| 00:04.47 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:05.43 | mpruett | reister? |
| 00:06.14 | *** join/#asterisk CoffeeKid (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 00:06.25 | *** join/#asterisk reister (n=chatzill@m206-232.dsl.tsoft.com) |
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| 00:18.10 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
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| 00:19.59 | FuriousGeorge | hey all |
| 00:20.58 | NewSole | FuriousGeorge |
| 00:21.11 | NewSole | or is it CuriousGeorge |
| 00:21.18 | dlynes_office | SpuriousGeorge |
| 00:21.58 | dlynes_office | Just like strom is really storm...he just doesn't want to admit it :) |
| 00:22.04 | ManW|DaMetlBat | lol |
| 00:22.27 | dlynes_office | right, Strom_C? |
| 00:22.39 | FuriousGeorge | i wouldnt either. storm is best known as the female x-person |
| 00:23.06 | dlynes_office | ah |
| 00:23.25 | dlynes_office | i don't keep up on all the cartoons and comic books |
| 00:23.44 | drray | I'm a DC weenie not a Marvel stooge |
| 00:24.06 | FuriousGeorge | dlynes_office: i dont keep up on em but i did when i was 7 |
| 00:24.13 | dlynes_office | heh |
| 00:24.17 | dlynes_office | last time i read a comic book |
| 00:24.27 | dlynes_office | i think was when the Submariner was still popular |
| 00:24.44 | dlynes_office | It was around the time Heavy Metal Issue 1 had just come out |
| 00:25.05 | orlock | i used to like 2000AD |
| 00:25.13 | FuriousGeorge | lol |
| 00:25.28 | orlock | some of them were realy really well drawn |
| 00:25.32 | dlynes_office | orlock: you make it sound like 2000 was a long time ago |
| 00:25.54 | FuriousGeorge | 200ad? was that the comic about the post y2k-bug apocolypse |
| 00:27.03 | FuriousGeorge | i'll be here all week |
| 00:27.42 | orlock | dlynes_office: well, i read it like, 15 years ago or something :) |
| 00:27.46 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
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| 00:37.31 | copland | <PROTECTED> |
| 00:37.31 | copland | <PROTECTED> |
| 00:37.33 | copland | <PROTECTED> |
| 00:38.16 | *** join/#asterisk rbd (n=robbyd@cpe-066-057-011-095.nc.res.rr.com) |
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| 00:40.38 | rbd | hi guys, I'm trying to play a gsm file via AGI from /var/lib/asterisk/sounds (debian, .asterisk 1.2.1...) with the files already there, I can play them no problem (e.g. I play 'beep' for instance), however if I try to move a file into that folder, (say from voicemail/1234/busy.gsm, and play 'busy', it can't find the file...however trying the file at its previous location, 'voicemail/1234/busy' works fine) any ideas? |
| 00:41.17 | Strom_C | rbd: sounds like you're mistyping the path somewhere |
| 00:41.22 | rbd | the permissions and the owner are all the same |
| 00:41.32 | Strom_C | and also: holy shit, 1.2.1 is WAY OLD |
| 00:41.49 | dlynes_office | well, not only that |
| 00:41.55 | dlynes_office | 1.2.1 has a severe memory leak bug |
| 00:41.58 | Jason99 | Anyone know what this means? Jun 14 20:38:41 WARNING[15194]: chan_sip.c:2542 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) |
| 00:42.31 | dlynes_office | Jason99: type 4 is ulaw, 256 is g729 |
| 00:42.41 | Jason99 | ah ok |
| 00:42.44 | dlynes_office | Jason99: you don't have g729 trancoder codec installed |
| 00:42.52 | Jason99 | thats right.. I dont |
| 00:42.54 | Jason99 | hehe |
| 00:42.54 | Jason99 | thanks |
| 00:43.05 | dlynes_office | Jason99: at the cli, you can type show codecs to see what the bit values are for the various codecs |
| 00:43.39 | dlynes_office | I just happen to have those two memorized because they're so common |
| 00:43.47 | Jason99 | but if both phones support g729 it should work right? |
| 00:43.53 | dlynes_office | Jason99: correct |
| 00:44.06 | dlynes_office | Jason99: assuming you don't tell asterisk to stay in the media path |
| 00:44.17 | dlynes_office | Jason99: then you might have issues, or you might now |
| 00:44.18 | dlynes_office | Jason99: then you might have issues, or you might not |
| 00:44.43 | Jason99 | From what I've read it should pass through without having the codec installed.. but it wont transcode which is what I'm doing.. |
| 00:45.10 | Jason99 | I thought both phones were setup for g729a, I'll have a look thanks |
| 00:45.11 | dlynes_office | well, yeah, cause you're trying to convert g729 to ulaw |
| 00:45.31 | dlynes_office | Jason99: I would check your sip.conf file, too...not just your phones |
| 00:45.48 | Jason99 | Can I force g729 on both phones in SIP ? |
| 00:45.50 | dlynes_office | Jason99: also, is any sip traffic going to be going outside your firewall? |
| 00:45.52 | Jason99 | oops.. sip.conf |
| 00:45.53 | dlynes_office | Jason99: yes |
| 00:46.17 | dlynes_office | Jason99: keep in mind, if your phones are using g729 |
| 00:46.28 | dlynes_office | Jason99: and you want to check your voicemail with them |
| 00:46.34 | dlynes_office | Jason99: you'll probably run into problems |
| 00:46.50 | dlynes_office | Jason99: voicemail is usually recorded in pcm, or gsm |
| 00:46.52 | Jason99 | dlynes_office: I'd like the phones to use G729 if they can, and if they can't use g711.. is that possible? |
| 00:47.01 | dlynes_office | Jason99: yes, it is |
| 00:47.14 | Strom_C | Jason99: do you have the bandwidth to support G711 all the time? |
| 00:47.27 | dlynes_office | Jason99: disallow=all ; allow=g729 ; allow=ulaw |
| 00:47.40 | Jason99 | Strom_C: no I don't I'd rather not use g711 for all calls |
| 00:47.41 | dlynes_office | Jason99: however, if you can handle g711 all the time, i'd forget g729 |
| 00:47.51 | dlynes_office | Jason99: g729 is horrible quality |
| 00:48.00 | dlynes_office | Jason99: it was invented to save bandwidth |
| 00:48.21 | Strom_C | it sounds like a really bad cellphone connection |
| 00:49.03 | Jason99 | It will be on selective endpoints that are on bad quality connections |
| 00:49.08 | dlynes_office | Strom_C: well, actually...if you're calling india |
| 00:49.18 | dlynes_office | Strom_C: g729 still sounds better than a land line |
| 00:49.23 | Strom_C | hahah true |
| 00:49.31 | Strom_C | but thats like being the tallest midget |
| 00:49.36 | Jason99 | lol |
| 00:49.48 | Jason99 | thanks for your help guys |
| 00:50.05 | Jason99 | actually.. does the order of the allows mean anything? |
| 00:50.50 | dlynes_office | Jason99: yes...hte order indicates the order of preference |
| 00:51.15 | Jason99 | thanks it works now |
| 00:51.51 | dlynes_office | heh...now try checking your voicemail :p |
| 00:51.59 | dlynes_office | or listening to your music on hold ;) |
| 00:52.19 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 00:53.13 | Jason99 | argg... |
| 00:53.15 | Jason99 | lol |
| 00:53.31 | *** join/#asterisk saftsack (n=saftsack@p54A7FED0.dip.t-dialin.net) |
| 00:53.32 | Jason99 | Is there a way around that? |
| 00:54.25 | dlynes_office | well, that's why i suggested using ulaw for your default, not g729 |
| 00:55.01 | Jason99 | Can you force a codec in a context? |
| 00:55.29 | Jason99 | I agree that ulaw is the best.. but in my case I have to use g729 for some endpoints |
| 00:55.46 | dlynes_office | yes, you can |
| 00:55.53 | Jason99 | what command would I lookup? |
| 00:56.00 | dlynes_office | bweschke wrote a patch for doing just that |
| 00:56.22 | dlynes_office | go to bugs.digium.com and do a search for codec |
| 00:56.33 | Jason99 | thanks |
| 00:56.33 | dlynes_office | you should be able to find bweschke's patch for it there |
| 00:56.42 | dlynes_office | it's a patch against trunk though |
| 00:56.53 | dlynes_office | so you might have to manually patch instead of using the patch tool |
| 01:05.03 | *** join/#asterisk Eric-xx (i=ericx@cm83.epsilon192.maxonline.com.sg) |
| 01:07.40 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 01:08.07 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 01:14.26 | websae | I am curious.....if I have 2 polycoms in an office........Polycom A and Polycom B....when i put a call on hold on polycom A, can i pick up that hold call on polycom b? |
| 01:14.29 | websae | is that possible? |
| 01:14.46 | dlynes_office | websae: yes |
| 01:14.50 | dlynes_office | websae: with asterisk, no |
| 01:15.03 | websae | hrm...not possible with asterisk darn... |
| 01:15.08 | websae | what does one need to do? |
| 01:15.25 | dlynes_office | well, you might be able to do that starting with asterisk 1.4 |
| 01:15.35 | dlynes_office | there's a feature for shared line appearances going into 1.4 |
| 01:15.40 | dlynes_office | I just don't know the particulars |
| 01:15.45 | *** join/#asterisk JSabines (n=alancast@201.138.163.186) |
| 01:15.56 | websae | ahh |
| 01:16.11 | dlynes_office | websae: it'll probably be released sometime in july |
| 01:16.16 | dlynes_office | but no firm date's been set yet |
| 01:16.17 | Dandan | bleh |
| 01:16.18 | Dandan | re all |
| 01:16.23 | rbd | okay with this missing prompts problem. Say I have a file /var/lib/asterisk/sounds/vm-youhave.gsm ... playing this file through AGI works fine (the file name is 'vm-youhave')...however if I cp vm-youhave.gsm to test.gsm in the same sounds directory, and try to play 'test'. I get "ast_openstream_full: File test does not exist in any format" |
| 01:16.55 | Dandan | rbd: are permissions ok? |
| 01:17.03 | dlynes_office | rbd: and? did you try the suggestions Strom_C made earlier? |
| 01:17.19 | rbd | yeah guys, upgraded to asterisk 1.2.7 |
| 01:17.26 | rbd | permissions are the exact same as the other files |
| 01:17.29 | dlynes_office | rbd: that was only one of the suggestions |
| 01:17.30 | dlynes_office | ah |
| 01:17.30 | Dandan | still that's 2.1 versions behind |
| 01:17.49 | dlynes_office | Dandan: yeah...taht's the price you pay for not using the source code |
| 01:18.05 | websae | hrm... |
| 01:18.06 | Dandan | dlyn |
| 01:18.07 | websae | okay |
| 01:18.12 | Dandan | dlynes_office: i always use it :) |
| 01:18.19 | rbd | hmm that's the newest debian has in 'testing'...interesting its so out of date for testing |
| 01:18.21 | Dandan | i even wrote my own compilation scripts |
| 01:18.46 | dlynes_office | Dandan: same here, but most peeps rely on debian's, or redhat's or insert-your-distro-here's binary packages |
| 01:19.08 | Dandan | BLAH :) |
| 01:19.11 | Dandan | slackware rulez :) |
| 01:19.22 | dlynes_office | Dandan: i use binary packages on slackware, too |
| 01:19.24 | Dandan | it takes 10 mins to update and upgrade and restart asterisk :) |
| 01:19.31 | dlynes_office | Dandan: but i compile it once on one machine |
| 01:19.36 | Dandan | dlynes_office: yeah, but only precompiled by me... |
| 01:19.42 | Dandan | EXXXXACTLY :) |
| 01:19.43 | dlynes_office | Dandan: create a package, and then deploy it by packages on all the other machines |
| 01:19.50 | Dandan | yup yup yup :) |
| 01:19.54 | Strom_C | rbd: just compile the damned thing from source already |
| 01:19.57 | schirpich | I have a TE410P currently using 3 span and I am attempting to hook up a 4th T1 in the 4th span... I believe its working. How can I actually test and verify its working as it should be? |
| 01:20.01 | dlynes_office | Strom_C: lol |
| 01:20.21 | rbd | Strom: the version I was trying this with earlier was compiled from source :) ... then I got too lazy and started using the deb |
| 01:20.24 | Dandan | i also add many expensive-optimizations and funrolls to the scripts to even further optimize asterisk :) |
| 01:20.25 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 01:20.28 | rbd | I'll try 1.2.9.1 then |
| 01:20.48 | Strom_C | schirpich: does it appear to work? |
| 01:20.51 | Dandan | rbd: tar xfvz asterisk*.tar.gz && ./configure && make && make install :) |
| 01:20.52 | Strom_C | can you place calls? |
| 01:20.58 | dlynes_office | rbd: anyways...the problem you're running into is not a problem with 1.2.7, 1.2.9.1 or 1.2.1 |
| 01:21.05 | dlynes_office | rbd: and it's not going to be solved by upgrading |
| 01:21.13 | dlynes_office | rbd: it's a problem with your configuration |
| 01:21.26 | schirpich | zap show channels shows all of them and the zttool shows its ok |
| 01:21.47 | schirpich | but other than getting a high volume of calls, how else could i test it? |
| 01:21.56 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 01:21.56 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:22.02 | rbd | dlynes_office, yup...never have run into this kind of things before...it's almost if the sounds are being cached somewhere else. I will mess with the paths |
| 01:22.07 | dlynes_office | Dandan: yeah...i just stage addons, asterisk, and sounds all in the same directory |
| 01:22.07 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 01:22.18 | dlynes_office | rbd: i suspect it's probably an issue with your dialplan |
| 01:22.23 | Dandan | oh, I didn't think of that :) |
| 01:22.29 | Dandan | that's easy though :) |
| 01:22.39 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 01:22.47 | dlynes_office | Dandan: that way i've got one integrated package |
| 01:23.12 | Dandan | right... |
| 01:23.19 | dlynes_office | Dandan: libpri and zaptel otoh i can't put into that package type stuff |
| 01:23.20 | Dandan | i see :) gotta try that |
| 01:23.31 | dlynes_office | Dandan: those i have to compile for each platform I'm targetting |
| 01:23.34 | Dandan | i had to drop zaptel from packages though... damn sangoma drivers... |
| 01:23.42 | dlynes_office | Dandan: well, that's not hte only reason |
| 01:23.52 | Dandan | ? |
| 01:23.53 | dlynes_office | Dandan: sometimes i use sangoma, sometimes digium, sometimes x100p |
| 01:24.02 | dlynes_office | Dandan: and sometimes ztdummy |
| 01:24.09 | Dandan | but you can compile zaptel for all of them |
| 01:24.13 | Dandan | same with libpri |
| 01:24.17 | dlynes_office | nah |
| 01:24.26 | dlynes_office | compoile once, install sangoma, compile again |
| 01:24.34 | Dandan | right, for sangoma |
| 01:24.42 | Dandan | but for x100/ztdummy? |
| 01:24.47 | dlynes_office | and so if i have something compiled for sangoma |
| 01:24.53 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 01:24.57 | dlynes_office | i can't use that on x100p/ztdummy/digium |
| 01:25.07 | Dandan | of course you can't |
| 01:25.13 | Dandan | that's sangoma thing... :/ |
| 01:25.14 | dlynes_office | Dandan: well, there you go |
| 01:25.16 | Dandan | unfortunately... |
| 01:25.25 | *** join/#asterisk mgob (n=goldenol@c-24-17-240-110.hsd1.wa.comcast.net) |
| 01:25.29 | mgob | hi |
| 01:25.34 | dlynes_office | bye |
| 01:25.42 | Dandan | but if, let's say you have only x100/zaptel/ztdummy tyhen you can pre-package the whole thing |
| 01:25.45 | Dandan | :) |
| 01:25.58 | Dandan | and libpri doesn't change anyway :) |
| 01:26.01 | dlynes_office | Dandan: if they all have the same kernel, yeah |
| 01:26.02 | Dandan | that can be precompiled |
| 01:26.13 | Dandan | dlynes_office: which I keep the same :) |
| 01:26.15 | dlynes_office | Dandan: certain machines i've customized kernels for though |
| 01:26.20 | mgob | any other artificial way of raising the volume on a prompt other then rx/tx or re-encoding the file? |
| 01:26.30 | Dandan | dlynes_office: but you keep it up to date? |
| 01:26.36 | dlynes_office | Dandan: because those super slow piece of crap via c3's i optimize the piss out of for the kernel |
| 01:27.07 | Dandan | lol, you can compile the kernel on a quad opteron and copy to c3... :) |
| 01:27.07 | dlynes_office | Dandan: the newest kernel i'm running is 2.6.15.5 |
| 01:27.23 | dlynes_office | Dandan: you wanna buy me a quad opteron? |
| 01:27.36 | CoffeeKid | hello, I need help making the agi service run on a address different then 127.0.0.1:5038 .. I need to make it run on anything within my subnet, what file do i need to edit to do this? :) |
| 01:27.36 | Dandan | well, maybe not at the moment |
| 01:27.41 | dlynes_office | hahaha |
| 01:27.42 | rbd | found the problem... stupid me...two seperate sounds directories... it was reading from an old one |
| 01:27.43 | Dandan | but wait a few... years... decades... :) |
| 01:28.25 | CoffeeKid | does that make any sense? :) |
| 01:29.15 | Dandan | CoffeeKid: not really |
| 01:29.16 | Dandan | :) |
| 01:29.17 | dlynes_office | CoffeeKid: manager.conf |
| 01:29.57 | CoffeeKid | dlynes_office: once i modify it,do i need to do a reload? or a total service restart? |
| 01:30.29 | dlynes_office | CoffeeKid: no idea...never used it |
| 01:30.38 | Dandan | i used it ONCE :) |
| 01:30.52 | dlynes_office | oh wait |
| 01:30.55 | dlynes_office | i modified it once |
| 01:30.58 | dlynes_office | so i could disable it |
| 01:30.59 | Dandan | for a cool experiment called yaacid - Yet Another Asterisk CallerID :) |
| 01:31.04 | Dandan | which was pretty cool :) |
| 01:31.05 | CoffeeKid | i have the following lines in there.. does it look correct? |
| 01:31.09 | Dandan | a pop up when a call came :) |
| 01:31.11 | CoffeeKid | deny=0.0.0.0/0.0.0.0 |
| 01:31.16 | CoffeeKid | permit=10.0.1.0/255.255.255.0 |
| 01:31.27 | dlynes_office | CoffeeKid: nope |
| 01:31.34 | dlynes_office | CoffeeKid: that netmask looks fubar |
| 01:31.44 | Dandan | ? |
| 01:31.44 | CoffeeKid | on which line? the permit? |
| 01:31.47 | Dandan | which one? |
| 01:31.49 | Dandan | i have: |
| 01:31.53 | dlynes_office | CoffeeKid: it should probably be more like permit=10.0.1.0/255.255.0.0 |
| 01:31.54 | Dandan | deny=0.0.0.0/0.0.0.0 |
| 01:31.54 | Dandan | permit=127.0.0.1/255.255.255.255 |
| 01:31.54 | Dandan | permit=10.1.1.1/255.255.255.0 |
| 01:31.54 | Dandan | permit=10.0.0.1/255.255.255.0 |
| 01:32.10 | Dandan | you can subnet 10.0.0.0/8 the way you want it |
| 01:32.13 | Dandan | no probs |
| 01:32.30 | CoffeeKid | okay, so if i do: |
| 01:32.30 | dlynes_office | yeah...but then what's the point of running a class A subnet, instead of a class C? |
| 01:32.32 | Dandan | at my place 10.0.0.0/24 is our data network and 10.0.1.0/24 is our voice :) |
| 01:32.52 | CoffeeKid | okay.... hmm |
| 01:33.15 | CoffeeKid | so probably 10.0.1.0/8 would work? |
| 01:33.26 | Dandan | dlynes_office: if you need max. 254 devices on your net |
| 01:33.32 | Dandan | then /24 is adequate :) |
| 01:33.41 | dlynes_office | Dandan: 253 |
| 01:33.42 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 01:33.48 | dlynes_office | Dandan: don't forget your gateway |
| 01:34.04 | CoffeeKid | then, how would i restart the service so that it listens on that ip address? |
| 01:34.04 | Dandan | CoffeeKid: 10.0.1.0/24 is enough anything 10.0.1.1-254 is gonna get to it :) |
| 01:34.08 | mgob | any other artificial way of raising the volume on a prompt other then rx/tx or re-encoding the file? |
| 01:34.17 | Dandan | CoffeeKid: restart now? |
| 01:34.29 | Dandan | dlynes_office: yeah, gw is another network host :) |
| 01:34.43 | dlynes_office | Dandan: yeah, but it's sacrificed for routing |
| 01:34.46 | CoffeeKid | restart asterisk? or just do a reload? |
| 01:35.02 | Dandan | dlynes_office: not necessarily, my voice net has no gw, it is not routed outside the company |
| 01:35.04 | dlynes_office | shutdown -r Now NOW!!!!!!! |
| 01:35.16 | Dandan | dlynes_office: don'[t be that RADICAL :) |
| 01:35.19 | CoffeeKid | no way, this is a production box :P |
| 01:35.22 | dlynes_office | lol |
| 01:35.31 | Dandan | do restart when convenient then |
| 01:35.54 | Dandan | # uname -a |
| 01:35.54 | Dandan | Linux voip 2.6.14.4 #1 SMP Wed Dec 21 12:13:09 EST 2005 i686 unknown unknown GNU/Linux |
| 01:36.01 | riddlebox | has anyone heard of a company called quintum? |
| 01:36.03 | Dandan | you are right, I am behind the schedule too |
| 01:36.07 | dlynes_office | riddlebox: never |
| 01:36.09 | Dandan | riddlebox: not me |
| 01:36.41 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 01:36.46 | dlynes_office | Quintum's one of the oldest voip hardware companies, riddlebox |
| 01:36.47 | riddlebox | they are selling sip gateways and ata devices that will do sip or h.323 |
| 01:37.04 | dlynes_office | riddlebox: and their prices are off the charts |
| 01:37.08 | Dandan | ~quintum |
| 01:37.21 | Dandan | jbot: ~quintum |
| 01:37.34 | riddlebox | they contacted my company the other day asking if we will support asterisk boxes for one of their customers |
| 01:37.34 | CoffeeKid | OMG! that worked, you guys are awesome! |
| 01:37.36 | Dandan | blah :) |
| 01:37.45 | Dandan | CoffeeKid: aren't we :) |
| 01:37.54 | Dandan | jbot: :P |
| 01:38.26 | riddlebox | dlynes_office, have you configured one of their tenor products? |
| 01:38.33 | dlynes_office | riddlebox: ask them to give you a couple free quintum boxes for testing with, and you'll think about it :) |
| 01:38.47 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-250.cybersurf.com) |
| 01:38.53 | dlynes_office | riddlebox: nah...never worked with their equipment at all |
| 01:39.01 | dlynes_office | riddlebox: but all their stuff is pretty high end |
| 01:39.06 | riddlebox | dlynes_office, the one gateway they sell will provide sip to 800 connections |
| 01:39.21 | riddlebox | look out broadvoice lol! |
| 01:39.32 | dlynes_office | riddlebox: yeah...they have other stuff for four ports, too |
| 01:39.43 | dlynes_office | riddlebox: they have the low end and the high end |
| 01:39.47 | Dandan | BV rox :) |
| 01:39.51 | dlynes_office | riddlebox: all high quality, and all high prices |
| 01:40.00 | riddlebox | yeah they gave us a two port and 4 port, but their documentation sucks |
| 01:40.17 | dlynes_office | riddlebox: i guess you've never tried an epygi gateway :p |
| 01:40.17 | riddlebox | the 2 port was like $130 I didnt think it was that bad |
| 01:40.23 | dlynes_office | riddlebox: those things freaking blow |
| 01:40.25 | Dandan | they are high quality too :) |
| 01:40.31 | Dandan | with HiFi sound :) |
| 01:40.45 | dlynes_office | Dandan: grandstream? high quality? who the hell are you trying to kid? |
| 01:41.10 | orlock | Hmm.. Has anybody hereused a netgear managedswitch? |
| 01:41.11 | Dandan | dlynes_office: i only had like 2 bad phones out of 75 :) |
| 01:41.20 | Dandan | orclo: i have 5 of them |
| 01:41.22 | Dandan | afair :) |
| 01:41.31 | orlock | Dandan: yes, but could you kill somebody with a grandstream? |
| 01:41.34 | Dandan | orlock: i have 5 of them |
| 01:41.37 | orlock | the cisco's feel like you could |
| 01:41.40 | dlynes_office | Dandan: i've only had one bad phone out of 25 or so |
| 01:41.44 | dlynes_office | Dandan: but the voice quality sucks |
| 01:41.51 | orlock | Dandan: do you know the default management ip/port for the web interface? |
| 01:42.00 | Dandan | hmmmm |
| 01:42.08 | Dandan | i do not, of the top of my head |
| 01:42.09 | Dandan | nmap them |
| 01:42.11 | Dandan | :) |
| 01:42.18 | orlock | dunno the ip :) |
| 01:42.18 | Dandan | there is this utility |
| 01:42.26 | Dandan | that recognizes them by mac... |
| 01:42.30 | orlock | yeah, there is always this utility.. FOR WINDOWS! |
| 01:42.34 | Dandan | it is avail. on netgear's website |
| 01:42.40 | Dandan | oh right |
| 01:42.40 | dlynes_office | orlock: nmap -v -Ss 192.168.1.0/24 |
| 01:42.41 | Dandan | windows :) |
| 01:42.54 | Dandan | then VmWare :) |
| 01:42.59 | Dandan | the player IS free :) |
| 01:43.35 | orlock | dlynes_office: :P |
| 01:43.41 | dlynes_office | orlock: ? |
| 01:43.53 | Dandan | he meant dandan: :P :) |
| 01:44.04 | orlock | dlynes_office: do you _know_ its on the 192.168.1 range? :) |
| 01:44.07 | dlynes_office | nah...I think he's scared of you |
| 01:44.17 | dlynes_office | orlock: nope...it was just an example |
| 01:44.21 | orlock | yes |
| 01:44.34 | orlock | i dont wanna have to sit there sniffing packets |
| 01:44.39 | orlock | maybe i will look for the box... |
| 01:44.47 | Dandan | orlock |
| 01:44.58 | dlynes_office | orlock: maybe walk over to the computer and type '/sbin/ifconfig'? |
| 01:44.58 | Dandan | orlock: it would be faster to install win :) |
| 01:45.01 | Dandan | prolly :) |
| 01:45.26 | dlynes_office | oh...nvm |
| 01:45.30 | dlynes_office | it's a crappy cisco phone |
| 01:45.30 | Dandan | orlock: hmmm... 8080? 8081? |
| 01:45.32 | Dandan | or 80 :) |
| 01:45.51 | Dandan | dlynes_office: it is a bit too expensive :) - cisco phone |
| 01:46.00 | Dandan | we were switching from 15 yrs old NVM phone system |
| 01:46.02 | Dandan | to voip :) |
| 01:46.05 | dlynes_office | can't you find out the ip address of a cisco by hitting a few buttons on its keypad? |
| 01:46.09 | dlynes_office | nvm? |
| 01:46.12 | dlynes_office | nortel voicemail? |
| 01:46.15 | Dandan | Nitsuko NVM |
| 01:46.19 | dlynes_office | ewewww |
| 01:46.27 | Dandan | or something similar :) |
| 01:46.37 | dlynes_office | we have some joker that was trying to get us to to become a dealer for nitsuko |
| 01:46.44 | dlynes_office | i thought those crappy pbxes died a long time ago |
| 01:46.55 | trelane | dlynes_office, <settings> <3> |
| 01:47.12 | dlynes_office | orlock: there ya go, orlock |
| 01:47.17 | dlynes_office | orlock: no need to use nmap |
| 01:47.29 | Dandan | lol |
| 01:47.34 | Dandan | nitsuko is alive and well |
| 01:47.41 | Dandan | their POS phones are $400/piece... |
| 01:47.59 | dlynes_office | so how many companies in your city have a recent model of nitsuko? |
| 01:48.58 | Dandan | hm, definetely quite a few... |
| 01:49.08 | Dandan | i know a dealer still marketing them... |
| 01:51.47 | Dandan | Hot, humid weather will be the rule on Sunday with highs of 90+ degrees away from Long Island Sound. |
| 01:51.51 | Dandan | BLAH lovely new england :/ |
| 01:53.05 | dlynes_office | heh....high of 66 for us :) |
| 01:53.40 | *** join/#asterisk ManxPower (i=ewieling@38.sub-70-210-92.myvzw.com) |
| 01:53.42 | Dandan | BC? Heh... |
| 01:53.42 | Dandan | nice |
| 01:53.48 | dlynes_office | Dandan: but...your weather is the best you know why? |
| 01:53.57 | dlynes_office | Dandan: the hotter the weather, the shorter the skirts :) |
| 01:53.58 | Dandan | dlynes_office: why? |
| 01:54.11 | Dandan | dlynes_office: LOL! :) I am married though :) |
| 01:54.16 | dlynes_office | too bad :) |
| 01:54.30 | Dandan | I know :) |
| 01:54.33 | Dandan | oh well :) |
| 01:54.34 | dlynes_office | doesn't mean you're not allowed to look though :) |
| 01:54.44 | dlynes_office | just can't touch :) |
| 01:54.53 | ManxPower | In the USA many television stations are switching to digital and broadcasting multiple "stations". So, I think to myself, "Self, I wonder how much a DTV tuner for my standard TV would cost?" and apparently the answer is "You'll have to wait since you can't buy one." |
| 01:55.21 | dlynes_office | and? |
| 01:55.39 | Dandan | well... looking but no toucing? :) i got used to touching though :) |
| 01:55.41 | dlynes_office | the cable station will only too gladly sell you one |
| 01:55.58 | denon | ManxPower: you can buy standalone HD tuners |
| 01:56.00 | Dandan | ManxPower: ATCS? or CATV? |
| 01:56.03 | denon | OTA HD |
| 01:56.05 | Dandan | *ATSC |
| 01:56.18 | dlynes_office | forget all that crap |
| 01:56.23 | Dandan | mine came with OTA HD, but I have no antenna :) |
| 01:56.25 | dlynes_office | digital satellite hdtv is where it's at |
| 01:56.34 | denon | yeah, Ive got dish HD personally |
| 01:56.36 | ManxPower | noky, not OTA HD, OTA SD |
| 01:56.45 | dlynes_office | everything's digital, instead of that crappy analog/digital hybrid setup that cable has |
| 01:56.46 | ManxPower | I don't have an HD TV, I don't expect to get an HD tv. |
| 01:56.47 | Dandan | OTA SD sux! :) |
| 01:56.52 | Dandan | but it is digital though :) |
| 01:57.12 | Dandan | ManxPower: i got a great deal on 40 inch lcd and i have an hd tv... |
| 01:57.19 | ManxPower | But I would like to see the 24-hour weather feed many stations have in addition to their normal feed. |
| 01:57.20 | Dandan | it's for my wife though :) |
| 01:57.27 | Dandan | I IRC from 15 inch CRT :) |
| 01:57.45 | dlynes_office | Dandan: damn...i irc from dual 17" flat screens |
| 01:57.48 | ManxPower | danalien, I want to pipe the output thru a closed coax system to analog TVs. |
| 01:58.25 | ManxPower | now that we all understand I want a TUNER, not a TV, we can proceed. |
| 01:58.25 | dlynes_office | Dandan: you're an alien now |
| 01:58.57 | Dandan | yeah i see :) |
| 01:59.32 | Dandan | ManxPower: hmmm... no idea if there are OTA HD/digital receivers |
| 01:59.38 | Dandan | to use your tv as a monitor only |
| 01:59.42 | ManxPower | Oh, and it needs RCA audio/video out, since that's all the agile TV converters I have can handle. |
| 02:00.47 | Dandan | requirements: single rich and availavle :) |
| 02:00.53 | Dandan | *available :D |
| 02:00.55 | ManxPower | Dandan, DTV people say "no, people won't have to buy new televisions, they can buy a low-cost box for use with their standard televisions", but like the unicorn or an honest politicion it seems like they are a myth |
| 02:01.34 | ManxPower | And people that watch OTA TV tend not to be a wealthy group of people. |
| 02:01.45 | Dandan | ManxPower: true, besides, i used to own a cox digital cable with all this HDTV crap - sorry, but it is not worth spending extra money on just like 10 channels with digital quality 720p/1080i... |
| 02:02.25 | Dandan | otoh, if you want a nice HD sig, install a dish and play with Bev :) |
| 02:02.34 | Dandan | they have a pretty nice sat signal :) |
| 02:03.01 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 02:03.21 | ids2500 | 800-225-0046 |
| 02:03.53 | orlock | we have a dtv box here |
| 02:04.06 | orlock | only cost $100 or something |
| 02:04.14 | Dandan | catv/sat? |
| 02:04.29 | Dandan | ids2500: nice number, what is it? |
| 02:04.37 | ids2500 | lol |
| 02:04.37 | ids2500 | not sure |
| 02:05.01 | orlock | Dandan: standard rf signal |
| 02:05.16 | ManxPower | Dandan, I could not care less about HD. I want all the cool multicast channels/ |
| 02:05.19 | dlynes_office | what a stupid phone number |
| 02:05.26 | ManxPower | orlock, where are you located? |
| 02:05.26 | dlynes_office | devil's paradise? |
| 02:05.27 | dlynes_office | wtf? |
| 02:05.55 | orlock | australia |
| 02:06.14 | ManxPower | See this: http://arstechnica.com/news.ars/post/20060614-7055.html |
| 02:06.38 | Dandan | dlynes_office: did you call? :) |
| 02:06.41 | dlynes_office | damn...32MB's to install asterisk 1.2.9.1 |
| 02:06.44 | dlynes_office | Dandan: yeah |
| 02:06.54 | dlynes_office | Dandan: it was a really stupid autoattendant |
| 02:06.55 | Dandan | i didn't have no phone in my basement |
| 02:06.57 | *** join/#asterisk iq|mobile (n=iq@71-215-58-212.omah.qwest.net) |
| 02:07.02 | Dandan | i have a reflashed at&t dlink though :) |
| 02:07.10 | ManxPower | that URL is why I started looking for DTV tuner |
| 02:07.15 | Dandan | callvantage :) |
| 02:07.44 | ManxPower | Hell, in the UK they have like 15 free DTV stations called FreeView |
| 02:07.49 | Dandan | for only $30 you get a fully blown voip (sip)+router |
| 02:08.00 | Dandan | ManxPower: that's uk, we are miles behind :) |
| 02:08.03 | Dandan | like with cell phones |
| 02:08.10 | Dandan | i always get one when I go to europe :) |
| 02:10.46 | *** join/#asterisk mrtwister (n=manopulu@107.250.broadband5.iol.cz) |
| 02:12.47 | techman97_andy | hey y'all, I have my test * box registering to my prod * box...I can recieve calls just fine, but when I dial from the test * box, I get a SIP circuit-busy message on the test box...and nothing on the prod box - any ideas? |
| 02:13.30 | ManxPower | techman97_andy, "sip debug" |
| 02:20.09 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 02:21.29 | *** join/#asterisk okdo (n=goldenol@c-24-17-240-110.hsd1.wa.comcast.net) |
| 02:21.33 | okdo | hi |
| 02:21.55 | okdo | this is going to sound retarded but I can not figure out the syntax for the life of me, does anyone have an example of using sox to boost the volume of a gsm file? |
| 02:22.14 | okdo | i can't get vol to actually make it sound louder so I am either a moron or I am a moron :) |
| 02:22.57 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 02:23.16 | Qwell | okdo: The second one |
| 02:23.57 | Qwell | okdo: and it's just -v |
| 02:24.04 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 02:24.34 | okdo | sox in.gsm -v2 out.gsm ? |
| 02:24.39 | Qwell | something like that |
| 02:26.54 | Jason99 | Does anyone know of a way to only do a reinvite if the gateway/phone is behind nat? |
| 02:32.02 | *** join/#asterisk nexstar (n=nexstar@ip68-111-77-138.oc.oc.cox.net) |
| 02:32.59 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 02:36.19 | ManxPower | Jason99, I doubt it. |
| 02:36.38 | *** join/#asterisk sergiovel (n=Sergio@24-119-73-67.cpe.cableone.net) |
| 02:36.40 | *** join/#asterisk newsmafia (n=newsmafi@wsip-70-166-5-130.sd.sd.cox.net) |
| 02:36.48 | ManxPower | Jason99, if both sides are behind NAT then you would have to port forward at least |
| 02:36.57 | sergiovel | hello everyone |
| 02:37.45 | ManxPower | one router |
| 02:37.47 | sergiovel | I have a quick question, I am visitin Boise Idaho, usa. Does anyone know of a shop here that sells voip phones? |
| 02:38.51 | Dandan | online? |
| 02:38.57 | Dandan | or brick and mortar? |
| 02:39.05 | Qwell | voip is illegal in Idago |
| 02:39.06 | sergiovel | brick and mortar |
| 02:39.08 | Qwell | Idaho too |
| 02:39.13 | sergiovel | really? |
| 02:39.17 | Dandan | brick and mortar? |
| 02:39.18 | Dandan | compusa? |
| 02:39.19 | Qwell | no |
| 02:39.21 | Dandan | best buy? |
| 02:39.24 | Dandan | circuit city? |
| 02:39.26 | Qwell | Dandan: yeah...right |
| 02:39.32 | Dandan | frys? |
| 02:39.33 | Dandan | :) |
| 02:39.38 | Qwell | Dandan: unlikely |
| 02:39.40 | sergiovel | ok, thanks, I am from argentina and not used to here |
| 02:39.41 | Dandan | no idea, i'd rather buy it online |
| 02:39.45 | Dandan | with next day delivery |
| 02:39.55 | Dandan | sergiovel: welcome to the us :) |
| 02:39.58 | sergiovel | just on a business trip but wanted to take advantage of the trip |
| 02:39.59 | sergiovel | thanks |
| 02:40.08 | Dandan | sergiovel: it is better to buy something online |
| 02:40.14 | Dandan | (and cheaper) |
| 02:40.24 | sergiovel | it sounds very good |
| 02:40.30 | Dandan | and have it delivered to your hotel/business the next day |
| 02:40.33 | Dandan | it is possible/doable |
| 02:40.41 | Dandan | it all depends what you are looking for... |
| 02:40.51 | sergiovel | sure will, any site you recomend? |
| 02:41.04 | sergiovel | looking for a voip phone |
| 02:41.13 | Dandan | start with www.pricegrabber.com - that is like... kelkoo or any other site that compares prices... |
| 02:41.33 | sergiovel | beautiful guys, thanks a bunch!!! |
| 02:41.51 | Qwell | might as well just hit voipsupply |
| 02:41.55 | ManxPower | any kind of IP phone you can buy at a store will be locked to a provider |
| 02:42.04 | sergiovel | right |
| 02:42.06 | Qwell | ManxPower: indeed |
| 02:42.21 | Dandan | yeah www.voipsupply.com |
| 02:42.46 | Dandan | call them while ordering and tell them that you want to have it delivered to your hotel/business and sent immediately :) |
| 02:42.57 | sergiovel | good idea |
| 02:43.27 | Dandan | besides, I was married by a priest from argentina :) |
| 02:43.35 | sergiovel | you are kidding |
| 02:43.37 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 02:43.41 | schirpich | when I run "pri show span 4" the 'Status' shows "Status: Provisioned, Down, Active |
| 02:43.41 | schirpich | " How do i change it from down to up? it is a new T1 btw |
| 02:43.42 | Dandan | no, really |
| 02:43.48 | sergiovel | wow |
| 02:43.53 | Dandan | :) |
| 02:44.05 | Dandan | very nice guy though :) |
| 02:44.06 | sergiovel | where in las vegas? :) |
| 02:44.11 | Dandan | somewhere from... grenada...? |
| 02:44.20 | sergiovel | we all are nice guys ;) |
| 02:44.22 | Dandan | sergio, naaah, never been to west coast... :) |
| 02:44.33 | sergiovel | ok |
| 02:44.38 | Qwell | You've NEVER been to Vegas? |
| 02:44.41 | Dandan | sergiovel: lol, all priests in argentina? :) (just kidding)? |
| 02:44.56 | Dandan | Qwell: nah, my farthest point west is niagara falls :) |
| 02:45.01 | Qwell | wtf |
| 02:45.09 | Dandan | and farthest south is new jersey :D |
| 02:45.19 | Dandan | (except costa rica and aruba ofc) |
| 02:45.21 | Qwell | You don't get out much... |
| 02:45.21 | sergiovel | prists in argentina are really not that good. sorry I hope i dont offend anyone |
| 02:45.41 | Dandan | Qwell: I do, it is not going west yet :) |
| 02:45.53 | Dandan | i go to europe 2-ice a year and some caribbean :) |
| 02:46.07 | Dandan | sergiovel: yeah, like everywhere, there are exceptions though :) |
| 02:46.18 | Dandan | those, that 'do it' really out of faith |
| 02:46.20 | sergiovel | you are right |
| 02:46.38 | Dandan | (do not confuse with fanatics) :D |
| 02:47.25 | sergiovel | im concern about the ammount of them that have a lot to do with child abuse down here, you know? |
| 02:47.33 | Dandan | I would say that the quote of my lifetime is: "Any form of fanatism is dangerous) |
| 02:47.35 | sergiovel | but there are exceptions |
| 02:47.49 | Dandan | sergiovel: same here |
| 02:48.01 | sergiovel | oh well |
| 02:48.09 | Dandan | there has been a big scandal couple years ago... the diocese of boston had to pay millions :) |
| 02:48.15 | Dandan | but, that's just politics... |
| 02:48.20 | sergiovel | yeah i remember |
| 02:49.00 | Dandan | so, anyway, did you find what you were looking for @ voipsupply? |
| 02:49.49 | schirpich | after installing a new T1 on the last span of my TE410P do i need to restart asterisk to get the new T1 up and active? |
| 02:50.09 | Dandan | don't you? |
| 02:50.20 | Dandan | restart when convenient should help |
| 02:50.30 | Dandan | unless you messed with kernel modules/config |
| 02:51.01 | Jason99 | Qwell: do you code at all? |
| 02:51.11 | Dandan | he prolly is too lazy to code :D |
| 02:51.21 | schirpich | well, span's 1-3 are working and have been for quite some time. we're just expanding and adding a last t1 to this card |
| 02:51.31 | Qwell | Dandan: I am, but I do anyways |
| 02:51.38 | Qwell | Jason99: why? |
| 02:51.48 | Dandan | then i would say you have to restart asterisk |
| 02:51.55 | *** join/#asterisk \lart (i=nunya@neo.jasons.org) |
| 02:52.08 | Dandan | Qwell: lol, I am too :D |
| 02:53.53 | Jason99 | Qwell: Unless this already exists.. it would be a good thing to add. A peice of code to compare the "real" ip compared to the ip received in the SDP. If the IPs are different Asterisk would have a variable in readable in a context that would tell us that it's behind NAT. This way you can make a simple if statement telling it to reinvite or not. |
| 03:00.51 | *** part/#asterisk newsmafia (n=newsmafi@wsip-70-166-5-130.sd.sd.cox.net) |
| 03:01.04 | *** join/#asterisk newsmafia (n=newsmafi@wsip-70-166-5-130.sd.sd.cox.net) |
| 03:07.59 | sergiovel | sorry dandan, I was looking at voipsuppy |
| 03:08.25 | sergiovel | yes, I am trying to find something that would work for a doorphone at my house |
| 03:08.36 | sergiovel | a mixture of fxo and a doorbell |
| 03:08.45 | sergiovel | or just a cheap voip phone |
| 03:09.05 | sergiovel | that i can connect to asterisk |
| 03:09.38 | Dandan | :) |
| 03:09.57 | asterboy | use some paging equipment |
| 03:10.04 | asterboy | I bought mine from ebay |
| 03:10.12 | sergiovel | what did you buy? |
| 03:10.18 | asterboy | Bogen PCM |
| 03:10.28 | sergiovel | hmm |
| 03:10.34 | sergiovel | let me see |
| 03:11.12 | asterboy | you have to hookup a talkback device like a mic. Sound is played over the loud speaker. |
| 03:11.30 | asterboy | nightringer is used to indicate solicitation |
| 03:12.10 | sergiovel | and to open the door...does it have a relay or smth? |
| 03:13.38 | *** join/#asterisk pigpen2 (n=mark@209.159.234.250) |
| 03:20.01 | *** join/#asterisk onixx (i=1000@Quebec-HSE-ppp3620919.sympatico.ca) |
| 03:20.26 | onixx | hi All ! anyone figured out if early dial works with an ata-286, latest firmware |
| 03:23.17 | Qwell | god I love these new mac commercials |
| 03:24.14 | Qwell | "I can run macos 10 or windows" "touche" "No, you aren't using that word right..." "touche" "no, see, you have to make a point first, then I make a counterpoint, then you can say it." "touche" "...you haven't made a point yet." |
| 03:24.44 | onixx | This early-dial thing drives me nuts !! It's been broken for years !!! |
| 03:28.26 | *** join/#asterisk juice (n=juice@209.33.104.91) |
| 03:32.21 | rbd | hey guys, when I run the AGI 'festival' command, it runs fine (the synthesized voice says whatever I supplied), but I get a response "RESULT_LINE: 510 Invalid or unknown command" |
| 03:34.11 | Dandan | anyone knows how to change bearer capabilities on PRI line? |
| 03:38.06 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-85-84.buckeyecom.net) |
| 03:49.52 | *** join/#asterisk froguz (i=froguz@177-138-222-201.adsl.terra.cl) |
| 03:50.24 | froguz | is a real geek question but, does anybody have a cool * wallpaper? |
| 03:51.55 | froguz | i want it for my office PC XD |
| 03:55.24 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 04:00.15 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 04:01.32 | [hC] | Hey guys... I have a DID coming in off a PRI, and im sending it into a macro, and want to execute a command on hangup, however, i only want it done after this particular DID hangs up, is it possible to do like an exten => somedid,h, or smething? |
| 04:01.45 | [hC] | or would i have to use a Goto() to go to a new context to handle it all, or something? |
| 04:02.06 | [hC] | its for fax to email, it needs to do rxfax, then after hangup, send it off via email, just cant think of the right way to do that |
| 04:02.14 | [hC] | wthout hijacking the global h,1, exten. |
| 04:03.14 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 04:05.13 | [hC] | guess i could do it in the macro.. |
| 04:05.21 | pdunkel | hC: I did it in a seperate context with a goto. Works fine. Unless you know the caller id there is no <somedid>,h,1. |
| 04:06.02 | pdunkel | You could not do it in the Macro, since the specials h,t,T,i,o,... are just not relevant within a macro. (They'll never be called) |
| 04:06.23 | [hC] | Yeah... I'll have it call a goto, which then calls the macro |
| 04:06.25 | [hC] | and do it in the goot. |
| 04:06.27 | [hC] | er goto. |
| 04:06.49 | pdunkel | The Only way you could do it in a macro would be to use TryExec(RxFax(<file>)) |
| 04:07.16 | pdunkel | and then check the REMOTEID which will only be set on successful receive. |
| 04:07.40 | *** join/#asterisk WeeZyyy (n=liquidni@rrcs-71-41-69-70.se.biz.rr.com) |
| 04:07.46 | [hC] | yah. |
| 04:07.50 | [hC] | This is easier with the goto. |
| 04:07.51 | [hC] | thanks. |
| 04:08.01 | pdunkel | I just dont get why you would even need a macro at all if you'll do it via goto |
| 04:08.12 | *** join/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-24-49-168-129.albyny.adelphia.net) |
| 04:08.18 | pdunkel | just exten=>,,Goto(fax,s,1) |
| 04:08.20 | pdunkel | then |
| 04:08.30 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:08.31 | pdunkel | exten=>s,1,rxfax(yfiel) |
| 04:08.37 | [hC] | cause i want a generic fax processing macro that will take parameters |
| 04:08.42 | pdunkel | exten=>h,1,System(sendit) |
| 04:08.52 | [hC] | for a handfull of DIDs for various people |
| 04:09.06 | pdunkel | You could do that using Channel Variables or the exten |
| 04:09.19 | pdunkel | Goto(faxctx,<param>,1) |
| 04:09.31 | pdunkel | exten=>_X.,1,rxfx(${EXTEN} |
| 04:09.48 | Jason99 | Qwell: are you still around |
| 04:09.53 | pdunkel | or exten=>Set(PARAM='VALUE') |
| 04:09.57 | [hC] | nod, could do that. |
| 04:09.59 | pdunkel | goto(fax,,) |
| 04:10.10 | pdunkel | rxfax(${PARAM}) |
| 04:10.48 | pdunkel | Well, anycase. Got to rush. |
| 04:10.54 | *** part/#asterisk pdunkel (n=pdunkel@213.235.192.21) |
| 04:17.24 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 04:17.57 | x86 | anyone have multiple fax lines working with spandsp? |
| 04:18.02 | x86 | care to share? |
| 04:18.03 | variable_office | in order to get from zaptel card to sip what is the order of things i need to configure? zaptel.conf and then extensions.conf or is there more than that that i am missing? |
| 04:18.55 | x86 | that's pretty much it... maybe sip.conf too |
| 04:19.08 | x86 | and modules.conf to slim out the stuff you dont need ;) |
| 04:19.24 | x86 | like MGCP, SCCP, H323, etc |
| 04:20.07 | variable_office | when i type zap show channels i still just get pseudo default is that correct? |
| 04:21.03 | variable_office | when i do ztcfg -vvvv i get 1 channels configured, so i asume i did it right? |
| 04:22.35 | x86 | that i'm not sure of |
| 04:22.42 | x86 | i only use SIP and IAX trunks |
| 04:22.52 | variable_office | ah, no zaptel eh? |
| 04:23.49 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 04:24.51 | x86 | right |
| 04:25.12 | variable_office | well how would you make an iax talk to an sip? |
| 04:25.19 | x86 | automagically |
| 04:25.24 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 04:25.29 | x86 | you dont have to manually tell it to convert |
| 04:26.01 | variable_office | no i mean say you have a main iax line coming in, and you want xx number to go to xxx on sip how would you do that? |
| 04:26.43 | x86 | in your inbound context (extensions.conf) |
| 04:26.49 | x86 | you can route on destination |
| 04:28.47 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:39.49 | P-NuT | Hi all. |
| 04:43.54 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 04:44.39 | *** join/#asterisk rushowr (n=rushowr@cpe-24-26-133-106.columbus.res.rr.com) |
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| 04:45.08 | rushowr | hey all, anyone experienced with AEL? Have a stupid issue with context not being merged, and don't know what the issue is |
| 04:47.46 | rushowr | even a pointer as to how I might figure out why it won't parse the extensions.ael file would help...maybe an app out there that test parses the file? |
| 04:47.47 | *** join/#asterisk juice (i=1000@209.33.106.97) |
| 04:49.42 | asterboy | I have an AEL button on my Camera. |
| 04:49.48 | asterboy | Does that count? |
| 04:49.56 | rushowr | lol |
| 04:49.58 | rushowr | I wish |
| 04:50.02 | rushowr | grrrrrrrrrr |
| 04:50.22 | asterboy | I use bash to parse my stuff |
| 04:50.33 | asterboy | awk, sed and grep |
| 04:50.51 | rushowr | Unfortunately, I don't know where I'd begin there |
| 04:51.15 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 04:51.16 | rushowr | oh shit I see where the prob is |
| 04:51.21 | asterboy | well you can start by 'cat extensions.ael |grep <whatever>' |
| 04:51.53 | rushowr | the voip-info page's example for if statements didn't show the need for $[...] |
| 04:51.58 | rushowr | it's in a note at the bottom... |
| 04:52.22 | asterboy | ya voip-info leaves a lot to the imagination. |
| 04:52.51 | rushowr | yep, unfortunately, I haven't found anything else even slightly in-depth on AEL, and I just started using it |
| 04:52.53 | asterboy | the learning curve is at a pace akin to walking in mud |
| 04:52.56 | rushowr | ah well ;-) |
| 04:53.08 | asterboy | ~docs |
| 04:53.09 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:53.10 | rushowr | I've got the standard syntax practically down to a science |
| 04:53.20 | asterboy | any of the other sites might have info |
| 04:53.47 | rushowr | thanks, I hadn't seen astmasters.net before... I got it loaded, thanks mates |
| 04:54.27 | asterboy | ya the owner of that site is on here often...he HATES skype |
| 04:54.34 | rushowr | right on |
| 04:55.11 | rushowr | damnit that still wasn't it |
| 04:55.12 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.static.techex.net.au) |
| 04:55.15 | Mavvie | evening |
| 04:55.19 | rushowr | spoke to soon I guess :D |
| 04:55.39 | rushowr | damned stray ('s |
| 04:56.13 | Mavvie | anybody here (a little bit) experience with app_rxfax? |
| 04:57.08 | asterboy | Anyone on that knows how to fix garbled talk resulting from overlap when people talk at the same time? |
| 04:57.40 | asterboy | Seems * is good to go, it's the VOIP providers. |
| 04:58.03 | *** part/#asterisk rushowr (n=rushowr@cpe-24-26-133-106.columbus.res.rr.com) |
| 04:58.24 | asterboy | wondering if * has a way to mitigate or balance the call |
| 04:59.34 | asterboy | ok, no takers there... |
| 04:59.43 | asterboy | how about services for the blind? |
| 05:00.17 | asterboy | well blind is a misnomer anyway. |
| 05:00.23 | asterboy | no such thing really. |
| 05:00.34 | Mavvie | aha, problem resolved for them then. |
| 05:00.58 | asterboy | but for those with poor optics that can't read a Polycom phone display. |
| 05:01.14 | asterboy | would be nice to have some voice command prompts. |
| 05:01.22 | asterboy | is that a festival thing? |
| 05:01.43 | asterboy | or skinny |
| 05:02.39 | asterboy | dam no takers there either... |
| 05:02.48 | asterboy | ok, fuck it...time to smoke dope |
| 05:03.03 | asterboy | been a long day anyway |
| 05:04.41 | Mavvie | rxfax returns -1 if the user hangs up. |
| 05:05.06 | Mavvie | now I have to find a way to ignore that 01 |
| 05:05.07 | Mavvie | 01 |
| 05:05.08 | Mavvie | -1 |
| 05:08.28 | [hC] | hmm. i sent a fax, came thru great |
| 05:08.35 | [hC] | now all of a sudden like 5 times in a row rxfax fails |
| 05:10.41 | variable_office | is zaptel something that needs to be started seperatly from asterisk, or does it start with asterisk? |
| 05:12.29 | Mavvie | variable_office: it's the device driver for the digium cards. |
| 05:12.44 | Mavvie | so it needs to be there before asterisk is started (unless you don't have digium card) |
| 05:14.13 | *** join/#asterisk DrkShdw (n=scorpio@fl-209-26-20-205.sta.sprint-hsd.net) |
| 05:14.16 | variable_office | Mavvie i am just trying to setup a test box to forward my calls from my generic x100p to a single sip user. do you know if i need to edit any other files but extensions.conf and zaptel.conf? |
| 05:14.39 | Mavvie | zapata.conf is my guess. |
| 05:15.02 | variable_office | whats zapata.conf do? |
| 05:15.27 | Mavvie | it tells asterisk what is in your zaptel.conf |
| 05:15.46 | variable_office | ah, ic |
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| 05:19.10 | *** join/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net) |
| 05:19.29 | websae | does anyone know how to do shared line appearances? |
| 05:19.52 | websae | so polycom a can put call on hold....and polycom b can pick it up |
| 05:20.20 | drray | call parking |
| 05:20.23 | drray | ? |
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| 05:25.03 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 05:26.59 | variable_office | in the extensions.conf file what is the context of each line? |
| 05:30.20 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 05:30.41 | satlan32 | good morning |
| 05:30.46 | satlan32 | anyone heer? |
| 05:31.28 | *** join/#asterisk Splat (n=Splat@220-253-100-70.TAS.netspace.net.au) |
| 05:34.33 | *** join/#asterisk clive- (n=pirch@dsl-146-69-243.telkomadsl.co.za) |
| 05:38.04 | variable_office | anyone here using realtime? |
| 05:39.04 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
| 05:39.14 | MikeJ[Laptop] | nope |
| 05:39.21 | MikeJ[Laptop] | yep |
| 05:39.43 | variable_office | what? |
| 05:39.45 | MikeJ[Laptop] | :D |
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| 05:43.54 | variable_office | i keep getting "pbx.c: Requested contexts didn't get merged" |
| 05:44.03 | variable_office | what does that mean/how can i fix this? |
| 05:46.18 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
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| 05:47.22 | Mavvie | aha. |
| 05:47.40 | Mavvie | the h extension has to happen in the original incoming context |
| 05:47.43 | variable_office | Mavvie you know this error? |
| 05:48.01 | Mavvie | I know about it now. |
| 05:49.08 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 05:49.17 | variable_office | so would that be defualt,h,1,xxx,xxx |
| 05:49.18 | variable_office | ? |
| 05:49.20 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
| 05:50.29 | niZon | damn wiki is down again |
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| 06:18.21 | Mavvie | heh... hylafax/rxfax is funny. |
| 06:18.57 | Mavvie | faxes send with hylafax to an rxfax machine get incorrectly terminated, so that rxfax sees it as a faail transfer. |
| 06:19.00 | Mavvie | failed |
| 06:19.49 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 06:20.35 | *** join/#asterisk lplatypus (n=luked@203.32.185.200) |
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| 06:21.20 | *** part/#asterisk febb (n=febb@201.98.23.242) |
| 06:23.27 | lplatypus | my work is currently installing CISCO VOIP equipment for the internal phone system... will asterisk let me write a bot which can make phone calls? |
| 06:23.42 | MikeJ[Laptop] | yes |
| 06:28.06 | lplatypus | cool... I'm dipping into the Asterisk book now (any other pointers would be welcome) |
| 06:31.29 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 06:33.41 | DrkShdw | lplatypus, check your private messages. I gave you a bit of advice. |
| 06:33.42 | MikeJ[Laptop] | ~docs |
| 06:33.51 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 06:33.59 | MikeJ[Laptop] | read a lot is my advice |
| 06:34.28 | lplatypus | thanks guys |
| 06:35.12 | *** join/#asterisk Keybuk (n=scott@quest.netsplit.com) |
| 06:36.45 | Keybuk | Hi, I'm having a problem and can't find a useful answer in the docs so far ... have asterisk sitting between a SIP/PSTN proxy (voip.co.uk) and a softphone (ekiga) ... the voice from the softphone can be heard fine on a pstn phone; but the pstn phone voice cannot be heart ... doesn't matter which dials first |
| 06:37.02 | Keybuk | if I set up an Echo in the dial plan, it works for both the pstn and softphone |
| 06:38.35 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 06:40.05 | *** join/#asterisk lorinc (n=ang@caracas-4834.adsl.interware.hu) |
| 06:48.06 | *** join/#asterisk Jameno123 (n=james@ddsl-216-68-219-38.fuse.net) |
| 06:49.03 | Jameno123 | point me to the right location, but, when i do "/usr/sbin/asterisk -rx "show queues"" shouldnt it return the exact same thing, as if i did asterisk -r, then typed "show queues" ? |
| 06:49.35 | Jameno123 | cuz, well, its not.. |
| 06:49.49 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 06:49.58 | Jameno123 | it returns about half the info, and cutts it off, right in the middle of the result |
| 06:50.13 | Jameno123 | almost as if theres a max string size, on the "remote unix connect" |
| 06:50.33 | littleball | hello. who can tell me how to reset a specific ZAP channel without restart the asterisk system. Because other channels works fine |
| 06:52.03 | *** join/#asterisk dlynes_home (i=1000@S0106001217014b92.vc.shawcable.net) |
| 06:52.36 | clive- | littleball does soft hangup work ? |
| 06:54.31 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 06:54.59 | Keybuk | hmm, debugging with tcpdump suggests that the voice from the pstn is arriving at the asterisk server, but not getting sent to the softphone ... and the softphone is sending its voice directly back to the pstn gateway, not via the asterisk server |
| 06:55.51 | *** join/#asterisk Eggplants (i=No@dsl-216-155-214-007.cascadeaccess.com) |
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| 07:05.55 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 07:06.24 | cjk | hi, is there a way to have voicemail on an external server so that i dont need to start those mpg processes? |
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| 07:45.10 | dlynes_home | cjk: mpg? what does mpg have to do with voicemail? |
| 07:46.57 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-164-253.lsanca.fios.verizon.net) |
| 07:47.16 | dlynes_home | Gamercjm: no |
| 07:47.23 | Gamercjm | .. |
| 07:47.32 | Gamercjm | wait till i ask something before you say no |
| 07:47.35 | Gamercjm | ;) |
| 07:47.43 | dlynes_home | well, whatever it was you were gonna ask |
| 07:47.46 | dlynes_home | the answer's no :P |
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| 08:10.28 | cjk | dlynes_home: sorry, i meant musiconhold |
| 08:11.20 | dlynes_home | cjk: yeah, you can do streaming |
| 08:12.10 | cjk | ok, i will search for it |
| 08:12.10 | cjk | thanks |
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| 08:15.47 | satlan32 | i guys, |
| 08:16.00 | satlan32 | i need help with running mysql query from the dialplan |
| 08:16.16 | satlan32 | when i use this: MYSQL(Fetch foundRow ${resultid} firstname) |
| 08:16.43 | satlan32 | if there is a result, foundRow will get the value of "1" |
| 08:16.49 | satlan32 | am i correct? |
| 08:17.08 | satlan32 | and then i can use GotoIf($[${foundRow} = 1]?23:20) ? |
| 08:17.59 | dlynes_home | satlan32: just so you know, there's not a lot of active people on right now |
| 08:18.44 | satlan32 | i can see ;) |
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| 08:33.41 | fourcheeze | probs with asterisk logging. It seems to have got stuck logging to a deleted file |
| 08:34.15 | fourcheeze | anyway I can encourage it to unstick short of a restart? |
| 08:37.45 | fourcheeze | is there a module that does the logging? |
| 08:39.07 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 08:39.08 | hads|home | fourcheeze: help logger |
| 08:39.26 | fourcheeze | ahhh |
| 08:39.27 | fourcheeze | thanks |
| 08:39.33 | hads|home | No probs :) |
| 08:39.49 | fourcheeze | so should I run logger rotate each night? |
| 08:40.20 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 08:40.51 | hads|home | If you like. |
| 08:41.40 | fourcheeze | or maybe I should just log to syslog |
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| 08:45.47 | dlynes_home | fourcheeze: i usually do an asterisk -rx "logger rotate" once a week, myself |
| 08:46.15 | fourcheeze | is it possible to get * to put a date next to the log? |
| 08:47.07 | shadebob | hi, I encount a little problem. In my csv cdr, and mysql cdr, dstchannel is truncated. I put a w after dialstr like the 0005861 report in bugs.digium.com |
| 08:48.02 | shadebob | but I have always the same problem. Someone can help me? |
| 08:48.38 | dlynes_home | fourcheeze: it already does |
| 08:49.32 | dlynes_home | fourcheeze: check your dateformat= line in your logger.conf...maybe you've got it set to empty |
| 08:51.41 | fourcheeze | I seem to have: |
| 08:51.42 | fourcheeze | [general] |
| 08:51.42 | fourcheeze | dateformat=%y%m%d-%H%M%S |
| 08:51.47 | Dr-Linux | dlynes_home: how to re-set spa-2100? |
| 08:51.48 | fourcheeze | in my logger.conf |
| 08:51.56 | dlynes_home | Dr-Linux: define reset |
| 08:52.10 | dlynes_home | fourcheeze: yeah...so it should be logging dates and times then |
| 08:52.10 | fourcheeze | * * * * R E S E T |
| 08:52.19 | fourcheeze | for a factory reset |
| 08:52.42 | Dr-Linux | dlynes_home: whre define reset? |
| 08:52.53 | dlynes_home | fourcheeze: check your messages log file |
| 08:53.17 | dlynes_home | Dr-Linux: check your manual. It'll tell you how to do a user reset, factory reset, password reset |
| 08:53.26 | fourcheeze | dlynes_ahh ok it seems to do it now I restarted the logger |
| 08:53.32 | dlynes_home | Dr-Linux: but you never stated which particular reset you're wanting |
| 08:53.56 | Dr-Linux | dlynes_home: i need admin password reset |
| 08:54.17 | dlynes_home | Dr-Linux: so do the user reset then |
| 08:54.34 | dlynes_home | Dr-Linux: you'll need to reprogram your user1, user2 settings though |
| 08:55.05 | Dr-Linux | dlynes_home: how can i do that, if i can't logged in |
| 08:55.26 | dlynes_home | Dr-Linux: plug a phone into phone jack 1 on the sipura 2100 |
| 08:55.26 | Dr-Linux | have this: |
| 08:55.27 | Dr-Linux | http://www.sipura.com/products/spa2100.htm |
| 08:55.34 | dlynes_home | Dr-Linux: unplug the network cable |
| 08:55.47 | dlynes_home | Dr-Linux: hit **** to enter the sipura dtmf menu |
| 08:55.57 | dlynes_home | Dr-Linux: and then I can't remember what the code is for user reset offhand |
| 08:56.22 | dlynes_home | I think it was 73987 or something |
| 08:56.51 | dlynes_home | Dr-Linux: download the user manual from www.sipura.com |
| 08:58.17 | Dr-Linux | dlynes_home: **** doesn't do anything, |
| 08:58.42 | Dr-Linux | network cable is out, and analog phone is connected to line1 |
| 08:59.02 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 08:59.44 | dlynes_home | Dr-Linux: plug the network cable back in and try line 1 again, then |
| 08:59.57 | Dr-Linux | ok |
| 09:00.29 | dlynes_home | Dr-Linux: if you get a pap2 though, i don't think that works with it |
| 09:01.27 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:01.40 | fourcheeze | dlynes_reset is 73738 - spells reset on the phone |
| 09:02.38 | dlynes_home | fourcheeze: yeah, but that's factory reset, not user reset |
| 09:02.42 | Dr-Linux | dlynes_home: you mean **** won't work on spa-2100? |
| 09:02.46 | dlynes_home | fourcheeze: that'll reset everything |
| 09:02.54 | dlynes_home | Dr-Linux: no, it will |
| 09:02.57 | dlynes_home | Dr-Linux: just not on pap2's |
| 09:03.29 | dlynes_home | fourcheeze: a user reset will only reset the pages accessible by the 'user' username |
| 09:03.32 | Dr-Linux | dlynes_home: i pluged back the network cable, but sitll **** doesn't do anything, phone is connected to line1 |
| 09:03.39 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.47 | dlynes_home | Dr-Linux: i don't knwo what to tell you then |
| 09:03.52 | dlynes_home | Dr-Linux: it works just fine for me |
| 09:04.53 | Dr-Linux | dlynes_home: it worked |
| 09:05.07 | Dr-Linux | what's the code to reset admin password? |
| 09:05.23 | dlynes_home | oh yeah...nvm...he needs to reset the admin password |
| 09:05.29 | dlynes_home | not the user settings |
| 09:05.34 | dlynes_home | yeah...so what fourcheeze said |
| 09:05.41 | dlynes_home | i'm half awake here |
| 09:06.03 | Dr-Linux | <PROTECTED> |
| 09:06.24 | dlynes_home | fourcheeze dlynes_reset is 73738 - spells reset on the phone |
| 09:07.15 | Dr-Linux | dlynes_home: yes that's spells reset, but not admin password reset |
| 09:07.31 | dlynes_home | Dr-Linux: if you forgot your admin password |
| 09:07.38 | dlynes_home | Dr-Linux: that's the only way you're going to reset it |
| 09:07.47 | dlynes_home | Dr-Linux: so, next time |
| 09:07.53 | dlynes_home | Dr-Linux: don't forget your admin password |
| 09:08.58 | Dr-Linux | dlynes_home: i did 73738 , it said, option 1 2 confirm or exit |
| 09:09.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 09:09.07 | Dr-Linux | i press 1 2 |
| 09:09.12 | Dr-Linux | it said scucessfull |
| 09:09.19 | dlynes_home | 1 to confirm |
| 09:09.20 | Dr-Linux | not sure what's gone successfull? |
| 09:09.22 | dlynes_home | not 1 2 confirm |
| 09:09.39 | dlynes_home | after hitting 1, you need to hit # to accept |
| 09:09.39 | Dr-Linux | but it said confirmed |
| 09:09.55 | dlynes_home | ok, so log into it now then |
| 09:09.55 | Dr-Linux | but what it will do? |
| 09:09.56 | Dr-Linux | 73738 |
| 09:10.04 | dlynes_home | full system reset of the sipura unit |
| 09:10.08 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 09:10.13 | dlynes_home | so all information on it is defaulted now |
| 09:10.29 | dlynes_home | including passwords, sip registrars, ... |
| 09:11.34 | Dr-Linux | 1 2 is confirmed |
| 09:11.41 | Dr-Linux | you were wrong there |
| 09:11.48 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:11.51 | Dr-Linux | 1 2 confirm |
| 09:11.59 | Dr-Linux | * 2 is exit |
| 09:11.59 | dlynes_home | I've done it enough times |
| 09:12.04 | dlynes_home | It's 1 to confirm |
| 09:12.15 | dlynes_home | the voice on the ivr on the sipura unit is horrible |
| 09:12.16 | Dr-Linux | hhm.. |
| 09:12.31 | Dr-Linux | it is perfectly fine for me |
| 09:12.32 | Dr-Linux | hhm.. |
| 09:12.37 | dlynes_home | I've gone trhough the sipura 2000, 2002, 2100, and pap2 |
| 09:12.46 | dlynes_home | not to mention the sipura 3000 |
| 09:12.56 | Dr-Linux | dlynes_home: now i have only ethernet port on my system, not sure how to access the sipura |
| 09:13.11 | Dr-Linux | dlynes_home: can i access the spa-2100 on local network? |
| 09:13.17 | dlynes_home | Dr-Linux: yes |
| 09:13.23 | dlynes_home | Dr-Linux: hook your phone back up |
| 09:13.23 | Dr-Linux | how? |
| 09:13.30 | dlynes_home | Dr-Linux: then hit **** to enter the ivr menu |
| 09:13.41 | Dr-Linux | done |
| 09:13.44 | dlynes_home | Dr-Linux: then hit 110# to find out what your ip address is |
| 09:13.55 | Dr-Linux | i know the ip |
| 09:14.12 | dlynes_home | so go to http://ip.address.of.sipura/admin/advanced |
| 09:14.22 | dlynes_home | then type in admin for the username, and no password |
| 09:14.34 | Dr-Linux | Opss |
| 09:14.44 | Dr-Linux | after resetting, old IP address is not working |
| 09:15.08 | dlynes_home | i told you |
| 09:15.17 | dlynes_home | ****, 110# to find out what the ip address is |
| 09:15.23 | dlynes_home | when you default it, it goes to dhcp |
| 09:15.49 | Dr-Linux | aww |
| 09:15.56 | Dr-Linux | dlynes_home: it says 0.0.0.0 |
| 09:15.57 | Dr-Linux | :S |
| 09:16.00 | littleball | hello, how to force end an active channel? |
| 09:16.04 | littleball | ZAP channel |
| 09:16.36 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 09:17.11 | dlynes_home | Dr-Linux: i guess you're not running a dhcp server, or your ethernet cable is unplugged |
| 09:17.48 | dlynes_home | littleball: soft hangup 1-1 |
| 09:17.53 | dlynes_home | littleball: or whatever your channel is |
| 09:17.53 | Dr-Linux | i'm running DHCP |
| 09:18.09 | dlynes_home | littleball: erm soft hangup Zap/1-1 i mean |
| 09:19.55 | dlynes_home | Dr-Linux: unplug the ethernet cable then, and plug it back in |
| 09:20.02 | dlynes_home | Dr-Linux: then unplug the power and plug it back in |
| 09:20.09 | dlynes_home | Dr-Linux: then after the lights finish blinking on it |
| 09:20.16 | Dr-Linux | :) |
| 09:20.20 | *** join/#asterisk JDofFED (n=jd@c-69-243-134-140.hsd1.in.comcast.net) |
| 09:20.23 | dlynes_home | Dr-Linux: do the **** and the 110# to find out what hte ip address is, again |
| 09:20.32 | Dr-Linux | thanks |
| 09:20.49 | dlynes_home | Dr-Linux: if it still doesn't work, do a **** and then 100# to find out whether dhcp is enabled or not |
| 09:21.07 | dlynes_home | Dr-Linux: if you need to enable it, you can do a **** and then 101#, and then 1# to enable |
| 09:21.31 | Dr-Linux | dlynes_home: it's enabled |
| 09:21.44 | dlynes_home | Dr-Linux: did you check? |
| 09:21.52 | dlynes_home | Dr-Linux: or are you assuming? |
| 09:22.46 | Dr-Linux | i checked |
| 09:23.26 | JDofFED | i am having problems install asterisk on a freebsd port |
| 09:24.19 | dlynes_home | JDofFED: on ports tree? |
| 09:25.40 | JDofFED | yes |
| 09:25.54 | JDofFED | i even updated the ports via cvsup |
| 09:26.05 | *** join/#asterisk Yalla-One (n=yallaone@unaffiliated/yalla-one) |
| 09:26.39 | wintix | sorry to bother, but i need to aks a simple question. what does the ${EXTEN:3} in the following line do: exten => _8.,2,Dial(IAX2/15...@fwdOUT/${EXTEN:3},60,r) |
| 09:26.52 | Dr-Linux | dlynes_home: problem is DHCP server, that assigns IP address on MAC base |
| 09:26.55 | dlynes_home | cd /usr/ports/misc/zaptel ; make ; make install ; cd /usr/ports/misc/libpri ; make ; make install ; cd /usr/ports/comm/asterisk ; make ; make install or something like that |
| 09:26.56 | puzzled | JDofFED: iirc there is an asterisk-bsd list. perhaps they can help you. check the digium website |
| 09:27.03 | Dr-Linux | not sure about spa-2100's mac |
| 09:27.09 | dlynes_home | Dr-Linux: so add the mac of the sipura to your list |
| 09:27.26 | Dr-Linux | yeah |
| 09:27.27 | Dr-Linux | okey |
| 09:27.32 | dlynes_home | Dr-Linux: the mac is on the label stuck to the underside of the sipura unit |
| 09:27.35 | JDofFED | •puzzled• okay |
| 09:27.51 | Yalla-One | What's the best "mini-implementation" of Asterix for home use (one small family) to control what times other people can call us, and basic features only? |
| 09:27.53 | dlynes_home | JDofFED: iow, compile and install zaptel, then libpri, then asterisk |
| 09:28.05 | drray | Yalla - asterisk from svn |
| 09:28.13 | hads|home | wintix: It strips the first three digits from the number stored in ${EXTEN} |
| 09:28.23 | JDofFED | •dlynes_home• okay, I will try that |
| 09:28.26 | dlynes_home | Yalla-One: I would try freepbx for that |
| 09:28.31 | dlynes_home | Yalla-One: try #freepbx |
| 09:28.49 | dlynes_home | Yalla-One: it's not well suited to complex installations, but for something simple like that, it should be just fine |
| 09:29.01 | Yalla-One | dlynes_home, Thanks - from what I can see it seems like the full-blown asterisk might be overkill on a small linux box for a 3 person family :) |
| 09:29.16 | Yalla-One | will check #freepbx - thanks much for quick pointers! |
| 09:29.24 | dlynes_home | Yalla-One: freepbx is still full blown asterisk, but it includes a nice easy to use web management system |
| 09:29.41 | dlynes_home | Yalla-One: if you install Asterisk@Home, it includes freepbx and a bunch of other stuff all from one nice easy to install cd |
| 09:30.11 | Yalla-One | dlynes_home, I don't want asterisk@home as it comes with its own distribution, and I want to run it just as another service on my fine-tuned Slackware server :) |
| 09:30.24 | dlynes_home | Yalla-One: ah |
| 09:30.30 | wintix | hads|home: are there the numbers after the 8 stored? (if i set .8_ ) |
| 09:30.32 | dlynes_home | Yalla-One: then you want asterisk, not amp |
| 09:30.41 | dlynes_home | Yalla-One: amp is a huge freaking pain in the ass to set up on slackware |
| 09:30.57 | dlynes_home | Yalla-One: because it's got huge dependency issues |
| 09:30.59 | Yalla-One | dlynes_home, Uhm - amp ? |
| 09:31.06 | dlynes_home | Yalla-One: amp/freepbx |
| 09:31.13 | dlynes_home | Yalla-One: amp is the old name, freepbx is the new name |
| 09:31.22 | dlynes_home | Yalla-One: Asterisk Management Portal |
| 09:31.23 | Yalla-One | aha - OK - so I'll just go with asterisk then and install it as a package under slackware. |
| 09:31.27 | shadebob | Anyone with cdr dstchannel truncated? |
| 09:31.34 | littleball | hello, for SIP, whether the REGISTER is authenticated or INVITE should be authenticated? |
| 09:31.48 | dlynes_home | Yalla-One: if you want, i have a package already made for slackware 10.2 |
| 09:32.00 | Yalla-One | I don't want web-GUIs etc - all I want is to configure it easily via some .conf files and get it going for managing my incoming phones (ie block sales people, and stop anyone calling after 2300 at night) |
| 09:32.03 | hads|home | wintix: eg. if you dialled 8223000 then ${EXTEN:3} would be 3000 |
| 09:32.09 | dlynes_home | Yalla-One: you'll need to compile and install zaptel and libpri though |
| 09:32.14 | Yalla-One | dlynes_home, That would be most kind of you. Does it include a slackBuild ? |
| 09:32.24 | dlynes_home | Yalla-One: nah...I just built it myself |
| 09:32.33 | wintix | hads|home: thanks for your help |
| 09:32.40 | hads|home | No probs :) |
| 09:32.40 | dlynes_home | Yalla-One: It's got all the asterisk-addons, sounds, documentation, ... |
| 09:32.43 | Yalla-One | dlynes_home, ./configure&&make&&checkinstall? |
| 09:32.50 | dlynes_home | Yalla-One: heh...not quite |
| 09:32.53 | Yalla-One | :) |
| 09:33.01 | dlynes_home | Yalla-One: no such thing as configure for asterisk, unless your'e using turnk |
| 09:33.07 | Yalla-One | dlynes_home, If you have it available - yes please :) |
| 09:33.13 | dlynes_home | Yalla-One: and even then, it's non-gnu autoconfigure |
| 09:33.48 | Yalla-One | dlynes_home, I don't have any PRI stuff - I've got IP telephony from an IP only operator, and will terminate SIP directly in asterisk, so probably won't need libpri. zaptel I never head of... will google |
| 09:34.26 | littleball | hello, for SIP, whether the REGISTER is authenticated or INVITE should be authenticated? |
| 09:34.45 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:35.44 | dlynes_home | Yalla-One: you can do cd /usr/local/src/zaptel-1.2.6 ; make ; make install ; cd /usr/local/src/asterisk-1.2.9.1 ; make ; make install ; make samples ; make progdocs ; cd doc ; mkdir -p /usr/doc/asterisk-1.2.9.1 ; cp -R . /usr/doc/asterisk-1.2.9.1 ; cd /usr/local/src/asterisk-sounds-1.2.1 ; make install ; cd /usr/local/src/asterisk-addons-1.2.3 ; make ; make install ; cd format_mp3 ; make ; make install ; cd ../asterisk-ooh323c ; . |
| 09:35.44 | dlynes_home | /configure --prefix=/usr ; make ; make install |
| 09:36.10 | dlynes_home | Yalla-One: oh yeah...forgot libpri-1.2.3 in between zaptel and asterisk |
| 09:36.15 | Yalla-One | "that's it" ? ;) |
| 09:36.21 | dlynes_home | yeah |
| 09:37.00 | dlynes_home | Yalla-One: anyways...you'll probably want zaptel so you can install ztdummy, and you'll need crc_ccitt and rtc modules installed from your kernel |
| 09:37.06 | Yalla-One | dlynes_home, Maybe I should try to make a slackBuild from it ... |
| 09:37.17 | Yalla-One | OK - thanks - and ztdummy is? |
| 09:37.48 | dlynes_home | Yalla-One: if you want to install any telephony cards, you'll want to compile libpri as well (libpri isn't used by them, but chan_zap.so won't get compiled unless you have libpri installed) |
| 09:38.10 | dlynes_home | Yalla-One: it's a timing source for when you don't have any telephony hardware installed |
| 09:38.36 | hads|home | chan_zap compiles without libpri |
| 09:39.11 | dlynes_home | hads|home: so why would it not compile on certain systems then, until after you've compiled and installed libpri? |
| 09:39.25 | dlynes_home | hads|home: or is it zaptel drivers installed that it's dependent on? |
| 09:39.29 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:39.46 | hads|home | That I do not know, but I've never installed libpri and use TDM400's often. |
| 09:40.02 | dlynes_home | hads|home: ah...must be dependent on the drivers being installed then |
| 09:40.27 | hads|home | I just compile zaptel and install and then compile asterisk and install. |
| 09:41.39 | dlynes_home | Yalla-One: also, if you're using linux 2.6.x, make sure udev 071 or newer is installed |
| 09:42.00 | dlynes_home | Yalla-One: run udevinfo -V to see which version you're running |
| 09:42.23 | hads|home | dlynes_home: Sorry, I missed one of you lines. Yes, it is the Zaptel drivers that chan_zap is dependant on. |
| 09:44.16 | Dr-Linux | dlynes_home: my sipura device has grabed an ip address via dhcp 192.168.0.4 |
| 09:44.33 | Dr-Linux | but when i do http://192.168.0.4/admin/advanced |
| 09:44.44 | Dr-Linux | Not Found |
| 09:44.44 | Dr-Linux | The requested URL /admin/advanced was not found on this server. |
| 09:45.52 | dlynes_home | Dr-Linux: how about http://192.168.0.4/admin/ ? |
| 09:46.00 | Dr-Linux | dlynes_home: same |
| 09:46.11 | Dr-Linux | dlynes_home: but i guess, it won't work from network |
| 09:46.20 | Dr-Linux | dlynes_home: bcoz once already i have expereinece with that |
| 09:46.35 | Dr-Linux | dlynes_home: but you said, so i tried again |
| 09:46.39 | Dr-Linux | but nothing seems to work |
| 09:46.41 | dlynes_home | Dr-Linux: how about telnet 192.168.0.4 80 and then type in GET /admin/advanced HTTP/1.0? |
| 09:46.41 | Nugget | telnet is eeeeeeevil! |
| 09:46.46 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 09:46.49 | Yalla-One | dlynes_home, Am on latest -current so I have udev 071. Thanks |
| 09:47.01 | Dr-Linux | :S |
| 09:47.03 | Dr-Linux | lemme try |
| 09:47.16 | dlynes_home | Dr-Linux: and then show me what it says for the http server version |
| 09:47.18 | SheriF_WorK | <PROTECTED> |
| 09:47.24 | SheriF_WorK | and my asterisk only supports g723 |
| 09:47.28 | SheriF_WorK | what is slin anyway !? |
| 09:47.42 | dlynes_home | SheriF_WorK: there's no legal g723 codec transcoder for asterisk |
| 09:48.26 | dlynes_home | slin is signed linear |
| 09:48.48 | Gamercjm | if i got a "PCI X100P FXO" does that mean i can use my analog phone? or do i still need to get a phone adapter? |
| 09:49.11 | JDofFED | what is this looking for? ===> Patching for zaptel-0.11 |
| 09:49.11 | JDofFED | -e: not found |
| 09:49.11 | JDofFED | *** Error code 127 |
| 09:49.14 | dlynes_home | Gamercjm: it gives you an analog passthrough port, not an fxs port |
| 09:49.27 | dlynes_home | JDofFED: it means you've got a bug in your shell script |
| 09:49.39 | JDofFED | nice, what shell should I be using? |
| 09:49.48 | dlynes_home | JDofFED: probably bash or bourne |
| 09:49.54 | Dr-Linux | dlynes_home: http://pastebin.com/710359 |
| 09:49.55 | JDofFED | ahh, I was using sh |
| 09:50.17 | dlynes_home | JDofFED: your sh is a symbolic link? |
| 09:50.24 | dlynes_home | JDofFED: or is it bourne shell? |
| 09:50.24 | JDofFED | no |
| 09:50.35 | dlynes_home | ah...it's bourne shell |
| 09:50.36 | JDofFED | default root shell, i can't remember which |
| 09:50.40 | Dr-Linux | dlynes_home: are you sure, it can be accessible via network? |
| 09:50.41 | dlynes_home | yeah...try bash instead |
| 09:50.45 | dlynes_home | see if that fixes it |
| 09:50.51 | JDofFED | okay, ty |
| 09:51.00 | dlynes_home | Dr-Linux: of course it can |
| 09:51.28 | Dr-Linux | dlynes_home: so what my PB says? |
| 09:51.29 | *** join/#asterisk ghenry (n=ghenry@81-174-209-84.pth-as2.dial.plus.net) |
| 09:51.32 | dlynes_home | Dr-Linux: you're a dumbass |
| 09:51.39 | dlynes_home | Dr-Linux: that's your redhat server, not your sipura unit |
| 09:51.49 | *** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 09:51.51 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 09:52.06 | JDofFED | : that did not work, should I be using make or gmake? |
| 09:52.10 | swytch | someone with experience using the Cdr events in the manager API? |
| 09:52.20 | dlynes_home | JDofFED: i would imagine gmake |
| 09:52.40 | JDofFED | : Makefile:26: *** missing separator. Stop. |
| 09:53.02 | JDofFED | : that does not work at all |
| 09:53.34 | dlynes_home | JDofFED: which freebsd version are you using, anyways? |
| 09:53.36 | *** join/#asterisk ghenry (n=ghenry@81-174-209-84.pth-as2.dial.plus.net) |
| 09:54.00 | Dr-Linux | dlynes_home: so what you suggest now? |
| 09:54.14 | dlynes_home | Dr-Linux: telling me the correct ip address? |
| 09:54.17 | *** join/#asterisk LH-euhost (n=LH-euhos@L6c13.l.strato-dslnet.de) |
| 09:54.25 | dlynes_home | Dr-Linux: instead of the ip address of your redhat server? |
| 09:54.25 | Dr-Linux | yes |
| 09:54.41 | Dr-Linux | sipura device ip address is 192.168.0.4 |
| 09:54.45 | dlynes_home | it's not |
| 09:54.53 | dlynes_home | look at your pastebin output |
| 09:55.03 | JDofFED | dlynes_home : FreeBSD asterisk1.local 5.4-RELEASE FreeBSD 5.4-RELEASE #0: Sun May 8 10:21:06 UTC 2005 |
| 09:55.04 | dlynes_home | it's clear as day that 192.168.0.4 is a redhat server, and not a sipura unit |
| 09:55.20 | swytch | hello, with my 1.0.7-BRIstuffed-0.2.0-RC7k i have problems with my M-APIs Cdr durations. Some calls seem to overlap. |
| 09:55.24 | dlynes_home | JDofFED: ah...yeah...i've only ever run asterisk on freebsd 6.0 |
| 09:55.37 | Dr-Linux | dlynes_home: it's clear that 192.168.0.4 is sipura device ip address |
| 09:55.46 | dlynes_home | Dr-Linux: look at your pastebin output |
| 09:55.58 | JDofFED | dlynes_home: intresting. It seems like the makefiles are broken |
| 09:56.03 | Dr-Linux | red hat is production server and it has 192.168.0.106 IP, that's asterisk server :) |
| 09:56.04 | X-Rob | # |
| 09:56.04 | X-Rob | <address>Apache/2.0.46 (Red Hat) Server at i2c-RHEL-B.i2c.com Port 80</address> |
| 09:56.05 | X-Rob | Yeah |
| 09:56.09 | X-Rob | That's a sipura all right |
| 09:56.10 | X-Rob | *laff* |
| 09:56.22 | Dr-Linux | that's not sipura |
| 09:56.23 | dlynes_home | X-Rob: yeah...he's very stubborn |
| 09:56.31 | dlynes_home | Dr-Linux: now you agree with me |
| 09:56.34 | dlynes_home | Dr-Linux: make up your mind |
| 09:57.02 | X-Rob | dlynes_home, it could be that he's got a transparent proxy grabbing everything. But I think it's pretty unlikely. |
| 09:57.06 | Dr-Linux | dlynes_home: well, i just typed what you told me |
| 09:57.09 | X-Rob | why didn't he use 'GET' anyway |
| 09:57.14 | Dr-Linux | i telenet the sipura device |
| 09:57.21 | Dr-Linux | and i issued the command you given |
| 09:57.25 | Dr-Linux | let me try from windows |
| 09:57.35 | dlynes_home | Dr-Linux: yeah, you telnetted the "sipura" device, and you connected to apache 2.0 on a redhat machine |
| 09:57.57 | dlynes_home | Dr-Linux: type nslookup i2c-rhel-b.i2c.com |
| 09:58.01 | X-Rob | no |
| 09:58.02 | X-Rob | that won't work |
| 09:58.06 | dlynes_home | Dr-Linux: see if you don't get 192.168.0.4 back |
| 09:58.08 | X-Rob | who knows what his DNS is |
| 09:58.23 | dlynes_home | yeah, true |
| 09:58.52 | X-Rob | Dr-Linux, What -exactly- makes you think ghat 192.168.0.4 is the sipura? You've gone ****110# into the phone? |
| 09:58.55 | dlynes_home | yeah...just realized the 'GET' command was missing before the path |
| 09:59.01 | swytch | hello, i wander what extension i can add to enshure that my Cdr durations are not passing the reality. |
| 09:59.05 | dlynes_home | X-Rob: of course not |
| 09:59.07 | swytch | => h,1,Hangup ? |
| 09:59.11 | dlynes_home | X-Rob: he wants to keep us guessing :p |
| 09:59.20 | X-Rob | swytch, I think it's ResetCDR |
| 09:59.27 | X-Rob | uh |
| 09:59.30 | X-Rob | 'passing the reality' |
| 09:59.39 | X-Rob | dlynes_home, heh |
| 10:00.08 | swytch | X-Rob: ok. durations are sometimes evidently too big in the M-API Cdrs. they overlap the next call from the same caller. |
| 10:00.18 | Dr-Linux | dlynes_home: yes i'm 100% sure |
| 10:00.32 | dlynes_home | Dr-Linux: well, you need to fix your network then |
| 10:00.34 | hads|home | lol |
| 10:00.37 | Dr-Linux | let me ping the 192.168.0.4 and unplug the sipura cable |
| 10:00.44 | dlynes_home | Dr-Linux: because you telnetted into a redhat machine, not a sipura unit |
| 10:01.19 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 10:01.40 | SheriF_WorK | dlynes_home: sorry i didn't get it :-s? |
| 10:01.49 | X-Rob | swytch, 'ResetCDR(w)' I believe should do it |
| 10:02.01 | SheriF_WorK | dlynes_home: i'm trying to integrate asterisk with Multitech voip device |
| 10:02.14 | wintix | i am trying to link two asterisk servers together. on the one side, i configured an extension 50 with a host=dynamic and a password, this pbx has DMZ to make sure nothing comes in it's way. on the other side, i have the following entry in sip.conf: register=> 50:password@host but this pbx doesn't register with the first pbx, any ideas what might be the problem? |
| 10:02.19 | Dr-Linux | dlynes_home: that's very strange |
| 10:02.27 | X-Rob | wintix, why register? |
| 10:02.48 | X-Rob | you only need to register if there's not a fixed IP address |
| 10:02.55 | Dr-Linux | ok sure |
| 10:02.56 | Dr-Linux | brb |
| 10:02.57 | dlynes_home | SheriF_WorK: the multitech only does g723? no other codecs? |
| 10:03.00 | wintix | there is no fixed ip adress, X-Rob |
| 10:03.09 | X-Rob | dlynes_home, OMG! He finally believed you! |
| 10:03.27 | X-Rob | <address>Apache/2.0.46 (Red Hat) Server at i2c-RHEL-B.i2c.com Port 80</address> |
| 10:03.35 | dlynes_home | No, dood!!!! |
| 10:03.36 | X-Rob | I can't see how ^^^ that can't give it away |
| 10:03.39 | swytch | X-Rob: => h,1,ResetCDR(w) ? |
| 10:03.44 | X-Rob | swytch, yeah |
| 10:03.54 | dlynes_home | Apache/2.0.46 (Red Hat) Server is a synonym for Sipura 2100 |
| 10:03.58 | X-Rob | Aaaaah |
| 10:04.08 | X-Rob | That must be the new confuse-o-matic firmware |
| 10:04.11 | X-Rob | I've been avoiding that one. |
| 10:04.20 | dlynes_home | It's Cisco's ultra new secret codename for their new Sipura product |
| 10:04.28 | dlynes_home | But shhhhh...don't let anyone else know |
| 10:04.29 | X-Rob | l33t c1sc0. |
| 10:04.38 | SheriF_WorK | dlynes_home: g723 / g729 / net coder |
| 10:04.48 | wintix | ah, i think i get my problem. is it ok to register the one pbx with the other or do i have to register them both against each other? |
| 10:04.50 | dlynes_home | SheriF_WorK: tell it to use g729 then, instead |
| 10:05.02 | dlynes_home | SheriF_WorK: asterisk only supports g723 in passthrough mode |
| 10:05.06 | X-Rob | wintix, registration is to tell the other machine where you are, so it can send calls to you. |
| 10:05.07 | SheriF_WorK | i know that g729 i should buy it for asteirks 10 USD per chanel ? |
| 10:05.18 | dlynes_home | SheriF_WorK: exactly...that's still pretty cheap |
| 10:05.33 | swytch | X-Rob: ah, i see from the "doc" that that will then erase the duration from an eventual existing (Up) call.. |
| 10:05.44 | X-Rob | switch, yes |
| 10:05.45 | SheriF_WorK | dlynes_home: will talk with boss about it :) |
| 10:05.45 | swytch | X-Rob: ..from the _same_ caller? am i correct? |
| 10:06.00 | X-Rob | from the call that's doing that stuff in the dialplan |
| 10:06.16 | X-Rob | it's very hard to affect another call from the dialplan |
| 10:06.45 | dlynes_home | SheriF_WorK: well, if he doesn't mind spending a bit of cash on the multitech, i can't see why he wouldn't splurge for about ten g729 licenses or so |
| 10:07.03 | swytch | so if a new call from the same callerid is hitting asterisk, then asterisk will clear the duration from an existing call from that caller and store that cleared Cdr. |
| 10:07.16 | X-Rob | swytch, no |
| 10:08.43 | swytch | i sont understand from the "doc" what the command does exactly.. do you know it? |
| 10:08.56 | swytch | <PROTECTED> |
| 10:10.29 | X-Rob | swytch, it ensures that the CDR is closed when the call is hung up. |
| 10:10.45 | *** join/#asterisk Sonderblade (n=mah@host-213.131.147.169.addr.tdcsong.se) |
| 10:11.04 | swytch | X-Rob: but i wonder why i have my problem in the first place. is there a way to fix the real problem? seems like calls are not preperly hung up. |
| 10:11.30 | *** join/#asterisk SparFux (n=player@e182017229.adsl.alicedsl.de) |
| 10:11.50 | swytch | X-Rob: so i actually needs to use the ResetCDR without the w option (since i want to know about the duration). |
| 10:12.54 | fourcheeze | my provider uses g729 which is what I want mostly. Is it possible to ask for g711 part way through a call, and if so how? |
| 10:13.28 | swytch | seems like anyone using asterisk as src for CDR should have =>h,1,Hangup and h,2,ResetCDR |
| 10:13.39 | hads|home | fourcheeze: I don't believe it's possibly to change codecs on the fly at present. |
| 10:13.56 | fourcheeze | hads|home: is that an asterisk limitation or a SIP one? |
| 10:14.17 | swytch | fourcheeze: SIP lets you change stuff in SDP with re-INVITE |
| 10:14.42 | fourcheeze | but there's no way to get * to do this? |
| 10:14.47 | fourcheeze | except to write a module perhaps |
| 10:15.14 | swytch | fourcheeze: so (i didnt know) apparently * (which _can_ do re-INVITE) dont let you change that specific thing, eg. codec |
| 10:15.30 | hads|home | fourcheeze: I'm not sure. I'll need to do some reading. |
| 10:15.48 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 10:16.04 | swytch | X-Rob: thanks for grat help. and -- am i correct in my "conclusion" that it seems like anyone using asterisk as src for CDR should have =>h,1,Hangup and h,2,ResetCDR |
| 10:16.58 | SheriF_WorK | dlynes_home: yes he should ;-) |
| 10:17.05 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 10:19.27 | X-Rob | 'h' is already after it's hung up |
| 10:19.32 | X-Rob | 'h' is already after it's hung up <-- swytch |
| 10:20.31 | swytch | so it would be better to do like .,1,Dial + .,2,Hangup + .,3,ResetCDR ? |
| 10:20.55 | fourcheeze | I've never needed to ResetCDR - am I missing somethng? |
| 10:21.13 | dlynes_home | Dr-Linux: figured out where your sipura unit is yet? |
| 10:21.47 | swytch | fourcheeze: my problem is that i see in my Cdrs (event from the Manager API) that according to the durations of my calls i have calls from the same caller overlapping. |
| 10:25.51 | swytch | but maybe all my problem is that i didnt have an explisit Hangup after the Dial. i had h,1,Hangup wich i now understand is nonsense |
| 10:26.08 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 10:26.53 | wintix | hm. i try to register a pbx with another, but i get the following error: Got SIP response 481 "Call Leg/Transaction Does Not Exist" |
| 10:27.47 | wintix | user and password are correct, foreign pbx is reachable and has an entry for the connectiong pbx in the sip.conf |
| 10:28.16 | wintix | any ideas what i do wrong? |
| 10:32.40 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 10:35.09 | LH-euhost | hello all, i have a problem with callback on *. When i get called back, i am able to enter a number like "01733516818". But often the * recognices a wrong number like "017335168118" (double 1, although i never entered two 1 digits). any hints? |
| 10:35.19 | pjo | wintix: that response typically means there's no matching INVITE for a BYE or CANCEL packet. do you have any firewalls inbetween you and them? |
| 10:35.57 | RoyK | mmm. strawberries |
| 10:36.14 | Aurs | RoyK: .no or .be? :P |
| 10:36.19 | RoyK | .no |
| 10:36.33 | Aurs | hmm.. so it's not too early for .no strawberries? |
| 10:36.38 | RoyK | not at all |
| 10:36.48 | RoyK | 28 degrees in oslo the other day |
| 10:36.52 | Aurs | I'm used to living in the cold north you know |
| 10:37.02 | Aurs | I know. I live in Oslo too |
| 10:37.02 | Aurs | :P |
| 10:37.33 | RoyK | stemmer, du er norsk.... |
| 10:37.34 | RoyK | jeje |
| 10:37.44 | Aurs | yes, i'm norsk |
| 10:37.50 | Aurs | hehe |
| 10:37.50 | Dr-Linux | dlynes_home: i just change the ip for spa device |
| 10:37.56 | Dr-Linux | it's not 192.168.0.243 |
| 10:38.24 | swytch | RoyK,Aurs: ich auch |
| 10:38.39 | Aurs | swytch :) |
| 10:38.51 | Dr-Linux | dlynes_home: |
| 10:38.52 | Dr-Linux | [root@LHR-PBX root]# telnet 192.168.0.243 80 |
| 10:38.52 | Dr-Linux | Trying 192.168.0.243... |
| 10:38.52 | Dr-Linux | telnet: Unable to connect to remote host: Connection refused |
| 10:39.19 | swytch | RoyK,Aurs: /j?eg oxo/ (c8 |
| 10:39.21 | RoyK | swytch: nei, ikke tysk |
| 10:40.06 | swytch | RoyK: ikkj?e tusj nei(n) |
| 10:40.19 | Aurs | lol |
| 10:41.18 | RoyK | rotfl |
| 10:41.41 | Dr-Linux | any idea? apache is not running on spa-2100 |
| 10:41.54 | swytch | yeah. i live_d_ in oslo too. now i live on the french riviera (c8 |
| 10:42.44 | Aurs | hva har tusjer med saken å gjøre.. hehe |
| 10:42.52 | swytch | trykkleif |
| 10:43.04 | Dr-Linux | brb |
| 10:43.16 | swytch | *tysker. jeg har selvsagt mitt norkse tastebrett |
| 10:44.11 | Aurs | french riviera... senior citizen? ;) |
| 10:44.31 | Aurs | hmm.. no.. that's spain.. my bad |
| 10:44.35 | swytch | nope. im anno 76 |
| 10:46.38 | swytch | im here trying to cope with a few callshops (calling to algerie etc) using asterisk, and audiocodes GW in the callshops. |
| 10:49.18 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 10:50.44 | swytch | MatsK: ennu en skandinav (c8 |
| 10:51.00 | MatsK | Jep |
| 10:58.02 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 10:59.40 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 11:04.21 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 11:04.26 | Dr-Linux | dlynes_home: around? |
| 11:10.53 | X-Rob | Dr-Linux, you manage to figure out the correct IP for your sipura yet? |
| 11:11.25 | Dr-Linux | X-Rob: yes, but dlynes_home was wrong .. |
| 11:11.39 | Dr-Linux | spa-2100 can't be accesible from network |
| 11:11.53 | Dr-Linux | i plugged the PC cable and i accessed it |
| 11:11.53 | X-Rob | no he wasn't. that machine you were connected to was a linux box, running red hat and apache. |
| 11:12.11 | Dr-Linux | and it's now confiugred with 2 different sip servers |
| 11:12.31 | Dr-Linux | X-Rob: yes that's correct |
| 11:12.35 | swytch | X-Rob: could my problem in the Cdr come from the fact that i often do reload * ? |
| 11:12.38 | Dr-Linux | but later i changed the IP address |
| 11:12.50 | X-Rob | swytch, possibly yes |
| 11:13.20 | Dr-Linux | X-Rob: everything is configured |
| 11:13.21 | swytch | X-Rob: i do a "sip reload" every 15 min from cron to synch my user-list |
| 11:13.29 | Dr-Linux | X-Rob: 2 problems: |
| 11:13.39 | X-Rob | switch, hmmm. Dunno. |
| 11:13.49 | X-Rob | you might be better to be using realtime |
| 11:14.06 | Dr-Linux | 1. how can i access the spa-2100 from network, however i can access it via PC :) |
| 11:14.19 | Ciber311 | hey guys |
| 11:14.19 | swytch | X-Rob: my "sip reload" is causing bad Cdrs then? |
| 11:14.30 | X-Rob | swytch, I don't know. |
| 11:14.34 | Dr-Linux | X-Rob: 2nd, when i dial any number it takes some time to send the number |
| 11:14.36 | Ciber311 | i hear like a crackling type noise in all my sip calls |
| 11:14.45 | Ciber311 | through like 3 providers |
| 11:14.49 | X-Rob | Dr-Linux, dial '#' to finish |
| 11:14.49 | Ciber311 | could it be the codec? |
| 11:14.52 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:14.58 | X-Rob | and read up on sipura dialplans |
| 11:15.12 | X-Rob | with your not being able to access it, no idea. |
| 11:15.48 | swytch | Dr-Linux: your problem is not finding the correct ip address of a device? |
| 11:15.59 | Dr-Linux | X-Rob: i can access it when i connect PC cable, and i access it via internal IP |
| 11:16.13 | Dr-Linux | switch: no that's not a problem |
| 11:16.34 | Dr-Linux | only thing, i can't access my spa-2100 from local network |
| 11:16.46 | swytch | NAT problem? |
| 11:17.22 | Dr-Linux | switch: what do you mean NAT problem? |
| 11:17.31 | Dr-Linux | both spa and my system is on same subnet |
| 11:17.34 | Dr-Linux | local subnet |
| 11:23.35 | *** join/#asterisk samourai1 (n=shadebob@84.16.31.10) |
| 11:25.52 | samourai1 | plz can u help me i have a sip provider registration connected to asterisk when i tape sip show registry the line is registered ,i can make outbound calls but when i call to my number |
| 11:25.58 | samourai1 | it doesn't work |
| 11:26.36 | samourai1 | it seems not to be registred |
| 11:30.29 | samourai1 | plz can anyone help me |
| 11:34.00 | mitcheloc | X-Rob: i thought you don't like godaddy? |
| 11:34.06 | X-Rob | I don't. |
| 11:34.10 | X-Rob | but they're still cheap. |
| 11:34.21 | mitcheloc | the new name is registered with them no? |
| 11:34.26 | mitcheloc | haha, true |
| 11:34.40 | mitcheloc | did you ever recover your files? |
| 11:35.43 | *** join/#asterisk kreenaa (n=dsfghrft@62.76.244.194) |
| 11:35.49 | kreenaa | HI |
| 11:35.50 | kreenaa | there |
| 11:36.36 | kreenaa | anyone here to help me |
| 11:36.55 | kreenaa | is it possible to integrate crystalvoice voi with asterisk |
| 11:37.16 | drray | what transport does crystalvoice use? |
| 11:37.35 | *** join/#asterisk backblue (n=moo@87-196-46-149.net.novis.pt) |
| 11:37.41 | kreenaa | http://www.crystalvoice.com/ |
| 11:37.43 | kreenaa | not surew |
| 11:37.45 | kreenaa | not sure |
| 11:38.01 | kreenaa | i just got the news abt this technology |
| 11:38.12 | kreenaa | anyone else heard abt it |
| 11:40.34 | kreenaa | ?? |
| 11:40.53 | samourai1 | plz can u help me i have a sip provider registration connected to asterisk when i tape sip show registry the line is registered ,i can make outbound calls but when i call to my number it doesn't work seems not to be registred |
| 11:41.28 | samourai1 | plz i have a sip provder registration connected to asterisk |
| 11:41.49 | samourai1 | xhen i tape sip show registered |
| 11:42.00 | samourai1 | the line is registered |
| 11:42.34 | samourai1 | when i make calls on outbound it work but when i call the number of the line it doesn't work |
| 11:43.19 | *** join/#asterisk effective (n=nick@mail.mercyministries.co.uk) |
| 11:43.24 | effective | hello |
| 11:43.57 | effective | any uk isdn junkies out there? |
| 11:45.31 | RoyK | effective: what exactly is an ISDN junkie? |
| 11:45.32 | RoyK | :) |
| 11:45.47 | X-Rob | poor bastards that can't get ADSL, I guess. |
| 11:45.50 | X-Rob | (Heya RoyK) |
| 11:46.37 | effective | heh |
| 11:46.50 | effective | this is for asterisk |
| 11:47.03 | RoyK | bingo |
| 11:47.19 | effective | (hence being in the asterisk chan ;)) |
| 11:47.35 | RoyK | another bingo |
| 11:47.38 | effective | heh |
| 11:47.40 | effective | anyway... |
| 11:47.52 | kreenaa | anyone know abt crystalvoice? |
| 11:48.09 | effective | I've not (ever) touched isdn but am needing to do an asterisk install on a bt 2e. Anyone have any experience with that? |
| 11:48.22 | samourai1 | no one can help me with sip registration |
| 11:48.30 | *** join/#asterisk saftsack (n=saftsack@p54A7FED0.dip.t-dialin.net) |
| 11:48.48 | *** join/#asterisk \lart (i=nunya@neo.jasons.org) |
| 11:49.29 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 11:50.14 | effective | just don't really know what card i'll need to get etc... |
| 11:50.41 | effective | isdn seems to be a world of acronyms |
| 11:51.19 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:52.44 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
| 11:52.52 | effective | ah well - guess noone knows |
| 11:54.20 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 11:54.46 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 11:55.58 | effective | anyone know any regulars who might - so i can watch out for em |
| 11:55.58 | \lart | effective, it's a world of TLAs (three letter acronyms). what are you looking to know about isdn cards? |
| 11:56.16 | \lart | while not a super-genius on the topic, i know a thing or 2. |
| 11:57.15 | effective | basically. I'm just starting the install and am wanting to know the cheapest compatable card for the system. I've got a diva card (only now to find out there are server and client cards :/) |
| 11:57.24 | \lart | bri or pri? |
| 11:57.34 | effective | Installed is an isdn2e from bt with 2 channels ptp |
| 11:57.42 | \lart | ok, that's bri |
| 11:57.48 | effective | cool |
| 11:57.56 | \lart | which card, specifically? |
| 11:58.19 | effective | just the eicon diva client pci isdn card |
| 11:58.29 | \lart | it has no model #? |
| 11:58.45 | effective | i'd heard that diva cards worked before i realised that they come in 2 flavours |
| 11:59.12 | effective | diva 2.01 pci |
| 11:59.24 | effective | 800-362 |
| 11:59.53 | effective | ah there it is... DIVA ISDN 2.01 PCI S/T UK |
| 12:00.29 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:00.40 | *** join/#asterisk senaque (n=roger@202.1.119.20) |
| 12:01.39 | \lart | from what i'm reading, you might just be buggered with that particular card. everything i see is for the diva server cards |
| 12:01.50 | effective | yeah that's what i thought |
| 12:02.18 | effective | do you know the cheapest card which is compatable with that setup? |
| 12:02.35 | \lart | looks like from those guys it's diva server v-bri and diva server bri for the 1 line cards |
| 12:02.36 | *** join/#asterisk beyond (n=lulongo@201-42-23-77.dsl.telesp.net.br) |
| 12:02.49 | \lart | have a look at the wiki - http://www.voip-info.org/wiki/view/Asterisk+hardware |
| 12:02.52 | \lart | it's got bri cards |
| 12:03.43 | \lart | although, now i see that you might be able to use the client card if it's got hardware echo cancellation, at least according to that page |
| 12:04.44 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 12:04.51 | Ciber311 | there any reason to use g729 other than to save BW? |
| 12:04.55 | \lart | this other page may also prove helpful - http://www.voip-info.org/wiki/view/zaptelBRI |
| 12:05.19 | luke-jr_ | Ciber311: bandwidth is the only reason to use compression... |
| 12:05.30 | \lart | Ciber311, I was pondering the same thing... I've got 2 stations, one on ethernet, the other 802.11g. I suspect no. |
| 12:05.40 | Ciber311 | well just checking |
| 12:05.58 | Ciber311 | cause i keep hearing like a crackling pop type noise in all my sip calls |
| 12:06.11 | Ciber311 | and it's driving me nuts |
| 12:06.15 | luke-jr_ | I generally try to stick to ulaw and gsm where I need it |
| 12:06.25 | effective | hmm i'll see if i can work out if it has |
| 12:06.26 | Ciber311 | using alaw/ulaw in my configs |
| 12:06.49 | Ciber311 | luke-jr_: any ideas? 3 providers... same noise through all |
| 12:07.17 | luke-jr_ | Ciber311: bad phones on your end? |
| 12:07.37 | SheriF_WorK | <PROTECTED> |
| 12:07.40 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-136-117.phlapa.east.verizon.net) |
| 12:07.46 | Ciber311 | polycom 501's |
| 12:07.50 | kreenaa | any way to distribute bandwidth among several providers using asterisk?? |
| 12:08.01 | Ciber311 | get the sound through both speakerphone and handset |
| 12:08.17 | Ciber311 | the noise is not particularly loud |
| 12:08.27 | Ciber311 | but i can hear it in the background if i raise the volume |
| 12:08.39 | McLazarus | Ciber311: what is the problem you are having with poly 501's? |
| 12:08.54 | kreenaa | <PROTECTED> |
| 12:08.57 | Ciber311 | i don't think it's the phone |
| 12:08.59 | McLazarus | I just joined, but I have about 200 poly 501's deployed with asterisk. |
| 12:09.04 | luke-jr_ | kreenaa: meaning? |
| 12:09.05 | Ciber311 | just some sort of background noise |
| 12:09.13 | Ciber311 | through sip calls |
| 12:09.20 | luke-jr_ | kreenaa: if you're going to spam a question that often, you'd better rephrase it |
| 12:09.31 | McLazarus | Ciber311: like a hum? |
| 12:09.48 | McLazarus | does it override audio, or is it only in the silences? |
| 12:10.08 | Ciber311 | in the silence pretty much |
| 12:10.14 | Ciber311 | like crackling popping |
| 12:10.27 | McLazarus | is it SIP to SIP calls? or SIP to PSTN? |
| 12:10.37 | Ciber311 | sip to sip |
| 12:10.41 | Ciber311 | well |
| 12:10.43 | kreenaa | is it possible to use 2 providers to make more stimultaneous calls using asterisk>>> |
| 12:10.46 | Ciber311 | my providers are sip |
| 12:10.51 | Ciber311 | people i call are on PSTN |
| 12:11.01 | Ciber311 | both incoming and outgoing calls |
| 12:11.04 | luke-jr_ | kreenaa: sure, but most providers don't limit them |
| 12:11.06 | Ciber311 | through different providers |
| 12:11.24 | McLazarus | so you have had the problem with different providers? |
| 12:11.34 | Ciber311 | oh and it's not there in the echo test |
| 12:11.47 | kreenaa | is it possible with asterisk...lets say |
| 12:11.49 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 12:11.55 | Ciber311 | yeah i get it through incoming calls from voxbone |
| 12:12.06 | McLazarus | my first thought is that it would be a problem on their connection to the PSTN, maybe their PRI card (or whatever) is broken or timing is wrong. |
| 12:12.06 | Ciber311 | and outgoing and incoming through axvoice |
| 12:12.26 | McLazarus | are the using compression do you know? |
| 12:12.32 | McLazarus | well more than ulaw |
| 12:12.33 | Ciber311 | no idea |
| 12:12.39 | kreenaa | at some time the bandwith is not sufficient ....then the next calls will be routed through another providr |
| 12:12.41 | McLazarus | that could effect sound quality. |
| 12:12.43 | Ciber311 | well they both support g729 |
| 12:12.57 | Ciber311 | well when the person talls |
| 12:12.58 | Ciber311 | talks |
| 12:13.02 | Ciber311 | it sounds VERY clear |
| 12:13.31 | McLazarus | what version of software is on the 501s? |
| 12:13.39 | Ciber311 | 1.6.5 |
| 12:14.00 | McLazarus | hmm |
| 12:14.11 | McLazarus | that is the newest, or at least it is the one I am running on mine. |
| 12:14.21 | Ciber311 | they are up to 1.6.6 now |
| 12:14.30 | Ciber311 | but the sobs don't release it to us peons :P |
| 12:14.33 | McLazarus | I'll have to see what was "fixed" |
| 12:14.47 | McLazarus | yeah, my reseller gets it. |
| 12:15.03 | Ciber311 | get it for us little people ;) |
| 12:15.39 | McLazarus | hah, well I don't want to annoy polycom in my official capacity |
| 12:16.07 | Ciber311 | hehe |
| 12:16.27 | McLazarus | Not sure what that could be. Strange that you had the problem through different providers. |
| 12:16.44 | McLazarus | my poly's are clear even in silences. |
| 12:17.05 | Ciber311 | i'm way too lazy to test with the other one |
| 12:17.17 | Ciber311 | these phones are way too annoying to deal with |
| 12:17.28 | Ciber311 | 10 minute reboots for everything ;) |
| 12:17.51 | Ciber311 | OMG I BREATHED ON IT NOW I GOTTA WAIT 5 MINS |
| 12:18.19 | luke-jr_ | kreenaa: whos bandwidth? |
| 12:18.32 | luke-jr_ | kreenaa: if your bandwidth is not sufficient, using another provider won't help |
| 12:20.28 | kreenaa | hi |
| 12:20.38 | kreenaa | why |
| 12:20.58 | kreenaa | i mean get another provider so there will be more bandwith |
| 12:20.59 | kreenaa | rite |
| 12:21.00 | kreenaa | ? |
| 12:21.08 | Ciber311 | ... |
| 12:21.19 | kreenaa | pls explain |
| 12:21.22 | kreenaa | i am new to this |
| 12:21.26 | Ciber311 | luke-jr_: i think he/she means get a new ISP |
| 12:21.31 | kreenaa | yes |
| 12:21.33 | *** join/#asterisk marvy (n=marvy@c220-237-79-137.kelvn1.qld.optusnet.com.au) |
| 12:21.35 | kreenaa | new ISP |
| 12:21.38 | kreenaa | 2 ISP |
| 12:21.44 | kreenaa | will asterisk support 2 ISP |
| 12:21.57 | kreenaa | if 1 ISP not sufficient |
| 12:22.12 | kreenaa | then following calls will be routed to 2nd ISP |
| 12:22.28 | mutilator | yerp |
| 12:22.59 | kreenaa | ok..so its possible...will it be difficult to implement |
| 12:24.00 | RoyK | kreenaa: that's not an asterisk issue, but a routing issue |
| 12:24.04 | mutilator | no |
| 12:24.22 | RoyK | no? |
| 12:24.25 | RoyK | http://lartc.org/ |
| 12:25.07 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-3-12.w81-248.abo.wanadoo.fr) |
| 12:25.15 | kreenaa | oh |
| 12:25.29 | kreenaa | its diffrent from asterisk... |
| 12:25.36 | kreenaa | it has nothing to do with asterisk |
| 12:25.50 | kreenaa | do u know abt crystalvoice |
| 12:25.54 | *** part/#asterisk Yalla-One (n=yallaone@unaffiliated/yalla-one) |
| 12:26.04 | mutilator | RoyK: i think he's talking a sip or iax provider |
| 12:26.11 | mutilator | routing calls thru 2 diff people |
| 12:27.06 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:27.17 | RoyK | erm. ok |
| 12:27.20 | kreenaa | crystalvoice enable voip to be used even with dial up |
| 12:27.22 | kreenaa | connetion |
| 12:27.25 | RoyK | shouldn't be too hard |
| 12:30.20 | [TK]D-Fender | Ciber311 : Shouldn't complain about the waits on the web interface... you should wake up and learn how to provision them :) |
| 12:31.17 | Ciber311 | hehe i know how to use the configs |
| 12:31.31 | Ciber311 | but the reboots are still anything over the smallest change |
| 12:31.38 | Ciber311 | err annoying |
| 12:33.45 | SheriF_WorK | [TK]D-Fender: my saviuor what do u think about that ? -- Got SIP response 302 "Redirecting..." back from 212.103.170.XXX -- Now forwarding SIP/108-13d4 to 'Local/20026@67888' (thanks to SIP/67888-23c3) <--- this redrecting should be me to Local .. why it add local ?? " where 67888 is the context for some kind of MVP Sip server . asterisk is registering as a client. but i don't know why asterisk trying to do Local/ channel when it get redirecti |
| 12:33.46 | SheriF_WorK | ng ? |
| 12:34.03 | SheriF_WorK | saveior * as i think my english is bad anyway :D |
| 12:34.27 | SheriF_WorK | <PROTECTED> |
| 12:34.27 | SheriF_WorK | Jun 15 15:41:05 NOTICE[32239]: app_dial.c:232 wait_for_answer: Unable to create local channel for call forward to 'Local/20026@67888' |
| 12:34.46 | SheriF_WorK | it's not local it's a phoonbook on the SIp server. |
| 12:37.43 | nortex | OT: Has Anyone been able to get Polycom phones to send status to a Micro$oft Live Comm server? I heard this feture was coming out, but I have not heard if it is out and working. |
| 12:39.12 | [TK]D-Fender | SheriF_WorK : No idea... |
| 12:39.42 | [TK]D-Fender | nortex : Its in SIP 2.0 coming out momentarily |
| 12:40.47 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 12:40.58 | SheriF_WorK | :-S |
| 12:41.03 | [TK]D-Fender | Ciber311 : I've learned to live with it since I have a habit of doing things right the first time :) |
| 12:41.06 | nortex | [TK]D-Fender, Thanks |
| 12:42.00 | [TK]D-Fender | nortex : M$ LCS... *shudder*..... Blue Sine-Wave of Death? :D |
| 12:42.19 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 12:42.51 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 12:43.41 | Ansonmus | Hello, we encounters some strange problems |
| 12:44.11 | mutilator | wow V for Vendetta got a 8.2/10 rating on imdb, was it that good? |
| 12:44.24 | Ciber311 | i liked it |
| 12:44.48 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-85-84.buckeyecom.net) |
| 12:45.02 | Ansonmus | when connecting to our asterisk using putty sometimes it takes much time. Also there are connection problems with our Grandstream phones to asterisk. Losing registrations and so on. Anyone an idea or an idea to begin debug this? |
| 12:45.07 | coppice | people are just worried about a vendetta if they say something bad :-) |
| 12:45.39 | nortex | [TK]D-Fender, The 2003 version has run really well, but I'm also looking at Wildfire with spark clients. I just don't think Polycom will send status updates to it, like out to lunch or in meeting. |
| 12:45.57 | Corydon76-home | Ansonmus: sounds like an overloaded network. Any Windows machines with infections? |
| 12:46.02 | mutilator | .. |
| 12:46.23 | Ansonmus | Corydon-w: whe have tested it with only a asterisk + switch + hardphones |
| 12:46.45 | Corydon76-home | switch or hub? |
| 12:46.59 | Ciber311 | [TK]D-Fender: how soon is momentarily? |
| 12:47.03 | Corydon76-home | I'd still go with network problems above and beyond anything else |
| 12:47.36 | Ansonmus | We run asterisk at home on a celeron 2.8 Ghz |
| 12:47.51 | Corydon76-home | ~aah |
| 12:48.11 | jbot | aah is, like, AMP and amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp) |
| 12:48.11 | Corydon76-home | ~freepbx |
| 12:48.14 | jbot | rumour has it, freepbx is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 12:48.42 | Ciber311 | isn't it trixbox now? lol |
| 12:48.44 | Ansonmus | Corydon-w: tested with a hub and a switch |
| 12:48.53 | Ciber311 | whoever came up with that name needs to be shot |
| 12:49.03 | Corydon76-home | Ansonmus: see the previous messages from jbot |
| 12:49.07 | *** join/#asterisk Lino` (n=Lino@i577BFC4A.versanet.de) |
| 12:49.33 | Ansonmus | Corydon-w: why should I not ask this question here? Do you think this is a freepbx related question? |
| 12:50.01 | Corydon76-home | I think this is related to you running AAH and not Asterisk |
| 12:50.30 | [TK]D-Fender | nortex : SIP 2.0 has HUGE LCS compliance add-ons. |
| 12:50.31 | Ciber311 | Ansonmus: it's a penis issue :P |
| 12:50.36 | [TK]D-Fender | Ciber311 : Within a month. |
| 12:51.02 | Ciber311 | so we'll finally get shared line appearances and all that jazz? :P |
| 12:51.23 | Ansonmus | can you tell why you think it is AAH related? On an other machine AAH works fine |
| 12:51.45 | mitcheloc | [TK]D-Fender: are there any documents/articles you can point me to about SIP 2.0 & LCS? |
| 12:51.56 | *** join/#asterisk oej (n=oej@213.115.215.54) |
| 12:52.00 | [TK]D-Fender | Ansonmus : sounds like its bandwidth is being choked out globally... |
| 12:52.10 | Corydon76-home | Ansonmus: this channel is for asterisk questions. You're running AAH, not Asterisk. |
| 12:52.13 | [TK]D-Fender | Ansonmus : Check all of your routing and do a network scan. |
| 12:52.37 | [TK]D-Fender | mitcheloc : its only in the SIP 2.0 Beta release notes. |
| 12:53.00 | mitcheloc | beta release notes bundled with LCS? |
| 12:53.42 | *** part/#asterisk noky (n=noky@200.69.211.18) |
| 12:54.34 | Ansonmus | aah does not change the asterisk build so I think it is not forbidden to ask questions about asterisk here |
| 12:55.11 | [TK]D-Fender | mitcheloc : no, the Polycom SIP 2.0 Beta release notes like I just said... |
| 12:55.14 | Ciber311 | Ansonmus: you're wasting your time... they don't care :P |
| 12:55.35 | Ansonmus | Ok, must i say: I don't have AAH ? :p |
| 12:56.00 | Ciber311 | too late |
| 12:56.03 | mitcheloc | Ansonmus: if you read the topic, you will see that questions like those should be directed elsewhere, it's like saying an volkswagon and a honda use the same engine, let's bug the engine manufacter and not the car manufacturer.... |
| 12:56.06 | [TK]D-Fender | Ansonmus : You have not offered ANYTHING of use in defining your problem let alone showing that its * related at all.. SSH being slow and getting cut has nothing to do with *. |
| 12:56.08 | Ciber311 | go change your name and come back |
| 12:56.09 | Ciber311 | :P |
| 12:57.02 | Ansonmus | are you all in real life the same as virtual? |
| 12:57.08 | [TK]D-Fender | Ansonmus : Please do tell what SSH being slow to connect has to do with that? Have you done bandwidth monitoring? Did you even tell us where the PC you were connecting to it from was relative to *? ANYTHING?! |
| 12:58.18 | Ciber311 | [TK]D-Fender: or even what's the load on the server |
| 12:58.20 | [TK]D-Fender | Ansonmus : How about you start over, be thorough in your description and point us to what you think is failing in your scenario exactly... |
| 12:58.28 | [TK]D-Fender | Ciber311 : Don't get fancy now! |
| 12:58.33 | Ciber311 | :P |
| 13:00.24 | Ansonmus | Hmm my english is not so good that doesn't help me. But I'm trying to help my collegua out of the shit. 1 hardphone direct connected to the asterisk server will sometimes not register. All incoming ISDN calls from outside comes in on the asterisk machine. The machine is a intel celeron 2.8 Ghz. I hope you will not fire me but help me at my nivo |
| 13:00.38 | *** join/#asterisk saftsack (n=saftsack@p54A7FED0.dip.t-dialin.net) |
| 13:01.15 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:01.33 | [TK]D-Fender | Ansonmus : You meantioned "Grandstream" for a phone , so by "direct connected" I presume you mean on a local LAN right next to *? |
| 13:02.04 | Ansonmus | I mean a crossover cable between phone and networkport of * server |
| 13:02.58 | mutilator | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=8945503275&ru=http://search.ebay.com:80/8945503275_W0QQfromZR40QQfviZ1 |
| 13:03.20 | [TK]D-Fender | Ansonmus : And you're running DHCPD on your server? |
| 13:04.06 | [TK]D-Fender | mutilator : Old joke, sold "as new" :) |
| 13:05.50 | mutilator | anyone know if there were any solar flares hittin earth this morning? our 5.8ghz radios have been going nuts |
| 13:06.02 | mutilator | signal is slowly getting better now |
| 13:07.07 | Ansonmus | [TK]D-Fender: normally we run dhcp on a adsl router. But for the test phone direct to * was the phone configured on a fixed ip. I don't think dhcpd is running. I don't know how to check it |
| 13:07.20 | [TK]D-Fender | mutilator : Wow... one bad day and you think the whole UNIVERSE is out to get you... sheesh |
| 13:07.40 | Ciber311 | mutilator: yeah one nearly blew the building next to me in half! |
| 13:07.52 | [TK]D-Fender | Ansonmus : Ok, well if its jsut SSH being slow not sure what to tell you.... Where are you connecting from? |
| 13:07.53 | mutilator | well |
| 13:07.55 | *** join/#asterisk lunk (n=lunk@negative-influence.com) |
| 13:08.02 | mutilator | these radios cover the northern half of michigan |
| 13:08.05 | mutilator | from saginaw up |
| 13:08.23 | mutilator | and all the ones in the northern part of that are acting weird |
| 13:08.27 | Ciber311 | well google it :P |
| 13:08.28 | CoaxD | hate. oracle. |
| 13:08.32 | Ciber311 | should not be hard to find out |
| 13:08.47 | Ciber311 | it could just be the black helicopters blocking your signal |
| 13:08.50 | CoaxD | Everybody commercially says "Postgres sucks. MySQL sucks. Go oracle!". Oracle sucks just as frickin bad. |
| 13:09.00 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 13:09.14 | Ciber311 | CoaxD: EVERYONE SUCKS!!! YAY!!! |
| 13:09.36 | CoaxD | yes. now we can all be happy |
| 13:09.39 | Ansonmus | [TK]D-Fender: My questions and answers are not clear I know. Problem 1: Losing connections or registrations between phone and *. Problem 2: (but we think related) is that connecting to SSH using putty takes sometimes relative much time and sometimes is fast |
| 13:09.46 | mutilator | .. |
| 13:10.02 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 13:10.02 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:10.14 | [TK]D-Fender | CoaxD : Typically few people trash PostgreSQL users... its a very respectable product that didn't cut a lot of corners... True Oracle is huge and for those needing it, hey... but MySQL's infractions are well known, and their sacrifices shunned by many. |
| 13:10.33 | [TK]D-Fender | Ansonmus : Where are you connecting FROM? |
| 13:11.09 | [TK]D-Fender | Ansonmus : and the quick analysis says something globally affecting networking is at fault there.... not *. |
| 13:11.25 | drray | plug a sip phone into the local subnet |
| 13:11.45 | mutilator | Ciber311: i don't think they post that kinda stuff that fast |
| 13:11.50 | mutilator | less you know of something i don't? |
| 13:12.05 | Ciber311 | i remember seeing a site that posted those |
| 13:12.16 | Ciber311 | had like sattelite graphs or something |
| 13:12.21 | Ciber311 | but they were constantly updated |
| 13:13.06 | Ciber311 | http://www.sec.noaa.gov/rt_plots/xray_5m.html |
| 13:13.47 | Ciber311 | got that from http://www.spaceweather.com/ |
| 13:14.33 | Ansonmus | [TK]D-Fender: I've not enough information to tell about the SSH problem. I'm now not on the location where the * server is. Now connection from my computer (windows xp + putty + adsl) to that server (adsl from the same provider). But that is not a big problem. But the first, losing connection between phone and server, also when phone is direct plugged in into * server is our big problem |
| 13:14.49 | mutilator | so there was more activity today according to that.. |
| 13:15.05 | mutilator | no idea what that graph is of but.. |
| 13:15.14 | Ciber311 | mutilator: http://www.n3kl.org/sun/noaa.html |
| 13:16.28 | Ciber311 | [TK]D-Fender: are SIP-B and SIP 2.0 the same thing? |
| 13:16.38 | mutilator | well the graph trends match the radio outtages |
| 13:16.43 | mutilator | so thats probably what it was then |
| 13:16.48 | Strom_C | good morning |
| 13:16.53 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 13:17.23 | boch | how can i read the return code of a system() cmd ? |
| 13:17.46 | Ciber311 | why hello to you too! |
| 13:17.57 | mutilator | and you guys thought i was crazy |
| 13:18.01 | mutilator | pffft |
| 13:18.21 | Ciber311 | lol |
| 13:18.29 | Strom_C | mutilator: you're in #asterisk. Of course you're crazy. |
| 13:18.31 | Strom_C | :) |
| 13:19.05 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 13:21.22 | [TK]D-Fender | Ansonmus : So you're saying that if you use that phone to directly call an * application like VoiceMail that it'll drop all by itself while you're playing around? Not using ANY other tech like a line or VOIP connection? |
| 13:21.28 | [TK]D-Fender | Ciber311 : NO. |
| 13:21.29 | mutilator | o_O |
| 13:21.48 | boch | how can i read the return code of a System() cmd in extensions.conf ? |
| 13:22.00 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 13:22.09 | Ciber311 | [TK]D-Fender: what's the difference? there's like 0 results for sip 2.0 on google |
| 13:22.10 | [TK]D-Fender | Ciber311 : SIP 2.0 is referring to the Polycom SIP applicatio version number much like Cisco is at version 8.x (Maybe they'll get it right somewhere in the double digits...) |
| 13:22.25 | Ciber311 | ah ok |
| 13:22.38 | [TK]D-Fender | Ciber311 : Its a closed Beta right now... you SHOULDN'T find anything on it... |
| 13:23.39 | Ciber311 | so are we ever gonna get shared line appearances etc in asterisk? |
| 13:24.07 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:24.32 | LH-euhost | hello all, i have a problem with callback on *. When i get called back, i am able to enter a number like "01733516818". But often the * recognices a wrong number like "017335168118" (double 1, although i never entered two 1 digits). any hints? |
| 13:24.51 | [TK]D-Fender | Ciber311 : Planned with * 1.4 |
| 13:25.12 | Strom_C | LH-euhost: what kind of channel is the call going over? |
| 13:25.19 | LH-euhost | sip |
| 13:25.32 | [TK]D-Fender | Ciber311 : Polycom-Asterisk integration is going to get so much better shortly including a hope merge for bwesche's ACD patch. |
| 13:25.37 | Strom_C | what kind of equipment is at the terminating end of the call |
| 13:25.38 | boch | [TK]D-Fender: do you know if is there a var with the last return code ? |
| 13:25.42 | Ciber311 | and we're at 1.2.9.... *dies* |
| 13:25.57 | [TK]D-Fender | boch : Not a clue.... last I recall System always returns -1 |
| 13:25.58 | Strom_C | Ciber311: 1.4 is scheduled for release soon |
| 13:26.01 | LH-euhost | a cellphone |
| 13:26.18 | Ciber311 | they're skipping 1.3.x? :P |
| 13:26.25 | [TK]D-Fender | Ciber311 : 1.4 is currently expected this summer. |
| 13:26.34 | Strom_C | Ciber311: 1.3 is the development branch which will become 1.4 release |
| 13:26.35 | Ansonmus | [TK]D-Fender: I will test things again. For now I will thank you for your patience and time. But the problem is not easy. 70% of time all things are ok so we spend much time to test and so. I hoped someone says: yeah that is a know problem or something like that. |
| 13:26.37 | [TK]D-Fender | Ciber311 : Do YOU remember * 1.1.x? |
| 13:26.50 | Ciber311 | sort of :P |
| 13:26.50 | [TK]D-Fender | Ciber311 : Get with the program! |
| 13:27.00 | Ciber311 | haha |
| 13:27.32 | Ciber311 | [TK]D-Fender: that's good about the polycom integration |
| 13:27.39 | Ciber311 | you don't work for polycom right? :P |
| 13:28.43 | [TK]D-Fender | Ciber311 : You'd think so :) But no, I just promote the WORTHY. If they turn on us, I turn on them. Simple as that. So far * is the PBX catching my eye... FreeSWITCH is becoming very interesting now, so in the future, who knows? |
| 13:29.00 | LH-euhost | Strom_C: and i have alaw, ulaw, g729, gsm and sliniear allowed for codecs in sip.conf |
| 13:29.07 | LH-euhost | maybe this is not good? |
| 13:29.49 | mutilator | we're not worthy, we're not worthy! |
| 13:30.06 | Strom_C | LH-euhost: which codec is the call actually using? |
| 13:30.10 | [TK]D-Fender | Ciber311 : To me the most valuable aspect of *'s position in the PBX world is its REPLACABILITY. If I buy a standard T1 card, standard SIP phones following the greater volume of RFC's and well built, then no piece is dependant on the other and NOBODY owns me. |
| 13:30.29 | [TK]D-Fender | Ciber311 : Thats what prevails with me. |
| 13:30.43 | [TK]D-Fender | mutilator : Not now... maybe later ;) |
| 13:31.56 | boch | cant belive all commands returns an exit code, and there is not a way to read it |
| 13:32.11 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:32.12 | tzanger | boch: yep, it's totally gay |
| 13:32.14 | Ciber311 | [TK]D-Fender: just looked at FreeSWITCH site... lol |
| 13:33.49 | ids2500 | LH-euhost: inband or rfc2833 dtmf ? |
| 13:34.19 | LH-euhost | Strom_C, i am not sure which codes is used. can you tell me how to see which codes is used? is this output in debug mode? |
| 13:34.30 | ids2500 | it doesn't matter what codec you're using euhost |
| 13:34.35 | ids2500 | it matters what dtmfmode you're usig |
| 13:35.01 | LH-euhost | then other question, where can i check which dtmfmode is being used? |
| 13:35.03 | ids2500 | if you're using rfc2833, then asterisk's rfc2833 implementation is known to be broken |
| 13:35.05 | ids2500 | euhost: sip.conf |
| 13:35.08 | ids2500 | dtmfmode = |
| 13:35.21 | LH-euhost | i didnt set anything there.. is there a default value for it? |
| 13:35.27 | ids2500 | no idea |
| 13:35.40 | LH-euhost | oh, i am wrong. its set to: dtmfmode=inband |
| 13:35.45 | ids2500 | okay |
| 13:35.56 | ids2500 | if your endpoints can use rfc2833 |
| 13:35.59 | ids2500 | try that instead |
| 13:36.01 | ids2500 | see if it works better |
| 13:36.05 | LH-euhost | ok, will try |
| 13:36.40 | LH-euhost | thx! |
| 13:38.11 | boch | tzanger: and whats the return code for? do you know |
| 13:38.35 | [TK]D-Fender | Ciber311 : Whats so funny? |
| 13:40.12 | Katty | morning |
| 13:40.14 | Ciber311 | the wheels thing |
| 13:40.50 | *** join/#asterisk nazgool (n=oli@dip-109-202.bras.dsl.breisnet.com) |
| 13:40.50 | nazgool | hi all |
| 13:41.07 | LH-euhost | ids2500: i tried rfc2833 and we have still the same problem. any further hints? |
| 13:41.10 | [TK]D-Fender | Ciber311 : It is a better wheel, no? :) |
| 13:41.21 | Strom_C | LH-euhost: maybe the cellphone sucks |
| 13:41.36 | Strom_C | LH-euhost: do you get the same problem with other phones? |
| 13:42.49 | Ciber311 | [TK]D-Fender: lot of places are saying 1.4 in july? |
| 13:42.59 | Strom_C | Ciber311: that's the goal, yes |
| 13:44.43 | tzanger | boch: it's not used *at all* and htere is no way to get at it |
| 13:45.06 | Ciber311 | guessing it would be pointless for me to test with my 501's using 1.6.5? |
| 13:45.17 | [TK]D-Fender | Ciber311 : enough |
| 13:45.28 | [TK]D-Fender | Ciber311 : huh? |
| 13:45.58 | [TK]D-Fender | Ciber311 : What are you running on the IP 501 now? |
| 13:46.05 | Ciber311 | 1.6.5 |
| 13:46.33 | [TK]D-Fender | Ciber311 : I was distributed with SIP 1.6.2 from factory, so no reason not to bump to 1.6.6 as it contains a LOT of little fixes from 1.6.5 |
| 13:46.51 | [TK]D-Fender | Ciber311 : but that has nothing to do with * 1.4 .... |
| 13:47.17 | Ciber311 | i'd love to get 1.6.6, but polycom won't release it to us peons :P |
| 13:47.46 | [TK]D-Fender | Ciber311 : Don't have a reseller? Just ask VoipSupply or Atacomm, and they'll typically refer you to their FTP where you can just DL it. |
| 13:48.23 | Ciber311 | voipsupply doesn't have it yet |
| 13:48.33 | nazgool | in the example sip.conf there is a section sip_proxy that is supposed to work only for incoming calls and a section sip_proxy-out supposed to work only for outgoing calls. what if i'd like to do both with the same external sip account and same proxy? do i have to fill out both these sections? |
| 13:48.45 | Ciber311 | suppose i could check with atacomm |
| 13:51.28 | LH-euhost | Strom_C: callback on a fixed line phone doesnt have this issues.. only a callback on a cellphone (its always only a problem with one or two digits, that they are double).. hmm :( |
| 13:51.34 | Ciber311 | ah |
| 13:51.40 | Ciber311 | atacomm has it on their ftp |
| 13:52.14 | Strom_C | LH-euhost: figure out which dtmfmode you're using |
| 13:52.56 | LH-euhost | i tried it with inband and now rfc2833. is there any other i can try? or any other setting to tune? |
| 13:53.15 | Hmmhesays | so my abstinence from excessive drinking is going well |
| 13:53.32 | Katty | woah |
| 13:53.33 | Strom_C | LH-euhost: do you have the same problem with different cellphones? |
| 13:53.35 | Katty | what now, Hmmhesays? |
| 13:53.45 | LH-euhost | strom_c, yes. |
| 13:53.57 | Hmmhesays | I've had 1 beer this week |
| 13:54.24 | Katty | oh. |
| 13:54.29 | Katty | Hmmhesays: whyfor? |
| 13:54.38 | Hmmhesays | I dunno |
| 13:54.43 | Hmmhesays | just haven't felt like drinking |
| 13:54.56 | Katty | weird. |
| 13:57.03 | Hmmhesays | been playing a lot of guitar though |
| 13:57.05 | Hmmhesays | and video games |
| 13:57.07 | Hmmhesays | weird |
| 13:57.14 | Hmmhesays | its like i'm regressing |
| 13:59.28 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 13:59.30 | nazgool | i get a strange warning: |
| 14:00.16 | nazgool | chan_sip.c:12708 reload_config: Empty context specified at line 54 for domain '10.0.0.200' |
| 14:00.58 | satlan32 | hi |
| 14:00.59 | nazgool | what did i do wrong? |
| 14:01.01 | satlan32 | need help please |
| 14:01.08 | satlan32 | in this line MYSQL(Fetch foundRow ${resultid} firstname) the var foundRow get the value 1 if there is a line found? |
| 14:01.13 | nazgool | i said domain=10.0.0.200 in my sip.conf |
| 14:01.40 | Ciber311 | stupid fedex drivers |
| 14:01.52 | FreezeS | nazgool, shouldn't you also specify context=... somewhere ? |
| 14:01.53 | Ciber311 | Jun 15, 2006 9:20 AM |
| 14:01.53 | Ciber311 | Delivery exception |
| 14:01.53 | Ciber311 | NEW YORK, NY |
| 14:01.55 | Ciber311 | Customer not available or business closed |
| 14:02.02 | Ciber311 | what the hell is up with that |
| 14:02.10 | Ciber311 | 40 mins ago i was sitting right here |
| 14:02.17 | Ciber311 | bastard never rang the door bell |
| 14:02.29 | *** part/#asterisk sevard (i=sev@merrill-49-169.resnet.ucsc.edu) |
| 14:02.34 | nazgool | in the [general] section. and there's a context=default as the very first line of that same section |
| 14:02.41 | FreezeS | Ciber311: can't you give them a call ? |
| 14:02.44 | Ciber311 | i'm getting the last laugh though, just called to have his ass re-attempt |
| 14:02.52 | Ciber311 | FreezeS: yeah |
| 14:03.04 | Ciber311 | paid for priority overnight, no way in hell i'm waiting another day |
| 14:03.36 | FreezeS | here usually the courier has my cellphone # and calls me when he is near |
| 14:04.39 | satlan32 | any1 can help?? |
| 14:04.40 | boch | tzanger: and have you used ${SYSTEMSTATUS} ? |
| 14:05.31 | Ciber311 | FreezeS: they're sending him back to re-attempt |
| 14:05.36 | Ciber311 | freaking loser |
| 14:05.46 | Ciber311 | bet it's a temp |
| 14:06.07 | Ciber311 | never have a problem with the regulars |
| 14:06.21 | FreezeS | is your door clearly labeled ? |
| 14:06.34 | Ciber311 | yes |
| 14:06.44 | Ciber311 | it's a dumb temp, i'm sure of it |
| 14:06.51 | Ciber311 | or a new guy |
| 14:06.56 | oej | nazgool, that is a bug that I've fixed recently |
| 14:07.03 | Dandan | anyone knows how to change bearer capabilities on PRI line? |
| 14:07.05 | oej | add domain=<domain>,<context> |
| 14:07.07 | Ciber311 | i know the regular fedex/ups guys by name :p |
| 14:07.19 | FreezeS | :)) |
| 14:07.44 | Ciber311 | afk |
| 14:08.07 | *** join/#asterisk lorinc (n=ang@caracas-2155.adsl.interware.hu) |
| 14:09.38 | nazgool | oej: recently as in "more recently than 1.2.9.1" ? |
| 14:09.51 | oej | Yes, after that |
| 14:10.00 | nazgool | ok thx |
| 14:11.16 | nazgool | so i'll have to build an svn version |
| 14:12.09 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 14:12.11 | Hmmhesays | anyone familiar with using libtool? |
| 14:13.59 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 14:15.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:17.07 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 14:21.14 | tRSS | what is the purpose of context = default in queues.conf. the comments say that the agent can press a single digit extension to get out of the queue. what is a single digit extension/ |
| 14:23.01 | [TK]D-Fender | tRSS : its exactly what it says.. you define a context for your queue and if a caller in the queue presses a single digit that matches an exten in that context the queue will abort and the caller will then continue being processed on that exten. |
| 14:23.33 | [TK]D-Fender | tRSS : For options like "You may press 1 at any time to leave us a voicemail" |
| 14:24.21 | tRSS | [TK]D-Fender: does this mean the agent would be out of the queue but still be on the phone the ongoing call and may be able to transfer this call to some other extension |
| 14:24.51 | [TK]D-Fender | tRSS : This isn't an option for AGENTS, its for callers in the queue. |
| 14:25.09 | *** join/#asterisk fgravato (n=frank@office-nat.choopa.net) |
| 14:25.14 | tRSS | ooh ok.. let me re-read the comments again :) |
| 14:25.19 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 14:25.44 | fgravato | anyone in here flash 7970G from CallManger to SIP |
| 14:26.28 | tRSS | ooh ... now I get it. so if I am in the queue for the past 20 mins (as a caller, obviously), then I would have the option to press 1 and leave a voicemail for the agents to later retrieve and may be call me back.. or something along these lines!? |
| 14:26.50 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.147.Dial1.SanJose1.Level3.net) |
| 14:27.30 | [TK]D-Fender | tRSS : Yes the "I don't feel like waiting around, let me out" option... you can have it do whatever you want naturally as its just more dialplan to execute. It leaves an entry in the queuelog with "EXITWITHKEY" |
| 14:30.08 | tRSS | [TK]D-Fender: that clears a lot of the things. thanks |
| 14:31.09 | tRSS | oh one more thing: what is the timeout option in queues.conf file. I thought if we are using round robin, it should keep ringing all agents until any of them picks it up, all the while putting the caller on hold? |
| 14:33.12 | nazgool | there's one thing i don't understand in the example sip.conf: |
| 14:33.22 | nazgool | (from 1.2.9.1) |
| 14:33.36 | [TK]D-Fender | trrs : Timeout is how long before passing to the next agent |
| 14:33.40 | nazgool | <PROTECTED> |
| 14:33.40 | nazgool | <PROTECTED> |
| 14:33.56 | key2 | [TK]D-Fender: do you know for what reason I could have an echo of the sound |
| 14:33.59 | nazgool | (the number at the left is the line number) |
| 14:34.21 | nazgool | <PROTECTED> |
| 14:34.21 | key2 | [TK]D-Fender: I basically means that when A calls B, A says something to B and A hears half of a second later what he said |
| 14:34.28 | nazgool | <PROTECTED> |
| 14:34.43 | nazgool | and then |
| 14:34.44 | nazgool | <PROTECTED> |
| 14:34.44 | nazgool | <PROTECTED> |
| 14:34.55 | nazgool | so that seems like a contradiction to me |
| 14:34.58 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:35.08 | tRSS | nazgool: peer: some that places calls to us (us being the asterisk box) and user: someone that we place call to (i.e. from asterisk to the user) |
| 14:35.37 | [TK]D-Fender | key2 : Try actually telling me useful DEATILS about the situation.... A & B doesn't mean anything. |
| 14:35.56 | [TK]D-Fender | nazgool : Please don't spam the channel like that, use pastebin |
| 14:35.57 | [TK]D-Fender | ~pb |
| 14:35.58 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 14:36.03 | nazgool | then shouldn't the type on line 321 in [proxy-out] be user instead of peer? |
| 14:36.49 | nazgool | since it says " we only want to call out, not be called" |
| 14:36.55 | [TK]D-Fender | tRSS : You are backwards on that... |
| 14:37.04 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 14:37.17 | [TK]D-Fender | nazgool : You use a PEER to call OUT to the provider. You use a USER to RECEIVE calls from them. |
| 14:37.27 | tRSS | [TK]D-Fender: i just realized that. nazgool: i meant it shold be the other way |
| 14:37.52 | nazgool | ok then, then shouldn't line 316 be type=user? |
| 14:38.06 | nazgool | i just mean the example sip.conf uses peer in both cases |
| 14:39.19 | [TK]D-Fender | nazgool : pastebin the larger part of that sample |
| 14:39.23 | De_Mon | nazgool you going to side with the example or a real talking human? |
| 14:39.26 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 14:41.36 | nazgool | http://pastebin.ca/65854 |
| 14:42.14 | nazgool | the numbers added on the left are line numbers |
| 14:42.33 | nazgool | i don't understand why there's peer in both cases |
| 14:42.44 | De_Mon | typo |
| 14:43.27 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 14:43.34 | nazgool | ok |
| 14:43.45 | nazgool | so on line 316 it would be type=user ? |
| 14:44.13 | De_Mon | youre a smart person, I think you can figure this out without handholding |
| 14:47.27 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 14:48.32 | *** join/#asterisk aze (n=aze@ACayenne-101-1-3-12.w81-248.abo.wanadoo.fr) |
| 14:49.18 | smackus | if i use AgentMonitorOutgoing for all of the calls dialing outbound, for example: |
| 14:49.18 | smackus | exten => _1XXXNXXXXXX,2,AgentMonitorOutgoing(c) |
| 14:49.18 | smackus | exten => _1XXXNXXXXXX,3,Dial(Zap/g2/${EXTEN:1}) |
| 14:49.18 | smackus | How do I record the agent calls inbound? My goal is to have the agent information and such in the file name of the recording, as well as the other information added to the CDR. |
| 14:49.18 | *** join/#asterisk skrusty (i=muad@xdev.net) |
| 14:49.38 | smackus | works for outbound correctly. |
| 14:49.41 | skrusty | anyone know if 07096 is pn1 or pn2? |
| 14:51.09 | Hmmhesays | does anyone know of rhel caches the rpms it uses anywhere ? |
| 14:51.43 | gaupe | Hmmhesays: try /var/spool/cache |
| 14:51.46 | asterboy | One of my VOIP providers is real choppy. Can * compensate for the bad connection? You know...a miracle kinda thing. |
| 14:54.26 | smackus | i have read that agentmonitoroubound is for outbound recording only, is there an equivalent for inbound monitoring? |
| 14:54.38 | key2 | [TK]D-Fender: In what situation could asterisk send back the RTP to the one sending it? |
| 14:54.45 | [TK]D-Fender | smackus : "MixMonitor" |
| 14:55.04 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 14:57.08 | *** join/#asterisk _Adam^ (i=adgi@orion.black.pl) |
| 14:58.55 | *** join/#asterisk wunderkin (n=wunderki@69.26.192.234) |
| 14:59.34 | _Adam^ | does anybody help me ? |
| 15:00.21 | *** part/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 15:00.31 | hypnox | you have to ask a question first _Adam^ |
| 15:00.35 | Hmmhesays | i don't know do they? |
| 15:00.58 | [TK]D-Fender | _Adam^ : WWW.DRPHIL.COM |
| 15:01.32 | _Adam^ | hypnox: i installed asterisk on OpenSuse (Dell Power Edge 2850, P4 3G 1G RAM) |
| 15:01.52 | *** join/#asterisk umay (n=chris@71-208-188-148.hlrn.qwest.net) |
| 15:02.01 | _Adam^ | when i call to asterisk to post mail , voice is very slowly |
| 15:03.12 | _Adam^ | music on gold is too very slow |
| 15:03.19 | RoyK | don't use suse for servers |
| 15:03.25 | RoyK | don't bang head against walls |
| 15:03.30 | RoyK | don't piss in pants |
| 15:04.10 | _Adam^ | RoyK: which distro dou you prefer ? |
| 15:04.31 | Strom_C | music on golf |
| 15:04.32 | Strom_C | er |
| 15:04.33 | Strom_C | gold |
| 15:04.34 | Strom_C | hahah |
| 15:04.44 | RoyK | debian |
| 15:04.48 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:04.52 | _Adam^ | ups ;-) my mistake ;-) music on hold ;-) |
| 15:04.53 | [TK]D-Fender | _Adam^ : Take your pick of CentOS, RHEL, Debian, Slackware |
| 15:05.05 | Strom_C | woot, debian! /me high-fives RoyK |
| 15:06.12 | _Adam^ | why not Suse ? |
| 15:06.45 | *** join/#asterisk ghenry (n=ghenry@81-174-209-161.pth-as2.dial.plus.net) |
| 15:07.41 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:08.37 | *** part/#asterisk clive- (n=pirch@dsl-146-69-243.telkomadsl.co.za) |
| 15:09.04 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 15:11.47 | *** join/#asterisk lorinc (n=ang@caracas-1410.adsl.interware.hu) |
| 15:12.47 | _Adam^ | ok thx |
| 15:14.00 | *** join/#asterisk tlowe_ (n=tlowe@bgp.terrorist.net) |
| 15:16.53 | asterboy | Suse is ok, but any distro with a 'try to be everything for everybody' setup...well sucks. |
| 15:17.21 | asterboy | imnsho |
| 15:17.46 | stephane_ | re |
| 15:18.15 | [TK]D-Fender | asterboy : .... he left |
| 15:19.36 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 15:19.53 | znoG_ | hi all. I have a PAP2-NA ATA and I'm trying to figure out if it supports T.38 .. does anyone know? |
| 15:20.08 | znoG_ | and my second question is whether Asterisk 1.2.x can be patched for t.38 passthrough? |
| 15:21.26 | LH-euhost | Strom_C + ids2500, i fixed my problem with false recognition of digits on a callback (callback on a callphone): we now set only to use the alaw & ulaw codecs + dtmfmode=auto - not it works perfectly afaik |
| 15:21.58 | *** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 15:22.39 | Ateboy | Hi there, Little problem about voicemail. I didn't override the default e-mail format in voicemail.conf and I only get the callerID name, not the number, is this normal? |
| 15:22.58 | Ateboy | I'd like to have the number as well, and the default format seems to be supposed to include it... |
| 15:23.32 | CunningPike | Ateboy: Do you get name in the message body? |
| 15:24.14 | *** join/#asterisk thermf (i=fadaasfa@d14-69-149-97.try.wideopenwest.com) |
| 15:24.19 | Ateboy | CunningPike: yes, I do get the name in msg body |
| 15:24.55 | CunningPike | Ateboy: OK - then yes, we had the same issue, and I made a change to the subject line to get name to appear - give me a minute to look up what we did |
| 15:25.03 | Strom_C | LH-euhost: awesome |
| 15:25.54 | mutilator | anyone know anything about private stockholding laws? |
| 15:26.12 | Ateboy | CunningPike: I dont really want name in subject... I want number in body... |
| 15:26.32 | CunningPike | Ateboy: Oh - do you get number in the subject? |
| 15:27.22 | *** join/#asterisk fgravato (n=frank@office-nat.choopa.net) |
| 15:28.11 | fgravato | has anyone manage to convert 7970G callmanager to SIP using the latest firmware? |
| 15:28.39 | dpryo | Hm.. Are there sip-images for 7970? |
| 15:28.48 | Qwell | yes, but they suck |
| 15:29.46 | dpryo | Figures ;P |
| 15:29.56 | Ateboy | CunningPike: no, I get "[PBX]: New message 1 in mailbox 0" in the subject |
| 15:30.01 | dpryo | Thing is.. I have a dusty 7970 somewhere |
| 15:30.14 | Qwell | dpryo: It works fine with chan_skinny, in my branch :p |
| 15:30.25 | Qwell | or you could donate it to the cause |
| 15:30.40 | Qwell | "the cause" being "the maintainer of chan_skinny" |
| 15:30.52 | Ateboy | CunningPike: and in the body I get ...0:11 long message (number 1) in mailbox 0 from NAME LASTNAME, on Thursday,... |
| 15:31.11 | mutilator | or just donate to me |
| 15:31.13 | dpryo | Qwell: I rather use my cisco callmanager ;P |
| 15:31.14 | mutilator | :O |
| 15:31.22 | CunningPike | Ateboy: OK - we use emailsubject=Voicemail message from ${VM_CIDNAME} <${VM_CIDNUM}> in our voicemail.conf |
| 15:31.25 | Qwell | dpryo: get me an ethereal dump :P |
| 15:31.43 | Ateboy | CunningPike: and I'd like to have ...0:11 long message (number 1) in mailbox 0 from NAME LASTNAME, 555-555-5555, on Thursday,... |
| 15:31.43 | mutilator | how bout i just mail you the 50 cds? |
| 15:31.53 | dpryo | Qwell: Sure.. But I need to get my 7970 to boot first.. It's kind of broken. |
| 15:32.04 | Qwell | dpryo: yeah...I could fix it :p |
| 15:32.08 | Qwell | but, then I'd keep it :D |
| 15:32.09 | Ateboy | CunningPike: I know this could be a workaround, but is this a bug? |
| 15:32.12 | dpryo | Qwell: hehe |
| 15:32.50 | CunningPike | Ateboy: We use the default message body - we get: Just wanted to let you know you were just left a 0:32 long message (number 1) in mailbox xxxx from NAME, on Wednesday, June 14, 2006 at 11:13:27 AM so you might |
| 15:32.55 | CunningPike | want to check it when you get a chance. Thanks! |
| 15:33.48 | *** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br) |
| 15:34.16 | Ateboy | CunningPike: if you look http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf, in the emailbody parameter |
| 15:34.34 | CunningPike | Ateboy: You have set yours? |
| 15:34.38 | Ateboy | nope |
| 15:34.41 | CunningPike | Ateboy: If so, what to? |
| 15:35.31 | Ateboy | Cunning: no, I only set format, serveremail, emaildateformat and minmessage |
| 15:35.47 | Ateboy | Cunning: so it should use the default one right? |
| 15:36.05 | Ateboy | Cunning: (so... in mailbox ${VM_MAILBOX}\nfrom ${VM_CIDNAME} (${VM_CIDNUM}), on ${VM_DATE}... |
| 15:36.33 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 15:36.42 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:36.51 | Ateboy | Cunning: So it should look like " in mailbox xxxx from NAME 555-555-5555, on... no? |
| 15:36.52 | CunningPike | Ateboy: Yes, which, in the example I posted, doesn't include number. You need to set a custom message using emailmessage that includes VM_CIDNUM and VM_CIDNAME |
| 15:37.04 | CunningPike | Ateboy: That'll work |
| 15:37.16 | CunningPike | Ateboy: Or something like it.... |
| 15:37.48 | Ateboy | Cun ning: but, according to the link I posted, the default includes VM_CIDNUM |
| 15:39.47 | Ciber311 | hmm |
| 15:40.13 | Ciber311 | [TK]D-Fender: don't know if it's a coincidence, but that noise is now gone using 1.6.6 |
| 15:42.00 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:42.02 | CunningPike | Ateboy: Not everything in the wiki is accurate :) We use the default and it doesn't contain number - we know VM_CIDNUM works because we use it in our subject (VM_CIDNUM and VM_CIDNAME were broken in older versions of asterisk - pre 1.2, iirc) |
| 15:44.43 | znoG_ | anyone doing fax passthru with Asterisk? |
| 15:45.13 | Hmmhesays | building my toolchain building my toolchain |
| 15:45.53 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 15:46.23 | *** join/#asterisk alunt2003 (n=alunt200@host81-158-181-242.range81-158.btcentralplus.com) |
| 15:46.52 | Dandan | anyone knows how to change bearer capabilities on PRI line? |
| 15:47.22 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
| 15:51.17 | tRSS | quick question: when I am setting up my queues.conf file and when I mention members, e.g. members=>SIP/user0; members=>SIP/user1, then I define these members in sip.conf instead of agents.conf, correct? and when I do a members=>Agents/user0, then these users will be defined in agents.conf, right? |
| 15:51.51 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 15:51.56 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 15:52.20 | McLazarus | Ciber311: you upgraded to 1.6.6 and it got rid of that noice? |
| 15:52.22 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 15:52.23 | McLazarus | noise |
| 15:52.30 | Ciber311 | yup |
| 15:52.43 | McLazarus | hm, maybe I shouldn't upgrade then, since I don't have the noise now :) |
| 15:52.46 | alunt2003 | anyone know what i need to do, to show an incoming call from "012345" as "Dave"? |
| 15:53.00 | Blackthorn | I have a spa-2000 unit that is droping calls both incoming and outcoming to the local dialing area (* with DID Pri service). How can I log or determine why calls are being droped? |
| 15:53.37 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 15:54.00 | Tili | Does Echo Cancellation on Sangoma work with 2.4 kernel? |
| 15:54.08 | Blackthorn | alunt2003: thers a callerid field in the sip config i belive, and theres a way inthe extenions.conf that you set the clalder-id. |
| 15:55.09 | Ateboy | Cun ning: thanks... |
| 15:55.11 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:55.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:55.16 | Blackthorn | i think this may help http://www.voip-info.org/wiki-CallerID |
| 15:55.16 | CunningPike | Ateboy: It worked? |
| 15:55.22 | Ateboy | cunning: trying |
| 15:55.23 | Ciber311 | McLazarus: what version are you on right now? |
| 15:55.31 | CunningPike | Ateboy: OK - great |
| 15:55.36 | alunt2003 | Blackthorn: Thanks,i'll have a look |
| 15:56.23 | McLazarus | 1.6.5 |
| 15:56.40 | McLazarus | looking at 1.6.6 release notes now. Doesn't look to exciting |
| 15:57.21 | LoRez | would it be illegal to claim that a company is closed on a DID that a lot of people misdial? |
| 15:57.38 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1168029373.dsl.bell.ca) |
| 15:57.42 | momelod | hello people |
| 15:58.07 | [TK]D-Fender | Tili : Yes. Zaptel EC works on anything zaptel runs on, and Sangoma HWEC is well... HWEC... nothing to do with Kernel even |
| 15:58.12 | Ciber311 | LoRez: can't that fall under libel or something?? |
| 15:58.16 | momelod | anyone know of a place where i can read reviews on all the available hard phones compatible w/ asterisk? |
| 15:58.16 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 15:58.37 | LoRez | Ciber311: could it? what if I didn't specify the name of the company? |
| 15:58.51 | *** join/#asterisk mopri (n=jjohn@201.192.107.57) |
| 15:58.53 | Ciber311 | LoRez: do you really want to risk getting sued? |
| 15:58.54 | blaylock | would anyone know what this warning means? WARNING[8816]: format_wav.c:247 update_header: Unable to find our position |
| 15:58.56 | Ciber311 | remember |
| 15:58.59 | Ciber311 | this is america |
| 15:59.05 | Ciber311 | they don't even need to be right :P |
| 15:59.10 | LoRez | sued for idiots calling the wrong number... |
| 15:59.17 | [TK]D-Fender | McLazarus : 1.6.6 fixed a LOT of bugs in 1.6.5 and previous. Als inclresaes presence support to a functional level for receptionists on IP601 + Att modules |
| 15:59.41 | Ciber311 | fixed my weird noise too :P |
| 15:59.43 | LoRez | maybe I should just record a message saying "I'm sorry, but you're a dumbass, please check the number and dial again" |
| 15:59.45 | Strom_C | LoRez: just pass along a recorded message without supervising |
| 15:59.52 | Strom_C | that way the call never gets charged for |
| 16:00.05 | [TK]D-Fender | LoRez : If its your DID, you can whatever the heck yuou want with it... |
| 16:00.15 | McLazarus | [TK]D-Fender: yeah I do see the presence stuff. |
| 16:00.26 | LoRez | Strom_C: why would I want to do that? |
| 16:00.34 | [TK]D-Fender | McLazarus : 1.6.6 is worth it no matter what if your on 1.6.x at all |
| 16:00.34 | LoRez | [TK]D-Fender: that's what I would think. |
| 16:00.51 | Strom_C | LoRez: well I assume you're paying for inbound calls on that DID, right? |
| 16:01.21 | tRSS | [TK]D-Fender: may be you can help me. if I say members=>SIP/user0 in queues.conf, then user0 would be defined in sip.conf and when I say members=>Agents/user0, then user0 would be defined in agents.conf, correct? |
| 16:01.24 | McLazarus | [TK]D-Fender: cool, I'll have to check it out. The onlything in their release notes that interests me is the FTP thing. Sometimes the config doesn't download. |
| 16:01.31 | LoRez | it's on a PRI that's paid for, it's not costing me more to tell them off :) |
| 16:01.39 | Strom_C | heh alright |
| 16:01.49 | Strom_C | still, I like the not-supervising idea |
| 16:02.01 | Strom_C | but then again thats just my bell-shaped head |
| 16:02.19 | LoRez | unfortunately the pri doesn't terminate in a * box :( |
| 16:02.23 | Ateboy | I just tested using emailsubject=Voicemail message from ${VM_CIDNAME} <${VM_CIDNUM}> in my voicemail.conf, but it only applies to the e-mail, not the pager notification |
| 16:02.25 | LoRez | not this one. |
| 16:02.35 | Ateboy | Anyone knows how to apply it to both? |
| 16:02.52 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 16:03.37 | Ciber311 | LoRez: you can always play some pr0n sounds as an answering msg ;) |
| 16:03.38 | samourai1 | got 200 ok on register that isn't a register |
| 16:03.48 | LoRez | rofl |
| 16:03.48 | samourai1 | who knows the issue |
| 16:03.55 | samourai1 | for this bug |
| 16:04.07 | samourai1 | plz can anyone help me |
| 16:04.13 | Ciber311 | calm down dude |
| 16:04.22 | Strom_C | samourai1: calm the hell down |
| 16:04.27 | samourai1 | on asterisk sip registration |
| 16:04.40 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 16:04.50 | samourai1 | stom_c:want u to help me? |
| 16:04.56 | samourai1 | strom_c:want u to help me? |
| 16:05.20 | Strom_C | samourai1: don't worry about it if it isn't causing you problems. |
| 16:05.45 | samourai1 | i can't receive calls Strom_c |
| 16:06.13 | [TK]D-Fender | Ateboy : Pager doesn't have a body or attachments IIRC.. you've have to include it in the topic... |
| 16:06.28 | samourai1 | and instead i can make outbound calls |
| 16:06.38 | samourai1 | Strom_c:can u help me |
| 16:06.58 | [TK]D-Fender | samourai1 : How about you pastebin the SIP debug of a failed incoming call attempt so we have something to help you debug this... |
| 16:07.03 | Ateboy | TKD-Fender -> from what I can see on my cell, it has a subject only... |
| 16:07.18 | samourai1 | asterisk don't detect incoming calls |
| 16:07.24 | samourai1 | no debug |
| 16:07.37 | [TK]D-Fender | samourai1 : then its a networking problem. |
| 16:08.01 | [TK]D-Fender | Ateboy : So tahts where you'll have to put the name/number |
| 16:08.09 | samourai1 | i'm behind a nat 10.150.6.245 |
| 16:08.33 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 16:08.37 | samourai1 | strom_c:but when i put externip in sip.conf |
| 16:08.45 | samourai1 | it doesnt work too |
| 16:08.46 | [TK]D-Fender | samourai1 : pastebin your [general] section of sip.conf |
| 16:08.46 | [TK]D-Fender | ~pb |
| 16:08.52 | jbot | somebody said pb was 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/ |
| 16:08.52 | Ateboy | TK-D: I put it in the subject and it didn't change |
| 16:08.52 | samourai1 | ok |
| 16:08.55 | [TK]D-Fender | Ateboy L Hmmm |
| 16:09.00 | [TK]D-Fender | Ateboy : Dunno... |
| 16:09.37 | Ateboy | TK: I'll try in the body |
| 16:10.11 | [TK]D-Fender | Ateboy : I though we just confirmed that the body doesn't get sent to pager accounts |
| 16:11.16 | samourai1 | TKD-Fender:its too long my sip.conf what do u want to know about |
| 16:11.37 | Hmmhesays | nothing is too long |
| 16:11.52 | [TK]D-Fender | samourai1 : How can the [general] section be "too long"? I've pastebined 400+line diallplans.... |
| 16:11.58 | tzanger | Hmmhesays: you have not spoken to my gf then. :-P |
| 16:12.09 | [TK]D-Fender | samourai1 : And what I saked for should only be a dozen tops.. |
| 16:12.14 | Hmmhesays | she complaining about your pink sock? |
| 16:12.27 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 16:13.30 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:14.17 | *** join/#asterisk Teeli (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 16:14.23 | samourai1 | [general] |
| 16:14.24 | samourai1 | outboundproxy=enterprise.voip.meditel.ma |
| 16:14.24 | samourai1 | outboundproxyport=5060 |
| 16:14.24 | samourai1 | port=5060 |
| 16:14.24 | samourai1 | nat=yes |
| 16:14.24 | samourai1 | autocreatepeer=yes |
| 16:14.26 | samourai1 | externip=84.16.31.10 |
| 16:14.30 | samourai1 | canreinvite=yes |
| 16:14.32 | samourai1 | localnet=10.150.6.245/255.255.255.240 |
| 16:14.33 | Strom_C | STOP THAT |
| 16:14.34 | samourai1 | defaultexpirey=1800 |
| 16:14.36 | samourai1 | maxexpirey=1800 |
| 16:14.38 | samourai1 | ;context=incoming |
| 16:14.40 | samourai1 | bindport=5060; UDP Port to bind to (SIP standard port is 5060) |
| 16:14.42 | Strom_C | samourai1: STOP NOW |
| 16:14.42 | samourai1 | bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) |
| 16:14.44 | samourai1 | srvlookup=yes; Enable DNS SRV lookups on outbound calls |
| 16:14.46 | samourai1 | language=fr; Default language setting for all users/peers |
| 16:14.48 | samourai1 | ;insecure=very |
| 16:14.50 | samourai1 | dtmfmode = auto |
| 16:14.52 | samourai1 | ;context=sip2_context |
| 16:14.54 | samourai1 | context=inbound-sip |
| 16:14.56 | samourai1 | ;compactheaders = yes; send compact sip headers. |
| 16:15.00 | samourai1 | ;sipdebug = yes; Turn on SIP debugging by default, from |
| 16:15.02 | samourai1 | authdebug = yes |
| 16:15.02 | Strom_C | holy catsex |
| 16:15.04 | samourai1 | realm=voip.meditel.ma |
| 16:15.06 | samourai1 | ; Gestion des appels entrants |
| 16:15.08 | samourai1 | register=>20404040:*****:20404040@voip.meditel.ma@voip.meditel.ma:5061/20404040 |
| 16:15.10 | samourai1 | register=>20303030:******:20303030@voip.meditel.ma@voip.meditel.ma/20303030 |
| 16:15.12 | samourai1 | [voip.meditel.ma] |
| 16:15.14 | samourai1 | context=incoming |
| 16:15.16 | samourai1 | type=friend |
| 16:15.18 | samourai1 | host=voip.meditel.ma |
| 16:15.20 | samourai1 | fromdomain=voip.meditel.ma |
| 16:15.22 | samourai1 | from=voip.meditel.ma |
| 16:15.24 | samourai1 | dtmfmode=auto |
| 16:15.25 | De_Mon | thankyou.. |
| 16:15.25 | droops | no saying catsex in #asterisk |
| 16:15.26 | samourai1 | outboundproxy=enterprise.voip.meditel.ma |
| 16:15.30 | samourai1 | outboundproxyport=5060 |
| 16:15.32 | samourai1 | port=5060 |
| 16:15.34 | samourai1 | ;disallow=all |
| 16:15.36 | samourai1 | allow=all |
| 16:15.38 | samourai1 | nat=yes |
| 16:15.40 | samourai1 | canreinvite=yes |
| 16:15.42 | samourai1 | qualify=no |
| 16:15.44 | samourai1 | insecure=no |
| 16:15.46 | Ciber311 | lol this guy... |
| 16:15.46 | samourai1 | username=20404040@voip.meditel.ma |
| 16:15.48 | samourai1 | fromuser=20404040 |
| 16:15.50 | samourai1 | authname=20404040 |
| 16:15.52 | samourai1 | secret=****** |
| 16:15.53 | Strom_C | droops: do you have a notification set up for catsex? |
| 16:15.54 | samourai1 | ;projet ipbx----------------*****************------------------- |
| 16:15.56 | Ateboy | can someone kick this guy out? |
| 16:15.56 | samourai1 | [tel1] |
| 16:15.57 | Ciber311 | omg... |
| 16:15.58 | samourai1 | username=tel1 |
| 16:16.02 | samourai1 | type=friend |
| 16:16.04 | samourai1 | secret=1234 |
| 16:16.06 | samourai1 | allow=all |
| 16:16.08 | samourai1 | host=dynamic |
| 16:16.10 | samourai1 | context=sip1_context |
| 16:16.11 | droops | no notification, just a 6th sense |
| 16:16.12 | samourai1 | nat=yes |
| 16:16.14 | Strom_C | haha |
| 16:16.14 | samourai1 | [tel2] |
| 16:16.16 | samourai1 | username=20404040@voip.meditel.ma |
| 16:16.18 | samourai1 | type=friend |
| 16:16.20 | samourai1 | fromuser=20404040 |
| 16:16.22 | samourai1 | ;fromdomain=voip.meditel.ma |
| 16:16.24 | samourai1 | callerid="20404040" <20404040> |
| 16:16.26 | samourai1 | allow=all |
| 16:16.28 | samourai1 | secret=Ave404040 |
| 16:16.32 | samourai1 | host=dynamic |
| 16:16.34 | samourai1 | ;externip=81.16.31.10 |
| 16:16.34 | Ciber311 | this guy is gonna paste the entire source code at this rate |
| 16:16.36 | samourai1 | ;outboundproxy=enterprise.voip.meditel.ma |
| 16:16.38 | samourai1 | ;outboundproxyport=5060 |
| 16:16.40 | samourai1 | context=sip2_context |
| 16:16.42 | samourai1 | dtmfmode=rfc2833 |
| 16:16.44 | samourai1 | qualify=no |
| 16:16.46 | samourai1 | nat=never |
| 16:16.48 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:16.48 | samourai1 | TKDfender :this is my sip.conf |
| 16:16.48 | De_Mon | no. just a very large sip.conf |
| 16:16.50 | samourai1 | i'm sorry Strom_c |
| 16:16.52 | samourai1 | this guy want some help |
| 16:17.01 | Strom_C | samourai1: NEVER NEVER NEVER DO THAT AGAIN |
| 16:17.03 | droops | hey strom, im bringing lowtek_mystik with me, and the wife might tag along, although i doubt it |
| 16:17.06 | Strom_C | cool |
| 16:17.11 | Ciber311 | De_Mon: i'm joking :P |
| 16:17.15 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 16:17.15 | Strom_C | samourai1: pastebin.ca |
| 16:17.27 | TripleFFFF | hey guys.. anyway to see actual realtime trascoding ? .. |
| 16:17.37 | TripleFFFF | liek if a bridge is being transcoded |
| 16:17.38 | TripleFFFF | or not |
| 16:17.51 | De_Mon | samourai1 you may not get kicked now, but you will. incase you wonder why, this is the reason. |
| 16:18.09 | Hmmhesays | dr fraiser on stargate sg-1 is teh hot |
| 16:18.12 | Hmmhesays | i'd hit that hit that |
| 16:18.28 | droops | tommorow is goign to be a fun day |
| 16:18.38 | De_Mon | uh, you like old people eh? |
| 16:19.33 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 16:21.04 | TripleFFFF | also why does each menu in an IVR trigger a CDR |
| 16:21.07 | TripleFFFF | this is weird |
| 16:21.15 | tzanger | yeah she's kind of cute |
| 16:21.17 | TripleFFFF | so if you press 1 then 3 then 4.. you get 4 cdr records |
| 16:21.20 | TripleFFFF | why |
| 16:22.07 | Strom_C | droops: so tomorrow, realistically, I should expect like six carloads of people to be tagging along with you |
| 16:22.13 | TripleFFFF | ???? |
| 16:22.44 | Ciber311 | Hmmhesays: fraiser? |
| 16:22.54 | Hmmhesays | on stargate, the doctor |
| 16:22.54 | momelod | anyone know of a place where i can read reviews on all the available hard phones compatible w/ asterisk? |
| 16:22.56 | droops | me, another nerd, and maybe my wife |
| 16:23.00 | droops | and 5 other car loads |
| 16:23.03 | Hmmhesays | google |
| 16:23.08 | momelod | :/ |
| 16:23.15 | Ciber311 | Hmmhesays: didn't she die like seasons ago? |
| 16:23.23 | Hmmhesays | hush i'm only on season 4 |
| 16:23.28 | [TK]D-Fender | momelod : Company use? |
| 16:23.28 | Ciber311 | lol |
| 16:23.31 | droops | at around noon |
| 16:23.32 | Ciber311 | that would do it |
| 16:23.34 | TripleFFFF | just rename #asterisk to #blahblah |
| 16:23.35 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 16:23.38 | momelod | [TK]D-Fender: yes |
| 16:23.38 | coppice | i think he's talking about Frasier, not Fraiser :-) |
| 16:23.51 | Ciber311 | yeah she's dead :P |
| 16:24.01 | Hmmhesays | stuff it |
| 16:24.02 | momelod | [TK]D-Fender: i wanted things like message waiting, call transfer, hold.. |
| 16:24.12 | momelod | oh, and a headset would be nice too |
| 16:24.27 | [TK]D-Fender | momelod : Only I'd suggest for you are Polycom's lineup. |
| 16:24.27 | Ciber311 | Hmmhesays: don't worry she doesn't die for 3 more seasons |
| 16:24.38 | [TK]D-Fender | momelod : You have/want PoE? What kind of call volume? |
| 16:24.39 | dlynes_home | momelod: on the asterisk wiki |
| 16:24.57 | momelod | what is PoE? |
| 16:25.06 | dlynes_home | momelod: only ones worth looking at, unless you've got a boatload of cash for ciscos are: Aastra and Polycom |
| 16:25.12 | momelod | ive got a call center with 12 employees |
| 16:25.22 | [TK]D-Fender | momelod : Power Over Ethernet. So you can power them through their Rj45 wiring and not have to plug in a power brick |
| 16:25.24 | coppice | you've never heard of Edgar Alan PoE? :-) |
| 16:25.42 | momelod | wow, thats possible?! do i need a special switch for that? |
| 16:25.48 | dlynes_home | momelod: yes |
| 16:25.49 | [TK]D-Fender | momelod : Yes. PM |
| 16:26.33 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 16:26.37 | Ciber311 | i need to get a smaller POE switch |
| 16:26.55 | Ciber311 | got a 24 port in my room and it's loud as hell :P |
| 16:27.17 | [TK]D-Fender | Ciber311 : You don't need smaller, only quieter :) Often smaller means NOISIER. Look at 1U rac servers :) |
| 16:27.34 | Ciber311 | i meant smaller as in like 4 ports :P |
| 16:27.48 | Ciber311 | i don't need 24 POE ports in here |
| 16:27.52 | Ciber311 | well not usually |
| 16:27.58 | Ciber311 | at one point i used them all... :P |
| 16:28.30 | Ciber311 | or maybe i'll just open it up and unplug the fans ;) |
| 16:28.37 | Ciber311 | air conditioned anyway |
| 16:29.08 | dlynes_home | Ciber311: you could donate it to me, if you want |
| 16:29.16 | coppice | how come by air con moves several kilowatts around the room and is almost silent, but the fans driving a couple of hundred watss from my PC drive me nuts? something is seriously wrong with PC cooling |
| 16:29.31 | Ciber311 | lol |
| 16:29.37 | momelod | because the fans are smaller they have to spin faster |
| 16:30.09 | dlynes_home | coppice: because the fans in your aircon use grease and metal ball bearings; the fan in your pc is plastic ball bearings and air |
| 16:30.20 | Ciber311 | these damn switches need to be tempeture controlled |
| 16:30.30 | coppice | but they don't have to be. the designs are silly. you have a big back panel, with little noisy fans on it. |
| 16:31.16 | coppice | I think Daikin should try making PCs |
| 16:35.07 | nazgool | i have domain=10.0.0.200 listed in the general section of my sip.conf. still i get a strange error when my sip phone (inside the lan) wants to register to my asterisk: |
| 16:35.30 | nazgool | chan_sip.c:11043 handle_request_register: Registration from '<sip:fritzbox@10.0.0.200>' failed for '10.0.0.91' - Not a local SIP domain |
| 16:35.45 | nazgool | any clue what i'm doing wrong? |
| 16:36.21 | nazgool | 10.0.0.91 is the ip of the sip phone, 10.0.0.200 the ip of my asterisk server |
| 16:37.08 | momelod | nazgool whats your subnet mask? |
| 16:37.34 | *** join/#asterisk Twister (n=bob@host197.nextsub.ncn.net) |
| 16:38.25 | nazgool | 255.255.255.0 |
| 16:39.17 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 16:39.36 | Twister | hey all, im trying to compile asterisk 1.2.9.1 on a fedora core 3 box, ive compiled zaptel but when i compile asterisk it errors out when trying to copile ap_curl.so and says /usr/bin/ld Cannot find -lidn, whats your reccomendation? |
| 16:39.44 | [TK]D-Fender | nazgool : Check your SIP realm. that seems to be the problem, and its not on *'s side |
| 16:40.40 | nazgool | ok thx |
| 16:42.29 | *** part/#asterisk gambolputty (n=gambolpu@cblmdm72-240-85-84.buckeyecom.net) |
| 16:43.09 | tRSS | question: I have defined members=>Agents/user0 in queues.conf and I have defined user0 as agent=>101,101,user0 in agents.conf. how do these agents register their softphones with * box or how do they log into the queue, even if I register the softphone for them somehow? |
| 16:45.30 | [TK]D-Fender | tRSS : "show application agengcallbacklogin" |
| 16:46.05 | tRSS | never mind, found a page about the same stuff at asteriskguru, thanks for the help anyhow Fender :) |
| 16:46.08 | dlynes_home | ~justinu |
| 16:46.13 | jbot | justinu is probably some other d00d |
| 16:46.14 | dlynes_home | ~seen justinu |
| 16:46.29 | jbot | justinu <n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net> was last seen on IRC in channel #asterisk, 2d 15h 55m 9s ago, saying: '~seen sevard'. |
| 16:46.29 | Hmmhesays | ok me and buildroot are fighting again |
| 16:46.29 | Hmmhesays | libc/sysdeps/linux/common/vfork.c:13: error: `fork' undeclared here (not i |
| 16:46.29 | Hmmhesays | nction) |
| 16:46.35 | dlynes_home | ~seen justinu|laptop |
| 16:46.37 | jbot | justinu|laptop <n=Justin@12.44.122.130> was last seen on IRC in channel #asterisk, 1d 16h 54m 7s ago, saying: 'spa2100 runs my phones at home, and canada rules for providing such a wonderful crop'. |
| 16:47.42 | dlynes_home | Twister: you need to install libidn |
| 16:48.03 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:48.28 | dlynes_home | ~lart Qwell[] |
| 16:48.46 | Qwell[] | ~kill dlynes_home |
| 16:48.49 | jbot | ACTION shoots a magneto-ionized fluxproton gun at dlynes_home |
| 16:48.59 | dlynes_home | bbiab |
| 16:51.44 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 16:53.00 | *** join/#asterisk Kte2 (n=root@sipx.ica.net) |
| 16:54.43 | Kte2 | anyone know how to compile zaptel on FC5? says im missing sources for 2.6.15 but i have 2.5.16? |
| 16:55.04 | Qwell[] | 2.5.16? |
| 16:55.13 | Kte2 | er, sorry, 2.5.16 |
| 16:55.24 | Qwell[] | That's what I just said |
| 16:55.34 | Kte2 | >.< i cant type right now |
| 16:55.34 | CunningPike | Kte2: Bet you don't have kernel-devel installed |
| 16:55.37 | *** join/#asterisk wunderkin (n=wunderki@69.26.192.234) |
| 16:55.41 | Kte2 | 2.6.16 |
| 16:55.49 | Kte2 | i check |
| 16:56.13 | CunningPike | Kte2: Or, you don't have a symlink in your /usr/srtc |
| 16:56.21 | CunningPike | s/srtc/src/ |
| 16:56.34 | Kte2 | symlink? |
| 16:59.18 | Kte2 | ive got the kernel-devel package |
| 17:00.17 | *** join/#asterisk sparkleytone (n=abstephe@206.27.17.51) |
| 17:00.30 | *** mode/#asterisk [+o file] by russellb |
| 17:05.23 | *** join/#asterisk Tanker_ (n=Tanker@rrcs-24-172-41-210.se.biz.rr.com) |
| 17:06.11 | Dandan | topic |
| 17:06.14 | Dandan | oops :) |
| 17:06.54 | sparkleytone | i can't seem to get conferencing with PINs to work :/ |
| 17:07.27 | sparkleytone | if i don't set any PINs, then the conference works beautifully...but when i set either or both user/admin pins, i get disconnected once it is connected |
| 17:07.57 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 17:08.32 | *** join/#asterisk Sammich (n=brian@elk-en0.intercom.net) |
| 17:08.59 | tRSS | when I use the AgentLogin() application in my extensions.conf, the agent is unable to login. It just keeps saying that agent login incorrect, please try again. I have member => Agent/101 in queues.conf and agent => 101,101,user0 in agents.conf, but surprisingly, the agent is unable to login |
| 17:09.19 | mountainm2k | sparkleytone: any output from verbose / debug on the CLI? |
| 17:10.14 | key2 | what's this error: Jun 15 18:56:24 WARNING[6550]: chan_sip.c:1216 retrans_pkt: Maximum retries exceeded on transmission 6D8186BB-D2FA-4DF6-86AC-9699E493BC46@192.168.4.35 for seqno 2 (Non-critical Response) |
| 17:10.23 | tRSS | what am I missing that doesn't allow the agent to login? |
| 17:10.34 | key2 | tRSS: everything |
| 17:10.49 | tRSS | key2: good one ;) |
| 17:10.56 | sparkleytone | mountainm2k: i'll post two on pastebin...one of a working conference with no pins and one with a broken with pins |
| 17:12.25 | sparkleytone | hmmm...think i found it |
| 17:12.29 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-82.cpe.netcabo.pt) |
| 17:12.47 | sparkleytone | working: http://pastebin.com/711082 broken: http://pastebin.com/711083 |
| 17:13.05 | sparkleytone | enter-conf-pin-number... |
| 17:13.15 | sparkleytone | is that an audio file that i somehow don't have? |
| 17:13.36 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 17:13.51 | [TK]D-Fender | sparkleytone : You won't find much help with FreePBX/AMP here.. read the cahnnel topic... |
| 17:14.17 | sparkleytone | hehe i am in there too...they don't really respond to questions that are more advanced |
| 17:15.08 | sparkleytone | how did you know i was using fpbx anyway? you whois me? |
| 17:15.31 | [TK]D-Fender | sparkleytone : No, the blatantly obvious parts of your dial-plan exectution. |
| 17:15.47 | sparkleytone | hehe |
| 17:16.00 | [TK]D-Fender | sparkleytone : I can smeel that junk a mile away, especially from those trying to hide the fact thats what they're using... |
| 17:16.42 | sparkleytone | i'm not hiding anything, btw. if its any consolation, i'm generally not a noob. just in asterisk i am. but i am more than capable of helping others help me ;) |
| 17:16.49 | [TK]D-Fender | sparkleytone : contexts like "from-internal", "from-sip" are somewhat unique to them. |
| 17:17.05 | sparkleytone | [TK]D-Fender: gotcha. |
| 17:17.15 | [TK]D-Fender | sparkleytone : let me look at that a sec for you.... |
| 17:17.17 | dlynes_office | [TK]D-Fender: nah...I had those contexts, too...they were leftovers from the previous asterisk admin |
| 17:17.24 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 17:17.28 | dlynes_office | [TK]D-Fender: we were never using freepbx or @home |
| 17:17.34 | sparkleytone | is the enter-conf-pin-number an audio file i need to find? |
| 17:18.20 | sparkleytone | hmmm...they are made in my src already...i thought i installed them |
| 17:18.46 | [TK]D-Fender | dlynes_office : Yeah... AMP *leftovers*, EQUALLY trustworthy ;) |
| 17:18.59 | sparkleytone | hehe |
| 17:19.08 | sparkleytone | at least AMP let you edit configs directly...>:o |
| 17:19.30 | [TK]D-Fender | sparkleytone : Yup, go huinting for that file.. its clearly not there... |
| 17:19.38 | *** join/#asterisk trimi` (i=aaa@62.162.242.231) |
| 17:19.38 | sparkleytone | well i found the file |
| 17:19.44 | sparkleytone | now i just gotta find out where its looking |
| 17:19.47 | Qwell[] | I found him too |
| 17:19.48 | sparkleytone | guess i'll find gsm files |
| 17:19.54 | Qwell[] | :D |
| 17:20.00 | [TK]D-Fender | sparkleytone : So does vi/emacs/pico/nano/jed/mc/gedit.................................................. .ETC |
| 17:20.01 | sparkleytone | /var/lib/asterisk |
| 17:20.04 | Qwell[] | sparkleytone: /var/lib/asterisk/sounds/ |
| 17:20.18 | sparkleytone | nano, ftw ;) |
| 17:20.32 | Qwell[] | real men use ed |
| 17:20.39 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 17:20.44 | marl | hi there folks |
| 17:20.44 | sparkleytone | hehe |
| 17:20.56 | sparkleytone | TextWrangler.app |
| 17:21.07 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:21.26 | trimi` | errr can any1 tell me why i have very high latency when i use g729 codec? The voice comes after 1 sec. |
| 17:21.41 | Qwell[] | trimi`: are you on an incredibly slow box? |
| 17:21.44 | dlynes_office | trimi`: i don't think it's g729 causing that |
| 17:21.55 | Qwell[] | like, 8mhz |
| 17:22.03 | Qwell[] | or...a sunfire T2000 :D |
| 17:22.04 | trimi` | Qwell[] 1GHz |
| 17:22.15 | trimi` | i dont have prob with other codec |
| 17:22.16 | sparkleytone | can i just drop everything from asterisk-sounds to my /var/lib/asterisk/sounds? |
| 17:22.19 | trimi` | only with g729 |
| 17:22.24 | Qwell[] | sparkleytone: yes, just run a make install |
| 17:22.58 | sparkleytone | Qwell[]: could have sworn i had done that when i set up the box on tues...guess we'll see |
| 17:23.15 | sparkleytone | ah...nice little sh script |
| 17:23.35 | sparkleytone | interesting i hadn't run across a sounds issue until now |
| 17:24.33 | LH-euhost | Hello, i have a callback set up with * and use the DISA feature to establish a second phone line. How can i get the entered phonenumber in extension.conf after the "exten => _.,7,DISA(no-password|custom-callback)" entry? in which variable is the number saved? |
| 17:25.16 | Qwell[] | LH-euhost: line 1 of TFM answers this question |
| 17:25.23 | Qwell[] | and using _. is STUPID |
| 17:25.35 | sparkleytone | thanks for the help guys/girls |
| 17:25.40 | Dandan | is anyone on the asterisk-user link? can someone check if my post is there? |
| 17:25.41 | Dandan | :/ |
| 17:26.22 | dlynes_office | Dandan: huh? |
| 17:26.37 | Dandan | dlynes_home: i decided to post my problems to asterisk-user |
| 17:26.38 | Dandan | :) |
| 17:26.44 | Dandan | *asterisk-users |
| 17:26.50 | Dandan | and I can't see my own post :) |
| 17:27.01 | sparkleytone | hmmm...the sound doesn't say to press the # key :/ |
| 17:27.53 | LH-euhost | Qwell, can you point me to an URL? ${EXTEN} is not working |
| 17:28.07 | Qwell[] | LH-euhost: I assure you, it works |
| 17:28.08 | file | what's not working about it? |
| 17:28.13 | Qwell[] | thousands of people use it daily |
| 17:28.19 | tzanger | heh |
| 17:28.28 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 17:28.29 | dlynes_office | Dandan: Dan Elder? |
| 17:28.36 | dlynes_office | oh...nvm |
| 17:28.41 | SexyKen | Hey guys -- is it possible to execute a macro using the manager API? |
| 17:28.45 | Dandan | http://lists.digium.com/pipermail/asterisk-users/2006-June/155866.html |
| 17:28.48 | LH-euhost | lol, but in ${exten} i have "s" inside.. not the number entered from the user |
| 17:28.49 | Dandan | just arrived |
| 17:28.56 | dlynes_office | My folder hasn't finished refreshing yet |
| 17:29.02 | dlynes_office | I've got way too many messages in it |
| 17:29.05 | Dandan | I still have no email though... |
| 17:29.05 | Qwell[] | LH-euhost: like I said...using _. is STUPID |
| 17:29.15 | Dandan | dlynes_home: maildir go go go! |
| 17:29.31 | dlynes_office | Dandan: i am using maildir |
| 17:29.37 | dlynes_office | Dandan: over imap |
| 17:29.47 | LH-euhost | qwell, any hint what to use instead? |
| 17:29.49 | Qwell[] | over atm, over pppoe, over ssh |
| 17:29.56 | Qwell[] | LH-euhost: how about a proper pattern match? |
| 17:29.58 | SexyKen | Hey guys -- is it possible to execute a macro using the manager API? |
| 17:29.59 | Dandan | dlynes_home: i am mbox user over imapS :) |
| 17:30.07 | dlynes_office | Dandan: but that particular folder has about 15K messages right now, because I haven't had time to optimize it |
| 17:30.19 | Dandan | whoa :) |
| 17:30.25 | mopri | does anyone know how make asterisk work with callerid from alcatel..? I live in CostaRica, and the telephone Co, works with alcatel equipment. Ive tried the cidsignalling=bell,v23,dtmf and the cidstart with polarity and ring. any suggestions? |
| 17:30.30 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 17:30.48 | dlynes_office | Dandan: it's got all the asterisk-users mail from June 28, 2005 onwards |
| 17:30.59 | LH-euhost | is there a difference if i write _XXX. or _. ? if i want match all phone numbers |
| 17:31.03 | Dandan | mopri: CR is so BEAUTIFUL! :) i have been there. :) OTOH they just lost to ecuador 3:0 :D |
| 17:31.05 | dlynes_office | Dandan: minus the messages I've already sorted out into subfolders |
| 17:31.18 | Qwell[] | LH-euhost: yes, but neither are "proper" |
| 17:31.19 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:31.25 | Dandan | dlynes_home: sizeable :) |
| 17:31.36 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:31.36 | Qwell[] | ~docs |
| 17:31.46 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:31.46 | Qwell[] | LH-euhost: start there |
| 17:31.47 | Dandan | jbot is asleep |
| 17:32.20 | LH-euhost | thx then i go back to manuals |
| 17:32.38 | mopri | Dandan:.. CR sucked in the world cup.. :S |
| 17:33.34 | Twister | dlynes_home: thank you for your help |
| 17:33.42 | dlynes_office | Dandan: oops....62,591 emails in my main asterisk-users folder |
| 17:34.11 | dlynes_office | Dandan: 466MB's of email |
| 17:34.29 | Qwell[] | That's why I don't subscribe to -users ;) |
| 17:34.53 | [TK]D-Fender | dlynes_home : Thats why I just browse it on www.asteriskguru.com :) |
| 17:35.24 | dlynes_office | Qwell[]: well, only 105MB's in code commits :) |
| 17:35.53 | Dandan | mopri: they still have to play my home country |
| 17:35.55 | Dandan | :) |
| 17:36.01 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 17:36.20 | mopri | ..:S, yeah but we-re both out of the worldcup.. :S |
| 17:36.35 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 17:36.37 | Dandan | mopri: heh, true, oh well, RSA here we come :) |
| 17:36.44 | Dandan | (Rep. of S. Africa) |
| 17:37.02 | SexyKen | Yooohooooo? |
| 17:37.03 | dlynes_office | Dandan: only 1.7GB's of email, so asterisk-users is a considerable portion of it |
| 17:37.07 | SexyKen | Are you guys not here? |
| 17:37.20 | dlynes_office | SexyKen: no, we're definitely not |
| 17:37.22 | Dandan | dlynes_home: rm -rf ~ :> |
| 17:37.26 | SexyKen | Sure seems lik eit |
| 17:37.32 | Dandan | SexyKen: no, those are our talk-bots |
| 17:37.33 | Dandan | :) |
| 17:38.05 | Dandan | from changelog, i would say that slack 11.0 is coming! |
| 17:38.13 | *** part/#asterisk Tanker_ (n=Tanker@rrcs-24-172-41-210.se.biz.rr.com) |
| 17:38.23 | dlynes_office | Dandan: ummmm |
| 17:38.30 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 17:38.31 | dlynes_office | Dandan: it's been coming for more than six months now |
| 17:38.36 | dlynes_office | ~lart dandan |
| 17:38.52 | Dandan | dlynes_home: yeah, but recently Pat made quite a few updates |
| 17:38.54 | Dandan | to the tree |
| 17:38.57 | *** join/#asterisk reister (n=chatzill@m206-232.dsl.tsoft.com) |
| 17:39.01 | marl | ive been looking around th enet trying to find a solution to this problem but havnt found anything that seams to relate to it :( im getting a LOT of distortion on some of the recordings from monitor, can anyone give me any pointers to try and work out whats happening? in using sox/soxmix to combine in and out bound channels into sterio, and am finding that both channels are distorted |
| 17:39.10 | Dandan | jbot, you dumbass :) |
| 17:39.14 | dlynes_office | I'll believe it when I get a shiny new set of cds or dvds in the mail. |
| 17:39.31 | Qwell[] | with Gentoo |
| 17:39.33 | Dandan | dlynes_home: :) i am staying on bleeding edge :) |
| 17:39.43 | Dandan | GENTOO! BLAAAH! |
| 17:39.56 | Qwell[] | wtf, you just said you wanted bleeding edge :P |
| 17:39.58 | dlynes_office | Dandan: he's kidding...I think he uses slackware, too :) |
| 17:40.00 | marl | running AMD2200+/512Mb Ram/ only running asterisk |
| 17:40.01 | Dandan | I do not have a whole week to get my distro installed |
| 17:40.11 | Qwell[] | week...pfft |
| 17:40.14 | Qwell[] | 3 hours |
| 17:40.21 | Dandan | Qwell: i have bleeding edge with slackware :D |
| 17:40.28 | Dandan | Qwell: with stage 3? |
| 17:40.29 | Dandan | no way! |
| 17:40.35 | Qwell[] | there is only stage 3 now |
| 17:40.36 | wintix | debian ftw!1 ;) |
| 17:40.45 | dlynes_office | Qwell[]: yeah...on a Sunfire 15000Ghz, octal processor smp machine maybe |
| 17:40.56 | Qwell[] | on my athlon xp 2000 |
| 17:40.56 | Dandan | lol :) |
| 17:41.14 | Dandan | all the way? |
| 17:41.14 | Qwell[] | dlynes_office: I should add it to my distcc though :D |
| 17:41.32 | Dandan | :) |
| 17:41.43 | Qwell[] | Dandan: gentoo takes no time to install |
| 17:41.45 | SexyKen | Anyone here ever do a click-to-dial script using PHP? |
| 17:41.50 | SexyKen | click to call |
| 17:41.52 | dlynes_office | so i'm guessing the initial install of gentoo with all software that you would need for a normal desktop machine comes as binary packages, not source code on gentoo? |
| 17:42.03 | Qwell[] | dlynes_office: depends.. |
| 17:42.04 | [TK]D-Fender | Dandan: Is he finally caving to 2.6 as default stock kernel yet? Its been years now... |
| 17:42.13 | dlynes_office | [TK]D-Fender: probably |
| 17:42.14 | Qwell[] | gnome etc can be binary |
| 17:42.17 | dlynes_office | [TK]D-Fender: he would have to |
| 17:42.26 | dlynes_office | [TK]D-Fender: i would think, anyways |
| 17:42.27 | Qwell[] | or, you can compile everything (including glibc and gcc) |
| 17:42.34 | [TK]D-Fender | dlynes_home : No he doesn't... look how long he's lasted :) |
| 17:42.55 | Dandan | [TK]D-Fender: it is in testing |
| 17:42.55 | [TK]D-Fender | dlynes_office : I just have HOPES... |
| 17:42.55 | Dandan | and I like it that way :) |
| 17:43.01 | Dandan | 2.4 is MUCH more stable |
| 17:43.09 | [TK]D-Fender | I'm just afraid to upgrade my kernels :) |
| 17:43.11 | Qwell[] | does slackware still not have pam? |
| 17:43.16 | Qwell[] | if not...yeah...wtf |
| 17:43.34 | Dandan | it NEVER had |
| 17:43.39 | Dandan | and NEVER will :) |
| 17:46.11 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 17:46.20 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:46.31 | SexyKen | Hello |
| 17:46.32 | SexyKen | Can you |
| 17:46.35 | SexyKen | Execute a macro |
| 17:46.41 | SexyKen | Using whatever it's called |
| 17:46.43 | SexyKen | Manager API |
| 17:47.56 | *** join/#asterisk twilson (n=terry@69.17.122.227) |
| 17:54.56 | Ateboy | Where can I find what is the default setting for 'messagebody' in voicemail.conf? |
| 17:55.07 | Qwell[] | Ateboy: app_voicemail.c |
| 17:55.13 | dlynes_office | [TK]D-Fender: the kernel is the only thing on slackware that I change right away |
| 17:55.26 | Ateboy | qwell: ok |
| 17:55.40 | Dandan | dlynes_home: well... :) |
| 17:55.48 | Dandan | i do too :) |
| 17:56.25 | dlynes_office | Dandan: I'm just not using anything higher than 2.6.15 right now...i've seen too many people having problems in this channel trying to use 2.6.16 |
| 17:56.47 | dlynes_office | [TK]D-Fender: btw...2.6.15.5 seems to be working fine with sangoma a200d |
| 17:56.53 | dlynes_office | [TK]D-Fender: at least the driver's loading, anyways |
| 17:57.03 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:57.13 | dlynes_office | [TK]D-Fender: stupid thing still doesn't recognize that I have udev installed though (i even upgraded to the latest udev, and still no dice) |
| 17:57.46 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 17:57.47 | Ateboy | qwell: no way to see its voicemail.conf equivalent? |
| 17:58.05 | dlynes_office | Ateboy: take a look at the sample voicemail.conf that ships with make samples |
| 17:58.42 | dlynes_office | Ateboy: if you're running a prepackaged binary distribution of asterisk, it might be called voicemail.conf.sample, or voicemail.conf-dist, or something similar |
| 17:59.07 | Ateboy | dynes :got it |
| 17:59.28 | Ateboy | So it is not sending the CALLIDNUM by default... |
| 17:59.39 | Qwell[] | Ateboy: read the note |
| 17:59.49 | timscott | dlynes_office: I still use 2.4.32 as my kernel of choice. |
| 18:00.13 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 18:00.23 | rene- | hi, which package hold MOH? |
| 18:00.30 | Qwell[] | rene-: asterisk |
| 18:00.39 | rene- | really? i cant find it |
| 18:00.43 | rene- | in Mac OX |
| 18:00.45 | rene- | OS X |
| 18:00.47 | Ateboy | Qwell: oh, 512-char limit? |
| 18:00.48 | Qwell[] | install zaptel |
| 18:00.58 | Qwell[] | Ateboy: no, the note about CIDNAME |
| 18:01.00 | rene- | zaptel? really? |
| 18:01.15 | Qwell[] | rene-: meetme won't be compiled unless zaptel is installed |
| 18:01.23 | rene- | i dont want meetme |
| 18:01.28 | rene- | i want MOH |
| 18:01.33 | Ateboy | Qwell: nice |
| 18:01.36 | Qwell[] | then it'll be there by default |
| 18:03.02 | Ateboy | Qwll: how will I force it to send the VM_CIDNUM every time? |
| 18:03.16 | Qwell[] | by putting it in the message? |
| 18:03.46 | Ateboy | but if the CIDNAME is null, I will get CIDNUM twice, no? |
| 18:03.52 | Qwell[] | no |
| 18:03.57 | Qwell[] | reread the note |
| 18:04.09 | Ateboy | ; The following definition is very close to the default, but the default shows |
| 18:04.09 | Ateboy | ; just the CIDNAME, if it is not null, otherise just the CIDNUM, or "an unknown |
| 18:04.09 | Ateboy | ; caller", if they are both null. |
| 18:04.49 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 18:05.52 | Ateboy | I guess your are suggesting I use something like "\n\tHi ${VM_NAME},\n\n\tYou have a ${VM_DUR} long new voicemail message (number ${VM_MSGNUM}) in mailbox ${VM_MAILBOX}\nfrom ${VM_CIDNAME} (${VM_CIDNUM}), on ${VM_DATE}\nso you might want to check it when you get a chance.\n\n"? |
| 18:06.07 | Qwell[] | sure |
| 18:06.25 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 18:06.40 | rene- | nothing in /var/lib/asterisk/mohmp3 |
| 18:06.43 | rene- | weird |
| 18:06.49 | Qwell[] | rene-: trunk? |
| 18:07.15 | rene- | 1.2.9.1 |
| 18:07.20 | Ateboy | k |
| 18:07.23 | rene- | did not made samples |
| 18:07.24 | Qwell[] | they were probably taken out |
| 18:07.31 | Qwell[] | rene-: go find some MP3s to put in there |
| 18:07.34 | rene- | ok |
| 18:07.38 | rene- | i did skip make samples |
| 18:07.41 | Qwell[] | or use native, and find another format |
| 18:08.45 | Ateboy | Another question: what makes Asterisk send the busy message instead of unavailable? When I'm already on the phone? |
| 18:08.58 | Qwell[] | show application voicemail |
| 18:09.37 | Qwell[] | or look at macro-stdexten |
| 18:10.44 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 18:11.14 | Bullseye_Network | anybody know what happened to the addmailbox script in 1.2.9.1 I dont see it anywhere |
| 18:11.25 | Ateboy | Qwell: in s-BUSY |
| 18:11.34 | Qwell[] | Bullseye_Network: it hasn't been needed in well over a year |
| 18:11.36 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:12.08 | Bullseye_Network | ok, whats the eaisyest way to create the mailboxes then? |
| 18:12.19 | Qwell[] | Bullseye_Network: add it to voicemail.conf |
| 18:12.23 | rene- | note to all Asterisk1.2.x requires asterisk-addons to be installed for MOH to work |
| 18:13.07 | Qwell[] | rene-: No it doesn't |
| 18:13.07 | Bullseye_Network | it will create the dir its self? |
| 18:13.07 | Qwell[] | Bullseye_Network: yes |
| 18:13.07 | Bullseye_Network | Hmmm. |
| 18:13.08 | Bullseye_Network | Maybe thats why im having problems with it not recording. |
| 18:13.13 | rene- | Qwell[]: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 18:13.20 | Qwell[] | rene-: okay, well it's wrong |
| 18:13.22 | rene- | it is in bold |
| 18:13.29 | Qwell[] | I'm telling you, that that isn't the case :) |
| 18:13.54 | Qwell[] | If you want mp3s to work with native moh, then yeah, you'll need format_mp3, but any other sound file will work just fine |
| 18:14.25 | Qwell[] | ie; ogg vorbis |
| 18:14.49 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 18:15.06 | Ateboy | What triggers s-BUSY? |
| 18:15.18 | Qwell[] | Ateboy: reread the macro |
| 18:15.31 | Qwell[] | it's very clear |
| 18:17.01 | Bullseye_Network | ok it did NOT create the directorys for the voicemail |
| 18:17.04 | Ateboy | Qwell: all I can see is that I have a section of s-BUSY that invokes Voicemail() with "b" and a section of s-NOANSWER that invokes Voicemail() with a "u" |
| 18:17.12 | Bullseye_Network | or do I have to completely exit asterisk |
| 18:17.13 | Qwell[] | Bullseye_Network: it creates them when you leave a message |
| 18:17.21 | rene- | ah cool i am putting mp3 in those |
| 18:17.21 | Qwell[] | you do need to reload voicemail.conf though, obviously |
| 18:17.25 | Ateboy | so I must set my phone to DND to get the busy message? |
| 18:17.28 | Qwell[] | Ateboy: there is a goto |
| 18:17.34 | Bullseye_Network | ls |
| 18:17.38 | Bullseye_Network | ops |
| 18:17.43 | reister | Does anyone know of a way to install asterisk + zaptel on a virtual private server if you don't have access to the kernel source? |
| 18:17.53 | Bullseye_Network | says it cant accept messages |
| 18:19.17 | Ateboy | Qwell: I can see it, it is based on the ${DIALSTATUS} |
| 18:19.31 | Ateboy | but what makes that the ${DIALSTATUS} is set to one or the other? |
| 18:20.05 | Qwell[] | Ateboy: show application dial |
| 18:21.50 | Ateboy | Qwell: I'm there |
| 18:22.53 | Bullseye_Network | its definately not creating the sirectory |
| 18:22.56 | Bullseye_Network | directory |
| 18:23.00 | rene- | install a kernel source inside your VPS for the version of the OS your VPS is running? |
| 18:23.03 | Bullseye_Network | unless a reload is not sufficient |
| 18:23.16 | *** part/#asterisk oej[home] (n=oej@apollo.webway.se) |
| 18:23.23 | Qwell[] | Bullseye_Network: It creates it when you leave a message |
| 18:23.48 | Bullseye_Network | I called and it says: it cant accept messages |
| 18:25.10 | Qwell[] | pastebin the cli |
| 18:25.30 | Ateboy | Qwell: I can't find anything relevant there, I also looked in the book and in README.variables |
| 18:25.34 | Bullseye_Network | It appears that reload does NOT reload voicemail.conf |
| 18:28.22 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 18:28.47 | Ateboy | BTW, my pager body works perfectly now :) |
| 18:29.40 | mountainm2k | Silly question: any dialup solutions available for a * with a PRI? iaxmodem looked a bit promising, but it appears to _only_ do fax -- not regular modem calls... |
| 18:30.01 | rene- | Qwell: my files were mp3 so i did needed to install addons after all |
| 18:30.01 | Qwell[] | mountainm2k: use something else |
| 18:30.17 | mountainm2k | Like? |
| 18:30.22 | Qwell[] | Don't use asterisk for the sake of using asterisk |
| 18:30.26 | mountainm2k | an external modem on a Zap? |
| 18:30.50 | mountainm2k | I want to use asterisk because I'm already implimenting it -- won't be a ton of traffic... Like, one port is fine... |
| 18:31.48 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 18:32.19 | MikeJ[Laptop] | you want to try to do modem traffic on the TDM400 cards? |
| 18:32.23 | MikeJ[Laptop] | does that work well? |
| 18:32.57 | mountainm2k | Dunno, havn't tried it... Gotta believe it's better than, say, an IAXy or other ata |
| 18:33.12 | MikeJ[Laptop] | for sure... |
| 18:33.24 | mountainm2k | mostly this is for "I'm in the boonies, and need a connection"... It'll almost _never_ get used... |
| 18:33.36 | MikeJ[Laptop] | I would bet a t1 card w/ a channel bank would work better. |
| 18:33.46 | mountainm2k | I only bothered asking because I see it _will_ handle ISDN dialups |
| 18:34.02 | mountainm2k | hah, yeah, well, that's a lot more expensive... I'm talking about a single port here, not an ISP, heh |
| 18:34.33 | *** join/#asterisk NewSole (n=dave@d226-105-226.home.cgocable.net) |
| 18:34.42 | MikeJ[Laptop] | hello NewSole |
| 18:34.46 | NewSole | hi |
| 18:38.26 | Bullseye_Network | ok here is whats happening with voicemail.... When I call the mail box from an internal extention it works fine. When I call from outside the office it syas the mailbox cannot take more messages |
| 18:39.26 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 18:40.14 | Bullseye_Network | OR it says its recording but there is no file in the box |
| 18:40.54 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 18:42.14 | *** join/#asterisk opus_ (n=opus@68.216.187.60) |
| 18:42.17 | [TK]D-Fender | Bullseye_Network : Pastebin your whole config... |
| 18:42.18 | [TK]D-Fender | ~pb |
| 18:42.20 | jbot | [pb] 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/ |
| 18:42.35 | opus_ | hey guys have you ever expereniced astierks music on hold just dying after about a day? |
| 18:42.40 | gregj | whenever I try to call someone via asterisk, I get: Jun 15 11:42:06 WARNING[52387] app_dial.c: Dial requires an argument (technology/number) |
| 18:42.46 | gregj | what is it, and what might cause it ? |
| 18:42.52 | gregj | I am using SIP |
| 18:43.33 | dlynes_office | gregj: you've got a line in your dialplan such as exten => 1,1,Dial, or exten => 1,1,Dial() |
| 18:43.50 | dlynes_office | gregj: the dial command needs arguments, just as the error tells you |
| 18:44.11 | dlynes_office | gregj: so like exten => 1,1,Dial(SIP/100) |
| 18:44.32 | gregj | I am setting it up via astbill/realtime |
| 18:44.48 | dlynes_office | gregj: well, obviously you're not doing it correctly, then |
| 18:45.39 | gregj | honestly, I can't find any straight forward howto |
| 18:45.43 | gregj | so I am doing things blindly |
| 18:46.08 | CunningPike | gregj: What is your dial command? |
| 18:46.24 | dlynes_office | CunningPike: he's using astbill/realtime to generate it |
| 18:46.37 | gregj | dunno, I am trying to dial between two accounts using linphone |
| 18:46.38 | dlynes_office | CunningPike: so i'm guessing it's something fubar in his astbill setup |
| 18:46.46 | CunningPike | dlynes_office: Aha |
| 18:47.07 | dlynes_office | but i know absolutely nothing about astbill |
| 18:47.24 | gregj | I am trying to setup small office pbx, that will have SIP gateway |
| 18:47.45 | gregj | but since we change things a lot, I want it to run over SQL db, so I thought astbill would be the best solution |
| 18:47.47 | opus_ | has anyone else had problems with Music On Hold just dying after a while? |
| 18:47.57 | dlynes_office | woah |
| 18:47.57 | dlynes_office | nice |
| 18:48.02 | dlynes_office | just found a great system admin web site |
| 18:48.08 | dlynes_office | http://www.tech-recipes.com/ |
| 18:48.21 | gregj | so in case I would like to add someone to the list, the whole service wouldn't need to go down |
| 18:48.28 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-164-253.lsanca.fios.verizon.net) |
| 18:49.20 | [TK]D-Fender | gregj : You shouldn't ahve to take everything down jsut to add someone... |
| 18:49.42 | Gamercjm | i know this isnt about asterisk, but any one know about micro relays? |
| 18:49.45 | Gamercjm | or relays |
| 18:50.05 | *** part/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 18:50.09 | *** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) |
| 18:50.14 | gregj | [TK]D-Fender: but astbill sounds like the best solution here, managable via www, etc |
| 18:50.20 | gregj | great deal for dumb managers here |
| 18:50.48 | dlynes_office | gregj: if they're that dumb, i, personally wouldn't let them anywhere near the phone system |
| 18:51.09 | gregj | there's not many smart managers |
| 18:51.17 | gregj | at least not in my life expierence there was |
| 18:51.24 | dlynes_office | gregj: a lot of managers just appear dumb on purpose |
| 18:51.34 | Qwell[] | dlynes_office: ha...no |
| 18:51.36 | dlynes_office | gregj: so that they can get the upper hand |
| 18:52.21 | CunningPike | Those who can, do. Those who can't, manage |
| 18:52.32 | [TK]D-Fender | gregj : You really want dumb managers managing a PBX? |
| 18:52.34 | Qwell[] | no, no, no |
| 18:52.37 | dlynes_office | Qwell[]: well, your statement is true in a large company |
| 18:52.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:52.51 | Qwell[] | Those who can, do. Those who can't, teach. Those who can't teach, manage. |
| 18:52.55 | dlynes_office | Qwell[]: in a smaller company, the owner/manager is usually the brains of the operation |
| 18:53.07 | Qwell[] | dlynes_office: maybe |
| 18:53.21 | Qwell[] | but, all I really know is the corp world |
| 18:53.28 | Qwell[] | and let me tell you...most of my managers have been idiots |
| 18:53.37 | gregj | so, anyway |
| 18:53.41 | Qwell[] | ...I get a new one on a yearly basis |
| 18:53.45 | gregj | I add two users via astbill, |
| 18:53.49 | dlynes_office | Qwell[]: well, that's because managers higher people that are dumber than them so that they have job security |
| 18:54.00 | gregj | what should I do to let them call each other |
| 18:54.06 | gregj | in mysqldb |
| 18:54.17 | dlynes_office | Qwell[]: the guy at the top doesn't have to worry about that, cause he's the guy at the top |
| 18:55.42 | *** join/#asterisk Teeli (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 18:56.06 | [TK]D-Fender | gregj : Don't expect much help on GUI's around here, especially the less popular ones... |
| 18:56.44 | gregj | so which one should I use :) |
| 18:56.49 | NewSole | Mike you there |
| 18:57.10 | dlynes_office | hey....who the hell was it that said vista and longhorn were the same thing? |
| 18:57.24 | justinu|laptop | offtopic: what's the current state of the art in repartitioning software? still partition magic? |
| 18:57.31 | *** join/#asterisk jsolares (n=jsolares@125.209.191.2) |
| 18:57.34 | dlynes_office | justinu|laptop: and parted |
| 18:57.35 | Qwell[] | justinu|laptop: parted |
| 18:57.43 | gregj | justinu|laptop: on windows, yes |
| 18:57.44 | Qwell[] | qtparted rocks |
| 18:58.03 | justinu|laptop | <PROTECTED> |
| 18:58.09 | dlynes_office | Qwell[]: parted also does ntfs formatting? |
| 18:58.13 | Qwell[] | yep |
| 18:58.20 | Qwell[] | qtparted does anyhow |
| 18:58.31 | justinu|laptop | well, i assume I can boot some live CD distro and use parted instead of having to pay for partition magic |
| 18:58.33 | Qwell[] | I've used it several times to resize an ntfs partition |
| 18:58.40 | Qwell[] | justinu|laptop: knoppix comes with it |
| 18:58.43 | justinu|laptop | werd |
| 18:58.50 | justinu|laptop | thx |
| 18:58.53 | Qwell[] | there is also fips, but...may it die a long slow death |
| 18:59.06 | dlynes_office | justinu|laptop: slackware comes with it, too...it's in the extra directory |
| 19:00.26 | dlynes_office | Qwell[]: seems kinda stupid that qtparted supports ntfs formatting, but the linux kernel doesn't |
| 19:00.49 | Qwell[] | dlynes_office: I think it links to something that's lgpl |
| 19:00.55 | dlynes_office | Qwell[]: or rather mkfs |
| 19:00.57 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 19:01.06 | dlynes_office | The kernel is lgpl too |
| 19:01.15 | gregj | linux ? |
| 19:01.18 | Qwell[] | dlynes_office: hard to link to userspace stuff in the kernel though :) |
| 19:01.22 | gregj | is it just a joke ? |
| 19:01.34 | dlynes_office | gregj: yes, it is. That's why you're using it. |
| 19:01.37 | gregj | kernel is GPL with one exception - it allows to run non GPL stuff in userspace |
| 19:01.39 | gregj | and that's it |
| 19:01.53 | gregj | not LGPL at all |
| 19:01.55 | dlynes_office | ah...thought it was lgpl |
| 19:01.59 | gregj | nope |
| 19:02.07 | dlynes_office | maybe it was gpl v2 then? |
| 19:02.14 | gregj | it is GPL licence, and at the bottom of it it says: |
| 19:02.18 | mountainm2k | many a hardware vendor (tivo, linksys) has been bitten by that |
| 19:02.39 | mountainm2k | tivo now impliments their media filesystem as a user-space application, rather than a kernel module |
| 19:02.48 | gregj | dlynes_office: http://pastebin.com/711301 |
| 19:03.00 | gregj | dlynes_office: that's is what it says on top of the license, than plain GPL license follows |
| 19:03.49 | justinu|laptop | knoppix says it uses libntfs+fuse for NTFS write access |
| 19:04.02 | dlynes_office | ah...never read the license for the kernel, because i've never had the need to know |
| 19:04.18 | gregj | that was the fist thing I read before browsing the source |
| 19:04.22 | gregj | but that was in 2.0 days |
| 19:04.22 | gregj | :] |
| 19:04.33 | gregj | anyway, I still can't get bloody thing to work |
| 19:04.38 | gregj | nor help from anyone on it |
| 19:04.44 | gregj | nor good docs |
| 19:05.00 | justinu|laptop | what? |
| 19:05.17 | dlynes_office | gregj: still can't get what bloody thing to work? |
| 19:05.18 | Dr-Linux | dlynes_home: my spa-2100 is working fine :) |
| 19:05.20 | gregj | asterisk+astbill |
| 19:05.32 | gregj | dlynes_office: well, not the kernel obviously |
| 19:05.34 | dlynes_office | ah...i would think that would be easier than writing kernel drivers |
| 19:05.39 | justinu|laptop | Dr-Linux: hey man |
| 19:05.39 | Dr-Linux | dlynes_office: where can i find, the spa-2100 ivr codes? |
| 19:05.50 | dlynes_office | Dr-Linux: in the spa-2100 user guide |
| 19:05.52 | gregj | what is "the trunk" in asterisk terms ? |
| 19:06.05 | Dr-Linux | hey justinu|laptop: how are you friend? |
| 19:06.05 | gregj | it says everywhere that I should delete and recreate "the trunk" |
| 19:06.05 | dlynes_office | gregj: the latest bleeding edge that's sorta stable |
| 19:06.06 | justinu|laptop | gregj: latest development repository |
| 19:06.15 | gregj | http://astbill.com/node/606 |
| 19:06.22 | justinu|laptop | Dr-Linux: ups and downs :/ |
| 19:06.31 | justinu|laptop | my spa-2100 works great tho :) |
| 19:06.40 | Dr-Linux | justinu|laptop: happy with jen? |
| 19:06.58 | justinu|laptop | yeah |
| 19:07.02 | dlynes_office | gregj: trunk == phone line |
| 19:07.06 | gregj | oh |
| 19:07.09 | redondos | I need serious help, guys. Asterisk doesn't even start: http://pastebin.com/711313 |
| 19:07.13 | gregj | and how can I delete it ? |
| 19:07.14 | Dr-Linux | justinu|laptop: you know Pakistan main provider installed a software to stop the voip traffic :( |
| 19:07.18 | gregj | from mysql |
| 19:07.24 | justinu|laptop | ah, that kind of trunk |
| 19:07.24 | dlynes_office | gregj: they mean the trunk configuration |
| 19:07.29 | *** join/#asterisk aze (n=aze@ACayenne-101-1-3-12.w81-248.abo.wanadoo.fr) |
| 19:07.32 | justinu|laptop | Dr-Linux: wow, that really sucks |
| 19:07.32 | dlynes_office | gregj: not the trunk itself |
| 19:07.42 | Dr-Linux | our h323 voip devices are not working since 2 weeks |
| 19:07.58 | justinu|laptop | anyone know why trunks are called such? |
| 19:07.59 | dlynes_office | gregj: you're gonna need to find someone on here that knows asterisk realtime configuration to help you |
| 19:08.05 | Dr-Linux | justinu|laptop: but just yesterday, SIP starting work |
| 19:08.21 | gregj | delete from asttrunk; might do it |
| 19:09.45 | dlynes_office | justinu|laptop: a trunk is usually defined as a group of communications banded together to form a trunk |
| 19:09.54 | justinu|laptop | yeah, but why "trunk" |
| 19:10.06 | dlynes_office | justinu|laptop: but for whatever reason, the telecom industry also refers to it as a single copper pair for one analog phone line, too |
| 19:10.29 | justinu|laptop | well, in real telco vernacular a trunk is a intermachine connection |
| 19:10.37 | justinu|laptop | and a line goes to a customer |
| 19:10.53 | Dr-Linux | justinu|laptop: www.syednetworks.com << i placed here the PK provider stuff, it was in today's URDU newspaper |
| 19:11.20 | dlynes_office | justinu|laptop: well, what is the base of a tree called? you know hte woody part that carries the water and food to all the extremeties of the tree's branches? |
| 19:11.38 | *** join/#asterisk gby (i=gby@l192-117-111-92.broadband.actcom.net.il) |
| 19:11.47 | justinu|laptop | anyways, the explanation I heard is that trunks are trunks because the very first coax that could carry FDM traffic was as thick as an elephant trunk |
| 19:12.04 | justinu|laptop | 4-5 inches in diameter |
| 19:12.11 | ids2500 | lol |
| 19:12.12 | ids2500 | LOL |
| 19:12.13 | ids2500 | no |
| 19:12.14 | ids2500 | hahahaha |
| 19:12.32 | Dr-Linux | justinu|laptop: can you include 1 inch more :P |
| 19:12.46 | dlynes_office | Dr-Linux: He's only got 5 inches |
| 19:12.50 | drray | no way, then my penis would be a trunk |
| 19:12.56 | drray | ooops lag! |
| 19:13.22 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 19:13.25 | dlynes_office | drray: go blow on your trunk, then |
| 19:13.33 | *** join/#asterisk Samoied (n=Samoied@200.180.6.202) |
| 19:13.34 | Dr-Linux | drray: what codecs? :S |
| 19:14.16 | drray | capn crunch |
| 19:14.26 | Qwell[] | oh boy |
| 19:14.33 | Qwell[] | drray: bad idea to say that name in here :P |
| 19:14.49 | Dr-Linux | Qwell[]: what name? |
| 19:14.59 | Qwell[] | Dr-Linux: oh no, you aren't going to trick me |
| 19:15.00 | dlynes_office | Dr-Linux: cap'n crunch |
| 19:15.19 | Qwell[] | crunch man == ultraperv |
| 19:15.24 | Dr-Linux | Qwell[]: i don't know what's cap`n crunch |
| 19:15.38 | Dr-Linux | ~dict cap`n crunch |
| 19:15.39 | dlynes_office | Dr-Linux: a high profile phreaker from the late 60's/early 70's |
| 19:15.52 | dlynes_office | ~wiki cap'n crunch |
| 19:15.54 | justinu|laptop | why is it a bad idea ? |
| 19:16.08 | Qwell[] | gby: ...too nice |
| 19:16.09 | dlynes_office | ~wiki john draper |
| 19:16.30 | Qwell[] | c'mon, read the last paragraph in the wiki article.. |
| 19:16.43 | Qwell[] | I dare you :D |
| 19:16.49 | drray | :) |
| 19:17.01 | *** join/#asterisk martijn_ (n=martijn@i155156.upc-i.chello.nl) |
| 19:17.20 | justinu|laptop | draper told me he phreaked a call to richard nixon |
| 19:17.20 | dlynes_office | Qwell[]: whistle that was, at the time, Qwell's favorite toy.... |
| 19:17.36 | drray | while he was blowing 2600hz |
| 19:17.45 | Dr-Linux | sorry guys, |
| 19:17.45 | drray | I apologize to all for starting this |
| 19:17.45 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 19:17.51 | Qwell[] | he blows more than 2600hz... |
| 19:17.59 | drray | trunksmoker |
| 19:18.05 | gby | justinu|laptop: yeah, he also says he once reached the whitehouse secretery and asked for toilet paper... |
| 19:18.12 | justinu|laptop | yeah |
| 19:18.36 | Dr-Linux | gby: who gave him toilet paper? Bush? :S |
| 19:18.37 | justinu|laptop | afaik, a lot of telco guys blow more than whistles |
| 19:19.02 | gby | Dr-Linux: try tricky dicky. it was the 70s ;-) |
| 19:19.08 | drray | I think this would be nixon |
| 19:19.21 | drray | since this story is older than a lot of you |
| 19:19.29 | drray | :) |
| 19:19.32 | Qwell[] | justinu|laptop: this is true |
| 19:19.48 | justinu|laptop | worst kept industry secret ever |
| 19:19.51 | Qwell[] | but...umm... |
| 19:19.55 | Dr-Linux | gby: :S sorry? |
| 19:19.55 | Qwell[] | captain crunch is just scary |
| 19:19.58 | drray | keep that secret |
| 19:20.02 | justinu|laptop | heh |
| 19:20.07 | Dr-Linux | gby: i know dick word though |
| 19:20.20 | gby | Dr-Linux: it's a nick name for Richard Nixon |
| 19:20.30 | justinu|laptop | dick is a nickname for anyone named richard |
| 19:20.39 | Dr-Linux | who the fuck is Richard Nixon? |
| 19:20.40 | justinu|laptop | how the hell did that get started, anyways? |
| 19:20.41 | drray | and a few people who are not named richard |
| 19:20.43 | Qwell[] | pardon the pun... |
| 19:20.58 | Dr-Linux | awww |
| 19:21.00 | Qwell[] | but, if crunch learned any secret information, while calling Nixon... |
| 19:21.05 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 19:21.06 | Qwell[] | he could blow the whistle on dick? |
| 19:21.08 | Dr-Linux | one of our manger name is Richard ... |
| 19:21.11 | justinu|laptop | Dr-Linux: 37th president of the USA |
| 19:21.13 | drray | deep throat! |
| 19:21.13 | Dr-Linux | let me ask him |
| 19:21.19 | Dr-Linux | i see |
| 19:21.31 | justinu|laptop | he resigned because of impending impeachment in 1974 |
| 19:21.38 | Qwell[] | okay, nobody liked my pun(s).. |
| 19:21.42 | drray | I did! |
| 19:22.20 | drray | here is a virtual bluebox I wrote |
| 19:22.25 | drray | sound 2600,5 |
| 19:22.28 | drray | the end |
| 19:22.31 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 19:22.35 | Dr-Linux | ~dict Richard |
| 19:22.36 | justinu|laptop | dr-linux: his most famous line was: "I am not a crook" |
| 19:22.56 | Dr-Linux | ~dict crook |
| 19:23.06 | justinu|laptop | lol |
| 19:23.09 | justinu|laptop | wrong |
| 19:23.12 | drray | :) |
| 19:23.17 | justinu|laptop | crook == a criminal |
| 19:23.23 | justinu|laptop | a theif |
| 19:23.30 | Dr-Linux | i see |
| 19:23.32 | Qwell[] | i before e.. |
| 19:23.50 | Qwell[] | stop teaching our foreign friend bad grammar :p |
| 19:23.59 | drray | nixon ordered wiretapping of DNC headquarters, then covered it up |
| 19:24.22 | Dr-Linux | Qwell[]: i know criminal and theif meanings, bcoz we have many here :P |
| 19:24.27 | Qwell[] | thief! |
| 19:24.32 | Qwell[] | justinu|laptop: See what you've done? |
| 19:24.37 | justinu|laptop | except words like vein, freight, deceive, etc. |
| 19:24.39 | Qwell[] | You broke him. :P |
| 19:24.46 | justinu|laptop | lol |
| 19:24.49 | Dr-Linux | heh |
| 19:24.57 | drray | which is odd because deep throat, was also arrested and sent to prison for wiretapping political adversaries |
| 19:25.10 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 19:25.41 | redondos | I solved my previous problem: a card was not tight enough, it appears that the owner was mangling it earlier today. |
| 19:25.50 | viler | It is possible to config a Queue with a member sip account from another host ? member => SIP/USER1@another host ???? |
| 19:25.51 | justinu|laptop | deep throat turned out to be FBI associated director |
| 19:26.00 | drray | who was passed over when Hoover died |
| 19:26.02 | justinu|laptop | i never heard he went to jail |
| 19:26.18 | drray | reagan pardoned felt |
| 19:26.18 | redondos | Can you help me creating reports from the logs generated by asterisk? Is there some documentation about this? |
| 19:26.23 | drray | so he might not have gone to jail |
| 19:26.42 | Dr-Linux | how can i cross my ISP's software? :S |
| 19:26.43 | justinu|laptop | tough guy: http://upload.wikimedia.org/wikipedia/en/4/4c/Felt1958.jpg |
| 19:27.03 | drray | heh |
| 19:27.09 | mpruett | He didn't go to jail - they just found who deep throat was a several months ago |
| 19:27.26 | drray | I just wonder what would ahve happened had Felt not been passed over, nixon would not ahve been charged |
| 19:27.33 | justinu|laptop | yeah, it's true that reagan pardoned him |
| 19:27.42 | Dr-Linux | drray: you are a bad guy or nice one? :S |
| 19:27.53 | drray | Dr-Linux, probably bad |
| 19:28.09 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 19:28.10 | Dr-Linux | drray: grrrr, you like Bush? |
| 19:28.17 | drray | not so much |
| 19:28.23 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 19:28.37 | C4T3l | hello all |
| 19:28.44 | Dr-Linux | drray: someone told me he is a gay :S |
| 19:28.53 | justinu|laptop | can't be |
| 19:28.55 | justinu|laptop | he's not in telco |
| 19:29.00 | Dr-Linux | lol |
| 19:29.08 | drray | well, unless it was the guy getting 2 1/2" inches from him, I'd dispute it |
| 19:29.17 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 19:29.19 | Dr-Linux | awww |
| 19:29.36 | justinu|laptop | besides, the guys at the top have a lot worse fetishes than homosexuality |
| 19:29.45 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 19:30.00 | drray | power |
| 19:30.09 | Chotaire | greetings all.. short question: is it possible to set Call Duration in a .call file? |
| 19:30.24 | Chotaire | like call shall be disconnected after specified amount of minutes.. I'm missing that feature. |
| 19:30.49 | Dr-Linux | Chotaire: i don't think it can be done. |
| 19:31.31 | justinu|laptop | since outgoing spool calls don't go thru app_dial, probably no |
| 19:31.32 | justinu|laptop | t |
| 19:32.01 | gby | anyone knows how to keep busydetect from detecting flase detecting the telco ring tone as a busy signal? |
| 19:32.19 | gby | anyone knows how to keep busydetect from false detecting the telco ring tone as a busy signal? |
| 19:32.23 | Chotaire | hm, that would suck ;) |
| 19:32.56 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 19:33.00 | gby | Chotaire: it happens with the biggest Israeli telco :-/ |
| 19:33.01 | Delvar | iv got to say it INGERLAAAAAND! 2 nill :D |
| 19:33.48 | *** join/#asterisk Greek-Boy (n=Greek-Bo@193.220.93.162) |
| 19:33.54 | gby | muk_ibook : it works with the qemu in CVS or so reported on the mailing list |
| 19:33.56 | Chotaire | gby: you changed indications? |
| 19:34.23 | Chotaire | hm, I wonder if that would have any impact on busydetect though |
| 19:34.24 | gby | Chotaire: as far as i could see from the code busydetect does not *use* indications... |
| 19:34.58 | gby | #qemu : muk_ibook it works with the qemu in CVS or so reported on the mailing |
| 19:34.59 | gby | <PROTECTED> |
| 19:35.03 | gby | gr... |
| 19:35.07 | gby | stupid IRC client |
| 19:35.20 | Chotaire | gby: no other idea, sorry. |
| 19:35.31 | gby | Chotaire: thanks anyway |
| 19:35.46 | Chotaire | open a bug report on that one. |
| 19:36.08 | gby | Chotaire : i prefer to open bug reports only when i can attach the patch to fix it;-) |
| 19:36.29 | justinu|laptop | gby: so how does it work then? hardcoded cadence? |
| 19:36.31 | Chotaire | since there were problems with different dial parameters, I was forced to use .call files instead.. now that .call files won't handle "call duration", I am getting pissed by the dial command again.. I think I should keep bugging people about fixing app_dial too |
| 19:36.45 | justinu|laptop | Chotaire: trust me, you're not alone |
| 19:37.27 | Chotaire | justinu: I have had to use dial with several mixed parameters in a very specific configuration. they did not work together, they were simply ignored. |
| 19:37.31 | Chotaire | hopefully that will be fixed one day. |
| 19:37.34 | gby | justinu|laptop : as far as i can tell it measures the differences between high cadence and low one and if it falls in certain ranges it flags it as a "busy" beep |
| 19:38.10 | justinu|laptop | "high cadence"? |
| 19:38.29 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 19:38.54 | gby | justinu|laptop : grr... english is not my mother tongue, sorry. |
| 19:38.57 | justinu|laptop | no prob |
| 19:39.11 | justinu|laptop | cadence is the rhythm of the tones |
| 19:39.42 | justinu|laptop | it's usually a lot easier to detect a tone based on its cadence, than to analyze the frequencies in the tone with an FFT or something |
| 19:39.47 | dlynes_office | Greek-Boy: hey...still having problems, eh? |
| 19:41.07 | Greek-Boy | dlynes; yes :( |
| 19:41.16 | Greek-Boy | but I applied for my service contract |
| 19:41.17 | gby | justinu|laptop : the source for busydetects has a remark about being in "half cadance" |
| 19:41.21 | Greek-Boy | now i gotta wait 2 weeks! |
| 19:41.45 | Greek-Boy | yeah thats cisco! unbelievable how the biggest networking company on earth can treat us like this :( |
| 19:41.56 | dlynes_office | Greek-Boy: because they're cisco |
| 19:42.09 | dlynes_office | Greek-Boy: they know you'll put up with their bs, and still love them afterwards :p |
| 19:42.16 | justinu|laptop | what's the problem? |
| 19:42.23 | Greek-Boy | cisco = network version of microsoft |
| 19:42.36 | gby | justinu|laptop : anyways, basically yes, except it only cares about detecting alternating silence and tone in a constant rythm, it does not seem to care what that rythm is and the Israeli dial tone sadly fits the pattern :-( |
| 19:42.45 | Strom_C | difference being that cisco products actually tend to work |
| 19:42.53 | Qwell[] | Strom_C: pretty much.. |
| 19:43.20 | dlynes_office | I notice how Qwell[] wasn't too much in the affirmative on that one... :) |
| 19:43.30 | justinu|laptop | gby: so your "ringback" indication and "busy" indication are very similar in israel? |
| 19:43.33 | Greek-Boy | what's the best way to record all calls in mp3 format (all channels) ? |
| 19:43.53 | dlynes_office | Greek-Boy: probably mix-monitor, or monitor |
| 19:43.55 | gby | justinu|laptop : not to a human ear, but yes for the busydetect algorythm :-) |
| 19:43.56 | justinu|laptop | i don't think there's any code in ast that can encode an MP3 |
| 19:43.58 | Qwell[] | Greek-Boy: Nothing |
| 19:44.00 | Strom_C | Greek-Boy: use monitor, then sox to mix em, then lame to encode em |
| 19:44.06 | dlynes_office | oh...nvm |
| 19:44.11 | justinu|laptop | or mixmonitor and cut out the sox step |
| 19:44.12 | dlynes_office | i didn't see the mp3 |
| 19:44.23 | Strom_C | mixmonitor only does gsm though |
| 19:44.27 | Strom_C | and that's icky sounding |
| 19:44.37 | Dr-Linux | i'm still using sox package |
| 19:44.37 | Qwell[] | you sure? |
| 19:44.38 | justinu|laptop | we lay down calls in PCM ulaw with mixmonitor |
| 19:44.45 | Strom_C | oh, does it do that now? |
| 19:44.46 | dlynes_office | mixmonitor will do any format that asterisk is capable of saving in |
| 19:44.50 | Strom_C | last I checked it was only gsm |
| 19:44.56 | Qwell[] | hell, it still uses the old Monitor format |
| 19:45.18 | gby | justinu|laptop : anyway, instead of trying to analyze the tones or rythm, i figure it will be much simpler to introduce a channel variable that will determine if busydetect will operate or not on specific calls |
| 19:45.20 | dlynes_office | well, any format that asterisk is capable of transcoding to, i men |
| 19:45.23 | dlynes_office | s/men/mean/ |
| 19:45.37 | Greek-Boy | so in the internal context i can setup recording in each extension or just add it to my main macro. and then for incoming context |
| 19:45.56 | mountainm2k | Speaking of recording, anybody have a good way to put a feature button on a SIP phone that starts recording a call when pressed? |
| 19:46.05 | gby | justinu|laptop : then i can only turn it on for incoming calls and not for outgoing calls. sort of a hack, but a generic one for all the places where busydetects misbehaves but still is useful |
| 19:46.06 | *** join/#asterisk izod (n=izod@mail.crowdercollege.net) |
| 19:46.41 | gby | mountainm2k : any button that you can program to send DTMF of "*8" will do that with a default asterisk install and proper dialplan |
| 19:47.25 | izod | anyone have an agi or dialplan solution for mass dialing radio contests? |
| 19:47.27 | mountainm2k | default asterisk, but the book had me write a dialplan from scratch... |
| 19:47.34 | justinu|laptop | izod: lol |
| 19:47.38 | mountainm2k | izod lol |
| 19:47.41 | mountainm2k | nice idea |
| 19:47.48 | dlynes_office | izod: heh...i used to use my telix autodialer for that |
| 19:47.57 | justinu|laptop | i was doing that 10 years ago with dialogic D240SCT1 cards |
| 19:48.00 | mountainm2k | I used to do that w/ old PBX and PRI -- redial was so damn fast it was unbelievable, I won all kinds of crap |
| 19:48.03 | izod | gotta ask, eh? |
| 19:48.20 | izod | :) |
| 19:48.54 | [TK]D-Fender | dlynes_home : As in the old DOS comm prog? |
| 19:49.13 | izod | we've got a nortel pbx here, but I'm not the telecom guy, so I don't get to jack with it much |
| 19:49.22 | dlynes_office | izod: why not just have an extension you dial into that dials a certain phone number, then loops when it's busy, hangs up, and redials? |
| 19:49.31 | izod | but I'm running an asterisk box on this DS-3 connection |
| 19:49.35 | dlynes_office | [TK]D-Fender: yeah, DOS and Windows...never used the windows version |
| 19:49.44 | justinu|laptop | bah... real BBSers used telemate |
| 19:49.45 | dlynes_office | [TK]D-Fender: minicom would work though, too |
| 19:49.56 | dlynes_office | justinu|laptop: yeah...that's what i used later on |
| 19:50.10 | dlynes_office | justinu|laptop: and then zcomm or something like that later, in OS/2 |
| 19:50.23 | justinu|laptop | lol viva OS/2! |
| 19:50.24 | izod | dlynes_office: yeah. could do that... I was looking for an easy way to light up 50 or so lines and be connected to the one(s) that didn't get a busy |
| 19:50.25 | [TK]D-Fender | dlynes_home : I remember using it way back it was my favourite. Qmodem (I believe it was called) was another, as well as one I wrote based on those. |
| 19:50.51 | justinu|laptop | gparted has a liveCD that's only 32meg |
| 19:50.52 | dlynes_office | [TK]D-Fender: yeah...the best all round one for DOS thought was Telemate |
| 19:50.59 | [TK]D-Fender | justinu|laptop : Yesh, Telemate was fn... that one had psuedo-multitasking IIRC) |
| 19:51.09 | justinu|laptop | d-fender: indeed |
| 19:51.13 | dlynes_office | [TK]D-Fender: i just liked telemate cause it had a 60 line mode |
| 19:51.19 | [TK]D-Fender | justinu|laptop : And "window'd" etc |
| 19:51.26 | justinu|laptop | i liked that you could edit a text file while downloading something |
| 19:51.28 | dlynes_office | oh yeah...and it had a windowing mode |
| 19:51.31 | dlynes_office | forgot about that |
| 19:51.33 | [TK]D-Fender | dlynes_home : Mine rocked, but then again... I did write it myself :) |
| 19:51.38 | izod | my favorite term program was JRComm on the Amiga... slick stuff |
| 19:51.47 | justinu|laptop | JRcomm was ok |
| 19:51.56 | justinu|laptop | the real shiznit on amiga was Cnet BBS software |
| 19:52.02 | dlynes_office | ewwwww |
| 19:52.04 | izod | ayep... ran a Cnet bbs for a while |
| 19:52.18 | dlynes_office | i had to log into that crap on some loser's bbs that was running cnet bbs on a c64 :) |
| 19:52.46 | izod | haha... I think I ended up with a $600 phone bill one month calling BBS's all over the country |
| 19:52.56 | justinu|laptop | directory opus was also very cool |
| 19:52.57 | dlynes_office | izod: yeah...$500 here |
| 19:53.06 | dlynes_office | izod: but i rang up $500/mo bills regularly |
| 19:53.09 | justinu|laptop | apparently they're still developing it for windoze! |
| 19:53.12 | izod | dir opus rocked. |
| 19:53.33 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 19:53.36 | gbodemantv | hi all |
| 19:53.43 | izod | dlynes_office: ouch. I hovered around $150 usually... $500 every month would have killed me |
| 19:54.00 | justinu|laptop | http://www.gpsoft.com.au/Intro.html |
| 19:54.24 | gbodemantv | any ever get 482 Loop detected?? |
| 19:54.36 | gbodemantv | not allowing me to dial my sip peers |
| 19:54.45 | gbodemantv | just goes to voicemail |
| 19:54.57 | gbodemantv | chan_local.c:496 local_alloc: No such extension/context 8201@default creating local channel |
| 19:54.59 | dlynes_office | izod: well, this was $500Cdn/mo, but still |
| 19:55.16 | dlynes_office | izod: i lived in Thunder Bay, so every decent BBS was long distance |
| 19:55.16 | Greek-Boy | i already have a receipt for my service contract even though I dont have the number. Lets hope that can stand in lawsuits if there is any cisco employees in here. lol |
| 19:55.17 | gbodemantv | app_dial.c:473 wait_for_answer: Unable to create local channel for call forward to 'Local/8201@default' (cause = 0) |
| 19:55.18 | justinu|laptop | real men just hacked a tymnet account and used local outdials to call remote BBSes :P |
| 19:55.22 | gbodemantv | any clues??? |
| 19:55.37 | dlynes_office | izod: i specialized in Amiga, Macintosh, OS/2 and programming software |
| 19:55.42 | *** join/#asterisk elriah (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 19:55.51 | justinu|laptop | man, the glory days of X.25 |
| 19:56.07 | elriah | Hi all. For some reason, after doing a 1.07 to 1.2 upgrade, my directory won't find name matches from voicemail.conf, any suggestions? |
| 19:56.31 | izod | dlynes_office: yeah... I was in the middle of nowhere... it was cheaper to call out of state bbs's than in-state |
| 19:56.35 | dlynes_office | elriah: might help to pastebin your voicemail.conf file |
| 19:57.01 | izod | justinu|laptop: heheh.. local bbs's? ha! none around unfortunately |
| 19:57.02 | justinu|laptop | if you had a local indial to an X.25 network, it was almost as good as the internet |
| 19:57.03 | dlynes_office | izod: well, everywhere was long distance though for me...even other cities in the same province |
| 19:57.11 | elriah | It's just a bunch of entries like this: 801 => 4444,First Last |
| 19:57.17 | justinu|laptop | and thanks to compuserve, there were quit a few indials to x.25 nets |
| 19:57.20 | dlynes_office | izod: most of the time, i usually called to Alabama and Texas |
| 19:57.43 | dlynes_office | izod: they were the bbses with the largest file troves, and they ran Courier HST's, so I could connect at 14.4K |
| 19:57.45 | izod | dlynes_office: yeah... I usually hit New York, New Jersey and California |
| 19:57.51 | elriah | Is there something I have to do to turn on the directory in 1.2? |
| 19:58.00 | dlynes_office | oh yeah...and Rusty 'n Edie's in California...forgot about them :) |
| 19:58.02 | justinu|laptop | courier HST, telebit trailblazers |
| 19:58.06 | justinu|laptop | those were the days |
| 19:58.25 | justinu|laptop | i remember spending 600 bucks on a courier dual standard |
| 19:58.29 | dlynes_office | they had a lot of great FLI files :) |
| 19:58.38 | elriah | Got it. Wrong context. lol |
| 19:58.46 | dlynes_office | Yeah...I spent $700 on an HST, myself |
| 19:59.01 | dlynes_office | justinu|laptop: i ugess you must've bought that dual standard after the prices had started to come down |
| 19:59.08 | mountainm2k | OK, I have my test *, with a single FXO, and a single IP phone... Only I could be using it... But still I get a fast busy sometimes when dialing out -- what can I do to see why? |
| 19:59.15 | justinu|laptop | i remember getting some kinda "sysop" discount |
| 19:59.25 | dlynes_office | I still have my HST sitting on my bookshelf :) |
| 19:59.39 | justinu|laptop | then I had to go buy some 1336 16550 UARTs so I could run the com ports at 56k |
| 19:59.42 | Nugget | the sysop prices were 50% off ($500 or so for a classic HST) |
| 19:59.42 | justinu|laptop | or 115k |
| 20:00.01 | justinu|laptop | s/1336/1337/ |
| 20:00.05 | dlynes_office | I have no idea why I lugged the HST all the way out here from thunder bay, though |
| 20:00.13 | justinu|laptop | nostalgia |
| 20:00.14 | Nugget | http://slacker.com/photos/computers/IMG_0864 |
| 20:00.17 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 20:00.23 | dlynes_office | It's a 3500 mile trip to get here |
| 20:00.31 | justinu|laptop | the USR modems could dial the fastest, and detect busy faster than anything I ha |
| 20:00.34 | Nugget | http://slacker.com/~nugget/stuff/courierhst.txt |
| 20:00.48 | tzanger | oh man |
| 20:00.51 | tzanger | USR |
| 20:01.20 | C4T3l | what is q921 protocol? |
| 20:01.30 | tzanger | with v32bis and what? |
| 20:01.36 | justinu|laptop | i got a supra v32bis modem that ran so hot I could cook my breakfast on it |
| 20:01.40 | tzanger | normally you would say MNP5 but it doesn't say that |
| 20:01.46 | tzanger | justinu|laptop: I had a friend with one of those |
| 20:02.05 | tzanger | the supra 2400 you could set s11 so low that it'd dial faster than the telco could discriminate |
| 20:02.07 | justinu|laptop | C4T3l: you mean q931? |
| 20:02.10 | Qwell[] | Nugget: motd for USR? |
| 20:02.14 | Qwell[] | Did they pay you? heh |
| 20:02.16 | Nugget | I miss the round-led hst with the mechnical relay that went *click* whenever it connected at 9600. |
| 20:02.20 | justinu|laptop | tzanger: USR was like that too |
| 20:02.30 | Nugget | Qwell[]: no, but the modems were half-price if you did that. |
| 20:02.34 | justinu|laptop | i never had one that old |
| 20:02.35 | Nugget | saved $500. |
| 20:02.36 | Qwell[] | ahh...cool |
| 20:03.10 | Qwell[] | jeebus |
| 20:03.35 | justinu|laptop | yeah, even at half price they were insanely expensive for 15 year olds |
| 20:03.39 | alunt2003 | Guys,Im trying to get caller-id to show names instead of numbers. I have "exten => 6458155,2,Set(CALLERID(07929902xxx)=Alun Mobile)" but i get this error "Unknown callerid data type" Any ideas? |
| 20:04.03 | Qwell[] | alunt2003: Set(${CALLERID(name)}) |
| 20:04.05 | Qwell[] | erm |
| 20:04.05 | dlynes_office | damn....microsoft is so confusing, even the windows-heads over at ##windows don't even know what the difference is between longhorn and vista :p |
| 20:04.11 | Qwell[] | alunt2003: Set(${CALLERID(name)}=blah) |
| 20:04.18 | [TK]D-Fender | Qwell : Equally bad |
| 20:04.24 | Qwell[] | [TK]D-Fender: ? |
| 20:04.29 | file | silly people |
| 20:04.30 | alunt2003 | Qwell[]: Thanks i'll try that |
| 20:04.31 | [TK]D-Fender | alunt2003 : Set(CALLERID(name)=blah) |
| 20:04.33 | file | Set(CALLERID(name)=blah) |
| 20:04.35 | *** part/#asterisk opus_ (n=opus@68.216.187.60) |
| 20:04.36 | Qwell[] | oh, duh |
| 20:04.40 | gby | dlynes_office : longhorn is the engineering code name for what marketing vall vista |
| 20:04.50 | dlynes_office | gby: see? you don't know either |
| 20:05.11 | Qwell[] | I know the difference! |
| 20:05.14 | dlynes_office | gby: Vista is the Desktop/Home Edition version of the next version of Windows. Longhorn is the server version |
| 20:05.19 | Qwell[] | longhorn was going to have every feature under the sun |
| 20:05.23 | Qwell[] | vista has none |
| 20:05.37 | gby | dlynes_office : i am no exactly what you would call a windows expert. or Windows user for that matter :-) |
| 20:05.57 | justinu|laptop | did you guys see that vista premium will require hybrid harddrives? |
| 20:05.58 | dlynes_office | everybody and their dog seems to think that longhorn is the codename, and vista is the release name, including everyone in ##windows |
| 20:06.04 | mountainm2k | repost: OK, I have my test *, with a single FXO, and a single IP phone... Only I could be using it... But still I get a fast busy sometimes when dialing out -- what can I do to see why? |
| 20:06.10 | dlynes_office | it's pretty funny |
| 20:06.16 | mountainm2k | I've even restarted asterisk, totally restarted, not reload |
| 20:06.24 | mountainm2k | seems like that channel is just wedged |
| 20:06.25 | gby | dlynes_office : the better questions is: who cares? :-) |
| 20:06.28 | Qwell[] | dlynes_office: Did you ever think, that maybe you're the one that's wrong? :D |
| 20:06.29 | mountainm2k | and I can't un-wedge it |
| 20:06.34 | dlynes_office | Qwell[]: nope |
| 20:06.35 | Qwell[] | 10 out of 10 people can't be wrong ;) |
| 20:06.43 | justinu|laptop | dlynes_office: wait, you think that windows iwll market a product called "longhorn"? |
| 20:06.48 | justinu|laptop | s/windows/microsoft |
| 20:06.53 | dlynes_office | justinu|laptop: yes |
| 20:06.58 | dlynes_office | justinu|laptop: they already are |
| 20:07.02 | Qwell[] | the server version is called "Vista Server edition" |
| 20:07.05 | justinu|laptop | so I can go buy longhorn? |
| 20:07.09 | dlynes_office | justinu|laptop: longhorn and vista are both in beta 2 |
| 20:07.13 | justinu|laptop | and the box says longhorn on it? |
| 20:07.20 | justinu|laptop | i'm so there |
| 20:07.20 | dlynes_office | I would imagine so, yeah |
| 20:07.37 | Qwell[] | http://www.microsoft.com/windowsvista/getready/editions/default.mspx |
| 20:07.40 | dlynes_office | They just unveiled both of them at WINHEC |
| 20:08.10 | izod | well... off to figure out this radio contest thing... whee! |
| 20:08.21 | izod | cya... |
| 20:08.28 | nazgool | i have my asterisk on a machine that does nat/firewall. i'd like it to be able to place and receive calls with a sip provider (e.g. sipgate), but that no one else can log in to sip on my asterisk. what do i have to do to make this sure? |
| 20:08.35 | justinu|laptop | ~nat |
| 20:08.40 | jbot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 20:09.24 | Qwell[] | http://www.microsoft.com/presspass/press/2005/jul05/07-22LHMA.mspx |
| 20:09.29 | nazgool | note: my asterisk isn't operating from behind the nat a priory (should i do that?) |
| 20:09.32 | Qwell[] | :D |
| 20:09.43 | mountainm2k | why is there no "zap debug" ? |
| 20:09.52 | Qwell[] | http://www.microsoft.com/windowsserver/bulletins/longhorn/beta1.mspx |
| 20:09.56 | Qwell[] | dlynes_office: I could go on.. |
| 20:10.01 | justinu|laptop | nazgool: "a priory"? |
| 20:10.29 | justinu|laptop | forgive my ignorance |
| 20:10.59 | dlynes_office | Qwell[]: then why are they still referring to the server edition as longhorn and the desktop edition as vista? |
| 20:11.18 | Qwell[] | ask them |
| 20:11.27 | Qwell[] | but, those articles were pretty clear about the name.. |
| 20:11.32 | VoicePulse | Because the server edition doesn't have a retail name yet. |
| 20:11.39 | alunt2003 | Qwell[]: I put in exten => 6458155,2,Set(${CALLERID(07929902xxx)}=Alun Mobile) but still no joy. |
| 20:11.48 | VoicePulse | In keeping with established naming conventions, the next version of Windows Server, currently codenamed Longhorn Server, will retain the Windows Server 200x moniker used its predecessors, Microsoft officials said at the TechEd 2006 trade show this week in Boston Massachusetts. Given its projected late 2007 release date, Longhorn Server will therefore be named Windows Server 2007 or Windows Server 2008. |
| 20:11.51 | Qwell[] | alunt2003: remove the ${} |
| 20:11.52 | file | this is interesting |
| 20:12.02 | alunt2003 | Qwell[]: Ok |
| 20:12.15 | nazgool | justinu|laptop: ahm "a priori" actually (typo). means something like "unless there should be a very good and unknown/unstated reason for the opposite to be true" |
| 20:12.26 | justinu|laptop | nazgool: ah |
| 20:12.29 | dlynes_office | Qwell[]: there ya go, longhorn is not hte same thing as vista |
| 20:12.37 | dlynes_office | Qwell[]: just ask voicepulse :p |
| 20:12.43 | Qwell[] | well, voicepulse are idiots |
| 20:12.50 | justinu|laptop | lol |
| 20:12.52 | file | LOL |
| 20:13.05 | justinu|laptop | voicepulse: thems fightin' words |
| 20:13.09 | Qwell[] | :D |
| 20:13.30 | nazgool | so anyways, my asterisk isn't behind a nat. i could place it behind the nat though, since it's on the same machine that does the nat. should i put it behind the nat (i always thought asterisk behind a nat means more problems) ? |
| 20:13.45 | vader-- | hmmm any of you guys have a like a asterisk going live check list |
| 20:13.45 | vader-- | hehe |
| 20:14.23 | justinu|laptop | nazgool: no, your initial instinct is right |
| 20:14.41 | nazgool | ok thx |
| 20:14.44 | justinu|laptop | however, does that machine have multiple ethernet interfaces? |
| 20:14.50 | nazgool | yup 2 |
| 20:15.03 | justinu|laptop | there might be some issue with asterisk binding to the wrong one in certain situations |
| 20:15.03 | nazgool | one internal (lan), one external (ppp) |
| 20:15.07 | justinu|laptop | i remember having problems with that before |
| 20:15.11 | dlynes_office | "Longhorn Server will feature the Windows Workflow Foundation, IIS 7, Serial ATA, Communications Foundation, Federated Identity, Network Access Protection, Dynamic Partitioning, Windows Virtualization Hypervisor, Service Hardening Windows Firewall, a next generation TCP/IP stack, and enhanced Terminal Services" - Bob Muglia, Sr. VP of Server and Tools Business at Microsoft |
| 20:15.24 | nazgool | well so far it seems to work with respect to that |
| 20:15.28 | justinu|laptop | i still think it's a codeword |
| 20:15.32 | dlynes_office | That's a quote from last month, Qwell |
| 20:15.35 | mountainm2k | it appear's I have accidently joined a MSFT channel... |
| 20:15.47 | alunt2003 | Qwell[]: I removed the brackets. I now get a new error "Function $CALLERID not registered" |
| 20:15.52 | dlynes_office | mountainm2k: lol |
| 20:16.18 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 20:16.27 | file | algorithmn: Set(CALLERID(name |
| 20:16.29 | file | GAH |
| 20:16.34 | file | algorithmn: Set(CALLERID(name)=My CallerID Name) |
| 20:16.40 | file | Set(CALLERID(num)=8005551212) |
| 20:16.45 | file | there, setting name and number |
| 20:17.24 | dlynes_office | Qwell[]: even microsoft's calling it Longhorn: http://www.microsoft.com/windowsserver/longhorn/default.mspx |
| 20:17.24 | justinu|laptop | or even Set(CALLERID(all)=John Doe <8005551212>) |
| 20:18.05 | justinu|laptop | dlynes_office: the html title of that page is 'Microsoft Windows Server code name "Longhorn" Beta 2 Home' |
| 20:18.28 | dlynes_office | justinu|laptop: yeah, and mentions nothign about vista |
| 20:18.41 | alunt2003 | file: but my original line was "exten => 6458155,2,Set(CALLERID(07929902xxx)=Alun Mobile)" thats correct isnt it,but it doesnt work |
| 20:18.51 | file | no, it's not correct |
| 20:19.03 | nazgool | is it longHORN? i thought it was longWAIT ? |
| 20:19.07 | *** join/#asterisk MartianLobster (n=clarks@m815f36d0.tmodns.net) |
| 20:19.11 | justinu|laptop | nazgool: heh, exactly |
| 20:19.12 | droops | hey im trying to use set, Set(callagent=12223334444) but then i do a noop(${callagent}) and i get nothing |
| 20:19.25 | MartianLobster | what is a good way to restart the asterisk server from trixbox? |
| 20:19.33 | MartianLobster | safe_asterisk, doesn't seem to do it |
| 20:19.34 | justinu|laptop | ~trixbox |
| 20:19.39 | MartianLobster | justinu|laptop: ok thanks |
| 20:19.42 | dlynes_office | MartianLobster: shutdown -r now NOW!!!!!!! |
| 20:19.46 | justinu|laptop | wtf is trixbox?? |
| 20:19.54 | dlynes_office | justinu|laptop: the new name of AAH |
| 20:19.57 | justinu|laptop | oh boy |
| 20:19.58 | *** join/#asterisk S^P (n=masood@203.148.73.236) |
| 20:20.13 | justinu|laptop | silly wabbit, trix are for kids! |
| 20:20.31 | Katty | rut roh |
| 20:20.32 | Katty | i need help. |
| 20:20.33 | dlynes_office | justinu|laptop: it's #freepbx's way of trying to trick us into helping their little army |
| 20:20.48 | Katty | http://pastebin.com/711473 <- that's my usb flash drive. |
| 20:20.51 | Katty | how do i mount it |
| 20:21.08 | file | mount /dev/sda1 /mnt |
| 20:21.23 | file | provided you want it mounted on /mnt |
| 20:21.24 | mountainm2k | yeah, that'd do it |
| 20:21.27 | dlynes_office | or mount /dev/sda1 /mnt -t vfat |
| 20:21.34 | Katty | file: dankou |
| 20:21.41 | S^P | hi I'm configuring a call center, and wana knw about the number of g.729 license we need . setup is => call recieving using SIP and forward on a SIP channel to Quintum. |
| 20:22.05 | mountainm2k | It appears I waited long enough, my ZAP channel un-wedged itself -- wish I knew what was causing that, as it's very irritating... |
| 20:22.08 | file | huh |
| 20:22.10 | S^P | I enabled CDR functionality of asterisk (MOH) |
| 20:22.12 | justinu|laptop | "wedged"? |
| 20:22.12 | file | when did I become opped? |
| 20:22.20 | justinu|laptop | a while ago |
| 20:22.32 | mountainm2k | "wedged" in that it won't call out -- when I try, I get a fast-busy |
| 20:22.43 | mountainm2k | although incoming calls still work |
| 20:22.43 | dlynes_office | file: russell opped you in asterisk-dev; i don't know who opped you here |
| 20:22.52 | mountainm2k | only one phone, and only one line, so it's got to be me... |
| 20:22.56 | file | I have access for dev... oh well! |
| 20:22.59 | justinu|laptop | mountainm2k: what's in your debug files when you try and dial out? |
| 20:23.05 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 20:23.48 | mountainm2k | just says "Executing Dial(), then Called <number> and then Zap/4-1 answered |
| 20:23.50 | mountainm2k | that's it |
| 20:23.55 | Netgeeks | Hey folks, quick question, is there a general consensus on asterisk performance on Hyperthread systems, in other words HT on or off preferred? |
| 20:24.08 | dlynes_office | Netgeeks: off |
| 20:24.15 | justinu|laptop | mountainm2k: it says that when you get a fast busy? |
| 20:24.15 | Qwell[] | Netgeeks: Which NIC did you say you were using in the e4500? |
| 20:24.24 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:24.24 | Netgeeks | bcm57xx |
| 20:24.34 | file | ooh broadcom |
| 20:24.38 | mountainm2k | justinu|laptop: yes, I can pastebin if you want, but it's really only that simple... |
| 20:24.44 | Qwell[] | ok |
| 20:24.46 | dlynes_office | broadcom kicks ass |
| 20:24.52 | mountainm2k | and it doesn't actually dial out, it's not the co giving me that... |
| 20:24.52 | dlynes_office | especially if you're using a digium card |
| 20:25.11 | [TK]D-Fender | ;) |
| 20:25.16 | justinu|laptop | mountainm2k: so what is giving you fast busy? |
| 20:25.17 | dlynes_office | my firmware's toast on my broadcom network card :((( |
| 20:25.36 | dlynes_office | something went super screwy under freebsd and then my broadcom went bye-bye |
| 20:25.52 | dlynes_office | had to throw a couple of 3com's in there, instead |
| 20:26.02 | mountainm2k | Hmm, actually... I just dialed out from SIP, got fast busy, and while I was listening to it I dialed in from outside world, got a busy... So maybe it _IS_ the CO giving me that... |
| 20:26.12 | Qwell[] | Netgeeks: and that's a quad port, right? |
| 20:26.20 | justinu|laptop | mountainm2k: if you're reall interested in solving it, it can be solved |
| 20:26.30 | dlynes_office | [TK]D-Fender: actually, my install for 2.6.15.5 for sangoma a200d went quite smooth |
| 20:26.35 | Netgeeks | I'm using the 5700 which is a two port card |
| 20:26.49 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:27.04 | mountainm2k | justinu|laptop: Well, I obvioulsy need to solve it... |
| 20:27.13 | justinu|laptop | mountainm2k: if you want to verify whether it's CO generated fast busy, or something else, turn on sip debug peer <yoursipphone> |
| 20:27.15 | Netgeeks | there are some quad port broadcoms, but you have to be careful, some of them use 'software' to drive all 4 ports and some have hardware |
| 20:27.26 | justinu|laptop | make your call, pastebin the sip dialog |
| 20:27.31 | [TK]D-Fender | dlynes_office : So it was only that bad UDEV version hmm? |
| 20:27.43 | Qwell[] | Netgeeks: tried a cassini? |
| 20:27.50 | Netgeeks | I don't know the details myself, but I was warned to make sure I got the right card |
| 20:27.52 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 20:27.56 | syzygybsd | has anyone installed zaptel on 2.6.15 or later? |
| 20:28.18 | Netgeeks | Qwell: cassini? um the only thing that I know that has that name is a space probe of some sort |
| 20:28.28 | Qwell[] | heh |
| 20:28.28 | syzygybsd | after looking up the error it looks like an option was removed from the kernel that it is trying to set, I can't compile it with this kernel |
| 20:28.40 | Qwell[] | It's a NIC the Sun techs have suggested using |
| 20:29.08 | Netgeeks | ah, no, never tried it |
| 20:29.11 | justinu|laptop | another offtopic: if I have a bootable ISO image, and a USB key, how do I put that ISO onto the USB key so I can boot off it? |
| 20:29.22 | Netgeeks | the 4500 has been turned off for over a month now :( |
| 20:29.27 | Qwell[] | Netgeeks: heh |
| 20:29.32 | gby | dlynes_office : /quit |
| 20:29.33 | alunt2003 | file: Nope,i'm horribly confused. If i put "exten => 6458155,2,SetCIDName(Alun Mobile)" it works-unfortunatly for any number that calls they become "Alun Mobile". If i use exten => 6458155,2,Set(CALLERID(07929902xxx)=Alun Mobile) it just wont work |
| 20:29.38 | mountainm2k | http://pastebin.com/711514 -- from CLI, debug 10 and verbose 10 |
| 20:29.46 | Qwell[] | Netgeeks: I get to test Linux on a new sunfire next week (or tomorrow, with any luck) |
| 20:29.57 | Netgeeks | Qwell: what model? |
| 20:30.03 | Qwell[] | T2000, 8 core ;) |
| 20:30.15 | file | alunt2003: which is all correct in behavior, you just don't understand exactly how it all works and what's right/not - what are you trying to do? |
| 20:30.20 | justinu|laptop | mountainm2k: so yeah - it looks like that fast busy is coming from CO |
| 20:30.26 | Netgeeks | nice, I guessing you will have some serious issues getting linux to work on it |
| 20:30.41 | mountainm2k | Grrr... OK, test-set in hand.... Back in a few... |
| 20:30.46 | Qwell[] | Netgeeks: nah, I have 100% support from Sun...and that's straight from Jonathan Schwartz. :) |
| 20:30.51 | justinu|laptop | mountainm2k: i thought your sip phone itself might generate the busy, but the SIP dialog says otherwise |
| 20:31.00 | alunt2003 | file: Im trying to get my own mobile number to show up as "Alun Mobile" |
| 20:31.05 | mountainm2k | fast busy for ALL numbers I try, not just ofc... |
| 20:31.31 | Netgeeks | Qwell: hrm, nice, keep me up to date. I'm interested in how well a non-transcoding asterisk will run on the T1/2000 |
| 20:31.40 | file | alunt2003: when you're placing outgoing calls? |
| 20:31.47 | file | or... |
| 20:31.48 | justinu|laptop | mountainm2k: how about "0"? |
| 20:31.53 | alunt2003 | file: No incoming |
| 20:32.03 | *** join/#asterisk cardiffit (n=sb@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 20:32.10 | cardiffit | wassup |
| 20:32.16 | dlynes_office | [TK]D-Fender: no, i suspect it was other "issues" |
| 20:32.25 | file | exten => 6458155/07929902xxx,1,Set(CALLERID(name)=Alun Mobile) |
| 20:32.28 | dlynes_office | [TK]D-Fender: including a botched sangoma install |
| 20:32.34 | dlynes_office | [TK]D-Fender: that I couldn't seem to undo |
| 20:33.30 | [TK]D-Fender | dlynes_office : Well its all sounding "gold" now... |
| 20:33.45 | *** join/#asterisk X-Rob (n=rob@CPE-60-231-85-52.qld.bigpond.net.au) |
| 20:34.13 | dlynes_office | [TK]D-Fender: hrm...got asterisk all configured for basics |
| 20:34.25 | dlynes_office | [TK]D-Fender: but i'm getting wrtdm Board 1 UNCONFIGUR 0 0 0 |
| 20:34.36 | vader-- | asterisk doesn't include sound files for regular numbers 0-9 with their sound addon? |
| 20:34.42 | dlynes_office | [TK]D-Fender: when i do a zap show status |
| 20:34.51 | Strom_C | vader--: those come with asterisk :) |
| 20:35.02 | vader-- | in the lady's voice? |
| 20:35.05 | Strom_C | yes |
| 20:35.11 | vader-- | whats the sound file for number nine? |
| 20:35.14 | Strom_C | her name is Allison |
| 20:35.18 | Qwell[] | digits/9.gsm |
| 20:36.06 | vader-- | nice |
| 20:36.07 | vader-- | thank you |
| 20:36.08 | vader-- | :) |
| 20:36.17 | file | Strommy Boy! |
| 20:36.58 | [TK]D-Fender | BBIAB |
| 20:37.24 | mountainm2k | <PROTECTED> |
| 20:37.34 | mountainm2k | Plugged my test-set between FXO and the CO |
| 20:37.39 | mountainm2k | verified I can dial |
| 20:38.05 | mountainm2k | what it sound slike is the Zap isn't waiting for dial-tone before actually dialing |
| 20:38.15 | mountainm2k | so it goes offhook and dials immediatly |
| 20:38.23 | mountainm2k | and the CO is sometimes missing the first digit |
| 20:38.45 | mountainm2k | also we have 10-digit dialing here, and it's dialing 9 digits, then a short delay (500ms maybe) and then the last one |
| 20:39.04 | Qwell[] | mountainm2k: Dial(Zap/g1/w${EXTEN}) |
| 20:39.23 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 20:39.29 | mountainm2k | the "w" says wait for dialtone, Im guessing? |
| 20:39.33 | justinu|laptop | mountainm2k: yeah, w will add a short pause to your dialstring |
| 20:39.34 | justinu|laptop | 500ms, iirc |
| 20:39.48 | *** join/#asterisk mmealling (n=michael@c-24-98-171-50.hsd1.ga.comcast.net) |
| 20:39.50 | alunt2003 | file: That works perfect for my mobile now,but times out for normal phones so i guess i have to add more rules or something. error: Timeout, but no rule 't' in context 'inbound' |
| 20:39.51 | justinu|laptop | your CO must have a damn slow digit receiver |
| 20:39.51 | Qwell[] | justinu|laptop: I want to say 250ms...but...I may be wrong |
| 20:40.12 | justinu|laptop | 250 sounds more correct |
| 20:40.15 | mountainm2k | it's weird, some of our lines in this area are like that and some aren't |
| 20:40.18 | file | alunt2003: essentially what it does is for priority 1, if the callerid matches your mobile it executes that instead, while other calls go to the regular priority 1 |
| 20:40.31 | file | alunt2003: so based on what I said... you should be able to put two and two together and figure something out |
| 20:40.32 | file | hopefully |
| 20:41.45 | justinu|laptop | mountainm2k: probably depends on what type of switch the line originates at |
| 20:42.19 | justinu|laptop | just add more "w" until it works reliably |
| 20:42.23 | syzygybsd | hmmm, I see a patch for fixing my problem in 2.16 but I am running 2.15 upgrade the kernel or patch the source... |
| 20:42.23 | justinu|laptop | if indeed that was the problem |
| 20:42.24 | alunt2003 | file: Yeah,I'll read my "Asterisk,The Future Of Telephony" book. It's very good at explaining dial plans,but i just couldn't work out the caller-id setting. Thanks again |
| 20:44.32 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-42.dynamic.qsc.de) |
| 20:44.44 | Katty | :> |
| 20:45.56 | Katty | :< |
| 20:47.23 | nazgool | tickle fetish? |
| 20:48.40 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 20:49.55 | *** join/#asterisk gigahz (n=Arno@ti131310a080-7446.bb.online.no) |
| 20:50.16 | mountainm2k | OK, that seems to have fixed my problem -- just one W |
| 20:50.31 | mountainm2k | it still dials 9 digits, then a short delay, then the 10th |
| 20:50.37 | mountainm2k | but that doesn't matter too much |
| 20:52.04 | justinu|laptop | cool |
| 20:52.14 | syzygybsd | there a good link for how to create a patch using svn |
| 20:52.19 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:52.28 | syzygybsd | I did one using cvs but now I need to do anohter |
| 20:58.32 | *** join/#asterisk kitche (n=dragon@pool-70-16-34-92.buff.east.verizon.net) |
| 21:01.59 | mountainm2k | OK, now, here's another question -- how can I simply record ALL incoming and/or outgoing calls? Just drop a call to monitor() in front of dial() ? |
| 21:02.41 | justinu|laptop | yeah, but use mixmonitor |
| 21:03.38 | mountainm2k | what diff? *7 works now (*R, hah), it recorded two files, then mixed them... |
| 21:04.00 | justinu|laptop | mixmonitor will take the two records and mix them together |
| 21:04.06 | justinu|laptop | monitor requires you to do it yourself |
| 21:04.27 | mountainm2k | Hmm, the automon function must use mixmonitor then -- it mixed them for me, I didn't do it... |
| 21:04.38 | justinu|laptop | yeah, probably does |
| 21:04.44 | justinu|laptop | monitor is old and silly :) |
| 21:04.57 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 21:04.57 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
| 21:04.59 | okdo | hi |
| 21:05.14 | okdo | is it possible to use sox within the asterisk dialplan so I can automatically raise the volume on the file? |
| 21:05.27 | Qwell[] | I bet mixmon could go away, by just adding an option to monitor |
| 21:06.07 | *** join/#asterisk planet_guru (n=chris@brezhnev.spiration.co.uk) |
| 21:06.37 | *** part/#asterisk mmealling (n=michael@c-24-98-171-50.hsd1.ga.comcast.net) |
| 21:08.00 | mishehu | I have libpri 1.2.3 installed, zaptel 1.2.6 installed, and then I built and installed asterisk 1.2.9.1. why might it be that chan_zap.so is reporting that 'pri_cpe' is an unknown signalling type? |
| 21:08.56 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 21:09.03 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 21:09.55 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 21:10.53 | tgrman | anyone else using wav49 as their voicemail storage format with trunk? |
| 21:10.59 | justinu|laptop | mishehu: switchtype set? |
| 21:11.41 | *** join/#asterisk techie (n=gus@voipops.net) |
| 21:12.16 | fnordian | tgrman: i remember some problems with wav49, but that was in 1.2.7 |
| 21:12.22 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 21:12.22 | dlynes_office | there we go |
| 21:12.30 | dlynes_office | getting asterisk set up on a P75 :) |
| 21:12.57 | mishehu | justinu|laptop: it's set for pri_cpe |
| 21:13.06 | mishehu | err |
| 21:13.07 | mishehu | national I mean |
| 21:13.08 | tgrman | I think that there may be something screwy with the gsm conversion being done with the wav49 format as the file generated will not play in windows media player. things work fine with 1.2 |
| 21:14.13 | smackus | is it possible to SetVar(CALLFILENAME=${the unique cdr id for this call here}? or is the cdr unique id created after the call is completed? |
| 21:15.07 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-171-241.dhcp.southerncoastalcable.net) |
| 21:15.27 | QbY_ | Is Find Me Follow Me not installed by default with Asterisk? |
| 21:15.35 | tgrman | the wav49 file will play fine through app_voicemail and using the play command in Linux, although sox does complain about an invalid gsm frame size |
| 21:18.30 | tgrman | anyone willing to help confirm this? don't want to submit a bug if it's not really a bug, although looks like a bug at present |
| 21:19.40 | *** join/#asterisk iq (n=iq@71-215-58-212.omah.qwest.net) |
| 21:22.09 | smackus | who is a good sip provider, low rate high quality. |
| 21:22.41 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 21:24.34 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:27.13 | *** join/#asterisk outsrc (n=fais@host202-147-186-58.khi.dancom.net.pk) |
| 21:27.25 | outsrc | hi room |
| 21:28.16 | Strom_C | h hi |
| 21:28.36 | outsrc | i am having a trouble in inbound call that when i receve call it says unable to authanticate unknown |
| 21:28.59 | outsrc | what could be the possible reason |
| 21:29.05 | Strom_C | what kind of channel? |
| 21:29.09 | justinu|laptop | sounds like iax? |
| 21:29.33 | outsrc | now its g729 |
| 21:29.42 | Strom_C | no, thats the codec |
| 21:29.44 | justinu|laptop | g729 is a codec |
| 21:29.49 | Strom_C | what protocol are you using? |
| 21:29.51 | justinu|laptop | SIP or IAX? |
| 21:29.54 | outsrc | sip |
| 21:30.00 | Strom_C | justinu|laptop: ha! |
| 21:30.03 | justinu|laptop | do you have the proper friend/user entry in sip.conf? |
| 21:30.12 | outsrc | yep i do |
| 21:30.56 | outsrc | i had same conf working on my other mechine which carsh so i turn my backup mechine up but i having this trouble |
| 21:31.04 | shmaltz | how can I change the PCI latency for a device? |
| 21:31.55 | outsrc | and when i dial my # it singel buzy |
| 21:32.47 | justinu|laptop | outsrc: turn on sip debug, make your inbound call, paste the output to pastebin.ca |
| 21:33.09 | justinu|laptop | outsrc: and pastebin your sip.conf friend/user entry, with password obscured |
| 21:34.05 | outsrc | and then? |
| 21:34.16 | justinu|laptop | someone looks and finds the problem, hopefully |
| 21:34.34 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 21:34.52 | outsrc | i am new here where this pastbin.ca ? |
| 21:34.58 | justinu|laptop | ~pb |
| 21:35.00 | jbot | [pb] 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/ |
| 21:35.14 | TripleFFFF | cdr_msql has the userfield=1 and loguniqueid=1 |
| 21:35.27 | TripleFFFF | however they both empty |
| 21:36.37 | twilson | TripleFFFF: Right, then on all incoming calls, use SetCDRUserfield(${CALLERID(DNID)} and you will be good. |
| 21:37.11 | twilson | TripleFFFF: if you have an inbound context where all calls come in, you can just set it once there. |
| 21:38.47 | TripleFFFF | yeah trying |
| 21:39.34 | TripleFFFF | <PROTECTED> |
| 21:39.36 | TripleFFFF | hmm |
| 21:39.42 | TripleFFFF | BAD ( |
| 21:39.43 | TripleFFFF | ;) |
| 21:40.04 | TripleFFFF | ok fixed.. trying |
| 21:40.12 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:40.58 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:41.48 | *** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 21:42.21 | TripleFFFF | ok its there |
| 21:42.39 | Dr-Linux | anybody know Multitech VOIP gateway? |
| 21:43.13 | TripleFFFF | ok then i mod my billing system to user userfield right ? |
| 21:43.19 | TripleFFFF | what happened in last revisions ? |
| 21:43.37 | twilson | TripleFFFF: trying to bill off of lastapp, etc. is a pain. never understood why the actual dialed number was never in the cdr. :-) |
| 21:44.04 | twilson | TripleFFFF: should work. last revisions? |
| 21:44.06 | Dr-Linux | Lastdata |
| 21:44.07 | Corydon-w | amdtech: on 7357, you tested the patch? |
| 21:44.07 | TripleFFFF | well.. not only that.. but i get weird shit.. like.. S, !< i mean i get multiple CDR's for same channel |
| 21:44.13 | amdtech | yes |
| 21:44.16 | TripleFFFF | each time it goes into queuen.. or jumps somewhere |
| 21:44.28 | TripleFFFF | same channel id |
| 21:44.52 | amdtech | i think the !ast_fileexists is looking at the text file which does exist |
| 21:45.01 | amdtech | so it's not even dropping into there |
| 21:45.17 | Corydon-w | amdtech: no, ast_fileexists specifically does NOT look at the txt file |
| 21:47.18 | TripleFFFF | thanks mate |
| 21:48.15 | Corydon-w | ast_fileexists only looks at registered file types |
| 21:49.39 | TripleFFFF | php question |
| 21:50.22 | rene- | if php is the question the answer is no |
| 21:50.39 | TripleFFFF | lol |
| 21:50.46 | TripleFFFF | nam i need to add : $blah="\${CALLERID(DNID)}"; |
| 21:50.54 | TripleFFFF | do i need ot escape the { too ? |
| 21:51.19 | amdtech | hhhmmm |
| 21:54.35 | justinu|laptop | i can't believe IBM hadn't figured out how to make a bootable cd by the time they releassed OS/2 warp 4 |
| 21:55.19 | justinu|laptop | i guess i shouldn't be surprised... the factory install routines for the thinkpad still uses .BAT files |
| 21:55.21 | justinu|laptop | lots of them |
| 21:57.10 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 21:57.51 | gigahz | hi all. is this channel good for asking on trixbox? |
| 21:58.04 | dlynes_office | gigahz: type /topic |
| 21:58.30 | TripleFFFF | hey |
| 21:58.38 | TripleFFFF | twilson ? |
| 21:59.07 | twilson | TripleFFFF: yeah? |
| 21:59.16 | TripleFFFF | well |
| 21:59.25 | TripleFFFF | if i use that i need to mod my whole set of apps |
| 21:59.29 | TripleFFFF | so i wanna be sure |
| 21:59.37 | TripleFFFF | $this->aNumber |
| 21:59.38 | TripleFFFF | oups |
| 21:59.51 | TripleFFFF | ${CALLERID(DNID)} |
| 21:59.55 | TripleFFFF | is in and out right ? |
| 22:00.07 | TripleFFFF | if i dual out 1231231234 that wat it will contain ? |
| 22:00.59 | *** join/#asterisk rikstah (n=rick@87.113.88.49.bbplus.pte-ag2.dyn.plus.net) |
| 22:02.28 | *** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 22:02.40 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:03.32 | twilson | to the best of my knowledge, barring some bug I'm not aware of. It will be on all incoming calls to asterisk, so since your phones and your peers will all dial into asterisk, you should be fine. |
| 22:04.15 | TripleFFFF | k |
| 22:04.18 | TripleFFFF | modding CDRtool |
| 22:04.19 | TripleFFFF | that pos |
| 22:04.32 | SwK[Work] | anyone had to deal with a nat'd sip-ua behind a microsoft internet connection sharing thing on SBS? |
| 22:04.37 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 22:05.49 | twilson | TripleFFFF: Of course, the best thing to do is run through every type of call that you allow your users to make and examine the CDRs that are produced. Billing w/ Asterisk is always a little tricky. Especially if you through in transfers, etc. |
| 22:06.18 | twilson | TripleFFFF: wow, I meant throw... not through. |
| 22:06.24 | *** join/#asterisk mtaht4 (n=m@adsl-75-10-213-145.dsl.pltn13.sbcglobal.net) |
| 22:07.31 | *** join/#asterisk JoseBravo (n=jdbravo@200.69.108.180) |
| 22:08.10 | TripleFFFF | kik\ |
| 22:08.15 | TripleFFFF | i mean lol |
| 22:08.30 | JoseBravo | How can I configure my asterisk, for aoutgoing calls. I have bought SIP channel with g729? |
| 22:10.31 | AlexCTI | Someone knows how can I increase the priority of a call to make it keeps his position on the queue? |
| 22:10.38 | *** part/#asterisk kitche (n=dragon@pool-70-16-34-92.buff.east.verizon.net) |
| 22:11.43 | dlynes_office | AlexCTI: you mean you want to be able to hang up, call back, and maintain your original position in the queue? |
| 22:13.08 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:13.10 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 22:13.30 | AlexCTI | dlynes, I'll make better my question, How can i make the a call comes into a queue with hight priotity than others that actually be there before? |
| 22:14.10 | JoseBravo | dlynes_home, may be can you help me with my question? |
| 22:14.15 | dlynes_office | AlexCTI: you'd have to write your own code for that |
| 22:14.31 | dlynes_office | AlexCTI: someone's already written code to be able to hang up, call back, and maintain original position in queue |
| 22:14.39 | JoseBravo | please |
| 22:14.45 | mfedyk | crap, I forgot the local milliwat number in the PSTN |
| 22:14.47 | dlynes_office | AlexCTI: you might be able to look at that code to consider how to write your code |
| 22:15.07 | dlynes_office | JoseBravo: take a look at the sample extensions.conf file that comes with asterisk |
| 22:15.13 | dlynes_office | JoseBravo: also, check the wiki |
| 22:15.15 | dlynes_office | ~wiki |
| 22:15.16 | dlynes_office | ~docs |
| 22:15.20 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 22:15.20 | dlynes_office | ~wikis |
| 22:15.22 | jbot | [wikis] http://www.voip-info.org |
| 22:15.37 | dlynes_office | holy crap is google spreadsheets ever damned buggy |
| 22:17.01 | AlexCTI | dlynes, actually i'm trying to play with ${PRIORITY} but i'm not sure how tell to that variable attach with the call that comes high |
| 22:17.24 | tgrman | anyone notice that the latest trunk produces wav49 files that WMP can't play when the ones 1.2 generated worked fine? |
| 22:17.30 | dlynes_office | same here, but then again, i don't use queues or agents, either |
| 22:19.29 | shmaltz | how can I change the pci Latency for a specific device? |
| 22:21.33 | *** join/#asterisk loonacy (n=loonacy@24-117-254-250.cpe.cableone.net) |
| 22:21.55 | *** join/#asterisk Eggplant (n=none@dsl-216-155-214-007.cascadeaccess.com) |
| 22:22.11 | loonacy | Anyone know where i can find a sample dialtone plan for North America? |
| 22:22.23 | shmaltz | loonacy, to do what? |
| 22:22.36 | shmaltz | loonacy, I think you mean dialplan and not dialtone plan |
| 22:22.44 | loonacy | I have an ATA that has chinese dialtones on it. |
| 22:23.27 | shmaltz | loonacy, that should be a setting in the ATA, it has nothing to do with asterisk |
| 22:25.20 | loonacy | I know it's on my ATA, i was just hoping someone could point me to a standard NA dialtone plan... I've been googling for an hour. |
| 22:25.50 | shmaltz | loonacy, I think that /etc/asterisk/indications might help you |
| 22:25.53 | Strom_C | 350+440 |
| 22:25.57 | Strom_C | hz |
| 22:26.13 | Strom_C | is dial tone in north america |
| 22:27.06 | Strom_C | Dial Tone is 350 Hz and 440 Hz held steady at -13 dBm0/frequency. |
| 22:27.13 | Strom_C | Audible Ring Tone is 440 Hz and 480 Hz for 2 seconds on and 4 seconds off at -13 dBm0/frequency. |
| 22:27.48 | Strom_C | Low Tone is 480 Hz and 620 Hz at -24 dBm0/frequency. |
| 22:27.57 | Strom_C | Line Busy Tone is Low Tone on and off every .5 seconds. |
| 22:28.40 | Strom_C | Reorder Tone is Low Tone on and off every .25 seconds |
| 22:29.04 | Strom_C | loonacy: does that answer your question, or do i need to throw the entire Bellcore book at you? :) |
| 22:31.49 | Dr-Linux | anybody know Multitech VOIP gateway? |
| 22:32.39 | gmfm | if I start monitoring a channel in the dialplan when the call is set up, is it possible to use features.conf to provide a dynamic way to stop monitoring? |
| 22:33.14 | gmfm | I tried this (stopmon => *7,callee,StopMonitor) but it doesn't stop it |
| 22:36.51 | Bullseye_Network | I've been having problems with VoiceMail All day. I had it working fine and then after a while it stops working again. |
| 22:37.01 | Bullseye_Network | my voicemail.conf --- http://pastebin.com/711774 |
| 22:37.19 | Bullseye_Network | it says this mailbox cannot accept messages |
| 22:37.30 | Bullseye_Network | and there are no messages in the box. |
| 22:37.57 | Bullseye_Network | I deleted all the boxes and let the system recreate them and it might work for a while and then back to the ssame problem |
| 22:38.48 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 22:39.57 | Bullseye_Network | Sometimes if I do a reload it will start working again |
| 22:42.06 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 22:42.21 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 22:43.18 | hads | Anyone understand fxotune more than me and is able to tell me why I always get 1=5,0,0,0,0,0,0,0,0 |
| 22:43.48 | hads | fxotune from trunk BTW |
| 22:44.40 | *** join/#asterisk bmac2 (n=bmac2@c-67-186-254-63.hsd1.co.comcast.net) |
| 22:45.12 | bmac2 | I am setting up Asterisk and need to know how and where to get a phone number assigned to my server so I can test it. I have vonage and am looking ot replace it iwth asterisk |
| 22:45.18 | bmac2 | where can I get a phone number? |
| 22:45.46 | loonacy | Strom_C: Thanks, everything sounds a lot better now. |
| 22:46.05 | *** join/#asterisk SRCR (n=Peter@a80-127-87-78.adsl.xs4all.nl) [NETSPLIT VICTIM] |
| 22:46.05 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 22:46.05 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM] |
| 22:48.11 | *** join/#asterisk SRCR (n=Peter@a80-127-87-78.adsl.xs4all.nl) |
| 22:48.36 | TripleFFFF | worked |
| 22:48.39 | TripleFFFF | oh im good |
| 22:48.40 | TripleFFFF | brb |
| 22:50.05 | twilson | TripleFFFF: I suppose that means I'm good too? |
| 22:51.13 | JoseBravo | I bought a SIP account for international callings. Now I want to configure my asterisk for use this account like channel. How can I do that? |
| 22:51.45 | mountainm2k | what your dialplan look like? |
| 22:51.48 | mountainm2k | extensions.conf ? |
| 22:52.11 | mountainm2k | I have a seperate context for outbound-long-distance |
| 22:52.22 | mountainm2k | you just need one for outbound-intl, like |
| 22:52.46 | mountainm2k | exten => _9011xxxxxxsomething,1,Dial(SIP/something) |
| 22:53.03 | mountainm2k | then include that context in your internal context |
| 22:54.24 | *** join/#asterisk Sedorox (i=sedorox@smartserv/cna/Sedorox) [NETSPLIT VICTIM] |
| 22:54.27 | bmac2 | mountainm2k, can you direct me to where to get a phone number assigned to my asterisk so I can test it? |
| 22:55.16 | mountainm2k | heh, wish I could... I'd like to get a block of a few numbers so I can test DID, but I havn't been able to as of yet... |
| 22:55.32 | bmac2 | so yours is set up to be outgoing only? |
| 22:55.35 | mountainm2k | I'm using a TDM board with an FXO module, and a spare phone line... |
| 22:55.43 | mountainm2k | I'm not doing any VoIP to the outside world |
| 22:55.47 | bmac2 | oh ok |
| 22:55.58 | mountainm2k | but Dial() is Dial() |
| 22:56.04 | *** join/#asterisk braniff (n=dfjk@unaffiliated/braniff) |
| 22:56.11 | justinu|laptop | how you gonna do DID with an FXO module? |
| 22:56.13 | bmac2 | ok thanks |
| 22:56.15 | mountainm2k | so if you have your dialplan set up for int'l already, just need to tell it to use the SIP... |
| 22:56.24 | mountainm2k | I'm not -- that's why I havn't tested it yet... |
| 22:56.35 | mountainm2k | Eventually I'm going to get a T1 board and a PRI line |
| 22:56.41 | justinu|laptop | werd |
| 22:56.44 | justinu|laptop | analog sucks |
| 22:56.47 | mountainm2k | not keen on getting all my phone lines from VoIP |
| 22:56.55 | justinu|laptop | VoIP is one thing |
| 22:57.02 | justinu|laptop | voice over internet is a whole different world |
| 22:57.10 | mountainm2k | VoIP over the internet is ... Yeah, what yo said |
| 22:59.37 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 22:59.51 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:01.47 | *** join/#asterisk dan__t (n=dant@72.232.74.146) |
| 23:01.52 | dan__t | Hi. |
| 23:02.43 | dan__t | I'm for sure out of my league here, so for sake of sounding like a 'tard, I'll just go ahead with it - we have a rather large Intertel PBX system, which I'm not too fond of. I'm wondering what kind of role Asterisk can play in the administration of that system |
| 23:04.04 | charles___ | dan__t: nothing |
| 23:04.18 | dan__t | Figured as much. |
| 23:04.20 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:04.21 | DrkShdw | asterisk would replace your current pbx.. |
| 23:04.27 | generalhan | whats up all !? |
| 23:04.34 | charles___ | dan__t: Asterisk can be your pbx but not manage your current pb |
| 23:04.36 | charles___ | obx |
| 23:04.38 | charles___ | pbx |
| 23:04.44 | dan__t | Ahh ok, that's what I was looking for. |
| 23:05.07 | generalhan | im looking for a really good wireless conference room phone ... anyone had good experiences with a particular one and Asterisk ? |
| 23:05.10 | braniff | what's a good linux softphone ?? |
| 23:05.16 | dan__t | I'd rather not get all new equipment to interface all the existing phones and stuff with Asterisk. |
| 23:05.16 | Qwell[] | wireless conf phone? |
| 23:05.20 | Qwell[] | can't say I've seen any |
| 23:05.34 | generalhan | youve never even seen any Qwell[] >? |
| 23:05.40 | Qwell[] | nope.. |
| 23:05.47 | generalhan | Polycom makes like 3 i think |
| 23:05.57 | Qwell[] | wifi, or like analog? |
| 23:06.06 | generalhan | analog |
| 23:06.16 | generalhan | with a WIRED base ... but a WIRELESS station |
| 23:06.17 | Qwell[] | oh...well, yeah |
| 23:06.20 | generalhan | lol |
| 23:06.23 | generalhan | thats what im talking about |
| 23:06.25 | charles___ | dan__t: come on, you said big company, you guys have the bucks |
| 23:06.29 | Qwell[] | so just get a polycom :p |
| 23:06.38 | generalhan | wireless just as in my boss doesnt want a trailing wire running across the huge table |
| 23:06.38 | Qwell[] | charles___: You must've never worked for a big company |
| 23:06.50 | DrkShdw | your intertel pbx, is it voip? |
| 23:07.08 | generalhan | Qwell[]: i knew about the Plycoms but i just wanted to know if anyone had used it and liked it ... or if someone knew of one better |
| 23:07.31 | charles___ | Qwell: I've worked for the biggest company group from latin america |
| 23:07.38 | braniff | can someone recommend a good linux softphone ?? |
| 23:07.46 | mountainm2k | gnophone |
| 23:07.50 | mountainm2k | IAX support :-) |
| 23:07.58 | braniff | ok |
| 23:08.06 | braniff | cool |
| 23:09.44 | rene- | i am scripting the manager interfase, i am issuing originate events with async, after origination i want to catch call link events, i think i can have some race conditions here if the number of calls originate is large, as in i can miss, linkages before the origination is over |
| 23:10.07 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 23:10.14 | rene- | so the question is what is best to use threads in process or different processes |
| 23:10.15 | mopri | hi anyway had anyluck setting up caller id in europe o latinamerica? |
| 23:10.15 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 23:10.38 | rene- | mopri: you mean forging callerid |
| 23:11.01 | dan__t | charles___, I said nothing of a big company heh |
| 23:13.00 | *** join/#asterisk klasstek (n=nunyobiz@c-67-177-199-232.hsd1.co.comcast.net) |
| 23:14.30 | *** join/#asterisk knarfly (n=Knarf980@c-69-180-98-189.hsd1.fl.comcast.net) |
| 23:15.09 | dan__t | So I've got a few boxes of crap. This is going to be fun |
| 23:15.35 | knarfly | Can anyone with a FWDNET account give me a test call? |
| 23:16.54 | jsaunders | That'd be cool if you could set the outbound codec w/ the Dial() cmd, or something along those lines, instead of on a peer or general basis. |
| 23:17.05 | jsaunders | Or am I just retarded and missing the big picture here. Hmm. |
| 23:23.45 | *** join/#asterisk saftsack (n=saftsack@p54A7ED94.dip.t-dialin.net) |
| 23:24.35 | *** join/#asterisk iq|mobile (n=iq@71-215-58-212.omah.qwest.net) |
| 23:25.26 | justinu|laptop | jsaunders: no, i think a per call codec feature would be useful for a lot of people |
| 23:25.50 | jsaunders | Amen brotha |
| 23:26.02 | justinu|laptop | of course, you could define two peers, with two different allows, but that wouldn't work for dynamic devices that register |
| 23:26.46 | generalhan | Qwell[]: would you recommend an IP conference room phone or an analog ?? cause i have that room wired with a connectection from my TDM and my TE210 |
| 23:27.18 | justinu|laptop | we use an analog polycom soundpoint connected to a Sipura ATA |
| 23:27.48 | justinu|laptop | a lot cheaper than going out and buying an soundpoint IP |
| 23:28.04 | generalhan | justinu|laptop: well i wouldnt need the Sipura since i have the TDM ports ... but i just dont know if it would be better to have an IP version rather than an analog |
| 23:28.56 | generalhan | justinu|laptop: let me put it this way then i guess .. the IP4000 is only $100 more than the Soundstation2W ... would it be a $100 benefit to use the IP4000 rather than the analog ? |
| 23:29.42 | justinu|laptop | not sure... the analog phone will work in more places, and be easier to setup |
| 23:29.56 | justinu|laptop | depends on how much you wanna eat your on VoIP dogfood :) |
| 23:30.07 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 23:30.07 | justinu|laptop | s/on/own/ |
| 23:30.29 | generalhan | justinu|laptop: yea ?? is that really what you meant to say ? lol |
| 23:30.36 | justinu|laptop | yeah |
| 23:30.52 | justinu|laptop | choice was easier for us, because we had a sounpoint already, and no way were they gonna buy another |
| 23:31.33 | mopri | hi anyway had anyluck setting up caller id in europe o latinamerica?, i can-t get the callerid from the local pstn analog line |
| 23:32.51 | mopri | i think i'll look for callerid translators.. maybe they are |
| 23:34.11 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 23:39.31 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 23:42.17 | *** part/#asterisk dan__t (n=dant@72.232.74.146) |
| 23:45.08 | mountainm2k | OK, another question... I'm trying XLite soft phone -- works, nice... But if I use same extension for hard-IP phone and soft-phone, it seems like the last one to connect is the one that gets incoming calls -- they don't both ring |
| 23:45.28 | mountainm2k | any way to make Dial(SIP/xxx) go to all those signed in under that name? |
| 23:45.39 | DrkShdw | umm.. you want ring groups |
| 23:46.06 | DrkShdw | 1 extension per phone, then configure the ring group accordigly |
| 23:46.31 | justinu|laptop | mountainm2k: asterisk doens't support multiple presence like that... make your phones register as seperate peers and use a Dial(SIP/device1&SIP/device2) |
| 23:46.45 | mountainm2k | Hmmm... Havn't done that yet, as it wasn't covered in the book, heh... More reading for me! |
| 23:47.05 | mountainm2k | Ahhh... OK, that'd work... |
| 23:48.22 | *** part/#asterisk umay (n=chris@71-208-188-148.hlrn.qwest.net) |
| 23:48.46 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:51.18 | knarfly | Can anyone with a FWDNET account help me with a test call? |
| 23:51.33 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 23:53.49 | braniff | i tried to set up asterisk....when i try to run it by typing "asterisk", nothing happens, and the process does not start (pgrep asterisk shows nothing)...how should i start debugging this ? |
| 23:54.29 | generalhan | braniff: type 'safe |
| 23:54.35 | generalhan | 'safe_asterisk' |
| 23:54.41 | braniff | ok |
| 23:54.48 | generalhan | or asterisk -cvvv |
| 23:55.47 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 23:55.51 | braniff | ah that gave me lots of errors to work with |
| 23:55.56 | generalhan | lol |
| 23:56.14 | generalhan | you already compiled all that you needed (ie zaptel and libpri) |
| 23:56.36 | braniff | i loaded the fedora package |
| 23:56.46 | generalhan | then asterisk and thats it ? |
| 23:57.19 | braniff | i meant i loaded the fedora asterisk rpm and all its dependencies... |
| 23:57.33 | generalhan | braniff: what kind of setup are you using on your * server ? like any digium hardware, ect |
| 23:58.15 | braniff | i'm trying to use Ekiga softphone with asterisk on the same laptop |
| 23:58.23 | braniff | very simple setup |
| 23:58.41 | generalhan | using a VoIP provider ? |
| 23:58.45 | braniff | yes |
| 23:59.08 | generalhan | ok i think that you need to modprobe ztdummy if youre not using any hardware |
| 23:59.21 | braniff | ok |
| 23:59.49 | braniff | looks like i don't have that module... |
