| 00:01.52 | *** join/#asterisk Magicblaze0071 (n=sony@fl-67-235-208-62.dhcp.embarqhsd.net) |
| 00:02.14 | *** join/#asterisk Alfio (n=Amunoz@190.6.138.96) |
| 00:03.29 | LemensTS | amd opteron 246 2ghz, 2gb ram |
| 00:05.13 | citywok | single core 2gb proc? |
| 00:05.21 | citywok | good luck with 500 calls on that |
| 00:05.50 | *** join/#asterisk LeddyHM (n=NONE@you.cant.hack.thisbox.org) |
| 00:06.13 | citywok | and 100mbit may not actually be 100mbit at serverbeach, hosting facilities like that like to worse than comcast |
| 00:06.23 | citywok | like to oversell* worse than comcast |
| 00:26.09 | raden | besides the oreilly asterisk book is there any good documentation ? |
| 00:27.15 | beek | raden: What's wrong with the oreilly book? |
| 00:27.35 | raden | i have the first version :( |
| 00:27.53 | beek | Download the second, then purchase a copy of it. |
| 00:27.53 | LemensTS | citywok...what is an ideal processor? |
| 00:28.00 | beek | ~book |
| 00:28.01 | infobot | [~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 at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:28.02 | LemensTS | i can have them switch servers |
| 00:29.21 | *** join/#asterisk exsync (n=UserNick@pdpc/supporter/active/exsync) |
| 00:29.32 | beek | infobot: tell raden about book |
| 00:29.57 | raden | yeah which means i have to order it no book store around has it soo far by having this up by weekend |
| 00:30.11 | beek | raden: DOWNLOAD THE PDF |
| 00:31.45 | raden | im confussed i just downlaoded the entire book from downloads.oreilly.com/books/9780596510480.pdf for FREE ????????? |
| 00:31.55 | beek | Yes. |
| 00:31.56 | *** join/#asterisk s14ck (n=s14ck@190-76-112-68.dyn.movilnet.com.ve) |
| 00:31.57 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 00:32.19 | beek | Now do the right thing and purchase a copy of it so you don't have to hover over your monitor to read it. |
| 00:32.26 | beek | But that'll get you rolling before the weekend. |
| 00:32.41 | raden | beek, i going to buy a used one for 3.99 from amazon |
| 00:32.52 | beek | Ensure it's the second edition. |
| 00:32.53 | raden | why do they let u download it for free ???? |
| 00:33.02 | raden | yeah it is i called the place as well |
| 00:33.06 | beek | Because this is opensource... |
| 00:33.16 | beek | .. and the authors are doing a great service. |
| 00:33.42 | beek | Used books don't, however, provide any revenue for the authors. |
| 00:33.43 | raden | just weird the publisher would allow it thanks :) |
| 00:34.04 | beek | oreilly doesn't give 'em all away... this one, however... |
| 00:40.03 | joat | they've been giving this one away forever |
| 00:40.18 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 00:40.52 | Orbixx | How can I set up a call queue to ring one extension, then if said extension doesn't pick up in X seconds, it defers to another extension? |
| 00:41.11 | beek | Orbixx: in the same queue? |
| 00:41.17 | Orbixx | Yes. |
| 00:41.22 | beek | timeout |
| 00:41.36 | beek | Do you have a specific order? |
| 00:42.13 | Orbixx | Yes, one must definitely ring first, then another if no answer. |
| 00:42.28 | joat | why not ring 'em all? |
| 00:42.33 | beek | Give them an order of priority... |
| 00:42.41 | beek | Are these agents or members? |
| 00:43.00 | Orbixx | beek: Having read up on agents, members seems a better option for me. |
| 00:43.11 | Orbixx | joat: Because I want to have a chance to answer before anybody else does. |
| 00:43.19 | joat | ah |
| 00:43.25 | beek | Then just put a number after the member: member => DAHDI/1,1 |
| 00:43.31 | beek | Then DAHDI/2,2 |
| 00:43.34 | beek | DAHDI/3,2 |
| 00:43.34 | Orbixx | And if I'm not around, I want it to defer to somebody else. |
| 00:43.46 | Orbixx | Thanks. |
| 00:43.48 | beek | Orbixx: I just went through this. |
| 00:43.57 | Orbixx | beek: I was explaining to joat. |
| 00:43.58 | raden | !providers |
| 00:44.17 | beek | Orbixx: .... no, I meant the *I* just went through the same requirement. |
| 00:44.24 | Orbixx | Oh right. |
| 00:44.25 | Orbixx | :> |
| 00:44.33 | raden | anyone recommend good origination provider ? |
| 00:44.48 | beek | infobot: tell raden about itsp |
| 00:45.55 | Orbixx | beek: Would you mind sharing your queue config files? |
| 00:46.02 | Alfio | he can use strategy=linear |
| 00:46.26 | Alfio | and put in configuration the extensions in order to ring |
| 00:46.57 | beek | Alfio: If he wants them one at a time... otherwise he needs to number them. |
| 00:47.44 | beek | Orbixx: Are you going to have them ring in order, or you first and then everyone else? |
| 00:47.56 | Alfio | Orbixx> Yes, one must definitely ring first, then another if no answer. |
| 00:48.10 | *** join/#asterisk dug (n=chatzill@ppp-71-139-42-138.dsl.snfc21.pacbell.net) |
| 00:48.16 | dug | why are my sip calls ringing busy? http://pastebin.com/m7734a0d7 is says busy congested but the line is free |
| 00:48.16 | Alfio | if he always will have that order he can use it as startegy |
| 00:48.16 | Orbixx | Either works really, it's just two members. |
| 00:48.49 | beek | Orbixx: then Alfio is correct. Use strategy=linear, and put the member entries in the order in which you want them to rung. |
| 00:48.53 | beek | s/rung/ring/ |
| 00:49.02 | Orbixx | O.o |
| 00:49.09 | dug | I am using pstn with digium tdm400 and inbound calls work fine |
| 00:49.32 | Alfio | dug those are not sip calls you are having trouble placing calls outbound |
| 00:49.39 | Alfio | not sip problems |
| 00:50.14 | dug | Alfio: I understand its not sip problems but I dont know why my outbound route rings busy |
| 00:57.50 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 01:01.15 | beek | GN All |
| 01:01.20 | jaytee | nite beek |
| 01:01.27 | beek | CU jaytee |
| 01:13.14 | raden | ~itsplist-us |
| 01:13.15 | infobot | i guess itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 01:13.59 | *** join/#asterisk BadHAL (n=nn@174-156-14-222.pools.spcsdns.net) |
| 01:14.21 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 01:15.24 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-96ff06a2048359d6) |
| 01:16.41 | *** join/#asterisk haryv (i=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:17.15 | haryv | Evening or morning depending where you are at. |
| 01:17.42 | haryv | Anyone know of a virtual secretary service that can represent two or more business using voip technoligy? |
| 01:18.07 | drmessano | virtual secretary? |
| 01:18.23 | Alfio | you mean digital recepcionist |
| 01:18.31 | Alfio | haryv |
| 01:18.31 | haryv | yes, some one who is not on a per hour or salery basis. |
| 01:18.37 | haryv | no |
| 01:18.40 | haryv | not digital |
| 01:18.43 | haryv | live person. |
| 01:18.50 | drmessano | Accurate Messages |
| 01:19.17 | haryv | I cannot afford the rates a one person secretary would require and my startup calls are not often enough. |
| 01:19.42 | drmessano | I just gave you a name |
| 01:19.50 | haryv | okay |
| 01:21.01 | [TK]D-Fender | drmessano: back |
| 01:21.07 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-166-138.dsl.stlsmo.sbcglobal.net) |
| 01:21.08 | haryv | not really a answetring serive but somone who can take the customers contact,finacial and other details. Answering service is just that. Take name and number |
| 01:21.25 | drmessano | haryv: NEGATIVE.. Read the website |
| 01:21.56 | [TK]D-Fender | haryv: "Outsourced Inbound Call-center" <- no-one said they had to be too smart |
| 01:22.23 | drmessano | Accurate Messages will even take customer ORDERS |
| 01:22.26 | [TK]D-Fender | haryv: www.aheeva.com <- |
| 01:22.40 | drmessano | http://www.accuratemessages.com/products.htm |
| 01:22.51 | drmessano | They're also very reliable |
| 01:22.51 | [TK]D-Fender | haryv: And they are a Digium training partner as well as a call-center ( w/ Atelka.com ) |
| 01:23.29 | haryv | Good to know. |
| 01:24.12 | haryv | woman some times get skittish when I need physical address or contact information. Probebly would be better to forward that call to that service. |
| 01:24.34 | drmessano | [TK]D-Fender: rue_mohr can suck the big one, BTW.. his own DMM shows he's got a -11db on the damn line.. as well as his tests in dahdi, which he claims are false due to bug/errors |
| 01:24.41 | *** join/#asterisk xp_prg (n=xp_prg3@99.2.31.217) |
| 01:24.45 | xp_prg | anyone use drbd here? |
| 01:25.12 | [TK]D-Fender | drmessano: He provided the info that shoots him down? |
| 01:25.17 | drmessano | Yes |
| 01:25.29 | drmessano | http://eds.dyndns.org/~ircjunk/images/p1010034.jpg |
| 01:25.31 | [TK]D-Fender | looks around for his ClueBat... |
| 01:26.09 | drmessano | He states the .25 is DB... for some fucking reason.. thats the AC VOLTS reading at the top.. the -10.17 is the calculated DBM reading, which VERIFIES what DAHDI has been telling him ALL ALONG |
| 01:26.28 | *** join/#asterisk ingenius (n=alektro@host95.190-229-175.telecom.net.ar) |
| 01:26.41 | [TK]D-Fender | drmessano: We hide it in the "big print" ;) |
| 01:26.50 | drmessano | Dahdi isn't off by -11db.. HE IS |
| 01:26.58 | drmessano | and his OWN DMM PROVES IT |
| 01:27.26 | Alfio | xp_prg i do but if you want some info about it you can go to drbd channel |
| 01:27.36 | xp_prg | I did nobody is talking in there |
| 01:27.38 | drmessano | For months hes been bitching about phones, Digium, the cards, how everyone is so stupid in here.. that we know nothing about how to take a reading, line current/voltage, etc |
| 01:27.47 | drmessano | And his own fucking meter showed how dumb he is |
| 01:28.18 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 01:28.20 | drmessano | "At least I got mah chicken" |
| 01:28.21 | [TK]D-Fender | drmessano: Actually.... *I* know nothing about that :P |
| 01:28.34 | [TK]D-Fender | drmessano: Then again.. I keep nice & queit on those points :) |
| 01:28.40 | drmessano | lol |
| 01:29.05 | Orbixx | Man. |
| 01:29.05 | haryv | Now I need to market a better way. |
| 01:29.10 | [TK]D-Fender | drmessano: You've got a natural edge with your ham and electrical background |
| 01:29.11 | Orbixx | I love big rage-rants on idiots. |
| 01:29.27 | [TK]D-Fender | Orbixx: This one is about 8 months in the making. |
| 01:29.29 | haryv | Im also a ham and FCC licenced |
| 01:29.32 | drmessano | This is like piper69 finding out his parents are brother and sister, and him ignoring it |
| 01:29.54 | *** join/#asterisk mascool (n=mascool@c-76-112-230-56.hsd1.mi.comcast.net) |
| 01:30.05 | Orbixx | An idiocy beyond belief case, eh? |
| 01:30.17 | [TK]D-Fender | hears "Dueling Banjos" playing in the distance... |
| 01:30.42 | haryv | drmessano what level are you |
| 01:30.59 | [TK]D-Fender | Orbixx: Sometimes fairly smart people are capable of being the most hard-nosed morons. He out-thought himself into a corner. |
| 01:31.09 | drmessano | Orbixx: You can google rue_mohr and select terms for zaptel, dahdi, bugs, 11db, tdm800, etc.. and hes all about how Digiums card sucks, theres bugs in dahdi_monitor, how no one in here knows shit about electronics and how to test a line with a meter |
| 01:31.20 | drmessano | haryv: General |
| 01:31.25 | haryv | TK, you mean those who are not open minded |
| 01:31.31 | haryv | Tech+ here |
| 01:31.47 | [TK]D-Fender | <- 2-Bit hack |
| 01:31.52 | haryv | But need to upgrade...just the motivation and time is a issue. |
| 01:32.15 | drmessano | haryv: If it wasnt for the BS about the bands and the ionosphere, I would have my extra.. The technical stuff is second nature.. |
| 01:32.18 | haryv | plus need to get my canadian licence out of the way. |
| 01:33.29 | haryv | I had to obtain my FCC radio Telephone repair and operators licence year ago as a requirment to repair Avionics equipment. The proggram was very tough. Down to the componet level analog/digital troubelshooting. |
| 01:34.27 | haryv | Again, blame it on the US goverment for deregulating the big three industries in the late 80s for aviation falling on its face in 93. |
| 01:34.28 | haryv | :) |
| 01:35.18 | drmessano | You've not lived until you've spark tested a tube-type AM broadcast transmitter and its bridge rectifier boards by tapping them with a screwdriver |
| 01:35.36 | *** join/#asterisk bluecrow76 (n=MSharp@ip68-105-153-45.br.br.cox.net) |
| 01:35.52 | haryv | mmm those crystal sets were very cool |
| 01:35.52 | drmessano | The "Kapow" means youre close |
| 01:35.59 | haryv | that goes a LONG way back |
| 01:36.13 | haryv | Built a 300kv tesla coil once. |
| 01:36.16 | drmessano | Im talking about 5000 watt transmitters.. |
| 01:36.20 | haryv | okay |
| 01:36.21 | haryv | :) |
| 01:36.27 | haryv | realy rf power |
| 01:37.08 | drmessano | Largest I had the benefit of were 20,000 watt FM transmitters |
| 01:37.16 | drmessano | Those are the ones that make you pee a little |
| 01:37.22 | haryv | Commercial? |
| 01:37.26 | drmessano | Yeah |
| 01:37.36 | haryv | you have your FCC licence? |
| 01:38.21 | drmessano | No, you dont need a GROL or an operator license for commercial broadcast anymore |
| 01:38.23 | haryv | interesting. |
| 01:39.21 | drmessano | I dont think using a spare teflon spacer from a piece of 3 inch copper line to hold down the failsafe on a transmitter so you can keep the door open during testing is an answer you would see on the test anyway |
| 01:42.05 | haryv | FCC is more in line with band limits, transmission types for that band, bandwith and other laws. This is more aimed for the technician. I could have taken the radar endorsment part of the test but did not. |
| 01:42.36 | *** join/#asterisk LeddyHM (n=NONE@you.cant.hack.thisbox.org) |
| 01:42.57 | drmessano | FCC is more concerned about compliance.. They dont care how you stay compliant, but they demand that it be done |
| 01:44.22 | drmessano | I was the Chief Operator for 5 FMs and 2 AMs for a number of years and it was interesting how rule changes seemed to move more and more towards "We're not going to tell you how to monitor, or how to document.. but we expect to get the info we need, if/when we need it..period" |
| 01:45.49 | drmessano | Went from checking meter readings on transmitters every 2 hours, then 3 hours, then to "However long it takes to maintain compliance and to ensure if you do fall out of tolerance, the condition is addressed or corrected within 3 hours of it occuring" |
| 01:46.23 | drmessano | and also being able to turn said transmitter completely on and off remotely, within 3 hours of a notification from the FCC that you were out of tolerance |
| 01:46.45 | haryv | fun |
| 01:47.16 | haryv | Is that still the case today? |
| 01:47.26 | drmessano | The beauty part is.. in order to catch the intolerance in a 3 hour window, you either need 24/7 automatic monitoring with notification, or you need manual monitoring every 2 hours, 59 mins, and 59 seconds to ensure you can meet the 3 hour window |
| 01:47.28 | drmessano | Yes |
| 01:48.25 | haryv | Probebly in most cases, its automatic right? |
| 01:48.26 | *** join/#asterisk Kumbang (n=chic@rusnas.paume.itb.ac.id) |
| 01:48.54 | haryv | Dont want spurious harmonics to interfear with the frequency of another station :) |
| 01:49.18 | drmessano | Youre required to run a test of the Emergency Alert System once a week, document it.. along with a test recieved from all the sources you are required to monitor for EAS, along with the monthly tests, and any emergency messages retransmitted |
| 01:49.50 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:49.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:50.08 | drmessano | Spurs are not really much of a problem.. the problems are things like power going high or low by 8%, and modulation/tower current readings in the case of AM's |
| 01:50.30 | drmessano | Most monitoring is remote, but only some is automatic |
| 01:51.05 | drmessano | It also requires contact to the outside, which is often in the form of a phone line which is suseptible to being taken out by lightning and going unnoticed for hours or days |
| 01:51.08 | haryv | Right. AES is interesting. I volinteered for a ARES unit and the main county emergency ops showed me the equipment and how it worked. |
| 01:52.03 | haryv | EAS system I mean |
| 01:52.09 | drmessano | EAS is horrible abomination.. Designed from the start as complete crap.. the whole topology model is flawed, and even 12 years into it, most emergency managers are clueless on how its works and how to use it |
| 01:52.48 | drmessano | The new version is coming out here soon.. lot more based on internet standards.. |
| 01:52.53 | haryv | ours was very knowlagable. Former B52 officer who was in charge of the center. |
| 01:53.08 | drmessano | Most of the boxes will use Web based interfaces and the new protocol is XML based |
| 01:53.29 | haryv | interesting |
| 01:53.56 | drmessano | Its still gonna sound like shit being mistuned into the airchain of a 75 yr old AM station |
| 01:54.06 | drmessano | But its progress, nonetheless |
| 01:55.14 | haryv | I could have still volinteered for ARES but work was interfering with the times I wanted to attend. |
| 01:55.32 | drmessano | Ive actually been giving some though to being able to parse the new EAS in asterisk and use TTS to generate calls in asterisk to alert staff or in use for an alerting system in itself |
| 01:56.45 | haryv | you trust asterisk for that role? |
| 01:57.17 | drmessano | I trust Asterisk more than the sources of the information |
| 01:57.35 | drmessano | Have you ever seen the movie Wargames? |
| 01:58.23 | drmessano | Whole premise of the movie.. Cant trust the men in the silo's.. Need to get the men out of the silo's so we build this supercomputer that gets a mind of its own, right.. |
| 01:58.43 | drmessano | The biggest problem with EAS is the men in the silo's |
| 01:58.49 | drmessano | TURN YOUR KEY SIR |
| 01:58.52 | drmessano | TURN YOUR KEY |
| 01:59.03 | drmessano | EAS fails due to humans.. |
| 01:59.35 | coppice | many would consider that a success |
| 01:59.53 | *** join/#asterisk Alfio (n=Amunoz@adsl-50-82.tricom.net) |
| 02:00.17 | haryv | dr, ever been in a silo? |
| 02:00.31 | drmessano | Sadly, no.. |
| 02:00.49 | jaytee | If John Spencer hadn't have been smoking dope with his buddhist girlfriend before the missile exercise he would've turned the key. |
| 02:00.59 | coppice | I had a picture of the inside of a silo, but it was very grainy |
| 02:01.07 | haryv | Closest I have been to nukes was 1/4 mile away. 1/2 of the countries spare nukes stored on my base..just a guess :) |
| 02:01.21 | drmessano | Jaytee: Maybe Michael Madsen wouldn't have been so pissed |
| 02:01.22 | jaytee | I go to tour one back in the 70's when I was in the USAF |
| 02:01.33 | jaytee | s/go/got |
| 02:01.48 | haryv | Got caught up in a nuke transport. So the armed SP said no, you cannot go back on the road :) |
| 02:01.52 | drmessano | Jaytee: Stupid trivia.. that was his first movie role.. and he's completely uncredited |
| 02:02.13 | jaytee | I loved the guy as Leo on West Wing |
| 02:02.57 | haryv | I had a option to work in the space defence command. |
| 02:02.57 | drmessano | I still use that line as an inside joke.. even though I find myself explaining it more and more to these young whippersnappers |
| 02:02.58 | jaytee | kind of oddly prophetic that in the last season his character has a heart attack and survives and several months later he has a real one and doesn't |
| 02:03.06 | drmessano | TURN YOUR KEY SIR |
| 02:03.09 | drmessano | TURN YOUR KEY |
| 02:03.29 | haryv | Are silos still manned |
| 02:03.30 | haryv | ? |
| 02:03.33 | jaytee | yes |
| 02:03.40 | drmessano | They are? |
| 02:03.43 | drmessano | No WOPR? |
| 02:03.54 | jaytee | still the old two man rule in silos and on subs |
| 02:04.10 | haryv | I wonder how many today vs the height of the cold war |
| 02:04.12 | [TK]D-Fender | No, due to political-correctness they are now "personned" :p |
| 02:04.13 | drmessano | wants one of the plastic snappy things with the launch codes inside |
| 02:04.36 | drmessano | CPE1704TKS == WIN |
| 02:05.29 | drmessano | http://www.zazzle.com/cpe1704tks_tshirt-235576213815588207 |
| 02:05.30 | drmessano | YESH |
| 02:06.01 | jaytee | and back when the movie was new all the geeks envied Matthew Brodrick's rig with the dual IMSAI 8" floppy drives and 64K 8-bit CPM cpu. |
| 02:06.10 | [TK]D-Fender | Bueller? |
| 02:06.15 | [TK]D-Fender | Bueller? |
| 02:06.19 | [TK]D-Fender | Bueller? |
| 02:06.19 | jaytee | Anyone? |
| 02:06.36 | drmessano | Christy Swanson FTW |
| 02:06.44 | jaytee | "That's right, I'm Abe Froman, the sausage king of Chicago" |
| 02:07.16 | haryv | I once walked up to one of the three fences that were the perimiter to the nuclear warhead storage for the country. Very quick way to die if you start to climb it. I think one had 50,000 volt going though it. Heard a story of a procupine walking into it. Set off alarms ect. For some reason, first jolt did not kill it. Second one did when it walked into it again. |
| 02:07.18 | haryv | "_ |
| 02:07.52 | jaytee | where was this? Holloman AFB in New Mexico? |
| 02:08.29 | haryv | now this is good. Big ballon festival going on 10 miles to the NW. One of them decended behind the fence. With 10 MPS holding there M-16s at them :) |
| 02:08.33 | haryv | no |
| 02:08.57 | drmessano | "Um, he's sick. My best friend's sister's boyfriend's brother's girlfriend heard from this guy who knows this kid who's going with the girl who saw Ferris pass out at 31 Flavors last night. I guess it's pretty serious." |
| 02:09.08 | drmessano | Yay, found it |
| 02:09.24 | haryv | AF stories are great :) |
| 02:11.31 | haryv | not that some one dieing is. guy disolved in a C-130 herk fuel tank once. Pilots and crew chiefs could not figure why the engines were flaming out. |
| 02:29.22 | *** join/#asterisk Carlos_PHX (n=carlos@ip68-3-162-244.ph.ph.cox.net) |
| 02:32.08 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:32.16 | leifmadsen | omg sjobeck |
| 02:32.18 | leifmadsen | :) |
| 02:32.35 | leifmadsen | drmessano: yoink! |
| 02:38.52 | sjobeck | leifmadsen: word up |
| 02:39.12 | sjobeck | leif: how things? good to see you. |
| 02:39.34 | leifmadsen | oh not bad -- just working on getting a PHP SOAP interface working... |
| 02:43.16 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
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| 02:51.30 | darkmadda | does any one have method of dialing out via gizmo->googlevoice->PTSN ... Inbond is working, and i thought there was a way to do it. (Possibly via having GV dialback(using http request) to gizmo->Asterisk and then connecting the dialback to the outgoing call)... I think that should work i just don't know how to do it. Anyone already doing this? Or know how to do it? |
| 02:54.55 | *** join/#asterisk dongs (n=lol@l212168.ppp.asahi-net.or.jp) |
| 02:55.03 | dongs | whats _X in context? |
| 02:55.30 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:57.42 | leifmadsen | dongs: pattern match |
| 03:06.28 | [TK]D-Fender | dongs: Agains any single digit |
| 03:09.37 | drmessano | Anyone know of a specific issue where MWI works on 1.6 after a reload, but then stop after some period of time.. even with the poll mailboxes set to yes? |
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| 03:20.34 | darkmadda | any good forums for asterisk? |
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| 04:07.53 | jplank | Every box I upgraded from zaptel to dahdi, have all got echo after the upgrade. Not one of them had echo with zaptel. All digium hardware (mostly 2400Ps and TE122s). The echo module shows booted in dmesg, as well as theirs no errors. Could I be doing something wrong? |
| 04:13.16 | [TK]D-Fender | jplank: Distinctive lack of PASTEBIN is one.... |
| 04:17.30 | jplank | http://pastebin.ca/1502760 |
| 04:18.07 | jplank | keep in mind this is an upgrade from zaptel to dahdi, so there is still some zaptel references I need to clean up (like context names) |
| 04:18.44 | jplank | would a dmesg help also? |
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| 04:24.55 | jplank | should I just downgrade the boxes to zaptel? |
| 04:28.11 | [TK]D-Fender | jplank: echocanceller=mg2,1-23 <-- doesn't sound like HWEC to me. |
| 04:29.21 | jplank | Thats the default that dahdi_genconfig puts in, but digium told me that if I have hardware echo can, it over-rides any software provisioned |
| 04:30.00 | jplank | and dmesg shows the hw echo can working - VPMADT032: Present and operational (Firmware version 117) |
| 04:30.34 | jplank | I have no problem removing that from system.conf though |
| 04:33.10 | jplank | commented them all out, still the same |
| 04:33.32 | citywok | does anybody know how to configure Bria's dhcp options? |
| 04:34.37 | [TK]D-Fender | citywok: Bria is a soft-phone last I cheked... when would a SF ever have anything to do with controlling DHCP? |
| 04:34.55 | citywok | it uses DHCP to discover the SIP URL it uses to provision itself |
| 04:35.26 | citywok | they say to use option 120, but i havent found much documentation on it |
| 04:41.09 | jplank | would downgrading from dahdi to zaptel be as easy as removing chan_dahdi.so, then recompiling zaptel and then asterisk? |
| 04:42.37 | [TK]D-Fender | jplank: I'd advise you to use Digium support to figure this bit out. |
| 04:43.04 | [TK]D-Fender | jplank: DAHDI should work just fine. Its basically a rename + more features which you don't use anyway |
| 04:43.26 | jplank | they couldn't, they said they had to escalate it to their developers, and I haven't heard back |
| 04:45.29 | jplank | yea, thats what I don't get. If it worked with zaptel before upgrading, I don't see why it wouldn't work with dahdi. Whats weird is its across the board, I see this behavior on 3 different boxes, all running current 1.4 and dahdi |
| 04:46.47 | jplank | nice 1.4.12 version of zaptel automatically removes dahdi, might be worth trying |
| 04:54.42 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 04:56.34 | [TK]D-Fender | citywok: maybe it has access to the host's options sonhow... |
| 04:57.31 | jplank | some trade off. Its either half duplex audio because of HWEC on zaptel, or echo on dahdi :( |
| 04:59.28 | jplank | either that or I ditch digium and just get a sangoma and try with that |
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| 05:21.12 | [TK]D-Fender | checkout time, later |
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| 05:54.34 | joelsolanki | hey guys. |
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| 05:54.49 | joelsolanki | we have a running asterisk setup with zyxel ip dslams. |
| 05:55.10 | AlmightyOatmeal | yay? |
| 05:55.23 | joelsolanki | the problem is when customer press flash key on their telephone lines which are connected to ip dslams and ip dslams connected to asterisk it doesnt work |
| 05:55.53 | joelsolanki | means when there is 2nd call on their line they try to press flash to accept that call but the current call disconnects |
| 05:56.09 | joelsolanki | so is there any configuration for flash or dtmf on asterisk side ? |
| 05:56.28 | AlmightyOatmeal | did you check the ip phone dialplan? |
| 05:56.41 | AlmightyOatmeal | i assume you're using ip phones right? |
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| 05:58.11 | joelsolanki | no it is analog phone connected to ip dslams thru copper line |
| 05:58.27 | joelsolanki | if i tested with softphone and ipphone it works |
| 05:58.40 | joelsolanki | but things behind ip dslams with analog phone doesnt work :( |
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| 05:59.39 | AlmightyOatmeal | analog phones using ata adapters? |
| 06:00.47 | joelsolanki | some thing link linksys pap2 which gives out 2 analog lines |
| 06:00.49 | AlmightyOatmeal | i guess i don't know that much about dslams but it doesn't make much sense that a dslam would be causing such an issue |
| 06:00.54 | joelsolanki | so ip dslam is samme type |
| 06:00.59 | joelsolanki | ok |
| 06:01.20 | AlmightyOatmeal | i would check the dialplan config of the pap2 adapters for switching lines |
| 06:01.41 | AlmightyOatmeal | there should be a config option for switching lines and you need to figure out what your phone sense the pap2 adapter to switch |
| 06:01.55 | AlmightyOatmeal | if i understand your setup right |
| 06:03.05 | jplank | joelsolanki: I assume the dslam and * are connected with an IP connection? |
| 06:03.37 | jplank | if so, the dslam would have to be doing the flashhook, not asterisk, you can't really do a flash hook over an SIP (or whatever) trunk |
| 06:06.19 | joelsolanki | yes it is connected with ip connetiont |
| 06:06.38 | jplank | is the second call just ringing? |
| 06:06.46 | joelsolanki | oh |
| 06:06.53 | jplank | and the enduser off the dslam here the call waiting tone? |
| 06:07.05 | joelsolanki | yes it will ring and as soon as the we press flash the current call disconnects |
| 06:07.32 | jplank | that would be an issue with the dslam, maybe theres a call waiting setting inside it?' |
| 06:07.32 | joelsolanki | yes he listens it |
| 06:07.49 | joelsolanki | yes we activated it but nothing yet :( |
| 06:07.56 | jplank | what do you see at the CLI? |
| 06:07.57 | joelsolanki | so u think it must be on dslam side ? |
| 06:08.00 | jplank | yea |
| 06:08.03 | jplank | would have to be |
| 06:08.11 | joelsolanki | on cli it just says hangup |
| 06:08.14 | *** join/#asterisk Kihote (n=Kihote@203.113.138.75) |
| 06:08.18 | joelsolanki | let me see dslam setting |
| 06:08.18 | jplank | * doesn't know anything about the flashhook on the dslam side of the call |
| 06:09.10 | jplank | just to confirm (PSTN)----Asterisk----[ip connection]----DSLAM----[POTS]----enduser? |
| 06:10.30 | joelsolanki | yes corrects |
| 06:11.36 | jplank | the flashhook is on the DSLAM side, if the call is ringing, the asterisk side is done, if the enduser is hearing a call waiting tone, you seem to be half way there with dslam |
| 06:12.50 | joelsolanki | yes agre |
| 06:13.08 | Kihote | Hello |
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| 06:37.11 | WindowsUser | dhcp option 120 is apparently just a sip server |
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| 06:58.44 | tehokie_ | hi |
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| 07:45.34 | pif | hi, is it possible to have a different voicemail message depending on the time of day? |
| 07:45.53 | pif | (without creating a new voicemail account) |
| 07:47.06 | tzafrir_laptop | a cron to modify the file itself in the voicemail box? |
| 07:48.24 | pif | not bad :) |
| 07:48.47 | pif | or use the 'b'usy message (which is never used otherwise) |
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| 08:00.15 | pif | tzafrir_laptop: do you have good feedback on your 1.6.1.x debian package in large production settings? |
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| 08:06.33 | tzafrir_laptop | pif, not much |
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| 08:13.18 | tamiel | hello, is there a way to limit sip registering on a sip account ? |
| 08:13.40 | tamiel | (no call limit but register limit) |
| 08:32.09 | pif | when using labels with a Goto, is there a fallthrough after the label? |
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| 08:39.54 | pif | to test the existence of ${ARG3}Â in a macro should I simply test its emptiness? |
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| 09:13.52 | seggie | hi |
| 09:14.31 | seggie | if anyone has documentation about implenting TLS for Asterisk I'd be interested |
| 09:14.56 | seggie | google wasnt my friend on this one :-/ |
| 09:21.32 | tzafrir_laptop | Thread-Level Socket? |
| 09:22.01 | tzafrir_laptop | (the one that is an improvement over the Single-Socket-Level interface?) |
| 09:22.03 | seggie | nop, the security protocol |
| 09:22.15 | seggie | eeeuh |
| 09:22.37 | seggie | yes the SSLv3 if I remind correctly |
| 09:23.49 | tzafrir_laptop | anyway, if Google fails, try Yahoo? http://search.yahoo.com/search?p=asterisk+tls |
| 09:23.50 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:24.04 | seggie | transport layer security |
| 09:24.07 | seggie | haha will do |
| 09:24.21 | tzafrir_laptop | ah, not Thread Local Storage, then |
| 09:25.57 | seggie | guess I'm not good enough in English to get the jokes at first :P |
| 09:27.04 | *** join/#asterisk datacompboy (n=datacomp@213.187.251.250) |
| 09:27.50 | datacompboy | Hi all! Anybody knows what to do with g729? i have setup driver OK, core show transition are OK, voice ok, but console flooded with [Jul 22 09:25:42] WARNING[18017]: chan_sip.c:3707 sip_write: Asked to transmit frame type 64, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
| 09:29.27 | datacompboy | Have tried several different codecs -- all provide same result. call are bridged via app_conference. |
| 09:31.02 | seggie | anyway if anyone has successfully implemented TLS on an Asterisk system, I am interested as there is not a HOWTO on the net, just forums with unresolved problems :) |
| 09:37.08 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 09:37.19 | Micc | I'm going nuts trying to figure out a way to check if an extension exists. |
| 09:38.04 | Micc | I'm trying to goto a local dial if it is a call to a number in our system. |
| 09:38.21 | Micc | I have been using Dial(Local/${EXTEN}@inbound) |
| 09:39.05 | Micc | but it doesn't work well, some phones get garbled audio through the local channel. |
| 09:39.06 | Micc | I just want a way to see if the extension exists then goto it. |
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| 10:18.20 | WeazelON | hey guys, I got some mind bugling crazy question ! whenever i'm dialing Queue, and the calee hits * the call is being disconnected, and its not Feature related since its only for incoming queue calls i can't see the * being pressed in either of the logs nor the verbose anyone has an idea how I find what is causing * to disconnect ? (its not Hh in the Queue() ) |
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| 10:23.18 | WeazelON | kaii: no answer :( pretty dead here |
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| 10:41.58 | seggie | "SSL cert error", if it rings a bell... My asterisk.pem is the concatenation of key.pem and certificate.pem |
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| 10:45.51 | AlmightyOatmeal | from the united states, calling canada shouldn't require a different dialplan, right? |
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| 11:15.53 | kaii | WeazelON: i joined bit late, i dont have your pastebin in the backlog. can you give the url again pls ? |
| 11:25.23 | AlmightyOatmeal | i keep getting the error that the device is not registered to place calls on the network (x-lite softphone) :( |
| 11:25.33 | AlmightyOatmeal | it was working before i restarted the asterisk daemon |
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| 11:38.06 | ttl- | are numbers allowed in the sip.conf headings? |
| 11:45.04 | tzafrir_laptop | yes |
| 11:45.24 | tzafrir_laptop | section headings? [123456] ? |
| 11:45.26 | tzafrir_laptop | yes |
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| 11:47.44 | memph | hi |
| 11:55.09 | drmessano | So who can tell me from 1.2 or 1.4 > 1.6, what changes do I need to make for MWI to work, in general.. |
| 11:55.36 | drmessano | I know the pollmailboxes and pollfreq in voicemail.conf.. Any device changes.. anything else? |
| 11:56.14 | drmessano | and actually.. if my messages are not being modified outside of Asterisk, pollmailboxes is irrelevant |
| 11:56.23 | drmessano | correct? |
| 12:00.10 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:00.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:05.44 | *** join/#asterisk markwaters (n=markwate@192.249.39-62.rev.gaoland.net) |
| 12:06.27 | markwaters | I am trying to run a shell script in the dialplan , it reports that its called correctly but doesn't seem to work |
| 12:06.42 | markwaters | I can run the script from the console via ! script and it works fine |
| 12:06.45 | markwaters | any ideas ?> |
| 12:06.48 | *** join/#asterisk Lyma (n=Lyma@unaffiliated/lyma) |
| 12:07.25 | markwaters | script is in /usr/bin and is chmodded wit +x |
| 12:09.14 | [TK]D-Fender | markwaters: How does it report that it is called correctly? What user is * running as? (common mistake). SHOW US. |
| 12:09.16 | [TK]D-Fender | ~pb |
| 12:09.17 | infobot | [~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 , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 12:09.19 | [TK]D-Fender | ^^^ |
| 12:10.55 | markwaters | thanks [TK]D-Fender , asterisk is running as a user called asterisk who has a valid login and a shell set to /bin/sh |
| 12:11.10 | markwaters | I can su asterisk and run the script without a problem |
| 12:11.13 | [TK]D-Fender | markwaters: And what about your SCRIPT? |
| 12:11.21 | [TK]D-Fender | markwaters: Show us <- |
| 12:12.30 | markwaters | the script contains just "ssh mark@core3 nzbget -R 100" , I have also tried executing that command direct like system(ssh mark@core3 nzbget -R 100) |
| 12:13.08 | *** join/#asterisk Dovid (n=chatzill@tony09-118-62.inter.net.il) |
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| 12:13.30 | markwaters | in the dialplan as it executes I see 'Executing [s@macro-out-uri:2] TrySystem("SIP/mark-085b89c0", "remote_control_core3-slow") in new stack' |
| 12:14.27 | markwaters | I have tried putting just the script name and including the full path to /usr/bin too |
| 12:14.43 | markwaters | its flummoxed me now |
| 12:17.12 | kaii | markwaters: try TrySystem("/usr/bin/remote_control_core3-slow >/tmp/remotecontrol.out 2>&1"), then make a call, and provide a pastebin with the contents of /tmp/remotecontrol.out please |
| 12:17.39 | markwaters | ok , now I have run it as the asterisk user from the shell I am being asked about the ssh keys so I guess its working now |
| 12:17.59 | markwaters | kaii: ahh , that's a good idea , then I can see the logs |
| 12:18.03 | markwaters | ok , brb |
| 12:18.11 | drmessano | [TK]D-Fender: I have a transcendental asterisk question for ya |
| 12:19.26 | kaii | markwaters: if you can run the command with ! this doesnt mean anything, as your remote CLI (asterisk -r) must not run as the same user as asterisk does. |
| 12:19.47 | kaii | markwaters: and absolute pathes are always a good thing |
| 12:20.30 | [TK]D-Fender | drmessano: Any time now... :p |
| 12:20.32 | drmessano | [TK]D-Fender: Going from a working 1.4 system to a 1.6 system.. and ignoring all else.. If I told you my MWI on my phones worked great.. Now when I start/reload asterisk, I get an MWI.. but it times out after x number of minutes and stays in that state until I reload/restart |
| 12:20.45 | markwaters | kaii: [TK]D-Fender - its working now , thanks guys |
| 12:21.23 | drmessano | So the lights go out.. I call Alison and I have 8 new messages.. i reload and it picks right up that I have the new messages waiting |
| 12:21.25 | markwaters | kaii: try , I am running the asterisk console as root and not asterisk user , thanks for pointing that out |
| 12:21.41 | [TK]D-Fender | drmessano: Not a permissions thing? |
| 12:22.12 | [TK]D-Fender | drmessano: Otherwise i might suspect a bug in the specific vour you're moving to |
| 12:23.03 | drmessano | Well, not sure.. there is an initial detection of the new messages on reload/restart.. but then I guess theres supposed to be a repetitive notification.. or perhaps asterisk suddenly decides there are no new messages |
| 12:23.48 | drmessano | All I know is reload/restart.. that process detects.. the phones get a light.. and its correct for the ones that have new messages.. some magic doesnt happen from there |
| 12:24.13 | drmessano | What is this bug you speak of? |
| 12:24.27 | drmessano | or are you saying it sounds like it could be of that line? |
| 12:25.37 | [TK]D-Fender | drmessano: Could be a VM MWI issue with your .6 ver. I don't know anything specific, just a thought |
| 12:25.56 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 12:26.39 | drmessano | You know.. whats funny.. |
| 12:27.03 | *** part/#asterisk w3wt (n=sean@24-247-31-58.dhcp.bycy.mi.charter.com) |
| 12:27.23 | [TK]D-Fender | drmessano: a poodle with an angora sweater & a mohawk? |
| 12:28.12 | drmessano | When I tried IMAP Storage for VM.. I had recompile and go back to file based because ALL my phones were getting constant notifications of new messages.. and in the case of the ATA's that had dumb(er) POTS phones, they had studder tones every 60 seconds.. no decay, no timeout |
| 12:28.29 | drmessano | Even phones NOT CONFIGURED for VM |
| 12:28.46 | drmessano | I wrote it off as an IMAP Storage problem |
| 12:28.52 | drmessano | But this seems fishy and maybe related |
| 12:29.00 | drmessano | Like MWI is the issue |
| 12:30.10 | drmessano | Come to think of it, 60 secs was my pollfreq I had set |
| 12:34.05 | *** part/#asterisk dongs (n=lol@l212168.ppp.asahi-net.or.jp) |
| 12:37.54 | maxagaz | i have a strange problem with asterisk in china, when i make long distance call, as soon as the person i'm calling picks up the phone after some tonalities, i get nothing, it doesn't work |
| 12:38.01 | maxagaz | does someone have an idea ? |
| 12:38.09 | drmessano | Communism? |
| 12:38.33 | maxagaz | drmessano, the problem is definitely technical |
| 12:39.05 | maxagaz | drmessano, i don't have this problem for short distance call |
| 12:39.23 | maxagaz | neither when calling abroad |
| 12:39.24 | drmessano | Same provider? |
| 12:39.46 | maxagaz | drmessano, yes |
| 12:39.53 | maxagaz | drmessano, called CNC |
| 12:40.15 | maxagaz | drmessano, they say everything is fine from their side |
| 12:40.20 | drmessano | If you can call local, and call abroad, but long distance doesnt work, take it up with them.. its not a PBX problem |
| 12:40.29 | drmessano | They're wrong |
| 12:40.39 | drmessano | All you are doing is sending them calls.. they handle the switching |
| 12:41.07 | drmessano | Calls are being sent in the proper format? |
| 12:41.17 | drmessano | As in, the formatting of the number |
| 12:43.15 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:43.48 | maxagaz | drmessano, ok, then i'll call them back for the nth time tomorrow |
| 12:44.12 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:44.13 | maxagaz | drmessano, today they told me very few people use our system, that's why it could come from it! |
| 12:44.26 | jaytee | just read an interesting article about Nortel selling off it's Enterprise Solutions Business to Avaya for $745 million |
| 12:44.27 | maxagaz | but it used to work before |
| 12:45.27 | drmessano | maxagaz: it worked before, and you changed nothing on your end? |
| 12:46.35 | maxagaz | drmessano, i can't be sure |
| 12:46.49 | maxagaz | drmessano, but our install is quite basic |
| 12:47.08 | maxagaz | drmessano, i mean, we have a dedicated server for that |
| 12:47.25 | maxagaz | drmessano, with asterisk 1.4 |
| 12:48.00 | drmessano | If you feel confident youre sending the calls correctly.. and whatever changes you made wouldnt affect that.. and with local and abroad working, and LD being the issue.. I would suspect provider, provider, provider |
| 12:48.05 | afink | can an extension be a variable? |
| 12:48.49 | maxagaz | drmessano, ok thanks for your help, i'll bother them again and again tomorrow |
| 12:49.12 | afink | for instance exten => ${userexten},1,Dial(${userexten},20) |
| 12:49.56 | maxagaz | drmessano, today i also recompiled libpri to make some tests (get rid of a patch), but unsuccessfully |
| 12:50.22 | drmessano | ${EXTEN} |
| 12:50.45 | drmessano | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 12:51.41 | [TK]D-Fender | afink: No |
| 12:52.45 | afink | yep just tried no go |
| 12:53.07 | afink | wishful thinking |
| 12:54.11 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 12:54.12 | [TK]D-Fender | afink: Really? Where to do you think that variable comes from? Variables exist during the call. |
| 12:56.11 | afink | variables that I previously assigned |
| 12:56.33 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:57.56 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 12:59.40 | jaytee | drmessano, shouldn't you be doing your up-to-the-minute live on scene morning traffic report? |
| 12:59.51 | drmessano | lol |
| 13:00.33 | drmessano | Im supposed to be at work.. I played hookie because I wanted to wake up with difficulty breathing and to have to go to the doctor.. It was my master plan |
| 13:00.43 | jaytee | hehe |
| 13:00.48 | *** part/#asterisk markwaters (n=markwate@192.249.39-62.rev.gaoland.net) |
| 13:00.54 | drmessano | I was in bed thinking, I want to wake up gasping for air and in pain.. and BAM |
| 13:00.58 | jaytee | could be the swine flu |
| 13:01.18 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:f552:cc04:2162:3074) |
| 13:01.47 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:02.35 | drmessano | Could be |
| 13:02.59 | drmessano | Could be I sucked in 2 gallons of dust in the last two days installing workstations |
| 13:03.26 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:06.04 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:06.39 | *** join/#asterisk brad_mssw (n=brad@216.155.101.90) |
| 13:07.16 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:10.39 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
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| 13:13.03 | [TK]D-Fender | afink: Variables die at the end of a call and are local to just that call. |
| 13:15.06 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 13:16.08 | Katty | hi |
| 13:16.37 | leifmadsen | howdy! |
| 13:17.59 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:18.31 | Katty | it's going to be a beautiful day |
| 13:18.38 | Katty | full of bunnies, and sunshine. |
| 13:18.39 | Katty | and hugs. |
| 13:18.41 | Katty | hugs [intra]lanman |
| 13:18.56 | [intra]lanman | hugs Katty back |
| 13:18.58 | [intra]lanman | hi Katty |
| 13:19.09 | coppice | today started with limited sunshine - an eclipse |
| 13:20.09 | Katty | better limited sunshine than moonshine |
| 13:20.17 | [intra]lanman | says who!! |
| 13:20.26 | Katty | :P |
| 13:23.47 | guax | any ultimate guide for sip trunking between asterisk servers? im getting general problems with autentication and cache in astdb. It seems that asterisk is ignoring the peer configuration in favour of a deprecated config thats out of sip.conf but is in astdb |
| 13:25.52 | afink | [TK]D-Fender: globals |
| 13:26.04 | Katty | guax: i handed out my last copy 5 minutes ago. |
| 13:26.20 | guax | =~ |
| 13:26.21 | guax | =P |
| 13:26.33 | [TK]D-Fender | afink: You'll still hve to come up with something else... |
| 13:27.18 | afink | I did. it was pointless anyways. |
| 13:27.34 | Katty | do something not pointless! |
| 13:27.49 | Katty | go have a walk. |
| 13:30.55 | leifmadsen | ~lbnc |
| 13:31.13 | leifmadsen | infobot: lbnc is "luser brain not connected" |
| 13:31.14 | infobot | leifmadsen: okay |
| 13:31.26 | leifmadsen | (not in reference to anyone here -- just found it online :)) |
| 13:32.18 | coppice | and PBD == post brain death |
| 13:33.20 | *** join/#asterisk unasi7 (n=unasi7@80-218-32-110.dclient.hispeed.ch) |
| 13:33.44 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 13:35.11 | Xetrov` | What is a good bluetooth headset for use with softphone? |
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| 13:45.16 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
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| 13:47.41 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:48.18 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
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| 13:48.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:50.12 | FreakGuard | what's the avantage over specifing the priority over just follow by order in the file? |
| 13:50.54 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:51.53 | leifmadsen | eh? |
| 13:52.05 | leifmadsen | FreakGuard: you mean 1,2,3,4 vs. 1,n,n,n ? |
| 13:52.12 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 13:52.33 | leifmadsen | FreakGuard: or do you mean, "why is there priority numbers at all?" |
| 13:52.35 | FreakGuard | leifmadsen: I'm more like wondering why they even exist |
| 13:52.45 | leifmadsen | FreakGuard: how would you go to a particular line? |
| 13:53.03 | FreakGuard | leifmadsen: specify labels? |
| 13:53.04 | leifmadsen | lets say I wanted to jump to the 5th priority... how would i get there without it? |
| 13:53.11 | leifmadsen | FreakGuard: labels exist, yes. |
| 13:53.20 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:53.31 | leifmadsen | exten => extension,n(label),Application(argument) |
| 13:54.00 | [TK]D-Fender | FreakGuard: How would you store your dialplan in a DB if it didn't know the order of the instructions? |
| 13:54.52 | FreakGuard | [TK]D-Fender: just ordered by lines? |
| 13:55.05 | [TK]D-Fender | FreakGuard: DB's don't have an "order" |
| 13:55.29 | [TK]D-Fender | FreakGuard: FreakGuard Unless the order IS a field, hence "priorities" |
| 13:56.28 | FreakGuard | So why do we need them explicitly in the files? The order is given by following \n |
| 13:57.05 | leifmadsen | FreakGuard: probably the most logical reason is because there used to be priority numbers, and then we got rid of them in the dialplan, but asterisk still had the concept of a list of lines, so we just created the placeholder 'n' and added labels. Otherwise you'd have a totally different format to "upgrade" to, and using the format I just specified allowed the dialplan formatting to change very little |
| 13:57.36 | *** join/#asterisk propellerhead (n=yogurt2u@host224.190-136-235.telecom.net.ar) |
| 13:57.43 | FreakGuard | oke, accepted ;-) |
| 13:57.50 | leifmadsen | FreakGuard: you probably would be more comfortable with AEL if the priority labeling stuff really bothers you that much |
| 13:58.09 | FreakGuard | leifmadsen: doesn't bother me. I just want to know why |
| 13:58.10 | leifmadsen | priority numbers are a relic of asterisk history |
| 13:58.17 | FreakGuard | tought so |
| 13:58.37 | [TK]D-Fender | leifmadsen: A constant fact more like ;) |
| 13:58.48 | [TK]D-Fender | leifmadsen: AEL is a hack :p |
| 13:58.49 | leifmadsen | whatevs |
| 13:58.59 | leifmadsen | [TK]D-Fender: and so are you! |
| 13:59.03 | FreakGuard | [TK]D-Fender: it's good enough? |
| 13:59.17 | [TK]D-Fender | leifmadsen: thats 2-BIT hack to YOU sir! |
| 13:59.32 | [TK]D-Fender | FreakGuard: AEL is a fake-out over extensions.conf |
| 13:59.36 | leifmadsen | FreakGuard: it is just a dialplan abstraction that gives you a different format, but converts back to "dialplan" internally |
| 13:59.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:59.48 | leifmadsen | FreakGuard: it's an alternative format for the same subsystem |
| 13:59.50 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 14:00.15 | [TK]D-Fender | FreakGuard: 1 more layer susceptable to failure and while it makes some things cleaner it makes debugging more difficult and functionally limits you vs straight dialplan. |
| 14:00.45 | *** join/#asterisk academy (n=adam@unaffilated/academy) |
| 14:00.56 | FreakGuard | [TK]D-Fender: I'm pretty aware of the impacts of abstraction layers, thanks ;-) |
| 14:00.57 | leifmadsen | FreakGuard: although lots of people do like to use it, so if you end up liking it, go for it |
| 14:01.26 | *** join/#asterisk thegoat (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net) |
| 14:01.30 | academy | If I take a DID from a wholesale voip provider, do I need to maintain a trunk to the provider or do I give the provider my sip address and they connect as needed? |
| 14:01.30 | *** join/#asterisk moa_ (n=moa_@65-19-228-168.vnet-inc.com) |
| 14:01.59 | leifmadsen | academy: if just taking incoming you need to define a [peer] and register => to the provider |
| 14:02.02 | [TK]D-Fender | FreakGuard: Just be aware that syntax on that layer is more likely to change/break/etc so you'll have to be even more careful when upgrading. |
| 14:02.11 | leifmadsen | the register tells them where you are, the peer will authenticate the incoming call |
| 14:02.24 | academy | leifmadsen: I thought register was only needed behind nat |
| 14:02.25 | leifmadsen | [TK]D-Fender: I think he gets it |
| 14:02.45 | [TK]D-Fender | leifmadsen: I like the look of my own typeface ;) |
| 14:02.48 | leifmadsen | academy: register is always needed unless you have a static IP, and the provider has configured it at their end to always send to that IP |
| 14:02.55 | [TK]D-Fender | uses Comic--Sans |
| 14:03.05 | leifmadsen | academy: register == "telling the other end where I am" |
| 14:03.10 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 14:03.14 | thegoat | has anyone integrated asterisk with skype? |
| 14:03.27 | academy | leifmadsen: I do have a static IP. Is it normal procedure not to register in this case? |
| 14:03.48 | leifmadsen | yes |
| 14:03.53 | academy | ok, thanks |
| 14:04.05 | leifmadsen | well, not uncommon anyways |
| 14:04.10 | tamiel | Is there a way to limit number of registered sessions on same sip account ? |
| 14:04.18 | leifmadsen | thegoat: yes, Digium has with SFA |
| 14:04.31 | leifmadsen | tamiel: you can only have 1 registered session per account |
| 14:04.39 | leifmadsen | asterisk is a B2BUA, not a proxy |
| 14:04.48 | [TK]D-Fender | thegoat: ... |
| 14:04.50 | [TK]D-Fender | ~skype |
| 14:04.51 | infobot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 14:04.52 | [TK]D-Fender | ^^^^^ |
| 14:05.08 | [TK]D-Fender | thegoat: SkypeForAsterisk is not available yet, no ETA |
| 14:05.19 | leifmadsen | [TK]D-Fender: soon |
| 14:07.13 | [TK]D-Fender | leifmadsen: Next spring... SHARP! |
| 14:07.27 | leifmadsen | whatever |
| 14:07.29 | tamiel | leifmadsen: yes but for example, if I have on session and register a second session, the second session overwrite the first one |
| 14:07.38 | tamiel | on/one |
| 14:07.43 | leifmadsen | tamiel: huh? |
| 14:07.52 | leifmadsen | tamiel: that just sounds like a re-REGISTER |
| 14:07.59 | coppice | soon on a geological scale |
| 14:08.12 | *** join/#asterisk pawpro (n=pawpro@213.166.12.73) |
| 14:08.42 | thegoat | i've been trying to get on the skype sip beta program, but that is like pulling teeth |
| 14:09.03 | pawpro | Hello everybody! I'm using autodialout in 1.6.1 is it possible to "SIPAddHeader" while using autodialout in Asterisk? |
| 14:09.10 | leifmadsen | thegoat: there are already enough people in the closed beta program. You'll probably have to wait for the open beta program. |
| 14:09.28 | leifmadsen | wonders what "autodialout" is |
| 14:09.37 | jaytee | was just wondering the same thing :-) |
| 14:10.14 | pawpro | .call files |
| 14:10.53 | pawpro | actually i dont use the file extension but I ment putting a file into /var/spool/asterisk/outgoing with the callstring inside |
| 14:12.22 | [TK]D-Fender | pawpro: Yes... but not when using a SIP "Channel:". Go stare at the list of * channel types funny till it hits you... |
| 14:12.50 | leifmadsen | pawpro: use Local channels to do some dialplan logic before dialing the SIP channel |
| 14:13.02 | pawpro | ha! |
| 14:13.51 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 14:14.27 | leifmadsen | runs off to do some issue tracker work |
| 14:15.34 | tamiel | leifmadsen: I use a sip softphone, register with it and 1 min after, register with sip hard phone on same account : hard phone overwrite soft phone registering . |
| 14:16.00 | leifmadsen | tamiel: exactly -- you can't register two devices to the same peer definition |
| 14:16.04 | leifmadsen | like I said earlier |
| 14:16.07 | leifmadsen | they need to be separate peers |
| 14:16.14 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 14:18.00 | *** join/#asterisk TommyBJ (n=tommy@217-14-12-26-dhcp-osl.bbse.no) |
| 14:18.46 | TommyBJ | Is it possible to gain information about which queue agent that the caller got assigned to? |
| 14:19.26 | TommyBJ | Information that can be used in the dialplan. |
| 14:21.06 | *** join/#asterisk N3tw0rk (n=kris@r74-192-40-158.vctrcmta01.vctatx.tl.dh.suddenlink.net) |
| 14:23.05 | [TK]D-Fender | TommyBJ: Make it par of the exten you have it dial in the dialplan to reach them. |
| 14:23.12 | [TK]D-Fender | part* |
| 14:24.12 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:26.00 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
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| 14:26.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:26.08 | tamiel | leifmadsen: yes I agree with you. Thanks |
| 14:27.40 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 14:27.42 | angryuser | Hello, i am looking for a voip solution carrier grade, something like http://www.mera-systems.com. Any advices ? Thank you. |
| 14:27.52 | TommyBJ | [TK]D-Fender: I'm not sure I understood that. A end user enters the queue without knowledge to any of the extension for the queues agents. |
| 14:27.54 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 14:28.16 | kaii | angryuser: you might try yate or freeswitch on this. hopefully i wont be kickbanned now. :-) |
| 14:28.21 | [TK]D-Fender | TommyBJ: And how is your agent called? |
| 14:28.31 | *** join/#asterisk ntbourey (n=ntbourey@c-75-74-236-16.hsd1.fl.comcast.net) |
| 14:28.36 | leifmadsen | eyes the kickban button |
| 14:28.37 | ntbourey | Morning everyone |
| 14:29.20 | ntbourey | Has anyone ever had an issue with SMC Cable Modems/Routers? |
| 14:29.34 | angryuser | kaii, i need somthing with billing & ready to go, load balancing with failover , biz is not a problem |
| 14:29.50 | TommyBJ | [TK]D-Fender: The agent is a member in the queue. Static |
| 14:29.52 | kaii | angryuser: why not buy mera systems then? |
| 14:30.02 | [TK]D-Fender | TommyBJ: what KIND of member? |
| 14:30.37 | angryuser | for around 1000 calls simultanious maximum to start, for mera, i just never heard about them, i wonder if there are any alternatives |
| 14:31.03 | [TK]D-Fender | angryuser: Looked at SER? |
| 14:31.04 | angryuser | kaii, ^ |
| 14:32.03 | TommyBJ | [TK]D-Fender: Not sure what you mean by what kind of member. The members are defined in agents.conf and associated in queues.conf . |
| 14:32.20 | [TK]D-Fender | TommyBJ: "member =>" <--- |
| 14:32.28 | jplank | doubt you want to do IP Centrix with SER alone |
| 14:32.45 | jplank | unless SER implemented class 5 features....? |
| 14:32.49 | kaii | angryuser: of course there are alternatives .. nortel for example. but i dont have a good overview of carrier market |
| 14:33.00 | angryuser | [TK]D-Fender, yea, cdrtool + mediaproxy stuff, i took them also into consideration, but the billing system they provide is somehow ugly xD |
| 14:33.47 | jplank | angryuser: metaswitch, coppercom, nextone (if you only need routing capabilities), ser+* |
| 14:33.48 | TommyBJ | [TK]D-Fender: member => SIP/Tommy for instnace |
| 14:34.01 | angryuser | jplank, thank i will google that |
| 14:34.11 | coppice | Its a bit late to buy coppercom :-) |
| 14:34.13 | [TK]D-Fender | TommyBJ: that is not an "Agent" then. and I don't believe there is anything yuo can do for that. |
| 14:34.25 | jplank | FYI - I also posted it by cost, starting from high to low |
| 14:34.43 | jplank | a good metaswitch can cost you 200k + |
| 14:34.59 | angryuser | bah |
| 14:35.44 | jplank | If you aren't already well versed in VOIP and * and or SER, I really advise against trying to run a service based business with it |
| 14:36.03 | jplank | it sure is powerfull enough, but one typo could take out all your clients |
| 14:36.14 | jplank | and there wont be anyone to troubleshoot it for you |
| 14:36.17 | jplank | ect |
| 14:36.28 | Kobaz | heh |
| 14:36.42 | Kobaz | it's like starting a construction company without knowing how to use a hammer |
| 14:36.46 | angryuser | jplank, i am used to ser(opensips) and asterisk |
| 14:36.47 | jplank | exactly |
| 14:36.52 | TommyBJ | [TK]D-Fender: If I change it then.. does it matter? |
| 14:36.53 | *** join/#asterisk tokozedg (n=toka@95.104.37.29) |
| 14:37.46 | kaii | [TK]D-Fender, TommyBJ: would be possible to dial the agents via local channel (e.g. Local/Tommy@agent-dispatcher) and do a Dial() with M() answering macro. |
| 14:37.47 | jplank | if your well versed in ser and *, then why say "i need somthing with billing & ready to go, load balancing with failover" |
| 14:37.57 | jplank | thats not the opensource way of life |
| 14:38.09 | kaii | TommyBJ: depends on where/when in your dialplan you need the information "who" answered the call. and what you want to do with it. |
| 14:38.10 | [TK]D-Fender | TommyBJ: If it goes via dialplan you can make it part of the extension you process. |
| 14:38.12 | jplank | thats the go spend 200k+ on a softswitch way of life |
| 14:38.53 | Kobaz | asterisk can function as a switch, it can function as a pbx... one of the best uses of it, is as an application server |
| 14:38.53 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:38.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:38.56 | coppice | hey, what's $200k these days? a politician's annual bill for hookers? |
| 14:39.04 | TommyBJ | What I'd like to achieve is execute a external script ... or even a func_curl with a DTMF and the one who answered the call. |
| 14:39.20 | tokozedg | hi, if i sent fax over sip trunk, and on the other side i redirect incoming call to fxs gateway connected to fax, will it receive fax? |
| 14:39.21 | angryuser | coppice, haha |
| 14:39.22 | Kobaz | it does make a pretty good pbx/switch |
| 14:39.38 | Kobaz | you just have to test everything, to make sure the config and setup fits your needs |
| 14:39.55 | TommyBJ | tokozedg: Yes.. just be sure that the codec is not some crappy GSM :) |
| 14:40.49 | angryuser | thank you for help everyone have a nice day |
| 14:40.54 | kaii | TommyBJ: as said, you could dial your "agent" (aka static queue member, not agent) via a local channel ... in this local channel you dial your static queue member (SIP/Tommy) and use Dial option M to execute an answering macro. in this macro you have access to the information WHO answered the call. you can execute your script there. |
| 14:40.59 | ketema | hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this? |
| 14:41.15 | kaii | TommyBJ: its possible but ugly as hell, believe me. |
| 14:41.16 | tokozedg | TommyBJ, ok and as i saw there is way to look if it`s fax send to another client, and if its not sent to another? |
| 14:41.22 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 14:41.56 | jplank | kobaz I have an * box on the back end of our VOIP network using it as a feature server. built and maintained right, it works great. I haven't broke the 500 concurrent call mark yet, but I dont see it being a problem (all media reinvited out of the way unless needed) |
| 14:42.01 | TommyBJ | kaii: True enough.. but then I'd circumvent the queue functionality as such. And I don't want that. I wat the caller to stay in queue and be automatically assigned an agent. |
| 14:42.12 | kaii | TommyBJ, tokozedg: you will possibly loose some data (lines) in the fax due to timing issues. |
| 14:42.43 | kaii | fax over SIP is not performing well without the help of T.38 |
| 14:43.02 | TommyBJ | True |
| 14:43.05 | kaii | TommyBJ: it wont affect how your queue works |
| 14:43.40 | kaii | TommyBJ: caller will hear music and will be automatically assigned .. just like normal |
| 14:43.41 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 14:44.11 | TommyBJ | kaii: Ah... NOW I understand you ... and yes.. that's genius.. almost ;) |
| 14:44.20 | kaii | TommyBJ: with this local construct you can do a lot of other very cool things. for example print the callers wait time in the callerid display of your agent. |
| 14:44.29 | TommyBJ | Exactly |
| 14:44.34 | TommyBJ | Hmm... cool. |
| 14:44.40 | TommyBJ | Definetly a way to go. |
| 14:44.45 | kaii | i did that for a call center few month ago. with asterisk 1.2 :PÜ |
| 14:44.55 | TommyBJ | Will probably spawn A LOT of local channels from time to time.. but that's worth it |
| 14:44.57 | *** join/#asterisk imcdona (n=imcdona@c-24-19-203-112.hsd1.wa.comcast.net) |
| 14:44.58 | TommyBJ | hehehe |
| 14:45.06 | TommyBJ | I have 1.4.21 to work with :) |
| 14:45.09 | TommyBJ | Lucky me |
| 14:45.16 | kaii | lucky you. |
| 14:46.26 | TommyBJ | Thanks alot for the tips. That saved me from using my head. |
| 14:46.43 | kaii | :-P |
| 14:47.19 | kaii | and it saved me from continuing work on my ugly 200+ lines sql queries |
| 14:47.45 | TommyBJ | hehe |
| 14:47.50 | TommyBJ | In that case.. You're welcome ;) |
| 14:49.32 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:49.46 | *** join/#asterisk imcdona (n=imcdona@c-24-19-203-112.hsd1.wa.comcast.net) |
| 14:52.07 | *** join/#asterisk encbladexp (n=stefan@p5495F8F1.dip.t-dialin.net) |
| 14:52.10 | encbladexp | hello |
| 14:52.28 | encbladexp | anybody here who uses Asterisk 1.4 and chan_capi? |
| 14:52.44 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 14:54.18 | tokozedg | is it enough to set t38pt_udptl = yes in sip.conf for user and enable T.38 in fxs to enable T.38 for fax? |
| 14:54.54 | *** join/#asterisk tokozedg (n=toka@95.104.37.29) |
| 14:55.23 | tokozedg | is it enough to set t38pt_udptl = yes in sip.conf for user and enable T.38 in fxs to enable use T.38? |
| 14:56.26 | ntbourey | Has anyone ran into issues with AGI not getting correct DTMF tones when executing a "wait for digit"? |
| 14:57.35 | *** join/#asterisk imcdona (n=imcdona@c-24-19-203-112.hsd1.wa.comcast.net) |
| 14:58.04 | ariel_ | tokozedg: have you read this? http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 |
| 14:58.36 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:59.50 | kaii | encbladexp: that was a metaquestion. i assume lot of people do. just ask your question. |
| 15:00.09 | encbladexp | ok, i live in Germany |
| 15:00.16 | encbladexp | when i vall a Handy which is not reachable |
| 15:00.26 | tokozedg | ariel_, yes but there is not what i actually want, as i guess |
| 15:00.29 | encbladexp | i get an Error Answer "Called Party balbalba".... |
| 15:00.42 | Chainsaw | encbladexp: Best use cellphone instead of "Handy". |
| 15:00.44 | encbladexp | when i call with Dial(.../b) i hear this Message |
| 15:00.58 | encbladexp | s/Handy/cellphone/g |
| 15:01.00 | encbladexp | ;-) |
| 15:01.01 | Chainsaw | encbladexp: That word isn't used outside of Germany, no matter how english you guys make it sound when you say it :) |
| 15:01.28 | encbladexp | Chainsaw, Handy feels like an english Word to us :-D |
| 15:01.38 | Chainsaw | encbladexp: But your telco is playing a voice announcement instead of giving a proper signalling tone, basically? |
| 15:01.49 | encbladexp | exactly |
| 15:01.57 | encbladexp | that is okay for me |
| 15:02.00 | *** join/#asterisk many (n=many@dslb-188-098-012-145.pools.arcor-ip.net) |
| 15:02.19 | ariel_ | thinks Handy was a name, I would have never think it was a cell phone. |
| 15:02.22 | encbladexp | but, if i call without /b, i get a ring Signal Indication on the calling party |
| 15:02.38 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 15:02.41 | encbladexp | ups, wrong |
| 15:02.48 | encbladexp | i get NOTHING on the calling party |
| 15:02.56 | encbladexp | no noise, no ringing, nothing |
| 15:03.09 | encbladexp | but i know that i should get such Voicemessage |
| 15:03.11 | Chainsaw | ariel_: Yeah, one of those things you just have to know :) |
| 15:03.38 | encbladexp | anotherproblem ist a 1-2 Seconds Delay if i answer a call |
| 15:03.48 | encbladexp | or a call i make gets answered |
| 15:04.09 | encbladexp | i call 0164648468 an dont hear the "Hello" from the Other side :-( |
| 15:04.32 | encbladexp | but, if i enable the Dial() m-Flag, it works (i hear the Hello) |
| 15:04.49 | encbladexp | but, in this case i dont hear the "Phone not reachable" Message :-( |
| 15:04.56 | ariel_ | So you have 2 issues, one when you dial out to that service that has a cell on it, you don't hear the ringing. But does the cell phone actually ring? |
| 15:05.02 | kaii | encbladexp: what are your options for Dial() ? could you provide the full extension? |
| 15:05.23 | encbladexp | ariel_, the Cellphone is switched off |
| 15:05.25 | tokozedg | thanks guys for your attention |
| 15:05.43 | encbladexp | i want to hear the "Not reachable Message", or a Busy Tone, or Something else |
| 15:06.48 | encbladexp | http://paste.pocoo.org/show/7JGJGvGXpMKg8psQEGL1/ this is the related part for calling out of my Asterisk Box |
| 15:07.07 | encbladexp | (Hardware ISDN is a AVM B1 PCI 4.0 Controller with latest chan_capi 1.1.2) |
| 15:07.20 | encbladexp | takes some food, i am Back in 5 Minutes |
| 15:08.25 | kaii | as i'm not using CAPI i dont know what /b means |
| 15:08.55 | *** join/#asterisk scruz (n=scruz@196.216.253.116) |
| 15:08.59 | scruz | hey everyone |
| 15:10.35 | scruz | is there a place where you can modify where asterisk picks up call files from? i have asterisk running, but it isn't processing my call files. i've even rebooted, and the file's still there |
| 15:11.11 | *** join/#asterisk imcdona (n=imcdona@96.9.161.246) |
| 15:12.43 | scruz | any ideas? |
| 15:13.13 | [TK]D-Fender | scruz: Could be timestamp, file locking when it got there, rights, etc |
| 15:15.27 | encbladexp | kaii, /b means "Early B3" |
| 15:15.54 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 15:16.10 | encbladexp | also known as "inband call progress" |
| 15:16.37 | encbladexp | you get the Ringing Indication and much more from the Telco, and not from Asterisk |
| 15:16.43 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-166-138.dsl.stlsmo.sbcglobal.net) |
| 15:16.56 | encbladexp | so the "The called number is unavliable at the moment" Messages work |
| 15:17.06 | LemensTS | can sox do g729 to wav, and wav to g729? google is giving me mixed answers |
| 15:17.27 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:17.32 | encbladexp | the biggest Problem is the 1-2 Second Delay for Calls (inbound, outbound) |
| 15:17.47 | encbladexp | the first 1-2 Seconds are Missing, not more, not less |
| 15:17.58 | afink | how can I make it so that a caller can tell if the person they called is on the other line? |
| 15:19.36 | scruz | [TK]D-Fender: ok. will run it live from the server and see how that goes |
| 15:20.01 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-131.gtconnect.net) |
| 15:22.00 | *** join/#asterisk chendy (n=chendy@61.141.250.70) |
| 15:25.52 | Qwell | LemensTS: essentially no |
| 15:26.21 | Qwell | LemensTS: even if you had a codec and could convince it do so...you couldn't legally use it (unless you licensed said codec) |
| 15:27.03 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:27.39 | *** join/#asterisk klochan (n=Klochan@195.222.70.1) |
| 15:28.29 | coppice | the wonderful things about googling is some deranged idiot will have posted a comment somewhere in some forum saying practically anything. You'll even find comments that "Microsoft Works" |
| 15:29.28 | voipheroes | f*ck*ng heretik. |
| 15:29.31 | ariel_ | lol, almost makes you think that bing will work... but not. |
| 15:29.49 | [TK]D-Fender | ariel_: It'll work. Will anyone CARE? |
| 15:30.27 | *** join/#asterisk pmhaddad-lappy (n=Phil@adsl-99-169-190-209.dsl.applwi.sbcglobal.net) |
| 15:30.31 | coppice | I wonder what you get if you look up White Christmas on Bing? |
| 15:30.33 | ariel_ | at least there ads are dead on |
| 15:30.47 | pmhaddad-lappy | hi all. i am having an issue with connecting 2 pbx |
| 15:30.49 | [TK]D-Fender | coppice: Funneh |
| 15:31.10 | ariel_ | 2 pbx connections are fairly easy |
| 15:31.27 | pmhaddad-lappy | ariel_, hang on i hit enter too fast ;) |
| 15:31.36 | ariel_ | plug one in to the other either via E1 or T1,,, or are you doing this via sip.h323, or iax2 |
| 15:32.07 | *** join/#asterisk klochan (n=Klochan@195.222.70.1) |
| 15:33.15 | ariel_ | funny, asterisk is and isnot a pbx, it can be a g/w, and be both or many things in one. There is no end to what you can do with it...almost makes coffee. |
| 15:33.41 | Qwell | ariel_: s/almost// - it's been done. |
| 15:34.10 | ariel_ | Qwell: it can start a coffee maker, but it actually can't make the coffee....;-) |
| 15:34.32 | [TK]D-Fender | Asterisk is a Telephony & PBX toolkit. |
| 15:34.46 | [TK]D-Fender | ariel_: And there are plenty of things you can't do with it... |
| 15:34.51 | ariel_ | Qwell: your support department takes forever to reply to email....argh |
| 15:35.04 | Aw0L | is the only real requirement for asterisk (aside from hardware) a voip service? |
| 15:35.23 | pmhaddad-lappy | hi all. i am having an issue with connecting 2 pbx's via SIP. The relevant portions of sip.conf and the errors are posted here: http://pastebin.com/m265563fb |
| 15:35.25 | ariel_ | [TK]D-Fender: there are always things you can't do...with or without.... just depends on time, money and know-how... |
| 15:35.31 | pmhaddad-lappy | ariel_, ^^ |
| 15:35.45 | ariel_ | Aw0L: no |
| 15:35.53 | [TK]D-Fender | ariel_: Thats like saying "yes your car can fly, you just need to build a plane around it". |
| 15:36.07 | ariel_ | [TK]D-Fender: yep |
| 15:36.12 | [TK]D-Fender | ariel_: * isn't a SIP proxy. What would it take to make it one? Answer : complete rebuild. |
| 15:37.02 | [TK]D-Fender | ariel_: Just like the car, it wouldn't be "asterisk" any more unless Digium took those changes internally. For which anything remotely sizable is a major undertaking and of course quite possibly refused. |
| 15:37.10 | Aw0L | ariel_: is there a specific doc to read that would give me a good conceptual understanding of everything that's needed? |
| 15:37.30 | ariel_ | ~docs |
| 15:37.31 | infobot | i guess docs is for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 15:37.34 | [TK]D-Fender | Aw0L: Who said you needed a VoIP service? |
| 15:38.05 | Aw0L | [TK]D-Fender: no one, I just made an assumption....I really am clueless with any phone-related techs |
| 15:38.25 | *** part/#asterisk Carlos_PHX (n=carlos@ip68-3-162-244.ph.ph.cox.net) |
| 15:38.33 | [TK]D-Fender | Aw0L: What do you actually want to do? |
| 15:38.41 | ariel_ | pmhaddad-lappy: you should not have put your pw in the pastbin |
| 15:38.56 | [TK]D-Fender | pmhaddad-lappy: And you DIDN'T put the failed call in there. |
| 15:39.00 | *** join/#asterisk rue_mohr (n=Dennis@24.207.122.10) |
| 15:39.14 | ariel_ | [TK]D-Fender: you beat me to that post |
| 15:39.17 | pmhaddad-lappy | oooh crap |
| 15:39.19 | pmhaddad-lappy | sorry hang on |
| 15:39.21 | [TK]D-Fender | rue_mohr: Saw a nifty multimeter reading pic yesterday.... |
| 15:39.24 | rue_mohr | [Jul 22 08:36:31] ERROR[30498]: pbx.c:1565 ast_func_write: Function VOLUME not registered ??? |
| 15:39.42 | Aw0L | [TK]D-Fender: just a soho phone system - something with caller menus and redirects to my cell...I was considering a paid virtual pbx service (they're cheap) |
| 15:39.44 | rue_mohr | you saw mine I think |
| 15:39.47 | [TK]D-Fender | rue_mohr: What ver? |
| 15:39.49 | ariel_ | Aw0L: you can setup asterisk with E1, PSTN's, hard lines and many other things not using voip |
| 15:39.51 | [TK]D-Fender | rue_mohr: Yes, that'd be the one... |
| 15:40.00 | [TK]D-Fender | rue_mohr: that indicates your line is bad... |
| 15:40.04 | Aw0L | ariel_: aaaah.... |
| 15:40.20 | rue_mohr | 1.4.25.1 |
| 15:40.23 | [TK]D-Fender | Aw0L: What do you want calls to come IN on? |
| 15:40.31 | [TK]D-Fender | rue_mohr: func_volueme = 1.6+ |
| 15:40.40 | rue_mohr | no, what you saw was the milliwatt app from out the channelbank |
| 15:41.05 | rue_mohr | the milliwatt app (the new one) is out by -11db |
| 15:41.06 | pmhaddad-lappy | [TK]D-Fender, ariel_ the passwords are there... |
| 15:41.13 | [TK]D-Fender | rue_mohr: Well you might want to clarify that with drmessano |
| 15:41.14 | rue_mohr | milliwatt(o) works fine |
| 15:41.15 | Aw0L | [TK]D-Fender: that's what I'm not sure of - for some of the paid services, I would need no physical phone lines or service, which I don't have now side from a cell |
| 15:41.27 | rue_mohr | there is no point, me and russel are fixing it |
| 15:41.54 | rue_mohr | my line was out by 3.5db, I called the telco and had them adjust it |
| 15:41.58 | pmhaddad-lappy | ariel_, http://pastebin.com/m21ad4467 |
| 15:42.00 | rue_mohr | recently |
| 15:42.02 | pmhaddad-lappy | that has the cli output |
| 15:42.22 | rue_mohr | [TK]D-Fender, but I'm glad you saw the pic :) |
| 15:43.27 | [TK]D-Fender | rue_mohr: Well I personally didn't have any context from it, but it doesn't say much to me anyway |
| 15:43.35 | ariel_ | pmhaddad-lappy: failed message tells you why it's not working. How are you sending the calls between them? |
| 15:43.35 | rue_mohr | thats ok |
| 15:43.36 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:43.54 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:43.59 | [TK]D-Fender | pmhaddad-lappy: And we don't see your dial. Also note the call is from 1191 <-- not the right user name |
| 15:44.01 | rue_mohr | I'm glad to see someone ack'ing that I can read analog levels |
| 15:44.05 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 15:44.16 | [TK]D-Fender | rue_mohr: Oh, that pic doesn't prove what I'm looking at ;) |
| 15:44.30 | [TK]D-Fender | rue_mohr: And I wouldn't know what means what there anyway :) |
| 15:44.34 | [TK]D-Fender | <- not an electrician |
| 15:44.56 | rue_mohr | its the little dbm reading at the bottom of the lcd that important |
| 15:45.00 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:45.10 | ariel_ | I want Pizza.... |
| 15:45.18 | ariel_ | ops wrong window..... |
| 15:45.32 | coppice | [TK]D-Fender all it takes is a screwdriver, and a pair of wire cutters |
| 15:45.50 | *** join/#asterisk mascool (n=mascool@c-76-112-230-56.hsd1.mi.comcast.net) |
| 15:46.12 | Qwell | rue_mohr: should show the picture to coppice and see about getting his opinion of it :p |
| 15:46.13 | [TK]D-Fender | coppice: An axe will do just fine :) |
| 15:46.14 | rue_mohr | thats ok, I'm just trying to trace down where my various problems are, the office is really hard for me, cause its all sip, I cant meter anything, I dont know if the volume problem I'm having is the card, asterisk, or the polycom set, I have have no idea, |
| 15:46.23 | pmhaddad-lappy | [TK]D-Fender, i know - thats the issue... I'm not sure what else you need to see.. uh this is the dial: -- Executing [s@macro-outbound:2] Dial("SIP/1191-00162c10", "SIP/LakeLinden_ REMC/92311001") in new stack |
| 15:46.32 | rue_mohr | coppice, |
| 15:46.41 | pmhaddad-lappy | :( sorry didn't think it was going to do that |
| 15:46.48 | coppice | [TK]D-Fender you play guitar, so you must already have one of those |
| 15:47.05 | timeshell_atwork | Is there a way to get the directory function to use festival to say the name rather than spell it out? |
| 15:47.06 | rue_mohr | coppice, http://eds.dyndns.org/~ircjunk/images/p1010034.jpg |
| 15:47.18 | ariel_ | pmhaddad-lappy: are you using freepbx or something like that? |
| 15:47.23 | pmhaddad-lappy | ariel_, nope |
| 15:47.29 | pmhaddad-lappy | this is just asterisk 1.6.0.9 |
| 15:47.48 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.48 | Aw0L | ariel_: [TK]D-Fender thanks, I'll do some more reading |
| 15:47.49 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 15:47.49 | rue_mohr | I should make a text clip for that image |
| 15:47.51 | ariel_ | so your using a macro what is the actual exten => command you have for your dial? |
| 15:47.53 | pmhaddad-lappy | and basically i just need to send calls from remote extensions from the remote pbx out to the local one |
| 15:47.56 | [TK]D-Fender | pmhaddad-lappy: And timeshell_atwork get them to RECORD their name |
| 15:48.01 | [TK]D-Fender | timeshell_atwork get them to RECORD their name |
| 15:48.17 | pmhaddad-lappy | ariel_, oh one sec |
| 15:48.23 | timeshell_atwork | Really? |
| 15:48.24 | timeshell_atwork | lol |
| 15:48.31 | [TK]D-Fender | timeshell_atwork: SMRT :p |
| 15:48.33 | timeshell_atwork | Haven't tried that yet. |
| 15:48.46 | rue_mohr | the top row is my co line, the meter is on the 2nd channelbank line, the line is dialed into the new milliwatt signal, note is says -10.17dbm |
| 15:48.47 | ariel_ | I need to move to the lab, hope not to get desconnected but brb |
| 15:49.07 | pmhaddad-lappy | ariel_, exten => s,n,Dial(SIP/LakeLinden_REMC/${MACRO_EXTEN}) |
| 15:49.40 | rue_mohr | https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=15386 |
| 15:50.35 | drmessano | rue_mohr: Your meter reading says it all |
| 15:50.47 | pmhaddad-lappy | ariel_, that's the dial command in the macro |
| 15:51.18 | [TK]D-Fender | pmhaddad-lappy: when you use [LakeLinden_REMC] to call the other side.. that's the username it should be looking for. there is no entry with that name on the other side |
| 15:51.42 | pmhaddad-lappy | [TK]D-Fender, you mean in sip.conf? |
| 15:51.50 | [TK]D-Fender | pmhaddad-lappy: Yes. |
| 15:52.05 | rue_mohr | at hte office what I really need, softphone or not is a way to know that signal is getting to the phones properly |
| 15:52.08 | pmhaddad-lappy | so i should have a [LakeLindenREMC] context on the remote pbx's sip.conf |
| 15:52.27 | pmhaddad-lappy | its there |
| 15:52.33 | *** join/#asterisk mattbUK (n=mattbUK@77-98-137-62.cable.ubr07.stav.blueyonder.co.uk) |
| 15:52.42 | rue_mohr | I have the tdm800 properly adjusted for line volume, (needs to be set to 1.9db) but my receptionsts still cant properly hear callers |
| 15:52.48 | drmessano | rue_mohr: Your own meter is showing your line level at -10.17dbm |
| 15:52.51 | pmhaddad-lappy | [TK]D-Fender, its also in the pastebin |
| 15:53.09 | rue_mohr | drmessano, thats the signal from milliwatt |
| 15:53.22 | pmhaddad-lappy | unless i'm missing what you're saying |
| 15:53.30 | drmessano | and? |
| 15:53.39 | rue_mohr | if I do milliwatt(o) I get about the 0dbm I should |
| 15:54.00 | [TK]D-Fender | pmhaddad-lappy: You are missing it. the name is not the SAME |
| 15:54.06 | rue_mohr | nobody checked to make sure the new milliwatt signal code generated the proper amplitude |
| 15:54.21 | rue_mohr | I beleive its actaully been wrong for years now.. |
| 15:54.24 | pmhaddad-lappy | ... but they are different boxes... |
| 15:54.53 | drmessano | This is pointless |
| 15:54.58 | pmhaddad-lappy | my register string shouldn't look like LakeLinden_REMC:REM906!@172.16.20.11/LakeLinden_REMC |
| 15:55.04 | [TK]D-Fender | phix: You can't have 1 peer saying "I''m FRED" and the other side having a peer saying "FRED1" |
| 15:55.08 | [TK]D-Fender | pmhaddad-lappy: ^^^ |
| 15:55.16 | rue_mohr | the old code was replaced cause it generated 1Khz, which isn't used becuase you can get an alias with the 8khz sampling |
| 15:55.20 | pmhaddad-lappy | hmmm ok so that register string is wrong then? |
| 15:55.31 | [TK]D-Fender | pmhaddad-lappy: Register means NOTHING. it does not auth calls |
| 15:55.31 | rue_mohr | drmessano, getting tired of this sircle? |
| 15:55.45 | pmhaddad-lappy | basically do i need to change both contexts to be the same? |
| 15:55.46 | [TK]D-Fender | pmhaddad-lappy: Your PEER entries auth calls |
| 15:56.01 | pmhaddad-lappy | like they should both be LakeLinden_REMC |
| 15:56.05 | rue_mohr | drmessano, my line is fine, your seeing the output of a T1 channelbank with its gains all set to 0db |
| 15:56.13 | [TK]D-Fender | pmhaddad-lappy: If I say I'm FRED, then there better be a FRED entry on the other side to match |
| 15:56.17 | pmhaddad-lappy | hm ok |
| 15:56.40 | pmhaddad-lappy | i'm just confused then... do i need a seperate LakeLinden context too? |
| 15:56.50 | drmessano | 0db, AKA -10.17dbm? |
| 15:56.51 | pmhaddad-lappy | i'm pretty much copying this out of the book here |
| 15:56.59 | rue_mohr | no |
| 15:57.12 | drmessano | Thats what your meter reads |
| 15:57.15 | rue_mohr | the milliwatt() generates a signal that is -10.17 dbm |
| 15:57.33 | rue_mohr | if I use milliwatt(o) the meter shows about 0dbm |
| 15:57.57 | ruben23 | hi, having errors like this, on my asterisk cli--->http://pastebin.com/m189be3f1 |
| 15:59.00 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:59.02 | rue_mohr | [TK]D-Fender, I'm sorry dude, I'm having a problem here with the func_volume you mentioned, I cant find it anywhere on the system, is it a seperate patch? |
| 15:59.25 | pmhaddad-lappy | [TK]D-Fender, now i get this: Registration from '<sip:LakeLinden@172.16.20.111>' failed for '10.71.20.26' - No matching peer found |
| 15:59.34 | pmhaddad-lappy | after changing both contexts to be the same... |
| 15:59.48 | [TK]D-Fender | [11:40]<[TK]D-Fender>rue_mohr: func_volueme = 1.6+ <------------ |
| 15:59.49 | *** join/#asterisk many (n=many@dslb-188-098-012-145.pools.arcor-ip.net) |
| 16:00.06 | rue_mohr | that IS a typo, right!? |
| 16:00.13 | [TK]D-Fender | rue_mohr: duh :p |
| 16:00.19 | [TK]D-Fender | rue_mohr: I make plenty |
| 16:00.22 | *** part/#asterisk mattbUK (n=mattbUK@77-98-137-62.cable.ubr07.stav.blueyonder.co.uk) |
| 16:00.24 | ariel_ | pmhaddad-lappy: keep it simple, call one system1 2nd system2 no registration needed between |
| 16:00.37 | pmhaddad-lappy | huh? |
| 16:00.41 | rue_mohr | good, cause I been grepping and googling for func_volume |
| 16:00.41 | pmhaddad-lappy | is now totally lost |
| 16:00.44 | pmhaddad-lappy | :( |
| 16:01.15 | pmhaddad-lappy | can someone show me an example sip.conf with something like this working? |
| 16:01.27 | pmhaddad-lappy | because i'm now throughly confused |
| 16:01.54 | [TK]D-Fender | Lunch time... |
| 16:02.27 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 16:02.57 | ruben23 | hi, having errors like this, on my asterisk cli--->http://pastebin.com/m189be3f1 |
| 16:03.11 | rue_mohr | k, I'm gonna assume its a line for the gloabl section of extensions.conf |
| 16:03.17 | rue_mohr | pmhaddad-lappy, just a sec |
| 16:03.44 | drmessano | rue_mohr: WTF.. I googled and first hit got me this --> http://www.voip-info.org/wiki/view/Asterisk+func+volume |
| 16:03.52 | pmhaddad-lappy | rue_mohr, ok |
| 16:04.13 | ariel_ | pmhaddad-lappy: http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
| 16:04.27 | ariel_ | keep it simple for testing and working your way through the dial plan |
| 16:04.57 | rue_mohr | aha, new in 1.6 |
| 16:05.09 | drmessano | [TK]D-Fender SAID it was |
| 16:05.14 | rue_mohr | OH thats what he meant |
| 16:05.17 | drmessano | [12:00] <[TK]D-Fender> [11:40]<[TK]D-Fender>rue_mohr: func_volueme = 1.6+ <------------ |
| 16:05.26 | rue_mohr | looks like a config line |
| 16:05.37 | rue_mohr | I was gonna ask why it disn't specify rx or tx |
| 16:05.45 | drmessano | it does |
| 16:05.56 | ariel_ | ruben23: not enough info there, but it's just saying it's busy. |
| 16:05.57 | pmhaddad-lappy | ariel_, don't want to use IAX... |
| 16:06.10 | rue_mohr | no, there isn't an rx and tx release of asterisk 1.6 |
| 16:06.19 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 16:06.21 | drmessano | What? |
| 16:06.47 | rue_mohr | I thought tk was giving me a config line, not a applicable asterisk version |
| 16:06.48 | ariel_ | pmhaddad-lappy: works same with sip |
| 16:07.20 | ariel_ | pmhaddad-lappy: you can even send calls to the box just with it's IP and setting up proper rules in the default context |
| 16:07.31 | ruben23 | ariel_: whats the internal server error..? |
| 16:08.00 | ariel_ | ruben23: you did not post enough info on what is going on, use set verbose 99 and get more info |
| 16:08.04 | pmhaddad-lappy | ariel_, i dont see how this is really differnet from what i'm doing :( |
| 16:08.16 | ariel_ | keep it simple |
| 16:08.32 | rue_mohr | pmhaddad-lappy, http://www.pastebin.ca/1503349 |
| 16:08.39 | ariel_ | dial(sip/IP/extension,20) |
| 16:08.54 | rue_mohr | so its been backported but I'v have to patch |
| 16:09.10 | timeshell_atwork | [TK]D-Fender : Ok. I have a question for you about configuring the Polycom phones for call parking with asterisk. I want to have one of the buttons configured such that when someone is on a call and presses it, it transfers it to the main parking lot extension whereby asterisk will then return the assigned parking lot number. |
| 16:09.45 | timeshell_atwork | Where I see in the SIP info that call parking can be enabled on the polycom phones, I don't see where to configure it to work this way. |
| 16:09.47 | rue_mohr | so the only way I can dial the volume up for the receptionist is to increase the volume on the polycom over the 12db I already have it set to |
| 16:10.12 | ariel_ | bbl I need to run errands during lunch. |
| 16:10.43 | pmhaddad-lappy | rue_mohr, ya that just confused me even more :( |
| 16:11.18 | rue_mohr | thats a working sip.conf file |
| 16:13.07 | pmhaddad-lappy | for what? |
| 16:13.13 | pmhaddad-lappy | it makes no sense to me lol |
| 16:14.42 | rue_mohr | for my office |
| 16:14.49 | rue_mohr | what do you need to know |
| 16:18.08 | pmhaddad-lappy | ok. how do i connect a remote pbx to another pbx with sip? i dont want each ext from the remote box authenticating. the remote pbx should auth with the local one |
| 16:18.13 | pmhaddad-lappy | without having to auth each ext |
| 16:18.19 | pmhaddad-lappy | your sip.conf made no sense to me |
| 16:19.14 | pmhaddad-lappy | it just shows a sip user |
| 16:20.12 | rue_mohr | go read up on IAX |
| 16:20.34 | rue_mohr | its forf connecting phone systems |
| 16:20.58 | pmhaddad-lappy | i thought you could do this with sip |
| 16:21.09 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:21.24 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:22.45 | [TK]D-Fender | timeshell_atwork: Not supported. |
| 16:23.27 | [TK]D-Fender | pmhaddad-lappy: You can. |
| 16:23.33 | [TK]D-Fender | No need for IAX. |
| 16:23.36 | *** part/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 16:23.58 | pmhaddad-lappy | [TK]D-Fender, ok... then can you please showing me an example of how to do it? |
| 16:24.50 | [TK]D-Fender | pmhaddad-lappy: only 1 side should register, if at all. |
| 16:25.00 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:25.05 | [TK]D-Fender | pmhaddad-lappy: fix this and PB what you've got |
| 16:25.20 | pmhaddad-lappy | [TK]D-Fender, so how am i supposed to send traffic back and forth? |
| 16:25.33 | pmhaddad-lappy | maybe i'm not being clear about how this needs to work... |
| 16:25.46 | [TK]D-Fender | pmhaddad-lappy: .... Does * have to register to a SIP PHONE for it to work? NO. |
| 16:26.22 | pmhaddad-lappy | what does that have to do with anything? |
| 16:26.33 | pmhaddad-lappy | you are only succeeding in confusing the hell out of me |
| 16:26.55 | [TK]D-Fender | pmhaddad-lappy: SIP is SIP. Doesn't matter that it's between 2 PBX's. |
| 16:27.10 | [TK]D-Fender | pmhaddad-lappy: ITSP's don't have to register to you... you register to THEM. |
| 16:27.14 | [TK]D-Fender | pmhaddad-lappy: ONE SIDE. |
| 16:27.23 | [TK]D-Fender | pmhaddad-lappy: Capice? |
| 16:28.10 | [TK]D-Fender | pmhaddad-lappy: And registration is not even required. |
| 16:28.20 | pmhaddad-lappy | [TK]D-Fender, so should I delete the register statement from one of the boxes? |
| 16:28.36 | [TK]D-Fender | pmhaddad-lappy: Thats what I just told you |
| 16:29.01 | [TK]D-Fender | pmhaddad-lappy: And the side that registers should have the HOST filled into their peer entry |
| 16:29.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:32.14 | pmhaddad-lappy | [TK]D-Fender, ok i removed the register string from one of the boxes and im still getting the username mismatch |
| 16:32.40 | [TK]D-Fender | [12:25]<[TK]D-Fender>pmhaddad-lappy: fix this and PB what you've got |
| 16:33.35 | pmhaddad-lappy | [TK]D-Fender, all i did was comment out the register => string |
| 16:33.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:34.34 | pmhaddad-lappy | [TK]D-Fender, the boxes are registered if i do a sip show registry |
| 16:35.09 | pmhaddad-lappy | [TK]D-Fender, i get the mismatch when i try to place a call across the two boxes |
| 16:35.31 | [TK]D-Fender | pmhaddad-lappy: because the peer names don't match |
| 16:35.55 | pmhaddad-lappy | [TK]D-Fender, because its trying to use the phones user id instead of the boxes registration |
| 16:36.09 | [TK]D-Fender | pmhaddad-lappy: No, the peers don't match ANYWAYS. |
| 16:37.08 | pmhaddad-lappy | [TK]D-Fender, the peers do match with there respective sides thats how the boxes are registered |
| 16:37.43 | [TK]D-Fender | pmhaddad-lappy: Ok. I'm going to say this again. Please try to follow. Registration has NOTHING to do with authing calls. |
| 16:37.46 | [TK]D-Fender | NOTHING |
| 16:38.32 | pmhaddad-lappy | [TK]D-Fender, ok fine but i dont want to have all of the phones registrations on both boxes |
| 16:38.52 | [TK]D-Fender | pmhaddad-lappy: And you don't have to. |
| 16:39.14 | pmhaddad-lappy | [TK]D-Fender, okay im sry im not getting this im really trying to understand this over here |
| 16:39.44 | pmhaddad-lappy | [TK]D-Fender, so do you want me to remove both of the register strings from both boxes? |
| 16:39.54 | [TK]D-Fender | pmhaddad-lappy: You just brought up a point that isn't required and are only confusing yourself and are not paying attention |
| 16:40.09 | [TK]D-Fender | pmhaddad-lappy: I said ONE. |
| 16:40.09 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 16:40.19 | drmessano | You set a peer up and all youve done is connect the boxes.. youve not set anything up for calls to pass from box to box.. Get the peers correct FIRST |
| 16:40.29 | [TK]D-Fender | pmhaddad-lappy: You seem to have far too much difficulty with simple instructions. |
| 16:40.41 | drmessano | Not rocket science |
| 16:41.14 | [TK]D-Fender | drmessano: Raw-Cat Sigh Hence |
| 16:41.14 | pmhaddad-lappy | [TK]D-Fender, ok but i removed the register string from one box like you said and i still ahve the same problem |
| 16:41.32 | [TK]D-Fender | pmhaddad-lappy: I've said it. TWICE even... |
| 16:41.42 | [TK]D-Fender | ~wmmfpb |
| 16:41.43 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 16:42.12 | jaytee | <PROTECTED> |
| 16:42.41 | jaytee | hands [TK]D-Fender some Xanax in the giant economy size bottle |
| 16:43.31 | [TK]D-Fender | jaytee: No... Richard Dreyfuss said it best "Its a new technique Bob, its called 'Death therapy'!" |
| 16:43.41 | jaytee | hahahaha |
| 16:44.44 | ariel_ | pmhaddad-lappy: you have 2 boxes you know where there at via IP address, so you don't need to register them to talk with each other. That's the first step. |
| 16:45.38 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 16:45.54 | jasonwoot | If I run safe_asterisk manually from command line, will it restart asterisk? |
| 16:46.16 | [TK]D-Fender | jasonwoot: When? Why? |
| 16:46.43 | [TK]D-Fender | jasonwoot: What state is the server is before you do this? How exactly are you calling it? |
| 16:46.53 | [TK]D-Fender | jasonwoot: Who killed J.R>? |
| 16:47.05 | leifmadsen | jasonwoot: hey! just the man I was looking for :) |
| 16:48.07 | pmhaddad-lappy | [TK]D-Fender, http://pastebin.com/d11a792ed |
| 16:48.52 | [TK]D-Fender | pmhaddad-lappy: I still see 2 registers and you did not follow my instructs to set the host. |
| 16:49.29 | [TK]D-Fender | is probably just wasting his time.... |
| 16:49.34 | kaii | [TK]D-Fender: if you look deeper at character 1 of line 5, you will only see one register. |
| 16:49.40 | pmhaddad-lappy | [TK]D-Fender, i commented out one of the register statement what do you mean host |
| 16:50.09 | pmhaddad-lappy | [TK]D-Fender, do you want me to set the host= to the ip address |
| 16:50.34 | kaii | on the side which registers, yes |
| 16:50.45 | ariel_ | don't need to register |
| 16:50.54 | [TK]D-Fender | [12:29]<[TK]D-Fender>pmhaddad-lappy: And the side that registers should have the HOST filled into their peer entry |
| 16:50.57 | pmhaddad-lappy | [TK]D-Fender, did you want me to remove one of the [LakeLinden] or [LakeLinden_REMC] |
| 16:51.02 | carrar | No one. The shots just wounded him (It was the Kristin Shepard character) |
| 16:51.02 | carrar | Answer originally posted in response to Who killed JR on Dallas? |
| 16:51.04 | ariel_ | host=pbx1 on pbx2 and host=pbx2 on pbx1 |
| 16:51.08 | carrar | haha |
| 16:51.13 | carrar | damns paste |
| 16:51.14 | *** join/#asterisk hfb (n=hfb@pool-96-251-62-168.lsanca.dsl-w.verizon.net) |
| 16:51.57 | kaii | ariel_: but need to know the ip requests should be send to |
| 16:51.58 | kaii | pmhaddad-lappy: set "host=172.16.20.111" in "[LakeLinden_REMC]" .. this is what d-fender wanted to tell you. |
| 16:52.32 | ariel_ | in his post he has both boxes IP address |
| 16:52.44 | kaii | ariel_: very safe if someone else gains control of your IP |
| 16:53.00 | kaii | registering with credentials is safer and should be done |
| 16:53.06 | ariel_ | yes but lets get him started basic |
| 16:53.13 | ariel_ | work security from there |
| 16:53.33 | kaii | if he does what d-fender and i say, it should work too |
| 16:54.02 | kaii | anyway.. im gonna knock off |
| 16:54.08 | ariel_ | fine but registration is only needed if the other end does not know your IP |
| 16:54.55 | encbladexp | cya |
| 16:54.57 | *** part/#asterisk encbladexp (n=stefan@p5495F8F1.dip.t-dialin.net) |
| 16:54.58 | pmhaddad-lappy | [TK]D-Fender, okay i have set my host like you wanted |
| 16:55.12 | [TK]D-Fender | pmhaddad-lappy: http://pastebin.com/m2103fbb7 |
| 16:59.11 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 16:59.17 | jasonwoot | Fender, everyone has to go tinkle sometime... I just choose to post a very important question in freenode first |
| 16:59.22 | jasonwoot | it's like an OCD thing |
| 17:01.08 | jasonwoot | I stopped asterisk but didn't restart it from the init script, so I'd like to start safe_asterisk without restarting asterisk and dropping my calls |
| 17:01.29 | [TK]D-Fender | jasonwoot: You need to develop ADD to keep the OCD in check |
| 17:02.06 | [TK]D-Fender | jasonwoot: Well if it didn't restart, what would safe_asterisk be restarting? You've just alluded that it isn't even running. |
| 17:03.42 | pmhaddad-lappy | [TK]D-Fender, ok i copied what you had im not longer getting user mismatch im now getting failed to authenticate user |
| 17:04.30 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 17:04.32 | [TK]D-Fender | pmhaddad-lappy: And I don't see your failed call with SIP debug |
| 17:04.47 | [TK]D-Fender | pmhaddad-lappy: ALL of it. |
| 17:05.04 | jasonwoot | [TK]D-Fender: asterisk was started manually. If I run safe_asterisk will it stop/restart asterisk? |
| 17:05.16 | ariel_ | jasonwoot: no |
| 17:05.19 | [TK]D-Fender | jasonwoot: No, it will simply fail |
| 17:05.25 | pmhaddad-lappy | [TK]D-Fender, do you want it just for the peer Lakelinden |
| 17:05.33 | [TK]D-Fender | pmhaddad-lappy: calling side only is fine |
| 17:05.42 | [TK]D-Fender | for now) |
| 17:07.26 | pmhaddad-lappy | [TK]D-Fender, http://pastebin.com/d582c2c83 |
| 17:08.43 | Alfio | hi, i wanna know if i can hang up all my calls in a queue at 5:00 pm for example |
| 17:09.50 | [TK]D-Fender | Alfio: You can do it whenever and why ever you want |
| 17:10.24 | Alfio | [TK]D-Fender what option should i use ? |
| 17:10.46 | [TK]D-Fender | Alfio: "soft hangup [channel]" |
| 17:10.59 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 17:12.47 | [TK]D-Fender | pmhaddad-lappy: Sync the PW's, I missed fixing that |
| 17:16.11 | *** join/#asterisk vicscandl (n=Christop@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 17:16.41 | *** join/#asterisk [netman] (n=netman@136.Red-88-27-57.staticIP.rima-tde.net) |
| 17:17.58 | pmhaddad-lappy | [TK]D-Fender, OMG it works thanks a bunch I really appreciate your help |
| 17:18.49 | [TK]D-Fender | pmhaddad-lappy: Good. Next time please pay attention when people ask your for things when trying to help you. |
| 17:19.34 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:19.35 | pmhaddad-lappy | [TK]D-Fender, i will do my best sometimes you guys arent the clearest on what your asking for |
| 17:20.05 | [TK]D-Fender | pmhaddad-lappy: I was blatantly clear on mine. Repetitively. |
| 17:20.42 | [TK]D-Fender | pmhaddad-lappy: Either way calls should flow smooth from one to the other including whatever ext # and CID either side wants to present |
| 17:20.48 | pmhaddad-lappy | [TK]D-Fender, ok well might have been clear to you but I wasnt getting it. But I really appreciate you sticking with me on this |
| 17:21.03 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:23.53 | xp_prg | jblack you here, I have a question for you |
| 17:24.16 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
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| 17:29.45 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 17:29.50 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 17:30.32 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 17:31.14 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:37.37 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 17:42.40 | vicscandl | greetings asterisk folks; my boss wants to setup a calling-card type system using asterisk. Any thoughts on the available plug-ins for this? |
| 17:44.14 | *** join/#asterisk Mehh (n=Tweak@ip241-242-174-82.adsl2.static.versatel.nl) |
| 17:44.17 | Mehh | hi |
| 17:47.50 | [TK]D-Fender | vicscandl: Go lookup http://www.asterisk2billing.org/cgi-bin/trac.cgi |
| 17:49.33 | Mehh | i hope someone can help me, im really a n00b with this but i managed to get SIP working behind NAT etc... now i can call to others with 3CX... but now i want to able to use an anlogue line to call "outside" is this possible with the phone.conf? |
| 17:50.18 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:50.31 | [TK]D-Fender | Mehh: phone.conf is deprecated junk. Go geta supported FXO interface. |
| 17:51.00 | Mehh | i dont want to buy some expensive stuff... im just doing this for testing etc |
| 17:51.19 | *** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 17:51.51 | leifmadsen | if you want to call out on an analog line, you need some decent hardware, like an FXO-to-SIP converter from linksys or something |
| 17:51.59 | sumasuma | How can I log asterisk events to flat file without connecting to AMI |
| 17:52.04 | leifmadsen | Mehh: if you really have no money, get a cheap pre-paid SIP ITSP |
| 17:52.13 | sumasuma | Is there is anything i can enable in asterisk to store it to a specific location ? |
| 17:52.15 | leifmadsen | sumasuma: logger.conf |
| 17:52.23 | sumasuma | leifmadsen: thanks. |
| 17:52.44 | vicscandl | [TK]D-Fender: much <3 man; researching now. |
| 17:53.30 | Mehh | but i connected an old modem on /dev/ttyS0 and it looks like asterisk did something with it |
| 17:53.47 | Mehh | because more leds came on when i placed the device in phone.conf |
| 17:54.24 | leifmadsen | Mehh: that will not work -- just stop bothering. Everyone comes in here thinking they can get it to work, and it won't. |
| 17:54.42 | Mehh | damn |
| 17:54.52 | leifmadsen | that is a very old module, and is not maintained |
| 17:55.16 | leifmadsen | just get a cheap ITSP and be done with it |
| 17:55.22 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 17:56.00 | coppice | ~modem |
| 17:56.01 | infobot | i heard modem is (Modulator/Demodulator) A device to turn digital signals to analog ones and back again, so they can be transmitted and translated back to digital at another modem without loss. Used for communication through means of audio, telephone, CB, etc. Random disconnects? S10=255 sure to do the trick! |
| 17:56.47 | Mehh | hmm what about pci ISDN cards? does those work? |
| 17:57.27 | WindowsUser | i think some of them do |
| 17:58.29 | Mehh | what i just wanted... was to be able to call with my mobile phone (with wifi) to phone numbers using the voip server on my work |
| 17:58.41 | coppice | Mehh: http://www.soft-switch.org/cards.html#modems Several cheap ISDN cards work with *. A modem won't |
| 18:01.49 | [TK]D-Fender | Mehh: "just for testing"? O RLY. What will this "testing" prove? |
| 18:33.11 | *** join/#asterisk jamicque (i=jamicque@jam.bema.one.pl) |
| 18:33.39 | *** join/#asterisk ArchGT (n=ArchGT@190.149.127.201) |
| 18:33.54 | Alfio | [TK]D-Fender how i can use the "soft hangup" in my dialplan |
| 18:33.55 | Alfio | ? |
| 18:34.46 | jamicque | Hi can anyone help me, I'm having problems with asterisk 1.6.x t38 pass-thru (namly faxes). I've reacently swiche'd to 1.6.x from 1.4. On 1.4 the problems dosen't exist. |
| 18:35.17 | [TK]D-Fender | Alfio: System() |
| 18:36.39 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:37.41 | Alfio | [TK]D-Fender i can use it with an schedule to run at 5pm for example with the proper sentences right? |
| 18:37.53 | jamicque | t38 is negotiated, however it can't complete the training. The situation occur on Linksys VoIP gateways as on Commetrex Server. There must be something wrong with my settings (any help would be aprriciated. |
| 18:38.20 | [TK]D-Fender | Alfio: What would dialplan have to do with executing things on a schedule? |
| 18:38.48 | Alfio | i want to hang up all the calls in one queue at 5:00pm |
| 18:39.06 | [TK]D-Fender | Alfio: This command isn;t normally inteded to be called via dialplan anyway, |
| 18:39.22 | [TK]D-Fender | Alfio: So maybe you should think about what is going to handle your scheduling. |
| 18:39.43 | Alfio | yes i mean i can use a cron |
| 18:40.17 | [TK]D-Fender | Alfio: Then by all means... |
| 18:40.40 | Alfio | ok |
| 18:41.12 | Alfio | [TK]D-Fender thx i will try some test |
| 18:49.40 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 18:53.50 | jamicque | anyone has any expiriance with t38 on asterisk 1.6? |
| 19:01.02 | *** join/#asterisk sjobeck (n=Adium@97-120-50-240.ptld.qwest.net) |
| 19:02.36 | *** part/#asterisk sjobeck (n=Adium@97-120-50-240.ptld.qwest.net) |
| 19:02.48 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:03.51 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 19:05.33 | howie | How do i check what version im running in asterisk? |
| 19:06.28 | jamicque | howie: show version |
| 19:08.48 | *** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 19:12.34 | leifmadsen | howie: core show version |
| 19:12.45 | leifmadsen | (show version works in 1.2, and is deprecated, but works in 1.4) |
| 19:14.17 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:16.43 | jamicque | I'm having problems with asterisk 1.6.x t38 pass-thru (t38 negotiated by re-invite). I've recently switched to 1.6.x from 1.4. On 1.4 the problems doesnÂ’t exist. T38 is negotiated, however it can't complete the training. any help would be appreciated... |
| 19:19.09 | *** join/#asterisk tfrew|afk (n=tfrew@office.neteasyinc.com) |
| 19:22.04 | *** join/#asterisk JohnTeddy (i=unstable@tor/regular/sid) |
| 19:22.17 | guax | incominglimit and call-limit in sip.conf are the same thing in 1.4? |
| 19:23.08 | JohnTeddy | I have broadvoice, I use the analog<>digital box, and I have some crappy analog phone. I only have one phone right now. This isn't an asterisk question, but this channel is probably the best place to ask. At some point there will be 4 more people in my office, and not just me so I will have asterisk, but not yet. What is a good SIP phone base and headset combo to buy, that will eventually work well with asterisk/trixbox one day? |
| 19:23.39 | Qwell | ~phones |
| 19:23.40 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 19:23.45 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 19:25.05 | Qwell | JohnTeddy: as for headsets, the only real contender afaik is Plantronics |
| 19:25.05 | JohnTeddy | I think I almost bought an Aastra 480i last year when I had vonage, but I figured out vonage does mac address restricting, so they support SIP, but only on phone manufacturers that pay them money. |
| 19:25.15 | JohnTeddy | Hence I got rid of vonage, and bought a broadvoice account. |
| 19:25.38 | [TK]D-Fender | JohnTeddy: Polycom > All |
| 19:26.10 | JohnTeddy | I need to do a physical conference, so a nice speaker phone, but I also am mobile in the office, so I need a headset. |
| 19:26.16 | JohnTeddy | Which polycom is good for that? |
| 19:26.37 | *** join/#asterisk mattbUK (n=mattbUK@77-98-137-62.cable.ubr07.stav.blueyonder.co.uk) |
| 19:26.37 | Qwell | any, heh |
| 19:26.50 | Qwell | I think all the current models have speakerphone, don't they? |
| 19:27.00 | Qwell | maybe not the 3xx |
| 19:28.04 | JohnTeddy | http://salestores.com/polyco25.html ? |
| 19:28.25 | Qwell | salestores...wow. |
| 19:28.29 | Qwell | generic much? |
| 19:29.02 | JohnTeddy | I have no idea which one to get, hehe. That's why I came in here. |
| 19:29.08 | JohnTeddy | link me to a good one, I'll likely buy that one. |
| 19:29.09 | Qwell | the 301 is...err...old |
| 19:29.18 | Qwell | 320/330 |
| 19:29.33 | Qwell | (or 321/331 - I seem to recall those existing now) |
| 19:29.40 | kmem | can i build just the agi portion of 1.6 from source if i installed 1.6 via a package? |
| 19:30.17 | Qwell | JohnTeddy: http://www.polycom.com/products/voice/desktop_solutions/soundpoint/index.html |
| 19:30.23 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 19:30.24 | mattbUK | any one able to offer some suggestions to help me figure out why my cisco 7960 won't accept inbound calls from my externally hosted asterisk box? Cisco is behind not RTP ports and SIP port open. Outbound calls are fine, inbound doesn't ring and just shows congested: http://pastebin.com/d26fa7fec |
| 19:30.45 | Katty | has oreos and milk! |
| 19:30.45 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 19:31.07 | Qwell | [TK]D-Fender: You seen the SE-22x? O.o |
| 19:31.39 | Katty | JohnTeddy: i'd consider the 330s to be a standard user phone. |
| 19:31.57 | Katty | JohnTeddy: the 501 is bigger, has more buttons. bigger display. |
| 19:32.45 | Katty | JohnTeddy: the 320 and 330 is basically the same, except the 330 has another network jack if you have two network devices and only one drop. |
| 19:32.48 | JohnTeddy | There are no prices listed. That is always bad. |
| 19:32.58 | Katty | JohnTeddy: voipsupply.com |
| 19:32.59 | Qwell | JohnTeddy: They don't sell to end-users directly |
| 19:33.04 | JohnTeddy | I see, ok. |
| 19:33.23 | Katty | the 330s were around 100ish last time i checked |
| 19:33.35 | kmem | can i build just the agi portion of 1.6 from source if i installed 1.6 via a package? |
| 19:33.49 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 19:34.33 | Katty | JohnTeddy: if you're going to be having a receptionist transfering around calls, i'd go with the 501 |
| 19:34.52 | [TK]D-Fender | Qwell: Analog? GAH :p |
| 19:34.52 | Katty | JohnTeddy: the 320/330 display is so small you won't be able to see more than 1 call at a glance without scrolling down. |
| 19:35.04 | Qwell | oh it's analog? |
| 19:35.44 | Qwell | that looks too fancy to be analog. |
| 19:35.45 | Katty | JohnTeddy: we use bluetooth headsets around here by platronics. |
| 19:35.46 | [TK]D-Fender | Katty: I'f Qwell Yup |
| 19:35.48 | [TK]D-Fender | askhgfa |
| 19:35.52 | [TK]D-Fender | asplodes |
| 19:35.56 | Katty | comforts [TK]D-Fender |
| 19:36.00 | JohnTeddy | http://www.voipsupply.com/polycom-ip-330 |
| 19:36.01 | JohnTeddy | ok, I will buy this base. |
| 19:36.08 | [TK]D-Fender | Katty: IP 501 = disco |
| 19:36.22 | Katty | 550 now? |
| 19:36.26 | JohnTeddy | And the headset shouldn't aslo be from polycom, it should be from Plantronics? |
| 19:36.32 | [TK]D-Fender | Katty: 450 more likely. |
| 19:36.35 | Qwell | Katty: the whole 5xx series is hard to justify |
| 19:36.37 | Katty | ah |
| 19:36.45 | Katty | JohnTeddy: dont' forget the ac adaptor. |
| 19:36.49 | [TK]D-Fender | JohnTeddy: Back that train up and proper describe your working environment |
| 19:36.53 | [TK]D-Fender | +ly |
| 19:37.01 | Katty | Qwell: we have all 501s here. |
| 19:37.03 | [TK]D-Fender | Qwell: Indeed |
| 19:37.09 | [TK]D-Fender | Katty: Decrepit! |
| 19:37.11 | Katty | well some are 500 |
| 19:37.18 | Katty | from eleventybillion years ago |
| 19:37.20 | [TK]D-Fender | Katty: WORSE! |
| 19:37.25 | Katty | *hee* |
| 19:37.31 | Katty | does the 450 look like the 501? |
| 19:37.39 | JohnTeddy | [TK]D-Fender: It's me in an office by myself. I have broadvoice. I'd like to get a phone that works with broadvoice(SIP?) and a head set. |
| 19:38.00 | JohnTeddy | Katty: wtf, they don't give me the ac adapter when I buy the phone? |
| 19:38.33 | [TK]D-Fender | JohnTeddy: What kind of volume? For you. for them? |
| 19:38.36 | Katty | JohnTeddy: it's listed on the page. |
| 19:39.38 | JohnTeddy | weird |
| 19:39.45 | Katty | we should just buy all 650s with color lcd |
| 19:40.06 | Katty | JohnTeddy: if you get more than one call at a time, i'd steer clear of the 300 series. |
| 19:40.12 | Katty | JohnTeddy: you'll just be annoyed. |
| 19:41.47 | *** join/#asterisk kuku1 (n=ingo@c-67-175-3-155.hsd1.il.comcast.net) |
| 19:41.48 | Katty | woah |
| 19:41.50 | Katty | wirelsss phones |
| 19:41.52 | Katty | hotthottthotttt |
| 19:42.08 | kuku1 | On 1.6, with monitor, I get stuck with two wav files after the conversation, they do not get merged |
| 19:42.26 | Katty | izzocute! |
| 19:44.28 | WindowsUser | kuku1: monitor(something.wav,m) or monitor(something.wav)? |
| 19:44.56 | kuku1 | ,m |
| 19:45.08 | WindowsUser | is sox installed? |
| 19:45.11 | kuku1 | I didnt have sox installed :) |
| 19:45.15 | WindowsUser | ah |
| 19:45.17 | kuku1 | Thank you ! |
| 19:45.31 | kuku1 | Now, I just need to join the other recorded converstations into one file somehow |
| 19:45.38 | *** join/#asterisk G_ERWIN (n=gk@j41048.upc-j.chello.nl) |
| 19:45.43 | Katty | mani need one of these now |
| 19:46.04 | WindowsUser | sox -m a.wav b.wav out.wav |
| 19:46.11 | *** join/#asterisk stix (n=stix@212.27.20.29.bredband.3.dk) |
| 19:46.31 | stix | Is it possible to put a call on hold via Asterisk manager? |
| 19:46.57 | Katty | http://42ndgeekstreet.blogspot.com/2008/12/converting-wav-to-gsm-for-asterisk.html |
| 19:47.45 | kuku1 | sox: invalid option -- m |
| 19:48.01 | Katty | kuku1: see blogspot post. |
| 19:48.11 | Katty | kuku1: and use mixmonitor for that |
| 19:48.30 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 19:49.09 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:50.49 | jamicque | Anyone run t38 passthru on asterisk 1.6.x ? |
| 19:51.20 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-ab939fe9bd4940e3) |
| 19:54.36 | mmattice | anybody tried setting up 1.6 on debian lenny with dahdi? |
| 19:55.40 | *** join/#asterisk De_Mon (i=de_mon@fl-76-4-141-167.dhcp.embarqhsd.net) |
| 19:55.58 | mmattice | I'm getting odd errors, like dahdi-source is out of sync with dahdi-linux, but they're both 1:2.2.0~dfsg~rc5 |
| 19:56.22 | xp_prg | hi all, I need to test bandwidth on a server, anyone know a good way to do that with linux on the command line? |
| 19:57.42 | afink | ok I have to ask. Anybody want rickroll.gsm? |
| 19:57.48 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 19:58.01 | Xetrov` | haha |
| 19:58.19 | [TK]D-Fender | afink: Sounds like it comes with copyrightInfringement.so |
| 19:58.21 | Xetrov` | that just gave me an awesome idea |
| 19:58.45 | afink | [TK]D-Fender: didn't think about that... |
| 19:59.05 | afink | [TK]D-Fender: it is for testing and educational purposes only |
| 20:00.02 | Alfio | afink tell that to the FBI |
| 20:00.14 | [TK]D-Fender | afink: So its legal if I shoot you in the hed jsut to learn how fast your brain will empty onto the floor for "educational purposes'? |
| 20:00.31 | afink | sure |
| 20:06.14 | jamicque | Anyone uses t38 on asterisk 1.6 ?? :) |
| 20:07.45 | *** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 20:09.04 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:10.28 | *** join/#asterisk FreakGuard (n=freak@unaffiliated/freakguard) |
| 20:13.33 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-230-234.telecom.net.ar) |
| 20:13.41 | [TK]D-Fender | jamicque: No more that when you asked 15 minutes ago, or a few before that... |
| 20:16.08 | jamicque | ok I'll wait patiently... |
| 20:16.55 | spackle | D-fender waits vigilantly, I was his victim yesterday |
| 20:19.44 | *** join/#asterisk ingenius (n=alektro@190.247.156.247) |
| 20:19.55 | spackle | xp_prg: what kind of bandwidth test, throughput? |
| 20:20.03 | Katty | fender's just bored easily. |
| 20:20.12 | spackle | hi Katty! |
| 20:20.18 | Katty | hi spackle (= |
| 20:20.18 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 20:20.22 | kuku1 | Katty: I don't have soxmixer |
| 20:20.58 | spackle | awkwardly hugs Katty |
| 20:21.00 | putnopvut | I cannot seem to find in the Polycom Administrator's guide for an IP 430 how to change the Expires header for presence-related SUBSCRIBE requests. Does anyone know if there is a way to change it from the default value of 3600 seconds? And if so, could you please share such info? Thanks. |
| 20:22.01 | Katty | hugs on spackle |
| 20:22.08 | Katty | kuku1: whatabout a sock sorter |
| 20:22.29 | Katty | putnopvut: personally, i'd just call polycom for that one. |
| 20:22.41 | Katty | putnopvut: put in a ticket requesting inflimation |
| 20:22.48 | kuku1 | Katty: nvm , I have soxmix :) |
| 20:22.52 | spackle | putnopvut: what about one of their other phone manuals? |
| 20:22.55 | putnopvut | Katty: thanks. Never hurts to ask in here first, though :) |
| 20:23.17 | putnopvut | spackle: You mean for something other than a 430? |
| 20:23.25 | putnopvut | spackle: I could check it out, I s'pose. |
| 20:23.39 | spackle | putnopvut: they have oodles of parameters to change on their phones once you are in the XML or the web interface. |
| 20:24.10 | putnopvut | spackle: yeah, I was specifically looking for an xml parameter to place in the <pres/> element. |
| 20:25.12 | spackle | putnopvut: can you change it on the web interface, export the config and check for the diff? |
| 20:25.15 | putnopvut | It kind of sucks when the test I need to run involves having to restart Asterisk every few minutes. I also need to reboot the phone so that it will hurry up and send the SUBSCRIBE. |
| 20:25.31 | putnopvut | spackle: Good idea. I haven't even looked at the web interface. |
| 20:25.37 | spackle | putnopvut: you probably know polycoms take forever to cycle. |
| 20:25.57 | putnopvut | spackle: yep. |
| 20:26.16 | [TK]D-Fender | people ho use Polycom's web interface to configure phones should be dragged out and shot |
| 20:26.22 | [TK]D-Fender | Survivors should be shot AGAIN. |
| 20:26.38 | *** part/#asterisk vicscandl (n=Christop@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 20:26.40 | spackle | putnopvut: it has been a long time since I looked at polycom configs. |
| 20:32.41 | Orbixx | How can I allow a caller to dial an internal extension? |
| 20:34.13 | WindowsUser | http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA |
| 20:34.14 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 20:34.56 | WindowsUser | or maybe background(dialtoooooonnnnnnnne) |
| 20:36.30 | [TK]D-Fender | Orbixx: Put an exten in the context they're dialing in |
| 20:36.42 | [TK]D-Fender | WindowsUser: CRAZY-PERSON |
| 20:37.30 | [TK]D-Fender | Orbixx: And there is no such thing as an "internal extension" |
| 20:37.35 | Orbixx | Ok, an extension. |
| 20:38.07 | howie | how do i get the ztdummy driver? |
| 20:40.06 | [TK]D-Fender | howie: Install Zaptel and modprobe ztdummy |
| 20:40.13 | beek | [TK]D-Fender: Using polycom's web interface is its own punishment. |
| 20:40.18 | xp_prg | spackle in particular transfer speed |
| 20:40.50 | [TK]D-Fender | beek: No, it then spreads to our hearing about it... |
| 20:41.03 | [TK]D-Fender | checkout time, heading home, BBIAB |
| 20:41.52 | spackle | xp_prg: ftp transfer might give you a ballpark idea? |
| 20:42.21 | xp_prg | spackle unfortunately I am trying to verify the bandwidth performance of a server, that is all I know how to tell you what I am doing :( |
| 20:43.45 | N3tw0rk | xp_prg: try jperf |
| 20:44.35 | spackle | xp_prg: on local network or accross a WAN or VPN connection? |
| 20:45.55 | N3tw0rk | xp_prg: will work either its runs on two machines and you point one at the other and it will measure useable bandwitdh |
| 20:46.36 | spackle | N3tw0rk: Thanks for the tip, I'll have a look at that |
| 20:47.50 | N3tw0rk | its is java and requires gui...i m trying to find one that will work all cli |
| 20:49.16 | xp_prg | well, the server exists on the internet, I am trying to figure out its max bandwidth ability |
| 20:49.31 | xp_prg | must I have another side to test that on? |
| 20:52.41 | spackle | xp_prg: xp_prg if you are trying to test the network bandwith, and not the disk bandwidth or anything else, you need to have two sides. |
| 20:53.20 | N3tw0rk | xp_prg: if you wnt to see the avail bandwidth to the inet then use somthing like http://www.speedtest.net or somthing of that nature |
| 20:53.21 | xp_prg | oh ok |
| 20:53.36 | xp_prg | yes that is exactly what I am doing cool N3tw0rk |
| 20:53.36 | xp_prg | ! |
| 20:53.47 | spackle | or you can use wget to see how fast a file can download |
| 20:54.10 | N3tw0rk | i use jperf before i install new voip clients between our colo and the new clients |
| 20:55.10 | xp_prg | I heart you N3tw0rk |
| 20:55.23 | N3tw0rk | xp_prg: lol |
| 20:56.16 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:59.25 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.162.169) |
| 21:00.08 | [TK]D-Fender | \o/ Cell IRC |
| 21:01.20 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:04.00 | Qwell | [TK]D-Fender: oh my... |
| 21:05.09 | [TK]D-Fender | just sitting down for dinner @ indian restaurant |
| 21:05.22 | Qwell | [TK]D-Fender: get the lamb |
| 21:05.57 | [TK]D-Fender | vegitarian place... chaat papri & bhatura |
| 21:06.43 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 21:07.57 | Qwell | pfft |
| 21:09.37 | [TK]D-Fender | oh I'm a full-blooded carnivore, don't get me wrong, but the fodd is awesome and dirt cheap |
| 21:10.03 | ariel_ | wonders what he missed.... |
| 21:10.31 | Qwell | ariel_: he's having salad or something |
| 21:10.54 | ariel_ | oh on a diet |
| 21:10.59 | [TK]D-Fender | salad? hell no... |
| 21:11.21 | Qwell | I dunno - I heard "vegetables" and my eyes glazed over |
| 21:11.22 | [TK]D-Fender | Indian vegitaria != lad |
| 21:11.35 | ariel_ | lol |
| 21:11.50 | *** join/#asterisk Laureano (n=Laureano@190.245.101.140) |
| 21:11.55 | *** part/#asterisk Laureano (n=Laureano@190.245.101.140) |
| 21:12.21 | file | sends [TK]D-Fender to Schwartz's |
| 21:13.53 | beek | Arrgh! Why do my queue members continue to go (paused)? http://www.pastebin.ca/1503635 |
| 21:14.00 | [TK]D-Fender | file: you missed my pi for last wed before you left! |
| 21:14.13 | file | [TK]D-Fender: yeah, I was incredibly busy |
| 21:14.41 | file | had an awesome time though |
| 21:15.31 | spackle | file: greetings |
| 21:15.37 | file | waves to spackle |
| 21:17.08 | [TK]D-Fender | all done,paid&out! |
| 21:17.11 | *** join/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 21:18.17 | Slade- | hey this isnt exactly related, but has anyone here heard of something that will allow a normal phonecall to execute an app on someones cellphone? |
| 21:18.52 | Slade- | delta apparently has a service like this |
| 21:19.01 | stope | any recommendations for a good QOS router for an office setting so the phone aren't choppy? Or is there a better way to do QOS? |
| 21:19.09 | timeshell_atwork | anyone know if mpg123 can play wma streams? |
| 21:19.37 | Corydon76-dig | timeshell_atwork: it definitely cannot |
| 21:19.44 | timeshell_atwork | gah |
| 21:20.01 | timeshell_atwork | anything that can play wma stream as moh in linux? |
| 21:20.22 | Corydon76-dig | Highly doubtful |
| 21:20.50 | Corydon76-dig | You have to realize that streaming is only half of it. It must also be able to resample to 8000Hz |
| 21:21.21 | Corydon76-dig | and while there are streamers for wma on Linux, they typically interface directly to audio hardware and don't resample |
| 21:23.53 | timeshell_atwork | figures |
| 21:24.05 | beek | Guys... what conditions cause Asterisk to pause members of a queue? I have the queues defined as 'autopause=no', I've tried five minute timeouts... nothing seems to stop various queue members ending up as paused. |
| 21:24.33 | Corydon76-dig | beek: have you checked the queue.log ? |
| 21:25.17 | beek | Corydon76-dig: All I see is an entry that states 'PAUSE' |
| 21:25.20 | [TK]D-Fender | Home! |
| 21:26.37 | [TK]D-Fender | brb |
| 21:26.38 | *** part/#asterisk [TK]D-Fender (n=zsirc@161.216.162.169) |
| 21:26.54 | beek | Corydon76-dig: that is to say, I see an entry that shows each of the extensions set for PAUSE, just not a reason why that would be. |
| 21:27.24 | beek | My configs (and queue show output) is at http://pastebin.ca/1503635 |
| 21:27.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:28.02 | [TK]D-Fender | \o/ |
| 21:28.35 | beek | [TK]D-Fender: IRC on your cell phone? That's decadent. |
| 21:28.36 | Corydon76-dig | beek: could be because somebody paused via the manager interface |
| 21:28.45 | Corydon76-dig | beek: or because of a ring-no-answer |
| 21:29.03 | [TK]D-Fender | beek: hAD THE PHONE FOR A YEAR & HALF ALREADY.... COPPICE WOULD SAY IT WAS AN OLD PHONE WHEN i GOT IT... |
| 21:29.08 | [TK]D-Fender | darn caps :p |
| 21:29.22 | beek | Corydon76-dig: Doesn't 'autopause=no' override the 'ring-no-answer'? |
| 21:29.34 | Corydon76-dig | beek: not according to the code, no |
| 21:29.49 | beek | I'm using these queues to ring a group of phones, some of which are unoccupied at times. |
| 21:29.49 | [TK]D-Fender | beek: Do you have a dialplan option to pause? |
| 21:29.52 | [TK]D-Fender | beek: Mine do |
| 21:30.03 | Orbixx | How can I get Record(...) and whatever follows the respective extension to continue to execute even after the caller hangs up mid-recording? |
| 21:30.17 | beek | [TK]D-Fender: Queue(OPERATOR1,t,,,20) is how I typically use it. |
| 21:30.56 | [TK]D-Fender | beek: I mean a dialplan ext to LET them pause themselves |
| 21:31.29 | beek | [TK]D-Fender: No, I don't. I'm just using a queue to hold calls while one or more manned phones ring. |
| 21:32.04 | beek | It is possible for a member to not take a call for a long time, yet I want it rung everytime. Queue apparently won't let me get away with that. |
| 21:32.49 | Corydon76-dig | beek: so either somebody is pausing the member through the manager interface or they're running PauseQueueMember through the dialplan |
| 21:34.40 | beek | Corydon76-dig: Hmmm... the only MI code is from Openfire's plugin module, and it's not tracking most of those members. And I know that PauseQueueMember isn't in the dialplan anywhere. |
| 21:36.33 | beek | Is my understanding of Queue correct? I have three phones (members) in a queue that may or may not have someone at that desk at that time. I'm using the ringall strategy so that they all ring. I thought that autopause=no would ensure that Asterisk wouldn't pause them, but perhaps my understanding is incorrect? |
| 21:36.40 | Corydon76-dig | beek: Note that if it calls the AMI with a blank queue name, it pauses all members of all queues |
| 21:37.30 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 21:38.14 | beek | Corydon76-dig: Openfire is being used to track the 'On Phone' status of the individuals, so its getting events from the AMI. I don't think it's issuing any commands. |
| 21:38.47 | Corydon76-dig | beek: you could add additional debugging to confirm WHAT is pausing your members |
| 21:39.12 | beek | Corydon76-dig: What do I need to do to enable that additional debugging? |
| 21:41.39 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 21:42.06 | Corydon76-dig | beek: http://asterisk.drunkcoder.com/patches/20090722__queue_pause_debug.diff.txt |
| 21:42.48 | beek | Corydon76-dig: This is for 1.6.0.10? |
| 21:43.00 | Corydon76-dig | 1.4, but should be similar enough |
| 21:43.16 | beek | I'll give it a try and see how it works. Thanks! |
| 21:49.17 | _ShrikE | I am running asterisk 1.4.25.1 and am having a problem accessing CDR(billsec) from the h extension in the dialplan. I have endbeforehexten=yes in cdr.conf, but billsec still returns 0 in the dialplan. It is storing the proper time in the cdr however. Any suggestions? |
| 21:49.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:52.34 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
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| 22:45.51 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 22:46.37 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 22:47.19 | [TK]D-Fender | \o/ Timmeh |
| 22:49.32 | bmoraca | why would Asterisk not be able to send the qualify messages to a SIP phone, and yet be able to communicate with it in every other aspect (i.e. the SIP phone can make calls and receive calls if qualify is turned off)? |
| 22:49.50 | sumasuma | what is the maximum size of the each asterisk variable? |
| 22:49.59 | sumasuma | what is the maximum size of the each asterisk variable can hold? |
| 22:49.59 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 22:50.02 | Orbixx | When WaitExten() is called, how can I accept any input to dial extensions 200 thru 299, even if only some exist? |
| 22:50.23 | [TK]D-Fender | Orbixx: You can't. Go make a pattern that can match |
| 22:50.30 | leifmadsen | pattern matches ftw |
| 22:50.41 | [TK]D-Fender | Dialplan basics <- |
| 22:51.26 | Orbixx | Not familiar with patterns. |
| 22:51.49 | [TK]D-Fender | Orbixx: Go read Chapter 5 a few dozen times. |
| 22:52.15 | [TK]D-Fender | Orbixx: this is beyond "bread & butter". |
| 22:53.10 | Orbixx | What? |
| 22:54.11 | [TK]D-Fender | Orbixx: Not knowing dialplan patterns is like preparing to take a Ferrari out on a test drive and asking "what's a clutch?" |
| 22:55.06 | jaytee | lol |
| 22:55.12 | beek | Hello jaytee |
| 22:55.17 | Orbixx | Is there an example I can see of a pattern? |
| 22:55.27 | jaytee | hi beek |
| 22:55.39 | jaytee | there's a bunch of them in the book |
| 22:55.42 | [TK]D-Fender | Orbixx: CHAPTER 5 |
| 22:55.45 | jaytee | ~book |
| 22:55.46 | infobot | [~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 at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:55.50 | beek | [TK]D-Fender: Your Pause/Unpause dialplan instruction... do you give any visual notification that they're 'paused?' |
| 22:56.22 | [TK]D-Fender | beek: aside from seeing the command execute? not AFAIK |
| 22:56.48 | beek | I'm interested in how you report that to the user... do you say "Paused" or anything to them? |
| 22:57.14 | beek | The reason I ask is that our propietary PBX has ACD and has a N/A button for not available, which shows that status. |
| 22:59.06 | *** join/#asterisk vicscandl (n=Christop@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 22:59.28 | beek | Anyway, I've cranked up verbosity and debug output and now capturing AMI events to a log. By damned I'll find out how these bastards are getting paused. |
| 22:59.41 | bmoraca | So...asterisk cannot communicate with my peer as shown here: http://www.pastebin.org/3669 and yet I can ping the peer from the asterisk box and receive calls as long as qualify=no and place calls regardless...what would cause these "OPTIONS" messages from not reaching the peer? |
| 23:02.40 | [TK]D-Fender | bmoraca: What are you using? Where is it located |
| 23:03.02 | bmoraca | it's a Cisco 7940 and it's located across a VPN |
| 23:04.13 | [TK]D-Fender | bmoraca: I'd wonder about your routing |
| 23:05.05 | bmoraca | [TK]D-Fender: as I stated, as well, I can verify connectivity in both directions via ping and other SIP messages transfer just fine |
| 23:05.38 | bmoraca | [TK]D-Fender: i get 2-way audio when placing calls out the trunk and I can call extension to extension if I set qualify=no |
| 23:05.42 | bmoraca | i'm baffled at this one |
| 23:06.42 | [TK]D-Fender | bmoraca: test with a softphone at that site |
| 23:07.26 | bmoraca | why would a softphone make any difference? the condition exists with all 28 cisco phones there...phones which I know work, as I've used them in other implementations |
| 23:08.08 | [TK]D-Fender | bmoraca: Isolate if the phones are any issue here |
| 23:08.10 | bmoraca | in fact, i set this up as a test yesterday and had no problems. yet I can't find a difference between my config now and my config then |
| 23:08.13 | [TK]D-Fender | bmoraca: Sanity check |
| 23:09.04 | *** join/#asterisk demiv (n=demiv@190.158.82.12) |
| 23:09.18 | Lyma | Hi! i'm looking for an eclipse plugin to write the dialplan (syntax highlighting only would be ok)... anyone knows? |
| 23:10.02 | vicscandl | is there a hardware compatability chart for AsteriskNOW? (having issues getting the ISO to boot up on a Dell PE2650 [test machine, not production]) |
| 23:11.18 | *** join/#asterisk K3rN3L (n=dam@infapen.com) |
| 23:11.23 | K3rN3L | hello |
| 23:11.32 | K3rN3L | somebody help me? |
| 23:11.41 | K3rN3L | im making a bot in AGI + asterisk |
| 23:11.50 | K3rN3L | i use php like a script language |
| 23:12.07 | K3rN3L | i dial to a extension on my pbx an this call other phone |
| 23:12.26 | K3rN3L | i need can call to phone a play a welcome sound |
| 23:12.54 | K3rN3L | how i can do this? |
| 23:13.12 | *** join/#asterisk demiv (n=demiv@190.158.82.12) |
| 23:14.10 | *** join/#asterisk demiv (n=demiv@190.158.82.12) |
| 23:16.14 | [TK]D-Fender | vicscandl: Its basically CentOS 5.2(+/-) + * |
| 23:17.25 | vicscandl | yea, thats what i thought it was, just odd that this "bonus" server that i have is being like "Rolling Stone", too many issues.... |
| 23:17.29 | bmoraca | uhg this is infuriating! the phone receives the messages, but asterisk doesn't get anything back from the phone...trying a firmware flash |
| 23:17.48 | vicscandl | thanks [TK]D-Fender. :) |
| 23:18.02 | [TK]D-Fender | vicscandl: Just roll your own |
| 23:18.40 | vicscandl | yea, thats what tomorrow is going to bring... was hoping for the easy way out.. :) |
| 23:18.50 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
| 23:18.52 | vicscandl | puts on his digital waders and dives in... |
| 23:19.39 | K3rN3L | exit |
| 23:21.26 | bmoraca | [TK]D-Fender: do you think this might have something to do with the fact that i have two NICs in the asterisk server and both are on separate networks? does asterisk have issues handling that? |
| 23:22.59 | [TK]D-Fender | bmoraca: GAH |
| 23:26.34 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:27.00 | bmoraca | [TK]D-Fender: although, if i disable the second nic, the issue still persists |
| 23:29.41 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:30.05 | [TK]D-Fender | bmoraca: Go prove its not the phones |
| 23:31.42 | bmoraca | [TK]D-Fender: telnetting into the phone and issuing a debug shows the OPTIONS messages being received by the phone and shows the attempt to send an OK back. the OK never gets there, yet connectivity (again) is verified on the phone. additionally, the phones can place and receive calls. |
| 23:31.42 | Nugget | telnet is eeeeeeevil! |
| 23:32.51 | [TK]D-Fender | bmoraca: you've got routing & firewalls to be checking... |
| 23:32.59 | [TK]D-Fender | bmoraca: Trace those packets... |
| 23:33.06 | [TK]D-Fender | bmoraca: And go do the sanity check |
| 23:33.09 | bmoraca | [TK]D-Fender: connectivity is not an issue. |
| 23:33.43 | Alfio | bmoraca i had an issue like you have and was my ips in the firewall |
| 23:33.54 | bmoraca | there are no firewalls in between. the VPN connection is not firewalled on either side |
| 23:34.55 | [TK]D-Fender | bmoraca: These are all dodgy non-test based answers |
| 23:35.13 | bmoraca | non-test based answers? do you want me to pastebin my traceroutes and pings for you? |
| 23:35.13 | *** part/#asterisk vicscandl (n=Christop@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 23:35.15 | [TK]D-Fender | bmoraca: Its called a sanity check because you do them even when you don't have a specific reason to. |
| 23:35.25 | [TK]D-Fender | bmoraca: Stop short-changing the debugging process |
| 23:35.35 | [TK]D-Fender | bmoraca: People miss stuff because of assumptions. |
| 23:36.36 | *** join/#asterisk field64 (n=6667@p5B22F010.dip.t-dialin.net) |
| 23:39.10 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 23:39.37 | bmoraca | [TK]D-Fender: I've done every test that I can think of to verify that connectivity exists between the two. the fact that I can MAKE PHONE CALLS ensures that I have connectivity. what else is there for me to check? if the issue was a phone issue, it would not be happening to 28 different phones. |
| 23:42.30 | [TK]D-Fender | bmoraca: So there are 27 other identical models on that same side of the VPN that are good? |
| 23:42.59 | bmoraca | no, there are 27 other phones which exhibit the exact same symptoms |
| 23:43.36 | [TK]D-Fender | bmoraca: So far it sounds pretty global on that subnet to me... |
| 23:45.01 | field64 | good evening all |
| 23:45.50 | bmoraca | right...and what are the common elements? connectivity (which can be ruled out) and asterisk. asterisk is not receiving or acknowledging the OKs from my phones. tomorrow I can attempt to set up another VPN to a test bench and test other phones, but I suspect the issue will persist because this is not a connectivity problem. |
| 23:45.51 | field64 | wondering about DAHDI dummy that shows results of "dahdi_test" 4.6777% |
| 23:45.58 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 23:46.24 | [TK]D-Fender | bmoraca: Rule out NOTHING |
| 23:46.37 | SkramX | can anyone recommend a good 'ol SIP softphone? I don't need to register with someone else's service, just want to be able to call out to SIP URIs |
| 23:46.43 | SkramX | d'oh - for iPhone |
| 23:46.44 | bmoraca | why would I continue to test something I know and have already verified to be correct? |
| 23:47.10 | [TK]D-Fender | bmoraca: verified correct? As in how? What alternative tests have you done besides those phones/ |
| 23:47.30 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 23:47.57 | bmoraca | [TK]D-Fender: connectivity is an all or nothing kind of thing...either it's there or it's not. it's there. end of story. |
| 23:48.28 | bmoraca | aside from that, i know that my phones are sending and receiving all applicable messages. asterisk is simply not receiving them. why, i can't say. |
| 23:48.37 | [TK]D-Fender | bmoraca: Ok that is a very hollow statement that I would never ever take at face value and tends to make stuff get over-looked. |
| 23:49.16 | [TK]D-Fender | bmoraca: And every time I hear you give more empty reinforcement to it I trust it even LESS |
| 23:49.28 | [TK]D-Fender | (Yes, I go negative) |
| 23:51.55 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 23:52.05 | Micc | Is there anyway to check if an extension exists? |
| 23:52.07 | bmoraca | [TK]D-Fender: for fuck's sake, do I need to teach you networking 101? i've been a cisco network technician for a long time. i know 100%, without question, that this VPN is set up correctly and WORKING. my issue is asterisk. end of story. it is not receiving certain messages. it does receive some (INVITES and registration requests), but it does not receive others. that's not a... |
| 23:52.09 | Micc | So I can goto it if it does? |
| 23:52.10 | bmoraca | ...connectivity issue. that's an asterisk issue. whether it's not actually listening on all IPs like I've configure it to or whether it's just fundamentally useless, the problem is with asterisk. |
| 23:52.17 | Micc | From the dialplan that is. |
| 23:52.52 | jaytee | telnet |
| 23:54.03 | *** join/#asterisk Iamnacho (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 23:55.03 | field64 | @bmoraca - just jumped in here, apologies if im missing the topic - but did you take a sniffer trace at the asterisk interface ? presume you've checked the ethernet stats, and ensured there are no errors, etc...perhaps a clever filter on wireshark would enable to you prove/disprove if the frames are arriving-sending/not, if you're correct, this would cut the circuit in half - sounds like you've already gone that far though. |
| 23:55.32 | [TK]D-Fender | bmoraca: * can't make SIP packets fail to arrive back. THAT is insanity |
| 23:56.21 | [TK]D-Fender | bmoraca: And I've had plenty of people tell me "I've been doing this for 20 years". Congratulations for them, 20 years and they still couldn't do it right. |
| 23:56.32 | field64 | @bmoraca - do the VPN GW have SIP ALGs active ? |
| 23:56.43 | [TK]D-Fender | bmoraca: you are putting yourself in front of a 2 minute test. |
| 23:56.54 | bmoraca | field64: there is no NAT on the VPN, so no, there are no ALGs. |
| 23:57.40 | field64 | @bmoraca - you can make calls both ways, just lose the keepalives ? |
| 23:57.45 | KavanS | lol |
| 23:58.13 | KavanS | define: cisco network technician |
| 23:58.15 | KavanS | ccna? |
| 23:58.19 | field64 | :) |
| 23:58.32 | KavanS | heh, I work with cisco developers... |
| 23:58.44 | field64 | do tell..KavanS |
| 23:58.46 | KavanS | and they aren't always the sharpest tools in the shead ;) |
| 23:58.51 | KavanS | *shed |
| 23:58.51 | field64 | :) |
| 23:58.54 | Micc | bmoraca, did you say you configured asterisk to listen on multiple ports? Last I checked it can't listen on multiple ports. |
| 23:59.00 | KavanS | so...such things are to be taken worth a grain of salt/sand :) |
| 23:59.10 | bmoraca | multiple IPs. |
| 23:59.13 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 23:59.30 | field64 | multiple ip/multiple MAC ? |
| 23:59.36 | KavanS | anyways...time for me to call it a day |
| 23:59.40 | Micc | ok, thats different then. So do you receive all invites from some IP's and not others? |
| 23:59.53 | Alfio | well i think you will need cisco call manager 4 :) |
