| 00:00.04 | sbrath | the thing is if it's just a simple setup, a freepbx distro is ok, but when you want more features your kinda boxed in. |
| 00:00.25 | Kobaz | it would be nice if emacs wasn't written almost entirely in lisp... it's quite slow for some things |
| 00:00.30 | bobisa | ok because im new in this world and i want to know what are my option |
| 00:00.47 | ChannelZ | nano nano |
| 00:00.50 | sbrath | most of the freepbx distros have limitations, that require you to Buy a higher version to get more than like 1 queue. or more than x lines. |
| 00:00.58 | Kobaz | ChannelZ: it used to be pico |
| 00:00.58 | Slugs_ | just start w/ asterisk and edit cfg's would be my advice |
| 00:01.12 | ChannelZ | yes and I still call it pico |
| 00:01.22 | ChannelZ | pico -w xxxx is a force of habit |
| 00:01.28 | Kobaz | back when everyone used pine |
| 00:01.33 | Slugs_ | ohh pine |
| 00:01.35 | bobisa | there is some site with good tutorial ? |
| 00:01.42 | vader-- | anyone in here happen to work with adtran units? looking for configuration samples on getting it to work with asterisk |
| 00:01.46 | ChannelZ | vi makes me sad inside |
| 00:01.49 | Kobaz | i think it was silly that the pine people didn't want to split pico off into a seperate project |
| 00:01.49 | vader-- | i have an Adtran TA924e |
| 00:01.54 | Kobaz | hence nano spawned |
| 00:01.55 | Slugs_ | bobisa, what distro |
| 00:01.59 | vader-- | alittle confused with how to configure it |
| 00:02.27 | bobisa | asterisk now |
| 00:02.38 | Slugs_ | k.. |
| 00:02.40 | Kobaz | and also some licensing issues |
| 00:02.56 | Slugs_ | so its all up, u just need to configure it? |
| 00:02.57 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.02 | bobisa | yes |
| 00:03.05 | Slugs_ | k |
| 00:03.15 | ChannelZ | if you JUST want a simple PBX and don't want to do much of anything cool, use freepbx.. |
| 00:03.20 | Kobaz | vader--: adtran makes a whole boatload of stuff... you'll have to be more specific |
| 00:03.34 | bobisa | and before buying some exta stuff i want to cofigure it with x-lite |
| 00:03.48 | Slugs_ | http://cdn.oreilly.com/books/9780596510480.pdf |
| 00:03.53 | Slugs_ | there you go |
| 00:03.58 | *** join/#asterisk trentcreek (~kvirc@red1.cs.panam.edu) |
| 00:04.04 | vader-- | TA924e voip gateway |
| 00:04.12 | ChannelZ | ~book |
| 00:04.12 | 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 |
| 00:04.13 | vader-- | im diagraming in visio what i want it to do right now |
| 00:04.22 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 00:04.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:04.29 | ChannelZ | Slugs_: infobot is our bitch |
| 00:04.34 | Slugs_ | thx |
| 00:04.35 | Slugs_ | ; |
| 00:04.36 | Slugs_ | ;) |
| 00:05.37 | bobisa | thanks |
| 00:05.37 | Slugs_ | slaps infobot |
| 00:06.39 | Kobaz | vader--: you'll likly have to dig up your manual, set the sip credentials on it and asterisk |
| 00:07.17 | vader-- | well i think the way i want to use the TA924e is different than the way most use it |
| 00:07.23 | ruben23 | hi |
| 00:07.27 | bobisa | what are difference between distribution |
| 00:07.52 | Slugs_ | bobisa, what now? |
| 00:08.17 | Slugs_ | asterisk distros? |
| 00:08.21 | bobisa | before you ask me what disribution that i have |
| 00:08.47 | ChannelZ | The difference is mostly preference |
| 00:08.58 | bobisa | ?? |
| 00:09.03 | ChannelZ | Run whatever the hell you want |
| 00:09.25 | *** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk) |
| 00:09.46 | ruben23 | hi setup DID # for incoming call, DID is register to the carreir but when i test incoming calls i get this error-----> http://pastebin.com/Wmvcafwq |
| 00:10.21 | ChannelZ | you have some mess there |
| 00:12.04 | ChannelZ | their auth doesn't match your config and I dunno what the UNAVAILABLE is all about |
| 00:13.23 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 00:14.00 | sbrath | bobisa: they are all basicly FreePBX + extras. Like trixbox charges for more features and is available using asterisk 1.6.x and PBX In A Box is also available with asterisk 1.6 and some extras. My preference would be PBX in a box as it also has a firewall configured by default. |
| 00:19.08 | hardwire | The proper use of figlet |
| 00:19.09 | hardwire | http://i.imgur.com/pbopP.jpg |
| 00:20.34 | vader-- | kobaz this is what i want to do |
| 00:20.35 | vader-- | http://tinypic.com/r/s3hx4y/5 |
| 00:21.02 | vader-- | I want all my FXS channels for analog phones, fax machines, etc go into the TA924e |
| 00:21.10 | vader-- | i want the PRI line from the telco to come into it |
| 00:21.28 | vader-- | and then have the TA924e connect to my internal lan and present that to the Asterisk box |
| 00:21.55 | vader-- | so if a call comes in from the PRI it can be sent to either the cisco IP phones or to an FXS channel |
| 00:22.11 | vader-- | and vice versa if a call is initiated from the inside it can go to the PRI line |
| 00:33.45 | sbrath | vader--: Good luck :) |
| 00:34.50 | sbrath | as long as the ta924e can establish a SIP trunk connection to the asterisk should be possible, but will take a fair amount of SIP routing. |
| 00:35.09 | sbrath | why didn't you just get a T1 card and a FXS card for the asterisk box? |
| 00:37.16 | Corydon76-dig | He felt he wasn't spending enough cash... |
| 00:37.27 | *** join/#asterisk lmsteffan (~laurent@reef.ac-noumea.nc) |
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| 00:41.59 | vader-- | i don't want any hardware tied to the asterisk box |
| 00:42.08 | vader-- | it's going to run in a VM |
| 00:44.02 | sbrath | I saw your irc log from back in january asking about the adtran |
| 00:44.55 | vader-- | ya the engineers at adtran said it could be done |
| 00:45.06 | sbrath | sorry, I've never setup the adtran for anything other than a V.35 T1 inbound connection. |
| 00:45.42 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 00:45.44 | sbrath | I can also turn my van into a hovercraft, but it'll take some time and hard work installing the fans :) |
| 00:46.42 | sbrath | I'd start with just configuring the adtran as a SIP server, since you have to get the PRI into a device that can route the calls to a SIP trunk |
| 00:47.06 | sbrath | start with the simple stuff, make the adtran work without the Asterisk. |
| 00:48.04 | vader-- | well what i wanted to do first was try and get an IP phone internally to call a FXS line |
| 00:48.25 | sbrath | Since the FXS ports on the Adtran will have to route calls to the local devices, and the dialplan/routing will need to be smart enough to route local extensions to the FXS ports or to the SIP trunk into the asterisk. |
| 00:48.25 | vader-- | i can setup all the FXS ports |
| 00:48.35 | doctorray | dumb question.. how do I disable the macro-stdexten? It's not in any of my config/dialplan files |
| 00:48.41 | sbrath | can you setup a SIP trunk on the adtran ? |
| 00:48.44 | doctorray | it's taking priority over my local context |
| 00:49.20 | vader-- | you can setup Trunk Groups and Trunk Accounts |
| 00:49.30 | Corydon76-dig | doctorray: First, uninstall FreePBX |
| 00:49.42 | vader-- | right now i have a Trunkgroup - SIP and Trunkgroup - PRI |
| 00:49.57 | doctorray | Corydon76-dig: never installed; setup asterisk from source |
| 00:50.01 | vader-- | Trunkgroup - SIP has a trunk account called Trunk - SIP |
| 00:53.14 | doctorray | I'll rephrase. app_macro is taking over my local extension dialing and looking for macro-stdexten for macro stdexten, which doesn't exist |
| 00:53.21 | doctorray | unload app_macro? |
| 00:54.51 | doctorray | ...which makes it say "no application 'macro' ..." |
| 00:56.41 | doctorray | aha, the pbx_extension_helper |
| 00:56.59 | doctorray | now to figure out how to modify that around.. |
| 00:59.25 | hardwire | any special way of including special extensions like i,t,h into another context from another context? |
| 01:10.54 | VoIP-Penguin | exten => i,1,Goto(other,i,1) perhaps? |
| 01:11.10 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:12.07 | *** join/#asterisk moy (~moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:16.11 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 01:18.46 | trentcreek | Is there anything to do to make it so I can see what an AGI acript is doing in the AMI? I have "agi set debug on" and debugging and other gooddies turned on, but I only am seeing it LOG ON and LOG OFF |
| 01:19.29 | Beave | I have a old phone with a iaxy.. pulse works fine, yadda yadda.. does anyone remember if the iaxy had enough power to driver the ringer? like on a old Western Electric phone? |
| 01:20.37 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 01:23.02 | hardwire | VoIP-Penguin: that might have to do |
| 01:24.35 | *** join/#asterisk nickaugust (~anonymous@167.83.189.72.cfl.res.rr.com) |
| 01:36.35 | *** join/#asterisk keith4_ (~keith@unaffiliated/keith4) |
| 01:46.17 | *** join/#asterisk thecardsmith (~doug@65-183-130-234-dhcp.burlingtontelecom.net) |
| 01:47.02 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 01:49.23 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 01:50.01 | aceio | how can i tell asterisk to bar international calls |
| 01:50.15 | *** join/#asterisk epaphus (~name@201.199.62.74) |
| 01:50.34 | VoIP-Penguin | Don't create an extension for them. |
| 01:51.03 | VoIP-Penguin | Or specifically create an extension for them that does not dial out. |
| 01:52.27 | epaphus | Hello. Iam using ekiga/twinkle with integrated sound cards analogue headsets on different PCs running ubuntu. Codec is GSM and PCMU.. and sound prefs are set to max. Even like this sound is very low.. for me and for the other party that hears me. Skype works crystal clear. Also using USB headsets improves everything a lot. Is there anything that I can do on the asterisk level? |
| 01:52.53 | aceio | VoIP-Penguin: okay cheers, and it that simple |
| 01:55.04 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 01:55.56 | Katty | hi |
| 01:59.20 | *** part/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 01:59.40 | jaytee | hi |
| 02:00.08 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 02:00.16 | *** join/#asterisk epaphus (~name@201.199.62.74) |
| 02:00.24 | norrec | how do you change the xmit speed for the app_fax mod? |
| 02:00.27 | norrec | module* |
| 02:00.42 | epaphus | sorry i got disconnected... |
| 02:00.44 | epaphus | did anybody reply? :) |
| 02:03.13 | hardwire | jaytee said: hi |
| 02:03.53 | jaytee | yes, but I was saying hi to Katty's hi |
| 02:10.46 | ChannelZ | I completely solved your problem, but I guess now you'll never know. |
| 02:13.02 | keith4_ | if I want to set up a console channel on another box, what are my options? I'd like to avoid installing asterisk on it. Maybe some sort of network sound daemon? |
| 02:13.56 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 02:14.09 | Slugs_ | . |
| 02:14.11 | keith4_ | can * open multiple alsa console channels? like, if I had two sound cards? |
| 02:14.33 | Slugs_ | with wget how can i grab all files in a dir? |
| 02:14.59 | keith4_ | what does that have to do with asterisk? |
| 02:15.02 | keith4_ | read the wget man page |
| 02:17.07 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 02:20.43 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 02:31.14 | *** join/#asterisk jmcdowell (~nooe@173-140-28-98.pools.spcsdns.net) |
| 02:34.08 | ChannelZ | there's -r for recursive |
| 02:37.06 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 02:51.08 | trentcreek | Is there anything to do to make it so I can see what an AGI acript is doing in the AMI? I have "agi set debug on" and debugging and other gooddies turned on, but I only am seeing it LOG ON and LOG OFF |
| 02:54.10 | ChannelZ | AGI and AMI are two different things |
| 02:54.28 | ChannelZ | see 'manager debug' |
| 03:03.52 | trentcreek | ChannelZ: yes, I turned on the agi debugger |
| 03:04.30 | trentcreek | It only shows the script logging in an dout |
| 03:04.34 | ChannelZ | AGI is one thing. AMI is another. |
| 03:04.38 | ChannelZ | Please re-read what I said |
| 03:04.41 | trentcreek | I know |
| 03:05.01 | trentcreek | but the script is inside the AGI directory, logging into the AMI |
| 03:16.49 | *** join/#asterisk TheTosh (~Lynx@unaffiliated/thetosh) |
| 03:17.20 | *** join/#asterisk GoRK (~gork@c75-111-94-145.amrlcmta01.tx.dh.suddenlink.net) |
| 03:22.42 | GoRK | hello everyone; having some audio trouble wondered if anyone had any suggestions -- remote user is on satellite isp (hughesnet) with polycom 601 behind NAT. asterisk 1.6.0 with other phones on LAN in private ip space (so * is rtp proxy in this situation) -- codec is g.729. audio downlink to the phone on satellite sounds perfect, but audio coming back from the satellite phone is choppy. Don't really know where to start tweaking. |
| 03:22.42 | GoRK | running internal timing but also have DAHDI hardware available; tried both ways |
| 03:24.03 | ChannelZ | Upload from satellite is the suck. Like, a modem. |
| 03:25.12 | keith4_ | whoa. voip over satellite? seriously? |
| 03:25.14 | ChannelZ | Worse actually |
| 03:25.16 | GoRK | i understand the bandwidth constraint; i dont think that is the problem here; i can switch the codec to ulaw and actually improve the sound quality |
| 03:25.27 | ChannelZ | It's bandwidth AND latency probably |
| 03:25.37 | Katty | hi |
| 03:25.56 | epaphus | hey guys.. i have PCs with ekiga/twinkle connected with asterisk.. they all sound low on volume.. is it possible to increase that on the server side? |
| 03:26.00 | ChannelZ | you could crank up the jitter buffers which will add delay and might make the audio less choppy |
| 03:26.54 | GoRK | im thinking that because * is in the rtp stream its not letting the polycoms jb take care of the problem; but the jb cant be enabled per channel can it? |
| 03:27.12 | GoRK | polycom to polycom with no asterisk is fine also -- ie dial by ip |
| 03:29.47 | ChannelZ | If * is in the middle then its jitter buffers are probably somewhat more important than the phone's I'd have to guess |
| 03:31.11 | GoRK | well i was hoping that when asterisk does packet2packet bridging it would be just relaying the packets as they arrived and the jb on the phone would be able to compensate |
| 03:32.11 | GoRK | is the jitterbuffer or internal timing/rtp improved in releases after the 1.6.0 series? |
| 03:35.30 | ChannelZ | I think, but I don't know for sure, if * is in the middle it's outputting a steady stream.. there just might happen to be silence, coming from the source side jitter |
| 03:36.33 | *** join/#asterisk Rajmohan (~raj@122.165.25.171) |
| 03:37.03 | Rajmohan | hi, can anyone guide me how to setup asterisk behind nat router |
| 03:37.05 | ChannelZ | but I could be wrong, maybe it does just pass packets along as they come from the other end in whatever fashion they arrive in which case the phone should be able to figure it out. Dunno |
| 03:37.11 | ChannelZ | ~sipnat |
| 03:37.11 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:37.47 | trentcreek | ~burn book |
| 03:37.48 | infobot | ACTION pours gasoline all over book, ignites the fire, and then enjoys some toasty marshmallows with the glorious blaze |
| 03:38.48 | GoRK | that is sort of what it sounds like; as i said switching to ulaw improves it; im thinking since there is simply more data to send it gets more tx timeslots and the data arrives in a more regular fashion |
| 03:40.48 | GoRK | well ill try messing with some jitter options and also try to get asterisk to get out of the rtp stream and see what i can accomplish; just wanted to check and see if anyone else had some experience with this particular situation before i started shooting |
| 03:45.48 | ChannelZ | Rajmohan: if your firewall is a Linux box it might also help to use nf_conntract_sip - though I was trying to play with it this weekend and actually am baffled by the fact that without any NAT options on the server or my client (or the connection tracker) my RTP is working... makes it hard to test |
| 03:46.42 | antiwire | You just load that and it does stuff? |
| 03:47.14 | Rajmohan | i dont have the firewall enabled rightnow |
| 03:47.24 | Rajmohan | only the router has static ip with nat |
| 03:47.39 | Rajmohan | now i need the asterisk, freebpx to work with the static ip |
| 03:47.55 | Rajmohan | i forwarded the port 80 from the router to the asterisk box |
| 03:47.59 | Rajmohan | it does not work yet |
| 03:48.29 | ChannelZ | antiwire: well the idea is that like other connection trackers, it 'watches' the SIP traffic through the firewall and will automatically port forward and/or fudge stuff in the packets to make things work |
| 03:49.55 | Rajmohan | can you help with this pls |
| 03:50.19 | [TK]D-Fender | Rajmohan: What does port 80 have to do with this? |
| 03:50.48 | ChannelZ | Rajmohan: port 80 is only freepbx's management.. but all things being equal that should work if you're trying to hit up the management interface from the outside |
| 03:51.31 | Rajmohan | how do i setup with a router configured with static ip |
| 03:51.40 | ChannelZ | (or at least I assume, does freepbx use port 80 by default? I know AsteriskNOW does) |
| 03:51.45 | [TK]D-Fender | rajYou've been linked to the guide already |
| 03:53.21 | ChannelZ | Rajmohan: if you want people/phones from the outside to connect to it, you'd have to port-forward 5060 for SIP, a range of RTP ports per rtp.conf... |
| 04:07.51 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
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| 04:35.24 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 04:37.03 | *** join/#asterisk Aven (~avenger@78.36.107.48) |
| 04:37.04 | sawgood | from an end point (SIP phone) on an Asterisk 1.6 box, *98 feature code goes to voicemail ... from this can you simply 'listen' to your greetings without having to re-record them? |
| 04:37.19 | *** join/#asterisk Peste_Bubonica (~eduardo@200-171-87-11.dsl.telesp.net.br) |
| 04:37.22 | Peste_Bubonica | Hi all... |
| 04:37.31 | sawgood | everytime I do this ... I am only prompted to record them |
| 04:37.44 | Peste_Bubonica | Asterisk Supports video conference? over sip or h323 or another proto? |
| 04:37.48 | sawgood | I believe one is recorded already .... but I cannot hear it |
| 04:38.15 | sawgood | this is to a virtual mailbox with no phone assigned, so I cannot simply 'call the extension' and no answer |
| 04:40.45 | VoIP-Penguin | If there is no phone, then you CAN call the extension and get no answer. |
| 04:41.08 | sawgood | As soon as I dial 400 ... I get a system prompt saying the extension is not vaild |
| 04:41.11 | sawgood | its very strange ... |
| 04:41.48 | VoIP-Penguin | Then it must not be valid. Call an extension that runs VoicemailMain() and enter the mailbox number when prompted. |
| 04:42.38 | VoIP-Penguin | I know exactly what the problem is... |
| 04:42.41 | sawgood | VoIP-Penguin: I do that by using *98 from another phone |
| 04:43.01 | VoIP-Penguin | You're using FreePBX and wanting help in the Asterisk channel. |
| 04:43.08 | sawgood | then I put in extension 400 ... then the password |
| 04:43.24 | sawgood | the stuff I am asking in FreePBX is closely related but not the exact same |
| 04:43.35 | VoIP-Penguin | You put in extension 400 and then the password... which doesn't make a bit of sense to me. |
| 04:43.36 | sawgood | it is for another concern I have on another FreePBX box |
| 04:43.53 | sawgood | *98 ... then 400 then the VM password ... |
| 04:44.11 | VoIP-Penguin | So then 400 is the mailbox number. |
| 04:44.14 | sawgood | This does work ... I am able to get to the VM of 400 from another phone by using *98 |
| 04:44.24 | sawgood | yes, 400 is the mailbox number ... |
| 04:44.28 | VoIP-Penguin | You tried to convince me 400 was the extension, but it clearly wasn't. |
| 04:44.31 | sawgood | I cannot dial 400 though from another phone |
| 04:44.34 | antiwire | check your dtmf settings |
| 04:44.48 | VoIP-Penguin | So... what was the problem again? |
| 04:44.51 | sawgood | to me ... an extension is not a phone ... but rather a call route |
| 04:45.02 | *** join/#asterisk Neo31 (~Neo31@unaffiliated/neo31) |
| 04:45.20 | antiwire | it sounds like his phone isn't sending dtmf in the expected format |
| 04:45.24 | sawgood | I would like to 'listen' to the recorded default greeting of 400 without having to overwrite it |
| 04:45.27 | VoIP-Penguin | Extensions are those numbers and/or letters in extensions.conf that make things happen when you call them from a device. |
| 04:46.34 | antiwire | sawgood: are you sure that what you are hearing happen isn't the VM autoattendant timing out instead of receiving key pushes? |
| 04:46.53 | VoIP-Penguin | Okay, I understand your goal, but I would have to poke around in VoicemailMain to know if you can listen to your own greeting. |
| 04:47.10 | sawgood | There is no AA on this box |
| 04:47.18 | VoIP-Penguin | I would think you could listen to it. |
| 04:47.30 | sawgood | I think this might be the answer ... |
| 04:47.44 | sawgood | I recorded a greeting for 400 ... very generic ... |
| 04:48.06 | sawgood | The office manager did *98 to get to 400, and she automatically heard my greeting |
| 04:48.12 | VoIP-Penguin | Optionally, you could pull the greeting sound file from the file system and play it. |
| 04:48.23 | sawgood | I think she over-wrote the greeting, but did not save it ... thus erasing the greeting |
| 04:48.42 | sawgood | now, when you do *98 400 ... you do not automatically hear anything 'because the greeting is gone' |
| 04:49.15 | sawgood | I think that might be what is happening ... I double check this by doing it in my LAB |
| 04:49.36 | sawgood | VoIP-Penguin: neat answer ... I think I'll do that just for fun! |
| 04:50.36 | *** join/#asterisk chuchete (~chuchete@247.47.27.77.dynamic.mundo-r.com) |
| 04:50.57 | chuchete | hello |
| 04:51.28 | chuchete | same one can help me with a Linksys spa3102? |
| 04:51.48 | VoIP-Penguin | chuchete: Ask the question. |
| 04:51.52 | *** join/#asterisk igorg (~igorg@net182.255.92-116.dynamic.omsk.ertelecom.ru) |
| 04:52.43 | chuchete | before all thanks and sorry my bad english. I am spanish.... |
| 04:54.48 | chuchete | well I am trying to configure a spa3102, put at the moment I dont want to use a Asterisk implementantio, the only I need ate the moment is just use a ip phone with a spa3102 th make regualr phone calls |
| 04:55.11 | sawgood | SPA3102 = standard ATA? |
| 04:55.21 | chuchete | thats is |
| 04:55.30 | chuchete | is that possible? |
| 04:55.35 | VoIP-Penguin | The 3102 is a VoIP gateway, with Ethernet, FXO and FXS ports. |
| 04:55.39 | sawgood | Well, most ATA and SIP phones require a 'registration' ... |
| 04:56.14 | sawgood | The ATA would need an 'account' on some SIP proxy and/or a FXO line .. to make an outbound call |
| 04:56.15 | chuchete | I am trying with a softphone |
| 04:57.22 | chuchete | I just want to use the spa as a gw betwing my softpphone and the ptsn |
| 04:57.33 | sawgood | that is do-able ... |
| 04:57.40 | VoIP-Penguin | You'll have to use the VoIP-to-pstn gateway settings to be able to call from an IP phone to the pstn. |
| 04:57.52 | sawgood | Do you have a standard 'telephone' line inserted into the ATA? |
| 04:58.01 | chuchete | yes |
| 04:58.31 | sawgood | on your ATA box, put in the information for the softphone under some 'account' tab |
| 04:58.32 | VoIP-Penguin | If it wasn't going on 00:00, I would dig out the 3102 documentation and figure it out. |
| 04:58.33 | chuchete | I can actualy me phone calls from my standard analog tenephone pluged on the spa |
| 04:59.09 | sawgood | Then on your softphone ... put in the ATA information ... and your softphone should 'register' to the ATA as an extension (at least that is how Grandstream ATA boxes work) |
| 04:59.16 | chuchete | here it is 6 in the morning ;) |
| 04:59.20 | sawgood | no, wait a second ... |
| 04:59.31 | sawgood | I was wrong |
| 04:59.59 | sawgood | You probably cannot 'register' a softphone onto an ATA unless the ATA has something special/different than I am thinking |
| 05:00.13 | VoIP-Penguin | It would have to be a registrar. |
| 05:00.21 | chuchete | I can conect to the spa from the softphone, but if i try to dial, nothing happend |
| 05:00.22 | sawgood | VoIP-Penguin is right ... I was wrong |
| 05:01.01 | chuchete | is not enough to be a sip client to connect to the spa? |
| 05:01.02 | *** join/#asterisk sourcode (~code@ppp-61-90-15-12.revip.asianet.co.th) |
| 05:01.14 | sawgood | most softphone (esp. Xlite; EyeBeam; Bria) require the phone be registered to a 'server' before they can make a call |
| 05:01.35 | sawgood | been down this road a few times |
| 05:01.37 | VoIP-Penguin | You have to configure the ATA with device/user information, then place calls to it. It would then dial out to the pstn if configured to do it. |
| 05:01.55 | chuchete | tha is ATA? |
| 05:02.00 | sawgood | I'm sure that is absolutly right |
| 05:02.04 | chuchete | wha is ATA? |
| 05:02.18 | VoIP-Penguin | SPA-3102 is an ATA |
| 05:02.20 | chuchete | waht u mean with ATA? sorry |
| 05:02.20 | ChannelZ | anus tickling aparatus |
| 05:02.23 | chuchete | ahhh ok |
| 05:02.48 | chuchete | well the ata has factory defaults, except the wan settings |
| 05:02.57 | VoIP-Penguin | I wonder if any of the softphone apps have registrar stuff built in. |
| 05:03.22 | chuchete | I have no problems withe registrations issues |
| 05:03.23 | VoIP-Penguin | If so, I guess that would solve the problem. |
| 05:03.44 | chuchete | the softphone is configurated to not register |
| 05:04.10 | VoIP-Penguin | There are so many settings in the 3102, it is easy to get confused. |
| 05:04.26 | chuchete | In my case is pretty simple |
| 05:04.29 | sawgood | I wanted one time to make two EyeBeam softphones to work talking over the Internet to each other, but they have to have a SBC in the middle |
| 05:04.33 | sawgood | kind of a drag ... |
| 05:05.09 | chuchete | I just need to use a sip client to place regular calls using de ATA |
| 05:06.17 | chuchete | try to find google info but all the info refers to asterisk implementation |
| 05:06.21 | sawgood | chuchete: you are probably 'missing' one piece of the puzzle ... or you have to do some 'trick' on the ATA to make it work the way you want it to |
| 05:06.54 | chuchete | I supposed this was the simplest way to use it |
| 05:07.00 | sawgood | to me, you only have 2 of the 3 requirements met |
| 05:07.26 | VoIP-Penguin | You can probably configure the PSTN user tab and have the softphone register to the ATA. |
| 05:07.37 | chuchete | yes |
| 05:07.49 | chuchete | but..? |
| 05:08.45 | chuchete | why the ata does not place the call that I am tring to do from the softphone? |
| 05:09.28 | sawgood | chuchete: you have to do more work on the ATA first ... before things will work |
| 05:10.46 | chuchete | jejejej I kown my friend, but after been reading a lot a playing a lot as well I am a bit confused |
| 05:11.20 | chuchete | looks like the simpliest way to use it, its not supported |
| 05:11.30 | VoIP-Penguin | I think it might be supported. |
| 05:11.33 | sawgood | wow ... you didn't read what VoIP-Penguin said |
| 05:12.03 | chuchete | sorry what u mean sawgood ? |
| 05:12.18 | VoIP-Penguin | I'm just not in the mood to dig out the instruction manual and get a route to a 3102 so I can login. |
| 05:12.23 | sawgood | You need to follow the TIP VoIP-Penguin gave to you |
| 05:12.45 | VoIP-Penguin | Later, maybe; just not right now. |
| 05:13.25 | chuchete | ok, thank u very much, I will keep trying |
| 05:14.15 | VoIP-Penguin | Maybe at 21:00 :) |
| 05:16.59 | VoIP-Penguin | chuchete: I mean 21:00 your time. |
| 05:18.29 | chuchete | ok, I´ll be back by that time then and comment my progress |
| 05:19.59 | sawgood | The RPM for Asterisk 1.6.0 ... is updated in the repo ... took my box to 1.6.0.26 ... very COOL!! |
| 05:41.52 | simcop2387 | with asterisk 1.6.1 how would i check how long a call has been up? (call was over sip if that matters, i'm not seeing anything in sip show channel) |
| 05:51.03 | sawgood | I have an Asterisk 1.6.0 box running with a SIP trunk as the only way to send/receive calls |
| 05:51.11 | sawgood | I can make 2 way audio outbound calls just fine |
| 05:51.44 | sawgood | When I call the DID ... I see in Asterisk the call arrives at the IP PBX, but I am getting a teleco error telling me the call cannot be completed as dialed |
| 05:51.54 | sawgood | any tips to why I cannot receive an incoming call |
| 05:52.47 | sawgood | "The number you have dialed ... is not in service" .... |
| 05:52.59 | sawgood | I'm not sure if this is an Asterisk system prompt or a teleco prompt |
| 05:53.19 | VoIP-Penguin | You probably don't have an extension for the call. |
| 05:53.35 | VoIP-Penguin | But without a sip debug, how will we ever know? |
| 05:54.11 | sawgood | I have an extension for the call (my softphone) ... which can make outbound calls |
| 05:54.21 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 05:55.29 | sawgood | is the troubleshooting command simply ... sip debug? |
| 05:57.16 | *** join/#asterisk PARAG (~PARAG@mail2.eastnets.com) |
| 05:57.21 | sawgood | <PROTECTED> |
| 05:57.36 | sawgood | So, I take it the system prompt I am hearing is an Asterisk system prompt |
| 05:57.45 | VoIP-Penguin | sawgood: phones are not extensions. |
| 05:57.58 | VoIP-Penguin | I repeat, phones are NOT extensions. |
| 05:58.17 | sawgood | so, I do not have an extension which can take the call |
| 05:58.20 | VoIP-Penguin | If your call does not work, it is probably because you do not have an extension for it. |
| 05:58.36 | sawgood | how do I 'see' if my softphone is an extension ... |
| 05:58.50 | VoIP-Penguin | phones are NOT extensions. |
| 05:59.04 | sawgood | ok ... let me try another re-wording then |
| 05:59.22 | sawgood | right now, the inbound route is to 'go to' extension 800 ... I have my softphone registered as extension 800 |
| 05:59.25 | sawgood | is that correct? |
| 05:59.37 | VoIP-Penguin | Close, but no. |
| 05:59.49 | VoIP-Penguin | Phones are not extensions. Phone cannot register as extensions. |
| 06:00.05 | VoIP-Penguin | Extension 800 can Dial your softphone. |
| 06:00.12 | *** join/#asterisk smooth_penguin (~smoove@59.95.19.29) |
| 06:00.19 | sawgood | very nice answer ... ok ... |
| 06:00.33 | sawgood | so, when a call comes into extension 800, how come my softphone does not ring? |
| 06:00.41 | sawgood | is there a way to test this? |
| 06:01.20 | VoIP-Penguin | Maybe extension 800 does not exist. |
| 06:01.30 | sawgood | when I dial *65 from the softphone, I hear the system prompt tell me, "your extension number is 800" |
| 06:01.33 | VoIP-Penguin | Maybe exten 800 does not have a Dial command to dial your phone. |
| 06:01.51 | VoIP-Penguin | Until you give me a sip debug, I still don't know why there is a failure. |
| 06:02.02 | sawgood | how do I turn on SIP debug |
| 06:02.10 | sawgood | I tried: sip debug .... debug sip ... |
| 06:02.14 | sawgood | nothing worked |
| 06:02.17 | VoIP-Penguin | sip set debug on |
| 06:02.19 | sawgood | thank you |
| 06:02.56 | sawgood | ok ... it is on ... should I call the DID now? |
| 06:03.02 | VoIP-Penguin | yes |
| 06:03.41 | sawgood | ok ... I have done it ... now how do I get the 'text from this" simply do a copy paste? |
| 06:03.53 | VoIP-Penguin | ~pb |
| 06:03.53 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 06:04.02 | VoIP-Penguin | Copy it and pastebin it. |
| 06:04.50 | VoIP-Penguin | Even if I can determine why the behavior exists, I doubt I can tell you how to fix it because you are using FreePBX and we can't support that. |
| 06:05.03 | VoIP-Penguin | ~freepbx |
| 06:05.04 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 06:05.14 | sawgood | actually this is trixbox CE ... a different PBX |
| 06:05.23 | VoIP-Penguin | It's still using FreePBX. |
| 06:05.52 | sawgood | http://pastebin.org/120811 |
| 06:06.01 | sawgood | well, I'll keep my fingers crossed ... |
| 06:07.32 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:07.38 | VoIP-Penguin | It looks like a perfectly good call to me. |
| 06:08.34 | sawgood | oh ... |
| 06:08.36 | sawgood | cool ... |
| 06:08.48 | sawgood | the call arrives ... but it does not ring the softphone ... |
| 06:09.01 | sawgood | I tried changing the extension to a 'hard phone', but the same result happens |
| 06:09.09 | sawgood | Maybe if I create a ring group |
| 06:09.19 | VoIP-Penguin | Call came in, looked for 5105501404 in from-sip-external, then executed 5105501404@from-sip-external. |
| 06:09.45 | VoIP-Penguin | It appears that there isn't anything for it to do there. |
| 06:10.33 | PARAG | ~pb |
| 06:10.34 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 06:10.54 | PARAG | ~voip |
| 06:10.55 | infobot | it has been said that voip is Voice over IP |
| 06:11.20 | VoIP-Penguin | The channel was answered, then played back the sound file like you said, then played the congestions tones... |
| 06:11.54 | sawgood | 'maybe' this is a SIP trunk provider concern? |
| 06:12.04 | VoIP-Penguin | If you weren't using a GUI, it would be a simple thing to fix. |
| 06:12.19 | sawgood | I notice on the SBC ... the 'orgin' IP is 127.0.0.1 ... when it is normally the static public IP address of the IP PBX |
| 06:12.37 | VoIP-Penguin | It would be as simple as exten => 5105501404,1,Dial(SIP/800,30) |
| 06:13.15 | sawgood | what does the 30 represent? |
| 06:13.17 | VoIP-Penguin | Your call is making it to your box just fine. |
| 06:13.27 | VoIP-Penguin | 30 second timeout on the Dial() application. |
| 06:13.34 | sawgood | thank you for your help |
| 06:14.05 | VoIP-Penguin | You could leave off the ,30 and it would timeout because of another timeout value anyway. |
| 06:14.11 | sawgood | I know on the SBC ... all the other 'working' accounts do not have 127.0.0.1 for an orgin IP ... they have the IP address of the IP PBX, but to me that would seem to mean the call would not arrive to the IP PBX at all |
| 06:16.28 | sawgood | VoIP-Penguin: do you work for Digium in Alabama? |
| 06:16.37 | VoIP-Penguin | no |
| 06:16.58 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 06:22.09 | VoIP-Penguin | Good luck in your configurations... time for sleep for me. (if I don't go now, I may not get to go at all.) |
| 06:30.09 | sawgood | thanks |
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| 07:41.01 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 07:42.54 | doolittlework | hi there i have the following 1 lines of code exten => _[*0-9].,n,GotoIf(${VALID_EXTEN(mainout,${EXTEN},1)}?mainout,${EXTEN},1) and i am battling to make sence of it, could someone please enlighten me |
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| 07:43.32 | doolittlework | wat does the $VALID_EXTEN bit do |
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| 07:53.03 | kaldemar | doolittlework: core show function VALID_EXTEN will tell you. but that exten won't work because of a syntax error. |
| 07:59.16 | chuchete | It wooorkkkkkks !!!! I did itttt |
| 08:00.53 | doolittlework | thx kaldemar |
| 08:01.14 | *** join/#asterisk LnxBil (~LnxBil@p5099b332.dip0.t-ipconnect.de) |
| 08:01.18 | LnxBil | Hi everybody. |
| 08:02.16 | LnxBil | I've problems with dialplan from LDAP on asterisk 1.6.2.0-1. It doesn't get loaded. |
| 08:02.55 | LnxBil | I use 'extensions => ldap,"dc=XXX,dc=XXX",extensions' in extconfig, but the querylog of my ldap server doesn't show any queries |
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| 08:14.37 | doolittlework | kaldemar: Goto(restart) can this restart the dialplan if placed at the end or the context? |
| 08:17.40 | *** join/#asterisk Toommi (~name@geldern.screenwork.de) |
| 08:17.49 | Toommi | hello |
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| 08:19.10 | ChannelZ | doolittlework: no that would jump to a priority label called 'restart' |
| 08:19.20 | Toommi | is there an event or something similar which is called when a phone is registering on asterisk |
| 08:19.30 | kaldemar | doolittlework: there is no such thing as restarting a dialplan. and it doesn't matter if something is at the end of a context. |
| 08:20.07 | ChannelZ | Toommi: Only in Manager really |
| 08:23.18 | Toommi | ChannelZ can you specify please |
| 08:23.50 | ChannelZ | http://lmgtfy.com/?q=asterisk+manager&l=1 |
| 08:24.26 | ChannelZ | the Asterisk Manage Interface - too expansive to explain in a few lines here |
| 08:24.29 | ChannelZ | And I'm going to bed anyways |
| 08:24.39 | *** join/#asterisk ice (~pisg@intrush.pl) |
| 08:25.05 | ChannelZ | but if you setup an AMI user properly and login to AMI with it, you will get 'events' about certain things |
| 08:25.07 | Toommi | imagine i googled allready :) |
| 08:25.45 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:26.18 | ChannelZ | in particular look at the Events command |
| 08:27.50 | Toommi | ok thank you ;) |
| 08:30.30 | ice | hi, iuse FAQ: http://blog.jploh.com/2007/01/28/asterisk-callback-disa/ to set up a callback, but hat i must setting in callback.call, when my outline is exten => _X.,1,Dial(DAHDI/g0/${EXTEN}) |
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| 08:43.12 | doolittlework | what the hell will this do? "(${nolb_${n}}?" |
| 08:43.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:44.49 | kaldemar | doolittlework: nothing |
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| 08:50.57 | doolittlework | lol |
| 08:51.03 | kaldemar | doolittlework: show some more if you want to know what something does. that just _maybe_ holds some value. |
| 08:51.19 | kaldemar | and is surrounded by irrelevant characters. |
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| 09:01.44 | doolittlework | we had someone setup a iax trunk for us before, but he passed away, sad to say, now I must decode it, must say having fun with asterisk, but quite a huge task if i only started linux three weeks ago, I need to change this over to a sip trunk |
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| 09:05.15 | doolittlework | kaldemar: http://pastebin.com/gLUR1sFn |
| 09:12.35 | doolittlework | brb |
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| 09:58.24 | aruntomar | can't hear voice on incoming call, using pri line with asterisk and redfone appliance, but the outgoing works fine |
| 09:58.37 | davix | how can I limit chanspy to only a range of extensions? |
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| 10:06.29 | ManxPower-work | ~answers |
| 10:06.30 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
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| 10:35.09 | redax | hi. |
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| 10:48.47 | ice | someone have good how to create callback asterisk ? |
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| 11:05.11 | ManxPower-work | A well known april fool's joke: Left Handed Whoppers: In 1998, Burger King ran an ad in USA Today, saying that people could get a Whopper for left-handed people whose condiments were designed to drip out of the right side.[8] Not only did customers order the new burgers, but some specifically requested the "old", right-handed burger. |
| 11:08.06 | *** join/#asterisk AsteriskNoob (~AsteriskN@host217-43-21-195.range217-43.btcentralplus.com) |
| 11:08.16 | AsteriskNoob | Hello all. |
| 11:08.30 | AsteriskNoob | Quick question about TFTP if anyone can help. |
| 11:08.49 | yang | ManxPower-work: they can only trick americans with such tricks |
| 11:09.25 | ManxPower-work | "Phone call: In 1998, UK presenter Nic Tuff of West Midlands radio station Kix 96 pretended to be the British Prime Minister Tony Blair when he called the then South African President Nelson Mandela for a chat. It was only at the end of the call when Nic asked Nelson what he was doing for April Fools' Day that the line went dead." |
| 11:22.14 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:26.32 | redax | what does asterisk do with user pressed flash button on a phone, using PAP2 ATA, pap2 sends the Flash button via AVT/INFO ? |
| 11:28.24 | *** join/#asterisk dwarken (~chatzilla@1405ds1-svo.0.fullrate.dk) |
| 11:30.47 | c0rnoTa | redax: it sends Hold via DTMF signalling (AVT/INFO/InBand) |
| 11:34.06 | ManxPower-work | That is incorrect. |
| 11:34.58 | ManxPower-work | Well, OK. Sort of correct. It sends a hold message, NOT DTMF. However other than that it is all handled on the ATA, not Asterisk |
| 11:37.03 | ManxPower-work | Polycom phones, for example, default to sending an invite with a a=recvonly in the SDP |
| 11:37.20 | ManxPower-work | That causes the server to play hold music to the caller |
| 11:38.21 | c0rnoTa | Yeap, ManxPower-work, thanks for correction |
| 11:41.45 | dwarken | where do i put a dialplan for incoming calls? |
| 11:41.56 | c0rnoTa | ~thebook |
| 11:41.57 | infobot | i guess thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 11:42.15 | c0rnoTa | dwarken: thebook tells you |
| 11:43.06 | ManxPower-work | dwarken, Where you set your system up to send them to. You must design that yourself. |
| 11:47.07 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 11:49.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:49.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:55.25 | redax | ManxPower-work: just asked what happens if I turn off the 3way conference, and call waiting service in the ATA, and it sends FLASH via sip info to the asterisk. |
| 11:58.22 | *** join/#asterisk TheTosh (~Lynx@unaffiliated/thetosh) |
| 12:08.13 | *** join/#asterisk ddefrenne (~ddefrenne@83.101.71.187) |
| 12:11.00 | dwarken | some take a look at http://pastebin.com/VAHQh1v0 ? :) i'm lost |
| 12:13.48 | *** part/#asterisk aruntomar (~test@61.17.193.163) |
| 12:14.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:14.09 | *** join/#asterisk cellZero (~stelio@dsl-185-111-172.dynamic.wa.co.za) |
| 12:14.19 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 12:15.08 | ManxPower-work | dwarken, you don't do this in the dialplan. How are calls getting into the system? SIP, PRI, CAS T-1, POTS? |
| 12:15.12 | cellZero | anyone know how to pass variable between AGI scripts? |
| 12:15.30 | ManxPower-work | cellZero, set a dialplan variable |
| 12:16.26 | dwarken | ManxPower-work: i'm using SIP... |
| 12:16.59 | ManxPower-work | dwarken, then the context= line in the sip.conf entry for that incoming device specifies where in extensions.conf the call will land. This is only for calls coming from that device. |
| 12:17.01 | cellZero | ManxPower-work: I'm using phpagi, and the set_variable function, but when i call the get_variable function it complains about Undefined variable. |
| 12:17.28 | [TK]D-Fender | cellZero: then you're calling it wrong |
| 12:17.34 | *** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk) |
| 12:17.42 | Toommi | any idea why my callrecording does not work, intern is works perfectly by if i call from extern it does not work, extension config is similar wW option is set |
| 12:17.43 | ManxPower-work | cellZero, You must be doing something wrong then. I am assuming these are multiple AGIs run during the SAME CALL? |
| 12:17.59 | ManxPower-work | Toommi, you are not setting DYNAMIC_FEATURES |
| 12:18.09 | cellZero | ManxPower-work: Yes |
| 12:18.22 | Toommi | oh ofcouse i set it :) |
| 12:18.29 | Toommi | [globals] |
| 12:18.29 | Toommi | DYNAMIC_FEATURES=automon |
| 12:18.33 | ManxPower-work | cellZero, then you are doing it wrong or there's a problem in phpagi (which would not suprized me at ALL) |
| 12:18.48 | Toommi | and automon in feature conf ist = *1 |
| 12:19.09 | [TK]D-Fender | ManxPower-work: No, the AGI lib has been very stable for a long time |
| 12:19.12 | Toommi | and not marked out :X |
| 12:19.26 | ManxPower-work | [TK]D-Fender, No, phpagi sucks. 8-| |
| 12:19.40 | cellZero | let me pastebin, and if anyone has a moment to look at my script, please do |
| 12:19.42 | [TK]D-Fender | ManxPower-work: I've never seen a problem that was user-based |
| 12:19.52 | ManxPower-work | For one thing it only supports a fairly small set of AMI options. |
| 12:20.41 | *** join/#asterisk thecardsmith (~doug@pool-71-161-218-3.burl.east.myfairpoint.net) |
| 12:21.26 | ManxPower-work | Toommi, now is the time where you paste the CLI output, the dialplan parts and your features.conf to pastebin.ca |
| 12:21.57 | cellZero | if anyone knows phpagi, please take a look here http://pastebin.com/53ZHY2bz |
| 12:22.36 | Toommi | intern it works if i call from 16 to ext 12, but if i been called from extern (in a queue to an agent) for example agent 16 who can record intern, it doesnt work |
| 12:23.02 | ManxPower-work | cellZero, I don't see any errors |
| 12:23.04 | Toommi | intern cli output : > User hit '*1' to record call. filename: wav,auto-1269350568-16-12,m |
| 12:23.08 | dwarken | ManxPower-work: cant find any context= line in the sip.conf ?? |
| 12:23.23 | ManxPower-work | Toommi, I'm waiting. |
| 12:23.38 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:23.44 | ManxPower-work | dwarken, I guess that's why it doesn't work then. Now STOP. And go READ THE BOOK. |
| 12:24.40 | Toommi | ManxPower-work: features.conf : http://pastebin.com/G2aLaMRW |
| 12:25.40 | cellZero | ManxPower-work: I'm using PHPAGI 2 from eder.us, thinking there might be a bug when retrieving variables |
| 12:26.01 | Toommi | ManxPower-work: cli for intern recording where it works http://pastebin.com/Mksv3pnG |
| 12:26.45 | ManxPower-work | Toommi, that is useless. do a "set verbose 3" and try it again |
| 12:27.07 | [TK]D-Fender | Toommi: And you aren't showing the complete call. |
| 12:27.24 | Toommi | http://pastebin.com/U4YdLfvp |
| 12:27.42 | Toommi | here is complete call for the extern call i hit the record key but nothing happens :X |
| 12:27.47 | ManxPower-work | Toommi, the first dialplan priority I want to see is priority 1. |
| 12:28.07 | ManxPower-work | not priority 7. [s@queue-call:7] |
| 12:28.24 | ManxPower-work | You need to show the complete call, not parts of it. |
| 12:28.44 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 12:29.32 | Toommi | ok sry |
| 12:29.47 | Toommi | http://pastebin.com/zqx4sjKe |
| 12:30.58 | ManxPower-work | Toommi, I do not see that W or w on the Queue line. |
| 12:31.44 | Toommi | oh god i am so stupid i put it in the call Executing [16@queue-call-phone:2] Dial("Local/16@queue-call-phone-b5d5;2", "SIP/16,15,tTwW") in new stack |
| 12:33.16 | *** join/#asterisk ddefrenne (~ddefrenne@83.101.71.187) |
| 12:33.51 | [TK]D-Fender | cellZero: I certainly have no reason to believe that |
| 12:34.29 | Toommi | thank you very much it works now, i didnot thought enough :x |
| 12:34.32 | *** join/#asterisk bobisa (~boboboboo@66.234.24.142) |
| 12:35.10 | bobisa | hi, i want to buy some ip phone, and i dont know whitch one to buy. have some suggestion ? |
| 12:35.43 | [TK]D-Fender | bobisa: Polycom > All |
| 12:36.20 | *** join/#asterisk styelz (~yoohoo@m0o0.mooo.com) |
| 12:36.25 | *** join/#asterisk dwarken (~chatzilla@1405ds1-svo.0.fullrate.dk) |
| 12:36.47 | ManxPower-work | ~phones |
| 12:36.48 | 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, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 12:37.32 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 12:40.25 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 12:41.25 | LnxBil | Anyone using LDAP for Dialplan? |
| 12:41.45 | [TK]D-Fender | LnxBil: Never that I've heard of |
| 12:42.02 | AsteriskNoob | I'd say forget Cisco 79xx... Config nightmare for me so far.. |
| 12:43.43 | [TK]D-Fender | AsteriskNoob: They aren't that big a deal... just that the end value isn't usually better than Polycom. |
| 12:44.04 | [TK]D-Fender | AsteriskNoob: Most would say Polycom is harder (and I agree) to start... but its the end that counts |
| 12:47.51 | ManxPower-work | "I do not want people to be agreeable, as it saves me that trouble of liking them." --Jane Austen |
| 12:48.16 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 12:48.56 | AsteriskNoob | Fender: Well, I've got a brick sitting on my desk right now. New to the whole SIP phone thing myself, but looking forward to having full SIP implementations soon. Just not happy with the configuration steps required like TFTP servers, etc... Telnet or a web page would have been nice. |
| 12:48.57 | Nugget | telnet is eeeeeeevil! |
| 12:49.22 | c0rnoTa | I'v found queue.conf, where members described like "Agent", but there are no agent describtion lines in agents.conf , could be this configuration useful for queue ? Is there another way to describe agents (not realtime method - there are no realtime engines configured)? |
| 12:49.30 | AsteriskNoob | Evil yes, but it's a simpe option if used locally. |
| 12:49.58 | [TK]D-Fender | Corydon76-dig: Huh? |
| 12:50.20 | [TK]D-Fender | c0rnoTa: Huh? |
| 12:50.30 | devoid | Haha |
| 12:50.47 | LnxBil | [TK]D-Fender: It works, at least it did once on a test machine. Unfortunately, I'm not able to reproduce it. Authentication works, VoiceMail works, but Dialplans don't work |
| 12:51.27 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 12:53.06 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 12:53.18 | Skeeter- | Morning yall |
| 12:54.20 | bobisa | ok, but im new in this world, for now is for test purpose, but i want to keep it after, also i want to have access to my anolog line and my skype sip, does polycom still the best choice ? |
| 12:55.01 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 12:55.02 | ManxPower-work | bobisa, Polycom is a SIP phone. Just like all the other SIP phones out there. |
| 12:55.21 | *** join/#asterisk TimeRider (steve@5ac31820.bb.sky.com) |
| 12:55.42 | ManxPower-work | bobisa, you really should read the Asterisk Book. |
| 12:55.44 | c0rnoTa | [TK]D-Fender: queue.conf have "member => Agent/32" line, but in agents.conf definition of the agents isn't exist. As I know, call to this queue wouldn't be useful because there are no members in queue. Am I wrong? |
| 12:55.45 | ManxPower-work | ~answers |
| 12:55.46 | infobot | somebody said answers was Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:55.57 | [TK]D-Fender | bobisa: For * to use Skype, there is a $66 USD pay-only per channel cost. |
| 12:56.06 | [TK]D-Fender | bobisa: And "test" means nothing |
| 12:56.14 | [TK]D-Fender | bobisa: you buy a phone to USE. |
| 12:56.27 | [TK]D-Fender | bobisa: To qualify for "test", just install a soft-phone |
| 12:56.47 | bobisa | i know but i want to start the right way, |
| 12:57.10 | [TK]D-Fender | c0rnoTa: then I guess you'd better go DEFINE that Agent, then shouldn't you" |
| 12:57.13 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 12:57.20 | [TK]D-Fender | bobisa: What do you actually want to do>? |
| 12:57.49 | bobisa | i want to replace my existing pbx that got trouble on it, for asterisk |
| 12:57.59 | bobisa | i want to know if that will be my best choice |
| 12:58.06 | bobisa | or i buy a new pbx |
| 12:58.13 | jblack | You're asking in #asterisk? Of course asterisk is the best choice. |
| 12:58.45 | LnxBil | Is there any way to debug the Realtime dialplan lookup? |
| 12:58.52 | jblack | If you don't go with asterisk, your hard drives will explode and you'll be fired. You'll be so sad about that... |
| 12:59.08 | jblack | that you won't look both ways when crossing the street on the way home, and youll get hit by a car... |
| 12:59.19 | c0rnoTa | [TK]D-Fender: ok, thx |
| 12:59.25 | jblack | thus killing you. So, when you look at it carefully, it's a matter of life or death for you to install asterisk. |
| 12:59.39 | [TK]D-Fender | bobisa: No pressure :) |
| 13:00.13 | bobisa | i know, but the cost will make my choice, that why i have so much question before starting anything |
| 13:00.39 | [TK]D-Fender | bbogo isntall *. Downlaod the book. install a softphone. Play around. |
| 13:00.41 | jblack | Oh, it's a matter of life and death.. but it's a matter of your death. We'll keep on living, even without you. |
| 13:00.43 | [TK]D-Fender | bobisa: ^ |
| 13:00.48 | jblack | Change that order. |
| 13:01.04 | jblack | Download the book. Read the book, install a soft phone, then install asterisk and play around. =) |
| 13:02.04 | *** join/#asterisk voipmonk (~shido6@dsl-69-172-110-65.acanac.net) |
| 13:02.34 | *** join/#asterisk thecardsmith (~doug@pool-71-161-218-3.burl.east.myfairpoint.net) |
| 13:03.12 | gego | jblack: Actually I sometimes was so desperately trying to get things working in * that I nearly got hit by a car ... |
| 13:03.23 | gego | jblack: how do you explain that ? |
| 13:03.40 | jblack | You -didn't- get hit by a car, =because you had worked on asterisk= |
| 13:03.51 | c0rnoTa | Oh! Another way to define agents is users.conf - i'v forgot it. |
| 13:04.00 | jblack | Had you done something else, your timing crossing the street would have been different, and you woulda gotten squahsed. |
| 13:04.08 | [TK]D-Fender | ~users.conf |
| 13:04.08 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 13:04.10 | gego | Ah, now I see ... thanks |
| 13:04.35 | jblack | Don't thank me. I don't work on the code. |
| 13:05.22 | *** join/#asterisk nickaugust (~anonymous@167.83.189.72.cfl.res.rr.com) |
| 13:06.24 | gego | no, but you gave me these deep insights of how things work ... in a bigger picture |
| 13:07.19 | c0rnoTa | What means "toaster grade"? :) |
| 13:08.20 | c0rnoTa | PBX like a toaster? :) |
| 13:08.34 | [TK]D-Fender | c0rnoTa: Yes... a "dumb appliance" |
| 13:09.28 | c0rnoTa | I'm solidarity with u |
| 13:09.52 | c0rnoTa | Don't like "users.conf way" |
| 13:09.57 | jblack | oh man we're FUCKED! |
| 13:10.08 | jblack | Bill Gates is now getting invovled in nuclear reactor development |
| 13:10.37 | coppice | ah, the blue smoke of death |
| 13:10.58 | c0rnoTa | that's why i'v forget about users.conf, cause i don't use it. So, [TK]D-Fender, thank you again! |
| 13:11.02 | [TK]D-Fender | SOMEBODY SET HIM UP THE BOMB |
| 13:16.00 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:19.52 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:20.08 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
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| 13:32.04 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:39.04 | PARAG | Guys, is it easy to sniff the SIP Passwords ?? |
| 13:40.17 | Toommi | install tshark and try yourself ^^ |
| 13:40.32 | file | they aren't sent in plain text. |
| 13:40.42 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:40.43 | PARAG | Yes it is encrypted |
| 13:41.21 | PARAG | Toommi, did you try ever ? |
| 13:41.51 | file | you use wireshark or whatever to get the MD5 hash if just using UDP or TCP, but that's not the password |
| 13:41.54 | Toommi | no but i am installing :) |
| 13:42.11 | tzafrir | PARAG, the passwords are not sent in the clear. Only a md5 checksum of them, along with that of the realm (domain) and a random "nonce" is sent |
| 13:42.40 | PARAG | tzafrir, that is correct. But do you know any any way we can sniff it ? |
| 13:42.46 | tzafrir | So not encrypted. Hashed |
| 13:43.15 | tzafrir | And no, you can only sniff the hash |
| 13:43.28 | PARAG | tzafrir, and what can i do with hash ? |
| 13:43.32 | PARAG | :) |
| 13:43.42 | tzafrir | smoke it |
| 13:44.03 | PARAG | i don't want to do that......so there is no way u mean |
| 13:44.04 | coppice | or make cookies |
| 13:45.39 | Naikrovek | cookies/ |
| 13:45.40 | Naikrovek | ? |
| 13:45.47 | Toommi | cakes in small :) |
| 13:45.55 | Naikrovek | someone said cookies? i'll take 10 |
| 13:47.52 | *** join/#asterisk Akiraa (~Akira@92.81.192.172) |
| 13:48.07 | [TK]D-Fender | spins up "Nookie" by Limp Bizkit |
| 13:51.01 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 13:51.19 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
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| 13:57.07 | *** part/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 13:58.01 | Toommi | PARAG: http://img32.imageshack.us/img32/1334/package.png |
| 13:58.09 | *** join/#asterisk ACK-NAK (~Miranda@home.chicagoventure.com) |
| 13:58.28 | *** join/#asterisk Toommi (~name@geldern.screenwork.de) |
| 13:58.31 | Toommi | re^^ |
| 13:59.16 | Naikrovek | md5 hashes are the way you can validate passwords without transmitting them |
| 13:59.43 | Toommi | yes but md5 is not really secure |
| 13:59.46 | Naikrovek | but with enough time md5 hashes can be brute forced. takes a loooong time though |
| 13:59.51 | *** join/#asterisk epaphus (~name@190.10.68.228) |
| 13:59.55 | Toommi | or rainbow tables |
| 14:00.00 | Toommi | with much luck ;) |
| 14:00.06 | patrb | PARAG: an md5 hash may as well be a plain text password unless you salt it |
| 14:00.13 | Naikrovek | yes |
| 14:00.19 | epaphus | hey guys.. i have PCs with ekiga/twinkle connected with asterisk.. they all sound low on volume.. is it possible to increase that on the server side? |
| 14:00.22 | epaphus | ive done everything i can on the client machines |
| 14:00.27 | Naikrovek | the no-salt md5 are already all brute forced, up to like 16 chars |
| 14:00.33 | patrb | yarr |
| 14:00.52 | patrb | oh and dont salt it with something stupid like their username |
| 14:00.56 | patrb | looks at microsoft |
| 14:01.01 | Toommi | ^^ |
| 14:01.09 | coppice | Toomi: the safe makers understand the nature of security better than computer people. they know nothing is totally safe. its all just a scale you need to rate things on |
| 14:02.18 | Toommi | yes i know it |
| 14:02.40 | patrb | i think that mentality is a cop out, if you're a developer...you should understand security |
| 14:02.44 | coppice | so you know that "yes but md5 is not really secure" is a meaningless statement |
| 14:03.15 | *** join/#asterisk voipmonk (~shido6@dsl-69-172-110-65.acanac.net) |
| 14:03.32 | coppice | MD5 is very secure against attacks by my mum, and rather less secure against attacks by the NSA when they have the bit between their teeth |
| 14:04.41 | *** join/#asterisk ktwilight_ (~ktwilight@91.180.32.245) |
| 14:04.58 | Toommi | i tried to say that md5 is not soooooooooooooooooooooo secure that when you hash your passwords that it is really save |
| 14:05.00 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 14:05.32 | Toommi | but in this case it is enough secure |
| 14:07.03 | coppice | really safe against who? |
| 14:07.50 | patrb | buy seriously...md5 may protect you from your mom, but even a 12 year old with a bad attitude could figure out how to crack it |
| 14:07.57 | patrb | ..unless its salted |
| 14:08.06 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:08.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:08.32 | coppice | MD5 still ain't that easy to crack. very few demonstrations of faking MD5 have been staged |
| 14:08.45 | patrb | thats an ignorant statement |
| 14:08.54 | coppice | why? |
| 14:09.03 | patrb | 1 sec, getting a link |
| 14:09.04 | Naikrovek | coppice: it's not easy to reverse, but it's easy to brute force |
| 14:09.27 | Naikrovek | coppice: md5 is fast, you just md5 the dictionary (to start with) and try to match hashes |
| 14:09.56 | patrb | http://www.defcon.org/images/defcon-17/dc-17-presentations/defcon-17-matt_weir-sudhir_aggarwal-cracking_passwords.pdf |
| 14:10.32 | patrb | If you have Itunes, download the talk from Defcon 17 called "Cracking 400,000 Passwords or How to Explain to Your Roommate why the Power Bill is a Little High" |
| 14:10.37 | patrb | Its free |
| 14:11.06 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 14:12.58 | patrb | coppice: there are also websites that let you paste your md5 hash...then the site servs you the cracked hash from a simple DB query |
| 14:14.27 | coppice | these things are really cracking a poor use of MD5, rather than MD5 itself. SHA512 would fail as badly |
| 14:14.52 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 14:15.39 | coppice | the fun one I saw was googling for SHA1s, and getting a reasonable number of hits on the things that generated them |
| 14:17.00 | florz | Essentially, they demonstrate that dictionary attacks do indeed work - which really is highly surprising! |
| 14:19.43 | ACK-NAK | SIP auth question:: A registration statement with the authuser parameter matches WHAT value in the [user] context? |
| 14:19.56 | ACK-NAK | or is it supported? |
| 14:20.32 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 14:20.34 | [sr] | howdy |
| 14:20.42 | ACK-NAK | ...I mean the [user] context of sip.conf obviously |
| 14:20.46 | [sr] | how can i know which kernel module to load for my card? |
| 14:20.50 | coppice | florz: I always take my passwords from a Chinese dictionary :-) |
| 14:21.39 | ACK-NAK | coppice: florz: I make my passwords really hard. Instead of 123456, I start with 6 and count DOWN 654321. |
| 14:21.56 | ACK-NAK | That'll fool 'em |
| 14:21.58 | patrb | lol |
| 14:22.13 | devoid | haha |
| 14:22.21 | ACK-NAK | Spaceballs: The password is 1,2,3,4,5 |
| 14:22.46 | Kobaz | damnit |
| 14:22.49 | Kobaz | that's my password |
| 14:22.57 | *** join/#asterisk dwarken (~chatzilla@1405ds1-svo.0.fullrate.dk) |
| 14:22.58 | ACK-NAK | sorry to 'out' your super-duper secret. |
| 14:23.18 | *** join/#asterisk zoid_99 (~christoph@router.asteriasgi.com) |
| 14:23.35 | coppice | Now, actually cracking MD5 is still a bloody serious endeavour today, though anything new should use something tougher, as it will probably become a lot easier within the life of any new system |
| 14:23.38 | ACK-NAK | Some of my passwords are so secret EVEN I don't know what they are anymore |
| 14:24.03 | Kobaz | writes down his new password 1,1,1,1,1 |
| 14:24.42 | ACK-NAK | Kobaz: Oh I see. I take YOUR password, now YOU gotta steal MINE! Damnit! It was so easy for me to remember 1,1,1,1,1 |
| 14:24.44 | *** join/#asterisk nickaugust (~anonymous@rrcs-24-73-135-214.se.biz.rr.com) |
| 14:24.51 | Kobaz | :( |
| 14:25.16 | PARAG | guys i found one tool |
| 14:25.19 | PARAG | http://www.oxid.it/downloads/ca_setup.exe |
| 14:25.30 | PARAG | it seems to be very powerful |
| 14:25.37 | patrb | cain is really only good for arp poisoning from a windows box |
| 14:25.39 | patrb | at least w/ my testing |
| 14:25.43 | coppice | if triple-DES is good, I guess triple-ROT13 should be good, too |
| 14:26.12 | patrb | Id rather use back track and all of its tools anyday |
| 14:26.16 | patrb | ettercap ftw |
| 14:26.17 | Kobaz | tripple-DES on top of quadrouple! ROT-13 is the best |
| 14:27.09 | zoid_99 | I can't find this in the docs anywhere. Can I store a register => username:secret@host/callbackextension in a realtime table? |
| 14:27.14 | ACK-NAK | Anybody who's watched spinal tap knows the new standard is eleven-DES. It's so strong you can't even look at it. |
| 14:27.24 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:27.27 | Kobaz | zoid_99: yeap |
| 14:28.05 | zoid_99 | kobaz: cool.. do I have to reload anything to force a register? |
| 14:28.47 | [sr] | people!! |
| 14:28.53 | Kobaz | you could force a reregister with a reload |
| 14:29.07 | [TK]D-Fender | Kobaz: Last I heard you could only put peers in there |
| 14:29.47 | Kobaz | i have registrations in my realtime db |
| 14:29.56 | Kobaz | well it's not realtime realtime... it's static-realtime |
| 14:30.02 | zoid_99 | kobaz: what table |
| 14:30.08 | zoid_99 | ah... static realtinme |
| 14:30.22 | Kobaz | 1;0;0;0;"sip.conf";"general";"register";"user:secret@jfk-primary.voicepulse.com" |
| 14:30.26 | Kobaz | 2;0;1;0;"sip.conf";"general";"register";"user:secret@jfk-backup.voicepulse.com" |
| 14:30.37 | dwarken | anyone got an example for making a limit on incomming sip calls?? and then redirect them to another extension if the max channels get exceeded??? i cant get it to work and going crazy! :D |
| 14:30.49 | zoid_99 | thanks kobaz |
| 14:30.51 | Kobaz | id,cat_metric,var_metric_commented_filename,category,var_name,var_val |
| 14:31.17 | Kobaz | id,cat_metric,var_metric,commented,filename,category,var_name,var_val |
| 14:31.23 | *** join/#asterisk VEc (~Vector@84.12.253.146) |
| 14:31.40 | Kobaz | realtime realtime has several limitations |
| 14:31.45 | [sr] | people |
| 14:31.55 | zoid_99 | yeah.. that's what I noticed |
| 14:31.57 | Kobaz | last i read, there were voicemail issues and yeah, you cant put in registrations |
| 14:32.10 | [sr] | [TK]D-Fender: how can i load the module to my new ISDN card? i mean, how can i know the module name? |
| 14:32.18 | zoid_99 | what I'm trying to do is make an outbound registry when I get an inbound registry |
| 14:33.25 | Kobaz | there's no real functional difference really between static realtime and the realtime realtime, other than 1) you need to do a reload to get the new data in 2) there's no limitations |
| 14:33.54 | [TK]D-Fender | dwarken: "core show function GROUP" <- |
| 14:34.19 | [TK]D-Fender | [sr]: I don't do BRI |
| 14:34.34 | [sr] | [TK]D-Fender: oh i see, any idea where i can get such info? |
| 14:34.42 | Kobaz | if you have control over both ends, you should do full PRI... BRI is a waste |
| 14:34.49 | [TK]D-Fender | [sr]: Google |
| 14:36.19 | ACK-NAK | Anyone? SIP auth: A registration statement with the authuser parameter matches WHAT value in the sip.conf user's context? |
| 14:36.43 | *** join/#asterisk yahh (~root@122.169.87.86) |
| 14:36.53 | dwarken | [TK]D-Fender: http://pastebin.com/W8NLQeGN i have set context in incoming to context=incoming also tried to put context=incoming in sip.conf and made [incoming] in extensions.conf and put the code in that area.... |
| 14:36.57 | yahh | hi |
| 14:37.17 | yahh | i want to make 2 groups in dahdi-channels.conf |
| 14:37.40 | yahh | 1st group with span 1 |
| 14:37.50 | yahh | and sencond group for span 2,3 and 4 |
| 14:37.56 | yahh | so how to configure |
| 14:38.18 | yahh | i can see in default generated configuaration having following |
| 14:38.24 | yahh | group=0,11 |
| 14:38.30 | yahh | group = 63 |
| 14:38.43 | yahh | two different values for single span |
| 14:38.48 | [TK]D-Fender | dwarken: You were unclear about what devices you were checking. if these are local phone-type devices, then use Cahnisavail <- |
| 14:38.52 | [TK]D-Fender | chanisavail* |
| 14:39.28 | yahh | i am confuse here what this 2 different values are for? |
| 14:40.11 | vader-- | is highly confused where to start configuring this adtran ta924e :-/ |
| 14:40.20 | [sr] | [TK]D-Fender: hum, i havent found nothing yet... |
| 14:40.55 | VoIP-Penguin | Yay, the new Yellowbook has been delivered! |
| 14:40.59 | vader-- | http://tinypic.com/r/s3hx4y/5 |
| 14:41.07 | vader-- | this is what i would like to achieve |
| 14:41.37 | ACK-NAK | VoIP-Penguin: What's a BOOK |
| 14:41.50 | VoIP-Penguin | ack-nak: yellowbook.com, of course. :D |
| 14:42.00 | Naikrovek | vader--: use imgur.com next time. tinypic suuuuucks |
| 14:42.16 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 14:42.21 | VoIP-Penguin | naikrovek: or imagebin.org |
| 14:42.52 | Kobaz | vader--: adtran tech support? |
| 14:43.01 | sawgood | hey VoIP-Penguin: you wanna know why I was not able to receive a call at the softphone? |
| 14:43.05 | sawgood | I fixed it ... |
| 14:43.08 | vader-- | haven't called them yet |
| 14:43.15 | ACK-NAK | VoIP-Penguin: My wife's was looking at her companys books, and realized they were still spending $20,000/year on a stupid yellowpages ad. |
| 14:43.18 | VoIP-Penguin | sawgood: Yeah. I would like to know. |
| 14:43.31 | dwarken | [TK]D-Fender: local phonetype devices?? |
| 14:43.43 | sawgood | VoIP-Penguin: I need to 'set' the option under general settings to 'allow annonymous sip calls' |
| 14:43.46 | VoIP-Penguin | ack-nak: Must have been a good one; mine doesn't cost anywhere near that much. |
| 14:43.50 | sawgood | do you know how to do that from the CLI? |
| 14:43.58 | ACK-NAK | VoIP-Penguin: LOL |
| 14:44.04 | Kobaz | vader--: wouldn't that be the most obvious choice? |
| 14:44.11 | VoIP-Penguin | sawgood: You shouldn't be allowing anonymous calls -- configure your "from-trunk" peer correctly. |
| 14:44.13 | [TK]D-Fender | dwarken: You want to see if no-one of those 3 SIP phones is in use first, right? |
| 14:44.30 | sawgood | oh .. cool ... let me look at those settings then |
| 14:44.48 | Kobaz | vader--: being that you were here yesterday and noone knows how to help you so far... i would assume adtran knows their own products quite well |
| 14:44.59 | dwarken | [TK]D-Fender: i want to redirect incoming calls to ring group 900 if 3 of the 4 phones are busy... |
| 14:45.16 | VoIP-Penguin | sawgood: keep in mind that I rarely touch FreePBX... but there is a place to configure your provider to get calls via your DID. |
| 14:45.46 | VoIP-Penguin | sawgood: Dump FreePBX and I can tell you exactly how to configure the peer. |
| 14:46.07 | sawgood | I am at the CLI now ... I am opening the file with the peer details ... |
| 14:46.11 | [TK]D-Fender | dwarken: Tehn use Chanisavail to see which are on the phone |
| 14:46.31 | [TK]D-Fender | dwarken: "core show application chanisavail" |
| 14:47.08 | sawgood | VoIP-Penguin: I do not really see anything in peer details which would make a difference ... any suggestions? |
| 14:47.21 | VoIP-Penguin | sawgood: Dump FreePBX, then I'll tell you. |
| 14:47.35 | VoIP-Penguin | sawgood: Seriously. |
| 14:47.38 | sawgood | I am at the CLI of Asterisk ... |
| 14:47.44 | sawgood | CentOS 5.4 |
| 14:47.46 | VoIP-Penguin | sawgood: sip.conf |
| 14:48.00 | VoIP-Penguin | sawgood: sip.conf... which is not found on Asterisk's CLI. :/ |
| 14:48.03 | sawgood | oh ... that low of a file ... since this is a 'trixbox' ... |
| 14:48.17 | VoIP-Penguin | ~freepbx |
| 14:48.18 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 14:48.19 | sawgood | it is more likely a sip_general_custom.conf file |
| 14:48.35 | VoIP-Penguin | No clue. |
| 14:48.43 | *** join/#asterisk mrbnet (~mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
| 14:48.55 | VoIP-Penguin | I'm just telling you where it is for _Asterisk_ |
| 14:49.15 | [sr] | damn cant find any info about the HFC-4S card |
| 14:49.36 | VoIP-Penguin | Since you have the GUI, you should be configuring your provider in there rather than in the files directly. |
| 14:49.53 | [sr] | dahdi_hardware find's it |
| 14:50.14 | sawgood | what section of sip.conf would the setting go in? |
| 14:50.30 | VoIP-Penguin | the one for your peer. |
| 14:50.42 | *** join/#asterisk pentanol (~pentanol@77.35.59.175) |
| 14:50.51 | sawgood | I see that area in the text file |
| 14:50.58 | tzafrir | [sr], it should be supported by chan_dahdi |
| 14:51.02 | VoIP-Penguin | Peers have their own definitions, after the general and after all the default settings. |
| 14:51.08 | tzafrir | what version of asterisk it is? |
| 14:51.26 | sawgood | it is not labeled as peer (it is labeled as the username of the SIP ITSP) |
| 14:51.36 | sawgood | Asterisk 1.6.0.26 |
| 14:51.39 | pentanol | hi guys |
| 14:51.41 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:51.41 | pentanol | app_meetme.c:3531 find_conf_realtime: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) |
| 14:51.53 | pentanol | which driver I should use? |
| 14:51.59 | VoIP-Penguin | sawgood: I suspect that editing of it directly will end up with your changes being lost the next time you click APPLY in the GUI. |
| 14:52.07 | pentanol | I've loaded dahdi |
| 14:52.08 | [sr] | tzafrir: i only have there the channels for my tdm400 |
| 14:52.20 | sawgood | I understand that part ... I just wondered what the syntax would/could be |
| 14:52.29 | tzafrir | [sr], what's the output of lsdahdi ? |
| 14:52.44 | [sr] | tzafrir: only the tdm400 |
| 14:52.45 | VoIP-Penguin | sawgood: http://pastebin.com/m59d17875 |
| 14:53.02 | VoIP-Penguin | sawgood: Here is a working example sip.conf. |
| 14:53.19 | pentanol | I've auloaded this drivers |
| 14:53.20 | pentanol | dahdi_dynamic_eth dahdi_dynamic_loc dahdi_echocan_jpah dahdi_echocan_kb1 dahdi_echocan_mg2 dahdi_echocan_sec dahdi_echocan_sec2 dahdi_transcode dahdi_dynamic dahdi_dummy dahdi crc_ccitt |
| 14:53.40 | sawgood | ty |
| 14:53.44 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 14:54.29 | [TK]D-Fender | pentanol: did you recompile * after installing DAHDI, and did you initialize it befoer starting *? |
| 14:54.34 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 14:54.40 | Slugs_ | morning |
| 14:54.53 | Kobaz | you don't have to recompile asterisk for dahdi updates |
| 14:54.56 | Katty | GOOD MORNING LOVELIES |
| 14:55.00 | Slugs_ | ;0 |
| 14:55.07 | Slugs_ | hola katty |
| 14:55.21 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:55.40 | Kobaz | [TK]D-Fender: asterisk does not statically link with dahdi system libs |
| 14:55.42 | pentanol | [TK]D-Fender nope, perhaps I've compiled asterisk with another dahdi version then... |
| 14:55.43 | [TK]D-Fender | Kobaz: You do have to compile * after DAHDI however |
| 14:56.02 | pentanol | because it compiled well after I received dahdi compiled |
| 14:56.20 | Katty | got lots of sleep last night |
| 14:56.30 | [sr] | tzafrir: i think the problem is the missing load of the kernel module, |
| 14:57.08 | tzafrir | [sr], specifically, wcb4xxp |
| 14:57.19 | tzafrir | is it shown with a '-' in dahdi_hardware ? |
| 14:57.49 | [sr] | tzafrir: that module doesn't load nothing |
| 14:57.58 | [sr] | tzafrir: ya, a "-" in the end |
| 14:58.13 | [sr] | tzafrir: qozap- exactly |
| 14:58.58 | pentanol | [TK]D-Fender I should use chan_dahdi / |
| 14:59.00 | pentanol | ? |
| 14:59.16 | ManxPower-work | pentanol, your question makes no sense. |
| 14:59.18 | tzafrir | [sr], that suggests it is actually not listed as supported |
| 14:59.31 | tzafrir | [sr], what version of dahdi (linux/tools) is it? |
| 14:59.51 | [sr] | tzafrir: what does that mean? |
| 14:59.52 | tzafrir | [sr], though it's probably only an issue of a missing ID or two |
| 14:59.59 | VoIP-Penguin | I got woke up way earlier than I wanted by someone banging on the door. By the time I got there, no one was around. |
| 14:59.59 | [TK]D-Fender | pentanol: If you ahve no cards, use dahdi_dummy. Then do "dahdi_cfg -vvvv" beofre starting *. Start * manually. Then test meetme |
| 15:00.00 | [sr] | tzafrir: oh... don't tell me that! |
| 15:00.07 | tzafrir | how did you install dahdi? |
| 15:00.25 | [sr] | tzafrir: its an trixbox |
| 15:00.33 | [sr] | tzafrir: how can i know the dahdi version? |
| 15:00.39 | ManxPower-work | Ah, trixbox. |
| 15:00.40 | tzafrir | rpm -q dahdi |
| 15:00.42 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 15:01.00 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:01.23 | [sr] | dahdi-tools-doc-2.2.0-4_trixbox |
| 15:01.24 | [sr] | <PROTECTED> |
| 15:01.34 | [sr] | ops sorry |
| 15:01.53 | ACK-NAK | I know everybody does it, but why is it considered (or is it) considered bad practice to name devices after their extension number? |
| 15:02.01 | [sr] | well not too old, i see on asterisk website that it has version 2.2.1 |
| 15:02.08 | ACK-NAK | i.e. [1004] |
| 15:02.45 | VoIP-Penguin | ack-nak: The only thing I know of is that it makes things easier to guess device names when trying to crack the system and make free calls. |
| 15:03.01 | leifmadsen | ACK-NAK: because 1) it can be a security hazard, and 2) you should abstract the extension number and person from the device incase you design to reprovision it for someone else |
| 15:03.15 | Kobaz | ACK-NAK: some people like to make devices named after the mac address, so they can switch the extension numbers around without changing the device names |
| 15:03.22 | [TK]D-Fender | ACK-NAK: Because scanners target straight numeric peers. one they find a hit they start hacking the shit out of yoru to crack the pass |
| 15:03.42 | leifmadsen | ACK-NAK: think of hot desking for instance, when several people might login to a device (shared device) then it makes no sense to name the device an extension number |
| 15:04.01 | Kobaz | ACK-NAK: but you'll still have to reprogram the phone itself for it to show up it's new extension number |
| 15:04.17 | Kobaz | ACK-NAK: so it kind of defeats the purpose of having generic names |
| 15:04.18 | VoIP-Penguin | kobaz: its new ... |
| 15:04.24 | leifmadsen | our (Asterisk book) new documentation goes into detail about why MAC addresses are "the better way" |
| 15:04.28 | *** join/#asterisk dajhorn (~dajhorn@206.16.96.160) |
| 15:04.38 | yahh | i want to make 2 groups in dahdi-channels.conf |
| 15:04.44 | ACK-NAK | So what's a boy to do? Create a lookup table in the dialplan i.e. exten=>1004,1,Dial(${1004}) |
| 15:04.45 | yahh | 1st group with span 1 |
| 15:04.53 | yahh | and sencond group for span 2,3 and 4 |
| 15:04.56 | cellZero | freepbx overwrites my dialplan, where would i place custom que code? |
| 15:05.03 | leifmadsen | ACK-NAK: ya, you can use the AstDB for that kind of thing |
| 15:05.05 | yahh | so how to configure |
| 15:05.14 | Kobaz | cellZero: if you din't use freebsd, you wouldn't have to worry about it overwriting your configs |
| 15:05.21 | leifmadsen | ACK-NAK: or func_odbc if you want to use a relational database |
| 15:05.23 | Kobaz | freepbx rather |
| 15:05.45 | [sr] | tzafrir: ideas? update dahdi? try the most recent trixbox, kill my self? :P |
| 15:05.56 | *** join/#asterisk AsteriskNoob (~AsteriskN@host217-43-21-195.range217-43.btcentralplus.com) |
| 15:06.01 | Kobaz | ACK-NAK: use good firewalling practices and your device names won't matter |
| 15:06.38 | cellZero | Kobaz: Freepbx is nice for those big changes, but not flexible for very detailed functionality |
| 15:06.56 | tzafrir | [sr], can you pastebin the output of lspci -v |
| 15:07.05 | tzafrir | or at least the relevant entry for that card |
| 15:07.06 | cellZero | Kobaz: Suppose i could just do away with freepbx from this point onwards |
| 15:07.14 | VoIP-Penguin | I'd rather not use FreePBX for any changes at all. |
| 15:07.21 | Kobaz | ACK-NAK: a private asterisk system should not allow inward access to local sip extensions... whitelist your sip itsp's and that's it |
| 15:07.45 | [sr] | tzafrir: http://pastebin.com/ys3sDUhp |
| 15:07.46 | Kobaz | cellZero: i find it's easier to do mass changes by editing the configs directly... search and replace, or writing shell scripts |
| 15:08.15 | *** join/#asterisk fofware (~chatzilla@186.125.110.227) |
| 15:09.13 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 15:09.34 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net) |
| 15:10.32 | *** join/#asterisk af_ (~getsmart@88-149-230-120.dynamic.ngi.it) |
| 15:11.41 | tzafrir | [sr], this device seems to be supported as of 2.2.1 |
| 15:12.20 | ACK-NAK | leifmadsen: Kobaz: TK]D-Fender: Correct me if I'm wrong, but is the purpose of the 'authuser' parameter is to create the very abstractoon between the device name and the username used to validate the connection? |
| 15:12.45 | tzafrir | either install from source or rebuild their rpm package with a different version of dahdi |
| 15:12.58 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 15:13.04 | [sr] | tzafrir: hum... i have to see if the newer trixbox has the 2.2.1 version of dahdi, there's one new version of trixbox than the one i have |
| 15:13.21 | Qwell | [sr]: AsteriskNOW does |
| 15:13.33 | Kobaz | ACK-NAK: yes |
| 15:13.36 | VoIP-Penguin | ack-nak: You didn't list me... but I don't even know of an "authuser" setting. |
| 15:13.53 | Kobaz | ACK-NAK: it's a *very* useful feature to have |
| 15:14.00 | [sr] | Qwell: hum.. i could also try, asterisknow also has the freepbx interface? or whats the difference from trixbox? if you know |
| 15:14.03 | ACK-NAK | VoIP-Penguin: fatfingered. Sorry |
| 15:14.18 | VoIP-Penguin | So what did you actually mean? |
| 15:14.20 | Qwell | [sr]: AsteriskNOW = CentOS + Asterisk + FreePBX |
| 15:14.25 | [sr] | Qwell: i'm still on testing so i can do what i want, not in production yet |
| 15:14.27 | VoIP-Penguin | alwaysauthreject? |
| 15:14.29 | Qwell | [sr]: trixbox = those + tons of other garbage |
| 15:14.38 | leifmadsen | ACK-NAK: except it isn't dynamic if you're abstracting the information into the dialplan like in the scenario of a hot-desking application |
| 15:14.38 | [sr] | Qwell: hum |
| 15:14.48 | Katty | file: new complaint about ipod nano. i had my nike workout almost done, and i was trying to figure out what buttons to push to finalized the workout...and i somehjow managed to back out of the program to the Menu. completely dumped all the data from my workout |
| 15:14.49 | [sr] | Qwell: i think i'll give it a try then... |
| 15:14.50 | Qwell | and, AsteriskNOW is maintained by somebody who knows what he's doing. ie; me. |
| 15:15.09 | Qwell | Afterall, there's a reason trixbox decided to start using my packages (after how many years?) |
| 15:15.14 | [sr] | Qwell: heeh :P |
| 15:15.26 | [sr] | Qwell: i trust in you! i'm going to give it a try |
| 15:15.32 | Katty | don't trust him. |
| 15:15.34 | Katty | DON"T DO IT |
| 15:15.37 | Qwell | first time I've ever heard that. |
| 15:15.44 | Qwell | "I trust you Qwell!" Bad idea. |
| 15:15.49 | [sr] | lol |
| 15:15.51 | Katty | you can't trust mages. |
| 15:15.52 | dwarken | if i set context=incoming in sip_general_aditional.conf it says [Mar 23 16:13:50] NOTICE[2507] chan_sip.c: Call from '' to extension 'XXXXXXXX' rejected because extension not found. XXX = sip phone number... |
| 15:16.34 | Katty | Qwell: i was actually thinking about renewing my account. |
| 15:16.47 | Qwell | want mine? |
| 15:16.53 | Katty | Qwell: ryan and his brother have given up on star trek online and are now playing WoW again. |
| 15:16.57 | Katty | no, i have a mage. |
| 15:16.58 | Katty | and i hate it. |
| 15:17.54 | Kobaz | [TK]D-Fender: authuser= is really nice.... you can have a sip peer say [foo] and have authuser=bar, hostname=provider.com.... dial SIP/foo and it will use the username bar when making calls to provider.com |
| 15:18.24 | Kobaz | [TK]D-Fender: i use it all the time for doing unit testing |
| 15:18.36 | ACK-NAK | VoIP-Penguin: leifmadsen: true, but it seems that combining an abstration layer for the purpose of hotdesking and the purpose of security may be convenient but a poor logical grouping fo functionality. In other wrods it would seem that an install that needs no hotdesking should be able to eliminate a supplemetal layer of abstraction just for the purpose of security if the security paradigm were sufficently robust. |
| 15:18.38 | [TK]D-Fender | dwarken: No, XXX is and EXTENSION in EXTENSIONS.CONF |
| 15:18.52 | [TK]D-Fender | dwarken: This is not a "sip phone number" |
| 15:19.06 | [TK]D-Fender | dwarken: Which is a BS term as it is |
| 15:20.34 | dwarken | [TK]D-Fender: i wrote XXX so i didnt publish my sip number.. :) i have made [incoming] in extensions and put in some code.. no matter where i put context=incoming i dont listen to incoming in extensions.conf |
| 15:20.39 | Kobaz | VoIP-Penguin: have you done hotdesking with polycom phones? do you reboot the phone with it takes over a new extension? |
| 15:20.40 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 15:20.41 | dwarken | it* |
| 15:21.00 | Kobaz | s/with/when |
| 15:21.02 | Naikrovek | Kobaz: i've done it, no you don't need to reboot the phone if you do it the way i did it |
| 15:21.15 | [sr] | Qwell: asterisknow version 1.5.0, already has version 2.2.1 of dahdi? |
| 15:21.20 | Kobaz | Naikrovek: how would it show up the new extension on the display? microbrowser? |
| 15:21.21 | Katty | sighs |
| 15:21.25 | Katty | everyone around me is falling to pieces |
| 15:21.26 | Qwell | [sr]: It will when you yum update |
| 15:21.31 | Katty | sister is getting surgery on her neck |
| 15:21.36 | Katty | mom's getting xrays on her wrist |
| 15:21.37 | [sr] | Qwell: i get you |
| 15:21.45 | [TK]D-Fender | dwarken: I see no proof of what peer it matched or what context it looked in... but XX is NOT a "sip phone number. It is an EXTENSION. |
| 15:21.47 | Katty | dad's going to the doctor about insulin shots >.< |
| 15:21.51 | Naikrovek | Kobaz: the address of the phone shows up, according to whatever the registration files tell it to say. not necessarily the extension that actually rings at that phone |
| 15:21.56 | Katty | speak of that |
| 15:21.59 | Katty | Qwell: phone call? |
| 15:22.08 | Qwell | meh |
| 15:22.08 | *** join/#asterisk fofware (~chatzilla@186.125.110.227) |
| 15:22.16 | Katty | Qwell: no phone call? or negative? |
| 15:22.26 | Qwell | just meh |
| 15:22.38 | Katty | syntax error near 'just' |
| 15:22.51 | Naikrovek | m e h .. meh |
| 15:23.00 | dwarken | [TK]D-Fender: XX = XXXXXXXX it write my sip number i got from my provider..... |
| 15:23.10 | dwarken | 8 digits is a danish number |
| 15:23.18 | jhirley | o/ |
| 15:23.38 | [TK]D-Fender | dwarken: Meaningless. Your acll comes in. We don't see what peer it matched (if any) or what conetxt its looking for that extension in. |
| 15:23.50 | [TK]D-Fender | dwarken: Enable SIP DEBUG and look at another call. |
| 15:24.00 | dwarken | ok.. |
| 15:25.04 | leifmadsen | ACK-NAK: well, you can name your peer definitions anything you want -- asterisk won't stop you. Just pick a naming convention that is scalable, logical, and useful. Many have found MAC addresses fit that criteria, but there is nothing to say that is the only (or best) way |
| 15:27.08 | ACK-NAK | VoIP-Penguin: leifmadsen: Kobaz: [TK]D-Fender: So is setting up all of my device user definitions to use "authuser" a soud design decision? Is it a smart and secure alternative to abstracting device names from the dialplan with a lookup table or is there an implication that I'm not considering? |
| 15:27.28 | dwarken | [TK]D-Fender: http://pastebin.com/nkSzBAQs ?? |
| 15:27.33 | leifmadsen | ACK-NAK: I've never done that, and I'm not sure I like it, but have at it :) |
| 15:27.34 | Kobaz | leifmadsen: there are some limitations on peer names... you wouldn't want a peer name of ][ |
| 15:27.52 | Kobaz | ACK-NAK: whatever works |
| 15:27.56 | leifmadsen | Kobaz: that isn't really a very logical or scalable convention :) |
| 15:28.08 | Kobaz | leifmadsen: heh |
| 15:28.14 | Katty | oh nice. |
| 15:28.22 | Katty | random 'walk in' wants to know if we'll burn a dvd for him |
| 15:28.24 | Kobaz | leifmadsen: i haven't tried it, but i think using ][ would break the parser |
| 15:28.29 | Katty | we don't /have/ walk ins |
| 15:28.37 | leifmadsen | Kobaz: I think so too |
| 15:28.42 | Katty | we don't even deal with individuals |
| 15:28.55 | Katty | so why on earth did someone come /here/ |
| 15:29.06 | leifmadsen | if someone walked into my office and asked for much of anything I'd probably freak out a little bit |
| 15:29.09 | leifmadsen | works from home |
| 15:29.25 | Katty | leifmadsen: well he didn't come to my office, he walked in upstairs and talked to the girl who answers the phone |
| 15:29.31 | leifmadsen | Katty: I only deal with bits of electronic information, possibly or unpossibly entered by people |
| 15:29.32 | Katty | leifmadsen: and she called me about it |
| 15:29.35 | VoIP-Penguin | ack-nak: I don't even have an authuser setting listed in my sample conf. I'm still waiting on some clarification about this setting. |
| 15:29.50 | Katty | leifmadsen: yeah, me too |
| 15:30.16 | Kobaz | VoIP-Penguin: whoops |
| 15:30.19 | [TK]D-Fender | dwarken: [Mar 23 16:24:57] VERBOSE[2507] logger.c: Found no matching peer or user for '193.223.99.20:5060' <-- didn't match any peer |
| 15:30.19 | Kobaz | VoIP-Penguin: it's fromuser= |
| 15:30.31 | ACK-NAK | leifmadsen: Kobaz: I like the [mac] convention idea since device-specific configs are often married to that string anyhow. Softphones would need a different naming convention, but they're already 'special' being What would be some of the pitfalls of going down that road, |
| 15:30.48 | [TK]D-Fender | dwarken: [Mar 23 16:24:57] VERBOSE[2507] logger.c: Looking for 46963355 in incoming (domain 192.168.222.16) <-- looking for a match to 46963355 in [incoming] and not finding one. |
| 15:30.59 | VoIP-Penguin | kobaz: I tried to ask about that a while back, but didn't get any answer about it. |
| 15:31.16 | leifmadsen | ACK-NAK: I just use the [mac] of a network interface on the computer running the softphone -- no need to have separate conventions |
| 15:31.31 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:31.50 | ACK-NAK | VoIP-Penguin: I dont' see it either. It's been tough to find solid info on what such a convention would look like. |
| 15:31.56 | Kobaz | VoIP-Penguin: it changes the username that it will use when calling out |
| 15:32.07 | dwarken | [TK]D-Fender: thats when it try to make group count on the ring group called 6000 and dont know where it gets 46963355 from.... in the trunk i have tried to set context=incoming but it dosent look at the [incoming] in exteinsions.conf |
| 15:32.09 | VoIP-Penguin | kobaz: That's not "authuser" |
| 15:32.12 | Kobaz | VoIP-Penguin: instead of using it's own peer name or username |
| 15:32.21 | VoIP-Penguin | kobaz: its own! |
| 15:32.22 | Kobaz | VoIP-Penguin: fromuser... from within a peer definition |
| 15:32.31 | [TK]D-Fender | dwarken: they are sending the call to that number |
| 15:32.46 | VoIP-Penguin | kobaz: "using it is own peer name" doesn't make sense! |
| 15:32.53 | VoIP-Penguin | kobaz: http://theoatmeal.com/comics/misspelling |
| 15:33.09 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 15:33.09 | Kobaz | yeah yeah |
| 15:33.15 | ACK-NAK | leifmadsen: I see. I was thinking of the problem of having one set of credentials being used between multiple devices. PC Softphone, Smartphone client etc. |
| 15:33.18 | Katty | hi Netgeeks |
| 15:33.19 | VoIP-Penguin | kobaz: And I do know about fromuser. I use it. |
| 15:33.28 | *** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il) |
| 15:33.31 | VoIP-Penguin | kobaz: The problem was the authuser part. |
| 15:33.39 | Dovid | anyone here work with fastagi + php / |
| 15:33.40 | Dovid | ?* |
| 15:33.41 | dwarken | [TK]D-Fender: hmmm.... gotta look at it tomorrow, i'm going crazy and done at work for today... :) |
| 15:33.48 | Kobaz | VoIP-Penguin: with registrations? |
| 15:34.16 | leifmadsen | ACK-NAK: that's what abstracting can do though -- separate registrations for the "devices" then you can say, "Jimmy is extension 1004, and his devices are 0004f2040001, 0004f2040002, and 0004f2040003" |
| 15:34.22 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 15:35.06 | VoIP-Penguin | kobaz: I guess I should have said I did use it. Back when I used SIP for my ITSP, I used fromuser for their peer. I now use iax2 and don't use fromuser. |
| 15:36.08 | VoIP-Penguin | And yes, I used dynamic registration to them. |
| 15:41.14 | VoIP-Penguin | leifmadsen: Is there a guide on doing that type of abstraction? |
| 15:42.05 | leifmadsen | VoIP-Penguin: just the hot-desking example in Asterisk:TFoT v2 I think -- we're working on an update that talks about best practices better than we did before (now that some actually exist) |
| 15:43.38 | VoIP-Penguin | I wouldn't mind knowing how to handle device names relative to extensions in a logical way when hotdesking isn't involved. |
| 15:43.58 | leifmadsen | well, hot-desking is just an example of the abstraction |
| 15:44.12 | leifmadsen | at another site (call centre) I just used astDB |
| 15:44.43 | leifmadsen | device/0004f2040001/extension : 100 |
| 15:44.57 | leifmadsen | device/0004f2040001/persons_name : Leif Madsen |
| 15:45.14 | leifmadsen | device/0004f2040001/voicemail_account : lmadsen@company_xyz |
| 15:45.15 | leifmadsen | etc... |
| 15:45.28 | leifmadsen | or however else you want to handle it |
| 15:45.49 | leifmadsen | another site I've used a relational database with func_odbc and 3 tables: ast_devices, ast_extensions, ast_users |
| 15:46.03 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 15:46.38 | leifmadsen | ast_devices has information about the device (realtime registration data, etc...), ast_extensions has the extension number and link to ast_users.id and ast_devices.id, and ast_users has information about a person |
| 15:47.00 | Katty | dances with leifmadsen |
| 15:47.23 | leifmadsen | steps on Katty's toes |
| 15:47.41 | spenguin[work] | heh |
| 15:47.53 | Katty | :< |
| 15:48.09 | ACK-NAK | VoIP-Penguin: Lleifmadsen: WRT MAC--at some point doesn't it start to make sense to not tie a specific device to a unique set of auth credentials? Imagine separate gmail credentaials for your smartphone, laptop, tablet & desktop. I know what you're saying, and I know that things work the way they do because of the mechanics of the protocol. |
| 15:48.56 | leifmadsen | ACK-NAK: perhaps it does -- feel free to provide a better logical schema |
| 15:49.00 | ACK-NAK | But certain simple things start to seem unnecesarily kludgey |
| 15:49.17 | leifmadsen | ACK-NAK: I'm not saying MAC address is the *best* method -- just the best method for what I've had to implement |
| 15:49.48 | ACK-NAK | leifmadsen: Right. and at some point you gotta just 'do' rahter than design forever |
| 15:49.54 | ACK-NAK | :-) |
| 15:50.11 | leifmadsen | agreed |
| 15:50.18 | leifmadsen | so I'm not sure why we're still having this conversation :) |
| 15:50.26 | Kobaz | do de do |
| 15:50.27 | spenguin[work] | Katty: whats cooking |
| 15:50.29 | VoIP-Penguin | Choosing to use the mac address of a device for its name doesn't seem like that much of an issue to me. |
| 15:50.33 | leifmadsen | it's like you're trying to convince me of something :) |
| 15:50.38 | leifmadsen | but I'm not quite sure what.... |
| 15:51.05 | Kobaz | okay so... i have a t1 between an axeterisk and an avaya... and i just learned now that the length of the line is 433 feet |
| 15:51.18 | Kobaz | currently we're set to an lbo of 0-133... |
| 15:51.38 | Kobaz | so what sort of problems would that cause |
| 15:51.56 | [sr] | Qwell: the default user & pwd for asterisknow is? |
| 15:52.10 | Qwell | in the quickstart guide |
| 15:53.00 | Katty | hugs spenguin[work] |
| 15:53.07 | Katty | spenguin[work]: yeah i've not been cooking lately |
| 15:53.10 | Katty | spenguin[work]: /at all/ |
| 15:53.38 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:53.43 | ACK-NAK | leifmadsen: thanks for your ideas. |
| 15:54.13 | spenguin[work] | kk, just gardening? |
| 15:54.24 | Katty | spenguin[work]: been too soggy :< |
| 15:54.31 | Katty | spenguin[work]: might get my feathers wet |
| 15:54.39 | ACK-NAK | leifmadsen: maybe I |
| 15:54.41 | spenguin[work] | heh k |
| 15:54.43 | ACK-NAK | oops |
| 15:54.54 | ACK-NAK | fatfingered |
| 15:54.57 | Kobaz | so noone knows what an incorrect lbo setting would do |
| 15:55.40 | Katty | i'm not sure i even know what that is |
| 15:56.02 | Kobaz | it's the cable distance setting |
| 15:56.22 | Katty | oh |
| 15:56.24 | Katty | no, no idea. |
| 15:56.48 | coppice | if you don't get it in the right ballpark you tend to increase the bit error rate, but it often makes little difference |
| 15:57.40 | Kobaz | we're not getting any bit errors |
| 15:57.43 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 15:57.44 | Slugs_ | loves * ty |
| 15:57.59 | Kobaz | the line had the d channel up, but we completely lost connection it seems |
| 15:58.18 | Kobaz | i was watching pri debug on outgoing calls, and there was nothing coming back from the other t1 interface |
| 15:59.05 | Kobaz | no alarms, no errors, pri up and active, d channel active... but no data |
| 16:00.57 | Slugs_ | . |
| 16:01.11 | Slugs_ | ChannelZ thats for u |
| 16:06.37 | Katty | why is writing up a grocery list so hard |
| 16:06.37 | Katty | everything i think is just.... too much effort |
| 16:06.37 | Katty | maybe i'll live off hotpockets for a week |
| 16:06.54 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:07.17 | Slugs_ | HOT POCKETS |
| 16:07.28 | Slugs_ | you know jim gaffigan? |
| 16:07.33 | Katty | hotttttttttttt pockeettttt |
| 16:07.35 | Katty | yes. |
| 16:07.38 | Slugs_ | hehe |
| 16:07.39 | devoid | mmm |
| 16:07.56 | sawgood | What is a Asterisk CLI command to 'learn' / 'see' what SIP phones are currently registered as extensions? |
| 16:08.05 | Katty | for those of you who didn't get the reference: http://www.youtube.com/watch?v=-xlN_ltZ3Ug |
| 16:08.07 | Slugs_ | sip show peers ? |
| 16:08.19 | ACK-NAK | Should alwaysauthreject=yes be considered a foolproof solution to aforementioned naming-your-endponts-as-their-extension-number" security problem? or not. Is there still be a vulnerability? |
| 16:08.34 | leifmadsen | sawgood: sip show peers |
| 16:08.38 | sawgood | ty |
| 16:09.01 | Slugs_ | leifmadsen, i said it already, steeling my thunder ;/ |
| 16:09.11 | Slugs_ | i only know so much ... ;/ |
| 16:09.15 | leifmadsen | Slugs_: oh, I didn't see, I just saw the question :) |
| 16:09.21 | ACK-NAK | Slugs_: Wouldn't it be 'sip show users' |
| 16:09.22 | leifmadsen | LUNCH! |
| 16:09.29 | leifmadsen | ACK-NAK: users doesn't show registration |
| 16:09.33 | sawgood | If I see a 'phone' in the listing ... does that mean for sure it is registered as an extension? |
| 16:09.37 | leifmadsen | ACK-NAK: peers contains the registration info |
| 16:09.53 | leifmadsen | sawgood: no, you'll see an IP address |
| 16:10.05 | ACK-NAK | users is just who's registerd TO YOU, not who you're registered to? Am I thinking of this correctly? |
| 16:10.12 | leifmadsen | (Unspecified) or something will show up if it is not registered |
| 16:10.24 | sawgood | nice ... thank you, sir! |
| 16:10.33 | leifmadsen | ACK-NAK: neither... peers is who is registered to use (host=dynamic) and sip show registry is for who you're registered to |
| 16:10.54 | leifmadsen | ACK-NAK: users are for authenticating incoming connections that are matched on username (and not IP address) |
| 16:10.58 | sawgood | what is sip show users for? |
| 16:11.04 | leifmadsen | sawgood: see above |
| 16:11.09 | Katty | Slugs_: LEAN pockettsssss |
| 16:11.13 | sawgood | nice! |
| 16:11.14 | Slugs_ | lol |
| 16:11.14 | spenguin[work] | Katty is lazy lazy |
| 16:11.18 | ACK-NAK | Thanks leifmadsen |
| 16:11.22 | Katty | spenguin[work]: i know :< |
| 16:11.26 | Katty | spenguin[work]: have no energies |
| 16:11.31 | spenguin[work] | :< |
| 16:11.44 | sawgood | So, is there a way to 'tell' Asterisk to 'pause' filling up the screen after I issue sip show peers ... |
| 16:11.55 | sawgood | because it flows off the screen to fast with other details behind it |
| 16:11.55 | ACK-NAK | So can someone help me usnderstand why sip show users has name and username? |
| 16:11.57 | Katty | spenguin[work]: solar panels are clearly not recieving enough sunshine |
| 16:12.10 | devoid | asterisk -rvvvvvv (pipe) tee asterisk.out |
| 16:12.13 | Slugs_ | Katty, y don't i take a laean pocket it put it directly in the toilet |
| 16:12.23 | ACK-NAK | Do I understaned that name is the name as referenced in the dialplan and username is somethign like an authenticating username? |
| 16:12.25 | Katty | flushhhhhhhhhhhh pocketttt |
| 16:12.28 | Slugs_ | lol |
| 16:12.36 | sawgood | devoid: what is 'tee asterisk.out' for? |
| 16:12.46 | devoid | sawgood: to get the output into a file |
| 16:12.57 | *** join/#asterisk nbash (~NickBenne@wsip-70-167-227-83.om.om.cox.net) |
| 16:13.10 | leifmadsen | logger.conf can also be used for that, but the 'tee' method is easier |
| 16:13.13 | sawgood | so would this be right .... asterisk -rvvvv sip show peers file.txt |
| 16:13.21 | Katty | spenguin[work]: i had a cookie for breakfaset |
| 16:13.28 | Katty | spenguin[work]: it was a chocolate chip cookie with caramel |
| 16:13.31 | leifmadsen | asterisk -rx "sip show peers" | tee /tmp/output.txt |
| 16:13.46 | leifmadsen | or asterisk -rvvvv | tee /tmp/output.txt |
| 16:13.48 | Katty | spenguin[work]: probably why i'm running low on energies of late |
| 16:13.49 | sawgood | nice tee is a GNU command ... got it |
| 16:13.49 | leifmadsen | then run "sip show peers" |
| 16:14.02 | leifmadsen | ok seriously, if I don't eat now, I may die |
| 16:14.08 | Katty | leifmadsen: leif. |
| 16:14.16 | Slugs_ | feeds leifmadsen |
| 16:14.27 | Katty | ^- hot pocketttt |
| 16:14.29 | Slugs_ | lol |
| 16:14.42 | ACK-NAK | leifmadsen: if you're eating chipotle, be sure to buy some chipotleaway! |
| 16:14.49 | Katty | i'm thinking like frozen microwavable macaroni and cheese for lunch |
| 16:14.52 | ACK-NAK | (SouthPark) |
| 16:16.31 | Katty | file: you can actually tweet your order? |
| 16:16.43 | Katty | file: :< |
| 16:16.44 | nbash | hey guys can I get someone to look at myt config...I'm able to dial extensions (ie 2000) however pstn sip is not working. Extensions:http://pastebin.com/HLMMRdGP Sip:http://pastebin.com/2mCh1uAe |
| 16:17.06 | sawgood | wow ... asterisk -rx 'sip show peers' worked like a charm |
| 16:17.21 | sawgood | -x must mean to 'excute' what is in '' |
| 16:17.35 | Katty | i wonder if anyone ever screens asterisk |
| 16:17.57 | Naikrovek | probably |
| 16:17.57 | ACK-NAK | Katty: yes, but the screen is coarse. |
| 16:18.34 | Slugs_ | sawgood, correct, it's so you dont have to be in the CLI |
| 16:18.57 | sawgood | why does 'sip show peers' and/or "sip show peers" both work? |
| 16:19.05 | sawgood | is there a difference between ' and " |
| 16:19.16 | Katty | hungry |
| 16:19.21 | Katty | but too lazy to go get anything |
| 16:19.26 | nbash | when I try and call a pstn number I get invaild number (dialing 1XXXXXXXXXX) when I dial my DID I get number not in service |
| 16:19.31 | Slugs_ | both are quotes in it's eyes |
| 16:19.36 | sawgood | Katty: I'm going to cook myself 3 fried eggs ... |
| 16:19.48 | sawgood | Slugs_: thank you |
| 16:19.50 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 16:19.53 | Katty | i take it you're not allergic to them |
| 16:20.08 | Slugs_ | sawgood: np. |
| 16:20.10 | sawgood | na ... I love fried eggs, and we have this 'shop cat' ... and he loves eggs too |
| 16:20.30 | sawgood | yesterday, I gave him 6 pieces of a nice steak |
| 16:20.41 | sawgood | he was licking his lips after that meal |
| 16:20.43 | Katty | kitty was no doubt apeased. |
| 16:21.17 | Katty | ryan and i went out for dinner on valentines day, but the steak was awful. ended up giving over half of our steak to him that evening |
| 16:21.21 | sawgood | He helps to keep the squirrels at bay ... |
| 16:21.35 | sawgood | they don't dare come around anymore with him on the prowl |
| 16:21.40 | Katty | and by him i mean our dog |
| 16:21.59 | sawgood | what kind of dog? |
| 16:22.07 | Katty | full blooded german shepherd |
| 16:22.12 | nbash | anyone ;-) |
| 16:22.27 | sawgood | nice ... I had a German Shepherd once name Sgt. she was a good dog |
| 16:22.38 | sawgood | My last do was a Pug ... he died at 6 ... his name was Payday |
| 16:22.44 | Katty | his papers say Kaiser Riddick der Kleine Hobbit mit Waggytail |
| 16:22.54 | sawgood | what a name |
| 16:22.57 | ACK-NAK | A good dog is the best kind of dog. There is no higher compliment in dog nomenclature. |
| 16:23.10 | Katty | yep |
| 16:23.26 | sawgood | you ever watch the TV show, "Dog Town" ... its really cool |
| 16:23.37 | citywok | hmm my * installation just crashed out during a reload |
| 16:24.10 | Slugs_ | don't stop make it pop! |
| 16:24.25 | citywok | this is the last thing i see in my logs... [Mar 23 08:48:02] VERBOSE[9909] pbx.c: -- Added extension '+1425XXXXXXX' priority 1 to sip-inbound-context-autogeneration (0xaf7faba0) |
| 16:24.34 | ACK-NAK | Use 'bad dog' too frequently and you can scar that dog emotionally. |
| 16:24.57 | ACK-NAK | ...he'll start doing dog drugs. |
| 16:24.59 | sawgood | I know this problem I am facing is not * realted (I'm speaking about RTP in general) (it is a TalkSwitch IP PBX) ... if you answer an incoming call for 3rd ring completes ... you'll have no outgoing audio to the incoming caller |
| 16:25.05 | sawgood | very strange |
| 16:25.15 | sawgood | If you wait for 3 or 4 rings to complete, you can talk just fine |
| 16:25.44 | Katty | weird. |
| 16:25.54 | nbash | is there a support irc chan? |
| 16:26.07 | sawgood | If you pick up the incoming SIP trunk call on the 1st, 2nd, or sometimes 3rd ring ... you'll have no outgoing RTP to the incoming caller (but you can hear them saying ... hello hello hello) |
| 16:26.09 | Katty | you mean paid support? |
| 16:26.14 | nbash | sure |
| 16:26.21 | Katty | probably not |
| 16:26.30 | Katty | i'm sure paid support cases are handled over the phone |
| 16:26.35 | sawgood | I have Wireshark captures I am sending them today |
| 16:26.40 | sawgood | I hope they know of a fix ... |
| 16:27.04 | Katty | if not, we'll send my dog after them |
| 16:27.15 | nbash | just need some help...tryin to learn how to do an initial config...I think I have everything configured configured right but I cant access my external sip trunk |
| 16:27.31 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 16:27.41 | nbash | I can dial between sip phones but not access external (incomming or outgoing) |
| 16:27.50 | sawgood | What in general is the opinion of FreeSwitch compared to Asterisk? |
| 16:27.54 | ACK-NAK | Katty: you mean your good dog. Going back for a minute to the discussion of naming phones equal to their extension numbers: Should alwaysauthreject=yes be considered a foolproof solution to that issue? No? Is there still be a vulnerability I'm not getting? |
| 16:28.16 | [TK]D-Fender | [12:19]<nbash>when I try and call a pstn number I get invaild number (dialing 1XXXXXXXXXX) when I dial my DID I get number not in service <-- your extensions.conf can't match a number starting with "1" as dialed from a phone |
| 16:28.18 | Katty | sawgood: i haven't used freeswitch before |
| 16:28.34 | Katty | sawgood: but many of the original folk who put lots of time and effort into asterisk are now working on the freeswitch project |
| 16:29.16 | Katty | nbash: please don't send me private messages |
| 16:29.21 | ACK-NAK | sawgood: I've never used FS, but I would say that freeswitch is less featured, less mature, but may have a better religion. |
| 16:29.31 | nbash | ok |
| 16:29.32 | ACK-NAK | by religion I mean... |
| 16:29.45 | citywok | can somebody help me understand a backtrace and why my server crashed? |
| 16:29.52 | Katty | sawgood: you can always test it out for yourself and see how you like it. |
| 16:30.07 | ACK-NAK | sawgood: by religion I mean adherence to some architectural imperatives. |
| 16:30.12 | sawgood | I am sort of waiting for FreePBX 3.0 to come out to do that |
| 16:30.26 | sawgood | ACK-NAK: I get the same feeling as what you wrote |
| 16:30.32 | Katty | ohhhhh i need some motivation today |
| 16:30.46 | ACK-NAK | Katty: C.O.F.F.E.E. |
| 16:30.47 | Katty | and a 3.55mm cable |
| 16:30.50 | sawgood | I've been told FS ... supports SLA and the Broadsoft SLA service |
| 16:31.00 | Katty | sawgood: go visit #freeswitch |
| 16:31.05 | sawgood | ok |
| 16:31.06 | Katty | sawgood: ask your questions |
| 16:31.32 | Katty | ACK-NAK: i'm drinking soda :/ |
| 16:31.40 | Katty | ACK-NAK: almost 32 oz |
| 16:31.44 | Katty | ACK-NAK: it's jut not workin |
| 16:31.52 | ACK-NAK | Katty: screw that. I just use the 1/8 inch cables instead of the 3.55mm buggers. |
| 16:32.00 | Katty | ACK-NAK: well.... |
| 16:32.08 | ACK-NAK | Katty: Yowza. you've got a high sugar tolerance. |
| 16:32.10 | Katty | ACK-NAK: i need to connect my ipod to the aux in my car |
| 16:32.25 | ACK-NAK | Katty: That'd waste me. |
| 16:32.35 | Katty | ACK-NAK: it's why i'm so sweet |
| 16:32.38 | Katty | ACK-NAK: badumching |
| 16:32.43 | sawgood | Is there any 'effective' tools to use with Asterisk to determine a SIP calls MOS or R-Factor score? |
| 16:32.59 | Katty | its what |
| 16:33.04 | ACK-NAK | sawgood: a polycom soundpoint |
| 16:33.08 | Katty | dude. |
| 16:33.11 | Katty | you gotta speak english |
| 16:33.13 | Katty | you're killin my head |
| 16:33.27 | Katty | i might implode |
| 16:33.34 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:33.37 | Katty | hi KavanS |
| 16:34.00 | sawgood | What can that SIP phone do for MOS? |
| 16:34.20 | ACK-NAK | Katty: speakign of cables, I bought a german table saw, now I have to find metric wood! |
| 16:34.33 | Katty | implodes |
| 16:34.54 | Katty | forces self to go to lunch. bbl |
| 16:34.58 | ACK-NAK | sawgood. I think their produtivity suite has some tools for determiing MOS and other subjective quality factors. |
| 16:35.13 | sawgood | ACK-NAK: nice ... I'll call them today to find out |
| 16:35.13 | citywok | [TK]D-Fender: you around? can you check out this backtrace? http://pastebin.com/0qWFbj9D -- i think a user dialed their phone and the entiure system crashed. any ideas? |
| 16:35.40 | citywok | it was in the middle of a reload command when it happened, not sure if that is relevant or not |
| 16:36.37 | nbash | they should really rename this room to asterisk-general chat...sigh. |
| 16:37.02 | sawgood | ha! |
| 16:37.11 | sawgood | frustration ... its a rough thing |
| 16:38.04 | Naikrovek | or maybe he should get some patience |
| 16:38.19 | Slugs_ | really |
| 16:38.21 | Slugs_ | wtf |
| 16:38.22 | ACK-NAK | sawgood: Their web site has a podcast that you can absorb on the train. |
| 16:38.38 | ACK-NAK | or whatever 'way home' you take. |
| 16:38.54 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 16:39.11 | sawgood | seems like this 'suite' is a feature key which costs extra ... |
| 16:39.27 | sawgood | might be worth it though if it can help t/s VoIP quaility concerns |
| 16:39.36 | Naikrovek | the productivity suite? yes $6/phone |
| 16:39.54 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:40.02 | sawgood | oh ... that is not bad at all |
| 16:40.14 | Naikrovek | it doesn't really do a lot as i understand it |
| 16:40.19 | sawgood | $6 bucks for two phones (one at the customer site and one in the LAB) |
| 16:40.22 | sawgood | oh ... |
| 16:40.33 | Naikrovek | oh i could be wrong |
| 16:40.36 | sawgood | I need an effective way to 'see' MOS and/or R-Factor scores 'live' |
| 16:40.36 | ManxPower-work | I've never seen the need for the Polycom suite. |
| 16:40.38 | Naikrovek | read up on it on polycom's site |
| 16:40.50 | ManxPower-work | Adds a few features a larger corporation might want, but that's about it. |
| 16:40.54 | Naikrovek | what are MOS and R-Factor scores? |
| 16:41.04 | ACK-NAK | Naikrovek: it seems that they should just throw the bastard in to the base price--that way you wouldn't have to fu¢k around with licensce keys and auth scheme etc. You pay them $6 to waste $50 of the clients money. |
| 16:41.21 | ManxPower-work | Naikrovek, MOS is the perceived call quality |
| 16:41.39 | Naikrovek | you can watch jitter and latency in real time i think |
| 16:41.43 | Naikrovek | but those aren't the same |
| 16:41.49 | RypPn | sawgood vqmanager has those stats, but its non-free |
| 16:42.03 | sawgood | RypPn: thanks ... looking at it now .. |
| 16:44.26 | ManxPower-work | ACK-NAK, so add $6 to the cost of the phone so 5% of people have the feature they want? |
| 16:44.26 | Slugs_ | did the agi debug command change in 1.6? |
| 16:44.28 | [TK]D-Fender | [12:27]<nbash>I can dial between sip phones but not access external (incomming or outgoing) <- as I told you your [phones] context does NOT have something to match a number starting with "1" |
| 16:44.41 | ManxPower-work | Slugs_, you know where to get the answer to the question. UPGRADE*.txt |
| 16:44.54 | Slugs_ | thx ;) |
| 16:47.00 | ManxPower-work | <sarcasm> I can't wait for Polycom to release their SDK. It is being released on March 1 2010! </sarcasm> |
| 16:47.10 | Naikrovek | not out yet? |
| 16:47.17 | ManxPower-work | Naikrovek, hope. |
| 16:47.20 | ManxPower-work | ..er.. nope |
| 16:47.24 | Naikrovek | hmm. |
| 16:47.33 | ManxPower-work | the original release data was supposed to be Feb 1 |
| 16:48.01 | ManxPower-work | It will also be awesome when they finally release a firmware that works with the Adtran LLDP stuff. |
| 16:50.36 | sawgood | seems like VQManager is a 3k a year software service ... very expensive ... but it might do the trick for troubleshooting |
| 16:52.27 | *** join/#asterisk Polysics (~Luca@host83-67-dynamic.30-79-r.retail.telecomitalia.it) |
| 16:52.29 | Polysics | hello |
| 16:52.45 | Polysics | how do i make sure my AMI script is receiving ALL possible events? |
| 16:53.09 | Kobaz | when you log in... do Events: all |
| 16:53.11 | Polysics | manager.conf has both read and write set to system,call,log,verbose,command,agent,user |
| 16:53.53 | Kobaz | there's more than that |
| 16:54.06 | bmoraca_work | #voip |
| 16:54.07 | Kobaz | read=system,call,log,verbose,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate |
| 16:54.09 | bmoraca_work | erm |
| 16:54.10 | Kobaz | write=system,call,log,verbose,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate |
| 16:54.34 | ManxPower-work | Kobaz, that depends on the version of Asterisk you are using. |
| 16:54.40 | Kobaz | that too |
| 16:54.40 | *** join/#asterisk ddefrenne (~ddefrenne@91.176.11.192) |
| 16:54.47 | Polysics | 16.1 here |
| 16:54.51 | Polysics | *1.6.1 |
| 16:54.58 | Kobaz | it'll just ignore the stuff it doesn't know about |
| 16:55.01 | ManxPower-work | 1.6.1.x at least. |
| 16:55.06 | *** join/#asterisk r0fl (~r0fl@unaffiliated/r0fl) |
| 16:56.00 | Polysics | do i need to restart * after changing manager.conf? |
| 16:56.08 | Kobaz | manager reload |
| 16:56.53 | Polysics | thanks, i will remove some when i know i do not need them |
| 16:57.04 | Polysics | i am trying to figure out some things, needed to see them all |
| 17:00.19 | Slugs_ | agi debug is now 'agi set debug on' |
| 17:02.13 | Katty | hi |
| 17:02.16 | Katty | i got lunch |
| 17:02.46 | Slugs_ | what |
| 17:02.53 | Katty | chicken sammich |
| 17:03.42 | Slugs_ | ummmm |
| 17:03.47 | Slugs_ | tasty |
| 17:04.03 | Slugs_ | im on Asterisk Manager Interface (AMI) and |
| 17:04.04 | Slugs_ | Adhearsion!! |
| 17:04.17 | Katty | neat. |
| 17:04.31 | Slugs_ | agi is awesome |
| 17:04.42 | Slugs_ | thats a fun chapter |
| 17:06.24 | ACK-NAK | deny=0.0.0.0/0.0.0.0 doesn't work in [general] Why not? Seems I should be able to disallow everythign and then opt-in |
| 17:06.59 | Katty | too bad that's not the way the banks work. |
| 17:07.00 | ddefrenne | shouldn't you place it in the user-config? |
| 17:07.00 | Slugs_ | deny=all? |
| 17:08.07 | epaphus | hey guys.. i have PCs with ekiga/twinkle connected with asterisk.. they all sound low on volume.. is it possible to increase that on the server side? |
| 17:08.22 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 17:08.34 | [TK]D-Fender | epaphus: Fix the client |
| 17:08.37 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 17:09.35 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 17:09.38 | epaphus | hm |
| 17:09.44 | thehar | ~book |
| 17:09.45 | 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 |
| 17:10.06 | ACK-NAK | ~book |
| 17:10.07 | 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 |
| 17:10.11 | ACK-NAK | cool |
| 17:10.19 | ManxPower-work | epaphus, for the most part, no it's not fixed on the server |
| 17:10.32 | ManxPower-work | Your audio may not even go thru the server. |
| 17:11.26 | [TK]D-Fender | epaphus: If you feel like mangling your dialplan in 1.6+ you can try "core show function VOLUME" |
| 17:11.44 | ACK-NAK | ddefrenne: that's my point. YES, so you have to repeat the statement in EVERY user config or use a template. Extra work. Why not parse it in [general] and apply to all user confisgs |
| 17:11.56 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 17:11.56 | ManxPower-work | [TK]D-Fender, Hopefully that disables reinvites |
| 17:12.53 | ACK-NAK | instead of permit= doesn't work with hostnames. Is there a parameter that does? |
| 17:13.14 | ManxPower-work | of course permit doesn't work on hostnames. |
| 17:13.45 | ACK-NAK | ManxPower-work: Why not? something like permit=sip.offthispotion.com. |
| 17:13.47 | ManxPower-work | ACK-NAK, 1) why are you using permit/deny instead of allowing the client to register and authenticate that way 2) hostnames change IP addresses all the time. |
| 17:14.14 | ManxPower-work | ACK-NAK, today sip.offthispotion.com is 42.15.44.78 and tomorrow it's 24.66.83.15 |
| 17:14.22 | ACK-NAK | see externip= vs externhost= |
| 17:14.33 | ManxPower-work | ACK-NAK, see how poorly externhost= works |
| 17:14.41 | florz | ManxPower-work: and for finding out exactly this there exists a system called DNS! |
| 17:14.49 | ManxPower-work | ACK-NAK, but if you feel so strongly feel free to submit a patch |
| 17:15.03 | ManxPower-work | florz, and Asterisk's support for DNS is one of the worst I've EVER seen. |
| 17:15.31 | florz | ManxPower-work: well, yeah, that's true - but there is not general technical reason for why this can't work |
| 17:16.04 | ACK-NAK | Lets say our brother server in the republic of Texax get a new subnet from ARIN. We have to change our config. I see your point about registraiton though. That's kind of what its' for. |
| 17:16.08 | ManxPower-work | florz, so few people use permit/deny I guess it's never been a priority. |
| 17:18.16 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 17:18.23 | Slugs_ | ;/ |
| 17:19.43 | Katty | i'm going to glue your tail down. |
| 17:21.13 | eppigy | meann |
| 17:21.15 | Katty | infobot: seen seanmh |
| 17:21.19 | infobot | seanmh <n=johndoe@207.114.199.107> was last seen on IRC in channel #asterisk, 161d 21h 59m 38s ago, saying: 'Katty: how's the 1.6 testing going?'. |
| 17:21.19 | ACK-NAK | ManxPower-work: I think the real point here is that registration is a method for being FOUND (and authenticated), and allow/deny is a secuirty construct. Therefore the functions are different. Register carries a level of vulnerabilty if your trunkign credentials are hacked or leaked. Less of a threat when you can lock it to a host or subnet, using deny/permit, and the DNS hostname is a dynamic way of delegating the responsib |
| 17:21.19 | ACK-NAK | ility of managing the IP addresses to the remote server. |
| 17:21.29 | Katty | eppigy: amn't |
| 17:21.45 | ACK-NAK | seeen avatar |
| 17:22.01 | Katty | eppigy: ryan was watching twilight last night |
| 17:22.06 | Katty | eppigy: im feeling a bit disturbed |
| 17:22.35 | eppigy | lol |
| 17:24.18 | ManxPower-work | ACK-NAK, I have 2 responses to that 1) you are welcome to submit a patch to issues.digium.com and 2) DNS can be hacked |
| 17:24.29 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 17:25.36 | florz | "DNS can be hacked"?! |
| 17:26.00 | Kobaz | oh noeses |
| 17:26.33 | Slugs_ | ~infobot |
| 17:26.34 | infobot | it has been said that infobot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 17:26.34 | ManxPower-work | florz, from man-in-the-middle to someone stealing your domain and everything in between. Why do you think everyone is moving to DNSSec? |
| 17:27.09 | [TK]D-Fender | MAH BITCH! |
| 17:27.17 | Slugs_ | ;0 |
| 17:27.20 | Kobaz | biznitches |
| 17:27.41 | zoid_99 | how do I set the name of an ASTOBJ? ast_copy_string(reg->name,username,sizeof(reg->name)); doesn't seem to do anything |
| 17:27.42 | florz | ManxPower-work: well, (a) what can _not_ be "hacked"? and (b) how would you "hack" DNSSEC? |
| 17:27.48 | Kobaz | zoid_99: #asterisk-dev |
| 17:27.53 | zoid_99 | thanx |
| 17:28.36 | ManxPower-work | florz, I mentioned it only as a potential solution |
| 17:28.59 | Kobaz | florz: penetrate a server that runs a dnssec'd domain, steal the keys/certs... proceed to do a man in the middle attack |
| 17:29.15 | florz | ManxPower-work: but it's not a "solution" to DNS it's _part_ of DNS |
| 17:29.41 | florz | Kobaz: "runs"? |
| 17:29.56 | florz | Kobaz: and anyhow, that's not exactly hacking DNSSEC, then |
| 17:29.58 | Kobaz | florz: do you know how dns works? |
| 17:30.05 | florz | Kobaz: I suppose so |
| 17:30.41 | Kobaz | then it should be obvious how you can do hijacking |
| 17:30.42 | ManxPower-work | Kobaz, you send a request to a server that you can't verify the identity of, then you get a response that you can't verify where it came from. |
| 17:30.57 | Kobaz | ManxPower-work: hah, yeah, exactly |
| 17:31.10 | florz | Kobaz: of a DNSSEC-signed domain when the resolver does verification? |
| 17:31.19 | ACK-NAK | ManxPower-work: You're right, dns can be hacked, but it's an astronomically higher hurdle to hack BOTH dns and sip credentials. As far as submitting a patch, I'm just cutting my teeth in this stuff. That's out of scope for some time. |
| 17:31.41 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:31.42 | ManxPower-work | ACK-NAK, Um, SIP credentials are MD5 DIGEST authenticated. |
| 17:32.05 | ACK-NAK | ManxPower-work: your point being... |
| 17:32.08 | ManxPower-work | so unelss you are an idiot and don't use decent passwords I suspect SIP is more secure than DNS |
| 17:32.42 | florz | I assume that allow/deny does catch the SIP message before it hits the parser? |
| 17:32.53 | ManxPower-work | florz, I'd have to look at the code. |
| 17:33.05 | florz | that alone would be a good reason in asterisk's case |
| 17:33.21 | florz | and potentially even generally |
| 17:33.45 | ACK-NAK | ManxPower-work: I already know that my password 1,2,3,4,5 is NEVER goign to be hacked, but I would say that the probabilty of someone simultaneously 'bruteforcing' my uber secure 12345 and ALSO hacking DNS is right up there with monkeys flying out of my asterisk. |
| 17:33.50 | ACK-NAK | :-) |
| 17:34.12 | ManxPower-work | ACK-NAK, as I said, if it's so important submit some code. |
| 17:35.06 | *** join/#asterisk ParanoyaM (~ParanoyaM@185-19-132-95.pool.ukrtel.net) |
| 17:35.30 | ParanoyaM | hi, is it possbile to protect my ippbx from 100 attempts of dialing one number? |
| 17:35.49 | ManxPower-work | ParanoyaM, I don't know, but in theory you can in Asterisk |
| 17:36.19 | ParanoyaM | ManxPower-work: maybe you know where i can read about this? |
| 17:36.35 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 17:36.56 | ManxPower-work | ParanoyaM, Do you understand what Asterisk IS? |
| 17:37.00 | ManxPower-work | ~toolkit |
| 17:37.01 | infobot | toolkit is, like, Remember Asterisk isn't really a PBX. Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 17:37.59 | ManxPower-work | ParanoyaM, you CODE it in your dialplan. Check the time, check the number, store the information in a global variable or data store, check it the next time a call happens. |
| 17:38.23 | ParanoyaM | ManxPower-work: problem in that: number changes |
| 17:38.23 | ManxPower-work | Why do you not want your users to dial the same number 100 times? Can't you just smack them upside the head if they do that? |
| 17:38.44 | ParanoyaM | because it is gray termination |
| 17:38.51 | codefreeze-lap | I wrote an app to do that! ;) |
| 17:38.52 | Kobaz | gray? |
| 17:38.57 | Kobaz | as opposed to yellow or blue? |
| 17:39.04 | ManxPower-work | Kobaz, "unofficial, usually illegal" routes. |
| 17:39.12 | Kobaz | heh |
| 17:39.13 | ParanoyaM | when one call goes out from 40 sims it leads to blocking sim |
| 17:39.34 | ManxPower-work | ParanoyaM, the dialplan is where you would code this. |
| 17:39.39 | ParanoyaM | codefreeze-lap: can you share your solution? |
| 17:39.43 | Kobaz | ParanoyaM: are you one of those fradulent robodialer companies? |
| 17:39.52 | ParanoyaM | Kobaz: no |
| 17:39.54 | Kobaz | i know they all use asterisk |
| 17:40.01 | Kobaz | they use the default music on hold |
| 17:40.11 | Slugs_ | ha! |
| 17:40.22 | ManxPower-work | Kobaz, and fail one out of 10 calls to collect the CNAM dip revenue? |
| 17:40.36 | codefreeze-lap | parnoyaM: the app to smack someone upside the head? |
| 17:40.46 | ACK-NAK | All of my asterisk robodialers are chaged to use the Microsoft OC MOH |
| 17:40.48 | ParanoyaM | codefreeze-lap: no |
| 17:40.59 | Kobaz | heh |
| 17:41.12 | ManxPower-work | ParanoyaM, have you read the Asterisk Book |
| 17:41.26 | Slugs_ | ~book |
| 17:41.27 | 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 |
| 17:41.34 | ACK-NAK | let me let me.... awwww I wanted to do ti. |
| 17:41.42 | Slugs_ | ;0 |
| 17:41.42 | Katty | heh |
| 17:41.58 | ACK-NAK | somebody else pls ask about the book. I'll get ready so I can do it! |
| 17:41.58 | Katty | okay someone gets to help me for a change |
| 17:42.06 | Katty | i have a very serious issue |
| 17:42.11 | Katty | and it's been bother me for HOURS now |
| 17:42.12 | spenguin[work] | run shome |
| 17:42.14 | Katty | my grocery list is empty |
| 17:42.14 | ACK-NAK | have youread the ~book |
| 17:42.19 | Slugs_ | lol |
| 17:42.21 | spenguin[work] | Katty: brocolli |
| 17:42.24 | Katty | and i just ....can't deal with it |
| 17:42.27 | spenguin[work] | spinach |
| 17:42.29 | Katty | IT NEEDS TO BE RESOLVED |
| 17:42.32 | codefreeze-lap | lay it on Katty now, ACK-NAK!!! |
| 17:42.40 | ACK-NAK | ~book |
| 17:42.41 | 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 |
| 17:42.43 | ACK-NAK | WOO HOO! |
| 17:42.49 | Katty | puts book on grocery list |
| 17:42.53 | outtolunc | hands katty a pen/pencil |
| 17:42.54 | Katty | oh wait, i already have one |
| 17:43.04 | ACK-NAK | let me get you a... |
| 17:43.06 | ACK-NAK | ~pencil |
| 17:43.06 | infobot | it has been said that pencil is something you write with |
| 17:43.08 | ACK-NAK | nope |
| 17:43.18 | ACK-NAK | where's infobot when you need 'em |
| 17:43.28 | Katty | he's right there |
| 17:43.32 | ManxPower-work | There is nothing wrong with playing with the bot, but it is best done in private and wash your hands after. |
| 17:43.33 | Katty | at least i assume infobot is a male |
| 17:43.35 | Katty | infobot: are you a male? |
| 17:43.47 | Katty | infobot: are you a female? |
| 17:43.49 | ACK-NAK | ~male? |
| 17:44.21 | ACK-NAK | infobot: [male]Yep, I'm a dude. |
| 17:44.34 | Katty | i smell impersonation |
| 17:44.48 | Kobaz | i dunno, it's hard to tell |
| 17:44.49 | ACK-NAK | ahhhhhhh. That's not all you smell |
| 17:45.06 | codefreeze-lap | It smacks of a turing test to me... |
| 17:45.07 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 17:45.09 | Katty | actually i smell febreeze |
| 17:45.16 | Katty | the container says.... |
| 17:45.18 | ACK-NAK | Actually the infobot just hijacked my account for a sec!! WFT? |
| 17:45.19 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:45.31 | Katty | Spedial Edition caramel cream and bliss |
| 17:45.44 | ACK-NAK | That pefectly explains the comment above being preceded by my nick |
| 17:45.53 | Katty | ACK-NAK: of course, dear. of course. |
| 17:46.23 | ACK-NAK | That infobot... Damn he's clever. He also hates to be anthropomorphized. He totally hates taht. |
| 17:46.31 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:46.52 | Katty | hi anonymouz666 |
| 17:47.18 | ACK-NAK | anonymouz666? Impersonating the devil---anonymously. |
| 17:47.20 | Katty | stares at blank grocery list |
| 17:47.31 | Katty | puts down broccoli |
| 17:47.55 | anonymouz666 | lol |
| 17:48.38 | eppigy | RIBEYE STEAK |
| 17:48.56 | Katty | eppigy: will you cook it for me? |
| 17:49.01 | Katty | actually, i bet ryan would cook it |
| 17:49.06 | Katty | that's a wonderful idea, dear. |
| 17:52.55 | bmoraca_work | i have some ribeyes in the freezer |
| 17:53.01 | bmoraca_work | should thaw them out and cook them |
| 17:55.38 | *** join/#asterisk geneticx_wrk (~geneticx_@host-208-88-126-198.biznesshosting.net) |
| 17:56.07 | Carlos_PHX | Damn, I'm hungry. No steak in the house, but some spicy Italian sausage on the grill would work for lunch. |
| 17:56.25 | Carlos_PHX | Let Katty talk about veggies again, that will kill my hunger. |
| 17:59.51 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:59.56 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-128-92.socal.res.rr.com) |
| 18:01.36 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 18:02.17 | bmoraca_work | wtb a voice wholesaler on the west coast that doesn't suck |
| 18:03.53 | Katty | what are you going to do |
| 18:03.56 | Katty | buy a new voice? |
| 18:04.06 | Katty | Carlos_PHX: broccoli and carrots |
| 18:04.12 | Katty | Carlos_PHX: i a rich cheese sauce |
| 18:04.23 | Katty | Carlos_PHX: with a crunchy cornflake topping, and bacon |
| 18:04.30 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 18:04.31 | Katty | Carlos_PHX: feeling less hungry? |
| 18:08.13 | hardwire | bmoraca_work: mmm.. steak |
| 18:08.18 | hardwire | for.. breakfast? |
| 18:08.23 | hardwire | heh |
| 18:08.34 | bmoraca_work | steak and eggs, baby! |
| 18:08.50 | hardwire | I love that about camping |
| 18:08.52 | Katty | heh. |
| 18:08.56 | hardwire | steak + eggs + potatoes |
| 18:09.31 | Katty | from Reddit: Today, I receveived a letter from Blue Cross informing me that my son would be dis-enrolled from my health plan because he'll be 22. I think I'm going to receive another letter soon. Thank you, Mr. President. |
| 18:09.33 | DJF5 | maybe not an asterisk question perse... but how can the process `asterisk -gcvvvv` keep running even if i tried stopping it with all |
| 18:09.40 | DJF5 | even kill -9 $PID |
| 18:09.52 | DJF5 | top sais: 27513 root 6 -8 0 19192K 14708K STOP 1 0:00 0.00% asterisk |
| 18:09.59 | DJF5 | keeps hanging in STOP |
| 18:11.42 | Kobaz | DJF5: something is badly locked up at the OS level, for that process |
| 18:11.52 | DJF5 | :S |
| 18:12.02 | DJF5 | ok |
| 18:14.21 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:14.39 | Qwell | This is why you don't use kill -9. |
| 18:14.40 | sbrath | If I set a CALLERID(name) in a cid-macro from my incoming context, and then send the call to new context to begin a IVR or a Queue processing, I loose the over-ridden CALLERID(name) does CALLERID(name) changes have a scope? |
| 18:14.56 | citywok | for some reason each time asterisk does a reload, a lot of my aastra phones go to no service and it takes a few minutes for them to re-register |
| 18:15.06 | citywok | any ideas? |
| 18:15.27 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.167.140) |
| 18:16.23 | sbrath | DJF5: if kill -9 dosen't work then asterisk is locked and holding onto something. You'd have to init 6 to fix it. |
| 18:16.56 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 18:16.58 | sbrath | that is very unusual. |
| 18:17.17 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.38 | DJF5 | yes, my thoughts exactly :p |
| 18:18.19 | DJF5 | @Qwell, i tried stopping it every way without -9 |
| 18:18.24 | DJF5 | then even -9 didnt work |
| 18:19.18 | afo0l | hi guys, anyone know what causes "zaphfc: hfc busy." when modprobing zaphfc? |
| 18:19.51 | [TK]D-Fender | sbrath: No, CID does not get choved back like that. Show us the actual problem happening |
| 18:25.01 | Katty | Limbaugh said that he never said that he was going to move there but that, .Once all this gets implemented, I am going there for my health care.. <- http://www.youtube.com/watch?v=QqczVe7GX2U <- also completely off topic. |
| 18:25.05 | Katty | ^- amusing |
| 18:25.41 | Katty | why would you say you never said something, when your exact words are plasted all over youtube |
| 18:25.47 | Katty | plastered. |
| 18:25.54 | Katty | does not seem to make good sense. |
| 18:26.03 | Katty | but he's not moving |
| 18:26.12 | Slugs_ | plsated would work if your from boston |
| 18:26.16 | Katty | the limbaugh lawfirm looked same as always when i drove by it this morning |
| 18:26.18 | Slugs_ | plasted |
| 18:26.22 | [TK]D-Fender | Katty: He needs gravity to be repealed first ;) |
| 18:26.38 | Katty | i should call the lawfirm and ask them when he's moving |
| 18:27.01 | bmoraca_work | it'd be way slick if USAA had health insurance...my premiums are going up 25% in april...ick |
| 18:27.29 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.238.130.dsl.dyn.forthnet.gr) |
| 18:27.34 | Katty | bmoraca_work: you can buy into the exchange |
| 18:27.44 | Slugs_ | don't get sick, no nned for insurache |
| 18:27.49 | Slugs_ | insurance |
| 18:27.55 | Katty | Slugs_: yeah, try to tell that to my appendix. |
| 18:27.58 | Katty | Slugs_: erm, lack of appendix |
| 18:28.02 | bmoraca_work | Slugs_: actually, by law, you must have insurance or pay a fine now |
| 18:28.08 | Katty | bmoraca_work: not yet |
| 18:28.12 | Katty | bmoraca_work: that starts in 2014 |
| 18:28.13 | Naikrovek | no |
| 18:28.17 | Naikrovek | 2014 |
| 18:28.24 | Naikrovek | .. what katty said |
| 18:28.31 | Katty | but that's the same as China |
| 18:28.36 | Katty | everyone there has to carry insurance too |
| 18:28.37 | Slugs_ | Katty, i got in a car accident in july, my insurance lapsed few months eairlier, 200k bill |
| 18:28.39 | Slugs_ | BAM |
| 18:28.45 | Katty | Slugs_: ouch |
| 18:28.46 | bmoraca_work | either way, the bullshit that was passed on sunday is bullshit and everyone who voted for it needs to be lit on fire |
| 18:28.48 | Katty | Slugs_: good job |
| 18:28.57 | Slugs_ | ;/ |
| 18:29.02 | Katty | Slugs_: <3 |
| 18:29.11 | Katty | bmoraca_work: thanks, but i'll take my bullshit. |
| 18:29.24 | Katty | bmoraca_work: and the bullshit will make sure that my father gets the insulin shots he needs. |
| 18:29.24 | Slugs_ | yeah i kmnow how much it sux not to have insurance that's for sure |
| 18:29.58 | bmoraca_work | Katty: i have no problem with health insurance or medical care reform, but the bill that was passed is NOTHING except a big payout to insurance companies. it doesn't do anything to make anything better for anyone, except to CUT medicare benefits. |
| 18:30.15 | *** part/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 18:31.49 | Katty | bmoraca_work: it does do anything but make sure my father can't be dropped because he has diabetes? |
| 18:32.07 | Katty | bmoraca_work: it doesn't do anything, but make sure children can't be barred and excluded from coverage for preexisting conditions? |
| 18:32.13 | Katty | bmoraca_work: pull your head out of your ass dude (= |
| 18:32.18 | Katty | bmoraca_work: we can talk about it in private if you want |
| 18:32.24 | bmoraca_work | i'd rather not. |
| 18:32.29 | Naikrovek | bmoraca_work: read: http://www.reuters.com/article/idUSN1914020220100319 |
| 18:32.29 | Katty | k |
| 18:32.33 | bmoraca_work | debating politics sets my teeth on edge. |
| 18:32.37 | Ad-Hoc | hi ppl |
| 18:32.40 | Katty | i can see that. |
| 18:33.27 | bmoraca_work | i rest by my statement that anyone who voted for this bill should be summarily executed in the worst possible fashion. i'm particular to drowning in an oil fire, myself. |
| 18:33.39 | Katty | well then line me up for the firing squad |
| 18:33.43 | Naikrovek | me as well |
| 18:34.05 | bmoraca_work | Katty: you're not a congressman, so I don't need to :P |
| 18:34.10 | Naikrovek | don't let fox news do your thinking for you. if you came to your opinion on your own, that's fine, just don't let limbaugh, hannity or anyone on fox think for you |
| 18:34.19 | Naikrovek | or anyone at all, really |
| 18:34.22 | Katty | fox news should be shot. |
| 18:34.37 | Katty | they should have entertainment purposes only disclaimers on the bottom of their screen |
| 18:34.44 | bmoraca_work | Naikrovek: i don't watch opinion news programs. i watch CNBC in the morning for financial news and that's all. |
| 18:35.07 | Naikrovek | conservatives should be all over this because it does more to balance the budget than anything done since 1993 |
| 18:35.14 | Naikrovek | that was clinton, btw. |
| 18:35.21 | [TK]D-Fender | [14:33]<Naikrovek>don't let fox news do your thinking for you. if you came to your opinion on your own, that's fine, just don't let limbaugh, hannity or anyone on fox think for you <- as has been well covered, those hosts are not "Fox News". Those are Fox Opinion Commetators |
| 18:35.26 | Naikrovek | fiscal conservatism and conservatives do not mix |
| 18:35.39 | Katty | is that how they get away with it? |
| 18:35.44 | Katty | it's all Opinions |
| 18:35.47 | bmoraca_work | Naikrovek: modern day republicans != conservatives |
| 18:35.50 | [TK]D-Fender | Katty: Yup |
| 18:35.56 | Katty | crazy stuff |
| 18:36.08 | bmoraca_work | Naikrovek: also...$1T in increased spending != balancing the budget. |
| 18:36.09 | Katty | bmoraca_work: what's your opinion of the tanning bed stuff? |
| 18:36.16 | [TK]D-Fender | Katty: You should become a TDS & TCR addict like me :) |
| 18:36.33 | Naikrovek | bmoraca_work: it will reduce the deficit by $1.8B |
| 18:36.37 | Naikrovek | bmoraca_work: read some more |
| 18:36.47 | Katty | healthcare is eating us alive |
| 18:36.53 | bmoraca_work | Katty: i don't think the government has any right (the CotUS supports this) to tell me what I can and cannot do, who I should or should not be able to spend my money with, or whether I should or should not have insurance |
| 18:37.04 | bmoraca_work | Katty: that's fine, but subsidies (what this bill is) are not the way to fix it. |
| 18:37.07 | Katty | bmoraca_work: what does that have to do with tanning beds? |
| 18:37.14 | *** join/#asterisk thecardsmith (~doug@pool-71-161-218-3.burl.east.myfairpoint.net) |
| 18:37.27 | bmoraca_work | Katty: the government shouldn't dictate whether i can or cannot use a tanning bed or to tax my use of it. |
| 18:37.40 | Naikrovek | bmoraca_work: then you're saying that the income tax, social security, and all of those things are all unconstitutional? |
| 18:37.48 | Katty | along with cigarette tax |
| 18:37.53 | Naikrovek | because you dont' ahve a choice with those either |
| 18:37.54 | Katty | and all tax, basically |
| 18:38.19 | bmoraca_work | Naikrovek: social security, yes. income tax, not necessarily. it's written into the CotUS. government control of business is specifically written OUT of the CotUS. |
| 18:38.27 | *** part/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:38.35 | bmoraca_work | Katty: sin taxes are supposed to be levied at the state level, not the federal level. |
| 18:38.43 | [TK]D-Fender | Naikrovek: Income tax is specifically uncontitutional. |
| 18:38.54 | *** mode/#asterisk [+m] by Qwell |
| 18:38.58 | Qwell | Back to Asterisk. Thanks. |
| 18:39.06 | *** mode/#asterisk [-m] by Qwell |
| 18:39.06 | Slugs_ | lol |
| 18:39.08 | bmoraca_work | lol |
| 18:39.09 | Naikrovek | hehe |
| 18:39.11 | Naikrovek | thanks qwell |
| 18:39.47 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:39.49 | Slugs_ | #politics |
| 18:39.54 | Slugs_ | ;0 |
| 18:39.55 | Katty | we have one of those? |
| 18:40.04 | bmoraca_work | my blood pressure's high enough already |
| 18:40.06 | Slugs_ | i just typed it |
| 18:40.16 | Katty | well there are actually a lot of people in there |
| 18:40.18 | *** join/#asterisk Aven (~avenger@78.36.107.48) |
| 18:40.25 | Slugs_ | lol |
| 18:40.33 | Katty | i left |
| 18:40.43 | Naikrovek | that place would be a madhouse |
| 18:40.51 | Slugs_ | lol |
| 18:41.02 | Slugs_ | yes it would |
| 18:41.30 | Katty | 13:41 -!- Nick MadHatter is already in use :< |
| 18:41.45 | Slugs_ | hehe |
| 18:42.04 | Slugs_ | there ya go |
| 18:43.17 | Katty | so how about that grocery list |
| 18:43.18 | Slugs_ | i dont take sides, i just want people to that 'need' help and art lazy, get help |
| 18:43.37 | Katty | so far i have steak and broccoli |
| 18:43.43 | Slugs_ | corn |
| 18:43.45 | Slugs_ | umm |
| 18:43.54 | Slugs_ | mashed potatoes |
| 18:43.56 | Katty | TEA |
| 18:44.32 | bmoraca_work | pie is always good, too. claim jumper dutch apple pies...mmmmm |
| 18:44.32 | Katty | Slugs_: have you seen those new steamfresh bags in the frozen section which is mashed potatos? |
| 18:44.43 | Katty | Slugs_: you microwave it and then mush it, and BAM mashed potatos |
| 18:44.49 | Slugs_ | lol yeah |
| 18:45.16 | Slugs_ | or the granulated mess add water or milk and BAM mashed taters |
| 18:45.41 | bmoraca_work | Katty: have you seen the frozen bags in freezer sections that are veggies+sauce? they have broccoli and cheese...microwave and ready! |
| 18:45.51 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 18:45.52 | Katty | bmoraca_work: i certainly have! |
| 18:45.59 | Katty | bmoraca_work: some of them are quite tasty |
| 18:45.59 | bmoraca_work | they're pretty good |
| 18:46.21 | bmoraca_work | i like the red potatos & green beans in the rosmary sauce |
| 18:46.22 | bmoraca_work | it's good |
| 18:46.28 | ACK-NAK | Difference between deny and contactdeny? |
| 18:46.51 | ACK-NAK | Users vs peers? |
| 18:47.03 | ACK-NAK | Confirmation anyone? |
| 18:47.36 | *** join/#asterisk BreezBl0k (~BreezBl0k@5e0e9985.bb.sky.com) |
| 18:47.43 | Katty | gets a post it note, writes confirmation on it in crayon, and hands it to ACK-NAK |
| 18:47.54 | Katty | you can haz confirmation |
| 18:48.03 | ACK-NAK | lolkatty |
| 18:49.12 | ACK-NAK | katty: but is also true that permit restricts BOTH and contactpermit only restricts user registration address? |
| 18:49.27 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:49.45 | Katty | ACK-NAK: i have no idea |
| 18:49.45 | BreezBl0k | Hi I have Yealink T-28P phones all work fine apart for one extension, you can ring it when it first powers on but shortly after it will go to voicemail without ringing but you can make calls out, if you reboot it its works both ways for a few mins again |
| 18:50.10 | ACK-NAK | ACK-NAK hands confirmation note back to katty |
| 18:50.11 | Katty | BreezBl0k: what happens when you replace the phone |
| 18:50.16 | Katty | ACK-NAK: :< |
| 18:50.17 | BreezBl0k | same thinh |
| 18:50.20 | BreezBl0k | *think |
| 18:50.22 | Katty | puts postitnote on forhead |
| 18:50.31 | BreezBl0k | god dammit thing! |
| 18:50.32 | Katty | i are confirmed. |
| 18:51.08 | BreezBl0k | i swapped phone and same issue moved phone to where a behaving phone is connected as well |
| 18:51.16 | Katty | BreezBl0k: do you have two devices trying to register as the same extension |
| 18:51.33 | BreezBl0k | no i even change secret to rule that out |
| 18:52.05 | bmoraca_work | ACK-NAK: according to http://www.asterisk.org/doxygen/trunk/Config_sip.html, contactpermit/deny are used for registration, whereas permit/deny are used as an ACL. i can't really think of a normal situation where the two would be different. |
| 18:52.14 | BreezBl0k | its realy stumping me i changed firmware as well to no avail |
| 18:52.20 | Katty | BreezBl0k: when the phone starts messing up, does it list an ip address in sip show peers |
| 18:52.28 | BreezBl0k | yes |
| 18:53.47 | Katty | debug |
| 18:53.50 | ACK-NAK | bmoraca_work: but they make two just so it's simple. :-) |
| 18:55.46 | VoIP-Penguin | Crap. I forgot to have lunch. |
| 18:56.55 | bmoraca_work | ACK-NAK: think about NAT for a minute...the ACL is going to be based on the global IP whereas the registration contact could be the inside IP |
| 18:57.00 | Slugs_ | gives VoIP-Penguin a hot pocket |
| 18:57.08 | VoIP-Penguin | What kind!? |
| 18:57.19 | Slugs_ | pizza! |
| 18:57.25 | bmoraca_work | ACK-NAK: like I said, though...i can't really think of a reason why they would need to be different...but the flexibility is there |
| 18:57.26 | VoIP-Penguin | I likes me some ham and cheese. |
| 18:57.40 | Katty | hotttttttt pockettttt |
| 18:57.41 | Slugs_ | man i was goingg to give you that one too |
| 18:57.52 | VoIP-Penguin | Pizza is good, too, as long as it isn't the old one. The new pizza is far better. |
| 18:57.56 | BreezBl0k | asterisk debug information states Extension 4152 is not available to be called |
| 18:57.59 | bmoraca_work | caliente pockets... |
| 18:58.20 | ACK-NAK | bmoraca_work: My brain hurts. |
| 18:58.30 | Qwell | ham and cheese. pfft. they started cubing the ham. |
| 18:58.44 | Slugs_ | Qwell, lol yeah |
| 18:59.01 | ACK-NAK | bmoraca_work: contactpermit=hostname (vs IP) didn't freak out the way permit=hostname does on reload. That could be useful. |
| 18:59.03 | VoIP-Penguin | No more sliced ham? |
| 18:59.12 | Slugs_ | neg |
| 18:59.16 | VoIP-Penguin | suck |
| 18:59.32 | *** join/#asterisk trevorsharrison (~tharrison@70.88.150.241) |
| 18:59.51 | ACK-NAK | bmoraca_work: wrong. Nevermind |
| 19:00.37 | ACK-NAK | bmoraca_work: wait, maybe not. |
| 19:01.01 | bmoraca_work | ACK-NAK: i don't think it does reverse lookups like that |
| 19:01.24 | Katty | trevorsharrison: are you ford's brother? |
| 19:01.44 | *** join/#asterisk chuckf_ (~chuckf@ubuntu/member/chuckf) |
| 19:01.52 | trevorsharrison | :( |
| 19:02.00 | Katty | :> |
| 19:02.13 | ACK-NAK | Yes, I agree. but it should. |
| 19:02.20 | ACK-NAK | bmoraca_work: ^^^^^ |
| 19:02.41 | bmoraca_work | ACK-NAK: i disagree...too time consuming |
| 19:03.32 | ACK-NAK | bmoraca_work: how so? Externrefresh=x |
| 19:03.58 | ACK-NAK | bmoraca_work: or DNSACLRefresh=x or somethign like that |
| 19:04.25 | bmoraca_work | ACK-NAK: you want to perform a reverse lookup on every single packet that comes in? not a good idea, imo. ACLs are done by IP for a number of reasons |
| 19:04.32 | bmoraca_work | anyway, off to lunch |
| 19:04.48 | ACK-NAK | bmorca_work: right. Cache. |
| 19:06.16 | BreezBl0k | Katty http://pastebin.com/XV4U9UVy is what debug is saying |
| 19:06.56 | [TK]D-Fender | BreezBl0k: ... |
| 19:06.58 | [TK]D-Fender | ~freepbx |
| 19:06.59 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:07.00 | [TK]D-Fender | ^^^^ |
| 19:08.45 | Kobaz | okay, i have a T1 LBO question... we have a 440ft cable between an avaya and an asterisk. I can set the lbo on asterisk to 440-550, but on the avaya i have settings in dB. What's the lbo eqivalencies to decibals? |
| 19:10.22 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 19:10.44 | Qwell | Kobaz: somewhere between 0 and -7.5, it looks like |
| 19:10.58 | Qwell | maybe. |
| 19:11.04 | Qwell | What options are there? |
| 19:13.48 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 19:15.26 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 19:15.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:16.39 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.29 | Kobaz | 0 -15 and... sec |
| 19:17.37 | trevorsharrison | My asterisk setup with a 4 port analog digium card just started having problems with inbound calls not getting audio (can't hear ivr greeting or audio from my polycom ext). Rebooting/powering off asterisk server + telecom equip hasn't fixed it. Outbound works fine. Don't have anything right now to independently test the lines with. Probably dialplan corruption? |
| 19:17.40 | bmoraca_work | Kobaz: leave it default unless you have a problem |
| 19:17.47 | Kobaz | bmoraca_work: we have problems |
| 19:18.11 | Qwell | Kobaz: I would guess -15 or -22.5 |
| 19:18.15 | Kobaz | 0, -7.5, -15, -22.5 |
| 19:18.16 | bmoraca_work | gotcha |
| 19:18.32 | Qwell | 440-550 isn't a valid lbo though afaik |
| 19:18.45 | Kobaz | that's in feet, on a sangoma card |
| 19:18.46 | Qwell | the closest would be 399-533 |
| 19:20.32 | Kobaz | 110-220 220-330 330-440 440-550 550-660 0DB 7.5DB 15DB 22.5DB |
| 19:20.35 | Kobaz | are the sangoma options |
| 19:21.12 | Kobaz | what if i just set the sangoma to be 15DB and the other side to be 15DB |
| 19:21.22 | Kobaz | and just not mess with the distance-based settings |
| 19:22.29 | bmoraca_work | i had fun yesterday when some idiot plugged my PBX into a data T1...trying to figure out why all my channels are ringing at the same time and i'm not getting DNIS information...freakin idiots |
| 19:23.17 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 19:23.25 | Kobaz | heh |
| 19:24.28 | Naikrovek | trevorsharrison: hang tight someone will help i'm sure |
| 19:24.37 | Naikrovek | patience is key in here sometimes |
| 19:25.38 | [TK]D-Fender | [15:17]<trevorsharrison>My asterisk setup with a 4 port analog digium card just started having problems with inbound calls not getting audio (can't hear ivr greeting or audio from my polycom ext). Rebooting/powering off asterisk server + telecom equip hasn't fixed it. Outbound works fine. Don't have anything right now to independently test the lines with. Probably dialplan corruption? <- no |
| 19:26.12 | [TK]D-Fender | trevorsharrison: go prove audio from the TDM to * direct. use Record & Playback for this. If that works, then the problem is between * and your phones. |
| 19:27.50 | Netgeeks | Hi Katty Long time no chat (responding to you from yesterday - lol) |
| 19:28.14 | trevorsharrison | I'm a little confused. Are you thinking about a (sigh, I can never remember the correct TLA) FXS vs a FXO? My digium is to connect to the PSTN for inbound phone lines. |
| 19:28.51 | [TK]D-Fender | trevorsharrison: there are 2 calls happening. One between your TDM card and *. The second is from * to your phones. |
| 19:29.09 | [TK]D-Fender | trevorsharrison: We need to isolate which leg has the problem. |
| 19:29.38 | trevorsharrison | k. lemme give you a few more data points. |
| 19:29.59 | trevorsharrison | When I call in, if I get routed to an IVR, I don't hear the IVR recorded greeting. |
| 19:30.20 | trevorsharrison | If I navigate to, for instance, the interactive directory, I don't hear any prompts. |
| 19:30.58 | trevorsharrison | Looking at the verbose logging in asterisk -rvvvv, I can see my DTMF effecting stuff. |
| 19:31.52 | trevorsharrison | If (calling in from outside using my cellphone) I connect to an extension, I can hear sounds from the cell phone on my ext, but no sound going the other way. |
| 19:32.07 | [TK]D-Fender | trevorsharrison: Forget DTMF. Prove AUDIO. |
| 19:32.38 | trevorsharrison | the last thing I typed doesn't? |
| 19:32.46 | [TK]D-Fender | trevorsharrison: If you have no IVR audio then you likely either have a HWEC issue where its spooling all the audio, or you have a gain issue |
| 19:33.07 | Katty | Netgeeks: yes, it's been HOURS |
| 19:33.26 | trevorsharrison | k. I can start with the zaptel config. |
| 19:33.42 | Netgeeks | hehe, I just had my client open and about 2/3ds of the way up the scroll bar I see a little marker that means someone said my name |
| 19:33.53 | Netgeeks | so I scroll up, and there you were |
| 19:33.54 | *** join/#asterisk knarfly (~vlad@c-98-242-237-166.hsd1.fl.comcast.net) |
| 19:33.54 | [TK]D-Fender | trevorsharrison: What precise card do yuo ahve. What * Version, What Zaptel version |
| 19:34.05 | trevorsharrison | ancient... hold on.... |
| 19:34.36 | knarfly | It's been a while since I configured Asterisk on FreeBSD...I made a simple install from the ports but the console keeps showing me this |
| 19:34.39 | knarfly | WARNING[7324]: db.c:57 dbinit: Unable to open Asterisk database '/var/db/asterisk/astdb': No such file or directory |
| 19:34.39 | knarfly | [Mar 23 15:31:39] WARNING[7324]: db.c:498 ast_db_gettree: Database unavailable |
| 19:34.44 | trevorsharrison | [TK]D-Fender: # rpm -qa | grep zap |
| 19:34.44 | trevorsharrison | zaptel-modules-1.4.12.1-1.2.6.18_92.1.18.el5 |
| 19:34.44 | trevorsharrison | zaptel-1.4.12.1-2 |
| 19:35.05 | tzafrir | knarfly, permissions issue? |
| 19:35.05 | Qwell | trixbox? |
| 19:35.17 | knarfly | tzafrir: as root? |
| 19:35.24 | [TK]D-Fender | trevorsharrison: I don't believe those versions are necessarily compatible |
| 19:35.34 | tzafrir | asterisk not running as root but that directory is woned by root? |
| 19:35.37 | tzafrir | hmm.... |
| 19:36.04 | knarfly | tzafrir: asterisk can only be run as root. |
| 19:36.05 | tzafrir | maybe the directory does not exist? |
| 19:36.11 | tzafrir | knarfly, why? |
| 19:36.15 | knarfly | let me check |
| 19:36.29 | tzafrir | runs Asterisk as 'asterisk' |
| 19:36.36 | Qwell | runs Asterisk as 'bob' |
| 19:36.37 | trevorsharrison | [TK]D-Fender: asterisk-1.4.22-3 |
| 19:36.43 | trevorsharrison | (sorry its taking so long) |
| 19:36.46 | VoIP-Penguin | knarfly: Really? |
| 19:36.51 | tzafrir | (and the init.d script fails you if you instruct it to run asterisk as root) |
| 19:37.09 | VoIP-Penguin | knarfly: I guess you better send that memo to all the people that don't run it as root. |
| 19:37.19 | tzafrir | (unless you give root an id that is not 0) |
| 19:37.35 | knarfly | VoIP-Penguin: sorry my inexperience is showing...this is my setup |
| 19:37.55 | VoIP-Penguin | knarfly: As a matter of fact, I think most people recommend to NOT run it as root. |
| 19:38.09 | tzafrir | knarfly, anyway, the point is that asterisk attempts to generate the astdb on startup |
| 19:38.13 | knarfly | the /var/db/asterisk directory did not exist...that directory was always created during the installation of previous copies |
| 19:38.31 | tzafrir | another sanity check: is there free disk space? writable mount? |
| 19:39.00 | Qwell | FreeBSD. I'm sure there's some config/path screwiness. |
| 19:39.22 | knarfly | the console is quiet now...it was this missing directory. this is the first time I've ever had to manually create it though |
| 19:39.23 | VoIP-Penguin | I have Asterisk 1.4 on FreeBSD, and I don't have the /var/db/asterisk directory. |
| 19:39.58 | trevorsharrison | [TK]D-Fender: I just noticed a "zaptel: no version for "oslec_echo_can_traintap" found: kernel tainted." in my /var/log/messages. I'm not sure that warning was in the log before (the uptime was 290+ days since last reboot) and the log files have been rotated since. |
| 19:40.11 | knarfly | 1.6.0.21 running here |
| 19:40.39 | Katty | where do you guys get your plane tickets |
| 19:40.44 | Qwell | Katty: expedia! |
| 19:40.52 | Katty | DOT COM |
| 19:40.58 | VoIP-Penguin | Travelocity |
| 19:41.12 | knarfly | will never buy Grandstream phones again....2 out of 3 have just stopped working |
| 19:41.20 | Qwell | knarfly: they do that. |
| 19:41.23 | VoIP-Penguin | ~gs |
| 19:41.24 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:41.34 | Netgeeks | depends on where I'm going, but then again, I'm more concerned about convenience than price in most cases |
| 19:41.40 | knarfly | infobot: live and learn |
| 19:42.03 | knarfly | the bad taste of poor quality remains long after the sweet taste of a cheap price fades |
| 19:42.14 | ManxPower-work | trevorsharrison, there is nothing wrong with a tainted kernel |
| 19:42.55 | trevorsharrison | [TK]D-Fender: ah, finally found info about the card: "Wildcard TDM400P REV E/F" |
| 19:43.48 | trevorsharrison | ManxPower-work: yeah, I agree, but the message seemed to indicate that the echo cancelation missing was due to kernel taint. |
| 19:44.00 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.238.130.dsl.dyn.forthnet.gr) |
| 19:45.18 | *** join/#asterisk jaytee (~465bd509@gateway/web/freenode/x-qchbeiomecvctvrw) |
| 19:47.28 | jaytee | hi |
| 19:47.48 | Slugs_ | hi |
| 19:48.06 | Katty | woah. flight to dallas: 2hrs. amtrak to dallas: 15.5hrs |
| 19:48.19 | jaytee | what's the diff in price? |
| 19:48.24 | Katty | 200 vs 90 |
| 19:48.45 | Katty | i think i'll just pay the 200 bucks and be there in time for breakfast |
| 19:48.51 | jaytee | how much is 13.5 hours of your time worth to you? |
| 19:49.00 | Netgeeks | convenience wins over price again! |
| 19:49.08 | Katty | i'm not going to spend 15.5 hrs on a train |
| 19:49.22 | Katty | http://www.fossilrim.org/ <- my destination |
| 19:49.39 | Katty | ^- goin with my mamma |
| 19:49.48 | jaytee | sitting in the same seat for 15.5 hours could never be enjoyable and in some cases might be fatal |
| 19:50.34 | VoIP-Penguin | I get tired of sitting in the same spot after just a couple hours. |
| 19:50.38 | trevorsharrison | afk for abit |
| 19:50.40 | Netgeeks | Katty, if you hadn't mentioned texas, I would have looked at that and said, 200 bucks to go to africa? no way |
| 19:50.52 | Katty | Netgeeks: that's more liek 4grand |
| 19:51.24 | Netgeeks | and don't forget the vaccinations and the doctor bill for the hospital stay when you get back |
| 19:51.35 | Katty | :< |
| 19:51.52 | jaytee | if I was going to travel outside the US I'd only go to countries that don't hate us so that only leaves me with Australia |
| 19:52.14 | VoIP-Penguin | What about India? |
| 19:52.24 | Katty | i don't want to go to india |
| 19:52.31 | jaytee | well, they don't hate us so much as the whole scorn thing |
| 19:53.05 | Netgeeks | hey now, there are other countries that don't hate the US... like..... New Zealand! okay, new zealand is part of australia kinda like canada is part of the us |
| 19:53.18 | jaytee | yeah, Kiwis are cool people |
| 19:53.27 | *** join/#asterisk Z_God (~julius@130.89.234.72) |
| 19:54.05 | jaytee | Indians are nice too |
| 19:54.45 | Netgeeks | it seems to me that india is either too hot or too wet.... sometimes both at the same time |
| 19:55.11 | jaytee | yes, but there is naan |
| 19:55.56 | hardwire | ahoy beasties |
| 19:56.06 | jaytee | ahoy, matey |
| 19:57.44 | hardwire | anybody using followme and cdr_adaptive_odbc? followme in 1.6.2.6 is crashing when attempting to use cdr resources that include cdr_adaptive_odbc. |
| 19:57.51 | hardwire | even if I specify nocdr |
| 19:59.31 | *** join/#asterisk Mango (~iMango@d154-20-86-138.bchsia.telus.net) |
| 20:00.31 | Mango | I'm trying to set up DNS SRV records so that my devices will fail over to a backup Asterisk server if the primary is unavailable. |
| 20:00.47 | VoIP-Penguin | That should be easy enough. |
| 20:00.52 | Mango | The problem is, my SPA921 doesn't seem to respect priority. I set one to priority 10 and one to priority 20, but it randoms between the two. |
| 20:01.12 | Mango | I set weight to 0 for both - is that right? |
| 20:02.01 | hardwire | weight is per priority |
| 20:02.08 | hardwire | so you did gud |
| 20:02.12 | Mango | OK. |
| 20:02.16 | hardwire | except if they are in the wrong order :) |
| 20:02.38 | Mango | I don't care about the order at this point - I'd settle for the phone picking one and sticking to it :P |
| 20:02.56 | Mango | The phone acts as if the primary server was unreachable...but it wasn't. |
| 20:03.19 | beek | hugs Katty -- waves to jaytee. |
| 20:03.59 | VoIP-Penguin | mango: What happens if you set both to the same priority and set the weight of one a little higher than the other? |
| 20:04.09 | Mango | Let's find out. |
| 20:04.25 | Mango | a little as in how much? |
| 20:04.35 | VoIP-Penguin | 1 |
| 20:04.42 | Mango | ok, so 1 and 2. one moment. |
| 20:04.46 | VoIP-Penguin | 5, 10, whatever |
| 20:05.00 | Mango | 5 and 10.' |
| 20:06.43 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:06.50 | VoIP-Penguin | I would normally not do it that way, but given your problem, I would at least test it. |
| 20:07.26 | VoIP-Penguin | It seems like a lot of clients don't work well with SRV for some reason. |
| 20:07.47 | jaytee | hi beek |
| 20:08.38 | Mango | ok, registered to primary |
| 20:09.08 | beek | jaytee: How goes the new gig? |
| 20:09.09 | VoIP-Penguin | primary is weighted to 10 and the secondary is weighted to 5? |
| 20:09.12 | Mango | yes. |
| 20:09.29 | VoIP-Penguin | And the priority is 10 on both? |
| 20:09.40 | Mango | yes |
| 20:09.48 | VoIP-Penguin | SOunds good. See how it works. |
| 20:09.53 | Mango | it re-registered to the primary |
| 20:09.57 | Mango | and then registered to the secondary |
| 20:10.01 | VoIP-Penguin | hmm |
| 20:10.09 | Mango | secondary again |
| 20:10.22 | VoIP-Penguin | So it still doesn't know what it should be doing. |
| 20:10.52 | VoIP-Penguin | Is there any chance that it's a DNS and host issue rather than the client being borked? |
| 20:11.10 | Mango | theoretically possible. the host name is _sip._udp.sip2.toao.net |
| 20:11.15 | jaytee | beek, it's good although sometimes a bit overwhelming |
| 20:11.20 | Mango | it looks right but I'd appreciate a second set of eyes |
| 20:11.54 | VoIP-Penguin | mango: Looks okay to me. |
| 20:11.58 | Mango | It doesn't seem to want to switch back to the primary. |
| 20:12.17 | beek | jaytee: I know the feeling. |
| 20:13.07 | *** join/#asterisk nickaugust (~anonymous@rrcs-24-73-135-214.se.biz.rr.com) |
| 20:13.32 | VoIP-Penguin | mango: I would go ahead and put the records back to the "normal" way since this way didn't seem to change anything. |
| 20:13.56 | Mango | ok, that is at _sip._udp.sip.toao.net |
| 20:14.10 | VoIP-Penguin | Oh, okay. |
| 20:14.11 | Deeewayne | jaytee, this comment is a little late, but the country of Georgia does not hate the US either |
| 20:14.29 | Mango | maybe I'll experiment with proxy redundancy method |
| 20:14.38 | Deeewayne | and they like to party too |
| 20:14.57 | Mango | set to "Based on SRV port". so far we're on the primary. |
| 20:15.17 | VoIP-Penguin | mango: That might be the only alternative if the client won't cooperate. At least you know the proxy is (supposed to be) designed to do it right. |
| 20:15.36 | Mango | ah, nope. just jumped to the secondary. |
| 20:15.51 | Mango | and back to the primary. lol. |
| 20:15.53 | VoIP-Penguin | based on SRV port? Shouldn't it be based on something else? |
| 20:16.05 | VoIP-Penguin | Like the SRV priority? |
| 20:16.15 | Mango | That's "Normal" |
| 20:16.22 | Mango | the only other option is "Based on SRV port" |
| 20:16.30 | VoIP-Penguin | hmm |
| 20:17.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:21.29 | hardwire | Mango: dns-srv seems to be a lost technology for some phones/switches |
| 20:21.42 | hardwire | the actual implementation of it is.. strange. |
| 20:21.43 | hardwire | :P |
| 20:21.44 | Mango | Apparently so :-/ |
| 20:22.37 | Mango | http://forum.voxilla.com/linksys-sipura-voip-support-forum/anyone-using-dns-srv-records-redundancy-12779.html |
| 20:22.39 | Mango | just found that |
| 20:22.43 | Mango | at least I'm not the only one :P |
| 20:23.20 | VoIP-Penguin | It's a pretty good thing if your devices will just use it right. |
| 20:27.58 | vader-- | would this violate anything in asterisk's 1.6.2 dial plans |
| 20:28.00 | vader-- | exten => 130,1,Macro(stdexten,130,SIP/001759CB88AC-01&SIP/SALESVOIPGW-01-X130) |
| 20:28.30 | VoIP-Penguin | Looks okay to me, but Macro is being deprecated in favor of GoSub() and Return(). |
| 20:28.47 | vader-- | hmmm |
| 20:33.45 | Kobaz | GHETTO |
| 20:33.46 | Kobaz | hah |
| 20:34.01 | Kobaz | this avaya is so old, and has an old t1 board, that LBO cannot be configured |
| 20:35.10 | Slugs_ | Kobaz: ive completed the ~book should i be an expert now :) |
| 20:35.16 | Kobaz | nice |
| 20:35.55 | bmoraca_work | Slugs_: read it 3 more times and implement a few production systems first |
| 20:36.10 | Slugs_ | yeah exactly |
| 20:36.35 | Slugs_ | this agi stuff is pretty facinating |
| 20:41.44 | Katty | so is anyone in dallas texas |
| 20:42.12 | beek | Katty: There are lots of people in dallas texas! |
| 20:42.21 | Katty | but is anyone /here/ in dallas |
| 20:42.23 | beek | ba-doom-chick |
| 20:42.35 | beek | Ahh... different question. ;-) |
| 20:42.41 | Katty | details...details... |
| 20:42.47 | *** join/#asterisk DotHack (~dothack@213.51.110.35) |
| 20:42.49 | beek | opens at the comedy club Friday night. |
| 20:43.10 | Katty | beek: i'll be sure to bring my pillow |
| 20:43.16 | beek | :( |
| 20:43.21 | Katty | tickles beek |
| 20:43.24 | DotHack | anyone has experience with gsm hardware? |
| 20:43.25 | beek | giggles |
| 20:44.11 | Corydon76-dig | DotHack: Yes, I use an Android phone on a near-daily basis |
| 20:44.46 | Katty | i'm not old enough to rent a car :< |
| 20:44.50 | Katty | this makes me sad. |
| 20:45.01 | DotHack | Corydon76-dig: foolish me. IRC asks for a more specific approach. I mean icm asterisk :) |
| 20:45.13 | bmoraca_work | lol @ http://thedailywtf.com/Articles/Else-where.aspx |
| 20:45.15 | *** join/#asterisk jhutchins_lt (~jonathan@64-151-37-66.dyn.everestkc.net) |
| 20:45.42 | *** part/#asterisk jhutchins_lt (~jonathan@64-151-37-66.dyn.everestkc.net) |
| 20:45.49 | Mango | Katty: if you're over 21, Enterprise will rent to you. |
| 20:45.53 | Mango | Most other places you have to be 25. |
| 20:46.14 | jaytee | I think ACE will rent at 21 also |
| 20:46.23 | jaytee | Hertz and Avis won't though |
| 20:46.40 | Mango | Yeah. |
| 20:46.53 | Mango | Also, there's a loophole at Enterprise |
| 20:47.05 | Mango | if you're married and your spouse is over 25, you can both drive the car, and you don't pay the underage rate. |
| 20:47.33 | Mango | We will be renting a car the day AFTER my birthday this year. It will be the first year I won't have to resort to loopholes :) |
| 20:47.39 | Katty | it's just me and my mother |
| 20:47.45 | Katty | looks like she's going to have to rent it in her name |
| 20:47.54 | Katty | this place requires 26+ |
| 20:47.56 | bmoraca_work | jaytee: they will if you pay an extra $40/day |
| 20:48.25 | Mango | $40!? At Enterprise it was only $14 |
| 20:48.32 | Mango | High, but not outrageous |
| 20:48.42 | bmoraca_work | Mango: i ran into that when renting a car on my honeymoon |
| 20:48.43 | *** join/#asterisk alaskappa (~MJCin49@101-182-58-66.gci.net) |
| 20:48.57 | bmoraca_work | Mango: thankfully, i rented through USAA and they waived all those extra fees for me |
| 20:48.58 | mrbnet | Is it possible to route different trunks through different network interfaces within asterisks config? or would I need to utilize static routes for this? |
| 20:49.21 | Mango | bmoraca_work: excellent. |
| 20:49.32 | Mango | The other thing to avoid is the 3rd party liability. |
| 20:49.51 | Mango | It was vastly cheaper to get it through my regular auto insurance. If I rent a car two times in a year, it's paid for. |
| 20:50.19 | Katty | this place is about 45/day |
| 20:50.23 | Katty | does that sound about average? |
| 20:50.35 | DotHack | mrbnet: i thick routing is the best option, asterisk uses the routing table of the os to connect to sip trunk |
| 20:50.50 | Mango | Katty, what city/state? |
| 20:50.55 | Katty | Mango: dallas, tx |
| 20:53.09 | VoIP-Penguin | That is a fairly standard rate. |
| 20:53.25 | Mango | They tend to get lower if you buy a week at a time. A bunch of places have $269/week, taxes included. |
| 20:53.40 | knarfly | can anyone recommend a good, reliable, and inexpensive VOIP provider these days |
| 20:53.46 | Mango | VoIP.ms |
| 20:53.58 | ACK-NAK | knarfly: I have great luck with Vitelity |
| 20:54.02 | VoIP-Penguin | knarfly: VoIP.ms or Flowroute |
| 20:54.03 | Katty | Mango: hrmm. we won't be out there a full week, but i will keep that in mind regardless |
| 20:54.10 | mrbnet | DotHack: That is what I figured, thx. Now I get to figure out how to handle routing through multiple default gateways. |
| 20:54.17 | Katty | infobot: itsp-list |
| 20:54.18 | infobot | [itsp-list] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:54.19 | ACK-NAK | knarfly: I like them because thy can set CNAM. That's rare. |
| 20:54.29 | ACK-NAK | knarfly: Teliax sucks IMO |
| 20:54.35 | Katty | I use bandwidth |
| 20:54.40 | Netgeeks | wicked camper rentals in Australia gave you an extra free day's rental if you picked up the camper in the buff |
| 20:54.46 | VoIP-Penguin | There's no guarantee that the CNAM will be looked up, though. |
| 20:54.46 | ACK-NAK | knarfly: Voicepusle keeps getting more and more expensive! |
| 20:54.49 | DotHack | mrbnet: you cat have multiple default gateways! |
| 20:54.54 | DotHack | cant |
| 20:55.04 | VoIP-Penguin | For example, if you call me on AT&T, I bet I won't see your name. |
| 20:55.21 | *** join/#asterisk brezular (~brezular@bband-dyn80.178-40-13.t-com.sk) |
| 20:55.24 | Mango | So Katty - how old are you? ;) |
| 20:55.36 | hardwire | 18! |
| 20:55.42 | Mango | ducks whatever Katty threw at him |
| 20:55.56 | ACK-NAK | VoIP-Penguin: if you're calling anyone on the PSTN with Callerid then YES, it will be looked up. Any RBOC will do that dip even if two-bit ITSP's are too cheap to pay the dipping fees. |
| 20:56.04 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 20:56.05 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 20:56.05 | Gugge | DotHack: you can use source routing though, and have a default gateway pr source ip |
| 20:56.11 | hardwire | nowait.. 30 flirty and thriving. |
| 20:56.21 | DotHack | mrbnet: sorry, it is possible just google it |
| 20:56.26 | beek | Mango is playing with fire. |
| 20:56.53 | VoIP-Penguin | ack-nak: How can you guarantee that AT&T is going to lookup your CNAM from an LiDB that your CNAM has been entered into? |
| 20:57.07 | knarfly | Katty: I'm a native Dallas Texan transplanted to Miami...would love to be back in Big D again |
| 20:57.20 | ACK-NAK | VoIP-Penguin: because you pay to insert it into their database. |
| 20:57.30 | VoIP-Penguin | hmm |
| 20:57.46 | mrbnet | DotHack: err, right, that doesn't make sense. I have one interface connected to the Internet the other in on an internat network with many different subnets. Is there a way to make the route go back out the nic it came in on, easily? |
| 20:58.13 | VoIP-Penguin | ack-nak: I tried to get AT&T to enter my name into their DB, but no one there seems competant enough to get it done for me. Any tips? |
| 20:59.17 | ACK-NAK | VoIP-Penguin: Of course you couldn't. The big boys dont' give a SHIT about you or about the tiny ITSP's of the world but when Vitelity partners with a very large carrier and slip LIDB assignment as a legal technicalitiy THEN they can set CNAM For most numbers, not all. So far I'm at about 90% 100% for regional numbers. |
| 20:59.45 | ACK-NAK | I run in to trouble in the rural telcos of the world. |
| 21:00.17 | VoIP-Penguin | I use VoIP.ms, which is a Vitelity reseller. Maybe I can get Vitelity to do it for me. |
| 21:00.20 | DotHack | mrbnet: yes thats possible but i have to look that up |
| 21:00.40 | DotHack | mrbnet: are you using firewall? |
| 21:00.57 | ACK-NAK | VoIP-Penguin: Possible. They charge a 1x $10 fee. Takes over a month. Some iTSP's can inject your number into CNAM using a '911 technicality', but that workaround carries monthly fees. |
| 21:01.45 | ACK-NAK | VoIP-Penguin: Telasip.com can set CNAM for numbers that Vitelity can't but their 'fix' carries a MRC. Not so VIte. |
| 21:02.01 | ACK-NAK | vite... |
| 21:02.03 | ACK-NAK | vitel. |
| 21:02.14 | VoIP-Penguin | ack-nak: I think I'll take one more shot with getting AT&T to do it for me for free before I try to get Vitelity to do it. |
| 21:02.26 | ACK-NAK | You can use any two-bit for thermination, but origination requires careful selection. |
| 21:02.35 | ACK-NAK | VoIP-Penguin: You're wasting your time. |
| 21:02.49 | ACK-NAK | VoIP-Penguin: You are absolutely positiviely wasting your time. |
| 21:03.17 | DotHack | mrbnet: what exactly are you trying to do? |
| 21:03.19 | mrbnet | DotHack: Yes, the only way I can think is to copy the static routes from the firewall to the asterisk box. All the end points are behind the the firewall with the asterisk box. Phones are in multiple locations on multiple subnets. I would like to peer my asterisk box with another but avoid the natting firewall. |
| 21:03.59 | *** join/#asterisk teknoprep (~Chris@unaffiliated/teknoprep) |
| 21:04.07 | teknoprep | hey all |
| 21:04.43 | DotHack | so you have 2 asterisk boxen with 2 natting firewalls between them? |
| 21:04.49 | DotHack | mrbnet: |
| 21:05.22 | ACK-NAK | VoIP-Penguin: I used to port my numbers TO AT&T (RCF) just long enough to get the CNAM set. THen I'd port to the ITSP of choice. THen the record would sit out there as an orpahn. Years ago they started 'killing orphans'. They don't want to help you. People like us are killing their core business . |
| 21:05.33 | *** join/#asterisk jro (~jaredo@ganondorf.loclhst.com) |
| 21:05.52 | Katty | http://i.imgur.com/EAzRQ.jpg <- the voices of Megatron (left) and Optiumus Prime (Right) |
| 21:05.59 | jro | Are there any setups or modules that have the line dispaly "on another call" or similar when dialing an extension and the person is on a line? |
| 21:08.14 | DotHack | is it possible with a sip phone to show the name of the callee on the callers phone? |
| 21:08.20 | DotHack | and if so how? |
| 21:08.22 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 21:08.30 | mrbnet | DotHack: only one natting firewall, I would also prefer to put those asterisk services on another IT |
| 21:08.36 | mrbnet | DotHack: IP* |
| 21:09.28 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:09.33 | DotHack | mrbnet: but how do you plan to circumvent the natting firewall? |
| 21:10.22 | mrbnet | DotHack: through a second NIC connected directly to the net, firewalled of course |
| 21:11.28 | DotHack | mrbnet: you can add ip of the peering asterisk box in your routing table with a gateway of the net |
| 21:11.46 | DotHack | and let your default 0.0.0.0 routing alone |
| 21:12.10 | hardwire | when I first got started with asterisk the Sipura SPA-3000 was my main device. |
| 21:12.33 | hardwire | what was weird is I kept seeing spa3k as an abbreviation.. and I kept wondering what idiot kept mispelling and leeting the word 'speak' |
| 21:12.39 | hardwire | sigh. |
| 21:13.19 | Katty | http://i.imgur.com/cOen3.jpg <- long, but so totally worth it |
| 21:13.29 | Katty | ^- not recommended with those who have religious sensitivities |
| 21:13.35 | Katty | s/with/for/ |
| 21:13.54 | mrbnet | DotHack: I would route that IP though the default route of the internet NIC, right? |
| 21:14.12 | DotHack | yes |
| 21:14.15 | *** join/#asterisk thecardsmith (~doug@65-183-130-234-dhcp.burlingtontelecom.net) |
| 21:14.23 | DotHack | mrbnet: yes that would be correct |
| 21:15.00 | DotHack | mrbnet: but not the nic that is connected to the nat fw |
| 21:15.33 | mrbnet | DotHack: right, I will give that a try |
| 21:17.09 | mrbnet | DotHack: I was hoping to stay away from managing some static routes if the number of peers starts to grow |
| 21:17.48 | DotHack | mrbnet: maybe then you have to consider removing the asterisk box from the nat |
| 21:20.01 | *** join/#asterisk hesco (~hesco@c-24-99-160-121.hsd1.ga.comcast.net) |
| 21:21.34 | hesco | is there any experience documented somewhere related to sizing servers to provide capacity for the meetme application? |
| 21:25.00 | Kobaz | strange |
| 21:25.18 | Kobaz | it looks like TrySystem SYSTEMSTATUS in very unreliable |
| 21:26.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:27.43 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:28.52 | ecrane | Katty: Nice one |
| 21:30.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:31.04 | hardwire | it's [TK]D-Fender ! |
| 21:31.10 | hardwire | he'll fix it! |
| 21:36.11 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 21:37.27 | Katty | grumbles at the bank |
| 21:38.05 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 21:39.19 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 21:39.43 | Deeewayne | Kobaz, I think it depends on the return code of the application and not all applications return consistent results, like app_queue |
| 21:42.56 | Kobaz | this isn't tryexec |
| 21:42.59 | Kobaz | this is trysystem |
| 21:43.05 | Kobaz | trysystem(test -r /foo/bar) |
| 21:43.11 | Kobaz | returns very inconsistant results |
| 21:43.53 | Deeewayne | yeah, tryexec....sorry |
| 21:43.53 | Kobaz | i haven't looked at the code, but i think SYSTEMSTATUS doesn't take into account the return code of the command |
| 21:44.00 | Deeewayne | you're correct |
| 21:44.08 | Deeewayne | my mistake |
| 21:44.28 | bn-7bc | can anuone point me to a place in europe I (as a peuvate person) cab buy a 877w with advanced ip cervixesc pre installed? |
| 21:44.36 | Kobaz | i'm getting SUCCESS even if the file doesn't exist |
| 21:44.50 | Kobaz | digs |
| 21:44.59 | Deeewayne | shrugs |
| 21:45.24 | Kobaz | but... i'm getting either APPERROR or SUCCESS... randomly... on a file that doesn't exist |
| 21:45.57 | Kobaz | so that's kinda useless |
| 21:46.03 | bn-7bc | I tried to find a reseler localy (ib borway) but they dont sell to non bizz users :( |
| 21:50.02 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 21:51.29 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 21:52.10 | Kobaz | i have a question about the question for questioning questions that result in questions |
| 21:52.13 | Kobaz | damnit |
| 21:52.14 | Kobaz | wrong window |
| 21:52.17 | Kobaz | er |
| 21:54.43 | mmlj4 | Kobaz: still funny, though |
| 21:54.47 | Kobaz | heh |
| 21:54.57 | vader-- | can you do parts of your confiruation in realtime? |
| 21:55.03 | Kobaz | sure |
| 21:55.11 | vader-- | like if i only want to do sip.conf in a database? |
| 21:55.14 | Kobaz | half my stuff is realtime, half my stuff is still config files |
| 21:55.33 | vader-- | what do you have in realtime and whats in config files? |
| 21:55.46 | Kobaz | realtime is sip/iax/extensions/voicemail |
| 21:56.14 | hardwire | I like using views for realtime extensions :) |
| 21:56.15 | hardwire | it's evil |
| 21:56.16 | Kobaz | config files is the base stuff like loggers, indications, etc |
| 21:56.32 | Kobaz | my stuff just calls stored procedures |
| 21:56.52 | hardwire | create view select * from sippeers and turn it into extensions format crazyness |
| 21:57.03 | Kobaz | yeah |
| 21:57.39 | hardwire | I stay away from stored procedures since they aren't inter-DB proof enough. |
| 21:57.57 | hardwire | since I assume I'll change the backend DB at any given moment :) |
| 21:59.36 | vader-- | do you guys use any front end to manage the databases? |
| 21:59.42 | Kobaz | pgadmin3 |
| 22:00.14 | Kobaz | hardwire: if you're going to change the db backend at any moment... i would say that's not a very good development methodology |
| 22:00.24 | bn-7bc | bardon my questions they where ment for #cisco |
| 22:01.46 | hardwire | Kobaz: creating a product for multiple walks of life.. is all :) |
| 22:02.56 | Kobaz | heh |
| 22:03.10 | Kobaz | so your users may change the backend at any moment |
| 22:03.45 | hardwire | I'd like to see something work well between DBs if it ever turns into a product |
| 22:04.03 | hardwire | otherwise I'd be pimping python in postgresql :) |
| 22:06.11 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 22:12.36 | Slugs_ | . |
| 22:12.55 | Slugs_ | hey hardwire |
| 22:13.02 | Slugs_ | que pasa |
| 22:20.24 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 22:22.36 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:22.36 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 22:25.25 | bmoraca_work | rofl @ http://thedailywtf.com/Articles/These-Go-To-Fourteen.aspx |
| 22:25.42 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 22:26.40 | *** join/#asterisk crizzly (~ttt@g230053167.adsl.alicedsl.de) |
| 22:27.08 | crizzly | hey ho. |
| 22:27.58 | crizzly | is it possible to add certain colors in asterisk CLI to VERBOSE messages ? |
| 22:28.40 | crizzly | also why in last csv in verbose-level 0 or off, i still these "-- Locally bridging SIP/...." |
| 22:29.28 | citywok | is there a functional, still findable outlook dialer for * |
| 22:30.06 | *** join/#asterisk rooky (~rooky@p5B179C87.dip0.t-ipconnect.de) |
| 22:31.04 | bmoraca_work | citywok: snap-a-number is the best i've seen |
| 22:31.19 | Kobaz | but every dial to a local channel is more calls on the stack |
| 22:31.22 | Kobaz | er |
| 22:31.40 | bmoraca_work | i know that digium bastardized it, but you might be able to find a download in the wild |
| 22:33.13 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 22:33.16 | citywok | thanks i'll take a look |
| 22:33.23 | citywok | of course i have x64 beta 2010, gotta find an 07 test boxc |
| 22:34.24 | *** join/#asterisk s4msung (~s4msung@dice.s4msung.de) |
| 22:34.43 | hardwire | Slugs_: hi |
| 22:35.00 | hardwire | dislikes programming IVR |
| 22:39.07 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 22:39.50 | adnc | i get a "Got SIP response 400 "Bad Request" back from 83.169.182.1" although sip show peer shows in the status field OK |
| 22:40.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:40.27 | adnc | Registration for '05312575555@reg01.kabelphone.de@proxy.kabelphone.de' timed out, trying again |
| 22:40.52 | adnc | i get these all the time and |
| 22:41.49 | jdoe | how can I troubleshoot cdr_odbc? odbc.ini etc. is setup properly since I can connect with isql... cdr_odbc.conf has the same information, res_odbc.conf uses the dsn in odbc.ini and is connected... but calls aren't being logged, csr_odbc complains that it can't find the database handle. |
| 22:43.42 | *** join/#asterisk Digr (~kvirc@ppp079166061056.dsl.hol.gr) |
| 22:43.49 | Digr | Hi |
| 22:45.32 | Digr | i have a problem with my asterisk gui. i have installed asterisk gui but when i try http://192.168.2.20:80/asterisk/static it says Access Denied sorry i cannot let you do that Dave |
| 22:45.45 | Digr | can you plz give some help? |
| 22:46.36 | bmoraca_work | Digr: you're not going to find any help here with that |
| 22:46.48 | bmoraca_work | Digr: try #asterisk-gui i think it is |
| 22:47.11 | Digr | ok thank you bmoraca |
| 22:47.33 | *** part/#asterisk Digr (~kvirc@ppp079166061056.dsl.hol.gr) |
| 22:48.14 | doctorray | hooray for successful test calls! |
| 22:48.25 | citywok | bmoraca_work: that works well on my x32 office 2007 |
| 22:48.32 | citywok | beta x64 2010 not so much |
| 22:48.39 | bmoraca_work | not surprised |
| 22:51.59 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:52.49 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 22:55.58 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 23:09.41 | jdoe | anyone use cdr_odbc? |
| 23:10.00 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 23:19.37 | *** join/#asterisk nickaugust (~anonymous@34.124.188.72.cfl.res.rr.com) |
| 23:20.16 | ACK-NAK | Alaysauthreject = yes, not yet supported in iax2? |
| 23:20.46 | *** join/#asterisk Mango (~iMango@d154-20-86-138.bchsia.telus.net) |
| 23:23.31 | Katty | hi |
| 23:23.44 | Katty | i just got back from running |
| 23:26.50 | ACK-NAK | Good for you Katty! |
| 23:26.53 | ACK-NAK | Seriosuly I mean it. |
| 23:27.02 | ACK-NAK | I used to run more. Now I bike. 12 Marathons. |
| 23:28.04 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 23:29.16 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 23:32.00 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-128-92.socal.res.rr.com) |
| 23:32.49 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 23:33.45 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 23:34.04 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 23:34.51 | *** part/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 23:36.15 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 23:37.43 | *** join/#asterisk xpot-mobile (~james@66.60.101.91) |
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| 23:47.06 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 23:47.19 | bmoraca_work | i wonder...is Asterisk Y2K38 compliant? |
| 23:47.28 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 23:47.51 | jstapleton | is there a good way to tell what dtmfmode is being used in a conversation? |
| 23:51.16 | wart___ | is there a way to trace individual ports? I have reason to suspect that the router I'm sitting behind or the Cable ISP provider is doing strange things. My various softphones are pretty intermittent when it comes time to make a call. |
| 23:52.01 | Mango | Is there any way to forward port 5061 to 5060 using a router, for outgoing connections? |
| 23:52.21 | Mango | My device wants to connect on 5061 but this Asterisk server is only open on 5060. |
