| 00:00.25 | RipeR-81 | nevermind found it |
| 00:04.44 | phix | umm where is the asterisk reference? |
| 00:05.02 | ManxPower | ~book |
| 00:05.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:05.59 | phix | oh it is in there? |
| 00:07.01 | mvanbaak | bye all |
| 00:07.25 | phix | hmmmmm |
| 00:12.36 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 00:13.32 | *** join/#asterisk Porks (i=Porks@200-148-39-249.dsl.telesp.net.br) |
| 00:13.41 | phix | Porks! |
| 00:14.10 | phix | Been a while ay :) |
| 00:14.16 | Porks | phix, hi! |
| 00:14.32 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 00:15.16 | esaym | what is a decent software phone for linux that works with asterisk? I need to test my setup and my hardware phones are still in the mail... |
| 00:22.33 | phix | esaym: anything that uses SIP |
| 00:22.44 | phix | xlite isn't bad |
| 00:22.50 | phix | but not opensource |
| 00:23.36 | esaym | I can't really find anyting |
| 00:23.56 | JunK-Y | use zoiper. |
| 00:24.06 | *** join/#asterisk RoyK (n=roy@91.149.13.232) |
| 00:25.11 | phix | esaym: can you find xlite? |
| 00:25.45 | esaym | yea I am looking right npow |
| 00:30.25 | _ShrikE | phix: devstate is not in 1.2 |
| 00:32.38 | phix | :( |
| 00:32.54 | *** part/#asterisk Porks (i=Porks@200-148-39-249.dsl.telesp.net.br) |
| 00:33.13 | Olobola | is fastagi ok in a busyish production environment? |
| 00:33.29 | phix | ok well I dont use the vairable that dial status sets, I use dial+101 (where dial is set to the umm number where the Dial call is :)) |
| 00:34.46 | phix | _ShrikE: the problem is dial+101 gets jumped to if the person being called hangs up on a mobile instead of answering |
| 00:35.04 | phix | Any was to tell the difference? |
| 00:35.16 | mosty | phix, it's best not to use priority jumping, better to use ${DIALSTATUS} |
| 00:35.46 | phix | mosty: is that the same thing though? |
| 00:36.15 | phix | I mean with ${DIALSTATUS} tell me that the person being called press the red button instead of the green one? :) |
| 00:36.17 | esaym | hmm this might work: http://www.gizmoproject.com/download-linux.html |
| 00:36.53 | *** join/#asterisk EvilDeshi (n=deshi@75-130-24-153.dhcp.mdsn.wi.charter.com) |
| 00:37.05 | phix | Awesome hold music, too bad it is breaching copyright :P |
| 00:37.18 | EvilDeshi | is there a tutorial somewhere for getting flite to work with *? |
| 00:37.48 | mosty | phix, the code that performs priority jumping does so by using ${DIALSTATUS}, so you can use it to cause exactly the same behaviour if you wish |
| 00:38.19 | mosty | Olobola, i thought the whole idea of fastagi was for busy servers |
| 00:38.31 | phix | where is the list of result codes for DIALSTATUS? |
| 00:38.45 | mosty | on the wiki i think |
| 00:39.43 | phix | ok |
| 00:42.25 | *** join/#asterisk RoyK (n=roy@91.149.13.232) |
| 00:43.36 | lmadsen | _ShrikE: hey, just trying to figure out how to schedule a shipment on fedex.com :) |
| 00:44.11 | _ShrikE | lmadsen: you mean like scheduling a pickup? |
| 00:44.16 | Olobola | mosty: I keep running into complaints about phpagi/fastagi. Would perl make much of difference over PHP for a fairly simple phone maze (db connections etc)? |
| 00:44.20 | lmadsen | _ShrikE: aye |
| 00:44.47 | lmadsen | Olobola: why not just use the dialplan and func_odbc for DB connections? |
| 00:45.33 | phix | yay, it is CHANUNAVAIL |
| 00:45.48 | phix | so CHANUNAVAIL == 101 ? |
| 00:45.58 | phix | hmm now to find out what the numbers mean |
| 00:46.13 | *** part/#asterisk RoyK (n=roy@91.149.13.232) |
| 00:47.02 | Olobola | lmadsen: I've written 500 lines of code already. If have to switch I will, just not sure if I absolutely need to. |
| 00:47.10 | lmadsen | Olobola: gotcha |
| 00:47.20 | lmadsen | just use the built in DB connections for PHP |
| 00:47.34 | lmadsen | I didn't read the scrollback though, so I'm sure I've missed a bunch of the conversation |
| 00:47.57 | _ShrikE | lmadsen: I think I can do that for you. Do you want it picked up at your APT? |
| 00:47.58 | lmadsen | _ShrikE: hrmmm... well, I'm waiting for fedex.com to email me my login, so it might just be quicker to walk up to a shipping centre tomorrow |
| 00:48.05 | lmadsen | _ShrikE: sure, that'll work too :) |
| 00:48.10 | lmadsen | I'll be here all day tomorrow |
| 00:48.17 | EvilDeshi | how do i load app_flite i added it to modules.conf and still when i try the command flite it says that application was not found |
| 00:48.35 | Olobola | lmadsen: I just don't want to have to rewrite crap! Everything works fine as is, just not sure about scalability. |
| 00:48.41 | lmadsen | *CLI> module load <module>.so |
| 00:48.55 | lmadsen | Olobola: ahhh.. gotcha -- SIPp and test it :) |
| 00:49.15 | lmadsen | no one can answer your question about scalability -- it can only be answered by actually testing |
| 00:49.20 | EvilDeshi | lmadsen it said it did not register itself and could not load |
| 00:49.32 | Olobola | ok, thank you. |
| 00:49.34 | lmadsen | then the module probably doesn't exist in /usr/lib/asterisk/modules/ |
| 00:49.40 | lmadsen | ~sipp |
| 00:49.40 | jbot | [sipp] a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/. If you really want to know how many channels your Asterisk box can do, learn how to utilize this program. |
| 00:49.44 | EvilDeshi | it does as i put that module there |
| 00:49.58 | lmadsen | then there is something possibly wrong with the module if it won't load |
| 00:50.24 | EvilDeshi | ok where can i get the latest verion of that app? |
| 00:50.36 | lmadsen | I have never heard of it, so it's obviously not standard asterisk |
| 00:50.38 | EvilDeshi | i honestly grabbed it from some rpm |
| 00:50.46 | lmadsen | google probably is your best bet |
| 00:50.58 | EvilDeshi | yeah been there done that which is why i came here oh well no big deal |
| 00:51.06 | EvilDeshi | im sure i can just use festival |
| 00:51.10 | lmadsen | then that app is probably very old and not supported by anyone |
| 00:51.16 | EvilDeshi | I was messing with a weather AGI script |
| 00:51.27 | EvilDeshi | its new app |
| 00:51.29 | _ShrikE | lmadsen: any time better than another? |
| 00:51.34 | lmadsen | _ShrikE: anytime after 10am :) |
| 00:51.35 | EvilDeshi | some new version of festival TTS |
| 00:52.06 | lmadsen | _ShrikE: although even before then is fine too, just make a note to "knock loud" because my apartment has a long hallway, so I can't always hear the door |
| 00:52.58 | drmessano | Weather AGI? |
| 00:52.59 | _ShrikE | lmadsen: Done.. tomorrow between 11:00am and 3:00pm |
| 00:53.01 | drmessano | From where? |
| 00:53.24 | EvilDeshi | drmessano it was a nerd vittles post |
| 00:53.26 | lmadsen | _ShrikE: awesome -- can you msg my your fedex number so I can mark the "Bill duties and taxes to" box? |
| 00:53.36 | EvilDeshi | http://bestof.nerdvittles.com/applications/weather-world/ |
| 00:53.39 | lmadsen | oh nevermind |
| 00:53.42 | lmadsen | I see it in the box above :) |
| 00:53.50 | drmessano | Oh |
| 00:53.53 | drmessano | Thats not new lol |
| 00:54.03 | EvilDeshi | oh its not? |
| 00:54.08 | lmadsen | maybe new to you... |
| 00:54.10 | EvilDeshi | im wondering if i can just edit the perl script |
| 00:54.15 | EvilDeshi | and replace flite with festival calls |
| 00:54.18 | *** join/#asterisk tripps (n=ss@c-76-31-153-101.hsd1.tx.comcast.net) |
| 00:54.58 | drmessano | I dunno.. I use the U.S. one |
| 00:55.05 | drmessano | It works pretty well |
| 00:55.09 | EvilDeshi | got a link for that one? |
| 00:55.28 | drmessano | http://bestof.nerdvittles.com/applications/weather-zip/ |
| 00:55.36 | EvilDeshi | yeah i got that one too |
| 00:55.48 | EvilDeshi | it too uses flite |
| 00:55.52 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582030.dsl.bell.ca) |
| 00:56.16 | drmessano | yeah |
| 00:56.18 | drmessano | It works |
| 00:56.24 | EvilDeshi | well i cant get flite to load |
| 00:56.28 | EvilDeshi | that is the problem i am having |
| 00:56.32 | drmessano | oh |
| 00:56.36 | EvilDeshi | the module wont load |
| 00:58.45 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:59.32 | *** part/#asterisk asr33 (n=asr33@dsl-207-112-74-61.tor.primus.ca) |
| 01:00.59 | lmadsen | interesting, for some reason this packing tape smells like chocolate |
| 01:01.13 | lmadsen | oh! I know what it is now... cocoa puffs |
| 01:01.18 | _ShrikE | its to throw off the dogs |
| 01:01.39 | lmadsen | smells exactly like it... which kinda scares me, because that probably means some chemical in the glue is also in cocoa puffs :) |
| 01:01.43 | lmadsen | _ShrikE: lol |
| 01:02.20 | *** join/#asterisk georgem11 (n=g@64.19.182.18) |
| 01:02.57 | georgem11 | Hey if anyone is alive I am having a wierd issue that I could not find a similar case of it happening to anyone else on the forums. I have a successful install of asterisk for 2 years now without any hiccups. However today, I am only getting one way voice when making outgoing calls (they can't hear me, but I can hear them). The only recent change I did was add an extension. I am really stumped by this and our telco provider swears up and down it is not th |
| 01:03.38 | lmadsen | ideally you should find another provider to test with.. also, you can debug with 'rtp debug' to make sure the RTP packets are going in both directions |
| 01:03.46 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 01:04.35 | georgem11 | testing with another provider isnt really an option, everything is setup with a t1 pri at the moment on the server |
| 01:05.13 | lmadsen | all you can do is verify the rtp is flowing then |
| 01:05.29 | lmadsen | you're getting one way audio on your T1? |
| 01:05.41 | georgem11 | we have 5 T1's bonded into the Telco's adtran |
| 01:05.44 | jblack | Perhaps someone controls a firewall between you and the endpoint, and recently started blocking more ports. |
| 01:05.57 | georgem11 | from there its a pri handoff into the t1 card on asterisk |
| 01:05.57 | lmadsen | sounds like he's not using VoIP.......... so I'm confused |
| 01:06.09 | georgem11 | yeah its not voip externally, just internally |
| 01:06.31 | lmadsen | all you can do is what I suggested I guess |
| 01:06.39 | lmadsen | that's all I do when I check for one-way audio |
| 01:07.07 | lmadsen | then I verify the configuration of my peers (canreinvite=no, nat=yes) |
| 01:07.31 | georgem11 | how do I do that debug command, Ive never done one before? |
| 01:07.44 | lmadsen | 'rtp debug' |
| 01:07.47 | lmadsen | as I stated.... |
| 01:08.01 | lmadsen | and 'rtp no debug' to turn off |
| 01:09.54 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 01:10.07 | georgem11 | once I do that in the cli it shows up in the full debug log? |
| 01:11.41 | TJNII | georgem11: So you have a T1 to the telco, but what kind of phones? |
| 01:12.13 | georgem11 | on this call I called out to my mobile but I was unable to hear myself when talking on the mobile to my desk phone |
| 01:12.17 | georgem11 | Jan 15 20:10:53 VERBOSE[4543] logger.c: Got RTP packet from 10.34.200.147:10030 (type 0, seq 58832, ts -485918959, len 160) |
| 01:12.18 | georgem11 | Jan 15 20:10:53 VERBOSE[4543] logger.c: Sent RTP packet to 10.34.200.147:10030 (type 0, seq 11301, ts 155200, len 160) |
| 01:12.18 | georgem11 | Jan 15 20:10:53 VERBOSE[4543] logger.c: Got RTP packet from 10.34.200.147:10030 (type 0, seq 58833, ts -485918799, len 160) |
| 01:12.18 | georgem11 | Jan 15 20:10:53 VERBOSE[4543] logger.c: Sent RTP packet to 10.34.200.147:10030 (type 0, seq 11302, ts 155360, len 160) |
| 01:12.20 | georgem11 | Jan 15 20:10:53 VERBOSE[4543] logger.c: Got RTP packet from 10.34.200.147:10030 (type 0, seq 58834, ts -485918639, len 160) |
| 01:12.27 | georgem11 | that was some of the rtp logs, not sure how to interpret them |
| 01:12.44 | georgem11 | we use polycom 501 SIP's |
| 01:12.48 | jblack | Yay autoignore. |
| 01:12.52 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d663e74a195e255b) |
| 01:13.02 | JT | thanks for that georgem11 |
| 01:13.33 | TJNII | georgem11: Are the polycoms on the same network? |
| 01:13.38 | *** join/#asterisk joe (n=nnnnnnnn@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 01:13.48 | georgem11 | yeah its's all on a flat lan with the pbx |
| 01:13.58 | JT | georgem11: please do not flood the channel again |
| 01:14.14 | georgem11 | sorry, i only meant to copy and paste two but a few more came with it |
| 01:15.11 | georgem11 | can anyone tell me what the RTP logs mean? |
| 01:15.32 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 01:15.39 | joe | what's the name of the voicemail subsystem for asterisk? |
| 01:15.48 | lirakis | joe: ? |
| 01:16.00 | lirakis | joe: the applications you can call? |
| 01:16.07 | jblack | joe: Commedian mail? |
| 01:16.17 | JT | georgem11: it means it sent and received packets |
| 01:16.28 | _ShrikE | georgem11, it looks like asterisk is sending and receiving audio to and from 10.34.200.147, which I assume is your polycom? |
| 01:16.44 | joe | jblack: ah right! thanks |
| 01:17.07 | jblack | bleh. Someone with a noisy line is sitting on coffeehouse. I suspect he's not getting packets either, so nobody can tell him his line is crappy. |
| 01:17.19 | jblack | Well, either that, or he shouldn't make calls from within power supplies. ;) |
| 01:17.25 | joe | hehe |
| 01:17.27 | tripps | ~dynamic |
| 01:17.29 | JunK-Y | pwd |
| 01:17.38 | jblack | /home/junk-y |
| 01:18.18 | georgem11 | yes, 10.34.200.147 is the polycom on my desk |
| 01:18.43 | georgem11 | i tested from other 501's around the office and a few 301's and had the same results. rebooted them as well and same results |
| 01:18.45 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 01:19.00 | TJNII | georgem11: Is it only incoming/outgoing? Does an echo test work? |
| 01:19.18 | georgem11 | incoming calls to the pbx are fine, voice works both way |
| 01:19.28 | JunK-Y | jblack: i know i was lost, thanks :) |
| 01:19.36 | _ShrikE | If you truly havent changed anything I would start to suspect the card |
| 01:19.45 | tripps | just to clarify - setting host to fixed IP for a friend rather than dynamic would stop the device from registering with *? also can you have static host and qualify=yes? |
| 01:19.45 | _ShrikE | or the provider equipment |
| 01:19.48 | georgem11 | outgoing calls to anywhere , they usually cant hear me, or sometimes i cant here them |
| 01:20.15 | TJNII | Just outgoing calls on yout T1 lines |
| 01:20.22 | georgem11 | yup |
| 01:20.28 | _ShrikE | not incoming? |
| 01:20.30 | georgem11 | nope |
| 01:20.35 | _ShrikE | buy thats odd |
| 01:20.47 | _ShrikE | s/boy/buy/ |
| 01:20.49 | tripps | ~book |
| 01:20.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:21.17 | georgem11 | thats why I dont suspect the card, since incoming calls are fine |
| 01:21.24 | _ShrikE | yeah |
| 01:21.35 | georgem11 | i would like to blame the provider, but want to cover all of the bases before they come out tomorrow |
| 01:22.21 | _ShrikE | the'll put a tberd on it and have an answer for your quickly |
| 01:22.25 | georgem11 | those rtn debug logs show that voice is going both ways from those snipets i showed? |
| 01:22.49 | georgem11 | or at least asterisk thinks voice is going both ways |
| 01:22.51 | _ShrikE | yes, you see the "sent packet to" and "got packet from" entries |
| 01:23.25 | georgem11 | yeah, the telco said they would put a t-berd on it any know right away if it was the router |
| 01:23.25 | jblack | Hmm. Perhaps its me. |
| 01:23.38 | *** join/#asterisk lters (n=tech@mrtcdsl-433.mis.net) |
| 01:23.44 | TJNII | The fact that it is intermittent on outgoing only means that it probably isn't a SIP/RTP issue to the phone. |
| 01:23.48 | _ShrikE | georgem11, thats part of what you pay them for. |
| 01:24.16 | lters | http://svn.digium.com/svn/asterisk/branches/1.4 seems to not exist? |
| 01:24.32 | mosty | what channel variables (if any) are passed to the server with iax? |
| 01:24.34 | georgem11 | yeah, but you know how the blame game goes for telco's, they will probably come out here and it will work fine |
| 01:24.37 | georgem11 | and I am back to square one |
| 01:25.18 | joe | is there a limit on the number of sip phones * can support? |
| 01:25.28 | georgem11 | lmadsen: do you have any more ideas for the rtp debug? |
| 01:26.35 | georgem11 | or any other ideas in general? |
| 01:30.14 | TJNII | zomg.... I think I just found my new moh .... "Banned Barbershop ballads" |
| 01:30.17 | *** part/#asterisk shtoom (n=godson@59.93.117.175) |
| 01:30.54 | tripps | i guess let me re-ask my question a different way; is it a best practice with fixed endpoints (i.e., deskphones) to give them static IPs, not have them register with * and set host=<IP> and turn off qualify to cut down on SIP traffic on the LAN? |
| 01:30.58 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 01:31.16 | jblack | Hmm. I don't think they can hear me. |
| 01:32.04 | TJNII | tripps: Why do you need static IPs for that? |
| 01:32.19 | TJNII | Oh, nm. I misread that. |
| 01:34.43 | tripps | i suppose it depeneds on the size of the network though - if you're deploying dozens of endpoints it maybe easier to use DHCP and register in the tftp config or whatever |
| 01:35.40 | *** join/#asterisk goatmilk (n=goatmilk@ip68-100-115-83.dc.dc.cox.net) |
| 01:36.09 | TJNII | Yea, if you have enough that you have to care about SIP traffic you have enough you want to have them autoprovision. |
| 01:36.51 | tripps | i don't suppose it's possible to configure the sip debug on the CLI so that it only logs certain kinds of messages . . . when debugging call issues it's a pain when you've got dozens of REGISTER messages to cull through . . . of course you could increase the interval i suppose |
| 01:36.51 | goatmilk | ergo... |
| 01:39.30 | mosty | tripps: you can tail -f /var/log/asterisk/full | grep -v 'patterns you dont want to see' |
| 01:39.59 | georgem11 | <PROTECTED> |
| 01:40.01 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:40.54 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 01:41.53 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:42.15 | phix | What is the point of NoOp? just for logging purposes? |
| 01:43.08 | *** join/#asterisk jblack (n=jblack@pool-71-181-185-193.sctnpa.east.verizon.net) |
| 01:43.15 | mosty | phix: logging, and also useful where you want a priority label but no particular action |
| 01:43.38 | jblack | Hmm. I'm getting a metric boatload of droped tcp packets with a dest of 25932. Could that be * related? |
| 01:43.43 | phix | ok |
| 01:43.46 | phix | thank you :) |
| 01:44.05 | phix | haha metric boatload :) I like |
| 01:44.23 | *** join/#asterisk BeeBuu (n=beebuu@219.135.43.116) |
| 01:45.06 | jblack | Yeah, coming from all over the interwebs. |
| 01:45.17 | phix | is that format of measurement similar to the length of a metric peice of string? |
| 01:45.24 | phix | piece even |
| 01:45.27 | TJNII | That's 1.567 standard boatloads for you imperial types |
| 01:45.32 | phix | hehe |
| 01:45.37 | phix | funny |
| 01:48.16 | tripps | mosty: yeah that's what i already do . . . .the pain isn't on the * end in my case, it's on the sip gateway end where the SIP debug messages aren't nearly well as organized and it's almost impossible to cull them out without some fancy block regex filtering |
| 01:49.17 | tripps | s/well as/as well/ |
| 01:49.38 | jblack | Nah, it can't be *. It's coming from a metric boatload of different hosts. Must be a new worm or something. |
| 01:51.30 | mosty | tripps, if you find a good way, let me know :) |
| 01:52.13 | tripps | my dad has a NEC KSU in his house with fancy nec phones but wants to play with voip - i'm wondering if I can splice in a sipura 3000 or equivalent, put * on the lan and still be able to use those phones somehow . . . |
| 01:52.43 | *** join/#asterisk RoyK (n=roy@91.149.17.219) |
| 01:53.38 | tripps | that is without installing a $$$$ voip card nec would love to sell . . . |
| 01:53.54 | TJNII | I wish wget's man page was in alphabetical order... |
| 01:54.31 | Qwell | I should make a SIP card for Asterisk. I'd make millions off of idio^H^H^H^Hcustomers |
| 01:54.51 | tripps | heh |
| 01:55.22 | *** join/#asterisk lzhang (n=lzhang@66-90-152-164.dyn.grandenetworks.net) |
| 01:55.51 | TJNII | Qwell: Just repackage NICs |
| 01:55.55 | tripps | googling phone systems and asterisk gives useless results since the pages always talk about the asterisk key on the phone ;) |
| 01:56.09 | puck | yes... |
| 01:56.11 | Qwell | TJNII: that's what I'm thinking. $10 realtek |
| 01:56.40 | file | Qwell: O.O |
| 01:56.43 | Qwell | :p |
| 01:56.47 | tripps | i'm thinking put the NEC sys on ebay and get polycoms . . . . probably what i'll end up doing |
| 01:56.58 | fujin | right, anyone familiar with diagonising a segmentation fault in Asterisk with a backtrace? |
| 01:57.06 | fujin | i've created the backtrace, (bt, bt full) |
| 01:57.07 | Qwell | file: we could sell it under the Telecomjoshvoxmart, Inc. brand |
| 01:57.13 | JunK-Y | fujin: read backtrace.txt |
| 01:57.14 | file | Qwell: good idea |
| 01:58.25 | fujin | bugger it, i wish you could redirect output from gdb |
| 01:59.10 | tripps | sweet current bid on nec electra phone sys on ebay with hours to go is $3.25 . . . . |
| 01:59.50 | Qwell | tripps: I wouldn't pay more than $3.18 |
| 02:00.03 | tripps | lol |
| 02:00.11 | lters | Qwell: where is the 1.4 svn branch? |
| 02:00.22 | Qwell | lters: gone for now |
| 02:00.35 | tripps | ah http://www.voip-info.org/wiki/view/Asterisk+legacy+integration |
| 02:00.36 | outtolunc | flew-da-coop |
| 02:00.45 | Qwell | lters: down for maintenance |
| 02:01.21 | lters | a server or vserver or something? |
| 02:03.22 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177649755.dsl.bell.ca) |
| 02:03.43 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 02:04.22 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 02:05.50 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 02:08.14 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 02:08.39 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-86-16-34.socal.res.rr.com) |
| 02:08.50 | jjshoe | is anyone aware of an application to see if a context exists? |
| 02:10.59 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 02:12.06 | fujin | how the hell do you add an attachment to a digium bugreport |
| 02:13.12 | *** join/#asterisk jblack (n=jblack@pool-71-181-149-51.sctnpa.east.verizon.net) |
| 02:13.18 | JunK-Y | jjshoe: DIALPLAN_EXISTS() and VALID_EXTEN() functions |
| 02:13.20 | jblack | Hmmm. Nobody can hear me on coffeehouse. |
| 02:13.30 | jblack | Yet I can hear myself on fwd echotest just fine. |
| 02:13.30 | mosty | fujin, there's an upload file thing on the bug page if you are the reporter, not sure if anyone can upload |
| 02:13.32 | JunK-Y | fujin: after you report it, return to ur bug and attach it. |
| 02:15.42 | voiper1 | is there anything wrong with running zaptel 1.4.x and asterisk 1.2.x? |
| 02:16.35 | mosty | voiper1, i believe it should work |
| 02:17.35 | voiper1 | it does im just wondering whether it will cause issues been different versions. |
| 02:17.36 | fujin | ah yep. |
| 02:17.50 | fujin | anyone with lots of Asterisk debug experience feel like taking a look at http://bugs.digium.com/view.php?id=11775 |
| 02:17.54 | fujin | ;] |
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| 02:19.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:20.47 | jjshoe | JunK-Y those looked to squeek in in the end of december, anything in 1.2? |
| 02:21.22 | JunK-Y | fujin: in gdb, can you do "p bridgepeer->name" ? |
| 02:21.36 | JunK-Y | bridgepvt = (struct sip_pvt *) 0x0 |
| 02:21.53 | JunK-Y | the private struct is NULL, so thats the problem, now why! |
| 02:21.55 | jblack | Just to be clear about this, sip can come in anywhere from port 10,000 to 20,000, right/ |
| 02:22.10 | JunK-Y | jjshoe: not that I know for 1.2, sorry. |
| 02:22.24 | jjshoe | jblack sip can be any port at all, along with rtp, which can be any port. |
| 02:22.38 | jblack | hmm. I must be blocking out sip calls then. |
| 02:22.52 | jjshoe | sip is usually 5060 |
| 02:22.56 | fujin | JunK-Y: http://bugs.digium.com/view.php?id=11775#80721 |
| 02:23.00 | TJNII | jblack: SIP is usually bound to one port, RTP is usually 10k-20k |
| 02:23.30 | jblack | I'll read over the firewall howto again to make sure I'm letting the right things through. |
| 02:23.36 | JunK-Y | so the chan is <ZOMBIE> that might be a reason why. |
| 02:23.48 | TJNII | jblack: Is the server on a public IP? |
| 02:23.58 | JunK-Y | can you reproduce it easily or you just let it run and it crash at some point? |
| 02:24.13 | fujin | JunK-Y: that's just normal running throughout the day |
| 02:24.44 | fujin | I don't know what is causing it, as such, cannot reproduce it |
| 02:24.53 | jblack | Yes, it is. I've set my firewall up to drop packets by default. I've opened up a handful of things (udp: 4520, 4569, 5060, 10000:20000, tcp: 10000:20000). |
| 02:25.07 | jblack | I can hear myself on fwd echotest, but people can't hear me in fwd coffeehouse. |
| 02:25.29 | jblack | I also allow through packets with a state of RELATED,ESTABLISHED |
| 02:25.55 | fujin | JunK-Y: any ideas? |
| 02:26.22 | TJNII | jblack: fwd -> free world dialup? |
| 02:26.38 | jblack | Yup |
| 02:26.42 | TJNII | mmkay |
| 02:26.56 | TJNII | If the server is on a public IP you shouldn't need to mess with the client side firewall |
| 02:27.10 | [TK]D-Fender | jblack, pastebin up your sip.cofn |
| 02:27.12 | [TK]D-Fender | conf* |
| 02:27.17 | TJNII | Are echotest and coffeehouse on the same server? I don't use FWD |
| 02:27.40 | jblack | wtf, the firewall comments don't match the firewall rules. |
| 02:27.44 | jblack | Case in point: |
| 02:27.48 | JunK-Y | fujin: the only "normal" <ZOMBIE> ive seen are redirected chans. |
| 02:27.50 | jblack | # SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well |
| 02:27.50 | jblack | iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT |
| 02:27.58 | JunK-Y | are you using redirect ? |
| 02:28.19 | fujin | JunK-Y: I initially thought that aswell, as I had noticed crashes before when the phones have SIP 302 callforwards turne don |
| 02:28.26 | jblack | I haven't defined redirect one way or another. |
| 02:28.26 | fujin | but went and checked 716 and it didn't have it on |
| 02:28.48 | TJNII | Jblack: Are echotest and coffeehouse on the same server? I don't use FWD |
| 02:29.00 | JunK-Y | i could write you a quick hack, but that wont be a proper fix, just a way to avoid * to crash |
| 02:29.12 | [TK]D-Fender | jblack, .....pastebin up your sip.conf.... |
| 02:29.13 | jblack | They sip to the same place, so probably. I can hear other people talk, they just can't hear me. |
| 02:29.18 | jblack | Sure thing. |
| 02:29.29 | fujin | JunK-Y: well, I'd rather try and work out what is causing it (if it's something I can control, perhaps) |
| 02:29.32 | jblack | Oh, I have canreinvite turned off |
| 02:29.41 | [TK]D-Fender | jblack, GOOD |
| 02:29.44 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:29.58 | TJNII | jblack: If the sip connection for echotest and coffehouse go to the same place, and one works then the problem is not your firewall. |
| 02:30.28 | JunK-Y | fujin: start logging full (with verbose) maybe you will see what makes it crashes. |
| 02:32.10 | jblack | http://pastebin.com/m307e8ae9 |
| 02:32.43 | jblack | TJNII: That's what I'd think. Yet I heard several other people talking with no problem, but seemingly no one hear me. |
| 02:32.50 | JunK-Y | fujin: like maybe 716 did something |
| 02:32.52 | jblack | s/hear/heard |
| 02:33.17 | TJNII | jblack: Is that a giant chat room? |
| 02:33.21 | jblack | [TK]D-Fender: I removed the local phone contexts from sip.conf. |
| 02:33.35 | jblack | TJNII: It is a chatroom. I don't think it ever gets giant in size. ;) |
| 02:33.49 | TJNII | Are you sure you don't have to do something to speak? |
| 02:34.08 | fujin | JunK-Y: I went up there as soon as I saw it.. and well, he didn't appear to have done anything |
| 02:34.10 | jblack | No, I can't say I'm sure of that, but I wouldn't think so. |
| 02:34.18 | fujin | JunK-Y: perhaps he did a transfer, I don't know |
| 02:34.26 | fujin | I can't read that gdb output to understand what happened |
| 02:34.43 | jblack | Perhaps it's my insecure=very that's doing it. |
| 02:35.06 | jjshoe | JunK-Y do you know of any way to 'hook' show dial plan? |
| 02:35.28 | JunK-Y | hook? show dialplan? |
| 02:35.37 | JunK-Y | i dont understand the question |
| 02:35.40 | TJNII | jblack: If echo test works the problem is not your config. |
| 02:35.52 | TJNII | FWD's echotest, that is. |
| 02:36.57 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 02:37.36 | jblack | TJNII: LIke I said, plenty of other people talking.. |
| 02:37.42 | jblack | I've got to think that _something_ isn't right. |
| 02:38.34 | TJNII | yes, but you said echotest works. So your connection obviously works. |
| 02:38.42 | jjshoe | JunK-Y to see if a context exists? |
| 02:41.23 | JunK-Y | jjshoe: i already answered that question earlier.\ |
| 02:41.40 | JunK-Y | fujin: in gdb: p p->owner |
| 02:42.03 | jjshoe | JunK-Y you answered if there was some way to get the output of show dialplan in asterisk, via exec for example? |
| 02:42.29 | JunK-Y | sure: asterisk -rx'show dialplan' |
| 02:42.39 | JunK-Y | or via the AMI. |
| 02:44.12 | fujin | JunK-Y: $1 = (struct ast_channel *) 0x827cf10 |
| 02:44.40 | fujin | it says that the line of chan_sip.c is 14062 if (bridgepvt->t38.state == T38_ENABLED) { |
| 02:44.46 | fujin | can I dsiable t38 globally or something? |
| 02:44.52 | fujin | I'm not doing any faxing |
| 02:45.49 | jjshoe | heh back to voip-info for some real help :P |
| 02:46.05 | fujin | why would you need to see if a context exists? |
| 02:46.13 | jblack | Lessee.. * has tcp 5038, ucp 2727 open. I certainly don't want to open 5038. 2727 is mgcp. Hmmm. it looks like I should have that one open. |
| 02:48.44 | mosty | can asterisk take an iax g729 call and convert it to a sip g729 call without transcoding? |
| 02:50.27 | *** join/#asterisk cpatry (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 02:52.41 | thedonvaughn | hey. I have an asterisk server, dual span T1. Zap/g1 (span 1 group) and Zap/g2 (span 2 group). I want to be able to dial out of Zap/g1, and if no Zap channels available then dial Zap/g2 and so forth. I do not want to trunk them, as I like to have control of each span. I am currently switching depending on DIALSTATUS CONGESTION and CHANUNAVAIL. It has been working, but feels like a cheap way (maybe not? :). Is there a prefered way of doing |
| 02:54.40 | jblack | ohhhh. |
| 02:54.59 | fujin | thedonvaughn: two dial statements |
| 02:55.02 | jblack | My rtp.conf was missing. The conf file says the defaults are rtpfstart=5000, and rtpend=31000. |
| 02:55.15 | jblack | Since I opened 10k-20k, some calls would work, and some wouldn't |
| 02:55.53 | thedonvaughn | fujin: just a exten = blah,n,Dial(Zap/g1/blah) then nextline exten = blah,n,Dial(Zap/g2/blah) ? |
| 02:57.54 | thedonvaughn | fujin: the second dial statement will only be dialed if the first one can't? |
| 03:00.20 | *** join/#asterisk jblack (n=jblack@pool-71-181-145-13.sctnpa.east.verizon.net) |
| 03:03.22 | fujin | thedonvaughn: correct |
| 03:03.29 | jblack | drats. everyone left coffeehouse |
| 03:03.33 | fujin | the macro/context will end on Hangup |
| 03:04.04 | thedonvaughn | fujin: simple enough thanks :) |
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| 03:25.57 | [TK]D-Fender | jblack, was away sorry. glad you found it. No need to leave such a hugh RTP hole up there if you don't need. |
| 03:26.38 | jblack | Nope, that wasn't it. |
| 03:27.00 | jblack | I thought so, but the fix didn't fix anything. It's rather flummoxing. |
| 03:28.18 | jblack | I think I'm on the right track though. The firewall also does masq for internal machines. Perhaps masq is getting confused and redirecting packets (I know it's not dropping) |
| 03:30.36 | jblack | Hmm. Looks like I have a dialplan flaw, from what I just saw. |
| 03:30.53 | jblack | yup |
| 03:31.25 | jblack | you should be able to get in now |
| 03:32.57 | jblack | guest IAX was trying to drop you into a non-existant incoming context. |
| 03:34.05 | jblack | Woot! "Deposit PAYPAL" |
| 03:37.46 | *** join/#asterisk hades123 (n=wqwsqww@d57-199-17.home.cgocable.net) |
| 03:40.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 03:41.14 | esaym | anybody run the asterisk package in debian etch? It won't let me run /etc/init.d/asterisk start as root, what the heck? |
| 03:41.52 | esaym | so how do I run it under the user asterisk if the user doesn't have a password? |
| 03:42.21 | mosty | esaym, /etc/init.d/asterisk start works fine for me |
| 03:42.32 | mosty | did you edit /etc/default/asterisk first? |
| 03:45.59 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:45.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:47.33 | russellb | [TK]D-Fender: what'd you do? |
| 03:47.53 | jblack | Made me sell out my soul.. or at least my privacy. |
| 03:48.25 | hades123 | Da Devil |
| 03:52.21 | jblack | I like kinder eggs. |
| 03:52.40 | hades123 | sorry jblack, I am keeping it all |
| 03:52.49 | jblack | 'tis ok |
| 03:53.17 | hades123 | well then, stop looking at it |
| 03:53.51 | jblack | pardwon moi! |
| 03:53.56 | hades123 | :) |
| 03:56.01 | hades123 | The toy was Shrek |
| 03:56.04 | hades123 | YAY |
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| 04:15.25 | *** part/#asterisk admin0 (n=admin@116.90.228.34) |
| 04:15.50 | *** join/#asterisk CarlFK (n=carl@76.29.25.210) |
| 04:18.53 | phix | hey, if I call a macro which runs Dial, after the macro call will $DIALSTATUS be set? |
| 04:19.08 | phix | or is it scoped only in the context where the Dial is run? |
| 04:19.15 | lmadsen | it's a channel variable, so it'll be available, yes |
| 04:21.21 | phix | yay |
| 04:21.42 | phix | cause I am attempting to write a fallback dial macro |
| 04:21.54 | phix | that calls a dial macro |
| 04:22.02 | riddlebox | what would lspci show if you had a tdm card in the pci slot? |
| 04:22.40 | phix | riddlebox: words |
| 04:23.11 | phix | 00:0c.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 04:23.22 | riddlebox | phix, hrmm mine is showing that |
| 04:23.23 | phix | that is what mine says anyway |
| 04:23.49 | riddlebox | I also have configs that I took from one pc and moved them over to a new one and now ztcfg -vv shows 0 |
| 04:23.58 | phix | ph |
| 04:23.59 | phix | oh |
| 04:24.11 | phix | lmadsen: you have a fallback dial example ?: ) |
| 04:24.30 | lmadsen | nada |
| 04:24.40 | phix | :( |
| 04:25.38 | lmadsen | Dial(SIP/provider/5195915119) |
| 04:25.55 | lmadsen | Dial(SIP/provider2/5195915119) |
| 04:26.07 | lmadsen | if Dial completes, it won't continue on in the dialplans |
| 04:26.57 | lmadsen | just perform a GotoIf() or MacroIf() based on the ${DIALSTATUS} variabl |
| 04:27.00 | lmadsen | seems pretty straight forward |
| 04:27.27 | [TK]D-Fender | typically don't even need to care about the GotoIf, and can fly right on into dialing out #2 |
| 04:27.42 | lmadsen | yep... as my first example showed :) |
| 04:28.39 | drmessano | [TK]D-Fender: I have a much better nightmare user for you |
| 04:28.57 | [TK]D-Fender | drmessano, I have a sharper sword... gimme a sec |
| 04:29.09 | jblack | Is there a way to get D() to dial dtmf _after_ the call has been bridged? |
| 04:29.17 | lmadsen | change the source? :) |
| 04:29.23 | drmessano | [TK]D-Fender: Pentium 1, 260MB HD, 32MB Ram, Windows 98, browser takes 5 minutes to change pages in FreePBX |
| 04:29.27 | jblack | heh. That's always an answer. :) |
| 04:29.31 | drmessano | I win |
| 04:30.07 | lmadsen | jblack: umm.... maybe try using G() and then using SendDTMF() -- I can't remember if you can execute dialplan logic after -- I think you can execute like one line or something... |
| 04:30.13 | hades123 | I still have a 486 lying around |
| 04:30.18 | drmessano | Hes been calling me sir, but I told him to call me by my real name: Donatello Raphael |
| 04:30.23 | drmessano | Gotta love Ninja Turtles |
| 04:30.37 | lmadsen | my grandma still uses her 286/16 |
| 04:30.58 | UnixDog | lll |
| 04:30.59 | hades123 | was my first PC, good ol days |
| 04:31.14 | phix | lmadsen: well I am having issues |
| 04:31.21 | drmessano | I have a 286/12 |
| 04:31.22 | lmadsen | sorry to hear that |
| 04:31.25 | drmessano | A DTK |
| 04:31.36 | hades123 | I remeber I was stunned at prince or persia when it first came out |
| 04:31.45 | jblack | Hmm |
| 04:31.49 | phix | lmadsen: actually I have another problem, one of my VoIP providers expects numbers in a certain format (COUNTRYCODE STATECODE NUMBER) |
| 04:31.49 | drmessano | Castles |
| 04:32.03 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 04:32.03 | lmadsen | phix: that's not really a problem... |
| 04:32.17 | lmadsen | can be easily solved with a pattern match |
| 04:32.17 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-198-40.dsl.irvnca.pacbell.net) |
| 04:32.48 | jblack | .. "You cannot use any additional action post-answer options in conjunction with this option". |
| 04:32.50 | phix | lmadsen: I have regex in my first dialplan, which calls my dialoutWithFallback macro, then I run the dialoutFunnyVoip macro which has regex in it again to change the numbers in expected format? |
| 04:32.57 | jblack | I understand what each of those words means, yet I can't quite decipher it |
| 04:33.27 | phix | lmadsen: when I run the macro s,1, is run first right? what do I put in there to make it match regex? |
| 04:33.33 | phix | goto(${ARG1}) ? |
| 04:34.21 | lmadsen | phix: sounds like you're making it too complicated |
| 04:34.26 | lmadsen | anyways, good luck, I'm out |
| 04:34.47 | phix | I dont want complication! how do I simply do it! |
| 04:34.49 | phix | crap he left |
| 04:35.21 | phix | who wants to take lmadsen's place? :) |
| 04:37.19 | fujin | phix: what are you trying to do |
| 04:37.21 | fujin | spell it out |
| 04:37.25 | fujin | and pastebin your current configs |
| 04:37.26 | fujin | ~pb |
| 04:37.27 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
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| 04:38.25 | jblack | TODO: Hook my scanner up to an extension |
| 04:38.38 | fujin | lol |
| 04:38.41 | riddlebox | hrmm, I wonder why I cannot get this to work, lspci sees the card, the fxo port lights up on it, I am using configs that I pulled from another system, basically my asterisk system moved from one machine to another, but ztcfg -vv shows 0 channels |
| 04:38.43 | fujin | I'm heading home |
| 04:38.47 | fujin | phix: if you're around in 45 mins |
| 04:42.27 | skopii | hello what is the difference between dial(sip/trunk/ext) and dial(sip/ext@trunk) |
| 04:45.03 | [TK]D-Fender | riddlebox, go check zaptel.conf |
| 04:45.43 | [TK]D-Fender | phix, you don't "goto" out of a macro. Thats the first rule of structured programming |
| 04:46.13 | JunK-Y | and u dont use goto at all its the 2nd rule :) |
| 04:46.14 | [TK]D-Fender | phix, and like I said, if you want a fallback, just dial out the second place IMMEDIATELY following the first. |
| 04:46.36 | [TK]D-Fender | JunK-Y, And the 3rd rule of Fight Club is... you don't talk about Fight Club! |
| 04:46.50 | *** part/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 04:47.22 | JunK-Y | zip it :) |
| 04:47.38 | jwh | so doing callerid manipulation, and then a Goto() is bad? |
| 04:48.39 | riddlebox | [TK]D-Fender, zaptel.conf is fine, I pulled the card and the config files from one machine(one that worked) and put them in another and now I cannot get the ztcfg to work |
| 04:48.43 | kyron | pbzip it |
| 04:48.53 | *** join/#asterisk x86- (n=x86@ool-18b88770.dyn.optonline.net) |
| 04:48.54 | [TK]D-Fender | riddlebox, PASTEBIN everything. |
| 04:49.40 | x86- | How can I remove modules directory contents |
| 04:50.17 | [TK]D-Fender | x86 : "man rm" <- |
| 04:50.23 | kyron | The only place I've seen GOTOs warranted is in kernel driver code |
| 04:50.30 | riddlebox | [TK]D-Fender, http://pastebin.ca/856941 |
| 04:50.55 | [TK]D-Fender | riddlebox, .... and the rest? |
| 04:50.58 | x86- | ahh.. so I have to just remove all the modules |
| 04:51.03 | skopii | bind uses goto |
| 04:51.13 | [TK]D-Fender | x86-, I never said that. |
| 04:51.22 | riddlebox | [TK]D-Fender, what else do you want, zapata.conf as well? |
| 04:51.25 | x86- | the modules there, aren't compild against the same version of asteirsk |
| 04:51.30 | [TK]D-Fender | riddlebox, Everything. |
| 04:52.10 | x86- | so i'm just trying to fix that problem.. I was told to remove module directory contents .. so if I just rm * asterisk and recompile it everything should work correctly? |
| 04:52.13 | [TK]D-Fender | x86-, if you say so... |
| 04:52.26 | riddlebox | [TK]D-Fender, http://pastebin.ca/856942 thats both of them |
| 04:52.27 | JunK-Y | x86-: yes |
| 04:53.05 | [TK]D-Fender | riddlebox, Show me how your card registered on boot, show me that the module is even loaded, etc. |
| 04:53.27 | JunK-Y | rm -rf /usr/lib/asterisk/modules && make install will reput all ur .so in that dir. |
| 04:53.33 | x86- | my question is this though...why would I have to remove module contents? wouldn't they get removed when I delete asterisk and then reinstall (compile) |
| 04:53.35 | JunK-Y | with ur specific * ver |
| 04:54.07 | JunK-Y | x86-: nope |
| 04:54.08 | [TK]D-Fender | x86-, Any matching one would get overwritten, thats all |
| 04:54.20 | drmessano | [TK]D-Fender: Tell me I am not crazy: |
| 04:54.21 | drmessano | chan_sip.c: Registration from 'desktop <sip:1005@192.168.1.5>' failed for '192.168.1.2' - ACL error (permit/deny) |
| 04:54.31 | drmessano | First impression? |
| 04:54.31 | [TK]D-Fender | x86-, Anything extra would stay behind and cause possible issues on "autoload=yes" |
| 04:54.33 | JunK-Y | so when you boot ur * you have problem with ur modules. |
| 04:54.57 | [TK]D-Fender | drmessano, how about the glaring "ACL error (permit/deny)" <- ? |
| 04:55.11 | drmessano | So im not crazy |
| 04:55.13 | drmessano | TY |
| 04:55.21 | drmessano | This guy is KILLING ME |
| 04:55.37 | [TK]D-Fender | drmessano, I didn't say that. You just asked what my first impression of that statement was. |
| 04:55.50 | riddlebox | [TK]D-Fender, http://pastebin.ca/856943 |
| 04:55.52 | [TK]D-Fender | drmessano, You are still crazy regardless of your being right or not :p |
| 04:55.56 | drmessano | ROFL |
| 04:56.07 | JunK-Y | drmessano: yeah [TK]D-Fender is [TK]D-Fender ! |
| 04:56.23 | JunK-Y | but you know what? i love you [TK]D-Fender ! |
| 04:56.28 | x86- | I just rm -rf /usr/lib/asterisk/modules && make install this is what happend http://pastebin.com/m3f6d1902 |
| 04:57.20 | JunK-Y | x86-: be in /usr/src/asterisk/ |
| 04:57.30 | JunK-Y | or whatever the source of ur asterisk source are. |
| 04:57.48 | [TK]D-Fender | riddlebox, "cat /proc/interrupts" , "ztcfg -vvvv" and before doing the latter, please remove all the commented out junk out of both of those files permanently. |
| 04:58.08 | JunK-Y | and whats app_flite.c ? |
| 04:58.10 | [TK]D-Fender | outtolunc, You know my friends too?!?! |
| 04:58.43 | outtolunc | any of them on american idol? and like to say 'vicorious'? |
| 04:58.48 | outtolunc | er +t |
| 04:58.53 | x86- | alright i'm in usr/src/asterisk-1.4 |
| 04:59.19 | JunK-Y | now do make install |
| 04:59.45 | JunK-Y | at the end ls /usr/lib/asterisk/modules/ should be filled correctly again. |
| 04:59.46 | x86- | thnx |
| 05:00.05 | *** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.ca.comcast.net) |
| 05:00.46 | JunK-Y | so everything fixed x86- ? |
| 05:01.23 | [TK]D-Fender | outtolunc, some of them worship strange idols and have delusions of adequacy.... does that count? |
| 05:01.34 | kyron | gee, building * is like building a kernel ;) |
| 05:01.44 | kyron | apparantly, even has a menuconfig! |
| 05:01.46 | kyron | :P |
| 05:01.53 | kyron | how easy can it get? |
| 05:01.58 | [TK]D-Fender | kyron, Turn up the heat, wait for it to pop, then smother it in butter :p |
| 05:03.11 | kyron | I love commenting on stuff I haven't done yet....like building my own *...but it will come...and I will pop up here more noisily ;) |
| 05:03.14 | kyron | mwehehe |
| 05:03.52 | kyron | the build porcess is nice but also a PITA for integrating into Gentoo... sigh... |
| 05:03.53 | outtolunc | [TK]D-Fender: close enough |
| 05:03.58 | [TK]D-Fender | kyron, in a Jiffy I'm sure.... |
| 05:04.51 | kyron | [TK]D-Fender, large jiffy...I have to type up my masters before I take up anything serious like that :P |
| 05:05.24 | riddlebox | [TK]D-Fender, http://pastebin.ca/856950 |
| 05:06.07 | [TK]D-Fender | riddlebox, that is messed up |
| 05:06.21 | riddlebox | what do you mean? |
| 05:06.34 | [TK]D-Fender | riddlebox, you say "zaptel.conf", it has no channels, and looks like the rest of zapata.conf in there, much of which is wrong |
| 05:07.05 | x86- | app_flite.c:260: warning: function declaration isn't a prototype |
| 05:07.07 | x86- | make: *** [app_flite.o] Error 1 |
| 05:07.43 | riddlebox | [TK]D-Fender, zaptel.conf has channel =>1 |
| 05:08.03 | [TK]D-Fender | riddlebox, look at your last pastebin and tell me what I should be seeing in there. |
| 05:08.22 | [TK]D-Fender | riddlebox, "channel => 1" is ZAPATA.CONF syntax! |
| 05:08.42 | [TK]D-Fender | riddlebox, Put. Down. The. Crack. Pipe. (c) JerJer |
| 05:10.20 | jblack | Anyone have time to take and receive a sip call? |
| 05:10.21 | phix | fujin: I am still here :) |
| 05:10.27 | [TK]D-Fender | jblack, sure |
| 05:10.30 | jblack | Cool. |
| 05:10.48 | jblack | awesome |
| 05:11.57 | phix | [TK]D-Fender: If I dialout immediately following the first then it tries the next channel even if the person being called cancels the call instead of picking up |
| 05:12.45 | riddlebox | [TK]D-Fender, I have fxsks=1, loadzone=us,defaultzone=us , thats what the book says to use? |
| 05:13.13 | phix | [TK]D-Fender: I only want to fallback if the channel is unavailable, that is the SIP provider cannot register or DNS / Internet is down |
| 05:14.09 | hmmhesays | jblack: $1 a minute |
| 05:14.31 | hmmhesays | phish tha tis pretty simple |
| 05:15.12 | phix | hmmhesays: it is simple? |
| 05:15.42 | phix | goto(s-${DIALSTATUS},1) ? |
| 05:15.58 | phix | or somethign better than that? |
| 05:16.35 | *** join/#asterisk catch23 (n=catch23@69.60.124.109) |
| 05:16.52 | phix | catch23--; |
| 05:16.52 | fujin | phix: now what was the problem? did you end up pastebinning it? |
| 05:17.01 | catch23 | hi, anyone here know if callweaver supports fastagi (fastogi in their world) |
| 05:17.08 | phix | fujin: not a problem, I am just trying to figure out how to do this |
| 05:17.18 | phix | fujin: fallback that is on CHANUNAVAIL |
| 05:17.31 | phix | s/that is// |
| 05:17.31 | hmmhesays | that simple |
| 05:17.42 | phix | thnx jbot :) |
| 05:17.43 | fujin | uh, what's so hard about that? |
| 05:17.58 | phix | I dont know how to do it |
| 05:18.00 | fujin | goto (s-${DIALSTATUS},1) and be done with it |
| 05:18.21 | phix | fujin: ummm but I need to fallback again if tha doesnt work |
| 05:18.34 | phix | I have 3 different channels |
| 05:18.41 | phix | PSTN, VOIP1, VOIP2 |
| 05:18.45 | fujin | I prefer to use switch(${DIALSTATUS}) { case blah: {} default: {} } |
| 05:18.51 | fujin | phix: oh, you don't need dialstatus then - |
| 05:18.59 | fujin | just use dial(); dial(); dial(); |
| 05:18.59 | phix | what do I need then ? |
| 05:19.18 | phix | fujin: but I only want it to dial on chanunavail |
| 05:19.30 | fujin | it'll only fall through on chanunavail |
| 05:19.38 | riddlebox | [TK]D-Fender, found the problem, I had the right file, in the wrong place, I had my good copy zaptel.conf in /etc/asterisk |
| 05:19.43 | fujin | afaik, anyway |
| 05:19.45 | phix | fujin: ? |
| 05:19.54 | x86- | I just recompiled asterisks .. still getting the same error trying to get the flite module to work for 1.4.x anyone get it working? |
| 05:20.11 | JunK-Y | x86-: what is app_flite exactly? |
| 05:20.15 | phix | fujin: no it will fall through if the end party hangs up instead of picking up |
| 05:20.20 | phix | I dont want that |
| 05:20.33 | fujin | what end party? |
| 05:20.36 | fujin | the dialee? |
| 05:20.39 | x86- | It reads text |
| 05:20.53 | x86- | it's a module to read text to voice |
| 05:21.06 | fujin | phix: then explain to me why http://rafb.net/p/JMBkQ699.html works |
| 05:22.14 | phix | fujin: the person you are dialing |
| 05:22.15 | JunK-Y | x86-: ha the stuff from nerd viffle or something like that? |
| 05:22.27 | fujin | (note: there really needs to be a better way to group together outgoing sip peers) |
| 05:22.29 | phix | fujin: if they cancel the call instead of picking up then it will try the next dial |
| 05:22.45 | fujin | phix: yeah, so? |
| 05:22.48 | fujin | so they cancel it twice |
| 05:22.52 | fujin | or they accept it th esecond time |
| 05:23.22 | x86- | JunK-Y yup excatly man that stuff is sweet over there.. |
| 05:23.37 | fujin | phix: here, take a look at this http://rafb.net/p/SV6pEG10.html - sounds like it does what you need. I use that for internal dialling. |
| 05:23.40 | x86- | Junk.. I figured it was instability due to the module being compiled under a diff version of asterisk.. |
| 05:23.41 | phix | fujin: but it will be using different channels, I dont want that, each channel has its own pricing |
| 05:23.42 | fujin | although you could obviously adapt it |
| 05:23.52 | phix | fujin: PSTN is more expensive than VOIP1 |
| 05:23.53 | x86- | I rm'd and recompiled asterisks like you said .. still no luck. |
| 05:24.16 | fujin | phix: ugh, use your brain |
| 05:24.23 | fujin | I just gave you a perfect example for modification |
| 05:24.27 | x86- | <PROTECTED> |
| 05:24.53 | x86- | any idea? all the links are right .. I did ldconfig .. |
| 05:25.19 | JunK-Y | x86-: its probably a module for 1.2 |
| 05:28.13 | drmessano | ACL = Asterisk Can't Lie |
| 05:28.42 | drmessano | ACL = Ain't Configged, Loser |
| 05:28.46 | x86- | ahhhh... |
| 05:29.42 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 05:31.08 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 05:41.18 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 05:50.19 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 06:00.57 | phix | fujin: grrrrr |
| 06:05.00 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:06.56 | tengulre | Loading... ... |
| 06:08.13 | vn | anyone has luck with * and TDDs (telephone device for deaf)? |
| 06:22.54 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 06:23.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:35.25 | jblack | Dear fwd: You're like a girlfriend after a year. You were so promising, but you're really teh sux0rs |
| 06:35.36 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 06:36.19 | [TK]D-Fender | jblack, more like Monica Lewinski...... sure it blows now, but it'll screw you later ;) |
| 06:38.17 | [TK]D-Fender | OK, CHECKOUT TIME. |
| 06:38.19 | [TK]D-Fender | LATER ALL |
| 06:38.25 | [TK]D-Fender | </caps> |
| 06:39.48 | jblack | LOL@@ |
| 06:43.18 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:44.18 | jblack | LOL |
| 06:46.04 | fujin | phix: still can't manage a little bit of extensions code? |
| 06:46.10 | fujin | you might aswell give up while you're ahead tbh |
| 06:46.13 | fujin | you fail, at life |
| 06:49.18 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 06:52.31 | drmessano | FWD is less reliable than telegraph |
| 06:52.46 | jblack | hmmm. |
| 06:52.58 | jblack | I can't argue with you right now. |
| 06:53.16 | drmessano | fujin: I have a bigger fail |
| 06:54.07 | drmessano | 4 days trying to get a PAP2 (easy, wut?) set up with a FreePBX ext |
| 06:54.11 | drmessano | I mean |
| 06:54.17 | drmessano | Thats like.. |
| 06:54.25 | drmessano | X-lite is "harder" than a pap2 |
| 06:54.28 | fujin | gtfo, take it to #freepbx |
| 06:54.36 | fujin | you fail at everything |
| 06:54.40 | fujin | pap2 is like |
| 06:54.42 | fujin | point click done |
| 06:54.57 | drmessano | fujin, quit being such a Nazi |
| 06:54.57 | fujin | even easier than configuraing a spa9x2 |
| 06:55.13 | fujin | 4days is pretty epic though |
| 06:55.28 | fujin | even on my first day of asterisk I managed to reinstall it and get the office functional again (sure, only 4 extensions, but hell) |
| 06:55.32 | fujin | fail is fail |
| 06:56.14 | drmessano | Yeah, wasnt my box.. you fail at trolling me.. again |
| 06:56.16 | jblack | I thought the spa8k was surprisingly difficult to configure. Not because the required amount of work to get them working is much (it really is trivial), but because it's loaded with so many options, finding the 3 things that need to be setup are flooded out. |
| 06:56.36 | fujin | jblack: same with the spa9x2's though.. so many options |
| 06:56.37 | jblack | That, and the various ui bugs that infest the thing. |
| 06:56.45 | fujin | and you have to go admin login -> advanced to get anywhere |
| 06:57.15 | fujin | drmessano: why not nuke it and install plain old *? |
| 06:57.23 | drmessano | Not my box |
| 06:57.26 | drmessano | Some dude in India |
| 06:57.44 | jblack | what is he running, |
| 06:57.52 | jblack | freepbx. |
| 06:57.54 | drmessano | TB with a PAP2 |
| 06:57.55 | drmessano | I mean |
| 06:57.59 | drmessano | HOW HARD IS THAT |
| 06:58.14 | jblack | No idea. I have neither one nor the other. =) |
| 06:58.14 | drmessano | Thats training wheels with no bike |
| 06:58.32 | drmessano | Oh god |
| 06:59.48 | jblack | In all fairness, * itself isn't very difficult.... one you spend 3 weeks getting a solid grasp of the terminology, methodology and paradigms |
| 07:01.46 | drmessano | Im getting web access to this box now |
| 07:01.53 | drmessano | This should be absolutely fascinating |
| 07:02.03 | drmessano | "Wate, wut?" |
| 07:03.41 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 07:04.01 | slavon_net | where 1.4 brbranch? in svn i see only 1.2 =( or its closed? |
| 07:04.30 | jblack | lol. "iMetal heaphones" |
| 07:05.06 | drmessano | I need a bigger keyboard wrist rest |
| 07:05.13 | jblack | I can't wait for this "let's put an i on the front" fad to be over. |
| 07:05.21 | drmessano | My forehead keeps hitting the desk |
| 07:05.56 | jblack | Heh. Go get some sleep. :) |
| 07:06.04 | drmessano | iAsteriskd 2.0 |
| 07:06.35 | drmessano | I cant wait for the 2.0 names with ER or ED at the end missing the E |
| 07:06.39 | drmessano | FLIPPR |
| 07:06.44 | drmessano | FUCKR |
| 07:07.06 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.102.57) |
| 07:07.47 | *** join/#asterisk jblack (n=jblack@pool-71-181-145-13.sctnpa.east.verizon.net) |
| 07:10.43 | *** join/#asterisk magumbade (n=magumbad@p5497D570.dip.t-dialin.net) |
| 07:16.20 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 07:16.35 | drmessano | Im close to shoving an Asterisk install guide in front of this guy and telling him to call me in a week |
| 07:22.36 | jblack | Give him the contents of ~book |
| 07:22.53 | UnixDog | ~ book |
| 07:22.54 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 07:23.02 | drmessano | I almost stroked out on him.. his extension had a diff password than what he gave me |
| 07:23.09 | drmessano | But apparently he changed it in the device too |
| 07:23.31 | UnixDog | night |
| 07:23.35 | drmessano | Night |
| 07:27.48 | drmessano | jblack |
| 07:28.03 | drmessano | Seriously ... need.... gangsta rap |
| 07:28.22 | jblack | Try afroman |
| 07:29.00 | jblack | It's Cali rap rather than gansta rap. It'll help you calm down. Especially if you have a blunt handy |
| 07:30.01 | jblack | hmm. |
| 07:30.11 | jblack | That wakeup script you gave me only works on the minute |
| 07:30.55 | xachen | afroman :) |
| 07:31.54 | *** join/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 07:32.15 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 07:32.17 | jochien1 | !libpri |
| 07:33.07 | jochien1 | hi -- can some 1 tell me why libpri is important for 1.4 i m trying to install it on etch |
| 07:33.41 | jblack | jochien1: I'm sure it's used with zapata (a type of hardware) devices. |
| 07:33.41 | jochien1 | <PROTECTED> |
| 07:34.35 | jochien1 | jblack: and how is it different from zaptel-1.4-current |
| 07:36.08 | *** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it) |
| 07:36.39 | jblack | I lied. It's for isdn |
| 07:36.54 | jblack | (apt-cache show libpri1) |
| 07:37.06 | jochien1 | i have got important links for these here if any1 wanted to know http://www.asterisk.org/downloads |
| 07:37.42 | jochien1 | jblack: thanks tho -- i had to do more research since u had said u belived... ;) thanks man |
| 07:38.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:39.08 | *** join/#asterisk adjohn (n=adjohn@p1089-ipad76marunouchi.tokyo.ocn.ne.jp) |
| 07:40.18 | adjohn | hello, I have a basic install of asterisk on my server. If I use a client like x-lite to connect to it via localhost, it works fine. If I put the IP address instead of localhost it won't connect. This is without a firewall running, and everything is on the same machine. Any ideas? |
| 07:41.43 | jochien1 | adjohn: where do u out the IP address? |
| 07:42.19 | adjohn | I am putting the IP address in the "domain" field of the x-lite client setup |
| 07:42.57 | adjohn | I am trying to troubleshoot a different problem, of no one externally being able to connect to my server as well. |
| 07:43.16 | drmessano | Sorry, was in another channel |
| 07:43.21 | drmessano | I got that guy in india up |
| 07:43.51 | *** join/#asterisk ionix (n=ionix@p2240-ipbfp02motosinmat.mie.ocn.ne.jp) |
| 07:44.39 | drmessano | He didnt APPLY |
| 07:44.41 | drmessano | No |
| 07:44.47 | drmessano | Seriously |
| 07:44.50 | jochien1 | adjohn: reserve the ip addr for the sip server |
| 07:44.53 | drmessano | 4 days |
| 07:44.55 | drmessano | no apply |
| 07:45.03 | adjohn | jochien1, how do I do that? |
| 07:45.35 | jochien1 | adjohn: i mean put the ip of ur server in the sip server filed on xlite |
| 07:46.16 | jochien1 | adjohn: and domain field you can put ut TLD |
| 07:46.25 | jochien1 | adjohn: ur* |
| 07:46.54 | adjohn | in sip.conf? |
| 07:47.08 | jochien1 | adjohn: no on xlite |
| 07:47.28 | adjohn | on the sip accounts, I only see domain for server address |
| 07:47.54 | jochien1 | adjohn: hold on |
| 07:47.59 | adjohn | okay thanks |
| 07:50.15 | jblack | Thanks drmessano |
| 07:51.32 | adjohn | jochien1, I was able to get it working on another server.. Sort of, here is the log from asterisk: http://pastebin.ca/857021 |
| 07:52.05 | adjohn | joshien1, but on the client, it appeared to not ever connect and eventually timed out. |
| 07:53.37 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582302.dsl.bell.ca) |
| 07:54.13 | jochien1 | adjohn: this can help http://www.google.co.ug/search?hl=en&q=xlite+user+guide&btnG=Google+Search and http://www.scribd.com/doc/14835/XLite-and-Asterisk |
| 07:54.27 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:54.32 | adjohn | jochien1, thanks |
| 07:57.37 | jochien1 | any body tried 1.4.17 |
| 07:57.40 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 07:57.53 | mosty | jochien1, of course |
| 07:58.13 | jochien1 | mosty: no bugs? |
| 07:58.29 | jochien1 | mosty: any trouble i mean? |
| 07:59.26 | mosty | there are a few, no showstoppers for the most common setups/scales though |
| 07:59.30 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 07:59.32 | mosty | that i know of, at least |
| 08:01.19 | jochien1 | mosty: i wldnt want to use something with trouble now init? anyways what about 1.4-current |
| 08:01.55 | mosty | use 1.2 if you want something a bit less buggy, but it has a lot fewer features |
| 08:02.10 | jochien1 | mosty: i have been using 1.2 but now i want to switch to the latest version fro better features |
| 08:02.29 | mosty | try it out on a test box |
| 08:02.44 | jochien1 | mosty: ok thanks man |
| 08:03.08 | jochien1 | mosty: if i have trouble whr do u think i shld first run too ;) |
| 08:05.39 | mosty | run to voip-info.org |
| 08:05.56 | jochien1 | <PROTECTED> |
| 08:06.02 | mosty | and read the 1.4 upgrade docs in the asterisk source/on the wiki |
| 08:06.06 | jochien1 | Oooops |
| 08:06.17 | jochien1 | mosty: ok |
| 08:08.26 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:09.27 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 08:14.22 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 08:22.35 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:23.05 | jochien1 | mosty: considering 1.4.17 it is less buggy u said? |
| 08:23.25 | mosty | i have no idea what you're trying to ask |
| 08:24.59 | jochien1 | mosty: 1.4.17 is not so problematic? |
| 08:25.37 | jochien1 | mosty: i just read tht it fixes the bugs from 1.4 but does it have its own bugs |
| 08:25.42 | *** join/#asterisk ronr (n=ron@82-170-109-196-static.dsl.ip.tiscali.nl) |
| 08:26.56 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 08:27.25 | mosty | it's not bug-free |
| 08:27.32 | mosty | no large piece of software ever is |
| 08:32.22 | jochien1 | mosty: thts true |
| 08:32.52 | jochien1 | mosty: i hope it has good documentation for these bugs or maybe 1.4 has better documentation? |
| 08:33.02 | mosty | ~thebook |
| 08:33.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 08:33.48 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 08:34.26 | jochien1 | mosty: thanks man |
| 08:37.13 | J4zen | Does anyone know of an Xen Asterisk image? Did anyone make one yet? |
| 08:37.43 | nixguy | there is one |
| 08:37.50 | nixguy | on digiums site |
| 08:37.53 | J4zen | is it available? |
| 08:37.55 | J4zen | oh really? |
| 08:38.24 | nixguy | yup |
| 08:38.40 | nixguy | or at least there was 2 months ago when i downloaded it and tryed it out .. |
| 08:38.55 | J4zen | what were your expierences? |
| 08:39.11 | nixguy | well |
| 08:39.17 | nixguy | im quite new to the asterisk world |
| 08:39.27 | J4zen | ah |
| 08:39.30 | nixguy | so i went back and installed a fresh asterisk from packages instead |
| 08:39.39 | J4zen | ok, too bad :) |
| 08:39.42 | nixguy | to get a feeling for things instead of trying out something prebuilt |
| 08:39.42 | J4zen | thanks for the info though |
| 08:39.43 | nixguy | http://asterisknow.org/downloads |
| 08:39.46 | nixguy | is where you get int |
| 08:39.47 | nixguy | it |
| 08:40.27 | drmessano | YAY... Coin toss for Asterisk, 8 lines |
| 08:40.43 | J4zen | thanks nixguy |
| 08:41.08 | nixguy | youre welcome! |
| 08:43.26 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 08:43.55 | nixguy | J4zen: my initial thought thouhg is |
| 08:44.12 | nixguy | that if you wanna use your xen asterisk with some real hardware (like a pri card) |
| 08:44.20 | nixguy | you will need a loot of fiddling |
| 08:45.12 | J4zen | yeah |
| 08:45.16 | J4zen | but im not going to |
| 08:45.30 | J4zen | i'll be using a SIP-trunk in stead of PRI interfaces |
| 08:45.40 | nixguy | ok |
| 08:45.51 | J4zen | i tried going down the BRI/PRI road, but it would be far too much work for this particulair situation |
| 08:45.59 | nixguy | J4zen: hehe yes |
| 08:46.02 | nixguy | mygod |
| 08:46.11 | nixguy | so poorly documented how to install PRI cards |
| 08:46.19 | J4zen | not so much configering, but maintenance and new implementations would be an arse when i leave the company next year lol |
| 08:46.23 | J4zen | yeah its tricky |
| 08:46.37 | J4zen | you'll get there though ;) just mess around and watch it light up :D |
| 08:46.40 | nixguy | i went the pri way , a little bit afraid i would have timing issues with sip... |
| 08:47.02 | J4zen | < Quadbri |
| 08:47.14 | nixguy | the pri part is up and running but im still a noob so reading about dialplan planning right now |
| 08:47.28 | J4zen | hehe yeah i hear ya |
| 08:47.40 | J4zen | only have about 4 months expierence myself |
| 08:47.54 | J4zen | came in as a windows user lol |
| 08:48.05 | J4zen | although this probably wouldnt be the best channel to mention this on |
| 08:48.05 | nixguy | hehe ok |
| 08:49.53 | J4zen | lol thats odd |
| 08:50.15 | J4zen | my PBX or my SIP-trunk is crashing my phone when i call it with my mobile lol |
| 08:50.46 | EvilDeshi | can a sip user be in more then one context? |
| 08:50.47 | jochien1 | i have only been given a chance to prove my sys admin by installing asterisk n i m doing it in vmware from a cisco backgroud. its now 3 weeks |
| 08:51.19 | J4zen | jochien1: afaik VMware definatly is not recommended , nor efficient |
| 08:51.23 | jochien1 | although i hv used linux |
| 08:51.59 | jochien1 | J4zen: it has been working for me |
| 08:52.27 | J4zen | how many sip-channels active on averige? |
| 08:52.41 | *** join/#asterisk bmg505 (n=leon@196.209.176.150) |
| 08:54.18 | nixguy | coming in from the windows world thats though its qyite differente |
| 08:56.07 | J4zen | oh yes |
| 08:56.43 | J4zen | you catch on rather fast though |
| 08:56.52 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
| 08:58.12 | *** join/#asterisk admin0 (n=admin@202.161.147.14) |
| 09:02.56 | J4zen | Jailtime.org is down? |
| 09:03.05 | J4zen | wrong channel, sorry |
| 09:09.06 | xachen | J4zen: its working for me |
| 09:10.38 | jochien1 | is 1.4.17 now a stable version |
| 09:11.16 | BBHoss | dunno if its stable or not, but its the latest :) |
| 09:11.35 | BBHoss | apparently they are getting better though |
| 09:12.00 | jochien1 | i thought i give it a try |
| 09:12.03 | *** join/#asterisk Stefan1979 (n=stan@4204ds2-vby.0.fullrate.dk) |
| 09:12.08 | *** join/#asterisk Federico2 (n=fede@pdpc/supporter/base/Federico2) |
| 09:12.16 | BBHoss | go ahead, report bugs plz :) |
| 09:12.44 | drmessano | Hmmm |
| 09:13.10 | drmessano | Im thinking Rock, Paper, Scissors for Asterisk |
| 09:13.44 | jochien1 | mosty r u still with us? |
| 09:13.59 | mosty | yes |
| 09:14.50 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 09:15.36 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 09:17.34 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 09:17.41 | FuriousGeorge | hey all |
| 09:18.05 | FuriousGeorge | anyone else notice 1.4 missing from svn |
| 09:19.02 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4aaf4b9263be4ba4) |
| 09:21.39 | FlatFoot | morning all |
| 09:24.39 | jochien1 | mosty: is 1.4.17 a stable version |
| 09:24.42 | BBHoss | FuriousGeorge: yeah i noticed that. Wonder why? |
| 09:25.38 | FuriousGeorge | BBHoss: perhaps new release imminent? |
| 09:25.43 | mosty | jochien1, you don't need my permission to try it, see for yourself |
| 09:26.59 | BBHoss | FuriousGeorge: well the bugtracker has versions up to 1.4.20 if that means anything |
| 09:27.20 | jochien1 | mosty, ok |
| 09:27.47 | BBHoss | but the last release was 01/02 i wouldnt think there would be another release within two weeks |
| 09:27.49 | FuriousGeorge | BBHoss: i still think all the releases are just a way to get us to reboot often |
| 09:27.54 | BBHoss | heh yeah |
| 09:29.17 | BBHoss | from logs today : <mvanbaak> markit: asterisk svn is down at the moment |
| 09:29.41 | BBHoss | no details though, and zaptel 1.4 is still there |
| 09:29.45 | FlatFoot | what were those blokes on when they recorded the music on hold ????? |
| 09:29.58 | FlatFoot | my mate in the office wants to know cos he wants some |
| 09:30.31 | *** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk) |
| 09:31.12 | BBHoss | FlatFoot: meth |
| 09:31.25 | FlatFoot | aha i shall pass on the info |
| 09:31.52 | BBHoss | and weekend doses of PCP |
| 09:32.03 | BBHoss | it should do the trick |
| 09:32.03 | FlatFoot | :) |
| 09:32.58 | BBHoss | night all or morning :) 3:33AM |
| 09:34.28 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:37.10 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:45.53 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 09:46.20 | Dr-Linux | i'm using asterisk ver 1.4.17 but i don't have app_jabber.so? |
| 09:46.24 | Dr-Linux | how can i get it? |
| 09:54.05 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 09:54.57 | phix | ~paste |
| 09:54.57 | jbot | paste is, like, http://rafb.net/paste/, or see also pb |
| 09:55.03 | phix | pb |
| 09:55.06 | phix | ~pb |
| 09:55.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 09:56.50 | phix | ok, here is my attempt at a fallback dialing macro |
| 09:56.53 | phix | http://rafb.net/p/6iVycT49.html |
| 09:57.11 | phix | It seems to work, but I would like to feedback, I have probably done some stupid or redundant things in it :) |
| 09:59.40 | phix | another question, how do I get my softphone ringing tone the same as my PSTN ? |
| 10:07.56 | phix | I eargily await your comments |
| 10:08.05 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:10.51 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
| 10:11.19 | *** join/#asterisk tsabi (n=tsabi@gw.creditexpress.hu) |
| 10:11.46 | phix | ay |
| 10:15.15 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:22.56 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 10:27.13 | phix | http://rafb.net/p/6iVycT49.html |
| 10:33.44 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-47217ff327020a7d) |
| 10:36.08 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 10:36.26 | jmls | trying to use #exec to get some dialplan from a script |
| 10:37.02 | jmls | from the bash prompt, "/etc/asterisk/myapp/getdata.sh getqueuedialplan?dialplantype=external" |
| 10:37.07 | jmls | returns the data |
| 10:37.24 | jmls | if I use #exec with the exact same command, it fails |
| 10:37.36 | *** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it) |
| 10:37.51 | jmls | [Jan 16 10:32:15] WARNING[7096]: config.c:806 process_text_line: No '=' (equal sign) in line 1 of /var/tmp/exec.1200479535.31992720 |
| 10:37.57 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 10:38.06 | jmls | I use #exec in the globals section with no problem |
| 10:38.20 | phix | ? |
| 10:38.56 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 10:49.26 | J4zen | Does anyone know of an AGI script for Asterisk and such that will e-mail or call on a trunk-failiure? |
| 11:00.42 | FlatFoot | i am seeing voice drop from * to sip phones ( snom ) all equip is on external ( real IP's ) running 1.2.13 on debian has anyone come across this ? can anyone point me in the right direction to investigate ? |
| 11:01.18 | FlatFoot | there is no drop from phone to * which is why i am perplexed |
| 11:02.06 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 11:09.18 | defswork | what needs restarting after a change to zapata.conf ? |
| 11:13.01 | tzafrir | defswork, depends what you change |
| 11:13.14 | defswork | I set priindication=outofband |
| 11:13.15 | tzafrir | If you change signalling or span configuration: yes |
| 11:13.26 | tzafrir | yes, this probably requires restart |
| 11:13.38 | defswork | of just asterisk or drivers too ? |
| 11:13.48 | tzafrir | or maybe just: module unload chan_zap.so ; module load chan_zap.so |
| 11:18.41 | *** join/#asterisk _ys (n=_ys@91.151.196.254) |
| 11:21.37 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 11:24.31 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 11:32.58 | defswork | anyone know what CAS signalling on span 2 conflicts with HDLC with FCS check on channel 16 is about ? (sangoma twin E1) |
| 11:34.21 | tzafrir | defswork, what version of zaptel? |
| 11:34.33 | defswork | I'm not sure |
| 11:34.52 | tzafrir | That is typically a bogus error message issues by some earlier versions of ztcfg |
| 11:34.53 | defswork | system was workign fine - I just rebooted after adding that priiindication entry :o |
| 11:35.12 | Alexandre_fr | Hello guys |
| 11:35.15 | tzafrir | do you happen to have a digital (E1/T1) card that is placed after an analog one? |
| 11:35.20 | defswork | no |
| 11:35.56 | defswork | single dual port E1 sangoma |
| 11:37.40 | defswork | any ideas ? asterisk won't start - fails to load chan_zap :o |
| 11:44.00 | *** join/#asterisk zaur_pronet (n=zaur_pro@85.132.55.134) |
| 11:44.01 | *** join/#asterisk glen2 (n=glen@212.54.184.253) |
| 11:44.59 | *** join/#asterisk bmg505 (n=leon@196.209.176.150) |
| 11:46.14 | glen2 | Is there an AUG in London? |
| 11:47.30 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-32-175.lns10.syd7.internode.on.net) |
| 11:48.29 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 11:50.41 | jblack | glen2: Every year. |
| 11:50.47 | tzafrir | Ausies group? |
| 11:53.20 | glen2 | London Asterisk users group. |
| 11:53.41 | glen2 | London in Englandshire. |
| 11:55.02 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:58.30 | *** join/#asterisk zaur_pronet (n=zaur_pro@85.132.55.134) |
| 11:59.38 | *** join/#asterisk zaur_pronet (n=zaur_pro@85.132.55.134) |
| 12:00.33 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 12:01.56 | franck | one thing I don't understand, how can I have echo between two VoIP phones connected to asterisk?how do I fix that? |
| 12:02.17 | RoyK | glen2: LAUG - Norwegian word meaning 'group' or 'guild' or similar :P |
| 12:05.20 | creativx | aslaug |
| 12:12.32 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 12:14.05 | fors1 | Hi! We're running web-meetme on our asterisk pbx, but there are som problems. Asterisk doesn't always recognize the DTMF tones sent from skypeout users. Is there any way I can tweak my asterisk pbx to make it recognize skypeout users? |
| 12:15.46 | RoyK | creativx: that's a name... Åslaug, Ås is from the the norse word for god... |
| 12:23.03 | creativx | i know RoyK |
| 12:24.07 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.64.193) |
| 12:26.49 | defswork | hmm now I'm getting app_dial.c: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 12:28.08 | *** part/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 12:28.49 | *** join/#asterisk oej (n=olle@213.115.215.130) |
| 12:43.09 | tzafrir | franck, echo can also be generated at the handset |
| 12:43.29 | tzafrir | defswork, that's a generic error message |
| 12:43.42 | tzafrir | What was the Dial command? |
| 12:43.47 | defswork | somehow my zap trunk changed from working as g1 |
| 12:43.49 | tzafrir | Please provide a more complete trace |
| 12:43.55 | defswork | changed to g0 and now it works |
| 12:44.07 | defswork | I've no idea why |
| 12:44.18 | defswork | or even where the g0/g1 reference is tied |
| 12:44.35 | defswork | this box hasn't been touched since last march |
| 12:51.40 | tzanger | :-( |
| 12:51.50 | tzanger | the driver tries to sleep while atomic |
| 12:52.59 | tzafrir | tzanger, what version? |
| 12:53.15 | tzafrir | what rev.? |
| 12:53.17 | tzanger | moment |
| 12:53.48 | tzanger | 1265 |
| 12:54.07 | tzanger | it says I'm at revision 1270 actually |
| 12:54.19 | tzafrir | What problem? |
| 12:54.29 | tzafrir | ah, ok |
| 12:54.40 | tzanger | BUG: sleeping function called from invalid context at include/asm/semaphore.h:99 |
| 12:54.43 | tzafrir | not a nice place to sleep at |
| 12:54.44 | tzanger | in_atomic():1, irqs_disabled():1 |
| 12:54.48 | tzanger | whenever zaptel disables the echo can due to tone |
| 12:55.05 | tzanger | I haven't dug into it only because it's not killing mre |
| 12:55.07 | tzanger | er me |
| 12:55.18 | tzafrir | Do you happen to have the backtrace? |
| 12:56.12 | tzanger | yep |
| 12:56.36 | *** join/#asterisk ToTo (n=ToTo@207.176.6.159) |
| 12:56.55 | tzanger | http://pastebin.ca/857181 |
| 12:57.01 | tzanger | there's a bunch of 'em :-) |
| 12:57.45 | tzafrir | I'll try to look at it later. But generally David Rowe is quite responsive |
| 12:58.17 | tzanger | indeed |
| 13:06.27 | defswork | jblack: I need that |
| 13:06.52 | defswork | jblack: for a hotel - but they will need web interface to view/set etc.. too |
| 13:07.29 | jblack | I dial mine in. It's pretty easy to setup a web interface to do that. Look into "asterisk outgoing spool" |
| 13:07.55 | jblack | Drop a specially formatted file into the right dir, and based on the modification time of the file, it dials out. |
| 13:07.59 | *** join/#asterisk Victor_Yure (n=Victor_Y@189.67.176.9) |
| 13:11.56 | jblack | Perhaps it's a good thing for society that * isn't trivial to setup. Asterisk practically begs users to abuse the phone system |
| 13:12.12 | jblack | * in the hands of a 13 year old could be a disaster. |
| 13:15.30 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:15.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:15.36 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 13:15.48 | *** join/#asterisk suvir (n=suvir@ppp-124.120.138.127.revip2.asianet.co.th) |
| 13:19.53 | *** join/#asterisk oej (n=olle@213.115.215.130) |
| 13:20.07 | defswork | jblack: I've read about attended alarm setups - you dial in the time etc.. |
| 13:23.59 | tzafrir | tzanger, convert that lock to a spinlock? :-( |
| 13:26.02 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:27.14 | *** join/#asterisk gby (n=gby@line25-90.adsl.actcom.co.il) |
| 13:29.58 | *** part/#asterisk gby (n=gby@line25-90.adsl.actcom.co.il) |
| 13:30.00 | *** join/#asterisk gby (n=gby@line25-90.adsl.actcom.co.il) |
| 13:31.49 | *** part/#asterisk gby (n=gby@line25-90.adsl.actcom.co.il) |
| 13:32.00 | *** join/#asterisk Op3r (n=Op3r@203.177.233.86) |
| 13:32.43 | Op3r | hello |
| 13:33.12 | Op3r | can anyone tell me why Im getting error like this even though my sip account is correct? -- Executing Dial("IAX2/edwin-5", "SIP/vd1/89055107081||60|tTo") in new stack |
| 13:33.13 | Op3r | <PROTECTED> |
| 13:33.13 | Op3r | Jan 16 16:28:48 WARNING[2242]: chan_sip.c:9870 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"edwin" <sip:asterisk@192.168.100.6>;tag=as73ba0b0d' |
| 13:33.13 | Op3r | <PROTECTED> |
| 13:33.13 | Op3r | <PROTECTED> |
| 13:35.02 | tzafrir | Someone is being anti-social and doesn't like your invitations |
| 13:35.38 | *** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it) |
| 13:35.48 | Op3r | err |
| 13:35.50 | Op3r | :( |
| 13:36.27 | tzafrir | What are you connecting to? |
| 13:37.12 | Op3r | to my gateway |
| 13:37.28 | Op3r | let me paste my sip.conf entry for the gateway |
| 13:37.34 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:39.44 | Op3r | tzafrir: http://pastebin.com/d6aa1a5ad |
| 13:40.10 | Op3r | is that correct? |
| 13:41.34 | tzafrir | username=vd2 |
| 13:41.38 | tzafrir | Why is that? |
| 13:41.39 | Op3r | tzafrir: oh sorry here's the correct 1 |
| 13:42.11 | Op3r | grrrr\ |
| 13:42.15 | Op3r | now I see it |
| 13:42.16 | Op3r | :( |
| 13:44.13 | Op3r | but I still have this error |
| 13:44.14 | Op3r | <PROTECTED> |
| 13:44.14 | Op3r | Jan 16 16:43:52 WARNING[2242]: chan_sip.c:9870 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"edwin" <sip:asterisk@192.168.100.6>;tag=as5cbe4dfc' |
| 13:44.14 | Op3r | <PROTECTED> |
| 13:44.50 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1096745723.dsl.bell.ca) |
| 13:45.22 | tzafrir | did you reload? |
| 13:45.43 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 13:47.22 | *** join/#asterisk beasty_ (n=jdecoste@about/apple/macbook/beasty) |
| 13:47.23 | beasty_ | hi there |
| 13:47.29 | beasty_ | anyone ever saw this error before ? |
| 13:47.34 | beasty_ | [Jan 16 14:32:47] WARNING[11140]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination |
| 13:47.37 | Op3r | yes |
| 13:47.46 | oej | too many times... |
| 13:47.47 | [TK]D-Fender | beasty_: By itself is meaningless |
| 13:47.54 | beasty_ | i know |
| 13:48.17 | [TK]D-Fender | beasty_: PASTEBIN the complete CLI output of your call at verbose 10, and sip debug enabled. |
| 13:48.19 | [TK]D-Fender | ~pb |
| 13:48.19 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:48.20 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 13:48.21 | *** join/#asterisk ming_zym (n=ming_zym@124.14.235.182) |
| 13:49.28 | Op3r | [TK]D-Fender: kindly please check the config i did? http://pastebin.com/m44ab171d im getting this error Jan 16 16:43:52 WARNING[2242]: chan_sip.c:9870 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"edwin" <sip:asterisk@192.168.100.6>;tag=as5cbe4dfc' |
| 13:49.28 | Op3r | <Op3r> -- SIP/vd1-08d4a110 is circuit-busy |
| 13:49.44 | beasty_ | [TK]D-Fender: http://rafb.net/p/hIYOCn19.html |
| 13:50.34 | [TK]D-Fender | beasty_: I see no sip debug in there.. show "sip show peer dbwifi" as well please |
| 13:51.10 | [TK]D-Fender | Op3r: provide everything I just asked of beasty_ |
| 13:52.24 | Op3r | ok |
| 13:53.14 | beasty_ | [TK]D-Fender: http://rafb.net/p/QqYy6783.html |
| 13:55.43 | [TK]D-Fender | beasty_: and the reason I didn't see any sip debug in your first PB? |
| 13:56.27 | beasty_ | i didn't set the sip debug on |
| 13:59.54 | *** join/#asterisk [koss] (i=koss@adsl-75-36-15-24.dsl.bcvloh.sbcglobal.net) |
| 14:00.04 | [TK]D-Fender | beasty_: Then go do it |
| 14:02.59 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 14:04.58 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:06.06 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 14:06.32 | x86 | what commands do i use with dig to find the DNS servers for a given IP range? |
| 14:06.35 | x86 | dig c.b.a.in-addr.arpa. does not seem to show what i want |
| 14:10.08 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 14:13.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:14.29 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:15.21 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1096745723.dsl.bell.ca) |
| 14:17.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:20.58 | *** join/#asterisk wayfearer (n=chatzill@p57B6624D.dip.t-dialin.net) |
| 14:21.15 | [koss] | can someone suggest any good alternative to something like fonality |
| 14:21.30 | wayfearer | hey there can anyone tell me what i can do with a asterisk server in the web |
| 14:21.49 | mosty | wayfearer, like what? |
| 14:21.54 | wayfearer | like netPBX |
| 14:22.16 | wayfearer | i don't really understand in which way it could be useful for me |
| 14:22.23 | mosty | what's netpbx? |
| 14:22.31 | mosty | what do you mean by "in the web" ? |
| 14:22.55 | wayfearer | i've got a root server |
| 14:23.11 | wayfearer | and have the option to install it with Asterisk NOW |
| 14:23.35 | wayfearer | i know that asterisk is for providing telephone services |
| 14:23.37 | tzanger | nice, my cell calling dialplan is much better now |
| 14:23.49 | mosty | wayfearer, yes |
| 14:24.23 | wayfearer | but how does it work |
| 14:24.35 | wayfearer | i call the server and it calls my friend for free ? |
| 14:24.57 | mosty | wayfearer, no it won't magically let you call the regular telephone network for free |
| 14:25.11 | wayfearer | but ... |
| 14:25.26 | mosty | wayfearer, but if you and your friend are both connected to it using voip phones, you can call your friend for free |
| 14:25.31 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:25.34 | wayfearer | ah okay |
| 14:26.00 | mosty | wayfearer, or you could connect asterisk to a paid service that lets you call to/from the regular telephone network |
| 14:26.14 | wayfearer | so if i have a internet flat and a voip phone or a terminal adapter and all my friends too, we can call us for free |
| 14:26.21 | wayfearer | okay |
| 14:28.20 | mosty | you can connect to the the regular telephone network in various ways, some requiring hardware, some not, but they all cost money |
| 14:31.17 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:31.17 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:31.55 | wayfearer | okay thanks ! ;-) |
| 14:32.25 | beasty_ | damn |
| 14:32.28 | beasty_ | brb |
| 14:32.59 | [koss] | can someone suggest any good alternative to something like fonality |
| 14:33.16 | lirakis | I have a dial macro that forwards a call to my cell if i dont answer my extension. When the call forwards to my cell, I can NEVER hear the caller, but they can hear me. I have no issues in any other situation with one way audio, and my pbx is on public ip... so im uncertain what else could be causing this. |
| 14:33.22 | mosty | [koss], not since you last asked |
| 14:33.41 | [koss] | lol mosty |
| 14:33.54 | mosty | lirakis, is nat involved? |
| 14:34.25 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:34.37 | lirakis | mosty: no .. pbx is on public ip. caller can be calling from pstn or a voip phone.. same issue |
| 14:35.23 | mosty | what channel type do you use to dial your cellphone? |
| 14:35.33 | lirakis | mosty: sip |
| 14:35.42 | lirakis | mosty: out to my service provider |
| 14:36.02 | mosty | lirakis, do you have canreinvite=no in the service provider's setup? |
| 14:36.39 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 14:37.08 | *** join/#asterisk mog (n=mog@216.207.245.1) |
| 14:37.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:37.10 | *** join/#asterisk luygy (n=ogmious@ANantes-152-1-19-194.w83-195.abo.wanadoo.fr) |
| 14:37.26 | lirakis | mosty: not sure ... but would reinvite be required? since its terminating to a cell?... |
| 14:37.41 | lirakis | mosty: ill check and add it if it is not, then try it out |
| 14:38.06 | *** join/#asterisk gardo (n=gardo@121.97.142.167) |
| 14:39.03 | mosty | your calls will still work with canreinvite=no, it just forces the audio to always go through asterisk |
| 14:39.31 | lirakis | mosty: i know what reinvites are. |
| 14:39.40 | lirakis | mosty: i was saying, i dont think it will make any difference |
| 14:39.45 | ManxPower | reinvites do not generally work if there is NAT involved. |
| 14:40.10 | mosty | lirakis, when you say the happens for pstn calls too, what type of pstn connection do you have? |
| 14:40.11 | lirakis | mosty: call -> my* -> attempt ext.... attempt cell |
| 14:40.53 | lirakis | mosty: the endpoints arent changing so there is no need to reinvite for any reason |
| 14:41.07 | ManxPower | lirakis: So you have a Zaptel card in your server? An ITSP to terminate the call? |
| 14:41.49 | ManxPower | lirakis: reinvites is not usually about the end points changing, it's about getting the server out of the call and letting the endpoints send audio direct, rather than thru the server. |
| 14:41.51 | *** join/#asterisk luygy (n=ogmious@ANantes-152-1-19-194.w83-195.abo.wanadoo.fr) |
| 14:41.59 | luygy | <PROTECTED> |
| 14:42.11 | lirakis | mosty: .. okay .. so you arent listening, or just dont get what i'm saying.. ;) .. no worries i do not have any kind of pstn connection in my pbx.. i said "caller can be calling from pstn or a voip phone" as in the person calling me .. can be calling from a phone on the pstn unrelated to my pbx all together. |
| 14:43.08 | ManxPower | lirakis: SIP Phone <-> Asterisk <-> NAT <-> Internet <-> VoIP Provider <-> PSTN |
| 14:43.12 | mosty | lirakis, so the call comes in via sip, and might be sent back out via sip, and when that happens you get one way audio? |
| 14:43.25 | ManxPower | Is THAT your setup? If so then reinvites could be happening |
| 14:43.42 | lirakis | ManxPower: no NAT ... |
| 14:43.55 | mosty | reinvites can break even without nat |
| 14:44.02 | ManxPower | lirakis: So all your devices are on public IP addresses. |
| 14:44.10 | *** join/#asterisk luygy (n=ogmious@ANantes-152-1-19-194.w83-195.abo.wanadoo.fr) |
| 14:44.20 | lirakis | mosty: no the call can come in from my provider (from some land line or whatever) and terminate back out my provider to reach my cell. |
| 14:44.22 | ManxPower | mosty: maybe so, but broken reinvites when no NAT and no firewall is involved is not all that common. |
| 14:45.00 | lirakis | ManxPower: .. not all devices, some handsets can be, but since this issue is about when the pbx attempts to redirect a inbound call to my cell phone, NAT is not involved |
| 14:45.18 | mosty | ManxPower, setting canreinvite=no is a quick and easy test, probably better to just try it imo |
| 14:45.33 | ManxPower | mosty: Rather than arguing about it. |
| 14:45.38 | mosty | exactly |
| 14:45.48 | lirakis | ManxPower: call from .. wherever -> my provider -> internet -> my * ->internet-> my provider-> cell |
| 14:45.48 | ManxPower | lirakis: BTW, the term for that is a "hairpin" |
| 14:46.08 | ManxPower | lirakis: I assume there is also no firewall involved? |
| 14:46.13 | lirakis | ManxPower: no |
| 14:46.33 | ManxPower | and nothing on the asterisk console to indicate a problem? |
| 14:46.53 | lirakis | ManxPower mosty: FYI i did try with canreinvite=yes and no and have the same result |
| 14:47.40 | lirakis | ManxPower: no , it says attempting native bridge .... and the call sets up.. then when i hangup everything hangs up fine.. i just get no audio on my cell phone. but the caller can hear me. |
| 14:47.44 | mosty | lirakis, next step i would try personally is to run a packet logger on your asterisk box, and verify that rtp packets are coming in/going out with the correct source/dest addresses |
| 14:48.13 | fiXXXerMet | chan_zap shows Depends on: res_smdi(M), zaptel_vldtmf(E), zaptel(E) Does M mean What does M and E mean? It looks like chan_zap is good as it is marked (*) |
| 14:48.25 | ManxPower | lirakis: you have the classic symptom if a NAT problem -- but without any NAT. |
| 14:48.33 | lirakis | ManxPower: yeah, i know |
| 14:48.39 | lirakis | ManxPower: which is why im confused |
| 14:48.40 | lirakis | ha ha |
| 14:48.40 | ManxPower | lirakis: does your asterisk box have more than one IP address? |
| 14:48.46 | lirakis | ManxPower: nope just one |
| 14:49.27 | lirakis | ManxPower: i should setup a different carrier and attempt to terminate out that ... maybe my provider doesnt like it (shrug) |
| 14:50.35 | lirakis | ManxPower: ill bump up debug and see if i can get anythign else on the cli before i start messing with that though |
| 14:53.18 | *** join/#asterisk j_wizworks (n=n1wil@mail.gatehousemgt.com) |
| 14:54.24 | j_wizworks | hello, having some trouble getting Asterisk registered on Broadvoice can anyone assist? |
| 14:56.10 | mosty | j_wizworks, broadvoice perhaps? |
| 14:56.18 | *** part/#asterisk jmls (n=jmls@81.138.42.77) |
| 14:56.29 | j_wizworks | mosty: yes indeed. |
| 14:56.44 | j_wizworks | My register string looks like this: |
| 14:58.08 | j_wizworks | register => 6178121984:password@sip.broadvoice.com:password:6178121984@sip.broadvoice.com |
| 14:58.33 | j_wizworks | I get the following in the CLI: |
| 14:59.59 | lirakis | <PROTECTED> |
| 15:00.14 | j_wizworks | Jan 16 09:59:35 NOTICE[2665]: chan_sip.c:5431 sip_reg_timeout: -- Registration for '6178121984@sip.broadvoice.com' timed out, trying again (Attempt #91) |
| 15:00.17 | J4zen | i was wondering the same thing lol |
| 15:01.27 | mosty | j_wizworks, why do you have everything in your register line twice? |
| 15:02.18 | mosty | lirakis, do you have an incorrect externhost or related setting? |
| 15:02.48 | x86 | what's the cheapest headset that will work well with a Polycom IP301? |
| 15:03.13 | lirakis | mosty: no i have no externhost set |
| 15:03.50 | lirakis | mosty: some time back i had it set, then my dns changed and the whole system went berzerk.. so i no longer set it for production systems ha ha |
| 15:03.54 | lmadsen | afternoon all -- I've got a bunch of Mitel 5220 phones here that have been "bricked", but apparently can be reflashed. If anyone knows how to reflash these phones, please msg me, and if it works, I'll send you a free 5220 |
| 15:04.18 | lirakis | lmadsen: ahhhh mitel |
| 15:04.23 | lmadsen | :) |
| 15:04.35 | tzanger | hahahah |
| 15:04.47 | tzanger | help me fix my crappy phones and I'll give you one! |
| 15:04.50 | fiXXXerMet | Ever since I installed ztdummy and recompiled asterisk for zaptel support, I can no longer hear sound from files, such as the voicemail prompt, or conference prompts. I have tried installing zaptel both from source and packages, and still - no sound. Ideas? |
| 15:04.54 | lmadsen | I have a 5220 on my desk (the only one of the batch that works), and it's a pretty decent phone... but I have this box of them in my storage locker, and it's a shame because I could donate them to TAUG |
| 15:05.03 | x86 | tzanger: haha |
| 15:05.13 | tzanger | I'm hooked on polycom |
| 15:05.15 | ManxPower | fiXXXerMet: you have a problem with your kernel |
| 15:05.22 | mosty | lmadsen, do the bricked phones go into a reboot loop? |
| 15:05.23 | mocker | ugh, that reminds me. I need to prepare for my lug meeting this month. |
| 15:05.26 | fiXXXerMet | ManxPower: Oh yeah? How so? |
| 15:05.28 | x86 | tzanger: ever use headsets with your polycom phones? :) |
| 15:05.34 | tzanger | nope |
| 15:05.35 | x86 | tzanger: if so, what did you use? |
| 15:05.38 | tzanger | but they have the ports I know |
| 15:05.38 | x86 | damn |
| 15:05.41 | mocker | er, aug meeting. |
| 15:05.42 | lmadsen | mosty: I think that's what I was told (honestly I haven't turned one on, but that sounds very familiar) |
| 15:05.50 | x86 | yeah i know, but I'm not sure what i need... |
| 15:05.59 | mosty | lmadsen, there's a page on the wiki with a solution |
| 15:06.03 | ManxPower | fiXXXerMet: no idea, I don't use ztdummy. Your problem is not unique. Check the mailing list archives or (maybe) the Wiki |
| 15:06.06 | lmadsen | mosty: the mitel wiki? |
| 15:06.18 | mosty | lmadsen, no voip-info |
| 15:06.19 | fiXXXerMet | ManxPower: OK, thanks. |
| 15:06.30 | lmadsen | mosty: ok cool -- I'll look for it |
| 15:09.18 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 15:10.09 | lmadsen | mosty: ok... so perhaps I'm blind, but I can't find the page you're referring to :) |
| 15:10.16 | [TK]D-Fender | x86 : Plantronics M22 Amps + H263 Binaural headsets w/ polaris quick-connect |
| 15:10.23 | mocker | Ugh, I think I may have to get some Grandstreams. |
| 15:10.43 | x86 | [TK]D-Fender: looking for cheap |
| 15:11.00 | [TK]D-Fender | ~cheap |
| 15:11.01 | jbot | methinks cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 15:11.10 | [TK]D-Fender | x86 : What phone do you have? |
| 15:11.18 | x86 | [TK]D-Fender: IP301 |
| 15:11.19 | mocker | [TK]D-Fender: Do you know how much a Polycom 330 is in Sofia, Bulgaria? :P |
| 15:11.36 | mocker | Oh, I thought you were doing the ~cheap to me. |
| 15:11.37 | x86 | [TK]D-Fender: the M22 is $82 alone... |
| 15:11.42 | [TK]D-Fender | mocker: Whats the average air-speed velocity of a swallow? |
| 15:11.56 | x86 | [TK]D-Fender: was looking for something like the GN2010 |
| 15:12.09 | x86 | [TK]D-Fender: can i use the GN2010 (which is half the price of the M22 amp alone) |
| 15:12.12 | [TK]D-Fender | x86 : you can try to find an RJ9 > 2.5mm adapter, but unamped it sucks. |
| 15:12.18 | mocker | [TK]D-Fender: African or European? |
| 15:12.29 | [TK]D-Fender | x86 :un-amped = bad. Try it and find out. |
| 15:12.38 | [TK]D-Fender | mocker: I don't know! |
| 15:12.40 | *** join/#asterisk AndyGraybeal (n=andy@node53.34.251.72.1dial.com) |
| 15:12.41 | x86 | [TK]D-Fender: any cheaper solution? |
| 15:12.41 | [TK]D-Fender | aaaaaaaaarrrrrrrrrrrggggggghhhhhhhhhhh |
| 15:12.56 | [TK]D-Fender | x86 : Don't be a cheap-ass. |
| 15:14.03 | RoyK | ~seen inspired |
| 15:14.05 | jbot | inspired <n=mikael@62.141.128.222> was last seen on IRC in channel #debian, 17d 6h 37m 24s ago, saying: 'thanks'. |
| 15:14.38 | j_wizworks | mosty: sorry boss called me for a few monutes... The reg string I have is the same string format that BV recommends on their support page. |
| 15:14.57 | mosty | j_wizworks, best give them a call then |
| 15:15.42 | j_wizworks | Already done that... they have stated to me that they support asterisk, but it's the user's responsibility to properly configure it. |
| 15:16.21 | mosty | j_wizworks, is your asterisk box behind nat? if so try setting externip |
| 15:16.22 | j_wizworks | I have tested the SIP acct on a soft SIP phone and it works... so I know the account info is good. I just can't seem to get asterisk to register. |
| 15:16.38 | lirakis | j_wizworks: .. did you try what we reccomended? i.e. not having a duplicate domain |
| 15:16.45 | j_wizworks | mosty: externip is set to WAN IP. |
| 15:16.54 | mosty | j_wizworks, what lirakis said |
| 15:17.50 | AndyGraybeal | all day is asterisk day today :) |
| 15:17.59 | j_wizworks | yes tried removing that... and no difference... at home (on another BV acct) I have an Asterisk @home box and it uses the same reg string format (but on a diff. account). This ast. box here is installed on a Debian 4 system using the apt-get packages. |
| 15:18.24 | j_wizworks | lirakis: yes tried that, no success. |
| 15:18.26 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 15:18.35 | esaym | is a sound card needed with asterisk? |
| 15:19.03 | esaym | and is this a problem: chan_iax2.c: Unable to open IAX timing interface: No such file or directory |
| 15:19.11 | lirakis | esaym: no |
| 15:19.17 | esaym | using asterisk 1.2 |
| 15:19.35 | mosty | esaym, no sound card needed. you need a zaptel card for some applications like meetme, or iax trunking |
| 15:20.20 | esaym | well I connect to the voip provider with iax. I don't know if that is trunking though. Either way it is not working :-/ |
| 15:20.47 | esaym | and calling the demo extensions does nothing, no sound or anything but I think it answers |
| 15:20.49 | mosty | esaym, well that's just a warning that you can ignore for now then. set trunk=no in iax.conf |
| 15:21.02 | mosty | your error is elsewhere |
| 15:21.11 | _ShrikE | lmadsen: http://www.voip-info.org/wiki/view/Asterisk+phone+Mitel+5220... Ive done the supkey thingy in the past. |
| 15:21.50 | j_wizworks | this is the first time I'm trying to set up asterisk without a GUI (as like ASterisk @ Home). Plus I prefer a Debian based OS underneath it as opposed to a RH like OS (CentOS) |
| 15:21.55 | *** part/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 15:23.27 | x86 | [TK]D-Fender: works perfectly fine with an un-amped headset |
| 15:23.35 | j_wizworks | the files I've modified are : /etc/hosts (inserted the sip.broadvoice.com record), /etc/asterisk/sip.conf, and /etc/asterisk/extensions.conf |
| 15:23.43 | x86 | [TK]D-Fender: i stole one from an analog VXI dialpad phone.. works great |
| 15:23.52 | j_wizworks | according to BV's support page. |
| 15:24.17 | j_wizworks | would there be any other file or setting I would need to check/change in the configs? |
| 15:24.21 | lirakis | <PROTECTED> |
| 15:24.30 | j_wizworks | pasting...: |
| 15:24.35 | j_wizworks | Jan 16 09:59:35 NOTICE[2665]: chan_sip.c:5431 sip_reg_timeout: -- Registration for '6178121984@sip.broadvoice.com' timed out, trying again (Attempt #91) |
| 15:24.40 | lirakis | <PROTECTED> |
| 15:24.55 | lirakis | j_wizworks: so, can you ping broadvoice.com? |
| 15:24.59 | j_wizworks | yes |
| 15:25.18 | lirakis | j_wizworks: so get some sip msg debugging going on |
| 15:25.28 | j_wizworks | reply: 64 bytes from sip.broadvoice.com (147.135.32.221): icmp_seq=1 ttl=249 time=6.81 ms |
| 15:25.53 | lirakis | j_wizworks: asterisk -vvvvvvvvvvvvvr .. then do sip debug peer 147.135.32.221 |
| 15:25.56 | *** part/#asterisk mog (n=mog@216.207.245.1) |
| 15:26.03 | j_wizworks | ok lemme try that.... |
| 15:27.02 | lirakis | <PROTECTED> |
| 15:27.04 | lirakis | not peer |
| 15:27.18 | j_wizworks | No such peer '147.135.32.221' |
| 15:27.26 | j_wizworks | ok |
| 15:28.03 | j_wizworks | output: |
| 15:28.10 | j_wizworks | Retransmitting #6 (NAT) to 147.135.32.221:5060: |
| 15:28.12 | j_wizworks | REGISTER sip:sip.broadvoice.com SIP/2.0 |
| 15:28.13 | j_wizworks | Via: SIP/2.0/UDP 209.213.70.66:5060;branch=z9hG4bK7a2bf52c;rport |
| 15:28.15 | j_wizworks | From: <sip:6178121984@sip.broadvoice.com>;tag=as5d8bad5f |
| 15:28.16 | j_wizworks | To: <sip:6178121984@sip.broadvoice.com> |
| 15:28.18 | j_wizworks | Call-ID: 2c9a13f9042a7ab22549a90f0a27aef1@sip.broadvoice.com |
| 15:28.19 | j_wizworks | CSeq: 277 REGISTER |
| 15:28.21 | j_wizworks | User-Agent: Asterisk PBX |
| 15:28.22 | j_wizworks | Max-Forwards: 70 |
| 15:28.24 | j_wizworks | Expires: 120 |
| 15:28.25 | j_wizworks | Contact: <sip:s@209.213.70.66> |
| 15:28.27 | j_wizworks | Event: registration |
| 15:28.28 | j_wizworks | Content-Length: 0 |
| 15:28.34 | mosty | don't paste here |
| 15:28.35 | lirakis | <PROTECTED> |
| 15:28.37 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:28.39 | lirakis | ~pastebin |
| 15:28.40 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:28.49 | j_wizworks | sorry |
| 15:28.52 | j_wizworks | new to IRC |
| 15:28.55 | j_wizworks | go easy on me please. |
| 15:31.24 | j_wizworks | ok sent to the pastebin, how to send the pastebin to another? |
| 15:31.29 | j_wizworks | just send the URL? |
| 15:31.47 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:32.20 | fiXXXerMet | j_wizworks: yes, the url |
| 15:32.38 | j_wizworks | ok here goes: http://pastebin.com/d2ce78143 |
| 15:32.40 | j_wizworks | thanks |
| 15:32.42 | lirakis | anyone know if ASTCC is still maintained? it seems dead |
| 15:32.54 | lirakis | lol |
| 15:33.08 | lirakis | j_wizworks: thats like .. have a register message |
| 15:33.12 | lirakis | *half |
| 15:33.40 | j_wizworks | lirakis: yeah looks to be so, but for whatever reason I get the failed registration. |
| 15:34.18 | lirakis | j_wizworks: come on man.... we need to see the error.. not just the first part saying "okay..im going to try to register now" |
| 15:34.34 | j_wizworks | ok let me paste the error int he bin... |
| 15:35.07 | lmadsen | _ShrikE: thanks for the tip -- I'm hoping I can give this a try tonight and get some of the phones working |
| 15:36.02 | lirakis | j_wizworks: paste in all!!! |
| 15:36.33 | j_wizworks | lirakis: how do go back to seeing the error - it is still showing the reg attempts from the sip debug ip 147.135.32.221 command? |
| 15:36.39 | j_wizworks | <--- new to CLI |
| 15:37.18 | lirakis | j_wizworks: ... |
| 15:37.25 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 15:37.34 | teknoprep | where do i get the firmware for polycom phones? |
| 15:37.38 | teknoprep | i have an ip320 |
| 15:37.46 | lirakis | j_wizworks: scroll up.. or just sit there .. it will attempt to register again |
| 15:37.51 | lirakis | teknoprep: uhh... polycom |
| 15:37.54 | mosty | teknoprep, polycom's website |
| 15:38.09 | _ShrikE | teknoprep: you can get older firmware on the polycom website, if you want the new stuff you need to talk to your reseller |
| 15:38.12 | [TK]D-Fender | j_wizworks: pastebin your sip.conf masking only passwords. |
| 15:38.12 | teknoprep | yeah no crap... where on there site.. and do i have to pay for it like cisco ? |
| 15:38.23 | [TK]D-Fender | teknoprep: from your reseller |
| 15:38.40 | j_wizworks | ok D-fender |
| 15:38.40 | mosty | teknoprep, support, phones, model |
| 15:39.24 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:40.22 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1096745723.dsl.bell.ca) |
| 15:40.44 | j_wizworks | lirakis: here is the debug and the error: http://pastebin.com/d71817e41 |
| 15:41.27 | Uni | anyone know offhand of where I could find info on setting the callerID of all outgoing calls to a single ID string? I've found some docs and played with SetCallerID but haven't had any luck so far |
| 15:41.43 | [TK]D-Fender | j_wizworks: that tells very little, now provide what I ask for. |
| 15:42.01 | [TK]D-Fender | Uni : "show function CALLERID" |
| 15:42.26 | j_wizworks | d-fender, copying the file and masking passwords... |
| 15:43.24 | esaym | Where do I start trouble shooting if none of the demo extensions in the default configs work? Or atleast I don't hear any sound... |
| 15:43.44 | *** join/#asterisk itguru (n=gabriel@82.108.189.18) |
| 15:43.58 | lirakis | j_wizworks: sigh again.. you are not giving all the messaging... |
| 15:44.05 | esaym | That is with the gizmo soft phone though, I can't get the x-lite soft phone to register |
| 15:44.08 | itguru | Hello all you asterisk people! |
| 15:44.21 | eric_hill | esaym: on the asterisk console (asterisk -r) do a "set verbose 9" and see if that helps. |
| 15:44.25 | lirakis | j_wizworks: its clear you dont understand what is important in the data... so leave it all in and let us decide what to look for, instead of cutting out all that you think isnt relevant. |
| 15:44.46 | itguru | I've got 24 hours to get a working VoIP setup up and running, and I'm in need of some serious help |
| 15:45.09 | j_wizworks | lirakis: I'm new yes I do not understand, but I'm trying, being new to IRC adds another layer of complication to the issue... I'm pasting the sip.conf file and it's taking time brb |
| 15:45.53 | kyron | j_wizworks, just make sure you don't paste here ;) |
| 15:45.56 | itguru | Has anyone ever got an Avaya handset running, or a Cisco 7910 connected to an asterisk box? |
| 15:46.25 | alrs | itguru getting Cisco stuff to work can be a bit of a bottomless pit |
| 15:46.50 | alrs | itguru: I've helped set up a trunk directly from an Avaya Definity system to an Asterisk box |
| 15:47.38 | eric_hill | itguru: Practically all of the Cisco phones I've tried work just fine. 7905, 7910, 7940, 7960, etc. |
| 15:47.46 | j_wizworks | that's why it's taking me forever to copy the sip.conf file out of the putty window... into the pastbin, is there an easier way to get the contents of this file out and paste? |
| 15:47.48 | esaym | eric_hill: console says " -- Executing Playback("SIP/111-0818f978", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en')" when I dail the 500 extension but I don't hear anything |
| 15:48.07 | [TK]D-Fender | j_wizworks: 1 swipe with the mouse grabs everything.... |
| 15:48.42 | eric_hill | esaym: If asterisk thinks it's playing the file, chances are that it is and your phone isn't decoding the audio correctly. Softphone? Physical phone? |
| 15:48.50 | *** join/#asterisk qdk (n=qdk@193.164.155.7) |
| 15:48.57 | j_wizworks | I have the file open with the nano text editor... (debian OS) I have to scroll the file, then highlight and it';s a pain... is there a simpler method? |
| 15:49.05 | esaym | eric_hill: crappy softphone |
| 15:49.18 | esaym | ata adapter coming in the mail today |
| 15:49.30 | *** join/#asterisk Havokmon (n=None@mail.valeoinc.com) |
| 15:49.44 | eric_hill | esaym: Try the X-Lite SIP phone: http://www.asteriskguru.com/tutorials/xlite_softphone.html |
| 15:50.31 | esaym | eric_hill: ye i have tried that, I couldn't get it to reg, it timed out |
| 15:50.44 | *** join/#asterisk spid3r_ (n=spid3r@229.87.modemcable.oricom.ca) |
| 15:51.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:52.03 | jim`` | wengophone has worked best for me for testing |
| 15:52.08 | spid3r_ | is there a way to know which sip user is making a call with a predefined variable? |
| 15:52.37 | [TK]D-Fender | j_wizworks: cat it from CLI copy & paste. 5 seconds |
| 15:52.46 | mosty | spid3r: set a channel variable in sip.conf |
| 15:53.15 | j_wizworks | sweet... lemme try that. |
| 15:53.25 | spid3r_ | mosty: thanks i'll give it a look |
| 15:53.43 | [TK]D-Fender | spid3r_: just parse out ${CHANNEL} |
| 15:53.57 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:54.30 | lirakis | esaym: make sure you set proxy and register host on xlite even if they are the same. ive never had any problems with it |
| 15:55.43 | esaym | lirakis: Yea I am messing with it right now, it just keeps timing out |
| 15:56.21 | esaym | nothing in the logs, |
| 15:56.40 | esaym | my network sniffer shows it sending packets to the server but it never responds |
| 15:57.43 | [TK]D-Fender | esaym: enable SIP debug on * CLI and pastebin the full call attempt. |
| 15:59.13 | [TK]D-Fender | SUN Buys MySQL , wheee! ---> http://blogs.mysql.com/kaj/2008/01/16/sun-acquires-mysq |
| 15:59.50 | drmessano | yep |
| 16:00.06 | drmessano | Not sure if thats good or bad yet |
| 16:00.10 | nixguy | for 100 million |
| 16:00.13 | nixguy | or something like that |
| 16:00.14 | Qwell | 1B |
| 16:00.15 | nixguy | niiice :) |
| 16:00.21 | Qwell | they were worth more, IMO |
| 16:00.28 | Qwell | quite a bit more |
| 16:00.30 | nixguy | mysql is a swedish company |
| 16:00.32 | nixguy | *proud* |
| 16:00.49 | tzanger | hopefully sun'll bury it |
| 16:00.50 | nixguy | actually they used to sit in the office i worked in a couple of years back |
| 16:00.53 | tzanger | <-- NOT a fan of mysql |
| 16:01.09 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 16:01.35 | AndyGraybeal | nixguy: that's pretty cool |
| 16:02.28 | drmessano | Maybe they they buy PHP too.. so the crap will work better together lol |
| 16:02.35 | drmessano | they will* |
| 16:03.57 | Havokmon | better? How so? |
| 16:03.58 | cappiz | someone here knows how i can "hack" asterisk to send DID? my IAX provider doesnt send DID... but iax debug shows CALLING NUMBER: |
| 16:04.10 | cappiz | with the correct DID |
| 16:04.43 | drmessano | Theres still a LOT of issues with PHPs MySQL modules |
| 16:04.44 | lmadsen | tzanger: pgsql, what?! |
| 16:04.48 | esaym | [TK]D-Fender: one second, I am on to something... |
| 16:05.03 | drmessano | Need of a group up rewrite, IMO |
| 16:05.09 | lmadsen | cappiz: uhhh... you mean callerid? |
| 16:05.33 | lmadsen | and if the provider doesn't send it (or allow you to send it -- it's not clear what direction you're going) -- there is nothing Asterisk can do about it |
| 16:05.47 | Havokmon | drmessano: link? I don't doubt you, I just want to know what to look out for. |
| 16:05.55 | cappiz | i called form my cell... to my pbx... IAX debug shows the number i called form my cell under "CALLING NUMBER:" |
| 16:06.16 | lmadsen | your provider isn't sending the CID probably then -- that's a provider issue |
| 16:06.19 | j_wizworks | http://pastebin.com/d1e9ad099 |
| 16:06.23 | lmadsen | or your dialplan is wrong -- use CALLERID() function |
| 16:06.25 | drmessano | http://www.google.com |
| 16:06.32 | drmessano | You find it here and there |
| 16:06.37 | j_wizworks | the sip reg error |
| 16:06.57 | [TK]D-Fender | j_wizworks: go into your sip.conf and permanently wipe out every commented-out line and repastebin it. |
| 16:06.59 | Havokmon | drmessano: lol I didn't want to sift through idiots who're causing their own problems :P |
| 16:07.03 | drmessano | lol |
| 16:07.18 | drmessano | There isnt ONE PAGE thats says PHP + MYSQL SUXORS HERES WHY |
| 16:07.22 | drmessano | Just need to look |
| 16:07.50 | cappiz | isn't DID the my pbx phonenumber? |
| 16:07.54 | Havokmon | That's what I wanted ;) "don't use if..." |
| 16:08.12 | j_wizworks | actually this is the correct post: |
| 16:08.14 | j_wizworks | http://pastebin.com/d32395005 |
| 16:08.19 | lmadsen | cappiz: DID is Direct Inward Dialing -- it is the number that Asterisk should be matching on |
| 16:08.27 | j_wizworks | sorry bout the confusion. |
| 16:08.34 | lmadsen | cappiz: I think you're using the wrong term |
| 16:08.38 | lmadsen | a DID is a phone number |
| 16:08.48 | lmadsen | (that you would dial and would be routed "somewhere") |
| 16:09.07 | lmadsen | from what I can tell, you're talking about CallerID |
| 16:09.08 | cappiz | yeah... thats what im thinking about |
| 16:09.21 | cappiz | no i want to use inbound routes |
| 16:09.30 | docelmo | Wow someone asking VERY basic questions.. Aparently he didnt bother to check for himself anywhere.. |
| 16:09.37 | cappiz | and route call to different direction depening on which number the user dialed |
| 16:09.53 | cappiz | but my iax providers doesnt provide me with DID |
| 16:10.09 | docelmo | cappiz who's your IAX provider? |
| 16:10.12 | cappiz | i have several iax/sip trunks |
| 16:10.30 | cappiz | docelmo, a norwegian provider |
| 16:10.32 | cappiz | cbktele |
| 16:10.36 | docelmo | ahh.. |
| 16:10.45 | RoyK | cappiz: lite firma? |
| 16:11.06 | cappiz | we'll do norwegian in a private chat |
| 16:11.11 | RoyK | heh |
| 16:11.22 | itguru | alrs: I've got 30 Cisco phones :( - and about |
| 16:11.38 | docelmo | itguru sorry to hear that |
| 16:11.50 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:12.30 | j_wizworks | d-fender: is there an easy quick way to wipe out those lines? |
| 16:13.00 | *** join/#asterisk RipeR-81 (n=ircap8@190.53.33.10) |
| 16:13.13 | [TK]D-Fender | j_wizworks: to the more knowledgeable, yes. just do it.... it shouldn't have taken even the 5 minutes since I asked to to it the :hard" way |
| 16:13.18 | RipeR-81 | anyone available ? im integrating ccm6 and asterisk |
| 16:13.44 | RipeR-81 | having asterisk as a gateway and ccm as my pbx using cisco ip phones 7941 |
| 16:13.53 | j_wizworks | if I make a copy of the file and process the copy thru sed will that work? |
| 16:14.07 | j_wizworks | (just not sure of the sed syntax) |
| 16:14.18 | j_wizworks | and also trying to save you time. |
| 16:14.23 | RipeR-81 | i am not able to send the calls from cisco call manager 6 to asterisk |
| 16:15.20 | itguru | docelmo: Why sorry?! Do I have a lot of work ahead of me?? |
| 16:15.45 | *** join/#asterisk ayrjola (n=ayrjola@cs181173201.pp.htv.fi) |
| 16:16.08 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 16:16.23 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:16.59 | iratik | Considering buying a TDM800P.... So I can have 8 FXS ports or 8 FXO ports... or have 2 cards - one with 8 FXS one with 8 FXO? |
| 16:17.44 | [TK]D-Fender | iratik: Zaptel FXS = ASS <- |
| 16:17.47 | Qwell | iratik: TDM2400, get all of them |
| 16:17.57 | [TK]D-Fender | iratik: and at 8 FXO you should look at partial PRI |
| 16:18.08 | [TK]D-Fender | Qwell: Got that in PCMCIA? ;) |
| 16:18.11 | iratik | I've got voice t1 coming in |
| 16:18.23 | [TK]D-Fender | iratik: then whats 8 FXO for? |
| 16:18.31 | Qwell | [TK]D-Fender: I did see a PCMCIA analog card once by some company.. no zaptel/asterisk drivers though |
| 16:18.34 | Qwell | neat idea, but...meh |
| 16:18.36 | iratik | not using all the lines |
| 16:18.49 | [TK]D-Fender | iratik: for your FXS : Linksys SPA-8000 |
| 16:19.12 | eric_hill | RipeR-81: Cisco CCM uses CSSP to negotiate with the phones, but it will connect to remote SIP targets. What have you tried? |
| 16:19.29 | anonymouz666 | [TK]D-Fender: is it the same PAP2 firmware? |
| 16:19.52 | iratik | I think I like the TDM2400 better so far |
| 16:20.07 | [TK]D-Fender | anonymouz666: More like SPA-2102 expanded |
| 16:20.11 | tzafrir | iratik, it would generally be recommended to have just one card |
| 16:20.12 | *** join/#asterisk jochien1 (n=jochieng@217.194.147.193) |
| 16:20.44 | j_wizworks | d-fender: http://pastebin.com/d631cac79 |
| 16:20.48 | j_wizworks | how's this? |
| 16:20.52 | RipeR-81 | eric_hill no i havent . i just followed the guide http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration |
| 16:20.53 | jochien1 | hi- friends. i m looking for a free sip telephony provider account! |
| 16:20.55 | [TK]D-Fender | iratik: Zaptel FXS handling = BLEH, Wiring costs = bleh, COST = bleh, size of the card = bleh. |
| 16:20.58 | j_wizworks | all commented lines removed. |
| 16:21.17 | iratik | so the FXS ports will not work properly? |
| 16:21.32 | RipeR-81 | eric_hill all the phones are connect to the ccm 6... are able to call within them selves |
| 16:21.40 | Qwell | [TK]D-Fender: howso? |
| 16:21.41 | tzafrir | And plently of BLEH devices with bleh handling, because the handling is not under your control |
| 16:21.59 | iratik | but FXO ports will work fine... so I will have to pretty much end up hooking up IP phones or ATAs to the network ? |
| 16:22.00 | [TK]D-Fender | j_wizworks: You have to move your REGISTER from 30 to 34, and set canreinvite=no under [genera] and every other section. |
| 16:22.17 | Qwell | I've never had any trouble with zaptel fxs O.o |
| 16:22.26 | eric_hill | RipeR-81: Have you set up the SIP trunk like the web page shows? |
| 16:22.35 | jochien1 | some pls help with my inquiry |
| 16:22.47 | iratik | I can run the individual lines into the many FXO ports the TDM2400 will provide me... then I will have a zaptel trunk ... then I can hook-up ipphones or ATAs to the network |
| 16:22.51 | j_wizworks | ok I'll try that... |
| 16:23.13 | [TK]D-Fender | Qwell: Load on your server, no redundancy, more to configure, DTMF transfers/conferences =ew |
| 16:23.22 | eric_hill | RipeR-81: If CCM is sourcing the call correctly, then you need to make sure your asterisk sip.conf file puts the inbound calls into the correct context. |
| 16:23.32 | Qwell | zaptel has like no load... |
| 16:23.36 | [TK]D-Fender | Qwell: and on top costs more and is on that HUGE ass card that won't fit in many servers. |
| 16:23.47 | eric_hill | RipeR-81: The asterisk console should show an inbound call. |
| 16:23.49 | RipeR-81 | eric_hill i have put the sip trunk just like the page shows |
| 16:23.52 | itguru | Any idea how to get Cisco 7910 phones working with *isk |
| 16:23.55 | [TK]D-Fender | Qwell: Adds cards to a box that doesn't need it. |
| 16:23.55 | RipeR-81 | eric_hill no.. wont show inbound |
| 16:24.03 | jochien1 | hi- friends. i m looking for a free sip telephony provider account! where can i get 1 ;) |
| 16:24.07 | Qwell | and it's less to configure. ATAs are a PITFA |
| 16:24.09 | RipeR-81 | eric_hill thats what i think the problem might be |
| 16:24.32 | eric_hill | RipeR-81: Can you watch a call from the CCM? I've only worked with CallManager Express, not the full Call Manager. |
| 16:24.37 | [TK]D-Fender | Qwell: 5 mins to setup, all said & done |
| 16:24.39 | RipeR-81 | si |
| 16:24.40 | Qwell | not only would you have to configure the ATA, you'd have to configure sip.conf |
| 16:24.43 | iratik | ATA's would probably stink anyway |
| 16:24.45 | [TK]D-Fender | jochien1: www.fwdnet.net |
| 16:24.45 | Qwell | sure, same with zaptel :p |
| 16:24.46 | RipeR-81 | eric_hill yeah i can see the call |
| 16:24.51 | iratik | thats why i'd like to use FXS ports |
| 16:24.57 | Qwell | it's like 2 lines in zaptel.conf/zapata.conf |
| 16:25.01 | [TK]D-Fender | Qwell: unless you run into IRQ issuse, kernel source, etc.... |
| 16:25.02 | RipeR-81 | eric_hill i can even make phone calls from asetrisk extensions to CCM extensions |
| 16:25.07 | iratik | that way ... the client's employees have no clue that they ever switched the PBX out |
| 16:25.18 | jochien1 | [TK]D-Fender: thanks let me check it out |
| 16:25.20 | [TK]D-Fender | Qwell: bad timers, and the million other things then never EVER happen with PCI solutions ;) |
| 16:25.25 | Qwell | [TK]D-Fender: if you have IRQ issues with a Digium card, I'll let you call me personally :p |
| 16:25.31 | Qwell | I will personally handle your case ;) |
| 16:25.40 | eric_hill | itguru: http://www.voipuser.org/forum_topic_5258.html http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx |
| 16:25.42 | Qwell | (no, not really) |
| 16:25.48 | [TK]D-Fender | Qwell: Well played :p |
| 16:25.59 | [TK]D-Fender | Qwell++ |
| 16:26.10 | iratik | If I were considering hosting an asterisk installation on Amazon EC2 or hosted provider with a much bigger bandwidth pool what would you sayu |
| 16:26.28 | eric_hill | RipeR-81: So what does the call show when CCM->asterisk? |
| 16:27.01 | *** part/#asterisk ayrjola (n=ayrjola@cs181173201.pp.htv.fi) |
| 16:27.37 | [TK]D-Fender | iratik: No, more bandwidth is bad... |
| 16:27.43 | iratik | why? |
| 16:27.51 | iratik | why more bandwidth==bad? |
| 16:27.52 | [TK]D-Fender | iratik: </sarcasm> |
| 16:27.57 | iratik | oh kay |
| 16:28.13 | [TK]D-Fender | iratik: WTF do you think we're going to say "Yes your idea of a massive uprgade is a bad thing?!" |
| 16:28.20 | kyron | meh, bandwidth is useless if you don't have latency |
| 16:28.38 | RipeR-81 | eric_hill let me check |
| 16:28.44 | iratik | well... I figured i might get some type of argument against putting a PBX on a hosting provider like EC2 |
| 16:28.45 | *** join/#asterisk hades123 (n=wqwsqww@d57-199-17.home.cgocable.net) |
| 16:28.49 | [TK]D-Fender | kyron: Latency is bad, so not having latency, and having bandwidth is GOOD <- |
| 16:28.50 | j_wizworks | D-Fender: http://pastebin.com/d490067c5 |
| 16:28.52 | j_wizworks | no change. |
| 16:28.56 | [TK]D-Fender | kyron: ...:p |
| 16:29.14 | eric_hill | iratik: EC2 instances can be shut-down by Amazon on a whim. |
| 16:29.24 | kyron | [TK]D-Fender, I forgot _low_ latency ;) |
| 16:29.39 | eric_hill | iratik: They'll start another one up for you, but I wouldn't like my phone system going down "just because" |
| 16:29.39 | kyron | or guaranteed latency at that matter |
| 16:29.40 | RipeR-81 | eric_hill i beieve one of my errors is not setting the incoming context... |
| 16:29.52 | RipeR-81 | eric_hill is there a guide? im googling |
| 16:30.07 | [TK]D-Fender | j_wizworks: under [sip.broadvoice.com] you should have "nat=no". Also what have you forwarded to your * box? |
| 16:30.18 | eric_hill | RipeR-81: The incoming context name is set in the sip.conf. The context should match a section in the extensions.conf. |
| 16:30.31 | eric_hill | RipeR-81: Can you pastebin those two please? |
| 16:30.52 | RipeR-81 | eric_hill sure |
| 16:30.57 | iratik | so ... not a good idea --- i'll find a dedicated hosting provider .. kay |
| 16:30.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:31.21 | RipeR-81 | eric_hill i havent set the incoming context on extensions.conf only have the ccm as friends on sip.conf |
| 16:31.25 | iratik | one last question... can anyone point me to a resource of howto's with AMI .... like in particular... how to use reinvite |
| 16:31.38 | j_wizworks | ok I'll change that setting ... so far 5060 is forwarded to the * box. |
| 16:32.39 | RipeR-81 | eric_hill the url is http://pastebin.com/m3f9ee33d |
| 16:33.09 | [TK]D-Fender | j_wizworks: 5060 "what"? |
| 16:33.44 | j_wizworks | 5060 UDP |
| 16:33.47 | eric_hill | RipeR-81: So your inbound calls should drop into the default context. Pastebin your extensions.conf so I can see what's in the default context. |
| 16:33.59 | itguru | eric_hill: Thanks for the URLs, the one on the wiki, I've seen that before, but I didn't find it helpful enough |
| 16:34.05 | RipeR-81 | ok |
| 16:35.10 | itguru | eric_hill: I know that I need to have a SIP phone image, and get that image onto the handset, but once that is on the handset, how do I configure it? |
| 16:35.30 | [TK]D-Fender | j_wizworks: you did not put "nat=no" under [sip.broadvoice.com] , and you need to forward 5060, 10000-20000 all UDP to * |
| 16:35.32 | RipeR-81 | eric_hill this is my extensions.conf default context http://pastebin.com/d564670fd |
| 16:35.40 | eric_hill | itguru: I don't think the 7910 has a SIP image available... please hold... |
| 16:36.13 | eric_hill | RipeR-81: does the call come in as 75xx or 79xx? |
| 16:36.15 | j_wizworks | ok I'll forward that range as well... |
| 16:36.23 | [TK]D-Fender | j_wizworks: and you'll have to make sure your firewall is not in the way |
| 16:36.31 | eric_hill | RipeR-81: duh - scratch that. The calls are coming in on 75xx, right? |
| 16:37.02 | RipeR-81 | ok 75xx are asterisk extensions |
| 16:37.10 | RipeR-81 | 79xx are ccm extensions |
| 16:37.31 | RipeR-81 | eric_hill i believe im missing something here... |
| 16:39.17 | j_wizworks | D-Fender: ok checked all those things and made the additional range forward - no other firewalls in the way. Still unable to register: http://pastebin.com/d44ef05b0 |
| 16:39.19 | eric_hill | itguru: Still trying to log in to Cisco |
| 16:39.49 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 16:40.54 | eric_hill | itguru: Confirmed. 7905/7912/7940/7960 has SIP firmware. 7910 is SCCP only. |
| 16:40.56 | [TK]D-Fender | j_wizworks: what router are you using? |
| 16:41.04 | j_wizworks | m0n0wall |
| 16:41.28 | [TK]D-Fender | j_wizworks: did you setup a hosts entry for sip.broadvoice.com? |
| 16:41.34 | j_wizworks | same as I'm using at home. and the Ast@home box works even without the port forwards on another BV account. |
| 16:41.39 | j_wizworks | yes |
| 16:41.40 | eric_hill | itguru: Wow. Not only that, but the 7910 firmware is *old*. Jun 30, 2005. |
| 16:41.46 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:41.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:41.47 | j_wizworks | there is a hosts entry |
| 16:42.01 | j_wizworks | pointing sip.broadvoice.com to the nyc proxy |
| 16:42.02 | [TK]D-Fender | j_wizworks: ok, I'm at a loss at this point then |
| 16:44.30 | itguru | eric_hill: Thanks for the tip! I really appreciate that. Now I just have to figure out how to get SCCP working |
| 16:44.39 | j_wizworks | the only difference with thos box and the one at home is this one is one is my 1st attempt to build an * box on Debian using the APT repos. I like the debian OS better than the RH based ones like CentOS, Fedora, etc. |
| 16:44.47 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 16:45.40 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:46.08 | j_wizworks | wish there was a debian based version of *@home. |
| 16:46.12 | j_wizworks | would be sweet. |
| 16:48.05 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 16:50.04 | *** part/#asterisk SexyKen (n=sexy@c-24-4-238-80.hsd1.ca.comcast.net) |
| 16:51.19 | AndyGraybeal | does xlite run slow on others computers? |
| 16:51.23 | [TK]D-Fender | j_wizworks: you sure that externip is right? |
| 16:52.10 | [TK]D-Fender | AndyGraybeal: Yeah it does... and it take 5 minutes for FreePBX to change pages on this P90 win98 machine! Can you haelp plaese?!?!?! |
| 16:52.24 | [TK]D-Fender | drmessano++ |
| 16:53.00 | hmmhesays | is one of the tza's around here I could use some blackfin advice |
| 16:53.20 | RipeR-81 | eric_hill |
| 16:53.21 | RipeR-81 | ? |
| 16:53.25 | j_wizworks | yes that 209 address is our WAN. |
| 16:53.48 | tzanger | hmmhesays: hahah yes |
| 16:54.43 | *** join/#asterisk Victor_Yure (n=Victor_Y@200.166.132.131) |
| 16:55.10 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 16:55.20 | hmmhesays | tzanger: is there some why I can find out what version of gcc was used to compile my current blackfin firmware. I'm trying to set up my devel environment |
| 16:55.23 | eric_hill | RipeR-81: sorry, had a call come up. At this point you need to trace the call and see how far it gets. |
| 16:55.26 | hmmhesays | strings tells me nothing |
| 16:55.28 | [TK]D-Fender | j_wizworks: ok, try another BV peer... |
| 16:55.35 | [TK]D-Fender | j_wizworks: mayeb NYC = DOA |
| 16:55.57 | eric_hill | RipeR-81: CCM should tell you /why/ a call didn't go through, and asterisk (console, set verbose 9) should show you a failed inbound call, or no call at all. |
| 16:56.21 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 16:56.21 | j_wizworks | ok I will try that but I used an Xlite softphone to test the acct on that BV peer and the xlite phone fired right up. |
| 16:57.01 | bsdwarrior | I want to play a message every 30 seconds when someone is in a queue. im doing Wait(30) but this cuts off the hold music |
| 16:57.16 | tzanger | hmmhesays: I already told you how, with strings on the asterisk binary that has not yet been stripped |
| 16:57.33 | hmmhesays | tzanger: if I had access to one |
| 16:58.35 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 16:58.44 | tzanger | ahh but you do |
| 16:58.58 | tzanger | build_xxx/asterisk/main/asterisk I think |
| 16:59.02 | tzanger | main may not be right |
| 16:59.19 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:59.55 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 17:00.15 | eric_hill | bsdwarrior: Instead of Wait, look at the "announcement" argument of a Queue. |
| 17:00.46 | x86 | ugh |
| 17:00.53 | bsdwarrior | eric_hill thanks. |
| 17:01.11 | bsdwarrior | eric_hill im learning, slowly |
| 17:01.18 | x86 | why can I icmp ping a peer by IP address, but chan_iax2 refuses to connect to the host with "UNREACHABLE" messages? |
| 17:01.24 | x86 | that's highly misleading, imho |
| 17:01.30 | x86 | since it's certainly reachable |
| 17:02.13 | bsdwarrior | eric_hill, that will play the message, but I want it to repeat every 30 seconds |
| 17:02.16 | x86 | any ideas how to force chan_iax2.so to connect to a host that it thinks is unreachable, but it certainly is reachable? |
| 17:02.44 | aydiosmio | How do I test for a Dial() timeout int he dialplan? Do I check ${DIALSTATUS}? I don't see a timeout return code for it |
| 17:03.18 | *** join/#asterisk kkn088 (n=kikoun@77.204.209.7) |
| 17:03.27 | Qwell | aydiosmio: it would probably be NOANSWER |
| 17:03.38 | hmmhesays | tzanger: I don't have access to the machine this firmware was built on |
| 17:03.43 | aydiosmio | thanks |
| 17:03.48 | tzanger | oh |
| 17:04.06 | tzanger | you may very well be screwed then. I am sure there is some identifying mark held somewehere in the ELF image, even stripped, but I don't know where to look |
| 17:04.14 | hmmhesays | tzanger: hence my delima |
| 17:04.17 | eric_hill | bsdwarrior: In that case, how about the "periodic-announce" feature of a regular queue. |
| 17:04.19 | hmmhesays | this image came for xorcom |
| 17:04.20 | eric_hill | bsdwarrior: http://www.voip-info.org/wiki-Asterisk+config+queues.conf |
| 17:04.25 | aydiosmio | so I can do: exten => s-NOANSWER,n,NoOp() |
| 17:04.30 | aydiosmio | in case of a timeout |
| 17:04.46 | tzafrir | hmmhesays, so ask Xorcom? |
| 17:05.12 | hmmhesays | I'm going to |
| 17:09.44 | bsdwarrior | eric_hill hmm, hope that works with the queues being in postgres |
| 17:09.54 | itguru | Is SCCP standard in an asterisk install, or do I need to add it? |
| 17:10.53 | Qwell | itguru: chan_skinny in asterisk 1.4 or higher |
| 17:11.03 | aydiosmio | if I do exten => s,n,Goto(s-${DIALSTATUS},1) and say s-BUSY does not exist in the dialplan, what happens to the call flow? |
| 17:11.12 | *** join/#asterisk ozant (n=ozanturk@85.104.1.153) |
| 17:11.21 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:12.57 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 17:13.14 | Ritzerisk | so anyways in my iaxmodem logs i get.... Unable to pass the full buffer onto the device file |
| 17:13.19 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 17:14.36 | [TK]D-Fender | aydiosmio: Go find out |
| 17:15.13 | itguru | I got 1.4.9 so, I should be okay @ Qwell |
| 17:17.23 | itguru | Qwell: I'm still lost as to how to actually get my Cisco Phone to communicate with my asterisx box :( |
| 17:17.29 | itguru | way too much reading to do |
| 17:17.53 | badcfe | my asterisk says its playing 'digits/hundred' (language 'fr'), but its _not_ reading my /var/lib/asterisk/sounds/fr/digits/hundred.alaw, just the /var/lib/asterisk/sounds/digits/hundred.alaw |
| 17:17.54 | drmessano | lol |
| 17:18.31 | badcfe | how do i get it looking into the language directory for digits (for non-digits messages it _does_) |
| 17:18.36 | drmessano | [TK]D-Fender: I am having a small problem installing Asterisk.. do I want to run the Live CD or install to Hard Drive? |
| 17:18.37 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 17:18.44 | drmessano | j/k |
| 17:19.09 | [TK]D-Fender | drmessano: with all this atlk about hard & floppy.... shouldn't we be having more fun? ;) |
| 17:19.15 | drmessano | lol |
| 17:19.22 | hades123 | I have kind of a design question guys |
| 17:19.35 | hades123 | Now asterisk works as B2BUA right |
| 17:19.47 | hades123 | can I set it up so it only handles signanling |
| 17:19.59 | drmessano | That guy that ate 4 days trying to get a PAP2 working on um.. that Green Boxes PBX thing system... yeah.. comes back.. "HOW WORK GIZMO TRUNK drmessano, halp me no?" |
| 17:20.01 | [TK]D-Fender | badcfe: /var/lib/asterisk/sounds/fr/digits/hundred.alaw should be /var/lib/asterisk/sounds/digits/fr/hundred.alaw |
| 17:20.02 | hades123 | when two internal extensions |
| 17:20.12 | hades123 | wants to talk to each other |
| 17:20.23 | Qwell | drmessano: HALP YOU YES SEND MONEYZ |
| 17:20.27 | badcfe | [TK]D-Fender: thanks, by the way this is 1.4.13 |
| 17:20.28 | drmessano | Yes lol |
| 17:20.29 | [TK]D-Fender | hades123: * is a B2BUA. Period. End of story. |
| 17:20.33 | drmessano | 4 days.. for a PAP2 |
| 17:20.54 | drmessano | Im thinking.. I can see the pages in my sleep.. WTF can he be doing wrong.. |
| 17:20.58 | drmessano | 3 days later |
| 17:21.03 | drmessano | "U SUCK, DIE" |
| 17:21.08 | badcfe | [TK]D-Fender: heh, ill make symlinks as "compatibility hack". -- or are symlinks not followed explisitly by asterisk? |
| 17:21.34 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 17:21.55 | hades123 | [TK]D-Fender : it will help in scaling the number phones, and reducing the oad immensely |
| 17:21.57 | drmessano | Sad part is.. I couldnt tell him. "Maybe hand config asterisk isnt for you... you know Fonalit....." Nope, hes was already there |
| 17:21.59 | hades123 | load* |
| 17:22.09 | drmessano | s/hes/he |
| 17:23.47 | drmessano | Now he want Gizmo "CUZ ASKERISK HALP ME MAKE FREE CALLS" |
| 17:24.09 | drmessano | The power of open source |
| 17:24.38 | hades123 | seems this topic has been discussed here alot |
| 17:24.52 | drmessano | I would seriously love an Asterisk YouTube video |
| 17:24.56 | drmessano | With umm |
| 17:24.57 | mikkel | Is it possible to buy a BRI card for ISDN connections and a TDM40B and connect Analog phones and make them ring ? |
| 17:25.08 | Qwell | mikkel: sure |
| 17:25.21 | Qwell | you need FXS modules on the TDM400p |
| 17:25.22 | drmessano | Guy getting a bill with a $20 long distance call on it.. then he installs asterisk and AT&T sends him a CHECK |
| 17:25.47 | hades123 | drmessano: What? |
| 17:25.49 | mikkel | Qwell, So Asterisk will convert between Digital and Analog ? |
| 17:25.56 | mikkel | Qwell, Yes 4 FXS on it. |
| 17:26.03 | hades123 | drmessano: Never heard about at&t giving money to anybody |
| 17:26.03 | Qwell | mikkel: it does them as two separate unrelated channels |
| 17:26.16 | Qwell | it's not so much "conversion" as it is...well...yeah, it's conversion :p |
| 17:26.20 | drmessano | hades123: A play on those that think they install Asterisk and all calls forever are suddenly "just free" |
| 17:26.27 | Qwell | basically every channel type in asterisk ends up as an "asterisk channel" in the core |
| 17:26.44 | hades123 | drmessano: LOL, didn't get the joke |
| 17:26.48 | Qwell | so yeah, you can connect many different types of things, with many different other types |
| 17:26.49 | hades123 | at first |
| 17:26.57 | Qwell | if that...answers your question |
| 17:27.09 | mikkel | Qwell, But it will be able to answer a ISDN connection and transfer it to an analog phone on the TDM40B card ? |
| 17:27.14 | Qwell | yes |
| 17:27.20 | mikkel | Qwell, Cool, thanks. |
| 17:27.29 | drmessano | IP = no phone lines = no charges from AT&T = Free calls worldwide forever YAY |
| 17:27.37 | Qwell | or a SIP device, or IAX2 device, or back out another ISDN channel...whatever, doesn't matter |
| 17:27.51 | hades123 | drmessano: at one point , this will actually happen |
| 17:27.59 | hades123 | drmessano: no dought |
| 17:28.15 | drmessano | Sure.. if everyone is using Asterisk |
| 17:28.18 | drmessano | or VoIP |
| 17:28.21 | Qwell | drmessano: give it time |
| 17:28.27 | hades123 | drmessano: it will be one charge , your internet , with it comes entertianment and phones ..etc |
| 17:28.34 | drmessano | I have no doubt it will happen |
| 17:28.55 | *** join/#asterisk ronr (n=ron@82-204-104-197.fttx.bbeyond.nl) |
| 17:29.08 | drmessano | Problem is, someone downloads Trashbox and they jump on IRC wanting to know "Y CALL NO FREE? WARE FREE CALL?" |
| 17:29.28 | drmessano | and thats the ONLY reason they googled asterisk in the first place |
| 17:30.21 | drmessano | "CAN U HALP ME? YES NO YES? I GIVE U SSH AND ROOT PASSWORD" |
| 17:30.29 | drmessano | Experience, outtolunc |
| 17:32.07 | Nugget | heh |
| 17:32.20 | aydiosmio | [TK]D-Fender: yessir, I will report back with my findings! |
| 17:32.47 | drmessano | OMG |
| 17:33.03 | drmessano | "Ok, I go now.. Let me know you want SSH remote access" |
| 17:33.07 | drmessano | NEVER |
| 17:33.09 | drmessano | DIE |
| 17:33.19 | *** join/#asterisk AndyGraybeal (n=andy@node53.34.251.72.1dial.com) |
| 17:35.34 | x86 | anyone ever mess with the Adit 600? |
| 17:36.00 | ronr | ~book |
| 17:36.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:36.06 | _ShrikE | x86: yes |
| 17:36.22 | Navion | Has anyone played with OpenVox cards? |
| 17:36.28 | Qwell | ~cheap |
| 17:36.28 | jbot | i guess cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 17:36.28 | x86 | _ShrikE: I bought one and had it shipped to a branch office... will it self-configure to a T1 just like a Rhino will? |
| 17:36.40 | drmessano | ~free |
| 17:36.41 | jbot | somebody said free was stuff might take awhile to get done |
| 17:36.56 | drmessano | ~askerisk |
| 17:36.59 | drmessano | bah |
| 17:37.01 | _ShrikE | x86: dont think there is any self configure in the adit |
| 17:37.21 | x86 | Navion: if by play you mean "take side cutters and tin snips to", then yes ;) |
| 17:38.02 | lirakis | ~lastseen DarylVoip |
| 17:38.04 | Navion | Hmmm... Well, that wasn't exactly what I ment. |
| 17:38.17 | justdave | I'm trying to set up with a new SIP provider, and when I get inbound calls from them, they come back with a different authentication username than the one we're registering with. What do I need to tweak to get the call recognized? |
| 17:38.57 | justdave | do I need the authentication name they're using as a sip [name] entry in the sip.conf or is there a field I can add to the entry for them to specify it? |
| 17:39.41 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 17:40.12 | justdave | I suspect adding their [authname] as a section to sip.conf would work but trying to avoid that if there's another way to do it because that would make things confusing to have a different name for their channels on inbound than the outbound ones, especially when their inbound 'name' is just a number that has nothing to do with their name or our phone number |
| 17:40.25 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 17:40.50 | aiksa[LV] | hi everyone. its been a while since my last visit here. |
| 17:41.18 | aiksa[LV] | today i encountered following message in asterisk CLI: Got SIP response 400 "SIP Parser Error : Unexpected 'N', line 3, column 9"Got SIP response 400 "SIP Parser Error : Unexpected 'N', line 3, column 9" |
| 17:41.35 | aiksa[LV] | could be fualty sip message? |
| 17:41.41 | aiksa[LV] | sorry, faulty. |
| 17:42.49 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-74-61.tor.primus.ca) |
| 17:43.07 | [TK]D-Fender | aiksa[LV]: Thats what its telling you.... |
| 17:43.29 | aiksa[LV] | okay, so wheteher it's Nokia's stack or 3com's |
| 17:43.54 | aiksa[LV] | i guess 3com is more likely to have caused that |
| 17:45.08 | hades123 | speaking about 3com , what do you guys thing about the 3com v3000 |
| 17:45.19 | hades123 | or 3com voip products in general |
| 17:45.51 | hades123 | please tell me it's shit |
| 17:46.34 | aiksa[LV] | urgh sorry, not the 3com. its micronet |
| 17:47.01 | aiksa[LV] | so - i am not suprised at all. taking into account any other their IP product i have ever used |
| 17:48.22 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 17:48.57 | rhombus | I hear that Digium has blocked the use of |
| 17:49.06 | rhombus | "asterisk" in Google Adwords |
| 17:50.09 | rhombus | Found out through a friend. If true, that's a sure way to destroy your business. I wonder whose idea that was? |
| 17:50.47 | AndyGraybeal | can you explain why it's a sure way to destroy your business? |
| 17:51.05 | drmessano | lol |
| 17:51.14 | drmessano | So, Digium has no rights to Asterisk then? |
| 17:51.20 | drmessano | ic |
| 17:51.58 | rhombus | AndyGraybeal: because it prevents all the partners and people who sell related services from marketing through Adwords |
| 17:52.18 | rhombus | and it destroys community goodwill |
| 17:52.28 | AndyGraybeal | interesting, thank you for explaining. |
| 17:53.02 | rhombus | This is especially true if you do something like that without 1) notifying people and 2) offering the community an alternative |
| 17:53.07 | aydiosmio | Asterisk is a Digium trademark |
| 17:53.14 | aiksa[LV] | emm, so right now partenrs are encourged to use 'wildcard' instead of 'asterisk'? |
| 17:53.23 | aydiosmio | Diguium has to protect its marks |
| 17:53.23 | rhombus | aydiosmio: Yes, yes, we've heard that |
| 17:53.35 | defswork | you can't trademark an everyday word |
| 17:53.37 | aydiosmio | authorized resellers woudl be allowed to use asterisk as a an adword |
| 17:53.47 | rhombus | aydiosmo: oh? Says who? |
| 17:53.48 | aydiosmio | asterisk isn't an everyday word |
| 17:53.54 | rhombus | aydiosmio: yes it is |
| 17:53.55 | aiksa[LV] | aydiosmio: oh, yes it is. |
| 17:53.58 | defswork | of course it is |
| 17:54.00 | aydiosmio | says normal corporate policy |
| 17:54.05 | defswork | it's in the dictionary |
| 17:54.09 | aiksa[LV] | it has been around well before digium |
| 17:54.17 | _ShrikE | the use of the word asterisk would be nearly impossible to enforce |
| 17:54.42 | aydiosmio | aiksa[LV]: That's not what defines that portion of the trademark law |
| 17:54.43 | rhombus | aydiosmio: "normal corporate policy"? Whose policy? |
| 17:54.51 | aydiosmio | lost of normal everyday words are trademarked |
| 17:54.58 | aydiosmio | lots* |
| 17:55.15 | Shaun2222 | asterisk doesnt have the ability to append to a gsm/wav file does it? |
| 17:55.15 | Uni | defswork: I beg to differ on the "<defswork> you can't trademark an everyday word" comment |
| 17:55.17 | _ShrikE | Those trademarks still dont prevent the everday use of the words |
| 17:55.29 | Uni | Apple is a registered trademark, http://www.apple.com/legal/trademark/appletmlist.html |
| 17:55.43 | Uni | and apple is a much more common word than asterisk |
| 17:55.44 | _ShrikE | so I cant offer an add selling apples? |
| 17:55.46 | aydiosmio | rhombus: when you do business with a company, it's standard procedure to be allowed use of product images and trademarks for marketing purposes |
| 17:55.49 | defswork | Uni: Windows isn't - unless you spell it with a capital W |
| 17:56.02 | aydiosmio | _ShrikE: not if it's a computer-related ad |
| 17:56.07 | rhombus | aydiosmio: are you working for Digium? |
| 17:56.21 | rhombus | aydiosmio: you are speculating |
| 17:56.24 | aydiosmio | trademark protections only apply really when companies that are inthe same business conflict |
| 17:56.27 | Uni | defswork: agreed, but I don't think the courts are case sensitive |
| 17:56.35 | rhombus | aydiosmio: what matters is Google's policy and Digium's policy |
| 17:56.36 | aydiosmio | creating a situation where a consumer would be "confused" by usage |
| 17:56.38 | *** join/#asterisk Victor_Yure (n=Victor_Y@200.166.132.131) |
| 17:56.42 | rhombus | and Google has blocked use of "Asterisk" wholesale |
| 17:56.45 | Uni | or at least, not necessarily case sensitive |
| 17:56.47 | defswork | Uni: that's the whole point - they are |
| 17:56.50 | aiksa[LV] | aydiosmio: i never said it cann not be used as trademark |
| 17:56.57 | defswork | thats why windows isn't a trademark and Windows is |
| 17:57.01 | aydiosmio | now |
| 17:57.05 | aiksa[LV] | aydiosmio: i was refering to whole google adwords thing |
| 17:57.05 | aydiosmio | let's get down to brass tacks |
| 17:57.22 | aydiosmio | what are YOU sellign that requires adwords advertising to market that has to do with asterisk? |
| 17:57.30 | aiksa[LV] | that adding apple to 'black list' for non apple affiliates would seem outright wrong, wouldnt it? |
| 17:57.43 | rhombus | aydiosmio: I am selling Asterisk consulting services, but I don't use Google Adwords |
| 17:57.53 | aiksa[LV] | aydiosmio: furthermore apple is also a treadmark of a record comapny established by beatles |
| 17:57.56 | defswork | rhombus: bad you!! |
| 17:57.57 | Uni | defswork: so your saying that the use of the word windows as the first word in a sentence would fall under that? |
| 17:57.58 | rhombus | However -- I still find this action to be self-defeating, and frankly, dumb |
| 17:58.08 | defswork | Uni: no - you need context |
| 17:58.12 | aydiosmio | aiksa[LV]: and they've beenin trademakr disputes for several years now |
| 17:58.18 | rhombus | since lots of businesses are in the Asterisk ecosystem and market their product specifically for use with it |
| 17:58.21 | aydiosmio | because they'reboth now in the music business |
| 17:58.27 | rhombus | I can think of some biggies like Aastra |
| 17:58.40 | Uni | sure, and even in a computer related ad, it would still be the same. ie. "Windows compatible hardware device for sale" |
| 17:58.41 | aiksa[LV] | aydiosmio: i am pretty aware of that. |
| 17:58.51 | *** join/#asterisk ghenry (n=ghenry@85-189-244-101.daisydsl.managedbroadband.co.uk) |
| 17:59.08 | aydiosmio | rhombus: yes and again, companies that market products specifically for Asterisk are often given explicit permission to use trademarks for marketing purposes |
| 17:59.09 | rhombus | defswork: Yes, bad me. I should be shot for trying to make a living off my experience. This is exactly why this move is a mistake. |
| 17:59.12 | aiksa[LV] | thats another problem with using common words for brands |
| 17:59.27 | rhombus | aydiosmio: You are guessing, friend. There is NO evidence that Google is making exceptions for anybody. |
| 17:59.32 | aydiosmio | Digum has prevented in the past copmanies form using the Asterisk mark to sell certain products |
| 17:59.32 | rhombus | This is a wholesale change |
| 17:59.36 | aydiosmio | it's not unreasonable |
| 17:59.42 | aiksa[LV] | aydiosmio: i guess it would even make it easier to declare it as a 'common name' in a trademark sense |
| 17:59.57 | rhombus | aiksa[LV]: bingo |
| 18:00.37 | rhombus | aydiosmio: Digium is only hurting itself by this action. |
| 18:00.50 | aydiosmio | I'm sure they feel horrible about it too. |
| 18:00.57 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 18:01.04 | rhombus | aydiosmio: they will when it impacts their sales |
| 18:01.09 | aydiosmio | it won't |
| 18:01.10 | Uni | defswork: I suppose my point all in all was that common words can indeed be trademarked, however I do agree that there remains possibility of the courts finding one way or another based on the casing of said word. |
| 18:01.11 | hades123 | hmmm .. I agree with rhombus |
| 18:01.57 | Uni | the validity of that, or of this particular alleged move by digium/google is another matter I don't think I'll weigh in on |
| 18:02.01 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 18:02.06 | rhombus | The number of businesses reselling Digium hardware through Adwords can't be counted on hands and feet |
| 18:02.11 | aydiosmio | http://www.digium.com/en/company/view-policy/5 |
| 18:02.45 | aydiosmio | read this |
| 18:02.45 | *** join/#asterisk otherwiseguy (n=otherwis@CPE-75-81-49-192.kc.res.rr.com) |
| 18:02.48 | aydiosmio | then get back with me |
| 18:02.57 | aiksa[LV] | afk 5min |
| 18:03.13 | defswork | rhombus: so sangoma for instance cannot sell "Asterisk compatible PRI" on google ? |
| 18:03.24 | rhombus | defswork: exactly |
| 18:03.28 | defswork | rhombus: I disagree |
| 18:03.33 | *** join/#asterisk ronr (n=ron@82-204-104-197.fttx.bbeyond.nl) |
| 18:03.43 | rhombus | defswork: hey, I disagree too -- but that's what's happened |
| 18:03.45 | defswork | rhombus: that's not masquerading as Asterisk or pretending to be asterisk |
| 18:03.49 | rhombus | EVERYTHING is blocked |
| 18:04.00 | defswork | rhombus: so theres no violation of anything - only a reference |
| 18:04.04 | rhombus | defswork: i know -- are you sure you're arguing with the right person? |
| 18:04.07 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 18:04.07 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:04.10 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 18:04.13 | rhombus | defswork: man, you're preaching to the choir here |
| 18:04.16 | defswork | rhombus: you can legally reference any trademark |
| 18:04.19 | defswork | sorry not arguing |
| 18:04.21 | defswork | discussing :) |
| 18:04.24 | rhombus | all I'm saying is that that's what they've done |
| 18:04.37 | x86 | _ShrikE: how do i configure it for my T1? |
| 18:04.39 | rhombus | and I'm saying it's 1. wrong, 2. a mistake, 3. will hurt them more than it helps them |
| 18:04.42 | defswork | rhombus: if that is the case then it does truly suck |
| 18:05.03 | rhombus | defswork: with respect to the Digium staff here on the channel: not the smartest thing they've ever done |
| 18:05.04 | _ShrikE | x86: I need a better idea of what you are trying to do with it exactly. |
| 18:05.21 | lirakis | isnt there a "last seen" command for jbot? |
| 18:05.22 | rhombus | and made infinitely worse by the lack of communication on the subject. No press release, no community notice, nothing. |
| 18:05.24 | x86 | _ShrikE: i'm not in front of the unit so I have no idea how to tell someone how to set it up |
| 18:05.24 | drmessano | If your business is that fragile that a lack of a google adword is going to crush it, maybe now would be a good time for self-evaluation |
| 18:05.29 | rhombus | Gee, where have we heard that before? |
| 18:05.37 | defswork | rhombus: where is this published - that google won't allow anyone other than digum to refer to asterisk ? |
| 18:06.09 | x86 | _ShrikE: I've got an Adit 600 configured with 24 FXS ports |
| 18:06.10 | defswork | they've not just paid for an exclusive word have they - nothing to do with trademarks etc.. ? |
| 18:06.17 | rhombus | There's a discussion ongoing on the biz list -- people who have been advertising using Asterisk in the ad for years have had their ads blocked with a trademark warning |
| 18:06.22 | x86 | _ShrikE: I want to connect it to an asterisk server over T1 interface |
| 18:06.45 | _ShrikE | x86: You will need someone to get into it with a console cable so you an give it an IP address. Then you can set it up remotely via telnet |
| 18:06.47 | rhombus | drmessano: my business will do fine -- I don't use adwords |
| 18:06.59 | rhombus | drmessano: but that doesn't make this idea any less stupid |
| 18:07.22 | defswork | rhombus: but according to the digum trademark policy unless you sell AsteriskManager or the like theres no violation |
| 18:07.28 | esaym | I can't seem to get a box with 1.2.13 to register with a box with 1.2.24 with iax. Did something change with iax between these two versions. The 1.2.24 box sends the auth but the 1.2.13 box sends back inval |
| 18:07.53 | rhombus | defswork: only aydiosmio thinks that using Asterisk in a Google ad is a trademark violation |
| 18:07.56 | esaym | like the password is wrong but it isn't |
| 18:08.00 | rhombus | so again, you're preaching to the choir |
| 18:08.28 | x86 | _ShrikE: what if there is no console cable? |
| 18:08.38 | rhombus | anyway, I've said my bit |
| 18:08.54 | _ShrikE | x86: screwed, or try to guess the ip thats on the unit |
| 18:08.57 | rhombus | I hope Digium wakes up and at least says something about it |
| 18:09.01 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 18:09.08 | x86 | _ShrikE: it has no default config? |
| 18:09.11 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 18:09.14 | drmessano | rhombus: then it's a pointless argument of semantics with no real world data to back up the implications |
| 18:09.19 | ajohnson | Does anyone know if app_mysql is threaded or not? |
| 18:09.27 | drmessano | I guess so |
| 18:09.48 | ajohnson | I seem to get one DB connection that hangs and everything that uses it stops executing |
| 18:10.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:10.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:10.06 | ajohnson | uses app_mysql hangs I mean |
| 18:10.43 | drmessano | Digium has to make money too.. if they want to buy an exclusive on an adword, more power to em |
| 18:10.48 | drmessano | But thats my .02 |
| 18:11.15 | *** join/#asterisk catpants (n=pREIXK@12-202-220-194.client.mchsi.com) |
| 18:11.28 | _ShrikE | x86: by default I think it maps all the fxs ports in order to the T1 interface, I dont know what the default IP is though, and you will need to get into it. |
| 18:11.39 | hades123 | from somebody who is savy in theinternet marketing business |
| 18:11.39 | _ShrikE | x86: especially if its not new |
| 18:11.54 | hades123 | I am telling you rhombus is right |
| 18:12.45 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:14.19 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 18:14.26 | *** join/#asterisk AndyGraybeal_ (n=andy@node53.34.251.72.1dial.com) |
| 18:15.33 | *** join/#asterisk vrtk (n=bb@189.21.178.20) |
| 18:17.26 | *** join/#asterisk corramor (n=corramor@216.207.245.1) |
| 18:17.44 | Shaun2222 | what source file contact the background cmd? |
| 18:17.50 | *** join/#asterisk dr0ck (n=dr0ck@216.207.245.1) |
| 18:19.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:20.18 | *** join/#asterisk AndyGraybeal_ (n=andy@node53.34.251.72.1dial.com) |
| 18:20.59 | hmmhesays | src/apps/app_background.c probably |
| 18:23.47 | tzafrir | Shaun2222, hint: grep for the help text in */*.c in the source directory |
| 18:24.27 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:25.47 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 18:26.10 | *** join/#asterisk AndyGraybeal (n=andy@node53.34.251.72.1dial.com) |
| 18:27.33 | *** join/#asterisk Woifi1988 (n=anon@M1425P023.adsl.highway.telekom.at) |
| 18:28.07 | Woifi1988 | hi |
| 18:28.31 | Woifi1988 | is there a way to build or install packets on a astlinux embedded os? |
| 18:28.33 | *** join/#asterisk hades123 (n=wqwsqww@d57-199-17.home.cgocable.net) |
| 18:29.44 | bkruse | Woifi1988: what OS are you thinking? |
| 18:30.21 | Woifi1988 | astlinux |
| 18:30.56 | bkruse | I mean, embedded device? |
| 18:31.38 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 18:31.51 | Woifi1988 | i have an image for an i386 architecture. it surely is similar to the image for wrap |
| 18:32.02 | bkruse | ahh, wrap boards, nice. |
| 18:32.19 | Woifi1988 | no its an alix |
| 18:32.24 | bkruse | ugh |
| 18:32.31 | Woifi1988 | but there is no image for my alix available |
| 18:32.47 | Woifi1988 | and wrap is too old ;-> |
| 18:32.51 | bkruse | pssh |
| 18:32.59 | bkruse | I am not sure, personally, I would just install deb minimal (if its i386 and you can) and then build asterisk |
| 18:33.11 | bkruse | or build the binaries and ship em over (just like debian packages actually do) |
| 18:33.27 | Woifi1988 | okay thats a good idea |
| 18:33.36 | Woifi1988 | i have to use astlinux |
| 18:33.42 | Woifi1988 | for my academic gradues |
| 18:33.59 | Woifi1988 | i have to try many asterisk versions on many platforms |
| 18:34.00 | bkruse | ahh, I understand |
| 18:34.13 | bkruse | to see which ones...work? or something? |
| 18:35.04 | Woifi1988 | yes to test performances and availability and also things like diffculty and time it consumpts |
| 18:35.20 | Woifi1988 | the most time I spent for the alix |
| 18:35.28 | Woifi1988 | it was really difficult |
| 18:35.59 | bkruse | Ahh, I understand. That makes sense |
| 18:36.06 | bkruse | is it a project? or just for your knowledge? |
| 18:36.15 | Woifi1988 | it's a project |
| 18:36.24 | x86 | _ShrikE: ok, got serial, but no ethernet |
| 18:36.36 | Woifi1988 | its combined wlan (with handover and qos) and voip |
| 18:36.43 | x86 | _ShrikE: there is no ethernet available in the area where my phone system is (they are all used already) |
| 18:37.07 | x86 | _ShrikE: so how do i set this thing up with just serial? |
| 18:37.09 | Woifi1988 | just have a look: www.t2u.at |
| 18:37.14 | Woifi1988 | but it's german! |
| 18:37.17 | x86 | _ShrikE: I'm having trouble finding documentation for it |
| 18:37.38 | bkruse | Woifi1988: that is interesting |
| 18:37.45 | Woifi1988 | thanks |
| 18:37.52 | aydiosmio | Is there a way to specify the name of the voicemail file to be saved with VoiceMail()? |
| 18:37.59 | Woifi1988 | but it finishes in 3 month |
| 18:38.05 | bkruse | Woifi1988: I would personally try the debian way, then you can build the binaries on another box VERY quickly with a bash script, even overnighted |
| 18:38.08 | Woifi1988 | and that i go for my final exams |
| 18:38.11 | bkruse | scp ; make install |
| 18:38.58 | Woifi1988 | bkruse: yes i tried ubuntu server. it was quite easy but i had some problems with configuring |
| 18:39.02 | esaym | anyone know why I can't register my asterisk 1.2.13 box with a 1.2.24 box using iax? |
| 18:39.03 | Woifi1988 | special with zapata |
| 18:39.16 | esaym | it acts like the password is inval but it is right |
| 18:39.17 | lmadsen | _ShrikE: FedEx just left with your box! |
| 18:39.45 | _ShrikE | !! |
| 18:39.49 | bkruse | Woifi1988: configuring what exactly? |
| 18:39.58 | lmadsen | _ShrikE: so you should be able to track it shortly |
| 18:40.33 | Woifi1988 | bkruse: i tried to install the zapata drivers and there was a problem because of the ubuntu headers |
| 18:41.16 | bkruse | Woifi1988: apt-get install kernel-headers |
| 18:41.28 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 18:42.28 | Woifi1988 | it was not that easy |
| 18:43.18 | Woifi1988 | there has been a conflict with some slinks... it was a stupid mistake but cost me much time |
| 18:43.38 | _ShrikE | x86: do a show a:1 to get the current setup on the T1 |
| 18:44.33 | _ShrikE | lmadsen: thanks |
| 18:44.34 | twisted | http://www.pcworld.com/article/id,141410-c,windowsbugs/article.html <-- bit ballsy, ey? |
| 18:44.53 | lmadsen | _ShrikE: np -- thank you! |
| 18:46.39 | *** join/#asterisk Patrickz_ (n=patrickz@ppp-124-121-58-20.revip2.asianet.co.th) |
| 18:46.53 | Patrickz_ | Hello all |
| 18:47.10 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-81.static-ip.m.telepacific.net) |
| 18:47.14 | Patrickz_ | first time here! |
| 18:47.27 | Patrickz_ | hello |
| 18:47.27 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-81.static-ip.m.telepacific.net) |
| 18:50.42 | Patrickz_ | :/ |
| 18:53.01 | [TK]D-Fender | Patrickz_: Don't expect everyone to just jump upp |
| 18:53.33 | Patrickz_ | yeah... I think so.. |
| 18:54.25 | mvanbaak | heeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeey Patrickz_ !!!!!! |
| 18:54.28 | mvanbaak | welcome ! |
| 18:55.06 | Patrickz_ | hello mvanbaak! just newbie here |
| 18:55.23 | mvanbaak | Patrickz_: dont worry. we all were once |
| 18:55.25 | Patrickz_ | never use irc almost 10 years |
| 18:55.29 | mvanbaak | cept [TK]D-Fender |
| 18:55.35 | mvanbaak | he was born a legend |
| 18:56.29 | Patrickz_ | I'm Asterisk newbie, just visit IRC community. |
| 18:57.57 | Havokmon | Welcome to the wall, flower. :) |
| 18:58.44 | Patrickz_ | I from Thailand |
| 18:59.32 | mvanbaak | twisted: this is big news as well: http://unixsadm.blogspot.com/2008/01/sun-microsystems-buys-mysql.html |
| 19:01.25 | Patrickz_ | that's big news, thanks |
| 19:01.52 | *** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk) |
| 19:03.03 | aiksa[LV] | mvanbaak: it aint a silly joke or aanything? |
| 19:03.15 | mvanbaak | no, it's true |
| 19:03.23 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 19:03.30 | mvanbaak | http://www.mysql.com/news-and-events/sun-to-acquire-mysql.html |
| 19:03.36 | mvanbaak | it's on their website |
| 19:04.07 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 19:04.13 | Yourname`` | Hello errbody. |
| 19:04.24 | aiksa[LV] | well, well |
| 19:04.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:04.36 | mvanbaak | let's hope they can beat the crap out of some f the more silly bugs in mysql |
| 19:04.46 | twisted | lol |
| 19:05.03 | twisted | they're just gonna turn it into a java bean and make it use more resources than necessary |
| 19:05.11 | brodiem | ugh wtf |
| 19:05.13 | mvanbaak | hahahahaha |
| 19:05.23 | jpsharp | twisted: That's messed up. True, but messed up. |
| 19:05.26 | twisted | lol |
| 19:05.30 | mvanbaak | rewrite mysql in java |
| 19:05.32 | aiksa[LV] | twisted: as if it did not already |
| 19:05.34 | mvanbaak | yeah, that will be fun |
| 19:05.37 | brodiem | twisted: and spill pages and pages of useless debugging info. |
| 19:06.00 | twisted | it's not useless if you know how to read it, but it is a PITA to follow the trail |
| 19:06.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:06.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:06.52 | *** join/#asterisk jblack (n=jblack@pool-71-181-145-13.sctnpa.east.verizon.net) |
| 19:07.21 | brodiem | yea but you usually can't rely on the logs it generates to find anything useful.. |
| 19:07.59 | Yourname`` | Hi. How easy is it to creat click2dial for about 10 agents? |
| 19:08.29 | aiksa[LV] | on related news tomorrow: SCSO states that they suspect mysql *might* use some of propertiary code from the unix branch they are the IP holders of |
| 19:08.48 | jpsharp | Wait, SCO is still around? |
| 19:09.01 | aiksa[LV] | jpsharp: just kidding |
| 19:09.01 | mvanbaak | aiksa[LV]: huh? SCO has money left to go to court ??? |
| 19:09.35 | aiksa[LV] | i just remebered good ol' times and the fun everyone made at them |
| 19:10.03 | jpsharp | The only useful thing about SCO was the couple of grand I made shorting the stock. |
| 19:10.04 | jpsharp | :) |
| 19:10.06 | mvanbaak | oej: you tried to call me yesterday at around 5/6 PM CEST ? |
| 19:10.06 | aiksa[LV] | SCO they might get a loan for giong to court |
| 19:10.19 | aiksa[LV] | using secret source codes as security |
| 19:10.20 | Corydon76-dig | Unlikley |
| 19:10.34 | oej | mvanbaak: No, not me |
| 19:10.42 | mvanbaak | then it was Erik ;) |
| 19:10.58 | Corydon76-dig | Given that they're likely to lose, no bank in the world is going to make that loan |
| 19:11.07 | mvanbaak | I only have 2 numbers listed under the companyname that showed up in my screen |
| 19:11.14 | KermitTheFragger | Corydon76-dig: maybe the citibank :) |
| 19:11.19 | mvanbaak | but I was busy so could not answer, and I got no mail |
| 19:11.23 | aiksa[LV] | Corydon76-dig: they put a bet on subprime though. |
| 19:11.33 | Corydon76-dig | Citi didn't make most of those loans... they only bought them |
| 19:11.54 | aiksa[LV] | when all the common sense indicated against that |
| 19:11.57 | Corydon76-dig | and if/when they can prove fraud was involved, they can reverse the transaction |
| 19:12.37 | aiksa[LV] | as a matter of fact I start liking the idea of court process financing |
| 19:13.33 | aiksa[LV] | with a margin of 300% a year, you'll even be okay with loosing every second case |
| 19:13.47 | brodiem | anyone know if there have been any updates/improvements/patches/etc to being able to use local channels as dynamic agents with queues? I.e. the ability to use Local with /n option and have the ability to transfer a call without it keeping the agent tied up? |
| 19:14.22 | aiksa[LV] | At least thats (the margin) they are giving for small unsecured loans to a persons with low credit worthness over here |
| 19:15.18 | aiksa[LV] | in other related news: Sun announces strategic partnership with Steorn which will be powering next edition of Mysql with mystical ORBO |
| 19:17.34 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 19:17.41 | jblack | How is steorn doing? |
| 19:18.47 | [TK]D-Fender | brodiem: Only way to trasfer and free up the agent is via DTMF. |
| 19:18.49 | jochien1 | just got 1.4.17 running, how do i connect it to the other asterisk server |
| 19:19.04 | [TK]D-Fender | jochien1: Go lookup "asterisk dual servers" on the WIKI |
| 19:19.16 | jblack | Same old same old. |
| 19:19.22 | jochien1 | ok |
| 19:20.27 | aiksa[LV] | jblack: same old, same old also regarding orbo |
| 19:22.02 | esaym | there is an asterisk svn right? for both 1.2 and 1.4? |
| 19:22.31 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 19:25.14 | [TK]D-Fender | BBIAB |
| 19:25.56 | *** join/#asterisk enjay5150 (n=chatzill@74.202.4.2) |
| 19:26.25 | enjay5150 | Im doing some testing with the Asterisk Appliance, and when Im using MixMonitor to record calls there is severe static on the recordings (not in the live conversation) has anyone experienced this? |
| 19:26.59 | Navion | What's the best solution for a single FXO and a single FXS at a remote site? |
| 19:27.24 | enjay5150 | TDM400 card with a single FXO and a single FXS module? |
| 19:27.53 | Navion | And a computer and asterisk and a keyboard and display... |
| 19:27.57 | Navion | Really? |
| 19:28.04 | enjay5150 | you asked... |
| 19:28.47 | mvanbaak | brodiem: asterisk trunk has this solved by giving you the possibility to monitor another device then using in your member device |
| 19:29.09 | mvanbaak | that way you can use /n and still monitor the agents device to determen 'inuse' state |
| 19:29.33 | Navion | So the Linksys SPA3102? Sipura 3000? are not viable options? |
| 19:29.44 | enjay5150 | not familiar with either. |
| 19:30.33 | mvanbaak | Navion: I haven't used any of them, but I hear good reports from all of them |
| 19:31.46 | Navion | OK, I need to be able to set up a one person office remotely. Really don't want to send a whole computer out there. |
| 19:32.23 | mvanbaak | Navion: yeah. use the linksys |
| 19:32.43 | Navion | I see the Sipura 3000 is advertized for more money than the Linksys but that might just be because they aren't being made anymore. |
| 19:33.27 | KermitTheFragger | does anybody know how to really disable the imap storage ? im still getting imap related errors with --without-imap |
| 19:33.34 | *** join/#asterisk CrazyYoss (n=luther@206.176.230.250) |
| 19:33.53 | mvanbaak | in configure ? |
| 19:33.58 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 19:34.18 | KermitTheFragger | mvanbaak: i did, but im still seeing app_voicemail_imap.c:74:21: error: imap4r1.h: No such file or directory |
| 19:34.28 | jblack | drmessano: Ping |
| 19:34.39 | mvanbaak | KermitTheFragger: the error is in configure or in make ? |
| 19:34.45 | KermitTheFragger | make |
| 19:35.18 | Patrickz_ | Anyone used Cisco AS5300? |
| 19:36.12 | fiXXXerMet | If I have the ztdummy and zaptel modules installed, I can't hear any sound from Playback(), voicemail, and what not. As soon as I rmmod those modules, I can hear the sound just fine. I saw a few occurrences of this in the mailing list, but no definite solution. Any ideas? |
| 19:38.18 | tzafrir | fiXXXerMet, when the module is loaded, what is the output of zttest ? |
| 19:38.45 | mvanbaak | KermitTheFragger: try: make menuselect |
| 19:38.52 | mvanbaak | there you can disable stuff |
| 19:38.56 | fiXXXerMet | tzafrir: Without -v, nothing - it just seems to keep going. With -v, hold on - let me run it again. |
| 19:39.22 | x86 | _ShrikE: you around? |
| 19:39.30 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 19:40.02 | tzafrir | fiXXXerMet, it's the same with and without -v |
| 19:40.08 | KermitTheFragger | mvanbaak: ugh ok that was stupid of me, thanks! |
| 19:40.10 | fiXXXerMet | hmm. |
| 19:40.22 | fiXXXerMet | How long does a pass take, tzafrir? |
| 19:40.24 | tzafrir | lsmod | grep zaptel |
| 19:40.42 | tzafrir | A few seconds, the most |
| 19:40.56 | fiXXXerMet | Then nothing. When I kill it (ctrl+c), it was "results after 0 passes" |
| 19:41.04 | *** join/#asterisk RoyK (n=roy@91.149.17.65) |
| 19:41.24 | fiXXXerMet | tzafrir: http://pastebin.com/m1afc6305 |
| 19:41.41 | *** join/#asterisk mgaal (n=Mike@c-24-5-165-3.hsd1.ca.comcast.net) |
| 19:41.58 | mgaal | hello friends, i have a query |
| 19:42.21 | tzafrir | fiXXXerMet, cat /proc/zaptel/* |
| 19:42.35 | mvanbaak | KermitTheFragger: ur welcome |
| 19:42.56 | fiXXXerMet | tzafrir: Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: RTC) 1" |
| 19:42.58 | mgaal | i need a hosted service that will provide me with a) a 1-800 number b) the ability to dial in to it, put in a code, and let the person calling be able to call anywhere through that number on my dime - basically a glorified hosted calling card |
| 19:43.13 | mgaal | anyone know of anything that fits the bill? |
| 19:43.50 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:47.04 | CrazyYoss | mgaal: I dont know of any service, sorry |
| 19:47.20 | mgaal | GADGETTTTT |
| 19:47.40 | *** join/#asterisk magumbade (n=magumbad@ppp-82-135-10-85.dynamic.mnet-online.de) |
| 19:48.30 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:48.30 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:51.49 | aiksa[LV] | leaving, bye |
| 19:52.26 | fiXXXerMet | tzafrir: Any other ideas? |
| 19:52.55 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 19:53.25 | *** join/#asterisk hi365_w (n=hi365@mail.pcgeula.co.il) |
| 19:53.30 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 19:54.40 | Toerkeium | hello everyone. Does anyone knows a good/cheap sip+gsm gateway? |
| 19:55.19 | mvanbaak | Toerkeium: good and cheap dont go together |
| 19:55.43 | Toerkeium | :) |
| 19:55.52 | Toerkeium | a reasonable cheap one? and reasonable good? |
| 19:56.23 | mvanbaak | we use the 2CN VoiceBlue with success |
| 19:56.42 | Toerkeium | how many chips support the 2CN VoiceBlue? |
| 19:56.49 | mvanbaak | 4 |
| 19:57.00 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 19:57.02 | Toerkeium | ahh, each one? |
| 19:57.22 | mvanbaak | no, they have 2 and 4 SIM models last time I checked |
| 19:57.37 | mvanbaak | oh, you can also have a look at the GSM cards from junghanns.net |
| 19:57.39 | tzafrir | fiXXXerMet, what version of Zaptel is it? |
| 19:57.47 | tzafrir | What kernel version? |
| 19:57.54 | mvanbaak | you put them in your asterisk machine |
| 19:58.05 | ManxPower | Does anyone know any dialplan differences between an attended/supervised transfer and a blind/unsupervised transfer? |
| 19:58.08 | Toerkeium | thanks mvanbaak, gonna check that |
| 19:58.37 | mvanbaak | ManxPower: they both can be done on the phone ? |
| 19:58.43 | [TK]D-Fender | ManxPower: as in? |
| 19:58.43 | mvanbaak | no need for a dialplan |
| 20:00.53 | _ShrikE | _x86: Im back |
| 20:01.02 | ManxPower | [TK]D-Fender: call comes into the operator/receptionist. If the receptionist does a supervised transfer (on the polycom) and the destination does not answer the call goes into their voicemail. If the receptionist does a blind transfer, the call times out to the directory. |
| 20:01.26 | ManxPower | I'm trying to find out what is different about the two types of transfer from the dialplan standpoint. |
| 20:01.34 | [TK]D-Fender | ManxPower: shouldn't be any at all |
| 20:01.42 | ManxPower | [TK]D-Fender: Well, that's what *I* thought. |
| 20:01.49 | [TK]D-Fender | ManxPower: 1st guess : She's jsut doing it wrong |
| 20:02.01 | ManxPower | [TK]D-Fender: not as far as we can tell. |
| 20:02.16 | [TK]D-Fender | ManxPower: get'em Eagle-eye! |
| 20:02.33 | ManxPower | This has been going on for weeks. Thought it was a race condition because sometimes it happened and sometimes it doesn't. turns out different receptionists do blind or supervised. |
| 20:02.52 | [TK]D-Fender | ManxPower: look at how "Directory" gets called in your setup... |
| 20:03.01 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:03.01 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:03.23 | x86 | _ShrikE: /query for a minute? |
| 20:03.49 | ManxPower | [TK]D-Fender: It gets called if the receptionist does not answer the original call. It's LOOKING like the call is falling back to the original extension's dialplan |
| 20:04.05 | Yourname`` | Is this a good format in manager.conf for "permit=127.0.0.1/255.255.255.0,65.22.33.123,85.43.123.128"?? |
| 20:04.27 | [TK]D-Fender | ManxPower: She's probably punching in the target exten and HANGING UP like old key-system reflexes have you do |
| 20:05.35 | ManxPower | [TK]D-Fender: I guess when I told her to press BLIND (she didn't know about the button) she could be doing that contrary to every other transfer this person has ever done. |
| 20:06.16 | [TK]D-Fender | ManxPower: if the reports of this happening are constant, then watch her do it. |
| 20:07.02 | ManxPower | It's constant only if they are using polycom Blind button during a transfer. |
| 20:07.20 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:07.41 | ManxPower | the thing is, why is the call continuing in the dialplan on the receptionist's extension? |
| 20:07.58 | [TK]D-Fender | ManxPower: SIP debug & CLI.... |
| 20:08.05 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 20:08.34 | ManxPower | [TK]D-Fender: you want to help me debug the sip debug? |
| 20:09.01 | ManxPower | I really need to wait until the system is not busy or it will be hell to try to figure it out. |
| 20:09.41 | ManxPower | as you know my macros are already spagetti code. |
| 20:10.47 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 20:10.47 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 20:11.21 | waverly360 | Hey guys, have any of you ever run into an issue on a pri where inbound calls throw a "Ring requested on channel 0/3 already in use on span 1", but outbound calls work just fine? I have 22 b channels available on my pri, but when I get about 6 of them in use, any inbound calls just get a busy signal. |
| 20:11.23 | *** join/#asterisk AndyGraybeal (n=andy@node53.34.251.72.1dial.com) |
| 20:13.38 | ManxPower | waverly360: Are you using g or G as your group indicator i.e. G1 or g1 |
| 20:14.02 | ManxPower | and when calls come in from the telco are they coming starting at the lowest numbered channel or the highest numbered channel? |
| 20:14.32 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 20:15.44 | fiXXXerMet | Are there any extra configuration steps for Asterisk when using ztdummy? |
| 20:15.53 | waverly360 | ManxPower: That entire PRI is setup in group 2, so I'm using g2 as the group. |
| 20:16.09 | waverly360 | ManxPower: It looks like all inbound calls start at the lowest numbers |
| 20:16.44 | ManxPower | waverly360: use G2 then |
| 20:17.03 | waverly360 | ManxPower: That starts them from the other end? |
| 20:17.03 | [TK]D-Fender | ManxPower: Hey remember htat issue where running ztdummy you get no audio? (no cards in system) and rmmod-ing it solves that. Do you know how to get it to WORK? |
| 20:17.29 | ManxPower | People will tell you that glare cannot happen on PRIs, but the way Asterisk handles channels on PRIs makes glare possible |
| 20:18.14 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 20:18.19 | ManxPower | [TK]D-Fender: I vaguely recall it has to do with a weird kernel RTC htz setting. |
| 20:18.20 | *** join/#asterisk ddunavant (n=David@68-244-175-48.area3.spcsdns.net) |
| 20:18.25 | jpsharp | I hate when my circuits glare at me. |
| 20:19.57 | *** join/#asterisk phocus (n=phocus@www.healthtech.net) |
| 20:20.18 | phocus | i am trying to telnet to asterisk, but i dont know wich account it wasnt me to log in with |
| 20:20.37 | jblack | Hey guys. Look what I found: http://www.scdlink.com/Details.cfm?ProdID=2789&category=23&cf=fr |
| 20:20.47 | jblack | That looks perfect to hook up to a * system |
| 20:21.10 | phocus | does anyone know how to figiure out wha account it is using |
| 20:21.17 | phocus | i am trying to write a caller ID plugin for MCE |
| 20:21.42 | jblack | phocus: you can figure out what user asterisk is running as by typing "ps aux | grep asterisk" |
| 20:21.42 | ManxPower | phocus: ssh into the server, su to root, ps -aux | grep asterisk, look at the userid of the process. |
| 20:21.52 | phocus | k |
| 20:21.56 | ManxPower | but really, that is a LINUX question not an ASTERISK question |
| 20:22.52 | phocus | i thought it had unerlying accounts it used for remote communication |
| 20:23.51 | CrazyTux[m] | Does anyone know if * lets me change timezones (GMT) for specific contexts/mailboxes? |
| 20:23.56 | jpsharp | Yes. |
| 20:24.09 | jpsharp | the tz option in voicemail.conf |
| 20:24.23 | CrazyTux[m] | jpsharp, I just specify the offset i.e. -8 or? |
| 20:24.37 | phocus | it says its runnig as a user asterisk, what is the default pw ? |
| 20:25.04 | AndyGraybeal | jblack: haha that site gave me an interesting idea.... somehow hook up the security system so if there is an intruder in the building, the security system "tracks" the intruder through the building, and "rings" the phones that he is near.... so if he walks through the building/buildings, the phones he walks past ring.... and if he picks it up it says something about the police are on their way.. etc. |
| 20:25.28 | jpsharp | CrazyTux[m]: You can specify a timezone. I have tz=central and tz=pacific in mine. |
| 20:25.33 | jblack | AndyGraybeal: Sure, that's doable. IR detection devices. |
| 20:25.39 | jpsharp | For users around the US. |
| 20:25.43 | CrazyTux[m] | jpeeler, ah, from /usr/share/zoneinfo |
| 20:26.12 | AndyGraybeal | jblack: that would be crazy to be some kinda theif... and walk through a building and every phoen that your in proximity with rings |
| 20:26.30 | jblack | I think it's not worth the effort, imho. |
| 20:26.43 | AndyGraybeal | jblack: yea yea, i guess i was having fun with my thoughts. |
| 20:26.44 | jpsharp | It'd be fun, though. |
| 20:27.52 | jblack | Sure, fun. |
| 20:28.10 | jblack | More useful would be to ring _every_ extension in the joint, all at once. |
| 20:28.33 | jblack | And the employees, and the police. |
| 20:28.52 | jpsharp | And the intercoms. |
| 20:29.37 | AndyGraybeal | jpsharp: intercoms, :) |
| 20:29.44 | AndyGraybeal | jblack: employees homes, yes nice! |
| 20:29.49 | AndyGraybeal | very fun thoughts. |
| 20:30.00 | jblack | Here's something I bet you would love doing, that would be really useful. Setup snmp in your system, and if a system disapears, gets overly loaded, the drives fill up, etc, have * call you with the last known stats for the troubled machine. |
| 20:30.32 | AndyGraybeal | nice that is awesome |
| 20:30.38 | mvanbaak | jblack: we already done that :) |
| 20:30.50 | jpsharp | You could have nagios drop a .call file into * |
| 20:31.02 | jblack | mvanbaak: I'm sorry. I meant to address that to andrygraybeal. Sorry for confusing you. |
| 20:31.42 | AndyGraybeal | i made my first asterisk call today from one xlite phone to another xlite phone on my machine :) |
| 20:31.47 | mvanbaak | jpsharp: or let it connect to the AMI to originate a call |
| 20:31.59 | jpsharp | mvanbaak: Either way. |
| 20:32.03 | [TK]D-Fender | fiXXXerMet: ManxPower>[TK]D-Fender: I vaguely recall it has to do with a weird kernel RTC htz setting. <---- |
| 20:32.46 | [TK]D-Fender | AndyGraybeal: And after being here for only what.... a YEAR? ;) |
| 20:33.02 | *** join/#asterisk _Vile (n=vile@208.100.152.234) |
| 20:33.10 | AndyGraybeal | [TK]D-Fender: yes yes, that is correct |
| 20:33.18 | jpsharp | Some people are just late bloomers. |
| 20:33.31 | jblack | Most people bloom before they're gray. |
| 20:33.34 | AndyGraybeal | now to get xlite to dial my phone connected to the fxo |
| 20:33.50 | Yourname`` | Does anyone have any experience with : http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script? |
| 20:36.30 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 20:36.54 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:38.21 | waverly360 | ManxPower: I'm sorry..I got pulled away from my pc for a bit. I just did a lookup on glare. That sounds like something that would happen intermittently..but this wasn't intermittent. |
| 20:38.55 | waverly360 | ManxPower: It was happening constantly, and I couldn't receive any inbound calls. I ended up restarting asterisk, and now that problem seems to have gone away. |
| 20:39.25 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 20:40.27 | jblack | drmessano: drmessano, wherefore are thou, drmessano? Deny thy telivision and refuse thy commercials |
| 20:40.31 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:40.31 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:40.37 | jblack | whoah! I summoned him! |
| 20:40.53 | lmadsen | o.O |
| 20:41.17 | jblack | Not you. :) |
| 20:41.29 | hi365_w | file ping |
| 20:41.39 | *** join/#asterisk d-k-t (n=dt@60.176.192.94) |
| 20:41.55 | AndyGraybeal | lmadsen: your name is in my asterisk book!! |
| 20:42.05 | AndyGraybeal | it's on the xlite phone config i'm looking at right now! |
| 20:42.08 | lmadsen | AndyGraybeal: w00t! :) |
| 20:42.12 | AndyGraybeal | :) |
| 20:42.20 | lmadsen | jblack: darn, I felt special there for a second |
| 20:42.23 | jblack | hi365_w: 10582 bytes from /etc/passwd: icmp_seq=1 ttl=1 time=0.00ms |
| 20:42.39 | enjay5150 | Im doing some testing with the Asterisk Appliance, and when Im using MixMonitor to record calls there is severe static on the recordings (not in the live conversation) has anyone experienced this? |
| 20:42.53 | jblack | lmadsen: You are special. Your very words are within 2 feet of my body, approximately 15 hours a day, six days a week. |
| 20:43.12 | jblack | In the _very_ high ranking spot of "right next to my coffee cup". |
| 20:43.13 | AndyGraybeal | lol |
| 20:44.24 | *** join/#asterisk ghenry (n=ghenry@85-189-244-101.daisydsl.managedbroadband.co.uk) |
| 20:44.30 | jblack | Considering the volume and the quality of my library, you should consider that high praise. |
| 20:45.17 | jblack | So, I summoned drmessano, but he's mute. /me tries another strategy |
| 20:45.24 | jblack | drmessano-LT: Speak, boy! Speak! |
| 20:45.28 | lmadsen | jblack: :) |
| 20:45.37 | fiXXXerMet | [TK]D-Fender: A weird kernel RTC htz setting? |
| 20:45.50 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 20:45.50 | lmadsen | and you can all have your very own autographed copy for the low low low price of $257.93 |
| 20:46.02 | AndyGraybeal | :) |
| 20:46.28 | jblack | lmadsen: If only I could afford it these days. So far, I've spent well over $700 to bilk verizon out of $30 a month. |
| 20:46.40 | lmadsen | lol |
| 20:46.43 | lmadsen | don't ya love that? :) |
| 20:46.56 | jblack | I have to admit, yes. |
| 20:46.58 | waverly360 | whois anthm |
| 20:46.59 | lmadsen | hence my reason for paying $25 a month to Rogers for an HD PVR... |
| 20:47.05 | waverly360 | dah... |
| 20:47.08 | lmadsen | whoisn't anthm |
| 20:47.08 | waverly360 | mistake :) |
| 20:47.22 | jblack | Certainly, though, * is pure crack. You start off with a softphone. Soon, you have softphones on all the computers in your house. |
| 20:47.42 | mvanbaak | indeed |
| 20:47.43 | jblack | Then, you need to get your fix by getting an ATA and more phones. Which leads to running more lines. |
| 20:47.45 | lmadsen | jblack: it really is... that's why I've been doing it for nearly 6 years now :) |
| 20:47.46 | mvanbaak | even on the PSP |
| 20:47.47 | *** join/#asterisk ZX81_ (n=ZX81@202.49.106.158) |
| 20:47.48 | mvanbaak | :) |
| 20:47.51 | *** join/#asterisk bmg505 (n=leon@196.209.183.100) |
| 20:47.54 | jblack | Before you know it, you're mainlining polycoms. |
| 20:47.54 | lmadsen | heh |
| 20:48.03 | lmadsen | my cell phone has a SIP client on it :) |
| 20:48.10 | lmadsen | and I have 7 hard phones on my desk |
| 20:48.15 | jblack | 7? |
| 20:48.19 | lmadsen | yes |
| 20:48.27 | lmadsen | actually... 6 now.. I sold one to _ShrikE :) |
| 20:48.27 | jblack | Oh, for testing and debugging. |
| 20:48.29 | file | eh? |
| 20:48.51 | lmadsen | actually 5.5 phones... the Cisco 7912 is barely a phone |
| 20:50.16 | nhuisman_work | neither is the 7910 |
| 20:51.14 | nhuisman_work | i use a cell phone, screw having 15 phones in the house. |
| 20:51.20 | [koss] | do polycoms support VLANs? |
| 20:51.27 | lmadsen | pretty sure they do |
| 20:51.44 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 20:51.57 | enjay5150 | yes they do. |
| 20:52.08 | jblack | nhuisman_work: I don't want to pay $50 a month just for one phone. |
| 20:52.08 | lmadsen | jblack: yah -- 1 phone for a company I consult for, another phone for my own PBX, and 3-4 phones at any time for consulting gigs (setting up clustered environments and such) |
| 20:52.20 | *** join/#asterisk iulius_ (n=iulius@mail1.technologieshq.com) |
| 20:52.30 | nhuisman_work | so you don't have a cell phone then? |
| 20:52.36 | jblack | Nope. |
| 20:53.06 | nhuisman_work | Yeah I'm to used to having instant phone anytime. |
| 20:53.35 | jblack | About three years ago, I cut back on my verizon habit. Canceled the $70/mo cell phone. Cancled the $60/mo aircard. Terminated the $30/mo local phone service. Cut the $120/mo DSL to $40 a month. |
| 20:53.59 | jblack | I went from $280 a month down to $40 a month. |
| 20:54.12 | jblack | No, that was closer to 2 years ago. |
| 20:54.38 | nhuisman_work | what's an aircard? |
| 20:54.57 | jblack | That's a pcmcia card that provides interwebs access over the cellular network. |
| 20:55.01 | nhuisman_work | oh.. |
| 20:55.20 | jblack | Think of it as crappy dsl, with 1500ms latency. |
| 20:55.22 | nhuisman_work | yeah I just have my $55 cell, cable i split with a few other people so that's like $10 |
| 20:55.39 | jblack | Dont' get caught "sharing" cable. |
| 20:55.52 | nhuisman_work | not like that |
| 20:55.53 | jblack | You're not screwing with the riaa. They jail people that catch doing that. |
| 20:56.00 | nhuisman_work | we have cable internet and cable in our house |
| 20:56.03 | nhuisman_work | and more then one person lives there |
| 20:56.26 | nhuisman_work | i'm pretty sure they can go fuck themselves in that situation. |
| 20:56.47 | jblack | In one house? I imagine you're fine. |
| 20:56.56 | nhuisman_work | yep |
| 20:56.58 | nhuisman_work | one house |
| 20:57.11 | nhuisman_work | my gf and I + a roommate in the extra room. |
| 20:57.58 | mvanbaak | I have wires running to 2 neighbour houses |
| 20:59.36 | nhuisman_work | hehe |
| 20:59.42 | nhuisman_work | in college I bought dsl in the dorms |
| 20:59.50 | nhuisman_work | and then ran lines to two other dorms |
| 21:00.01 | nhuisman_work | ran some packet shaping stuff and gave them a slice |
| 21:01.08 | *** part/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 21:02.01 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:07.18 | hi365_w | anyone working woth polycom's? my new 650's keep getting stuck at "Checking Application" |
| 21:08.27 | *** join/#asterisk AndyGraybeal (n=andy@node178.34.251.72.1dial.com) |
| 21:09.34 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
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| 21:10.34 | mvanbaak | nhuisman_work: yeah. my openbsd altq setup is shaping stuff nicely |
| 21:11.18 | *** join/#asterisk cheGGo (n=snafu@dslb-088-068-103-079.pools.arcor-ip.net) |
| 21:11.31 | cheGGo | hi there |
| 21:12.38 | fiXXXerMet | [TK]D-Fender: Interesting, because I am getting lots of rtc: lost some interrupts at 1024Hz. in dmesg |
| 21:16.20 | [TK]D-Fender | fiXXXerMet: should be 1000hz, not 1024 |
| 21:16.41 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 21:16.42 | [TK]D-Fender | fiXXXerMet: that appears to be the issue |
| 21:16.47 | fiXXXerMet | Is that the RTC setting you mentioned? |
| 21:16.56 | [TK]D-Fender | fiXXXerMet: Yes |
| 21:19.56 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 21:22.29 | fiXXXerMet | [TK]D-Fender: Do I have to recompile the kernel to fix that, or something? |
| 21:22.38 | ManxPower | hi365_w: chances are the sip.ld on your server is not compatible with the 650 |
| 21:22.39 | fiXXXerMet | Guess I should direct my question to #ubuntu. |
| 21:22.43 | *** join/#asterisk wishbone__ (n=wishbone@189.70.22.252) |
| 21:22.47 | [TK]D-Fender | fiXXXerMet: Yes, you should |
| 21:22.53 | fiXXXerMet | Thank you. |
| 21:23.43 | *** part/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 21:24.42 | [TK]D-Fender | ok, heading home... |
| 21:24.43 | [TK]D-Fender | bbiab |
| 21:24.53 | JayTee52 | good luck with Ubuntu, that channel is usually jam packed and scrolls at lightspeed |
| 21:26.03 | fiXXXerMet | yeah :( |
| 21:26.06 | fiXXXerMet | And they're pointing me back to here. |
| 21:27.18 | wishbone__ | hi all., can I use asterisk in a telecom company to play like a telephony switch ? |
| 21:27.31 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 21:29.00 | mocker | Guh, I hate reading through sip debug logs |
| 21:29.38 | cheGGo | who not? ;P |
| 21:30.33 | JayTee52 | fiXXXerMet, what version of Ubuntu are you running Asterisk on? |
| 21:30.42 | fiXXXerMet | The most recent, JayTee52 |
| 21:31.00 | fiXXXerMet | 7.10 |
| 21:31.00 | JayTee52 | 7.10 Gutsy Gibbon? Server or Desktop |
| 21:31.04 | fiXXXerMet | Server. |
| 21:31.40 | JayTee52 | there is a RT kernel image available in the repositories. |
| 21:32.12 | fiXXXerMet | linux-image-2.6.22-14-rt ? |
| 21:32.18 | JayTee52 | yes |
| 21:35.02 | *** join/#asterisk trippss (n=sean@72.20.150.196) |
| 21:35.05 | JayTee52 | I was running Asterisk 1.2 on Ubuntu but we just migrated to 1.4 on a new Dell PowerEdge Quad Core Xeon that came with 64 bit RHEL 5 |
| 21:36.05 | lmadsen | mmmm |
| 21:36.15 | lmadsen | that's what I run too, but s/RHEL 5/CentOS 5 |
| 21:36.24 | lmadsen | Dell PowerEdge 2950 |
| 21:36.27 | JayTee52 | 64 bit? |
| 21:36.35 | lmadsen | ya, but running 32bit OS |
| 21:36.42 | lmadsen | (64 bit stuff wasn't stable at install) |
| 21:37.02 | JayTee52 | ah, this came with RHEL 5 64 bit and get this.....Gnome installed by default. |
| 21:37.09 | *** join/#asterisk itguru (n=gabriel@5ac302c9.bb.sky.com) |
| 21:37.40 | *** join/#asterisk AJaymn (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 21:37.46 | lmadsen | hawt |
| 21:38.02 | lmadsen | wouldn't have mattered what was isntalled -- I would have reinstalled it (even if I was gonna use RHEL 5 again) |
| 21:38.29 | JayTee52 | if you want linux factory installed by Dell so you get their support for it (my boss did, I didn't care) it only comes with 64 bit. I had a bit of trepidation but I compiled the libpri, zaptel and Asterisk and she's up and running like a top. |
| 21:38.31 | fiXXXerMet | JayTee52: Should I recompile zaptel or asterisk after installing linux-image-2.6.22-14-rt ? |
| 21:38.39 | JayTee52 | yep |
| 21:38.45 | fiXXXerMet | Both or just *? |
| 21:38.50 | lmadsen | JayTee52: yah, I have my stuff running on 64bit here too at home |
| 21:39.32 | AJaymn | If you had a choice of SIP or IAX from a provider what one would you use? and why is one better over the other? |
| 21:39.55 | JayTee52 | fiXXXerMet, I'd backup your /etc/asterisk/ config files and then recompile all of it then copy your configs back into /etc/asterisk |
| 21:40.30 | fiXXXerMet | aye, ok. thanks. |
| 21:40.44 | JayTee52 | If I'm running Asterisk and my provider offers IAX I'd go with IAX. |
| 21:41.13 | AJaymn | just curious.. whats the benifet? |
| 21:41.42 | JayTee52 | IAX trunks, SIP doesn't trunk in a real sense, just kind of a kludged sense |
| 21:41.55 | nhuisman_work | JayTee52, how so |
| 21:42.24 | jwh | IAX on asterisk == pain |
| 21:42.47 | jwh | try dealing with 2000 calls/minute over IAX ;) |
| 21:42.49 | AJaymn | well ive been having issues using trying to use IAX to Vitelity.. is i use SIP to them i have less nasty call-ness ;) |
| 21:44.07 | JayTee52 | nhuisman_work, I'd go into more detail as I understand it but it's quittin time for me so I gotta scoot. |
| 21:44.14 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:44.18 | JayTee52 | later all |
| 21:44.21 | nhuisman_work | kk |
| 21:45.47 | ZX81_ | weird why did xchat flash at me :) |
| 21:45.51 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 21:46.31 | tzafrir | because you were at a different desktop? |
| 21:46.44 | *** part/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 21:48.46 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:51.41 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 21:52.59 | adeel | hi, i'm having a very trivial problem, but i can't seem to figure it out....where do i define the incoming call handling? extensions.conf? |
| 21:55.01 | [TK]D-Fender | adeel, ALL call handling = extensions.conf |
| 21:55.20 | adeel | that's what i thought... |
| 21:57.15 | *** part/#asterisk ozant (n=ozanturk@85.104.1.153) |
| 22:00.52 | bsdwarrior | periodic_announce_frequency im assuming is in seconds ? |
| 22:01.18 | [TK]D-Fender | bsdwarrior, yes |
| 22:02.18 | bsdwarrior | I can't get it to play the message. |
| 22:04.02 | bsdwarrior | im using realtime queues and I set periodic_announce and periodic_announce_frequency in the db |
| 22:04.16 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 22:04.21 | *** part/#asterisk itguru (n=gabriel@5ac302c9.bb.sky.com) |
| 22:04.58 | *** part/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 22:06.38 | bsdwarrior | tkd-fender, I cant get this to play the periodic message. |
| 22:07.06 | [TK]D-Fender | bsdwarrior, Yes its incredible. Almost like I didn't hear you say that 5 minutes ago! |
| 22:07.52 | bsdwarrior | tkd-fender, sorry man |
| 22:08.33 | drmessano-LT | lol |
| 22:14.46 | enjay5150 | Im doing some testing with the Asterisk Appliance, and when Im using MixMonitor to record calls there is severe static on the recordings (not in the live conversation) has anyone experienced this? |
| 22:15.25 | jblack | drmessano-LT: Look what I found for us: http://www.scdlink.com/Details.cfm?ProdID=2789&category=23&cf=fr |
| 22:20.55 | bsdwarrior | I can't figure out for the life of me how asterisk is reading the queue_table from a database |
| 22:21.30 | bsdwarrior | its commented out in extconfig.conf |
| 22:21.42 | bsdwarrior | ;queues => odbc,asterisk |
| 22:23.31 | *** part/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 22:23.36 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 22:31.20 | *** join/#asterisk gene2 (n=vasya@ool-4350bce8.dyn.optonline.net) |
| 22:31.37 | gene2 | has something happened to svn? |
| 22:33.55 | gene2 | anyone here? |
| 22:34.55 | tzafrir | Everyone were eaten by the Subversion Monster |
| 22:35.17 | gene2 | oh |
| 22:35.19 | gene2 | that monster |
| 22:35.21 | gene2 | i better run |
| 22:35.46 | tzafrir | yeah, it's currently recovering from some unplanned maintinance... |
| 22:35.59 | gene2 | i see |
| 22:37.28 | *** join/#asterisk Giofe (n=chatzill@201.230.177.77) |
| 22:39.42 | wishbone__ | hi all, please somebody break me a leg! Can I use asterisk in a telecom company to play like a telephony switch ? |
| 22:39.56 | jblack | Yes. |
| 22:40.29 | wishbone__ | jblack, can u tell some good reading about it? |
| 22:40.35 | [TK]D-Fender | ~book |
| 22:40.36 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:41.34 | wishbone__ | something about how to build it? |
| 22:41.44 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 22:45.24 | *** join/#asterisk Dayver (n=user@rrcs-76-79-65-114.west.biz.rr.com) |
| 22:45.28 | Dayver | does anyone has a list / site of main US SIP providers ? |
| 22:45.35 | [TK]D-Fender | wishbone__, Go read the book |
| 22:45.53 | [TK]D-Fender | Dayver, WIKI has a large list |
| 22:46.07 | Dayver | Thanks |
| 22:49.23 | *** part/#asterisk RoyK (n=roy@91.149.17.65) |
| 22:50.14 | mvanbaak | zzzz time |
| 22:58.39 | drmessano-LT | I R USE AKERIST? |
| 22:58.59 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 22:59.30 | lmadsen | Shaun2222: yes, there is an option to dial for GoSub() |
| 22:59.40 | lmadsen | Shaun2222: just look at the options and you would have seen it |
| 22:59.54 | putnopvut | It's the 'U' option |
| 23:00.07 | lmadsen | putnopvut: shhhhhhhhh :) |
| 23:00.14 | Shaun2222 | haha |
| 23:00.28 | putnopvut | I agree though, it's good to check the options before asking. |
| 23:00.36 | Shaun2222 | i've been looking at VOIP's docs... guess i should be looking at 'core show application dial' |
| 23:00.45 | putnopvut | Yep, that's how I got the answer. |
| 23:00.48 | *** join/#asterisk jburbage (i=jburbage@dhcp-64-58-3-6.mho.net) |
| 23:00.53 | Shaun2222 | even though that doesnt show me the option |
| 23:00.59 | Shaun2222 | is the U option only in trunk |
| 23:01.02 | putnopvut | Yes. |
| 23:01.20 | Shaun2222 | ok, does trunk have any major issues... am i going to regret bumpin up to it? |
| 23:01.37 | Shaun2222 | oh oh oh... does background work in gosub?!@!!!@ |
| 23:01.52 | Shaun2222 | backgroun/waitexten broken with dial+macro's |
| 23:03.07 | lmadsen | Shaun2222: there is warning not to use trunk in production -- use at your own risk |
| 23:07.01 | Shaun2222 | ya sure that warning always exists |
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| 23:09.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:10.23 | jburbage | anyone here who can answer some questions about call queues? |
| 23:11.08 | Shaun2222 | i can answer that question... yes |
| 23:11.10 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 23:11.51 | jburbage | I read that AgentCallbackLogin will be deprecated soon, so I tried to design my queue as described in the queues-with-callback-members.txt file |
| 23:12.06 | Shaun2222 | hmm trunk is bitching about a few app, chan and func modules that exist it didnt install... |
| 23:12.10 | Shaun2222 | where these ditched? |
| 23:12.21 | Shaun2222 | app_hasnewvoicemail.so, app_lookupblacklist.so |
| 23:12.27 | lmadsen | Shaun2222: /usr/lib/asterisk/modules/ |
| 23:12.30 | jburbage | but I also want to use FOP (asternic) to monitor the queue, and it won't recognize the agent login if it's loggint into Local/${EXTEN}@agents |
| 23:12.44 | lmadsen | rm -f /usr/lib/asterisk/modules/* && make install |
| 23:12.48 | Shaun2222 | lmadsen: ya but some of these look important.... like func_moh.so |
| 23:12.58 | lmadsen | they will be reinstalled |
| 23:13.02 | Shaun2222 | ok |
| 23:13.03 | lmadsen | but the right modules this time |
| 23:13.14 | lmadsen | assuming you selected them in menuselect |
| 23:13.17 | lmadsen | make a backup first |
| 23:13.31 | Shaun2222 | menuselect? |
| 23:13.34 | Shaun2222 | is that new? |
| 23:13.40 | Shaun2222 | i just ran a ./configure && make && make install |
| 23:13.40 | lmadsen | (as you should always do when someone you don't know on IRC tells you to remove all the files in a directory) |
| 23:13.44 | Shaun2222 | no that stuff isnt there... |
| 23:13.50 | lmadsen | menuselect is new in 1.4... so... over a year old |
| 23:13.58 | Shaun2222 | lmadsen: i have the old source a can just do a make install on it |
| 23:14.05 | lmadsen | ./configure && make menuselect && make install |
| 23:14.10 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 23:14.19 | jburbage | hey, I've met lmadsen, I'll delete whatever he tells me to >.> |
| 23:14.21 | Shaun2222 | didnt know about that... guess i'll check it out |
| 23:14.31 | jburbage | shaun2222: did you see my question? |
| 23:14.36 | lmadsen | jburbage: that lmadsen guy is bad news |
| 23:14.45 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 23:14.47 | lmadsen | Shaun2222: you should read TFoT2 |
| 23:15.45 | Shaun2222 | app_hasnewvoicemail.so isnt in the list, my guess is it was removed or combine with app_voicemail.c |
| 23:16.07 | [hC] | god damn... these vendors are killing me. its really hard to provide solid voip services when phones crash and fuck up all the time |
| 23:16.26 | [hC] | This polycom slowdown and eventual reboot when monitoring +20 sip extensions is going to be the death of me. |
| 23:18.08 | Shaun2222 | lmadsen: menuselect have pretty much everything selected.. |
| 23:18.11 | Shaun2222 | by default |
| 23:18.50 | lmadsen | right |
| 23:18.58 | lmadsen | but you learned something |
| 23:19.01 | Dayver | Hey I am using voip.ms as my main sip provider, does anyone can suggest any good SIP providet out there. |
| 23:20.17 | Shaun2222 | well func_moh or whatever isnt there anymroe.. |
| 23:22.16 | Shaun2222 | actually i look to be ok |
| 23:22.23 | Shaun2222 | none of the modules i have set to load are missng |
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| 23:25.01 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 23:31.51 | cappiz | is it possible to make an ssh tunnel to SIp |
| 23:32.11 | cappiz | like ssh ssh-server 5060:sip-server:5060 ? |
| 23:32.34 | JayTee52 | don't think so |
| 23:32.50 | JayTee52 | might be possible though |
| 23:33.05 | cappiz | HUM |
| 23:33.07 | lmadsen | cappiz: try it |
| 23:33.17 | lmadsen | I think it should work... just not sure how latent it would be |
| 23:33.18 | JayTee52 | by default SSH uses port 22 |
| 23:33.37 | lmadsen | I create tunnels for connection to MySQL servers all the time |
| 23:33.50 | lmadsen | ssh -L 5432:myserver:5432 lmadsen@myserver |
| 23:34.01 | *** join/#asterisk Docfxit (n=Docfxit@ip-64-32-143-214.lax.megapath.net) |
| 23:34.30 | lmadsen | so see no reason you couldn't do it for SIP... think the connection has to be initiated from the same box that tunnel is created from though |
| 23:34.52 | JayTee52 | worth a try anyways |
| 23:35.19 | Docfxit | HI, Does anyone know how to enter the default gateway and DNS servers into a cfg file for Polycom phones? |
| 23:36.10 | cappiz | hum... doesnt looke like it works... you think it might be an TCP/UDP issue= |
| 23:41.54 | [hC] | any of you guys watching 20+ hints on a polycom with firmware 2.0+ ? |
| 23:42.13 | *** join/#asterisk jblack (n=jblack@pool-71-181-145-13.sctnpa.east.verizon.net) |
| 23:45.09 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
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| 23:50.47 | trippss | ~test |
| 23:50.47 | jbot | Oh, no! There's a test and I haven't studied! |
| 23:53.58 | cappiz | i get this in my debug: Received incoming SIP connection from unknown peer to XXXXXX (SIP username from provider) and then it plays: "ss-noservice" |
| 23:54.08 | cappiz | what would be the most common reason? |
| 23:56.44 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 23:58.23 | jblack | whoah. The fwd<=>packet8 gateway actually works. |
| 23:58.46 | Qwell | jblack: occasionally |
| 23:59.45 | jblack | fair enough. |
| 23:59.53 | jblack | whoah. The fwd<=>packet8 gateway is actually working at this minute! |