| 00:00.12 | FarrisG | ok so maybe this door phone won't QUITE get me what I want... It seems that when you push the button, it "picks up" the line, but also sends some kind of signal (probably just an impedance change, since it's passive). So if I hold down the button, the sip agent that picks up the call gets some loud noise for a bit and then a very quiet, but distorted, two-way signal from the door phone. Any ideas? |
| 00:00.29 | *** join/#asterisk ateoh211 (n=kalupson@c-68-33-231-126.hsd1.md.comcast.net) |
| 00:00.41 | ateoh211 | I installed asterisk(from debian package) |
| 00:00.51 | Strom_C | ateoh211, thats your first mistake |
| 00:00.58 | ateoh211 | hehe |
| 00:01.31 | ateoh211 | I configured 2 lines and we can make iax calls(just playing on a private network) |
| 00:02.09 | ateoh211 | but when I configure a conference room and dial it, I hear a message (that is not a valid conference number) |
| 00:02.22 | Strom_C | did you edit meetme.conf? |
| 00:02.35 | Strom_C | also, do you have a zaptel card? |
| 00:03.00 | ateoh211 | I do not have a zaptel card |
| 00:03.24 | ateoh211 | I did etic meetme.conf |
| 00:04.02 | Strom_C | ateoh211, do you have zaptel and ztdummy loaded? |
| 00:06.13 | ateoh211 | no, I do not... |
| 00:06.29 | hfb | Strom_C, Before I forget,. Thanks for giving the talk on Saturday. |
| 00:06.37 | Strom_C | very welcome |
| 00:06.44 | Strom_C | ateoh211, you need to load those kernel modules |
| 00:07.33 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 00:07.52 | ateoh211 | ok, working on that..do you know the exact names so I can insmod them? |
| 00:08.18 | nickswanjan_ | How do I classify an inbound call to oh323 by ip source so it gets into the right context? Can't find the syntax, and [x.x.x.x] doesn't work... |
| 00:08.23 | jsaunders | Does anyone know the physical dimenions of a Digum tdm2400? |
| 00:08.24 | hfb | Strom_C, I was the guy trying to stress to Brian and Kurt how easy it was to build and use ir transmitters. I actually gave them both one. I just home brian uses it sometime before 2010. |
| 00:08.26 | jsaunders | Digium even |
| 00:08.37 | ateoh211 | think I have to build the zaptel stuff, I'll work on |
| 00:08.38 | hfb | s/home/hope/ |
| 00:08.54 | hfb | Hmm. |
| 00:09.08 | Strom_C | jsaunders, it's a full-length card |
| 00:09.25 | Qwell[] | full-length and full-height |
| 00:09.30 | Qwell[] | it's a big beastie |
| 00:09.36 | hfb | That was neat. never sean that before. |
| 00:09.50 | jsaunders | Excuse my ignorance... anyone know the actual dimensions? I'm looking but have not found yet. |
| 00:09.59 | hfb | s/sean/seen/ |
| 00:10.19 | x86 | w000 |
| 00:10.35 | x86 | found a bug in asterisk 1.2.4... not that it really matters anymore since 1.2.6 is out :P |
| 00:12.31 | jsaunders | For those who are curious. Full length pci apparently is 5.45"Wx 1.70"Hx 13.43"D. |
| 00:14.16 | Qwell[] | 1.7"? |
| 00:14.37 | Qwell[] | I'm only counting two dimensions that matter |
| 00:14.59 | Qwell[] | Though, I guess there does have to be a limit, for other cards to fit |
| 00:15.10 | Shaun222 | when i say have Background(thanks) where would that audio file be? |
| 00:15.19 | Qwell[] | question is, is it 1.7" both directions, or just top or bottom? |
| 00:16.55 | jsaunders | Understandable, the confusion w/ width and height. I'm sure you can figure it out. ;) |
| 00:17.52 | nickswanjan | Shaun222: /var/lib/asterisk/sounds/ |
| 00:18.00 | *** part/#asterisk phpmattk_ (n=phpmattk@ip-216-7-118-114.fireserve.net) |
| 00:18.34 | Shaun222 | can i use std .wav files or do i need to convert these |
| 00:18.42 | Shaun222 | it looks like theirs nothing but gsm files in their right now. |
| 00:18.56 | Strom_C | Shaun222, standard wav files |
| 00:19.00 | Strom_C | 8khz 16 bit |
| 00:19.17 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:20.21 | nickswanjan | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 00:20.29 | Snake-Eyes | any ideas why my mailbox wont take my password/ogin? I put in my login/username and then it asks for password, so put that in and then it says fail.. |
| 00:22.05 | Strom_C | nickswanjan, screw that |
| 00:22.12 | Strom_C | I use real PCM files in my setup |
| 00:22.14 | Strom_C | sounds better |
| 00:22.25 | Strom_C | none of this GSM garbage |
| 00:22.49 | dlynes | Strom_C, can you actually use wav files for the Background() dialplan function/ |
| 00:22.57 | Qwell[] | dlynes: among many others |
| 00:22.57 | Strom_C | hell yes |
| 00:22.59 | austinnichols102 | strom_c: moved the te110p to another slot and everything came right up |
| 00:23.01 | Strom_C | thats what I just said |
| 00:23.25 | dlynes | qwell: oh...thought it was only limited to gsm for that function, but wav files, mp3's, ... for Playback, et al |
| 00:24.40 | austinnichols102 | strom_c: right. so thanks! |
| 00:25.36 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 00:25.47 | nickswanjan | Strom_c, the link was for reference - it says you can do either |
| 00:26.02 | asterboy | Anyone getting a good price for the Grandstream GXP-2000? |
| 00:26.23 | asterboy | Have it widdled down to $84 USD |
| 00:27.13 | asterboy | going once... |
| 00:28.26 | asterboy | going twice... |
| 00:28.33 | Strom_C | ...dogballs! |
| 00:28.46 | asterboy | SOLD! |
| 00:28.56 | Shaun222 | i have a problem right now where it cuts off the first quick bit of my intro, is their a way to correct this with out adding 1 second of air? |
| 00:29.10 | asterboy | to the guys with dogsballs for a paltry $84. |
| 00:29.28 | Strom_C | Shaun222, do an Answer first? |
| 00:30.15 | Shaun222 | i do |
| 00:30.15 | nickswanjan | I've had the best luck with adding 1 second of air to the front of the recording. |
| 00:30.25 | Strom_C | Shaun222, Answer followed by Wait(1) |
| 00:33.15 | Shaun222 | Strom_C: http://pastebin.com/629988 |
| 00:33.49 | Strom_C | Shaun222, what about Playback() instead of Background() |
| 00:33.50 | Shaun222 | now when i call it gives a auto fallthrough message with status unknown |
| 00:33.52 | Shaun222 | and hangs up |
| 00:33.59 | Strom_C | er |
| 00:33.59 | Strom_C | wait |
| 00:34.06 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 00:34.10 | Strom_C | why the hell do you have n instead of priority numbers? |
| 00:34.22 | Shaun222 | whats the diffrance between playback and background |
| 00:34.25 | nick125 | anyone here able to put people on hold in xlite without opening a new call? |
| 00:34.36 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 00:34.38 | Shaun222 | Strom_C: it was from the example they had |
| 00:34.39 | Strom_C | also, your _NXXNXXXXXX extension has two entries with priority 2 |
| 00:34.46 | *** part/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 00:34.47 | Strom_C | Shaun222, you're supposed to replace n with priorities |
| 00:34.54 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 00:35.00 | Shaun222 | whoops |
| 00:35.08 | Shaun222 | that fixed it it looks like |
| 00:35.10 | Strom_C | examples are good, but using your brain helps too |
| 00:35.13 | _-Jon-_ | hey everyone |
| 00:35.38 | Shaun222 | Strom_C: trying to learn as much as i can... the docs for asterisk out their are confusing... their is just too much junk |
| 00:36.02 | _-Jon-_ | having a slight problem I'm hoping someone can help me with.. I have an old HandyTone-286 kicking around that I hooked up but whenever I dial, I get this error" Unable to find a codec translation path from g723 to ulaw" |
| 00:36.09 | Shaun222 | voip's wiki makes me want to commit suicide.. |
| 00:36.23 | _-Jon-_ | Now the thing is, in sip.conf I have diallow=all and allow=ulaw but it still won't work |
| 00:36.35 | Strom_C | _-Jon-_, is your handytone set to ulaw? |
| 00:36.57 | Shaun222 | Strom_C: what did you mean by this? "<Strom_C> why the hell do you have n instead of priority numbers?" |
| 00:37.10 | _-Jon-_ | Strom, it has a bunch of codec settings but none are ulaw or g711 |
| 00:37.19 | Strom_C | _-Jon-_, what are your options |
| 00:37.24 | _-Jon-_ | Is asterisk not able to conver it from whatever to ulaw? |
| 00:37.29 | _-Jon-_ | hold on let me list them |
| 00:37.45 | Strom_C | Shaun222, you know how extension lines work, right? |
| 00:37.52 | Strom_C | exten => extension,priority,action |
| 00:38.01 | _-Jon-_ | Strom_C, pcmu, pcma, g726-32, g729, g723, and ilbc |
| 00:38.11 | Strom_C | _-Jon-_, pcmu |
| 00:38.20 | Shaun222 | Strom_C: right. |
| 00:38.27 | Strom_C | ulaw == PCM with ulaw companding |
| 00:38.39 | _-Jon-_ | Strom_C, strange then, because PCMU is it's number 1 choice |
| 00:38.41 | Strom_C | Shaun222, instead of 2,3,4 you had n |
| 00:39.04 | Shaun222 | ahh, ya got that fm a example |
| 00:39.08 | asterboy | Anyone seen the new Grandstream GXP-3000 video phone? |
| 00:39.16 | Strom_C | like I said, using your brain tends to help |
| 00:39.20 | justinu | ~asterboy |
| 00:39.21 | jbot | hmm... asterboy is a weed smoker |
| 00:39.21 | Shaun222 | Strom_C: so then i should always use a number... |
| 00:39.29 | Strom_C | Shaun222, of course |
| 00:39.37 | asterboy | ~justinu |
| 00:39.38 | jbot | hmm... justinu is some other d00d |
| 00:39.38 | justinu | there's nothing wrong with priority "n" |
| 00:39.39 | Strom_C | Shaun222, look at the example files |
| 00:40.13 | Strom_C | justinu, oh? |
| 00:40.20 | Shaun222 | Strom_C: have been thats why i used n |
| 00:40.22 | justinu | n is "next", so you don't have to count, and then renumber them when you modify the order |
| 00:40.31 | asterboy | that GXV-3000 looks smart. |
| 00:40.37 | Strom_C | justinu, ah |
| 00:40.42 | _-Jon-_ | Strom_C, any ideas on why it might not be working? Or is there some way for me to allow Asterisk to convert from one codec to the other/ |
| 00:40.48 | justinu | you can also assign a label to an n |
| 00:41.03 | justinu | as in exten => 1,n(main-menu),Playback(main-menu) |
| 00:41.09 | orlock | e |
| 00:41.10 | Strom_C | _-Jon-_, is there an option to force it to use pcmu? |
| 00:41.14 | justinu | exten => i,1,Goto(main-menu) |
| 00:41.28 | Strom_C | because it looks like its trying to use g723 |
| 00:42.22 | _-Jon-_ | Hmm I'm not sure. g723 is #5 on the list so I have no idea why it's so stuck on that one. Maybe I'll try resetting it completely and starting over. I know it worked before |
| 00:42.31 | Shaun222 | anybody know if voicepulse has a limit to incomming and outgoing conncurrent calls? |
| 00:42.36 | Strom_C | Shaun222, 4 |
| 00:42.41 | Shaun222 | for both/ |
| 00:42.43 | Strom_C | yes |
| 00:42.44 | Shaun222 | or total in/out |
| 00:42.47 | Strom_C | 4 total |
| 00:43.11 | Shaun222 | gonna need that raised, if i stick with them anyway |
| 00:43.19 | Strom_C | go with someone else then |
| 00:43.28 | Shaun222 | they wont raise it? |
| 00:43.32 | Strom_C | i doubt it |
| 00:43.45 | _-Jon-_ | Thanks Strom |
| 00:43.55 | Shaun222 | i'll pay more... :) |
| 00:43.56 | Strom_C | _-Jon-_, did that work? |
| 00:43.57 | Shaun222 | hah |
| 00:43.59 | Shaun222 | i'll talk to them |
| 00:44.16 | _-Jon-_ | Strom_C, not sure yet, I'm going to try in a minute :P |
| 00:44.20 | Strom_C | oh |
| 00:44.26 | Strom_C | well make sure it works before you thank me |
| 00:44.46 | nick125 | Hey, I'm having a weird issue where a user says they can hear me, but, I can't hear them, but, they say their mic indicator in xlite is flashing when they are talking..any suggestions? |
| 00:45.00 | _-Jon-_ | But you tried, so I got to thank you for that at least |
| 00:45.38 | Strom_C | nick125, SIP or IAX? |
| 00:45.45 | nick125 | SIP |
| 00:45.56 | Strom_C | nick125, is either party behind NAT? |
| 00:46.18 | nick125 | I'm behind a nat, but, I try my "test" extension, which is MoH, and I hear sound fine |
| 00:46.29 | Strom_C | is the other party behind a NAT? |
| 00:46.35 | nickswanjan | Any OH323 users out there? How do I classify an inbound call to oh323 by IP source so it gets into the right context? All my inbound calls are ending up in the default voip-h323 context, regardless of the source IP... |
| 00:46.35 | nick125 | nope.. |
| 00:46.52 | Strom_C | nick125, is your asterisk box behind NAT? |
| 00:47.22 | nick125 | and, when I put the person on hold, they didn't hear hold music either.. |
| 00:47.25 | nick125 | and no, its not |
| 00:47.33 | Strom_C | odd |
| 00:47.35 | Strom_C | which codec? |
| 00:47.38 | nick125 | GDM |
| 00:47.40 | nick125 | *GSM |
| 00:47.42 | Strom_C | ew |
| 00:47.46 | nick125 | on both parties |
| 00:47.52 | Strom_C | does it work if you try a different codec? (ulaw?) |
| 00:47.53 | justinu | if you can't hear them, you're not recieving their RTP stream |
| 00:48.02 | justinu | either they're sending it to the wrong place, or you're blocking it. |
| 00:48.25 | nick125 | if I put the guy on hold, I get a ton of these errors: |
| 00:48.26 | nick125 | Mar 29 17:48:37 NOTICE[10874]: rtp.c:564 ast_rtp_read: Unknown RTP codec 98 received |
| 00:48.56 | *** join/#asterisk testshifter (n=testshif@203.172.17.212) |
| 00:49.13 | testshifter | help! |
| 00:49.29 | testshifter | when i run make i got the following error: |
| 00:49.31 | testshifter | /usr/bin/ld: cannot find -lssl |
| 00:49.31 | testshifter | collect2: ld returned 1 exit status |
| 00:49.31 | testshifter | make: *** [asterisk] Error 1 |
| 00:49.38 | Strom_C | tessier, install the ssl libraries |
| 00:49.59 | testshifter | what packages do i need for ssl libraries? i am a newbie |
| 00:50.12 | Strom_C | testshifter, the download page tells you what you need |
| 00:50.17 | dlynes | testshifter, openssl |
| 00:50.18 | *** join/#asterisk jpablo (n=jpablo@200.94.130.197) |
| 00:50.37 | testshifter | i have openssl running on my FC4 |
| 00:50.37 | jpablo | hey people, what' s the cheapest place to get a rhino channel bank with 16 fxo |
| 00:50.57 | Strom_C | Jim's House of Disreputable Telecom Equipment? :) |
| 00:51.33 | *** join/#asterisk _andre (n=andre@20151201162.user.veloxzone.com.br) |
| 00:51.40 | _andre | good evening |
| 00:51.57 | _andre | anyone here using a database to store queue logs? |
| 00:52.00 | jpablo | Strom_C what's his url ? :P |
| 00:52.06 | Strom_C | jpablo, that was a joke |
| 00:52.33 | testshifter | guyz please help me: |
| 00:52.34 | testshifter | /usr/bin/ld: cannot find -lssl |
| 00:52.35 | testshifter | collect2: ld returned 1 exit status |
| 00:52.35 | testshifter | make: *** [asterisk] Error 1 |
| 00:52.42 | jpablo | Strom_C: i knew :P |
| 00:53.16 | _andre | testshifter: you need to install openssl |
| 00:53.19 | x86 | anyone have outbound E164 lookups working? |
| 00:54.02 | testshifter | i am using FC4, and when i do this: rpm -qa | grep OpenSSL i got this: pyOpenSSL-0.6-1.p24.4 |
| 00:54.02 | x86 | i stole the macro from the wiki, and it seems to work well IF it can find an ENUM record (or multiple records), but if it does not find any, it goes into an infinite loop where asterisk will use about 85% CPU |
| 00:54.25 | x86 | testshifter: that's just the python bindings, you need openssl |
| 00:54.32 | x86 | testshifter: yum install openssl |
| 00:54.52 | testshifter | yum install openssl |
| 00:54.52 | _andre | and probably openssl-devel too |
| 00:55.11 | _-Jon-_ | strom_c, okay so it works, but not really |
| 00:55.16 | x86 | oh yeah that's a good one too ;) |
| 00:55.49 | jsaunders | Yeah, that package seperation between base & devel made me scratch my head once, one day. Heh. Got er all figer'd now. |
| 00:56.14 | _-Jon-_ | strom_c, it actually connects now, but the audio is VERY choppy and then after 45-60 seconds the call just drops |
| 00:56.19 | jsaunders | "What'dya mean it's not installed... I just installed it! Oh, I forgot the -devel aswell. Ugh" |
| 00:56.50 | Strom_C | _-Jon-_, what kind of bandwidth do you have between you and the asterisk box? |
| 00:57.02 | justinu | -devel is associated header files, that's it |
| 00:57.19 | jsaunders | Required for compiling no doubt. Good to know, thanks justinu. |
| 00:58.48 | x86 | i've been using Gentoo to compile asterisk for me, but the latest version available is 1.2.4 and I'd like to upgrade to 1.2.6. Is there a tutorial on how to compile Asterisk, complete with asterisk-addons, res_config_mysql, and cdr_addon_mysql? |
| 00:59.15 | Shaun222 | testshifter: you only need the -devel package.. |
| 01:00.49 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:01.18 | Shaun222 | how can i make asterisk so that when a user presses 9 it gives them a outside line to dial out of ... like on a normal pbx. |
| 01:01.41 | Strom_C | Shaun222, in your outbound dial plan |
| 01:01.50 | Strom_C | exten => _91NXXNXXXXXX,1,Whatever |
| 01:01.54 | nick125 | okay, now I got it working, but, now, it echos horribly |
| 01:02.15 | Strom_C | nick125, is one or both parties using speakers and a mic? |
| 01:02.20 | x86 | exten => _9X.,1,Dial(SIP/user:pass@provider/${EXTEN:1}|20|t) |
| 01:02.27 | testshifter | guyz what is the right package for openssl? |
| 01:02.32 | nick125 | Strom_C: nope :P, when we used skype, it worked without echo.. |
| 01:02.48 | Shaun222 | testshifter: yum install openssl-devel |
| 01:02.51 | Shaun222 | should be all you need |
| 01:03.14 | Ariel_ | hello folks hope everyone one is doing great today. |
| 01:03.31 | testshifter | the box doesnt have internet connection |
| 01:05.17 | Shaun222 | i think my extentions are wrong or somthing... |
| 01:05.26 | Strom_C | Shaun222, pastebin please |
| 01:05.53 | Strom_C | could it be that you're misspelling "extension"? we may never know |
| 01:05.53 | Shaun222 | right now if a user dials a number xxx-xxx-xxxx it throughs them into the main system... i understand why it does this because it's matching.. |
| 01:05.57 | Shaun222 | just not sure what to do about it. |
| 01:06.00 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 01:07.01 | Shaun222 | http://pastebin.com/630032 |
| 01:07.30 | Strom_C | Shaun222, what context do your phones live in? |
| 01:07.33 | mattwj2005 | so when is #asterisk going to have it own official conference room? :P |
| 01:07.41 | Shaun222 | Strom_C: default |
| 01:07.44 | Strom_C | mattwj2005, heh, that might be fun to set up |
| 01:07.50 | Strom_C | Shaun222, ok, so in default: |
| 01:07.58 | mattwj2005 | it would be awesome :D |
| 01:08.06 | nick125 | I've been playing around with meetme..hehe |
| 01:08.22 | Strom_C | exten => _91NXXNXXXXXX,1,Goto(outgoing,${EXTEN:1},1) |
| 01:08.33 | mattwj2005 | yeah I have been trying to as well |
| 01:08.37 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:08.49 | mattwj2005 | I think something didn't get compiled |
| 01:08.53 | Strom_C | meetme is screwed on my colo'd server...otherwise I'd set one up right now |
| 01:09.02 | Strom_C | there's a timing problem and the conference audio drifts |
| 01:09.35 | Shaun222 | Strom_C: would it be a better idea to move incomming calls out of default and into my incomming or somthing |
| 01:09.41 | Strom_C | Shaun222, always |
| 01:09.42 | Shaun222 | and leave default for the phones |
| 01:09.47 | Shaun222 | ok |
| 01:09.52 | Strom_C | always always have different contexts for inbound and outbound |
| 01:09.57 | Shaun222 | ok |
| 01:10.05 | Shaun222 | that makes sense. |
| 01:12.26 | Shaun222 | Strom_C: what does ${EXTEN:1} mean? |
| 01:12.33 | mattwj2005 | so any of you guys play around with the video functions? |
| 01:12.51 | Strom_C | it means "the dialed extension less the first digit" |
| 01:12.55 | Shaun222 | does the :1 chop off the beggining number |
| 01:12.58 | Shaun222 | oh ok |
| 01:13.03 | Strom_C | so if they dial 913115552368, it will pass 13115552368 |
| 01:13.07 | Shaun222 | right |
| 01:14.30 | Shaun222 | Strom_C: this doesnt exactly do what i wanted though, on most systems (at least that i've used) you press 9, the system then connects you to a line and gives a dialtone... |
| 01:14.32 | Shaun222 | is that possible to do |
| 01:15.17 | Strom_C | Shaun222, ZAP channels? |
| 01:15.37 | Shaun222 | dunno, i'm new to this phone system stuff |
| 01:15.47 | Strom_C | Shaun222, what kind of phone are you using |
| 01:16.03 | Shaun222 | right now just a software phone in linux kiax. |
| 01:16.08 | *** join/#asterisk pgw (n=pgw@mail.otani-shokai.com) |
| 01:16.27 | Strom_C | Shaun222, since the dialtone is generated locally by the phone, you cant pass dialtone on |
| 01:16.38 | Shaun222 | i will probably have some cisco phones or somthing soon... gotta look into it |
| 01:16.40 | Strom_C | you could theoretically play a recording of dialtone |
| 01:18.09 | Shaun222 | Strom_C: also, unless i dial the full number the system doesnt understand it... for short numbers should i do exten => _91NXXXXXX,1,Goto(outgoing,${EXTEN:1},1) |
| 01:18.34 | pgw | does anyone know a channel to discuss hylafax problems? |
| 01:18.35 | Strom_C | Shaun222, are you going out over a POTS line or something? |
| 01:18.48 | Shaun222 | no |
| 01:19.03 | Strom_C | well, what "short numbers" will you be dialing? |
| 01:19.03 | Shaun222 | out via voicepulse (iax) |
| 01:19.22 | Shaun222 | local numbers.. |
| 01:19.23 | Strom_C | well, voicepulse wants the full ten digit number |
| 01:19.24 | Shaun222 | so 949. |
| 01:19.28 | Shaun222 | ok |
| 01:19.31 | Shaun222 | then i could do somthing like... |
| 01:19.32 | Strom_C | voicepulse is not AT&T |
| 01:19.42 | Splat | I need to have any 625133XX numbers go through a specific trunk.. but any other NXXXXXXX can go through either of 2 trunks.. should I change the NXXXXXXX to not include the 5133 or can I just ad a specific route that will be checked first and if it matches it goes no further? |
| 01:19.48 | Strom_C | unless you're in that unfortunate part of 949 that's Verizon |
| 01:19.52 | Shaun222 | exten => _9NXXXXXX,1,Goto(outgoing,1949${EXTEN:1},1) maybe? |
| 01:19.58 | x86 | anyone have outbound E164 lookups working? |
| 01:20.02 | x86 | i stole the macro from the wiki, and it seems to work well IF it can find an ENUM record (or multiple records), but if it does not find any, it goes into an infinite loop where asterisk will use about 85% CPU |
| 01:20.39 | Strom_C | Shaun222, personally I say just get used to dialing 1+10 |
| 01:21.08 | Shaun222 | Strom_C: thats ghetto and confusing for hundreds of people... |
| 01:21.31 | Strom_C | Shaun222, jeez, how many people are you setting this box up for? |
| 01:21.31 | Shaun222 | Strom_C: you live in the 949? |
| 01:21.38 | Strom_C | Shaun222, I live in Los Feliz |
| 01:21.50 | Shaun222 | dunno where that is |
| 01:21.57 | Strom_C | just east of Hollywood |
| 01:22.00 | Strom_C | city of Los Angeles |
| 01:22.04 | x86 | Strom_C: you should check out my dialplan and help me debug this ENUM macro ;) |
| 01:22.06 | Shaun222 | ok |
| 01:22.42 | Shaun222 | Strom_C: your not too far, 2 hours or so from me (dana point) |
| 01:24.05 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:27.02 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 01:27.36 | *** join/#asterisk Darwin_35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 01:27.37 | testshifter | guyz i have already installed the asterisk |
| 01:27.45 | testshifter | how will i test or make it active??? |
| 01:27.55 | testshifter | guyz i have already installed the asterisk, how will i test or make it active??? |
| 01:28.05 | Darwin_35 | stop repeating |
| 01:28.13 | Darwin_35 | ask once and wait for a reply |
| 01:28.23 | testshifter | sorry :) |
| 01:28.31 | ManxPower | ~book |
| 01:28.32 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:28.33 | ManxPower | ~thebook |
| 01:28.35 | jbot | thebook is, like, Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 01:28.37 | Darwin_35 | AND IF YOU READ THE MANUAL IT TELLS YOU HOW TO START ASTERISK |
| 01:28.41 | Darwin_35 | sorry |
| 01:28.45 | ManxPower | testshifter, Read the book |
| 01:28.54 | testshifter | do i need to install ztdummy if i will only use it for VOIP?? |
| 01:29.04 | Strom_C | tessier, and for the love of God don't ask the same question twice in a row |
| 01:29.07 | ManxPower | testshifter, all your answers are in the book |
| 01:29.13 | Darwin_35 | Test I explained this to you yesterday |
| 01:29.33 | Darwin_35 | only if your using meetme |
| 01:29.45 | testshifter | sorry, i am an new bie to linux and asterisk |
| 01:29.56 | Darwin_35 | thats what manuals are for |
| 01:30.10 | Darwin_35 | they teach you what you need to know |
| 01:30.19 | testshifter | i am using the book from oreilly.. |
| 01:30.24 | Darwin_35 | and find the local linux users group and join it |
| 01:30.37 | testshifter | is it a good reference or can you suggest better reference |
| 01:30.40 | Darwin_35 | they will help teach you linux |
| 01:31.04 | Darwin_35 | the asterisk handbook |
| 01:31.07 | x86 | ManxPower: you said you have E164 working for internal stuff right? |
| 01:31.16 | ManxPower | x86, yes |
| 01:31.19 | testshifter | where can i obtain the asterisk handbook |
| 01:31.19 | x86 | ManxPower: can you send me your ENUM lookup macro? |
| 01:31.20 | testshifter | ?? |
| 01:31.23 | ManxPower | in a very sick and twisted way |
| 01:31.54 | x86 | ManxPower: i stole the macro from the wiki, which works quite well IF the number has an E164 DNS entry, but causes asterisk to go into an infinite loop if the lookup fails |
| 01:31.58 | Darwin_35 | www.asterisk.org |
| 01:32.17 | Darwin_35 | voip-info.org/wiki |
| 01:32.45 | Darwin_35 | asteriskgeek.com I believe also |
| 01:33.09 | ManxPower | http://pastebin.ca/47545 |
| 01:33.30 | ManxPower | testshifter, The asterisk handbook is terribly out of date |
| 01:33.44 | testshifter | where can i obtain a good reference? |
| 01:33.55 | Darwin_35 | I just gave you 3 |
| 01:34.06 | [av]bani | voip-info's wiki software is horrible, the versioning sucks |
| 01:34.09 | x86 | ManxPower: thanks man :) |
| 01:34.22 | x86 | ManxPower: of course you dial non-ENUM numbers too right? |
| 01:34.33 | ManxPower | x86, yes. |
| 01:35.01 | x86 | ManxPower: holy crap this is about 30 lines shorter than my macro ;) |
| 01:35.14 | ManxPower | x86, My needs are much more simple. |
| 01:35.15 | x86 | ManxPower: my macro has loops and all kinds of craziness |
| 01:35.39 | ManxPower | that's because if an enum look up fails you can always route to PSTN. |
| 01:35.40 | x86 | right, you only have one server to lookup on, and you only have one outbound trunk to go out of |
| 01:35.43 | ManxPower | you can't do that if dialing extensions |
| 01:35.56 | ManxPower | x86, actually, I have 8 or so servers. |
| 01:36.04 | x86 | 8 DNS servers? |
| 01:36.06 | x86 | ouch ;) |
| 01:36.12 | ManxPower | Nodren, 2 DNS servers |
| 01:36.22 | ManxPower | <PROTECTED> |
| 01:36.28 | x86 | right, but not 2 different zones either |
| 01:36.35 | x86 | same zone |
| 01:36.41 | ManxPower | This was complicated enough |
| 01:36.52 | x86 | ManxPower: want to see mine? |
| 01:36.53 | x86 | heh |
| 01:37.08 | ManxPower | x86, no, |
| 01:37.17 | nick125 | Is it possible to use meetme without a zaptel device (or the ztdummy plugin)? |
| 01:37.23 | Strom_C | nick125, no |
| 01:37.26 | x86 | ManxPower: pfft |
| 01:37.30 | Strom_C | ztdummy, or zaptel card |
| 01:37.48 | Strom_C | nick125, get a Real Server(tm) |
| 01:37.52 | x86 | 309 people in here and no one does E164 for what it was intended to do :P |
| 01:38.00 | nick125 | I do have a real server..I run the VPS (the actual box the VPS is on) |
| 01:38.12 | nick125 | I jsut run it in a VPS for security reasons |
| 01:38.54 | x86 | eh.. VMWare GSX/ESX, anyone? |
| 01:38.55 | ManxPower | oh, VPS, not VPM |
| 01:38.57 | x86 | :P |
| 01:39.05 | nick125 | xen :D |
| 01:39.12 | x86 | TMFACAFO |
| 01:40.56 | ManxPower | we need to start using VMware server edition and reduce some of our NOC clutter. |
| 01:41.23 | FarrisG | does anyone make an exterior door phone that is just a speaker and a button that "picks up" the line? That would be perfect, but all of the door phones I've seen have a button that sends a call signal and then it's up to the switch/router to actually pick up the line |
| 01:41.41 | ManxPower | FarrisG, it's been discussed several times on the asterisk-users mailing list |
| 01:41.52 | FarrisG | ManxPower: Ok, checking now |
| 01:42.16 | *** join/#asterisk AlexCTI (n=alex@adsl-70-244-110-121.dsl.ksc2mo.swbell.net) |
| 01:42.47 | x86 | oooooooo door phone... |
| 01:43.00 | Strom_C | FarrisG, Western Electric 2554 :) |
| 01:43.04 | x86 | i could make a buzzer for the front door ;) |
| 01:43.36 | x86 | Strom_C: nice joke ;) |
| 01:45.49 | *** join/#asterisk future (n=future@virtual.phuture.sk) |
| 01:45.59 | tainted_ | does asterisk support t.38? |
| 01:46.02 | nick125 | is htere something like meetme that doesn't need zaptel? |
| 01:46.05 | nick125 | :/ |
| 01:47.16 | nick125 | kernel modules dont want to compile for me :( |
| 01:47.55 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 01:49.01 | *** join/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 01:49.02 | *** join/#asterisk AsteriskNewbie (n=linux_ba@63.250.111.200) |
| 01:49.29 | AsteriskNewbie | hello ... any DIGIUM staff around here? |
| 01:49.40 | AsteriskNewbie | anyone seen Mogorman? |
| 01:50.26 | AsteriskNewbie | Anyone has experience with the TDM2400 |
| 01:50.28 | AsteriskNewbie | ? |
| 01:52.00 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 01:52.29 | AsteriskNewbie | Any geekheads arounds? |
| 01:52.46 | AsteriskNewbie | Need help getting an TDM2400 up and running ... |
| 01:52.55 | Strom_C | AsteriskNewbie, sigh, I can try |
| 01:53.00 | dokhench | what point you having difficulty at? |
| 01:53.11 | AsteriskNewbie | Hmm .. trying to configure a TDM2400 .. |
| 01:53.15 | AsteriskNewbie | this is what zaptel says .. |
| 01:53.39 | Strom_C | the answer is "dogballs" |
| 01:53.55 | Darwin_35 | storm_c play nice |
| 01:53.56 | mattwj2005 | yeah meet me can be a pain |
| 01:53.56 | dokhench | you sure it isn't 42 Strom_C? |
| 01:54.52 | dokhench | heh =) |
| 01:55.12 | jbroome | on the internet, the answer to most things is 69 |
| 01:55.29 | Strom_C | in soviet russia, 69 answers you |
| 01:55.52 | AsteriskNewbie | Sorry .. when I try to start zaptel .. I get ... |
| 01:55.53 | AsteriskNewbie | Loading zaptel hardware modules:No functioning zap hardware found in /proc/zaptel, loading ztdummy |
| 01:55.53 | AsteriskNewbie | Running ztcfg: ZT_CHANCONFIG failed on channel 21: No such device or address (6) |
| 01:55.55 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 01:56.07 | AsteriskNewbie | that's for the TDM2400 with one FXO card ... |
| 01:56.31 | AsteriskNewbie | zaptel.conf is like: fxsks=21-24 |
| 01:56.31 | AsteriskNewbie | loadzone=us |
| 01:56.32 | AsteriskNewbie | defaultzone=us |
| 01:56.37 | Strom_C | AsteriskNewbie, do you see the card when you do lspci? |
| 01:56.49 | Strom_C | Darwin_35, welcome to North America. Get used to it. |
| 01:57.00 | AsteriskNewbie | Strom_ .. no I don't |
| 01:57.44 | Strom_C | AsteriskNewbie, well there's your problem. |
| 01:57.44 | Strom_C | try a different PCI slot |
| 01:57.44 | AsteriskNewbie | Hmm .. but when I started ... it was recognised ... |
| 01:57.44 | AsteriskNewbie | by kudzu .. |
| 01:57.44 | AsteriskNewbie | And I did see the lights blink on and off ...? |
| 01:58.10 | dokhench | asterisknewbie, you did plug the molex power connector to it, right? |
| 01:58.11 | Darwin_35 | Nanpa should have no control over *xx extensions. makes it a pain to write dialplans |
| 01:58.12 | Strom_C | AsteriskNewbie, take the card out, put the FXO module in position 1, reseat, and try again |
| 01:58.22 | Strom_C | dokhench, thats only if you're using FXS cards |
| 01:58.31 | Darwin_35 | NPA NXX is fine |
| 01:58.48 | AsteriskNewbie | Strom_C: ... hmm ... ok .. except the machine is remoted :( |
| 01:58.56 | Strom_C | AsteriskNewbie, what?? |
| 01:59.07 | Strom_C | draw circle, bang head here |
| 01:59.14 | AsteriskNewbie | I connected via ssh is what I mean. And yeah .. the FXO card is already in position 1 ... |
| 01:59.20 | Strom_C | Darwin_35, NANPA should have control over all aspects of the dial plan, or else it would be anarchy |
| 01:59.30 | nick125 | Anyone know how to not make MoH choppy (using native)? |
| 01:59.43 | Strom_C | AsteriskNewbie, well, then your FXO should be on channels 1-6, not 21-24 |
| 01:59.46 | Strom_C | er |
| 01:59.48 | Strom_C | 1-4 |
| 02:00.08 | ManxPower | Um NANPA has no control over *xx, only specifications, they are not going to put you in jail if you don't use them |
| 02:00.40 | AsteriskNewbie | Hmm .. that's what I thought .... but it didn't work there either. Gave the message: ZT_CHANCONFIG failed on channel 1: No such device or address |
| 02:00.52 | Strom_C | AsteriskNewbie, how far away is the computer? |
| 02:01.07 | AsteriskNewbie | let's say ... 45 minutes driving :p |
| 02:01.18 | Strom_C | and why did you get a TDM2400 for only 4 CO lines? |
| 02:01.52 | AsteriskNewbie | Well ... just for testing .. the idea being to add more modules once we are up .... |
| 02:01.54 | dokhench | the tdm2400 is the only digium card and it only has one fxo module, right? |
| 02:02.04 | Strom_C | no, the TDM2400 is the 24 port analog card |
| 02:02.11 | AsteriskNewbie | That's right .. |
| 02:02.14 | Strom_C | the TDM400 is the 4 port analog card |
| 02:02.22 | Strom_C | then there are a ton of T1 cards |
| 02:02.44 | dokhench | strom_C: was asking newbie what was in his machine. ;) |
| 02:03.11 | Strom_C | dokhench, well then address the question to AsteriskNewbie and/or word it more explicitly |
| 02:03.58 | AsteriskNewbie | TDM2400 can have up to 6 modules ... each module = 4 channels .. and modules can be fxo or fxs |
| 02:04.06 | AsteriskNewbie | Card has an amphenol head ... |
| 02:04.15 | Strom_C | AsteriskNewbie, what kind of machine is it in? |
| 02:04.21 | Strom_C | (what size case) |
| 02:04.28 | *** join/#asterisk file (n=jcolp@mctnnbsa24w-142167058031.pppoe-dynamic.nb.aliant.net) |
| 02:04.28 | AsteriskNewbie | Strom_C: Its an IBM x250 series ... |
| 02:04.34 | dokhench | asterisknewbie: grep -v ^# /etc/zaptel.conf |
| 02:04.37 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-83-26.dsl.chcgil.sbcglobal.net) |
| 02:04.44 | dokhench | asterisknewbie: what comes back? |
| 02:05.17 | Strom_C | AsteriskNewbie, will you please pastebin your entire zaptel.conf? |
| 02:05.19 | *** join/#asterisk Splat (n=Splat@220-253-97-29.TAS.netspace.net.au) |
| 02:05.39 | AsteriskNewbie | Right now .. |
| 02:05.40 | AsteriskNewbie | fxsks=1-4 |
| 02:05.40 | AsteriskNewbie | loadzone=us |
| 02:05.40 | AsteriskNewbie | defaultzone=us |
| 02:06.07 | AsteriskNewbie | earlier on .. I had it on 21-24 channel .. and the error message was different .. |
| 02:06.11 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-23-97.dsl.tulsok.swbell.net) |
| 02:06.17 | Strom_C | ~pastebin |
| 02:06.21 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 02:06.42 | Zodiacal | anyone know if its posible to get hints to display who is calling who? |
| 02:07.07 | AsteriskNewbie | ok |
| 02:07.08 | Zodiacal | cisco 7960 sccp |
| 02:07.26 | dokhench | asterisknewbie: lsmod | grep zaptel |
| 02:07.49 | dokhench | asterisknewbie: do the pastebin as strom_c says |
| 02:08.34 | *** join/#asterisk Jon335 (i=Jon335@ottawa-hs-209-217-119-86.d-ip.magma.ca) |
| 02:08.41 | *** join/#asterisk joonysam (n=a@brom-245-243.flexabit.net) |
| 02:08.47 | joonysam | anyone have problems with 7.4 image on the cisco 79xx phones? |
| 02:08.50 | [av]bani | Zodiacal: nope, not possible |
| 02:08.59 | AsteriskNewbie | Did you see it? |
| 02:09.07 | Zodiacal | avbani darn, i thought i read somewhere that you can do that |
| 02:09.23 | dokhench | asterisknewbie: you have to tell us where you pastebin'd it |
| 02:09.26 | Zodiacal | just to display the caller next to the callee |
| 02:09.42 | [av]bani | Zodiacal: nope. |
| 02:09.45 | AsteriskNewbie | http://pastebin.com/630109 |
| 02:09.51 | Zodiacal | avbani okie,thanks! |
| 02:10.15 | dokhench | asterisknewbie: rmmod ztdummy |
| 02:10.16 | AsteriskNewbie | Funny thing is ... I don't see any entry for the 2400 in /etc/sysconfig/zaptel ... shouldn't I? |
| 02:10.31 | AsteriskNewbie | ok .. done .. |
| 02:10.46 | dokhench | asterisknewbie: modprobe wctdm24xxp |
| 02:10.54 | dokhench | asterisknewbie: ztcfg -vvf |
| 02:12.14 | ManxPower | the README in the zaptel source dir lists what cards go with which drivers. |
| 02:12.19 | AsteriskNewbie | Done: http://pastebin.com/630111 |
| 02:13.10 | dokhench | asterisknewbie: there ya go. you'll need to make sure whatever distro you run loads up the right modules at boot time. |
| 02:13.24 | AsteriskNewbie | hMM .. it doesn't mention the 2400 in the readme but I can't imagine that it isn't supported? |
| 02:13.39 | AsteriskNewbie | dohkhench .. hmm ... lemme see .. gimme a sec .. am running CentOD .. |
| 02:13.50 | dokhench | http://kb.digium.com/entry/13/23/ |
| 02:14.33 | dokhench | welpers, lost is on.. later ppl. =) |
| 02:14.43 | AsteriskNewbie | Hmm .. thanks dokhench ... I'll be reading that .... |
| 02:15.34 | Flauto | i tried to connect the pstn line to my asterisk through sipura spa 3000 today |
| 02:15.37 | Flauto | it did not work |
| 02:15.44 | Flauto | anyone is willing to help |
| 02:15.47 | *** join/#asterisk stormfr (n=StorM@213.161.192.210) |
| 02:22.06 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 02:28.18 | tzanger | hmm. how the hell do I use Dial() to ring a zap channel using a distinctive ring? |
| 02:28.58 | Strom_C | Dial(ZAP1r2) |
| 02:29.01 | Strom_C | er |
| 02:29.04 | Strom_C | Dial(ZAP/1r2) |
| 02:29.13 | tzanger | Strom_C: hmm interesting |
| 02:29.20 | tzanger | Strom_C: danke, you saved me some googling |
| 02:29.38 | Strom_C | it's not Strom...it's not Google....it's STROOGLE! |
| 02:29.39 | joonysam | can anyone tell me why a cisco 7940 with the 7.4 image will not attempt to register to * even though it is set to register and the proxy address is correct? |
| 02:33.55 | *** join/#asterisk denon (i=root@synapse.subneural.net) |
| 02:33.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:36.31 | ManxPower | is it the 7.4 SIP image? |
| 02:38.27 | nick125 | for native MoH, which is better: ogg or mp3? |
| 02:38.41 | Strom_C | mp3 I believe |
| 02:39.04 | *** join/#asterisk jo3sm1th (n=joesmith@203-206-181-202.perm.iinet.net.au) |
| 02:39.11 | nick125 | I'm getting alot of gaps when I'm using native MoH, mp3, low quality.. |
| 02:39.18 | jo3sm1th | how do you alter extensions.conf to not automaticaly hang up the call once it connects |
| 02:39.27 | jo3sm1th | because when I dial a # it goes to it, answers, and in the CLI says "hangingup |
| 02:39.34 | jo3sm1th | and the call gets ended too! |
| 02:43.02 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 02:43.22 | *** join/#asterisk angom_h (n=angom@red-corp-200.79.154.31.telnor.net) |
| 02:44.59 | jo3sm1th | Someone email the solution to jo3sm1th@safe-mail.net and I'l re-imburse you |
| 02:45.10 | Strom_C | ah, such patience |
| 02:45.15 | nick125 | yes |
| 02:48.33 | *** join/#asterisk ken^^ (n=ken@host-66-59-246-138.lcinet.net) |
| 02:48.36 | ken^^ | crazy |
| 02:48.56 | Strom_C | lunatic |
| 02:49.05 | ken^^ | got a quick question, someone here should know -- what's the "best" native format to put MOH files in ? |
| 02:49.24 | Strom_C | didnt you just ask that? |
| 02:49.32 | Strom_C | er, no, someone else did |
| 02:49.35 | ken^^ | Strom_C: what ? |
| 02:49.37 | Strom_C | I like mp3 |
| 02:49.54 | ken^^ | Strom_C: yeah, me too, but i have a bunch of ogg vorbis files that asterisk doesn't support that i need to convert |
| 02:50.02 | ken^^ | Strom_C: so i may as well do it right from the get-go |
| 02:50.07 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 02:51.01 | ManxPower | ken^^, ulaw |
| 02:51.15 | ken^^ | ManxPower: you sure now ? |
| 02:51.18 | *** join/#asterisk powerchip (n=powerchi@197.80-202-229.nextgentel.com) |
| 02:51.24 | ken^^ | ManxPower: aiight, ulaw it be ! |
| 02:51.27 | ManxPower | ken^^, because it's a LOSSLESS format |
| 02:51.38 | ken^^ | ManxPower: righty-o |
| 02:51.41 | ManxPower | WAV, is usually ulaw |
| 02:51.43 | *** join/#asterisk jo3sm1th (n=joesmith@203-206-181-202.perm.iinet.net.au) |
| 02:51.55 | ManxPower | really the best format is whatever codec most of your users use |
| 02:51.59 | Strom_C | no no |
| 02:52.01 | jo3sm1th | I need that command to put in IAX.CONF so it keeps the call connected and won't display the "hanging up" msg in CLI |
| 02:52.04 | Strom_C | wav is linear PCM |
| 02:52.07 | jo3sm1th | Anyone had this problem before please help me |
| 02:52.07 | Strom_C | :) |
| 02:52.10 | Strom_C | ulaw is ulaw |
| 02:52.11 | ManxPower | and remember many MOH files really sound horrible over a compressed codec |
| 02:52.23 | ken^^ | ManxPower: to be honest, i'd rather keep em as ogg vorbis, but asterisk doesn't like non-mono orbis files ... go figure |
| 02:52.30 | Strom_C | jo3sm1th, pastebin your iax.conf and extensions.conf |
| 02:52.42 | ManxPower | ken^^, you can easily hack it in |
| 02:53.00 | jo3sm1th | Hold one second... |
| 02:53.13 | ManxPower | do a custom mp3 class (1.2.x) and rind a player that will decode your ogg files to raw audio on STDOUT |
| 02:53.16 | ken^^ | ManxPower: hmm -- interesting idea, i could, but you see, i'm very, very lazy and a 1 line shell command will convert everything |
| 02:53.38 | ManxPower | ken^^, it's just morally wrong to convert ogg or mp3 into a different format. |
| 02:53.48 | ken^^ | ManxPower: i like that idea with the custom mp3 class -- what command do you think will do that ? |
| 02:54.06 | ken^^ | ManxPower: although, i really don't like the idea of PIPING data into asterisk either ... |
| 02:54.10 | jo3sm1th | http://pastebin.com/630145 |
| 02:54.16 | jo3sm1th | I had altered this setting in iax.conf before |
| 02:54.18 | ManxPower | ken^^, I've never used it. 8-). Do you have a command line ogg player? |
| 02:54.38 | ManxPower | ken^^, it's either going to pipe the raw audio or it's going to read it off disk. |
| 02:54.43 | ken^^ | ManxPower: hmmm ... i'm going to guess yes -- do i know what it is? no |
| 02:54.49 | jo3sm1th | I need that command to put in IAX.CONF so it keeps the call connected and won't display the "hanging up" msg in CLI |
| 02:54.51 | ManxPower | piping might actually be faster since your are less disk I/O bound. |
| 02:55.05 | ken^^ | ManxPower: how do you figure that ? |
| 02:55.11 | ManxPower | jo3sm1th, you are either 1) very very stupid or 2) a troll. |
| 02:55.21 | jo3sm1th | when I call someone it kills the conversation after 10 seconds |
| 02:55.38 | Strom_C | jo3sm1th, will you please shut up while I look at your files? |
| 02:55.39 | ManxPower | ken^^, leaving the files in ogg format would mean there would be MUCH less data read from the disk. remember raw audio data is like 1 meg per second |
| 02:55.51 | Qwell | second? |
| 02:55.53 | ManxPower | well, ok, if it's ulaw it's only 64k per second, but still.... |
| 02:55.54 | Qwell | minute.. |
| 02:56.02 | ken^^ | ManxPower: true, very true ... |
| 02:56.22 | Strom_C | jo3sm1th, where is your extensions.conf? |
| 02:56.22 | jo3sm1th | I just miss the good old days |
| 02:56.29 | ManxPower | Qwell, ulaw/alaw would be closer to 1 meg per min, I think. |
| 02:56.30 | jo3sm1th | when I could pay someone to do this instead o fbeing insulted |
| 02:56.46 | ManxPower | jo3sm1th, Digium will be happy to do it for pay |
| 02:56.48 | Qwell | ManxPower: ~128k mp3s are 1mb/min, if I'm not mistaken |
| 02:57.03 | ManxPower | Qwell, *nod* so raw data would be MUCH more. |
| 02:57.10 | Qwell | 60x though? |
| 02:57.24 | ManxPower | let me see if I have a server that still records VM in .WAV |
| 02:57.28 | ken^^ | ManxPower: i think i'm just going to convert to ulaw, and say fuck it |
| 02:57.34 | jo3sm1th | Here: http://pastebin.com/630150 |
| 02:57.41 | Qwell | ManxPower: 8hkz will be less |
| 02:57.54 | Qwell | I assume? heh |
| 02:58.14 | jo3sm1th | That has nothing to do with it |
| 02:58.25 | jo3sm1th | I had this problem before there is one command like hangup=no to put in iax.conf |
| 02:58.37 | ManxPower | jo3sm1th, there is no such command. |
| 02:58.39 | jo3sm1th | So if Im stupid for remembering this exists than !kickban me now you fat ugly nerds |
| 02:58.42 | ken^^ | anyone hooked up a wintv radio decoder and piped that through to asterisk on MOH ? |
| 02:58.43 | Qwell | jo3sm1th: try workalways=please |
| 02:58.50 | ManxPower | perhaps you mean notransfer=yes, but that has NOTHING TO DO WITH HANGING UP A CALL |
| 02:58.53 | jo3sm1th | Yes |
| 02:58.54 | jo3sm1th | THAT |
| 02:58.57 | jo3sm1th | *HUG* |
| 02:59.07 | Strom_C | what about cocksucking=yes |
| 02:59.07 | ManxPower | so you were using the totally wrong terms and looking like a moron |
| 02:59.15 | jo3sm1th | Yea i know |
| 02:59.20 | ManxPower | perhaps a better description of the issue.... |
| 02:59.24 | Qwell | Strom_C: If there is a strom.conf... |
| 02:59.28 | jo3sm1th | This is like riding a bicycle for the first time in 2 years |
| 02:59.28 | Strom_C | woohoo |
| 02:59.54 | Strom_C | jo3sm1th, riding a bicycle for the first time in 2 years was as if I rode one yesterday |
| 02:59.59 | jo3sm1th | I want to read more and educate myself so next time I wont sound so stupid so is wiki the best place for a glossary of this kind of stuff |
| 03:00.06 | ManxPower | ~thebook |
| 03:00.09 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 03:00.09 | ManxPower | ~docs |
| 03:00.10 | jbot | [docs] 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 |
| 03:00.11 | ManxPower | ~mailinglist |
| 03:00.13 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 03:01.21 | jo3sm1th | hate my questions |
| 03:01.25 | jo3sm1th | hate my analogies I dont care |
| 03:01.26 | jo3sm1th | you |
| 03:01.30 | jo3sm1th | re all probably fat nerds |
| 03:01.44 | jo3sm1th | Expect ManxPower |
| 03:01.47 | jo3sm1th | Long live ManxPower |
| 03:01.58 | ManxPower | jo3sm1th, I'm the meanest of them all |
| 03:02.16 | Strom_C | jo3sm1th, I weigh less than 130 pounds |
| 03:02.19 | ManxPower | if you had asked your question again you would have been put on my /ignore list |
| 03:02.49 | jo3sm1th | BTW I paid nufone $20 and I havent ogt the email yet is Greg here |
| 03:03.55 | *** join/#asterisk rbridal (n=rbridal@12.160.149.6) |
| 03:04.09 | rbridal | hiya |
| 03:04.34 | *** join/#asterisk appdevx (n=appdevx3@203.172.17.212) |
| 03:04.39 | rbridal | I know this has probably already been asked - I'm on FC5, new install, doing make rpm on the asterisk 1.2.6 - it gives an error about astman and asterisk.8.gz .. any ideas on how to fix this? |
| 03:05.02 | jo3sm1th | ManxPower is there anything similar to the notransfer function |
| 03:05.07 | appdevx | guyz help! i have already installed asterisk in my server box.. how will i test it in my client?? |
| 03:05.17 | ManxPower | jo3sm1th, similar in what way. |
| 03:05.37 | appdevx | guyz help! i have already installed asterisk in my server box.. how will i test it in my client?? |
| 03:05.58 | Strom_C | appdevx, please stop asking the same thing repeatedly |
| 03:06.03 | Qwell | appdevx: ask again every minute, you'll get an answer soon! |
| 03:06.09 | Qwell | don't let anybody miss it! |
| 03:06.12 | jo3sm1th | The problem is: every number I dial connects and then the call goes silent hand in hand with the CLI announcing "hanging up" |
| 03:06.33 | Strom_C | we should have the bot kick anyone who says the same thing >1 time in 60 seconds :) |
| 03:06.34 | jo3sm1th | and I had this happen before I thought for sure it was notransfer to be placed in the context of iax.conf but apparently that wasnt the answer |
| 03:06.46 | ken^^ | ah hah |
| 03:06.52 | ken^^ | Converting to sln format |
| 03:06.52 | ken^^ | Starting from Asterisk 1.2.0, the .sln (SLINEAR) format seems to be the preferred format. |
| 03:06.52 | ken^^ | To convert wav file to sln, use the following command: |
| 03:07.00 | ken^^ | ManxPower: so uh ... yeah |
| 03:07.03 | Strom_C | a-ha! |
| 03:07.54 | jo3sm1th | I will Western Union anyone some $ to help me fix that problem (which I wont repeat) |
| 03:08.05 | appdevx | guyz care to help a newbie here! thanks! |
| 03:08.17 | Strom_C | jo3sm1th, how much? |
| 03:08.21 | jo3sm1th | and one more with my DID not working for accepting calls, these are both so minor but I cant micromanage everything if I was in practice it would be done in 10 mins but Im not so insult me if you like or msg me if you love |
| 03:08.53 | Qwell | $175/hour, 2 hour minimum |
| 03:09.06 | jo3sm1th | Any other bidders? |
| 03:09.15 | Strom_C | I've got a better bargain! I bid $173, 2 hour minimum |
| 03:09.29 | Qwell | same price, 1.8 hour minimum! |
| 03:09.35 | Qwell | wait, no |
| 03:09.44 | appdevx | guyz help! i have already installed asterisk in my server box.. how will i test it in my client?? |
| 03:09.46 | *** join/#asterisk rufoz (i=rufoz@200.226.54.243) |
| 03:09.53 | Qwell | ~lart appdevx |
| 03:10.01 | Qwell | damn bot :P |
| 03:10.09 | Strom_C | appdevx, either go away or restate your question more clearly |
| 03:10.48 | appdevx | Strom_c, do i need to configure my sip.conf if i will test it in my client? |
| 03:10.52 | jo3sm1th | Ok $300.00 it is |
| 03:10.57 | appdevx | ~lart |
| 03:11.07 | jo3sm1th | just msg me if you are interested thats 3 times the price I had to pay Greg from nufone but Im desperAte |
| 03:11.11 | Strom_C | jo3sm1th, you fail at multiplication |
| 03:11.15 | Qwell | now, what's the problem? |
| 03:12.44 | jo3sm1th | Mine? |
| 03:12.51 | ManxPower | $300? that's a little over 2 hours at my standard billable rate |
| 03:13.04 | rbridal | anyone received an error on the make rpm command? every time I try, I get an error about astman and the asterisk man page .. rather than pulling the lines out of the spec file, I'd like to fix it .. any ideas? |
| 03:13.24 | ManxPower | But honestly, it's 9pm at night and I'm not doing anything I don't want to do |
| 03:13.38 | ManxPower | rbridal, nobody here uses make rpm |
| 03:13.45 | rbridal | doh |
| 03:13.46 | ManxPower | report it as a bug on bugs.digium.com |
| 03:13.46 | Strom_C | ManxPower, I thought it was 9 PM in the morning :) |
| 03:13.51 | kfuq | http://www.centos.org/modules/news/article.php?storyid=127 <-- rofl |
| 03:13.59 | Qwell | Strom_C: 9PM in the afternoon...get with it |
| 03:14.05 | Strom_C | ok ok |
| 03:14.06 | Qwell | kfuq: welcome to last week |
| 03:15.45 | kfuq | thanks! |
| 03:19.14 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 03:20.58 | justinu | my thinkpad finally shipped... rock |
| 03:22.05 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:24.31 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
| 03:34.22 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 03:35.01 | jo3sm1th | I have one small problem left |
| 03:35.40 | jo3sm1th | According to my iax.conf there is one thing wrong in the context Im pasting on PasteBin which dis allows my calls to get accepted |
| 03:35.59 | jo3sm1th | http://pastebin.com/630179 |
| 03:37.14 | jo3sm1th | Wow troubleshooting finall ypaid off |
| 03:37.45 | jo3sm1th | thank you! |
| 03:38.59 | rbridal | are there any downfalls to installing asterisk @ home as opposed to asterisk? |
| 03:40.08 | *** join/#asterisk Lino` (i=Lino@i577BC54E.versanet.de) |
| 03:40.27 | Strom_C | AMP |
| 03:40.37 | Strom_C | thats the biggest downfall :) |
| 03:41.07 | jo3sm1th | Amp? |
| 03:42.02 | jo3sm1th | Lets say I want to make an Asterisk server at a hotel Im staying at |
| 03:42.11 | jo3sm1th | do you thin kit will work behind their pay daily Inet service |
| 03:42.24 | jo3sm1th | firewalled, or however they have it set up |
| 03:42.52 | nick125 | could you setup VPN? |
| 03:43.19 | nick125 | I tihnk you would need ports forwarded for that.. |
| 03:44.31 | encode | isnt that what a STUN server is for? |
| 03:44.58 | nick125 | I never had to set this up behind a firewall I have no control over.. |
| 03:47.19 | Lino` | forget amp |
| 03:47.42 | justinu | all STUN does is tell you what your routeable IP is, and what type of NAT you're behind |
| 03:47.51 | Strom_C | amp == assrape management package |
| 03:48.05 | orlock | i find Sail looks pretty nice |
| 03:48.09 | rbridal | their pay daily inet service, is it just a captive portal thing? |
| 03:48.09 | Lino` | somethign like that |
| 03:48.15 | orlock | but its made for sme |
| 03:48.25 | Lino` | whoever invented amp must have no common sense whatsoever. |
| 03:49.00 | Lino` | but hey, the good thing is AMP does not exist anymore. |
| 03:49.16 | Lino` | bad news: they just changed names, now its fr***BX |
| 03:50.18 | *** join/#asterisk Sedorox (n=penbra67@smartserv/cna/Sedorox) |
| 03:51.01 | Strom_C | Lino`, name got too tarnished? |
| 03:51.35 | *** join/#asterisk bmg505 (n=leon@dsl-146-30-165.telkomadsl.co.za) |
| 03:51.36 | rbridal | shake your junk |
| 03:51.42 | rbridal | no wait, thats the amp wireless shit |
| 03:53.31 | *** join/#asterisk bla (n=bla@netblock-66-218-41-231.dslextreme.com) |
| 03:57.40 | Lino` | maybe ;) |
| 03:58.07 | Lino` | fact: amp makes customizing asterisk a horrible nightmare |
| 03:58.33 | Lino` | fact: amp users are a) lazy, b) stupid, c) both |
| 04:00.15 | rbridal | ok .. I'm a network admin, not a voip guy, so this is gonna sound odd .. I'm setting up an asterisk box to talk directly to one of our core voip switches, so the switch can send calls to it, and vice versa .. what do I call the link between the core switch and the asterisk box? |
| 04:00.59 | Strom_C | a trunk? |
| 04:01.03 | Strom_C | an ethernet link? |
| 04:01.26 | Sedorox | when you say switch... do you mean a normal ethernet switch.. or a actual voip switch (like cisco call manager) |
| 04:01.28 | rbridal | well, its not an ethernet link, since its gonna go to either our chicago or a vegas core |
| 04:01.39 | Strom_C | rbridal, leased line? |
| 04:01.40 | Lino` | uplink |
| 04:01.41 | Strom_C | T1? |
| 04:01.45 | Strom_C | DS3? |
| 04:01.47 | Lino` | ok then its a trunk |
| 04:01.49 | rbridal | an actual voip switch, like an excel switch |
| 04:02.36 | rbridal | Strom_C .. the media is kinda irrelevant .. its gig from the asterisk to a core ethernet switch here, then fiber to the net, then god knows what, and fiber down to an excel switch |
| 04:02.53 | rbridal | so I call it a trunk between the asterisk server and the core excel switch? |
| 04:03.10 | Sedorox | yea.. because your putting multiple calls over it.. |
| 04:03.21 | rbridal | ok, sweet .. thanks |
| 04:03.55 | rbridal | don't wanna look like a dumbass tomorrow when the voip tech I'm working with gets into the office :) |
| 04:04.03 | Sedorox | hehe |
| 04:04.30 | x86 | gah |
| 04:04.43 | x86 | can someome please help me debug an infinite loop in my dialplan? |
| 04:04.46 | justinu | excel? gah |
| 04:04.52 | justinu | i wrote a ton of code for those |
| 04:04.53 | x86 | https://office.shellshark.net:7960/extensions.conf |
| 04:05.08 | *** part/#asterisk rufoz (i=rufoz@200.226.54.243) |
| 04:05.12 | Sedorox | The requested URL /extensions.conf was not found on this server. |
| 04:05.22 | Lino` | huh?= |
| 04:05.24 | rbridal | we have a ton of versatel switches too, hope you've never had to write code for those |
| 04:05.32 | Lino` | versatel does make switches? |
| 04:05.39 | x86 | Sedorox: doh... hold |
| 04:05.40 | rbridal | yeah, shitty ones that won't fucking turn up |
| 04:05.55 | Sedorox | lol |
| 04:06.02 | x86 | Sedorox: try now :) |
| 04:06.05 | rbridal | we've got 17 of them out there, and maybe 5 are working |
| 04:06.24 | Sedorox | ok.. and where are you having the provblem? |
| 04:06.26 | Lino` | crap |
| 04:06.30 | Lino` | look at my hostmask |
| 04:06.44 | Lino` | i577BC54E.versanet.de |
| 04:06.49 | Lino` | versanet = versatel |
| 04:07.02 | Sedorox | lol |
| 04:07.09 | Lino` | so i am actually having a deal with the devil himself. |
| 04:07.09 | rbridal | so I can blame you? :) |
| 04:07.14 | rbridal | ooh, ok .. lol |
| 04:07.35 | Lino` | i pay 50 euro for 3 BRI lines, one 2mbit line and one 20 mbit line |
| 04:07.41 | Lino` | so its quite a good deal |
| 04:07.45 | Sedorox | damn |
| 04:07.48 | rbridal | uhm, different versatel I think |
| 04:07.55 | Lino` | maybe |
| 04:07.55 | x86 | Sedorox: macro-enum-dial |
| 04:08.01 | rbridal | the versatel that makes core telecom switches is canadian |
| 04:08.04 | Lino` | this one is a dutch isp |
| 04:08.15 | Lino` | dutch like goldmember |
| 04:08.16 | Maxxed | hah |
| 04:08.22 | Maxxed | must be some nice switches |
| 04:08.31 | x86 | Sedorox: did it work? |
| 04:08.32 | Maxxed | rate limit at the gateway |
| 04:08.37 | Maxxed | let the lan be fast |
| 04:08.37 | Sedorox | yea it loaded |
| 04:08.46 | Sedorox | Maxxed: yea.... my net admins are stupid |
| 04:08.52 | Sedorox | they force 100/half on the client |
| 04:08.54 | Maxxed | h4x0r the next |
| 04:08.58 | Maxxed | net* |
| 04:09.04 | Lino` | huh? |
| 04:09.07 | Maxxed | if there stupid im sure it wont be to hard to own |
| 04:09.19 | Lino` | how could one person be possibly so stupid |
| 04:09.21 | Sedorox | and apparently they have rate limiting on the switches... because I can't do more then 1MB/s to my roomate.. on the same switch.. one port over from me |
| 04:09.27 | Maxxed | brute force snmp rw strings |
| 04:09.27 | Maxxed | heh |
| 04:09.29 | Sedorox | lol |
| 04:09.34 | Lino` | why should one need 100/half? |
| 04:09.49 | Maxxed | stupid ass admin ndeed |
| 04:09.55 | x86 | Sedorox: any ideas? |
| 04:09.56 | Lino` | (i believe your network equipment is good enough to handle full duplex) |
| 04:09.56 | Sedorox | Lino`: I was told.. that it is good network practise to force the switches at the speed.. and then half-duplex for clients |
| 04:10.03 | Lino` | lol |
| 04:10.11 | Sedorox | x86: still looking.. but that is wayyyy over my head.. never seen anything so complex before :p |
| 04:10.13 | bsdfreak | heh |
| 04:10.14 | Lino` | i was told that dictatorship is evil. |
| 04:10.16 | Sedorox | Lino`: duh :p |
| 04:10.23 | Sedorox | all cisco 3550+ for local |
| 04:10.26 | Sedorox | 6509's at core |
| 04:10.35 | x86 | err |
| 04:10.46 | x86 | Sedorox: s/local/edge/ ? |
| 04:10.57 | Sedorox | well yea... |
| 04:11.05 | Lino` | forcing 100/half is dictatorship, does that mean your switch is a rogue nation? |
| 04:11.12 | x86 | actually, distribution is probably what you're looking for :) |
| 04:11.13 | Sedorox | ahahahha |
| 04:11.38 | Sedorox | I dunno.. all I know is if I transfer a file.. at 1MB/s (if I'm lucky) to my roommate.. I cannot to anything else |
| 04:11.44 | Lino` | ;) |
| 04:11.53 | x86 | yeah half duplex sucks my balls |
| 04:11.58 | Sedorox | and thats not even close to 100/half's speed... |
| 04:12.03 | Lino` | hm |
| 04:12.05 | Sedorox | should still be better then that |
| 04:12.10 | Sedorox | last year I was able to pull 8MB/s easy |
| 04:12.13 | x86 | oh they're doing rate limiting on a per-port basis |
| 04:12.21 | x86 | weak :( |
| 04:12.27 | Lino` | well thats the effect caused by a fixed speed @ the switch and @ the nic |
| 04:12.42 | x86 | Sedorox: get a switch to hook you and your roomate up together :P |
| 04:12.47 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 04:12.52 | Lino` | like i do 100/full the switch does 100/half and then we beat the crap out of each other |
| 04:12.56 | Sedorox | x86: oh we do :? |
| 04:12.57 | Nugget | or just crimp up a crossover cable. |
| 04:12.57 | Sedorox | :> |
| 04:13.05 | Nugget | or buy macs. :) |
| 04:13.11 | x86 | Nugget: ? |
| 04:13.13 | Lino` | not a good idea |
| 04:13.23 | Sedorox | lol |
| 04:13.25 | Sedorox | well.. |
| 04:13.33 | Lino` | mac users (at least the mac users i know) would install amp |
| 04:13.36 | Sedorox | we have a decent size private network that we use for transfering shit |
| 04:13.51 | Sedorox | cisco routers.. switches (I'm gong for ccna.. he's going for ccnp) |
| 04:13.58 | Lino` | lol |
| 04:14.04 | Sedorox | 7 boxes on it... 4 doing distcc |
| 04:14.16 | Nugget | I'm not happy until my home network is so complicated that it makes my ears bleed. |
| 04:14.32 | rbridal | my entire home network sits inside of a night stand :) |
| 04:14.38 | Lino` | one german company has found the ultimate cash cow. they are providing internet access for university students and charge for bandwith. |
| 04:14.48 | Lino` | 25 cent / GB |
| 04:15.03 | Sedorox | lol |
| 04:15.06 | Lino` | just for routing the traffic from the dorm network into the (free) university network |
| 04:15.08 | Nugget | so you earn money for each pirated song and dvd. :) |
| 04:15.12 | Sedorox | ahahah |
| 04:15.35 | Nugget | http://slacker.com/~nugget/stuff/SlackerNOC.pdf |
| 04:16.04 | Lino` | yeah |
| 04:16.16 | Sedorox | lol |
| 04:16.21 | Sedorox | I need to take pics of our stuff |
| 04:16.30 | Sedorox | I had my home.. but its pitiful... (sp) |
| 04:16.44 | justinu | at least it's home |
| 04:16.47 | Sedorox | I'm on a WS-C3548-XL ot be exact |
| 04:16.55 | Lino` | lol |
| 04:16.55 | Sedorox | which goes back to the 6509's |
| 04:17.03 | Sedorox | (yes.. they leave CDP on) |
| 04:17.08 | Sedorox | also.... they leave EIGRP open... |
| 04:17.20 | Lino` | where did you get that stress machine? - dunno, just met that guy at the elevator and he went mental, so i just stole it. |
| 04:17.35 | Lino` | "you stole it?" - "yes!" - "but that's STEALING!" |
| 04:17.46 | Lino` | lol it crowd is the best ;) |
| 04:17.47 | Sedorox | lol |
| 04:17.50 | Sedorox | I knos! |
| 04:17.57 | Sedorox | I have all them saved :) |
| 04:18.01 | Sedorox | wish they would release more... |
| 04:18.20 | Lino` | let me guess, you've also "found" them (just like me) |
| 04:18.31 | Sedorox | hehe... yea.. hehe |
| 04:18.37 | Sedorox | I tried a UK proxy.. but too slow |
| 04:18.38 | Lino` | in a stream which is another word for torrent ;) |
| 04:18.52 | Sedorox | well.. I pulled them off some other site that had them up |
| 04:18.56 | Lino` | :D |
| 04:18.59 | Sedorox | my roomate I think got it off of torrent |
| 04:19.28 | *** join/#asterisk parf (n=parf@c-24-60-165-111.hsd1.ma.comcast.net) |
| 04:19.59 | Lino` | but thats stealing ;) |
| 04:20.06 | Sedorox | ahah |
| 04:20.07 | Lino` | (as if i would care *gg*) |
| 04:20.13 | Sedorox | damn.. so much shit floating around resnet.... |
| 04:20.42 | Lino` | :D |
| 04:21.03 | Sedorox | good think I run linux.. dont' have to worry about the viruses and shit |
| 04:21.21 | Sedorox | 'tho when I reinstalled windows last year.. 2 secs after inital install it got hit.... |
| 04:22.36 | Maxxed | aigh fellas im out |
| 04:22.40 | Maxxed | headed to the his |
| 04:22.44 | Maxxed | hiz* |
| 04:22.48 | Maxxed | halla at chall l8r |
| 04:23.44 | Lino` | :D |
| 04:24.35 | Lino` | at the sound of the tone the time will be exactly 6:25 in the morning. which means 72 hours without sleep |
| 04:25.00 | Sedorox | lol |
| 04:25.10 | Sedorox | well.. I'm off.. gonna go play some farcry before I turn in |
| 04:25.12 | Sedorox | night |
| 04:25.37 | Lino` | :-P |
| 04:27.10 | Darwin_35 | man my dialplan is now 45 pages |
| 04:27.23 | Strom_C | Darwin_35, includes are your friend |
| 04:27.25 | Darwin_35 | I just printed it |
| 04:29.36 | *** join/#asterisk Tili (i=Tili@61.144.20.233) |
| 04:29.50 | exten123 | It's linux got some editor can just show line that is not start with ;. |
| 04:30.08 | rbridal | its got grep |
| 04:31.28 | Lino` | ;) |
| 04:32.02 | Lino` | thats one reason why i love unix since i was 6 years old *g* |
| 04:32.32 | Lino` | all those pretty useful tools which windows doesnt have |
| 04:32.32 | rbridal | honestly, I've got about 10 years now with linux, and I'm still a windows guy |
| 04:32.45 | Lino` | well it depends actually |
| 04:32.56 | Lino` | for end user interfacing, windows is great |
| 04:33.02 | Lino` | for data processing etc linux is great |
| 04:33.31 | rbridal | unless your developers can only code visual basic .. then you're screwed and your dp boxes run f'in windows |
| 04:33.57 | Lino` | i do software development for banks etc, a lot of cheque processing etc. windows 16cpu box can process 2mio cheques / hour, linux can do 3,5mio / hour with 1 cpu |
| 04:34.11 | Lino` | 3.5 |
| 04:35.01 | rbridal | not to put you down on this, but since you're a linux guy, I'm sure there are windows devs out there who can push that hourly up significantly |
| 04:35.13 | Lino` | well of course that would be possible |
| 04:35.34 | Lino` | but you'll have to do much optimizing etc, things i almost dont have to do @ linux |
| 04:36.01 | rbridal | I would bet that, given the right developers on each side, a program running on a windows server will come fairly close, but never exceed, what a good linux developer could accomplish |
| 04:36.24 | Lino` | (main problem in this case is a big fat oracle db running on the same system, linux can distribute the compute power, windows cant. oracle just hugs it all up) |
| 04:36.45 | rbridal | oracle on a windows box? but - why? |
| 04:36.51 | Lino` | without oracle the 16cpu box is faster than the 1 cpu linux box (haha) |
| 04:37.00 | Lino` | thats what i ask myself every single day |
| 04:37.29 | Lino` | (i dont really like oracle - its a big fat hard to administrate expensive slow and ugly system) |
| 04:37.51 | rbridal | figure, you pay that much for licensing .. especially on a 16 proc box (fuck that'd be expensive) .. why not throw solaris at it? |
| 04:38.08 | Lino` | dunno, im just a developer, not the decision maker. |
| 04:39.05 | Lino` | if i would make the decisions everything would be better (almost) - at least i wouldnt switch storage to SAN when already facing performance problems. |
| 04:39.10 | justinu | apprantly 10g is much easier to run |
| 04:39.19 | Lino` | its still 9i |
| 04:39.29 | justinu | yeah... we're running 8i |
| 04:39.34 | justinu | which is kinda shitty |
| 04:39.46 | Lino` | but hey, we still have 200k windows nt boxes in the network. |
| 04:39.53 | Abydos313 | we run 9i also and it's on windows servers.. using dataguard and works perfectly |
| 04:40.12 | Lino` | Abydos313, it works great as long as you only run oracle on the server. |
| 04:40.29 | Abydos313 | yeah that's all those dells do is oracle |
| 04:40.31 | Lino` | if you have 2 or more concurrent applications on that server oracle f**ks up everything. |
| 04:40.46 | rbridal | 9i on ours, too .. we'd never have it if one of our vendors didn't require it for least cost routing :( |
| 04:40.47 | enots | :) |
| 04:40.53 | Abydos313 | what was teh other program? |
| 04:41.07 | Lino` | recostar |
| 04:41.13 | Lino` | and my application |
| 04:41.22 | Abydos313 | never heard ofthat app |
| 04:41.41 | Lino` | it recognizes handwriting on cheques and other transaction slips |
| 04:42.04 | Abydos313 | ok |
| 04:42.06 | Lino` | (german banks love to use forms, they all have to be processed) |
| 04:43.09 | Lino` | now look at this. somebody was clever while designing this database. he declared a 255 char long field for a zip code (always 5 chars long ) - ~20 mio datasets in the table *gg* |
| 04:43.28 | Abydos313 | heh |
| 04:44.05 | rbridal | damn europeans |
| 04:44.15 | Lino` | hey, shut up :-P |
| 04:44.28 | rbridal | lol, sorry man |
| 04:44.42 | Lino` | :D |
| 04:44.46 | Lino` | np |
| 04:44.49 | enots | lino, did u finish this design with bitmap index? :) |
| 04:45.03 | Lino` | huh? |
| 04:45.09 | Lino` | 7970? |
| 04:45.10 | bkw_ | http://video.google.com/videoplay?docid=-8260059923762628848 |
| 04:45.14 | bkw_ | watch that |
| 04:45.27 | Lino` | which design @ enots? |
| 04:45.52 | enots | 255 chars for zip :) |
| 04:46.20 | Lino` | no |
| 04:46.27 | Lino` | i'm revising it. |
| 04:46.36 | Lino` | i would not be that stupid |
| 04:47.31 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 04:47.41 | Lino` | right now i'm correcting it - that just costs 5 minutes per field ;) |
| 04:48.07 | *** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 04:56.25 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:58.42 | *** join/#asterisk b66mer (n=b66mer@204.9.61.37) |
| 04:58.49 | *** join/#asterisk bjweeks (n=bjweeks@24.137.156.198) |
| 04:59.59 | Idle | where can I get more info on calling from a zaptel FXO line? I cant seem to find much docs on Wildcards.. |
| 05:01.01 | Lino` | which card? |
| 05:01.12 | Idle | TDM100 IIRC |
| 05:01.17 | Idle | the 4 port, :) |
| 05:01.18 | b66mer | just do a search for asterisk dialplans |
| 05:01.41 | b66mer | what kind of wild cards do you need? |
| 05:01.56 | wasim | Idle: Dial(Zap/1/123948273940 |
| 05:02.00 | Idle | I have one, :) I just need to figure out how to use it |
| 05:02.16 | Idle | wasim: 1 being the channel, and the # being the phone # to dial I assume? |
| 05:02.25 | wasim | Idle: modprobe zaptel, modprobe wctdm, ztcfg -vvv |
| 05:02.30 | b66mer | http://voipspeak.net/index.php?/content/view/33/2/ |
| 05:02.44 | b66mer | you can download the o'reilly asterisk book for free |
| 05:02.57 | b66mer | has all of the wildcards and explainations |
| 05:03.51 | Idle | wasim: its working, I just need the dialplan stuff |
| 05:04.02 | b66mer | exten => _91800NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:${TRUNKMSD}}) |
| 05:04.02 | Idle | ok |
| 05:06.01 | b66mer | if I want to have a SIP call initiated from the internet come in to my asterisk behind a firewall... is all I need 5060/udp and 10000-20000/udp? |
| 05:07.08 | wasim | Idle: _9.,1,Dial(Zap/g1/${EXTEN:1}) will send out on zap group 1 (which you can define in zapata.conf) every dialed extension beginning with 9, and strip the 9 |
| 05:07.29 | Idle | hm |
| 05:07.35 | Idle | groups eh... |
| 05:08.16 | wasim | group=1 before your channel=1-4 definition |
| 05:08.21 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:09.06 | Idle | yea |
| 05:09.27 | Idle | how does it know where to split them? |
| 05:09.52 | bjweeks | is there a list of commands that can be run from phones like *60, *61 etc...? |
| 05:12.38 | Idle | hm... can the zaptels send caller id's on fxo lines so that I can set my own caller ID instead of my telco? |
| 05:13.07 | file[laptop] | you can't set outgoing caller ID on analog FXO lines |
| 05:13.54 | b66mer | if I want to have a SIP call initiated from the internet come in to my asterisk behind a firewall... is all I need 5060/udp and 10000-20000/udp? |
| 05:14.31 | Idle | ok |
| 05:16.24 | Idle | does it round-robin the groups? or go by channel #? |
| 05:21.03 | exten123 | izit igorepat only work for zap interface? |
| 05:22.19 | *** join/#asterisk lepine (n=Miranda@Toronto-HSE-ppp3774609.sympatico.ca) |
| 05:22.30 | lepine | good evening everyone |
| 05:22.34 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 05:23.13 | lepine | has anyone ever worked with a piece of software called uScreen / uReach ? |
| 05:23.27 | Lino` | oh yeah |
| 05:23.49 | Lino` | now i revised the database. a query which used to take 42 seconds now takes 0.1 second |
| 05:24.19 | lepine | the company i work for got many 'unique number' services to try to have stats on their marketting campaigns |
| 05:24.32 | lepine | the software supplied kind of sucks ... |
| 05:24.52 | lepine | i joyfully noticed the desktop software uses XML |
| 05:25.17 | ManxPower | aclu.tv |
| 05:25.41 | lepine | is that sort of thing common? documented? |
| 05:26.22 | lepine | i wrote up a quick php script that makes a call to get the voicemail messages ... |
| 05:26.35 | lepine | Event: message-list |
| 05:26.36 | *** join/#asterisk Op3r (n=op3r@202.71.189.90) |
| 05:26.40 | Darwin_35 | brb |
| 05:26.47 | Op3r | any can help me with call transfer?" |
| 05:27.06 | lepine | could anyone point me to docs about which type of events might be available so i could get the number of attempted calls etc etc ?? |
| 05:27.48 | Lino` | hm? |
| 05:28.37 | ManxPower | Op3r, press the TRANSFER button on your phone. |
| 05:28.53 | ManxPower | The follow the instructions for transfering that came with your SIP phone. |
| 05:28.54 | Op3r | ManxPower: we are using analog phones connected to quintum |
| 05:28.57 | Op3r | <PROTECTED> |
| 05:29.33 | ManxPower | Is the Quintum a Channelbank or an analog/SIP gateway |
| 05:31.01 | Op3r | analog/SIP |
| 05:31.50 | ManxPower | Op3r, then you MUST find the information from the Quintum becuase the Quintum is what is signaling the transfer |
| 05:33.26 | lepine | where could i find a list of possible events for a subscribe command ? |
| 05:35.04 | Op3r | any other way to deal with it on the extensions.conf? |
| 05:35.40 | b66mer | if I want to have a SIP call initiated from the internet come in to my asterisk behind a firewall... is all I need 5060/udp and 10000-20000/udp? |
| 05:43.29 | ManxPower | Op3r, no, there is not because Asterisk will not even SEE the transfer until the Quintum sends it the request |
| 05:44.12 | ManxPower | the quintum is acting as (a rather dumb) PBX |
| 05:50.14 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 05:51.38 | *** join/#asterisk Tili (i=Tili@61.144.20.233) |
| 06:04.22 | *** join/#asterisk nixbox (n=nixbox@129.110.198.134) |
| 06:04.24 | nixbox | hi all |
| 06:05.06 | nixbox | i have configured asterisk on my machine, and xlite on another, xlite is able to register with asterisk but when i call from xlite, it immediately says 404 not found, client mistakes |
| 06:05.11 | nixbox | what am i doing wrong? |
| 06:05.40 | Strom_C | is the extension in the same context as the number you're trying to dial? |
| 06:06.04 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 06:06.51 | nixbox | i have not specified context |
| 06:06.55 | nixbox | its the default |
| 06:07.17 | nixbox | i have put entries for both extensions in extensions.conf under a [default] block |
| 06:08.23 | Strom_C | pastebin your extensions.conf |
| 06:08.28 | Strom_C | ~pastebin |
| 06:08.30 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 06:10.09 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 06:11.50 | nixbox | Strom_C, http://pastebin.com/630295 |
| 06:12.31 | Strom_C | well theres your problem |
| 06:12.40 | Strom_C | you have two contexts labeled default |
| 06:13.49 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 06:22.31 | nick125 | when you record a call, where does it usually go? |
| 06:22.35 | nick125 | (where is it stored?) |
| 06:24.39 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 06:24.44 | Strom_C | /var/spool/asterisk/monitor |
| 06:25.47 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:26.03 | kmilitzer | Morning everyone ... |
| 06:26.19 | nick125 | I really should setup voicemail *sighs* |
| 06:27.04 | nixbox | Strom_C, thanks, its ringing now, but on linux when i press accept it says that i should close any application using the soundcard though i have no applications using the soundcard, i am using kphone |
| 06:27.22 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 06:27.23 | Strom_C | *shrug* i cant debug your softphone |
| 06:28.22 | salaud | I just installed Asterisk 1.2.1 on one of my servers and I can no longer get DTMF through IAX channels.. any ideas? |
| 06:29.29 | X-Rob | don't use 1.2.1 |
| 06:29.30 | X-Rob | use 1.2.6 |
| 06:29.54 | salaud | X-Rob: fixes? |
| 06:30.02 | X-Rob | *shrug* |
| 06:30.04 | salaud | But, not in debian |
| 06:30.06 | X-Rob | but it's 6 months old. |
| 06:30.12 | X-Rob | Hah |
| 06:30.13 | X-Rob | Well |
| 06:30.16 | X-Rob | go ask debian support |
| 06:30.33 | Strom_C | salaud, yeah. install 1.2.6 |
| 06:30.46 | Strom_C | www.asterisk.org |
| 06:30.50 | salaud | Hmm... |
| 06:30.53 | Strom_C | dont be a wimp! compile it yourself! |
| 06:31.25 | salaud | Strom_C: I have many times... but, now we have a directive (by the network server manager) to stay with what's in debian |
| 06:31.36 | salaud | The issue is that it works on another 1.2.1 machine |
| 06:31.42 | Strom_C | tell your network server manager to cram it ;) |
| 06:31.43 | salaud | Has anyone heard of this problem? |
| 06:31.52 | Strom_C | salaud, what codec are you using? |
| 06:32.02 | salaud | ulaw |
| 06:32.14 | Strom_C | what type of phones? |
| 06:32.32 | salaud | This is a call coming in from a PSTN gateway |
| 06:33.00 | salaud | hits a context that does, Set(TIMEOUT(digit)) and waits, etc.. |
| 06:33.22 | salaud | they used to work fine... I just noticed that now when a caller hits DTMF it don't work over IAX |
| 06:33.36 | salaud | when the call comes in over SIP from inside the phone system, DTMF works fine |
| 06:33.54 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 06:34.53 | salaud | it's just bizarre |
| 06:36.22 | salaud | I like to take a shot a fixing the problem as things stand in 1.2.1 .. before I screw up all the nice CFengine work that the network manager has in place by compiling something ... especially since 1.2.1 DTMF seems to work fine on another box |
| 06:36.26 | salaud | any ideas? |
| 06:42.54 | dokhench | anyone in channel that has done faxing from an iaxy over local lan to asterisk then out zap interface care to comment on the reliability of the faxing? |
| 06:44.19 | testshifter | help |
| 06:44.55 | testshifter | ls |
| 06:44.56 | Strom_C | dokhench, I've done something similar |
| 06:44.59 | testshifter | ~docs |
| 06:45.00 | jbot | i guess 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 |
| 06:45.27 | konfuzed | hey can I get a channel bank of iaxy s any where |
| 06:45.55 | Strom_C | you can get a channel bank |
| 06:46.03 | Strom_C | what do you want to do, konfuzed? |
| 06:47.12 | konfuzed | how rude ebay says connection refused |
| 06:49.53 | konfuzed | i want to do (well i was tricked into) move 40 4j11 jacks in a 40 unit apartment over to the basement connection as iaxy to an * gateway IAX to DID provider |
| 06:50.30 | Strom_C | konfuzed, get several channel banks and a quad-span digium T1 card |
| 06:50.33 | konfuzed | then they can keep there sme phone set and not even have to think |
| 06:50.47 | Strom_C | but first make sure you can even port all those numbers |
| 06:51.00 | brettnem | konfuzed: where are you located? |
| 06:51.14 | konfuzed | toronto canada |
| 06:51.21 | brettnem | doh |
| 06:51.34 | brettnem | I hear the ADIT 600 will someday support SIP. |
| 06:52.02 | brettnem | You can try the MGCP ADIT 600..it'll do what you want.. one box. 40 Pots.. not IAX tho.. MGCP. |
| 06:52.10 | konfuzed | should be number portability but it would not surprise me if the tenants had issue with their existing phone bill |
| 06:52.31 | konfuzed | screw that if its not iax im not doing |
| 06:52.35 | konfuzed | it |
| 06:52.37 | konfuzed | or IT |
| 06:52.51 | brettnem | well.. stock up on some iaxys then |
| 06:52.57 | brettnem | sounds like a plan |
| 06:53.02 | konfuzed | yeah all taped to a board |
| 06:53.06 | konfuzed | hahahaaaaa |
| 06:53.20 | brettnem | why so determined to use IAX? |
| 06:53.30 | konfuzed | rhino looks good |
| 06:53.33 | brettnem | just put a server on prem |
| 06:53.40 | konfuzed | have to |
| 06:53.45 | konfuzed | i would nto do it any other way |
| 06:53.54 | brettnem | why not? bandwidth? |
| 06:53.58 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 06:54.52 | konfuzed | they can have the one time hardware of pbx and channel bank setup and what to give them a voicemail account and free IP to IP only everything else would be extras |
| 06:55.10 | konfuzed | they tried to stick me with $3/unit/mth |
| 06:55.24 | brettnem | ok, that didn't answer the question |
| 06:55.27 | konfuzed | but i stretched it to 5 |
| 06:55.31 | konfuzed | oh |
| 06:55.32 | konfuzed | well |
| 06:55.34 | konfuzed | yeah bandwidth |
| 06:55.40 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 06:55.43 | brettnem | why not just do g.729? |
| 06:55.43 | konfuzed | last mile scenario |
| 06:55.55 | brettnem | I'm very familiar with last mile |
| 06:56.14 | konfuzed | well it fits my model very well. |
| 06:56.31 | brettnem | how do you know if it fits your model if you don't have hardware in your model yet? |
| 06:57.06 | konfuzed | we operate an openBSD connection load balancer for say 6 ADSLs and a lan to the 40 units |
| 06:57.14 | konfuzed | well will soon to be that is |
| 06:57.29 | X-Rob | konfuzed, boy, is your life going to suck soon. |
| 06:57.39 | konfuzed | oh yeah |
| 06:57.39 | brettnem | hmm.. I do 400 units over 1 T1 with g.729 |
| 06:57.45 | konfuzed | mmmmm |
| 06:57.53 | X-Rob | konfuzed, that works with TCP because TCP is stateful. UDP isn't. |
| 06:57.59 | X-Rob | VoIP is UDP. |
| 06:58.15 | X-Rob | a sensible load balancer will send all it's udp traffic over one interface |
| 06:58.20 | brettnem | so you are doing it for $5CDN? |
| 06:58.26 | konfuzed | they also tied in the wireless |
| 06:58.33 | X-Rob | which means it's just the same as using one adsl. |
| 06:58.36 | brettnem | wow |
| 06:58.46 | brettnem | you are going to have some serious jitter |
| 06:58.57 | X-Rob | he's gunna have some serious not-working. |
| 06:58.59 | konfuzed | mmmmm uh yeah as an add on to the 29/mth/unit for managed lan |
| 06:59.19 | konfuzed | i told them not to mention voip dont sell them voip |
| 06:59.24 | brettnem | X-Rob: agreeded. |
| 06:59.27 | konfuzed | they dived on the voip |
| 06:59.48 | konfuzed | i wonder if it is possible to get a t1 out there |
| 06:59.51 | brettnem | doen't sound like you have the network infrastructure to support your need |
| 07:00.00 | brettnem | sure it's possible. what are you going to pay for it |
| 07:00.19 | konfuzed | yeah i figured it had to jump to at least 10 dsl to make that start to be viable |
| 07:00.32 | konfuzed | even at 5Mb down times 10 |
| 07:00.52 | konfuzed | that 50Mb rating is going to pull in probably less than half |
| 07:01.05 | konfuzed | 8^( |
| 07:01.11 | brettnem | you don't seriously think that concatonating DSL lines is going to work, do you? |
| 07:01.27 | konfuzed | i would kinda like to put a DSLAM in the building |
| 07:01.33 | konfuzed | but I think that blows the budget |
| 07:01.47 | brettnem | you need bandwidth to the property |
| 07:01.47 | Strom_C | DS3 is the way to go :) |
| 07:01.54 | X-Rob | konfuzed, DLSAM's are usually cheaper than 2 runs of RJ45 to each room. |
| 07:02.07 | orlock | dslam will mean you can us ephone cables |
| 07:02.09 | konfuzed | why is that the average person thinks phone calls are weightless invisble thought process or something |
| 07:02.27 | orlock | but going some smart etheernet switch/router would be cheaper, but would need cat5 or better |
| 07:02.31 | brettnem | I didn't realize the average person thought that. |
| 07:02.42 | brettnem | konfuzed: I hope you get some bandwidth before you do this project |
| 07:02.42 | konfuzed | well then the 40 ADSL modems in each unit |
| 07:02.47 | orlock | the average person is a fucking moron |
| 07:02.56 | brettnem | orlock: amen |
| 07:03.00 | konfuzed | i try to stear them away |
| 07:03.02 | orlock | and just remember, half of people are below that average |
| 07:03.13 | *** join/#asterisk nixbox (n=nixbox@129.110.198.134) |
| 07:03.20 | brettnem | well this sounds like a setup for a disaster |
| 07:03.42 | nixbox | is there a way whereby asterisk won't seize my soundcard, i want to run a client on the same machine running asterisk. |
| 07:03.45 | brettnem | konfuzed: check out tut-systems.. but for the love of all that is good.. get some real bandwidth before you do this |
| 07:03.55 | konfuzed | but then my buddy made arrangement with another budy to stick a link on support.microsoft.com for all those that microsoft no longer will give support to |
| 07:03.57 | brettnem | nixbox: noload => chan_oss |
| 07:03.58 | brettnem | .so |
| 07:04.15 | konfuzed | i dont think they can get real bandwidth out there |
| 07:04.25 | brettnem | konfuzed: then they can't get voip out there |
| 07:04.39 | konfuzed | those guys blew it I told them not to mention voip |
| 07:04.55 | konfuzed | internet surfing no problem |
| 07:04.55 | brettnem | it doesn't matter what they SAY. it WONT WORK on the backend |
| 07:05.09 | konfuzed | a dozen chicks on the phone and forget it |
| 07:05.18 | brettnem | it'll probably suck for internet surfing as well |
| 07:05.28 | brettnem | X-Rob: am I missing something here? |
| 07:05.39 | X-Rob | brettnem? |
| 07:05.43 | konfuzed | well just for home use |
| 07:05.55 | X-Rob | You're saying it won't work for UDP. You're right. |
| 07:05.56 | konfuzed | internet will be just fine |
| 07:05.56 | brettnem | konfuzed: for 40 homes use |
| 07:06.09 | X-Rob | it'll work for TCP because TCP is stateful, and the adsl router can remember which source IP was for which connection |
| 07:06.19 | X-Rob | with UDP, the machine has NO WAY of telling what stream it's attached to. |
| 07:06.23 | konfuzed | well yeah 1 5mb line for 4 units |
| 07:06.28 | brettnem | X-Rob: I'm not convinced it'll work for 40 units even with TCP. |
| 07:06.32 | x86 | is there a Return function? |
| 07:06.34 | brettnem | 4 units? |
| 07:06.37 | brettnem | yes |
| 07:06.40 | brettnem | there is |
| 07:06.41 | X-Rob | brettnem, oh yeah, it will. That's actually quite easy. |
| 07:06.49 | X-Rob | web browsing should work ok |
| 07:06.56 | konfuzed | yeah 10 dsl lines at 5 MB each hooked into a lan of 40 units |
| 07:07.00 | brettnem | no I'm not convinced his 5mb isn't really 1mb |
| 07:07.05 | X-Rob | it'll be clunky and any web server that does session tracking will break |
| 07:07.26 | brettnem | that sounds like "not working" to me.. ;) |
| 07:07.28 | x86 | brettnem: can i return to the next priority in line? |
| 07:07.31 | X-Rob | What I'd do is I'd have one ADSL line per 4 units |
| 07:07.35 | konfuzed | i hoping to get 2.5Mb per line in real life |
| 07:07.45 | X-Rob | Seperate machine, and if they want to do VoIP, have an asterisk machine for each one |
| 07:07.48 | konfuzed | well yeah 1 - 5mb line for 4 units |
| 07:07.53 | X-Rob | asterisk machine does NAT and Firewall |
| 07:07.54 | brettnem | konfuzed: you have no idea what kind of concentration on the adsl lines your provider does |
| 07:07.57 | konfuzed | dats what i said |
| 07:08.00 | brettnem | a 5mb line isn't dedicated bandwidth |
| 07:08.19 | X-Rob | so each 4 units has it's own ADSL line, and a linux box running asterisk. |
| 07:08.22 | konfuzed | well im friendly with the isp for the past 6 years |
| 07:08.37 | konfuzed | well one pbx will do just fine |
| 07:08.42 | X-Rob | no, it won't. |
| 07:09.17 | brettnem | If I'm installing DSLAMS, I'd run a 10M to one DSLAM and put 100 5M customers on it |
| 07:09.26 | X-Rob | konfuzed, you don't understand layer 2 stuff, and you're trying to do stuff that isn't going to work, or is going to work purely by coincidence. |
| 07:09.27 | x86 | brettnem: s,n,Dial(blah) s,n,Goto(s-${DIALSTATUS}) s,n,Dial(otherprovider) s,n,Goto(s-${DIALSTATUS}) s,n,Congestion() s-ANSWERED,n,Hangup() s-BUSY,n,Return() |
| 07:09.29 | konfuzed | load balance the whole phat 10 DSLs together and some body will pull down 8Mb with multiple downloads |
| 07:09.39 | brettnem | btw, that's just an example.. I'd have to think about the concentrations a bit more.. |
| 07:09.41 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-233.claranet.co.uk) |
| 07:09.53 | X-Rob | konfuzed, YOU CAN'T LOAD BALANCE UDP |
| 07:09.59 | Qwell | konfuzed: or if one of the 4 uses 5MB, the other 4 get screwed. Either way is bad |
| 07:10.01 | X-Rob | ARE YOU NOT PAYING ATTENTION? |
| 07:10.07 | brettnem | x86: what do you want that return to do? |
| 07:10.13 | brettnem | X-Rob: he isn'... |
| 07:10.18 | Qwell | and that's only if you actually get 10x5 :) |
| 07:10.21 | x86 | brettnem: continue down the dialplan |
| 07:10.25 | Qwell | (which you won't) |
| 07:10.30 | konfuzed | uh well i dont let this out too often but |
| 07:10.38 | brettnem | oh here it comes |
| 07:10.47 | x86 | brettnem: i want to dial 3 different providers in a specific order, but if the call is answered by any one of them i want to stop traversing the dialplan |
| 07:10.47 | X-Rob | konfuzed, you better have a condom on that. |
| 07:11.06 | Qwell | x86: check ${DIALSTATUS} |
| 07:11.14 | brettnem | X-Rob: hah |
| 07:11.15 | konfuzed | ive got at least 4 seperate DSLs on seperate lans that each connect more than 15 users |
| 07:11.18 | konfuzed | they are 5mb connections |
| 07:11.22 | x86 | Qwell: *nod* read my example ;) |
| 07:11.29 | X-Rob | konfuzed, and their VoIP doesn't work. |
| 07:11.35 | X-Rob | or it'll only work for one person |
| 07:11.36 | x86 | Qwell: s,n,Dial(blah) s,n,Goto(s-${DIALSTATUS}) s,n,Dial(otherprovider) s,n,Goto(s-${DIALSTATUS}) s,n,Congestion() s-ANSWERED,n,Hangup() s-BUSY,n,Return() |
| 07:11.45 | konfuzed | and now udp wont load balance but thats ok |
| 07:11.53 | konfuzed | you dont need to load balance one voip call |
| 07:11.57 | brettnem | x86: yeah, dialstatus is the right way to do that.. check dialstatus each time and decide what to do. |
| 07:12.06 | konfuzed | you need to laod balance the tcp/ip connections |
| 07:12.14 | brettnem | konfuzed: what you have described won't work for 40 units |
| 07:12.21 | konfuzed | all those peer-to-peer on udp can go to hell |
| 07:12.22 | x86 | brettnem: right, but i want to use gotos |
| 07:12.25 | X-Rob | konfuzed, if you're not talking about voip, what are you doing in #asterisk? |
| 07:12.32 | X-Rob | why don't you go to #networking-101 or something |
| 07:12.39 | x86 | brettnem: as you can see in my example, i already am using dialstatus |
| 07:12.48 | brettnem | x86: check Qwell's example above |
| 07:12.52 | Qwell | x86: Then what's the question? |
| 07:13.06 | x86 | brettnem: i dont see it ;) |
| 07:13.08 | X-Rob | x86, GotoIf([${DIALSTATUS}=ANSWERED]?end) |
| 07:13.14 | konfuzed | im working with * internally for my own uses and experimenting for small pbxs |
| 07:13.16 | x86 | X-Rob: thanks :) |
| 07:13.45 | konfuzed | really i told those guys that I refuse to talk to them about providing voip services |
| 07:13.45 | brettnem | x86: or exten => s-CHANUNAVIL,1,Dial(SIP/provider2) |
| 07:13.48 | Qwell | I swear to god, if I see/hear one more AT&T commercial... |
| 07:13.50 | Qwell | I'm gonna go nuts |
| 07:13.58 | brettnem | me too |
| 07:14.13 | Qwell | "All around th.." shut the fuck up with that already...I get it |
| 07:14.15 | Qwell | ugh |
| 07:14.26 | X-Rob | *song song* AT&T is Really nice.. WE don't eat your kids.. We promise that you'll love us, just don't query your bills! *song song* |
| 07:14.29 | brettnem | they can take their fancy logo and go stick it |
| 07:14.40 | Qwell | brettnem: It's just that damn song that bugs me |
| 07:14.40 | konfuzed | im saying at best you could give them some toys to play with but not replace a phone system |
| 07:15.06 | brettnem | konfuzed: people who don't know they are getting voip and are expecting a regular phone line just love TOYS |
| 07:15.07 | X-Rob | konfuzed, you better make sure you tell them that what you're planning will make voip IMPOSSIBLE for the clients |
| 07:15.22 | x86 | X-Rob: can i put some kind of OR in there, where if the number was busy it also goes to end? |
| 07:15.33 | X-Rob | x86, just do another gotoif |
| 07:15.38 | x86 | hmm |
| 07:15.38 | X-Rob | oh |
| 07:15.39 | x86 | ok |
| 07:15.40 | X-Rob | sorry |
| 07:15.40 | X-Rob | yes |
| 07:15.48 | x86 | just use a pipe? |
| 07:15.53 | x86 | or colon? |
| 07:15.54 | brettnem | no pipe |
| 07:16.03 | brettnem | just do another gotoif.. it's safe.. |
| 07:16.04 | X-Rob | [ [ ${DIALSTATUS = ANSWERED ] || [ ${DIALSTATUS} = BUSY ] ] |
| 07:16.09 | x86 | w00t |
| 07:16.11 | x86 | thanks :) |
| 07:16.16 | brettnem | ok fine whatever.. haha |
| 07:16.18 | X-Rob | put quotes around all that though |
| 07:16.25 | X-Rob | "${ .. }" = "ANSWERED" |
| 07:16.35 | brettnem | I hate all the damn symbolic nesting in asterisk logic.. it's rediculous |
| 07:16.37 | konfuzed | ok. if they manage to sign the deal next month for 5years at $40/unit/mth and the factoring company coughs up 60 % upfront then the wire install and equipement set up should be sufficiently looked (crossing my fingers) |
| 07:16.53 | konfuzed | nah its too tight isnt it |
| 07:17.05 | brettnem | your BACKHAUL is totally F-D! |
| 07:17.05 | konfuzed | like I said I was tricked into this |
| 07:17.10 | konfuzed | yeas it is |
| 07:17.13 | X-Rob | konfuzed, say no. Walk away. |
| 07:17.19 | X-Rob | you'll just end up being suet. |
| 07:17.22 | X-Rob | sued, too. |
| 07:17.25 | brettnem | konfuzed, so no, RUN AWAY |
| 07:17.45 | brettnem | where did 40/unit come from? |
| 07:17.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:17.52 | brettnem | btw, is this CDN? |
| 07:18.04 | Qwell | Why not just make them get internet on their own? |
| 07:18.05 | konfuzed | ah some dumb ass co-op at the far end of town. |
| 07:18.23 | konfuzed | funnily enough not to far from microsoft support |
| 07:18.33 | brettnem | funnily |
| 07:18.37 | Qwell | Is this for apts, or what? |
| 07:18.41 | konfuzed | but like o the opposite corner that never got the telecom upgrade |
| 07:18.53 | brettnem | konfuzed: get a T1 |
| 07:18.56 | konfuzed | yeah apt housing co-op |
| 07:19.02 | brettnem | konfuzed: it won't take an upgrade |
| 07:19.04 | Qwell | meh, let them get their own internet |
| 07:19.07 | x86 | X-Rob: what about if it rings ok at provider 1, but user lets it ring and ring until the timeout on the Dial() command... is there a TIMEOUT dialstatus? |
| 07:19.11 | Qwell | If I lived there, I would get my own anyhow |
| 07:19.21 | brettnem | look, students are bandwidth whores |
| 07:19.23 | Qwell | but, guess what...I wouldn't be allowed to, because you have a dumbass contract with the telco |
| 07:19.32 | X-Rob | x86, you seem to be unaware of the magic command 'show applcation dial' |
| 07:19.33 | Qwell | congratulations, you've fucked me |
| 07:19.38 | konfuzed | stupid place has 5 buildings with 2 units per floor for 4 floors all connected via passment |
| 07:19.43 | x86 | X-Rob++ |
| 07:19.43 | konfuzed | uh basement |
| 07:19.48 | x86 | X-Rob: thanks man :) |
| 07:19.49 | brettnem | x86: it'll just go to the next priority |
| 07:20.01 | x86 | brettnem: right, which i dont want ;) |
| 07:20.15 | X-Rob | you want to check if dialstatus = noanswer |
| 07:20.20 | brettnem | konfuzed: that's pretty typical setup |
| 07:20.20 | X-Rob | <PROTECTED> |
| 07:20.20 | X-Rob | <PROTECTED> |
| 07:20.20 | X-Rob | <PROTECTED> |
| 07:20.29 | x86 | ooooooo :) |
| 07:20.48 | konfuzed | i prefer one building with 8 to 16 units per floor |
| 07:20.53 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 07:21.07 | brettnem | konfuzed: you don't frequently get to do what you want.. |
| 07:21.13 | brettnem | I think that was a song.. |
| 07:21.19 | konfuzed | the contact there is like oh I just got into the IT field 2 years ago |
| 07:21.27 | Qwell | brettnem: with slightly different lyrics. ;) |
| 07:21.28 | konfuzed | we should get in cisco and fiber optics |
| 07:21.35 | X-Rob | yeah. Goo cisco! |
| 07:21.44 | X-Rob | Buy Cisco stuff! |
| 07:21.51 | X-Rob | Pay me lots of money to fix it! |
| 07:21.52 | konfuzed | what a nut bar |
| 07:21.53 | Qwell | X-Rob: cisco sucks! |
| 07:21.56 | Qwell | oh...wait... |
| 07:22.06 | X-Rob | Qwell, it only sucks because you haven't paid me ENOUGH MONEY! |
| 07:22.08 | brettnem | konfuzed: this is a small complex.. it shouldn't be too hard |
| 07:22.15 | Qwell | I'm the the who gets everybody to start using sccp :P |
| 07:22.26 | X-Rob | Qwell, well, that's because you're mentally unbalanced. |
| 07:22.28 | konfuzed | i know its only 40 units, the lan is no problem |
| 07:22.34 | Qwell | X-Rob: heh, indeed |
| 07:22.42 | brettnem | konfuzed: the voice shouldn't be too hard either.. really.. |
| 07:22.43 | Qwell | Nugget is mad at me, I think. ;) |
| 07:22.46 | X-Rob | I have my two new SPA941's here |
| 07:22.54 | X-Rob | Qwell, did you actually fix chan_sccp? |
| 07:22.56 | konfuzed | and they want kewl wifi and so secure locustworld meshaps will be in the picture |
| 07:22.56 | Qwell | He now has to buy another 7960, to try out sccp, because I rant and rave about it all the time |
| 07:23.05 | Qwell | X-Rob: fuck no, it's a sinking ship |
| 07:23.14 | konfuzed | i have to sorta rearrange my lego pieces now though |
| 07:23.17 | Qwell | X-Rob: now, chan_skinny on the other hand... |
| 07:23.20 | X-Rob | Aaha. |
| 07:23.28 | brettnem | I want my 7920 to work with asterisk.. :( |
| 07:23.37 | Qwell | Will be teh rox0r by May, I'm hoping... |
| 07:23.54 | brettnem | my 7920 used to crash asterisk.. barf |
| 07:23.56 | Qwell | brettnem: I'm working on it |
| 07:23.57 | konfuzed | so this place has such old wiring that I'll run a second set of cat6 just for telephony |
| 07:24.04 | X-Rob | I was at a computer shop last week, and they guy is talking about VoIP, and I mentioned that I know a bit about it (*snigger*) |
| 07:24.09 | brettnem | konfuzed: sounds like overkill to me |
| 07:24.18 | X-Rob | it's amazing how much crap people will spout to make themselves seem to know more than they do |
| 07:24.25 | konfuzed | the guy wanted to jump on over to IPTV cause their cable contract was up for renewal |
| 07:25.06 | konfuzed | so he insists to add a clause for upgrade to such fancy services before the 5 years is up |
| 07:25.09 | brettnem | is it just me, or has konfuzed hit every corner of technology for his property without the possibility of actually using any of it? |
| 07:25.14 | konfuzed | the guy is a nut |
| 07:25.34 | konfuzed | I doubt even 4 of them have any clue about iptv |
| 07:25.43 | X-Rob | brettnem, you know, I was just thinking that. |
| 07:25.45 | Strom_C | brettnem, he's the guy who calls the special Graybar number for people who have no fucking clue what they're getting into |
| 07:25.52 | konfuzed | i know these people are insane |
| 07:25.57 | brettnem | : |
| 07:25.57 | konfuzed | im gonna run |
| 07:25.59 | brettnem | :) |
| 07:26.03 | konfuzed | thanks for the advice |
| 07:26.05 | X-Rob | I know, you should use wireless too |
| 07:26.06 | exten123 | why I until now I still cannot browse iaxtel.com what the problem? |
| 07:26.10 | konfuzed | i'll sleep better now |
| 07:26.14 | X-Rob | just coz that won't do you any good either |
| 07:26.20 | brettnem | I think you should sell it all wifi voip over adsl |
| 07:26.24 | konfuzed | i can tell them to dig them selves out |
| 07:26.29 | brettnem | I know.. |
| 07:26.34 | konfuzed | they did sell that |
| 07:26.42 | Strom_C | konfuzed, dont forget to add single AND multi-mode fiber into the mix!@ |
| 07:26.47 | *** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it) |
| 07:26.49 | brettnem | how about a wiki voip video phone running h323 |
| 07:26.50 | X-Rob | wifi _and_ fibre |
| 07:26.53 | X-Rob | OMG!@#$ |
| 07:26.54 | Strom_C | AND PoE |
| 07:26.56 | X-Rob | What a good idea! |
| 07:26.56 | konfuzed | im serious it was sort of sad |
| 07:27.00 | X-Rob | PoEoFibre |
| 07:27.01 | Qwell | brettnem: wiki? |
| 07:27.11 | brettnem | you totally need some SONET in there.. and maybe some DWDM |
| 07:27.16 | Strom_C | and auto wiki blogging software |
| 07:27.18 | Qwell | MPLS, *IN* the phone! |
| 07:27.18 | brettnem | Qwell: no thanks, I'm full |
| 07:27.24 | Strom_C | bliki! |
| 07:27.27 | brettnem | Qwell: DUDE |
| 07:27.31 | konfuzed | their all gonna get a reality check soon but I dont know if theyll be able to wrap their tiny collosums around it |
| 07:27.35 | Qwell | too much? |
| 07:27.38 | brettnem | oh I bet they will |
| 07:27.47 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:27.53 | X-Rob | konfuzed, Say this: |
| 07:27.56 | brettnem | oh Qwell: haha I meant Wifi |
| 07:27.59 | X-Rob | 'Dude. You guys are fucked. Seeya' |
| 07:28.01 | Qwell | ahh |
| 07:28.02 | X-Rob | then walk out |
| 07:28.13 | X-Rob | You might want to pad it out a bit |
| 07:28.28 | Qwell | X-Rob: three seconds between each should about do it |
| 07:28.50 | konfuzed | yeah im goin to go plug my iaxy into microsofts support rejects queue instead |
| 07:28.52 | brettnem | and when you're done, you can totally bling it out http://www.myblingring.com/ |
| 07:29.00 | konfuzed | hahhahaaaa |
| 07:29.30 | brettnem | um |
| 07:29.31 | brettnem | guys |
| 07:29.32 | X-Rob | Tell them that when the $40/month mob screws it totally up, you can do it properly, but expect to be paying for a t3 OR SOMETHING |
| 07:29.38 | brettnem | I don't think he gets it |
| 07:29.53 | X-Rob | wups |
| 07:29.56 | X-Rob | sorry for souting |
| 07:30.07 | X-Rob | but that kinda worked.. t3.. OR SOMETHING!#@! |
| 07:30.18 | brettnem | thank you QBERT |
| 07:30.22 | X-Rob | You can get the spittle flying if you want. That'll be funny. |
| 07:30.22 | Strom_C | HAHAH |
| 07:30.35 | Strom_C | QBERT for the win |
| 07:30.36 | konfuzed | well maybe only 10 of them will actually be interested in figuring out how to use it |
| 07:30.46 | brettnem | QBERT for IRC |
| 07:30.55 | brettnem | how to use WHAT? |
| 07:30.56 | X-Rob | So get a 3600 cisco in there or something, something that's upgradable. |
| 07:31.06 | X-Rob | 3640's are cheap, and can handle a DS3 worth of data. |
| 07:31.11 | brettnem | Hello, and welcome to vaporCo |
| 07:31.16 | konfuzed | yeah |
| 07:32.30 | *** join/#asterisk batman2 (n=asdfd@ip70-181-90-193.oc.oc.cox.net) |
| 07:32.41 | konfuzed | really i just wanted mini-pbx for me and my dog and to make it look like some one else answers my phone cause im too busy to talk about win98 |
| 07:32.54 | brettnem | win98? wha? |
| 07:33.06 | konfuzed | ms support rejects |
| 07:33.10 | Strom_C | konfuzed, please stop smoking all that crack |
| 07:33.10 | brettnem | does your dog.. actually have it's own extension |
| 07:33.29 | konfuzed | dos to win31 to win 95 98 and ME |
| 07:34.01 | brettnem | yay.. a support queue for antiquated softwares |
| 07:34.39 | konfuzed | microsoft tells on the phone that they can buy a supported OS or bugger off and call my buddy |
| 07:34.59 | brettnem | I'm sure MS loves to promote your buddy's business |
| 07:35.12 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 07:35.13 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:35.19 | konfuzed | hey its a fact of life or an unfortunate reality |
| 07:35.52 | brettnem | or a whimsical fairyland |
| 07:36.05 | konfuzed | well really my buddy answers the support line at ms support |
| 07:36.15 | brettnem | does your buddy live in india? |
| 07:36.22 | konfuzed | no in toronto |
| 07:36.28 | Strom_C | same thing |
| 07:36.31 | brettnem | does he share a cubicle with mr.gates? |
| 07:36.36 | brettnem | haha |
| 07:36.38 | konfuzed | but now he works at the missasauga call center |
| 07:37.05 | konfuzed | i dont think the phone reps ever get to go to seattle |
| 07:37.08 | brettnem | I bet he's got some serious pull at MS then, right? |
| 07:37.13 | konfuzed | no |
| 07:37.28 | brettnem | seattle? hmm.. and I thought had all that stuff in redmond |
| 07:37.58 | konfuzed | just enough to tell the callers they wont support where they can go to get help with their confusions |
| 07:38.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:38.17 | konfuzed | seatles play town isnt it |
| 07:38.27 | konfuzed | i dont bother with geography much |
| 07:38.47 | Strom_C | konfuzed, remember, india and botswana share a border along the yellow river |
| 07:38.56 | konfuzed | hahahaaaa |
| 07:39.08 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 07:39.29 | konfuzed | i think he's picking up the french canadian callers |
| 07:39.37 | konfuzed | that cant handle that in india |
| 07:40.23 | brettnem | I bet he gets a lot of calls |
| 07:40.41 | konfuzed | so i still cant really imagine anyone actually wanting to take those calls in the first place |
| 07:41.25 | brettnem | just ask your buddy |
| 07:41.29 | konfuzed | i said I would help them all download ubuntu and reinstall |
| 07:42.51 | konfuzed | getting a salary to support new microsoft technologies is one thing but |
| 07:43.24 | konfuzed | getting all the left over calls for whats no longer supported has to byte |
| 07:44.19 | konfuzed | myself i think I would be better off working at a chocolate shop |
| 07:44.36 | konfuzed | so i dont have to talk about computers to anyone |
| 07:48.09 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:48.37 | x86 | http://pastebin.ca/47573 |
| 07:49.17 | x86 | X-Rob: for some reason after the first dial attempt fails and it gets to the GotoIf, my SIP phone that's trying to dial via the context gets a 603 declined error |
| 07:54.51 | shido6 | mic check |
| 07:55.04 | x86 | 1212 |
| 07:55.13 | shido6 | scheduled outage here |
| 07:55.50 | brettnem | x86: Check your spaces around the = sign.. I think they might cause a parsing error.. |
| 07:55.58 | brettnem | ok, goodnight all.. it's been swell |
| 07:59.47 | kippi | Hey |
| 08:00.39 | kippi | I have just tried to start asterisk and got an error, I have tried to use ztcfg -v and get this error line 0: Unable to open master device '/dev/zap/ctl' anyideas? |
| 08:00.55 | ToTo | hi all, i wold to use fax with *, is it possible? |
| 08:01.12 | vgster | would people recommend acpi on or off? |
| 08:01.23 | vgster | ToTo yes it is possible |
| 08:02.07 | ToTo | vgster, there is an ad-hoc module? |
| 08:03.32 | ToTo | vgster, do you know where i can find some usefull doc? |
| 08:03.48 | vgster | http://www.voip-info.org/tiki-index.php |
| 08:04.01 | vgster | look at iaxmodem/hylafax and spandsp |
| 08:04.37 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:05.54 | vgster | different faxing solutions, depends which way you want to go and how many etc... |
| 08:06.14 | ToTo | vgster, tnx |
| 08:06.49 | vgster | np |
| 08:07.36 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:08.43 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 08:14.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:15.39 | wasim | ick asterisk crashing |
| 08:21.29 | x86 | mine too :( |
| 08:21.51 | x86 | on certain numbers i try to do ENUM lookups to (that have E164.org records) it crashes the server |
| 08:22.00 | x86 | works fine consistantly for other numbers |
| 08:22.02 | *** join/#asterisk SHad|Work (n=kvirc@popust.net) |
| 08:22.12 | x86 | same number length, same country, etc |
| 08:22.34 | SHad|Work | I need a little theoretical help with cdr_mysql in asterisk |
| 08:22.55 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 08:23.14 | SHad|Work | Is it possible to somehow make it record the call atempts that were placed to a SIP extensions that is not registered at the time? |
| 08:27.44 | *** join/#asterisk somethingbad (i=vladi@cnika.bg) |
| 08:27.56 | somethingbad | hello, i have a problem with asterisk-ooh323 |
| 08:28.21 | somethingbad | with callerid |
| 08:28.38 | somethingbad | *CLI> -- Executing Set("SIP/4912500-2b84", "CALLERID(number)=4912500") in new stack |
| 08:28.38 | somethingbad | <PROTECTED> |
| 08:29.09 | somethingbad | but "Unknown" number is displayed on the phone that i'm calling |
| 08:29.51 | *** join/#asterisk _medic_ (i=medic@p54B1A662.dip0.t-ipconnect.de) |
| 08:30.07 | _medic_ | Good Morning |
| 08:31.37 | *** join/#asterisk darkskiez (n=darkskie@213.190.164.48) |
| 08:31.57 | Tili | some guy suggested that using VoIP over VPN is better as ISPs can't block your calls as they wont know it. Does that sound right? VPN is TCP and then we will have UDP over TCP? |
| 08:31.59 | SHad|Work | any asterisk developers here at the moment? :) |
| 08:31.59 | Ansonmus | Good morning |
| 08:32.11 | _medic_ | Good Morning ... I suffer a serious asterisk problem, which i cannot debug,and i couldn't find any help in the web .. I can SIP without a problem, but approx 5 - 10 seconds after i took a CAPI-Call offhook, the whole box freezes :-( it happened withasterisk 1.2.4 as well with 1.1.x |
| 08:33.13 | _medic_ | i also tried capi-cm 0.6.4 as 0.5.4 .. no change :-( |
| 08:34.06 | Ansonmus | Can anyone help me with SIP? There must be sort groups like "helpdesk" and "secretary" how to config this? |
| 08:34.14 | Ansonmus | which stategy |
| 08:34.29 | wasim | Ansonmus: dialplan, extensions.conf |
| 08:35.11 | _medic_ | if you need groups, you maybe want to consider agents.conf and queue.conf too |
| 08:36.56 | Ansonmus | at client side, must a client enter the group sip like sip:helpdesk and sip:ansonmus or only sip:ansomnmus ? |
| 08:38.15 | Ansonmus | with X-lite I can enter two account but there is no option to differ the dialtone or so |
| 08:39.12 | _medic_ | ?? |
| 08:40.44 | Ansonmus | at this moment we have a normal phonecentral? with systemphones. my phone rings on helpdesk, and on company1.. I can see at some leds which type of incoming call it is. How can I realize this with sip |
| 08:40.49 | Ansonmus | or iax |
| 08:42.29 | _medic_ | with a softphone ? |
| 08:42.46 | Ansonmus | asterisk + softphone |
| 08:43.17 | _medic_ | you can't.. dont knowany softphone whichwould support that... but you can have some popup, telling you |
| 08:43.32 | _medic_ | triggered by asterisk |
| 08:43.48 | Ansonmus | hmm strange.. in the diagnostic log I can see which address is called |
| 08:44.37 | Ansonmus | It is imo a standard option in normal (non voip) telephonesystems |
| 08:45.30 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 08:45.31 | _medic_ | in the softphone ?..sure.. but you would either need a softphone, which supports multiple SIP-Accs, or some softphone wich supports kind of messaging ... |
| 08:45.41 | backblue | * |
| 08:45.45 | backblue | morning all |
| 08:46.02 | _medic_ | Ansonmus, no,its not .... System phones use proprietary extensions |
| 08:46.06 | _medic_ | hi backblue |
| 08:46.20 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 08:46.29 | _medic_ | and VOIP is not comparable to old style telephones |
| 08:47.00 | _medic_ | to beprecise, SIP is completely different |
| 08:47.33 | Ansonmus | but SIP is imo developped to solve such problems? |
| 08:47.47 | _medic_ | No |
| 08:48.13 | iDunno | SIP is nasty stuff. |
| 08:48.41 | _medic_ | you could use some formof h323, which is basically isdn protocol |
| 08:48.41 | iDunno | it's fine on internal networks. across the interweb it's 'fun' with firewall tastic foo. |
| 08:48.56 | X-Rob | Ansonmus, you want a 'key system' |
| 08:49.03 | X-Rob | Asterisk can't do that. Sorry. |
| 08:49.13 | Ansonmus | what do you mean by a key system? |
| 08:49.30 | X-Rob | What you described above is called a key system. With one light per line. Asterisk can't do that. |
| 08:49.32 | X-Rob | Thank you, come again. |
| 08:52.38 | *** join/#asterisk hensema (n=erik@scrat.hensema.net) |
| 08:52.40 | Ansonmus | but, I'm far I can see in the diagnostics which address is called. (Have entered 2 accounts eg. "helpdesk" and "ansonmus") But only the client doesn't show that. |
| 08:52.59 | Op3r | anyone tried IVR on asterisk? |
| 08:53.06 | hensema | hi, does MeetMe require working zap channels? |
| 08:53.14 | X-Rob | hensema, it needs a zap timing source. |
| 08:53.19 | hensema | ah |
| 08:53.27 | X-Rob | ztdummy is fine (in fact, better than a _Real_ timing source) on 2.6 kernels |
| 08:53.46 | X-Rob | Ansonmus, perhaps I don't understand. Are you talking about caller ID? |
| 08:54.06 | hensema | X-Rob: ok, thanks |
| 08:54.19 | batman2 | how can i connect my linksys to my asterisk@home? |
| 08:54.30 | batman2 | do i need to create a sip trunk? |
| 08:54.38 | Ansonmus | X-Rob: sort of but not caller id but callED id (we call it here destination address) |
| 08:55.23 | X-Rob | Ansonmus, not easily. You could prefix the caller ID with something.. You can do that easily with Asterisk@Home |
| 08:55.41 | Ansonmus | we experimenting with aah |
| 08:56.24 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F2DE2.dip0.t-ipconnect.de) |
| 08:56.51 | Ansonmus | In normal situations group addresses like 'sales' and 'support' are they forwarded to persons, or do clients register with their own and a group account? |
| 08:57.01 | X-Rob | you use ring groups for that |
| 08:58.28 | wasim | going medieval on chanspy crashes |
| 08:58.55 | X-Rob | 3 crop rotation? Maypole dances? |
| 08:59.56 | riksta | Op3r: nearly everyone does some form of IVR |
| 09:02.33 | Ansonmus | I've in the Asterisk Management Portal of a@h. I can send incoming calls to a ring group. Can I choose a different ring group on a different PSTN destination address? |
| 09:02.44 | Ansonmus | I've = I am |
| 09:04.39 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 09:07.47 | _medic_ | Ansonmus, if its not important to see where the call is aiming at, before you take it,you can have a voice telling the callee ... |
| 09:08.48 | riksta | b/j #radrails |
| 09:08.50 | riksta | oops |
| 09:11.30 | _medic_ | does by chance anybody know, how to debug a freezing box, when i accept an incoming isdn call ? |
| 09:13.16 | Ansonmus | Can I use inbound routing on incoming calls? |
| 09:14.03 | Ansonmus | So I can use DID to ring different ring groups |
| 09:14.24 | batman2 | i will pay anybody to help me get my Linksys PAP2-NA working with my Asterisk@home |
| 09:16.52 | *** join/#asterisk smeevil (n=smeevil@gateway.office.sod.nl) |
| 09:16.54 | smeevil | hello |
| 09:17.52 | tainted_ | how do i manipulate dialplan based on datetime? |
| 09:20.43 | wasim | oyster*CLI> show application GotoIfTime |
| 09:21.16 | wasim | batman2: GBP2000 / hr or any part thereof |
| 09:23.28 | wasim | http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging |
| 09:23.29 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 09:23.31 | smeevil | when compiling asterisk 1.2.6 i run into an error like : chan_zap.c:10934: warning: passing argument 1 of 'pri_set_error' from incompatible pointer type , any suggestions ? |
| 09:23.43 | tainted_ | anyone use SER w/ asterisk? |
| 09:24.01 | wasim | tainted_: lotsa folks |
| 09:24.28 | tainted_ | i need to scale my asterisk box and want to know how to proceed |
| 09:24.40 | tainted_ | i've heard great things about SER as a SIP proxy |
| 09:24.46 | wasim | batman2: free tip #32 - get rid of aah |
| 09:24.53 | wasim | tainted_: indeed |
| 09:25.08 | wasim | tainted_: http://www.voip-info.org/wiki-Asterisk+at+large |
| 09:25.23 | asterboy | Need some help getting my call forwarding working. |
| 09:25.39 | asterboy | I want * to do the call forwarding instead of my telco. |
| 09:26.09 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 09:26.23 | asterboy | call comes in on ZAP and I want to transfer to another ZAP number from my Polycom phone |
| 09:26.36 | asterboy | help |
| 09:26.58 | asterboy | here is a pasty: http://pastebin.ca/47006 |
| 09:27.31 | hwt | hey, how can i enable internationalized voicemail prompts? |
| 09:27.41 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:27.42 | wasim | hwt: SetLanguage() |
| 09:27.49 | Dandre | hello all, |
| 09:27.50 | hwt | swedish and norwegian grammar isn't like english. |
| 09:28.08 | hwt | wasim: or language=xx in sip.conf, etc., right? those are the same? |
| 09:28.09 | RoyK | hwt: big surprice :) |
| 09:28.11 | wasim | hwt: yah, its a known issue with * ... folks are working on a replacement |
| 09:28.25 | asterboy | holy shit, know one knows anything about call forward on *? |
| 09:28.44 | hwt | wasim: yeah, but swedish and norwegian support appears to have been implemented in upstream *. |
| 09:28.54 | wasim | hwt: ask RoyK |
| 09:28.54 | asterboy | Guess I should ask, Does * do call forwarding? |
| 09:28.57 | smeevil | i'm trying to use asterisk with a bristuff zaphfc....on loading asterisk i get the following error :Mar 30 11:28:24 ERROR[19469]: chan_zap.c:10546 setup_zap: Unknown signalling method 'bri_cpe' |
| 09:29.10 | RoyK | hwt: just bug wasim about it. it's his fault |
| 09:29.12 | hwt | wasim: i already discussed this with him in #asterisk-no :) |
| 09:29.21 | wasim | hwt: he's not very helpful is he ... |
| 09:30.04 | asterboy | I know it can forward using *72, but can it act a forwarding server? |
| 09:30.15 | RoyK | http://karlsbakk.net/fun/wasimlane.gif |
| 09:30.17 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 09:30.51 | hwt | wasim: i'll let RoyK know how this turns out, so that he can relay the information to other people. |
| 09:31.10 | asterboy | hello hello |
| 09:31.14 | Dandre | I have a project to use the asterisk manager API to design a tool that will allow a user to initiate transfer. I want to include some control so that the user should be able to do that only for its extension. Is this possible? |
| 09:34.31 | Ansonmus | When I use CID name prefix in a ring group the name will be replaced by the prefix and not prefixed.. is this a bug of a@h? or do I something wrong |
| 09:40.34 | Ansonmus | where are the ring groups saved? |
| 09:40.37 | Ansonmus | in asterisk? |
| 09:41.16 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 09:41.27 | RoyK | Ansonmus: that's queues |
| 09:42.43 | _medic_ | noone here who can help me with freezing asterisk ? ( I already turned the heater on ) |
| 09:45.02 | Ansonmus | I can't find my defined ring groups |
| 09:49.17 | Dandre | In the wiki, there seem to be poor security for the manager api. (password are even transfered in plain text). Is there some work on enhancing this? |
| 09:51.42 | riksta | Dandre: you can use a hash for authentication, it's documented |
| 09:52.23 | wasim | aww ... i was just gonna tell him to light a fire under it ... |
| 09:52.36 | Dandre | ok I haven't seen. Is there a per extension granularity possibility? |
| 09:52.47 | wasim | Dandre: there is talk of SSL for manager |
| 09:53.05 | riksta | Dandre: http://www.voip-info.org/wiki-Asterisk+manager+API look at the first post on the comments at the bottom |
| 09:54.19 | Dandre | ok fine. |
| 09:54.33 | riksta | Dandre: and read the last comment too (more info about the challenge) |
| 09:55.36 | Dandre | I want to allow my authenticated user to manage only its extension. Is there a tip for that? |
| 09:55.48 | *** join/#asterisk Eggplant (i=No@dsl-745.cascadeaccess.com) |
| 09:58.55 | smeevil | hello, i'm trying to run asterisk build on bristuff 0.3.0.PRE1 , all compiling went fine, but starting asterisk results in the follow error : [chan_zap.so]Mar 30 11:56:17 WARNING[25800]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_suspend_acknowledge , does anyone have an idea what might be the problem |
| 09:58.58 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 10:01.54 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 10:04.47 | *** join/#asterisk saftsack (n=saftsack@p54A7E996.dip.t-dialin.net) |
| 10:13.00 | x86 | morning :P |
| 10:13.28 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 10:15.23 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 10:17.12 | smeevil | found a solution, had to update libpri :) |
| 10:17.25 | *** join/#asterisk blkremedy (n=ur3rdeye@142M28.oasis.mediatti.net) |
| 10:22.38 | xbit` | i have a living connection through a voip service, but there is no voice both ways. maybe the firewall's config not right? asterisk is behind nat |
| 10:23.11 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 10:23.13 | x86 | whoa how do you get a living connection? |
| 10:23.18 | x86 | you mean it eats and stuff? |
| 10:23.25 | x86 | and like, breathes? :P |
| 10:24.05 | xbit` | i'm sure you know ;) |
| 10:28.15 | riksta | xbit`: SIP? |
| 10:28.57 | fourcheeze | xbit`: google for sip and nat |
| 10:29.06 | fourcheeze | there's a tutorial that explains what that probably is |
| 10:29.24 | riksta | xbit`: you need to forward port 5060 and 10,000 to 20,000 (standard) all UDP |
| 10:29.33 | Ansonmus | I've plugged a ISDN line from the NT1 to a BRI card in asterisk. All outgoing calls are not accepted, we get a message from the telco that the number is not recognized |
| 10:30.03 | xbit` | riksta, 2 IPphones |
| 10:30.35 | asterboy | tons of questions getting asked here and not much support going on. |
| 10:31.54 | riksta | it's night time for most people here |
| 10:32.06 | x86 | i was going to say he could pipe in any time and help ;) |
| 10:32.15 | riksta | me too, but i refrained |
| 10:32.32 | *** part/#asterisk guyboertje (n=guy@213-131-125-116.onyx.net) |
| 10:39.58 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 10:41.59 | *** join/#asterisk saftsack (n=saftsack@p54A7E996.dip.t-dialin.net) |
| 10:45.13 | fourcheeze | anyone set up SER as a stateful proxy on the front of * ? |
| 10:45.30 | fourcheeze | if so can they describe how they did it in 100 words or less ;-) |
| 10:47.47 | x86 | fourcheeze: why would you want to do that? |
| 10:47.58 | x86 | fourcheeze: and couldnt you just run a SIP trunk between them? |
| 10:48.05 | fourcheeze | well |
| 10:48.15 | fourcheeze | I'm considering scaling options |
| 10:48.27 | fourcheeze | I'm thinking that if I could get clients to register with SER |
| 10:48.40 | fourcheeze | which proxies registration through to * |
| 10:48.49 | x86 | that seems pointless ;) |
| 10:48.49 | fourcheeze | then SER can handle the inter-sip stuff |
| 10:48.53 | fourcheeze | why? |
| 10:49.04 | x86 | eh i dunno just does ;) |
| 10:49.10 | fourcheeze | what's your scaling solution? |
| 10:49.14 | x86 | why not setup two asterisk boxes ? |
| 10:49.29 | fourcheeze | well, I'll need more than 2 eventually |
| 10:49.41 | fourcheeze | however |
| 10:49.43 | fourcheeze | ok, 2 boxes |
| 10:49.44 | x86 | keep adding them and use DUNDi or E164 |
| 10:49.56 | fourcheeze | ok |
| 10:50.02 | x86 | use MySQL for the backend |
| 10:50.08 | fourcheeze | already doing the mysql thing |
| 10:50.13 | fourcheeze | it's how to route dynamically |
| 10:50.15 | x86 | setup a mysql server on both boxes, and have them cluster |
| 10:50.33 | x86 | share the SIP, IAX, extensions and voicemail databases |
| 10:50.37 | x86 | and you're set |
| 10:50.49 | fourcheeze | ok so user A logged into box A |
| 10:50.54 | fourcheeze | user B logged into B |
| 10:50.57 | fourcheeze | or rather registered with |
| 10:51.00 | fourcheeze | B calls A |
| 10:51.06 | fourcheeze | how does dundi know where A is? |
| 10:51.23 | x86 | isnt that how DUNDi works? |
| 10:51.36 | fourcheeze | yo utell me |
| 10:51.55 | x86 | AFAIK it allows you to port numbers between servers |
| 10:52.00 | fourcheeze | ok |
| 10:52.06 | fourcheeze | but can it do that every few minutes |
| 10:52.10 | fourcheeze | when people reregister? |
| 10:52.29 | x86 | i think it's realtime |
| 10:52.41 | fourcheeze | I've looked for this kind of information but can't find it |
| 10:52.54 | fourcheeze | one thing I've thought of doing is using the sip full contact url |
| 10:53.00 | fourcheeze | which is stored in realtime |
| 10:53.05 | fourcheeze | however that's not always right |
| 10:53.11 | fourcheeze | sometimes it has a private IP range on it |
| 10:53.11 | x86 | http://www.voip-info.org/wiki-Asterisk+DUNDi+Call+Routing |
| 10:53.18 | fourcheeze | yeah, read that one |
| 10:53.56 | fourcheeze | I don't see anything dynamic in there |
| 10:54.15 | x86 | thats the whole point of DUNDi man :) |
| 10:54.57 | x86 | of course, even without dundi, a user will effectively be registered on both servers :) |
| 10:55.09 | x86 | using mysql clustering and realtime |
| 10:56.36 | fourcheeze | how is that any use? |
| 10:56.47 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 10:56.57 | x86 | because you can handle more call volume |
| 10:57.06 | fourcheeze | I mean |
| 10:57.16 | fourcheeze | if the user is registered on both servers |
| 10:57.24 | fourcheeze | (which I'm not sure is really true) |
| 10:57.35 | fourcheeze | how does the call know where to go |
| 10:57.49 | fourcheeze | x86: do you have a setup like this? |
| 10:57.53 | x86 | the idea is to setup DNS in a round-robin |
| 10:58.06 | fourcheeze | yeah, I know about the round robin thing |
| 10:58.29 | x86 | where you have, for example, voip.mydomain.net that points to 10.1.2.3 and 10.4.5.6 |
| 10:58.37 | fourcheeze | yeah |
| 10:58.40 | x86 | a call can hit either one |
| 10:58.45 | fourcheeze | yep |
| 10:58.54 | x86 | since the user is registered at both, it'll make it to them ;) |
| 10:58.59 | fourcheeze | hmm |
| 10:59.06 | fourcheeze | do you have this working in practice? |
| 10:59.14 | x86 | seen it done |
| 10:59.15 | *** join/#asterisk fulgas (n=fulgas@207.226.175.10) |
| 10:59.20 | x86 | havent done it myself yet though |
| 10:59.38 | fourcheeze | I might have to play with it |
| 10:59.45 | xbit` | ok, portforwarding was missing, thanks |
| 11:00.04 | fourcheeze | however I would have the mysql server on a different box entirely |
| 11:00.07 | fourcheeze | and a backup for it |
| 11:00.22 | x86 | 4 servers, 2 mysql, 2 asterisk |
| 11:00.26 | x86 | sounds good :) |
| 11:00.35 | fourcheeze | and then that should scale |
| 11:00.42 | fourcheeze | up to N asterisk |
| 11:00.59 | x86 | so would having mysql on the same box :) |
| 11:01.07 | x86 | but i can understand not wanting to do that |
| 11:01.26 | fourcheeze | well, I think mysql will scale better if given boxes of its own |
| 11:01.29 | x86 | as it would increase load and what not |
| 11:01.31 | x86 | right |
| 11:01.40 | fourcheeze | I single mysql server with high spec can probably serve all these boxes |
| 11:02.06 | wasim | yeah, but 1 is 1 too little |
| 11:02.21 | fourcheeze | you need 1 and a spare |
| 11:02.28 | x86 | err |
| 11:02.36 | x86 | setup a active-active cluster |
| 11:02.43 | x86 | do DNS round robin for mysql too |
| 11:02.52 | fourcheeze | not sure that would work too well if one went down |
| 11:03.06 | fourcheeze | I suppose heartbeat can help |
| 11:03.11 | x86 | err |
| 11:03.14 | x86 | no... |
| 11:03.22 | x86 | look into binary logging with mysql |
| 11:03.23 | fourcheeze | however I would use a single server and then batched updates to the other one |
| 11:03.32 | fourcheeze | more efficient |
| 11:03.33 | x86 | you can do active active and not have to have linux-ha or anything |
| 11:03.46 | fourcheeze | yeah |
| 11:03.49 | fourcheeze | I know about that |
| 11:03.50 | x86 | that's niether more efficient or a pretty solution ;) |
| 11:04.04 | x86 | you want realtime data for realtime applications |
| 11:04.04 | fourcheeze | I'm not sure mysql is quite up to that just yet |
| 11:04.07 | x86 | not batched ;) |
| 11:04.10 | x86 | sure it is |
| 11:04.12 | x86 | i've done it |
| 11:04.14 | fourcheeze | batching is only for the backup |
| 11:04.26 | x86 | ah |
| 11:04.31 | fourcheeze | mysql is quite capable of serving many thousands of queries per second |
| 11:04.36 | wasim | http://www.seriema-systems.com/mysql-ha/ |
| 11:04.36 | fourcheeze | you shouldn't need a cluster |
| 11:04.38 | x86 | so yeah setup a 2 node mysql cluster |
| 11:04.53 | x86 | fourcheeze: redundancy in case one dies? |
| 11:04.53 | [ProB]CrazyMan | question, in asterisk 1.2.X was an chage with the jumping ... how does it work now, just go ahead with next priority insted of jump n+101 ? |
| 11:05.08 | x86 | fourcheeze: you put all your eggs in one basket you'll get fried ;) |
| 11:05.09 | fourcheeze | x86: maybe |
| 11:05.15 | fourcheeze | hence the backup |
| 11:05.26 | x86 | right, which isnt realtime ;) |
| 11:05.34 | x86 | hence my point about a cluster |
| 11:05.36 | fourcheeze | it doesn't matter for these purposes |
| 11:05.46 | fourcheeze | your server shouldn't be going down all the time |
| 11:05.59 | fourcheeze | if you're unlucky it will happen once per year |
| 11:06.00 | x86 | components die over time |
| 11:06.13 | x86 | and every time it happens that's data lost |
| 11:06.21 | fourcheeze | not very much |
| 11:06.27 | x86 | if you're willing to take that risk, do it your way ;) |
| 11:06.28 | x86 | well |
| 11:06.36 | fourcheeze | I'll have a look at both and benchmark |
| 11:06.46 | x86 | if you're to the point of needing dual asterisk servers, you need clustered mysql |
| 11:07.06 | fourcheeze | x86: do you use realtime extensions? |
| 11:07.14 | x86 | yes |
| 11:07.27 | fourcheeze | I'm just wondering about doing that myself |
| 11:07.33 | fourcheeze | or whether I should just use some AGI |
| 11:07.40 | fourcheeze | or odbc calls |
| 11:07.42 | x86 | realtime sippeers, sipfriends, and iax also |
| 11:07.53 | fourcheeze | yeah, I do those |
| 11:07.53 | x86 | odbc == realtime |
| 11:07.56 | fourcheeze | and voicemail users |
| 11:08.01 | fourcheeze | odbc != realtime |
| 11:08.16 | fourcheeze | realtime extensions means that your extensions.conf is replaced by a database |
| 11:08.17 | x86 | odbc is just an accessor to a database, like mysql |
| 11:08.31 | smeevil | could someone please spare some time to help me find why asterisk does not detect any digits that are being given (using softphones) |
| 11:08.56 | fourcheeze | smeevil: check dtmf settings on your softphones and * are the same |
| 11:09.47 | fourcheeze | x86: you can do odbc lookups via a static extensions.conf |
| 11:09.51 | fourcheeze | which one do you use? |
| 11:09.56 | smeevil | fourcheeze, sip.conf has dtmfmode = rfc2833 , the softphones are using that as well |
| 11:10.07 | x86 | fourcheeze: which driver does that? |
| 11:10.27 | fourcheeze | x86: well either use AGI or func_odbc.so |
| 11:10.39 | fourcheeze | however I've not managed to get func_odbc.so so return a value properly yet |
| 11:10.49 | x86 | AGI != ODBC :P |
| 11:11.06 | wasim | although its just as twisted |
| 11:11.13 | fourcheeze | x86: I didn't say it did |
| 11:11.15 | x86 | RoyK: join the club :P |
| 11:11.24 | fourcheeze | quite sunny here |
| 11:14.52 | wasim | and well wishes ofcourse |
| 11:16.35 | smeevil | fourcheeze, any other suggestions ? |
| 11:16.56 | fourcheeze | try setting both to sip info |
| 11:17.01 | fourcheeze | smeevil: ^^ |
| 11:17.28 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 11:17.34 | smeevil | fourcheeze, dtmfmode=info ? |
| 11:17.40 | fourcheeze | yeah, I think so |
| 11:18.08 | x86 | smeevil: dtmfmode=auto works better :) |
| 11:18.13 | x86 | smeevil: 1.2 only |
| 11:19.07 | smeevil | info works :D thank you ! |
| 11:21.21 | smeevil | seems the softphones only work when using the info setting ( even if dtmfmode=auto) |
| 11:22.52 | hwt | i have that dtmf problem with a snom 360. |
| 11:22.57 | hwt | could that be the same? |
| 11:23.17 | wasim | let it snow, let it snow, let it snow ... |
| 11:23.30 | *** join/#asterisk pebiman (i=pebiman@209.205.82.140) |
| 11:23.48 | pebiman | Hello, somebody here? |
| 11:23.55 | wasim | pebiman: nope, just us bots |
| 11:24.11 | wasim | all 320 of 'em |
| 11:24.25 | wasim | correction 319 ... |
| 11:25.38 | wasim | tsk tsk |
| 11:25.42 | wasim | he didn't like that |
| 11:27.08 | zoa | us bots bknow better |
| 11:27.32 | X-Gen | wasim: stop scaring the tourists |
| 11:29.35 | mut | MMMmmmm breadstix for breakfast |
| 11:32.27 | x86 | mmm pizza for breakfast |
| 11:32.31 | x86 | :P |
| 11:32.38 | wasim | mmm snowcones! |
| 11:33.36 | x86 | just dont eat the yellow ones |
| 11:33.40 | x86 | they are special ;) |
| 11:36.26 | *** join/#asterisk Aurs (n=aurs@a217-118-40-143.bluecom.no) |
| 11:38.51 | *** join/#asterisk saftsack (n=saftsack@p54A7E996.dip.t-dialin.net) |
| 11:49.49 | mut | heh |
| 11:49.54 | mut | i have a pizza sub in the fridge here |
| 11:49.57 | mut | thats lunch |
| 11:50.19 | mut | gf's friend last night was like "i'm buying you guys dinner get what ya want" |
| 11:50.20 | hwt | hm, make mpg123 breaks on ubuntu. known problem? |
| 11:50.23 | hwt | or am i missing packages? |
| 11:50.38 | mut | he orders a small pizza, 2 super salads a order of breadsticks and a pizza sub |
| 11:51.00 | mut | i'm like.. holy take advantage of that why don't ya |
| 11:51.01 | hwt | make[3]: *** No rule to make target `\ |
| 11:51.01 | hwt | ', needed by `mpg123'. Stop. |
| 11:51.05 | hwt | not really informative. |
| 11:54.20 | *** part/#asterisk billings (n=billings@pdpc/supporter/active/billings) |
| 11:55.32 | *** join/#asterisk coppice (n=chatzill@184.157.17.210.dyn.pacific.net.hk) |
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| 12:46.41 | cacun | exit |
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| 12:54.13 | Modcuts | would it be a bad idea to choose gsm over g729 |
| 12:55.35 | wasim | Modcuts: yes, if the endpoint only did g729, or if you had less than 26kbps |
| 12:57.41 | *** join/#asterisk batman2 (n=asdfd@ip70-181-90-193.oc.oc.cox.net) |
| 12:58.10 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 12:58.19 | clive- | can you belive I can't figure where the sounds directory is |
| 12:58.32 | wasim | /var/lib/asterisk/sounds |
| 12:58.58 | batman2 | i keep getting Can't connect to login server from my linksys pap2-na |
| 12:59.05 | batman2 | what can i do to solve this? |
| 12:59.21 | xermesx | hi all |
| 12:59.24 | clive- | wasim thanks...I was lookingthere, but my asterisk doesnty seem to be looking there...acky |
| 12:59.27 | clive- | wacky |
| 12:59.32 | wasim | asterisk.conf |
| 12:59.38 | Modcuts | wasim: um.... if my phones support it my voip support it and asterisk supports it when could it fall? And is it about twice as network heavy as g729? |
| 12:59.40 | xermesx | how do i apply language patch say.patch ? |
| 13:00.03 | wasim | Modcuts: no, gsm is 13 kbps, g729 is about 9 |
| 13:00.23 | wasim | Modcuts: but gsm is a hell of lot lighter on the cpu |
| 13:00.26 | batman2 | wasim: i keep getting Can't connect to login server from my linksys pap2-na, hoow can i solve this? |
| 13:00.29 | clive- | asterisk .conf? |
| 13:00.41 | wasim | Modcuts: show translation and http://www.convergence.com.pk/iax2/ for more details |
| 13:00.41 | Modcuts | wasim: surley thats a good thing using less cpu |
| 13:01.13 | Op3r | can anyone point me to how to do IVR on asterisk? |
| 13:01.16 | Modcuts | wasim: that is wicked!!! |
| 13:01.26 | wasim | ~docs |
| 13:01.29 | jbot | i guess 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 |
| 13:02.39 | Modcuts | wasim: thank you that looks very good |
| 13:04.06 | Modcuts | wasim: do you think i would gain much in quality from moving over to gsm? |
| 13:05.13 | *** join/#asterisk taec (n=phil@eventhorizon.hosting365.ie) |
| 13:05.39 | smeevil | could someone please tell me if this is possible : a call is being transferred to someone , his phone rings and in stread of picking it up the person presses 1, 2 or 3 , which each plays a message like , i cannot pickup right now, or i'm allready on the phone and will call you back , or do not disturb .... just as an example |
| 13:06.23 | wasim | smeevil: yes |
| 13:06.24 | taec | Is the queue_log format documented anywhere? I can make sense out of most of it, but just some fields are a bit off |
| 13:06.24 | taec | Sorry, not a bit off, but hard to understand from the data. |
| 13:06.44 | smeevil | wasim, could you possible direct me to some documentation regarding such a setup ? |
| 13:07.42 | wasim | smeevil: the call is transferred to an extension, that extension instead of dialing the person, launches either an agi or someother way to call person in a separate channel, presents a menu and choices, you can pass the original transfered channel as VAR |
| 13:08.18 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F2DE2.dip0.t-ipconnect.de) |
| 13:09.19 | smeevil | wasim, ok, but that would require the user to actually pick up the phone right ? |
| 13:09.36 | smeevil | wasim, i was hoping to use the phone's special buttons to reply to such an action |
| 13:09.45 | wasim | smeevil: oh, right, depends on the phone then |
| 13:10.15 | smeevil | wasim, Grandstream BudgeTone 102 IP Phone |
| 13:11.00 | wasim | smeevil: doubit ... we had that type of functionality in farfon, but alas ... |
| 13:11.59 | wasim | taec: /usr/src/cvs-src/asterisk/doc/queuelog.txt |
| 13:12.50 | wasim | taec: i'm having jdg put something similar, but not quite, in moziax |
| 13:12.56 | wasim | s/taec/smeevil |
| 13:12.58 | smeevil | wasim, hmm, wll see later on then.....i saw something like s-DONTCALL and s-TORTURE in the demo extension....but i cant seem to grasp how it is supposed to get those statusses |
| 13:13.00 | xermesx | how do i apply language patch say.patch ? |
| 13:13.45 | *** join/#asterisk CleanerX (n=nix@p54A3C4D4.dip0.t-ipconnect.de) |
| 13:14.27 | clive- | did something change that asterisk is not looking in the /var/lib/asterisk/sounds directory ? |
| 13:15.23 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:17.03 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 13:18.44 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:20.26 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:21.36 | taec | wasim, thanks a million, that's perfect |
| 13:22.33 | nettie | Hi guys.. I have a couple of polycom phones.. each of them has 3 channels. What's the best way to define them as extensions? I mean, maybe name the phone and then create an extension in extensions.conf and point it to the 3 channels ? any suggestion please? |
| 13:24.34 | *** join/#asterisk coppice (n=chatzill@141.197.17.210.dyn.pacific.net.hk) |
| 13:26.06 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 13:26.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:27.00 | PoWeRKiLL | hi |
| 13:28.42 | caio1982 | coppice: hello steve... want some more unicall/r2 problems like "whoa, i'm owned but found (26) in read [codes from the call]" ? everything looks fine now, but sometimes it stop with protocol failures, sometimes it works, but with this message |
| 13:29.01 | *** join/#asterisk Mauro__ (n=mauro@oliver.altascumbres.cl) |
| 13:29.20 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F2DE2.dip0.t-ipconnect.de) |
| 13:29.22 | caio1982 | coppice: it used to complain about the cas bit pattern (which seems to be fine) |
| 13:30.48 | coppice | only a couple of people have reported that "whoa, i'm owned" thing. I assume it is something specific about their setup, but I don't know what |
| 13:31.10 | hwt | hmm, my snom360 won't transmit DTMF in voicemail menu. |
| 13:31.25 | hwt | i have tried dtmfmode = auto, info, inbound and rfc. |
| 13:32.05 | wasim | whats the snom set to? |
| 13:32.11 | caio1982 | coppice: worth a pastebin to check the log from a successful call and one from a failed? i suspect about our setup too, but i have to prove that hehe |
| 13:32.35 | coppice | ok |
| 13:32.43 | Mauro__ | can I get the status of a responded call with asterisk? |
| 13:33.11 | Mauro__ | if it s a fax, answering machine, a human |
| 13:33.29 | [TK]D-Fender | nettie : Consider each phone a single "extension" and assign all 3 line keys to that registration. Subsequent calls will "roll-over" to the next available line key when you're on a call. |
| 13:34.29 | hwt | wasim: rfc2833. |
| 13:35.40 | nettie | [TK]D-Fender so I need to define it just once in sip.cong and replicate the auth informations on each phone channel? |
| 13:35.52 | caio1982 | coppice: that's the failed call log: http://pastebin.com/630713 |
| 13:36.05 | rbd | hey guys, I was looking info asterisk (SIP) load balancing... I have a trunked connection coming in from a sip provider to a group of asterisk machines (this is incoming calls only)... I'd like to be able to round robin across these asterisk boxes. I saw that vovida had a sip proxy but it looks dead...other possibilities may be LVS or DNS load balancing?? any ideas? |
| 13:36.20 | wasim | hwt: and dtmfmode=rfc2833 doesn't work? did ya reload? |
| 13:36.28 | wasim | hwt: what does * console or sip debug show? |
| 13:37.14 | wasim | rbd: http://www.voip-info.org/wiki-Asterisk+at+large |
| 13:40.26 | clive- | wasim there is no place in asterisk.conf for a sound file location directory :( |
| 13:40.37 | rbd | wasim, ahh, thanks |
| 13:40.52 | *** join/#asterisk pengyong (n=lala@222.185.18.93) |
| 13:41.07 | hwt | wasim: hm, i rewrote the dialplan on the snom, and now it appears to be working. |
| 13:41.36 | clive- | very strange.....hmmm |
| 13:41.49 | wasim | indeed ... eerie |
| 13:42.10 | clive- | eerie..lol..well does this give any clues: file.c:509 ast_openstream_full: File vm-toenternumber.gsm does not exist in any format |
| 13:42.32 | PoWeRKiLL | it is possible to do Dial(number@context) instead of doing Goto(context,number) ? |
| 13:42.35 | [TK]D-Fender | nettie : No, you don't duplicate for each "line". You define 1 registration using 3 line keys, with 1 call per line key |
| 13:42.49 | coppice | 24000 quad core opterons - I guess that could transcode quite a bit of G.729 :-) |
| 13:43.12 | clive- | coppice what are you doing with 24000 opertons?....:) |
| 13:43.17 | Mauro__ | can I get the status of a responded call with asterisk, if it s a fax, answering machine, a human? I m dialling outgoing calls with asterisk |
| 13:43.21 | wasim | toast a few marshmallows? |
| 13:43.37 | wasim | Mauro__: NVDetect |
| 13:43.42 | Mauro__ | thanks |
| 13:43.43 | Mauro__ | :D |
| 13:43.49 | coppice | cray are going to use 24000 quad core opterons to make a PC for the US government |
| 13:43.50 | nettie | [TK]D-Fender you can do it in the phone? |
| 13:44.17 | clive- | wow, who is supplying the power station next door for it |
| 13:44.20 | nettie | [TK]D-Fender because at the moment I call them poly1-1 poly1-2 poly1-3 and so on.. poly2-1 |
| 13:44.29 | [TK]D-Fender | nettie : what do you mean "in the phone"? |
| 13:44.43 | [TK]D-Fender | nettie : remove the other 2 line definitions. are you web-configuring it? |
| 13:45.21 | nettie | [TK]D-Fender in sip.conf I have poly1-1 poly1-2 and poly1-3. I replicate this in line1 line2 and line3 on the phone. |
| 13:45.25 | nettie | yes I'm webconfign the, |
| 13:45.27 | nettie | them |
| 13:45.57 | nettie | you're suggesting me to define only poly1 in line1 and the others will be usable automatically? they'll rollover and so on? |
| 13:45.58 | [TK]D-Fender | you should only define the phone once. You don't want 3 distict registrations on it. |
| 13:46.07 | nettie | ok |
| 13:46.10 | nettie | I'll do it now |
| 13:46.12 | [TK]D-Fender | nettie : Yes, thats what I suggest. |
| 13:46.34 | [TK]D-Fender | nettie : remember = 1 reg, 3 line keys, 1 call / line key |
| 13:46.59 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:48.06 | nettie | [TK]D-Fender sounds fine for me .. the question is.. does the phone automatically uses a single reg for 3 lines or I have to specify that somewhere? and when 2 regs are useful? when I need to register the phone in a different context? |
| 13:49.15 | [TK]D-Fender | nettie : you need to specify it. |
| 13:49.49 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:49.55 | [TK]D-Fender | nettie : 2 regs are good when your phone is registering to 2 seperate servers or you are sharing the phone with someone else and have the other reg ring differentl, etc. |
| 13:54.59 | PoWeRKiLL | it's seems to me that Goto(context,number) then do a Dial is working better than Dial(Local/number@context) ? |
| 13:55.15 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 13:56.47 | Katty | mew. |
| 13:57.30 | puzzled | hi all |
| 13:57.44 | coppice | caio1982: are you using spandsp-0.0.2pre25? |
| 13:57.48 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 13:58.32 | caio1982 | coppice: yes |
| 13:59.10 | caio1982 | coppice: we're looking the r2 signalling documentation here but without any success so far |
| 13:59.47 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 14:00.01 | *** join/#asterisk b66mer (i=fwuser@blackhole.c5i.com) |
| 14:00.52 | coppice | ah. that's the problem. I screwed up some updates. i thought I had made a change to the mfc/r2 code some time ago, and i changed something related to this in spandsp-0.0.2pre25 thinking the other half of the change in mfc/r2 was out in everyone's systems already. I can give you the change for mfcr2.c, or you can use spandsp-0.0.2pre23 |
| 14:01.55 | caio1982 | coppice: what do you think is the right choice for now? i can test both, but i thought the newer spandsp release was "better" to use |
| 14:02.05 | clive- | coppice, any idea why I get this message....:file.c:509 ast_openstream_full: File vm-toenternumber.gsm does not exist in any format |
| 14:02.41 | coppice | caio1982: the change to mfcr2.c is only a few lines. |
| 14:02.57 | MikeJ[Laptop] | clive-, it's probably to do with: File vm-toenternumber.gsm does not exist in any format |
| 14:03.14 | caio1982 | coppice: ok, can you pastebin them? |
| 14:03.20 | clive- | mike:), lol. yup, but that file isthere and does exsit, thats the wierd thing |
| 14:03.36 | MikeJ[Laptop] | it's not |
| 14:04.02 | MikeJ[Laptop] | or you don't have any formats loaded maybe |
| 14:04.13 | wasim | heh |
| 14:04.22 | clive- | Mike, its in: /var/lib/asterisk/sounds/vm-toenternumber.gsm |
| 14:04.32 | wasim | clive-: read perms? |
| 14:05.07 | nettie | [TK]D-Fender where shoul I specify it ? in the webconf? |
| 14:05.16 | wasim | clive-: do a s,1,Playback(vm-toenternumber.gsm) |
| 14:05.22 | wasim | err, withouth the .gsm bit |
| 14:05.24 | *** join/#asterisk doolph (n=doolph@200.75.208.199) |
| 14:05.29 | nettie | [TK]D-Fender under num per line keys? |
| 14:05.32 | doolph | anyone can help me with sipura 3000 ? |
| 14:06.05 | clive- | wasmin, lets try |
| 14:06.46 | Splat | anyone happen to have used sipdiscount at all? |
| 14:07.59 | coppice | caio1982: i just created 0.0.3pre9 which has only this change making it different from 0.0.3pre8 |
| 14:08.50 | caio1982 | coppice: thanks, i'll take a look at it then i'll report to you later |
| 14:09.25 | [TK]D-Fender | Katty: mew. |
| 14:09.36 | [TK]D-Fender | nettie : yes |
| 14:12.44 | clive- | wasim, my hero! |
| 14:12.47 | clive- | its works |
| 14:14.28 | nettie | [TK]D-Fender so basically I need to add 3 to: calls per line key and to num lines per key |
| 14:15.00 | nettie | uhmm |
| 14:15.07 | nettie | didnt work.. phones are white |
| 14:15.09 | nettie | all 3 phones |
| 14:15.18 | macTijn | hmm |
| 14:15.24 | nettie | maybe somethin wrong in the config |
| 14:15.43 | macTijn | can anyone tell me what type of signalling is used on belgium pots lines ? |
| 14:16.59 | nettie | forgot to reload maybe ;p |
| 14:17.04 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 14:17.05 | puzzled | macTijn: in case of isdn should be euroisdn |
| 14:17.10 | *** join/#asterisk scrambray8927 (n=scrambra@12.104.121.147) |
| 14:17.13 | wasim | clive-: i'm confused now |
| 14:17.21 | wasim | clive-: where were you getting the original error? |
| 14:17.25 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 14:17.37 | wasim | macTijn: prolly loopstart |
| 14:17.39 | scrambray8927 | Anyone have any experience with signal processing or software based modems? |
| 14:17.51 | nettie | nice |
| 14:17.59 | nettie | works |
| 14:18.32 | coppice | scrambray8927: nope. nobody here has any experience like that |
| 14:18.40 | scrambray8927 | I'm looking for a software based approach to determine signals heard on a call via on my asterisk box (such as busy, fax, answering machine, carreir, etc...) |
| 14:18.55 | scrambray8927 | coppice can you point me in the right direction? |
| 14:18.58 | [TK]D-Fender | nettie : if the phone is "white" then you didn' |
| 14:19.09 | [TK]D-Fender | t enter the reg info properly for the remaining reg. |
| 14:19.25 | [TK]D-Fender | nettie : it should be 3 line keys, *1* call per line key. |
| 14:20.07 | nettie | uhmm |
| 14:20.08 | clive- | wasim, i shouldnt have included the .gsm |
| 14:20.15 | nettie | what 3 calls per line does? |
| 14:20.58 | wasim | clive-: ugh ... sorry, should have caught that earlier when is said blah.gsm not found in any format, it should have said blah not found in any format |
| 14:21.16 | lepine | could someone help me out with querying a SIP server ? |
| 14:21.25 | lepine | what i have working so far : http://pastebin.com/630779 |
| 14:21.28 | coppice | scrambray8927: * does some of those things, though not always very well. answering machine detection never works. that's just a waste of time |
| 14:22.01 | macTijn | puzzled: POTS, no ISDN |
| 14:22.05 | trelane_ | and just remember if language x is so great why's it written in C? |
| 14:22.12 | trelane_ | eek mischan |
| 14:22.14 | macTijn | Wasim: I was guessing loopstart indeed, just need to know for sure |
| 14:22.15 | trelane_ | sorry :( |
| 14:22.29 | [TK]D-Fender | nettie : you don't want that. that will mean that each line key will have a 3 level call-waiting on it before scrolling to the next key (which is the "natural" way) |
| 14:22.49 | lepine | could someone point me to docs listing the commands you gan give to get certina information back ? |
| 14:23.00 | scrambray8927 | coppice: you say it does some of the things i mentions, could you provide me with a link in the wiki or a term to look up in the * documentation that would allow me to read about *'s current support of these features? |
| 14:23.01 | lepine | missed calls for example |
| 14:23.06 | caio1982 | coppice: awesome, everything went fine, even multiple calls at the same time (which was taking 100% of my cpu slices)... thanks once again (i shall diff -ruN the changes now hehe) |
| 14:23.44 | nettie | btw: I have a problem with music on hold.. when I call a user trought my voip provider, the sound of stop and start contnuously. I noticed that when the user make some kind of noise he hear the sound perfectly when he stop talking the sound drops continuosly. Do you think it's dependant on my voip carrier codec? |
| 14:24.00 | nettie | [TK]D-Fender thanx a lot I got it.. |
| 14:24.12 | nettie | [TK]D-Fender damn you know a lot about polycoms ehehe |
| 14:24.56 | scrambray8927 | caio1982: what are the specs on your * box? |
| 14:25.01 | [TK]D-Fender | nettie : Got to be good at something I guess :) I do strongly advise that you download the admin guide and lear how to provision them throughthe XML files though... |
| 14:25.37 | caio1982 | scrambray8927: it's just a test machine, what specs do you want to know? you probably owns a better box |
| 14:26.41 | scrambray8927 | i have a 3ghz dual core intel, 2 gig ram, and a 1.5mb/1.5mb down/up modem. |
| 14:27.38 | scrambray8927 | i'm interested in war dialing my own lines about once every month or two to inventory my pstn |
| 14:27.53 | nettie | [TK]D-Fender yeah you're definitely good ehehe :) uhmm provisioning via xlm look sinteresting.. I'm downloading thr guide right now.. :) |
| 14:28.05 | scrambray8927 | make sure there are no modems hiding in the wings that are letting people in |
| 14:29.28 | scrambray8927 | so i'm curious about signal processing or a software based modem that will allow me to interperate the tones that asterisk hears when it calls all of these lines. In addition I wanted to have asterisk initiate more than 1 call at a time (I'm assuming i could have about 20 simultaneiously with the specs i gave above but I wanted to make sure). |
| 14:30.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:31.22 | nettie | [TK]D-Fender anyway the best way is to name the phones and then specify the extension in extensions.conf and point the phone ny registration name right? |
| 14:32.59 | rbd | as an extension to a previous question I had...given a SIP trunk with incoming calls...I was looking at a way to round robin these calls across set of unconnected (unclustered) asterisk boxes (the incoming call is for an ivr, no specific users to reach)....it looks like something as simple as round robin DNS or a SER redirect server setup may work instead of a full fledged SER proxy...basically incoming call 1 goes to asterisk ivr server |
| 14:33.00 | rbd | <PROTECTED> |
| 14:33.09 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 14:34.41 | _DAW | Hello. I am having a problem with hdlc on a TE110P. I get WARNING: Error inserting zaptel (/lib/modules/2.6.9-34.EL/extra/zaptel.ko): Unknown symbol in module but only after uncommenting define config_zapata_net. Any suggestions? |
| 14:35.09 | Katty | iDunno: bouncer. |
| 14:35.19 | iDunno | Katty: uh huh :) |
| 14:36.28 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 14:36.38 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 14:37.25 | MattH | Greetings all... perhaps someone can help me here. I am having an aweful time with queues/agents. I have my queues setup... and agents... but for some reason asterisk is letting me log in with agents that dont' exist in the agents.conf file, and not prompting me for passwords, and even though a show queues will show the agent logged into the queue show agents shows they aren't logged in!! |
| 14:37.29 | MattH | running 1.2.5 |
| 14:40.19 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 14:40.32 | *** join/#asterisk apardo (n=apardo@87.218.44.118) |
| 14:43.07 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:44.51 | [TK]D-Fender | nettie : Not sure what you mean there..... |
| 14:45.46 | *** join/#asterisk Denmark (n=fake@62.242.24.182) |
| 14:46.36 | zoa | almost none |
| 14:46.50 | Denmark | whats digium? |
| 14:47.06 | macTijn | ... |
| 14:47.07 | coppice | i often wonder how many of the people here work at all :-) |
| 14:47.13 | fenlander | digium - it's some kind of new dance I think |
| 14:47.14 | macTijn | hey coppice :) |
| 14:47.21 | Denmark | olli: Dunno, lets ask him. |
| 14:47.55 | wasim | left, right, right, forward, left, back, left, left, right, backward, bow ... repeat ... |
| 14:48.28 | brad_mssw | coppice: oh, we all work ... or at least we make it seem like we do :) |
| 14:49.13 | coppice | wasim: the GW Bush dance goes goes right, right, right, right, right.... |
| 14:49.42 | caio1982 | hah |
| 14:49.50 | Denmark | coppice : I thought it was up up up |
| 14:50.14 | Denmark | Ys |
| 14:50.43 | brad_mssw | as long as the number of rights he makes isn't divisible by 3 ... he's happy (uhh, 3 rights makes a left, no?) |
| 14:51.08 | scrambray8927 | coppice: can you elaborate on *s ability to determine tones on a call? |
| 14:51.21 | Denmark | Lets discuss politics instead of phones; much easier to start a fight. :-) |
| 14:51.42 | coppice | we start fights about phones pretty easily |
| 14:51.55 | Denmark | how? |
| 14:52.01 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
| 14:52.04 | Denmark | "My phone is bigger than your phone!" ? |
| 14:52.29 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-217.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 14:52.51 | Denmark | uriahhee2 : he said "when" httpd is suff. |
| 14:53.46 | wasim | i wonder if 24000 quad-opts will handle my mgcp ua without crashing |
| 14:54.02 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 14:54.12 | coppice | wasim: nah, not if you have more than 2 E1s |
| 14:55.08 | wasim | hmm |
| 14:55.56 | Denmark | Btw. do you have an oppion about sippura? |
| 14:56.35 | Denmark | I wonder what the difference between 841 and 941 is .. and if any other brand is compareable - qualitywise. |
| 14:57.02 | Denmark | (and/or pricewise) |
| 14:57.15 | coppice | 941 - 841 = 100 |
| 14:57.35 | Denmark | coppice : You're joking? |
| 14:57.49 | [TK]D-Fender | Denmark : 941 is a Cisco-ized 841. Its "not bad", but I'd suggest Polycom over it any day. |
| 14:58.17 | [TK]D-Fender | Denmark : No, coppice's math skills are UNDENIABLE.... |
| 14:58.30 | Denmark | So if I can get hold of 841, I am better off? |
| 14:58.55 | Denmark | (I noticed that 941 requires license to use the last 2 lines, which strikes me as greedy) |
| 14:59.01 | [TK]D-Fender | Denmark : I'd suggest a Polycom IP 501 over either of those... |
| 14:59.16 | Denmark | tkd: It supports SIP? |
| 14:59.24 | fenlander | the 841 needs a licence as well iirc |
| 14:59.26 | [TK]D-Fender | Denmark : In order to use them like for seperate registrations, yes. |
| 14:59.31 | [TK]D-Fender | Denmark : Naturally |
| 14:59.35 | *** join/#asterisk Tili (i=Tili@218.19.64.138) |
| 14:59.38 | macTijn | does anyone have pointers on what the belgian national dial plan looks like ? |
| 15:00.12 | Denmark | [TK]D-Fender: Is it much more expensive or in the same prince range? |
| 15:00.36 | zoa | mactijn |
| 15:00.40 | zoa | what do you need to know ? |
| 15:01.32 | Denmark | http://www.polycom.com/products_services/1,1443,pw-34-182-10533,00.html |
| 15:01.51 | zoa | i think trixter has an international list on his website |
| 15:02.09 | macTijn | zoa: everything |
| 15:02.10 | macTijn | :) |
| 15:02.15 | macTijn | where's that ? |
| 15:03.01 | macTijn | I need number lengths, area code + lengths, special prefixes like 1-8xx and 1-9xx |
| 15:03.13 | [TK]D-Fender | Denmark : The IP 501 is about 15% more expensive than the 941/ |
| 15:03.17 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:03.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:03.27 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:03.35 | Denmark | [TK]D-Fender : Doesn't look bad at first glance |
| 15:03.44 | zoa | im looking for it |
| 15:03.46 | zoa | cant find it |
| 15:03.51 | macTijn | hmm |
| 15:03.53 | zoa | how is that called again ? |
| 15:03.57 | zoa | aaah |
| 15:04.04 | macTijn | ?! |
| 15:04.23 | Denmark | [TK]D-Fender : Whats the main advantages about this model? |
| 15:04.23 | zoa | http://www.0xdecafbad.com/uploads/bigcountry.csv.gz |
| 15:04.26 | zoa | have fun |
| 15:04.27 | zoa | :) |
| 15:05.05 | macTijn | nice domain name ;) |
| 15:05.09 | macTijn | cool, thanks :) |
| 15:05.11 | zoa | got a list of 579,887 different numbers around the world. |
| 15:05.22 | zoa | thats why i cannot remember it :) |
| 15:05.26 | macTijn | oh, just 41 mb :) |
| 15:05.39 | macTijn | let's see how hard gedit will crash :) |
| 15:05.52 | zoa | but there is number portability in belgium |
| 15:06.01 | zoa | and some destinations might cost more than others |
| 15:06.57 | Denmark | (I like the design .. though the sippura design is also nice) |
| 15:07.18 | [TK]D-Fender | Denmark : Superior feel / sound quality, 2nd ethernet port, better call control, better use of LCD for conveying information / controls, more configureable. |
| 15:08.28 | Denmark | [TK]D-Fender : I seem to have misplaced my pdf-reader .. whats the 2nd ethernet port for? |
| 15:09.18 | [TK]D-Fender | Denmark : in case you don't have extra jacks where you need them, you can plug it in-line with your computer, etc. |
| 15:10.09 | [TK]D-Fender | Denmark : my company misplanned its customer service department and I only had 1 jack for 4 people so I daisy-chained them all together. |
| 15:10.23 | Denmark | ahh .. kinda like a one-port switch. Neat. |
| 15:10.41 | Katty | i like Danes. |
| 15:10.58 | Katty | also! |
| 15:11.03 | Denmark | Katty : :-) |
| 15:11.13 | Katty | is there any neat way of, when a call comes in, to make a shiny little script come up on a computer? |
| 15:11.19 | Katty | like when line 123-123-1234 rings... |
| 15:11.35 | Katty | up pops, "thank you for call alaska blahblahblahblah foobarlblahbalh kthxbi" |
| 15:11.54 | Katty | i can make windows machines have a little popup thingy, cause of the windows messenger service |
| 15:12.08 | Katty | but i don't know how big of a message i can get in teh leetle window. :< |
| 15:12.19 | [TK]D-Fender | Katty : in your dial-plan do a System call that will issue an IM message from CLI. |
| 15:12.35 | Denmark | Katty : Thats always the problem, isn't it? ;-) |
| 15:12.36 | Hmmhesays | indeed |
| 15:13.23 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 15:13.46 | [TK]D-Fender | Katty : Another way is to make a small script on the * box that will update a file with the content of the message. On the client side have them load a web-page with a JS auto-reload that will steal focus and resize for the message |
| 15:14.14 | macTijn | zoa: thanks, you have been a *great* help |
| 15:14.39 | zoa | np |
| 15:14.57 | macTijn | <- off to the pub |
| 15:15.20 | Denmark | macTijn : Lucky you. |
| 15:15.28 | iDunno | bah! |
| 15:17.35 | vgster | does any use CONFIG_ZAPTEL_MMX with x86_64 cpus? |
| 15:17.48 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:18.24 | Denmark | [TK]D-Fender : Seems it has one more line than the sippura also. (Unless you buy the license or find some other way to "open" the phone) |
| 15:20.51 | trixter | zoa i think trixter has an international list on his website -- astbill.com hosts that :) |
| 15:21.23 | trixter | I havent posted the additions I got after they took it, and what is on my site is older than what I gave them iirc.. I really dont know its out of date either way |
| 15:21.35 | *** join/#asterisk TheoC (n=theochao@68-191-219-240.dhcp.dntn.tx.charter.com) |
| 15:21.52 | Katty | [TK]D-Fender: hmm, that's a neat idea. |
| 15:21.55 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:21.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:22.01 | Katty | anthm: (= |
| 15:22.15 | Katty | [TK]D-Fender: i'm definately liking that idea. |
| 15:22.19 | anthm | hi |
| 15:22.29 | anthm | yay |
| 15:22.31 | trixter | hi katty |
| 15:22.31 | [TK]D-Fender | Katty : I was going to do that for screen-pops for my CSR's during Queue calls based on IVR input. |
| 15:22.34 | Katty | hey trixter |
| 15:22.49 | Katty | [TK]D-Fender: is there anyway to make it RealTime |
| 15:22.50 | [TK]D-Fender | Katty : fairly easy to do, even for me.... |
| 15:23.06 | TheoC | Does anyone know if you can connect via ssh to an * box that only has a pri line going into it? |
| 15:23.33 | Katty | [TK]D-Fender: or is javascript realtime? i'm thinking about a meta refresh |
| 15:23.40 | `Sauron | theoC: No. |
| 15:23.45 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 15:23.50 | `Sauron | Which actually means "Depends." |
| 15:23.53 | [TK]D-Fender | Katty : make a small Java app to receive a TCP connection immediately from the dial-plan to "push" the screen. Also not too hard, but definately beyond me (but not my ability to rip off other people's code :D) |
| 15:24.03 | Katty | one small problem. |
| 15:24.06 | Katty | i don't know java. |
| 15:24.11 | TheoC | `Sauron: depends on what? |
| 15:24.17 | [TK]D-Fender | Katty : hence the "ripping off" part! |
| 15:24.28 | Katty | sounds complicated. |
| 15:24.33 | santoshr | how to implement callwaiting |
| 15:24.36 | Katty | i like this text idea with meta refresh a leetle better. |
| 15:24.41 | [TK]D-Fender | Katty : www.hotscripts.com .... I'm sure there are snippets in places like that that would serve as the base... |
| 15:25.03 | Katty | probably. |
| 15:25.58 | wasim | TheoC: if that PRI has IP on it, why not? |
| 15:26.21 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 15:26.23 | [TK]D-Fender | Katty : JS auto-reload off a PHP page is the easiest. The "polling" page would reload itself every 5 seconds lets say and in PHP if there is new content, it would include a URL to "pop-up" and steal focus. the monitoring page would never need focus and would have to remain open at all times to check for them. |
| 15:26.46 | *** join/#asterisk _medic_ (i=medic@p54B1A662.dip0.t-ipconnect.de) |
| 15:27.27 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:27.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:27.36 | X-Gen | Katty: y dont u like the sending an IM to the user idea ? |
| 15:27.57 | Katty | X-Gen: why do you care? |
| 15:28.14 | X-Gen | im just a caring kinda guy |
| 15:28.19 | TheoC | If the PRI line is going into a digium card - will that accept ssh connections? |
| 15:28.21 | X-Gen | :) |
| 15:28.30 | _medic_ | good evening |
| 15:28.37 | santoshr | how can call waiting be implemented with flash. |
| 15:30.03 | eric_hill | TheoC: SSH needs TCP/IP to run. A PRI line is simply a method of channelizing a T1/E1 connection; it doesn't use TCP. |
| 15:30.58 | santoshr | how can call waiting be implemented with flash |
| 15:31.39 | trelane_ | santoshr, what kind of phone do you have? |
| 15:32.00 | santoshr | its a normal phone not a two line or anything |
| 15:32.07 | trelane_ | how is it terminated? |
| 15:32.08 | Katty | X-Gen: you don't care. |
| 15:32.11 | Katty | X-Gen: don't lie to me. |
| 15:32.16 | santoshr | terminated ? |
| 15:32.19 | `Sauron | Mew |
| 15:32.22 | TheoC | eric_hill: If that's the case - how do you access the * box remotely? With a seperate broadband connection? |
| 15:32.23 | Katty | mew. |
| 15:32.34 | Katty | X-Gen: you're probably just looking for an arguement. |
| 15:32.44 | trelane_ | santoshr, iaxy? zap? channel bank? sangoma? |
| 15:32.59 | wunderkin | terminate: (v) to kill |
| 15:33.13 | trelane_ | wunderkin, terminate: (v) to end |
| 15:33.22 | trelane_ | kthx |
| 15:33.36 | santoshr | through sip fxs device |
| 15:33.38 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:33.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:33.44 | wunderkin | i like kill better |
| 15:33.45 | Denmark | monty python? |
| 15:33.54 | Katty | Denmark: you are right out! |
| 15:34.01 | Katty | anthm: and you appear to need duct tape. |
| 15:34.11 | _medic_ | I suffer a serious asterisk problem, which i cannot debug,and i couldn't find any help in the web .. I can SIP without a problem, but approx 5 - 10 seconds after i took a CAPI-Call offhook (I dial from PSTN into the BRI, andtransmitit to a SIP client on my workstation), the whole box freezes :-( it happened withasterisk 1.2.4 as well with 1.1.x |
| 15:34.38 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:34.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:34.40 | wunderkin | i missed |
| 15:34.50 | Denmark | wunderkin : A lot .. |
| 15:35.20 | santoshr | trelane thr are sip fxs devices and they are called directly dial(sip/ ) |
| 15:35.40 | Katty | wunderkin: :< |
| 15:36.01 | Hmmhesays | i enjoy duct tape |
| 15:36.04 | trelane_ | santoshr, you're going to have to find out if that device supports 3 way calling, and if so asterisk should support it |
| 15:36.11 | Hmmhesays | well I figured out how i'm going to use lcdial, it's pretty sweet |
| 15:36.24 | Denmark | [TK]D-Fender : They seem to have a 601 also, and 301 .. etc. etc. |
| 15:36.26 | trelane_ | santoshr, my iaxy spports 3 way calling |
| 15:36.40 | santoshr | ok... |
| 15:36.51 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-186.buckeyecom.net) |
| 15:37.47 | Katty | Hmmhesays: oh? |
| 15:37.50 | gambolputty | Hi. Is there a way in * to log the debug console lines of individual calls to individual log files instead of one big giant log file like is currently done? |
| 15:37.53 | Hmmhesays | I want to kill however decided 7 digit dialing is a good idea though |
| 15:37.55 | Katty | Hmmhesays: i'm more curious about that duct tape comment. |
| 15:37.58 | Hmmhesays | *whoever |
| 15:38.21 | iDunno | what's wrong with 7 digit dialling? |
| 15:39.07 | Denmark | I dunno? |
| 15:39.11 | Hmmhesays | because when I have terminating gateways spread out in different states and the users dials 7 digits, but the local terminating gateway is full... I have to go and figure out what areacode i'm supposed to dial to get to it from a different state |
| 15:39.23 | wunderkin | Denmark likes it rough |
| 15:39.34 | Hmmhesays | YOU SMELL WHAT THE ROCK IS COOKING |
| 15:39.50 | Katty | Hmmhesays: simmer down, boy. |
| 15:39.57 | Denmark | wunderkin : If you spare the cane, you spoil the child. |
| 15:39.58 | [TK]D-Fender | Denmark : they do, its just that I was suggesting something comperable. The IP 301 is a great budget phone. The IP 601 is mostly for receptionists, etc. |
| 15:40.13 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 15:40.42 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 15:40.44 | Denmark | [TK]D-Fender : So, the IP 301 lacks features... but the sound quality is still comparable to the sippura? |
| 15:41.07 | [TK]D-Fender | Denmark : Better than Sipura. The only real feature it lacks is speakerphone. |
| 15:41.35 | Denmark | Oh .. there is no loudspeaker? |
| 15:41.47 | [TK]D-Fender | Denmark : No speakerphone in the 301 |
| 15:42.01 | [TK]D-Fender | thats why I suggested the 501. best mid-range phone. |
| 15:42.17 | Denmark | speakerphone means you can make the room listen in on the convo, right? |
| 15:43.28 | _medic_ | noone here who can help me with freezing asterisk ? ( I already tried turning the heater on ) |
| 15:43.43 | Denmark | heh :) |
| 15:43.46 | [TK]D-Fender | Denmark : Not just listen, but speak as well. The 301 has a speaker, but no mic. so if you only want everyone to HEAR the call, then yes. |
| 15:43.58 | Hmmhesays | anyone use cmd mysql? |
| 15:44.19 | Denmark | [TK]D-Fender : Ahh .. so you could have the shout to the receiver? |
| 15:44.36 | Denmark | s/the/them/ |
| 15:44.41 | [TK]D-Fender | Denmark : Something like that.... so again, thats why i suggested the 501. |
| 15:44.57 | RaYmAn-Bx | Hmmhesays: yep |
| 15:45.19 | Denmark | [TK]D-Fender : if the price difference is violent, I might consider the 301. |
| 15:45.33 | RaYmAn-Bx | Hmmhesays: Not for anything particular fancy though. (just speeddial lookup) |
| 15:45.45 | [TK]D-Fender | Denmark : depending where you are there is a bit of a difference. |
| 15:45.59 | Denmark | [TK]D-Fender : I am in Denmark. :-) |
| 15:46.02 | Katty | Hmmhesays: you have any Jamiroquai? |
| 15:46.08 | [TK]D-Fender | Denmark : best avg price here is $115 for the 301, $170 for the 501. |
| 15:46.11 | Denmark | I am not sure I can get it at all. |
| 15:46.12 | Hmmhesays | Katty: not a fan |
| 15:46.14 | Katty | kk |
| 15:46.20 | Hmmhesays | RaYmAn-Bx any problems? |
| 15:46.26 | Katty | [TK]D-Fender: do you have any jamiroquai? |
| 15:46.27 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 15:46.40 | [TK]D-Fender | Katty : Trip-hop free here..... |
| 15:46.47 | Katty | [TK]D-Fender: mew? |
| 15:46.50 | Denmark | [TK]D-Fender : The 501 seems to hit the price of the sippura 941 here. |
| 15:46.50 | Katty | [TK]D-Fender: that does not parse. |
| 15:46.58 | Hmmhesays | i really dig this new weezer tune |
| 15:46.58 | RaYmAn-Bx | Hmmhesays: no, works fine..admittedly I haven't upgraded my asterisk in quite a while but still.. |
| 15:47.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:47.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:47.07 | hackeron | Hey, can someone recommend a phone that works really well with asterisk? -- like the voicemail LED, pickup groups whre you can see how you're picking the phone up for, etc? |
| 15:47.22 | Hmmhesays | I may have to use it if I have to allow 7 digit dialing on this network |
| 15:47.45 | Denmark | hackeron : Maybe the 501? I havn't read the specs yet.. |
| 15:47.45 | [TK]D-Fender | Katty : I don't do = dance, hip-hop, rap, trance, house, fusion, jazz, big-band, and a few others..... I'm a rocker, through and through. |
| 15:47.58 | RaYmAn-Bx | Hmmhesays: the only real annoyance is that you have to keep the password in your dialplan. |
| 15:48.05 | Katty | [TK]D-Fender: k'then (= |
| 15:48.21 | Hmmhesays | [TK]D-Fender: you should sometime |
| 15:48.22 | *** join/#asterisk heka (n=heka@82.114.68.126) |
| 15:48.24 | hackeron | Denmark: is it much better than the 301? -- I heard its a nightmare behind a NAT |
| 15:48.31 | [TK]D-Fender | hackeron : Polycom. |
| 15:48.45 | Hmmhesays | one band here does a medly of three little pigs by green jelly and don't phunk with my heart |
| 15:48.47 | hackeron | [TK]D-Fender: we have some IP 301 phones that arent that hot |
| 15:48.51 | Hmmhesays | the crowd loves it |
| 15:48.54 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:48.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:49.11 | [TK]D-Fender | Hmmhesays : The 301 is the low end. 501 is where it really starts. |
| 15:49.21 | Denmark | hackeron : If the network is configured correctly, using STUN usually fixes nat problems .. |
| 15:49.40 | *** join/#asterisk ronn (n=zakforev@84-45-132-117.no-dns-yet.enta.net) |
| 15:49.50 | hackeron | [TK]D-Fender: hmm, ok, thanks - I'll get some 501s :) |
| 15:49.54 | [TK]D-Fender | Hmmhesays : "and along came Rambo who said "eat lead wolfy!" |
| 15:50.22 | Denmark | Hmmhesays : Whats your problems with the 301? |
| 15:50.49 | Denmark | (I wonder if it would affect the scenarios I could imagine) |
| 15:51.13 | ronn | is there a way to disable cdr loggin ? |
| 15:51.23 | Hmmhesays | why are you asking me? |
| 15:51.34 | Hmmhesays | [TK]D-Fender lol |
| 15:52.20 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 15:52.31 | Denmark | [TK]D-Fender : Not meaning to sound cheap .. but if 501 has no advantage to what I want .. why pay more. They had a customer segment to build the 301 for, I assume. :-) |
| 15:53.07 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.60) |
| 15:53.49 | eric_hill | Cisco has the great 7905G series similar to the 301's. They're cheap, they work, and when you put them on users' desks, the users complain that the speakerphone is broken... No mics either. Get the 501s.... |
| 15:53.51 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 15:54.24 | Denmark | Hmmhesays : Didn't you say you had some 301's that weren't hot? |
| 15:54.29 | eric_hill | We had to go back and purchase another 15 or so 7940's to give some of our "special" users speakerphone. |
| 15:54.53 | Hmmhesays | Denmark: not that I recall |
| 15:55.22 | Denmark | <hackeron> [TK]D-Fender: we have some IP 301 phones that arent that hot |
| 15:55.25 | Hmmhesays | RaYmAn-Bx: the password thing is trivial to modify I bet |
| 15:55.37 | Denmark | Hmmhesays : Sorry .. my mistake. :-) |
| 15:55.38 | Hmmhesays | just borrow some code from cmd authenticate |
| 15:56.17 | Denmark | hackeron: What was your problem(s) with the 301? |
| 15:57.02 | Denmark | eric_hill : Well .. in this case the users are not used to speakerphone. |
| 15:57.19 | Hmmhesays | so who wants to help me modify app_lcdial? |
| 15:57.22 | [TK]D-Fender | Denmark : its just that the LCD is character based on the 301 instead of pixel-based so its not as readable or as appealing. |
| 15:57.25 | Hmmhesays | my C skills kind of suck |
| 15:57.38 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 15:57.41 | hackeron | Denmark: donno, never used it myself, hate the grandstream gxp2000 I have here but people in the NY office been complaining about the 301, it wouldnt register the password correctly, there are still working phones for which asterisk constantly reports invalid login, voicemail button doesnt work, speakerphone volume is low, etc |
| 15:57.54 | frenzy | how do I create a SIP trunk that goes to local asterisk? |
| 15:58.14 | Denmark | [TK]D-Fender : That might be an issue .. they have bad eyes, so it needs to be easy to read. (Its for elderly people) |
| 15:58.35 | [TK]D-Fender | Denmark : Pay for the 501 then.... it si a better phone... |
| 15:58.40 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc117.dialup.mindspring.com) |
| 15:58.48 | SpaceBass | morning folks |
| 15:58.48 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:59.04 | Denmark | hackeron : how is sound on the grandstream? I always wondered about that? |
| 15:59.41 | brodiem | Denmark, sub-par |
| 15:59.45 | [TK]D-Fender | Grandstream = Shit-on-a-stick |
| 15:59.49 | brodiem | Denmark, and speakerphone is horrible |
| 15:59.51 | Denmark | [TK]D-Fender : Right. Now you have me worry if the display is too small though. ;-) I need to find some pictures of the phone I guess. Is there back-light in the display? |
| 16:00.17 | hackeron | Denmark: the sound is god awful - the 301 sounds much better, biggest problem is it picks up sounds from the room like crazy, someone typing on the desk opposite is picked up as loud clicking |
| 16:00.31 | [TK]D-Fender | Denmark : no back-light. Very few phones have that. Snom, and Grandstream.... but they suck as phones.... |
| 16:00.55 | [TK]D-Fender | hackeron : You find the 301 is too sensitive? |
| 16:01.12 | hackeron | [TK]D-Fender: no, the grandstream, I never used the 301 myself, just phoned others using it :) |
| 16:01.22 | RoyK | ~nickometer [TK]D-Fender |
| 16:01.34 | santoshr | trelane_ .. i just tried a flash with the current setup.. 2 fxs and 1 fxo..dialed a fxs first then flashed and dialed the fxo ... which happened.. now how do i do a call waiting with this |
| 16:01.50 | Denmark | [TK]D-Fender : So the only real choices are sippura, polycom or manure on a stick? |
| 16:01.55 | brodiem | [TK]D-Fender, don't like snom? I have a 360 here and found it to be pretty nice |
| 16:02.03 | Denmark | sippura/cisco |
| 16:02.05 | Denmark | even. |
| 16:02.09 | salviadud | i use a handytone |
| 16:02.23 | salviadud | works fine for me |
| 16:02.39 | Denmark | salviadud : Have you compared sound quality with other phones? |
| 16:02.50 | salviadud | i happen to own a sipura |
| 16:02.58 | salviadud | a sipura 3000 |
| 16:02.59 | Denmark | (I didn't know handytone made ip-phones . htough I knew they make ATA) |
| 16:03.06 | salviadud | and a spa 841 |
| 16:03.19 | jbalcomb | [TK]D-Fender you get my e-mail about my 'no audio' problem? |
| 16:03.30 | salviadud | they all sound pretty much the same |
| 16:03.30 | Denmark | How does sound quality compare to the 841? |
| 16:03.33 | *** join/#asterisk tamp4x (n=Lab@64.201.13.170) |
| 16:03.41 | Denmark | ok |
| 16:03.45 | [TK]D-Fender | brodiem : Don't have personal experience with Snom, but enough word-of-mouth as to lower voice quality, flakey firmware and horrific use of the LCD. |
| 16:03.54 | salviadud | the spa 841 sounds a bit too much in my opinion |
| 16:03.55 | vgster | does mpg123 compile on x86_64? |
| 16:04.11 | salviadud | kinda loud... |
| 16:04.15 | [TK]D-Fender | jbalcomb : Yeah, congrats on the fix although it makes no sense to me :) |
| 16:04.31 | [TK]D-Fender | Denmark : 841 is a cheap-ass crappy phone... |
| 16:04.35 | coppice | "horrific use of the LCD" :-\ They show HP LOvecraft movies on it? :-\ |
| 16:04.38 | [TK]D-Fender | 841 feels like junk. |
| 16:04.46 | salviadud | i agree d-fender |
| 16:04.52 | salviadud | it does feel like junk |
| 16:04.57 | [TK]D-Fender | coppice : just loking at it makes me scream, is that enough? ;) |
| 16:05.09 | santoshr | how to implement call waiting and to be able to toggle with flash |
| 16:05.21 | [TK]D-Fender | santoshr : What interface? |
| 16:05.45 | *** join/#asterisk GuruDom (n=domiplus@66-202-165-66.rev.knet.ca) |
| 16:05.47 | santoshr | sip fxs devices |
| 16:05.50 | nettie | [TK]D-Fender seems I have a problem with moh when I call to the pstn trough my voip carrier. Local is fine. Do you think could e a codec problem related to my carrier? If the remote party talks or make noises the playback is smooth, if he doesnt the playback is jumpy.. uhmm any idea please? thanx! |
| 16:06.01 | salviadud | i got this setup, i have the grandstream connected to my analog switchbox, so i can dial from my spa 841 to the grandstream, and do an internal call, it sounds like crap on the spa 841 |
| 16:06.10 | Denmark | right now I have a sippura spa2k |
| 16:06.17 | salviadud | if i use a regular phone and call the grandstream for a voip call, sounds pretty good |
| 16:06.25 | Denmark | and use "legacy" (is that the right word?) phones. |
| 16:06.28 | jbalcomb | [TK]D-Fender yeah, definitely wierd. I'm going to contact digium about it today. |
| 16:06.45 | [TK]D-Fender | nettie : no clue... |
| 16:06.46 | jbalcomb | [TK]D-Fender calling dell too about my 'PCI Parity Error' |
| 16:06.46 | salviadud | now, i don't understand fender, why you dissin' on the grandstream? |
| 16:07.19 | Denmark | salviadud : So grandstream is better than the 841? |
| 16:07.48 | salviadud | yeah, definetely |
| 16:07.52 | [TK]D-Fender | salviadud : The whole world admits that they "mean well" and are "inexpensive", but that they cut so many corners I'd rather pay double for something decent!@ |
| 16:08.07 | salviadud | haha |
| 16:08.08 | Denmark | OK ... I forget about the 841 |
| 16:08.19 | [TK]D-Fender | Denmark : 841 is close to the GS's, but I would suggest against either. |
| 16:08.33 | salviadud | besides the 841 is just a sip phone |
| 16:08.37 | puzzled | get a 941 or a polycom or cisco |
| 16:08.45 | puzzled | or snom |
| 16:08.48 | salviadud | the grandstream is an ata with fxs and fxo ports |
| 16:08.52 | Denmark | oh .. 941 > 841 ? |
| 16:08.57 | coppice | anyone seen the grandstream video phone? |
| 16:08.58 | [TK]D-Fender | 941 is a waste. Its too close to the IP 501 to consider. |
| 16:09.00 | salviadud | does a lot more stuff |
| 16:09.10 | Hmmhesays | i think i've fallen in lurve with freepbx |
| 16:09.23 | [TK]D-Fender | Denmark : the 941 is a "nicer" version of the 841. Better, but not worth it. |
| 16:09.29 | Denmark | I am not sure what fxs and/or fxo is. |
| 16:09.54 | puzzled | [TK]D-Fender: but afaik the 941 is cheaper than the 501 and is much easier to configure |
| 16:10.03 | salviadud | denmark, welcome to school :) |
| 16:10.05 | Denmark | I thought ATA was a box where you connect legacy phones (Old PSTN stuff) |
| 16:10.54 | [TK]D-Fender | puzzled : A tiny bit easier, a little bit cheaper, but physically inferior. The 501 is worth the extra 20$ hands-down. |
| 16:11.37 | puzzled | [TK]D-Fender: agree when you talk 1 to 5 phones but when you talk about 1000 phones it requires way more $$$ |
| 16:11.41 | De_Mon | why is there a cdr_database.so AND cdr_odbc.so? |
| 16:11.55 | puzzled | De_Mon: to confure the innocent |
| 16:11.59 | ronn | is there a way to disable cdr loggin completely? |
| 16:12.00 | puzzled | confuse even |
| 16:12.03 | Denmark | puzzled : Esp. the configuration stuff. |
| 16:12.05 | [TK]D-Fender | puzzled : You mean as to volume pricing? |
| 16:12.11 | puzzled | yup |
| 16:12.19 | De_Mon | do they both use the same driver for the given database? |
| 16:12.42 | puzzled | polycom sucks in the volume pricing department while cisco will go a long way to get the deal |
| 16:12.55 | [TK]D-Fender | puzzled : then again, companies want QUALITY typically, and Polycom is better managed than Sipura.\ |
| 16:12.56 | Denmark | De_Mon : ODBC is a way to unify the interface to different databases on varios platforms. |
| 16:13.32 | [TK]D-Fender | puzzled : Cisco makes great phones, but they still are considerably more expensive than Polycom, and are troublesome with their SIP images still last I heard. |
| 16:13.33 | ronn | anyone knows how to disable cdr loggin ? |
| 16:13.33 | coppice | sipura is cisco. companies believe the sun shines from the rear of cisco |
| 16:13.37 | puzzled | [TK]D-Fender: my experience is that the top brass want quality and couldn't care less about the rest |
| 16:14.01 | puzzled | coppice: yup |
| 16:14.10 | Denmark | puzzled : So when we talk one phone, its the 501 that is supposed to win the race ,no doubt? |
| 16:14.13 | De_Mon | ronn edit modules.conf and change load => cdr_foo to noload => cdr_foo |
| 16:14.15 | [TK]D-Fender | puzzled : In mine they care a normal amount on agverage and would rather pay a bit more to know they won't be changing their minds later. |
| 16:14.18 | *** join/#asterisk jac]Z[oby (n=me@62.218.44.178) |
| 16:14.41 | puzzled | Denmark: not necessarily since with one phone you would better buy the polycom 601 :) |
| 16:14.43 | [TK]D-Fender | Denmark : All depends on how much saving a few bucks in the big picture factors in. |
| 16:14.50 | jac]Z[oby | anyone can help me with a question concerning ooh323 with asterisk and innovaphone PBX? |
| 16:14.55 | ronn | De_Mon: what's the cdr module for the cdr loggin in Master.csv ? |
| 16:15.04 | jac]Z[oby | having troubles with the keepalive |
| 16:15.05 | [TK]D-Fender | Denmark : You won't regret a 501, I'll tell you that. I regretted my 941, and am glad to be rid of it :) |
| 16:15.24 | puzzled | [TK]D-Fender: seems you have a customer with a brain :) |
| 16:15.25 | Denmark | puzzled : Hehe .. you confusing me .. but the 601 is much more expensive, right? |
| 16:15.41 | puzzled | Denmark: yes but you get what you pay for |
| 16:15.57 | De_Mon | ronn actualy I just looked in etc (you know, like you should have done first) cdr.conf has the option 'enable=yes' change that |
| 16:16.04 | [TK]D-Fender | Denmark : the IP 601 is a GREAT phone (I run an office on just those), but probably over-kill for you. |
| 16:16.26 | *** join/#asterisk ToTo (n=ToTo@host31-145.pool874.interbusiness.it) |
| 16:16.31 | De_Mon | but for performance sake, don't load modules you arn't using! |
| 16:16.46 | ronn | De_Mon: thanks will try that |
| 16:17.04 | Denmark | [TK]D-Fender : Yeah .. I am kinda new to the extra features that comes with IP .. since my friends and family just recently started getting seriously interesed. |
| 16:17.05 | puzzled | so the TC400P is going to cost around $1,999. anyone with experience in the dsp card biz know if that's a reasonable prive? |
| 16:17.07 | *** join/#asterisk nubbrage (n=blitzrag@12.174.37.165) |
| 16:17.24 | salviadud | errrr, that's not cheap |
| 16:17.28 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 16:17.36 | Denmark | s/erese/ereste/ |
| 16:17.43 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc117.dialup.mindspring.com) |
| 16:17.44 | salviadud | so, it'll probably work nicely |
| 16:17.45 | [TK]D-Fender | Denmark : the 601 adds PoE, 6 lines, MicroBrowser, and a higer-res screen. |
| 16:17.56 | ronn | De_Mon: I don't have cdr.conf .. can I just create the file ? |
| 16:18.03 | puzzled | [TK]D-Fender: wish they would add a backlit screen |
| 16:18.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:18.27 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 16:18.45 | [TK]D-Fender | puzzled : thats a common complaint which I think Polycom is strongly considering. They should also ditch the PoE adapter cable and go integrated and just charge moe. its worth it. |
| 16:19.04 | Denmark | [TK]D-Fender : How is the MicroBrowser usefull? Just bragvalue or? |
| 16:19.29 | puzzled | Denmark: stream real time stock pricing to the phones of the brass. they love it |
| 16:19.55 | Denmark | (higher resolution screen sounds tempting though) |
| 16:20.15 | [TK]D-Fender | Denmark : Good for displaying your company logo (which you can do on the 501 without it), and interactive content. |
| 16:20.24 | Denmark | puzzled : Oh .. yeah.. whether info .. stuff like that. :-) |
| 16:20.26 | jac]Z[oby | someone pro in ooh323 ? |
| 16:20.34 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 16:20.35 | [TK]D-Fender | puzzled : Yup, perfect sample. I use mine to show live queue stats. |
| 16:20.38 | fu3 | hey lads |
| 16:20.43 | [TK]D-Fender | Denmark : exactly |
| 16:21.22 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:21.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:22.08 | Denmark | [TK]D-Fender : Is the letters bigger in the 601? |
| 16:22.25 | [TK]D-Fender | Denmark : Not sure, don't think so... just room for more. |
| 16:22.47 | [TK]D-Fender | Denmark : I have little experience with the 501 & 301. |
| 16:23.07 | fu3 | Why do I live in the only town in America that cannot get PRI |
| 16:23.11 | [TK]D-Fender | I think its bigger on the 501 because of being 1/2 the res. |
| 16:23.35 | Denmark | [TK]D-Fender : Its a hardsome phone, the 601. |
| 16:23.51 | Denmark | very clean design. |
| 16:24.21 | [TK]D-Fender | Denmark : they're not bad... the Cisco's are nicer, just not 100$ nicer :) |
| 16:24.23 | santoshr | [TK]D-Fender: is callwait a function. i mean can i call it in case of busy |
| 16:24.53 | [TK]D-Fender | santoshr : what is "it" What equipment are you referring to with regards to CW? |
| 16:25.52 | santoshr | it is a fxs device sort of a ip phone.. |
| 16:26.07 | [TK]D-Fender | santoshr : Sorry, you are just not making any sense.... |
| 16:26.17 | Denmark | fxs fxo ... lemme google those .. I was confused by the "just of sip-phone" comment. |
| 16:26.42 | santoshr | fxo would be something which would be sending analog to ip.. and fxs vice versa |
| 16:26.43 | nettie | uhmm |
| 16:26.55 | nettie | the moh is jumpy also calling landlines |
| 16:27.19 | nettie | could be that my mp3 is encoded at 320kbit/sec? |
| 16:27.34 | nettie | s/that/because |
| 16:28.46 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 16:29.06 | santoshr | so if a has connected with b both being the same kinda fxs devices and if c dialed to b ... ? but i know for a fact tht i am able to flash a calll directly by pressing the on-hook button |
| 16:29.21 | jac]Z[oby | lol |
| 16:30.07 | jac]Z[oby | http://en.wikipedia.org/wiki/Foreign_exchange_station |
| 16:30.32 | jac]Z[oby | aka FXS, FXO on same page |
| 16:30.52 | Denmark | jac]Z[oby : thanks |
| 16:31.01 | jac]Z[oby | np |
| 16:31.06 | coppice | at one of these foreign exchange stations, no actual foreign exchange is possible |
| 16:31.23 | jac]Z[oby | :) |
| 16:31.26 | docelm0 | Hay I got a interesting question for someone.. Can chan_sip bind more than one port at a time? |
| 16:31.39 | wasim | or you get a real crappy rate ... best case ... |
| 16:32.32 | coppice | the names fxs and fxo were chosen to obscure rather than illuminate |
| 16:32.33 | santoshr | [TK]D-Fender: i am a bit new to asterisk ? now tht u all hvae had a laugh.. can u throw some light on cW |
| 16:33.40 | docelm0 | Anyone on the chan_sip question? |
| 16:34.32 | wasim | docelm0: not that i'm aware of |
| 16:34.50 | docelm0 | I didnt think so damnit |
| 16:35.07 | [TK]D-Fender | santoshr : Who's call waiting are we talking about? on an analog phone line in from the telco? as a feature on an ATA to handle multiple calls? |
| 16:35.16 | wasim | docelm0: you could iptables it |
| 16:37.27 | santoshr | no all the devices are sip boxes |
| 16:38.06 | [TK]D-Fender | santoshr : what is a "sip box", and who's CW are we talking about? |
| 16:38.17 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 16:38.21 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:38.55 | *** join/#asterisk apardo (n=apardo@87.218.44.253) |
| 16:39.30 | santoshr | those are devices whch i am proxying to the * box. |
| 16:39.44 | santoshr | and cw would be on tht device... |
| 16:40.32 | [TK]D-Fender | santoshr : You are failing BADLY. Give me the model # of the device who's use of call waiting are are asking about! Is it on a normal analog line, or a feature on an ATA?!! |
| 16:41.23 | [TK]D-Fender | santoshr : If I say "my transmission is bad", I could be talking about my CAR, my TV, my internet upstream rate, ANYTHING!!!! |
| 16:42.04 | trelane_ | woot! I just clobbered /usr/bin/perl! |
| 16:42.06 | Denmark | Satelite disc .. |
| 16:42.07 | santoshr | ok.. wiat |
| 16:42.09 | Abydos313 | true, but most people would immediately think of car transmission ;) |
| 16:42.28 | GerbilNut | or, us nerds would think internet upstream rate |
| 16:42.46 | *** join/#asterisk RoyK (n=roy@85.166.27.37) |
| 16:42.56 | santoshr | http://www.welltech.com/product_e_12.htm |
| 16:43.02 | [TK]D-Fender | Abydos313 : Most people are STUPID and should require a license to think.... |
| 16:43.08 | santoshr | [TK]D-Fender: can u please vist the aboce mentioned url |
| 16:43.13 | Abydos313 | heh |
| 16:44.36 | [TK]D-Fender | santoshr : Fine so you're talking about an ATA that supports Call-waiting as a way of handling multiple calls on an analog phone. now that I know you're not talking about an analog LINE, what about this? |
| 16:44.43 | Abydos313 | that device says rfc3261.. i though * used rfc2833 |
| 16:45.35 | wasim | 3261 is SIP |
| 16:46.03 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 16:46.51 | Abydos313 | ok |
| 16:46.52 | santoshr | <PROTECTED> |
| 16:47.14 | [TK]D-Fender | santoshr : Did you try reading the manual? |
| 16:47.20 | wasim | heh |
| 16:48.35 | _medic_ | I suffer a serious asterisk problem, which i cannot debug,and i couldn't find any help in the web .. I can SIP without a problem, but approx 5 - 10 seconds after i took a CAPI-Call offhook (I dial from PSTN into the BRI, and transmitit to a SIP client on my workstation), the whole box freezes :-( it happened with asterisk 1.2.4 as well with 1.1.x .. anyone who has any advice ? |
| 16:49.33 | santoshr | but wont i hvae to do anything in the extensions.conf |
| 16:50.19 | _medic_ | or at least someone who knows what to do with |
| 16:50.21 | _medic_ | <PROTECTED> |
| 16:50.22 | _medic_ | <PROTECTED> |
| 16:50.36 | _medic_ | why is result disabled ? |
| 16:51.17 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 16:51.21 | Denmark | _medic_ : Most of what you say is greek to me, but have tried stracing or ptracing (or whatever) asterix to find out what makes it crash? |
| 16:52.19 | [TK]D-Fender | santoshr : this is not *'s job its more of a "how do I make my POS ATA do more than it was designed to".... |
| 16:52.28 | santoshr | ok |
| 16:52.30 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:52.33 | _medic_ | Denmark, not yet, because i am not sure what to trace .. after all, it deadlocks the kernel ... |
| 16:52.43 | *** join/#asterisk kokey (n=ubunture@201.153.63.87) |
| 16:52.54 | santoshr | thnks for the help dude. |
| 16:52.58 | [TK]D-Fender | santoshr : I just downloaded the manual (something you should have done) and read it. There appears to be NO call-waiting feature for jumping between 2 calls. |
| 16:53.10 | Denmark | _medic_ : Which kernel? |
| 16:53.32 | Denmark | _medic_ : I mean - which version of the kernel? |
| 16:53.48 | _medic_ | Denmark, so i am not sure, if a should debug kernel (BIG mess) .. or only asterisk .. but would have to do it2-3times, to ensureit crashes at the same position |
| 16:53.51 | Denmark | Katty : Have you watched the "the net" movie? |
| 16:53.52 | *** join/#asterisk ^HeLL^ (i=root@80.91.87.66) |
| 16:53.56 | _medic_ | os[medic@growsome, Linux 2.6.15-gentoo-r7 i686] |
| 16:54.00 | ^HeLL^ | hello all! |
| 16:54.01 | Katty | Denmark: yes'm, it's grand. |
| 16:54.09 | santoshr | thanks [TK]D-Fender |
| 16:54.11 | Denmark | :) |
| 16:55.07 | Denmark | _medic_ : Try with 2.6.16 if possible? |
| 16:55.22 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 16:55.39 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 16:56.07 | _medic_ | surely its possible ... |
| 16:56.09 | Denmark | Katty : Bullocks made a few great. |
| 16:56.17 | Katty | (= |
| 16:56.17 | rbd | is it possible to stick a queue in front of a meetme conference that has a maximum user size (i.e. if the conference is already full, queue users up until someone in the conference either leaves or is kicked) |
| 16:56.44 | _medic_ | *sighs* but i doubt its the kernel .. sip-only works great... |
| 16:57.53 | _medic_ | its as soon as capi (and therefore Fritz!PCI) come to play... but the card never made any problems yet ... i am just stuck |
| 16:58.15 | Denmark | _medic_ : The kernel shouldn't deadlock usually.. |
| 16:58.39 | Denmark | _medic_ : Besides, I think there are a few security patches in .16 |
| 16:59.34 | _medic_ | *sighs* i HATE kernel debugging :-((( but if thats the only way :-/ |
| 16:59.38 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 16:59.52 | ^HeLL^ | anybody has signed up into AsteriskCounter? |
| 16:59.56 | Denmark | _medic_: I am just suggesting to upgrade your kernel, not debug it. :-) |
| 17:00.26 | _medic_ | Denmark, i doubt, upgrading the kernel will change anything |
| 17:00.33 | wasim | rbd: cute, but you'd need an intermediary channel, agent |
| 17:00.47 | wasim | rbd: who is logged into the meetme and the queue |
| 17:01.10 | Denmark | _medic_ : Its one of those small things support will ask you to do ... are you afraid of your uptime? |
| 17:01.12 | rbd | wasim, yeah that's what it looked like to me from the queue docs...oh well |
| 17:01.34 | _medic_ | even upgrading chan_capi-cm to 0.6.4 or asterisk from 1.1.x to (marked unstable) 1.2.4 didnt change anything.. |
| 17:01.41 | Denmark | _medic_ : I just wasted 107+ days of uptime to upgrade my kernel ... |
| 17:02.02 | Denmark | _medic_: Uptime is nice .. but when an upgrade is needed .. don't hesitate to waste it. |
| 17:02.06 | _medic_ | Denmark, no .. its only a few hours, because i triedseveral things already this morning :) |
| 17:02.15 | Denmark | ok |
| 17:02.45 | ^HeLL^ | i'm making a screenshot of this channel to insert it into an article about Asterisk, so say hello! :) |
| 17:02.46 | _medic_ | but since this a important box, i wanted to leave it as "stable" as possible |
| 17:03.00 | file | hello! |
| 17:03.12 | _medic_ | Hi :-) |
| 17:03.27 | vgster | does anyone have any knowledge of pri hdlc errors? and in particular how I can get around PRI got event: HDLC Abort (6) |
| 17:04.50 | ^HeLL^ | vgster: have you got bit sync right? |
| 17:04.56 | vgster | bit sync? |
| 17:04.58 | Denmark | _medic_: You can reuse the config file, and keep the old kernel for reboot. But I would run strace or something on asterix first. |
| 17:05.15 | ^HeLL^ | vgster : second parameter on zaptel span |
| 17:06.02 | _medic_ | i will.. |
| 17:06.30 | _medic_ | and i will trynewest bleeding edge on asterisk and chan_capi |
| 17:06.30 | Denmark | ok.. I am off to try and enable encryption in my SPA2k :-) |
| 17:06.40 | Denmark | _medic_: :-) |
| 17:10.14 | vgster | ^HeLL^ > yes ive tried various options, seems to be the mobo, putting the car din the pci-x slot seems to make it go away |
| 17:11.22 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 17:11.23 | jbalcomb | Has anyone had to adjust thier 'silencethreshold' for voicemail above the default '96' to deal with people getting cut off while leaving voicemails? |
| 17:11.24 | backblue | hi |
| 17:11.50 | backblue | what its the channel of ser? |
| 17:11.52 | backblue | openser |
| 17:12.22 | ^HeLL^ | screenshot done., thanks to all... |
| 17:12.29 | *** join/#asterisk Darwin_35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 17:12.31 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 17:14.18 | wasim | pah, build your own kernel, don't let gentoo do it |
| 17:14.24 | backblue | ppl, if i need a sip cluster, i can do it with openser and mediagateway without asterisk? |
| 17:14.52 | wasim | yes |
| 17:15.27 | backblue | so asterisk only exists to interconnet to telcos and to do disa and voicemail? |
| 17:16.06 | wasim | no |
| 17:16.32 | backblue | wasim: what more? :D |
| 17:16.48 | ^HeLL^ | backblue : i use asterisk to control the temperature of my acuarium |
| 17:17.15 | backblue | ^HeLL^: mine it's better, i use asterisk to control my both acuarium's. |
| 17:17.26 | ^HeLL^ | when i travel |
| 17:17.41 | ^HeLL^ | with the same controller? |
| 17:17.54 | ^HeLL^ | :D |
| 17:17.56 | backblue | yes |
| 17:18.05 | ^HeLL^ | curious |
| 17:19.00 | *** join/#asterisk saftsack (n=saftsack@p54A7E996.dip.t-dialin.net) |
| 17:19.07 | ^HeLL^ | asterisk is used to make spam. XD |
| 17:19.12 | ^HeLL^ | too |
| 17:19.26 | *** join/#asterisk RoyK (n=roy@ti211310a080-6949.bb.online.no) |
| 17:20.10 | ^HeLL^ | good, now return back |
| 17:20.12 | ^HeLL^ | ... |
| 17:22.08 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 17:22.52 | jbalcomb | ok, so if i think Asterisk is falsely timingout when someones leaving a voiceamil should I be raising or lowering the silencethreshold? |
| 17:23.26 | jbalcomb | the default is '128' but our's is set to '96', i'm not getting from the option description which way to adjust this. |
| 17:23.38 | mut | lower |
| 17:23.56 | [TK]D-Fender | jbalcomb : You need to stop playing with the gain on your PRI cards.... |
| 17:24.07 | [TK]D-Fender | jbalcomb : What are they set at these days? |
| 17:24.52 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:27.33 | jbalcomb | [TK]D-Fender rxgain=-4.5, txgain=-8.0 |
| 17:27.37 | [TK]D-Fender | EEK.... |
| 17:28.06 | [TK]D-Fender | jbalcomb : You bought a HWEC capable card... you shouldn't be having to mess with that.... |
| 17:28.13 | jbalcomb | [TK]D-Fender yes, yes, yes.. keep in mind I have been round and round with settings trying to find a balance between echo and low volumes |
| 17:28.46 | [TK]D-Fender | jbalcomb : You mean you still have to do that even with the HWEC?! |
| 17:29.01 | jbalcomb | [TK]D-Fender i dunno. the new server isn't in yet and i'm not sure what card the current production server is |
| 17:29.50 | jbalcomb | [TK]D-Fender ultimately i have determined that the echo problems are because of the gxp-2000s but adjusting the gains helped so i'm running with it |
| 17:30.15 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 17:30.19 | [TK]D-Fender | jbalcomb : Well if you're not on a better card yet then I guess you'll just leave it. |
| 17:30.57 | jbalcomb | [TK]D-Fender which way do i adjust the silencethreshold to make it less likely to think something is silence? |
| 17:32.10 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 17:32.33 | [TK]D-Fender | jbalcomb : Lower. a high threshold tolerates MORE noise before considering as "voice". |
| 17:33.20 | [TK]D-Fender | jbalcomb : As you are already below the norm (as is required by your nasty down-tuned gain), just keep digging :) |
| 17:35.58 | *** join/#asterisk slak- (i=slak@i686.us) |
| 17:35.59 | slak- | hey |
| 17:36.18 | slak- | the bosses wife is on my ass, she wants written instructions on how to check vm, change the greeting, change passwoird |
| 17:36.21 | slak- | etc |
| 17:36.27 | slak- | anyone know of such a document on voip-info? |
| 17:36.31 | slak- | i searched and searched |
| 17:36.38 | [TK]D-Fender | slak- : cmd voicemailmain <- |
| 17:36.45 | slak- | ? |
| 17:36.59 | [TK]D-Fender | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain |
| 17:37.12 | slak- | ok |
| 17:37.12 | slak- | ty |
| 17:37.22 | [TK]D-Fender | print & hand it over. and hand-wriote on the # to access it. |
| 17:37.23 | jbalcomb | [TK]D-Fender cool. hopefully this will get one more person off my back. ;) |
| 17:37.40 | [TK]D-Fender | jbalcomb : Are they that numerous?> |
| 17:39.41 | jbalcomb | [TK]D-Fender no, just one person but they are the rep for a client who provides 44% of our revenue. :/ |
| 17:39.58 | [TK]D-Fender | jbalcomb : know that feeling... |
| 17:40.05 | *** join/#asterisk srt (n=nobody@gw0-cgn.reucon.net) |
| 17:40.19 | [TK]D-Fender | I think I'm going to take the plunge and buy some Polycom phones for home.... |
| 17:40.46 | [TK]D-Fender | I've been running off a borrowed IP 600 for too long... |
| 17:40.50 | slak- | man |
| 17:40.56 | [TK]D-Fender | So time to get a 301 & 501 |
| 17:40.56 | slak- | i just told the bosses wife |
| 17:40.59 | slak- | 'you are not my boss' |
| 17:41.02 | slak- | and she got so pissed |
| 17:41.05 | slak- | SOO pissed |
| 17:41.09 | slak- | ahhaha |
| 17:41.18 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 17:41.29 | [TK]D-Fender | slak- : thats what some would call an ECM (Ending Career Move) |
| 17:41.37 | slak- | really? |
| 17:41.44 | slak- | i call it PMS |
| 17:41.55 | [TK]D-Fender | slak- : He sleeps with her so you CAN catch flack for that... |
| 17:42.05 | slak- | well |
| 17:42.18 | slak- | if i hand over this nicely formatted word document within the enxt 10min |
| 17:42.21 | slak- | maybe i'll be OK |
| 17:43.12 | salviadud | slak, did you say it nicely? |
| 17:43.27 | salviadud | or was it in a mean sort of way |
| 17:43.30 | *** join/#asterisk vaewyn (i=freeman@mail.parrishmachine.com) |
| 17:43.42 | salviadud | like "fuck off bitch" |
| 17:43.57 | scrambray8927 | what are Level 3 networks? |
| 17:44.23 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:44.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:44.29 | slak- | salviadud: it was just under my breath |
| 17:44.33 | slak- | as she was leaving my office |
| 17:44.41 | *** join/#asterisk ebag (n=gabe@adsl-69-239-166-49.dsl.renocs.pacbell.net) |
| 17:44.46 | slak- | she had a shitfit cos she found some cd from an LCD projector laying around |
| 17:44.48 | slak- | not labelled |
| 17:44.56 | slak- | and not in its 'designated folder' |
| 17:45.05 | salviadud | women |
| 17:45.15 | slak- | and this is all during lunch |
| 17:45.19 | slak- | she had the nerve to storm in here |
| 17:45.27 | slak- | acuse me of misplacing this fucking cd ive never seen before |
| 17:45.38 | slak- | and then i said 'ok fine give it to the secretary, she files these things/ |
| 17:45.42 | slak- | and she got pissed then |
| 17:45.52 | slak- | and started going off about how she doesnt have voicemail instructions still |
| 17:46.07 | slak- | and i said, its pretty self explanatory, i mean the operator voice tells you what to do |
| 17:46.14 | slak- | and then she got even more pissed |
| 17:46.16 | salviadud | haha, what a dumbass |
| 17:46.23 | slak- | and thats when i said, you're not my boss and im at lunch |
| 17:46.32 | salviadud | good work slak |
| 17:46.52 | salviadud | i believe your boss has to live through that everyday |
| 17:47.00 | slak- | ua he's totally bitched out |
| 17:47.02 | slak- | i feel for this guy |
| 17:47.16 | slak- | OCD is a serious psychiatric disorder |
| 17:47.20 | slak- | it effects all of us here |
| 17:47.22 | slak- | ;( |
| 17:47.26 | salviadud | OCD? |
| 17:47.42 | slak- | obsessive compulsive disorder |
| 17:47.47 | salviadud | ohhh, that |
| 17:47.47 | slak- | is what i think the real problem is |
| 17:47.58 | trelane_ | I have two users with Cisco 7960 phones running the latest sip firmware (8.2). When attempting to access their voicemail or any other part of the ivr system their DTMF tones (dtmf=inband/ulaw) are not being sent by the phone, or asterisk isn't picking them up, does anyone have any experience with these phones that has seen this issue? dialing out and back in works, however direct dialing an internal extension and then following prompts does no |
| 17:47.58 | trelane_ | t. |
| 17:48.17 | slak- | she said to me |
| 17:48.17 | salviadud | and what's the name of the company? |
| 17:48.25 | slak- | 'dont you know who pays your salary?; |
| 17:48.26 | slak- | ahahha |
| 17:48.35 | Darwin_35 | everyone that uses asterisk suffers from ocd |
| 17:48.48 | salviadud | no way, i don't suffer from ocd |
| 17:49.03 | slak- | if you saw my office, i clearly dont suffer from ocd |
| 17:49.07 | salviadud | so, she got all power crazy |
| 17:49.09 | slak- | its the opposite |
| 17:49.14 | slak- | i dont give a shit about anything |
| 17:49.22 | salviadud | cause she's the boss's wife |
| 17:49.24 | slak- | ya |
| 17:49.25 | slak- | she said |
| 17:49.33 | *** join/#asterisk sterfablod (n=sterfabl@216-106-179-17.ds1.mia1.net.ststelecom.com) |
| 17:49.34 | *** join/#asterisk HamYaI (n=HamYai@125.24.12.236) |
| 17:49.35 | slak- | 'and dont you EVER tell me that im not your boss again' |
| 17:49.38 | slak- | all loud like |
| 17:49.58 | sterfablod | anyone know of a good prepaid billing app for asterisk? |
| 17:50.10 | salviadud | tell your boss he needs to put a leash on that bitch |
| 17:50.31 | slak- | im going to tell him that my work orders need to arrive ina more calm and structured manner |
| 17:50.39 | slak- | and my lunch hour should be respected |
| 17:50.58 | salviadud | where the hell do you work man? |
| 17:51.06 | slak- | this moms and pops startup |
| 17:51.08 | slak- | fuck this place |
| 17:52.08 | salviadud | what's your astrological sign slak? |
| 17:52.08 | slak- | pieces |
| 17:52.08 | salviadud | i can imagine the tension... |
| 17:52.17 | salviadud | dude, if you can get a better job |
| 17:52.20 | salviadud | go for it |
| 17:52.24 | slak- | i hnestly havent looked |
| 17:52.36 | slak- | but you know i got this job at 18-19 |
| 17:52.38 | slak- | 3 years ago |
| 17:52.44 | slak- | and they treat me like im their kiddo |
| 17:52.48 | slak- | not so much anymore |
| 17:52.56 | slak- | but occassionally she gets that way |
| 17:53.46 | salviadud | so, it's a small place |
| 17:54.04 | slak- | like 20 employees |
| 17:54.18 | salviadud | and you're the sysadmin? |
| 17:54.25 | slak- | ya and java dev |
| 17:54.49 | scrambray8927 | Is being on a "level 3 network" a benefit when seraching for a VoIP termination service? |
| 17:55.12 | slak- | being on a network with the lowest latency is |
| 17:55.25 | slak- | i'd traceroute over a few days to all of your final choices |
| 17:55.27 | denon | level3 is a voip provider |
| 17:55.30 | slak- | and pick based on that |
| 17:55.34 | Darwin_35 | ok who here is using postgress and realtime ? |
| 17:55.36 | denon | and they usually aren't well known for their quality |
| 17:55.49 | shido6 | being on a network peered with other MAJOR isps and minotr isps |
| 17:56.02 | scrambray8927 | slak-, how would I go about finding the IP's to traceroute for each possible VoIP provider? |
| 17:56.03 | shido6 | peering and latency |
| 17:56.23 | *** join/#asterisk scrambray8927 (n=scrambra@12.104.121.147) |
| 17:56.55 | scrambray8927 | slak-, would I traceroute the same IP as their web site? |
| 17:57.40 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 18:00.49 | *** join/#asterisk saftsack (n=oliver@p54A7E2C0.dip.t-dialin.net) |
| 18:01.09 | *** join/#asterisk rue_mohr (n=not@24.207.0.99) |
| 18:01.12 | rue_mohr | hello |
| 18:01.15 | scrambray8927 | hi |
| 18:01.16 | slak- | scrambray8927: nah thats not a reliable way |
| 18:01.21 | slak- | i'd email them |
| 18:01.28 | slak- | and ask for the ip |
| 18:01.33 | scrambray8927 | thanks |
| 18:01.35 | slak- | tell them why you want it |
| 18:01.47 | rue_mohr | I have an unlocked pap2 here, can anyone point me at information on how to tie it to our asterisk system? |
| 18:01.54 | slak- | some common providers, the folks in here might have the information |
| 18:02.13 | scrambray8927 | would anyone in here happen to have the IP's for VoIP providers so I can check the latency? |
| 18:03.20 | scrambray8927 | slak-, how much of a difference in ms for the latencey am I going to want to be concerned with? 100ms? |
| 18:03.26 | slak- | well |
| 18:03.29 | slak- | i use asterlink |
| 18:03.35 | slak- | and my latency to them is like 20ms |
| 18:03.37 | slak- | ;) |
| 18:03.40 | slak- | and thats what i like to see |
| 18:03.46 | slak- | 17ms |
| 18:04.00 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:04.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:04.24 | rue_mohr | anyone? |
| 18:04.32 | scrambray8927 | nice |
| 18:07.01 | xbit` | could you advise a softphone for linux? |
| 18:07.18 | *** join/#asterisk shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 18:07.28 | jbalcomb | trelane sounds like you need to set those accounts to dtmf iva rfc2833 in the sip.conf |
| 18:07.35 | [TK]D-Fender | xbit` : Ekiga |
| 18:07.38 | slak- | scrambray8927 |
| 18:07.41 | slak- | try this |
| 18:07.46 | slak- | switch.asterlink.org |
| 18:07.48 | slak- | ping it |
| 18:07.51 | scrambray8927 | k |
| 18:08.03 | slak- | er |
| 18:08.05 | slak- | .com? |
| 18:08.09 | slak- | ya |
| 18:08.10 | slak- | .com |
| 18:08.21 | slak- | <PROTECTED> |
| 18:08.26 | slak- | pretty good |
| 18:08.48 | jbalcomb | scrambray8927 Level 3 is an ISP |
| 18:08.52 | slak- | also you can join #asterlink and ask them whatever questions you might have |
| 18:08.57 | slak- | im out to lunch peaqce |
| 18:09.01 | scrambray8927 | slak-, time=42ms TTL=53 |
| 18:09.08 | scrambray8927 | slak-, thanks, cya |
| 18:09.29 | scrambray8927 | jbalcomb, do they have VoIP temination services? |
| 18:09.57 | jbalcomb | scrambray8927 i do not know. i only deal with the for network peering and uplinks |
| 18:09.58 | rue_mohr | anyone able to offer me any help with a linksys PAP2? |
| 18:10.18 | scrambray8927 | jbalcomb, thanks |
| 18:10.22 | shaun2222 | scrambray8927: L3 has voip services if thats what your asking |
| 18:10.26 | shaun2222 | skype uses them... |
| 18:11.11 | *** join/#asterisk QbY (i=user@cm-12-197-169-221.dhcp.geo-sc.southerncoastalcable.net) |
| 18:11.40 | scrambray8927 | shaun2222, I'm looking for a voip termination account that will accept my asterisk callerID and reflect it externally and allow me to have more than 4 outgoing lines open at once per account |
| 18:11.42 | *** join/#asterisk chrustinho (n=christop@p54AA09A2.dip0.t-ipconnect.de) |
| 18:12.24 | scrambray8927 | nufone only allows 4 outbound per account (so I'd probably have to write a script to swtich account information if all 4 outbound lines were being used - unless somethign like this exists already)) |
| 18:13.11 | shaun2222 | scrambray8927: your in the same boat i am in, i actually just dialed voicepulse to ask them what the limit was and if it could be raise for more money. |
| 18:13.17 | QbY | Server running kernel version 2.6.9 or 2.6.11.9, no other options, no USB, how can I get a "timer" to utilize conferencing? |
| 18:13.36 | scrambray8927 | shaun2222, i meant voicepulse connect, i haven't been able to get in touch with nufone |
| 18:13.39 | GerbilNut | setup the next priority with the other account info, should switch to it |
| 18:13.54 | scrambray8927 | GerbilNut, thanks |
| 18:14.00 | [TK]D-Fender | QbY : with a 2.6 Kernel you can use RTC for timing. |
| 18:14.26 | shaun2222 | scrambray8927: i was thinking just signing up with multiple accounts with them, and then rolling my 800 number accross the local numbers |
| 18:14.29 | QbY | [TK]D-Fender -- Link? Docs? Does it require a module to be compiled into the Kernel? (I'm limited there also) |
| 18:14.37 | shaun2222 | so if ones maxed out it will just go onto the next |
| 18:14.45 | justinu | wow, async RTP is finally commited to svn trunk |
| 18:14.54 | justinu | what universe did I wake up in this morning? |
| 18:14.54 | shaun2222 | 12 connections would be plenty for me. |
| 18:14.57 | shaun2222 | 3 accounts.. |
| 18:15.39 | [TK]D-Fender | QbY : WIKI.... search for it... |
| 18:16.07 | scrambray8927 | technically, if you dont' assign caller id to the lines, do they have phone #'s once they terminat on the PSTN? |
| 18:16.13 | justinu | qby: you want ztdummy... |
| 18:16.21 | scrambray8927 | or would it show up as "unavailable?" on my landline caller id? |
| 18:16.43 | scrambray8927 | shaun2222, do you think you're going to go with voicepulse connect? |
| 18:16.44 | slak- | scrambray8927: what you want is asterlink :) |
| 18:16.49 | slak- | its cheap too |
| 18:16.54 | slak- | no monthly fee |
| 18:16.56 | GerbilNut | shaun2222, look into telcommone.net, they have $0.01 /minute charges for outbound calls. Just don't know if they will let you do the CID or how many calls |
| 18:17.02 | slak- | <2cents /minute |
| 18:17.27 | scrambray8927 | GerbilNut, how many lines can you have active going outbound simultaneous? |
| 18:17.30 | slak- | and CID workds |
| 18:17.36 | shaun2222 | scrambray8927: you may want to look at thsi...http://www.level3.com/VoIPTAP/ |
| 18:17.40 | shaun2222 | might be what youw ant |
| 18:17.43 | GerbilNut | I don't know, i haven't contacted them about that yet |
| 18:17.46 | slak- | and you can have as many simultaneous calls as youur bandwidrth allows |
| 18:17.48 | scrambray8927 | slak-, asterlink.com? i'm int he chan |
| 18:17.52 | slak- | yes |
| 18:17.56 | slak- | talkk to file |
| 18:18.00 | slak- | tell him slak sent ya |
| 18:18.07 | slak- | and ask for your 5 dollar discount |
| 18:18.12 | slak- | or credit |
| 18:18.22 | *** part/#asterisk rue_mohr (n=not@24.207.0.99) |
| 18:19.15 | scrambray8927 | slak-, on what server? |
| 18:20.16 | slak- | wha? |
| 18:20.44 | mut | does the voicemail system let you bypass the password? |
| 18:20.47 | mut | or not? |
| 18:20.59 | mut | like in the dialplan, if exten 1005 dials voicemail, it pops right to it |
| 18:21.02 | scrambray8927 | i thought you meant talk to file in the #asterlink chan, but i now realize you meant in here |
| 18:21.23 | file | talking to me is fruitless, I no longer work for Asterlink |
| 18:21.32 | file | your best bet is MikeJ or bkw |
| 18:21.50 | slak- | you got fired? |
| 18:21.54 | file | no |
| 18:22.07 | scrambray8927 | thanks |
| 18:23.12 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:23.36 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:24.22 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 18:24.42 | *** join/#asterisk stoffell (n=stoffell@d51A4D5AD.access.telenet.be) |
| 18:28.04 | *** join/#asterisk vaewyn (i=ejabberd@jabber.andrews.edu) |
| 18:28.22 | *** join/#asterisk Jon335 (i=Jon335@ottawa-hs-209-217-93-41.d-ip.magma.ca) |
| 18:28.57 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 18:29.45 | vaewyn | Booyah |
| 18:29.53 | *** join/#asterisk wolfson (n=fredugy@65.174.122.198) |
| 18:29.54 | vaewyn | or something like that... |
| 18:32.35 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 18:33.42 | salviadud | what are you cheerin' about? |
| 18:34.09 | vaewyn | I'm actually on the channel again after... ohh... 2+ months |
| 18:34.27 | vaewyn | work has sucked my other-work time :P |
| 18:35.45 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 18:35.47 | salviadud | oh, i see |
| 18:35.56 | FarrisG | Looking for a wireless SIP phone, but they all seem to be 802.11-based. I want the phone presence, the base-station, to be wired, and just have a wireless handset. Does anyone make such a beast? |
| 18:36.00 | Dr-Linux | anybody is using AstGUIclient? |
| 18:36.17 | stoffell | FarrisG, something like Kirk does, with the IP600 |
| 18:36.48 | [av]bani | [TK]D-Fender: how do you set polycom to do blind transfer by default like cisco and grandstream and oh, every other phone on the planet defaults to? |
| 18:36.48 | Hmmhesays | ahh Kirk, the ultimate ladies man |
| 18:37.04 | vaewyn | FarrisG: http://www.voipsupply.com/product_info.php?products_id=923 something like that? :} |
| 18:37.25 | stoffell | [av]bani, does grandstream 'default' to blind transfer? only if you use the transfer button 'first', so.. is that default? |
| 18:37.39 | [av]bani | stoffell: grandstream defaults to blind |
| 18:37.54 | [av]bani | FarrisG: you want an aastra 480-ct |
| 18:38.04 | stoffell | [av]bani, if you use the transfer button, and 'then' dial, yes.. if you first pick a line, then dial, then transfer, no.. |
| 18:38.27 | scrambray8927 | what's the average call bandwith utiliaztion for SIP vs IAX? i.e. 64kb vs 92kb... |
| 18:38.32 | [av]bani | FarrisG: http://www.aastra.com/enterpriseip/pro_243.asp |
| 18:38.39 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 18:38.45 | *** join/#asterisk asterboy (n=root@S010600485480f4be.ed.shawcable.net) |
| 18:39.08 | [TK]D-Fender | [av]bani : Not sure you can chance the default behaviour |
| 18:39.14 | [av]bani | [TK]D-Fender: yay polycom |
| 18:39.21 | asterboy | Anyone suggest a good book to assist me in fixing my forwarding problems? |
| 18:39.31 | FarrisG | [av]bani: That looks great, but I don't need the desk phone |
| 18:39.36 | vaewyn | poly's are very nice :P |
| 18:39.43 | _Paulo_ | scrambray8927, bandwidth comsumption is codec related. |
| 18:39.50 | [av]bani | vaewyn: cisco is better |
| 18:39.53 | [TK]D-Fender | [av]bani : not a big deal... |
| 18:39.56 | stoffell | vaewyn, they are okay, but not that good looking :) |
| 18:40.06 | scrambray8927 | _Paulo_, what's the default codec used |
| 18:40.08 | [av]bani | [TK]D-Fender: it's just annoying to have some phones default one way, and other phones another way |
| 18:40.30 | [TK]D-Fender | [av]bani : then you clearly need to ditch all your other phones and go "all Polycom"! |
| 18:40.35 | vaewyn | Who cares about looks... they have the best speaker phone... and the voice quality rocks normally also |
| 18:40.38 | stoffell | if I call out (through zap, BRI) and hangup before the other partie picks up, the ringing doesn't stop, any idea why this is? (asterisk CLI does say "Hungup ZAP/x) |
| 18:40.40 | [av]bani | [TK]D-Fender: all cisco |
| 18:40.40 | wasim | i wonder if you can get asterix comics in pdf online |
| 18:40.50 | [av]bani | [TK]D-Fender: i would never, ever, go "all polycom" |
| 18:41.35 | _Paulo_ | scrambray8927, you will have do do a little research, it can be different on distinct versions or systems. |
| 18:41.43 | stoffell | [av]bani, why not all poly? |
| 18:41.51 | vaewyn | [av]bani, Actually I think polycoms look better... ciscos look too much like overgrown blackberrys or overgrown cellphones |
| 18:42.01 | [av]bani | stoffell: because cisco is better constructed, far more capable, and better value for the money |
| 18:42.05 | _Paulo_ | scrambray8927, go to http://www.voip-info.org/ |
| 18:42.11 | scrambray8927 | _Paulo_, thanks |
| 18:42.43 | [av]bani | vaewyn: polycoms look like a giant chocolate bar |
| 18:43.03 | stoffell | [av]bani, hm, okay. for less then 40 users (not high volume calling) I'd recommend Thomson's |
| 18:43.16 | vaewyn | [av]bani, hun? you talking about the IPx00 IPx01? units |
| 18:43.20 | _Paulo_ | scrambray8927, look at http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 18:43.21 | [av]bani | vaewyn: ip601 |
| 18:43.40 | [av]bani | vaewyn: in the office, we have gxp2000, ip601, cisco 7970, and snom 360 |
| 18:43.43 | Zodiacal | anyone know if most company's callerid name is accurate? |
| 18:43.47 | vaewyn | [av]bani, nah... more like a car console :P |
| 18:44.06 | [av]bani | vaewyn: the 7970 looks like it belongs on the bridge of the starship enterprise |
| 18:44.15 | [av]bani | thus, my background image is NCC-1701 |
| 18:44.26 | stoffell | hehe [av]bani :) |
| 18:44.37 | vaewyn | [av]bani, anyways... I'll take the call quality and features of the 601 for it's price over any of the Ciscos.... but that is just me |
| 18:45.12 | [av]bani | vaewyn: well, since i have all these phones in the office to compare with, i by far prefer the cisco |
| 18:45.42 | stoffell | [av]bani I think it's personal choice, i prefer the polycom 501 and thomson st2030 above cisco |
| 18:45.48 | [av]bani | and heh, dont ask me about the snom |
| 18:45.48 | asterboy | Anyone suggest a good SIP book? |
| 18:45.51 | FarrisG | Anyone know what the range is like on the Aastra 480i CT? |
| 18:46.35 | vaewyn | [av]bani, Well... Polycoms have all the features we need... the 601 especially... and they are so darn cheap :} and once again... the call quality especially on speakerphone is the best |
| 18:46.46 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:46.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:47.33 | Zodiacal | anyone? callerid name accurate for companies? |
| 18:47.46 | wasim | Zodiacal: it depends |
| 18:47.51 | Zodiacal | usualy? |
| 18:48.05 | [av]bani | vaewyn: the construction of the cisco is better than the ip601, the phone is much faster, the handset is much better, the BLF's are nicer, the screen is better, the buttons are better, the xml support is more comprehensive |
| 18:48.40 | [av]bani | oh, and the default ringtone for the cisco is a ring, not a polycom "fart" |
| 18:48.42 | [av]bani | :) |
| 18:48.46 | eric_hill | Zodiacal: It's hard to answer that question. If CompanyA owns CompanyB, when CompanyB makes a call, sometimes CompanyA comes up on caller ID because CompanyA pays the bills. |
| 18:49.01 | Zodiacal | gotcha |
| 18:49.06 | Zodiacal | thank you! |
| 18:49.06 | stoffell | told you it's personal choice.... :) |
| 18:49.19 | Zodiacal | vaewyn cisco sip doesn't have hints |
| 18:49.21 | [av]bani | stoffell: well, everyone in the office feels the same way... |
| 18:49.33 | [av]bani | stoffell: so its not just me |
| 18:49.54 | vaewyn | [av]bani, I'd rather drop a 601... I havn't noticed a speed difference... handset no one seems to care about here... screen is nicer... but not worth that much more money... buttons are ennh.. and the microbrowser on the IP-601 beats the XML crud on the Cisco to @#$#@$ |
| 18:50.25 | vaewyn | Heck... we are using our 601s as timeclocks the microbrowser is so flexible |
| 18:50.32 | vaewyn | just ripping the handset off |
| 18:50.34 | scrambray8927 | Can someone explain a relatively simple yet confusing concept to me? |
| 18:50.36 | [av]bani | vaewyn: the menus in the 601 are slow... and the phone takes AGES to boot. and er no, the microbrowser in the cisco is MUCH better. the ip601's is VERY limited (i've had to write applications for the 601) |
| 18:50.36 | stoffell | [av]bani, sure, they 'are' superb phones, but i do think it depends on the people.. |
| 18:50.55 | asterboy | nothing can beat two tin cans and a string. |
| 18:50.57 | vaewyn | and the budget :P |
| 18:51.03 | [av]bani | vaewyn: the cisco's xml is far, far, far, far more capable than the xhtml of the polycoms |
| 18:51.11 | scrambray8927 | If I subscribe to a VoIP termination service for outbound calls from my asterisk box and I don't assign caller ID to the extension I'm dialing out from, what type of identification does this call have when it his the PSTN network (say if I call my home phone)? will it show up on my home PSTN caller ID as "Unavailable" or will it have a phone#? |
| 18:51.13 | vaewyn | 3 601s or 2 ciscos... hmmm... :} |
| 18:51.48 | [av]bani | vaewyn: oh, and cisco's firmware policy doesn't suck. polycom can lick my balls |
| 18:51.51 | vaewyn | [av]bani, how so? I havn't come up with an app yet I can't do on the 601... |
| 18:51.53 | wasim | scrambray8927: depends on him |
| 18:52.10 | wasim | scrambray8927: your voip termination services i.e. |
| 18:52.13 | vaewyn | [av]bani, cisco's firmware policy sucks royally if you didn't buy the phones new.... |
| 18:52.26 | [av]bani | vaewyn: nope. anyone can buy a $8 smartnet. i did. |
| 18:52.53 | vaewyn | [av]bani, yes you can buy that... and then if you admit why you did it they cancel it |
| 18:52.55 | [av]bani | whereas if you buy ip601's from ebay, you are F U C K E D |
| 18:52.57 | vaewyn | [av]bani, been there... done that |
| 18:53.16 | vaewyn | hun? You can get firmware for the 601s from tons of places |
| 18:53.18 | scrambray8927 | wasim, do you mean that in some cases the voip term serv will have my outgoing call show up as a # on a PSTN caller id in some cases and in other cases it won't have a number associated with it? |
| 18:53.19 | [av]bani | eh? cisco doesnt sell smartnet anymore. |
| 18:53.23 | [av]bani | its through resellers. |
| 18:53.31 | [av]bani | and resellers don't interrogate you why you're buying smartnet. |
| 18:53.41 | asterboy | [av]bani, you can get the firmware from other sites. |
| 18:53.55 | [av]bani | asterboy: not 'legally'. polycom gets upset |
| 18:53.55 | vaewyn | no... but cisco asked us when they reviews the support contracts for our switches |
| 18:53.57 | asterboy | ~smartnet |
| 18:54.02 | asterboy | ah |
| 18:54.16 | asterboy | cisco wants money yearly. |
| 18:54.17 | [av]bani | asterboy: polycom's firmware policy is bullshit |
| 18:54.21 | scrambray8927 | [av]bani, why are you in a bad spot if you buy ip601's from ebay? |
| 18:54.34 | asterboy | ya they wure make it unnecessarily dificult. |
| 18:54.45 | [av]bani | scrambray8927: beause the only place you are authorized to get polycom firmware upgrades is through the vendor who sold you the polycom. |
| 18:54.51 | asterboy | coffee is screwing up me spelling |
| 18:54.52 | scrambray8927 | gotcha |
| 18:54.52 | [av]bani | scrambray8927: and on ebay, that isnt likely to happen |
| 18:55.05 | scrambray8927 | [av]bani, so it's illegal? |
| 18:55.13 | [av]bani | scrambray8927: polycom thinks so |
| 18:55.14 | asterboy | I wanted to buy Polycom from a vendor and they asked if I was certified. |
| 18:55.19 | [av]bani | scrambray8927: to them, it's "warez" |
| 18:55.25 | vaewyn | whereas with cisco... legally you arn't allowed to get it in any way shape or form for used equipment... you can cheat with smartnet... but.... |
| 18:55.28 | asterboy | I said yes...thanks to ebay. |
| 18:55.49 | asterboy | waht is smartnet? |
| 18:55.53 | [av]bani | vaewyn: wrong. smartnet is a support contract. you're _supposed_ to buy it for used equipment. duh! |
| 18:56.02 | asterboy | yuk |
| 18:56.02 | *** join/#asterisk b66mer (i=fwuser@blackhole.c5i.com) |
| 18:56.07 | scrambray8927 | i see |
| 18:56.12 | [av]bani | vaewyn: thats the whole POINT of smartnet |
| 18:56.13 | asterboy | no thanks |
| 18:56.21 | vaewyn | [av]bani, read the smartnet contract... doesn't apply to firmware for those phones |
| 18:56.59 | vaewyn | now in reallity... either company it doesn't matter... we all work around it |
| 18:57.17 | vaewyn | and btw... if it did apply... you also have to have 1 smartnet per phone |
| 18:57.25 | [av]bani | yes... its cheap |
| 18:57.33 | vaewyn | not if you have 2500 phones |
| 18:57.36 | [av]bani | whereas with polycom, it's illegal |
| 18:57.43 | [av]bani | that is, if you ebay them |
| 18:57.55 | Katty | Hmmhesays: come watch rocky horror at the university center with me! |
| 18:57.59 | vaewyn | It's not illegal... it's discouraged |
| 18:58.06 | scrambray8927 | k |
| 18:58.11 | [av]bani | if polycom finds out, they throw a fit |
| 18:58.21 | vaewyn | I didn't sign any agreement to buy my new phones from them |
| 18:58.24 | [av]bani | because it's "unauthorized" |
| 18:58.35 | vaewyn | and they gave me the firmware with no EULA or click-wrap |
| 18:58.40 | [av]bani | vaewyn: well gee, apply that argument to cisco then.. voila |
| 18:58.45 | vaewyn | yeah.. and no legal leg to stand on |
| 18:58.48 | nettie | hey guys, how do I enable direct extension dial within the welcome message please? Sometimes I hear something like "if you know your party extension you can dial it at any time..) |
| 18:58.55 | *** join/#asterisk backblue (n=moo@87-196-64-142.net.novis.pt) |
| 18:58.56 | [av]bani | vaewyn: thanks for justifying used cisco purchases :) |
| 18:58.57 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 18:59.02 | vaewyn | [av]bani, cisco does have an agreement when you log in to get the firmware |
| 18:59.15 | vaewyn | click-wrap |
| 18:59.17 | rbd | kind of OT, but I'll be pulling h323 network data off the wire...anyone know of an h323 parsing library I could use...or should I just use the openh323's project's header files (etc) and make my own? |
| 19:00.01 | vaewyn | Ethereal if you just want to play with the data.... something else if you are trying to eavesdrop |
| 19:00.44 | [av]bani | vaewyn: there's no sccp/sip "license" for the phone beyond the smartnet you buy. you should be complaining about the per-phone call manager license :) |
| 19:00.59 | vaewyn | who uses call manager? |
| 19:01.13 | vaewyn | crisco call mangler sucks |
| 19:01.38 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:01.51 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 19:02.45 | *** part/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 19:03.23 | asterboy | lol call mangler |
| 19:03.24 | [av]bani | apparently, millions of customers use call mangler |
| 19:03.47 | vaewyn | yes... and millions of people are below average intelligence... that doesn't mean they are right |
| 19:04.10 | vaewyn | trust me... try it for 2500+ phones... then get back to me |
| 19:04.37 | [av]bani | you have 2500+ phones? |
| 19:04.51 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 19:05.01 | vaewyn | Yep... and growing |
| 19:05.14 | nettie | vaewyn you're loaded! what are you using? |
| 19:05.16 | vaewyn | only 1400+ voip so far |
| 19:05.27 | vaewyn | loaded... hahahahah |
| 19:05.29 | [av]bani | sorta doubt you're using asterisk with 1400 phones... |
| 19:05.30 | nettie | :) |
| 19:06.02 | vaewyn | several asterisk... all feed via Ts into a norhell Option 11c |
| 19:06.39 | vaewyn | when we hit 2500 then the norhell just becomes a glorified channel bank |
| 19:06.40 | stoffell | vaewyn, using dundi ? |
| 19:06.46 | vaewyn | yes |
| 19:06.48 | brettnem | a couple Ts to a Options 11C? sounds like a small setup on the backend for all that |
| 19:06.54 | asterboy | ~norhell |
| 19:07.04 | asterboy | ~callmangler |
| 19:07.08 | vaewyn | well... 2 Ts per landing server |
| 19:07.10 | stoffell | lol |
| 19:07.10 | asterboy | ~crisco |
| 19:07.12 | jbot | [crisco] a nice slimy brand of router, suitable for cooking with. |
| 19:07.12 | [av]bani | yeah, T1 sounds fishy. you should be T3 for 2500 |
| 19:07.23 | vaewyn | 14 landing server |
| 19:07.27 | [av]bani | unless theyre not all at the same location |
| 19:07.32 | brettnem | are these business customers? |
| 19:07.38 | vaewyn | university |
| 19:07.51 | vaewyn | we are the telco :P |
| 19:07.53 | brettnem | you could do that with 10-15 T1s |
| 19:07.59 | vaewyn | for a small exchange that is :P |
| 19:07.59 | brettnem | where are you located? |
| 19:08.04 | vaewyn | SW michigan |
| 19:08.07 | brettnem | you are a CLEC? |
| 19:08.08 | [av]bani | brettnem: around 8 T1s is break even where t3 is more cost effective |
| 19:08.17 | vaewyn | no... not CLEC |
| 19:08.27 | brettnem | vaewyn: so what kind of telco are you then? |
| 19:08.31 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 19:08.32 | vaewyn | should have been... but... no one did it when the getting was good |
| 19:08.37 | brettnem | [av]bani: right.. actually, I've seen it closer to 4 T1s. |
| 19:08.48 | brettnem | [av]bani: but that's for the cost of the transport, not the hardware.. |
| 19:08.57 | vaewyn | brettnem, a grandfathered branch exchange is the only thing keeping us from being a normal business user |
| 19:09.01 | justinu | he said it's a university |
| 19:09.08 | [av]bani | hmm st2030 isnt sold in north america |
| 19:09.31 | brettnem | hmm.. I thought he said he was a telco for a university.. |
| 19:09.50 | vaewyn | We could have gotten CLEC but... they missed it when the getting was good |
| 19:10.04 | brettnem | you'd be a CLEC if it did the right things for you. |
| 19:10.19 | vaewyn | it would... we missed the boat |
| 19:10.24 | vaewyn | we are in SBC land so... |
| 19:10.27 | brettnem | so? |
| 19:10.38 | brettnem | I'm in SBC land.. I'm probably missing something here.. |
| 19:10.45 | vaewyn | Trust me... costs about 5 times as much time and money now to do the CLEC work |
| 19:10.49 | brettnem | btw, who isn't in SBC land? please stand up? |
| 19:10.53 | vaewyn | use to be free |
| 19:10.56 | vaewyn | hehehe |
| 19:10.57 | vaewyn | true |
| 19:11.04 | brettnem | vaewyn: I've been building CLECs for about 10 years now |
| 19:11.05 | [av]bani | no point in going CLEC unless you plan to be facilities based. and that is massive capital expenses. |
| 19:11.16 | [av]bani | just going CLEC for resale isnt worth it |
| 19:11.21 | brettnem | [av]bani: agreed |
| 19:11.41 | brettnem | I don't know if I'd say it's massively capital intensive, but it is capital intensive |
| 19:11.41 | vaewyn | We don't resale... we are the telco for campus :} and owned by the campus |
| 19:11.44 | nettie | http://en.wikipedia.org/wiki/CLEC |
| 19:12.01 | nettie | just for the other users :) |
| 19:12.02 | [av]bani | brettnem: well, ss7 switches arent cheap. nor is colo |
| 19:12.03 | brettnem | vaewyn: you buy PSTN access from someone |
| 19:12.12 | [av]bani | and the lawyers arent cheap either |
| 19:12.17 | brettnem | [av]bani: depends who you are. :) |
| 19:12.21 | brettnem | and who you know. :) |
| 19:12.28 | [av]bani | if youre lucky, yes |
| 19:12.36 | brettnem | and who said ss7 is a requirement? don't let them brainwash you. |
| 19:12.37 | [av]bani | if youre entering unfamiliar areas, no |
| 19:12.38 | vaewyn | brettnem, SBC lands us T3... and we also have a couple peering agreements with local VoIP providers |
| 19:12.55 | vaewyn | SS7 isn't a big deal... $$$... but easy to do |
| 19:13.04 | vaewyn | easier to just stay on the T3s though :P |
| 19:13.05 | [av]bani | brettnem: well, you can go isdn but they will look at you funny and you'll lose functionality |
| 19:13.18 | vaewyn | PRI baby! |
| 19:13.19 | vaewyn | hehehe |
| 19:13.32 | dpryo | I have a (rather big) ss7-switch for sale.. |
| 19:13.32 | brettnem | [av]bani: do you know anyone interconnecting with PRI? |
| 19:13.40 | brettnem | dpryo: so do I, what do you have? :) |
| 19:13.49 | [av]bani | brettnem: only for ptp t1's, not for voice |
| 19:14.03 | brettnem | with PRI? |
| 19:14.04 | dpryo | brettnem: 5ESS :) |
| 19:14.14 | brettnem | Ah,well mine's small compared to that.. |
| 19:14.16 | [av]bani | though really, most clecs prefer to use atm for bridging circuits these days... |
| 19:14.24 | brettnem | I'm selling a Tekelec T7000 |
| 19:14.30 | brettnem | eww! atm |
| 19:14.31 | [av]bani | though its really the ilecs who dictate that |
| 19:14.41 | brettnem | [av]bani: where are you located? |
| 19:14.51 | [av]bani | brettnem: atm fits neatly into the tdm model the ilecs are comfortable with. |
| 19:14.52 | b66mer | anybody setup asterisk to answer for their gizmo account? know why i am getting an error on the asterisk console: 408 "Request Timeout" |
| 19:14.58 | brettnem | I don't know what you mean about a ptp T1 with PRI not for voice. |
| 19:14.58 | vaewyn | heck.. the only reason we have this setup is because we are HEAVILY grandfathered :P |
| 19:15.10 | [av]bani | ilecs are tdm mindset. all this newfangled packet switching confuses them |
| 19:15.23 | brettnem | [av]bani: Not SBC.. they have a voip tarriff |
| 19:15.49 | [av]bani | brettnem: that's very new. qwest has a voip tariff too, but they cant actually deploy it anywhere yet because they can't figure out how to program their switches yet. |
| 19:16.07 | brettnem | [av]bani: where are you located? |
| 19:16.10 | [av]bani | oregon |
| 19:16.16 | [av]bani | qwest territory |
| 19:16.20 | brettnem | texas, sbc here |
| 19:16.25 | brettnem | or whoever they are now |
| 19:16.32 | vaewyn | we have voice via SBC... data via private fiber to michnet... and a couple backup data Ts from Qwest |
| 19:16.59 | vaewyn | and soon I am hoping to hit JerJer up for private fiber to him :P |
| 19:17.11 | [av]bani | qwest has voip on the tariff list, but they don't know how to deploy it. and wont till they finish sending all their techs to voip classes. |
| 19:17.15 | brettnem | I've got a load of interconenction trunks to SBC all over texas |
| 19:17.49 | [av]bani | its quite funny and sad |
| 19:18.02 | brettnem | what is? |
| 19:18.10 | [av]bani | qwest having tariffs they cant deploy |
| 19:18.18 | [av]bani | because they lack the technical expertise |
| 19:18.21 | *** join/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net) |
| 19:18.21 | brettnem | it's just political.. not so they can use it. |
| 19:18.42 | [av]bani | well, they have "OMG VOIP NOW!1!!111" on their website |
| 19:19.00 | [av]bani | but then thats just qwest :) |
| 19:19.08 | brettnem | nah.. it's more like "we wrote the book on voip and if you don't do it like this you f-d" |
| 19:19.26 | [av]bani | actually the first round of "voip" they planned to deploy h323... |
| 19:19.58 | [av]bani | till they realized no end user phones speak h323 anymore |
| 19:20.23 | brettnem | I don't think the original voip tariffs where intended for end user connections. |
| 19:20.23 | [av]bani | i dont think theyve even settled on CPE stuff yet |
| 19:20.36 | *** part/#asterisk vaewyn (i=ejabberd@jabber.andrews.edu) |
| 19:20.50 | [av]bani | though whatever they pick will be unbelievably shitty as usual |
| 19:21.33 | b66mer | if I want to allow SIP calls from the internet to my asterisk... is 5060/udp and 10000-20000/udp enough/everything? |
| 19:21.41 | b66mer | to pass my firewall |
| 19:21.46 | [av]bani | asterisk isnt compatible with the interweb |
| 19:22.13 | b66mer | huh |
| 19:22.16 | kardecallan | _Paulo_ I have already installed the unicall module and mfc-5c protocol is working well. |
| 19:22.22 | jaike | b66mer: 4569 for iax |
| 19:22.22 | b66mer | what do you mean? |
| 19:22.23 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 19:22.32 | b66mer | yea... I need it for SIP only |
| 19:22.50 | jaike | that should work |
| 19:22.52 | [av]bani | rtp ports are up to your CPE... |
| 19:23.01 | [av]bani | some you can configure, others you cant |
| 19:23.23 | b66mer | I am having a problem where when I call a GIZMO number that forwards to my asterisk box... but it seems to work intermittently |
| 19:23.37 | kardecallan | The problem was with TDM400P plate. |
| 19:23.40 | b66mer | my asterisk is registered to the gizmo account |
| 19:23.57 | b66mer | 75% of time it goes to voice mail |
| 19:24.02 | _Paulo_ | kardecallan, good to hear everthing is working now. |
| 19:24.10 | b66mer | 25% I receive the call |
| 19:24.27 | b66mer | any ideas? |
| 19:24.30 | _Paulo_ | kardecallan, the tdm400p was faulty? |
| 19:25.53 | *** part/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 19:25.56 | *** join/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 19:26.30 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 19:27.42 | b66mer | buhler? |
| 19:27.44 | b66mer | buhler? |
| 19:30.30 | Darwin_35 | tomany new flybynight voip providers... |
| 19:31.04 | *** part/#asterisk rnovotny22 (n=bob@198.57.19.126) |
| 19:34.16 | justinu | anyone ever upgrade a CentOS 4.2 box to kernel 2.6.13? |
| 19:34.21 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-193-10.telkomadsl.co.za) |
| 19:34.26 | justinu | currently, they run 2.6.9 and apparently I need 2.6.13 to use the RTC |
| 19:34.47 | justinu | (according to ztdummy.c) |
| 19:35.46 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 19:36.43 | *** join/#asterisk jmacz (n=jmacz@201.244.192.16) |
| 19:37.23 | trixter | or what asterisk will crash? |
| 19:38.01 | justinu | good question |
| 19:38.18 | trixter | but the better question is will it crast with 2.6.13 |
| 19:38.19 | trixter | :P |
| 19:38.25 | trixter | er crash |
| 19:38.39 | *** join/#asterisk _andre (n=andre@fosforo.k8.com.br) |
| 19:38.47 | _andre | hello |
| 19:39.02 | justinu | i've done kernel upgrades in the 1.4 days, but that was with slackware |
| 19:39.08 | justinu | none of this package management stuff |
| 19:39.13 | justinu | kinda worried about the whole thing |
| 19:39.23 | _andre | is it possible to have the general sip configs in sip.conf and only the users in a mysql table using extconfig.conf? |
| 19:40.36 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:40.40 | nettie | justinu |
| 19:40.43 | nettie | uhmm |
| 19:41.06 | qseek | mog_work: hi I had a question on app_amd |
| 19:41.07 | nettie | I'm running centos 3.6 |
| 19:41.12 | nettie | and also 4.2 |
| 19:41.19 | justinu | k |
| 19:41.21 | nettie | you dont need to recompile anything imho |
| 19:41.28 | qseek | which file is the one I should use, app_amd or app_amd2 |
| 19:41.34 | qseek | there are two attachments |
| 19:41.36 | justinu | the latest packages I'm seeing are kernel 2.6.9-34 tho |
| 19:41.40 | justinu | i need 2.6.13 |
| 19:41.43 | uwe | hello, im trying to test asterisk , but i dont have a mic , so i want to redirect testing sound to /dev/mixer, which i suppose is the mic device, right ? |
| 19:42.03 | justinu | just get a mic |
| 19:42.08 | justinu | or a phone |
| 19:42.28 | terrapen | anyone using SER here? |
| 19:42.31 | nettie | justinu for what reason you need 2.6.13? |
| 19:42.35 | nettie | I dont see th epoint |
| 19:42.42 | scrambray8927 | does anyone know if major companies offer VoIP termination for IAX like Verizon, AT&T or Comcast? |
| 19:42.44 | nettie | just downlaod and install zapata |
| 19:43.05 | justinu | nettie: according to ztdummy.c, USE_RTC should be enabled w/ kernel version 2.6.13 |
| 19:43.12 | nettie | in 2.4 it uses usb-uhci, zaptel and ztdummy |
| 19:43.38 | justinu | excerpt from ztdummy.c: http://pastebin.ca/47620 |
| 19:43.38 | nettie | on 2.6 somethign is changed.. you no longer need ztdummy I think |
| 19:43.39 | nettie | uhmm |
| 19:43.44 | justinu | no, i definitely need ztdummy |
| 19:43.48 | nettie | lemme doubleck |
| 19:43.52 | justinu | and i'm using it now, and it doesn't work right. |
| 19:44.08 | nettie | hangon I'm sure I reado somethign on the matter |
| 19:44.11 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 19:44.39 | justinu | the problem I have is that when SIP channels are in a Meetme conf, the latency gets worse and worse as the call progresses |
| 19:44.43 | justinu | eventually it's unusable |
| 19:44.48 | justinu | 2 second latency |
| 19:44.49 | justinu | etc. |
| 19:45.11 | justinu | apparently having ztdummy "USE_RTC" is the solution, but the reason my ztdummy isn't using it is because kernel is not 2.6.13 |
| 19:45.29 | justinu | i mean, i guess i could just force it to do it |
| 19:45.40 | justinu | but there's probably a reason they had that #if in there |
| 19:46.11 | nettie | just |
| 19:46.15 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:46.46 | nettie | justinu you're right.. on 2.6 it doesnt use uhci anymore but needs the internal timer |
| 19:46.47 | nettie | The kernel has to be recompiled with CONFIG_CRC_CCITT=y or CONFIG_CRC_CCITT=m. To do this, run 'make menuconfig' and change it under 'Library Routines -> CRC-CCIT functions'. If you compiled it as a module, |
| 19:46.48 | nettie | load it, otherwise reboot. If the problem persists, enable RTC in the kernel. |
| 19:46.52 | [av]bani | * just needs to ditch ztdummy and use other timers |
| 19:47.13 | justinu | i've got CRC_CCITT in there |
| 19:47.28 | justinu | "other timers" |
| 19:47.32 | justinu | like a zap card? |
| 19:47.33 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:47.35 | [av]bani | no |
| 19:47.37 | nettie | justinu is it compiled? |
| 19:47.49 | justinu | it's a module |
| 19:47.52 | nettie | laode it |
| 19:47.54 | nettie | load it |
| 19:47.58 | nettie | and try |
| 19:48.10 | justinu | it's already loaded |
| 19:48.16 | justinu | zaptel loaded it |
| 19:48.18 | nettie | and you're having problems? |
| 19:48.24 | nettie | no.. |
| 19:48.30 | nettie | you had problems because you didnt have it beforE? |
| 19:48.36 | nettie | didnt use it before |
| 19:48.37 | justinu | it's always been loaded |
| 19:48.41 | nettie | uhmmm |
| 19:48.44 | *** join/#asterisk ToTo (n=ToTo@host31-145.pool874.interbusiness.it) |
| 19:48.57 | justinu | # lsmod | grep zap |
| 19:48.57 | justinu | zaptel 196740 5 ztdummy |
| 19:48.57 | justinu | crc_ccitt 6081 1 zaptel |
| 19:49.07 | nettie | should be fine.. |
| 19:49.16 | justinu | meetme is useless on this machine :( |
| 19:49.16 | nettie | and it stil llags? |
| 19:49.18 | justinu | yep |
| 19:49.21 | nettie | uhmm |
| 19:49.33 | nettie | what version are u using? 2.6.9-34? |
| 19:49.43 | nettie | yeah |
| 19:49.45 | nettie | got ur msg |
| 19:49.46 | [av]bani | justinu: like hpet |
| 19:50.06 | justinu | [av]bani: I am unfamiliar |
| 19:50.18 | nettie | just |
| 19:50.21 | nettie | justinu |
| 19:50.31 | GerbilNut | anyone know if you can send the caller id in a dial statement? |
| 19:50.36 | nettie | is genrtc loaded? |
| 19:50.43 | b66mer | anybody know of a good commercially oriented provider of phone services supporting IAX or SIP, asterisk friendly? |
| 19:50.48 | justinu | nettie: no |
| 19:51.03 | nettie | and module rtc is not there? |
| 19:51.12 | justinu | nope |
| 19:51.26 | nettie | CONFIG_HPET_EMULATE_RTC=y |
| 19:51.26 | nettie | CONFIG_RTC=y |
| 19:51.32 | nettie | mine is buil tinto the kernel |
| 19:51.35 | justinu | hmm |
| 19:51.41 | nettie | that's /boot/config-2.6.9-34.EL |
| 19:51.43 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:51.43 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:51.48 | justinu | ah |
| 19:51.58 | justinu | actually, this system is running 2.6.9-22.ELsmp |
| 19:52.01 | nettie | ah |
| 19:52.03 | nettie | hanogn |
| 19:52.03 | [av]bani | justinu: http://www.intel.com/hardwaredesign/hpetspec.htm |
| 19:52.04 | justinu | i suppose I need to move up to 34 |
| 19:52.05 | nettie | I got tht too |
| 19:52.07 | nettie | w8 |
| 19:52.24 | nettie | <PROTECTED> |
| 19:52.24 | nettie | CONFIG_HPET_EMULATE_RTC=y |
| 19:52.24 | nettie | CONFIG_RTC=y |
| 19:52.24 | nettie | CONFIG_SENSORS_RTC8564=m |
| 19:52.24 | nettie | CONFIG_SND_RTCTIMER=m |
| 19:52.24 | nettie | same |
| 19:52.59 | justinu | same results here |
| 19:53.09 | justinu | however, no rtc modules are loaded according to lsmod |
| 19:53.34 | nettie | it's builtin |
| 19:53.53 | nettie | see if you have /proc/driver/rtc |
| 19:54.06 | nettie | or /proc/sys/dev/rtc |
| 19:54.23 | justinu | yep, those exist |
| 19:54.32 | justinu | does that mean I can recompile ztdummy.c w/ USE_RTC? |
| 19:54.46 | justinu | even though it seems to indicate 2.6.13 is required? |
| 19:54.52 | nettie | sue |
| 19:54.53 | nettie | sure |
| 19:54.55 | nettie | I think so |
| 19:54.56 | nettie | try |
| 19:54.57 | nettie | please |
| 19:55.00 | nettie | it's there |
| 19:55.02 | justinu | ok nettie, i'll give it a shot... thanks for the help |
| 19:55.05 | nettie | it's in the kernel |
| 19:55.10 | nettie | np |
| 19:55.25 | nettie | I really hope it works for u |
| 19:55.43 | generalhan | anyone using 1.2.6 yet ? |
| 19:55.50 | nettie | I do |
| 19:56.04 | generalhan | any noticable differeneces for .5 ? |
| 19:56.14 | generalhan | for = from |
| 19:56.43 | nettie | well I'm not very used to asterisk, kinda new.. and I have a very limited usage |
| 19:56.44 | generalhan | im still using 1.2.1 and i was about to do the upgrade to 1.2.5 this weekend and i saw 1.2.6, so i might as well just use that one ! |
| 19:56.54 | nettie | for what I do it's fine |
| 19:57.04 | nettie | but my usage is very basic |
| 19:57.26 | nettie | you should upgrade if you have problems or if you need additionals features |
| 19:57.27 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 19:57.35 | nettie | check in the changelog if you find anything interesting |
| 19:57.44 | nettie | or a good reason to upg |
| 19:59.19 | justinu | nettie: no good... setting USE_RTC in ztdummy.c broke all my async RTP now |
| 19:59.25 | justinu | no indications before answer :( |
| 19:59.33 | nettie | uhmmm |
| 19:59.34 | bkw_ | broken? |
| 19:59.37 | bkw_ | WHAT? |
| 19:59.38 | bkw_ | NEVER! |
| 19:59.42 | justinu | :P |
| 19:59.53 | nettie | are you just doing make linux26 right? |
| 19:59.58 | justinu | just make install |
| 20:00.06 | *** join/#asterisk linlin (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 20:00.18 | justinu | but it works fine for everything except meetme (if I don't define USE_RTC) apparently |
| 20:00.29 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 20:01.00 | nettie | it's very strange and it's getting pretty advanced this topic.. it's strange ppl using centos/rhel didnt complain about that.. |
| 20:01.04 | nettie | very very strange |
| 20:01.11 | justinu | yeah... i dunno what the deal is |
| 20:01.19 | nettie | lemme see if I can find some more infor for you |
| 20:01.22 | justinu | someone advised me to apply a patch to ztdummy |
| 20:01.26 | justinu | patch 5971 |
| 20:01.27 | [av]bani | anyone know how to disable call waiting on polycoms? |
| 20:01.33 | nettie | you did tht? |
| 20:01.49 | justinu | but I realized that the patch wouldn't be affective unless #define USE_RTC is in the source |
| 20:01.54 | justinu | because it's all inside ifdefs |
| 20:02.14 | justinu | yeah, i did that... and I also applied the latest async RTP patches to 1.2.6 |
| 20:02.18 | justinu | as advised |
| 20:02.31 | linlin | im looking for a cheap provider of 800 (or any toll-free) numbers for use on voip, any suggestions? reccomendations? |
| 20:02.47 | redondos | Where can I read about Asterisk outputting some statistics about incoming calls with caller ID, what they did on the IVR, and so on? |
| 20:03.12 | Hmmhesays | they put lettuce on my quarter pounder |
| 20:03.12 | generalhan | what are these zaptel-1.2.5-patch and asterisk-1.2.6-patch files on digiums ftp ? anyone know ? |
| 20:03.29 | scrambray8927 | i'm pining some voip service provider's ip's to get the latency between my * box and their servers, each ping gives me a different response (5ms, 6ms, 22ms) how many should I try before I average them together? |
| 20:03.50 | justinu | nettie: the other patch is 5374 |
| 20:04.12 | generalhan | hmmhesays: you got a QP/Big Mac combination |
| 20:04.30 | Hmmhesays | i guess so |
| 20:04.35 | generalhan | QP/BigMac hybrids |
| 20:04.40 | Hmmhesays | freaky |
| 20:04.41 | generalhan | they are all the rage no-a-daws |
| 20:04.43 | generalhan | lol |
| 20:04.55 | Hmmhesays | the times they are a changing |
| 20:05.00 | generalhan | hahaha |
| 20:05.01 | generalhan | yea they are |
| 20:05.15 | generalhan | hmmhesays: have you upgraded to 1.2.6 yet ? |
| 20:05.28 | Hmmhesays | yeah I just installed 1.2.6 on a new server |
| 20:05.38 | Hmmhesays | it kind of made my loins tingle |
| 20:05.59 | generalhan | well im going to upgrade my production server tomorrow evening from 1.2.1 and im a bit nervous ! lol |
| 20:06.04 | Hmmhesays | why? |
| 20:06.17 | generalhan | ive never done an * upgrade before ... only installed newer versions on new servers |
| 20:06.40 | Hmmhesays | i wouldn't worry about it |
| 20:06.43 | generalhan | and im looking at digiums ftp and there are patch files to go with asterisk-1.2.6, and i dont know what the hell those are for |
| 20:06.55 | generalhan | do i need them ? |
| 20:07.01 | Hmmhesays | well you could always read what they are fo |
| 20:07.16 | generalhan | but then i have to read ! lol, yea i could do that |
| 20:07.17 | Hmmhesays | i am reading the most hilarious totalfark thread right now |
| 20:07.22 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:07.26 | Hmmhesays | i *almost* shot soda out my nose |
| 20:07.32 | generalhan | im in the middle of a huge wiring job for my boss right now |
| 20:07.35 | generalhan | HAHA |
| 20:07.54 | Hmmhesays | put a dialer on his phone so every time he tries to call his girlfriend he gets his wife |
| 20:08.21 | generalhan | Hahahahaha |
| 20:08.50 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:08.58 | Hmmhesays | i'm evil, i know |
| 20:09.07 | *** join/#asterisk Mauro__ (n=mauro@oliver.altascumbres.cl) |
| 20:09.11 | iDunno | EEEEEEEVIL! |
| 20:09.15 | iDunno | don't be evil! |
| 20:09.38 | jbalcomb | Is it ok if i set 'pbx_ael.so' to noload? |
| 20:09.56 | file | Hmmhesays & generalhan: 1.2.7 is going to be released soon fyi |
| 20:10.06 | Hmmhesays | cool |
| 20:10.07 | generalhan | by tomorrow ? |
| 20:10.11 | generalhan | lol |
| 20:10.11 | eric_hill | justinu: Do you have ntpd running? |
| 20:10.15 | file | potentially, yes |
| 20:10.23 | Hmmhesays | we came up with a wonderful band name last night in our drunken stupor |
| 20:10.30 | Hmmhesays | "I did what?!?" |
| 20:10.31 | generalhan | little tiny wierd things are starting to mappen on my system so i want to upgrade first change i get |
| 20:10.39 | justinu | eric_hill: not at the moment |
| 20:10.42 | scrambray8927 | has anyone used VoIPJET.com? |
| 20:10.44 | file | well you can upgrade now and see... change to 1.2.7 won't be bad |
| 20:10.50 | Hmmhesays | yeah I use voipjet fairly heavily |
| 20:10.52 | generalhan | sweetness |
| 20:11.01 | eric_hill | justinu: Do you think that clock skew might be part of the problem? |
| 20:11.05 | scrambray8927 | Hmmhesays, is it free |
| 20:11.19 | justinu | eric_hill: never thought about it... worth a shot, i suppose. |
| 20:11.20 | Katty | oooh, a file |
| 20:11.20 | generalhan | file: when i upgrade (talking only about the * part not zaptel) i just have to shut it down and do a make install again right ? |
| 20:11.21 | Katty | :>>> |
| 20:11.22 | Mauro__ | hmm how can I use GET DATA on a outgoing call? |
| 20:11.33 | generalhan | easy as that ? |
| 20:11.49 | file | generalhan, aye... and make sure you don't have any leftover modules that might conflict and cause trouble... but it'll warn you about that |
| 20:11.51 | file | Katty, !!!!!!!!!!!!! |
| 20:11.53 | Hmmhesays | is voipjet free? nice quesiton dude |
| 20:12.20 | scrambray8927 | Hmmhesays, well on their site it says "Free IAX Account" but then it says .25$ below it |
| 20:12.37 | generalhan | ok i have Hmmhesays & files blessing on the upgrade ... i have no worries now ! |
| 20:12.42 | Hmmhesays | grand |
| 20:12.58 | generalhan | haha |
| 20:13.22 | file | scrambray8927, it says - and I quote "Instant IAX(tm) Test Account Free $0.25 usage Click Here" |
| 20:13.57 | Katty | iDunno: yeah, i tend to not like random guys hugging on me |
| 20:14.04 | Katty | but i make an exception for some people |
| 20:14.16 | justinu | meow |
| 20:14.21 | asterboy | pussy? |
| 20:14.22 | file | like me? |
| 20:14.36 | Mauro__ | hmm how can I use GET DATA on a outgoing call made by asterisk?? |
| 20:14.40 | scrambray8927 | file, thanks |
| 20:14.43 | Katty | file: just like you, dear. |
| 20:14.51 | Qwell[] | eww |
| 20:15.00 | Qwell[] | Katty: You don't know where he's been |
| 20:15.13 | Katty | Qwell[]: i've shared his internet connection |
| 20:15.18 | Katty | Qwell[]: how much more personal can ya get? |
| 20:15.20 | generalhan | wow... file comes out then everyone comes outta the woodwork , Katty and Qwell[] are here now ... |
| 20:15.31 | Katty | file's a lovely boy. |
| 20:15.33 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-133-128.red.bezeqint.net) |
| 20:15.38 | justinu | it's kinda like high school |
| 20:15.39 | file | yay lovely |
| 20:15.41 | asterboy | get a hotel room |
| 20:15.45 | Katty | we did. |
| 20:15.50 | asterboy | do tell |
| 20:15.52 | Qwell[] | ooo |
| 20:15.53 | Katty | never. |
| 20:16.33 | iDunno | :) |
| 20:16.57 | eric_hill | Katty: did you take pictures? ;) |
| 20:17.03 | asterboy | I gotta get my daughter on here. |
| 20:17.12 | justinu | w00t |
| 20:17.14 | asterboy | Might actually get my forwarding problems solved. |
| 20:17.15 | eric_hill | asterboy: that would be a BAD idea. |
| 20:17.16 | justinu | fresh meat |
| 20:17.17 | file | noes! no pictures |
| 20:17.22 | Katty | no, there were no pictures. |
| 20:17.30 | asterboy | lol |
| 20:17.35 | Hmmhesays | wow my should is farked up |
| 20:17.36 | jbalcomb | Can someone look at the errors I get on a 'reload' and advise me on how to deal with or get rid of them please? http://pastebin.com/631411 |
| 20:17.37 | iDunno | just a full length feature film? *grin* |
| 20:17.41 | Hmmhesays | I think i'm going to buy a bass tonigght |
| 20:17.48 | Qwell[] | Hmmhesays: tasty |
| 20:18.02 | jbalcomb | Hmmhesays: bass ale, bass fish, or bass guitar? |
| 20:18.07 | Hmmhesays | i heard ice ice baby last night at the bar and I got the urge to play it |
| 20:18.10 | file | jbalcomb, they are pretty self explanitory... |
| 20:18.11 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:18.31 | asterboy | or Bass as in big as $1000s of dollars instrument. |
| 20:18.37 | jbalcomb | file that is the least helpful thing I can imagine you wasting your time telling me. |
| 20:18.49 | Katty | uh, no. |
| 20:18.52 | Katty | no full length films either. |
| 20:19.09 | scrambray8927 | Can anyone offer some advice/experience? I'm deciding what VoIP termination account to go with with Asterlink, JunctionNetworks, Teliax or VoicePulse Connect? |
| 20:19.22 | eric_hill | Katty: where's the fun in that? |
| 20:19.33 | file | jbalcomb, then ask a a better question? if a module can't find it's own configuration file, then chances are it'll complain... so if you're not using the module - then don't load it |
| 20:19.34 | generalhan | i used VoicePulse and would tell everyone that has even heard of them to kill themslves now |
| 20:19.39 | asterboy | jbalcomb, looks like your missing a bunch of config files in /etc/asterisk. |
| 20:19.42 | file | as for zaptel stuff some of it can not be changed by a reload, so it tells you |
| 20:19.45 | iDunno | Katty: the fact that you qualified that with "no full length films" *suggests* that there are some that aren't full length... |
| 20:19.46 | Katty | eric_hill: dun you worry, there was plenty fun to be had. |
| 20:19.47 | Damin | jbalcomb: Read the following document, which will tell you how to remove unused / unconfigured modules from Asterisk so you can get rid of those messages: http://www.voip-info.org/wiki/view/Asterisk+Slimming |
| 20:19.48 | file | and same applies for other modules |
| 20:19.59 | Katty | right file? |
| 20:19.59 | eric_hill | Katty: I can only hope ;) |
| 20:20.11 | file | ooh it was most excellent |
| 20:20.12 | file | Katty is the best |
| 20:20.22 | iDunno | I can believe that. |
| 20:20.27 | scrambray8927 | generalhan, could you give me some specifics about voicepulse that you weren't happy with? |
| 20:20.35 | jbalcomb | Damin thanks, i'll have a look at that. |
| 20:21.17 | jbalcomb | asterboy yeah, this system was built before i got here so i'm not sure if we should be using things were not or if the previous admin just didn't care about being tidy. |
| 20:22.45 | Hmmhesays | what a clusterfuck |
| 20:23.25 | eric_hill | I built a virtual cluster in VMware once and named it "fuck". <g> |
| 20:23.38 | Hmmhesays | wow I havne't actually |
| 20:24.51 | asterboy | jbalcomb, what I suggest you do is this: 1. Backup you /etc/asterisk directory *very important* for obvious reasons. |
| 20:24.55 | Hmmhesays | ARGH, i almost hung up the phone on these morons |
| 20:25.12 | asterboy | 2. goto your asterisk source directory and do make samples. |
| 20:25.29 | Hmmhesays | someone needs to write chan_strangle |
| 20:25.29 | asterboy | This will overwrite your /etc/asterisk directory which is pretty fucking annoying. |
| 20:25.38 | Hmmhesays | but it makes backups |
| 20:25.54 | asterboy | 3. Make a backup of the new sample conf files in /etc/asterisk, call it samples. |
| 20:26.14 | jbalcomb | asterboy i have the configs in svn and am working on building a new server from scratch by only bringing over what I really need from this server. |
| 20:26.19 | asterboy | 4. now copy from your original backup into the /etc/asterisk directory. |
| 20:26.31 | justinu | enabling ntpd seems to not have helped |
| 20:26.39 | justinu | after 4 minutes in a meetme, latency is above 1 second |
| 20:26.39 | asterboy | This will overwrite the files with the original config but put in place the samples. |
| 20:26.48 | jbalcomb | asterboy gotcha, no make sure i have the added configs that might be missing. |
| 20:26.55 | asterboy | yep |
| 20:27.23 | asterboy | then just look at the output again and play with the ones that complain. |
| 20:27.31 | asterboy | i.e. Dundi etc. |
| 20:27.36 | jbalcomb | asterboy ok, i'll go ahead and do that so i can get a cleaner view of the logs and reduce my 'what the hell is that error?' time. ;) Thanks. |
| 20:27.58 | *** join/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net) |
| 20:28.14 | asterboy | Ya I did that and the only message I get now is Mar 30 13:28:07 WARNING[21345]: res_musiconhold.c:1029 load_moh_classes: The old musiconhold.conf syntax has been deprecated! Please refer to the sample configuration for information on the new syntax. |
| 20:28.42 | asterboy | I tried the new syntax but then my MOH stopped working. |
| 20:29.10 | asterboy | so I just put it back for now and live with one red W*WARNING* Will Robinson message. |
| 20:30.01 | asterboy | Right now I really need to get * working as a forwarding service. |
| 20:30.16 | asterboy | It's realy wasy when the telco provides the ol' *72... |
| 20:30.26 | *** join/#asterisk _asr (i=asr@pimpbox.latency.net) |
| 20:30.46 | asterboy | ...but in my case, I'm the one who needs to switch people from one ZAP channel to another. |
| 20:30.58 | asterboy | So far, * does not seem to be able to do it. |
| 20:31.09 | justinu | say what? |
| 20:31.42 | [TK]D-Fender | justinu : just ordered an IP 301 & 501 for home / consulting :) |
| 20:31.50 | justinu | whee |
| 20:32.10 | [TK]D-Fender | asterboy : You can do it, you just need to take up 2 channels at a time for the duration of the call. |
| 20:32.12 | justinu | got some GXP2000's for the bathroom ;) |
| 20:32.17 | justinu | bathrooms |
| 20:32.19 | *** join/#asterisk noky (n=Noky@200.69.211.18) |
| 20:32.21 | noky | hi |
| 20:32.27 | [TK]D-Fender | justinu : Strangely appropriate ;) |
| 20:32.31 | justinu | yep |
| 20:32.46 | nettie | hey [TK]D-Fender :) |
| 20:32.50 | nettie | thanx for today |
| 20:32.51 | *** join/#asterisk saftsack (n=saftsack@p54A7E996.dip.t-dialin.net) |
| 20:32.53 | asterboy | [TK]D-Fender, I have the ZAP channels to do that, but HOW is that done? |
| 20:33.04 | Qwell[] | asterboy: dial()? |
| 20:33.14 | justinu | send the call back down another zap channel |
| 20:33.29 | asterboy | yes, I have the dial plans setup, but I must have something messed up in the config. |
| 20:33.47 | justinu | asterisk /can/ do this |
| 20:33.50 | justinu | just FYI |
| 20:34.06 | noky | i want to know how configurate a basical extend that allow to comunicate two user SIP (i have an sip users registered right now in ast*) |
| 20:34.16 | noky | a basical extension.conf* |
| 20:34.18 | noky | please |
| 20:34.20 | *** join/#asterisk iq (n=iq@71-214-1-59.omah.qwest.net) |
| 20:34.23 | asterboy | When I pickup my Polycom IP600 and hit [Forward] and enter the number I want to call, it gives me an error message. |
| 20:34.26 | [TK]D-Fender | asterboy : its all dial-plan... how is it SUPPOSED to work in your mind? (what would you have them do to enable/disable it?) |
| 20:34.40 | justinu | forward works on my polycom ip601 |
| 20:34.47 | [TK]D-Fender | asterboy : what gives the error? * or Polycom? |
| 20:34.53 | asterboy | * |
| 20:34.59 | asterboy | Here is the pastebin: http://pastebin.ca/47006 |
| 20:35.02 | [TK]D-Fender | asterboy : What error? |
| 20:35.16 | justinu | that's a dialplan |
| 20:35.17 | justinu | not an error |
| 20:35.21 | asterboy | NOTICE[12947]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 20:35.24 | scrambray8927 | What VoIP provider do you guys use? |
| 20:35.41 | justinu | that message comes from ast_request |
| 20:35.42 | asterboy | justinu, the error message is in the description of the pastebin. |
| 20:35.57 | justinu | for whatever reason ast_request won't create your channel |
| 20:35.57 | asterboy | added the dial plan to see if you can find the screw up |
| 20:36.05 | noky | i want to know how configurate a basical extension that allow to comunicate two user SIP (i have an sip users registered right now in ast*) |
| 20:36.08 | noky | lpease |
| 20:36.09 | justinu | is the channel you're trying to seize on in use? |
| 20:36.17 | asterboy | no |
| 20:36.23 | justinu | you positive? |
| 20:36.28 | asterboy | yes |
| 20:36.42 | Katty | where's a good vacationy spot? |
| 20:36.43 | asterboy | I have 3 incoming FXO Lines. |
| 20:36.46 | justinu | what happens if you use the exact same dial string, but use it on an exten like "123" |
| 20:36.49 | justinu | dial 123, what happens? |
| 20:36.58 | justinu | katty: i liked thailand |
| 20:37.07 | justinu | hong kong was dope too |
| 20:37.42 | asterboy | I get a message from the telco that the phone number I'm trying to reach is not in service. |
| 20:37.56 | justinu | that's whack, yo |
| 20:38.04 | Nugget | I love visiting japan, but it's so ungodly expensive. |
| 20:38.23 | [TK]D-Fender | asterboy : ok, thats because you have 1 line associated with each registration on your IP601 and it tries to send the call out the same reg it comes in on. Since ther is 1 outside line associated with each reg (your dialout X''s) there is no free line! |
| 20:38.35 | asterboy | ah! |
| 20:38.38 | Nugget | my last two vacations were tokyo and nyc. I need to go someplace cheap next. |
| 20:38.38 | justinu | really? |
| 20:38.41 | asterboy | that would make total sense. |
| 20:38.53 | justinu | all hail d-fender |
| 20:38.58 | [TK]D-Fender | asterboy : Call comes in on line 2 and the polycom dials it out on the reg that only has access to line 2.... |
| 20:39.08 | asterboy | got it... |
| 20:39.16 | asterboy | now I have something to chew on. |
| 20:39.22 | [TK]D-Fender | asterboy : you need to rethink your line strategy |
| 20:39.28 | asterboy | yep |
| 20:39.28 | [av]bani | [TK]D-Fender: how do you disable call waiting on the polycom? |
| 20:39.31 | justinu | heh |
| 20:39.38 | justinu | you can't, apparently |
| 20:39.43 | [av]bani | nice |
| 20:39.43 | justinu | same thing with eyebeam |
| 20:39.49 | [TK]D-Fender | [av]bani : the "beep" or the ability to take more than 1 call period? |
| 20:39.52 | justinu | beep |
| 20:39.53 | [av]bani | [TK]D-Fender: the beep |
| 20:39.59 | [TK]D-Fender | I think you can... 1 sec |
| 20:40.01 | [av]bani | it's incredibly annoying |
| 20:40.02 | justinu | i looked |
| 20:40.07 | justinu | couldn't find any option related to i |
| 20:40.09 | justinu | t |
| 20:40.13 | justinu | unless it's very new in 1.6.5 |
| 20:40.13 | [av]bani | more astounding excellence from polycom? |
| 20:43.15 | *** join/#asterisk rikishi (n=shigueta@69.65.149.190) |
| 20:43.15 | asterboy | so in sip.conf I need to make multiple lines per registration? |
| 20:43.31 | [TK]D-Fender | hmmm guess not... |
| 20:43.51 | redondos | What should I use in extensions.conf to perform an action when the line doesn't answer? |
| 20:43.56 | [TK]D-Fender | asterboy : many ways to fix this... |
| 20:43.59 | redondos | It is 't'? |
| 20:44.01 | generalhan | youre trying to make the stupid call waiting beep go away ? |
| 20:44.10 | justinu | no, t is timeout waiting for input |
| 20:44.12 | justinu | from an IVR |
| 20:44.19 | redondos | I see. |
| 20:44.21 | generalhan | i use incomingcalllimit=1 in sip.conf and that solved that problem for my reps |
| 20:44.24 | justinu | generalhan: yes |
| 20:44.25 | noky | i want to know how configurate a basical extension that allow to comunicate two user SIP (i have an sip users registered right now in ast*) |
| 20:44.29 | asterboy | point my nose in the direction and I'll be like a hound on a fox hunt. |
| 20:44.29 | redondos | So what is it, justin? |
| 20:44.41 | justinu | redondos next priority after dial |
| 20:44.47 | redondos | Right. |
| 20:44.51 | [av]bani | generalhan: i just want to disable the _beep_, not the ability to take more calls |
| 20:44.51 | redondos | Thanks a bunch. |
| 20:44.53 | justinu | exten => 1,1,Dial(SIP/1,30) |
| 20:44.57 | rikishi | hi, i have a question about asterisk: i have 3 FXO X100P cards with 1 POTS line connected to each of them. i´d like to just answer incoming calls and transfer them to extensions, and also make calls... question: should i create 3 outbound routes or just one with 3 trunks? |
| 20:45.04 | justinu | exten => 2,1,VoiceMail(u1) |
| 20:45.08 | justinu | er exten 1,2 |
| 20:45.12 | [av]bani | uhoh, incomingcalllimit no longer works... |
| 20:45.17 | justinu | lol |
| 20:45.18 | [av]bani | depreciated |
| 20:45.23 | justinu | deprecated |
| 20:45.32 | [av]bani | there goes that idea |
| 20:45.37 | [av]bani | next? |
| 20:45.40 | generalhan | ohh crap .... since when ? im still using 1.2.1 and im about to upgrade |
| 20:45.49 | [av]bani | heh |
| 20:46.10 | [av]bani | i guess youre supposed to use groups now |
| 20:46.41 | [av]bani | doesnt work anymore at least as recent as march 2005 |
| 20:46.51 | asterboy | I setup the rikishi, I'm setup with 3 outbound routes and need to change to one with 3 trunks because I want to be able to automagically pickup the next free extensions. |
| 20:47.09 | redondos | justinu: I see, thanks. So what if the line *did* answer, and the call made by Dial() ended. How would I avoid the user going to the voicemail, in your example? |
| 20:47.19 | asterboy | s/I setup the//g |
| 20:47.43 | asterboy | so I guess it's all in what your trying to accomplish. |
| 20:47.44 | justinu | redondos: normal if dial succeeeds, the channel hangsup after dial |
| 20:47.51 | [av]bani | yep incomingcalllimit no longer exists in 1.2.5 |
| 20:47.51 | justinu | regardless of what the next steps are |
| 20:47.53 | [av]bani | at all |
| 20:47.54 | [av]bani | gone |
| 20:48.00 | b66mer | can I have a phone that has multiple different extensions? |
| 20:48.09 | [av]bani | poof |
| 20:48.09 | redondos | justinu: Oh, so the channel *always* hangs up after Dial()? |
| 20:48.16 | justinu | redondos: there's a way to change that behaviior tho |
| 20:48.17 | redondos | justinu: Can I avoid that and present a menu back again? |
| 20:48.18 | [av]bani | generalhan: guess you better not upgrade :) |
| 20:48.21 | Katty | b66mer: yes. |
| 20:48.21 | justinu | redondos: see option g to dial |
| 20:48.27 | b66mer | which phone? |
| 20:48.28 | rikishi | asterboy: so you mean for my case i just should set 1 outbound route with 3 trunks, and with this i handle incoming calls to the 3 POTS lines, and also make calls with an available line,right? |
| 20:48.33 | Katty | b66mer: some phones, like the polycom 500, support dual logins. |
| 20:48.38 | b66mer | cool |
| 20:48.39 | Katty | b66mer: voip-supply.com has a lot of phones. |
| 20:48.41 | b66mer | thanks |
| 20:48.47 | generalhan | [av] dont tell me that ... i NEED to upgrade ive been having little issues popping up all over lately |
| 20:48.49 | Katty | b66mer: they usually say how many logins a phone supports at a time. |
| 20:48.54 | asterboy | b66mer, get a multiline SIP phone like a Grandstream GXP-2000 or a better quality polycom IP600 and program the lines to map to extensions on *. |
| 20:48.57 | b66mer | ohhh I just bought a bunch of phones for testing |
| 20:48.57 | [av]bani | generalhan: sorry to rain on your parade :)) |
| 20:49.06 | justinu | if you need help, ask me in the channel |
| 20:49.13 | generalhan | so now i need to find another way to get rid of the call waiting beep in my reps ears |
| 20:49.15 | justinu | don't /msg me unless you're paying US Dollars |
| 20:49.18 | *** join/#asterisk darkskiez (n=darkskie@194.164.233.141) |
| 20:49.22 | rikishi | i read that when configuring more than 1 trunk to a route, it uses the first one and if it "fails", then choses the second... does "fail" means also if the line is in use? |
| 20:49.26 | b66mer | I got a Poly 600, 2 x 7960s, Aastra 480i, and a Cisco 7910 |
| 20:49.33 | b66mer | for testing |
| 20:49.42 | b66mer | just waiting for UPS to deliver |
| 20:49.58 | justinu | nice |
| 20:50.06 | justinu | i've used all but the ciscos |
| 20:50.15 | redondos | justinu: Ok. Cool. One last question: what if the line was busy when I used Dial()? |
| 20:50.30 | b66mer | I need to build a phone system for our office... trying to figure out what I am going to deploy |
| 20:50.34 | generalhan | i have 15 7960's .. pain in the ass to set up but a great phone once your firmware is all setup |
| 20:50.34 | justinu | then check ${DIALSTATUS}, i think |
| 20:50.35 | asterboy | rikishi, ya I think that's exactly right. |
| 20:51.08 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 20:51.40 | eKo1 | anybody know how of a function that will give me chan->name without the appended hex digits at the end? |
| 20:52.11 | generalhan | eKo1: you can use the CUT() cmd to get the chan name without the digits at theend |
| 20:52.25 | asterboy | eKo1, sounds like you need to do some string manipulation on ${extension} |
| 20:52.38 | asterboy | cut(), sweet. |
| 20:53.16 | eKo1 | let me check that out |
| 20:53.35 | Corydon-w | And if your channel name contains a hyphen (-), you can do some arithmetic with FIELDQTY() |
| 20:54.07 | generalhan | eKo1: http://generalhan.pastebin.ca/47625 take a look at that ... this is the way i got the channel name with out the hex |
| 20:54.07 | asterboy | another nice one...eKo1 paste what works there, I have to do that also. |
| 20:54.28 | shaun2222 | i remember their being some module that i could use to write code in other languages.. |
| 20:54.32 | shaun2222 | what was it called |
| 20:54.39 | eKo1 | i have chan names that have multiple hyphens in them... |
| 20:54.39 | Corydon-w | AGI |
| 20:54.54 | justinu | bani: maybe you can do something totally lame like check the extension hints before you dial it? |
| 20:55.13 | shaun2222 | do you know if i just have my extentions send to AGI/perl and do all my stuff from with in that? |
| 20:55.21 | Corydon-w | ${CUT(CHANNEL,,1-$[${FIELDQTY(CHANNEL,-)} - 1])} |
| 20:55.24 | generalhan | eKo1: try the way that i have it set up in my pastebin and just let me know if it works ... this is how i do in on my production server and it hasnt failed me yet |
| 20:55.26 | [av]bani | argh, you can't. |
| 20:55.33 | [av]bani | stupid stupid stupid stupid polycom |
| 20:57.31 | eKo1 | hmmm...but that will only work on Sip channels. What about Zap, IAX, etc? |
| 20:57.35 | *** join/#asterisk |omni| (i=cathode@216.64.178.146) |
| 20:58.02 | generalhan | same deal |
| 20:58.21 | generalhan | its just that all my phones are set up as SIP/EXTEN so i have to say SIP |
| 20:58.59 | eKo1 | if i have ax/julio-mete/2, i just want julio-mete |
| 20:59.12 | eKo1 | err, iax/julio-mete/2 |
| 20:59.19 | shaun2222 | i'm trying to setup asterisk so that a incomming call comes in and then asterisk will dial out to my cell phone and connect the call. How can i do this? |
| 20:59.19 | _Paulo_ | ~pb |
| 20:59.21 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:00.05 | Corydon-w | eKo1: are you using asterisk 1.0? |
| 21:00.24 | generalhan | shaun2222: exten => whatever,1,Dial(g1/you_cell_number) ... is how i do it |
| 21:00.37 | eKo1 | not anymore but my knowledge of chan->name is from 1.0 |
| 21:00.57 | Corydon-w | Well, the IAX2 channel name changed in 1.2 to follow the same format as SIP and Zap |
| 21:01.14 | Corydon-w | For precisely this reason |
| 21:01.19 | asterboy | man some people are so stupid on ebay. |
| 21:01.24 | eKo1 | Ahh...I see. |
| 21:01.36 | asterboy | just watch someone buy a TDM card USED for more than its worth NEW. |
| 21:01.48 | noky | asterisk's gurues |
| 21:01.50 | generalhan | Well [av]bani, we need to figure out how to turn off call waiting on certain lines now that i cant use my incomingcalllimit ... damnit |
| 21:02.08 | noky | i need to create some dialplan for my ast* that allow comunicate two users sips |
| 21:02.12 | noky | how can i do that?!??!?!? |
| 21:02.36 | [TK]D-Fender | generalhan : Turning off call waiting is easy enough.... |
| 21:02.59 | eKo1 | where is the documentation for this? |
| 21:03.02 | redondos | Why would asterisk report a Dial() as ANSWERED when it wasn't actually answered, via an FXS card? |
| 21:03.06 | shaun2222 | generalhan: whats the g1/ mean? should that be changed to something else? |
| 21:03.08 | generalhan | [TK]D-fender: there is no way to turn off call waiting on these Aastra phones that my reps have, ive already talked with them about that |
| 21:03.14 | generalhan | thats why i had to use the incomingcalllimits |
| 21:03.25 | [TK]D-Fender | generalhan : Just set the registration to "callsperlinekey=1", "linekeys=1" |
| 21:03.32 | generalhan | g1 is Group1 as in how my ZAP lines are set up |
| 21:04.15 | shaun2222 | generalhan: i want it to dial out of the voicepulse i have setup (context outgoing-call] |
| 21:04.28 | generalhan | [TK]D-fender: please tell me you can elaborate on that for me ... or shoot me a link, cause im willing to do anything to make the reps stop complaining to me about a beep in their ear |
| 21:04.42 | generalhan | gross you use VoicePulse ?? |
| 21:04.48 | generalhan | connect or whoilesale ? |
| 21:04.55 | shaun2222 | connect |
| 21:05.13 | shaun2222 | generalhan: not in production, they where just simple and cheap for testing purposes. |
| 21:05.43 | shaun2222 | not sure if i'll keep them, so far it's worked ok.. |
| 21:05.46 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 21:05.48 | generalhan | then im not real sure. pastebin your outgoing-call context and lemme see |
| 21:06.11 | generalhan | cause with wholesale we had a customer number we had to use on our outgoing calls so im not as familiar with the connect |
| 21:07.41 | [av]bani | [TK]D-Fender: that prevents you from being able to put calls on hold and originate new calls... |
| 21:07.50 | shaun2222 | generalhan: i figured it out. |
| 21:07.52 | [av]bani | [TK]D-Fender: that is to say, that's a totally sucky way of "disabling" call waiting on polycom |
| 21:07.58 | shaun2222 | generalhan: whats your nick from? |
| 21:08.04 | shaun2222 | i've heard it before. |
| 21:08.04 | generalhan | lol |
| 21:08.10 | shido6 | generalhan |
| 21:08.11 | shaun2222 | stargate or somthing? |
| 21:08.18 | generalhan | ive had this same nick on everything i use since i was like 7 LOL |
| 21:08.28 | [TK]D-Fender | [av]bani : You can initiate a conference and never complete it as a way of calling out... |
| 21:08.32 | [av]bani | i'm looking at a better way, because callsperlinekey is an unacceptable way of doing it |
| 21:08.36 | generalhan | heya shido |
| 21:08.39 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 21:08.59 | [av]bani | and :P @ polycom for being 'tards |
| 21:09.02 | [TK]D-Fender | [av]bani / generalhan : another option is to put a "ChanIsAvail" check in your dial-plan to see if they are already on a call. |
| 21:09.06 | Zodiacal | anyone know if i am reading this correctly? install instructions 1a) says that the speed dial buttons on the 7960 won't work if i install a 7914 addon? |
| 21:09.10 | [av]bani | [TK]D-Fender: chanisavail doesnt do that. |
| 21:09.18 | [av]bani | read up on the chanisavail docs. :P |
| 21:09.25 | [TK]D-Fender | [av]bani : Yes it does. |
| 21:09.39 | [TK]D-Fender | [av]bani : I've done it personally. |
| 21:09.39 | [av]bani | all chanisavail checks is if a call can go through. it knows nothing about # of calls |
| 21:09.47 | [av]bani | i use chanisaval too. it doesn't do that. |
| 21:09.51 | Zodiacal | oh heres the page: http://www.voip-info.org/tiki-index.php?page=Cisco+7914-HOWTO |
| 21:09.52 | Zodiacal | duh |
| 21:09.52 | [TK]D-Fender | [av]bani YOU read up on it :) |
| 21:10.34 | [av]bani | it checks if routing is possible. it doesnt check if channel is "in use" because that's not what it does. read app_chanisavail.c to see why |
| 21:11.09 | [av]bani | eg, if you're using it to "check channel is busy", it wont work the way you think it does |
| 21:11.21 | [TK]D-Fender | [av]bani : http://pastebin.ca/47627 <- please note option "s" |
| 21:11.39 | [TK]D-Fender | [av]bani : If they are on a call AT ALL you can say "just don't call" |
| 21:12.15 | [av]bani | [TK]D-Fender: all asterisk can know is if a call is being routed _through asterisk_ |
| 21:12.18 | [TK]D-Fender | [av]bani : thats what I did in an early implementation of AgentCallbackLogin style queue's here |
| 21:12.32 | [av]bani | [TK]D-Fender: if a phone is talking to another device not through asterisk (think reinvite), asterisk can't possibly know. |
| 21:12.33 | [TK]D-Fender | [av]bani : this will say if they are on a call period. |
| 21:12.42 | [TK]D-Fender | [av]bani : Reinvite= EVIL! |
| 21:12.48 | [TK]D-Fender | [av]bani : NEVER USE IT |
| 21:12.58 | [av]bani | [TK]D-Fender: route 1400 sessions through asterisk. gotcha. |
| 21:13.37 | [av]bani | [TK]D-Fender: also... you are saying, only use one asterisk pbx ever. (think failover) |
| 21:13.37 | [TK]D-Fender | [av]bani : You have that many do you? And with Gigabit, who cares? |
| 21:14.09 | [TK]D-Fender | [av]bani : Sure, keep extending to "worst case scenario"..... |
| 21:14.15 | [av]bani | thats hardly worst case |
| 21:14.24 | [av]bani | but anyway i found out a way to do it |
| 21:14.32 | generalhan | lies |
| 21:14.32 | [av]bani | (disable call waiting) |
| 21:14.34 | generalhan | lol |
| 21:14.37 | generalhan | haha |
| 21:14.43 | [av]bani | obviously [TK]D-Fender doesn't need it :) |
| 21:14.46 | [TK]D-Fender | [av]bani : Sure it is... Asterisk will already have blown up after that much traffic, so how much worse could it get? ;) |
| 21:15.58 | shaun2222 | when using queues can i have a cell phone number as part of the queue of reps? |
| 21:16.09 | [av]bani | why not? |
| 21:16.25 | shaun2222 | dunno, it seamed as if queues where only for phones that had registered them selves |
| 21:16.37 | stoffell | [av]bani, just wondering, what other 'main' components beside asterisk and dundi you use to get a redundant setup? |
| 21:16.57 | [av]bani | stoffell: multiple registrations on your phones |
| 21:16.58 | nubbrage | shaun2222: end point can be anything you want -- member => SIP/my_provider/cell_number |
| 21:17.03 | nubbrage | doesn't need to be registered |
| 21:17.10 | shaun2222 | ok |
| 21:17.15 | justinu | well... it appears my meetme problems have something to do w/ the async RTP stuff |
| 21:17.18 | shaun2222 | time to dig into queues i guess |
| 21:17.24 | justinu | because it's working fine without that patch |
| 21:17.37 | stoffell | [av]bani, so like a "backup" SIP server, or seconday sip server... anything else ? (this is usefull !) |
| 21:18.43 | [av]bani | stoffell: thats about it |
| 21:19.15 | [av]bani | one of these days soemone will make a load balancing switch that can handle SIP |
| 21:19.24 | stoffell | [av]bani, thanks, because multiple servers to me also seems like 'the good way' to go.. |
| 21:19.33 | stoffell | i now have some material to get our lab started :) |
| 21:20.03 | [av]bani | stoffell: if you want "more 9's", then dual ether / dual switches also |
| 21:20.14 | [av]bani | thats overkill though :) |
| 21:20.50 | Mauro__ | how do I use GET DATA? |
| 21:20.51 | stoffell | [av]bani; it is :) i just want a lot of 9's on the servers :) the rest is easily (and quickly) replaceable |
| 21:21.04 | Mauro__ | something like exten => *1235,1,get data /var/lib/asterisk/sounds/custom/rut.wav 2000 9 |
| 21:21.54 | eKo1 | Corydon-w: you said "the IAX2 channel name changed in 1.2 to follow the same format as SIP and Zap" but they're all different as far as I can see. |
| 21:21.58 | [av]bani | stoffell: for high reliability i just built solid-state asterisk servers |
| 21:22.06 | [av]bani | stoffell: running from flash |
| 21:22.20 | eKo1 | how big a flash? |
| 21:22.28 | file | justinu, one of the reasons that patch isn't in yet |
| 21:22.39 | [av]bani | eKo1: 1gb... about $50 |
| 21:22.43 | stoffell | [av]bani; yes, also have that in mind.. 4 'big sites' with a "real" server, and several (approx. 10) with a solid-sate solution, and then get dundi rocking, oh boy.. |
| 21:23.09 | Qwell[] | [av]bani: speeddial support :) |
| 21:23.16 | stoffell | [av]bani, is dundi in a lab-environment a good representation of "real-life" in your opinion? (a well-constructed lab that is) |
| 21:23.25 | [av]bani | eKo1: enough for a full asterisk + sources, compiler, kernel + sources, apache, tftp, firmware + configs for phones, php scripts, and several 100's of mb's left over |
| 21:23.37 | [av]bani | Qwell[]: dynamic? |
| 21:23.51 | Qwell[] | no hints yet |
| 21:24.04 | [av]bani | 7970? |
| 21:24.09 | Qwell[] | maybe :p |
| 21:24.17 | *** join/#asterisk _deg_ (n=deg@200.250.222.8) |
| 21:24.27 | [av]bani | "Line In Use" is so boring, i want "Bridged call to SIP/FXO2" |
| 21:24.30 | [av]bani | :) |
| 21:31.02 | *** join/#asterisk chr|s_ (n=chris@217.171.53.147) |
| 21:31.10 | xbit` | can asterisk be used without root privilege? |
| 21:31.11 | [av]bani | yay disabled polycom callwaiting tone |
| 21:31.19 | chr|s_ | ok people, I am ready to make my first asterisk call! hurrah, well, in 10 minutes I think |
| 21:31.20 | [av]bani | a big phlbbbbt @ [TK]D-Fender |
| 21:31.24 | generalhan | please tell me how ... |
| 21:31.28 | [av]bani | :)) |
| 21:31.47 | generalhan | i dont have polycoms though so hopefully the same trick will work with my Aastras |
| 21:31.55 | [av]bani | not likely |
| 21:32.07 | chr|s_ | polycom handsets? |
| 21:32.38 | chr|s_ | I just got two of those, haven't tried them out yet |
| 21:32.45 | [TK]D-Fender | [av]bani : Do share.... |
| 21:34.01 | [av]bani | [TK]D-Fender: i thought you're happy with ChanIsAvail() :)) |
| 21:34.16 | [av]bani | <- meanie |
| 21:35.56 | [TK]D-Fender | [av]bani : Hey, for the amount I help everyone else in here INCLUDING you..... |
| 21:36.01 | [av]bani | lies |
| 21:37.23 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
| 21:37.24 | file | now now, be nice all |
| 21:37.27 | stoffell | lol |
| 21:37.44 | [av]bani | <CALLWAITING se.pat.callProg.6.name="call waiting" se.pat.callProg.6.inst.1.type="silence" se.pat.callProg.6.inst.1.value="0"/> |
| 21:37.47 | [av]bani | <CALLWAITINGLONG se.pat.callProg.7.name="long call waiting" se.pat.callProg.7.inst.1.type="silence" se.pat.callProg.7.inst.1.value="0"/> |
| 21:37.50 | [av]bani | i'll bill you later |
| 21:38.03 | shaun2222 | in the queues.conf for member => if i wanted to use a context that was in iax.conf i would do member => iax/context ? |
| 21:38.11 | generalhan | oh god .. i have nothing even close to what that looks like for the Aastra phones |
| 21:38.12 | generalhan | boo |
| 21:38.17 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 21:38.23 | [av]bani | i hear aastra actually has a support department though |
| 21:38.29 | generalhan | they do |
| 21:38.31 | Hmmhesays | i don't want anybody else, when i think about you I touch myself |
| 21:38.37 | generalhan | and they told me, ther eis no way to turn it off |
| 21:38.44 | [av]bani | boo? |
| 21:38.45 | *** join/#asterisk lorinc (n=ang@caracas-2906.adsl.interware.hu) |
| 21:38.54 | [av]bani | yell at them to add a config switch |
| 21:39.00 | Hmmhesays | turn what off? |
| 21:40.13 | *** join/#asterisk |omni| (i=rob@216.64.178.146) |
| 21:41.10 | [av]bani | even grandstream lets you turn off call waiting |
| 21:41.49 | Hmmhesays | do it in asterisk |
| 21:41.54 | Hmmhesays | if you must |
| 21:42.22 | generalhan | im trying to figure out how to do that |
| 21:42.26 | [av]bani | or just dont buy aastra :) |
| 21:42.37 | [av]bani | till they remove head from anus |
| 21:43.04 | generalhan | i cant help that now... my boss had me buy 40 of them and they are already being used |
| 21:43.06 | generalhan | lol |
| 21:43.16 | [av]bani | generalhan: can you replace indications? |
| 21:43.28 | generalhan | what do you mean ? |
| 21:43.34 | *** part/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 21:43.42 | [av]bani | does aastra let you replace indications with custom wav files |
| 21:44.01 | generalhan | nope |
| 21:44.06 | shaun2222 | in the queue.conf if i wanted to just have it dial a extention would i do member => Iax/user.name ? |
| 21:45.04 | [av]bani | generalhan: seems to me you could use the 'call forward if busy' on the aastra |
| 21:45.23 | [av]bani | forward it direct to voicemail |
| 21:45.26 | generalhan | hmm |
| 21:45.38 | generalhan | you reading that from a link somewhere ? |
| 21:45.46 | [av]bani | 480i sip user guide |
| 21:45.53 | generalhan | hmm |
| 21:45.56 | [av]bani | page 23 |
| 21:45.59 | generalhan | im using 9112i's |
| 21:46.24 | generalhan | like im gonna give 40 lowly reps 480is !!!!! |
| 21:47.26 | Hmmhesays | so this pirate walks into a bar with a steer wheel sticking out of his pants. The bartender says "Sir you have a steering wheel sticking out of your parts" the pirate responds "Argh! I know its driving me nuts!" |
| 21:47.42 | generalhan | ..... |
| 21:47.43 | generalhan | haha |
| 21:47.56 | file | ha ha ha |
| 21:48.58 | *** join/#asterisk ian_k[bootcamp] (n=ian@gateway.digium.com) |
| 21:49.22 | *** part/#asterisk ian_k[bootcamp] (n=ian@gateway.digium.com) |
| 21:50.16 | *** join/#asterisk veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 21:50.39 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 21:52.40 | websae | anyone looking for termination/origination on the LEVEL 3 network? |
| 21:52.42 | websae | :) |
| 21:53.09 | shaun2222 | websae: nice intro... |
| 21:53.17 | websae | and good afternoon all |
| 21:53.27 | websae | shaun: thank you if i do say so myself |
| 21:54.09 | websae | how are you doing? |
| 21:54.09 | shaun2222 | websae: little advice.. get ride of the hotmail address on your domain. |
| 21:54.09 | websae | don't have a hotmail address |
| 21:54.13 | websae | but thanks for the advice lol |
| 21:54.17 | shaun2222 | <PROTECTED> |
| 21:54.25 | shaun2222 | websae: talking about the whois info. |
| 21:54.36 | websae | ohh -- yeah ixwebhosting did that |
| 21:54.41 | websae | don't know what that is about |
| 21:54.52 | shaun2222 | also, change the tech info so yahoo is no longer listed |
| 21:55.17 | shaun2222 | somebody recommended your site to me the other day... that info immediatly made me look away.. |
| 21:55.22 | websae | yeah - great call, because that's hindering my business performance? |
| 21:55.26 | justinu | file: you've had the same troubles, then? |
| 21:55.32 | justinu | w/ meetme + async RTP |
| 21:55.38 | shaun2222 | that and the fact the domain was registered 6 months ago or so... |
| 21:55.46 | file | ...I haven't touched that |
| 21:55.48 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-20-208.slkc.qwest.net) |
| 21:55.57 | Hmmhesays | is rollergrrl a chick? |
| 21:56.46 | rollergrrl | Calling someone a 'chick' sure is flattering. |
| 21:56.49 | justinu | file: but you mentioned that being one reason the async RTP patch hasn't been commited? |
| 21:57.01 | file | justinu, I have read over the bug... and heard the talk about it |
| 21:57.02 | Hmmhesays | <PROTECTED> |
| 21:57.28 | justinu | file: ok... i'm getting conflicting reports... some people say patch 5374 helped their problems with meetme latency |
| 21:57.40 | justinu | file: but my experience is not that |
| 21:58.58 | Juggie | justinu, one could assume meetme is the problem then :) |
| 21:59.15 | justinu | juggie: actually, meetme works ok without that patch applied, it seems |
| 21:59.24 | Katty | Hmmhesays: does that make you flakey? |
| 21:59.31 | justinu | perhaps repatching it in a different sequence will help :dunno: |
| 21:59.40 | Juggie | justinu, that patch is a required change to * |
| 21:59.43 | Katty | Hmmhesays: crumby, maybe? |
| 21:59.52 | Juggie | sending out RTP timed on the incomming stream is bad |
| 22:00.12 | justinu | yeah |
| 22:00.19 | justinu | agreed... and I need to to play early media to my customers |
| 22:00.27 | rollergrrl | makes him good to eat with soup |
| 22:00.39 | justinu | L3 doesn't provide a tx RTP stream before the call has been 200 OK'ed |
| 22:00.48 | justinu | so early media is totoally async |
| 22:01.22 | Juggie | indeed |
| 22:01.25 | Juggie | its just bad practice |
| 22:01.33 | justinu | indeed ;) |
| 22:01.33 | Juggie | i would claim meetme has many inherant flaws :) |
| 22:01.45 | rikishi | what does this line means? [from-sip-external] |
| 22:01.45 | rikishi | include => from-pstn |
| 22:01.49 | justinu | unfortunately i'm only forced to use meetme because I can't bridge the 2 calls any other way |
| 22:01.51 | rikishi | on extensions.conf |
| 22:02.02 | websae | justinu: how are you doing? |
| 22:02.15 | Juggie | justinu, oh no, this again ;) |
| 22:02.18 | *** join/#asterisk heka (n=heka@82.114.68.126) |
| 22:02.19 | justinu | yep |
| 22:02.20 | Juggie | whats wrong with Dial() |
| 22:02.31 | *** join/#asterisk zx225 (n=joel@65.183.42.3) |
| 22:02.35 | justinu | dial bridges the calls together on the basis of who answers first |
| 22:02.57 | justinu | that isn't the criteria i'm using to determine which outbound leg gets bridged |
| 22:03.01 | heka | Hello, Im using slav`s jitter buffer patch for sip, I have enable it in sip.conf. How can I check if jb is enabled? |
| 22:03.13 | Juggie | justinu, did you run a macro on the dial? |
| 22:03.27 | rollergrrl | Can't you gut out the stuff you need from app_meetme? |
| 22:03.28 | justinu | yep... the macro only executes on the first channel that answers |
| 22:03.36 | justinu | all other outbound dials are released at that time |
| 22:03.42 | Juggie | ahhh ok. |
| 22:03.45 | Juggie | thats what i was afraid of |
| 22:03.53 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 22:03.57 | Juggie | well there's also an app_bridge on mantis |
| 22:03.58 | justinu | yep... so anyways... i've achieved the call flow I needed via AMI |
| 22:04.07 | justinu | oh really? |
| 22:04.08 | rollergrrl | there's that app_icd too |
| 22:04.15 | nubbrage | rollergrrl: wierd -- you haven't been around in a while |
| 22:04.26 | Juggie | alternatively you could patch app_dial so that it didnt stop trying to dial until someone completed the macro with an active call. |
| 22:04.29 | rollergrrl | nubbrage: true |
| 22:04.32 | Juggie | though that would be a little bit of work. |
| 22:04.36 | justinu | juggie: that's what i've been thinking about |
| 22:04.41 | justinu | but yeah - it appears like a fair bit of work |
| 22:04.43 | file | nubbrage, is jsmith about? |
| 22:04.44 | nubbrage | rollergrrl: welcome back :) |
| 22:04.47 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 22:04.48 | nubbrage | file: he IS! |
| 22:04.48 | Juggie | basically make it run the macro for whoever answers |
| 22:04.58 | file | nubbrage, tell him I took care of the MOH bug he posted in 1.2 and trunk |
| 22:04.58 | nubbrage | file: let me *page* him for you |
| 22:04.59 | Juggie | and only bridge once the macro completes to who ever completes the macro first |
| 22:05.04 | justinu | juggie: that might involve threads and all sorts of things I'm not qualified to be mucking with |
| 22:05.16 | Juggie | justinu, possible. |
| 22:05.25 | justinu | i'm still hazy as to the inner workings of ast |
| 22:05.32 | nubbrage | file: message passed on! |
| 22:05.39 | nubbrage | file: he says *thanks!* |
| 22:05.44 | file | aye |
| 22:05.49 | file | it just ignores all past the limit :P |
| 22:06.10 | *** join/#asterisk Ludanek (n=nechci@rb1j38.chello.upc.cz) |
| 22:06.31 | rollergrrl | nubbrage: surprised anyone remembers me |
| 22:06.56 | justinu | app_bridge looks like exactly what I need |
| 22:07.08 | nubbrage | rollergrrl: I remember everyone :D |
| 22:07.23 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:07.26 | Corydon-w | Why not? Rollergirl is a famous DJ |
| 22:07.38 | asterboy | Thanks Fender for helping get call forwarding/conferencing working. |
| 22:08.05 | Ludanek | hi, is possibble to run billion isdn S0 card on a@h 2.7 ???? |
| 22:08.12 | justinu | rollergirl (the dj) is very hot |
| 22:08.50 | asterboy | After both partie's hangup however, the channels remain open...any ideas on how to fix that? |
| 22:09.28 | justinu | sip? |
| 22:09.34 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 22:09.42 | asterboy | yes |
| 22:09.43 | Ludanek | rtp timeout ? |
| 22:09.51 | justinu | sounds like the bye is not being received |
| 22:10.34 | rollergrrl | good music too |
| 22:10.37 | asterboy | Oh I get this message after the transfer, but the call still goes through and seems to work fine: -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.18 |
| 22:10.44 | justinu | i like a few of her songs |
| 22:11.11 | asterboy | that IP is the SIP Polycom IP600 phone. |
| 22:11.23 | asterboy | That initiated the transfer |
| 22:12.24 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 22:17.00 | asterboy | rtp timeout did not end the ZAP channels used in the transfer....hmmmm.. |
| 22:17.22 | justinu | it's a zap channel now? |
| 22:17.27 | asterboy | lol |
| 22:17.47 | asterboy | yes, I use Polycom phones and the outgoing lines are ZAP |
| 22:17.54 | justinu | describe the call flow then |
| 22:18.32 | *** join/#asterisk veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 22:19.28 | asterboy | outside caller ----> ZAP Channel 1 ----> Polycom IP600 Line 1 --- TRANSFER ---> ZAP Channel 2 -----> Polycom IP600 Line 2 ----HANGUP---- ZAP Channels 1 & 2 remain open |
| 22:19.58 | asterboy | Using a regular phone to listen in on pots line...getting really loud fast beeps. |
| 22:20.10 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-154.apple.com) |
| 22:20.49 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-154.apple.com) |
| 22:22.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:24.13 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:24.56 | asterboy | justinu, what other variables exit for timeouts on ZAP channels? |
| 22:25.16 | asterboy | rtp timeout is great for SIP |
| 22:27.03 | justinu | that call flow confuses me |
| 22:27.21 | justinu | how does it go from zap/2 to the polycom? |
| 22:27.55 | asterboy | zap/2 is programmed to dial another polycom extension. |
| 22:28.21 | justinu | yeah, but why are you transfering the call from zap/1 to zap/2? |
| 22:28.37 | asterboy | just for testing purposes. |
| 22:29.09 | justinu | it probably has something to do with the lack of disconnect supervision on the zap channels. |
| 22:29.10 | justinu | not sure |
| 22:29.25 | asterboy | ya that sounds like a good hint. |
| 22:29.27 | justinu | you can't really do stuff like that on FXO lines, because the signalling isn't designed for it. |
| 22:29.40 | justinu | it would work fine if you had a PRI or E&M T1 |
| 22:30.09 | asterboy | If I transfer outside of the * box, should it be ok? |
| 22:30.27 | asterboy | actually, it won't matter. |
| 22:30.44 | justinu | well... try it... see if the disco passes thru |
| 22:30.47 | justinu | i bet it doesn't |
| 22:30.48 | asterboy | cause I still need to pickup another ZAP channel to make the transfer since I'm doing the forwarding. |
| 22:31.02 | justinu | you're trying to get too cute with FXO... you need more |
| 22:31.38 | asterboy | more what? |
| 22:31.46 | justinu | more signalling, scotty |
| 22:31.50 | asterboy | lol |
| 22:32.02 | asterboy | give me all you got!, kirk out. |
| 22:32.22 | asterboy | need S7 |
| 22:32.26 | asterboy | need SS7 |
| 22:32.29 | justinu | heh, not quite |
| 22:32.33 | asterboy | lol |
| 22:32.36 | asterboy | for 3 lines. |
| 22:32.51 | justinu | you need digital trunks |
| 22:33.02 | justinu | or 4 wire loops, which don't exist anymore afaik |
| 22:33.15 | asterboy | well my telco lines are actually VOIP ATAs |
| 22:33.29 | justinu | heh |
| 22:33.30 | asterboy | but convert to pots like lines before * |
| 22:33.35 | justinu | lol |
| 22:33.45 | justinu | no wonder it doesn't work |
| 22:34.04 | justinu | those ATAs probablyu don't provide the break in loop current that kewlstart needs to detect disco |
| 22:34.08 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 22:34.10 | SpaceBass | hey folks |
| 22:34.19 | SpaceBass | anyone using A@H 2.7 |
| 22:34.21 | SpaceBass | ? |
| 22:34.23 | asterboy | yes, that is most likely it. |
| 22:34.27 | *** join/#asterisk vazir (i=anton@82.198.21.17) |
| 22:34.31 | asterboy | and the fact that SIP is involved. |
| 22:35.00 | asterboy | I'm just using this in LAB. |
| 22:35.12 | asterboy | In real world, the lines will be true POTS |
| 22:35.28 | justinu | you're not working in a very good lab, my friend |
| 22:35.35 | vazir | Hi! anyone knows if Asterisk can be managing just signalling traffic and not tap-in into RTP? |
| 22:35.39 | asterboy | yes, money is the issue there. |
| 22:36.09 | asterboy | that's twice today someone has asked a question closely related to the issue I just had |
| 22:36.26 | justinu | that's why this place exists |
| 22:36.39 | asterboy | I'm in the zone. |
| 22:37.01 | vazir | Hi! anyone knows if Asterisk can be managing just signalling traffic and not tap-in into RTP? |
| 22:37.52 | Denmark | wow |
| 22:39.11 | Denmark | [TK]D-Fender : Looks like in Denmark, the ip-501 is close to double price of a 941! |
| 22:39.34 | *** join/#asterisk _DAW (n=bob@adsl-150-58-20.msy.bellsouth.net) |
| 22:43.14 | asterboy | seems to work excellent for conferencing. |
| 22:44.30 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-154.apple.com) |
| 22:46.02 | asterboy | What is going on is that the zap channels are talking to each other and ignoring signalling from the calling party. |
| 22:46.10 | tdonahue | does anyone know of RFC2833 problems with an asterisk box talking to a vega 400 |
| 22:46.12 | *** join/#asterisk Fieldy (i=trash@gentoo/contributor/Fieldy) |
| 22:46.12 | tdonahue | ? |
| 22:46.29 | asterboy | I need a way to kill those zap channels once the call discos |
| 22:47.18 | asterboy | This must be possible otherwise noone could use call forwarding from *. |
| 22:47.31 | Nugget | \o/ ^o^ /c_ /o\ |
| 22:47.35 | Nugget | It's fun to stay at the |
| 22:47.36 | Nugget | \o/ ^o^ /c_ /o\ |
| 22:47.50 | asterboy | :P |
| 22:47.54 | MooingLemur | hehe |
| 22:49.24 | asterboy | /o/ ^o/ /o^ <o^ <o> |
| 22:49.43 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 22:49.49 | asterboy | thats kinda fun |
| 22:49.49 | tdonahue | no one has any bad experiences with the vega stream products? |
| 22:50.48 | *** part/#asterisk QbY (i=user@cm-12-197-169-221.dhcp.geo-sc.southerncoastalcable.net) |
| 22:54.34 | Denmark | [TK]D-Fender (took me a while to find a danish dealer) |
| 22:57.26 | *** part/#asterisk vazir (i=anton@82.198.21.17) |
| 22:57.28 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:58.28 | Denmark | [TK]D-Fender : Hmm ... seems its the dealer thats expensive .. he wants more for IP 301, than others want for sippura 941 |
| 22:59.21 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-154.apple.com) |
| 22:59.55 | asterboy | Is there a way to monitor the ZAP channels for an "out of service" (fast beep) and automagically disconnect? |
| 23:00.19 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-154.apple.com) |
| 23:00.29 | qseek | hi is there a way to kill all outgoing calls without taking down asterisk |
| 23:00.52 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
| 23:00.57 | justinu | soft hangup |
| 23:00.59 | asterboy | that is the third time. |
| 23:01.26 | qseek | that would be for one call, but say you have an auto dialer and u want to kill it how would u go about doing it |
| 23:01.48 | justinu | dunno |
| 23:02.22 | qseek | k |
| 23:02.40 | asterboy | Zap/2-1 s@6692862:1 Up Bridged Call(Zap/1-1) |
| 23:02.44 | asterboy | Zap/1-1 611@Home:2 Up Dial(Zap/g1/611|90|twTW) |
| 23:03.36 | Denmark | later tkd? |
| 23:03.43 | justinu | kbi |
| 23:04.02 | Denmark | kbi? |
| 23:04.15 | justinu | mew |
| 23:04.25 | Denmark | mew? |
| 23:04.36 | justinu | ~mew |
| 23:04.37 | jbot | A MIME mail reader for Emacs/XEmacs. URL: http://www.mew.org/ |
| 23:04.41 | Denmark | justnulling2: are you trying to confuse me? |
| 23:04.41 | justinu | heh |
| 23:04.45 | justinu | perhaps |
| 23:04.47 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 23:04.50 | Denmark | =) |
| 23:04.57 | asterboy | the soft hangup works to hangup both lines when I issue it for just one channel or the other. |
| 23:04.58 | Denmark | ~kbi |
| 23:05.25 | justinu | because they're bridged in asterisk |
| 23:05.25 | asterboy | so most like justinu is right that it will cleanup once connected to true pots lines. |
| 23:05.42 | justinu | it's because kewlstart isn't working |
| 23:05.44 | asterboy | exactly |
| 23:05.47 | justinu | probably because of your ATA |
| 23:05.58 | asterboy | screws up the signaling |
| 23:06.08 | justinu | http://ourproject.org/docman/view.php/116/144/faq.html#TDM%20&%20Analog_1 |
| 23:06.19 | asterboy | That is a good thing to know. |
| 23:06.28 | asterboy | Wonder if anyone else has had this issue. |
| 23:06.34 | justinu | of course |
| 23:06.47 | asterboy | did they find a solution? |
| 23:06.55 | justinu | yeah, hook it up to a real switch that supports it |
| 23:07.19 | asterboy | that would be nice...but the idea is to get out of the telco realm |
| 23:07.50 | justinu | so why are you playing with analog then? |
| 23:08.05 | asterboy | some clients insist on it. |
| 23:08.19 | asterboy | for me thought, I'd like to have VOIP in house. |
| 23:08.28 | asterboy | s/thought/though/g |
| 23:09.01 | Denmark | dialtone |
| 23:09.04 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 23:09.53 | terrapen | ring ring ring ring ring ring ring |
| 23:09.55 | terrapen | BANANAPHONE |
| 23:10.14 | asterboy | BANANAPHONE who? |
| 23:10.53 | terrapen | I'm definitiely having a BANANAPHONE extension in our new corporate phone system |
| 23:11.00 | Denmark | BANANAPHONE dialtone |
| 23:11.16 | terrapen | dial 226262 |
| 23:11.44 | Denmark | "the number you have dialed, have not been inserted!" |
| 23:11.49 | terrapen | heh |
| 23:11.52 | asterboy | lol |
| 23:13.29 | Denmark | s/have not been/is not/ |
| 23:14.09 | Denmark | "Please check the number and try again!" *fast busy* |
| 23:15.10 | Denmark | What does that mean? I get it if I call a new number I bought from my VoIP provider (who is supposed to run asterix). I am guessing they didn't assign the number to me (yet)? |
| 23:15.44 | justinu | either that or you're returning a 404 when they try and forward the call to you |
| 23:16.25 | Denmark | oh.. I didn't assign the number to a phone .. I assigned it to "dialtone" |
| 23:17.08 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 23:18.10 | *** join/#asterisk Harm|w (i=harm@67.58.10.48) |
| 23:19.02 | Denmark | I guess I will try to assign it to a phone, and then assign one of my other numbers to dialtone, and see if its the same answer I get. :-) |
| 23:19.26 | Denmark | (I do plan to run asterisk myself, but I am not ready) |
| 23:19.47 | asterboy | Found this nice tid bit for the problem of disconnect supervision: http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision |
| 23:20.03 | justinu | oh... then they haven't inserted your number yet :P |
| 23:20.30 | Harm|w | anyone know if its possible to upgrade a Cisco 7970G straight from version 2.5 to 8.0sip? or it has to be incremental like with 7940/60s? |
| 23:20.30 | Denmark | justnulling2: Hehe :) |
| 23:20.34 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 23:20.50 | *** join/#asterisk Liquid_Ic (n=Liquid_I@ool-4573cc11.dyn.optonline.net) |
| 23:20.51 | Denmark | s/nulling2/inu/ |
| 23:21.59 | *** join/#asterisk blehbo (n=el_terri@do.you.like.my.frippers.com) |
| 23:22.15 | Liquid_Ic | Been spending today setting up Asterisk@Home. Calls to other extentions work good, but any outgoing calls give me "all circuits busy |
| 23:22.22 | Denmark | I don't understand "dialtone" quite, and jbot didn't know either. |
| 23:22.46 | Liquid_Ic | i setup an account with nuphone and followed their directions to setup the trunk and outgoing routing, but still no luck |
| 23:22.52 | justinu | bah: http://pastebin.ca/47639 |
| 23:23.01 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-217.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:24.50 | X-Rob | Liquid_Ic, see the topic? |
| 23:25.13 | X-Rob | (*hint*) |
| 23:28.16 | Harm|w | anyone know if its possible to upgrade a Cisco 7970G straight from version 2.5 to 8.0sip? or does it have to be incremental like with 7940/60s? |
| 23:28.52 | Denmark | Harm|w : No idea, but there was a guy just before asking exact the same question. |
| 23:28.56 | Harm|w | probably him |
| 23:29.12 | blehbo | youre trippin Harm|w |
| 23:29.17 | Harm|w | lies. |
| 23:32.18 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 23:32.22 | Zodiacal | sorry got disconnected.. |
| 23:32.22 | Zodiacal | anyone know of a way to get sccp phones to dial out immedialy insted of waiting for the dialplans timeout? |
| 23:34.07 | Zodiacal | the sip firmware allowed me to press the dial softkey |
| 23:34.54 | tgrman | Zodiacal: if the SCCP based Cisco phones are like the SIP based ones, you can setup a dialplan.xml file on your TFTP server that specifies dial timeouts/patterns, etc |
| 23:35.17 | Harm|w | try hitting the pound (#) key when you're done dialing |
| 23:35.40 | Zodiacal | harm|w nope.. |
| 23:36.35 | blehbo | while we're on questions, is there any way to set the DNIS that asterisk sends out a PRI T1 to a pbx such as an Intertel Axxess, its sending the string "@2" where the interntel expects the DNIS |
| 23:37.20 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-154.apple.com) |
| 23:38.11 | Zodiacal | tgrman i guess thats the only way to do it.. i would have to create a complex dialplan that can detect local numbers and excludes internal extentions.. |
| 23:38.17 | Zodiacal | hoping there was an easier way :P |
| 23:38.44 | Zodiacal | err not excludes internal extions but recognizes them |
| 23:39.47 | tgrman | blehbo: does using SetCallerID before you do a Dial not do what you want? |
| 23:40.38 | blehbo | well the intertel needs to know what number is dialed, not what number is dialed from |
| 23:41.01 | blehbo | well it needs to know both, but its not being told what incoming number is dialed |
| 23:41.19 | tgrman | Zodiacal: no other way of which I am aware. Cisco phones are nice and all but Polycom phones are _MUCH_ better, IMHO |
| 23:41.57 | Zodiacal | tgrman sccp driver makes cisco phones a lot better, except for this one anoying thing.. sip didn't do this.. |
| 23:42.26 | [av]bani | sccp doesnt do nat at all... |
| 23:42.35 | [av]bani | nor does it do anything resembling security :) |
| 23:43.56 | tgrman | cisco phones were really designed to be used within an enterprise, where you theoretically wouldn't have NATs etc and have total control over the network, using LOTS of cisco gear, of course |
| 23:45.47 | [av]bani | using ONLY cisco gear, period |
| 23:46.30 | Zodiacal | eh? |
| 23:46.42 | tgrman | for the price, you get a lot more functionality with the polycom phones |
| 23:46.45 | Zodiacal | avbani whats sip do for security? |
| 23:47.00 | [av]bani | Zodiacal: it at least has passwords, and sips |
| 23:48.07 | Zodiacal | u mean a sccp phone can't connect to a phone that is nat'ed? |
| 23:48.29 | Zodiacal | or server rather |
| 23:48.35 | [av]bani | sccp has no explicit nat support, you need to shim it up with hacks |
| 23:48.50 | tgrman | nat is a bad idea for voip in general |
| 23:48.51 | [av]bani | and with asterisk, there's no reinvite |
| 23:49.10 | [av]bani | so all your rtp streams are forced to route through your asterisk, whether you like it or not |
| 23:49.30 | [av]bani | for stuff like remote pbxes, this is unacceptable |
| 23:49.32 | Zodiacal | when would you have a topology that put the server on another network than the phones? |
| 23:49.52 | Zodiacal | ic |
| 23:49.58 | Zodiacal | yeah, my stuffs all on the same net |
| 23:49.59 | [av]bani | often? remote pbx is quite common. |
| 23:50.09 | [av]bani | theres lots of companies selling outsourced pbx |
| 23:50.29 | [av]bani | just plonk phones on your network, register them with the pbx. companies like it because they dont have to manage anything |
| 23:50.29 | tgrman | service provider, user connects phone at home, which registers to Asterisk at office, etc |
| 23:51.16 | tgrman | i though everyone liked paying Avaya to come manage their PBX :-) |
| 23:51.27 | [av]bani | without reinvite support, your rtp streams have to go back to the remote pbx just to reach a phone in the same room |
| 23:51.40 | [av]bani | which is the case currently with sccp and asterisk |
| 23:52.05 | tgrman | unless you have a smart NAT or SBC, the same will happen if you're behind a NAT, even with reinvite |
| 23:52.21 | [av]bani | as stated before, sccp does not do nat |
| 23:52.34 | [av]bani | you can mangle it to, with baseball bats and sledgehammers |
| 23:52.42 | [av]bani | but it's not native, unlike sip |
| 23:52.51 | Zodiacal | hehe |
| 23:53.15 | Zodiacal | cisco sip firmware is criple ware.. |
| 23:53.27 | [av]bani | cisco is punishing users for daring to use sip |
| 23:53.31 | Zodiacal | yep |
| 23:53.36 | [av]bani | sccp is THE ONE TROO PROTOCOL, PEONS |
| 23:53.44 | tgrman | the next version of call manager is supposedly going to support SIP |
| 23:53.56 | tgrman | what "suppot" means, i'm not sure |
| 23:53.58 | [av]bani | i bet cisco is angry about it |
| 23:54.01 | *** join/#asterisk |omni| (i=rob@216.64.178.146) |
| 23:54.01 | Zodiacal | interesting.. u think they would update the sip firmware for the phones too? |
| 23:54.09 | [av]bani | sip kicked sccp's ass, and cisco is grudgingly supporting it now |
| 23:54.24 | [av]bani | YOU WANT SIP? WE WILL GIVE YOU SIP HEHEHE MORONS |
| 23:54.36 | [av]bani | Cisco CrippleSIP(tm) |
| 23:54.51 | Zodiacal | SIP should sue saying unless you give these range of sip features its not really SIP |
| 23:54.59 | Zodiacal | so that they have to say Cisco CrippleSIP |
| 23:55.22 | tgrman | for NAT support IAX is better than SIP |
| 23:55.30 | [av]bani | cisco would just call it Cisco Intermanager Protocol(tm) orsomething |
| 23:55.35 | [av]bani | which just "happens to work with sip" |
| 23:55.38 | |omni| | what issues are you having with SIP on Cisco hardware? |
| 23:55.39 | asterboy | I've followed the direction for CALL PARKING here: http://www.voip-info.org/wiki/index.php?page=Asterisk+call+parking |
| 23:55.46 | Zodiacal | omni no hints |
| 23:55.48 | Zodiacal | for one |
| 23:55.50 | [av]bani | |omni|: cisco doesnt support blf with sip, at all. period. |
| 23:55.57 | asterboy | But when I press # and 700 I don't get to call park...nothing happens. |
| 23:56.10 | [av]bani | |omni|: that is really a killer. |
| 23:56.19 | |omni| | blf? |
| 23:56.26 | [av]bani | hints/presence |
| 23:56.31 | asterboy | What is the prcedure to setup call park? |
| 23:56.36 | |omni| | got it |
| 23:56.53 | tgrman | for features like that, use a Polycom |
| 23:57.07 | [av]bani | about 90% of the features of sip are not supported with cisco's sip |
| 23:57.10 | [av]bani | you can make calls, thats about it |
| 23:57.21 | [av]bani | no g729 either iirc |
| 23:57.42 | asterboy | I want to be able to put a call on hold and pickup at another SIP Polycom phone. |
| 23:57.47 | tgrman | it does g729 |
| 23:57.50 | asterboy | call park does that nicely, no? |
| 23:57.56 | [av]bani | nicely? thats relative |
| 23:58.01 | asterboy | lol |
| 23:58.20 | [av]bani | sip-b would be better |
| 23:58.31 | asterboy | I think the polycom is eating my "#" for * to pickup that call park. |
| 23:58.54 | asterboy | if that is what is needed, (seems to be the default). |
| 23:59.36 | tgrman | with park, what Asterisk is really doing is transferring a call to a parking extension, then you dial that extension from the other phone to retrieve the call |
| 23:59.52 | [av]bani | BLA would be better than park |
| 23:59.52 | Zodiacal | anyone know if 911 operators get mad if you test dial 911? :) |
| 23:59.57 | [av]bani | coming summer 2006 :/ |
| 23:59.58 | |omni| | uhh..ya |
