| 00:03.12 | Strom_C | livesNbox: are you using plain-vanilla asterisk, or are you using FreePBX/AMP/Trixbox/A@H? |
| 00:04.23 | livesNbox | Trixbox at the moment |
| 00:04.37 | livesNbox | I think I might have it close to working.. I'm getting an error due to congestion though |
| 00:04.39 | Strom_C | livesNbox: see the topic of this channel |
| 00:04.50 | livesNbox | and the sound of the playback on the recordings is very strange.. kind of stuttering.. |
| 00:05.03 | livesNbox | Yeah I saw the topic.. I'm not looking for line by line assistance here just some direction |
| 00:09.11 | toerkeium | does anyone knows of an example of calling AGI from php? |
| 00:11.42 | quid2478 | Is there any code out there that interfaces a website with Asterisk... as in a new user can create an Acct on * via a web interface? |
| 00:11.58 | quid2478 | In other words "sign up" to your box? |
| 00:12.48 | *** join/#asterisk Terlouw (n=Terlouw@80.126.223.172) |
| 00:13.48 | Terlouw | does anyone know of a voip provider that offers CID "spoofing" ... ? |
| 00:14.01 | engineeer | voipjet |
| 00:14.19 | Terlouw | that was fast ... :) |
| 00:14.26 | engineeer | your welcome |
| 00:15.22 | engineeer | there is even a script to do it on the fly |
| 00:15.47 | Terlouw | that would be nice... |
| 00:16.04 | engineeer | search for cidspoof.agi |
| 00:16.50 | quid2478 | who does Voipjet use as a provider... their link to the UK was out a few nights ago, kinda sucked |
| 00:17.28 | Strom_C | cidspoof.agi? it's two lines in extensions.conf |
| 00:17.42 | Strom_C | you don't need a bloody AGI script for that |
| 00:18.02 | quid2478 | Strom: Yeah, but a PITA if you want to change it every time |
| 00:18.07 | engineeer | the agi allows U to dial an ext put in the "from" number then put in the "to" number on the fly |
| 00:18.13 | Strom_C | quid2478: pattern matching |
| 00:18.18 | Strom_C | substrings |
| 00:18.25 | quid2478 | Yeah, you're right/// I;ve seen it |
| 00:18.40 | quid2478 | I did see one AGI script that would generate a random CID based on the area code you are calling |
| 00:20.15 | engineeer | voipjet does track the cid's on thier system though in call records |
| 00:20.30 | Strom_C | engineeer: so does EVERYONE |
| 00:24.19 | docelmo | ya seriously |
| 00:28.53 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 00:30.16 | andymul | Anyone interested in some PHP/Asterisk work please PM me |
| 00:31.44 | Sponge_bob | its free PHP/Asterisk work too! |
| 00:32.26 | andymul | I'll pay in Monopoly money of your color choice |
| 00:32.29 | docelmo | I wrote parts of phpagi what do you need done? Also how much do you wanna pay? |
| 00:32.36 | andymul | docelmo, PM me |
| 00:33.03 | andymul | I actually have a few AGI scripts I need written, and I use PHPAGI right now |
| 00:34.15 | livesNbox | hey guys ever since I enabled my digium te100p card my system seems to be having a problem playing recordings back. they are really choppy and slow |
| 00:34.53 | livesNbox | any ideas ? |
| 00:34.55 | andymul | You running the latest Zaptel? |
| 00:34.58 | Strom_C | livesNbox: TE100P? |
| 00:36.12 | livesNbox | sorry: TE110P |
| 00:36.24 | livesNbox | andymul: yes just downloaded and compiled a day or two ago |
| 00:36.27 | Strom_C | ok, I was going to say... |
| 00:36.44 | andymul | Does anyone in here have any high volume servers? Sometimes through AMI "show channels" reports incorrect data. |
| 00:36.46 | Strom_C | livesNbox: zaptel 1.2 release branch or trunk? |
| 00:37.03 | livesNbox | 1.2.6 |
| 00:37.07 | livesNbox | release |
| 00:37.14 | livesNbox | (i think) :) |
| 00:37.51 | Strom_C | run zttest |
| 00:38.44 | livesNbox | what am I looking for ? |
| 00:39.01 | livesNbox | -460.119629% -459.985352% -459.716797% -459.521484% -459.472656% |
| 00:39.03 | livesNbox | so far |
| 00:39.06 | Strom_C | what the hell!? |
| 00:39.07 | livesNbox | heheh |
| 00:39.07 | *** join/#asterisk evilbit (i=hhoffman@gateway/tor/x-7a150cf2d9469550) |
| 00:39.08 | Strom_C | something is seriously bonkers with your system |
| 00:39.13 | Qwell | at least it's kinda consistent :P |
| 00:39.14 | hads | heh |
| 00:39.14 | livesNbox | agreed.. |
| 00:39.19 | livesNbox | what should I look at ? |
| 00:39.32 | livesNbox | I'm just trying to setup this card to work with a PRI |
| 00:39.38 | evilbit | hi, is spandsp still available and meant to be used for faxing? or is there something else? |
| 00:39.47 | Strom_C | livesNbox: example of good results: |
| 00:39.48 | Strom_C | 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 100.000000% 100.000000% |
| 00:39.56 | livesNbox | showoff |
| 00:39.58 | livesNbox | ! |
| 00:40.09 | Strom_C | you asked what you were supposed to be looking for |
| 00:40.31 | livesNbox | heh |
| 00:40.42 | livesNbox | any ideas what I have gotten messed up ? |
| 00:41.07 | Strom_C | did you say you were running trixbox, or is this a regular linux install with asterisk and freepbx |
| 00:41.23 | livesNbox | trixbox.. but if you can just point me in the right direction |
| 00:42.16 | Strom_C | well, this may just be coincidence, but I had RAM in a box die immediately after installing trixbox last week (client wanted an AMP-ectomy, so I wanted to see if it was easy to do) |
| 00:42.37 | Terlouw | another question :) |
| 00:42.38 | livesNbox | the box runs fine until I load the zap modules. |
| 00:42.50 | Terlouw | i signed up for voipjet |
| 00:42.51 | Strom_C | livesNbox: what are the specs on the box? |
| 00:43.01 | Terlouw | and i can "spoof" the phone number |
| 00:43.04 | Terlouw | but.... |
| 00:43.06 | livesNbox | I don't exactly know -- but it's pretty beefy.. |
| 00:43.10 | Terlouw | how do i remove the +1 ? |
| 00:43.12 | livesNbox | dual processors.. I think 2 GB of ram.. |
| 00:43.18 | Terlouw | it needs to be +31 |
| 00:43.19 | livesNbox | like 750GB sata raid drives |
| 00:43.48 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
| 00:43.54 | Strom_C | Terlouw: voipjet is a U.S. provider... |
| 00:44.11 | Terlouw | so? |
| 00:44.20 | engineeer | +1 only shows on a cell phone wont show or reg phone or at least it does not show on mine |
| 00:44.46 | docelmo | yes |
| 00:45.24 | Strom_C | livesNbox: for shits and giggles, i'd like to see whether the problem exists on a vanilla linux / asterisk install...no trixbox, no freepbx |
| 00:46.34 | livesNbox | I don't have that luxury unfortunately. |
| 00:46.46 | livesNbox | I've got to get this working how it sits. |
| 00:47.08 | Strom_C | livesNbox: back up the system, try it with plain vanilla linux, and then restore it? |
| 00:47.15 | Terlouw | no way to change the +1 ? |
| 00:47.54 | andymul | Terlouw: Caller ID is sent through the end carriers following a specification, so ultimately...no, you can't change it |
| 00:48.11 | Terlouw | :( |
| 00:48.43 | andymul | The format that is, most of the smaller VoIP providers will honor whatever CID you send them |
| 00:49.28 | Terlouw | any known where i can use a +31 cid? |
| 00:49.30 | Un1x | when you have this |
| 00:49.30 | Un1x | exten => s,1,Dial(Zap/1,15) |
| 00:49.41 | Un1x | thaty signifys ring zap channels 1 to 15 correct? |
| 00:49.52 | Qwell | no |
| 00:50.02 | hads | Un1x: No. It will ring Zap/! for 15 seconds |
| 00:50.08 | Un1x | ahh ok |
| 00:50.11 | hads | erm. Zap/1 |
| 00:50.19 | wunderkin | zap/!oneoneone |
| 00:50.24 | hads | :) |
| 00:50.31 | Un1x | weird tdm400p cards is backwards |
| 00:50.36 | Un1x | i have my phone plugged into port 4 |
| 00:50.42 | Un1x | and it still rings :p |
| 00:50.44 | *** join/#asterisk lakasub (n=lakasub@195.18.158.140) |
| 00:50.52 | Strom_C | Un1x: uh |
| 00:51.02 | Strom_C | Un1x: what modules do you have on it |
| 00:51.06 | Un1x | 2fxs 2 fxo |
| 00:51.09 | Un1x | the tdm22b |
| 00:51.14 | Qwell | green is fxs |
| 00:51.34 | hads | Un1x: I don't mean to be rude, but you said you have read the book, if you had you would understand this. |
| 00:51.56 | Un1x | i didn't read the nabdook |
| 00:52.00 | Un1x | i read users manual lol |
| 00:52.05 | Un1x | im starting to read the handbook |
| 00:52.09 | Un1x | ive only finished chapter 1 :/ |
| 00:52.19 | Qwell | ~book |
| 00:52.21 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:52.25 | Un1x | these days the manuals/handbooks start gettuign bigger and bigger |
| 00:52.30 | Un1x | qwell thats the one im reading |
| 00:52.38 | Un1x | i have it downloaded, on my comp as PDF. |
| 00:53.02 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 00:53.21 | livesNbox | is it a no no to run ztdummy with real zaptel modules loaded ? |
| 00:53.28 | livesNbox | because I just noticed that's what i'm doing... |
| 00:53.31 | *** join/#asterisk Winkie (n=urmom@gateway.duclicsic.com) |
| 00:53.51 | Strom_C | livesNbox: uh, i would not do that |
| 00:53.54 | Qwell | livesNbox: it's a bit pointless, yeah |
| 00:53.57 | Strom_C | livesNbox: only run what you need |
| 00:54.02 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 00:54.02 | Un1x | Hmm Qwell wich card you using? |
| 00:54.03 | livesNbox | but it wouldn't hork the whole system right ? |
| 00:54.30 | Qwell | Un1x: 3x5" post |
| 00:54.54 | Un1x | :? wich card is that |
| 00:54.59 | hads | heh |
| 00:55.56 | Un1x | hmm |
| 00:55.59 | Un1x | Asterisk Business Edition™ if i buy that |
| 00:56.00 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 00:56.05 | Un1x | i get support for ASterisk :P? |
| 00:56.21 | Strom_C | you can also buy paid support for regular asterisk |
| 00:56.46 | Un1x | :O |
| 00:57.00 | Un1x | heh i should buy it have em help me configure this thing :P |
| 00:57.02 | Strom_C | digium hardware comes with free installation support |
| 00:57.33 | Un1x | hmmmm... yes installation to install the hardware |
| 00:57.36 | Un1x | but not to configure |
| 00:57.40 | Un1x | or get running ... |
| 00:57.48 | Strom_C | configuration of the hardware is included |
| 00:57.48 | Un1x | like they wouldn't help me with my dialplan i dont think so |
| 00:57.51 | Strom_C | dialplan configuration is not |
| 00:57.51 | Un1x | :O |
| 00:58.12 | Un1x | hmm maybe i'll give them a call and ask them if ive configured anything wrong tomorow,. |
| 00:58.18 | Strom_C | Un1x: if you're so desperate to get it working, hire me as a consultant |
| 00:58.23 | Un1x | roflmfao |
| 00:58.26 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 00:58.39 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 00:58.47 | Un1x | my phone is plugged into Two (2) FXO Modules (red) |
| 00:59.09 | Un1x | the rewd modules if im look at it right weird eh |
| 00:59.09 | Strom_C | your telephone set? |
| 00:59.10 | Un1x | yea |
| 00:59.10 | Strom_C | why? |
| 00:59.21 | Un1x | it's the only port it works on for some odd reason :S |
| 00:59.26 | Strom_C | uh |
| 00:59.29 | evilbit | telephone set gets plugged into FXS, correct? |
| 00:59.32 | Strom_C | yes |
| 00:59.38 | Strom_C | Un1x: and it gives a dialtone? |
| 00:59.39 | evilbit | k |
| 00:59.45 | Un1x | yea |
| 00:59.54 | evilbit | o_0 |
| 00:59.54 | Qwell | immediate=yes |
| 00:59.56 | Strom_C | Un1x: then it isn't plugged into an FXO module |
| 00:59.59 | Qwell | s,1,Dialtone() :D |
| 01:00.06 | hads | You may have the port/module numbering backwards. |
| 01:00.14 | Un1x | hmm wait |
| 01:00.21 | Un1x | well according to the pic |
| 01:00.29 | Un1x | port 1 on the module is port one onth e back of the thing |
| 01:00.35 | evilbit | so, it's been a while since I've used asterisk... is the fax stuff still spandsp? or is it something else? |
| 01:00.39 | Qwell | look at the card itself... |
| 01:00.40 | Un1x | oh wait nevermind im looking at it upside down lol |
| 01:00.59 | Un1x | but serioulsy i made calls earlier while it was plugged into port 4 |
| 01:01.05 | Un1x | weird tho i had a tone and everything |
| 01:01.08 | Strom_C | uh, i don't believe that |
| 01:01.08 | Qwell | ~lart Un1x |
| 01:01.09 | Un1x | just noe one else could call me |
| 01:01.32 | Un1x | :| |
| 01:01.39 | Un1x | ~lart Qwell |
| 01:01.42 | Un1x | heh |
| 01:02.00 | Un1x | anyway i gotta go i'll speak yo you guys later imma get this printed |
| 01:02.01 | Strom_C | Qwell is invincible |
| 01:02.06 | Qwell | okay, I call BS on Quizno's commercials |
| 01:02.09 | Un1x | or go buy a print copy of ASterisk TFOT and read it |
| 01:02.14 | Qwell | their subs DO NOT look ANYTHING like that |
| 01:04.22 | livesNbox | like when I dial my voicemail it says Pa a a a a a a a s s s s s s s s s s s w ww o o o o o r r r r d dd d d d |
| 01:04.30 | livesNbox | any idea what would cause something like that? |
| 01:04.47 | quid2478 | Lives: Are you running VMWare? |
| 01:04.50 | Strom_C | livesNbox: is the jitter issue only when calling recordings? |
| 01:04.55 | livesNbox | yes only recordings |
| 01:04.58 | livesNbox | and no, not vmware |
| 01:05.11 | Strom_C | are zaptel and the disk controller sharing an interrupt maybe? |
| 01:05.14 | livesNbox | but it's also like a lot slower than it should be |
| 01:05.22 | hads | Wouldn't -460% on zttest cause that? ;) |
| 01:05.26 | livesNbox | i mean "password" might take 2 seconds to say.. this takes like 6.. |
| 01:05.36 | livesNbox | hads: that's what i"m thinking |
| 01:05.39 | livesNbox | Strom_C: how do I check ?! |
| 01:05.48 | Strom_C | livesNbox: first off, lsmod | grep zaptel |
| 01:05.52 | Strom_C | and tell me what it says |
| 01:06.01 | livesNbox | zaptel 196740 57 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 01:06.02 | livesNbox | crc_ccitt 6081 1 zaptel |
| 01:06.11 | Terlouw | what is the best provider for a +44 number ? (and with best... i mean cheapest... :) ) |
| 01:06.21 | Strom_C | why the crap do you have anything but zaptel and wcte11xp loaded? |
| 01:06.36 | livesNbox | Strom_C: because I am a fool! |
| 01:06.45 | livesNbox | i'll rmmod them for now |
| 01:07.26 | livesNbox | zaptel 196740 51 wcte11xp |
| 01:07.26 | livesNbox | crc_ccitt 6081 1 zaptel |
| 01:07.30 | Un1x | ive only got zaptel and wctdm |
| 01:07.31 | Un1x | nothing else |
| 01:07.51 | hads | That's lovely Un1x |
| 01:09.41 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 01:09.47 | livesNbox | so how do I tell if it's an interrupt problem ? |
| 01:10.01 | Strom_C | livesNbox: did removing the other modules do anything? |
| 01:10.25 | livesNbox | no |
| 01:10.53 | Qwell | I remember somebody having this problem before... |
| 01:11.00 | Qwell | if they took the card out, it worked fine |
| 01:11.10 | Qwell | or, perhaps, even if they unloaded the zaptel module |
| 01:11.21 | wunderkin | is the t1 hooked up to the telco and receiving timing? |
| 01:11.35 | Qwell | I think it ended up being a hardware problem... |
| 01:11.46 | livesNbox | It's hooked up and should be working.. if I plug it into my old nortel switch it comes back up and works. |
| 01:11.54 | Qwell | livesNbox: did it just start happening? Is it a new card? |
| 01:11.54 | Strom_C | wunderkin: oh duh, of course....livesnbox, where do you have zaptel.conf configured to get timing from? |
| 01:12.02 | wunderkin | if you do, so you dont |
| 01:12.55 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:13.02 | livesNbox | Strom_C: I'm not sure.. |
| 01:13.19 | Strom_C | livesNbox: show me your span statement in zaptel.conf |
| 01:13.30 | livesNbox | span=1,1,0,esf,b8zs |
| 01:13.43 | Strom_C | ok, so it |
| 01:13.51 | Strom_C | it's set to get timing from the pri |
| 01:13.59 | livesNbox | ok.. |
| 01:14.09 | livesNbox | maybe that's wrong ? |
| 01:14.18 | Strom_C | livesNbox: what happens differently when it's plugged into the nortel switch? |
| 01:14.21 | Strom_C | and who is the telco? |
| 01:14.39 | livesNbox | SBC |
| 01:14.52 | Strom_C | which flavor of SBC? |
| 01:14.54 | livesNbox | and I dunno what happens differently... |
| 01:15.00 | Qwell | Strom_C: strawberry |
| 01:15.02 | Qwell | duh |
| 01:15.04 | Strom_C | mmmm |
| 01:15.12 | Strom_C | Strawberry Bell Corporation |
| 01:15.12 | livesNbox | I don't konw |
| 01:15.13 | livesNbox | know |
| 01:15.20 | Strom_C | livesNbox: which state are you in |
| 01:15.38 | livesNbox | OH |
| 01:15.44 | Strom_C | ok, so SBC Midwest |
| 01:15.45 | livesNbox | as in ohio.. not as in "OH I GET IT!" |
| 01:15.53 | Strom_C | are you on a 5E or a DMS? |
| 01:16.08 | livesNbox | no sure.. |
| 01:16.11 | livesNbox | I think 5e |
| 01:16.14 | Strom_C | sigh |
| 01:16.19 | livesNbox | I've definitely heard the sbc guys talk about 5e |
| 01:16.29 | Strom_C | what area code and prefix does your DID pool live in? |
| 01:16.54 | livesNbox | 937264 |
| 01:17.14 | Strom_C | dayton, ohio |
| 01:17.17 | Strom_C | 5ESS |
| 01:17.19 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 01:17.24 | livesNbox | yeah that's it |
| 01:17.26 | livesNbox | ;) |
| 01:18.08 | livesNbox | thanks a lot for taking the time to work on this with me.. I really appreciate it |
| 01:18.15 | robl^ | Dayton, OH? ACK!!!!! *flashback of nightmares in Kettering, OH* |
| 01:18.36 | livesNbox | hey I live in kettering |
| 01:18.43 | livesNbox | boyeyeyeyeeeee |
| 01:18.46 | Strom_C | livesNbox: pastebin the output of lspci |
| 01:18.57 | Strom_C | ~pb |
| 01:18.58 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 01:20.19 | livesNbox | waiting for pastebin.com...... |
| 01:20.30 | Strom_C | try pastebin.ca |
| 01:21.10 | livesNbox | http://pastebin.ca/107485 |
| 01:22.32 | Strom_C | also pastebin cat /proc/interrupts |
| 01:23.22 | livesNbox | http://pastebin.ca/107489 |
| 01:23.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:24.48 | Qwell | I still think it's a hardware issue |
| 01:25.03 | Strom_C | yeah, i cant see anything in the config that might be a problem |
| 01:25.08 | livesNbox | Qwell: do you mean a bad card or ? |
| 01:25.12 | Qwell | or something |
| 01:25.22 | Qwell | is it a fairly new card? |
| 01:25.38 | livesNbox | brand new.. never been used |
| 01:25.50 | livesNbox | so it "could" be bad out of the box. |
| 01:26.15 | Strom_C | livesNbox: can you test it in another box? |
| 01:26.22 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 01:26.31 | livesNbox | yeah I could tommorrow... |
| 01:26.36 | wunderkin | livesNbox, so do you or do you not have something plugged into it |
| 01:26.37 | Strom_C | like...something less insanely hemorrhaging-edge than what it's in now? |
| 01:26.41 | livesNbox | are we officially out of ideas ? |
| 01:26.41 | *** join/#asterisk RoyKa (n=roy@122.80-203-78.nextgentel.com) |
| 01:26.44 | Qwell | I would call Digium tomorrow (after testing in another box), and see if they can find out what's wrong |
| 01:27.57 | *** part/#asterisk popvoxdave (n=popvoxda@c-71-206-59-174.hsd1.md.comcast.net) |
| 01:29.30 | livesNbox | ok.. I'll do that then.. but everything looks setup OK ? |
| 01:29.39 | livesNbox | the timing and all that |
| 01:29.50 | wunderkin | yes if you have something plugged in |
| 01:29.52 | Strom_C | livesNbox: is there a PRI plugged into it right now? |
| 01:30.03 | livesNbox | yes |
| 01:30.46 | Strom_C | yep...time for the ol' eight seven seven linux me trick |
| 01:30.53 | livesNbox | the led on the back of the card blinks faster and faster until it's almost solid.. then slows down until it's almost solid off for a second or so.. then repeats |
| 01:31.08 | Strom_C | livesNbox: did you check your t1 cable? |
| 01:31.47 | livesNbox | well if I unplug it from my card in my asterisk box and plug it into my nortel switch the switch works. |
| 01:31.57 | livesNbox | and it looks like the correct pinout.. |
| 01:31.59 | livesNbox | 12 45 |
| 01:32.21 | Strom_C | is it a factory-made cable or is it the crimp-it-yourself special? |
| 01:33.00 | livesNbox | well I'm taking the 2 foot cable SBC installed and coupling it to a factory made cat5 cable |
| 01:33.06 | livesNbox | that's about 20 feet long |
| 01:33.20 | Strom_C | didn't sbc give you a smartjack? |
| 01:33.43 | livesNbox | no.. comes right off a 66block and terminates to a rj45 plug |
| 01:33.47 | livesNbox | male |
| 01:34.07 | Strom_C | I'd like to officially shoot whoever installed that in the face |
| 01:34.17 | livesNbox | hahah |
| 01:34.32 | livesNbox | I'd settle for just taking my business away from them |
| 01:34.47 | Strom_C | I like SBC |
| 01:35.10 | Strom_C | they're worlds better than...oh i don't know...the former GTE bits of Verizon. or god forbid Sprint |
| 01:35.13 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 01:35.19 | livesNbox | heh.. |
| 01:35.48 | tengulre11 | Hi,all! good monday! |
| 01:36.23 | sevard | still sunday. |
| 01:36.40 | tengulre11 | :( |
| 01:36.42 | Strom_C | bruises. delivered. |
| 01:36.50 | sevard | it's motherfucking 105 degrees. |
| 01:36.57 | sevard | in minnesota. |
| 01:37.01 | sevard | :'( |
| 01:37.16 | sevard | mmmmmm yeah baby hurt me more. |
| 01:37.37 | tengulre11 | anybody know how to connect two asterisk with IAX2? (ServerA In A City, ServerB in B city, they have static ip). |
| 01:37.50 | Strom_C | tengulre11: it's ridiculously easy |
| 01:38.11 | sevard | tengulre11: it's very simple, too simple of a question to ask really, check out voip-info.org |
| 01:38.17 | tengulre11 | Strom_C: can you give me some tips! |
| 01:38.24 | Strom_C | tengulre11: read the following |
| 01:38.25 | Strom_C | ~docs |
| 01:38.27 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:38.27 | Strom_C | ~book |
| 01:38.28 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:38.47 | tengulre11 | but I can not registry ServerB users in Server A. |
| 01:39.05 | Strom_C | what do you mean |
| 01:39.17 | livesNbox | well.. I unloaded the zaptel module that card uses (wcte11xp) and restarted asterisk and now the recordings are sweet again. |
| 01:39.38 | Strom_C | livesNbox: yeah, like we said...call digium |
| 01:39.39 | wunderkin | so it was plugged in bot not active |
| 01:39.48 | livesNbox | yeah |
| 01:40.00 | wunderkin | thanks for the info thats your problem |
| 01:40.14 | tengulre11 | I want registry a IAX2 user ,like 8001:8001@ServerB.IP/8001 , but I get Timeout in 'iax2 show registry' |
| 01:40.14 | livesNbox | what is ? |
| 01:40.38 | Strom_C | tengulre11: you can't do that unless you specify the user on that specific server |
| 01:40.50 | Strom_C | tengulre11: iax2 is for trunking; it does not make two asterisk boxes behave as a single box |
| 01:41.07 | tengulre11 | Strom_C: OH! |
| 01:41.10 | wunderkin | livesNbox, ill try again, is the t1 circuit active and working? |
| 01:41.23 | livesNbox | wunderkin yes |
| 01:41.45 | livesNbox | but when I activate that card the recordings get choppy and slow... |
| 01:42.05 | tengulre11 | Strom_C: do u think which protocol can be usinged! |
| 01:42.21 | Strom_C | tengulre11: I already gave you a solution |
| 01:43.07 | tengulre11 | but I can not found any useful in that single doc! |
| 01:43.22 | sevard | holy shit. |
| 01:43.23 | wunderkin | livesNbox, so you are taking timing from the telco, the circuit was plugged in and working, you can make calls on it |
| 01:43.31 | Strom_C | <Strom_C> tengulre11: you can't do that unless you specify the user on that specific server |
| 01:43.38 | Strom_C | thanks |
| 01:43.45 | tengulre11 | :( |
| 01:43.56 | livesNbox | wunderkin: no -- never able to make calls through the asterisk server. |
| 01:44.05 | wunderkin | livesNbox, why |
| 01:44.15 | livesNbox | wunderkin: not sure. |
| 01:44.24 | livesNbox | I kind of assumed it was related to the funky recording playback. |
| 01:44.32 | livesNbox | and the -459% zttest results |
| 01:44.47 | wunderkin | umm mk |
| 01:44.55 | tengulre11 | Strom_C : can you give me more detail info. Thanks very very ....very much! I come from china. my english too bad! so.. |
| 01:44.58 | Strom_C | wunderkin: I'm fairly sure it's a hardware problem, hence "call digium" |
| 01:45.04 | livesNbox | am I off base ? |
| 01:45.09 | Strom_C | tengulre11: read the documentation |
| 01:45.09 | tengulre11 | or sample! |
| 01:45.11 | wunderkin | ok i see, well i just wanted to make sure it was getting timing |
| 01:45.19 | Strom_C | tengulre11: or hire me as a consultant :) |
| 01:45.21 | wunderkin | i guess i was not here for the rest |
| 01:45.24 | sevard | Strom_C: give him some more detail info. |
| 01:45.28 | livesNbox | wunderkin: andI really don't know.. |
| 01:45.49 | livesNbox | wunderkin: I am guessing that since Strom_C knew it was a 5e the timing stuff was OK. |
| 01:45.52 | tengulre11 | Strom_C: :-D), how much are you want? |
| 01:46.13 | sevard | tengulre11: I can do the same thing you're asking Strom_C to do but I'll charge you 50% less |
| 01:46.24 | litage | is it possible to use sendText() on asterisk's CLI? |
| 01:46.38 | Un1x | OMFG |
| 01:46.43 | tengulre11 | sevard: LOL!!!! |
| 01:46.44 | Un1x | i fucked up my system NOOOO! |
| 01:46.53 | livesNbox | should I try switching the timing source or something ? |
| 01:48.31 | livesNbox | wunderkin? |
| 01:49.10 | rob0 | Let's see a bidding war! |
| 01:49.24 | wunderkin | just call digium |
| 01:49.26 | sevard | I bid rob0's girlfriend! |
| 01:49.50 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 01:49.56 | livesNbox | heh.. ok.. Thanks for your help guys.... "i'll be back..." |
| 01:50.02 | livesNbox | but right now I got to GET TO DA CHOPPA! |
| 01:50.17 | sevard | .. |
| 01:51.27 | livesNbox | http://babychoppa.ytmnd.com |
| 01:51.40 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-148-103.client.mchsi.com) |
| 01:52.11 | sevard | you're the man now, dog! |
| 01:52.25 | Strom_C | punch the keys for god's sake! |
| 01:53.12 | sevard | YTMND?!?!? |
| 01:53.34 | Strom_C | sevard: did you ever see "you're the voicemail now, dog:? |
| 01:53.40 | sevard | Nope |
| 01:53.57 | sevard | Silly phone phreak. |
| 01:54.14 | Strom_C | http://voicemailnowdog.ytmnd.com/ |
| 01:54.58 | sevard | hahahahahahah |
| 01:55.08 | Qwell | That better not be running sip |
| 01:55.08 | sevard | that's a cisco 7960 if i'm not mistaken |
| 01:55.39 | Qwell | sevard: 7960, non g :p |
| 01:55.54 | Strom_C | ive got two non-G 7960s |
| 01:56.05 | Strom_C | my speakerphone button is labeled SPEAKER |
| 01:56.13 | Strom_C | my messages button is labeled MESSAGES |
| 01:56.28 | Qwell | eh? |
| 01:56.37 | Qwell | speaker button says speaker? |
| 01:56.48 | Strom_C | the label above it says SPEAKER |
| 01:56.53 | Qwell | You have a china knockoff :P |
| 01:56.55 | Strom_C | the actual button has a speaker pictogram on it |
| 01:56.56 | Qwell | oh, ok |
| 01:57.46 | sevard | the speeddial button I have for Strom_C is a pictogram of two figures buggering eachother |
| 01:57.54 | Strom_C | awww |
| 01:58.06 | Strom_C | i get my own buttfucking pictogram |
| 01:58.07 | Strom_C | <3 |
| 01:58.21 | sevard | stop hitting me with large books, jerk. |
| 01:58.54 | Strom_C | it's only a yellow pages |
| 01:59.03 | Strom_C | it's not like I'm hitting you with Telcordia SR-2275-4 |
| 01:59.24 | rob0 | In this town the yellow pages isn't that big. |
| 01:59.30 | sevard | same |
| 01:59.47 | sevard | like 8 fly swatters |
| 01:59.50 | Qwell | in LA, they deliver it by freight |
| 02:00.08 | rob0 | oh yeah ... watch out for those LA books |
| 02:00.17 | tengulre11 | Qwell: Nice to meet you! |
| 02:00.24 | sevard | manhatten books crush children |
| 02:00.44 | Strom_C | at least they break the los angeles book up into multiple directories |
| 02:00.55 | Strom_C | the las vegas metro area has ONE yellow pages volume |
| 02:01.01 | Strom_C | it's bonkers-huge |
| 02:01.11 | Qwell | only because of all the "escort services" |
| 02:01.18 | Qwell | there are only like 8 people that live in vegas |
| 02:01.20 | sevard | yeah, they could drop the las vegas yellow pages book on iraq |
| 02:01.37 | rob0 | haha KC's was pretty big when I lived there (more than 10 years ago) |
| 02:01.41 | Qwell | tengulre11: Don't msg me |
| 02:01.57 | tengulre11 | :( |
| 02:02.13 | rob0 | but I think they were starting to split it up then. |
| 02:02.48 | sevard | I just noticed my new white pages section comes complete with not only bold print but apparently you can buy highlited entries |
| 02:02.50 | rob0 | I left in time to miss the 10-digit local dialing :) |
| 02:03.09 | sevard | so it looks like you ran a friggen highlighter over a particular buisness |
| 02:03.25 | sevard | way to go assholes, you just made an already almost useless tool totally useless, I threw it out. |
| 02:03.30 | tengulre11 | Qwell: how can connect two asterisk?(ServerA in city A, ServerB in city B, a user in A want to dial other users in B) |
| 02:03.42 | Qwell | tengulre11: hire Strom_C |
| 02:03.43 | *** join/#asterisk fritz5150 (n=erik@72.174.226.238) |
| 02:03.53 | Strom_C | i'm digium-certified! |
| 02:04.00 | tengulre11 | :( |
| 02:04.05 | Qwell | Strom_C: heh |
| 02:04.05 | tengulre11 | I m a student! |
| 02:04.15 | partition | I'm a potato! |
| 02:04.25 | partition | yes, Strom has been dCAPitated |
| 02:04.26 | sevard | Holy shit, a talking potato(e)! |
| 02:04.28 | Strom_C | I'm a meatbag! |
| 02:04.36 | sevard | <PROTECTED> |
| 02:04.55 | sevard | i'm friggen reference man i swear. |
| 02:05.02 | Strom_C | spaceballs?! |
| 02:05.07 | Strom_C | oh shit...there goes the planet |
| 02:05.08 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 02:05.08 | sevard | no. |
| 02:05.18 | tengulre11 | nobody can help me! |
| 02:05.28 | sevard | the first one was dan quale the second one was the fifth element. |
| 02:05.44 | *** part/#asterisk fritz5150 (n=erik@72.174.226.238) |
| 02:05.48 | Strom_C | sevard: the line in fifth element is "I am a meat popsicle" |
| 02:05.51 | rob0 | That's MISTER Potato to YOU. |
| 02:05.56 | sevard | jesus are you serious? |
| 02:06.02 | Strom_C | sevard: yes |
| 02:06.12 | sevard | Strom_C: I'm going in to the doctor in a week or so to see if I need a hearing aid |
| 02:06.14 | tengulre11 | all people talking here for money? |
| 02:06.18 | sevard | So that would make sense. |
| 02:06.30 | Strom_C | tengulre11: what you're asking for is ridiculously basic |
| 02:06.34 | sevard | tengulre11: Yes. Strom_C also offers sexual services iirc. |
| 02:06.41 | Strom_C | tengulre11: you can easily read the documentation and figure this out |
| 02:06.45 | Qwell | sevard: you most certainly rc |
| 02:07.04 | Strom_C | tengulre11: so if you are unwilling to do that, then most of us are unwilling to do such a mundane task for free |
| 02:07.28 | tengulre11 | Strom_C: Thanks! |
| 02:08.05 | sevard | bbiab - heading over into a nice cool basement. |
| 02:08.13 | Strom_C | tengulre11: please stop ending all your sentences with exclamation points! it's really really annoying! |
| 02:08.25 | sevard | <tengulre11> I hate everyone in here!!! don't talk with me! |
| 02:08.39 | sevard | ^ /msg |
| 02:08.41 | tengulre11 | :( |
| 02:09.01 | tengulre11 | I m only point sevard! |
| 02:09.09 | sevard | ...what |
| 02:09.18 | tengulre11 | <sevard> Strom_C is expensive. I told you I'm cheaper. |
| 02:09.25 | sevard | Sexually. |
| 02:09.55 | Strom_C | also, sevard asks me for help all the time |
| 02:10.14 | sevard | and I might be have to take twice as long on acount of you being so stupid. thus twice the charge. |
| 02:26.09 | rob0 | I'll charge triple *and* destroy your system completely ... but I'm good at sales so the PHB will be happy with me. |
| 02:26.56 | *** join/#asterisk icyfire0573 (n=jason@ool-44c1d110.dyn.optonline.net) |
| 02:27.50 | Qwell | I'll charge 4 times as much, and you'll still have the EXACT same problem |
| 02:28.00 | Qwell | and I'll do it within 5 minutes |
| 02:28.42 | icyfire0573 | I'm having serious problems doing * to * connections. I've tried various parts and I've tried looking on google, but I'm either not searching for the right thing or I just can't follow it. I currently have a configuration that gives me a no registration for peer error but returns nothing on the ''client'' asterisk machine. |
| 02:29.22 | Qwell | icyfire0573: You don't really want * boxes to register with each other. Are the IPs static? |
| 02:31.30 | icyfire0573 | Qwell its kindof static. I'm using a vpn tunnel but I cant gaurantee every time the vpn tunnel comes up it will have the same IP. |
| 02:32.44 | Qwell | Do they have DNS? |
| 02:32.54 | Qwell | I would change it from host=dynamic to the hostname |
| 02:34.46 | *** join/#asterisk ged (n=ged@dsl093-040-165.pdx1.dsl.speakeasy.net) |
| 02:35.10 | Strom_C | holy god, am I in the mood for fish tacos |
| 02:35.14 | icyfire0573 | I'm working on that right now, since its a VPN tunnel and i'm the endpoint I think I can make it static given half a chance. |
| 02:35.20 | icyfire0573 | Fish Tacos? |
| 02:35.35 | Strom_C | however, I wonder if there's a closer fish taco place than Rubio's in Glendale... |
| 02:35.52 | Qwell | Strom_C: You're in LA for christs sake...of course there is |
| 02:38.10 | icyfire0573 | They are both static now, and i have host host=10.9.0.1 (the remote host) in the configuration file for the ''local'' machine. I can get rid of the registration line though right? |
| 02:38.27 | Strom_C | Qwell: the taco stands within walking distance are closed |
| 02:38.42 | Qwell | lam |
| 02:38.43 | Qwell | e |
| 02:38.46 | Qwell | or lamb |
| 02:39.44 | icyfire0573 | How would I dial the remote server from the ''local'' one |
| 02:39.59 | Qwell | icyfire0573: You want to dial an exten on the remote server? |
| 02:40.03 | Qwell | ie, another user |
| 02:40.09 | icyfire0573 | Affirmative. |
| 02:41.13 | Qwell | IAX2/hostname/exten should work |
| 02:41.30 | Qwell | or IAX2/user@hostiniax.conf/exten@context |
| 02:42.29 | icyfire0573 | Thank you so much. |
| 02:42.43 | Qwell | bunch of different (valid) ways |
| 02:45.15 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:45.53 | yxa | for 1.2, do I need to load res_musiconhold.so ? |
| 02:46.03 | Qwell | yxa: Only if you want MoH |
| 02:46.31 | yxa | Qwell I do. but under modules.conf, it says noload |
| 02:46.45 | Qwell | So take out the noload |
| 02:48.48 | yxa | Qwell thanks |
| 02:50.28 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:01.09 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-59.dslextreme.com) |
| 03:12.09 | rowter | anyone know why with some phones asterisk does native bridge and with other no, I need to know when it hangs up, but with native bridge it drops one channel and make it one.. |
| 03:20.53 | *** join/#asterisk NEXTGroup (n=stevetho@adsl-70-240-211-215.dsl.hstntx.swbell.net) |
| 03:24.18 | rowter | there is no way to disable native bridge?!! |
| 03:27.08 | [TK]D-Fender | rowter : Native bridge is what * does on any channel that is not doing a reinvite. |
| 03:27.36 | [TK]D-Fender | rowter : Thats the whole point of a B2BUA which is the core definition of *. |
| 03:28.07 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 03:28.10 | russellb | no, that's not what a native bridge is ... a reinvite *is* a native bridge |
| 03:28.33 | [TK]D-Fender | russellb : ... or the exact opposite! |
| 03:28.52 | russellb | a native bridge is when the two channels are the same type, and they have a *native* way to bridge together |
| 03:28.55 | implicit | russelb: actually reinvite has to do with signalling only in SIP, native bridge can happen regardless or signalling path |
| 03:29.17 | [TK]D-Fender | implicit : I was using that as a sample, but sure. |
| 03:29.31 | russellb | a reinvite is a type of native bridge. |
| 03:29.37 | implicit | noname32, wrong |
| 03:29.41 | implicit | sorry |
| 03:29.41 | rowter | I see, there si some way to disable for sip?, cause the h, exten is executed right away.. |
| 03:29.44 | implicit | no russellb you are wrong |
| 03:29.51 | implicit | a reinvite is NOT a type of a native bridge |
| 03:30.08 | implicit | reinvite is a retransmission of an INVITE message to renegotiate SDP or something else in the dialog |
| 03:30.10 | russellb | implicit: are you serious? i know what i am talking about :) |
| 03:30.22 | russellb | yes, which is the SIP method of doing a native bridge |
| 03:30.22 | implicit | yes i am serious, i think you are mixing up terms though |
| 03:30.30 | implicit | no it is not actually the SIP method |
| 03:30.36 | implicit | it is part of what is used in * to do a native bridge |
| 03:30.44 | blitzrage | russellb: pffft... like you've programmed any significant parts of Asterisk... |
| 03:30.56 | implicit | but they are quite different things |
| 03:30.58 | [TK]D-Fender | blitzrage : ! ! ! |
| 03:30.59 | Qwell | ooo! |
| 03:31.13 | Qwell | blitzrage, russellb: Remember that dive bar we went to? :D |
| 03:31.23 | blitzrage | [TK]D-Fender: I'm moving out of hell!^H^H^H^H^HMississauga! |
| 03:31.25 | Qwell | When my boss lived in Anaheim...he was a regular there |
| 03:31.34 | implicit | hey Qwell how're you doing |
| 03:31.35 | blitzrage | Qwell: oh yes, that place rocked :) |
| 03:31.36 | [TK]D-Fender | blitzrage : OMGZ! Where to? |
| 03:31.48 | blitzrage | [TK]D-Fender: near Fort York in Toronto, right downtown |
| 03:31.59 | blitzrage | pretty pumped, should have a good view |
| 03:32.09 | rowter | now with native bridge there is no way I could know when a call is being really hangup, cause it executes h exten right way.. |
| 03:32.17 | [TK]D-Fender | blitzrage :: Yikes... right into "south-central".... get armed quick! |
| 03:35.22 | russellb | implicit: sorry, got a phone call. a "native bridge" in asterisk terminology is when you want to bridge two channels of the same type, and they have some native way of talking to each other directly. |
| 03:35.36 | russellb | so yes, a "reinvite", is how chan_sip does a native bridge. |
| 03:36.53 | russellb | i am confident i know what I am talking about in this case :) |
| 03:37.50 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.69.159) |
| 03:37.58 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 03:38.54 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.69.159) |
| 03:38.59 | *** join/#asterisk niteowldave (n=dave@203.82.162.40) |
| 03:40.50 | blitzrage | [TK]D-Fender: I don't understand why... seems like a pretty decent area |
| 03:41.03 | blitzrage | [TK]D-Fender: not like I"m at Jane/Finch |
| 03:44.15 | *** join/#asterisk mitcheloc (n=mitchelo@c-24-23-37-212.hsd1.ca.comcast.net) |
| 03:45.48 | [TK]D-Fender | blitzrage : I like it just a little outside the madhouse... |
| 03:46.14 | [TK]D-Fender | blitzrage : Nothing wrong with Mississauga. Far enough away for sanity, close enough to go in for a show. |
| 03:46.51 | [TK]D-Fender | blitzrage : Much like my bein in the West Island. We've got it all here, and a 15 minute drive to downtown. |
| 03:48.12 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 03:48.59 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 03:50.53 | *** join/#asterisk bmg505 (n=leon@dsl-146-41-248.telkomadsl.co.za) |
| 03:53.25 | blitzrage | [TK]D-Fender: yah well, with Mississauga, it takes me 2 hours by public transit to get downtown -- Mississauga is shit in my opinion (especially if you're a single bachelor like myself) |
| 03:54.46 | blitzrage | downtown is going to be great -- Molson Indy is right beside the apartment, the Ex is about 500m away, the beer festival is held at Fort York right across the road, and the Molson Amphitheatre is less a 1km away (already got tickets to see Tool in September), then there's the Air Canada Centre (Raptors/Leafs) and the Skydome (Jays) |
| 03:55.28 | blitzrage | oh, and apparently Jim says there's an open air ice rink right there too, and of course the park that runs along Lakeshore -- bike path starts at my door step pretty much (right across the road) |
| 03:55.38 | blitzrage | ...so yah, I'm excited :) |
| 03:55.57 | blitzrage | and I just finished reconciling my business end of year books! only took me all day, but it's done |
| 03:57.19 | [TK]D-Fender | blitzrage : Cool.... you should consider getting a licence.... or I guess with you bing in town now its time to get a riced-up Segway! ;) |
| 03:58.20 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 04:02.52 | rowter | why native bridge work on some sip phones and not in others? my cisco 7960 does not do native bridge but my grandstream it always do it when I answer with it. |
| 04:13.51 | *** join/#asterisk ghinckley6 (n=Spam@209.26.206.110) |
| 04:14.14 | MikeJ | hmmm |
| 04:14.18 | ghinckley6 | i need some help seting up a group on zap channels any one help |
| 04:14.51 | ghinckley6 | or just tell me where the manual is |
| 04:22.33 | [TK]D-Fender | ~book |
| 04:22.35 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:22.37 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:23.29 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 04:31.28 | rowter | anyone knows if there is some way to disable native bridge on rtp.c? |
| 04:32.26 | russellb | rowter: canreinvite=no in sip.conf |
| 04:33.17 | rowter | russellb, I have already that, but on some phones it does it.. let me see. |
| 04:35.21 | rowter | russellb, Local/101@agentringback-81b2,1 answered SIP/100-a006 it does it, and goes right away to h exten :( |
| 04:35.35 | rowter | russellb, Attempting native bridge of SIP/100-a006 and SIP/101-9052 |
| 04:36.05 | russellb | attempting does not mean it is actually done |
| 04:36.25 | russellb | chan_sip will refuse to do it |
| 04:36.49 | rowter | russellb, but it ends my deadagi and send to h exten.. :( |
| 04:37.12 | russellb | DeadAGI? what are you trying to do on a channel that is already hung up? |
| 04:38.34 | rowter | russellb, am trying to get the hangup stat.. after a dial.. |
| 04:39.09 | rowter | russellb, deadage exec a dial and after hangup updates a db. |
| 04:39.12 | russellb | so do it in the 'h' exten instead |
| 04:39.36 | russellb | or, optionally, you need to handle SIGHUP in your AGI script |
| 04:39.40 | russellb | that is what is killing your AGI |
| 04:40.40 | rowter | russellb, after answer , and attempt to bridge h is also excecuted.. |
| 04:40.58 | rowter | handle sighup., how can that be done? |
| 04:41.05 | russellb | i don't know, you tell me |
| 04:41.11 | russellb | it depends on the programming language you are using |
| 04:41.23 | russellb | you'll have to research signal handlers in the language you are using |
| 04:41.53 | rowter | ok russellb .. let me see |
| 04:43.12 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:44.42 | *** join/#asterisk parag_ast (n=root@dxb-b16259.alshamil.net.ae) |
| 04:45.04 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 04:50.15 | ghinckley6 | well got that ATFOT |
| 04:50.32 | ghinckley6 | it has a few chapter missing |
| 04:50.59 | ghinckley6 | any way figured out the group thing on my own not so hard |
| 04:52.14 | ghinckley6 | any one tell me where a good macro or scriptis to handle incoming calls |
| 05:01.06 | ghinckley6 | using playback comand how do you get to pause for a second |
| 05:01.52 | *** join/#asterisk andymul (n=andymul@cpe-69-203-217-237.nyc.res.rr.com) |
| 05:02.48 | blitzrage | ghinckley6: Wait() |
| 05:03.17 | russellb | well, using playback, you'd play a file of silence :D |
| 05:03.44 | blitzrage | pffft... if you wanna be fancy.... |
| 05:05.15 | ghinckley6 | must lookup wait |
| 05:05.40 | JT | did you download the single document version of TFOT |
| 05:05.44 | JT | one pdf in a zip |
| 05:05.51 | JT | i didn't find any chapters missing in that |
| 05:06.06 | ghinckley6 | yes i did hat was sarcasim |
| 05:06.37 | ghinckley6 | i actually also have the printed copy of the book |
| 05:06.52 | ghinckley6 | but it leaves a lot to be desired |
| 05:07.13 | ghinckley6 | chapters 7 and 8 or a waste of paper |
| 05:10.14 | ghinckley6 | awe think you for the wait tip |
| 05:10.17 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 05:10.35 | Kerry_G | anyone have voicemail attachments working with blackberrys? |
| 05:14.29 | parag_ast | Today i m saying with confirmation that ASTERISK CAN NOT BE USED FOR PRODUCTIONS |
| 05:14.55 | jbroome | wussy |
| 05:16.18 | *** join/#asterisk daysmen3 (n=primus@host86-140-208-99.range86-140.btcentralplus.com) |
| 05:17.22 | ghinckley6 | parag_ast I know of many asterisk sytems that are in use. I have several hundred plus seat call centers up and running on Asterisk all doin prodictive outbound dialing betting the hell out of the servers |
| 05:17.48 | ghinckley6 | Asterisk does work it is however a pain in the ass to setup |
| 05:18.12 | parag_ast | did u ever try DIGIUM FXO card |
| 05:18.17 | parag_ast | with asterisk |
| 05:18.23 | parag_ast | i implmented 4 places |
| 05:18.33 | parag_ast | all people wants to throw that boxes |
| 05:18.44 | parag_ast | It dosn't hangup |
| 05:18.57 | parag_ast | when i complained to digium they said |
| 05:19.12 | parag_ast | You are right parag, we hv not implemented for all the countries |
| 05:19.17 | Un1x | rofl |
| 05:19.18 | parag_ast | its still in devlopment |
| 05:19.21 | Un1x | im glad i live in us :P |
| 05:19.26 | Un1x | it works here perfectly |
| 05:19.34 | hads|home | ~enter |
| 05:19.36 | jbot | The enter key or return key is not a substitute for punctuation. Use a period '.', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 05:19.37 | ghinckley6 | yes have them in my system |
| 05:19.37 | parag_ast | I live in UAE..and tone plant is |
| 05:19.39 | file | even the call progress now for US is iffy... |
| 05:19.42 | parag_ast | ver different |
| 05:20.12 | ghinckley6 | well hell |
| 05:20.21 | parag_ast | BUT PRODUCT SHOULD BE ALWAYS STABLE |
| 05:20.32 | parag_ast | people can wait for email |
| 05:20.35 | parag_ast | but not for call |
| 05:20.36 | parag_ast | right |
| 05:20.39 | file | parag_ast: it's impolite to yell |
| 05:20.45 | Un1x | actualy i cant wait for either... |
| 05:20.49 | parag_ast | whenever |
| 05:20.52 | parag_ast | people try |
| 05:20.54 | ghinckley6 | you be running a us sytem before you know it. Bush has to years left in office give him a chance to invade ther as well |
| 05:21.07 | Un1x | lmao |
| 05:21.27 | Un1x | he can get reelected... |
| 05:21.27 | ghinckley6 | its what we do bomb people |
| 05:21.31 | ghinckley6 | no not really |
| 05:21.41 | ghinckley6 | luckly for us |
| 05:22.00 | hads|home | parag_ast: Disconnect supervision. |
| 05:22.08 | Un1x | actualy if i was Bush id invade those moslim schools that teach extremism rather tyhen go for countrys coz thats where it all begins according to BBC, and other news resources.. |
| 05:22.18 | parag_ast | Yes |
| 05:22.22 | *** join/#asterisk WeirdM (n=weirdm@udp079073uds.hawaiiantel.net) |
| 05:22.23 | parag_ast | DISCONNECT SUPERVISION |
| 05:22.24 | Corydon76-home | Saw a History Channel special about the apocalypse today, and it all became too obvious. |
| 05:22.29 | Corydon76-home | Bush is the Anti-Christ |
| 05:22.37 | ghinckley6 | yea well |
| 05:22.40 | ghinckley6 | probally |
| 05:22.45 | WeirdM | Where is a good place to get a card with FXS port? |
| 05:23.04 | WeirdM | I mean with an FXS port |
| 05:23.11 | parag_ast | I tried to change in zonedata.c |
| 05:23.14 | parag_ast | also but of no use |
| 05:23.44 | parag_ast | here in UAE our telco is using three switching device |
| 05:23.48 | parag_ast | Seimens, |
| 05:23.50 | ghinckley6 | ok Wait worked very good now only a few more |
| 05:23.51 | parag_ast | alkatel |
| 05:24.00 | ghinckley6 | details work out |
| 05:24.30 | ghinckley6 | UAE i would think the stuff is closer to British then anthinkg |
| 05:25.33 | parag_ast | it is closer to UK tone plant |
| 05:25.44 | ghinckley6 | yes British |
| 05:26.00 | ghinckley6 | Cable & Wirless set it all up after WW2 |
| 05:26.25 | ghinckley6 | its been up graded since but bascically the british stuff |
| 05:27.43 | Kerry_G | just because it doesnt work for YOU doesnt mean it doesnt work for thousands and thousands of other people |
| 05:28.15 | hads|home | Aye. |
| 05:35.31 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:36.27 | joelsolanki | Hello all. i have one question related to asterisk cdr. let me give the pastebin |
| 05:39.06 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:39.18 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 05:41.04 | *** join/#asterisk Supaplex (n=supaplex@shell.aros.net) |
| 05:42.18 | Supaplex | is there another channel for iaxcomm (~iaxclient)? If It's not OT here, I'm interested in suggestions for building on mac os x. I'm stuck at this bug: http://sourceforge.net/tracker/index.php?func=detail&aid=1531529&group_id=72851&atid=535894 |
| 05:45.46 | ghinckley6 | [outbound-long-distance] |
| 05:45.46 | ghinckley6 | exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 05:45.46 | ghinckley6 | exten => _91NXXNXXXXXX,2,Congestion( ) |
| 05:45.47 | ghinckley6 | exten => _91NXXNXXXXXX,102,Congestion( ) |
| 05:46.06 | ghinckley6 | that should do a long distance call correct |
| 05:46.17 | ghinckley6 | not a busy signal |
| 05:46.59 | ghinckley6 | and come up wit a 404 on the phone |
| 05:47.33 | parag_ast | ANYBODY TRIED WITH ASTERISK + QUINTUM |
| 05:54.13 | *** join/#asterisk LuckySeven (n=lucky7@218.208.207.191) |
| 05:56.08 | LuckySeven | Hi, I have just configured Asterisk with a basic dialplan. I have configured an [outgoing] context, when I press 9, I can dial out. However, the moment the call is bridged, I can hear an "echo" of my own voice. |
| 05:56.16 | LuckySeven | Is this something to do with echotraining? |
| 05:56.35 | parag_ast | asterisk1*CLI> show channels |
| 05:56.35 | parag_ast | Channel Location State Application(Data) |
| 05:56.35 | parag_ast | Zap/1-1 s-BUSY@macro-exten-v Up Busy() |
| 05:56.36 | parag_ast | SIP/10-0839cd20 (None) Up Bridged Call(Zap/3-1) |
| 05:56.36 | parag_ast | Zap/3-1 s@macro-dial:10 Up Dial(SIP/10&SIP/11&SIP/13&SIP/ |
| 05:56.36 | parag_ast | 3 active channels |
| 05:56.38 | parag_ast | 2 active calls |
| 05:56.59 | parag_ast | see |
| 05:57.02 | parag_ast | this problem |
| 05:57.16 | parag_ast | now Zap/1-1 and Zap/3-1 |
| 05:57.19 | parag_ast | got hanged |
| 05:57.28 | parag_ast | why is it happening |
| 05:57.42 | parag_ast | http://pastebin.ca/105545 |
| 05:59.23 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 05:59.34 | Un1x | lmao i tihnk my hdd just died :/ |
| 06:00.32 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 06:00.53 | joelsolanki | Hi all. i have question regarding cdr of asterisk. plz let me pastebin the cdrs |
| 06:01.32 | brookshire | parag: what version of asterisk? |
| 06:02.45 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 06:03.10 | joelsolanki | http://pastebin.ca/107974 |
| 06:03.36 | joelsolanki | please note the start and end time of all 3 calls. |
| 06:04.10 | joelsolanki | all 3 calls are origiated from same ip. i m confused how it is possible that one can make 3 calls at a time. |
| 06:04.20 | joelsolanki | customer has linksys pap2 with 2 ports in it. |
| 06:04.27 | joelsolanki | is this asterisk cdrs problem ? |
| 06:04.36 | yxa | guys, the CONF button on SIP does not actually using MeetMe right? |
| 06:04.50 | joelsolanki | i m using asterisk 1.2.0 |
| 06:04.52 | yxa | s/using/use |
| 06:05.03 | joelsolanki | any hints please |
| 06:05.52 | Strom_C | 1.2.0?? |
| 06:05.57 | Strom_C | upgrade!! |
| 06:06.43 | joelsolanki | yes |
| 06:06.57 | joelsolanki | will upgrade solve the issue. |
| 06:07.00 | joelsolanki | ? |
| 06:07.04 | Strom_C | maybe |
| 06:07.25 | joelsolanki | hmm ok. but did u check the cdrs which i pasted in pastebin ? |
| 06:07.30 | Strom_C | no |
| 06:07.46 | joelsolanki | plz check. if u could get some idea plz tell me |
| 06:07.54 | Strom_C | im on the phone |
| 06:08.04 | Qwell | Strom_C: You? On a phone? |
| 06:08.07 | joelsolanki | i m waiting then... |
| 06:10.22 | joelsolanki | also if my asterisk server has few calls running and if my power goes and server get down then calls will be disconnected but will that calls have the cdr ? |
| 06:10.36 | Qwell | uhh...no |
| 06:10.47 | Qwell | and you seriously can't expect them to be logged |
| 06:11.02 | Strom_C | joelsolanki: oh for god's sake, buy a UPS |
| 06:11.09 | daysmen3 | just wanted to know whether anyone has had success setting up asterisk in the UK for a call center - silly question maybe but hey why now |
| 06:11.18 | daysmen3 | ...why not |
| 06:11.20 | joelsolanki | oh |
| 06:11.45 | joelsolanki | Qwell: Strom_C : can u plz tell me the drawbacks when this thing happens ? |
| 06:12.19 | Qwell | joelsolanki: umm...lots |
| 06:12.30 | Qwell | complete and utter hardware fail, for one |
| 06:12.39 | creativx | thats always a winner |
| 06:12.54 | joelsolanki | ok and other major things ? |
| 06:12.58 | Qwell | joelsolanki: yes |
| 06:13.51 | joelsolanki | Qwell: can u plz tell me others. or can u redirect to some url so i can read it |
| 06:14.07 | Qwell | joelsolanki: google.com |
| 06:14.25 | joelsolanki | ok i will search. |
| 06:15.09 | joelsolanki | also i note that when my server gets powerdown and if few calls were running then in cdrs i found that calls didnt got ended. and asterisk billed wrongly to my customers. |
| 06:15.27 | joelsolanki | sometimes it was 8 to 12 hours calls which is not possible. |
| 06:15.31 | creativx | why in gods name would you allow your server to go down? |
| 06:15.33 | Strom_C | joelsolanki: why the hell would you not have a UPS on a production server? |
| 06:15.46 | creativx | hell/god etc |
| 06:15.55 | joelsolanki | Yes i have the UPS + power generator. |
| 06:16.00 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 06:16.23 | Supaplex | so you have a flakey $@$*( power supply that just cuts out? |
| 06:16.55 | joelsolanki | ups+ power generator are working but our sometimes the electricity voltage comes high / low and sometimes it fluctuates and that time server gets rebooted. |
| 06:17.15 | joelsolanki | i have complain lot of times to electricity provider but no soltuions :( |
| 06:17.25 | joelsolanki | so i m having bad time |
| 06:17.54 | creativx | hehe |
| 06:18.03 | creativx | how about an ups that can straighten out voltage spikes |
| 06:18.30 | joelsolanki | hmm yes. i m searching for some good ups |
| 06:18.45 | creativx | www.apc.com |
| 06:18.59 | creativx | should have what you need |
| 06:19.01 | Supaplex | two options. a true ups (not standby), or a power conditioner. orrrr... |
| 06:19.17 | creativx | your own powerplant |
| 06:19.21 | creativx | apc can provide that too |
| 06:19.43 | joelsolanki | yes i will surely look at it. |
| 06:20.20 | Supaplex | <sarcasm>But it must be a bug in asterisk!</sarcasm> |
| 06:20.38 | joelsolanki | but can u tell me when this occurs asterisk makes wrong cdrs ? |
| 06:21.02 | JT | joelsolanki: you need a double conversion online ups |
| 06:21.13 | JT | one that always generates power from battery voltage |
| 06:21.15 | joelsolanki | hmm yes. |
| 06:21.18 | JT | and always charges the batteries |
| 06:21.37 | joelsolanki | yes. |
| 06:22.11 | joelsolanki | does asterisk have stale calls ? |
| 06:26.47 | fuser | christ this is driving me nuts |
| 06:27.01 | fuser | why do some iax peers authenticate once a minute |
| 06:27.14 | fuser | i cant freakin debug anything with this noise on the console |
| 06:27.46 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 06:29.15 | *** join/#asterisk l0l1k (n=Freeman@212.248.91.229) |
| 06:29.23 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 06:30.02 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 06:30.22 | fuser | yeah, nooobody knows the answer to this one |
| 06:30.27 | fuser | identical iax.conf buildouts |
| 06:30.40 | fuser | but one will register, oh, say every hour or so |
| 06:30.52 | fuser | another will fill my console with garbage |
| 06:31.04 | yxa | is .gsm a wav format? |
| 06:31.15 | LuckySeven | fuser: are you using the same devices? |
| 06:31.26 | fuser | LuckySeven: multiple asterisk boxes |
| 06:31.53 | LuckySeven | fuser: how many boxes are connected to your main asterisk server? |
| 06:31.58 | fuser | hundreds |
| 06:32.13 | fuser | i have half a dozen that drive me insane |
| 06:32.35 | LuckySeven | fuser: Are all of these under your contrl? |
| 06:32.39 | LuckySeven | *control |
| 06:32.43 | fuser | yep |
| 06:32.58 | fuser | im considering upgrading a couple of them |
| 06:32.59 | JT | yxa: not, gsm is not a wav format |
| 06:33.02 | fuser | but they are so stable otherwise |
| 06:33.03 | JT | s/not/no/ |
| 06:33.14 | LuckySeven | I see, are they using the same config, all of them, how about the versions? |
| 06:33.20 | fuser | i really dont want to do that unless i have another reason to |
| 06:33.42 | fuser | LuckySeven: versioning is different |
| 06:34.07 | fuser | LuckySeven: three are running 1.2.7.1 |
| 06:34.09 | *** join/#asterisk ma_dzen (n=ma_dzen@217.66.17.141) |
| 06:34.12 | fuser | one is running 1.0.9 |
| 06:34.15 | LuckySeven | maybe there is a pattern to the authentication depending on the version. |
| 06:34.17 | fuser | im on 1.2.10 |
| 06:34.34 | LuckySeven | I have a few IAX devices here with me, nearly all of them authenticate every minute. |
| 06:34.40 | fuser | dont think it has much to do with the version unless there is some glitch that resurfaced from 1.0.9 |
| 06:34.45 | LuckySeven | I am guessing it is possibly for NAT compatibility. |
| 06:34.49 | fuser | but then i have some older boxen running the 1.0.9 w/o this problem |
| 06:35.02 | fuser | LuckySeven: even so i dont see where to modify this |
| 06:35.03 | LuckySeven | That is really strange. |
| 06:35.16 | fuser | you are telling me. i have been building these things for years |
| 06:35.37 | LuckySeven | :) |
| 06:37.14 | LuckySeven | Really sorry, can't help you out there. :( |
| 06:39.16 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 06:39.53 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 06:41.19 | ghinckley6 | [outbound-long-distance] |
| 06:41.19 | ghinckley6 | exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 06:41.19 | ghinckley6 | exten => _91NXXNXXXXXX,2,Congestion( ) |
| 06:41.19 | ghinckley6 | exten => _91NXXNXXXXXX,102,Congestion( ) |
| 06:41.23 | ghinckley6 | sorry |
| 06:41.28 | ghinckley6 | can i do this |
| 06:41.41 | LuckySeven | isn't that from the oreilly VoIP book? |
| 06:41.55 | ghinckley6 | ${arg1}&${ARG2} |
| 06:42.15 | ghinckley6 | yea sorry about that |
| 06:42.29 | LuckySeven | What is there to be sorry about. :) |
| 06:42.35 | ghinckley6 | the arg thing is what i am asking about |
| 06:42.44 | ghinckley6 | pated the wrong thing |
| 06:43.15 | LuckySeven | I see. Don't see a problem with that as long as arg1 and 2 is defined. Does both your phone ring? |
| 06:44.38 | ghinckley6 | well here is what i am trying to do i have every thing done exect incoming calls |
| 06:44.52 | ghinckley6 | i am to the point were it say enter the ext ... |
| 06:45.03 | ghinckley6 | how do i get to conect to the right ext |
| 06:45.19 | ghinckley6 | outbound works as does calling ext internaly |
| 06:45.21 | LuckySeven | what protocol are u using? |
| 06:45.31 | ghinckley6 | zap to sip |
| 06:45.41 | LuckySeven | i see, so you are doing an incoming config now? |
| 06:45.47 | ghinckley6 | yes |
| 06:46.03 | LuckySeven | There is this variable. ${EXTEN} or something. |
| 06:46.05 | LuckySeven | let me check |
| 06:46.06 | ghinckley6 | i have 2 tdm400p with 8 fxo ports |
| 06:46.26 | ghinckley6 | configured as a single group |
| 06:46.50 | LuckySeven | I see. I don't think there is a problem with your hardware. |
| 06:47.27 | LuckySeven | are all your internal clients using IAX or SIP? |
| 06:47.57 | ghinckley6 | ip |
| 06:48.00 | ghinckley6 | sip |
| 06:48.18 | ghinckley6 | no hardware problem i can call out on all ports and call all phones |
| 06:48.33 | LuckySeven | can't you just execute a Dial(SIP/${EXTEN}) |
| 06:48.43 | ghinckley6 | just need a script or the syntax to make the incoming calls go were they need to |
| 06:48.49 | ghinckley6 | yes internal |
| 06:49.22 | LuckySeven | In that case, you must have a predefined extension. |
| 06:49.37 | LuckySeven | One that when the user calls, all phones in that group will ring. |
| 06:49.40 | ghinckley6 | yes they are set up |
| 06:50.11 | ghinckley6 | when i get a call on a Zap chanel i need to route it to a Sip Channel |
| 06:50.11 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 06:50.39 | LuckySeven | i see. that should be relatively easy to do as long as you have setup the zap channel and SIP correctly. |
| 06:52.07 | ghinckley6 | yes both are |
| 06:52.17 | ghinckley6 | how does one do it |
| 06:53.07 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:53.19 | *** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com) |
| 06:53.29 | yxa | is there a tutorial on how to best convert a wav/gsm so that it sounds best over the phone? |
| 06:54.04 | FuriousGeorge | yxa: check the wiki, you gott use sox to sample it at 8 bits X 8khz |
| 06:54.13 | FuriousGeorge | (re)sample |
| 06:54.26 | LuckySeven | glen: I don't really get what you want to do. |
| 06:54.38 | LuckySeven | exten => Dial(SIP/exten) |
| 06:54.50 | LuckySeven | i think that should suffice right? |
| 06:54.53 | yxa | FuriousGeorge but wav is less cpu intensive right? |
| 06:55.09 | FuriousGeorge | anyone know when the new expected release date for 1.4 is? originally it was june to july |
| 06:55.27 | russellb | FuriousGeorge: no idea. |
| 06:55.41 | russellb | "when it's ready" is the official answer. |
| 06:56.06 | FuriousGeorge | yxa: i think i have heard that wav is less cpu intensive than compressed formats, yeah |
| 06:56.15 | FuriousGeorge | russellb: you work for digium right? |
| 06:56.21 | russellb | yes, i do |
| 06:56.26 | FuriousGeorge | uh oh :) |
| 06:56.56 | FuriousGeorge | j/k, so its been pushed back enough that we cant guestimate at all? |
| 06:56.57 | russellb | we figure it's best to wait until we have all the things completed that people want completed |
| 06:57.00 | ghinckley6 | russellb: can you point me to a good incoming call manual or script |
| 06:57.09 | mitcheloc | russellb: do you work on sight? |
| 06:57.10 | mitcheloc | * site |
| 06:57.23 | russellb | mitcheloc: no, but I will starting in January |
| 06:57.39 | mitcheloc | okay, cool, i was wondering cause i visited there a few weeks ago ;) |
| 06:57.46 | russellb | ghinckley6: try the O'Reilly book if you're looking for generic info ... |
| 06:57.51 | russellb | ~thebook |
| 06:57.53 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:57.53 | russellb | ~book |
| 06:57.55 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:57.55 | FuriousGeorge | russellb: cool, the more stuff the merrier. does that mean the new chan_sip is gonna make it in |
| 06:57.58 | ghinckley6 | no have that |
| 06:58.13 | russellb | FuriousGeorge: i'm not sure what new chan_sip you're talking about |
| 06:58.27 | russellb | i mean, it has gone through a lot of changes, sure |
| 06:58.43 | ghinckley6 | i am tring to route in coming calls from the Zap channels to the Sip phones |
| 06:58.54 | ghinckley6 | i dont quite understand the book |
| 06:59.11 | FuriousGeorge | russellb: well, i heard in here once that getting shared lines, call barge, etc working it would require a heavy rewrite of chan_sip that may not make it into 1.4 |
| 06:59.11 | russellb | ghinckley6: poked around the wiki? |
| 06:59.33 | russellb | FuriousGeorge: well, there is some shared line appearance code in there now ... |
| 06:59.41 | ghinckley6 | www.voip-info.org |
| 06:59.43 | ghinckley6 | that |
| 06:59.45 | FuriousGeorge | the hints you mean? |
| 06:59.49 | russellb | and as of a few days ago, "call whispering" support |
| 06:59.54 | russellb | which is what you mean by barging, i think |
| 06:59.57 | FuriousGeorge | whispering? |
| 07:00.06 | russellb | no, not hints, shared line appearances |
| 07:00.32 | russellb | whispering ... it's basically ChanSpy with the added ability to speak to the person |
| 07:00.35 | ghinckley6 | this is the last thing to get working for tonight |
| 07:01.08 | ghinckley6 | i am tired and nothing is making sence any more |
| 07:01.16 | FuriousGeorge | russellb: like for instance, mary's phone rings and steve can grab the call ala key system |
| 07:01.20 | FuriousGeorge | whats that called? |
| 07:01.33 | FuriousGeorge | i get all these abbreviations mixed up |
| 07:01.46 | russellb | FuriousGeorge: yeah, that's SLA, and mark wrote some code to do that |
| 07:01.52 | russellb | not sure how well it works, haven't tried it |
| 07:02.01 | Un1x | ~ports |
| 07:02.04 | jbot | from memory, ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 07:02.09 | Un1x | :| |
| 07:02.14 | Un1x | i ment wich ports asterisk uses |
| 07:02.16 | Un1x | anyway |
| 07:02.16 | Un1x | oh well |
| 07:03.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:03.53 | FuriousGeorge | russellb: when you say that SLA code is in there now, is this a patch that was written a few weeks ago to chan sip... i think i saw it on bugtracker |
| 07:03.54 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 07:04.22 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 07:04.33 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:05.22 | Qwell | did SLA ever actually hit the bug tracker? |
| 07:05.51 | FuriousGeorge | Qwell: i remember seeing something about "call stealing" |
| 07:06.02 | FuriousGeorge | whta was, i believe, a patch to chan sip |
| 07:06.49 | FuriousGeorge | i currently use a metermaid patch for 1.2.10 which is wunderbar |
| 07:07.39 | russellb | did not go through the bug tracker |
| 07:07.50 | Qwell | didn't think so.. |
| 07:08.02 | russellb | call stealing ... there is app_directed_pickup ... |
| 07:09.26 | FuriousGeorge | http://svn.digium.com/view/asterisk/team/markster/asterisk-sla/ |
| 07:09.50 | ghinckley6 | how does one do a dial zero for the operator |
| 07:10.07 | ghinckley6 | got the previous problemfixed |
| 07:10.17 | FuriousGeorge | exten => 0,1,dial(${OPERATOR}) |
| 07:10.29 | FuriousGeorge | exten => 0,2,panic() |
| 07:13.19 | FuriousGeorge | russellb: that code we were talking about is based on trunk, right? |
| 07:13.55 | *** join/#asterisk ghinckley6 (n=Spam@209.26.206.110) |
| 07:13.59 | russellb | yes |
| 07:14.42 | FuriousGeorge | russellb: thanks for the time |
| 07:15.16 | russellb | no problem |
| 07:15.58 | *** join/#asterisk knoppix_debian (n=jaitonys@201.19.110.100) |
| 07:16.09 | knoppix_debian | webmin for asterisk |
| 07:16.52 | russellb | knoppix_debian: trixbox |
| 07:17.03 | russellb | or wait, freepbx |
| 07:17.07 | russellb | i can't keep the names straight. |
| 07:17.30 | knoppix_debian | for debian |
| 07:17.57 | russellb | it can be installed on debian ... but there may not be a package for it |
| 07:18.04 | russellb | but that's you're only chance |
| 07:23.17 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) [NETSPLIT VICTIM] |
| 07:23.17 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) [NETSPLIT VICTIM] |
| 07:23.17 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) [NETSPLIT VICTIM] |
| 07:23.25 | *** join/#asterisk heath__ (n=heath__@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 07:24.23 | hads|home | knoppix_debian: You may be able to use packages from Xorcom Rapid. |
| 07:24.39 | Strom_C | why guis? |
| 07:24.46 | Strom_C | did typing go out of fashion or something? |
| 07:24.56 | knoppix_debian | humm |
| 07:24.56 | hads|home | Aparently |
| 07:24.56 | SwK_ | anyone know any good "blade servers" that offer a DC power option? |
| 07:25.53 | E-bola | Morning |
| 07:26.14 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
| 07:28.50 | knoppix_debian | something mais I would like uses the webmin about estou facilitate |
| 07:29.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:29.25 | Strom_C | wow...franglais |
| 07:31.35 | implicit | SwK_, howve u been |
| 07:35.26 | SwK_ | working too damned much... U? |
| 07:37.34 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 07:37.58 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79EAA.pool.t-online.hu) |
| 07:38.21 | *** part/#asterisk heath__ (n=heath__@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 07:38.41 | inspired | I'm thinking of demonstrating ad o-it-yourself virtual web pbx. does anyone know any good ones? |
| 07:38.56 | inspired | sign up, set up your dialplan and devices and off you go. |
| 07:39.40 | E-bola | #freepbx? |
| 07:39.49 | E-bola | or well trixbox i guess |
| 07:39.57 | inspired | nah, not like that |
| 07:40.05 | inspired | one that supports several users |
| 07:40.48 | inspired | not for setting up a pbx, but instead a pbx that's hosted elsewhere |
| 07:41.16 | E-bola | u mean to support multiple companies? |
| 07:41.16 | inspired | I remember I saw a really nice solution a time back, but can't seem to find it now |
| 07:41.22 | inspired | yes |
| 07:41.29 | E-bola | hmm dont know any sory |
| 07:42.22 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 07:44.27 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 07:50.26 | shido6 | how unique. |
| 07:51.22 | potsboy | hey all, its it possible to provide a campon tone on a zap channel? |
| 07:54.34 | *** join/#asterisk Blafasel (n=bpodszun@relay3.vistream.de) |
| 08:00.03 | *** join/#asterisk Xen^ (n=linux@202.5.145.56) |
| 08:00.40 | *** join/#asterisk vlt (n=dm@p54B344C6.dip0.t-ipconnect.de) |
| 08:01.55 | Blafasel | Hi there. I'm trying to route SIP calls over zaptel/ss7/isup to landline calls. While SIP <-> SIP works fine, SIP -> Landline calls succeed, but are muted (no audio). Any ideas how I can analyse/solve the problem? |
| 08:05.40 | potsboy | blafasel do you get audio if you call from the console? |
| 08:06.53 | Blafasel | I fear I don't know how to do that.. To be honest: I'm bloody new to this stuff. |
| 08:07.00 | Blafasel | The "On-Hold" music works |
| 08:07.01 | *** join/#asterisk _fa_ (i=faceoff@een.os3.kn.pl) |
| 08:07.24 | Blafasel | So if I put the call on hold in my sip client, the other side can listen to this fancy default music |
| 08:08.38 | potsboy | sounds like it more of a client issue, what client are you using? |
| 08:09.11 | Blafasel | I tried several, including X-Lite 2.0 on a mac, 3.0 on a windows machine, sjphone on a mac etc. |
| 08:10.59 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:12.03 | *** join/#asterisk waglik (n=mwegrzyn@polpak.litex.pl) |
| 08:13.49 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:14.46 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 08:16.08 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 08:19.45 | Blafasel | Is there any way to debug this call issue? |
| 08:20.44 | inspired | Blafasel, tried setting canreinvite=no on sip friends? do you still get audio? |
| 08:22.05 | Un1x | . |
| 08:22.26 | Blafasel | inspired: It's already using canreinvite=no. Only the hold music, nothing more |
| 08:22.54 | inspired | hmm |
| 08:26.46 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 08:27.13 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 08:30.55 | *** join/#asterisk svenna_ (n=svenna@p548D14B8.dip0.t-ipconnect.de) |
| 08:31.04 | svenna_ | hi all |
| 08:32.32 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:32.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:34.08 | svenna_ | after a user talks to my mailbox, i wanto to check for new mails with HasNewmail(). but when he talks to the box an hangs up, the dialplan ends befor it get to check for new mail... is there a way to fore * to go through the whole dialplan? |
| 08:34.25 | svenna_ | (i guess its hole, not whole...) |
| 08:34.48 | JT | it is whole |
| 08:35.05 | svenna_ | ok thx 4 that :-) |
| 08:37.20 | potsboy | svenna try h,1,blabla |
| 08:42.24 | svenna_ | in cmd dial()? |
| 08:45.27 | potsboy | h will execute on hangup |
| 08:49.03 | svenna_ | ahhh |
| 08:49.06 | svenna_ | ok i see |
| 08:53.13 | *** join/#asterisk nailbags|laptop (n=neil@c220-237-12-224.randw1.nsw.optusnet.com.au) |
| 08:54.13 | *** join/#asterisk foRza (n=tMs@firewall.hikt.no) |
| 08:55.05 | svenna_ | ok, that seems to work |
| 08:55.07 | svenna_ | thx |
| 08:55.30 | svenna_ | now i see, that i have to use vmcount instead of hasnewmail :-( |
| 08:55.48 | svenna_ | so i will see how that works... |
| 08:57.06 | *** join/#asterisk kiddy (n=achu@124.125.39.182) |
| 08:57.29 | kiddy | Can anybody clear the following doubt :? |
| 08:58.10 | kiddy | I have installed festival in my machine and configured it as in http://www.voip-info.org/wiki-Asterisk+Festival+installation |
| 08:58.40 | kiddy | but when I test as the above documentation says I can't hear anything from asterisk server |
| 08:58.58 | kiddy | please help me to do the festival in asterisk |
| 08:59.37 | *** join/#asterisk s0lid (n=jlq@124.6.176.100) |
| 09:01.44 | waglik | hello! |
| 09:02.21 | waglik | I have problem with hangup detection with Asterisk 1.2.9 and 1.2.10 |
| 09:03.05 | waglik | I have pri connection to my telco |
| 09:03.26 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:03.39 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:08.35 | *** join/#asterisk nailbags|laptop (n=neil@c220-237-12-224.randw1.nsw.optusnet.com.au) |
| 09:11.59 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:11.59 | *** join/#asterisk waba (n=waba@eott.enix.org) |
| 09:15.53 | waba | I'm facing a WTF. From README.variables, there is a ! negation operator. I have: Set(enable=$[ !${caller_shop_group} ]) with the variable being worth 0. At runtime I get enable worth "!0"... What I am doing wrong? |
| 09:16.02 | waba | (besides not RTFSing) |
| 09:16.42 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:20.32 | waba | (tried with () and/or "" and/or spaces, of course, though they shouldn't be needed) |
| 09:25.42 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:29.43 | L|NUX | Qwell : did you saw mog ? |
| 09:30.32 | L|NUX | ~seen mog |
| 09:30.40 | jbot | mog <i=ejabberd@68.62.237.103> was last seen on IRC in channel #asterisk, 1d 9h 40m 17s ago, saying: 'whats wrong pbuckley'. |
| 09:30.40 | L|NUX | !seen mog |
| 09:34.11 | Blafasel | This question might seem absolutely dumb, but since I'm bloody new and unable to get it answered with google: Do I need OSS or Alsa support? What for? Which one is better? |
| 09:36.38 | waba | my problem makes much more sense now: according to ast_expr2.fl, the ! op stayed warm & cozy in the imagination of the README.variables writer. *mutter* |
| 09:37.16 | waba | Blafasel: for asterisk? You need none of them. You *can* use ALSA for some things, but it's not at all used for making a VoIP/copper pair PBX |
| 09:37.57 | waba | and for a general workstation, ALSA is better than OSS (which is a legacy thing) |
| 09:38.03 | Blafasel | waba: Thanks for pointing that out. I'm still struggling to solve my no-sound problem and wondered if I missed a mandatory requirement here. |
| 09:38.17 | Blafasel | So I guess I need/want neither. |
| 09:38.30 | waba | agreed |
| 09:39.08 | waba | if I got it well, ALSA is only used if you want asterisk to do audio output to people that would be in the same physical room than the server |
| 09:39.14 | ghenry | what have you guys found to be the best web gui? |
| 09:39.16 | ghenry | for * |
| 09:40.12 | waba | haven't tried any of them yet, so can't tell |
| 09:46.24 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:58.55 | *** join/#asterisk s0lid (n=jlq@124.6.176.100) |
| 10:10.54 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 10:13.52 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 10:15.21 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:18.38 | *** join/#asterisk parag_ast (n=root@dxb-b16259.alshamil.net.ae) |
| 10:19.09 | parag_ast | <PROTECTED> |
| 10:19.10 | parag_ast | <PROTECTED> |
| 10:19.10 | parag_ast | <PROTECTED> |
| 10:19.10 | parag_ast | <PROTECTED> |
| 10:19.10 | parag_ast | <PROTECTED> |
| 10:19.10 | parag_ast | <PROTECTED> |
| 10:19.12 | parag_ast | <PROTECTED> |
| 10:19.33 | parag_ast | can anybody let me know exect reason why Zap/1-1 and Zap/3-1 got hanged up |
| 10:21.42 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 10:21.49 | fourcheeze | Heeeeelllllpp!!!! |
| 10:22.02 | fourcheeze | I had to fail over to another box for a while |
| 10:22.18 | fourcheeze | now I've switched back and I'm getting stale nonces everywhere |
| 10:22.35 | fourcheeze | any ideas? |
| 10:23.06 | fourcheeze | no-one seems to be authenticating |
| 10:23.47 | kiddy | fourcheeze : I have installed and configured festival as in voip wiki |
| 10:24.21 | kiddy | <PROTECTED> |
| 10:24.23 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:24.45 | puzzled | morning |
| 10:24.49 | kiddy | I can't hear anything and I find the reason in the log |
| 10:25.29 | kiddy | client(1) Mon Jul 31 05:41:11 2006 : accepted from serverclient(1) Mon Jul 31 05:41:11 2006 : disconnected |
| 10:25.46 | kiddy | can you help me to solve the problem ? |
| 10:26.02 | fourcheeze | kiddy: I'm just having a crisis right now and I know nothing about festival |
| 10:26.42 | kiddy | ok no problem |
| 10:26.54 | fourcheeze | anyone got any idea about my problem? |
| 10:27.08 | kiddy | I am expecting help from others |
| 10:29.43 | fourcheeze | what is a stale nonce and why am I getting them instead of registrations? |
| 10:33.40 | fourcheeze | what is a stale nonce? |
| 10:34.25 | *** join/#asterisk expat_iain (n=expat_ia@194.204.99.131) |
| 10:34.26 | fourcheeze | anyone home? |
| 10:34.30 | fourcheeze | ping ? |
| 10:34.36 | expat_iain | No, am at the office. |
| 10:34.39 | fourcheeze | hehe |
| 10:34.43 | expat_iain | Gnaaar. |
| 10:35.23 | fourcheeze | any idea what a stale nonce is? |
| 10:36.48 | _4d4m_ | the nonce is a field used in the headers for authentication |
| 10:36.53 | _4d4m_ | its value is created by asterisk |
| 10:37.08 | fourcheeze | ok |
| 10:37.10 | _4d4m_ | a stale nonce is one that is no longer valid (due to timeout i guess) |
| 10:37.18 | fourcheeze | so how do I refresh them all |
| 10:37.27 | _4d4m_ | i've only hit the problem once, and it was for a particular type of UA |
| 10:37.32 | fourcheeze | I'm guessing they are using the one that they used on the other system |
| 10:37.38 | mut | you usually get it when a device tries to login too quickly |
| 10:37.48 | mut | relogin* |
| 10:37.50 | _4d4m_ | and to solve it, i had to set pedantic=no in sip.conf and then reload |
| 10:37.56 | fourcheeze | ok |
| 10:38.35 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 10:39.07 | fourcheeze | what does that do? |
| 10:39.08 | kiddy | any one have any idea about this log : client(1) Mon Jul 31 05:41:11 2006 : accepted from server |
| 10:39.19 | kiddy | client(1) Mon Jul 31 05:41:11 2006 : disconnected |
| 10:39.38 | kiddy | I am getting it when I try to connect to festival |
| 10:40.12 | _4d4m_ | fourcheeze: forget the pedantic=no bit and force fresh registration from your UA's |
| 10:40.24 | fourcheeze | _4d4m_: that doesn't seem to work |
| 10:40.28 | _4d4m_ | hmm |
| 10:40.41 | fourcheeze | I keep getting "unauthorised" |
| 10:40.53 | _4d4m_ | what version of *? |
| 10:40.57 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:40.58 | fourcheeze | 1.2.4 |
| 10:40.59 | backblue | hi* |
| 10:42.13 | kiddy | fourcheeze : May I know what client you are using to connect to asterisk ? |
| 10:42.58 | fourcheeze | kiddy: snom360, sipuras, polycoms |
| 10:43.09 | fourcheeze | currently none of them working :/ |
| 10:43.21 | fourcheeze | _4d4m_: any more ideas? |
| 10:43.26 | kiddy | is there any "Send DTMF" field in configuration window ? |
| 10:43.35 | fourcheeze | no |
| 10:44.25 | LuckySeven | Is anyone here using a Genesis IP phone for connecting to Asterisk? |
| 10:45.44 | fourcheeze | I seem to have this problem: |
| 10:45.45 | fourcheeze | http://archives.free.net.ph/message/20051215.183515.e25c3841.en.html |
| 10:46.58 | Dr-Linux|work | why asterisk doesn't recognize CallerID in my country? Pak |
| 10:48.15 | _4d4m_ | fourcheeze: i've just looked at that.. but thats to do with a bug in an earlier version of asterisk.. |
| 10:49.21 | fourcheeze | basically things are not managing to authenticate |
| 10:49.32 | fourcheeze | asterisk always sends a 401 |
| 10:49.35 | fourcheeze | this sucks |
| 10:50.04 | fourcheeze | and it's now about an hour that people have been without phones :/ |
| 10:50.21 | fourcheeze | so my mobile is going mad! |
| 10:50.34 | fourcheeze | how do I give asterisk a slap around the face? |
| 10:50.58 | fourcheeze | I've currently got 2 online out of 54 clients |
| 10:51.41 | kiddy | fourcheeze : can you reboot asterisk once again and check the logs ? |
| 10:51.49 | _4d4m_ | fourcheeze: is http://lists.digium.com/pipermail/asterisk-dev/2006-January/018384.html of any relevance? |
| 10:52.40 | fourcheeze | interesting |
| 10:52.41 | fourcheeze | maybe |
| 10:53.02 | fourcheeze | doesn't look like there's a solution |
| 10:53.11 | fourcheeze | so shall I just pack up and go home? |
| 10:53.12 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 10:53.27 | fourcheeze | should I remove the astdb or something? |
| 10:54.59 | fourcheeze | _4d4m_: /me is about to give up and quit |
| 10:55.00 | _4d4m_ | does rebooting the UA have any effect? |
| 10:55.03 | fourcheeze | no |
| 10:55.19 | fourcheeze | where are the nonces stored? |
| 10:55.37 | *** part/#asterisk knoppix_debian (n=jaitonys@201.19.110.100) |
| 10:55.38 | fourcheeze | I've got 3 clients that can login |
| 10:55.57 | fourcheeze | those all seem to work |
| 10:55.59 | _4d4m_ | and what is different about them? |
| 10:56.06 | fourcheeze | I've no idea |
| 10:56.12 | fourcheeze | all my clients shoudl be the same |
| 10:56.23 | fourcheeze | at least the user credentials are the same |
| 10:56.23 | _4d4m_ | very wierd |
| 10:57.09 | _4d4m_ | you could try clearing astdb to clear registration info i guess |
| 10:58.03 | _4d4m_ | but i'm pretty much out of ideas now myself.. :-/ |
| 10:59.14 | fourcheeze | hmm |
| 10:59.48 | kiddy | please try 'pedantic=yes' in sip.conf and restart the asterisk box |
| 11:00.20 | _4d4m_ | when the 401 is sent out, the UA should start over and discard all old dialog information |
| 11:00.31 | _4d4m_ | and likewise the server should start afresh too |
| 11:00.36 | fourcheeze | yeah |
| 11:00.38 | _4d4m_ | or thats my understanding of it |
| 11:01.21 | kiddy | fourcheeze : please try 'pedantic=yes' in sip.conf and restart the asterisk box |
| 11:01.23 | fourcheeze | well that's what I thought |
| 11:01.27 | fourcheeze | kiddy: ok trying |
| 11:03.52 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 11:04.12 | *** part/#asterisk Mattwj2005 (n=Matt@user-12l3n0n.cable.mindspring.com) |
| 11:04.24 | fourcheeze | kiddy: no difference there |
| 11:04.41 | fourcheeze | _4d4m_: how do I refresh the astdb? |
| 11:05.11 | kiddy | hmm |
| 11:08.58 | kiddy | fourcheeze : Try the following |
| 11:09.09 | kiddy | stop asterisk server for a while |
| 11:09.26 | kiddy | unplug a phone and plug it |
| 11:09.50 | kiddy | then edit the configuration such as passwd,username etc again and submit it |
| 11:10.09 | kiddy | start asterisk server |
| 11:10.32 | kiddy | I think it will clear the more than one attempt to register to server |
| 11:11.24 | fourcheeze | kiddy: what do you think the problem is? |
| 11:15.44 | _4d4m_ | fourcheeze: you have actually restarted * right? not just reloaded? registration seedings live on through reloads in astdb |
| 11:15.52 | fourcheeze | yes restarted completely |
| 11:15.56 | _4d4m_ | thoughts so |
| 11:16.08 | fourcheeze | but maybe I need to delete the astdb or soemthing |
| 11:16.09 | fourcheeze | where is it? |
| 11:16.48 | kiddy | fourcheeze : look at this http://bugs.digium.com/view.php?id=4343 |
| 11:17.20 | fourcheeze | hmm |
| 11:17.23 | fourcheeze | yeah |
| 11:20.08 | kiddy | fourcheeze : Is it is helpful ? |
| 11:23.53 | *** join/#asterisk andew (n=andew@84-45-170-202.no-dns-yet.enta.net) |
| 11:24.37 | _4d4m_ | fourcheeze: you could always check 1.2.10 out and give it a whirl with your existing config and astdb? |
| 11:24.46 | fourcheeze | hmm |
| 11:26.12 | _4d4m_ | have you already used tethereal/ngrep/tcpdump to check the contents of the signalling traffic for registrations? |
| 11:26.18 | _4d4m_ | just to make sure nothing else is amiss? |
| 11:27.34 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 11:27.45 | fourcheeze | not tried that yet |
| 11:29.48 | expat_iain | I've a PRI card that zttool shows me as being internall clocked. My zaptel.conf file has the following setting: 'span=1,1,0,ccs,hdb3'... |
| 11:30.04 | expat_iain | ...and I'm providing a clock on the PRI inbound. |
| 11:30.11 | expat_iain | Have I missed something here?? |
| 11:32.43 | expat_iain | As far as I can tell, I'm supposed to be taking the clock from the PRI line, but it ain't playing fair. |
| 11:33.34 | champster | If I remember right, you have to alter and recompile zttool to get correct info from the card. |
| 11:34.08 | fourcheeze | _4d4m_: I'm trying a new server now |
| 11:34.11 | champster | But that alteration is not to be used while the phone system is in use. |
| 11:34.20 | fourcheeze | _4d4m_: got arp conflicts I think |
| 11:34.21 | expat_iain | Oh. This is installed on a Trixbox device. |
| 11:34.27 | fourcheeze | stuff not getting through from outside |
| 11:34.32 | fourcheeze | any ideas anyone? |
| 11:34.40 | champster | The unalterred zttool can be used awhile asterisk is running. |
| 11:34.40 | expat_iain | So you're suggesting I grab the sources and do the recompile? |
| 11:35.37 | champster | I do not remember the required change. it is either commented in to code, or there was a posting about what to change. |
| 11:36.09 | expat_iain | Cheers. Will check that out. |
| 11:36.36 | _4d4m_ | fourcheeze: flush the arp cache on your router? not sure.. |
| 11:37.16 | fourcheeze | yeah, not my router unfortunately |
| 11:37.17 | champster | Are my posts orange or is it just on my display.? (i was playing with the colors on my mIRC.) |
| 11:37.20 | Dr-Linux|work | why asterisk doesn't recognize CallerID in my country? Pak |
| 11:37.45 | Dr-Linux|work | anybody have any clue? |
| 11:38.30 | champster | I recall there being posts about callerID in India and Pakistan. you may want to check the list archives. |
| 11:39.25 | Dr-Linux|work | champster, i did some google, but i can't find any help |
| 11:39.54 | Dr-Linux|work | champster, actually i wanna allow calls throught my asterisk against caller id authentication. |
| 11:40.13 | champster | Try India instead of Pakistan. (if they use the same tech.) |
| 11:41.15 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 11:41.16 | *** join/#asterisk }btorch{ (n=btorch@c-66-176-87-59.hsd1.fl.comcast.net) |
| 11:41.31 | }btorch{ | what app do you guys use for music on hold ? |
| 11:41.38 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:41.42 | }btorch{ | mpg123 seems to crash asterisk |
| 11:42.22 | }btorch{ | I keep getting a "flexible rate not heavly tested" error if a user stays on hold for to long |
| 11:43.10 | champster | expat_iain -- http://bugs.digium.com/view.php?id=4186 -- note, the patch may not be usable on new code. |
| 11:43.56 | X-Rob | }btorch{, don't use VBR mp3's then. |
| 11:44.16 | }btorch{ | what's the other option ? |
| 11:44.26 | X-Rob | don't use MP3's, convert them to slin |
| 11:44.27 | }btorch{ | what's VBR by the way |
| 11:44.33 | X-Rob | variable bit rate. |
| 11:44.36 | nailbags | }btorch{: native MoH |
| 11:47.27 | }btorch{ | any other option? |
| 11:47.44 | X-Rob | don't use asterisk. |
| 11:49.03 | puzzled | hehe |
| 11:49.20 | champster | expat_iain -- actually, I haven't looked in a while, but my zttool shows the proper clock source. |
| 11:49.43 | champster | expat_iain -- what version of zaptel are you using? I am using 1.2.7 |
| 11:51.31 | Blafasel | My zttool shows the wrong state as well.. |
| 11:51.52 | champster | 1.2.7? |
| 11:51.58 | Blafasel | Yes |
| 11:52.10 | X-Rob | zttool does _not_ show you the proper clock source |
| 11:52.20 | X-Rob | it does in zaptel-trunk, apparently |
| 11:52.30 | X-Rob | but it's not something you need to 'diagnose'. |
| 11:52.37 | X-Rob | it's set to what you told it to. |
| 11:53.01 | champster | Mine shows - Sync Source: T4XXP (PCI) Card 0 Span 1 |
| 11:53.44 | champster | Maybe my zttool reports better since I switched from TE110P to TE410P? |
| 11:53.47 | Blafasel | Well, I'm just desperate so I try to look at every possible error ;) |
| 11:54.39 | champster | I also wish zttool would show you loop state and not just a loop button. |
| 11:54.49 | fourcheeze | anyone know if the nonce is held in realtime? |
| 11:56.27 | coppice | I wish they'd sort out the TE11x and TE41x drivers so they report all the errors from the card |
| 11:57.30 | waglik | hello again :) |
| 11:57.33 | champster | I would like to be able to monitor span errors externally so that I can react faster to a downed line. |
| 11:58.02 | waglik | I have a problem with asterisk not reacting to a hangup on a pri line |
| 11:58.03 | coppice | you can't monitor span errors anywhere right now |
| 11:58.17 | waglik | if the hangup happens before anyone pickups the line |
| 11:58.27 | waglik | did anyone encounter something similar? |
| 11:58.46 | waglik | I'm using Digium TE210P |
| 11:59.25 | waglik | to connect to the PRI, the telco is TP SA (Poland, euroISDN) |
| 12:01.31 | waglik | The asterisk seems to notice the hangup (this is what I get in the console): |
| 12:01.33 | waglik | Channel 0/12, span 1 got hangup ACK |
| 12:01.49 | waglik | but it does not stop to ring the target extension |
| 12:02.03 | }btorch{ | ok Moh does't work on mine |
| 12:02.26 | Blafasel | Is there any way to analyse what exactly runs on over my E1 link? Read: What data/what content? I still don't have a starting point to fix my issue here. |
| 12:02.51 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 12:08.07 | *** join/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil) |
| 12:11.49 | TypMic | Incoming audio to a digium TDM400 card from a Phone-Line is very low resulting in the receiving PC softphone barely hearing the user on the telephone. Anyone have any ideas how to improve incoming audio to Asterisk from a phone-line connection to a PSTN network and telephones on that network. |
| 12:12.34 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:13.18 | Blafasel | "mtp_put: Full MTP receivebuf, event lost." doesn't sound nice.. Neither does "Audio buffer underrun" or "Unable to write to alert pipe".. Any ideas what might be the cause? |
| 12:13.40 | puzzled | anyone know how to fix the vpm450m_fw.h not found when compiling zaptel-1.2.7? I can't even find it in the source |
| 12:15.54 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:18.19 | coppice | Blafasel: which SS7 are you running? |
| 12:18.39 | Blafasel | chan_ss7 |
| 12:19.13 | coppice | that seems to have problems like this. i think its a bit early in its development, still |
| 12:19.49 | Blafasel | Well, can you suggest any alternative that I can evaluate without buying it? |
| 12:19.52 | puzzled | Blafasel: no idea but I know there is a chan_ss7 list. perhaps ask there |
| 12:20.11 | coppice | Blafasel: not without buying |
| 12:21.34 | coppice | i heard there was a chan_ss7 list, but I never found it |
| 12:21.53 | coppice | still, half the planet can't even find the web site :-) |
| 12:22.27 | puzzled | coppice: http://www.sifira.dk/chan-ss7/ |
| 12:22.45 | puzzled | and the list is at http://bat.berlios.de/mailman/listinfo/chan-sccp-users |
| 12:22.50 | coppice | do you realise more than half the internet can't see that site? |
| 12:22.59 | *** join/#asterisk jaike (n=a@203.115.188.120) |
| 12:23.05 | puzzled | guess I'm at the right half :) |
| 12:23.17 | Blafasel | coppice: That's bad.. I'd buy one if it works, not before.. ;) |
| 12:23.49 | coppice | why is chan_sccp the place for chan_ss7? no wonder I couldn't find it |
| 12:24.52 | puzzled | coppice: ugh sorry. http://lists.digium.com/mailman/listinfo/asterisk-ss7 |
| 12:25.07 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 12:25.11 | coppice | nothing useful is ever posted to that list |
| 12:25.33 | puzzled | nod |
| 12:25.45 | jaike | are call files limited to only one channel per file? |
| 12:26.09 | jaike | or one call |
| 12:26.29 | coppice | Blafasel: then I think you won't get a reliable SS7 without developing it :-) If you are prepared to pay for SS7, there is a solid reliable one for *. |
| 12:26.38 | *** join/#asterisk bintut (n=bintut@cable-202-8-251-159.d-one.net) |
| 12:27.15 | puzzled | coppice: which one is that? cosini? |
| 12:27.21 | coppice | yep |
| 12:27.33 | bintut | what do you recommend: a digium tdma w/ 4 fxo or a sangoma tdma w/ 4 fxo? |
| 12:28.07 | puzzled | I recommend going digital as much as possible :) |
| 12:28.41 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 12:28.50 | Blafasel | coppice: Well, I'm not yet convinced that my problem is only chan_ss7 related and I usually try before I buy.. This is a showcase only for now. |
| 12:28.53 | coppice | oh, TDMA for FXO ports. now there's a novelty :-) |
| 12:29.30 | bintut | puzzled: we only have 4 lines to pstn here and we're planning that all our local lines would be through voip and using sipura analog adapters |
| 12:31.06 | bintut | puzzled: this is only a replacement for our existing pbx provided by our pstn and we're planning here to replace it with an asterisk box and having a tdma pci card with 4 fxo. what do you suggest? |
| 12:31.46 | puzzled | bintut: nothing. I don't do analog |
| 12:31.48 | coppice | Blafasel: maybe not. the problems with chan_ss7 usually show up as an overrunning signaling channel. however, it can easily overrun the audio channels too, when the signalling channels gets badly screwed up |
| 12:31.52 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:31.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:33.00 | Blafasel | coppice: Would that explain that I can establish calls without any audio/sound? |
| 12:33.18 | bintut | anyone here can share their advise on setting up an asterisk pbx? |
| 12:34.03 | coppice | Blafasel: maybe. the key problem i know about is the signalling channel overrunning at light loads. if its really busy it should work better. |
| 12:35.53 | Blafasel | Hmm.. Currently I'm at no load at all.. But I cannot put load on a system where my calls are "useless". ;) |
| 12:36.12 | *** join/#asterisk momelod (n=momelod@HSE-Montreal-ppp134142.qc.sympatico.ca) |
| 12:36.12 | coppice | ain't life a stinker :-) |
| 12:36.24 | momelod | g'morning peoples |
| 12:38.34 | momelod | q: if i want to tweak my zaptel kernel module (for example to add/remove aggressive suppression) is there a better way to do it then editing ztconfig.h and recompiling + installing it every time? |
| 12:38.47 | Blafasel | Anyway, thanks for the help.. I'll fiddle around with this stuff for a couple more hours.. |
| 12:39.57 | *** part/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil) |
| 12:41.08 | [TK]D-Fender | Was wondering if someone here can clarify a hing or two about Call Parking in * as I've never used it before. I have included the "parkedcalls" context at the lowest level of phone's context in the dialplan heirarchy and verified that the feature is un-commented out in features.con. From there though things get "grey" on how I can use it. |
| 12:41.39 | [TK]D-Fender | Do I need to use "tT" to allow them to use * native DTMF transfers to send the caller in, or can I use SIP transfers? |
| 12:42.23 | anthm | if you want to use sip transfers get app_valetparking |
| 12:43.31 | anthm | the onboard one only works on the embedded # transfer |
| 12:45.39 | [TK]D-Fender | anthm : So all I have to do is DL the .c file, manualty add it to the makefile, and away I go? |
| 12:46.40 | anthm | yah or if it's 1.2 you can do it all in 1 step |
| 12:48.07 | benjk_ | puzzled, analog suxx, doesn't it?! |
| 12:48.31 | [TK]D-Fender | anthm : Yes, 1.2.9.1. Looking at it now... |
| 12:50.29 | anthm | if your brave |
| 12:50.32 | anthm | perl /usr/src/asterisk/contrib/scripts/astxs -autoload -install http://www.pbxfreeware.org/app_valetparking.c |
| 12:51.22 | [TK]D-Fender | anthm : I'm going through the script install instructions off loligo. THose still applicable? (linked from Wiki) |
| 12:52.00 | Sonderblade | what settings control which language the IVR voice prompts should be in? |
| 12:52.19 | anthm | ya know, i never really can be sure |
| 12:52.19 | anthm | it changes a lot |
| 12:52.25 | [TK]D-Fender | anthm : I'll presume that the pbxfreeware is the best place knowing its origins |
| 12:52.45 | anthm | I had *all* my stuff working right before 1.2 came out |
| 12:53.01 | anthm | so that is your best bet but i dont knoe if anythign changed |
| 12:55.19 | [TK]D-Fender | anthm : So you've heard no reports either way as to its functioning at this point? |
| 12:55.44 | anthm | if it compiles it will function |
| 12:56.42 | anthm | there is a cool thing i added to it right before the last revision you can use app_dial to unpark |
| 12:56.44 | mut | everyone better go fill their gas tanks |
| 12:56.45 | mut | http://www.cnn.com/2006/WORLD/europe/07/31/russia.oilspill.ap/index.html |
| 12:56.55 | *** join/#asterisk ramtha (n=t@195.14.234.162) |
| 12:56.57 | ramtha | hy |
| 12:57.23 | ramtha | i have 10 snom phones behind a firewall |
| 12:57.35 | ramtha | ougoing everything works |
| 12:57.40 | ramtha | but incomming not |
| 12:58.00 | ramtha | my asterisk is locate in public internet |
| 12:58.09 | ramtha | and all phones are registered |
| 12:58.36 | ramtha | the incoming call goes to the firewall and there it doesn´t reach the voip phone |
| 12:58.44 | ramtha | portforwarding can not be the solution |
| 12:59.03 | ramtha | because i can not forward all ports to diffrent interneal ip adresses |
| 12:59.08 | *** join/#asterisk ACiDV (n=acidv@c66.110.128-170.clta.globetrotter.net) |
| 12:59.21 | anthm | hairpin calls from public to private dont work right cos asterisk refuses to have separate sip ua binded to both nic |
| 12:59.37 | ramtha | um |
| 12:59.45 | ramtha | really? |
| 13:00.00 | ramtha | how do several hostedpbx provider go arround this problem? |
| 13:00.14 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 13:00.15 | ramtha | they do not place the asterisk in your network |
| 13:00.15 | anthm | you can cheat if you clone chan_sip and hack the code so it registers chan SIPINT where it says SIP and sipint.conf where it says sip.conf |
| 13:00.19 | anthm | and compile it again |
| 13:00.29 | anthm | probably more than that |
| 13:00.41 | anthm | i had to do it on iax but i didnt need it for sip but same idea |
| 13:00.47 | [TK]D-Fender | anthm : Many thanks. Installed and am reading up on now... |
| 13:00.59 | anthm | np |
| 13:01.20 | ACiDV | Anyone use AddQueueMember with Local/ channel ? ex. AddQueueMember(sales|Local/${CALLERID(num)}@localdevices) ... I can log but never receive any call. |
| 13:01.22 | anthm | you need different settings on the internal one than you do on the external when you have that setup |
| 13:01.32 | *** join/#asterisk asteriskmonkey (n=phil@h216-235-8-130.host.egate.net) |
| 13:01.45 | ramtha | anthm: ok let me understand this right. |
| 13:02.21 | ramtha | anthm: the is no solution out of the box to get calls from several ip phones behind a router, without portforward to work? |
| 13:02.55 | bintut | is there a f/oss driver for sangoma a20002d cards + asterisk? |
| 13:02.58 | anthm | does it work when they call each other |
| 13:03.11 | ramtha | yes |
| 13:03.16 | anthm | then nope =D |
| 13:03.32 | anthm | http://bugs.digium.com/view.php?id=2611 |
| 13:03.37 | ramtha | ok, can you explain agian why? :) |
| 13:03.41 | anthm | i was shunned for reporting a similar bug |
| 13:03.50 | [TK]D-Fender | bintut : Sangoma uses their Wanpipe driver which interfaces with Zaptel. Nothing else needed. |
| 13:03.53 | anthm | for it being alreay a known issue and no major |
| 13:04.35 | bintut | [TK]D-Fender: thanks again.. maybe i should get this card.. :) |
| 13:04.58 | anthm | note i submitted it oct 2004 and it's my fault for not realizing it was already reported in 2 bugs that are both closed |
| 13:04.58 | Sonderblade | noone knows what options to set to not get english language voice prompts? |
| 13:05.34 | [TK]D-Fender | Sonderblade : * multi-lingual support is all documented on the WIKI. |
| 13:05.55 | anthm | sip has no way to bring authenticated calls in 1 nic and forward them out another on the same box |
| 13:05.56 | Sonderblade | [TK]D-Fender: badly documented |
| 13:06.04 | [TK]D-Fender | Sonderblade : And alternate official language packs are avaiable on Digium's FTP |
| 13:06.09 | anthm | unless yo ucan bind a specific UA to each ip |
| 13:06.24 | anthm | with it's own entire group of settings |
| 13:06.36 | ramtha | uhh |
| 13:06.37 | ramtha | i got it! |
| 13:06.43 | ramtha | in the config of snom |
| 13:06.51 | ramtha | i set kepp alive interval |
| 13:06.55 | ramtha | and now it works! |
| 13:07.01 | anthm | well lucky day |
| 13:07.56 | Sonderblade | i have set language= in sip.conf, iax.conf and indications.conf but i still get the default english prompts sometimes |
| 13:08.22 | tzanger | Sonderblade: do you have the sounds in the language you prefer? |
| 13:08.30 | [TK]D-Fender | Sonderblade : It will fallback to english if you do not have that soundfile in your language folder. |
| 13:08.45 | [TK]D-Fender | tzanger : Obviously not all.... |
| 13:08.45 | Sonderblade | i do have the correct sounds |
| 13:08.59 | [TK]D-Fender | Sonderblade : but your list is incomplete. thats the problem. |
| 13:09.06 | tzanger | morning [TK]D-Fender |
| 13:09.08 | Sonderblade | when calling from some extensions i get the prompts in my language but when calling from other extensions i get them in english |
| 13:09.46 | Sonderblade | its like asterisk is trying to do some stupid language auto-detection when the channel opens |
| 13:10.20 | [TK]D-Fender | tzanger : Mornin' *yawn* |
| 13:10.48 | [TK]D-Fender | Sonderblade : then you haven't set your language choice in all the appropriate places. |
| 13:11.19 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:11.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.51 | Sonderblade | [TK]D-Fender: which are those? i'm using sip so i thought setting language=se in [general] in sip.conf should be enough |
| 13:12.18 | [TK]D-Fender | Sonderblade : What about Zap & other techs? |
| 13:12.35 | [TK]D-Fender | Sonderblade : You'd have to isolate a specific example of a call to pin down what was missed |
| 13:13.52 | Sonderblade | [TK]D-Fender: how do i do that? |
| 13:14.20 | Sonderblade | [TK]D-Fender: the asterisk's console output isn't helpful |
| 13:14.42 | [TK]D-Fender | Sonderblade : Place a call from/to a phone you know doesn't react like you want it to and examine all the configs associated with that call. |
| 13:14.51 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 13:14.52 | festr_ | hello |
| 13:14.53 | [TK]D-Fender | Sonderblade : Hell yeah it is... |
| 13:15.18 | festr_ | i've problem with queue (asterisk 1.2.10). When leaving this application it want execute next priority |
| 13:15.21 | festr_ | http://pastebin.ca/108354 |
| 13:15.25 | festr_ | here is small debug |
| 13:15.30 | festr_ | is it bug or misconfiguration? |
| 13:15.31 | robl^ | Sonderblade: add more v's!! asterisk -rvvvvvvv and it will give ya more info |
| 13:15.49 | Sonderblade | robl^: im already running with verbose 10 |
| 13:17.13 | [TK]D-Fender | festr_ : that dialplan snippet doesn't match the CLI output... where's the "answer" we see being called? |
| 13:17.33 | [TK]D-Fender | festr_ : Jul 31 15:07:28 WARNING[11409]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'in_91_1' |
| 13:17.58 | [TK]D-Fender | And this tells me your showing us an "apples & oranges" scenario for sure... |
| 13:18.32 | festr_ | aaaaaaaaaaaah i'm such an idiot |
| 13:18.39 | festr_ | [TK]D-Fender: thank you |
| 13:18.44 | [TK]D-Fender | festr_ : np |
| 13:18.59 | [TK]D-Fender | <- Master of the blatantly obvious |
| 13:21.21 | Mercestes | So how do I "check debug for more info?" |
| 13:21.57 | [TK]D-Fender | Mercestes : .... on what? |
| 13:22.09 | festr_ | btw any info about release 1.4beta? |
| 13:22.21 | festr_ | does exist any roadmap? |
| 13:22.38 | Mercestes | res_config_mysql.so: update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 13:22.52 | coppice | i think the roadmap has some potholes :-) |
| 13:22.54 | Mercestes | I've like...compiled +debug, and hit debug level 99 and debug 99. I still see no debug msgs. |
| 13:24.30 | SpaceBass | Mercestes, on the cli? what kind of trunk |
| 13:24.55 | [TK]D-Fender | anthm : Really liking that app of yours.... much nicer way IMO. |
| 13:24.57 | Mercestes | Oh, it's RTA trying to read something from sip.conf in MySQL most likely. |
| 13:24.59 | festr_ | Mercestes: CLI> set debug |
| 13:25.09 | festr_ | Mercestes: CLI> set verbose |
| 13:25.09 | Mercestes | festr_ did that, set debug 99 |
| 13:25.18 | Mercestes | festr_: 99 as well |
| 13:25.36 | festr_ | Mercestes: edit /etc/asterisk/logger.conf console => notice,warning,error,verbose,debug |
| 13:25.48 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 13:25.55 | anthm | thx |
| 13:26.35 | cjk | hi, im trying to compile asterisk trunk and res_snmp but I just does not compile, does anyone know if I have to set some flags in the Makefile or .... |
| 13:26.52 | Mercestes | Woo, you rock, festr_ |
| 13:27.05 | festr_ | Mercestes: np |
| 13:28.33 | *** join/#asterisk mocker (n=ks@in.kansas.but.not.a.republi.cn) |
| 13:28.37 | ACiDV | Anyone use AddQueueMember with Local/ channel ? ex. AddQueueMember(sales|Local/${CALLERID(num)}@localdevices) ... I can log but never receive any call.. I have also add an hint line but the status is always 'Unknown' when I do a 'Show Queues' |
| 13:28.41 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 13:28.42 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.237.193.Dial1.SanJose1.Level3.net) |
| 13:28.49 | mocker | Does anyone know what needs to be reloaded to reread queues.conf? |
| 13:29.01 | pnk | "reload" ? |
| 13:29.09 | mocker | pnk: Damn. |
| 13:29.15 | ACiDV | mocker ... reload 'app_queues.so' |
| 13:29.15 | *** join/#asterisk s0lid (n=jlq@202.71.179.140) |
| 13:29.32 | ACiDV | sorry... 'reload app_queue.so' |
| 13:29.35 | mocker | For some reason my announce= line isn't being picked up, or doesn't work. |
| 13:29.54 | mocker | Under each queue it's just announce=soundfile, corret? |
| 13:30.31 | [TK]D-Fender | ACiDV : The queue system can't know the status of a Local channel because it isn't a device. it has no clue what tech it will dial if ANY AT ALL. |
| 13:30.54 | Sonderblade | when i call with a softphone and it first registers itself, then call the IVR i get swedish voice prompts, but if i call without the softphone having registered itself, i get english prompts |
| 13:30.58 | Sonderblade | how come? |
| 13:31.01 | [TK]D-Fender | mocker : What kind of announcement are you referring to? |
| 13:31.15 | mocker | [TK]D-Fender: An announcement to an agent of what queue the call is coming in on. |
| 13:31.18 | ACiDV | [TK]D-Fender ... so I must not use Local/ channel ? |
| 13:31.51 | [TK]D-Fender | Sonderblade : Globally you don't have a language set. You're probably going to want to try and do this the smart way and set the language at the START of your IVR.... |
| 13:32.00 | ACiDV | This wiki page use Local/ on queue ... http://www.voip-info.org/wiki/view/Asterisk+Queue+Information |
| 13:32.30 | Sonderblade | [TK]D-Fender: how do you do that? |
| 13:32.40 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.237.193.Dial1.SanJose1.Level3.net) |
| 13:34.07 | [TK]D-Fender | Sonderblade : Look at the LANGUAGE function. |
| 13:34.18 | ACiDV | Thanks, Will try to talk again with Ian @ Digium... or someone else... they send me a dialplan script last week on MSN but it doesn't look to work w/ AddQueueMember |
| 13:34.30 | [TK]D-Fender | ACiDV : Not that you shouldn't use it, just that it has limitations such as knowing its status. |
| 13:34.33 | trelane_ | is there a way to tie up a phone so that the user is only presented wtih one call at a time even while in queue? if I set the maximum to 1 the call immediately goes to voicemail |
| 13:34.37 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 13:34.51 | [TK]D-Fender | ACiDV : pastebin it, we'll take a look |
| 13:34.51 | Sonderblade | [TK]D-Fender: SetLanguage()? That doesn't seem like a smart way |
| 13:35.12 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 13:35.14 | [TK]D-Fender | Sonderblade : Sure its smart... and SetLanguage is deprecated. Us the FUCNTION |
| 13:35.17 | ACiDV | SetLanguage will be depreciated on 1.4 |
| 13:35.23 | mocker | [TK]D-Fender: Have you used that feature before? |
| 13:35.26 | [TK]D-Fender | ACiDV : Its deprecated NOW |
| 13:35.29 | ACiDV | [TK]D-Fender ok wait :) I will past my script (login) |
| 13:35.33 | [TK]D-Fender | mocker : Yes, plenty of times. |
| 13:35.57 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 13:36.02 | mocker | [TK]D-Fender: Hmm, is there a trick to it? |
| 13:36.06 | [TK]D-Fender | mocker : Being in a dual-language city/province/country its something of a reality for me. |
| 13:36.20 | hmmhesays | curse you canucks |
| 13:36.23 | [TK]D-Fender | mocker : Try putting it in your IVR ;) There's a trick! |
| 13:36.24 | Sonderblade | [TK]D-Fender: so with that in the dialplan i don't need it in any config files? |
| 13:36.56 | [TK]D-Fender | Sonderblade : Yes you do (for things like VM access), unless you want to set it for EVERY feature they dial. |
| 13:37.38 | hmmhesays | whoa oh china grove |
| 13:38.37 | [TK]D-Fender | hmmhesays : Don't forget to crank the delay :) |
| 13:38.57 | tzanger | oh great |
| 13:39.01 | tzanger | now I have that song in my head |
| 13:39.14 | tzanger | I wonder if I'd look suspicious playing air guitar in the office |
| 13:39.36 | [TK]D-Fender | hmmhesays : Oh, and I'm seriously loving my new MIDID master controller.... gott ditch my Roland HP-137 digital piano now ASAP to recoup some $ to I can have it pay for it and my next controller + stand |
| 13:39.56 | [TK]D-Fender | tzanger : Do what I do and just bring a real one in with you :) |
| 13:40.10 | tzanger | :-) |
| 13:40.20 | hmmhesays | [TK]D-Fender: haha yeah, and cool |
| 13:40.27 | Sonderblade | [TK]D-Fender: it seems like you would need to write Set(LANGUAGE()=language) multiple times in the dialplan to ensure that the function is called on all calls going through asterisk |
| 13:41.10 | [TK]D-Fender | Sonderblade : Or set it in each devices config (SIP/IAX/ZAP) |
| 13:41.10 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 13:41.12 | hmmhesays | [TK]D-Fender: I went to go play my les paul on friday with ghs boomer 0.10's on it, after playing my strat with .11's on it for the last month they felt let spaghetti noodles |
| 13:41.16 | ACiDV | [TK]D-Fender: http://pastebin.ca/108382 .. my script for ACD login/logout (mimic Agent/ with AddQueueMember).. all work except that agent doesnt receive call |
| 13:41.18 | [TK]D-Fender | Sonderblade : Which is what you should be doing |
| 13:41.33 | Sonderblade | [TK]D-Fender: but i just explained why that didn't work |
| 13:42.18 | [TK]D-Fender | Sonderblade : No you merely said that what you DID didn't work. I never said you did it RIGHT..... |
| 13:42.32 | RoyK | <PROTECTED> |
| 13:43.14 | [TK]D-Fender | RoyK : You don't say...... |
| 13:43.20 | sumasuma | hi how cani dial through the pstn port of sipura connected to the asterisk server |
| 13:43.39 | [TK]D-Fender | ACiDV : ok, so what in there does not work? |
| 13:44.11 | *** join/#asterisk waglik (n=mwegrzyn@polpak.litex.pl) |
| 13:44.36 | Sonderblade | [TK]D-Fender: i don't know what i'm doing wrong then |
| 13:45.01 | trelane_ | is there a chanspy function that works like remote pickup... ie I dial some prefix and then the phone's extension and chanspy follows Hint to find that channel? |
| 13:45.01 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 13:45.04 | ACiDV | [TK]D-Fender... the agent doesnt receive all... and state is always (Unknown) |
| 13:45.17 | hmmhesays | trelane_: you can setup chan spy to do that |
| 13:45.38 | [TK]D-Fender | ACiDV : Ok, the state thing is to be expected. so you're saying that a call goes into the queue but no agents ever get called? |
| 13:45.50 | ACiDV | [TK]D-Fender exact |
| 13:46.20 | [TK]D-Fender | ACiDV : OMG ScopServ..... |
| 13:46.36 | [TK]D-Fender | ACiDV : Shouldn't you be asking Joel for that stuff? :) |
| 13:46.46 | ACiDV | [TK]D-Fender : more exactly FireWorx... |
| 13:46.54 | ACiDV | [TK]D-Fender ... I'm Joel |
| 13:47.08 | trelane_ | hmmhesays, is there an example available, or am I going to have to set up ${SPYGROUP} for every phone? |
| 13:47.20 | [TK]D-Fender | ACiDV ..... <- Andrew @ Belanger :) |
| 13:47.26 | ACiDV | Ahhh =) |
| 13:47.27 | trelane_ | ok this is amusing |
| 13:47.28 | [TK]D-Fender | lol |
| 13:47.29 | ACiDV | Hi Andrew :) |
| 13:47.32 | [TK]D-Fender | salut mon ostie! |
| 13:47.34 | trelane_ | shouldn't you be asking $RANDOMGUY |
| 13:47.36 | trelane_ | that's me! |
| 13:47.37 | ACiDV | hehe =) |
| 13:47.44 | tzanger | <-- Andrew @ ... uh... |
| 13:47.46 | hmmhesays | trelane_: unfortunately it has been a long time since i've setup chanspy |
| 13:47.53 | trelane_ | <---- Andrew at allthingsit |
| 13:47.55 | trelane_ | ok wait |
| 13:47.59 | trelane_ | IS EVERYONE HERE NAMED ANDREW? |
| 13:48.01 | tzanger | heh |
| 13:48.07 | [TK]D-Fender | tzanger : My company...... |
| 13:48.16 | [TK]D-Fender | trelane_ : Resistance is futile! |
| 13:48.16 | tzanger | I'm not at your company |
| 13:48.52 | [TK]D-Fender | tzanger : Correct, merely in our company ;) |
| 13:49.13 | [TK]D-Fender | ACiDV : PB up the "show queues" for it.... |
| 13:49.57 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 13:50.35 | trelane_ | anyone that's used chanspy recently mind pointing me to a decent tutorial? voip-info.org isn't particularly useful on this one |
| 13:50.37 | fuser | SIP/2.0 407 Proxy Authentication Required |
| 13:50.43 | fuser | ive never had to deal with this before |
| 13:50.58 | [TK]D-Fender | ACiDV : And get rid of that "congestion" in your agent dial.. that will answer the channel and prevent the call from redistributing. |
| 13:51.30 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 13:51.34 | esculapio__ | they can help me please, who speaks espanol |
| 13:51.58 | [TK]D-Fender | ACiDV : and line 19 of your first PB has a /n I don't think you meant to put in there.... |
| 13:52.08 | esculapio__ | hola quien puede ayudarme |
| 13:52.35 | esculapio__ | awe6, |
| 13:52.37 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 13:53.13 | trelane_ | anyone want to start an asterisk andrew conspiracy group? |
| 13:53.18 | [TK]D-Fender | ACiDV : Which would probably be the reason it can't for the context to dial out into |
| 13:56.07 | *** join/#asterisk bXi (i=bluepunk@irssi.co.uk) |
| 13:57.01 | bXi | hi |
| 13:57.14 | bXi | i've got a server running with asterisk |
| 13:57.22 | bXi | but i need to test it now |
| 13:57.23 | bXi | any suggestions? |
| 13:57.56 | sevard | i heard sipsak is good |
| 13:58.31 | [TK]D-Fender | bXi : Test it "how" is the question.... what did youset up in *? |
| 13:58.45 | fuser | anyone here work at teliax |
| 13:58.48 | fuser | answer the damn phone |
| 13:58.56 | bXi | [TK]D-Fender: i work at a company |
| 13:59.09 | bXi | they want to make a product with * |
| 13:59.19 | bXi | its my job now to find out as much as possible about it |
| 13:59.37 | bXi | they already have a box running it somewhere but i cant look at their config files and such |
| 14:00.31 | fourcheeze | can someone remember the command in debian to use mpg123 instead of mpg321? |
| 14:00.35 | hmmhesays | bXi: good luck wit that |
| 14:00.38 | hmmhesays | *with |
| 14:00.41 | [TK]D-Fender | bXi : You can't test what you can't see really..... |
| 14:00.56 | hmmhesays | fourcheeze make install mpg123 in your asterisk source directory |
| 14:01.07 | fourcheeze | oh |
| 14:01.21 | [TK]D-Fender | fourcheeze : Just use Native MoH and forget MPG* altogether. |
| 14:01.33 | hmmhesays | that works too |
| 14:02.06 | hmmhesays | i never did get my wcusb gadget workign friday |
| 14:02.14 | fourcheeze | [TK]D-Fender: how do I do that? |
| 14:02.42 | [TK]D-Fender | fourcheeze : Look it up on the WIKI. its well documented and very easy |
| 14:02.49 | hmmhesays | fourcheeze you really don't have to do anything I think the config files are default to use it now |
| 14:03.04 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:03.08 | fourcheeze | this is 1.2.4 |
| 14:03.23 | fourcheeze | right now I just want to get moh as it was prior to meltdown |
| 14:03.43 | hmmhesays | what happened? |
| 14:03.50 | fourcheeze | [TK]D-Fender: does native MOH go in a loop? |
| 14:04.14 | fourcheeze | hmmhesays: well, I had a server crash, so I failed over to my backup but * kept sending 401s out |
| 14:04.27 | fourcheeze | and I noticed my backup is missing moh |
| 14:04.38 | [TK]D-Fender | fourcheeze : it cycles through them... seems to work great for everyone I've herd from. I use it in all my installs |
| 14:04.55 | hmmhesays | fourcheeze, you never put your backup into production did you... tsk tsk |
| 14:05.00 | fourcheeze | [TK]D-Fender: does it use the same stream for everyone, or a new one for each? |
| 14:05.12 | [TK]D-Fender | fourcheeze : New for each I believe |
| 14:05.14 | fourcheeze | hmmhesays: well it's only one little flaw |
| 14:05.19 | [TK]D-Fender | fourcheeze : Could be wrong.... go read! |
| 14:05.22 | fourcheeze | however I'm very cross with * today |
| 14:05.28 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 14:05.38 | hmmhesays | have a beer, swear at it and get your job done |
| 14:05.38 | fourcheeze | basically it just refused all my clients logins |
| 14:05.40 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 14:05.45 | hmmhesays | such is the life of an IT person |
| 14:06.41 | trelane_ | hey, if the software just worked, we wouldn't get paid an egregious amount of money to fix it! |
| 14:07.02 | fourcheeze | hmmhesays: no, it's not normally *this* bad |
| 14:07.10 | fourcheeze | I considered quitting the whole voip thing at least 3 times |
| 14:07.52 | hmmhesays | haven't been around long huh? |
| 14:07.57 | fourcheeze | :-) |
| 14:08.02 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:08.10 | fourcheeze | you have days like this too? |
| 14:08.24 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 14:08.34 | hmmhesays | of course |
| 14:08.34 | fourcheeze | somehow it's worse when people's phones are involved |
| 14:08.45 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79EAA.pool.t-online.hu) |
| 14:08.58 | hmmhesays | always, telephones have been around for 100 years and are generally a stable appliance |
| 14:09.15 | hmmhesays | so people have learned to rely on them |
| 14:09.21 | fourcheeze | sure |
| 14:10.13 | hmmhesays | they don't understand or care what medium their phones are operating on, they want the same reliability they've come to expect from the magic talking headset |
| 14:10.27 | fourcheeze | indeed |
| 14:10.42 | hmmhesays | does anyone have #asterisk log from friday? i'm looking for a link I posted |
| 14:11.00 | fourcheeze | I have one customer who, whenever there is a problem, give me the history of their dealings with our company |
| 14:11.14 | fourcheeze | this takes about 15 minutes before I get around to helping them |
| 14:11.59 | hmmhesays | yeah |
| 14:12.10 | hmmhesays | some people just like to hear themselves talk |
| 14:12.25 | hmmhesays | you can be sure that particular person doesn't have many friends that listen to him |
| 14:12.25 | jbroome | fourcheeze: i have a client that does the same thing. I let him drone on since we charge by the hr. |
| 14:12.34 | fourcheeze | hehe |
| 14:12.39 | nortex | Or they like to over descibe the problem |
| 14:12.46 | fourcheeze | jbroome: sounds like I need to work for your boss |
| 14:12.56 | hmmhesays | yeah because they think they have some idea what is going on |
| 14:13.14 | fourcheeze | hmmhesays: how's the mixer BTW? |
| 14:13.40 | nortex | Yeah it is hard to listen and keep in mind the fix you thought of after the first sentence. |
| 14:17.10 | Kerry_G | anyone have voicemail attachments working with blackberrys? |
| 14:17.49 | tzanger | Kerry_G: not me. I'd love a real SMS gateway that I don't have to pay horrendous amounts of money for just to send MWI notification |
| 14:18.37 | hmmhesays | fourcheeze: pretty good, discovered we had one unbalanced input plugged into it when the rest was balanced |
| 14:18.40 | hmmhesays | that is NO GOOD |
| 14:19.16 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:19.50 | fourcheeze | hmmhesays: that's probalby quite loud |
| 14:20.04 | hmmhesays | definately not good |
| 14:21.33 | [TK]D-Fender | hmmhesays : DirectBox++ |
| 14:21.49 | *** join/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 14:21.58 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 14:22.32 | [TK]D-Fender | syn : ack |
| 14:22.35 | syn | rst! |
| 14:22.36 | *** join/#asterisk Quintana (n=sylvain@office.proformatique.com) |
| 14:22.37 | syn | :) |
| 14:22.55 | [TK]D-Fender | syn : scripted? |
| 14:22.59 | syn | no |
| 14:23.13 | [TK]D-Fender | syn : just wondering... would have been funny... |
| 14:23.17 | syn | true |
| 14:23.23 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 14:23.24 | syn | but my bot is named ack ;) |
| 14:23.27 | [TK]D-Fender | "er" |
| 14:23.27 | hmmhesays | [TK]D-Fender: eh? |
| 14:24.01 | [TK]D-Fender | hmmhesays : get a direct box to take in your unbalanced signal. I should get one for my Boss GT-8 |
| 14:24.04 | bkidney | Has anyone seen this problem. When I dial from my Cisco 7960 to an external line, I still hear the cisco ringing in my ear after the other extension has picked up (we can talk and he does not here the ringing)? |
| 14:25.06 | syn | so |
| 14:25.15 | syn | I'm trying to build chan_misdn |
| 14:25.18 | hmmhesays | [TK]D-Fender: goes from unbalanced to balanced? |
| 14:25.33 | syn | but it looks like the API of the mISDN I installed doesn't match |
| 14:25.49 | syn | for example |
| 14:25.51 | syn | ie.c: In function `enc_ie_complete': |
| 14:25.51 | syn | ie.c:72: error: incompatible types in assignment |
| 14:26.09 | *** join/#asterisk Kernel-Kris (n=kkirklan@lfkn-fw.angelinacounty.net) |
| 14:26.12 | syn | line 72 is |
| 14:26.27 | syn | qi->sending_complete = p - (unsigned char *)qi - sizeof(Q931_info_t); |
| 14:26.34 | syn | the rvalue is an unsigned char * |
| 14:26.42 | syn | the lvalue is an ie_info_t ... |
| 14:26.46 | Kernel-Kris | any suggestions of fixing a slight echo on outgoing calles through a wildcard single port fxo |
| 14:27.14 | Kerry_G | have you used ztmonitor to set your levels? |
| 14:27.48 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 14:28.12 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 14:28.15 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79EAA.pool.t-online.hu) |
| 14:28.20 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 14:29.09 | Kernel-Kris | Kerry_G: no i havnt ive never had any problems out of these cards, when i run ztmonitor what should i look for and how would i fix it |
| 14:29.41 | [TK]D-Fender | hmmhesays : Correct |
| 14:29.49 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-127-74-61.aubnin.fios.verizon.net) |
| 14:31.16 | *** part/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 14:31.28 | [TK]D-Fender | hmmhesays : http://www.music123.com/Music123/Search/Default.aspx?Ntk=Default&Ntx=mode+matchall&Ntt=direct+box |
| 14:32.39 | *** join/#asterisk Decryptus (n=JDD@lau06-3-82-240-153-51.fbx.proxad.net) |
| 14:33.01 | *** join/#asterisk tempest1 (n=asf@adsl-153-43-12.chs.bellsouth.net) |
| 14:36.09 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:37.04 | *** join/#asterisk MikeJ (n=vircuser@204.250.115.159) |
| 14:37.50 | hmmhesays | [TK]D-Fender: I see |
| 14:38.23 | hmmhesays | bah I can't get flash to work in linux at all |
| 14:38.41 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 14:39.56 | Sonderblade | id like to implement a functionality to my asterisk so that after you transfer a call, if the extension you transferred it to didn't answer, the call "bounces back" to you, anyone know how to do that? |
| 14:40.02 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:40.41 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 14:41.07 | *** part/#asterisk Decryptus (n=JDD@lau06-3-82-240-153-51.fbx.proxad.net) |
| 14:41.20 | file | Sonderblade: I assume a blind transfer? |
| 14:41.36 | Sonderblade | file: no, attended call transfer |
| 14:42.01 | file | Sonderblade: don't you usually wait on the line until someone answers? |
| 14:42.04 | waglik | I'm still fighting with the asterisk not detecting caller hanging up before pickup on asterisk side |
| 14:42.04 | [TK]D-Fender | Sonderblade : just have it hang up if they don't answer. |
| 14:42.23 | *** part/#asterisk syn (i=syn@kenobi.sceen.net) |
| 14:42.29 | file | hanging up random calls is the answer to life's problems |
| 14:42.43 | Sonderblade | [TK]D-Fender: that would make the caller pissed off |
| 14:43.20 | waglik | I've connected an Alcatel OmniPCX to the second span of my digium card |
| 14:43.24 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:43.26 | waglik | and everything worked ok |
| 14:43.26 | Sonderblade | i.e: secretary transfers to boss, boss is not available, secretary get call back and transfer it to some manager |
| 14:43.37 | waglik | the only difference in logs is: |
| 14:43.53 | file | isn't that what an attended transfer is all about? |
| 14:44.14 | waglik | when the hangup is from Telco side: |
| 14:44.16 | waglik | Channel 0/1, span 1 got hangup ACK |
| 14:44.32 | Sonderblade | file: no because after you have transferred the call there is no way to take it back |
| 14:44.32 | waglik | when hanging up from OmniPCX: |
| 14:44.37 | *** part/#asterisk Kernel-Kris (n=kkirklan@lfkn-fw.angelinacounty.net) |
| 14:44.46 | waglik | Channel 0/31, span 2 got hangup request |
| 14:45.02 | file | Sonderblade: but you don't transfer the call until you know the boss can take it |
| 14:45.16 | waglik | it seems, the telco is signalling hangup incorrectly |
| 14:45.23 | waglik | am I right? |
| 14:46.05 | Sonderblade | file: the boss is in a different building, so how exactly are you going to figure that out? |
| 14:46.17 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 14:46.44 | file | Sonderblade: attended transfer is two parts - one consists of calling the other person and then you speak to them and say hey can you take this call? and then after that you either transfer the person who called you to them, or not |
| 14:46.45 | file | afaik |
| 14:48.19 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 14:49.03 | *** part/#asterisk jaike (n=a@203.115.188.120) |
| 14:49.13 | *** join/#asterisk PoWeRKiLL (n=powerkil@85.64.221.116.dynamic.barak-online.net) |
| 14:49.31 | Sonderblade | file: that is pretty complicated, and proprietary pbx:es have the bounce back call transfer feature |
| 14:50.31 | [TK]D-Fender | Sonderblade : I mean if you do an attended transfer and the person you dials doesn't answer, don't have it fall to VM. Just have it hang up. That will automatically drop you back to the call you were transferring |
| 14:50.37 | [TK]D-Fender | Sonderblade : Thats a blind transfer w/ callback |
| 14:53.41 | Sonderblade | [TK]D-Fender: But that requires me to first check if the extension im trying to transfer the call to is present. I don't want to do that step. |
| 14:53.57 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 14:54.33 | [TK]D-Fender | Sonderblade : well forget that term "attended". make a dial-plan pattern to do the dial & callback |
| 14:54.41 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 14:55.23 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 14:56.52 | fourcheeze | anyone doing any kind of clustering that allows for hints/presence etc? |
| 14:57.33 | Sonderblade | [TK]D-Fender: Yeah, that's the plan. But I haven't found a way to check if the call was transferred. You need to know that to check if you should drop to VM or "bounce back" |
| 14:57.37 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 14:58.31 | *** join/#asterisk s0lid (n=jlq@202.71.179.140) |
| 14:58.33 | [TK]D-Fender | Sonderblade : I have just templated the dialplan setup you'd need to do this including multi-ple call back. Not that difficult. |
| 14:59.38 | *** part/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 14:59.44 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 15:02.08 | Sonderblade | [TK]D-Fender: how do you check if the call was transferred? |
| 15:03.17 | [TK]D-Fender | Sonderblade : check the dialstatus upon dialing. on failure have it call back the person who transferred it. |
| 15:04.09 | Sonderblade | [TK]D-Fender: how do you know who transferred it? |
| 15:05.37 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 15:06.17 | Kerry_G | With Blackberry Enterprise Server, I can set wav as an allowed attachment but opening it gives a format error. Appearently in BES 4.1, there is only one accepted wave format and its not the default format from asterisk |
| 15:06.43 | [TK]D-Fender | Sonderblade : last hint for you : 2 ways. Either set a channel variable in the phone device, or do an attended transfer and match by callerID for the callback.. |
| 15:08.31 | *** join/#asterisk daysmen3 (n=primus@host86-140-208-99.range86-140.btcentralplus.com) |
| 15:09.08 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
| 15:09.13 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 15:13.21 | Sonderblade | [TK]D-Fender: By attended transfer do you mean "press TRANSER, press extension #, press SEND"? |
| 15:14.18 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:14.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:16.27 | *** join/#asterisk njan (n=james@about/security/staff/njan) |
| 15:19.07 | hmmhesays | ~seen zoa |
| 15:19.18 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 10d 1h 55m 43s ago, saying: 'SynUK: cisco.com'. |
| 15:19.41 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-127-74-61.aubnin.fios.verizon.net) |
| 15:21.07 | [TK]D-Fender | Sonderblade : Something like that. |
| 15:21.52 | *** join/#asterisk Blafasel (n=bpodszun@relay3.vistream.de) |
| 15:23.28 | Sonderblade | [TK]D-Fender: Then that doesn't work. Say the call is transferred to extension 150, the call no different from how it would have been if extension 150 had been dialled directly |
| 15:23.55 | [TK]D-Fender | Sonderblade : It can work, you need to think on it a bit more... |
| 15:24.05 | Blafasel | Hi there.. After failing horribly at SIP<->SS7 calls I tried something more basic.. Both the 500 default extension (asterisk demo call) and the 600 (echo latency evaluation call) are muted here. While SIP<->SIP phones work fine.. Any ideas where I can start to find out more about it? |
| 15:24.43 | [TK]D-Fender | Sonderblade : Think about the # you are going to dial. Thats the only piece of info you really need..... |
| 15:28.37 | eKo1 | Blafasel: how were you achieving SIP<->SS7? |
| 15:29.22 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.150) |
| 15:29.51 | Blafasel | eKo1: Softphone/mobile <-SIP-> Asterisk <-chan_ss7-> Provider |
| 15:31.52 | Supaplex | I know, let's us a pbx. :P |
| 15:31.56 | jarrod | anyone have algorithm for parsing country code from dialed strings? |
| 15:32.42 | Sonderblade | [TK]D-Fender: do you mean that the transferer should prepend a sigil to the number for it to work? so the transferer calls *150 instead of 150? |
| 15:34.18 | Juggie | algorithm? |
| 15:35.36 | sevard | !seen dlynes |
| 15:35.43 | eKo1 | Blafasel: Ah. And it didn't work out for you? |
| 15:35.44 | sevard | !seen dlynes-work |
| 15:35.50 | sevard | !seen dlynes~ |
| 15:35.55 | sevard | god damnit. |
| 15:36.08 | sevard | since dlynes isn't here, the voicemailking, can anyone answer this |
| 15:36.09 | sevard | ;4310 => -5432,Sales,sales@marko.net |
| 15:36.09 | eKo1 | jarrod: you can easily make on up. |
| 15:36.14 | sevard | what's the '-' flag? |
| 15:37.06 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.static.sasknet.sk.ca) |
| 15:37.15 | momelod | q: if i want to tweak my zaptel kernel module (for example to add/remove aggressive suppression) is there a better way to do it then editing ztconfig.h and recompiling + installing it every time? |
| 15:37.30 | Blafasel | eKo1: No, not at all. No audio |
| 15:38.07 | [TK]D-Fender | Sonderblade : You're catching on :) |
| 15:38.27 | Tall-guy | gents, I've got some jittery/bad sampling issues going on my zaptel TDM400's with ZAP to ZAP calls (no SIP!), would anyone care to listen to a small .WAV file sample of an inbound call and tell me what they think is the cause? |
| 15:38.34 | [TK]D-Fender | Sonderblade : or do an attended transfer to pick up that chan var. |
| 15:40.01 | sevard | [TK]D-Fender: do you know? ;/ |
| 15:40.14 | eKo1 | Blafasel: That sucks... |
| 15:40.31 | Blafasel | Uhm? Yes. |
| 15:41.02 | Blafasel | But failing to use even the simple voicemail stuff etc. is far worse.. |
| 15:41.33 | eKo1 | How is it failing? |
| 15:41.51 | Blafasel | I don't hear anything. It doesn't record anything. |
| 15:42.18 | Blafasel | And * spawns 2 or more mpg123 processes with each eating one cpu at 100%.. |
| 15:42.18 | [TK]D-Fender | sevard : Looks more like a typo |
| 15:43.55 | waglik | yes, yes, yes :) I found the reason of my hungup problems |
| 15:43.58 | Qwell | [TK]D-Fender, sevard: voicemail.conf says exactly what it does |
| 15:44.38 | eKo1 | Blafasel: What version of * are you using? |
| 15:44.49 | Qwell | file: nub |
| 15:45.00 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:47.07 | hmmhesays | idefisk for linux is not working well |
| 15:47.16 | hmmhesays | where is zoa dag nabit |
| 15:47.19 | Blafasel | eKo1: 1.2 |
| 15:48.06 | file | hmmhesays: lol |
| 15:48.34 | salviadud | have you guys heard the new tool album on MOH? it rocks |
| 15:48.48 | hohum | 10000 days? |
| 15:48.54 | hohum | that came out a while ago already |
| 15:49.03 | salviadud | i know |
| 15:49.08 | salviadud | i just heard it though |
| 15:49.19 | hohum | it isn't tool's best work IMHO |
| 15:49.55 | hohum | then again nothing beats songs like Lateralus, Stink Fist, etc |
| 15:50.01 | *** join/#asterisk sumasuma (n=sumase@cm222.omega183.maxonline.com.sg) |
| 15:50.09 | salviadud | Rosetta Stoned |
| 15:50.16 | salviadud | that song in particular, blew me away |
| 15:50.23 | salviadud | and made me laugh, at the same time |
| 15:50.37 | sumasuma | i could not make my sipura 3000 to register to my asterisk server |
| 15:50.45 | sumasuma | can anyone please help me ? |
| 15:50.51 | salviadud | sumasuma, jesus man, it's sip |
| 15:51.22 | salviadud | sumasuma, have you registered channels already? |
| 15:51.34 | sumasuma | yes |
| 15:51.41 | sumasuma | i connected with xlite and it is working fine |
| 15:51.46 | sumasuma | and can make outgoing calls |
| 15:51.56 | sumasuma | i want to connect sipura to use the fxo line |
| 15:52.05 | sumasuma | it is not at all registering |
| 15:52.57 | sumasuma | salviadud, can you please help me to get it done ? |
| 15:53.02 | PoWeRKiLL | someone have a nokia e61 ? |
| 15:53.14 | salviadud | pastebin your sip.conf file |
| 15:53.20 | sumasuma | sure |
| 15:53.36 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 15:54.04 | *** join/#asterisk watchy (n=gweg@office2.gwhsi.com) |
| 15:54.17 | *** join/#asterisk csplinter (n=csplinte@adsl-216-63-102-177.dsl.bumttx.swbell.net) |
| 15:54.19 | watchy | how come zap show channels isnt in my svn asterisk? |
| 15:54.33 | salviadud | sumasuma, i'm behind a firewall |
| 15:54.35 | Qwell | watchy: Do you have chan_zap loaded? |
| 15:54.49 | watchy | hmm |
| 15:54.55 | watchy | i woulda thought it loaded by default |
| 15:54.59 | watchy | its a fresh install |
| 15:55.02 | *** join/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 15:55.07 | Qwell | Did you install zaptel first? |
| 15:55.24 | csplinter | Were can I find a list of compatible phones? |
| 15:55.25 | watchy | yea |
| 15:55.25 | sumasuma | salviadud, got it ? |
| 15:55.52 | [TK]D-Fender | csplinter : Clarify that a bit if you please.... |
| 15:55.54 | watchy | maybe i know why qwell |
| 15:55.55 | watchy | brb |
| 15:56.58 | csplinter | <[TK]D-Fender>: sorry I hadnt heard of asterisk before 15 minutes ago, I guess what I want to know is my companys 3com voip phone good for testing with asterisk |
| 15:57.01 | watchy | i just ran under zaptel make install but its not showing up |
| 15:57.18 | Qwell | watchy: You have to recompile asterisk after installing zaptel |
| 15:58.28 | watchy | qwell:ah |
| 15:58.37 | eKo1 | csplinter: What signaling does it use? |
| 15:58.45 | watchy | should zaptel always be installed first? |
| 15:59.00 | Qwell | watchy: If you want things like chan_zap, or meetme |
| 15:59.05 | watchy | heh |
| 15:59.23 | watchy | ok zaptels installing now to recomp asterisk |
| 15:59.39 | csplinter | eKo1: you'll have to forgive me I'm not really familiar with all this, by signal what do you mean? I'll have to find out. |
| 15:59.53 | eKo1 | csplinter: SIP, H.323, IAX |
| 15:59.58 | puzzled | csplinter: afaik the 3com phones use a proprietary protocol but can use sip. the catch is that you need the 3com pbx to actually download the sip firmware into the phone |
| 16:00.00 | csplinter | oh |
| 16:00.06 | csplinter | ill find out |
| 16:00.44 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 16:00.55 | csplinter | hmm I see, so If I can get it use SIP signaling it will though? |
| 16:00.56 | Sonderblade | [TK]D-Fender: which chan var? |
| 16:01.05 | eKo1 | csplinter: Yes. |
| 16:01.13 | csplinter | great thx alot guys |
| 16:01.14 | [TK]D-Fender | Sonderblade : Invent one yourself. |
| 16:01.29 | puzzled | csplinter: not sure but at least it will theoretically talk to same protocol |
| 16:01.33 | [TK]D-Fender | Sonderblade : Read up on sip.conf |
| 16:02.02 | *** part/#asterisk Skram (n=MarkS@70.86.176.56) |
| 16:02.32 | [TK]D-Fender | csplinter : IIRC 3COM's use a specific build of H.323 and integration is probably going to be hard. Go verify what protocols are available for it. |
| 16:03.24 | watchy | hmm |
| 16:03.34 | watchy | what the hell zaptel shit still aint showing up |
| 16:03.43 | Qwell | watchy: Did you reinstall *? |
| 16:03.51 | file | the mailman is very very evil |
| 16:03.53 | watchy | yea i recompiled it and reinstalled it |
| 16:04.01 | Qwell | watchy: with `make install`? |
| 16:04.05 | Qwell | on zaptel and asterisk |
| 16:04.06 | watchy | yes |
| 16:04.11 | watchy | yes |
| 16:04.14 | Qwell | file: most are |
| 16:04.22 | watchy | ive never had this issue before |
| 16:04.23 | Qwell | watchy: and you restarted? |
| 16:04.35 | Qwell | (asterisk, not the server) |
| 16:04.41 | watchy | pbx asterisk # killall -9 asterisk |
| 16:04.42 | watchy | asterisk: no process killed |
| 16:05.14 | watchy | yea |
| 16:05.18 | watchy | what the hell |
| 16:05.26 | eKo1 | check for the presence of chan_zap.so |
| 16:05.35 | Qwell | bbl, work |
| 16:05.38 | Qwell | file: 2 weeks! |
| 16:05.48 | eKo1 | If it doesn't exist, the recompile didn't work. |
| 16:05.49 | watchy | where should it be? |
| 16:05.56 | file | Qwell: zomg |
| 16:06.00 | Qwell | :D |
| 16:06.07 | file | Qwell: get to work! |
| 16:06.12 | Qwell | pfft, why? |
| 16:06.14 | file | Qwell: prepare for madness! |
| 16:06.18 | Qwell | not like they're gonna fire me or anything :P |
| 16:06.24 | file | lawl |
| 16:06.35 | file | that's true |
| 16:06.39 | eKo1 | watchy: just enter `locate chan_zap.so' |
| 16:06.40 | Qwell | (they actually can't, now) |
| 16:07.17 | Qwell | I dread reading my email this morning |
| 16:07.37 | file | go! |
| 16:07.41 | Qwell | so many "omg, Qwell, noes!" |
| 16:07.42 | Qwell | meh |
| 16:07.56 | watchy | im running gentoo aint no locate built in and i have no idea what package it is |
| 16:08.02 | Qwell | slocate |
| 16:08.12 | file | Qwell: maybe they'll make your 2 weeks miserable and terminate your net access |
| 16:08.19 | Qwell | file: sa-weet! |
| 16:08.28 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 16:08.35 | Blafasel | watchy: Usually it's in /usr/lib/asterix/modules |
| 16:08.42 | watchy | pbx zaptel # slocate chan_zap.so |
| 16:08.42 | watchy | pbx zaptel # |
| 16:08.43 | *** join/#asterisk riddlebox (n=blah@24-207-167-238.dhcp.stls.mo.charter.com) |
| 16:08.43 | Qwell | Blafasel: try again |
| 16:08.44 | Blafasel | asterisk, of course ;) |
| 16:08.49 | Qwell | watchy: updatedb first |
| 16:08.53 | Qwell | bbl |
| 16:09.07 | eKo1 | watchy: then use find |
| 16:09.42 | watchy | well chan_zap aint under /usr/lib/asterisk/modules |
| 16:09.44 | watchy | hrm |
| 16:11.09 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:11.28 | watchy | zaptel is what makes chan_zap.so? |
| 16:12.05 | eKo1 | What does find say? |
| 16:12.13 | watchy | nothing |
| 16:12.14 | eKo1 | chan_zap.so is made by *. |
| 16:12.21 | Sonderblade | [TK]D-Fender: what has sip.conf to do with call transfering? |
| 16:12.28 | eKo1 | Well, that means you did not recompile properly. |
| 16:12.53 | watchy | im going to try again |
| 16:12.57 | watchy | to recomp ast |
| 16:12.57 | eKo1 | watchy: please do a make clean && make && make install in the * source dir. |
| 16:13.12 | watchy | Package configured for: |
| 16:13.13 | watchy | <PROTECTED> |
| 16:13.13 | watchy | <PROTECTED> |
| 16:13.19 | watchy | i just did make clean and ./configure |
| 16:13.27 | watchy | now im making |
| 16:14.17 | eKo1 | ./configure ? |
| 16:14.27 | watchy | when you type make it does it |
| 16:14.27 | eKo1 | Did you emerge * or something? |
| 16:14.41 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:14.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:14.47 | watchy | nope |
| 16:14.52 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:14.52 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:15.16 | watchy | if this dont work ill get * current instead of svn |
| 16:16.17 | *** join/#asterisk evisu (i=hIRC@bzq-88-155-80-250.red.bezeqint.net) |
| 16:17.26 | eKo1 | That may well be your problem. Oh boy... |
| 16:17.34 | watchy | wierd |
| 16:17.46 | watchy | it ocmpiles all channels in asterisk/channels |
| 16:17.51 | watchy | but chan_zap |
| 16:17.53 | *** join/#asterisk RoyK (n=roy@ti211310a080-0478.bb.online.no) |
| 16:18.29 | watchy | im gonna get current |
| 16:18.48 | eKo1 | Maybe zap isn't install properly. |
| 16:19.18 | watchy | i guess that could be the case but i dunno why |
| 16:19.30 | watchy | make clean; make linux26; make install |
| 16:19.31 | eKo1 | Do you have a zap card? |
| 16:19.50 | watchy | Zapata Telephony Interface Registered on major 196 |
| 16:19.50 | watchy | Zaptel Version: SVN-trunk-r1249 Echo Canceller: MG2 |
| 16:19.51 | watchy | yea |
| 16:19.57 | watchy | TDM400p thats loading fine |
| 16:20.09 | evisu | whats the max amount of recommended concurrent calls on one box for a 3ghz i686 ? |
| 16:20.21 | eKo1 | evisu: depends |
| 16:20.41 | evisu | on what factors? |
| 16:20.46 | Blafasel | hrhr.. zaptel from svn as well? |
| 16:20.52 | eKo1 | watchy: you want to do `make clean && make linux26 && make install' instead |
| 16:20.55 | watchy | blA: YEo |
| 16:20.56 | Blafasel | Someone's eager to try the bleeding edge |
| 16:20.57 | watchy | oops |
| 16:21.45 | watchy | yea im probably retarded going with that shit |
| 16:22.01 | watchy | i think i need to go back to current, of course i havent tried it yet |
| 16:22.07 | watchy | this is a new box |
| 16:22.19 | watchy | i thought goin with svn wouldnt hurt |
| 16:22.53 | evisu | eKo1... what does it depend on? |
| 16:23.11 | *** join/#asterisk madfactor (i=madfacto@74.128.34.115) |
| 16:23.25 | watchy | dloading current * now though |
| 16:24.13 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 16:24.40 | nortex | On queues, does ringall skip agents who are on the phone? |
| 16:24.46 | Sonderblade | [TK]D-Fender: I found the BLINDTRANSFER cha variable, is that the one you have been talking about? |
| 16:24.58 | eKo1 | evisu: a list of things that I don't care to enumerate right now |
| 16:25.44 | evisu | great, thanks. |
| 16:25.45 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:25.54 | eKo1 | evisu: all I can tell you is that there is no formula for it and if there is one, it will have many variables besides the clock speed of your CPU |
| 16:25.56 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 16:26.09 | evisu | besides bandwidth |
| 16:26.13 | evisu | and codecs |
| 16:26.16 | watchy | i hate fucking ie |
| 16:26.25 | watchy | now i cant visit some sites for some reason |
| 16:26.29 | watchy | i just get white blank pages |
| 16:26.32 | eKo1 | don't use it then |
| 16:26.52 | watchy | ive never used firebox but im swithcing right now |
| 16:26.53 | salviadud | use the fire |
| 16:27.16 | watchy | and ill use firefox till the new macbook pros come out the 8th |
| 16:27.20 | watchy | and then order me one of them |
| 16:27.57 | eKo1 | Use firefox or opera. |
| 16:30.25 | *** part/#asterisk mitcheloc (n=mitchelo@c-24-23-37-212.hsd1.ca.comcast.net) |
| 16:30.35 | *** join/#asterisk mitcheloc (n=mitchelo@c-24-23-37-212.hsd1.ca.comcast.net) |
| 16:31.01 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:31.01 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:33.05 | watchy | ok i just installed zaptel 1.2.7 or whatever |
| 16:33.10 | watchy | ocmpin asterisk now |
| 16:33.30 | Blafasel | Okay, I managed to do SIP<->SIP calls. They work fine and reliable. I can issue SIP<->SS7/ISUP calls as well, but I don't hear any audio. But - and that's the point where my brain starts to hurt: If I put the SIP<->SS7 call on hold on my softphone, the SS7/landline guy listens to my fancy default "on-hold" music. |
| 16:34.20 | Blafasel | Is there any way to check/change the way * "translates" between SIP and SS7? How does this reencoding/retransmission work? |
| 16:34.23 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.14) |
| 16:34.34 | *** join/#asterisk antiPosix (n=jkezar@159.105.109.200) |
| 16:34.54 | eKo1 | Blafasel: set your codecs to ulaw |
| 16:35.16 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 16:35.17 | Blafasel | eKo1: On the SIP side? Is there any other place where I'd need to do that? |
| 16:35.18 | smackus | how can I restrict calls... for example I want to restrict anything _19XXNXXXXXX |
| 16:35.19 | antiPosix | I compiled and started asterisk. What do I need to do to setup two XLite softphones to talk to each other? |
| 16:35.42 | antiPosix | on a scale of 1 to expert I am .1 (so bare with me) |
| 16:35.49 | hmmhesays | anyone got idefisk runnining in linux? |
| 16:35.56 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 16:36.02 | watchy | wtf is idefisk |
| 16:36.08 | denon | win32 IAX client |
| 16:36.41 | bpiper | antiPosix: do this exten => _1900.,1,Congestion |
| 16:36.45 | watchy | hmm asterisk current and zaptel current work |
| 16:37.25 | nortex | hmmhesays, I do in windows, let me try it real quick on my linux box. |
| 16:37.39 | antiPosix | bpiper: um...ok |
| 16:37.43 | Blafasel | eKo1: Restricting the SIP client to ulaw doesn't change anything, unfortunately. Still successful calls without audio |
| 16:38.32 | hmmhesays | denon: close but not quite |
| 16:38.39 | antiPosix | bpiper: in extensions.conf I take it |
| 16:38.45 | hmmhesays | its a mac/linux/win32 client |
| 16:38.55 | denon | hmmhesays: ah, well I only use it on win32 |
| 16:38.58 | denon | but its an IAX client |
| 16:39.01 | denon | whatever platform |
| 16:39.03 | noname32 | whats the correct syntax for creating an ext to use as a speed dial? i am trying to use this http://pastebin.ca/108647 i am getting prority issues what should it look like? |
| 16:39.30 | bpiper | anitPosix: yea, you may try the asterisk users email list |
| 16:39.35 | file | noname32: third line down has NO priority |
| 16:39.43 | hmmhesays | noname32: wonder why that is |
| 16:39.52 | file | noname32: silliness I say |
| 16:39.59 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 16:40.00 | noname32 | lol so should 3? |
| 16:40.05 | eKo1 | Blafasel: Are you using the latest stable versions of * and zapata? |
| 16:40.10 | hmmhesays | could |
| 16:40.42 | Blafasel | eKo1: * yes, zapata? No, not at all |
| 16:40.55 | eKo1 | Please upgrade. |
| 16:41.03 | RoyK | ~pb |
| 16:41.04 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 16:41.08 | Blafasel | No, you didn't get me here. I'm not using zapata |
| 16:41.24 | Blafasel | This is just zaptel/chan_ss7 |
| 16:41.34 | Blafasel | (both are up to date, yes) |
| 16:41.48 | hmmhesays | i'm still having trouble with this damn usb fxs device |
| 16:41.55 | eKo1 | Blafasel: ah, OK. |
| 16:42.13 | file | hmmhesays: not... that... |
| 16:42.32 | hmmhesays | file: huh 0_o |
| 16:42.43 | file | hmmhesays: wcusb? |
| 16:42.54 | hmmhesays | file: ja |
| 16:43.00 | hmmhesays | want to see my pastebin? |
| 16:43.00 | *** join/#asterisk oej (n=oej@65.197.203.67) |
| 16:43.11 | file | hmmhesays: I've never seen one, touched one, looked at the driver, but sure! |
| 16:43.27 | hmmhesays | http://pastebin.ca/104155 |
| 16:43.29 | hmmhesays | lol |
| 16:43.32 | *** join/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 16:43.42 | bkidney | <PROTECTED> |
| 16:43.48 | hmmhesays | file: i found this thing in a box and it looked like the thing from the old asterisk dev kit lite or whatever they called it |
| 16:44.08 | file | hmmhesays: device exists? udev fine? |
| 16:44.26 | *** join/#asterisk sp0n9e (n=sp0n9e@69.12.216.48) |
| 16:44.30 | hmmhesays | yep |
| 16:45.02 | hmmhesays | that drive auto loads when you plug it in too |
| 16:45.13 | coppice | Blafasel: are you trying to use chan_ss7 with a T1? |
| 16:45.16 | sp0n9e | i need to get 8-10 phone lines in an asterisk server...4FXO ports is what i want, right? |
| 16:45.27 | Blafasel | coppice: E1 |
| 16:45.31 | Blafasel | But - yes |
| 16:45.44 | coppice | then why did you restrict things to ulaw? |
| 16:46.07 | Blafasel | coppice: Because eKo1 just advised me to do that and I've no clue how to fix this issue.. |
| 16:46.08 | hmmhesays | sp0n9e: something wrong with your math there |
| 16:46.20 | [TK]D-Fender | Sonderblade : No. look at "setvar" and use your imagination.... |
| 16:46.21 | sp0n9e | hmmhesays: yeah, i'll need multiple cards |
| 16:46.41 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:46.42 | eKo1 | Hmm...I have a sip peer with fromuser = foo. However, when that peer makes a call, I don't get SIP/foo as the channel name. |
| 16:46.48 | hmmhesays | might be easier to go with an 8 port external gateway, rather than put two cards in |
| 16:46.49 | [TK]D-Fender | sp0n9e : we suggest you get a fractional PRI rather than analog line cards. |
| 16:46.57 | ghenry | anyone use http://www.jivesoftware.org/asterisk-im/ or other jabber things with *? |
| 16:47.02 | hmmhesays | or an 8 port fxo gateway |
| 16:47.02 | coppice | Blafasel: I think if you've no clue what to do its time to give up. that thing is only for people prepared to do deep debugging. :-) |
| 16:47.14 | [TK]D-Fender | sp0n9e : Or get an A200 w/ 8 ports and do it in one card. |
| 16:47.30 | sp0n9e | also, do i want hardware echo cancellation? |
| 16:47.44 | hmmhesays | who makes that? |
| 16:47.53 | Blafasel | coppice: I'd love to give up, go home, have a beer. My boss might have a different opinion, though ;) |
| 16:48.21 | *** join/#asterisk oej (n=oej@65.197.203.67) |
| 16:48.26 | eKo1 | Blafasel: contact the chan_ss7 devs about your problem. |
| 16:48.31 | eKo1 | Maybe they can help. |
| 16:48.36 | smackus | is there an SIP analog phone adapter? I have seen iax, but no sip |
| 16:49.15 | [TK]D-Fender | sp0n9e : More like "Do you want echo" :) |
| 16:49.39 | Blafasel | eKo1: I probably know the answer.. Something along the lines of "Use the source, Luke" |
| 16:49.42 | sp0n9e | lol |
| 16:49.55 | eKo1 | smackus: SIP ATAs are everywhere. |
| 16:50.25 | sp0n9e | [TK]D-Fender: have any links about how these gateways work? |
| 16:50.25 | coppice | eKo1: I think he's trying to use something that is a work in progress. |
| 16:50.36 | eKo1 | This fromuser business is funny. |
| 16:50.38 | sp0n9e | does it go from FXO to SIP? |
| 16:50.52 | *** part/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 16:51.11 | smackus | was just looking in the wiki about ATA and only saw the iax ones. can anyone recommend a good SIP adapter? |
| 16:51.31 | eKo1 | spa1001 |
| 16:51.41 | coppice | good and ATA don't go naturally together |
| 16:51.49 | csplinter | Ok, Iv'e decided that useing the IAX protocol is the best solution for us, now Iv'e seen the IAXY box, which is nice, now I'm trying to find IAX compatible phones, but I only see a handful, Is there a list somewere? |
| 16:52.02 | smackus | coppice: why is that? |
| 16:52.19 | eKo1 | csplinter: that is because there are only a handful of them |
| 16:52.33 | Sonderblade | [TK]D-Fender: and when exactly am i supposed to set that variable? |
| 16:52.49 | coppice | they range from shoddy to total crap. they lie endlessly about the specs. they have weird limitations, and they are buggy as hell |
| 16:53.06 | csplinter | eKo1: yea thats what I was afraid of, thats why i would like to see a list, so i don't miss one of the few there are |
| 16:54.24 | watchy | * and zap work fine now that im using current |
| 16:54.48 | smackus | coppice: so they way I want to use this is I want one analog single line cordless phone plugged into my entire asterisk roll out, will I get what i need with something like the SPA-1001? |
| 16:55.05 | csplinter | what do you guys think of the iaxy box? |
| 16:55.08 | smackus | and will it work |
| 16:55.34 | eKo1 | csplinter: i had the original; it got really hot and eventually stopped working |
| 16:55.50 | eKo1 | smackus: yes |
| 16:55.55 | smackus | ok |
| 16:55.56 | [TK]D-Fender | sp0n9e : no, the A200 is a PCI cardlike the TDM400. You put it righ in your * server |
| 16:56.00 | smackus | thanks for the recommendation |
| 16:56.03 | csplinter | eKo1: yea i read they have some problems with heat |
| 16:56.04 | [TK]D-Fender | Sonderblade : sip.conf........ |
| 16:56.10 | csplinter | does anyone have one of the new ones? |
| 16:56.20 | sp0n9e | [TK]D-Fender: i was looking at the gateway |
| 16:56.30 | smackus | i see the price is about $60 give or take, is that a good price to pay for the SPA-1001? |
| 16:56.36 | [TK]D-Fender | sp0n9e : Gateways are pricy... not much of a point in most cases. |
| 16:56.44 | [TK]D-Fender | sp0n9e : Which one exactly? |
| 16:56.54 | sp0n9e | claro and some others |
| 16:57.06 | sp0n9e | i just want the easiest way to get 8 phone lines into asterisk |
| 16:57.06 | [TK]D-Fender | smackus : SPA-1001 is a waste. Splurge a big and get the SPA-2002 for 10$ more. |
| 16:57.09 | *** join/#asterisk pengyong (n=lala@218.93.158.79) |
| 16:57.33 | csplinter | Does anyone know if the newer IAXY boxes still have heat problems? |
| 16:57.48 | Sonderblade | [TK]D-Fender: i wonder if im really the only one who has nfc what you are saying |
| 16:58.01 | [TK]D-Fender | sp0n9e : Never heard of them, which is a bad sign. If you REALLY want a gateway go with something more known like AudioCodes or Mediatrix. Otherwise I'd suggest an A200 for 8 lines and get the EC module with it |
| 16:58.09 | pengyong | any one can help me to play music ring to caller before answering the call |
| 16:58.23 | [TK]D-Fender | Sonderblade : No there are all sorts of people here with nfc. No need to feel left out ;) |
| 16:58.37 | [TK]D-Fender | Sonderblade : Go look up "config sip.conf" on the wiki and give it a GOOD read. |
| 16:58.46 | bkidney | <PROTECTED> |
| 16:58.47 | *** join/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 16:58.52 | hmmhesays | [TK]D-Fender: I use quintum 2nd generation hardware on all my external fxo gateway applications |
| 16:59.02 | [TK]D-Fender | pengyong : "show application dial" |
| 16:59.29 | [TK]D-Fender | sp0n9e : Oh yeas, asn hmmhesays reminds me, Quintum is pretty respectable as well. |
| 17:00.01 | eKo1 | not the analog quintums |
| 17:00.22 | hmmhesays | eKo1: 2nd generation analog quintums are all I use for external fxo apps |
| 17:00.36 | hmmhesays | and the perform quite well |
| 17:00.43 | eKo1 | oh good because the 1st gen. really sucked donkey balls |
| 17:00.53 | eKo1 | and since then, i haven't touched quintums |
| 17:00.57 | hmmhesays | eKo1: agreed, they had some issues |
| 17:01.08 | hmmhesays | 2nd gen is completely different hardware running a completely different OS |
| 17:01.56 | eKo1 | How is the SIP support on those? |
| 17:02.38 | hmmhesays | when they first came out? It honestly wasn't that great, a few firmware revisions later, and it is pretty good |
| 17:03.16 | sp0n9e | [TK]D-Fender: so i'd need a sangoma A200 with a daughter card? |
| 17:03.19 | hmmhesays | one thing I like is they allow me to add voip services to pbx's without the endusers having to change dialplans |
| 17:03.53 | RoyK | ~rtfm |
| 17:03.55 | jbot | hmm... rtfm is Read The F*cking Manual (TM) Interwebs-speak for 'Repeat the first message'. It is used when the message did not transfer over the Interwebs properly. If someone tells you to RTFM, be patient with them, and copy-and-paste your original message several times. (http://uncyclopedia.org/wiki/RTFM) |
| 17:03.55 | eKo1 | hmmhesays: example? |
| 17:04.14 | sp0n9e | lol |
| 17:04.30 | [TK]D-Fender | sp0n9e : Correct |
| 17:04.45 | [TK]D-Fender | sp0n9e : Takes up 1 PCI resource and 2 backplanes |
| 17:04.54 | sp0n9e | right |
| 17:04.55 | *** join/#asterisk trelane` (n=trelane@unaffiliated/trelane) |
| 17:05.05 | hmmhesays | eko1: they have fxs/fxo gateways. When you put them on the trunk side of the pbx they can pick up the calls and look for a sip route. if one is not found they just pass the call straight through to the pots line |
| 17:05.35 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 17:05.38 | [TK]D-Fender | sp0n9e : I use them exclusively for analog line usage in my consulting. |
| 17:05.40 | hmmhesays | so you can do some least cost routing with them, or have other sip endpoints out there |
| 17:05.46 | tzanger | wow... $140 ea for 50 ip430s. If I had experience with the ip430 I'd look at it. ip501 only here |
| 17:06.01 | sp0n9e | [TK]D-Fender: this beats the hell out of a traditional PBX |
| 17:06.04 | hmmhesays | eKo1: it is really quite nice |
| 17:06.18 | [TK]D-Fender | tzanger : I own one. Great price (tell me where!) and its a great little phone. Zippy firmware and boots twice as fast. |
| 17:06.26 | tzanger | on -biz |
| 17:06.39 | [TK]D-Fender | tzanger : OMGZ link meh! |
| 17:06.42 | hmmhesays | eKo1: if you ever have an install where you think you might need something like that, drop me a line |
| 17:06.54 | [TK]D-Fender | sp0n9e : My opinion as well. |
| 17:07.09 | eKo1 | hmmhesays: thanks! |
| 17:07.28 | hohum | anyone use NexTone products? |
| 17:09.38 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
| 17:09.52 | pengyong | [TK]D-Fender, i enable 'm' option, and the caller on the PRI side not here music and ring |
| 17:10.02 | pengyong | i have post a mail to user list |
| 17:10.22 | [TK]D-Fender | pengyong : pastebin CLI output of a call using that method so we can see whats going on. |
| 17:10.23 | [TK]D-Fender | ~pb |
| 17:10.24 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 17:11.12 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 17:11.22 | hmmhesays | ~hmmhesays |
| 17:11.23 | jbot | somebody said hmmhesays was not really here... |
| 17:11.25 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:11.25 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:11.57 | pengyong | [TK]D-Fender>, pls check: http://channels.debian.net/paste/3302 |
| 17:12.20 | hmmhesays | ~hmmhesays |
| 17:12.21 | jbot | i heard hmmhesays is not really here... |
| 17:12.29 | hmmhesays | bah that one sucks |
| 17:13.01 | [TK]D-Fender | ~[TK]D-Fender |
| 17:13.03 | jbot | [[tk]d-fender] rockin' the casbah !!! |
| 17:13.08 | [TK]D-Fender | b00y4h |
| 17:13.15 | hmmhesays | who sets those anyway? |
| 17:13.23 | mocker | ~mocker |
| 17:14.39 | [TK]D-Fender | pengyong : Maybe your MoH setup isn't right... says its trying... |
| 17:14.48 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 17:15.26 | Slugs_ | is the digital receptionist used to create an IVR type application |
| 17:15.28 | Slugs_ | ? |
| 17:15.55 | pengyong | [TK]D-Fender, if i answer the call, and hold the call, the caller will here MOH |
| 17:16.07 | mut | anyone know the string for spa's to follow daylight savings time? |
| 17:17.39 | pengyong | is PRI support Music ring before call is answered? |
| 17:17.44 | *** join/#asterisk RoyK (n=roy@ti211310a080-0478.bb.online.no) |
| 17:18.24 | [TK]D-Fender | pengyong : Don't think so.. you should do an "Answer" first I believe. |
| 17:18.44 | [TK]D-Fender | pengyong : Thats "early media" and I'm unsure on its impact on your scenario. |
| 17:19.04 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 17:19.05 | syzygyBSD | in a agi script, if I want to tell it to goto, is the command "exec goto context exten priority" or how do I seperate the options? comma, pipe, space? |
| 17:19.42 | pengyong | if i answered the call, the caller should pay although the call is not connected |
| 17:20.28 | [TK]D-Fender | pengyong : well it IS conencted, just not 2-way... like I said this is geting a little out of my level of expertise. |
| 17:22.40 | file | pengyong: I answered your post on the list already :) |
| 17:24.29 | hmmhesays | good lord yum is slow |
| 17:24.59 | file | hmmhesays: horribly so |
| 17:25.06 | file | I <3 Debian |
| 17:26.01 | hmmhesays | file: yeah i like debian too, just not for desktop os |
| 17:26.06 | hmmhesays | my workstation is fc5 |
| 17:26.47 | momelod | question: if i have a call comming in from a queue, is there a way to show which queue the call is from on the phone using cid? |
| 17:26.48 | RoyK | fedora == evil |
| 17:26.48 | hmmhesays | i just want to watch an episode of stargate on my lunch hour damnit |
| 17:27.16 | hmmhesays | RoyK: fedora is a great "close to windows" OS |
| 17:27.29 | [TK]D-Fender | momelod : Yup. |
| 17:28.20 | hmmhesays | i don't have to futz with my mouse, my multimedia keyboard works |
| 17:28.20 | momelod | :) can u give me a keyword that i can search ? |
| 17:29.03 | [TK]D-Fender | momelod : "show function CALLERID" |
| 17:29.09 | momelod | ty |
| 17:30.04 | salviadud | anybody watch adult swim? |
| 17:30.09 | momelod | wow, that show function is nice too! didnt know about that :D |
| 17:30.12 | hmmhesays | haha yeah |
| 17:30.21 | hmmhesays | buy peter wheel |
| 17:30.26 | Corydon-w | "show functions" |
| 17:30.54 | Sonderblade | [TK]D-Fender: ok i've read everything in config sip.conf on the wiki as you said, and there is nothing in it about call transfers |
| 17:30.59 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 17:31.02 | salviadud | hmmhesays, where do you download your stargate episodes? |
| 17:31.13 | hmmhesays | salviadud: usenet |
| 17:31.34 | salviadud | hmmhesays, jesus, they still exist |
| 17:31.55 | hmmhesays | people re-up them |
| 17:31.58 | [TK]D-Fender | Sonderblade : I never said what you were looking for was about call transfers so much as a way to ID the callback contact.... "SetVar". Read closer |
| 17:32.03 | hmmhesays | and I have 70 days retention on my giganews account |
| 17:32.27 | salviadud | hmmhesays, well that's pretty nice |
| 17:32.59 | hohum | hey! hey! |
| 17:33.05 | hohum | this is #Asterisk, not #Piracy |
| 17:33.25 | hohum | please discuss your criminal activities elsewhere |
| 17:33.34 | hohum | :) |
| 17:33.37 | Sonderblade | [TK]D-Fender: i think the callback extension number is stored in the BLINDTRANSFER variable |
| 17:34.39 | hmmhesays | hohum: stfu |
| 17:34.55 | hmmhesays | downloading episodes of stargate is not piracy, nor criminal if you own the original content |
| 17:35.28 | hohum | if you own the original why would you need to download it off of a usenet news group? why not just rip it? |
| 17:36.19 | hohum | distributing the film is ILLEGAL, whether or not you own it, you broke the law because you got it off of someone who was distributing it illegaly |
| 17:36.24 | [TK]D-Fender | hmmhesays : Questionable, but also ineffecient... just rip them yourself you lazy ass! |
| 17:36.50 | *** join/#asterisk digime (n=digime@70.230.196.197) |
| 17:36.56 | [TK]D-Fender | hohum : Thats the darker side of grey..... hmmhesays would be legit, but not the distributor. |
| 17:37.00 | digime | anyone know of an answering service/call center with IAX/SIP support? |
| 17:37.06 | hmmhesays | hohum: you are mistaken |
| 17:37.10 | digime | ideally usa/canada |
| 17:37.15 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:37.25 | hmmhesays | 12:36:23) hohum: distributing the film is ILLEGAL, whether or not you own it, you broke the law because you got it off of someone who was distributing it illegaly |
| 17:37.47 | hmmhesays | you did NOT break the law by getting it off of someone who is illegally distributing it |
| 17:38.17 | digime | anyone have experience linking request tracker with asterisk? |
| 17:38.23 | hmmhesays | if you own the original content you are entitled to copy or obtain a copy of that content for personal use |
| 17:38.29 | hohum | regardless, the conversation doesn't belong here |
| 17:39.07 | salviadud | hohum, well excuuuuuse me princess, you're the one mentioning piracy, not us |
| 17:39.08 | [TK]D-Fender | hmmhesays : Actually I think that'd be "receiving stolen merchandise" so actually yeah, not legit even though you posess RIGHTS to get have it. Its the means that counts :) Anyways, lets just end it at that shall we? |
| 17:39.25 | hmmhesays | if however in the process of obtaining that content, you inadvertently distribute that content ex: (p2p) applications. Then you are breaking the law |
| 17:39.31 | [TK]D-Fender | salviadud : Go back to mushroom-land! |
| 17:39.40 | *** part/#asterisk digime (n=digime@70.230.196.197) |
| 17:40.07 | hmmhesays | [TK]D-Fender: it bothers me when people take what the riaa and mpaa say without a grain of salt |
| 17:40.37 | hohum | it isn't what the RIAA and MPAA say, it is copyright law |
| 17:40.53 | hohum | I certainly am not on the MPAA's side |
| 17:40.55 | hmmhesays | maybe you should read the law a little bit closer |
| 17:41.07 | Blafasel | Now, please repeat after me: IANAL. |
| 17:41.14 | hmmhesays | there is a lot of gray area in there |
| 17:41.24 | [TK]D-Fender | hmmhesays : I take it with salt, msg, chinese 5-spices and THEN follow through by the requisite purge in the men's room :) |
| 17:41.25 | salviadud | IANAL |
| 17:41.29 | Blafasel | hrhr ;) |
| 17:41.35 | hmmhesays | [TK]D-Fender: sounds about right |
| 17:41.37 | hmmhesays | anyhoo |
| 17:41.37 | [TK]D-Fender | Blafasel : Yes, you are quite ANAL ;) |
| 17:41.49 | hmmhesays | the other thing that bothers me is when people whine about getting off topic |
| 17:41.52 | hmmhesays | this isn't paid support |
| 17:41.57 | *** join/#asterisk MikeJ (n=vircuser@204.250.115.212) |
| 17:42.03 | Blafasel | My sexual preference at least doesn't involve pirates. Well - costumes maybe, but.. |
| 17:42.13 | hmmhesays | fish net stockings |
| 17:42.14 | hohum | no, but there are plenty of other "general" chat channels around, go join one |
| 17:42.43 | hohum | then you justify to other people how you're not doing anything wrong until you're blue in the face |
| 17:42.51 | hohum | I don't care |
| 17:42.53 | salviadud | hohum, are we really stopping you from getting to know asterisk a little better? |
| 17:43.12 | salviadud | we'll go off-topic for a while, we're human |
| 17:43.22 | hmmhesays | people who come in here and are seriously complaining about being off topic need to step back, take a breath and relax |
| 17:43.40 | salviadud | hmmhesays, right with you buddy |
| 17:43.53 | hohum | salviadud: there's a difference between going off topic and discussing questionable subjects |
| 17:44.11 | hmmhesays | "questionable subjects" |
| 17:44.18 | hmmhesays | who are you to say what is questionable? |
| 17:44.36 | hmmhesays | maybe if we were talking about nambla or something like that |
| 17:44.43 | salviadud | or abortion |
| 17:44.46 | hohum | you said yourself that it's a grey area |
| 17:45.04 | hohum | I'm using your own in-context description no |
| 17:45.05 | hohum | now |
| 17:45.05 | hmmhesays | i said copyright law is a grey area |
| 17:45.45 | Bobcat_1966 | Hello All, does anybody have fax detection working on Asterisk Trunk |
| 17:46.15 | hmmhesays | asterisk trunk? |
| 17:46.24 | Bobcat_1966 | svn trunk |
| 17:46.24 | salviadud | like a zap trunk you mean? |
| 17:46.25 | Bobcat_1966 | sorry |
| 17:46.44 | salviadud | svn is not a channel |
| 17:47.01 | hmmhesays | ~svn |
| 17:47.02 | jbot | hmm... subversion is version control software. see http://subversion.tigris.org/ it aims to be a better CVS than CVS. |
| 17:47.12 | hmmhesays | but it sure is handy |
| 17:47.13 | Bobcat_1966 | I know but I have just uprgaded to Asteriks SVN Trunk and cannot get fax to work |
| 17:47.29 | file | Bobcat_1966: be more specific |
| 17:47.42 | hmmhesays | i'm going to guess he's using a@h, or trixbox |
| 17:48.22 | hmmhesays | and broke their fax detecting dp |
| 17:48.57 | Bobcat_1966 | Ok. I built my asterisk box from scratch using asterisk 1.2 and never installed NVFax, I just upgraded to the Newest SVN trunk and tryied to install NVFAX but get a compile error when I try. |
| 17:49.03 | Bobcat_1966 | app_nv_backgrounddetect.c:328: warning: no previous prototype for 'load_module' |
| 17:49.04 | Bobcat_1966 | app_nv_backgrounddetect.c:333: warning: no previous prototype for 'description' |
| 17:49.06 | Bobcat_1966 | app_nv_backgrounddetect.c:338: warning: no previous prototype for 'usecount' |
| 17:49.07 | Bobcat_1966 | app_nv_backgrounddetect.c: In function `usecount': |
| 17:49.09 | Bobcat_1966 | app_nv_backgrounddetect.c:340: warning: implicit declaration of function `STANDARD_USECOUNT' |
| 17:49.10 | Bobcat_1966 | app_nv_backgrounddetect.c: At top level: |
| 17:49.12 | Bobcat_1966 | app_nv_backgrounddetect.c:345: warning: function declaration isn't a prototype |
| 17:49.13 | Bobcat_1966 | make[1]: *** [app_nv_backgrounddetect.o] Error 1 |
| 17:49.15 | Bobcat_1966 | make[1]: Leaving directory `/usr/src/asterisk/apps' |
| 17:49.16 | Bobcat_1966 | make: *** [apps] Error 2 |
| 17:49.27 | file | you can't use that out of tree module with trunk |
| 17:49.58 | Bobcat_1966 | Im guessing NVFax is not a compatible app but just wanted to see if anybody know what I might be doing wrong |
| 17:50.15 | Bobcat_1966 | ahh |
| 17:50.15 | file | it hasn't been updated to the new loader changes |
| 17:50.46 | Bobcat_1966 | is there a new app_nvfax.so that you know of? |
| 17:50.52 | file | nope |
| 17:51.17 | salviadud | faxing is not as hot as e-mailing you know... |
| 17:51.17 | Bobcat_1966 | so does fax not work under Asterisk trunk? is there another solution? |
| 17:51.38 | hmmhesays | bobact pastebin man |
| 17:52.03 | file | Bobcat_1966: fax is very generic, that specific application will not work with trunk unless you update it to the new loader changes |
| 17:52.19 | Bobcat_1966 | Ok thanks |
| 17:52.43 | file | you can look at existing stuff in the official tree to see what changed, and muck with the module to get it to compile and work |
| 17:52.44 | file | if you want. |
| 17:53.03 | Bobcat_1966 | appriciate it file. |
| 17:53.08 | *** join/#asterisk RoyK (n=roy@ti211310a080-0478.bb.online.no) |
| 17:53.23 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
| 17:56.02 | philippel | hi all - I"m having a strange problem I'm wondering if anyone has seen? Asterisk box A connected with Pri to Box B on a quad span card, Box B simply bridges the call out another PRI span to the local telephone company. When the far end is reached into an IVR (non-asterisk and asterisk IVRs) there are random double dtmf digits seen even though Box A and Box B report sending the correct digits on the CLI with dtmf logging enabled |
| 17:56.43 | salviadud | seen or heard? |
| 17:57.12 | philippel | either - any helpful data on what might help to address:) |
| 17:57.33 | *** join/#asterisk anthm (n=anthm@000-410-949.area4.spcsdns.net) |
| 17:57.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:57.41 | salviadud | if i dial 1, it outputs 11? |
| 17:58.28 | philippel | example, if you reach the far end and dial 058514 it might repeat back 055855114 |
| 17:58.48 | philippel | but cli on both systems all report sending only 058514 |
| 18:00.33 | *** join/#asterisk mishkiz (n=janusmis@zeus.corsidian.com.br) |
| 18:01.20 | *** join/#asterisk ginvent (n=joseph@adsl-63-199-241-141.dsl.sndg02.pacbell.net) |
| 18:01.56 | ginvent | I have a polycom ip600 going to my asterisk box with a zap channel. The analog channel has call waiting, how to get it to switch over? |
| 18:02.24 | CunningPike | ginvent: Come again? |
| 18:02.28 | ginvent | Zap is configured with callwaiting = yes. |
| 18:02.44 | ginvent | When I am on a call... I can't switch over when I hear the call waiting beep |
| 18:03.17 | mishkiz | hello all...im having a little problem here with nat, and I already read a lot of documentation on web, but I still cant understand how to get asterisk and "external (on internet)" sip clients connected...could anybody give me a tip ? |
| 18:03.17 | ginvent | How do I command asterisk to "flash" the zap channel? |
| 18:03.47 | ginvent | I dial 9 to get out (the analog channel), but when I get a call waiting beep, I can't figure out how to switch to that call. |
| 18:04.21 | CunningPike | ginvent: Put your current call on hold, and then you should be able to press the 'Answer' softkey to pick up the other call |
| 18:04.38 | CunningPike | ginvent: Use the up and down arrow keys to select the other call if necessary |
| 18:04.39 | ginvent | It's the same analog line though. |
| 18:04.47 | ginvent | It doesn't come in as a seperate call. |
| 18:04.51 | [TK]D-Fender | CunningPike : Nope. He's talking about ANALOG call waiting.... |
| 18:05.05 | ginvent | [TK]D-Fender, exactly. |
| 18:05.09 | ginvent | Can I even do this? |
| 18:05.17 | CunningPike | Oh - I thought everyone had a PRI ;) |
| 18:05.29 | [TK]D-Fender | ginvent : There is an app for it but I'm unsure of its usability. maybe |
| 18:07.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.165) |
| 18:08.06 | DarKnesS_WolF | can someone point me to the application that makes each user has his own busy / unavilable message on the voicemail ? |
| 18:09.31 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 18:09.36 | nortex | On queues, does ringall skip agents who are on the phone? I set up penalties on a queue in 1.2.10.1 and the members in penalty 1 were not skipped even while on the phone. |
| 18:09.51 | hohum | CunningPike: My company operates a facilities-based CLEC and we have absolutly no TDM (no pris) on our network at all |
| 18:10.10 | *** part/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 18:10.17 | [TK]D-Fender | DarKnesS_WolF : There is no app for it, its a parm of VoiceMail |
| 18:10.39 | CunningPike | hohum: What are you using? |
| 18:10.51 | DarKnesS_WolF | [TK]D-Fender: ok thx :) i'll dig it up :) |
| 18:10.53 | hohum | SIP and H323 only |
| 18:11.06 | DarKnesS_WolF | [TK]D-Fender: u and x86 always point the light to me :P |
| 18:11.15 | *** join/#asterisk jhamlyn (i=jhamlyn@203.33.186.65) |
| 18:11.26 | hmmhesays | what company? |
| 18:11.33 | CunningPike | hohum: For PSTN termination, I mean |
| 18:11.35 | [TK]D-Fender | DarKnesS_WolF : I need to ramp my laser up a few more giga-watts ;) |
| 18:11.36 | jhamlyn | :-) |
| 18:11.37 | hohum | http://www.interceltelecoms.com |
| 18:11.41 | *** join/#asterisk burnproof (n=jsharryp@210.213.198.59) |
| 18:11.42 | hohum | cunning: SIP and H323... |
| 18:12.29 | DarKnesS_WolF | [TK]D-Fender: oh thats soo evil :P |
| 18:12.29 | hohum | cunningpike: We have interconnects with Global Crossing, Level 3, Primus, Qwest, iBasis, etc, all IP-based |
| 18:12.39 | CunningPike | hohum: Ah, I see |
| 18:13.33 | websae | yuck...IP - BASED |
| 18:14.02 | websae | how's that working out for you? |
| 18:15.48 | file | IP based and prefix based authentication... |
| 18:16.02 | *** join/#asterisk Lenusik (n=ln@office-181.telengy.net) |
| 18:16.41 | *** join/#asterisk apardo (n=apardo@87.217.146.142) |
| 18:17.08 | ginvent | So how do I get this to work? hmmm... I need to flash the analog line while I am on it. |
| 18:17.16 | hohum | it's working quite well, we send PINs |
| 18:17.23 | hohum | so its an IP/PIN combo |
| 18:17.24 | ginvent | I wonder if there is a zap channel command for that I can actuate from asterisk. |
| 18:17.30 | ginvent | or write a script? |
| 18:17.57 | hohum | we don't do routing over the public internet (although we have in the past and you can't tell the difference in most cases) |
| 18:18.12 | hohum | we have peering agreements in place with all of our carriers |
| 18:18.48 | Slugs_ | Once I setup and extension and then associate a digital receptionist with that extension, I then should be able to dial that extension and here the recordings for the digital receptionist. Is that correct?> |
| 18:19.15 | websae | well peering is a better solution |
| 18:19.23 | mog | yes Slugs_ |
| 18:19.29 | DarKnesS_WolF | is there any SIP web applets that able to connect to asterisk ? |
| 18:19.42 | [TK]D-Fender | Slugs_ : please read the channel topic..... |
| 18:19.50 | hohum | websae: it's the best solution, because you control the network right up to your carrier |
| 18:19.52 | mog | any sip device would likely be able to connect to asterisk DarKnesS_WolF |
| 18:20.09 | hohum | websae: with a well thought-out QoS policy it is just as good as TDM |
| 18:20.14 | DarKnesS_WolF | mog: u have a point here :) i wann the applet :P hehe :) |
| 18:20.26 | DarKnesS_WolF | [TK]D-Fender: find it thx ;-) |
| 18:20.26 | CunningPike | [TK]D-Fender is like a shark with blood when it comes to AAH - he can sniff out 1 ppm ;) |
| 18:20.29 | mog | google sip java client |
| 18:20.33 | mog | your sure to find some |
| 18:20.45 | file | mog: Java rocks, doesn't it? |
| 18:21.07 | websae | sure with point to point circuits , it's about the same as TDM, depending on hops |
| 18:21.09 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 18:21.19 | [TK]D-Fender | file : in terms of SINKING, yes ;) |
| 18:21.21 | CunningPike | [TK]D-Fender: trixbox users are friends, not food |
| 18:21.22 | DarKnesS_WolF | mog: thx googling alread :) |
| 18:21.28 | DarKnesS_WolF | already * |
| 18:21.42 | [TK]D-Fender | CunningPike : You're close.... just hard to swallow ;) |
| 18:21.48 | mog | first result DarKnesS_WolF https://sip-communicator.dev.java.net/ |
| 18:21.48 | CunningPike | lol |
| 18:21.52 | mog | right what you wanted |
| 18:22.01 | jhamlyn | Does anyone have problem downloading ftp://ftp.digium.com/pub/zaptel/b410p/misdn-b410p.tar.gz - keep getting a timeout - ?? |
| 18:22.14 | DarKnesS_WolF | mog: seems i used the wrong term... i was googling sip web applet :) |
| 18:22.26 | mog | sip java client |
| 18:22.29 | mog | was what i googled |
| 18:22.52 | DarKnesS_WolF | thx |
| 18:23.34 | hohum | websae: we own our backbone, so a carrier is never more than 4 hops at the most, and QoS policies/MPLS tagging are in place end-to-end |
| 18:23.50 | websae | that's great then :) |
| 18:25.13 | CunningPike | jhamlyn: Works fine for me |
| 18:25.48 | jhamlyn | hmmm - tku --- no firewall here and cant get it started vian any of the local machines... |
| 18:28.16 | *** join/#asterisk japerry (n=falc0n@216.231.51.209) |
| 18:34.38 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 18:35.56 | dlynes_office | I guess if I want to check the status of line 1 on 6 different phones that I would need to write an AGI script? |
| 18:36.16 | salviadud | would anybody be interested in receiving a encrypted e-mail from me containing some wav files |
| 18:36.22 | dlynes_office | Basically, I want to ring line 1 on all phones, only if none of them have line 1 open, and same for line 2, 3, and 4 |
| 18:36.23 | salviadud | well, just 1 wav |
| 18:36.33 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:36.41 | dlynes_office | Good morning, justinu|laptop |
| 18:36.57 | justinu|laptop | hey dlynes |
| 18:37.09 | salviadud | yeah, bon jour dlynes |
| 18:37.14 | salviadud | ca va |
| 18:37.18 | dlynes_office | buenas dias, salviadud |
| 18:37.31 | [TK]D-Fender | dlynes_office : Nope, all dialplan.... |
| 18:37.52 | dlynes_office | [TK]D-Fender: so I would just use a whole crapload of gotoif's then? |
| 18:38.39 | dlynes_office | i.e. gotoif(ischanavail(sip/101)) ... |
| 18:39.22 | jhamlyn | Has anyone got the B410P working .. I am have a number of compile issues.. What is the best kernel to use 2.4 or 2.6 |
| 18:39.29 | mishkiz | hello all...im having a little problem here with nat, and I already read a lot of documentation on web, but I still cant understand how to get asterisk and "external (on internet)" sip clients connected...could anybody give me a tip ? |
| 18:39.30 | salviadud | laters |
| 18:39.36 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net) |
| 18:40.08 | [TK]D-Fender | dlynes_office : that'd work. Maybe if you could describe the line layout a bit more we could narrow it down. |
| 18:41.48 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 18:41.53 | dlynes_office | [TK]D-Fender: 1 analog line, 4 sip lines, ring on all six phones during the day, autoattendant after hours, first call will come in on analog line, other calls will be trunked in on iax2, I want each call coming in on iax2 trunk showing up as a separate line appearance |
| 18:42.23 | dlynes_office | erm 4 iax2 lines i mean |
| 18:42.38 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 18:43.25 | mjmac | my, how things have changed in a few years... does anyone recommend a NA termination provider that doesn't require monthly fees or high minimum balances? i just want to test something, and even VPC has a $50 minimum now. i'm looking at the wiki, but wondering if anyone here has an opinion. |
| 18:44.05 | dlynes_office | mjmac: you could try www.five9snetwork.com |
| 18:44.09 | *** join/#asterisk _MDC_ (n=marcus@c-6efde255.06-72-6c6b7013.cust.bredbandsbolaget.se) |
| 18:44.29 | dlynes_office | mjmac: their policy is on their website as a downloadable file |
| 18:44.52 | _MDC_ | I've got trouble using the characters åäö from a CURL command in a GotoIf statement, any idea? |
| 18:48.43 | mjmac | jeez... i remember when this list had about 10 entries. now everyone and their brother-in-law is jumping on the VoIP bandwagon |
| 18:49.11 | antiPosix | what is the cheapest way to bridge a PBX network and PSTN |
| 18:49.17 | dlynes_office | mjmac: what list? www.calltermination.com? |
| 18:49.19 | antiPosix | in the form of a PCI card |
| 18:49.26 | mjmac | http://www.voip-info.org/wiki/index.php?page=VOIP%20Service%20Providers%20Business#NorthAmerica |
| 18:49.29 | mocker | When I do a sip reload it seems that people can't receive calls until they relaunch their SIP client. Is there an easy way to avoid that? |
| 18:49.32 | dlynes_office | ah |
| 18:50.06 | dlynes_office | mocker: must be an issue on your configuration |
| 18:50.16 | dlynes_office | mocker: I've never come across that problem |
| 18:50.27 | mocker | dlynes_office: I'm thinking it has to do with registration of the sip devices. |
| 18:50.37 | dlynes_office | mocker: when you do a sip reload, have you changed your clock? |
| 18:50.57 | mocker | dlynes_office: No |
| 18:51.22 | dlynes_office | mocker: after you do a sip reload, and you do a sip show peers, do you see status unreachable for any of your peers? |
| 18:52.07 | mocker | dlynes_office: Not sure. |
| 18:52.19 | mocker | Right now they are all "Unmonitored" |
| 18:52.26 | dlynes_office | ah |
| 18:52.46 | dlynes_office | are your sip devices all behind a nat? |
| 18:52.58 | dlynes_office | or are they on the same subnet as your asterisk box? |
| 18:53.26 | mocker | All behind nat. |
| 18:53.31 | dlynes_office | mocker: ok, make sure you have canreinvite=no, and nat=yes |
| 18:53.31 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 18:53.38 | [TK]D-Fender | dlynes_office :Why each as a seperate line appearance? Presence lighting on lines to show occupancy? |
| 18:53.40 | dlynes_office | mocker: or nat=route, depending on which works for you |
| 18:53.52 | hmmhesays | people always insist on paying for crap and I can never figure out why |
| 18:54.05 | [TK]D-Fender | antiPosix : X100P |
| 18:54.27 | [TK]D-Fender | mocker : What you need is "qualify=yes" |
| 18:54.31 | dlynes_office | mocker: also try qualify=300 |
| 18:54.43 | dlynes_office | mocker: qualify=300 works on every firewall I've tried |
| 18:54.56 | dlynes_office | mocker: qualify=yes (qualify=2000) doesn't necessarily work with all of them |
| 18:55.00 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.88) |
| 18:55.11 | [TK]D-Fender | dlynes_office : 300ms is extremely optimistic and highly unneccesary except in rare cases |
| 18:55.24 | dlynes_office | [TK]D-Fender: yeah, but i find it works for all cases |
| 18:55.47 | dlynes_office | [TK]D-Fender: only time it didn't was when I had some wrt54g's with buggy firmware |
| 18:56.06 | jbalcomb | When I try to call my X-Lite phone at home from my office (GXP-2000) I get "Got SIP response 488 "Not Acceptable Here" back from 234.56.45.121". What should I be looking for? |
| 18:56.11 | CunningPike | http://www.polycom.com/products_services/0,1443,pw-34-14992-14993,00.html?trackID=14993&track=pwHome |
| 18:56.35 | [TK]D-Fender | dlynes_office : it can work but timeouts would royally suck, and you're increasing packet load quite a bit. Load up your hosts and you'll be chewing bandwidth for nothing. |
| 18:56.47 | *** join/#asterisk s0lid (n=jlq@202.71.179.140) |
| 18:57.07 | justinu|laptop | jbalcomb: codec incompatibility |
| 18:57.08 | CunningPike | jbalcomb: Could be a codec mismatch |
| 18:57.09 | dlynes_office | [TK]D-Fender: anyways, basically because it's ringing on all phones, if someone's on 'line 2' for an outgoing call i don't want to ring that 'line' because that person won't see the incoming call |
| 18:57.37 | *** join/#asterisk evisu (i=hIRC@bzq-88-155-80-250.red.bezeqint.net) |
| 18:57.58 | *** join/#asterisk s0lid (n=jlq@202.71.179.140) |
| 18:58.11 | dlynes_office | [TK]D-Fender: and i figure it's probably easier to code the dialplan to try the same 'line number' on all phones, than to try the first available 'line' on each phone |
| 18:58.15 | [TK]D-Fender | dlynes_office : Don't know why poeple still feel the need to associate line keys with lines.... |
| 18:58.42 | dlynes_office | [TK]D-Fender: because they're small offices and they're used to keysystems |
| 18:58.52 | [TK]D-Fender | dlynes_office : Sad little creatures! |
| 18:58.58 | eKo1 | lol |
| 18:59.15 | dlynes_office | [TK]D-Fender: this same office however, is using call parking |
| 18:59.18 | eKo1 | People have a hard time accepting change. But that's just because they're lazy. |
| 18:59.24 | justinu|laptop | or stupid |
| 18:59.39 | dlynes_office | justinu|laptop: i think yours is a more accurate description |
| 19:00.25 | dlynes_office | I tried getting them to use autoattendant, too, but they want the receptionist to screen calls during the day |
| 19:00.41 | jbalcomb | justinu|laptop CunningPike: http://pastebin.ca/108887 and I have G711 uLaw set as the only Enabled codec in X-Lite. |
| 19:01.16 | [TK]D-Fender | eKo1, justinu|laptop : Not mutually exclusive... typically stupid AND lazy :) |
| 19:01.21 | justinu|laptop | yeah |
| 19:01.33 | jbalcomb | Shouldn't there be some sort of more specific error if the phone and the server can not negotiate an agreeable codec? |
| 19:01.41 | justinu|laptop | pastebin.ca isn't coming up for me |
| 19:01.59 | jbalcomb | [TK]D-Fender: Is the internet broken in Canada? |
| 19:02.10 | dlynes_office | nope |
| 19:02.25 | dlynes_office | pastebin.ca works fine for me from home, but not from the office |
| 19:02.30 | jbalcomb | dlynes_office: haha.. I can't convince my bosses of the autoattendant either. |
| 19:02.40 | dlynes_office | It all depends on whether your provider supports ipv6 or not |
| 19:02.42 | justinu|laptop | got itnow |
| 19:02.50 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 19:02.56 | CunningPike | jbalcomb: We don't have the Internet in Canada |
| 19:02.58 | dlynes_office | if they don't, you'll have problems with pastebin.ca (it uses ipv6 and ipv4 dns entries) |
| 19:03.06 | justinu|laptop | jbalcomb: turn on sip debug, and paste the sip dialog |
| 19:03.24 | [TK]D-Fender | jbalcomb : No.. its just you :) |
| 19:03.53 | justinu|laptop | what does ipv6 have to do with it? |
| 19:04.17 | jbalcomb | hexadecimal is awesome. |
| 19:04.25 | dlynes_office | if your router doesn't understand ipv6, and the dns server returns an ipv6 address, how are you doing to deal with it? |
| 19:04.42 | jbalcomb | dlynes_office: i'd kick its ass. |
| 19:05.10 | dlynes_office | jean claude van damme style :) |
| 19:05.12 | justinu|laptop | well, your IP stack won't be asking for ipv6 DNS resolution unless you specifically configure it to do so |
| 19:05.19 | justinu|laptop | IPV6 records are AAAA in DNS, not A |
| 19:05.21 | jbalcomb | point at it and tell it to get out cause i'm an american and i don't need no stinking ipv6 crap |
| 19:05.24 | dlynes_office | ah |
| 19:05.25 | dlynes_office | ok |
| 19:05.40 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
| 19:06.21 | CunningPike | Does anyone know the default username and password for the web interface for Polycom 501? |
| 19:06.23 | jbalcomb | of course, i did just have to fight tooth and nail to get a new /20 from my provider. |
| 19:06.31 | justinu|laptop | CunningPike: Polycom/456 |
| 19:06.43 | jbalcomb | CunningPike: Polycom:456 |
| 19:06.44 | CunningPike | OK - thanks |
| 19:07.19 | jbalcomb | if i had known i was going to end up a computer guy i wouldn't have played so much oregon trail in typing class.. |
| 19:07.34 | dlynes_office | oregon trail? |
| 19:07.53 | SpaceBass | AWESOME GAME! |
| 19:07.57 | *** join/#asterisk ffang (n=ffang@porcuswine.cbnco.com) |
| 19:07.58 | jbalcomb | yeah, its the best. "Your party has died of starvation" |
| 19:08.04 | lunaphyte_ | don't forget number crunchers :) |
| 19:08.11 | dlynes_office | because you didn't type fast enough? |
| 19:08.21 | SpaceBass | Civilization almost caused me to fail out of HS and college |
| 19:08.35 | dlynes_office | yeah....civilization kicks ass |
| 19:08.41 | SpaceBass | and now with Civ 4 I'm about to lose my job |
| 19:08.46 | jbalcomb | dlynes_office: yeah, and cause i wouldn't repeat people if i could look at the screen while i type 20 wpm. |
| 19:08.47 | [TK]D-Fender | CunningPike : Step. Away. From. The. BROWSER! |
| 19:09.15 | justinu|laptop | is civ4 better than civ3? i was pretty disapointed with 3 |
| 19:09.32 | dlynes_office | civ 2 wasn't all that great either |
| 19:09.38 | dlynes_office | civ 1 was the best |
| 19:09.51 | jbalcomb | ok, fix my SIP 488 error? |
| 19:10.09 | CunningPike | [TK]D-Fender: Relax - we're just exploring it as a possible option for providing end users with some basic UI functionality for their phones - call forwarding etc. Just humoring my boss, realy |
| 19:10.14 | dlynes_office | jbalcomb: fix your codec selection? |
| 19:10.24 | justinu|laptop | (12:03:09) justinu|laptop: jbalcomb: turn on sip debug, and paste the sip dialog |
| 19:10.31 | CunningPike | jbalcomb: Sorry - I couldn't connect to pastebin.ca |
| 19:10.47 | eKo1 | Use another pastebin then. |
| 19:10.57 | dlynes_office | try ipv4.pastebin.ca |
| 19:11.00 | jbalcomb | justinu|laptop: i'm working on that.. |
| 19:11.01 | [TK]D-Fender | CunningPike : Not funny......forwarding is only done on the phone itself anyways, or in your dialplan. Nothing for the boss to play with in there. |
| 19:11.22 | CunningPike | [TK]D-Fender: I know - he needs to 'discover' that for himself |
| 19:11.50 | [TK]D-Fender | CunningPike : Poor schmuck |
| 19:11.50 | dlynes_office | ~pb |
| 19:11.52 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 19:12.26 | CunningPike | [TK]D-Fender: He's a pretty clued-in boss - he just needs 'show me' every now and again |
| 19:12.50 | dlynes_office | CunningPike: No he's not. You're talking about a municipal government, here :) |
| 19:12.54 | CunningPike | pastebins are dying off - pastebin.com, now pastebin.ca........ |
| 19:12.58 | CunningPike | dlynes_office: We |
| 19:13.02 | dlynes_office | CunningPike: nobody that works for them is clued in |
| 19:13.16 | CunningPike | dlynes_office: We're not all the schmucks that the media would have you believe........ |
| 19:14.02 | dlynes_office | CunningPike: btw...I think i've determined that my problem with fop was that I was trying to do something that it's not capable of doing |
| 19:14.26 | CunningPike | dlynes_office: Ah - that'll break stuff every time :) |
| 19:14.40 | dlynes_office | yeah, but it's not smart enough to tell me that |
| 19:14.50 | dlynes_office | I think what I might end up doing when I get enough time |
| 19:15.02 | dlynes_office | Is write a Swing Java applet to accomplish the same thing |
| 19:15.30 | dlynes_office | and opensource the java code |
| 19:15.39 | dlynes_office | Then it's a lot more portable |
| 19:15.45 | mog | heh opensource java code... |
| 19:15.54 | dlynes_office | fop doesn't work in firefox on linux 90% of the time |
| 19:16.34 | dlynes_office | mog: well, you could run a decompiler on the .class file and get the java code anyways :) |
| 19:16.51 | mog | yeah |
| 19:17.07 | dlynes_office | why's that? |
| 19:17.20 | mog | well all other oss software i can run on all my hw |
| 19:17.25 | mog | i cant run java on my mips linux box |
| 19:17.28 | mog | or my arm one |
| 19:17.32 | dlynes_office | why not? |
| 19:17.33 | mog | or even my ppc one |
| 19:17.43 | mog | no java implement ation for mips or arm |
| 19:17.49 | CunningPike | dlynes_office: Finish your vm rewrite first please ;) |
| 19:17.50 | dlynes_office | I'm sure there is |
| 19:17.52 | mog | or at least easily obtainable |
| 19:18.04 | dlynes_office | I've got Java on my Nokia cellphone |
| 19:18.12 | mog | yeah |
| 19:18.14 | dlynes_office | It runs on MIPS I think |
| 19:18.18 | [TK]D-Fender | I have Java in my MUG! |
| 19:18.24 | dlynes_office | erm wait |
| 19:18.27 | dlynes_office | nvm |
| 19:18.27 | mog | good stuff tk-fender |
| 19:18.29 | dlynes_office | ARM, actually |
| 19:19.32 | mog | but im not holding my breath |
| 19:20.01 | file | mog: that would be bad, we wouldn't want you to be unconscious :( |
| 19:20.05 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:20.21 | justinu|laptop | dlynes: bringing up java won't make you popular |
| 19:20.27 | eKo1 | lol |
| 19:24.36 | file | Sonus... Sonus... who had the Sonus |
| 19:26.14 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 19:27.10 | syzygyBSD | what is wrong with java? that is what I learned OOP on |
| 19:27.24 | justinu|laptop | nothing, imo |
| 19:27.35 | justinu|laptop | but a lot of ppl like to diss it |
| 19:27.58 | *** join/#asterisk wangster (n=wangster@static-64-201-170-129.ptr.terago.ca) |
| 19:28.08 | eKo1 | java is fine, but then again, so is cobol |
| 19:29.04 | trelane` | logo for the win! |
| 19:29.16 | [TK]D-Fender | CPM > all |
| 19:29.21 | trelane` | lies! |
| 19:29.40 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3n0n.cable.mindspring.com) |
| 19:30.00 | justinu|laptop | logo is pretty cool |
| 19:30.48 | jbalcomb | justinu|laptop: Here is my `sip debug peer 4999` http://pastebin.ca/108944 |
| 19:33.14 | *** join/#asterisk asteriskbaby (n=undercov@58.65.193.69) |
| 19:34.24 | Mattwj2005 | hey guys |
| 19:34.57 | justinu|laptop | jbalcomb: hmmm... not much help there... asterisk is only offering g711u and the xlite client says 488 right away... trying to think of what else it could be |
| 19:35.13 | justinu|laptop | xlite has it's own debug console... have you tried turning that on? |
| 19:35.23 | asteriskbaby | HI EVeyrone |
| 19:35.39 | asteriskbaby | i am a newbie at asterisk@home |
| 19:36.04 | asteriskbaby | just need a little bit help is someone is ready to.. |
| 19:36.22 | Zodiacal | anyone know if its posible to saydigits before a blind transfer? |
| 19:36.39 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:36.55 | Slugs_ | this might be a shot in the dark, but has anybody here successfully added a sip trunk to Inter-tel using asterisk? |
| 19:37.32 | dlynes_office | justinu|laptop: bringing up java? |
| 19:38.17 | dlynes_office | justinu|laptop: I don't care whether it makes me popular or not...it's infinitely more portable than Flash |
| 19:38.20 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 19:38.28 | justinu|laptop | you said something about a swing version of fop |
| 19:38.30 | dlynes_office | justinu|laptop: and trust me...I hate Java with a passion |
| 19:38.34 | Zodiacal | is there a way to play a sound file during a call with someone? |
| 19:38.39 | Wazb^ | hi |
| 19:38.46 | dlynes_office | justinu|laptop: it's just the ideal platform to do something like that in |
| 19:38.52 | justinu|laptop | so why do you hate it? |
| 19:38.59 | dlynes_office | justinu|laptop: soooooooooooooooo boring |
| 19:39.11 | dlynes_office | it's like watching paint dry |
| 19:39.27 | justinu|laptop | i feel that way about everything computer related lately |
| 19:39.41 | Wazb^ | can anyone tell me any utility which can convert gsm file into g729 file |
| 19:39.43 | *** join/#asterisk andymul (n=andymul@cpe-69-203-217-237.nyc.res.rr.com) |
| 19:39.51 | dlynes_office | Wazb^: man sox |
| 19:40.01 | justinu|laptop | i don't think sox can do it |
| 19:40.11 | dlynes_office | hrm |
| 19:40.14 | justinu|laptop | g729 codecs aren't free, and there's no opensource utility to do it |
| 19:40.14 | dlynes_office | thought it could |
| 19:40.21 | asteriskbaby | can anyone tell me how u password protect my outbound calls through zap/g0 |
| 19:40.35 | dlynes_office | ooops |
| 19:40.39 | dlynes_office | that was g711, not g729 |
| 19:40.56 | Wazb^ | ya i saw that |
| 19:40.57 | dlynes_office | asteriskbaby: Authenticate() |
| 19:41.11 | mog | or vmauthenicate for even more security |
| 19:41.24 | justinu|laptop | not sure, but if you buy the g729 licenses, you might get access to that utility, Wazb^ |
| 19:41.29 | asteriskbaby | i am using asterisk@home |
| 19:41.39 | justinu|laptop | ~trixbox |
| 19:41.40 | jbot | somebody said trixbox was NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 19:41.43 | dlynes_office | asteriskbaby: ----> /join #freepbx |
| 19:41.53 | asteriskbaby | i have joined it |
| 19:42.04 | asteriskbaby | but they says its not AMP room |
| 19:42.11 | dlynes_office | so why not ask A@H specific questions there? |
| 19:42.57 | dlynes_office | asteriskbaby: did you join #freepbx, or #trixbox? |
| 19:43.12 | asteriskbaby | i have joined #freebox |
| 19:43.25 | dlynes_office | Yeah...it's #freepbx, not #freebox |
| 19:44.07 | Wazb^ | actaully i bought 10 licenses of g729, My calls from DID coming using g729 and going out through G729 |
| 19:44.30 | Wazb^ | i am using Calling Card application with Asterisk |
| 19:45.12 | *** join/#asterisk adorah (n=Administ@84.94.208.224.cable.012.net.il) |
| 19:45.47 | mog | asterisk trunk has all files in g729,gsm, ulaw, and slin if you want to get one of those |
| 19:45.53 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 19:46.14 | asteriskbaby | thanks dlynes |
| 19:46.43 | Slugs_ | this might be a shot in the dark, but has anybody here successfully added a sip trunk to Inter-tel using asterisk? |
| 19:47.14 | Wazb^ | <mog> from where ? |
| 19:47.39 | mog | if you check it out it has options to download it |
| 19:47.44 | eKo1 | Wazb^: Which one are you using? |
| 19:47.49 | mog | i dont know where you can wget it from |
| 19:47.52 | mog | but thats all it does |
| 19:49.03 | adorah | dlynes Hi |
| 19:49.31 | dlynes_office | good afternoon, adorah |
| 19:49.59 | dlynes_office | erm evening |
| 19:50.10 | adorah | <dlynes_office>The problem I've told u regarding the hoarse line.. |
| 19:50.11 | mog | morning |
| 19:50.55 | adorah | <dlynes_office>I allowed u-law a-law in addition to gsm in the remote ip phone and it cloged the bandwidth.. |
| 19:51.17 | dlynes_office | cool |
| 19:51.24 | adorah | <dlynes_office>so I allowed only gsm and it sounds clean now.. |
| 19:51.29 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 19:51.36 | Mattwj2005 | I have just recently figured out enum...pretty cool stuff :) |
| 19:51.48 | dlynes_office | I take it the bandwidth isn't too good in Israel atm? |
| 19:52.21 | adorah | very bad and now I think I have another BW issue: <dlynes_office> |
| 19:52.24 | *** join/#asterisk kamber (n=undercov@58.65.193.69) |
| 19:52.31 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 19:52.35 | dlynes_office | bw? |
| 19:52.44 | adorah | bandwidth.. |
| 19:52.44 | dlynes_office | oh...nvm |
| 19:52.53 | adorah | I have 2 iax trunks to a provider..2day it went off-line twice for hours while the provider was up and running..any idea why? |
| 19:52.55 | rene- | hello |
| 19:53.07 | Wazb^ | <mog> i am using Asterisk 1.2.9.1 |
| 19:53.10 | Mattwj2005 | hey rene |
| 19:53.18 | dlynes_office | adorah: the route between your server and theirs probably went down |
| 19:53.22 | rene- | is it possible to list in the CLI or in Realtime the names of the queues that the system has registered?? |
| 19:53.27 | *** join/#asterisk DaveHope (n=Dave@internal.davehope.co.uk) |
| 19:53.29 | mog | okay one sec Wazb^ ill help ya ^_^ |
| 19:53.42 | Corydon-w | "show queues" |
| 19:53.45 | rene- | Mattw|2005: hey |
| 19:53.50 | rene- | Corydon-w: nothing shows |
| 19:53.52 | rene- | mmm |
| 19:54.03 | adorah | <dlynes_office>their server was on-line coz I called them from another machine and it was ok.. |
| 19:54.06 | rene- | my question is poorly worded. |
| 19:54.14 | mog | http://ftp.digium.com/pub/telephony/sounds/releases/ |
| 19:54.15 | rene- | i meant to list in the cli or in ami the realtime queues |
| 19:54.23 | mog | there has the sounds in many formats |
| 19:54.56 | Corydon-w | Nope |
| 19:55.19 | dlynes_office | adorah: did you try tracerouting their server? |
| 19:55.20 | Corydon-w | There isn't an interface in realtime to list stuff, only to query for particular records |
| 19:56.48 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 19:57.48 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 19:59.15 | Vorondil | hi all, quick question: i've done some google'ing and poked around voip-info but i can't find anywhere that tells me if asterisk supports sip/simple (instant messaging) or not. does anyone around know one way or the other? |
| 19:59.47 | file | it supports it if the message is sent with the same callid as an active call |
| 19:59.48 | file | otherwise no |
| 20:00.04 | Vorondil | ah, i see |
| 20:00.28 | *** join/#asterisk WeirdM (n=weirdm@udp079073uds.hawaiiantel.net) |
| 20:00.30 | Vorondil | so gaim's sip/simple protocol plugin can just connect to asterisk and use it like an im server. |
| 20:00.44 | Vorondil | (or any simple client, for that matter) |
| 20:00.53 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 20:01.14 | WeirdM | Has anyone here decrypted their vonage username and password and got Asterisk working with it? I have my username and password but I can't get it to register. Anyone have a sample sip.conf? |
| 20:01.20 | Shaun2222 | in the extenstions.conf can i include files? say i wanted to seperate a bunch of contexts into seperate riles? |
| 20:01.22 | Shaun2222 | files* |
| 20:01.23 | Wazb^ | thanks <mog> |
| 20:01.40 | *** join/#asterisk The_LightSide (n=lightsid@wbs-196-2-110-38.wbs.co.za) |
| 20:04.31 | WeirdM | Vorondil: You should look in to setting up a Jabber server |
| 20:04.56 | NDT | Shaun2222: yeah #include "blahblah.conf" |
| 20:06.14 | Wazb^ | i have a cisco which is forwarding did to asterisk via sip g729 , asterisk send call to same cisco for terminating carrier. |
| 20:06.34 | Vorondil | WeirdM: yeah, if i was serious about getting in-house im running, that's what i'd do. my boss just asked if he could send text messages from adium on his mac book to a phone in the office (all sip) |
| 20:06.45 | mog | yeah Vorondil we already have jabber support for that kind of stuff |
| 20:06.56 | mog | you can use ser to do simple |
| 20:06.59 | Wazb^ | when a call hits asterisk i can hear promt and when call goes to cisco again i can hear nothing , not even a ring |
| 20:07.03 | mog | and you can even relay it over jabber |
| 20:07.06 | mog | but its pretty ugly |
| 20:07.59 | Vorondil | eh, s'all good. just curious. |
| 20:08.41 | WeirdM | Has anyone here decrypted their vonage username and password and got Asterisk working with it? I have my username and password but I can't get it to register. Anyone have a sample sip.conf? |
| 20:08.59 | Vorondil | anyway, thanks file, WeirdM, and mog |
| 20:09.18 | mog | no problem |
| 20:10.39 | *** join/#asterisk bmg505 (n=leon@c1-140-16.rndf.isadsl.co.za) |
| 20:12.36 | Wazb^ | <mog> i am woring on scenario like Cisco--DID--> Asterisk (callingcard) ---> Cisco --terminate--> Carrier via SIP g729 |
| 20:12.46 | *** join/#asterisk gugaiz (n=gugaiz@host200.200.61.156.ifxnw.com.ar) |
| 20:12.48 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 20:12.58 | *** join/#asterisk s0lid (n=jlq@124.6.176.100) |
| 20:13.29 | Wazb^ | when call goes to cisco , i get no audio |
| 20:13.53 | WeirdM | Are any of the devices behind NAT? |
| 20:14.01 | mog | well for one which cisco??? and two several questions like that |
| 20:14.04 | Wazb^ | no |
| 20:14.52 | *** join/#asterisk burnproof (n=burnproo@host-222-126-79-242.dhcp.infocom.ph) |
| 20:14.53 | Wazb^ | i have only one cisco 3845 |
| 20:15.34 | mountainm2k | any Polycom experts? I changed the displayName option in my .cfg file, and rebooted the phone, but it won't take it... |
| 20:15.53 | Wazb^ | i get no audio on outgoing call |
| 20:16.19 | [TK]D-Fender | mountainm2k :No doubt because its being overridden in sip.conf |
| 20:16.30 | gugaiz | when the call start, in the macro environment. Can I get the ip address from the user that start the call? |
| 20:16.40 | nortex | On queues, does ringall skip agents who are on the phone? I set up penalties on a queue in 1.2.10.1 and the members in penalty 1 were not skipped even while on the phone. |
| 20:17.19 | *** join/#asterisk adorah (n=Administ@84.94.122.203.cable.012.net.il) |
| 20:17.26 | mountainm2k | [TK]D-Fender: Nope, grep doesn't find it there... Also it _does_ take on another phone -- but I set this one by hand (on the phone itself) -- clearly that was dump... Now I'm "formatting" it... |
| 20:17.46 | [TK]D-Fender | mountainm2k : You should never set it on the phone anyways.... |
| 20:18.00 | mountainm2k | [TK]D-Fender: Also while you're here, how to load new rings? IE what format, where they go, what file(s) to tell the phone about them in... |
| 20:18.38 | [TK]D-Fender | mountainm2k : ULAW, and sip.cfg in your provisioning folder |
| 20:18.39 | mountainm2k | [TK]D-Fender: Heh, yeah, well, I was trying to figure out what option it actually is... It's displayName="soemthing" but I didn't know that -- a busted find-and-replace hosed that part of all my phone-blah.cfg's. |
| 20:18.50 | *** join/#asterisk adorah (n=Administ@84.94.122.203.cable.012.net.il) |
| 20:19.01 | rene- | Corydon-w: so no real way to get information on a queue that was defined via realtime? |
| 20:19.01 | mountainm2k | ULAW even though the one in there ends in .wav? |
| 20:19.11 | adorah | Re Re Re |
| 20:19.24 | mountainm2k | mountainm2k tries to find that section in the manual... |
| 20:19.29 | gugaiz | or I need SER? |
| 20:19.38 | [TK]D-Fender | mountainm2k : DL another firmware and examine how it should have been formatted and realize with * inbetween everything that you shouldn't be using the phone to do it anyways :) |
| 20:19.43 | rene- | adorah? |
| 20:19.48 | [TK]D-Fender | mountainm2k : Correct. |
| 20:20.01 | adorah | <rene: yup |
| 20:20.22 | Corydon-w | rene-: only by querying the underlying database directly |
| 20:21.22 | mountainm2k | [TK]D-Fender: You saying I should be setting the name, etc in * instead of on the phone? |
| 20:21.23 | mountainm2k | ;thinks... |
| 20:22.29 | mountainm2k | OK, the format fixed the displayName, heh... |
| 20:24.36 | *** join/#asterisk gugaiz (n=gugaiz@host200.200.61.156.ifxnw.com.ar) |
| 20:26.16 | rene- | Corydon-w: I have restored to show queues to get an agent paused status |
| 20:26.21 | rene- | resorted |
| 20:26.28 | rene- | how can i get that now? |
| 20:27.56 | *** join/#asterisk gugaiz (n=gugaiz@host200.200.61.156.ifxnw.com.ar) |
| 20:28.40 | SpaceBass | anyone have suggestions for a good DID provider in Spain? |
| 20:28.50 | SpaceBass | leaning towards voxbone at this point |
| 20:28.51 | crochat | Hello ! |
| 20:29.07 | crochat | I have a problem with my dialplan |
| 20:29.15 | dlynes_office | ~suggestions |
| 20:29.16 | jbot | from memory, suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite ... |
| 20:29.51 | rene- | agent status does not show an agent paused status, only 'show queues' or Action: Queues, now my queues are realtime based, how to get this information? is it possible? |
| 20:30.56 | [TK]D-Fender | mountainm2k : An yes you should definately be setting that in sip.conf. |
| 20:31.24 | [TK]D-Fender | BBIAB |
| 20:31.28 | gugaiz | Can I get (with variable) the ip address of the client sip, in the macro context? |
| 20:31.44 | crochat | When I call my Asterisk (registered on a SIP provider) server from normal phone network, the extensions don't work at all... when I push some keys on the phone, nothing appears... until the timeout :-( |
| 20:32.07 | gugaiz | or I need SER? |
| 20:32.13 | justinu|laptop | crochat: verify dtmf mode is set correctly |
| 20:33.25 | gugaiz | I really need to get start and stop of the call, its jobs is make by SER, or Can I make with asterisk? |
| 20:34.38 | eKo1 | gugaiz: Yes you can get the IP of the SIP UA in your dialplan. |
| 20:35.02 | gugaiz | in the start or stop ?I need to get the ip address, the call's unique id, and the number dialed. |
| 20:35.07 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 20:35.21 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 20:35.23 | gugaiz | eKo1, which is the variable name? |
| 20:35.28 | hmmhesays | never fear, I am back |
| 20:35.53 | eKo1 | gugaiz: I don't know :P |
| 20:35.57 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:36.27 | gugaiz | eKo1, ups.. |
| 20:36.29 | eKo1 | But you can get all that info. with the special dialplan variables. |
| 20:38.29 | gugaiz | eKo1, you know where can I found documentation? |
| 20:38.39 | eKo1 | voip-info.org |
| 20:38.58 | eKo1 | and the docs in the source code |
| 20:39.11 | justinu|laptop | yeah, there's a doc describing nearly all the dialplan variables |
| 20:39.16 | justinu|laptop | in the distribution tarball |
| 20:39.37 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 20:40.32 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 20:40.44 | *** join/#asterisk lorinc (n=ang@caracas-4732.adsl.interware.hu) |
| 20:45.08 | dlynes_office | justinu|laptop: not to mention voip-info.org |
| 20:48.13 | crochat | juanjoc: Nothing works :-( |
| 20:49.23 | crochat | justinu|laptop: Nothing works :-( the codec used seems to be forced on gsm, so I cannot use inband, and neither rfc2833 or info work at all... |
| 20:49.55 | *** join/#asterisk MoutaPT (n=MoutaPT@81.193.129.246) |
| 20:50.41 | sp0n9e | okay, i've got asterisk up and running, but it's not listening for SIP connections, where should i look next? |
| 20:50.46 | MoutaPT | Hi, can any one help me, i make modprobe ztdummy and my Meetme works fine, then i reboot and lost meetme conferences, i also can see that is a issue with autoload ztdummy |
| 20:50.56 | *** join/#asterisk lele (n=lele@rivendell.ipv6.windmill.it) |
| 20:52.54 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 20:55.03 | rene- | Corydon-w: you say i needed to query the database directly, my main problem is to check weather an agent is paused or not.. the realtime queues and realtime queuemembers do not have a provision for that. where is this info stored? astdb? or what did you meant by checking the underlying DB |
| 20:56.32 | brookshire | MoutaPT: add ztdummy to /etc/modules.conf |
| 20:56.56 | gugaiz | eKo1, you know something about get the start and the stop, of the current call |
| 20:57.37 | MoutaPT | brookshire I'm with Centos |
| 20:57.39 | justinu|laptop | crochat: rfc2833 over gsm should be correct... perhaps your provider is misconfigured? |
| 20:57.45 | MoutaPT | no /etc/modules.conf |
| 20:58.12 | eKo1 | gugaiz: Yes, that is a CDR variable. |
| 20:58.27 | brookshire | oh well.. google! |
| 20:58.36 | MoutaPT | echo "modprobe ztdummy" >> /etc/rc.d/rc.local solved my problem |
| 20:58.41 | MoutaPT | thks any way |
| 20:59.08 | rene- | Corydon-w: so, i would use static queuemembers cuz i need agent login, and if i used dynamic realtime queuemembers the database structure does not have a paused field, what gives? |
| 20:59.12 | Corydon-w | rene-: if they don't show up in 'show queues', then they're not paused |
| 20:59.29 | brookshire | MoutaPT: my bad.. it's /etc/modules |
| 20:59.35 | brookshire | modules.conf is in asterisk, lol |
| 21:00.00 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:00.09 | rene- | Corydon-w: remember that show queues dont show realtime based queues? |
| 21:00.22 | Corydon-w | Actually, it does |
| 21:00.29 | Corydon-w | but only queues which are loaded |
| 21:00.53 | Slugs_ | Has anybody successfully intergrated Inter-Tel PBX with asterisk? |
| 21:01.39 | rene- | mysql realtime status show that my connection is fine... my extconfig is named accordingly and i have rows with the mandatory name column with valid values, i think i will enable mysql debug to look for what asterisk is trying to send |
| 21:01.41 | Slugs_ | I want to accomplish this via sip or t1 |
| 21:02.16 | rene- | so it should show them |
| 21:03.15 | Corydon-w | Only if they're loaded |
| 21:06.48 | ghenry | This looks sweet: http://www.jivesoftware.org/asterisk-im/arch.jsp |
| 21:08.21 | gugaiz | eKo1, but with CDR I a can't get the variables when the call connect, I need to wait that call finish |
| 21:08.46 | gugaiz | is possible that get cdr in the same moment that call connect? |
| 21:09.27 | eKo1 | gugaiz: No because certain info. is not there until the call finishes, like the duration. |
| 21:10.35 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 21:11.25 | gugaiz | eKo1, ok, that is the problem because I only need the UA ip address, the number and unique id of call, but in the moment that call connect |
| 21:11.31 | FuriousGeorge | Jul 26 17:59:10 NOTICE[2965] chan_zap.c: Avoiding deadlock... |
| 21:11.37 | FuriousGeorge | i got a couple thousand of those in one shot |
| 21:11.38 | rene- | Corydon-w: this is weird, if i do a realtime load queues name 'queuename' i get my data but it doesnt get loaded in asterisk: srv5*CLI> realtime load queues name guayamayo02 --> http://pastebin.ca/109095 |
| 21:11.44 | FuriousGeorge | or one second i should say |
| 21:12.35 | gugaiz | there are any way to do that? |
| 21:13.25 | FuriousGeorge | then today asterisk crashed. it seems to do so every few weeks. im gonna have to cron it to restart |
| 21:13.30 | eKo1 | gugaiz: yes |
| 21:13.55 | eKo1 | FuriousGeorge: I restart * everyday at 5:00 AM. |
| 21:14.39 | FuriousGeorge | eKo1: i have a few different servers and it seems the more you use'em the more often you gotta restart them |
| 21:14.42 | FuriousGeorge | its a shame really |
| 21:15.20 | eKo1 | Yes it is, but you learn to live with it. |
| 21:15.28 | *** join/#asterisk xserve (n=jimmy@69.54.29.188) |
| 21:15.29 | gugaiz | eKo1, can you give a page or a word to find documentation |
| 21:15.38 | eKo1 | gugaiz: grep the source code |
| 21:15.39 | FuriousGeorge | i think its my zap hardware. sometimes fxo channels get in use, and there is no way to getem back without restarting the server |
| 21:15.55 | FuriousGeorge | well not the server but asterisk itself |
| 21:15.57 | xserve | Hi, I am trying to setup the Asterisk server, and I am assuming I will need to do some port forwarding since I am behind a router, but I am not sure which ports I need to forward |
| 21:16.03 | eKo1 | FuriousGeorge: nah, it isn't the hardware because I run into troubles on a pure VoIP * box. |
| 21:16.23 | eKo1 | xserve: search voip-info.org for asterisk firewall |
| 21:16.27 | FuriousGeorge | eKo1: does you CLI get really non-responsive, and funny acting |
| 21:16.37 | xserve | ok thanks |
| 21:16.42 | eKo1 | FuriousGeorge: Yes. |
| 21:16.52 | *** join/#asterisk AJaymn (n=Ya@70.59.126.206) |
| 21:17.08 | FuriousGeorge | eKo1: at least its not only me. today it was so bad i couldnt even restart now |
| 21:17.12 | FuriousGeorge | i had to reboot the server |
| 21:17.34 | FuriousGeorge | eKo1: so restarting at 5am should keep it up all the time, you think? |
| 21:17.35 | brookshire | furious: do you use agents? |
| 21:17.39 | FuriousGeorge | brookshire: no |
| 21:17.47 | *** part/#asterisk Mattwj2005 (n=Matt@user-12l3n0n.cable.mindspring.com) |
| 21:17.51 | brookshire | hmm.. nm then :) |
| 21:18.31 | FuriousGeorge | brookshire: i notice calling myself via zap will sometimes cause me to lose an fxo |
| 21:18.40 | FuriousGeorge | till i restart the sever |
| 21:18.47 | xserve | hrmm dang i need a range and my router wont let me do ranges |
| 21:18.49 | brookshire | can you reproduce it? |
| 21:19.59 | Zodiacal | anyone know if its posible to to create a applicationmap that transfers calls? i.e. a speed dial of sorts that will allow one button transfering |
| 21:20.08 | Zodiacal | transfer() doesn't seem to work for me |
| 21:20.10 | FuriousGeorge | brookshire: sure i can call myself a bunch of times till i see a starting simple switch that doesnt correspond to a "hungup..." |
| 21:20.10 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 21:20.22 | arkonadev | does anyone here have experience using fonality and pbxtra? |
| 21:20.39 | FuriousGeorge | eKo1: http://lists.digium.com/pipermail/asterisk-dev/2005-December/017552.html this guy seems to have the same issue as us. no one responded to him on the mailing list |
| 21:20.59 | brookshire | furious: if it is digium hardware, i would suggest calling support and showing them |
| 21:21.15 | brookshire | they might be able to fix it |
| 21:21.17 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 21:21.23 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 21:22.29 | eKo1 | FuriousGeorge: It would be wise for you to respond. |
| 21:22.48 | pigpen | Has anyone had issues with Polycom phones (601 to be specific) while transfering a call, it misses a digit upon entry of the exten? |
| 21:22.57 | brookshire | furiousgeorge: that looks like an agent/queue problem, are you using queues and agents? |
| 21:23.58 | brookshire | pigpen: the first digit? |
| 21:24.01 | pigpen | hmm...let me check... |
| 21:24.38 | *** join/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 21:24.56 | pigpen | It is the 3rd digit. |
| 21:25.14 | pigpen | like <transfer>123, it will miss the "3" |
| 21:25.23 | pigpen | or: |
| 21:25.29 | pigpen | like <transfer>123*, it will miss the "3" |
| 21:25.45 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 21:25.56 | brookshire | i've had problems with iax that were similar to this |
| 21:26.03 | brookshire | but i just updated asterisk and it worked again |
| 21:26.14 | pigpen | hmm...no iax...even internal stuff. |
| 21:26.35 | pigpen | I have had the issue from 1.2.0 -> 1.2.9.x |
| 21:26.46 | brookshire | it could be dialplan |
| 21:26.57 | pigpen | I just figured it was a sip software thing on the phone...but still no dice. |
| 21:27.05 | pigpen | hmm... |
| 21:27.10 | pigpen | dialplan...hmm... |
| 21:27.27 | brookshire | it could be set to drop the last digit |
| 21:27.40 | pigpen | oh...did I mention...it comes and goes... |
| 21:27.41 | pigpen | :) |
| 21:28.14 | pigpen | ...and I have had the issue on 3 differnet polycom 601's, across Rev: A - C |
| 21:28.19 | brookshire | hah |
| 21:28.30 | *** join/#asterisk overworked554 (n=overwork@atlantis.clearshout.com) |
| 21:28.38 | brookshire | polycom's are throughly tested with asterisk.. |
| 21:28.46 | brookshire | something has to be not right |
| 21:28.48 | pigpen | ...and supported! |
| 21:29.05 | brookshire | not right with a config |
| 21:29.19 | brookshire | what is your digitmap set to on your phone? |
| 21:29.31 | pigpen | well, I deployed the same "skeleton" dialplan for someone...but they use 4 digit not 3.... |
| 21:29.34 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 21:29.41 | pigpen | let me call them to see if they have noticed it. |
| 21:30.19 | *** part/#asterisk overworked554 (n=overwork@atlantis.clearshout.com) |
| 21:31.32 | pigpen | yeah...4 digit no issue. |
| 21:31.43 | hmmhesays | wiki wiki wild wild |
| 21:31.44 | brookshire | could be a digitmapping problem |
| 21:31.48 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:34.55 | pigpen | Digit mapping...hmm.... |
| 21:34.59 | pigpen | ok..so what the hell is that? |
| 21:35.10 | pigpen | Here is my "line" where it dials: exten => _1XX,6,Dial(SIP/${EXTEN},20,wW) |
| 21:35.23 | pigpen | note this is line 6 of 19 |
| 21:35.44 | *** part/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 21:36.02 | CunningPike | pigpen: Look in your sip.cfg (the Polycom one) for 'digitmap' |
| 21:36.49 | pigpen | ok....found it. |
| 21:37.04 | pigpen | I guess I will dig out the sip admin manual to "educate" my self |
| 21:37.25 | pigpen | Currently it is: |
| 21:37.54 | pigpen | <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/> |
| 21:38.15 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
| 21:38.45 | dos000 | i just did a load test on openser using sipp. the output is culmilating at 2600 cps .. with only registers ! not too shaby .. but i was expecting more |
| 21:40.20 | eKo1 | dos000: maybe that is a hardware limit |
| 21:40.31 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 21:40.32 | eKo1 | Test it on a more powerful setup |
| 21:42.05 | dos000 | eKo1, what would be reasonable ... this is a 3g machine with 1G ram on the server. also i am only sending register and authenticating if i get 401. what would asterisk be capable of ? |
| 21:42.13 | denon | make me a bootable cd and I'll test it on a quad-core xeon :) |
| 21:42.18 | denon | well, dual dual code |
| 21:42.19 | denon | core |
| 21:42.48 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 21:42.53 | *** part/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 21:43.01 | dos000 | eKo1, the test machine is doing 85%cpu and the server 75% |
| 21:43.18 | dos000 | denon, if i get that machine i'll mary it myself ;-) |
| 21:43.20 | TripleFFFF | darn sipsaking it ? |
| 21:44.25 | dos000 | TripleFFFF, with sipsak i could do 6000 option msg per sec |
| 21:45.01 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 21:45.04 | dos000 | eKo1, are you familiar with sipp ? |
| 21:45.10 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 21:45.23 | pigpen | denon, what, no 8 way? |
| 21:47.41 | AgiNamu | We have an 8-way we're gonna run asterisk on, but its via ESX Server :) |
| 21:48.01 | AgiNamu | Crossing my fingers, hoping the TDM cards will work with ESX 3 |
| 21:48.03 | denon | pigpen: actually, these boxes will be 8-way when the quad-core CPUs come out end of this year |
| 21:48.38 | pigpen | yeah...we are running asterisk on a few Dell 6850's with duals....8GB ram...very nice. |
| 21:48.53 | AgiNamu | Why 6850 |
| 21:49.00 | AgiNamu | can you get in 3 or 4 T1 cards |
| 21:49.11 | pigpen | The customer said "we want redundancy and high uptime" so hey...6850 is the way to go. |
| 21:49.27 | pigpen | the 6850 is fully hotswap, even down to the proc. |
| 21:49.35 | AgiNamu | Oh really |
| 21:49.39 | *** join/#asterisk anthm (n=anthm@204.250.115.207) |
| 21:49.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:49.41 | pigpen | Oh yes. |
| 21:50.10 | pigpen | But: one downside. No molex power connector. So you have to get external power to power fxs's |
| 21:50.17 | AgiNamu | you got a name for that or a link? Cause that's awesome information |
| 21:50.20 | pigpen | ^^kinda sucky in my opinion |
| 21:50.27 | *** join/#asterisk crocz (n=crocco@cc961619-a.groni1.gr.home.nl) |
| 21:50.34 | pigpen | name? |
| 21:50.43 | pigpen | See Dell.com |
| 21:50.44 | AgiNamu | like what they call hot swapping procs |
| 21:50.54 | AgiNamu | or where they say that |
| 21:51.05 | pigpen | Mind you you have to have atleast 2 physical procs... |
| 21:52.00 | AgiNamu | im looing at it, and all I see is hot power/fans/memory/disk |
| 21:52.01 | AgiNamu | pci |
| 21:52.33 | denon | hah - you guys see this? http://denon.cx/letterman-gates |
| 21:54.08 | AgiNamu | heh. |
| 21:54.27 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 21:54.33 | AgiNamu | anyways pigpen, how would you cover from a CPU failure? |
| 21:54.42 | AgiNamu | your system state would be unknown |
| 21:54.43 | sp0n9e | talk about a rare event. |
| 21:55.15 | dlynes_office | AgiNamu: you ask yourself why you were using a 286 to run asterisk? |
| 21:55.39 | AgiNamu | ? |
| 21:55.51 | sp0n9e | wouldn't it be a lot better to have redundant servers before a hotswap cpu... |
| 21:55.54 | pigpen | I haven't looked for "documention" regarding it. I have a kernel dev as a business partner...who inserted the code into the 2.6.x |
| 21:55.57 | dlynes_office | or why you overclocked the cpu to the hilt? |
| 21:56.19 | sp0n9e | dlynes_office: i've had more instances where my ram died before my cpu |
| 21:56.29 | AgiNamu | but theres not even any mention of it on the dell site |
| 21:56.32 | dlynes_office | sp0n9e: exactly my point |
| 21:56.39 | sp0n9e | :) |
| 21:56.44 | AgiNamu | yea, and the 2850s support chipkill and all that |
| 21:56.54 | dlynes_office | sp0n9e: i've had ram die a few times....i've had hard drives die a lot of times, but never had a cpu die |
| 21:56.58 | AgiNamu | so I'm just questioning what high availablility a dell 6850 has over a 2850 |
| 21:57.44 | pigpen | I would download the tech sheets of both. |
| 21:58.16 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 21:58.52 | rene- | Corydon-w: i needed to add members to the queue before i saw their status (doh!) |
| 21:59.33 | TripleFFFF | so i might be better |
| 22:01.12 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 22:01.40 | *** part/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 22:01.47 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 22:02.00 | Qwell[] | dlynes_office: I've had a CPU explode on me |
| 22:02.12 | Qwell[] | dlynes_office: of course...we were taking a blowtorch to it at the time, but... |
| 22:02.17 | dlynes_office | Qwell[]: heh |
| 22:02.28 | sp0n9e | it's fun when power supplies die. |
| 22:02.28 | TripleFFFF | btw wahts dif on onet et offnet |
| 22:02.29 | TripleFFFF | ;) |
| 22:02.40 | Qwell[] | TripleFFFF: try again |
| 22:02.48 | dlynes_office | Qwell[]: the only time I've ever seen that happen is on the old crappy Intel 386dx/33's |
| 22:03.08 | Qwell[] | It was an amd 386 :D |
| 22:03.15 | dlynes_office | Qwell[]: they didn't dissipate heat well, and so if they were on for too long a period in too warm of a room, their tops would pop |
| 22:03.22 | sp0n9e | you can't imagine the fun that is had when flames release the magic smoke of a previously working power supply. |
| 22:03.29 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:03.35 | dlynes_office | Qwell[]: nah...the amd's had ceramic tops, and would dissipate heat well |
| 22:03.42 | TripleFFFF | i mean whats differentce on ON NET and OFF NET |
| 22:04.01 | dlynes_office | TripleFFFF: on net means they have a pop there; off net means they don't |
| 22:04.16 | Qwell[] | dlynes_office: remember - blowtorch |
| 22:04.29 | AgiNamu | Yea but if a CPU dies, your system is dead, since there's no way it can recover. Switching a CPU out might be possible by not scheduling stuff on it |
| 22:04.36 | AgiNamu | but if it just failed, your system is down |
| 22:04.39 | dlynes_office | Qwell[]: i'll keep that in mind next time I want to blow up a cpu |
| 22:05.45 | dlynes_office | AgiNamu: they're playing into your fear, and you're gulping it up as fast as you can |
| 22:07.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 22:09.29 | AgiNamu | what fear?? |
| 22:09.52 | dlynes_office | your fear of a cpu dying :) |
| 22:12.31 | AgiNamu | haha i dont fear that |
| 22:12.50 | trelane | I had an intel DX4/100 blew up on me |
| 22:12.54 | trelane | s/blew/blow |
| 22:13.05 | AgiNamu | someone was just saying the 6850 was more highly available than a dell 2850 because you can hot swap a proc on a 6850 and I have never heard of that, and dont think it'd do any good anyways |
| 22:13.09 | dlynes_office | serves ya right for buying intel :) |
| 22:13.11 | Qwell[] | I had a 100mhz cpu melt a hole in the plastic under it |
| 22:13.17 | Qwell[] | That was tricky to fix... |
| 22:13.32 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:13.43 | Qwell[] | the actual connectors (and the CPU!) were still okay, so we heated up a sewing needle, and remelted it to fit |
| 22:14.05 | [TK]D-Fender | TripleFFFF : I presume you're asking about Unlimitel for on/offnet? |
| 22:15.43 | TripleFFFF | no |
| 22:15.44 | TripleFFFF | lol |
| 22:15.47 | TripleFFFF | on us |
| 22:15.54 | TripleFFFF | unlimitel ..hmm |
| 22:16.02 | TripleFFFF | i can do 0.9 on net |
| 22:16.09 | TripleFFFF | 1.1 off net canada |
| 22:16.15 | TripleFFFF | 0.9 on net canada |
| 22:16.17 | TripleFFFF | so not bad |
| 22:16.38 | dlynes_office | I'm getting 0.05 on net canada |
| 22:16.45 | dlynes_office | erm |
| 22:16.47 | dlynes_office | 0.5 |
| 22:18.45 | TripleFFFF | hey |
| 22:18.49 | TripleFFFF | commit ? |
| 22:18.56 | TripleFFFF | and where from ;) |
| 22:19.19 | file | Telcomjoshvoxmart has on net at 0.49 |
| 22:19.37 | Qwell[] | qwell communications has off net at 0.93 |
| 22:19.38 | dlynes_office | file: heh |
| 22:19.41 | TripleFFFF | telcominindia as on net to hell for 0.000666 |
| 22:19.51 | Qwell[] | What are we talking about anyhow? |
| 22:20.01 | dlynes_office | TripleFFFF: commit? |
| 22:20.08 | dlynes_office | TripleFFFF: what does svn have to do with minutes? |
| 22:20.13 | linlin | is there any way to use the asterisk command line to reboot a server? |
| 22:20.20 | file | dlynes_office: how much do you commit to I think he means |
| 22:20.22 | Qwell[] | linlin: !shutdown -r now |
| 22:20.32 | dlynes_office | file: oh |
| 22:20.37 | linlin | on the asteris kcli? |
| 22:20.38 | dlynes_office | zippo |
| 22:20.53 | dlynes_office | no minimum commitments |
| 22:21.10 | TripleFFFF | where from ;) |
| 22:21.17 | dlynes_office | www.five9snetwork.com |
| 22:21.31 | TripleFFFF | BC |
| 22:21.38 | dlynes_office | They're based out of Vancouver, and their SIP server's in Toronto's peer one facility |
| 22:21.45 | dlynes_office | yep |
| 22:22.37 | dlynes_office | bbiab |
| 22:22.41 | dlynes_office | just grabbing something to eat |
| 22:23.01 | TripleFFFF | Canada Quebec Montreal $0.0045 7/23/2006 |
| 22:27.50 | AgiNamu | Anyone from Asteria here? |
| 22:29.14 | dlynes_office | TripleFFFF: yeah..pretty good prices, eh? |
| 22:29.30 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 22:33.04 | folder | Any AstLinux users in here? I can't find the AstLinuxSetup windows CF-image writer tool. the url of the chaps homepage is dead :( |
| 22:34.21 | TripleFFFF | hmm |
| 22:34.28 | TripleFFFF | quality ? |
| 22:34.42 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 22:35.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:36.43 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:36.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:37.44 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:39.03 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 22:40.19 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 22:48.07 | *** join/#asterisk rockoo (n=rockoo@mail.copronet.com.pe) |
| 22:48.10 | rockoo | hi all |
| 22:48.22 | CunningPike | ~seen all |
| 22:48.33 | jbot | all <n=Elive_us@pc-82.160.232.108.olsznet.ec.pl> was last seen on IRC in channel #elive, 130d 3h 45m 46s ago, saying: 'hello kender ;)'. |
| 22:50.53 | rockoo | newbie question - sorry - if i want to call w/ a sipsoftphone to a external number i need a fxs-port? |
| 22:51.12 | E-bola | u need a something to connect sip to pstn |
| 22:52.39 | rockoo | ok - but this thingie needs a fxo-port to make sip to pstn? |
| 22:53.10 | E-bola | depends if u wanna connect ur sip phone to pstn urself |
| 22:53.19 | E-bola | i think the most normal thing to do is to use a pstn gateway |
| 22:53.43 | CunningPike | rockoo: You can either connect to the PSTN yourself with an FXO adapter, or use the services of an ITSP |
| 22:54.41 | dlynes_office | rockoo: you can use a usb->fxo adapter, pci->fxo adapter, fxo gateway, or an itsp |
| 22:55.46 | dlynes_office | rockoo: are you only using one sip softphone, and no other sip devices? |
| 22:55.55 | rockoo | ok the idea is to use asterisk for sip and connect a voip-gateway for the external calls |
| 22:56.33 | dlynes_office | rockoo: you're going ot use it for incoming calls as well? not just outgoing calls? |
| 22:56.49 | rockoo | for both |
| 22:56.53 | dlynes_office | rockoo: and it's only for one phone? |
| 22:57.05 | rockoo | for outgoing i need fxo - and for incomming fxs? |
| 22:57.15 | CunningPike | ~fxofxs |
| 22:57.16 | jbot | i guess fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 22:57.23 | rockoo | no for 4 to 16 lines |
| 22:57.23 | dlynes_office | for outgoing/incoming you need fxo, for an analog phone, you need fxs |
| 22:57.43 | dlynes_office | rockoo: and you want them all to be analog? no voip lines? |
| 22:57.47 | rockoo | fxs - is need if i want to connect a fax |
| 22:58.07 | dlynes_office | rockoo: fxs is if you want to frustrate yourself trying to get a fax to work |
| 22:58.19 | rockoo | dlynes_office: jep - overhere in peru voip is not much used ... |
| 22:58.50 | rockoo | sorry - i have to switch the office - be right in 30 min :) |
| 22:58.57 | dlynes_office | rockoo: yeah, so i would suggest getting either a digium tdm2400p w/hwec, or a sangoma a200d |
| 22:59.12 | dlynes_office | rockoo: forget the itsp and the gateway |
| 22:59.35 | rockoo | the idea was to connect 2,3 or 4 voip gateways ... |
| 23:00.05 | dlynes_office | rockoo: it's cheaper to go with a tdm2400p or a sangoma a200d than it is to go with a bunch of four port gateways |
| 23:00.18 | dlynes_office | rockoo: and then you don't have to play with all those power supplies |
| 23:00.48 | rockoo | dlynes_office: do you have ast. price of th tdm2400p? |
| 23:00.54 | dlynes_office | both the tdm2400p and the sangoma a200d can handle up to 24 phone lines |
| 23:01.18 | dlynes_office | ast.? |
| 23:01.24 | Qwell[] | est |
| 23:01.24 | rockoo | exstimate |
| 23:01.29 | dlynes_office | ah |
| 23:01.42 | rockoo | aaah - yes ... |
| 23:01.46 | dlynes_office | Not offhand, no |
| 23:01.54 | dlynes_office | I don't deal in retail prices; only wholesale |
| 23:02.09 | rockoo | ok - thx - be right back... |
| 23:02.30 | *** part/#asterisk sp0n9e (n=sp0n9e@69.12.216.48) |
| 23:02.33 | dlynes_office | but last time I checked, I think it was about $500-600 for a four port sangoma a200/tdm400 without a hardware echo can |
| 23:03.11 | E-bola | any1 tried cancelling echo on a sip only setup? |
| 23:03.28 | dlynes_office | Ebola: no such thing as echo on a sip-only setup |
| 23:03.40 | dlynes_office | Ebola: echo is generated by analog equipment |
| 23:03.42 | hads | Unless you have really bad handsets |
| 23:03.53 | dlynes_office | hads, handsets are analog :) |
| 23:03.59 | carrar | everyone phone is analog |
| 23:04.04 | hads | heh, true :) |
| 23:04.05 | dlynes_office | hads, they're just going through a digital base :) |
| 23:04.16 | CunningPike | E-bola: SIP-to-SIP echo is acoustic coupling in the set |
| 23:04.28 | CunningPike | s/is/is usually/ |
| 23:04.37 | dlynes_office | CunningPike: or caused by faulty setup on the itsp's end |
| 23:04.51 | dlynes_office | CunningPike: or itsp's that are too cheap to buy hwec :) |
| 23:05.18 | CunningPike | Don't know any of those ;) |
| 23:05.41 | dlynes_office | CunningPike: hehe...sure you don't :) |
| 23:05.46 | dlynes_office | CunningPike: you're not using hwec :) |
| 23:05.51 | dlynes_office | CunningPike: erm wait |
| 23:06.00 | CunningPike | Yes we are - the Mother of HWEC |
| 23:06.03 | dlynes_office | CunningPike: forgot...you went out and bought some after the fact, right? |
| 23:06.31 | CunningPike | dlynes_office: We gave up on the the Digium VPM and bought a Ditech |
| 23:06.48 | dlynes_office | CunningPike: thought digium just recently got hwec? |
| 23:07.18 | CunningPike | dlynes_office: We bought our board about a year ago - maybe less |
| 23:07.22 | dlynes_office | ah |
| 23:08.04 | CunningPike | Since we got the Ditech, it's "Echo? What echo?" :D |
| 23:08.17 | dlynes_office | CunningPike: same for us, since we got the a200d's |
| 23:08.25 | CunningPike | dlynes_office: Cool |
| 23:08.43 | dlynes_office | the a200d's are the analog cousin to the a104d |
| 23:08.52 | CunningPike | dlynes_office: Bet yours was a lot cheaper than ours :D |
| 23:09.07 | dlynes_office | CunningPike: about $550USD |
| 23:09.16 | dlynes_office | CunningPike: that was for four ports, and hwec |
| 23:09.22 | CunningPike | dlynes_office: Our Ditech was about $3k |
| 23:09.30 | dlynes_office | yeah, but that's a four port pri card |
| 23:09.35 | dlynes_office | not a four port analog tdm card |
| 23:09.48 | CunningPike | Correct - it's an external box, not a card |
| 23:09.53 | dlynes_office | damn |
| 23:09.55 | dlynes_office | a whole box |
| 23:10.00 | dlynes_office | suckage |
| 23:10.16 | dlynes_office | probably would've been cheaper just to get a sangoma a104d |
| 23:10.27 | CunningPike | dlynes_office: It's a little box - only about 8" deep - rack mount |
| 23:10.51 | dlynes_office | then you'd have a four port pri pci card (universal slot), and Octasics carrier grade echo can |
| 23:11.17 | CunningPike | dlynes_office: How long have they been out for? |
| 23:11.22 | dlynes_office | a while |
| 23:11.43 | dlynes_office | I think it was Sangoma's first card for the digital telephony market |
| 23:11.57 | dlynes_office | They've been making WAN hardware for many years |
| 23:12.04 | hads | Can't you put the new Digium HWEC on the older boards? |
| 23:12.23 | [TK]D-Fender | dlynes_office : Sangoma's been in the T1 world for decades |
| 23:12.30 | dlynes_office | ah |
| 23:12.49 | dlynes_office | so the a104d's been around for a long time, then? |
| 23:12.53 | [TK]D-Fender | dlynes_office : More popularly they started with the A101/102 single/double T/E1 |
| 23:13.02 | [TK]D-Fender | dlynes_office : Yeah... since last october :) |
| 23:13.06 | CunningPike | hads: Maybe - but it never worked well for us |
| 23:13.08 | dlynes_office | that's not a long time |
| 23:13.38 | CunningPike | dlynes_office: And not long enough ago for us....... didn't exist when we bought our TE411s |
| 23:14.14 | dlynes_office | but as far back as I remember, there's been sangoma driver support in the Linux kernel |
| 23:14.40 | hads | CunningPike: It's accademic now, since you have your solution. But you're talking about the old HWEC can right? Not the new Octasic one. |
| 23:14.55 | *** join/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 23:15.02 | dlynes_office | and I started configuring and compiling my own kernels starting with the 2.0 kernel |
| 23:15.17 | CunningPike | hads: I'm guessing so - we bought ours about a year ago |
| 23:15.27 | hads | Yeah. |
| 23:15.28 | dlynes_office | I've been using linux since before 1.0 mind you, though, and sangoma drivers were in the kernel then, too |
| 23:15.48 | CunningPike | dlynes_office is actually 104 years old |
| 23:16.01 | dlynes_office | CunningPike: ummm...foomaster |
| 23:16.08 | CunningPike | :D |
| 23:16.11 | dlynes_office | CunningPike: linux is only about 12 or 13 years old |
| 23:16.31 | dlynes_office | well...linux that you could actually use, i mean |
| 23:16.46 | dlynes_office | i.e. that would boot and had support software running with it |
| 23:17.01 | dlynes_office | it wasnt' terribly stable then, though :) |
| 23:21.06 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 23:21.51 | robl^ | I've been using Linux since kernel 0.99.pl9 |
| 23:24.05 | mog | well robl^ my name is linus |
| 23:24.08 | mog | so ppttth ^_^ |
| 23:24.40 | mog | i have only been using it since 99 |
| 23:24.57 | CunningPike | My name is Al Gore, and I invented this whole thing, so stfu the rest of you pipsqueaks |
| 23:26.42 | dlynes_office | robl^: there was never any such thing as kernel 0.99.pl9 |
| 23:27.01 | dlynes_office | robl^: kernel patches didn't really exist yet, at that point |
| 23:27.24 | dlynes_office | robl^: linus just tacked another decimal point on to the end of the version number instead |
| 23:28.00 | mog | burn.... |
| 23:28.13 | mog | i think i started at 2.0 or just pre 2.0 |
| 23:28.16 | mog | i cant rememeber |
| 23:28.24 | dlynes_office | mog: slackware 96? |
| 23:28.36 | CunningPike | Well, I started at kernel 2.4, so there :P |
| 23:28.53 | mog | no i started with debian |
| 23:28.54 | dlynes_office | CunningPike: yeha, but you started with windows 2000, too |
| 23:29.06 | CunningPike | dlynes_office: Actually no - Solaris |
| 23:29.08 | mog | i can figure it out |
| 23:29.35 | CunningPike | dlynes_office: Pre-NT 3.1 - I remember seeing my first NT box |
| 23:29.39 | dlynes_office | mog: yeah...i set aside some spare partitions when i set up my laptop...was thinking about putting debian on there |
| 23:29.54 | *** join/#asterisk jbalcomb (n=JimBalco@m495e36d0.tmodns.net) |
| 23:29.58 | robl^ | dlynes_office: actually they DID have pl's even a quick google will show you that |
| 23:30.02 | dlynes_office | Corydon-w: ooooooh.....apple ][ |
| 23:30.32 | Corydon-w | I'm just happy I started with 16-sector disks, instead of the earlier 13-sector disks from DOS 3.2 |
| 23:30.32 | jbalcomb | ~seen [TK]D-Fender |
| 23:30.46 | jbot | [tk]d-fender is currently on #asterisk (2h 30m 46s). Has said a total of 4 messages. Is idling for 17m 44s, last said: 'dlynes_office : Yeah... since last october :)'. |
| 23:30.52 | Corydon-w | 140k was a lot of disk space |
| 23:31.13 | dlynes_office | Corydon-w: apple DOS, or ProDOS? |
| 23:31.24 | Corydon-w | I started with Apple DOS |
| 23:31.29 | Corydon-w | ProDOS came later |
| 23:31.32 | dlynes_office | I think I was using Apple DOS 3.2 and ProDOS 3.3.1 |
| 23:31.54 | Corydon-w | I doubt you'd remember DOS 3.2, if you were using ProDOS |
| 23:32.17 | dlynes_office | Corydon-w: I sold my two Apple ][+'s in 1994. |
| 23:32.24 | Corydon-w | Uh, and ProDOS never made it past 2.1 |
| 23:32.49 | dlynes_office | Corydon-w: ok, then maybe I got the version number wrong....it was some long version number |
| 23:33.00 | Corydon-w | 2.0.3, perhaps? |
| 23:33.07 | dlynes_office | nope...don't think so |
| 23:33.12 | dlynes_office | that version number doesn't sound familiar |
| 23:33.20 | dlynes_office | maybe 2.0.1 though |
| 23:33.35 | dlynes_office | or maybe 1.3.1? |
| 23:33.41 | Corydon-w | Quite possible |
| 23:33.42 | dlynes_office | somehting with a 1 on the end |
| 23:34.20 | jbalcomb | [TK]D-Fender I found my X-Lite problem: "Not using codec g711u due to insufficient upstream bandwidth" |
| 23:34.28 | dlynes_office | heh |
| 23:34.35 | dlynes_office | good job :) |
| 23:35.05 | dlynes_office | jbalcomb: i guess you've got less than 100Kbps upstream bandwidth? |
| 23:35.16 | jbalcomb | guess my cellular dial-up isn't good enough. |
| 23:35.22 | dlynes_office | ah yeah |
| 23:35.29 | dlynes_office | that would be 50-70 kbps |
| 23:35.39 | mog | i started with debian 2.0 hamm so linux 2.0. something |
| 23:35.58 | mog | so i guess im not as cool as some of ya folk.... |
| 23:36.02 | Corydon-w | http://en.wikipedia.org/wiki/ProDOS |
| 23:36.04 | jbalcomb | dlynes_office: well, it would be nice to pay $30/mnth for unlimited cellular internet and also get unlimited phone calls too eh? |
| 23:36.13 | [TK]D-Fender | jbalcomb : Cell = suck :) |
| 23:36.22 | dlynes_office | jbalcomb: we've got something here called 'portable' internet now |
| 23:36.27 | mog | hey Corydon-w i just got an apple 2e |
| 23:36.28 | [TK]D-Fender | jbalcomb : What did I tell you about SANE testing parameters?! |
| 23:36.28 | jbalcomb | [TK]D-Fender yeah but its damn handy |
| 23:36.34 | mog | can you hooke me up with hw / sw |
| 23:36.53 | [TK]D-Fender | jbalcomb : Off for a bit back later. |
| 23:36.56 | dlynes_office | jbalcomb: it's gsm edge internet (55-70kbps), and you don't need a cellphone |
| 23:37.01 | jbalcomb | [TK]D-Fender: well, i thought it would suck but i didnt think x-lite would deny me the option |
| 23:37.05 | dlynes_office | jbalcomb: just a gsm modem and a sim card |
| 23:37.34 | [TK]D-Fender | jbalcomb : Will begin testing tonight. Are there any config changes scheduled or can I count "now" as a feature freeze? |
| 23:37.47 | jbalcomb | dlynes_office: that sounds just like what i have but faster |
| 23:37.52 | Corydon-w | mog: have a 800k floppy for it? |
| 23:37.57 | mog | yes |
| 23:38.00 | dlynes_office | jbalcomb: you in canada? |
| 23:38.05 | Corydon-w | mog: 1 MB memory? |
| 23:38.07 | mog | i got a bunch of educational titles for it |
| 23:38.11 | mog | im not sure on memory |
| 23:38.18 | mog | i have 80 column expansion card |
| 23:38.19 | Corydon-w | mog: Appleworks 6? |
| 23:38.22 | mog | and another set of memory |
| 23:38.27 | russellb | :q |
| 23:38.27 | robl^ | dlynes_office: http://www.manualy.sk/sag/node68.html check the boot log in this OLD version of the Linux SysAdmin Guide ;-) |
| 23:38.28 | mog | as well as a homebrew card |
| 23:38.30 | Corydon-w | mog: probably only 128k |
| 23:38.30 | mog | someone made |
| 23:38.32 | jbalcomb | [TK]D-Fender oh, this week i have to fix all those call queues so i have lots of changes to make |
| 23:38.52 | Corydon-w | mog: you can get a replacement for the 80column card which will give you 1MB memory |
| 23:38.53 | mog | im planing on getting serial card for it |
| 23:38.55 | jbalcomb | [TK]D-Fender: if you look at call queue tesuto you can see what they are going to all look like |
| 23:39.04 | mog | so i can use it as a terminal |
| 23:39.10 | mog | oh thats hot convey |
| 23:39.12 | mog | er Corydon-w |
| 23:39.29 | Corydon-w | mog: there's also an Ethernet card available |
| 23:39.40 | jbalcomb | [TK]D-Fender i don't beleive i'll be changing much else though |
| 23:39.42 | mog | yeah i saw that |
| 23:39.44 | Corydon-w | mog: although the drivers for the Ethernet card require a IIgs |
| 23:39.44 | [TK]D-Fender | jbalcomb : Ok, I'll DL everything as a reference point. |
| 23:39.46 | mog | but no programs for 2e |
| 23:39.49 | mog | only 2gs |
| 23:39.54 | [TK]D-Fender | jbalcomb : BIAB |
| 23:39.54 | mog | well they have drivers |
| 23:39.57 | mog | but nothing that can use it |
| 23:40.04 | Corydon-w | mog: only $155 for the Ethernet card |
| 23:40.10 | mog | i found em cheaper |
| 23:40.14 | mog | just worthless |
| 23:40.15 | Corydon-w | Where? |
| 23:40.23 | mog | there are two ethernet cards |
| 23:40.28 | mog | that different people made |
| 23:40.34 | Corydon-w | Ah |
| 23:40.43 | mog | one is cheaper |
| 23:40.50 | mog | bother are worthless as you need a gs |
| 23:40.55 | mog | to do anything with them |
| 23:40.59 | mog | or contiki |
| 23:41.06 | mog | which kinda defeats my purpose |
| 23:41.47 | dlynes_office | robl^: try this: ftp://ftp.kernel.org/pub/linux/kernel/Historic/v0.99/linux-0.99.15.tar.bz2 |
| 23:41.55 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:42.16 | Corydon-w | mog: what about a SCSI card? |
| 23:42.26 | Corydon-w | mog: or you could also get one of the internal IDE drives |
| 23:42.27 | TripleFFFF | +list |
| 23:42.37 | TripleFFFF | mog=moc with a mug ? |
| 23:42.37 | dlynes_office | TripleFFFF: you're not in #freeswitch |
| 23:42.42 | TripleFFFF | oups |
| 23:42.42 | TripleFFFF | lol |
| 23:42.47 | mog | nope |
| 23:42.52 | mog | mog = mogorman |
| 23:42.58 | mog | = matt ogorman |
| 23:43.08 | robl^ | dlynes_office: yeah.. but if you compile and boot it, it will show pl15 under uname -a.. that didn't change until 1.0.x |
| 23:43.11 | mog | i want to get serial working as someone has homebrew server you can use |
| 23:43.20 | dlynes_office | robl^: ah |
| 23:44.01 | mog | so you can just load programs over that |
| 23:46.08 | *** join/#asterisk agboris (n=dm_it24@203.215.180.254) |
| 23:46.36 | agboris | call destroyed by asterisk with ... |
| 23:46.51 | agboris | <-- SIP read from 63.210.114.5:5060: |
| 23:46.51 | agboris | ACK sip:8007010349@192.168.2.56;user=phone SIP/2.0 |
| 23:46.51 | agboris | Via: SIP/2.0/UDP 63.210.114.5 |
| 23:46.51 | agboris | Call-ID: 13748479@63.210.114.5 |
| 23:46.51 | agboris | From: <sip:4105639547@63.210.114.5;user=phone>;tag=10000000-0-2087316567 |
| 23:46.52 | agboris | To: <sip:8007010349@192.168.2.56;user=phone>;tag=as7e116b7d |
| 23:46.54 | agboris | CSeq: 1 ACK |
| 23:46.55 | robl^ | dlynes_office: btw -- just downloaded the 0.99.15 kernel it uses the patchlevel convention in CHANGELOG ;-) |
| 23:46.56 | agboris | Contact: <sip:4105639547@63.210.114.5> |
| 23:46.58 | agboris | Content-Length: 0 |
| 23:47.00 | agboris | --- (8 headers 0 lines)--- |
| 23:47.01 | Qwell[] | agboris: hi to you too |
| 23:47.02 | agboris | Destroying call '13748479@63.210.114.5' |
| 23:47.03 | file | agboris: that is very very rude |
| 23:47.04 | agboris | can any one help me .... |
| 23:47.18 | file | agboris: and that is a perfectly fine SIP message |
| 23:47.52 | agboris | why my phone dont ring and queue is empty as well |
| 23:47.57 | Corydon-w | mog: apparently, the ][e will work with the Ethernet card and has drivers |
| 23:48.08 | Corydon-w | mog: even for the ][+, if you're so inclined |
| 23:48.33 | Corydon-w | mog: http://www.a2retrosystems.com/a2UtherManual.pdf |
| 23:48.41 | agboris | when i use soft phone my call connects and i can even talk... |
| 23:48.46 | dlynes_office | somebody must've been bored, obviously |
| 23:48.47 | mog | yeah but no programs |
| 23:48.53 | file | agboris: but your provider never sends you the call? |
| 23:48.54 | robl^ | EWWW!!!! a ][+ on ether??? I hope you invest in a gigabit ether net swich cuz those computers can satureate yer bandwidth |
| 23:49.03 | Corydon-w | mog: it has a web browser, email, telnet, IRC... |
| 23:49.03 | Nugget | telnet is eeeeeeevil! |
| 23:49.03 | dlynes_office | robl^: no kidding |
| 23:49.08 | mog | for 2e? |
| 23:49.20 | Corydon-w | mog: yes, all of those |
| 23:49.26 | Corydon-w | mog: it even supports DHCP |
| 23:49.39 | TripleFFFF | so |
| 23:49.39 | mog | i knew that |
| 23:49.41 | dlynes_office | my ii+ had enough problems just trying to load the Lisa assembler, never mind trying to do ethernet |
| 23:49.43 | mog | i knew it did dhcp |
| 23:49.46 | robl^ | last time I used a 2e, I was in high scool |
| 23:50.00 | TripleFFFF | if i want somethign better then the manager crap i could hmmm either use agi or write a module |
| 23:50.09 | agboris | my provider is working fine my other PBX is going wel |
| 23:50.20 | dlynes_office | sold both of my apple ][+'s 12 years ago for the princely sum of $100Cdn :p |
| 23:50.40 | robl^ | which is $20US *ducks* |
| 23:50.44 | file | agboris: is it the same account? |
| 23:50.52 | agboris | <file> yes |
| 23:50.53 | dlynes_office | robl^: umm...I think you mean $120USD |
| 23:51.05 | dlynes_office | robl^: the USD isn't worth squat, anymore |
| 23:51.08 | agboris | but on other pbx |
| 23:51.15 | file | agboris: you're registering to the same account from two Asterisk installs? |
| 23:51.19 | robl^ | yeah.. our economy is sinking faster than SCO |
| 23:51.26 | dlynes_office | robl^: no kidding |
| 23:51.42 | agboris | <file> yes, but i have turned other one off... |
| 23:51.57 | dlynes_office | sco used to be a pretty good company, too |
| 23:51.57 | agboris | that is causing problem and i am preparing backup |
| 23:52.15 | robl^ | ohh!!! I love my new phone!! I've gone from Grandstream to Cisco to Snom... to.... Aastra! |
| 23:52.17 | dlynes_office | until they decided their main line of business was pursuing intellectual property infringements |
| 23:52.40 | agboris | <file> what could be the issue |
| 23:52.54 | dlynes_office | robl^: are you being sarcastic, or do you actually like Aastras? |
| 23:53.04 | file | agboris: sip show registry shows you registered from the box that you want calls to go to, but you get no calls there? |
| 23:53.10 | robl^ | dlynes_office: I am not sure they even have a clue what they are doing anymore. I think they just employ people to keep lawyers busy |
| 23:53.22 | agboris | yes ..... |
| 23:53.29 | file | agboris: then work with your provider! |
| 23:54.12 | agboris | <file> please guide me ...what to say or do.... |
| 23:54.18 | robl^ | dlynes_office: I actually like my Aastra! I have decomissioned 3 of my Snom 360s and replaced with Aastra 480is. Going to add 4 more Aastras.. prolly 9133i |
| 23:54.29 | dlynes_office | robl^: good luck on the 9133i's |
| 23:54.50 | dlynes_office | robl^: every supplier i've talked to is out of stock, and aastra's not shipping any new ones until mid august |
| 23:54.57 | dlynes_office | robl^: they're having a supply chain problem |
| 23:55.05 | file | agboris: I'm not going that far... |
| 23:55.06 | robl^ | dlynes_office: ohhh |
| 23:55.24 | robl^ | I haven't tried to order any yet. I just knew they were back orderd |
| 23:55.29 | agboris | <file> just give me clue... |
| 23:55.35 | dlynes_office | robl^: every supplier i've talked to in Canada doesn't have any, including Aastra |
| 23:55.36 | robl^ | new ones are about to be released? |
| 23:55.50 | dlynes_office | robl^: I ended up having to snag a couple of old demo units from Sayson Technologies |
| 23:56.11 | dlynes_office | robl^: don't think so...I think it's just a supply chain issue |
| 23:56.29 | file | agboris: clue for what? I'm all for helping people but if you don't know what to say to your provider to get help from them... that's not good |
| 23:56.53 | robl^ | dlynes_office: well I am not in a huge hurry at the moment.. but I will say is that the 480i has a great full duplex speaker |
| 23:57.03 | dlynes_office | robl^: so does the 9133i |
| 23:57.28 | dlynes_office | robl^: i haven't tried the 480i, 480iCT, or the 9112i yet |
| 23:57.30 | agboris | <file> thanks for ur help .... |
| 23:57.31 | dlynes_office | robl^: only the 9133i |
| 23:57.38 | dlynes_office | robl^: i've got 23 of them deployed |
| 23:57.52 | robl^ | dlynes_office: nice!! how did you desi them? |
| 23:57.56 | file | agboris: "I'm registered to you but I can't receive calls" how is that? |
| 23:57.57 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 23:58.00 | dlynes_office | robl^: desi? |
| 23:58.11 | dlynes_office | robl^: desi's a slang term for an East Indian |
| 23:58.30 | robl^ | dlynes_office: I noticed that the 9133i requre a special card for button labels.. with holes for the lights |
| 23:58.51 | robl^ | dlynes_office: desi - it's a button label |
| 23:59.03 | dlynes_office | robl^: yeah...it comes with the clear plastic window and the cards |
| 23:59.20 | dlynes_office | robl^: it's almost the same as the Nortel 7324's and the like |
| 23:59.37 | dlynes_office | erm 7208's I mean |