wireless handsets? |
| 04:01.05 | drmessano | So I am gonna run it to my cell, desk phone at work, and one emergency phone at home |
| 04:01.26 | drmessano | and other places at work as needed.. I got up to 8 and it was actually a bit much |
| 04:01.39 | drmessano | But I am trying to cut back on my cell minutes |
| 04:01.57 | drmessano | Damn calling me on my cell when I am at another building with a LL on the desk in front of me |
| 04:02.47 | husimon | isn't 49mhz like old cell phones? |
| 04:02.54 | drmessano | Cordless |
| 04:02.55 | drmessano | not cell |
| 04:03.25 | husimon | yeah but they don't make modern phones on that frequency anymore right? |
| 04:03.54 | drmessano | 49.61, .63, .67, .71, .73, .77, .81, .83, .93, .97 |
| 04:03.58 | drmessano | Nope |
| 04:04.02 | J4k3 | husimon: theres actually been a 'reemergance' of 49 mhz stuff. |
| 04:04.13 | drmessano | err |
| 04:04.15 | drmessano | 46 |
| 04:04.17 | drmessano | Not 49 |
| 04:04.25 | drmessano | 46 was the desk |
| 04:04.31 | drmessano | 49 was the handset |
| 04:04.35 | kyron | gnight all |
| 04:04.35 | J4k3 | yep |
| 04:04.39 | husimon | so the best you can get now for going through walls is 900mhz |
| 04:05.00 | [koss] | pull-up antennas <3 |
| 04:05.00 | kyron | (low freq phones might be a good idea) |
| 04:05.11 | drmessano | 46.61, .63, .67, .71, .73, .77, .81, .83, .93, .97 and 49.81, 49.83, 49.87, 49.93, and 49.97 |
| 04:05.11 | J4k3 | it covered a good sized chunk of my ~1000 home neighborhood |
| 04:05.22 | J4k3 | husimon: yes and no |
| 04:05.22 | kyron | [koss], yeah...except for that ;P |
| 04:05.32 | J4k3 | husimon: it penetrates best, but your noise at 900 may be very high and walls won't hold it out |
| 04:05.48 | husimon | J4k3, guess i'll just buy a few and find out |
| 04:05.49 | [koss] | haha i think we just remember a long range because the antenna was 4 ft long |
| 04:05.56 | drmessano | Higher freq, higher penetration |
| 04:06.23 | J4k3 | drmessano: longer the wavelength the better the effective penetration |
| 04:06.42 | J4k3 | but logically higher noise, since anything emitting that noise will actually penetrate |
| 04:06.58 | J4k3 | 2.4 crap... you get a couple layers of foil-backed foam insulation and the signal disappears ;) |
| 04:07.00 | drmessano | What do you define at effective penetration? |
| 04:07.05 | drmessano | as* |
| 04:07.12 | husimon | when the bitch moans |
| 04:07.22 | mihinomenest | (the longer the wavelength, the less energy is wasted on oscilation; more energy for penetration) |
| 04:07.29 | J4k3 | when she calls back the next day and she's still breathing hard? |
| 04:07.30 | J4k3 | :D |
| 04:07.34 | husimon | heh |
| 04:07.41 | husimon | when she calls back the next day, can't walk, but wants more. |
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| 04:08.50 | J4k3 | we're dirty! |
| 04:09.02 | Ritzerisk | whats a good dialer program to consider |
| 04:09.23 | husimon | so here's a question: what is the best basic sip phone |
| 04:09.30 | husimon | good deal for the money kind of phone |
| 04:09.35 | J4k3 | define 'best' and 'basic' |
| 04:09.48 | husimon | hmm |
| 04:09.50 | J4k3 | I think the best effective bang/buck is the grandsuck budgetnone 101 |
| 04:10.06 | J4k3 | but, its a grandsuck budgetnone 101... but damn its half the price of the next cheapest SIP handset |
| 04:10.12 | mihinomenest | husimon: the $85 grandstream GXP-2000 ? |
| 04:10.15 | husimon | we have 7940's |
| 04:10.17 | husimon | i don't want to buy more |
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| 04:10.37 | husimon | well hmm those are 189 now |
| 04:10.40 | husimon | cheaper then before |
| 04:10.45 | husimon | plus no stupid cisco license fee |
| 04:10.52 | mihinomenest | the grandstream will leave you wanting something with a richer featureset, though. |
| 04:10.57 | J4k3 | yeah |
| 04:11.04 | J4k3 | a bt101 will make you want... a phone |
| 04:11.10 | husimon | heh |
| 04:11.18 | J4k3 | but, its still a great deal |
| 04:11.26 | J4k3 | folks here love polycom, thats my next personal phone purchase |
| 04:12.27 | mihinomenest | Polycoms look cartoony, IMO. |
| 04:12.55 | husimon | it's those stupid red and blue buttons they use |
| 04:13.01 | husimon | ip301s? |
| 04:13.08 | mihinomenest | (and the handsets) |
| 04:13.16 | husimon | what handsets? |
| 04:13.34 | husimon | which ones have handsets i mean |
| 04:14.23 | mihinomenest | the desktop polycoms... |
| 04:14.48 | husimon | model? |
| 04:14.52 | mihinomenest | ip301. |
| 04:15.14 | [TK]D-Fender | IP301 = dead choice. |
| 04:15.44 | [TK]D-Fender | The only Polycom without a handset is the IP 4000, but thats a technicality |
| 04:15.49 | mihinomenest | anything on this page really: http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip.html |
| 04:15.54 | husimon | there is always 7970's :P what a waste of money |
| 04:16.00 | husimon | ~$500 for a phone |
| 04:16.08 | J4k3 | a 7970 is a fucking ripoff |
| 04:16.13 | husimon | no kidding |
| 04:16.14 | drmessano | HAH... http://www.howstuffworks.com/question333.htm |
| 04:16.16 | J4k3 | you can buy a PC with a 17" LCD to run a softphone on |
| 04:16.16 | husimon | oo man a color screen |
| 04:16.18 | J4k3 | for that price |
| 04:16.21 | drmessano | "Can anyone hear my baby monitor?" |
| 04:17.09 | J4k3 | "yes" |
| 04:17.23 | J4k3 | my friend in downtown austin says its at a whole different level now |
| 04:17.25 | drmessano | Yes, and we can hear you doing the nasty if the crib is still in your bedroom |
| 04:17.58 | mihinomenest | I had a customer connected to my 900MHz FWAP. turns out it was her neighbor's video baby monitor over 400yds away. |
| 04:17.58 | J4k3 | 900 and 2.4 NTSC cameras |
| 04:17.58 | J4k3 | voyeurism |
| 04:18.09 | jameswf-home | .me objects to the new night rider |
| 04:18.12 | husimon | interesting, the uniden phone with 5.8ghz handset |
| 04:18.14 | mihinomenest | I've also had way to many customers call up and complain that their internet stops working while they're on the phone. |
| 04:18.22 | jameswf-home | dof |
| 04:18.27 | husimon | mihinomenest, that's what you get for using 2.4ghz phones |
| 04:18.36 | drmessano | Yep |
| 04:18.39 | mihinomenest | "oh really, let me take a look" so I ping'em. "tick tick tick tick tick" |
| 04:18.54 | J4k3 | husimon: DECT is a nice choice |
| 04:19.01 | J4k3 | 1.9 ghz, penetrates like PCS cellular |
| 04:19.14 | mihinomenest | husimon: I've had more problems with 5.8GHz phones interfering with my 900MHz last mile than I have had 900MHz phones interfering. |
| 04:19.20 | J4k3 | I wanna say they allow about 40mW EIRP on it, maybe 100mW... something like that, at least |
| 04:19.33 | jameswf-home | <bevis> yeah yeah penetrate </bevis> |
| 04:19.36 | J4k3 | mihinomenest: because most "5.8" phones only do 5.8 one way |
| 04:19.38 | J4k3 | :P |
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| 04:19.51 | mihinomenest | J4k3: I know. |
| 04:20.14 | J4k3 | DECT stuff is all on 1.9ghz |
| 04:20.22 | J4k3 | its actually a 5 mhz sliver in the middle of the pcs cell band |
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| 04:26.18 | VJFROMGT | I sudently stop getting audio via iax trunk, signaling works but no audio |
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| 04:39.53 | jameswf-home | Plop plop fizzz fizzzz |
| 04:53.48 | drmessano | heh |
| 04:54.22 | J4k3 | if your plop fizzes, you should go to the clinic immediately. |
| 04:55.31 | [hC] | I dont suppose its possible to originate a call as g729 rather than slin? |
| 04:56.14 | [hC] | im trying to originate a call which plays a g729 file to a box which answers on SIP and accepts only g729, but it complains that it cant translate from slin to g729 (lack of g729 license) |
| 04:56.22 | [TK]D-Fender | [hC], your initial channel should use whatever codecs your peer (you ARE using a defined peer, RIGHT?) was set up to offer. |
| 04:57.53 | [TK]D-Fender | *b00m* |
| 04:58.12 | [hC] | [TK]D-Fender: im just using a call file which starts with dialing Local/999@default, which dials out (IAX) to a remote box that plays a file using g729 - this works when i call it with a g729 capable phone |
| 04:58.21 | [hC] | then the other leg of the call does another call to a different peer, using g729 as well |
| 04:58.25 | [hC] | and it still tries to transcode to slin |
| 04:58.47 | [hC] | ive also tried replacing one of the call-outs with a Playback(tt-monkeys) which only exists in g729, and it wants to transcode to slin |
| 04:58.53 | [TK]D-Fender | [hC], well I'd inspect your peer definition and the CLI output of your call..... |
| 04:59.07 | [hC] | presumably because all originated calls from within asterisk itself using call files start by trying to use slin.. |
| 05:00.00 | [hC] | calling from g729 capable phones, both local extensions work without need to transcode. but, trying to bridge two outgoing calls that were both made using g729 together is not working. |
| 05:00.13 | [hC] | All I want to do is originate a ton of calls out using g729 to test the load of a line |
| 05:00.23 | [hC] | but on this architecture, there is no g729 codec available. |
| 05:00.34 | [TK]D-Fender | [hC], Blah, blah, blah, blah, blah, blah, ok, fine, sure. Lest talk, more show! |
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| 05:00.47 | [TK]D-Fender | less* |
| 05:01.05 | [hC] | I'm trying to figure out a way to do it! It just doesnt look like its possible. |
| 05:01.10 | [TK]D-Fender | [hC], pastebin makes most things possible |
| 05:01.18 | [hC] | aside from placing all of these calls from an actual handset. |
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| 05:01.55 | [hC] | I can try to show you, sure, but i dont think there is a solution... Hold up, i'll pastebin it all |
| 05:08.22 | hmmhesays | sing us a song you're the piano man |
| 05:09.14 | [TK]D-Fender | 'cause we're all in a mood for a melody... |
| 05:09.54 | hmmhesays | oh that song never gets old |
| 05:10.04 | hmmhesays | I jammed last night on the big stage here |
| 05:10.15 | [hC] | Ok I think I may have found the problem |
| 05:10.20 | [hC] | Milliwatt() uses slin. :| |
| 05:10.45 | [TK]D-Fender | hmmhesays, if you acutally played that, I'm impressed |
| 05:10.48 | [hC] | Is there a way to generate g729-only tones that are constant? aside from using a .g729 file with a constant tone in it? |
| 05:10.52 | hmmhesays | [TK]D-Fender, I didn't |
| 05:10.58 | hmmhesays | I just played some pop/punk stuff |
| 05:11.01 | hmmhesays | a little blink 182 |
| 05:11.06 | drmessano | ewww |
| 05:11.07 | [TK]D-Fender | [hC], play milliwatt and encode it |
| 05:11.13 | [hC] | Yeah, I can do that. |
| 05:11.15 | [TK]D-Fender | hmmhesays, Bleh |
| 05:11.22 | drmessano | Blink 182 = blah |
| 05:11.27 | hmmhesays | beer and no warm up, and only getting 2 songs doesn't make for much of a jam |
| 05:11.34 | hmmhesays | blink always gets the crowd going |
| 05:11.37 | drmessano | heh |
| 05:12.27 | hmmhesays | anyway, I'm going to pick up some of these cheap ass yoki d drums on ebay |
| 05:12.39 | hmmhesays | we lost our practice area so now we have to be quiet |
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| 05:15.45 | markgreene | How can I output everything being sent to the CLI, to a file? So that I don't need to be looking at the CLI to see an error go by. |
| 05:15.59 | jameswf-home | markgreene: logger.conf |
| 05:17.00 | markgreene | jameswf-home: I'll google that then. Thanks |
| 05:17.28 | hmmhesays | the only way to quiet a drummer down is to make him go digital for practice |
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| 05:34.35 | jblack | the other way to quiet a drummer is to threaten them with a nuisance lawsuit |
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| 05:38.59 | markgreene | Anyone in here know mysql enough to tell me how I can sort call records by the number of digits in the dst field? Baiscly I only want to see where the destination was a 4 digit extension. |
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| 05:40.46 | micander | select * from tableName where Len(fieldName)=4; should work |
| 05:41.06 | drmessano | lol |
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| 05:43.38 | [hC] | [TK]D-Fender: so here's something relevant. When calling from my phone, this all works now (eliminated milliwatt) but when originating a call i get "No codecs found to offer, cancelling call" - SIP debug and traceroute both show that no attempt was made. Doing a sip show peer <x> it claims that it knows about g729. |
| 05:44.03 | [hC] | Maybe asterisk wont originate a call with only g729 available when no codec_g729a.so is loaded.. |
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| 05:45.22 | [hC] | yeah the problem is on the originating box, when you bring up the first leg of the call (before the second leg is up) it tries to speak slin. There's not gonna be any way around this. |
| 05:45.35 | [TK]D-Fender | [hC], Makes sense |
| 05:46.00 | [TK]D-Fender | [hC], Sure there is... buy a friggen license |
| 05:46.05 | [hC] | It does yeah, it would be nice if you could tell it not to try defaulting to slin, though. |
| 05:46.15 | [hC] | I'm doing this on blackfin, there is no codec_g729a.so available for it. |
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| 05:50.32 | hmmhesays | [hC], you're working on a blackfin also? |
| 05:50.51 | hmmhesays | fun |
| 05:51.00 | [TK]D-Fender | [hC], I am continually amused by your ability to stack improbably and doom concepts together in new and unusual ways that can only end in failure... |
| 05:51.39 | [hC] | [TK]D-Fender: how else would we learn where the limits are? :) you should be thanking me! |
| 05:51.52 | [hC] | hmmhesays: yeah, im doing a lot of work with the aa50 |
| 05:52.41 | [hC] | [TK]D-Fender: it shouldnt be that unreasonable to try to load test the capacity of an embedded box using g729 passthru! :) |
| 05:53.07 | hmmhesays | [hC], I'm working on getting spandsp working |
| 05:53.25 | [hC] | hmmhesays: wow, that sure is a challenge to bite off :) |
| 05:53.57 | [hC] | hmmhesays: im trying to figure out whats wrong with VLAN tagging on this thing, and would love to see a codec_g729a.so become available. |
| 05:54.09 | hmmhesays | [hC], compile one |
| 05:54.19 | hmmhesays | the sources are available |
| 05:54.22 | [hC] | hmmhesays: i just may have to. |
| 05:55.01 | hmmhesays | I have spandsp running with rxfax on my blackfin, but there is too much floating point code slowing things down |
| 05:55.18 | hmmhesays | coppice suggested I started converting equalizer to fixed point |
| 05:55.28 | hmmhesays | s/started/start/ |
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| 06:33.03 | drmessano | yawn |
| 06:37.12 | drmessano | jblack |
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| 06:52.04 | metfan2007 | Hi all, I have an issue in iax.conf, I setting callerid for a user like callerid="Name" <9393>, but the issue is when I try to get that values using ${CALLERID(name)} and ${CALLERID(num)}, a receive blank variables... any idea? |
| 06:58.40 | ManxPower | don't use quotes |
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| 07:00.10 | ifnotwhynot | hi there could anyone please give me the link to the new asterisk 1.4 book please |
| 07:02.20 | drmessano | ~book |
| 07:02.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 07:08.09 | ifnotwhynot | thanx |
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| 07:39.37 | BBHoss | sup fools |
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| 07:48.16 | jblack | drmessano: Sir? |
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| 08:16.28 | jblack | It's quiet tonight |
| 08:17.25 | BBHoss | sure is |
| 08:17.49 | BBHoss | probably too early for europeans to come on |
| 08:24.01 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
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| 08:27.59 | endre | it 9.27am |
| 08:28.01 | endre | it's |
| 08:28.04 | endre | in central europe |
| 08:28.13 | BBHoss | 2:28AM here |
| 08:28.22 | BBHoss | Central US :) |
| 08:28.23 | endre | hah that's early then |
| 08:32.41 | *** join/#asterisk BBHoss (n=hoss@c-71-207-173-38.hsd1.al.comcast.net) |
| 08:32.41 | BBHoss | that was odd |
| 08:34.37 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) [NETSPLIT VICTIM] |
| 08:36.34 | *** join/#asterisk Navion (n=billp@208-106-107-201.dsl-s.california.com) |
| 08:37.20 | Navion | Anyone had any trouble being able to break dial tone with Sangoma FXS's? |
| 08:39.32 | Navion | It seems like some DTMF digits are not being recognized while dial tone is present. Once the dial tone is broken all the digits are recognized, even the ones the card wouldn't decode with dial tone present. |
| 08:41.23 | Navion | So many people logged in to computers at work... Everyone is asleep here. |
| 08:41.40 | BBHoss | no |
| 08:41.47 | BBHoss | just can't help you :) |
| 08:42.13 | Navion | That's comforting. ;) |
| 08:42.24 | BBHoss | i would suggest calling sangoma |
| 08:43.06 | Navion | I sent them a tech eMail. Seems like I've sent those before without a response. |
| 08:43.28 | BBHoss | emails are a dime a dozen |
| 08:43.30 | Navion | maybe a phone call would be better. |
| 08:43.37 | BBHoss | exactly |
| 08:43.55 | Navion | Any idea where their tech support is? |
| 08:44.09 | BBHoss | no idea |
| 08:44.55 | Navion | I never know if it's better to call in daytime in the Americas, Europe or Asia |
| 08:45.12 | Navion | Of course it's all made in China... |
| 08:45.46 | Navion | Or Canada... |
| 08:46.12 | BBHoss | ontario |
| 08:46.45 | *** join/#asterisk grantm (n=grant@kolob.wingateservices.com) [NETSPLIT VICTIM] |
| 08:46.58 | BBHoss | Navion, they are after hours :( |
| 08:47.10 | Navion | I'll prpbably have to have a french translator |
| 08:47.12 | BBHoss | heh they have a conference |
| 08:47.28 | BBHoss | after-hours conference |
| 08:48.02 | Navion | Oh well....BBL |
| 08:48.58 | BBHoss | Navion, heh call the tech support |
| 08:49.07 | BBHoss | they have a call-back service :) |
| 08:49.30 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) [NETSPLIT VICTIM] |
| 08:50.38 | BBHoss | Navion, try adding relaxdtmf=yes to zapata.conf |
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| 08:59.48 | FlatFoot | morning all |
| 09:02.30 | *** join/#asterisk cjk (n=ldidelot@195.26.5.145) |
| 09:02.34 | cjk | hi, is there a variable that tells me if the channel is in t.38 ? |
| 09:03.51 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:07.40 | *** join/#asterisk sx|lappy (n=sxpert@lgit-1225.obs.ujf-grenoble.fr) |
| 09:08.40 | obnauticus | nuts |
| 09:09.25 | obnauticus | I just got some dell rapid vrails for a 7950 and epic failure. |
| 09:09.37 | obnauticus | vrails* |
| 09:12.28 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 09:18.49 | JT | rails? |
| 09:19.50 | *** join/#asterisk atop (n=user@oaktyres.force9.co.uk) |
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| 09:38.20 | nebojsajsimic | hi alll |
| 09:38.25 | BBHoss | sup dog |
| 09:40.42 | obnauticus | JT ya |
| 09:41.17 | FlatFoot | morning BBhoss |
| 09:41.22 | BBHoss | sup |
| 09:41.27 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:41.34 | obnauticus | BBHoss bows to no man! |
| 09:41.38 | FlatFoot | how's the detox program ;) |
| 09:41.45 | BBHoss | fsck that |
| 09:41.47 | obnauticus | lmao |
| 09:41.51 | JT | obnauticus: the rails don't fit? |
| 09:42.04 | obnauticus | no JT, they looked like rails for a 2-4U case |
| 09:42.13 | obnauticus | instead, they are just for a dell Poweredge 7950 |
| 09:42.23 | obnauticus | now they are useless unless i actually buy them |
| 09:42.23 | obnauticus | lmao |
| 09:42.33 | obnauticus | i got them for free |
| 09:42.34 | obnauticus | 3 kits of them |
| 09:43.00 | JT | ah ok |
| 09:43.04 | JT | 7950 must be huge |
| 09:43.07 | JT | and old? |
| 09:43.10 | obnauticus | no |
| 09:43.13 | obnauticus | they are new |
| 09:43.14 | obnauticus | and small |
| 09:43.24 | obnauticus | these rails look like they can take a 4U load |
| 09:43.43 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-9f44bcb4f4e2ad1f) |
| 09:44.22 | obnauticus | 1750 sorry |
| 09:44.50 | XnOSX | anybody here have problems with the fax in asterisk (T38)? |
| 09:45.16 | obnauticus | JT I got my 42U rack |
| 09:45.23 | obnauticus | it's actually like a 43.5U rack. |
| 09:45.39 | BBHoss | obnauticus, where the hell are you putting that? |
| 09:45.55 | obnauticus | BBHoss a lot of stuff |
| 09:46.05 | obnauticus | i have 3 servers right here |
| 09:46.09 | obnauticus | and a 2U switch |
| 09:46.15 | obnauticus | that's like 16U's |
| 09:46.18 | BBHoss | heh, sounds like my room |
| 09:46.20 | obnauticus | and im going to buy some ProLiant's off of ebay soon |
| 09:46.24 | XnOSX | when the calls enter in my pbx asterisk think that is a fax and hang up the call inmediatly |
| 09:46.25 | obnauticus | BBHoss i'm 16 though |
| 09:46.26 | obnauticus | lol |
| 09:46.35 | BBHoss | yeah, i remembered that |
| 09:46.47 | obnauticus | I'm thinking of getting an osciloscope just to put it in there |
| 09:46.53 | obnauticus | because i have nothing else to put in there |
| 09:47.01 | BBHoss | so are you just ordering this stuff fof learning/fun or an actual purpose |
| 09:47.07 | obnauticus | actual purpose. |
| 09:47.17 | obnauticus | slash fun |
| 09:47.19 | BBHoss | heh |
| 09:47.21 | BBHoss | always |
| 09:47.35 | BBHoss | you must be making some nice money from freelancing |
| 09:47.36 | obnauticus | hobbys other than sports are always good. |
| 09:47.50 | obnauticus | well im a cheapskate so i got this rack for $50 |
| 09:47.58 | BBHoss | damn that is cheap |
| 09:48.07 | obnauticus | you know what OSDL is right? |
| 09:48.09 | BBHoss | i need more room, right now i'm in a dorm room |
| 09:48.12 | BBHoss | yeah |
| 09:48.16 | obnauticus | their old uhh |
| 09:48.18 | obnauticus | datacenter |
| 09:48.28 | obnauticus | is like a few miles from me, and linus used to wokr there |
| 09:48.32 | BBHoss | yeah |
| 09:48.32 | obnauticus | Linus used the server cage i got |
| 09:48.40 | BBHoss | what kind of other fun shit do they have |
| 09:48.41 | BBHoss | cool |
| 09:48.48 | obnauticus | they used to have a lot of stuff |
| 09:48.51 | obnauticus | but i got an SAS array |
| 09:48.52 | *** join/#asterisk arthurlutz (n=arthur@logilab2-7-50.cnt.nerim.net) |
| 09:49.03 | BBHoss | heh |
| 09:49.06 | obnauticus | with 5 147.4GB HDD's in it |
| 09:49.10 | obnauticus | hot swappable |
| 09:49.14 | BBHoss | 15k or 10k? |
| 09:49.17 | obnauticus | 15K |
| 09:49.21 | BBHoss | very nice |
| 09:49.21 | obnauticus | for 200 duckets. |
| 09:49.22 | JT | oscilloscopes are pretty useful |
| 09:49.35 | obnauticus | JT, I know I made one out of an old cathode ray tube |
| 09:49.45 | obnauticus | I'm going to get a spectrum analyzer soon though. |
| 09:49.50 | obnauticus | and perhaps a GNURadio. |
| 09:49.54 | BBHoss | yeah ive always wanted an osc, but now that i'm at a university that has 100s, there is really no point |
| 09:50.11 | JT | i have 2 |
| 09:50.12 | JT | somewhere |
| 09:50.13 | obnauticus | I'm lucky that WSU lets me use their stuff for BGA soldering |
| 09:50.23 | JT | a dual channel CRO |
| 09:50.31 | obnauticus | oscilloscope? |
| 09:50.31 | JT | 100MHz i think? maybe 20? |
| 09:50.33 | BBHoss | hopefully this summer i can move into a house, then i can really jack the power bill up :) |
| 09:50.38 | JT | and a 10MHz LCD handheld one |
| 09:50.40 | obnauticus | BBHoss lmao |
| 09:50.40 | JT | yeah |
| 09:50.46 | obnauticus | I use solar panels. |
| 09:50.55 | BBHoss | yeah, if i buy we may do that |
| 09:50.55 | obnauticus | well i try to as much as possible |
| 09:51.01 | obnauticus | I have a single point of failure though |
| 09:51.05 | obnauticus | if power goes out im toast |
| 09:51.13 | obnauticus | well i have a 4U APC that will keep me up for 4 hours |
| 09:51.15 | BBHoss | im starting to get some more clients for my asterisk consulting |
| 09:51.18 | JT | put important stuff in a datacentre |
| 09:51.20 | obnauticus | other than that, there's an epic failure. |
| 09:51.28 | obnauticus | Jt, that's what colocation is for |
| 09:51.28 | obnauticus | lol |
| 09:51.40 | BBHoss | colo is too fscking expensive |
| 09:51.48 | obnauticus | not really |
| 09:51.58 | obnauticus | if you generate enough money for it, it will pay for it's self. |
| 09:52.02 | BBHoss | id like to open up a colo center in an old nuke silo, that would be cool |
| 09:52.07 | JT | colo is cheap these days |
| 09:52.10 | JT | as a customer |
| 09:52.16 | JT | not to build a datacentre |
| 09:52.18 | obnauticus | i think he's talking about like |
| 09:52.22 | obnauticus | two racks |
| 09:52.28 | BBHoss | yeah |
| 09:52.30 | obnauticus | That's what i would use |
| 09:52.39 | BBHoss | cheaper in bulk, but the small stuff gets expensive |
| 09:52.41 | obnauticus | and a lot of internet. |
| 09:52.42 | JT | i have one rack |
| 09:53.01 | obnauticus | I have one now |
| 09:53.20 | JT | i meant colo |
| 09:53.21 | obnauticus | for this next year it will look riduclous with only like 7 things in it though |
| 09:53.23 | JT | i also have one at home |
| 09:53.36 | obnauticus | Didn't I tell you what i saw in a colo one time? |
| 09:53.41 | obnauticus | like two nights ago |
| 09:53.41 | BBHoss | ive got a sun X4100 that i should colo, its just collecting dust because it is SOOOO loud |
| 09:53.55 | obnauticus | BBHoss ProLiant's are also extremely loud |
| 09:53.58 | obnauticus | i have one sitting outside my room |
| 09:54.02 | obnauticus | i cannot sleep with it on |
| 09:54.04 | BBHoss | dunno if they can top this one |
| 09:54.05 | BBHoss | same here |
| 09:54.10 | JT | BBHoss: is that 1RU? |
| 09:54.10 | obnauticus | Ya. |
| 09:54.11 | BBHoss | its like 40db MINIMUM |
| 09:54.14 | obnauticus | lmao |
| 09:54.15 | obnauticus | wtf |
| 09:54.16 | BBHoss | JT, yeah |
| 09:54.21 | obnauticus | mine sounds like a vaccuum cleaner. |
| 09:54.30 | obnauticus | with all what |
| 09:54.32 | JT | BBHoss: yeah, that's the one i saw in a datacente with a SATA enclosure |
| 09:54.33 | obnauticus | 12 fans it has in it |
| 09:54.35 | obnauticus | hot swappable. |
| 09:54.37 | JT | whole rack dedicated to it |
| 09:54.47 | obnauticus | that's cool |
| 09:54.57 | obnauticus | I have a rack near mine that's just all ATX cases stacked up |
| 09:54.57 | BBHoss | its got 6 small-diameter high-rpm fans that turn at like 6000rpms minimum |
| 09:55.01 | obnauticus | for about 4 feet |
| 09:55.21 | BBHoss | anyone ever try Asterisk on SPARC? |
| 09:55.25 | obnauticus | I dunno the specs of the priliant's. |
| 09:55.55 | obnauticus | DL380 G4's are really nice though |
| 09:55.58 | obnauticus | I like the iLO |
| 09:56.17 | BBHoss | oh the ILOM on the Sun is awesome |
| 09:56.20 | JT | the latest DL360s are pretty nice |
| 09:56.26 | obnauticus | I haven't tried those |
| 09:56.28 | JT | 1RU with redundant PSUs |
| 09:56.34 | obnauticus | I try to stick to IPMI though. |
| 09:56.38 | BBHoss | remote console, display, cdrom, harddrive, floppy, everything |
| 09:56.46 | obnauticus | remote console, nice. |
| 09:56.51 | obnauticus | I have the adapters you know |
| 09:56.54 | obnauticus | with the console server switch |
| 09:57.07 | BBHoss | oh its more than console, you can use it with GUIs too |
| 09:57.16 | obnauticus | lol X11 LOM |
| 09:57.20 | obnauticus | GUI LOM |
| 09:57.24 | BBHoss | yep |
| 09:57.26 | obnauticus | wait |
| 09:57.28 | obnauticus | i was just kidding |
| 09:57.37 | BBHoss | mostly for windows dudes |
| 09:57.41 | obnauticus | Oh |
| 09:57.41 | obnauticus | ew. |
| 09:57.52 | BBHoss | well who runs x11 on a server? |
| 09:57.53 | obnauticus | but when i fill this whole rack up |
| 09:57.58 | JT | i'm trying to decide what phone to put in my co-lo rack |
| 09:58.01 | obnauticus | BBHoss people who run gustygibbom |
| 09:58.05 | JT | it has to be a cheapish phone |
| 09:58.07 | JT | but not shit |
| 09:58.09 | obnauticus | Jt you are putting a phone in there |
| 09:58.11 | obnauticus | to annoy people? |
| 09:58.14 | obnauticus | that's what i would do |
| 09:58.14 | obnauticus | lmao |
| 09:58.14 | BBHoss | heh |
| 09:58.19 | obnauticus | put a speaker in there |
| 09:58.19 | JT | how would it annoy them? |
| 09:58.21 | obnauticus | blare music |
| 09:58.22 | obnauticus | call them |
| 09:58.24 | obnauticus | when tehy are near by |
| 09:58.27 | obnauticus | stream a webcam |
| 09:58.29 | obnauticus | through the internet |
| 09:58.32 | obnauticus | and yell at them when they walk by |
| 09:58.36 | JT | heh |
| 09:58.38 | JT | i see |
| 09:58.45 | obnauticus | I'll set that up in a week |
| 09:58.47 | BBHoss | obnauticus, just hook up asterisk to the OSS/ALSA sound card and plug a speaker in |
| 09:58.48 | obnauticus | and crap in the bottom of my rack |
| 09:58.54 | JT | more like when i want to make calls when i'm at the rack |
| 09:58.57 | obnauticus | play 8 bit music on hold stuff. |
| 09:59.01 | JT | hate using mobile phones when unnecessary |
| 09:59.05 | JT | also for testing |
| 09:59.08 | obnauticus | JT I know. |
| 09:59.15 | obnauticus | does your colo have WiFi? |
| 09:59.20 | obnauticus | you could just use that on your server. |
| 09:59.34 | JT | my rack doesn't |
| 09:59.38 | obnauticus | Mine doesn't have WiFi but a few punks setup some Aironet's in their racks |
| 09:59.42 | obnauticus | lol |
| 09:59.43 | JT | use it on my server for what? |
| 09:59.54 | JT | yeah i see some people have tried to hide aps in their racks |
| 09:59.58 | obnauticus | i mean use their WiFi so you can text a softphone |
| 10:00.06 | obnauticus | ya, the people can't locate it |
| 10:00.11 | obnauticus | they screwed it into their case i heard |
| 10:00.12 | obnauticus | it was funny |
| 10:00.20 | JT | and another rack has a pile of crappy gsm antennas sitting above their rack on the cable tray |
| 10:00.25 | obnauticus | wtf? |
| 10:00.30 | JT | and about 6 GSM cards |
| 10:00.38 | obnauticus | I'll get you a picture of the dudes who stacked the ATX cases in the rack |
| 10:00.42 | JT | multiport gsm card too |
| 10:00.44 | JT | heh |
| 10:00.46 | obnauticus | it's like crooked too |
| 10:00.49 | JT | i can't take pictures |
| 10:00.52 | obnauticus | not even properly stacked. |
| 10:00.57 | obnauticus | they don't let you? |
| 10:00.58 | JT | against datacentre rules |
| 10:01.03 | obnauticus | use your cell phone |
| 10:01.06 | JT | all photos must be approved by management |
| 10:01.12 | JT | people have done that before |
| 10:01.21 | JT | and security have gone up to them within 2 minutes |
| 10:01.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 10:01.31 | obnauticus | act like you are calling |
| 10:01.34 | obnauticus | and press the take picture button |
| 10:01.35 | obnauticus | lmao |
| 10:01.46 | JT | those pictures suck anyway |
| 10:01.49 | obnauticus | ya |
| 10:01.57 | obnauticus | Just hook a webcam to your server. |
| 10:02.00 | obnauticus | that would be entertaining. |
| 10:02.04 | JT | and someone may or may not have taken pictures near my rack before cctv went up there |
| 10:02.10 | JT | with a dslr |
| 10:02.34 | obnauticus | I can bring an SLR into mine |
| 10:02.39 | obnauticus | they don't care |
| 10:02.44 | JT | so can i |
| 10:02.46 | JT | except they care |
| 10:02.51 | obnauticus | Do racks exist with biometric locks? |
| 10:03.11 | JT | bit of a wank |
| 10:03.20 | JT | if the person got past building security |
| 10:03.22 | JT | there are issues |
| 10:03.56 | obnauticus | ya |
| 10:04.00 | obnauticus | mine generally aren't diccks |
| 10:04.04 | obnauticus | probably because of my age |
| 10:05.02 | JT | i have to swipe 7 times to get to my rack |
| 10:05.06 | JT | and pass the security desk |
| 10:05.16 | obnauticus | RFID and biometrics |
| 10:05.20 | obnauticus | at mine |
| 10:05.20 | JT | your face from the record is displayed on an lcd above the second entrance door |
| 10:05.22 | obnauticus | both insecure |
| 10:05.46 | JT | they have 24/7 security, better than any automated security measures |
| 10:05.49 | obnauticus | that's pretty cool |
| 10:05.56 | obnauticus | Disneyland-ish though |
| 10:06.14 | JT | minimum 4 security guards on after hours, 5+ business hours |
| 10:06.22 | JT | facility security guards that is |
| 10:06.32 | JT | some suites have private security too |
| 10:06.35 | obnauticus | I like watching the fat person at the NOC |
| 10:06.57 | JT | one floor of the datacentre is owned by the federal government here |
| 10:07.03 | JT | and they have armed guards |
| 10:07.05 | obnauticus | Oh that makes sense. |
| 10:07.16 | JT | which are a government agency |
| 10:07.20 | JT | not private security |
| 10:07.26 | obnauticus | lol what if they used tasers |
| 10:07.26 | obnauticus | miss! |
| 10:07.26 | obnauticus | fail!! |
| 10:07.39 | JT | shrug, that floor, the guards have guns |
| 10:08.33 | BBHoss | damn |
| 10:08.33 | obnauticus | what address range do you get? |
| 10:08.36 | obnauticus | and how much bandwidth you got? |
| 10:08.37 | BBHoss | guns? |
| 10:08.43 | JT | BBHoss: feds |
| 10:08.47 | BBHoss | ahh |
| 10:08.48 | JT | the have a whole floor |
| 10:08.57 | BBHoss | JT which datacenter do you use? |
| 10:09.15 | JT | this datacentre has 6.5 floors of colo space |
| 10:09.21 | JT | 1.5 floors of facilities |
| 10:09.26 | obnauticus | where is this |
| 10:09.27 | obnauticus | that is huge. |
| 10:09.29 | JT | sydney |
| 10:09.32 | JT | globalswitch.com |
| 10:09.34 | obnauticus | oh |
| 10:09.35 | BBHoss | ahh yes |
| 10:09.40 | JT | space for over 3000 racks |
| 10:09.51 | JT | power usage when full will be around 50MW |
| 10:10.00 | obnauticus | nuts. |
| 10:10.13 | JT | at the moment they are having issues with energy supply |
| 10:10.43 | JT | they have 2 redundant set of incoming power cables taking diverse routes, each running at 33kV |
| 10:10.53 | JT | and each has dual 33 to 11kV transformers |
| 10:10.59 | obnauticus | Ya, I need to up the amperage on my circuits to my room too |
| 10:11.09 | JT | the cables and globalswitch transformers can handle the power |
| 10:11.13 | obnauticus | mine's got n+1 with power |
| 10:11.15 | obnauticus | i think they are fine |
| 10:11.32 | JT | the transformers sending the power from the power company are getting maxxed out |
| 10:11.40 | JT | at the substation |
| 10:11.59 | obnauticus | Wonder what their power bill is per month eh |
| 10:12.03 | obnauticus | probably more than their backbone |
| 10:12.03 | obnauticus | lol |
| 10:12.07 | JT | dunno |
| 10:12.08 | JT | err |
| 10:12.15 | JT | globalswitch is strictly carrier neutral |
| 10:12.29 | obnauticus | how do your lines terminate |
| 10:12.29 | JT | they do not offer telco/Internet connectivity under any circumstances |
| 10:12.31 | obnauticus | in your cage? |
| 10:12.35 | BBHoss | i dunno why more of these companies don't do solar |
| 10:12.45 | JT | because solar is a waste of time |
| 10:12.54 | JT | for such high power |
| 10:12.56 | obnauticus | wind generator! |
| 10:13.02 | BBHoss | they used to be, but the new solar is much more efficient |
| 10:13.15 | JT | obnauticus: i get connectivity from the company that leases this cage |
| 10:13.26 | JT | globalswitch is just an it real estate provider |
| 10:13.33 | JT | it's pretty much the best arrangement |
| 10:14.38 | obnauticus | I dunno how good the guys are at montioring my usage |
| 10:14.38 | obnauticus | lol |
| 10:15.13 | obnauticus | they give me a port on a cisco switch and if i want to split it i can... |
| 10:15.23 | obnauticus | as long as i have the space to |
| 10:15.38 | obnauticus | it's only like 40/40 though |
| 10:18.07 | cjk | hi, is there a variable that tells me if the channel is in t.38 or not ? |
| 10:18.19 | BBHoss | has anyone here been getting a "your search looks like a bot" message from google? |
| 10:18.22 | BBHoss | http://sorry.google.com/sorry/?continue=http://www.google.com/search%3Fhl%3Den%26q%3Dasterisk%2Bconsulting%26btnG%3DGoogle%2BSearch |
| 10:18.35 | *** join/#asterisk ifnotwhynot (n=davidh@c1-211-5.rrba.isadsl.co.za) |
| 10:19.02 | ifnotwhynot | sorry to ask agian book for asterisk 1.4 link please |
| 10:19.10 | BBHoss | ~book |
| 10:19.10 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:19.15 | ifnotwhynot | thx |
| 10:22.46 | obnauticus | BBHoss i've gotten that once. |
| 10:23.01 | BBHoss | its happening constantly to me, so damn annoying |
| 10:23.19 | BBHoss | it happens whether i use the firefox-builitn or google.com |
| 10:23.59 | obnauticus | check your browser useragent |
| 10:24.07 | Alan1234 | Morning. Question: Can some recommend a company in the UK that can give me a UK number and SIP access to my Asterisk system? |
| 10:24.11 | obnauticus | Maybe that's what's causing it. |
| 10:24.25 | BBHoss | Alan1234, so you want a UK DID? |
| 10:24.52 | obnauticus | What kind of LOM do eservers havE? |
| 10:25.19 | obnauticus | OR do they have any :| |
| 10:25.22 | Alan1234 | BBHoss: i am just coming up to speed on all this terminology, so yes, i think DID is what i want |
| 10:26.02 | BBHoss | Alan1234, didww.us provides unlimited DIDs for a monthly fee, you might try them |
| 10:26.15 | Alan1234 | BBHoss: thank you, appreciate the pointer |
| 10:26.30 | BBHoss | Alan1234, there are TONS out there though, look around |
| 10:26.40 | Alan1234 | http://www.voip-info.org/wiki/view/DID+Service+Providers |
| 10:26.50 | Alan1234 | yup, now you told me the term to search for i see oodles! thanks |
| 10:28.08 | BBHoss | np |
| 10:28.21 | *** join/#asterisk badcfe_ (i=christia@alltid.dritings.no) |
| 10:28.26 | badcfe_ | hello everybody. i just installed the TC400B wild card. and modprobed it alright as i can see in dmesg with a "successfull" notice and all. now, i start the asterisk and write "show translation" in the CLI, and theres no new possibly translation apearing. also, theres no command such as "show transcoder" like written in the pdf doc of the card from digium. i have left out something, but nothing thats written in the doc afaik ... help? |
| 10:29.12 | agx | little question: when i fork a call with Dial(SIP/10&SIP/11) and i answer with SIP/10 on LCD of SIP/11 i've "missed call: 1" is there any option to avoid this? |
| 10:29.38 | jblack | If the phone rings, it rings. |
| 10:29.38 | badcfe | maybe one must add something in modules.conf ..? |
| 10:29.56 | jblack | Try one phone at a time |
| 10:30.16 | jblack | But still if a phone rings an you don't answer it on that phone, then it's a missed call on that phone |
| 10:30.27 | badcfe | or is there some --with-zaptel that must be specified when compiling asterisk? |
| 10:30.34 | agx | jblack, but i've answered onto another phone, so its not a missed call ;-) |
| 10:30.42 | BBHoss | agx, there is really not a great solution for it |
| 10:31.06 | agx | BBHoss, i could imagine but normal pbx does not have this little drawback |
| 10:31.09 | jblack | agx: Logic with me won't do you any good. Try making logic to the phone you didn't ansewr. |
| 10:31.34 | BBHoss | agx, it doesn't have anything to do with asterisk, it has to do with the phones |
| 10:32.13 | BBHoss | agx, if you hook a polycom up to one of those new-fangled nortel boxes, it will do the same thing |
| 10:32.16 | agx | BBHoss, is any phone able to distinguish between a SIP CANCEL and a "no answer" timeout? |
| 10:32.29 | BBHoss | agx, not sure |
| 10:32.53 | tzafrir | badcfe, no. normally configure should detect an installed zaptel |
| 10:33.36 | obnauticus | bed times! |
| 10:33.46 | obnauticus | cya BBHoss, JT |
| 10:33.52 | BBHoss | obnauticus, cya |
| 10:33.53 | badcfe | tzafrir: is it possible to check if my running asterisk is aware of zaptel? |
| 10:34.01 | obnauticus | BBHoss i bow down to no man. |
| 10:34.07 | BBHoss | heh |
| 10:34.11 | obnauticus | ya that's right |
| 10:34.12 | obnauticus | k |
| 10:34.26 | agx | BBHoss, should i ask on -dev? it seems if you send an optional header with the CANCEL the phone will not show the missed call |
| 10:34.36 | agx | BBHoss, for instance: Reason: SIP;cause=200;text="Call completed elsewhere" |
| 10:34.48 | tzafrir | badcfe, I figure: ls -l /proc/PID_OF_ASTERISK/fd | grep /dev/zap |
| 10:35.08 | BBHoss | agx, you could try, i dunno how you would customize it though, but it would certainly be nice if we had an option for that |
| 10:35.24 | JT | oh, obnauticus left :/ |
| 10:35.30 | JT | was going to url some pics of the dc |
| 10:35.36 | BBHoss | agx,you can disable the missed calls though |
| 10:35.48 | BBHoss | feature.8.enabled="0" will disable it |
| 10:36.24 | JT | BBHoss: http://80.68.88.208/~jon/DSC_3638.JPG |
| 10:36.25 | badcfe | tzafrir: that command (substituted the pid ofcourse) gave nothing. maybe a problem that i run it as asterisk user? |
| 10:36.33 | JT | http://80.68.88.208/~jon/DSC_3654.JPG |
| 10:36.44 | BBHoss | nice, thats huge |
| 10:36.56 | JT | http://80.68.88.208/~jon/DSC_3675.JPG |
| 10:37.00 | JT | yeah |
| 10:37.17 | tzafrir | It means asterisk has not opened any zaptel device file |
| 10:37.46 | tzafrir | This may be because it hasn't tried (no support) or because zaptel kernel support is not there |
| 10:37.59 | tzafrir | lsmod | grep ^zaptel |
| 10:38.12 | JT | BBHoss: those pics are from a year ago |
| 10:38.22 | JT | now there's much less light coming from the windows |
| 10:38.29 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 10:38.29 | BBHoss | i bet |
| 10:38.33 | JT | they board up the windows when an area becomes in use |
| 10:38.49 | BBHoss | damn, so security concious |
| 10:38.59 | badcfe | tzafrir: zaptel 192584 1 zttranscode |
| 10:39.04 | JT | yeah the 3rd floor is completely boarded up |
| 10:39.06 | BBHoss | what would they do if someone came in there guns blazing though? |
| 10:39.11 | JT | government |
| 10:39.15 | JT | hmm |
| 10:39.18 | BBHoss | so... :) |
| 10:39.20 | JT | first, how would they come in? |
| 10:39.43 | JT | the reception is about half a floor above the road |
| 10:39.49 | JT | there's a first door at road level |
| 10:39.50 | badcfe | tzafrir: when i compiled zaptel i had kernel headers, _not_ all the kernel _source_. is that an evil? |
| 10:39.51 | JT | steps |
| 10:39.54 | JT | a second door |
| 10:39.55 | tzafrir | badcfe, zaptel's there. I guess asterisk indeed has no support |
| 10:39.56 | BBHoss | well, ideally, they would have colo space to get access |
| 10:39.58 | JT | reception/security |
| 10:40.10 | JT | then mantrap tubes |
| 10:40.12 | tzafrir | badcfe, zaptel is OK, as you can see |
| 10:40.13 | BBHoss | or turn someone in security |
| 10:40.25 | JT | then lift or door to get to L2 colo corridor |
| 10:40.25 | BBHoss | burn i mean |
| 10:40.40 | badcfe | tzafrir: okay. i tried to run the asterisk daemon as root and without safe_shit. still the same. |
| 10:40.45 | JT | burn? |
| 10:41.13 | BBHoss | apparently its a CIA term, it was used in a book i read |
| 10:41.42 | JT | ah ok |
| 10:41.55 | badcfe | tzafrir: where is a config file that specifies how asterisk looks for zaptel stuff? |
| 10:42.05 | JT | let's say they get past reception |
| 10:42.12 | JT | which means smashing about 3 sets of glass |
| 10:42.21 | JT | then they have to get past fire doors |
| 10:42.28 | JT | or elevtor |
| 10:42.38 | JT | then there's more fire doors |
| 10:42.51 | BBHoss | sounds great |
| 10:42.54 | JT | to the rooms off the corridor on each floor |
| 10:43.04 | BBHoss | but does security have access to all of these things |
| 10:43.09 | JT | probably |
| 10:43.21 | JT | except the inside of some customers' cages i'm guessing |
| 10:43.27 | BBHoss | they never pay the security guys enough |
| 10:43.30 | tzafrir | badcfe, headers: /usr/include/zaptel |
| 10:43.58 | tzafrir | badcfe, you need codec_zap, right? |
| 10:44.18 | tzafrir | Do you have the module codec_zap.so? |
| 10:44.19 | JT | it would have to be stealthy |
| 10:44.30 | badcfe | tzafrir: i guess its something more lame i have forgotten, as i am a newbee to zaptel. but its something missing from the doc ive read. must i run ztcfg or something? |
| 10:44.35 | JT | so shooting up the place wouldn't work |
| 10:44.40 | JT | cops would be there in 3 minutes |
| 10:45.00 | BBHoss | JT, usually those places have great physical security, but you always have to remember the human element |
| 10:45.10 | tzafrir | badcfe, zaptel is OK |
| 10:45.14 | JT | true |
| 10:45.14 | tzafrir | Look into Asterisk |
| 10:45.22 | badcfe | tzafrir: show modules like zap gives only the app_zapateller.so |
| 10:45.23 | yang | When I am calling my mobile number from asterisk from the extension 059209586 (which should show this CALL-ID), I am seeing number 0338606057 http://openpaste.org/en/4978/ . I have specified callerid=059209586 ... |
| 10:45.33 | BBHoss | JT, but then again, thats true for every organization |
| 10:45.37 | tzafrir | basically, look into the results of configure |
| 10:45.40 | JT | indeed |
| 10:45.55 | BBHoss | its one of my hobbies to analyze how to break into a place using social engineering :) |
| 10:46.04 | JT | :) |
| 10:46.06 | badcfe | tzafrir: does the configure script log .. where? |
| 10:46.13 | BBHoss | just for fun though of course |
| 10:46.23 | BBHoss | it would be nice if i could get paid for it though :) |
| 10:46.30 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 10:46.33 | JT | let's say you got all the way into a cage |
| 10:46.45 | JT | then the racks have keys, and a lot of cages have customer run cctv |
| 10:46.51 | JT | that customer nocs monitor |
| 10:46.59 | sergee | are there any experienced spa962/932 users here? |
| 10:47.15 | BBHoss | JT, what kind of response time would you get out of CCTV though? |
| 10:47.27 | BBHoss | JT, or you could use something to disable it |
| 10:47.38 | BBHoss | massive infrared light or something |
| 10:47.49 | JT | building security could respond within minutes |
| 10:47.59 | sergee | i wonder, is it possible to use linebuttons from spa962 in the same way as buttons on spa932 (to monitor some lines)? |
| 10:48.01 | JT | well if it was been interfered with, they would probably send someone |
| 10:48.26 | BBHoss | maybe, but if they didn't see you they might just assume equipment malfunction |
| 10:48.33 | BBHoss | and send someone the next day |
| 10:48.40 | JT | maybe, depends on customer |
| 10:48.44 | JT | some are really paranoid |
| 10:49.03 | BBHoss | and what kind of keys to the racks have, like 2-pin locks? |
| 10:49.11 | JT | the cage next to me has over 200 racks in it, and over 30 cctv cameras run by the customer |
| 10:49.20 | JT | and 5 racks dedicated to the security gear |
| 10:49.42 | JT | and some of their sub cages have full rfi shielding mesh |
| 10:50.02 | yang | I am using these extensions - http://openpaste.org/en/4979/ |
| 10:50.07 | JT | depends on the rack that people use |
| 10:50.48 | BBHoss | the cheaper ones would just have like a minibar key that could be bumped easily |
| 10:51.12 | BBHoss | i guess the biometric ones would present a problem |
| 10:51.17 | yang | The VOIP provider admin has unlocked the CALLER-ID function for me, otherwise I am always going out as CALLERID as my primary number |
| 10:51.19 | JT | perhaps, but again if you're not meant to be there and they see you, you may be out of luck |
| 10:51.49 | BBHoss | it would definitely be easier to infiltrate the corporation that owns it |
| 10:52.09 | JT | heh |
| 10:53.27 | JT | it's funny in equinix sydney |
| 10:53.31 | JT | it's one big open floor |
| 10:53.35 | JT | some in cages |
| 10:53.37 | JT | most not |
| 10:53.42 | JT | but there's 2 suites |
| 10:54.02 | JT | so they're like grey square boxes with walls and roofs inside the co-lo area |
| 10:54.07 | BBHoss | heh |
| 10:54.14 | JT | one belongs to optus, the other to telstra |
| 10:54.29 | JT | the optus one has cctv cameras mounted on the cable trays pointing at the roof |
| 10:54.34 | JT | to monitor the exhaust air vents |
| 10:54.45 | JT | for the suite |
| 10:54.51 | BBHoss | thats cool |
| 10:55.21 | JT | equinix sydney is a joke facilities wise though, glad i only have 1RU there |
| 10:55.39 | JT | they only have 3 gensets iirc |
| 10:55.46 | JT | and no gas fire supression |
| 10:56.17 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:56.52 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 10:57.04 | jmls | wondering why my zttest is so bad |
| 10:57.17 | jmls | average is 99.977315 |
| 10:57.28 | BBHoss | jmls, what kind of mobo is this? |
| 10:57.29 | JT | that's on the edge of ok |
| 10:57.30 | jmls | running on a dell 2950. |
| 10:57.46 | jmls | it's a TE410P |
| 10:57.51 | jmls | or TE405P. |
| 10:57.59 | JT | BBHoss: did i mention that GS has 10 2.2MVA gensets atm? |
| 10:58.03 | JT | but they're upgrading |
| 10:58.13 | BBHoss | damn |
| 10:58.18 | BBHoss | what kind of AC? |
| 10:58.23 | JT | they already get noise complaints from residents |
| 10:58.29 | JT | even with 10metres of baffling |
| 10:58.38 | JT | and water sprays to suppress noise |
| 10:58.53 | JT | the gensets run at 11kV 3 phase |
| 10:58.56 | BBHoss | i know The Planet had plenty of power backup, but their AC wouldn't restart so they had to shut donw :) |
| 10:58.59 | jmls | how in god's name do I turn off the IO-APIC-level and revert back to a standard IRQ 1-13 ? |
| 10:59.17 | JT | BBHoss: the AC does not shut down at global switch in a power outage |
| 10:59.30 | jmls | i've got wct4xxp on IRQ 193 with IO-APIC-level |
| 10:59.37 | JT | upgrading to 22gensets eventually |
| 10:59.53 | JT | they use rotary upses |
| 11:00.17 | JT | they can power the chillers and all necessary equipment and customer gear before a genset starts |
| 11:00.25 | BBHoss | power shouldnt be a problem then :) |
| 11:00.38 | atop | what can cause low results for zttest? my system shows an average of 98.75 but the calls sounds ok. Is is something to worry about? |
| 11:00.38 | JT | each rotary ups lasts for 19seconds max |
| 11:00.48 | JT | genset takes 9 seconds max to start providing clean power |
| 11:00.55 | BBHoss | that works |
| 11:01.02 | JT | the gensets are hotstart |
| 11:01.04 | JT | it's insane |
| 11:01.24 | JT | i couldn't work out why the generator rooms were so noisy with them off |
| 11:01.37 | BBHoss | wow |
| 11:01.54 | BBHoss | what kind of fuel? |
| 11:01.57 | JT | until the guy said that they have electric motors constantly spinning the cranks at operating rpm, and coolant and lubricant is kept at operating temp 24/7 |
| 11:02.02 | Mavvie | Can some bugmanager put #11917 from feedback to open? |
| 11:02.09 | JT | these are massive Cat V16s too |
| 11:02.11 | JT | Diesel |
| 11:02.16 | BBHoss | ahh |
| 11:02.21 | BBHoss | of course |
| 11:02.40 | BBHoss | maybe they should see if they can get a test nuclear reactor to power it :) |
| 11:02.45 | JT | hah |
| 11:02.51 | JT | as if, no nucelar power in australia |
| 11:02.55 | JT | big political shitfight |
| 11:03.00 | BBHoss | oh really |
| 11:03.19 | BBHoss | i have one less than 20 miles from my location |
| 11:03.28 | JT | oh and the Thrane chillers are pretty impresive too |
| 11:03.28 | Mavvie | me too. |
| 11:03.31 | Mavvie | the only one in .au |
| 11:03.43 | JT | 3 installed, will be turned into 6 |
| 11:03.49 | JT | each chiller is 3.2MW |
| 11:04.02 | JT | Mavvie: that's not nuclear power though |
| 11:04.16 | Mavvie | JT: it's the closest thing there is to it. |
| 11:04.20 | JT | heh |
| 11:04.28 | BBHoss | what are you talking about? |
| 11:04.43 | JT | there's a nuclear reactor for medial/scientific purposes |
| 11:04.48 | JT | not power generation |
| 11:04.51 | JT | much smaller scale |
| 11:04.53 | BBHoss | yeah |
| 11:05.36 | JT | on cold mornings you can see clouds of steam rising from global switch |
| 11:05.42 | JT | ~60 cooling towers iirc |
| 11:06.46 | Mavvie | that reminds me that I have to find the real date of their power maintenance work. |
| 11:06.58 | BBHoss | we have 3 GE BWRs here, nearly 3840 megawatts of electricity |
| 11:07.03 | JT | maintenance eh? |
| 11:07.25 | JT | what are they doing? |
| 11:07.58 | Mavvie | Not really sure, but then went from room 1 to room 5. |
| 11:08.14 | Mavvie | skipped room 2 till 4 |
| 11:08.21 | JT | on l2? |
| 11:08.25 | Mavvie | yups. |
| 11:08.30 | JT | does it involve people losing power? |
| 11:08.43 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 11:08.44 | Mavvie | one day lead A, next day lead B |
| 11:09.02 | JT | loss of power? that's stupid |
| 11:09.15 | JT | they have static switches |
| 11:09.22 | JT | i don't see why they need to shut power down |
| 11:11.14 | Mavvie | got the email here: |
| 11:11.23 | Mavvie | As you may be aware changes in OH&S legislation related to electrical |
| 11:11.23 | Mavvie | works on switchboards require Global Switch to upgrade all level 2 |
| 11:11.23 | Mavvie | switchboards. |
| 11:11.25 | yang | When I am calling my mobile number from asterisk from the extension 059209586 (which should show this CALL-ID), I am seeing number 0338606057 http://openpaste.org/en/4978/ . I have specified callerid=059209586 ... |
| 11:11.29 | yang | I am using these extensions - http://openpaste.org/en/4979/ |
| 11:11.43 | JT | oh i see |
| 11:11.51 | XnOSX | i have a pbx asterisk machine, when a call try access asterisk think that its a fax and hang up the call, what can i do for recive my calls wihout problems??? |
| 11:11.54 | JT | glad i'm not on level 2 then |
| 11:13.31 | Mavvie | it's not my idea of a great time, in one rack we have two boards which are both drawing both 14 Ampere. If you turn one off, it will go to something > 16 Ampere and shutdown the board. |
| 11:13.35 | badcfe | i have zaptel, but my newly compiled and installed asterisk 1.4.17 does not see it. how can i proceed? |
| 11:13.40 | JT | no idea what the changes involve? |
| 11:13.46 | Mavvie | Not really sure how to resolve that one yet except using next racks powerboards. |
| 11:14.01 | JT | hrm |
| 11:14.13 | Mavvie | No, is on need to know base and we're just little fishies. |
| 11:14.16 | JT | tell them to give you temporary extra feeds |
| 11:14.19 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:14.21 | JT | hah |
| 11:14.34 | JT | it's just power works, you wouldn't think it'd be a huge secret |
| 11:14.39 | JT | may i'll ask security next time |
| 11:14.57 | Mavvie | ask the little round electrician :-) |
| 11:15.13 | JT | i'm never in the l2 colo space |
| 11:15.16 | JT | well i was today |
| 11:15.22 | JT | but that's not normal :) |
| 11:15.57 | badcfe | i see no /usr/lib/asterisk/modules/chan_zap.so ... should this have been there? |
| 11:16.14 | *** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar) |
| 11:17.10 | badcfe | my crying is maybe over. i figured out that i must /etc/init.d/zaptel start |
| 11:18.41 | badcfe | but no .. no one has /dev/zap/transcode open, and asterisk does not have any zap in it .. 8-( |
| 11:19.14 | BBHoss | badcfe, did you compile zaptel first then asterisk? |
| 11:20.06 | BBHoss | also, when you compile asterisk, make sure you make menuselect then go in and make sure the zaptel stuff is enabled |
| 11:20.38 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 11:23.05 | *** join/#asterisk nighty^ (n=nighty@p1052-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 11:24.51 | badcfe | BBHoss: yes. |
| 11:25.44 | badcfe | BBHoss: chan_zap has XXX is menuconfig 8-( |
| 11:25.56 | BBHoss | hmm, what distro is this |
| 11:26.35 | badcfe | debian etch with zaptel zaptel-1.2.23 and asterisk-1.4.17 on an amd64 |
| 11:26.43 | BBHoss | hmm |
| 11:27.32 | BBHoss | well apparently asterisk can't see that you have zaptel installed |
| 11:28.09 | BBHoss | the modules aren't loaded either are they? |
| 11:28.42 | badcfe | BBHoss: theyre not. my operating system has got zaptel module perfecly up and running tho |
| 11:29.01 | jblack | huh. i didn't know calls between callwithus numbers are free |
| 11:29.16 | badcfe | BBHoss: i now tried ./configure ; make menuselect from a shiny new 1.4.17 tar-ball. same thing |
| 11:29.30 | BBHoss | hmm |
| 11:30.09 | BBHoss | i guess go back to the zaptel direcotry |
| 11:30.11 | badcfe | codec_zap has XXX too, in the menuselect. and ive even started the zaptel init script! |
| 11:30.13 | BBHoss | ./configure |
| 11:30.15 | BBHoss | make |
| 11:30.17 | BBHoss | make install |
| 11:30.19 | BBHoss | make config |
| 11:30.26 | BBHoss | try it again, then try asterisk again |
| 11:30.33 | badcfe | ill do it .. |
| 11:30.59 | atop | you are running the asterisk ./configure before checking menuselect again right? |
| 11:31.03 | badcfe | should i nuke the previous zaptel build directory first? |
| 11:31.08 | *** join/#asterisk SteveTotaro (n=root@pool-70-22-26-147.balt.east.verizon.net) |
| 11:31.10 | BBHoss | make clean |
| 11:31.15 | badcfe | ok |
| 11:31.45 | badcfe | BBHoss: theres no configure script in zaptel |
| 11:32.03 | BBHoss | hmm |
| 11:32.35 | atop | there is in mine |
| 11:32.38 | atop | cd.. |
| 11:32.45 | atop | oops, wrong window |
| 11:32.48 | BBHoss | heh |
| 11:33.05 | BBHoss | badcfe, you are correct, just make and make install |
| 11:33.41 | badcfe | BBHoss: even tho previously i did see that zaptel was prefectly installed (aparently) in dmesg and all |
| 11:34.03 | atop | hang on, why are you using 1.2 zaptel with 1.4 asterisk? |
| 11:34.05 | BBHoss | is this just for ztdummy, or do you have a card |
| 11:34.19 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-224-150.nsw.bigpond.net.au) |
| 11:34.35 | badcfe | /usr/teas/src/zaptel-1.2.23/xpp/xbus-core.c:1254: warning: cast to pointer from integer of different size |
| 11:34.39 | badcfe | this is an amd64 |
| 11:34.58 | BBHoss | have you tried the 1.4.8 branch? |
| 11:35.00 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-224-150.nsw.bigpond.net.au) |
| 11:35.01 | badcfe | may be a problem? (i dont think so since the system got zaptel good) |
| 11:35.20 | badcfe | 1.4.8 of zaptel? |
| 11:35.23 | BBHoss | yes |
| 11:35.27 | badcfe | nope |
| 11:35.32 | badcfe | where can i fetch that? |
| 11:35.36 | BBHoss | or w/e the latest is |
| 11:35.50 | atop | I dont think asterisk 1.4 will 'see' a 1.2 zaptel, which would explain the lack of a zap channel |
| 11:36.02 | atop | http://downloads.digium.com/pub/zaptel/releases/zaptel-1.4.8.tar.gz |
| 11:36.20 | BBHoss | atop, it was my understanding that 1.4 will work with both 1.2 and 1.4 zaptel versions, but i may be incorrect |
| 11:36.30 | defswork | is IAX safe to running stragith over the internet ? or should it always be via VPN ? |
| 11:36.35 | defswork | straight* |
| 11:36.47 | BBHoss | defswork, its as safe as anything else |
| 11:36.56 | BBHoss | you can even add encryption in it |
| 11:37.04 | atop | BBHoss, you may well be right, I dont know either way, but unless there's a really good reason to use a 1.2 zaptel....? |
| 11:37.08 | BBHoss | you cant do that with SIP (in asterisk) |
| 11:37.30 | badcfe | i put MODULES="$MODULES wctc4xxp" in /etc/default/zaptel and comment the rest |
| 11:38.16 | badcfe | i dont need the other modules right? |
| 11:38.31 | BBHoss | well you have a wctc4xxp card correct> |
| 11:38.48 | badcfe | yes. |
| 11:38.59 | BBHoss | thats alll you should need then |
| 11:39.12 | badcfe | hmm. the "make configure" in zaptel tells me about a "/etc/sysconfig/zaptel" --- i dont have such a file |
| 11:39.24 | BBHoss | make configure? |
| 11:39.49 | BBHoss | on 1.4, run ./configure |
| 11:40.07 | badcfe | Zaptel has been configured. |
| 11:40.14 | badcfe | I think that the zaptel hardware you have on your system is: |
| 11:40.19 | badcfe | (nothing) |
| 11:40.34 | badcfe | thats what the make config in zaptel tells |
| 11:41.07 | BBHoss | ahh, you must have zap 1.4x for asterisk 1.4x |
| 11:41.34 | BBHoss | it says so in the USERS MANUAL |
| 11:41.52 | atop | yeah, I should read that one of these days |
| 11:41.55 | SteveTotaro | where can one find this USER'S MANUAL? |
| 11:42.19 | BBHoss | SteveTotaro, this is for the TC400P/M card |
| 11:42.28 | BBHoss | but digium makes users manuals for all of their products |
| 11:43.00 | SteveTotaro | That is a nice addition |
| 11:44.56 | BBHoss | badcfe, are you trying to install 1.4.8 now? |
| 11:46.17 | SteveTotaro | mmmm, first cup of coffee for the day... |
| 11:46.29 | mvanbaak | SteveTotaro: wow |
| 11:46.39 | badcfe | BBHoss: yes. i did make config too. which ends in I think that the zaptel hardware you have on your system is: |
| 11:47.16 | BBHoss | type: modprobe wctc4xxp |
| 11:47.26 | SteveTotaro | black coffee, no sugar, no cream... |
| 11:47.32 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 11:47.47 | badcfe | zttranscode 13968 1 wctc4xxp |
| 11:47.56 | badcfe | and all is there (as before) .. the system has it. |
| 11:48.16 | BBHoss | badcfe, then look at dmesg and see if you see something about wctc4xxp: Wildcard tc400P........\ |
| 11:48.33 | badcfe | Zaptel DTE (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) |
| 11:48.33 | badcfe | Found and successfully installed a Wildcard TC: Wildcard TC400P+TC400M |
| 11:48.47 | SteveTotaro | is that trixter? |
| 11:49.02 | BBHoss | badcfe, ok good so far |
| 11:49.05 | badcfe | (as before (zap 1.2.23)) |
| 11:49.12 | BBHoss | hmm |
| 11:49.30 | BBHoss | you need to remove all traces of 1.2.23 from your system |
| 11:49.30 | badcfe | whats the next step (maybe at some point i vomited on my system) |
| 11:49.48 | BBHoss | the next step is to compile asterisk |
| 11:49.55 | BBHoss | but since you already have it compiled |
| 11:50.03 | BBHoss | you probably need to make clean first |
| 11:50.06 | BBHoss | then make |
| 11:50.09 | BBHoss | then make install |
| 11:50.13 | badcfe | ok |
| 11:51.31 | badcfe | AND ITS GOOD! |
| 11:51.34 | BBHoss | ok |
| 11:51.38 | BBHoss | go ahead and make |
| 11:51.41 | BBHoss | then make install |
| 11:51.52 | badcfe | ok (was it the "old" zap that fooled me..) |
| 11:52.13 | BBHoss | yeah i believe since you were using 1.2 with 1.4 asterisk, it didnt see it |
| 11:53.22 | badcfe | BBHoss: but is that version of zap kind of development branch? |
| 11:53.28 | BBHoss | no |
| 11:53.36 | BBHoss | well, as much as 1.4 asterisk is |
| 11:58.02 | BBHoss | badcfe, does it work |
| 11:58.10 | BBHoss | you can check it out with show transcode |
| 11:58.53 | badcfe | BBHoss: yes it does. thank you. im eager to set up a call bridged by it now. a sec.. |
| 11:59.38 | ifnotwhynot | hi ther i hav a problem with my asterisk box if i want to go to my cdr directory it just hangs does anyone know how to solve this? |
| 11:59.56 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:00.01 | tzafrir | anyone managed to get a remote asterisk hung with completion lately? |
| 12:00.09 | tzafrir | e.g: I run: |
| 12:00.21 | tzafrir | 'originate Zap/1 application echo' |
| 12:00.46 | tzafrir | after that I type: 'so' and press Tab (to complete to 'soft hangup') |
| 12:00.54 | tzafrir | and my asterisk -r is hung |
| 12:01.19 | *** join/#asterisk elverkilde (n=jon@85.235.240.45) |
| 12:02.26 | XnOSX | have asterisk beta2 v1.6 support for T38??????? |
| 12:02.36 | elverkilde | Hi all - * clears the callerid on my trunk, setting it to "new user", any suggestions? |
| 12:02.40 | BBHoss | tzafrir, just tested on mine, works fine here |
| 12:02.53 | BBHoss | tzafrir, im running 1.6.0-beta2 |
| 12:03.07 | yang | When I am calling my mobile number from asterisk from the extension 059209586 (which should show this CALL-ID), I am seeing number 0338606057 http://openpaste.org/en/4978/ . I have specified callerid=059209586 ... |
| 12:03.12 | yang | I am using these extensions - http://openpaste.org/en/4979/ |
| 12:03.49 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:04.37 | yang | The VOIP operater told me something about spoofing CALLID numbers and that these are restricted with anti spoofing on my account |
| 12:04.43 | *** part/#asterisk jmls (n=jmls@81.138.42.77) |
| 12:04.45 | XnOSX | have asterisk beta2 v1.6 support for T38? anybody here know about this! |
| 12:05.25 | BBHoss | XnOSX, i think there was a patch to add SendFax and RecieveFax, but im not sure if it required spandsp or not |
| 12:06.38 | XnOSX | BBHoss: i have a problem with a some calls in my asterisk when a call enter, asterisk cant recognized if is a call or a fax and the call is hang up :( |
| 12:06.59 | BBHoss | yang, why aren't you setting the callerid in extensions.conf? |
| 12:07.21 | BBHoss | XnOSX, why do you think that asterisk thinks its a fax |
| 12:08.22 | XnOSX | because i was set a debug mode and the result say me that is a fax |
| 12:08.33 | XnOSX | sorry my english is not so good friend |
| 12:08.54 | BBHoss | do you use fax? |
| 12:09.09 | XnOSX | no |
| 12:09.38 | XnOSX | i only want to recive calls but the calls hang up inmediately friend |
| 12:10.05 | Uatec | hi, i'm using an SPA 922, when i press set call fowarding to another phone, sometimes this other phone goes straight to voicemail, so the first number goes straight to voicemail |
| 12:10.12 | Uatec | this is fine if it's just one phone |
| 12:10.14 | BBHoss | XnOSX, pastebin your zapata.conf and zaptel.conf |
| 12:10.19 | BBHoss | ~pb |
| 12:10.19 | jbot | [~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://www.rafb.net/paste |
| 12:10.30 | XnOSX | BBHoss: ok hold on please |
| 12:10.35 | Uatec | but if that phone is in a call group, none of the other phones ring, the call just goes straight to voicemail |
| 12:10.40 | Uatec | is there anything i can do about that? |
| 12:10.45 | Uatec | i don't have any idea how i could stop it really? |
| 12:12.18 | tzafrir | The one I'm currently using is Debian packages 1.4.17 |
| 12:12.29 | tzafrir | But I recall this on some earlier versions |
| 12:12.52 | BBHoss | tzafrir, i am not using zaptel, can you reproduce the issue using a sip channel? |
| 12:13.25 | *** join/#asterisk saftsack (n=saftsack@p4FC7504A.dip.t-dialin.net) |
| 12:13.45 | XnOSX | BBHoss: zapata.conf http://pastebin.ca/892419 |
| 12:14.19 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:14.26 | XnOSX | BBHoss: i dont have a zaptel.conf in this asterisk |
| 12:14.31 | BBHoss | k |
| 12:15.47 | BBHoss | XnOSX, try putting faxdetect=no in each of the groups |
| 12:16.26 | tzafrir | got it |
| 12:16.38 | BBHoss | tzafrir, what was it |
| 12:16.39 | tzafrir | it's when the originating channel doesn't answer |
| 12:16.54 | XnOSX | BBHoss: oki i´ll try it |
| 12:16.55 | yang | BBHoss: like ;exten => 59209583,1,Set(CALLERID(059209583)=Joe Smith <59209583>) , before the DIAL command ? |
| 12:17.04 | tzafrir | dialplan add extension noanswer,1,wait(60) into demo |
| 12:17.08 | XnOSX | and i´ll tell you latter the result |
| 12:17.13 | BBHoss | tzafrir, yep, same here |
| 12:17.13 | yang | hi tzafrir |
| 12:17.31 | tzafrir | originate Local/noanswer@demo application echo |
| 12:17.31 | BBHoss | tzafrir, but if it answers it unlocks it |
| 12:17.34 | tzafrir | so<tab> |
| 12:18.52 | BBHoss | think a bug-report is needed? |
| 12:20.46 | tzafrir | BBHoss, any chance you could report it? It will take me a while to set up trunk to replicate it there |
| 12:20.57 | tzafrir | yang, hi, what's up? |
| 12:21.11 | BBHoss | tzafrir, yeah |
| 12:21.46 | yang | tzafrir: ah well, I am dealing with dialplan |
| 12:21.53 | *** join/#asterisk goldenfox (n=goldenfo@125.60.235.202) |
| 12:22.18 | yang | tzafrir: on the other asterisk it works well simply with callerid= option, and on the new provider it doesn't work |
| 12:23.12 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 12:23.26 | SteveTotaro | some providers do not allow you to set your callerid |
| 12:23.38 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:23.42 | yang | this one allows, i talked with operator |
| 12:23.54 | yang | but he mentioned that it needs to be set correctly on my asterisk |
| 12:24.02 | *** join/#asterisk divs123 (n=chatzill@122.164.243.133) |
| 12:24.15 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:24.40 | *** part/#asterisk divs123 (n=chatzill@122.164.243.133) |
| 12:24.58 | SteveTotaro | did he clarify "correctly"? |
| 12:25.32 | yang | SteveTotaro: he was mentioning that I had enabled anti-spoof on their side, and now he has disabled it, but still my callerid doesnt work... |
| 12:26.12 | SteveTotaro | maybe they have to do a "reload" ;) |
| 12:26.30 | carrar | totoro? |
| 12:26.40 | SteveTotaro | si senior |
| 12:26.48 | carrar | never mind |
| 12:26.55 | SteveTotaro | lol |
| 12:27.04 | nixguy | tonari no totoro tototoro |
| 12:27.07 | nixguy | totoro totoro |
| 12:27.14 | carrar | heh |
| 12:27.19 | SteveTotaro | cornjulio |
| 12:27.25 | carrar | I just arrived here |
| 12:27.34 | nixguy | tokyo rox |
| 12:27.36 | carrar | full of sushi and sake |
| 12:27.42 | nixguy | and cute girls! |
| 12:27.42 | carrar | yeah it does |
| 12:27.47 | SteveTotaro | and hot asian women |
| 12:27.51 | carrar | in little plade skirts |
| 12:27.53 | nixguy | veeellli hot |
| 12:28.22 | BBHoss | heh |
| 12:28.31 | yang | SteveTotaro: no it was reloaded (I guess), since now i receive the number 0338606057, and previously I always received my "primary defined" number... |
| 12:28.50 | carrar | Mark is missing out |
| 12:28.57 | SteveTotaro | carra, qui est la? |
| 12:28.57 | carrar | on the hottie Asian scene |
| 12:29.04 | carrar | heh |
| 12:30.04 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
| 12:30.06 | BBHoss | tzafrir, alright, reported |
| 12:30.19 | tzafrir | what number? |
| 12:30.28 | BBHoss | 11927 |
| 12:31.54 | BBHoss | should be sufficient, especially since it's reproducible |
| 12:35.03 | BBHoss | tzafrir, fair enough |
| 12:35.43 | BBHoss | damn i'm fscking starving! |
| 12:37.18 | SteveTotaro | drink more coffee, that is my breakfast and lunch |
| 12:37.19 | BBHoss | damn, china is planning to rollout 300 gigawatts of PBMR reactors! |
| 12:37.57 | coppice | just keeping down the carbon footprint :-) |
| 12:38.15 | BBHoss | yeah, sure... |
| 12:38.46 | coppice | they could use some of that power this week |
| 12:38.56 | BBHoss | PBMR's are of an elegant design |
| 12:39.17 | coppice | elegant, but kinda untried |
| 12:40.53 | BBHoss | wonder how you would shut it off |
| 12:41.23 | *** join/#asterisk Tebi (n=tebi@gw.aller.fi) |
| 12:41.39 | coppice | lots of desigs looked good, until people saw how the whole lifecycle worked out |
| 12:41.42 | cpm | pbmrs are pretty neat. |
| 12:42.23 | BBHoss | it would be even better if Thorium could be used instead of the normal enriched uranium |
| 12:42.28 | cpm | 'nuclear power' lifecycles work out pretty nasty, but folks still love their nuke power. |
| 12:42.30 | coppice | the existing reactors in china were built by foreigners. I don't think there is a lot of local experience of any kind of reactor |
| 12:42.44 | BBHoss | heh, kind of scary eh? |
| 12:43.17 | cpm | isn't pbmr kinda proprietary anyway? I would expect it would be foreigners doing this work as well |
| 12:43.48 | BBHoss | apparently there have been a few made, one in Germany, China, and ZA |
| 12:44.17 | BBHoss | all of them are low-mW reactors though, wonder how it scales |
| 12:44.37 | cpm | well, time will tell, won't it? |
| 12:45.08 | SteveTotaro | meltdown, china syndrome |
| 12:45.22 | BBHoss | AAE holds a patent related to PBMRs in the US |
| 12:45.30 | cpm | pepsi syndrome more likely |
| 12:47.44 | BBHoss | who knew the germans have been fscking around with PBMRs since '66 |
| 12:48.16 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:48.17 | coppice | its more the pepsi syndrome that worries us about the french built PWR china has near here :-\ |
| 12:49.58 | SteveTotaro | as long as it isn't the McDonald's syndrome, I guess it will be OK |
| 12:50.11 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177611872.dsl.bell.ca) |
| 12:50.11 | cpm | heh |
| 12:54.06 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 12:54.41 | coppice | McDonald's syndrome? you mean truly dead end jobs? |
| 12:56.57 | SteveTotaro | that would be Walmart Syndrome |
| 12:57.38 | SteveTotaro | while deadened, a McDonald's manager can make $100k + |
| 12:59.27 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:00.08 | *** join/#asterisk F (i=f@unaffiliated/f) |
| 13:03.35 | hi365 | is there an eta for 1.14.18? (or is going to be released based on bugs, etc.) |
| 13:03.37 | *** part/#asterisk F (i=f@unaffiliated/f) |
| 13:04.05 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 13:04.46 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 13:05.50 | ifnotwhynot | i am having problems loging agents onto asterisk ask me for a username(accepts info) ask me for password(accepts info) ask me for new extension ?????????????????????? this is where they loose me, in cli get this error Extension '101' is not valid for automatic login of agent '101'??????? |
| 13:05.54 | ZaVoid | morning |
| 13:06.03 | SteveTotaro | i thought all of 1.4 was released based on bugs |
| 13:06.37 | yang | When I am calling my mobile number from asterisk from the extension 059209586 (which should show this CALL-ID), I am seeing number 0338606057 http://openpaste.org/en/4978/ . I have specified callerid=059209586 ... |
| 13:06.47 | yang | I am using these extensions - http://openpaste.org/en/4979/ |
| 13:09.33 | kyron | mning |
| 13:11.26 | ifnotwhynot | yang do you have a wifi phone with softphone installed on it? |
| 13:11.46 | ifnotwhynot | looks like you are dialaing a sip phone |
| 13:11.55 | yang | its a handset |
| 13:12.06 | ifnotwhynot | sip handset? |
| 13:15.30 | yang | yes |
| 13:15.41 | cappiz | is it possible to manipulate the SIP From header? |
| 13:15.45 | *** join/#asterisk michael-i (n=michael-@141.41.40.226) |
| 13:16.46 | *** part/#asterisk elverkilde (n=jon@85.235.240.45) |
| 13:17.43 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 13:17.57 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:18.14 | ifnotwhynot | try this yang http://openpaste.org/en/4981/ |
| 13:18.42 | *** join/#asterisk saftsack (n=saftsack@p4FC7504A.dip.t-dialin.net) |
| 13:20.22 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 13:20.25 | *** join/#asterisk sob0l (n=sobol@devel4.net) |
| 13:20.37 | sob0l | is it possible to add trustrpid collumn to sippeers ? |
| 13:21.08 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:21.18 | Alan1234 | i have some firewall issues i think. |
| 13:21.30 | Alan1234 | i have a remote user who is trying to connect to our Asterisk server. |
| 13:22.05 | Alan1234 | he can make a call to us; and we can make call to him. |
| 13:22.10 | Alan1234 | it gets connected. |
| 13:22.25 | Alan1234 | HOWEVER ... only outgoing voice can be heard; nothing he says gets through us to |
| 13:22.33 | Alan1234 | [Feb 5 14:24:27] WARNING[25488]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 13:22.50 | ifnotwhynot | port 5060 must be opened |
| 13:22.56 | Alan1234 | on which side? |
| 13:22.59 | BBHoss | Alan1234, make sure your NAT settings in sip.conf are configured |
| 13:23.02 | ifnotwhynot | remote |
| 13:23.06 | BBHoss | before you do ANYTHING else |
| 13:23.10 | ifnotwhynot | both must be open |
| 13:23.15 | BBHoss | my port 5060 is not open and it works fine |
| 13:23.19 | Alan1234 | yes, ours is open here fine. |
| 13:23.39 | BBHoss | Alan1234, configure the localnet option and the externip option in sip.conf |
| 13:23.41 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 13:23.44 | ifnotwhynot | looks like you context can be wrong |
| 13:23.47 | Alan1234 | BBHoss: what should they be? he is behind a NAT, we are not. |
| 13:24.00 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:24.03 | BBHoss | so your asterisk server is not behind NAT? |
| 13:24.08 | Alan1234 | no. |
| 13:24.18 | Alan1234 | but our remote user is. |
| 13:24.25 | BBHoss | Alan1234, hang on a quick sec |
| 13:24.32 | Alan1234 | sure thing. thx |
| 13:24.36 | lirakis | ahh.. (snifff) 2.6.22-14 ... fresh... :) |
| 13:25.00 | *** join/#asterisk Psychobilly (n=Fuzz@athedsl-4404063.home.otenet.gr) |
| 13:25.04 | BBHoss | Alan1234, do you have nat=yes in sip.conf for your remote user's peer? |
| 13:25.10 | BBHoss | or friend |
| 13:25.36 | BBHoss | also, what type of phone is he using? |
| 13:25.40 | Psychobilly | does asterisk support codec re-negotiation during a call? especially when it has to hadle fax data? |
| 13:25.56 | Psychobilly | i searched a bit on the net but didnt find much |
| 13:26.05 | Psychobilly | im asking about asterisk 1.4 |
| 13:26.29 | BBHoss | Psychobilly, it probably means that it doesn't support it :) but you might ask in #asterisk-dev |
| 13:26.58 | Psychobilly | thx BBHoss i ll try there too :) |
| 13:27.00 | Alan1234 | BBHoss: right i have set it now to =yes and he's using a softphone zoiper |
| 13:27.05 | BBHoss | hmm |
| 13:27.14 | BBHoss | he is using SIP, correct? |
| 13:27.54 | Alan1234 | BBHoss: He's checking that now. |
| 13:27.57 | ifnotwhynot | can you get incoming calls from the user alan1234?? |
| 13:28.26 | Alan1234 | ifnotwhynot: yes, but he can only hear me. |
| 13:28.35 | BBHoss | what kind of router is it? |
| 13:29.12 | Alan1234 | router? at his side? |
| 13:29.16 | BBHoss | yeah |
| 13:29.31 | Alan1234 | not sure, but he can port-forward whatever is required to his desktop |
| 13:29.40 | Alan1234 | got 5060 port forwarded |
| 13:29.44 | BBHoss | well |
| 13:30.05 | BBHoss | the only bad thing is for rtp, you have to forward 10000-20000 by default |
| 13:30.22 | BBHoss | you can try IAX since its zoiper, it only require 4569 |
| 13:30.50 | BBHoss | also you might want to enable stun server if using SIP |
| 13:31.09 | BBHoss | http://www.astguru.com/tutorials/zoiper.html |
| 13:31.20 | ZaVoid | sjphone works great too for sip clients |
| 13:31.50 | Alan1234 | thanks chaps -- will let you know how we get on |
| 13:32.05 | cjk | hi, is there a variable that tells me if the channel is in t.38 or not ? |
| 13:33.15 | *** join/#asterisk AndyGraybeal_ (n=andy@node37.36.251.72.1dial.com) |
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| 13:40.09 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:40.10 | *** join/#asterisk wishbone__ (n=wishbone@189.70.21.39) |
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| 13:45.23 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 13:47.43 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 13:48.03 | nebojsajsimic | hi all again |
| 13:48.17 | ifnotwhynot | hi nebojsajsimic |
| 13:48.18 | nebojsajsimic | please for little help vith AGI |
| 13:48.29 | nebojsajsimic | I can't get $agivar['agi_extension']; |
| 13:48.33 | nebojsajsimic | like variable |
| 13:48.45 | nebojsajsimic | $pozvani = $agivar['agi_extension']; |
| 13:48.55 | nebojsajsimic | in PHP |
| 13:49.12 | nebojsajsimic | any help??? |
| 13:50.59 | *** part/#asterisk Titanous (n=Jon335@unaffiliated/titanous) |
| 13:51.48 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 13:53.48 | jmls | hi guys |
| 13:54.10 | jmls | is there any way of completely disabling the blind transfer in features.conf ? If I ; it out, it defaults to # |
| 13:55.59 | jmls | aha. you post a question, the answer pops into your head |
| 13:56.05 | jmls | blindtransfer => |
| 13:57.15 | lirakis | exit |
| 13:58.06 | jmls | no |
| 13:58.10 | jmls | ;) |
| 13:58.40 | SteveTotaro | blindtransfer => #*#* |
| 13:59.23 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1167863865.dsl.bell.ca) |
| 14:00.39 | jmls | SteveTotaro: just setting "blindtransfer =>" disables it entirely by the looks of things |
| 14:01.26 | SteveTotaro | yes, i always leave stuff in but make it hard to access for some reason |
| 14:02.19 | eric2 | I have an issue calling some local numbers where the phone service is provided by the local cable company, other numbers work well, anyone else experience this? |
| 14:02.33 | SteveTotaro | i guess because someone is always sure to ask for whatever is disabled at some point |
| 14:02.46 | eric2 | I get the following message: Everyone is busy/congested at this time (1:0/1/0) |
| 14:03.39 | SteveTotaro | eric2: how are you calling them |
| 14:04.24 | jmls | SteveTotaro: yeah, but we just had a problem where an inbound caller "accidently" triggered the blind transfer function. Couldn't find any way of ensuring an inbound outside channel couldn't access the transfer :( |
| 14:04.38 | eric2 | SteveTotaro: exten => _705NXXXXXX,1,Dial(SIP/${EXTEN}@vpri_gw1) |
| 14:04.58 | eric2 | it works for people with the traditional land lines |
| 14:05.09 | eric2 | but for those with cable type phones.. fast busy |
| 14:05.33 | eric2 | and it works for those on voip too |
| 14:05.37 | SteveTotaro | so vpri_gw1 sends calls out a pri? |
| 14:05.39 | *** join/#asterisk HeXeD (n=hex@87-194-8-43.bethere.co.uk) |
| 14:05.40 | eric2 | yes |
| 14:05.46 | eric2 | virtual pri |
| 14:05.56 | SteveTotaro | but they are your pstn connectivity |
| 14:05.57 | SteveTotaro | ? |
| 14:06.08 | SteveTotaro | you need to contact them and let them know |
| 14:06.10 | eric2 | yes, I do belive so |
| 14:06.11 | eric2 | ok |
| 14:06.14 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
| 14:06.21 | SteveTotaro | give them some example numbers |
| 14:06.22 | eric2 | so something on their end? |
| 14:06.31 | eric2 | ya, I sent them 2 numbers yesterday... |
| 14:06.38 | eric2 | is this normal? |
| 14:07.00 | eric2 | a number is a number and it should work... my provider is a clec |
| 14:07.03 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:07.04 | michael-i | when i pick up a ringing ZAP channel, that extension's number is played back in dtmf (phone 1111 rings, I pick up, I hear 1111 then the call connects). Has anyone heard of this? (zaptel-bsd, asterisk 1.4.17) |
| 14:07.41 | eric2 | thanks SteveTotaro |
| 14:09.11 | SteveTotaro | i had that issue with GXing on a T3 |
| 14:10.37 | eric2 | did it take them long to fix? |
| 14:11.06 | SteveTotaro | damn wifi stinks in the fog |
| 14:13.06 | SteveTotaro | yes, i gave them example numbers and they found something wrong in their call routing |
| 14:13.06 | SteveTotaro | they should be able to reproduce it on their end I assume |
| 14:14.49 | *** part/#asterisk jmls (n=jmls@81.138.42.77) |
| 14:15.41 | SteveTotaro | not long to fix once i convinced them to try a couple of example numbers |
| 14:17.41 | SteveTotaro | but that took a while |
| 14:17.41 | SteveTotaro | ITSPs are probably alot more responsive than huge monolithic companies like GXing |
| 14:22.39 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 14:22.44 | *** join/#asterisk bantu (n=Miranda@p54A330FC.dip0.t-ipconnect.de) |
| 14:22.47 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177649575.dsl.bell.ca) |
| 14:26.37 | yang | Do I need to setup codecs in zapata.conf ? |
| 14:26.51 | yang | I had some problems with that calls were deaf |
| 14:31.00 | *** join/#asterisk tobias (n=tobias@cpe-076-182-087-105.nc.res.rr.com) |
| 14:32.29 | synthetiq | no doces in zapata.conf |
| 14:32.33 | synthetiq | codecs |
| 14:33.17 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 14:33.19 | jeremy_g | aright |
| 14:33.58 | ifnotwhynot | ca one asterisk to work with mysql? |
| 14:35.24 | ifnotwhynot | yes micheal |
| 14:38.35 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:39.24 | jeremy_g | does * support PRACK |
| 14:39.45 | freezey | whats a good IP phone for testing? |
| 14:39.47 | freezey | cheap |
| 14:43.35 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:44.09 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:44.56 | RoyK | freezey: softphone? |
| 14:46.50 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:47.28 | jeremy_g | whatsChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority |
| 14:47.37 | jeremy_g | what do we mean by asynch? |
| 14:49.15 | mosty | is the global variable FORWARD_CONTEXT used in asterisk 1.2? it works for me in 1.4 but not sure if it's used in 1.2 |
| 14:49.45 | Corydon76-dig | jeremy_g: meaning the Goto happens when whatever is executing returns, not immediately |
| 14:51.23 | *** join/#asterisk postel_ (n=jp@wikimedia/Postel) |
| 14:51.53 | cappiz | is it possible to have different values for remote-party-id and the From in SIP-headers? |
| 14:53.39 | freezey | RoyK: na i want a physical phone like maybe cisco or polycom |
| 14:54.15 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 14:56.59 | *** join/#asterisk __genis-bcn__ (n=genis@nat/fluendo/x-4897368605e27b23) |
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| 14:59.19 | RoyK | freezey: grandstream have some rather cheap ones, but I don't know how good they are |
| 14:59.31 | RoyK | SNOMs are great, but more expensive |
| 14:59.39 | RoyK | you get what you're paying for..... |
| 15:00.20 | styelz | just get an ata |
| 15:00.23 | badcfe | hello i have this transcoding wildcard. the pdf doc talk about a "mode" variable.. where is that to be set eventually? |
| 15:02.20 | badcfe | grep mode /etc/zaptel.conf /etc/asterisk/zapata.conf gived me nothing |
| 15:02.40 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
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| 15:03.55 | mosty | ~grandstream |
| 15:03.57 | jbot | grandstream is probably the Yugo of VoIP hardware. Run. Run away now. |
| 15:06.28 | ZaVoid | lol |
| 15:06.33 | ZaVoid | i like my gxp-2000 |
| 15:07.13 | freezey | RoyK: how are the polycoms? or the cisco 7906G |
| 15:07.47 | freezey | this one cisco phone says its a Sip phone |
| 15:07.51 | freezey | so i guess that would be pretty good |
| 15:09.37 | freezey | mosty: how do you feel about the cisco 7906G SIP phone? |
| 15:10.21 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 15:10.47 | mosty | freezey, personally i would go for polycom or snom if you want a sip phone |
| 15:10.56 | ifnotwhynot | is there a free monitoring application for asterisk? |
| 15:11.23 | *** part/#asterisk Jerzyk (i=jerzyk@83.12.113.66) |
| 15:12.09 | ifnotwhynot | i u looking at pricce the snom is not a bad price for good quality, cisco a bit pricy |
| 15:12.22 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-230-170-165.dsl.irvnca.pacbell.net) |
| 15:12.52 | freezey | ifnotwhynot: yeah i mean i want to just spend like 300 for 2 phones for testing |
| 15:13.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:13.26 | ifnotwhynot | 300 $ use snom i use them work great |
| 15:14.31 | ifnotwhynot | does anyone know if there is a free monitoring application for asterisk? |
| 15:15.44 | mvanbaak | nagios |
| 15:15.52 | *** join/#asterisk Atkins (n=atkins@216.80.0.58) |
| 15:16.05 | kyron | freezey, polycom |
| 15:17.00 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
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| 15:22.00 | *** join/#asterisk ManxPower (n=manxpowe@251.sub-70-223-214.myvzw.com) |
| 15:26.19 | eric2 | how much ram is required to handle 100 simultaneous calls using g.729? |
| 15:26.35 | Qwell | eric2: ram wouldn't be the issue there |
| 15:26.47 | eric2 | processor speed? |
| 15:26.58 | Qwell | pretty much |
| 15:27.00 | eric2 | ok |
| 15:27.01 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 15:27.13 | Qwell | and the answer to your next question - it depends |
| 15:27.17 | eric2 | hehe |
| 15:27.41 | ManxPower | eric2: Your question is like "how much bandwidth do I need?" |
| 15:27.56 | eric2 | ya, that's another issue I need to figure out |
| 15:28.19 | eric2 | I see 711 takes about 40k or so |
| 15:28.34 | RoyK | g.711 takes 64kbps + ip + udp + rtp |
| 15:28.44 | Qwell | eric2: it's about 80k |
| 15:28.45 | RoyK | with 20ms packetization, that is 80kbps |
| 15:28.51 | ZaVoid | eric2: i have 100 simultaneous calls on my boxes now with g729 |
| 15:28.56 | ZaVoid | but i'm not transcoding all of those |
| 15:29.03 | RoyK | with 40ms packetization, it's 72kbps |
| 15:29.15 | eric2 | so some you run at 711, some at 729 I guess |
| 15:29.47 | eric2 | as for getting 729 setup, the only way is through digium? |
| 15:29.50 | ZaVoid | no some come in g729 so i don't transcode them because all my calls go out g729 |
| 15:30.04 | eric2 | ah, interesting :) |
| 15:30.04 | kyron | eric2, http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 15:30.06 | ZaVoid | if calls come in at a different codec.. say g711.. i transcode them to g729 |
| 15:30.16 | ZaVoid | or if they come g723 i send them to a transcoding box first |
| 15:30.58 | eric2 | so if a call comes in on 729, your server doesn't have to do any conversions, you just pass them on as 729 to the end user.. correct? |
| 15:31.03 | ZaVoid | correct |
| 15:31.12 | ZaVoid | assuming the far end allows g729 |
| 15:31.26 | ZaVoid | just make sure you have allow=g729 |
| 15:31.38 | eric2 | I'm setting the up to use g729 with availability to use 711 if required |
| 15:31.43 | eric2 | *them |
| 15:31.44 | ZaVoid | most of my configs are disallow=all then allow=g729 |
| 15:32.17 | freezey | ifnotwhynot: which phones do you use cause i will just get the same? and are they out of the box ready? or no |
| 15:33.09 | kyron | Q: About transcoding: if 2 callers in a conference are 729 and 5 are 711, are the 2 calls transcoded to 729 or will I have to pay 7 licenses to transcode the conference to 729. I guess the other way of putting this, does * "upgrade" the codec or does it use the lowest common denominator? |
| 15:33.18 | eric2 | linksys ip spa942's is what I'm using |
| 15:33.58 | *** join/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 15:35.31 | *** part/#asterisk __genis-bcn__ (n=genis@nat/fluendo/x-4897368605e27b23) |
| 15:36.04 | freezey | eric2: are they good? and did the come out of the box ready to go? |
| 15:36.44 | eric2 | ya, easy to configure... I'm happy w/them |
| 15:36.53 | freezey | nice nice |
| 15:36.58 | freezey | gonna take a look at these as well |
| 15:37.01 | freezey | whos your VOIP provider? |
| 15:37.04 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 15:37.09 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 15:37.21 | fiXXXerMet | Can I change the conference recording location? |
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| 15:43.48 | Navion | Anyone had any trouble being able to break dial tone with Sangoma FXS's? It seems like some DTMF digits are not being recognized while dial tone is present. Once the dial tone is broken all the digits are recognized, even the ones the card wouldn't decode with dial tone present. |
| 15:46.09 | kyron | Navion, could it be VAD related? |
| 15:46.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:48.22 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 15:48.23 | teknoprep | hey all |
| 15:48.38 | teknoprep | i have some questions about g729 and dtmfmodes |
| 15:49.11 | teknoprep | i am trying to find out why i can not either have both voicemail digit recognition or outbound to real world recognition. |
| 15:49.27 | ManxPower | teknoprep: no questions needed. INBAND DTMF does not work in compressed codecs like G729. RFC2833 is the recommended DTMF mode, INFO should only be used as a last resort. |
| 15:49.50 | teknoprep | ManxPower, well if i set my dtmfmode to rfc2833 |
| 15:50.11 | teknoprep | ManxPower, i also use the g729 on all outbound calling. and ulaw on the internal network |
| 15:50.39 | teknoprep | ManxPower, i have this problem.. i tested this out by calling my cell phone... when i hit a button on the phone.. i hear it as if it was pressed 6 times |
| 15:50.46 | ManxPower | teknoprep: For ulaw, I actually suggest INBAND, then use RFC2833 for the PSTN legs |
| 15:51.01 | teknoprep | ManxPower, what about the phone setting? |
| 15:51.11 | ManxPower | teknoprep: remember Asterisk's RFC2833 support is...not...the greatest. |
| 15:51.11 | teknoprep | ManxPower, keep it on inband also.. i am using the cisco 7940 |
| 15:51.22 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-230-170-165.dsl.irvnca.pacbell.net) |
| 15:51.43 | teknoprep | ManxPower, i am going nuts with this.. everything works great with ULAW and inband dtmf |
| 15:51.58 | teknoprep | ManxPower, i am trying to switch to g729 so i can squeeze more calls onto a T1 |
| 15:52.00 | ZaVoid | is there bugs with asteirsk rfc2833 manx? |
| 15:52.24 | ManxPower | teknoprep: when you call the PSTN your phone is NOT calling the PSTN. The phone is calling Asterisk, think of that as the FIRST call. Then Asterisk calls the PSTN (sounds to me you are using an ISTP), that call can use a different DTMF mode. |
| 15:52.46 | teknoprep | ManxPower, i understand this |
| 15:53.07 | ManxPower | ZaVoid: I don't know exactly, I just hear people complaining about it and some carriers like Level 3 don't even work with RFC2833 |
| 15:53.13 | ManxPower | teknoprep: so what is the problem. |
| 15:53.18 | ZaVoid | ooo you know what |
| 15:53.24 | ZaVoid | since you mention level3 i do have DTMF issues with them |
| 15:53.36 | ManxPower | <-- smarter than he looks |
| 15:53.37 | ZaVoid | sometimes it works and sometimes i get the dreaded "user entered nothing" |
| 15:53.43 | ZaVoid | i love that |
| 15:54.01 | ManxPower | ZaVoid: If you are able to, use ulaw for your L3 connection and INBAND |
| 15:54.04 | *** join/#asterisk phsdshft (n=phsdshft@204.56.88.231) |
| 15:54.28 | teknoprep | ManxPower, istp is an internet phone provider ? |
| 15:54.35 | ZaVoid | http://bugs.digium.com/view.php?id=10058 |
| 15:54.37 | ZaVoid | i see it |
| 15:54.56 | ManxPower | ~itsp |
| 15:54.57 | jbot | [~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 |
| 15:55.38 | teknoprep | ManxPower, thats what i thought.. and yes i am using ISTP |
| 15:55.51 | teknoprep | ManxPower, its bandwidth.com |
| 15:56.39 | teknoprep | ManxPower, the problem is when i use compression codec ... and call someone... when a digit is pressed it sounds like its being pressed 6 times |
| 15:57.08 | teknoprep | ManxPower, so when i dial a phone number... it connects me. i get connceted to an IVR that says please press the number of the extension i would like |
| 15:57.13 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 15:57.15 | ManxPower | teknoprep: G729, I assume. What DTMF mode? |
| 15:57.31 | teknoprep | ManxPower, i press 1 - 0 - 2 ... and it does.... 1 1 1 1 1 1 0 0 0 0 0 0 2 2 2 2 2 2 |
| 15:57.35 | teknoprep | rfc2833 |
| 15:57.49 | teknoprep | ManxPower, if i use info it doesn't even work |
| 15:57.50 | ManxPower | teknoprep: post the sip.conf entries for both the provider and the phone |
| 15:58.27 | phsdshft | Hi. When I'm using a SIP connection to Broadvoice through a Checkpoint Firewall, After approximately ~180 seconds, I cannot make further outbound calls (they fail,) although I remain registered. What settings can resolve this (maxexpirey, defaultexpirey, rtptimeout,rtpholdtimeout..) |
| 15:58.38 | teknoprep | ManxPower, well i would have to reconfigure the system... right now its up and running and i can't change it.. let me make it up as if i was going to set it up.. but it will take a bit.. i had to change them back to ULAW before they opened shop today or they would have had alot of problems |
| 15:58.54 | *** join/#asterisk Bourrelle (n=Bourrell@132.207.156.100) |
| 15:59.28 | Bourrelle | Hello 1 quick question |
| 15:59.42 | ManxPower | teknoprep: just show me the entries |
| 15:59.50 | Bourrelle | is theres a conversation between 2 endpoints by an asterisk server |
| 16:00.15 | Bourrelle | and one of them put the call onHold |
| 16:00.40 | Bourrelle | does asterisk forward this "SIP event" to the other endpoint |
| 16:00.56 | Bourrelle | SIP event = onHold event |
| 16:00.58 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:01.47 | *** join/#asterisk neonerz (i=d1dc7757@gateway/web/ajax/mibbit.com/x-d79cf7a6cf10ea96) |
| 16:02.08 | ManxPower | Bourrelle: Asterisk NEVER forwards ANYTHING except for audio between the two endpoints. |
| 16:02.20 | ManxPower | SIP proxies forward events. Asterisk is not a SIP proxy. |
| 16:02.20 | neonerz | thats the new commands in sip.conf for 1.4 that allows DTMF to work with 1.2? |
| 16:02.24 | neonerz | a link would be nice |
| 16:02.39 | *** join/#asterisk arthurlutz (n=arthur@logilab2-7-50.cnt.nerim.net) |
| 16:02.47 | ManxPower | neonerz: there's nothing in sip.conf.example? |
| 16:02.52 | neonerz | nah |
| 16:02.59 | neonerz | I'm using my old sip.conf from 1.2 |
| 16:03.12 | neonerz | I made a backup of all the configs before hand |
| 16:03.18 | ManxPower | neonerz: then look at 1.4's sip.conf.example. Did you even read upgrade.txt ???? |
| 16:03.22 | neonerz | but somehow sip.conf got corrupted |
| 16:03.32 | neonerz | ahhh sip.conf.example |
| 16:03.42 | neonerz | yea |
| 16:04.34 | neonerz | I didn't upgrade btw |
| 16:04.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:04.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:05.03 | ManxPower | neonerz: read upgrade.txt even if you didn't upgrade your box. You are trying to upgrade your BRAIN with 1.4 info. |
| 16:06.31 | neonerz | I went through everything before the upgrade (thats how I knew about the command addtion) |
| 16:06.35 | neonerz | I just couldn't remember the commands |
| 16:07.22 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:07.22 | neonerz | sip.conf.example <-kick started my brain though |
| 16:07.31 | ManxPower | Honestly, I'm not aware of anything having to do with DTMF between 1.2/1.4 |
| 16:08.29 | file | neonerz: find it yet? |
| 16:09.05 | Bourrelle | ManxPower is there a way for an endpoint to know that he is onHold ?? |
| 16:09.14 | Bourrelle | is asterisk doesnt forward anything except audio |
| 16:09.18 | Bourrelle | if* |
| 16:09.30 | file | currently no. |
| 16:09.48 | file | it would be possible to forward that through... but nobody has written it |
| 16:10.29 | *** join/#asterisk ddunavant (n=David@pool-96-231-69-97.washdc.east.verizon.net) |
| 16:10.41 | neonerz | ManxPower: rfc2833compensate=yes <---- |
| 16:11.03 | ManxPower | neonerz: you're so smart |
| 16:11.05 | ManxPower | ! |
| 16:11.06 | Bourrelle | but asterisk must at least forward the SIP INVITE |
| 16:11.11 | file | Bourrelle: why? |
| 16:11.14 | ManxPower | Bourrelle: why? |
| 16:11.16 | neonerz | ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. |
| 16:11.24 | Bourrelle | how cant he forward another SIP commands |
| 16:11.37 | neonerz | I only have ONE client that connects to me with a asterisk |
| 16:11.38 | ManxPower | Bourrelle: Asterisk is considered a SIP endpoint. |
| 16:11.57 | file | Asterisk is a PBX... when you put someone on hold what do you expect? you expect that they hear your hold music... that is what happens, the INVITE is not forwarded on... Asterisk simply starts playing music to them |
| 16:12.41 | Bourrelle | ok Imagine Client1 - Asterisk - Client2 |
| 16:13.08 | ManxPower | Bourrelle: NO SIP PBX forwards SIP packets. |
| 16:13.15 | Bourrelle | Client1 wants to make an audio session with Client 2 |
| 16:13.52 | Bourrelle | he send an Invite with his SDP for codec negotiation |
| 16:13.54 | file | it doesn't, it makes a connection to Asterisk and then Asterisk makes a connection to Client 2... Asterisk sits in the middle, also known as a B2BUA (back to back user agent) |
| 16:14.03 | Bourrelle | but your saying that Client 2 doenst receive it |
| 16:14.05 | file | you aren't making a direct connection. |
| 16:14.12 | Bourrelle | how can he negociate his codecs ? |
| 16:14.25 | ManxPower | Bourrelle: no, asterisk generates it's own INVITE and sends THAT to the other end. |
| 16:14.38 | Bourrelle | ok perfect |
| 16:14.47 | Bourrelle | thx alot guys very appreciated |
| 16:14.52 | ManxPower | Bourrelle: the phones are not negotiating codecs with each other, they are doing that with the PBX. |
| 16:15.09 | ManxPower | Bourrelle: if you want SIP packets to be forwarded, use a SIP proxy like SER (SIP Experess Router) |
| 16:15.43 | Qwell | file: is that a term that we made up? O.o |
| 16:15.51 | Qwell | or are there other B2BUAs? |
| 16:15.54 | file | Qwell: wazzat? B2BUA? nosir |
| 16:16.13 | Qwell | or is that one of those things that only really exist in RFCs? |
| 16:16.27 | file | it exists in a world of dreams. |
| 16:16.56 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:17.52 | phsdshft | Hi. When I'm using a SIP connection to Broadvoice through a Checkpoint Firewall, After approximately ~180 seconds, I cannot make further outbound calls (they fail,) although I remain registered. What settings can resolve this (maxexpirey, defaultexpirey, rtptimeout,rtpholdtimeout..) |
| 16:19.36 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:20.33 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 16:20.40 | file | phsdshft: define "they fail"... do you have a sip debug? |
| 16:21.38 | phsdshft | yes. it looks like it is sending out INVITE's but not getting any response |
| 16:22.43 | file | your firewall might have a low time for dropping UDP mappings |
| 16:22.53 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:23.40 | file | do you forward ports or rely on Broadvoice's NAT stuff? |
| 16:23.58 | eric2 | for outgoing calls, if my sip provider gives me gw1 and gw2 in order for me to forward my calls to, what's the best setup for fail over in case gw1 goes off line as outbound calls should then be routed through gw2..? |
| 16:24.27 | file | eric2: two peer entries with qualify=yes to monitor the two servers, then dialplan logic that does failover |
| 16:25.02 | phsdshft | @file: It goes through a checkpoint firewall which is SIP aware.. it looks like it actually reads the SIP headers and makes decisions based on that.. |
| 16:25.04 | file | eric2: you could also use the RAND dialplan function I suppose to spread across the two servers as well... |
| 16:25.18 | phsdshft | if I changed the defaultexpirey (I think thats the one) it increased the timeout value |
| 16:25.29 | phsdshft | but I was hoping there is another setting that I can use? |
| 16:25.37 | phsdshft | in asterisk.. to increase the session length or whatever |
| 16:25.43 | phsdshft | or have it send keepalives of some type |
| 16:26.00 | eric2 | I think I"ll try to spread out the load and do some dial plan logic... tx file |
| 16:26.12 | ZaVoid | ooo load average 41.02 |
| 16:26.14 | ZaVoid | thats fun |
| 16:26.37 | eric2 | looks like a crazy load to me |
| 16:26.37 | ZaVoid | wonder what the hell is causing that |
| 16:28.14 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 16:31.50 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:32.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:33.08 | *** mode/#asterisk [+o putnopvut] by lmadsen |
| 16:33.11 | lmadsen | mwahahahaha |
| 16:33.40 | lmadsen | putnopvut is now free to answer all questions you may have in a private chat window |
| 16:33.52 | putnopvut | hardy har har |
| 16:39.21 | *** join/#asterisk noneo (n=artur@82.152.82.150) |
| 16:40.08 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 16:44.23 | Bourrelle | Can I ask one more thing |
| 16:44.31 | Qwell | Bourrelle: you just did |
| 16:44.55 | Bourrelle | does asterisk works the same way that a Cisco CallManager works ? |
| 16:45.14 | Qwell | what do you mean by "same way"? |
| 16:45.41 | mosty | Bourrelle, they work very differently internally |
| 16:45.46 | Bourrelle | hehe above they told me that asterisk doesnt forward any SIP call |
| 16:45.50 | Bourrelle | for exemple |
| 16:45.59 | Bourrelle | Client1 - asterisk - Client2 |
| 16:46.04 | mosty | Bourrelle, you can configure asterisk to forward sip calls |
| 16:46.07 | Bourrelle | client send an onHold messagr to asterisk |
| 16:46.13 | ManxPower | Bourrelle: no, it does not directly forward the SIP PACKETS. |
| 16:46.15 | Bourrelle | asterisk doesnt inform Client and just start playing music |
| 16:46.23 | Bourrelle | do you think that a cisco callmanager qworks the samwe way |
| 16:46.27 | mosty | Bourrelle, have you read the book? |
| 16:46.29 | mosty | ~book |
| 16:46.29 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:46.42 | ManxPower | Bourrelle: I don't know for sure, but that is pretty much the standard design of VoIP PBXs |
| 16:46.55 | ManxPower | Bourrelle: I suspect you think this is a problem. It is not a problem. |
| 16:46.57 | Bourrelle | no thx ill read it |
| 16:47.30 | file | if it was a problem tons of people would be rebelling |
| 16:48.32 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 16:49.07 | _Krieger_ | what is the condition[s] when CLI 'restart now' is not working? |
| 16:53.33 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 16:53.51 | lunaphyte_ | broken? |
| 16:54.24 | file | Asterisk could be deadlocked |
| 16:55.27 | twisted | it could be bound and gagged |
| 16:56.21 | Corydon76-dig | Compile with DONT_OPTIMIZE and DEBUG_THREADS and when that happens, run a 'core show locks', copy the output to pastebin, and signal us here |
| 16:56.46 | Qwell | it could be in a contention wait |
| 16:56.51 | Qwell | (I crack myself up) |
| 16:57.20 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 16:58.02 | *** part/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 16:59.11 | Bourrelle | mosty |
| 16:59.25 | lunaphyte_ | are the objects that entries in sip.conf and iax.conf define referred to as channels? |
| 16:59.33 | Bourrelle | how can I make it forwards SIP call |
| 16:59.37 | Bourrelle | never heard of that |
| 16:59.41 | mosty | Bourrelle, read the book |
| 16:59.49 | Bourrelle | kk |
| 17:03.58 | file | lunaphyte_: do you mean like peers and users? |
| 17:04.54 | lunaphyte_ | yes |
| 17:05.26 | file | they are not channels... they are entries used for authenticating inbound calls and placing outbound calls |
| 17:05.46 | lunaphyte_ | what is a channel? |
| 17:06.03 | ManxPower | lunaphyte_: A channel is one leg of a call for VoIP. |
| 17:06.11 | ManxPower | heck it is for PSTN too. |
| 17:06.27 | Qwell | a "channel" needn't even go anywhere |
| 17:07.35 | lunaphyte_ | so, for example, the media stream between a sip phone and whatever is at the other end of that stream, be it asterisk, or some other phone? |
| 17:08.24 | Qwell | a channel doesn't necessarily need to have media |
| 17:08.32 | lunaphyte_ | oh, no? |
| 17:08.38 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 17:08.42 | teknoprep | i am at this dental office |
| 17:08.50 | teknoprep | and i am having problems with DTMF and g729 |
| 17:09.17 | teknoprep | when i press a button on an outbound call... say i hit 1... it actually does 1 1 1 1 1 1 |
| 17:11.20 | mosty | teknoprep, what dtmf mode are you using? |
| 17:12.26 | *** join/#asterisk micander (n=Michael_@Full-Service-Travel-1157986.cust-rtr.pacbell.net) |
| 17:12.27 | teknoprep | ulaw is the codec on the LAN with inband dtmf |
| 17:12.36 | teknoprep | g729 on outbound calls with rfc2833 |
| 17:12.46 | lunaphyte_ | if i have a sip device with an fxo port that is registered w/ asterisk, and i'm sending extensions that start with certain prefixes out through it, would it be accurate to refer to that connection to the pstn as a "channel"? |
| 17:13.26 | mosty | lunaphyte, it's a sip channel from asterisk's point of view |
| 17:13.31 | ManxPower | lunaphyte: Only if you were too lazy to use the correct terms. |
| 17:15.23 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 17:16.01 | lunaphyte_ | well, i hope i'm not lazy - but i do know i'm uninformed. hopefully i can change that a bit. |
| 17:16.29 | lunaphyte_ | ManxPower: what would the correct terms be? |
| 17:19.09 | teknoprep | mosty any idea ? |
| 17:19.48 | mosty | teknoprep, try dtmfmode=auto |
| 17:20.13 | mosty | and perhaps dtmfmode=info on the lan side |
| 17:20.36 | teknoprep | mosty if i set dtmfmode=info on the lan side... i can not access voicemaiil |
| 17:20.37 | ManxPower | lunaphyte: well the sip.conf entry for the device would be called "peer" or device". The actual call between the asterisk and the ATA is one channel, then the call from the ATA to the PSTN would be a 2nd channel. |
| 17:21.30 | lunaphyte_ | ok, i understand. |
| 17:21.32 | mosty | teknoprep, can you set your phone to use SIP info for dtmf? |
| 17:21.51 | neonerz | ManxPowerlunaphyte: Only if you were too lazy to use the correct terms. <-awsome |
| 17:22.29 | teknoprep | mosty i have a cisco 7940 |
| 17:22.45 | mosty | teknoprep, good luck figuring that one out then :) |
| 17:22.47 | teknoprep | mosty # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) |
| 17:22.58 | ManxPower | lunaphyte_: Hardcore PBX people (and VERY weirdly enough GUI people) sometimes call it a route as well. |
| 17:23.09 | ManxPower | avt is RFC2833 |
| 17:23.11 | mosty | teknoprep, i can't help with cisco phones, sorry |
| 17:23.48 | neonerz | wouldn't route be more of a TDM term |
| 17:24.36 | teknoprep | ManxPower shoudl i set the phone to be AVT_ALWAYS |
| 17:24.52 | teknoprep | ManxPower then the asterisk ext to set dtmfmode=rfc2833 |
| 17:24.54 | ManxPower | teknoprep: I would assume so. |
| 17:24.58 | teknoprep | ManxPower then the outbound to rfc2833 |
| 17:25.02 | SomethingISOdd | anyone here using a2billing |
| 17:25.03 | teknoprep | ManxPower i'll try that |
| 17:25.08 | ManxPower | teknoprep: but everyone I've ever heard of using Cisoc phone did not have to screw with the DTMF mode on the phone |
| 17:25.39 | neonerz | doesn't cisco phones do rfc2833 by default? |
| 17:25.51 | teknoprep | ManxPower are they using g729 on there outbound channels |
| 17:26.14 | teknoprep | ManxPower this is the problem i am having... internal is fine. its going out over g729 . |
| 17:26.25 | neonerz | I know its not inband or info by default |
| 17:26.36 | neonerz | tekno are you using ast 1.2? |
| 17:26.39 | teknoprep | 1.4 |
| 17:26.43 | neonerz | oh |
| 17:26.44 | teknoprep | 1.4.17 |
| 17:26.48 | neonerz | that blows that thought away |
| 17:27.45 | neonerz | I have a bunch of 7940-60-70's with 1.4 and leaving cisco's default (which is rfc2833) seems to work without any changes |
| 17:28.09 | neonerz | its the outbound trunk that has a problem reading the DTMF? |
| 17:28.11 | *** join/#asterisk saftsack (n=oliver@p4FC7504A.dip.t-dialin.net) |
| 17:28.22 | teknoprep | i have no idea at this point |
| 17:28.37 | saftsack | hi, is a compact flash card fast enough for using asterisk? |
| 17:28.39 | mvanbaak | I use chan_skinny and it works fine :) |
| 17:28.44 | mvanbaak | saftsack: yes |
| 17:29.24 | neonerz | I came in pretty late, what exactly is the problem? |
| 17:29.58 | teknoprep | on internal dtmf with ulaw everything works great |
| 17:30.05 | saftsack | mvanbaak, so i can do queue and voicemail without using a ramdisk? |
| 17:30.23 | teknoprep | but on outbound over g729 dtmf is crazy... i call my cell phone per say to test this out... i press 1 on the cisco phone... it actually does 1 1 1 1 1 1 |
| 17:30.49 | neonerz | have you tried with a different phone? |
| 17:30.50 | mvanbaak | saftsack: depends. if you do recording of the queues and you want to push a lot of queue calls it will wear out the flashdisk pretty fast |
| 17:31.08 | mvanbaak | saftsack: but queue calls without recording should be no issue |
| 17:31.09 | neonerz | have you tried using 729 internally to see if it has the same problem? |
| 17:31.19 | teknoprep | neonerz yeah i have a few old polycom phones here also |
| 17:31.25 | ManxPower | teknoprep: that is a classic problem with DTMF mode to the carrier. |
| 17:31.26 | teknoprep | neonerz yes same problem |
| 17:31.27 | saftsack | mvanbaak, there aren't more than 2 similar voice recordings |
| 17:31.40 | ManxPower | but since you refuse to pastebin the relevant entries, I can't help you anymore. |
| 17:31.41 | teknoprep | ManxPower so you are saying it could be the carrier ? |
| 17:31.46 | neonerz | teknoprep: have your provider cap the RTP traffic to see what they ssee |
| 17:31.52 | teknoprep | ManxPower well i had to come into the shop. |
| 17:32.01 | file | teknoprep: adding dtmf to your console line in logger.conf and doing logger reload will tell you what Asterisk sees coming from a device, if it's RFC2833 going out then rtp debug will show the packets going out |
| 17:32.01 | mvanbaak | you might want to look into mounting some nfs share for your call recordings and mount the CF disk read-only |
| 17:32.02 | teknoprep | ManxPower i will pastebin now.. i was at home when i first talked about this |
| 17:32.17 | file | so if the DTMF coming from your device looks fine then you just eliminated half the possible problem... |
| 17:32.36 | ManxPower | I am SO glad we never have DTMF problems. |
| 17:32.37 | neonerz | file: he said internally DTMF seems to work - its just when he sends the call to his provider |
| 17:32.44 | teknoprep | @file dtmf on lan should be fine.. i can access voicemail np |
| 17:32.57 | neonerz | we had a ton of DTMF problems with ast 1.2 and nortal's new VSP cards |
| 17:32.59 | teknoprep | @file also when i call ext - ext and i press buttons they sound fine on the other end |
| 17:33.09 | neonerz | nortels* |
| 17:33.09 | file | I would bet on the carrier then as well |
| 17:33.40 | neonerz | teknoprep: have your carrier cap the RTP stream to see what they get for DTMF |
| 17:34.27 | teknoprep | neonerz i am using bandwidth.com they are supposted to be pretty good... but let me call them |
| 17:34.34 | neonerz | Verizon started upgrading all their nortel media gateway's with the newest firmware for the VSP cards, and one by one customers were breaking |
| 17:34.51 | file | DTMF doesn't really have to be that hard... |
| 17:35.03 | neonerz | I only ask that because you might see everything as fine, but they might see something you dont |
| 17:35.13 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 17:35.40 | Qwell | neonerz: L3 had the same "problem", for the same reason, I believe |
| 17:35.50 | neonerz | like before I knew there was a problem with rfc2833 and ast 1.2, it took us and Verizon weeks to figure it out |
| 17:36.07 | neonerz | Qwell: Nortel said its not a problem - Asterisk was the problem |
| 17:36.11 | neonerz | which I guess was true |
| 17:36.13 | Qwell | neonerz: they're right |
| 17:36.18 | file | sort of right. |
| 17:36.19 | neonerz | yea |
| 17:36.19 | Qwell | but it's been fixed in 1.4 |
| 17:36.24 | neonerz | sort of* |
| 17:36.26 | neonerz | exactly |
| 17:36.26 | Qwell | file: well...heh |
| 17:36.33 | neonerz | since it works with Nortel's older VSP cards |
| 17:36.33 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 17:36.41 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:36.43 | neonerz | it was just the latest and geatest it didn't work with |
| 17:36.47 | file | I still say looking at the actual amount of time between the begin and the end versus the duration value in the actual RFC2833 packet is silly |
| 17:36.49 | file | but that's just me |
| 17:36.53 | teknoprep | i'll be back laster |
| 17:37.05 | neonerz | file: doesn't matter what we think |
| 17:37.10 | mamep | how can i enable call forwarding? |
| 17:37.17 | neonerz | *78 |
| 17:37.24 | neonerz | *72 |
| 17:37.34 | file | mamep: without knowing what device/phone you are asking about, can't be answered with certainty |
| 17:37.55 | mamep | hmm |
| 17:37.57 | mamep | x-lite |
| 17:38.09 | neonerz | doesn't x-lite do the call forwading itself? |
| 17:38.22 | file | I don't know if x-lite supports it... might have been one of the features they took out |
| 17:38.32 | neonerz | no i was wrong |
| 17:38.34 | neonerz | your right |
| 17:38.38 | neonerz | it does do DND though |
| 17:38.44 | mamep | ? |
| 17:38.49 | mamep | do not distrub? |
| 17:38.53 | neonerz | yea |
| 17:39.05 | file | mamep: since X-Lite doesn't support it you will have to write dialplan logic in Asterisk to do call forwarding |
| 17:39.09 | file | (if you want it) |
| 17:39.10 | neonerz | mamep: some phones will let you do call forwarding directly from it |
| 17:39.16 | neonerz | some you have to build an app |
| 17:39.27 | mamep | yeah it's better to make it through asterisk |
| 17:39.31 | mamep | any guidance available? |
| 17:39.44 | file | this can be accomplished using the DB dialplan function and the Read dialplan application, plus setvar in sip.conf or the caller ID number |
| 17:40.26 | saftsack | mvanbaak, do you have a cf * server? |
| 17:40.26 | neonerz | yea |
| 17:40.34 | mvanbaak | saftsack: yes |
| 17:40.37 | file | plus the GotoIf dialplan application I suppose |
| 17:40.49 | neonerz | just have it write to the DB that the exten number is forwarded to whatever, then make it check the DB on inbound calls to that exten |
| 17:41.11 | saftsack | mvanbaak, do you do logging on a ramdisk/nfs partition or directly on flash? |
| 17:41.56 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 17:42.04 | mvanbaak | saftsack: remote syslog daemon |
| 17:42.35 | neonerz | mamep: http://pastebin.com/m7f1a9a47 <-the extension.conf app |
| 17:42.38 | mvanbaak | and CDR logs are stored using cdr_adaptive_odbc |
| 17:43.07 | mvanbaak | brb, have to go to trainstation to pickup wife there |
| 17:43.13 | saftsack | so i take a normal linux, and the only thing which is written very often are the logs or do i have to do some other special things so that the cf card isnt often written? |
| 17:43.16 | mamep | neonerz : aserisk 1.4? |
| 17:43.20 | neonerz | yea |
| 17:43.36 | neonerz | http://pastebin.com/m48eeffed <-how I check for it on my incoming trunk |
| 17:43.44 | neonerz | your probably not going to want to use my way |
| 17:44.22 | neonerz | on the cfon and off app |
| 17:44.28 | neonerz | you could leave out the first include |
| 17:44.34 | neonerz | I stole the app from FreePBX |
| 17:44.43 | *** join/#asterisk d-k-t (n=dt@125.120.139.9) |
| 17:45.18 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 17:45.30 | neonerz | mamep: I send all inbound calls to custom-check-cf and it rolls from there |
| 17:45.44 | neonerz | mind you though, that app sends the call out of the box |
| 17:54.31 | file | quite quiet... |
| 17:56.03 | Corydon76-dig | like the quiet before a storm? |
| 17:56.52 | file | perhaps |
| 17:57.35 | Corydon76-dig | <cue song> |
| 18:01.16 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 18:01.47 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 18:03.39 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:08.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:16.42 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:18.20 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-126-7-146.bflony.east.verizon.net) |
| 18:19.24 | *** join/#asterisk zobia (n=laurashr@222.212.67.44) |
| 18:19.53 | zobia | Qwell:good afternoon! |
| 18:20.17 | zobia | Qwell: i got a request from my carrier about the 200 ok message |
| 18:20.30 | zobia | Qwell: but i don't know how to change my sides's 200 ok message |
| 18:20.39 | zobia | Qwell: Here is what he said "Now in your 200 OK answer you are sending m=audio 41098 RTP/AVP 0 8 101. You need to send m=audio 41098 RTP/AVP 0. Means please do not send "101". |
| 18:20.59 | zobia | Qwell: can you help me to understand what he need me to change? |
| 18:22.10 | file | zobia: set dtmfmode to info or inband. |
| 18:22.23 | file | 101 is RFC2833 DTMF |
| 18:22.42 | file | and 8 is alaw in case you didn't know |
| 18:22.57 | zobia | @file:thank you very much |
| 18:23.14 | SuPrSluG | hi all, i'm getting heavy static on my phones. what generally causes static? |
| 18:24.07 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:24.42 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:26.33 | file | SuPrSluG: always? |
| 18:26.48 | SuPrSluG | just recently |
| 18:27.14 | file | I meant if you call remote devices, prompts, etc... |
| 18:28.04 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 18:28.23 | *** join/#asterisk joe (n=nnnnnnnn@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 18:33.04 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 18:35.10 | *** join/#asterisk tobias (n=tobias@cpe-076-182-087-105.nc.res.rr.com) |
| 18:36.50 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:40.54 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 18:41.27 | XnOSX | ast_rtp_read: RTP Read too short (WHY??????) |
| 18:42.40 | file | XnOSX: unknown... the remote device could be sending some sort of proprietary keep alive packet on the same port as the RTP... |
| 18:42.47 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:43.01 | file | the data read in simply was not big enough to be an RTP packet |
| 18:45.08 | tzafrir | $ lsb_release -i |
| 18:45.08 | tzafrir | Distributor ID: Debian |
| 18:45.24 | tzafrir | Sadly lsb_release is not guaranteed to be intalled everywhere... |
| 18:46.32 | XnOSX | im triying recive calls in my asterisk pbx server but for any reason when answer near phone grandstream the call is down |
| 18:48.05 | XnOSX | file: take a look http://pastebin.ca/892813 |
| 18:48.56 | file | you are doing T38? |
| 18:49.40 | *** join/#asterisk tobias (n=tobias@cpe-076-182-087-105.nc.res.rr.com) |
| 18:50.22 | XnOSX | file: no |
| 18:50.34 | XnOSX | i would like onlye send and recive calls |
| 18:50.56 | file | XnOSX: apparently this disagrees... disable it on your grandstream? |
| 18:51.01 | XnOSX | but for any reason this is the mensage in the debug sip mode in a CLI console |
| 18:51.40 | XnOSX | file: ummmm i dont know let me see |
| 18:52.54 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:53.46 | Daviey | jameswf: Why do you feel the need to tell everyone where you are? |
| 18:54.17 | file | play nice! |
| 18:55.20 | jameswf | because when i have -home it says dont call my boss..... otherwise I am on wok time |
| 18:55.33 | jameswf | s/wok/work |
| 18:56.13 | XnOSX | file: no this phone havent a T38 configurationç |
| 18:56.28 | file | something is doing T38. |
| 18:56.51 | XnOSX | :S |
| 18:58.46 | XnOSX | own telecom provider send calls with T38 or AUDIO, and they cant change these |
| 18:59.50 | file | you can try t38pt_udptl=no in sip.conf |
| 18:59.58 | file | in the general section |
| 19:00.10 | XnOSX | ummm ok let me see |
| 19:02.42 | lmadsen | atis_work: that is normal -- they are two separate channels. You need to use variable inheritance to pass them through. |
| 19:03.14 | lmadsen | Set(_SOMEVARIABLE=foo) <-- inherited to the next channel |
| 19:03.33 | lmadsen | Set(__SOMEOTHERVARIABLE=bar) <-- inherited to all following channels from that call |
| 19:04.48 | lmadsen | atis_work: I'd like to see the dialplan where this is not working, because I do channel variable inheritance all the time with Local channels |
| 19:05.14 | lmadsen | if variable inheritance didn't work, I'm sure I would have run into it by now... |
| 19:06.41 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 19:06.56 | lmadsen | atis_work: SIP (channel 1) --> Local (channel 2) --> Dial(SIP/...) (channel 3) |
| 19:07.08 | atis_work | lmadsen: no, it's about callfile |
| 19:07.12 | lmadsen | to get channel 3 to see the variables from channel 1, you need to do a double underscore |
| 19:07.16 | atis_work | otherwise it works great |
| 19:07.22 | atis_work | i know that ;) |
| 19:07.54 | lmadsen | either way, if it is a bug, it should be discussed here until a bug is opened on bugs.digium.com, at which point you can discuss in #asterisk-bugs |
| 19:08.15 | plik | lmadsen: is the underscore / double-underscore only used when Setting the variable, or when recalling its value - ie with ${__WHATEVER} ? |
| 19:08.23 | lmadsen | only when setting the variable |
| 19:08.26 | file | plik: only when setting |
| 19:08.28 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 19:08.32 | plik | thannks |
| 19:08.54 | atis_work | lmadsen: i create a callfile to Local/123@invite and send it to join,456,1 |
| 19:09.16 | atis_work | lmadsen: i'm not sure that this is bug, so i would want architectural confirmation from developers |
| 19:09.19 | lmadsen | where are you setting the inherited variables, and can I see the dialplan? |
| 19:09.33 | atis_work | and if it's a bug, i could probably get it fixed with some pointers |
| 19:09.43 | lmadsen | atis_work: no, you need to confirm it is a bug first, then you can open a bug, then you can confirm the architectural stuff |
| 19:09.52 | lmadsen | right -- bug needs to be confirmed, then filed first |
| 19:10.10 | lmadsen | otherwise #asterisk-dev becomes 2nd and 3rd tier support, which it is not |
| 19:10.18 | atis_work | i know :) |
| 19:10.23 | lmadsen | please paste the callfile and the relevant dialplan parts into a pastebin |
| 19:10.41 | atis_work | huh, that would be quite some effort, i have a lot of macros |
| 19:10.44 | kyron | lmadsen, got the book, a bit disappointed, it looks the same as the pdf... |
| 19:11.12 | lmadsen | atis_work: "relevant" parts -- this means the callfile you're using, and the dialplan portion that is setting the underscores |
| 19:11.15 | kyron | A bit heavier though :P |
| 19:11.27 | atis_work | lmadsen: still half-hour of work :p |
| 19:11.33 | lmadsen | kyron: the book and the PDF are the same -- the entire book is released under the creative commons license |
| 19:11.55 | lmadsen | atis_work: I think you're thinking about way too much of the dialplan, or your dialplan is quite twisted |
| 19:12.05 | atis_work | it is |
| 19:12.06 | zobia | @file:hello @file. do you know how to set the asterisk to autostart if there's deadlock happened? |
| 19:12.08 | lmadsen | atis_work: you don't have to, but the developers are either going to ignore you, or tell you to do what I just told you |
| 19:12.22 | kyron | lmadsen, I was teasing you :) |
| 19:12.35 | lmadsen | kyron: I see :) Humour can be lost on me when in IRC |
| 19:12.57 | file | zobia: you could setup something to monitor it... |
| 19:13.02 | file | zobia: I have never done it, and thus can not answer |
| 19:13.10 | kyron | lmadsen, sorry about that. |
| 19:13.19 | lmadsen | np... I guess I'll go back to doing some real work now :) |
| 19:13.47 | zobia | @file: ok if there's no strait config for asterisk to do so, i have to setup a program to monitor it. thanks for answer |
| 19:14.32 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:14.50 | Daviey | watchdog \o/ |
| 19:15.14 | kyron | What will I be slashed, tortured and burned with if I buy anything from www.nxtvox.com ? |
| 19:21.50 | Qwell | kyron: clone cards? only by your users/customers |
| 19:21.59 | Qwell | and you'll want to /wrists |
| 19:23.23 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 19:24.24 | plik | \o/ ^o^ o< /o\ |
| 19:24.57 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 19:25.23 | kyron | hehehehehe LOL |
| 19:26.49 | kyron | well, for home use I wouldn't question, but for any _real_ implementations I would be disturbed to see people buy such low-priced clones... |
| 19:29.34 | *** part/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 19:29.48 | zobia | @file: hello . do you know what's Driver for channel 'SIP/pbxmgr-vegas-0a015498' does not support indication 3, emulating it means? |
| 19:30.47 | file | 3 is ringing |
| 19:31.06 | file | it means chan_sip wants the core to provide the ringing sound as an audio stream |
| 19:31.13 | file | instead of telling the SIP device to generate ringing on it's side |
| 19:32.12 | zobia | @file. what i can do to avoid this error? |
| 19:32.31 | zobia | i have to generate ring? |
| 19:32.47 | zobia | @file: like use dialplan to ring(4)? |
| 19:33.09 | file | it's not an error |
| 19:33.13 | file | it is a debug message meant for developers |
| 19:33.36 | file | unless you are a developer, or a developer has told you to turn on debug logging... don't |
| 19:33.51 | file | it *will* freak you out |
| 19:34.09 | hmodes | ignore the code behind the curtain. |
| 19:34.15 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:35.44 | ZPertee | how long is one ring? I want to dial an extension for 4 rings what is that equal to in seconds? |
| 19:35.50 | atis_work | lmadsen: i posted dialplan pieces and log here: http://pastebin.com/m292ba9ae |
| 19:36.00 | *** join/#asterisk Synoptic (i=Synoptic@modemcable034.152-81-70.mc.videotron.ca) |
| 19:36.20 | atis_work | you can see that from init_vars - call_id should be initalized only once, then in child channel it should be inherited |
| 19:36.36 | atis_work | however with callfile - those channels seems unrelated |
| 19:37.02 | *** part/#asterisk Synoptic (i=Synoptic@modemcable034.152-81-70.mc.videotron.ca) |
| 19:38.27 | *** join/#asterisk Synoptic (i=Synoptic@modemcable034.152-81-70.mc.videotron.ca) |
| 19:38.50 | mvanbaak | debug messages are fun |
| 19:39.06 | mvanbaak | and the asterisk ones are pretty friendly |
| 19:40.23 | *** join/#asterisk uwe (n=uwe@213.244.124.16) |
| 19:41.05 | lmadsen | atis_work: got it loaded, just doing some testing with a client, so I'll try and check it between tests |
| 19:42.25 | atis_work | lmadsen: thanks for looking |
| 19:42.30 | atis_work | do any other developers have some thoughts on this? |
| 19:42.42 | *** join/#asterisk angryuser (i=nononon@df01t2-212-195-196-46.d4.club-internet.fr) |
| 19:42.56 | lmadsen | file: hrmmm... when doing a Redirect from AMI, should the channel that gets hung up still execute the 'h' extension as per normal? (Because that doesn't happen) |
| 19:43.31 | file | I would think so, but I do not know |
| 19:45.34 | lmadsen | file: ya... I think it's a bug... |
| 19:45.44 | lmadsen | lol, I was just about to respond to atis_work |
| 19:45.45 | lmadsen | # |
| 19:45.46 | lmadsen | [Feb 5 11:12:41] DEBUG[16068] channel.c: Copying hard-transferable variable call_id. |
| 19:45.46 | lmadsen | # |
| 19:45.46 | lmadsen | [Feb 5 11:12:41] DEBUG[16068] channel.c: Copying hard-transferable variable TRANSFER_CONTEXT. |
| 19:45.56 | lmadsen | I don't see where it isn't doing what it is told to do |
| 19:46.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:47.28 | lmadsen | atis_work: wb |
| 19:47.32 | atis_work | lmadsen: you can always redirect other channel by adding ExtraChannel |
| 19:48.11 | lmadsen | atis_work: where are you calling this? (I don't see the output in the cli log) |
| 19:48.12 | lmadsen | # |
| 19:48.12 | lmadsen | <PROTECTED> |
| 19:48.12 | lmadsen | # |
| 19:48.12 | lmadsen | <PROTECTED> |
| 19:48.13 | lmadsen | # |
| 19:48.15 | lmadsen | <PROTECTED> |
| 19:48.17 | lmadsen | # |
| 19:48.21 | lmadsen | <PROTECTED> |
| 19:48.32 | *** join/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
| 19:48.38 | atis_work | oh, i cutted that parts out, they really aren't relevant |
| 19:48.39 | mvanbaak | ~pb |
| 19:48.39 | jbot | [~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://www.rafb.net/paste |
| 19:48.45 | mvanbaak | ;) |
| 19:48.49 | kyron | LOL |
| 19:49.27 | mvanbaak | we should create pb.asterisk.org |
| 19:49.33 | lmadsen | atis_work: what isn't being inherited then? I see the [Feb 5 11:12:41] DEBUG[16068] channel.c: Copying hard-transferable variable call_id. stuff |
| 19:49.59 | lmadsen | mvanbaak: I'm getting this from pb, but it's being dumb and making 1 line copied = 2 lines pasted |
| 19:50.05 | atis_work | yes, but i guess - that's copied to SIP channel, not Local/@invite |
| 19:50.19 | lmadsen | atis_work: that's where it's set... not copied to.... |
| 19:50.30 | lmadsen | you're calling the Local channel with the set variables... |
| 19:51.05 | mvanbaak | lmadsen: I know, it does that to me too all the time |
| 19:51.10 | atis_work | uh, i mean Local/2601@invite-8302,1 - that's where join_conf is executing |
| 19:51.31 | atis_work | the bridged channel |
| 19:52.25 | atis_work | you see - i set __call_id in Local/2601@invite-8302,2 |
| 19:52.45 | lmadsen | yes I see that |
| 19:52.47 | atis_work | so, it seems that Local/2601@invite-8302,1 which is bridged gets initialized before |
| 19:52.49 | lmadsen | what version is this? |
| 19:52.52 | atis_work | 1.4.17 |
| 19:53.10 | lmadsen | So you're going calfile --> Local --> SIP --> Local ? |
| 19:53.15 | atis_work | yes |
| 19:53.15 | file | you're trying to inherit a variable up |
| 19:53.28 | atis_work | umm, no, down |
| 19:53.51 | atis_work | well, as it seems now - it's Callfile -> Local,2 -> SIP -> Local,1 |
| 19:54.32 | atis_work | i would want to inherit from Local,2 to Local,1 (and i don't get - why they are in reverse order) |
| 19:55.04 | ManxPower | you can't send variables across SIP easily. |
| 19:55.22 | file | 2 is executing the dialplan, 1 is/will be executing dialplan also (per your callfile) |
| 19:55.29 | file | you are trying to pass a variable from 2 to 1 |
| 19:55.37 | file | so that it appears on the other side |
| 19:55.58 | atis_work | well, but i would want that 1 is created first, as it's executed first |
| 19:56.52 | file | they are created at the same time. |
| 19:57.14 | atis_work | should they? |
| 19:58.08 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:58.28 | file | yes |
| 19:58.54 | file | you can't do two things at once on a channel... that's why chan_local creates two channels |
| 19:59.16 | file | one is executing dialplan, the other is (usually) bridged to another channel OR is also executing dialplan logic |
| 20:00.03 | zobia | @file: do you know "Didn't get a frame from channel" and then "channel.c: Bridge stops bridging channels " issue? |
| 20:00.21 | file | zobia: it almost always means that the channel hungup |
| 20:00.26 | zobia | @file: do you know the possible way that can cause this? |
| 20:00.58 | zobia | @file: can this be a signal of "deadlock" channel? |
| 20:01.11 | file | zobia: no, it's normal |
| 20:01.29 | file | zobia: just turn off debug now. |
| 20:01.30 | atis_work | file: so there's now way that this is gonna get fixed sometime.. it just makes passing variables between callfile-originated channels impossible (or very hard) |
| 20:01.33 | zobia | @file: thank you |
| 20:01.50 | *** join/#asterisk micander (n=Michael_@Full-Service-Travel-1157986.cust-rtr.pacbell.net) |
| 20:01.53 | zobia | @file: i have to turn on debug. cause i want to know what cause the channel deadlock |
| 20:01.59 | file | atis_work: in your specific scenario for what you want to accomplish, I do not believe there is a way |
| 20:03.25 | atis_work | file: i want to be able to link them together in CDR |
| 20:03.56 | atis_work | usually i'm doing it by checking variable call_id, and if it's empty set __call_id=${UNIQUEID} |
| 20:05.12 | zobia | @file: whenever i got "channel.c: Avoiding initial deadlock" the deadlock will spread to every channels. do you have a better way for me to tell what can cause this? |
| 20:05.14 | atis_work | file: i tried local channels without /n - that gives me SIP channel in the end, and variables are inherited to that, but again - there's problem that local channel migrates to SIP at some point after answer |
| 20:05.48 | file | zobia: it is currently normal to have that message come up |
| 20:06.22 | file | zobia: the way to debug this was previously mentioned, compile Asterisk with DONT_OPTIMIZE and DEBUG_THREADS and when a deadlock happens use core show locks to get the info and submit a bug |
| 20:07.12 | file | atis_work: I can only tell you how chan_local works and why it is doing what it is doing., |
| 20:07.22 | zobia | @file: yes i know that is for 1.4. but now i am running on 1.2 and could not shift to 1.4 until it become stable with all my features. so iam struggling with it with 1.2 right now. |
| 20:07.32 | zobia | @file: do you know what can cause " channel.c: Dropping duplicate answer!" |
| 20:08.05 | file | zobia: it received a duplicate answer? a lot of the messages you are seeing are perfectly normal... |
| 20:08.31 | zobia | @file duplicate answer are normal? |
| 20:08.55 | file | it is possible. |
| 20:09.20 | zobia | @file: whenever i got this duplicate answer, after a while this phone will dead and the channels will be gradutelly all deadlock. |
| 20:09.57 | file | zobia: okay. |
| 20:10.07 | zobia | @file. the phone which has "hangup duplicate answers" is with 2 lines. |
| 20:12.08 | zobia | @file. how to force the 1.2 to through a core dump whenever there's deadlock? |
| 20:14.30 | file | I do not remeber. |
| 20:15.17 | zobia | @file:ok. |
| 20:15.56 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 20:17.10 | atis_work | file: you don't have opinion that there should be some way how to pass variables from one channel of originate to another? |
| 20:17.19 | *** join/#asterisk JonMcN (n=Jon@cpc4-sout2-0-0-cust715.sotn.cable.ntl.com) |
| 20:18.00 | file | it's not originating that is your issue... it's the fact inheritance only happens at channel creation time, you want to pass variables from one channel to another (in this case Local channels) |
| 20:18.44 | JonMcN | Hi, can someone help me find out why zaptel-1.4 (svn snaphost) won't build? http://pastie.caboo.se/147894 |
| 20:18.45 | JonMcN | TIA |
| 20:18.59 | file | atis_work: it could be useful |
| 20:19.37 | JonMcN | ./configure ; make # works fine |
| 20:19.47 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 20:20.00 | JonMcN | just can't install >:( |
| 20:21.06 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177649251.dsl.bell.ca) |
| 20:21.09 | atis_work | file, lmadsen - thanks.. i guess i'll be going with generating some unique hash when creating callfile - and then mess around with db to get something working.. |
| 20:23.28 | lmadsen | atis_work: sorry :( |
| 20:24.01 | atis_work | lmadsen: you see why i wanted developers point of view ;) |
| 20:24.31 | lmadsen | atis_work: yes, but it still wasn't necessary appropriate for that channel until it was verified |
| 20:24.33 | atis_work | is this still classified as user-ish question? :p |
| 20:24.37 | lmadsen | yes it is |
| 20:24.55 | file | you're just lucky I'm helping in here today |
| 20:25.19 | file | and know how it works. |
| 20:25.53 | *** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar) |
| 20:26.23 | zobia | @file: yes i think i am also luck that met you today. |
| 20:26.28 | lmadsen | otherwise I would have just told you to file a bug, which would have been the next logical step |
| 20:26.45 | zobia | @file: i found the problem before deadlock. it's right after the user delete a voicemail |
| 20:27.06 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 20:27.26 | zobia | @file: do you think deleteing voicemail could cause channel deadlock? |
| 20:27.46 | JonMcN | and this if i use the current tarball: |
| 20:27.48 | JonMcN | http://pastie.caboo.se/147897 |
| 20:27.54 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:27.57 | file | zobia: I suppose it is possible, but I do not remember that code 'nor am I going down that road |
| 20:28.47 | zobia | @file. no problem. since you thinkg it's possible, i will go on finding the reason. |
| 20:29.04 | zobia | @file. i never knows that deleting a voicemail can cause this. |
| 20:35.01 | JonMcN | anyone? |
| 20:40.08 | Daviey | Hmm, do you have the kernel-headers installed properly? |
| 20:40.11 | JonMcN | yes |
| 20:40.19 | *** join/#asterisk SwK (n=SwK@user-69-73-16-126.knology.net) |
| 20:40.31 | Daviey | zaptel should compile fine against the headers |
| 20:40.39 | Daviey | you shouldn't need the full source |
| 20:40.50 | JonMcN | :( |
| 20:46.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:47.07 | *** join/#asterisk LakeSolon (n=blake@63.231.182.86) |
| 20:47.32 | LeBowlingAlley | Hello, I'm having an intermittent issue (1-3 times a day) of incoming calls being answered but there is no audio. Meanwhile, the outside caller can hear the person that answered. The * logs are reporting "Didn't get a frame from channel (whatever SIP phone tried to answer the call)". Anybody have any idea what could be happening? system has been working for about a year prior. |
| 20:47.56 | LakeSolon | Afternoon Folks |
| 20:48.48 | file | LeBowlingAlley: that means the channel hungup |
| 20:51.13 | LeBowlingAlley | so any idea what else i could be looking for that would be related to the issue? |
| 20:54.17 | hmmhesays | I bet the called party is answering just as the calling party is sent to voicemail, or something like that |
| 20:55.22 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 20:57.49 | *** join/#asterisk CVirus (n=GoD@82.201.174.232) |
| 20:57.51 | file | a complete console output would confirm or deny hmmhesays' hypothesis |
| 20:58.21 | LeBowlingAlley | i dunno. it just started happening. |
| 20:58.37 | hmmhesays | file: indeed |
| 20:59.26 | LeBowlingAlley | i can't imagine after a year, we would start having so many cases of that happening. |
| 21:05.59 | hmmhesays | <file> a complete console output would confirm or deny hmmhesays' hypothesis |
| 21:06.35 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 21:06.50 | hmmhesays | good lord mediatrix sucks for faxing |
| 21:06.54 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 21:07.19 | LakeSolon | So I have a weird question, and there may well be another way of going about it, but what I've come up with right now is to run two instances of Asterisk on the same box each bound to a different IP, and 'bounce' a trunk off of the second one so that half my traffic comes from each IP. |
| 21:07.45 | hmmhesays | why would you do that? |
| 21:07.47 | LakeSolon | For the purpose of load balancing over two WAN connections. |
| 21:08.12 | *** join/#asterisk Atkins (n=atkins@216.80.0.58) |
| 21:08.42 | LakeSolon | I can bind one IP to one WAN, and one IP to the other. |
| 21:08.59 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 21:10.45 | zobia | @file |
| 21:10.56 | zobia | @file: do you know "rtp read too short"? |
| 21:11.09 | file | I already said what it was earlier. |
| 21:11.19 | husimon-away | that's what she said. |
| 21:12.19 | zobia | @file: you don't know ? |
| 21:12.36 | file | the packet that was received is too small to be an RTP packet |
| 21:12.38 | plik | husimon-away: http://xkcd.com/174/ |
| 21:12.57 | husimon-away | laugh |
| 21:13.53 | zobia | @file:ok |
| 21:14.16 | plik | husimon-away: sorry about the confusion with 'k to park' yesterday - but this is where I read it: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial pesky wiki lies :/ |
| 21:14.29 | husimon-away | yeah no problem i've run into the same crap reading wikis |
| 21:15.04 | husimon-away | the truth is i needed "t" |
| 21:15.12 | plik | heh |
| 21:15.22 | husimon-away | because for some reason call transferring on my phone won't let asterisk playback the park digits |
| 21:15.29 | plik | when I learn a bit more I moght go on a wiki cleansing session |
| 21:15.34 | husimon-away | i guess k is for the "park hot key" ? |
| 21:15.43 | plik | ah, prolly |
| 21:15.47 | husimon-away | *73 |
| 21:15.55 | jameswf | anyone seen any issues mixinf t1 with e1 in asterisk |
| 21:16.00 | husimon-away | was what I saw in my features.conf |
| 21:16.05 | plik | I have the same siyuation as you with cisco 7949s.... |
| 21:16.24 | husimon-away | plik, yeah kind of annoying, i don't really want to use two different methods of call transfering |
| 21:16.28 | plik | dtmf transfer plays the digits, but the phones softkeys dont |
| 21:16.35 | husimon-away | but luckily only about one person in the building uses it |
| 21:16.42 | husimon-away | so I just tell him the other way to do it |
| 21:16.46 | husimon-away | actually |
| 21:16.50 | husimon-away | optionally I could just enable k |
| 21:17.19 | plik | only 1 cisco here so not too much of a prob... |
| 21:18.09 | zobia | you people are really luck only 1 cisco phone. here are full of which drives me mad. |
| 21:19.26 | plik | the initiall connfig was a pain, but once done I'd think it not tooo hard to replicate |
| 21:20.23 | plik | cisco's on ebay seem to be the cheapest way of getting a voip phone round here ( apart from grandstream 101) |
| 21:21.01 | plik | I could get 2 7940s for the price of a polycom 330 or linksys 942 :/ |
| 21:21.16 | Qwell | aren't the 330s like $90? |
| 21:21.37 | Qwell | I would doubt that even 1 used 7940 could be had for less than that... |
| 21:21.46 | plik | prolly, but they're also like GBP 90 |
| 21:21.53 | plik | thats the polycom |
| 21:22.03 | Qwell | plik: find a better reseller |
| 21:22.08 | plik | ciscos are like GBP 35-45 mostly |
| 21:22.23 | plik | Qwell: all pretty similar over here |
| 21:22.23 | Qwell | what's that in USD? |
| 21:22.41 | plik | abbout double |
| 21:22.49 | bkruse | valentines day is so expensive. |
| 21:22.52 | plik | so 70 -70 USD for a cisco 7940 |
| 21:23.02 | plik | 70-90 |
| 21:23.04 | Qwell | bkruse: just wait |
| 21:23.53 | bkruse | Qwell: lol. I have to go get some nice stuff, ideas ideas |
| 21:24.13 | bkruse | I do not know if I am fully aware where I am asking this question.... :P |
| 21:24.57 | Qwell | plik: that's pretty cheap for a cisco |
| 21:25.12 | plik | yeah - thats used on ebay though.... |
| 21:25.16 | Qwell | stuff |
| 21:25.19 | Qwell | erm, still |
| 21:25.22 | bkruse | plik: cosmetic scratches? |
| 21:26.07 | plik | bkruse: some worse than others - one I got is practiaclly mint --- just a little dusty |
| 21:26.24 | bkruse | plik: nice! sounds well worth the deal then |
| 21:26.54 | plik | it is - I just wish there was a nice alternative at a similarly good price ) |
| 21:27.11 | eric2 | faxing support over g.711... anyone have this going? |
| 21:27.36 | bkruse | :X |
| 21:27.46 | eric2 | I'm guessing I need the nv_faxdetect installed? |
| 21:27.51 | *** join/#asterisk findlay (n=justin@72.8.99.158) |
| 21:27.56 | findlay | in when using the Voicemailmain() application is it possible to forward a message to a user in a different voicemail context? |
| 21:28.09 | findlay | s/^in // |
| 21:28.12 | eric2 | ya, it's possible |
| 21:28.19 | findlay | how? |
| 21:28.25 | eric2 | I haven't done it, but it cannot be too hard |
| 21:28.34 | Qwell | there's an option in the menu |
| 21:28.40 | eric2 | just make sure you specify the user + context |
| 21:28.41 | bkruse | listen to the lady |
| 21:28.41 | Qwell | oh, context |
| 21:28.48 | Qwell | hmm, probably not |
| 21:28.51 | findlay | when you push 8 it asks for an extension, nothing about context |
| 21:30.50 | *** join/#asterisk neonerz (i=d1dc7757@gateway/web/ajax/mibbit.com/x-36c8f0beafcbfed0) |
| 21:31.29 | findlay | is it possible to replicate the functionality of Voicemailmain() with other applications? Maybe I could make it work manually? |
| 21:34.18 | cesar_CR | hi guys, FWD is for any type of calls PSTN? CELL??? |
| 21:34.19 | *** join/#asterisk [DS]LynxW (n=jzawacki@pool-71-191-163-40.washdc.fios.verizon.net) |
| 21:35.37 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-103-90.red.bezeqint.net) |
| 21:36.12 | Dovid | I just updated my kernel and I am trying to build zaptel-1.2 from trunk. can anyone help me with this error ? |
| 21:36.12 | Dovid | http://pastebin.ca/893071 |
| 21:37.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:38.28 | inv_arp[work] | Dovid: what is the make line you passed |
| 21:39.32 | Dovid | make clean |
| 21:39.51 | Nivex | cd shower;make clean |
| 21:40.09 | Dovid | same rror |
| 21:40.24 | Dovid | erro* |
| 21:40.30 | Dovid | there is no shower directory |
| 21:40.35 | Dovid | error* |
| 21:41.34 | Dovid | i checked it out from http://svn.digium.com/svn/zaptel/branches/1.2/ |
| 21:41.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:45.07 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 21:45.46 | iratik | in AMI , if I had started a telnet session with AMI and did not logoff and my client became terminated... I cannot logon from another client until the first (terminated) client logs off? |
| 21:45.46 | Nugget | telnet is eeeeeeevil! |
| 21:46.34 | eric2 | telnet shouldn't be used |
| 21:46.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:47.01 | eric2 | unless you're connecting to 127.0.0.1 |
| 21:47.05 | iratik | i am |
| 21:47.18 | eric2 | aye |
| 21:47.28 | *** join/#asterisk uwe (n=uwe@a21-96.adsl.paltel.net) |
| 21:47.51 | iratik | thats aside from the point.... how do i logoff orphaned sessions? |
| 21:47.58 | eric2 | linux? |
| 21:48.08 | eric2 | or w32? |
| 21:48.27 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 21:48.31 | eric2 | if you have console access (linux) kill the process |
| 21:48.46 | eric2 | kill -9 `ps -aux | grep telnet` |
| 21:48.58 | eric2 | pooOOFFFF, it'll be gone |
| 21:49.04 | iratik | but that doesn't issue "Action: logoff" from that client |
| 21:49.09 | eric2 | true |
| 21:49.11 | iratik | that would just orphan the session |
| 21:49.16 | eric2 | can you restart the service? |
| 21:49.21 | iratik | not now |
| 21:49.24 | iratik | calls going |
| 21:49.25 | *** join/#asterisk eric_hill (n=eric_hil@204.94.175.2) |
| 21:49.27 | eric2 | ah |
| 21:49.37 | iratik | there may be no way to do this |
| 21:49.45 | eric2 | did you google? |
| 21:49.57 | iratik | restarted |
| 21:50.01 | iratik | as soon as a call ended |
| 21:50.10 | iratik | that logs off all AMI instances |
| 21:53.16 | *** join/#asterisk anthm][ (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:53.17 | lirakis | later all |
| 21:53.21 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 21:53.44 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 21:55.00 | cesar_CR | hi guys, I am from costa Rica, any good voip service provider that you can recomen me ? |
| 21:55.18 | eric_hill | Anyone have any experience with Qwest VoIP service? |
| 21:57.39 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:59.20 | [DS]LynxW | Hello, I'm using a TE220 and am getting a "squelch" type sounds on inbound and outbound calls. Any ideas before I call Digium support? |
| 22:01.59 | *** join/#asterisk tuxfoo (n=tmmarini@pool-72-65-149-149.chrlwv.east.verizon.net) |
| 22:02.34 | [DS]LynxW | Anyone heard of this, even? |
| 22:06.36 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 22:07.18 | eric_hill | Squelch --- wow... that takes me back... |
| 22:07.36 | eric_hill | Is it just when the call starts up, or during the call? |
| 22:07.40 | [DS]LynxW | Well, I'm not sure that is the term... it's something digital. |
| 22:07.46 | [DS]LynxW | during the call. |
| 22:08.11 | eric_hill | I knew exactly what you meant. :) |
| 22:08.22 | [DS]LynxW | the system is mainly a "middleman" between the telco and a Nortel MICS |
| 22:08.26 | eric_hill | It's a problem with the codec on a line when the latency gets too high. |
| 22:08.45 | [DS]LynxW | Well, googling sqeltch and asterisk doesn't seem to help.. so I wasn't sure. |
| 22:08.51 | eric_hill | We had the same problem on our frame relay connections when the pipes were somewhat full. |
| 22:08.57 | Qwell | who had that probem earlier with compiling zaptel 1.2? |
| 22:09.03 | eric_hill | We fixed it with a good QoS classification and MPLS. |
| 22:09.23 | [DS]LynxW | Well, it wasn't a problem before the Asterisk system was installed. |
| 22:09.36 | [DS]LynxW | right now, it's Telco->Asterisk->Nortel MICS |
| 22:09.50 | Qwell | Dovid: Give this patch a try. http://pastebin.ca/893126 |
| 22:09.52 | [DS]LynxW | and even calls between Asterisk<->Nortel MICS seem to have them. |
| 22:10.25 | [DS]LynxW | I'm ready to install the latest zaptel drivers (probably tonight after 2nd shift leaves) and see if that helps. |
| 22:10.42 | [DS]LynxW | but, I don't remember it being there when I originally setup the system about 3 months ago. |
| 22:10.57 | eric_hill | Make sure you're running full duplex on all of your Ethernet connections. Half-duplex could cause that problem. |
| 22:11.25 | [DS]LynxW | let me check. |
| 22:11.52 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 22:11.58 | JonMcN | Is a consistent zttest of 98.547363% acceptable? |
| 22:12.03 | Qwell | JonMcN: no |
| 22:12.26 | JonMcN | Qwell, How can i find out why it is so low? |
| 22:13.12 | eric_hill | JonMcN: what's the output of "lspci -tv"? |
| 22:14.06 | neonerz | eric_hill: ahhh frame relay, thats so 1990 |
| 22:14.39 | eric_hill | neonerz: Agreed. We're literally 60% of the way through our MPLS migration. QoS is so nice. |
| 22:15.02 | eric_hill | neonerz: And 1995 to be precise :) |
| 22:15.13 | bkruse | you sharing irq's? |
| 22:15.22 | bkruse | does your call quality go down when you move your mouse? :[ |
| 22:16.00 | xp_prg | anyone use perl to interact with asterisk here? |
| 22:16.18 | JonMcN | eric_hill, http://pastie.caboo.se/147962 |
| 22:16.25 | [DS]LynxW | eric_hill: eth0: negotiated 100baseTX-FD, link ok |
| 22:17.15 | eric_hill | JonMcN: you're having problems with zttest because your card is hanging off that sub-PCI device. It's probably a PCI bridge. Try moving the card to another PCI slot. |
| 22:17.48 | [DS]LynxW | Hmm.. I thought that was a gigabit nic.. and it is.. It must not be in a gigabit port on the switch :( I'll have to fix that as well. |
| 22:18.19 | eric_hill | [DS]LynxW: A) what's the latency between the asterisk box and your Nortel? B) If you do a ping flood with large packets (+4k), do any packets get dropped? |
| 22:18.44 | [DS]LynxW | How can I test the latency? |
| 22:18.47 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584495.dsl.bell.ca) |
| 22:18.56 | JonMcN | eric_hill, will do, the problem is that the motherboard has a special riser daughter board to expand the PCI slots - so could they all be the same |
| 22:19.12 | eric_hill | [DS]LynxW: Ping response time. |
| 22:19.15 | JonMcN | It's also full height PCI slots, with the smaller style card |
| 22:19.15 | *** join/#asterisk d-tech (n=d-dtech@72.245.233.107) |
| 22:19.50 | [DS]LynxW | Sorry for my ignorance, how would I ping the Nortel? |
| 22:19.53 | eric_hill | JonMcN: Had exactly the same problem with a 2U server from HP. The "left" side was across a bridge, the "right" side was direct... Of course they didn't document that :) |
| 22:20.29 | JonMcN | eric_hill, i'll try that now - this is a 2U Supermicro |
| 22:20.29 | eric_hill | [DS]LynxW: Am I missing something? Are you talking to the Nortel over IP? Or is it a T1? |
| 22:20.31 | [DS]LynxW | JonMcN: I can't say for sure, but some riser cards give the first slot direct access. |
| 22:20.33 | JonMcN | thanks |
| 22:20.36 | [DS]LynxW | T1 |
| 22:20.38 | [DS]LynxW | Sorry. |
| 22:20.49 | [DS]LynxW | TE220 is dual T1, one for Telco, one for Nortel. |
| 22:20.52 | eric_hill | [DS]LynxW: Well it sure is hard to ping down a T1 :) |
| 22:20.55 | JonMcN | will try now and report back |
| 22:21.03 | eric_hill | [DS]LynxW: Analog T1, or PRI? |
| 22:21.08 | [DS]LynxW | PRI |
| 22:21.10 | SomethingISOdd | hello is there anyway to see live calls? that are connected to asterisk |
| 22:21.17 | eric_hill | [DS]LynxW: Who's doing the clocking? |
| 22:21.46 | eric_hill | [DS]LynxW: And are you using q.SIG (q.931), or some other framing? |
| 22:21.50 | [DS]LynxW | eric_hill: Dunno, how do I tell? |
| 22:22.42 | eric_hill | Is your signalling set to pri_cpe, or pri_net? |
| 22:23.44 | [DS]LynxW | pri_net |
| 22:23.46 | [DS]LynxW | hmm. |
| 22:24.01 | [DS]LynxW | actuall, telco is pre_cpe and Nortel is pri_net |
| 22:24.05 | Qwell | ~book |
| 22:24.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:24.08 | Qwell | SomethingISOdd: you should read that |
| 22:24.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:24.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:24.43 | [DS]LynxW | Yeah, I plan on buying that.. in bulk.. to give to customers as well.. This is my first T1 system. |
| 22:25.08 | [DS]LynxW | I followed the "Nortel-Asterisk-0.2.pdf" guide for this. |
| 22:25.12 | [DS]LynxW | and that is what it said to use.. |
| 22:25.39 | eric_hill | [DS]LynxW: Phone companies almost always supply the clock. I treat the clock similar to NTP: Let the clock direction follow the "client". |
| 22:25.44 | [DS]LynxW | but now that I think about it, if Asterisk is talking pri_cpe to the telco, wouldn't the Nortel be expecting the same thing? |
| 22:25.45 | *** join/#asterisk RoyK (n=roy@ip-2-29-149-91.dialup.ice.no) |
| 22:26.04 | SomethingISOdd | Qwell ok thanks |
| 22:26.16 | [DS]LynxW | So, would that be correct? |
| 22:26.22 | eric_hill | [DS]LynxW: If the asterisk box is going between the two locations, the asterisk should be set to pri_cpe for the PSTN side, and pri_net for the Nortel side. |
| 22:26.26 | JonMcN | eric_hill [DS]LynxW: bottom of 3 slots now (was middle) = Average: 99.951172 |
| 22:26.32 | JonMcN | \o/ |
| 22:26.54 | eric_hill | [DS]LynxW: So, yes, it's correct. |
| 22:27.01 | eric_hill | JonMcN: woot! |
| 22:27.06 | [DS]LynxW | :) |
| 22:27.24 | JonMcN | eric_hill, thanks |
| 22:27.44 | JonMcN | Although, i'm sure i've had better timers from ztdummy in the past :( |
| 22:28.31 | eric_hill | [DS]LynxW: What is the output of pri show span 1? |
| 22:28.41 | eric_hill | [DS]LynxW: And 2 of course. |
| 22:29.16 | [DS]LynxW | Anything I should be looking for? I don't have direct copy/paste into IRC :/ |
| 22:29.38 | eric_hill | [DS]LynxW: http://pastebin.com |
| 22:30.17 | eric_hill | [DS]LynxW: I'd look for things like "Retrans > 0". |
| 22:30.22 | husimon | [DS]LynxW, it shouldn't say no pri running on X |
| 22:30.31 | husimon | ;) |
| 22:30.36 | [DS]LynxW | http://pastebin.com/d554ed6d6 |
| 22:30.39 | [DS]LynxW | I hope that works. |
| 22:31.10 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 22:31.37 | eric_hill | [DS]LynxW: Any idea what that N200 Counter is? |
| 22:31.40 | [TK]D-Fender | [DS]LynxW, what ver of zaptel? |
| 22:31.47 | [DS]LynxW | Oh.. and doing a call recording doesn't "hear" the sound. |
| 22:32.10 | JonMcN | eric_hill, bah, on another machine ztdummy just produced: Best: 100.000000 -- Worst: 99.890137 -- Average: 99.949341, and hardware = Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952958.. Seems unfair :( |
| 22:32.23 | [DS]LynxW | [TK]D-Fender: can you tell from the source? I wasn't smart enough and left it named zaptel-1.4-current.tar.gz |
| 22:32.55 | eric_hill | JonMcN: of course. Production is never as happy as test... |
| 22:33.09 | JonMcN | :) |
| 22:33.21 | [TK]D-Fender | [DS]LynxW, Ok, its 1.4 series at least... go "up"?grade to the latest 1.4 release (non RC) |
| 22:33.22 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 22:33.33 | [DS]LynxW | [TK]D-Fender: I have 1.4.8 waiting to go.. but I can't play with it till tonight. |
| 22:33.50 | [TK]D-Fender | [DS]LynxW, ok, for sanity check, pastebin "cat /proc/interrupts" |
| 22:34.12 | *** join/#asterisk tuxfoo (n=tmmarini@pool-72-65-149-149.chrlwv.east.verizon.net) |
| 22:35.09 | [DS]LynxW | http://pastebin.com/d457e35ea |
| 22:35.23 | [TK]D-Fender | [DS]LynxW, and another with "dmesg" and your zaptel.conf. |
| 22:35.37 | husimon | hey [TK]D-Fender you mentioned before that using DTMF to do call transfers, etc was bad. Could you comment on why it is a bad idea? |
| 22:35.57 | *** join/#asterisk primenz (n=root@mail.primesoft.co.nz) |
| 22:36.27 | [TK]D-Fender | husimon, It means you are putting responsibilty for bread & butter stuff in *'s hands and delays passing DTMF to IVR's, etc. Let your phones do the work they are supposed to do by themselves. |
| 22:37.17 | husimon | k |
| 22:37.49 | [TK]D-Fender | husimon, Also part about why Zaptel FXS = ASS |
| 22:37.57 | [DS]LynxW | [TK]D-Fender: my dmesg is very messy right now.. I was doing ztdiags |
| 22:38.06 | [TK]D-Fender | [DS]LynxW, thats ok. |
| 22:38.07 | *** part/#asterisk RoyK (n=roy@ip-2-29-149-91.dialup.ice.no) |
| 22:38.15 | [TK]D-Fender | [DS]LynxW, dump it all, I'll sift through |
| 22:38.19 | [DS]LynxW | but I do have this message: Losing some ticks... checking if CPU frequency changed. |
| 22:38.36 | [TK]D-Fender | [DS]LynxW, Ah, have you shecked your kernel timer freq? |
| 22:38.40 | [TK]D-Fender | checked* |
| 22:38.52 | [DS]LynxW | Nope. |
| 22:38.56 | [TK]D-Fender | [DS]LynxW, do it! |
| 22:39.04 | [TK]D-Fender | [DS]LynxW, needs to be 1000. |
| 22:39.14 | *** join/#asterisk stanhope (n=roberto@host-84-221-126-91.cust-adsl.tiscali.it) |
| 22:39.26 | husimon | [TK]D-Fender, yeah i'm going to use linksys atas for my fxs. |
| 22:39.34 | [TK]D-Fender | husimon, Good call. |
| 22:39.53 | [TK]D-Fender | husimon, Anything below 24 ports = Linksys |
| 22:40.37 | husimon | [TK]D-Fender, yeah I only have about 8. For future reference what do you recommend past 24? |
| 22:40.49 | [TK]D-Fender | husimon, For 8-port : SPA-8000 <- |
| 22:40.54 | *** join/#asterisk angryuser (i=nononon@df01t2-62-34-201-4.d4.club-internet.fr) |
| 22:41.14 | [TK]D-Fender | husimon, For 24+ I would probably go for a Mediatrix 1124 or AudioCodes MP-124 |
| 22:41.21 | jm|laptop | hmm |
| 22:41.29 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 22:41.34 | eric_hill | [TK]D-Fender: Any thoughts on the Rhino channel banks? http://www.rhinoequipment.com |
| 22:41.36 | husimon | [TK]D-Fender, unfortunately the ports are spread across two floors and different patch panels so I just got 2 port models |
| 22:41.43 | jm|laptop | why is Asterisk appending @my-asterisk-ip to my SIP calls when I specify CALLERID(all)="name <number>" |
| 22:41.54 | husimon | [TK]D-Fender, yeah i was thinking mediatrix audio codes too |
| 22:41.55 | BBHoss | eric_hill, i've heard good things about Rhino in general |
| 22:41.55 | [TK]D-Fender | eric_hill, I reqally prefer not to, but if you have the extra T1 port, its not "bad". |
| 22:42.02 | *** join/#asterisk tristanbob_ (n=tristanr@oalug/member/tristanbob) |
| 22:42.04 | husimon | err "and audiocodes" |
| 22:42.30 | BBHoss | audiocodes's mediatrix is nice too |
| 22:42.31 | eric_hill | [TK]D-Fender: Unfortunately I have to support about 20-25 cheap-ass Wal-Mart phones on a factory floor... |
| 22:42.39 | [TK]D-Fender | eric_hill, I'll say as channel banks go they are very friendly and the company good to work with. |
| 22:42.48 | [TK]D-Fender | eric_hill, I jsut don't like the technology |
| 22:42.58 | BBHoss | channel banks are never fun |
| 22:42.59 | [TK]D-Fender | eric_hill, thats fine... Sip gateways <--- |
| 22:43.04 | BBHoss | especially when using CAS |
| 22:43.10 | uwe | um, i just purchased g729 codec from digium, and when i do show translation, it shows that converting from various codecs to g729 will take 48 milliseconds ! is this normal !!! this is a lot ! |
| 22:43.24 | eric2 | what's the cpu speed? |
| 22:43.26 | BBHoss | uwe, sounds really high to me |
| 22:43.36 | [TK]D-Fender | uwe, I'm not sure thats MS... thats just relative wieght as far as I know. |
| 22:43.37 | Qwell | uwe: 48 milliseconds for 1 second of audio |
| 22:43.45 | uwe | 2992.674 Mhz |
| 22:43.53 | angryuser | <[TK]D-Fender is there any way to detect if internet is failing and signal that to asterisk? or for example can * read a value from file (0|1) ? |
| 22:43.54 | husimon | uwe what type of chip... |
| 22:43.55 | BBHoss | that doesn't say much |
| 22:43.58 | eric2 | xeon, p4, opteron? |
| 22:44.01 | husimon | uwe, dual core xeon, etc etc |
| 22:44.03 | eric2 | 386? |
| 22:44.07 | BBHoss | 286? |
| 22:44.10 | Qwell | and yeah, if you're going from like ilbc or speex to g729, it will take a while |
| 22:44.10 | BBHoss | 8088? |
| 22:44.12 | eric2 | x8086? |
| 22:44.25 | husimon | eric2 i challenge you to find anything below a p4 that goes to 2992mhz : |
| 22:44.25 | BBHoss | UNIVAC? |
| 22:44.31 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:44.32 | husimon | eric2 in the intel line... |
| 22:44.34 | Qwell | husimon: Sparc |
| 22:44.35 | [TK]D-Fender | angryuser, its not * job to look for this, and you can have your external process warn * any way you want. |
| 22:44.35 | eric_hill | C=64... |
| 22:44.36 | eric2 | I don't know the intell lineup |
| 22:44.40 | BBHoss | husimon, you can get a celeron that high |
| 22:44.42 | Qwell | amd |
| 22:44.43 | eric2 | I stay away from intel |
| 22:44.47 | uwe | Intel(R) Pentium(R) 4 CPU 3.00GHz |
| 22:44.58 | husimon | yeah p4 3ghz is a little slow |
| 22:45.00 | uwe | nothing, plain p4 machine |
| 22:45.01 | [DS]LynxW | [TK]D-Fender: Sorry, how do I check the kernel frequency? |
| 22:45.03 | BBHoss | probably a p4c800 |
| 22:45.19 | [TK]D-Fender | Qwell, can you fill [DS]LynxW in on that? |
| 22:45.32 | Qwell | no idea, and I don't think it matters anymore |
| 22:45.34 | angryuser | <[TK]D-Fender> what is the simpliest way? agi? php sript? |
| 22:45.48 | BBHoss | uwe, try core show translation recalc |
| 22:45.55 | [DS]LynxW | Google is telling me to add clock=pmtmr to the kernal boot options. |
| 22:46.01 | eric_hill | [DS]LynxW: What flavor of linux? |
| 22:46.04 | [TK]D-Fender | angryuser, there is this huge "depends". You need to tell me very specifically how and where you would check for this. |
| 22:46.12 | uwe | [TK]D-Fender, show translation --> Translation times between formats (in milliseconds) |
| 22:46.30 | [DS]LynxW | CentOS 4 AKA Trixbox. |
| 22:46.31 | [TK]D-Fender | uwe, Ok, I wasn't 100% sure on it, but thanks for the clarification. |
| 22:46.35 | [TK]D-Fender | EW! |
| 22:46.36 | [DS]LynxW | 2.2.3 |
| 22:46.38 | uwe | same results BBHoss |
| 22:46.39 | [TK]D-Fender | EWWWWWWWWWWWWWW!!!!!!!!!!!!!!!!!!! |
| 22:46.43 | [DS]LynxW | Yeah.. I know.. |
| 22:46.56 | [DS]LynxW | :/ |
| 22:46.57 | Qwell | uwe: what are the slin > g729 and g729 > slin times? |
| 22:47.04 | [DS]LynxW | But, it's fast and easy.. for the most part. |
| 22:47.11 | [DS]LynxW | And works fine for analog. |
| 22:47.15 | Qwell | and which version of the codec did you use? |
| 22:47.21 | eric_hill | Ubuntu + apt-get install asterisk :) |
| 22:47.23 | [TK]D-Fender | [DS]LynxW, Believe me, that doesn't spare you any funny looks at all... |
| 22:47.23 | [DS]LynxW | this is the first problem I have had with a stable system.. if there is such a thing. ;) |
| 22:47.27 | eric_hill | https://www.centos.org/modules/newbb/viewtopic.php?forum=34&topic_id=3098&viewmode=thread |
| 22:47.38 | angryuser | <[TK]D-Fender> ok for example we ping www.google.com if timeout internet is down, so i want to use gotoif, the question is, how to change the value in * db from external program |
| 22:47.49 | *** part/#asterisk primenz (n=root@mail.primesoft.co.nz) |
| 22:47.51 | [DS]LynxW | yeah, I know.. CentOS5+atrpms.net = yum install asterisk as well. |
| 22:47.51 | [TK]D-Fender | [DS]LynxW, if you're running a Trixbox install there is no need to look for anything else. |
| 22:48.04 | husimon | what are the benefits of using g729 vs g711?, it's just low bandwidth? |
| 22:48.05 | *** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose) |
| 22:48.06 | BBHoss | my processor is like a celeron 1.4ghz and i get 13200 slin>g729 |
| 22:48.11 | [TK]D-Fender | [DS]LynxW, The kernel was primed for *. Its jsut a question of your MB & other hardware |
| 22:48.19 | [TK]D-Fender | husimon, correct |
| 22:48.26 | drako | is there a way to listen conversation live? |
| 22:48.35 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 22:48.36 | BBHoss | and g729>slin is 2800 |
| 22:48.37 | eric2 | like a spy? |
| 22:48.38 | uwe | Qwell, 2 and 47 |
| 22:48.41 | BBHoss | thats microseconds though |
| 22:48.42 | [DS]LynxW | drako: chanspy |
| 22:48.47 | eric2 | spying is bad |
| 22:48.51 | [TK]D-Fender | husimon, anything other than G.711 is only there for bandwidth savings (except G.722 which is there to make you think you should waste more for no good reason) |
| 22:48.52 | Qwell | uwe: and which codec module did you use? |
| 22:49.09 | [DS]LynxW | eric2: depends on your job. |
| 22:49.35 | uwe | Qwell, codec_g729a_v33_i686.tar.gz |
| 22:49.37 | [DS]LynxW | [TK]D-Fender: Well, This wasn't an issue in the beginning.. it's just gotten to this point. |
| 22:49.40 | uwe | if this is what you mean |
| 22:49.45 | Qwell | it is |
| 22:49.48 | angryuser | <[TK]D-Fender> no idea ? |
| 22:49.52 | [DS]LynxW | Or at least not this bad.. or not as noticable.. or I wouldn't have put it into production. |
| 22:50.08 | Qwell | uwe: asterisk 1.4? |
| 22:50.20 | uwe | yes Qwell |
| 22:50.36 | eric_hill | [DS]LynxW: Is your call plan transcoding anything? I.e., is it a CPU-bound problem? |
| 22:50.42 | [DS]LynxW | [TK]D-Fender: not it seems like every call has it.. and only one side can hear it.. but it might be internal or external, it seems random.. or maybe its the originating party. |
| 22:50.47 | uwe | yesterday i tried the opensource g729, and it was 2 - 9 i think |
| 22:50.51 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:50.56 | codejunky | hello, I have a sip hardware phone connected, "sip show peers" show the phone with status: OK (300 ms), 300ms is really slow for normal ethernet, right? |
| 22:51.16 | [TK]D-Fender | angryuser, "how to change the value in * db from external program" = asterisk -rx "database set system down 1" |
| 22:51.17 | [DS]LynxW | eric_hill: Well, load average: 0.01, 0.02, 0.00 |
| 22:51.20 | [DS]LynxW | so I don't think so. |
| 22:51.27 | Qwell | uwe: does the i586 version change the numbers at all? |
| 22:51.36 | uwe | i can try ... |
| 22:51.52 | drako | [DS]LynxW, ty |
| 22:51.53 | BBHoss | uwe, on my anemic celeron 1.7ghz i get 13.2ms slin>g729 |
| 22:52.01 | eric_hill | [DS]LynxW: Are you out of magic pixie dust? |
| 22:52.19 | [DS]LynxW | eric_hill: aside from trying to upgrade the drivers tonight.. I think so. |
| 22:52.20 | [DS]LynxW | :/ |
| 22:52.22 | uwe | creepy ... |
| 22:52.44 | [DS]LynxW | I just don't want to have to rebuild the system at 3am to get it back up before people come in. |
| 22:52.53 | eric_hill | [DS]LynxW: If it /was/ working and then started acting up, you might have a shitty interface card. |
| 22:53.11 | eric_hill | [DS]LynxW: Can you swap the two ISDN PRI cards and see if that helps? Or is it a two-port card? |
| 22:53.18 | [TK]D-Fender | g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 |
| 22:53.28 | [DS]LynxW | two-port card. |
| 22:53.30 | eric_hill | [DS]LynxW: Or can you run a diag on the Nortel card? |
| 22:53.31 | [TK]D-Fender | <PROTECTED> |
| 22:53.40 | [TK]D-Fender | Mine seems fine |
| 22:53.45 | [DS]LynxW | Shoot.. I hate the Nortel.. |
| 22:53.48 | BBHoss | hey whats that site that lets you give your stuff away to other people in the area (not craigslist) |
| 22:53.59 | eric_hill | BBHoss: ebay? |
| 22:54.00 | [TK]D-Fender | BBHoss, Ebay? |
| 22:54.05 | plik | BBHoss: freecycle ? |
| 22:54.06 | Qwell | freecycle? |
| 22:54.07 | BBHoss | yeah |
| 22:54.12 | [DS]LynxW | freecycle :) |
| 22:54.20 | [DS]LynxW | anyway.. I'll try upgrading to night.. |
| 22:54.27 | [DS]LynxW | and stop in if I still have issues.. |
| 22:54.32 | [DS]LynxW | thanks for all the help. |
| 22:54.38 | husimon | i'm pretty sure my asterisk boxes are way more powerful then they need to be :P |
| 22:54.49 | angryuser | <[TK]D-Fender> thanks |
| 22:56.30 | JonMcN | gah, i reinstalled wanpipe, now back to 98.535156% >:/ |
| 22:57.13 | angryuser | codejunky i have one, what is the problem? |
| 22:57.27 | *** part/#asterisk [DS]LynxW (n=jzawacki@pool-71-191-163-40.washdc.fios.verizon.net) |
| 22:57.27 | uwe | Qwell, same same :( |
| 22:57.41 | codejunky | angryuser: sip show peers shows what tim in ms? |
| 22:57.54 | codejunky | angryuser: phone/phone 192.168.0.26 D 5060 OK (304 ms) |
| 22:57.57 | angryuser | codejunky wait |
| 22:58.19 | codejunky | angryuser: Is the webinterface *really* slow for you? |
| 22:58.45 | angryuser | codejunky yes 119 ms bigger that everyone else |
| 22:59.06 | uwe | codejunky, qualification ping (read sip qualify) |
| 22:59.16 | codejunky | hm, okay |
| 22:59.25 | uwe | http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+sip+show+peers :) |
| 22:59.43 | angryuser | <codejunky> i like it coz it's numerotating really fast |
| 22:59.44 | codejunky | I have the problem that the phone is sometimes unreachable |
| 22:59.51 | Qwell | uwe: can you pastebin the entire output of core show translations? |
| 22:59.52 | Qwell | ~pb |
| 22:59.52 | jbot | [~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://www.rafb.net/paste |
| 23:00.07 | uwe | one sec |
| 23:00.15 | codejunky | asterisk reports that the phone is not reachable, I do not know why |
| 23:00.16 | angryuser | <codejunky> firmware upgraded? |
| 23:00.23 | codejunky | angryuser: not done yet |
| 23:00.46 | codejunky | angryuser: I did not find any links on the homepage for the firmware files |
| 23:01.00 | codejunky | angryuser: and in the webinterface the button to upgrade does not work too :/ |
| 23:01.25 | angryuser | <codejunky> i did and the only thing annoing from that phone that i got mess in CLI "Got SIP response 405 "Method Not Allowed" back from 192.168.0.79" |
| 23:01.33 | *** join/#asterisk jcims (n=chatzill@rrcs-24-172-217-2.central.biz.rr.com) |
| 23:01.47 | codejunky | angryuser: hm okay |
| 23:02.03 | angryuser | <codejunky> i upgraded from interface |
| 23:02.09 | codejunky | angryuser: when did you the firmware upgrade? I was not able to find the proper files anymore |
| 23:02.17 | [hC] | bkruse: ping! |
| 23:02.20 | uwe | Qwell, http://pastebin.com/m75ae409c |
| 23:02.38 | bkruse | [hC]: pong! |
| 23:02.49 | bkruse | you just caught me, whats up? |
| 23:02.50 | [hC] | bkruse: hey! :) I had a couple questions for you about the aa50, if you have a sec? |
| 23:03.03 | bkruse | [hC]: shoot |
| 23:03.07 | angryuser | <codejunky> long time ago, half a yerar maybe |
| 23:03.15 | bkruse | I will try my best |
| 23:03.16 | codejunky | hm okay |
| 23:03.17 | uwe | there is an ugly shift in the table ... :) shift everything from the second raw to the left one col :) |
| 23:03.22 | codejunky | angryuser: I will try the qualify thing |
| 23:03.22 | Qwell | yeah |
| 23:03.42 | [hC] | bkruse: Did you say you got VLANs working from the ssh shell?... I have it "working" - but all I can seem to do is ping, any time i try to transfer data, it just hangs. |
| 23:03.59 | [hC] | bkruse: including telnetting to port 22, in either direction, and expecting to see an ssh header, for example. |
| 23:04.01 | bkruse | did you flush your iptables? (nat rules at least) |
| 23:04.08 | [hC] | yes. |
| 23:04.09 | drmessano | "Mobile VOIP" is not the same as "I've got my Asterisk box in the back seat" |
| 23:04.15 | tzanger | drmessano: hahaha |
| 23:04.17 | bkruse | iptables -t nat -F |
| 23:04.19 | [hC] | I killed udhcpd, dnsmasq, and iptables -t nat -F |
| 23:04.45 | bkruse | hmm, but your icmp requests are getting through? hmmm... did you iptables -F before starting? because it does not some DROP rules that may not be in nat |
| 23:05.04 | cappiz | i get : Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) when i try asterisk -r |
| 23:05.06 | [hC] | I did iptables -F and iptables -t nat -F |
| 23:05.12 | plik | drmessano: how about Asterisk running on my laptop using starbucks wifi? |
| 23:05.15 | cappiz | the file does exist, and asterisk is running |
| 23:05.15 | codejunky | uwe: I have no "sip qualify " on asterisk 1.4.17, any ideas why? |
| 23:05.23 | [hC] | bkruse: to set up the vlan, i simply did vconfig add eth1 20, then ifconfig eth1.20 ip netmask blah |
| 23:05.30 | [hC] | It seemed like all that was necessary. |
| 23:05.34 | bkruse | right right |
| 23:05.38 | bkruse | routes? |
| 23:05.42 | codejunky | uwe: sorry, got it |
| 23:05.47 | [hC] | bkruse: they're there, nothing weird in the way. |
| 23:06.10 | [hC] | bkruse: I exhausted all i could think of for a few hours and eventually gave up... are you using vlans on yours? |
| 23:06.41 | bkruse | [hC]: i will think about it....hmmm |
| 23:06.46 | drmessano | lol |
| 23:06.53 | uwe | codejunky, its a part of sip configuration |
| 23:06.58 | uwe | http://www.voip-info.org/wiki/view/Asterisk+sip+qualify |
| 23:07.10 | uwe | sorry, just got all responses at one time !! |
| 23:07.16 | codejunky | uwe: yeah, okay, thanks. :-) |
| 23:07.40 | codejunky | uwe: I disabled it, is it the right choice to solve my problems? |
| 23:07.48 | [hC] | Also - another thing I was looking at last night, is the aadk SVN branch does not seem to be autotag updated when the normal asterisk trunk is updated? Is that going to change? I was looking to take advantage of the newly opensourced res_phoneprov, and the old commercial one (which is much more limited) is still in the aadk image. |
| 23:08.01 | [hC] | Along with the old (broken) chan_skinny.so |
| 23:08.05 | drmessano | Im trying to remember who it was... Had asterisk VM's installed in Windows, and used X-Lite to connect to the asterisk VM, IAX from the Asterisk VM to an Asterisk VM on his sisters and mom's laptops and they had similar setups |
| 23:08.19 | uwe | um, im not sure, but i think it would be wise to set it to a hight value ... like 5000 |
| 23:08.32 | JonMcN | Does a PRI card need to be connected to the telco network, for the timing to work? |
| 23:08.35 | plik | insane |
| 23:08.44 | drmessano | So they each 3 had Asterisk VMs, IAX2 connected between them, and Xlite softphone to talk to each of their Asterisk Vms |
| 23:08.49 | [hC] | i wanted to update my aadk image but its svn tree is old. |
| 23:09.04 | drmessano | That a HUGE WTFFFFFFF |
| 23:09.07 | [hC] | bkruse/qwell ^^ (if either of you know?) |
| 23:15.26 | cappiz | i get : Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) when i try asterisk -r. File exist and asterisk is running |
| 23:15.39 | *** part/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 23:16.34 | husimon | drmessano, that sure sounds like a stupid way to do it |
| 23:17.29 | husimon | drmessano, did he also need a $1900 loan ? |
| 23:18.24 | husimon | and have $200k vans? |
| 23:19.59 | drmessano | lol |
| 23:20.28 | drmessano | Sadly, this wasn't just 1 stupid lump... this is actually TWO different people |
| 23:20.41 | husimon | damn |
| 23:20.43 | drmessano | <deity> help us! |
| 23:20.47 | husimon | lol |
| 23:21.04 | drmessano | Super Tuesday is so exciting |
| 23:21.06 | husimon | i'm so using that |
| 23:21.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:21.10 | jcims | best quality softphone? |
| 23:21.10 | husimon | yeah i bet |
| 23:21.14 | jcims | for winders |
| 23:21.20 | drmessano | "Digg predicts Ron Paul to win Republican nomination" |
| 23:21.21 | jcims | (free or $$$) |
| 23:21.22 | husimon | i'm just going to read the news tomorrow |
| 23:21.25 | jcims | lol |
| 23:21.28 | husimon | digg is full of shit |
| 23:21.29 | *** join/#asterisk CobraCommand (n=x@63.161.232.136) |
| 23:21.31 | drmessano | "..and be declared real winner of superbowl" |
| 23:21.42 | jm|laptop | why does callerID always show MY asterisk server after the @ ? |
| 23:21.47 | jm|laptop | for incoming SIP calls |
| 23:22.37 | jm|laptop | hm |
| 23:22.37 | husimon | drmessano, did digg also say that rudy giuliani is a real american hero and 911 is the answer for everything? |
| 23:22.39 | husimon | :P |
| 23:22.40 | CobraCommand | hi guys, I need some help, I have an asterisk which transfers incoming calls from an E1 to a SIP Phone, using a Digium TE220, but why does it sounds like a robotic voice? It seems Darth Vader is on the line |
| 23:22.53 | husimon | CobraCommand, are you sure it's not your father on the line? |
| 23:23.11 | CobraCommand | hahaha I'm sure |
| 23:23.17 | husimon | ;) |
| 23:23.18 | *** part/#asterisk PepOSX (n=angeldav@190.72.146.204) |
| 23:23.45 | drmessano | Probably |
| 23:24.28 | CobraCommand | any ideas? |
| 23:24.41 | *** join/#asterisk neoalex (n=neoalex@cpe-74-73-94-101.nyc.res.rr.com) |
| 23:25.32 | neoalex | hi guys, I'm having a problem when trying to call a SIP address from one of my extensions I get Failed to authenticate on INVITE |
| 23:25.39 | plik | CobraCommand: no, but you should take advantage of it and record some really cool sounding prompts ;) |
| 23:26.36 | Qwell | CobraCommand: what codec is the sip phone using? |
| 23:28.01 | [hC] | Qwell: hey, do you know if the aadk tree is going to start getting autotagged from the asterisk trees or possibly trunk tree? the 'asterisk' tree in the aadk repos is behind significantly.. |
| 23:28.08 | CobraCommand | QWell: it's an X-lite soft phone |
| 23:28.21 | Qwell | CobraCommand: what codec? |
| 23:28.46 | jcims | speaking of soft phones, is there a short list of 'best' softphones out there? |
| 23:28.57 | jcims | soft phone reviews or whatnot |
| 23:30.32 | plik | jcims: not sure about reviews, but I think there a list on the wiki.... you could always try a few and update it with your verdicts |
| 23:31.04 | jcims | that's always the hard part...coming back around to share your opinion :) |
| 23:31.14 | plik | always |
| 23:31.34 | plik | what platform you on , anyway? |
| 23:31.57 | jcims | windows for the most part. i'm trying to set up a small volunteer call center and want something that folks can use from home |
| 23:32.27 | jameswf | I like mozphone simple and easy |
| 23:32.40 | *** join/#asterisk CobraCommand (n=x@63.161.232.136) |
| 23:32.46 | plik | dunno then... prolly a toss up between x-lite, sjphone and wengophone |
| 23:32.49 | CobraCommand | QWell: it uses Broadvoice-32, Broadvoice 32-fec, g711 alaw, g711 ulaw, ilbc, Speex fec, speex wideband, speex wideband fec |
| 23:32.51 | jcims | jameswf: do you use a headset with it? |
| 23:32.57 | plik | or mozphone :) |
| 23:33.01 | jcims | plik: thanks...i'll start there :) |
| 23:33.16 | plik | good luck ..., and don't forget to report back ;) |
| 23:33.43 | jcims | haha....will do |
| 23:33.59 | mvanbaak | Qwell: ping |
| 23:34.47 | jcims | in the eternal words of Lord Farquad 'Someday I will repay you, unless of course I can't find you, or if I forget. ' |
| 23:35.14 | plik | heh |
| 23:39.07 | *** join/#asterisk CobraCommand (n=x@63.161.232.136) |
| 23:42.20 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 23:46.02 | *** part/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 23:57.52 | drmessano | Hmmm |
| 23:58.06 | drmessano | The internet is being goofy |
| 23:58.21 | drmessano | Sounds like a job for LEROOOY JENKINS |
| 23:58.55 | husimon | ive done exactly what leroooy did :) |
| 23:58.59 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 23:58.59 | drmessano | lol |
| 23:59.13 | Robba | Hi Guys |
| 23:59.26 | husimon | it is amazing how popular he got |
| 23:59.33 | drmessano | yep |
| 23:59.45 | Robba | I have a problem with asterisk recognizing a Zap Trunk |
| 23:59.50 | husimon | wow his name was even the answer to to a jeopardy question |
