| 00:00.00 | AdamB0122 | ok |
| 00:00.01 | JT | it is not used for T1 as far as i'm aware |
| 00:00.08 | AdamB0122 | i dont think so either |
| 00:00.23 | JT | you don't need zttranscode loaded |
| 00:00.41 | AdamB0122 | yea, |
| 00:00.43 | AdamB0122 | i keep removing it |
| 00:00.49 | AdamB0122 | but it keeps finding its way home somehow |
| 00:00.56 | AdamB0122 | and I'm not retarting the box |
| 00:01.17 | AdamB0122 | hm |
| 00:01.25 | AdamB0122 | every time i bounce Asterisk zttranscode restarts |
| 00:01.38 | JT | some script or something is setup wrong |
| 00:01.47 | *** join/#asterisk ManxPower (n=manxpowe@015-802-134.area5.spcsdns.net) |
| 00:02.19 | AdamB0122 | hm |
| 00:02.32 | AdamB0122 | What do you guys suggest for a front end. |
| 00:02.39 | JT | front end? |
| 00:02.47 | AdamB0122 | like, web gui or something |
| 00:02.55 | JT | vi or a text editor of your choice |
| 00:02.58 | AdamB0122 | or non-inteligent people can do simple tasks |
| 00:03.03 | AdamB0122 | yea, |
| 00:03.05 | AdamB0122 | I prefer Vi |
| 00:03.05 | JT | we don't use awful web guis here |
| 00:03.13 | AdamB0122 | but I can't exactly say that to the sales department |
| 00:03.25 | JT | why do they need to configure the pbx? |
| 00:03.47 | AdamB0122 | and I'll be damned if they're going to come crying to me every time they want to get a recording, or change someone's password, or extension |
| 00:04.36 | JT | do they configure the current pbx? |
| 00:04.49 | AdamB0122 | no, I do. (dont like that either) |
| 00:05.03 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:05.05 | JT | that's the way it should be |
| 00:05.27 | JT | you're asking for a world of hurt if you let non-technical people configure such an important device |
| 00:05.30 | AdamB0122 | i mean like |
| 00:05.34 | AdamB0122 | i dont mean configure |
| 00:05.39 | AdamB0122 | i mean things like web-based voicedmail |
| 00:05.43 | AdamB0122 | the recording that they do |
| 00:05.47 | AdamB0122 | configuration i'm fine with |
| 00:06.06 | AdamB0122 | I've got a sales department that LOVES the fact that they can record a call, and use it for training |
| 00:06.21 | AdamB0122 | Trixbox offers a "ARI" |
| 00:06.31 | AdamB0122 | which is neat, but i'm sure works only with trixbox |
| 00:06.40 | AdamB0122 | and trixbox is something i'd prefer to avoid |
| 00:06.58 | JT | yes, it's evil |
| 00:07.08 | AdamB0122 | yea, I've been playing with it |
| 00:07.20 | AdamB0122 | and its retarded method of making a million and half _additonal files pisses me off |
| 00:07.31 | AdamB0122 | + any time i edit ANYTHING in the command line, everything breaks |
| 00:07.37 | AdamB0122 | hastle after haslte. |
| 00:08.01 | JT | ~trixbox |
| 00:09.15 | jbot | trixbox is, like, a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 00:09.18 | JT | ;) |
| 00:09.18 | AdamB0122 | done any work with the new "AsteriskNow"? |
| 00:09.18 | JT | no |
| 00:09.18 | DrukenLPY | anyone know how i would "cut" the username out of the channel name? if the username is a fixed 6 length? |
| 00:09.18 | blitzrage | ManxPower: yo! |
| 00:10.02 | AdamB0122 | lol, bit of a late responce mr bot |
| 00:10.24 | ManxPower | AdamB0122: you are not supposed to edit trixbox config files |
| 00:10.34 | AdamB0122 | i know |
| 00:10.36 | AdamB0122 | and thats annoying |
| 00:10.41 | JT | AdamB0122: what's "responce"? ;) |
| 00:10.42 | AdamB0122 | trixbox in general is annoying |
| 00:10.49 | Innatech | Then run plain * . |
| 00:10.55 | AdamB0122 | Bad spelling and entirely too little sleep :p |
| 00:11.10 | JT | haha what time is it now? |
| 00:11.25 | ManxPower | ~zeeek |
| 00:11.58 | jbot | [zeeek] someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 00:11.58 | AdamB0122 | right now its 9pm, been here since 9am, and as you saw last night... i was up untill about 4am |
| 00:12.13 | AdamB0122 | ROFL |
| 00:13.02 | blitzrage | ManxPower: nice! Where you going? |
| 00:13.04 | ManxPower | That statement is true on SO many levels |
| 00:13.13 | ManxPower | blitzrage: damn fucking new orleans |
| 00:13.17 | ManxPower | AGAIN |
| 00:13.27 | blitzrage | ManxPower: what for? |
| 00:13.36 | ManxPower | blitzrage: work |
| 00:13.39 | blitzrage | fun stuff |
| 00:13.49 | ManxPower | a couple of asterisk upgrades and new installs |
| 00:14.06 | blitzrage | sound as busy as me |
| 00:14.15 | blitzrage | did you hear that you can start using 1.4 now? :) |
| 00:15.13 | ManxPower | blitzrage: Digium has finally moved their corporate PBX to an actual 1.4.x RELEASE? |
| 00:15.21 | blitzrage | yep :) |
| 00:15.41 | ManxPower | I'll bet you won't tell me how many bugs they fixed as a result of it, huh? |
| 00:16.08 | ManxPower | blitzrage: And actually, 1.4 is now a viable option because of it. |
| 00:18.28 | DrukenLPY | ManxPower: what variables are available by using SIP_HEADER? |
| 00:18.58 | ManxPower | DrukenLPY: Hell if I know. I never need to deal with SIP headers except for setting the alert info |
| 00:19.13 | ManxPower | Isn't it documented in README.variables |
| 00:19.18 | DrukenLPY | hehe honest answer :) |
| 00:19.37 | ManxPower | DrukenLPY: %99.999 of the time you do not need to KNOW anything about the actual protocol |
| 00:20.09 | blitzrage | unless you are using (Open)SER :) |
| 00:20.14 | ManxPower | DrukenLPY: You must be in an ITSP enviroment |
| 00:20.44 | blitzrage | ManxPower: not sure how many were fixed as a direct result, I've only been working for them for a couple weeks :) |
| 00:20.45 | ManxPower | blitzrage: With OpenSER you need to know enough to WRITE your own SIP proxy before being able to accomplish anything useful with it. |
| 00:20.46 | DrukenLPY | how'd ya guess... hehe |
| 00:20.54 | blitzrage | ManxPower: indeed |
| 00:21.07 | blitzrage | I'm still not that great at it |
| 00:21.09 | ManxPower | DrukenLPY: because pretty much only VERY large companies and ITSPs really care about things like SIP_HEADER |
| 00:21.29 | blitzrage | ya... and even I don't use it that much |
| 00:21.40 | *** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net) |
| 00:21.42 | ManxPower | I come from a corporate enviroment where even setting up DUNDi or ENUM is more work than manually managing routes |
| 00:21.49 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:22.20 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:22.20 | blitzrage | there were so many things I didn't use until I started doing clustered environment |
| 00:22.23 | ManxPower | Amtrak Train 19: 2 hrs behind schedule |
| 00:22.27 | blitzrage | ugh |
| 00:22.42 | ManxPower | blitzrage: I hate not having a car |
| 00:22.54 | blitzrage | ManxPower: ya... I don't mind it because I live in a city with good public transport |
| 00:23.12 | ManxPower | blitzrage: *nod* |
| 00:23.21 | ManxPower | I live in a city that does not even have cab service |
| 00:23.29 | blitzrage | ouch |
| 00:24.04 | Strom_M | blitzrage: you live in toronto, right? |
| 00:24.16 | ManxPower | blitzrage: MOST of the reason I hate not having a car is these damn business trips |
| 00:24.26 | ManxPower | I should have just rented a car for the week. |
| 00:24.39 | blitzrage | Strom_M: indeed :) |
| 00:24.44 | Strom_M | cool |
| 00:24.50 | Strom_M | i've never been there...i'd like to go sometime |
| 00:25.01 | blitzrage | ya, everyone I talk to seems to like it |
| 00:25.02 | Strom_M | i may visit montreal next month |
| 00:25.12 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 00:26.01 | tako-san | anyone know how to deal with the message "Internal RTCP NTP clock skew detected"? I have double checked my system time and also made sure that ntpd is running and configured properly. What else can I do? |
| 00:26.13 | DrukenLPY | Strom_M: any idea how i'd strip the username out of the channel ? |
| 00:26.23 | Strom_M | DrukenLPY: use Cut() |
| 00:26.26 | Strom_M | er, CUT() |
| 00:26.48 | AdamB0122 | hm |
| 00:26.55 | AdamB0122 | If I can place outbound calls fine |
| 00:27.09 | AdamB0122 | if I dial out, and I can hear myself fine, and have a conversation |
| 00:27.14 | ManxPower | I love Toronto |
| 00:27.21 | AdamB0122 | but if i dial in, it doesn't play any sounds |
| 00:27.29 | Strom_M | AdamB0122: SIP? |
| 00:27.34 | AdamB0122 | Yes |
| 00:27.36 | AdamB0122 | er |
| 00:27.39 | AdamB0122 | Zap external |
| 00:27.45 | Strom_M | that's not SIP! |
| 00:27.47 | AdamB0122 | T1 coming off a channel bank |
| 00:27.54 | AdamB0122 | sorry, its SIP internally |
| 00:27.56 | ManxPower | AdamB0122: If someone tells you to run Answer() first, they are an idiot and you should not listen to anything else they say. |
| 00:28.27 | AdamB0122 | <.< |
| 00:28.35 | AdamB0122 | Ok. |
| 00:29.26 | Innatech | tako-san : is it a new install? I saw that for the first day or so * was up, and then it went away (presumably NTP took a while to sync up.) |
| 00:29.28 | ManxPower | AdamB0122: We'll make a deal with you. We won't call you an idiot for not properly describing the problem and your environment if you restate your issue. |
| 00:29.40 | Strom_M | tako-san: run ntpdate |
| 00:29.45 | tako-san | Innatech: Not a new install |
| 00:29.50 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-245.shs.nyc.wayport.net) |
| 00:30.09 | ManxPower | You get bonus points for a 1 line ascii diagram as well |
| 00:30.12 | blitzrage | ManxPower: lol |
| 00:30.17 | Innatech | and maybe ntpq -p , if you want to verify your NTP setup. |
| 00:30.23 | ManxPower | Why does RTP care about NTP anyway?" |
| 00:30.33 | blitzrage | ManxPower: I found a use for Answer() today though... apparently Queue() doesn't automatically answer the line |
| 00:30.44 | wothinn | Does anyone know why FWD would be refusing my registration attempt on IAX2 using 863638 as my username and the password I set on my account as my password? The account is newly set-up. Is there a wait time? |
| 00:31.03 | tako-san | Strom_M: ! "no ntp servers found"! very strange. it has internet connectivity. |
| 00:31.09 | Strom_M | how about |
| 00:31.18 | Strom_M | where are you located? |
| 00:31.31 | tako-san | bc, canada |
| 00:31.34 | AdamB0122 | I have a asterisk pbx, with a TE120P T1 card... |
| 00:31.43 | Strom_M | ntpdate ca.pool.ntp.org |
| 00:31.54 | AdamB0122 | If I start a call, from the PBX, it goes out fine, and i can answer it on my cell, and talk no problem |
| 00:32.08 | tako-san | 24 Jul 17:31:53 ntpdate[7795]: the NTP socket is in use, exiting |
| 00:32.16 | Strom_M | stop ntpd? |
| 00:32.41 | AdamB0122 | I currenlty have from_pstn, the incoming context for a zap call, to goto a answer(), and play a sound. |
| 00:33.07 | AdamB0122 | If I call in from my cell phone, I see the start of the call on Zap/1-1 |
| 00:33.15 | Strom_M | AdamB0122: does that sound work if you call internally? |
| 00:33.16 | AdamB0122 | I see on the CLI that its "playing" a sound |
| 00:33.24 | AdamB0122 | but, the sound doens't come out of the phone |
| 00:33.27 | AdamB0122 | Strom_M > Yes |
| 00:33.30 | ManxPower | AdamB0122: So you have Channelized T-1 <-> Channel Bank <-> Asterisk Analog Cards <-> Asterisk <-> SIP Softphones? |
| 00:33.39 | tako-san | Strom_M: Thank you |
| 00:33.54 | AdamB0122 | ManxPower > yes |
| 00:34.05 | AdamB0122 | Strom_M > if I dial 7777, it goes to from_pstn, and i can hear the sound |
| 00:34.19 | ManxPower | AdamB0122: Why are you converting the T-1 into analog before going into Asterisk? |
| 00:34.25 | JT | ManxPower: it doesn't |
| 00:34.30 | Strom_M | ManxPower: he just said he had a TE120P |
| 00:34.31 | JT | it pulls it from WAN2 port |
| 00:34.34 | ManxPower | jt: He jus said he did. |
| 00:34.41 | JT | he is pulling a couple of timeslots off at the adit 600 |
| 00:34.41 | AdamB0122 | fuck I'm confused |
| 00:34.56 | JT | ManxPower: i spent ages working out his setup last night :) |
| 00:34.57 | ManxPower | JT: That was a test. He failed it. I won't help him now. |
| 00:35.17 | ManxPower | JT: I shall leave him in your expert hands then. |
| 00:35.26 | JT | ManxPower: what's the test? |
| 00:35.50 | AdamB0122 | apperently something about asterisk analog cards vis a 120P |
| 00:35.53 | AdamB0122 | vs* |
| 00:35.54 | DrukenLPY | Strom_M: you ever used cut? cause it's not making much sence to me... |
| 00:36.03 | Strom_M | DrukenLPY: yes, it's quite simple |
| 00:36.11 | ManxPower | I've been without internet for a few days. Turn out my verizon software is so old it was trying to connect to their test network for the higher speed internet service. |
| 00:36.16 | Strom_M | CUT(variable,delimiter,which-field-to-return) |
| 00:36.16 | ManxPower | JT: describing in the diagram something illogical. He said that was his configutation when it was not. |
| 00:36.30 | ManxPower | If he can't even properly know what he has He is far beyond my help |
| 00:36.32 | wothinn | http://pastebin.ca/632623 <-- my iax.conf and error message while trying to connect to FWD. |
| 00:36.34 | *** join/#asterisk stridernzl (n=neville@125-237-98-1.jetstream.xtra.co.nz) |
| 00:36.37 | DrukenLPY | Strom_M: feel like helping a guy out? can i paste you the two lines i think should give me it? |
| 00:36.41 | JT | ManxPower: ah |
| 00:36.43 | wothinn | If anyone has ideas, I'd really appreciate it. |
| 00:36.45 | Strom_M | DrukenLPY: syre |
| 00:36.46 | Strom_M | er, sure |
| 00:37.11 | JT | ManxPower: well it's channelised T1 > Adit 600 > TE120P Asterisk > softphones |
| 00:37.55 | AdamB0122 | So I presume the TE120P doesn't consitiute a "Asterisk Analog Card" |
| 00:38.04 | JT | no, T1s are digital |
| 00:38.07 | ManxPower | JT: What the ADIT? |
| 00:38.08 | JT | you should know this by now |
| 00:38.12 | AdamB0122 | channelbank |
| 00:38.23 | ManxPower | ...er... WHY is there an ADIT there? |
| 00:38.32 | AdamB0122 | i didn't do it |
| 00:38.34 | JT | ManxPower: the Adit pulls off a couple of channels for other lines, before arriving at the pbx |
| 00:38.52 | JT | perfectly acceptable channelised T1 behaviour :) |
| 00:38.56 | ManxPower | JT: Maybe he is not beyond hope, afterall. |
| 00:39.19 | ManxPower | JT: IT is the RECOMMENDED setup in my world for analog lines |
| 00:39.25 | *** join/#asterisk Pettson (i=andreas@seleya.sbin.se) |
| 00:39.43 | ManxPower | Fax, modem, CC machine? Channel bank to peel off some channels before the T-1 gets to Asterisk. |
| 00:39.50 | JT | ah |
| 00:39.58 | AdamB0122 | yea |
| 00:40.01 | ManxPower | Well, I have some ideas as to why there might not be sound. |
| 00:40.05 | JT | well what if you have a real service, ie. PRI? ;) |
| 00:40.24 | *** join/#asterisk galeras (n=root@201.245.103.169) |
| 00:40.28 | JT | can't pull off channels then |
| 00:40.31 | galeras | howdy |
| 00:40.45 | ManxPower | JT: My Telco will do things like 1-6 B-Chann, 7-12 FXO, 13-23 Internet, 24 D-chan |
| 00:40.55 | ManxPower | JT: The hell you can't. |
| 00:40.55 | AdamB0122 | oh god. |
| 00:41.00 | JT | wow :o |
| 00:41.06 | JT | ManxPower: most probably won't though |
| 00:41.10 | ManxPower | "will do things" == "will do things at our request" |
| 00:41.25 | galeras | i'm trying to connect my *box with an alcatel 4400 via E1 |
| 00:41.26 | Strom_M | AdamB0122: did you ever call the telco and find out what you have? |
| 00:41.36 | ManxPower | Most of the time it is just 1-20 B-Chan, 21-23 FXO, 24 D-Chan |
| 00:41.53 | JT | Strom_M: as far as they're concerned, he has a bunch of CAS lines, i doubt they know what his ADit is configured to take off |
| 00:42.04 | JT | ManxPower: what telco is that? |
| 00:42.10 | AdamB0122 | XO Communications |
| 00:42.15 | AdamB0122 | and No, they didn't know |
| 00:42.19 | ManxPower | JT: XFone, regional |
| 00:42.28 | galeras | my problem is, when alcatel try to get dialtone from my * box i get the message: |
| 00:42.28 | galeras | Extension '' in context 'from-alcatel' does not exist. |
| 00:42.28 | AdamB0122 | oh, the crazy one |
| 00:42.30 | Strom_M | AdamB0122: your telco doesn't know? who did you call? |
| 00:42.32 | JT | ManxPower: i don't think anyone does CAS in australia. |
| 00:42.42 | ManxPower | AdamB0122: Does the problem happen if you remove the channel bank for testing |
| 00:42.47 | galeras | and from-alcatel has _. extension |
| 00:42.59 | AdamB0122 | I can go try |
| 00:43.03 | AdamB0122 | one moment |
| 00:43.09 | ManxPower | galeras: _. never fixes anything |
| 00:43.12 | *** join/#asterisk javb (n=javb@190.80.235.113) |
| 00:43.25 | javb | any ideas on how to make asterisk load by default on ubuntu server when booting? |
| 00:43.56 | galeras | your right, what extension do i need to configure in that case? |
| 00:43.58 | snuff-work | generally u can find the init script under ur asterisk source /contrib |
| 00:44.01 | ManxPower | galeras: We really can't help you. |
| 00:44.04 | Strom_M | javb: /usr/src/asterisk/contrib/ and look for the init script |
| 00:44.25 | ManxPower | Since we have no idea how the alcatel is setup, how asterisk is set up, the line type, signalling and protocol used. |
| 00:44.46 | AdamB0122 | hm. |
| 00:44.46 | ManxPower | galeras: your problem is not in extensions.conf |
| 00:44.51 | javb | dont have 'asterisk' on /usr/src |
| 00:45.01 | ManxPower | javb: WHERE IS YOUR ASTERISK SOURCE CODE? |
| 00:45.09 | AdamB0122 | What all configuration changes need to happen now that I'm bypassing the ADIT? |
| 00:45.31 | ManxPower | AdamB0122: Do you have a red alarm? |
| 00:45.33 | JT | AdamB0122: none |
| 00:45.39 | javb | my asterisk source code.. mmm, where i download and uncompressed it? |
| 00:45.45 | AdamB0122 | Yes |
| 00:45.50 | Strom_M | javb: yes |
| 00:45.50 | ManxPower | javb: then that would be where the script is |
| 00:45.56 | AdamB0122 | Wildcard TE12xP Card 0 RED 53 0 0 |
| 00:46.07 | ManxPower | AdamB0122: Red alarm means "cable problem" |
| 00:46.11 | wothinn | Well, that's just dandy... I can log in with FWD.Communicator, but Asterisk won't register to FWD. |
| 00:46.12 | javb | So, should i copy that scrip somewhere? |
| 00:46.23 | ManxPower | chances are you have a T-1 crossover cable from the channel bank to Asterisk |
| 00:46.27 | AdamB0122 | yea |
| 00:46.32 | JT | ManxPower: yes, i told him to make one |
| 00:46.37 | AdamB0122 | I made it last night, |
| 00:46.41 | ManxPower | you would want a straight thru from the telco to Asterisk |
| 00:46.43 | AdamB0122 | use a standard Cat5? |
| 00:46.48 | JT | because he was getting red alarm using normal ethernet cable and was connect to the CB |
| 00:46.59 | AdamB0122 | yea |
| 00:47.15 | JT | AdamB0122: cat5 is the type of cable, i think you mean ethernet/TIA568A |
| 00:47.17 | ManxPower | you need crossover for Asterisk/Channelbank, but not Telco/Asterisk or Telco/Channelbank |
| 00:47.29 | AdamB0122 | Now, I can try this if it will help, but I really can't take off the channel bank, as we do have a CC machine and a fax machine that i'd rather not have to figure out with asterisk |
| 00:47.36 | ManxPower | A plain ethernet straight thru cable will work for straight thru T-1 |
| 00:48.11 | ManxPower | AdamB0122: I understand. I just want to make SURE the channel bank is not causing some sorts of issue and bypassing it should be a mind numbingly simple thing to do. |
| 00:48.39 | AdamB0122 | cool |
| 00:48.47 | AdamB0122 | give me a moment, let me run another cable |
| 00:49.10 | ManxPower | If bypassing it does not solve the problem, then you know you don't have to waste any time working on it. If bypassing the channel bank DOES solve the problems you now have a place to start looking. |
| 00:49.31 | Sedorox | T1 crossovers are fun |
| 00:50.04 | javb | ? |
| 00:50.19 | JT | 1 to 4, 2 to 5, 4 to 1, 5 to 2, what does that spell?!? |
| 00:50.30 | ManxPower | T-1 Crossovers are "look up the diagram on the wiki, follow it" |
| 00:50.59 | ManxPower | It is one of the very few things I consider the Wiki useful for. |
| 00:51.13 | Strom_M | i just remember my color codes and pinouts |
| 00:51.14 | JT | haha jaded on the outdated and inaccurate info |
| 00:51.32 | Sedorox | hmm |
| 00:52.00 | javb | i copy "rc.debian.asterisk" which was in "contrib/init.d" to "etc/init.d" .. but nothing, any ideas, please help. im kind of newbie |
| 00:52.02 | ManxPower | blitzrage: Also talk to me after a week of no restarts of the Digium PBX |
| 00:52.36 | ManxPower | javb: This really isn't a linux support channel |
| 00:52.41 | Sedorox | jmm |
| 00:52.53 | blitzrage | ManxPower: ok :) |
| 00:53.26 | Strom_M | ManxPower: i'm thinking of a two syllable word, one of which rhymes with "kite" and the other of which rhymes with "mud" |
| 00:53.31 | javb | ManxPower: i undestand, but im trying to make asterisk load by default on a linux distribution. |
| 00:53.31 | Strom_M | :) |
| 00:53.42 | Sedorox | Last reload: 11 weeks, 3 days, 23 hours, 45 minutes, 3 seconds |
| 00:53.44 | Sedorox | :D |
| 00:53.53 | wothinn | Does anyone know why registration with FWD would work with their Communicator package but not Asterisk? iax.conf at http://pastebin.ca/632623. |
| 00:54.25 | Strom_M | http://www.stromcarlson.com/misc/lolte410p-small.jpg |
| 00:54.38 | ManxPower | Strom_M: I think the term is "eat your own dog food |
| 00:54.42 | ManxPower | Sedorox: What verison of Asterisk? |
| 00:54.46 | Strom_M | ManxPower: i'm joshing |
| 00:55.02 | Sedorox | ManxPower: its out of date, badly |
| 00:55.13 | ManxPower | Sedorox: 1.4x or 1.2x? |
| 00:55.19 | Strom_M | 1.0.x! |
| 00:55.20 | Sedorox | Asterisk 1.2.14 |
| 00:55.48 | ManxPower | Sedorox: My point was that until Digium had the balls to upgrade their corporate PBX to 1.4, I was not going to upgrade any of my customers to 1.4 |
| 00:56.13 | Sedorox | ah |
| 00:56.18 | Sedorox | I haven't played with 1.4 yet |
| 00:56.29 | ManxPower | and my point to blitzrage is that many of the issues I've encountered with Asterisk over the years only happen after the system gets some usage. |
| 00:56.31 | Sedorox | honestly haven't played too much with 1.2 :/ |
| 00:56.37 | Sedorox | ah |
| 00:56.40 | AdamB0122 | Alright. |
| 00:56.53 | ManxPower | I am happy that Digium is/has upgrade(d) to 1.4.x |
| 00:56.53 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:56.57 | AdamB0122 | sorry about the wait, couldn't find anymore 20ft straight-through cables, so i had to make one |
| 00:57.20 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:57.20 | Sedorox | they have? |
| 00:57.22 | Sedorox | heh |
| 00:57.23 | ManxPower | AdamB0122: My suggestion was one of those "if it takes more than 10 mins to do it, it may not be worth the trouble" |
| 00:57.51 | *** join/#asterisk zapa (n=hzapa@201.147.132.2) |
| 00:59.05 | ManxPower | I WANT A CIGARETTE |
| 00:59.18 | AdamB0122 | hm |
| 00:59.26 | jgoddess | ManxPower we upgradef from 1.09 to v1.4 there have been some nice fixes but nice problems |
| 00:59.28 | AdamB0122 | now if i do ztcfg -vvvvv, it hangs |
| 00:59.32 | jgoddess | like vnak packet storming =P |
| 01:00.42 | Sedorox | Cig's are bad for you |
| 01:00.52 | ManxPower | jgoddess: I don't like it when my users come banging on my door, waving torches and screaming "burn the geek". They tend to do that when the PBX goes down. |
| 01:01.10 | ManxPower | Sedorox: It is also unhealthy to be around me when I am denied them. |
| 01:01.23 | Sedorox | true... |
| 01:01.25 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 01:01.28 | Sedorox | my parents smoke |
| 01:01.30 | Sedorox | can't stand it... |
| 01:01.35 | Sedorox | can't wait to be back at school |
| 01:01.41 | riddlebox | does anyone have the Aastra 9112I phone? |
| 01:01.51 | shido6 | whats wrong riddlebox? |
| 01:01.59 | ManxPower | I'm sure it is the fault of the tree hugging hippies and the gay martians! They will destroy this country! |
| 01:02.24 | riddlebox | shido6, I just want to know if it has the transfer key, and conference keys on the phone built in? |
| 01:02.49 | [TK]D-Fender | riddlebox, I seriously hope you're not planning on buying one..... |
| 01:03.12 | zapa | hi all, does any body know some compatibilty isue with PowerEdge SC 440 with TDM400 Digium cards |
| 01:03.23 | ManxPower | [TK]D-Fender: is it one of those 802.11 FiFi phones? |
| 01:03.35 | [TK]D-Fender | ManxPower, No, its just uber-low-end |
| 01:03.38 | riddlebox | [TK]D-Fender, whats wrong with them? |
| 01:03.45 | ManxPower | zapa: your extensive search of the mailing list archves was not helpful? |
| 01:03.47 | [TK]D-Fender | riddlebox, You can do a lot better for your money |
| 01:04.10 | riddlebox | [TK]D-Fender, what other phones do you suggest, these are just phones for my house |
| 01:04.45 | [TK]D-Fender | riddlebox, Polycom IP320 kills it |
| 01:05.03 | Strom_M | [TK]D-Fender: have you gotten your hands on a 320 yet? |
| 01:05.17 | [TK]D-Fender | Strom_M, no personally, but professionally. |
| 01:05.23 | Strom_M | close enough |
| 01:05.36 | [TK]D-Fender | Strom_M, All the usually Polycom goodness. |
| 01:05.44 | [TK]D-Fender | Strom_M, join/split rocks |
| 01:05.48 | zapa | Hi ManxPower sorry i don't find usefull information in list |
| 01:05.54 | shido6 | Go Polycom. |
| 01:05.56 | ManxPower | Some day the whole USA will have EVDO coverage -- Utopia will have arrived. |
| 01:05.57 | Qwell | is that the fake hold feature? |
| 01:06.29 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:06.44 | [TK]D-Fender | Qwell[], No, thats Polycom's way to take and 2 calls and merge for a 3-way call, and break any conference up into the seperate calls the same way |
| 01:07.01 | [TK]D-Fender | Qwell[], Its a feature I've never seen on any other phone |
| 01:07.07 | Qwell | 'splain |
| 01:07.26 | Qwell | by "conference", you mean on-phone |
| 01:07.42 | [TK]D-Fender | Qwell[], 2nd call comes in while you're on your first? NP, just hit [join] and your other call on hold and bingo, instant 3-way call |
| 01:07.49 | [TK]D-Fender | Qwell[], yes, phone based |
| 01:08.06 | [TK]D-Fender | Qwell[], basically being able to merge and toss around calls every which way. |
| 01:08.16 | ITiliti | I am able to get the DID information from the SIP Header written to the debug log, how can I get it written to the CDR database? |
| 01:08.34 | ManxPower | ITiliti: Uh you get the DID info in ${EXTEN} |
| 01:08.42 | ITiliti | We have a bunch of the aastra 57I series, I have to say they rock. |
| 01:09.05 | ManxPower | and that info should be in the CDR unless you are doing something bizarre. |
| 01:09.10 | AdamB0122 | ManxPower > Well, I've wired it directly into the T1 modem using a straight-thru cable, and now the wildcard has a YEL alamr |
| 01:09.13 | ITiliti | not sure. I am using this to get t5he DID info: exten => s,n,NoOp(${SIP_HEADER(TO)}) |
| 01:09.18 | [TK]D-Fender | ITiliti, I'd happily trade mine for an IP 301 <------------ |
| 01:09.39 | ITiliti | really? are you using hte XML sripts for ti that ar eavailable? |
| 01:09.54 | ITiliti | to extract the DID number. |
| 01:10.09 | *** join/#asterisk CVirus (n=GoD@62.135.96.251) |
| 01:10.15 | [TK]D-Fender | ITiliti, nope... and little need. |
| 01:10.17 | ITiliti | now way, why do tyou say that? I love the 57 i. Why don;t you like it? |
| 01:10.26 | AdamB0122 | did a ztcfg and then rebounced asterisk and the alarm is ok now |
| 01:11.03 | ITiliti | we ove the fact that we can XML binded to a button so people can easily page, intercom, or even pull XML from news feeds etc. |
| 01:11.16 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 01:11.30 | AdamB0122 | hm |
| 01:11.32 | AdamB0122 | Ok |
| 01:11.33 | AdamB0122 | I call in |
| 01:11.48 | AdamB0122 | and I hear about the first 1/4 of a second of "you are about to enter a echo test" |
| 01:11.55 | AdamB0122 | and then the call is dropped |
| 01:12.04 | ITiliti | ManxPower- How can I get the DID into EXTENSION? |
| 01:13.01 | *** part/#asterisk galeras (n=root@201.245.103.169) |
| 01:13.02 | ManxPower | ITiliti: ${EXTEN} always contains the DID when the call arrive in Asterisk. If you use Goto or macros, that would overwrite that information, of course. |
| 01:13.19 | ManxPower | AdamB0122: is that different? |
| 01:13.31 | ITiliti | we are using feepbx, so that is probably what is causing it.. |
| 01:13.36 | JT | ~freepbx |
| 01:13.37 | jbot | [freepbx] unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 01:13.42 | ManxPower | ~zeeek |
| 01:13.43 | jbot | i guess zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 01:13.47 | AdamB0122 | well. via directly in, I hear about 1/4 of 'you' and via the channel bank, i dont |
| 01:13.57 | ManxPower | ITiliti: thank you for wasting 5 mins of my life that I will never get back |
| 01:14.05 | jgoddess | well it would be easier to use the full logs |
| 01:14.09 | ManxPower | AdamB0122: any messages on the console |
| 01:14.13 | ITiliti | that may be true, but we are selling around 25-30 installs a month right now using it and clients love it... |
| 01:14.13 | jgoddess | or enable high verbose on the clie |
| 01:14.21 | jgoddess | to see what errors are actually happening |
| 01:14.24 | ManxPower | AdamB0122: I assume JT told you to NOT use callprogress= or busydetect= |
| 01:14.27 | AdamB0122 | And now if i call in, i get a all cirtcuts are busy |
| 01:14.28 | jgoddess | so you will know why it goes to dead air |
| 01:14.32 | JT | ITiliti: cool, just don't bother us with it then |
| 01:14.34 | AdamB0122 | but not the system |
| 01:14.47 | JT | ITiliti: we cannot support freepbx |
| 01:14.54 | AdamB0122 | yea, its keeping those lines open |
| 01:15.05 | ITiliti | ManxPower- don;t be so mean! There is plenty of areas in Freepbx to write custom apps, contexts, etc. |
| 01:15.11 | AdamB0122 | if i do a show channels |
| 01:15.12 | ManxPower | AdamB0122: yes he told you to not use them or yes you are using them |
| 01:15.28 | JT | ITiliti: freepbx is utter crap, shame on you for selling it to others in a consulting capacity |
| 01:15.30 | ManxPower | ITiliti: I'm sure there are. This is not the place to get support for freepbx |
| 01:15.32 | AdamB0122 | he didn't say anything about those commands) |
| 01:15.34 | jgoddess | haha |
| 01:15.40 | jgoddess | DOWN WITH FREEPBX |
| 01:15.41 | jgoddess | =P |
| 01:15.43 | AdamB0122 | and "yea" was a crap-word... didn't mean anything |
| 01:15.46 | ManxPower | AdamB0122: make sure you are not using either option |
| 01:15.54 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 01:15.56 | ManxPower | AdamB0122: what version of Asterisk? |
| 01:16.01 | ITiliti | not asking you to support freepbx. I am jst trying to figure out how to take this exten => s,n,NoOp(${SIP_HEADER(TO)}), and add it to a variable that I can add to the CDR database. |
| 01:16.06 | AdamB0122 | 1.2.20 |
| 01:16.30 | ManxPower | ITiliti: ANY question you have is freepbx support. With freepbx or any of the other guis ALL OF THE RULES are different. |
| 01:16.34 | ITiliti | nothing to do with freepbx... |
| 01:16.52 | ITiliti | fine |
| 01:16.54 | AdamB0122 | callprogress and busydetect would be in the zap configs, correct? |
| 01:17.01 | ManxPower | ITiliti: Bullshit. It has everything to do with a crap dialplan that is written so complex we can't even begin to diagnose the problem |
| 01:17.26 | ManxPower | ITiliti: this is how you do that: exten => _NXXNXXXXXX,1,NoOp(${SIP_HEADER(TO)}) |
| 01:17.34 | ITiliti | relax. who pissed in your wheaties... |
| 01:17.40 | ManxPower | now if your call is on exten => s, then the DID would be "s" |
| 01:17.53 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:17.54 | ITiliti | got it. Thanks... |
| 01:18.03 | ManxPower | You can try looking at ${DNID}, which in some situations would contain the ORIGINALLY dialed number. |
| 01:18.31 | ManxPower | for all of my macros, the first line is exten => s,1,Goto(${MACRO_EXTEN},1) to avoid the whole problem |
| 01:18.35 | ITiliti | That is another thought I was going to check out, but the asterisk log has the DID dialed for everyone I have tried after that last ,ine... |
| 01:18.50 | ITiliti | I will go ask the freepbx ppl. Thanks for you hel ManpowerX |
| 01:19.03 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 01:19.29 | *** part/#asterisk ITiliti (n=IceChat7@72.54.46.18) |
| 01:19.36 | ManxPower | ITiliti: You are calling Ford tech support for a problem with your Ford car. The problem is that the entire power train, engine, electrical system are all custom built by someone else. |
| 01:19.40 | ManxPower | bah! |
| 01:20.39 | Penggu | hi all. i've got the [default] extensions, which nobody should be reaching, as all my sip.conf lines have assigned (alternative) extensions, and my incoming zap calls also have assigned, alternative extensions. the [default] context is still there. I want to catch *anything* that somehow (by error or otherwise) gets there. what extensions would I use? |
| 01:21.07 | Penggu | I've already put an 's': s,1,NoOp(Someone shouldnt be here); s,n,HangUp() |
| 01:21.09 | jgoddess | holy cow |
| 01:21.10 | ManxPower | Penggu: You have just discovered one of the very, very few situations where you might consider exten => _. |
| 01:21.24 | Penggu | does _. cover 's' ? |
| 01:21.30 | ManxPower | Penggu: "s" is NOT a wildcard. |
| 01:21.36 | Penggu | i know.. |
| 01:21.45 | blitzrage | and yes, it matches EVERYTHING |
| 01:21.49 | AdamB0122 | http://rafb.net/p/irMeNn71.html |
| 01:21.50 | blitzrage | including 'h' |
| 01:21.53 | Penggu | hmm, k |
| 01:21.56 | AdamB0122 | Thats what the system does |
| 01:22.00 | Penggu | ill just use _. then |
| 01:22.02 | ManxPower | Penggu: _. covers everything. It will usually be run twice for a call because it also matches "h" |
| 01:22.02 | AdamB0122 | Where is the Zap/3-1 comeing from |
| 01:22.18 | Penggu | never had to use it |
| 01:22.35 | Penggu | what do you call them? default extensions? |
| 01:22.37 | AdamB0122 | and it says maybe 1/2 of the word "you" from the sound byte, and then kills the call. |
| 01:22.41 | ManxPower | AdamB0122: you have two calls coming in at exactly the same time. |
| 01:22.50 | ManxPower | I suspect a telco issue |
| 01:22.58 | AdamB0122 | Every single time that I call in from my phone? |
| 01:23.06 | AdamB0122 | like, Literally, EVERY time i call in, i get that |
| 01:23.12 | ManxPower | AdamB0122: YUP. |
| 01:23.26 | ManxPower | I assume the phone is a SIP phone? |
| 01:23.34 | AdamB0122 | cell phone |
| 01:23.39 | ManxPower | or are you calling into the line from your cell phone. |
| 01:23.46 | ManxPower | Call your telco, scream at them |
| 01:23.47 | AdamB0122 | Cell phone to land line |
| 01:24.14 | ManxPower | but before you do that add these to the beginning of your echo test part of your dialplan |
| 01:24.42 | ManxPower | Noop(EXTEN is ${EXTEN}, CALLERID(all) is ${CALLERID(all)}) |
| 01:24.47 | ManxPower | assuming your are using 1.4 |
| 01:24.54 | AdamB0122 | 1.2 |
| 01:25.08 | ManxPower | ..er...that will work in 1.2 as well |
| 01:25.11 | AdamB0122 | k |
| 01:25.55 | ManxPower | apparently there is a freight train dead on the tracks ahead of us |
| 01:26.14 | *** join/#asterisk ccesario_ (n=ccesario@201-0-53-167.dsl.telesp.net.br) |
| 01:26.43 | AdamB0122 | NoOp("Zap/1-1", "EXTEN is s| CALLERID(all) is "" <>") in new stack |
| 01:26.49 | AdamB0122 | thats what the NoOp outputted |
| 01:27.02 | ManxPower | AdamB0122: ONLY for Zap/1-1? |
| 01:27.06 | ccesario_ | WARNING[1103]: chan_sip.c:8023 check_auth: username mismatch, have <306>, digest has <305> |
| 01:27.11 | AdamB0122 | both |
| 01:27.17 | ccesario_ | somebody have idea ? |
| 01:27.18 | AdamB0122 | both Zap/3-1 and 1-1 |
| 01:27.19 | ManxPower | AdamB0122: you have a telco problem |
| 01:27.31 | JT | ManxPower: are you on a train? |
| 01:27.32 | ManxPower | ccesario_: only the obvious one |
| 01:27.36 | ManxPower | JT: Yes. |
| 01:27.42 | JT | ManxPower: what connectivity? |
| 01:27.50 | AdamB0122 | hm |
| 01:27.51 | AdamB0122 | gay. |
| 01:27.55 | AdamB0122 | telco problems are the suck |
| 01:27.57 | ManxPower | JT: Sprint 1xRTT |
| 01:28.04 | Penggu | is there a cmd to log a warning/error apart from NoOp() ? |
| 01:28.07 | ManxPower | I'll get EVDO in about 30 miles |
| 01:28.14 | Penggu | something with raised status |
| 01:28.17 | JT | heh ok |
| 01:28.22 | AdamB0122 | cause I would hate to fix the telco problem for me.... and land up killing the old phone system |
| 01:28.28 | Penggu | like, PanicStations() |
| 01:28.34 | AdamB0122 | cause they need the old system untill this one is ready to be put in place |
| 01:28.42 | JT | it could be a timing issue |
| 01:29.14 | AdamB0122 | JT > like, the T1 does timing instead of telco? |
| 01:29.53 | ManxPower | What is the signalling on the line? FXO, E&M, etc? |
| 01:30.00 | AdamB0122 | FXO |
| 01:30.15 | ManxPower | What country |
| 01:30.18 | AdamB0122 | US |
| 01:30.22 | AdamB0122 | er wait |
| 01:30.31 | AdamB0122 | fxs sorry |
| 01:30.36 | AdamB0122 | span=1,1,0,esf,b8zs |
| 01:30.36 | AdamB0122 | fxsks=1-5 |
| 01:30.52 | AdamB0122 | (and 1-5 is just because i know those 5 channels work) |
| 01:31.02 | JT | AdamB0122: T1s don't do timing, one end does timing |
| 01:31.30 | AdamB0122 | thats what I mean, sorry. the 120P doing the timing vs telco |
| 01:31.33 | Penggu | hmm |
| 01:31.40 | ManxPower | AdamB0122: I'm out of ideas, but if you paste your log output to the mailing list, describe what you tried, the symptoms, etc I'm pretty sure your problem is biazarre enough people will lose sleep over it |
| 01:31.41 | JT | AdamB0122: also zaptel has it's own timing issues if there are shared interupts, etc |
| 01:31.49 | Penggu | i don't like the double extension thing because of the _. triggering 'h' |
| 01:31.56 | JT | AdamB0122: try running zttest for a little bit |
| 01:32.10 | JT | AdamB0122: and take note of the lowest score and the normal score |
| 01:32.17 | AdamB0122 | getting almos solid 100% |
| 01:32.23 | AdamB0122 | i had 1 99.987 |
| 01:32.33 | AdamB0122 | and its got its own IRQ |
| 01:32.35 | ManxPower | So I asked the amtrak person "so there's no chance to let us out for a cigarette?", their answer is "No, we are in the middle of the woods." I came so close to saying "Listen here bitch, I smoke in the middle of the woods every day." |
| 01:32.57 | ManxPower | Penggu: if you have an exten => h in that context then _. won't match it |
| 01:33.08 | Penggu | before or after it? |
| 01:33.23 | *** join/#asterisk bjohnson (n=bjohnson@dsl-67-55-16-254.acanac.net) |
| 01:33.27 | ManxPower | order does NOT matter WITHIN a context |
| 01:33.28 | JT | AdamB0122: as long as it doesn't fall below 99.97% |
| 01:33.44 | AdamB0122 | no |
| 01:33.59 | AdamB0122 | lowest one is 99.98793 |
| 01:34.05 | JT | good |
| 01:34.10 | Penggu | so all 'hangups' end up under [default]? or just whatever extensions have been gathered ? |
| 01:34.13 | ManxPower | AHA! |
| 01:34.17 | AdamB0122 | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.997269 |
| 01:34.21 | Strom_M | ManxPower: take on me |
| 01:34.51 | ManxPower | Penggu: Um, I thought you had NO relation between [default] and any other context |
| 01:35.10 | Penggu | im testing getting to default (added an extension with a goto) |
| 01:35.22 | ManxPower | JT: I was going to ask you about CAS, but since you are in .au.... |
| 01:35.29 | Penggu | in case someone lands there by a dumb mistake |
| 01:35.31 | ManxPower | AdamB0122: I can only thing of one other thing. |
| 01:35.48 | JT | ManxPower: i've used it |
| 01:35.50 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 01:35.52 | JT | ManxPower: only on a channel bank |
| 01:36.00 | Penggu | quit |
| 01:36.05 | Penggu | woops.. this aint asterisk |
| 01:36.10 | ManxPower | AdamB0122: and that is that your T-1 is not a "normal" CAS T-1 and that the CAS bits are doing something weird |
| 01:36.10 | JT | telcos don't provide CAS here as far as i'm aware |
| 01:36.25 | AdamB0122 | Yea. something wierd is going on |
| 01:36.35 | ZX81 | still dropping calls? |
| 01:36.39 | AdamB0122 | I'm going to call our XO rep tomorrow and be like... 'wtf' |
| 01:36.49 | ManxPower | AdamB0122: can you double check EVERYTHING with regards to line type, signalling, etc on your channel bank to see if it is set to something weird that makes it work with your line? |
| 01:36.51 | ZX81 | heh that'll be fun |
| 01:36.55 | JT | ManxPower: it could be E&M and he's using the wrong mode |
| 01:37.29 | AdamB0122 | E&M instead of eft? |
| 01:37.31 | AdamB0122 | esf* |
| 01:37.34 | JT | no. |
| 01:37.52 | JT | instead of cas (fxo/fxsks) |
| 01:37.59 | ManxPower | E&M would be instead of FXS |
| 01:38.12 | JT | but it's probably better to check all your line specs with the telco |
| 01:38.24 | AdamB0122 | yea. |
| 01:38.27 | AdamB0122 | i'll just call them tomorrow |
| 01:38.38 | ManxPower | AdamB0122: check the settings on the Adit |
| 01:38.40 | Strom_M | AdamB0122: that's what i told you to do yesterday :/ |
| 01:38.49 | ManxPower | see what is different from your asterisk settings |
| 01:38.56 | JT | esf is extended super frame, and operates at a lower layer and isn't relevant to this problem |
| 01:39.24 | JT | ManxPower: what was the cas question? |
| 01:39.27 | Strom_M | oh baby, i love it when you b8zs me |
| 01:39.28 | *** join/#asterisk kroo (n=kroo@AToulon-152-1-94-212.w86-200.abo.wanadoo.fr) |
| 01:39.33 | kroo | hello everyone |
| 01:39.42 | kroo | I need a bit of help to install asterisk |
| 01:39.45 | kroo | is there anyone |
| 01:39.52 | kroo | ???\ |
| 01:39.57 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 01:39.57 | Strom_M | kroo: no, we're all busy eating cheese |
| 01:40.03 | JT | kroo: not with your level of patience |
| 01:40.04 | riddlebox | kroo what distro |
| 01:40.17 | kroo | win32 |
| 01:40.22 | JT | ... |
| 01:40.25 | JT | clown |
| 01:40.26 | flenders | ha |
| 01:40.29 | riddlebox | sorry |
| 01:40.32 | JT | asterisk doesn't run in windows |
| 01:40.44 | kroo | it does |
| 01:40.52 | ManxPower | new trainstatus 2hrs 10 mins late |
| 01:40.58 | ManxPower | kroo: not on this channel it doesn't |
| 01:40.59 | riddlebox | i thought there was a port |
| 01:41.02 | JT | kroo: wrong. |
| 01:41.14 | ManxPower | riddlebox: oh, I'm sure some lunatic did a port, but we don't support the port here |
| 01:41.24 | riddlebox | :) |
| 01:41.24 | JT | kroo: any attempt to run it under windows is a complete hack and is a miracle if it works at all |
| 01:41.38 | JT | i think the "port" involved a cygwin hack |
| 01:41.45 | ManxPower | JT: I've always suspected a pact with satan in situations like that |
| 01:41.51 | JT | heh |
| 01:41.56 | kroo | I wanna give a try I haven't got much choice |
| 01:42.13 | ManxPower | kroo: then I guess we don't have much choice but to /ignore you |
| 01:42.14 | JT | kroo: you always have choice, get an old desktop and put linux on it |
| 01:42.38 | ManxPower | Or pay Digium for consulting |
| 01:42.42 | flenders | kroo: try running on vmware |
| 01:42.48 | ManxPower | Be sure to ask for "russel" |
| 01:42.51 | flenders | I did it first time I installed asterisk |
| 01:43.32 | kroo | I'd like to do that but I'll do that remotely for someone and he's in France and I m not => no choice oterwise I will isnstall linux |
| 01:43.42 | JT | (yes, the document actually exists) |
| 01:43.45 | ManxPower | kroo: Please leave. |
| 01:44.13 | Strom_M | if you leave |
| 01:44.16 | Strom_M | please leave now |
| 01:44.21 | Strom_M | please don't take my spot away |
| 01:44.40 | Strom_M | </orchestral manouvres in the car park> |
| 01:44.46 | kroo | anyway, just could you tell me if I have to configure the zapata.conf file if i haven't got a zaptel hardware, waht does this file pls ? |
| 01:44.55 | Strom_M | kroo: read thebook |
| 01:44.55 | JT | nothing |
| 01:44.56 | Strom_M | ~thebook |
| 01:44.57 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:44.58 | ManxPower | Asterisk runs on Windows about as well as Bill Gates runs on water. |
| 01:45.01 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:45.11 | Strom_M | ManxPower: classic |
| 01:45.28 | ManxPower | kroo: you do not need to configure the config file if you don't have the hardware -- that goes for all of Asterisk |
| 01:45.45 | kroo | thanks |
| 01:45.46 | ManxPower | Strom_M: it is? I thought I just made it up. I must have read it somewhere. |
| 01:46.41 | Strom_M | no, i mean that's an instant classic |
| 01:46.55 | ManxPower | Oh! Cool! |
| 01:47.06 | ManxPower | Strom_M: see the other channel |
| 01:47.18 | riddlebox | lol |
| 01:47.41 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 01:49.59 | *** join/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 01:52.40 | *** join/#asterisk kimosabe (n=kimosabe@189.175.37.162) |
| 01:52.43 | ManxPower | Can't we just have a blanket policy of "you ask about Asterisk on Windows, you get banned for 1 week" |
| 01:53.03 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 01:53.18 | [TK]D-Fender | ManxPower, no, we need our gladatorial newb-crushing to help raise our spirits! |
| 01:53.56 | kimosabe | can some one lead me in the direction of a how to for x100p config via sip device |
| 01:54.18 | [TK]D-Fender | kimosabe, thsoe 2 things configs have nothing to do with each other |
| 01:54.43 | kimosabe | tk i want to be able to take pstn via a sip device |
| 01:54.59 | [TK]D-Fender | take?! |
| 01:55.06 | zapa | thanks for all |
| 01:55.59 | kimosabe | if i pick up the phone that is conected to the sipura for it to automatically be accesing the dial tone from my pstn |
| 01:56.45 | shido6 | thats what the sipura is for. |
| 01:57.37 | kimosabe | shido do you have a sample config of this sir |
| 01:58.25 | [TK]D-Fender | kimosabe, You'll have to read your SPA's guide for auto-dial (bat-phone capability), and from there jsut do "exten => whateveritdialed,1,Dial(Zap/1)" |
| 01:58.56 | shido6 | do u have an ast box at home? |
| 01:59.17 | kimosabe | yes |
| 01:59.43 | shido6 | how many digits does your local telco expect to complete a call? |
| 02:00.02 | kimosabe | 7 |
| 02:00.13 | kimosabe | but more if i make international and stuff |
| 02:00.38 | shido6 | how many Zap channels do you have at the house there? |
| 02:00.48 | shido6 | or wherever the ast box is... |
| 02:01.00 | kimosabe | i have to |
| 02:01.21 | kimosabe | the chanels will either come from a sipura 3000 |
| 02:01.24 | kimosabe | or x100p |
| 02:01.52 | shido6 | does anyone call into your x100p , ever? |
| 02:02.04 | kimosabe | yes that will be necesary |
| 02:02.29 | shido6 | is the sipura 3000 setup at all to place or receive calls? |
| 02:02.55 | kimosabe | yes it is |
| 02:03.02 | shido6 | what do you call it |
| 02:03.10 | shido6 | what label does it have in sip.conf ? |
| 02:03.30 | kimosabe | ahh ok i havent set up the 3000 via sip.conf yet |
| 02:03.38 | shido6 | :) |
| 02:03.54 | kimosabe | thats the info im looking for i had just tried back to back config and its not stable so know i have a x100p |
| 02:03.56 | shido6 | kimosabe1 a good label for it? |
| 02:04.02 | kimosabe | in ast box |
| 02:04.13 | kimosabe | pstn would be fine |
| 02:04.25 | ManxPower | This is starting to look like it might beat my previous worst BGirmingham NOLA Amtrak Trip |
| 02:05.16 | ManxPower | The worst was Chicago / New Orleans, which was 11 hours late |
| 02:05.39 | Innatech | Supposing I want to split a client's VSP service over multiple providers for greater fault tolerance, is a sensible approach to find a provider that will provide hunting once the allocated channels are all taken? And if so, who might such a provider be? (IAX trunking vastly preferred, closer to SoCal the better.) |
| 02:05.59 | Innatech | (talking about incoming calls.) |
| 02:09.02 | JT | ManxPower: you have Internet. It's not that bad. |
| 02:09.03 | ManxPower | That sounds like something that would happen to me. |
| 02:09.38 | ManxPower | JT: I can only blather on about silly things for so long and torturing newbies isn't as fun as it was a few hours ago |
| 02:09.57 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:11.50 | *** join/#asterisk ManxPower (n=manxpowe@015-855-517.area5.spcsdns.net) |
| 02:11.54 | ManxPower | And there has been a significant lack of interesting problems, with the exception of that one. |
| 02:12.01 | ManxPower | apparently I just hit EVDO |
| 02:12.49 | ManxPower | you'd think they could move between RTT and EVDO without totally messing up your connection |
| 02:12.53 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 02:13.18 | ManxPower | and now I'm back on 1xRTT LOL! |
| 02:15.12 | JT | nice |
| 02:17.03 | Innatech | EVDO is for shit, IMHO. Barely runs faster than 1xRTT for me, drops its connection fscking constantly. |
| 02:17.40 | Innatech | Then there's VZW's assholic tethering policies and crippled BT. |
| 02:22.28 | javb | anybody u made asterisk load by default in ubuntu server? (asterisk installed from source code) |
| 02:24.36 | ManxPower | My experience is EVDO is good |
| 02:25.17 | ManxPower | I frequently get 150 Kbps on EVDO (same as a T-1) |
| 02:25.37 | ManxPower | more commonly is around 90k |
| 02:26.01 | Innatech | Gah. I never see anything like that. |
| 02:26.03 | [TK]D-Fender | ManxPower, I thought EVDO / CDMA was really rare in the USA by comparison to GSM. |
| 02:26.10 | ManxPower | [TK]D-Fender: no |
| 02:26.11 | shido6 | kimosabe, the spoonfeeder 3000 is acting up, give it a few more seconds to churn out an example |
| 02:26.18 | Innatech | It's only VZW & Sprint. But the coverage is decent. |
| 02:26.40 | ManxPower | And osme regionals like Alltell |
| 02:26.42 | shido6 | kimosabe, http://pastebin.ca/632695 |
| 02:26.58 | [TK]D-Fender | shido6, hes GONE :) |
| 02:27.00 | Innatech | Plus, I can't stand listening to GSM's RFI in consumer electronics--like my car stereo. Seriously annoying. |
| 02:28.00 | Innatech | BTW, I had to break down and enter NTP offsets into those 501's web GUIs. They assiduously ignored everyway I could have possibly provisioned it. >shrug< |
| 02:28.32 | ManxPower | I'm currently tracking the train in realtime with a PC GPS device and google earth |
| 02:30.22 | [Outcast] | Innatech: I have had the same problem with 501's before |
| 02:30.57 | Innatech | Yeah. I got tired of wasting time futzing around with them. |
| 02:31.07 | *** join/#asterisk ManxPower (n=manxpowe@032-385-265.area5.spcsdns.net) |
| 02:32.27 | [TK]D-Fender | Innatech, web GUI? ICK!!!!!!! |
| 02:32.30 | *** join/#asterisk hoowa (n=chatzill@210.83.203.100) |
| 02:32.57 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 02:33.31 | Innatech | I know, I know. |
| 02:33.49 | Innatech | After three hours resetting and reprovisioning I didn't care any more. |
| 02:33.56 | *** join/#asterisk ManxPower (n=manxpowe@015-793-286.area5.spcsdns.net) |
| 02:34.08 | tzanger | I've never had issue with 501/601 ocnfig |
| 02:34.21 | tzanger | I think the trick is to have a base set of working files and fiddle from there |
| 02:34.26 | Innatech | I did that. |
| 02:34.48 | Innatech | In fact, one of the 430s was working fine with the same files, two days ago and one was misbehaving. |
| 02:35.04 | Innatech | *501s. |
| 02:35.06 | jgoddess | polycom phones? |
| 02:35.09 | Innatech | Yep. |
| 02:35.13 | jgoddess | what is the problem? |
| 02:35.25 | Innatech | They won't accept NTP offset from the provisioning files. |
| 02:35.31 | jgoddess | sorry not that I could help but mine tend to go bonkers sometimes |
| 02:35.32 | Innatech | At least, ONE of them won;t. |
| 02:35.46 | jgoddess | do you actually rename the config files |
| 02:35.48 | jgoddess | reset to factory |
| 02:35.51 | Innatech | So I formatted them, fed them the same configs--and one worked and the other didn't. |
| 02:35.56 | jgoddess | and then force it to relad the files again? |
| 02:35.57 | Innatech | I *formatted* them. |
| 02:36.00 | jgoddess | right.. |
| 02:36.03 | Innatech | Whatever, it's done. |
| 02:36.20 | jgoddess | typically anytime I've seen that problem is a hickup in the config it saves to itself |
| 02:36.27 | jgoddess | and if you use a server that has the dchp lease on it |
| 02:36.37 | *** join/#asterisk jkimball4 (n=jerrid@pc006629.mbsc.unomaha.edu) |
| 02:36.41 | jgoddess | make sure there isn't something in the config that could overright that |
| 02:36.43 | jgoddess | like on our server |
| 02:36.50 | jgoddess | we actually set the ntp from there |
| 02:36.55 | Innatech | Yeah. I did all that for hours on end. They're static. I've read all the docs, I see the cfg files when I close my eyes. Really, it's not worth talking about anymore. |
| 02:37.09 | jgoddess | also one thing is different phones might have been sent with different firmwares |
| 02:37.12 | jkimball4 | When calling QueueAdd on AMI, is there anything that can prevent QueueMemberAdded from being being raised? |
| 02:37.18 | Innatech | nope, pushed from the * tftp. |
| 02:37.27 | jgoddess | so if that is the case then you might want to make sure your sip.ltd and sipconfigs are different for it |
| 02:37.42 | jgoddess | hehe |
| 02:37.44 | jgoddess | Innatech ok |
| 02:37.48 | jgoddess | I'm actually going to bed |
| 02:37.55 | Innatech | g'nite. ;P |
| 02:37.56 | jgoddess | maybe a good kick down the street would help |
| 02:37.58 | jgoddess | ;) |
| 02:38.12 | Strom_M | no |
| 02:38.16 | Strom_M | fedex them to me |
| 02:38.22 | Strom_M | i'll gladly take them |
| 02:38.25 | jgoddess | maybe treathen it with fire |
| 02:38.28 | jgoddess | hehe =P |
| 02:38.38 | jgoddess | you better load or the other ones get it |
| 02:38.38 | Strom_M | you lose for horribly misspelling "threaten" |
| 02:38.45 | jgoddess | no |
| 02:38.47 | jgoddess | I never loose |
| 02:38.50 | Strom_M | lose |
| 02:38.51 | Strom_M | not loose |
| 02:38.53 | Strom_M | dorkus |
| 02:38.54 | jgoddess | I didn't say I was winning on the basis of spellilng |
| 02:38.59 | jgoddess | assmus? |
| 02:39.01 | *** part/#asterisk jkimball4 (n=jerrid@pc006629.mbsc.unomaha.edu) |
| 02:39.08 | jgoddess | its the interweb |
| 02:39.10 | jgoddess | who cares |
| 02:39.18 | jgoddess | not a english paper |
| 02:39.34 | Strom_M | lawl rite oh kay sew yarstirdey i wras dreybng downr tah stright rigta? |
| 02:39.45 | jgoddess | heh |
| 02:39.52 | jgoddess | typically on efnet that would make sense |
| 02:39.54 | jgoddess | =P |
| 02:40.15 | Strom_M | it may be the interweb, but since you've only got the text with which to communicate, it helps if you don't royally butcher it to death |
| 02:41.12 | jgoddess | I really didn't actually it would be normal dyslexic behavior since it was a switch of the t and th |
| 02:41.17 | jgoddess | alrighty then |
| 02:42.05 | Strom_M | hence why you can use jbot for regexps! |
| 02:42.07 | Strom_M | like so: |
| 02:42.09 | jgoddess | your from canada makes sense |
| 02:42.10 | Strom_M | s/so/boners/ |
| 02:42.16 | jgoddess | no really |
| 02:42.16 | Strom_M | no, i'm from los angeles |
| 02:42.23 | Strom_M | i'm only in canada for the week |
| 02:42.27 | jgoddess | what wouldn't be a programming replacement usage now would it |
| 02:42.37 | jgoddess | anyways |
| 02:42.45 | jgoddess | enjoy yourself night |
| 02:45.14 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 02:45.21 | *** join/#asterisk Shoeb (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 02:45.50 | RyanW | what do i need to change in zapata.conf to set the variable AST_BRIDGE_IGNORE_SIGS in rtp.c ? |
| 02:46.00 | Shoeb | Hello. What is the maximum number of simultaneous SIP calls allowed by Asterisk? |
| 02:46.33 | Strom_M | Shoeb: depends on your system |
| 02:47.01 | Shoeb | Strom_M: What's the best it can get, and what if we're on 711 ULAW? (Let's imagine we have the best server) |
| 02:47.42 | Strom_M | define "the best server" |
| 02:48.14 | Shoeb | Strom_M: Whatever the asterisk community deems as the best, heh. I dunno, like duocore 3.2ghz with 2gig ram or something like that? |
| 02:48.28 | Strom_M | i dunno. 700 calls? 1000 calls? |
| 02:48.46 | JT | Shoeb: get a xeon if you want a decent server |
| 02:49.03 | JT | 700 calls on one asterisk instance, i doubt it |
| 02:49.46 | Strom_M | Shoeb: for practical purposes, i'd say engineer your network such that no single box handles more than 200 calls at a time |
| 02:50.12 | Shoeb | JT: Done. Now how many can I push through, max? If I'm planning to deploy a dialer arch. that could dial 2000 simultaneous, I'm trying to see how many boxes would be good. |
| 02:50.33 | Strom_M | Shoeb: if you're planning on implementing an automated dialer, go die in a fire plzkthx |
| 02:50.37 | Shoeb | Strom_M: That looks like a reasonable enough answer. And that's on the powerful system? |
| 02:50.41 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:50.46 | Shoeb | lol |
| 02:51.26 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 02:51.38 | Shoeb | Well, it'll be scaleable upwards to atleast 5000 channels in due course (If it all works without crumbling to pieces.) |
| 02:53.02 | JT | 5000 |
| 02:53.11 | JT | what connectivity do you have to make 5000 calls/ |
| 02:53.23 | Penggu | guys, how would i jump to another context without specifying the extension - just let it fall through till it finds an extension based on what the person dialled? |
| 02:53.29 | javb | how do i set the defuault verbose level in asterisk? |
| 02:53.54 | Penggu | (eg person dials 0983407850, authenticate catches it, auth = ok, now follow through the extensions in another context) |
| 02:56.05 | Shoeb | JT: SIP. |
| 02:57.13 | Strom_M | Shoeb: that's like....400 megabits of bandwidth MINIMUM |
| 02:58.02 | *** join/#asterisk nath0099 (i=James@77-96-249-156.cable.ubr02.maid.blueyonder.co.uk) |
| 02:58.17 | Shoeb | Strom_M: I know. Bandwidth, and Hardware are not an issue. |
| 02:58.30 | Strom_M | and money is no object |
| 02:58.36 | Strom_M | and i'm the next vonage |
| 02:58.53 | Shoeb | When it comes to paying the people who would help me (us) set it up, yes. Money is no object. |
| 02:58.57 | Strom_M | i've heard it sixty thousand times :) |
| 02:59.09 | Shoeb | Don't wanna be no Vonage. :) |
| 02:59.19 | blitzrage | I wanna be the next Vonage! |
| 02:59.21 | Shoeb | I'm sure you have, bud. |
| 02:59.28 | blitzrage | can someone help me set it up? |
| 02:59.41 | JT | Shoeb: please don't call people bud, it's very patronising |
| 02:59.43 | ltdwk | i wanna be a fireman. |
| 02:59.55 | Sedorox | I wanna be a crackhead!... oh.. wait... |
| 02:59.58 | Shoeb | Anyway, moving on. So, Strom_M, you think 200 channels per box? |
| 03:00.12 | Shoeb | JT: My humble apologies. |
| 03:00.23 | ltdwk | JT: Not as patronising as champ, though |
| 03:00.29 | Shoeb | Or chief. |
| 03:00.38 | JT | ltdwk: depends what country your from i guess ;) |
| 03:00.54 | blitzrage | ~sipp |
| 03:00.54 | jbot | i guess sipp is a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/ |
| 03:01.18 | JT | Shoeb: what ITSP will handle 5000 simultaneous calls from you? |
| 03:01.29 | JT | Shoeb: do you have 5000 agents? |
| 03:01.31 | blitzrage | jbot_: sipp is a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/. If you really want to know how many channels your Asterisk box can do, learn how to utilize this program. |
| 03:01.34 | ltdwk | maybe he is a telco. |
| 03:01.40 | JT | ltdwk: doubtful |
| 03:02.22 | Shoeb | JT: 800. And we've invested in an ITPS that can, and does currently. |
| 03:02.25 | blitzrage | ~sipp |
| 03:02.25 | jbot | somebody said sipp was a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/. If you really want to know how many channels your Asterisk box can do, learn how to utilize this program. |
| 03:02.28 | blitzrage | there we go. |
| 03:02.29 | blitzrage | :) |
| 03:02.42 | Shoeb | blitzrage: You rock! |
| 03:02.49 | JT | Shoeb: i see |
| 03:03.28 | JT | 800 agents at the one location? |
| 03:03.49 | Shoeb | Yup. Might get maybe 400 more. |
| 03:04.21 | Penggu | is there a cross between DISA and GoTo where you send the channel to an context with a number they already dialled? (no dial-tone) |
| 03:04.24 | JT | getting into the "hard to believe" territory there |
| 03:05.08 | Shoeb | That |
| 03:05.11 | Shoeb | is ok. I'm not trying to prove anything here to anyone. |
| 03:05.16 | Shoeb | Just needed some answers. :) |
| 03:05.30 | JT | what sort of company has 800 person callcentres? |
| 03:05.55 | JT | not sure how that equates to 5000 calls either |
| 03:06.21 | Shoeb | Like the first one, given the best configuration of a server system, how many simultaneous calls can each Asterisk server handle. |
| 03:06.40 | RyanW | my telco is sending me a FRAME_CONTROL (5)-> Busy and sometimes FRAME_CONTROL(8)->Congestion mid conversation, and asterisk is terminating the audio bridge and hanging up the call. |
| 03:06.43 | JT | probably not much over 200, but you'll have to benchmark |
| 03:06.50 | ltdwk | the "Best" configuration might be like, a quad processor quad core box. |
| 03:06.56 | ltdwk | that would handle a few calls |
| 03:07.11 | RyanW | Is there some was i can filter out these control frames with an option in zaptel.conf perhaps ? |
| 03:07.11 | JT | ltdwk: where are these quad processor quad core motherboards? |
| 03:07.13 | Shoeb | JT: 5000 simulatenous calls don't necessarily give calls to 5000 agents. It can even be used for purposes such as Voice Blasting. Come on, I'm sure you know that. |
| 03:07.19 | ltdwk | but your bus bandwidth will limit you |
| 03:07.21 | JT | ltdwk: not with asterisk. |
| 03:07.33 | JT | ltdwk: asterisk has seriously problems with much over 200-300 calls |
| 03:07.38 | ltdwk | JT: when barcelona comes out next month |
| 03:07.40 | JT | unless he is doing virtualisation |
| 03:07.45 | Shoeb | ah |
| 03:07.46 | ltdwk | JT: they fit in standard socket F |
| 03:07.47 | JT | s/seriously/serious/ |
| 03:07.57 | Shoeb | So you'd think 200 calls will be good, and 300 would be pushing it? |
| 03:08.13 | *** join/#asterisk marc7 (n=marc@24.86.254.94) |
| 03:08.18 | JT | Shoeb: voice blasting, what is that, emergency broadcasting? |
| 03:08.25 | JT | Shoeb: probably |
| 03:08.37 | JT | ltdwk: socket f? |
| 03:08.45 | Shoeb | Voice Broadcasting. And yes, it could be for Emergency broadcasting. :) |
| 03:08.54 | Shoeb | Hmm, gotcha. |
| 03:09.00 | marc7 | brief question... I have asterisk Recording to G.711 .ulaw files... is there any way I can easily edit those in a waveform editor like audacity and re-encode them? |
| 03:09.09 | JT | Shoeb: what else is voice broadcasting used for? |
| 03:09.17 | ltdwk | JT: http://en.wikipedia.org/wiki/Socket_F |
| 03:09.21 | JT | marc7: sox |
| 03:09.25 | Shoeb | 200 calls. Does that lean quite a lot on the configuration of the system? Like will the duocore push 200, or will a regular server push 200? |
| 03:09.47 | JT | what is "duocore"? |
| 03:09.52 | Shoeb | JT: Who knows. I once got a voice broadcast asking me to call this number if I wanted my dick enlarged, lol. |
| 03:10.03 | JT | Shoeb: that sounds illegal/immoral |
| 03:10.10 | Shoeb | Dual Core Processor. |
| 03:10.22 | JT | Shoeb: you can get quad core xeons now |
| 03:10.29 | Shoeb | And yes, I know it sounds illegal/immoral, but I'm giving you an idea of what people use Voice broadcasting for, lol |
| 03:10.45 | Shoeb | I wish I could go and laugh at the guy behind it all, but oh well. |
| 03:10.50 | JT | ltdwk: oh, AMD |
| 03:11.08 | marc7 | JT: thanks |
| 03:11.11 | JT | ltdwk: no wonder it didn't ring any bells |
| 03:11.20 | ltdwk | JT: http://www.tyan.com/product_board_detail.aspx?pid=466 |
| 03:11.46 | Penggu | another question: can i turn autofallthrough=no for a particular context? if yes, can i: _X,1,authenticate(a user), and then set include => another_exten, and expect whatever the user dialled before the Auth to follow on through? |
| 03:12.00 | JT | ltdwk: 1207 pins, that's nutty |
| 03:12.09 | JT | ltdwk: is that mobo for a cpu that doesn't exist yet? |
| 03:12.24 | Shoeb | Alright JT. Thanks a ton for your help. Much appreciated. Same to you, Strom_M. |
| 03:12.41 | JT | Shoeb: you'll probably need a sip proxy too |
| 03:12.50 | Shoeb | Have a fantabulous night! |
| 03:12.54 | Shoeb | JT: Got it. :) |
| 03:13.03 | Shoeb | Err, I think like 29 of them. |
| 03:13.19 | Shoeb | Anyway, good night, good sir. |
| 03:13.19 | JT | also, for broadcasting, you'll cause serious system load spawning all calls in a system at once |
| 03:13.26 | Shoeb | ... |
| 03:13.29 | Shoeb | Now we're talking. |
| 03:13.37 | ltdwk | JT: No, it has CPU's but only dual core. AMD doesn't release their quad core (barcelona) until next month, but it fits in the same socket and will work fine with a bios update. |
| 03:13.53 | Shoeb | What do you mean system load will get affected? You mean the .call files stuff? |
| 03:13.53 | JT | ltdwk: ah ok |
| 03:14.04 | JT | ltdwk: are they quad core per die? |
| 03:14.10 | JT | Shoeb: .call or AMI originate |
| 03:14.20 | ltdwk | JT: yep.... monolithic quad core, not two dies stuck together |
| 03:14.31 | JT | well apparently, making 200 calls at the same time causes high load ;) |
| 03:14.58 | ltdwk | JT: Which no doubt means they will be very expensive due to the low yield |
| 03:14.58 | Shoeb | I'm sure. |
| 03:15.03 | Shoeb | Hmm. |
| 03:15.10 | JT | ltdwk: i will be curious to see how they compare to xeons |
| 03:15.14 | Shoeb | So what would be an easier approach, JT? |
| 03:15.17 | JT | ltdwk: well that sucks |
| 03:15.25 | JT | Shoeb: staggering the calls |
| 03:15.33 | Shoeb | JT: Into bursts? |
| 03:15.43 | JT | or a flow |
| 03:16.03 | Shoeb | Can you please tell me how a flow would be? |
| 03:16.39 | JT | it wouldn't be bursty |
| 03:17.13 | Shoeb | But then it would still cause CPU load, unless this flow mechanism is something I'm not able to understand. |
| 03:17.38 | JT | the load would be more steady insteak of causing spikes |
| 03:18.13 | Shoeb | And how can we achieve this steadiness? |
| 03:18.30 | Shoeb | By dialing a set number of calls every minute, or something? Or how exactly? |
| 03:18.33 | JT | by making calls in a stream |
| 03:18.41 | JT | well yes |
| 03:23.30 | javb | is there an special option for letting user change passwords of their own voicemails ? |
| 03:25.48 | [TK]D-Fender | javb, Yes, listen to the INSTRUCTIONS. |
| 03:26.56 | javb | [TK]D-Fender: im listening, but when selection THE INTRUCTION, it will say "record your personal greeting" ... so there is something wrong. |
| 03:27.25 | javb | if i select the right option for changing pass, it wont take me there.. |
| 03:29.11 | [TK]D-Fender | javb, then your prompts are out of date |
| 03:29.37 | javb | [TK]D-Fender: what do you mean? |
| 03:29.42 | javb | can u elaborate? |
| 03:30.12 | [TK]D-Fender | javb, What part of that statement isn't clear? The worded instructions CHANGED when Voicemail had some features added and I'm guessing your system has some OLD instruction sound files. |
| 03:33.49 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:35.08 | javb | [TK]D-Fender: it is not just the sounds (instructions) because, it start recording and all.. |
| 03:35.26 | javb | i change to the defaults, and still have the same error |
| 03:35.28 | javb | any idea? |
| 03:35.51 | [TK]D-Fender | javb, try option 5 |
| 03:36.26 | javb | nothing |
| 03:37.18 | javb | perfect. |
| 03:37.24 | javb | its on 5.. |
| 03:37.36 | javb | any ideas where can i find update intructions? |
| 03:37.51 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 03:42.42 | JT | pesky 4minute pri outages |
| 03:42.57 | JT | i rushed to the mdf but there was no-one fiddling with cables :/ |
| 03:43.20 | *** join/#asterisk Shoeb (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 03:43.46 | Shoeb | JT: My box crashed. Sorry for last time. One last question I had was, which one... asterisk 1.2, or 1.4, is better? |
| 03:44.07 | JT | there's some contention there |
| 03:44.10 | JT | 1.2 is more tested |
| 03:44.15 | JT | but 1.4 has some new features |
| 03:44.47 | Shoeb | What are some real good features of 1.4, do you know? |
| 03:46.04 | d3wayne | Shoeb: 1.2 moves to security fixes only in a little over a week |
| 03:47.08 | Shoeb | d3wayne: Aaah. But since 1.2 is heavily tested and widely used, vulns are not as public on 1.4, correcto? |
| 03:48.15 | d3wayne | vulns are not as public on 1.4 ? |
| 03:49.08 | Shoeb | Meaning, since 1.2 is famous.. more people look for vulnerabilities on there. Since 1.4 is not as widely used, the focus is just not there.. hence not so much importance given to security fixes on 1.4. |
| 03:49.30 | Shoeb | But other than secfixes, what are the better features of 1.4 that aren't in 1.2? |
| 03:50.31 | d3wayne | if you haven't read UPGRADE.txt, then it will tell you a lot |
| 03:51.19 | file | d3wayne: ! |
| 03:51.21 | Penggu | is there a way to detect that all the Zap lines are congested? |
| 03:51.27 | d3wayne | mr. file :-) |
| 03:51.30 | Shoeb | That's what I was looking for, thanks d3wayne! :) |
| 03:51.43 | Shoeb | Good night, JT and d3wayne. |
| 03:51.55 | file | d3wayne: I went and saw the movie Hairspray |
| 03:52.16 | file | it t'was good! |
| 03:52.20 | d3wayne | was Spider Pig in it ? |
| 03:52.21 | *** join/#asterisk bmg505 (n=leon@196.209.177.217) |
| 03:52.28 | file | no :( |
| 03:52.46 | d3wayne | I don't know if I saw a Hairspray trailer yet...let me google it |
| 03:52.50 | JackEStorm | d3wayne: that would be Harry Porker |
| 03:53.47 | d3wayne | oh yeah, I saw something about this |
| 03:54.37 | JackEStorm | spider pig spider pig does whatever a spider pig does look outtttt for the spider pig... |
| 03:54.42 | *** join/#asterisk mgplc (n=acchung@www.chaos-creations.com) |
| 03:54.45 | JackEStorm | I need to make that a ring tone... |
| 03:58.38 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 03:59.06 | *** join/#asterisk Corydon76-home (i=blue@pdpc/supporter/sustaining/Corydon76-home) |
| 03:59.06 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 03:59.11 | mgplc | Hello everyone. I'm new to IRC so please pardon any etiquette issues. I'm wondering if anyone here has installed 2 different types of Digium cards in the same box. I have 2 TDM400s and 1 TDP2400 installed. |
| 03:59.38 | mgplc | I can only get Asterisk to work if I load either the TDM400 drivers or the TDP2400 drivers but not both. |
| 04:00.10 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 04:00.24 | JT | not sure specifically, but usually no more than 2 digium cards are recommended in the one PC |
| 04:00.52 | flenders | did you try one TDM400 and the 2400? |
| 04:01.14 | mgplc | flender - I haven't yet. |
| 04:01.44 | mgplc | I'll try to give that a try. |
| 04:01.53 | JackEStorm | the TDP2400 wasn't first on the PCI bug was it? |
| 04:02.10 | JackEStorm | (and by "not work" you mean ZTCFG fails?) |
| 04:02.28 | JackEStorm | s/bug/buss/ |
| 04:02.34 | mgplc | JackEStorm - Asterisk doesn't load and it gives an error about not being able to open any of the zap channels. |
| 04:02.54 | Maliuta | mgplc: are the zaptel kernel mods loaded? |
| 04:02.58 | mgplc | I see the channels appear under /dev/zap/ |
| 04:03.08 | Maliuta | that is normally the cause of chan_zap not loading |
| 04:03.36 | Maliuta | mgplc: what is the [relevant] output of an lsmod? |
| 04:03.38 | JackEStorm | well forget about asterisk for now, does the zaptel driver load? and does ztcfg run fine? |
| 04:03.55 | mgplc | The zaptel drivers load fine. Let me dump an lsmod with only the TDM drivers. |
| 04:04.10 | mgplc | Module Size Used by |
| 04:04.10 | mgplc | wctdm 39232 8 |
| 04:04.10 | mgplc | zaptel 196388 19 wctdm |
| 04:04.10 | mgplc | crc_ccitt 6144 1 zaptel |
| 04:04.10 | mgplc | nvram 13832 0 |
| 04:04.11 | mgplc | edd 14620 0 |
| 04:04.13 | mgplc | speedstep_lib 8452 0 |
| 04:04.17 | mgplc | freq_table 8576 0 |
| 04:04.19 | mgplc | thermal 21896 0 |
| 04:04.20 | JackEStorm | mgplc: flooding is bad btw |
| 04:04.21 | mgplc | processor 30400 1 thermal |
| 04:04.23 | mgplc | fan 9348 0 |
| 04:04.25 | mgplc | button 12432 0 |
| 04:04.27 | mgplc | battery 15364 0 |
| 04:04.29 | mgplc | ac 10372 0 |
| 04:04.31 | mgplc | snd_pcm_oss 66728 0 |
| 04:04.31 | JackEStorm | and the bot is dead |
| 04:04.33 | mgplc | snd_mixer_oss 25216 1 snd_pcm_oss |
| 04:04.35 | mgplc | snd_intel8x0 37028 0 |
| 04:04.37 | Maliuta | we don't need all the modules |
| 04:04.37 | mgplc | snd_ac97_codec 76640 1 snd_intel8x0 |
| 04:04.39 | mgplc | snd_pcm 113284 3 snd_pcm_oss,snd_intel8x0,snd_ac97_codec |
| 04:04.40 | JT | mgplc: stop that |
| 04:04.41 | mgplc | snd_timer 31620 1 snd_pcm |
| 04:04.43 | Maliuta | just the relevant ones |
| 04:04.43 | mgplc | snd 70532 6 snd_pcm_oss,snd_mixer_oss,snd_intel8x0,snd_ac97_codec,snd_pcm,snd_timer |
| 04:04.47 | mgplc | soundcore 13792 1 snd |
| 04:04.49 | mgplc | snd_page_alloc 14600 2 snd_intel8x0,snd_pcm |
| 04:04.51 | mgplc | ipv6 272256 25 |
| 04:04.53 | mgplc | af_packet 26760 2 |
| 04:04.55 | mgplc | evdev 13184 0 |
| 04:04.55 | JT | mgplc: can you never, EVER do that again? |
| 04:04.57 | mgplc | joydev 13760 0 |
| 04:04.59 | mgplc | sg 42528 0 |
| 04:05.01 | mgplc | st 43164 0 |
| 04:05.03 | mgplc | sd_mod 22144 0 |
| 04:05.05 | mgplc | sr_mod 21156 0 |
| 04:05.07 | mgplc | scsi_mod 121412 4 sg,st,sd_mod,sr_mod |
| 04:05.09 | mgplc | ide_cd 44448 0 |
| 04:05.11 | mgplc | cdrom 42652 2 sr_mod,ide_cd |
| 04:05.13 | mgplc | ehci_hcd 35204 0 |
| 04:05.17 | mgplc | ohci_hcd 25604 0 |
| 04:05.19 | mgplc | sis_agp 12164 1 |
| 04:05.21 | mgplc | agpgart 37804 1 sis_agp |
| 04:05.23 | mgplc | subfs 12672 2 |
| 04:05.25 | mgplc | dm_mod 63104 0 |
| 04:05.25 | JackEStorm | Jesus H Frog dude.... |
| 04:05.26 | JT | fucking idiot |
| 04:05.27 | Maliuta | wctdm, zaptel and zttranscode are the relevant ones |
| 04:05.27 | mgplc | usbcore 120164 4 ehci_hcd,ohci_hcd |
| 04:05.29 | mgplc | sis900 24580 0 |
| 04:05.31 | mgplc | raid1 21632 1 |
| 04:05.33 | mgplc | reiserfs 265680 1 |
| 04:05.34 | Strom_M | someone kickban plz |
| 04:05.35 | mgplc | JackEStorm - sorry about that. |
| 04:05.36 | Strom_M | Qwell: |
| 04:05.37 | mgplc | JT - Not a problem. |
| 04:05.38 | Strom_M | Corydon76-home: |
| 04:05.40 | Strom_M | Corydon76-work: |
| 04:05.44 | Strom_M | file: |
| 04:05.48 | Maliuta | someone is running a distro kernel |
| 04:06.00 | tzafrir_laptop | Strom_M, in on line |
| 04:06.13 | Strom_M | yes true |
| 04:06.26 | Strom_M | and with that, i'm going out for Calgary's finest fried chicken |
| 04:06.28 | JT | mgplc: how is sending 50 lines to channel "relevant output"? |
| 04:06.33 | tzafrir_laptop | Maliuta, many are. What specific distro and version you have in mind? |
| 04:07.02 | JackEStorm | Back to wait a said: "well forget about asterisk for now, does the zaptel driver load? and does ztcfg run fine?" |
| 04:07.13 | Maliuta | tzafrir: I don't run a kernel I didn't compile, and anything that is going to be needed from boot I compile in rather than as a module |
| 04:07.27 | Maliuta | tzafrir: so i don't care _which_ distro |
| 04:07.34 | tzafrir_laptop | mgplc, pasting more than, say, 3 lines to the channel is considdered impollite. Don't do that. Use a pastebin such as http://pastebin.ca/ |
| 04:07.53 | JackEStorm | tzafrir: pasting more than one line is |
| 04:07.56 | file | hrm? |
| 04:08.00 | ZX81 | heh how come it didn't boot him? |
| 04:08.04 | JackEStorm | file: scroll up |
| 04:08.09 | ZX81 | ~pastebin |
| 04:08.09 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 04:08.13 | JackEStorm | JBOT is gone :( |
| 04:08.14 | mgplc | tzafrir_laptop, thanks, my apologies. |
| 04:08.21 | ZX81 | JackEStorm: where? |
| 04:08.36 | JackEStorm | ok not oped then |
| 04:08.41 | mgplc | Maliuta: So right now Asterisk and the 2 TDM400 cards work fine. |
| 04:08.43 | JT | Maliuta: just use modiles for zaptel |
| 04:08.52 | JT | don't compile it in |
| 04:08.57 | Maliuta | JT: same here |
| 04:09.47 | JackEStorm | tzafrir: I'm all etch here, even on production systems. |
| 04:09.55 | mgplc | Maliuta: Things don't work though when I do a modprobe of wctdm24xxp |
| 04:09.56 | JT | s/modiles/modules/ |
| 04:09.56 | tzafrir_laptop | JT, even with your own kernel I would recommend to build zaptel as external modules |
| 04:09.56 | Maliuta | mgplc: so the problem is? |
| 04:10.07 | JT | tzafrir_laptop: that's what i just said |
| 04:10.24 | Maliuta | tzafrir_laptop: I use unstable at home and etch at work |
| 04:10.43 | tzafrir_laptop | hmm... some undefined symbol? |
| 04:10.52 | JackEStorm | wiat a sec, when did the zaptel mods get put in to the kernel tree as a patch? |
| 04:10.56 | mgplc | Maliuta: I'm trying to get 16 channels up and running. I have 2 legacy TDM400s that I wanted to keep working since it provides 8 channels. The new TDP2400 is providing the other 8. |
| 04:11.20 | Maliuta | anyone using chan_skinny or chan_sccp with cisco 7940 (or 7960)? |
| 04:11.40 | Maliuta | mgplc: I think the TDP2400 uses a different module |
| 04:12.00 | Maliuta | mgplc: all three cards are in the one box? and show up in an lspci? |
| 04:12.10 | JackEStorm | mgplc: it should work, but for the 3RD and last time, did ztcfg run fine? did it bring up all the chans? or did it fail? |
| 04:12.25 | tzafrir_laptop | Maliuta, tail /var/log/kern.log will hopefully show that error |
| 04:12.31 | JT | mgplc: fxs or fxo? |
| 04:12.54 | Maliuta | and what JackEStorm said |
| 04:13.04 | mgplc | JackEStorm, ztcfg runs fine with only the TDM400 drivers loaded |
| 04:13.30 | mgplc | All the ports are running as FXS |
| 04:13.33 | JackEStorm | and you changed the config to refect pci buss order right? |
| 04:13.46 | Maliuta | tzafrir_laptop: I am not having any hardware issues :) |
| 04:14.23 | tzafrir_laptop | Maliuta, I suspected some software issues. What error you *do* get. |
| 04:14.27 | JackEStorm | zaptel.cfg needs to be defined in module load order, and that is pci buss order |
| 04:14.30 | mgplc | JackEStorm, which config are you referring to? On the bus the 2 TDM400s come first, then the TDP2400 |
| 04:15.01 | JackEStorm | mgplc: |
| 04:15.14 | JackEStorm | mgplc: err, /etc/zaptel.conf |
| 04:15.41 | Maliuta | mgplc: you probably also need to load wctdm24xxp |
| 04:16.09 | JackEStorm | zaptel kernel mods are loaded in pci buss order, you need to have your /etc/zaptel.cfg reflect that, else it will fail. |
| 04:16.53 | mgplc | JackEStorm, in my zaptel.cfg, the only config I have is fxsks=1-8 |
| 04:17.20 | mgplc | Is there a way to specify which channels belong to which card in zaptel.conf? |
| 04:17.56 | JackEStorm | no |
| 04:18.08 | shido6 | ? |
| 04:18.10 | JackEStorm | it's topdown order from the buss |
| 04:18.27 | Maliuta | tzafrir_laptop: none, my install is working fine, I am just curious about the skinny/sccp stuff because I have just got a cisco phone and I would like to be able to do more stuff with the menu (which means using the non-sip firmware on the phone) |
| 04:18.57 | JackEStorm | (and I say it this way, because there are other issues with digital and analog cards in the same system and digital needs to be first) |
| 04:19.24 | mgplc | JackEStorm, so then if I wanted to get 16 fxs ports I would use fxsks=1-16 correct? Even though the 16 ports are spanned across 3 cards? |
| 04:19.44 | JackEStorm | no |
| 04:20.08 | Penggu | if i match an extension: exten => s-.,1,, how do I get what s-. actually was ? |
| 04:20.09 | JackEStorm | 1-8 and then 9-16, your second card uses another module |
| 04:20.15 | Penggu | assuming the person dialled a number to call |
| 04:20.17 | JackEStorm | mgplc:you running 1.4? |
| 04:21.14 | mgplc | JackEStorm: I'm running a very old version of Asterisk v1.0.9.1 |
| 04:21.23 | JackEStorm | UGH |
| 04:21.24 | mgplc | Can't upgrade right now. |
| 04:21.45 | JT | can't or won't |
| 04:22.25 | Maliuta | mgplc: what zaptel version? |
| 04:22.39 | mgplc | Maliuta: zaptel-1.4.2.1 |
| 04:22.40 | JackEStorm | Maliuta: 1.0.x I bet |
| 04:22.51 | JT | mgplc: ... |
| 04:22.52 | JackEStorm | mgplc: back up your configs |
| 04:22.59 | JT | mgplc: why on earth would that work at all? |
| 04:23.07 | JackEStorm | mgplc: put all 3 cards in and run genzaptel |
| 04:23.16 | JT | more importantly |
| 04:23.20 | JT | mgplc: upgrade asterisk |
| 04:23.22 | JackEStorm | if you are really using 1.4 |
| 04:23.27 | JackEStorm | JT: nod |
| 04:23.42 | JT | JackEStorm: he's using asterisk 1.0.9.1 |
| 04:23.52 | JT | zaptel 1.4.2.1 is not the correct version to use |
| 04:24.07 | Maliuta | I agree, I don't think that the newer zaptel stuff with work with that ancient version of asterisk |
| 04:24.31 | JT | they usually don't work between minor releases, let alone 2 major releases |
| 04:24.33 | Maliuta | and if you are putting newer cards in it seems like the prime time to upgrade asterisk to something more modern |
| 04:24.43 | JackEStorm | nod to all |
| 04:24.47 | *** join/#asterisk FreddyPG (n=Freddy@125.164.201.38) |
| 04:24.48 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 04:25.21 | JackEStorm | JT: but if he is really using zap1.4, then genzaptelconf will help him, else, he needs to upgrade, so he can get help. |
| 04:25.45 | JT | JackEStorm: eh |
| 04:25.58 | Maliuta | 1.0 is ancient, I have only ever used 1.2 and up |
| 04:26.02 | JT | ast 1.0 with zap 1.4, genzaptelconf won't help him |
| 04:26.03 | JackEStorm | JT: but, I did get asterisk 1.2 to use zap 1.4 with no issues. |
| 04:26.39 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 04:26.50 | mgplc | Funny thing is that zap 1.4 is working with Asterisk 1.0.9.1 with either: the 2 TDM400s, or the TDP2400 by themselves. |
| 04:27.13 | JT | mgplc: maybe so, but you should try de-crackifying your setup first |
| 04:28.07 | Maliuta | and as has been said fix the zaptel.conf to reflect the hardware installed |
| 04:28.15 | JackEStorm | and looking back, if you want to use a DGM-TDM2xxxx you need to upgrade everything. |
| 04:28.25 | mgplc | Trying to do the genzaptel thing. |
| 04:28.32 | JackEStorm | Maliuta: nod |
| 04:28.44 | Maliuta | if it only knows about the first 8 channels don't bitch when it does't "automagically" find the other 8 |
| 04:28.46 | JackEStorm | then make sure /dev/zap has the right permissions. |
| 04:29.22 | Maliuta | telling zap about the other 8 channels, and loading the appropriate module might help |
| 04:30.11 | *** part/#asterisk FreddyPG (n=Freddy@125.164.201.38) |
| 04:32.10 | JackEStorm | Maliuta: I had some major issues putting a TDM400 and a TE120P in a system. |
| 04:32.12 | tzafrir_laptop | Maliuta, the format of zapata.conf hasn't really changed since 1.0 |
| 04:32.30 | tzafrir_laptop | so genzatelconf of recent zaptel will help you, actually |
| 04:32.50 | JackEStorm | tzafrir_laptop: yeah, span ordering |
| 04:33.40 | JackEStorm | (but I still have an issue where with cold boot vrs warm boot modual loading :( ) |
| 04:33.48 | tzafrir_laptop | zapata.conf, that is. Sorry |
| 04:34.03 | JackEStorm | tzafrir: I want to change everything about zaptel |
| 04:34.26 | JackEStorm | I want to break off digital spans from analog spans |
| 04:36.27 | JackEStorm | tzafrir: zapata isn't that bad, it's like sip.conf, ...but I'd like to see Asterisk define spans as I/O O or I, and then branch the tree from there. |
| 04:36.30 | mgplc | JackEStorm: I just ran the genzaptelconf and it regenerated my zaptel.conf and zapata-channels.conf |
| 04:36.54 | mgplc | I have been able to start Asterisk and I'm trying to see if all the ports are recognized. Looks good so far! |
| 04:37.07 | JackEStorm | mgplc: now look at the new zaptel.conf and see how if differs from the old one you where using |
| 04:37.45 | JackEStorm | mgplc: and 2nd time, verify the permissions of and in /dev/zap that they match asterisk runtime. |
| 04:40.44 | JackEStorm | tzafrir: and then over all, I'd like to see something that makes huntgroup defines cleaner ....you know * more as a SoftSwitch and not a PBX |
| 04:41.22 | *** join/#asterisk nain (n=nain@203.148.77.18) |
| 04:41.40 | [TK]D-Fender | JackEStorm, * is NEITHER |
| 04:41.46 | nain | Hi Everybody |
| 04:43.10 | mgplc | JackEStorm: Checked the permissions, those are good. The zaptel generator used 1 line for every port instead of using a range. |
| 04:43.34 | Maliuta | which is just as valid |
| 04:43.35 | JackEStorm | [TK]D-Fender: true, but I'd like to see more of a softswitch layout and feature set, so you can build a pbx out of it, if thats all you want |
| 04:43.45 | JackEStorm | mgplc: with all 3 cards in? |
| 04:44.09 | JackEStorm | mgplc: and the use/group that * runs as can read /dev/zap/*? |
| 04:44.22 | [TK]D-Fender | JackEStorm, Thats pretty much what we have NOW.... |
| 04:45.00 | nain | I am using Sangoma A200r 2 Port FXO card, I am having problem with callerid detection from zap channel ? while asterisk detect callerid from softphone but not from zap channel ? |
| 04:45.15 | mgplc | JackEStorm: Yes. All 3 cards are in. The first 8 ports are working correctly right now. |
| 04:45.36 | mgplc | I am trying to test the other 8 ports right now. |
| 04:46.14 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 04:46.19 | JackEStorm | [TK]D-Fender: 'cept zaptel does not branch off I/O groups and I groups and O groups...(groups/spans) [or digital interfaces from analog] |
| 04:46.45 | JackEStorm | mgplc: and ztcfg -vvvv showed all 16 chans? |
| 04:46.46 | JT | nain: there is no "detecting" callerid from a softphone, it's there in the sip headers |
| 04:47.19 | rbd | hi guys, I've installed the asterisk packages on ubuntu feisty.... I have some of my own custom prompts I want to use in asterisks, but it seems that I have both a /usr/share/asterisk/sounds and a /var/lib/asterisk/sounds. currently /var/lib/asterisk/sounds is empty, but can I use either or should I stick to /usr/share/asterisk/sounds? |
| 04:47.29 | JT | rbd: "asterisk" |
| 04:47.30 | mgplc | JackEStorm: Yes! ztcfg shows all 16 channels. |
| 04:47.52 | JackEStorm | JT: callerid zone could be off, noisy, being sent on the 2nd ring...etc... |
| 04:48.09 | nain | JT: Hmmmm! so how can i read callerid from zap Channel, I have set variable SET(Callerid=${CALLERID(NUM)}, but Callerid Variable is empty while call arrived from zap channel |
| 04:48.20 | JackEStorm | mgplc: ok, and /dev/zap* can be read by asterisk? |
| 04:48.43 | mgplc | JackEStorm: Yes. /dev/zap/* can be read by asterisk. |
| 04:49.22 | JT | JackEStorm: not sure what that has to do with what i said. |
| 04:49.41 | JackEStorm | mgplc: now setup zapata.conf in /etc/asterisk, if * fails the /var/log/asterisk/messages file will tell you why |
| 04:50.09 | tzafrir_laptop | rbd, Debian/Ubuntu packages set the datadir to /usr/share/asterisk |
| 04:50.38 | JackEStorm | JT: I read him as saying that the zap chan doesn't read CID to pass to the softphone |
| 04:50.43 | tzafrir_laptop | Which makes more sense, IMHO |
| 04:51.09 | JT | nain: what do you currently have in your dialplan to answer an incoming fxo call? |
| 04:51.14 | JT | nain: if >3 lines, use pastebin.ca |
| 04:51.21 | tzafrir_laptop | this can also be set at runtime in asterisk.conf |
| 04:51.46 | *** part/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
| 04:53.44 | Maliuta | nain: is your pstn provider sending callerid? |
| 04:53.59 | Maliuta | nain: I know mine charges extra for that feature, so I do without |
| 04:54.16 | Maliuta | $7/mnth for callerid isn't worth it |
| 04:54.30 | JT | Maliuta: haha telstra |
| 04:54.40 | Maliuta | hellstra |
| 04:54.49 | JT | Maliuta: can't get optus? |
| 04:55.08 | Maliuta | not if I want to stay with 'node for my dsl |
| 04:55.15 | JT | well that sucks |
| 04:55.41 | Maliuta | I encourage people to use my VoIP line anyhow, it's a brisbane number (which is where most people call me from) |
| 04:56.30 | Maliuta | I would look into moving to opt-arse, but I am going to move back to brisbane as soon as I can |
| 04:57.00 | Maliuta | it may be possible in theory to have optus do the re-sell thing on my line |
| 04:57.01 | nain | Maliuta: Zap Channel answer the call from incoming-zap context then i set the exten like this exten => _X.,n,Set(CALLERID=${CALLERID(num)}) to get the caller id but callerid variable or ${CALLERID(num)} variable is empty |
| 04:57.58 | Maliuta | nain: probably meaning that the provider isn't sending the callerid stuff down the line |
| 04:58.03 | nain | Maliuta: if SIP channel answer the call and i use the same exten to read callerid it shows the SIP User callerid in variable but not from zap channel |
| 04:58.45 | Maliuta | nain: SIP is different from zap, the zap stuff requires that the provider transmit the stuff down the line |
| 04:58.47 | nain | Maliuta: I Have discussed with other user with same provider and they are getting callerid |
| 04:59.13 | *** join/#asterisk levi_home (n=levi@levi.dsl.xmission.com) |
| 04:59.24 | Maliuta | nain: are the on the same "plan" as you, like I said my provider charges extra for it |
| 04:59.26 | nain | Maliuta: they told me that there could be some thing wrong with configration |
| 04:59.53 | Maliuta | nain: and have you told chan_zap to use callerid for that line? |
| 04:59.53 | [TK]D-Fender | nain, And the reason you've been whining all this time without SHOWING THEM TO US IS!??!?! |
| 04:59.57 | nain | Maliuta: yes that is in same plan and no extra charges for caller id |
| 05:00.03 | JT | nain: ok, let'ssimplify it. |
| 05:00.24 | [TK]D-Fender | nain, PASTEBIN your configs. |
| 05:00.24 | JT | nain: can you receive calledid when you hook up a callerid capable handset to the line? |
| 05:00.24 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 05:00.24 | JT | instead of asterisk |
| 05:00.24 | JT | callerid |
| 05:00.48 | nain | Maliuta: I am not sure, but i have defined in zapta.conf to usecallerid=yes, callerid=asrecieved ... |
| 05:01.16 | nain | JT: Yes I recieve the callerid with callerid capable handset but not with sangoma A200r fxo card |
| 05:03.17 | [TK]D-Fender | nain, callerid=asrecieved <- bad. callerid=asreceived <- good |
| 05:03.21 | [TK]D-Fender | nain, get your spelling right |
| 05:03.59 | Maliuta | lol |
| 05:04.03 | nain | [TK]D-Fender: in zapta.conf spell are correct, it was here by typo |
| 05:04.10 | [TK]D-Fender | trust-- |
| 05:04.48 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 05:05.02 | [TK]D-Fender | nain, do everyone a favour and PASTEBIN all your configs. |
| 05:05.09 | JT | nain: please use copy and paste. transcribing is just wasting our time. |
| 05:05.19 | JT | nain: we are busy people :) |
| 05:06.46 | nain | JT: I am sorry for that, unfortunatally i am not connected with machine at the moment. |
| 05:06.57 | JT | then what are we supposed to do? |
| 05:07.08 | JT | pray that it will fix itself |
| 05:07.21 | nain | JT: :) |
| 05:07.52 | Maliuta | JT: making random recommendations seems like the right move |
| 05:07.54 | Maliuta | :) |
| 05:08.24 | nain | well is there any thing else that cause the callerid not to work |
| 05:09.28 | JT | nain: yes, failing to wait a sufficient period of time before answering the call. |
| 05:09.52 | JT | using the wrong callerid mode |
| 05:09.56 | JT | a whole number of things |
| 05:10.11 | JT | but without configs and more details, it's a waste of time |
| 05:10.51 | Maliuta | yeah, I seem to remember putting a 2 second delay in for answering calls at some point |
| 05:11.02 | Maliuta | been ages since I tweaked that conf |
| 05:12.23 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 05:15.16 | nain | Malituat, JT: THanks for your valuable information, i will put the config in pastebin as well as soon i will connect to machine and I will try your suggestions as well .... |
| 05:27.45 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 05:31.13 | mgplc | Maliuta: JackEStorm: Thanks for all the help. Rebuilding the zapata.conf file seemed to do the trick. I've tested all 16 lines and they all seem to be functioning fine. |
| 05:32.03 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:32.07 | Keltus | how do I do something like this: call a number, if it doesn't pick up after 3 rings, call another number... do this until one of them picks up |
| 05:32.40 | Keltus | and then if there is a pick up, it bridges that call with an incoming call to asterisk |
| 05:34.26 | Strom_M | Keltus: look at the DIALSTATUS variable |
| 05:34.26 | Strom_M | and put a timeout on the Dial() app |
| 05:34.55 | Keltus | I looked at Dial() but it doesn't return anything to indicate whether or not the call was received |
| 05:36.22 | [TK]D-Fender | Keltus, <Strom_M> Keltus: look at the DIALSTATUS variable <-------------------------------------------- |
| 05:36.27 | Keltus | oh, right |
| 05:36.29 | Keltus | I'm looking |
| 05:36.43 | [TK]D-Fender | Keltus, And kee in mind the "g" option for Dial. |
| 05:36.56 | [TK]D-Fender | Anyways... way late.... heading to bed, later all |
| 05:37.35 | Keltus | ahh that's cool |
| 05:37.38 | Keltus | so you don't have to wait for timeout |
| 05:37.55 | Keltus | I'm doing this in AGI, so I hope I can get the DIALSTATUS variable still |
| 05:38.26 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 05:38.43 | *** join/#asterisk RSAMan (n=a@dsl-242-44-247.telkomadsl.co.za) |
| 05:39.07 | Keltus | would you recommend using the manager API to dial out from AGI, or to use the spool dir? |
| 05:40.01 | Keltus | I can't think of a way to know when the call is completed using the spool method |
| 05:44.00 | Keltus | does it delete the file when it's done maybe? |
| 05:44.50 | JT | what |
| 05:44.58 | JT | manager api does not dial out with agi |
| 05:45.05 | JT | it's the manager interface |
| 05:45.12 | nain | Any one would recommend good application that popup information on based of callerid ? |
| 05:52.45 | *** join/#asterisk tuzhila (n=kvirc@84.47.128.99) |
| 05:55.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 05:56.06 | Keltus | JT: I mean how to I dial out using the manager API when my script is running |
| 05:56.41 | Keltus | errrr I mean is it better to call the manager API or the spool dir |
| 05:58.26 | Keltus | .call files are kind of hard to deal with. I'm not sure how I can connect it to the current context... hmmm |
| 05:59.36 | JT | it's easy, you specify the context |
| 06:00.07 | JT | it's up to you which to use, i think using ami would be better for high volumes |
| 06:06.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:10.17 | Keltus | originally, I was going to have incoming calls go to the default context, and just pass all the variables to an AGI script to do the dirty work |
| 06:10.42 | Keltus | but then I realized the dialing out and bridging the calls is not easy in AGI so I think I will just do everything in the dialplan |
| 06:11.58 | *** join/#asterisk lsodi (n=lsodi@195.80.124.193) |
| 06:17.48 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 06:24.10 | *** join/#asterisk LoneShadow (n=a@c-76-103-55-28.hsd1.ca.comcast.net) |
| 06:32.52 | *** join/#asterisk sergee (i=kvirc@voip1.west-call.com) |
| 06:32.59 | *** join/#asterisk NirS_ (n=Nir@87.68.232.33.adsl.012.net.il) |
| 06:46.20 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 06:58.38 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:07.09 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
| 07:14.25 | *** join/#asterisk SXT40 (n=sxt40@cpe-65-25-148-111.columbus.res.rr.com) |
| 07:15.36 | SXT40 | I'm having a maybe odd problem... does anyone know what commonly causes "503 Service Unavailable" in *? |
| 07:20.15 | *** join/#asterisk CM3_1_2_632 (n=CM3_1_2_@n219076080194.netvigator.com) |
| 07:26.57 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:29.40 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:32.47 | RSAMan | hello |
| 07:32.55 | RSAMan | how can i check if asterisk is running ? |
| 07:33.35 | tuzhila | ps ax|grep asterisk |
| 07:33.55 | tuzhila | etc/init.d/asterisk status |
| 07:34.18 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.2.27.revip2.asianet.co.th) |
| 07:35.17 | HaMYaI | is Corydon76 or juggie awake? |
| 07:35.31 | RSAMan | kk |
| 07:35.32 | RSAMan | ta |
| 07:40.55 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:44.04 | RSAMan | i am on page 90 of that huge guide. they refer to the dialplan |
| 07:44.12 | RSAMan | is this a config file |
| 07:44.12 | RSAMan | ? |
| 07:44.38 | RSAMan | trying to configure a sip server.. |
| 07:44.38 | RSAMan | [internal] |
| 07:44.38 | RSAMan | exten => 100,1,Dial(SIP/john) |
| 07:44.38 | RSAMan | exten => 611,1,Echo( ) |
| 07:44.44 | RSAMan | they want me to add this to dailplan |
| 07:45.40 | JT | you must've missed something in the book |
| 07:45.49 | JT | extensions.conf |
| 07:45.54 | JT | and that's Echo() |
| 07:45.59 | JT | Echo is all you need |
| 07:47.12 | RSAMan | kk |
| 07:47.12 | RSAMan | thanks |
| 07:51.39 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:52.41 | RSAMan | erm if i edit the extensions.conf file should i restart asterisk |
| 07:52.42 | RSAMan | ? |
| 07:53.08 | JT | or type extensions reload |
| 07:53.14 | RSAMan | kk |
| 07:55.26 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 07:56.25 | RSAMan | sweet |
| 07:56.28 | RSAMan | i have a sip server |
| 07:56.29 | RSAMan | :) |
| 07:57.05 | JT | :) |
| 08:03.27 | jarod14 | I'm trying to find a way to match an extension of unknown length but ending with a # . There is my current code in extensions.conf : http://pastebin.fr/125 . I would like to replace the several pattern by only one. but I do not know how to wait for the # DTMF character indefinetely. Any idea ? |
| 08:05.29 | creativx | did you try . |
| 08:06.35 | JT | although inadvisable, _X.# |
| 08:06.35 | jarod14 | yes but it did not match the # character |
| 08:06.47 | jarod14 | tried too |
| 08:07.10 | jarod14 | it did not worked |
| 08:07.14 | JT | perhaps the answer is no then |
| 08:07.22 | JT | # has some default behaviour |
| 08:07.37 | JT | of dialplan fallthrough |
| 08:09.11 | jarod14 | mmm ok |
| 08:09.25 | jarod14 | what do you think about a while loop ? |
| 08:09.43 | JT | sounds hackish |
| 08:10.24 | jarod14 | yep it is ^^ |
| 08:10.59 | creativx | does it have to end with a pound |
| 08:11.17 | jarod14 | yes unfortunately |
| 08:11.36 | jarod14 | it's required |
| 08:12.01 | jarod14 | creativx what are you thinking about ? |
| 08:12.36 | creativx | nevermind |
| 08:16.24 | *** join/#asterisk ZeeRoCOOOL (n=ZeeRoCOO@196.203.146.148) |
| 08:16.36 | ZeeRoCOOOL | morning |
| 08:17.01 | jarod14 | hi ZeeRoCOOOL |
| 08:17.32 | ZeeRoCOOOL | please can any one help me ? I have a little problem with the Dial Command |
| 08:17.40 | jarod14 | creativx, JT thx for your help anyway |
| 08:18.51 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:27.09 | tzafrir_laptop | ZeeRoCOOOL, maybe if you'll actually ask your question |
| 08:31.47 | ZeeRoCOOOL | ok |
| 08:31.51 | ZeeRoCOOOL | thank's |
| 08:32.28 | ZeeRoCOOOL | On my dialplan I need to execute some PHP files with the suitable value of ${DIALSTATUS} |
| 08:32.44 | *** join/#asterisk vgster (n=vgster@91.103.131.90) |
| 08:33.04 | ZeeRoCOOOL | but when the Dial is called |
| 08:33.29 | ZeeRoCOOOL | any extension after it is ignored |
| 08:34.22 | ZeeRoCOOOL | I used the g option but no way |
| 08:34.40 | ZeeRoCOOOL | the M option doesn't work too |
| 08:35.35 | creativx | ZeeRoCOOOL: noop(${DIALSTATUS}) after dial is ignored? |
| 08:36.08 | ZeeRoCOOOL | yes |
| 08:36.28 | ZeeRoCOOOL | any thing after Dial... is ignored |
| 08:48.32 | mvanbaak | only when the call is answered |
| 08:51.09 | tzafrir_laptop | ZeeRoCOOOL, what version of Asterisk is it? |
| 08:52.17 | ZeeRoCOOOL | 1.2 |
| 09:01.04 | *** join/#asterisk lehel (n=sfdsd@86.125.81.206) |
| 09:01.13 | lehel | hello |
| 09:02.17 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-70-107-176-119.ny325.east.verizon.net) |
| 09:02.20 | tzafrir_laptop | ZeeRoCOOOL, in 1.2 you have a priorityjumping setting (I don't remember the exact name) |
| 09:03.12 | RSAMan | hi again |
| 09:03.54 | RSAMan | in the extensions.conf file , is [incoming] a context |
| 09:04.04 | RSAMan | and whats the difference between that and [internal] |
| 09:04.10 | RSAMan | i am readsing the manual btw |
| 09:06.17 | *** join/#asterisk blueneon (n=blueneon@dsl-146-29-190.telkomadsl.co.za) |
| 09:06.22 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-145.net.novis.pt) |
| 09:06.51 | mvanbaak | they are different contexts |
| 09:06.54 | tzafrir | the names are arbitrary |
| 09:06.56 | blueneon | hi, im trying to setup an extension rule that will dial a certain extension but if that extension is busy it will try another extension, how would i do that? |
| 09:08.46 | mvanbaak | blueneon: something like this should work http://pastebin.ca/632902 |
| 09:09.32 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 09:09.41 | blueneon | ta :) |
| 09:10.37 | blueneon | tho doesnt that only check the status after 45 sec? |
| 09:11.17 | blueneon | i would need it to divert to the other extension right away if the first one is already busy |
| 09:12.15 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 09:12.19 | kolian123 | hello |
| 09:12.40 | kolian123 | A question, is iax.conf IP based authentication work, exists? |
| 09:14.19 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 09:14.58 | creativx | blueneon: dial will give up immediately |
| 09:19.56 | *** join/#asterisk ZeRoCoOOL (n=ZeeRoCOO@196.203.146.148) |
| 09:20.36 | kolian123 | i suspect it's really broken |
| 09:23.35 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 09:25.01 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 09:25.50 | ZeRoCoOOL | Can I ask a question please ? |
| 09:26.06 | creativx | dont ask to ask plz |
| 09:26.10 | masus | :) |
| 09:26.19 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 09:27.00 | RSAMan | lol |
| 09:27.02 | RSAMan | yes |
| 09:27.08 | RSAMan | does that answer your question ? |
| 09:27.13 | masus | :D |
| 09:27.46 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 09:31.21 | *** join/#asterisk ZeeRoCOOOL (n=ZeeRoCOO@196.203.146.148) |
| 09:31.31 | ZeeRoCOOOL | so ? |
| 09:31.38 | creativx | no |
| 09:31.39 | creativx | don't ask. |
| 09:31.52 | blueneon | creativx: i just tried that rule, and it seems its waiting the full 45sec before it decides to divert to the other extention |
| 09:32.00 | ZeeRoCOOOL | I think that ur bot hate me lol |
| 09:32.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:33.08 | *** join/#asterisk nain (n=nain@203.148.77.60) |
| 09:33.12 | nain | HI |
| 09:33.55 | creativx | blueneon: ok. i might be remembering wrong |
| 09:34.26 | RSAMan | question, I am setting up a sip server, will most of the configurations be done in the dialplan ? |
| 09:34.37 | RSAMan | sorry i am new to this |
| 09:35.31 | creativx | what do you mean a "sip server" is |
| 09:35.32 | RSAMan | bause if i add 2 users to my sip.conf file , will they be abble to phone one another ? |
| 09:35.44 | RSAMan | without a dialplan |
| 09:35.47 | creativx | no |
| 09:35.51 | creativx | the dialplan is the heart of asterisk |
| 09:35.54 | RSAMan | kk |
| 09:36.00 | masus | hi all, is something like this possible. i have a mysql table with phone numbers . and if i make a call only the numbers in the table will be called ...Thanks |
| 09:36.01 | RSAMan | just getting a feel |
| 09:36.14 | creativx | in the dialplan you map extensions to logic and devices |
| 09:36.19 | tzafrir_laptop | ~ZeeRoCOOOL |
| 09:36.20 | jbot | from memory, zeerocoool is probably someone that should get some help! |
| 09:36.32 | creativx | heheh |
| 09:36.34 | masus | or maybe i ask another way ... -> is it possible to limit the callings to the phone numbers of my database |
| 09:37.03 | masus | i'll pay 100 usd |
| 09:37.05 | RSAMan | so i should say something like " exten => 123,1,Dial(SIP/Jack) |
| 09:37.22 | RSAMan | to map each new user in sip.conf to a number |
| 09:37.32 | masus | for someone who give me assistance ;) |
| 09:37.44 | RSAMan | but then is this necessary " exten => Jack,1,Dial(SIP/Jack) |
| 09:37.47 | RSAMan | ? |
| 09:38.14 | RSAMan | sorry i am confused |
| 09:38.17 | creativx | if you want your users to be able to dial the extension "Jack" then yes |
| 09:38.31 | pj_ | Users don't know Jack anyway |
| 09:38.31 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:101:382a:df3b:1f18:4150) |
| 09:38.36 | creativx | exactly |
| 09:38.36 | creativx | heheh |
| 09:38.37 | RSAMan | true that |
| 09:38.44 | creativx | the first line should suffice |
| 09:38.50 | creativx | punch in 123 in a phone |
| 09:38.58 | creativx | and user jack's device rings |
| 09:39.06 | creativx | that is the idea atelast |
| 09:39.14 | RSAMan | but for the moment all my clients are connecting via soft phone / sip |
| 09:39.56 | RSAMan | so the users dont know about one another until they are in the dialplan |
| 09:40.10 | RSAMan | ? |
| 09:40.48 | RSAMan | i just need to grasp on these concepts |
| 09:40.53 | RSAMan | sorry for sounding dumb |
| 09:46.50 | creativx | yes |
| 09:47.38 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 09:47.54 | creativx | god damn #¤")%)"% ip10s crap phones |
| 09:48.42 | *** join/#asterisk shinao1 (n=shinao1@41.205.184.212) |
| 09:49.18 | masus | again -> is it possible to limit the callings, to the phone numbers that are stored in mysql table? Thanks ... :) |
| 09:50.04 | nain | Hi Every body, |
| 09:51.48 | MrMister2 | Hi. Has anyone used a Siemens C450IP (or in the UK 460IP) wireless phone with Asterisk? |
| 09:52.55 | MrMister2 | I've seen a post on a forum saying that it works but there was no additional information and i wanted to know if the DTMF commands (transfer, pickup, etc...) worked with no problems. |
| 09:54.30 | nain | I am having problem while reading callerid from Zap Channel, |
| 09:54.53 | nain | ${Callerid(num)} shows empty variable |
| 09:57.04 | *** join/#asterisk twans (n=x@2001:5c0:8fff:fffe:0:0:0:6c4f) |
| 09:57.47 | nain | JT: would you please suggest that what's wrong with configration as callerid variable from zap channel is empty... see the [from-zaptel] |
| 09:57.48 | nain | include = default |
| 09:57.48 | nain | exten => s,1,Wait(3) |
| 09:57.48 | nain | exten => s,n,Answer() |
| 09:57.48 | nain | exten => s,n,Goto(callback,2007,1) |
| 09:58.04 | nain | sorry.... paste wrong clipboard |
| 09:58.15 | nain | it's http://pastebin.ca/632934 |
| 09:59.23 | JT | the dialplan in pastebin is different to what you pasted above |
| 09:59.52 | MrMister2 | Forget my question (not that anyone saw it :)), just found the answer on voip-info.org |
| 10:00.50 | nain | pastbin dialplan is actual, this info is for another config not related to this machine... |
| 10:01.31 | nain | JT: pastbin info is copy & paste from config files... |
| 10:01.44 | JT | nain: well the one in pastebin is wrong |
| 10:01.58 | nain | JT: which one... |
| 10:02.18 | JT | nain: how can goto go to another context without being told which context to go to? |
| 10:02.39 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 10:03.11 | *** join/#asterisk guillote_GNU (n=guillote@host111.190-30-66.telecom.net.ar) |
| 10:03.29 | nain | JT: yes, you are right, that is corrected now... actually i was playing with config files so context name was removed... |
| 10:03.44 | JT | show the new config in pastebin |
| 10:03.48 | nain | JT: yes, that is like exten => s,2,Goto(callback,2007,1) |
| 10:04.39 | JT | add Wait(3) to the start |
| 10:05.56 | JT | actually, make it Wait(5) |
| 10:06.15 | creativx | masus: do a sql lookup on the outbound extension dialed? |
| 10:07.02 | masus | is there a documantation |
| 10:07.03 | lsodi | two asterisk installations in one server, is that good idea? |
| 10:07.39 | creativx | masus: yes |
| 10:07.55 | masus | :) |
| 10:08.10 | creativx | masus: many ways to solve that. google the best ways to do sql lookups via the dialplan. either agi or other applications |
| 10:08.22 | nain | JT: I am doing wait(5).. |
| 10:08.25 | masus | creative : thanks |
| 10:08.59 | creativx | you could even CURL() it |
| 10:09.12 | creativx | the possibilities are... endless |
| 10:11.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 10:11.25 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:13.57 | masus | creativx: The Realtime Switch --> is this what i'm looking for |
| 10:14.00 | masus | :P |
| 10:14.15 | lsodi | have any one tryed to install two asterisks into one server? |
| 10:15.24 | JT | nain: put the new dialplan in pastebin.ca |
| 10:15.59 | *** join/#asterisk naain (n=nain@203.148.77.60) |
| 10:16.22 | JT | nain: put the new dialplan in pastebin.ca |
| 10:16.52 | naain | JT: ok hold on.. |
| 10:21.31 | naain | JT: http://pastebin.ca/632964 |
| 10:22.38 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:23.01 | JT | naain: it still doesn't work? |
| 10:23.24 | naain | JT: see the log |
| 10:24.12 | naain | JT: I have put all the variable but all are empty (exten => 2007,n,NoOp(${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, ${CALLERANI}, ${DNID}, ${RDNIS}) |
| 10:24.38 | JT | naain: what asterisk version? |
| 10:24.49 | naain | asterisk 1.4.5 |
| 10:25.09 | JT | well it would help if you don't use outdated variables |
| 10:25.14 | JT | all of them are gone in 1.4 |
| 10:25.21 | JT | use the new CALLERID function |
| 10:26.04 | naain | JT: let me know which one to use with 1.4.5 with Sangoma A200r fxo card |
| 10:26.14 | JT | naain: the card is irrelevant |
| 10:26.32 | JT | the CALLERID function is the same between technologies |
| 10:26.33 | JT | look it up |
| 10:26.54 | naain | JT: ok suggest me which variable to use with 1.4 to get callerid work. |
| 10:27.13 | JT | also get rid of this # |
| 10:27.14 | JT | exten => 2007,n,SET(CALLERID(NUM)=${CALLERIDNUM}) |
| 10:27.18 | creativx | naain: learn to use the CLI |
| 10:27.19 | JT | naain: LOOK it up |
| 10:27.29 | JT | voip-info.org |
| 10:27.34 | JT | or README.variables |
| 10:28.49 | naain | You mean it should be simple like: SET(CALLERID=${CALLERIDNUM}) |
| 10:29.16 | JT | no |
| 10:29.19 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 10:29.21 | JT | i mean DELETE IT |
| 10:29.21 | naain | JT: I try to understand from voip-info and asteriskguru but examples are given like above i am using.. |
| 10:29.27 | JT | that's what "get rid of" means |
| 10:29.37 | naain | JT: got it... |
| 10:30.06 | naain | JT: would you suggest the new variable or correct way. |
| 10:30.25 | JT | naain: you should not be trying to set the variable |
| 10:30.34 | JT | chan_zap will set it if it receives it from the PSTN |
| 10:30.48 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 10:30.49 | JT | so just delete it and stop being obtuse |
| 10:31.36 | naain | JT: after deleting that line where can i see that callerid is being detected ... |
| 10:32.08 | JT | naain: by fixing up the NoOp line using CALLERID functions |
| 10:32.20 | JT | i already said this |
| 10:36.13 | naain | JT: Ok see the pastbin now, is that correct? |
| 10:37.00 | JT | you haven't given me the url, how can i check? |
| 10:37.41 | naain | JT: Oh sorry, http://pastebin.ca/632974 |
| 10:38.55 | *** join/#asterisk klausdarilion (n=klausdar@nat.labs.nic.at) |
| 10:39.51 | JT | naain: now you have no way of seeing if you are getting the right callerid |
| 10:41.17 | klausdarilion | Hi all! I have problems to understand the meaning of ast_frame->offset/data/datalen and what for is AST_FRIENDLY_OFFSET. Is this somewhere documented or can someone give me short description please? |
| 10:41.24 | naain | JT: I will request you to correct the config where i can see the callerid as well, in pastbin.. |
| 10:41.46 | JT | naain: i will request you to do your own research unless you pay me |
| 10:41.54 | JT | i'm not going to make your dialplans for you |
| 10:42.01 | JT | you need to learn how to do it |
| 10:42.17 | naain | JT: in the meanwhile i have uncomment the dialplan but still NoOp Debug shows like this: -- Executing [2007@callback:2] NoOp("Zap/4-1", "| | | | | ") in new stack |
| 10:43.54 | JT | naain: i have told you about five billion times now that those variables are no longer used |
| 10:44.05 | JT | naain: look up the new function CALLERID |
| 10:44.32 | creativx | rofl |
| 10:44.37 | creativx | JT, paddling upstream again eh |
| 10:44.40 | naain | JT: ok let me checkout |
| 10:45.24 | klausdarilion | regarding: AST_FRIENDLY_OFFSET - solved. I just found a description in the mailing list archive |
| 10:46.01 | JT | creativx: :/ |
| 10:49.22 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:50.07 | naain | JT: is it the new usage of function : exten => 0,1,NoOp({CALLERIDNAME} = ${CALLERIDNAME}) |
| 10:50.08 | naain | exten => 0,2,NoOp({CALLERIDNUM} = ${CALLERIDNUM}) |
| 10:50.15 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 10:50.21 | santibiotico | hi |
| 10:50.48 | creativx | rofl |
| 10:50.56 | *** join/#asterisk Daviey (i=daviey@ubuntu/member/daviey) |
| 10:50.56 | creativx | JT: so, how do you do it? drugs? |
| 10:51.31 | creativx | naain: here's a tip that comes with a hefty invoice: 5*CLI> show function CALLERID |
| 10:51.51 | santibiotico | is there any way to detect if an extension is busy or unavailable? |
| 10:52.29 | Daviey | Hi, how many concurrent calls can you have with a single ISDN-BRI line (In the UK)? |
| 10:52.52 | JT | Daviey: 2 |
| 10:53.06 | Daviey | JT: really?! :( |
| 10:53.14 | JT | yes. |
| 10:53.22 | Daviey | How can 20-30 lines be achieved? |
| 10:53.48 | JT | PRI |
| 10:55.30 | JT | creativx: no such luxury :P |
| 10:55.36 | Daviey | ta |
| 10:58.55 | santibiotico | is there also any way to dial a number within a conversation? |
| 10:59.42 | santibiotico | i want to dial i.e. 1 during a a conversation and then process another dial plan statement for example |
| 11:00.06 | santibiotico | i cannot use the waitexten app as i want to do it during the conversation |
| 11:06.27 | creativx | mm lunch |
| 11:07.05 | Hymie | anyone have polycom 330s? |
| 11:07.51 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 11:08.11 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:10.18 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 11:10.45 | Hymie | s0ck: don't listen to them, they're all aliens.. every single one of them.. they are not to be trusted, unless you want to make sure that 12 is equal to 12.121! |
| 11:11.02 | *** part/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 11:11.04 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 11:11.31 | Hymie | s0ck: it's ok, you're now in the correct dimension |
| 11:13.03 | creativx | who needs float precision anyways |
| 11:13.31 | Daviey | Anybody where i can find a list of UK ISDN-PRI service providers? |
| 11:14.05 | JT | Daviey: i'm sure BT do it |
| 11:14.22 | JT | Daviey: don't you know what other telcos are around? |
| 11:15.00 | Daviey | Yeah.. our telco's are kinda messed up |
| 11:15.26 | Daviey | BT usually do the work, but there are often contracted by different suppliers |
| 11:15.28 | JT | here's a tip: ISDN30 |
| 11:15.33 | Daviey | thanks |
| 11:15.34 | Daviey | :) |
| 11:15.35 | JT | heh |
| 11:15.59 | JT | i have no idea if BT is your only PRI provider or not |
| 11:17.17 | ai-a | DaveCanoe: i cant say which is best, but we use Kingston tech. but they are crap on tech support. |
| 11:18.06 | Daviey | ai-a: thanks |
| 11:26.25 | *** join/#asterisk cfz (n=suxwin@mail1.mr-bricolage.bg) |
| 11:31.58 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:32.07 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:32.15 | DrukenLPY | morning everyone |
| 11:33.05 | Dr-Linux | hhm.. |
| 11:33.15 | DrukenLPY | hey Dr-Linux, ltns |
| 11:33.23 | Dr-Linux | ${DIALOPTS})) << this is a channel veriable or global variable? :S |
| 11:33.33 | Dr-Linux | what is DIALOPTS ? |
| 11:34.04 | Dr-Linux | DrukenLPY: ? |
| 11:34.46 | mvanbaak | prolly global variable |
| 11:34.54 | DrukenLPY | not sure... looks to me like it's a custom variable... |
| 11:34.58 | DrukenLPY | perhaps dial options? |
| 11:35.01 | mvanbaak | yeah |
| 11:35.10 | mvanbaak | set somewhere in extensions.conf |
| 11:35.17 | mvanbaak | that's my bet |
| 11:35.36 | DrukenLPY | could be used in a dynamic system for the dial options of a peer.... |
| 11:35.40 | DrukenLPY | i used to do that... |
| 11:36.17 | Dr-Linux | DrukenLPY: you are correct |
| 11:36.49 | Dr-Linux | but my question is should i use this variable something in dialplan, or i can use it directly :S |
| 11:36.58 | Dr-Linux | where can i set it's vaules :S |
| 11:37.02 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 11:37.23 | DrukenLPY | in the dialplan |
| 11:37.34 | Dr-Linux | how |
| 11:37.55 | naain | JT: I have read and try all the new variable as well, but still failed, see the pastbin http://pastebin.ca/633030 |
| 11:39.17 | Dr-Linux | aww |
| 11:39.18 | Dr-Linux | got it |
| 11:39.19 | Dr-Linux | Set(DIALOPTS=tTrwWL(60000:20000:5000)) |
| 11:39.44 | DrukenLPY | diging into someone else's dialplan ? |
| 11:40.30 | DrukenLPY | naain: sure you got callerid on the zap channels? |
| 11:40.36 | DrukenLPY | from your telco |
| 11:42.34 | naain | DrunkenLPY: Yes, when i connect lines to callerid capable set it shows callerid |
| 11:42.47 | JT | naain: what country are you in? |
| 11:42.48 | DrukenLPY | these are analog lines then ? |
| 11:42.56 | *** join/#asterisk davidcsiii (n=davidcsi@212.166.169.27) |
| 11:43.05 | naain | yes analog lines |
| 11:43.29 | DrukenLPY | what country as jt asked are you in ? |
| 11:43.56 | naain | Pakistan |
| 11:44.22 | davidcsiii | hello all, question: i'm sending peers to different context depending on IP address withOUT registration. i.e. 192.168.10.1 to context [server1] and any other ip ton context default. now, the question is: On what does asterisk bases to send peers to contextes? on the ip on SIPURI or on the TCP/IP layer IP address? |
| 11:45.05 | JT | naain: find out what callerid mode is used in pakistan |
| 11:45.15 | JT | naain: and set it in zapata.conf |
| 11:47.42 | DrukenLPY | JT, do you know if i can run lets say 10 PSTN, and 11 DATA on a single PRI, and actually use them both at the same time using a single sangoma card? |
| 11:48.33 | *** join/#asterisk kova (n=Koen@tech.quentris.be) |
| 11:48.35 | JT | DrukenLPY: i believe so, yes |
| 11:48.41 | DrukenLPY | sweet |
| 11:49.00 | kova | hi all! |
| 11:49.17 | kova | how can I reload rsa keys without restarting asterisk? |
| 11:49.22 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 11:49.49 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:50.40 | naain | JT: how can i find callerid mode is being used in Pakistan and how to set mode in zapta.conf (at which placed in config file) |
| 11:52.14 | JT | naain: research online or call your telco |
| 11:52.30 | JT | as for setting the mode, you need to find that out yourself |
| 11:53.05 | tzafrir_laptop | davidcsiii, you wanted to ask something? |
| 11:53.33 | tzafrir_laptop | <davidcsi> hello all, question: i'm sending peers to different context depending on IP address withOUT registration. i.e. 192.168.10.1 to context [server1] and any other ip ton context default. now, the question is: On what does asterisk bases to send peers to contextes? on the ip on SIPURI or on the TCP/IP layer IP address? |
| 11:59.48 | tzafrir_laptop | davidcsiii, I don't think you can do that ith asterisk. only by username |
| 11:59.52 | tzafrir_laptop | anybody? |
| 12:00.43 | ZeeRoCOOOL | ? |
| 12:01.05 | kova | indeed ... ? |
| 12:01.19 | kova | I thought context was defined in sip.conf |
| 12:01.40 | kova | or did I misunderstand the question |
| 12:01.57 | DrukenLPY | tzafrir: i belive you are correct.. i don't know of ip blocking to contexts |
| 12:02.06 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 12:03.30 | tzafrir_laptop | and you can't send a peer (chan_sip terminology) to a context without registration. That would be a user or a friend, BTW |
| 12:04.08 | tzafrir_laptop | I think that he's asking for a workaround |
| 12:06.22 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 12:09.07 | lsodi | is any one online who uses MOR free 0.4.10? |
| 12:11.26 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.31 | *** join/#asterisk CM3_1_2_632 (n=CM3_1_2_@pcd503191.netvigator.com) |
| 12:15.28 | CM3_1_2_632 | morning..... |
| 12:16.36 | creativx | mid day |
| 12:17.29 | CM3_1_2_632 | good mid-day..... |
| 12:20.27 | Maliuta | it's 22:20 |
| 12:20.35 | Maliuta | :P |
| 12:21.15 | naain | JT: I have discussed with one of other user with Same Teleco in Pakistan they are using Digium Card and getting Callerid with same zapta.conf file, but with sangoma card it shows nothing |
| 12:21.48 | kova | how is the experience here with chanskype? |
| 12:24.22 | kova | does anyone use chanskype? is it worth the money? |
| 12:24.25 | creativx | kova: slim to none. |
| 12:25.22 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 12:25.24 | hank | hi |
| 12:25.49 | kova | creativx, are there other solutions to connect with skype? |
| 12:25.53 | Maliuta | I object to skype on principle ... I don't like proprietory protocols |
| 12:26.21 | Maliuta | and I don't like some of the firewall avoidance stuff skype have been doing |
| 12:26.35 | CM3_1_2_632 | i liked it as long as it's free..... |
| 12:26.36 | Maliuta | as a sysadmin it makes my skin crawl |
| 12:26.52 | kova | Maliuta, I agree ... but now it's there and used by a lot of people, we have to live with it |
| 12:27.18 | hank | Over here we have asterisk with snom VoIP phones. When colleague1 is on the phone and at the same time colleague2 calls colleague1, colleague2 hears the 'free' tone and colleague1 is signalled that there is another call for him. Where would i change it so colleague2 hears a 'busy' tone? |
| 12:27.27 | Maliuta | kova: or ignore it :) |
| 12:27.52 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 12:28.35 | tzafrir_laptop | any alternative that uses standard protocols and Just Works? |
| 12:28.55 | kova | Maliuta, that's up to you ... I want to connect Skype to our internal telephony ... and I was thinking of using chanskype, hence my question |
| 12:29.36 | tzafrir_laptop | kova, it might provide you with audio, but not text messages |
| 12:30.43 | kova | tzafrirn that's the point ... can't do much with text messages on telephony ;-) |
| 12:31.10 | JT | kova: chan skype is a big hack |
| 12:31.12 | tzafrir_laptop | kova, SIP has SIMPLE. |
| 12:31.15 | JT | waste of time anyway |
| 12:31.19 | tzafrir_laptop | some SIP phones support it. |
| 12:31.42 | tzafrir_laptop | IAX has nothing |
| 12:32.26 | kova | JT, I have seen how it works ... but how does it run in general? is it reliable? |
| 12:32.41 | JT | no |
| 12:33.19 | JT | CM3_1_2_632: skype is rubbish, and only free as in beer |
| 12:34.08 | kova | that's a pity ... more and more companies are interested in connecting telephony equipment to skype |
| 12:34.10 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.163) |
| 12:34.16 | CM3_1_2_632 | JT: beer's good....skype's not beer so skype's not good.... |
| 12:34.44 | JT | kova: companies can be quite silly, just like people |
| 12:34.49 | kova | any other solutions then? |
| 12:36.03 | kova | JT, unfortunately the world is not just black and white ... |
| 12:36.26 | kova | if skype was black and sip was white, chanskype would be grey |
| 12:36.53 | JT | kova: i know the world is not black and white |
| 12:36.58 | JT | but skype is rubbish |
| 12:37.11 | JT | and all solutions to use it in asterisk are a hack |
| 12:37.23 | JT | that involve silly things like simulating sound cards |
| 12:38.12 | Rienzilla | hey do you guys know a way to use ventrilo w/ asterisk? |
| 12:39.07 | kova | since the protocol is proprietary, hacking the sound device is the only way for now I guess |
| 12:39.50 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:39.50 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:40.35 | JT | Rienzilla: no. how many times have you asked? |
| 12:40.47 | JT | Rienzilla: yet another proprietary voip protocol |
| 12:40.56 | JT | forget about using it with asterisk |
| 12:42.17 | Rienzilla | hm ok |
| 12:42.29 | Rienzilla | too bad |
| 12:42.52 | JT | for you perhaps |
| 12:43.01 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 12:43.04 | Rienzilla | yes |
| 12:43.05 | JT | i suspect most people don't care about ventrilo :) |
| 12:43.08 | Rienzilla | I know |
| 12:43.31 | Rienzilla | I just wanted to use my sip phone with headset to connect with ventrilo :) |
| 12:43.46 | kova | honestly, never heard of ventrilo before |
| 12:44.03 | Rienzilla | it's a voice conferencing thingy, mainly used for online gaming |
| 12:44.06 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 12:44.36 | kova | thanks for the update ... have to go now |
| 12:44.58 | *** part/#asterisk kova (n=Koen@tech.quentris.be) |
| 12:47.05 | [TK]D-Fender | Rienzilla: Have them connect to your * instead |
| 12:47.25 | creativx | jag sitter i ventrilo og spiller lite dota |
| 12:47.42 | Rienzilla | yeah thats the alternative, but that requires them to install different client software |
| 12:48.03 | Rienzilla | creativx? |
| 12:48.32 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 12:50.44 | [TK]D-Fender | Rienzilla: Maybe (I've read about how to have MSN Messenger attach), but its worth it. |
| 12:51.51 | [TK]D-Fender | Rienzilla: Because before you know it you'll end up using Skype for this, Ventrilo for that, another client for 1 stupid thing, and one for REAL TELEPHONY. |
| 12:52.33 | Rienzilla | yeah |
| 12:52.34 | Rienzilla | I agree |
| 12:52.50 | Rienzilla | the thing is that ventrilo is pretty well integrated in the community |
| 12:53.11 | Rienzilla | so it's hard to get others to install a different thing |
| 12:53.24 | Rienzilla | btw, what softphones would you guys recommend for windows? |
| 12:53.29 | blitzrage | X-Lite |
| 12:53.39 | blitzrage | or idefisk |
| 12:53.44 | creativx | Rienzilla: basshunter has a song about ventrilo |
| 12:55.33 | kkn088 | Rienzilla: do you know mumble, a GPL TS/ventrilo like |
| 12:55.50 | Rienzilla | no |
| 12:56.17 | kkn088 | http://sourceforge.net/projects/mumble/ |
| 12:56.29 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:57.44 | Rienzilla | hmm |
| 12:57.45 | tzafrir_laptop | I use twinkle on Linux. some like Ekiga. I think Ekiga has a Windows version as well |
| 12:59.01 | Maliuta | kiax on linux works better for me |
| 12:59.21 | Maliuta | though tinkle is probably the best SIP softfone I tried |
| 12:59.54 | shido6 | tinkle, eh? |
| 13:00.15 | Maliuta | whatever |
| 13:00.40 | Maliuta | I had eye surgery 12 hours ago, so bite my shiny metal butt :P |
| 13:00.46 | shido6 | aim at the bullseye |
| 13:00.54 | shido6 | or you'll be moppin the floor |
| 13:01.24 | Maliuta | shido6: depth perception require 2 working eyes |
| 13:01.53 | shido6 | use the force , whatever... |
| 13:01.55 | [TK]D-Fender | Maliuta: And a subject to observe other than Kate Moss ;) |
| 13:01.57 | shido6 | hehe |
| 13:02.33 | Maliuta | lol |
| 13:04.25 | [TK]D-Fender | "A German court has once again upheld the GPLv2 and convicted Skype (based in Luxembourg) of violating the GPL by selling the Linux-based VoIP phone 'SMCWSKP 100' without proper source code access. (Original is in German, link is a Google translation.) Skype later added a flyer to the phones' packaging giving a URL where the sources could be obtained; but the court found this insufficient... |
| 13:04.26 | [TK]D-Fender | ...and in breach of GPL section 3. The plaintiff was once again Netfilter developer Harald Welte, who runs gpl-violations.org. The decision is available in German at www.ifross.de (Google translation here)." |
| 13:05.03 | JT | NICE |
| 13:05.07 | pigpen | Is it possible to send a sip header that makes a polycom play a ring tone to the speaker? ie: like a bell system |
| 13:05.57 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 13:06.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.44 | JT | [TK]D-Fender: great news |
| 13:07.13 | [TK]D-Fender | pigpen: No. |
| 13:07.20 | VJFROMGT | any recomended asterisk manual out there? |
| 13:07.24 | pigpen | Oh well. Good idea though. |
| 13:07.27 | [TK]D-Fender | pigpen: Wait... you mean while NOT on a call? then yes |
| 13:07.35 | pigpen | Correct. |
| 13:07.56 | [TK]D-Fender | pigpen: Then its jsut a "Page", and you just need to record the ringing sound. |
| 13:07.58 | pigpen | Exampe: Kids are ready for the end of Chemistry. A bell needs to ring. A polycom is in the classroom. |
| 13:08.23 | pigpen | Yeah..that is what I have running now. I am using the allpage.agi. |
| 13:08.30 | [TK]D-Fender | pigpen: triggered call-file "ringing page" |
| 13:08.50 | pigpen | yeah..using the page app right? |
| 13:09.01 | Corydon76-home | We use IAXys connected to a legacy overhead paging system |
| 13:09.28 | Corydon76-home | The Polycoms aren't quite loud enough |
| 13:09.34 | pigpen | Corydon76-home, yeah..I have "added" a output to some polycom's to attach to the overhead in some buildings. |
| 13:09.46 | [TK]D-Fender | pigpen: When i say Page I wasn't really meaning the app. No sense in that. Just a call-file w/ autoanswer on the Polycom end where you playback a recording and hangup. |
| 13:09.52 | pigpen | I just thought a sip header would be nice and small. |
| 13:10.10 | [TK]D-Fender | Corydon76-home: They can be.... max the volume on speaker & tweak your recording :) |
| 13:10.15 | Corydon76-home | pigpen: the setup allows us to send pages to one building or all four... |
| 13:10.23 | pigpen | Ah..yeah...the only issue is that it is kinda slow. |
| 13:10.28 | pigpen | with the call file. |
| 13:10.44 | pigpen | ie: 80-100 phones takes about 30-60 seconds. |
| 13:10.48 | [TK]D-Fender | pigpen : AMI Originate then... same dael. |
| 13:10.51 | pigpen | schools don't like that . |
| 13:11.00 | creativx | cron script, ami originate: |
| 13:11.03 | Corydon76-home | High school, middle school, fine arts building, athletics building |
| 13:11.11 | pigpen | ami originate...ok... |
| 13:11.19 | creativx | i take it you wonder what ami is |
| 13:11.20 | creativx | hehe |
| 13:11.25 | creativx | then i wonder.. is there a |
| 13:11.27 | creativx | ~ami |
| 13:11.27 | jbot | extra, extra, read all about it, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 13:11.34 | creativx | :) |
| 13:11.39 | pigpen | Corydon76-home, yep..but unfortunatly, they didn't put the overhead into some of the buildings.... |
| 13:11.58 | pigpen | creativx, tks. |
| 13:12.16 | pigpen | yeah..Asterisk Manager Interface...sure...sorry. |
| 13:12.19 | pigpen | dense this am. |
| 13:12.23 | Corydon76-home | pigpen: right, they need to be separate systems... otherwise you have issues with differing ground references |
| 13:12.25 | *** join/#asterisk friedrich| (n=friedric@e177250070.adsl.alicedsl.de) |
| 13:15.16 | *** join/#asterisk myiagy (i=myiagy@189.4.123.131) |
| 13:16.37 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:17.53 | *** join/#asterisk af_ (n=getsmart@81-174-46-138.dynamic.ngi.it) |
| 13:19.46 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 13:20.33 | sopo2k4 | Im having problems with my Inbound number from VoipTalk, ive selected to have it routed to number@my-asterisk-box over SIP, and well i ring it and it just shows Number not available. I have it setup in sip.conf and extensions.conf however asterisk doesnt even show any log |
| 13:20.59 | creativx | sopo2k4: enable debugging |
| 13:21.01 | blitzrage | what does 'sip debug' from the CLI say? Is it even hitting the asterisk box? |
| 13:21.12 | blitzrage | did you open the right ports? |
| 13:21.13 | JT | sopo2k4: asterisk has a public ip address? |
| 13:21.25 | blitzrage | 5060 is signalling, and 10000-20000 is the (default) RTP |
| 13:21.34 | sopo2k4 | blitzrage ive DMZ'd the asterisk box |
| 13:21.40 | sopo2k4 | so all ports should be open |
| 13:21.44 | blitzrage | *should* |
| 13:21.50 | JT | sopo2k4: asterisk has a public ip address? |
| 13:21.51 | sopo2k4 | and reachable. and theres no debug in the CLI |
| 13:21.58 | sopo2k4 | i dont think its hittin the asterisk box for some reason |
| 13:22.03 | sopo2k4 | its 192.168.2.9 |
| 13:22.08 | JT | sopo2k4: it doesn't then |
| 13:22.09 | blitzrage | don't *think*... KNOW! |
| 13:22.13 | JT | sopo2k4: you must port forward. |
| 13:22.24 | blitzrage | DMZ is for boxes to get an external IP |
| 13:22.29 | sopo2k4 | hm |
| 13:22.33 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:101:382a:df3b:1f18:4150) |
| 13:22.35 | mosty | sopo2k4, are you registered to the sip server? |
| 13:22.38 | sopo2k4 | so could i set the asterisk box to listen on my main ip? |
| 13:22.40 | sopo2k4 | yes |
| 13:22.58 | sopo2k4 | if its been put on the dmz list |
| 13:23.05 | mosty | sopo2k4, have you checked with sip show registry? |
| 13:23.11 | blitzrage | sopo2k4: no.. you'd have to have a 2nd IP since the router has the primary address |
| 13:23.28 | *** join/#asterisk CuriosCat (i=stian@ninja.noc.host.net) |
| 13:23.29 | sopo2k4 | voiptalk.org:5060 84491991 105 Registered Wed, 25 Jul 2007 14:22:12 |
| 13:23.44 | sopo2k4 | ok |
| 13:23.48 | blitzrage | registration doesn't mean incoming will work |
| 13:23.48 | JT | actually |
| 13:23.54 | JT | you don't need to port forward |
| 13:24.00 | blitzrage | although it means your username/pass is probably ok :) |
| 13:24.04 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 13:24.05 | JT | i thought a client was trying to connect to your box for some reason |
| 13:24.19 | sopo2k4 | nah, im only trying to get my asterisk box to receive the incoming call |
| 13:24.24 | blitzrage | I never port forward on my Linksys routers |
| 13:24.34 | sopo2k4 | ive got a damn belkin |
| 13:24.36 | blitzrage | although I guess I'm talking about phones, and not Asterisk behind NAT |
| 13:24.37 | sopo2k4 | thats my problem in one i think. |
| 13:24.42 | JT | it's generally unnecessary |
| 13:24.54 | JT | sopo2k4: enable sip debug and see if you get anything when you try and call the box |
| 13:24.56 | Maliuta | sopo2k4: the incoming may come in on a differnt ID than the registered one |
| 13:25.11 | sopo2k4 | ok hold up let me enable sip debug |
| 13:25.33 | Maliuta | sopo2k4: I have to have 2 entries for my SIP connection, one is the outgoing (which registers) and the other is for the incoming |
| 13:25.41 | sopo2k4 | yes |
| 13:25.43 | msetim | hi guys, what make the res_monkey.c? |
| 13:25.45 | sopo2k4 | it shows something |
| 13:25.51 | sopo2k4 | quite alot actually |
| 13:25.56 | sopo2k4 | want a pastebin? |
| 13:26.04 | Maliuta | yeah, worth a look |
| 13:26.07 | sopo2k4 | hold up |
| 13:26.42 | JT | Maliuta: that's crazy |
| 13:26.51 | JT | Maliuta: why do you need to register to send calls? |
| 13:27.16 | mosty | JT: some sip servers require it, for some reason |
| 13:27.17 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 13:27.24 | Maliuta | JT: the register tells the provider where I am |
| 13:27.32 | hank | The Dial-Function does not seem to notice when one of our snom-phones is busy. What could be the problem? |
| 13:27.35 | [TK]D-Fender | sopo2k4: I'm quite sure I told you YESTERDAY to go HERE : |
| 13:27.38 | Maliuta | even though I have a static and never move |
| 13:27.40 | [TK]D-Fender | ~sipnat |
| 13:27.41 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:27.43 | hank | Do i HAVE to use ChanIsAvail? |
| 13:27.47 | sopo2k4 | http://pastebin.com/m1a209ba2 |
| 13:27.51 | mosty | hank: do you have qualify=yes for that sip peer? |
| 13:28.03 | hank | mosty: its in the general section of sip.conf |
| 13:28.19 | mosty | hank: what is the status of the peer when you do sip show peer peernamegoeshere |
| 13:28.24 | JT | Maliuta: hence crazy if you only make calls with that account, the provider does not need registrations for you to make a call |
| 13:28.37 | [TK]D-Fender | hank: You missed the point.... phones can handle multiple calls and will ACCEPT another if their on the phone up until the max # is reached (varies by model). |
| 13:29.04 | Maliuta | JT: mynetfone ... I have a DID on it aswell |
| 13:29.16 | JT | oh |
| 13:29.16 | [TK]D-Fender | hank: So if you want to prevent CW, etc, then you'll either have to check it yourself, or disable it on the phone, or place a sip.conf call-limit. |
| 13:29.23 | JT | mynetfone are a pack of idiots |
| 13:29.27 | Maliuta | sopo2k4: who is the provider? |
| 13:29.30 | JT | they have no idea how to run an ITSP |
| 13:29.33 | sopo2k4 | voiptalk.org |
| 13:29.34 | Maliuta | JT: yeah |
| 13:29.36 | mosty | hank: is the snom phone doing call waiting, or is it really busy? |
| 13:29.39 | JT | whoever setup their sip stuff is a moron |
| 13:29.56 | Maliuta | sopo2k4: the callerid setting looks like a brisbane phone number |
| 13:30.08 | JT | Maliuta: wouldn't your DID need to register? |
| 13:30.12 | sopo2k4 | hmm |
| 13:30.15 | creativx | mosty: as tk said, theres 3 ways this is limited, either in sip.conf with call-limit, the phone has a licence that says it can have X number of active lines, or call waiting on the phone |
| 13:30.18 | [TK]D-Fender | looking for 01962658744 in default (domain asterisk-uk.zapto.org) |
| 13:30.19 | [TK]D-Fender | <--- Reliably Transmitting (NAT) to 217.14.132.185:5060 ---> |
| 13:30.19 | Maliuta | JT: I am looking to move, their idea of untimed is 2hrs |
| 13:30.21 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 13:30.36 | JT | Maliuta: right |
| 13:30.37 | [TK]D-Fender | sopo2k4: You ARE apparently geting the call and your DIALPLAN is defective |
| 13:30.38 | mosty | creativx, i know. hank is the person asking |
| 13:30.41 | hank | mosty: OK |
| 13:30.42 | JT | Maliuta: they don't have infrastructure |
| 13:30.43 | sopo2k4 | hmm |
| 13:30.46 | JT | Maliuta: they just resell |
| 13:30.48 | creativx | mosty: please fwd np tnx |
| 13:30.49 | creativx | ;) |
| 13:30.57 | JT | have the brochure of the wholesalers here |
| 13:31.02 | [TK]D-Fender | sopo2k4: looking for 01962658744 in default (domain asterisk-uk.zapto.org) <--------------- pay attention |
| 13:31.03 | sopo2k4 | i knew it wasnt my router |
| 13:31.19 | JT | Maliuta: they mainly make the money by charging per minute instead of per second |
| 13:31.20 | sopo2k4 | ok |
| 13:31.28 | [TK]D-Fender | sopo2k4: it could have been, or it could have been your settings in sip.conf, etc. |
| 13:31.52 | [TK]D-Fender | sopo2k4: could be a lot fot hings, but you seemed to think not showing use your sip.conf was a good way to let us help you. |
| 13:32.53 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 13:32.57 | hank | [TK]D-Fender, mosty, creativx: Thanks. That did it :) |
| 13:33.05 | Maliuta | JT: at 3c/min to canada they are still smegloads cheaper than hellstra |
| 13:33.22 | Maliuta | JT: who would you recomend in .au? need a brisbane DID |
| 13:33.43 | sopo2k4 | <PROTECTED> |
| 13:33.57 | JT | Maliuta: not for calls to mobiles |
| 13:34.15 | JT | which is the most expensive phone call type in australia |
| 13:34.32 | JT | Maliuta: pennytel seems ok |
| 13:34.32 | [TK]D-Fender | sopo2k4: Well, we DON'T need it right now, but you SHOULD have provided it at the start, adn INCLUD the [general] section...... |
| 13:34.46 | [TK]D-Fender | sopo2k4: right now you know EXACTLY what you need to add, so go do it. |
| 13:35.07 | sopo2k4 | lol.... |
| 13:35.18 | Maliuta | sopo2k4: I am assuming you have a 01962658744 context in your extensions.conf |
| 13:35.28 | sopo2k4 | yup |
| 13:35.45 | Maliuta | JT: Iuse my mobile for mobile calls, I don't use all my cap anyhow |
| 13:35.53 | sopo2k4 | [01962658744] |
| 13:35.53 | sopo2k4 | exten => 01962658744,1,Dial(IAX2/james) |
| 13:35.58 | sopo2k4 | thats it... |
| 13:36.06 | Maliuta | JT: what about faktortel? they do IAX |
| 13:36.18 | JT | faktortel are complete morons |
| 13:36.25 | JT | avoid at all costs |
| 13:36.31 | JT | one of the worst ITSPs in .au |
| 13:36.52 | JT | their sydney dids had been down for 3 weeks, a few weeks ago, dunno if they ever fixed it |
| 13:37.06 | JT | and they abuse you if you "make too many calls" |
| 13:38.20 | Maliuta | hmm |
| 13:38.24 | [TK]D-Fender | Maliuta: WRONG ANSWER |
| 13:38.28 | [TK]D-Fender | sopo2k4: DITTO |
| 13:38.35 | Maliuta | node phone are way 'spensive |
| 13:38.38 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:38.40 | sopo2k4 | lol |
| 13:38.53 | JT | Maliuta: pennytel are about the cheapest in .au for rates |
| 13:39.00 | Hymie | -- Incoming call: Got SIP response 500 "Internal Server Error" back from x.x.x.x <-- anyone getting a lot of these messages back from polycoms after an asterisk restart? |
| 13:39.04 | [TK]D-Fender | Maliuta: YOU TOO :) |
| 13:39.15 | [TK]D-Fender | Hymie: Yup! Flooded by them! |
| 13:39.24 | [TK]D-Fender | Hymie: "Mostly Harmless" |
| 13:39.28 | Hymie | oh no |
| 13:39.45 | Maliuta | JT: I will look at them, thanks |
| 13:40.07 | JT | Maliuta: the prepay thing is annoying though |
| 13:40.17 | Maliuta | mynetfone are the same |
| 13:40.40 | JT | Maliuta: if DIDs are mission critical, forget most ITSPs |
| 13:40.46 | JT | Maliuta: for what type of calls? |
| 13:40.55 | CuriosCat | if DIDs are mission critical, get POTS lines :) |
| 13:41.09 | JT | wrong answer |
| 13:41.14 | JT | CuriosCat: ISDN |
| 13:41.23 | JT | pots is bad for lots of DIDs |
| 13:41.27 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 13:41.39 | Maliuta | JT: mostly I want it for the brisbane DID and calls to fixed lines, plus international calls to canada |
| 13:41.52 | CuriosCat | JT: Well, if it's lots, get a PRI. My point is don't rely on VOIP for mission-critical. :P |
| 13:42.24 | Maliuta | CuriosCat: yeah, because I can get a POTS DID from 2000kms away |
| 13:42.29 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 13:42.34 | Maliuta | it's my home setup |
| 13:42.41 | coolbeans | Anyone have a clue why in 1.2.18, when using static real-time and mysql, voicemail passwords aren't updated in the DB when changed with app_voicemail? It changes them in asterisk, but never updated to the db. Of course, a restart of app_voicemail restores whatever passwords are in the db. Any help would be appreciated. |
| 13:43.05 | CuriosCat | Maliuta: You can, it's just fantastically expensive. But why is a DID from 2000 kms away mission critical? |
| 13:43.16 | Maliuta | if this were for a business I'd recommend VoIP internal and a PRI/BRI solution |
| 13:43.17 | JT | Maliuta: i fail to see how the prices are the same |
| 13:43.43 | JT | Maliuta: pennytel: Canada 0.00878/m mynetfone: Canada Landline 1306 $0.019/m |
| 13:43.58 | zeeesh | i just donwloaded asterisk-1.4.8 ,,,getting error " |
| 13:43.58 | zeeesh | " |
| 13:44.12 | zeeesh | configure: error: *** termcap support not found |
| 13:44.43 | Maliuta | JT: I'm on a differnt plan to that mynetphone one |
| 13:44.46 | JT | zeeesh: libncurses-dev |
| 13:44.58 | [TK]D-Fender | zeeesh: yum install libtermcap libtermcap-devel newt newt-devel ncurses ncurses-devel |
| 13:44.58 | JT | Maliuta: how much are you paying? |
| 13:45.38 | zeeesh | No Match for argument: libtermcap? |
| 13:45.50 | Maliuta | JT: $9.95/mnth and $0.03/minute to canada |
| 13:46.01 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:46.09 | Maliuta | pretty sure, they did just change the charges, but not by that much |
| 13:46.24 | JT | Maliuta: that's even MORE expensive |
| 13:46.40 | JT | Maliuta: mynetfone also charge per 60seconds, which is awful |
| 13:46.43 | *** join/#asterisk jsbach (n=jsbach@fokus6150.fokus.fraunhofer.de) |
| 13:46.47 | Maliuta | JT: the monthly includes a DID and 100 fixed line calls |
| 13:46.58 | JT | Maliuta: pennytel charge per second with no minimum |
| 13:47.06 | Maliuta | the fixed line calls are untimed (read max 2hrs) |
| 13:47.41 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:48.25 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 13:48.26 | JT | Maliuta: that's not international is it? |
| 13:48.48 | Maliuta | no, the international rates are different |
| 13:49.06 | Maliuta | pennytel look better |
| 13:49.22 | *** join/#asterisk ccesario_ (n=ccc@189-19-9-100.dsl.telesp.net.br) |
| 13:49.24 | Maliuta | don't know why I didn't see them a year ago when I did the inital setup |
| 13:50.51 | JT | don't get me wrong |
| 13:51.01 | JT | pennytel is not a high end business ITSP |
| 13:51.07 | JT | but neither is mynetfone |
| 13:51.18 | JT | mynetfone is utterly useless for most businesses imho |
| 13:51.37 | JT | they can't work out how to do more than 1 simultaneous phone call per sip account |
| 13:51.39 | syzygyBSD | anyone know a good sip termination in NZ? |
| 13:52.14 | syzygyBSD | I mean, I can set one up in a couple months... but until then... |
| 13:52.27 | mosty | JT, they probably know how, but want more accounts, and make their money on account fees |
| 13:52.34 | jsbach | hei, i am calling an xlite client using asterisk in the middle. the prob is i am getting an 480 (?) from the client... anyone had this before? |
| 13:53.04 | JT | mosty: no, i really think that's just how they've technically set it up |
| 13:53.19 | JT | mosty: if you buy a business plan with 4 "lines" you must use 4 sip accounts |
| 13:53.23 | JT | which is stupid |
| 13:53.34 | mosty | JT, if that's true, then they really are stupid |
| 13:53.58 | JT | i rang them up |
| 13:54.05 | JT | and asked all about it |
| 13:54.12 | JT | that's what they told me |
| 13:54.28 | Maliuta | JT: not using it for business :) |
| 13:54.33 | mosty | maybe the person on the phone was just confused about what a "line" is |
| 13:54.57 | JT | mosty: they checked with "techs" |
| 13:54.58 | Maliuta | JT: yeah, most of the ITSP's I have looked at want to charge the earth for extra lines |
| 13:55.06 | mosty | JT, heh well that is crap then |
| 13:55.18 | JT | Maliuta: the one good thing about engin is you get unlimited inbound and outbound |
| 13:55.38 | jsbach | any ideas ppl? |
| 13:57.25 | mosty | jsbach, sip response 480 means "temporarily unavailable". ie it's a config thing on the x-lite i guess |
| 13:58.40 | *** part/#asterisk hank (n=hank@netwichtig.de) |
| 13:58.48 | jsbach | mosty, thanx for response, jah, firstly the xlite client is on, registered at asterisk (* console says so) and i allowed any calls from my domain.. |
| 13:58.58 | jsbach | i actually donnu what i should configure more.. |
| 13:59.09 | jsbach | configure more... on the client side.. |
| 13:59.21 | mosty | jsbach, show us your dial command |
| 13:59.36 | mosty | and does x-lite have any logs you can look at? |
| 14:00.55 | jsbach | mosty, not really.. sorry for the unpricise answer cuz it just opens a regular "my documents" folder with regular docs as i want to click "look at logs" which i am not used to from linux anyway :P |
| 14:01.35 | jsbach | mosty, u want to see the whole dialog? on pastebin? |
| 14:02.05 | [TK]D-Fender | sopo2k4: Reminder <------------- |
| 14:02.07 | [TK]D-Fender | [TK]D-Fender>looking for 01962658744 in default (domain asterisk-uk.zapto.org) |
| 14:02.09 | [TK]D-Fender | <--- Reliably Transmitting (NAT) to 217.14.132.185:5060 ---> |
| 14:02.10 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 14:02.18 | mosty | are you using x-lite on linux? don't- it sucks big time |
| 14:02.25 | jsbach | hei [TK]D-Fender ;) |
| 14:02.43 | [TK]D-Fender | mosty: Don't worry... it sucks everywhere else too! ;) |
| 14:02.56 | jsbach | mosty, no, i use some other on linux, but on windows at second client i have a xlite |
| 14:03.21 | *** join/#asterisk juxhi (n=juxhi@nathan.epi.usf.edu) |
| 14:03.22 | [TK]D-Fender | jsbach: IM IN UR PBX R00TING UR CALLZ! |
| 14:03.35 | mosty | [TK]D-Fender, i'm not a windows person, thankfully |
| 14:04.14 | jsbach | [TK]D-Fender, jah... |
| 14:04.53 | brodiem | What is a good way to implement call waiting in *? On my old install I just took dialparties.agi from AMP but in rebuilding on 1.4, I'm not sure how to get the device state of a SIP ext without using Manager API and I'm using realtime for the SIP users. |
| 14:05.36 | brodiem | Going to try fun_devstate, not sure if it works using ARA though |
| 14:06.06 | coolbeans | Anyone have a clue why in 1.2.18, when using static real-time and mysql, voicemail passwords aren't updated in the DB when changed with app_voicemail? It changes them in asterisk, but never updated to the db. Of course, a restart of app_voicemail restores whatever passwords are in the db. Any help would be appreciated. |
| 14:06.55 | puzzled | coolbeans: I would check the changelog from 1.2.23 if it has been fixed in the mean time (if it really is a bug) |
| 14:07.04 | mosty | brodiem, the phones handle it, don't they? |
| 14:07.47 | jsbach | mosty, http://paste.css-standards.org/20578 |
| 14:08.11 | jsbach | mosty, the 480 contains a warning header: Warning: 399 devnull "User reject" |
| 14:08.17 | brodiem | mosty Yeah I had just hoped to keep it pbx-based since there's a number of phone vendors in the mix, at a couple different locations, etc... but I can if I have to |
| 14:09.03 | JT | brodiem: call waiting is a function of the phone |
| 14:09.05 | JT | not asterisk |
| 14:09.06 | mosty | jsbach, what's your dial command look like? |
| 14:09.15 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:09.40 | jsbach | and how can i say asterisk not to manipulate from tags to "caller" <sip:asterisk@ip_address:8060> ? |
| 14:09.57 | jsbach | mosty, my invite is at the beginning of that paste uri |
| 14:10.04 | jsbach | INVITE alice@my_domain.com |
| 14:10.06 | mosty | brodiem, i think call waiting on the asterisk side would be a nasty hack. i would recommend to the clients that they pick a phone that supports multiple simultaneous calls, it should just work |
| 14:10.30 | mosty | jsbach, can i see your dial command? |
| 14:10.38 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:10.46 | jsbach | mosty, INVITE sip:alice@semiconductor.jsbach SIP/2.0 |
| 14:10.59 | mosty | jsbach, i mean the dial command in your dialplan |
| 14:11.14 | jsbach | ok, i get it |
| 14:11.40 | jsbach | mosty, exten => alice,1,Dial(SIP/alice,10) |
| 14:11.41 | brodiem | mosty It did work good using dialparties.agi - it would just do an API ExtensionState and depending on the result would decide to send the call or not. I guess its not that big of a deal, its just all of the phones themselves have CW enabled by default rather than disabled, and their dial plans send all feature codes to the PBX directly |
| 14:12.10 | brodiem | I guess I'll just save the headache and do it on the phones.. |
| 14:14.11 | jerlique | I'm having problems with * listening to DTMF from a channel bank, the sip debug from * says "Unauthorised", any hints? |
| 14:14.14 | *** join/#asterisk wunderkin (n=wunderki@ip68-2-62-143.ph.ph.cox.net) |
| 14:14.29 | jsbach | mosty, any diagnosis there ? |
| 14:14.59 | mosty | jsbach, looks odd to me. since [TK]D-Fender tells us that x-lite is crappy on windows also, i'd recommend trying another client first |
| 14:15.14 | creativx | why is x-lite crappy? |
| 14:15.26 | creativx | am I using it wrong since its working fine? :p |
| 14:15.34 | jsbach | creativx, cuz i get a 480 to my sip invite.. |
| 14:15.42 | jsbach | creativx, and everything looks fine.. |
| 14:15.52 | brodiem | creativx and why did they pull 729 support from the free one |
| 14:16.18 | jsbach | mosty, [TK]D-Fender , which client is good then? |
| 14:16.53 | mosty | creativx, on linux it's full of memory leaks and race conditions and other fun things. i have no first hand experience with windows but i hear it's crap there too |
| 14:17.10 | creativx | ah.. it has been treating us nice in windows land |
| 14:17.24 | creativx | even works nice with the plantronics usb headsets |
| 14:17.29 | creativx | using ulaw |
| 14:18.20 | [TK]D-Fender | jsbach: idefisk |
| 14:18.46 | [TK]D-Fender | creativx: Oh it IS stable and works flawlessly (IME) but is LIMITED |
| 14:18.48 | mosty | on linux i use twinkle occasionally |
| 14:20.45 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:20.48 | creativx | [TK]D-Fender: that might be a point =) |
| 14:21.09 | [TK]D-Fender | creativx: no transfer, multi-call capability, no conference, etc. |
| 14:21.19 | creativx | yeah |
| 14:21.21 | [TK]D-Fender | creativx: idefisk gives you all that, plus IAX2. |
| 14:21.23 | creativx | i see why they work so well for us |
| 14:21.48 | creativx | transfer is done on-screen, with a live user list pulled from the ami, we don't do conferences =) |
| 14:23.45 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 14:24.16 | *** join/#asterisk Strom_M (n=strom@h72-2-22-215.bigpipeinc.com) |
| 14:25.36 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:27.41 | jmls | is there any way of specifying a slot to park a call in, or is it always the first available ? |
| 14:28.10 | kombi | how does one move the configs to somewhere else? changing astetcdir in asterisk.conf seems to have no effect |
| 14:31.00 | *** join/#asterisk oratelecom (n=davidsfe@bl10-160-38.dsl.telepac.pt) |
| 14:31.07 | oratelecom | hi!!! |
| 14:31.18 | oratelecom | someone can help me please?!?!!?!? |
| 14:31.47 | oratelecom | I have a problem in my server. |
| 14:31.47 | jmls | yup. The nearest burger king is in the high street, right next to Mcdonalds |
| 14:32.36 | creativx | im glad its inside your server, not outside of it. |
| 14:32.38 | oratelecom | how can i check if a phone is in use... if the phone suports multicalls |
| 14:32.47 | oratelecom | creativx... so funny! |
| 14:33.17 | mosty | oratelecom, you could use hints |
| 14:33.27 | oratelecom | specify that please.. |
| 14:33.37 | mosty | oratelecom, look it up on the wiki |
| 14:34.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:34.33 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:35.13 | oratelecom | thanks mosty but i already did.. |
| 14:35.24 | *** join/#asterisk shtoom (n=shtoom@59.181.111.61) |
| 14:35.49 | oratelecom | and i still can't find a solution for my problem |
| 14:36.33 | mosty | oratelecom, what is the problem exactly? |
| 14:37.15 | creativx | oratelecom: theres 3 ways multicalls are limited, either in sip.conf with call-limit, the phone has a licence that says it can have X number of active lines, or call waiting on the phone |
| 14:37.18 | JT | < brodiem> creativx and why did they pull 729 support from the free one |
| 14:37.29 | *** part/#asterisk shtoom (n=shtoom@59.181.111.61) |
| 14:37.37 | JT | ^ well that's one's obvious, G.729 codec licenses cost real money |
| 14:38.08 | oratelecom | creativx are you familiar with linksys phones? |
| 14:38.25 | mosty | oratelecom, what are you trying to do, that isn't working? |
| 14:38.35 | creativx | oratelecom: non |
| 14:38.43 | oratelecom | mosty |
| 14:39.37 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:40.04 | mosty | oratelecom |
| 14:40.05 | oratelecom | the problem is that when i check if the phone is availiable, it returns true, even when someone is using it, because the phone has call waiting suport. |
| 14:40.19 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.109.135) |
| 14:40.31 | zeeesh | installing asterisk-1.4.8 ... 1st untar 2nd ./configure and what is 3rd and 4th ? |
| 14:40.37 | mosty | oratelecom, so you want to limit the number of simultaneous calls to that client to 1? |
| 14:40.45 | oratelecom | yes |
| 14:40.46 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 14:40.50 | twitchnln | morning |
| 14:40.50 | mosty | zeeesh, no no no, first is read the docs |
| 14:41.03 | mosty | oratelecom, then set call-limit in sip.conf for that sip client |
| 14:41.13 | oratelecom | ok |
| 14:41.17 | oratelecom | thanks mosty |
| 14:41.20 | oratelecom | i will try that |
| 14:41.36 | oratelecom | then i'l feedback u |
| 14:41.55 | zeeesh | <mosty>: train left the station ... i did .. ./configure then make clear .. then make .. then make install .. |
| 14:42.02 | zeeesh | let c what will happen |
| 14:42.41 | *** join/#asterisk antimoof (n=dogcow@netbsd/developer/antimoof) |
| 14:43.09 | zeeesh | man its done ... why u were saying nononon |
| 14:43.16 | zeeesh | anyway thnx |
| 14:43.25 | antimoof | any of y'all know if digium has any documentation lurking about that compares wtf the differences are between their various PCI cards? |
| 14:43.37 | twitchnln | i'm attempting to setup automatic acd reports for my * but when i try to break the outbound out of it by src extension, it doesn't count transferred calls as being from an extension but the originating number, how can i break them out by extension? |
| 14:43.44 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 14:44.06 | mosty | antimoof, their website has product descriptions |
| 14:44.44 | antimoof | yes, all the descriptions are there - but you have to go and look at each item individually. |
| 14:44.53 | antimoof | I was hoping for more of a matrix view kinda thing. |
| 14:45.17 | [TK]D-Fender | antimoof: Sorry, no "quick comparison sheet". You're just going to have to cope with reading a few pages... |
| 14:45.22 | antimoof | you know. bullet time. cool jackets. trinity. and "this card does E1, this card does echo cancellation, and this does NA BRI". |
| 14:45.27 | mosty | antimoof, i think a matrix would not be very helpful, there would be so many empty spots |
| 14:45.59 | JT | antimoof: just describe your card requirements. |
| 14:46.08 | antimoof | I don't _know_ what my card requirements are yet. :) |
| 14:46.21 | mosty | then what are your functional requirements? |
| 14:46.24 | JT | analogue |
| 14:46.25 | JT | pri |
| 14:46.26 | JT | bri |
| 14:46.34 | [TK]D-Fender | antimoof: My chicken just ate your egg. |
| 14:46.40 | antimoof | but, for instance, the TE405P and TE410P have exactly the same copy. |
| 14:46.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:47.08 | antimoof | I know it's all digital telephony, and I know I want echo cancellation, and I want it to work in North America. prolly end up getting the 412 or 420. |
| 14:47.19 | mosty | antimoof, if i were you i'd just go straight to the sangoma website and look there instead |
| 14:47.22 | syzygyBSD | mmmm, chicken |
| 14:47.24 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:47.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:47.50 | [TK]D-Fender | antimoof: For single PRI I highly recommend the Sangoma A101d. |
| 14:47.58 | antimoof | oh, hell - the digium stuff is just rebadged sangoma? |
| 14:48.19 | JT | no. |
| 14:48.19 | mosty | antimoof, no digium cards just are not as good as the sangoma cards |
| 14:48.41 | mosty | antimoof, and from memory the sangoma models that end in "d" have echo cancellatin |
| 14:48.50 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 14:48.58 | antimoof | huh. 'k. (I'm an utter voip newbie; I'm merely going to be the sysadmin assembling and scripting and soforth. the guy who knows what he's talking about with PRIs and PBXs and all that crap isn't awake yet.) |
| 14:49.32 | JT | antimoof: the TE405P is 5v pci, the TE410P is 3.3v pci |
| 14:50.03 | JT | sangoma cards are universal voltage |
| 14:50.13 | antimoof | but I was thinking about getting my company (all praise the company!) to get a couple of cards and a server to use as a dedicated conference call box - but I dunno how much of the stream is actually processed on the CPU, how much is just shuffling bits around, etc. |
| 14:50.33 | antimoof | 5v only? eek! that's good to know. |
| 14:50.33 | JT | antimoof: depends if you're doing much |
| 14:50.38 | syzygyBSD | antimoof: how high of a load? |
| 14:50.46 | JT | antimoof: transcoding is what pulls real load |
| 14:50.58 | *** join/#asterisk illc0mm (n=bill@uslec-63-243-117-243.cust.uslec.net) |
| 14:51.12 | antimoof | why would I want/need to transcode? why can't I stick with just one codec? |
| 14:51.35 | antimoof | (the other idea was to eventually dump all our traffic directly into the PBX instead of terminating the VOIP stuff elsewhere. or whatever the proper term is.) |
| 14:51.48 | illc0mm | you can, as long as all the devices support the same codec |
| 14:52.01 | illc0mm | and you're not conferencing or modifing anything |
| 14:52.40 | JT | illc0mm: why would conferencing really impact? |
| 14:52.43 | antimoof | oh. hmmmm. yeah, I could see how for conferencing you'd have to sum all the channels together. nasty. good test of CPU or DSP or whatever. |
| 14:52.53 | illc0mm | Yeah it's shouldn't sorry |
| 14:52.58 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 14:52.58 | mosty | JT, well you have to superimpose channels i guess |
| 14:53.04 | JT | mosty: that's it |
| 14:53.06 | illc0mm | I was thinking strictly pass through |
| 14:53.12 | JT | and perhaps a transcode to slin |
| 14:54.03 | illc0mm | for the most part though, if your devices all support the same codec, no issue |
| 14:54.15 | antimoof | one of the nasty bits about getting bootstrapped with hardware in asterisk is figuring out what hardware does what - e.g. which cards are strictly for talking PRI, which actually do sound calculation offloading, etc. |
| 14:54.53 | JT | only one does offloading, the TC400B is a G.729/G.723 transcoding card |
| 14:54.57 | illc0mm | hey, anyone got bandwidth.com SIP trunks working? |
| 14:55.10 | illc0mm | sorry, anyone here.... I know someone has... hah |
| 14:55.12 | antimoof | how many of the end-user phones are gonna have g.279/g.273, though? |
| 14:55.13 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 14:55.34 | JT | antimoof: a lot of phones are G.729 capable |
| 14:55.41 | illc0mm | antimoof: cisco / polycom / snom |
| 14:55.44 | JT | doesn't mean you have to use it |
| 14:55.46 | illc0mm | to name a few |
| 14:56.04 | mosty | antimoof, i recommend you pick phones that support at least g729, gsm and g711 |
| 14:56.07 | Optic | g.729 r0x0rs my b0x0rz |
| 14:56.13 | Optic | ilbc is good food too |
| 14:56.17 | JT | mosty: hard requirement, close to none support gsm. |
| 14:56.20 | antimoof | well, I'd think that for conferencing, you'd want all the offloading you can get; if you have, say, 16 people all blathering at once, that's a lot of streams to decode/mux together. |
| 14:56.22 | illc0mm | g.729 is good for low bandwidth |
| 14:56.24 | Optic | but pretty much only grandstream supports it :) |
| 14:56.47 | illc0mm | ugh |
| 14:56.47 | neverblue | any VOIP providers in the channel? |
| 14:56.53 | mosty | JT, i primarily work with snoms. playing with some polycoms, though they seem optimised for g711 |
| 14:56.55 | neverblue | looking to try service |
| 14:57.04 | JT | mosty: they have g.729 too |
| 14:57.06 | Optic | illc0mm: haha, "ugh" grandstream? :) I agree |
| 14:57.11 | JT | mosty: i only run g.711 to my phones |
| 14:57.21 | Optic | i gave away my budgetone that I was using as a test handset |
| 14:57.28 | mosty | JT, yes but the polycom g729 doesn't sound so good in my opinion |
| 14:57.31 | Optic | it was the world's most ugly and horrible phone |
| 14:57.32 | JT | antimoof: you don't offload conferences |
| 14:57.44 | JT | antimoof: they are done by host cpu |
| 14:58.04 | coppice | mosty: G.729 never sounds as good as G.711 |
| 14:58.06 | mosty | JT, on my smaller installs i use g711, on my larger installs bandwidth becomes an issue |
| 14:58.09 | JT | mosty: just use G.711 |
| 14:58.25 | mosty | coppice, obviously. but some g729 implementations sound better than other g729 implementations |
| 14:58.55 | coppice | some must be badly broken, then :-\ |
| 14:58.55 | Optic | hiya |
| 14:59.02 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 14:59.09 | illc0mm | Optic: have several grandstream phones |
| 14:59.21 | illc0mm | Optic: not a fan |
| 14:59.26 | lirakis | .. i lovem |
| 14:59.27 | mosty | JT, polycoms support gsm don't they? |
| 14:59.43 | lirakis | i have some cisco/sipurra phones.. they suck |
| 14:59.46 | Optic | no, i bought a budgetone when i was trying * for the first time ever, because it was low risk |
| 14:59.56 | JT | lirakis: hope you're joking |
| 14:59.57 | JT | mosty: no |
| 14:59.59 | Optic | then we bought a pile of SPA-841's which were also piles of shit :P |
| 15:00.00 | lirakis | budgetone blows.. |
| 15:00.03 | JT | mosty: almost bothing does |
| 15:00.04 | lirakis | but .. the gxp-2000 |
| 15:00.06 | lirakis | is good |
| 15:00.08 | illc0mm | lirakis: dont get me wrong, for the price the grandstreams can't be beat. but there are a bit of a hit and miss |
| 15:00.17 | *** join/#asterisk el_critter (n=chatzill@190.74.124.133) |
| 15:00.18 | JT | lirakis: no, the GXP-2000 is utter unadulterated rubbish |
| 15:00.21 | JT | ~gs |
| 15:00.22 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:00.24 | JT | ^^^^^^^^^^^^6 |
| 15:00.24 | illc0mm | lirakis: yeah the gxp-2000 is better |
| 15:00.29 | mosty | optics: yeah the 8XX's are terrible. i hear the 9XX's are much better, but i still don't like them much |
| 15:00.37 | Optic | we've ben using Polycom IP430's lately, which I'm *very* happy with |
| 15:00.46 | Optic | nice screen, great sound, good build quality |
| 15:00.47 | illc0mm | Optic: Yeah? Been looking at those |
| 15:00.49 | Optic | and PoE! |
| 15:00.54 | illc0mm | Optic: how's the speaker phone? |
| 15:00.58 | el_critter | do you know good wireless phones? |
| 15:01.00 | coppice | anyone seen a grandstream 2020? they look a lot better in the pictures |
| 15:01.06 | Optic | usable, no speakerphone is great |
| 15:01.09 | lirakis | JT: i used to work at a large voip hardware providor.. ive used virtually all the big name phones.. the gxp-2000 is the simplest quickest and highest value for the entry/hobbyists dollar |
| 15:01.22 | Optic | we have 5 of the IP430's now |
| 15:01.22 | JT | lirakis: rubbish |
| 15:01.24 | [TK]D-Fender | IP430 = virtually pointless. IP 320 = budget killer phone. |
| 15:01.25 | mosty | my major gripe with polycoms at the moment is the web interface is horrible |
| 15:01.25 | Optic | the users all really like them |
| 15:01.26 | JT | lirakis: Polycom IP320 |
| 15:01.33 | Optic | ooh, I haven't used a 320 |
| 15:01.33 | JT | unbeatable price/performance |
| 15:01.43 | illc0mm | lirakis: Yeah, hobbiest, you got that right |
| 15:01.44 | *** join/#asterisk karleeto (n=karl@gentoo.karlhaines.com) |
| 15:01.50 | Optic | polycom provisioning is easy too |
| 15:01.53 | Optic | one software for all models |
| 15:01.54 | lirakis | JT: [Tk]D-fender and i have had this discussion before.. i wont run it into the ground again |
| 15:01.56 | illc0mm | lirakis: that being said, I have a Cisco 7960 at home |
| 15:01.56 | Optic | one config file for all models |
| 15:01.58 | JT | i wouldn't wish a grandstream upon any hobbyist |
| 15:02.06 | Optic | just needs DHCP and an FTP server |
| 15:02.26 | [TK]D-Fender | lirakis: And no.. the GXP really isn't.. ip 320 is a FEW dollors more for a HUGELY better phone. |
| 15:02.26 | illc0mm | JT: it makes the hobbiest learn, thats why I like it |
| 15:02.26 | JT | lirakis: you could work for the biggest company in the world, it doesn't preclude you from being wrong |
| 15:02.34 | JT | illc0mm: haha learn about what can go wrong? :) |
| 15:02.48 | lirakis | [TK]D-Fender: im putting ip320's into the next call center i am setting up... |
| 15:02.50 | illc0mm | lirakis: yeah, enron was a big company, I wouldn't hire their accountants. :) |
| 15:02.52 | [TK]D-Fender | lirakis: Oh yea... you were working on that non-existant list of the features the GXP hasn't over Polycom! ;) |
| 15:02.54 | Optic | we have 301's, 501's, 500's, 300's, a 4000, and some 430's |
| 15:03.01 | Optic | and I've never had any problems with configuration :) |
| 15:03.03 | lirakis | [TK]D-Fender: .. far to busy lol |
| 15:03.06 | mosty | Optic, snom provisioning is good also |
| 15:03.13 | [TK]D-Fender | lirakis: I'm not blue yet! |
| 15:03.16 | Optic | i've never used a snom phone |
| 15:03.17 | lirakis | [TK]D-Fender: ha ha |
| 15:03.25 | *** join/#asterisk cullenincrease (n=cp@c-75-64-44-200.hsd1.tn.comcast.net) |
| 15:03.27 | mosty | the snom web interface is very good |
| 15:03.28 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 15:03.37 | Optic | ah, i've never used the polycom web interface :) |
| 15:03.37 | cullenincrease | where can i read about the difference between 1.2 and 1.4 i can't find anything online |
| 15:03.39 | mosty | the downside of snom phones is the headsets are useless |
| 15:03.40 | lirakis | Optic: yeah .. the older snoms are kinda .. ehh |
| 15:03.45 | lirakis | Optic: your lucky |
| 15:03.53 | Optic | i don't see the point |
| 15:03.55 | Optic | just use the config files |
| 15:03.59 | [TK]D-Fender | IP320 kills pretty much every non-manager/receptionist phone out there. |
| 15:04.01 | lirakis | exactly |
| 15:04.12 | mosty | cullenincrease, 1.4 has newer features but is buggier at the moment. what are you interested in specifically? |
| 15:04.13 | lirakis | (to optic) |
| 15:04.22 | cullenincrease | which one is best for me |
| 15:04.26 | Rienzilla | I have a snom 360 |
| 15:04.27 | cullenincrease | i'm using it for a 5 agent call center |
| 15:04.31 | Rienzilla | just playing around with it |
| 15:04.44 | lirakis | cullenincrease: i use 1.2 .. i had stability issues with 1.4 |
| 15:04.44 | SwK | anyone from teliax hang out in here? |
| 15:04.47 | Optic | excellent |
| 15:04.51 | cullenincrease | ok cool deal |
| 15:04.55 | Optic | well, next time I order some phones I will get some IP320s :) |
| 15:05.03 | cullenincrease | 1.2 it is! |
| 15:05.04 | mosty | Optic, well one thing i like to do is provision the base settings, and let people customise the other settings on their phone (eg ring tones) |
| 15:05.07 | [TK]D-Fender | Optic: http://www.telephonydepot.com/product_p/105-058-320.htm |
| 15:05.13 | lirakis | Optic: they are cheap .. high qual phones. |
| 15:05.15 | [TK]D-Fender | Optic: $87.5<-----------0 |
| 15:05.20 | Optic | do they come in a 5-pack? |
| 15:05.28 | lirakis | .. maybe i love gxp's so much because its what i started with... ahh.. nostaligia |
| 15:05.39 | lirakis | or maybe.. b/c ive never had a problem with them |
| 15:05.39 | Optic | $87 *and* PoE? |
| 15:05.40 | lirakis | ha ha |
| 15:06.11 | JT | lirakis: the audio on grandstreams is awful |
| 15:06.13 | lirakis | Optic: fyi .. gxp-2000 have that too ;) lol |
| 15:06.25 | lirakis | JT: speaker phone is terrible |
| 15:06.30 | JT | and handset |
| 15:06.37 | lirakis | JT: never had that issue |
| 15:06.48 | Nivex | sweet, I rejoined to glean equipment advice and the topic is immediately there :) |
| 15:07.01 | lirakis | JT: id like to try the BT-200 and the gxp-2020 .. supposed to be higher aoustic quality |
| 15:07.16 | JT | lirakis: it's easy to think a phone sounds good when you haven't heard a better phone |
| 15:07.17 | lirakis | JT: but im not in the market for more personal phones now |
| 15:07.28 | lirakis | JT: .. i used to work at a voip supply store |
| 15:07.30 | JT | lirakis: i'd like to throw all grandstreams in an incinerator ;) |
| 15:07.32 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 15:07.37 | lirakis | JT: ive used.. a lot of phones.. |
| 15:07.39 | JT | lirakis: i really don't care what you worked for |
| 15:07.45 | MrTelephone | any idea why 1 out of 8 channels on the adit 600 won't detect dtmf? |
| 15:07.46 | JT | you must've sold a lot of shite |
| 15:07.59 | lirakis | JT: .. you assumed i hadnt used any thing else.. you assumed wrong |
| 15:08.06 | *** join/#asterisk whatwherewhen (i=whatwher@196.211.34.3) |
| 15:08.14 | JT | lirakis: no, i assumed you'd used nothing *better* |
| 15:08.20 | lirakis | JT: and we sold.. and i used daily a polycom .. |
| 15:08.46 | JT | maybe your hearing range isn't all that great :) |
| 15:09.04 | lirakis | (shrug) .. i doubt it |
| 15:09.25 | lirakis | .. okay.. enough banter.. i do have other work to tend to.. |
| 15:09.28 | JT | also grandstreams are ugly as all hell |
| 15:09.32 | *** join/#asterisk ZeeRoCOOOL (n=ZeRoCoOL@196.203.146.148) |
| 15:09.40 | Optic | mmm polycoms |
| 15:09.41 | ZeeRoCOOOL | good afternoon |
| 15:09.45 | Optic | ok, i need to work again too :) |
| 15:09.53 | *** join/#asterisk ccesario_ (n=ccc@189-19-9-100.dsl.telesp.net.br) |
| 15:11.24 | Nivex | so basically I'm hearing that the Polycom IP 320 are the in thing for a small deployment? |
| 15:11.38 | ZeeRoCOOOL | I have a problem can any one help me please |
| 15:11.46 | JT | Nivex: or 330 if you need an additional ethernet port |
| 15:11.48 | *** join/#asterisk Op3r (n=Op3r@125.212.63.101) |
| 15:12.31 | [TK]D-Fender | Nivex: Though for the price difference you should jsut pay someone to do another ethernet drop. |
| 15:12.41 | Nivex | I was just about to say, ethernet ports are cheap :) |
| 15:12.50 | JT | ethernet drops are not THAT cheap ;) |
| 15:13.47 | MrTelephone | do channel banks transfer dtmf usinb RBS bits? |
| 15:14.01 | Nivex | hmm... PoE only. Will have to figure that into the budget. |
| 15:14.20 | [TK]D-Fender | JT : 25$ for the equipment cost.... |
| 15:14.55 | brodiem | Could someone _please_ have a look at http://pastebin.ca/633202 - I'm having a problem with some remote users at a particular location keeping SIP registrations alive. If you look at this exchange (fromm asterisk doing its sip_poke's) you can see something really weird is happening. To asterisk, it isn't getting these replies. |
| 15:15.02 | *** join/#asterisk `paul (n=aldee@124.107.13.212) |
| 15:15.17 | [TK]D-Fender | Nivex: If you have no PoE and no secondary drop, THEN the ip 430 looks like a choice. Then again, so does the IP 501 for a few bucks more. |
| 15:15.40 | pigpen | When using the n-priority, am I correct that using GoTo's, such as jumping to priority 25 (which is labeled with a n-priority) will still work correctly right? |
| 15:16.22 | pigpen | ie: even though I use goto's, I don't need to keep up the old numbering priority. |
| 15:16.35 | Nivex | [TK]D-Fender: good to know. thanks |
| 15:16.45 | [TK]D-Fender | pigpen: you lose the point of BEING "n". Use labels for your gotos if you insist on using "n" |
| 15:16.46 | blitzrage | pigpen: don't do that -- use labels |
| 15:17.00 | blitzrage | exten => 100,n(my_label),NoOp() |
| 15:17.01 | Nivex | the linksys/sipura phones _look_ pretty sweet, but it sounds like they are still crap |
| 15:17.03 | `paul | can you change entries in extension.conf based on the time of the day? (ie redirect to someone on morning and another one on evenings) |
| 15:17.08 | blitzrage | exten => 100,n,Goto(my_label) |
| 15:17.24 | blitzrage | `paul: yes... use GotoIfTime() |
| 15:17.36 | [TK]D-Fender | `paul: Change entries? Sort-of (time based includes), or use "show application gotoiftime" |
| 15:18.26 | pigpen | k. got it. |
| 15:18.39 | [TK]D-Fender | Nivex: http://www.telephonydepot.com/product_p/105-058-501.htm |
| 15:18.54 | blitzrage | the point of priority 'n' and labels is so that you don't use numbered priorities anymore |
| 15:19.00 | [TK]D-Fender | Nivex: Costs a fair bit more, but your get a better phone for it.... |
| 15:19.21 | JT | [TK]D-Fender: in many commercial environments, the cabling cost is a lot higher than just the equipment cost |
| 15:19.28 | blitzrage | technically you can use the priority number, but why? when you add a line, now you have to change all references to that line |
| 15:19.41 | coppice | Nivex: close up the sipura phones look like junk |
| 15:19.46 | antimoof | my next inane question: if I don't actually use digium hardware, do I still need the zaptel kernel module? |
| 15:20.17 | [TK]D-Fender | JT : how much more is the question, plus consideration for the long term added valu of having it, and the simplification of wiring, reduced points of failure, etc. |
| 15:20.27 | `paul | how do i do time based includes? |
| 15:20.34 | *** join/#asterisk blueneon (n=blueneon@dsl-146-29-190.telkomadsl.co.za) |
| 15:20.37 | [TK]D-Fender | antimoof: You do if you want MeetMe conferencing, or IAX2 trunking |
| 15:20.46 | [TK]D-Fender | `paul: Time to hit the BOOK |
| 15:20.48 | [TK]D-Fender | ~book |
| 15:20.49 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:20.53 | antimoof | why only for those two? the timer bits? |
| 15:21.01 | [TK]D-Fender | antimoof: yup |
| 15:21.01 | illc0mm | Damn, cant get these SIP trunks to work inbound.... |
| 15:21.19 | antimoof | it seems kinda odd that you have to resort to a kernel module to get a software timer - but perhaps that's just me. |
| 15:21.50 | JT | [TK]D-Fender: certified outlets by independent cabling contractors cost around USD$100 here iirc |
| 15:21.55 | mosty | antimoof, userspace software timers probably aren't accurate enough |
| 15:22.10 | blueneon | hi.. on an older version of asterisk whenever i was talking to someone i could press R on the handset and that would allow me to make another call while having the caller on hold, the caller would hear my onhold music while on hold... now that i'm using 1.4.4 when i press R the caller hears absolutly nothing.. any ideas? |
| 15:22.59 | [TK]D-Fender | JT : Damn.... you guys suck :) My prices were like $65 around here.... |
| 15:23.11 | JT | heh |
| 15:23.54 | Nivex | well dang, I just found a 4 PoE port switch for $65. That's not much more than the two PoE adapters :) |
| 15:24.10 | JT | poe adapters are a rip off |
| 15:25.09 | illc0mm | Nivex: where did you find that POE switch at? |
| 15:25.32 | *** join/#asterisk ManxPower (n=manxpowe@015-799-378.area5.spcsdns.net) |
| 15:26.27 | *** join/#asterisk CunningPike (n=arodgers@209.17.159.211) |
| 15:27.02 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 15:27.12 | Nivex | http://www.provantage.com/trendnet-tpe-s44~7TDWE00V.htm |
| 15:27.16 | Nivex | illc0mm: ^^^ |
| 15:27.27 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 15:30.40 | illc0mm | Nivex: thx |
| 15:32.37 | blueneon | anyone able to help me out at all? |
| 15:32.53 | Qwell[] | ~logs |
| 15:32.54 | jbot | extra, extra, read all about it, logs is apt/ibot/infobot/jbot/purl all log daily to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz |
| 15:32.55 | Qwell[] | JT: ^^ |
| 15:33.47 | JT | asterisknerds, QUIT |
| 15:33.57 | [TK]D-Fender | blueneon: "R" is not a button that I know of by another name, and you have not mentioned all of the relevant related hardware. |
| 15:34.21 | sopo2k4 | how to read a variable using SayDigits |
| 15:34.22 | sopo2k4 | ? |
| 15:35.05 | JT | asterisknerds: stop |
| 15:35.55 | [TK]D-Fender | sopo2k4: You don't. SayDigits TALKS. |
| 15:36.02 | [TK]D-Fender | sopo2k4: "show application read" |
| 15:36.47 | *** join/#asterisk yacoob (i=yacoob@hell.pl) |
| 15:36.49 | *** join/#asterisk centrex (i=centrex@nat/digium/x-c453a4be9e95e2da) |
| 15:36.50 | yacoob | Hi there. |
| 15:36.53 | oratelecom | hell |
| 15:37.01 | *** join/#asterisk renier (n=renier@24.138.203.120) |
| 15:37.16 | yacoob | can anyone take a look at http://www.davidpashley.com/blog/2007/07/25#sip-integration (not mine) and tell whether it's feasible? :) |
| 15:37.21 | yacoob | (idea is nice) |
| 15:38.37 | Strom_M | um, how about the simple solution: |
| 15:38.53 | Strom_M | a softphone on the PC that blinks annoyingly when a call comes in |
| 15:39.40 | sopo2k4 | fender i know that |
| 15:39.48 | sopo2k4 | i meant, talk the digits back down the line |
| 15:39.50 | yacoob | that's one solution, but the best would be if you only get a poke on the screen, while the stand alone phone handles the call |
| 15:39.57 | sopo2k4 | i want it to say the digits, stored inside the variable called digits |
| 15:40.04 | [TK]D-Fender | sopo2k4: Well thats what it does. |
| 15:40.12 | Strom_M | sopo2k4: SayDigits(${digits}) |
| 15:40.17 | sopo2k4 | cheers strom |
| 15:40.20 | sopo2k4 | thats what i was looking for |
| 15:40.33 | [TK]D-Fender | sopo2k4: Go read the chapter on using VARIABLES now :) |
| 15:40.49 | [TK]D-Fender | yacoob: Doable, but will require programming. |
| 15:40.52 | Strom_M | sopo2k4: at the CLI, type "core show application SayDigits" |
| 15:41.30 | sopo2k4 | ok let me see :P |
| 15:41.38 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 15:41.42 | yacoob | [TK]D-Fender, that's not a problem, I think. Where would the "logic" sit, at the asterisk server as some extension, or outside of it, polling the state or getting notifications? |
| 15:43.42 | Qwell[] | ~channels |
| 15:43.49 | yacoob | http://www.voip-info.org/wiki/view/ADM+-+Asterisk+Desktop+Manager hm :> |
| 15:44.00 | neverblue2 | any VOIP providers in the channel today? looking to try someones service |
| 15:44.01 | blueneon | if i use Dial(ZAP/1,180,m) there is on-hold music until 1 is answered, but if i put the caller on hold via the handset using the (R) button the caller hears silence instead of onhold music, what am i doing wrong? |
| 15:44.34 | [TK]D-Fender | yacoob: You'd setup a small Java applet that listens to AMI for dial-outs to your phone. It would then mess with your mixer and do a pop-up. |
| 15:45.41 | jsbach | if i send an invite with From: "bob" <sip:bob@blablubber.de> , asterisk changes to "bob" <sip:asterisk@blablubber.de> as it relays to callee.. how can i tell asterisk not to touch a "From" header field? |
| 15:45.42 | Strom_M | blueneon: is it an analog phone? |
| 15:45.44 | [TK]D-Fender | blueneon: You have STILL not told us what ahrdware you are using. |
| 15:46.54 | yacoob | [TK]D-Fender, right. Thanks. As for now I don't have even a test environment, but it looks promising :) |
| 15:47.07 | blueneon | yes its an analog phone. but this used to work fine with the older asterisk |
| 15:47.23 | Strom_M | blueneon: what does the CLI say when you put the call on hold? |
| 15:47.27 | blueneon | im using a digium tdm / zap |
| 15:48.05 | MrTelephone | one way static on a channel bank, could it be a punch down? |
| 15:48.13 | blueneon | the cli just shows an (R) and i get dial tone (and can dial out).. the caller is put on hold, but hears silence until i press (R) again |
| 15:48.26 | Strom_M | blueneon: no... |
| 15:48.29 | Strom_M | blueneon: the asterisk CLI |
| 15:48.34 | blueneon | oh my bad |
| 15:48.34 | Strom_M | not the display on the phone |
| 15:48.35 | blueneon | sec |
| 15:48.49 | Strom_M | so R probably stands for "recall" |
| 15:49.52 | [TK]D-Fender | blueneon: Go prove that your MoH is even setup right in this upgrade of yours |
| 15:50.13 | Strom_M | [TK]D-Fender: stuff a sock in it |
| 15:50.49 | blueneon | [TK]D-Fender: it does work, like i said if i use Dial(ZAP/1,180,m) music is heard etc |
| 15:51.00 | tzanger | [TK]D-Fender: you're getting a lot of flack lately... did you sleep with these guys wives or something? |
| 15:51.06 | blueneon | Strom_M: im just trying to get pastebin working |
| 15:51.08 | blueneon | :/ sec |
| 15:51.16 | yacoob | [TK]D-Fender, just a question to be sure: how much "invasive" such method is, as is how much rights do I need to achieve this (on the Asterisk)? This ADM thingy (http://adm.hamnett.org/?q=node/5) requires... full access? |
| 15:51.20 | DrukenLPY | anyone have experince with asterisk now? |
| 15:51.32 | Qwell[] | I had some experience with asterisk earlier today |
| 15:51.38 | yacoob | (yeah, I know, RTFM :) |
| 15:51.41 | blueneon | Strom_M: http://pastebin.ca/633247 |
| 15:51.47 | DrukenLPY | asterisknow smart ass |
| 15:51.52 | DrukenLPY | :) |
| 15:51.57 | [TK]D-Fender | yacoob: Non-invasive. Setup an AMI proxy and let this account have listen-only priviledges |
| 15:52.00 | Strom_M | blueneon: pastebin your musiconhold.conf |
| 15:52.10 | blueneon | when i press the recall button on the handset i get a dialtone, the caller is put on silence hold and i get that in the CLI |
| 15:52.13 | blueneon | ok |
| 15:52.31 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:52.36 | yacoob | [TK]D-Fender, uhu. Well, we'll see about that :) Thanks! |
| 15:52.47 | ManxPower | blueneon: you get what on the CLI? |
| 15:52.49 | *** join/#asterisk enioreh (n=enioreh@core.kahmm.net) |
| 15:52.50 | blueneon | http://pastebin.ca/633248 |
| 15:52.52 | Strom_M | tzanger: I think we're all getting tired of [TK]D-Fender's childish, irascible nonsense |
| 15:53.05 | blueneon | thats my musiconhold.conf |
| 15:53.17 | tzanger | Strom_M: actually I rather like [TK]D-Fender's banter. |
| 15:53.25 | Strom_M | tzanger: it gets old quickly |
| 15:53.28 | [TK]D-Fender | Strom_M: More sock, sir? :) |
| 15:53.36 | ManxPower | blueneon: I assume you are using analog phone connected into a digium analog card? |
| 15:53.37 | blueneon | ManxPower: http://pastebin.ca/633247 i get this on the asterisk console |
| 15:53.41 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 15:53.53 | blueneon | ManxPower: correct |
| 15:53.54 | tzanger | I imagine after months and months of helping out here he gets tired of the same core communication issues people have with explaining their problems |
| 15:54.03 | tzanger | I know I do... one tends to get... impatient |
| 15:54.12 | ManxPower | blueneon: you need your music on hold class BEFORE the channel line in /etc/asterisk/zapata.conf |
| 15:54.25 | blueneon | ? |
| 15:54.33 | Mercestes | I think he models his online personality after the people described in "how to ask a smart question." |
| 15:54.34 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-ac4efe16ec5fccb4) |
| 15:54.35 | ManxPower | tzanger: I call it "support burnout" |
| 15:54.39 | Mercestes | I know I do |
| 15:54.42 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 15:54.42 | [TK]D-Fender | ManxPower: Beware, I may be approaching your level of BOFH-ness! |
| 15:54.44 | enioreh | hi |
| 15:54.49 | blueneon | i have musiconhold=default for each channel listed in zapata |
| 15:54.54 | ManxPower | [TK]D-Fender: I am so proud! |
| 15:54.57 | tzanger | ManxPower: I call it crotchety old man... I'm in training for when I retire |
| 15:55.00 | Mercestes | [TK]D-Fender, In your dreams. no one out BOFH's Manx. |
| 15:55.02 | [TK]D-Fender | ManxPower: as I knew you would be! |
| 15:55.05 | enioreh | has somebody successfully used the manager to hangup some calls ? |
| 15:55.05 | tzanger | already have my lawn chair and my shaking onion |
| 15:55.08 | tzanger | dang kids |
| 15:55.24 | [TK]D-Fender | enioreh: Yup |
| 15:55.36 | enioreh | I get a reponse message telling me that the action was successfull but the call is not hung up :/ |
| 15:55.37 | ManxPower | blueneon: pastebin your zapata.conf |
| 15:55.40 | tzanger | my finger-waggling course hasn't been approved yet |
| 15:55.59 | Mercestes | tzanger, =/ |
| 15:56.07 | ManxPower | [TK]D-Fender: When users behave I'm not much of a BOFH. |
| 15:56.08 | renier | hey. looking for a pointer to avoid having 3 second empty voicemail when there is a missed call. Is there a way to filter this automagically? |
| 15:56.12 | enioreh | [TK]D-Fender: is there anything to do after sending the Hangup action ? |
| 15:56.20 | Mercestes | tzanger, Didn't Hugh Heffner make a video training series for that? |
| 15:56.22 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:56.25 | ManxPower | When they don't behave, I have found that it is best to cause them pain until they do behave. |
| 15:56.35 | tzanger | Mercestes: that's a different kind of waggling |
| 15:56.43 | Mercestes | there's another kind? |
| 15:56.48 | ManxPower | renier: yes, see the min voicemail length in voicemail.conf |
| 15:56.53 | [TK]D-Fender | enioreh: I never used that method directly. I always issue CLI through AMI to do "soft hangup [channel]" |
| 15:56.58 | tzanger | renier: there are options in voicemail.conf for that... minmsglen and such |
| 15:56.59 | renier | ManxPower: ah, thnx |
| 15:57.00 | blueneon | ManxPower: http://pastebin.ca/633255 |
| 15:57.33 | enioreh | [TK]D-Fender: ok, i ll check that |
| 15:58.26 | enioreh | damn, it works well in the asterisk cli |
| 15:59.04 | [TK]D-Fender | enioreh: use the COMMAND AMI call. |
| 15:59.18 | [TK]D-Fender | enioreh: you can do a LOT with that + text parsing |
| 15:59.28 | enioreh | [TK]D-Fender: right, thanks for the tip :) |
| 15:59.43 | Voicemeup | can asterisk listen or bind on 2 ports ? |
| 16:00.09 | illc0mm | Damn, I'm going crazy with this SIP Trunking issue |
| 16:00.15 | blueneon | ManxPower: see anything wrong in my zapata? |
| 16:00.15 | [TK]D-Fender | Voicemeup: it can bind to a single, or all, but not multiple individual |
| 16:00.21 | ManxPower | blueneon: It looks right to me. |
| 16:00.24 | blueneon | hmm |
| 16:00.34 | blueneon | i wonder why its not working :( |
| 16:00.42 | ManxPower | illc0mm: There is no such thing as a "sip trunk" in Asterisk. That might be why you are having problems. |
| 16:00.45 | Voicemeup | like some providers (cable) are blocking 5060 now to block competition and was wondering the easiest way to proxy this |
| 16:00.46 | illc0mm | haha |
| 16:00.47 | Voicemeup | as 15060 |
| 16:01.02 | Voicemeup | rinetd is tcp .. so id need a udp proxy right ? |
| 16:01.11 | illc0mm | ManxPower: right, it's not technically a SIP trunk, but that's what Bandwidth.com calls it |
| 16:01.14 | Voicemeup | or amke it listen on 2 ports |
| 16:01.17 | ManxPower | And calling it a sip trunk makes think you rode the short bus to VoIP School |
| 16:01.28 | illc0mm | yeah, well, I probably did |
| 16:01.39 | illc0mm | how about SIP "trunk" |
| 16:01.44 | ManxPower | illc0mm: I don't care. Use the terms we use or you will have problems getting people to understand you and help you. |
| 16:02.01 | illc0mm | wow, such hostility. simple mistake |
| 16:02.01 | Voicemeup | sip gateway |
| 16:02.08 | Voicemeup | sip peer |
| 16:02.10 | Voicemeup | sip user |
| 16:02.12 | ManxPower | illc0mm: How about "sip connection" or "sip account" or |
| 16:02.16 | illc0mm | there you go |
| 16:02.27 | ManxPower | illc0mm: FreePBX and their ilk call it "sip trunk" |
| 16:02.32 | illc0mm | okay okay |
| 16:02.35 | illc0mm | enough of that |
| 16:02.54 | Voicemeup | yeah .. lol and hteyr default ocnfig makes the inbound part use from -pstn instead of from-trunk |
| 16:02.54 | Voicemeup | lol |
| 16:02.56 | illc0mm | seriously, I'm just calling it what the provider called it, mostly looking for someone using the same provider |
| 16:03.03 | coppice | define a trunk. its a pretty vague term |
| 16:03.27 | Voicemeup | he main structural member of a tree. |
| 16:03.29 | ManxPower | coppice: It is the main piece of a tree |
| 16:03.32 | illc0mm | yes |
| 16:03.34 | Voicemeup | a chute or conduit, or a watertight shaft connecting two or more decks. |
| 16:03.42 | illc0mm | my VOITree is not working |
| 16:03.46 | Voicemeup | but.. in telecommunications, has a number of closely related meanings. |
| 16:03.50 | ManxPower | It is also a type of connection to the telco using analog or T-1 technology (circuit switched) |
| 16:03.57 | Voicemeup | http://en.wikipedia.org/wiki/Trunking |
| 16:04.00 | coppice | ManxPower: I thought it was an elephant's nose |
| 16:04.14 | illc0mm | I'm glad we can get a course on semantics here, it's very helpful. Please continue. :) |
| 16:04.19 | ManxPower | coppice: shows what you know. *tease* |
| 16:04.27 | Voicemeup | ok well in 1912 the term trunk... |
| 16:04.28 | Voicemeup | j/k |
| 16:04.32 | coppice | oh, gosh, wikipedia has a definition. we have an authoritative answer :-) |
| 16:04.40 | illc0mm | coppice: haha, I'll fix that |
| 16:04.50 | Voicemeup | authored by 1000 noobs on a moonless night |
| 16:04.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:05.02 | illc0mm | okay, so here is the situation |
| 16:05.20 | illc0mm | bandwidth.com, provides a service called "sip trunk" which is just a bunch of sip accounts |
| 16:05.21 | ManxPower | illc0mm: it may seem silly, but is like going to a mechanic and saying "the catalytic converter is not working" when you really mean "My muffler has a hole in it" |
| 16:05.38 | ManxPower | the mechanic will waste time and money trying to fix the wrong problem. |
| 16:05.55 | ManxPower | and since we do this support for FREE, one of the rudest things you can do is waste our time. |
| 16:06.02 | illc0mm | ManxPower: I get it, I don't need 30 lines saying the same thing. I'm not being ungrateful but I understand. |
| 16:06.25 | illc0mm | ManxPower: I'm over that part, understood, 10-4, roger, over. |
| 16:07.05 | coppice | ManxPower: if you don't want to waste your time, WTF are you doing here? :-) |
| 16:07.13 | drako | How I can see active calls and hang one? |
| 16:07.22 | Strom_M | drako: "show channels" |
| 16:07.29 | Strom_M | and "soft hangup [channelname]" |
| 16:09.04 | illc0mm | Simple mistake, it doesn't help to have 30 people jump in and repeat it, seriously. I give out free support too for other projects, and I don't have that attitude. If you don't want to help, fine, but I don't need the ubiquitous open source lecture of how you're donating your time and how we must follow the proper protocol, lest I get bashed and called a noob. I understand everyone's time is valuable, and I think all of our times would |
| 16:09.34 | Strom_M | illc0mm: this seems to be "irascible bitch morning" here in #asterisk |
| 16:09.40 | Strom_M | i wouldn't take it too hard |
| 16:10.00 | Strom_M | it's usually not quite this bad |
| 16:10.10 | *** join/#asterisk NirS_ (n=Nir@87.68.232.33.adsl.012.net.il) |
| 16:10.27 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 16:10.37 | illc0mm | Strom_M: I'm not, but I see this stuff all the time and there is no need for it. I'm just as invested in this stuff as everyone else, a community supported product is only as good as the community that supports it. If we don't have that, we're no better than the "other" guys. |
| 16:10.41 | enioreh | [TK]D-Fender: Thank for you tip , it worked perfectly :) |
| 16:10.51 | [TK]D-Fender | enioreh: You're welcome |
| 16:12.24 | ManxPower | coppice: Don't interrupt my diatribe with logic! |
| 16:13.50 | twitchnln | <PROTECTED> |
| 16:14.16 | twitchnln | or can i? |
| 16:14.44 | ManxPower | I dunno. I don't bill for calls 8-) |
| 16:16.05 | *** join/#asterisk Jiboneus (n=Jibone@60.54.54.71) |
| 16:17.11 | Strom_M | twitchnln: try looking into the ACCOUNTCODE variable |
| 16:17.11 | illc0mm | twitchnln: you using CDR? |
| 16:17.18 | *** join/#asterisk j-goddess (n=humblein@phrank.aus.us.siteprotect.com) |
| 16:18.28 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:19.01 | killfill | hi.. |
| 16:19.11 | killfill | is it my connection? |
| 16:19.15 | killfill | cannot get 1.4.9 |
| 16:19.17 | killfill | Opening fileinfo database failed |
| 16:19.24 | killfill | same with the vul pdf's... |
| 16:20.38 | killfill | actually.. |
| 16:20.43 | killfill | its the http |
| 16:20.55 | killfill | htp://ftp.digium.com has an issue |
| 16:21.26 | neverblue2 | any VOIP providers in the channel today? looking to try someones service |
| 16:21.26 | Qwell[] | bbryant: ^^? |
| 16:22.02 | [TK]D-Fender | killfill: user the secondary... |
| 16:22.22 | [TK]D-Fender | neverblue2 : Just contact one directly. |
| 16:22.31 | killfill | secondary? |
| 16:22.44 | neverblue2 | [TK]D-Fender, do you have an issue with me asking in the channel? |
| 16:22.49 | [TK]D-Fender | killfill: asterisk.org has a few mirros up |
| 16:22.53 | Qwell[] | ftp1 appears to be having dns problems also... |
| 16:23.08 | killfill | ftp2 too :P |
| 16:23.13 | Voicemeup | http://en.wikipedia.org/wiki/Comparison_of_VoIP_software |
| 16:23.13 | Voicemeup | nice list |
| 16:23.15 | [TK]D-Fender | neverblue2: You've asked repeatedly with no apparent answer and you could get one if you just tried the direct approach. |
| 16:23.37 | neverblue2 | [TK]D-Fender, you didnt answer my question :) |
| 16:24.04 | [TK]D-Fender | neverblue2: Ask a the same question a few MORE times and ask me that again ;) |
| 16:24.12 | Strom_M | neverblue: [TK]D-Fender is 25% helpful and 75% irascible bitch. What do you think? |
| 16:24.19 | [TK]D-Fender | neverblue2: Right now I'm just guiding the lost :) |
| 16:24.45 | neverblue2 | like 90/10 |
| 16:24.46 | MrTelephone | being a phone provider is stressful |
| 16:24.47 | neverblue2 | lmao |
| 16:24.53 | neverblue2 | i mean 10/90 :/ |
| 16:24.55 | [TK]D-Fender | Strom_M: You seem rather irked today give the massive volume of help I disperse here..... |
| 16:25.11 | Voicemeup | sorry for pm |
| 16:25.14 | Voicemeup | lol |
| 16:25.36 | neverblue2 | hey Voicemeup :) |
| 16:25.36 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 16:25.44 | Strom_M | [TK]D-Fender: that still doesn't give you a good reason to behave like a six year old half the time :) |
| 16:25.44 | bbryant | killfill, should be fixed, sorry |
| 16:25.44 | *** join/#asterisk KpoH (n=AID@host-89-41-66-13.moldtelecom.md) |
| 16:25.59 | Voicemeup | hey , got your refund ok ? |
| 16:26.06 | killfill | thanks! |
| 16:26.58 | neverblue2 | Voicemeup, i havent got an email about it |
| 16:27.12 | funkmaster | hi there ppl :) |
| 16:27.20 | neverblue2 | Voicemeup, i could check with my man. and see if the CC has the refund :) |
| 16:27.29 | Voicemeup | yeah should be on its way |
| 16:27.38 | funkmaster | i got a problem with the config of one of my sip providers, when ppl call me they can hear me but i can not hear them |
| 16:28.00 | Voicemeup | billing dept taking are of it.. oh .. and you got a bye problem, if you don't fix htat first youll never get what oyur looking for |
| 16:28.02 | funkmaster | i was wondering if someone could take a look at my sip.conf and extensions.conf and help me out alittle please? |
| 16:28.26 | funkmaster | the provider i have this problem with is sipgate.de |
| 16:28.30 | [TK]D-Fender | Strom_M: I'm quite far from that level thank-you, but I do know some who definately apply. |
| 16:28.36 | funkmaster | http://pastebin.ca/633302 there is my sip and extensions.conf |
| 16:28.38 | Voicemeup | ;) |
| 16:28.55 | neverblue2 | bye? |
| 16:29.02 | twitchnln | funkmaster: is it just one provider or multiple? because if it's multiple providers, then it sounds like a fw issue |
| 16:29.05 | Rienzilla | hmmm |
| 16:29.08 | funkmaster | just the one |
| 16:29.10 | Voicemeup | n/m |
| 16:29.13 | Rienzilla | is app_conference supposed to work? :) |
| 16:29.17 | *** join/#asterisk CunningPike (n=arodgers@209.17.159.211) |
| 16:29.28 | neverblue2 | we are working fine with our current providers, so for me looking for others, I will try :) |
| 16:30.22 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:31.50 | MrTelephone | someone needs a provider? |
| 16:32.08 | funkmaster | a free one? |
| 16:32.19 | funkmaster | g |
| 16:32.23 | lirakis | so .. i missed the flame/conversation... what do people say is the real name for a "sip trunk" |
| 16:32.27 | MrTelephone | how do you go about getting local area codes tho |
| 16:32.47 | funkmaster | for which country? |
| 16:32.52 | MrTelephone | canada? |
| 16:33.05 | funkmaster | u have already a provider? |
| 16:33.15 | *** join/#asterisk Modcuts (n=modcuts@lan.proporta.com) |
| 16:33.18 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:33.18 | MrTelephone | i need to learn more about what it takes to be a provider |
| 16:33.28 | neverblue2 | MrTelephone, i was looking for a provider |
| 16:33.30 | funkmaster | usually u can choose on their site, if they don't offer it, it means they just give u a voip number... |
| 16:33.44 | MrTelephone | every call you would make from my system would cost 4 cents a minute |
| 16:33.50 | funkmaster | MrTelephone: being a provider is not difficult |
| 16:33.51 | Modcuts | Does anybody know if there is any wifi phones that actually work well on the market? |
| 16:33.52 | j-goddess | yea but doesn't telus monopolize everything in canada =P |
| 16:33.56 | j-goddess | besides voip providers |
| 16:34.03 | MrTelephone | how are companies getting dids in different areas? |
| 16:34.07 | Voicemeup | telus sucks |
| 16:34.13 | Voicemeup | rogers too |
| 16:34.19 | funkmaster | MrTelephone: depends on the country |
| 16:34.21 | j-goddess | that is the resp orgs |
| 16:34.23 | Voicemeup | BTW rogers is closing the wholesale market |
| 16:34.26 | MrTelephone | im working in a town and i have 60% of the people moved over to our system |
| 16:34.32 | Voicemeup | got a meeting tomorow regarding this.. |
| 16:34.32 | funkmaster | not all countries offer or allow reginal dids |
| 16:34.33 | j-goddess | they are the ones in charge of distributing numbers to companies |
| 16:34.42 | Voicemeup | they gonna only do retail now |
| 16:34.56 | Voicemeup | other big ones to follow .. GT etc |
| 16:34.58 | MrTelephone | you need to be a true competitor to get the services cheap enough |
| 16:35.14 | funkmaster | u just wanna serve canadian numbers? |
| 16:35.17 | MrTelephone | bell monopolizes in ontario |
| 16:35.46 | j-goddess | so you would get the numbers from whomever you get your PRI from |
| 16:35.46 | MrTelephone | if I could get a 1000 customers i'd be driving a porsche to work |
| 16:35.50 | Voicemeup | BELL MCE got sold for 56 billion |
| 16:35.53 | j-goddess | nice |
| 16:35.58 | funkmaster | is anyone using sipgate.de? |
| 16:35.59 | j-goddess | you going to pick me up on the way there? |
| 16:36.01 | Voicemeup | and they want a profit in 5 years. so im sure they will cut down on all |
| 16:36.03 | MrTelephone | why not |
| 16:36.06 | MrTelephone | :P |
| 16:36.11 | Voicemeup | and outsource even more of the sales/support to 3rd coutnried |
| 16:36.19 | lirakis | sip outbound channel? |
| 16:36.22 | MrTelephone | my pri provider will give me out of area dids? |
| 16:36.33 | j-goddess | depends on what they have the permission to do |
| 16:36.35 | MrTelephone | no way then i could just ask for a number in every town so i don't have to pay long distance |
| 16:36.36 | Voicemeup | telephone service is getting into a big fight game.. with big pockets trying to rule the world |
| 16:36.36 | tzanger | MrTelephone: sure, if you're willing ot pay IX charges |
| 16:36.52 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 16:36.52 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 16:37.06 | MrTelephone | how are companies selling LD for 1 cent a minute? thats friggin crazy |
| 16:37.11 | j-goddess | hi Corydon76-work :) |
| 16:37.16 | j-goddess | haha |
| 16:37.17 | j-goddess | yeah it is |
| 16:37.30 | j-goddess | I laugh at all the cable companies that sell phone server for 40 or 50 bucks |
| 16:37.41 | MrTelephone | im selling dialtone for 25 bucks/month |
| 16:37.46 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 16:37.47 | coppice | Voicemeup: the world is very pro-robber-baron right now. what do you expect? |
| 16:37.56 | Voicemeup | flowers and peace |
| 16:37.56 | j-goddess | because you know at their office it is asterisk and I want to barf just thinking about paying that much |
| 16:38.02 | j-goddess | hey candy |
| 16:38.06 | sweeper | MrTelephone: eh, clec I used to work for got LD at .1 cents a minute |
| 16:38.07 | j-goddess | come on chocolate |
| 16:38.12 | Voicemeup | heck this is why we concentrate on B2B solutions and not mainstream retail |
| 16:38.12 | MrTelephone | well i looked into getting a carrier class switch and the cheapest one was 100K |
| 16:38.18 | asterisknerds | <PROTECTED> |
| 16:38.20 | Voicemeup | look as sunrocket and allo.com |
| 16:38.24 | Voicemeup | both went belly up in last week |
| 16:38.36 | MrTelephone | so I'll become a clec but then you need a facility to get true cost cuts |
| 16:38.50 | MrTelephone | whats b2b? |
| 16:38.56 | Voicemeup | Biz 2 Biz |
| 16:39.01 | sweeper | or find a clec that will sell to you for .5 cents :D |
| 16:39.03 | Voicemeup | B2C = biz 2 consumer |
| 16:39.04 | Voicemeup | etc |
| 16:39.12 | coppice | I think there's a carrier class carrier in the harbour outside. maybe the US navy could offer you a deal on that :-) |
| 16:39.28 | sweeper | coppice: is is OVER NINE THOUSAND?!?!?!? |
| 16:39.50 | MrTelephone | carrier class softswitch that supports packetcable technology |
| 16:40.08 | MrTelephone | the best way to do voip for us is cable modems because they do QOS and are never Natted |
| 16:41.04 | coppice | cable modems are shared media devices, and can never guarantee their QoS |
| 16:41.48 | MrTelephone | docsis 1.1 |
| 16:41.56 | MrTelephone | Unsolicted grant service flows |
| 16:41.57 | *** join/#asterisk ryant (n=ryant@4.17.197.118) |
| 16:42.04 | ryant | hello all |
| 16:42.25 | ryant | I've got a question about using asterisk to bridge modem traffic |
| 16:42.38 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:42.42 | MrTelephone | when you pick up the phone the modem sends a qos request to the ubr and opens up a seperate service flow for that traffic, its awesome |
| 16:42.45 | ryant | ANY way to bridge two analog lines/modems with asterisk over a data network |
| 16:42.46 | KpoH | does ChanIsAvail(SIP/peer@host) whould work? I meant with remoute host |
| 16:43.09 | sweeper | NO INTERNET FOR A WEEK~ |
| 16:43.31 | MrTelephone | if u hate rogers you can take a node down by generating a 40db 0-42mhz garbage signal on your coax |
| 16:43.45 | MrTelephone | but don't do it for more than 30 minutes at a time :-/ |
| 16:44.00 | sweeper | I think 110v AC would be more effective |
| 16:44.08 | sweeper | AND I have adaptors for AC -> coax |
| 16:44.21 | MrTelephone | taps don't allow ingress voltage |
| 16:44.33 | MrTelephone | or they shouldn't |
| 16:44.35 | MrTelephone | :-/ |
| 16:44.41 | sweeper | probably true! |
| 16:44.51 | ryant | can anyone help me? |
| 16:44.51 | coppice | of course they do. you need to try harder |
| 16:45.04 | MrTelephone | trunk amps amplify reverse modem traffic until it reaches the head end |
| 16:45.06 | sweeper | well, I DO have that flyback transformer lying around... |
| 16:45.27 | *** part/#asterisk Modcuts (n=modcuts@lan.proporta.com) |
| 16:46.58 | coppice | that's the spirit. car ignitions work well too |
| 16:47.04 | renier | ok, another voicemail question. is it possible to have a voicemail greeting depending on the time of day? |
| 16:47.05 | KpoH | people, how about ChanIsAvail(SIP/peer@host)? will it work with remote host? |
| 16:49.59 | kolian123 | yes |
| 16:50.49 | *** join/#asterisk rhombus (i=user13@74.12.124.179) |
| 16:51.10 | rhombus | Is anyone else having trouble downloading from the Digium FTP? |
| 16:51.41 | rhombus | sorry, it's up again -- finally |
| 16:52.59 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 16:53.06 | Rienzilla | hmm |
| 16:53.47 | Rienzilla | any ideas why I cannot get app_conference to work? I compiled the module, and added a conference to my dialplan. I can join the conference fine, but conference memebers cannot hear each other... |
| 17:01.35 | Corydon76-work | You might want to try app_meetme |
| 17:01.35 | mvanbaak | anything in the logs ? |
| 17:01.35 | mvanbaak | app_meetme needs zaptel timer. That's not available in all setups |
| 17:01.35 | MrTelephone | mvanbaak, i took a look at that authlibmysql and i'll need to improve my c skills before i tackle that |
| 17:01.35 | Corydon76-work | It's actually not the timer, but the mixer. But anyway |
| 17:01.36 | Rienzilla | I need a hardware timer or some kernel module in order to use meetme |
| 17:01.36 | Rienzilla | uhm |
| 17:01.36 | Rienzilla | logs say a couple of suspicious things |
| 17:01.36 | Rienzilla | do we have a pastebin here? |
| 17:01.36 | Corydon76-work | ~pb |
| 17:01.43 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:01.43 | mvanbaak | MrTelephone: :) |
| 17:01.43 | Corydon76-work | and if the bot would answer... |
| 17:01.43 | mvanbaak | http://pastebin.ca |
| 17:01.43 | Rienzilla | http://pastebin.com/m53e9898a |
| 17:01.44 | Rienzilla | i' |
| 17:01.44 | Rienzilla | I'm concerned about the translator path warnings, and maybe the 'unanticipated delivery time' is something wrong |
| 17:01.44 | Rienzilla | need any more info? |
| 17:01.44 | Rienzilla | (i.e. sip.conf or extensions.conf?) |
| 17:02.25 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 17:02.42 | ccesario_ | somebody have idea about howto fix this error ? http://pastebin.ca/633341 |
| 17:02.53 | mvanbaak | no idea Rienzilla |
| 17:03.01 | Rienzilla | hmm ok :( |
| 17:03.12 | pj_ | no idea either |
| 17:03.50 | *** join/#asterisk zpertee (n=zach@oh-69-34-21-229.sta.embarqhsd.net) |
| 17:04.11 | zpertee | has anyone tried any of the digium clone cards? |
| 17:04.19 | Qwell[] | ~cheap |
| 17:04.59 | jbot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 17:05.30 | *** part/#asterisk Strom_M (n=strom@h72-2-22-215.bigpipeinc.com) |
| 17:05.35 | *** join/#asterisk Strom_M (n=strom@h72-2-22-215.bigpipeinc.com) |
| 17:05.36 | Strom_M | oops |
| 17:06.12 | mvanbaak | ~gs |
| 17:06.40 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 17:06.40 | Strom_M | heh |
| 17:06.53 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 17:06.57 | brettnem | hey all |
| 17:07.18 | Corydon76-work | ~cisco |
| 17:07.19 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 17:07.45 | kolian123 | ~asterisk |
| 17:07.46 | jbot | somebody said asterisk was the best free PBX in the world |
| 17:07.47 | brettnem | hey anyone using AMD in asterisk 1.4? |
| 17:08.52 | *** part/#asterisk yacoob (i=yacoob@hell.pl) |
| 17:09.10 | brettnem | anyone... anyone.. ?? |
| 17:09.14 | kolian123 | ~ibm |
| 17:10.06 | jbot | well, ibm is International Business Machines - a very spiffy company - who happens to make AIX, OS/400, OS/2, and other cool operating systems, not to mention some of the most superior hardware on the market.. I Blame Microsoft. I Buy Macintosh |
| 17:10.22 | rhombus | is anybody from Digium here? |
| 17:10.27 | Mercestes | ~mercestes |
| 17:11.06 | jbot | mercestes is definitely a total nub |
| 17:11.07 | Qwell[] | rhombus: several |
| 17:11.07 | rhombus | I can't pull packages from the digium ftp server with wget |
| 17:11.07 | rhombus | I get 404s, even though the URLs work in other http clients |
| 17:11.07 | Qwell[] | tried in the last 20 minutes? |
| 17:11.07 | Mercestes | Did I misspell my own name??? |
| 17:11.07 | rhombus | I am bringing this up because it started shortly after the server failure |
| 17:11.07 | rhombus | yeah |
| 17:11.07 | rhombus | I did it just 90 seconds ago |
| 17:11.10 | Mercestes | or did jbot finally add me to his ignore list? |
| 17:11.13 | Mercestes | oh! There he goes... |
| 17:11.24 | Mercestes | He was having performance anxiety |
| 17:11.30 | ryant | isn't anyone using modem/data connections via asterisk/VoIP? for briding a broken analog line, etc? |
| 17:11.54 | Mercestes | ~8ball Will I get laid by the cleaning ladies? |
| 17:12.43 | jbot | Unsure. |
| 17:12.46 | Qwell[] | bbryant: ^^? It's unhappy |
| 17:12.46 | brettnem | blah |
| 17:12.46 | blitzrage | ryant: perhaps... but modem/fax doesn't work well over a packetized network |
| 17:13.11 | sweeper | blitzrage: but we have this awsome t.38 thing! |
| 17:13.30 | blitzrage | ya... but that doesn't work the same way :) |
| 17:13.35 | waKKu | sweeper r u using t.38 with an ATA ? |
| 17:13.46 | sweeper | waKKu: all signs point to no |
| 17:13.52 | waKKu | i had tried use it with linksys pap2na but with no succss |
| 17:14.15 | rhombus | Qwell[]: what happened to the ftp? |
| 17:14.20 | Qwell[] | dunno |
| 17:14.23 | rhombus | http://pastebin.ca/633378 |
| 17:14.43 | bbryant | rhombus, try again in 60 seconds |
| 17:14.54 | rhombus | bbryant: okay |
| 17:14.56 | Qwell[] | I wonder what happened to ftp1... |
| 17:15.03 | Qwell[] | it like...disappeared |
| 17:15.15 | waKKu | I give up of t.38 from pap2na and now i'm using hylafax + iaxmodem + winprint hylafax that's working perfectly ;) |
| 17:15.37 | bbryant | Qwell, they took the other server off of the round robin |
| 17:15.42 | Qwell[] | ahh |
| 17:16.28 | ryant | hylafax rocks, can't wait to set it up again |
| 17:16.51 | waKKu | yeah.. :) |
| 17:16.58 | rhombus | What does "Mercestes" mean? |
| 17:17.03 | waKKu | mail instead of lots of papper ownz;D |
| 17:17.15 | Mercestes | rhombus, Long story. |
| 17:17.28 | rhombus | put it in pastebin! :P |
| 17:17.29 | Mercestes | In short it's an Internet character I created a long time ago for an online RPG. |
| 17:17.41 | rhombus | Ah. Does it have anything to do with a Mercedes? |
| 17:17.43 | Mercestes | I kinda liked the character so it became an online persona. |
| 17:17.44 | blitzrage | *coughnerdcoughcough* |
| 17:17.47 | Mercestes | not a thing. |
| 17:17.51 | rhombus | Darn. |
| 17:18.03 | Mercestes | I'm not a nerd...I'm a geek. I have a social circle. |
| 17:18.05 | rhombus | You should create an Internet character that does. |
| 17:18.13 | rhombus | I'm a nerd, and I have a social circle too. |
| 17:18.18 | Mercestes | Mercedes Lackey? |
| 17:18.28 | blitzrage | Geek is to Star Wars as Nerd is to Star Trek |
| 17:18.36 | rhombus | bbryant: It worked. What did you do? |
| 17:18.37 | Mercestes | Nah. |
| 17:18.41 | Mercestes | It's one nerd and several geeks. |
| 17:18.48 | rhombus | Mercestes: Wasn't she in "The Fisher King"? |
| 17:19.00 | Mercestes | I thought she was a writer. |
| 17:19.14 | rhombus | Mercestes: oh. THAT Mercedes :D |
| 17:19.21 | Mercestes | lol |
| 17:19.26 | bbryant | rhombus, added a 200 OK header |
| 17:19.31 | coppice | iaxmodem seems to be working pretty well, now. just a few more quirks to fix |
| 17:19.34 | rhombus | Mercedes Ruehl was in "The Fisher King". |
| 17:20.50 | rhombus | Thanks, bbryant -- and thanks everyone else for the entertainment. Back to work for me. |
| 17:20.54 | *** part/#asterisk rhombus (i=user13@74.12.124.179) |
| 17:21.49 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-221-126.dsl.irvnca.pacbell.net) |
| 17:21.54 | BSD_Tech | morning |
| 17:22.14 | BSD_Tech | any one have a good document on linking 2 diff server conf rooms |
| 17:23.01 | Rienzilla | bweh |
| 17:23.36 | BSD_Tech | ? |
| 17:24.20 | Rienzilla | oh sorry |
| 17:24.28 | Rienzilla | I'm annoyed that my asterisk doesnt do what I wanted |
| 17:24.35 | Rienzilla | had nothing to di with your question :) |
| 17:28.28 | Blackthorn | I have asterisk up and working on my ubentu system however, I havn't got it so it starts up automaticly upon boot. I followed some instructions in the wicki. |
| 17:28.50 | Corydon76-work | Blackthorn: which distro? |
| 17:28.55 | Blackthorn | talks about adding a script to with the commend "sudo vi /etc/even.d/asterisk" |
| 17:29.28 | Corydon76-work | Uh, you mean /etc/init.d/asterisk |
| 17:29.41 | Blackthorn | 6.06.1 lts |
| 17:30.09 | Blackthorn | thats not what the doc say... http://www.voip-info.org/wiki-Asterisk+Starting+and+Stopping |
| 17:30.28 | BSD_Tech | /etc/init.d |
| 17:30.35 | BSD_Tech | rc.local |
| 17:30.36 | Innatech | believe me, it doesn't say even.d |
| 17:30.39 | Blackthorn | kk, i'll go see what i can do |
| 17:30.53 | *** join/#asterisk blueneon (n=blueneon@dsl-146-29-190.telkomadsl.co.za) |
| 17:30.55 | BSD_Tech | and at the bottom put safe_asterisk |
| 17:34.38 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 17:34.46 | Blackthorn | so do i add the script thats listed in the doc url above to the rc.local file? |
| 17:35.12 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 17:37.27 | Blackthorn | or create a directory in init.d called asterik and create a file called rc.local placed in that directory with the script listed in the above docs? |
| 17:38.55 | ryant | does the IAXy device handle fax/data any better than SIP ??? |
| 17:39.15 | BSD_Tech | I waould use iaxmodem |
| 17:39.20 | BSD_Tech | and hylafax |
| 17:40.21 | ryant | I'm not doing fax though it's data from PSTN to a Modem without an analog line inbetween them |
| 17:40.25 | BSD_Tech | or asterisk 1.4.x with t.38 passthrew |
| 17:40.28 | coppice | ryant: nope |
| 17:40.36 | BSD_Tech | nope |
| 17:40.45 | BSD_Tech | you can not usae a modem over asterisk |
| 17:40.58 | renier | hello. is there a way to provide voicemail greetings based on time of day? |
| 17:41.12 | BSD_Tech | you can not do analog over digital it wont work right |
| 17:42.49 | ryant | dang |
| 17:44.52 | Qwell[] | huh? |
| 17:45.12 | ryant | what? |
| 17:45.36 | BSD_Tech | he wants to do analog modem data over a digital voip connection |
| 17:45.59 | BSD_Tech | not faxing |
| 17:46.01 | ryant | basically a line that is buried was cut a long time ago and never fixed properly. |
| 17:46.07 | Qwell[] | umm |
| 17:46.11 | Qwell[] | what do you think faxing...is? |
| 17:46.19 | Qwell[] | fax == modem |
| 17:46.22 | BSD_Tech | gax is short burst |
| 17:46.28 | BSD_Tech | fax |
| 17:46.39 | ryant | we have a piece of machinery that has a modem and some outside guys need to dial in and do maintenance on it |
| 17:46.58 | ryant | so I was bridging to FXS >> Asterisk >> IAXy >> Modem |
| 17:47.15 | ryant | of course it's not working well enough to finish the connection |
| 17:47.25 | ryant | would SIP work better than IAX in this respect? |
| 17:47.52 | coppice | ryant: nope |
| 17:51.50 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 17:53.58 | BSD_Tech | QWell 2400/9600 seems to work half the time but if you try to push higher I have seen major line connection errors |
| 17:54.13 | BSD_Tech | when pushing fax over voip |
| 17:54.33 | *** join/#asterisk Mrtaz (n=wcl@h-68-167-99-74.cmbrmaor.covad.net) |
| 17:55.00 | *** join/#asterisk cygar (n=cygar@200.26.191.3) |
| 17:55.08 | BSD_Tech | and I had tried to do what he wants. we have a alarm systems and I tried to push it over voip and it failed way to often to make the connection |
| 17:55.09 | Corydon76-work | fax over voip is as good of an idea as voice over avian carrier |
| 17:55.21 | BSD_Tech | lol so true |
| 17:55.33 | Rienzilla | *sigh* |
| 17:55.49 | Rienzilla | app_conference is being a bitch :/ |
| 17:56.33 | Strom_M | [TK]D-Fender: stop jittering the pigeons |
| 17:56.36 | [TK]D-Fender | Rienzilla: MeetMe doesn't fit your needs? |
| 17:56.48 | Corydon76-work | [TK]D-Fender: isn't that usually known as a "shotgun"? |
| 17:56.53 | Nugget | fax + voip = endless headaches culminating in your termination. |
| 17:57.24 | [TK]D-Fender | Corydon76-work: Nah... magnetron :) Oh... and you'll want to get replcement credit cards while you're at it ;) |
| 17:57.33 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:57.33 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:57.50 | Corydon76-work | [TK]D-Fender: not likely. I don't use the magnetic strips all that much anyway |
| 17:57.53 | Rienzilla | [TK]D-Fender: it would, but I need to modify my kernel for it |
| 17:58.07 | [TK]D-Fender | Rienzilla: Why is that? What are you running? |
| 17:58.30 | Rienzilla | I read I needed some timer source |
| 17:58.42 | Rienzilla | (i'm running asterisk 1.2 on debian sarge |
| 17:58.54 | [TK]D-Fender | Rienzilla: Zaptel compiles a kernel module. You don't need to actually compile a new KERNEL |
| 17:59.13 | [TK]D-Fender | Rienzilla: Just apt-get your kernel source & headers and you're good to go. |
| 18:00.10 | Strom_M | you don't even need the source; just the headers will do |
| 18:00.33 | Rienzilla | I'll try that |
| 18:00.47 | Rienzilla | but it doesnt seem to compile cleanly |
| 18:00.58 | Strom_M | how so? |
| 18:01.43 | [TK]D-Fender | Rienzilla: try again and pastbin the failure |
| 18:01.45 | Rienzilla | asdfadasiojd |
| 18:01.48 | Rienzilla | oops |
| 18:01.53 | Strom_M | exactly |
| 18:01.53 | [TK]D-Fender | pastebin* |
| 18:01.59 | Rienzilla | I'll fiddle around with it, I think it cant find the headers |
| 18:02.07 | Strom_M | Rienzilla: try |
| 18:02.14 | BSD_Tech | TK you know of a good document of linking 2 conf rooms on 2 diff servers |
| 18:02.16 | Strom_M | apt-get install kernel-headers-`uname -r` |
| 18:02.22 | Rienzilla | I have those |
| 18:02.42 | Strom_M | Rienzilla: on a standard debian install it should work fine then |
| 18:02.56 | Rienzilla | I'll retry |
| 18:03.41 | Rienzilla | oh nm |
| 18:05.18 | *** join/#asterisk cygar (n=cygar@200.26.191.3) |
| 18:06.06 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com) |
| 18:07.07 | *** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 18:14.32 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 18:15.06 | magic_hat | anybody have suggestions re better VOIP providers than Broadvoice? |
| 18:15.36 | blitzrage | Unlimitel or NuFone |
| 18:15.55 | magic_hat | blitzrage: how's call quality and asterisk integration? |
| 18:16.00 | blitzrage | good |
| 18:16.03 | [TK]D-Fender | BSD_Tech: Best way I can think of is a double-ended Originated call between the 2 systems |
| 18:16.14 | *** join/#asterisk SwK_ (n=SwK@63.96.55.2) |
| 18:16.25 | [TK]D-Fender | BSD_Tech: Pretty simple, its a question of how you want to terminate the link. |
| 18:17.07 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 18:17.12 | [TK]D-Fender | magic_hat: iaxtel comes recommended better than most around here. |
| 18:17.20 | Qwell[] | teliax you mean |
| 18:17.57 | [TK]D-Fender | Yeah, oops :) |
| 18:18.38 | magic_hat | cool |
| 18:18.45 | BSD_Tech | TK it basicly needs to go like this |
| 18:19.35 | BSD_Tech | caller calls the conf room on main system and then the main systen needs to connect to 2 other conf rooms on ther servers and wee are thinking iax |
| 18:19.46 | BSD_Tech | and gsm |
| 18:19.52 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 18:20.24 | [TK]D-Fender | BSD_Tech: I would use SIP if I were you. You're doing this to reduce timer load, etc... |
| 18:20.55 | [TK]D-Fender | BSD_Tech: as I mentioned its only a question of what will initiate & teardown the bridge. |
| 18:21.06 | [TK]D-Fender | BSD_Tech: The actual task is remarkably easy. |
| 18:21.14 | [TK]D-Fender | BSD_Tech: its the CLEANUP ;) |
| 18:21.19 | BSD_Tech | what would you suggest this is new to me never done this |
| 18:22.01 | [TK]D-Fender | BSD_Tech: Just told you.... Originate w/ dialplan on both sides. |
| 18:22.03 | BSD_Tech | I find no good howto documents |
| 18:22.08 | BSD_Tech | ok |
| 18:22.18 | [TK]D-Fender | BSD_Tech: there is no how-to, because its just 2 calls. |
| 18:22.33 | [TK]D-Fender | BSD_Tech: Set it to bypass logins & name recording, and presto |
| 18:22.38 | [TK]D-Fender | BSD_Tech: Real easy |
| 18:22.50 | BSD_Tech | ok how t omake the conf dectect that a user is in it and dial to the other conf |
| 18:23.11 | sopo2k4 | anyone understand why, when i press 2 this goes to priority 20 instead of what ive set. |
| 18:23.20 | sopo2k4 | exten => 01962658744,6,GotoIf($["${option}" = "1"]?19:20:21) |
| 18:23.20 | sopo2k4 | exten => 01962658744,7,GotoIf($["${option}" = "2"]?8:9:10:11:12:13:14:15:16:17:18) |
| 18:23.29 | [TK]D-Fender | BSD_Tech: Cron job + time delayed trigger check. |
| 18:23.52 | ai-a | sopo2k4: thanks for the phone number ;) |
| 18:24.02 | sopo2k4 | lol |
| 18:24.03 | sopo2k4 | np |
| 18:24.12 | *** join/#asterisk blueneon (n=blueneon@dsl-146-29-190.telkomadsl.co.za) |
| 18:24.23 | [TK]D-Fender | sopo2k4: pastebin the whole dialplan section and the CLI output of your failed attempt please. |
| 18:24.27 | sopo2k4 | ok |
| 18:24.29 | sopo2k4 | hold up |
| 18:24.38 | BSD_Tech | hmmm |
| 18:24.49 | ai-a | GotoIf($["${option}" = "1"]?19:20:21) <- WRONG |
| 18:24.57 | ai-a | if its '1' it goes 19,,, else it goes 20 |
| 18:25.10 | ai-a | GotoIf(condition?label1[:label2]) <- you can only have 2 choices. |
| 18:25.22 | ai-a | press '6' it goes to 20 too. |
| 18:25.29 | [TK]D-Fender | sopo2k4: priority 7 will never get executed |
| 18:25.32 | ai-a | press anything apart from '1' it goes 20 |
| 18:25.48 | sopo2k4 | ok |
| 18:25.50 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:25.54 | sopo2k4 | so if i wanted todo it specifically for 1 / 2 |
| 18:25.58 | sopo2k4 | id use ?19 |
| 18:26.15 | ai-a | sopo2k4: what do you expect it to do ? if they press 2, you want ? and what if they press what ? |
| 18:26.16 | sopo2k4 | only for = "1" so on for the other options |
| 18:26.26 | sopo2k4 | like a menu sort of thing |
| 18:26.35 | ai-a | write loads of them. |
| 18:26.35 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:26.37 | BSD_Tech | ok TK I will pay you 20 bucks via paypal to help me get this working . I not grep the full |
| 18:26.38 | sopo2k4 | press 1 does something, press 2 does something else, press 3 does something else |
| 18:27.09 | ai-a | exten => 01962658744,6,GotoIf($["${option}" = "{insert digit you want}"]?{ do this }) .. next line .... |
| 18:27.26 | ai-a | sopo2k4: read -> http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf |
| 18:27.28 | flujan_ | hi all. |
| 18:27.36 | flujan_ | guys I installed a sip trunk on my asterisk pbx |
| 18:27.39 | [TK]D-Fender | BSD_Tech: Tonight when I'm home.... |
| 18:27.44 | BSD_Tech | ok |
| 18:27.45 | flujan_ | asterisk is making the calls |
| 18:27.50 | flujan_ | I answer the pstn telephony |
| 18:27.57 | anonymouz666 | [TK]D-Fender your home is here. |
| 18:28.00 | anonymouz666 | this channel. |
| 18:28.03 | anonymouz666 | lol |
| 18:28.09 | flujan_ | everything I say on the pstn phone i listen on my asterisk extension. |
| 18:28.26 | flujan_ | everything I say using the asterisk extensions appear muted to the pstn phone. |
| 18:28.28 | flujan_ | any ideas? |
| 18:28.38 | blueneon | i've put this line in my dialing plans, exten => 77,1,Pickup(ZAP/3-1), its meant to allow me to pickup any calls that are ringing on the internal zap extension number 3, but i keep getting an error: pickup_exec: No target channel found for ZAP/3-1, any ideas what i might be doing wrong? |
| 18:28.43 | Strom_M | flujan_: is your asterisk box behind NAT? |
| 18:28.44 | flujan_ | I need to enable masquerade or something like that on the asterisk pbx? |
| 18:28.48 | sopo2k4 | cheers ai-a got it working |
| 18:29.06 | flujan_ | Strom_M: no... Asterisk have a network card connecting it directly with the sip trunk. |
| 18:29.17 | Strom_M | blueneon: try just ZAP/3 not ZAP/3-1 |
| 18:29.24 | blueneon | tried that |
| 18:29.27 | blueneon | same result |
| 18:29.31 | Strom_M | flujan_: so asterisk has only one network connection? |
| 18:29.49 | Strom_M | blueneon: hold on a sec, i'll help you out |
| 18:29.52 | flujan_ | two |
| 18:30.05 | flujan_ | one with my network and another with the sip/trunk |
| 18:30.08 | Strom_M | flujan_: and are you certain the SIP calls are routing out the non-NAT conection? |
| 18:30.44 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 18:31.02 | magic_hat | if my calls work great most of the time and are echoing/distorted about 20% of the time, am I right in thinking i should be blaming my ISP or Broadvoice rather than my * setup? |
| 18:31.30 | Strom_M | blueneon: read the documentation for the Pickup() application |
| 18:31.44 | Strom_M | magic_hat: yes |
| 18:31.56 | blueneon | Strom_M: i have :( |
| 18:32.00 | magic_hat | any way to pin it down to ISP or BV? |
| 18:32.06 | [TK]D-Fender | magic_hat: Usually. What phones are you using? |
| 18:32.12 | magic_hat | X-Lite. |
| 18:32.24 | Strom_M | blueneon: then you should know that you're supposed to specify an extension, not a channel |
| 18:32.33 | blueneon | i tried the extension also |
| 18:32.40 | blueneon | again same results |
| 18:32.55 | blueneon | mind checking out my extensions.conf? |
| 18:32.59 | Strom_M | pastebin it |
| 18:33.13 | blueneon | kk |
| 18:33.27 | *** join/#asterisk JD_2007 (n=Abc@cpe-75-82-48-60.socal.res.rr.com) |
| 18:33.42 | [TK]D-Fender | magic_hat: Do some tests between internal phones. if that doesn't echo, then you can blam BV. |
| 18:34.10 | Strom_M | http://plif.andkon.com/archive/wc034.gif |
| 18:34.13 | ai-a | where do i set the channel to use for DChannel ? |
| 18:34.16 | ai-a | seems to be set to 0... |
| 18:34.24 | Strom_M | ai-a: you set that in zaptel.conf |
| 18:34.33 | magic_hat | TKD-Fender: good advice. |
| 18:34.43 | ai-a | its said dchan=16, but its not using that for some reason Strom_M |
| 18:34.50 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:35.02 | flujan | Strom_M: yeap... now I get the problem, I have a externip parameter on my sip.conf |
| 18:35.17 | Strom_M | flujan: read this |
| 18:35.20 | Strom_M | ~sipnat |
| 18:35.20 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:35.39 | Strom_M | you may need to set the bindaddress also |
| 18:36.08 | Strom_M | i'm only guessing at this; i've never done it on a box with multiple network interfaces |
| 18:36.31 | blueneon | Strom_M: http://pastebin.ca/633481 |
| 18:36.39 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 18:37.32 | flujan | Strom_M: thanks for the tip. :) |
| 18:37.52 | Strom_M | blueneon: first off, I would strongly advise you /not/ to use single-digit extensions for your numbering plan |
| 18:38.26 | blueneon | (ok, will change that later, this is for testing purposes atm) |
| 18:38.51 | *** join/#asterisk angom (n=angom@red-corp-201.143.81.252.telnor.net) |
| 18:39.03 | *** part/#asterisk zpertee (n=zach@oh-69-34-21-229.sta.embarqhsd.net) |
| 18:39.05 | ai-a | Strom_M: when i type zap show channels, should i see 16 as d-channel? i see 1->15, 17->19 (18 lines) |
| 18:39.29 | Strom_M | but yeah, i'm stumped, partially because ive never used Pickup()...my only guess might be that it has something to do with the use of the macro |
| 18:39.51 | Strom_M | as a test, try an extension that dials without using a macro, then see if pickup() works on that |
| 18:40.06 | Strom_M | ai-a: no, you should not see D-channels in "zap show channels" |
| 18:40.17 | *** join/#asterisk CunningPike (n=arodgers@209.17.159.211) |
| 18:40.18 | ai-a | how can i confirm it ? |
| 18:40.25 | *** part/#asterisk angom (n=angom@red-corp-201.143.81.252.telnor.net) |
| 18:40.27 | ai-a | as the intense debug seems to show dchan as 0 |
| 18:40.30 | Strom_M | pri show span 1 |
| 18:41.02 | ai-a | oh i see, thanks,, says 16. |
| 18:41.24 | Strom_M | 0 is your E1 framing channel :) |
| 18:41.46 | ai-a | im dialing out, and getting Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (resource unavailable) (2) ] |
| 18:42.06 | Strom_M | what channel are you trying to use? |
| 18:42.50 | ai-a | says 2 i think. |
| 18:43.04 | Strom_M | don't "think" - be certain :) |
| 18:43.07 | ai-a | call 32770 on channel 2 enters state 1 (Call Initiated) |
| 18:43.14 | MrTelephone | no thinking allowed |
| 18:43.19 | ai-a | lol ;) |
| 18:43.24 | Strom_M | turn off pri debug |
| 18:43.30 | ai-a | why ? |
| 18:43.37 | Strom_M | because at this point we don't need it |
| 18:43.56 | ai-a | well, i got to that point, i think. |
| 18:44.08 | ai-a | it all seems to work, i was about to phone the teleco and ask them to check their end. |
| 18:44.26 | Strom_M | ai-a: pastebin the following: |
| 18:44.30 | Strom_M | - zaptel.conf |
| 18:44.33 | Strom_M | - zapata.conf |
| 18:44.36 | Strom_M | - extensions.conf |
| 18:46.21 | Strom_M | MrTelephone: no thinking allowed only when "think" means "I'm guessing at something I can easily find out for certain" |
| 18:46.29 | ai-a | http://pastebin.ca/633492 |
| 18:47.37 | Strom_M | several questions |
| 18:47.41 | Strom_M | (1) are you using a GUI? |
| 18:47.45 | ai-a | yes. |
| 18:47.50 | ai-a | and partly no :0 |
| 18:48.03 | Strom_M | which gui? |
| 18:48.06 | ai-a | boss wants the gui, im using linux at the moment to get it going. |
| 18:48.10 | ai-a | installed *now. |
| 18:48.17 | ai-a | hence you dont want to see the extension.conf file ;) |
| 18:48.22 | Strom_M | sigh |
| 18:48.30 | Strom_M | second, why are you using round-robin dialing? |
| 18:48.41 | Strom_M | and also, why are you not using all 30 b-channels of the E1? |
| 18:49.01 | ai-a | we've tried all different, started with just Zap/g1 |
| 18:49.23 | ai-a | dont have 30 channels, only 18 |
| 18:52.31 | Strom_M | ok |
| 18:52.35 | Strom_M | try Zap/G1 |
| 18:52.38 | Strom_M | see what happens |
| 18:52.48 | ai-a | tried it already. same result each time. |
| 18:52.56 | Strom_M | well i'd call the telco then |
| 18:53.01 | ai-a | we're guessing its failing on the other side, but we dont get any alarms here. |
| 18:59.02 | blueneon | Strom_M: I managed to find a work around, just added the zap chans to the same groups and use *8 to pick them up |
| 18:59.06 | blueneon | thanks anyways :) |
| 18:59.23 | Strom_M | blueneon: ehhhhhhhh, thats not the best way to do it |
| 19:01.52 | blueneon | i cant get it to work the other way tho |
| 19:01.53 | blueneon | :/ |
| 19:02.06 | *** join/#asterisk logyati (n=logyati@201.29.26.188) |
| 19:02.42 | logyati | [TK]D-Fender, hey, yesterday i had i problem, and you suggested me to do one thing that you dont like to do in dialplan |
| 19:03.30 | logyati | [TK]D-Fender, but it was late and i had to go home |
| 19:03.50 | logyati | [TK]D-Fender, unfortunally, i dont record logs of irc conversations, so, can you repeat please? it was something like putting auth on dialplan |
| 19:04.14 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:04.19 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 19:04.23 | *** join/#asterisk saftsack (n=saftsack@pD9E07124.dip.t-dialin.net) |
| 19:04.58 | Daviey | Hi, i need some assistance spec'ing a server for *, 150 SIP extensions and 60 channels. How would i got about spec'ing this? |
| 19:05.37 | [TK]D-Fender | logyati: Ditch your peer endty and use "Dial(SIP/user:pass@host/extentodial) for your dial-out. |
| 19:05.53 | [TK]D-Fender | Daviey: Depends on transcoding mostly. |
| 19:06.19 | Daviey | hopefully i wouldn't need to do too much. |
| 19:06.24 | [TK]D-Fender | Daviey: If you're talking standard internal use, jsut a basic modern processor, 2 gig, and a decently fast HD will do. |
| 19:06.36 | [TK]D-Fender | Daviey: No need for SCSI. |
| 19:06.48 | Daviey | 'internal' use? |
| 19:07.04 | [TK]D-Fender | Daviey: If your users are local to your server. |
| 19:07.09 | Daviey | ah |
| 19:07.13 | Daviey | split over 3 sites |
| 19:07.28 | logyati | [TK]D-Fender, oh, ty |
| 19:07.44 | caio1982 | russellb: can i talk to you in pvt? it's about a freenode' staff request regarding the channel #asteriskbrasil.org (and it needs to be handled by an #asterisk operator) |
| 19:08.16 | Daviey | [TK]D-Fender: why should the location of the users matter? |
| 19:08.31 | Qwell[] | caio1982: Russell isn't around right now. What do you need? |
| 19:08.37 | Qwell[] | or, you can msg me, I guess |
| 19:08.42 | caio1982 | ok :) |
| 19:08.48 | [TK]D-Fender | Daviey: that can have bandwitdh and consequently transcoding load in many cases |
| 19:09.27 | [TK]D-Fender | Daviey: You're looking for this remote site to use your single PBX? |
| 19:10.05 | saftsack | hi this is my newest project. atm goals: driver is already loaded, just asterisk is the thing i have to compile for the machine |
| 19:10.07 | saftsack | http://img411.imageshack.us/my.php?image=foto69ot0.jpg |
| 19:10.17 | BSD_Tech | TK pong me when you get home |
| 19:10.21 | Daviey | [TK]D-Fender: yeah, 3 sites sharing a server. 100Mb leased lines between them |
| 19:10.28 | saftsack | asus wl500gp, minipci to pci adaptor, hfc-s BRI isdn card |
| 19:10.54 | Daviey | [TK]D-Fender: one lcoation only has 20 users, other 40, and the rest where the server is located |
| 19:11.33 | BSD_Tech | I would use a vpn |
| 19:11.38 | [TK]D-Fender | Daviey: Ok, nevermind then, you're fine :) |
| 19:11.57 | [TK]D-Fender | Daviey: And yes... VNP those sites |
| 19:11.59 | [TK]D-Fender | SPN* |
| 19:12.02 | BSD_Tech | Daviey, you vpning the offices |
| 19:12.04 | [TK]D-Fender | VPN* |
| 19:12.06 | Daviey | [TK]D-Fender: oh, were you concerned about using a standard net connection between them? |
| 19:12.06 | [TK]D-Fender | hfdsldslfdlashdf |
| 19:12.11 | Strom_M | ~cohujibuggle |
| 19:12.12 | jbot | cohujibuggle is, like, gublgubbglggugglbuglgbugblgbgbgbgbglbglgbulgblugbgubgublgbglulllbgbb |
| 19:12.26 | Daviey | BSD_Tech: no.. private leased line.. no nead for vpn? |
| 19:12.40 | [TK]D-Fender | Daviey: Was more for local networking, port forwarding / keep-alives, etc... |
| 19:12.52 | BSD_Tech | you could cut down on cost by going normal network and doing a vpn |
| 19:13.01 | [TK]D-Fender | Daviey: Basically your lan segments are routed private, right? |
| 19:13.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-220-132.cablep.bezeqint.net) |
| 19:13.07 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 19:13.08 | Daviey | [TK]D-Fender: yes |
| 19:13.15 | [TK]D-Fender | Daviey: Ok, then fine as-is |
| 19:13.17 | Daviey | a glorified cat5 cable imo |
| 19:13.19 | saftsack | no coments to my project? |
| 19:13.33 | BSD_Tech | Saft ? |
| 19:13.36 | Daviey | thanks guys |
| 19:13.51 | [TK]D-Fender | saftsack: Cute,what is it? |
| 19:14.14 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 19:14.18 | saftsack | i wrote what it is ;) but in the future it should be a replacement for my patton gateway |
| 19:14.40 | BSD_Tech | some of us did not see so explain |
| 19:14.52 | *** join/#asterisk GothAlice (n=amcgrego@190.140.153.199) |
| 19:15.41 | GothAlice | I currently have ports 4569, 5060-6000, and 10000-20000 forwarded to my Asterisk box behind a Linksys Wireless Firewall/Router. Control channel stuff works like a hot damn, but I get no audio. WTF? |
| 19:15.52 | GothAlice | (On Linksys SPA962 phone off-site.) |
| 19:16.37 | Optic | sip works way better if the server isn't behind NAT |
| 19:16.47 | Jingles | this is very true. |
| 19:16.49 | [TK]D-Fender | GothAlice: Read here : |
| 19:16.51 | [TK]D-Fender | ~sipnat |
| 19:16.52 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:17.15 | [TK]D-Fender | SIP usually works fine regardless of which side(s) are NAT'd |
| 19:17.17 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 19:17.33 | GothAlice | K |
| 19:18.52 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 19:19.08 | Optic | it's the scotch, mmmm |
| 19:19.10 | Optic | oops |
| 19:20.46 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-943d21106b3327f2) |
| 19:21.54 | *** part/#asterisk JD_2007 (n=Abc@cpe-75-82-48-60.socal.res.rr.com) |
| 19:21.59 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:22.44 | CuriosCat | Seems I missed something in the tutorial here. |
| 19:22.49 | CuriosCat | Asterisk isn't picking up the phone. |
| 19:24.15 | killfill | hi |
| 19:24.29 | killfill | is there a wey to setup queues, so agents can reviece mora than 1 call at a time? |
| 19:24.48 | CuriosCat | It says "Asterisk is ready" but doesn't give an indication it's seeing the incoming ring. So maybe I configured the line wrong? |
| 19:25.06 | CuriosCat | it's a Digium 401P card with one FXO module connected to a POTS line |
| 19:25.22 | Strom_M | ...you mean a digium TDM01B |
| 19:25.34 | Strom_M | pastebin your zaptel.conf zapata.conf and extensions.conf |
| 19:26.27 | CuriosCat | One sec. |
| 19:26.57 | Mercestes | CuriosCat, also try doing a core set verbose 10 first |
| 19:27.25 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-221-126.dsl.irvnca.pacbell.net) |
| 19:27.25 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 19:27.48 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:30.41 | *** join/#asterisk gzero (n=gzero@81.175.82.2) [NETSPLIT VICTIM] |
| 19:31.22 | *** join/#asterisk karleeto (n=karl@gentoo.karlhaines.com) |
| 19:31.23 | CuriosCat | Strom_M: http://sentinel.host.net/voip/ |
| 19:31.32 | *** join/#asterisk Shoeb (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 19:31.33 | CuriosCat | (has all three files in there) |
| 19:32.16 | *** join/#asterisk astguy (n=astguy@c-24-8-95-194.hsd1.co.comcast.net) |
| 19:33.30 | Strom_M | CuriosCat: ok...just for next time, may I suggest you make copies of the sample files and then start fresh in a clean file? |
| 19:33.31 | Shoeb | Hi. From what I know, when asterisk is installed.. the basics need to be made sure of. Basics such as one extension can call the other, and vice-versa. But when you get an error on your softphone while doing that ... what can be wrong? The dial plan seems to be fine. |
| 19:33.34 | Strom_M | muuuuuuuuuuch easier to read that way |
| 19:33.46 | Strom_M | Shoeb: depends on the error |
| 19:34.07 | Strom_M | CuriosCat: you have the drivers loaded and whatnot, right? |
| 19:34.13 | *** join/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 19:34.23 | *** join/#asterisk icallandy (n=andy@rrcs-24-105-128-186.nyc.biz.rr.com) |
| 19:34.23 | astguy | Hi. I'm looking for some help with .call files. Any gurus out there with a minute to help out a newbie? |
| 19:34.36 | Strom_M | astguy: just ask your question |
| 19:34.36 | Shoeb | Hi Strom_M. Xlite softphone gives the following error: "Call failed: Address incomplete".. and the lady's voice says "The person you are calling is unavailable, please try again." |
| 19:34.48 | Strom_M | Shoeb: are you using a GUI? |
| 19:34.54 | Shoeb | Strom_M: Nope. |
| 19:35.00 | astguy | How can I tell if a .call file resulted in a busy signal? |
| 19:35.05 | Shoeb | Just vanilla asterisk. |
| 19:35.14 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-221-126.dsl.irvnca.pacbell.net) |
| 19:35.14 | Strom_M | Shoeb: what does the CLI show when you place the call? |
| 19:35.19 | Strom_M | with verbosity set to 10 |
| 19:35.22 | Shoeb | Lemme pastebin that, please. |
| 19:35.37 | Shoeb | Actually, with SIP debug not on, and verbosity to 10.. there's nothing. |
| 19:35.53 | Shoeb | But when I do sip debug, I *do* see some stuff going on, although hard for me to understand/decypher. |
| 19:36.01 | Strom_M | pastebin your extensions.conf and sip.conf and then tell me what you're dialing |
| 19:36.10 | frigidzephyr | astguy: you should get some kind of output on the CLI like with any other call, make sure verbosity is up and |
| 19:36.29 | logyati | [TK]D-Fender, i did what u said and it didnt work, please look http://www.pastebin.ca/633525 |
| 19:36.33 | astguy | Checking out CLI with a busy call now, thanks. |
| 19:36.42 | Shoeb | Ok. |
| 19:36.55 | MrTelephone | i can honestly nothing i install works 100%. i should drive off a cliff |
| 19:37.05 | MrTelephone | honestly say |
| 19:37.42 | logyati | [TK]D-Fender, ops wrong paste, let me correct |
| 19:38.08 | [TK]D-Fender | logyati: this is not a valid # to dial "ricardo@caerj.proderj.rj.gov.br" |
| 19:38.30 | [TK]D-Fender | logyati: exten => 1,1,Dial(SIP/asterisk:xxxxx@caerj.proderj.rj.gov.br/1265345243654321,30,r) |
| 19:38.35 | logyati | [TK]D-Fender, this is the right http://www.pastebin.ca/633529 |
| 19:38.41 | astguy | OK, .call on a busy gives me: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 0 |
| 19:38.49 | logyati | [TK]D-Fender, i did copy the wrong line, i have this exten to call paulo |
| 19:39.27 | *** join/#asterisk emily_25 (n=skdbjf@89.129.159.80) |
| 19:39.29 | emily_25 | hi! |
| 19:39.39 | Mercestes | Hi guy with a girl's name. |
| 19:40.01 | frigidzephyr | astguy: what type of channel is the outgoing leg of the call? |
| 19:40.11 | emily_25 | ? |
| 19:40.20 | frigidzephyr | astguy: as in sip, zap, iax ? |
| 19:40.23 | Mercestes | emily_25, Sorry, I'm mostly reminding myself of the odds. |
| 19:40.23 | astguy | frigidzephyr: IAX2 |
| 19:40.32 | Mercestes | Bad experiences and all |
| 19:40.44 | emily_25 | am |
| 19:41.21 | [TK]D-Fender | logyati: Agin, the format is bad. you should not have an "2" in your target exten. |
| 19:41.24 | [TK]D-Fender | @* |
| 19:41.39 | frigidzephyr | astguy: youll probably want to turn on iax2 set debug and take a look at that, I have not done any iax debugging tho |
| 19:41.46 | emily_25 | this is not a channel for flirt, isnt it? what matters guy or girl? |
| 19:42.11 | zone | hi |
| 19:42.22 | logyati | [TK]D-Fender, its there isnt it? Dial(SIP/asterisk:xxxxx@caerj.proderj.rj.gov.br/paulo@caerj.proderj.rj.gov.br,30,r) |
| 19:42.51 | logyati | [TK]D-Fender, ok let me try |
| 19:43.08 | astguy | frigidzephry: thanks. But if I see something in the CLI, how do I act on it programmatically -- that is with a script or something? |
| 19:43.08 | [TK]D-Fender | NO |
| 19:43.39 | [TK]D-Fender | logyati: paulo@caerj.proderj.br <- should be a NUMBEr |
| 19:43.59 | j-goddess | wow |
| 19:44.33 | Mercestes | zone: hi. |
| 19:44.41 | zone | mi asterisk isnt hanging up the sip chanels, not even when i phone the voicemail to retrieve messages |
| 19:44.52 | logyati | [TK]D-Fender, number??? im calling from pstn to sip user paulo@caerj.proderj.rj.gov.br, i dont have a number |
| 19:44.53 | frigidzephyr | astguy: this may be more what your looking for http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS |
| 19:45.06 | *** join/#asterisk drzed (n=drzed@synflood.homelinux.org) |
| 19:45.07 | zone | and olso keeping the outside calls without hang up when i hung up the phone |
| 19:45.09 | drzed | hi there |
| 19:45.13 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.137) |
| 19:45.16 | Mercestes | weird |
| 19:45.26 | astguy | frigidzephyr: cool, thanks. Checking it out. |
| 19:45.28 | Mercestes | what V of asterisk and what type of phones?? |
| 19:45.34 | [TK]D-Fender | logyati: That is a non-standard dial the : Dial(SIP/paulo@caerj.proderj.rj.gov.br) |
| 19:45.35 | drzed | what does "Timeout, but no rule 't' in context 'default'" mean? |
| 19:45.45 | zone | asterisknow :S and snom 370 and 300 |
| 19:46.16 | Mercestes | Yea...you know better...:P |
| 19:46.23 | Mercestes | See, if you were a girl I'd still help you. |
| 19:46.28 | Mercestes | Anyways... |
| 19:46.29 | [TK]D-Fender | drzed: Means you're running an IVR and the person took too long to react and * doesn't have anything to handle their lack of response |
| 19:46.39 | Mercestes | Pastebin a CLI output of verbose 10 of a call *with the hangup* |
| 19:46.55 | zone | ok |
| 19:46.57 | zone | not in the office now |
| 19:47.01 | zone | so i cant get it |
| 19:47.03 | logyati | [TK]D-Fender, my scenario is, asterisk should use openser to call paulo@caerj.proderj.rj.gov.br. So, asterisk should login with asterisk:xxxxx into openser, then call |
| 19:47.08 | Mercestes | Then we can't help you. :( |
| 19:47.09 | zone | i will came back when i can |
| 19:47.16 | zone | ok no worry |
| 19:47.18 | Mercestes | Sounds good. |
| 19:47.19 | zone | thx anyway |
| 19:47.26 | j-goddess | heh might want to turn on sip debugging as well ;) |
| 19:47.31 | Mercestes | np |
| 19:47.38 | zone | thx j-goddess |
| 19:47.44 | Mercestes | j-goddess, not yet. I just want to see asterisk going "Hangup" in the CLI first. |
| 19:47.49 | logyati | [TK]D-Fender, thats why i had that peer inside sip.conf |
| 19:47.53 | j-goddess | :) |
| 19:48.06 | j-goddess | I <3 debug it makes haxoring fun |
| 19:48.08 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 19:48.10 | Mercestes | I avoid sip debug as often as possible. |
| 19:48.11 | Shoeb | Hi Strom_M. Here's the pastebin. |
| 19:48.12 | zone | its a frequent problem? |
| 19:48.13 | Shoeb | http://pastebin.ca/633541 |
| 19:48.18 | Mercestes | zone: no. |
| 19:48.28 | zone | fuking asterisknow |
| 19:48.29 | zone | xD |
| 19:48.35 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:48.43 | zone | i will install debian and asterisk |
| 19:48.46 | Mercestes | If it were I would have gone, "oh! I know that! Just recalibrate your woozeinator and realign your calculator." |
| 19:48.50 | [TK]D-Fender | logyati: Ok, I think you're better off with someone else to assist as I have no experience with proxying |
| 19:48.52 | zone | but im afraid i cant configure it al all |
| 19:48.55 | BSD_Tech | Asterisk now is only 40% there |
| 19:48.59 | Mercestes | That's what the book and the wiki is for. |
| 19:49.01 | Mercestes | ~docs |
| 19:49.02 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 19:49.06 | Mercestes | ~thebook |
| 19:49.07 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:49.07 | BSD_Tech | it has alot of growing up to do |
| 19:49.10 | Mercestes | ~mercestes |
| 19:49.10 | jbot | mercestes is definitely a total nub |
| 19:49.11 | j-goddess | gammah |
| 19:49.13 | j-goddess | :) |
| 19:49.14 | Mercestes | Damnit! |
| 19:49.14 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 19:49.20 | zone | lol |
| 19:49.21 | zone | thx |
| 19:49.22 | Mercestes | I gotta quit typing that. |
| 19:49.26 | j-goddess | =P |
| 19:49.27 | logyati | [TK]D-Fender, oh, anyway, ty |
| 19:49.43 | logyati | does anyone here has experience with proxying? |
| 19:50.03 | centrex | Just a suggestion, when pasting configuration files, it will be a lot clearer if you leave everything commented out out of the pastebin usually. grep -v '^;' /etc/asterisk/zapata.conf for example will output everything starting with a semicolon. |
| 19:50.06 | Mercestes | logyati, is that a new age term for swinging? |
| 19:50.24 | centrex | everything *not* starting with a semicolon. |
| 19:50.49 | logyati | Mercestes, lol |
| 19:50.55 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 19:51.15 | zone | if i install a debian and asterisk, is better an apt-get asterisk or just download and install? |
| 19:51.15 | [T]ank | is there a way from the dial plan to do a three way call? |
| 19:51.30 | [T]ank | i know the phones can do it, but I want to be able to do it from the dial plan for an application I am building. |
| 19:51.35 | Mercestes | zone: google debian asterisk there is an apt-get asterisk depends or something. |
| 19:51.41 | centrex | zone, The current version that comes with debian is still in the 1.2 series. it will be seriously outdated. |
| 19:51.46 | Mercestes | zone: But after tha tmanu source dl and install is my vote. |
| 19:52.01 | zone | i can use not the stable version |
| 19:52.04 | zone | a sid version or testing |
| 19:52.12 | Mercestes | you probably want 1.4 |
| 19:52.16 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 19:52.20 | centrex | zone, Even with unstable I believe they only have 1.2. |
| 19:52.26 | zone | really? |
| 19:52.29 | zone | wo... |
| 19:52.31 | zone | oks |
| 19:52.40 | centrex | zone, Debian is known for it's stability, but a lot of time doesn't have the most bleeding edge packages. |
| 19:52.52 | zone | okk |
| 19:53.16 | zone | freepbx is good for configuring? |
| 19:53.23 | jkiff | <PROTECTED> |
| 19:53.27 | zone | i may install asterisk 1.4 and freepbx |
| 19:53.28 | [TK]D-Fender | zone: change your nick back already.... |
| 19:53.44 | zone | why? |
| 19:53.47 | [TK]D-Fender | zone: and all GUI's suck. HARD |
| 19:53.58 | blitzrage | ooooo! |
| 19:54.06 | centrex | [TK]D-Fender, even asterisk-gui?! |
| 19:54.08 | blitzrage | I've been going about this all wrong then looking for a g/f |
| 19:54.10 | centrex | blasphemy! |
| 19:54.21 | blitzrage | I just need a GUI! |
| 19:54.22 | centrex | blitzrage, wow. |
| 19:54.24 | [TK]D-Fender | centrex: I thought my statements scope was pretty clear.... |
| 19:54.34 | centrex | [TK]D-Fender, i was just kidding. |
| 19:54.42 | Mercestes | zone Debian is fine. |
| 19:54.51 | Jingles | gooey |
| 19:54.58 | zone | im afraid i cant configure all whitout a fui |
| 19:54.58 | zone | :s |
| 19:55.04 | zone | gui |
| 19:55.06 | blitzrage | untrue |
| 19:55.07 | centrex | zone, You'll have to learn asterisk. |
| 19:55.13 | blitzrage | be scared |
| 19:55.19 | zone | i know centrex |
| 19:55.24 | zone | but no time |
| 19:55.32 | blitzrage | oh good... you can forget everything about that :) |
| 19:55.45 | centrex | oh and i wasn't knocking debian... I'm a debian fan, I was just saying he would probably be better to use the latest asterisk packages instead of debian's apt packages. |
| 19:55.49 | blitzrage | asterisk is for those with too much time |
| 19:56.03 | CuriosCat | strom_m: I have a backup of the sample files. |
| 19:56.06 | zone | lol xD |
| 19:56.30 | zone | blitzrage i have time to lear, but i have no time to finish that job |
| 19:56.57 | zone | im thinking even in trixbox |
| 19:56.58 | CuriosCat | strom_m: I loaded two drivers, zaptel and wctdm |
| 19:57.49 | Shoeb | Strom_M: http://pastebin.ca/633541 |
| 19:57.49 | MrTelephone | can you get distortion from an improper Line Built Out? |
| 19:57.49 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 19:57.49 | centrex | is Strom_M even here? I can't see him saying anything...... |
| 19:57.53 | zone | blitzrage i mean i want to learn more asterisk, but i have to finish an installation in no time, later i can learn more |
| 19:58.02 | Shoeb | centrex: He bloct you. |
| 19:58.08 | centrex | =( |
| 19:58.11 | blitzrage | blocked* |
| 19:58.20 | awannabe | hey guys, has anyone had problems with the snoms (snom 360 in this case), not being able to park calls with #1 using the DTMF feature? |
| 19:58.32 | CuriosCat | centrex: He's said stuff earlier. Probably got bored of waiting for me and others to respond :P |
| 19:58.35 | zone | what about trixbox? |
| 19:58.42 | CuriosCat | (I got sidetracked when my CEO walked into my office) |
| 19:58.46 | Mercestes | Oh please god, no trixbox |
| 19:58.58 | zone | ums |
| 19:59.32 | centrex | zone: By the time you downloaded and installed trixbox you could have read asterisk docs =) |
| 19:59.42 | centrex | enough to get up and running, at least |
| 19:59.56 | zone | umm not sure, but i dont think i will understant them at first read |
| 19:59.56 | zone | xD |
| 20:00.24 | [TK]D-Fender | Shoeb: Looking for 200 in outgoing (domain 64.34.139.102) |
| 20:00.33 | [TK]D-Fender | Shoeb: SIP/2.0 484 Address Incomplete |
| 20:00.35 | zone | okok, i will try debian |
| 20:00.43 | Shoeb | That's what it says to me [TK]D-Fender |
| 20:00.50 | [TK]D-Fender | Shoeb: You have no exten to match in [outgoing] |
| 20:01.02 | zone | but u have to be here all day 24h!! ¬¬' |
| 20:01.04 | zone | xD |
| 20:01.07 | Shoeb | Meaning? :S |
| 20:01.24 | [TK]D-Fender | Shoeb: http://pastebin.ca/633557 |
| 20:01.31 | [TK]D-Fender | Shoeb: Meaning there is no 200! |
| 20:01.32 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 20:01.43 | [TK]D-Fender | Shoeb: meaning try to dial something VALID |
| 20:02.24 | *** join/#asterisk |dennis| (n=dennis@bze-dist-rou03-natpool37.btl.net) |
| 20:02.57 | Shoeb | Ok. |
| 20:03.00 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 20:03.08 | Shoeb | So it should be in [outgoing |
| 20:03.15 | Shoeb | Could you point it out to me please? |
| 20:03.18 | Shoeb | As in how to do it. |
| 20:03.32 | Shoeb | I thought by putting it in [users] it would enable me to do that. |
| 20:05.00 | [TK]D-Fender | Shoeb: See this? http://pastebin.ca/633561 |
| 20:05.08 | Shoeb | yessir |
| 20:05.15 | [TK]D-Fender | Shoeb: Your phoones are pointing to [outgoing] <------------------ |
| 20:05.24 | Shoeb | Ok... |
| 20:05.38 | [TK]D-Fender | Shoeb: So why would you think they are going to use [users]? |
| 20:05.42 | *** join/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV) |
| 20:06.06 | Shoeb | You rock! |
| 20:06.11 | Shoeb | I'll point it to users! |
| 20:06.35 | [TK]D-Fender | Shoeb: You're welcome |
| 20:06.37 | VxJasonxV | Would anyone in here happen to have a Polycom 330 or 320? For whatever reason, my 320's register, but it seems that dialing externally never reaches the asterisk server. |
| 20:06.51 | VxJasonxV | Internal (extension-to-extension calls) work just fine. |
| 20:08.07 | Shoeb | [TK]D-Fender: And now Xlite says "Call failed: Declined" |
| 20:08.16 | [TK]D-Fender | VxJasonxV: likely your dialplan on the phone itself |
| 20:08.37 | VxJasonxV | [TK]D-Fender, I would believe so... but I'm confused as to why that would be |
| 20:08.41 | [TK]D-Fender | Shoeb: You're gonna have to repastebin your whole new setup & output. |
| 20:08.57 | Shoeb | [TK]D-Fender: All I changed was sip.conf from context outgoing to users |
| 20:09.05 | VxJasonxV | 9[2-9]xxxxxxxxx <-- is the relevant dialplan that should be matched. |
| 20:09.08 | [TK]D-Fender | Shoeb: provide new info please |
| 20:09.19 | Shoeb | http://pastebin.ca/633570 |
| 20:09.24 | [TK]D-Fender | VxJasonxV: you see notihng with SIP debug enabled? |
| 20:09.38 | [TK]D-Fender | Shoeb: I need to see the CALL.... |
| 20:09.55 | VxJasonxV | No. I have debug up to 4 (assuming there's a level deeper than 1 to begin with), and it never reaches the server so far as I can tell |
| 20:09.56 | Shoeb | ah sorry |
| 20:10.49 | [TK]D-Fender | VxJasonxV: I'm talking "sip debug" here... |
| 20:10.56 | VxJasonxV | sorry |
| 20:11.00 | VxJasonxV | just enabled it on the peer. |
| 20:11.06 | Shoeb | [TK]D-Fender: http://pastebin.ca/633573 |
| 20:11.09 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 20:12.17 | [TK]D-Fender | Shoeb: looks fine, the call is accepted and you have dialplan errors like it says |
| 20:12.45 | VxJasonxV | hmmm |
| 20:12.53 | VxJasonxV | it's issuing a 404 apparently |
| 20:12.58 | Shoeb | Where does it say that? :S |
| 20:13.24 | Pagautas | Shoeb: do you have context named users? |
| 20:13.29 | Shoeb | Pagautas: Yup |
| 20:13.39 | [TK]D-Fender | <PROTECTED> |
| 20:13.41 | [TK]D-Fender | <PROTECTED> |
| 20:13.46 | Shoeb | ah |
| 20:13.52 | Pagautas | pastebin it |
| 20:14.08 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 20:14.11 | [TK]D-Fender | Pagautas: thats 1 line :) |
| 20:14.24 | Pagautas | :) |
| 20:14.36 | [TK]D-Fender | Shoeb: We clearly see it accepting the call and you have dialplan errors |
| 20:14.49 | Shoeb | Pagautas: http://pastebin.ca/633581 |
| 20:14.54 | Shoeb | I think it's the druidexten thing. |
| 20:15.11 | Shoeb | I copied that from another DRUID maintained pbx system. (DRUID the web interface thing) |
| 20:16.23 | VxJasonxV | Looking for 9303####### in default (domain voip.domain.com) |
| 20:16.25 | VxJasonxV | And that 404's |
| 20:16.47 | Pagautas | Shoeb: could you show all extensions.conf? |
| 20:16.51 | [TK]D-Fender | VxJasonxV Very clearly a dialplan error then |
| 20:16.57 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:17.01 | VxJasonxV | that would be a /server/ dialplan issue though, yes? |
| 20:17.06 | [TK]D-Fender | Shoeb: You didn't cut& paste enough then! |
| 20:17.35 | [TK]D-Fender | Shoeb: Thats like taking the engine out of 1 car, putting it on a wagon and trying to start it without the rest of the assembly :) |
| 20:17.52 | [TK]D-Fender | VxJasonxV: Yes, * dialplan |
| 20:18.08 | Shoeb | Pagautas: http://pastebin.ca/633592 |
| 20:18.25 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 20:18.28 | Shoeb | lol [TK]D-Fender .. I was modifying as I went along. |
| 20:18.29 | centrex | comments, gross |
| 20:18.38 | Shoeb | But this druidexten confused me. so I left it |
| 20:19.06 | Pagautas | ant what is druidexten? |
| 20:19.14 | [TK]D-Fender | Shoeb: Well you're calling a macro that doesn't exist and your call stops. As easy as that. |
| 20:19.21 | Pagautas | yes |
| 20:19.29 | Shoeb | I pulled it from extensions.conf that was created by druid (whch is a gui) |
| 20:19.35 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 20:19.44 | Shoeb | [TK]D-Fender: How can I improvise? |
| 20:20.06 | Pagautas | use something like this |
| 20:20.06 | Pagautas | [users] |
| 20:20.06 | Pagautas | exten => 100,1,Dial(SIP/${EXTEN}) |
| 20:20.07 | Pagautas | exten => 200,1,Dial(SIP/${EXTEN}) |
| 20:20.11 | [TK]D-Fender | Shoeb: Yes, but you don't HAVE that macro. Your call is DEAD. do you understand at all? Its like a book telling you to go to chaapter 5 and there IS NO CHAPTER 5 |
| 20:20.26 | [TK]D-Fender | Shoeb: Time to actually learn how to write a dialplan... |
| 20:20.28 | [TK]D-Fender | ~book |
| 20:20.29 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:20.30 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:20.34 | Pagautas | i think youre trying to do that now |
| 20:20.46 | Shoeb | I understand that. And now that I don't have that macro, how can I improvise? |
| 20:20.52 | centrex | Shoeb, You have to write the macro. |
| 20:20.54 | Shoeb | Pagautas: Let me try that, please. |
| 20:21.17 | zone | im afraid writing asterisk config from scratch |
| 20:21.20 | Pagautas | there is no need to use macros if its just a simple context |
| 20:21.21 | [TK]D-Fender | Shoeb: forget "improvise", time to learn. You've skipped to most important and basic part about *. |
| 20:22.55 | Shoeb | Pagautas: Thanks!! That worked! |
| 20:23.24 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 20:23.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:23.57 | Shoeb | [TK]D-Fender: I'm in the process of reading that book :) |
| 20:26.59 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 20:27.02 | *** join/#asterisk Grapsus (n=grapsus@135.224.100-84.rev.gaoland.net) |
| 20:27.10 | *** join/#asterisk AdamB0122 (n=Adam@207.200.28.175) |
| 20:27.10 | Grapsus | Hello ! |
| 20:27.16 | [TK]D-Fender | telnet |
| 20:27.19 | Nugget | telnet is eeeeeeevil! |
| 20:27.20 | [TK]D-Fender | ;) |
| 20:27.26 | Nugget | heh |
| 20:27.39 | Shoeb | Got the call working, and now the other party can't hear me. |
| 20:27.41 | Shoeb | NAT issues? |
| 20:27.43 | [TK]D-Fender | Nugget: I've got your number now :D |
| 20:27.44 | AdamB0122 | hm |
| 20:27.49 | [TK]D-Fender | Shoeb: ... |
| 20:27.51 | [TK]D-Fender | ~sipnat |
| 20:27.52 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:27.59 | AdamB0122 | if my telco uses em_w as the signalling |
| 20:28.05 | AdamB0122 | what do i use in zaptel.conf? |
| 20:28.21 | [TK]D-Fender | ok, I'm off... later all |
| 20:28.37 | Shoeb | Well, asterisk itself is on a server that is not firewalled at all. |
| 20:28.44 | Shoeb | The SIP phones however, are behind NAT. |
| 20:29.21 | AdamB0122 | Sip behind nat can be a pain. |
| 20:30.05 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 20:30.42 | Shoeb | Oh, great.. before I could atleast see in the CLI when I was dialing a PSTN phone number. And now I can't even see that. |
| 20:30.47 | Shoeb | And all I changed was the users thing. |
| 20:31.24 | Shoeb | Hey Pagautas, need your expertise again if you don't mind please? |
| 20:31.30 | centrex | Shoeb, did you restart asterisk and forget to reset the verbose? |
| 20:31.42 | Shoeb | centrex: Yes, I did restart.. |
| 20:31.54 | centrex | did you remember to set verbosity up again? |
| 20:32.20 | centrex | if not, try set verbose 9 or core set verbose 9 |
| 20:33.08 | Shoeb | Oh shit lemme do that. |
| 20:33.20 | Shoeb | But still, with verbosity on 3 I used to be able to see it. |
| 20:33.34 | Strom_M | "Verbosity was 0 and is now 9" |
| 20:33.40 | Strom_M | i'm putting 50 bucks on that |
| 20:33.44 | centrex | Yeah I think verbosity 3 is the max, but I'm not sure, so I always do set verbose 9999999 for the heck of it. |
| 20:33.45 | Strom_M | (canadian) |
| 20:33.52 | Shoeb | Verbosity was 3 and is now 10 |
| 20:33.57 | Strom_M | damn |
| 20:34.03 | Shoeb | Sorry to disappoint Strom_M :P |
| 20:34.12 | Strom_M | this is like roulette :( |
| 20:34.16 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 20:34.24 | Shoeb | haha |
| 20:34.45 | Strom_M | at least it was canuck bucks and not real dollars |
| 20:34.51 | Shoeb | centrex: I still don't see it in the CLI, even with verbose 10. I'm guessing this is the dialplan being a bad boy? |
| 20:34.59 | BSD_Tech | ok app_meetme seems to have issue |
| 20:35.13 | BSD_Tech | its not playing back the names that users record |
| 20:35.16 | centrex | Shoeb, would depend on what errors you're seeing. Does the call even go through? |
| 20:35.23 | Shoeb | centrex: Nope. |
| 20:35.26 | Shoeb | Nothing. |
| 20:35.33 | *** join/#asterisk SXT40 (n=root@cpe-65-25-148-111.columbus.res.rr.com) |
| 20:35.49 | Shoeb | Xlite says "Call failed: Not found" |
| 20:35.50 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:36.08 | SXT40 | anyone know what might cause this: res_agi.c: Could not find application (ChanIsAvail) ? I checked the modules directory... everything looks good perms-wise... :? |
| 20:37.05 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 20:37.31 | Shoeb | centrex: I set the outgoing dialplan. Now, how do I make it so when I dial out from my extension.. I can dial my local extensions around here and the external phone number? |
| 20:38.23 | Innatech | I'm rusty on dialplan commands. How do I play a sound and then dial an extension? |
| 20:38.41 | shido6 | play the sound as the first priority then dial the 2nd |
| 20:38.59 | Innatech | thx |
| 20:39.34 | centrex | Shoeb, You will have to put an extension that matches on wildcards and sends that information out your pstn channel. |
| 20:39.56 | Shoeb | centrex: I use voip SIP channels. |
| 20:40.13 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 20:40.52 | shido6 | http://en.pastebin.ca/633607 |
| 20:40.54 | shido6 | there |
| 20:43.11 | *** join/#asterisk y7n (n=na@81-179-157-41.dsl.pipex.com) |
| 20:44.05 | *** join/#asterisk link55 (n=andy@rrcs-24-105-128-186.nyc.biz.rr.com) |
| 20:45.32 | y7n | Is there a way asterisk can alert me if a call is ended unexpectedly? |
| 20:46.10 | y7n | Maybe run a small batch file, send an email or something |
| 20:46.25 | Strom_M | y7n: I don't think there's a Q.931 cause code for "Telephone slammed down by angry customer" :) |
| 20:47.14 | y7n | hmmm |
| 20:47.21 | Shoeb | Two issues I'm having. Two sip clients behind NAT are connecting to an unfirewalled asterisk server, and these two sip clients are able to call each other (thanks Pagautas ) but now no one can hear each other. The second issue is when I call a regular landline phone number I don't even see it in the CLI, verbose 10 and sip debug. |
| 20:52.54 | karleeto | i have quite a few voip systems installed around town, all using the same hardware (polycom 501s, a few linksys spa941s, with Digium TDM cards).. I dont have any echo problems at my other locations, but for some reason I'm hearing a slight echo of everything that I say (NOT of what the person on the other line is saying) |
| 20:53.03 | karleeto | does anyone have any idea what would cause this? |
| 20:53.28 | karleeto | it has to be something specific to this location, since i'm not having this problem anywhere else with the same hardware |
| 20:55.00 | Mercestes | karleeto, fxotune and adjust your rxgain and txgain values. Also check ztmonitor |
| 20:55.38 | karleeto | ALSO: i've got a job coming up that involves 3 different physical locations, with a private VPN in between.. I've done jobs with 2 locations and an IAX trunk in between, but i'm having trouble visualizing how I would go about doing this, is there anyone who would be willing to chat with me a minute and give me some ideas? |
| 20:55.48 | karleeto | Mercestes: THANKS! |
| 20:56.11 | Mercestes | Np |
| 20:56.24 | Mercestes | and once you have your VPN tunnel created, it should be just a local link to your vpn endpoint. |
| 20:56.49 | Mercestes | All yoru devices would be on the same "LAN" connected via VPN. Nothing special there. |
| 20:59.13 | *** join/#asterisk sxt40 (n=sxt40@cpe-65-25-148-111.columbus.res.rr.com) |
| 20:59.22 | mvanbaak | ~pb |
| 20:59.22 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:59.33 | sxt40 | whoot! got my problem fixed :) adding modules that you want to use, to modules.conf is a good thing :P |
| 21:00.05 | sxt40 | one last question, however. Odd codec stuff... any ideas on this: Unable to find a codec translation path from ilbc to slin ? |
| 21:00.41 | *** join/#asterisk zpertee (n=zach@oh-69-34-21-229.sta.embarqhsd.net) |
| 21:00.57 | zpertee | does anyone know of any commercial grade sip ata devices? |
| 21:01.16 | Strom_M | linksys pap2t |
| 21:01.26 | Strom_M | spa-2002 |
| 21:01.30 | Strom_M | etc etc etc etc etc |
| 21:01.40 | Jingles | Strom_M : would you choose a SIP ata over an IAXy? |
| 21:02.35 | Strom_M | Jingles: depends on the application |
| 21:03.26 | *** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net) |
| 21:05.16 | Jingles | Strom_M : that surprises me, quite honestly. What would make you choose one over the other? (in general terms) |
| 21:05.35 | Jingles | because I've got ATAs as well as IAXys in production - but they were deployed on an 'at the boss's whim' basis. |
| 21:05.41 | Jingles | not to meet some specific/special need. |
| 21:06.19 | Jabroni | to the guys that admin asterisk.org(digium?).. guess they missed to updated the download page to reflect the latest version (1.4.9).. right now that page still says 1.4.8 (and i installed that thinkin it was the last.. dooh!) |
| 21:07.38 | Strom_M | Jingles: if firewall or NAT traversal is an issue, then IAXy all the way |
| 21:08.03 | Jingles | Strom_M : would that be because it's easier to manage IAX2 since it doesn't need a SIP port, and an RTP range? |
| 21:08.06 | Jingles | or is there some other reason? |
| 21:08.12 | Strom_M | if i need polarity reversal on supervision, then sipura |
| 21:09.00 | Strom_M | Jingles: exactly |
| 21:09.04 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 21:09.14 | punkgode | hi anyone with Asterisk Realtime experience? |
| 21:09.27 | flujan | punkgode: yeap... what exactly do you need? |
| 21:09.33 | punkgode | http://bugs.digium.com/view.php?id=10305 |
| 21:09.38 | punkgode | that :P |
| 21:09.48 | flujan | people, someone with experience using asterisk 1.4 to receive and send fax? |
| 21:09.50 | Jingles | Strom_M : *nods* ok. that's good to know. It sounds then like I should convert all my remote offices to IAX2 (I've read mark spencer's docs on SIP vs. IAX2, and how IAX2 is more latency friendly as well) |
| 21:10.06 | punkgode | flujan, nope, just 1.2 |
| 21:11.12 | flujan | punkgode: hum... I were using it with odbc and postgresql on version 1.2 on 1.4 just use the postgresql module. |
| 21:11.38 | flujan | voip-info.org says that version 1.4 comes with a fax over ip feature... |
| 21:11.59 | Mercestes | T.38 |
| 21:12.03 | Mercestes | ~foip |
| 21:12.04 | jbot | methinks foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject |
| 21:12.04 | punkgode | flujan, I guess I'll try that, just to be sure that res_mysql_config is what's causing this |
| 21:12.09 | Jabroni | on which package is the ast_debug() ?? or on what ver/rev was it introduced?? |
| 21:12.09 | Dan0maN_Work | hello. my company is VERY interested in moving to * for their telephony solution. i'm currently reading the online book about * to get familiar with it. quick question came to me though. our current PBX has an overhead paging feature that we implemented. it basically broadcasts out to over 200-300 speakers in the ceiling. is there a way to implement this with *? |
| 21:12.25 | waKKu | flujan i got a lot of problems with FoIP these days... i'm using hylafax + iaxmodem now... working great |
| 21:12.26 | Strom_M | Dan0maN_Work: YES |
| 21:12.27 | Strom_M | er, yes |
| 21:12.28 | flujan | punkgode: good point... |
| 21:12.30 | Strom_M | sorry...caps |
| 21:12.47 | Jabroni | Dan0maN_Work i think that with grandstream phones.. u can set them up to autopickup the speaker... |
| 21:13.11 | Jabroni | thats what i readed a while back when i was checking that.. not sure if they came with a better solution |
| 21:13.15 | waKKu | i have a Linksys PAP2-na with (teorically) support of T.38... but, I cant make it works |
| 21:13.55 | punkgode | Dan0maN_Work, there are SIP speakers out there |
| 21:14.10 | Strom_M | or you can just hook asterisk into your existing paging amplifier |
| 21:14.21 | Strom_M | no need to go all crazy with the hardware |
| 21:14.26 | Dan0maN_Work | thanks for the answers |
| 21:14.34 | Dan0maN_Work | Strom_M: that's what i was looking for |
| 21:14.49 | Dan0maN_Work | i will investigate that at a later date |
| 21:15.53 | flujan | waKKu: I just want to use somethink built-in asterisk... It is sad that this is not working at all... |
| 21:16.18 | sopo2k4 | anyone know what usually causes this error? chan_iax2.c:7182 socket_process: Call rejected by 195.66.85.55: No authority found |
| 21:16.33 | Strom_M | sopo2k4: bad password, perhaps |
| 21:16.38 | sopo2k4 | it isnt. |
| 21:17.01 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 21:17.05 | Sci_05 | sopo2k4: bad context on one side not pointing to the right one? |
| 21:17.22 | sopo2k4 | was working earlier |
| 21:17.26 | sopo2k4 | so shouldnt be |
| 21:17.44 | Mercestes | sopo2k4, Did you update your service packs? |
| 21:18.01 | sopo2k4 | asterisk? |
| 21:18.06 | Mercestes | Yes. |
| 21:18.18 | sopo2k4 | only installed the 1.4.8 |
| 21:18.30 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 21:18.30 | Mercestes | Then I'm going to guess your configs are in error. |
| 21:18.49 | sopo2k4 | surely it wouldnt have worked earlier then |
| 21:18.50 | sopo2k4 | lol |
| 21:18.57 | Mercestes | Yea, normally I'd agree... |
| 21:19.06 | Mercestes | but if you knew what you were doing you'd know that Asterisk doesn't have "service packs." |
| 21:19.32 | Mercestes | Here is a serious question: |
| 21:19.34 | sopo2k4 | well ubuntu is up to date. |
| 21:19.40 | Mercestes | What *did* change between the time it worked and the time it didn't work. |
| 21:19.48 | Mercestes | yea, ubuntu doesn't have "service packs" either. |
| 21:19.56 | sopo2k4 | ok |
| 21:20.26 | sopo2k4 | between the time it worked and didnt work, changing few priority's to 101,102,103 |
| 21:20.27 | Mercestes | I would pretty much guarantee *something* changed from working to not working. |
| 21:20.32 | punkgode | flujan, I'm not sure if T.38 protocol is integrated into asterisk |
| 21:20.41 | Mercestes | Priority jumping is deprecated in 1.4 |
| 21:20.42 | j-goddess | sopo2k4 can you put your iax.conf and extensions.conf on pastebin.ca? |
| 21:20.53 | sopo2k4 | ok |
| 21:20.54 | punkgode | flujan, that's what you need to fax over IP |
| 21:20.55 | sopo2k4 | let me do that now |
| 21:20.59 | sopo2k4 | u can see for yourself then :P |
| 21:21.36 | j-goddess | I will say Mercestes is correct |
| 21:21.41 | flujan | punkgode: according to voip-info it is on 1.4 |
| 21:21.48 | j-goddess | that is the only time I have come across that error |
| 21:22.12 | j-goddess | but will see ;) |
| 21:22.31 | centrex | sopo2k4, and if you could, before you paste them, run grep -v '^;' /etc/asterisk/extensions.conf (makes it's more readable for everyone!) |
| 21:22.32 | sopo2k4 | http://pastebin.com/d23cf8052 |
| 21:22.38 | sopo2k4 | blah |
| 21:22.38 | sopo2k4 | ok |
| 21:22.39 | sopo2k4 | hold |
| 21:23.05 | centrex | personal quirk =) |
| 21:23.39 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 21:24.04 | MrTelephone | i kind of got my pri t1 distortion problems narrowed down to the pri card.. seems that I have to reload asterisk/kernel modules and everything is fine for a week or 2 |
| 21:24.21 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:24.25 | MrTelephone | It's probably a good idea to restart asterisk regularily I guess but.. |
| 21:24.26 | Mercestes | I see no comments. |
| 21:24.26 | MrTelephone | heh |
| 21:24.45 | sopo2k4 | i see non either |
| 21:24.47 | centrex | Mercestes, I said it before he pasted it |
| 21:25.08 | Mercestes | I read it after he pasted it. |
| 21:25.11 | Mercestes | that's what counts. |
| 21:25.12 | centrex | almost everyone uses the sample configs and pastes all that info too :-/ |
| 21:25.19 | Mercestes | lol |
| 21:25.21 | Mercestes | very true |
| 21:25.25 | sopo2k4 | i dont have the samples |
| 21:25.26 | sopo2k4 | in it |
| 21:25.27 | MrTelephone | does anyone use tdm over ip? |
| 21:25.31 | sopo2k4 | if thats what u wanted removed centrex :P |
| 21:25.55 | centrex | sopo2k4, sorry, it's just almost everyone includes all the comments from the sample files, and it's a real headache to read pastebins with all that. |
| 21:26.03 | sopo2k4 | yeah i bet :P |
| 21:26.20 | Mercestes | top entry is still in iax.conf? |
| 21:26.34 | fujin | Hi there, I can't quite remember how to do this but I have done it before - what is the command used to record a wav file to the local system, in extensions.conf? So that I might configure an extension like _99xx,1,recordsoundlawwwlzz(xx) |
| 21:26.35 | sopo2k4 | yup |
| 21:26.51 | Mercestes | K |
| 21:27.04 | Mercestes | fujin, monitor |
| 21:27.38 | centrex | sopo2k4, is it all iax users, or just the james user? Or |
| 21:27.49 | fujin | isn't monitor for recording entire configs? |
| 21:27.50 | sopo2k4 | just james |
| 21:27.55 | fujin | err, calls no tconfigs |
| 21:28.13 | sopo2k4 | refresh it |
| 21:28.17 | sopo2k4 | i added the CLI output |
| 21:28.18 | sopo2k4 | at the bottom |
| 21:28.24 | centrex | sopo2k4, thought so. You don't have an s extension in your context he's defined it. Try making an extension that says exten => s,1,playback(demo-congrats) and see if that works. |
| 21:28.46 | Strom_M | centrex: for fun, you should have them play spam.gsm instead |
| 21:28.55 | centrex | never heard that one |
| 21:29.05 | fujin | haha, go figure, I wanted Record(/path/to/lol) |
| 21:29.12 | Strom_M | "thank you for calling specal price analysis and marketing..." |
| 21:29.23 | centrex | oh is that the enlarging pills? |
| 21:29.28 | Strom_M | yes |
| 21:29.31 | centrex | aha |
| 21:29.31 | sopo2k4 | so add a s, before the first priority |
| 21:29.32 | sopo2k4 | ? |
| 21:29.38 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:29.48 | Mercestes | sopo2k4, After installing the service pack. |
| 21:29.59 | centrex | sopo2k4, No, add an s extension. Just copy and paste what I typed and put it in your dialplan right under [outgoing] |
| 21:30.07 | sopo2k4 | could u find the link |
| 21:30.07 | Mercestes | sopo2k4, in [outgoing] put the line exten => s,1,Playback(tt-monkeys) |
| 21:30.08 | sopo2k4 | ok |
| 21:30.28 | Mercestes | then exten => s,2,Hangup() |
| 21:30.29 | a1fa | how many telephone lines can come in through one RJ11 wire? |
| 21:30.41 | Strom_M | RJ11? one |
| 21:30.46 | Strom_M | RJ11 is 6P2C |
| 21:30.50 | a1fa | how about CAT6e? |
| 21:30.58 | Strom_M | RJ12 is two-line service on a 6P4C jack |
| 21:31.08 | Strom_M | RJ14 is three-line service on a 6P6C jack |
| 21:31.14 | Mercestes | a1fa, three on a Cat6e. |
| 21:31.20 | Strom_M | cat5 can handle four lines on an 8P8C jack |
| 21:31.24 | sopo2k4 | still get the same |
| 21:31.30 | Mercestes | a1fa, Just count the wires and divide by 2. |
| 21:31.35 | a1fa | hm.. so, one FXO port is for one line only |
| 21:31.37 | Strom_M | Mercestes: 8/2 = 4 |
| 21:31.38 | Mercestes | Strom_M, oh yea, 4. |
| 21:31.40 | Strom_M | LOLMATH |
| 21:31.44 | Mercestes | ~mercestes |
| 21:31.45 | jbot | mercestes is definitely a total nub |
| 21:31.54 | Mercestes | I forgot the null wires in cat. |
| 21:31.55 | Mercestes | :( |
| 21:32.06 | a1fa | Digium TDM01B can only have one line? |
| 21:32.19 | a1fa | since its single FXO port |
| 21:32.27 | Strom_M | yes |
| 21:32.30 | centrex | if it's just one fxo port, then it's just one line to the telco. |
| 21:32.55 | Qwell[] | Mercestes: null wires? |
| 21:33.17 | a1fa | centrex : that sucks.. i hate how telco splits up their lines into multiple wires |
| 21:33.28 | Strom_M | a1fa: ...... |
| 21:33.30 | a1fa | :P |
| 21:33.34 | Strom_M | you can't |
| 21:33.35 | Strom_M | have |
| 21:33.41 | Strom_M | multiple lines on a single pair!!! |
| 21:33.41 | a1fa | digital > analog |
| 21:33.48 | Strom_M | (with analog anyway) |
| 21:33.50 | Qwell[] | Strom_M: You *could* :P |
| 21:33.56 | a1fa | you could |
| 21:33.57 | Qwell[] | but let's not get into that |
| 21:34.01 | a1fa | they did that back in the day |
| 21:34.04 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 21:34.10 | a1fa | you and your neighbour shared one wire |
| 21:34.11 | Strom_M | oh sure, radio frequency division multiplexing |
| 21:34.17 | Strom_M | a1fa: it was still one line |
| 21:34.25 | Strom_M | that's called "party line service" |
| 21:34.27 | a1fa | but two different numbers |
| 21:34.32 | Mercestes | Qwell[], Yea, all the wires that don't do anything....like....the other 6. >.> |
| 21:34.33 | Strom_M | for coded ringing only |
| 21:34.37 | Qwell[] | Mercestes: 6? |
| 21:34.40 | Strom_M | you couldn't talk on the phone at the same time |
| 21:34.40 | Qwell[] | you mean 4 |
| 21:34.41 | centrex | a1fa, if you want digital, you don't want a tdm card. there are separate cards for digital/analog. |
| 21:34.45 | MrTelephone | party in your pants service :-/ |
| 21:34.50 | a1fa | centrex : T1 card? |
| 21:34.58 | a1fa | E1/T1 |
| 21:35.05 | centrex | a1fa, yes. |
| 21:35.08 | data23 | in the uk, bt still have 'dacs' units on old analogue lines, causes havoc if one wants to get broadband, as it needs to be removed an a new line installed |
| 21:35.19 | a1fa | yeah, i dont need 24 lines :P |
| 21:35.22 | a1fa | 4 is enough |
| 21:35.25 | MrTelephone | what happens if you have the inproper LINE BUILD OUT on your T1? |
| 21:35.43 | Mercestes | qwell[]: w/e You know how long it's been since I had to play with network wires?. |
| 21:35.43 | a1fa | MrTelephone : you go boom! |
| 21:35.47 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 21:35.47 | Mercestes | no wait...that was last week. |
| 21:36.00 | MrTelephone | no seriously |
| 21:36.03 | MrTelephone | what happens |
| 21:36.09 | centrex | Strom_M, I got my backpack! |
| 21:36.12 | MrTelephone | could you lose your clocking? |
| 21:36.18 | Strom_M | centrex: sweeeeeeeeeeeeeeeeeeeeeeeeeet |
| 21:36.29 | centrex | Strom_M, but it didn't have any of the gear in it.... just a backpack =( |
| 21:36.35 | Strom_M | :( |
| 21:36.42 | Strom_M | steal it from the supply cabinet |
| 21:36.48 | centrex | :/ |
| 21:36.50 | Strom_M | i mean, uh, request it nicely! |
| 21:37.45 | flujan | someone using heartbeat with asterisk? |
| 21:37.48 | Mercestes | White Orange/ Orange, White Green/ Blue, White Blue/ Green, White Brown/Brown. |
| 21:37.53 | Mercestes | see...I *can* do it... |
| 21:37.59 | Strom_M | blue orange green brown slate |
| 21:38.00 | flujan | I am searching from a ha resource script to start asterisk automaticallly |
| 21:38.01 | Mercestes | and no, I didn't google it. |
| 21:38.04 | Strom_M | white red black yellow violet |
| 21:38.11 | Strom_M | bell operators give better service |
| 21:38.13 | Mercestes | That's cat 3 isn't it? |
| 21:38.17 | Strom_M | why run backwards? you'll vomit |
| 21:38.22 | centrex | flujan, can you not have heartbeat just start the safe_asterisk script? |
| 21:38.27 | Strom_M | Mercestes: that's 25-pair color code |
| 21:38.32 | Mercestes | oh. |
| 21:38.43 | Mercestes | the RBYV threw me off. |
| 21:38.44 | flujan | centrex: yeap... but it not working.... |
| 21:38.51 | Mercestes | tho I thougth it was...red, black, yellow green. |
| 21:38.55 | Mercestes | ..but again, cat 3. w/e |
| 21:38.55 | centrex | flujan, does asterisk start normally with an asterisk -c / |
| 21:38.59 | Strom_M | Mercestes: nooooooooooooooo |
| 21:39.06 | Mercestes | no? |
| 21:39.06 | Strom_M | cat 3 uses the same color code as cat 5 |
| 21:39.13 | Mercestes | not the cat 3 I have. |
| 21:39.27 | Strom_M | TIA 568 A/B just uses a specific pinout for 8-position connectors |
| 21:39.29 | Mercestes | I do have some 8 pair cat 3 tho |
| 21:39.31 | centrex | flujan, that / was supposed to be a ? |
| 21:39.35 | MrTelephone | does cisco IOS support for SIP -> T1 on their access routers? |
| 21:39.41 | Mercestes | I should read my boxes more =/ |
| 21:39.49 | Mercestes | 8 pair..gah 4 pair. |
| 21:39.58 | Mercestes | leave me alone, Qwell, your throwing off my mojo |
| 21:40.25 | flujan | centrex: mpg123: no process killed |
| 21:40.50 | flujan | centrex: i am running it in two nodes... when I kill node 2 node one runs the stop script. :( |
| 21:42.01 | centrex | flujan, You have heartbeat setup so node 2 monitors node 1, and if asterisk stops on node 1, you shutdown on node 2 also? Doesn't seem like a very useful cluster..... |
| 21:42.40 | flujan | centrex: I don't desire this behavior... |
| 21:43.33 | flujan | centrex: node 1 falls so node 2 tryes to load the resources... even if it is the primary node of the cluster... very strange behavior. |
| 21:44.11 | Rienzilla | bweh |
| 21:44.12 | a1fa | anybody know a good asterisk ready appliance? |
| 21:44.18 | Rienzilla | zaptel won't compile indeed |
| 21:44.35 | Rienzilla | Strom_M: you still there? |
| 21:44.49 | *** join/#asterisk Dr-linux|home (n=Dr-Linux@203.99.189.222) |
| 21:45.07 | Dr-linux|home | can anyone filed error: http://phpfi.com/252269 |
| 21:45.17 | anonymouz666 | the musicclass on [general] overwrite the specific one? |
| 21:45.19 | Dr-linux|home | what's wrong it's not sending calls though |
| 21:46.06 | fujin | MrTelephone: we run SIP on our AS5400 |
| 21:46.15 | fujin | works perfect |
| 21:46.18 | fujin | two E1's, stepping |
| 21:46.22 | MrTelephone | SIP->PSTN? |
| 21:46.40 | centrex | flujan, hrm. Maybe you could try a heartbeat/mon configuration instead? Have heartbeat monitor node 1, and if it goes down, have it start mon, and then mon can detect if asterisk is running? |
| 21:46.41 | MrTelephone | what do you use for the call routing |
| 21:46.42 | fujin | yes, the universal gateway terminates the PRI's over the dual E1's, and Sip->asterisk |
| 21:46.46 | fujin | Asterisk |
| 21:46.54 | MrTelephone | nice |
| 21:46.56 | fujin | I don't do any advanced call routing, really |
| 21:47.02 | MrTelephone | i should have doen that instead of mess with these queer t1 cards |
| 21:47.06 | flujan | centrex: yeap will try it. |
| 21:47.07 | fujin | heh |
| 21:47.10 | fujin | expensive, though. |
| 21:47.24 | fujin | but we have the benefit of 2 hour onsite replacement if the entire thing breaks |
| 21:47.28 | flujan | centrex: but indeed it is a very strange behavior to stop a secondary node and the cluster stops the primary one. |
| 21:47.29 | fujin | and dual power supplies, dual E1's |
| 21:47.33 | centrex | flujan, That's really the only thing I can think of. I've used mon and heartbeat, but I'm not the guru by far. |
| 21:47.39 | Dr-linux|home | fujin: any clue? |
| 21:47.39 | Dr-linux|home | http://phpfi.com/252269 |
| 21:48.08 | MrTelephone | sip supports fallback servers.. why not setup 2 asterisk boxes? |
| 21:48.24 | fujin | I have two asterisk boxes |
| 21:48.34 | fujin | oh you're talkinm to someone else |
| 21:48.35 | fujin | doh ;( |
| 21:48.41 | shido6 | Zzz |
| 21:49.10 | MrTelephone | fujin? |
| 21:49.14 | Mercestes | Rienzilla, Soekris boxes with Astlinux |
| 21:49.24 | fujin | pass |
| 21:49.24 | fujin | :D |
| 21:49.28 | Rienzilla | ? |
| 21:49.37 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 21:49.38 | Mercestes | asterisk ready appliances.. |
| 21:49.39 | fujin | heartbeat is a much better solution that SIP SRV records, I'd say |
| 21:49.51 | Mercestes | Digium also has some out of the box.........thing. |
| 21:50.11 | fujin | Dr-linux|home: what's the problem? |
| 21:50.18 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.145.25) |
| 21:52.03 | Dr-linux|home | fujin: i'm unable to send calls through this sip peer |
| 21:52.19 | centrex | Mercestes, the asterisk appliance |
| 21:52.25 | Mercestes | yea, that. |
| 21:52.35 | Dr-linux|home | fujin: can you see anything helpfull log in the pastebin? |
| 21:52.45 | Dr-linux|home | Mercestes: :) |
| 21:52.58 | Mercestes | Yea, Mr. Linux. |
| 21:52.59 | fujin | uh |
| 21:53.01 | fujin | not really |
| 21:53.03 | Mercestes | s/Yea/Heya |
| 21:53.03 | fujin | didn't look at it |
| 21:53.28 | Dr-linux|home | http://phpfi.com/252269 << here you go |
| 21:53.33 | Dr-linux|home | Mercestes you too :P |
| 21:53.46 | fujin | no, there is nothing helpful |
| 21:53.49 | fujin | paste some asterisk output |
| 21:53.53 | fujin | what doesn't happen? |
| 21:54.16 | Dr-linux|home | i pasted in your pvt |
| 21:54.18 | Dr-linux|home | aww |
| 21:54.21 | Dr-linux|home | in Mercestes as well :P |
| 21:54.32 | Mercestes | your peer is offline |
| 21:54.45 | fujin | bingo |
| 21:54.46 | fujin | lol |
| 21:54.47 | Mercestes | There is literally, "no route to destination" which means, * cannot contact the destionation. |
| 21:54.51 | fujin | or your peer is unconfigured |
| 21:55.00 | fujin | dial 10000 will yield the same thing |
| 21:55.03 | CuriosCat | So..my TDM401B is showing me a green LED, even without a POTS line plugged in. |
| 21:55.12 | CuriosCat | (it stays green when I plug in the POTS line) |
| 21:55.15 | Daejeo1 | which one is good to go with "Cisco ATA-188" or Linlsys adapter dual fxs? |
| 21:55.40 | Daejeo1 | linksys* |
| 21:56.04 | fujin | i've had good experience with the Linksys gear, although the cisco stuff is probably better for an enterprise grade environment |
| 21:56.13 | fujin | where you're able to ask Cisco for support |
| 21:56.58 | Daejeo1 | I have to use at home |
| 21:57.02 | Strom_M | Rienzilla: what's u[ |
| 21:57.04 | Strom_M | up |
| 21:57.15 | Mercestes | fujin, Only when referring to switches and routers |
| 21:57.16 | Daejeo1 | fujin: which one do you reco |
| 21:57.35 | fujin | well, do you have a partnership with cisco? |
| 21:57.37 | fujin | if you do, then cisco |
| 21:57.40 | fujin | if not, Linksys |
| 21:57.53 | punkgode | Daejeo1, I agree with fujin, I work with Linksys and the quality they offer is is very good for the price |
| 21:58.05 | fujin | I just rolled out 50 linksys spa942's |
| 21:58.15 | fujin | they are on par with the ciscos, apart from BLF support/sidecar support |
| 21:58.25 | fujin | good quality calls, tactile/sturdy handsets |
| 21:58.39 | fujin | haven't had much experience with the ATA's, but I have read that they are good, and easily configurable/centrally provisonable |
| 21:58.57 | punkgode | fujin, they have one disadvantage.. provisioning is a pain. |
| 21:59.16 | fujin | is it not similar to the spa942's? |
| 21:59.25 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:59.46 | fujin | All I had to do was supply one DHCP option, the tftp server.. and then create the generic/per-MAC configs |
| 22:00.18 | Daejeo1 | I dealt with Cisco Ip phone7960g before . so I have little experience |
| 22:00.27 | fujin | ah |
| 22:00.41 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 22:00.43 | Daejeo1 | i guess it is the same stuff |
| 22:00.44 | fujin | if you're not a cisco partner, I'd go with Linksys ;) |
| 22:00.46 | punkgode | fujin, I upload it with HTTP posts, how do you provision port forwardings? |
| 22:01.03 | fujin | port forwardings? I don't. |
| 22:01.08 | fujin | port forwarding is stupid |
| 22:01.09 | fujin | (imho) |
| 22:01.45 | *** join/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 22:01.56 | Daejeo1 | why do I need a partnership with cisco in order to use cisco stuff |
| 22:01.58 | punkgode | fujin, how do you access a VNC service behind the router? If that's the only thing you have? |
| 22:02.11 | fujin | I don't have a VNC service behind any router |
| 22:02.20 | fujin | and if I did, it'd be over a VPN |
| 22:02.31 | punkgode | fujin, that's what I said to my boss |
| 22:02.35 | punkgode | fujin, lol |
| 22:02.40 | pigpen | Correct me if I am wrong, priority "labeling" is only used for Goto handling, when using the "n" priority method? |
| 22:02.48 | pigpen | Or maybe confirm. |
| 22:03.02 | n00dle | Here's an interesting one: Using idefisk 1.31, trying to dial comedian mail, i'm getting all of my dtmf input doubled. The * box is 1.2.6 |
| 22:03.02 | pigpen | Just trying to wrap my feeble little brain around this method. |
| 22:03.05 | *** join/#asterisk stridernzl (n=neville@125-237-98-1.jetstream.xtra.co.nz) |
| 22:03.07 | punkgode | fujin, do you use the webadmin interface at all? |
| 22:03.11 | fujin | no |
| 22:03.16 | fujin | for what? |
| 22:03.23 | fujin | the phones? |
| 22:03.31 | punkgode | fujin, port forwarding? xD |
| 22:03.34 | n00dle | Anyone know when 2.0 (zoiper) will be out for linux? |
| 22:04.03 | punkgode | fujin, port forwarding configuration, that's the only way... I hate to do a machine job |
| 22:04.10 | fujin | like I said |
| 22:04.11 | Daejeo1 | fujin: one more thing i want to ask "unlock the linksys adapter" |
| 22:04.17 | fujin | Daejeo1: don't know, sorry :) |
| 22:04.20 | fujin | buy a legit one |
| 22:04.22 | fujin | they come unlocked |
| 22:04.44 | fujin | take a look on |
| 22:04.46 | fujin | http://forum.voxilla.com/ |
| 22:04.51 | fujin | they have an excellent Linksys section |
| 22:05.01 | Daejeo1 | thankyou |
| 22:05.15 | fujin | they even have a shop, where you can buy them :P |
| 22:07.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 22:08.12 | Daejeo1 | fujin: still i could not find a cheap video phone |
| 22:08.33 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:09.45 | snuff-work | Daejeo1, cheapest hard vid phone ur going to get is a grandstream.. they have shit speaker phone though |
| 22:09.54 | snuff-work | aka speaker phone = useless |
| 22:10.08 | Mercestes | ~gs |
| 22:10.08 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:10.13 | Mercestes | ~phones |
| 22:10.13 | jbot | methinks phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 22:10.37 | *** join/#asterisk janinge (n=janinge@211.80-202-239.nextgentel.com) |
| 22:10.51 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:11.05 | *** join/#asterisk glacid (i=unknown@evool.com) |
| 22:11.12 | snuff-work | yes yes Mercestes ... but if u want cheap vid phone that's the best i've seen |
| 22:11.32 | glacid | snuff: what phone? |
| 22:11.53 | Mercestes | snuff-work, s/best/cheapest/ |
| 22:12.32 | snuff-work | glacid, look on granstreams website there is only 1.. think its gxv3000 |
| 22:13.14 | glacid | sorry, i just joined in the middle of that |
| 22:13.16 | snuff-work | like ppl say though grandstream are sucky.. but if u want the best of the cheaper vid phones.. that is it |
| 22:14.23 | glacid | yikes, for that price i'd just buy a good webcam |
| 22:14.41 | Shoeb | Two issues I'm having. Two sip clients behind NAT are connecting to an unfirewalled asterisk server, and these two sip clients are able to call each other (thanks Pagautas ) but now no one can hear each other. The second issue is when I call a regular landline phone number I don't even see it in the CLI, verbose 10 and sip debug. |
| 22:15.07 | glacid | i know the point is video phone, but if the quality is as bad as i think i'd be for a cheap $250 video phone.. i'd rather stick to a web cam haha |
| 22:16.11 | centrex | Shoeb, can you pastebin one of the sip.conf users? |
| 22:17.14 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 22:17.55 | Shoeb | centrex: http://pastebin.ca/633745 |
| 22:18.20 | glacid | having problems with my X100P and CID (within the USA) |
| 22:18.22 | glacid | http://pastebin.com/d1d7d9cc2 |
| 22:18.36 | centrex | Shoeb, you have qualify and nat set to yes, that is what I was going to recommend, no idea. |
| 22:19.40 | Shoeb | centrex: Yeah, one of the manuals suggested that. |
| 22:24.40 | *** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-943d21106b3327f2) |
| 22:26.29 | fujin | does an agent have to have a password? |
| 22:28.36 | AdamB0122 | Back |
| 22:28.36 | AdamB0122 | Quick question |
| 22:28.36 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:28.36 | AdamB0122 | i've got a DataT1 |
| 22:28.36 | AdamB0122 | sorry |
| 22:28.36 | AdamB0122 | a VoiceT1 |
| 22:28.36 | Innatech | anyone have a VSP they like that offers hunting to another number on incoming trunks once the channels are all in use? |
| 22:28.38 | AdamB0122 | b8zs, and esf |
| 22:28.45 | AdamB0122 | and the signalling is em_w |
| 22:28.55 | AdamB0122 | Now, in zapata.conf i have the signalling set |
| 22:29.12 | AdamB0122 | but for the zaptel.conf, what do i need to declare for those channels? |
| 22:29.27 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 22:29.27 | *** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) [NETSPLIT VICTIM] |
| 22:29.30 | AdamB0122 | since their em_w, I'm pretty sure i dont use fxs or anything like that |
| 22:29.37 | AdamB0122 | do i use dchan? |
| 22:29.51 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 22:29.51 | *** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) [NETSPLIT VICTIM] |
| 22:30.18 | Innatech | span= , bchan= , dchan= |
| 22:30.46 | AdamB0122 | bchan = the normal em_w channels, and dchan being the dataline? |
| 22:31.06 | Innatech | mm...I'm awfully fuzzy on the details. I just remember the parameters. |
| 22:31.30 | AdamB0122 | pretty sure thats right |
| 22:31.36 | Strom_M | AdamB0122: NO NO and NO |
| 22:31.42 | AdamB0122 | oh |
| 22:31.45 | Strom_M | AdamB0122: bchan and dchan are only used for ISDN PRI |
| 22:31.45 | AdamB0122 | damn it |
| 22:31.47 | Innatech | heh. |
| 22:31.53 | *** part/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 22:32.08 | AdamB0122 | ok, thats not what i have |
| 22:32.14 | AdamB0122 | this is being pulled off a channel bank |
| 22:32.25 | Innatech | then you do want fxo settings, I think. |
| 22:32.59 | AdamB0122 | 24 is a fax line, can't touch it, 23 is my data line, and 1-22 are my em_w / esf,b8zs lines |
| 22:33.35 | Innatech | fxoks=1-22 |
| 22:33.38 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 22:33.38 | *** join/#asterisk lilalinux (i=e-trolle@langweiligneutral.deswahnsinns.de) [NETSPLIT VICTIM] |
| 22:33.38 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 22:33.38 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) [NETSPLIT VICTIM] |
| 22:33.38 | *** join/#asterisk zined (i=dadrian@pitlab.de) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk festr__ (n=festr@ns.regnet.cz) |
| 22:33.39 | *** join/#asterisk killfill (n=killfill@201.238.233.3) |
| 22:33.39 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk Eimann (i=eimann@chaconne.etherkiller.de) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk Hymie (n=Hymie@l8r.net) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk ltd (n=z@nox.amused.net) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk Infested (n=infested@24.148.112.10) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk legis (i=legis@unaffiliated/legis) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 22:33.39 | *** join/#asterisk FlatFoot (i=FlatFoot@80.88.192.83) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk errr (n=errr@fedora/errr) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk jarrod (i=anon@theos.org) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk Pettson (i=andreas@seleya.sbin.se) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk bmg505 (n=leon@196.209.177.217) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk levi_home (n=levi@levi.dsl.xmission.com) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-221-126.dsl.irvnca.pacbell.net) [NETSPLIT VICTIM] |
| 22:33.39 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 22:33.40 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) [NETSPLIT VICTIM] |
| 22:33.40 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) [NETSPLIT VICTIM] |
| 22:33.40 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 22:33.40 | *** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) [NETSPLIT VICTIM] |
| 22:33.40 | AdamB0122 | in the zaptel.conf, and in zapata, set singalling to em_w? |
| 22:33.40 | Innatech | signalling=fxo_ks |
| 22:34.03 | *** join/#asterisk glacid (i=unknown@evool.com) |
| 22:34.03 | centrex | AdamB0122, I haven't ever seen that type of setup, but I'd definitely like to know for future reference if you figure it all out. |
| 22:34.03 | Innatech | the last time I did this was a loooong time ago, though. |
| 22:34.24 | AdamB0122 | yea, its been a pretty tricky setup |
| 22:34.44 | AdamB0122 | i've been stuck here for like three days now, only being able to do any work after hours when all the sales & support people leave |
| 22:34.54 | AdamB0122 | + I have no experience with T1's either, so that doesn't help anything. |
| 22:35.03 | Innatech | :( |
| 22:35.06 | shido6 | heheh |
| 22:35.29 | AdamB0122 | I got ahold of my telco |
| 22:35.44 | AdamB0122 | and they say off the channelbank is E&M Wink signalling |
| 22:35.52 | AdamB0122 | and that i can use E&M wink or E&M instant |
| 22:36.25 | AdamB0122 | so I'm going to presume that singalling has to be e&m since fxo and e&m are so different |
| 22:36.34 | Sci_05 | <PROTECTED> |
| 22:36.42 | AdamB0122 | XO |
| 22:36.44 | Innatech | hmmm....I remember doing it differently, and voip-wiki says fxo_ks, but whatever. If it doesn't work one way, try the other. |
| 22:37.36 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 22:37.38 | centrex | AdamB0122, what kind of card is it? |
| 22:37.40 | AdamB0122 | when i use fxo_ks signalling |
| 22:37.54 | AdamB0122 | the T1 doesn't recognize that the Asterisk box has picked up |
| 22:37.56 | AdamB0122 | and its a TE120P |
| 22:38.04 | centrex | AdamB0122, Call digium support. |
| 22:38.20 | Sci_05 | oh ok, I got quest who sold me a "DSS" circuit (its a e&m wink) and i am having a problem when i dial out. Have to wait 10-20 sec before their side picks up |
| 22:39.00 | centrex | AdamB0122, Seriously, digium gives free installation support for the purchase of all the cards. |
| 22:39.10 | AdamB0122 | k |
| 22:39.15 | AdamB0122 | lemme give them a call |
| 22:39.21 | Sci_05 | AdamB0122: whats going on? |
| 22:39.33 | AdamB0122 | Funky Telco |
| 22:39.48 | centrex | I'm one of the newer guys here, so I don't know, but someone here will. |
| 22:39.49 | Sci_05 | ahh |
| 22:39.56 | Sci_05 | good luick, cause its never their fault |
| 22:40.02 | AdamB0122 | eh |
| 22:40.27 | AdamB0122 | its more of funky telco + asterisk configuration |
| 22:40.32 | glacid | anyone have any suggestions for CID on an X100P (USA) |
| 22:40.35 | glacid | http://pastebin.com/d1d7d9cc2 |
| 22:40.38 | AdamB0122 | the telco uses em_w as their signalling |
| 22:41.08 | AdamB0122 | and I can't seem to get em_w signalling to work with ztcfg -v |
| 22:41.41 | [TK]D-Fender | glacid, if that doesn't work, then you've got a flakey card. The X100P is NOTORIOUS for that exact problem (assuming you've proven with a seperate phone that CID signalling is indeed being sent) |
| 22:42.14 | [TK]D-Fender | glacid, But your configs are fine |
| 22:42.18 | glacid | it is being sent, i went ahead and ordered a digium TDM400 |
| 22:42.23 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 22:42.53 | *** join/#asterisk Dr-linux|home (n=Dr-Linux@203.99.189.123) |
| 22:43.06 | glacid | i'm migrating my home phone lines off of analog phones to all voip phones, so i know that it is still being sent |
| 22:43.08 | Daejeo1 | [TK]D-Fender |
| 22:43.10 | Innatech | So...VSP? With hunting? Anyone got one? |
| 22:43.17 | Daejeo1 | hi :) |
| 22:43.36 | centrex | AdamB0122, I'm trying to find an answer for you, hold on a sec and we'll see. |
| 22:43.38 | Sci_05 | AdamB0122: what kind of card do you have in the box? |
| 22:43.57 | [TK]D-Fender | glacid, so you're ditching analog phones, but keeping your analog lines? |
| 22:43.58 | AdamB0122 | grrr |
| 22:44.02 | AdamB0122 | TE120P |
| 22:44.06 | blitzrage | Innatech: Mix Networks has vPBX's with Huntgroups... is that what you're looking for? |
| 22:44.10 | AdamB0122 | sorry, grr was for AIM, not here |
| 22:44.31 | Sci_05 | AdamB0122: give me a sec and I will post my config that works with e&m wink |
| 22:44.39 | AdamB0122 | thanks a ton |
| 22:45.23 | Innatech | blitzrage: no, I'm just looking for a VSP that will provision an incoming trunk such that once the channels are exhausted addtional calls will roll over to another number (on a second VSP.) We're looking to split up the traffic, and have redundency. |
| 22:46.24 | glacid | [TK]D-Fender, I actually have service through VoicePulse and Vitelity, but I wanted an emergency line in case something goes wrong |
| 22:46.26 | Sci_05 | AdamB0122: check this out http://www.pastebin.ca/633787 |
| 22:47.34 | [TK]D-Fender | glacid, You mean your analog? A TDM400 is a big expense to keep an analog line... in the end you seem to be just spending MORE. Thats what I have a Cell for. |
| 22:47.51 | AdamB0122 | whoa |
| 22:47.51 | AdamB0122 | wierd |
| 22:48.05 | Shoeb | [TK]D-Fender: Two issues I'm having. Two sip clients behind NAT are connecting to an unfirewalled asterisk server, and these two sip clients are able to call each other (thanks Pagautas ) but now no one can hear each other. The second issue is when I call a regular landline phone number I don't even see it in the CLI, verbose 10 and sip debug. |
| 22:48.24 | AdamB0122 | do you actually NOT have anything declared besides loadzone and span in your zaptel? |
| 22:48.25 | glacid | [TK]D-Fender: I know, i'm ridiculous like that :( |
| 22:48.35 | AdamB0122 | op |
| 22:48.36 | [TK]D-Fender | Shoeb, ... |
| 22:48.36 | AdamB0122 | there it is |
| 22:48.38 | [TK]D-Fender | ~sipnat |
| 22:48.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:48.38 | Innatech | It seems like there has to be a provider that can provide this. It's kinda elementary. |
| 22:48.42 | AdamB0122 | it actually has the & symbol |
| 22:48.46 | Shoeb | [TK]D-Fender: I |
| 22:48.50 | [TK]D-Fender | glacid, the first step is admitting you have a problem ;) |
| 22:48.54 | glacid | [TK]D-Fender: Also, i can get away with a little bandwidth reduction if i am forwarding a call to my cellphone |
| 22:49.01 | Shoeb | I have looked at it.. and I think qualify=yes and nat=yes should basically do it. |
| 22:49.32 | glacid | [TK]D-Fender, you know, whatever i can say to lie to myself and justify the cost ;) |
| 22:49.35 | [TK]D-Fender | Shoeb, then you are MISTAKEN. Read it again. |
| 22:49.48 | Shoeb | ok |
| 22:49.49 | [TK]D-Fender | glacid, to yourself... only to yourself :) |
| 22:50.06 | Rienzilla | hmz |
| 22:50.07 | [TK]D-Fender | glacid, but we do what we feel we must... |
| 22:50.13 | Rienzilla | zaptel won't compile indeed here on sarge :/ |
| 22:50.25 | Shoeb | [TK]D-Fender: My problem according to voipinfo is #9. And that's what it says. |
| 22:50.30 | glacid | well, i do have an iphone (yes i heard about the vulnerabiity), so this isn't 100% an exercise in cost savings |
| 22:50.31 | Innatech | [TK]D-Fender: know any VSP's that will do hunting to another number upon trunk channel exhaustion? |
| 22:50.35 | centrex | AdamB0122, here's the answer I found here: In zaptel.conf, set it up as a normal t1 like span=1,1,0,esf,b8zs (change to your specifics) then under that set e&m=channelnumbershere (that's zaptel.conf) |
| 22:50.37 | glacid | i'm going for omnipotence |
| 22:50.39 | [TK]D-Fender | Shoeb, Follow the OTHER link. |
| 22:50.49 | [TK]D-Fender | Innatech, nope. |
| 22:50.53 | Innatech | k, thanks |
| 22:50.55 | centrex | AdamB0122, And then in zapata.conf, set it up as a normal t1 line, but with signalling as em_w and no switchtype. |
| 22:51.05 | [TK]D-Fender | glacid, Ok, now we're talking ;) |
| 22:51.16 | [TK]D-Fender | glacid, I'm waiting for OpenMoko's rev 2 release |
| 22:51.30 | glacid | is that the green phone? |
| 22:51.45 | [TK]D-Fender | glacid, Nope. www.openmoko.com |
| 22:52.10 | [TK]D-Fender | glacid, Actually.... was there a combo w/ green? Don't THINK so... |
| 22:52.15 | glacid | [TK]D-Fender: oh sweet.. |
| 22:52.26 | glacid | [TK]D-Fender: no there's also an open source green phone that runs linux |
| 22:52.33 | [TK]D-Fender | glacid, just like this one. |
| 22:52.44 | [TK]D-Fender | glacid, the Green one was that QT based one. |
| 22:52.54 | Shoeb | [TK]D-Fender: Done. But it mentions how the asterisk is behind NAT. In our case, Asterisk is not behind NAT, and the sip softphones are behind NAT. And they are not able to talk to each other. |
| 22:53.06 | glacid | [TK]D-Fender: right.. qtopia, i had to search my bookmarks for it |
| 22:53.24 | Innatech | Well, if I can't get hunting, I'd be interested in people's favorite IAX trunk providers. Feel free to /msg if you like. |
| 22:53.40 | [TK]D-Fender | Shoeb, Each case is well documented |
| 22:53.49 | [TK]D-Fender | Shoeb, read the samples CLOSELY. |
| 22:54.01 | Shoeb | From the first link, right? |
| 22:54.08 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 22:54.11 | coldsteal | hello |
| 22:54.13 | [TK]D-Fender | Shoeb, yes |
| 22:54.14 | blitzrage | I'm just curious why a VSP would want to do that... because if you use all the trunks you're paying for, doesn't it cost them a "trunk" to forward the call somewhere else? (i.e. they still have to deal with the call?) |
| 22:54.16 | *** join/#asterisk msetim (i=msetim@200-140-230-235.ctame705.dsl.brasiltelecom.net.br) |
| 22:54.18 | Shoeb | Gah, I'm sure it's something small and simple and I'm overlooking it! |
| 22:54.23 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 22:54.24 | blitzrage | unless I'm misunderstanding what you're wanting to do... |
| 22:54.29 | *** part/#asterisk msetim (i=msetim@200-140-230-235.ctame705.dsl.brasiltelecom.net.br) |
| 22:54.32 | *** join/#asterisk elriah (n=e@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 22:55.09 | elriah | Do any of you use madplay? I can't get mine to start, it won't even show up in the task list. I have the full path in my musiconhold.conf section definition. Asterisk 1.2, any suggestions? |
| 22:55.15 | coldsteal | im just wondering if in sip.conf where i specify the context if i can specify 2 |
| 22:55.35 | coldsteal | like context=local and outbound |
| 22:55.53 | [TK]D-Fender | coldsteal, no. |
| 22:55.58 | blitzrage | no, it wouldn't make sense |
| 22:56.11 | blitzrage | you would specify a third context, that included both of those contexts |
| 22:56.11 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:56.17 | blitzrage | [my_context_for_special_phone] |
| 22:56.22 | blitzrage | include => local |
| 22:56.24 | blitzrage | include => outbound |
| 22:56.44 | coldsteal | o thats makes sence |
| 22:56.46 | coldsteal | thanks |
| 22:57.04 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:57.14 | [TK]D-Fender | Shoeb, canreinvite=no |
| 22:57.14 | [TK]D-Fender | ; IMPORTANT! phones must not be allowed to attempt to |
| 22:57.14 | [TK]D-Fender | ; directly connect with each other |
| 22:57.24 | Innatech | blitzrage: first off, I'd pay for the feature, and second I'm not sure that's true. I don't know how redirection works on the PSTN. |
| 22:57.55 | Innatech | For instance, if I engage call forwarding, I can still use my line after someone calls and is forwarded, even if they haven't released their circuit. |
| 22:58.01 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:58.12 | Innatech | (talking about trad. lines, now. ) |
| 22:58.12 | blitzrage | Innatech: I count that as 2 calls |
| 22:58.17 | Shoeb | LOL, I was *just* looking at that one and thinking "this is the only one part I don't know much about" |
| 22:58.27 | Innatech | blitzrage: except my pair isn't ever seized. |
| 22:58.27 | blitzrage | ya... I don't deal with traditional telephony, I do SIP |
| 22:58.47 | blitzrage | Innatech: right... your circuit is free to make a call because the forwarding is done upstream |
| 22:59.32 | Innatech | blitzrage: which suggests that the PSTN connects the caller directly to a different endpoint than they dialed. |
| 22:59.32 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 22:59.42 | Innatech | >shrug< I'm not really sure. I just would like the feature if I can find it. |
| 23:00.12 | [TK]D-Fender | Shoeb, which is why I put a giant flashing disclaimer around it :) |
| 23:00.16 | *** join/#asterisk galeras (n=root@201.245.103.169) |
| 23:00.22 | blitzrage | ya... just not sure if a VSP can do that because they are not usually the physical connection, which means they are probably going to have to accept the incoming connection and send another back to their upstream provider |
| 23:01.02 | *** join/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 23:01.04 | Innatech | yes, I've been wondering about that. Whether it works kind of like an ICMP redirect, or wheter they have to effectively proxy it for you. I just don't know. But I know what functionality I want. |
| 23:01.08 | blitzrage | not sure if a 302 back the upstream provider causes the VSP to be utilizing one of *their* trunks or not... |
| 23:01.10 | anonymouz666 | anyone know if in sip.conf the musicclass=default on [generals] overwite the specific one? I configured my own music class but it is always playing the damn calm-river |
| 23:01.18 | blitzrage | I work on the other side of the softswitch :) |
| 23:01.23 | centrex | Innatech, I have heard of something like that before, where the telco can recognize that a number they send to a pbx is being forwarded back out and just connect them directly.... I have no idea how it would be implemented, however. |
| 23:01.24 | Innatech | blitzrage: yup. In any case, I'd pay for it. |
| 23:01.47 | Innatech | centrex: me neither. I'm soft on legacy telephony. |
| 23:01.48 | blitzrage | Innatech: sent you a msg with an email to contact to ask |
| 23:01.54 | Innatech | blitrage: thanks much. |
| 23:01.56 | blitzrage | np |
| 23:02.22 | centrex | Innatech, Strom_M might know, I'd try again when he's back here. |
| 23:02.28 | Innatech | Alright, thanks. |
| 23:02.35 | Strom_M | what might I know? |
| 23:02.53 | JT | yes there's a few names for it |
| 23:02.56 | JT | ECT is one |
| 23:03.01 | JT | Explicit Call Transfer |
| 23:03.10 | Strom_M | 2BCT |
| 23:03.15 | Strom_M | two b-channel transfer |
| 23:03.46 | JT | asterisk's pri stack is pretty low on features, so naturally doesn't support it :) |
| 23:03.54 | Shoeb | [TK]D-Fender: It worked!!!! :) |
| 23:04.00 | Strom_M | JT: actually it does work on 5ESS |
| 23:04.03 | [TK]D-Fender | Shoeb, z0mg! |
| 23:04.04 | Shoeb | [TK]D-Fender: Just trying clarity issues now. |
| 23:04.05 | jerlique | I'm having problems with * listening to DTMF from a channel bank, the sip debug from * says "Unauthorised", any hints? |
| 23:04.30 | JT | Strom_M: well that's a miracle ;) |
| 23:04.35 | [TK]D-Fender | jerlique, SIP debug has nothing to to with channel banks |
| 23:04.41 | Innatech | Strom_M: so are there VSPs that will allow me to have incoming calls hunt to a second number if my trunk's channels are all in use? That was the original question. |
| 23:05.04 | Shoeb | [TK]D-Fender: The other sip agent is trying to call my extension now and I see this error: |
| 23:05.06 | Shoeb | Jul 25 19:05:00 NOTICE[5119]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:05.11 | JT | Innatech: you talking sip? |
| 23:05.22 | Innatech | Strom_M: then there was some speulation about whether or not that would cost the VSP another line. |
| 23:05.30 | Innatech | JT: preferably IAX |
| 23:05.34 | [TK]D-Fender | Shoeb, the message doesn't give any hint why, we need to see what CAUSED it |
| 23:05.38 | jerlique | tk]d-Fender: Considering * communicates to the channel bank with sip - it does |
| 23:05.46 | Shoeb | [TK]D-Fender: When I dial to him, it goes well.. but when he calls me it gives that error. |
| 23:05.51 | JT | Innatech: that significantly limits your choice |
| 23:06.00 | JT | Innatech: why on earth do you need line hunt on VoIP? |
| 23:06.02 | Innatech | JT: Yeah, I can use SIP is neccessary. |
| 23:06.05 | jerlique | how else do calls get made/received - sip is the signalling protocol/ |
| 23:06.09 | Shoeb | And then it says: == Auto fallthrough, channel 'SIP/200-08d3d638' status is 'CHANUNAVAIL' |
| 23:06.13 | Innatech | JT: to split up service over multiple providers. |
| 23:06.15 | [TK]D-Fender | jerlique, then its not a channel bank, its just a high-density SIP gateway. Channel bank is a term reserved for TDM T1 type equipement |
| 23:06.24 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.142) |
| 23:06.25 | [TK]D-Fender | jerlique, What model are you using? |
| 23:06.32 | JT | jerlique: you must mean PRI to SIP gateway, NOT channel bank. |
| 23:06.39 | Innatech | *if neccesary, not is neccessary. |
| 23:06.39 | jerlique | oh ok - sorry than! Vega 50 6x4 |
| 23:06.49 | anonymouz666 | should I use the parameter musicclass= or musiconhold= in sip.conf? |
| 23:06.57 | jerlique | Yes its a BRI/FXO -> SIP gateway |
| 23:07.00 | [TK]D-Fender | JT : no, he never said what was on the other side :) |
| 23:07.01 | centrex | Innatech, told ya Strom_M would know =) |
| 23:07.04 | JT | Innatech: for inbound or outbound? |
| 23:07.06 | ManxPower | anonymouz666: what does sip.conf.sample show you? |
| 23:07.13 | AdamB0122 | anyway to change the verbosity level AFTER you've started asterisk? |
| 23:07.20 | JT | [TK]D-Fender: ok, looks like FXS/FXO to SIP |
| 23:07.21 | Innatech | JT: inbound. Using multiple outbound trunks is as easy as sleepwalking. |
| 23:07.22 | AdamB0122 | IE: i dont care to see all the remove unix connection crap now |
| 23:07.26 | Strom_M | Hi. I'm Strom. I have a phone problem. |
| 23:07.27 | centrex | AdamB0122, set verbose numberhere |
| 23:07.31 | AdamB0122 | verbose |
| 23:07.32 | AdamB0122 | thats it |
| 23:07.32 | AdamB0122 | thanks |
| 23:07.34 | Strom_M | (Hi Strom) |
| 23:07.36 | [TK]D-Fender | Strom_M, Hi Strom! |
| 23:07.37 | centrex | AdamB0122, or core set verbose numberhere |
| 23:07.40 | JT | Innatech: why would that be possible over multiple providers? |
| 23:07.44 | Innatech | JT: as in, I have 6 channels on my main trunk. When it's full, calls roll over to a secondary trunk on another provuder.s |
| 23:07.53 | AdamB0122 | JT : turns out my Telco was stupid |
| 23:08.00 | JT | Innatech: there is no such thing as sip trunks |
| 23:08.07 | JT | AdamB0122: yeah? |
| 23:08.13 | AdamB0122 | JT : they didn't use normal signalling, and their channel layout was a bit whacky |
| 23:08.19 | AdamB0122 | JT : yea, E&M signalling |
| 23:08.23 | Innatech | JT: What? |
| 23:08.36 | centrex | AdamB0122, Did you see what I showed you? About how to configure zaptel/zapata for E&M? |
| 23:08.39 | AdamB0122 | JT : working 100% now, thanks for all the help you've given me over the last three days. |
| 23:08.48 | JT | Innatech: that is not possible anyway, unless your inbound DID provider routes the numbers to another telco |
| 23:08.50 | AdamB0122 | centrex - > uhh, no, i must have missed it |
| 23:08.56 | Innatech | JT: That's the idea. |
| 23:09.10 | Innatech | JT: Just like the hunt groups we have now. |
| 23:09.13 | centrex | AdamB0122, ah. I had went around asking because I was curious. |
| 23:09.13 | AdamB0122 | centrex > I got it working thou, turns out that zaptel.conf uses E&M while zapata.conf just uses EM |
| 23:09.16 | ManxPower | Many of my users have 100 messages in their INBOX in voicemail. |
| 23:09.20 | JT | Innatech: that stupid, why don't you just get them all sent from your provider? |
| 23:09.29 | centrex | AdamB0122, zapata should use em_w for em wink. |
| 23:09.31 | ManxPower | Tomorrow they will have an incentive to not have that many messages in their mailbox. |
| 23:09.33 | AdamB0122 | centrex > yea |
| 23:09.38 | JT | Innatech: hunt groups is a circuit switch concept. VoIP is NOT circuit switched |
| 23:09.43 | AdamB0122 | but the "E & M" part is just represented as EM |
| 23:09.54 | AdamB0122 | vs E&M in zaptel.conf, which is what screwed me up |
| 23:10.04 | JT | AdamB0122: cool, did you work out which channels were which? |
| 23:10.07 | AdamB0122 | didn't figure zaptel would use the & symbol, since most dont. |
| 23:10.09 | centrex | AdamB0122, yeah, e&m in zaptel, and em_w in zapata |
| 23:10.10 | AdamB0122 | JT > yea |
| 23:10.21 | JT | AdamB0122: Ear and Mouth ;) |
| 23:10.21 | *** join/#asterisk fabliz (n=francois@34.pool85-49-252.dynamic.orange.es) |
| 23:10.27 | ManxPower | AdamB0122: you were the one with the Adit, right? |
| 23:10.30 | AdamB0122 | centrex > yup yup. ^.^ |
| 23:10.35 | Innatech | JT: Yes, I'm aware of the difference between PSTN and VOIP. Calls come in from the PSTN to a DID--it should be possible to send them to another PSTN number (which is actually a VOIP DID) if the first one is at capacity |
| 23:10.39 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 23:10.39 | AdamB0122 | ManxPower > yes, and thank you for your help as well |
| 23:11.04 | JT | Innatech: it would make incredibly more sense for them to just send you all the calls |
| 23:11.16 | JT | Innatech: in any case, it's something you'd have to get them to agree to do |
| 23:11.18 | Innatech | JT: Yeah, I'm getting that picture. |
| 23:11.20 | JT | which may be hard |
| 23:11.35 | Innatech | yeah, I know that there's no real incentive to provide it. I just want it. Heh. |
| 23:11.46 | jerlique | SO any ideas about this SIP->BRI gateway? * can receive/make call via the gateway, it even receives the sip DTMF info messages, it just says UNauthorised to them. |
| 23:12.00 | JT | providers will often call forward if you have a fault, or if you pay them |
| 23:12.03 | fabliz | hello, anyone knows how I could compile zaptel with my custom options (i want to disable the firmware download of cards i don't use) without entering menuselect (automated install)? |
| 23:12.13 | Innatech | JT: would a full trunk qualify as a fault? |
| 23:12.19 | JT | Innatech: no |
| 23:12.23 | Innatech | gargh. |
| 23:12.26 | AdamB0122 | fabliz > pretty sure there is a walkthrough on voip-info |
| 23:12.29 | JT | Innatech: it would qualify as insufficient provisioning |
| 23:12.45 | JT | and there really is no such thing as a sip trunk |
| 23:12.45 | Innatech | Yeah. That's what I need to provide for. |
| 23:12.46 | anonymouz666 | the musiconhold is only working when configured in [general] the sip devices musicclass= does not work |
| 23:12.51 | anonymouz666 | always get the default moh |
| 23:12.51 | JT | calls are connections setup at will |
| 23:12.55 | anonymouz666 | anyone has an idea? |
| 23:12.57 | Penggu | hi all. i dont know where else to ask: i have auto-answer on snom320... but I wanted it to beep/ring first, at least once... any ideas how to do it the non-asterisk way? |
| 23:12.58 | JT | there is no persistance |
| 23:13.07 | Innatech | JT: Well, I'm being miseld somewhat by the * terminology, then, when it comes to SIP trunks. |
| 23:13.21 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 23:13.28 | JT | Innatech: there are a lot of noobs, like freepbx and some business types who call it sip trunking |
| 23:13.30 | fabliz | AdamB0122, I searched but didn't find yet... Don't you have a link? |
| 23:14.03 | keith4_ | the default transfer button is #, right? |
| 23:14.16 | keith4_ | can i make "flash" do the transfer? |
| 23:14.32 | Innatech | JT: Yeah, and the vendors tend to call it that too. I'll try and be more precise. In any case, what would be the best practice here? Retain a PSTN line and use traditional hunting to distribute calls to VOIP DIDs? |
| 23:14.43 | AdamB0122 | http://www.voip-info.org/wiki/index.php?comment_page=2&page_id=1081&maxComments=10&comments_maxComments=10&comments_sort_mode=commentDate_desc&comments_style=flat |
| 23:14.55 | anonymouz666 | dammit the musicclass= does not work for peers and friends |
| 23:14.56 | AdamB0122 | fabliz > its down below "asterisk zaptel installation" |
| 23:14.57 | JT | Innatech: best practice for reliable DIDs is to get a PRI circuit |
| 23:14.59 | anonymouz666 | unless setmusiconhold is used |
| 23:15.03 | AdamB0122 | that explains how to compile the src |
| 23:15.09 | anonymouz666 | it only works on general |
| 23:15.16 | JT | you don't really need "line hunt" on a pri, as it's all just different timeslots |
| 23:15.18 | AdamB0122 | it might not be the exact changes, but thats the compile walkthrough |
| 23:15.20 | [TK]D-Fender | keith4_, it would be REALLY great if you would tell us what equipment you are using before asking advise on its use.... |
| 23:15.21 | Innatech | JT: Heh. And when the client won't pay for that? Just go with one VSP and live with it? |
| 23:15.42 | JT | Innatech: they client is an idiot then, ITSPs are unreliable |
| 23:15.44 | keith4_ | [TK]D-Fender: fair enough. |
| 23:15.54 | fabliz | AdamB0122, I saw this one, they don't talk about menuselect at all, kind of outdated |
| 23:15.55 | Innatech | I don't think I can put that in a memo. |
| 23:15.58 | Innatech | LOL. |
| 23:15.59 | keith4_ | using plain old crappy analog phone at the moment |
| 23:16.07 | keith4_ | with digium analog hardware |
| 23:16.38 | [TK]D-Fender | keith4_, I do believe you can using "flash" (transfer=yes ; in zapata) |
| 23:16.42 | Innatech | Alright, I'm getting the picture. There's no money in this for the VSPs. I'll call around hat in hand and see if anyone will take extra money to provide it. |
| 23:16.43 | fabliz | i just want to know, if i could select my make options modifying a file instead of compiling menuselect |
| 23:16.52 | JT | Innatech: you just need to tell them there are a lot of variables that can lead to telephony unavailability with an ITSP, and it's probably not your fault if they have issue |
| 23:16.56 | Shoeb | [TK]D-Fender: You rock, let me tell you that. Now, my second issue is giving me a little trouble. Dialing an external number. |
| 23:17.14 | Innatech | yeah. They nod sagely now. When it happens, they'll scream bloody murder. |
| 23:17.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 23:17.31 | JT | Innatech: why don't you just get more channels with the provider? |
| 23:17.38 | keith4_ | [TK]D-Fender, the problem is that I can't use the # key when navigating other peoples' menus, because asterisk intercepts it as a transfer, and I don't know how to make it stop doing that |
| 23:18.02 | Shoeb | [TK]D-Fender: Verbose 10 doesn't bring anything up either when I dial the external phone number. I guess we're now leaving this to SIP debug. |
| 23:18.04 | JT | keith4_: get rid of the tT from the Dial line? |
| 23:18.24 | ManxPower | JT: I just updated my std-exten macros to play "I'm sorry, but the user's mailbox cannot accept any more messages" AND not even ring the person's phone, if they have 100 messages in their INBOX. |
| 23:18.50 | JT | ManxPower: you have some problem users/ghosts? |
| 23:19.04 | [TK]D-Fender | keith4_, Stop using "tT" in your dial options |
| 23:19.11 | Innatech | JT: They know they want multiple outgoing providers, so it would be easy to leverage that, for starters. Past that, in case one has network problems it'd be nice to be able to put in a blanket forward and get the calls on the second provider. And, for that matter, I'm used to thinking in terms of hunting when it comes to these needs so that's the direction I went naturally. |
| 23:19.20 | [TK]D-Fender | Shoeb, clearly |
| 23:19.24 | ManxPower | JT: it slows down the daily backups |
| 23:19.34 | Shoeb | [TK]D-Fender: Let me get you the debug. |
| 23:19.54 | ManxPower | and yes, we have many off site users that do not check their voicemail |
| 23:20.08 | JT | Innatech: if they're that worried about their phones working they should get a PRI |
| 23:20.22 | Innatech | JT: Yeah. I've been over that road more than once with them. |
| 23:20.28 | [TK]D-Fender | ManxPower, You should auto-tarball & email them and clear them off :) |
| 23:20.43 | ManxPower | [TK]D-Fender: 8-) |
| 23:20.50 | Innatech | JT: It was murder to get them to agree to a pure data t1 for VOIP and 'net. They wanted to carry everything over a freaking business DSL. |
| 23:20.54 | JT | Innatech: i wouldn't trust an ITSP to reliably redirect anything |
| 23:21.05 | ManxPower | I suggested putting their cell in the voicemail notification system so everytime they get a voicemail it calls their cell phone to tell them |
| 23:21.22 | [TK]D-Fender | ManxPower, you can't make all of the people happy all of the time as they say.... hope your solution makes them the happiest |
| 23:21.28 | Innatech | JT: yeah. This idea was courtesy of the something-is-better-than-nothing-and-it-sounds-like-it-should-work department. |
| 23:21.31 | ManxPower | I would not trust an ITSP to shine my shoes. |
| 23:21.36 | Innatech | Hah! |
| 23:22.05 | ManxPower | [TK]D-Fender: every 2 months or so we send out an e-mail to the users telling them to clean out their mailbox. I'm tired of it. |
| 23:22.08 | [TK]D-Fender | ManxPower, Oh course you wouldn't... its not like they do THAT for a living! Learn to outsource properly! |
| 23:22.18 | keith4_ | [TK]D-Fender, JT, I want to be able to transfer, I guess... but I don't want # to do it |
| 23:22.21 | [TK]D-Fender | ManxPower, I mean e-mail out the VM's themselves! |
| 23:22.45 | ManxPower | keith: Do you really have to transfer that many calls that you have dialed that go to outside numbers? |
| 23:22.45 | [TK]D-Fender | keith4_, I told you what to add to Zapata, and what to remove from dial. Do both and you'll be fine |
| 23:22.50 | JT | keith4_: then use your sip phone's transfer feature |
| 23:22.56 | ManxPower | [TK]D-Fender: most of them would get very confused. |
| 23:23.09 | [TK]D-Fender | JT : he's not USING a SIP phone..... you have stopped reading again! ;) |
| 23:23.27 | [TK]D-Fender | ManxPower, You say that... as though it weren't a permanent state of being for them ;) |
| 23:23.37 | JT | [TK]D-Fender: i thought that was likely, but maybe he should get a less on-crack setup ;) |
| 23:23.46 | ManxPower | keith4_: You DO understand that tT is an ugly hack mostly used by 1) newbies 2) people that do not have TRANSFER or FLASH button on their phone, right? |
| 23:24.40 | keith4_ | ManxPower: I do now ;-) |
| 23:25.36 | ManxPower | well, I need to make a run to Desterhan LA |
| 23:25.39 | ManxPower | yippee |
| 23:26.52 | keith4_ | [TK]D-Fender, how do I transfer a call now? |
| 23:27.59 | Shoeb | [TK]D-Fender: Sorry it's taking a while. I got booted off.. just trying to get back in. |
| 23:28.16 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 23:29.13 | *** join/#asterisk kje (n=kje@62-99-209-38.c-vzollerg.xdsl-line.inode.at) |
| 23:29.18 | [TK]D-Fender | keith4_, You just finished asking how to use flash to do transfers, I tell you how, and NOW you are asking the same question AGAIN! |
| 23:29.19 | BSD_Tech | TK you home yet |
| 23:29.31 | [TK]D-Fender | BSD_Tech, ummmm... no? |
| 23:29.35 | BSD_Tech | ok |
| 23:29.37 | [TK]D-Fender | ;) |
| 23:29.40 | [TK]D-Fender | j/k |
| 23:29.59 | BSD_Tech | TK has no real home |
| 23:30.15 | keith4_ | [TK]D-Fender, well, then you should conclude that using flash to do a transfer isn't working, obviously |
| 23:31.43 | BSD_Tech | Tk when you have time let me know |
| 23:32.52 | Innatech | oh...wait. We're talking about hardware flash. Nevermind. |
| 23:33.02 | [TK]D-Fender | keith4_, have you completely restarted *? "reload" alone isn't enough |
| 23:33.27 | [TK]D-Fender | BSD_Tech, ina bit |
| 23:33.34 | keith4_ | [TK]D-Fender, yes |
| 23:33.44 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:33.49 | [TK]D-Fender | keith4_, When you hit flash, what happens? |
| 23:34.03 | keith4_ | it behaves like call waiting |
| 23:34.08 | [TK]D-Fender | keith4_, you should also have threewaycalling=yes, etc in there.... |
| 23:34.21 | keith4_ | the call in progress went away, and i get a dial tone |
| 23:34.38 | [TK]D-Fender | keith4_, you should get a 2nd tone, dial your target. Once ringing, hangup and that should complete the transfer |
| 23:34.46 | keith4_ | .... oh |
| 23:35.04 | keith4_ | i didn't realize it was going to be three way |
| 23:36.41 | Shoeb | [TK]D-Fender: Now the box won't let me back in. I guess this will have to wait till later. |
| 23:37.28 | Shoeb | But one quick question, very general.. would chmodding /var to 777 cause any havoc to the system? (Someone was just trying to create .call files and it wasn't letting him do it so he did the chmod) |
| 23:37.47 | [TK]D-Fender | Shoeb, ....... |
| 23:37.56 | andrewg_fm | Shoeb: uhhh, can do :) |
| 23:38.04 | andrewg_fm | bad idea though to do it :p |
| 23:38.27 | JT | Shoeb: actually asterisk often doesn't like 777 callfiles |
| 23:38.52 | JT | Shoeb: and yes, it's stupid to 777 /var |
| 23:38.54 | Shoeb | then why wasn't it letting him create the .call files? |
| 23:39.06 | JT | because he didn't have permission? |
| 23:39.10 | *** join/#asterisk saftsack (n=saftsack@pD9E07124.dip.t-dialin.net) |
| 23:39.10 | JT | 777 is not the answer |
| 23:39.13 | Shoeb | He was root |
| 23:39.34 | [TK]D-Fender | Shoeb, because you aren't supposed to creat them in the spooled folder, you're supposed to MOVE them there after creation. |
| 23:39.39 | JT | and what are the permissions of the dir? |
| 23:39.59 | Shoeb | JT: The permissions WERE drwxr-xr-x |
| 23:40.12 | Shoeb | [TK]D-Fender: Aaah.. |
| 23:40.18 | JT | Shoeb: and owner and group? |
| 23:40.38 | Shoeb | root root |
| 23:40.45 | JT | Shoeb: the callfile documentation clearly states to never make callfiles in the spool directory |
| 23:40.57 | Shoeb | [TK]D-Fender: But he was trying to move them there. Not create there. |
| 23:41.21 | shido6 | ZzZz |
| 23:42.25 | Shoeb | You know what, wrong story. I'm just being told that he made a script to create callfiles and move them to spool by the user apache. |
| 23:42.45 | Shoeb | So when someon said create on the index.php file, it created the files (created by apache) and tried to move it to spool |
| 23:43.12 | Shoeb | PS: For general knowledge, why does a 777'd call file cause a problem? |
| 23:43.54 | *** part/#asterisk galeras (n=root@201.245.103.169) |
| 23:44.38 | Nugget | Just as a general rule it's bad form to have a file flagged executable when it isn't, but I can't think of any way that would directly cause a problem other than the fact that people will point at it and laugh. Being world-writable is surely not the best choice, either. |
| 23:47.09 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 23:47.31 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 23:48.29 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:48.29 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 23:48.32 | blitzrage | work sucks |
| 23:48.44 | shido6 | :( |
| 23:48.57 | Nugget | I love my jobs. |
| 23:49.24 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 23:51.37 | Shoeb | lol Nugget |
| 23:51.44 | Shoeb | Thanks. |
| 23:57.55 | coldsteal | im trying to forward an incoming call when busy |
