| 00:00.08 | nhuisman_work | *snicker* |
| 00:00.08 | pcrane | or, more specifically, a PBXtra channel? |
| 00:00.29 | pcrane | yeah |
| 00:00.30 | pcrane | I know |
| 00:00.32 | pcrane | not my choice |
| 00:00.45 | pcrane | is working for a client using it, and I need to get AMD working on it |
| 00:01.11 | keith4_ | can it be set to *not* route? |
| 00:02.44 | lanning | keith4_: don't plug in the external ethernet? |
| 00:02.55 | keith4_ | ooh. good point |
| 00:06.59 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:06.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:07.55 | pcrane | feels dirty... |
| 00:08.16 | pcrane | is in the trixbox channel |
| 00:11.43 | voxter | russellb: hey russ, thanks for the email the other day. very helpful stuff. One more question, does IAX in 1.6 support transfers while staying in the signalling path, but removing itself from the media path now? ( i know this was an issue before ) |
| 00:12.07 | russellb | Yes. In fact, 1.4 does as well |
| 00:12.30 | voxter | I kinda thought so. |
| 00:12.41 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 00:12.58 | russellb | I think you configure transfer=mediaonly in iax.conf ... |
| 00:13.09 | voxter | Ah, i was just about to ask that. Thanks :) |
| 00:13.21 | voxter | I put together notes for what i was goign to cover, and my iphone ate them. :( i'm starting over |
| 00:15.23 | russellb | eep |
| 00:15.26 | russellb | that's no good |
| 00:15.33 | *** part/#asterisk DarrenCT (n=darren@adsl-065-007-146-085.sip.asm.bellsouth.net) |
| 00:15.40 | russellb | as long as I don't have a conflict, I'll try to be at your presentation |
| 00:15.49 | voxter | yeah.. thank god its still fairly fresh in my head, so its not that bad.. |
| 00:15.52 | voxter | Great |
| 00:16.21 | voxter | It's my first time speaking at a conference, but hopefully i can have some fun with it :) |
| 00:17.33 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 00:17.42 | unpaidbill | ~astlog |
| 00:17.46 | unpaidbill | ~irclogs |
| 00:18.00 | voxter | russellb: do you know of any commercial vendors who have implemented IAX in media gateways? |
| 00:18.46 | unpaidbill | ~logs |
| 00:18.47 | jbot | All conversations are logged to http://ibot.rikers.org/channel, where "channel" is replaced by the URL-encoded channel name, such as %23freenode for #freenode. Lines starting with spaces are not logged. |
| 00:18.47 | russellb | Only ones that are using Asterisk ... |
| 00:19.36 | *** join/#asterisk bsdfreak (n=alex@ninja.paranode.net) |
| 00:19.39 | bsdfreak | hi. |
| 00:20.50 | UD | hi |
| 00:20.56 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 00:27.56 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:35.47 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 00:35.47 | *** join/#asterisk Yourname (i=Yourname@unaffiliated/yourname/x-837320) |
| 00:36.00 | rasterix | :O corydon is the maintainer of vim config files??? |
| 00:36.03 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 00:39.33 | Qwell | rasterix: what ever gave you that idea? |
| 00:41.44 | rasterix | qwell: voip-info |
| 00:42.06 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 00:47.00 | *** join/#asterisk emer08 (i=emerzone@122.55.66.8) |
| 00:47.50 | nhuisman_work | vim config files? |
| 00:48.01 | emer08 | anyone here who can help me about asterisk? I'm a newbie. |
| 00:48.18 | nhuisman_work | oh, syntax highlighting |
| 00:48.35 | nhuisman_work | emer08, have your read the book yet? |
| 00:48.37 | nhuisman_work | ;) |
| 00:49.00 | nhuisman_work | http://downloads.oreilly.com/books/9780596510480.pdf |
| 00:50.39 | emer08 | thanks sir |
| 00:51.15 | emer08 | sir can i you a question about how to connect this one? |
| 00:51.27 | bsdfreak | anyone seen Call leg/Transaction Does Not Exist when Asterisk drops calls after about 20s? |
| 00:51.35 | emer08 | PSTN > Asterisk > PSTN |
| 00:52.33 | nhuisman_work | emer08, you mean use analog phones with asterisk? |
| 00:52.49 | nhuisman_work | I'd say you need some ata converters |
| 00:52.54 | nhuisman_work | linksys pap2 are not bad |
| 00:53.13 | emer08 | we're using TE410P |
| 00:53.18 | nhuisman_work | they act as sip devices from asterisk and provide an analog phone line for non-ip phones |
| 00:53.57 | Wayhigh | anyone here have any ideas of how to lower the post dial delay on zaptel/tdm400's? |
| 00:54.01 | nhuisman_work | yeah, that gets you ptsn -> asterisk |
| 00:54.36 | nhuisman_work | now you need FXS ports |
| 00:54.48 | Qwell | what? |
| 00:54.59 | nhuisman_work | wait, let me make sure i'm not confusing those two again :P |
| 00:55.23 | Qwell | Wayhigh: tell your telco to answer faster |
| 00:55.28 | nhuisman_work | yeah that's correct. |
| 00:55.44 | Wayhigh | qwell: eh? |
| 00:55.47 | nhuisman_work | emer08, http://www.voipsupply.com/index.php?cPath=96_124 |
| 00:56.00 | Qwell | where did he mention analog? |
| 00:56.00 | bsdfreak | Is Asterisk error 481 indicative of a specific configuration problem? |
| 00:56.04 | nhuisman_work | something like this : http://www.voipsupply.com/product_info.php?products_id=713 |
| 00:56.33 | Wayhigh | qwell: you think it's because the PSTN isn't fast enough? |
| 00:56.46 | nhuisman_work | Qwell, doesn't ptsn -> asterisk -> ptsn mean analog? |
| 00:56.53 | Qwell | umm, no |
| 00:56.57 | Wayhigh | I thought about testing ground start instead of kewlstart |
| 00:57.21 | nhuisman_work | Qwell, I guess it could be routing the whole t1 ? |
| 00:59.17 | *** join/#asterisk Entr4nced_ (n=IMG001@cpe-76-190-138-70.neo.res.rr.com) |
| 01:01.04 | *** join/#asterisk genji1981 (n=gej@203.152.122.50) |
| 01:02.15 | emer08 | how would you do an inbound call from pstn will be accepted through E1 by asterisk and will be automatically transfer to another E1 for the outbound call to another pstn? asterisk server will just act as a CDR recorder. |
| 01:03.27 | *** join/#asterisk VaNNi (n=VaNNi___@38.104.76.90) |
| 01:04.01 | *** join/#asterisk VaNNi (n=VaNNi___@38.104.76.90) |
| 01:04.34 | *** part/#asterisk dawalama (n=dawalama@wsip-68-105-250-226.sd.sd.cox.net) |
| 01:05.02 | JayTee52 | emer08, just enable CDR recording and then point all incoming calls from the first E1 to a context that dials a call out the Zap group of the other E1 |
| 01:07.13 | JayTee52 | emer08, but you might want to consider building a "failover queue" to queue calls to the outbound E1 if there are no channels available. If you have excess capacity you might not need to bother. |
| 01:08.01 | emer08 | thanks jaytee |
| 01:08.08 | JayTee52 | yw |
| 01:08.54 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:09.52 | emer08 | i tried that but the asterisk is quickly releasing the call. Is there a problem with my dial plan? |
| 01:10.11 | JayTee52 | pastebin your zapata.conf and your extensions.conf |
| 01:10.15 | JayTee52 | ~pastebin |
| 01:10.16 | 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 |
| 01:10.33 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 01:10.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:13.42 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:14.08 | tvirus | Is it possible, and I'm guessing it is but I don't know how, to have someone dial in and then, after going through an IVR and selecting support, have it ring all extensions that are considered 'support'. Essentially, Someone calls, they press 1 for support, and it rings extensions 100, 101, 102, 103 and 104. |
| 01:14.19 | tvirus | Gonna take a look at one of those visual dial plan thingys |
| 01:15.23 | JayTee52 | tvirus, yes but it would be better to setup a queue with agents for something like that. |
| 01:15.37 | emer08 | sorry jaytee i'm on leave. i'm not on the office. I will just work on it when i come back would you give me a sample on how the dial plan will be? |
| 01:15.38 | *** join/#asterisk bollarice (n=Bolla_Ri@m2d5336d0.tmodns.net) |
| 01:16.15 | JayTee52 | emer08, do you already have your zapata.conf setup and your E1's working? |
| 01:16.48 | tvirus | JayTee52: I think that's what our eventual goal is. |
| 01:17.09 | JayTee52 | have you read through any of the book yet? |
| 01:17.19 | tvirus | Yeah, working my way through them :) |
| 01:17.26 | tvirus | Haven't gotten to queues yet. |
| 01:17.28 | nhuisman_work | tvirus, yeah that's very easily possible |
| 01:17.53 | tvirus | Cool, I'll have to start researching. |
| 01:18.04 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 01:18.17 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 01:18.18 | nhuisman_work | tvirus, queues are the way to do it call queues or just manually ring all the numbers |
| 01:18.22 | JayTee52 | tvirus, queues with agents aren't hard to setup and the book has some good examples, plus there are some good tips on voip-info.org for them |
| 01:18.29 | nhuisman_work | oops messed up that sentence |
| 01:18.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3607feb459b25b8c) |
| 01:18.39 | nhuisman_work | i meant call queues are the best way but you can also just manually do it. |
| 01:18.40 | emer08 | yes @jaytee |
| 01:18.42 | exothermc | Anyone here have a local SIP trunk in China or HK? |
| 01:19.19 | JayTee52 | exothermc, same answer as last week and the week before. Nope! :-) |
| 01:19.55 | exothermc | JayTee52: Guess I should have been here last week. |
| 01:20.02 | tvirus | Thanks for the info, I'm reading up on them at voip-info |
| 01:20.28 | JayTee52 | ok, maybe not every week consecutively but I remember you asking that before. |
| 01:20.29 | irieKen | I have an aastra which is getting "SIP response 400" errors with Asterisk. Anyone experience this before? |
| 01:20.45 | exothermc | JayTee52: hmm I think you may have the wrong person. |
| 01:20.51 | JayTee52 | nope |
| 01:21.05 | exothermc | goes hunting in his logs for his lost memory |
| 01:21.07 | JayTee52 | maybe a couple weeks back |
| 01:21.49 | tvirus | Follow up on queues, if someone has been in a queue for say, 30 seconds, can it be escalated to another extension? I'm guessing something like Wait(30), exten => 105,1,Dial(...)? |
| 01:22.01 | JayTee52 | emer08, PM? |
| 01:22.48 | JayTee52 | tvirus, yes but queues have settings that will let help you do that more efficiently with less coding in the dialplan |
| 01:23.15 | mchou | irieKen: pastebin the full sip debug |
| 01:23.51 | *** join/#asterisk VaNNi (n=VaNNi___@38.104.76.90) |
| 01:23.57 | tvirus | Nice. That makes me happy. Thanks, JayTee52 |
| 01:25.10 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 01:25.45 | *** join/#asterisk Assimilate (n=Assimila@24-116-182-58.cpe.cableone.net) |
| 01:26.16 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:26.17 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 01:29.14 | JayTee52 | tvirus, your welcome |
| 01:32.18 | *** join/#asterisk propellerhead (n=yogurt2u@200.117.254.94) |
| 01:32.32 | irieKen | mchou: I am trying to get the sip debug now. I have never done this before, so sip set debug, and copy+paste what gets spit out? |
| 01:34.01 | propellerhead | hello people |
| 01:34.11 | propellerhead | whats |
| 01:35.10 | irieKen | Hi Propellerhead. |
| 01:35.30 | propellerhead | what is the command o funtion to see how many the channels actives are in one momment |
| 01:36.00 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:36.00 | propellerhead | is not a english speaker |
| 01:36.45 | propellerhead | I cannot find this information in the doc |
| 01:36.49 | irieKen | Propellerhead: Sip show channels |
| 01:37.07 | propellerhead | Yes iratik, it is en the cmd line |
| 01:37.31 | propellerhead | but i want to know in the dialplan |
| 01:39.26 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:39.28 | propellerhead | for example: exten => XXXX,n,gotoif(number_of_channes_on > 20|na12) |
| 01:42.30 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:42.42 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 01:43.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 01:43.30 | irieKen | mchou: I figured out my problem. I had accidentally set the CID incorrectly, and the Aastra didn't like that. |
| 01:47.23 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 01:47.46 | *** join/#asterisk chendy (n=chatzill@121.35.51.25) |
| 01:48.43 | exothermc | JayTee52: Well just grep'd through all my logs, no mention of me asking that before |
| 01:49.18 | exothermc | JayTee52: but anyway, didn't know it would be so damn hard to parse colloguy transcripts |
| 01:50.23 | nhuisman_work | propellerhead, you might look at : http://www.voip-info.org/wiki/view/Asterisk+func+group |
| 01:50.28 | JayTee52 | exothermc, I apologize and stand corrected. |
| 01:51.00 | exothermc | JayTee52: no problem learned me some sed in the process ;) |
| 01:51.14 | nhuisman_work | that example seems to imply you can check how many calls are on the trunk group |
| 01:51.19 | nhuisman_work | using GROUP_COUNT |
| 01:51.24 | JayTee52 | sed, awk and grep, how could we live without them. |
| 01:51.41 | nhuisman_work | i'm guessing you could use that to make asterisk say the number |
| 01:51.42 | exothermc | bah it is fact we can't. |
| 01:51.55 | nhuisman_work | I think I use sed very little |
| 01:52.05 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 01:52.20 | nhuisman_work | hmm |
| 01:52.30 | nhuisman_work | well I guess I use the standard syntax sed uses in vi a lot |
| 01:52.34 | exothermc | well you can get away with using very little of either sed or awk, but if you don't use much of both then you don't do very much. |
| 01:52.47 | propellerhead | tks nhuisman_work |
| 01:52.48 | exothermc | ;) |
| 01:53.00 | JayTee52 | exothermc, I've got a couple good pdf cheat sheets in pdf format for sed and awk |
| 01:53.00 | propellerhead | I'll see this now |
| 01:53.32 | nhuisman_work | I think I use vi as a replacement to sed |
| 01:53.52 | JayTee52 | I've also got a good bash-vi-editing cheatsheet too |
| 01:55.06 | jblack | I bet somewhere out there, is a vi shell |
| 01:56.02 | nhuisman_work | discusting... heh |
| 01:56.15 | nhuisman_work | you would break your : key |
| 01:56.28 | jblack | I'm sure you'd be in line mode |
| 01:56.31 | nhuisman_work | you can always use the shell inside vi |
| 01:56.35 | nhuisman_work | !command |
| 01:56.41 | nhuisman_work | not quite the same I guess |
| 01:56.46 | nhuisman_work | I use it sometimes for file lists |
| 01:58.19 | nhuisman_work | oo time to go paddling |
| 02:09.12 | *** join/#asterisk jameswf-home (n=james@ip24-56-23-35.ph.ph.cox.net) |
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| 02:09.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:12.07 | JayTee52 | ~101 |
| 02:12.08 | jbot | 101 is probably Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 02:13.37 | *** join/#asterisk irieKen_ (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 02:15.50 | *** join/#asterisk dwayne (i=dwayne@76.29.245.9) |
| 02:16.59 | scooby2 | Anyone use ZapScan() to listen to calls? |
| 02:23.14 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:24.46 | _ShrikE | uses chanspy to listen to calls |
| 02:26.32 | scooby2 | ahh, looks like that may have replaced zapscan |
| 02:27.16 | scooby2 | with zapscan you marked channels with a group and then ran zapscan(group) so only certain could be listened to |
| 02:28.57 | scooby2 | guess they arent the same. |
| 02:30.13 | Alton2 | Folks, a question if I may, |
| 02:30.25 | *** join/#asterisk grufft3ch (n=grufft3c@70.114.145.96) |
| 02:30.40 | Alton2 | thinking about a setup spanning 20 or so campuses, one asterisk box at each one, |
| 02:31.19 | Alton2 | for the normal maintenance of sip.conf and extensions.conf I'm wondering if there's anything web-based and quite basic out there, |
| 02:31.41 | Alton2 | if not I can write something with a few PHP pages, just wondering. |
| 02:31.43 | Alton2 | . |
| 02:37.47 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
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| 02:39.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:42.47 | *** join/#asterisk AiT (n=airani@66-146-175-59.skyriver.net) |
| 02:44.13 | AiT | I just purchased some G.729 Licenses from Digium, what's the codec_g729a_v35_nocona.tar.gz vs. codec_g729a_v35_opteron.tar.gz ? |
| 02:44.29 | h3x | isnt that fairly obvious |
| 02:44.41 | h3x | amd opteron, intel nocona architecture |
| 02:45.21 | AiT | ive got a QuadCore AMd, is the opteron optimized? |
| 02:45.43 | h3x | isnt that part of the opteron series |
| 02:48.35 | brodiem | I always stick with the i386 build to ensure compatibility.. |
| 02:55.58 | Qwell | brodiem: compatibility with 1994? |
| 02:56.44 | Qwell | AiT: is it a 64-bit install? |
| 02:58.55 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 03:02.33 | [TK]D-Fender | Alton2: No. |
| 03:02.42 | UD | het [TK]D-Fender |
| 03:03.23 | UD | i tried the xorcom live cd |
| 03:03.29 | AiT | Qwell, yeh, 64bit |
| 03:03.45 | Qwell | then opteron |
| 03:03.58 | UD | but it still wont aknowledge my fxo channel |
| 03:04.27 | AiT | show G729 (per instructions) is not working from the command line. was that command changed in version 1.4? |
| 03:05.08 | Qwell | AiT: did it load successfully? |
| 03:05.17 | [TK]D-Fender | UD: install a normal distro and normal asterisk, and normal zaptel like the rest of us |
| 03:05.27 | AiT | Qwell, dunno...trying to find out |
| 03:05.41 | UD | now i simply get ZT_CHANCONFIG FAILED failed on channel 1: No such device or address (6) |
| 03:05.48 | Qwell | it probably didn't. unload it, and load it again while Asterisk is running. You should see errors if it failed |
| 03:06.27 | UD | well i will i needed a last resort now because apt-get decided to delete my kernel and not give me a new one |
| 03:06.28 | AiT | Qwell....maybe it was still loading, now all is well :) |
| 03:06.44 | UD | no more menu.lst for grub initrd nothing |
| 03:07.26 | UD | then i found the only kernel i had on disk laying around and tried to boot it and got errors so i just gave up |
| 03:07.33 | UD | besides my install was lame anyway |
| 03:08.07 | UD | im starting to think i got punked by whoever sold me this card |
| 03:08.15 | [TK]D-Fender | UD: Maybe you should pick a distro that you are able to manage this time. |
| 03:08.36 | UD | manage? |
| 03:08.47 | UD | i dont know what i need to manage is the thing, to make this work |
| 03:08.57 | UD | i did just like it said in the oreilly book |
| 03:09.14 | [TK]D-Fender | UD: If your kernel is falling out from under you and you can't keep it under wraps perhaps that distro wasn't a great choice for you |
| 03:09.44 | [TK]D-Fender | UD: the O'Reilly book (TFOT) does not make managing your DISTRO its business |
| 03:09.45 | UD | well |
| 03:09.57 | UD | i dont hack apt stuff |
| 03:10.03 | UD | or really delete anything |
| 03:10.11 | UD | i'd expect it to do what its supposed to by default |
| 03:10.46 | UD | if i do a apt-get install kernel-image.... i expect it not to delete all my necessary stuff for booting |
| 03:11.29 | UD | so i dunno what that was all about |
| 03:11.54 | UD | it just doesnt have any data worth saving |
| 03:12.07 | UD | i kinda like the menu though |
| 03:12.13 | UD | on this livecd |
| 03:18.26 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 03:24.54 | genji1981 | ~pastebin |
| 03:24.55 | 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 |
| 03:25.19 | AiT | hmm...asterisk not dialing or recving with G729 enabled only... |
| 03:25.29 | AiT | any have voicepulse? |
| 03:29.55 | [TK]D-Fender | AiT: Feel free to share actual debug data with us... |
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| 03:38.07 | jameswf-home | psh data puhlese |
| 03:39.04 | jblack | Thankfully, I speak zombie. I believe jameswf-home said "Push data, please" |
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| 03:46.29 | *** part/#asterisk spokra (n=spokra@74-61-17-236.sea.clearwire-dns.net) |
| 03:47.55 | [TK]D-Fender | sits back to listen to the crickets |
| 03:53.30 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:55.29 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 03:55.34 | jblack | It is kind of quiet. |
| 03:55.41 | jblack | checks the confernece |
| 03:56.38 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 04:01.47 | AiT | http://pastebin.com/d5b171862 |
| 04:01.52 | AiT | "all circuits are busy now" when allow=G729 When I insert GSM in there, it dials through |
| 04:02.08 | Alton2 | Fender, thanks. |
| 04:02.37 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-133-147.dsl.teksavvy.com) |
| 04:04.28 | [TK]D-Fender | AiT: That is not SIP debug and is of no use. |
| 04:05.01 | [TK]D-Fender | AiT: pastebin the failed call including SIP DEBUG, along with "show modules like codec" |
| 04:07.04 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 04:07.17 | UD | http://pastebin.com/m5c28c2f9 |
| 04:07.31 | AiT | [TK]D-Fender, that is sip debug...here is another call with show modules like codec http://pastebin.com/m5b9cb1a1 |
| 04:09.46 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 04:10.40 | [TK]D-Fender | AiT: No, that is not SIP debug for your call attempt |
| 04:13.11 | [TK]D-Fender | UD: What card do you have? |
| 04:14.09 | UD | a wildcard x100p |
| 04:14.29 | UD | pci:0000:01:04.0 wcfxo- 1057:5608 Wildcard X100P |
| 04:14.51 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 04:15.24 | [TK]D-Fender | UD: modprobe the driver and the retry ztcfg -vvvv. Also "rmmod ztdummy" |
| 04:15.30 | [TK]D-Fender | (first) |
| 04:15.42 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 04:16.44 | AiT | [TK]D-Fender, http://pastebin.com/dc8567e8 |
| 04:17.48 | UD | same error |
| 04:18.00 | UD | i've been trying the module i just forgot i rmmod it |
| 04:18.03 | UD | wcfxo |
| 04:18.36 | UD | maybe it doesnt like this card |
| 04:20.12 | AiT | [TK]D-Fender, doesn't look like its using the G729 codec, huh? |
| 04:22.49 | [TK]D-Fender | AiT: http://connect.voicepulse.com/Features.aspx |
| 04:23.18 | [TK]D-Fender | AiT: next time read your providers big print. |
| 04:24.17 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:24.17 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:24.37 | AiT | [TK]D-Fender, WTF, I thought G729 was to be the standard?!? |
| 04:24.52 | [TK]D-Fender | ~assume |
| 04:24.53 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 04:27.35 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 04:31.50 | AiT | man...that's such a drag...i really like VoicePulse... |
| 04:33.04 | AiT | [TK]D-Fender, can you recommend anyone? |
| 04:33.06 | ManxPower | AiT: it costs money to use G729 |
| 04:33.17 | ManxPower | AiT: How many G729 licenses have you purchased? |
| 04:33.26 | AiT | 20 licenses |
| 04:33.28 | UD | so [TK]D-Fender would you pretty much say my card is garbage? |
| 04:33.40 | UD | it was boxed up in the garage for a while |
| 04:33.47 | AiT | $10 a license is a steal... |
| 04:34.26 | [TK]D-Fender | AiT: And why did you buy 20 licences? |
| 04:34.29 | ManxPower | AiT: now go convince Voicepulse to spend some money for G729 licenses for THEIR stuff. |
| 04:34.52 | [TK]D-Fender | ManxPower: Lets see what we can figure from the start of this... |
| 04:35.07 | AiT | [TK]D-Fender, it's a office stepup...we've got about 35 extensions |
| 04:35.13 | AiT | stepup |
| 04:35.24 | [TK]D-Fender | UD: Its common to consider the card garbage even if it WAS working. |
| 04:35.37 | [TK]D-Fender | AiT: What kind of connection to the internet? |
| 04:35.41 | *** join/#asterisk Miguellinux (n=cecica@201.230.9.33) |
| 04:35.49 | AiT | [TK]D-Fender, T1 |
| 04:36.10 | UD | they have bad rep for being junk? |
| 04:36.19 | UD | i should take it out and see if any of the caps are bad |
| 04:36.31 | [TK]D-Fender | AiT: that would indeed barely survive the # of channels you were looking at. |
| 04:36.45 | [TK]D-Fender | UD: Its a craptastic winmodem. |
| 04:36.53 | UD | i have a whole room full of electronic parts |
| 04:36.56 | implicit- | lol |
| 04:37.04 | AiT | ManxPower, that's such a turn off...I can't believe they don't support G729 |
| 04:37.06 | UD | winmodem doesnt mean crap though |
| 04:37.08 | implicit- | no g729 that's ridiculous |
| 04:37.16 | UD | well it means crap but not necessarily malfunction |
| 04:37.20 | [TK]D-Fender | AiT: Feel free to punish yourself for not looking. |
| 04:37.25 | implicit- | if you want, try out my service i'll give you some free dough on it to play around - www.flowroute.com |
| 04:38.10 | [TK]D-Fender | AiT: GSM would still fit |
| 04:38.19 | [TK]D-Fender | AiT: and would be a much lower load on your server |
| 04:38.57 | AiT | [TK]D-Fender, I've got a QuadCore X64 9800GHz 8GB mem system... |
| 04:39.12 | AiT | load is under 1% |
| 04:39.20 | implicit | :) |
| 04:39.45 | [TK]D-Fender | AiT: Of course it is... you can't put the transcoding load on it to test because they don't even ACCEPT it. |
| 04:39.49 | AiT | implicit, how long your service been around? |
| 04:39.54 | implicit | one year |
| 04:40.13 | implicit | we have custs who spedn 20 bucks a month |
| 04:40.16 | implicit | and ones who spend 15k/mo |
| 04:40.26 | [TK]D-Fender | AiT: thats like saying I have a high-performace car, lookat at it, it isn't straining at all... and its PARKED |
| 04:40.40 | [TK]D-Fender | AiT: Just go try GSM already |
| 04:40.47 | AiT | implicit, 760 number? I'm in San Diego... |
| 04:40.48 | implicit | we optimize our media paths directly to the gateways worldwide, to minimize jitter & latency. We also do t.38 for fax |
| 04:40.51 | implicit | really? cool |
| 04:40.53 | implicit | oceanside |
| 04:41.23 | AiT | [TK]D-Fender, I've been using GSM, but we're hitting the bandwidth limit |
| 04:41.33 | AiT | [TK]D-Fender, we've been using GSM so far with VoicePulse |
| 04:41.34 | implicit | you can increase frame size |
| 04:42.38 | [TK]D-Fender | AiT: You are pushing the limits on your T1..... RTP+GSM= 34kbps/channel |
| 04:42.55 | [TK]D-Fender | implicit: Does VP support that? |
| 04:42.59 | implicit | i don't nkow |
| 04:43.02 | implicit | we support variable frame size |
| 04:43.09 | implicit | you can set 60ms frames |
| 04:43.11 | implicit | any size you want |
| 04:43.14 | implicit | to reduce ip overhead |
| 04:43.22 | implicit | we use Sonus gateways, which do it pretty well |
| 04:43.29 | implicit | and our signaling side doesn't touch the media |
| 04:43.40 | AiT | <implicit, who and were is your backbone? |
| 04:43.54 | implicit | we pass most of our traffic to what used to be MCI |
| 04:44.21 | implicit | we have locations in LA, NY, Toronto, Colombia, and Brazil |
| 04:44.24 | implicit | for gateways |
| 04:44.29 | scooby2 | [TK]D-Fender: ever used ZapScan() in 1.4? |
| 04:44.35 | implicit | depending on where traffic is going, so your media will go point to point |
| 04:49.12 | [TK]D-Fender | scooby2: Nope |
| 04:55.32 | scooby2 | if i can figure out how to record transferred calls and get ZapScan() working again, today will have been quite a success |
| 04:56.37 | jblack | Hmmm. Monitor stops working after transfer? |
| 04:58.34 | scooby2 | believe so, yes |
| 05:04.03 | scooby2 | atleast mixmonitor is known to stop recording when transferred |
| 05:04.12 | *** part/#asterisk AiT (n=airani@66-146-175-59.skyriver.net) |
| 05:06.50 | scooby2 | appears when the call is transferred it loses almost everything. callerid, recording, etc |
| 05:09.21 | scooby2 | looks like the hack someone figured out is to start recording when the call is passed to the user/agent/ext |
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| 05:09.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:12.09 | [TK]D-Fender | bed time. Lter all |
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| 05:45.49 | UD | hi again |
| 05:46.00 | techman97 | yo |
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| 05:54.44 | UD | im not sure if im misunderstanding this or what |
| 05:54.59 | UD | can asterisk accept incoming sip calls? |
| 05:55.12 | kaldemar | UD: yes |
| 05:55.52 | *** join/#asterisk hi365_m (n=hi365@213.151.50.174) |
| 05:55.53 | UD | i was reading that book, and all over the net. i get examples like this http://blog.lithiumblue.com/2007/07/receiving-incoming-sip-calls-from.html |
| 05:56.26 | UD | but none of them are very good because they dont say where i put my info in |
| 05:57.10 | UD | so do i need a seperate program to answer the call and pipe it over via iax? |
| 05:57.54 | kaldemar | what? |
| 05:57.58 | jblack | ud: I think you might do good reading the book a second time. Did you read this one? : |
| 05:58.01 | jblack | ~book |
| 05:58.02 | 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, or see ~buybook |
| 05:58.15 | kaldemar | even that example mentions sip.conf. |
| 05:58.25 | techman97 | UD: yeah, you need to research a little more - asterisk does SIP, IAX, PRI/PSTN...all in one. |
| 05:58.56 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-114-16-171.satx.res.rr.com) |
| 05:59.03 | UD | yea kaldemar but when i sign up for a sip service |
| 05:59.10 | UD | how does it know to go to me? |
| 05:59.25 | techman97 | you configure the SIP settings in sip.conf |
| 05:59.31 | kaldemar | you tell the service how. |
| 05:59.35 | techman97 | exactly. |
| 05:59.41 | jblack | rearead the book pleae. I'll give you the chapter to reread most carefully if you're patient |
| 05:59.51 | techman97 | it's a little overwhelming for the beginner, but read, read, read. |
| 06:00.37 | kaldemar | to repeat the original answer, you definitely have misunderstood and yes, asterisk can accept sip calls. |
| 06:00.49 | UD | you know where i could find an example config? |
| 06:01.03 | UD | yes that would help jblack. |
| 06:01.08 | jblack | As example comes with asterisk. |
| 06:01.51 | jblack | chapters 8, 4, 5 and 6 would be good starts. |
| 06:03.53 | drmessano | 7,10,and 3 are almost pornlike |
| 06:05.13 | jblack | bleh. cnn is whoring for Palin. |
| 06:06.27 | UD | i've read the book over and over |
| 06:06.35 | UD | its a waste of money i hate oreilly |
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| 06:07.05 | jblack | oh boy. |
| 06:07.39 | UD | chapter 4 begins with, "how sip works" basically |
| 06:08.08 | UD | then goes on to a useless configuration and i think "hell yea any second its going to tell me where to put my sip proxy and user name" |
| 06:08.19 | UD | then it goes on to inbound IAX |
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| 06:08.45 | jblack | makes popcorn |
| 06:09.20 | jblack | If it's that hard for you, I can set up a working system for you for appropriate financial considerations. msg me if interested. =) |
| 06:09.52 | techman97 | UD: you may be more apt to use Asterisk in a flash, Trixbox, FreePBX...etc... |
| 06:09.55 | techman97 | it's all point and click |
| 06:10.05 | techman97 | core Asterisk is nitty gritty text file manipulation |
| 06:10.10 | kaldemar | UD: are you using plain asterisk or some GUI on it? |
| 06:11.47 | UD | am i going to be able to point and click my way to somewhere i can put my info? |
| 06:12.03 | UD | actually lemme read the 2nd edition of the book |
| 06:12.15 | UD | the 1st edition is what i have |
| 06:13.02 | jblack | also, appendix D; configuration files. |
| 06:14.35 | kaldemar | UD: page 97 has a header "Connecting to a SIP Service Provider" |
| 06:14.58 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 06:15.14 | kaldemar | UD: and configuration examples. |
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| 06:15.36 | kaldemar | chapter 4, like jblack said. |
| 06:16.21 | jblack | That's how I know he didn't read the book. What he wants is in it |
| 06:17.02 | UD | i am looking at the 2nd edition book |
| 06:17.08 | UD | the 1st doesnt mention it |
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| 06:25.14 | UD | i think the book is down :( |
| 06:25.36 | kaldemar | no, it's not. |
| 06:25.53 | UD | well, doesnt load for me |
| 06:26.00 | UD | i think i have a voodoo spell on my computer |
| 06:26.14 | kaldemar | http://downloads.oreilly.com/books/9780596510480.pdf |
| 06:27.37 | slider750 | UD: O'Reilly helped me obtain and complete several jobs, just my two cents. It's more reference material, not step by step to get you connected to broadvoice or whatever you are trying to connect to. What is the business purpose of your asterisk project anyways? |
| 06:28.36 | *** join/#asterisk chigital (n=chigital@tmo-127-1.customers.d1-online.com) |
| 06:30.43 | UD | that link is much more in depth than the book |
| 06:30.49 | UD | that host=line doesnt even exist |
| 06:31.05 | UD | i was going to try that but i figured i'd ask first |
| 06:31.40 | UD | slider750 i would like to configure farm equipment and poll for environmental data |
| 06:31.57 | UD | via phone |
| 06:32.24 | slider750 | so does the farm equipment phone home or something like that? |
| 06:32.48 | slider750 | UD: if you want to share, please do... there may be some options for ya |
| 06:33.24 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-8c28b1f119aba498) |
| 06:34.27 | slider750 | UD: also your SIP provider may provide asterisk instructions |
| 06:34.58 | *** join/#asterisk af_ (n=getsmart@88-149-240-50.dynamic.ngi.it) |
| 06:37.20 | UD | i dont have any schematics if thats what you mean |
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| 06:39.44 | *** part/#asterisk Lanh (i=lanh@unaffiliated/lanh) |
| 06:39.55 | belendax | when I run modprobe zaptel it says FATAL: Module zaptel not found . why ? |
| 06:40.07 | jblack | belendax: You don't have the module. |
| 06:40.55 | kaldemar | looks around for captain obvious |
| 06:41.19 | jblack | He's been idle for a while. |
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| 06:42.00 | belendax | jblack: doesn't modules of zaptel adds automatically in kernel 2.6 ?! |
| 06:42.13 | kaldemar | no |
| 06:42.33 | *** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) |
| 06:42.53 | belendax | so how can I add modules ? |
| 06:43.07 | kaldemar | you need to install zaptel if you want to use it. |
| 06:43.19 | belendax | kaldemar: I installed it |
| 06:44.24 | kaldemar | how? |
| 06:45.08 | belendax | kaldemar: from source , with ./configure ..... make install |
| 06:45.14 | jblack | I'm not in the mood for questions of this calibre tonight. |
| 06:45.16 | dandre | Hello, |
| 06:46.10 | jblack | the questions are so broad and diffuse, it's like swatting at fog. |
| 06:47.11 | dandre | I have search google to find a solution to inbound sip connection showing 'check_auth: username mismatch' when insecure is set to invite. I haven't found. Is there anything I can do? |
| 06:49.06 | jblack | sounds to me like you have set the authuser or user option in sip.conf |
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| 06:57.09 | mort_gib | Dandre: What is your problem |
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| 07:00.20 | dandre | IfI set insecure=invite in sip.conf, on inbound calls I get this error message: |
| 07:00.20 | dandre | check_auth: username mismatch, have <ovh>, digest has <anonymous> |
| 07:00.27 | felipex | hi at all |
| 07:02.19 | dandre | basically the problem is that if I have 2 sip accounts on the same provider, all inbound call are shown to be from only one account |
| 07:02.22 | mort_gib | Why do you need insecure=invite |
| 07:02.37 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 07:02.44 | dandre | sorry if I set insecure=no |
| 07:03.08 | dandre | I must set insecure=invite to get inbounds calls |
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| 07:12.00 | felipex | is it possible to delete gap on music on hold ? |
| 07:12.23 | felipex | sometimes there are some gaps on the stream |
| 07:21.56 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 07:26.43 | dandre | If I put in my inbound sip context |
| 07:26.44 | dandre | exten = _X!,... |
| 07:27.02 | dandre | the call is rejected but not if I put |
| 07:27.10 | dandre | exten = s,.. |
| 07:27.35 | dandre | why? |
| 07:30.57 | mchou | dandre: I suggest you pastebin your sip.conf (mask usernames passwords) |
| 07:31.41 | mchou | dandre: no way for anyone to diagnose for sure unless they see this |
| 07:32.06 | dandre | If I set sip debug I get INVITE w/o did |
| 07:32.08 | mchou | dandre: also your dialplan |
| 07:32.54 | dandre | ok I'll arrange for this by I use users.conf instead of sip.conf |
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| 07:33.27 | dandre | the sip provider is base on cirpack |
| 07:35.42 | mchou | dandre: sip.conf should be pastebin'ed |
| 07:37.20 | mchou | dandre: I'm dubious anyone is really going to give a darn about your users.conf |
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| 08:06.05 | dandre | mchou: here are the relevant infos: |
| 08:06.06 | dandre | http://pastebin.com/d4f93da4f |
| 08:07.38 | dandre | if I add exten = s,... lines in trunk-inbound-kwtk-100006 context, I get inbound calls |
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| 08:19.33 | emer08 | what link will i go if i'm not familiar with the terms in asterisk or VOIP? |
| 08:19.45 | maxhbp2005 | hi everyone |
| 08:20.40 | maxhbp2005 | i have a issue regarding MOH |
| 08:20.47 | maxhbp2005 | can anyone help me? |
| 08:21.20 | EmleyMoor | What is the issue? |
| 08:21.45 | maxhbp2005 | thanks for your reply |
| 08:22.26 | linuxstb | emer08: I would suggest the Asterisk book - http://www.asteriskdocs.org/ (or buy a hard copy) |
| 08:22.26 | maxhbp2005 | when i got incoming call i have setup a script that will dial local channel to my sip user and i have put moh instead of ring |
| 08:22.42 | maxhbp2005 | but i am not able to hear any moh from end |
| 08:22.52 | maxhbp2005 | and on asterisk it is playing fine |
| 08:22.56 | emer08 | thanks |
| 08:23.58 | maxhbp2005 | any ideas? |
| 08:24.11 | maxhbp2005 | becuase provider side there is no issue of codecs |
| 08:24.30 | maxhbp2005 | i have also checked that incoming call and with same did it is giving me MOH |
| 08:24.43 | maxhbp2005 | but on working server it is not giving me MOH |
| 08:24.55 | dandre | I have this invite command from my provider: http://pastebin.com/d9125fbb |
| 08:25.34 | dandre | How can I get the data from the To:... Line in my dialplan (extensions.conf) ? |
| 08:28.08 | SwK | dandre, thats the s extension to match it... you're probably registering out to them in sip.conf... |
| 08:28.30 | SwK | if you want to match on that account change your register line to be somehting like |
| 08:28.53 | SwK | register => user:password@host/WHAT_IM_LOOKINGFOR |
| 08:29.14 | SwK | and they'll change that s@ to WHAT_IM_LOOKINGFOR@ |
| 08:31.18 | dandre | ok and if I have 2 did numbers on this account, how will it work? |
| 08:35.29 | SwK | what provider? |
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| 08:38.41 | mort_gib | If I have CONFIG_HZ_250=y and CONFIG_HZ=250 in /boot/conf will it cause issues with * |
| 08:44.29 | mort_gib | <maxhbp2005>You want to have a look at queus |
| 08:44.45 | maxhbp2005 | ok |
| 08:44.50 | maxhbp2005 | i am checking that |
| 08:44.53 | maxhbp2005 | thanks |
| 08:46.35 | maxhbp2005 | that will affect to all incoming call or one of them? |
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| 08:49.17 | dandre | SwK: ovh.com |
| 08:52.06 | dandre | the register line works ok |
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| 09:38.33 | fbnts | Hi, I was wondering if there was a way to set a queue to request that the agent press a number to accept the call? |
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| 10:10.54 | jkirby | hello, I'm trying to setup port forwarding from our ADSL router to a internal PABX, we use SIP accounts on the asterisk box (which we then make calls via POTS trunks) - what port do I forward to to allow registration/functionality of this? |
| 10:11.54 | jblack | 5060, and the range you have in rtp.conf |
| 10:12.17 | jblack | The default in rtp.conf is... excessive. I'd trim it down to 3x the number of concurrent calls you want |
| 10:12.39 | jkirby | and it would be UDP? |
| 10:12.53 | jblack | did you even try to look this up? |
| 10:13.00 | genji1981 | jblack: no kidding. my router log shows sip: ports exceeded |
| 10:13.40 | jblack | 'this machine has 65k ports. I know! I want 15% of 'em. That'll be plenty!' |
| 10:13.54 | genji1981 | and every once and a while, i get disconnected. |
| 10:14.14 | jblack | cut it down in rtp.conf. a hundred is probably more than you'll ever need. |
| 10:14.26 | genji1981 | why was it set to default of such a big range? |
| 10:14.37 | jkirby | jblack: yes I did, and applied what I found but it still aint working. |
| 10:14.47 | jblack | genji1981: Too much crack cocaine, is my guess. |
| 10:15.06 | genji1981 | ah.. making up for something. |
| 10:15.20 | genji1981 | 'my, its so huge!' |
| 10:15.34 | jblack | heh. that may be. |
| 10:17.01 | fbnts | Hi, I was wondering if there was a way to set a queue to request that the agent press a number to accept the call? |
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| 10:51.07 | jblack | Damnit. My doctor is trying to kill me. Again. |
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| 10:58.53 | _Roman | Can anyone tell me where the queue_log is documented? |
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| 11:48.44 | lesouvage | Does any of you have good experience with Polycom Soundpoint IP330 interfacing with ldap to retrieve all the info to fill the contact book so users can call by name and not by number? |
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| 12:11.11 | patrick-- | hey all, im seeking an asterisk webfrontend where people can check/manage their voicemailboxes |
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| 12:16.47 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:21.53 | matt_keys | i've got a customer in florida who has a tdm400 with two analog lines. outbound calls work, but inbound from a cell phone rings a half of a cadence then stops. on the console it shows all the phones ringing. when the customer picks up the phone she hears dialtone. on my end of the cell phone, i hear nothing past the 1/2 ring |
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| 12:22.35 | matt_keys | if i call from our office lines (on another asterisk box with outbound through PRI), the phone actually rings down there and I can talk to them |
| 12:22.46 | matt_keys | the console looks the same as the cell phone call |
| 12:24.46 | matt_keys | zttool shows that the card is fine, nothing on the logs indicating anything is wrong |
| 12:25.16 | matt_keys | any ideas what's going on with it? i'm wondering if it got zapped by lightning |
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| 12:38.22 | Blackvel | hi all. is there any offical german/international lookup service which asterisk can use to pstn / calling e.164 number to company name mapping? |
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| 12:45.58 | bminish | hello I have a sangoma card that keeps resetting PRI with short write errors. some log extracts here http://pastebin.com/m5cc4d0d4 |
| 12:46.30 | jblack | word to the wise? If you have a job, try to keep it. |
| 12:47.02 | jblack | bminish: No idea. |
| 12:47.08 | bminish | we suspect a PRI issue but provider says no. |
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| 12:48.23 | jblack | You have the money. The provider wants it. |
| 12:49.42 | jblack | You can tell them 'your fault or no, if we can't this fixed, then we're not getting what we need' |
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| 13:02.26 | fantasticmrfox | Lo, in my DNS am I mean to use... e2u+iax2 |
| 13:02.35 | fantasticmrfox | or e2u+iax ... for ENUM resolution |
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| 13:03.15 | fantasticmrfox | e2u+iax:iax2 maybe? |
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| 13:18.28 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon 2008! #astric |
| 13:18.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:18.48 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon |
| 13:21.01 | Alton2 | good |
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| 13:28.59 | amorphka77 | Do I have to modify zaptel.conf in order to create channel groups or it is enough to do it in zaptel.conf? I have a ISDN PRI connection. |
| 13:29.37 | LemensTS | Im trying to get 2 phones to connect to an asterisk at another location (nat ewww) I set the phones to NAT in sip.conf, then at the location that the asterisk box is at i set the DSL router to forward port 5060 to the asterisk box. Then on the phones i give them the public IP address of the DSL Modem... and i get this http://pastebin.com/m56540735 |
| 13:29.56 | LemensTS | It registers them, then right away they go unreachable |
| 13:30.48 | LemensTS | amorphka77: zapata.conf |
| 13:31.21 | [TK]D-Fender | ~sipnat |
| 13:31.22 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:31.23 | [TK]D-Fender | ^^^^^ |
| 13:31.59 | amorphka77 | LemensTS: How can I make the changes in zapata.conf take effect? Asterisk restart is enough? |
| 13:32.00 | [TK]D-Fender | amorphka77: Are you reverring to grouping channel within a PRI, or crossing spans? |
| 13:32.13 | [TK]D-Fender | amorphka77: Yes, restart * or reload chan_zap.so |
| 13:32.36 | amorphka77 | Fender: No crossing, I have only 1 span. It is Didium TE120p |
| 13:34.28 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:34.49 | [TK]D-Fender | amorphka77: then that would be zapata.conf only |
| 13:36.15 | *** join/#asterisk oilinki (n=oil@ppp-124-120-10-118.revip2.asianet.co.th) |
| 13:36.47 | amorphka77 | Fender: Thanks! What do You mean when you say "restart *"? |
| 13:39.20 | *** join/#asterisk riddlebox (n=adfad@67-65-158-106.ded.swbell.net) |
| 13:39.34 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:40.50 | [TK]D-Fender | amorphka77: Restart Asterisk. How much more is there to say on this? |
| 13:41.17 | *** join/#asterisk henningw (n=henningw@et-1-4.gw-nat.bs.ka.oneandone.net) |
| 13:41.33 | *** part/#asterisk henningw (n=henningw@et-1-4.gw-nat.bs.ka.oneandone.net) |
| 13:41.59 | amorphka77 | Fender: Sorry, but I would like to be sure. Is it true that if I want to create channel groups to route incomming calls from different extensions to different context, than it is not enough to configure Astersik only. I have to tell my provider too to route calls to the appropriate channels. Right?! |
| 13:42.29 | LemensTS | TK: thanks i got the NAT working from that first site. |
| 13:43.01 | [TK]D-Fender | amorphka77: You have the telco direct certain DID's to certain B-chans already? |
| 13:43.08 | UD | does ipkall have limit on amount of calls made to it? |
| 13:44.02 | *** join/#asterisk Knightfal (n=jjj@66.178.134.235) |
| 13:44.48 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
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| 13:46.52 | amorphka77 | Fender: No grouping on telco side. |
| 13:47.29 | amorphka77 | Fender: Incomming calls just hit Asterisk on a random channel, and that's it... |
| 13:48.08 | *** part/#asterisk LemensTS (n=matthew@adsl-70-130-218-70.dsl.stlsmo.swbell.net) |
| 13:48.51 | [TK]D-Fender | amorphka77: Since you see the DID that was dialed what are you hoping to do by grouping your channels? |
| 13:51.10 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 13:53.16 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:53.28 | eric_hill | I'm trying to get the right combination of call-limit and limitonpeers so that I can have multiple outbound calls from a phone, but only one inbound call at a time to a phone. What's the right way to do this? |
| 13:53.42 | amorphka77 | Fender: I would like to dedicate certain channels to certain DID, in order to limit the channel consumption of a DID. Because the two DIDs are concurrents to eachother. I want to provide constant resources. |
| 13:54.19 | lowtek | amorphka77: GROUP() |
| 13:54.26 | amorphka77 | Fender: new PBX or new card is not an option |
| 13:54.55 | lowtek | amorphka77: If you want to limit channels by did your only option is to use groups ... |
| 13:55.26 | [TK]D-Fender | amorphka77: Ok, I just wanted to confirm your goals. Indeed the telco has to restrict channel usage on THEIR side, and you may want to use zapata groupings to restrict your outbound dialing possibly as well. |
| 13:55.54 | [TK]D-Fender | lowtek: No, this is not under your control, it is the telco |
| 13:56.24 | [TK]D-Fender | eric_hill: just check if they are on a call before actually dialing them. |
| 13:56.29 | lowtek | [TK]D-Fender: Ahh. |
| 13:56.35 | amorphka77 | Fender: Ok, that is what I wanted to hear :) Thanks a lot! |
| 13:56.41 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:57.37 | *** join/#asterisk grufft3ch (n=grufft3c@70.114.145.96) |
| 13:58.54 | eric_hill | [TD]D-Fender: I assume you just mean ChanIsAvail/GotoIf before the Dial, right? |
| 13:59.07 | *** join/#asterisk moy (n=moy@nat/ibm/x-7b33d2688557d58f) |
| 13:59.47 | grufft3ch | ~sipnat |
| 13:59.47 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:02.11 | *** join/#asterisk XnOSX (n=XnOSX@212.145.175.26) |
| 14:04.20 | amorphka77 | eric_hill: ChanIsAvail can be a solution but if you make channel groups no need for it. |
| 14:05.00 | *** part/#asterisk amorphka77 (i=icechat5@wlan-217-79-139-207.pannon.hu) |
| 14:06.51 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 14:07.12 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 14:07.40 | [TK]D-Fender | eric_hill: Yes |
| 14:08.23 | grufft3ch | I've setup the nat settings recommeneded on the ~sipnat article from the bot here, and those didn't seem to make a difference. My Polycom 330 is able to register, and even start a call, however, there is no sound. |
| 14:08.28 | grufft3ch | I've also read that Polycom's have crappy NAT due to RTP -- I've disabled ALL inspection on our Office Firewall (CiscoASA5505 & Allowed All ports / protocols to the server( for now, will tighten up once I fix this) And it still doesn't seem to work. |
| 14:08.32 | *** join/#asterisk IsUp (n=nocturne@88.243.216.154) |
| 14:08.37 | IsUp | hello |
| 14:09.07 | grufft3ch | Even on a Whitelisted server, completely, Why would i be having sound problems? The phone and server ARE able to talk, shown by the fact that the remote phone is able to register |
| 14:09.11 | grufft3ch | as well as start a call |
| 14:09.18 | *** part/#asterisk matt_keys (i=smatt@usaregs.com) |
| 14:09.23 | grufft3ch | however, Sound hates upon me |
| 14:09.30 | [TK]D-Fender | grufft3ch: Would help if you described all of the networking involved and show us your configs |
| 14:10.15 | IsUp | I am using ss7. my telco provides CICs to me which stars from 1 to 255. but i am not able to see some CICs when i type "ss7 show channels" |
| 14:10.27 | mort_gib | What happens if Kernel timing is not 1000 hz ?? |
| 14:10.47 | lowtek | grufft3ch: Once registered, if your phone can continue to send sip packets to asterisk (i.e., if you sip debug your peer and try to make a call and see the decoded packets) then focus on RTP. I don't know about the ASA, but make sure there's no cisco-specific SIP stuff enabled or configured. |
| 14:10.52 | IsUp | for example, CIC 31 is not shown, CIC 48 not shown too. |
| 14:13.00 | [TK]D-Fender | mort_gib: You get timing issues for MoH, zaptel (if using ztdummy), etc |
| 14:13.05 | lowtek | grufft3ch: it's going to be with the ASA obviously, Polycom phones work great with NAT. |
| 14:13.06 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:14.24 | mort_gib | OK, but suppose that it looks like the default kernel has been setup with 250hz and MOH and Conference calls are fine... |
| 14:14.40 | mort_gib | Like not stutter or anything. |
| 14:15.51 | [TK]D-Fender | mort_gib: What are calls coming in on? |
| 14:15.58 | mort_gib | Sangoma cards |
| 14:16.12 | [TK]D-Fender | mort_gib: Now please read what I said again. |
| 14:16.39 | mort_gib | :-) ISDN2 -> Sangoma MDG -> * |
| 14:16.42 | [TK]D-Fender | mort_gib: This only applies when using ZTDUMMY for your timing source. You aren't |
| 14:17.17 | mort_gib | Well, I have had to load ztdummy because * wont start using IAX in trunk mode until I do so. |
| 14:18.00 | mort_gib | I know Sangoma cards comes with their own timing device |
| 14:18.47 | [TK]D-Fender | mort_gib: :/ |
| 14:19.06 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:19.07 | *** join/#asterisk mog (n=mog@nat/digium/x-1c90d8a34887e30b) |
| 14:19.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:19.27 | *** join/#asterisk brettnem (n=brettnem@user-387oe5t.cable.mindspring.com) |
| 14:19.43 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:19.49 | [TK]D-Fender | mort_gib: well as long as its working, whatever right? |
| 14:20.01 | IsUp | I am using chan_ss7. my telco provides 1 to 255 CICs to me. but i am not able to see some CICs when i type "ss7 show channels", like CIC 33, CIC 48.. |
| 14:20.40 | mort_gib | Yes and NO! I would prefer to understand 100% |
| 14:22.01 | *** part/#asterisk retak (n=kater@dslb-084-059-137-147.pools.arcor-ip.net) |
| 14:22.39 | *** join/#asterisk pud (n=jkatka@sea01-v502-nat.marchex.com) |
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| 14:27.54 | *** join/#asterisk brettnem (n=brettnem@user-387oe5t.cable.mindspring.com) |
| 14:28.22 | IsUp | i need some help about chan_ss7 configuration. anyone able to help? |
| 14:28.40 | *** join/#asterisk stencil (n=stencil@76-10-147-63.dsl.teksavvy.com) |
| 14:29.32 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:29.32 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:30.45 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:32.22 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 14:33.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:34.08 | Dr-Linux|home | When i dial in through my SIP provider and enter any DTMF .. asterisk revices each digits twice, like if i dial 4092 my asterisk understand it as 44009922, how can i fix this DTMF problem? |
| 14:35.01 | lmadsen | Dr-Linux|home: what version of asterisk? |
| 14:35.34 | Dr-Linux|home | lmadsen: 1.4.17 |
| 14:35.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:39.56 | lowtek | Dr-Linux|home: Also post your sip.conf and give us some details about your provider. |
| 14:40.18 | lmadsen | almost sounds like you're getting both inband and out of band signalling |
| 14:41.59 | Dr-Linux|home | lowtek: well, currently i'm using dtmfmode=2833 in my sip.conf and having this issue, but if i use inband or info then asterisk doesn't any dtmf |
| 14:42.09 | *** join/#asterisk brettnem (n=brettnem@user-387oe5t.cable.mindspring.com) |
| 14:42.27 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 14:43.06 | *** join/#asterisk ManxPower (n=manxpowe@11.sub-70-222-50.myvzw.com) |
| 14:43.36 | Dr-Linux|home | lowtek: can you please any question about sip.conf you want to ask? |
| 14:43.51 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 14:44.06 | *** join/#asterisk Mind-Riot (n=James@65.111.208.79) |
| 14:44.12 | UD | i've been trying to get sip.conf to work for a few days now :( |
| 14:44.51 | *** join/#asterisk Great_Anta_Baka (i=c4219f6d@gateway/web/ajax/mibbit.com/x-1a5eccde584a9c1d) |
| 14:44.58 | UD | and all i want to do is Answer() a call and Playback(hello-world) |
| 14:48.17 | linuxstb | UD: You'll need to provide a little more info than that... e.g. is the sip device registering OK with asterisk? What is the console output when you dial your test extension? Pastebin your sip.conf... |
| 14:49.12 | ManxPower | UD: put a copy of your sip.conf and extensions.conf on pastebin.ca. Mask ONLY passwords. |
| 14:50.14 | Dr-Linux|home | lowtek: it just works fine if i call from landline |
| 14:50.38 | ManxPower | I will only ask once. |
| 14:51.37 | UD | ok |
| 14:52.38 | *** join/#asterisk spokra (n=spokra@host093-179-144.sea0.speakeasy.net) |
| 14:52.38 | techman97 | UD: you need to read the book they were talking about last night, or visit the Asterisk WIKI |
| 14:52.53 | UD | first though i should mention that i am not using a SIP "device" but rather i have a free SIP provider and am using ipkall as the "device" |
| 14:52.55 | IsUp | I am using chan_ss7. my telco provides 1 to 255 CICs to me. but i am not able to see some CICs when i type "ss7 show channels", like CIC 33, CIC 48. |
| 14:53.02 | techman97 | what you're looking to do is rudimentary, and you'll have to be able to do it to do anything else imho |
| 14:53.09 | ManxPower | IsUp: I doubt anyone here uses SS7. |
| 14:53.14 | UD | ekiga to be specific about the service |
| 14:53.48 | ManxPower | IsUp: It sucks, I know. Your best bet might be the Asterisk users or maybe asterisk-ss7 (if it's still alive) mailing lists. |
| 14:53.59 | ManxPower | see lists.digium.com or |
| 14:54.02 | ManxPower | ~mailinglist |
| 14:54.02 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 14:54.12 | UD | also the service works great when i use a soft phone or something to recieve calls |
| 14:54.28 | ManxPower | UD: nobody is going to help you without pastebina |
| 14:54.31 | UD | that being said, here comes my config files. ' |
| 14:54.44 | UD | ipkall just gives me a busy signal when i dial it |
| 14:55.30 | *** join/#asterisk spokra (n=spokra@host093-179-144.sea0.speakeasy.net) |
| 14:55.33 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 14:56.12 | IsUp | ManxPower thanks |
| 14:56.27 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 14:56.38 | ManxPower | IsUp: SS7 is very very uncommon in the USA for customer links. |
| 14:56.48 | [TK]D-Fender | Very uncommon period |
| 14:56.59 | JT | not uncommon in telco networks |
| 14:57.21 | [TK]D-Fender | JT : jsut talking normal consumers here... |
| 14:57.22 | ManxPower | [TK]D-Fender: In some parts of the world the telco prefers SS7 over PRI. |
| 14:57.32 | [TK]D-Fender | ManxPower: Yeah, I met that guy too ;) |
| 14:57.39 | IsUp | It's common in Turkey :> |
| 14:57.45 | [TK]D-Fender | ManxPower: Now he need only become a plural ;) |
| 14:58.48 | Mind-Riot | ok heres a question for anyone who wants to take a stab at it... I have a sip peer out on our WAN that is calling a sip phone I have here on my server LAN through our asterisk server. When I call through the asterisk server its choppy as all hell. Using G711u always. The thing is when I call the phone using IP-to-IP call its perfectly clear. Anyone know where I should start troubleshooting this? |
| 14:58.51 | UD | http://pastebin.com/m2bf35eec |
| 14:58.56 | variable_office | Huh, strange issue here. I have a client now that in the sip packet the from is marked as unknown@<ip> the client is registered with the proxy though, any ideas? |
| 14:59.34 | ManxPower | Mind-Riot: loading ztdummy could cause that. |
| 15:00.55 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 15:01.09 | lowtek | Mind-Riot: ulaw will require 64k both directions, do you have the bandwidth available? What's the latency look like? |
| 15:01.14 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 15:01.30 | Mind-Riot | Bandwidth is there. Latency around 35ms |
| 15:01.31 | ManxPower | UD what ports did you forward? |
| 15:01.35 | Mind-Riot | Jitter 9ms |
| 15:01.57 | lowtek | Mind-Riot: Have you tried a lower bandwidth codec to see if it improves? |
| 15:02.10 | Mind-Riot | yes |
| 15:02.11 | ManxPower | Mind-Riot: does "lsmod" show ztdummy? |
| 15:02.17 | UD | 5060 |
| 15:02.19 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:02.19 | lowtek | ManxPower: Just because I really don't know, how would ztdummy cause this? |
| 15:02.21 | Mind-Riot | Tried g729 and same results |
| 15:02.24 | *** join/#asterisk sant0sk1 (n=sant0sk1@wsip-68-15-235-89.om.om.cox.net) |
| 15:02.33 | ManxPower | lowtek: happens when the kernel HZ is not 1000 |
| 15:02.50 | lowtek | ManxPower: Ahh. Interesting. Learn something new every day. |
| 15:02.51 | UD | and on the same system as asterisk is installed i ran a soft phone program for troubleshooting purposes, call sent and recieved audio fine |
| 15:02.51 | ManxPower | usually when it's kernel HZ 100 or something in many distros. |
| 15:02.57 | JT | lowtek: actually, ulaw will require close to 85kbit/s over sip with 20ms packetisation |
| 15:02.59 | ManxPower | lowtek: it's a very common issue. |
| 15:03.06 | lowtek | JT: I stand corrected. |
| 15:03.21 | ManxPower | UD: I really can't help you as you must not have read all the NAT docs. |
| 15:03.23 | JT | it has more overhead than TDM |
| 15:03.26 | ManxPower | You need to forward more ports. |
| 15:03.31 | ManxPower | ~sipnat |
| 15:03.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:03.36 | ManxPower | go read it AGAIN |
| 15:04.19 | ManxPower | lowtek: the symptom is typically "playback is choppy", but it could cause the issue Mind-Riot is having. Too bad he doesn't want my help. |
| 15:04.33 | UD | i tried without NAT first i've just been trying different things. im not sure why i did that one. |
| 15:04.40 | UD | by forward, i meant forward in my router |
| 15:04.50 | ManxPower | UD: you never forwarded the RTP port. |
| 15:05.01 | UD | no |
| 15:05.06 | ManxPower | well, the RTP audio UDP ports. |
| 15:05.15 | ManxPower | It should all be documented in the docs. |
| 15:05.35 | UD | why would i need to if it works for other softphone with the same service |
| 15:05.41 | ManxPower | UD: I'm not here to tutor you. |
| 15:05.48 | ManxPower | UD: because a softphone is not a sip server. |
| 15:06.15 | UD | i made a call to the sip phone with my cell phone |
| 15:06.16 | ManxPower | you don't need to do any port forwarding if you use a SIP client, only if you have a SIP server behind NAT. |
| 15:07.12 | Mind-Riot | Anyone know how to uload ztdummy? |
| 15:07.25 | ManxPower | UD: But I'm not going to argue with you and I'm not going to go tell you to read the specific docs that were written to help with your specific issues. |
| 15:07.35 | ManxPower | Mind-Riot: stop asterisk, rmmod ztdummy |
| 15:07.52 | Mind-Riot | ManxPower: thanks |
| 15:10.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:11.00 | UD | heh what i want to do is exactly what i have been doing successfully with another program so you figure its because i didnt open ports |
| 15:11.30 | variable_office | what would cause a sip client that normally forwards its <username>@<ip> to send unknown@localhost |
| 15:11.51 | ManxPower | variable_office: username= in the [sipconfsections] |
| 15:12.01 | [TK]D-Fender | UD: its all layed out in the guide you have been provided. Maybe you should read it |
| 15:12.05 | UD | i had port 10000->20000 open before and it did not work, but since i stopped forwarding those ports used the service to ANSWER a call with one program, i figured i could do so with another program |
| 15:12.14 | ManxPower | [TK]D-Fender: I gave up on him. |
| 15:12.35 | variable_office | ManxPower, nothig on the server has changed, and nothing on the ata as far as I know |
| 15:13.06 | ManxPower | variable_office: then you are hallucinating. Things don't just break when nothing is changed. |
| 15:14.23 | variable_office | no configurations changed at least |
| 15:14.50 | UD | ManxPower since you're not here to tutor anyone, why even respond. |
| 15:15.46 | ManxPower | UD: to help people with actual questions that might not be in the documentation. |
| 15:16.02 | phix | hi |
| 15:16.11 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:17.05 | variable_office | ManxPower, it is the strangest thing, just happened this morning, and now everything is being passed as from client unknown from the one peer |
| 15:17.09 | [TK]D-Fender | UD: Go follow the guide. When you done so, show us useful backup of your problem. |
| 15:17.52 | ManxPower | UD: once you get your system setup according to the guide, if you have further problems I will be happy to help you. |
| 15:17.58 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 15:18.26 | ManxPower | [TK]D-Fender: he spends much of his time talking about things not relevant to the issue and tends to tell you why you can't be right. Quite annoying actually. |
| 15:20.11 | [TK]D-Fender | ManxPower: I'll just wait and see if the dust settles. |
| 15:20.24 | *** part/#asterisk sant0sk1 (n=sant0sk1@wsip-68-15-235-89.om.om.cox.net) |
| 15:21.14 | Blackvel | manager api. how can I debug it? |
| 15:21.19 | *** join/#asterisk AiT (n=airani@66-146-175-59.skyriver.net) |
| 15:21.31 | Blackvel | any cli command for debuggin? |
| 15:21.34 | [TK]D-Fender | Blackvel: RAID :) |
| 15:21.50 | Blackvel | rtmf? :) |
| 15:22.14 | Blackvel | err rtfm ? :) |
| 15:22.16 | Blackvel | whats raid? |
| 15:22.17 | ManxPower | Blackvel: I'm not aware of any manager debug stuff. You might be able to turn on high verbosity or debug level. Other than that you would have to put debugging statements in your code. |
| 15:22.31 | ManxPower | Blackvel: the answer to your question is not in any docs I know of. |
| 15:22.44 | Blackvel | this phonesuite cti software does something and nothing happens... |
| 15:22.59 | Blackvel | would be cool to check on asterisk server. |
| 15:23.16 | Blackvel | ManxPower: thanks. I almost feared that I miss something :) |
| 15:24.02 | ManxPower | Blackvel: maybe you can tcpdump the interface the software uses to talk to manager? |
| 15:24.05 | brodiem | Blackvel: dump packets on the manager's listening port |
| 15:24.07 | ManxPower | or wireshark really |
| 15:24.24 | Blackvel | sometimes I can even feel that I haven't done anyhting the last 3 years with asterisk :) |
| 15:24.27 | Mind-Riot | ManxPower: I unloaded ztdummy and is still choppy as hell. Any other thoughts? |
| 15:24.37 | Blackvel | maybe theres a log option in phonesuite |
| 15:24.42 | ManxPower | Blackvel: RAID is a product in the USA to kill 6 legged bugs |
| 15:24.42 | Blackvel | quite cool software btw |
| 15:25.03 | Blackvel | oh |
| 15:25.11 | Blackvel | nah sw is not buggy |
| 15:25.22 | Blackvel | I swear it was working before (at least together with snom) |
| 15:25.34 | ManxPower | Mind-Riot: diagram your setup. Like "PRI -> Sangoma/Asterisk -> LAN -> SIP phone" or whatever is correct for your system. |
| 15:26.39 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:30.21 | IsUp | ah i like using Raid too. summer nights, you know. so many bugs |
| 15:30.28 | IsUp | raid'em all |
| 15:31.06 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:31.06 | Mind-Riot | ManxPower: SIP Phone (PAP2) -> Router -> WAN Backbone (45mbit) -> Router -> L3 Switch -> Asterisk -> L3 Switch -> Sip Phone (PAP2)... This scenario is choppy. |
| 15:31.11 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:31.41 | Mind-Riot | ManxPower: SIP Phone (PAP2) -> Router -> WAN Backbone (45mbit) -> Router -> L3 Switch -> Sip Phone (PAP2)... IP-IP Call This scenario is perfect. |
| 15:32.17 | ManxPower | Mind-Riot: You might not be happy with any of the things I can think of to try. |
| 15:32.47 | ManxPower | Many motherboards have incredibly HORRIBLE built in ethernet interfaces. Try disabling the one on the motherboard and try with a card you know works well. |
| 15:32.51 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 15:33.32 | ManxPower | We had a mail server that could not keep a connection for more than 30 seconds and an SSH session would disconnect anytime lots of data was being displayed. Put in a good network card and it worked great. |
| 15:34.42 | Qwell | ManxPower: did you see the one where email couldn't be sent to some *physical* locations? |
| 15:34.52 | Qwell | ie: SF>NY was bad, SF>LA was fine |
| 15:35.12 | ManxPower | Qwell: Nope. I'd just go back to bed if I heard of someone with that problem. |
| 15:35.27 | Qwell | ManxPower: turns out...it was actually happening |
| 15:36.25 | Qwell | I forget exactly what it was, but it was something with very short mail timeouts. Email to close locations could be delivered fine, but places too far away would timeout before it could finish |
| 15:36.35 | ManxPower | Qwell: I did hear of some brand of router used in Aussieland that would crash if it saw some specific string in an FTP transaction. |
| 15:37.17 | *** join/#asterisk sircco (n=sircco@nlab.iskon.hr) |
| 15:37.25 | sircco | hello, anyone here using openvox card? |
| 15:38.27 | sircco | if i use misdn do i have to get good result from ztcfg ? |
| 15:38.42 | ManxPower | sircco: very few people use that card |
| 15:38.43 | brodiem | Qwell: a screwy ECN stack maybe? |
| 15:38.49 | ManxPower | at least very few people on this channel. |
| 15:39.10 | sircco | ManxPower: well then... anyone using misdn then |
| 15:39.46 | *** join/#asterisk viperdude (n=viperdud@87.127.75.241) |
| 15:39.48 | ManxPower | sircco: if you can't find help here, try the mailinglist archives. |
| 15:39.52 | ManxPower | ~mailinglist |
| 15:39.52 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 15:40.10 | sircco | i'll try here first, looking on google at same time, tnx! :) |
| 15:40.12 | JT | misdn doesn't use ztcfg |
| 15:41.55 | sircco | jk: ok tnx, i know this is basic stuff :) |
| 15:42.27 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 15:44.41 | UD | maybe theres another pbx software thats somewhat comparable to asterisk? |
| 15:45.27 | ManxPower | UD: YATE is one. OpenPBX is another. |
| 15:45.54 | JT | openpbx is long gone |
| 15:46.00 | JT | at least the name you're thinking of |
| 15:46.03 | ManxPower | Now called Callweaver |
| 15:46.31 | UD | im starting to think i have a corrupted asterisk |
| 15:47.27 | JT | callwear dev seems pretty dev |
| 15:47.39 | JT | most people seem to be working on other projects like freeswitch atm |
| 15:47.48 | JT | s/pretty dev/pretty dead/ |
| 15:49.19 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:50.13 | jeev | pump, pump the jam |
| 15:52.57 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-e9f375096de868fa) |
| 15:52.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:54.10 | *** part/#asterisk moy (n=moy@nat/ibm/x-7b33d2688557d58f) |
| 15:56.43 | jaytee | oh dear god |
| 15:57.15 | jeev | seriously |
| 15:57.20 | jeev | what are we oh dear god'ing at |
| 15:57.30 | UD | slaps jeev around a bit with a large penis pump |
| 15:57.31 | Qwell | you |
| 15:57.42 | jaytee | yes, what Quell said |
| 15:58.03 | [TK]D-Fender | UD: "corrupted asterisk"? Where do you get this bizarre idea from? |
| 15:58.13 | jaytee | maybe he's in politics? |
| 15:58.13 | jeev | die |
| 15:58.32 | jeev | you damn mccain fans |
| 15:59.05 | UD | well i have no idea whats wrong but i'm about to call it quits. everyone links me to documents. i try the examples, they dont work. i dont know what's wrong, nobody in here knows whats wrong, this is obviously a problem |
| 15:59.08 | variable_office | http://pastebin.ca/1194929 --> this doesnt make sense, it says it is dialing from sipuser3, then it doesnt pass the correct username when dialing out? |
| 15:59.35 | Qwell | ~book |
| 15:59.35 | 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, or see ~buybook |
| 15:59.37 | Qwell | UD: try there |
| 15:59.37 | [TK]D-Fender | UD: And you admit not having forward needed ports, and not once did you actually provide SIP debug for a call attempt. |
| 15:59.40 | jeev | UD: i swear that's not my bag baby |
| 16:01.12 | [TK]D-Fender | UD: I have a few pretty good ideas what it wrong and you aren't looking at crucial stuff. |
| 16:01.34 | UD | i did forward the ports, just now for the heck of it. i had it opened earlier |
| 16:02.00 | UD | anyway where do i get the debugging info from? |
| 16:02.16 | [TK]D-Fender | UD: * CLI - "sip debug" |
| 16:03.33 | UD | does this save this to a log or something? |
| 16:03.55 | variable_office | UD, it is displayed onscreen |
| 16:05.05 | UD | i have a lot of crap on screen when i do that. im guessing its old stuff though |
| 16:05.55 | UD | SIP Debugging re-enabled |
| 16:05.56 | UD | debian*CLI> |
| 16:06.01 | UD | i do a call and get nothing |
| 16:08.08 | ManxPower | It's not going to work at all if you don't forward the RTP ports. |
| 16:08.46 | UD | they're forwarded |
| 16:08.47 | ManxPower | I'll even quote it for you: "Our Asterisk server will also have to have ports 5060 (UDP), and the port range specified in ârtp.confâ (typically 10000-20000 UDP) forwarded to it. NEITHER of the remote phones connecting to our server need any forwarding." |
| 16:11.03 | ManxPower | UD: you almost always have to register with a provider. That, I imagine, is your next step. |
| 16:11.40 | IsUp | how can i trigger a context from Console? |
| 16:11.47 | IsUp | or an AGI |
| 16:11.51 | ManxPower | IsUp: you really can't. |
| 16:11.58 | UD | as i've stated earlier, i have a provider that i have setup with my sip phone software on the same machine that successfully recieves calls |
| 16:12.00 | ManxPower | define "trigger a context" |
| 16:12.23 | ManxPower | UD: and as I stated earlier that does not mean much. |
| 16:12.43 | ManxPower | Now do you want my help to proceed or not? |
| 16:12.46 | IsUp | so what's the Dial command on Asterisk CLI? |
| 16:12.58 | UD | sure |
| 16:13.05 | UD | what does register with a provider mean? |
| 16:13.17 | ManxPower | IsUp: there is only a CLI dial if you have alsa/oss installed. |
| 16:13.21 | variable_office | ManxPower, i post a pastebin that shows what i was talking about. it runs dial(sipuser3, num) but then in the sip debug it says from: unknown that doesn't make sense |
| 16:13.21 | JT | UD: do you have a register line in sip.conf? |
| 16:13.24 | ManxPower | IsUp: asterisk is not a softphone. |
| 16:13.41 | ManxPower | UD: usually it would be register => username:password@providersipserver.com |
| 16:13.54 | ManxPower | in sip.conf [general] You can see examples of this in sip.conf.sample. |
| 16:14.13 | ManxPower | variable_office: I am not a SIP guru. |
| 16:14.26 | ManxPower | And even if I was, debugging SIP messages is not something I do for free. |
| 16:14.59 | variable_office | it isnt really debugging sip messages, asterisk goes from knowing the username to being clueless |
| 16:15.13 | iXombie | lol |
| 16:15.16 | IsUp | on Dial command i get |
| 16:15.17 | IsUp | Unable to re-open DSP device /dev/dsp: No such file or directory |
| 16:15.19 | IsUp | any ideas? |
| 16:15.33 | ManxPower | IsUp: you do not have your sound drivers loaded |
| 16:15.40 | IsUp | ah |
| 16:15.41 | iXombie | Is there a way to get voicemail to notify multiple extentions without using a voicemail group? |
| 16:15.47 | ManxPower | Do you have speakers/headset and microphone connected? |
| 16:15.50 | IsUp | via SSH i cant hear sounds anyway |
| 16:15.58 | ManxPower | iXombie: yes, |
| 16:16.00 | IsUp | okay Manx, thanks |
| 16:16.34 | ManxPower | iXombie: well, not multiple EXTENSIONS, but multiple DEVICES yes. |
| 16:16.36 | iXombie | ManxPower: What would I change for this? |
| 16:16.46 | *** join/#asterisk i9tech (n=arthurh@70-57-250-186.albq.qwest.net) |
| 16:16.49 | ManxPower | iXombie: mailbox=vmbox@vmcontext |
| 16:16.50 | iXombie | that would be be good too |
| 16:17.23 | iXombie | on the device? or in voicemail.conf? |
| 16:17.27 | [TK]D-Fender | UD: You do not register to IPKALL. You provide them a full URI to dial. If you have forwarded all the proper ports on your router & you do not have a bad firewall in the way (including your * machine itself), then you have either set up IPKALL improperly on their service, or the traffic is being blocked |
| 16:17.41 | ManxPower | if you want that peer to be notified of multiple mailboxes than mailbox=vmbix1@vmcontext1,vmbox2@vmcontext2 |
| 16:18.05 | ManxPower | [TK]D-Fender: I think he is trying to get incoming working too. |
| 16:18.19 | [TK]D-Fender | ManxPower: IPKALL only does incoming... |
| 16:18.20 | iXombie | ManxPower: Where do i put this? in voicemail.conf? |
| 16:18.41 | ManxPower | iXombie: NO! sip.conf, zapata.conf, mgcp.conf, sccp.conf, h323.conf |
| 16:18.51 | ManxPower | The DEVICE config files in Asterisk |
| 16:19.03 | ManxPower | [TK]D-Fender: interesting |
| 16:19.35 | iXombie | ManxPower: Ahh, sweet thank you. Still learning. There is a lot to take in. |
| 16:19.57 | ManxPower | iXombie: much of it is in The Good Book. |
| 16:19.59 | ManxPower | ~book |
| 16:20.00 | 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, or see ~buybook |
| 16:20.23 | iXombie | ManxPower: havent had time to read it. Most of it I have been able to wing. |
| 16:20.25 | ManxPower | bbiab |
| 16:20.37 | ManxPower | iXombie: No. You do not have time to NOT read it. |
| 16:20.44 | iXombie | lol |
| 16:20.50 | ManxPower | Learning the book will save you countless hours of work |
| 16:22.49 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 16:25.13 | UD | i did see the register => line in the sip.conf.sample, under the ;------ OUTBOUND SIP REGISTRATIONS ------ section. Does this mean i still have to use it even though i'm not planning on making any outbound calls? |
| 16:26.13 | [TK]D-Fender | UD: What did I just tell you? You do not register to IPKALL. You provide them the URI to dial to reach you. |
| 16:26.38 | [TK]D-Fender | UD: and registering has nothing to do with outbound calls. |
| 16:27.03 | UD | i was replying to JT |
| 16:27.29 | UD | [TK]D-Fender thats what i though, i dont use ipkall |
| 16:27.47 | [TK]D-Fender | UD: well thats who I see you talking about. |
| 16:27.48 | lowtek | Wow. Google Chrome is fast. Too bad it's only for windows. |
| 16:27.54 | UD | err i mean |
| 16:27.59 | UD | yes, i do use it |
| 16:28.02 | UD | but this is not the problem |
| 16:28.03 | [TK]D-Fender | UD: Did I miss your mentioning someone else earlier? |
| 16:28.14 | [TK]D-Fender | UD: Perhaps you could rediscribe your problem. |
| 16:28.16 | UD | nah sorry brainfart |
| 16:28.34 | [TK]D-Fender | UD: If you can't keep your thoughts together then we're all wasting time. |
| 16:29.10 | UD | ok i use ipkall but lets pretend i dont because as i've said over and over the procedure works with anything else SIP but asterisk |
| 16:29.16 | *** part/#asterisk stencil (n=stencil@76-10-147-63.dsl.teksavvy.com) |
| 16:29.25 | rob0 | Wasting time, in IRC? Say it's not so!!! |
| 16:29.41 | UD | i dial my ipkall # on my cell phone and my soft phone is running with my info, it works |
| 16:30.01 | UD | so i think we can safely illiminate troubleshooting ipkall, unless i am missing something |
| 16:30.06 | rob0 | ipkall in * is simple enough that I was able to do it |
| 16:30.43 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 16:32.29 | [TK]D-Fender | UD: What exactly did you forward on your router to *? |
| 16:32.36 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 16:33.04 | UD | 5060/udp/tcp |
| 16:33.13 | [TK]D-Fender | UD: Check your local firewall |
| 16:33.14 | UD | 10000-20000/udp/tcp |
| 16:33.28 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 16:33.28 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:34.29 | UD | in the same section is port 22 forwarded that suddenly makes ssh stop working from external locations when i unforward it |
| 16:38.13 | UD | firewall is disabled |
| 16:38.29 | UD | the router is set on default settings |
| 16:39.10 | UD | and i have no local firewall on the machine |
| 16:40.28 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 16:40.37 | SuPrSluG | hello all |
| 16:41.45 | hardwire | anybody seen any VoIP hardware that uses multicast dns for service discovery? |
| 16:41.55 | [TK]D-Fender | UD: pastebint he CLI output of an inbound attempt at verbose 10, SIP debug enabled. |
| 16:42.08 | SuPrSluG | OT: try to upgrade an old cisco phone (bargain) anyone know wher to get old discontinued cisco firmware. the upgrade needs to be a step process |
| 16:43.39 | *** join/#asterisk pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 16:44.32 | UD | verbose 10? |
| 16:44.50 | [TK]D-Fender | UD: yes |
| 16:45.00 | *** part/#asterisk jip|werk (n=asdf@h66-38-133-13.gtconnect.net) |
| 16:45.03 | UD | how do i set that? |
| 16:45.16 | [TK]D-Fender | UD: "set verbose 10" , "sip debug" |
| 16:49.18 | *** join/#asterisk steliosk (n=Stelios@79.107.85.140) |
| 16:50.13 | *** join/#asterisk s0ck (n=c1@unaffiliated/s0ck) |
| 16:50.37 | Blackvel | hey guys. please don't hit me but what do you think about the apstel visual dialplan tool for e.g IVR? |
| 16:50.52 | *** join/#asterisk hfb (n=hfb@96.247.52.127) |
| 16:51.04 | [TK]D-Fender | Blackvel: Why would you think any of us actually use it? |
| 16:51.15 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:51.30 | Blackvel | honestly I think you don't |
| 16:51.54 | s0ck | does anyone have blf working on a snom360? i can get the button to blink when it's ringing but it doesn't pick up the call. latest firmware. |
| 16:52.06 | Blackvel | aint much faster and extendable coding everything from hand (as I do right now)? |
| 16:52.11 | UD | there is no debugging information |
| 16:52.41 | Blackvel | aint it much faster and more extendable coding everything from hand? |
| 16:52.43 | s0ck | i think it might be a problem with subscribe...? |
| 16:52.47 | [TK]D-Fender | UD: If you see nothing then it is a networking issue and nothing to do with *. |
| 16:53.01 | UD | oops here it comes just lagged or something |
| 16:53.35 | [TK]D-Fender | UD: Either you did not put the proper IP in IPKALL's URI to reach you, your router's forwarding failed, or the packets never reached your router in the first place. |
| 16:54.07 | UD | http://pastebin.com/m3c8a0700 |
| 16:54.45 | [TK]D-Fender | UD: "options" is not a call attempt |
| 16:55.39 | UD | do you think ipkall is figuring out im using this for asterisk and preventing it? |
| 16:56.00 | [TK]D-Fender | UD: No. |
| 16:56.46 | [TK]D-Fender | UD: paste the URI you provided to IPKALL. |
| 16:58.25 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 16:58.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon |
| 16:58.50 | UD | well i didnt give it to ipkall in any sort of uri form |
| 16:58.54 | UD | i went to the site |
| 16:58.58 | UD | signed up |
| 16:59.10 | UD | gave it my sip proxy, ekiga.net, user name, and password |
| 16:59.34 | [TK]D-Fender | UD: Well gee isn't that nice to know after all this time... |
| 16:59.51 | [TK]D-Fender | UD: And where is all of your work to set up * for Ekiga's service? |
| 17:00.02 | [TK]D-Fender | UD: because that you DO have to register with <--- |
| 17:00.34 | UD | are you implying that ipkall isnt working? |
| 17:00.51 | UD | work? |
| 17:00.56 | UD | where is it supposed to be? |
| 17:01.34 | [TK]D-Fender | UD: No I'm saying that all the normal minimal method for setting them up is not the method you are using and you've now introduced your use of Ekiga's service after wasting hours not letting us know that something else might be wrong. |
| 17:02.40 | [TK]D-Fender | UD: pastebin "sip show registry" |
| 17:02.46 | pids | Anyone have any idea why two asterisk servers passing dtmf lose the timing? The call comes in SIP with the correct length, however when I pass it to another asterisk server via iax it arrives with a 0ms time. |
| 17:03.16 | UD | nothing |
| 17:03.17 | UD | Host Username Refresh State Reg.Time |
| 17:03.26 | s0ck | does anyone know if the target peer should acknowledge an inbound sip INVITE |
| 17:03.35 | s0ck | it doesn't seem to... |
| 17:03.43 | [TK]D-Fender | UD: great, a total waste of time. |
| 17:04.05 | [TK]D-Fender | UD: Go set up * to register with Ekiga's service and set up the peer to receive calls against |
| 17:04.51 | [TK]D-Fender | s0ck: BFL != SLA |
| 17:04.55 | UD | forgive me, but this is what i've been asking how to do all morning |
| 17:05.04 | s0ck | http://pastebin.com/d26b0a9d6 |
| 17:05.23 | s0ck | agreed... |
| 17:06.07 | s0ck | i have been reading up on how blf should work *in theory* but i'm still sketchy |
| 17:06.16 | s0ck | any literature explaining it would be great |
| 17:07.03 | [TK]D-Fender | s0ck: BLF is not sued to pick up a call. |
| 17:07.08 | [TK]D-Fender | used* |
| 17:07.40 | s0ck | right |
| 17:08.05 | s0ck | damn? |
| 17:08.10 | s0ck | i need bloody call pickup |
| 17:08.21 | [TK]D-Fender | UD: http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net |
| 17:08.55 | [TK]D-Fender | s0ck: And you think you're going to be able to control that with FreePBX owning your ass? Good luck. |
| 17:09.54 | s0ck | probably not but i'm gonna give it a bash :) |
| 17:10.01 | s0ck | (and a few minutes of your time, if you will) |
| 17:10.03 | s0ck | :P |
| 17:10.43 | s0ck | i guess gs were combining blf+call pickup then |
| 17:10.51 | s0ck | and wrapping it up and calling it blf |
| 17:10.56 | [TK]D-Fender | s0ck: I have absolutely no motivation currently to look twice at a FreePBX dialplan issue. |
| 17:11.10 | [TK]D-Fender | s0ck: What does "gs" have to do with this? |
| 17:11.43 | s0ck | blf, as i understood it less than 10 minutes ago does this blf/call pickup thing within five minutes of unboxing it |
| 17:11.54 | s0ck | i no longer use them tho, call quality etc |
| 17:12.20 | s0ck | i must have tried 5 different snom firmwares |
| 17:12.22 | [TK]D-Fender | s0ck: Unboxing? HUH!? |
| 17:12.29 | s0ck | and every option available to get call pickup working |
| 17:13.13 | [TK]D-Fender | s0ck: Call pickup is a dialplan issue and since you're using FreePBX, not even one we are going to be helping you with. Go ask in their support channel. |
| 17:13.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 17:13.38 | s0ck | i understand |
| 17:13.48 | s0ck | more interested in reading how call pickup should work tho |
| 17:13.54 | s0ck | got anything handy? |
| 17:15.40 | x86 | http://jalopnik.com/assets/images/jalopnik/2008/09/Timmay.jpg |
| 17:15.41 | x86 | rofl |
| 17:16.05 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 17:16.21 | lowtek | That's funny $h*t |
| 17:16.24 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 17:17.45 | Assimilate | Having a odd issue. I have a dialplan that sets a ringtone to a specific one based off of callerid. That works if I just ship the call to a Sip device, but if I add that call to a queue I lose the custom ringtone. Does the queue command change the alert_info? |
| 17:18.51 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:20.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:20.28 | [TK]D-Fender | s0ck: "core show applications like pick" |
| 17:21.05 | UD | im starting to see why hans reiser killed his wife |
| 17:21.06 | [TK]D-Fender | Assimilate: Pastebin the CLI output of your call attempt from beginning to end |
| 17:21.13 | [TK]D-Fender | UD: and that is? |
| 17:21.23 | lowtek | UD: ?? |
| 17:21.33 | lowtek | UD: Problems at home? |
| 17:21.44 | lowtek | UD: Try "core show application divorce" |
| 17:22.25 | UD | heh |
| 17:22.28 | *** part/#asterisk JenniferAkemi (n=akemi@206-248-161-97.dsl.teksavvy.com) |
| 17:22.29 | UD | <PROTECTED> |
| 17:22.47 | UD | im sure it was reiserfs bugs that invoked it |
| 17:22.55 | UD | he admitted himself that it sucked |
| 17:23.02 | UD | although the only thing he admitted to |
| 17:23.28 | lowtek | UD: The dude was clearly wacked. |
| 17:24.47 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:25.49 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 17:26.32 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 17:30.02 | lesouvage | Is it correct that SNOM and Linksys voip phones support LDAP out of the box? |
| 17:30.07 | Assimilate | [TK]D-Fender, http://pastebin.com/d7225da15 |
| 17:30.45 | Blackvel | lesouvage: I think I read about ldap on 370 |
| 17:31.03 | Blackvel | have not tried it yet |
| 17:31.14 | [TK]D-Fender | Assimilate: Executing Dial("Local/104@qexten-8161,2", "SIP/104|20|thm") in new stack <- |
| 17:32.00 | [TK]D-Fender | Assimilate: You are setting the ring info before you call queue. Inside of your queue that var gets passed (at best) to the LOCAL CHANNEL that is your agent. That channel in turn calls the actual SIP device but it is indeed lost by that point |
| 17:32.13 | [TK]D-Fender | Assimilate: You should do this in the extens that actually call your agents. |
| 17:32.49 | Assimilate | ah ok so all the settings are lost because its a new stack |
| 17:33.00 | [TK]D-Fender | Assimilate: Yup |
| 17:33.20 | [TK]D-Fender | Assimilate: modify the extens that call the actual devices |
| 17:33.26 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 17:33.28 | Assimilate | sweet thanks, now I got to wrap my head around the best way to do that. |
| 17:33.38 | Assimilate | cause I use the same qexten for all queue calls |
| 17:33.39 | UD | im going to install one of these in my computer http://www.pyramydair.com/images/acc/SDMSD87s.jpg |
| 17:33.44 | Mahmoud | any one here connects his asterisk to any of the online iax trunk providers? |
| 17:33.52 | [TK]D-Fender | Assimilate: 104@qexten <- good start |
| 17:33.57 | Mahmoud | i would like to connect my asterisk to one, so far i'm thinking of vyke |
| 17:34.31 | lesouvage | For Polycom you need extra licenses for ldap support bt I'm not sure what you get for the extra licenses. Does one of you have experience with ldap support on a Polycom Soundpoint IP330 and would you like to share that. |
| 17:36.04 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 17:40.03 | Blackvel | is there any benefit? |
| 17:40.46 | Blackvel | why not just include ldap on * (my plan is to do that for callerid(name) setting next days)? |
| 17:46.54 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 17:47.27 | Blackvel | bbl |
| 17:47.57 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpfg.cable.mindspring.com) |
| 17:48.51 | lowtek | Is there a good way to SELECT based on a value to the LEFT of a character? If my value is "www.whatever.com" is there an easy way to select whatever is to the LEFT of the first period (.)? |
| 17:49.08 | Mahmoud | any iax trunk provider? |
| 17:49.24 | lowtek | Oops, sorry guys, wrong window. |
| 17:49.25 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 17:50.52 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:51.32 | *** join/#asterisk airjump (n=airjump@p508ACB1E.dip.t-dialin.net) |
| 17:52.19 | *** join/#asterisk ManxPower (n=manxpowe@65.sub-70-220-105.myvzw.com) |
| 17:52.45 | Nugget | lowtek: in postgresql you'd do "SELECT WHERE substr(field,1,(strpos(field,'.')-1) = 'www'; :) |
| 17:56.30 | lowtek | Nugget: substring_index() is what I was after in mysql ... But thanks anyway ... So you prefer postgres? |
| 17:57.15 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 18:00.10 | ManxPower | *scream* Why do people think echo cancelers screws up faxes? |
| 18:00.21 | *** part/#asterisk airjump (n=airjump@p508ACB1E.dip.t-dialin.net) |
| 18:00.57 | coppice | maybe because they can |
| 18:01.10 | ManxPower | Every single EC out there, including the ones that work with Asterisk immediately and unconditionally disable themselves if they detect a fax tone. This has been the case since EC and faxes started in the 1970s |
| 18:01.34 | coppice | assuming the disable works |
| 18:01.43 | ManxPower | coppice: I've never seen it fail in Asterisk. |
| 18:02.47 | ManxPower | easy enough to confirm with a "zap show channel X" during a zap call. |
| 18:03.11 | coppice | they rely on seeing the 2100Hz tone. many systems conspire to stop that happening |
| 18:03.48 | ManxPower | coppice: If the 2100Hz tone doesn't get thru, I sort of doubt any other part of the fax will get thru. |
| 18:04.21 | coppice | why? lots of funky setups mute the start of calls |
| 18:04.57 | ManxPower | coppice: How often have you personally (as you are the fax god) seen the EC in Asterisk mess up faxes? |
| 18:05.30 | coppice | i haven't looked recently, but it used to happen in a lot of the audio i was sent |
| 18:05.36 | lowtek | Why, why won't faxing die ... |
| 18:05.42 | ManxPower | lowtek: because it is lowtech |
| 18:05.51 | lowtek | lol |
| 18:06.20 | lowtek | I've seen beautiful asterisk deployments get butchered beyond belief and bastardized because someone MUST send a couple faxes a month |
| 18:06.37 | coppice | why should faxing die? its perfectly serviceable for its purpose, and far easier to use that desktop computer alternatives |
| 18:06.44 | ManxPower | lowtek: because it is almost always faster to fax the document than to scan it in on a scanner, then e-mail it. |
| 18:06.44 | *** join/#asterisk mgdm_ (n=michael@river.mgdm.net) |
| 18:07.11 | ManxPower | and of course if the other party needs to sign it they will print it out, fax it back to you. |
| 18:07.41 | coppice | if the asterisk deployments were genuinely beautiful, faxes would go through. its only ugliness that hurts them |
| 18:07.44 | ManxPower | (I shall leave discussions of the idiocy of relying on faxed signatures for another time) |
| 18:08.15 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.14) |
| 18:08.16 | *** join/#asterisk bradleyprice86 (n=bprice@fw.datafax.net) |
| 18:08.20 | ManxPower | coppice: we always have one fax machine directly connected to a telco pots line. |
| 18:08.42 | coppice | that's a sad admission of product shortcomings |
| 18:09.12 | ManxPower | coppice: It is what simply works. No problems, no hacking, no tweaking. It. Simply. World. |
| 18:09.17 | ManxPower | and Works too. |
| 18:09.21 | *** join/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info) |
| 18:09.31 | lowtek | I'd have thought that paper based inter-company commerce would have been instinct by now ... |
| 18:09.39 | coppice | a 100 year old POTS line would work fine, but your super duper new system won't. isn't that rather sad? :-\ |
| 18:09.45 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:09.45 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:09.56 | *** join/#asterisk Entr4nced (i=IMG001@54.sub-75-218-124.myvzw.com) |
| 18:10.02 | ManxPower | And I'm too old to waste my time getting something that will have issues working. I'll stick to keeping the office fax machine on a POTS line. |
| 18:10.07 | lowtek | At any rate, good points all. |
| 18:10.12 | ManxPower | (my users also can get faxes on their DID) |
| 18:10.17 | lesouvage | ManxPower: see http://www.voip-info.org/wiki/view/Asterisk+fax This technique works only when the configured codec is G.711 with no voice activity detection (VAD) and no echo cancellation (EC), |
| 18:10.21 | lowtek | ManxPower: Using? |
| 18:10.23 | brodiem | coppice: Are you implying you can fax reliably without POTS? |
| 18:10.28 | coppice | FAXes have many benefits for a company these days. for example, they are not recorded :-) |
| 18:10.30 | ManxPower | coppice: patches are welcome 8-) |
| 18:10.49 | tzafrir_laptop | brodiem, T.37? |
| 18:10.55 | ManxPower | lowtek: using spandsp, rxfax, ghostscript, and a custom script. |
| 18:11.09 | lowtek | ManxPower: How reliable is it (seriously, no bs) |
| 18:11.18 | jaytee | I once got a help desk call from a VP who was upset that a network printer was down and he was getting ready to leave on a trip and needed to print out about 30 emails he'd recieved but not read. what a waste of paper and a VP salary. |
| 18:11.37 | ManxPower | lowtek: reliable enough for users to usually have their faxes come into their DID. If they have a problem they simply use the pots fax machine. |
| 18:11.48 | coppice | huge numbers of people print every e-mail |
| 18:12.00 | lowtek | ManxPower: If you could throw a percentage at it, just a good guess, what would it be? |
| 18:12.33 | ManxPower | lowtek: I cannot give a percentage. Users are bitchslapped and told to "use the damn office fax" if they complain about their DID fax problems |
| 18:12.39 | lowtek | lol |
| 18:13.04 | UD | so is anyone here using a free service to get incoming SIP or IAX calls? |
| 18:13.17 | UD | i guess ekiga isnt going to work :( |
| 18:13.38 | UD | and looks likke im not the only one |
| 18:13.41 | ManxPower | The major issues that I had to deal with are getting the fax into a PDF form that worked well and the fact that we get 50 page faxes and that turned into a PDF will frequently larger than the max message size on the e-mail server. |
| 18:14.20 | lowtek | I could get rxfax+spandsp to work well over PRI, but not if I switched/forwarded the call to another server. |
| 18:14.25 | lowtek | via SIP |
| 18:14.36 | lowtek | rather, FROM a PRI channel |
| 18:14.45 | ManxPower | lowtek: no fax ever goes over voip on my systems |
| 18:15.04 | *** join/#asterisk hadronzoo (n=user@ppp-70-247-168-51.dsl.rcsntx.swbell.net) |
| 18:15.33 | lesouvage | lowtek: it i not only ho many but what faxes. I have almost been crusified because a normally properly working fax has mess up when high placed people were communication by fax over multimilion EUR contract. Wrong moment for failure. |
| 18:15.35 | *** join/#asterisk cryptnix (n=andrew@firewall.kasltechnology.net) |
| 18:15.44 | ManxPower | In fact, my macros if they see that the channel is not Zap won't even try to detect a fax tone. |
| 18:16.10 | hadronzoo | Can anyone tell me how to forward voicemail message notifications from multiple voicemail boxes to the same phone? |
| 18:16.14 | ManxPower | lesouvage: that is why I use POTS for fax. I never get screamed at. |
| 18:16.31 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 18:16.38 | ManxPower | hadronzoo: where are you trying to forward them to |
| 18:17.27 | lesouvage | ManxPower: I totally agree, use POTS for FAX is a proper oneliner for a T-shirt. But sometimes costumers has other ideas and suddenly came with a fax wish on a place without pots. |
| 18:17.56 | *** join/#asterisk hadronzoo (n=user@ppp-70-247-168-51.dsl.rcsntx.swbell.net) |
| 18:18.03 | ManxPower | lesouvage: then I tell them to get a different consultant. They are told up front I won't do fax over voice over ip |
| 18:18.05 | hadronzoo | sorry, my client dropped |
| 18:18.13 | ManxPower | (1:16:39 PM) ManxPower: hadronzoo: where are you trying to forward them to |
| 18:18.37 | ManxPower | lesouvage: I have no interest in setting things up for a customer I KNOW they will have issues with. |
| 18:19.05 | hadronzoo | ManxPower: I would like to forward notifications to an Aastra 57i |
| 18:19.33 | ManxPower | hadronzoo: in the [sipconfentry] for the Aastra put mailbox=whatevervmbox@whatevervmcontext you want. |
| 18:19.34 | [TK]D-Fender | hadronzoo: Specifiy each box in your sip.conf peer |
| 18:19.43 | ManxPower | I'm outta here. |
| 18:19.47 | lesouvage | ManxPower: I agree again but life isn't always that well defined. |
| 18:20.01 | hadronzoo | I'll try that, thanks |
| 18:20.56 | UD | ekiga sucks |
| 18:20.59 | UD | :( |
| 18:21.22 | coppice | this is a widely held belief |
| 18:21.35 | lowtek | ManxPower: What's the most stable spandsp afayk? |
| 18:21.58 | UD | coppice is there something better i can use? |
| 18:21.59 | VJFROMGT | i am looking for someone to fix a iax / zaptel issue right now ,, if anyone interested please pm me |
| 18:22.07 | UD | free? or at least try before you buy? |
| 18:22.27 | lowtek | UD: What are you trying to accomplish overall? The big picture? |
| 18:22.48 | [TK]D-Fender | UD: Which "Ekiga" are you talking about? |
| 18:23.01 | hadronzoo | [TK]D-Fender: So, I just add multiple mailbox=[ext]@default lines in sip.conf? |
| 18:23.07 | [TK]D-Fender | hadronzoo: Yes |
| 18:23.11 | UD | the one asterisk wont register to |
| 18:23.38 | hadronzoo | [TK]D-Fender: wow, that was easy. And I can have the same mailbox from two different extensions? |
| 18:23.44 | Maliuta | VJFROMGT: not really interested, it 0430 on a saturday morning, I just got home and am so pissed it took me 5 mins to type this reply :P |
| 18:23.46 | UD | is there more than 1? |
| 18:23.58 | Maliuta | 2 is more than 2 :) |
| 18:24.01 | [TK]D-Fender | hadronzoo: Any device can be informed on as many boxes as you want |
| 18:24.03 | Maliuta | well 1 at least |
| 18:24.07 | VJFROMGT | lol |
| 18:24.14 | [TK]D-Fender | UD: Ekiga the SOFTPHONE, or "ekiga.net" SIP service |
| 18:24.32 | hadronzoo | [TK]D-Fender: wonderful. Thanks so much! |
| 18:24.35 | VJFROMGT | well i urgently need someone , so if anyone there, please pm me |
| 18:24.37 | [TK]D-Fender | UD: Sort of like Ford. |
| 18:24.45 | Maliuta | ekiga is shite,zoiper at least uses IAX |
| 18:24.49 | [TK]D-Fender | UD: I hate Ford cars, but love Ford models :) |
| 18:25.14 | Maliuta | sounds like VJFROMGT needs a date ;P |
| 18:25.25 | VJFROMGT | lol |
| 18:25.30 | UD | the provider |
| 18:25.39 | Maliuta | covers his date |
| 18:25.59 | VJFROMGT | 2 servers linked via iax, bandwith was amlost symetrical, suddenly it is now 1:10 (down/up) and i am no expret |
| 18:26.00 | [TK]D-Fender | UD: They are jsut a P2P SIP proxy anyways, what did you need them for? |
| 18:26.13 | Maliuta | watches one transformers ep before 'leepies |
| 18:26.19 | UD | well |
| 18:26.24 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 18:26.31 | UD | i want to have asterisk answer an incoming call |
| 18:27.23 | [TK]D-Fender | UD: And from where we left off, you simply didn't configure it at all, and I even wasted time Googleing the instruction on how to do so and HAND FED them to you and still you seem to have nothing to show for it. |
| 18:27.37 | ManxPower | hadronzoo: those are not extensions, they are devices. |
| 18:27.41 | UD | let me show you then |
| 18:27.57 | UD | i used that example |
| 18:28.13 | UD | see the thing is i cannot find any good documentation, solid in one location |
| 18:28.26 | UD | like that example, i think you mean http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net |
| 18:28.33 | UD | leaves out the rest of the file |
| 18:28.48 | hadronzoo | ManxPower: Right, I mean SIP devices, not extensions. |
| 18:29.19 | UD | i can keep screwing around with it but i get the -- Got SIP response 606 "Not Acceptable" back from 86.64.162.35 |
| 18:29.26 | ManxPower | UD: THERE IS NO REST OF THE FILE REQUIRED |
| 18:29.27 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 18:29.47 | ManxPower | UD: not acceptable means "no codecs are available on both sides" |
| 18:30.03 | ManxPower | Like if you only allowed ulaw and the other side only accepts GSM you would get the message. |
| 18:30.57 | ManxPower | I would put a disallow=all and an allow=ulaw |
| 18:32.24 | *** join/#asterisk dawalama (n=dawalama@wsip-68-105-250-226.sd.sd.cox.net) |
| 18:32.45 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:33.06 | dawalama | Good morning all good people. Do you guys know if accountcode (in CDR ) gets truncated if you are using RA to store them in Database |
| 18:33.16 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 18:33.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:33.30 | ManxPower | UD: you seem to think you have ONE problem. You don't. You have many problems and if you can't dedicate the time to solve each of them one at a time, you'll have a miserable life using Asterisk. |
| 18:33.35 | ManxPower | It doesn |
| 18:33.47 | ManxPower | It doesn't help that you seem to not even bother to respond to stuff. |
| 18:33.52 | lmadsen | RA? Remote Access BBS system FTW |
| 18:35.03 | ManxPower | You're on your own UD. |
| 18:35.43 | ManxPower | I have better things to do than hope some random person I'm helping will respond. |
| 18:41.11 | [TK]D-Fender | UD: And you're not showing us complete debug. Singular messages like that are worthless |
| 18:42.34 | *** join/#asterisk ant (n=aburton@209.11.127.13) |
| 18:44.49 | UD | i dont know what to paste |
| 18:45.10 | UD | i just commented out the whole sip.conf and did sip reload |
| 18:45.15 | UD | got identical debug info |
| 18:46.06 | [TK]D-Fender | UD: If you commented out your whole sip.conf you have NO CONFIG |
| 18:46.29 | [TK]D-Fender | UD: and the reason you are not showing US the full debug after I've asked you for it repeatedly is...? |
| 18:47.19 | UD | [TK]D-Fender, yes i agree |
| 18:47.25 | UD | that was the reason i commented it out |
| 18:47.43 | UD | where is my debugging info |
| 18:48.11 | UD | i have debug info that says that same unacceptable line and the exact same stuff, that i can pastebin again if you want, but hasnt changed |
| 18:48.14 | [TK]D-Fender | UD: fix your configs, enable SIP debug and show us a failed call. |
| 18:48.28 | ManxPower | (1:29:20 PM) UD: i can keep screwing around with it but i get the -- Got SIP response 606 "Not Acceptable" back from 86.64.162.35 |
| 18:50.12 | [TK]D-Fender | UD: If you've commented out everything in sip.conf you are dead in the water. Follow the instructions I've already handed you and show us the failed call with sip debug |
| 18:50.30 | UD | i did that to debug the debugger |
| 18:51.03 | [TK]D-Fender | UD: that is a single useless error message, not a SIP conversation. |
| 18:52.28 | *** join/#asterisk angryuser (n=Miranda@36.249.146.195.dynamic.adsl.abo.nordnet.fr) |
| 18:53.10 | *** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137) |
| 18:53.45 | *** join/#asterisk molkmin (i=d17c33d4@gateway/web/ajax/mibbit.com/x-ae8c48187872c816) |
| 18:54.00 | UD | http://pastebin.com/m4e4f30a1 |
| 18:54.05 | rgsteele||work | Quick question, hopefully. If you make changes to either sip.conf or extensions.conf, and you execute an asterisk -rx "reload", if asterisk doesn't like the config, will it just not reload (like Apache), or will it puke? |
| 18:55.07 | TJNII | It will reload and complain about the line it doesn't like. |
| 18:55.11 | M1s3ry | rgsteele||work, it will reload, the new config may not work though if it's not set up right |
| 18:55.13 | *** part/#asterisk molkmin (i=d17c33d4@gateway/web/ajax/mibbit.com/x-ae8c48187872c816) |
| 18:55.16 | [TK]D-Fender | UD: that is only the ANSWER to your register attempt, not the START of it |
| 18:55.33 | rgsteele||work | So, under no circumstances will it refuse to reload if the config is munged in some way? |
| 18:55.44 | M1s3ry | I've yet to see it happen |
| 18:55.57 | M1s3ry | so... nope |
| 18:56.00 | [TK]D-Fender | rgsteele||work: Bad option will just get discarded and the rest will load. How functional that leaves you will vary accordingly |
| 18:56.41 | TJNII | [TK]D-Fender++ |
| 18:57.24 | angryuser | good day |
| 18:58.24 | UD | http://pastebin.com/m1c8542cd |
| 18:59.31 | [TK]D-Fender | UD: Contact: <sip:101@192.168.0.124> <-- you have clearly not followed the instructions we pointed you to repeatedly many hours ago about setting your * system up properly to operate behind NAT. |
| 18:59.32 | [TK]D-Fender | ~sipnat |
| 18:59.33 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:59.34 | [TK]D-Fender | ^^^^^^ |
| 19:00.15 | ManxPower | [TK]D-Fender: after like 6 hours he still has not even gotten it working to the point of that doc. |
| 19:00.19 | rgsteele||work | [TK]D-Fender: Thanks for the tip, appreciate it. |
| 19:00.25 | ManxPower | he thinks he has one problem. |
| 19:00.26 | [TK]D-Fender | UD: And hearing you say you've commented everything out lends me to believe you are continuosly taking 1 step forward, and 5 steps back with every try |
| 19:00.33 | angryuser | i have a person with 2 analog lines, when they are connected with analog phones directly all is fine, but when they get connected to TDM2400p fxo port, his phone company says 'line unavailable' and nothing under cli, his dialplan is sane, any ideas ? |
| 19:00.51 | UD | i am taking 0 steps forward |
| 19:01.11 | UD | i think you must have missed my last message saying the debugger is spewing false info |
| 19:01.15 | UD | register => 872074:mypasswordhere@ekiga.net/101 |
| 19:01.19 | UD | thats the line i used |
| 19:01.20 | [TK]D-Fender | UD: Before you say you weren't even setting up a register. now you seem to have. |
| 19:01.33 | [TK]D-Fender | UD: Go fix all the NAT settings you undid |
| 19:01.33 | ManxPower | angryuser: add "ww" to the beginning of the number to Dial in the dialplan |
| 19:01.43 | UD | i didnt undo anything |
| 19:01.44 | tzafrir_laptop | angryuser, how do those lines show in /proc/zaptel/* |
| 19:01.46 | tzafrir_laptop | ? |
| 19:01.49 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 19:02.18 | UD | i even tried my actual user name rather than that # |
| 19:02.19 | ManxPower | angryuser: each "w" (on FXO PORTS ONLY) dill pause for .5 second, usually 1 second is enough time to let the telco start to detect DTMF |
| 19:02.23 | angryuser | tzafrir_laptop : yes , zapata/zaptel is configured&working he got some fxs also on the same card, no problem |
| 19:02.26 | UD | and im not loading an empty file |
| 19:02.33 | ManxPower | UD: nothing is going to work until your NAT stuff is set. |
| 19:03.08 | UD | <[TK]D-Fender> UD: Contact: <sip:101@192.168.0.124> <-- ya i noticed that too but its not what i typed |
| 19:03.37 | tzafrir_laptop | angryuser, something, somewhere is not OK. |
| 19:03.41 | [TK]D-Fender | UD: Ekiga is complaining because you didn not set your system up properly to work behind NAT. Follow the guide you'd now been linked to a dozen times |
| 19:03.47 | angryuser | ManxPower : give me example of that w plz :) |
| 19:04.15 | UD | you just got done telling me that i only need those 2 lines in that file |
| 19:04.19 | ManxPower | exten => 5551212,1,Dial(Zap/g1/ww${EXTEN}) |
| 19:04.21 | UD | ManxPower ^ |
| 19:04.41 | ManxPower | UD: I'm not really helping you anymore as you are not dedicating the time required. |
| 19:05.03 | [TK]D-Fender | UD: Go follow the guide now. Anything else is a waste of time. |
| 19:05.03 | UD | fine |
| 19:05.17 | UD | i really have been sort of ignoring you since you contradict everything you say and seem to generally not know what you're talking about |
| 19:05.29 | ManxPower | UD: Yes, I don't really know much about Asterisk. |
| 19:05.32 | [TK]D-Fender | UD: I have not contradicted myself |
| 19:05.41 | UD | i was talking to ManxPower |
| 19:05.49 | UD | he jumps back and forth |
| 19:05.54 | [TK]D-Fender | UD: then stop talking to him and do what I've told you. |
| 19:05.58 | *** join/#asterisk jcoffi (n=Justin_C@75.147.154.145) |
| 19:06.05 | angryuser | tzafrir_laptop: , yes he had same missconfigurations at start, but nothing serious, 'Zap show status' is ok cat /proc/zaptel/* is ok ztcfg -vvv is ok he got some irq misses, but nothing serious |
| 19:06.22 | [TK]D-Fender | UD: You are the one jumping and whining. I've wasted hours handing you information and you waste this time complaining with him. |
| 19:07.26 | UD | http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net i followed that guide exactly and it does not work |
| 19:07.34 | scooby2 | anyone use ZapScan? I add channels to a group called SPY and they show up in "group show channels" as part of that group but then when I try and do ZapScan(SPY) it is just dead air. Using ZapScan() lets me change through all zap calls with the # but I only want certain calls to be spied on. |
| 19:08.03 | [TK]D-Fender | UD: and I linked you to the other guide to handle the NAT portion *10 TIMES* |
| 19:08.04 | hardwire | I'm guessing Intrepid Ibex and it's zaptel source aren't friendly atm? |
| 19:08.13 | [TK]D-Fender | ~sipnat |
| 19:08.14 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:08.18 | hardwire | the zaptel package is still using some depreciated stuff? |
| 19:08.20 | [TK]D-Fender | ^^^ this guide is IMPORTANT |
| 19:09.19 | UD | and you dont think i tried that? |
| 19:09.42 | [TK]D-Fender | UD : Does a cookbook tell you how to open the door to your oven so you can bake a cake? No. Because they only deal with the parts that they should have to worry about. Should a guide to Ekiga.net have to care that you may or may NOT be behind NAT? No, they only provide the pertinent settings that are 100% specific to them. the rest is YOUR job. |
| 19:09.54 | ManxPower | [TK]D-Fender: I don't think he understands that he needs to set up his config as a merge of BOTH documents. |
| 19:09.58 | *** join/#asterisk legis (n=estar@unaffiliated/legis) |
| 19:10.25 | [TK]D-Fender | UD>and you dont think i tried that? <--- You say you commented everything out. that says you keep trying to do half the job and then UNDO IT |
| 19:10.26 | dawalama | what is a good practice for setting CDR UserField .. I tried AppendCDRUserField, SetCDRUserField, and SET(CDR(userfield)=value) |
| 19:10.31 | dawalama | None of them worked |
| 19:10.46 | [TK]D-Fender | UD: And you sure as heck aren't showing us your configs so we can see what you've done either. |
| 19:11.10 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 19:11.11 | dawalama | I get a 'No such application' error in asterisk . I got these commands from googling |
| 19:11.13 | *** join/#asterisk TheMahmoud (n=foo@unaffiliated/mahmoud) |
| 19:11.15 | jaytee | ohhh, I'm pissed now! I just discovered my car's owner's manual doesn't tell me how to tune up my Briggs and Stratton lawnmower. I'm gonna sue! |
| 19:11.16 | *** join/#asterisk Jason_m0000 (n=hansonsu@openbrick.kicks-ass.net) |
| 19:11.30 | [TK]D-Fender | dawalama: Then you are calling dialplan applications that do not exist. |
| 19:11.50 | legis | "Unable to find a codec translation path from ulaw to g729" <- does that mean I need g729 codecs? |
| 19:12.07 | [TK]D-Fender | dawalama: and "googling" does not imply that what you are reading applies to anything, let alone your current scenario |
| 19:12.15 | [TK]D-Fender | legis: Yes |
| 19:12.19 | UD | i am starting from scratch. my config files are going to be deleted |
| 19:12.22 | Jason_m0000 | hello nufone support...... |
| 19:12.26 | angryuser | ManxPower: dawalama Set(${Callerid(name)}) ?? |
| 19:12.37 | angryuser | oh not to manxpower |
| 19:12.39 | ManxPower | dawalama: "show applications" show what applications are available for YOUR version of Asterisk |
| 19:12.40 | [TK]D-Fender | UD: Then go follow both guides and integrate accordingly. |
| 19:12.43 | legis | [TK]D-Fender: when a peer doesn't support a codec, the call should at least ring right? |
| 19:12.47 | UD | yes |
| 19:12.48 | UD | i will |
| 19:12.50 | [TK]D-Fender | legis: No |
| 19:12.59 | legis | k |
| 19:13.16 | angryuser | mhahaha ^^^^ |
| 19:13.17 | [TK]D-Fender | legis: Why should it ring when it knows it can't talk anyways? |
| 19:13.26 | ManxPower | Jason_m0000: this is not nufone support. Take the 2nd door, down the hall, to the left. |
| 19:13.43 | bradleyprice86 | I am in the process of setting up a 2nd asterisk system to replace the first, I am going to test the new system, and I wanted to somehow only setup a few clients on the new system to test it for a few days. How would I route traffic from the 2nd system back to the first? |
| 19:13.51 | dawalama | Great . Thanks ManxPower .. I am very new to Asterisk , and trying to get this working has been challenging |
| 19:14.00 | legis | [TK]D-Fender: because codec negotiation is done when the call is answered? |
| 19:14.03 | [TK]D-Fender | angryuser>ManxPower: dawalama Set(${Callerid(name)}) ?? <--- wrong on so many levels. |
| 19:14.33 | dawalama | ? |
| 19:14.39 | angryuser | [TK]D-Fender : his question in unclear also |
| 19:15.10 | [TK]D-Fender | angryuser: Didn't seem to have a real question. Also jsut showed an error without bothering to show which application call CAUSED it. |
| 19:15.18 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:16.33 | dawalama | Hey, all the commands I used above exists in the application list. Looking at SetCDRUserField , it advises me to use Set(CDR(userfield)=...) .. which didn't work |
| 19:17.05 | ManxPower | dawalama: I guess you did not read UPGRADE.txt and UPGRADE-1.2.txt. |
| 19:17.06 | [TK]D-Fender | dawalama: Show us. |
| 19:17.15 | [TK]D-Fender | dawalama: PASTEBIN <- |
| 19:17.40 | [TK]D-Fender | dawalama: because the line you just showed us now should certainly not throw the error message you showed us ealier |
| 19:18.55 | dawalama | I am using asterisk-java |
| 19:18.57 | dawalama | http://pastebin.com/d154f435b |
| 19:19.25 | dawalama | I am trying to store a really long identication string in userfield .. since CDR's accountcode is only 20 char long |
| 19:19.56 | Jason_m0000 | has he been here? |
| 19:20.05 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 19:20.12 | dawalama | This is what I get in the log |
| 19:20.12 | dawalama | [Sep 5 07:58:44] WARNING[1345]: pbx.c:5155 ast_pbx_run_app: No such application 'Set(CDR(userfield)=0000011176b47d03000a002f0034002a67e612ed)' |
| 19:20.12 | [TK]D-Fender | dawalama: CLI output only please. |
| 19:20.33 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 19:20.49 | [TK]D-Fender | dawalama: Its taking the full line as if it were the application, which it is not. |
| 19:21.20 | [TK]D-Fender | dawalama: and Asterisk-java is not supported here |
| 19:21.54 | [TK]D-Fender | dawalama: originateAction.setApplication("Set(CDR(userfield)=0000011176b47d03000a002f0034002a67e612ed)"); <--- and this is not an "application. This is a complete DIALPLAN App call. |
| 19:22.09 | dawalama | I see |
| 19:22.37 | eric_hill | What settings affect the "Hangup ACK requested on unconfigured channel number" message? |
| 19:22.54 | dawalama | Thanks [TK]D , I will do more research now |
| 19:23.13 | [TK]D-Fender | dawalama: Parameters belong somewhere else |
| 19:23.27 | eric_hill | It's followed by a pair of numbers (28/14 span 1). The 14 is the correct channel number, but I don't have a 28 listed anywhere in the config.... |
| 19:25.03 | UD | [TK]D-Fender |
| 19:25.07 | UD | you are going to hate me |
| 19:25.13 | dawalama | On a little more asterisk oriented question, is there a way to allow asterisk to increase the size for accountcode |
| 19:25.21 | UD | i forgot i had another router in line |
| 19:25.34 | [TK]D-Fender | UD: Improper use of tense. |
| 19:25.41 | UD | i just looked at my router's gateway |
| 19:25.44 | UD | lol |
| 19:26.04 | [TK]D-Fender | UD: and it doesn't matter because you couldn't even do the part you are responsible for properly so extra issues don't even facttor in yet |
| 19:27.15 | [TK]D-Fender | UD: And double-NAT on one side can be accounted for, though can pose problems if you lack conrol over both or your routers suck |
| 19:30.21 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 19:32.56 | VJFROMGT | I have a IAX link between 2 servers and all of a sudden the bandwidht is no longer symetrical, i am looking for someone who cna fix it , please pm me if you can,, urgent |
| 19:33.02 | scooby2 | [TK]D-Fender: anyway to check the * config before doing a reload? |
| 19:33.20 | [TK]D-Fender | scooby2: God gave you eyes. Thats all you've got. |
| 19:34.06 | Jason_m0000 | has jeremy been around? |
| 19:34.26 | TJNII | Jason_m0000: ask the bot. (~seen) |
| 19:36.44 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:38.05 | bkw_ | "god gave you eyes"? damn |
| 19:38.28 | [TK]D-Fender | damns bkw_ |
| 19:38.44 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:38.50 | bkw_ | someone woke up on the bitch side of the bed today |
| 19:38.52 | bkw_ | :P |
| 19:39.09 | bkw_ | anyway that was funny |
| 19:39.10 | [hC] | hey! bkw! |
| 19:39.18 | [hC] | how are ya? |
| 19:39.22 | TJNII | bkw_: Give him some slack. After chasing UD in circles all afternoon I'd be grumpy too. |
| 19:39.32 | legis | if my softphone uses ulaw to my asterisk, and asterisk uses g729 to a ITSP, I need a g729 codec for that? |
| 19:39.37 | bkw_ | TJNII: I can totally understand |
| 19:40.01 | bkw_ | legis: yes |
| 19:40.16 | [TK]D-Fender | TJNII: He asked, I delivered. Some people just don't know how to be happy... |
| 19:40.38 | legis | bkw_: k, that is what is called transcoding? |
| 19:40.44 | [TK]D-Fender | legis: yes |
| 19:40.54 | TJNII | [TK]D-Fender: Sometime I don't know how you have the patience to be here all the time. |
| 19:41.07 | legis | ok |
| 19:41.30 | [TK]D-Fender | TJNII: I really don't sometimes. I'm not sure you can comprehend how much I've held back today. |
| 19:42.09 | bkw_ | [TK]D-Fender: if you're that frustrated its vacation time.... or margarita time. |
| 19:42.15 | bkw_ | likes margarita |
| 19:42.28 | [TK]D-Fender | bkw_: vacation it is.... I don't drink |
| 19:42.29 | TJNII | [TK]D-Fender: Probably not. I was gone most the afternoon and when I came back I was amased he had made no noticeable progress. |
| 19:42.42 | bkw_ | [TK]D-Fender: I don't see why not... its great! |
| 19:43.06 | [TK]D-Fender | bkw_: Offers me nothing, and costs a lot. |
| 19:43.21 | bkw_ | its cheap.. do it at home. |
| 19:43.26 | bkw_ | hehe |
| 19:44.22 | [TK]D-Fender | bkw_: Too much work :) |
| 19:44.41 | [hC] | bkw, are you gonna be at astricon again this year? I want to chat about freeswitch with you while im there |
| 19:44.44 | [TK]D-Fender | bkw_: Food is either 2 steps tops at home, or eaten out :) |
| 19:45.19 | *** join/#asterisk gerhard7 (n=gerhard@82-170-16-80.ip.telfort.nl) |
| 19:45.27 | bkw_ | [hC]: yes i'll be there. |
| 19:45.40 | [hC] | bkw_: they better have golf carts again. |
| 19:45.44 | bkw_ | haha |
| 19:45.46 | scooby2 | Anyone have issues with 'agent logoff Agent/#' not working all the time? |
| 19:46.18 | scooby2 | even from the cli sometimes it takes doing it two or three times |
| 19:48.10 | scooby2 | or I need to figure out how to use AgentCallbackLogin to logout |
| 19:48.27 | *** join/#asterisk i3inary (n=shaman@cpe-76-88-94-149.san.res.rr.com) |
| 19:50.29 | scooby2 | no one uses the old way of logging agents on and off? |
| 19:51.53 | *** join/#asterisk ESCulapio__ (n=elvyn@mail.abaca.com.do) |
| 19:53.32 | [TK]D-Fender | scooby2: Go call AgentCallbackLoginand use it to log out |
| 19:55.42 | hardwire | nom snom snom |
| 19:56.20 | *** join/#asterisk jetole (n=Joe@irc.fossl.net) |
| 19:57.45 | jetole | hey fellas, I am trying to put together an estimate for asterisk in our office and I was hoping someone can recommend a quality sip provider? |
| 20:02.20 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 20:04.31 | *** join/#asterisk TheMahmoud (n=foo@unaffiliated/mahmoud) |
| 20:07.00 | hadronzoo | how can I override a config option in a particular section within an included config file? I tried the [section](+) format, but that doesn't appear to be working. |
| 20:09.24 | [TK]D-Fender | hadronzoo: You don't |
| 20:10.30 | *** join/#asterisk r0bis_ (n=roberts@host86-168-186-96.range86-168.btcentralplus.com) |
| 20:10.42 | hadronzoo | [TK]D-Fender: oh, ok. thanks |
| 20:13.19 | r0bis_ | Hello all! I've got a problem - when I call my asterisk box from outside line asterisk responds alright and does what is required. However when asterisk hangs up the phone I am calling from (both mobile and landline) do not know the call has ended. Could it be zaptel not loaded with option for UK? Many thanks |
| 20:14.12 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:18.38 | *** part/#asterisk jcoffi (n=Justin_C@75.147.154.145) |
| 20:19.02 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 20:19.45 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 20:25.51 | [TK]D-Fender | ok, day is done, I'm out |
| 20:26.59 | r0bis_ | Hello all! I've got a problem - when I call my asterisk box from outside line asterisk responds alright and does what is required. However when asterisk hangs up the phone I am calling from (both mobile and landline) do not know the call has ended. Could it be zaptel not loaded with option for UK? |
| 20:28.30 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:34.04 | *** join/#asterisk FlyboySR22 (n=rsears@searsair.fw.americanis.net) |
| 20:34.57 | *** join/#asterisk thx_man (n=thx_man@port-83-236-207-89.static.qsc.de) |
| 20:45.18 | thx_man | hi, does anyone know what that menuselect error depends on? beta9 have no menuselect error, only rc1 |
| 20:45.26 | thx_man | menuselect_newt.c: In function `run_menu': |
| 20:45.26 | thx_man | menuselect_newt.c:321: error: `NEWT_KEY_ESCAPE' undeclared (first use in this function) |
| 20:45.26 | thx_man | menuselect_newt.c:321: error: (Each undeclared identifier is reported only once |
| 20:45.26 | thx_man | menuselect_newt.c:321: error: for each function it appears in.) |
| 20:45.26 | thx_man | make[1]: *** [menuselect_newt.o] Error 1 |
| 20:46.02 | tzafrir_laptop | thx_man, I'd say there's an earlier warning about some missing include file |
| 20:48.34 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:49.32 | *** join/#asterisk thx_man (n=thx_man@port-83-236-207-89.static.qsc.de) |
| 20:49.35 | tzafrir_laptop | ~pb |
| 20:49.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:56.23 | eric_hill | ~seen |
| 20:59.11 | *** part/#asterisk patrick-- (n=patrick@212.112.255.47) |
| 21:02.18 | tzafrir_laptop | ~seen eric_hill |
| 21:02.20 | jbot | eric_hill is currently on #asterisk. Has said a total of 3 messages. Is idling for 5m 57s, last said: '~seen'. |
| 21:02.42 | eric_hill | Yes I am :) |
| 21:03.42 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 21:03.45 | khronos | Hi guys. |
| 21:04.15 | tzafrir_laptop | thx_man, do you actually have chan_zap explicitly disabled by you? or was there a problem with the build dependencies? |
| 21:04.42 | khronos | How do I get Asterisk to generate a busy signal to a caller if I have a customer who has two lines with me, but they are sharing a pool of lines? |
| 21:04.53 | khronos | when the third call comes in? |
| 21:05.54 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 21:07.20 | *** join/#asterisk dFence (n=chatzill@p5496B0C4.dip0.t-ipconnect.de) |
| 21:07.40 | dFence | hey guys! is there a channel where all the siemens-hicom/hipath gurus hide at? |
| 21:08.12 | mvanbaak | ~ask |
| 21:08.13 | jbot | rumour has it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:08.33 | thx_man | <PROTECTED> |
| 21:09.04 | eric_hill | dFence: Haven't all the siemens-hicom/hipath guys moved to Cisco CM? :) |
| 21:09.39 | mvanbaak | eric_hill: no, they all moved to asterisk + chan_skinny |
| 21:09.40 | dFence | eric_hill: then i need a veteran ;D i just can't get this fat ugly bitch to work the way i want it to... |
| 21:10.07 | mvanbaak | dFence: replace it with asterisk |
| 21:10.25 | dFence | mvanbaak: too expensive right now... trust me - was my first thought ^^ |
| 21:10.33 | mvanbaak | eh ? |
| 21:10.42 | mvanbaak | it's like 500 $ |
| 21:10.46 | eric_hill | dFence: I'd vote for a catastrophic failure in the core (fire would work nicely) followed by a nice insurance telephone call. |
| 21:11.00 | mvanbaak | lol eric_hill |
| 21:11.15 | dFence | eric_hill: and yep, came to me as well :) |
| 21:11.18 | eric_hill | Last time I worked on a HiPath system, I nearly slit my wrists. |
| 21:11.21 | tzafrir_laptop | thx_man, the problem is probably because you don't have some pre-requirement of chan_zap . Do you have zaptel installed? |
| 21:11.43 | mvanbaak | s/zaptel/DAHDI/ |
| 21:11.50 | dFence | mvanbaak: i ran the calculations once... would be somewhere around 1800bugs including rewiring the whole house |
| 21:12.00 | thx_man | jep |
| 21:12.24 | mvanbaak | dFence: but you get a system that's scalable, flexible, robust |
| 21:12.32 | mvanbaak | how much bucks is that ? |
| 21:13.03 | dFence | mvanbaak: you don't have to lecture me on that topic... but my boss ;D and any investition > 500 is way beyond questioning... |
| 21:13.05 | *** join/#asterisk neurosys (n=neurosys@c-66-229-91-186.hsd1.fl.comcast.net) |
| 21:13.13 | eric_hill | mvanbaak: I've had * enter a CPU race condition twice now that required a physical power-cycle of the box. That's not exactly roses in my book... |
| 21:13.46 | mvanbaak | eric_hill: did you file a bug report for it on bugs.digium.com ? |
| 21:14.36 | dFence | while i'm kinda holding this beast together and it sorta-kinda does what i want i'm afraid ill have to keep working with it... but i already told em that i won't be no part in any attempt to fix it after it kicks the bucket -> then it's * time |
| 21:15.49 | mvanbaak | dFence: dd if=/dev/zero of=/dev/hipath |
| 21:16.22 | eric_hill | dFence: What's your problem with the HiPath? Phone? Trunk? Service? Queue? |
| 21:16.35 | dFence | eric_hill: i can't get the s0-bus to work properly... |
| 21:16.47 | eric_hill | dFence: Do you get an error? |
| 21:17.03 | dFence | eric_hill: does fritzCard-software crashing count? |
| 21:17.18 | dFence | i don't know how to talk to capi directly so i don't get a proper error-msg |
| 21:17.27 | dFence | it just crashes... |
| 21:17.37 | eric_hill | dFence: What flavor of firmware are you running? Something in the 5.x range at least, right? |
| 21:17.49 | dFence | eric_hill: on the hicom? |
| 21:17.54 | eric_hill | dFence: Si. |
| 21:18.24 | dFence | eric_hill: at least > 2.0.2... where does it reveal it's sw-version? |
| 21:18.46 | dFence | it's an almost ancient hicom 118E |
| 21:18.46 | eric_hill | dFence: Or maybe it was 1.5.x... My mind is like water - cold, dripping, and periodically turns to steam. |
| 21:18.49 | mvanbaak | CLI> core show status |
| 21:18.50 | mvanbaak | ;) |
| 21:19.00 | *** join/#asterisk moy (n=moy@nat/ibm/x-302d9413159197f8) |
| 21:19.11 | mvanbaak | hey moy |
| 21:19.26 | mvanbaak | bye lesouvage |
| 21:19.31 | eric_hill | dFence: Ah, I worked on the big big brother of that system. Never anything that small... |
| 21:19.45 | moy | hey mvanbaak how you doing? |
| 21:20.13 | *** join/#asterisk Paige (n=Paige@2001:470:b:aa:21f:c6ff:fe48:8ce3) |
| 21:20.18 | mvanbaak | moy: not too bad. bought a house today ! |
| 21:20.29 | moy | yay! |
| 21:20.42 | Qwell | moy: http://blogs.digium.com/2008/09/05/another-reason-to-believe-in-open-source-development/ |
| 21:21.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:21.33 | moy | hey, that's me |
| 21:21.37 | dFence | eric_hill: it has the size of a friggin microwave - you call that _small_ |
| 21:21.38 | moy | heh |
| 21:22.10 | *** part/#asterisk Paige (n=Paige@2001:470:b:aa:21f:c6ff:fe48:8ce3) |
| 21:22.35 | eric_hill | dFence: Yes, I worked on a system that came in a box the size of a refrigerator. 200+ person call center with skills based routing, call recording, etc... |
| 21:22.57 | eric_hill | dFence: It was pretty cool, but a royal bitch to get working right. |
| 21:23.22 | dFence | eric_hill: it's german and it's made by siemens - what. the. hell. did. you. expect?! |
| 21:23.23 | eric_hill | dFence: The installers were at the site for weeks straight, and back dozens of times to iron out other issues. |
| 21:23.44 | mvanbaak | moy: good job ! |
| 21:24.09 | moy | thanks :) .... but in honor to the truth is not quite accurate though :) ... I never met Mark at an event |
| 21:25.32 | mvanbaak | btw: http://www.openbsd.org/items.html#44 |
| 21:25.51 | mvanbaak | </advertisement> |
| 21:26.05 | Qwell | how do I download the CD? |
| 21:26.15 | Qwell | oh, that's right, you can't. :p |
| 21:26.42 | mvanbaak | Qwell: you can |
| 21:26.47 | Qwell | mvanbaak: is there a song to go along with this release? |
| 21:27.48 | mvanbaak | Qwell: wget ftp://ftp.openbsd.org/pub/OpenBSD/snapshots/i386/install44.iso |
| 21:28.01 | Qwell | ahh |
| 21:28.31 | mvanbaak | Qwell: and the song is here: http://www.openbsd.org/lyrics.html#44 |
| 21:28.55 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 21:28.57 | jasonwoot | woot |
| 21:28.59 | Qwell | groans |
| 21:29.04 | mvanbaak | whehehehehe |
| 21:29.19 | jasonwoot | is dragon naturally speaking worth a crap? |
| 21:29.35 | mvanbaak | dragon is a fork of Open |
| 21:30.14 | mvanbaak | that is, if you are talking bout dragonflybsd |
| 21:30.33 | Corydon76-dig | Dragon Naturally Speaking is fine, but it is not suited for telephony projects |
| 21:31.01 | Corydon76-dig | If you want a telephony-based voice system, use Lumenvox |
| 21:31.34 | Corydon76-dig | The parameters for each are completely different. Lumenvox is not well suited for arbitrary word recognition |
| 21:32.01 | Corydon76-dig | but the price/performance works out well for limited word recognitiion |
| 21:32.51 | *** join/#asterisk Sweeper (n=sweeper@74.51.109.60) |
| 21:32.54 | Corydon76-dig | The issue with Dragon for telephony is that Dragon, while good with arbitrary word recognition, requires a period of training that is not possible (too long!) on voice calls |
| 21:33.17 | Sweeper | http://www.magicjack.com/9/index.asp <-- anyone tried one of these? it looks like something that could almost be decent but is probably shit |
| 21:33.44 | jasonwoot | quality control has been asking me for transcripts for a year, and my knee jerk answer is "no" |
| 21:33.51 | jasonwoot | it's a tough concept to wrap my brain around |
| 21:34.12 | Corydon76-dig | transcripts of what? |
| 21:34.20 | dFence | hrm... great - mailserver just kicked it, nothing's working at all... i'm gonna hit the sheets... NIGHT |
| 21:34.37 | jasonwoot | we record upwards of 5000 calls per day... |
| 21:35.01 | Corydon76-dig | jasonwoot: take a page from the medical transcription industry. Contract off shore for transcriptionists |
| 21:35.35 | mvanbaak | Qwell: you cant say that song is bad |
| 21:35.35 | tzafrir_laptop | Corydon76-dig, sea dragons? |
| 21:35.39 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:35.45 | jasonwoot | lol... naturally, they're wanting same-day information... |
| 21:35.46 | mvanbaak | it's actually the best song they did |
| 21:35.50 | Bananaskin | Sweeper n3glv has one of the magic jacks, $20 usd for a years calls |
| 21:36.00 | Corydon76-dig | jasonwoot: how much are they willing to pay? |
| 21:36.13 | jameswf-home | is adding ManxPower's quote to my collection: "Echo Cancelers do not screw up faxes, people screw up faxes. 8-)" |
| 21:36.13 | Qwell | Corydon76-dig: they want it free, naturally |
| 21:36.23 | Corydon76-dig | jasonwoot: you can hire US workers to do transcription, but it gets VERY expensive |
| 21:36.43 | mvanbaak | who needs faxes anyway ? |
| 21:36.52 | Corydon76-dig | Minimum pay in the US is a LOT higher than minimum pay, in, say, the Phillipines |
| 21:37.07 | mvanbaak | BIG NEWS: there is this new thing called email ! |
| 21:37.07 | jameswf-home | mvanbaak: not everyone has heard of email it is still pretty new |
| 21:37.34 | Corydon76-dig | jasonwoot: you may even want to do hybrid solution... figure out which recordings are high priority, get those done locally, and everything that can wait 48 hours, send overseas |
| 21:37.55 | jameswf-home | mvanbaak: email has yet to prove it self ready for primetime and is only really used by nerds geeks and their respective grandmothers |
| 21:37.56 | jasonwoot | transcription just makes me go eeewwwwww |
| 21:38.01 | mvanbaak | we trashed our fax 2 years ago. and we did not need it after that |
| 21:38.13 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) |
| 21:38.15 | Corydon76-dig | jasonwoot: humans are currently the only way to do it reliably |
| 21:38.28 | jameswf-home | KILL ALL HUMANS |
| 21:39.04 | mvanbaak | I really dont understand why ppl want fax capabilities |
| 21:39.23 | jasonwoot | I've used voice recognition via training for one speaker, but I just don't know how to capture a 2-way convo |
| 21:40.05 | Corydon76-dig | You don't. |
| 21:40.19 | Corydon76-dig | Not with the current software, anyway |
| 21:40.34 | jasonwoot | Ok, mind if I quote you Corydon76? |
| 21:40.40 | jasonwoot | what's your phone number, btw |
| 21:40.41 | Corydon76-dig | Go ahead |
| 21:41.02 | Corydon76-dig | Uh, you don't call me directly |
| 21:41.03 | mvanbaak | 1-800-Corydon |
| 21:41.15 | Corydon76-dig | mvanbaak: I almost got that one, actually |
| 21:41.22 | mvanbaak | for real ? |
| 21:41.23 | mvanbaak | cool |
| 21:41.26 | Corydon76-dig | for real |
| 21:41.35 | mvanbaak | that would be awesome |
| 21:41.36 | Corydon76-dig | Black hole got it, though |
| 21:41.44 | mvanbaak | blegh |
| 21:42.07 | legis | is there a sip CLI softphone do tests? |
| 21:42.25 | mvanbaak | legis: use asterisk with chan_console |
| 21:42.30 | Corydon76-dig | The one of which I'm most proud is for a gay youth group |
| 21:42.41 | Corydon76-dig | 877-ONE-IN-TN |
| 21:42.49 | jasonwoot | ok, that's my cue... have a great weekend |
| 21:42.49 | mvanbaak | gheh |
| 21:43.02 | legis | mvanbaak: can you provide an example on how to dial from asterisk CLI? can' find how |
| 21:43.05 | mvanbaak | Corydon76-dig: you be at astricon ? |
| 21:43.12 | Corydon76-dig | mvanbaak: of course |
| 21:43.28 | Corydon76-dig | I'm presenting, remember? |
| 21:43.42 | mvanbaak | legis: console dial <exten>@<context> |
| 21:43.50 | mvanbaak | Corydon76-dig: ah, course |
| 21:43.59 | mvanbaak | Corydon76-dig: we'll finally meet |
| 21:44.17 | Corydon76-dig | woohoo |
| 21:44.35 | mvanbaak | it's still unreal to me |
| 21:44.43 | mvanbaak | this week is weird |
| 21:45.13 | mvanbaak | I'm thinking about what to bring to astricon, I bought a house today (first time of my life) |
| 21:45.15 | Corydon76-dig | plays the theme from Twilight Zone |
| 21:45.47 | mvanbaak | and now I hear I'm going to meet the legendary Corydon76-dig |
| 21:45.49 | mvanbaak | damn |
| 21:46.20 | Corydon76-dig | mvanbaak: I'll tell you what I told Malcolm: STOP THAT |
| 21:46.32 | mvanbaak | lol |
| 21:46.39 | Corydon76-dig | Makes me feel weird to be called legendary |
| 21:46.48 | mvanbaak | but but but |
| 21:46.51 | mvanbaak | you are ! |
| 21:47.07 | Corydon76-dig | goes to throw up |
| 21:47.35 | mvanbaak | looks at func_odbc, cdr_adaptive_odbc and bows before the master |
| 21:47.51 | Corydon76-dig | Ugh |
| 21:48.04 | mvanbaak | oh wait ! |
| 21:48.40 | mvanbaak | looks at res_curl and bows again |
| 21:49.16 | mvanbaak | too bad we aren't sharing a room ;) |
| 21:49.19 | Corydon76-dig | Go worship file or something |
| 21:51.05 | mvanbaak | nah, he's in #geeks |
| 21:51.23 | mvanbaak | eeeew |
| 21:51.30 | mvanbaak | and #drupal |
| 21:51.43 | Corydon76-dig | blinks |
| 21:52.19 | mvanbaak | we are coding Covide. that's a CMS as well |
| 21:52.24 | Corydon76-dig | btw, I have something for you to test |
| 21:52.37 | mvanbaak | sure thing. what is it ? |
| 21:52.38 | Corydon76-dig | team/tilghman/realtime_failover/ |
| 21:53.06 | Corydon76-dig | For specifying multiple lines in extconfig for each realtime family, with priorities |
| 21:53.11 | mvanbaak | that is: if A fails, try B |
| 21:53.12 | mvanbaak | ? |
| 21:53.15 | Corydon76-dig | Right |
| 21:53.27 | mvanbaak | awesome ! |
| 21:53.32 | mvanbaak | I'll test this weekend |
| 21:53.34 | Corydon76-dig | and enhancements to res_odbc, such that if A fails, stop retrying contantly for N seconds |
| 21:53.35 | scooby2 | taking from Colbert - Asterisk: The Future of Telephony (Second Edition) - Great book or Greatest book? |
| 21:53.39 | mvanbaak | that is freaking great |
| 21:53.58 | mvanbaak | scooby2: greatest book |
| 21:54.15 | mvanbaak | Corydon76-dig: I'll give it a shot this weekend |
| 21:54.19 | Corydon76-dig | mvanbaak: Nah, next edition will be greater |
| 21:54.43 | scooby2 | does it cover 1.4 much in the second edition? |
| 21:54.48 | mvanbaak | Corydon76-dig: I just setup a testbed with 3 master<->master database servers and 8 asterisk servers |
| 21:54.56 | Corydon76-dig | k |
| 21:55.18 | mvanbaak | so easy to do |
| 21:55.27 | mvanbaak | just take $random_db offline |
| 21:55.33 | mvanbaak | and see how it works |
| 21:55.50 | mvanbaak | xm destroy db1.vanbaak.info |
| 21:55.55 | mvanbaak | \o/ |
| 21:56.36 | mvanbaak | I got myself a SUN X2200M2 |
| 21:56.47 | mvanbaak | for all the vm's |
| 21:56.59 | scooby2 | xen is pretty nice |
| 21:57.10 | mvanbaak | scooby2: I love it |
| 21:57.44 | scooby2 | I broke down 13 servers into 3 xen boxes. Quite cost effective. |
| 21:57.52 | mvanbaak | yup |
| 21:58.11 | r0bis_ | can ayyone help: after asterisk hangup line still is open until the phone from which I call asterisk is put on hook (i.e. disconnected). Asterisk CLI though says |
| 21:58.13 | mvanbaak | but we still use physical boxen for our firewalls |
| 21:58.13 | scooby2 | also let me fit our datacenter into two racks instead of three:) |
| 21:58.13 | r0bis_ | <PROTECTED> |
| 21:58.20 | tzafrir_laptop | mvanbaak, what do you run dom0 on? |
| 21:58.31 | mvanbaak | tzafrir_laptop: debian stable |
| 21:58.46 | Corydon76-dig | mvanbaak: so for ODBC, only the first connection will wait a bit, then it will failover completely until the timeout happens... then try again on the first one... |
| 21:58.48 | scooby2 | mvanbaak: definitely. We use 2 1u's redundant with heartbeat. I would love to figure out how to make asterisk as redundant. |
| 21:59.20 | mvanbaak | Corydon76-dig: cool. just what I needed |
| 21:59.24 | Corydon76-dig | mvanbaak: so basically, once the failed database comes back up, Asterisk will, after the timeout, switch back |
| 21:59.40 | mvanbaak | Corydon76-dig: you my man. |
| 21:59.59 | mvanbaak | Corydon76-dig: so I can put all my db servers in there |
| 22:00.07 | mvanbaak | Corydon76-dig: and asterisk will take care of it |
| 22:00.11 | Corydon76-dig | Right |
| 22:00.18 | *** join/#asterisk neurosys (n=neurosys@c-66-229-91-186.hsd1.fl.comcast.net) |
| 22:00.19 | mvanbaak | Corydon76-dig: because they are all mastel-master |
| 22:00.27 | mvanbaak | 4 node setup |
| 22:01.15 | r0bis_ | can anyone pls help: after asterisk hangup line still is open until the phone from which I call asterisk is put on hook (i.e. disconnected). Asterisk CLI though says: -- Hungup 'Zap/1-1' |
| 22:01.24 | mvanbaak | scooby2: we have 3 OpenBSD firewalls with relayd on it to loadbalance the shit. combine that with realtime and dundi ... |
| 22:05.20 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 22:08.16 | ManxPower | r0bis_: Asterisk will never hangup and FXS until the phone is physically put back on hook. This does not apply to FXO. |
| 22:08.46 | mvanbaak | but now I'm off to bed |
| 22:10.27 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 22:11.24 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:18.40 | legis | can I make a incoming call be directed to asterisk itself? |
| 22:20.17 | ManxPower | legis: Um, where else would it go? Mars? |
| 22:20.20 | r0bis_ | ManxPower: Thank you very much for reply. Maybe I am wrong, but just thinking - line goes into FXO interface. Asterisk receives call through it. Caller goes through menus with dtmfs and upon a certain choice Asterisk comes to Hangup() in dialplan. But thi hangup does not happen - if the other party does not put down the receiver, asterisk keeps the line open (busy). Shouldn't I get Asterisk to send a busy signal after |
| 22:20.20 | r0bis_ | hangup() or something? |
| 22:20.51 | legis | ManxPower: I mean, i would like to answer the call with the asterisk cli |
| 22:21.02 | ManxPower | r0bis_: once asterisk hangs up the line it is up to the telco to disconnect the call. Asterisk cannot force the telco to do anything |
| 22:21.12 | ManxPower | legis: google for "console phone asterisk" |
| 22:21.54 | ManxPower | you must have oss/alsa loaded, the console phone configured, and the sound drivers loaded. If the sound dev files were not installed when you installed Asterisk then you will have to install them and rebuild Asterisk. |
| 22:22.57 | legis | ManxPower: I have those, I can make calls with consolo dial, want to test incoming now |
| 22:23.09 | r0bis_ | ManxPower: right; however when I put down my ordinary phone receiver, the telco (BT) disconnects the line. I would then need to be able to emulate this with Asterisk? |
| 22:23.11 | ManxPower | legis: Console/1 I think is the channel |
| 22:23.32 | ManxPower | r0bis_: Asterisk does this by DEFAULY. |
| 22:23.52 | ManxPower | Maybe you have your signalling screwed up or something. |
| 22:24.19 | ManxPower | legis: but that is NOT directing the call to "asterisk itself", it is directing the call to the console phone. |
| 22:26.18 | r0bis_ | ManxPower: Sorry, maybe I'm silly. If I call home from mobile and reach my analog phone - as soon as I hangup my analog phone I hear busy tone on my mobile and line is released. If I call home and reach asterisk and asterisk does hangup() mobile remains in connected state and line is busy if I try to reach asterisk from another mobile |
| 22:26.26 | legis | ManxPower: Ok thanks! |
| 22:26.43 | ManxPower | rob0: there you are confused. The line is not released, the line is open, active and making a busy sound. |
| 22:27.15 | jaytee | is listening to Seether - Broken [6:06 (17%)] |
| 22:27.30 | r0bis_ | ManxPower: this :)) starts to sound like light at the end of the tunnel |
| 22:27.40 | jaytee | oops, wrong window |
| 22:27.43 | jaytee | sorry about that |
| 22:29.19 | r0bis_ | ManxPower: so if an analog phone is put back on hook where does the busy signal in the other phone that called it come from - it then must be provided by telco |
| 22:30.44 | r0bis_ | ManxPower: if so the analog phone line must be released to be able to receive new calls. However this is not the case. If I call from another mobile, the line that was hangup() by asterisk is still busy |
| 22:32.05 | jaytee | I've run into that issue before with cheap X100P cards but I was using them for analog lines from another PBX. |
| 22:32.06 | rob0 | is confused, indeed :) |
| 22:33.05 | jaytee | I had to drop using the Answer() and Hangup() apps for that extension's context. |
| 22:33.16 | r0bis_ | yup :( - where could I read up about how this happens? |
| 22:36.45 | jaytee | r0bis_, dunno, maybe google for Asterisk disconnect supervision, FXO as keywords |
| 22:38.06 | r0bis_ | jaytee, ok, thank you |
| 22:38.13 | ManxPower | r0bis_: Asterisk should, automatically, and by default, with no need for you to make changes, hang up the line EXACTLY like an analog phone does. I do not know why this is not happening in your situation. |
| 22:38.58 | ManxPower | It is common for Asterisk to not DETECT when the FAR END HANGS UP, but if that is your situation -- well you are far, far beyond my help. |
| 22:39.03 | *** part/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 22:40.54 | r0bis_ | ManxPower: maybe loading wctdm with debug=1 option will be of some help. I am sorry, but I have no idea how I ended up in this situation - and the more I google the more I discover that problem is exactly as you say - usually the opposite |
| 22:42.00 | ManxPower | r0bis_: maybe the driver does not know hat country you are in |
| 22:42.14 | ManxPower | hat = what |
| 22:42.26 | *** join/#asterisk vonkleist (n=gcontrer@189.155.176.253) |
| 22:42.59 | ManxPower | r0bis_: I would suspect wiring being reversed or something like that where an anlog phone would not care. |
| 22:43.56 | r0bis_ | ManxPower: well, as for the driver : dmesg: [134489.792112] Module 1: Installed -- AUTO FXO (UK mode) |
| 22:44.45 | *** join/#asterisk mgdm (n=michael@river.mgdm.net) |
| 22:44.52 | r0bis_ | ManxPower: also zaptel.conf says loadzone and defaultzone = uk |
| 22:44.56 | *** join/#asterisk vipcarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 22:44.58 | vipcarrier | hi |
| 22:45.18 | vipcarrier | does any one has a php script to provision cisco 79XX |
| 22:50.27 | legis | ls\ |
| 22:52.06 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-e9f375096de868fa) |
| 22:52.08 | jaytee | no such file or directory :-) |
| 22:54.00 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 22:54.29 | *** join/#asterisk astassistant (n=skip@216.160.91.74) |
| 22:55.22 | *** join/#asterisk robba (n=robert@203.214.68.53) |
| 22:55.32 | robba | hey guys |
| 22:55.56 | robba | does nyone here know about bristuff like installation and so forth? |
| 23:03.21 | *** join/#asterisk sevard (n=sev@multimedia.dvc.edu) |
| 23:03.28 | sevard | Does anyone have any ANI readback #s? |
| 23:03.35 | sevard | I'm at a loss. |
| 23:08.21 | *** join/#asterisk Psychobilly (n=moi@adsl79-56.kln.forthnet.gr) |
| 23:08.23 | *** join/#asterisk Steven_B (n=Steven_B@dslb-088-077-213-110.pools.arcor-ip.net) |
| 23:08.42 | *** part/#asterisk astassistant (n=skip@216.160.91.74) |
| 23:09.18 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:12.48 | tzafrir_laptop | robba, some do |
| 23:12.58 | tzafrir_laptop | do you have a question? |
| 23:13.53 | robba | i have to install asterisk with a b410P card |
| 23:14.09 | robba | and i have never done an intall with a BRI card |
| 23:14.50 | robba | i have tried mISDN an that didn't seem to work |
| 23:15.04 | robba | well the card came online for a few seconds then died |
| 23:15.38 | ManxPower | robba: Digium provides support for their cards. |
| 23:16.30 | robba | yeah i didn't find it real helpful i tried the make b410p and it seemed to do everything but didn't seem to work |
| 23:17.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:20.22 | *** join/#asterisk kj4acm (n=nr4q@24-183-225-98.dhcp.kgpt.tn.charter.com) |
| 23:20.54 | scooby2 | Any way to listen in without stopping mixmonitor? |
| 23:21.22 | kj4acm | what username/password belongs in /etc/asterisk/phpagi.conf ? |
| 23:23.24 | [TK]D-Fender | kj4acm: only thing that could would be for AMI |
| 23:24.58 | kj4acm | i've hosed this up somehow |
| 23:25.34 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:26.30 | [TK]D-Fender | kj4acm: And managed to tell us nothing about your actual problem in the process. Impressive... |
| 23:28.35 | kj4acm | fender: oh. well i was scratching my head for a second. changing the default password of "amp111" to another password in /etc/manager.conf gives me a bunch of php errors when i try and view my asterisk settings (trixbox) |
| 23:29.11 | [TK]D-Fender | ~wglwat |
| 23:29.12 | jbot | well, wglwat is well, good luck with all that |
| 23:29.43 | kj4acm | haha, yeah. i've not done anything with it really. easiest thing for me to do would be just start from scratch again except there's no monitor on the machine |
| 23:30.42 | ManxPower | kj4acm: perhaps you could ask on the correct channel? |
| 23:30.51 | ManxPower | take a look at the /topic |
| 23:31.32 | kj4acm | that'd probably be the best approach wouldn't it |
| 23:31.59 | [TK]D-Fender | Best approach is silently from behind with a sharp knife. |
| 23:32.33 | kj4acm | i cut myself about a month ago with a spyderco knife... still healing |
| 23:33.58 | jaytee | haha, http://img172.imageshack.us/img172/5267/blocksjb0.png |
| 23:34.30 | jaytee | I'd say this guy is a bit on the irritated side |
| 23:34.40 | *** join/#asterisk gambolputty (n=BC43599@cpe-24-175-90-49.tx.res.rr.com) |
| 23:38.52 | *** join/#asterisk angryuser (n=Miranda@36.249.146.195.dynamic.adsl.abo.nordnet.fr) |
| 23:39.32 | angryuser | is it sane to write like this ? some user on forum has deadlock problems exten => 500,n,Dial(SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506&SIP/600,60,rtT) |
| 23:40.32 | *** join/#asterisk UD (i=OHNOUDID@unaffiliated/underdawg) |
| 23:41.56 | *** join/#asterisk _zoomy_ (n=_zoomy_@dahlin.csbnet.se) |
| 23:42.48 | ManxPower | it's never sane to use "r" and both "t" and "T" at the same time. |
| 23:43.20 | [TK]D-Fender | ManxPower: Unrelated.... "r" is evil all by itself. |
| 23:43.41 | tzafrir_laptop | BTW: http://openbts.sourceforge.net/ |
| 23:44.06 | tzafrir_laptop | ("bts" does not stand for Bug Tracking System there) |
| 23:46.31 | *** join/#asterisk vonkleist (n=vonkleis@189.155.176.253) |
| 23:46.46 | _zoomy_ | Hello. Im trying to connect to a postgresql database and do a write via Set() in the dialplan. But I get this error message: [Sep 6 01:42:28] ERROR[8809]: func_odbc.c:128 acf_odbc_write: No database handle available with the name of 'asterisk-connector' (check res_odbc.conf). The weird thing is that I have the cdr setup with the exact same dsn. Are they (cdr and the dialplan) using the same config file /etc/odbc.ini? |
| 23:51.18 | ManxPower | [TK]D-Fender: there should have been a comma in my statement. Yes, "r" is evil all the time. |
| 23:51.30 | *** join/#asterisk adr3nalin3_ (n=afink@asa.redglaze.com) |
| 23:52.04 | _zoomy_ | The cdr is working well. This is what *CLI> odbc show gives me: Name: asterisk, DSN: asterisk-connector, Pooled: no, Connected: yes. |
| 23:52.06 | *** join/#asterisk LemensTS (n=matthew@adsl-70-130-218-70.dsl.stlsmo.swbell.net) |
| 23:52.38 | [TK]D-Fender | ManxPower: and "tT" is also perfectly valid... |
| 23:52.49 | LemensTS | I got an asterisk behind a nat, and 2 sip phones on a different network behind a nat. When i do 'sip show peers' it shows both sip phones having the wan IP of there nat, but also the same port...they should have different ports right? |
| 23:53.00 | ManxPower | [TK]D-Fender: yes, but it is very seldom you want both ends of the call to be able to transfer |
| 23:53.16 | [TK]D-Fender | ManxPower: not inherently wrong, just equally sad :) |
| 23:54.04 | ManxPower | LemensTS: you have a braindead NAT router. See if you can change the SOURCE PORT to something other than 5060 on one of the phones, that might make your router do the right NAT thing |
| 23:54.45 | *** join/#asterisk kdas (n=ME@c-67-161-44-222.hsd1.ca.comcast.net) |
| 23:55.51 | kdas | can someone tell me why my asterisk server can't regonize any keys from my softphone or analog phone ? |
| 23:56.29 | ManxPower | kdas: for the softphone a DTMF mode mismatch would do it. no idea about analog unless you set your rxgain way too low. |
| 23:57.20 | kdas | ManxPower, it has to do when i call my voicemail. I use voicemailmain(blah|s) |
| 23:57.56 | kdas | ManxPower, it says "Please press 1 to listen to your new messages" but when i press one nothing happens |
| 23:59.38 | _zoomy_ | Does anyone know if there are multiple files which config connections to database? |
