| 00:02.18 | *** part/#asterisk neal29 (n=np20433@nat/sun/x-c484fe5d35b22940) |
| 00:05.17 | *** part/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 00:20.54 | *** join/#asterisk mkillebrew (n=mkillebr@edge.50ae.net) |
| 00:21.38 | mkillebrew | anyone know what VoIP providers provide ANI with a toll free number and not just CPN? |
| 00:24.43 | mkillebrew | or does voicepulse provide ANI on inbounds to toll free and I just have asterisk improperly configured to take advantage of it |
| 00:26.40 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
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| 00:37.35 | *** join/#asterisk captiancrash (n=jonmoore@adsl-074-181-189-229.sip.owb.bellsouth.net) |
| 00:38.33 | *** join/#asterisk cosmo83 (n=arava@117.195.166.232) |
| 00:39.14 | cosmo83 | hi guys looking for some help in call recording through asterisk . can someone point me to the right direction .??? |
| 00:40.40 | mkillebrew | yes |
| 00:41.08 | mkillebrew | Monitor() |
| 00:42.05 | [TK]D-Fender | cosmo83: "core show application monitor" |
| 00:42.11 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:43.16 | [TK]D-Fender | mkillebrew: Check your SIP headers on the invite.... |
| 00:48.20 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-1c14e3823364ed55) |
| 00:57.17 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
| 00:58.36 | *** join/#asterisk riverman (n=samusweb@c-71-197-92-231.hsd1.ca.comcast.net) |
| 00:58.47 | riverman | hello. I have a question about the AGI manager api. |
| 00:59.30 | riverman | is there anyone here who can help me? |
| 01:00.06 | [TK]D-Fender | ~ask |
| 01:00.07 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:00.21 | riverman | ah. my bad |
| 01:00.54 | riverman | is it possible to do third party call connect using the manager api method 'originate'? |
| 01:01.17 | riverman | i.e. place a call to an outside line, once it picks up, connect it with another number |
| 01:06.22 | [TK]D-Fender | riverman: that is the entire point of Originate |
| 01:06.46 | riverman | okay. thank you, D-Fender. |
| 01:08.01 | [TK]D-Fender | riverman: You're welcome. It calls the Channel: you specify and on answer executes either a single application 9almost worthless), or more commonly dumps them into the dialplan where you tell it. This includes you specifying an exten that will dial out to some other resource. Useful for "click-to-call" type scripts, etc |
| 01:08.07 | [TK]D-Fender | darn |
| 01:08.12 | [TK]D-Fender | All that typing waste |
| 01:08.16 | [TK]D-Fender | dd |
| 01:21.51 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e5ec0ee34d191f8a) |
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| 01:25.57 | beek | [TK]D-Fender: If you have a moment -- http://www.pastebin.ca/1429486 |
| 01:26.32 | beek | This is a debug capture of the PRI->*->PRI bridge with and without Monitor(); |
| 01:27.51 | beek | Right in the middle of both traces, where it says 'Requested indication' is where it starts to diverge. |
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| 01:38.17 | beek | GN ALL |
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| 01:51.01 | rhassing | I'm trying to get Auto Answer to work on my Polycom IP330 |
| 01:51.17 | rhassing | I've added the SIP header "Alert-Info: Auto Answer" |
| 01:51.34 | rhassing | But my phone does not respond as expected :-( |
| 01:52.56 | carrar | Your polycom sip config the line <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/> in it? |
| 01:52.59 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.5) |
| 01:53.05 | [TK]D-Fender | rhassing: Have you set up the appropriate section of your provisioning as well to match? |
| 01:53.39 | rhassing | [TK]D-Fender, Yes... I think so... |
| 01:53.47 | [TK]D-Fender | rhassing: Pastebin it |
| 01:53.48 | [TK]D-Fender | ~pb |
| 01:54.08 | 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/ |
| 01:54.21 | rhassing | http://pastebin.com/m7d0bbcc2 |
| 01:55.07 | [TK]D-Fender | rhassing: <alertInfo voIpProt.SIP.alertInfo.1.value="Auto Answer" voIpProt.SIP.alertInfo.1.class=""/> <-- class should be "3" |
| 01:55.28 | rhassing | [TK]D-Fender, Ok, I'll try |
| 01:57.27 | rhassing | [TK]D-Fender, Thank, that did the trick :) |
| 01:58.00 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
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| 02:11.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 02:43.08 | chrisb | i want to use a teleconference company that uses asterisk |
| 02:43.17 | chrisb | do you know one? |
| 02:44.53 | *** join/#asterisk BeeBuu (n=beebuu@219.135.43.212) |
| 02:45.02 | *** part/#asterisk mkillebrew (n=mkillebr@edge.50ae.net) |
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| 02:46.42 | thehar | chrisb: just conference bridging or video as well? |
| 02:47.42 | chrisb | thehar: just voice telecon |
| 02:48.06 | thehar | chrisb: mind if i /query? |
| 02:48.13 | chrisb | thehar: go |
| 02:50.16 | BeeBuu | how can i play a sound file to all users in a room? |
| 02:51.15 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 02:51.33 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 03:02.30 | [TK]D-Fender | chrisb: Why does it matter what this 3rd party uses? |
| 03:12.07 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 03:13.38 | jblack | rotfl. A scammer is going to sue me for "liable" for posting the mailing address for his domain. :) |
| 03:17.19 | jaytee | isn't it spelled libel? |
| 03:17.37 | jblack | That, or defamation. |
| 03:18.20 | thehar | uhm. good luck. |
| 03:18.31 | thehar | did he have whois "privacy" set? |
| 03:18.48 | jblack | Nope. Publicly available. Pulled it straight out of whois. |
| 03:19.22 | thehar | heh. |
| 03:19.32 | thehar | darwin award goes toooo |
| 03:19.47 | jblack | Yes. Someone that going to hold me libel for liable, who housed a scam in his own home, is comin after me. |
| 03:20.31 | jblack | I didn't figure it out tonight, until I checked my spam filter. The threat's like a week old.... I mailed back with a "thanks for the joke". |
| 03:20.46 | drmessano | 419 works for me |
| 03:22.14 | jblack | This guys' scam was that he set up a paypal account to setup payments to sell the Kindle dx at 1/2 price. These days, his site, kindledxpromo.com, just redirects to the site. |
| 03:22.32 | *** join/#asterisk cosmo83 (n=arava@59.161.187.98) |
| 03:22.57 | jblack | He's been interviewed a couple times since. He's apparently pissed that the forums, amazon and such came down on him like a ton of bricks for "running a legitimate business" |
| 03:33.31 | cosmo83 | guys can somebody point me to the right place for call recording !!! |
| 03:34.08 | cosmo83 | i want to record calls that are going to a existing EPBX |
| 03:35.00 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:35.12 | carrar | sounds criminal |
| 03:45.08 | [TK]D-Fender | cosmo83: "core show application monitor" |
| 03:50.17 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 03:50.29 | cosmo83 | [TK]D-Fender: its a vanilla PBX not an asterisk PBX |
| 03:50.48 | Qwell | cosmo83: what? |
| 03:51.05 | [TK]D-Fender | cosmo83: What is a "vanilla" PBX? Does it have sprinkles on top? |
| 03:51.11 | cosmo83 | iam looking to record calls of a DIgital PBX |
| 03:51.21 | Qwell | #mydigitalpbx? |
| 03:51.41 | [TK]D-Fender | cosmo83: And what kind of advise are you expecting? |
| 03:51.49 | [TK]D-Fender | advice* |
| 03:52.04 | [TK]D-Fender | cosmo83: And do be more specific than "digital PBX" |
| 03:52.08 | cosmo83 | i have an existing PBX, iam asked not to touch it |
| 03:52.25 | cosmo83 | and asked to record incoming an outgoing calls of the same |
| 03:52.39 | [TK]D-Fender | cosmo83: So shove something in front that will take call from/to it and record |
| 03:53.12 | cosmo83 | can asterisk do that retaining the extensions ? |
| 03:53.27 | [TK]D-Fender | cosmo83: We don't even know what you have |
| 03:53.56 | [TK]D-Fender | cosmo83: And if you put * between your other PBX at telco's it can record all calls to the PSTN |
| 03:54.19 | [TK]D-Fender | cosmo83: Which I suppose might help if you described how it is that your current PBX even does this. |
| 03:55.10 | cosmo83 | so what kind of hardware iam looking |
| 03:55.41 | cosmo83 | i have heard about the sangoma cards that can act as CPE and also the NET side |
| 03:55.43 | [TK]D-Fender | cosmo83: Can't answer because YOU have not told us what your current PBX uses |
| 03:55.59 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
| 03:56.17 | cosmo83 | my current PBX has two PRIs and some analog lines |
| 03:56.47 | cosmo83 | and is a seimens PBX box |
| 03:57.56 | drmessano | I'll take Catch These Men for 100 |
| 03:58.15 | cosmo83 | [TK]D-Fender: anymore info u need ? |
| 03:58.20 | [TK]D-Fender | cosmo83: So much more informative. You'd then need a 4 port PRI card. Sangoma and Digium both make cards for this |
| 03:59.07 | cosmo83 | OK and how do i configure my asterisk extensions for recording . |
| 03:59.09 | [TK]D-Fender | cosmo83: So you'd configure * to act as CPE on ports 1&2, and as NET on 3&4 and pass in/out from each direction and record |
| 03:59.11 | *** join/#asterisk X-Rob (n=rob-x@18.139-27-211.static.dsl.brs.iprimus.net.au) |
| 03:59.23 | [TK]D-Fender | cosmo83: You go read the BOOK and learn how * works . |
| 03:59.25 | [TK]D-Fender | ~book |
| 03:59.26 | 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 |
| 03:59.49 | [TK]D-Fender | cosmo83: And go play around with some small scale test implementation |
| 04:00.06 | [TK]D-Fender | cosmo83: Its all in your DIALPLAN. this is where all call processing takes place |
| 04:00.11 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-7cf59cfd6e395be6) |
| 04:00.15 | cosmo83 | [TK]D-Fender: i agree with you on reading the book. my knowledge on the same is weak. |
| 04:00.31 | [TK]D-Fender | cosmo83: So 2 real things to configure : the hardware interface, and the call processing. |
| 04:00.54 | cosmo83 | iam worried on the call processing |
| 04:01.37 | [TK]D-Fender | cosmo83: Take call from 1 span, enable recording, dial out the other span |
| 04:01.44 | [TK]D-Fender | cosmo83: Nothing to worry about |
| 04:02.00 | jaytee | until something goes wrong :-) |
| 04:02.20 | [TK]D-Fender | jaytee: No, then the worry is gone, and the fear is realized :) |
| 04:02.22 | cosmo83 | OK, but the call will be in progress while recording right ? |
| 04:02.26 | jaytee | haahaha |
| 04:02.32 | [TK]D-Fender | cosmo83: Yes |
| 04:03.08 | cosmo83 | so lets say the agent has not lifted the phone my recording will just have rings in it |
| 04:03.32 | [TK]D-Fender | cosmo83: this has NOTHIGN to do with what you do on your PBX aside from calls coming in or actually going OUT. |
| 04:04.12 | [TK]D-Fender | cosmo83: Its stting between the PBX & the telco. Internal stuff is irrelevant. |
| 04:04.31 | cosmo83 | [TK]D-Fender: OK, i will try this out and bug u guys if iam stuck somewhere |
| 04:04.44 | cosmo83 | [TK]D-Fender: any ideas on the analog lines ? |
| 04:04.58 | [TK]D-Fender | cosmo83: that is more of a PITA, but doable as well... |
| 04:05.22 | cosmo83 | i need an FXO and FXS card ? |
| 04:05.30 | [TK]D-Fender | cosmo83: You'll need a matching set of FXS & FXO ports on some interface to do the same thing as you are going to do with PRI |
| 04:06.03 | cosmo83 | FXO goes to the telco and FXS goes to the existing PBX |
| 04:06.11 | Micc | why does chan_features not show up in make menuselect in 1.6.1.0?? |
| 04:06.27 | [TK]D-Fender | cosmo83: Yes and * serves the same role |
| 04:06.43 | [TK]D-Fender | Micc: becase its RES_features |
| 04:08.07 | cosmo83 | [TK]D-Fender: also i have been going through the sangoma pages i have found a PRI tapping device |
| 04:08.19 | cosmo83 | [TK]D-Fender: will that help |
| 04:08.28 | [TK]D-Fender | coscwhat does it say its for> |
| 04:08.36 | [TK]D-Fender | cosmo83: what does it say its for? |
| 04:08.59 | *** join/#asterisk xpot (n=james@70.91.210.233) |
| 04:09.05 | cosmo83 | it says its a T1 tapping device connected again to a sangoma dual port card |
| 04:09.14 | [TK]D-Fender | cosmo83: what does it say its for? <-------- |
| 04:09.30 | [TK]D-Fender | cosmo83: "tapping" is not an all encompasing magic word |
| 04:09.33 | X-Rob | [TK]D-Fender, cosmo83 is da feds. |
| 04:09.43 | [TK]D-Fender | (unless you're talking about Jessica Alba's ass) |
| 04:09.56 | [TK]D-Fender | ZING! |
| 04:09.57 | Micc | TKD-Fender, there is no res_features here either. |
| 04:10.03 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 04:10.07 | cosmo83 | X-Rob: hope everyone is still staying in the room |
| 04:11.43 | cosmo83 | they say its tap connection adapter |
| 04:12.27 | [TK]D-Fender | cosmo83: And you keep repeating those 2 words like they mean something. |
| 04:12.53 | [TK]D-Fender | cosmo83: You appear to have no ability to read tht e description blurb they put on their own product page |
| 04:13.05 | Qwell | TAPI? |
| 04:13.24 | X-Rob | chortles |
| 04:13.28 | X-Rob | is that what he means? |
| 04:13.33 | Qwell | throws $10 on the table. |
| 04:13.47 | Qwell | cosmo83: Is the word on the site "TAPI"? |
| 04:14.53 | X-Rob | tee aie pee eye |
| 04:16.17 | Micc | TKD-Fender, its always been chan_features, its never been res_features. |
| 04:16.18 | cosmo83 | Qwell: No, they mention the word "Line Monitoring" |
| 04:16.31 | Qwell | that's 2 words. |
| 04:16.34 | Qwell | context? |
| 04:17.00 | cosmo83 | Qwell: context is of call recording |
| 04:17.33 | Qwell | That mirrors the port. |
| 04:17.48 | Qwell | doesn't help you at all |
| 04:18.28 | cosmo83 | Qwell: If it mirrors the port can it be used to record the same . |
| 04:18.32 | Micc | features show at the CLI seems to work still so maybe chan_features moved to a more core part of asterisk. |
| 04:18.41 | Qwell | cosmo83: You...still need software. |
| 04:19.03 | [TK]D-Fender | Qwell: That device makes references to a PCI card taht we never SEE and gives an impression that it is NOT a separate ATF card necessarily and just decodes G.711 streams off it "somehow" |
| 04:19.14 | [TK]D-Fender | Qwell: with APIS for Linux/Win |
| 04:19.15 | Qwell | [TK]D-Fender: it's a splitter |
| 04:19.32 | cosmo83 | Qwell: will * do the job ? |
| 04:19.38 | [TK]D-Fender | Qwell: I know what it LOOKS like, but when you read the docs it makes it sound like that pic is only half of what it comes with |
| 04:19.43 | Qwell | cosmo83: Everything everybody has told you is correct. |
| 04:19.54 | [TK]D-Fender | Qwell: Its dangerous allusions ALL OVER :) |
| 04:20.17 | [TK]D-Fender | Qwell: Or as I prefer to call it : fucking vague shit! |
| 04:20.19 | drmessano | Im not wearing any pants |
| 04:20.27 | [TK]D-Fender | Qwell: this time, Sangoma = FAIL |
| 04:20.35 | [TK]D-Fender | drmessano: TMI |
| 04:20.37 | rob0 | drmessano==Paul |
| 04:20.57 | rob0 | /j #abbeyroad |
| 04:21.10 | drmessano | Wasnt that shoes? |
| 04:21.12 | cosmo83 | [TK]D-Fender: From what i understood, along with this so called splitter u need a dual port PRI , which again with wanpipe installed can send the RTP info to softwares like orec |
| 04:21.17 | rob0 | oh hmmm ... does this mean drmessano is dead? |
| 04:21.40 | [TK]D-Fender | cosmo83: If you still need a separate AFT WTF is this good for? |
| 04:21.47 | drmessano | I cant be dead, I am the walrus |
| 04:21.57 | Qwell | koo koo ka-chu |
| 04:21.59 | cosmo83 | [TK]D-Fender: thats what seems to be vague |
| 04:22.25 | [TK]D-Fender | cosmo83: If you want to know about that device it has nothing to do with * as far as I can tell |
| 04:22.35 | [TK]D-Fender | cosmo83: I'v already told you what I know will work |
| 04:22.46 | cosmo83 | [TK]D-Fender: as u have suggested before i will look into the MIM, bug u guys later |
| 04:24.24 | rob0 | oh yeah shoes, I think I was confusing it with the satire, the Rutles |
| 04:24.40 | cosmo83 | thanks guys |
| 04:24.45 | *** part/#asterisk cosmo83 (n=arava@59.161.187.98) |
| 04:27.09 | drmessano | This is boring.. someone send trentcreek back to #centos |
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| 04:44.06 | bbryant | yes |
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| 05:36.04 | KyleK | oh just great, calls go out as username@did.voip.les.net and calls come in as number@did.voip.les.net, now to figure out how to make this SPA3102 work with that |
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| 08:22.39 | WeazelON | hi guys, Can someone tell me how to skip the option in MeetMe for "Press 1 to accept this recording" ["Playing 'vm-review' (language 'en')"] i want to say my name and directly go inside the Meetme |
| 08:24.53 | WeazelON | anyone ? |
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| 08:31.50 | KyleK | worst case you peek at app_meetme.c |
| 08:33.49 | WeazelON | allow me to rephrase the question so it'll be a bit more clear |
| 08:34.13 | WeazelON | i want it to NOT ask for name confirmation when recording names on Asterisk 1.2 i think its "I" that replaces "i" |
| 08:34.20 | WeazelON | question is that possible for asterisk 1.2 ? |
| 08:39.54 | WeazelON | KyleK, where is the elusive app_meetme.c ? |
| 08:40.04 | WeazelON | (asterisk 1.2.17) |
| 08:40.37 | *** join/#asterisk hi365 (n=hi365@94.159.178.51) |
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| 08:51.30 | WeazelON | no buyers ? |
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| 08:52.32 | ck_28 | hiall |
| 08:52.44 | ck_28 | 0x40 (slin) what does it means |
| 08:53.04 | ck_28 | it begins like that then it switch to 0x100(g729) |
| 08:58.56 | tzafrir_laptop | what does? |
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| 09:56.10 | mahiti-irc | hi |
| 09:56.27 | mahiti-irc | im using asterisknow 1.5.0 and digium tdm400p |
| 09:56.59 | mahiti-irc | but when i make a call via analog line i just get full ring |
| 09:57.09 | mahiti-irc | its not answered !! |
| 09:57.32 | mahiti-irc | any idea? |
| 10:03.35 | tzafrir_laptop | mahiti-irc, what module: FXS (connected to an analog phone) or FXO (connected to PSTN)? |
| 10:04.06 | mahiti-irc | fxs |
| 10:11.02 | tzafrir_laptop | FXS? you have to pick up the phone |
| 10:11.56 | tzafrir_laptop | The FXS side (Asterisk) does not decide when the line is connected. It is the phone that does |
| 10:12.42 | tzafrir_laptop | Unless you actually confuse FXS and FXO |
| 10:23.24 | mahiti-irc | ? |
| 10:28.55 | mahiti-irc | tzafrir_laptop, actually tried swapping the phone line with both the modules |
| 10:29.00 | mahiti-irc | but its the same |
| 10:29.05 | mahiti-irc | i get a full ring |
| 10:29.09 | mahiti-irc | no answer |
| 10:29.48 | tzafrir_laptop | do you have an analog phone or do you try connecting to a PSTN provider? |
| 10:30.10 | mahiti-irc | analog phone connected to the tdm card |
| 10:40.42 | mahiti-irc | ? |
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| 10:56.09 | zeeesh | compiling zaptel ... getting error "/usr/src/svn/zaptel-1.4.10.1/kernel/xpp/xdefs.h:120: error: conflicting types for âboolâ |
| 10:56.09 | zeeesh | include/linux/types.h:36: error: previous declaration of âboolâ was here"? |
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| 11:43.09 | shyam_k | anyone know whether can i make dahdi cards work in bayonne?(don't kill me);-) |
| 11:43.42 | iksik | kills ya |
| 11:43.44 | iksik | ;P |
| 11:47.27 | kaldemar | shyam_k: i guess #bayonne would be a better place to ask |
| 11:51.41 | shyam_k | kaldemar am waiting for dyfet there for couple of days now.. i feel lonely there.. will mail bayonne-devel as soon as its free net time;-) |
| 11:52.25 | shyam_k | anyway i found this thing in todo list:) |
| 11:53.05 | shyam_k | i mean in bayonne/TODO. |
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| 12:04.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:30.35 | beek | [TK]D-Fender: morning... |
| 12:31.09 | [TK]D-Fender | beek: Mornin' |
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| 12:36.21 | beek | morning jaytee |
| 12:36.24 | jaytee | morning beek |
| 12:43.49 | timeshell_atwork | Happy Thursday! |
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| 12:46.27 | ariel_ | Morn |
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| 13:01.11 | mweichert | hello, I'm having a problem trying to get my dialplan patterns to work. I'm using the pattern 220[1-5] to refer to extensions 2201-2205, but if I dial any of those extensions I get "your call cannot be completed as dialed". |
| 13:01.20 | mweichert | I'd appreciate any help - here is the config and log: http://pastie.org/485265 |
| 13:02.25 | file | you have to put _ in front to indicate that it is a pattern match |
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| 13:02.44 | mweichert | file, oh, ok... thanks! |
| 13:02.44 | file | or am I crazy... |
| 13:03.10 | file | nope, I'm not |
| 13:04.15 | [TK]D-Fender | and I LOVE people who talk about messages they get which they feel should be taken for granted and ... yup... its ANOTHER GUI box... |
| 13:05.00 | [TK]D-Fender | file: And no... you are still crazy, you just happen to be right too ;) |
| 13:05.22 | file | crazy like a banana! |
| 13:06.04 | [TK]D-Fender | file: A mixed-metaphor a day is worth 2 in a bush ;) |
| 13:06.18 | file | rabid foxes are eating my foot. |
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| 13:06.51 | *** part/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net) |
| 13:07.28 | seste | hi all |
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| 13:09.36 | seste | I'm having problems with chan_mobile....no audio is sent on both directions. In the log I got a erro 'sco_read() error 104' here is a log: http://pastebin.com/m3faf56ef |
| 13:10.03 | seste | this erro means connection reset by peer. |
| 13:10.51 | seste | I'm using asterisk and addons 1.6.1.0 |
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| 13:17.26 | seste | some one knows why is this happening?? |
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| 13:42.24 | timeshell_atwork | seste Any particular reason you're using 1.6.1? |
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| 13:51.15 | Aiatek | do i need internet conection to install Dahdi? |
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| 13:53.04 | tzafrir_laptop | Aiatek, with the default settings: yes . Those things that get downloaded are firmwares for digium cards. Do you have one? |
| 13:53.17 | tzafrir_laptop | Do you actually have a problem with that downloading? |
| 13:54.06 | tzafrir_laptop | (the debian package is patched as an autobuilder should not assume to have an internet connection) |
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| 13:59.47 | seste | timeshell_atwork: no reason...is the one that i got from the website, Why?? |
| 14:00.39 | [TK]D-Fender | seste: 2 words : bleeding edge |
| 14:00.54 | timeshell_atwork | Personally I consider1.6.0 more stable |
| 14:02.38 | tzafrir_laptop | timeshell_atwork, I personally go for "go for a specific version and stick with it for a while" |
| 14:03.00 | timeshell_atwork | even if it doesn't work |
| 14:03.08 | tzafrir_laptop | If you have a while till production, 1.6.1 is probably the one to start with |
| 14:03.55 | Aiatek | yes |
| 14:04.06 | Aiatek | i was installing in my house |
| 14:04.30 | Aiatek | and give me an erro bvecause i dont have internet |
| 14:04.32 | Aiatek | error |
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| 14:10.55 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 14:11.14 | Vec | Does asterisk support multiple IAX users per IAX trunk ? |
| 14:11.49 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:12.03 | Vec | i.e. if A and C is making calls to B does B only need to have 1 IAX trunk setup ? |
| 14:13.11 | *** join/#asterisk clintc (n=clintc@n128-227-34-74.xlate.ufl.edu) |
| 14:14.22 | timeshell_atwork | sounds right |
| 14:15.11 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
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| 14:18.16 | [TK]D-Fender | Vec: If your abuse of the term "trunk" means "a single peer entry on the receiver", then sure. Of course you won't know which system the call is from |
| 14:18.30 | [TK]D-Fender | Vec: and that only applies to inbound calls |
| 14:18.51 | Vec | [TK]D-Fender : cool thanks, yeh I am abusing the term trunk :), I have set trunk=no |
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| 14:24.58 | eppigy | D: |
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| 14:32.23 | jaytee | D: |
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| 14:36.10 | eppigy | 8[] |
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| 14:46.17 | IPGHOST | hi buddies , do dahdi supports dialogic cards ? |
| 14:49.10 | seanbright | no |
| 14:54.32 | *** join/#asterisk sharp (n=sharp@sauropod.org) |
| 14:54.46 | sharp | what is a good way to provide sip service on multiple ports? |
| 14:57.20 | [TK]D-Fender | sharp: Whats the point? |
| 14:57.43 | jaytee | the sharp end of a spear or sword? |
| 14:58.49 | timeshell_atwork | jaytee : Pencil |
| 14:59.14 | beek | jaytee opens this week in Atlantic City |
| 14:59.19 | *** join/#asterisk philippel (n=p_lindhe@pool-98-111-74-92.sttlwa.fios.verizon.net) |
| 14:59.31 | sharp | [TK]D-Fender: i want to have sip running on 5060 |
| 14:59.37 | Great_Anta_Baka | does fax passthrough have to be configured on both the outbound trunk and the ata device in the asterisk config files? |
| 14:59.38 | sharp | because its the default |
| 14:59.45 | [TK]D-Fender | sharp: thats funny, thats what it NORMALLY runs on |
| 14:59.57 | sharp | but my housemates use a stupid vonage router |
| 15:00.10 | sharp | that eats anything with a src port of 5060 |
| 15:00.26 | [TK]D-Fender | sharp: "bindport=5061" under [general] |
| 15:00.44 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:00.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:00.45 | sharp | but then i'm not serving sip on 5060 anymore |
| 15:00.47 | therealcircut | hey all |
| 15:00.49 | [TK]D-Fender | sharp: Of course you'll have to tell all your devices that reg to it to use that |
| 15:01.22 | sharp | so i can no longer tell people "call me at sip:sharp@server.org" |
| 15:01.38 | therealcircut | if i keep getting this in my console: |
| 15:01.40 | therealcircut | pbx02*CLI> ^M[May 20 09:39:03] WARNING[28666]: chan_dahdi.c:7086 handle_init_event: Unable to play dialtone on channel 50, do you have defaultzone and loadzone |
| 15:01.40 | sharp | its now "sharp@server.org:5061" |
| 15:01.43 | therealcircut | defined? |
| 15:02.10 | sharp | [TK]D-Fender: i |
| 15:02.17 | sharp | [TK]D-Fender: i'd like to run on two ports |
| 15:02.18 | therealcircut | where do i fix it? I've already added those 2 lines to my chan_dahdi.conf file, under each context as well as the defaults |
| 15:02.25 | [TK]D-Fender | sharp: Not possible |
| 15:02.37 | sharp | maybe with SER, but i don't know if theres another way |
| 15:02.46 | [TK]D-Fender | sharp: Setup a proxy if you really want to waste time on what seems like a very small need |
| 15:03.01 | [TK]D-Fender | sharp: * will not bind to 2 ports for this |
| 15:03.09 | kaldemar | or play with iptables if there's a module to mangle sip packets. |
| 15:03.23 | [TK]D-Fender | therealcircut: those 2 lines don't belong in chan_dahdi.conf |
| 15:03.33 | philippel | question, tested on the latest app_queue.c 1.4 svn, when I setup queue members in queues.conf, Asterisk appears to re-order them as it sees fit which breaks ring strategies like rrmemory or the backport of linear. Anyone aware of this issue? I did one test putting the members in two different orders and both times show queue has them in it's own order? |
| 15:03.44 | therealcircut | [TK]D-Fender: ?? |
| 15:03.56 | therealcircut | they are also in the system.conf file for dahdi |
| 15:04.21 | kaldemar | maybe a mere prerouting from 5060 to 5061 would do. |
| 15:04.27 | [TK]D-Fender | therealcircut: Got an indications.conf file as well? Care to pastebin your configs so we can SEE what you're doing? |
| 15:05.05 | *** join/#asterisk stijnbe (n=stijnbe@78-22-110-114.access.telenet.be) |
| 15:05.30 | philippel | putnopvut ping |
| 15:05.35 | putnopvut | philippel: pong |
| 15:05.46 | therealcircut | [TK]D-Fender: which configs u want? |
| 15:05.48 | philippel | my comment above, are you aware of the issue or am I confused? |
| 15:05.55 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:05.59 | philippel | order of memebers in queues.conf |
| 15:06.11 | therealcircut | [TK]D-Fender: i do have an indications.conf file |
| 15:06.53 | putnopvut | philippel: I'm taking a look... |
| 15:07.11 | philippel | putnopvut thanks, going to run get a cup of coffee, will be back within 5 |
| 15:07.36 | therealcircut | [TK]D-Fender: you want my system.conf file |
| 15:08.21 | putnopvut | philippel: yeah, the problem is that a hashtable is used for queue members, and we can't guarantee an order that the members will be stored in the container. You can change this behavior by modifying the first param to ao2_containter_alloc in init_queue to be 1 instead of 37. |
| 15:09.06 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 15:10.02 | putnopvut | That way, all members are placed into one hash bucket and the order used in queues.conf will be preserved. |
| 15:12.35 | philippel | putnopvut wouldnt' this be considered a bug? does the same problem exist in 1.6? (where linear is supported and this would be a bug for that reason)? |
| 15:13.02 | *** join/#asterisk hi365 (n=hi365@94.159.176.165) |
| 15:13.12 | putnopvut | philippel: Well, the linear strategy is the only one in which the order of members matters, and this is dealt with in the code of 1.6 |
| 15:13.20 | jaytee | wow, second news item from the top on voip-info.org. I'm having trouble suppressing the laughter and the stream of urine running down my leg |
| 15:13.25 | jaytee | Why manufactors of a Windows based PBX belives that it trumps a Linux based appliance or proprietary PBX http://www.3cx.com/news/softswitch-advantages.html |
| 15:13.29 | hi365 | amyome using an audiocodes mp-114? im trying to figure out if I can, from asterisk, tell it which port to use on outbound calls |
| 15:13.44 | hi365 | (i.e. instead of using the audiocodes routing) |
| 15:16.03 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 15:16.04 | philippel | putnopvut well that is why I brought up linear - however, one could argue that to some extent rrmemory matters as well. Although you can't gurarnttee where it starts, I could imagine there might be some people who have an expectation of order still |
| 15:16.54 | rob0 | jaytee, I won't have time to read the paper, but ISTM that there might be some valid points, albeit poorly thought-out and based on some bad assumptions. |
| 15:16.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:17.28 | putnopvut | philippel: Yeah, I can see how one might expect that to be true, but (AFAIK) it has never been documented that the rrmemory strategy behaves that way. |
| 15:18.00 | rob0 | One valid point is that "competent" Windows people are a dime a dozen, competent Linux people are rare, and it's nearly impossible for a non-techie to determine who's truly competent. |
| 15:18.42 | coppice | competant people of any discipline are pretty rare |
| 15:19.13 | asteriskmonkey | how can you do a call confirmation with Dadhi.. i used to use Zap/g1c/yadda but the c option no longer works in dahdi |
| 15:19.37 | philippel | putnopvut hmm, let me take a look at rrmemroy (as well as making your change) in the mean time, I had another question (maybe better for the dev channel but ...) - how difficult woudl it be to add an option to app_queues.c that effectively had 'noanswer' members be treated as if they were busy/unavailble - so that one could use penalties in a way that agent not answering there phone behaved as unavail? |
| 15:20.46 | asteriskmonkey | how can you make an agent a cell phone with # confirmation on pickup, the c option no longer works in Dahdi |
| 15:21.39 | putnopvut | philippel: that would be trickier than it sounds, given the way app_queue works right now. |
| 15:22.37 | philippel | putnopvut so there is not one place where it could be 'spoofed' so to speak, if the paramter is set, treat a disposition of NOANSER as BUSY? |
| 15:23.02 | philippel | putnopvut I know using a local channel I have breadboarded it ot spoof it that way from the channel? |
| 15:23.58 | [TK]D-Fender | asteriskmonkey: M() |
| 15:24.11 | philippel | putnopvut also - I took a look at the a02_container_alloc, looks like the backported linear patch does set it to 1 but the members are still out of order, weird. (note - grabed 1.4 svn of app_queue.c only, not the whole branch, for this test I did) |
| 15:25.50 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 15:26.31 | asteriskmonkey | M? [TK]D-Fender, can you point me to the updated dial spec for that? |
| 15:26.49 | jaytee | core show application dial |
| 15:28.48 | asteriskmonkey | oh cool thanks, i see now |
| 15:29.31 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
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| 15:30.05 | putnopvut | philippel: that may be because you are allocating the container before you know that the strategy will be linear. |
| 15:30.28 | putnopvut | So the container is being allocated with 37 hash buckets, then you see that the strategy will be linear. Had to work around that in 1.6 as well. |
| 15:30.37 | therealcircut | can anyone help me with the following error message: |
| 15:30.42 | therealcircut | WARNING[28666]: chan_dahdi.c:7086 handle_init_event: Unable to play dialtone on channel 50, do you have defaultzone and loadzone |
| 15:30.43 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 15:31.50 | philippel | putnopvut let me look at that - I just put in a debug as I was going to check on that, any thoughts on the penalty from my comment, or is the trick I breadboarded in the local channel not something easily 'spoofed' in the application? |
| 15:32.25 | [TK]D-Fender | asteriskmonkey: get coding little monkey! |
| 15:32.51 | putnopvut | philippel: It's not something easily spoofed in the application, unfortunately. |
| 15:33.27 | philippel | so that would be the reason why this has been desired by many for so long but up to now hasn't been able to be done I assume? |
| 15:33.57 | Anth8708 | guys, i have a weird problem. i have a tdm808p and a TE122 in a box and on reboot, they have been "swapping" places, changing my channel setup. Is this normal? |
| 15:34.15 | putnopvut | philippel: Because most people can easily work around it either in a way like you have or by exploiting the "autopause" option. |
| 15:34.25 | *** join/#asterisk hi365 (n=hi365@94.159.176.165) |
| 15:35.03 | philippel | putnopvut doesn't autopause force them to re-engage though? |
| 15:36.03 | putnopvut | Yes, an "unpause" operation is necessary to make the member reachable again. |
| 15:36.17 | philippel | btw you are right from the debug message I added, it didn't call the modified alloc - I guess that patch is slightly flawed - what is the 'consequence' of changing the hash form 37 to 1 always? |
| 15:36.22 | putnopvut | The method by which that is accomplished is left to the admin. |
| 15:37.05 | putnopvut | If you had a queue with lots of members, you may see some slowdown since finding a member would be an O(N) operation instead of a constant time operation. |
| 15:37.16 | putnopvut | (That was with regards to changing to 1 from 37) |
| 15:37.52 | philippel | so I assume for small queues, no real cost (small being maybe less than 60 members?) |
| 15:38.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:38.12 | putnopvut | philippel: I would agree with that. |
| 15:38.56 | philippel | so what was teh workaround, conceptually, for 1.6 and the linear issue. Was it bascially making sure this was set to 1 some how or something more elaborate? |
| 15:42.06 | putnopvut | philippel: we make sure we know the strategy that will be used for a queue before doing any other initializations for the queue. If using linear, then allocate a hash table with only 1 bucket, otherwise use the default 37. |
| 15:43.56 | philippel | so you bascially read in the full config before allocating anything and check for linear strategies? I assume that involves a bit more re-arranging of the code, not a super trivial change to this 1.4 backport of the linear patch? |
| 15:44.52 | anonymouz666 | putnopvut: if we set VOLUME(rx/tx) before Monitor, does that will affect the volume on the record calls? I bet so |
| 15:45.02 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:45.06 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 15:45.16 | putnopvut | philippel: Actually, the config is read in anyway when loading queue data. It's just that now we actually do some parsing before doing the init_queue operation. |
| 15:45.38 | putnopvut | anonymouz666: that is an excellent question. I'm not sure the order in which audiohooks are processed. |
| 15:45.53 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552044.dsl.bell.ca) |
| 15:47.39 | putnopvut | anonymouz666: after looking at the audiohook code, I think you are correct. |
| 15:48.18 | anonymouz666 | wonderful. thanks. |
| 15:49.00 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:49.22 | nephfl | it seems like i can use a call file to get one channel into a meetme conference but I cant add a second channel that way or the meetme dies |
| 15:49.46 | putnopvut | anonymouz666: actually, now that I take a closer look, I think I am wrong. It looks like the "spy" hooks are written to, and then after that, any "manipulate" audiohooks are used on the audio. I think this means that MixMonitor will receive the audio without the volume modification. |
| 15:50.00 | putnopvut | anonymouz666: You should be able to test this pretty easily, though. ;) |
| 15:50.00 | atis_work | philippel: what exactly do you want to do with noanswer/busy? |
| 15:50.29 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 15:50.53 | anonymouz666 | putnopvut: this is the same for Monitor? |
| 15:51.04 | philippel | atis_work - I would like to use penalties, but in such a way that if an agent does not answer the phone, they are considered as if they are not available thus triggering the excallation to the next higher penalty |
| 15:51.35 | philippel | right now, if they don't answer the phone (thus NOANSWER) then the queue will never try the next penalty level |
| 15:52.13 | philippel | it's a very common desire for many uses of queues in SMB environments that are not true call centers |
| 15:52.47 | atis_work | so, agent A with penalty 1 and agent B with penalty 2 are logged in |
| 15:52.55 | atis_work | and there's C with penalty 1 |
| 15:53.03 | atis_work | so if now A doesn't answer - it goes to B? |
| 15:54.07 | atis_work | ooh, i think i'm starting to get it.. if A doesn't answer it goes to C and then back to A, right? |
| 15:54.41 | philippel | if you have A and B with penalty 1, and C with penalty 2, the normal behavior is never try C if A or B are available, and not answering the phone is considered available. This is proper for standard call centers, agents shoudl always answer or be busy talking |
| 15:55.07 | philippel | however, there are many situaitons where you may want to change the behavior and say if A or B don't answer there phone, then it shoudl go to C |
| 15:55.14 | atis_work | you could tweak it with wrapuptime |
| 15:55.24 | philippel | how |
| 15:55.59 | philippel | from what I understand and have played with, it wil never escllate up to C if A and B can be run unless both are busy |
| 15:56.04 | atis_work | well, if wrapuptime is high enough, A missing call will get lastcall set to current time, and then wrapuptime kicks in |
| 15:56.27 | atis_work | also, there's feature in development, that will allow to change wrapuptime temporarily for agent |
| 15:56.46 | *** join/#asterisk Corydon76-dig (i=white@pdpc/supporter/bronze/Corydon76-home) |
| 15:56.46 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 15:56.59 | atis_work | so, when you get NOANSWER in dialplan, you just set wrapuptime for next call to 1 minute, and let ring other agents |
| 15:57.11 | atis_work | or 5 minutes - depending on count of agents |
| 15:57.12 | asteriskmonkey | looking for help on this, the old zap c option would ring and wait for a #, it # was pressed it would connect the call, the M option with the new dial command dosnt offer up any scenario that would make this the same.... any pointers to a solution ? |
| 15:57.33 | Chuggs | ~book |
| 15:57.34 | 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 |
| 15:57.50 | philippel | atis_work interesting approach, doens't wrapupttime only apply if they answered the call though? |
| 15:58.15 | atis_work | well.. i'm confident to change that :D |
| 15:58.40 | atis_work | actually - wrapuptime applys always based on lastcall |
| 15:58.42 | cesar_CR | hello guys, any recomendation... I am looking for a calling-card software for an asterisk in a "cyber-cafe" |
| 15:59.06 | atis_work | so, i'm thinking of letting user to update lastcall upon missed call (optionally) |
| 15:59.20 | atis_work | so, wrapuptime gets added - and agent still can't be called |
| 15:59.50 | atis_work | but yeah, that's quite complex system.. not for easy enable-and-works |
| 16:00.16 | atis_work | or few custom settings could be created based on this mechanism |
| 16:00.21 | asteriskmonkey | atis did you find an answer to the multiple agents dial/macro exection issue in 1.6 |
| 16:00.26 | atis_work | putnopvut: are you reading? :) |
| 16:01.39 | philippel | atis_work well there are various ways to get the starts and moons to line up that may get thigns working, but it would be really nice to find a way to get penalties to work as described, ideally on a per queue basis but on an all or nothing basis woudl be quite acceptable as well (e.g. in general, usepenaltyonnoanswer=yes, somethign like that) |
| 16:02.33 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 16:03.01 | atis_work | umm, just thought of some caveats |
| 16:03.43 | atis_work | what if there's just one agent with higher penalty and he logs out while call is escalated to search for agent of next penalty.. then both previous agents are not ringed because of wrapuptime.. |
| 16:03.58 | atis_work | but probably that's ok, because they were away before.. |
| 16:04.06 | therealcircut | bunch a bums! |
| 16:04.45 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:05.30 | mike345 | argh... all the analog modem connections that go in and out of my asterisk on PRI's are not handshaking this morning. voicew calls seem fine. argh... worked fine for a few years...today no handshakes. any ideas? I've restarted the fax machine, modem banks, etc, but no luck. waiting for no call volume to restart asterisk. anyone else have any ideas? |
| 16:07.38 | *** join/#asterisk buzzyd (n=psp-man@86.54.239.113) |
| 16:07.51 | *** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
| 16:08.18 | JerJer | does anyone know if http://www.e164.org/wiki/AsteriskSRTP is the latest info for SRTP shit on asterisk ? |
| 16:08.22 | buzzyd | Hi all, Does anyone know if meetme in dynamic mode can play the conference room number when joining? |
| 16:08.55 | buzzyd | Seems like a simple thing but can't see anything on voip-info.org about it |
| 16:13.22 | *** join/#asterisk ctooley (n=ctooley@76.22.158.92) |
| 16:14.10 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 16:14.12 | ctooley | 3 boxes on the same subnet... BoxA<--->BoxB<--->BoxC |
| 16:14.12 | ctooley | Box A calls Box B via SIP. Box B calls Box C via SIP. Box A and Box C are bridged together |
| 16:14.25 | ctooley | canreinvite=no is set on every single SIP definition in sip.conf for all servers |
| 16:14.34 | ctooley | however the RTP gets sent from BoxA<--->BoxC |
| 16:14.42 | ctooley | What's the setting to make BoxB stay in the RTP stream? |
| 16:15.02 | ctooley | should know this |
| 16:15.54 | JerJer | canreinvite=no in [general] is typically what i use |
| 16:16.05 | ctooley | JerJer, that's there too |
| 16:19.36 | Corydon76-dig | ctooley: sounds like a bug |
| 16:20.24 | ctooley | Corydon76-dig, 1.6.0.3 |
| 16:20.31 | ctooley | Will update to the later version |
| 16:20.46 | *** part/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 16:27.00 | *** join/#asterisk dbcooper1 (n=User@64.203.244.146) |
| 16:31.14 | ctooley | :-) Compiling 1.6.0.9 that has 729 active calls going on is a bit slower than one would hope |
| 16:31.59 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:32.05 | anonymouz666 | 729 active calls? |
| 16:32.56 | anonymouz666 | that's nice for no trascoding |
| 16:33.39 | Qwell | I hear ctooley runs like 40 million concurrent channels on a single box :D |
| 16:33.48 | Qwell | give or take a few |
| 16:33.56 | Corydon76-dig | I heard 40 Billion |
| 16:34.30 | *** join/#asterisk aquarat (n=cowalski@196.211.186.114) |
| 16:34.35 | anonymouz666 | ctooley: can you run 10.000 active calls? then you can have a dinner with digium people |
| 16:34.45 | Corydon76-dig | Kajillion. or something like that |
| 16:34.47 | anonymouz666 | heh |
| 16:34.49 | rob0 | McD's stopped keeping count, now it's just "Billions and Billions Served". |
| 16:35.32 | ctooley | is attempting to see just how many calls he _can_ run on a single box. |
| 16:35.49 | Corydon76-dig | rob0: if only they could keep count of UNIQUE customers |
| 16:36.22 | Corydon76-dig | ... who have not yet had to have a triple coronary bypass |
| 16:37.03 | buzzyd | Does anyone know if meetme in dynamic mode can play the conference room number when user is joining? |
| 16:37.49 | Corydon76-dig | I don't believe meetme ever plays the conference room number |
| 16:38.11 | buzzyd | Can anyone rewrite it to do it :) |
| 16:38.21 | buzzyd | such a simple task surely |
| 16:38.55 | buzzyd | can't believe it doesn't do by default |
| 16:39.07 | Corydon76-dig | Why not just do a Playback prior to entering MeetMe? |
| 16:40.24 | buzzyd | I guess because I don't know how :) |
| 16:40.45 | rob0 | The Unix way, don't add irrelevant features to every application, when there is already a simpler application which can do it. |
| 16:40.50 | Corydon76-dig | Dialplan change... no code change required |
| 16:41.05 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 16:41.32 | buzzyd | something like play message please enter room number play it back then hand it to meetme app? |
| 16:41.55 | Corydon76-dig | Sounds fine here |
| 16:42.26 | buzzyd | I can do this without agi? |
| 16:42.32 | Corydon76-dig | Correct |
| 16:42.39 | Corydon76-dig | Read and Playback are all you need |
| 16:42.54 | buzzyd | cool off to read up on it thanks |
| 16:43.08 | rob0 | core show application playback |
| 16:43.52 | buzzyd | even better :) |
| 16:46.37 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:54.40 | *** join/#asterisk pzn (n=pzn@189.103.18.3) |
| 16:54.55 | pzn | Hi. I'm first time user |
| 16:55.03 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 16:55.45 | jameswf | pzn: Quick run!! |
| 16:56.51 | pzn | (sorry for hitting enter key wrongly in my previous post) I'm first time user of asterisk, can you point me to a reference about how to use asterisk to "convert" codecs? there will be only two sip clients, to communicate each other. one supports g711u and the other g729. |
| 16:57.40 | *** join/#asterisk neurosys (n=vinix@c-66-229-93-196.hsd1.fl.comcast.net) |
| 16:59.18 | pzn | the g711u works in the same ethernet as asterisk, so no problems. the other that is (g729 or g711) is in internet, I do not have satisfatory results using 711u on both due to the bandwidth of the device that is internet. So I need some reference to read about using asterisk as a codec converter. |
| 17:01.52 | mweichert | I'm trying to originate a call from one extension to another on the same system using Asterisk Manager API. I'm not sure what to use for a channel... I've tried using SIP/localhost/2206 but I get an error about a loop detected. I'm sure that I'm just missing something simple |
| 17:07.13 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 17:11.19 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096590412.dsl.bell.ca) |
| 17:12.02 | atis_work | mweichert: not localhost.. SIP/2206 (assuming there's entry for 2206 in sip.conf) and SIP/2207 - for another one |
| 17:12.59 | atis_work | oh, you probably have to send another part to some context in dialplan, or just use application Dial in originate |
| 17:15.44 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.108) |
| 17:21.53 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 17:22.15 | mweichert | SIP/2212 doesn't work: "Unable to request channel SIP/2212" |
| 17:22.37 | dbcooper1 | any have any experience with Sangoma's USB FXO device (u100)? I'm thinking of ordering one for testing - its like $125 or so... |
| 17:22.58 | dbcooper1 | anyone* |
| 17:23.12 | mweichert | aha, the Local channel is what I need :) |
| 17:23.51 | mweichert | hmm, maybe not. :/ |
| 17:24.00 | buzzyd | trying to check for varconf which is a read entry for * used GotoIf($[ ${varconf} = *]?11:1 |
| 17:24.33 | buzzyd | anyone spot my newbie mistake? |
| 17:25.01 | dbcooper1 | ) |
| 17:25.40 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 17:26.14 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:26.25 | buzzyd | tried with "*" that not work either |
| 17:26.44 | *** part/#asterisk dbcooper1 (n=User@64.203.244.146) |
| 17:27.35 | [TK]D-Fender | buzzyd: WHITESPACE |
| 17:28.20 | buzzyd | exten => s,10,GotoIf($[ ${confauth} = "*"]?11:1) |
| 17:28.37 | buzzyd | exten => s,10,GotoIf($[ ${confauth}="*"]?11:1) better? |
| 17:29.53 | ruben23 | hi where do i cehck log errors on my asterisk box |
| 17:30.10 | ruben23 | hi where do i check log errors on my asterisk box |
| 17:30.20 | thehar | uhm /var/log/asterisk has logs |
| 17:30.27 | buzzyd | on mine anyway |
| 17:30.50 | buzzyd | \/var/log/asterisk/ |
| 17:30.55 | buzzyd | mm |
| 17:31.30 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 17:32.13 | buzzyd | [TK]D-Fender neither work :( |
| 17:36.11 | *** join/#asterisk kc8pxy (n=gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
| 17:36.24 | mweichert | I'm trying now to originate a call from cli. i want to call extension 2216 and connect it to extension 2206. To do this, I'm trying: originate SIP/2212 extension 2206@from-internal-custom - but I still get "unable to request SIP/2216". Why would I be unable to request a channel? |
| 17:37.14 | kaldemar | buzzyd: use "" around the variable |
| 17:37.44 | mweichert | I have no active channels right now |
| 17:38.04 | file | mweichert: the originate like you just put above has SIP/2212, not SIP/2216 |
| 17:39.07 | buzzyd | exten => s,10,GotoIf($[ "${confauth}"="*"]?11:1) ? |
| 17:39.16 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-e9a044aae661b6f9) |
| 17:39.16 | mweichert | file, sorry, that's a typo - everything is SIP/2212 |
| 17:39.57 | [TK]D-Fender | buzzyd: WHITESPACE <----- |
| 17:40.04 | kaldemar | buzzyd: yes. but you still have one ugly space in there. |
| 17:40.22 | buzzyd | exten => s,10,GotoIf($["${confauth}"="*"]?11:1) ? |
| 17:40.29 | buzzyd | dananaaaaaah :) |
| 17:40.35 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 17:40.50 | buzzyd | * ducks but miss times it |
| 17:41.08 | kaldemar | the whitespace doesn't screw it up, it's just ugly. |
| 17:41.21 | buzzyd | haha fair enough |
| 17:41.36 | buzzyd | works though :) |
| 17:41.47 | [TK]D-Fender | kaldemar: Whitespace does screw stuff up |
| 17:42.11 | kaldemar | [TK]D-Fender: not anymore. :) |
| 17:42.33 | mweichert | file, if I use Local/2212, things look a lot different: http://pastie.org/485580 |
| 17:43.49 | buzzyd | I have whitespace in my other checks and it works also worked with this one it was "" around the Var that sorted it, but I know nothing about these things :) |
| 17:52.26 | *** join/#asterisk swiftkick (i=eeeeeeee@mail.beanproducts.com) |
| 17:52.49 | buzzyd | Thanks for you help all loving asterisk just wish I had more time to play :) |
| 17:53.03 | *** part/#asterisk buzzyd (n=psp-man@86.54.239.113) |
| 17:54.16 | swiftkick | hi, is there an easy way to capture the debug output of "sip set debug peer ip x.x.x.x" to a file ??? |
| 17:56.04 | *** join/#asterisk Alborracho (n=chatzill@190.25.135.1) |
| 17:56.12 | Alborracho | hi everyone |
| 17:56.36 | swiftkick | ive got some chump who guessed a password for one of our SIP peers and is using it with an autodialer to attempt credit card fraud |
| 17:56.39 | Alborracho | im getting this message in a sip call "Non-codec capabilities (dtmf)" can anyone give some clue about this error? |
| 17:57.21 | Alborracho | im using g729 |
| 18:00.04 | [TK]D-Fender | Alborracho: Who says that is an ERROR? |
| 18:00.34 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 18:00.39 | Alborracho | my telco company because we cannot recognize DTMF in rfc2833 |
| 18:00.49 | KyleK | swiftkick: i thought file was an option in the setting of debug |
| 18:00.51 | Alborracho | i was looking in the sip messages and thi called my attention |
| 18:01.18 | swiftkick | KyleK i dont see it under "help sip" |
| 18:01.28 | [TK]D-Fender | Alborracho: perhaps you should pastebin a complete call for use to look at and your configs |
| 18:01.33 | [TK]D-Fender | ~pb |
| 18:01.34 | 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/ |
| 18:01.35 | [TK]D-Fender | ^^^^^66 |
| 18:01.36 | jayrod422 | can anyone tell how to fix a problem where asterisk 1.4.23 crashes randomly until moderate load (30 calls with media from softphones behind nats).. it just dies - nothing in the logs |
| 18:01.38 | Alborracho | ok |
| 18:01.56 | KyleK | swiftkick: might be better off running tcpdump with a filter for that IP address |
| 18:01.56 | swiftkick | KyleK and asterisk doesnt support shell style file redirection |
| 18:02.04 | swiftkick | kylek good idea. thanks |
| 18:02.43 | Alborracho | [TK]D-Fender: here it is http://pastebin.ca/1430128 |
| 18:02.47 | [TK]D-Fender | jayrod422: That provides no usable details and is not on the current version. |
| 18:03.16 | jayrod422 | im kind of afraid to go to 1.6 |
| 18:03.18 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 18:03.19 | [TK]D-Fender | Alborracho: CONFIGS please |
| 18:03.27 | [TK]D-Fender | jayrod422: I never said 1.6 |
| 18:03.30 | Alborracho | ok |
| 18:03.36 | KyleK | jayrod422: i think he means 1.4.24.2 or something close to that |
| 18:03.51 | jayrod422 | im on 1.4.23.1 |
| 18:04.01 | jayrod422 | so update first then come ask questions |
| 18:04.08 | timeshell_atwork | ~tk |
| 18:04.09 | infobot | ACTION snipes $herlo with a straw and rolled up piece of paper |
| 18:04.32 | KyleK | jayrod422: well if its random, it might have been a screwy problem that was fixed |
| 18:05.04 | jayrod422 | true |
| 18:05.21 | jayrod422 | i only have this issue on boxes with all remote softphones |
| 18:06.21 | Alborracho | [TK]D-Fender: http://pastebin.ca/1430134 it has sip.conf extensions.conf and the sip trace |
| 18:07.48 | [TK]D-Fender | Alborracho: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) |
| 18:07.58 | KyleK | jayrod422: best i can suggest with my noob knowledge is to try and monitor CPU and memory usage before and during a crash |
| 18:08.07 | jayrod422 | its always low |
| 18:08.26 | [TK]D-Fender | Alborracho: They don't seem to offer rfc2833, you aren't even match a peer entry, and DTMF over inband RTP on G.729 = FAIL |
| 18:08.31 | jayrod422 | is there a way to enable a crash log or something |
| 18:08.33 | [TK]D-Fender | Alborracho: I recommend switching to ALAW |
| 18:09.34 | Alborracho | they don want to use g711 because of bandwith |
| 18:09.42 | [TK]D-Fender | Alborracho: they are offering it. |
| 18:09.52 | [TK]D-Fender | Alborracho: I recommend you accept |
| 18:10.02 | Alborracho | yes we are in test now, but in production they want to use only g729 |
| 18:10.03 | [TK]D-Fender | Alborracho: because otherwise I don't see DTMF coming over... |
| 18:10.13 | [TK]D-Fender | Alborracho: It is possible they are using SIP-INFO for this. |
| 18:10.23 | [TK]D-Fender | Alborracho: Somthing you can try at least |
| 18:10.29 | Alborracho | the thing here is that they are not offering rfc2833 right? |
| 18:10.38 | [TK]D-Fender | Alborracho: debug seems to say so |
| 18:10.47 | Alborracho | ive tried info and doesnt work |
| 18:10.55 | Alborracho | ok thanbks fender |
| 18:11.09 | [TK]D-Fender | Alborracho: Whats on the other side? |
| 18:11.41 | Alborracho | a nokia siemens networks |
| 18:11.54 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 18:12.01 | Alborracho | wich connects to the cellphones network 3gsm |
| 18:12.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:12.11 | [TK]D-Fender | Alborracho: What you could do is put another device at the remote side to act as a gateway |
| 18:12.40 | [TK]D-Fender | Alborracho: and have that run ALAW + inband, and convert at which point you can even use IAX2 trunking to save even more bandwidth |
| 18:12.54 | [TK]D-Fender | Alborracho: If you run multiple simultaneous channels |
| 18:13.07 | Alborracho | ok i see, ill talk to them about this, |
| 18:13.21 | Alborracho | what should i get from them to use rfc2833? |
| 18:14.44 | [TK]D-Fender | Alborracho: get from them? they don't aoffer it. If there is an option to configure their PBX for this I don't know it |
| 18:15.11 | [TK]D-Fender | Alborracho: But I'm saying the inband+alaw will work and you can use a local * box to act as a midway point for your connection |
| 18:15.14 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 18:16.21 | Alborracho | my * box is inside the telco datacenter im connected directly to their platform |
| 18:16.37 | [TK]D-Fender | Alborracho: "Sucks to be you" |
| 18:16.45 | Alborracho | but they refuse to use g711 |
| 18:17.07 | [TK]D-Fender | Alborracho: tell them to fix it then. |
| 18:19.33 | Alborracho | ok thanks Fender |
| 18:20.46 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:24.51 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 18:29.02 | *** join/#asterisk stijnbe (n=stijnbe@78-22-110-114.access.telenet.be) |
| 18:32.00 | *** part/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 18:37.00 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 18:38.32 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 18:38.52 | neurosys | Are the Grandstream SIP Gateways any good? |
| 18:39.38 | [TK]D-Fender | ~gs |
| 18:39.39 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:40.17 | neurosys | ouch. ok.. any recommendations? I need to provide a sip gateway for 26 analog phones. |
| 18:41.16 | ariel_ | are all analog phones at the same location? |
| 18:41.48 | neurosys | Spread out over 3 buildings. It's a small resort. |
| 18:41.51 | *** join/#asterisk intralanman (n=intralan@99-207-161-255.pools.spcsdns.net) |
| 18:42.43 | [TK]D-Fender | neurosys: Mediatrix 4100 series |
| 18:44.12 | neurosys | [TK]D-Fender: Sorry, but did you once tell me voip-supply was one of the better shops? |
| 18:44.28 | [TK]D-Fender | neurosys: God selection, iffy price |
| 18:45.07 | *** join/#asterisk plq (n=plq@88.250.169.4) |
| 18:45.07 | carrar | TK, how does Mediatrix stack up to AudioCodes? |
| 18:45.35 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:46.20 | *** join/#asterisk [acer]lanman (n=intralan@70-10-82-254.pools.spcsdns.net) |
| 18:46.55 | carrar | Looks like they might be cheaper |
| 18:49.41 | neurosys | Thx TK - good stuff :) |
| 18:54.54 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 18:55.27 | jplank | dont forget how annoying voip-supply gets once they get your email |
| 18:57.50 | mmlj4 | forget? that'll never happen |
| 18:57.51 | mmlj4 | :-( |
| 18:57.58 | SuPrSluG | hello |
| 18:58.53 | *** join/#asterisk ctp (n=ctp@brsg-d9bef7f3.pool.mediaWays.net) |
| 18:59.51 | SuPrSluG | will chanspy work after a call is transferred or does it like monitor and end when the channel ends? |
| 19:01.08 | *** join/#asterisk ctp (n=ctp@brsg-d9bef7f3.pool.mediaWays.net) |
| 19:02.13 | jplank | i'm pretty sure CHANspy that particular channel. If you transfer the call, your probably lose that channel....well unless your monitoring the channel it came in on I guess |
| 19:02.58 | jplank | but if your using chanspy to monitor an extension, and that extension transfers the call to a different extension, you'll lose the monitor |
| 19:03.15 | swiftkick | did cisco or asterisk ever devise a way to select alternate ringtones on cisco 79x0's using the Alert-Info header? (besides the <bellcore> ringtones) ?? |
| 19:05.22 | SuPrSluG | jplank, that's what I figured,thx |
| 19:05.41 | swiftkick | i came up with a quick hack to give me exactly one alternate ringtone on the 79x0's: set the alternate ringtone desired on the phone. then create a "local" extension that adds the Alert-Info: to change to a bellcore ringtone. |
| 19:06.23 | swiftkick | presto: two different ringtones. |
| 19:06.32 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 19:06.41 | swiftkick | but still no way for asterisk to invoke any of the 20 other ringtones the phone grabs from tftp |
| 19:15.12 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-146-113.telkomadsl.co.za) |
| 19:18.59 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 19:20.25 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
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| 19:22.34 | *** part/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 19:31.33 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 19:31.52 | *** join/#asterisk Tene_ (n=sweeks@poipu/supporter/slacker/tene) |
| 19:32.06 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:33.09 | Tene_ | I'm having a lot of trouble with using imap storage for voicemail. Specifically, I can't figure out how to configure it to speak ssl. |
| 19:33.34 | Tene_ | I can set imapport to 993, but afaict, it's still trying to speak IMAP, not SSL. |
| 19:34.00 | Tene_ | ldd on app_directory_imap.so shows that it is linked against libssl |
| 19:34.10 | Tene_ | so it was properly compiled with ssl support, afaict. |
| 19:35.54 | dni | Hello., I was hoping someone could help me with a musicOnHold issue im having,. I keep on getting |
| 19:35.54 | dni | [May 21 15:33:18] NOTICE[3649]: res_musiconhold.c:542 monmp3thread: Request to schedule in the past?!?! |
| 19:35.54 | dni | <PROTECTED> |
| 19:35.58 | j_kroon | you could "fake" it using stunnel. |
| 19:36.28 | Tene_ | ... ouch. |
| 19:36.30 | Tene_ | Maybe. |
| 19:36.31 | j_kroon | @ Tene_ |
| 19:36.43 | j_kroon | agreed :) |
| 19:36.52 | Tene_ | I'll keep that as a backup plan. |
| 19:37.18 | KyleK | will Background() let me press * while its playing the message and then jump somewhere? (like to extension "a") |
| 19:37.56 | KyleK | dni: do the mp3's have a constant bitrate or variable? |
| 19:38.08 | j_kroon | KyleK, yes. |
| 19:38.32 | j_kroon | but it'll jump to extension * |
| 19:38.36 | [TK]D-Fender | dni: Stop using mpg123 |
| 19:38.43 | dni | KyleK, im trying to stream somaFM application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -s --mono http://streamer-dtc-aa05.somafm.com:80/stream/1018 |
| 19:39.00 | dni | [TK]D-Fender, can i do the same with format_mp3 fromt he addonn's package ? |
| 19:39.09 | [TK]D-Fender | dni: Not streaming |
| 19:39.15 | dni | hmm |
| 19:39.21 | dni | ok |
| 19:39.41 | *** join/#asterisk intralanman (n=intralan@173-133-91-66.pools.spcsdns.net) |
| 19:42.35 | j_kroon | hmm, i'm sending Originate requests via the web manager api, specifically I'm passing /rawman?action=Originate&Channel=SIP%2F199&Exten=0126624479&Context=callorigination&Priority=1 ... but get an error "Channel not specified" - this feels like a typo but I can't spot it. |
| 19:44.37 | KyleK | dni: maybe stream to a temp file and play the file? |
| 19:50.52 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 19:54.25 | [TK]D-Fender | dni: That is normally just a warning. Does it pose an actual problem for you somewhere |
| 19:54.27 | [TK]D-Fender | ? |
| 19:54.42 | dni | [TK]D-Fender, i dont hear any audio when i place the current call on hold |
| 19:54.50 | dni | but i can open it fine with a web browser |
| 19:55.05 | [TK]D-Fender | dni: That would be a problem... madplay is usually recommended for this thing... give it a shot |
| 19:55.23 | dni | ok will do, thanks |
| 19:55.35 | neurosys | [TK]D-Fender: You use anyone in particular for your voip gear? |
| 19:56.05 | [TK]D-Fender | neurosys: www.telephonydepot.com - usually |
| 19:56.19 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:57.01 | neurosys | [TK]D-Fender: At the very least the site doesnt look like my 5 year old designed it :P |
| 19:57.14 | Tene_ | ah, looks like it's a problem with my c-client |
| 19:57.15 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 19:57.57 | j_kroon | https://issues.asterisk.org/view.php?id=14208 *sigh* |
| 19:58.11 | j_kroon | fortunately it has been fixed :) |
| 19:59.54 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 20:01.21 | *** join/#asterisk grndslm (n=grndslm@96.19.110.120) |
| 20:02.30 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:02.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:03.07 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.0 (2009/04/28), Asterisk 1.6.0.9 (2009/04/06), 1.4.25 (2009/05/21), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 20:03.19 | leifmadsen | Asterisk 1.4.25, and Asterisk-Addons 1.6.0.2 are now available for download! |
| 20:05.41 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 20:05.56 | watchy | anyway to get the directory to announce the name with the owners voice? |
| 20:09.40 | [TK]D-Fender | watchy: Yes, tell them to RECORD it |
| 20:09.41 | Katty | HAI WATCHY! |
| 20:10.47 | riddlebox | Katty, whats up |
| 20:11.19 | Katty | riddlebox: umm. opposite of down? |
| 20:14.04 | eppigy | hi |
| 20:14.11 | eppigy | so sleepy |
| 20:14.22 | eppigy | my sleep schedule is so mess up |
| 20:14.31 | Katty | hello, sleepybones. |
| 20:14.35 | eppigy | yesh |
| 20:14.46 | Katty | how did your schedule get messed up |
| 20:14.51 | eppigy | well |
| 20:14.57 | eppigy | I was at work late one day |
| 20:15.01 | Katty | :< |
| 20:15.06 | eppigy | and the next morning I was going to wake up late |
| 20:15.12 | eppigy | and someone called and woke me up |
| 20:15.18 | eppigy | so when I went home from work that day |
| 20:15.25 | eppigy | I fell asleep at 6;30PM |
| 20:15.37 | eppigy | and it has been screwy since |
| 20:15.41 | Katty | :< |
| 20:15.45 | Katty | weekend recovery? |
| 20:17.04 | eppigy | well |
| 20:17.05 | eppigy | :[ |
| 20:17.13 | eppigy | I promised my dad I would visit them |
| 20:17.16 | eppigy | in jacksonville |
| 20:17.31 | eppigy | we will see |
| 20:20.20 | watchy | HAI KATTY! |
| 20:20.32 | watchy | want 2 date? |
| 20:20.41 | Nugget | huggles Katty |
| 20:22.34 | neurosys | Hiya Katty |
| 20:22.42 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 20:27.55 | [TK]D-Fender | checkout time, BBIAB |
| 20:32.39 | Katty | watchy: i'm already dating someone, thanks. |
| 20:32.42 | Katty | hugs Nugget |
| 20:32.44 | Katty | neurosys: hello. |
| 20:33.25 | leifmadsen | Katty: what... we can't wife swap? :) |
| 20:33.55 | Aiatek | can i install dahdj without internet access? |
| 20:34.09 | Aiatek | dahdi |
| 20:35.39 | beek | Aiatek: assuming that you can get the tarball, yes. |
| 20:37.01 | *** part/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 20:40.34 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 20:41.25 | Tene_ | Okay, still having problems with imap voicemail. I updated c-client and recompiled asterisk... but still getting "connection timed out", which is a lie |
| 20:41.33 | Tene_ | I can connect just fine with openssl s_client |
| 20:41.51 | Tene_ | and I ran a packet capture, and the server *is* responding. |
| 20:42.21 | Tene_ | I can connect just fine on port 143, but not on 993 |
| 20:52.03 | Tene_ | the outbound packets look like they're not ssl-encrypted. |
| 20:52.16 | Tene_ | i get the same packets whether i'm connecting to port 993 or port 143 |
| 20:53.34 | florz | why should that depend on the port?! |
| 20:54.14 | Tene_ | florz, because I can't find any configuration options to ask it to speak ssl |
| 20:55.00 | florz | then, maybe, it doesn't speak SSL? |
| 20:55.41 | Tene_ | 1) that would be horrible. 2) the docs talk about how to compile asterisk to make sure it supports SSL. |
| 20:55.59 | florz | well, if it does support SSL, then maybe it uses STARTTLS? |
| 20:56.50 | Tene_ | It doesn't. :( It's being rejected from the IMAP server because the server refuses plaintext auth when not inside of SSL. |
| 20:57.21 | *** join/#asterisk yjtmas (n=nux@host59-183-dynamic.6-79-r.retail.telecomitalia.it) |
| 20:57.27 | yjtmas | hi all |
| 20:57.40 | *** join/#asterisk SebastianS (n=schu@adsl-dyn58.78-98-36.t-com.sk) |
| 20:59.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:03.19 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:07.05 | *** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net) |
| 21:07.41 | *** join/#asterisk ToTo (n=ToTo@212.118.143.91) |
| 21:08.12 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 21:08.30 | florz | Tene_: well, that's rather bad, then =:-) |
| 21:08.59 | [TK]D-Fender | ToTo: I don't think you're in Kansas anymore.... |
| 21:09.24 | Tene_ | florz: looks like it actually is using tls, and I just can't read. |
| 21:09.25 | Tene_ | :) |
| 21:10.18 | florz | IC =:-) |
| 21:10.29 | hardwire | Added extension '_1888220xxxx' priority 1 (CID match '_+1NXXNXXXX') to inbound |
| 21:10.39 | hardwire | will it match the '+' char in the callerid num? |
| 21:10.48 | florz | Tene_: well, you not being able to read is the point of TLS ;-) |
| 21:11.02 | Tene_ | lol ;) |
| 21:11.12 | [TK]D-Fender | hardwire: yes |
| 21:12.57 | watchy | katty: your making me sad and i don't know if i want to live without you |
| 21:13.51 | watchy | I'm thinking about buying a puppy, what should i get |
| 21:15.19 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 21:17.11 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-ee917ce98e477329) |
| 21:22.43 | hardwire | [TK]D-Fender: moo.. thanks. |
| 21:28.35 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:33.38 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 21:36.55 | *** join/#asterisk lost_soul (n=shawn@cpe-67-241-68-104.twcny.res.rr.com) |
| 21:37.08 | *** join/#asterisk Ziaeon (n=ziaeon@75-149-177-2-Miami.hfc.comcastbusiness.net) |
| 21:37.37 | Ziaeon | Hey, someone mentioned a dial code that lead them to a prompt where they could put an extension and listen in on the convo, kinda like the FOP panel but on the phone. Hows this done? |
| 21:39.51 | *** join/#asterisk rvhi (n=chatzill@207.2.110.6) |
| 21:40.34 | rvhi | hi, in queue, is there a way to let agent press a button before allowing to take a call? |
| 21:40.58 | rvhi | e.g. if the agent's voicemail kicks in, i don't want the system think that it is a person answering |
| 21:44.15 | Ziaeon | 555, gotcha |
| 21:44.17 | yjtmas | hi |
| 21:44.18 | *** part/#asterisk Ziaeon (n=ziaeon@75-149-177-2-Miami.hfc.comcastbusiness.net) |
| 21:44.27 | yjtmas | I'm using asteriskgui 2.0 |
| 21:45.04 | yjtmas | with asterisk 1.6.0.9, they are compatible? |
| 21:52.01 | *** join/#asterisk n3hxs (n=HAMming@pool-173-66-115-234.washdc.fios.verizon.net) |
| 21:52.41 | Katty | AHHHHHHHHHHHH!!!!!!!!!!!!!!!!!!!!! |
| 21:52.45 | Katty | asplodes. |
| 21:57.42 | Pan3D | ducks |
| 22:01.07 | *** join/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 22:01.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:01.59 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 22:03.32 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 22:05.05 | Optic | i have a no-sip-audio problem :( |
| 22:07.55 | Optic | but I don't have NAT, so that might make it a bit easier ;) |
| 22:08.18 | bkw_ | sounds liek firewall |
| 22:08.22 | bkw_ | cuz audio is over RTP not sip :) |
| 22:08.34 | Optic | yeah, i hate iptables |
| 22:08.45 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 22:09.44 | Optic | -A RH-Firewall-1-INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT |
| 22:09.44 | Optic | i figured that would do it |
| 22:09.52 | Optic | my sip call is answered fine, i put it in the demo context |
| 22:09.56 | Optic | it tries to play the demo greeting just fine, but I don't hear it ;) |
| 22:13.27 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:18.51 | Optic | from tcpdump it looks like i'm sending rtp properly |
| 22:18.54 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:18.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:19.11 | Optic | 18:16:56.883871 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 200) xxxxx.privatedns.com.15852 > sip.ca2.link2voip.com.25926: UDP, length 172 |
| 22:19.41 | jameswf | Qwell: any reason you dont have a raid KS |
| 22:21.33 | Qwell | nobody has asked for it |
| 22:21.47 | Qwell | but, partitioning is all manual, can't you choose raid options in there? |
| 22:24.12 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:31.44 | Optic | any hints? :( |
| 22:34.11 | Optic | Sent RTP packet to 66.51.110.210:25996 (type 00, seq 010517, ts 062400, len 000160) |
| 22:34.11 | Optic | Got RTP packet from 66.51.127.163:27192 (type 00, seq 030629, ts 061040, len 000160) |
| 22:43.06 | jameswf | manual raid is a pita... I like to do as little as possible durring the OS install.. |
| 22:43.18 | Optic | it is so strange, it looks like rtp is getting through just fine |
| 22:44.00 | Optic | unless the remote isn't accepting my rtp packets, which I guess contain the audio i'm sending |
| 22:46.22 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 22:49.04 | hardwire | is anybody here like.. sitting next to a payphone? |
| 22:49.29 | Chainsaw | DECT phone, SIP phone, cellphone... *looks around further* |
| 22:49.32 | Chainsaw | Nope. Sorry. |
| 22:51.30 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 22:53.07 | hardwire | chainsaw: me too! |
| 22:53.11 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 22:53.24 | Optic | is it normal for the rtp source and destination to be different ips? |
| 22:53.37 | hesco | how do I get my .call file generated calls to show up in the cdr? |
| 22:54.26 | hardwire | hesco: they are no different than other calls. |
| 22:54.33 | hardwire | if I'm not totally mistaken. |
| 22:55.46 | hesco | incoming calls and calls generated from a handset seem to show up in the cdr table, but the .call file generated ones don't seem to |
| 22:56.18 | hardwire | how are they different as far as asterisk is concerned? |
| 22:56.43 | hesco | I guess that is what I'm trying to figure out. |
| 22:56.50 | hardwire | it's calling a phone on behalf of the system and dispatching it to a context/extension. |
| 22:56.59 | hesco | running yet another test |
| 22:57.01 | hardwire | are you turning off CDR anywhere? |
| 23:01.16 | hesco | ok, just sent a call through the system, the incoming call was recorded but not the outgoing call |
| 23:01.43 | hardwire | well.. watching the debug log may help |
| 23:01.48 | hesco | how is it I might have turned off CDR? |
| 23:02.00 | hesco | how do I access the debug log? |
| 23:02.43 | Corydon76-dig | Turn it on in logger.conf, logger reload, and core set debug 4 |
| 23:03.08 | *** join/#asterisk vvuja (n=vvuja@79-175-71-114.adsl-a-1.sezampro.yu) |
| 23:07.34 | Optic | is frustrated |
| 23:07.48 | hardwire | punches you in the arm. |
| 23:07.48 | Optic | asterisk usually works fine for me |
| 23:09.26 | vvuja | hello |
| 23:10.10 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:12.21 | *** part/#asterisk vvuja (n=vvuja@79-175-71-114.adsl-a-1.sezampro.yu) |
| 23:15.24 | *** join/#asterisk colinm_ (n=colinm@174-17-23-165.phnx.qwest.net) |
| 23:16.12 | hesco | well, that hasn't shown me the issue with CDR yet, but it is showing up other issues already |
| 23:16.29 | Optic | gives up for now ;( |
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| 23:27.30 | orpheee | hi |
| 23:36.03 | orpheee | hello i've a problem with my conference and meetme you can the configuration and error there http://pastebin.com/d4c91be1e |
| 23:36.42 | orpheee | thks for help |
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| 23:43.55 | telnettech | <PROTECTED> |
| 23:44.00 | telnettech | sorry |
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| 23:53.08 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |