| 00:00.01 | fujin | see everything* else* |
| 00:01.40 | [TK]D-Fender | fujin, No way. |
| 00:03.47 | fujin | bugger. That's a shame. |
| 00:05.05 | [hC] | cesar_CR: hey :) i am kinda from costa rica too |
| 00:05.33 | cesar_CR | [hC], well hello !!! from where ??? :) |
| 00:05.52 | [hC] | cesar_CR: my wife is from costa rica, i lived there for a few years and now i am back in vancouver in canada.. she lives in san pedro |
| 00:05.56 | [hC] | cesar_CR: do you work for ICE? |
| 00:06.13 | cesar_CR | [hC], yes ... |
| 00:06.42 | [hC] | cesar_CR: cool... I have always wanted a contact at ICE :) are you guys using asterisk? |
| 00:07.37 | cesar_CR | in prod envirement no... |
| 00:08.35 | [hC] | cesar_CR: what are you testing it for? voicemail, or business services? |
| 00:08.44 | cesar_CR | but I am traying to show everybody the benef. of *, I think I'll start in my office |
| 00:08.51 | [hC] | cesar_CR: i have some asterisk installations in costa rica |
| 00:09.13 | [hC] | cesar_CR: i had to patch zaptel to recognize the tones used there, but after that everything worked |
| 00:09.30 | cesar_CR | really ??? good to know... |
| 00:09.59 | cesar_CR | wich kind problems ? |
| 00:10.03 | [hC] | cesar_CR: if you need any help i would be happy to help you if i can |
| 00:10.33 | [hC] | costa rica just wasnt listed in the tonezone.c file for using analog lines... it would overlay a north american dial tone over top of the dialtone from the CO |
| 00:11.07 | cesar_CR | well, this year I will start my first implemtation... |
| 00:11.19 | [hC] | cesar_CR: what do you do for ICE normally? |
| 00:12.35 | *** join/#asterisk donzx (n=donzx@S0106006097940f68.vw.shawcable.net) |
| 00:13.20 | donzx | does anyone know how hard would it be to play a message then capture the DTMF tones ( digits pressed ) and pass these onto external application ? |
| 00:14.55 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 00:15.11 | [TK]D-Fender | donzx, "show application read" |
| 00:16.22 | *** join/#asterisk jsb- (n=jsb@neptune.uqtr.ca) |
| 00:19.34 | donzx | [TK]D-Fender, once I use cmd Read to store DTMF input into variable, how do I pass this onto application which for example could store this in database ? |
| 00:19.57 | fujin | use the MYSQL() application, or the built in Set DB |
| 00:19.58 | [TK]D-Fender | donzx, "show application system" |
| 00:20.01 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1096745492.dsl.bell.ca) |
| 00:20.16 | [TK]D-Fender | donzx, Or use any of the other DB related dialplan applications. |
| 00:20.27 | Qwell | HP_Owner...heh |
| 00:20.34 | Qwell | nkoth: had your box serviced recently, have we? |
| 00:20.36 | [TK]D-Fender | donzx, You should actually stop and read all of *'s applications.... |
| 00:20.48 | jsb- | or he never changed the default preload |
| 00:20.52 | Qwell | it's a postload |
| 00:20.56 | jsb- | is it |
| 00:20.56 | Qwell | service load |
| 00:20.58 | jsb- | heh |
| 00:20.59 | Qwell | yep |
| 00:21.10 | jsb- | hp sucks then :) |
| 00:21.14 | Qwell | that's what they put on when they replace your drives, or whatever |
| 00:21.17 | Qwell | yeah, you're telling me |
| 00:21.21 | jsb- | well, their consumer products do |
| 00:21.34 | jsb- | their enterprise line is quite nice - but i'd rather use ibm |
| 00:21.40 | Qwell | Entry date / time: 12/03/2007 9:20am PT |
| 00:21.41 | jsb- | not sure about lenovo yet |
| 00:21.42 | Qwell | Expected Delivery Date: 01/14/2008 |
| 00:21.46 | Qwell | ...freaking HP |
| 00:22.02 | jsb- | i hate meridian mail |
| 00:22.08 | jsb- | takes freakin forever to restore |
| 00:22.10 | Qwell | welcome to the club |
| 00:23.10 | jsb- | but i like my m2616 :( |
| 00:23.32 | jsb- | Qwell: what'd you order |
| 00:23.44 | Qwell | I didn't order anything. that's a replacement part |
| 00:23.49 | *** join/#asterisk crmcheese (n=brie@CPE0015e969c5ed-CM0014048e9604.cpe.net.cable.rogers.com) |
| 00:23.53 | crmcheese | hello everyone |
| 00:23.54 | jsb- | wow, that long for a freakin fru |
| 00:23.58 | Qwell | a $15 remote control, that "due to the holidays" is out of stock |
| 00:24.12 | Qwell | jsb-: that's nothing - I've been dealing with this crap since October |
| 00:24.19 | jsb- | buy ibm or dell |
| 00:24.21 | jsb- | :) |
| 00:24.32 | Juerd_ | My MOH doesn't work. I see mpg123 when I use "ps ax", but I hear nothing on the phone. I'm using Dial's m |
| 00:24.32 | fujin | dell ftw |
| 00:24.39 | Juerd_ | Any hints? |
| 00:24.41 | fujin | Juerd_: use native mp3 |
| 00:24.43 | Qwell | No, I don't buy Dell :p |
| 00:24.44 | fujin | from asterisk-addons |
| 00:24.55 | Juerd_ | fujin: Tried that first. Also, no sound. |
| 00:25.10 | jblack | And all was well with the world. |
| 00:25.18 | jsb- | Juerd_: the sound of silence.mp3 is not a good moh file |
| 00:25.31 | fujin | no sound with native mp3? |
| 00:25.32 | Juerd_ | jsb-: I use a different file :) |
| 00:25.34 | fujin | you've got issues dude ;> |
| 00:25.46 | jsb- | maybe you've got DRM on that mp3 |
| 00:25.54 | Juerd_ | No, I just made the mp3 myself |
| 00:25.58 | Juerd_ | Exported it from Audacity |
| 00:26.11 | jsb- | i'm obviously kidding |
| 00:26.50 | crmcheese | Does anyone here use AsteriskNow? |
| 00:27.00 | jsb- | ~asterisknow |
| 00:27.00 | jbot | somebody said asterisknow was based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 00:27.01 | Qwell | crmcheese: your best bet would be #asterisknow |
| 00:27.24 | crmcheese | ah thanks |
| 00:27.40 | jsb- | be a man, code your own dial plan |
| 00:27.42 | crmcheese | I haven't installed it yet and am open to suggestions. What are the drawbacks to using asterisknow? |
| 00:27.43 | jsb- | heh |
| 00:27.45 | jsb- | :) |
| 00:27.52 | jsb- | you learn less |
| 00:28.16 | jsb- | found a few bugs last time i played with it... |
| 00:28.33 | crmcheese | How much knowledge would one have to have to use either Asterisk or ..Now effectively? I'm not a programmer |
| 00:28.44 | Juerd_ | Any further hints on the silent moh? |
| 00:28.48 | Qwell | crmcheese: a little common sense goes a long way :) |
| 00:28.53 | Qwell | if you've got that, you're golden |
| 00:29.04 | Qwell | (and a bit of patience helps too...) |
| 00:29.26 | jsb- | Juerd_: see anything pertinent in console? |
| 00:29.27 | crmcheese | Patience and stubborness I have in droves... lol |
| 00:29.36 | CrazyTux | Patience is virtue :) |
| 00:29.40 | Qwell | crmcheese: it's just like learning any other program. there's a bit of a learning curve |
| 00:30.07 | Juerd_ | jsb-: No : |
| 00:30.09 | Juerd_ | :| |
| 00:30.50 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-075-182-081-167.nc.res.rr.com) |
| 00:31.22 | jsb- | Juerd_, create an extenstion with MusicOnHold() |
| 00:31.26 | jsb- | see if that works |
| 00:31.33 | jsb- | e.g. |
| 00:31.34 | jsb- | <PROTECTED> |
| 00:31.34 | jsb- | <PROTECTED> |
| 00:32.06 | jsb- | hmm. meridian mail hard drive is making funny noises |
| 00:32.16 | Juerd_ | An application's install script just borked my config |
| 00:32.28 | Juerd_ | destar apparently wants to control everything, not caring about what I did manually |
| 00:32.52 | Juerd_ | Ah, it made backups. |
| 00:32.52 | jsb- | of course |
| 00:33.11 | jsb- | i was going to ask what was destar, but i figured it was a gui |
| 00:33.18 | Juerd_ | It is :) |
| 00:33.53 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 00:33.56 | Juerd_ | restored |
| 00:33.59 | Juerd_ | Now going to try your idea |
| 00:34.59 | RyanW | I'm getting SIP/2.0 484 Address Incomplete when trying to place a call using com_zimbra_asterisk in Zimbra, any ideas? |
| 00:34.59 | Juerd_ | Hm, yes, that does work. |
| 00:35.13 | Juerd_ | jsb-: MusicOnHold does give me music. But Dial(,,m) apparently doesn't. |
| 00:36.01 | jsb- | ok, so that part works |
| 00:36.07 | Qwell | ~book |
| 00:36.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:36.10 | Qwell | crmcheese: ^^ |
| 00:36.12 | jsb- | post your Dial() string |
| 00:36.26 | Qwell | download that - there is no better place to start |
| 00:36.34 | Juerd_ | jsb-: exten => xs4all,n,Dial(IAX2/testphone,,m) |
| 00:37.08 | jsb- | what are you hearing when you call that extension? |
| 00:37.17 | [TK]D-Fender | Juerd_, pastebin the entire extension. |
| 00:37.30 | jsb- | yeah, what he said |
| 00:37.35 | Juerd_ | jsb-: Nothing, while it's ringing |
| 00:37.47 | [TK]D-Fender | Juerd_, better yet... pastebin the CLI output of the entire failed call. |
| 00:37.52 | [TK]D-Fender | ~pb |
| 00:37.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:37.54 | [TK]D-Fender | ^^^^^^^^^^^ |
| 00:37.55 | jsb- | try adding a timeout value just for kicks |
| 00:38.03 | [TK]D-Fender | Juerd_, along with your musiconhold.conf |
| 00:38.17 | Juerd_ | [TK]D-Fender: There's supposed to be CLI output? |
| 00:38.35 | [TK]D-Fender | Juerd_, when a call is being processed? Yes. clearly... |
| 00:38.43 | Juerd_ | TJNII: I get none |
| 00:38.50 | Juerd_ | It just stays at hostname*CLI> |
| 00:39.01 | TJNII | Huh? |
| 00:39.10 | [TK]D-Fender | Juerd_, "set verbose 10" |
| 00:39.21 | Juerd_ | http://tnx.nl/6878XSXD is the paste by the way |
| 00:39.34 | Juerd_ | <PROTECTED> |
| 00:39.38 | [TK]D-Fender | Juerd_, your 1st line is commented out. |
| 00:39.47 | [TK]D-Fender | Juerd_, your exten is broken |
| 00:39.48 | Juerd_ | [TK]D-Fender: Yes, but it didn't work with the Answer either |
| 00:40.00 | Juerd_ | What's broken? :) |
| 00:40.15 | Juerd_ | Oh, no priority 1, right? |
| 00:40.16 | [TK]D-Fender | Juerd_, you can't just start an exten with an "n" priority. |
| 00:40.31 | Juerd_ | Heh, it did ring so I hadnt noticed it. Thanks. |
| 00:40.31 | [TK]D-Fender | Juerd_, Correct... no CLI = Where is my call?! |
| 00:40.32 | Juerd_ | Let's see |
| 00:40.41 | [TK]D-Fender | Juerd_, get your act together. |
| 00:40.55 | Juerd_ | [TK]D-Fender: Oh, I know why it kept working. Forgot to reload after commenting. |
| 00:41.13 | Juerd_ | But while we're at this, should I answer the call before Dial()ing? |
| 00:41.16 | [TK]D-Fender | Juerd_, please pastebin the entire failed attempt, and the configs |
| 00:41.28 | [TK]D-Fender | Juerd_, also another sample of MoH working elsewhere. |
| 00:41.38 | [TK]D-Fender | Juerd_, Yes |
| 00:42.06 | Juerd_ | jsb-: Adding timeout didn't change the situation, by the way. |
| 00:43.00 | *** part/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 00:43.24 | jsb- | just a guess |
| 00:45.01 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 00:45.36 | Juerd_ | http://tnx.nl/6879PDXE # console output for a call |
| 00:48.46 | [TK]D-Fender | Juerd_, And the rest...... |
| 00:49.22 | Juerd_ | [TK]D-Fender: I'll paste one with the entire timeout no pick/hangup |
| 00:49.58 | Juerd_ | [TK]D-Fender: http://tnx.nl/6880OXFW |
| 00:50.18 | Juerd_ | jblack: What did you do? |
| 00:50.28 | jblack | Got the spa8k to work reliably. |
| 00:50.30 | [TK]D-Fender | Juerd_, and the working sample & your configs... |
| 00:50.38 | [TK]D-Fender | jblack, excellent :) |
| 00:50.39 | Juerd_ | [TK]D-Fender: What do you mean by working sample? |
| 00:50.49 | Juerd_ | Whoa, one of our servers tries to skinny-connect. |
| 00:51.07 | jsb- | jblack: nice piece of eqpt |
| 00:51.08 | [TK]D-Fender | Juerd_, You said you have MoH working elsewhere in your setup. I want to see that CLI output. And your musiconhold.conf |
| 00:51.10 | jblack | All I need to figure out now is how to use the spa * codes for transferring. |
| 00:51.20 | jsb- | try *98 (no joke) |
| 00:51.22 | Juerd_ | [TK]D-Fender: Ah, okay. That's an extension I dial. |
| 00:51.25 | Juerd_ | [TK]D-Fender: Hold a sec. |
| 00:51.26 | jblack | I did. |
| 00:51.42 | jsb- | does it configure like a pap2? |
| 00:51.47 | Juerd_ | Oh, heh, the "skinny" connect was a portscan. |
| 00:51.53 | jblack | No idea. I do't have a pap2 |
| 00:52.10 | jblack | It does say *98 is the code, and it is turned on for line 1 |
| 00:52.17 | [TK]D-Fender | jblack, transfer = [flash] Dial a # (ringing) [flash] |
| 00:52.38 | [TK]D-Fender | jblack, Blind transfer = [flash] Dial a # (ringing) [flash] |
| 00:52.48 | ZX81_ | hi all, just had a customer call complaining that the first word of a sentence was chopped off when he got a call transferred - I'm assuming its the rtp stream taking time to set up - any ideas? |
| 00:52.52 | [TK]D-Fender | jblack, Attended transfer = [flash] Dial a # (ringing) (answer) (talk) [flash] |
| 00:52.56 | [TK]D-Fender | jblack, typically |
| 00:53.05 | Juerd_ | [TK]D-Fender: http://tnx.nl/6881VGCD |
| 00:53.25 | jblack | On this, call #1, flash, call #2, flash performs 3-way calling |
| 00:53.42 | [TK]D-Fender | jblack, Sorry.. |
| 00:53.53 | [TK]D-Fender | jblack, Attended transfer = [flash] Dial a # (ringing) (answer) (talk) [hangup] |
| 00:54.01 | jblack | I'll try that. |
| 00:54.14 | jsb- | Blind transfer = [flash] *98 number (doot doot) [dialtone] |
| 00:54.26 | jsb- | that's what all my (default) spa's do |
| 00:54.36 | jblack | another thing I haven't figured out yet is why there's a 15 second delay before calls start ringing through |
| 00:54.36 | jsb- | *98 should give you a weird dialtone |
| 00:54.45 | jsb- | dialplan? |
| 00:55.09 | [TK]D-Fender | jsb-, wasn't *98 to my memory.. thats a CLASS reserved code |
| 00:55.17 | jsb- | i know |
| 00:55.25 | [TK]D-Fender | jsb-, but yes there was typically a * code for it. |
| 00:55.28 | jsb- | i just logged in to a spa2000 |
| 00:55.36 | jsb- | and *98's the code for blind transfer |
| 00:55.39 | jblack | Yup, there's transfer. |
| 00:55.40 | jsb- | as stupid as it may sound |
| 00:55.50 | [TK]D-Fender | jblack, you should have a config page specifying what is used for it as you can override it as well |
| 00:55.50 | jsb- | it totally breaks class |
| 00:55.51 | jblack | flash, second, answer, hangup does great. |
| 00:56.01 | jsb- | oh, that's even better |
| 00:56.15 | [TK]D-Fender | jblack, if you hang up during ringing I believe it goes blind |
| 00:56.23 | [TK]D-Fender | jblack, test that while you're at it |
| 00:56.24 | *** join/#asterisk __freedom__lover (n=eduardo@201-26-102-134.dsl.telesp.net.br) |
| 00:56.36 | jsb- | don't you have admin/advanced mode in web config? |
| 00:56.51 | [TK]D-Fender | jblack, and don't test codes blindly. the lowere level ATA's have a config page that spells it all out for you |
| 00:57.07 | [TK]D-Fender | Juerd_, And that sample works? |
| 00:57.13 | jblack | jsb-: Yeah. I'm looking at it right now. Out of the ..~30 codes there, I could care less about 30 of them. |
| 00:57.33 | jsb- | most if not all are useless |
| 00:57.38 | jsb- | but can be disabled in the lines page |
| 00:57.38 | __freedom__lover | hi all, i'm running asterisk1.2.13 on a freebsd 6.1 with realtime configuration to extensions and sippeers |
| 00:57.41 | jblack | [TK]D-Fender: That's where I'm getting the codes from. I won't want to accidentally instruct the spa to reboot itself. |
| 00:57.41 | Juerd_ | [TK]D-Fender: Yes |
| 00:57.55 | [TK]D-Fender | jblack, well... if you could care "less" about them then you care at least a little! a start! |
| 00:57.57 | Juerd_ | [TK]D-Fender: It plays nice music, continuously, until hung up |
| 00:58.06 | jblack | So the only features.conf item I need to retain is automon. Woot. |
| 00:58.09 | *** join/#asterisk aut (n=aut@modemcable086.237-80-70.mc.videotron.ca) |
| 00:58.24 | [TK]D-Fender | Juerd_, and you only hear regular ringing on your other dial? you let it go for at least 10s? |
| 00:58.28 | aut | where does asterisk store peer information? even after i restart asterisk, it's showing an old ip address for one of the sip peers |
| 00:58.28 | __freedom__lover | when i change a peer from clear password (secret) to encrypted (md5secret), my peer don't connect anymore |
| 00:58.39 | jblack | Now, just to figure why there's a 15 second delay between finishing dialing the number, and the number being dialed. |
| 00:58.41 | __freedom__lover | even i reboot my ata |
| 00:58.43 | Juerd_ | [TK]D-Fender: No ringing or any sound, whatsoever. I let it go for 10 seconds and it timed out correctly. |
| 00:58.45 | [TK]D-Fender | jblack, thereby removing "tT" from the picture leaving "w" |
| 00:58.56 | jblack | exactly correct. |
| 00:58.58 | jsb- | jblack: how much did you pay for that? |
| 00:59.01 | [TK]D-Fender | Juerd_, well you didn't have a timeout before. |
| 00:59.02 | Juerd_ | [TK]D-Fender: The softphone that I'm dialing does ring |
| 00:59.07 | jsb- | jblack: and can you post config screenshots? |
| 00:59.09 | Juerd_ | [TK]D-Fender: Correct. I added it because jsb- suggested it. |
| 00:59.12 | __freedom__lover | i have to reload the sip module to it connect.. |
| 00:59.18 | jblack | I blew $250 on the spa8k, and another $200 on phones. |
| 00:59.23 | jsb- | what did i do? |
| 00:59.28 | jblack | Stickin' it to the man can be an expensive proposition. |
| 00:59.30 | [TK]D-Fender | Juerd_, tell you what... put a Wait(5) between your dial and answer... or heck, a PLAYBACK of something.... |
| 00:59.35 | __freedom__lover | someone does know why? |
| 00:59.38 | jsb- | jblack: what did you replace |
| 00:59.45 | [TK]D-Fender | jblack, expensive phones... |
| 00:59.53 | [TK]D-Fender | jblack, how many? |
| 00:59.58 | Juerd_ | [TK]D-Fender: I'll try, hold on |
| 01:00.01 | [TK]D-Fender | jblack, maybe NOT expensive depending.. |
| 01:00.02 | jblack | since I started this adventure? a skype 802.11g phone. |
| 01:00.12 | [TK]D-Fender | jblack, RMA it :) |
| 01:00.25 | jblack | I got a dect6 panasonic with 3 handsets, and a dect6 ge with 2 handsets. |
| 01:00.33 | [TK]D-Fender | jblack, and since you're becoming happy with the SPA... I wouldn't call that "blown" |
| 01:00.55 | [TK]D-Fender | jblack, if the DECT base only has 1 port then its a dead-end... |
| 01:01.01 | Juerd_ | [TK]D-Fender: If I add a playback in front, the music works. |
| 01:01.03 | jblack | [TK]D-Fender: Trust me buddy. My bank account is blown like a tire. |
| 01:01.07 | [TK]D-Fender | jblack, multi handset "home systems" = BS |
| 01:01.23 | Juerd_ | [TK]D-Fender: If I put a Wait in, it doesn't |
| 01:01.24 | jblack | I didn't get them for "home systems" reasons. That's nonsense. |
| 01:01.27 | [TK]D-Fender | jblack, Well blown budget is a little different than a wasted budget. |
| 01:01.29 | jblack | (i.e. I agree). |
| 01:01.47 | [TK]D-Fender | jblack, then you have my condolences... now pawn them off as close to top dollar as you can find. |
| 01:01.55 | jblack | I did that so that I can still get calls when both of my machines in the office are suspended, and also get calls while in the bedroom. |
| 01:02.02 | jblack | I saved myself a lot of misery by not running cable. |
| 01:02.04 | [TK]D-Fender | Juerd_, Well you SHOULD see * answer... then wait 5 sec. |
| 01:02.18 | *** part/#asterisk aut (n=aut@modemcable086.237-80-70.mc.videotron.ca) |
| 01:02.25 | Juerd_ | [TK]D-Fender: It does |
| 01:02.41 | jsb- | bah, i wanted ip phones in the other rooms |
| 01:02.43 | jsb- | without cabling |
| 01:02.48 | jblack | Here's my big lesson, that I learned today. * is _crack_ |
| 01:02.49 | jsb- | apple WDS ftw |
| 01:03.08 | jblack | It starts off with a soft phone. It's such a rush. So, you do another soft phone, and another. |
| 01:03.16 | jblack | before long, you have to mainline hard phones to get your fix. |
| 01:03.26 | [TK]D-Fender | jblack, Hey... I bought 3 Polycom phones for HOME. Thats about 600$ CDN after taxes, etc I've dropped on mine. |
| 01:03.41 | jblack | I believe it. I'll probably get real phones some day too. |
| 01:03.46 | jsb- | i have 2 7960g's, a 7940g and a mitel 5055 |
| 01:03.50 | [TK]D-Fender | jblack, I took a loss on an SPA-942, SPA-3000, SPA-2000 as well |
| 01:03.50 | jsb- | 1 pap2 and 2 spa2k's |
| 01:03.53 | jsb- | all for home use :( |
| 01:03.58 | jsb- | that's besides the meridian 1 |
| 01:04.17 | [TK]D-Fender | jblack, let me say unless you're getting a ton of calls.. there's no point. |
| 01:04.31 | [TK]D-Fender | jblack, I could be perfectly happy with a cordless analog phone + ATA |
| 01:04.47 | jblack | I get very few calls, and I call very few people. |
| 01:05.05 | [TK]D-Fender | jblack, then you're set with your SPA-8000 |
| 01:05.06 | jsb- | but you can't beat picking your outbound CLID using the phone's line kyes |
| 01:05.07 | jsb- | keys |
| 01:05.26 | [TK]D-Fender | jsb-, assuming you even have the option going outbound |
| 01:05.51 | jblack | Oh yeah, the flexibility is staggering. |
| 01:06.03 | jsb- | if you're going to play, you have to go for the whole nine yards |
| 01:06.40 | jblack | I'd say going from 1 phone in the house to... 9 in two weeks, that's not insignificant |
| 01:07.00 | [TK]D-Fender | jblack, You're right... ALMOST overkill.... time for a PRI next :p |
| 01:07.27 | jblack | I need more. Those empty ports on the 8k look so lonely. |
| 01:07.32 | [TK]D-Fender | Juerd_, so even with 10s wait and a playback it only rings? |
| 01:07.46 | Juerd_ | [TK]D-Fender: It stays silent on the SIP side, it rings the softphone |
| 01:07.52 | jsb- | jblack: do you like the vintage look? |
| 01:08.05 | Juerd_ | [TK]D-Fender: But with a playback(tt-weasels) instead of the wait, it plays both the weasels file and the MOH perfectly. |
| 01:08.10 | jblack | I know. I _hate_ my neighbor. Maybe I can get a discarded payphone, and offer free phone service facing his house. Put a big sign on it that says "Drug dealers call free!" |
| 01:08.37 | [TK]D-Fender | Juerd_, So do this : "Playback(silence/1) |
| 01:08.54 | [TK]D-Fender | Juerd_, I think you need solid audio even if "null" |
| 01:08.57 | mhiku | what linux distro is common in using as an asterisk server? |
| 01:09.03 | [TK]D-Fender | Juerd_, I THINK that might do it for you. |
| 01:09.32 | jsb- | jblack: hook a 1a2 to that bad boy |
| 01:09.35 | [TK]D-Fender | mhiku, CentOS, Debian are the 2 biggest, followed by Gentoo, FC, Slackware, etc |
| 01:09.55 | [TK]D-Fender | jblack, is the unit nice to handle? |
| 01:10.01 | Juerd_ | [TK]D-Fender: That did it! :) |
| 01:10.03 | Juerd_ | [TK]D-Fender: Thanks! |
| 01:10.10 | [TK]D-Fender | Juerd_, You're welcome :) |
| 01:10.20 | Juerd_ | [TK]D-Fender: Any idea why it wouldn't work without? |
| 01:10.21 | mhiku | oh, ubuntu isnt mentioned lol, ok thnx |
| 01:10.52 | mhiku | centos ftw |
| 01:10.56 | [TK]D-Fender | mhiku, yes, your GUI-Happy craze has not gone unnoticed. Ubuntu is more complex to get * set up on. |
| 01:11.32 | jblack | [TK]D-Fender: The phones? They're reasonable. A bit busy in the keypads... |
| 01:11.32 | jsb- | why would one want to run * on ubuntu |
| 01:11.51 | [TK]D-Fender | jblack, I meant the SPA-8000 itself. |
| 01:12.42 | [TK]D-Fender | jsb-, because the server itself would be "X" by default and give some glimmer of hope that it'll be completely manageable without CLI intervention. |
| 01:12.51 | jsb- | bah |
| 01:12.55 | jblack | I found setting it up awkward. The machine gets sluggish while registration, which is itself sluggish. |
| 01:13.18 | jblack | between that and the 15 second delays before calls are placed made debugging painful. |
| 01:13.20 | jsb- | anyone got any experience using spa-941/942/962 phones? |
| 01:13.33 | alrs | jsb- I don't like those phones |
| 01:13.49 | alrs | jsb- echo-prone |
| 01:13.50 | jblack | Once I got much more patience, and started blowing off things I didn't recognize, and stopped trying to requier auth, it became easy to manage. |
| 01:14.04 | jblack | I like it much more, now that it works |
| 01:14.04 | jsb- | good to know |
| 01:14.12 | jsb- | i guess i'll stick with cisco |
| 01:14.20 | jsb- | how about the new 3911 sip phones |
| 01:14.38 | alrs | jsb- Have you had good luck with Cisco? I'm doing everything Polycom at the moment. |
| 01:14.48 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 01:14.48 | jsb- | excellent luck |
| 01:15.04 | ZX81_ | hi all, just had a customer call complaining that the first word of a sentence was chopped off when he got a call transferred - I'm assuming its the rtp stream taking time to set up - any ideas? |
| 01:15.11 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ec51c941565181c6) |
| 01:15.11 | jblack | I think with familiarity, it's a tool that I can really enjoy. It has about 13 million options. So, overwhelming for the first 3 hours. After that, not such a problem. |
| 01:15.32 | jsb- | alrs: mitel is fairly good too - but i like Cisco more |
| 01:15.35 | [TK]D-Fender | jblack, the dial delay is entirely configurable |
| 01:15.46 | jblack | I figure. It's just going through the mountain of options |
| 01:15.53 | [TK]D-Fender | jblack, soon you'll be learning how to handle its dialplan. |
| 01:15.58 | alrs | jsb- I've found that Cisco stuff is really fussy and moving firmware versions around is a pain |
| 01:16.10 | ZX81_ | FXO == TDM400P, EC == HPEC, SIP == Grandstream 8 port FXS, Reception == Snom 320 |
| 01:16.16 | jsb- | alrs: agreed, but once they're stable they're great |
| 01:16.31 | jsb- | they're picky but i'm used to it - the phones operate flawlessly |
| 01:16.33 | alrs | jsb- what keeps you away from Polycom? |
| 01:16.41 | [TK]D-Fender | ZX81_, Who is the end picking up the call? |
| 01:16.41 | jblack | That's what * is for. :) |
| 01:16.44 | jsb- | frankly, never tried em |
| 01:16.48 | ZX81_ | :) Adrian |
| 01:16.59 | ZX81_ | the grandstream gateway |
| 01:17.00 | [TK]D-Fender | jsb-, Polycom kills pretty much everything else. |
| 01:17.11 | alrs | jsb- even at the low end |
| 01:17.12 | [TK]D-Fender | ZX81_, That says it all... |
| 01:17.19 | ZX81_ | call comes in via fxo with answer(1000), then goes to reception (no problems), then to gxw |
| 01:17.24 | ZX81_ | [TK]D-Fender, really? |
| 01:17.25 | jsb- | i'll take a look at the newer 300 series |
| 01:17.26 | [TK]D-Fender | ~gs |
| 01:17.27 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:17.28 | [TK]D-Fender | ^^^^^^^^^ |
| 01:17.33 | ZX81_ | lol |
| 01:17.43 | jsb- | yes, i hate grandstream and similar crap |
| 01:17.45 | ZX81_ | sucky |
| 01:17.47 | alrs | jsb- I've got a 320 here that sounds great, even on speaker phone |
| 01:17.58 | alrs | jsb- really good for a ~$100 phone |
| 01:18.00 | jsb- | alrs: they've got stellar speakerphones, i'll give you that |
| 01:18.10 | ZX81_ | they've been fine (except for 5 or 6 that the firmware didn't take on) |
| 01:18.12 | jsb- | but i have been very impressed by mitel's design and audio |
| 01:18.18 | jblack | man this thing is configurable |
| 01:18.18 | ZX81_ | have deployed quite a few |
| 01:18.19 | jsb- | my old beat-up 5055 is very hard to beat |
| 01:18.24 | ZX81_ | moved now to linksys |
| 01:18.26 | jsb- | jblack: just like a pap2 |
| 01:18.30 | [TK]D-Fender | Polycom = speakerphone kings |
| 01:18.39 | ZX81_ | heh obviously |
| 01:18.39 | jsb- | [TK]D-Fender: obviously |
| 01:18.43 | jsb- | haha |
| 01:18.47 | ZX81_ | at $100000000000000000000000 anyone could be |
| 01:18.47 | jblack | There's even a "More Echo Supression" option. |
| 01:18.56 | [TK]D-Fender | jsb-, For basic stuff, almost all the Linksys ATA's are pretty much the same. |
| 01:19.07 | jblack | That should come with an <img src> of a dial with 11 on it |
| 01:19.21 | [TK]D-Fender | ZX81_, Actually in North America, Polycom is less expensive than Snom. |
| 01:19.26 | ZX81_ | no way |
| 01:19.29 | ZX81_ | crazy |
| 01:19.32 | [TK]D-Fender | ZX81_, Very way. |
| 01:19.39 | ZX81_ | ok I'm buying from stateside then |
| 01:19.52 | [TK]D-Fender | ZX81_, where are you located? |
| 01:19.59 | ZX81_ | :) New Zealand - Matt Riddell |
| 01:20.00 | ZX81_ | :) |
| 01:20.01 | jsb- | i ain't trusting snom either |
| 01:20.22 | [TK]D-Fender | ZX81_, FAR.... shipping would suck, but if you bought enough it might eb worth it. |
| 01:20.29 | mosty | [TK]D-Fender, is there a way to get polycom phones to use ldap for a directory service? |
| 01:20.35 | ZX81_ | yeah just have to do bulk |
| 01:20.37 | *** join/#asterisk Morrocco (n=ivan@189.164.93.68) |
| 01:20.38 | [TK]D-Fender | mosty, Microbrowser <- |
| 01:20.52 | jblack | Yeah. I don't see where to get rid of that 15 sec delay. That's ok. It's not a dealbreaker |
| 01:21.00 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 01:21.17 | [TK]D-Fender | jblack, You'll figure it out... its all about the MGCP dialplan and the 2 timeouts |
| 01:21.17 | mosty | [TK]D-Fender, so i'd have to write a web page that presents the contents of my ldap server? |
| 01:21.26 | [TK]D-Fender | mosty, yup |
| 01:21.37 | mosty | hmm ok |
| 01:21.38 | [TK]D-Fender | mosty, I've done it for non LDAP |
| 01:21.47 | jblack | Ahhh. Here we go. |
| 01:22.26 | mosty | yeah, i've seen walkthroughs for non-ldap stuff, but i would like this to integrate with some of the other non-polycom phones that support ldap |
| 01:22.33 | [TK]D-Fender | jblack, Trust me, you'll learn to like it before long. Little bits will come to you at a good and steady pace. |
| 01:22.49 | jblack | Yeah. I think I'm going to like this a lot. |
| 01:23.02 | [TK]D-Fender | mosty, well if you can code it the MB is the way. The internal directory is phone-local only |
| 01:23.19 | jblack | well, I'm not seeing anything that screams dialplan at me, though |
| 01:23.46 | mosty | [TK]D-Fender, i figured, the internal directory isn't really centralised |
| 01:23.53 | jsb- | jblack, do you have any tabs at the upper section of the screen? |
| 01:23.57 | Juerd_ | What does "bridged" mean? |
| 01:24.00 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
| 01:24.02 | [TK]D-Fender | jblack, it'll look something like "(xxxxxxxxxT|*XX|1XXXXXXXX)" and so on |
| 01:24.02 | jblack | I do. Router, and Voice. That's it |
| 01:24.16 | jsb- | hmm |
| 01:24.34 | jsb- | no sub-tabs, or submenus (line 1 through 8) |
| 01:25.19 | jblack | Under router, there's status (read only), wan setup (rather configurable), lan setup (not configurable at all (!)), application, which contains typical port forwarding management |
| 01:25.34 | jsb- | hmm |
| 01:25.42 | jsb- | try http://ip.address/admin/advanced |
| 01:25.55 | jsb- | or just /admin |
| 01:26.24 | ZX81_ | Juerd_, connected |
| 01:27.00 | jblack | Voice has info (ro), System (remote syslog and administrative passwords), sip (with and incredible number of options), provisioning, regional, and L1-L8 |
| 01:27.01 | Juerd_ | Ah, makes sense :) |
| 01:27.12 | Juerd_ | I sometimes wonder why this world has its own words :) |
| 01:27.15 | jblack | admin/advanced drops into router, and /admin is a 404 |
| 01:27.28 | jsb- | go to L1 |
| 01:27.40 | jsb- | bottom of page should give you a dialplan string |
| 01:27.52 | jblack | Oh, there's the dialplan. |
| 01:27.54 | jsb- | btw, screenshots would be appreciated :) |
| 01:27.55 | jblack | It's a single string. |
| 01:27.58 | jsb- | yep |
| 01:28.59 | [TK]D-Fender | jblack, Should be a string/port |
| 01:29.32 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 01:32.42 | jblack | Hmmm. |
| 01:37.17 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 01:40.01 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:44.11 | mosty | i'm having trouble with BLF in asterisk 1.4 (works fine with 1.2). i've added limitonpeers=yes and call-limit=999 in sip.conf, is there anything else i might need to change? |
| 01:46.28 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 01:46.39 | donzx | what is exten,s in asterisk ? |
| 01:46.55 | keith4_ | s for start |
| 01:47.08 | donzx | ah is that at begining of call keith4_ ? |
| 01:47.25 | keith4_ | you start at s when you don't have any input |
| 01:50.46 | mosty | with asterisk 1.4 when i do "show hints" in the console, it says all my registered sip clients are idle, but i can see that there are active bridged sip channels. is this likely to be a bug? |
| 01:52.46 | *** join/#asterisk centrex (n=centrex@c-71-207-165-154.hsd1.al.comcast.net) |
| 01:55.13 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 01:55.21 | *** join/#asterisk skopii (n=skopii@69.31.131.51) |
| 01:57.14 | [TK]D-Fender | mosty, limitonpeers=yes <- remove, and make sure you are using "type=peer" |
| 01:57.30 | [TK]D-Fender | donzx, This is all explained in chapter 5 of THE BOOK. |
| 01:57.31 | [TK]D-Fender | ~book |
| 01:57.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:58.06 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 01:58.36 | mosty | i'm using type=friend, and the wiki says that you should have limitonpeers=yes in this case |
| 01:58.55 | donzx | [TK]D-Fender, thanks, I basically need to read 5 dtmf digits ( after playing a file then pass this onto perl script as argument ), I think it should be simple to do |
| 02:00.16 | skopii | Hi, i got asterisk working with a 2-party video conference. I am wondering if it supports a 3 or more party video conference? |
| 02:00.20 | [TK]D-Fender | donzx, yes |
| 02:00.51 | fujin | donzx: so do it? :D |
| 02:01.38 | fujin | read(agent,agent-user,5); system(/path/to/perl ${agent}); |
| 02:03.12 | Shaun2222 | any idea why i'm seeing these warnings? http://pastebin.com/m5790fab0 |
| 02:03.30 | Shaun2222 | my extensions.conf plays a goodbye message ang calls hangup |
| 02:04.01 | mosty | hmm, i have a call between two sip friends, only the called end is marked as "InUse", why isn't the calling end also "InUse" ? |
| 02:04.46 | [TK]D-Fender | mosty, .... PASTEBIN.... |
| 02:08.19 | keith4_ | hmm, i seem to have broken my outbound calling. but internal stuff works fine. log shows "app_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown)" |
| 02:08.41 | Shaun2222 | [TK]D-Fender: any ideas about my pastebin? |
| 02:08.52 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 02:09.43 | [TK]D-Fender | Shaun2222, no idea |
| 02:09.54 | [TK]D-Fender | keith4_, See above... |
| 02:10.11 | [TK]D-Fender | keith4_, "load chan_zap.so" while you're at it... |
| 02:10.37 | keith4_ | i have some analog lines internally that work fine on zap channels |
| 02:10.51 | keith4_ | this doesn't make any sense |
| 02:11.49 | [TK]D-Fender | keith4_, start PASTEBINNING..... |
| 02:11.56 | keith4_ | mmmm, pastebin |
| 02:12.37 | drmessano | Can you pastebin me a link to pastebin? |
| 02:12.49 | keith4_ | ... |
| 02:13.13 | keith4_ | try rafb.net |
| 02:13.16 | [TK]D-Fender | ~pb |
| 02:13.17 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:13.18 | [TK]D-Fender | EW! |
| 02:13.25 | [TK]D-Fender | rafb = bleh. You can't edit stuff after |
| 02:14.59 | keith4_ | no ads though |
| 02:15.20 | drmessano | Its not that hard to implement a non-shitty pastebin |
| 02:16.32 | *** join/#asterisk metfan2007 (n=metfan20@189.135.120.78) |
| 02:16.37 | mosty | [TK]D-Fender, http://pastebin.com/d6d0f0b38 |
| 02:17.42 | keith4_ | http://pastebin.com/m4d68883e |
| 02:17.55 | [TK]D-Fender | mosty, I said "type=peer" <------ |
| 02:18.49 | mosty | [TK]D-Fender, won't it work with type=friend? the wiki suggests that it can |
| 02:18.56 | [TK]D-Fender | keith4_, CLI output, not debug log |
| 02:19.07 | [TK]D-Fender | mosty, and the WIKI is *NEVER* wrong.... |
| 02:19.22 | keith4_ | heh |
| 02:19.26 | metfan2007 | Hi everybody... has anyone successful integrated an Avaya Definity and Asterisk via H.323 trunk??? I don't have idea how to start :S |
| 02:20.52 | keith4_ | [TK]D-Fender: eh.... CLI output? |
| 02:22.49 | metfan2007 | anybody??? H.323 trunk? |
| 02:22.55 | Shaun2222 | whats ${EXTEN} end up beign set too if your in a s,1,blah context? |
| 02:24.17 | Juerd_ | Does anyone know a sipphone with good dbus support? |
| 02:24.27 | Juerd_ | (That'll be a softphone) |
| 02:25.09 | keith4_ | does ekiga or xlite not have dbus support? |
| 02:25.24 | Juerd_ | openwengo and ekiga have dbus support, ekiga doesn't do multiple lines, and openwengo can't transfer calls through the dbus interface |
| 02:25.27 | Juerd_ | I'll have a look at xlite |
| 02:25.35 | keith4_ | it's.... a binary lump |
| 02:25.47 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 02:26.00 | Juerd_ | Oh, I'm restricted to open source |
| 02:26.09 | Juerd_ | (And I think that's great) |
| 02:26.43 | donzx | does anyone know why It would say password incorrect with exten => 51,1,VoiceMailMain(1234), in voicemail.conf I have 1234 => 7878,Example Mailbox, .... ? |
| 02:26.44 | AndyGraybeal_ | Juerd_: that so looks like a conversation that you had yesterday! |
| 02:26.46 | AndyGraybeal_ | or the day before |
| 02:27.12 | Juerd_ | AndyGraybeal_: Really? I just installed asterisk today... |
| 02:27.19 | AndyGraybeal_ | yea, weird, nevermind me |
| 02:27.34 | Juerd_ | I have designed an interface for a phone, but now need a way to make it work :) |
| 02:27.37 | donzx | when I enter 7878 I get -- Incorrect password '' for user '1234' |
| 02:27.42 | *** join/#asterisk saftsack (n=saftsack@p4FC76A9C.dip.t-dialin.net) |
| 02:27.54 | Juerd_ | Since asterisk obviously can't tell a phone to pick itself up (grin), I'll need to drive a softphone I think. |
| 02:28.13 | keith4_ | you just need "autoanswer" |
| 02:28.50 | *** join/#asterisk Felix27 (n=test123@rrcs-76-79-65-114.west.biz.rr.com) |
| 02:28.50 | Juerd_ | keith4_: That's my last resort, if I can't find anything else, I'll just use a separate extension and transfer calls when they should be "picked up" |
| 02:29.47 | Juerd_ | Hi caller, here's some music while you're in the humanless limbo. Your call may be transferred to the operator... or not ;) |
| 02:29.51 | Juerd_ | >:) |
| 02:29.56 | [TK]D-Fender | donzx, because your DTMF mode is set incorrectly and * can't even see you entering anything <- |
| 02:30.35 | Felix27 | Hello, I install trixbox but I don't see where I can add queues and groups. How can I install this part? thank you |
| 02:31.36 | drmessano | lol |
| 02:32.43 | keith4_ | maybe my trunk line is dead? |
| 02:32.46 | keith4_ | hmmm |
| 02:33.44 | *** join/#asterisk mike-ekim (n=mike@24.238.28.248) |
| 02:33.56 | mike-ekim | im trying to do a Queue() but it tells me on console output that Queue application not found |
| 02:34.02 | donzx | [TK]D-Fender, how do I fix this ? |
| 02:34.05 | mike-ekim | show application Queue also returns that it is not a registered |
| 02:34.08 | SplasPood | mike-ekim: load app_queue.so ? |
| 02:34.15 | mike-ekim | ah! |
| 02:34.25 | SplasPood | mike-ekim: if that does it, check your modules.conf |
| 02:34.34 | Felix27 | any idea about trixbox? |
| 02:34.41 | mike-ekim | wow that was easy |
| 02:34.51 | metfan2007 | anybody? h.323 experience? |
| 02:35.01 | JunK-Y | Felix27: see #trixbox, we dont support it here. |
| 02:35.02 | donzx | [TK]D-Fender, dtmfmode = rfc2833 in sip.conf ? |
| 02:35.17 | *** join/#asterisk Rias (n=VANDORGE@65.240.61.219) |
| 02:35.37 | Felix27 | thanks |
| 02:37.44 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
| 02:38.23 | *** part/#asterisk Rias (n=VANDORGE@65.240.61.219) |
| 02:42.46 | Juerd_ | Cool, you can Dial() with music on hold, without answering |
| 02:43.21 | Juerd_ | And actually, when I call with my mobile phone, it continues to say "dialing" until the softphone that I Dial() picks up. |
| 02:43.39 | Juerd_ | I wonder if for billing purposes, it counts as a call. |
| 02:44.15 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 02:44.35 | Juerd_ | Oh, I should still have that prepaid sim card. I can test! |
| 02:47.02 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:48.52 | Juerd_ | Well isn't that sweet, it's free |
| 02:49.33 | Juerd_ | Now I only have to forward calls originating from my mobile phone to a radio stream and I have free mobile radio :D |
| 02:49.50 | Juerd_ | (Nevermind that radio is already free over the air :)) |
| 02:52.07 | *** join/#asterisk Blinkiz (n=niklas@h-89-233-204-231.wholesale.rp80.se) |
| 02:53.54 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:53.59 | keith4_ | oy |
| 02:54.15 | *** join/#asterisk obnauticus (n=obnautic@c-24-22-14-101.hsd1.or.comcast.net) |
| 02:54.42 | Juerd_ | Oh, it doesn't update even if I call a normal phone number |
| 02:55.01 | Juerd_ | So disregard my saying it's free. The balance check isn't real time. |
| 02:55.55 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-075-182-081-167.nc.res.rr.com) |
| 02:58.48 | keith4_ | can I set variables to other variables? |
| 02:58.56 | Juerd_ | Yes |
| 02:58.58 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 02:58.58 | *** part/#asterisk jsb- (n=jsb@neptune.uqtr.ca) |
| 02:59.11 | Juerd_ | You can copy values. You can't alias variables, as far as I known. |
| 02:59.24 | keith4_ | so I have PHONE6=Zap/6, and then I can say FAX=PHONE6 ? |
| 02:59.44 | keith4_ | er, i misspoke |
| 03:00.04 | keith4_ | wait, no. that's what i have |
| 03:00.28 | Juerd_ | Set(AFX=${PHONE6}) |
| 03:01.04 | keith4_ | can't do that globally though, right? |
| 03:01.11 | Juerd_ | Add ,g |
| 03:01.21 | keith4_ | ooh. thanks |
| 03:18.36 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 03:21.51 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 03:28.17 | *** join/#asterisk optize (n=tyler@hosting.nerdie.net) |
| 03:28.19 | donzx | where does cmd Read look for file, in which dir ? |
| 03:28.51 | tzafrir_home | Basically under the sound files dir |
| 03:29.03 | tzafrir_home | Unless you give it an absolute path |
| 03:29.03 | optize | is there a way to limit all calls to a certain max time, like 1 hour? |
| 03:29.13 | tzafrir_home | And it optionally adds the language |
| 03:30.50 | *** join/#asterisk pixelstick (n=pixel@d-74-214-40-54.metrocast.net) |
| 03:30.58 | *** join/#asterisk AndyGraybeal (i=minDlabs@node154.34.251.72.1dial.com) |
| 03:31.54 | optize | hmph |
| 03:33.56 | AndyGraybeal | [TK]D-Fender... welp i'm buying it right now.. the linksys spa3102 right now |
| 03:34.25 | AndyGraybeal | thanks alot for hte recomendation |
| 03:34.46 | AndyGraybeal | hopefully i can get asterisk running with it |
| 03:45.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:45.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:00.04 | drmessano | yawn |
| 04:05.46 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:08.54 | mosty | [TK]D-Fender, i just changed my sip client from type=friend to two entries (type=peer and type=user respectively), reloaded sip, and BLF still does not work |
| 04:09.08 | *** join/#asterisk Realmy (n=Realmy@CPE-24-167-200-199.wi.res.rr.com) |
| 04:09.48 | Realmy | Is this the appropriate channel to ask a technical asterisk question, or should I be somewhere else? |
| 04:09.59 | mosty | when sip user 340340 makes a call, "show hints" shows 340340 as idle |
| 04:10.06 | mosty | Realmy, ask here |
| 04:10.17 | Realmy | mosty, thanks! |
| 04:10.32 | *** join/#asterisk aliverius (n=DEVS@ipd108.52.tellas.gr) |
| 04:10.43 | aliverius | heya fellas |
| 04:11.03 | Realmy | I want to send e-mail notifications when I have a voicemail via a gmail account- i.e. asterisk logs into gmail's smtp server to send the mail. Can anybody point me in the right direction, if this is possible? |
| 04:11.10 | aliverius | i wanted to know how far i can go with asterix and a capi2 isdn modem |
| 04:12.31 | mosty | Realmy, do you want all mail from your asterisk box to go via smtp.google.com ? |
| 04:12.53 | Realmy | mosty, yes, that would be ideal :) |
| 04:13.49 | mosty | realmy: configure your mail server to do that, it's not something you can configure in asterisk |
| 04:15.16 | Realmy | mosty, can you point me in the right direction to figure out how to do that (perhaps some good google keywords to search?) |
| 04:15.28 | Realmy | mosty, just sendmail gmail smtp, or..? |
| 04:15.30 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 04:16.06 | mosty | realmy: depends what smtp server you are running, find it's website, read the docs |
| 04:16.37 | Realmy | mosty, fair enough, I'll start digging :) thanks! |
| 04:16.39 | keith4_ | i'm trying to do fax detection and send the call to a fax machine connected on a zap channel. using a free-fax website to test it. http://pastebin.com/m4134840e |
| 04:19.44 | keith4_ | thoughts? |
| 04:20.26 | keith4_ | Realmy: nullmailer *might* do it... if you only ever send TO gmail |
| 04:21.56 | Realmy | keith4_, Thanks, I'll look into nullmailer. I just want to dump messages to a gmail account- I'm really the only one who uses the * box, and I don't really want to mess around with all of the spam restrictions from sending e-mail from a residential IP :) |
| 04:22.52 | keith4_ | you might have to use your ISPs smtp server |
| 04:24.58 | Realmy | keith4_, iirc gmail uses a nonstandard port for its smtp server, so you can usually just go directly through it. That's how I have my thunderbird configured anyway :) I think I'm on the right track with the google searching- thanks for the direction! |
| 04:25.23 | keith4_ | heh... good luck |
| 04:25.33 | rob0 | 587 (submission) is indeed a standard port. You have to AUTH. |
| 04:26.01 | rob0 | most any MTA can be set up to do that. |
| 04:28.50 | SwK | is there a reinvite issue in the 1.4branch? |
| 04:29.36 | mosty | Realmy, smtp.google.com uses the standard ports. but you will need to use authentication |
| 04:30.01 | Realmy | keith4_ and rob0, could I use ssmtp to accomplish this by editing the mailcmd= line in voicemail.conf? |
| 04:32.15 | SwK | ok... anyone... asterisk treating a codec change re-invite as a new call... SVN-branch-1.4-r96932 |
| 04:32.39 | lmadsen | SwK: thought that was a known issue.... |
| 04:32.46 | SwK | is it? |
| 04:32.48 | lmadsen | was thinking there was an open bug for it |
| 04:32.53 | SwK | I'm asking hah |
| 04:33.08 | lmadsen | could be wrong, but it sounds vaguely familiar |
| 04:33.25 | SwK | off the mantis i go |
| 04:35.47 | *** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:36.13 | SwK | well I dont see it |
| 04:36.21 | SwK | not under channels/chan_sip/* |
| 04:36.23 | SwK | but |
| 04:43.16 | *** join/#asterisk tengulre (n=tengulre@124.42.50.54) |
| 04:44.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 04:45.27 | Realmy | ah! perfect! |
| 04:47.19 | Realmy | Just in case this ever comes up again: I configured ssmtp to send the mail through the gmail account that I specified. From there all I had to do was edit that mailcmd= line to use ssmtp |
| 04:47.23 | Realmy | works like a charm :) |
| 04:50.40 | keith4_ | anyone have faxing working through asterisk? |
| 04:50.50 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 04:50.51 | keith4_ | http://pastebin.com/m4134840e |
| 05:00.09 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 05:12.14 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 05:16.05 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 05:17.57 | metfan2007 | is there any way to connect to a odbc database from the dial plan? |
| 05:18.24 | keith4_ | http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_dbodbc |
| 05:18.35 | russellb | func_odbc is a better option ... |
| 05:19.15 | metfan2007 | why func_odbc is a better option? |
| 05:41.00 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 05:52.15 | *** join/#asterisk MSpin (n=jjhuff@gw.mspin.net) |
| 05:52.40 | MSpin | wow, I haven't been here in *ages* |
| 05:53.18 | jblack | Then you should visit more often. ;D |
| 05:54.06 | MSpin | heh, not a huge reason since I'm not running a * install ATM |
| 05:54.38 | MSpin | until the boss started bitching about his dorky PBX at home :) |
| 06:01.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
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| 06:28.56 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
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| 07:25.40 | yidiyuehan | hello guys. |
| 07:25.59 | yidiyuehan | any one knows how i can add a delay between Dial command for ISDN2 channel? |
| 07:26.23 | dominic1 | hi, wait is your friend |
| 07:26.24 | yidiyuehan | I am used to use 'w' for zaptel channel but it doesn't work for ISDN2 channel |
| 07:33.52 | yidiyuehan | I am used to use 'w' for zaptel channel to insert a delay with Dial command, but it doesn't work for ISDN2 channel. any one knows other command for ISDN2? |
| 07:35.19 | *** part/#asterisk gamoreland (n=gamorela@c-67-169-140-102.hsd1.ca.comcast.net) |
| 07:35.54 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:37.19 | mosty | yidiyuehan, Wait() ? |
| 07:37.47 | jblack | Yay! I found the book for the spa 8000! |
| 07:39.11 | yidiyuehan | hi mosty, wait cannot be used for dial command right? |
| 07:39.31 | jblack | wait and dial are two seperate commands. |
| 07:39.41 | jblack | what do you want? To specify how long to let the dial happen before you give up? |
| 07:39.42 | yidiyuehan | exten=> _852X.,1,Dial(mISDN/1/18005606060wwwww28952747#ww${EXTEN}#) |
| 07:39.51 | mosty | yidiyuehan, just Wait before the dial command? |
| 07:39.52 | yidiyuehan | that's what i want to have |
| 07:40.00 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:40.21 | yidiyuehan | i want to call the number 18005606060, then wait for 4 secs, then dial 28952747 automatically |
| 07:40.34 | jblack | Oh, dialing pauses. |
| 07:40.36 | yidiyuehan | then wait for 1 sec, and dial the numbers |
| 07:40.59 | yidiyuehan | yes, it works for zap analog line, but doesn't work for ISDN2 digital line.... |
| 07:41.02 | jblack | Perhaps it's , |
| 07:41.28 | yidiyuehan | any idea mosty and jblack? |
| 07:42.05 | jblack | http://www.velocityreviews.com/forums/t236300-inserting-pauses-into-asterisk-dialing-rules.html |
| 07:42.14 | jblack | then search for "This bit of Asterisk macro" |
| 07:43.57 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 07:44.05 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 07:46.49 | yidiyuehan | thank jblack, that maybe work for digital line. |
| 07:47.07 | *** part/#asterisk dev00 (n=burkon@office-gw.ush.de.tnib.net) |
| 07:47.11 | yidiyuehan | i will try and see how. just wondering whether there is a simple command like "w" will insert the pause. |
| 07:47.19 | yidiyuehan | for ISDN digital line |
| 07:49.57 | mosty | yidiyuehan, why don't you just use the Wait command? |
| 07:51.08 | yidiyuehan | mosty, wait command doesn't work in this case as i need to send out the digits for outgoing call. |
| 07:51.24 | *** join/#asterisk af_ (n=getsmart@88-149-241-230.dynamic.ngi.it) |
| 07:51.43 | yidiyuehan | or either I am stupid that don't know how to use wait for my case like this. |
| 07:51.52 | mosty | yidiyuehan, what are you trying to do exactly? i think i am missing the point |
| 07:54.04 | yidiyuehan | exten=> _852X.,1,Dial(mISDN/1/18005606060wwwww28952747#ww${EXTEN}#) |
| 07:54.18 | yidiyuehan | i want to call the number 18005606060, then wait for 4 secs, then dial 28952747 automatically |
| 07:54.24 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:54.28 | yidiyuehan | then wait for 1 sec, and dial the numbers |
| 07:55.05 | yidiyuehan | the command "w" works for analog line, but not mISDN driver |
| 07:55.40 | mosty | yidiyuehan, Dial(mISDN/...etc), Wait(1) Dial(somethingelse) ? |
| 07:56.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:57.45 | yidiyuehan | let me try again, as I tried before asterisk will only execute first dial command unless it's busy or unavailable. |
| 07:58.30 | mosty | you can check ${DIALSTATUS} after the first dial command |
| 08:01.52 | *** join/#asterisk Maliuta (i=nikolai@119.11.97.205) |
| 08:05.31 | yidiyuehan | sorry mosty, it won't execute next command. |
| 08:06.21 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:09.15 | mosty | why not? |
| 08:11.20 | yidiyuehan | as the fist Dial command will bridge the call and asterisk thinks they are talking so it won't execute next commands. the status is answered status |
| 08:11.51 | yidiyuehan | it's very strange that even like this: exten=> _852X.,1,Dial(mISDN/1/18005606060,tr) |
| 08:12.01 | yidiyuehan | asterisk will also say error on it. |
| 08:12.07 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:12.12 | *** join/#asterisk whymarkwhy (n=koos@nas-mid.nashuamp.co.za) |
| 08:12.14 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:12.20 | yidiyuehan | only way using mISDN driver is exten=> _86X.,1,Dial(mISDN/1/18005606060) |
| 08:13.03 | yidiyuehan | does it because all the dial options like tr are for zap channels only> |
| 08:13.03 | *** join/#asterisk Maliuta (i=nikolai@119.11.97.205) |
| 08:13.05 | mosty | what error? |
| 08:14.06 | yidiyuehan | -- Executing Dial("Local/865584042168@trydundi-73b6,2", "mISDN/1/18005606060|tr") in new stack |
| 08:14.16 | yidiyuehan | <PROTECTED> |
| 08:14.21 | yidiyuehan | P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE] |
| 08:14.26 | yidiyuehan | <PROTECTED> |
| 08:14.28 | whymarkwhy | Hi there need help with a strange one, had a working asterisk ivr system linked to 8 fxo trunks, it seems the link between asterisk and the extension.conf file has disapeared. i can not do a extension reload nor does it seems that asterisk uses it to load when restarting |
| 08:14.32 | whymarkwhy | any help welcome |
| 08:15.55 | SwK | yidiyuehan, its ,,tr |
| 08:16.26 | SwK | Dial(mISDN/1/18005606060,tr) will error cause you are trying to use tr as your dial timeout |
| 08:16.41 | SwK | Dial(mISDN/1/18005606060,,tr) is the correct usage for no dial timeout |
| 08:18.13 | yidiyuehan | yes! SwK, thank you, by the way do you know how I can use "w" to insert a delay for mISDN channel |
| 08:22.59 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:24.32 | whymarkwhy | Hi there need help with a strange one, had a working asterisk ivr system linked to 8 fxo trunks, it seems the link between asterisk and the extension.conf file has disapeared. i can not do a extension reload nor does it seems that asterisk uses it to load when restarting |
| 08:25.06 | mikkel | Hi is it possible to make people wait on line (listen to music) will finishing another call. If you only have one line in ? Do I need as many lines as I want people in line ? |
| 08:29.15 | whymarkwhy | not if you are using sip mikkel yes if you are using fxo lines : mikkel |
| 08:31.13 | mikkel | whymarkwhy: I'm pretty new at this, so fxo and sip is still not clear to me. sip is internet ? and fxo is landlines ? |
| 08:31.39 | mikkel | I'm using a normal analog line. |
| 08:32.14 | *** join/#asterisk vgster (n=vgster@psc.navonline.net) |
| 08:34.02 | whymarkwhy | no you would not be able to play moh for a second because you can only have one call per analog line, but if you have more than one sip extension you can play moh for the second caller |
| 08:35.30 | mikkel | whymarkwhy: I have a Wildcard TDM22B (2 FXO, 2 FXS). But it is not possible on analog lines, you say I can only have one call per line ? |
| 08:36.22 | mikkel | whymarkwhy: So to make a people able to call in and wait in line, I need ISDN with as many lines as I want people waiting ? |
| 08:37.06 | mikkel | whymarkwhy: I don't want any SIP (internet calls for now) |
| 08:39.39 | whymarkwhy | yes on a basic rate isdn you can have 2 calls at the same time |
| 08:39.55 | whymarkwhy | on e1 you can have up to 30 calls on hold |
| 08:40.26 | whymarkwhy | if you want two calls on hold on fxo you need more than one fxo line |
| 08:42.16 | mikkel | whymarkwhy: With 2 fxo is it possible to have them dial the same number or is there two numbers ? |
| 08:42.37 | JT | the telco can setup line hunt |
| 08:43.05 | whymarkwhy | you may ask your telko to link the two lines with one number called hunting |
| 08:51.59 | mikkel | whymarkwhy: So I could use analog lines and make the teleco setup line hunt. But if I need, lets say 8 people on hold, I should use ISDN (E1) ? |
| 08:52.40 | J4zen | How does asterisk perform on VPS? |
| 08:52.59 | J4zen | For example, if i wanted to run multiple asterisk nodes on one physical server |
| 08:53.01 | J4zen | VPS would be the way to go? |
| 08:55.20 | tzafrir_laptop | What type of VPS? |
| 08:55.53 | J4zen | yet to be determained, all theorycraft here |
| 08:55.58 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 08:57.08 | tzafrir_laptop | mikkel, yes, it is possible to put people "on hold". But they will still use an incoming line of yours |
| 08:58.12 | tzafrir_laptop | In fact, if you have two lines you can easily connect a call through those two lines (and wonder why the ^&*^&* you cannot dial out) |
| 08:59.13 | tzafrir_laptop | J4zen, you can have multiple asterisk processes on the same server without a VPS |
| 08:59.26 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com) |
| 09:05.48 | FlatFoot | morning all |
| 09:05.52 | J4zen | tzafrir: Thanks |
| 09:08.28 | *** join/#asterisk BdPuK (n=ben@83-244-135-2.cust-83.exponential-e.net) |
| 09:14.46 | *** join/#asterisk Federico2 (n=fede@pdpc/supporter/base/Federico2) |
| 09:14.49 | Federico2 | hi there |
| 09:17.01 | whymarkwhy | Hi there need help with a strange one, had a working asterisk ivr system linked to 8 fxo trunks, it seems the link between asterisk and the extension.conf file has disapeared. i can not do a extension reload nor does it seems that asterisk uses it to load when restarting |
| 09:17.24 | whymarkwhy | what is the command for extension reload in asterisk 1.4.8 |
| 09:17.25 | whymarkwhy | ? |
| 09:17.48 | ronr | whymarkwhy: dialplan reload? |
| 09:18.12 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:18.48 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 09:19.21 | whymarkwhy | asterisk tells me there is no such a command |
| 09:20.24 | ronr | whymarkwhy: try asterisk -vvvr and use tab completion on d<tab> to figure out what your asterisk does support |
| 09:22.07 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
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| 09:28.50 | squigly | whymarkwhy, just a reload should do it? |
| 09:29.53 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:31.10 | whymarkwhy | if i start asterik without zaptel driver i can reload using dialplan reload, but if i start the zaptel driver and the try ,, the dialplan reload is missing in the command selection |
| 09:31.42 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 09:31.52 | whymarkwhy | how does one uninstall asterisk and zaptel to start from scratch? |
| 09:32.02 | whymarkwhy | using asterisk 1.4.8 |
| 09:32.46 | ronr | whymarkwhy: find <myasterisksrcdir> -type f | xargs grep uninstall |
| 09:33.59 | mikkel | whymarkwhy: Thanks for the help. (Looking at all the expensive cards from digium :) |
| 09:34.12 | Federico2 | I need a sort of a "reverse ExtenSpy": when user A calls B, asterisk should also call C and let him snoop the conversation |
| 09:35.24 | whymarkwhy | does not work ronr/ |
| 09:40.42 | ronr | whymarkwhy: really, if I do it I find there's an uninstall option in the Makefile (which in turn tells me the odds are very much in my favor that make uninstall will uninstall asterisk) |
| 09:42.32 | Federico2 | is it possible to dial out multiple destinations and letting more than one user answering? |
| 09:45.21 | ronr | Federico2: take a look at MeetMe (page 164 of the book) |
| 09:46.03 | Federico2 | ronr: I did, but I tought meetme require user interaction... am I wrong? |
| 09:47.21 | ronr | Federico2: I haven't used MeetMe (yet), but I guess if you have a room at 600 you can just Dial(phone1&phone2&phone3, 600) thrice |
| 09:47.41 | Federico2 | thank you! |
| 09:48.51 | whymarkwhy | thx ronr i just made a huge typo |
| 09:56.11 | fujin | Is there a way for me to use svn to manage the current trunk release? |
| 09:56.23 | fujin | i.e.; instead of downloading asterisk-14.1.17.tar.gz etc whenevr it comes out |
| 09:56.30 | fujin | I'd just like to use svn |
| 09:57.27 | *** join/#asterisk FOMICIANO (n=FOMICIAN@189-015-178-034.xd-dynamic.ctbcnetsuper.com.br) |
| 09:59.50 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
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| 10:01.01 | fujin | nevermind, managed to work it out |
| 10:01.10 | *** join/#asterisk harpal (n=Harpal@124.125.79.212) |
| 10:01.14 | fujin | svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 10:01.24 | fujin | afaict, pulls the latest stable equivalent of 1.4.17 |
| 10:03.01 | Federico2 | ronr: after the MeetMe exten is run, the user is stuck there so the Dial exten are never executed |
| 10:05.10 | *** join/#asterisk BdPuK (n=ben@83-244-135-2.cust-83.exponential-e.net) |
| 10:08.08 | ronr | Federico2: ah, of course, maybe there's a flag for the dial application to force it to continue? |
| 10:13.21 | *** join/#asterisk stuarta (n=stuarta@unaffiliated/stuarta) |
| 10:13.39 | tzafrir_laptop | well, latest 1.4 has som changes that were not included in 1.4.17 (commited later) |
| 10:14.54 | Federico2 | ronr: also the meetme should continue |
| 10:15.41 | awk | honestly will digium ever fscking release a version of 1.4 asterisk without the constant shit and bugs that come with it |
| 10:15.50 | awk | I thought dead locks where fixed in 1.4.17, I guess I was wrong |
| 10:16.04 | awk | nice... fucking whole call centre fucking out once again... |
| 10:18.05 | ronr | Federico2: the MeetMe only sets up the room, you want to add people to it right? |
| 10:18.53 | *** join/#asterisk FOMICIANO (n=FOMICIAN@189-015-187-004.xd-dynamic.ctbcnetsuper.com.br) |
| 10:18.56 | ronr | Federico2: how about having your dialplan call some AGI and have this AGI generate call files that connect a bunch of phones to the room |
| 10:22.18 | badcfe | argh, what do i need in order to compile asterisk .. i get *** termcap support not found |
| 10:24.45 | badcfe | found it, guess its libncurses5-dev |
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| 10:36.08 | fujin | badcfe: apt-get build-deps asterisk |
| 10:36.58 | whymarkwhy | awk whats wrong? |
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| 10:51.57 | awk | whats wrong? |
| 10:52.18 | awk | erm lets think.. since i have started on 1.4 I have not had 1 release that fucking does not have some or other serious bug.. |
| 10:59.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 11:10.04 | whymarkwhy | what type of bugs do you get? |
| 11:10.32 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:11.02 | awk | well in 1.4.17 I have a whole bunch of channels that dont hang up |
| 11:11.12 | *** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 11:11.18 | awk | and it causes major issues with agents and calls not being passed to them |
| 11:11.25 | awk | I cant soft hangup those channels either |
| 11:11.30 | awk | let me restart asterisk. wow its fixed... |
| 11:11.44 | awk | its a 24 hour call centre, you cant just restart it.. |
| 11:11.55 | shazaum | any know phone soyo? |
| 11:12.09 | shazaum | i need upgrade firmware |
| 11:12.33 | *** join/#asterisk appelza (n=d@dsl-240-173-87.telkomadsl.co.za) |
| 11:12.41 | appelza | where can I set the full log max size limits? |
| 11:13.16 | shazaum | i see in site of the phone, "upgrade firmware" |
| 11:13.46 | awk | whymarkwhy where in za are you from |
| 11:14.09 | shazaum | But when the whole load firmware back as he was and not update |
| 11:15.00 | shazaum | Anyone know how I can see the version of the chip is open the phone???? |
| 11:20.09 | appelza | <PROTECTED> |
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| 11:22.32 | shazaum | appelza: maybe set a rotate in cron |
| 11:22.54 | shazaum | after in cron rm :D |
| 11:25.12 | shazaum | appelza understand? |
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| 11:29.14 | *** join/#asterisk felipex (n=nonloso@88-149-172-71.static.ngi.it) |
| 11:29.20 | felipex | hi at all |
| 11:29.28 | felipex | how can i debug voicemail? |
| 11:29.40 | felipex | the message is recorded but not sent in email |
| 11:30.26 | *** part/#asterisk stuarta (n=stuarta@unaffiliated/stuarta) |
| 11:30.36 | shazaum | fujin: but u have server mail? |
| 11:31.19 | shazaum | u can set debug in logger |
| 11:31.33 | felipex | shazaum i done it |
| 11:31.52 | felipex | but i can't see anything on console about mailing |
| 11:31.52 | awk | is there a way to soft hangup a queue call where there is no zap channel associated |
| 11:32.01 | awk | thats what bug im getting in 1.4.16/17 |
| 11:32.29 | awk | if I get agents to log out and log in it seems to work again but then randomly it picks agents and it stops working till they log out and in again |
| 11:32.33 | awk | but as i restart asterisk it works |
| 11:32.56 | awk | for 3/4 days till i get that channel that has no zap association and it starts screwing with things big time! |
| 11:33.31 | shazaum | felipex: use "tail -f" in file debug |
| 11:36.27 | felipex | shazaum debug file under /var/log/ ? |
| 11:37.18 | shazaum | yes |
| 11:37.22 | shazaum | /var/log/asterisk |
| 11:37.49 | shazaum | see the logger.conf |
| 11:38.03 | shazaum | has more options |
| 11:38.10 | felipex | shazaum no there isn't that file under /var/log/asterisk |
| 11:39.04 | shazaum | oh man |
| 11:39.34 | shazaum | see asterisk.conf |
| 11:39.44 | shazaum | search by astlogdir |
| 11:39.56 | shazaum | type.... |
| 11:40.10 | shazaum | in my conf |
| 11:40.13 | shazaum | astlogdir => /var/log/asterisk |
| 11:40.57 | shazaum | u can specify other directory |
| 11:41.21 | felipex | yes also for me but debug file there isn't |
| 11:41.43 | awk | oh great, now I see it locks uip the CLI too |
| 11:42.30 | felipex | shazaum ok the problem was on logger.conf |
| 11:42.36 | felipex | now i have it |
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| 11:43.36 | felipex | shazaum nothing about email |
| 11:43.47 | felipex | i see the message recorded |
| 11:43.57 | felipex | but than nothing about mailing it |
| 11:44.44 | shazaum | k |
| 11:44.53 | shazaum | but u can serv mail ? |
| 11:45.03 | shazaum | for send the mail? |
| 11:45.26 | felipex | shazaum what you mean for serv mail ? |
| 11:46.18 | shazaum | mail server |
| 11:46.25 | shazaum | :D |
| 11:46.32 | shazaum | my english is bad |
| 11:46.53 | felipex | shazaum going to check it |
| 11:48.14 | shazaum | ok |
| 11:52.13 | awk | I wish I had the simple problems that everyone else is suffering here.. not problems out of my control |
| 11:52.23 | awk | not massive bugs in asterisk |
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| 12:04.38 | felipex | shazaum the server doesn't send email |
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| 12:04.50 | felipex | i need to use a smtp |
| 12:04.57 | felipex | it is possible? |
| 12:06.16 | shazaum | yes |
| 12:06.27 | shazaum | use a client |
| 12:06.39 | shazaum | i use the native client |
| 12:06.44 | shazaum | the OS |
| 12:07.01 | shazaum | type, debian etch has exim |
| 12:07.26 | felipex | i use debian etch |
| 12:07.32 | felipex | ok i have to configure exim |
| 12:07.52 | felipex | later i try |
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| 12:08.00 | felipex | thanks for all at the moment |
| 12:08.03 | felipex | bye |
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| 12:35.55 | Faustov | hi, to what should i set allow= in sip.conf if i have to use codec g729 ? |
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| 12:40.08 | awk | allow=g729 |
| 12:40.47 | Faustov | is notation like allow=g729,ulaw correct? |
| 12:40.52 | Faustov | to allow those 2? |
| 12:44.36 | waKKu | i'm sure u can use 2 lines to specify it.. dont sure that you line is correct |
| 12:44.51 | waKKu | allow=g729 |
| 12:44.53 | waKKu | allow=ulaw |
| 12:44.56 | Faustov | oh |
| 12:44.58 | Faustov | that way |
| 12:45.06 | Faustov | so the first one gets higher priority? |
| 12:45.13 | waKKu | yeah |
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| 12:48.10 | Faustov | cool, thanks |
| 12:48.41 | Faustov | but it seems i dont have g729 - is it a part of asterisk or do i have to install something else? |
| 12:50.32 | mosty | you have to download it seperately |
| 12:50.52 | mosty | and you will need to buy licences from digium |
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| 13:10.22 | Faustov | mosty: are you sure i need some special license for g729? I can use this codec from some linksys voip gw, but if i run asterisk i need a special license? |
| 13:11.11 | mosty | if you want asterisk to transcode g729, you need to buy a licence from digium. if you want asterisk to do g729 in passthru mode only, no licence is required |
| 13:11.23 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 13:12.11 | Faustov | and what needs to be done to achieve the latter? |
| 13:12.48 | mosty | allow=g729 in iax.conf or sip.conf, and don't attempt to transcode in your dialplan |
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| 13:20.46 | felipex | shazaum are yout here? |
| 13:21.02 | ZaVoid | morning all |
| 13:21.48 | felipex | hi at all |
| 13:21.57 | felipex | how to debug voicemail ? |
| 13:22.10 | felipex | the file is recorded but not sent to email box |
| 13:22.28 | felipex | and i don't understand why |
| 13:23.26 | mosty | felipex, pastebin your voicemail.conf |
| 13:23.33 | ZaVoid | sup mosty |
| 13:23.33 | felipex | mosty ok |
| 13:23.34 | waKKu | does a MTA configured? |
| 13:23.57 | felipex | waKKu yes from command line i can send mail with mutt |
| 13:24.15 | Faustov | mosty: how can i disable transcoding? It must be enabled by default, because i haven't set it up... Also what would be the downside of this? |
| 13:24.58 | mosty | faustov: transcoding occurs when you try to play non-g729 sounds to a sip client that only supports g729 |
| 13:25.29 | mosty | Faustov, or when you try to record a channel in non-g729 format when the channel only supports g729 |
| 13:26.36 | Faustov | mosty: i am getting this after i set disallow=all, allow=g729 |
| 13:26.38 | Faustov | [Jan 8 14:56:38] WARNING[4505]: chan_sip.c:2963 sip_call: No audio format found to offer. Cancelling call to 00226121601 |
| 13:26.41 | *** join/#asterisk dundel (n=daniel@200.2.161.172) |
| 13:28.06 | mosty | faustov: what is the source and destination of that call? and can you pastebin your sip.conf? |
| 13:29.43 | felipex | mosty http://www.pastebin.ca/846627 |
| 13:31.28 | Faustov | mosty: just a moment |
| 13:31.33 | mosty | felipex, pastebin.ca does not work for me, can you use a different pastebin site? |
| 13:32.04 | felipex | mosty which one? |
| 13:32.20 | mosty | any other pastebin site |
| 13:32.40 | felipex | pastebin.com is ok? |
| 13:33.00 | mosty | should be |
| 13:33.06 | felipex | http://pastebin.com/m67d2a8b1 |
| 13:33.20 | mosty | pastebin.ca appears blank for me, no idea why |
| 13:33.39 | felipex | mosty no problem |
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| 13:36.16 | mosty | felipex, is the voicemail file recorded ok, and if so what do your mail logs show? |
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| 13:37.57 | felipex | mosty the voicemail is recorded |
| 13:38.07 | felipex | mail logs don't show nothing |
| 13:38.22 | Faustov | mosty: http://pastebin.ca/846633 here's part of my sip.conf and extensions.conf; i'm dialing 361 |
| 13:39.01 | mosty | felipex, try putting attach=yes in your user line |
| 13:39.23 | mosty | Faustov, pastebin.ca does not work for me, try pastebin.com |
| 13:40.32 | Faustov | mosty: here: http://pastebin.com/m2a9eba6b |
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| 13:41.13 | waKKu | your exten configured at voicemail.conf is 1234, is not ? Why are you dialing 361 ? |
| 13:42.30 | mosty | Faustov, what device is calling easycall? |
| 13:43.03 | Faustov | waKKu: i am not using voicemail.conf |
| 13:43.33 | Faustov | mosty: agent [0012] registered in sip.conf |
| 13:43.38 | waKKu | ops.. :) |
| 13:44.25 | mosty | Faustov, add the [0012] section of sip.conf to your pastebin and show us again |
| 13:44.41 | felipex | mosty which user line ? |
| 13:45.06 | mosty | felipex, the one that defines the voicemail account |
| 13:45.14 | Faustov | mosty: http://pastebin.com/m496678d3 |
| 13:45.45 | felipex | [default] |
| 13:45.45 | felipex | 1234 => 4242,Ricc Inf,info@enriconeri.com |
| 13:45.53 | felipex | mosty this lines? |
| 13:46.14 | mosty | Faustov, try allow=g729 in the [0012] section |
| 13:47.00 | felipex | mosty or in my sip.conf ? |
| 13:47.10 | mosty | yes |
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| 13:48.53 | Faustov | mosty: that brings me back to the initial problem (unable to dial via easycall, which uses g729), here's the log: http://pastebin.com/m174bf68c |
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| 13:50.31 | mosty | Faustov, what codecs does "sip show peer 0012" mention? |
| 13:51.55 | mosty | also, i don't see any g729 errors in that last pastebin. easycall is rejecting your call for some reason |
| 13:52.27 | Faustov | mosty: Codecs : 0x8010e (gsm|ulaw|alaw|g729|h263) |
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| 13:53.13 | Faustov | mosty: what could be the reason it is rejecting my calls? |
| 13:53.38 | mosty | Faustov, do you have the correct login details? is it registered properly? |
| 13:53.51 | Faustov | mosty: yeah |
| 13:54.29 | mosty | do you get any warnings or errors when you dial with the current setup? |
| 13:54.49 | Faustov | no, just busy tone and all that i pasted from the log |
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| 13:55.44 | mosty | Faustov, get in contact with easycall's support, they will be able to tell you |
| 13:55.56 | Faustov | yeah doing that |
| 13:56.16 | mosty | everything looks ok on your end |
| 13:58.44 | Faustov | ye and it was working |
| 14:03.56 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 14:03.56 | Faustov | oh god |
| 14:04.09 | fiXXXerMet | I know, I'm here. |
| 14:04.15 | Faustov | :< |
| 14:04.17 | fiXXXerMet | Horrible, huh? |
| 14:04.23 | Faustov | i'm about to start crying |
| 14:04.34 | *** join/#asterisk jcanfield (n=jcanfiel@68.109.242.162) |
| 14:04.43 | Faustov | they give 603 when someone's out of credit... i was so used to "you're out of credit" form a different provider |
| 14:04.47 | Faustov | that this never came to my mind |
| 14:04.52 | Faustov | spent 3h... |
| 14:06.10 | yang | Are BRI cards with HFC-4S chip working well supported with asterisk? |
| 14:07.27 | mosty | yang: i'd recommend a sangoma BRI card instead |
| 14:08.31 | yang | mosty: sangoma...do yxou know the around prices for it? |
| 14:09.21 | yang | mosty: do you recommend sangoma over digium, I find digium bri cards rather expenssive |
| 14:10.17 | felipex | mosty i added mailbox=1234@default on each account in sip.conf |
| 14:10.34 | felipex | but nothing |
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| 14:14.05 | mosty | yang, digium only make one bri card as far as i know, and it requires the misdn driver which everybody seems to hate |
| 14:14.45 | mosty | felipex: can you copy+paste an example from the wiki that has attach=yes in the voicemail user definition? |
| 14:18.31 | felipex | mosty ok i see |
| 14:18.40 | felipex | mosty do you have a url? |
| 14:18.50 | mosty | voip-info.org |
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| 14:19.09 | RoyK | ~wiki |
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| 14:23.38 | fiXXXerMet | Question. |
| 14:23.57 | fiXXXerMet | Trying to place an outbound call using our iax2 provider, voipstreet. |
| 14:24.17 | fiXXXerMet | They're setup in the iax.conf file |
| 14:24.34 | felipex | mosty like this ? http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf |
| 14:25.05 | mosty | felipex, yes |
| 14:26.07 | tzanger | wtf |
| 14:26.18 | tzanger | iax2 one-way delay |
| 14:26.21 | tzanger | this is so weird |
| 14:26.36 | tzanger | A: zap/iax -- B: iax/zap |
| 14:26.45 | tzanger | iax2 show netstats on A and B |
| 14:26.57 | tzanger | A's jitter sticks at about 832 (?!) and the delay keeps growing |
| 14:26.59 | tzanger | B is constatnt |
| 14:27.09 | tzanger | a->b audio is fine, b->a audio is delayed increasingly more |
| 14:27.52 | tzanger | A and B are on the same lan |
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| 14:30.30 | felipex | mosty ok it works |
| 14:30.39 | felipex | mosty thank you |
| 14:31.11 | fiXXXerMet | Forget my question - I figured it out. :) |
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| 15:01.48 | yang | mosty: Which drivers have you tested with Sangoma A500 ? |
| 15:02.15 | mosty | all the sangoma cards use wanpipe |
| 15:02.24 | mosty | and in turn, zaptel |
| 15:02.49 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:03.54 | yang | but in debian which package would i need - zaptel ? |
| 15:04.11 | [TK]D-Fender | yang, .... WANPIPE + ZAPTEL |
| 15:04.20 | *** join/#asterisk s0lid (n=_freq@210.213.243.83) |
| 15:04.24 | yang | I get no hits for wanpipe |
| 15:04.52 | mosty | you have to download it from sangoma, and build a .deb package which you can install |
| 15:04.56 | yang | ok |
| 15:05.33 | yang | These cards are all expenssive alternatives...There is a HFC-4S ISDN card which costs only 20 euro |
| 15:06.00 | yang | But I don't know how good it is |
| 15:06.19 | mosty | at that cost, it is definitely junk |
| 15:06.36 | mosty | if it works with asterisk at all |
| 15:07.25 | yang | It is I think the manufacturer is Beronet |
| 15:09.21 | *** join/#asterisk fuzzbawl (n=fuzzbawl@blackhole.cyberlinkint.com) |
| 15:09.39 | fiXXXerMet | ~book |
| 15:09.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 15:12.58 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 15:12.59 | tdonahue-laptop | hi all |
| 15:15.49 | tdonahue-laptop | does anyone know where the list of numbers representing codecs in the SIP m attributes are defined? |
| 15:16.06 | *** join/#asterisk deltaray2 (n=Deltaray@mayhem.kiva.net) |
| 15:16.21 | hi365 | any suggestions for an auto-dialer? |
| 15:16.44 | deltaray2 | Is it normal to have a 2-3 second delay initially when I call someone? It drops to about 1/2 second after the conversation starts, but initially it takes a second or two. |
| 15:17.14 | *** join/#asterisk workinit (n=test@205.241.238.186) |
| 15:18.54 | workinit | Anyone around for a quick question |
| 15:21.27 | J4zen | Depends on how quick the quick question is ;) |
| 15:23.42 | fuzzbawl | hey all. I'm having an issue with the directory. I hit # and all I get is "No directory entries match your search" repeatedly. Any ideas? |
| 15:24.54 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 15:26.10 | tzafrir_laptop | yang, HFC-4S card that costs 20 euros? I think you confuse it with a HFC-S card (single-span cards) |
| 15:26.20 | workinit | haha. I can't call my external extensions anymore |
| 15:29.29 | tzafrir_laptop | And the question is? |
| 15:29.47 | workinit | Yeah I guess that wasnt a question |
| 15:29.56 | admin0 | :) |
| 15:30.01 | admin0 | that was a comment i gues |
| 15:30.19 | workinit | I used to be able to call my external extensions until my co-worker switched the numbers and now I cant. I can call out but not to the extensions |
| 15:30.24 | tzafrir_laptop | Still, pastebin a trace from the CLI when you try to call out, and the relevant parts of your dialplan |
| 15:30.30 | workinit | like asterisk isnt picking them up as extensions |
| 15:30.56 | fiXXXerMet | Are there any good software ways to deal with echo? |
| 15:31.32 | [TK]D-Fender | fiXXXerMet, OSLEC |
| 15:31.43 | [TK]D-Fender | fiXXXerMet, and Digium's HPEC |
| 15:31.46 | [TK]D-Fender | ~hpec |
| 15:31.46 | jbot | hpec is probably Digium's High Performance Echo Cancellation software - http://www.digium.com/en/products/software/hpec.php - Free for Digium cards under warranty; US$10 per channel otherwise. |
| 15:31.54 | [TK]D-Fender | ~oslec |
| 15:31.54 | jbot | oslec is, like, Open Source Line Echo Canceller. See http://www.rowetel.com/ucasterisk/oslec.html . |
| 15:33.03 | Faustov | guys, is it within asterisk's possibilities to have some behaviour whenever a specific error (like 603) occurs when i make a call? |
| 15:33.11 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:33.48 | coppice | the best way to deal with echo is to have a lawyer send it a cease and desist letter |
| 15:34.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:34.47 | [TK]D-Fender | Faustov, You've got the source... |
| 15:35.05 | *** join/#asterisk PepOSX (n=pepOSX--@190.79.246.105) |
| 15:35.42 | Faustov | [TK]D-Fender: sooo, instead of browsing the source, i'm just asking if a function exists. Furthermore, i need some behaviour for a very specific case |
| 15:36.05 | [TK]D-Fender | Faustov, No, clearly nothing exists |
| 15:36.36 | [TK]D-Fender | Faustov, * has the dialplan & AMI. Thats it. * doesn't have triggers for generic stuff that you can just latch onto |
| 15:36.47 | FlatFoot | coppice: lol |
| 15:39.17 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 15:39.44 | *** join/#asterisk mtryfoss (n=mtryfoss@6.81-166-192.customer.lyse.net) |
| 15:39.51 | *** join/#asterisk dundel (n=daniel@200.2.161.172) |
| 15:40.03 | fuzzbawl | any ideas about the directory problem? permissions on voicemail.conf are 665 |
| 15:41.55 | badcfe | in one asterisk i can see any dtmf that is beeing relayed. in the others no .. how do i controll tris? aparently not with the core debug level ... |
| 15:42.27 | [TK]D-Fender | badcfe, yes, should be core debug |
| 15:46.13 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:49.31 | *** join/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 15:51.55 | *** join/#asterisk kamanashisroy (n=root@202.56.7.139) |
| 15:52.09 | kamanashisroy | hi, thanks everyone for this wonderful software |
| 15:52.16 | kamanashisroy | now, I wanted to send a call to a sip peer .. sip peer name is "5143572514" .. proxy "sip02.unlimitel.ca" and realm is "unlimitel.ca" .. |
| 15:52.23 | kamanashisroy | I dialed with "SIP/545143889985@5143572514" |
| 15:52.30 | kamanashisroy | and the sip debug shows it is sending data to proxy "sip02.unlimitel.ca" |
| 15:52.42 | kamanashisroy | Attempting call on SIP/545143889985@5143572514 |
| 15:53.14 | kamanashisroy | but it is sending something like, INVITE sip:545143889985@sip02.unlimitel.ca SIP/2.0 |
| 15:53.31 | *** join/#asterisk ProofPack (n=no@208.255.225.67) |
| 15:53.37 | kamanashisroy | should not it send something like "sip:545143889985@unlimitel.ca" ? |
| 15:53.53 | kamanashisroy | because the realm is unlimitel.ca not sip02.unlimitel.ca |
| 15:54.12 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 15:55.15 | [TK]D-Fender | kamanashisroy, "fromdomain=unlimitel.ca" <- And you have 2 extra digits in front of your phone # dialed.... not good |
| 15:56.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:56.10 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:56.31 | *** part/#asterisk fuzzbawl (n=fuzzbawl@blackhole.cyberlinkint.com) |
| 15:56.44 | *** join/#asterisk Victor_Yure (n=aaa@189.40.73.45) |
| 15:57.02 | *** join/#asterisk s0lid (n=_freq@210.213.243.83) |
| 15:57.43 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:57.43 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:58.14 | kamanashisroy | [TK]D-Fender: extra digits means |
| 15:58.15 | kamanashisroy | ? |
| 15:58.35 | [TK]D-Fender | kama"sip:545143889985@unlimitel.ca" ? <--- whats with the 54 in front? |
| 15:59.04 | *** join/#asterisk vgster (n=vgster@psc.navonline.net) |
| 15:59.05 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-1290dbf01cdda09c) |
| 15:59.22 | [TK]D-Fender | kamanashisroy, that isn't a valid dial pattern with them. |
| 16:00.40 | kams | [TK]D-Fender: confused.. give 5 minutes to check |
| 16:02.28 | *** join/#asterisk lirakis (n=lirakis@65.200.189.220) |
| 16:03.40 | kams | [TK]D-Fender:actually I am using realtime .. |
| 16:04.17 | kams | [TK]D-Fender: realm: unlimitel.ca, outboundproxy: sip02.unlimitel.ca and host: sip02.unlimitel.ca |
| 16:05.10 | *** join/#asterisk vgster (n=vgster@psc.navonline.net) |
| 16:06.49 | kams | and note that "From: "Shuva" <sip:5143572514@unlimitel.ca>;tag=as2d2f5042" is correct .. |
| 16:07.12 | kams | but To: <sip:545143889985@sip02.unlimitel.ca>;tag=as4cc7fabf is not correct (not sure ..) |
| 16:07.41 | kams | I think it should be something like To: <sip:545143889985@unlimitel.ca>;tag=as4cc7fabf |
| 16:07.51 | kams | [TK]D-Fender: what do you think ? |
| 16:09.09 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 16:09.41 | __freedom__lover | have someone configured voicemail for brazil? |
| 16:10.57 | caio1982 | yes, and you can directly join #asterisk-br if you need specific help on portuguese "tropicalization" of asterisk :) |
| 16:12.31 | __freedom__lover | do you know if i can use the number of the months? |
| 16:12.56 | __freedom__lover | caio1982: i see you are brazillian, hunf? |
| 16:13.27 | workinit | is there a good ebook on asterisk 1.2? |
| 16:14.44 | russellb | ~thebook |
| 16:14.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:14.51 | russellb | the 1st edition was based on 1.2 |
| 16:14.51 | __freedom__lover | workinit: yes, that.. |
| 16:15.05 | russellb | but i still recommend reading the 2nd edition. |
| 16:15.15 | workinit | yeah I'm reading the 2nd now |
| 16:15.26 | workinit | but our system is 1.2 and I dont have a gui |
| 16:15.28 | russellb | you shouldn't be doing any new installs on 1.2 ... |
| 16:15.46 | workinit | its already installed |
| 16:15.55 | russellb | well you should upgrade :-p |
| 16:16.08 | workinit | admin wont upgrade because he'd have to upgrade the compiler or something like that |
| 16:16.19 | workinit | << Not a linux guru still learning |
| 16:16.28 | [TK]D-Fender | kams, I thin you shouldn't have a "54" in front of your normal 10-digit number. Care to tell me why they are there? |
| 16:17.03 | ProofPack | I'm having some difficulty with calls timing out and going to Asterisk voicemail when a Polycom Phone is set for call forwarding (sip) to a cell phone. Any ideas? |
| 16:17.09 | ez` | how can i kill a sip channel now ; soft hangup does nothing |
| 16:17.43 | kams | [TK]D-Fender: sorry it is a number I am testing .. I do not know any more .. I think 10 is Ok for destination !! is not it ?? |
| 16:18.01 | [TK]D-Fender | kams, what is the 54 in front for? |
| 16:18.02 | workinit | I had trixbox running in 30 minutes or so because it was self explanatory but he wants me to learn the way its set up now with no gui |
| 16:18.38 | yang | I am having difficulties connecting my handset to asterisk...I am using asterisk as a bouncer....which connects to voip provider. But I don't know which parameters I must put into my phone handset - here is the config http://openpaste.org/en/4585/ |
| 16:19.21 | kams | [TK]D-Fender: 545143889985 this is the number I wanted to reach.. that is all .. |
| 16:19.34 | [TK]D-Fender | kams, Where is this number located? |
| 16:20.08 | kams | [TK]D-Fender: no location is necessary .. just a destination .. |
| 16:20.39 | [TK]D-Fender | kams, ..... Since when is a number like that LEGAL? |
| 16:21.04 | [TK]D-Fender | kams, That is not a legitimate North American # format, nor is it international. |
| 16:21.18 | kams | [TK]D-Fender: let me check |
| 16:22.12 | russellb | workinit: please do not message me for help directly |
| 16:22.36 | russellb | workinit: that's what this channel is for |
| 16:22.50 | *** join/#asterisk Havokmon (n=rick@64-198-2-66.ip.mcleodusa.net) |
| 16:23.31 | [TK]D-Fender | workinit, You can't learn the way it is set up "now". Now = Trixbox which if full of such bloated BS scripts. |
| 16:24.11 | *** join/#asterisk jblack (n=jblack@pool-71-181-185-97.sctnpa.east.verizon.net) |
| 16:24.19 | *** join/#asterisk FOMICIANO (n=FOMICIAN@189-015-213-047.xd-dynamic.ctbcnetsuper.com.br) |
| 16:25.01 | kams | [TK]D-Fender: I set the field fromdomain=unlimitel.ca still no luck .. |
| 16:25.23 | kams | INVITE sip:545143889985@sip02.unlimitel.ca !! |
| 16:25.32 | [TK]D-Fender | kams, And you're still dialing an improper number, not showing us your new configs, etc. |
| 16:25.49 | [TK]D-Fender | Anyways, I'm packing up to head to the office. Back later. |
| 16:26.25 | workinit | [TK]D-Fender, Trixbox was used just to see if it worked. We're just using asterisk 1.2 now |
| 16:26.33 | workinit | well thats what we've been using |
| 16:26.53 | [TK]D-Fender | workinit, If you're telling me you completely gutted it out, MAYBE.... |
| 16:26.59 | [TK]D-Fender | anyways, BBIAB |
| 16:27.46 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:27.48 | yang | Is it possible to use asterisk as a bouncer, which connects to voip provider number. And the handset connects to asterisk |
| 16:29.04 | jblack | I do that. |
| 16:29.31 | *** join/#asterisk didz_ (n=voce@201.19.91.122) |
| 16:30.38 | RoyK | yang: isn't that normal bridging? |
| 16:33.30 | kams | [TK]D-Fender: it is working .. it was all my fault sorry for interrupt you .. |
| 16:33.59 | workinit | Is there an easy fix for not being able to dial an external extension eventhough the external extensions can dial internals |
| 16:34.25 | *** part/#asterisk kams (n=root@202.56.7.139) |
| 16:34.54 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 16:35.40 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1088827360.dsl.bell.ca) |
| 16:35.46 | *** part/#asterisk lirakis (n=lirakis@65.200.189.220) |
| 16:36.25 | *** join/#asterisk lirakis (n=lirakis@65.200.189.220) |
| 16:39.14 | yang | RoyK: but my phone does like to connect to asterisk this way, what am i doing wrong? |
| 16:39.55 | RoyK | yang: I'm not sure if I understand what you're trying to do |
| 16:42.10 | *** join/#asterisk mog (n=mog@nat/digium/x-c33073a690a951ff) |
| 16:42.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:42.40 | yang | RoyK: if you scroll up there is my paste link |
| 16:42.41 | *** join/#asterisk nighty^ (n=nighty@66-163-28-100.ip.tor.radiant.net) |
| 16:43.18 | RoyK | hm |
| 16:43.21 | RoyK | couldn't find it |
| 16:48.09 | *** join/#asterisk HarryR (n=Harry@77.240.56.17) |
| 16:48.59 | Havokmon | Hmmm once I set 'externip' - every sip client must be nated ? Looks like if I use a hostname, it doesn't send the hostname to resolve, but resolves it itself :/ |
| 16:49.58 | *** join/#asterisk JaminCollins (n=jcollins@151.101.5.95) |
| 16:51.23 | jblack | ~nat |
| 16:51.24 | jbot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 16:51.38 | jblack | That's not what I wanted |
| 16:53.53 | *** join/#asterisk tsabi (n=tsabi@gw.creditexpress.hu) |
| 16:54.12 | JaminCollins | Is it possible with 1.4 to have the agent configuration stored in a realtime DB? |
| 16:56.09 | yang | RoyK: http://openpaste.org/en/4585/ |
| 16:56.26 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:56.44 | Qwell | ~sipnat |
| 16:56.44 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:56.45 | yang | RoyK: this setup i have on asterisk, but if i enter the user-id number into my phone I cannot register on asterisk |
| 16:57.28 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:57.32 | RoyK | yang: that's understandable - to be able to register, host must be set to dynamic |
| 16:57.46 | RoyK | and don't use insecure unless you have to |
| 16:58.10 | nny_1 | is it sane or possible to run 2 X 4 port TDM cards (or 2 X 8, 1 8 port, 1 4 port) on the PCI bus without timing issues? Read somewhere in the previous book that could be a problem, wasn't surte if it still is... |
| 16:59.00 | Havokmon | Qwell: yeah, except if your phone is behind nat with the *, it can't connect to it, because it tries to connect to the external IP. |
| 16:59.03 | [TK]D-Fender | nny_1: 2 cards is usually fine |
| 16:59.12 | Havokmon | not a biggie, I gotta setup another firewall, but a hassle :/ |
| 16:59.31 | nny_1 | [TK]D-Fender: cool thanks |
| 17:02.46 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:02.55 | yang | RoyK: but if I set host = dynamic , how will asterisk know , how to connect to voip provider? |
| 17:03.25 | RoyK | well |
| 17:03.27 | RoyK | ~book |
| 17:03.28 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:03.32 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:03.43 | RoyK | obviously, you have to use one sip peer for your client and one for the provider |
| 17:03.46 | *** join/#asterisk jblack (n=jblack@pool-71-181-185-97.sctnpa.east.verizon.net) |
| 17:04.22 | yang | RoyK: I dont understand, so 2 contexts ? |
| 17:05.12 | RoyK | you can probably share the context in extensions.conf if you like, but that'll be more work and probably uglier, but you can't share the sip peer in sip.conf |
| 17:05.55 | yang | So I just copy&paste that sip context and I make one with host=IP and another with host=dynamic? |
| 17:05.55 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 17:06.05 | nny_1 | any value between the AEX and TDM series cards (pci express vs. pci) |
| 17:06.10 | [TK]D-Fender | yang: yes, seperate contexts. * takes calls in from each of these different sources and treats them how YOU tell it to based on the context you point them to and the extens you put in. |
| 17:06.12 | nny_1 | ooh |
| 17:06.17 | [TK]D-Fender | yang: * is a ... |
| 17:06.19 | [TK]D-Fender | ~b2bua |
| 17:06.19 | jbot | extra, extra, read all about it, b2bua is a back 2 back user agent |
| 17:06.19 | nny_1 | nm hardware echo on the AEX |
| 17:06.25 | [TK]D-Fender | yang: go read THE BOOK |
| 17:07.09 | [TK]D-Fender | and don't use the words "SIP CONTEXT" that is "SIP ACCOUNT ENTRY". |
| 17:07.29 | yang | yeah, i am reading the book slowly, it only has 600 pages |
| 17:07.33 | [TK]D-Fender | yang: the term "context" is pretty much reserved in reference to the dialplan (extensions.conf" |
| 17:07.52 | [TK]D-Fender | yang: We'll try to make the next one bigger for you then.... |
| 17:08.12 | FlatFoot | hello all , what is the correct format for the following on 1.4.`` ? |
| 17:08.15 | workinit | [TK]D-Fender, I was talking about 2 totally seperate systems earlier. :) |
| 17:08.20 | FlatFoot | GotoIf($[${DIALSTATUS}=BUSY]?101:3) |
| 17:08.53 | RoyK | FlatFoot: first of all, don't use 101:3 - set labels instead |
| 17:09.10 | [TK]D-Fender | workinit: Then just setup 1.4 as 1.2 is no longer supported. |
| 17:09.26 | lmadsen | FlatFoot: use priority labels |
| 17:09.31 | [TK]D-Fender | FlatFoot: Exactly that |
| 17:09.34 | nny_1 | lol crap, the asterisk AEX series descriptions all read the same for varying pricing |
| 17:09.47 | nny_1 | er digium/asterisk |
| 17:09.53 | workinit | [TK]D-Fender, admin wont upgrade because he'd have to upgrade the compiler and have to reset that entire box back up. |
| 17:09.54 | FlatFoot | RoyK: ok i shall investigate . it's just that * is moaning .... unexpected '=', expecting $end |
| 17:09.55 | lmadsen | exten => 100,n(my_label),NoOp() |
| 17:10.06 | [TK]D-Fender | workinit: That makes NO sense... |
| 17:10.12 | workinit | you're telling me |
| 17:10.16 | workinit | sounds like he just doesnt want to do it |
| 17:10.28 | [TK]D-Fender | FlatFoot: pastebin your entire exten. |
| 17:10.34 | RoyK | FlatFoot: GotoIf($["${DIALSTATUS}"="BUSY"]?101:3) |
| 17:10.47 | [TK]D-Fender | FlatFoot: probably becasue DIALSTATUS is null. |
| 17:10.48 | RoyK | oh, and drop priority numbers |
| 17:10.51 | lmadsen | ya, ${DIALSTATUS} must have been empty |
| 17:10.57 | nny_1 | lol seriously someone go to http://store.digium.com/products.php?category_id=42 and spot the differences between the single Quad FXO ones |
| 17:10.57 | *** join/#asterisk Victor_Yure (n=aaa@200.166.132.131) |
| 17:11.04 | RoyK | like lmadsen said, use (mylabel) and gotoif....?mylabel |
| 17:11.06 | lmadsen | anytime you have a variable that could be empty, enclose it in quotes |
| 17:11.10 | FlatFoot | i see i had missed the "" around DIALSTATUS |
| 17:11.24 | workinit | [TK]D-Fender, I can explain more once my co-worker gets here |
| 17:11.29 | RoyK | all variables may be null |
| 17:11.33 | workinit | maybe i'm saying it wrong |
| 17:11.35 | RoyK | you never know |
| 17:11.48 | jblack | Oh what a disaster. |
| 17:12.08 | FlatFoot | RoyK: i am going through the process of upgrading from 1.2.13 new box etc |
| 17:12.13 | FlatFoot | trying to clean up a bit |
| 17:12.15 | [TK]D-Fender | FlatFoot: pastebin the entire exten (not just THAT priority) |
| 17:12.25 | [TK]D-Fender | jblack: O? |
| 17:12.46 | FlatFoot | ~pb |
| 17:12.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:13.21 | lmadsen | RoyK: unless you check with EXISTS() or ISNULL() (which I do on a regular basis in my dialplan because I'm pulling data from the DB, and then I check to see if it exists before I do any comparisons and then send it to an error handler if I have an unexpected value) |
| 17:13.24 | yang | RoyK: http://openpaste.org/en/4587/ |
| 17:13.32 | jblack | Those changes that we did lastnight to my dialplan have made debugging a nightmare. :) |
| 17:13.39 | FlatFoot | TK , http://pastebin.com/d14f82b44 |
| 17:14.39 | RoyK | yang: read the docs, please. you have to understand that asterisk treats all sip users the same, whether they're a $200k switch or a softphone. asterisk bridges between them as you've told it to in extensions.conf |
| 17:15.40 | nny_1 | lol go digium go |
| 17:15.54 | yang | RoyK: So I need to modify extensions.conf ? |
| 17:15.58 | nny_1 | the differences are the same as the tdm series, but their marketing company just c/pd the descriptions |
| 17:16.08 | [TK]D-Fender | FlatFoot: Stop checking dialstatus BEFORE your dial <-- |
| 17:16.10 | nny_1 | so AEX844E is 1 FXO 1 FXS |
| 17:16.18 | jbot | send killer squirrels to attack yang for RTFM, courtesy of royk |
| 17:16.19 | nny_1 | 840 1 FXS 1 FXO |
| 17:16.23 | nny_1 | er 1 FXS 0 FXO |
| 17:16.31 | jbot | chops yang in half with a free AOL CD for _not_ RTFM, courtesy of royk |
| 17:16.33 | nny_1 | but the descriptions all read for 0 FXS 1 FXO |
| 17:16.36 | nny_1 | should i tell them? |
| 17:16.40 | nny_1 | heh' |
| 17:17.05 | FlatFoot | [TK]D-Fender: yep that might make sense |
| 17:17.36 | yang | RoyK: If I just knew what you are trying to suggest to me, I understand the read the book part |
| 17:18.19 | [TK]D-Fender | yang: And why do you have 2 entries for the same thing? |
| 17:19.36 | [TK]D-Fender | FlatFoot: While you're at it : SetMusicOnHold is deprecated, Dial option "r" = EVIL, Priority jumping after dial?! UMMM No. Thats what dialstatus is for. |
| 17:19.41 | yang | ok you suggested separate contexts...good |
| 17:19.50 | yang | I have the same context no wi can see it |
| 17:20.10 | [TK]D-Fender | FlatFoot: Voicemail2(u4367) should also be Voicemail2(4367@default,u) |
| 17:20.50 | yang | But I am trying to use asterisk as a bridge.... |
| 17:22.26 | FlatFoot | [TK]D-Fender: see i told you i was trying to clean up , this box hasn't really been looked at since v 1.2.xx |
| 17:22.50 | [TK]D-Fender | yang: Thats fine... a call comes in to * via 1 entry in sip.conf, goes into the DIALPLAN which processes the call and makes the decision to dial OUT another entry in some way |
| 17:23.11 | coppice | I generally use this as a bridge. it works very well for me - http://en.wikipedia.org/wiki/Image:Bridge1.jpg |
| 17:23.48 | yang | ah |
| 17:23.49 | yang | ok |
| 17:24.21 | _ShrikE | ls |
| 17:25.52 | yang | [TK]D-Fender: what I need is just extension like exten => 2970,1,Dial(SIP/2<here another SIP account number>) |
| 17:26.11 | yang | and then it will forward |
| 17:26.24 | *** join/#asterisk s0lid (n=_freq@210.213.243.83) |
| 17:29.04 | jblack | I need a way to see what extensions are in a context after includes are taken into account. |
| 17:30.19 | [TK]D-Fender | yang: and it is not "forwarding". * is merely calling the destination and bridging the call and is sitting in the middle |
| 17:30.33 | dbtid_ | in the section of the telephony book that discusses "Settup Up the Dialplan for Some Test Calls" |
| 17:30.34 | [TK]D-Fender | yang: But yes, and extension similar to that |
| 17:30.40 | dbtid_ | it creates an echo extension at 500 |
| 17:30.55 | dbtid_ | i can dial 500, debug tells me things are happening, but there's no echo on my phone. |
| 17:31.03 | dbtid_ | what have i done wrong? |
| 17:31.05 | [TK]D-Fender | jblack: there is no composite way of seeing them all together... |
| 17:31.45 | dbtid_ | if i do console dial i hear "weasels have eaten our telephone system" |
| 17:31.51 | jblack | ok |
| 17:34.40 | [TK]D-Fender | dbtid_: console dial is not your phone. its using your soundcard. |
| 17:34.48 | dbtid_ | i realize that |
| 17:34.59 | dbtid_ | i guess what i was pointing out is the tt-weasel exists. |
| 17:35.06 | [TK]D-Fender | dbtid_: What phone are you using exactly, and how exactly is it connected to *? |
| 17:35.20 | dbtid_ | well, it's a hitachi wirelessip 5000 e-a |
| 17:35.30 | dbtid_ | it's connecting via SIP |
| 17:35.34 | dbtid_ | (i hope that answers your question) |
| 17:35.35 | [TK]D-Fender | dbtid_: Located where relative to Asterisk? |
| 17:35.38 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 17:35.41 | dbtid_ | about 3 feet away. |
| 17:35.46 | dbtid_ | maybe 2 |
| 17:35.55 | [TK]D-Fender | dbtid_: NETWORKING WISE. |
| 17:36.07 | jblack | Nothing can dial anything. |
| 17:36.16 | [TK]D-Fender | jblack: huh? |
| 17:36.17 | dbtid_ | phone -> wireless AP -> etherswitch -> asterisk host |
| 17:36.29 | [TK]D-Fender | jblack: Everything was perfect alst night. What happened? |
| 17:36.37 | jblack | I slept. |
| 17:36.39 | drmessano | [TK]D-Fender |
| 17:36.46 | dbtid_ | sip debug reveals email-like sip call messages |
| 17:36.58 | jblack | I'll figure out what's going on. |
| 17:37.05 | [TK]D-Fender | dbtid_: set up a softphone to test it with |
| 17:37.12 | drmessano | can you specificy CID in a dial string? |
| 17:37.14 | jblack | At this moment I'm ensuring things are dropping into the right context. |
| 17:37.15 | dbtid_ | i'd be happy to |
| 17:37.24 | jblack | after that, I'll have to go over those includes for outgoing. |
| 17:37.26 | dbtid_ | i have the ekiga.net softphone. would that work? |
| 17:37.28 | [TK]D-Fender | drmessano: No. "show function CALLERID" |
| 17:37.33 | [TK]D-Fender | dbtid_: Sure |
| 17:37.36 | dbtid_ | thanks |
| 17:37.38 | drmessano | Damn |
| 17:37.46 | [TK]D-Fender | drmessano: why "damn"? |
| 17:37.58 | [TK]D-Fender | drmessano: Whats the need for it to be passed through app_dial? |
| 17:38.09 | drmessano | Well, if I am using a custom extension and I want specific a CID for that call |
| 17:38.17 | drmessano | a specific |
| 17:38.24 | drmessano | and my provider lets me spoof |
| 17:38.34 | [TK]D-Fender | drmessano: Then set it before your dial. |
| 17:38.35 | jblack | aha |
| 17:38.44 | workinit | If someone were to help me get Internal extensions to dial external extensions what conf files would they need info from? It works the other way around. |
| 17:38.54 | jblack | no |
| 17:38.57 | [TK]D-Fender | workinit: Chapter 5 <--------- |
| 17:39.09 | workinit | 2nd edition book |
| 17:39.13 | workinit | ? |
| 17:40.03 | [TK]D-Fender | workinit: BOTH |
| 17:40.07 | workinit | ty |
| 17:40.22 | jblack | I found it. I _do_ need to answer before dialing out |
| 17:41.02 | jblack | we took too many answers out |
| 17:42.13 | drmessano | [TK]D-Fender |
| 17:42.24 | drmessano | I got this from someone on twitter.. |
| 17:42.26 | drmessano | Time to re-familiarize myself with Asterisk. I'm now in charge of the office PBX. |
| 17:42.58 | drmessano | I'll direct him your way, k? |
| 17:43.11 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:43.58 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:47.00 | jblack | Ok tk, I found the problem with a high level of certainty. |
| 17:47.19 | jblack | I _must_ answer before I call dial. |
| 17:48.45 | [TK]D-Fender | jblack: definielty shouldn't... |
| 17:49.05 | [TK]D-Fender | drmessano: My rates are very accessable ;) |
| 17:49.05 | jblack | Ok, then it's masking a different problem, I guess. |
| 17:49.28 | jblack | If I plop an answer before extensions, I can reach them, if I don't, I can't. |
| 17:51.21 | [TK]D-Fender | jlbFunny it was all working yesterday... |
| 17:53.25 | dbtid_ | [TK]D-Fender: ok, i'm using the ekiga client. if i call sip:500@10.0.24.108 i see the call on asterisk, but it doesn't last very long. i get the message "Call from '' to exention 500 rejected because extension not found." |
| 17:53.56 | [TK]D-Fender | dbtid_: Go fix your dialplan. |
| 17:53.57 | dbtid_ | when i do 'dialplan reload' it tells me "Added extension '500' priority [1-3] to internal' |
| 17:54.25 | [TK]D-Fender | dbtid_: then clearly thats not the context that ekiga was told you use. |
| 17:54.34 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:55.00 | dbtid_ | well, i called 'sip:s@<ip addr>' and i heard the weasel message |
| 17:55.11 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 17:56.54 | dbtid_ | the example dial plan has a [default] section which defines 's' and an [internal] section which defines 500 |
| 17:57.00 | dbtid_ | is 500 not an extension? |
| 17:57.26 | [TK]D-Fender | dbtid_: Go prove what context its searching in |
| 17:57.33 | dbtid_ | i don't know how to do that |
| 17:57.50 | [TK]D-Fender | dbtid_: You can show me 100 dialplan contexts each with a perfectly valid "500" that will never get called. |
| 17:58.03 | [TK]D-Fender | dbtid_: "sip debug". Look at the call come in |
| 17:58.12 | *** join/#asterisk AndyGraybeal (n=mind@casanueva.wifi.frognet.net) |
| 17:58.17 | *** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk) |
| 17:58.32 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:58.34 | AndyGraybeal | russellb, i got that Linksys SPA3102... ordered it last night, hopefully i'll get it by friday |
| 17:58.52 | AndyGraybeal | russellb, i'm excited, but it looks like a steep learning curve for asterisk |
| 17:58.57 | *** join/#asterisk Victor_Yure (n=aaa@200.166.132.131) |
| 17:59.18 | [TK]D-Fender | AndyGraybeal: you've been here for what.. at least a year now? |
| 17:59.33 | AndyGraybeal | [TK]D-Fender .. in and out .. asking questions based on what i have read |
| 17:59.40 | AndyGraybeal | i've never used asterisk in my life though |
| 17:59.45 | AndyGraybeal | i've only been researching using it |
| 17:59.50 | AndyGraybeal | so finally, i'm taking my first step |
| 17:59.53 | [TK]D-Fender | AndyGraybeal: And you never got off your ass to actually install * and even jsut test with a softphone? |
| 18:00.07 | AndyGraybeal | [TK]D-Fender, didn't realize i could do that |
| 18:00.16 | [TK]D-Fender | AndyGraybeal: .... no comment :| |
| 18:00.30 | AndyGraybeal | [TK]D-Fender, that's okay, i am the grand idiot |
| 18:00.57 | jblack | [TK]D-Fender: Yeah, everything worked last night, until a full restart of *. |
| 18:01.01 | [TK]D-Fender | AndyGraybeal: That was you cue to walk away from the topic queitly ;) |
| 18:01.15 | jblack | Then I started having problems with ekiga, which needs insecure=auth, and other such things. |
| 18:01.17 | AndyGraybeal | [TK]D-Fender, i don't mind broadcasting it, but thanks for the hint |
| 18:01.31 | [TK]D-Fender | jblack: Makes no sense... we did full reloads andd since the only 3 configs you use are indeed full restarted from that, it shouldn't be the issue |
| 18:01.31 | jblack | It's all good now. I just guarded dials with answers. |
| 18:01.40 | [TK]D-Fender | AndyGraybeal: All part of the service :) |
| 18:01.46 | AndyGraybeal | :P |
| 18:01.51 | [TK]D-Fender | jblack: jsut the outside ones I hope |
| 18:01.58 | jblack | I was on * the whole call. perhaps there's a subsystem that doesn't reload during calls. |
| 18:02.15 | jblack | I need it for internal too. with answer, I can call between internals. without, I cannot |
| 18:02.17 | [TK]D-Fender | jblack: Nope, not amongst extensions.con, sip.conf, iax2.conf |
| 18:02.30 | [TK]D-Fender | jblack: Tha makes no sense.... |
| 18:02.31 | *** join/#asterisk mog (n=mog@nat/digium/x-ac4cba2839785d86) |
| 18:02.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:02.40 | AndyGraybeal | [TK]D-Fender, i've got the old modem bandwidth anyway.. i could never use a softphone at home :( |
| 18:03.15 | [TK]D-Fender | AndyGraybeal : softphone direct to * same lan. Who said anything about "internet"? |
| 18:03.21 | jblack | elf needs food badly. |
| 18:03.34 | AndyGraybeal | [TK]D-Fender, ah never thought of it that way either |
| 18:04.27 | [TK]D-Fender | Orc kicks Elf in the nads for 3d20 + 5,000,000 damage!!!! |
| 18:04.33 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 18:04.50 | AndyGraybeal | is digium's tdm400p better than linksys spa3102 ? |
| 18:05.35 | AndyGraybeal | (not talking about the router part on the spa3102 at all btw) |
| 18:05.37 | [TK]D-Fender | AndyGraybeal: I'd say yes, dpending on your application. |
| 18:05.56 | AndyGraybeal | [TK]D-Fender, okay cool... but i can't afford that sucker, and this way atleast i can afford it |
| 18:06.05 | AndyGraybeal | i'm thinking about getting the digium for the business then |
| 18:06.12 | [TK]D-Fender | AndyGraybeal: Yes, and I was the one who referred you to it. |
| 18:06.55 | dbtid_ | [TK]D-Fender: have you ever gone through the book "Asterisk - The Future of Telephony?" Are you familiar with it's order? |
| 18:06.57 | AndyGraybeal | [TK]D-Fender i know i know :) |
| 18:07.13 | [TK]D-Fender | dbtid_: I've scanned it before. What about it? |
| 18:07.25 | dbtid_ | well, i'm going through it, following its examples and instructions. |
| 18:07.49 | dbtid_ | it leads one to believe that when i get through certain sections, by following those examples, things like calling extension 500 should work. |
| 18:08.24 | dbtid_ | if you've never actually gone through it, and don't understand what it's leading people to do, then you don't understand the lack of depth that the book has up to a certain point. |
| 18:08.29 | *** join/#asterisk trippss (n=ss@72.20.150.196) |
| 18:09.45 | nny_1 | anyone know when the cookbook is gonna be out? |
| 18:10.02 | *** join/#asterisk gaero (n=theo@unaffiliated/balkantools) |
| 18:10.08 | gaero | hi there |
| 18:10.09 | [TK]D-Fender | dbtid_: and I told you to enable sipd debug on * CLI and that would revel what SIP peer it matched against, and what dialplan contextit was searching in. |
| 18:10.36 | gaero | does somebody know the telecommunications protocols used in algeria ? |
| 18:10.38 | [TK]D-Fender | dbtid_: And strangely I don't have a sign of you having found it or putting it in a pastebin for me to point out what you missed. |
| 18:10.40 | [TK]D-Fender | ~pb |
| 18:10.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:10.42 | [TK]D-Fender | ^^^^^^^^^ |
| 18:10.42 | dbtid_ | well, you didn't reveal all that information; you just said to enable sip debug; also, dialplans are VERY sketchily mentioned. |
| 18:10.57 | dbtid_ | [TK]D-Fender: i'd be happy to point out what section of the book i'm working from, or paste it for you. |
| 18:11.15 | [TK]D-Fender | dbtid_: just enable sip debug, try your call again and pastebin the whole mess |
| 18:11.15 | mvanbaak | no, we need your actual implementation |
| 18:12.05 | dbtid_ | ok |
| 18:12.28 | *** join/#asterisk h4lt (n=Gustavo@geness.funcitec.rct-sc.br) |
| 18:12.33 | nny_1 | [TK]D-Fender: it's actually a goose, stuffed inside a turkey, stuffed inside a quail.. the last part gets messy |
| 18:12.46 | dbtid_ | for which call? the 500 or the 's' call? |
| 18:12.56 | dbtid_ | 's' works -- i hear the weasel message |
| 18:12.58 | dbtid_ | 500 does not |
| 18:13.16 | [TK]D-Fender | dbtid_: the 500 that doesn't work. And while you're at it include the entire context you think it should ahve been looking in. |
| 18:13.16 | *** join/#asterisk Greek-B0y (n=email@41.221.58.5) |
| 18:13.56 | [TK]D-Fender | dbtid_: And never both doing "dial" from the console. Always use an actual configured device like a softphone, etc |
| 18:14.38 | dbtid_ | ok |
| 18:14.44 | dbtid_ | what's the "context" you keep talking about? |
| 18:14.49 | dbtid_ | my extensions.conf? |
| 18:15.19 | nny_1 | wow god |
| 18:15.27 | workinit | [TK]D-Fender, Thanks for the help. My extensions.conf was missing the extensions we were adding under [incoming] |
| 18:15.40 | nny_1 | I would hate to be the company that uses a TDM2406E, and pays for 24 analog lines vs T1, etc |
| 18:15.42 | dbtid_ | [TK]D-Fender: http://rafb.net/paste/p/TakaTi27.html |
| 18:15.45 | [TK]D-Fender | dbtid_: Yes. |
| 18:15.52 | dbtid_ | bah |
| 18:15.53 | dbtid_ | one sec |
| 18:16.04 | dbtid_ | http://rafb.net/p/TakaTi27.html |
| 18:16.17 | dbtid_ | that's the call debug |
| 18:16.41 | [TK]D-Fender | dbtid_: first : Found no matching peer or user for '10.0.24.131:5086' <-- * can't ID a sip peer entry to match your phone. |
| 18:16.54 | [TK]D-Fender | dbtid_: And can you please pastebin your dialplan... |
| 18:16.58 | dbtid_ | sure |
| 18:17.06 | [TK]D-Fender | dbtid_: and heck while you're at it you sip.conf masking only passwords. |
| 18:17.11 | *** join/#asterisk FOMICIANO (n=FOMICIAN@189-015-213-106.xd-dynamic.ctbcnetsuper.com.br) |
| 18:17.26 | dbtid_ | extensions.conf is http://rafb.net/p/OqspjA90.html |
| 18:17.40 | dbtid_ | i didn't understand your last statement. |
| 18:18.07 | [TK]D-Fender | dbtid_: No the next thing is that : Looking for 500 in default (domain 10.0.24.108) <- its looking for 500 in [default]. And clearly it isn't in there. |
| 18:18.25 | [TK]D-Fender | dbtid_: And your sip.conf file and change only the passwrods for security reasons |
| 18:19.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:19.11 | [TK]D-Fender | jblack: Do you know what ingredient makes up 53% of a McDonalds Chicken McNugget? |
| 18:19.24 | file | sawdust? |
| 18:19.26 | jblack | I would imagine either fat or water |
| 18:19.30 | [TK]D-Fender | jblack: CORN <- |
| 18:19.45 | jblack | GOOD! Corn needs to learn a lesson too. |
| 18:19.48 | dbtid_ | [TK]D-Fender: the extensions.conf file gives [default] (defines 's') and [internal] which defines 500 |
| 18:19.54 | [TK]D-Fender | jblack: Between the "breading", HFCS, oils, etc... its friggen CORN. |
| 18:20.15 | dbtid_ | and why do i need to define a sip peer? i configured it to use 1001 which is in sip..conf which i will paste |
| 18:20.20 | [TK]D-Fender | dbtid_: [default] doesn't have the "500" you dialed. Welcome to "a dead end". |
| 18:20.25 | nny_1 | is the asterisk now appliance easy to customize? or is it designed to only be configured via the web interface |
| 18:20.35 | [TK]D-Fender | dbtid_: Doesn't matter if you have one in [internal], it isn't looking in three. |
| 18:20.39 | nny_1 | ~aa |
| 18:20.39 | jbot | aa is, like, at gg |
| 18:20.41 | jblack | come to pappa, little corn nuggets |
| 18:20.47 | dbtid_ | i understand |
| 18:20.51 | AndyGraybeal | [TK]D-Fender, thanks alot for the help though.. this is going to be great, i'm pretty excited. |
| 18:20.58 | dbtid_ | i think i got it now [TK]D-Fender |
| 18:21.00 | [TK]D-Fender | nny_1: The appliance is just an ARm based box with * + AsteriskGUI |
| 18:21.12 | dbtid_ | sip.conf is at http://rafb.net/p/aGVenN56.html |
| 18:21.20 | dbtid_ | it has a [general] which says 'context=default' |
| 18:21.27 | dbtid_ | if i said 'context=internal' |
| 18:21.34 | dbtid_ | then it would be able to find 500, correct? |
| 18:21.49 | *** join/#asterisk grEvenX (n=even@85.80-203-122.nextgentel.com) |
| 18:21.52 | nny_1 | [TK]D-Fender: yeah i have an ip04 here with astfin.. just trying to match apples to apples.. wasn't sure where a standard box with a real proc and ram/ custom asterisk stood against the AA |
| 18:21.59 | jblack | [TK]D-Fender: I believe it. I think that's a good thing. It means I'm getting veggies too. ;) |
| 18:22.09 | [TK]D-Fender | dbtid_: We should look back to the first problem : you didn't set Ekiga up right an * doesn't see ti as being one of those accounts you set up |
| 18:22.51 | [TK]D-Fender | dbtid_: thing is your call shouldn't be falling back to [general]. It should match a device you defined in sip.conf, but it isn't |
| 18:23.06 | [TK]D-Fender | dbtid_: So go fix your ekiga |
| 18:23.27 | [TK]D-Fender | dbtid_: Then the rest looks fine. |
| 18:24.02 | [TK]D-Fender | dbtid_: And remember that "dial" from the * CLI = waste of time as a test. Doesn't mean it'll ever get hit by anything else. |
| 18:24.22 | [TK]D-Fender | nny_1: More like comparing lemons to lemons ;) |
| 18:24.27 | nny_1 | [TK]D-Fender: heh |
| 18:25.16 | nny_1 | [TK]D-Fender: makes it hard to sell an 8 port analog card and box when the card alone is 800 and the hardware is 400-500 |
| 18:25.40 | nny_1 | [TK]D-Fender: figure the AA isn't as expandable/ customizable |
| 18:26.30 | [TK]D-Fender | nny_1: the appliace has limited storage, a weak CPU that won't survive transcoding, no expandability and is a pidgeon-holed dead-end. Oh... and one with heat issues. That about sum it up? |
| 18:26.57 | [TK]D-Fender | nny_1: Any if any 1 part of it dies.. its all dead. |
| 18:28.10 | nny_1 | [TK]D-Fender: trust me, I have no desire to install or support them, just looking at such things as pricing points.. i will say they offer full replacement as well as 15 incidents etc etc free email support blah blah for 899 a year (with the AA + 1 year support platinum (8 port analog) at 2500.00 |
| 18:28.37 | nny_1 | [TK]D-Fender: but yeah I can see where the minute you want it to do something nifty, it shits the bed |
| 18:29.12 | *** part/#asterisk JaminCollins (n=jcollins@151.101.5.95) |
| 18:29.16 | nny_1 | [TK]D-Fender: the 500 MHZ blackfin protype we have here doesn't even handle CID yet, and the whole under system is a slew of engineering feats just to get it to run on such a small proc |
| 18:30.45 | jblack | I think I'm going to replace the old copper plant in this house with cat5. |
| 18:30.46 | [TK]D-Fender | jblack: Sure thing Wile-E |
| 18:31.17 | jblack | I'm not _sure_ yet, but I think that the extensions were all run from source to destination. |
| 18:31.34 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 18:31.44 | [TK]D-Fender | jblack: Cat5e is always a good thing... and you can still run your Rj11's over it. |
| 18:31.59 | [TK]D-Fender | jblack: All in how you handle your patch panel |
| 18:32.03 | jblack | Does rj11 need 5e, rather than 5? |
| 18:32.28 | [TK]D-Fender | jblack: no, gigbit does, so if you're going to do a job.... don't skimp |
| 18:32.37 | [TK]D-Fender | jblack: Cat6 even |
| 18:34.25 | jblack | with luck, I can get a punchdown board at lowes or home depot. |
| 18:34.52 | AndyGraybeal | the telephony world prior to Asterisk... sounds like the HVAC world right now |
| 18:35.14 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:35.23 | Qwell | AndyGraybeal: are you proposing HoIP? |
| 18:35.28 | jblack | AndyGraybeal: Huh? No, this a asterisk problem, in "last 100 ft" style |
| 18:35.42 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 18:36.01 | AndyGraybeal | Qwell, lol :) |
| 18:36.06 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 18:36.07 | jmls | when a call is hung up, does the h extension only execute on the originating leg of the call ? |
| 18:36.38 | jblack | Andy wants us to stick wireless adaptors in our brain so that we can skip the hardware entirely. ;) |
| 18:36.49 | [TK]D-Fender | jmls: it only executes if the dialed end hung up. |
| 18:36.59 | AndyGraybeal | jblack: i'm a little confused at your comments! |
| 18:37.11 | jmls | yeah, but on which channel ? The originating ? |
| 18:37.12 | jblack | which I find tempting. I could google while buried under the blankets. Porn would be amazing. |
| 18:37.22 | *** join/#asterisk mog (n=mog@nat/digium/x-80ad0c7d64f1981b) |
| 18:37.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:37.44 | jblack | AndyGraybeal: Your original reply to my having to rewire thing... "telephony world prior to asterisk" |
| 18:38.40 | [TK]D-Fender | jmls: as I said, on the one doing the Dial. |
| 18:39.03 | AndyGraybeal | jblack: i am comparing the telephony world to the hvac world... hvac needs an open source solution like asterisk |
| 18:39.16 | AndyGraybeal | even an open source community for hvac solutions |
| 18:39.58 | AndyGraybeal | hvac is extremely proprietary world.. and the technicians are very appauled to giving out any information to 'hobbyists'... |
| 18:42.18 | *** join/#asterisk cjk (n=cjk@d90-129-39-85.cust.tele2.lu) |
| 18:42.37 | cjk | hi, is it possible to update the calleridon an established call. this is quite important with attended transfers.... |
| 18:45.56 | jblack | ahhh. |
| 18:46.47 | AndyGraybeal | jblack :) |
| 18:49.08 | [TK]D-Fender | cjk: No. You should be doing Blind transfers if you want them to see the originating CID |
| 18:53.14 | *** join/#asterisk cjk (n=cjk@d90-129-39-85.cust.tele2.lu) |
| 18:53.42 | cjk | [TK]D-Fender, i understand the logic behind, but most traditional pabx'es support it. so users are used to it and asking for it |
| 18:54.07 | [TK]D-Fender | cjk: Asterisk doesn't, tell them to get used to it. |
| 18:57.32 | nny_1 | [TK]D-Fender: is that only with polycoms? I have heard that snoms can transfer the CID |
| 18:57.40 | Daviey | cjk: worth reading - http://www.freepbx.org/forum/freepbx/users/passing-caller-id-to-extensions-upon-transfer |
| 18:58.09 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 18:58.12 | cjk | Daviey, yep thats where i come from |
| 18:58.20 | [TK]D-Fender | nny_1: he's talking about * passing remote CID on an ATTENDED transfer. * doesn't support that period. We are not talking about blind transfers here. |
| 18:58.40 | nny_1 | [TK]D-Fender: ok |
| 18:59.24 | nny_1 | ahh nice statement from polycom |
| 18:59.40 | nny_1 | the Caller ID presented to the third person is the transferee's info when doing a blind transfer, but the transferor's info when you do an attended transfer. We've considered changing this, but everyone has a different idea of what's correct... -- |
| 18:59.41 | nny_1 | Stephen Sprunk |
| 18:59.41 | nny_1 | Polycom Systems Engineer |
| 19:00.27 | nny_1 | so it is by design, and thats "the way it is".. good enough, I can train a hamster to blind transfer... |
| 19:01.28 | jblack | but can you train the blind to transfer hamsters? |
| 19:01.44 | creativx | or can you transfer the blind together with the hamsters? |
| 19:01.57 | Daviey | bah, why would you want incoming CID on an attended :S |
| 19:02.11 | nny_1 | that seems to be the exact reason |
| 19:02.14 | jblack | And what do you do if you find out the hamsters are in the blind, during a transfer? |
| 19:02.56 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 19:02.57 | iratik | Alright..... Every softphone i've tried (I've tried over 20), is horrible. DIAX is working very well though... its simple and customizable...... but I can't seem to read the sip debug output to see why i can't receive an incoming call on it. I ran "sip set debug on" .... and copied the output of asterisk -r when i dialed 513(an IAX extension) from 512(sip extension)....... http://pastie.caboo.se/136805 ..... any idea what happened so th |
| 19:02.58 | iratik | at i'm getting unavailable --- ExtensionState4... can't see any headers though in the output |
| 19:03.08 | nny_1 | jblack: i am actually working on a blind hamster transfer right now.. I wish they would stop throwing them so hard though. It ruin the sound quality of the hamster |
| 19:04.23 | *** part/#asterisk jmls (n=jmls@81.138.42.77) |
| 19:04.34 | nny_1 | iratik: is that amportal or something? |
| 19:04.50 | iratik | I don't know what amportal is |
| 19:04.54 | fiXXXerMet | sip show peers shows my two devices as 'Unmonitored'. How do I monitor them, and what is the difference between monitored and unmonitored? |
| 19:05.12 | [TK]D-Fender | iratik: SIP DEBUG please... that CLI output is meaningless. |
| 19:05.18 | [TK]D-Fender | iratik: and I told you that yesterday |
| 19:05.24 | iratik | I ran set sip debug on |
| 19:05.37 | iratik | ahh |
| 19:05.39 | iratik | sip set debug on |
| 19:07.24 | iratik | okay.... its alot of output |
| 19:07.26 | iratik | but here it is |
| 19:07.26 | iratik | http://pastie.caboo.se/136807 |
| 19:08.18 | iratik | I probably need to filter for my IP |
| 19:08.35 | [TK]D-Fender | iratik: actually... deed IAX2 debug. that IAX/513 its calling -- Executing [s@macro-dial:10] Dial("SIP/512-096a1938", "IAX2/513|15|tr") in new stack |
| 19:08.35 | jblack | I wonder if a pci slot dust guard would make for an ok punchdown tool. |
| 19:08.42 | nny_1 | so whats dialparties.agi? From what I can google, it is a program that calls status from asterisk DB etc. and seems to be relates to "Asterisk Management Portal" |
| 19:08.51 | [TK]D-Fender | iratik: And do "iax2 show peer 513" please |
| 19:09.09 | [TK]D-Fender | nny_1: FreePBX BS <- |
| 19:09.18 | nny_1 | [TK]D-Fender: ahh ok ok.. thatswhatithought |
| 19:09.33 | nny_1 | ewww |
| 19:09.47 | iratik | [TK]D-Fender: here is iax2 show peer 513: http://pastie.caboo.se/136810 |
| 19:10.31 | [TK]D-Fender | iratik: Addr->IP : (Unspecified) Port 0 <--- not even registered |
| 19:10.48 | [TK]D-Fender | iratik: Status : UNKNOWN <- |
| 19:11.40 | iratik | ahhhhhhhhhh |
| 19:12.14 | iratik | I'm an idiot |
| 19:12.35 | creativx | atleast you said it yourself |
| 19:12.36 | creativx | :) |
| 19:12.53 | [TK]D-Fender | iratik: 11 steps to go... |
| 19:13.08 | iratik | it works! |
| 19:13.20 | iratik | DIAX has a "register" checkbox |
| 19:13.25 | iratik | no wonder it wasn't working |
| 19:13.28 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:14.25 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:16.41 | ProofPack | I'm having some difficulty with calls timing out and going to Asterisk voicemail when a Polycom Phone is set for call forwarding (sip) to a cell phone. Is there a way in the dialplan to detect this is happening and reset the call timer? |
| 19:17.10 | [TK]D-Fender | ProofPack: pastebin the call attempt with sip debug enabled and verbose 10 |
| 19:17.11 | [TK]D-Fender | ~pb |
| 19:17.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:17.13 | [TK]D-Fender | ^^^^^^^^^ |
| 19:22.54 | ProofPack | http://pastebin.com/d642b387b is one snipet of the CLI and dialplan |
| 19:24.38 | [TK]D-Fender | ProofPack: Ok, looks like on forward it is still bound by the 20s timeout on dial. |
| 19:24.43 | [TK]D-Fender | ProofPack: Nobody picked up in 20000 ms |
| 19:24.51 | dbtid_ | sigh; i hate meetings |
| 19:24.56 | ProofPack | and I am working on a capture with the sip debug |
| 19:25.08 | [TK]D-Fender | ProofPack: No need actually |
| 19:25.11 | ProofPack | yes, that is exactly what is happening. |
| 19:25.19 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 19:25.29 | dbtid_ | [TK]D-Fender: ok, i'll look at my ekiga setup; i thought i told it to use '1001' as its login |
| 19:25.34 | ProofPack | any known way to reset the timer if it is redirected? |
| 19:25.36 | dbtid_ | i think that's what you were getting at. |
| 19:25.43 | [TK]D-Fender | ProofPack: Only way to bypass that is to have it so that your exten leading to the PSTN does an Answer before doing its own dial. |
| 19:26.13 | [TK]D-Fender | dbtid_: Yes I've told you many times that Ekiga was not setup and * has no idea who you think it should be. |
| 19:26.41 | ProofPack | oh, very elegant, send the redirect to it's own context, answer the call and then patch it through... |
| 19:26.48 | dbtid_ | you keep forgetting that i'm trying to follow more or less a tutorial. |
| 19:27.05 | [TK]D-Fender | ProofPack: Or stop using the SIP redirect and make a dialplan based forwarding. |
| 19:27.33 | [TK]D-Fender | dbtid_: If you didn't setup your SIP client with the login credentials well, there isn't much that can be done. |
| 19:29.06 | dbtid_ | well, ekiga was told to use '1001' as its user and authentication. |
| 19:29.14 | dbtid_ | i have 1001 in sip.conf |
| 19:29.17 | ProofPack | [TK]D-Fender, I do plan, long term, on dialplan based forwarding. |
| 19:29.29 | [TK]D-Fender | dbtid_: not properly apparently. |
| 19:30.56 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 19:33.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:33.10 | *** join/#asterisk n0rus (n=zupeq@gw.valiyev.net) |
| 19:33.24 | [TK]D-Fender | dbtid_: So keep an eye on your SIP debug during test calls until you get your auth right |
| 19:33.44 | dbtid_ | how do you specify that 10.0.24.131 is a valid peer |
| 19:33.45 | dbtid_ | ? |
| 19:35.13 | [TK]D-Fender | dbtid_: Fill in the blanks right in ekiga and things will be fine. |
| 19:35.27 | n0rus | People, help me out here. I have two video SIP phones: one in Norway, the other one is in France. Both are NATed. Asterisk server (1.2) is running Linux with iptables. So far I have managed to make regular calls, they work great. However, the video does not start. The phone has three lines and the video works when I call to work (also video). So it looks like that the iptables firewall is blocking somewhere. |
| 19:35.59 | dbtid_ | [TK]D-Fender: which part of ekiga? there's a configuration assistant that says nothing about the sip server. there's the account, which has everything AFAIK it's supposed to have. |
| 19:36.02 | [TK]D-Fender | n0rus: here : |
| 19:36.04 | [TK]D-Fender | ~sipnat |
| 19:36.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:36.12 | dbtid_ | wait |
| 19:36.22 | dbtid_ | page 3 "username" on the sip server. that's the 1001 from the sip.conf right? |
| 19:36.29 | n0rus | been there, read that |
| 19:36.31 | dbtid_ | and i don't have a password b/c i'm just playing now. |
| 19:36.47 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:37.06 | [TK]D-Fender | dbtid_: cLOSE eKIGA, OPEN IT AGAIN AND IT SHOULD TRY TO REGISTER. sEE WHO ITS TRYING TO REGISTER AS |
| 19:37.13 | dbtid_ | ok |
| 19:37.19 | n0rus | do I have to forward RTP traffic to the phone or do I have to allow it on localhost (asterisk server=? |
| 19:38.52 | n0rus | all right, found the answer on http://www.aocomputing.net/?p=3 |
| 19:38.58 | n0rus | will try |
| 19:39.05 | n0rus | thans folks |
| 19:39.07 | n0rus | thanks |
| 19:39.14 | *** part/#asterisk dbtid_ (i=qwi0tu5z@cpe-71-72-252-171.columbus.res.rr.com) |
| 19:40.17 | *** join/#asterisk dbtid (n=dbtid@ip67-91-21-162.z21-91-67.customer.algx.net) |
| 19:40.27 | dbtid | [TK]D-Fender, http://rafb.net/p/vfAb9k18.html |
| 19:41.55 | dbtid | does this indicate a successful registration? |
| 19:42.44 | [TK]D-Fender | dbtid: That looks good. It should also not be scubscribing to * for MWI |
| 19:43.21 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:43.36 | dbtid | i don't see 'MWI' in there... what's MWI? |
| 19:44.06 | [TK]D-Fender | dbtid: Ok, don't worry about that... now try a call with it. you should only put "500" in the dial bad |
| 19:44.08 | [TK]D-Fender | bar* |
| 19:44.24 | dbtid | ok |
| 19:45.09 | dbtid | but i'm confused. i thought 500 wasn't a valid extension for 'default' context? |
| 19:45.09 | dbtid | i thought 's' was the extension from the extensions.conf [default] section? |
| 19:45.09 | [TK]D-Fender | dbtid: Correct, but now * seems to ID the phone properly and the [1001] entry DID point to the context that had it |
| 19:45.51 | dbtid | [1001] has context = phones |
| 19:45.55 | dbtid | is that what you mean? |
| 19:46.59 | dbtid | ah i think i see now |
| 19:47.08 | [TK]D-Fender | dbtid: Yes, and in your dialplan [phones] has "include => internal" which DOES have your 500 |
| 19:47.17 | dbtid | in sip.conf, [1000] and [1001] both have context=phones |
| 19:47.28 | dbtid | in extensions.conf, [phones] has include => internal |
| 19:47.31 | [TK]D-Fender | dbtid: Correct so either account should be cabale of dialing 500 |
| 19:47.34 | dbtid | and [internal] has exten 500 etc |
| 19:47.35 | [TK]D-Fender | capable* |
| 19:47.42 | dbtid | ok and 500 is the echo extension |
| 19:47.58 | dbtid | can you please explain the [default] in extensions.conf where 'exten => s,...'? |
| 19:48.04 | [TK]D-Fender | dbtid: better phrased : an exten you made that does an echo test |
| 19:48.12 | [TK]D-Fender | dbtid: not now... |
| 19:48.16 | dbtid | ok |
| 19:48.23 | [TK]D-Fender | dbtid: lets get this working with 500 first |
| 19:49.09 | nny_1 | just for the record some guy is trying to tell me that DMZ = Own Public IP for * setups |
| 19:49.11 | dbtid | i do not have a mic on my pc so i can see it connect, and it shows me it's connecting to Echo somehow |
| 19:49.16 | nny_1 | i cry bs |
| 19:49.25 | dbtid | if i try from the WiFi phone, i get the same types of messages when i dial 500 |
| 19:49.37 | dbtid | <------------> |
| 19:49.38 | dbtid | <PROTECTED> |
| 19:49.38 | dbtid | <PROTECTED> |
| 19:49.38 | dbtid | <PROTECTED> |
| 19:49.48 | dbtid | then the call gets destroyed after a time |
| 19:50.06 | dbtid | do those two lines indicate that * correctly set up the call? |
| 19:50.18 | [TK]D-Fender | dbtid: Excellent. I will consider that a full success. Sure you don't have a Mic, but your sip.conf and extensions.con are in agreement, and your Ekiga is set up right |
| 19:50.28 | dbtid | seems to be |
| 19:50.39 | [TK]D-Fender | dbtid: Congratulations on your first successful call. |
| 19:50.40 | dbtid | i'll bring in a Mic from home tomorrow |
| 19:50.41 | dbtid | thanks |
| 19:51.11 | dbtid | since i haven't read as far into the * book, some bits were missing from the sip.conf <-> extensions.conf |
| 19:51.14 | dbtid | but i see how that works now |
| 19:51.15 | nny_1 | i am looking for a way to make an asterisk box a router, without having to fsck directly with iptables (some kind of web based firewall package that doesn't have it's own distro, like Clarkconnect..) I could install asterisk on an CC box (it uses centos) but they have their own repos. etc etc. |
| 19:51.25 | nny_1 | but yeah DMS ! public IP AFAIK |
| 19:51.33 | dbtid | now i need to figure out how to make the echo part of this work with my Hitachi 5000-EA phone |
| 19:51.40 | dbtid | it dials, connects, but i hear nothing. |
| 19:52.26 | dbtid | i have a feeling that the [1000] is not right in sip.conf |
| 19:53.49 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 19:55.17 | nny_1 | seems smoothwall can be built on top of the system if needed |
| 19:55.31 | *** join/#asterisk bllzm (n=inforx@S0106006097940f68.vw.shawcable.net) |
| 19:55.56 | Havokmon | Hmm sip show peer has 'N' under nat, but there isn't anywhere I've set nat=no |
| 19:56.04 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:56.16 | bllzm | using agi/fastagi how can one stream a file, read DTMF input then get result in a variable, I can do this using cmd Read, but not sure how to emulate this inside agi, is it get_data ? |
| 19:56.18 | Havokmon | sip.conf nat=yes, and under each ext nat=yes What else is there? |
| 19:56.35 | Havokmon | under each ext in sip.conf that is. |
| 19:57.18 | Havokmon | qualify=yes is set as well |
| 19:58.18 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:58.57 | dbtid | ok, if i add extension 501 to extensions.conf and i set it up to play tt-weasel, i can dial 501 and everything works fine |
| 19:59.11 | dbtid | i have set up 500 per the example of the echo, but i hear nothing when i dial 500. |
| 19:59.43 | [TK]D-Fender | dbtid: call from your WifI to Ekiga |
| 20:00.27 | hmmhesays | I'm trying to figure out if there is a way to tell if a call was transfered from another phone or not |
| 20:00.38 | [TK]D-Fender | dbtid: And add an "Answer" before you call Echo |
| 20:00.51 | [TK]D-Fender | hmmhesays: channelvariables.txt <- |
| 20:01.18 | [TK]D-Fender | Havokmon: pastebin is your friend |
| 20:02.33 | cjk | hi, when doing a blind transfer it seems that its done out of the context where the user was when he dialed the first number, but i would like that it start out of the users default context. any idea? |
| 20:02.58 | hmmhesays | [TK]D-Fender: yeah i'm not transferring with asterisk with cmd transfer though |
| 20:07.49 | [TK]D-Fender | cjk: Transfers use the context of the transferrer |
| 20:08.23 | bllzm | anyone knows how to do cmd Read inside AGI using perl, I tried $agi->stream_file and $agi->get_data but they dont work |
| 20:08.45 | cjk | [TK]D-Fender, well lets say my context in sip.conf is DEFAULT, but when dialing an outside number i end up in context GLOBAL. When i do a blind transfer now, its not starting in DEFAULT but in GLOBAL |
| 20:09.05 | [TK]D-Fender | cjk: pastebin it.... |
| 20:09.07 | Havokmon | http://pastebin.com/m5714d566 - My nat doesn't seem to take |
| 20:11.12 | cjk | [TK]D-Fender, http://pastebin.com/m44acb5f2 |
| 20:11.45 | [TK]D-Fender | Havokmon: You are missing "qualify=yes" for those phones. |
| 20:12.08 | [TK]D-Fender | cjk: sip.conf please.... |
| 20:12.34 | hmmhesays | [TK]D-Fender: this one is a conundrum for sure |
| 20:12.44 | cjk | [TK]D-Fender, users are in realtime |
| 20:12.46 | hmmhesays | dumpchan() isn't showing me any variables I can use |
| 20:13.40 | [TK]D-Fender | cjk: well I need to see how they are setup and I don't see a tranfer in your CLI output. try again. |
| 20:14.02 | Havokmon | [TK]D-Fender, only 6000, because I was playing with that one. They're all behind nat, and all show the same issue |
| 20:14.17 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1088827360.dsl.bell.ca) |
| 20:14.24 | cjk | [TK]D-Fender, line 52 |
| 20:14.39 | [TK]D-Fender | Havokmon: and 6000 does not have nat=yes |
| 20:14.54 | Havokmon | yeah - because I was playing with it :P |
| 20:15.19 | Havokmon | 6000 is my pc, 6206 is right next to me, neither show as nat |
| 20:15.34 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:15.45 | [TK]D-Fender | haveAnd your configs are broken like I jsut told you |
| 20:15.59 | dbtid | [TK]D-Fender, in order to call Ekiga from my WiFi, i'd have to set up extensions for them in extensions.conf, right? |
| 20:16.13 | cjk | [TK]D-Fender, sip user paste here: http://pastebin.com/m472970b |
| 20:16.14 | [TK]D-Fender | dbtid: Yup |
| 20:16.16 | dbtid | thanks |
| 20:16.22 | Havokmon | broken? because I removed nat=yes for one sip entry? |
| 20:16.34 | Havokmon | * can't read the next one entirely? |
| 20:17.10 | hmmhesays | so has anyone tried out counterpaths new soft phone? |
| 20:17.11 | [TK]D-Fender | Havokmon: All of those phones had better have both. |
| 20:17.44 | Havokmon | They did before I realized it wasn't working. Then I made changes to see what would happen. |
| 20:18.14 | AndyGraybeal | are there currently any open hardware projects that show "how to build your own fxo box?" |
| 20:18.54 | hmmhesays | throw an fxo card inside asterisk/callweaver/fs? |
| 20:18.57 | dbtid | [TK]D-Fender, adding the Answer() before Echo() worked perfectly, as i'm sure you knew it would; i wonder if the author knows that... |
| 20:18.59 | Corydon76-dig | First step is to buy a TDM804... |
| 20:19.31 | *** join/#asterisk G-nerd (n=AskMe@dhcp-077-249-041-129.chello.nl) |
| 20:20.05 | G-nerd | Yo wazup G's |
| 20:20.06 | AndyGraybeal | hmmhesays: no i mean like a parts list for building an fxo |
| 20:20.18 | AndyGraybeal | hmmhesays; maybe an eagle pcb layout too or something |
| 20:20.35 | G-nerd | is eagle for linux free, I mean opensource |
| 20:20.47 | *** part/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 20:20.50 | AndyGraybeal | G-nerd: i don't think so, but it's the only thing that came to my head |
| 20:21.01 | G-nerd | oooh ok ;) |
| 20:21.03 | AndyGraybeal | i don't know of a opensource pcb design alternative off the top of my head |
| 20:21.56 | G-nerd | i think there is, but I guess that was a windows version, but opensource |
| 20:22.14 | G-nerd | some graphic expert wrote that, a retty good one realy |
| 20:22.44 | hmmhesays | oh polycom you are not helping me today |
| 20:23.07 | G-nerd | anyway we are here about asterisk |
| 20:23.11 | G-nerd | hihihi |
| 20:25.25 | hmmhesays | [Jan 8 12:26:12] DEBUG[29114]: chan_sip.c:13130 attempt_transfer: SIP transfer: Succeeded to masquerade channels. |
| 20:25.37 | hmmhesays | the way polycom's handle transfers is killing me today |
| 20:26.34 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 20:27.03 | hmmhesays | they don't make asterisk set any of the transfer variables |
| 20:27.46 | hmmhesays | unless you do a blind transfer |
| 20:28.06 | dexpdx | are they ways to execute call control applications for the CLI |
| 20:28.21 | dexpdx | such as zapbarge hangup etc? |
| 20:29.34 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 20:29.59 | hmmhesays | is there any way to make a polycom phone only do blind transfers? |
| 20:30.09 | *** join/#asterisk ZX81_ (n=ZX81@202.49.106.158) |
| 20:30.46 | [TK]D-Fender | hmmhesays: Nope, but you can make it default to blind. |
| 20:31.21 | drmessano-LT | IM BLIND..... transfer |
| 20:31.50 | drmessano-LT | AT&T comes in to install two new T1s |
| 20:31.53 | dexpdx | hmmhesays: a blindfold on the LED screen? |
| 20:32.03 | drmessano-LT | New cabinet doesnt have lightning protection on it |
| 20:32.06 | G-nerd | Hello my friends, are most ATAs suited enough to use it for Asterisk? |
| 20:32.20 | dexpdx | ATAs? |
| 20:32.20 | drmessano-LT | "See ya next week!" "Next week?" "Oh, nothing" |
| 20:32.32 | drmessano-LT | Most ATA's will work with Asterisk |
| 20:32.39 | G-nerd | Analog Telephony Adapter I guess |
| 20:32.47 | drmessano-LT | I can hook my toaster to Asterisk.. it's THAT good |
| 20:33.01 | drmessano-LT | But seriously, ATAs will work |
| 20:33.32 | G-nerd | ok, so why should I use telephone card of Digiumm, while I can use ATA too, right? |
| 20:33.58 | hmmhesays | [TK]D-Fender: how do you do that? |
| 20:34.00 | [TK]D-Fender | G-nerd: dEPENDS WHO YOU INCLUDE IN "MOST". pERHAPS YOU CAN TELL US WHAT YOU ACTUALLY WANT TO DO AND WHAT PRODUCTS YOU'VE ALREADY BEEN CONSIDERING |
| 20:34.04 | G-nerd | Is tere some benefit to use Digium stuff or is it the support |
| 20:34.04 | drmessano-LT | Depends on the task |
| 20:34.12 | [TK]D-Fender | hmmhesays: Go check the admin guide... you'll find it fast enough |
| 20:34.17 | hmmhesays | k |
| 20:34.27 | drmessano-LT | Digium cards work for what they are designed for |
| 20:34.28 | G-nerd | D-Fender, are you angry at me? |
| 20:34.31 | drmessano-LT | As does an ATA |
| 20:34.34 | hmmhesays | blindxfer actually sets a variable I can use to check if the call was transferred |
| 20:34.39 | hmmhesays | at that point I can do all kinds of crazy sh1t |
| 20:34.48 | drmessano-LT | Thats like asking "Do I need a car if my scooter has headlights?" |
| 20:35.13 | G-nerd | Well I want to start my own small business and try to find out what are the possibilites |
| 20:35.22 | *** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 20:35.33 | drmessano-LT | ATAs are good hockey pucks for using Analog phones on Asterisk |
| 20:35.42 | G-nerd | I'm really sorry if my questions seems to be stupid |
| 20:35.43 | drmessano-LT | But not the ideal solution |
| 20:35.44 | ThatKidKel | does anyone know if it is possible to generate the "this call is being recorded" reminder beeps with asterisk? |
| 20:35.53 | *** join/#asterisk homeinfo (n=home@196.211.34.2) |
| 20:36.16 | drmessano-LT | "Digium cards" can be used for terminating T1s, for interfacing to PSTN, for connecting analog phones, etc, etc, etc, etc |
| 20:36.21 | drmessano-LT | Thats too broad a question |
| 20:36.24 | dexpdx | ThatKidKel: use your dial play with Background or any other playback |
| 20:36.31 | dexpdx | before monitor command |
| 20:36.46 | drmessano-LT | Again, if you need a car, get a car.. if you want a burger, make a burger |
| 20:36.55 | lirakis | G-nerd: it sounds like .. from what youve said here... you shouldnt get near a tdm card .. |
| 20:37.02 | drmessano-LT | lol |
| 20:37.07 | drmessano-LT | ~book |
| 20:37.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 20:37.10 | drmessano-LT | READ |
| 20:37.12 | ThatKidKel | dexpdx.. i'm not sure i follow.. i need it to continously beep every 30seconds, or so so that the user knows its being recorded |
| 20:37.13 | lirakis | G-nerd:what do you want to do ? hook up a phone to a voip service provider? |
| 20:37.13 | drmessano-LT | then reread |
| 20:37.21 | drmessano-LT | and when you have it all figured out |
| 20:37.27 | drmessano-LT | READ AGAIN, because you DONT |
| 20:37.42 | G-nerd | ok ok I got the point |
| 20:37.44 | lirakis | G-nerd: if thats what you want .. get an ATA or a voip phone |
| 20:37.53 | drmessano-LT | and for the hell of it, read it a 4th time on the toilet... it makes a good bathroom read |
| 20:38.02 | drmessano-LT | Thats where most info soaks in anyway |
| 20:38.12 | G-nerd | hahahahaa |
| 20:39.00 | hmmhesays | the admin guide seems to have no reference to that |
| 20:39.09 | drmessano-LT | Im thinking of getting a toilet on casters to use as an office chair, so I can know as much about asterisk as [TK]D-Fender |
| 20:40.24 | G-nerd | ok, I fouyd out that there are some devices I don't know yet, Like FXO, FXS TDM...? In Holland we only have pstn, isdn and voip which is pretty new to us |
| 20:40.45 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:40.53 | [TK]D-Fender | drmessanoAnd here I figured its where you do your real thinking anyways, so you'd best let it rub off on your other endeavours (not literally of course... that would be gross) |
| 20:41.02 | drmessano-LT | [TK]D-Fender is going to throw the PDF at you |
| 20:41.11 | drmessano-LT | lol |
| 20:41.15 | G-nerd | I already have it |
| 20:41.20 | drmessano-LT | No |
| 20:41.20 | G-nerd | thnx |
| 20:41.25 | drmessano-LT | He will download a new one |
| 20:41.27 | drmessano-LT | and throw it |
| 20:41.28 | homeinfo | pstn is your telko the people respomsible for giving your line service beeing fxo-alaog line in or isdn basic or primary |
| 20:41.33 | [TK]D-Fender | G-nerd: Again, what are you needs? |
| 20:41.34 | drmessano-LT | He has an endless supply |
| 20:41.48 | homeinfo | there is my share of info for tonight |
| 20:42.05 | G-nerd | Ok this is my plan. I want to deliver small biznez pbx |
| 20:42.06 | ProofPack | TDM (Time Division Multiplexing) would likely be an E1 and isdn basic and Primary |
| 20:42.15 | G-nerd | And I want to use Asterisk for it |
| 20:42.31 | *** join/#asterisk puppet (n=puppet@c-c215e155.1511-4-64736c10.cust.bredbandsbolaget.se) |
| 20:42.53 | G-nerd | In holland most biznez comapny use only pstn or isdn basic, with 3 telnumbers and one faxnumber |
| 20:43.00 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:43.01 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 20:43.01 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:43.11 | Havokmon | Hrmph. nat=yes means sip show peers will have N for nat, not Y. That was obvious. |
| 20:43.22 | G-nerd | and I want to do this as earning extra money, not to be my real core biznez, at least not yet |
| 20:43.53 | G-nerd | btw I already have a job |
| 20:43.55 | [TK]D-Fender | G-nerd: If you're thinking of offereing telecom servces you should invest in a clue. |
| 20:44.08 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 20:44.08 | ProofPack | design considerations G-nerd, your connection to the outside world? VOIP? user considerations? IP phones? likely to be the easiest entry point. |
| 20:44.08 | homeinfo | sure g-nerd you get the customenrs and i will supply the dialplans for some weed |
| 20:44.21 | [TK]D-Fender | drmessano, jblack: Anyone else you know in IT that already has another job and has business aspirations? :) |
| 20:44.23 | G-nerd | yes I know, but I just started this week man, so I am not familiar to it |
| 20:44.32 | AndyGraybeal | thank god for the "asterisk: future of telephony" book! it is wonderful for a newb like myself |
| 20:44.40 | homeinfo | read the manual first |
| 20:44.40 | G-nerd | hahaha |
| 20:44.52 | AndyGraybeal | i'm at the end of chapter 2 |
| 20:45.00 | AndyGraybeal | homeinfo: thank you i will read the manual also |
| 20:45.07 | hmmhesays | Yeah there is no reference here about making blind xfer default |
| 20:45.34 | homeinfo | thats where we come in |
| 20:45.38 | G-nerd | and because I have some knowledge of telecommunications (graduated in 2000) |
| 20:45.40 | ProofPack | get yourself a server and install asterisk (asterisk: future of telephony is a good source) and set up some sip softphones. Branch out from there with a voip account and sip hardphones. |
| 20:46.08 | G-nerd | I already have installed Asterisk, but I am also new to Linux. so configuring things costs me a lot of time |
| 20:46.18 | G-nerd | and I want to learn the hard way not using GUI |
| 20:46.21 | iratik | Is there anyway to dial out calls on a regular modem using asterix? |
| 20:46.42 | ProofPack | think of it as an investment of time and not a cost. |
| 20:46.47 | G-nerd | ProofPack that is my first step to do it yes |
| 20:46.57 | homeinfo | has anyone checked out the politics chanel on the freenode channel weird bunch up in arm about the soon to be first lady president(yeah right) |
| 20:47.15 | G-nerd | yes you are right Proofpack I really agree with you |
| 20:47.18 | homeinfo | thats the best way g-nerd |
| 20:47.29 | ProofPack | yes, it is IMHO the easiest was of getting yourself a grounding point, you can branch out from there once you get that running |
| 20:47.40 | hmmhesays | hmm it says blindxfer preffered is only on the ip 320/330's |
| 20:47.46 | homeinfo | finding faults works best with command line edititing |
| 20:47.59 | homeinfo | how's that for spelling |
| 20:48.09 | iratik | anyone? |
| 20:48.45 | [TK]D-Fender | iratik: No. |
| 20:48.47 | homeinfo | yes iratik |
| 20:48.50 | ProofPack | It's a little difficult to start with but it's really not that hard, and there alot of help available in the mailing list and voip-info once you've committed to getting something working. |
| 20:49.14 | _ShrikE | Polycom question. I am using a 601 and side car. I would like to have the speed dial entries start on the side car, instead the ones I create take the next available position regardless of the index I specify. Can I change this? |
| 20:49.17 | iratik | homeinfo: any resources you could point me to? |
| 20:49.17 | *** join/#asterisk Shazaum (n=machine_@189.30.108.128) |
| 20:49.27 | hmmhesays | [TK]D-Fender: have you tried making any of your polycom's prefer blindxfer? |
| 20:49.28 | homeinfo | regarding? |
| 20:49.38 | [TK]D-Fender | _ShrikE: Nope. First come, first served |
| 20:49.54 | [TK]D-Fender | hmmhesays: by DEFAULT, no... standard usage, all the time |
| 20:49.59 | _ShrikE | thanks |
| 20:50.25 | ProofPack | _ShrikE I always fill up the slots with stupid things to get them happily in the right position |
| 20:50.28 | hmmhesays | [TK]D-Fender: the docs say that you can only do that on the ip 320/330 |
| 20:50.50 | homeinfo | regarding what iratik?? |
| 20:50.50 | iratik | homeinfo: using a traditional modem to dial out calls |
| 20:50.53 | G-nerd | yes I knwo ProofPack, I like it that is why I want to learn. At my job I'm working on a tracking device using ARM microcontroller and this is totally different but switching from microcontroller programming and configuring Asterisk on Linux is sometimes confusing |
| 20:51.16 | homeinfo | what modem |
| 20:51.37 | *** join/#asterisk m160858 (n=m160858@200.48.6.67) |
| 20:52.02 | iratik | a regular 56k modem |
| 20:52.13 | ProofPack | G-nerd, once linux is up and running with asterisk on it, you can pretty much get by without much linux knowlegde IMHO. It's all text config files and alot of trial and error. |
| 20:52.15 | [TK]D-Fender | iratik: NO <- |
| 20:52.23 | _ShrikE | ProofPack: Yeah, I am probably going to do blank entries mapped to a contact that plays back a message saying that button doesnt work. |
| 20:52.27 | iratik | I was thinking not.... thanks tho |
| 20:52.38 | G-nerd | and especially if my kids want their attention and when they sleep my wife wants also some attention, or I might end up with me and my laptop |
| 20:52.46 | [TK]D-Fender | hmmhesays: 1 Polycom is the same as another... |
| 20:53.21 | ProofPack | G-nerd, I'm living proof of the fact that knowing linux isn't a pre-requisite |
| 20:53.28 | homeinfo | google iratik |
| 20:53.33 | iratik | i'm at it |
| 20:54.15 | G-nerd | I believe on your words Proofpack |
| 20:54.29 | AndyGraybeal | is there a proper form of "backup" of asterisk? can i use something like Amanda to handle everything? |
| 20:54.35 | G-nerd | btw what is IMHO, my English is not that good you know |
| 20:55.18 | [TK]D-Fender | AndyGraybeal: Sure |
| 20:55.26 | AndyGraybeal | g-nerd: "in my humble opinion" |
| 20:55.31 | ProofPack | G-nerd, In my humble opinion, it's a phrase to say you aare only stating your opinion, not neccessarily others, or common consensus. |
| 20:55.36 | AndyGraybeal | [TK]D-Fender: rad thank you |
| 20:55.50 | G-nerd | aaah well I like that |
| 20:56.01 | G-nerd | thank you very much |
| 20:56.31 | G-nerd | Ok I will read Asterisk doc to understand more what to do, anyway thank you man! |
| 20:56.37 | G-nerd | thank you you all |
| 20:56.40 | hmmhesays | I don't think you can do this on an ip 601 |
| 21:02.06 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 21:04.55 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 21:05.07 | *** join/#asterisk arooni-mobile (n=arooni__@c-24-19-232-203.hsd1.mn.comcast.net) |
| 21:05.09 | arooni-mobile | hey folks |
| 21:05.13 | arooni-mobile | what does this mean: [Jan 8 21:04:51] NOTICE[8215]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 1 |
| 21:05.23 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 21:05.31 | file | Deeewayne: O.O |
| 21:05.31 | Deeewayne | O.o |
| 21:05.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:06.00 | Deeewayne | arooni-mobile: did you copy or move the call file ? |
| 21:06.31 | arooni-mobile | Deeewayne, move |
| 21:08.39 | arooni-mobile | Deeewayne, could it be because i don't have enough outgoing lines from my SIP provider? |
| 21:08.53 | arooni-mobile | no i'm still getting those failed call messgaes |
| 21:09.27 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 21:09.53 | hmmhesays | anyone happen to have the latest and greatest polycom firmware laying around? |
| 21:11.54 | alexcf | asdfasdfasdfasdfasdfasdfasdfasdfasdfasdfasdfasdfasdfasdfasdf/w 8 |
| 21:11.56 | alexcf | oops |
| 21:12.04 | drmessano-LT | So.. |
| 21:12.22 | drmessano-LT | HP laptop with XP.. If I plug a wired connection in, my wireless dies |
| 21:12.29 | drmessano-LT | Not to self: HP Suxors |
| 21:12.53 | drmessano-LT | Note to self: learn to type "note" |
| 21:14.32 | Qwell | drmessano-LT: \o/ |
| 21:15.15 | Qwell | drmessano-LT: dv6500? |
| 21:15.55 | [TK]D-Fender | arooni-mobile: Enable all appropriate debug and try your call again. An error line like that is worthless |
| 21:16.08 | trippss | anyone know if a good point to point T1 provider that's cheap? |
| 21:16.24 | drmessano-LT | nx9420 |
| 21:16.31 | Qwell | pfft, business line |
| 21:16.41 | drmessano-LT | Fingerprint scanner works... partly |
| 21:16.50 | drmessano-LT | Its a dual core, so it runs half as fast |
| 21:17.10 | drmessano-LT | and its a widscreen, so I have to turn my head as a type |
| 21:17.13 | Qwell | when I got my laptop replaced, I really wanted a Compaq model... |
| 21:17.26 | drmessano-LT | It weighs twice as much as it should |
| 21:17.26 | Qwell | erm, no, I wanted an HP model |
| 21:17.34 | Qwell | but they said I had to get the same line.. |
| 21:17.40 | drmessano-LT | and it takes 3 days to shut down |
| 21:17.40 | drmessano-LT | lol |
| 21:17.52 | drmessano-LT | I hate them both |
| 21:17.55 | drmessano-LT | I want an Acer |
| 21:17.57 | Qwell | of course, they don't have the 14" ones in that line anymore, so they had to upgrade me...meh |
| 21:18.10 | Qwell | I *wanted* a 14". I *bought* a 14". friggen HP |
| 21:18.18 | drmessano-LT | lol |
| 21:18.32 | drmessano-LT | Toshiba if I had a choice |
| 21:18.50 | drmessano-LT | Acer is probably the best they will do if I ask for another one.. 11 months after getting this one |
| 21:18.52 | drmessano-LT | Oh |
| 21:18.56 | drmessano-LT | The BEST PART |
| 21:19.11 | drmessano-LT | The left and right click buttons are RUBBER |
| 21:19.21 | drmessano-LT | and worn down to the white plastic inside |
| 21:19.26 | drmessano-LT | So I guess its rubber coated |
| 21:19.37 | Qwell | rubberesque |
| 21:19.39 | drmessano-LT | So now my fingers rub rubber and plastic |
| 21:19.56 | drmessano-LT | Doesnt chap my fingertips AT ALL |
| 21:20.04 | Havokmon | [TK]D-Fender, Didn't you tell me on the sonicwall to disable SIP Transformations? |
| 21:21.59 | [TK]D-Fender | Havokmon: Yup |
| 21:22.28 | Havokmon | Ok. That kept my NAT from working right on the Sonicwall Pro :) |
| 21:22.39 | Havokmon | it would drop the udp pings |
| 21:22.56 | puppet | anyone know any smaller phpthingy to edit realtime? or do i have to code my own ? :) |
| 21:23.58 | Havokmon | puppet, FCKeditor ? |
| 21:25.33 | puppet | Havokmon: dont think u hit right on any of it :=) |
| 21:25.53 | Havokmon | lol |
| 21:26.00 | m160858 | hi |
| 21:26.15 | m160858 | i in since many days ago i ask about the different between AgentLogin() and AgentCallBack() |
| 21:26.53 | [TK]D-Fender | m160858: AgentLogin requires you to stay on that CALL, and you will get "beeped" to answer a call |
| 21:27.10 | m160858 | i know that |
| 21:27.12 | [TK]D-Fender | m160858: AgentCallBackLogin places a call into the dialplan to contact agents. |
| 21:27.20 | m160858 | yes |
| 21:27.44 | drmessano-LT | Jeez |
| 21:27.51 | drmessano-LT | "this page wont load" |
| 21:27.55 | drmessano-LT | "So?" |
| 21:28.03 | drmessano-LT | "You're the IT director, help me" |
| 21:28.30 | drmessano-LT | "Reboot, call me tomorrow" |
| 21:28.38 | [TK]D-Fender | drmessanoHey... we had an IT director in here jsut the other day! |
| 21:28.52 | drmessano-LT | I dont cook fried |
| 21:28.54 | drmessano-LT | I dont cook fries |
| 21:28.58 | [TK]D-Fender | :D |
| 21:29.02 | [TK]D-Fender | ok, I'm outta here... |
| 21:29.04 | [TK]D-Fender | BBIAB |
| 21:29.07 | drmessano-LT | ok |
| 21:29.37 | arooni-mobile | how do i enable all appropriate debug |
| 21:29.48 | arooni-mobile | how do i tell asterisk i want a good level of granularity |
| 21:29.51 | arooni-mobile | in my error messages |
| 21:30.52 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 21:33.43 | lirakis | okay.. this day is over |
| 21:33.45 | lirakis | alter |
| 21:33.50 | lirakis | err.. later |
| 21:34.14 | *** part/#asterisk lirakis (n=lirakis@65.200.189.220) |
| 21:38.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:43.36 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:44.08 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 21:44.15 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:44.36 | pigpen | So...using realtime voicemail...when a user changes their password...it changes the password for every record in the db table.... |
| 21:44.39 | pigpen | ideas? |
| 21:45.01 | pigpen | I mean, sure, you can imagine the fun...like an inter-office lottery..guess the password now??? |
| 21:45.27 | pigpen | Was working fine....then, well, stopped working fine. |
| 21:45.31 | drmessano-LT | Oh oh oh |
| 21:45.38 | drmessano-LT | Is it 1234? |
| 21:45.41 | drmessano-LT | NO WAIT |
| 21:45.42 | drmessano-LT | 1111 |
| 21:45.45 | pigpen | anything you want baby. |
| 21:46.02 | drmessano-LT | Make it 4321... they'll NEVER guess it |
| 21:47.14 | drmessano-LT | Sadly, I have no recommendations.. So best I can offer you is comic relief |
| 21:47.16 | pigpen | yeah, and with 192 voicemail boxes...you can imagine the odds! |
| 21:47.21 | drmessano-LT | lol |
| 21:47.47 | pigpen | yeah, I find that realtime "help" is few and far between. |
| 21:53.08 | *** join/#asterisk __freedom__lover (n=eduardo@189-68-9-206.dsl.telesp.net.br) |
| 21:53.59 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:59.16 | Corydon76-dig | ~thebook |
| 21:59.16 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 21:59.22 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 21:59.30 | robl^ | ~book |
| 21:59.31 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:04.08 | Rawplayer | doesnt work |
| 22:04.16 | Rawplayer | de last url |
| 22:04.43 | [TK]D-Fender | Rawplayer, Does for me... |
| 22:04.53 | Rawplayer | http://tfot.leifmadsen.com |
| 22:04.54 | Rawplayer | that one |
| 22:05.06 | [TK]D-Fender | Rawplayer, I'm on it now |
| 22:05.17 | Rawplayer | oh |
| 22:05.18 | Rawplayer | hmm |
| 22:06.22 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 22:07.08 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 22:07.38 | *** join/#asterisk keulin (n=cray@cray.pck.nerim.net) |
| 22:10.51 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 22:11.44 | *** join/#asterisk Slingky (n=Maxime@modemcable111.80-201-24.mc.videotron.ca) |
| 22:12.15 | Slingky | hi guys! could you tell me how to know why registration failed (sip trunk) ??? |
| 22:12.25 | Slingky | which command or file to check ? |
| 22:12.50 | Slingky | when i type "sip show registry", i get request sent |
| 22:12.57 | dexpdx | /var/log/asterisk/messages ? |
| 22:14.12 | Slingky | ok, registration timed out |
| 22:14.23 | Slingky | ping time was like 600ms and more |
| 22:14.27 | Slingky | but now is like 40ms |
| 22:14.32 | Slingky | still same error message |
| 22:14.41 | Slingky | is there a way to set a bigger timeout? |
| 22:14.48 | hmmhesays | [TK]D-Fender: you wouldn't happen to have sip 2.2 firmware laying around would you? |
| 22:16.02 | [TK]D-Fender | Slingky, pastebin the CLI output of your registration attempts with SIP DEBUG enabled |
| 22:16.14 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:16.40 | [TK]D-Fender | hmmhesays, maybe somewhre |
| 22:17.21 | hmmhesays | it seems 2.0 does not support making blind xfer default |
| 22:18.04 | Slingky | [TK]D-Fender: i run * in a vmware machine, don't know if i can copy-paste |
| 22:19.25 | Slingky | i see something that starts with: retransmitting #6 (no NAT) to 66.49.255.42:5060: |
| 22:19.51 | [TK]D-Fender | Slingky, pastebin the whole mes. Being in a VM has no impact |
| 22:21.42 | [TK]D-Fender | ~pb |
| 22:21.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:21.44 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 22:22.28 | brodiem | I'm trying to remember, but is there a setting somewhere that will tell the remote endpoint to start transmitting RTP media _first_ always? |
| 22:23.01 | brodiem | or wait for it to send RTP first rather before Ast starts delivering RTP |
| 22:24.52 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:24.52 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:27.21 | *** join/#asterisk socken23 (n=socken@231-103.77-83.cust.bluewin.ch) |
| 22:27.52 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 22:27.52 | socken23 | Hi all! I'm in the middle of writing a report. I'm looking for other examples of projects like trixbox. Any quick ideas?? |
| 22:28.01 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:28.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:28.16 | drmessano-LT | Report? |
| 22:28.25 | grandpapadot | What's the best way to switch calls between two adjacent asterisk servers before dialing out to a provider? |
| 22:28.26 | drmessano-LT | Like "Failed attempts at bundling Asterisk"? |
| 22:28.26 | socken23 | a diploma thesis actually ;-) |
| 22:28.31 | drmessano-LT | Oh |
| 22:28.40 | drmessano-LT | Elastix |
| 22:28.51 | drmessano-LT | EasyVoxBox |
| 22:28.52 | socken23 | Elastix!! That's the one I was looking for :-) Thanks! |
| 22:28.56 | drmessano-LT | np |
| 22:29.01 | grandpapadot | i.e., to see if the number (extension) exists before going outside? |
| 22:29.32 | socken23 | ok, have to work ;-) Thanks & CU |
| 22:29.35 | Slingky | you have adminsparadise also |
| 22:29.47 | fiXXXerMet | How can I create an outbound extension that requires a 9 before the number? |
| 22:29.59 | fiXXXerMet | Just the patten. Something like _91NXXNXXXXXX? |
| 22:30.24 | fujin | that's how. |
| 22:30.35 | fujin | and then you can ${EXTEN:1} to strip the 9 |
| 22:30.38 | *** part/#asterisk Slingky (n=Maxime@modemcable111.80-201-24.mc.videotron.ca) |
| 22:30.48 | fiXXXerMet | ahh |
| 22:31.16 | fiXXXerMet | I was missing the :1 |
| 22:31.45 | fujin | :) |
| 22:31.58 | G-nerd | hello guys, FXO and FXS are just stupid interfaces for POTS, I thought these were futuristic virtual scientific hightechnological phones |
| 22:32.21 | fujin | lol? |
| 22:32.23 | G-nerd | all the drama about reading a book |
| 22:32.39 | fujin | no, they're just different parts of the PSTN. |
| 22:32.51 | G-nerd | but anyway I figured it out |
| 22:32.52 | fujin | ~fxo |
| 22:32.52 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 22:32.54 | fujin | ~fxs |
| 22:32.55 | jbot | methinks fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 22:33.30 | G-nerd | but that is just abc kids stuff |
| 22:33.42 | fujin | no shit |
| 22:33.42 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-239-242.dhcp.embarqhsd.net) |
| 22:33.51 | fujin | I stay a LONG way away from fxs/fxo ports |
| 22:34.21 | G-nerd | In holland we don't have a name for it, we just call connectors to the phone or wall |
| 22:34.30 | fujin | so does everyone else |
| 22:34.41 | fujin | the only time you'll ever hear of fxs/fxo is when you need to know what they are |
| 22:34.55 | G-nerd | well I guess with Asterisk |
| 22:35.08 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 22:37.47 | [TK]D-Fender | G-nerd, Or any other piece of telecom equipement like a PBX |
| 22:38.32 | G-nerd | yes, but for me it all have the same connector |
| 22:39.21 | G-nerd | the same cable I can connect with an analog phone to the wall, I can use that same cable to connect to a pbx too |
| 22:40.29 | G-nerd | And T1 or E1 is nothing more than a modem, right? |
| 22:41.01 | G-nerd | It uses a carrier to transmit voice information |
| 22:41.15 | G-nerd | In holland we call it just Modem |
| 22:41.58 | [TK]D-Fender | G-nerd, No, T1 isn't a "modem" |
| 22:42.08 | [TK]D-Fender | G-nerd, And you really don't understand much at all |
| 22:42.26 | [TK]D-Fender | G-nerd, You're burying yourself with each claim you make... |
| 22:43.27 | G-nerd | I guess the systems you use in America are very different than in Holland |
| 22:43.48 | G-nerd | or nameconvention |
| 22:44.11 | tzafrir_home | G-nerd, well, you can call it a modem, in the very wide sense of the word. But I wonder why... |
| 22:44.22 | tzafrir_home | Sounds like an insult :-) |
| 22:44.26 | puppet | rofl |
| 22:44.32 | puppet | G-nerd doesnt seem to have a clue |
| 22:44.34 | Havokmon | what would a t1 modulate/demodulate ? |
| 22:45.00 | puppet | Havokmon: a t1-modem is like the new version of adsl, but just older |
| 22:45.09 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 22:45.13 | G-nerd | modem puts the information signal on a carrier for transmission, like AM for Radio |
| 22:45.17 | tzafrir_home | TDM on the line to "whatever-the-driver-makes-of-it" |
| 22:45.23 | [TK]D-Fender | puppet, EQUALLY lost |
| 22:45.32 | puppet | [TK]D-Fender: u dont think i was joking? ;P |
| 22:45.44 | G-nerd | and demodulator takes away the carrier so to keep the information needed |
| 22:45.44 | [TK]D-Fender | Modems are technology applied over ANALOG links. |
| 22:45.46 | tzafrir_home | G-nerd, in that sense your ethernet adapter is a modem |
| 22:45.58 | Havokmon | Finally. Something I can agree with Fender on ;) |
| 22:46.11 | [TK]D-Fender | tzafrir_home, And so is my TELEGRAPH |
| 22:46.17 | G-nerd | no, not my ethernet adapter, but my cable modem does |
| 22:46.20 | puppet | [TK]D-Fender: bump :) |
| 22:46.41 | tzafrir_home | [TK]D-Fender, well, the term "ADSL modem" and hence later "Cables modem"is also quite common |
| 22:46.51 | G-nerd | ethernet adapter is simply ones or zeroes |
| 22:46.55 | G-nerd | on off |
| 22:47.00 | Havokmon | tzafrir_home, yeah, and it's wrong - like crackers are called hackers :P |
| 22:47.16 | [TK]D-Fender | tzafrir_home, Well ADSL is carried as a modulated signal over an analog line, same with Cable |
| 22:47.16 | fujin | not quite, G-nerd |
| 22:47.43 | G-nerd | yes that is right |
| 22:47.54 | [TK]D-Fender | G-nerd, Stop now you've hit bottom, no need to start digging. |
| 22:47.58 | tzafrir_home | Well, a T1 adapter modulates bits over a certain analog medium |
| 22:48.03 | puppet | [TK]D-Fender: he may find oil |
| 22:48.08 | tzafrir_home | (and likewise an ethernet adapter) |
| 22:48.19 | G-nerd | We do have ISDN, wich works digitally, but I assume it is also modulated over the line |
| 22:48.27 | [TK]D-Fender | ~assume |
| 22:48.27 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 22:48.28 | [TK]D-Fender | ^^^^ |
| 22:48.49 | puppet | i just got a spammail from some photoservice, and there was alink "change settings" and my password stood there in CLEARTEXT, idiots |
| 22:49.32 | Qwell | puppet: but did they send the username in cleartext? |
| 22:49.41 | puppet | Qwell: yes... |
| 22:49.44 | puppet | Qwell: all of it |
| 22:50.23 | G-nerd | tzafrir, I don't think ethernet adapter sends data modulated, unless you mean the modemcard |
| 22:50.31 | Havokmon | MBNA, until last month or so, put your whole credit card number in the link to get a pdf of your bill |
| 22:51.28 | puppet | Havokmon: nice move .x |
| 22:52.01 | G-nerd | Are some guys who studied telecommunications? |
| 22:54.02 | Havokmon | later all |
| 22:54.06 | CrazyTux | Has anyone here ever had an issue where a human voice sometimes was mistaken as a DTMF tone? |
| 22:54.16 | puppet | CrazyTux: nope |
| 22:55.00 | G-nerd | no hahaha, but it could be possible |
| 22:55.55 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 22:56.11 | [TK]D-Fender | CrazyTux, I once whistled up a 300 baud connection :) |
| 22:56.25 | CrazyTux | [TK]D-Fender, :) lol, I have this one guy |
| 22:56.28 | G-nerd | hahahaha |
| 22:56.41 | CrazyTux | Where his voice is being mistaken as DTMF on a DAILY basis |
| 22:56.44 | [TK]D-Fender | CrazyTux, And yes with poor line conditions and a shitty DTMF detection routine it could happen |
| 22:56.48 | puppet | CrazyTux: >< |
| 22:57.10 | CrazyTux | [TK]D-Fender, any advice on "best preventing" |
| 22:57.11 | [TK]D-Fender | CrazyTux, DAILY? Have you checked to see if he accidentally swallowed a cell phone? ;) |
| 22:57.20 | CrazyTux | [TK]D-Fender, lol |
| 22:57.23 | puppet | or a wissle? |
| 22:57.27 | G-nerd | Anyway I 've searched further what E1 is (E1 is for Europe, T1 for USA) and found out that E1 is also used for ISDN (developed by a dutch Telecom company) |
| 22:57.27 | CrazyTux | [TK]D-Fender, no seriously daily. |
| 22:57.37 | G-nerd | And ISDN is familiar here |
| 22:58.18 | CrazyTux | puppet, lol |
| 22:58.31 | pigpen | [TK]D-Fender, hey, mind a quick question? |
| 22:58.34 | [TK]D-Fender | G-nerd, E1 = 31 ISDN channels |
| 22:58.38 | [TK]D-Fender | pigpen, fast |
| 22:58.45 | pigpen | So...using realtime voicemail...when a user changes their password...it changes the password for every record in the db table.... |
| 22:58.52 | pigpen | ideas? |
| 22:58.55 | [TK]D-Fender | G-nerd, T1=24 analog |
| 22:58.58 | G-nerd | 30 data channel and one B--cahnnel which is for signaling |
| 22:59.14 | [TK]D-Fender | G-nerd, No, that is E1-PRI <--- |
| 22:59.28 | puppet | pigpen: wait ill test it here im running realtime voicemail |
| 22:59.33 | [TK]D-Fender | G-nerd, you simply said E1 which is the BASE. You don't need to be using PRI signalling over it |
| 22:59.39 | puppet | pigpen: or wait ;P not in asterisk, but should be same |
| 22:59.40 | [TK]D-Fender | pig nope |
| 22:59.45 | G-nerd | well here in holland we have 2 types, that is with 2 D and 1 B and 30 D and 1 B |
| 22:59.55 | pigpen | puppet, I have been running it for 6 months...no issue. |
| 23:00.03 | pigpen | out of the blue...boomn |
| 23:00.09 | [TK]D-Fender | G-nerd, And you are very good at mixing your terms up and will mislead anyone who listens to you. |
| 23:00.21 | [TK]D-Fender | G-nerd, 2B+d = ISDN-BRI |
| 23:00.45 | G-nerd | D-Fender, it is not fair to talk about me like that |
| 23:00.50 | *** join/#asterisk tc3driver-nii (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 23:00.58 | G-nerd | yes ok |
| 23:01.08 | G-nerd | we have the basic and premium |
| 23:01.15 | G-nerd | or primary |
| 23:01.24 | puppet | G-nerd: u read on wiki and copy pasted, or? |
| 23:01.38 | G-nerd | no man, I learnt that at school for a long time |
| 23:01.43 | [TK]D-Fender | G-nerd, then for your reference those are (in your case) ISDN-BRI, and ISDN-E1-PRI |
| 23:01.50 | puppet | pigpen: enable full sql logging, and check the query? |
| 23:01.59 | [TK]D-Fender | ~pri |
| 23:02.00 | jbot | pri is probably Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 23:02.16 | G-nerd | ok, but I just want to mention that we never used the term E1 or T1 or whatever |
| 23:02.27 | [TK]D-Fender | G-nerd, however E1 can be FULLY channelized wirthout a D chan. |
| 23:02.51 | [TK]D-Fender | G-nerd, then you were being too "street" with your terms |
| 23:02.51 | puppet | [TK]D-Fender: i have no clue of theese things, what does the T-standard use the D channel for? |
| 23:03.10 | G-nerd | D means data and B is for signalling etc... |
| 23:03.16 | [TK]D-Fender | puppet, both T1 & E1 can use PRI signalling requiring a D chane |
| 23:03.24 | puppet | [TK]D-Fender: ah |
| 23:03.32 | *** join/#asterisk obnauticus (n=obnautic@c-24-22-14-101.hsd1.wa.comcast.net) |
| 23:03.32 | G-nerd | also for alarm things they use the B channel |
| 23:03.36 | [TK]D-Fender | puppet, the D-chan carries all of the call control and channel switching information, passing on DID data, etc |
| 23:03.41 | bkruse | [TK]D-Fender: fully channelized e1 without a d channel, but not t1? |
| 23:03.52 | [TK]D-Fender | bkruse, Sure, CAS T1 |
| 23:04.02 | puppet | [TK]D-Fender: ah ok, so its kinda just the voice that goes in the B channels? and rest in D? |
| 23:04.11 | bkruse | [TK]D-Fender: I thought you could do that with t1 also |
| 23:04.16 | G-nerd | D-Fender are working for a telecom comapny? |
| 23:04.21 | [TK]D-Fender | bkruse, I was restricting the conversation to standards they at least knew... but apparently not well enough :) |
| 23:04.49 | [TK]D-Fender | G-nerd, No, I consult however and have worked with this stuff in * for many years. |
| 23:05.04 | G-nerd | oooh that's why you get so upset |
| 23:05.12 | bkruse | [TK]D-Fender: hehe :] its just that I test test test new cards, so you have to do ALL of the types of framing/codings, even though <1% may use it :] |
| 23:05.28 | G-nerd | you might think, what kind of idiot does he think about me |
| 23:05.28 | bkruse | [TK]D-Fender: for sake of his learning, you are def correct and thank you for helping in this channel so much :] |
| 23:05.36 | [TK]D-Fender | G-nerd, No... only when you KEEP getting it wrong more and more, and then use the equivalent of "kiddie terms" to pass on "knowledge" :) |
| 23:05.36 | Qwell | [TK]D-Fender: what do you do for a living, anyhow? just consulting? |
| 23:05.40 | Qwell | no "day job"? |
| 23:06.07 | bkruse | Qwell: asterisk consulting pays well :] |
| 23:06.11 | [TK]D-Fender | bkruse, there's tons still that I don't know.... I just pass on what I do :) |
| 23:06.12 | Qwell | indeed it does |
| 23:06.26 | G-nerd | Well D-Fender, sorry for that, but that was certanly not my intention |
| 23:06.29 | [TK]D-Fender | Qwell, Full-time IT for a plumbing importer & manufacturer |
| 23:06.37 | Qwell | plumbing importer? |
| 23:06.46 | [TK]D-Fender | G-nerd, And still you succeeded gloriously! Let us rejoice! |
| 23:06.50 | Qwell | so you've got a lot of free time :p |
| 23:07.03 | bkruse | [TK]D-Fender: nono, I was just saying I am not sure but I think there is, and to thank you :] |
| 23:07.06 | [TK]D-Fender | Qwell, "I'm just the cook" - Steven Seagal (Under Seige) |
| 23:07.08 | G-nerd | no no please, I don' t want to be remebered like that |
| 23:07.08 | [TK]D-Fender | Qwell, ;) |
| 23:07.48 | [TK]D-Fender | G-nerd, s'ok take you time and learn here with us. It is a group process. |
| 23:08.03 | [TK]D-Fender | alrighty.... or to martial arts... pray I don't have an accident :) |
| 23:08.12 | G-nerd | pfew, I'm glad you take it from the positive side |
| 23:08.36 | [TK]D-Fender | G-nerd, I do. |
| 23:09.27 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 23:09.52 | *** part/#asterisk gaero (n=theo@unaffiliated/balkantools) |
| 23:10.02 | G-nerd | Well I got to go, thank you all my friends! And especially D-Fender for his patient! |
| 23:11.23 | hmmhesays | hrm, well the 2.2 firmware is producing some interesting results |
| 23:11.54 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 23:12.11 | hmmhesays | my buddy watch seems to have changed its behavior |
| 23:14.33 | brodiem | I'm trying to remember, but is there a setting somewhere that will wait for the remote endpoint to start transmitting RTP media _first_? |
| 23:15.16 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 23:18.59 | hmmhesays | my directory doesn't buddy watch anymore hmmm |
| 23:20.07 | hmmhesays | found it |
| 23:20.42 | brodiem | anyone? fighting a one-way audio prob but only affects incoming calls, not outgoing.. only thing I can come up with is the fact that on incoming the first RTP is incoming to the endpoint so it's not getting into a NAT pinhole and causing future packets to be rejected |
| 23:23.11 | *** join/#asterisk abaci (n=IceChat7@ool-4b7fc532.static.optonline.net) |
| 23:23.40 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 23:27.45 | *** join/#asterisk jgoo (n=chris@77.49.253.16) |
| 23:28.08 | puppet | brodiem: its a firewall problem i bet |
| 23:29.21 | brodiem | puppet: yes it is being blocked, that is why I'm asking about waiting for the client to start sending rtp media first since there is only a problem on inbound calls.. |
| 23:29.53 | puppet | brodiem: or fix the firewall ? |
| 23:30.22 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 23:30.51 | *** join/#asterisk bjingles (n=brett@24.68.237.193) |
| 23:30.54 | brodiem | there is nothing more that can be done on it, like many other crappy routers out there that insist on its stateful inspection crap that doesn't work |
| 23:31.27 | puppet | get a noncrappy router or is it a clients? |
| 23:31.41 | brodiem | client, one of the watchguard firebox's that aren't cheap.. |
| 23:32.06 | puppet | haha, you would think that a noncheap router would be good, and not messs to much up |
| 23:33.20 | brodiem | if I could only delay the outgoing RTP from asterisk for a second it'd be done with, I thought for sure there was a built-in way of doing it but I might be thinking of osmething else.. |
| 23:33.24 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 23:34.36 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 23:35.48 | bjingles | does 'misc file error Error is 0x20" mean anything to anyone? |
| 23:36.01 | bjingles | after provisioning a polycom phone that's what I get |
| 23:37.17 | hmmhesays | anyone ever remap keys on a polycom phone? |
| 23:37.22 | ManxPower | brodiem: no, there is no way to do that. |
| 23:37.41 | ManxPower | Asterisk expects routers, firewalls, and NAT to be non-broken. |
| 23:38.14 | ManxPower | bjingles: it means you screwed up your config files for the phone. |
| 23:40.50 | bjingles | is there any way to figure out which one? |
| 23:41.17 | bjingles | I mean there's sip.conf phone1.cfg and the 000000 (Mac address one) |
| 23:41.24 | nny_1 | ~fonality |
| 23:44.57 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 23:49.04 | ManxPower | bjingles: No idea. In my experience it's usually a missing or misplaced quote. |
| 23:49.55 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 23:50.28 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 23:50.33 | nny_1 | getting build_tools/genudevrules: line 3: udevinfo: command not found on zaptel 1.4.7 |
| 23:50.34 | *** join/#asterisk bjweeks (n=bjweeks@unaffiliated/bjweeks) |
| 23:50.37 | *** join/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 23:50.49 | nny_1 | saw a bug report filed, but it says it was fixed in 1.4.6 |
| 23:50.59 | bjweeks | is the firmware on Polycom's support page the latest? |
| 23:51.47 | *** join/#asterisk coppice (n=chatzill@230.202.17.210.dyn.pacific.net.hk) |
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| 23:53.00 | bjweeks | never mind, it isn't |
| 23:53.16 | bjingles | manxpower thanks so do you recommend just rebuilding from scratch until it goes? |
| 23:53.37 | bjingles | rebuilding config files I mean |
| 23:55.13 | mocker | Can a Sangoma card and a Digium card coexist? |
| 23:55.33 | russellb | i don't think so |
| 23:55.38 | mocker | or can a Digium card still work w/ the patched zaptel that Sangoma uses? |
| 23:55.41 | russellb | you may open up some kind of black hole to hell |
| 23:55.55 | mocker | russellb: I'm sitting next to a cabinet full of Avaya equipment. |
| 23:55.59 | Corydon76-dig | mocker: no, they corrupt the driver pretty badly |
| 23:56.00 | mocker | Hell is here, and it is hot. |
| 23:56.15 | russellb | you would certainly lose your digium card support by patching zaptel. |
| 23:56.17 | dennis- | ~book |
| 23:56.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 23:56.25 | russellb | and like Corydon76-dig said, it's not clear whether the patch is safe |
| 23:56.38 | russellb | because it conflicts with some other code in zaptel that the digium T1 cards use |
| 23:57.10 | mocker | Gotcha. |
| 23:57.23 | Corydon76-dig | We'd welcome a sane patch from them, but we're not going to write it for them |
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| 23:58.31 | Corydon76-dig | Evening, Katty |
| 23:59.01 | Katty | allo (= |
