| 00:00.18 | zpertee | Hello. I have a question. What is the average hourly cost for pbx consulting? If I ask $20/hour is that too much if I am acting as a sub contractor to a company? |
| 00:00.30 | fujin | USD? |
| 00:00.35 | zpertee | yeah |
| 00:00.55 | fujin | just for consulting, or actual work? |
| 00:01.02 | fujin | consulting and contracting are two different things you see |
| 00:01.16 | zpertee | both. I designed and installed |
| 00:01.20 | fujin | I charge upwards of 90$usd/h for asterisk-based VoIP contracting |
| 00:01.22 | De_Mon | zpertee i've never met a computer consultant that charged less than $40 (and that is low imho) |
| 00:01.34 | fujin | but I do awesome stuff, so, go figure ;> |
| 00:01.53 | De_Mon | I do between 60-80 depending on the PITA factor |
| 00:02.22 | fujin | yeah, fully depends (for me) on the size of the target |
| 00:02.26 | zpertee | okay good. I'm still young and have a lot to learn so I'm trying to charge low. but a customer is complaining about $20/hour and I thought that was more than reasonable |
| 00:02.49 | fujin | Find a new customer ;P |
| 00:03.03 | De_Mon | zpertee the young learn fast, they are taking advantage of you! |
| 00:03.03 | fujin | 20/h is pretty base-rate cheap. |
| 00:03.21 | zpertee | yeah I hear ya, problem is it is family friends so it's a little more complicated |
| 00:03.28 | De_Mon | I got paid $20 for computer work at 10-15 |
| 00:03.30 | fujin | ugh; |
| 00:03.33 | fujin | I hate doing that kind of work. |
| 00:03.36 | De_Mon | RUN AWAY |
| 00:03.36 | *** join/#asterisk bmcghee (n=brentmcg@d66-183-250-149.bchsia.telus.net) |
| 00:03.42 | bmcghee | im getting this error. http://www.pastebin.ca/834172 |
| 00:03.46 | bmcghee | Asterisk 1.4.14 |
| 00:03.57 | bmcghee | trying to do ext to ext calling |
| 00:03.58 | zpertee | do you charge for any research needed? |
| 00:04.05 | De_Mon | don't charge friends and family. Either help for free and remind them as much when the complain, or don't help at all. |
| 00:04.17 | fujin | I can't see any errors bmcghee |
| 00:04.42 | bmcghee | it wont ring on the other end |
| 00:04.45 | De_Mon | zpertee sometimes... |
| 00:05.16 | zpertee | De_Mon: ok I had to do some research as I was trying to implement asterisk with a legacy pbx that I wasn't quite familiar with |
| 00:05.31 | fujin | fuck I hate legacy pbx's. |
| 00:05.35 | zpertee | amen |
| 00:05.37 | fujin | I just have someone plot out what usually happens |
| 00:05.38 | De_Mon | I wouldn't charge for 100 hours of "research" but I'd definaly do 10-20% of the actual time I spent, depending on how basic the research was |
| 00:05.39 | fujin | and replicate it with * |
| 00:06.08 | Downchuck | is using a dedicated SDSL line for a 15 person office a bad idea? |
| 00:06.11 | zpertee | ok makes sense. that's about what I did. Thanks for the reassurance |
| 00:06.27 | De_Mon | i smell food -- afk |
| 00:06.30 | Downchuck | i figure i can get off with one SDSL and one Cable, use the backup for their porn browsing, and as a backup in case of crap. |
| 00:10.22 | [TK]D-Fender | Downchuck, You didn't say what bandwidth your SDSL was or what you planned on passing OVER it. |
| 00:10.48 | Downchuck | 1.5M, voice |
| 00:10.56 | Downchuck | I'd imagine i could go with gsm |
| 00:11.08 | a1fa | does anybody know if avaya's power brick is 802.3af compliant? |
| 00:11.18 | zpertee | fujin: is 80-90 hours too long to charge if I have to personally purchase all items (most at store), install them, run wire, setup and configure asterisk, and implement it with legacy pbx? |
| 00:11.30 | Downchuck | was thinking the digium appliance |
| 00:11.52 | fujin | well, was it actually 80-90 hours? |
| 00:12.07 | fujin | I always charge exactly what I do |
| 00:12.19 | fujin | subtracting a bit here or there if I was lazy |
| 00:12.20 | zpertee | yes, but was wondering if that sounded totally ridiculous or not? |
| 00:12.21 | Downchuck | zpertee: your customer is being a bad customer. |
| 00:12.28 | fujin | mm, I think that's the case. |
| 00:12.43 | fujin | 80-90 hours at 20/h for a VoIP contractor/consultant is pretty cheap tbh. |
| 00:12.47 | fujin | sounds like you've done a pretty intense job |
| 00:13.10 | a1fa | in 802.3af, 7&8th wires are power? |
| 00:13.43 | fujin | googleit? |
| 00:13.47 | zpertee | alright I get it. I'm obviously over worrying about nothing! sorry to bother you and thanks for all of your help. |
| 00:13.59 | a1fa | fucking thing sparks everytime i hook it up |
| 00:14.06 | fujin | ha |
| 00:14.09 | fujin | you're doing it wrong. |
| 00:14.14 | a1fa | how so? |
| 00:14.28 | fujin | I dunno. |
| 00:14.30 | fujin | two plugs |
| 00:14.33 | fujin | one end to switch |
| 00:14.35 | fujin | one end to $device |
| 00:14.36 | fujin | && done |
| 00:14.39 | fujin | sparks = wrong |
| 00:14.41 | fujin | no sparks = good |
| 00:14.52 | a1fa | i have a poe brick dude |
| 00:14.59 | fujin | oh, fail |
| 00:15.01 | fujin | definitely doing it wrong |
| 00:15.22 | a1fa | ok genius |
| 00:17.35 | *** join/#asterisk MrFollies (n=Miranda@60-242-243-193.static.tpgi.com.au) |
| 00:18.29 | a1fa | i think this power supply is only charging 7&8 wire |
| 00:18.58 | a1fa | 7th wire -48, 8th wire +48 |
| 00:21.04 | a1fa | anybody have any experience with poe? |
| 00:23.52 | a1fa | what a bullshit |
| 00:23.57 | a1fa | anyone uses valcom products here? |
| 00:28.10 | Downchuck | repost.. 1.5mbps SDSL w/ 15 active calls using GSM -- over the digium appliance.. Will that work out for me? |
| 00:28.30 | Downchuck | I think that the 64kbps codec would max out at 10 calls.. |
| 00:29.38 | craigk | is there any trick to re-parking a previously parked call which I have picked up? I park a call using the one step parking feature, then pick the call back up again - and then i can not park it again :( |
| 00:31.08 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) [NETSPLIT VICTIM] |
| 00:32.58 | *** join/#asterisk w3n0y (n=chatzill@241644hfc229.tampabay.res.rr.com) [NETSPLIT VICTIM] |
| 00:32.58 | *** join/#asterisk data23 (i=data@92.b6.3845.static.theplanet.com) [NETSPLIT VICTIM] |
| 00:32.58 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 00:32.58 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) [NETSPLIT VICTIM] |
| 00:32.59 | *** join/#asterisk wothinn (n=Allfathe@vs1.svartalfheim.net) [NETSPLIT VICTIM] |
| 00:32.59 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 00:32.59 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:33.00 | *** join/#asterisk Yourname` (i=Myztic@unaffiliated/yourname/x-837320) [NETSPLIT VICTIM] |
| 00:33.00 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) [NETSPLIT VICTIM] |
| 00:33.00 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) [NETSPLIT VICTIM] |
| 00:33.00 | *** join/#asterisk pepse (n=pepse@71-223-124-101.phnx.qwest.net) [NETSPLIT VICTIM] |
| 00:33.00 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 00:33.36 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:38.03 | *** join/#asterisk CpuID (n=cpuid@gentoo/contributor/cpuid) |
| 00:38.56 | CpuID | hey ppls, ive got a TDM400 card with a single FXO and single FXS, ive got the home landline connected to the FXO and a cordless to the FXS, theres also an IAX handset in the mix there, if i call the landline the CID of my mobile shows fine on the IAX phone, but not the cordless (which has a display for it and all)... |
| 00:39.31 | CpuID | each of the IAX phone and the cordless both have extensions (200 and 201 respectively), so i can call between the 2 for testing as well |
| 00:39.37 | CpuID | any ideas as to why the CID isnt working on my FXS? |
| 00:39.45 | CpuID | its a panasonic cordless, pretty basic |
| 00:41.46 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9ceceaececb68237) |
| 00:43.53 | tzafrir_laptop | do you actually get CID? |
| 00:46.10 | [TK]D-Fender | CpuID, Means you didn't set your zapata.conf right for your fxs channel. pastebin the whole file please. |
| 00:46.11 | [TK]D-Fender | ~pb |
| 00:46.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:46.13 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 00:50.50 | bmcghee | <PROTECTED> |
| 00:50.54 | bmcghee | asterisk is running |
| 00:50.59 | bmcghee | but its reporting that its not |
| 00:51.43 | bmcghee | Verbosity is at least 3 |
| 00:51.49 | bmcghee | is what asterisk consol sayin |
| 00:54.09 | bmcghee | anyone |
| 00:54.10 | bmcghee | ? |
| 00:57.07 | CpuID | sec [TK]D-Fender |
| 00:57.42 | [TK]D-Fender | bmcghee, Not sure I follow... |
| 00:58.00 | CpuID | http://pastebin.com/m428a0090 |
| 00:58.01 | CpuID | [TK]D-Fender, |
| 00:58.42 | *** join/#asterisk Tebi_ (n=tero@gw.aller.fi) |
| 01:01.17 | tzafrir_laptop | CpuID, you need to use: callerid=asreceived |
| 01:01.18 | *** join/#asterisk techie (n=techie@adsl-76-214-31-16.dsl.lsan03.sbcglobal.net) |
| 01:01.24 | tzafrir_laptop | for the FXO channel |
| 01:02.33 | CpuID | ah sec |
| 01:02.39 | [TK]D-Fender | CpuID, http://pastebin.com/m626f217c <- give that a whirl after restarting * |
| 01:03.09 | CpuID | tzafrir_laptop, funny thing is if i call from the iax phone to the cordless i should expect the IAX phones cid, but i dont get that :P |
| 01:03.10 | tzafrir_laptop | for callerid= setting: I think a reload should do |
| 01:03.14 | *** join/#asterisk mechanicus01 (n=none@189.149.76.68) |
| 01:03.14 | CpuID | so asreceived wouldnt quite help there |
| 01:03.17 | CpuID | sec checking pastebin |
| 01:03.35 | CpuID | bit cleaner btw [TK]D-Fender :) |
| 01:03.40 | CpuID | less assumptions lol |
| 01:04.02 | mechanicus01 | hi everybody, anyone know where i can download trixbox softphone? |
| 01:04.15 | bmcghee | [TK]D-Fender, asterisk can recieve calls into the IVR but you cannot go to an EXT, nor can a extention logged in as a agent recieve the calls. also i cannot call a ext from my ext. i get DENIED |
| 01:04.25 | bmcghee | the panel says asterisk isnt running but i see it running |
| 01:05.07 | mechanicus01 | or maybe HUDlite softphone |
| 01:05.33 | *** join/#asterisk MrFollies (n=Miranda@60-242-243-193.static.tpgi.com.au) |
| 01:06.22 | tzafrir_laptop | good? no idea. atcom sell IAX phones and ATAs. |
| 01:06.38 | MrFollies | atcom? |
| 01:06.40 | [TK]D-Fender | bmcghee, There are way to many things going on there. First lets jump to the fact you have QUEUE?AGENT issues. Thats its own little world. but before that you say you can't even pick an OPTION on an IVR thats supposed to lead to that QUEUE? |
| 01:06.47 | tzafrir_laptop | I have one such ATA and it looks reasonable |
| 01:07.07 | [TK]D-Fender | bmcghee, and what "panel" are you talking about? |
| 01:07.18 | bmcghee | freepbx |
| 01:07.18 | [TK]D-Fender | MrFollies, No. |
| 01:07.29 | bmcghee | freepbx isnt connecting to asterisk |
| 01:07.33 | [TK]D-Fender | bmcghee, You already know whats coming next.... |
| 01:07.37 | bmcghee | ya |
| 01:07.42 | *** join/#asterisk jwh (i=jwh@scarlett.lon.rewt.org.uk) |
| 01:07.44 | bmcghee | but im having asterisk problems as well |
| 01:07.57 | bmcghee | and freepbx chan is dead. no one answering |
| 01:08.00 | Qwell | unless you have configs you wrote 100% by yourself, it's a freepbx problem |
| 01:08.12 | tzafrir_laptop | bmcghee, ask asterisk questions |
| 01:08.17 | [TK]D-Fender | bmcghee, and I guess you don't see fit to tell us what your call is coming in on either.... this may have something to dow ith your IVR issues... |
| 01:08.31 | bmcghee | everything worked until i changed passwords in manager.conf and amportal.conf |
| 01:08.42 | Qwell | sounds like a freepbx problem to me |
| 01:08.47 | [TK]D-Fender | Qwell, Like that isn't a ready-made contradiction in terms the moment you uttered it :) |
| 01:08.48 | bmcghee | ivr works fine |
| 01:08.53 | bmcghee | ivr to que works |
| 01:08.57 | tzafrir_laptop | bmcghee, asterisk -r |
| 01:08.58 | bmcghee | que to EXT dont work |
| 01:09.12 | tzafrir_laptop | do you have messages about failed manager login attempts? |
| 01:09.14 | bmcghee | k im -r |
| 01:09.16 | [TK]D-Fender | bmcghee, "asterisk can recieve calls into the IVR but you cannot go to an EXT," <--- you just said you can't dial an ext.... |
| 01:09.37 | Qwell | tzafrir_laptop: the answer is that the password in the database is wrong. |
| 01:09.44 | Qwell | it's clearly not an asterisk problem... |
| 01:09.47 | [TK]D-Fender | bmcghee, And yes this is clearly a FreePBX issue. All of this. You mucked around and can't bail yourself out. |
| 01:12.30 | MrFollies | tzafrir_laptop: Any idea where I can buy these things? |
| 01:13.50 | [TK]D-Fender | MrFollies, No. <- |
| 01:14.00 | CpuID | [TK]D-Fender, weird...same issue |
| 01:14.08 | CpuID | file is much cleaner now at least but yea :P |
| 01:14.17 | [TK]D-Fender | CpuID, Do you get the CID from the IAX phone? |
| 01:14.22 | CpuID | yep |
| 01:14.35 | CpuID | called the landline/fxo and got my mobiles cid showing on the iax phone |
| 01:14.36 | [TK]D-Fender | CpuID, Are you WAITING for the CID to appear from both ends? |
| 01:14.51 | CpuID | ive got it set to ring both handsets at the same time when a call comes in on the landline (doesnt run Answer first either) |
| 01:15.01 | CpuID | waiting...elaborate? |
| 01:15.16 | [TK]D-Fender | CpuID, You need to wait a few sec for your analog phone to get the CID info.... |
| 01:15.24 | [TK]D-Fender | CpuID, Usually between the 1st & 2nd ring |
| 01:15.30 | CpuID | ah sec |
| 01:15.33 | CpuID | exten => s,1,Wait(1) |
| 01:15.33 | CpuID | exten => s,2,Dial(IAX2/nathanhome1&${ANALOGHSZAP},120) ; Call Both Office Phone And Kitchen Analog Phone For Incoming Calls |
| 01:15.34 | tzafrir_laptop | MrFollies, their site is http://www.atcom.cn/ , which should give you an idea of the available models. |
| 01:15.36 | [TK]D-Fender | CpuID, Are you letting it ring 2-3 times? |
| 01:15.39 | tzafrir_laptop | Just look for them |
| 01:15.49 | [TK]D-Fender | atcom = BLEH |
| 01:15.54 | tzafrir_laptop | For some strange reason the first hit I got was amazon UK |
| 01:15.54 | CpuID | ya i let the fxs ring a few times |
| 01:16.04 | CpuID | as i know it sometimes takes a few rings for the cid to appear on the cordless |
| 01:16.11 | CpuID | the iax phones cid is instant of course |
| 01:16.14 | bmcghee | it put all passwords back to there default |
| 01:16.16 | bmcghee | still not working |
| 01:16.40 | CpuID | ive pretty much eliminated any issues with the fxo by calling between the iax phone and the cordless...cid shows on iax while not on cordless even with internal calls |
| 01:16.52 | [TK]D-Fender | CpuID, Do you ever see CID on the FXS? |
| 01:17.01 | MrFollies | K, thanks... |
| 01:17.10 | tzafrir_laptop | [TK]D-Fender, their A188 is nice |
| 01:17.22 | CpuID | hmm |
| 01:17.23 | CpuID | [Dec 28 11:16:48] WARNING[16783]: chan_zap.c:4125 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. |
| 01:17.26 | CpuID | i did just see that though |
| 01:17.32 | CpuID | when tryign to call the cordless from the iax phone |
| 01:17.43 | CpuID | and i did see it before as well when trying to call the landline/fxo |
| 01:17.51 | [TK]D-Fender | tzafrir_laptop, Can you link it? |
| 01:17.53 | CpuID | nah i never see it on the fxs... |
| 01:18.00 | CpuID | unless the cordless is plugged into a standard landline :P |
| 01:18.06 | tzafrir_laptop | http://www.atcom.cn/En_products_AG188.html |
| 01:18.06 | CpuID | (bypassing *) |
| 01:18.14 | [TK]D-Fender | CpuID, Well that proves its the FXS, not the phone |
| 01:18.22 | [TK]D-Fender | CpuID, pastebin your zaptel.conf |
| 01:18.57 | [TK]D-Fender | tzafrir_laptop, Single-port FXS? |
| 01:19.07 | CpuID | sec |
| 01:19.32 | CpuID | ill pastebin just the uncommented lines (i used a basic example and uncommented lines, mofo comments :)) |
| 01:19.48 | [TK]D-Fender | CpuID, trash everything commented out permanently THEN pastebin it |
| 01:19.52 | CpuID | hehe :P |
| 01:20.38 | fujin | sed 'd/^#.*/' |
| 01:20.45 | CpuID | lol outta like 100+ lines the config was like 4 lines :P |
| 01:20.47 | CpuID | pastebining now |
| 01:20.49 | tzafrir_laptop | [TK]D-Fender, yes |
| 01:21.05 | CpuID | http://pastebin.com/d734d56d7 |
| 01:21.33 | CpuID | the loadzone/defaultzone are giving me AU dialtones/busy tones which is mainly what i wanted (make it sound more local to the average person using the phone) |
| 01:21.43 | CpuID | and its a standard AU cordless, nothing special bout it |
| 01:21.53 | [TK]D-Fender | tzafrir_laptop, Well their PA1688 stuff was BLEH. phone felt like garbage and sounded much the same. I suppose an ATA can be an inherently better experience. They work ok? |
| 01:22.07 | [TK]D-Fender | CpuID, You in AU with that? |
| 01:22.12 | tzafrir_laptop | [TK]D-Fender, yes |
| 01:22.39 | CpuID | ya, australia |
| 01:22.47 | [TK]D-Fender | tzafrir_laptop, Well I'll take your first-hand accounting of it then. How intuitive are calling features compared to say a Linksys? |
| 01:23.02 | [TK]D-Fender | CpuID, I've heard PLENTY of issued with AU CID with Zaptel.... |
| 01:23.10 | [TK]D-Fender | issues* |
| 01:23.12 | tzafrir_laptop | well, I just use it as a phone |
| 01:23.47 | tzafrir_laptop | what cid method is used in australia? |
| 01:23.49 | [TK]D-Fender | tzafrir_laptop, well.. yeah! But give the basics a try : 3-way, CW-CID, Blind & attended transfer, etc |
| 01:24.28 | CpuID | cant remember what CID method is used here |
| 01:24.39 | CpuID | the standard landline/fxo's cid is fine here...as i mentioned :) |
| 01:24.58 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 01:25.00 | CpuID | its just the fxs thats the problem, which both ends are local/in my control (to an extent i spose, cant mess with the cordless's settings hehe) |
| 01:34.34 | CpuID | no ideas...? :) |
| 01:38.24 | [TK]D-Fender | CpuID, Go check the WIKi for more info on AU CID and see what you can turn up |
| 01:38.31 | [TK]D-Fender | CpuID, And Google it as well |
| 01:38.37 | CpuID | k |
| 01:39.03 | rob0 | ACID |
| 01:39.18 | rob0 | that's IT ... hallucinating!! |
| 01:49.44 | *** part/#asterisk simonr (n=simonr@mail.ingleinsurance.com) |
| 02:00.36 | *** join/#asterisk deltaray2 (n=deltaray@adsl-76-248-67-30.dsl.bltnin.sbcglobal.net) |
| 02:01.34 | deltaray2 | Hi, I'm just learning asterisk. I have a system that someone setup for me and then he had to move away. Anyways, I've setup another extension on it and I can call out from it, but none of the other phones can call it. Did I miss some configuration some where? |
| 02:03.27 | Notre1 | is there a way to do some sort of distictive ring on zap phones (using Digium FXS card) in lieu of a mwi lamp? |
| 02:08.08 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 02:09.50 | [TK]D-Fender | deltaray2, This is typical of an incorrect setup where NAT is involved |
| 02:10.23 | [TK]D-Fender | deltaray2, is there a NAT between * and that phone? |
| 02:14.12 | deltaray2 | No, its all on the same network segment. |
| 02:14.42 | deltaray2 | All the phones and the asterisk server are on the same segment. |
| 02:15.32 | deltaray2 | To create the new extension, I copied one of the other extension's configs in the config files where I found that extension and changed the relevant values. |
| 02:15.38 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 02:16.11 | MrFollies | What type of phone is it? Softphone, hardware, zap, sip??? |
| 02:16.57 | deltaray2 | Hardware: linksys SPA942. All three phones are this model. |
| 02:17.13 | MrFollies | Is the SPA registering with asterisk OK? |
| 02:17.42 | deltaray2 | It seems like it. Would it need to register to be able to place a call to another phone? |
| 02:18.13 | MrFollies | registering tells asterisk that you are ready to receive calls. |
| 02:19.07 | MrFollies | Run the command : asterisk -r -x "sip show peers" |
| 02:19.11 | MrFollies | What does it tell you? |
| 02:19.36 | deltaray2 | phone3/phone3 192.168.1.56 D N 5060 OK (5 ms) |
| 02:19.42 | deltaray2 | That's the phone I setup. |
| 02:20.35 | MrFollies | Then it's registered OK. The entry in extensions.conf is it the same as the others? |
| 02:20.50 | *** join/#asterisk cymon (n=cymon@pool-71-245-67-120.prvdri.fios.verizon.net) |
| 02:20.54 | MrFollies | What does it look like? |
| 02:21.23 | cymon | okay, quick IAX2 question about FWD |
| 02:21.36 | cymon | I'm getting registration refused messages from their IAX2 host |
| 02:21.47 | cymon | phone number and pass are correct |
| 02:21.54 | cymon | and I've only allowed ulaw as a codec |
| 02:21.55 | CpuID | [TK]D-Fender, sendcalleridafter=2 |
| 02:21.56 | CpuID | :) |
| 02:22.01 | CpuID | http://www.voip-info.org/wiki/view/Australia+Asterisk+Details |
| 02:22.07 | [TK]D-Fender | CpuID, :) |
| 02:22.12 | deltaray2 | MrFollies: Whoops: exten => 1003,1,Dial(SIP/phone2, 15) |
| 02:22.17 | deltaray2 | I missed changing that line. |
| 02:23.25 | deltaray2 | COOL! It works now. Thanks. |
| 02:23.30 | deltaray2 | Phones ARE exciting. |
| 02:23.52 | MrFollies | That line is ideed crutial :) |
| 02:24.46 | MrFollies | phone2 should of course be phone3 |
| 02:25.20 | deltaray2 | Yep |
| 02:26.02 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 02:26.03 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 02:27.10 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 02:47.00 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:49.05 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:51.56 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:57.35 | hmmhesays | ack my polycom ip 320 hung on updating initial configuration |
| 02:58.47 | hmmhesays | when there is no tftp server present anyone else run into this? |
| 03:05.26 | *** join/#asterisk ReD-MaN (i=root-rox@172-220.static.golden.net) |
| 03:06.43 | *** join/#asterisk BBHoss (n=jack@76.73.251.16) |
| 03:08.08 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 03:08.22 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 03:08.36 | objective | Does anyone know why I can see my * console output scrolling by even when i haven't logged into my server? |
| 03:09.22 | fujin | started with -c? |
| 03:12.45 | objective | fujin -- hmm, that could be it... i'll have to check... |
| 03:12.58 | fujin | ps aux|grep asterisk |
| 03:13.00 | fujin | will soon tell you |
| 03:15.38 | *** part/#asterisk phalacee (n=phalacee@123-2-59-211.static.dsl.dodo.com.au) |
| 03:18.17 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 03:18.28 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-12-95.lns10.syd7.internode.on.net) |
| 03:18.35 | objective | fujin -- yup, asterisk -vvvgcd |
| 03:18.50 | objective | thanks |
| 03:19.49 | objective | i only noticed because i kvm'd into a box and could run CLI commands without having to ever login... |
| 03:36.43 | BBHoss | anybody know of an aastra configuration tool |
| 03:38.30 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 03:39.31 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 03:47.08 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 03:47.58 | brimstone | if i have one call, go zap->asterisk -iax->asterisk is there a way i can make the last asterisk system tell the first asterisk system that it can't handle the call and it should continue it's dialplan? |
| 03:48.09 | brimstone | ping Corydon76-dig |
| 03:48.25 | fujin | not cleanly |
| 03:48.37 | brimstone | yeah, autofallthrough isn't working for me |
| 03:49.22 | fujin | although if a dial fails for whatever reason, it'll fall through to the next priority quite happily |
| 03:50.04 | brimstone | it's not though, the first asterisk system just says "oh, a hangup, i like those" and drops the call |
| 03:50.22 | fujin | umm? |
| 03:50.33 | fujin | it shouldn't do, afaik |
| 03:50.42 | fujin | I dial twice for redundancy |
| 03:50.59 | fujin | different peers |
| 03:51.01 | fujin | and that works fine |
| 03:52.07 | brimstone | ah, stupid Answer() got in the way on the 2nd system |
| 03:52.15 | brimstone | no Answer() and it's happy |
| 03:52.21 | brimstone | thanks fujin! i'm not completely crazy |
| 03:52.33 | fujin | haha |
| 03:52.33 | fujin | yeah. |
| 03:52.35 | fujin | awesome ;P |
| 03:54.19 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:56.28 | [T]ank | anyone interested in a sangoma 104d T1 card? looking to sell it. |
| 03:58.18 | hmmhesays | post on the asterisk forums or ebay |
| 03:58.30 | hmmhesays | i've always gotten pretty good prices on ebay |
| 03:58.41 | [T]ank | will do... just offering it here first. thanks |
| 03:58.44 | hmmhesays | mutually beneficial prices I should say |
| 04:02.30 | [T]ank | yeah... i am seeing that too |
| 04:05.13 | *** part/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 04:07.38 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:15.51 | *** join/#asterisk anonymouz667 (i=hoje@201.23.212.246) |
| 04:16.20 | *** part/#asterisk anonymouz667 (i=hoje@201.23.212.246) |
| 04:22.04 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 04:23.53 | *** join/#asterisk bhrobinson (n=Flagg732@198.211.206.162) |
| 04:24.19 | bhrobinson | hey is anyone in here familiar with the astribank? |
| 04:35.25 | *** join/#asterisk Itiliti (n=Itiliti@76.29.84.107) |
| 04:36.15 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 04:36.33 | Itiliti | I am having a "wabble sound " on my server. It is running th eSMP kernal, and the latest Zaptel, asterisk, etc. |
| 04:37.04 | Itiliti | When people are talking it is fine. It is only during quesues, and announcements, etc. |
| 04:37.13 | Itiliti | Anyone have any ideas what could be causing it? |
| 04:39.35 | Corydon76-dig | Are you running X or frame buffer console? |
| 04:39.57 | fujin | Itiliti: check the duplex settings |
| 04:40.08 | Corydon76-dig | Anything that eats interrupts for breakfast could cause that |
| 04:40.42 | drfreeze | Hi |
| 04:40.47 | fujin | Itiliti: are they mp3's? |
| 04:41.12 | drfreeze | Is there an ISO that will install linux and asterisk for a quick start? |
| 04:41.24 | fujin | asterisknow |
| 04:41.45 | fujin | ubuntu && apt-get install asterisk |
| 04:41.45 | fujin | ;P |
| 04:42.19 | nhuisman_work | yeah but you do get a nice installer and web gui with asterisknow |
| 04:42.29 | fujin | meh |
| 04:42.33 | fujin | they're more likely to break stuff. |
| 04:42.44 | fujin | clicky buttons and asterisk in the same sentence is so wrong |
| 04:42.44 | nhuisman_work | unfortunately i wouldn't work with my hardware since the version of glibc was too old |
| 04:42.49 | nhuisman_work | i=it |
| 04:43.14 | drfreeze | jj |
| 04:43.18 | nhuisman_work | i dunno, I think having some basic small subset of commands for day to day use in gui wouldn't be that bad. |
| 04:43.46 | nhuisman_work | definitely most of it should be cli configured |
| 04:44.01 | drfreeze | fujin: :). thanks |
| 04:44.24 | Itiliti | no fX or frame buffer console. |
| 04:44.48 | Itiliti | no they ar enot mp3's. They are just the recordigns in PCM 8bit, 16Khz, mono. |
| 04:44.58 | fujin | checked your duplex? |
| 04:45.09 | nhuisman_work | on the servers soundcard? |
| 04:45.14 | Itiliti | what do you mean duplex? for the soundcard? |
| 04:45.14 | nhuisman_work | do they make half duplex cards these days? |
| 04:45.28 | nhuisman_work | or do you mean on the network connection to the phone/server |
| 04:45.57 | nhuisman_work | time for a crash course in asterisk to replace our current voip |
| 04:46.00 | Itiliti | there is a sound card there, but this isnt an issue on the overhead speakers thetr. it is even happening when people call in and hear the main IVr. Would the Soundcard have something to do with that? |
| 04:46.08 | nhuisman_work | our primary call manager server just died right before I was ready to upgrade. |
| 04:46.58 | Itiliti | I will admit, they do hear some gargling on the overhead speaker throught soundcard, but I thought it would be a zaptel timing issue, or something weird going on with the SMP.. |
| 04:47.18 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a3911ff9bf26304e) |
| 04:47.47 | fujin | I mean the network |
| 04:47.48 | fujin | nic |
| 04:47.49 | fujin | duplex |
| 04:57.08 | *** join/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
| 04:58.02 | n3glv | tzafrir_home, u around m8 ? |
| 05:03.37 | Itiliti | fujin it is running in Full Duplex. on a Gig switch. |
| 05:03.57 | Itiliti | and at the full 1000MBit port. |
| 05:04.41 | Notre1 | How do I play a sound file as a voicemail greeting? |
| 05:05.26 | Itiliti | I do notice when running a ZTTEST, I am getting down to th 70% mark. it iswings between 99 and 70 and then 80 then 70 then 99, then 70, etc. |
| 05:06.03 | *** join/#asterisk fiXXXerMet (n=meep@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 05:06.14 | Itiliti | it is definitely an issue with the zaptel.. How can I trouble shooot it? |
| 05:07.03 | fiXXXerMet | I have a few asterisk questions, if anyone has time. I want to setup a pure VoIP system and use either vonage or voipstreet for my phone number. I have a broadband connect, a VoIP phone, and I can sign up for either service. I don't need any other hardware, do I? |
| 05:07.36 | fiXXXerMet | And if so, does anyone have a good asterisk+gui VoIP howto for ubuntu/debian? |
| 05:09.45 | remmo | apt-get install asterisk |
| 05:09.58 | fiXXXerMet | GUI? |
| 05:10.03 | remmo | whats a gui? |
| 05:10.09 | fiXXXerMet | Like a web interface |
| 05:10.13 | fiXXXerMet | For administration. |
| 05:10.30 | fiXXXerMet | I also want to have this use MySQL to log call information. |
| 05:10.46 | remmo | there really is none web interface that really works well |
| 05:10.58 | remmo | mysql integration is not that hard. www.voip-info.org |
| 05:11.05 | remmo | has all the answers for you |
| 05:11.16 | fujin | There's a free MySQL CDR interface which is quite reasonable |
| 05:11.24 | fujin | although formulating your own would probably be preferable to that. |
| 05:11.31 | fiXXXerMet | How come? |
| 05:11.43 | remmo | cause all the user interfaces are SHIT |
| 05:11.48 | fujin | you get what you need, and nothing that you don't. |
| 05:11.52 | fiXXXerMet | Oh :) |
| 05:12.00 | fujin | asterisk is really set-it-and-forget-it. |
| 05:12.11 | fujin | configure it once off, and you'll never really have to touch it short of adding new devices |
| 05:12.16 | fujin | if you do it right, of course |
| 05:12.21 | fiXXXerMet | I think I'll setup a virtual server for asterisk so if I screw up, I can start over |
| 05:12.37 | *** join/#asterisk Maliuta (n=nikolai@119.11.102.235) |
| 05:12.47 | fiXXXerMet | Unless it really is as easy as apt-get install asterisk |
| 05:12.49 | fiXXXerMet | ? |
| 05:13.04 | fujin | That'll do the basics, and install the samples. |
| 05:13.12 | fujin | I prefer to build asterisk. |
| 05:13.22 | fiXXXerMet | Why? |
| 05:13.30 | fujin | Get everything I do need, and nothing I don't. |
| 05:13.33 | remmo | i prefer to build. i add extra settings to rtp.c |
| 05:13.41 | fiXXXerMet | I always build dspam because it's very site/install/user specific. Is it like that? |
| 05:13.59 | fujin | one would hope so |
| 05:14.06 | fiXXXerMet | Hmm. |
| 05:14.19 | fujin | and the version of asterisk in deb/ubu isn't up-to-date |
| 05:14.31 | fujin | and you can't do cool stuff like drop in russelb's backported func_devstate.c and build it, etc |
| 05:14.33 | fujin | I dunno. |
| 05:14.36 | fiXXXerMet | I'll try building. Do I need zaptel or libpri or anything like that? |
| 05:14.40 | fujin | I wouldn't build Postfix |
| 05:14.48 | fujin | you'll need whatever you need~! |
| 05:14.58 | fiXXXerMet | For pure voip? |
| 05:15.03 | fujin | Just asterisk :) |
| 05:15.13 | fiXXXerMet | great |
| 05:15.25 | [TK]D-Fender | Zaptel is advised |
| 05:15.30 | fiXXXerMet | Why? |
| 05:15.32 | fujin | for app_page, meetme etc. |
| 05:15.44 | fujin | only if you intend on using functionality requiring a dummy timing interface |
| 05:15.54 | [TK]D-Fender | IAX2 trunking, better MoH timing, a few things.. |
| 05:16.01 | fiXXXerMet | Oh yes, I remember now. |
| 05:16.02 | fujin | you need zaptel for iax2 trunking?? |
| 05:16.12 | fujin | I haven't seen any moh issues without ztdummy |
| 05:16.24 | [TK]D-Fender | fujin, Which when you get down to it typically settles as a "hell yeah!" |
| 05:16.26 | remmo | zaptel is only used for timing |
| 05:16.33 | remmo | conferences and trunking iax |
| 05:16.47 | [TK]D-Fender | fujin, It does work BETTER. Fewer timing complains, but it isn't absolutely necessary |
| 05:16.58 | fujin | I see |
| 05:17.11 | fujin | ho-hum |
| 05:19.16 | *** join/#asterisk mechanicus01 (n=none@189.149.76.68) |
| 05:50.18 | fiXXXerMet | Question about Call Detail Recording. I see cdr+pgsql, but not cdr_mysql. Should I use cdr_custom? |
| 05:50.30 | fujin | no, cdr_mysql is in asterisk-addons |
| 05:50.53 | fiXXXerMet | I'm in make menuselect right now, so should I unselect all of those? |
| 05:51.01 | fiXXXerMet | And then do I add that later on? |
| 05:51.45 | fujin | [Dec 28 18:50:58] NOTICE[6929]: chan_sip.c:15647 sip_poke_noanswer: Peer 'wxc' is now UNREACHABLE! Last qualify: 0 |
| 05:51.49 | fujin | how can I force another poke? |
| 05:51.51 | fujin | it doesn't even try |
| 05:51.56 | fujin | and I know the peer is standing up |
| 05:52.09 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:55.40 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:56.38 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583811.dsl.bell.ca) |
| 06:07.51 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-653c61810c7fb57d) |
| 06:15.05 | nhuisman_work | does anyone know how to use an mgcp gateway |
| 06:15.25 | nhuisman_work | i have a cisco vg200 with a t1 pri in it and i'm wondering if asterisk can talk to it |
| 06:16.07 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-10a05256521fa020) |
| 06:23.56 | bhrobinson | tzafrir_home are you around? I need a hand with your Xorcom driver |
| 06:24.10 | nhuisman_work | hey how are you folks partitioning your asterisk boxes? |
| 06:24.25 | nhuisman_work | ... /boot, /, and /var ? |
| 06:25.00 | bhrobinson | I have all but the initialize_registers to kick in |
| 06:26.22 | bhrobinson | I am not getting the XPD folder in the proc/xpp/XBUS |
| 06:26.34 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 06:36.16 | nhuisman_work | grr wtf |
| 06:36.24 | nhuisman_work | asterisk be installer crashed with a python error |
| 06:39.53 | *** join/#asterisk [Outcast] (n=bill@219-89-206-239.adsl.xtra.co.nz) |
| 06:44.14 | *** join/#asterisk mechanicus01 (n=none@189.149.76.68) |
| 06:52.41 | Corydon76-dig | nhuisman_work: are you using one of the supported distros? |
| 06:52.50 | nhuisman_work | it is the cd that came with it |
| 06:52.53 | nhuisman_work | rpath linux |
| 06:53.01 | Corydon76-dig | Oh, okay |
| 06:53.10 | Corydon76-dig | Call support in the morning |
| 06:53.18 | nhuisman_work | yeah trying the installer one more time |
| 06:53.24 | nhuisman_work | i saw one bug report of the same error |
| 06:53.28 | nhuisman_work | and then they couldn't repeat it |
| 06:53.31 | nhuisman_work | so trying mine one more time |
| 06:54.03 | nhuisman_work | ugggg |
| 06:54.32 | nhuisman_work | same crap |
| 06:55.00 | nhuisman_work | maybe i'll try expert install instead of custom |
| 06:56.53 | fiXXXerMet | OK, I've setup asterisk, added an extension, and plugged in my ip phone, which has an ip address. Now what? :) |
| 06:56.56 | nhuisman_work | Corydon76-dig, do they supply some sort of source or packages on the cds they give you? |
| 06:57.10 | nhuisman_work | Corydon76-dig, to let you install it on a different version of linux |
| 06:57.51 | craigk | anybody know why i can not re-park a parked call that i have picked up ? |
| 07:00.56 | hmmhesays | cause its a shitty bug |
| 07:01.00 | fiXXXerMet | omg. the phone itself has a web interface! |
| 07:01.05 | fiXXXerMet | i just came. |
| 07:04.22 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 07:09.18 | *** join/#asterisk harpal (n=Harpal@124.125.79.212) |
| 07:10.38 | fiXXXerMet | Ugh, still can't figure out how to register my phone to asterisk |
| 07:13.42 | fiXXXerMet | What is an Auth ID (SIP ID)? |
| 07:16.31 | nhuisman_work | interesting |
| 07:16.40 | nhuisman_work | using express mode works |
| 07:16.42 | nhuisman_work | feh |
| 07:18.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-df7d59ed3aedd97b) |
| 07:18.41 | ReDNeQ | sup everyone |
| 07:19.19 | fiXXXerMet | yao |
| 07:19.51 | ReDNeQ | happy holidays/new year |
| 07:20.09 | marexz | thnx |
| 07:20.53 | fiXXXerMet | thank you |
| 07:21.08 | *** join/#asterisk dkatz334 (n=guest@66.238.199.82.ptr.us.xo.net) |
| 07:21.09 | nhuisman_work | hey does anyone know if it is possible to use mgcp as a gateway |
| 07:21.15 | nhuisman_work | not just as a phone interface |
| 07:21.18 | nhuisman_work | ie a t1 pri gateway |
| 07:21.25 | fiXXXerMet | Can anyone help? I need help setting up my first phone (linksys spa941) with my first asterisk install |
| 07:21.36 | nhuisman_work | it seems like asterisk is only setup to use mgcp for endpoints |
| 07:21.42 | fiXXXerMet | Do I need to setup some kind of SIP ID or something? |
| 07:21.49 | ReDNeQ | yes fiXXXerMet |
| 07:21.57 | ReDNeQ | you need to setup a SIP account for the phone |
| 07:22.03 | ReDNeQ | with the asterisk server |
| 07:22.16 | fiXXXerMet | Os that all within sip.conf? |
| 07:22.29 | ReDNeQ | some.. |
| 07:22.48 | ReDNeQ | you need to read up on this if this is really your first time. or you may want to try FreePBX as well |
| 07:22.58 | ReDNeQ | its a grapchical interface to help people that are new to this |
| 07:23.11 | fiXXXerMet | I installed asterisk-gui as well but I don't see any sip conf stuff there |
| 07:23.52 | ReDNeQ | yea. im not too familiar with the asterisk-gui.. does it have a section that uses or calls them extensions? |
| 07:24.29 | fiXXXerMet | It has a users section, where I added an extension |
| 07:25.29 | fiXXXerMet | hmm, now I have a dial tone! |
| 07:28.38 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 07:33.32 | fiXXXerMet | I wish my phone would show my name. :( |
| 07:38.43 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 07:39.58 | *** join/#asterisk Sargun (i=nobody@atarack/staff/sargun) |
| 07:40.05 | Sargun | haven't visited you guys in forever. |
| 07:40.32 | fujin | hello |
| 07:40.34 | fujin | sargun |
| 07:40.36 | fujin | how are you today? |
| 07:40.44 | Sargun | good, you? |
| 07:41.00 | fujin | I'm not too bad. |
| 07:41.05 | fujin | on a wild hunt to find some marijuana. |
| 07:41.09 | fujin | Not here, obviously :) |
| 07:41.27 | fujin | are you an IRC operator here? |
| 07:42.31 | Sargun | Nope. |
| 07:43.12 | n3glv | why u need an op? |
| 07:46.29 | fujin | no, just wondering lol |
| 07:46.33 | fujin | i though the staff/ |
| 07:46.52 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b18580aa9c5ba7a8) |
| 07:47.31 | denon | he figures he'll only yack about weed in the channel if there's nobody around who will kick him out |
| 07:47.52 | fujin | LOL |
| 07:47.55 | fujin | no way~ |
| 07:48.20 | fujin | something wrong with weed? |
| 07:48.21 | fujin | l> |
| 07:48.32 | denon | yes |
| 07:50.26 | denon | I'm headed to bed - if you'd like a detailed explanation, call the police, give them your current location, and tell them you've got a bag of weed |
| 07:50.31 | denon | they'll fill you in |
| 07:51.32 | drmessano | I am a pretty free thinking person.... |
| 07:51.35 | drmessano | But... |
| 07:51.45 | drmessano | Legalizing stupidity was a bad move |
| 07:52.44 | *** join/#asterisk bmcghee`home (n=brentmcg@S010600179a29f419.ok.shawcable.net) |
| 07:53.16 | bmcghee`home | CLI is saying "<SIP/2590-b74056f0> Playing 'vm-password' (language 'en')" but there is no audio coming out |
| 07:54.28 | *** join/#asterisk NolanG (n=ngarrett@75.148.58.161) |
| 07:56.01 | fiXXXerMet | Setting up my VoicePulse account now. |
| 07:58.17 | nhuisman_work | bah |
| 07:58.25 | fiXXXerMet | bah? |
| 07:58.28 | nhuisman_work | not to yo |
| 07:58.34 | fiXXXerMet | phew |
| 07:58.38 | nhuisman_work | stupid distro that asterisk be comes with has an old version of glibc too |
| 08:01.05 | nhuisman_work | man this is lame |
| 08:01.25 | nhuisman_work | this stupid redfone gateway software requires a new version of glibc and none of the supported asterisk be distros have it |
| 08:04.47 | *** part/#asterisk NolanG (n=ngarrett@75.148.58.161) |
| 08:07.09 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 08:12.22 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 08:16.26 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:22.28 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 08:29.07 | fujin | nhuisman_work |
| 08:29.11 | fujin | install the new glibc in a chroot |
| 08:29.15 | fujin | debootstrap or something |
| 08:29.24 | fujin | is it really necesary to run the redfone softwar on the same server? |
| 08:33.41 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:35.01 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 08:41.00 | *** join/#asterisk dkatz334 (n=guest@66.238.199.82.ptr.us.xo.net) |
| 08:41.56 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 08:43.43 | Alexandre_fr | somebody knows how to handle the url in the queue ? |
| 08:44.36 | Alexandre_fr | I put an url in the command queue but when the agent answer I don't have anithing |
| 08:48.07 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 08:49.16 | FlatFoot | morning all |
| 08:56.36 | Sargun | hi |
| 08:56.39 | Sargun | URL in the queue |
| 08:59.28 | *** join/#asterisk kiscokid (n=kiscokid@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 09:03.29 | Alexandre_fr | exten => 1,n,Queue(support|t|www.google.com) |
| 09:03.45 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:05.22 | Alexandre_fr | but when my agent answer there is no popup of a browser, or an url on the softphone display |
| 09:06.42 | nhuisman_work | fujin is that very hard? Running it in a chroot |
| 09:26.52 | *** join/#asterisk RoyK (n=roy@ip-213-15-149-91.dialup.ice.no) |
| 09:29.22 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:30.27 | badcfe | can i use #include file in extensions.conf |
| 09:30.28 | badcfe | ? |
| 09:30.54 | badcfe | i have many related contexts that i want in a separate file |
| 09:31.55 | *** join/#asterisk mechanicus01 (n=none@189.149.76.68) |
| 09:32.01 | *** join/#asterisk b_d (n=brian@209.240.42.151) |
| 09:33.05 | *** part/#asterisk kiscokid (n=kiscokid@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
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| 09:55.09 | badcfe | i want to include a whole file as dialplan, how do i do this? |
| 09:56.03 | *** join/#asterisk qdk_ (n=qdk@195.242.194.41) |
| 10:03.32 | RoyK | badcfe: see the samples |
| 10:03.45 | RoyK | badcfe: yes, you can use #include |
| 10:06.44 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 10:06.46 | shadebob | hi |
| 10:17.43 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 10:21.50 | *** join/#asterisk R1ck (i=rick@belphegor.deadlysins.nl) |
| 10:22.01 | R1ck | hello, any idea why outbound cid doesnt work, the interface says to use the magic string 'hidden' to hide it, but that doesnt work, also, if I set one of our other numbers, it doesnt work, it only (and always) uses our main number for outbound dialing.. the line is an isdn2 line, and I use a junghanns.net quadbri ISDN card |
| 10:22.57 | FlatFoot | R1ck: are you in the uk ? if so have you got TON5 setup on your lines ? |
| 10:23.12 | R1ck | no i'm in .nl |
| 10:23.42 | R1ck | whats TON5, could it be that there is something similar here? |
| 10:23.56 | FlatFoot | ah ok , in the uk you need to sign a ton5 agreement so that you can present any outgoing number BUT agree that you will not send daft numbers etc |
| 10:24.04 | R1ck | ah right |
| 10:24.51 | R1ck | the weird thing is that it does work when I connect the lines to our old Siemens PBX, it sends a different number and is able to call out with different numbers |
| 10:25.30 | FlatFoot | in that case i would check with the isdn supplier that you are able to send out many numbers |
| 10:25.54 | R1ck | i can, the Siemens pbx can do it.. so should Asterisk.. |
| 10:26.14 | FlatFoot | does the seimens use the same line then ? |
| 10:26.37 | R1ck | yes |
| 10:26.58 | FlatFoot | can you show your outgoing context ? pastebin |
| 10:27.01 | R1ck | i either patch the lines to the Siemens box or the Asterisk box |
| 10:27.01 | FlatFoot | ~pb |
| 10:27.02 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:27.28 | R1ck | hmm just a minute |
| 10:31.30 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 10:31.42 | *** part/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 10:31.57 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 10:33.41 | R1ck | ugh, its too complex (trixbox) |
| 10:34.22 | FlatFoot | R1ck: trixbox , i should wander off to #trixbox for help on that |
| 10:34.48 | mvanbaak | ~trixbox |
| 10:34.49 | jbot | trixbox is probably a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 10:35.27 | R1ck | yeah I tried there but nobody is responding |
| 10:37.39 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 10:38.09 | mvanbaak | probably too busy to fix their own systems ;) |
| 10:38.21 | FlatFoot | lol |
| 10:40.14 | R1ck | i guess :) |
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| 10:49.04 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b18580aa9c5ba7a8) |
| 10:57.35 | nhuisman_work | anyone know where to start in installing linux-ha on rpath linux? |
| 10:57.40 | nhuisman_work | (the distro that comes with abe) |
| 10:57.58 | nhuisman_work | guess just install it via source |
| 10:58.09 | mvanbaak | apt-get install heartbeat2 ? |
| 10:58.21 | mvanbaak | geez, cant type this morning |
| 11:00.37 | nhuisman_work | um rpath doesn't have apt-get |
| 11:00.49 | mvanbaak | ow |
| 11:00.55 | mvanbaak | any other package manager ? |
| 11:00.59 | nhuisman_work | conary |
| 11:01.13 | nhuisman_work | which points at digiums repository which only contains asterisk stuff |
| 11:01.42 | mvanbaak | maybe you can add the default repositories ? |
| 11:01.53 | nhuisman_work | trying to find a repository |
| 11:02.09 | mvanbaak | http://wiki.rpath.com/wiki/Conary |
| 11:02.57 | nhuisman_work | looks pretty doubtful |
| 11:03.30 | mvanbaak | I dont want to look into it ;) |
| 11:04.00 | nhuisman_work | i think I need to call them tomorrow to see what options I have |
| 11:04.24 | mvanbaak | you can call digium for free with IAX |
| 11:05.02 | mvanbaak | or go for the source way |
| 11:05.11 | nhuisman_work | well I did buy their software so I should be able to call them |
| 11:05.11 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:05.29 | nhuisman_work | kind of disappointing that the other distros they support are fedora and rhel |
| 11:05.33 | nhuisman_work | 3 and 4 |
| 11:05.53 | badcfe | how do i set multiple variables in ami, for example an originate action? |
| 11:06.09 | badcfe | how do i separate the assignments? |
| 11:07.00 | mvanbaak | yeah |
| 11:07.08 | mvanbaak | they should start supporting debian |
| 11:08.00 | nhuisman_work | no kidding |
| 11:08.07 | nhuisman_work | or a new friggin version of fedora |
| 11:08.43 | mvanbaak | I tried fedora but it's not my fav |
| 11:08.54 | FlatFoot | try FreeBSD |
| 11:09.01 | nhuisman_work | i'm just trying to find a way to upgrade to glibc 2.4.x |
| 11:09.02 | FlatFoot | thats what we run works a treat |
| 11:09.15 | nhuisman_work | doesn't look like I will be able to using the "supported" distros |
| 11:09.25 | nhuisman_work | unless I figure out a way to upgrade glibc in a chroot |
| 11:10.01 | mvanbaak | FlatFoot: we use debian and openbsd |
| 11:10.10 | mvanbaak | FlatFoot: debian where we need PRI hardware |
| 11:10.14 | mvanbaak | openbsd otherwise |
| 11:10.29 | FlatFoot | FreeBSD is able to run pri np |
| 11:10.48 | FlatFoot | i am running a 2 port pri card on FreeBSD 4.2 |
| 11:10.59 | nhuisman_work | i like debian quite a bit more then most distros |
| 11:11.04 | nhuisman_work | ubuntu for my desktop use |
| 11:11.06 | FlatFoot | the old * was debian but the hardware has seen better days |
| 11:12.09 | mvanbaak | FlatFoot: I know, but we already have to support 2 different operating systems |
| 11:12.14 | mvanbaak | that's enough for us |
| 11:12.55 | mvanbaak | and like nhuisman_work I'm running ubuntu on my laptops |
| 11:15.58 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:19.57 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:20.02 | tzafrir_home | bhrobinson, here? |
| 11:20.03 | FlatFoot | mvanbaak: i have oly in the past year started to use freebsd , come from a windoze backgroud |
| 11:20.10 | FlatFoot | *background |
| 11:20.20 | mvanbaak | ah |
| 11:20.52 | mvanbaak | I started with Debian back in the days where you had kernel only unnumbered releases of debian |
| 11:20.57 | FlatFoot | all the other blokes here are UNIX from the start |
| 11:21.18 | mvanbaak | then couple of years ago I started with openbsd for redundant firewalling (2 weeks after they imported pf) |
| 11:21.24 | FlatFoot | my first intro to the world of computers was a VIC20 |
| 11:21.35 | mvanbaak | mine a Sinclair Spectrum |
| 11:21.41 | FlatFoot | about 25 years back |
| 11:21.57 | mvanbaak | I still have my sinclair. and it's still working |
| 11:22.08 | FlatFoot | ah but i had the 16k ram pack with mine , it was the rolls royce of the vic20 |
| 11:22.27 | mvanbaak | 12:21 < FlatFoot> about 25 years back |
| 11:22.36 | mvanbaak | I was 4 back then |
| 11:22.41 | FlatFoot | swapped mine for a stereo |
| 11:22.41 | mvanbaak | so no computer for me ;) |
| 11:22.47 | FlatFoot | well i'm an old git m8e |
| 11:23.00 | FlatFoot | i was 14 |
| 11:23.01 | mvanbaak | git clone FlatFoot |
| 11:23.03 | mvanbaak | ;) |
| 11:23.34 | mvanbaak | I only played on the sinclair for 6 months. Then my dad bought a 386 |
| 11:23.47 | mvanbaak | with a whopping 1 MB of RAM and a 40MB harddrive |
| 11:24.08 | FlatFoot | i moved on to a commodore 64 |
| 11:24.25 | mvanbaak | we had a commodore 64 as well, but only for games and game programming |
| 11:24.29 | FlatFoot | i still have my monochrome 286 laptop ( orange screen ) |
| 11:24.39 | mvanbaak | the 386 was for the real work. qbasic and all |
| 11:24.51 | FlatFoot | i used to use that to decode sattelite television |
| 11:25.54 | mvanbaak | and when I went to collage I got my IBM Cyrix 166 mhz. There I moved to debian and never left the *NIX side |
| 11:26.30 | mvanbaak | only windows pc I sometimes have to touch is the one on my parents house |
| 11:26.34 | FlatFoot | ah i left the world of computers until i started work at the channel tunnel programming in access v1 |
| 11:26.52 | FlatFoot | that was about 15 years back |
| 11:27.10 | nhuisman_work | you know, I guess I could just run asterisk in a non-supported distro to get glibc 2.4 |
| 11:27.18 | nhuisman_work | i mean asterisk business edition |
| 11:27.29 | nhuisman_work | what's the point of paying for support then though, i guess. |
| 11:27.48 | mvanbaak | nhuisman_work: I wont run ABE on a non-supported distro |
| 11:27.57 | FlatFoot | is anyone actually working today ? or are we all in reminising mode ? |
| 11:28.11 | mvanbaak | I'm working a little bit |
| 11:28.21 | mvanbaak | only support calls get attention |
| 11:28.28 | mvanbaak | all the other stuff is suspended till Jan 7 |
| 11:28.40 | nhuisman_work | mvanbaak, are you a digium employee? |
| 11:28.47 | mvanbaak | nhuisman_work: unfortunatelly no |
| 11:29.09 | mvanbaak | nhuisman < |
| 11:29.09 | nhuisman_work | guess I need to call redfone and say wtf, make me a glibc 2.3 version of your latest software |
| 11:29.14 | mvanbaak | nhuisman <-- dutch ? |
| 11:29.19 | nhuisman_work | sounds that way doesn't it |
| 11:29.23 | mvanbaak | yup |
| 11:29.25 | nhuisman_work | pretty sure my family came from there |
| 11:29.29 | nhuisman_work | i'm in Hawaii |
| 11:29.36 | mvanbaak | yeah, noticed that |
| 11:29.39 | mvanbaak | 12:28 host | n=nhuisman@aeko.IfA.Hawaii.Edu |
| 11:30.23 | mvanbaak | I do know how digium works, so I dont think you'll get support when running ABE on non-supported OS |
| 11:30.44 | mvanbaak | hhmm, RHEL does not include ha ? |
| 11:31.01 | nhuisman_work | yeah I think I can get ha working with rhel |
| 11:31.05 | nhuisman_work | with a separate repository |
| 11:31.19 | FlatFoot | mvanbaak: your website is very LOUD ;) |
| 11:31.27 | mvanbaak | it is ? |
| 11:31.44 | mvanbaak | what's loud about it ? |
| 11:31.44 | FlatFoot | xs4all.nl , or is that just your provider ? |
| 11:31.55 | mvanbaak | ah |
| 11:31.55 | mvanbaak | yeah |
| 11:31.56 | nhuisman_work | I need to figure out how to run glibc in a chroot |
| 11:31.58 | mvanbaak | that's my provider |
| 11:32.02 | mvanbaak | it's ugly |
| 11:32.11 | mvanbaak | FlatFoot: mine is http://michiel.vanbaak.info |
| 11:32.19 | FlatFoot | i now have a headache looking at that |
| 11:32.36 | mvanbaak | FlatFoot: yeah, xs4all website always makes me grab my sunglasses |
| 11:32.54 | FlatFoot | yours is very subtle choice of colours |
| 11:33.03 | FlatFoot | we are www.orbital.net |
| 11:33.26 | mvanbaak | nice |
| 11:33.33 | mvanbaak | my company is http://www.terrazur.nl |
| 11:33.52 | mvanbaak | hahahahahaha |
| 11:34.04 | mvanbaak | orbital needs to start using stripslashes() |
| 11:34.11 | mvanbaak | ORBITAL chooses \"THE BUNKER\" - |
| 11:34.16 | mvanbaak | OUCH |
| 11:34.52 | FlatFoot | didn't notice that . don't really look at the site much |
| 11:35.00 | FlatFoot | normally out working too much |
| 11:35.06 | mvanbaak | yeah |
| 11:35.14 | mvanbaak | I dont look at our website neither |
| 11:35.29 | mvanbaak | and I'm not responsible for it, so I dont bother |
| 11:35.47 | FlatFoot | spend most of my time building a wireless network around the county |
| 11:36.04 | mvanbaak | I spend most of my time programming |
| 11:36.24 | FlatFoot | don't get to do so much programming now |
| 11:36.29 | FlatFoot | kinda miss it |
| 11:36.37 | rob0 | "I am a wireless lineman for the county"? |
| 11:36.43 | mvanbaak | lol |
| 11:38.20 | FlatFoot | rob0: what gear do you use ? |
| 11:40.45 | rob0 | nonono I was just trying to sing the Glen Campbell song. |
| 11:41.03 | FlatFoot | ah ok |
| 11:44.39 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 11:45.08 | FlatFoot | rob0: i do use linesman pliers if that helps |
| 11:54.52 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 12:07.34 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:12.09 | *** join/#asterisk Skarmeth (n=Skarmeth@201.9.74.32) |
| 12:37.48 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 12:38.44 | *** join/#asterisk coppice (n=chatzill@137.192.17.210.dyn.pacific.net.hk) |
| 12:39.56 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
| 13:02.52 | *** join/#asterisk carrello (n=salvator@81-174-56-54.static.ngi.it) |
| 13:03.25 | carrello | hi all |
| 13:03.34 | mvanbaak | yo |
| 13:03.45 | *** join/#asterisk Tourinho (n=tourinho@fw01.telehumana.com) |
| 13:03.50 | Tourinho | good day all |
| 13:04.02 | mvanbaak | ello |
| 13:04.03 | *** join/#asterisk CVirus (n=GoD@82.201.178.84) |
| 13:04.19 | carrello | I write from italy |
| 13:04.31 | mvanbaak | .nl here |
| 13:04.33 | Tourinho | when I start my Asterisk, it start a mpg123 process automatically, where can I found this rule? I looked in /etc/asterisk and didnt found anything |
| 13:04.47 | mvanbaak | Tourinho: musiconhold.conf |
| 13:05.03 | Tourinho | mvanbaak: thanks... Ill take a look |
| 13:07.26 | Tourinho | mvanbaak: ok.. i found it.. just the last question, after some time running (ie. 1 month) the mpg123 process doent stop running.. doesnt asterisk need to kill the process? |
| 13:07.48 | mvanbaak | it does |
| 13:08.22 | Tourinho | if I run ps aux on my system, i can see more then 20 mpg123 process |
| 13:08.27 | mvanbaak | normally asterisk kills the mpg123 process |
| 13:08.32 | mvanbaak | but it goes wrong sometimes |
| 13:08.36 | mvanbaak | I noticed that too |
| 13:08.47 | Tourinho | humm.. right |
| 13:08.50 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:08.54 | mvanbaak | dont use mpg123 |
| 13:11.01 | mvanbaak | use the file based moh |
| 13:11.09 | mvanbaak | http://svn.digium.com/view/asterisk/branches/1.4/configs/musiconhold.conf.sample?view=markup |
| 13:11.14 | mvanbaak | that has some nice text about it |
| 13:12.44 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 13:19.22 | Tourinho | mvanbaak: ok.. Ill take a look, tahnks |
| 13:19.35 | Tourinho | had to go now.. thanks for your help. cya |
| 13:19.41 | *** join/#asterisk kn0x (n=pinochle@75.127.83.151) |
| 13:19.44 | mvanbaak | latero |
| 13:22.11 | *** join/#asterisk biw (i=ben@colchester-lug/member/Ben) |
| 13:22.53 | biw | hi, I'm writing some macros and I'm jumping from one to another. I can't track down whether variables I define in the first macro are available to be updated once I'm into the second? |
| 13:29.57 | shido6 | so use a no op |
| 13:30.00 | shido6 | to display the variable |
| 13:30.12 | shido6 | so when you are debugging |
| 13:30.16 | shido6 | you can keep track |
| 13:38.54 | tzafrir_home | anybody get to try that Astercon2 ? |
| 13:39.59 | mvanbaak | I dont even know what that is ;) |
| 13:40.56 | tzafrir_home | http://astercon.0420.com/ |
| 13:41.53 | tzafrir_home | announced today on voip-info by James Zhu from OpenVox |
| 13:43.21 | mvanbaak | website is freaking slow |
| 13:43.26 | shido6 | very |
| 13:44.25 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-2211b0f36f816eba) |
| 13:46.12 | shido6 | please tell me its not all in chinese |
| 13:46.44 | *** join/#asterisk ronr (n=ron@82-170-109-196-static.dsl.ip.tiscali.nl) |
| 13:47.02 | mvanbaak | lol |
| 13:47.03 | mvanbaak | indeed |
| 13:47.21 | mvanbaak | waiting 68 seconds for a screenshot just to look at some chinese characters |
| 13:47.29 | Qwell | that name won't last long |
| 13:47.50 | mvanbaak | gheh |
| 13:48.06 | ronr | in asterisk 1.4, how do can I do something conditional on timeout of Dial, so Dial(<something>, 10) success Hangup(), timeout, Dial(<something else>, 20)? |
| 13:48.55 | mvanbaak | Dial will end dialplan processing once the call is connected |
| 13:49.15 | mvanbaak | so as long as the call is not connected in 10 seconds it will move on to the next priority in your dialplan |
| 13:49.26 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 13:49.41 | ronr | ah, I see, actually a lot easier than I though :) |
| 13:50.12 | mvanbaak | ronr: maybe it's a good idea that you reed THE book first |
| 13:50.14 | mvanbaak | ~book |
| 13:50.14 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 13:51.25 | shido6 | this scares me |
| 13:51.26 | shido6 | we allow Chinese word and English word mix on page! |
| 13:51.34 | jengelh | haha |
| 13:51.52 | shido6 | I dont want to configure something and hit a brick wall because the 2nd half is in chinese |
| 13:52.12 | mvanbaak | gheh |
| 13:52.50 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 13:53.16 | Pagautas | hey |
| 13:53.28 | Pagautas | anybody online? |
| 13:53.40 | jengelh | No, I just rebooted the Internet. |
| 13:53.52 | hmmhesays | Then I pulled the ups out |
| 13:53.53 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 13:53.55 | Pagautas | i've updated asterisk to 1.4.16.2 and host based auth doesnt work anymore |
| 13:54.11 | Pagautas | is there any way to work around this? |
| 13:54.21 | Pagautas | i'm using asterisk with cisco router |
| 13:54.42 | hmmhesays | define host based auth? |
| 13:54.54 | Pagautas | and cisco router is authorized by host not by user and passwd |
| 13:54.56 | markit | hi, I would like that incoming call wait 10 seconds then I answer with a message... I use wait(10), but caller don't hear dialtone during waiting... what can I do? is it normal? |
| 13:55.25 | hmmhesays | you want the caller to hear dialtone for 10 seconds? |
| 13:55.38 | markit | hmmhesays: yes, during the wait time |
| 13:55.48 | shido6 | wtf |
| 13:55.49 | markit | now is silence until my message plays |
| 13:55.52 | hmmhesays | are you looking for user input at that point? |
| 13:55.55 | Pagautas | hmmhesays: host based i mean when username and passwd is not used, just ip address |
| 13:56.01 | markit | hmmhesays: no, just have to wait |
| 13:56.14 | markit | hmmhesays: and I don't want to answer (yet) |
| 13:56.15 | hmmhesays | ~playtones |
| 13:56.15 | mvanbaak | why wait 10 seconds ? |
| 13:56.23 | tzafrir_home | mvanbaak, it may eb not so trivial: the database also needs to know how to store and process multi-byte characters |
| 13:56.24 | shido6 | here we go....... |
| 13:56.32 | markit | mvanbaak: is just an example |
| 13:56.33 | tzafrir_home | Previous versions of mysql had problems with that |
| 13:56.34 | hmmhesays | bah doesn't jbot have the cmd list |
| 13:56.40 | hmmhesays | markit, look at playtones |
| 13:57.02 | markit | hmmhesays: thanks (does it work even if I've no answered? ) |
| 13:57.07 | mvanbaak | tzafrir_home: it depends on what backend you use |
| 13:57.11 | hmmhesays | you have to answer to play audio |
| 13:57.19 | badcfe | im sorry, what do one normally call the # character in the us? |
| 13:57.19 | mvanbaak | myisam and innodb both support utf-8 storage |
| 13:57.27 | hmmhesays | badcfe "pound" |
| 13:57.32 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 13:57.37 | badcfe | hmmhesays: thanks |
| 13:57.39 | hmmhesays | you must be one of those "hash" people |
| 13:57.40 | markit | hmmhesays: mmm no, I just want it to work like a regular phone... you call someone and you hear dialtone until someone answers |
| 13:57.46 | tzafrir_home | Right. But it a bit more complicated than "HTML allows unicode so it should just work" |
| 13:57.56 | hmmhesays | markit, I think you are trying to say ring tone |
| 13:58.06 | mvanbaak | tzafrir_home: tell me about it |
| 13:58.09 | markit | hmmhesays: right! sorry,ring tone! |
| 13:58.18 | hmmhesays | your phone should generate the ring tone |
| 13:58.20 | mvanbaak | tzafrir_home: I'm core dev of a webbased CRM tool |
| 13:58.22 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com) |
| 13:58.32 | tzafrir_home | ah, I forgot |
| 13:58.44 | mvanbaak | we use UTF-8 as default charset |
| 13:58.46 | hmmhesays | as long as you don't answer the call first |
| 13:59.04 | mvanbaak | been a pain to switch from LATIN-1 to UTF-8 |
| 13:59.06 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 13:59.11 | markit | hmmhesays: voip provider -> asterisk -> mysipphone |
| 13:59.14 | mvanbaak | all kind of detection and conversion shit |
| 13:59.33 | hmmhesays | and on what side of the call are you having the ring tone problem? |
| 13:59.57 | markit | hmmhesays: Someone calls (no ring tones!) -> voip provider -> asterisk -> [wait(19) -> mysipphone |
| 14:00.55 | hmmhesays | is there a reason you aren't using cmd dial to call 'mysipphone' right away? |
| 14:01.02 | badcfe | hmmhesays: and the * character is it commonly called star, or asterisk? |
| 14:01.12 | hmmhesays | badcfe star |
| 14:01.39 | markit | hmmhesays: yes, I would wite a certain number of time based upon other condition, and after that time answer / play a message / send to voicemail |
| 14:01.56 | hmmhesays | markit, you might try cmd ringing before your wait |
| 14:02.11 | markit | hmmhesays: thanks, I try and let you know :) |
| 14:02.22 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:02.23 | Pagautas | so any help with asterisk and cisco router? |
| 14:02.53 | hmmhesays | you are having the host based auth problem? |
| 14:02.58 | hmmhesays | what exactly is failing? |
| 14:03.02 | Pagautas | yes |
| 14:03.22 | Pagautas | Failed to authenticate user |
| 14:03.38 | hmmhesays | pastebin a sample sip.conf entry |
| 14:04.18 | markit | hmmhesays: perfect!!! thanks very very much |
| 14:04.31 | hmmhesays | markit, that works? |
| 14:04.44 | markit | hmmhesays: yes, like a charm :) |
| 14:05.14 | Pagautas | http://pastebin.ca/834804 |
| 14:05.55 | *** join/#asterisk PepOSX (n=pepOSX--@201.248.215.16) |
| 14:05.56 | Pagautas | my network looks like [many sip users] --- [asterisk] -- [cisco] --- [pstn] |
| 14:06.10 | Pagautas | users cant get call from pstn |
| 14:06.31 | Pagautas | cisco has not way to auth with username/passwd |
| 14:06.51 | markit | btw, when I record the call (voicemail), I have a flow of these messages in the console: Dec 28 15:06:16] WARNING[8902]: rtp.c:891 ast_rtcp_read: RTCP Read too short |
| 14:07.17 | Pagautas | cant turn allowguest=yes because server is public accessable |
| 14:07.49 | Qwell | Pagautas: using realtime? |
| 14:07.59 | markit | (and the message is recorded briefly until this flow of WARNINGS start, sigh) |
| 14:08.02 | Qwell | if so, use latest 1.4 svn |
| 14:08.24 | hmmhesays | good lord this computer is dying |
| 14:09.05 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 14:09.16 | hmmhesays | are you sure the ip you are trying to register from is the same ip you have in the host field? |
| 14:09.28 | Pagautas | yes i'm sure |
| 14:09.49 | hmmhesays | change type from friend to peer |
| 14:09.53 | hmmhesays | try again |
| 14:11.51 | bhrobinson | tzafrir_home are you there? |
| 14:12.23 | *** join/#asterisk carrello (n=salvator@81-174-56-54.static.ngi.it) |
| 14:13.15 | Pagautas | nothing changes |
| 14:13.17 | Pagautas | Failed to authenticate user "123456" <sip:123456@cisco_ip> |
| 14:13.32 | carrello | hi all again |
| 14:14.32 | hmmhesays | wait, you are regexten=<some_ip> ? |
| 14:14.56 | hmmhesays | try [123456] regexten=123456 |
| 14:15.03 | hmmhesays | that would make more sense to me |
| 14:16.10 | Pagautas | hmmhesays: but what if number is not 123456 |
| 14:16.44 | hmmhesays | so you want any number coming from <cisco_ip> to be registered? |
| 14:17.17 | carrello | I have a problem: gsm files played by background on asterisk 1.4.14 sounds very bad when a call arrive, while musiconhold (wav files) no...why according to you? |
| 14:17.29 | *** part/#asterisk biw (i=ben@colchester-lug/member/Ben) |
| 14:17.58 | Pagautas | my config where working well untill i've updated from 1.4.4 to 1.4.16.2 |
| 14:18.30 | hmmhesays | hmm I have no idea |
| 14:18.33 | hmmhesays | sorry |
| 14:19.57 | carrello | ops...my version is 1.4.16.2 |
| 14:20.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.06 | *** part/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 14:22.27 | hmmhesays | carrello: what does your cpu usage look like while you are playing back those files? |
| 14:23.22 | carrello | i have to see |
| 14:25.37 | carrello | hmmmm: top shows no load |
| 14:25.58 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:27.31 | hmmhesays | ok so thats not a problem |
| 14:27.49 | Pagautas | heh i've found what was wrong |
| 14:27.55 | hmmhesays | Pagautas: do tell |
| 14:28.02 | Pagautas | insecure=invite,port was missing |
| 14:28.10 | carrello | hmmm: i don't believe; I'm trying only with a call |
| 14:28.26 | Pagautas | everything else left the same |
| 14:28.27 | hmmhesays | Pagautas: that doesn't make much sense in regards to a Register |
| 14:28.35 | hmmhesays | or is it registering fine and failing on invite |
| 14:28.49 | hmmhesays | is/was |
| 14:29.12 | mvanbaak | hmmhesays: I dont think it has anything to do with registering |
| 14:29.19 | Pagautas | hmmhesays: cisco router doesnt register to asterisk |
| 14:29.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:29.38 | hmmhesays | mvanbaak: thats where I got confused, I thought it was failing auth on register |
| 14:29.43 | mvanbaak | no |
| 14:29.46 | Pagautas | no |
| 14:30.15 | *** join/#asterisk ronr (n=ron@82-170-109-196-static.dsl.ip.tiscali.nl) |
| 14:30.26 | hmmhesays | thats what I get for not paying attention |
| 14:30.31 | mvanbaak | lol |
| 14:30.40 | Pagautas | :) |
| 14:31.23 | carrello | I have a problem: gsm files played by background on asterisk 1.4.16.2 sounds very bad when a call arrive, while musiconhold (wav files) no...why according to you? |
| 14:31.29 | *** join/#asterisk lmoreira (i=TieFight@189.70.243.31) |
| 14:31.59 | mvanbaak | carrello: transcoding I think |
| 14:32.13 | mvanbaak | carrello: try using wav files for background as well |
| 14:32.27 | hmmhesays | he has, they have no problem |
| 14:32.28 | mvanbaak | or the gsm files are very bad quality :) |
| 14:33.27 | carrello | i have a digium tdm400P |
| 14:35.06 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-0b8f70f6ab47b871) |
| 14:35.20 | *** part/#asterisk putnopvut (n=putnopvu@nat/digium/x-0b8f70f6ab47b871) |
| 14:35.39 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-0b8f70f6ab47b871) |
| 14:36.44 | carrello | I have a problem: gsm files played by background on asterisk 1.4.16.2 sounds very bad when a call arrive, while musiconhold (wav files) no (digium tdm400P)...why according to you? |
| 14:37.26 | mvanbaak | please stop repeating your question |
| 14:37.49 | carrello | excuse all |
| 14:39.24 | toresbe | Hey |
| 14:39.42 | toresbe | does anyone here have a modem connected to something SIP or IAXable? |
| 14:40.15 | *** join/#asterisk minkus (n=minkus@pool-71-182-32-147.clrkwv.east.verizon.net) |
| 14:41.30 | [TK]D-Fender | toresbe: Havew before. Highly inadvisable |
| 14:41.46 | *** part/#asterisk minkus (n=minkus@pool-71-182-32-147.clrkwv.east.verizon.net) |
| 14:41.50 | toresbe | [TK]D-Fender: We're only going to run 300 baud FSK. |
| 14:41.57 | *** join/#asterisk clusco (n=clus@77.120.49.60.cbj03-home.tm.net.my) |
| 14:42.05 | [TK]D-Fender | toresbe: Might survive on G.711 |
| 14:42.28 | [TK]D-Fender | toresbe: Most connections are very sensitive to delays, echo, jitter, PL, etc |
| 14:42.52 | *** join/#asterisk minkus (n=minkus@2001:5c0:9968:5b91:21b:77ff:fe23:178f) |
| 14:44.30 | hmmhesays | [TK]D-Fender: have you ever seen a polycom ip 320 just hang on "updating configuration" screen when the tftp server is unavailable? |
| 14:44.46 | [TK]D-Fender | hmmhesays: Can happen if you mangle their config files |
| 14:45.08 | [TK]D-Fender | hmmhesays: And all it can do is keep trying to load a broken one locally |
| 14:45.14 | hmmhesays | [TK]D-Fender: The phone booted fine grabbed its config and worked fine |
| 14:45.29 | [TK]D-Fender | hmmhesays: Dunno what to say at that point then |
| 14:45.37 | hmmhesays | [TK]D-Fender: yeah I don't know either |
| 14:48.24 | hmmhesays | god myspace is annoying 45 http requests just to log in |
| 14:48.35 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:48.41 | mvanbaak | duh |
| 14:49.00 | grandpapadot | Hi all. Is there a channel variable in 1.2 that represents the sip peer user from sip.conf? |
| 14:51.10 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:52.22 | [TK]D-Fender | grandpapadot: Chop it off the channel name |
| 14:52.54 | grandpapadot | Got it. Thanks, TK. |
| 14:56.54 | *** join/#asterisk punkgode (i=Sr@gateway/tor/x-d79fb1ce97621166) |
| 14:57.37 | mvanbaak | or use the setvar stuff in sip.conf |
| 15:01.33 | punkgode | hi, what should contain the "channel" variable when a call is terminated ? I'm having different behaviours, sometimes is the caller channel and sometimes is the callee channel. Apparently it's random |
| 15:02.20 | *** join/#asterisk ACiDV (n=acidv@97-147.dr.cgocable.ca) |
| 15:06.33 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:06.42 | Zeeek | join voip-users-conference |
| 15:06.50 | Zeeek | muhahaha |
| 15:06.55 | grandpapadot | TK: Wow, the CUT function is pretty powerful.. wonder how I missed it up to now.. |
| 15:07.30 | Zeeek | granpapdot you should see the PASTE! |
| 15:07.57 | jengelh | don't forget COPY |
| 15:10.33 | Zeeek | by the way, where IS the asterisk pasteboard? |
| 15:11.37 | mocker | ~pastebin |
| 15:11.38 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:12.12 | Zeeek | naw, that's not a pasteboard, that's a pasteBIN |
| 15:13.25 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 15:13.36 | a1fa | anybody familiar with 802.3af (PoE) here? |
| 15:14.04 | *** join/#asterisk ddunavant (n=David@66.170.97.28) |
| 15:14.50 | [TK]D-Fender | a1fa: http://www.networkworld.com/details/4681.html |
| 15:15.10 | a1fa | [TK]D-Fender : hi buddy |
| 15:15.41 | toresbe | Anyone around who have a modem, a way to connect it to asterisk, and spare time? |
| 15:15.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:15.51 | a1fa | i have an avaya poe brick that sends power over wires 7,8 |
| 15:16.00 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 15:16.05 | a1fa | how crazy |
| 15:16.08 | Zeeek | [TK]D-Fender HAPPY XMAS and a Merry Niu Year! |
| 15:17.33 | [TK]D-Fender | toresbe: You don't have enough to test your setup yourself? |
| 15:18.27 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:18.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:23.53 | Bladerunner05 | hola all, using asterisk 1.4.16.2, tdm400p and a wellcome gsm sound create with sox -r 8000 -c 1, but when I play it the sound is not clear |
| 15:24.20 | Qwell | Bladerunner05: by "not clear" do you mean "terrible"? |
| 15:24.42 | Qwell | if so, you've probably been hit with what we think is a bug in gcc 4.2 - check what you used to compile asterisk |
| 15:26.12 | Bladerunner05 | @Qwell: I used gcc version 4.2.3 20071123 (prerelease) |
| 15:26.12 | mvanbaak | gcc has bugs ? |
| 15:26.23 | mvanbaak | Bladerunner05: then indeed you hit the bug |
| 15:26.28 | Qwell | Bladerunner05: you're going to want to recompile with 4.1... |
| 15:26.47 | Bladerunner05 | @Qwell: Thanks I'll do that |
| 15:27.27 | tzafrir_home | Bladerunner05, any reason you use gsm rather than wav? |
| 15:27.36 | Qwell | of course - there is also the possibility that sox would also be affected by this |
| 15:27.37 | tzafrir_home | gsm means lowe quality |
| 15:27.52 | Qwell | tzafrir_home: are you aware of a way to tell what version of gcc a deb package was built with? |
| 15:28.26 | Bladerunner05 | If I use wav file instead of gsm I resolve this ? |
| 15:28.32 | tzafrir_home | no. But normally packages are just built with "gcc" of that distro |
| 15:28.35 | Qwell | Bladerunner05: sort of |
| 15:29.26 | Qwell | Bladerunner05: people have reported that this bug has also affected even alaw<>ulaw transcodings |
| 15:29.44 | Qwell | you might get lucky, but to be perfectly honest, the only thing I would recommend is recompiling with gcc 4.1 |
| 15:30.16 | tzafrir_home | or convert the file with asterisk instead? |
| 15:30.21 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 15:30.23 | Bladerunner05 | @Qwell: sure thanks |
| 15:30.26 | tzafrir_home | to see if this has caused the problem? |
| 15:30.33 | mvanbaak | I dont think you can see what version of gcc was used |
| 15:30.35 | Qwell | tzafrir_home: I'm certain of the problem :) |
| 15:31.55 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:33.19 | af_ | hfc chipset isdn card (1 bri) are directly supported by zaptel framework now? |
| 15:34.45 | tzafrir_home | af_, you need bristuff |
| 15:35.02 | Qwell | mattf actually did write a driver in zaptel, heh |
| 15:35.19 | Qwell | I have no idea if it works - probably not |
| 15:35.24 | tzafrir_home | and mattf is working on BRI support in trunk |
| 15:35.30 | Qwell | yep |
| 15:35.33 | af_ | I am reading a thread in the ml, but didn't understood which modules are needed |
| 15:35.34 | tzafrir_home | PTP works well. ptmp doesn't |
| 15:36.23 | af_ | the digium isdn adapters are hfc based? |
| 15:36.48 | tzafrir_home | yes (on the same HFC 4S chip as the junghanns and bero.net cards) |
| 15:37.26 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 15:38.41 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 15:39.35 | af_ | infact uses misdn framework too |
| 15:39.47 | af_ | continue to not understand the thread in the ml |
| 15:40.37 | tzafrir_home | From the little I know, qozap as-is won't work with the Digium cards. e.g: look at this little adjustment that is needed for bero.net cards: |
| 15:41.02 | af_ | I would like to not use bristuff |
| 15:41.06 | *** join/#asterisk PepOSX (n=pepOSX--@190.79.246.105) |
| 15:41.34 | af_ | but I guess is the only alternative available |
| 15:41.45 | coppice | like to use misdn == masochism |
| 15:41.51 | af_ | why? |
| 15:42.02 | coppice | misdn == poo poo |
| 15:42.21 | af_ | do not understand what means |
| 15:42.34 | af_ | poo meaning? |
| 15:42.40 | coppice | misdn == crappy by design |
| 15:42.45 | af_ | ah |
| 15:42.50 | tzafrir_home | http://blog.eth0.cc/zaptel-patchwork . I was actually a bit amazed to read that patch to qozap |
| 15:43.02 | mvanbaak | BRI == crappy by design |
| 15:43.27 | coppice | what is wrong with BRI |
| 15:43.49 | _ShrikE | Anyone using signalogic C5561 with asterisk? |
| 15:44.22 | mvanbaak | a lot |
| 15:44.35 | Zeeek | gentlemen, start your engines |
| 15:45.02 | mvanbaak | unreliable, lots of different implementations, impossible combinations of signalling etc |
| 15:45.23 | javb | guys, im having this error, "Failed to authenticate user <sip:8093681228@200.58.241.220>;tag=1c341680962" ... where "8093681228 is the callerid of my cellphone... cant understand this error... |
| 15:45.38 | coppice | there is a lot wrong with various implementations, but there's nothing wrong with BRI itself |
| 15:45.38 | javb | The call is comming from a SIP trunk via a SIP service provider. |
| 15:45.40 | javb | Any ideas? |
| 15:46.05 | Zeeek | coppice if you had one voip-related wish for 2008, what would it be? |
| 15:46.16 | [TK]D-Fender | javb: yeah, actually show us the entire call from beginning to end with SIP debug and your config masking only passwords.... |
| 15:46.17 | [TK]D-Fender | ~pb |
| 15:46.18 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:46.19 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 15:48.15 | javb | [TK]D-Fender: the call: http://dpaste.com/29213/ |
| 15:48.30 | javb | [TK]D-Fender: What do you mean with "config masking only passwords?" |
| 15:48.42 | coppice | Zeeek: that people wake up and stop trying to put PCI cards in PCs for telephony. embrace VoIP properly, and use gateway boxes for those interconnect jobs. :-) |
| 15:49.03 | [TK]D-Fender | javb: Means show us the SIP.conf and only change the passwords. |
| 15:49.20 | Zeeek | coppice gateway boxes like what? |
| 15:51.11 | *** join/#asterisk minkus (n=minkus@2001:5c0:9968:5b91:21b:77ff:fe23:178f) |
| 15:51.19 | javb | [TK]D-Fender: sip.conf -> http://dpaste.com/29214/ |
| 15:52.09 | [TK]D-Fender | javb: add "insecure=port,invite" to your [sdq1] section |
| 15:52.39 | [TK]D-Fender | javb: And retry. Also please specify your codecs for it, and remove the callerID entry |
| 15:54.52 | [TK]D-Fender | Zeeek: AudioCodes Mediant, etc.... |
| 15:55.59 | [TK]D-Fender | coppice: In your opinion, what are the better makes & models of SIP gateways (FXO & Digital)? |
| 15:56.01 | coppice | Zeeek: you catch on fast :-) That is the other part of the puzzle. You develop a set of standardised low cost gateway boxes for analogue, BRI. PRI, etc. and get various Chinese makers to churn them out at low cost :-) |
| 15:56.32 | javb | [TK]D-Fender: Changes made. but didnt work. Here is the sip debug: http://dpaste.com/29216/ |
| 15:56.53 | javb | (yes, i did sip reload) |
| 15:57.26 | [TK]D-Fender | javb: BIG PRINT : Looking for 8092024084 in from-dgtec (domain 190.6.144.109) <---------- SIP/2.0 404 Not Found |
| 15:57.54 | Zeeek | coppice the way things are headed, is there still going to be a market for high end hardwxare devel? |
| 15:58.27 | javb | [TK]D-Fender: i`m sorry ? |
| 15:58.46 | [TK]D-Fender | javb: Means "fix your dialplan " |
| 15:58.56 | *** join/#asterisk brpvieira (n=bernardo@c9118288.static.bhz.virtua.com.br) |
| 16:00.03 | coppice | Zeeek: what is your definition of "high end hardware devel"? |
| 16:00.10 | javb | [TK]D-Fender: i have, in context [from-dgtec] exten 's' .. just that. .. can u help see the problem? |
| 16:00.35 | [TK]D-Fender | javb: its not LOOKING for "s". its telling you its looking for "8092024084". Go make it |
| 16:00.57 | Zeeek | coppice "if you have to ask..." :) |
| 16:01.28 | Zeeek | it means I was talking through my cheap modem |
| 16:01.46 | coppice | Zeeek: oh, you mean the "fleece the suckers" hardware bracket |
| 16:02.17 | Zeeek | heh, eggs-acly |
| 16:06.52 | javb | [TK]D-Fender: u are right. Can you tell me what part of the sip debug did you saw it was looking for that exten ? and i didnt give me the error ("uknown exten"), instead, it was giving me "failed to authenticate" ? |
| 16:07.06 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 16:07.37 | *** part/#asterisk fiXXXerMet (n=meep@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 16:11.18 | [TK]D-Fender | javb: I copy & pasteed it for you... go get some coffee |
| 16:11.33 | *** join/#asterisk DarylVoip (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net) |
| 16:12.19 | javb | Ok, perfect. Thanks. |
| 16:12.58 | coppice | Zeeek: the blood sucking segment seems to have moved from hardware to services. even the highest end hardware seems to be pretty cheap these days, as so much of it can leverage the low cost of commodity bits and pieces. services seem to bleed people pretty well, though |
| 16:14.25 | Zeeek | coppice well, it's no accident that the verb "to service" means what it means ;) |
| 16:16.22 | coppice | if you look that the current telephony cards, those are still blood sucking, but that's not going to be a long lived thing. A card that costs <$100 to make selling for $1500 is a a dreamlike markup for most of the electronics industry :-) |
| 16:18.40 | Zeeek | such is life |
| 16:19.01 | Zeeek | but as the saying goes, "what the hooker earns, she spends in makeup" - applies to VOIP too |
| 16:19.44 | Zeeek | Oh, spam time: VOIP Users Conference is in 40 minutes. Info: http://VoipUsersConference.org |
| 16:19.56 | coppice | yeah, companies making huge markups have a remarkable ability to disipate them :-) |
| 16:20.07 | rob0 | ~Zeeek |
| 16:20.08 | jbot | zeeek is, like, someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 16:20.17 | Zeeek | -no joke, what can possible be NEW in VOIP for 2008? |
| 16:20.21 | Zeeek | uhhhhhh |
| 16:20.34 | Zeeek | I was soooo young when I wrote that :) |
| 16:20.45 | rob0 | :) |
| 16:22.23 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 16:22.23 | *** mode/#asterisk [+o twisted] by ChanServ |
| 16:22.47 | coppice | in a sane world new in VoIP in 2008 would be people taking the high ground with VoIP, instead of being a me too for the PSTN at a lower cost and crappier quality |
| 16:22.54 | *** join/#asterisk ronr (n=ron@82-170-109-196-static.dsl.ip.tiscali.nl) |
| 16:23.26 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 16:23.47 | Zeeek | you mean - gasp - create something new instead of a clone of century-old technology? Nah |
| 16:24.55 | *** join/#asterisk killfill (n=killfill@200.55.220.3) |
| 16:24.56 | killfill | hi |
| 16:25.11 | coppice | Well embracing wideband, which the PSTN has never been able to really get to grips with, would be a start. actually try to make the telephony experience better, instead of cheaper but crappier |
| 16:25.23 | killfill | my mashine is telling "Channel x/y, span 1 got hangup request, cause 16" |
| 16:25.34 | killfill | and cause 102 too. |
| 16:25.39 | killfill | what does the codes mean? |
| 16:25.43 | clusco | hi everyone.... im damn newbies about all this Voip stuff.... where should i start 1st???? |
| 16:25.44 | mvanbaak | cause 16 is normal clearing |
| 16:25.52 | killfill | (is there a dictionary for error codes somwhere? |
| 16:25.55 | mvanbaak | clusco: go read THE book |
| 16:25.57 | Zeeek | coppice funny you mention that. A lot of people are asking for it on phone-in podcasts now. Which you can get - almost - with Skype |
| 16:25.59 | clusco | does it sound good for me to start from asterisknow ???? |
| 16:26.00 | mvanbaak | ~book |
| 16:26.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:26.22 | Zeeek | ~nook |
| 16:26.23 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:26.33 | clusco | i had dload that book.... |
| 16:26.40 | killfill | mvanbaak: whats the 102Â? |
| 16:26.42 | Zeeek | I SAID Nook, not Book |
| 16:27.01 | clusco | ~nook |
| 16:27.02 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:27.11 | clusco | ~wook |
| 16:27.12 | Zeeek | ~crook |
| 16:27.18 | Zeeek | ~pook |
| 16:27.24 | clusco | just nook |
| 16:27.47 | holiday_42 | ~implode |
| 16:27.48 | jbot | ACTION implodes |
| 16:27.55 | Zeeek | whew, thanks |
| 16:28.08 | Zeeek | ~VOIPUsersConference.org |
| 16:28.16 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 16:28.38 | clusco | Zeeek: does asterisknow good for newbies like me ???? |
| 16:28.52 | coppice | Zeeek: once you go wideband, you never go back :-) |
| 16:28.57 | Zeeek | seriously? Try it and see. It makes it so easy to just try it out |
| 16:29.43 | mvanbaak | killfill: I'm looking for the conversion table |
| 16:29.51 | mvanbaak | sorry I cant answer you from memory |
| 16:29.52 | mvanbaak | hang on |
| 16:29.58 | Zeeek | clusco the try it out was for you |
| 16:30.02 | Zeeek | coppice Even if I do wideband, my providers need to do it or no one hears it. Like the tree falling in the forest |
| 16:30.07 | killfill | heh |
| 16:30.15 | clusco | Zeeek: thanks |
| 16:30.21 | killfill | mvanbaak: if you found where that table is online, please tell! |
| 16:30.30 | coppice | Zeeek: providers are just a passing fad |
| 16:30.43 | Zeeek | clusco I think there are a couple of CD you can try asterisk on without even installing it. Someone ? |
| 16:31.05 | clusco | Zeeek: do you mean livecd ??? |
| 16:31.05 | Zeeek | coppice truer words were never spoken. But then life itself is temporary |
| 16:31.11 | Zeeek | Ya |
| 16:31.26 | Zeeek | clusco I'm pretty sure there are a few |
| 16:31.36 | clusco | there's no livecd for asterisknow 6 yet |
| 16:31.40 | Zeeek | come to the conference in 30 min and ask |
| 16:32.06 | mvanbaak | killfill: 102 is AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE |
| 16:32.36 | killfill | mvanbaak: timer?.. i.e d-channel? |
| 16:32.41 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:32.41 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:32.45 | mvanbaak | think so |
| 16:32.48 | mvanbaak | hey russellb |
| 16:32.53 | killfill | ooh |
| 16:33.07 | killfill | that could be good, becouse it measn the error is on the telco.. :P |
| 16:33.26 | russellb | jbot: wave to mvanbaak |
| 16:33.27 | jbot | Bye, to mvanbaak |
| 16:33.27 | mvanbaak | killfill: you can find the causes here: |
| 16:33.30 | mvanbaak | http://svn.digium.com/view/asterisk/branches/1.4/include/asterisk/causes.h?view=markup |
| 16:33.30 | killfill | mvanbaak: your see that on the code? |
| 16:33.32 | russellb | ... |
| 16:33.34 | killfill | ah |
| 16:33.35 | russellb | silly. |
| 16:33.42 | mvanbaak | russellb: you want me to leave ? |
| 16:33.44 | mvanbaak | *sniff* |
| 16:33.47 | russellb | mvanbaak: no! |
| 16:33.50 | Qwell | n't |
| 16:33.50 | mvanbaak | you broke my heart |
| 16:33.53 | russellb | mvanbaak: it was supposed to be a wave hello |
| 16:34.06 | mvanbaak | ;) |
| 16:34.07 | AlexTO | Hi everyone, i'm trying to setup the CDRs on *NOW for 64, but when i try to make the install it show me error |
| 16:34.08 | Zeeek | russellb I spoke to Mark a few minutes ago and he ordains that you guys join in in 20 min |
| 16:34.24 | clusco | Zeeek: which one that was with web based admin ???? |
| 16:34.30 | coppice | russellb: as opposed to an ogg hello? |
| 16:34.34 | Zeeek | wkae up and smell th'e bakelite insulation burning |
| 16:34.40 | AlexTO | Does anyone know about it, who can help me out? |
| 16:34.48 | russellb | Zeeek: is he on today? |
| 16:34.57 | Zeeek | isn't there an asterisknow IRC channel? |
| 16:35.03 | Zeeek | russellb no next week |
| 16:35.19 | twisted | russellb! |
| 16:35.28 | AlexTO | OK, |
| 16:35.39 | russellb | twisted: ! |
| 16:36.06 | mvanbaak | looks like it's wakuptime in the us |
| 16:36.28 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:36.28 | twisted | dunno about wakuptime, but it ranges from 8am to 11am in the CONUS |
| 16:36.28 | Zeeek | don't you hate that spam keeps changing countries? So now I'm reading about vi@gr in Italian |
| 16:36.39 | killfill | mvanbaak: what else coult it be?.. (apart of a telco problem) |
| 16:37.10 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
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| 16:37.10 | *** mode/#asterisk [+o file] by irc.freenode.net |
| 16:37.11 | mvanbaak | killfill: I have no idea. I dont have much experience with zaptel |
| 16:38.53 | Zeeek | if my cell is so smart, how come it asks me the time and date when I turn it on? And why is the default date 2002? I think I need a new "smart" phone |
| 16:39.25 | jengelh | I think you need a new battery. |
| 16:40.30 | holiday_42 | don't turn it off ;) |
| 16:41.16 | killfill | anyone know why would on get an AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, on a zaptel call? |
| 16:42.09 | Zeeek | One minute to launch |
| 16:42.24 | coppice | Zeeek: that is not a phone issue. its a network issue. take it to India, for example, and it will set the time automatically |
| 16:42.30 | Zeeek | holiday_42 actually it's rarely on |
| 16:42.37 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 16:42.39 | mvanbaak | Zeeek: talkshoe conference ? |
| 16:42.42 | shadebob | hi |
| 16:42.50 | Zeeek | my other phone sets the time on the same nw |
| 16:43.06 | Zeeek | mvanbaak yes: http://VoipUsersConference.org |
| 16:43.08 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 16:43.27 | mvanbaak | can I listen from there ? |
| 16:43.36 | Zeeek | <PROTECTED> |
| 16:43.44 | coppice | Zeeek: phones typically have a config option to never set the time from the network |
| 16:43.46 | Zeeek | I believe there's a Flash badge there |
| 16:44.03 | jwh | not all networks support retrieving he timethough |
| 16:44.05 | Zeeek | coppice I know, this is an old Nokia candy bar. |
| 16:44.07 | jwh | +space |
| 16:44.31 | Zeeek | I now have a free cell service from my DSL provider |
| 16:44.45 | Zeeek | well, free received calls. 10 minutes of calls per month free |
| 16:44.52 | mvanbaak | <PROTECTED> |
| 16:44.56 | mvanbaak | that one ok as well ? |
| 16:45.16 | Zeeek | gotta run, you guys are all welcome to come and be brilliant in #voip-users-conference |
| 16:45.21 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:46.08 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 16:49.00 | *** join/#asterisk a_pyles (n=chatzill@rbuv-164-107-249-200.resnet.ohio-state.edu) |
| 16:50.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:53.17 | *** part/#asterisk a_pyles (n=chatzill@rbuv-164-107-249-200.resnet.ohio-state.edu) |
| 16:55.34 | ReDNeQ | is there a way to clear cf on an ext in config files.. Phone has turned it off but * still says it is on |
| 17:00.01 | De_Mon | what is cf? |
| 17:00.29 | ReDNeQ | call forwarding |
| 17:00.49 | jengelh | CompactFlash. |
| 17:01.06 | rob0 | Cystic Fibrosis. :( |
| 17:01.28 | De_Mon | how do you see that in asterisk? |
| 17:01.44 | ReDNeQ | asterisk -vvvvvcr |
| 17:01.46 | [TK]D-Fender | De_Mon: brace for impact :) |
| 17:01.56 | ReDNeQ | when a call is made to that ext that is what is reported |
| 17:02.00 | De_Mon | ReDNeQ sorry what? |
| 17:02.03 | [TK]D-Fender | ReDNeQ: pastebin the entiore call where yuo see this occuring from beginning to end |
| 17:02.14 | ReDNeQ | [TK]D-Fender, ok... |
| 17:02.15 | [TK]D-Fender | ~pb |
| 17:02.16 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:02.27 | De_Mon | ReDNeQ could there be another phone that is forwarding the call maybe (thats what pastebin will tell us) |
| 17:02.36 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com) |
| 17:03.22 | De_Mon | I could keep talking, but will just look at the pastebin instead |
| 17:03.30 | [TK]D-Fender | De_Mon: Good call |
| 17:03.45 | De_Mon | I learned THAT lesson yesterday talking about macros and call files! |
| 17:04.40 | [TK]D-Fender | De_Mon: When in doubt, sit back and enjoy the show, and let them incriminate themselves. Asking for anything but the raw evidence is a waste of time in 99% of cases |
| 17:04.48 | ReDNeQ | ok here it is http://pastebin.ca/835009 |
| 17:05.02 | toresbe | [TK]D-Fender: well, I don't have a remote modem, no |
| 17:05.13 | ReDNeQ | the main problem i am have is that the hunt is only working on the last extension in the group and is bypassing ext 10 and 130 |
| 17:05.43 | [TK]D-Fender | ReDNeQ: As suspected that FREEPBX BS and has nothing to do with Asterisk. |
| 17:06.18 | ReDNeQ | so this is strictly something to do with FreePbx... |
| 17:06.29 | [TK]D-Fender | ReDNeQ: Yes. |
| 17:06.53 | ReDNeQ | and how do you come to that, just because i use it.. I mean i really need some guidance? |
| 17:07.28 | [TK]D-Fender | ReDNeQ: I know it because I see that their dialparties AGI is pulling DB vars up in determining what it FELLS LIKE DOIND. |
| 17:07.32 | De_Mon | ~freepbx |
| 17:07.33 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:07.39 | [TK]D-Fender | "FEELS LIKE DOING"* |
| 17:07.42 | *** join/#asterisk simond (n=simon@208.68.95.5) |
| 17:07.58 | simond | w/hois jerjer |
| 17:08.00 | simond | oops |
| 17:08.00 | ReDNeQ | De_Mon, you dont have to bot me with quotes i know and understand what is and isnt |
| 17:08.02 | [TK]D-Fender | ReDNeQ: FreePBX is not supported here and you should know better than to ask. |
| 17:08.10 | simond | is Jared around here somewhere? |
| 17:08.25 | ReDNeQ | [TK]D-Fender: i really thought this was in asterisk... That is only why i asked |
| 17:08.50 | [TK]D-Fender | ReDNeQ: Of course you did.... based on what might I ask? |
| 17:08.51 | De_Mon | ReDNeQ knowing and not doing, means you don't really KNOW |
| 17:08.53 | ReDNeQ | i see dialparties.agi and thought that was * |
| 17:09.18 | jwh | hm |
| 17:09.29 | ReDNeQ | De_Mon, no shit! maybe thats why I ask, if I knew I'd be a FUCKED GOD like you? |
| 17:09.33 | jwh | how would one limit asterisk to certain contexts when using sip uri dialling? |
| 17:09.36 | [TK]D-Fender | ReDNeQ: its a friigen AGI! Thats 3rd party code no different thatn dialplan sand gets CALLED by the dialplan. |
| 17:09.54 | jwh | as basically, I can dial any number@sip.blah.com regardless of which context its in |
| 17:10.01 | ReDNeQ | [TK]D-Fender: as alwasy thanks for straigting me out.. |
| 17:10.05 | ReDNeQ | sorry i asked |
| 17:10.09 | jwh | which also means outbound calls can be made without authentication |
| 17:10.31 | De_Mon | don't get pissed at me you're the one that said you KNEW. Geez quite a little phylosophy and he bites my head off. |
| 17:10.32 | ReDNeQ | [TK]D-Fender, your help is appreciated really no j/k |
| 17:10.50 | *** join/#asterisk ralfep (n=ralfe@vc-196-207-35-48.3g.vodacom.co.za) |
| 17:10.52 | [TK]D-Fender | De_Mon: ok, tone it down a bit please... |
| 17:10.52 | ReDNeQ | De_Mon: no i didnt say I knew, i said i thought! |
| 17:11.10 | De_Mon | s/phylosophy/philosophy/ |
| 17:11.28 | [TK]D-Fender | jwh: That places an un-authed call to "target". And the receiving end does whatever it feels like with it. |
| 17:12.03 | [TK]D-Fender | jwh: and a "URI" is not a thing in a "context". |
| 17:12.10 | jwh | [TK]D-Fender: ok |
| 17:12.20 | De_Mon | sorry I type slow and was replying to an earlier comment... |
| 17:12.22 | [TK]D-Fender | jwh: Perhaps the dial statement with that URI is.... |
| 17:12.46 | jwh | [TK]D-Fender: but the main problem is, as i've obviously allowed outbound calls for customers, if someone dials 01xxxx@ip, it sends outbound calls out via the pstn without any authentication |
| 17:13.21 | [TK]D-Fender | jwh: then you should pay attention to the context you set under [general} <-- |
| 17:13.49 | ralfep | hi all. I'm new to Asterisk. Could someone help me with a SIP problem? When my SIP phone tries to connect, asterisk says "Peer is not supposed to register". What does that mean? |
| 17:14.08 | mintee | I've got an auto answer on an exten, but it picks up so fast the message is trunked... how can I put like a 2 second pause before my exten => 533,1,Answer |
| 17:14.09 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 17:14.18 | mintee | or between the Answer and Playback? |
| 17:14.32 | [TK]D-Fender | ralfep: Means you set "host=[something specific other than dynamic]" and then your device is trying to register to that account |
| 17:14.36 | jwh | [TK]D-Fender: tyes, the context i've set contains just inbound ddi's |
| 17:14.53 | ralfep | so i must set that line to "host=dynamic"? |
| 17:15.03 | [TK]D-Fender | jwh: You've clearly set something wrong so pastebin a sample of a call that goes bad, and your dialplan & sip.conf |
| 17:15.15 | [TK]D-Fender | ralfep: if you're expecting your device to register, yes. |
| 17:15.31 | ralfep | [TK]D-Fender: Thanks. I'll try that now. |
| 17:16.01 | [TK]D-Fender | mintee: Answer, Wait(2), Playback |
| 17:16.36 | mintee | Thanks [TK]D-Fender |
| 17:17.45 | mintee | perfect |
| 17:18.48 | [TK]D-Fender | mintee: np |
| 17:19.17 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 17:20.01 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 17:28.49 | jwh | [TK]D-Fender: sec |
| 17:32.01 | Bladerunner05 | using asterisk 1.4.16.2 I get this error: file.c:643 ast_readaudio_callback: Failed to write frame |
| 17:34.44 | De_Mon | Bladerunner05 and? |
| 17:35.37 | *** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk) |
| 17:36.52 | AlexTO | does anyone set add_on in 64? |
| 17:37.17 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 17:38.19 | Bladerunner05 | •De_Mon• and what ? |
| 17:38.50 | De_Mon | Bladerunner05 you haven't mentioned why you're shaing this information |
| 17:40.24 | *** join/#asterisk jdspencer (n=jdspence@12.37.95.91) |
| 17:41.07 | jdspencer | Anybody alive? |
| 17:41.39 | De_Mon | nobody here but us bots |
| 17:41.44 | jdspencer | Nice |
| 17:42.19 | jdspencer | Can anybody help me with what seems to be a hardware incompatibility issue? (Digium TE412P and Dell PowerEdge 2950) |
| 17:42.26 | jdspencer | I've burnt out two motherboards |
| 17:42.37 | jdspencer | Only when the two are paired |
| 17:42.47 | fugitivo | change dell for another brand |
| 17:43.06 | jdspencer | Love to... but I have a production system that needs to come back up while the new system is on order. |
| 17:43.35 | jdspencer | Any recommendations on brand? I'm thinking IBM. |
| 17:43.39 | jdspencer | xSeries |
| 17:44.09 | De_Mon | what do you mean burnt out two motherboards? |
| 17:44.21 | De_Mon | literaly dead even after removing the card? |
| 17:44.22 | jdspencer | Specifically the PCI bus fails |
| 17:44.33 | jdspencer | indeed, even after the card is gone |
| 17:44.42 | jdspencer | Dell is pointing the finger at Digium |
| 17:44.51 | jdspencer | Nobody is available at Digium because of the holiday |
| 17:45.00 | holiday_42 | ? |
| 17:45.25 | jdspencer | I just called to purchase support and was told nobody was in the office who could help me with that purchase. |
| 17:45.50 | jdspencer | This really builds confidence in Digium/Asterisk for my management. |
| 17:45.58 | jdspencer | But I'm sure it's all Dell's fault |
| 17:46.11 | De_Mon | I see a big "for use only with a 3.3 volt PCI slot" warning but crap that sucks |
| 17:46.25 | jdspencer | The voltages match |
| 17:48.15 | *** join/#asterisk ariel_ (n=ariel_@70-46-87-154.ftl.fdn.com) |
| 17:48.45 | killfill | is there any way to tell ${CURL(url)} a timeout value? i.e. 2 seconds |
| 17:48.46 | ariel_ | hello everyone |
| 17:48.55 | jdspencer | hey ariel |
| 17:48.55 | Bladerunner05 | using the latest asterisk and addons in specially cdr_mysql how can I popule uniqueid ? |
| 17:49.04 | *** join/#asterisk marcan (i=1337@29.Red-88-9-94.dynamicIP.rima-tde.net) |
| 17:49.26 | jdspencer | blade: uniqueid usually populates itself |
| 17:49.56 | Bladerunner05 | in my case not |
| 17:50.02 | De_Mon | you can't, see the documentation for the list of modifiable cdr values |
| 17:50.11 | jdspencer | blade: have you been through this document? -- http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 17:50.45 | jdspencer | blade: looks like a compile flag needs to be edited, but I'm not sure how up to date the info on that page is |
| 17:51.33 | jdspencer | blade: you could also cheat and set that field to AutoIncrement |
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| 17:57.00 | Bladerunner05 | •jdspencer• I intend the uniqid of asterisk not auto_increment of mysql filed |
| 17:57.06 | Bladerunner05 | (field) |
| 17:57.35 | jdspencer | blade: so you are logging multiple asterisk servers to one database? |
| 17:58.08 | jdspencer | ooooh |
| 17:58.15 | jdspencer | You mean the uniqueid Asterisk assigned! |
| 17:58.52 | jdspencer | You'll need to follow the instructions in that link I pasted under the heading "Storing the Unique ID" |
| 18:00.14 | ariel_ | I don't know why they never just defaulted to have that included to start with. |
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| 18:07.57 | jdspencer | Anyone have a further thought on why a TE410P and/or TE412P would be causing the PCI bus in a Dell PE2950 to hardware failure? |
| 18:09.10 | ariel_ | yes seem that before on the 2850. Give me a minute my network is about to get reset. I'll be back. |
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| 18:09.36 | *** join/#asterisk ariel_ (n=ariel_@70-46-87-154.ftl.fdn.com) |
| 18:09.49 | ariel_ | Ok back |
| 18:09.55 | jdspencer | sweet |
| 18:11.40 | Bladerunner05 | Using 1.4.16.2 I notice a lot of file.c:643 ast_readaudio_callback: Failed to write frame and then hangup |
| 18:12.22 | Bladerunner05 | I'm using the tdm400p with zaptel-1.4.7.1 |
| 18:13.12 | russellb | that is usually when the far end hangs up |
| 18:13.26 | russellb | make sure you don't have callprogress turned on ... |
| 18:13.35 | russellb | or busydetect while we're at it |
| 18:14.57 | ariel_ | jdspencer, we had to run linux without the acpi support |
| 18:15.13 | *** join/#asterisk rnovotny22 (n=rnovotny@99-203-56-203.area2.spcsdns.net) |
| 18:15.16 | jdspencer | and afterward all was happy? |
| 18:15.43 | ariel_ | So far we have about 10 2850 with them up and running no issues with them any more. |
| 18:16.01 | jdspencer | excellent to hear |
| 18:16.19 | jdspencer | that's just a kernel arg of "acpi=off" right, or did you recompile without acpi support? |
| 18:16.43 | ariel_ | I have about 12 2850,s 5 1850 and 4 1950s all running with acpi=off |
| 18:16.44 | ariel_ | yes |
| 18:16.55 | Bladerunner05 | @russellb sure now it works |
| 18:17.20 | jdspencer | And your issue was basically a PCI bus failure? |
| 18:17.22 | Bladerunner05 | but it didn't reach the hang up first of 10 seconds when I hang up |
| 18:18.01 | Bladerunner05 | @russelb: sorry It continue as I didn't hang up..... |
| 18:18.15 | ariel_ | jdspencer, frame slips |
| 18:18.25 | jdspencer | we're getting those too |
| 18:18.52 | jdspencer | I'm still confused about how anything could translate into needing a new motherboard |
| 18:19.18 | jdspencer | But if it happens again we'll drop that system for a different line of hardware. |
| 18:19.28 | jdspencer | Thanks for your help! |
| 18:20.42 | ariel_ | any time |
| 18:21.23 | *** join/#asterisk nny_1 (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 18:22.25 | nny_1 | quick q, looking for a place to research the idea of how to setup a round robin between two extensions.. not trying to use the Queue application, as the extensions don't report their status to *. Just want asterisk to use 1, than the other, than the 1st one. |
| 18:22.56 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-a5858d684bfcc3ff) |
| 18:22.59 | jdspencer | nny_1 you could maintain a global variable in your dialplan to track which one is up next |
| 18:23.32 | [TK]D-Fender | nny_1: "show function DB" , "show application gotoif" , "show application set" |
| 18:23.55 | jdspencer | that's right, you could also use the internal DB |
| 18:24.12 | De_Mon | ariel_ is a frame slip when frames get "off by one" ? |
| 18:24.40 | De_Mon | every once in a while I'll see warnings audio 240 frame 350 next waring is audio 350 audio 120 and so on... |
| 18:25.30 | De_Mon | apci is something you can disable in the bios too isn't it? |
| 18:25.40 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
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| 18:26.34 | ariel_ | De_Mon, depends on the mb |
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| 18:28.00 | De_Mon | is what I described above a "frame slip"? |
| 18:29.35 | davidnicol | playback in 1.4 does not appear to be sending RTP -- is there a playback.conf or something? |
| 18:30.26 | *** join/#asterisk nny_1 (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) [NETSPLIT VICTIM] |
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| 18:30.55 | nny_1 | weee! |
| 18:30.59 | nny_1 | lol splits... |
| 18:31.00 | hardwire | weee?! |
| 18:31.16 | Bladerunner05 | there is alternative to busydetect=yes bucause without this tdm400p don't recognize hangup and with this asterisk hang up the channel |
| 18:31.18 | nny_1 | eh my client just stated that 210 people left the channel |
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| 18:31.51 | AlexTO | Hi, i'm having a problem setting up the add_on on 64, looks like the destination folders are differents that it has already setup. |
| 18:32.12 | AlexTO | when i make the "make install" |
| 18:32.32 | De_Mon | davidnicol your talking about the Playback() application? |
| 18:32.49 | De_Mon | net split |
| 18:32.52 | AlexTO | does any one familiar with it that cn help me |
| 18:33.05 | nny_1 | [TK]D-Fender: I am assuming the use of the DB is for consistency, or loss through a restart? In other words, If i defined my magic variable as one value at startup, and then changed it back and forth, it would still work |
| 18:33.14 | nny_1 | of course i plan to have a failsafe or otherwise afterwords |
| 18:33.35 | nny_1 | if this=this than dial foo, if this=that than dial bar, else dial =foo |
| 18:33.56 | nny_1 | er if this=this than dial foo (set this=that) |
| 18:34.05 | nny_1 | sorry i know thats not the most eloquent way of putting it |
| 18:34.16 | jdspencer | nny: you're making sense |
| 18:34.18 | De_Mon | the asterisk database is persistant across restarts (or can be) |
| 18:34.49 | jdspencer | nny: so you could do it using your magic variable if you don't care about start-state consistency |
| 18:35.00 | De_Mon | and its for tracking data, how else are you going to know which extension to call first, a global variable? |
| 18:35.03 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 18:35.14 | nny_1 | jdspencer: ahh ok thanks |
| 18:35.25 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 18:35.31 | nny_1 | jdspencer: gonna work on setting it up now. |
| 18:35.41 | nny_1 | good to know the variable could be consistent, though |
| 18:36.20 | AlexTO | http://pastebin.ca/835136 |
| 18:38.33 | *** part/#asterisk simond (n=simon@208.68.95.5) |
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| 18:42.44 | [TK]D-Fender | nny_1: Yes |
| 18:43.37 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 18:48.39 | nny_1 | [TK]D-Fender: heh cool, i think i got it.. gonna test now |
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| 18:50.23 | nny_1 | heh i just thought about this, and it seems noobish, but it would be nice if there was a dialplan editing app that, eh i guess auto correct is a poort choice of words, but like a spell checker for the asterisk language... just a thought |
| 18:51.40 | De_Mon | nny_1 you can use vim with hilighting that will tell you if the keywords are spelled wrong |
| 18:52.53 | nny_1 | De_Mon: ahh nice have to try that out, been told it is far superior to nano when dealing specifically with code |
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| 18:55.14 | *** join/#asterisk suvir (n=chatzill@ppp-124.120.129.129.revip2.asianet.co.th) |
| 18:59.34 | De_Mon | syntax hilighting is very nice |
| 18:59.38 | davidnicol | De_Mon: yes, exactly; and derivatives, such as DateTime |
| 19:00.08 | davidnicol | I am now able to route RTP via the system, but cannot play files from it |
| 19:00.47 | davidnicol | yes nny_1 vim seems to know .conf syntax OOTB |
| 19:01.03 | De_Mon | davidnicol make sure you are Answer() ing any channels first before you try to play audio over |
| 19:01.26 | davidnicol | De_Mon have that, yes. |
| 19:01.29 | twisted | oej: you around? |
| 19:01.31 | De_Mon | it does, but there are specific syntax definitions for /etc/asterisk/extensions.conf and friends |
| 19:02.08 | davidnicol | I answer, wait one second, DateTime; the console says "playing day-5" or something like that and sends exactly one RTP packet |
| 19:02.14 | davidnicol | (debug rtp) is on |
| 19:02.23 | davidnicol | no stream of them |
| 19:02.46 | De_Mon | hrmph 1.4 trunk by chance? |
| 19:02.51 | nny_1 | davidnicol: thanks |
| 19:02.57 | De_Mon | erm branch |
| 19:03.01 | davidnicol | yes, exactly |
| 19:03.16 | davidnicol | or at least a recent 1.4 release |
| 19:03.19 | De_Mon | as in, not a release like 1.4.16 |
| 19:03.25 | nny_1 | jdspencer: [TK]D-Fender thanks, it works beautifully... |
| 19:03.35 | davidnicol | not sure .. checking |
| 19:04.00 | davidnicol | 1.4.15 release |
| 19:04.22 | davidnicol | is this fixed in .16? |
| 19:04.48 | De_Mon | duno, im using 1.4.13 and its working normally |
| 19:05.22 | *** join/#asterisk frenzy (i=user@unaffiliated/frenzy) |
| 19:05.33 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 19:07.00 | nny_1 | anyone have any thoguhts on the sercurity risks of using proftpd? |
| 19:07.09 | nny_1 | i.e. plaintext passwords, etc |
| 19:07.28 | Qwell | nny_1: I think proftpd has had remote root exploits in the past |
| 19:07.39 | Qwell | there's always some risk |
| 19:08.14 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 19:08.31 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 19:09.09 | nny_1 | Qwell: indeed |
| 19:10.08 | nny_1 | discussing how to handle clients provisioning on smaller systems (1-5 phones)... be nice to have smaller systems just ftp cfg and firmware updates from our ftp server... |
| 19:12.32 | nny_1 | eh at least the version we have here is 1.3.0, which I seems to have the root exploits fixed |
| 19:12.56 | nny_1 | there is, however, a local overflow exploit, but not too concerned about local accoutn access |
| 19:13.52 | *** join/#asterisk kc1028 (n=mrprotoc@66-117-145-58.lmi.net) |
| 19:14.48 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:14.49 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:17.54 | killfill | is there any way to tell ${CURL(url)} a timeout value? |
| 19:19.06 | killfill | im loading an URL from the dialplan. but it i have no network, the thing stays in the curl call. cannot permit this.. :S |
| 19:19.48 | [TK]D-Fender | killfill: "show function CURL" |
| 19:20.41 | killfill | [TK]D-Fender: cannot see a timeout there.. :S |
| 19:20.49 | [TK]D-Fender | killfill: then tahts your answer. |
| 19:21.00 | killfill | :S |
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| 19:24.27 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:25.25 | Bladerunner05 | I'm looking for italian configuration for busy detect using zapata.conf |
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| 19:48.26 | hardwire | do most of y'all use rate-engine for LCR? |
| 19:49.13 | hardwire | I can see how well it would work.. but I'm totally boggled by what other options there would be for straightforward LCR for large area deployments. |
| 19:49.35 | hardwire | nny_1: nah.. I grew up with southern parents.. but I'm a Colorado native (living in Alaska) |
| 19:49.55 | nny_1 | hardwire: hehe ok.. yeah i was born in NY, raised in CA, lived in SC for 10 years now |
| 19:50.18 | nny_1 | i have no freaking clue what i am :) |
| 19:50.18 | hardwire | nny_1: an oddity if you were in the civil war. |
| 19:50.18 | nny_1 | lol |
| 19:50.19 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 19:50.35 | hardwire | nny as in johnny? |
| 19:50.42 | nny_1 | indeed |
| 19:50.47 | hardwire | homicidal? |
| 19:50.52 | nny_1 | well as in johnen vasquez.. yeah |
| 19:50.55 | hardwire | :) |
| 19:51.00 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 19:51.09 | nny_1 | er yeah johnny, johnen vasquez's character |
| 19:51.17 | nny_1 | LOL |
| 19:51.18 | nny_1 | nice |
| 19:51.56 | davidnicol | upgraded to 1.4.16.2 and still doesn't work |
| 19:52.26 | mmlj4 | any teredo users? how well does this work for road warriors, say with iax? |
| 19:52.27 | davidnicol | the console seems color-enhanced now though |
| 20:02.12 | De_Mon | davidnicol this over sip, iax..? |
| 20:02.57 | davidnicol | sip |
| 20:03.26 | davidnicol | and RTP will route for calls through, including to and from registered softphones |
| 20:03.41 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 20:04.02 | mvanbaak | did you issue Answer before running Playback ? |
| 20:04.08 | nny_1 | this is gonna be a weird q, but does anyone have a cool way to c/p an entire config file over an ssh session without using scp? |
| 20:04.38 | davidnicol | nny_1: cat > newfile |
| 20:04.49 | mvanbaak | open a new buffer in vim and edit a file on the remote machine with it |
| 20:04.50 | davidnicol | end with ctrl/d |
| 20:05.08 | nny_1 | davidnicol: er sorry, thinking I guess to local clipboard |
| 20:05.17 | mvanbaak | :e ssh://mvanbaak@server/etc/asterisk/sip.conf |
| 20:05.34 | davidnicol | how do yuo have a local clipboard over a ssh session? |
| 20:05.55 | nny_1 | davidnicol: like if i needed to post a log file that was 10 pages deep from a remote server to pastebin.. i usually scp it over, open it in abiword, and then edit select all --> copy --> paste |
| 20:07.07 | davidnicol | you can't select the scroll buffer in your terminal? |
| 20:07.25 | davidnicol | putty might let you log the session |
| 20:07.33 | davidnicol | to a file |
| 20:08.18 | mvanbaak | vim scp://hostname/path/to/file |
| 20:08.23 | mvanbaak | like that |
| 20:08.26 | nny_1 | davidnicol: er using gterm on local nix machine |
| 20:08.40 | nny_1 | mvanbaak: hmm i like that |
| 20:08.45 | mvanbaak | select all |
| 20:08.47 | davidnicol | good for you. Can't select the scroll buffer? |
| 20:08.52 | mvanbaak | paste in a new buffer |
| 20:08.58 | mvanbaak | save new buffer to local file |
| 20:09.12 | nny_1 | davidnicol: heh sorry no not that i can see |
| 20:09.27 | mvanbaak | nny_1: when in vim command mode type this: :help netrw |
| 20:09.33 | nny_1 | just a q.. always figure i have been doing it the hardway for a long time |
| 20:10.45 | *** join/#asterisk javb (n=javb@190.80.201.55) |
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| 20:12.00 | javb | Hi, have a custumer, who doesnt have enough money for polycom / snom sets. Has anyone here worked with grandstream gpx2000 ? BT-102 !@#$ ... any idea... |
| 20:13.12 | mvanbaak | ~gs |
| 20:13.13 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:13.34 | nny_1 | javb: you can score some nice polycom 320's for under 100 bucks |
| 20:13.45 | mvanbaak | I did have some customers that really really wanted to use the gxp2000 |
| 20:13.50 | mvanbaak | and they do work |
| 20:14.11 | mvanbaak | hell, even the indication leds and stuff work |
| 20:14.37 | mvanbaak | but they break easily for no reason, lockup, screen gets blank, quality is sucky |
| 20:14.44 | javb | nny_1: polycom 320 has just 1 ethernet port. i need two. and 330, its 105... plus taxes and taking them to custumers country reach 175 |
| 20:14.58 | davidnicol | so it still appears that the playback RTP is trying different UTP than the ones in use by the NAT system: |
| 20:15.40 | mvanbaak | and you really really really dont want to use the BT-102 |
| 20:15.42 | nny_1 | javb: indeed |
| 20:17.25 | nny_1 | javb: basically.. form what i have seen, (Check voip-info.org on the gs stuff) is yes! they are garbage, but they apparently are capable of handling basic needs.. i would be twice worried if i needed to send something like that to a place I couldn't physically be.. I have a client in Panama with 10 snoms, and thew damn things issue a password challenge for the sip user whwnever they have an issue connecting... |
| 20:17.35 | *** join/#asterisk windsor1 (n=win@adsl-75-24-215-230.dsl.pltn13.sbcglobal.net) |
| 20:18.11 | nny_1 | javb: so i have to remote desktop in, hit the interface, and change it from whatever extensions the client was tryng to dial when the snom issued the challenge,... it's.. furstrating to say the least... |
| 20:18.17 | nny_1 | sorry spelling sucks today |
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| 20:18.57 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:19.24 | javb | lol... well, thanks |
| 20:19.52 | mvanbaak | nny_1: that snom issue is irritating indeed |
| 20:20.40 | nhuisman_work | does anyone here use redfone? |
| 20:20.49 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:21.27 | davidnicol | on 1.2, the packet goes to the same port, the first one, then the rest of them. On 1.4, the first packet goes to a different port from the port where the sounds from the phone come from. |
| 20:21.45 | davidnicol | where do I report this if not here? |
| 20:22.39 | davidnicol | or does 1.4 have additional subtlety to nat=yes canreinvite=no? |
| 20:23.10 | De_Mon | if you haven't already read update.txt for 1.2 and 1.4 that might tell you |
| 20:23.20 | De_Mon | what you describe does sound like nat issues |
| 20:25.13 | nny_1 | mvanbaak: yeah.. i'd give anything for some kind of resolution on it.. been pouring through the snom wiki looking for a way to disable or at least break it... don't get me wrong the phones *aren't* that bad, but they aren't that good either |
| 20:26.16 | mvanbaak | indeed |
| 20:26.25 | mvanbaak | I have not found a way to 'fix' it |
| 20:28.02 | nny_1 | hopefully there is some movement to change that at snom... for now I have a machine dedicated as a remote desktop just to login and fix it when it happens... unfortunately for snom, I won't be buying another one fo their phones until the issue is resolved.. |
| 20:30.11 | nny_1 | some of the stuff on them isn't so bad.. the config menus are robust.. you can do test dials viz the web interface.. these things are good.. but that one issue makes the rest of those pretty much useless |
| 20:30.15 | nny_1 | buit yeah /rant |
| 20:31.32 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:31.43 | tzafrir_home | cisco phones are also like that, right? |
| 20:32.31 | nny_1 | heh yay even there email support a question stuff is way over zealous.. |
| 20:32.44 | nny_1 | gonna try to contact them and get some kind of statement on the matter |
| 20:32.56 | davidnicol | changing canreinvite to nonat instead of no make no difference |
| 20:33.02 | *** join/#asterisk jblack (n=jblack@pool-71-181-136-33.sctnpa.east.verizon.net) |
| 20:34.44 | mvanbaak | tzafrir_home: maybe the sip load. I have not seen it with my 38 skinny loads |
| 20:36.29 | nny_1 | sent them an email.. if any good comes of it, i'll forward it along |
| 20:37.19 | [TK]D-Fender | Polycom > All |
| 20:37.25 | nny_1 | indeed |
| 20:38.10 | nny_1 | need to try out the kirks.. i use Aastra for cordless (non 2.4) needs.. it works great, but I hate having polycoms and kirks in the network.. confuses the users |
| 20:38.16 | nny_1 | er polycoms and aastra* |
| 20:38.27 | nny_1 | i won't deal with 2.4 sip phones |
| 20:38.34 | nny_1 | er 2.4 GHZ |
| 20:42.12 | tzafrir_home | davidnicol, please pastebin sip.conf |
| 20:46.19 | De_Mon | I have an aastra that use to work till i upgraded to 1.4 then it just stopped. very weird but never had time to look into it :( |
| 20:47.19 | AlexTO | there is anyopne familiar with CDRs in MySQL? |
| 20:47.53 | nny_1 | De_Mon: by stopped working you don't mean the right menu button 86's the phone do you? |
| 20:47.58 | davidnicol | ~pb |
| 20:47.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:51.19 | mvanbaak | AlexTO: I use it |
| 20:54.36 | davidnicol | http://pastebin.ca/835361 |
| 20:56.01 | [TK]D-Fender | davidnicol, You have no localnet clause, no externIP, and a ton of other appropriate missing stuff. |
| 20:56.03 | [TK]D-Fender | ~sipnat |
| 20:56.04 | 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 |
| 20:56.07 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 20:59.15 | mvanbaak | damn |
| 20:59.20 | mvanbaak | you beat me on that one |
| 21:00.33 | davidnicol | it's getting the correct IP, and routing through it -- I am listening to the digium MOH via my X-lite, via the box in question. It's just playback that doesn't work. |
| 21:01.36 | davidnicol | IAX2/guest@misery.digium.com/asterisk-dev i will return presently |
| 21:02.57 | *** join/#asterisk FlatFoot (n=chatzill@80.88.218.4) |
| 21:04.09 | FlatFoot | evening all |
| 21:05.01 | *** join/#asterisk outtolunc (n=4a3ea968@gateway/web/cgi-irc/ircatwork.com/x-694fc929575d85c7) |
| 21:07.07 | nny_1 | how do i check which codecs (for example in the IVR) asterisk is using ? |
| 21:07.10 | nny_1 | in console |
| 21:07.22 | nny_1 | nm lol |
| 21:07.26 | nny_1 | love asterisk console |
| 21:07.35 | nny_1 | at least that it is intuitive |
| 21:09.13 | nny_1 | hmm.. i have gsm support, i have the pls-hold-while-try.gsm file in my sounds dir, but when console gets to that part is says filenot found in any format... i just compiled asterisk add-ons into the system, could something else have broke? |
| 21:09.30 | nny_1 | fwiw it worked before i comipled * addons |
| 21:11.04 | nny_1 | meh my biz partner did "make samples [STRANGLE] during the process, let me just restore the conf backups |
| 21:12.08 | De_Mon | snicker |
| 21:13.18 | nny_1 | playback_exec: ast_streamfile failed on Zap/1-1 for pls-hold-while-try |
| 21:13.21 | nny_1 | hmm |
| 21:13.25 | nny_1 | googletime |
| 21:13.34 | Havokmon | I have a pri to another pbx, by default does asterisk verify dialtone prior to placing a call over that pri? |
| 21:14.10 | Havokmon | I should say, I have a pri from asterisk to another pbx |
| 21:14.26 | nny_1 | <PROTECTED> |
| 21:14.47 | nny_1 | so even though gsm shows up in codecs, it is not trying to use it.. now to figure out what borked |
| 21:15.31 | nny_1 | well the file exists at least |
| 21:17.25 | nny_1 | eww woah |
| 21:17.38 | nny_1 | the invalid extension message sounds like poop |
| 21:18.13 | nny_1 | which is pbx-invalid.gsm |
| 21:18.20 | nny_1 | so it plays *that* gsm file |
| 21:18.24 | nny_1 | meh wtf |
| 21:20.48 | jblack | Hello. I'm having a slight problem with setting up sip for the first time. |
| 21:21.28 | jblack | I set up ekiga to dial into a new asterisk server. I set the context in sip.conf for the sip connection to go to phones, but it seems to be going to the default context instead |
| 21:22.49 | nny_1 | errm |
| 21:22.58 | [TK]D-Fender | jblack, how are you entering the @ to dial into it? |
| 21:23.06 | [TK]D-Fender | # |
| 21:23.48 | nny_1 | so is gsm supposed to sound like garbage? |
| 21:23.51 | nny_1 | :) |
| 21:23.52 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582783.dsl.bell.ca) |
| 21:24.00 | Qwell | nny_1: rebuild asterisk with gcc 4.1 |
| 21:24.05 | Qwell | 4.2 sucks |
| 21:24.10 | nny_1 | Qwell: interesting |
| 21:24.16 | Qwell | it's broken, heh |
| 21:24.36 | nny_1 | Qwell any way to see what version of gcc I am using nowq? |
| 21:24.38 | nny_1 | now* |
| 21:24.41 | Qwell | gcc -V |
| 21:24.55 | jblack | d-fender: I'm just dialing to the ip of the asterisk server. |
| 21:25.01 | jblack | I can connect _to_ the sip phone. |
| 21:25.32 | jblack | My goal is that when an internal sip phone dials into the server, it gets a different context for tings like outgoing calls. |
| 21:25.34 | [TK]D-Fender | jblack, .... how EXACTLY are you entering it into the field? |
| 21:25.38 | nny_1 | Qwell: well.. you guesses what version I am using, so you win the prize.. well I win the prize, i get to downgrade gcc and recompile -_- |
| 21:25.39 | nny_1 | guessed* |
| 21:25.47 | jblack | sip:192.168.2.2 |
| 21:25.53 | Qwell | what's my prize? |
| 21:26.19 | jblack | (My thinking is that I can give authenticated sip phones a different context than unauthenticated connections from outside) |
| 21:26.26 | tzafrir_home | Qwell, what's so wrong with 4.2? Could you please be more specific? |
| 21:26.31 | [TK]D-Fender | jblack, exactly. |
| 21:26.36 | Qwell | tzafrir_home: nope, it's just broken with transcoding |
| 21:26.48 | [TK]D-Fender | jblack, if you dial it as a full URI w/ an "@" then its an unauthed call |
| 21:26.52 | Qwell | there's a bug report on gcc's tracker |
| 21:26.56 | tzafrir_home | Any specific bug? |
| 21:27.04 | Qwell | no, they want us to find it |
| 21:27.08 | davidnicol | [TK]D-Fender: I have a type-9 installation, neither of those are needed |
| 21:27.24 | jblack | With @, unauthenticated. Without an @, authenticated. Correct? |
| 21:27.31 | nny_1 | Qwell: me bitching and complaining about gcc at this point.. :) so this is a common issue? I am experiencing to issues atm.. gsm sounds like garbage (ex: stock voicemail responses, etc.) and for some reason asterisk doesn't see pls-hold-while-try.gsm as even existing, in spite of the fact that 1.) it is in my sounds dir and 2.) it has the same perms as every other gsm file in there |
| 21:27.34 | tzafrir_home | Qwell, in what distro? |
| 21:27.38 | Qwell | tzafrir_home: all |
| 21:27.47 | Qwell | well, presumably |
| 21:27.50 | [TK]D-Fender | davidnicol, pastebin your actual peer entries and do a complete job this time, and include CLI output, SIP debug, etc. |
| 21:28.18 | tzafrir_home | There are no bugs abotu it in Debian |
| 21:28.22 | Qwell | nny_1: are you trying to play "pls-hold-while-try.gsm" or "pls-hold-while-try"? |
| 21:28.40 | tzafrir_home | And we ship gsm sounds by default, sadly |
| 21:28.54 | jblack | Ok. Regardless of using @ or not in front of the ip, I get the same result. My authenticated sip phone is dropping straight to the default context |
| 21:29.06 | nny_1 | Qwell: sry it is pls-hold-while-try |
| 21:29.09 | *** join/#asterisk alrs (i=non-knav@pozug.com) |
| 21:29.20 | nny_1 | Qwell: I assume it always seeks the best format based on settings |
| 21:29.24 | Qwell | it does |
| 21:29.40 | Qwell | tzafrir_home: http://bugs.digium.com/view.php?id=11243 |
| 21:29.41 | nny_1 | Qwell: funny thing is it just worked* 1 hour ago until my biz partner compiled in asterisk addons |
| 21:29.48 | davidnicol | what would one look for in SIP debug? Are there "this-is-my-port" entries in the SIP messages? |
| 21:29.59 | jblack | I put my extensions.conf and sip.conf at http://rafb.net/p/MS8Rzv21.html |
| 21:31.16 | nny_1 | Qwell: seems fair enough |
| 21:31.36 | nny_1 | Qwell: bug report describes most of my issues, i will deal with the missing gsm file that isnt afterwards |
| 21:32.10 | tzafrir_home | Qwell, thanks |
| 21:32.12 | syzygyBSD | anyone know where I can get better rejection line recordings |
| 21:32.54 | outtolunc | quit |
| 21:33.05 | [TK]D-Fender | syzygyBSD, thats a poor business decision for the phone-sex industry..... |
| 21:33.09 | nny_1 | syzygyBSD: my personal voicemail has some GREAT rejection recordings |
| 21:33.26 | nny_1 | :) |
| 21:33.44 | Havokmon | lol |
| 21:34.34 | syzygyBSD | nny_1: ya, I got one last night.... someone drunk dialed the wrong number... I was so confused |
| 21:34.38 | nny_1 | Qwell: have to recompile for snmp and mysql cdr support anyways.. will work on that this weekend.. nothing life threatening atm... i'll post back if anything unusual happens |
| 21:34.49 | syzygyBSD | [TK]D-Fender: you are back! |
| 21:36.05 | nny_1 | syzygyBSD: lol nice.. you think you were confused.. person that called you probable saved a load of embarassment on a DUI (Dialing under the influence) |
| 21:36.35 | jblack | Any suggestions of things I can look at? |
| 21:36.38 | Qwell | until said person brings it up to the intended recipient. |
| 21:36.51 | Qwell | "What? You never broke up with me over the phone..." |
| 21:37.05 | syzygyBSD | ya, too bad they didn't give me the right phone number, I would have delivered the message for them |
| 21:37.15 | syzygyBSD | every minute... for an hour |
| 21:37.42 | nny_1 | lol |
| 21:41.52 | *** join/#asterisk ZX81 (n=ZX81@121.90.79.233) |
| 21:42.41 | nhuisman_work | man I wish I had a few more months to wait for asterisk be version C, the current version is way out of date. |
| 21:43.05 | [TK]D-Fender | jblack, pastebin the complete CLI output of a failed attempt at verbose 10, sip debug enabled |
| 21:44.48 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:45.01 | jblack | Ok. |
| 21:45.14 | ZX81 | fujin: hi - shouldn't you be out enjoying the weather? |
| 21:45.23 | fujin | probably |
| 21:45.26 | fujin | only just woke up ;] |
| 21:45.46 | fujin | might go for a drive, soon. find some weed. |
| 21:46.33 | *** join/#asterisk ZX81_ (n=ZX81@121.90.79.171) |
| 21:46.55 | fujin | ugh. |
| 21:47.02 | ZX81_ | indeed |
| 21:47.13 | *** part/#asterisk ddunavant (n=David@66.170.97.28) |
| 21:47.24 | ZX81_ | downloading 2000 emails over a ~56k connection sucks |
| 21:47.42 | fujin | thank god for HSDPA |
| 21:47.48 | ZX81_ | heh |
| 21:48.02 | drmessano | 56k.. go find a Taco Bell |
| 21:48.15 | ZX81_ | lol yeah |
| 21:48.26 | ZX81_ | supposed to be faster but isnt |
| 21:48.46 | Qwell | [TK]D-Fender: send me one, I'll try it out |
| 21:48.52 | ZX81_ | :) |
| 21:49.13 | ZX81_ | bye all |
| 21:49.25 | [TK]D-Fender | Qwell, I would... but its be roaming where you are and you'd get smashed on data :) |
| 21:49.35 | Qwell | surely they can be unlocked :p |
| 21:49.54 | fujin | bleh, htc touch |
| 21:49.57 | [TK]D-Fender | Qwell, There is no data plan where you are that competes with mine :) |
| 21:49.59 | jblack | [KB]: Pastebin is still down, so I pasted to: http://rafb.net/p/C6bBiG72.html |
| 21:50.06 | Qwell | idc |
| 21:50.07 | Qwell | :p |
| 21:50.12 | fujin | reverse enginered freetouch dll ftw |
| 21:50.26 | [TK]D-Fender | fujin, Works wonderfully for me. Plays all my music & videos, browse the web, do e-mail, Google Maps on demand.... |
| 21:50.36 | Qwell | gmaps + gps? |
| 21:50.43 | fujin | touch doesn't have gps nah |
| 21:50.45 | fujin | but you can BT -> gps |
| 21:50.52 | Qwell | really? |
| 21:50.53 | [TK]D-Fender | Qwell, no gps.. bu supports an external one. |
| 21:50.55 | fujin | only tytn II has built in gps |
| 21:51.46 | jblack | [tk]d-fender: Ok, I have it up at http://rafb.net/p/C6bBiG72.html |
| 21:52.11 | [TK]D-Fender | jblack, Found no matching peer or user for '192.168.2.97:5088' |
| 21:52.16 | [TK]D-Fender | jblack, not authing.... |
| 21:52.21 | jblack | hmm. |
| 21:52.36 | jblack | Calls from IPKall are going to the phone though. |
| 21:52.53 | jblack | Oh, I know. I think I'm running two softphones on two machines at once. perhaps that confused things |
| 21:53.07 | JerJer | prolly need insecure=port |
| 21:53.16 | [TK]D-Fender | jblack, Yes, we are definitely confused now. |
| 21:53.20 | JerJer | or port=5088 ... in the peer |
| 21:53.56 | JerJer | but i have no real clue what is being talked about here - just popping in |
| 21:54.12 | jblack | I'll relog it, with just one softphone, with insecure=port added to sip.conf under the [jblack] entry |
| 21:56.40 | jblack | woot. insecure=port solved it |
| 21:58.17 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 22:09.43 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 22:11.01 | [T]ank | anyone interested in a sangoma a104d t1 card? only used for 1 month. then switched it out for SIP Provider. purchase price was about $2000. Would let it go for substantially less. |
| 22:25.10 | *** part/#asterisk nny_1 (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 22:39.13 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 22:39.29 | *** join/#asterisk RoyK (n=roy@ip-213-15-149-91.dialup.ice.no) |
| 22:40.29 | *** join/#asterisk RoyK (n=roy@ip-213-15-149-91.dialup.ice.no) |
| 22:49.49 | *** join/#asterisk RoyK (n=roy@ip-213-15-149-91.dialup.ice.no) |
| 22:54.55 | *** join/#asterisk nirz (n=nir@194.90.229.88) |
| 22:56.23 | Havokmon | I've connected a Fujitsu pbx to Asterisk via a PRI. when I try and call myself from the Fujitsu, asterisk says: Extension '6204' in context 'fujitsu' from '' does not exist. Rejecting call on channel 0/9, span 2 |
| 22:56.35 | Havokmon | Are the extensions context sensitive? |
| 22:56.49 | *** part/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 22:57.41 | Qwell | you mean will it look for 6204 in other contexts than fujitsu? no, it won't |
| 22:59.08 | *** join/#asterisk RoyK (n=roy@ip-213-15-149-91.dialup.ice.no) |
| 22:59.28 | *** join/#asterisk k2nt23 (n=rubsoft@190.40.226.242) |
| 23:01.53 | Havokmon | ok. |
| 23:03.11 | De_Mon | Havokmon you have to Include => context for it to do something like that |
| 23:03.17 | De_Mon | [fujitsu] |
| 23:03.23 | De_Mon | include => local-numbers |
| 23:03.38 | Havokmon | That worked. Is there another way to label your channels? |
| 23:03.50 | De_Mon | other than... |
| 23:04.08 | Havokmon | De_Mon - No just used the context to label my channels, so I knew what was what.. and I'm using some particular software package to create extensions ;) |
| 23:04.19 | Havokmon | contexts :) |
| 23:05.04 | Havokmon | hmm ok, I see there is a from-internal context.. I'll just play with it |
| 23:05.05 | Havokmon | Thanks :) |
| 23:05.12 | Havokmon | Have a good weekend all! |
| 23:05.16 | De_Mon | you too |
| 23:12.30 | jblack | Thanks for the help. ;) |
| 23:26.20 | _ShrikE | anyone here used signalogic boards with asterisk? |
| 23:27.44 | *** join/#asterisk Winkie (n=urmom@general-ld-220.t-mobile.co.uk) |
| 23:33.54 | nhuisman_work | does skinny support normally come with asterisk or do I need to compile it in? |
| 23:34.03 | *** join/#asterisk Einsteinium (n=99@hosted.serverspy.net) |
| 23:34.54 | lmadsen | should compile if you select it in menuselect |
| 23:35.12 | nhuisman_work | i'm using binaries |
| 23:35.15 | nhuisman_work | this is business edition |
| 23:37.20 | *** join/#asterisk vetetix (n=vetetix@83.222.34.12) |
| 23:38.51 | *** join/#asterisk RoyKa (n=roy@ip-154-11-149-91.dialup.ice.no) |
| 23:41.38 | lmadsen | hrmmm.... |
| 23:41.47 | lmadsen | oh, I don't think ABE has skinny |
| 23:41.55 | SwK | anyone have DIDs in Iraq? |
| 23:41.59 | nhuisman_work | every so clever of them. |
| 23:42.00 | lmadsen | SwK: I do |
| 23:42.13 | SwK | lmadsen, you do or lmh does? |
| 23:42.14 | lmadsen | nhuisman_work: it's because only complete channel drivers and features are included in ABE |
| 23:42.20 | lmadsen | SwK: neither -- I was lying |
| 23:42.25 | nhuisman_work | makes sense |
| 23:42.25 | SwK | haha |
| 23:42.29 | SwK | lmadsen, hows it going |
| 23:42.36 | lmadsen | SwK: not too shabby! you? |
| 23:42.48 | SwK | lmadsen, not too bad |
| 23:42.53 | nhuisman_work | man this upgrade from cisco to asterisk is going to be a one way rollercoaster to hell |
| 23:43.02 | nhuisman_work | there is no rolling back the phones if all goes sour |
| 23:43.17 | lmadsen | I'd probably do one and test then |
| 23:43.22 | nhuisman_work | yeah i'm going to |
| 23:43.30 | nhuisman_work | i guess one of each type of phone |
| 23:43.32 | lmadsen | Cisco phones can be switched from SCCP to SIP and back again |
| 23:43.37 | lmadsen | I've done it |
| 23:43.56 | nhuisman_work | anyone use cisco phones and know what version of the sip firmware is best? |
| 23:44.26 | lmadsen | ~bestquestions |
| 23:44.33 | lmadsen | ~best |
| 23:44.34 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 23:44.57 | nhuisman_work | laugh |
| 23:45.14 | nhuisman_work | best for a balance of stability and features |
| 23:45.24 | lmadsen | I've used 8.8 without a crash |
| 23:45.51 | nhuisman_work | i remember having fun with older versions |
| 23:45.58 | nhuisman_work | pressing lots of buttons power cycled the phones |
| 23:46.07 | nhuisman_work | 8.8 is the latest eh |
| 23:46.29 | lmadsen | yep |
| 23:46.55 | lmadsen | running it on my 7960, but it's just a test phone. I register it and use it to place and receive calls |
| 23:47.00 | lmadsen | I don't use anything fancy on it |
| 23:47.11 | lmadsen | 7.4 had no problems with long calls |
| 23:47.11 | nhuisman_work | you got a link for how to downgrade and put skinny back on a cisco phone? |
| 23:47.17 | lmadsen | haven't done a long call on this 8.8 though |
| 23:47.26 | De_Mon | crap crap crap. I misplaced my bridge() backport |
| 23:47.26 | lmadsen | I think you just reflash it from tftp |
| 23:47.32 | lmadsen | De_Mon: you bastard |
| 23:47.36 | lmadsen | let me find it again for you |
| 23:47.48 | De_Mon | I have that, but I improved it |
| 23:47.51 | De_Mon | and lost that one :( |
| 23:48.33 | lmadsen | ahhh |
| 23:48.36 | lmadsen | improved? |
| 23:48.39 | lmadsen | and you didn't send it back to me?! |
| 23:49.16 | nhuisman_work | anyone know of a nice little application to handle phones and their firmware? I know I can use tftp and build all the mac adress files and such but something that works and has a little gui or something would be nice. |
| 23:50.05 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:53.04 | nhuisman_work | can 7935's be upgraded to sip? |
| 23:55.37 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 23:56.50 | De_Mon | lmadsen nothing major, I swiped the original commit by russle with cleaned up formatting and massaged it to merge cleanly with the other debian patches for asterisk |
| 23:57.24 | ariel_ | nhuisman_work, look at a combo release for asterisk and freepbx like CentPBX it might be what your looking for |