| 00:09.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:11.16 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 00:13.52 | jarrod | zc |
| 00:16.48 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 00:23.44 | vlt | I want to answer incoming voip calls with a voice "You are on hold on position x ..." and then hand the call over to an existing ISDN/analog PBX with 16 analog phones, 4 ISDN BRI channels (2 of them connected to external line, the other 2 internal). Can I do this with asterisk? What hardware do I need to connect asterisk to the ISDN lines (maybe to one of the external channels, too)? ...? |
| 00:25.54 | *** join/#asterisk anthm (n=anthm@h4608a74c.area4.spcsdns.net) |
| 00:25.54 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 00:36.26 | *** part/#asterisk NDT (n=noone@cpe-24-195-66-214.nycap.res.rr.com) |
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| 00:42.07 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 00:42.19 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 00:44.25 | rushowr | murf, anybody who knows AEL2 inside and out |
| 00:44.33 | rushowr | need a quick question answered |
| 00:44.34 | *** join/#asterisk Tbb (n=ccorbin@bravo.telebarbados.com) |
| 00:44.40 | rushowr | desperately |
| 00:44.43 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:45.28 | rushowr | anyone know the difference between the 'jump' and 'goto' statements in AEL2? |
| 00:47.02 | rushowr | I think i know but....ah fuck it |
| 00:47.03 | rushowr | cheers |
| 00:47.22 | *** join/#asterisk chreese (n=chatzill@bridalveil.istep.com) |
| 00:51.16 | *** join/#asterisk stinkpad (n=cunted@cpc2-broo4-0-0-cust398.renf.cable.ntl.com) |
| 00:51.43 | stinkpad | anyone got zaptel compiled correctly on sparc64? |
| 00:53.16 | *** join/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 01:01.04 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 01:01.23 | jtexter3 | If I have two channels bridged, what is the best way to disconnect them? |
| 01:01.30 | jtexter3 | ast_softhangup? |
| 01:04.10 | Snake-Eyes | hmm i have question about softhangup too |
| 01:04.58 | Snake-Eyes | yet to get clear answer about how softhangup should be used |
| 01:06.14 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 01:07.24 | jtexter3 | Snake-Eyes, what have you run into with softhangup? |
| 01:08.40 | Lyfe | can anyone tell me if the syntax for this is correct (to go to priority 20 if dialing 918001234567): GotoIf($["${EXTEN:2:4}" = "800"]?20) |
| 01:09.30 | Lyfe | presuming you're after the 800 part, that is. |
| 01:11.36 | Lyfe | nevermind, answered my own question, noop() is my friend. |
| 01:13.12 | nailbags|laptop | i'm getting a weird echo from my tdm400 ... it only happens after an attended transfer! any ideas? |
| 01:13.21 | nailbags|laptop | :q |
| 01:13.22 | Snake-Eyes | jtexter3, Ive been trying to use it in a macro to get it to hangup all calls on a sip trunk, yet it doesnt seem seem to hang up a call |
| 01:17.42 | *** part/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 01:17.44 | jtexter3 | looking at various pieces of code, maybe ast_softhangup only hangs up if its AST_SOFTHANGUP_DEV, or a time |
| 01:18.31 | *** join/#asterisk michaelo (n=michaelo@adsl-147-45-179.gsp.bellsouth.net) |
| 01:19.33 | Snake-Eyes | AST_SOFTHANGUP_DEV ? |
| 01:22.01 | jtexter3 | well, looking at the code for the actual API, you can pass it several soft hangup reasons. Looks like the SoftHangup dialplan app won't actually hangup an application |
| 01:22.22 | Snake-Eyes | the really scary part is how no one is concern that softhangup might not work, seeing how its used for e911 /000 calls |
| 01:22.37 | Snake-Eyes | hmm |
| 01:23.11 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:23.30 | Snake-Eyes | jtexter3, I want it to hangup a call on a trunk, not app |
| 01:24.18 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 01:25.29 | Snake-Eyes | I want it to hangup acall on a trunk so a emergency call can go through yet in the test ive done its not hanging up any calls ... |
| 01:27.06 | blitzrage | has anyone used the Local/ channel in Asterisk Business Edition? |
| 01:27.27 | blitzrage | the 1.0 release |
| 01:27.40 | P-NuT | Hi guys. I'm using a cisco 7905 phone, with asterisk. When I call somewhere with an IVR, i try to press the buttons for the options, but it doesn't recognise that i'm pressing any. |
| 01:27.46 | P-NuT | Is this an asterisk issue? |
| 01:27.52 | P-NuT | or something else? |
| 01:29.15 | Snake-Eyes | P-NuT, have you checked what you have set the dtmf for in asterisk and the phone? |
| 01:30.39 | P-NuT | umm.. |
| 01:30.39 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 01:30.39 | Snake-Eyes | blitzrage, whats the difference between normal Asterisk and Asterisk Business Edition? |
| 01:30.48 | P-NuT | no. What should it be? |
| 01:31.53 | Snake-Eyes | well, if you're using g729 it has to be rfc2833, else you can use inband |
| 01:32.24 | P-NuT | i'm using g711 and in sip.conf I have this.. |
| 01:32.25 | Snake-Eyes | P-NuT, I would set dtmf on both to rfc2833 first and if that doesnt work then inband |
| 01:32.26 | P-NuT | dtmfmode=rfc2833 |
| 01:32.50 | Snake-Eyes | ok, then in your cisco phone set it to rfc2833 |
| 01:33.09 | P-NuT | umm.. actually a setting "ON" the phone? |
| 01:33.15 | P-NuT | or in sip.conf? |
| 01:33.28 | Snake-Eyes | "ON" the phone |
| 01:33.46 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 01:34.37 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 01:34.42 | quid246 | is it just me or are alot of calling-card outfits run by shady guys? |
| 01:34.55 | P-NuT | hmm |
| 01:35.00 | P-NuT | dont know if I can do that |
| 01:35.04 | justinu|laptop | its not you |
| 01:35.29 | quid246 | justina: I saw one board and seems like every 5th message says "don't deal Mohammed he'll rip you off". |
| 01:35.32 | [TK]D-Fender | quid246 : Nope... most do evil stuff like hijack other people accoutns, run over credit limits, frequently bankrupt and restart.... |
| 01:35.42 | Snake-Eyes | P-NuT, look at the manual for the phone, most voip phones have a web interface |
| 01:35.43 | quid246 | or insert name |
| 01:35.50 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 01:36.18 | quid246 | They are definitely a rung below cellphone salesmen |
| 01:36.23 | justinu|laptop | heh |
| 01:36.44 | quid246 | atleast the cellphone guys wear a suit, albeit a cheapie |
| 01:37.24 | Snake-Eyes | cellphone guys wearing suites ? |
| 01:37.56 | Snake-Eyes | all the ones ive seen where pants and a open collar shirt with something hanging from there neck |
| 01:38.04 | Snake-Eyes | where=wear |
| 01:38.46 | quid246 | haha yeah true... I haven't dealt with them since the days of the brick, so they were more slick then |
| 01:39.30 | quid246 | which reminds me... Cingular is charging a surcharge for people with old analog phones. Way to take advantage of the retro-lover... haha |
| 01:40.56 | quid246 | I wish I still had my brick... would make for a great sight gag. Excuse me I have to make a call, and pull the behemoth out of your briefcase |
| 01:41.08 | omal | my understanding is that an older phone costs them more to serice than a new one |
| 01:41.10 | *** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au) |
| 01:41.30 | quid246 | omal: Well, yeah... since they are analog they take up a heck of alot more bandwidth than the digital call |
| 01:41.58 | omal | so i wouldnt really call it taking advantage |
| 01:42.09 | quid246 | I was ebing sarcastic |
| 01:42.31 | quid246 | long-live the bag phone |
| 01:44.54 | *** join/#asterisk [hC] (n=hardcore@190.10.9.191) |
| 01:45.31 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
| 01:46.21 | quid246 | SOIP? |
| 01:46.51 | omal | Sarcasm Over IP :D |
| 01:47.06 | [TK]D-Fender | ~book |
| 01:47.09 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:47.24 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 01:47.35 | Snake-Eyes | ~booked |
| 01:47.39 | omal | its also found on my coffee table |
| 01:48.01 | quid246 | haha that's what I thought |
| 01:48.24 | Snake-Eyes | hmm |
| 01:48.30 | Snake-Eyes | ~bot |
| 01:48.32 | jbot | I ain't no stinkin' bot. I am a finely tuned and hand crafted tool. Oh wait... I guess I am a bot (that you should not abuse). |
| 01:48.40 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 01:48.43 | Snake-Eyes | haha |
| 01:48.47 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:48.50 | Qwell | ~botabuse |
| 01:48.51 | jbot | Stop tormenting me! |
| 01:50.30 | file | ~botsnack |
| 01:50.30 | jbot | thanks, file |
| 01:51.53 | Snake-Eyes | ~botlove |
| 01:51.55 | jbot | ACTION hugs CIA-5 |
| 01:51.55 | [TK]D-Fender | ~lart himself |
| 01:52.06 | [TK]D-Fender | :D |
| 01:52.36 | Snake-Eyes | hehehe |
| 01:53.17 | harryvv | Anyone seen a case of a working zap with no errors just suddenly not respond ? My cli output was right. Well I just restarted the server and now its working. |
| 01:54.31 | rene1 | ~lart twisted[asteria] |
| 01:55.01 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 01:56.38 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 01:57.04 | rene1 | ~lart brookshire |
| 01:57.09 | rushowr | hey all, got another question or two, shouldn't be hard...lemme know if you mind trying to help :) |
| 01:57.14 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 01:57.31 | murf | rushowr: hey there, still got the AEL question? |
| 01:57.36 | rushowr | hel;l yea |
| 01:57.38 | quid246 | it's funny when you look at it... with the exception of windowed clients... IRC hasn't really changed much in 10 years |
| 01:57.40 | rushowr | there you are! |
| 01:57.48 | rushowr | let me get you in a pm |
| 02:02.09 | harryvv | quid, actually it has in one respect. The servers dont crash as much from abuse. |
| 02:02.25 | *** join/#asterisk userdefined (n=jross@cpe-24-169-131-0.rochester.res.rr.com) |
| 02:02.42 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 02:03.32 | quid246 | harryvu: Touche on that one. I remember having to regain my channel on splits. |
| 02:03.52 | userdefined | hola. can anyone point me to an extensions.conf example showing how to route a call based on the calling party? |
| 02:04.04 | userdefined | i've been googling for a while but apparently don't know the magic words =/ |
| 02:07.01 | hads | userdefined: exten => 1234/5432,1,NoOp() |
| 02:07.11 | hads | Where 5432 is calling party. |
| 02:08.04 | userdefined | ah! thanks very much |
| 02:08.09 | hads | np |
| 02:09.48 | harryvv | quid, yea those were the fustrating days |
| 02:16.24 | omal | [22:15:06] <Damin> Digium just got 13.2 million in Venture Capital funding! ;) |
| 02:16.40 | file | 13.8 |
| 02:16.45 | dlynes_laptop | omal: that was a misprint; it's 13.8 |
| 02:18.10 | omal | hooray! |
| 02:19.26 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:19.38 | dlynes_laptop | well, not a misprint; a misreport; they later restated it when Mark corrected them |
| 02:26.03 | [hC] | Cool, i wonder what they'll do with it |
| 02:26.04 | [hC] | :) |
| 02:26.25 | file | we're going to buy $13.8 million dollars worth of muffins |
| 02:26.36 | Nivex | file: here I was gonna say a really friggin' big keg |
| 02:26.56 | file | pfft |
| 02:27.41 | Nivex | hmmm... no updates to Pound Key lately |
| 02:27.46 | harryvv | vm needs a option to dial the number after listening to the vm |
| 02:28.26 | file | harryvv: what if it already has it? |
| 02:30.01 | harryvv | file I have not seen it. Which option is it? the |
| 02:30.24 | file | ; callback=fromvm ; Context to call back from |
| 02:30.24 | file | <PROTECTED> |
| 02:31.01 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 02:31.08 | harryvv | In options it would be nice to have it say press 1 to call this number, press 6 to listen to new messages ect. |
| 02:31.22 | foo | We have a VoIP talk at our local LUG right now ... /join #sgvlug ... the stream URL is in the topic. Free, open source. :) |
| 02:31.22 | mitcheloc | if you can stream this, KG is giving a presentation right now on asterisk ;) http://stream.sgvlug.org:8800/ |
| 02:31.28 | foo | HAHA. |
| 02:31.29 | mitcheloc | meh |
| 02:31.30 | foo | mitcheloc: :P |
| 02:34.17 | harryvv | file this in voicemail.conf |
| 02:34.42 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:35.19 | file | I just grabbed it from the example config in trunk |
| 02:35.54 | file | it is also in 1.2 |
| 02:36.41 | [hC] | i have to go thru the example configs again some time |
| 02:36.45 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:36.52 | [hC] | so many features since i did my 'standard' configs back in 2004 :P |
| 02:37.35 | rushowr | •murf• hey murf I remembered what I was gonna ask, if you've got a sec |
| 02:37.37 | rushowr | PM me |
| 02:41.21 | *** join/#asterisk und3rtug4 (n=undertug@88.157.68.236) |
| 02:41.37 | *** part/#asterisk und3rtug4 (n=undertug@88.157.68.236) |
| 02:45.24 | *** join/#asterisk froguz (i=froguz@214-142-222-201.adsl.terra.cl) |
| 02:47.20 | harryvv | hard to believe that a ipod could be used to blow up a plane. |
| 02:49.30 | mitcheloc | Kerry_G live on webcam... -> http://stream.sglug.org:8800 (seriously) |
| 02:49.44 | mitcheloc | ** Kerry_G live on webcam... -> http://stream.sglug.org:8800 (seriously) |
| 02:49.46 | mitcheloc | ** http://stream.sgvlug.org:8800/ |
| 02:49.54 | mitcheloc | heh, clumsy me |
| 02:50.38 | Snake-Eyes | .... |
| 02:50.43 | *** join/#asterisk rrittenhouse (n=tad@24.55.244.254) |
| 02:50.52 | [hC] | not only may i say, "wtf" - but your link is dead. |
| 02:51.36 | mitcheloc | [hC]: use the last one, i missed the "v" twice in a row *doh* |
| 02:52.15 | [hC] | Guess it doesnt work on safari |
| 02:52.47 | [hC] | what is it anyways? |
| 02:53.30 | mitcheloc | [hc]: supposedly, you need a proper video player, like VLC or videolan, KG is doing a presentation on Asterisk to the LUG @ CalTech |
| 02:54.54 | [hC] | who is kg? |
| 02:54.58 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:55.05 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:55.17 | [hC] | they're doing slides on freepbx right now |
| 02:55.34 | mitcheloc | i agree....eww |
| 02:55.43 | mitcheloc | he runs www.voipspeak.net |
| 02:57.09 | *** join/#asterisk nortex (n=barracud@adsl-68-93-160-132.dsl.amrltx.swbell.net) |
| 02:57.44 | [hC] | he |
| 02:57.47 | [hC] | 'he's not very funny |
| 02:57.52 | [hC] | and seems like a douchebag. |
| 02:58.29 | mitcheloc | if you can do better you are welcome to |
| 02:58.50 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 02:59.46 | [hC] | are you there? |
| 03:00.45 | mitcheloc | [hC]: yes |
| 03:01.05 | file | am I Supposed to get audio with this? |
| 03:01.14 | [hC] | mitcheloc: are you in the frame? |
| 03:01.19 | [hC] | theres audio yeah |
| 03:01.23 | [hC] | im using vlc |
| 03:01.23 | mitcheloc | file: i think so, yes, join #sgvlug and ask |
| 03:01.36 | [TK]D-Fender | KG promotes Linksys & Trixbox a wee bit too much... |
| 03:02.16 | mitcheloc | [TK]D-Fender: he did a disclaimer before he started =P |
| 03:02.22 | [TK]D-Fender | :O |
| 03:02.32 | [TK]D-Fender | I'm trying to get in through VLC and failing.. |
| 03:02.45 | [hC] | mitcheloc: are you the ibm laptop in the bottom left? i think i see bitchx |
| 03:03.40 | mitcheloc | [hC]: i don't know, just a sec |
| 03:04.04 | [hC] | wow, a girl's there? haha |
| 03:04.51 | mitcheloc | [hC]: a couple, why are you being so negative? |
| 03:04.56 | nortex | What are ya'll up to? |
| 03:05.28 | [hC] | im not being negative... Didnt mean it in a negative way |
| 03:05.30 | Qwell | What is this stream? |
| 03:05.33 | mitcheloc | nortex: Kerry_G is webcasting, join #sgvlug and read the chan topic for the url |
| 03:05.35 | [hC] | Just surprising, not a lot of girls go to LUG meetings. |
| 03:05.48 | mitcheloc | I'll wanr you all that he is doing a fair bit on TB/FreePBX |
| 03:05.53 | mitcheloc | * warn |
| 03:05.59 | Qwell | That stream quality is complete crap... |
| 03:06.11 | dlynes_laptop | no kidding |
| 03:06.14 | dlynes_laptop | and fwiw |
| 03:06.28 | dlynes_laptop | it works if you have mplayer plugin for firefox |
| 03:06.32 | mitcheloc | Qwell: probably because 90% of the people here in the room are watching it at the same.as seeing it live |
| 03:06.41 | Qwell | :p |
| 03:07.06 | [hC] | .... why would you do that??? :P |
| 03:07.13 | *** join/#asterisk AJaymn (i=AJaymn@70.59.126.198) |
| 03:07.22 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 03:07.24 | dlynes_laptop | [hC]: and that's surprising? |
| 03:07.34 | Qwell | somebody needs to shout something obscene |
| 03:07.35 | mitcheloc | [hC]: i don't understand the reasoning either as i'm not doing it |
| 03:07.44 | mitcheloc | Qwell: haha |
| 03:07.45 | dlynes_laptop | [hC]: have you been to any of the vanlug meetings? |
| 03:07.53 | [hC] | dlynes_laptop: a long time ago. |
| 03:08.00 | [hC] | I presented at one |
| 03:08.21 | dlynes_laptop | man, a lot of those guys are such a bunch of children |
| 03:08.35 | dlynes_laptop | major fanaticism |
| 03:08.48 | [hC] | yep, with nothing better to do than sneer over linuxy things. |
| 03:08.48 | [hC] | its the biggest nerd fest ive ever seen. |
| 03:08.50 | dlynes_laptop | exactly |
| 03:09.03 | dlynes_laptop | no realism; only linux fanaticism |
| 03:09.07 | [hC] | yep. |
| 03:09.23 | Qwell | "Programming"...PFFT |
| 03:10.03 | *** join/#asterisk phifli_ (n=w0w00@nc-71-53-104-47.dhcp.embarqhsd.net) |
| 03:10.20 | dlynes_laptop | btw |
| 03:10.31 | dlynes_laptop | where's 'sgv'? |
| 03:10.43 | [hC] | san gabriel valley |
| 03:10.48 | Qwell | hmm |
| 03:10.48 | dlynes_laptop | california? |
| 03:10.49 | [hC] | california |
| 03:10.51 | mitcheloc | if anyone has questions for Kerry, I can go ahead and ask him? i.e. "why not use conf files?" would be a great one :) |
| 03:10.51 | [hC] | yeah |
| 03:10.52 | dlynes_laptop | ah |
| 03:11.02 | mitcheloc | it's near so-cal, "caltech" |
| 03:11.40 | Qwell | hey, I am in sgv :D |
| 03:12.12 | [hC] | "Would you say that trixbox is like, Asterisk for dummies?" |
| 03:12.13 | [hC] | :P |
| 03:12.19 | Qwell | ~thebook |
| 03:12.21 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:12.36 | mitcheloc | Kerry said yes |
| 03:12.37 | Qwell | Asterisk for...idiots |
| 03:12.40 | [hC] | HA! |
| 03:12.46 | Qwell | trolling, kthx |
| 03:13.02 | [hC] | I didnt think you;'d actually ask :) teehee |
| 03:13.42 | mitcheloc | i've only got 19 minutes of batter life left... *sigh* |
| 03:13.53 | Qwell | woo, nano! |
| 03:14.02 | [hC] | I edit config files with a hex editor |
| 03:14.11 | [hC] | Ask if he edits extensions.conf with windows notepad |
| 03:14.12 | [hC] | :P |
| 03:14.13 | [hC] | j/k |
| 03:14.41 | mitcheloc | i've actually used notepad...go figure |
| 03:14.53 | Qwell | just unix2dos and dos2unix it |
| 03:14.54 | [hC] | what is a DID Alert Info? :P |
| 03:14.55 | *** join/#asterisk tlow (n=tlowe@voip.terrorist.net) |
| 03:14.59 | nortex | Same here |
| 03:15.22 | nortex | Same as SIP Header I think. |
| 03:15.23 | mitcheloc | there you go :) |
| 03:15.36 | Qwell | You should mention that you're getting questions from IRC :p |
| 03:15.37 | *** join/#asterisk laurence (n=laurence@DHCP-112-206.caltech.edu) |
| 03:15.42 | Qwell | and that Qwell says "w00t" |
| 03:16.13 | mitcheloc | Kerry carpooled with me, i'm sure he knows i'm not asking these questions... |
| 03:16.17 | Qwell | heh |
| 03:16.22 | rrittenhouse | I bet you get this a lot. I want to mess with asterisk pretty bad. I have no phone line coming in nor do i have any sip phones haha. What *could* i technically do with it (just curious) |
| 03:16.23 | mitcheloc | ;) |
| 03:16.30 | Qwell | snow-m? |
| 03:16.34 | [hC] | You may note that the questions im asking are coming all the way from san jose costa rica today |
| 03:16.34 | [hC] | :P |
| 03:16.37 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:16.47 | [hC] | Eek, stream stalled. |
| 03:17.04 | nortex | rrittenhouse: You could download a sip softphone and get an account with an ITSP |
| 03:17.09 | rrittenhouse | ah |
| 03:17.38 | rrittenhouse | well time warner just took over our adelphia and im thinking about getting their "digital phone" if its worth it |
| 03:17.44 | [hC] | Is he talking about the spa941"? |
| 03:17.45 | Qwell | rrittenhouse: don't do it |
| 03:17.49 | Qwell | [hC]: yeah |
| 03:17.51 | rrittenhouse | but i need to look into an ITSP |
| 03:17.54 | Qwell | or 841? |
| 03:17.55 | [hC] | they're really not that good. :( |
| 03:17.59 | Qwell | nope... |
| 03:18.03 | [hC] | They're echoey |
| 03:18.07 | [hC] | crackly |
| 03:18.08 | rrittenhouse | oh man :( |
| 03:18.17 | [hC] | For the price, the polycom ip430 blows it out of the water. |
| 03:18.22 | Qwell | so I hear |
| 03:18.37 | [hC] | aw man.. my stream stalled right before he answered the coolness factor question |
| 03:19.31 | [hC] | Its not good for your eyes to work in a dungeon |
| 03:20.01 | Qwell | chan_skinny has unlimited softkeys |
| 03:20.08 | Qwell | Thank you very much |
| 03:20.15 | [hC] | ooh chan_sccp was abaondoned somewhat recently |
| 03:20.23 | [hC] | Im going to have to switch to chan_skinny after al |
| 03:20.29 | [hC] | that is, if i can find enough friggin time |
| 03:20.45 | mitcheloc | time, huh? |
| 03:20.50 | [hC] | ........ he didnt like polycom?!?! |
| 03:20.57 | Qwell | polycom sidecars |
| 03:20.59 | [hC] | aw man |
| 03:21.00 | [hC] | snom |
| 03:21.01 | Qwell | short o.. |
| 03:21.13 | dlynes_laptop | heh |
| 03:21.18 | dlynes_laptop | he's complaining about polycom |
| 03:21.23 | dlynes_laptop | and raving about linksys |
| 03:21.24 | dlynes_laptop | wtf? |
| 03:21.31 | Qwell | dude |
| 03:21.32 | [hC] | hello blackberry messenger |
| 03:21.32 | dlynes_laptop | oh....and now loves grandstream |
| 03:21.34 | Qwell | there are like...5 people there |
| 03:21.38 | nortex | There is not even a Poly on the desk |
| 03:21.43 | dlynes_laptop | wtf? |
| 03:21.46 | Qwell | mitcheloc: raise your hand :p |
| 03:21.46 | dlynes_laptop | THE GRANDSTREAM SUCKS |
| 03:21.52 | Qwell | non-chalant(sp) |
| 03:22.00 | file | MyFirstVoIPPhone(tm) |
| 03:22.03 | mitcheloc | I'm actually not in the picture |
| 03:22.06 | Qwell | lame |
| 03:22.14 | dlynes_laptop | woah |
| 03:22.17 | dlynes_laptop | the picture's clear now |
| 03:22.21 | mitcheloc | see the guy in the bottom right? i'm just behind him to the right |
| 03:22.29 | Qwell | scoot over, heh |
| 03:22.38 | [hC] | as far as what? |
| 03:22.45 | [hC] | oh asterisk |
| 03:22.51 | rrittenhouse | any suggestions on a softphone to mess with asterisk (under linux) |
| 03:22.58 | phifli_ | was anyone here at the con in europe |
| 03:23.00 | [hC] | what sorts of problems do polycoms have |
| 03:23.01 | [hC] | ask him that |
| 03:23.05 | [hC] | whats wrong? |
| 03:23.10 | Qwell | umm |
| 03:23.13 | [hC] | it has blf |
| 03:23.14 | Qwell | polycom HAS blf...troll |
| 03:23.20 | nortex | rrittenhouse: Check out xlite from counterpath. |
| 03:23.20 | Qwell | call him on it ;/ |
| 03:23.22 | mitcheloc | okay, i got a short battery life left |
| 03:23.25 | file | trolllllllll |
| 03:23.31 | rrittenhouse | nortex, thx |
| 03:23.32 | mitcheloc | i'm going to sleep my laptop, come on in a bit and i'll wave at you all ;) |
| 03:24.05 | nortex | mitcheloc: later |
| 03:24.17 | Qwell | ~blf |
| 03:24.18 | jbot | somebody said blf was Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 03:24.18 | dlynes_laptop | wtf? |
| 03:24.18 | [hC] | ask ummmmmm |
| 03:24.22 | Qwell | dude |
| 03:24.24 | [hC] | blind lamp function? |
| 03:24.24 | Qwell | CALL HIM ON IT |
| 03:24.25 | dlynes_laptop | now he's saying he'll use polycoms |
| 03:24.41 | dlynes_laptop | he says they suck, but he still uses them |
| 03:24.43 | dlynes_laptop | wtf???? |
| 03:25.25 | Qwell | laurence: yell at him, please |
| 03:25.31 | Qwell | "Busy Lamp Field" |
| 03:26.02 | dlynes_laptop | now buddy in the green shirt's picking his nose on national tv |
| 03:26.03 | dlynes_laptop | wtf? |
| 03:26.07 | Qwell | haha |
| 03:26.09 | [hC] | ask if he knows, will asterisk show my receptionist if line1 is in use, or line2? |
| 03:26.11 | [hC] | :P |
| 03:26.26 | [hC] | I think buddy in the green shirt earlier asked a question |
| 03:26.32 | [hC] | and i thought he was a 11 year old girl in the audience |
| 03:26.34 | [hC] | im not sure. |
| 03:26.49 | dlynes_laptop | and it's 'neesh', not 'nitch' |
| 03:26.50 | dlynes_laptop | sheesh |
| 03:26.54 | nortex | He waived at the camera a bit ago |
| 03:27.18 | Qwell | ooo, I like that PROXIMA fade in |
| 03:28.07 | nortex | A grand for a killer server??? |
| 03:28.31 | [hC] | erm |
| 03:28.37 | [hC] | dell sc430 = approx 450 bucks |
| 03:28.52 | nortex | Would you call that a killer server? |
| 03:28.59 | [hC] | p4 2.6ghz with 1gb ram and 120gb sata drive |
| 03:29.04 | [hC] | its pretty damn good for the price. |
| 03:29.25 | Damin | Where is the video stream for this presenation? |
| 03:29.27 | Qwell | without what? |
| 03:29.28 | Qwell | http://stream.sgvlug.org:8800/ |
| 03:29.56 | [hC] | a 'restart now' works even quicker. |
| 03:30.00 | Qwell | omfg |
| 03:30.01 | [hC] | or, restart when convenient' |
| 03:30.06 | Qwell | "wait for no calls" |
| 03:30.10 | Damin | What the hell is it streaming with? |
| 03:30.16 | Qwell | Damin: mplayer? |
| 03:30.27 | dlynes_laptop | [hC]: i thought a restart now has a potential for memory leaks? |
| 03:30.28 | Qwell | or, the server part? |
| 03:30.30 | [hC] | Damin: a Logitech Quickcam from 1994. |
| 03:30.45 | laurence | Damin: You mean the codec? Ogg/Theora. |
| 03:31.00 | [hC] | Hahaha. |
| 03:31.01 | laurence | Or leave off the port and you get a java applet. |
| 03:31.01 | [hC] | 'Wife' |
| 03:31.04 | [hC] | Have you seen the audience? |
| 03:31.09 | [hC] | (no offense mitcheloc) |
| 03:31.20 | [hC] | These guys arent out of school yet! |
| 03:31.21 | JT | i'd want at least raid hard drives and a redundant power supply in my killer asterisk server |
| 03:31.34 | Qwell | ask about e911 |
| 03:31.38 | Qwell | with voip |
| 03:31.53 | [hC] | oh ive got a good one |
| 03:31.55 | file | haha |
| 03:32.03 | [hC] | "Ive got a Cisco 7970, but i dont have the firmware for it, can i get a copy from you?" |
| 03:32.07 | Qwell | heh |
| 03:32.14 | [hC] | :P |
| 03:32.17 | [hC] | </troll> |
| 03:32.18 | nortex | :P |
| 03:32.40 | dlynes_laptop | what's his nick on irc? |
| 03:32.48 | Qwell | dlynes_laptop: asterisk_nub, I imagine |
| 03:32.53 | dlynes_laptop | heh |
| 03:32.56 | dlynes_laptop | woah |
| 03:32.56 | [hC] | Kelly_G or something |
| 03:33.10 | dlynes_laptop | Almost at the end of the presentation, he finally mentions digium cards |
| 03:33.14 | dlynes_laptop | wtf? |
| 03:33.25 | [hC] | So far i still refuse to use digium cards |
| 03:33.35 | [hC] | maybe after this round of financing, they'll step it up |
| 03:33.36 | dlynes_laptop | heh |
| 03:33.39 | nortex | Otherwise he might have to explain fxo and fxs to then |
| 03:33.41 | [hC] | Ooh |
| 03:33.43 | Qwell | ha |
| 03:33.45 | dlynes_laptop | xed2 admits to being the guy who picked his nose in the front |
| 03:33.49 | dlynes_laptop | smart |
| 03:34.13 | [hC] | Ask him 'how can i solve the issue of my internal office network being saturated, say by downloads or a virus, and killing my phone calls? ie QoS.. |
| 03:34.19 | [hC] | especially when you're using the phone as a switch |
| 03:34.19 | [hC] | :) |
| 03:34.31 | mitcheloc | i asked about 911.... watching the room over laurences shoulder.... ;) |
| 03:34.40 | Qwell | mitcheloc: he completely avoided the question |
| 03:34.56 | mitcheloc | maybe he is leading up to it |
| 03:35.02 | mitcheloc | got to sleep laptop... |
| 03:35.07 | [hC] | chow |
| 03:35.18 | [hC] | Ask about sangoma :) |
| 03:35.25 | [hC] | hee hee |
| 03:36.24 | Qwell | and if your net connection is down, then what? |
| 03:36.39 | [hC] | Hah, or 'hey so ive got trixbox installed here but i cant get my xlite to register, can you help me?' |
| 03:36.42 | [hC] | :P |
| 03:36.43 | Damin | SPA 3000 lets you do 911 pass through.. |
| 03:36.55 | Qwell | Damin: true, if you have an analog line |
| 03:37.00 | Qwell | but, he's talking specifically about SIP |
| 03:37.02 | Qwell | ITSP |
| 03:37.06 | Damin | It also has a no-power cross connect between the FXO and FXS ports.. |
| 03:37.20 | Damin | So if you don't have power, it bridges the two.. |
| 03:37.26 | Qwell | Damin: yeah |
| 03:37.33 | dlynes_laptop | Damin: and a no-network cross connect too |
| 03:37.51 | dlynes_laptop | Damin: epygi's also do that |
| 03:37.53 | [hC] | ask him how much cpu per call |
| 03:37.58 | Damin | dlynes_laptop: So that if the network is down it hardwires the ethernet to the FXS? :) |
| 03:38.17 | dlynes_laptop | Damin: if the network is down, it hardwires the fxs to the fxo |
| 03:38.26 | dlynes_laptop | Damin: also known as a lifeline |
| 03:38.52 | Damin | dlynes_laptop: Hehe.. I know.. I was just messing with you.. |
| 03:39.19 | dlynes_laptop | Damin: dood...you weren't messing with me...you were messing your pants...and now you stink |
| 03:39.22 | dlynes_laptop | Damin: get away |
| 03:39.51 | [hC] | Haha |
| 03:40.32 | Qwell | yeah, because emergencies don't happen during the summer when it's hot |
| 03:40.35 | Qwell | ... |
| 03:40.48 | Idle | they dont |
| 03:40.53 | [hC] | wonder what his biggest install is |
| 03:41.00 | nortex | And a hour of being down is no big deal :) |
| 03:41.19 | Damin | nortex: Unless you are a phone-sex provider.. |
| 03:41.21 | nortex | He said something about 90 phones |
| 03:41.31 | Qwell | "couple grand"? |
| 03:41.34 | Idle | how can you cluster asterisk? |
| 03:41.35 | [hC] | Whhhhhattt |
| 03:41.39 | Qwell | NO! |
| 03:41.42 | Qwell | 1500 |
| 03:41.44 | [hC] | This is why i buy sangoma. |
| 03:41.45 | nortex | Or you company likes to make sales by phone. |
| 03:41.48 | Damin | Idle: He meant "cluster fuck" |
| 03:42.01 | [hC] | a102d = $600 bucks. |
| 03:42.05 | Idle | what? I was actually asking how you would... :S |
| 03:42.13 | [hC] | THANK you |
| 03:42.18 | Qwell | heh |
| 03:42.31 | dlynes_laptop | now he mentions sangoma :) |
| 03:42.39 | [hC] | ...but dont use them, because that would be a travesty... |
| 03:42.46 | dlynes_laptop | no |
| 03:42.54 | dlynes_laptop | don't use them, because file's heart will be broken |
| 03:43.04 | [hC] | er.. |
| 03:43.05 | [hC] | ./Setup install |
| 03:43.07 | [hC] | is pretty easy |
| 03:43.08 | [TK]D-Fender | [hC] : EC version of 102? |
| 03:43.08 | [hC] | :) |
| 03:43.13 | [hC] | [TK]D-Fender: yeah! |
| 03:43.24 | [TK]D-Fender | [hC] :news to me.. going to look for now.... |
| 03:43.29 | [hC] | I realized not too long ago I have a non EC a102 |
| 03:43.32 | [hC] | and want to replace it |
| 03:43.34 | [TK]D-Fender | 600 is unbelieveably cheap |
| 03:43.43 | [hC] | I got the price from my sangoma rep |
| 03:43.48 | Qwell | rhino channelbank does rock, I hear |
| 03:43.52 | [hC] | they announced the card a while ago but i couldnt find anywhere to buy it on line. |
| 03:43.59 | [TK]D-Fender | [hC] : thats the price of the NORMAL one... |
| 03:44.13 | Qwell | it IS NOT $2k |
| 03:44.14 | nortex | hC Voipsupply had it last I checked. |
| 03:44.26 | file | I don't care what you use :P |
| 03:44.28 | [hC] | [TK]D-Fender: I may be ~$100 off, I just remember it was around the 600 mark |
| 03:44.30 | file | use what works for you! |
| 03:44.32 | Qwell | bah, sissies need to call him on it |
| 03:44.46 | Qwell | laurence: That means you :P |
| 03:44.50 | [TK]D-Fender | [hC] : I don't care if its $1000...... |
| 03:44.59 | dlynes_laptop | or mitcheloc |
| 03:45.04 | [TK]D-Fender | [hC] : its much cheaper than 1/2 the A104D |
| 03:45.12 | dlynes_laptop | come on mitch!!! |
| 03:45.16 | Qwell | go up there |
| 03:45.29 | nortex | mitch's laptop went to sleep |
| 03:45.37 | file | lol |
| 03:46.19 | [TK]D-Fender | [hC] : the 102u goes for 770$ |
| 03:46.28 | [TK]D-Fender | [hC] : I want to see it in print :) |
| 03:46.35 | [hC] | Where did you find the price? |
| 03:46.45 | [hC] | I get -25% from sangoma directly |
| 03:46.46 | [hC] | im a reseller |
| 03:46.55 | dlynes_laptop | [hC]: same here |
| 03:47.03 | dlynes_laptop | [hC]: i would never pay $770 |
| 03:47.23 | dlynes_laptop | [hC]: about $550-600USD for a 102u |
| 03:47.27 | [TK]D-Fender | [hC] : Atacomm. |
| 03:47.40 | [TK]D-Fender | [hC] : I can't find the A102d listed anywhere... |
| 03:47.54 | dlynes_laptop | [TK]D-Fender: because it doesn't exist |
| 03:48.07 | dlynes_laptop | [TK]D-Fender: the a101d/a102d don't exist |
| 03:48.25 | [hC] | 102d does infact, it was announced the same time as the 104d |
| 03:48.26 | dlynes_laptop | YET |
| 03:48.32 | [hC] | I just ordered one |
| 03:48.36 | dlynes_laptop | eh? |
| 03:48.51 | dlynes_laptop | why'd they tell me it doesn't, when I talked to my rep? |
| 03:49.03 | file | dlynes_laptop: you talked to someone in another universe |
| 03:49.16 | dlynes_laptop | file: no...I talked to someone in the sangoma office |
| 03:49.16 | [hC] | when did you ask? |
| 03:49.20 | [hC] | I ordered 4 days ago |
| 03:49.24 | [hC] | although |
| 03:49.24 | [TK]D-Fender | [hC] : yeah I see recent references, but no official product page anywhere. So its pending. |
| 03:49.25 | dlynes_laptop | about 1-1/2 months ago or so |
| 03:49.33 | Damin | Creepy? |
| 03:49.35 | [hC] | I didnt do it personally, my hardware ordering dude did |
| 03:49.41 | [hC] | so I may need to confirm with him |
| 03:49.49 | dlynes_laptop | [hC]: umm...announced the same time as the 104d? the 104d's been out for a while |
| 03:50.13 | [hC] | dlynes_laptop: it was announced at the same time, not released at the same time. |
| 03:50.33 | [hC] | is that mitcheloc? |
| 03:50.34 | dlynes_laptop | [hC]: ah...well, the a200 hwec was announced a while ago, too |
| 03:50.38 | *** join/#asterisk niZon (n=ilt@S0106beefd4cecc3d.wp.shawcable.net) |
| 03:50.41 | dlynes_laptop | [hC]: but it didn't come out until June |
| 03:50.46 | [hC] | Yeah. |
| 03:50.54 | laurence | [hC]: It's not him. |
| 03:50.55 | dlynes_laptop | same for the a108d |
| 03:50.59 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-67-79-2-146.sw.biz.rr.com) |
| 03:51.15 | [hC] | Blackberry can play wav |
| 03:51.17 | CoffeeIV | what does this mean: "Unable to find a codec translation path from g729 to slin" |
| 03:51.19 | [hC] | 8700 can anwyays. |
| 03:51.30 | file | CoffeeIV: your Asterisk install can't transcode G729 |
| 03:52.27 | laurence | Qwell: I started nodding off, I'm not awake enough to ask your questions. Plus I'm not feeling particularly mean, so I'm not going to "call him" on it. |
| 03:52.35 | Damin | CoffeeIV: It means that asterisk was unable to find a codec translation patch from g729 to Signed Linear. |
| 03:52.37 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:52.41 | Qwell | well, he's giving incorrect information... |
| 03:52.44 | Qwell | and that isn't cool |
| 03:52.50 | CoffeeIV | file: does that mean that inorder to handle calls, I need to either get g729 or make my provider send me calls in another codec ? |
| 03:53.06 | Damin | He's freaking stealing my lines.. |
| 03:53.12 | dlynes_laptop | hehe |
| 03:53.16 | phifli_ | haha |
| 03:53.20 | Damin | I use that line in my Intro to Asterisk presentations.. |
| 03:53.29 | Qwell | Damin: which? "lowered the bar"? |
| 03:53.36 | [hC] | Qwell: haha. |
| 03:53.37 | [hC] | um |
| 03:53.37 | file | CoffeeIV: either both sides need to use the same codec (g729) or you need to not use g729, or buy a license |
| 03:53.40 | Damin | "Thank God for cell phones, because they have lowered expectations" |
| 03:53.43 | Qwell | yeah |
| 03:53.43 | [hC] | vonage is not pronounced that wya |
| 03:53.47 | CoffeeIV | ok |
| 03:53.47 | [hC] | for the love of god |
| 03:53.50 | Qwell | heh |
| 03:54.04 | Qwell | even $900B in advertising can't get people to pronounce it right |
| 03:54.08 | Damin | Did he say "Vohnaj"? |
| 03:54.15 | phifli_ | haha |
| 03:54.17 | Qwell | Damin: Voh-naj-ee |
| 03:54.45 | [hC] | vonajh |
| 03:55.02 | Damin | Who is this guy? |
| 03:55.08 | Qwell | Damin: some troll :p |
| 03:55.16 | Qwell | he's telling like...half-truths |
| 03:55.20 | Damin | No.. the Svlug guy.. |
| 03:55.20 | [hC] | he runs voipspeak.net |
| 03:55.34 | [hC] | so, some nerd who has been using trixbox for 6 mos |
| 03:55.36 | [hC] | :) |
| 03:55.40 | suma | is there is any IAX FXO device available ? |
| 03:55.58 | [hC] | get outta here. |
| 03:56.00 | [hC] | gpl. |
| 03:56.13 | [hC] | yes because im sure customers give a shit about GPL. |
| 03:57.01 | laurence | [hC]: Chill. He's saying that because it's a LUG. It's a running joke for this talk. :-) |
| 03:57.08 | [hC] | I know :) |
| 03:57.14 | Damin | suma: Not sure. |
| 03:57.30 | Damin | suma: the Iaxy is just FXS.. |
| 03:57.51 | suma | Damin: yes, i'm familier with that, but looking for a FXO device |
| 03:57.53 | Damin | suma: But I think that there was an australian company that did an IAX ata w/ FXo.. |
| 03:58.16 | suma | Damin: you got their name ? |
| 03:58.20 | Qwell | until the end of the year or so |
| 03:58.59 | [hC] | chan_twocansandstring.so |
| 03:59.20 | [hC] | I use it every day! :) |
| 03:59.46 | [hC] | I find it really ironic how he was doing the GPL joke at the LUG, yet his laptop runs winxp. :) |
| 03:59.51 | Damin | suma: I'm looking.. |
| 03:59.53 | dlynes_laptop | who's the doofus up there speaking, now? |
| 03:59.58 | dlynes_laptop | is it mitcheloc? |
| 04:00.04 | Qwell | no |
| 04:00.06 | [hC] | apparently not |
| 04:00.12 | laurence | [hC]: See? It's ironic! :-) |
| 04:00.26 | dlynes_laptop | oh |
| 04:00.28 | dlynes_laptop | it's laurence |
| 04:00.35 | laurence | The "doofus" is Matti. |
| 04:00.39 | dlynes_laptop | hehe |
| 04:00.44 | dlynes_laptop | ah...the asterisk developer? |
| 04:00.54 | Qwell | huh? |
| 04:00.57 | laurence | Yah, I'm typing telepathically while you see me standing there. :-) |
| 04:01.14 | dlynes_laptop | nvm |
| 04:01.29 | Damin | suma: It was virbiage: http://www.virbiage.com/3010.php But it is only an FXS device. |
| 04:01.31 | dlynes_laptop | i thought there was somebody in #asterisk-dev with the nick matti or something like that |
| 04:01.54 | Qwell | You can use vonage with asterisk... |
| 04:02.00 | Qwell | It's...SIP |
| 04:02.24 | Damin | Qwell: But why would you want to? :) |
| 04:02.31 | Qwell | Damin: he's saying you can't |
| 04:02.51 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 04:03.13 | Qwell | haha |
| 04:03.54 | Qwell | BS!! |
| 04:04.04 | Qwell | "Every cell provider" |
| 04:04.27 | [hC] | damn, what did i miss? I went to appease my wife |
| 04:04.47 | file | QW |
| 04:04.49 | Qwell | [hC]: "You can set your cidnum as the callee, and get right into their VM. It works with every cell provider" |
| 04:04.50 | file | Qwell: !!!!!!!!!!!!!!!! |
| 04:04.56 | [hC] | Hah |
| 04:05.01 | [hC] | Maybe in 1998 |
| 04:05.24 | Sedorox | sprint it does |
| 04:05.27 | Sedorox | verizon I know doesn't |
| 04:05.28 | Sedorox | nor nextel |
| 04:05.30 | Qwell | Sedorox: No it doesn't |
| 04:05.31 | Sedorox | nor cingular |
| 04:05.33 | Sedorox | hehe |
| 04:05.34 | Qwell | not sprint |
| 04:05.40 | Sedorox | well thats what I was told |
| 04:05.43 | Sedorox | I haven't tried it myself |
| 04:05.47 | Qwell | I have :) |
| 04:05.53 | Sedorox | hehe so I'll go wirh your word then |
| 04:06.07 | Sedorox | tmobile is dumb enough to do that |
| 04:06.10 | laurence | Qwell: So what date shall I put you down for your own Asterisk talk? |
| 04:06.15 | [hC] | tmobile does not |
| 04:06.16 | [hC] | i tried |
| 04:06.19 | Qwell | some (most) providers, however, DO check "in network calling" via cidnum |
| 04:06.19 | [hC] | cingular does not |
| 04:06.20 | Sedorox | lol |
| 04:06.21 | Qwell | laurence: schedule me |
| 04:06.30 | [hC] | they all check ani now |
| 04:06.30 | Qwell | laurence: I'm right here in wsco |
| 04:06.44 | Qwell | claude...umm...who? |
| 04:06.52 | [hC] | junky? |
| 04:06.53 | Qwell | klaud, even? |
| 04:06.54 | laurence | Qwell: Easily done, the schedule is pretty open. It's a popular subject, we could just wait a couple of months and it would go over big. |
| 04:06.56 | Qwell | [hC]: no :p |
| 04:07.05 | Qwell | laurence: I'm only in wsco until the end of Oct or so |
| 04:07.19 | Qwell | wait, no, person I'm thinking of is "Klause", heh |
| 04:07.27 | laurence | Qwell: Are you close enough to Pasadena to join us at BC? |
| 04:07.33 | Qwell | laurence: wsco.. |
| 04:07.52 | laurence | Qwell: mitcheloc wants you to come buy..... |
| 04:08.06 | laurence | wsco? |
| 04:08.09 | Qwell | What, tonight? heh |
| 04:08.12 | Qwell | West Covina.. |
| 04:08.47 | laurence | Qwell: Ah. Well, go down the 210, Get off at Lake, go South to California. We're often there until 1AM talking on the sidewalk after BC closes. :-) |
| 04:08.55 | Qwell | BC? |
| 04:09.07 | laurence | Burger Continental. |
| 04:09.13 | Qwell | never heard of it |
| 04:09.27 | laurence | In spite of the name it's kind of middle-eastern. |
| 04:09.50 | laurence | Qwell: So, are you going to come by? |
| 04:09.53 | [hC] | haha. oh yes, security people dont use email. |
| 04:10.07 | Qwell | laurence: another time, sure. I don't drive :) |
| 04:11.02 | laurence | Qwell: Email me your contact info so we can set up a talk: dustin@laurences.net. |
| 04:11.26 | file | moo |
| 04:11.36 | [hC] | ahh, internet taking a crap. |
| 04:11.56 | Qwell | laurence: see msg |
| 04:12.05 | Qwell | and, keep that to yourself for about...2 days :) |
| 04:12.22 | Qwell | (between the two of you) |
| 04:15.39 | Qwell | WTF?! |
| 04:15.47 | Qwell | No drivers for debian?! |
| 04:15.51 | Qwell | IT'S LINUX |
| 04:15.59 | Qwell | laurence: I rest my case. :) |
| 04:17.35 | mitcheloc | okay real quick i'll wave haha |
| 04:17.48 | Qwell | nub |
| 04:17.51 | [hC] | mitcheloc was the bald guy? |
| 04:18.03 | mitcheloc | is it still on? |
| 04:18.06 | mitcheloc | no i'm not balk! |
| 04:18.07 | Qwell | yeah |
| 04:18.08 | mitcheloc | * bald |
| 04:18.09 | [hC] | yep |
| 04:18.13 | mitcheloc | qwell you can see me then? |
| 04:18.15 | Qwell | yep |
| 04:18.20 | [hC] | you're kinda kevin rose lookin |
| 04:18.20 | Qwell | can you see us? |
| 04:18.30 | file | yay |
| 04:18.43 | mitcheloc | i'm in the bottom right |
| 04:18.44 | mitcheloc | haha |
| 04:18.51 | mitcheloc | anyways battery is seriously low |
| 04:18.52 | [hC] | ooh that girl looks like she might be kinda cute |
| 04:18.53 | Qwell | it's off |
| 04:18.56 | mitcheloc | Qwell: too bad you can't make it tonight |
| 04:19.02 | Qwell | come get me :P |
| 04:19.04 | [hC] | im still streaming |
| 04:19.08 | mitcheloc | Qwell: i can |
| 04:19.10 | [hC] | ah now it died. |
| 04:19.16 | Qwell | mitcheloc: I wish, heh |
| 04:19.21 | Qwell | work tomorrow ;/ |
| 04:19.25 | mitcheloc | Qwell: pmed my cell if you change your mind |
| 04:19.27 | file | last day! |
| 04:19.27 | Qwell | and my toe is broken, heh |
| 04:19.53 | Qwell | indeed :) |
| 04:20.07 | shido6 | erf? |
| 04:20.08 | mitcheloc | Qwell: don't be too anti-kerry, he may not have had everything right, but he still presented most of the basics ;) |
| 04:20.22 | Qwell | mitcheloc: I'm not.. just wish he wouldn't have misstated :) |
| 04:20.49 | Qwell | mitcheloc: PM me that # again |
| 04:21.57 | *** part/#asterisk tlow (n=tlowe@voip.terrorist.net) |
| 04:24.45 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 04:28.51 | Qwell | What? ;/ |
| 04:28.55 | *** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home) |
| 04:28.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 04:28.56 | file | wazzzzzzzup |
| 04:29.07 | *** join/#asterisk tlow (n=tlowe@voip.terrorist.net) |
| 04:34.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
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| 04:43.40 | *** part/#asterisk tlow (n=tlowe@voip.terrorist.net) |
| 04:51.00 | *** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 05:03.14 | *** join/#asterisk Jenna (i=CherryRe@gateway/tor/x-1b29eab1b23fa613) |
| 05:03.37 | *** part/#asterisk Jenna (i=CherryRe@gateway/tor/x-1b29eab1b23fa613) |
| 05:07.15 | Snake-Eyes | any one want to have another crack at softhangup problem I have :) |
| 05:09.29 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 05:25.36 | x86 | maybe if someone had awesome telepathy they could know what the problem was, and if you were lucky enough, they might have the answer too... until then, you probably wont get much help ;) |
| 05:30.20 | *** join/#asterisk [hC] (n=hardcore@190.10.9.191) |
| 05:43.10 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:44.41 | *** join/#asterisk daysmen3 (n=primus@host86-138-214-197.range86-138.btcentralplus.com) |
| 05:47.10 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 05:48.07 | Snake-Eyes | ha ha ha |
| 05:48.54 | Snake-Eyes | well now that i have baited you :P |
| 05:52.46 | Snake-Eyes | x86, this now the 3-4 time ive asked (loosing count now). Is softhangup ment to hangup all calls on a trunk ? eg I have macro that calls softhangup and doesn't seem hangup any other calls on the SIP trunk when its called (tried SoftHangup(SIP/trunk-sx-1) SoftHangup(SIP/trunk-sx|a) |
| 05:55.13 | hads|home | According to 'show application softhangup' that's how it should work |
| 05:55.41 | Snake-Eyes | well its not .... :( |
| 05:56.23 | Snake-Eyes | mog was saying something about sub channels but he went to sleep and i couldnt get any more out of him |
| 05:58.43 | Snake-Eyes | hads|home, any ideas ? |
| 06:03.33 | hads|home | Not really, looks like it should work. |
| 06:06.16 | x86 | why not just use Hangup() |
| 06:07.44 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 06:08.24 | rushowr | hey all, it's me again :) anyone in here have a good understanding of macros, particularly about variable inheritance? |
| 06:09.12 | rushowr | I've got macros calling macros, which in turn sometimes call other macros. It doesn't always work, and I'm figuring it's due to inheritance of variables |
| 06:09.17 | rushowr | but looking for confirmation |
| 06:09.33 | rushowr | and yes I've read the Asterisk Macros page on voip-info |
| 06:11.37 | Snake-Eyes | x86, I don't want to hangup the call, I want to hangup another call/channel on the trunk |
| 06:11.42 | Snake-Eyes | http://pastebin.ca/126159 |
| 06:12.50 | hads|home | Snake-Eyes: You should get a warning message "Soft hanging %s up." for each channel |
| 06:14.04 | phifli_ | set verbose 9999 |
| 06:14.05 | phifli_ | ;) |
| 06:14.13 | hads|home | Also, regarding your dialplan logic, what if the call on that trunk is already an emergancy call? You will kill their call. |
| 06:14.17 | rushowr | •phifli_• who? |
| 06:14.22 | phifli_ | do that on asterisk |
| 06:14.25 | phifli_ | set debug 999 |
| 06:14.26 | phifli_ | lol |
| 06:14.28 | phifli_ | if you wanna see all messages |
| 06:14.30 | Snake-Eyes | hads|home, not getting that warning |
| 06:14.39 | rushowr | •phifli_• I take it you're not talking to me...... |
| 06:14.57 | phifli_ | speaking in general to see error messages |
| 06:14.58 | Snake-Eyes | hads|home, that dial plan is for testing, ill change it later once i know softhangup works |
| 06:15.05 | phifli_ | has|home said "Soft haing up o %s" |
| 06:15.06 | phifli_ | whatever |
| 06:15.07 | *** join/#asterisk _omer (n=_omer@202.38.51.2) |
| 06:15.16 | rushowr | ah, yeah not me :) |
| 06:15.26 | _omer | hi |
| 06:15.29 | rushowr | I guess no one knows about the macro thing |
| 06:15.46 | Snake-Eyes | phifli_, ok |
| 06:16.08 | rushowr | •Snake-Eyes• Softhangup works |
| 06:16.16 | hads|home | Snake-Eyes: Well does 'show channels' show a SIP/back-trunk-ulaw-1 channel? |
| 06:16.20 | _omer | how do I know if the number or DTMF I am getting is in E.164 standard?? |
| 06:16.32 | rushowr | test it |
| 06:16.55 | rushowr | •_omer• the newer trunk revisions have the function REGEX |
| 06:17.20 | _omer | REGEX ??? |
| 06:17.22 | rushowr | •Snake-Eyes• I use Softhangup constantly, it works |
| 06:17.34 | Snake-Eyes | hads|home, yes but i double check in a sec with debug set nice and high |
| 06:17.41 | _omer | let me check wiki ... |
| 06:17.42 | rushowr | •_omer• As in Regular Expression, for testing your data |
| 06:17.43 | Snake-Eyes | rushowr, ok |
| 06:18.09 | Snake-Eyes | rushowr, can you give me example of your working softhangup? |
| 06:18.20 | rushowr | •Snake-Eyes• Softhangup(${CHANNEL}) |
| 06:18.32 | rushowr | SE, hangs up the current channel |
| 06:18.48 | _omer | rushowr : I just meant to say...if given number is with COUNTRYCODE or without COUNTRY CODE.. |
| 06:19.20 | rushowr | •_omer• test the length of the var, or maybe pass it to a context...thgere's many ways |
| 06:19.26 | rushowr | •sevard• Softhangup(${chan_to_hang}) would hangup whatever channel is spec'd |
| 06:19.36 | rushowr | oops, I meant that for Snake |
| 06:20.10 | Snake-Eyes | rushowr, is there a way to hangup the first channel being used? |
| 06:20.26 | rushowr | •Snake-Eyes• hrm... explain what you mean? |
| 06:20.34 | _omer | if number is for United Arab Emirates .....and second number is for UK ....then shouldn't be same..:-/ |
| 06:20.34 | rushowr | •Snake-Eyes• by first channel being used |
| 06:21.01 | rushowr | •_omer• there's soooo many ways to test the value of the DTMF |
| 06:21.29 | rushowr | •_omer• for instance: (I use AEL2, so bear with me) |
| 06:21.35 | *** join/#asterisk FlyBoySR22 (n=rsears@gateway.americanis.net) |
| 06:21.39 | Snake-Eyes | rushowr, 1st person places a call then a 2nd person places a call, when 3rd makes a call using the macro the 1st persons call is hangup |
| 06:21.48 | *** join/#asterisk neo (n=neo@kessel.ordrejedis.net) |
| 06:21.51 | phifli_ | well use channel macros rather than global |
| 06:21.58 | neo | hello :) |
| 06:21.59 | phifli_ | read the newest README on macros |
| 06:22.10 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.102.189) |
| 06:22.18 | rushowr | •Snake-Eyes• basically you just gotta pull the value of the channel name for the one you want to kill |
| 06:22.31 | rushowr | •Snake-Eyes• haven't ahd to do it yet, so don't have a fast answer on that |
| 06:23.18 | rushowr | •_omer• basically, I'm not trying to be an ass, but read up on asterisk dialplan functions on voip-info it's all there, all the crap you need to use for testing values |
| 06:23.20 | m_a_g_o | good evening folks, I just updated the SVN version and suddenly I'm getting this messages when loading the g729 and g723 modules... Segmentation fault (core dumped)... anyone has a clue, any idea? |
| 06:23.35 | rushowr | •m_a_g_o• yeah I hear there's a prob with 'em |
| 06:23.42 | phifli_ | heh |
| 06:23.42 | rushowr | •m_a_g_o• unfortunately that's all I know |
| 06:23.50 | phifli_ | i'd hack up the source if i had a system to do it on haha maye another day ={ |
| 06:23.51 | phifli_ | =P |
| 06:24.08 | rushowr | •phifli_• *chuckle* |
| 06:24.14 | Snake-Eyes | rushowr, damn, i was hoping '|a' would do it |
| 06:24.18 | JT | man, what is with your nick addressing, rushowr? |
| 06:24.27 | m_a_g_o | rushowr... :$ |
| 06:24.29 | neo | hum, has one of you already tested a Load Balancing Solution for Asterisk ? |
| 06:24.30 | rushowr | •JT• I've got some old IRC script running |
| 06:24.50 | JT | it looks like an inverse colour U is on each side of the nick |
| 06:25.04 | rushowr | over here it's white dot on the black backgroun |
| 06:25.17 | rushowr | thanks for the heads up though, I'll play with it |
| 06:25.35 | JT | well i have a white on black console |
| 06:25.39 | rushowr | lol |
| 06:25.46 | rushowr | that's fun, eh? |
| 06:25.46 | m_a_g_o | rushowr, do you know how I can get the curl function with 1.2? |
| 06:25.50 | neo | same here :) |
| 06:25.55 | JT | yeah, standard console |
| 06:26.04 | Qwell | m_a_g_o: install curl-dev |
| 06:26.05 | JT | i'd wager that the majority of the channel does :P |
| 06:26.09 | rushowr | •m_a_g_o• hrm...have you tried asterisk-backports? |
| 06:26.10 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 06:26.12 | rushowr | the site's great |
| 06:26.14 | rrittenhouse | i'm still trying to get xlite to connect to asterisk haha. |
| 06:26.18 | Snake-Eyes | hads|home, btw I will eventually have emergency calls on a seperate trunk but both trunks going to the same machine thus killing random calls on trunk A wont effect calls on emergency trunk :) |
| 06:26.19 | rushowr | or at least the idea |
| 06:26.23 | phifli_ | xlite works with asterisk |
| 06:26.26 | phifli_ | make sure you change the auth user |
| 06:26.38 | rrittenhouse | yeah but ive never messed with asterisk before and im just kinda trying to get the hang of whats what |
| 06:26.44 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 06:26.51 | rushowr | well, anyway mates, I gotta run and get back to this code, have a good one :) |
| 06:27.05 | m_a_g_o | rushowr: first time I hear about it... could you give me the address? |
| 06:27.07 | rrittenhouse | i see you add a "friend" in the sip.conf and then an extention in the extensions.conf or something..... |
| 06:27.13 | rushowr | one sec mGO |
| 06:27.14 | rushowr | mago |
| 06:27.22 | phifli_ | you can add em as user if you want |
| 06:27.29 | phifli_ | just make sure you turn off incoming calls :) |
| 06:27.45 | phifli_ | or you can add user/peer separately to get more contorl over it |
| 06:27.47 | phifli_ | control |
| 06:27.52 | m_a_g_o | Qwell, thanks, but could you point me where I can download the source? |
| 06:27.57 | phifli_ | but friend is easier for not customizing the setup |
| 06:28.00 | neo | sounds like nobody knows sip load balancing :'( |
| 06:28.02 | phifli_ | www.asterisk.org |
| 06:28.03 | rushowr | asterisk-backports.org |
| 06:28.04 | Qwell | m_a_g_o: What distro are you on? |
| 06:28.12 | m_a_g_o | fc3 |
| 06:28.13 | rrittenhouse | holy crap i got it :P |
| 06:28.18 | Qwell | yum install curl-devel |
| 06:28.19 | rushowr | cheers |
| 06:28.48 | Qwell | and obviously curl needs to be installed too |
| 06:28.52 | rrittenhouse | woo this is ..neat now that i got a user logged in hahah |
| 06:29.32 | phifli_ | lol |
| 06:29.38 | phifli_ | you'll get used to it =P |
| 06:29.45 | m_a_g_o | Qwell: thx, installing now... |
| 06:30.17 | neo | Qwell: sip load balancing, do you know something about it ? |
| 06:30.36 | Qwell | neo: could use something like ser |
| 06:31.54 | neo | hum, yes |
| 06:32.06 | neo | i would try something more like IPVS |
| 06:34.41 | *** join/#asterisk [hC] (n=hardcore@190.10.9.191) |
| 06:34.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:35.01 | Snake-Eyes | I see why rushowr has rush in his name lol |
| 06:37.46 | phifli_ | whys that |
| 06:37.57 | phifli_ | owr should be hour =/ |
| 06:38.00 | phifli_ | even our haha |
| 06:38.13 | phifli_ | gnite |
| 06:39.54 | dlynes_laptop | ~seen kerry_g |
| 06:39.57 | jbot | kerry_g <n=Kerry_G@ip70-187-129-227.oc.oc.cox.net> was last seen on IRC in channel #asterisk, 5d 5h 50m 41s ago, saying: 'no, you have spent 4 hours begging people to fix the problems you made'. |
| 06:40.09 | Qwell | heh |
| 06:40.40 | Juggie | ~seen theplot |
| 06:40.41 | jbot | theplot <i=ThePlot@202.164.38.210> was last seen on IRC in channel #asterisk, 4d 12h 41m 50s ago, saying: 'I did set in the address field to match the username too'. |
| 06:40.50 | Snake-Eyes | night phifli_ |
| 06:41.15 | phifli_ | take it easy |
| 06:41.26 | Snake-Eyes | he rushes in and out of the channel asking and answering questions |
| 06:51.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:04.32 | *** part/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:04.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:16.48 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:20.18 | suma | is asterisk working wonderful ? |
| 07:20.25 | suma | whu this channel is so silent ? |
| 07:20.43 | Dico_ | hello everybody |
| 07:20.58 | suma | hi dico |
| 07:20.59 | Dico_ | <PROTECTED> |
| 07:21.04 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 07:21.10 | suma | what is your problem ? |
| 07:21.11 | Dico_ | withouth the AMI ? |
| 07:21.18 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:21.18 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:21.21 | suma | linux they says, don't ask to ask |
| 07:21.27 | Dico_ | humm, i've been told there is a problem |
| 07:21.57 | Dico_ | suma, hello |
| 07:22.04 | Snake-Eyes | suma, the asterisk channel was silent for 39 mins :P |
| 07:22.07 | Dico_ | i dunno why the channe is so silent |
| 07:22.20 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 07:22.26 | Dico_ | but yes asterisk works :) |
| 07:23.07 | Dico_ | dlynes_laptop, are you around ? |
| 07:23.22 | Snake-Eyes | AU are going home, the US is still asleep and Euo is still waking up :P |
| 07:23.40 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:23.47 | Dico_ | 9.23 |
| 07:23.57 | Dico_ | it takes 23 min to drink a coffee ? |
| 07:24.04 | e-ddie | yeah |
| 07:24.08 | e-ddie | i didnt even get mine :( |
| 07:24.20 | Dico_ | humm, may be : the time to enjoy the office managers attached to the coffee ;) |
| 07:24.51 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:25.03 | Dico_ | hehe : small survey : |
| 07:25.08 | e-ddie | got it now : |
| 07:25.09 | e-ddie | :D |
| 07:25.23 | Dico_ | in your company, what is an office manager : 1 person ? |
| 07:25.27 | Dico_ | 1 team ? |
| 07:25.38 | Dico_ | is it equivalent to a secretary ? |
| 07:25.47 | e-ddie | the one responsible for installing office |
| 07:26.05 | Dico_ | lol, not the soft |
| 07:26.11 | Dico_ | real people i mean :p |
| 07:26.12 | e-ddie | ;) |
| 07:26.27 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:26.46 | Dico_ | I know there is cyber s... but real one is still better :p |
| 07:26.52 | Dico_ | ok, anyway |
| 07:27.13 | Dico_ | have you already tested the app_queue ? |
| 07:32.01 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 07:35.47 | *** join/#asterisk teapot (n=tandrews@mail.grok.org.za) |
| 07:35.59 | teapot | morning |
| 07:37.55 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 07:38.53 | Snake-Eyes | any one know if you can define a variable in sip,conf ? |
| 07:40.30 | teapot | only time I've ever seen that is being able to define an environment variable in odbc.conf |
| 07:40.35 | E-bola | Morning |
| 07:42.07 | Snake-Eyes | hmm |
| 07:48.24 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:50.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:01.20 | inspired | anyone else having problems with dtmf on 1.2.10? it worked fine on 1.0.7. after upgrading to 1.2.10 my apps seldom catch more than 3-4 digits before they just exit |
| 08:03.16 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 08:03.58 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-52-43.hsd1.il.comcast.net) |
| 08:04.11 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-144-231.dsl.pltn13.sbcglobal.net) |
| 08:04.12 | kuku5 | which file do I specify the different 10000-20000 port range? |
| 08:05.11 | darkgamer20 | is it possible to install freepbx on a custom installed asterisk system? where can I find the instructions to do so |
| 08:05.23 | hads|home | rtp.conf |
| 08:06.23 | hads|home | darkgamer20: freepbx.org? |
| 08:06.39 | adorah | http://www.freepbx.org/trac |
| 08:06.53 | darkgamer20 | hads|homes: i tried the guides/help link but it shows something about trac |
| 08:08.00 | Un1x | havent you guys read, the Topic |
| 08:08.16 | hads|home | Yes, |
| 08:08.18 | Un1x | FreePBX/Asterisk@home etc not supported here, join #freepbx |
| 08:08.41 | darkgamer20 | oops sorry didnt notice that |
| 08:14.01 | hads|home | Un1x: You can talk! |
| 08:14.26 | Un1x | lol, i was watching a movie debating weather i should watch another, or go sleep.. atm |
| 08:25.04 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 08:25.22 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 08:28.01 | teapot | what's the 'correct' way to yield/sleep in a module? |
| 08:28.33 | teapot | I use sched_yeild() but grepping for it I find only one other instance in the code |
| 08:38.21 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 08:42.13 | *** join/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com) |
| 08:43.54 | *** join/#asterisk jailbreaker (n=teodory@mail.jetfinanceintl.com) |
| 08:47.55 | phearless | hiiiiii |
| 08:50.34 | phearless | how can I set summer time / winter time on a cisco 7960 ? |
| 08:51.05 | phearless | i am in london but in summer we are at GMT+1 |
| 08:54.02 | mitcheloc | phearless: i'm guessing that you can try setting up a time server and the phones can synchronize off of it? |
| 08:54.37 | phearless | err.. i do not know |
| 08:54.52 | phearless | sntp_mode: "anycast" |
| 08:54.52 | phearless | sntp_server: "asterisk" |
| 08:54.53 | phearless | time_zone: "CET" |
| 08:54.57 | phearless | I use this for the monent |
| 08:55.01 | phearless | moment* |
| 08:55.08 | phearless | asterisk is my asterisk server |
| 08:55.14 | stinkpad | "BST"? |
| 08:55.27 | phearless | i will try BST |
| 08:56.09 | *** join/#asterisk Alex|Work (n=hauntedu@gentoo/user/Alex) |
| 08:58.48 | phearless | if I put BST in the SIPDefault, on the phone, it is GMT |
| 08:58.54 | phearless | this is weird |
| 09:02.47 | phearless | i really do not know how this time thing works |
| 09:03.37 | x86 | cant you use UTC ? |
| 09:04.07 | x86 | hmm, probably no hard-coded timezone will automatically adjust for DST though |
| 09:05.41 | *** join/#asterisk gandalfcome (n=wkerzend@p213.54.181.161.tisdip.tiscali.de) |
| 09:05.52 | *** part/#asterisk x86 (n=x86@p3m/member/x86) |
| 09:06.10 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 09:06.13 | Bert- | hello there |
| 09:06.37 | Bert- | just a little question : for what asterisk is listening on tcp port 2000 please ? |
| 09:09.06 | hads|home | It doesn't AFAIK unless you have told it to. |
| 09:09.20 | Bert- | AFAIK ? |
| 09:09.30 | gandalfcome | I want to set up a linux server with two isdn cards, one to recieve calls from the existing isdn connection and the second isdn card two call with a normal isdn phone via voip. Is this possible? Is that doable in a reasonable amount of time with good linux knowledge? Are there better packages for this task? Thanks in advance |
| 09:09.30 | hads|home | As Far As I Know. |
| 09:09.35 | Bert- | lol ok |
| 09:09.54 | hads|home | 2000 = # Sieve mail filter daemon |
| 09:10.07 | Bert- | ?? |
| 09:10.20 | hads|home | That's what /etc/services says |
| 09:10.21 | Bert- | in asterisk there is a mail filter daemon !? |
| 09:10.25 | Bert- | ho ok |
| 09:10.40 | Bert- | netstat -atnp shows that is asterisk listening onthat port |
| 09:10.55 | gandalfcome | with asterisk of course. |
| 09:11.17 | hads|home | Try `grep 2000 /etc/asterisk/*` see if it is set up to listen on that port. |
| 09:11.28 | Bert- | skinny |
| 09:11.32 | Bert- | done :) |
| 09:11.58 | Bert- | thx:! |
| 09:12.10 | hads|home | Ah, I don't use skinny. :) |
| 09:12.26 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 09:12.57 | Bert- | neither di I |
| 09:13.02 | Bert- | s/di/do |
| 09:13.27 | fourcheeze | Assid: did you find out any more about realtime clustering? |
| 09:13.41 | phearless | <x86> cant you use UTC ? <-- london is not in UTC time |
| 09:13.58 | hads|home | Bert-: You can noload => chan_skinny.so in modules.conf |
| 09:14.44 | Bert- | ho |
| 09:14.51 | Bert- | I deleted the conf file to disable it |
| 09:14.56 | Bert- | bad way ... |
| 09:15.32 | hads|home | Probably want to add a noload for it, otherwise it will just use it's default settings. |
| 09:15.52 | Bert- | hmm in fact if asterisk doesn't find the conf file, it disable the module |
| 09:16.23 | hads|home | Not nessecarily I don't think |
| 09:16.46 | Bert- | for skinny, mgcp, sql i'm sure |
| 09:17.08 | hads|home | OK, fair enough. Like I said I don't use them so I wouldn't know ;) |
| 09:20.15 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.27) |
| 09:20.20 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
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| 09:23.54 | Assid | fourcheeze: havent looked at it.. been busy trying to play with antispam settings |
| 09:24.25 | fourcheeze | Assid: I'm starting to hope that dundi may be the answer |
| 09:24.54 | *** join/#asterisk index1 (n=gobblin@124x34x136x123.ap124.ftth.ucom.ne.jp) |
| 09:25.12 | *** part/#asterisk index1 (n=gobblin@124x34x136x123.ap124.ftth.ucom.ne.jp) |
| 09:26.19 | fourcheeze | Assid: http://forums.digium.com/viewtopic.php?p=4820&] |
| 09:26.21 | phearless | what is the phone number to call to listen to the voicemail ? |
| 09:26.26 | fourcheeze | Assid: http://forums.digium.com/viewtopic.php?p=4820& |
| 09:26.39 | fourcheeze | second one |
| 09:27.01 | fourcheeze | phearless: whatever you make it |
| 09:27.18 | phearless | i can't find this number in the config files |
| 09:27.45 | Assid | fourcheeze: well.. dundi still wont know where the user logged in from to route the calls |
| 09:27.54 | phearless | I can see my messages in trixbox(asterisk recording interface) |
| 09:28.06 | phearless | but I can't find where is the phone number to listen to it |
| 09:28.38 | fourcheeze | phearless: I have this: |
| 09:28.41 | fourcheeze | exten => 8500,1,VoicemailMain(${CALLERIDNUM}|s) |
| 09:28.56 | Assid | actually.. i added the @context as well |
| 09:29.17 | Assid | since i got calls for 1 semi office within the main office |
| 09:29.36 | fourcheeze | Assid: I think dundi can be updated in real time, but I've not looked too hard at it |
| 09:29.43 | Assid | hrmm |
| 09:29.54 | Assid | will check it in a bit.. im just too zonked to think .. |
| 09:29.58 | Assid | was up till 5 or so |
| 09:30.31 | fourcheeze | Assid: what's your time now? |
| 09:30.31 | Assid | brb |
| 09:30.36 | Assid | 3.00 pm |
| 09:31.43 | phearless | 200 => 123,alex,,,attach=no|saycid=yes|envelope=no|delete=no |
| 09:31.45 | phearless | in /etc/asterisk/voicemail.conf |
| 09:32.38 | phearless | fourcheeze: fro mwhich file does come from your line ? |
| 09:34.06 | *** join/#asterisk yojanl (n=yoja@212.78.183.150) |
| 09:35.32 | yojanl | I have a rather complicated problem I think, in short: I cannot call out with a grandstream gxp-2000, I think its an issue with the grandstream software and the asterisk version because I can call out with a sipsoftware in this network. But I cannot call out with the grandstream... |
| 09:36.19 | yojanl | incoming calls are working fine, only outgoing. Ive tried to understand the sip debug output and it looks like only invites are coming in, but i dont see an ack |
| 09:36.34 | phearless | fourcheeze ? |
| 09:36.49 | yojanl | an other grandstream, with 1.0.1.9 software workes fine from an other network (havent tested it in this network) |
| 09:36.54 | fourcheeze | phearless: extensions.conf |
| 09:37.17 | fourcheeze | phearless: everything you dial goes in extensions.conf (or the realtime equivalent) |
| 09:38.13 | phearless | /etc/asterisk/applications.conf:exten => ${APP-MESSAGECENTER},5,VoiceMailMain(${VMCONTEXT}) |
| 09:38.13 | phearless | /etc/asterisk/applications.conf:exten => _${APP-MESSAGECENTER}X.,4,VoiceMailMain(${EXTEN:3}@${VMCONTEXT}) |
| 09:38.13 | phearless | /etc/asterisk/applications.conf:exten => ${APP-MESSAGECENTER-DIRECT},5,VoicemailMain(${CALLERIDNUM}@${VMCONTEXT}) |
| 09:38.13 | phearless | /etc/asterisk/extensions.conf:exten => a,n,VoiceMailMain(${ARG1}@${VMCONTEXT}) |
| 09:38.23 | phearless | I got that stuff in my extensions |
| 09:38.28 | phearless | something is missing ? |
| 09:38.50 | phearless | I don't see the phone number for the voicemail in this, so something should be missing |
| 09:40.22 | *** join/#asterisk Bambr (n=Bambr@213-35-236-196-dsl.end.estpak.ee) |
| 09:40.24 | pif | yojanl : with the gxp you signed-up for a lot of pain |
| 09:41.14 | yojanl | pif: the other gxp I had worked fine, but now... |
| 09:41.21 | *** join/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com) |
| 09:41.30 | yojanl | Im getting creazy! |
| 09:41.35 | pif | some work, other don't |
| 09:42.01 | pif | they're like women: moody |
| 09:43.35 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 09:44.30 | *** join/#asterisk prh (n=paul@X80.mjr.org) |
| 09:44.32 | yojanl | haha, but I think it can be solved |
| 09:44.40 | yojanl | i have 2 here, and they both do the same |
| 09:44.56 | yojanl | somehow this version of the software f*cks something up with the request |
| 09:45.01 | yojanl | incoming is working smooth |
| 09:45.22 | yojanl | where could I go to solve this? does grandstream have tech support? |
| 09:47.37 | phearless | I just added the module Voicemail in http://asterisk/admin/config.php?type=tool&display=modules |
| 09:47.56 | fourcheeze | phearless: where did all that stuff in extensions.conf comefrom? |
| 09:48.13 | phearless | i do not know |
| 09:48.19 | phearless | from trixbox.. |
| 09:48.22 | fourcheeze | ahhh |
| 09:48.34 | fourcheeze | I think you want #freepbx then |
| 09:48.44 | fourcheeze | see the topic |
| 09:49.04 | phearless | yes i know |
| 09:49.16 | phearless | but the config of the voicemail is in asterisk |
| 09:49.17 | fourcheeze | if you don't want to be using trixbox (and why would you) then just start off with a blank extensions.conf |
| 09:49.24 | phearless | freepbx is just an interface |
| 09:49.34 | phearless | okay fourcheeze |
| 09:49.37 | fourcheeze | it's an interface that puts a load of stuff in your extensions.conf |
| 09:49.45 | phearless | i will probbly do this when I will understand better asterisk |
| 09:49.51 | phearless | probably* |
| 09:49.53 | hads|home | No, it's an interface... Yeah, what he said. |
| 09:50.25 | fourcheeze | hads|home: so trixbox doesn't need anything in your extensions.conf to work? |
| 09:50.46 | fourcheeze | phearless: it's a steeper curve to go for plain old asterisk but you'll get there quicker :-) |
| 09:51.09 | hads|home | No, it does, I was halfway through saying what you said. But I'm slow tonite :) |
| 09:51.19 | fourcheeze | ahhh |
| 09:51.36 | hads|home | mmm.. beer |
| 09:52.57 | fourcheeze | yeah, beer will speed you up ;-) |
| 09:53.28 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
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| 09:56.15 | *** join/#asterisk kanelbullar (n=pas@83.240.200.92) |
| 09:56.37 | kanelbullar | Hello all |
| 09:57.18 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 09:57.29 | kanelbullar | Does anyone know if e&m signalling supports ANI? |
| 10:00.37 | kanelbullar | zapata.conf: siginalling=em |
| 10:01.00 | kanelbullar | zaptel.conf: e&m=1-24 |
| 10:05.38 | phearless | anybody got some fun cisco 7960 logos ? |
| 10:12.36 | nibbler_de | if i don't want to "Answer" a call. what was the other command again? |
| 10:21.13 | *** join/#asterisk erwinism (n=erwinpog@61.9.118.37) |
| 10:21.36 | erwinism | hello, i have cisco 7960, can anyone help me connect this phone to my asterisk box? |
| 10:21.44 | erwinism | it is brand new |
| 10:23.16 | *** join/#asterisk Tommmo (n=tps@203.62.181.52) |
| 10:23.24 | Rawplayer | erwinism: why? |
| 10:23.38 | Tommmo | for RealTime is it necessary to define each realtime context in extensions.conf? |
| 10:23.41 | Rawplayer | just add a client |
| 10:23.52 | erwinism | i dont have idea how to connect this thing. its not on its manual |
| 10:24.32 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
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| 10:32.20 | shadebob | hi. I have a little problem with the transfer of an agent call. When I tranfer the call, I have only silence |
| 10:33.29 | *** part/#asterisk sergee (n=opera@195.94.224.197) |
| 10:35.30 | *** join/#asterisk danast (n=abc@c-68-61-93-107.hsd1.mi.comcast.net) |
| 10:35.44 | tinyviolin | shadebob: at least it's easier to close the sale that way, without any objections |
| 10:40.49 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 10:41.30 | kuku5 | erwinism: you need the sip image |
| 10:41.43 | kuku5 | voip-info.org |
| 10:41.49 | kuku5 | or you can use sccp i guess |
| 10:42.21 | erwinism | kuku5: ok i will search on that |
| 10:42.24 | phearless | okay *98 works for the voicemail now ... COOL ! |
| 10:48.13 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 10:48.27 | mut | mornin |
| 10:48.29 | Winkie | is it just me or is chan_agent the most horrible thing ever when using the manager interface? |
| 10:48.30 | Winkie | mornin |
| 10:48.33 | Winkie | i've been up all night |
| 10:48.45 | Winkie | and now i intend to hunt down whoever is responsible for zombie and masq channels and kill them |
| 10:49.14 | mut | heh *shrug* |
| 10:49.22 | Winkie | if you are in this channel and are responsible please pm me home address and list of greatest fears thx |
| 10:50.08 | tinyviolin | Winkie: wait 'til you got 40 agents on and it deadlocks and you're getting screamed at |
| 10:50.25 | Winkie | tinyviolin: i'm putting way more than 40 onto it |
| 10:50.33 | Winkie | however we will have live backup systems |
| 10:50.41 | Winkie | i'm talking purely from a tracking point of view |
| 10:50.43 | tinyviolin | how many? |
| 10:50.48 | Winkie | 140 or so? |
| 10:50.51 | erwinism | kuku5: thanks for the help |
| 10:50.52 | Winkie | not on a single box like |
| 10:50.53 | tinyviolin | one box? |
| 10:50.55 | tinyviolin | oh |
| 10:50.58 | Winkie | no, 3 most likely |
| 10:51.04 | Winkie | 1x40 2x50 |
| 10:51.20 | Winkie | have you ever read the stuff it spews out on the manager interface? |
| 10:51.36 | tinyviolin | yeah, liberal use of UserEvent can help make it a little more sane |
| 10:51.48 | tinyviolin | then you can disable a bunch of the perms and have way less crap to look at |
| 10:51.54 | Winkie | irrelevant in our situation unfortunately |
| 10:52.10 | tinyviolin | are you using cmd Queue? |
| 10:52.13 | Winkie | our calling plan is extremely simple, our agents do most of the work with transfers and internal bounces |
| 10:52.17 | Winkie | yes |
| 10:53.01 | Winkie | i don't have any problem with the queue portion, but when it tries to connect to an agent it pseudo bridges it using some horrific Local channel hackery it seems |
| 10:53.38 | Winkie | it seriously needs to be rewritten and if it does deadlock on me people will die |
| 10:53.42 | tinyviolin | i use Bridge .. it's simpler, but you have the do the metrics yourself |
| 10:53.52 | tinyviolin | but you have a lot more freedom |
| 10:54.12 | Winkie | indeed, i've already written one patch to add in an AgentAssociate event which allows you to link the inbound channel to the agent channel |
| 10:55.24 | tinyviolin | cool |
| 10:55.49 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-69ae571ed7daed97) |
| 11:01.50 | hads|home | I don't use it so this may be bogus, but aparently there is known trouble with queues and it is better in trunk. |
| 11:02.31 | hads|home | erm... queues/chan_agent something. As I said, I don't use it. |
| 11:02.32 | tinyviolin | well yeah, i shouldn't bitch, because the guys on our team that still use agents/queues haven't had the deadlocks in a while |
| 11:02.52 | *** join/#asterisk vlt (n=dm@p54B345FA.dip0.t-ipconnect.de) |
| 11:03.53 | Winkie | well hopefully i'll be writing a paper on this implimentation i'm doing, pretty much 99% linux, offices in 3 countries with over 200 asterisk managed callcentre workers |
| 11:03.56 | Winkie | going to be a while though |
| 11:04.57 | tinyviolin | you know, i just looked and app_queue is 4000 lines of code |
| 11:04.59 | tinyviolin | why? |
| 11:05.13 | Winkie | yeah that's what i can't figure out, check out chan_agent |
| 11:06.20 | hads|home | 2500 |
| 11:09.53 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 11:11.46 | *** join/#asterisk Rawplayer (n=kevin@braadharing.oom-killer.org) |
| 11:11.58 | Rawplayer | does anyone of you guys have a snom 300 voip telephone? |
| 11:12.25 | Rawplayer | when i pick up the horn it makes a brummy noise |
| 11:14.28 | *** join/#asterisk McLazarus (n=mcallist@72.78.37.124) |
| 11:25.14 | Tommmo | for RealTime is it necessary to define each realtime context in extensions.conf? |
| 11:25.35 | Tommmo | i find that it only works when the config file has been updated to have the realtime context defined |
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| 11:45.30 | ManxPower | *sigh* NPR is doing another story on New Orleans |
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| 11:52.47 | phearless | hwo can i switch my message status between busy, unavailable,temporary ? |
| 11:55.33 | ManxPower | phearless, for the most part you don't switch between busy and unavail in the system. |
| 11:55.38 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 11:55.43 | *** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:55.51 | ManxPower | Asterisk plays the message type based on the u or b prefix to the call to voicemail |
| 11:56.36 | phearless | i do not know waht is u and b |
| 12:00.48 | ManxPower | phearless, Voicemail(b1234) |
| 12:01.00 | ManxPower | in extensions.conf |
| 12:01.11 | phearless | ok i will have a look |
| 12:01.26 | ManxPower | Unless you are using FreePBX/AMP/Asterisk@Home/Trixbox. If you are using one of those, I do not know how you set the option. |
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| 12:17.13 | danast | b stands for busy and u stands for unavailable playbacks |
| 12:18.39 | [TK]D-Fender | "show application voicemail" |
| 12:19.40 | danast | I am using asterisk realtime voicemail, I cannot get app_directory to work and cannot change vm password from the phone. \n I am using postgresql |
| 12:19.59 | danast | any suggestions |
| 12:22.33 | Ahrimanes | anyone having trouble with nested macros ? |
| 12:23.24 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 12:25.22 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 12:25.25 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 12:30.07 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 12:32.00 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.74) |
| 12:34.20 | Tommmo | does the number of contexts defined in extensions.conf significantly impact the performance of asterisk? |
| 12:34.30 | Tommmo | even if most contexts have nothing in them? |
| 12:34.34 | Tommmo | (or very little) |
| 12:35.42 | ManxPower | Tommmo, it should not |
| 12:36.14 | Tommmo | thanks |
| 12:36.41 | ManxPower | you should, of course test that. |
| 12:36.54 | Tommmo | will try |
| 12:37.52 | Assid | hahahahaha |
| 12:37.52 | Assid | <PROTECTED> |
| 12:38.04 | neo | asterisk ? |
| 12:38.27 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 12:38.47 | Assid | nope.. email |
| 12:39.11 | neo | spamd ? |
| 12:39.18 | [TK]D-Fender | I can't imagine there being a reason it should place a load of any significane on a system.... (Aside from a WRT maybe in a worst case scenario) |
| 12:39.26 | [TK]D-Fender | (if even) |
| 12:39.41 | ManxPower | 'morning [TK]D-Fender |
| 12:39.51 | Tommmo | because it appears i can't create contexts on the fly for Realtime |
| 12:39.56 | Tommmo | it looks like they have to exist first. |
| 12:40.25 | Assid | spamd .. can do that to you |
| 12:40.36 | neo | oh yeah |
| 12:40.42 | *** join/#asterisk jaike (n=a@203.115.188.120) |
| 12:40.46 | neo | can kill your box so easily :) |
| 12:41.21 | *** join/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 12:41.25 | jaike | anyone experiencng mixmonitor problems with 1.2.10? one-way audio or one side delayed |
| 12:44.01 | *** join/#asterisk stp1800 (n=tmoore@216.64.53.141) |
| 12:44.06 | [TK]D-Fender | ManxPower: *yawn* mornin' |
| 12:44.32 | [TK]D-Fender | Tommmo: That is something different.... |
| 12:44.52 | Tommmo | [TK]D-Fender: is there another way around it? |
| 12:45.04 | Tommmo | because it seems each context for realtime has to be put into extensions.conf |
| 12:45.04 | Tommmo | e |
| 12:45.21 | Tommmo | eg [CONV001-2] |
| 12:45.22 | Tommmo | switch => Realtime |
| 12:48.42 | fourcheeze | Assid: what's your MTA - exim has an option to stop receiving mail when load is high |
| 12:49.23 | Assid | fourcheeze: qmail |
| 12:49.34 | fourcheeze | ahh |
| 12:49.44 | Assid | its averaging around 12-14 right now |
| 12:49.54 | Assid | mostly because of clamd |
| 12:49.58 | fourcheeze | yeah |
| 12:50.02 | fourcheeze | BTDT |
| 12:50.05 | ManxPower | The problem with qmail is that I'm basically lazy. |
| 12:50.16 | fourcheeze | the problem with it is that it's so non-standard |
| 12:50.17 | Assid | dtbt? |
| 12:50.21 | Assid | btdt ? |
| 12:50.26 | fourcheeze | been there done that |
| 12:50.39 | ManxPower | qmail's license does not allow distros to include the binary, which means I have to build it myself and, as I said, I'm lazy. |
| 12:50.48 | fourcheeze | ahh yeah |
| 12:51.05 | fourcheeze | it also puts itself in stupid places IIRC |
| 12:51.16 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 12:51.17 | fourcheeze | my first ever MTA was qmail - then I did apt-get install exim and saw the light |
| 12:51.36 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 12:52.09 | Assid | nah.. im happy wit the likes of qmail.. nice speed.. |
| 12:52.56 | ManxPower | d |
| 12:58.59 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 13:00.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:02.24 | ionix | qmail is so gay |
| 13:02.27 | ionix | Exim is better |
| 13:02.35 | ionix | it installs in like 20 dirs |
| 13:03.35 | mut | o_O |
| 13:06.58 | mut | Assid... how much mail is that thing processing? |
| 13:06.58 | Nivex | postfix > exim |
| 13:06.58 | mut | or is it just a crap server? |
| 13:06.58 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 13:06.59 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 13:07.28 | creativx | hm |
| 13:07.42 | creativx | the quickes way to grep for the queuememberstatus "status" codes? |
| 13:07.58 | Assid | maybe around 10 mails per second.. including spam |
| 13:08.49 | Assid | running clamd/spamd etc on it |
| 13:08.56 | fourcheeze | postfix is good, but exim is a bit friendlier |
| 13:09.05 | blitzrage | pffft... sendmail! |
| 13:09.06 | fourcheeze | exim has some good hooks in it for spam/virus etc |
| 13:09.15 | fourcheeze | ha |
| 13:09.43 | fourcheeze | Any irc channel can turn into a MTA war |
| 13:09.57 | quid246 | MTA? |
| 13:10.42 | quid246 | I thought MTA wars only happened on the NCY subway |
| 13:10.42 | quid246 | NYC |
| 13:10.42 | fourcheeze | NYC? |
| 13:10.43 | fourcheeze | http://www.google.co.uk/search?q=define%3A+MTA&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-GB:unofficial |
| 13:10.54 | Assid | koay i think that box doesnt wanna respond anymore |
| 13:11.06 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.14 | Assid | uhho... __alloc_pages: 0-order allocation failed (gfp=0xf0/0) |
| 13:12.12 | Assid | __alloc_pages: 0-order allocation failed (gfp=0x1d2/0) |
| 13:12.13 | Assid | VM: killing process perl5.8.4 |
| 13:12.18 | Assid | thats not good is it? |
| 13:12.59 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:15.38 | fourcheeze | well it's trying to help |
| 13:15.56 | fourcheeze | if you won't kill processes yourself ;-) |
| 13:17.59 | Assid | need more ram? |
| 13:21.28 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 13:22.01 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:27.24 | fourcheeze | Assid: will it let you type "free" right now? |
| 13:28.05 | hmmhesays | sevard |
| 13:28.16 | hmmhesays | I need someone to test this calling card app |
| 13:28.19 | fourcheeze | Assid: but generally more ram would be good |
| 13:29.21 | *** join/#asterisk paryl (n=chatzill@www.admiralexpress.com) |
| 13:29.52 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 13:29.54 | SpaceBass | hey folks |
| 13:29.57 | Assid | yeah.. tons of caching and buffering.. as usual |
| 13:30.01 | SpaceBass | got a strange one-way audio issue |
| 13:30.02 | yojanl | anybody has great knowledge about grandstream gxp-2000's? I cannot make an outgoing call, very strange, response doesnt come back to the phone but in this network everything is okay. From a software client I can make outgoing calls so I guess its the software in the GXP |
| 13:30.34 | SpaceBass | I have 2 wifi phones on a subnet that is seperated by nat/firewall... when I call to the phone, audio works fine, when I call from it I get no audio in either direction....firewall is setup to allow all traffic |
| 13:30.46 | SpaceBass | but here is what is strange, when I turn on rtp debugging, it works perfectly |
| 13:30.58 | hmmhesays | SpaceBass: |
| 13:31.07 | paryl | i'm having issues with agents... now and then one will try to log on, and it will get stuck... for lack of better words. it sits there and nothing happens... so they naturally try to login again and again... and finally i have to restart asterisk altogether because there are 10 login tries just sitting there in the system. is it possible to kill a specific call that's taking place from the... |
| 13:31.08 | SpaceBass | hey hmmhesays |
| 13:31.08 | paryl | ...asterisk CLI? |
| 13:31.31 | hmmhesays | SpaceBass: you want to test this calling card app for me? |
| 13:32.24 | SpaceBass | hmmhesays, why not? |
| 13:32.42 | hmmhesays | http://www.thelostpacket.org/tricks.php |
| 13:33.23 | SpaceBass | hmmhesays, give me a few |
| 13:33.28 | SpaceBass | and I'll let you know |
| 13:33.50 | hmmhesays | testing your own stuff never works out well |
| 13:34.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:35.30 | cbrake | I have a Linksys SPA-941 and a cheap analog phone connected to a TDM400-FXS port calling out over NuFone. People I call say the Linksys phone sounds garbled and the analog phone sounds fine. I can't tell any difference on the receiving end. Any ideas why the linksys phone does not sound as good on the receiving end? |
| 13:36.10 | hmmhesays | um call yourself and test it |
| 13:38.01 | cbrake | hmmhesays: yeah, I've done that and the Linksys phone does sound worse. Trying to figure out why, or what I can do to fix it. Review suggest the SPA941 has decent voice quality ??? |
| 13:39.02 | cbrake | hmmhesays: but the sound coming into the Linksys phone sounds fine. Just seems to be an issue with sound going out of the linksys phone. |
| 13:39.16 | hmmhesays | vad enabled on the linksys? |
| 13:39.37 | paryl | please help guys... is there a way to kill a call in progress? |
| 13:39.44 | hmmhesays | ctrl-c |
| 13:39.59 | benjk | yeah, hangup |
| 13:40.24 | paryl | but i have, and they still show up in 'show channels verbose' |
| 13:40.39 | benjk | zombies |
| 13:40.43 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:40.49 | paryl | how do i get rid of them? |
| 13:40.55 | benjk | wait |
| 13:41.03 | cbrake | hmmhesays: what is vad? |
| 13:41.12 | paryl | dude, i am waiting... but they are staying around |
| 13:41.14 | hmmhesays | vad or silence suppression |
| 13:42.29 | paryl | i have one 'zombie' that's been active 8 minutes now |
| 13:42.48 | paryl | and no other calls can get in because of it |
| 13:42.58 | benjk | is this on a zaptel interface? |
| 13:43.16 | paryl | actually... it agents who are logging in |
| 13:43.32 | cbrake | hmmhesays: silence suppression is turned off |
| 13:43.41 | hmmhesays | what voice codec? |
| 13:44.15 | cbrake | hmmhesays: G711u from the phone to asterisk |
| 13:44.25 | cbrake | hmmhesays: and GSM from asterisk to nufone |
| 13:44.32 | hmmhesays | check your packet time |
| 13:44.52 | cbrake | hmmhesays: can this be done w/ ping? |
| 13:44.58 | hmmhesays | no |
| 13:45.13 | benjk | the queue mgt system has a wind up parameter to specify the time an agent needs to clean up his stuff after each call |
| 13:45.46 | paryl | benjk... is that in response to me? |
| 13:45.52 | benjk | yeah |
| 13:46.03 | hmmhesays | cbrake: is is the amount of voice that goes into each packet |
| 13:46.26 | benjk | read up on configuration for agents and queues |
| 13:46.53 | mut | god i hate to tell my bosses i told ya so |
| 13:47.07 | mut | but they grounded all of our copper we laid in this town wrong |
| 13:47.32 | paryl | yes... i have. this isn't what you're thinking. like i said above... it works great most of the time, but now and then the login process stalls, for lack of better words. |
| 13:47.37 | paryl | creating what you called a zombie |
| 13:47.46 | benjk | should have listened to you and roll out fiber instead of copper :) |
| 13:47.51 | paryl | then every login/logout attempt after that simply stalls too |
| 13:48.02 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 13:48.12 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 13:48.12 | paryl | i just upgraded to 1.2.10 last night, and it's happened twice this morning |
| 13:48.27 | benjk | did it work before? |
| 13:48.38 | mut | heh yea benjk |
| 13:48.47 | mut | we shoulda |
| 13:48.59 | benjk | copper prices are going up too |
| 13:49.05 | benjk | some strike in Chile |
| 13:49.11 | cbrake | hmmhesays: is packet time part of the SPA941 or asterisk config or both? |
| 13:49.22 | mut | it would just be the most hi tech city in northern michigan or something |
| 13:49.34 | mut | fiber to home before most of the rest of the state gets it |
| 13:49.43 | mut | in a rural town |
| 13:49.56 | mut | be good publicity anyway |
| 13:50.04 | benjk | well, you better get used to fiber because more and more countries rolling out fiber, its going to become not-quite-so-hi-tech-anymore |
| 13:50.17 | mut | heh |
| 13:50.24 | mut | we're JUST getting dsl here |
| 13:50.27 | mut | and it's us providing it |
| 13:50.32 | mut | not any telco |
| 13:50.42 | *** join/#asterisk yogurt2ungue (n=charlie@200.69.250.91) |
| 13:51.00 | benjk | cool |
| 13:51.18 | hmmhesays | cbrake: compare the two |
| 13:51.22 | mut | from what i hear now verizon wants to start doing dsl tho |
| 13:51.27 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 13:51.28 | mut | but that'll be 2 years off atleast |
| 13:51.29 | paryl | has anyone else seen the issue i described above? please? |
| 13:51.47 | benjk | did it work before 1.2.10 |
| 13:52.05 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:52.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:52.41 | paryl | it works all the time, yes... this is just a periodic issue now and then. it's just that after upgrading to 1.2.10, it's happened twice in the last hour |
| 13:53.07 | paryl | i never could track it down, but now because of the timing, i have to |
| 13:53.10 | benjk | then go back to the version you were using before |
| 13:53.24 | paryl | which had IAX2 and vm issues |
| 13:53.56 | benjk | higher version numbers are not going to cure your penis envy, in case you suffer from that |
| 13:54.19 | paryl | god, are you the channel troll or what? dude, i'm looking for answers |
| 13:54.37 | paryl | i said the previous version had major issues i needed fixed |
| 13:54.43 | paryl | now this is the issuee |
| 13:54.49 | benjk | people should be using the software that works for them, not just because the version numbers are fancy |
| 13:55.27 | benjk | no you said that your problems have become worse |
| 13:55.28 | blitzrage | snakes on a plane! |
| 13:55.49 | blitzrage | hehehe |
| 13:57.23 | tzanger | fucking bluetooth |
| 13:57.29 | tzanger | wtf is up with that snakes on a plane hting |
| 13:57.30 | tzanger | I don't get it |
| 13:58.17 | [TK]D-Fender | tzanger: Its a new movie with Samuel L Jackson. |
| 13:58.41 | Dr-Linux|work | tzanger looks angry! |
| 13:58.47 | [TK]D-Fender | tzanger: It was a "working title first" and then when they wanted to change it, SLJ said NO WAY! So it stuck, and its kinda catchy :) |
| 13:58.50 | hmmhesays | snakes on a plane muthafuckah |
| 13:59.05 | [TK]D-Fender | tzanger: Thats also literally the plot line. |
| 13:59.11 | file | moo |
| 13:59.22 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 14:01.28 | mut | touch file |
| 14:01.35 | hmmhesays | whoa... |
| 14:01.40 | file | eep |
| 14:02.13 | mut | -_- |
| 14:02.16 | Rawplayer | -,- |
| 14:02.29 | cbrake | hmmhesays: SPA941 RTP Packet size = 0.03s (is this "packet time") ? |
| 14:02.31 | *** join/#asterisk ToyMan (n=stuq@ool-44c7b88e.dyn.optonline.net) |
| 14:02.34 | tzanger | [TK]D-Fender: ahhh okay |
| 14:02.45 | tzanger | I thought it was maybe a web 2.0 thing |
| 14:02.46 | hmmhesays | cbrake yup |
| 14:02.50 | hmmhesays | 30 milliseconds |
| 14:03.17 | e-ddie | i use web 3.1 |
| 14:03.34 | cbrake | hmmhesays: ok thx, now to find the nufone size ... |
| 14:03.43 | hmmhesays | compare them between your two phones |
| 14:03.53 | hmmhesays | nufone size doesn't matter |
| 14:04.05 | creativx | i use web 3.11 for workgroups |
| 14:04.09 | creativx | its AJAX2 powered |
| 14:04.16 | hmmhesays | if your linksys is sending out larger packets, they might be exceeding the mtu size on your modem |
| 14:04.22 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 14:04.23 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:04.43 | Dr-Linux|work | who owns nufone? |
| 14:04.47 | [TK]D-Fender | tzanger: In there he supposedly says quite directly "I wan these MOF snakes off the MOFO plane!" |
| 14:04.58 | hmmhesays | bwhaha |
| 14:05.02 | [TK]D-Fender | tzanger: Because he NEVER speaks inappropriately ;) |
| 14:05.25 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:07.02 | blitzrage | I can't believe you guys are still talking about snakes on a plane :) |
| 14:07.18 | mut | i can;t believe they remade it |
| 14:07.24 | mut | :'( |
| 14:07.32 | tzanger | Dr-Linux|work: jerjer |
| 14:07.38 | tzanger | [TK]D-Fender: indeed :-) |
| 14:07.42 | tzanger | ahem |
| 14:07.46 | tzanger | REMADE snakes on a plane? |
| 14:08.00 | hmmhesays | that was all the rage on totalfark for like a month |
| 14:08.30 | hmmhesays | so where do I post this calling card app so I can get some criticism on it |
| 14:08.33 | trelane_ | SNAKES ON A PLANE?!? |
| 14:08.47 | *** join/#asterisk sp0n9e_ (n=sp0n9e@phpurge.com) |
| 14:08.53 | trelane_ | Dr-Linux|work, Jermey McNamera |
| 14:09.44 | cbrake | hmmhesays: how do I determine the * -> Nufone packet size? Can't find any options in iax.conf and can't convince the console to tell me? |
| 14:09.48 | tzanger | god damn I hate Palm's insistence to lock you out of your own hardware |
| 14:09.52 | tzanger | it's almost as bad as Microsoft |
| 14:10.03 | blitzrage | aye |
| 14:10.14 | hmmhesays | cbrake: i meant compare between the two phones you are using |
| 14:10.26 | mut | yea |
| 14:10.28 | mut | from book |
| 14:10.43 | mut | i can't believe all the publicity such a retarded movie is getting |
| 14:10.57 | hmmhesays | hey, slj is in it |
| 14:11.02 | mut | well yea |
| 14:11.07 | mut | that'll make it a winner right there eh |
| 14:11.19 | hmmhesays | the 51st state was fantastic |
| 14:11.22 | benjk | I hate Vodafone's insistence not to allow me to view any photos that haven't been taken with one of the phone's two cameras |
| 14:11.30 | benjk | and any other content for that matter |
| 14:11.43 | benjk | you can send yourself voicemail by email attachment for example |
| 14:11.57 | benjk | well, you can, but the phone will not play it, cause it isn't signed by Vodafone |
| 14:12.16 | yatesy | vodaphones branding really sucks |
| 14:12.39 | benjk | if it was just the branding I couldn't care less |
| 14:12.41 | yatesy | my dad ended up reflashing his phone back to manufacturer standard |
| 14:13.19 | benjk | I would like to know how to do that |
| 14:13.22 | quid246 | I hate Cellphones... how they lock everything you do, a *real* phone would let you upload ringtones/pictures/etc via USB or BlueTooth with no "service fees" |
| 14:13.35 | yatesy | i can do that on mine |
| 14:13.39 | benjk | I don't care about ringtones |
| 14:13.56 | benjk | but I want to be able to use voicemail by email attachments |
| 14:14.27 | tzanger | quid246: yep |
| 14:14.37 | benjk | its a Symbian based phone so I figured some Java guru could somehow remove the need for the signature |
| 14:14.38 | tzanger | this new Nokia 6265i I have apparently only supports one bluetooth device at a time |
| 14:14.47 | benjk | but every Java guru I asked said it wasn't possible |
| 14:15.05 | tzanger | if my headset is connected it does not participate in any other BT activities... i.e. I cannot use my Palm to dial through my phone if the headset is on |
| 14:15.16 | benjk | that sucks |
| 14:15.32 | Pj_ | polygamy rulez |
| 14:15.34 | benjk | my Nokia doesn't do that |
| 14:15.37 | tzanger | not to mention that Palm refuses to let me have full control over the init and dial strings, so CDMA phones don't work as a general rule |
| 14:15.57 | tzanger | but I can hack the palm library to use the correct BT profile (HFAG instead of DUN) |
| 14:16.01 | benjk | I can be on the phone via BT headset while my MacBook is using the phone for a data connection also over BT |
| 14:16.02 | hmmhesays | bah cleaning out my old code folder, I got a lot of half written crazy sh1at in here |
| 14:16.10 | tzanger | benjk: which phone is that? |
| 14:16.18 | tzanger | hmmhesays: :-) We all have those kinds of folders :-) |
| 14:16.30 | benjk | I am not sure, I think 6680, here in Japan they change the model numbers |
| 14:16.37 | tzanger | yeah same here in Canada |
| 14:16.41 | tzanger | it's CDMA though which is the problem |
| 14:16.52 | benjk | the Vodafone model number is Nk700MkII |
| 14:16.56 | tzanger | it's funny, it has a WCDMA SIM card socket but the manual says not to connect anything into it |
| 14:17.01 | benjk | or Nk702MkII |
| 14:17.28 | benjk | this is a multi-protocol multi-band phone |
| 14:17.47 | benjk | it does both GSM in four frequency bands and WCDMA |
| 14:18.16 | benjk | you need that over here cause Japan doesn't have any GSM whatsoever |
| 14:18.29 | cbrake | hmmhesays: other phone is analog TDM400 FXS channel. How do I determine the packet size when using this phone? |
| 14:18.51 | tzanger | benjk: NICE |
| 14:18.52 | cbrake | hmmhesays: I suppose I could also run some tests w/ xlite as well. |
| 14:18.57 | hmmhesays | good call |
| 14:19.02 | tzanger | I might have to send this phone back, this blows goats |
| 14:19.09 | tzanger | Motorola RAZR maybe |
| 14:19.09 | hmmhesays | also do an echo test, see if the problem shows up there |
| 14:19.14 | ionix | man, I hate coding forms |
| 14:19.23 | benjk | but as I said, becuase of Vodafone's greed I cannot listen to voicemail attachments because my Asterisk server cannot sign them with Vodafone's digital signature |
| 14:19.24 | hmmhesays | ionix: yep |
| 14:19.26 | ionix | form validation, error notification and all that shit |
| 14:19.38 | ionix | someone should make a killer class for forms |
| 14:19.58 | ionix | like we just specify the forms with preg match and validation tools and it does it automatically |
| 14:20.03 | tzanger | I hate form programming as well |
| 14:20.09 | tzanger | benjk: yuck. |
| 14:20.14 | benjk | yeah |
| 14:20.19 | ionix | doing a signup form is soo boring then you have to code a information update form after the signup |
| 14:20.26 | ionix | pukes |
| 14:20.30 | hmmhesays | dreamweaver is pretty nice for generating forms |
| 14:20.36 | hmmhesays | nvu is ok too |
| 14:20.40 | ionix | well, the form is easy |
| 14:20.43 | ionix | the form validation is shit |
| 14:20.49 | hmmhesays | forms with associated validation |
| 14:20.59 | benjk | also, OSX comes with a synchronisation utility which installs an agent on a Symbian phone by sending the Java applet to the phone by BT |
| 14:21.02 | ionix | in dreamweaver? |
| 14:21.10 | tzanger | benjk: nice |
| 14:21.16 | benjk | I can't install this applet because its not signed by Vodafone |
| 14:21.20 | ionix | ah wtf |
| 14:21.24 | ionix | I thought I was on ##php |
| 14:21.25 | ionix | sorry |
| 14:21.27 | benjk | and this means I cannot synchronise addresses and stuff |
| 14:21.29 | hmmhesays | ionix, LOL |
| 14:21.34 | hmmhesays | nice |
| 14:21.36 | ionix | damn, I am tired tonight |
| 14:21.41 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 14:21.41 | hmmhesays | where are you? |
| 14:21.44 | ionix | Japan |
| 14:21.44 | tzanger | I'm gonna see if I can fuck around with the phone with the nokia phone suite |
| 14:21.45 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 14:21.59 | benjk | When I tried to bring this to Vodafone's attention they said the phone doesn't support Mac connectivity |
| 14:22.00 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:22.01 | hmmhesays | I have a linksys wip-300 that is a phone tof ark with |
| 14:22.17 | benjk | I said, Apple does support it, you block their support |
| 14:22.24 | jtexter3 | I'm currently writing a module for asterisk. If I have two channels bridged together, what is the correct way to break that bridge so I can go on and do other things with the channels? |
| 14:22.27 | benjk | they don't get it |
| 14:22.37 | benjk | its not on the CD, so it can't work, they say |
| 14:22.50 | hmmhesays | I'm guessing look at the transfer app |
| 14:26.14 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:27.00 | jtexter3 | unfortunately, app_transfer just uses the underlying technology to connect to a separate channel, i.e. a sip redirect |
| 14:27.00 | fourcheeze | can anyone see the writing on the wall with the windows communications server 2007? |
| 14:27.16 | hmmhesays | ok so who wants to test this prepaid dialplan |
| 14:27.20 | hmmhesays | I need some input |
| 14:27.29 | jtexter3 | it looks like may ast_softhangup is what I want, but I can't tell for sure |
| 14:28.09 | brodiem | just curious, what causes the first part of the audio on a local extension to be cut off, i.e. if you are dialing an extension that just does a Playback(). Unless you put a Wait() beforehand the first .5 second or so of the audio is cut off. Is there a sip.conf parameter that will correct this? |
| 14:28.39 | hmmhesays | the first part of the audio for pretty much any extension will be cut off without a wait |
| 14:29.07 | brodiem | I've concluded that already :) |
| 14:30.25 | *** join/#asterisk w32 (n=w32@c-71-193-124-77.hsd1.il.comcast.net) |
| 14:30.38 | brodiem | I would just love to know why it cannot just wait until the call is setup before it starts running through |
| 14:31.32 | *** part/#asterisk jaike (n=a@203.115.188.120) |
| 14:31.37 | benjk | brodiem, jtodds silence recordings are your friend |
| 14:31.42 | *** join/#asterisk zedkatuf (n=audela@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 14:31.54 | benjk | www.loligo.com |
| 14:32.09 | benjk | look for sounds |
| 14:32.14 | benjk | directory silence |
| 14:32.40 | benjk | play 1 sec of silence before your prompt and you'll never experience a cutout |
| 14:33.20 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:33.22 | benjk | Wait() doesn't help, btw |
| 14:33.31 | brodiem | benjk why not just use Wait()? |
| 14:33.41 | benjk | well, it doesn't fix the cutout |
| 14:34.18 | brodiem | i was just curious to know if this happens with everyone and if there was some parameter I could set that would start audio after a channel is completely setup or something |
| 14:34.32 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 14:34.35 | brodiem | benjk it always works for me? |
| 14:34.36 | dyn | hi |
| 14:35.05 | dyn | faq: I have no "dial" command on the asterisk console.. which module am I missing? |
| 14:35.05 | benjk | never worked for me |
| 14:35.14 | benjk | app_dial.so |
| 14:35.15 | dyn | is it 'modem'? |
| 14:35.24 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.118.Dial1.SanJose1.Level3.net) |
| 14:35.28 | dyn | thanks benjk, checking |
| 14:35.42 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 14:36.02 | benjk | maybe your Linux box has a sense for tidyness |
| 14:36.08 | dyn | benjk: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_dial.so: undefined symbol: ast_bridge_call |
| 14:36.40 | benjk | app_dial is amongst the most horrible code on the planet |
| 14:36.47 | SpaceBass | hmmhesays, I'm still going to try and test for ya |
| 14:36.51 | MatsK | dyn: do you have a soundcard and oss or alsa configed |
| 14:36.51 | SpaceBass | my "day job" is getting in the way |
| 14:36.53 | *** part/#asterisk jailbreaker (n=teodory@mail.jetfinanceintl.com) |
| 14:36.56 | benjk | ast_bridge_call is in res_features |
| 14:36.58 | SpaceBass | not to mention my one-way audio issue |
| 14:37.07 | hmmhesays | SpaceBass: cool |
| 14:37.16 | hmmhesays | it is pretty simple |
| 14:38.04 | benjk | make sure you have res_features.so loaded |
| 14:38.14 | dyn | MatsK: I have but now as you say, something is not right with my alsa config (cannot even invoke alsamixer though I have a /dev/dsp0). ok, I'll check this, thank you |
| 14:38.20 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 14:38.42 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 14:38.46 | benjk | the error message you pasted has nothing to do with ALSA |
| 14:39.24 | benjk | you have res_features.so missing |
| 14:39.30 | dyn | okay |
| 14:40.34 | *** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) |
| 14:40.43 | sudhir492 | Hi All |
| 14:40.50 | sudhir492 | Anyone using PAP2-NA |
| 14:41.00 | SplasPood | Anyone ever had an issue where *1 on a Queue/Agent call will cause the call to hangup? I can duplicate it every time |
| 14:41.13 | benjk | its not an issue |
| 14:41.17 | benjk | its a feature |
| 14:41.33 | SplasPood | :P |
| 14:41.36 | hmmhesays | well my yum is completely farked up |
| 14:41.40 | benjk | show application dial |
| 14:41.44 | benjk | will tell ya |
| 14:41.59 | benjk | just don't pass the h and H flags |
| 14:42.00 | SplasPood | *1 starts on the fly recording |
| 14:42.07 | SplasPood | and those flags are not passed |
| 14:42.13 | SplasPood | this is an Agent call |
| 14:42.15 | SplasPood | in a queue.. |
| 14:42.32 | benjk | then there's a bug in app_dial or ast_bridge_call |
| 14:42.45 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:42.58 | benjk | because the hangup is caused by the code that is supposed to hangup when you press * if you pass those flags |
| 14:43.07 | SplasPood | hrm yea |
| 14:43.13 | SplasPood | lemme try remapping the feature |
| 14:43.15 | SplasPood | to something else |
| 14:43.22 | SplasPood | isn't it hangup on # tho? |
| 14:43.30 | benjk | app_dial is pretty horrific, its possible that some circumstances cause it to confuse the status |
| 14:43.40 | benjk | no, # is for transfer |
| 14:43.45 | SplasPood | nope |
| 14:43.46 | SplasPood | yea |
| 14:43.46 | SplasPood | * |
| 14:43.59 | benjk | * for hangup, # for transfer |
| 14:44.25 | macTijn | ~paste? |
| 14:44.26 | jbot | i heard paste is see http://paste.husk.org |
| 14:44.31 | macTijn | ~pastebin? |
| 14:44.32 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts |
| 14:44.34 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 14:44.56 | benjk | and with all them gotos and breaks in the app_dial, there are situations where it does something that it shouldn't do |
| 14:45.22 | macTijn | ~pastebin is also http://paste-it.net |
| 14:45.24 | jbot | macTijn: okay |
| 14:45.31 | macTijn | ~paste is also http://paste-it.net |
| 14:45.32 | jbot | okay, macTijn |
| 14:45.36 | macTijn | redundancy++ |
| 14:45.37 | macTijn | ;) |
| 14:46.37 | Rawplayer | ;) |
| 14:46.40 | benjk | SplasPood, you can always comment out the section of code in app_dial that handles the * DTMF, also in ast_bridge_call, then test again |
| 14:46.45 | zeedo | heh, the reply to ~pastebin floods more than mosts pates :-P |
| 14:46.52 | benjk | if the issue is gone, you know why |
| 14:47.54 | SplasPood | benjk: heh I *could* do that... thing is I want them to be able to hit *1 to record the call |
| 14:48.10 | benjk | I am not saying you should do it as a fix |
| 14:48.20 | SplasPood | ok, yea |
| 14:48.33 | benjk | I am saying you should do it as a test, to find out what the culprit is |
| 14:48.47 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 14:49.27 | SplasPood | hrm, how can features.conf be reloaded |
| 14:49.35 | SplasPood | (short of a full reload) |
| 14:50.20 | blitzrage | try "reload res_features.so" |
| 14:50.42 | blitzrage | then do a "show features" to see if your changes took |
| 14:50.49 | [TK]D-Fender | SplasPood: You're using AgentLogin for your Queu aren't you? |
| 14:51.19 | SplasPood | [TK]D-Fender: Yes |
| 14:51.45 | [TK]D-Fender | SplasPood: You're kinda screwed... * hangs up a call in that app.... |
| 14:51.55 | [TK]D-Fender | SplasPood: Pray you can remap that feature. |
| 14:52.16 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 14:52.32 | SplasPood | [TK]D-Fender: Where's that, as part of the Agent/ channel? |
| 14:52.39 | brodiem | i find its just easier to record all calls :) |
| 14:53.05 | SplasPood | brodiem: thats the path I'm going to take if necessary |
| 14:53.23 | SplasPood | [TK]D-Fender: show application AgentLogin doesn't mention it |
| 14:53.35 | SplasPood | oh |
| 14:53.35 | SplasPood | yes |
| 14:53.39 | SplasPood | it says 'the star key' |
| 14:55.53 | SplasPood | [TK]D-Fender: I'm actually using CallBackLogin |
| 14:55.55 | SplasPood | but same diff |
| 14:56.10 | SplasPood | doesn't appear to be a way, short of hacking the source, to disable it |
| 14:57.12 | benjk | that's what happens if you bolt on features without an overall design |
| 14:57.38 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:57.41 | benjk | sooner or later features will step on each others' toes |
| 14:57.47 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 14:58.25 | sudhir492 | Anyone here using PAP2-NA? |
| 14:58.57 | hmmhesays | paptuna |
| 14:59.22 | benjk | papertuna |
| 14:59.26 | benjk | papertiger |
| 14:59.33 | hmmhesays | spiceytuna |
| 14:59.34 | *** join/#asterisk _Guhit (n=amistry@am-productions.biz) |
| 14:59.34 | benjk | peppertuna |
| 14:59.38 | benjk | heh |
| 14:59.41 | hmmhesays | speaking of spicey tuna |
| 14:59.41 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.74) |
| 15:00.17 | hmmhesays | word to the wise, don't mess around with your girlfriend after eating hot wings and not washing your hands |
| 15:00.25 | SplasPood | [TK]D-Fender: where would this code be, chan_agent? |
| 15:00.31 | dyn | rotfl |
| 15:00.41 | hmmhesays | she WILL NOT be pleased |
| 15:00.48 | file | hmmhesays: or yourself |
| 15:00.48 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 15:00.56 | benjk | not washing your hands or not washing your mouth? |
| 15:00.57 | hmmhesays | file: lol |
| 15:01.22 | hmmhesays | benjk whatever part of you that you might be sticking down there that still has hot sauce on i |
| 15:01.22 | hmmhesays | t |
| 15:01.24 | benjk | You can always get a Korean girlfriend |
| 15:01.30 | SplasPood | [TK]D-Fender: nevermind, it is |
| 15:01.31 | benjk | they should be used to it |
| 15:01.34 | file | one of my friend's did that, he put a drop of something called "Satan's Blood" on his finger to taste it - went to the bathroom afterwards... |
| 15:01.34 | file | haha |
| 15:01.42 | hmmhesays | Korean girls like burning vagina's? |
| 15:01.53 | hmmhesays | file: LOL |
| 15:02.07 | hmmhesays | this happened to be "wild sauce" |
| 15:02.07 | benjk | not sure, but they eat this stuff called kimchi and other hot stuff all the time |
| 15:02.12 | *** part/#asterisk yogurt2ungue (n=charlie@200.69.250.91) |
| 15:02.30 | *** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au) |
| 15:02.38 | malverian | Anyone else getting frequent crashes with Asterisk 1.2.10? |
| 15:02.51 | hmmhesays | my 1.2.5 box crashed today |
| 15:02.51 | malverian | Seems to be a bug in channel.c that was introduced since 1.2.7 |
| 15:03.19 | malverian | I get a bunch of messages about "avoided initial deadlock" and then a few seconds later it crashes. |
| 15:03.24 | hmmhesays | anyone ever have the problem where you connect and you get no command line? |
| 15:03.36 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:03.37 | hmmhesays | License version 2 and other licenses; you are welcome to redistribute it under |
| 15:03.37 | hmmhesays | certain conditions. Type 'show license' for details. |
| 15:03.38 | hmmhesays | ========================================================================= |
| 15:03.41 | hmmhesays | thats it |
| 15:04.08 | brodiem | i've never had a crash yet knock on wood |
| 15:04.10 | brodiem | root@pbx:~# uptime |
| 15:04.10 | brodiem | <PROTECTED> |
| 15:04.18 | *** join/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil) |
| 15:04.21 | malverian | brodiem, I mean Asterisk crash.. not the actual box. |
| 15:04.28 | malverian | brodiem, asterisk -r -x "show uptime" |
| 15:04.33 | sp0n9e | damn there's a lot of warnings in the wanpipe driver. |
| 15:05.09 | brodiem | malverian, nothing impressive about mine there :) I had to restart when I added a PRI a couple of days ago |
| 15:05.34 | hmmhesays | CLI> show uptime |
| 15:05.34 | hmmhesays | System uptime: 49 seconds |
| 15:05.34 | hmmhesays | *CLI> |
| 15:06.05 | hmmhesays | I just restared |
| 15:06.12 | yojanl | hi, what should I fill in in the field "Use NAT IP" in the advanced setting of the gxp-2000? the public IP of our network, the ip of our asterisk server or the internal ip of the grandstream? |
| 15:06.23 | hmmhesays | nothing |
| 15:06.28 | hmmhesays | don't fill it in |
| 15:06.33 | hmmhesays | nat=yes in sip.conf |
| 15:07.16 | yojanl | okay, but I have a problem here with outgoing calls, I was hoping the Use Nat IP could help me solve the problem |
| 15:07.28 | hmmhesays | what problem? |
| 15:08.16 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:08.35 | yojanl | strange problem, outgoing calls time out, the grandstream gets no reaction from the server and the server cant get to the grandstream. First I thought it was something here in this network but Ive installed a sip soft client and i can call out with that sip soft client so my guess is its the grandstream... |
| 15:09.00 | yojanl | incoming cals work without problems... |
| 15:09.13 | sp0n9e | yojanl: do both of them have the same context? |
| 15:09.18 | hmmhesays | you have nat=yes in sip.conf? |
| 15:09.39 | yojanl | sp0nge: yes, I even used the same account for testing |
| 15:09.44 | yojanl | nat=yes is in sip.conf |
| 15:09.52 | SpaceBass | I have 2 wifi phones on a subnet that is seperated by nat/firewall... when I call to the phone, audio works fine, when I call from it I get no audio in either direction....firewall is setup to allow all traffic |
| 15:09.55 | SpaceBass | but here is what is strange, when I turn on rtp debugging, it works perfectly |
| 15:09.57 | hmmhesays | are you getting rtp back to the grandstream? |
| 15:10.01 | yojanl | its a very strange problem, im no beginner in asterisk but this is something ive never seen |
| 15:10.17 | yojanl | how can i check if the rtp gets back? |
| 15:10.21 | hmmhesays | SpaceBass: is it trying to reinvite? |
| 15:10.23 | yojanl | incoming works |
| 15:10.33 | SpaceBass | hmmhesays, good question...would I want it to, or not/ |
| 15:10.36 | hmmhesays | yojani, you also canreinvite=no |
| 15:10.43 | hmmhesays | not |
| 15:11.07 | hmmhesays | SpaceBass: call an echo test from the phone |
| 15:11.08 | yojanl | i think ive tried reinvite, but i will do it no just 2 be sure, ive changed almost all settings :-) |
| 15:11.08 | SpaceBass | hmmhesays, its set to no on both wifi phones |
| 15:11.21 | ManxPower | If there is nat involved then reinvites will NOT work. |
| 15:11.22 | hmmhesays | SpaceBass: call an echo test in asterisk |
| 15:11.37 | hmmhesays | ManxPower: no, they are unlikely to work |
| 15:11.47 | hmmhesays | you can get them to work with nat though |
| 15:12.05 | ManxPower | hmmhesays, the chances are so slow "will not" is a good term. |
| 15:12.31 | hmmhesays | for your average joe yeah |
| 15:12.35 | SpaceBass | i just cannot understand why enabeling rtp debug would fix it |
| 15:12.43 | hmmhesays | what version of asterisk? |
| 15:12.54 | yojanl | hmmhesays: reinvite no makes no problem |
| 15:12.56 | ManxPower | hmmhesays, have you ever gotten reinvites to work without doing portforwarding on the NAT routers? |
| 15:12.57 | SpaceBass | 1.2.9.1 |
| 15:13.07 | yojanl | Grandstream GXP2000 1.1.0.16 |
| 15:13.14 | hmmhesays | yojani: what? |
| 15:13.20 | yojanl | Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k |
| 15:13.30 | hmmhesays | yojani: makes no problem? |
| 15:13.42 | yojanl | hmmhesays: sorry, typing to quick. gives the same problem i meant |
| 15:13.50 | yojanl | times out again |
| 15:13.58 | hmmhesays | ManxPower: never tried |
| 15:14.07 | hmmhesays | when I use reinvites and nat I have control of the nat devices |
| 15:14.18 | ManxPower | *nod* |
| 15:14.40 | hmmhesays | yojanl: is the call even making it to asterisk? |
| 15:14.58 | yojanl | yes, the call is getting to asterisk |
| 15:15.12 | yojanl | i can send you the sip debug output if you think it helps |
| 15:15.19 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:15.23 | hmmhesays | put your your wan ip into the use nat field in the gxp |
| 15:15.30 | SpaceBass | hmmhesays, having problems with echo test currently |
| 15:15.40 | hmmhesays | problems specific to that phone? |
| 15:15.43 | _Guhit | I've got a problem with asterisk (asterisk 1.2.9.1, zaptel 1.0, FreeBSD) not hanging up with my X100P. After I modified the BATT_DEBOUNCE, I can see that asterisk does get a message like Hungup 'Zap/1-1' when the far end disconnects, even during the times that it does keep the line engaged. Any suggestions on how to force the card to hangup when it sees the "Hungup" message? |
| 15:15.45 | hmmhesays | or .. problems in general |
| 15:15.51 | SpaceBass | in general |
| 15:15.54 | hmmhesays | what? |
| 15:15.58 | hmmhesays | why? |
| 15:16.01 | SpaceBass | its not implemented correctly in trixbox I think |
| 15:16.18 | SpaceBass | got it working |
| 15:16.28 | hmmhesays | how is that possible? trixbox compiles asterisk from source |
| 15:16.47 | SpaceBass | yeah but the feature codes are odd |
| 15:16.49 | SpaceBass | regardless its working |
| 15:16.52 | SpaceBass | and I was able to hear myself |
| 15:16.55 | hmmhesays | what'd you do? |
| 15:16.57 | *** join/#asterisk anthm (n=anthm@000-446-609.area4.spcsdns.net) |
| 15:16.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:17.11 | *** join/#asterisk UlbabraB (n=filippo@host241-43-static.72-81-b.business.telecomitalia.it) |
| 15:18.00 | SpaceBass | i have canreinvite=no on the lan phone and the wifi (natted) phone |
| 15:20.03 | yojanl | hmmhesays: makes no difference :-( |
| 15:20.10 | yojanl | wan ip in the field |
| 15:20.18 | hmmhesays | so it is working now SpaceBass? |
| 15:20.26 | hmmhesays | yojani, odd |
| 15:21.06 | SpaceBass | hmmhesays, for the time being...but yeah |
| 15:21.50 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:21.56 | yojanl | hmmhesays: its very strange, i have a gxp in a differnet network which works fine, but that gxp has an old version of the firmware, i think its a firmware issue |
| 15:22.02 | hmmhesays | could be |
| 15:22.04 | hmmhesays | update it |
| 15:22.11 | hmmhesays | I got 1.2.9 running on this wrt here |
| 15:22.13 | ManxPower | I suspect the inconing call is not matching a sip.conf entry. |
| 15:22.27 | yojanl | the stupdig thing is, the old version works, this gxp has the newest version |
| 15:22.38 | hmmhesays | with a usb audio card running |
| 15:22.43 | ManxPower | The easy way to fix that is to put context=INVALID in sip.conf [general] then context=thecorrectcontext in each SIP device [section] |
| 15:22.47 | yojanl | and since i cannot downgrade to the old version im stuck |
| 15:22.52 | hmmhesays | i've turned my wrt into an ip phone bwhwhaa |
| 15:23.11 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 15:23.16 | ManxPower | I turned my WRT into a WAP |
| 15:23.21 | hmmhesays | haha |
| 15:23.30 | yojanl | the most important thing is incoming and that works, but it irritates my very very much that i cannot get outgoing to work, and outgoing normally is the simplest! |
| 15:23.45 | hmmhesays | currently i have my running asterisk with usb audio and a linksys wip-300 wifi phone registering to it |
| 15:23.48 | ManxPower | yojanl, paste the Dial command for outgoing |
| 15:24.17 | mut | is there a way to turn off events in the manager api? |
| 15:24.23 | mut | so i can send me commands and get results |
| 15:24.26 | mut | but not see everything else |
| 15:24.28 | ManxPower | Does "sip show peers" show the correct external IP address for the NATed phone? |
| 15:24.31 | SpaceBass | how is the wip-300? b/c the 330 sucks ass |
| 15:24.43 | yojanl | ManxPower: that works, if I use my other phone, or the local soft client on my laptop i can call outbound... the trouble is the grandstream doesnt receive the answer from the asterisk server |
| 15:24.51 | *** join/#asterisk naif (n=xyz@85-18-35-21.ip.fastwebnet.it) |
| 15:25.00 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 15:25.08 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 15:25.12 | ManxPower | yojanl, and "sip show peers" shows the correct IP address for the grandstream phone? |
| 15:25.35 | yojanl | yes, the right ip and right port |
| 15:25.59 | yojanl | cause incoming is working perfectly, so there can be a connection, outgoing is the headache |
| 15:26.34 | yojanl | context etc is correct, my guess is the grandstream does something new to the sip packages and in combination with this router or something the package doesnt get back |
| 15:26.42 | yojanl | i only see invites coming, no ack going back |
| 15:26.46 | hmmhesays | SpaceBass: i like the wip-300 what is wrong with the 330? |
| 15:27.08 | ManxPower | yojanl, incoming can work just fine even if there is no ip address listed in "sip show peers" for that device. |
| 15:27.15 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 15:27.22 | SpaceBass | hmmhesays, long laundry list....no stun which makes it worthless while traveling...inside my house (at least now) its working |
| 15:27.31 | SpaceBass | wpa supports only 27bit keys |
| 15:27.41 | yojanl | ManxPower, okay I didnt know that |
| 15:27.42 | SpaceBass | several of the buttons just do nothing.....theres lots of little bugs |
| 15:27.42 | ManxPower | yojanl, does your Asterisk have more than 1 ip address? |
| 15:28.04 | yojanl | ManXPower, I use normally one 1, but there are several ips on that server |
| 15:28.22 | ManxPower | yojanl, do you have a bindaddr or anything like that in sip.conf? |
| 15:28.27 | fourcheeze | yojanl: just use 1 ip number |
| 15:28.38 | yojanl | ive tried to work with the hostname, and the ip, now its set to the ip |
| 15:28.48 | ManxPower | remove the bindaddr. |
| 15:28.54 | yojanl | in sip says: bindaddr=0.0.0.0 |
| 15:29.02 | ManxPower | remove the bindaddr. |
| 15:29.07 | ManxPower | there is no need for it in your case |
| 15:29.09 | yojanl | okay, going to try now |
| 15:29.19 | fourcheeze | ManxPower: why remove it ? Why not set it to the IP number he's trying to access * with? |
| 15:29.51 | ManxPower | fourcheeze, beacuse bindaddr only applies to the SIP signaling packets. It has no effect on RTP, and that causes issues. |
| 15:29.59 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 15:30.00 | yojanl | manxpower: no difference |
| 15:30.19 | ManxPower | yojanl, I'm STILL waiting for you to paste the Dial command from Asterisk. |
| 15:30.22 | fourcheeze | yojanl: are you seeing any messages about stale nonces? |
| 15:30.54 | yojanl | ManxPower: you mean the Dial command in extenstions? |
| 15:31.03 | yojanl | fourcheeze: what is a stale nonce? |
| 15:31.03 | ManxPower | yojanl, correct. |
| 15:32.26 | ManxPower | Those look OK. |
| 15:32.39 | fourcheeze | yojanl: is the IP number you're talking to asterisk on the main IP number on your server? |
| 15:32.46 | fourcheeze | e.g. eth0 not eth0:4 |
| 15:32.59 | yojanl | fourcheeze: what is a stale nonce? I see Retransmitting #1 (NAT): |
| 15:33.06 | *** join/#asterisk Gregabyte (n=greg@gateway.digium.com) |
| 15:33.34 | fourcheeze | fourcheeze: you get a stale nonce when the client doesn't get the message to reregister |
| 15:33.52 | fourcheeze | i.e. the 401 |
| 15:33.56 | yojanl | fourcheeze: yes, its the main ip |
| 15:34.02 | sp0n9e | <PROTECTED> |
| 15:34.02 | ManxPower | yojanl, does it start working for a few mins when you reboot the natted phone? |
| 15:34.34 | *** join/#asterisk monkey13 (n=monkee13@69.7.217.155) |
| 15:34.42 | yojanl | ManxPower: it doesnt work all day, the phone is only a few days already and nothing, only incoming calls working like a charm |
| 15:34.42 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
| 15:34.50 | fourcheeze | hmm |
| 15:35.25 | }btorch{ | hey guys is it possible to create a conference room and while in conference call someone and make the person part of the conference |
| 15:35.50 | fourcheeze | yojanl: have you tried a stun server? |
| 15:36.04 | yojanl | yes, i think the stun.fwd.net or so |
| 15:36.11 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 15:36.14 | Bert- | hi again |
| 15:36.21 | hmmhesays | whoa bug in the cc app |
| 15:36.21 | fourcheeze | yojanl: have you tried removing it |
| 15:36.27 | yojanl | but ive tried so many things, ill try it again |
| 15:36.29 | yojanl | its now removed |
| 15:36.31 | fourcheeze | ok |
| 15:36.33 | quid246 | what kind of billing scales well.. alot of what I've read seems to speak against AGI? |
| 15:36.33 | fourcheeze | try it in then |
| 15:37.04 | Bert- | I have a little issue : with some phones, when we hangup, asterisk don't hangup with the other side, and the channel is still open |
| 15:37.23 | yojanl | is stun.fwd.net a valid one? and should i set the account to: NAT Traversal (STUN): yes? |
| 15:37.43 | Bert- | it hangs some time after, like 1 or 2 min |
| 15:37.44 | yojanl | stun.fwdnet.net is correct right? |
| 15:38.09 | benjk | }btorch{ create an extension that calls the conference room, then calls somebody and transfer them to that extension |
| 15:38.20 | benjk | correction, call somebody |
| 15:38.37 | hmmhesays | quid246: define scales? |
| 15:38.44 | benjk | fish scales |
| 15:39.13 | }btorch{ | benjk, yeah that's what i did |
| 15:39.15 | fourcheeze | yojanl: I've never used that one |
| 15:39.30 | }btorch{ | benjk, do I waste channels when doing that though ? |
| 15:39.38 | yojanl | i tried it now with that one, but i doesnt make any diffence |
| 15:39.38 | benjk | why? |
| 15:39.40 | }btorch{ | guess not if I hang up the call right |
| 15:39.52 | fourcheeze | yojanl: it seems to work |
| 15:40.16 | benjk | when you transfer the called party to the conference, they will be directly "connected" to the conference |
| 15:40.22 | quid246 | hmmmhesays: well... when you are running 200+ SIP/IAX channels with billing... won't be a serious drain on resources. |
| 15:40.33 | yojanl | I see something now, if I debug the sip ip a line saying: SIP/2.0 488 Not Acceptable Here is there |
| 15:40.39 | hmmhesays | what type of billing? |
| 15:40.47 | quid246 | prepaid |
| 15:40.52 | hmmhesays | using what app? |
| 15:41.04 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 15:41.23 | quid246 | That I am not certain about... I've looked at the current offerings, a2billing being the most "clean" looking app... but may end up just doing somethign myself |
| 15:41.36 | hmmhesays | I wrote a simple prepaid app with no agi |
| 15:42.00 | quid246 | yeah I remember you meantioning you used the DP |
| 15:42.05 | hmmhesays | very basic, but its got the basics a prepaid app needs |
| 15:42.06 | yojanl | so I guess that must be the trouble, asterisk or the phone says: |
| 15:42.06 | yojanl | SIP/2.0 488 Not Acceptable Here |
| 15:42.16 | hmmhesays | could be easily built upon |
| 15:42.21 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:42.29 | hmmhesays | http://www.thelostpacket.org/tricks.php if you want to give a whirl |
| 15:42.38 | quid246 | okay... lemme check it out |
| 15:42.51 | hmmhesays | i need some suggestions anyway |
| 15:42.51 | quid246 | nice domainname BTW |
| 15:43.01 | hmmhesays | quid246: thanks <chuckle> |
| 15:43.11 | [TK]D-Fender | yojanl: Thats a codec mismatch error |
| 15:43.22 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:43.59 | yojanl | [TK]D-Fender thats strange, cause I have nothing specific set, is there a way to debug this further? |
| 15:45.36 | yojanl | the prefered codecs in the GS are: PCMU, PCMA, G.723.1, G729A/B, GSM, PCMU |
| 15:46.08 | yojanl | and in sip.conf ive got |
| 15:46.20 | yojanl | disallow=all allow=ulaw allow=alaw allow=gsm |
| 15:48.18 | yojanl | so in short, it looks like it fails because asterisk says: Not acceptable here? Am I correct? |
| 15:48.38 | file | yojanl: full sip debug is good |
| 15:49.31 | yojanl | file: how can I enable full sip debug? |
| 15:50.05 | file | sip debug |
| 15:50.06 | file | :D |
| 15:52.06 | yojanl | file: okay, im already doing this :-) |
| 15:52.28 | file | pastebin it so others can see |
| 15:52.39 | yojanl | how do I pastebin? |
| 15:52.45 | file | ~pb |
| 15:52.46 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 15:52.51 | file | use pastebin.ca |
| 15:54.28 | SpaceBass | hmmhesays, and my one-way audio problem is back :) |
| 15:54.35 | hmmhesays | SpaceBass: lovely |
| 15:54.39 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:54.40 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:56.06 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:56.45 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:56.57 | smackus | how can I make it so that in a context of extensions.conf, if someone enters a non existent extension it transfers to a particular extension(receptionist)? |
| 15:57.35 | *** join/#asterisk morex (i=morex@host86-134-128-75.range86-134.btcentralplus.com) |
| 15:57.51 | trelane_ | exten => i,1,<something goes here> |
| 15:57.52 | morex | Hello all |
| 15:58.04 | hmmhesays | exten => i,1,Goto(sexy-stripper-voice,s,1) |
| 15:58.05 | *** join/#asterisk ToyMan (n=stuq@ool-45784f3b.dyn.optonline.net) |
| 15:58.06 | funxion | has anyone ever used ooh323c? |
| 15:58.07 | morex | I'm getting messages like these in /var/log/asterisk/messages |
| 15:58.11 | hmmhesays | funxion: yes |
| 15:58.12 | trelane_ | smackus, please look into exten => i (invalid) and exten => t (timeout) |
| 15:58.12 | hmmhesays | dialy |
| 15:58.17 | hmmhesays | *daily even |
| 15:58.20 | Assid | i want sexy stripper whose voice that is |
| 15:58.23 | funxion | does it work good |
| 15:58.33 | Assid | sup file |
| 15:58.41 | hmmhesays | funxion, I don't have much volume but it works well for me |
| 15:58.42 | smackus | thanks |
| 15:58.59 | file | WELL, nothing... at least that's what I'm saying - what's really sup? who knows |
| 15:59.01 | funxion | hmmhesays does it work with the current release? and is it no longer included in asterisk-addons? |
| 15:59.09 | sp0n9e | wooohooo. |
| 15:59.10 | sp0n9e | :( |
| 15:59.11 | hmmhesays | its not? |
| 15:59.16 | yojanl | this is what happens: http://pastebin.ca/126540 |
| 15:59.27 | funxion | I couldnt find it |
| 15:59.34 | hmmhesays | um it is in 1.2.3 |
| 15:59.35 | _Guhit | Argh! Ok, I can't believe I wasted sleep over this last night. It seems that the X100P was sharing the IRQ with the sound card and that was causing problems. Unloading the sound driver seems to have fixed all my problems with asterisk not hanging up and caller id issues. !!! |
| 15:59.39 | morex | cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files |
| 15:59.46 | sevard | morex: that's the problem with vi. |
| 15:59.51 | funxion | yeah |
| 15:59.52 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 16:00.06 | yojanl | the second not acceptable here comes because theres already a first invite, and reinvite is no i guess |
| 16:00.11 | morex | Any ideas? I think it's resulting in dropped calls... |
| 16:00.16 | hmmhesays | it works in 1.2.10 |
| 16:00.20 | Qwell | morex: turn up ulimit |
| 16:00.22 | morex | I think Asterisk is running out of file descriptors... |
| 16:00.28 | morex | Qwell: I thought I already did. |
| 16:00.37 | Qwell | before you run asterisk, and in the same shell |
| 16:00.42 | funxion | why can I not find it in 1.2.10 |
| 16:00.45 | sp0n9e | any ideas why ztcfg is segfaulting on me? |
| 16:00.53 | morex | I have ulimit -n 65536 |
| 16:00.54 | boch | i want to send a fax from command line, some .jpg or .bmp file, is it possible? where do i start reading? |
| 16:01.03 | Qwell | morex: -c I thought |
| 16:01.04 | morex | in my /etc/init.d/asterisk script |
| 16:01.06 | morex | Ah... |
| 16:01.20 | morex | Helpfully debian doesn't supply a ulimit man page |
| 16:01.22 | hmmhesays | boch, you using spandsp? |
| 16:01.45 | hohum | morex: that's because ulimit is built into your shell |
| 16:01.46 | yojanl | file: do you see anything that can help? |
| 16:01.52 | morex | OK |
| 16:02.04 | hmmhesays | funxion, it is in asterisk-addons 1.2.3 |
| 16:02.10 | funxion | ok |
| 16:02.14 | morex | ulimit -c specifies maximum core size. |
| 16:02.23 | yojanl | after further studying i think the grandstream doesnt receive or doesn respond to the confirm, it just start another invite |
| 16:02.27 | hohum | funxion: :P |
| 16:02.27 | funxion | does it install with addons or are there special parameters |
| 16:02.36 | morex | ulimit -n is the maximum number of open file descriptors |
| 16:02.43 | morex | From http://www.ss64.com/bash/ulimit.html |
| 16:02.44 | boch | hmmhesays: dont know, you tell me |
| 16:02.57 | morex | And I'm using ulimit -n... |
| 16:03.12 | boch | hmmhesays: i never dealed with asterisk and faxes |
| 16:03.14 | hmmhesays | you can accomplish it with spandsp and a perl or php script |
| 16:03.32 | yojanl | file: and others helping me, I have to go for half an hour (bring the girlfriend with the car :-) so be back in half an hour, hopefully someone know whats going on! |
| 16:03.46 | hmmhesays | nail her |
| 16:03.59 | file | hmmhesays: that's always on your mind! |
| 16:04.04 | hmmhesays | file: duh |
| 16:04.37 | boch | hmmhesays: ok thanks ill see some spandsp docs |
| 16:05.02 | hmmhesays | for some monetary compensation I could have a working demo for you in a couple hours |
| 16:05.14 | *** part/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil) |
| 16:05.14 | boch | hmmhesays: do i have to re-build * to support spandsp ? |
| 16:05.23 | hmmhesays | no |
| 16:05.30 | boch | awesome |
| 16:10.42 | sp0n9e | does sangoma have an irc chat room? |
| 16:10.48 | funxion | hmmhesays will addons 1.2.3 work with asterisk 1.2.10? |
| 16:10.57 | Qwell | funxion: yes |
| 16:10.57 | sp0n9e | funxion: yes. |
| 16:11.02 | funxion | thank you |
| 16:12.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:13.03 | funxion | hmmhesays how would I got about installing ooh323c from asterisk-addons? |
| 16:13.33 | funxion | I cant seem to find much documetation on it |
| 16:13.39 | funxion | can someone point me in the right direction |
| 16:14.13 | eKo1 | There should be documenation in the asterisk-addons package. |
| 16:14.37 | funxion | I did a grep -e ooh323c * |
| 16:14.39 | funxion | nothing |
| 16:14.56 | funxion | even in the channels dir |
| 16:15.34 | hmmhesays | funxion |
| 16:15.38 | funxion | yes |
| 16:15.41 | hmmhesays | what the crap man, it is in asterisk-addons |
| 16:15.44 | hmmhesays | not asterisk |
| 16:15.55 | funxion | I know |
| 16:15.59 | smackus | http://pastebin.ca/126550 when i play a meesage from exten => 5313 it offers the option to enter an extension... can anyone tell me what I have messed up so that it ignores the extension entered? |
| 16:16.03 | funxion | in /usr/src/asterisk-addons |
| 16:16.06 | charles___ | ohh my freaking head |
| 16:16.09 | morex | So ulimit -n doesn't work with Asterisk? |
| 16:16.12 | charles___ | so easyyyyy |
| 16:16.14 | smackus | nevermind.... I know what I am missing. |
| 16:16.36 | funxion | if I just install addons it will work |
| 16:16.48 | funxion | or do I need to do anything special for ooh323c specifically |
| 16:16.54 | smackus | wait... nope. ok, question still stands |
| 16:17.57 | funxion | hmmhesays does it require open_h323 and pwlib? |
| 16:18.40 | smackus | based of my pastebin, should i just be able to dial any extension in the same context as what I have called into? 5313? |
| 16:19.17 | smackus | so if i dial 5313, and I hear the message if you know your extension dial it now... then I dial 900, should that now work? |
| 16:19.26 | boch | anyone familiar with Sipura2100 and faxes through * ? |
| 16:23.09 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 16:23.28 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 16:23.52 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 16:24.06 | websae | faxes through asterisk can be difficult |
| 16:26.56 | hmmhesays | spandsp works ok for me |
| 16:27.59 | boch | always? |
| 16:28.28 | hmmhesays | always? |
| 16:28.38 | boch | my * only passes some faxes, not all |
| 16:28.51 | hmmhesays | oh yeah? |
| 16:28.54 | hmmhesays | why is that? |
| 16:29.05 | boch | thats a good question |
| 16:29.10 | eKo1 | Mine does the same. Faxing is not reliable on *. |
| 16:29.15 | hmmhesays | mine seems to work reliably |
| 16:29.27 | hmmhesays | what is your fax setup like? |
| 16:29.35 | boch | dont know if is a bw problem, or what. |
| 16:29.41 | eKo1 | Most of my faxes result in POOR LINE CONDITION |
| 16:29.50 | hmmhesays | hmm not mine |
| 16:30.04 | macTijn | what's your uplink? SIP ? |
| 16:30.07 | websae | i find it better to outsource faxing to a third party... |
| 16:30.10 | macTijn | and if so, what codec ? |
| 16:30.14 | sp0n9e | depends on if faxes get caught up in the echo cancellation. |
| 16:30.20 | sp0n9e | from what i've read. |
| 16:30.27 | websae | and not come within 10ft of it and asterisk |
| 16:30.35 | funxion | does ooh323c require open_h323 and pwlib? |
| 16:30.39 | hmmhesays | I use spandsp and sip reliably |
| 16:30.41 | hmmhesays | funxion: no |
| 16:30.45 | macTijn | hmmhesays: what codec ? |
| 16:30.55 | hmmhesays | ulaw |
| 16:31.02 | funxion | will it just install by make and make install asterisk-addons? |
| 16:31.05 | macTijn | that'll work, if there's no packet loss |
| 16:31.06 | funxion | and specialconfig? |
| 16:31.10 | eKo1 | My fax machine is connected to a SPA2002 set to use ulaw which sends the call to our main * box which then sends the call to our * box that is connected to our PRI lines. |
| 16:31.12 | hmmhesays | funxion read the config files |
| 16:31.24 | macTijn | eKo1: any transcoding ? |
| 16:31.31 | hmmhesays | transcoding fax? |
| 16:31.31 | hmmhesays | right |
| 16:31.38 | hmmhesays | funxion read the docs |
| 16:31.40 | coppice | faxing over VoIP channels will seldom work, whatever codec you choose |
| 16:31.52 | eKo1 | macTijn: no because the main * box sends the call to the PRI * box using ulaw. |
| 16:31.59 | eKo1 | So it goes ulaw all the way. |
| 16:32.01 | macTijn | coppice: works with ulaw or alaw |
| 16:32.03 | hmmhesays | coppice I fax over SIP/Ulaw all the time |
| 16:32.07 | macTijn | eKo1: that's good |
| 16:32.27 | coppice | if faxing over VoIP works, its by luck, not design |
| 16:32.36 | funxion | hmmhesays Im only finding info on jerjer's h323 |
| 16:32.36 | eKo1 | coppice: ditto |
| 16:32.37 | *** join/#asterisk kpettit (n=keith@adsl-70-241-67-181.dsl.hstntx.swbell.net) |
| 16:32.51 | hmmhesays | funxion it is in the asterisk addon package |
| 16:33.07 | funxion | in /usr/src/asterisk-addons/doc# |
| 16:33.09 | coppice | macTjin: the codec is only slightly relevant |
| 16:33.10 | websae | it's the downside of getting business clients to switch over to VoIP |
| 16:33.19 | macTijn | for fax |
| 16:33.28 | macTijn | works quite nice |
| 16:33.55 | boch | fucking outdated ppl who uses faxes |
| 16:34.00 | macTijn | uhuh |
| 16:34.08 | *** join/#asterisk Lead_one (n=wont@dslb-084-058-191-079.pools.arcor-ip.net) |
| 16:34.11 | coppice | macTjin: that is also by luck, since the digum cards don't sync |
| 16:34.26 | macTijn | apparently it's the only way to transfer legal-aimed docs |
| 16:34.57 | coppice | its sad the kind of crap that is acceptable as a legal document |
| 16:35.06 | macTijn | coppice: heh, true :) |
| 16:35.21 | macTijn | * raised my error rate on faxes |
| 16:35.28 | macTijn | which doesn't surprise me at all |
| 16:35.43 | coppice | telexes used to be legally acceptable, and any fool could fake one in a few minutes |
| 16:35.58 | macTijn | heh |
| 16:36.26 | macTijn | faxes are quite fun to do a man-in-the-middle attack with |
| 16:36.45 | brodiem | i hate fax |
| 16:36.53 | boch | whats the reinvite method for ? |
| 16:36.53 | *** join/#asterisk Kyler (n=chatzill@74.132.200.229) |
| 16:36.54 | macTijn | so do I |
| 16:37.10 | macTijn | but working at an ISP is not doable without fax |
| 16:37.16 | coppice | boch reinviting. what else? |
| 16:37.29 | macTijn | in .nl we still need a bunch of signed docs to register a .nl domain name |
| 16:37.34 | boch | coppice: but it is codec related ? |
| 16:37.58 | macTijn | boch: no, more NAT related |
| 16:38.03 | brodiem | I have to listen to people complain every day because a fax failed |
| 16:38.05 | boch | oh ok |
| 16:38.11 | coppice | reinvite can be used to change codecs, change routing, change between voice and T.38 FAX and other things |
| 16:38.15 | websae | i say just setup a proxy to a 3rd party fax service for you customers |
| 16:39.26 | websae | *your |
| 16:39.37 | funxion | why can I not find ooh323c in any of the svn tags? |
| 16:39.39 | websae | that way you don't even have to deal with fax problems |
| 16:40.13 | Kyler | A long time ago I recall seeing something about performing actions (in the dialplan) after the calling (SIP) party has terminated the call. I'm not finding that now. Pointers? |
| 16:40.23 | brodiem | websae, will that work while being able to still use a physical fax machine so that docs don't haveto be scanned/e-mailed? |
| 16:40.24 | macTijn | hmm |
| 16:40.26 | macTijn | isn't there a set of extensions for asterisk that permit FoIP ? |
| 16:40.43 | macTijn | should just be a modification of rxfax |
| 16:40.50 | coppice | not yet |
| 16:40.59 | macTijn | in the works ? |
| 16:41.08 | coppice | in testing now |
| 16:41.14 | Kyler | I'm coming into this fax discussion late but...Gafachi says they have T.38 service. I've been meaning to try it with Asterisk. |
| 16:41.16 | macTijn | cool |
| 16:41.33 | funxion | ok |
| 16:41.36 | funxion | Im a dumbass |
| 16:41.39 | funxion | sry hmmhesays |
| 16:42.23 | coppice | but we have found the T.38 passthrough stuff in trunk has problems with the real world. someone will have to sort that out, because I won't |
| 16:42.51 | Kyler | So my fuzzy recollection is that there's no way (or only a tricky way?) to do anything in the dialplan after the caller terminates the call. |
| 16:43.02 | hmmhesays | extension h |
| 16:43.12 | benjk | won't work in a macro though |
| 16:43.19 | Kyler | No, it's not that easy...is it? I thought there was a problem with that. |
| 16:43.20 | benjk | macros don't have h support |
| 16:43.31 | benjk | also, only if the called party hangs up |
| 16:43.33 | SpaceBass | anyone know why simply turning rtp debug on would solve audip problems? |
| 16:43.34 | Kyler | Ah! That's why it didn't work for me. O.k., workaround... |
| 16:43.46 | Kyler | benjk: Thanks! |
| 16:43.54 | macTijn | heh |
| 16:43.57 | benjk | if the calling party hangs you can't trap that |
| 16:44.02 | macTijn | skype on asterisk hack: http://slacker.com/~nugget/projects/asterisk/page12 |
| 16:44.06 | macTijn | quite funny solution :) |
| 16:44.15 | nortex | Can AGI call an asp page on a server? |
| 16:44.16 | benjk | I have a patch for app_macro that adds h support |
| 16:44.26 | macTijn | nortex: AGI can do anything you want |
| 16:44.38 | benjk | nope |
| 16:44.50 | sp0n9e | nortex: i would guess you would have to write some type of wrapper for the http request |
| 16:45.24 | benjk | hangup by the calling party immediately terminates the pbx |
| 16:45.35 | benjk | but I have a patch for that too |
| 16:45.51 | nortex | I really just want * to send the information to the outside app, not really needing a response. |
| 16:45.58 | benjk | so far its only working for calling party hangup before the bridge |
| 16:46.17 | benjk | still working on the part that traps hangup by calling party after the bridge |
| 16:46.30 | sp0n9e | nortex: you'll have to write an AGI script to do that, or get a cgi module for asp and use that |
| 16:47.23 | sp0n9e | #!/bin/php -q\nfile_get_contents('http://example.com/foo.asp'); |
| 16:47.33 | sp0n9e | where \n is a newline (and i forgot <?php |
| 16:47.45 | sp0n9e | that's a quick wrapper to request the page |
| 16:48.52 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 16:49.32 | macTijn | sp0n9e: that's not reading headers, you need to do that first |
| 16:50.02 | *** join/#asterisk marcster (n=rubber@203.87.184.218) |
| 16:50.16 | sp0n9e | macTijn: why would i need to read the headers? |
| 16:50.19 | marcster | hi. what is a device like sipura 3000 called? |
| 16:50.21 | sp0n9e | not using any of the information |
| 16:50.28 | marcster | is it a modem? |
| 16:50.36 | boch | now i can send from pstn to a SIP endpoint, but not backwards, what can be the problem ? |
| 16:51.24 | macTijn | sp0n9e: hm, wait, that's only if you are also sending commands to * |
| 16:51.28 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:51.28 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:51.43 | [TK]D-Fender | marcster: Its an ATA |
| 16:52.14 | macTijn | Analog Telephony Adapter |
| 16:52.17 | [TK]D-Fender | marcster: But for the fact it also has an FXO port (ATA is typically reserved for FXS devices) you could say "SIP gateway" |
| 16:52.27 | blitzrage | one way audio in an internal network is awesome! |
| 16:52.30 | marcster | i see |
| 16:52.43 | macTijn | blitzrage: firewall ? :> |
| 16:52.58 | macTijn | or is it on purpose ? :) |
| 16:53.00 | marcster | thanks. a google search on 'sipura ATA' the results i needed to get started. |
| 16:53.17 | macTijn | \o/ |
| 16:56.01 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 16:59.43 | blitzrage | macTijn: no FW on the internal network |
| 16:59.58 | blitzrage | I think asterisk is forwarding the RTP frames to a different subnet than what it should |
| 17:00.05 | macTijn | blitzrage: ahh |
| 17:00.08 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:00.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:00.09 | macTijn | <- gone |
| 17:00.14 | macTijn | l8r all |
| 17:01.48 | [TK]D-Fender | blitzrage: Missing localnet clause is what it sounds like.... |
| 17:01.50 | *** join/#asterisk ajaymn (n=Ya@70.59.126.206) |
| 17:03.10 | *** join/#asterisk nDuff (n=ccd@64.128.31.220) |
| 17:03.54 | *** join/#asterisk ki2k (n=ki2k_@207.231.83.242) |
| 17:04.07 | nDuff | Can I specify a behavior to take place immediately whenever all members of a queue are already on the line? |
| 17:04.13 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 17:04.22 | ki2k | anyone know if FWD.net is broken for new accounts? |
| 17:05.20 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:09.05 | nortex | It still suprises me what sound files are missing, things like night mode or closed |
| 17:10.02 | ki2k | anyone know of a way to allow the use of the @ sign as part of a password? |
| 17:10.35 | eKo1 | ki2k: good question. Maybe enclosing it in "" will help. |
| 17:11.08 | ki2k | nope |
| 17:11.18 | ki2k | i'm thinking of using html encoding |
| 17:12.29 | ki2k | nope, %40 doesnt work |
| 17:12.42 | ki2k | welp, thats a bug |
| 17:12.55 | ki2k | cant use : and @ as part of your password |
| 17:13.00 | ki2k | in asterisk |
| 17:13.46 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 17:14.40 | sb_mx | ki2k, we use : as part of our sip peers' password |
| 17:14.45 | eKo1 | ki2k: when you way password, what do you mean? |
| 17:14.53 | eKo1 | s/way/say |
| 17:15.09 | ki2k | register username:password@sip.server.com |
| 17:15.27 | eKo1 | Ah. Try single quotes then. |
| 17:15.31 | eKo1 | Maybe that'll help |
| 17:15.37 | [hC] | anyone have an example of having a recording play, instead of hearing a caller hear 'ringing'. maybe something that says 'please wait while we connect your call' |
| 17:15.53 | ki2k | try register username:'p@ssword'@server.com ? |
| 17:15.59 | eKo1 | [hC]: that is pretty easy to do. |
| 17:16.11 | eKo1 | ki2k: yeah |
| 17:16.25 | [hC] | Yeah, i imagine it is.. Is it a dial option, or do i have to use playback first? |
| 17:16.27 | ki2k | eKo1: nope |
| 17:16.41 | ki2k | iax2 show registry doesnt even see it |
| 17:17.10 | [hC] | oh i can use special hold music too |
| 17:17.12 | [hC] | that works |
| 17:17.15 | [hC] | so it will auto-repeat |
| 17:17.31 | *** join/#asterisk Gregabyte (n=greg@gateway.digium.com) |
| 17:17.42 | ki2k | eKo1: know of any other ways to escape out the @ ? |
| 17:18.01 | ki2k | my voip provider set up an account for me but he's now on vacation |
| 17:18.03 | [TK]D-Fender | [hC]: Give them actual MoH after playing the annoucment once |
| 17:18.08 | ki2k | i cant have him change hte passwd |
| 17:18.11 | ki2k | till monday |
| 17:18.38 | [hC] | [TK]D-Fender: yeah. now, playing the announcement once.. (i realize this probably sounds stupid but..) can I have dial play it, or do i have to hack it in some other way? |
| 17:19.02 | [hC] | I dont see any options in Dial() to do that. |
| 17:19.06 | sp0n9e | ki2k: encode @ into %40 |
| 17:19.20 | sp0n9e | p%40ssword |
| 17:19.35 | sp0n9e | it's a URI not a URL, but it works in some URIs |
| 17:19.36 | [TK]D-Fender | [hC]: Just sove it before your dial. |
| 17:19.43 | [TK]D-Fender | shove* |
| 17:20.01 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.74) |
| 17:22.47 | [hC] | [TK]D-Fender: yeah that works, the only thing im curious about in that way, is if people dial extensions off an ivr, i just do an include of the local sip context... I guess i could hook like... _XXXX, play the thing, then goto ${EXTEN} of the local sip context. |
| 17:24.59 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 17:25.04 | Wazb^ | hi |
| 17:25.10 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 17:25.11 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 17:25.39 | [TK]D-Fender | [hC]: Yup, you acn front-end it like that. and on failed GOTO then send them to "i" |
| 17:25.41 | Wazb^ | does anybody know any opensource wholesale billing with asterisk? |
| 17:26.14 | Beighto | Is there any way to put a delay on input dtmf? I am having a problem where if someone dials a 4 digit extension it would go the a 3 digit extension and then go to an invalid option immediatly after. Setting the digit timeout doesn't seem to effect this. |
| 17:26.27 | ki2k | sp0n9e: already tried it |
| 17:27.08 | }btorch{ | hey guys I have some horrible echo now aftern upgrading my zaptel and asterisk to the latest |
| 17:27.21 | }btorch{ | echo on iax2 to iax2 calls |
| 17:27.58 | eKo1 | check the latency |
| 17:28.16 | }btorch{ | how ? |
| 17:28.51 | }btorch{ | I'm using a one ear piece headset (cheap one) and idefisk .. whenever I lower the speaker volume echo seems to go away |
| 17:29.29 | }btorch{ | but if you try to hear it really hard you can .... MG2 seemed to make the echo on my zaps worse too before I used MG1 i think |
| 17:29.52 | smackus | I am using addqueuemember to log my agents into the phones... one issue I have is that when on a call, the queue will still deliver the calls to that phone, even though it is not idle. is there a way to stop that? |
| 17:30.03 | [TK]D-Fender | }btorch{: That's accoustic echo and has nothing to do with Zaptel. |
| 17:30.29 | blitzrage | [TK]D-Fender: the phones aren't talking to anything outside the network though -- seems a canreinvite=no resolves the one way audio (phones are SIP, but only talk to other SIP phones on the LAN or to chan_zap) |
| 17:30.29 | [TK]D-Fender | smackus: Using Local channel? |
| 17:30.58 | Wazb^ | does anybody know any opensource wholesale billing with asterisk? |
| 17:30.59 | [TK]D-Fender | blitzrage: You implied it spans subnets (even though local). My guess is the phones don't know how to route back. |
| 17:31.00 | hmmhesays | wow did I drink waay too much last night |
| 17:31.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:31.27 | Beighto | anybody using voxee? |
| 17:31.35 | blitzrage | [TK]D-Fender: the phones are on the same subnet though -- I was just theorizing that Asterisk was getting confused and was sending the audio out a different route the phones couldn't be reached at |
| 17:31.36 | smackus | [TK]D-Fender: I dont know how to do that |
| 17:31.39 | [hC] | I wonder if its cause ive mucked with the tonezone stuff here.. Calling users dont hear ringing indication when * is ringing sip phones |
| 17:31.40 | [hC] | very odd |
| 17:32.03 | blitzrage | [TK]D-Fender: I might still try the localnet in there later to see if I can get reinvites to work |
| 17:32.16 | *** join/#asterisk Akhilesh (n=akhilesh@203.76.184.38) |
| 17:32.24 | blitzrage | but thus far at least I've got a production box working again |
| 17:32.26 | }btorch{ | [TK]D-Fender, I believe that is true on the iax <-> iax calls but what about a call that comes in from a zap line to a meetme room |
| 17:32.43 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 17:32.49 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:32.49 | *** join/#asterisk riddlebox (n=blah@24-207-167-238.dhcp.stls.mo.charter.com) |
| 17:32.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:33.01 | [TK]D-Fender | blitzrage: That won't help on re-invite. The phone still needs to know where to go. |
| 17:33.22 | [TK]D-Fender | }btorch{: Who in the room gets echo? |
| 17:33.24 | feld_ | if I park calls I dont get any info from asterisk telling me where I parked it -- I'm just starting to try to config this feature... all I have is include => parkedcalls in my extensions.conf and briefly looked over features.conf |
| 17:33.31 | feld_ | what am I missing? I dont see it in the docs. |
| 17:33.35 | }btorch{ | maybe I shouldn't have added aggressive suppressor and changed to MG2 |
| 17:33.40 | }btorch{ | everyone |
| 17:33.54 | Akhilesh | Hi guys, I need some information. |
| 17:33.56 | }btorch{ | [TK]D-Fender, the zap caller hears his own echo |
| 17:34.00 | [TK]D-Fender | feld_: Screw Parking (at least the one that comes with *). Get app_valetparking from pbxfreeware.... |
| 17:34.01 | brodiem | }btorch{ doesn't agressive just cause it to go half duplex? |
| 17:34.09 | }btorch{ | [TK]D-Fender, and me (iax) and the other (iax) user too |
| 17:34.11 | [TK]D-Fender | }btorch{: When he's ALONE? |
| 17:34.14 | Akhilesh | As asterisk works only on linux, is there PBX on windows ? |
| 17:34.14 | feld_ | [TK]D-Fender: i'll check it out thx |
| 17:34.26 | }btorch{ | [TK]D-Fender, did not test that |
| 17:34.31 | }btorch{ | will do that .. brb |
| 17:34.48 | ki2k | Akhilesh: i'm running asterisk on a windows server |
| 17:34.55 | ki2k | Akhilesh: Microsoft Virtual Server |
| 17:34.57 | ki2k | hahahaha |
| 17:35.01 | ki2k | with ztdummy |
| 17:35.08 | Akhilesh | Where did u get binaries on Windows ? |
| 17:35.20 | ki2k | I didnt |
| 17:35.21 | Akhilesh | :) Ki2K.... I need it |
| 17:35.28 | Rawplayer | o god.. |
| 17:35.29 | [TK]D-Fender | Akhilesh: LOL. There are no "windows" binaries.... its all virtualized Linux.... |
| 17:35.34 | ki2k | yep |
| 17:35.35 | sp0n9e | yep |
| 17:35.36 | ki2k | it works |
| 17:35.47 | sp0n9e | you can get MS Virtual PC for free now |
| 17:35.52 | ki2k | yep |
| 17:35.53 | sp0n9e | don't know how well that would run it |
| 17:36.04 | ki2k | sp0n9e: testing now |
| 17:36.05 | smackus | [TK]D-Fender: ok, so is this what you are pointing me to? AddQueueMember(MyQueue,Local/123 at my-context) ) |
| 17:36.08 | ki2k | it seems to be ok |
| 17:36.16 | ki2k | it's a dual core ath 64 |
| 17:36.50 | Akhilesh | Is there soft PBX which is natively compiled on windows ? |
| 17:36.51 | sp0n9e | now if only i could get my sangoma card working :) |
| 17:37.25 | eKo1 | Akhilesh: * + cygwin |
| 17:37.39 | blitzrage | you'll just have timing issues regardless in a virtualized environment |
| 17:37.40 | sp0n9e | how's it run on cygwin? |
| 17:37.57 | ki2k | i dont even wanna bother with cygwin |
| 17:38.00 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
| 17:38.07 | ki2k | Akhilesh: why do you want a windows machine? |
| 17:38.14 | Akhilesh | Native ? Without cygwin ? |
| 17:38.21 | sp0n9e | http://www.asteriskwin32.com/ <<< looks pretty bad. |
| 17:38.53 | [TK]D-Fender | Akhilesh: Forget about * on windows. You want a windows VoIP server? Odds are it will cost you and not be this flexible., |
| 17:39.11 | smackus | is there a way to do something like this: exten => *1,2,AddQueueMember(test1,Local/<some variable for device>) |
| 17:39.41 | [TK]D-Fender | smackus: Clarify that var.... |
| 17:39.42 | feld_ | ARGH... |
| 17:40.02 | feld_ | my boss bought Grandstream phones after I specificly told home to get Polycoms |
| 17:40.10 | [TK]D-Fender | feld_: I've recently implemented the valetparking and it just rocks..... |
| 17:40.16 | [TK]D-Fender | feld_: ZOMG!!!! |
| 17:40.20 | feld_ | these things suck ass and they dont send DTMF right |
| 17:40.28 | feld_ | they are 101 in the config and they wont work with voicemail |
| 17:40.54 | smackus | [TK]D-Fender: am i following you correctly on using Local channel to correct my queue issue? |
| 17:41.59 | [TK]D-Fender | feld_: Get them returned... |
| 17:42.20 | [TK]D-Fender | smackus: Not sure.... not a qualified sample fo code to examine... |
| 17:42.44 | bcnl | [TK]D-Fender: valtparking? |
| 17:43.15 | bcnl | I still no longer have MoH when ppl are parked, I get a codec translation error, eventhough the calls are ulaw |
| 17:46.39 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 17:46.55 | ki2k | feld: fire your boss |
| 17:48.38 | [TK]D-Fender | feld_: Just return them. End of story.... |
| 17:50.38 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 17:52.42 | smackus | ok... the local thing is the answer to my queuing issue, thanks [TK]D-Fender: |
| 17:52.50 | smackus | now i just need to get the correct variable set up :-D |
| 17:53.06 | smackus | i have it so that it uses whatever i dialed to be the extension... which I dont want. |
| 17:53.15 | ki2k | so anyone else have any ideas about my password problem? |
| 17:54.52 | feld_ | ki2k: he already breaks enough stuff around here... and he has some financial issues that just surfaced to us employees, so i will be leaving as soon as i get another job lined up |
| 17:55.26 | feld_ | [TK]D-Fender: found that dtmfmode=auto fixes the issue... works for these phones and out xlite phones, so it will suffice... but man... just reading the asterisk wiki on these things makes me want to puke. |
| 17:55.34 | feld_ | *and our xlite |
| 17:56.05 | jsaunders | * You were kicked from #yate by killall-9 (killall-9) |
| 17:56.05 | jsaunders | - |
| 17:56.05 | jsaunders | #yate unable to join channel (address is banned) |
| 17:56.14 | jsaunders | :D Heheh. Wow, she really hates me. |
| 17:56.25 | file | what did you do? |
| 17:56.33 | file | and please don't do it here |
| 17:56.34 | file | :P |
| 17:56.36 | jsaunders | :) |
| 17:57.34 | jsaunders | I don't know what I did. I was given no reason. |
| 17:58.38 | jsaunders | I suspect I know why... because of differing opinions on a certain topic between Diana and myself. Was just a little suprised to see it from killall-9. Curious as to wether it was him who actually did it, or just her using his account. |
| 17:58.45 | feld_ | rofl my cow-orker changed his ringtone to Duke Nukem's "Get back to work, you slacker!" |
| 17:58.58 | jsaunders | Heheh, beauty. |
| 17:59.01 | jsaunders | I loved that game. |
| 17:59.18 | feld_ | yeah I cant wait for DNF to be released. Actually, nobody can wait. |
| 17:59.51 | jsaunders | Oh? DNF you say. News to me. *googles* |
| 17:59.52 | mog | jsaunders, what was all that about? |
| 18:01.00 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:01.00 | ki2k | DNF? ha |
| 18:01.11 | jsaunders | Diana claims to have given me all this help with regards to our skySwitch project. Apparently, according to her, we had an agreement that I would release our code if she helped us. That was never the agreement, yet she's adamant it was. I have several logs of conversations clearly showing otherwise but she's ignorant & stubborn and truly believes she is right. |
| 18:01.49 | jsaunders | If I was spiteful I'd put up a public website w/ the logs but alas... I don't think that would do much. |
| 18:02.06 | mog | ahh |
| 18:02.12 | mog | was funny argument |
| 18:02.18 | jsaunders | It's beyond all of that anyways. She simply doesn't like me now cuz I argue with her. :) |
| 18:02.24 | jsaunders | Oh, you saw? Heheh. |
| 18:02.29 | mog | yeah |
| 18:02.34 | jsaunders | Man, that's one of many, let me tell you. |
| 18:02.38 | mog | im in all pbx channels |
| 18:02.48 | jsaunders | Rediculous. It's been going on for over a half a year now. |
| 18:02.50 | jsaunders | Heheh |
| 18:02.59 | jsaunders | I try and be in all. Down 1 now. |
| 18:03.05 | jsaunders | Ah well, that's what bnc is for. |
| 18:03.22 | nDuff | My SIP phones appear not to be informing the server when the user is busy, but instead trying to push the incoming call through on an additional line. Is this typically fixable behavior? |
| 18:03.22 | file | mog: you spy you! |
| 18:04.09 | jsaunders | mog: Did she say anything after my last comment and before I was kicked? I don't have the channel logged and my client closed the window when kicked. |
| 18:04.10 | puppet | 20:01:09 < diana> hi puppet |
| 18:04.14 | puppet | ;P |
| 18:04.29 | mog | nothing really |
| 18:04.33 | jsaunders | kew, tnx mang |
| 18:05.05 | smackus | based on the wiki... (${EXTEN}: The current extension ) what do i use to setvar to the extension i am dialing from? |
| 18:07.08 | feld_ | what the hell |
| 18:07.10 | feld_ | load average: 13.41, 5.14, 1.96 |
| 18:07.29 | feld_ | ^^ from when I put 4 grandstream phones on the system |
| 18:09.00 | jsaunders | heheh |
| 18:10.51 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 18:11.46 | Seba_soy | hello, anyone has configure realtime sip in astrisk? |
| 18:11.47 | *** join/#asterisk ^Xypher (n=bentley@arnor.fornost.com) |
| 18:12.16 | *** part/#asterisk ^Xypher (n=bentley@arnor.fornost.com) |
| 18:12.21 | quid246 | seba; yeah, just follow the wiki... pretty easy |
| 18:12.37 | Seba_soy | can I register a sip account using realtime?, I can make calls but I can't registre the user |
| 18:13.02 | quid246 | seba; you mean create a user in the SQL table? |
| 18:13.07 | *** join/#asterisk jeebusmobile (n=jeebusmo@12.180.154.130) |
| 18:13.32 | *** join/#asterisk Un1x (i=Sean@CPE123456789123-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 18:13.33 | Seba_soy | well, I have x-lite, so I want to register it so it can see if it have voicemail and all that... |
| 18:13.59 | quid246 | then insert it into the table |
| 18:14.05 | Seba_soy | I have it inserted... |
| 18:14.36 | quid246 | if it's in the table & x-lite is configed properly, then check your DB settings in extconfig |
| 18:15.03 | Seba_soy | When I try to register, asterisk said NOT FOUND, but if I send a call directly, it goes OK, it uses extensions and all config I put on DB |
| 18:18.27 | javar | when modify manager.conf, i need reboot the server? |
| 18:18.53 | *** join/#asterisk Ixitxachitl (n=m@209.151.130.10) |
| 18:18.59 | Seba_soy | done |
| 18:19.16 | Seba_soy | I have to replace type=user by type=friend |
| 18:19.18 | Seba_soy | :) |
| 18:19.20 | Seba_soy | my mistake |
| 18:20.57 | smackus | ok... what am i doing wrong? http://pastebin.ca/126692 I need to have the variable for Local/$<variable> show the extension that I am logging in from how can I do that? |
| 18:21.22 | benjk | ${variable} |
| 18:21.30 | benjk | not $<variable> |
| 18:21.57 | Ixitxachitl | Wondering if anyone has any ideas: i've got a PRI connected to digium card, and inbound calls can be received and heard just fine. However, outbound calls are garbled to the point of incomprehension, but only to the phone calling out (the receiving line hears audio properly) |
| 18:22.00 | smackus | sorry... what am i doing wrong? http://pastebin.ca/126692 I need to have the variable for Local/${variable} show the extension that I am logging in from how can I do that? |
| 18:23.44 | *** join/#asterisk NDT (n=noone@cpe-24-195-66-214.nycap.res.rr.com) |
| 18:24.00 | *** join/#asterisk c4t3l (n=c4t3l@72.16.242.49) |
| 18:24.08 | sb_mx | smackus, what version of * are you running? i've found trunk doesnt support SetVar anymore |
| 18:24.39 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 18:24.45 | smackus | Asterisk 1.2.10 |
| 18:26.37 | sb_mx | smackus, found your error |
| 18:27.34 | sb_mx | change the AddQueueMember line to exten => *3,3,AddQueueMember(test1,Local/${CHANNEL}) and the next ones add +1 to their priority |
| 18:28.27 | smackus | ahhh crap |
| 18:28.50 | puppet | ANyone know how it is with g729 in europe? do you need to pay license here since it is software patent? |
| 18:29.04 | benjk | yes you do |
| 18:29.14 | ki2k | how does the licensing work? |
| 18:29.23 | benjk | you purchase the licenses from Digium |
| 18:29.26 | ki2k | any why havent anyone come up w/ a open src version? |
| 18:29.27 | puppet | well |
| 18:29.30 | puppet | there are other ways |
| 18:29.35 | puppet | the intel open729 thing |
| 18:29.36 | benjk | because it is patent encumbered |
| 18:29.48 | puppet | well there are no software patents in .eu |
| 18:29.51 | ki2k | what intel thing? |
| 18:29.55 | benjk | you have to pay the royalties to the patent holders no matter if its open source or not |
| 18:30.03 | benjk | nothing to do with software patents |
| 18:30.04 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 18:30.04 | puppet | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 18:30.11 | c4t3l | hence lic fees |
| 18:30.16 | benjk | for educational purposes only |
| 18:30.24 | benjk | any other use is illegal |
| 18:30.30 | ki2k | ah ok |
| 18:30.33 | c4t3l | hence lic fees |
| 18:30.37 | benjk | EU has signed trade agreements with the US |
| 18:30.38 | puppet | well isnt it really software patent? since it is a software they want lic fees for |
| 18:30.50 | benjk | that means all US patents are valid in the EU through the backdoor |
| 18:30.56 | puppet | gah |
| 18:30.59 | puppet | that sucks :( |
| 18:31.07 | benjk | and vice versa |
| 18:31.09 | *** part/#asterisk Ixitxachitl (n=m@209.151.130.10) |
| 18:31.14 | puppet | well well, then i just have to pay up and increase the thought price |
| 18:31.38 | smackus | it is still trying to set that variable to the extension *1. which is the dialed extension, not the extension i am dialing from. |
| 18:31.45 | benjk | or avoid using patent encumbered codecs in the first place |
| 18:32.05 | benjk | there are very good free codecs out there |
| 18:32.15 | puppet | benjk: speex? |
| 18:32.20 | c4t3l | yes |
| 18:32.23 | benjk | Speex has the same low bandwith footprint and quality |
| 18:32.27 | justinu|laptop | benjk: you live! |
| 18:32.35 | benjk | hi justin |
| 18:32.41 | justinu|laptop | how the heck have you been? |
| 18:32.43 | puppet | so i should forget about g729 and go speex? |
| 18:32.56 | benjk | yes, if you can |
| 18:33.10 | ki2k | also, since codecs are so cpu intensive, why hasnt people made coprocessor cards that sit on pci/pci-e or even on the digitum cards |
| 18:33.12 | benjk | I was busy and away |
| 18:33.21 | puppet | ki2k: thats true |
| 18:33.25 | puppet | ki2k: should be good |
| 18:33.36 | benjk | you never called me up, did you actually come to Japan after all? |
| 18:33.41 | justinu|laptop | no :( |
| 18:33.54 | ki2k | thats why dedicated pbx's are faster, dedicated hardware |
| 18:34.31 | c4t3l | ki2k: you have to get hardware manufacturers motivated to do so |
| 18:34.48 | }btorch{ | [TK]D-Fender, wierd the echo did go away once us both iax client disconnected from the meetme room and connected back using a regular phone |
| 18:34.49 | benjk | kik2, Digium has just announced a g729/g723 PCI card |
| 18:34.50 | puppet | c4t3l: well why cant digium do it? ;P |
| 18:35.01 | *** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au) |
| 18:35.02 | cybertrickle | I am trying to install zaptel, and its not making the .so files (thus I cant modprobe it). Anything I can try ? |
| 18:35.09 | c4t3l | :D |
| 18:35.39 | file | the TC400P |
| 18:35.43 | benjk | yep |
| 18:35.45 | }btorch{ | [TK]D-Fender, maybe IDEFISK isn't doing such a good job on echo cancelling or maybe it may be the echo acoustic due to the cheap headset |
| 18:35.48 | ki2k | benjk: is support in asterisk yet? |
| 18:36.11 | benjk | of course if its from Digoum it will be for Asterisk |
| 18:36.22 | benjk | but its not shipping yet |
| 18:36.24 | ki2k | yet as in already done |
| 18:36.31 | ki2k | already in the current distro |
| 18:36.31 | benjk | I heard it will ship within a month or so |
| 18:36.50 | file | current distro? some of the components required for it are in trunk |
| 18:37.12 | ki2k | ah ok |
| 18:37.25 | ki2k | anyway, what's w/ the mmx optimizations? |
| 18:37.36 | ki2k | them not compatible w/ AMD cpus? |
| 18:38.34 | cybertrickle | When you make zaptel, the drivers are in the .so files, not .ko files right ? |
| 18:38.45 | file | cybertrickle: uh... no |
| 18:38.53 | brodiem | depends if you're on 2.4 or 2.6 |
| 18:38.55 | smackus | so i am dialing *1 from extension 3562. I want it to log in Local/3562, not *1... what am I doing wrong? http://pastebin.ca/126707 |
| 18:39.18 | cybertrickle | file, ko then ?? |
| 18:39.32 | file | ko for 2.6, o for 2.4 I believe - but I haven't been on a 2.4 system in a longggg time |
| 18:39.40 | brodiem | that is correct |
| 18:39.48 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 18:41.06 | cybertrickle | these ko files go in /usr/lib/asterisk/modules, correct ?? |
| 18:41.11 | sb_mx | smackus, you want it to go to Local/{the-extension-you-dialed-from} right? |
| 18:41.18 | smackus | correct |
| 18:41.31 | file | cybertrickle: no, those are for Asterisk modules |
| 18:41.34 | sb_mx | smackus, instead of {CHANNEL} use {CALLERID(num)} |
| 18:41.42 | cybertrickle | file, where then ? |
| 18:41.43 | file | cybertrickle: all you should need to do for zaptel is make and make install, and the modprobe to load the modules |
| 18:41.54 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.74) |
| 18:42.01 | file | provided your install is capable of building kernel modules |
| 18:42.09 | cybertrickle | file, modprobe cant find them. But I do see them in the zaptel folder. |
| 18:42.39 | cybertrickle | modprobe says "FATAL: Module wct4xxp not found. |
| 18:42.39 | cybertrickle | " |
| 18:42.45 | file | did you do make install? |
| 18:42.49 | cybertrickle | yeah |
| 18:43.18 | file | what distro? |
| 18:44.29 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 18:44.29 | *** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net) |
| 18:45.00 | cybertrickle | file, Fedora Core 5 |
| 18:45.24 | file | have you called technical support to get your installation support? they know the tips'n'tricks for the different distros |
| 18:45.27 | ki2k | whats wrong w/ fedora? |
| 18:45.42 | cybertrickle | file, no I have not. |
| 18:46.10 | yojanl | hi im back, anyone seen my pastbin? |
| 18:46.14 | cybertrickle | file, if I copy the .ko files manually to the kernel modules folder I get the error "WARNING: Error inserting zaptel (/lib/modules/2.6.15-1.2054_FC5smp/zaptel.ko): Invalid module format" |
| 18:46.25 | cybertrickle | file, Ever seen that error before ?? |
| 18:47.11 | file | I don't use FC5, just Debian but people may have had that problem before... it would be on voip-info.org, the mailing lists, or wherever |
| 18:47.19 | file | and technical support has problem encountered it with someone |
| 18:47.26 | smackus | ok, CALLERID(num) will not work either, cuz its a call center.. all of the caller ids are set to the same number. i needs to be extension. |
| 18:48.19 | smackus | there has to be a variable that is for the device called from but what I have done it does it as SIP/ I need it to be local. |
| 18:49.16 | sb_mx | smackus, this is for inbound or outbound calls |
| 18:49.35 | sb_mx | smackus, or internal calls for that matter (ie: apps, ext2ext, etc) |
| 18:49.36 | smackus | inbound. using channel I can get it this close: Added interface 'Local/SIP/3562-0de3' |
| 18:49.47 | smackus | it is an inbound queue |
| 18:50.12 | sb_mx | smackus, and instead of SIP/3562-0de3 you only want the 3562? |
| 18:50.20 | file | smackus: setvar in sip.conf for each device, export a variable to the dialplan which is like the Local string to reach them or something |
| 18:50.57 | smackus | hmmm... ok interesting. |
| 18:51.02 | smackus | lemme try a few things. |
| 18:51.15 | file | setvar=MY_AGENT_CALLBACK=100@agents |
| 18:51.26 | file | AddQueueMember(support,Local/${MY_AGENT_CALLBACK}) |
| 18:51.58 | yojanl | I get this: Via: SIP/2.0/UDP [[my wan ip]]:6050;branch=z9hG4bKde22e969cb66db45;received=[[my wan ip]];rport=33297 |
| 18:52.09 | sb_mx | cybertrickle, do you get an error message when compiling zaptel? |
| 18:52.12 | file | it's a Via header, determines the path the packet took |
| 18:52.24 | file | packet... SIP message... meh |
| 18:52.35 | cybertrickle | sb_mx, nope |
| 18:52.36 | yojanl | what is the last rport? Maybee this is my problem, in a succesfull request the last rport is the same as the first port |
| 18:53.27 | yojanl | successfull: Via: SIP/2.0/UDP [[wan ip]]:6100;branch=z9hG4bKbb378c42e1365306;received=[[wan ip]];rport=6100 |
| 18:55.24 | yojanl | can I change a setting in my grandstream so it will trigger a differen rport? |
| 18:56.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:57.24 | smackus | file: so the setvar=MY_AGENT_CALLBACK would be done in the extensions.conf? or on each entry in the sip.conf? |
| 18:57.31 | file | sip.conf |
| 18:57.35 | smackus | awesome, thanks |
| 19:02.12 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 19:03.43 | Akhilesh | I need some small help |
| 19:04.13 | SkramX | Interesting.. a client wants me code a web interface and asterisk type script to work with http://liarliar.sourceforge.net/ .. |
| 19:04.22 | SkramX | could be cool... could be super hard |
| 19:04.31 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:04.39 | file | Akhilesh: asking a question goes a long way towards getting help |
| 19:05.09 | Lyfe | anyone have any ideas why i can receive a call out of a standard fxo card but can't place one? |
| 19:05.10 | Akhilesh | Since I am testing on a single machine, I can use only one softphone. But to test, is there anything I can do ? |
| 19:05.11 | Rawplayer | how does * perform on sparc64 compared to x86? |
| 19:05.13 | *** join/#asterisk ionix (n=ionix@p3101-ipbfp05miyazaki.miyazaki.ocn.ne.jp) |
| 19:05.23 | Akhilesh | Can I hear my own voice after some delay ? Like delayed echo ? |
| 19:05.28 | *** join/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131) |
| 19:06.41 | ArkonaDev-Mike | Anyone played with the IRC server imbedded into Fonality's Asterisk solution? |
| 19:06.55 | [TK]D-Fender | Delayed echo.. is that like a redundant repetition? ;) |
| 19:07.07 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 19:07.27 | Akhilesh | I mean, I want to setup a SIP user , to whom if I call I should hear back after some delay, say 3-4 seconds. |
| 19:08.01 | [TK]D-Fender | Akhilesh: Depends on the latency of the connection for Sip>SIP |
| 19:08.24 | Akhilesh | I wanted to test how it works... I am a total newbie to this voice stuff. |
| 19:08.31 | Akhilesh | I have setup two users |
| 19:08.51 | c4t3l | hello |
| 19:09.02 | Akhilesh | I registeresed on sip and connected using x-Lite. |
| 19:09.09 | Akhilesh | *registered |
| 19:09.56 | Akhilesh | Then, I dialed some random number, it said "The person u r calling is unavailable. Please try again." |
| 19:10.18 | [TK]D-Fender | Akhilesh: Means that you are not dialing a valid # |
| 19:10.48 | [TK]D-Fender | Akhilesh: that message is created by X-Lite, not # |
| 19:12.56 | Akhilesh | # ? Whats that ? |
| 19:13.05 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 19:14.32 | ArkonaDev-Mike | Anyone ever user Fonality's HUD software? |
| 19:16.39 | NDT | ArkonaDev-Mike: Saw screenshots never had any use for it though so never tried it |
| 19:16.44 | Akhilesh | oops... my context in sip.conf was wrong :) |
| 19:16.52 | Akhilesh | Thanks D-fender. |
| 19:17.05 | file | [TK]D-Fender gets a cookie! |
| 19:18.20 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 19:18.22 | [TK]D-Fender | arf! |
| 19:20.57 | [TK]D-Fender | Meant to say * on the 2nd one there... |
| 19:21.02 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:21.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:21.20 | [TK]D-Fender | anthm: Killed that peer good you did! |
| 19:21.30 | anthm | lol |
| 19:22.09 | cybertrickle | I got past my zaptel issues. I upgrade the kernel |
| 19:22.10 | [TK]D-Fender | Thats the best part about my God complex.... no peer pressure :D |
| 19:22.30 | Lyfe | anyone have any ideas why i can receive a call out of a standard fxo card but can't place one? (hopefully?) |
| 19:22.30 | cybertrickle | Has anyone ever had issues doing asterisk -r after its running ? |
| 19:22.45 | [TK]D-Fender | Lyfe: Show use your dialplan. |
| 19:22.55 | Lyfe | exten => _91NXXNXXXXXX,n,Dial(Zap/1/${EXTEN:1}) |
| 19:22.55 | Lyfe | exten => _91NXXNXXXXXX,n,Hangup |
| 19:22.55 | Lyfe | exten => _91NXXNXXXXXX,n+100,Hangup |
| 19:22.56 | [TK]D-Fender | us8 |
| 19:22.58 | [TK]D-Fender | us* |
| 19:23.08 | [TK]D-Fender | Lyfe: PASTEB IT. ALL OF IT |
| 19:23.19 | [TK]D-Fender | ~pb |
| 19:23.22 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 19:23.24 | [TK]D-Fender | dfsfhjvhnhnfasdlkjnhkdrnuqwe9rityvnioufytvg |
| 19:23.26 | [TK]D-Fender | BLARG |
| 19:23.31 | Lyfe | woops, forgot the noop on the top. |
| 19:23.51 | mog | hey i have a question |
| 19:23.52 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 19:23.57 | mog | anyone here know stuff about hints |
| 19:24.03 | Lyfe | d-fender: is there a preferred place to paste 5 lines? |
| 19:24.06 | [TK]D-Fender | mog : I do, what of them? |
| 19:24.11 | *** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 19:24.13 | file | mog: I'll give you a hint... |
| 19:24.18 | mog | ooh thanks |
| 19:24.19 | [TK]D-Fender | Lyfe: pastebin.ca |
| 19:24.37 | file | ut roh |
| 19:24.47 | [TK]D-Fender | Yeah file.... you work there... they have exclusive rights to you! |
| 19:25.40 | Lyfe | d-fender: http://pastebin.ca/126750 |
| 19:25.48 | brif8 | I have a remote SIP phone (cisco 7960) which has registered with the local * system, as ssen by sip show peers. (1) the phone can dial an extension on the * box and place an outbound call. (2) The phone will ring when it's extension is dialed but NO voice on incoming calls. and I get "SIP response 481 "Call Leg/Transaction Does Not Exist" What am I missing |
| 19:26.16 | shmaltz | Lify, change the dial line to read: |
| 19:26.18 | shmaltz | _91NXXNXXXXXX,n,Dial(Zap/1/ww${EXTEN:1}) |
| 19:26.39 | Lyfe | ww? interesting. |
| 19:27.25 | shmaltz | Lyfe, it inserts a pause, so that dialtone is present before it dials. |
| 19:28.51 | shmaltz | Lyfe, did that fix the problem? |
| 19:29.02 | brodiem | anyone recommend a good CRM w/ easy * integration? |
| 19:29.17 | Lyfe | i need a moment, to test this better, cause now it's sitting there, apparently. |
| 19:29.19 | *** join/#asterisk aydiosmio (n=aydiosmi@65.213.70.43) |
| 19:29.31 | shmaltz | brodiem, define easy |
| 19:30.04 | aydiosmio | you should use the User Logon to join an extension to a queue right? My phone keeps saying my extension is unavailable? |
| 19:30.38 | shmaltz | aydiosmio, your phone? then take it up with the manufacturer of your phone |
| 19:30.57 | aydiosmio | I mean the extension, asterisk reports it |
| 19:31.06 | shmaltz | ohic |
| 19:31.16 | shmaltz | aydiosmio, what does *asterisk* say? |
| 19:31.30 | aydiosmio | phone 200 is currently unavailable |
| 19:31.39 | yojanl | okay, one try again. I have a GXP-2000 and in this network I can receive calls but I cannot place calls, I have tried an other gxp-2000 here which works in a different network but i doesnt work here, the strange thing is that in this network if I try to call with firefly_thirdparty in the failing network i can place calls. The problem lays I think in the GXP with the Network here. This combination fails. What can I check? |
| 19:31.43 | shmaltz | aydiosmio, thank you, pb |
| 19:31.46 | shmaltz | ~pb |
| 19:31.48 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 19:31.50 | *** join/#asterisk e-MAC (n=YMironek@82.207.61.237) |
| 19:31.50 | [TK]D-Fender | aydiosmio: Clarify what he perceives as "unavailable". |
| 19:32.38 | yojanl | so outgoing is failing and I cant figure out why, incoming works like a charm |
| 19:32.46 | shmaltz | yojanl, what does your network topology look like? |
| 19:32.59 | shmaltz | yojanl, pb your sip show peers |
| 19:33.18 | aydiosmio | the featurecode plays back is-curntly-unavail.gsm |
| 19:33.22 | aydiosmio | that's it |
| 19:33.36 | shmaltz | aydiosmio, that means that asterisk is operating nicely |
| 19:33.38 | shmaltz | :P |
| 19:33.39 | aydiosmio | my softphone is registered and I can dial to it |
| 19:33.42 | yojanl | the asterisk server is on a private ip, then I have the wan with a router which forwards the udp ports to the static interal ip of the grandstream |
| 19:34.02 | shmaltz | yojanl, again your sip show peers |
| 19:34.02 | brif8 | For those who know sip debug here is the SIP debug peer 650 I get http://channels.debian.net/paste/3409 |
| 19:34.02 | Akhilesh | how to read voice mail >? |
| 19:34.07 | [TK]D-Fender | aydiosmio: Show use the CLI output leading to this error.... |
| 19:34.14 | Akhilesh | I dialed 8500, then it asked for my mailbox. |
| 19:34.24 | [TK]D-Fender | Akhilesh: "show application voicemailmain" |
| 19:34.49 | Akhilesh | I pressed 5190 as in sip.conf, I have written mailbox=5190 |
| 19:35.09 | Akhilesh | Then it asks for password, how do I dial password as my password is teststring ! |
| 19:35.32 | yojanl | http://pastebin.ca/126768 |
| 19:35.38 | [TK]D-Fender | Akhilesh: that makes no sense... its what you put in voicemail.conf |
| 19:35.56 | Akhilesh | nothing |
| 19:35.59 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 19:36.18 | shmaltz | yojanl, which one is the gpx? |
| 19:36.21 | Akhilesh | oops, realized my mistake. |
| 19:36.25 | [TK]D-Fender | Akhilesh: Go read up on how to configure your mailbox in the first place.... |
| 19:36.30 | yojanl | its the zs1 |
| 19:36.32 | Akhilesh | Thanx D-Fender. |
| 19:36.34 | yojanl | and the zs2 |
| 19:36.56 | [TK]D-Fender | Akhilesh: Great idea to DL the book first. It will show you the basics you may have skipped. |
| 19:36.57 | [TK]D-Fender | ~book |
| 19:36.59 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:37.06 | shmaltz | yojanl, on a different network, by that you mean to this asterisk, or another asterisk? |
| 19:37.42 | Lyfe | shmaltz: unfortunately, adding the ww in front of the number did not solve it. |
| 19:37.48 | yojanl | line 1 & 2, in my dialplan I can dial the sipura without problems with firefly_thirdparty in this network, but when I call in this network with the grandstream it fails |
| 19:37.56 | shmaltz | Lyfe, what do you get on the line? |
| 19:38.02 | aydiosmio | oh, I'm not set up as an agent. |
| 19:38.02 | shmaltz | and what does the CLI report? |
| 19:38.24 | Lyfe | i get a fast busy signal, and the only thing i see from the CLI is what's in the second part of the pastebin url i gave. |
| 19:38.26 | shmaltz | yojanl, can you read and answer what you are asked? |
| 19:38.49 | yojanl | shmaltz: asterisk is an external network |
| 19:39.16 | yojanl | but on a public IP |
| 19:39.17 | shmaltz | yojanl, thats *not* what I asked last |
| 19:39.27 | Lyfe | ('cept that now there's "ww" in front of the phone numbers) |
| 19:39.28 | shmaltz | yojanl, on a different network, by that you mean to this asterisk, or another asterisk? |
| 19:39.44 | yojanl | i have only 1 asterisk |
| 19:39.55 | yojanl | that asterisk is on a public internet server |
| 19:40.02 | yojanl | i am behind a router |
| 19:40.10 | shmaltz | yojanl, so those phones work on a different network with the *same* asterisk? |
| 19:40.13 | yojanl | yes |
| 19:40.36 | shmaltz | yojanl, what does sip debug show? |
| 19:40.52 | shmaltz | do a sip debug peer zs1, try making a call from zs1, then pb the cli |
| 19:41.12 | yojanl | k |
| 19:41.13 | yojanl | hold on |
| 19:41.33 | brodiem | shmaltz, didn't see your response.. easy as in I don't have to develop my own code to listen for manager API events |
| 19:41.40 | ki2k | anyone know what docs digium gives you when you buy their business package? what's in those neat binders? |
| 19:41.53 | shmaltz | brodiem, try FOP |
| 19:42.12 | Lyfe | wish i was simply bored.. i'm merely confused. |
| 19:42.20 | brodiem | i know FOP can launch a CRM, but isn't a CRM itself |
| 19:42.25 | shmaltz | ki2k, if you don't know how to read they wont help you, if you do know how to read then you don't need them :P |
| 19:42.28 | brodiem | and I don't like FOP :) |
| 19:43.22 | ki2k | shmaltz: huh? |
| 19:43.32 | ki2k | shmaltz: i just wanna know what they're bundling |
| 19:43.41 | shmaltz | ki2k, I got no clue |
| 19:43.50 | ki2k | they call it "Asterisk Technical Manual" |
| 19:43.57 | ki2k | and a quickstart guide |
| 19:44.02 | ki2k | http://www.digium.com/en/products/software/abe.php |
| 19:44.13 | yojanl | http://pastebin.ca/126778 |
| 19:44.53 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 19:45.29 | GerbilWrk | Anyone familiar with the Grandstream Handytone 286/386 or 488 products? |
| 19:46.10 | shmaltz | yojanl, you are having nat issues |
| 19:46.23 | shmaltz | yojanl, what does your sip.conf look like? |
| 19:46.45 | *** join/#asterisk TK9 (n=Miranda@p54B28EA9.dip0.t-ipconnect.de) |
| 19:47.53 | shmaltz | yojanl, pb your sip.conf |
| 19:48.15 | blitzrage | if I perform a SetGroup for a particular extension, is it possible for me to perform a CheckGroup() for that Group I just set from a different extension? |
| 19:48.34 | *** part/#asterisk TK9 (n=Miranda@p54B28EA9.dip0.t-ipconnect.de) |
| 19:48.39 | blitzrage | I don't think that's going to work because SetGroup just sets a channel variable... which is what CheckGroup is probably looking for... |
| 19:48.52 | yojanl | http://pastebin.ca/126784 |
| 19:49.02 | shmaltz | blitzrage of course |
| 19:49.28 | blitzrage | the easy way for me to do what I need is to use the Local channel, but of course thats no available in ABE 'A' |
| 19:49.58 | *** join/#asterisk kchrist (n=kwilson@bridalveil.istep.com) |
| 19:50.22 | shmaltz | yojanl, what type of router do you have there? |
| 19:50.39 | blitzrage | the issue is basically that I need a method to determine if a phone has a channel in use, and if so, not to include it in the group I'm dialing from an extension |
| 19:50.43 | *** part/#asterisk kchrist (n=kwilson@bridalveil.istep.com) |
| 19:50.54 | shmaltz | yojanl, I would disable the port forwarding to the zs1, it shouldn not be needed |
| 19:50.56 | Lyfe | anyone have any ideas for why I can receive calls from a zap channel, but cannot place calls on that zap channel: http://pastebin.ca/126750 ? I get fast-busy, that apparently is generated by asterisk. |
| 19:51.05 | shmaltz | blitzrage, use chanisavail |
| 19:51.27 | yojanl | smaltz: dont know for sure, its a corparte one, they have forwarded the ports i wanted to the static ip of the grandstream |
| 19:51.30 | yojanl | and incoming works fine |
| 19:51.41 | blitzrage | shmaltz: does that not just check to see if a channel is available (i.e. registered) and not whether it is using a line or not? |
| 19:51.44 | yojanl | I guess theres a win nt server somewhere |
| 19:51.48 | shmaltz | yojanl, it appears that that might be the problem |
| 19:52.15 | shmaltz | blitzrage, nope I use it for ecactly what you want to use it, I use it in my page context for app_page |
| 19:52.24 | yojanl | what do you think it can be? i can call the corporate network guy but I have to know what he has to change |
| 19:52.48 | yojanl | if you want i can post a succesfull outgoing call via a sip soft client on my laptop in the same network as the non functioning grandstream |
| 19:52.50 | shmaltz | yojanl, the firefly also has ports forwarded? |
| 19:53.17 | shmaltz | does that softphone have the ports forwarded? |
| 19:53.33 | yojanl | i tried on a random port and one of the forwarded ports, and in both situations firefly works for outgoing, incoming I havent tested |
| 19:54.02 | yojanl | sorry, the forwarded ports are on the grandstream so i tried it only on ports which havent been configured to point to my laptop |
| 19:54.10 | shmaltz | yojanl, change the ports of the gpx to one thats *not* forwarded |
| 19:54.17 | blitzrage | shmaltz: thanks -- unfortunately I'm building a DP for someone running ABE, and of course I don't have a copy of it, so I'm not exactly sure which features it actually has :) |
| 19:54.34 | shmaltz | blitzrage, doesn't Digium do that? |
| 19:54.43 | yojanl | shmalz: i already tried this, but it gives the same result. Just to be sure I can try again on a very random port, hold on |
| 19:56.56 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 19:56.58 | jhiver | hi all |
| 19:57.20 | jhiver | i have a strange issue: this morning my PAP2-NA was working fine, sending and receiving calls |
| 19:57.34 | jhiver | tonight, when i try to send calls, i see this on the asterisk CLI |
| 19:57.37 | jhiver | Aug 11 23:57:28 NOTICE[90384]: chan_sip.c:10372 handle_request_invite: Failed to authenticate user <sip:jhiver@lcr.ykoz.net>;tag=b1680c61c06a4e9fo0 |
| 19:57.52 | yojanl | schmalts: gives the same result |
| 19:58.15 | jhiver | any ideas what's going on? |
| 19:58.38 | yojanl | its so strange that only the grandstream has issues and not the firefly |
| 19:58.51 | yojanl | are the special settings on a win nt server that have to be set? |
| 19:59.30 | shmaltz | yojanl, I'm sure, but I can't tell you unless you tell me what type of router/firewall you have |
| 19:59.55 | yojanl | im trying to find out, but I can only telnet and it gives no info :-( |
| 20:00.05 | yojanl | and i dont have the telnet pswd |
| 20:00.25 | shmaltz | yojanl, looks like a cisco to me |
| 20:00.42 | shmaltz | yojanl, tell the netadmin (if it's cisco that is) to do a no fixup sip |
| 20:00.42 | yojanl | theres no http connection possible |
| 20:01.55 | yojanl | shmaltz: thank you very very much, im not a beginner in asterisk but this thing has got me troubled. I hate it when i cannot fix issues like this! |
| 20:02.05 | *** part/#asterisk neo (n=neo@kessel.ordrejedis.net) |
| 20:02.14 | shmaltz | np anytime |
| 20:02.56 | ki2k | anyone have a clue what error 423 means? |
| 20:03.05 | ki2k | interval too brief |
| 20:03.09 | yojanl | shmaltz: what exactly does no fixup sip do? you think it defaults to a different ip? |
| 20:03.32 | yojanl | and, sorry if i want to know too much, how come firefly has no troubles with this? |
| 20:03.42 | yojanl | is it a specific grandstream issue? |
| 20:03.46 | shmaltz | yojanl, it's a cisco firewall feature that does some things with sip traffic |
| 20:03.51 | shmaltz | nope |
| 20:03.56 | nDuff | ...not that grandstreams don't have plenty of issues. |
| 20:07.24 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:07.38 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-9.getnet.net) |
| 20:09.39 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:09.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:09.50 | [TK]D-Fender | blitzrage: "show application chanisavail" |
| 20:11.07 | shmaltz | blitzrage, you want to use it with the s option |
| 20:12.46 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:12.50 | Lyfe | anyone have any ideas for why I can receive calls from a zap channel, but cannot place calls on that zap channel: http://pastebin.ca/126750 ? I get fast-busy, that apparently is generated by asterisk. |
| 20:12.56 | [TK]D-Fender | bi bi :( |
| 20:13.11 | Lyfe | (yes, i know i'm repeating myself, but i'm not finding any luck on the web, either) |
| 20:14.00 | [TK]D-Fender | Lyfe: tried plugging a normal phone on that jack and dialing the #? Also please pastebin your zapata.conf |
| 20:14.22 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
| 20:14.28 | Lyfe | yes, the line has been tested. |
| 20:14.30 | dos000 | howdy |
| 20:16.00 | shmaltz | Lyfe, plug in a phone while asterisk goes offhook, listen to make sure you get all the 11 digits, also, if you have a butset try using the moniter to make sure that asterisk is not dialing to soon after going off hook, also try using relaxtdmf |
| 20:16.16 | shmaltz | Lyfe, is this a tdm400? |
| 20:16.33 | Lyfe | nope, it's a cheapo 1-port card. |
| 20:16.53 | shmaltz | hmmmmm, so I don't realy know, |
| 20:17.03 | Lyfe | and, since i'm 80mi away, i can't really plug a phone in and test it :\ |
| 20:17.43 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 20:17.45 | Lyfe | i had the guys at the datacenter test the line, but they might not take to well to finding a splitter, and listening while i try to dial. |
| 20:17.46 | shmaltz | Lyfe, why gas is soooooooooooo expensive :P |
| 20:18.08 | Lyfe | because visiting there once a month is cheaper than hauling t1 lines out here. |
| 20:18.11 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 20:18.27 | Lyfe | not to mention the power & cooling requirements. |
| 20:18.49 | shmaltz | Lyfe, where are you located? |
| 20:19.02 | Lyfe | in the worst possible place to buy a t1 for internet connectivity :P |
| 20:19.30 | shmaltz | Lyfe, IRAQ? |
| 20:19.34 | *** join/#asterisk MindHack (n=mindhack@unaffiliated/mindhack) |
| 20:19.48 | Lyfe | ok, guess i'm wrong about that. |
| 20:19.59 | Lyfe | rockford, IL, USA. |
| 20:20.00 | shmaltz | LOL |
| 20:20.13 | shmaltz | and the colo? Chicago? |
| 20:20.24 | Lyfe | milwaukee, actually. |
| 20:20.29 | Lyfe | prior contacts. |
| 20:20.39 | shmaltz | Milwaukee, is in IL? |
| 20:20.45 | Lyfe | sorry, milwaukee, WI. |
| 20:20.50 | shmaltz | oic |
| 20:21.10 | websae | lyfe: you live in milwaukee? |
| 20:21.17 | Lyfe | nope. |
| 20:21.20 | *** join/#asterisk Trakkasure (n=Sgemtum@69-163-145-203.atlsfl.adelphia.net) |
| 20:21.36 | Lyfe | we just have contacts with a company that we like who's datacenter is in milwaukee. |
| 20:22.06 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-6.getnet.net) |
| 20:24.16 | Lyfe | anyway.. getting back to it, i was asked to pastebin zapata.conf, and i swear that the only thing useful out of it (given that the [general] section is default, and the other sections are commented out) is this section that's in the url i pasted already. |
| 20:26.51 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:26.59 | shmaltz | gtg, cya guys |
| 20:27.04 | shmaltz | enjoy ur weekend |
| 20:28.24 | Lyfe | anyway..s ince it's requested: http://pastebin.ca/126808 |
| 20:30.17 | *** join/#asterisk zpertee (n=zach@cpe-65-25-51-117.neo.res.rr.com) |
| 20:30.57 | zpertee | hey does anyone have a moment to answer a couple of questions about g729 codecs |
| 20:30.58 | [TK]D-Fender | Lyfe: that is not zapata.conf |
| 20:31.06 | Lyfe | oh, my mistake... i read wrong. |
| 20:31.23 | eKo1 | zpertee: what ? |
| 20:31.55 | zpertee | eKo1, i have 2 channels of g729 but how do i support that and regular gsm |
| 20:32.20 | zpertee | eKo1, i want it to default to g729 |
| 20:32.22 | Lyfe | d-fender: my apologies, i've got too many dumb things going on around me right now and it's affecting my brain. |
| 20:32.46 | eKo1 | zpertee: disallow => all and allow => g279 |
| 20:33.08 | zpertee | eKo1, ok thank you so much for your time. I truly appreciate it. |
| 20:33.13 | Lyfe | d-fender: http://pastebin.ca/126811 |
| 20:34.43 | *** part/#asterisk zpertee (n=zach@cpe-65-25-51-117.neo.res.rr.com) |
| 20:34.59 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 20:36.11 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 20:36.58 | *** join/#asterisk sxpert (n=sxpert@vau75-1-81-57-130-155.fbx.proxad.net) |
| 20:37.02 | sxpert | heya |
| 20:37.55 | *** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 20:45.24 | sxpert | anyone home ? |
| 20:45.31 | *** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 20:45.32 | Lyfe | there used to be. |
| 20:45.48 | Lyfe | hell, if my brain was working, i might've even gotten a response to my initial inquiry. |
| 20:46.47 | NDT | was just a split they will be back |
| 20:47.24 | Lyfe | there was? |
| 20:56.01 | MindHack | Im with an organization looking to have more effective conferance calls. Skype is proving to be quite limiting, and quality is really bad when communicating with people overseas. |
| 20:56.18 | hmmhesays | i'm not home |
| 20:56.24 | hmmhesays | asterisk contract? |
| 20:56.38 | MindHack | Is it possible, or sensible, to use asterisk to only communicate pc to pc? With perhaps only one node actually dialing out? |
| 20:58.01 | hmmhesays | why not? |
| 20:58.09 | sxpert | hmmhesays: yeah.. some old friend of mine requests me to create and invoice a brand new system for him ; |
| 20:58.10 | sxpert | ;D |
| 20:59.00 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:59.29 | Lyfe | so, all that pasting and stupidity over a comma. |
| 21:00.01 | sxpert | ok, so what should I use for the base PC (I'll only be using G711) ? AMD64 or intel ? |
| 21:00.22 | Lyfe | or something. |
| 21:01.11 | Lyfe | nevermind, guess not. |
| 21:01.19 | hmmhesays | sxpert: I see |
| 21:01.40 | benjk | I'd say WRAP or Soekris board |
| 21:01.42 | hmmhesays | so we'll see you in here on ${install_date} + 1 |
| 21:01.50 | hads|home | heh |
| 21:01.53 | hmmhesays | half bald |
| 21:01.55 | sxpert | benjk: nah, it needs to look serious ;D |
| 21:01.56 | hmmhesays | no sleep |
| 21:02.17 | hmmhesays | I see quintum fixed their asterisk prolem |
| 21:02.20 | benjk | an embedded system looks a bazillion times more serious than a foolish desktop toy box |
| 21:02.22 | sxpert | benjk: it will need to handle like 4 E1 plus some analog ports for testing / monitoring |
| 21:02.41 | sxpert | benjk: that'd be a rack machine ;D |
| 21:03.02 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:03.07 | benjk | fair enough, I'd still settle for four embedded boxes though if it doesn't need to transcode |
| 21:03.26 | sxpert | it will need to do multiple people conferencing with lots of conferences, something like 24 conferences of 5 people |
| 21:03.39 | benjk | ok |
| 21:04.29 | hmmhesays | yep sxpert you will be pulling your hair out |
| 21:04.49 | sxpert | hmmhesays: heh |
| 21:04.50 | sxpert | lol |
| 21:05.04 | hmmhesays | this your first install? |
| 21:05.11 | sxpert | yeah |
| 21:05.22 | hads|home | Good luck then. |
| 21:05.26 | sxpert | lol |
| 21:05.27 | Lyfe | he's right then, you will be. |
| 21:05.48 | sxpert | obviously, everything *will* go wrong ;D |
| 21:05.57 | hads|home | 4 E1's is definitly throwing yourself in the deep end. |
| 21:06.00 | Lyfe | <PROTECTED> |
| 21:06.26 | sxpert | heh, we will start with 1 E1 anyhow... then add more E1s if the service is successful |
| 21:07.41 | sxpert | the E1s will either come from Completel or Colt |
| 21:09.33 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 21:14.07 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 21:14.32 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 21:15.31 | fiber0pti | I'm using the manager API to create a small java application. I'm trying to take a channel and transfer it directly to voicemail but I don't know how I can do it. I can transfer to other extensions, how can I transfer directly to a specified mailbox? |
| 21:16.45 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 21:18.12 | eKo1 | First of all, you can't use the manager API to create anything. |
| 21:18.23 | SplasPood | Hey.. Can anyone suggest to me how I might be able to hook Cepstral up to Asterisk in such a way so that I could send text to Cepstral via some external source and have it speak that text via an existing call.. Ie, I want to make something where one party of the call can be typing and doing TTS |
| 21:18.32 | eKo1 | Second of all, you just need to make your channel dial into a context that will enter voicemail. |
| 21:19.55 | eKo1 | SplasPood: There is no built-in functionality that will allow that. |
| 21:20.03 | eKo1 | You're going to have to hack it out. |
| 21:20.23 | SplasPood | eKo1: I know, I'm more than willing to do that |
| 21:20.40 | SplasPood | eKo1: just looking for suggestions |
| 21:20.51 | SplasPood | eKo1: Ideally I'd like to avoid coding app_mycepstral.c |
| 21:21.00 | eKo1 | app_cepstral already exists. |
| 21:21.06 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.205) |
| 21:21.09 | SplasPood | yes, but not the way I need it |
| 21:21.16 | eKo1 | Then hack it. |
| 21:21.24 | eKo1 | You can't avoid coding. |
| 21:21.34 | SplasPood | can an AGI be used that'd call Cepstral() app_cepstral when it got events from some external source? |
| 21:21.39 | quid246 | I wish Cepstral had a sultry voice... then I could develop a "dirty talk phone service" based on AI |
| 21:21.49 | SplasPood | eKo1: I can try and avoid coding C which is not my area |
| 21:21.58 | eKo1 | quid246: hehehe |
| 21:22.13 | eKo1 | SplasPood: Not your area? Then what is your area? |
| 21:22.22 | SplasPood | Just not my language |
| 21:22.29 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 21:22.37 | eKo1 | I take it you're not a programmer. |
| 21:22.45 | eKo1 | Your AGI idea seems feasable. |
| 21:22.45 | SplasPood | Not by trade, no |
| 21:23.00 | SplasPood | Ok |
| 21:23.31 | benjk | SplasPood, you can still use app_cepstral |
| 21:23.57 | benjk | Cepstral(${TEXT}) |
| 21:24.41 | benjk | and do asterisk -rx database put tts text "foo bar baz" |
| 21:24.52 | benjk | actually with quotes |
| 21:25.23 | benjk | asterisk -rx "database put tts text \"foo bar baz\"" |
| 21:25.28 | fiber0pti | eKo1: How would I specify the extension that I want to transfer too? |
| 21:26.10 | benjk | then have a small macro that does DBget to read the text from astdb and stick it into ${TEXT} |
| 21:26.42 | eKo1 | fiber0pti: Are you using the raw manager interface? |
| 21:26.44 | [TK]D-Fender | fiber0pti : Make an extens for each mailbox... |
| 21:27.06 | fiber0pti | eKo1: I'm using the asterisk-java API |
| 21:27.09 | benjk | so, no c coding needed, only a handful of dialplan |
| 21:27.20 | benjk | no AGI needed either |
| 21:27.26 | eKo1 | fiber0pti: sorry, I'm not familiar with that one. I work in raw mode only. |
| 21:27.28 | [TK]D-Fender | fiber0pti : exten => #1000,1,VoiceMail(u1000@default) , etc.... |
| 21:27.52 | GerbilWrk | Has anyone been sucessful sending CallerID with Voxee.com termination services? |
| 21:27.58 | fiber0pti | [TK]D-Fender: I understand what you're saying but I don't understand how to integrate that into the application I'm developing |
| 21:28.25 | fiber0pti | [TK]D-Fender: How can I dynmically pass the extension? |
| 21:28.43 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:29.01 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 21:30.35 | GerbilWrk | Anyone at all? We just get Unknown for the callerid with the Voxee service, it's kinda annoying |
| 21:31.16 | TrixVox | Name or number? |
| 21:31.44 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 21:32.01 | mitcheloc | GerbilWrk: voxee should provide free support to their customers |
| 21:32.02 | supjigatr | Hello. |
| 21:32.22 | GerbilWrk | mitcheloc, they do, through trouble tickets that will take a while, i'm trying to get something quick, just in case |
| 21:32.34 | TrixVox | No phone support? |
| 21:32.40 | GerbilWrk | TrixVox, all my cell phone shows is "Unknown" |
| 21:33.14 | GerbilWrk | no phone support apparently |
| 21:33.46 | quid246 | Well, even if you set it... it might not show depending on who in the termination pool Voxee connects to. |
| 21:33.52 | benjk | they probably have phne support but they don't accept calls from "Unknown" callers :) |
| 21:34.04 | supjigatr | Anyone have tips on troubleshooting Credit card and fax machines not working from FXS ZAP to PRI Zap bridge? |
| 21:34.39 | TrixVox | just use VoicePulse, CID works perfectly |
| 21:34.46 | eKo1 | supjigatr: faxing is not reliable |
| 21:34.51 | websae | VoicePulse ~ yuck |
| 21:34.59 | websae | my opinion |
| 21:35.08 | websae | faxing, just use a third party for that... |
| 21:35.10 | TrixVox | yeah, because you're a competitor |
| 21:35.31 | supjigatr | eKo1: I think you are refering to spandsp. I have a real fax machine and a few credit card machines on a analog channelbank and they are calling out on the PRI. |
| 21:35.57 | GerbilWrk | and how much does Voicepulse charge per minute? |
| 21:36.04 | supjigatr | I'm not using spandsp. I can't get a zap to zap bridge to work with a modem. |
| 21:36.04 | benjk | who cares |
| 21:36.24 | benjk | rates are low enough not to have to care about fractions of pennies |
| 21:36.33 | eKo1 | Supaplex: The faxes go through * right? |
| 21:36.50 | GerbilWrk | 1 cent a minute, versus 3 cents a minute on 8000 minutes a month, makes a difference |
| 21:37.02 | websae | what's the quality like? |
| 21:37.02 | benjk | if its a little more than elsewhere, it won't hurt, but it ensures that you get better service because all the suckers signed up with the ones that are 0.01 penny cheaper |
| 21:37.12 | websae | benjk: exactly :) |
| 21:37.14 | TrixVox | 0.005 - 0.019 depending on where you call, i average about 0.007 aka 7/10th of a penny |
| 21:37.31 | supjigatr | Eko1: They are a native ZAP bridge. NO IP involved. |
| 21:38.21 | GerbilWrk | TrixVox, and they work with Asterisk, and have unlimited channels at once? |
| 21:38.25 | benjk | and how much time are you now wasting trying to troubleshoot and you don't even get phone support for your 8000 mins |
| 21:38.27 | benjk | pffft |
| 21:38.38 | adorah | do u know where can I get cheap fxs modules for digium tdm400 card? |
| 21:39.07 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 21:39.12 | TrixVox | GerbilWrk: http://connect.voicepulse.com/ , everything is spelled out there |
| 21:39.14 | GerbilWrk | benjk, I JUST signed up for their service to try it out, i don't know how quick their support is, or if they have a phone support line I can reach them at |
| 21:39.28 | TrixVox | not www.voicepulse.com, that's the vonage-like service for home users |
| 21:39.44 | supjigatr | Eko1: I suspect my problem is timing but no real docs on taking timing from port 1 on a 104d and outputting on all other spans synced. |
| 21:39.46 | benjk | nah, you want connect.voicepulse.com |
| 21:40.36 | supjigatr | It seems all the 104d ports are not synced by default and run independant. |
| 21:40.44 | eKo1 | supjigatr: interesting. |
| 21:45.10 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 21:45.23 | TripleFFFF | guys.. what that option to group cdr into one entry |
| 21:45.27 | supjigatr | What I need is port 1 to accept timing, ports 2-4 to sync to 1 and transmit to slave Channel Banks. |
| 21:45.32 | javar | adorah: maybe here http://shop.ifax.com/-c-32_26.html?osCsid=8d5dae981104dce7243f3f6b6663734b |
| 21:45.43 | supjigatr | Has anyone done this with sangoma or digium card? |
| 21:47.26 | [TK]D-Fender | supjigatr : I do it on my A104d |
| 21:48.38 | supjigatr | How do you define timing with the A104d? |
| 21:48.55 | supjigatr | I want to accept timing on Port 1 and sync that with 2,3,4. |
| 21:49.30 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 21:51.41 | GerbilWrk | so TrixVox, do you work for Voicepulse or something? |
| 21:51.48 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:51.53 | [TK]D-Fender | supjigatr :pastebin your configs |
| 21:52.40 | TrixVox | nah, i've tried everyone and they all have their problems, the guys at vp have always picked up the phone when i called and at least made some effort to fix the problem, i'm done with all these bullshit email-only support providers |
| 21:53.01 | GerbilWrk | TrixVox, have you used Teliax before? |
| 21:53.14 | supjigatr | http://pastebin.ca/126892 |
| 21:53.35 | GerbilWrk | TrixVox, and how many times have you seen voicepulse go down? |
| 21:54.29 | TrixVox | it's gone down before, i don't think there's any provider that hasn't... but they answer the phones during an outage and there's a message up on the website pretty quickly with a status |
| 21:55.01 | [TK]D-Fender | supjigatr : and now your wanpipe confis.... |
| 21:55.18 | TrixVox | they used to have this crappy sip server early on that went down all the time (the iax ones didn't), but they upgraded everything a few months ago and it's been smooth sailing since |
| 21:56.15 | GerbilWrk | ok, the plan is to you use two servers for outbound calls, so I doubt they will both be down at the same time |
| 21:56.23 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 21:56.30 | TripleFFFF | guys.. my asterisk is not producing the cdr logs correctly... |
| 21:56.36 | TripleFFFF | just the first thing it does.. |
| 21:56.52 | TrixVox | yeah, you register to two servers for incoming and send outgoing through two servers |
| 21:56.54 | TripleFFFF | so 1,dial,(SIP/blah) |
| 21:56.54 | TripleFFFF | 2,dial(sip/1234123123@host) |
| 21:56.59 | TripleFFFF | will only make cdr for first |
| 21:57.06 | Mattwj2005 | hey guys |
| 21:57.14 | Juggie | TripleFFFF, because 2 will never run |
| 21:57.38 | Juggie | unless 1 fails |
| 21:57.39 | GerbilWrk | well incoming is just an 800 numbers which will forward to our local phone switch in the cities we can accept it, and then come in through junction networks most likely for the rest of the cities |
| 21:57.45 | TripleFFFF | yes but 1 fails |
| 21:58.09 | TripleFFFF | ok i meant.. Look.. i get hmm |
| 21:58.35 | TrixVox | junction's prices are crazy for a newbie service provider |
| 21:58.41 | TripleFFFF | i mean.. it should make a cdr log for every operation.. not just update the cdr record with latest operation |
| 21:58.58 | TripleFFFF | so.. if dial, dial ,dial and voicemail |
| 21:59.02 | TripleFFFF | hmm |
| 21:59.05 | TripleFFFF | ok i get it |
| 21:59.13 | GerbilWrk | well most of our 800 usage is in areas we can get the calls for free, we may spend $10 a month, even with their prices once we get everything squared away |
| 22:01.40 | Mattwj2005 | is it possible to hook up a bluetooth to usb adapter to an asterisk box and call directly from the server? |
| 22:02.03 | *** join/#asterisk NativeOnRye (n=terry_si@206.163.1.131) |
| 22:02.04 | eKo1 | Mattwj2005: call where? |
| 22:02.15 | Mattwj2005 | call anywhere |
| 22:02.20 | eKo1 | and what does the adapter have to do with anything? |
| 22:02.51 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 22:03.07 | Mattwj2005 | well that is how I would enable my server to have bluetooth abilities |
| 22:03.54 | eKo1 | I don't get it. |
| 22:04.41 | Mattwj2005 | I was thinking maybe I could dial out with festival |
| 22:06.36 | supjigatr | Pastebin appears to be locking up. |
| 22:06.58 | eKo1 | Festival does text-2-speech. How will it help you dial out? |
| 22:07.45 | Mattwj2005 | oops....doesn't asterisk have some speech2text abilities? |
| 22:08.51 | TripleFFFF | actyally |
| 22:08.51 | TripleFFFF | <PROTECTED> |
| 22:12.02 | sxpert | should I have lots of memory for an asterisk box, or the minimum is enough ? |
| 22:12.53 | sxpert | thinking either 1G or 2G per processor here (considering how cheap ram is these days) |
| 22:17.35 | eKo1 | 1G is good. |
| 22:18.12 | Mattwj2005 | it is possible to forward calls to a bluetooth headset? |
| 22:19.20 | [TK]D-Fender | Mattwj2005 : Sphinx. |
| 22:19.22 | eKo1 | forward calls to a headset? |
| 22:19.39 | eKo1 | that doesn't make sense |
| 22:19.48 | Lyfe | a guy i work with used his bluetooth headset as an audio device for his softphone on his laptop. |
| 22:20.05 | eKo1 | Lyfe: that makes sense |
| 22:20.14 | Mattwj2005 | well for example lets say someone gives me a call.....I pickup on a softphone on my computer....can I transfer to an extension (bluetooth headset) |
| 22:20.42 | eKo1 | the headset isn't a softphone Mattwj2005 |
| 22:20.58 | eKo1 | or a phone |
| 22:21.11 | eKo1 | I've never seen a voip bluetooth phone |
| 22:21.16 | eKo1 | that would be killer |
| 22:21.47 | Mattwj2005 | I am just trying to understand chan_bluetooth and chan_btp |
| 22:22.40 | *** join/#asterisk fgwaller (n=frank@65.105.5.40.ptr.us.xo.net) |
| 22:24.21 | carrar | My cell phone bluetooth ear piece also works with my laptop as a audio in/out device |
| 22:25.03 | Lyfe | carrar: that's what the guy i work with was doing.. same unit he used for his cell phone. |
| 22:25.28 | Mattwj2005 | yeah that is probably what I'll do....it wanted it to be cool (and geekier) than that |
| 22:25.29 | Mattwj2005 | :) |
| 22:25.30 | TripleFFFF | me |
| 22:25.31 | TripleFFFF | ;) |
| 22:25.42 | TripleFFFF | laptop with spinx + voice recgno + asterisk |
| 22:25.47 | TripleFFFF | + bluettoth |
| 22:25.59 | Mattwj2005 | lol |
| 22:25.59 | carrar | and a hologram imager |
| 22:26.00 | TripleFFFF | so i click.. recognize call home.. and idal home |
| 22:26.01 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 22:26.12 | sxpert | eKo1: ok. should I expect running the stats postgres on a second processor in the same box, or on some other box via the network ? |
| 22:26.35 | Mattwj2005 | and a holodeck and repacator |
| 22:26.38 | carrar | obi one kanobi, you're my only hope! |
| 22:26.46 | Mattwj2005 | :P |
| 22:26.56 | fgwaller | Two call at the same time having the same UniqueId, is that a Bug or a Feature... (colission in PRI channel) |
| 22:27.16 | Mattwj2005 | a Stargate or two would be nice |
| 22:27.22 | Mattwj2005 | lol |
| 22:28.30 | TripleFFFF | theroicaly an atom is in 2-3 ~ places in same time till you look at it.. ( quantum crap says it anyhow) so .. jjust dont look and the object will be elsewhere |
| 22:28.31 | TripleFFFF | ;) |
| 22:28.52 | Mattwj2005 | lol |
| 22:29.15 | *** join/#asterisk _DAW (n=bob@adsl-35-242-196.msy.bellsouth.net) |
| 22:29.28 | Mattwj2005 | wow we are all a bunch of nerds :-P |
| 22:29.29 | _DAW | HI all.. |
| 22:30.04 | eKo1 | TripleFFFF: not atoms, but its constituent particales |
| 22:30.18 | eKo1 | err, particles |
| 22:30.58 | fgwaller | I only understand Quark |
| 22:32.04 | TripleFFFF | maybe its neutrinos also |
| 22:32.58 | eKo1 | and that is due to the probabilistic nature of quantum mechanics |
| 22:33.25 | eKo1 | the position of any particle, say an electron can only be described by a probability density function. |
| 22:34.14 | TrixVox | It's called the Voxee Uncertainty Principle -- there are 2-3 customer service reps, but if you try to find one, they've moved elsewhere |
| 22:34.30 | benjk | TripleFFFF, those particles are doing that on purpose, just to piss you off |
| 22:35.04 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.205) |
| 22:36.05 | sxpert | fgwaller: do you want to play dabbo at quark's bar ? |
| 22:36.52 | _DAW | Has anyone here seen dtmf problems with asterisk over a very high latency circuit (satellite in this case). I have a ds1 to a satellite carrier with an IVR on my * box. When I call in and enter the extension I want to dial I get variations of the digits I dialed. Usually it is truncated. Suggestions? |
| 22:37.16 | sxpert | _DAW: press the dtmf key longer ? |
| 22:38.08 | Lyfe | heh.. very high latency indeed. |
| 22:41.17 | *** join/#asterisk chreese (n=chatzill@bridalveil.istep.com) |
| 22:45.05 | Lyfe | anyone setup a non-pri t1 that can tell me how to get the callerid information from the line (my provider says they're providing as much callerid information as they're receiving about it) |
| 22:45.35 | fgwaller | No, my wife would hire some Klingons to hun me... |
| 22:45.52 | Lyfe | i would think that simply setting "usercallerid=yes" in zapata.conf would solve this, but i eem to be mistaken. |
| 22:46.03 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 22:46.22 | benjk | pri debug is your friend |
| 22:46.43 | benjk | it'll show you the raw PRI messages coming in |
| 22:46.51 | Lyfe | psst.. "non-pri" |
| 22:46.55 | benjk | that'll tell you if caller ID is sent or not |
| 22:47.06 | fgwaller | sorry only PRI here |
| 22:47.07 | benjk | so what's the protocol then? |
| 22:47.11 | Lyfe | e&m wink |
| 22:47.22 | fgwaller | (bushdrums) |
| 22:47.29 | benjk | there gotta be some debug options for that |
| 22:47.34 | fgwaller | with CAS? |
| 22:47.53 | eKo1 | e&m wink <---- yuck |
| 22:47.57 | sxpert | anyone has experience with interfacing with completel or Colt E1s ? |
| 22:48.11 | Lyfe | eKo1: yeah, i know, we're migrating to a pri in a couple weeks, but it's bugging me righ tnow :P |
| 22:48.24 | justinu|laptop | in my experience, E&M wink doesn't work on asterisk |
| 22:48.39 | Lyfe | works fine. no callerid though. |
| 22:48.51 | justinu|laptop | then it doesn't work fine, does it |
| 22:49.06 | benjk | is there no debug option for e&m ? |
| 22:49.08 | Lyfe | fine, yes.. not good though |
| 22:49.37 | justinu|laptop | i actually fixed a bug regarding E&M a while back, but it was for immediate start |
| 22:49.44 | justinu|laptop | i gave up on trying to get winkstart to work |
| 22:49.51 | benjk | but does it have a debug option? |
| 22:50.01 | justinu|laptop | no, i added me own to debug the problem |
| 22:50.13 | benjk | pah, that really sucks then |
| 22:50.25 | benjk | shouldn't use anything that has no debug |
| 22:50.37 | Lyfe | well, i didnt' know any better wheni started. |
| 22:51.07 | benjk | debugging and logging are the most important items for anything software related |
| 22:51.35 | benjk | if you can't see whats going on, you're blind |
| 22:51.58 | benjk | would you start a job as a cab driver if you're blind? |
| 22:52.16 | Lyfe | actually, that might work, given cab drivers. |
| 22:52.22 | benjk | heh |
| 22:52.23 | eKo1 | hehehe |
| 22:56.02 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 22:56.27 | twisted[asteria] | actually e&m debugs just fine. |
| 22:56.41 | twisted[asteria] | and yes, e&m wink does work on asterisk |
| 22:56.42 | twisted[asteria] | we use it here |
| 22:56.57 | Lyfe | have information on how to tell what kinda stuff you're getting from it so i can see if i'm getting callerid info? |
| 22:57.21 | twisted[asteria] | you should get data in the form of *CALLERID*DNID* |
| 22:57.21 | twisted[asteria] | i do believe. |
| 22:57.28 | Lyfe | is it a variable? |
| 22:57.30 | twisted[asteria] | no |
| 22:57.38 | twisted[asteria] | turn on debug |
| 22:57.42 | twisted[asteria] | set debug 9 |
| 22:57.43 | twisted[asteria] | set verbose 9 |
| 22:57.48 | twisted[asteria] | call into one of the e&m wink channels |
| 22:57.52 | twisted[asteria] | you will see the digits pulsed to you |
| 22:58.32 | sxpert | *pulsed* ??? |
| 22:58.35 | sxpert | wow... |
| 22:58.43 | Lyfe | hrmm.. on the console you'd see it? (i don't see anything on there) |
| 22:58.44 | twisted[asteria] | pulsed, toned, wtf ever |
| 22:59.00 | twisted[asteria] | you have to turn on debug in the logger |
| 22:59.11 | *** part/#asterisk sp0n9e (n=sp0n9e@phpurge.com) |
| 22:59.34 | twisted[asteria] | but i'll give you a little hint |
| 22:59.41 | twisted[asteria] | our provider *SOMETIMES* provides caller id |
| 23:00.04 | twisted[asteria] | so you may or may not get it |
| 23:00.13 | twisted[asteria] | at first they told us they couldn't provide callerID on e&m wink |
| 23:00.19 | twisted[asteria] | but i called them out |
| 23:00.29 | Lyfe | well, my provider has said they are providing callerID on it. |
| 23:00.44 | twisted[asteria] | so turn on debug and see |
| 23:01.22 | *** join/#asterisk VetteC6 (n=info@216.143.192.69) |
| 23:01.48 | twisted[asteria] | e&m wink is so old and slow ... i despise it sometimes |
| 23:02.10 | Lyfe | so, set debug 9, set verbose 9, and hope that i see debug info? |
| 23:02.14 | twisted[asteria] | no |
| 23:02.17 | twisted[asteria] | turn it on in the logger |
| 23:02.23 | twisted[asteria] | you have to turn it on in the logger |
| 23:02.39 | twisted[asteria] | once you turn it on there (logger.conf) |
| 23:02.45 | Lyfe | ahh |
| 23:02.50 | twisted[asteria] | reload the logger, then do set verbose 9 and set debug 9 |
| 23:02.55 | twisted[asteria] | and viola, you have debug |
| 23:04.25 | *** join/#asterisk ogbi (i=ogbi@71.194.90.124) |
| 23:04.27 | *** join/#asterisk MatsK (n=mats@83.233.97.229) |
| 23:05.14 | Lyfe | well, how about that, just dnis digits. |
| 23:05.53 | twisted[asteria] | ;) |
| 23:06.28 | twisted[asteria] | now you can call your provider and complain :) |
| 23:06.35 | TripleFFFF | hehe.. hey |
| 23:06.40 | TripleFFFF | how you get glboal vars ? |
| 23:06.45 | TripleFFFF | i tough they where globals by default |
| 23:06.47 | twisted[asteria] | ${VARNAME} |
| 23:06.51 | TripleFFFF | nah |
| 23:06.51 | Lyfe | well, the question is, is the provider of the phone i tried to dial out with providing them with the callerid. |
| 23:06.53 | TripleFFFF | its not liking it |
| 23:07.12 | TripleFFFF | i do SET OldContext=${CONTEXT} |
| 23:07.15 | twisted[asteria] | Lyfe, good question. |
| 23:07.21 | twisted[asteria] | TripleFFFF, no, that won't work. |
| 23:07.23 | TripleFFFF | then a LOCAL/1234@mynewcontext |
| 23:07.29 | TripleFFFF | and in there i try to get Old |
| 23:07.35 | TripleFFFF | oh |
| 23:07.45 | TripleFFFF | setvar ? |
| 23:07.50 | TripleFFFF | changes all the time |
| 23:08.17 | twisted[asteria] | TripleFFFF, show application setglobalvar |
| 23:08.55 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 23:09.02 | Lyfe | cause this is what i should be expecting to have the callerid too, right? |
| 23:09.03 | Lyfe | Aug 11 18:08:53 VERBOSE[581] logger.c: -- Starting simple switch on 'Zap/1-1' |
| 23:09.06 | Lyfe | Aug 11 18:08:54 DEBUG[581] chan_zap.c: DTMF digit: 1 on Zap/1-1 |
| 23:09.08 | Lyfe | Aug 11 18:08:54 DEBUG[581] chan_zap.c: DTMF digit: 5 on Zap/1-1 |
| 23:09.11 | Lyfe | Aug 11 18:08:54 DEBUG[581] chan_zap.c: DTMF digit: 4 on Zap/1-1 |
| 23:09.12 | *** mode/#asterisk [+b %lyfe!*@*] by twisted[asteria] |
| 23:09.20 | twisted[asteria] | sorry, you're going to wind up flooding the channel with that |
| 23:09.22 | twisted[asteria] | use pastebin.ca |
| 23:09.33 | TripleFFFF | As of v1.2 SetVar is deprecated and we are back to Set. |
| 23:09.34 | TripleFFFF | lol |
| 23:09.36 | TripleFFFF | got |
| 23:09.44 | *** mode/#asterisk [-b lyfe!*@*] by twisted[asteria] |
| 23:09.53 | TripleFFFF | they say to use set with ,g |
| 23:10.01 | twisted[asteria] | ah k |
| 23:10.08 | *** join/#asterisk VetteC6 (n=info@216.143.192.69) |
| 23:10.18 | twisted[asteria] | see, you found your own answer ;) |
| 23:13.24 | *** join/#asterisk file (n=file@neutrino.joshua-colp.com) |
| 23:13.24 | *** mode/#asterisk [+o file] by ChanServ |
| 23:13.26 | *** mode/#asterisk [-b %lyfe!*@*] by twisted[asteria] |
| 23:13.41 | twisted[asteria] | strange |
| 23:13.45 | twisted[asteria] | i could swear i undid that already |
| 23:13.50 | Lyfe | freenode's weird. |
| 23:14.07 | twisted[asteria] | yeh. |
| 23:15.31 | chreese | hi, i was wondering if anyone could help w/ a fax issue. sending from a fax machine on an FXS card out via an FXO card, calling to a phone number on the same machine which is connected to an FXO. hope that makes sense. |
| 23:15.35 | *** part/#asterisk VetteC6 (n=info@216.143.192.69) |
| 23:15.49 | chreese | here is my ztmonitor stuff: http://pastebin.ca/126995 |
| 23:15.50 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 23:15.50 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 23:15.51 | Lyfe | sorry about the spam, wanted 3 lines, got 4, got caught. :P |
| 23:22.58 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 23:37.04 | *** join/#asterisk monrad (n=mmk@eth1.rt001a.cxnet.dk) |
| 23:41.56 | bkw_ | ALONE |
| 23:41.58 | bkw_ | wasabi peeps |
| 23:42.22 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 23:42.57 | NativeOnRye | <chreese> are you using the g.711 ulaw codec? |
| 23:43.45 | _Guhit | I'm trying to setup receiving faxes with my X100P and all I can seem to get when using rxfax is a small 8 byte file. There seems to several people with the same problem (via. google) but no solutions. |
| 23:44.00 | bkw_ | rxfax isn't ment to even be used for faxing |
| 23:44.12 | bkw_ | if you get 8 bytes then your libtiff is hozed |
| 23:44.49 | _Guhit | bkw_: What is rxfax supposed to be used for? |
| 23:45.14 | bkw_ | it was a simple test application to try out spandsp |
| 23:45.24 | bkw_ | it would be more prudent to use iaxmodem with hylafax |
| 23:45.28 | bkw_ | so you gain software ECM |
| 23:45.31 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.74) |
| 23:45.36 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 23:45.59 | _Guhit | bkw_: hmmm...ok, I'll try that then |
| 23:46.14 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.74) |
| 23:46.23 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.74) |
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