| 00:03.29 | *** join/#asterisk coppice (~chatzilla@m180-219-227-89.smartone-vodafone.com) |
| 00:09.33 | CRCinAU_ | if anyone here uses sccp, any hints on how to do a dialplan for it? |
| 00:09.44 | CRCinAU_ | it seems that after converting my cisco phones from SIP to SCCP, the dialplan.xml is no longer referenced. |
| 00:09.50 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
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| 00:13.11 | pabelanger | CRCinAU_: what did you do to get banned? |
| 00:14.27 | CRCinAU_ | pabelanger: I can't remember... it was back in 2009 aparently. |
| 00:14.34 | *** join/#asterisk jmordica (jmordica@c-68-63-214-171.hsd1.ms.comcast.net) |
| 00:14.47 | jmordica | what is the easiest cleanest way to do redundancy between two asterisk boxes on different networks? heartbeat with csync2? |
| 00:14.50 | CRCinAU_ | I think I disagreed with maintaining so many branches of asterisk at once or something |
| 00:16.30 | tzanger | on *different* networks? |
| 00:16.43 | tzanger | generally speaking heartbeat is for two boxes in close proximity to each other |
| 00:17.01 | tzanger | on different networks I'm wondering if you wouldn't want to use DUNDI and some dialplan logic/quick DNS updates |
| 00:17.38 | jmordica | yea different networks.. hmm.. like dns srv? |
| 00:18.17 | tzanger | what kind of redundancy are you after? random outside clients getting in? |
| 00:19.00 | jmordica | just if one server goes down the second server comes up. I think i can use dns srv in the cisco phones. there is an option for it. |
| 00:19.18 | jmordica | So i guess i'd have to copy my databases over every night so I would have the same exact box on both ends |
| 00:20.10 | jmordica | use rsync or something? |
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| 00:37.55 | Kobaz | http://www.thatvideosite.com/video/the_skin_gun |
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| 00:49.43 | jmordica | will mydns work for dns srv? |
| 00:59.10 | paulc | Kobaz: wow, that's pretty cool eh? |
| 01:01.03 | Kobaz | yeah |
| 01:13.10 | tzanger | jmordica: if you're setting up and concerned about a redundant SIP service I highly doubt you will be using mydns for anything related to said service |
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| 01:15.09 | quidpro | Hmm... anybody running 1.4 on CentOS 5.5? From TOP it looks like it's only using on core of a Dual-Xeon QuadCore. |
| 01:15.50 | WIMPy | Doing what? |
| 01:16.09 | quidpro | Just passing calls, some transcoding,e tc. |
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| 02:18.58 | CRCinAU_ | pooey |
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| 02:45.30 | jmordica | this may be a fairly dumb question. Do i have to have a 3rd server doing dns srv or can i install mydns on my primary and secondary server? |
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| 03:11.28 | jodrellblank | Hi; Say I have a deskphone which can only have one SIP server connection, but I want it to have two. Say it's connected to a local asterisk server as SIP/100 - can I get the asterisk server to connect to thirdparty.example.org and act as a SIP client, and sort of ... passthrough incoming calls to the phone? |
| 03:12.01 | thenk_ | yeah |
| 03:12.08 | thenk_ | that's what asterisk does |
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| 03:13.22 | jodrellblank | it is? |
| 03:13.36 | WIMPy | yes |
| 03:13.44 | WIMPy | ~b2bua |
| 03:13.44 | infobot | well, b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 03:15.55 | jodrellblank | I mean, I don't want to trunk to the other server. As far as the other server is concerned there should only be a SIP phone connected. Is that just a case of a config in sip.conf which can be handled by normal extension handling? |
| 03:16.18 | WIMPy | ~siptrunk |
| 03:16.18 | infobot | somebody said siptrunk was something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 03:16.57 | thenk_ | jodrellblank: an asterisk 'sip trunk' is exactly the same as the sip phone connecting to asterisk |
| 03:16.59 | thenk_ | it's just another peer |
| 03:17.30 | WIMPy | It's just as much of an SIP server as any phone. |
| 03:17.40 | *** join/#asterisk aiguojun (~agj@117.39.246.108) |
| 03:18.25 | jodrellblank | this sounds like it will work almost as easily as I want it to work |
| 03:18.33 | jodrellblank | that never happens with technology. What am I missing? |
| 03:19.01 | aiguojun | HELP!asterisk1.6.2.17,I used xlite 3.0 softphone , dtmfmod was seted to rfc2833 and auto,i must dial password very fast, otherwise, asterisk only receive one or two digit. |
| 03:19.05 | jodrellblank | I guess I'll get some login details tomorrow and try it |
| 03:20.05 | jodrellblank | so what's the point in IAX trunks? |
| 03:20.51 | thenk_ | to connect with IAX clients |
| 03:21.38 | thenk_ | jodrellblank: what you will need to do is setup one peer and one friend in sip.conf, then a dial plan to handle incoming calls |
| 03:22.18 | thenk_ | if you've never touched asterisk before, i'd recommend something like asteriskNOW or trixbox, as they come with configuration GUI and easy dialplan generators. |
| 03:24.00 | jodrellblank | I have touched it before, but I don't know asterisk/VOIP very well |
| 03:24.36 | jodrellblank | which is a shame, as I'm going to get lumbered with a "our VOIP calls have intermittent noise on them" problem tomorrow |
| 03:25.02 | aiguojun | connect with sip client |
| 03:29.58 | *** part/#asterisk aiguojun (~agj@117.39.246.108) |
| 03:32.16 | CRCinAU_ | hmmm |
| 03:32.25 | CRCinAU_ | anyone know if I can optimise this a bit better? http://fpaste.org/vM2W/ |
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| 03:38.05 | thenk_ | CRCinAU_: exten => 1[83]00XXXXXX |
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| 03:45.00 | KNERD | They need to change the name to AsteriskLATER |
| 03:45.22 | KNERD | It only installed CentOS...shit I did not need to download another copy of CentOS |
| 03:47.20 | CRCinAU_ | thenk_: good spotting ;) |
| 03:48.08 | Corydon76-home | thenk_: don't forget the underscore |
| 03:49.24 | thenk_ | oh yeah, that too |
| 03:49.36 | thenk_ | my bad :) |
| 03:50.46 | thenk_ | so i'm trying to make asterisk answer a phone call from h263 client, and play a h263 video file to them |
| 03:50.51 | thenk_ | playback() does not support video codecs? |
| 03:51.21 | thenk_ | ..Just says no file exists - but if I change it to some audio codec extension, it spits that out |
| 03:52.05 | WIMPy | I think, forwarding is the only thing, Asterisk can do with videos. |
| 03:52.35 | Corydon76-home | thenk_: it can, but you need both an audio AND a video codec |
| 03:52.39 | WIMPy | But that might change with the new media/codec handling stuff. |
| 03:53.01 | Corydon76-home | and no, you cannot play an h263 file to them. Asterisk doesn't know how to read that filetype |
| 03:53.21 | thenk_ | ah. |
| 03:53.25 | thenk_ | that's what I thought. |
| 03:53.26 | WIMPy | Do you need two files then? |
| 03:53.37 | thenk_ | What video codec can Asterisk play>? |
| 03:53.54 | Corydon76-home | Only thing Asterisk can do is store the literal frames and read those frames back off the disk, which is different from how the standard h263 format exists |
| 03:53.58 | thenk_ | the core says it knows all about video formats |
| 03:54.12 | thenk_ | ah, so there's no h 263 packetizer? |
| 03:54.20 | Corydon76-home | It knows its own stuff, not standard formats |
| 03:54.52 | thenk_ | so I setup an RTSP server, and trying to use app_rtsp to playback H263 |
| 03:54.59 | thenk_ | but that does not work either.. |
| 03:55.04 | Corydon76-home | I agree that it's non-optimal, just the same as that Asterisk cannot (yet) do video transcoding |
| 03:55.07 | thenk_ | app_rtsp seems to not work properly |
| 03:55.43 | Corydon76-home | Although in the second case, the problem is really more of a problem with the massive CPU it takes to transcode video |
| 03:56.15 | thenk_ | libavcodec is pretty efficient |
| 03:56.31 | thenk_ | plus, most of computers now have plenty of spare cores doing not much |
| 03:56.55 | Corydon76-home | Well, and then there's the problem of most standard video codecs having patents still in effect |
| 03:56.55 | thenk_ | transcode of a 10min video takes only a couple of seconds |
| 03:58.00 | Corydon76-home | With a for-profit entity backing up Asterisk, patent holders don't just look the other way, the way they might when small nonprofit open source projects use those patents |
| 03:58.17 | Corydon76-home | As they say in law school, never sue anybody who has no money |
| 04:00.53 | Corydon76-home | At some point, somebody will come out with a hardware transcoder for video for Asterisk, and since they pay the licensing fees with the hardware, it's much more possible to use that to deal with the patent issue |
| 04:02.33 | thenk_ | mm. |
| 04:02.48 | thenk_ | I just want to answer a SIP call and play a video |
| 04:07.55 | thenk_ | i've been able to make it happen using other software which gives video streams to Asterisk to use |
| 04:08.07 | thenk_ | but that's all licensed software, which costs a bundle |
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| 05:16.28 | rocksfrow | anybody here use polycom phones? |
| 05:16.38 | rocksfrow | soundpoints |
| 05:20.20 | rocksfrow | really? that's shocking... |
| 05:20.44 | kaldemar | many do. |
| 05:21.18 | rocksfrow | yeah that's why i was shocked nobody said yes: P |
| 05:21.21 | rocksfrow | i guess it's the hour... |
| 05:21.33 | rocksfrow | i was curious if anybody uses phoneprovisioning.com |
| 05:21.44 | rocksfrow | i'm currently using a local FTP server... but i love the idea of having something setup remote |
| 05:21.48 | rocksfrow | you ever hear of that site? |
| 05:22.06 | rocksfrow | anybody from here make it? :P ...it's free so i'd imagine its an open source geek ;) |
| 05:22.22 | kaldemar | the hour or the question. poll-like questions rarely get any answers. |
| 05:22.43 | rocksfrow | yeah i guess i could have been more straight-forward |
| 05:23.03 | rocksfrow | does anybody who use polycom phones have any thoughts/experience using phoneprovisioning.com's free config hosting? |
| 05:23.09 | rocksfrow | still too poll-like? haha |
| 05:23.18 | kaldemar | never used it. people tend to use their own servers for provisioning. |
| 05:23.59 | rocksfrow | yeah, i figure so... i've been doing so for a year or so (since i got the phones)... but i like the idea of having it hosted / easily editable |
| 05:24.21 | rocksfrow | i have some scripts somebody from here shard with me that i use... but its still sort of annoying |
| 05:24.33 | rocksfrow | i might play with it |
| 05:25.11 | rocksfrow | kaldemar, what's your opinion, do you see any security risks hosting this on something like that? i'm not sure anybody could really do any damage with my polycom config files? |
| 05:26.05 | rocksfrow | http://www.phoneprovisioning.com/setup-polycom.php |
| 05:26.10 | rocksfrow | it looks hella simple to use too |
| 05:27.04 | kaldemar | if you get the files without any encryption, it's a risk. |
| 05:27.36 | rocksfrow | i've got all pbx traffic blocked at the firewall, though |
| 05:27.49 | rocksfrow | it's a pretty small asterisk setup only like 20 phones |
| 05:27.52 | rocksfrow | and all local, one location |
| 05:28.04 | rocksfrow | with the exception of a couple specific IP opens |
| 05:28.45 | rocksfrow | kaldemar, could you elaborate on what the risk would be specifically? i guess if it weren't firewalled somebody would get the SIP info and register remotely? |
| 05:28.56 | rocksfrow | kaldemar, i'd love you to scare me out of it :P |
| 05:31.33 | kaldemar | someone could eavesdrop your data and use your box to make calls when they find a hole to your pbx. |
| 05:32.45 | rocksfrow | kaldemar, yea... |
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| 05:35.25 | rocksfrow | i GUESS that's good enough reason :P |
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| 06:10.05 | vk2dgy | good morning/evening. Can anyone offer me a word of advice with .call files? |
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| 06:31.29 | vk2dgy | good morning/evening. Can anyone offer me a word of advice with .call files? Specifically, I have call files working fine but I want to enforce a maximum duration of the call. Any suggestions? |
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| 06:46.57 | kaldemar | vk2dgy: use a local channel as the channel in your call file and set a max duration in the Dial application. |
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| 06:53.48 | vk2dgy | kaldemar: is that the only way to do it?? I have unknown numbers both called and calling which (I thought) makes it a little hard? |
| 06:57.24 | kaldemar | vk2dgy: only way that i can think of right now. what exactly makes it hard using a local extension? |
| 06:57.53 | vk2dgy | the fact that neither called nor calling number are guaranteed to be a local extension! |
| 06:58.43 | vk2dgy | I have an application where sometimes I want to (at a specific time) call person (1) and connect them to person (2). One may be a local extension, but often times won't be. |
| 06:59.33 | vk2dgy | for example, a staff member who's at home and needs to talk to someone overseas at a determined time (but don't want them to have to call on their dime, which could be very expensive) |
| 07:00.28 | vk2dgy | so a call file can call them (home or office), when they answer, call the other party. But I want to limit the call to (say) 30 mins. |
| 07:01.12 | kaldemar | vk2dgy: i think you misunderstood local channel. you can call an extension in your dialplan, it doesn't have to mean that the destination is one of your local phones. |
| 07:01.23 | thenk_ | maybe i'm crazy, but i'm pretty sure you can't make or receive phone without a local extension. |
| 07:01.54 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 07:02.23 | vk2dgy | you can, it works. |
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| 07:02.57 | kaldemar | vk2dgy: local channel is e.g. Local/12345@yourcontext, where you can put whatever you like as the extension and context. then you have an extension that does what you want. |
| 07:03.55 | vk2dgy | thanks, I'll experriment. |
| 07:06.48 | kaldemar | and don't think the extension as a plain number, you can put a channel identifier or whatever you wish in it, and just parse the EXTEN variable in the extension. |
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| 07:16.58 | kleszcz | mornin' |
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| 07:29.24 | dimm | please say some words about this line in logfile /var/log/asterisk/full |
| 07:29.49 | dimm | [Mar 10 10:29:27] DEBUG[7488] audiohook.c: Read factory 0xb5418610 and write factory 0xb5419030 both fail to provide 160 samples |
| 07:29.51 | dimm | [Mar 10 10:29:27] DEBUG[7471] audiohook.c: Read factory 0xb76e70a0 was pretty quick last time, waiting for them. |
| 07:29.59 | dimm | just some words :) |
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| 08:01.28 | daxt | hi guys , I have a softphone connected to asterisk server , which is connected to DID sip service . I want the softphone to dial a PSTN number, How do i set the dial plan for that ? |
| 08:03.26 | *** join/#asterisk myp (~andrew@mx2.admin.tomsk.ru) |
| 08:04.25 | kaldemar | daxt: make an extension that matches numbers that you want to send to PSTN and either put the extension in the phone's context or put it in another context and include it in the phone's context. i'd do the latter. |
| 08:05.19 | daxt | kaldemar what if number changes , do i have to add another extension to new number ? |
| 08:07.07 | kaldemar | that's why you use dialplan patterns that match more than one number: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 08:07.44 | kaldemar | be sure to notice "The period character (.) within a pattern..." in that article. |
| 08:09.13 | daxt | thanks kaldemar |
| 08:16.05 | daxt | kaldemar , i am unable to understand that article , can you help me to make the dial plan. the numbers i want to dial are 0123456789 and 9876543210 from the softphone. |
| 08:16.05 | daxt | so the probable dial plan would be exten => _XXXXXXXXXX,2,Dial(Sip/ ????? ) |
| 08:16.05 | daxt | how do i fill the question marks ? |
| 08:16.34 | kaldemar | that pattern would match all numbers that are 10 digits in length. is that what you really want? |
| 08:17.16 | kaldemar | to fill in the question marks, you need to know what you want to send to your provider. are the forementioned numbers appropriate? |
| 08:19.10 | p3nguin | Don't forget a peer to send them via. SIP/peer/extension If you want to send the numbers that match the pattern, you can use the ${EXTEN} variable. |
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| 08:22.26 | daxt | ahaa now we are getting there |
| 08:22.33 | daxt | thanks p3nguin |
| 08:22.42 | thenk | hehe |
| 08:22.56 | thenk_ | found the other browser window :P |
| 08:23.19 | thenk | i should really practise closing my browser tabs |
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| 08:38.58 | daxt | guys this was my extension "exten => _XXXXXXXXXXXXX,1,Answer() |
| 08:38.58 | daxt | exten => _XXXXXXXXXXXXX,2,(SIP/${EXTEN}@mydivert,30,Tt) |
| 08:38.59 | daxt | exten => _XXXXXXXXXXXXX,3,Hangup()" |
| 08:39.30 | daxt | i guess something wrong with this :( |
| 08:39.34 | p3nguin | Consider SIP/mydivert/${EXTEN} |
| 08:39.42 | p3nguin | And do not Answer() it. |
| 08:40.08 | p3nguin | exten => _XXXXXXXXXXXXX,1,Dial(SIP/mydivert/${EXTEN}) |
| 08:40.17 | p3nguin | exten => _XXXXXXXXXXXXX,n,Hangup() |
| 08:40.23 | daxt | ah |
| 08:40.27 | daxt | thanks p3nguin |
| 08:40.35 | p3nguin | Copy what I write. |
| 08:40.42 | daxt | sure |
| 08:41.30 | ChannelZ | yeah shut up and don't question him! |
| 08:42.35 | p3nguin | If I don't insist that it is copied exactly, all sorts of various shit will get modified. |
| 08:42.53 | daxt | "[Mar 10 03:38:11] WARNING[20495]: pbx.c:3696 pbx_extension_helper: No application '' for extension (default, 0094757009465, 1)" |
| 08:43.02 | p3nguin | And then I have to repeat myself multiple times trying to explain why it doesn't work and how to fix it. |
| 08:43.15 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 08:43.22 | p3nguin | Did you save the changes and then run "dialplan reload"? |
| 08:43.48 | daxt | asterisk -r and then reload |
| 08:43.55 | p3nguin | How about never do that again. |
| 08:44.17 | p3nguin | If you modify extensions.conf, run dialplan reload after you save the changes. |
| 08:44.38 | SiNGLer | I guess he just removed first priority, didn't renumber |
| 08:45.04 | daxt | exten => _XXXXXXXXXXXXX,1,(SIP/mydivert/${EXTEN}) |
| 08:45.04 | daxt | exten => _XXXXXXXXXXXXX,n,Hangup() |
| 08:45.07 | p3nguin | That's why I said to copy what I wrote. |
| 08:45.19 | p3nguin | Next time, copy what I said to copy. |
| 08:45.43 | p3nguin | And after you copy it, paste it into your screen. |
| 08:45.53 | p3nguin | s/screen/file/ |
| 08:46.01 | daxt | ya it works now |
| 08:46.03 | daxt | thanks guys |
| 08:46.25 | p3nguin | See? Even when I insist that it is copied, it STILL doesn't get copied. |
| 08:46.43 | p3nguin | So fuck me. I don't even know why I try to help people. |
| 08:47.09 | daxt | p3nguin , just dont bother about it , u will be rewarded for ur work |
| 08:47.51 | daxt | :) |
| 08:48.59 | dimm | how i can check formats available for dialing record? |
| 08:49.09 | dimm | such as wav, ulaw .... and others |
| 08:51.27 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 08:57.21 | thenk | p3nguin: have you attempted to work with app_rtsp before? |
| 08:57.50 | thenk | i'm aiming my question at you since you seem quite knowledgable |
| 08:57.59 | thenk | +e. |
| 08:58.02 | *** join/#asterisk alvdav (alvdav@gateway/shell/bshellz.net/x-ryfwjgoxinaffomt) |
| 09:00.46 | alvdav | #include means including a file and include means including from a extention? is this correct? |
| 09:01.31 | thenk | include => demo |
| 09:01.35 | thenk | includes the [demo] context |
| 09:01.45 | thenk | #include demo will include demo file |
| 09:02.25 | alvdav | thenk: thank you! |
| 09:02.42 | thenk | if you include => context, any extension after it will override, i think |
| 09:02.59 | thenk | work over, time to go home! |
| 09:03.14 | alvdav | bye |
| 09:03.29 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 09:10.48 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:14.54 | *** join/#asterisk AB3LS (~Geoff@pool-96-249-152-201.hrbgpa.fios.verizon.net) |
| 09:40.37 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:40.40 | schmidts | good morning |
| 09:41.48 | *** join/#asterisk RamaskeZ (~RamaskeZ@ip-87-86-154-74.easynet.co.uk) |
| 09:51.36 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 09:59.38 | dimm | really good |
| 09:59.55 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 10:13.04 | *** join/#asterisk daxt (~daxt@112.135.73.197) |
| 10:18.06 | *** join/#asterisk flashdeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 10:20.47 | flashdeluxe | Hi! I got a question, sometimes i got the problem, that calls are dropped without any obvious reason. As i analyzed the log, i found out that if the problem occurs, there is a message that SIP is "locally bridged" see: http://pastebin.de/15965 How can i avoid this? |
| 10:21.59 | *** join/#asterisk awclin_ (~alinford@80.169.133.251) |
| 10:24.29 | justdave | I've got a user that's not showing up in the Directory() app |
| 10:24.52 | justdave | according to the docs, the names are taken from voicemail.conf |
| 10:24.57 | justdave | and the user is listed there |
| 10:25.30 | justdave | any ideas what to look for? |
| 10:28.56 | kaldemar | flashdeluxe: set directmedia=no or canreinvite=no, depending on your asterisk version. |
| 10:31.52 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:32.56 | justdave | ooooh, she is listed. She has a recorded name on her mailbox that isn't her. |
| 10:33.07 | flashdeluxe | kaldemar: it is already set up "=no" i am using asterisk 1.8 |
| 10:33.24 | justdave | something tells me the previous owner of that extension didn't get their voicemail deleted properly |
| 10:33.51 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:34.20 | flashdeluxe | here is my sip.conf http://pastebin.de/15966 |
| 10:42.28 | justdave | flashdeluxe: you have canreinvite and not directmedia. 1.8 needs the directmedia one. canreinvite is for the older versions. |
| 10:43.08 | justdave | (I'm guessing so anyway, the docs for 1.8 don't even list canreinvite anymore) |
| 10:44.18 | kaldemar | canreinvite is still supported in the code. |
| 10:45.51 | flashdeluxe | klademar: are you sure? in asterisk 1.8, too? |
| 10:47.06 | kaldemar | flashdeluxe: i'm looking at chan_sip.c for 1.8.3 here. |
| 10:48.02 | kaldemar | flashdeluxe: on the other hand, your sip.conf paste shows a peer by the name 85 that has the setting, and the CLI output shows peers 53 and 41. |
| 10:48.21 | kaldemar | so, you still haven't shown that you have the setting... |
| 10:52.15 | *** join/#asterisk awclin_ (~alinford@80.169.133.251) |
| 10:57.02 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:58.12 | flashdeluxe | kaldemar: the settings are for all peers the same, i pasted only an output, because the sip.conf is very long, but if you want to be sure, i can paste the whole thing: http://pastebin.de/15968 |
| 11:03.27 | alvdav | hi again, im new to asterisk so i have a little problem to identifiy the extention wich will be used when someon calls from extern |
| 11:03.37 | alvdav | any hints how i can figure it out? |
| 11:04.54 | alvdav | problem is, i have a freepbx dialblan and i have to creat it for a asterisk installation |
| 11:05.01 | alvdav | dialplan* |
| 11:05.15 | alvdav | i have to recreate it* |
| 11:07.04 | kaldemar | alvdav: you'll have better luck in #freepbx. |
| 11:07.24 | alvdav | kaldemar: hmm yeah... sry |
| 11:07.30 | alvdav | but thx |
| 11:12.19 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 11:21.31 | skrusty | morning all |
| 11:22.35 | alvdav | morning skrusty |
| 11:22.49 | kleszcz | hi |
| 11:23.12 | skrusty | does anyone know if there is a realtime version of agents? i've seen in mantis (sorry, issue tracker) that there is a patch from 2005, but it seems it never made it into 1.6 |
| 11:23.21 | skrusty | although i am not sure... anyone got any hints? :) |
| 11:24.16 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 11:40.42 | *** join/#asterisk _zoom_ (~Admin@196.1.219.211) |
| 11:41.24 | _zoom_ | hello, in sip.conf does sip channel config override general configuration? |
| 11:42.45 | kaldemar | _zoom_: yes. but not all parameters are valid for both sections. |
| 11:42.52 | wdoekes2 | channel? you mean device |
| 11:46.37 | Lantizia | I've got an old 90's PC with a dial up modem, I'm thinking of connecting it to a PAP2T (2xFXS SIP adapter), and then perhaps registering something as an asterisk "peer" that can speak dial up to simulate a fake ISP... will only be going over this LAN, slow baud rate and good codec so should be ok... idea what this "something" than can speak dial up could be? |
| 11:46.44 | Lantizia | *any idea... |
| 11:47.41 | Lantizia | like a sip soft modem :) |
| 11:52.34 | *** join/#asterisk Diffen (~diffen@90-227-126-239-no16.business.telia.com) |
| 11:54.32 | Diffen | Evning im throwing out a long shot here but i have a linksys pap2 connect to my asterisk and when i receive a call and then the callee hangs up the call a bye are sent to the linksys but the call in the handset on the linksys pap2 are not terminated. Anyone here that have seen this problem? |
| 11:54.38 | _zoom_ | kaldemar: allow, disallow codecs |
| 11:55.45 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 11:57.23 | *** part/#asterisk _zoom_ (~Admin@196.1.219.211) |
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| 12:12.18 | *** join/#asterisk zooz (~zooz@188-220-41-110.zone11.bethere.co.uk) |
| 12:14.39 | wdoekes2 | Lantizia: I believe trying to do voice over a modem is a recipe for failure |
| 12:14.54 | wdoekes2 | but I may misunderstand what you're trying to do |
| 12:16.55 | Lantizia | yup you are :P |
| 12:17.19 | *** join/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 12:17.31 | *** part/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 12:33.39 | *** join/#asterisk cjk (~cjk@85.93.217.128) |
| 12:33.43 | cjk | j #sangoma |
| 12:34.52 | cjk | hello, does anyone know if the Digium Hybrid Cards BRI cards support Hardware Echo Cancellation on the BRI channels? |
| 12:35.15 | *** join/#asterisk ahuemer (ahuemer@83-65-26-208.static.xdsl-line.inode.at) |
| 12:36.04 | ahuemer | hi, beginner question. i just set up asterisk for the first time and tried exten 500, the iax2 demo, which does not work |
| 12:36.27 | ahuemer | are there any known problems with pbx.digium.com or is it likely that i made a configuration mistake ? |
| 12:37.01 | wdoekes2 | ~newbook |
| 12:37.01 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 12:37.11 | ahuemer | asterisk 1.8.2.3 on linux x86 btw |
| 12:37.47 | ahuemer | wdoekes2: is this a response to my question? |
| 12:38.02 | Dovid | ahuemer: Can you PB your config ? |
| 12:38.03 | Dovid | ~pb |
| 12:38.03 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 12:38.32 | ahuemer | sure, just extenstions.conf ? or others too ? |
| 12:38.41 | wdoekes2 | ahuemer: yes it is. I don't know about default configs, but I do know that the book should be quite helpful in setting up a basic config |
| 12:39.00 | Dovid | ahuemer: A debug will help the most. |
| 12:39.05 | Dovid | most likely a config issue |
| 12:39.27 | ahuemer | what do you mean by "a debug" ? |
| 12:39.50 | Dovid | sip set debug on |
| 12:40.18 | Dovid | oops. ur doing IAX |
| 12:40.19 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 12:42.09 | kaldemar | ahuemer: the sample configs should not be used. don't expect any sample extensions to work. |
| 12:43.54 | ahuemer | kaldemar: i adjusted the configs so that i got a dialtone on my cisco 12sp, the voicemail demo (8500), the echo demo (600) work as expected. the iax2 test gives me the woman's voice, but then |
| 12:44.03 | ahuemer | [Mar 10 13:41:14] WARNING[4699] chan_iax2.c: Call rejected by 216.207.245.8: Unable to negotiate codec |
| 12:44.06 | ahuemer | in the log |
| 12:44.22 | ahuemer | i just found out that there is a "iax2 set debug on" option |
| 12:44.27 | Lantizia | is T38 just for faxes or could it carry like a dial up internet connection? |
| 12:44.29 | ahuemer | should i pastebin the output? |
| 12:44.57 | Dovid | ahuemer: AH ok. than thats ur issue |
| 12:45.08 | ahuemer | o.O |
| 12:45.17 | ahuemer | wut? |
| 12:45.25 | Dovid | the device on: 216.207.245.8 does not like the codecs that you are using |
| 12:45.44 | Dovid | can you pb your iax.conf ? |
| 12:45.53 | ahuemer | sure |
| 12:46.41 | ahuemer | i did not do any modifications to this file, maybe that's the problem |
| 12:47.07 | Dovid | ahuemer: We know what the issue is. we just need to modify ur config's to what the remote IP wants |
| 12:47.45 | ahuemer | http://pastebin.com/cWT8KHCE |
| 12:50.15 | Dovid | ahuemer: where do you have 216.207.245.8 ? or is it going to something.digium.com ? |
| 12:50.47 | ahuemer | there is the following in extensions.conf: |
| 12:50.53 | ahuemer | exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo |
| 12:50.57 | Dovid | ah ok |
| 12:51.01 | ahuemer | and that resolved to that ip |
| 12:51.33 | Dovid | so when you make a call Unless you have it set in sip.conf or iax.conf (for instance [marko]). |
| 12:51.50 | Dovid | if you had exten => 500,n,Dial(IAX2/guest@marko/s@default) ; Call the Asterisk demo |
| 12:52.03 | kaldemar | ahuemer: you should allow some codecs. first disallow all and then allow what you want in the preferred order. |
| 12:52.03 | Dovid | then it will use the settings that you have under [marko] |
| 12:52.24 | Dovid | however whe you call to a URL or IP directly it uses what you have under the [general] seciton |
| 12:52.51 | Dovid | under there you have: |
| 12:52.53 | Dovid | # |
| 12:52.54 | Dovid | ;allow=all ; same as bandwidth=high |
| 12:52.54 | Dovid | # |
| 12:52.54 | Dovid | ;disallow=g723.1 ; Hm... Proprietary, don't use it... |
| 12:52.54 | Dovid | # |
| 12:52.54 | Dovid | disallow=lpc10 ; Icky sound quality... Mr. Roboto. |
| 12:52.54 | Dovid | # |
| 12:52.55 | Dovid | ;allow=gsm ; Always allow GSM, it's cool :) |
| 12:53.00 | Dovid | what you want to have there is: |
| 12:53.12 | kaldemar | ahuemer: but still, don't expect any samples to work. |
| 12:53.33 | Dovid | disallow=all |
| 12:53.33 | Dovid | allow=ulaw |
| 12:53.33 | Dovid | allow=alaw |
| 12:53.40 | kaldemar | Dovid: please don't flood. |
| 12:54.07 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 12:54.08 | Dovid | this way when you make the call to Digium you are telling them that you are willing to use Ulaw or alaw. these two codecs are supported by almost any every SIP device |
| 12:54.10 | mzb | turns off the alarm bells |
| 12:54.44 | Cadey | Hi guys, is CURL a way to use a website to return values i can place in variables on the dialplan? |
| 12:55.16 | Cadey | or is there a better way to use HTTP to ger variables into a dialplan |
| 12:56.04 | ahuemer | Dovid: kaldemar: thanks for the help so far |
| 12:56.08 | kaldemar | Cadey: func CURL is the best option if you want to use a dialplan function or application. |
| 12:56.30 | ahuemer | i am a bit confused since you two semm to talk about different things |
| 12:56.44 | ahuemer | i just tried various settings of allow and disallow in iax.conf |
| 12:56.58 | ahuemer | even with allow=all the demo did not work |
| 12:57.11 | ahuemer | kaldemar: i think you wanted me to do something else |
| 12:57.22 | Cadey | kaldemar : cool thanks mate. Am I also right in thinking all the page needs to return is XXX and XXX is what is CURL application returns? |
| 12:57.26 | ahuemer | i didn't get what that really was. could you please describe in different words? |
| 12:57.58 | Dovid | ahuemer: What what was ? |
| 12:58.08 | Dovid | my last post about codecs ? |
| 12:58.30 | ahuemer | Dovid: yes, i tried that |
| 12:58.46 | ahuemer | still came warning in the logs and no connection |
| 12:58.51 | kaldemar | ahuemer: well, as i said, you shouldn't expect it to work. it is just _a sample_. |
| 12:59.13 | Dovid | kaldemar: All he has is a codec issue so he needs to tweak that |
| 13:00.00 | kaldemar | Dovid: or digium has disallowed all codecs on their box... |
| 13:00.26 | ahuemer | kaldemar: i understand that. it seems to me that the provided examples are quite close to a working configuration. could you help me adjust the setttings to something useable? |
| 13:00.34 | Dovid | kaldemar: would need a trace for that. but why would they ? |
| 13:00.40 | ahuemer | kaldemar: well, yes, that was my first thought |
| 13:01.04 | kaldemar | Dovid: so that they wouldn't get a gazillion test calls a day for no purpose. |
| 13:01.10 | ahuemer | to be precise: i thought the iax2 service might not be available atm |
| 13:01.18 | Dovid | Then they would just turn it off ;) |
| 13:01.25 | Dovid | ahuemer: Run this |
| 13:01.26 | Dovid | iax2 set debug on |
| 13:01.26 | ahuemer | nmap isn't able to check without a doubt for a udp port... |
| 13:01.28 | Dovid | in asterisk |
| 13:01.31 | Dovid | then pb it |
| 13:01.53 | kaldemar | Cadey: i suppose so, but test it with app Verbose or NoOp. see also func CURLOPT. |
| 13:02.18 | Cadey | thanks mate :) |
| 13:02.43 | ahuemer | http://pastebin.com/JisykmQV |
| 13:02.59 | kaldemar | ahuemer: your warning said that the call was rejected by the far end. that means that there is a signaling connection and you got a reply. |
| 13:03.14 | ahuemer | thanks, that makes things clearer |
| 13:03.48 | Dovid | seems like ur just sending ulaw and nothing esle |
| 13:04.52 | *** join/#asterisk Akiraa (~Akira@82.76.237.183) |
| 13:09.19 | ahuemer | same with bandwith= medium or high |
| 13:09.43 | ahuemer | oh, allow=all seems to do the same |
| 13:10.29 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:10.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:10.59 | ahuemer | kaldemar: no other hints than the samples won't work? |
| 13:11.46 | kaldemar | ahuemer: no. don't bang your head against that wall. there are plenty of others waiting for you. |
| 13:12.41 | ahuemer | so i am going to read through all docs before i can make it work myself |
| 13:13.10 | *** part/#asterisk ahuemer (ahuemer@83-65-26-208.static.xdsl-line.inode.at) |
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| 13:25.40 | zooz | I have a number of internal SIP clients behind the NAT (single IP address) and they all need to connect to one VoIP gw via SIP, but for some reason, some of them randomly refuse to register. what could be the cause of that? |
| 13:31.00 | Cadey | Hi, did the patch for CURL to send it a timeout at paramater 3 get included into 1.8? |
| 13:31.07 | Cadey | as it doesnt seem to work |
| 13:31.17 | Cadey | https://issues.asterisk.org/view.php?id=12920 <-- this patch |
| 13:32.24 | Cadey | arrr sorry Kaldemar, i guess thats why you told me to look at CURLOPT!! |
| 13:32.25 | Cadey | my bad |
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| 14:04.22 | Katty | hi |
| 14:05.14 | schmidts | Morning Katty! |
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| 14:14.45 | Jinxed- | Has anyone gotten asterisk to connect with the 3cx softphone? |
| 14:15.20 | pigpen | Hi all. Would anyone know where I could find the realtime database table structure for the 1.8.x builds? I have google'ed for some time with no luck. I have found 1.6.1.18, but I have heard there are some new fields. |
| 14:17.10 | dimm | please say some words related to my problem. When i dialing to some number , then i can not hear tone signal at ~30 seconds, then i can hear my opponent. |
| 14:20.10 | pigpen | dimm, now, I just jumped in, but what you posted doesn't make sense. If you dialed "some" number, you would have had a ring after that. The dialtone is *before* you dial. Maybe it is just me....I do need more coffee. |
| 14:21.03 | dimm | pigpen, my english is to low to change 'some number' to 'some number at the city' , did your understand me? |
| 14:21.58 | pigpen | ah...I understand about the translation. I don't understand myself half the time. |
| 14:22.21 | dimm | pigpen, ok, i just try to use translate.google |
| 14:22.21 | pigpen | so, are you saying it doesn't ring after you make a call? |
| 14:22.53 | dimm | pigpen, it ringing, but i not hear opponent.... after 15 seconds i can hear opponent |
| 14:23.06 | dimm | ok |
| 14:23.07 | dimm | look |
| 14:23.26 | dimm | me try dial from softphone 1234567 |
| 14:23.46 | dimm | me start hearing in headphones |
| 14:23.52 | dimm | not hearing |
| 14:24.02 | dimm | after 15 seconds i can hear my opponent |
| 14:24.22 | dimm | i just ask them "how long you say me 'hello' ?" |
| 14:24.33 | dimm | she answering that approx. 15 seconds |
| 14:24.56 | pigpen | Oh, are you referring to a delay in audio? |
| 14:25.03 | dimm | yes |
| 14:25.18 | dimm | but only on some numbers :) |
| 14:25.47 | pigpen | Most delays are due to poor internet connection, high latency. Is this over the internet or a wide area network? |
| 14:26.26 | dimm | this is call via sip... hm... from my softphone, via asterisk, to city number |
| 14:26.56 | dimm | over the internet |
| 14:27.28 | pigpen | any chance the "city number" is a VOIP too? When you add latency of a VOIP call to another endpoint that is also a VOIP call, sometimes the issue is made worse, much worse. |
| 14:27.46 | pigpen | Are you hearing echo? (ie: yourself when you talk) |
| 14:28.27 | dimm | pigpen, not, i'm not hear myself |
| 14:28.40 | dimm | pigpen, now i try to know if my opponent use VoIP |
| 14:28.46 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:28.51 | pigpen | yeah, I bet that is what it is. |
| 14:29.04 | pigpen | and that would explain why some has the issue, and some don't. |
| 14:29.05 | benngard | can some1 tell me whats wrong with: exten => 0317998975,1,Set(CONNECTEDLINE(all)=Magnus Benngard <0317998975>) |
| 14:29.21 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 14:29.48 | *** join/#asterisk daxt (~daxt@112.135.93.223) |
| 14:30.11 | wdoekes2 | CALLERID(all)= ? |
| 14:30.36 | benngard | exten => 0317998983,1,Set(CONNECTEDLINE(all)=Asa Axelsson <0317998983>) <= that line works! |
| 14:30.37 | pigpen | yeah, I have never used CONNECTEDLINE, I had to google it. |
| 14:31.05 | benngard | i cant see the difference |
| 14:31.05 | pigpen | ....and still don't know what it does. |
| 14:31.12 | wdoekes2 | hehe |
| 14:31.17 | benngard | it send the info back to the caller |
| 14:31.33 | wdoekes2 | ah COLP or whatever it's called |
| 14:31.44 | dimm | pigpen, yes, my opponent use VoIP too , but another provider instead me. |
| 14:31.59 | benngard | where nice feature, i dial 983, translates into 0317998983 and i see Asa Axelsson... on my display |
| 14:32.41 | wdoekes2 | are you sure it's not working? perhaps your phone already has some other name cached? |
| 14:32.55 | pigpen | dimm, yeah, you are adding voip latency to voip latency on separate providers (ie: you and them). If it were all controlled by you, it wouldn't be as bad, but still probably have issues. |
| 14:32.55 | wdoekes2 | it this SIP? do a sip debug and watch the messages |
| 14:33.06 | pigpen | dimm, tell them to get a real phone. ;-) |
| 14:34.17 | pigpen | hmm..CONNECTEDLINE seems to be rather experimental. I wouldn't use it in production. |
| 14:34.39 | pigpen | as the docs say...."Regardless this feature has become complicated." |
| 14:35.00 | pigpen | This tells me: Don't use it if you want to get some sleep. |
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| 14:36.18 | *** join/#asterisk a_m_y (~IceChat7@121.96.33.130) |
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| 14:37.34 | benngard | [Mar 10 15:41:49] WARNING[23574]: pbx.c:3846 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 14:37.34 | benngard | [Mar 10 15:41:49] WARNING[23574]: pbx.c:3476 func_args: Can't find trailing parenthesis for function 'CALLERID(num'? |
| 14:38.05 | benngard | it claims t miss a } but where? |
| 14:38.16 | a_m_y | hi any howto for queues and how to log-in and log-out on that queue. thanks |
| 14:38.35 | a_m_y | ~queues |
| 14:38.35 | benngard | i have like 100 lines like that so it must be something on that line but i cant find what it is |
| 14:42.47 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:43.18 | pigpen | a_m_y, google. There are many good references. Also, check out the queues.conf, it is ridded with information. |
| 14:45.40 | a_m_y | yes tried queue remove member Agent/<exten> <queue> |
| 14:46.22 | a_m_y | thanks pigpen |
| 14:49.41 | Cadey | Hi guys, you can now use patterns on incomming calls in the dial plan? |
| 14:49.53 | Cadey | I ase because its not working for incomming calls for me? |
| 14:50.05 | WIMPy | now? |
| 14:50.20 | Cadey | sorry, not |
| 14:51.26 | Cadey | am i best to use the s extension and then use the ${EXTEN} variable to test things? |
| 14:52.50 | WIMPy | s will only match if no number has been received. Maybe you should be a little more specific. |
| 14:53.22 | Cadey | ok WIMPy, I have calls being sent directly to my PBX like so NUMBER@IP |
| 14:53.40 | Cadey | but when i receive numbers from that server i cannot use patterns in the dial plan |
| 14:53.43 | Cadey | so for example |
| 14:53.55 | Cadey | a call to 1999@IP i see as 1999 |
| 14:54.05 | Cadey | but i cant use say _199X to catch it |
| 14:54.10 | Cadey | it has to be exact |
| 14:55.28 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:55.38 | WIMPy | There must be something else going on. Try to increase verbos and debug to try and see what's going on. |
| 14:57.04 | Cadey | well thats the crazy part there is nothing in the output |
| 14:57.18 | Cadey | awww crap |
| 14:57.21 | Cadey | im so stupid! |
| 14:57.24 | Cadey | sorry :( |
| 14:57.41 | Cadey | I forgot the best pattern is matched regardless of the order they are in the dial plan |
| 15:00.24 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 15:05.58 | dimm | ok! |
| 15:06.16 | dimm | working day is finished... at my time zone :) |
| 15:06.22 | dimm | see you later |
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| 15:15.12 | *** join/#asterisk sourcode (~code@ppp-58-8-108-29.revip2.asianet.co.th) |
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| 15:21.13 | cjk | hello, does anyone know if the Digium Hybrid Cards BRI cards support Hardware Echo Cancellation on the BRI channels? |
| 15:21.46 | WIMPy | What's that one called? |
| 15:22.42 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 15:24.16 | m_tadeu | my asterisk server is behind a nat. I'll ask someone to check these settings for sip clients. client behind nat(nat=yes, qualify=yes), client has public ip(nat=no, qualify=no), client inside asterisk nat(nat=no, qualify=no)....is this correct? |
| 15:26.02 | *** join/#asterisk joachim_- (~joachim@ti0206a340-dhcp0479.bb.online.no) |
| 15:26.05 | joachim_- | Hi! |
| 15:26.29 | joachim_- | What is the difference between highpriority=yes|no in asterisk.conf ?? |
| 15:27.49 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 15:27.57 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 15:34.09 | schmidts | m_tadeu yes but you should use qualify also to check if a client is reachable and not only for nat keep alive. |
| 15:45.51 | m_tadeu | schmidts: I have an agent inside the local network of asterisk answering on a queue, and another client on another nat. when asterisk answers the call, the client can hear the answering machine and then going to the queue. But when the agent gets the call, no sound goes either way |
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| 15:48.10 | *** part/#asterisk edersohe (~edersohe@201.161.16.236) |
| 15:49.55 | joachim_- | What is the difference between highpriority=yes|no in asterisk.conf ?? When running a realtime system |
| 15:50.24 | joachim_- | Should it be set to yes or no in a production system with over 100 phones? |
| 15:51.44 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 15:51.57 | puzzled | hi |
| 15:53.59 | *** join/#asterisk path (~path@46.29.254.241) |
| 15:55.46 | schmidts | joachim_ only if you have only one cpu and an old system and if you run asterisk not as root |
| 15:56.40 | schmidts | joachim_ just set it to yes, this is what the default /etc/init.d will do, so why not ;) |
| 15:56.48 | a_m_y | ~sipnat |
| 15:56.49 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
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| 16:04.54 | jaytee | } is a brace not a parenthesis ) |
| 16:11.53 | leifmadsen | (parenthesis), [bracket], {brace} |
| 16:21.06 | *** join/#asterisk gpled (~gpled@69.1.105.42) |
| 16:25.41 | GTXComm | :q |
| 16:25.46 | GTXComm | woops |
| 16:29.30 | *** join/#asterisk javier_cintron (~javier_ci@189.169.211.101) |
| 16:30.05 | javier_cintron | Hi there. I think I have found a bug in asterisk 1.8.3, where could I report it? |
| 16:31.36 | leifmadsen | sip:31337@shifteight.org |
| 16:31.39 | ChannelZ | issues.asterisk.org - check first |
| 16:31.48 | leifmadsen | please call that if you can to help us test HD conferencing (G.722 not necessary) |
| 16:32.04 | leifmadsen | ISN: 31337*1273 as well |
| 16:32.19 | ChannelZ | HD conferencing? |
| 16:32.31 | The_Boy_Wonder | yep |
| 16:32.38 | The_Boy_Wonder | load testing, trying to break it |
| 16:33.07 | ChannelZ | oh. Whew. I thought maybe it meant you could still hear me farting even when I put my hand over the phone |
| 16:33.23 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 16:33.46 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 16:34.03 | ChannelZ | runs off to work - can call from there in 20 if you're still going |
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| 16:35.35 | javier_cintron | ChannelZ: Thank very much. |
| 16:39.43 | leifmadsen | testing complete |
| 16:39.45 | The_Boy_Wonder | thanks everyone! |
| 16:39.52 | GTXComm | Q: Sip URI setup, inbound and outbound.. Is this easy to set up? I have been looking for a little online; however, need some help. |
| 16:40.11 | leifmadsen | GTXComm: that is covered in ... |
| 16:40.14 | leifmadsen | ~newbook |
| 16:40.14 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 16:40.19 | GTXComm | thanks. |
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| 16:49.11 | *** join/#asterisk leafartn (~chatzilla@190.145.253.51) |
| 16:51.10 | three60 | Hi all. My setup is : astersisk server and SIP/Sam at location A, SIP/Owen at location B(outside of the local network). Both behind nat firewalls. Both phones can make and receive calls, but SIP/Owen has no audio. I don't see any errors from CLI. Here's the last few lines from the cli output and sip.conf (http://pastie.org/1653170). new to this stuff. Is this an astersisk issue, or is it more likely the settings for the SIP/Owen phone? |
| 16:52.55 | leafartn | Hi! in asterisk 1.8 the cmd agentCallbackLogin is removed, but how can I log in the agents with callback? |
| 17:00.19 | *** join/#asterisk hfb (~hfb@pool-98-112-220-203.lsanca.dsl-w.verizon.net) |
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| 17:00.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:04.30 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 17:04.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:05.07 | *** join/#asterisk shamelessn00b (~chatzilla@175.110.242.37) |
| 17:05.11 | shamelessn00b | ~itsp |
| 17:05.11 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 17:05.15 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 17:05.28 | shamelessn00b | ~itsplist-us |
| 17:05.29 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 17:08.08 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 17:10.25 | leifmadsen | leafartn: you build the dialplan to do that -- using Local channels and such |
| 17:11.20 | leifmadsen | leafartn: http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288932 |
| 17:11.53 | *** join/#asterisk _zoom_ (~Administr@41.218.33.142) |
| 17:12.27 | _zoom_ | hello how to replace ringback with music sound? |
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| 17:27.18 | leifmadsen | _zoom_ you should have waiting for a couple more minutes and I would have told you, "core show application dial" -- option m |
| 17:27.24 | leifmadsen | s/waiting/waited/ |
| 17:27.47 | leifmadsen | but you didn't, so now you'll have no idea |
| 17:29.59 | leafartn | leifmadsen: thank you! |
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| 17:41.17 | *** join/#asterisk nixxer (~nixxer@cpe-72-185-141-70.tampabay.res.rr.com) |
| 17:41.24 | nixxer | anybody know how to speed up the voicemail ivr? |
| 17:41.39 | nixxer | ..or record a new one? |
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| 17:51.50 | leifmadsen | nixxer: just record new prompts |
| 17:51.56 | mocker | nixxer: Did you take a look at the vm-*.gsm in /var/lib/asterisk/sounds/ |
| 17:52.07 | leifmadsen | mocker: you can't assume *.gsm |
| 17:52.15 | leifmadsen | I don't even install gsm prompts anymore |
| 17:52.28 | mocker | leifmadsen: True, was just trying to point him to the proper files. |
| 17:52.28 | leifmadsen | ls vm-* would be better |
| 17:52.29 | mocker | :) |
| 17:52.57 | leifmadsen | and if you're using 1.6.x or later, it'll likely be in /var/lib/asterisk/sounds/en/ (unless you're using french, spanish, or australian english) |
| 17:53.11 | mocker | psh, 1.4 for life! |
| 17:53.12 | mocker | er.. :) |
| 17:53.21 | nixxer | there's 113 files in sounds/en starting with 'vm-' |
| 17:53.43 | mocker | nixxer: Probably could script up something that speeds up the promps and does pitch shifting. |
| 17:53.57 | mocker | or... record a bunch of new sounds. |
| 17:54.00 | leifmadsen | ableton live is good at that :) |
| 17:54.03 | nixxer | how do i do that? |
| 17:54.21 | leifmadsen | nixxer: make sure you're just looking at a single format as there could be multiple formats listed |
| 17:54.31 | nixxer | they're all *.gsm |
| 17:54.47 | leifmadsen | you could see what 'sox' could do |
| 17:55.02 | leifmadsen | increasing the spead of the files without affecting pitch isn't all that trivial |
| 17:55.03 | mocker | That's what I would look into first. |
| 17:55.43 | mocker | http://markplusplus.wordpress.com/2006/10/01/pitch-correct-play-speed-with-mplayer/ |
| 17:56.10 | Cadey | can anyone see why this is happening please? |
| 17:56.24 | mocker | leifmadsen: Need pitch shift dialplan app. :P |
| 17:56.48 | Cadey | http://pastebin.com/T4hrZwRB |
| 17:56.57 | Cadey | im testing to see if the value of number is not "" |
| 17:57.14 | leifmadsen | mocker: core show function PITCH_SHIFT() in 1.8 and later |
| 17:57.15 | Cadey | and in the verbose you can see its no, but its still calling panicCall and not ringNumber? |
| 17:57.27 | leifmadsen | mocker: it does exist, it just changes pitch though, and not speed |
| 17:57.40 | mocker | leifmadsen: horrible. |
| 17:57.48 | leifmadsen | http://ofps.oreilly.com/titles/9781449303822/ch03.html#c03-AudioManipulation_id346128 |
| 17:57.59 | leifmadsen | The_Boy_Wonder: mocker says PITCH_SHIFT() is horrible |
| 17:58.18 | The_Boy_Wonder | WHAT |
| 17:58.21 | The_Boy_Wonder | well |
| 17:58.25 | The_Boy_Wonder | no that's pretty accurate |
| 17:58.32 | mocker | lol |
| 18:00.29 | mocker | Anyone know if updating from dCAP 1.2 to something more current requires the whole test or an abridged version? |
| 18:02.46 | pabelanger | mocker: Rewrite the test, I believe |
| 18:03.29 | nixxer | thanks guys |
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| 18:08.55 | leifmadsen | The_Boy_Wonder: :) |
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| 18:36.43 | citywok | has anybody used the new aastra firmware that was recently released? It caused problems for some of my phones. |
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| 18:39.50 | *** join/#asterisk Vrtigo1 (~Vrtigo1@vpn.lpga.com) |
| 18:41.25 | Vrtigo1 | Anyone know of a method to check DID availability? I.E. If I want 123-456-78XX, is there any way to check to see if those numbers are currently in use other than googling them / calling them? |
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| 18:54.32 | p3nguin | vrtigo1: There's a database for that sort of thing. I don't know how you can gain access to it, though. |
| 18:54.42 | Vrtigo1 | Do you know what the name of it is? |
| 18:57.10 | p3nguin | I unfortunately do not know. I know one person that might, but he's not here right now. |
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| 19:17.44 | Katty | peeks in |
| 19:18.01 | Katty | hello my asterisk does not work at alL HOW TO FIX PLS??? |
| 19:19.34 | chazzam | Katty, you obviously need http://www.youtube.com/watch?v=qRuNxHqwazs to fix it |
| 19:19.40 | *** join/#asterisk a_m_y (IceChat7@112.204.234.44) |
| 19:19.54 | a_m_y | ~book |
| 19:19.54 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 19:20.05 | chazzam | otherwise you won't be able to carry it out to the end |
| 19:23.00 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 19:25.52 | logicwrath | does anyone know of any softphones that support auto-answer from the SIP header? When I use TAPI I want my phone to auto-answer. |
| 19:28.03 | fauxalliance | logicwrath, no, sorry.. but they could dial into a queue from a softphone... and have the queue 'BEEP' and pass the call. |
| 19:28.50 | fauxalliance | idk of any softphone that can AA... google may differ |
| 19:28.59 | logicwrath | are there any softphones that support intercom and paging? |
| 19:29.17 | fauxalliance | i'd have to google |
| 19:29.25 | logicwrath | yea, np |
| 19:30.00 | fauxalliance | logicwrath, fwiw, i have to endeavour this project next week for the shed, else get a 'wireless door-chime' ;) |
| 19:30.18 | Katty | chazzam: what is that |
| 19:30.24 | Katty | chazzam: will that fix my asterisk???? |
| 19:30.57 | logicwrath | well, check out ninja pro softphone from global ip |
| 19:31.06 | logicwrath | i think they support it but there trial wont register on my server |
| 19:31.06 | fauxalliance | chazzam, fixed my proxy ;) |
| 19:31.08 | logicwrath | so i cant be sure |
| 19:31.26 | chazzam | Katty, it will fix your asterisk, and then the world! and it fixes proxies too! |
| 19:31.34 | chazzam | :) |
| 19:31.39 | Katty | that is just adorable. |
| 19:31.53 | fauxalliance | logicwrath, shame.... add +GPL to the search query |
| 19:32.07 | Katty | where is everyone, anyway? |
| 19:32.15 | Katty | Qwell: and you owe me an update. |
| 19:32.55 | fauxalliance | Katty, I owe you a HUG UPDATE! |
| 19:33.09 | fauxalliance | hugs Katty |
| 19:33.51 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 19:34.19 | *** join/#asterisk _zoom_ (~Administr@41.218.33.83) |
| 19:34.23 | _zoom_ | hello, can I do stopplaytones in other context where i did playtones? |
| 19:35.10 | fauxalliance | steps away |
| 19:36.56 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 19:37.14 | raden_work | there are files in my moh directory but i have no music playing on hold ? |
| 19:40.23 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:43.42 | p3nguin | vrtigo1: Talk to raden_work about access to that database. |
| 19:44.00 | Vrtigo1 | thx |
| 19:44.30 | Vrtigo1 | raden_work: p2nguin tells me you may be able to help me figure out a question I have regarding checking the availability of DID numbers. |
| 19:44.46 | Vrtigo1 | err, p3nguin...heh |
| 19:46.03 | raden_work | Vrtigo1, ? what you need |
| 19:46.27 | raden_work | p3nguin, any idea why musiconhold a no go ? ver 1.8 |
| 19:47.05 | raden_work | [default] |
| 19:47.05 | raden_work | mode=files |
| 19:47.05 | raden_work | directory=moh |
| 19:47.05 | Vrtigo1 | raden_work: trying to figure out if there is some way to perform a check for given number blocks, i.e. 123-456-78XX. I need additional DIDs and trying to find available blocks that would fit my existing numbering plan |
| 19:47.27 | raden_work | Vrtigo1, pm me numbers i can lookup quick |
| 19:48.10 | raden_work | Vrtigo1, what state ? |
| 19:48.39 | Vrtigo1 | raraden_work, florida...is there any way to do a wildcard match like 386-XXX-6XXX? |
| 19:49.04 | Vrtigo1 | and find any continuous blocks of 100 that are avalable? |
| 19:49.38 | raden_work | there are non that size available |
| 19:50.29 | raden_work | 386.663.1013 |
| 19:50.29 | raden_work | <PROTECTED> |
| 19:50.29 | raden_work | 386.663.1015 |
| 19:50.29 | raden_work | <PROTECTED> |
| 19:50.29 | raden_work | 386.663.1016 |
| 19:50.29 | Vrtigo1 | what about 386-XXX-X2XX |
| 19:50.29 | raden_work | <PROTECTED> |
| 19:50.31 | raden_work | 386.663.1017 |
| 19:50.33 | raden_work | <PROTECTED> |
| 19:50.37 | raden_work | 386.663.1018 |
| 19:50.39 | raden_work | <PROTECTED> |
| 19:50.41 | raden_work | 386.663.1019 |
| 19:50.43 | raden_work | <PROTECTED> |
| 19:50.45 | raden_work | 386.663.1023 |
| 19:50.47 | raden_work | <PROTECTED> |
| 19:50.49 | raden_work | 386.663.1025 |
| 19:50.53 | raden_work | <PROTECTED> |
| 19:50.57 | raden_work | 386.663.1034 |
| 19:50.59 | raden_work | that the tightest block im seeing |
| 19:51.11 | raden_work | 1035-1038 then 1040-1050 then 1205-1210 |
| 19:51.23 | raden_work | at best i can find a block of 10 in all of the 386 |
| 19:51.29 | Vrtigo1 | is this db you are checking against for a single provider or global availability? |
| 19:51.37 | raden_work | GLOBAL |
| 19:51.45 | raden_work | Were connected to everyone |
| 19:52.02 | raden_work | I have 1205 - 1263 |
| 19:52.08 | raden_work | 386-663 |
| 19:52.31 | raden_work | biggest block I can get yea |
| 19:52.46 | *** part/#asterisk _zoom_ (~Administr@41.218.33.83) |
| 19:52.56 | raden_work | 1.2 cents a min on thoose |
| 19:53.14 | Vrtigo1 | does it show #s that are not assigned to a lec or not assigned to an end user? sorry i am a bit naive on how this works |
| 19:53.42 | raden_work | shows me all numbers that are not in use and available in the pool |
| 19:54.01 | raden_work | its really complex how it works and i dont want to get into the tech on it |
| 19:54.26 | Vrtigo1 | ok...is there a difference between voip / non voip availability? |
| 19:54.34 | raden_work | SOmetimes |
| 19:54.45 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 19:54.55 | raden_work | but there is not even a 10 block available down there in that exchange for landline |
| 19:55.05 | raden_work | and the cost is nuts |
| 19:55.12 | *** join/#asterisk alex-s-77 (~alex-s-77@80-219-214-131.dclient.hispeed.ch) |
| 19:55.27 | raden_work | like through us DID's are $5 a month each 1.2 cents a min |
| 19:55.29 | Vrtigo1 | my provider has several blocks of 100 in 386 for $25/mo, just none that fit the numbering plan i would like to use |
| 19:55.45 | Vrtigo1 | so that is why i was wondering how the db you are looking at works... |
| 19:55.48 | raden_work | Then get them cause they are not showing up here |
| 19:56.26 | raden_work | if a local carrier has numbers and they dont report they are available its not seen basically |
| 19:56.55 | Vrtigo1 | ok, thats what i figured, why i asked if the db showed numbers not assigned to a lec or to an end user |
| 19:57.03 | raden_work | we work with XO , 3cx ,, att , tw telcom , MCI , Level 3 , global crossings and a few others |
| 19:57.06 | Vrtigo1 | thx, thats helpful |
| 19:57.23 | raden_work | be careful what carrier you use |
| 19:57.29 | Vrtigo1 | i imagine my provider gets theirs from one of them since they are a smaller company |
| 19:58.19 | raden_work | 760-386 there is a 100 block |
| 19:58.30 | raden_work | i see 98% of whats available |
| 19:58.35 | Vrtigo1 | 760-386 or 386-760? |
| 19:58.37 | raden_work | I can have lines provisioned in 5 minutes |
| 19:58.43 | Vrtigo1 | i am in 386 area code |
| 19:58.48 | raden_work | 760-386 lol |
| 19:58.51 | Vrtigo1 | oops |
| 20:01.34 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 20:02.03 | fullstop | Hi all. I'm looking into LumenVox, vestec etc integration with asterisk. Is it worth dealing with MRCP? |
| 20:02.23 | fullstop | It sounds like a nice thing to have.. but it is kind of complicated to set up. |
| 20:02.42 | *** join/#asterisk a_m_y (IceChat7@112.204.234.44) |
| 20:02.49 | *** join/#asterisk war9407 (war@71.6.165.232) |
| 20:05.45 | a_m_y | hi, wanted to know the exact package requirements for asterisk 1.4. tnx in advance |
| 20:06.02 | a_m_y | linux package requirement i mean. |
| 20:07.06 | tzafrir | a_m_y, what linux? |
| 20:07.18 | a_m_y | centos 5.5 i386 |
| 20:07.34 | a_m_y | and for 64bit |
| 20:09.32 | alex-s-77 | Good evening all! I'm trying to setup a 1.6.2.9-2 asterisk on Debian Squeeze 64bit. In my sip.conf, I've got a SIP device "2000" defined, but when I try to connect to it from my Mac with the Telephone app, it just doesn't connect. In the "asterisk -vvvvvc" console, I also see nothing. what to do? |
| 20:10.27 | paulc | alex-s-77 how's your dialplan? default context for SIP inbound, and having an extension defined that will hit your 2000 outbound |
| 20:10.28 | raden_work | is there something that needs to be enabled for music on hold in asterisk 1.8 ? |
| 20:10.39 | alex-s-77 | I can connect just fine to other SIP servers (like eg. pbxes.org â this is in no way related to my internet provider) |
| 20:11.02 | alex-s-77 | paulc: i'm new to all this. how do i answer your question best? |
| 20:11.20 | Vrtigo1 | alex-s-77, firewall between your computer and the * server? |
| 20:11.28 | alex-s-77 | Vrtigo1: no firewalls |
| 20:12.12 | alex-s-77 | what's strange - from this pbxes.org, I can make it connect just fine to my * server⦠:/ |
| 20:12.12 | paulc | alex-s-77: Where is your server, your 2000 extension end point device, and the computer (with soft phone?) that you're trying to connect from? Relevant if we're doing networky stuff.. |
| 20:12.29 | *** join/#asterisk eugeneoden (~goden@southeast-invest.oneringnetworks.net) |
| 20:12.33 | paulc | alex-s-77: if not... then it's down to the context that your inbound calls land in.. and whether than context has an extension defined to send calls to your 2000 device |
| 20:13.10 | alex-s-77 | paulc: my server is somewhere in the internet - it's a virtual server. my computer (softphone: Telephone -> http://code.google.com/p/telephone) is, uhm, at home behind a NAT wlan router |
| 20:13.33 | alex-s-77 | paulc: i'm not yet trying to do any calls. just "login". |
| 20:14.05 | paulc | alex-s-77: So you want to have a peer defined in sip.conf for your home device/account.. and get it to register successfully? |
| 20:14.32 | alex-s-77 | paulc: yes. as I said, I'm new to all this and try to follow "the asterisk book" http://www.the-asterisk-book.com/unstable/der-context.html |
| 20:14.33 | paulc | alex-s-77: Once you see it registered, then you want to make a test call (to music on hold, an outbound call, whatever, etc?) - trying to make sure I understand the issue |
| 20:15.10 | alex-s-77 | paulc: correct. first get it "registered", but my softphone always says, that it cannot login. |
| 20:15.56 | alex-s-77 | paulc: it's not related to the softphone, though â with SIPhon on iPhone, it also doesn't work. but these programs work just fine when trying to connect to other sip servers (which I did not setup, which might be the reason why they work *G*) |
| 20:16.24 | paulc | alex-s-77: So the steps are usually.... define a peer in sip.conf.. [homepc] or something.. give it a password.. host=dynamic.. nat=____ maybe.. then use the same credentials to register. |
| 20:16.51 | alex-s-77 | paulc: no work :( my sip.conf: http://tel.skwar.me/s/sip.conf |
| 20:16.56 | paulc | Do you see anything on the console about failed registration attempts etc? (you may need to increase logging - "core set debug 5" and "core set verbose 5") |
| 20:17.23 | alex-s-77 | paulc: no, nothing on console. I started it with -vvvvv |
| 20:17.48 | alex-s-77 | paulc: but when this pbxes.org connects, I do seeSaved useragent "PBX" for peer 2002 |
| 20:17.54 | paulc | hmm.. you could do "sip debug on" too.. but it's weird you're seeing nothing.. is there any software firewall on the server |
| 20:18.05 | alex-s-77 | paulc: no, no firewalls |
| 20:18.24 | alex-s-77 | paulc: No such command 'sip debug on' (type 'core show help sip debug on' for other possible commands) |
| 20:18.34 | Katty | hugs fauxalliance a bit belatedly >.< |
| 20:19.07 | fullstop | always with the hugs. |
| 20:19.16 | alex-s-77 | paulc: it's Asterisk 1.6.2.9-2+squeeze1, the way debian squeeze ships it |
| 20:19.21 | paulc | alex-s-77: sorry, "sip set debug on" |
| 20:19.50 | alex-s-77 | paulc: there's something |
| 20:20.34 | a_m_y | ~sipnat |
| 20:20.34 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 20:20.48 | alex-s-77 | paulc: but⦠how do I understand the output? http://pastebin.com/rVvscw1T |
| 20:22.25 | alex-s-77 | my asterisk is not behind a nat. it's in the internet. |
| 20:23.01 | paulc | alex-s-77 do you have a field in your soft phone for "auth user ID" or similar? |
| 20:25.14 | alex-s-77 | paulc: no, i think. i've got http://min.us/ljIUdc and http://min.us/ljE7yE |
| 20:26.12 | alex-s-77 | paulc: but with bink, I might have. |
| 20:26.18 | alex-s-77 | paulc: what should i enter? |
| 20:27.17 | paulc | alex-s-77: Deutsch? Ganz toll! ;) Have you tried just "2000" for Beschreibung? |
| 20:28.09 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 20:28.20 | paulc | alex-s-77: or in Erweitet, change SIP-Adresse to just "2000" |
| 20:28.32 | alex-s-77 | paulc: yep, deutsch *G* "Sorry" about this :-) i just now tried 2000 for "beschreibung" and "vollständiger name". didn't work :( |
| 20:29.14 | alex-s-77 | paulc: no, also 2000 for sip-adresse doesn't work. :( |
| 20:29.38 | alex-s-77 | thinks * is confusing⦠|
| 20:29.54 | paulc | alex-s-77: 2 secs, let me look at something here |
| 20:30.05 | alex-s-77 | paulc: if you want me to try a different app - tell me which one, and i'll gibe it a try. |
| 20:30.28 | alex-s-77 | paulc: also: feel free to try to connect to it⦠tel.skwar.me - 2000 - 1234 |
| 20:32.07 | paulc | alex-s-77: I just connected fine, using Asterisk as the registering end point. So it half points to the soft client configuration |
| 20:32.23 | paulc | alex-s-77: Maybe I'll set up an endpoint for you to try registering to |
| 20:33.09 | alex-s-77 | paulc: soft client? hm. they work just fine with pbxes.org and also other sip servers! |
| 20:33.15 | raden_work | this is odddd |
| 20:33.19 | raden_work | Qwell, you around bro ? |
| 20:33.38 | raden_work | Qwell, my music on hold will not work at all |
| 20:33.56 | alex-s-77 | paulc: additionally: three clients i tried (telephone, blink and siphon) all don't work. and they all work with other servers!?! |
| 20:34.39 | *** part/#asterisk gpled (~gpled@69.1.105.42) |
| 20:35.50 | raden_work | HOLD is not working with any of our 6133i phones or my 480i but its working with PAP2T when i flash over |
| 20:36.06 | raden_work | then again i think the pap2t does it itself |
| 20:36.46 | mocker | wonders if he can download the rough cut of the new asterisk book as an epub. |
| 20:37.37 | cobra2 | i did something nerdy today. |
| 20:38.12 | cobra2 | I took the asterisk-doc pdf, converted it to text, did a little editing and piped it though espeak to get an audio book split by chapters. |
| 20:38.19 | mocker | hha |
| 20:39.03 | cobra2 | i thought it was kinda cool, just thought i would share the idea |
| 20:41.05 | *** join/#asterisk digitolx (~ssureshot@dynamic-acs-24-154-156-11.zoominternet.net) |
| 20:41.36 | digitolx | what causes a phone to register to a asterisk box? is it the SIP boot file? |
| 20:41.43 | paulc | cobra2: how was the voice quality? |
| 20:41.48 | paulc | digitolx: what kind of phone? |
| 20:42.05 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 20:42.13 | raden_work | ok I can listen to my music on hold if i call the musiconhold extension my pap2t if i flash over it works but non of the phones will work |
| 20:42.19 | digitolx | cisco 7940 and 7960's' |
| 20:42.23 | raden_work | I put someone on hold there is nothing there |
| 20:42.30 | cobra2 | paulc: really good, a little robotic but that's to be expected |
| 20:42.43 | cobra2 | paulc: turned into like 13+ hours of audio |
| 20:43.02 | paulc | cobra2: I've used the Allison Ceptral voice for some dev prompts for an IVR I'm working on. Still makes me laugh the way some things are pronounced :) |
| 20:43.35 | paulc | digitolx: man, it's ages since I've touched those.. but from memory there's a config file you can edit, tied to MAC address.. when it boots, it'll fetch the config, which then contains the registration details |
| 20:43.36 | cobra2 | i used the scottish voice of espeak, just cause i didn't like the english voice |
| 20:43.59 | cobra2 | i also make it talk faster |
| 20:44.17 | paulc | now THAT's gotta be entertaining! :-D |
| 20:44.40 | cobra2 | paulc: very, i can pack it up into a torrent if you want |
| 20:45.24 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 20:46.08 | paulc | cobra2: haha, nah, you're alright.. not sure my head could take it.. I'd rather read the stuff as needed. Thanks though :) |
| 20:46.09 | digitolx | paulc: roger that... so If I have a second asterisk box as a backup and change all the SIP files in /ftfpboot to point to the new box the phones should register to the new asterisk server? |
| 20:46.19 | cobra2 | no worries |
| 20:47.02 | *** join/#asterisk PeterEs (~PeterEs@78-27-57-250.dsl.alice.nl) |
| 20:47.04 | paulc | digitolx: Sounds about right.. although from what I remember, don't the Cisco phones allow for a primary and a backup server, to deal with outages etc? (you wouldn't want users to have to reboot their phones to get the new provisioning information if you had a server outage, right?) |
| 20:48.01 | *** join/#asterisk Cadey (~x@host86-138-17-217.range86-138.btcentralplus.com) |
| 20:48.50 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 20:48.53 | digitolx | paulc: well yeah that sounds like a plan except it wasn't implemented and we have an outage right now so I'm kinda picking up the pieces |
| 20:49.35 | alex-s-77 | is frustrated, that asterisk doesn't work :-( |
| 20:49.45 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 20:50.00 | Cadey | Hi peeps, Ive read using GotoIf(Len($Blah) > 0 ?a:b) is the normal way to check if a variable is empty as the following jsut doesnt seem to work in the dialplan GotoIf($["${Blah}"] = ""?a:b) - is this correct, does the interpriter not like comparing the latter? |
| 20:50.07 | paulc | digitolx: ah, gotcha.. always great in hindsight innit.. for now then, yes, what you described will work and get you back up and running |
| 20:50.20 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:50.24 | paulc | Cadey: Your [] are in the wrong place |
| 20:50.49 | Cadey | I tried $["${Blah}" = ""]? |
| 20:50.56 | Cadey | and that didnt work also? |
| 20:51.12 | paulc | Cadey: GotoIf($["${SomeVar}"="Blah"]?a:b) should work |
| 20:51.21 | leifmadsen | GotoIf($["${Blah}" = ""]?true:false) |
| 20:51.29 | p3nguin | In Cisco SIPDefault.cnf, there's proxy1_address, proxy2_address, proxy3_address, etc. Maybe that's for configuring multiple Call Managers (Asterisks). |
| 20:52.04 | Cadey | arr ok well i tried that but i must have messed somthing up i guess |
| 20:52.19 | Cadey | im used to C# and this syntax is a little new to me :) |
| 20:52.19 | digitolx | paulc: thanks for the input.. and thank god I know how to use sed lol |
| 20:53.08 | paulc | digitolx: hehe no worries - good luck! :) |
| 20:53.47 | p3nguin | I have proxy1_address through proxy6_address all set to the same IP address (addr of Asterisk). I stopped using SIP on my phones, so I don't know if those are for multiple call control devices or not. |
| 20:53.54 | *** join/#asterisk jkroon (~jkroon@dsl-241-236-235.telkomadsl.co.za) |
| 20:54.20 | jkroon | hi guys, i'm seeing lots of log entries like NOTICE[26165] chan_sip.c: Call from '' to extension '00972599178715' rejected because extension not found in context 'default'. |
| 20:54.31 | *** join/#asterisk sahafeez (~sahafeez@4.53.128.211) |
| 20:54.39 | p3nguin | jkroon: That's a good thing. |
| 20:54.45 | jkroon | my issue is that I've got no users with context=default - so how did those INVITE requests even land in my dialplan? |
| 20:54.59 | jkroon | p3nguin, not if there is some memory leak in * eventually causing it to crash. |
| 20:55.00 | paulc | jkroon: what's your context in [general] of sip.conf? |
| 20:55.02 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 20:55.19 | p3nguin | jkroon: It means you accept anonymous users to send calls inbound, and they aren't able to match a valid extension and make calls on your dime. |
| 20:55.23 | jkroon | paulc, default - well spotted. |
| 20:55.46 | jkroon | ok, how do I just prevent anonymous sip invites from even being considered? |
| 20:55.57 | p3nguin | jkroon: There's no memory leak from anonymous calls. |
| 20:56.08 | p3nguin | jkroon: allowguest=no in sip.conf |
| 20:56.52 | p3nguin | If Asterisk crashes from a few phone calls, it's screwed up and needs updated/upgraded. |
| 20:57.57 | p3nguin | The allowguest setting controls the anonymous calls, which start in the context in the general section of sip.conf. |
| 20:58.24 | p3nguin | They start in the context in general because they haven't matched any explicitly configured peers. |
| 20:59.05 | raden_work | Anyone have a idea why music on hold would not work ? |
| 20:59.34 | raden_work | music on hold extension works |
| 20:59.47 | raden_work | PAP2t activates music on hold |
| 20:59.56 | paulc | p3nguin: generally a good idea to have allowguest=no in sip.conf then.. stops all those pesky people trying to guess valid credentials then outdial through you etc.. (been interesting using fail2ban to keep an eye on that previously) |
| 20:59.57 | raden_work | but aastra 9133i and 480i not working |
| 21:00.03 | raden_work | this makes no sense |
| 21:00.41 | paulc | raden_work Can you try with a 1.6? I noticed MOH issues with 1.8 using Dial with the "do macro on answer" thing, combined with M option.. inbound PSTN call didnt' get MOH with 1.8 but did with 1.6 |
| 21:01.02 | raden_work | <PROTECTED> |
| 21:01.11 | raden_work | did not have this issue till we went to 1.8 |
| 21:01.25 | raden_work | is there a fix ? |
| 21:02.39 | p3nguin | paulc: Most people I know need to have anonymous calls allowed because of unknown callers calling an allowed SIP URI on the system. E.g., I allow people that I don't know to call me by SIP URI. |
| 21:03.01 | p3nguin | But if you don't allow unknown people to make calls of any sort, allowguest=no is a must. |
| 21:03.37 | paulc | raden_work: Yeah, same here. So I stuck with 1.6 - is there a solid need/reason for you to run 1.8? |
| 21:03.43 | *** join/#asterisk jkroon (~jkroon@dsl-241-255-111.telkomadsl.co.za) |
| 21:03.52 | p3nguin | It's also a good idea to use the authreject setting to make sure the crackers trying to find valid credentials can't tell the difference between valid and invalid usernames. |
| 21:04.10 | jkroon | annoying network connection. |
| 21:04.11 | paulc | p3nguin: Thanks for the clarification - makes total sense.. and yeah, authreject was the one that was ringing a distant bell at the back of my mind.. |
| 21:04.32 | raden_work | paulc, ill have to download and revert tonight |
| 21:05.00 | jkroon | p3nguin, last I got was you saying there is no memory leak from anonymous calls - was there additional info I could use? like the authreject ? |
| 21:05.03 | raden_work | been this way for a few weeks just got to my desk today |
| 21:05.18 | p3nguin | (1455.45) <jkroon> ok, how do I just prevent anonymous sip invites from even being considered? |
| 21:05.21 | p3nguin | (1456.06) <p3nguin> jkroon: allowguest=no in sip.conf |
| 21:05.58 | paulc | raden_work: it's the little things eh.. I had an issue with AMI dropping calls after a second Redirect. Had to roll back to a version where it works.. kind of a "if it aint broke don't fix it" kinda deal.. but I'd prefer to be running something more "current" generally |
| 21:05.59 | jkroon | p3nguin, added thanks. the authreject thing also seems potentially useful? |
| 21:06.06 | p3nguin | (1457.55) <p3nguin> The allowguest setting controls the anonymous calls, which start in the context in the general section of sip.conf. |
| 21:06.09 | p3nguin | (1458.23) <p3nguin> They start in the context in general because they haven't matched any explicitly configured peers. |
| 21:06.42 | raden_work | paulc, we had to upgrade accounting server asterisk was on that just decided to go with new version but you know how it goes |
| 21:07.02 | p3nguin | Yes, authreject makes all attempts respond with the same thing, regardless if the username is valid or invalid. That makes it /slightly/ more difficult to crack accounts. |
| 21:07.32 | p3nguin | If you can tell which one is good and which one is bad, you can concentrate on the good ones and you have a better chance of gaining access. |
| 21:08.43 | jkroon | p3nguin, ok, so why should there be a difference by default? I would reckon (from limited experience with security systems) that either the username/password combination is valid or not - why would that differ by default with SIP? |
| 21:08.56 | *** join/#asterisk blakers8423232 (~blakers34@209.12.10.117) |
| 21:09.14 | raden_work | paulc, how does something like MOH get messed up i mean for real |
| 21:09.50 | jkroon | additionally, it seems (based on the logs I've got on my systems) that mostly they first nail you with hundreds and hundreds of REGISTER attempts (so subsequently I've set up a type of autoban using the IPs from the logs as a feeder) |
| 21:10.11 | jkroon | raden_work, ?? |
| 21:10.13 | blakers8423232 | I'm using asterisk v. 1.6.1.18. Everytime I send dtmf (rfc28333) asterisk changes the SSRC. How can I stop it from doing this? My carrier won't recognize my dtmf until i do. |
| 21:10.28 | paulc | raden_work: I have no idea. Similar issue with my AMI Redirect issue. I'm not a C coder, but I kinda want to go through the code to check shit out... just takes time, which isn't in abundant supply right now (is it ever?!) |
| 21:10.42 | raden_work | lol |
| 21:10.46 | raden_work | no its not |
| 21:10.49 | raden_work | or id be working on it |
| 21:11.01 | raden_work | i dont get why my PAP2t works but regular sip phones dont |
| 21:11.07 | m_tadeu | should I have externip and externhost defined in sip.conf? or just one of those? |
| 21:11.23 | p3nguin | jkroon: It's sort of like two cases: 1) if you enter in a user/passwd combination and the response is "this is an invalid user." You know the username is no good and you need not try to crack the password for it. 2) You enter a user/passwd and the response is "the password for this user is incorrect." Now you spend time trying to crack the password for a valid user... |
| 21:11.59 | jkroon | p3nguin, i get that, but why (by default) does the responses differ? What's the motivation for having them differ? |
| 21:12.01 | p3nguin | jkroon: The authreject makes all responses the latter -- it says it's a valid user and that the password is incorrect, regardless if the user is really valid or not. |
| 21:13.22 | p3nguin | jkroon: I have no idea what the reasoning was to have a difference. I believe I saw that the default for that parameter in 1.8 is now to show the same response for all attempts. In earlier versions, the default was to differentiate between the two. |
| 21:13.50 | jkroon | what I'm getting at is that if there is a good reason they should differ and that is why it is that way by default then I'll rather leave it and try to detect these hack attempts from the logs and use something like ban2ip to nail the crap at the network layer. |
| 21:13.58 | WIMPy | Where's the fun if you don't leave some backdoors? |
| 21:14.00 | raden_work | im running 1.8.2 what are odds this is fixed in 1.8.3 |
| 21:15.30 | p3nguin | jkroon: I've never seen a reason expressed to differentiate between the two failure types. I would make the adjustment to my system so that all failures look the same until I had a very good reason to keep them separate. |
| 21:15.40 | jkroon | WIMPy, having to have your systems go down two times in as many weeks due to crackers messing with you, having a third nail a client and cause serious financial harm, and having a fourth performing pen-tests against a web server in two weeks, and having to trace them is NOT my idea of fun. |
| 21:15.50 | Cadey | Hello again, We hear a grainy sound at the end of a background() call, it is 100% not in the audio file, does anyone else experience such a noise? the format is ulaw |
| 21:16.09 | jkroon | Cadey, no, never had that. |
| 21:16.24 | Cadey | if you can dial a number in the UK you will hear it |
| 21:16.32 | raden_work | leifmadsen, you around ? |
| 21:16.36 | WIMPy | jkroon: Then you know now that you need to carefully read EVERY configuration option. |
| 21:16.38 | blakers8423232 | I'm using asterisk v. 1.6.1.18. Everytime I send dtmf (rfc28333) asterisk changes the SSRC. How can I stop it from doing this? My carrier won't recognize my dtmf until i do. |
| 21:16.44 | p3nguin | It's a common practice to set alwaysauthreject to yes, so I guess that's why the default changed in 1.8. |
| 21:17.04 | Cadey | try calling 442031373662 |
| 21:17.11 | jkroon | WIMPy, that's what i'm busy trying to do. |
| 21:17.17 | Cadey | you will hear the gritty sound at the end of each background() |
| 21:17.29 | WIMPy | That's actually a very obvious one. There are more complicated ones. |
| 21:17.42 | jkroon | but the client i could do nothing about. using 1234 as your phone secrets is not my idea of secure. |
| 21:18.28 | jkroon | WIMPy, such as? |
| 21:18.39 | p3nguin | cadey: I hear just a little "shish" sound after some of the sentences. Nothing serious. |
| 21:18.51 | WIMPy | RTP, NAT and features |
| 21:19.03 | Cadey | yeah I realise its not serious but wondered what it could be? |
| 21:19.13 | Cadey | as if i can get rid of it i will :) |
| 21:19.55 | p3nguin | cadey: You could take the sound files and edit them with something like audacity; just chop off the last 0.25 second. |
| 21:20.15 | p3nguin | cadey: Is that your system that I called? |
| 21:20.28 | Cadey | p3ngiuin : but its not on the audio files, i checked that? |
| 21:20.33 | Cadey | yes mate its my systm |
| 21:20.41 | paulc | Cadey: Is it actually at the end? I think it's header data at the start of the audio files.. I've had something similar before |
| 21:20.55 | p3nguin | cadey: Did you happen to log the caller ID that I sent with my call? |
| 21:21.02 | paulc | Cadey: saving as "headerless" usually solves the issue - how did you create the prompts? |
| 21:21.18 | Cadey | we recorded them in our sound booth at work |
| 21:21.28 | Cadey | and then coverted them to ulaw from normal wavs |
| 21:21.37 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 21:22.05 | Cadey | p3nguin : nope |
| 21:22.08 | Cadey | did see your callerID |
| 21:22.16 | paulc | Cadey: It's that bit that's adding the header.. you want to save as raw PCM or headerless ulaw.. I think using CoolEdit doing it as wav adds the header |
| 21:22.36 | jkroon | WIMPy, ok, by the time a cracker can get to rtp stuff there is already much more significant problems, and I fail to see how NAT settings can affect system security (typically it'll cause crappy service by dropping calls, one-way voice and other artifacts), and to get to features you need to be able to establish a call not true? |
| 21:22.37 | Cadey | oh right i see paulc thanks |
| 21:22.38 | paulc | if you open the file as raw, you'll see a noisy "click" caused by the embedded header/metadata |
| 21:22.45 | p3nguin | cadey: Okay. I was hoping you did, so you could tell me what showed up. I rarely call international numbers directly, so I was curious if my caller ID was formed properly. |
| 21:23.00 | Cadey | well i can put it in a noop and tell you mat? |
| 21:23.29 | jkroon | Cadey, i always use sox to convert to .wav and then asterisk's file convert utility to convert to the formats that I support (system dependent but usually ulaw/alaw/gsm/g729) |
| 21:23.39 | p3nguin | Sure. Verbose() is my preferred way. I can call again if you want to check it for me. |
| 21:24.08 | WIMPy | jkroon: You don't need any SIP access to get owned via RTP! |
| 21:24.10 | Cadey | p3nguin : CallerID(num) the value you want? |
| 21:24.20 | p3nguin | cadey: yes |
| 21:24.23 | jkroon | WIMPy, do you have a ref I can read up from? |
| 21:24.36 | Cadey | call it again mate |
| 21:25.08 | Cadey | oh bugger sorry, missed the $ |
| 21:25.13 | p3nguin | :/ |
| 21:25.16 | paulc | Cadey: I'll give you a call too.. should come from 1604639xxxx |
| 21:25.20 | paulc | haha.. that pesky $! |
| 21:25.27 | Cadey | ok go again :( |
| 21:25.47 | Cadey | noop($CALLERID(num)) |
| 21:25.50 | Cadey | is that wrong ? |
| 21:26.01 | paulc | NoOp(${CALLERID(num)}) |
| 21:26.11 | Cadey | arggg |
| 21:26.13 | Cadey | sorry :) |
| 21:26.24 | p3nguin | Verbose(${CALLERID(num)}) |
| 21:26.28 | Cadey | ok and now |
| 21:26.33 | p3nguin | here goes again. |
| 21:26.43 | Cadey | 016046391550 |
| 21:26.49 | Cadey | 04019034562 |
| 21:27.19 | p3nguin | Is that the correct format if you wanted to call me in the US? |
| 21:27.34 | Cadey | 016 ? |
| 21:27.37 | Cadey | erm we would use 001 |
| 21:27.53 | WIMPy | jkroon: http://events.ccc.de/congress/2010/Fahrplan/events/4193.en.html |
| 21:27.55 | p3nguin | If you wanted to call a 10-digit US phone number, how would you dial it? |
| 21:27.55 | Cadey | in the US calling the UK you use 001144 right? |
| 21:28.02 | paulc | 011 44 xxxxx |
| 21:28.12 | p3nguin | I use 011 44 .... or 00 44 .... |
| 21:28.15 | Cadey | we would call 001 800 XXX YYY |
| 21:28.16 | paulc | is from the UK but living in Canada |
| 21:28.19 | paulc | well.. Canadian born.. |
| 21:28.31 | paulc | It irks me you can't do full national dialing here like you can in the UK |
| 21:28.47 | p3nguin | 011 and 00 are both valid international access codes for me. |
| 21:28.47 | paulc | that whole "1 means long distance" thing |
| 21:29.08 | Cadey | paulc i never understand why you have to put 011 before an international code on the american system |
| 21:29.17 | Cadey | here we jsut put a 0 then country code |
| 21:29.39 | paulc | Cadey: Yeah.. 0 for outside your local area, 00 for international.. like a lot of the rest of the world.. makes sense to me |
| 21:29.48 | p3nguin | cadey: So if you wanted to call me, you would dial 0 and then a 1, and then the 10-digit US number? |
| 21:29.56 | jkroon | WIMPy, thanks. |
| 21:30.05 | paulc | Cadey: Interestingly.. Germany usually uses 0 as "outside line" on PBXs, instead of the normal/common 9 |
| 21:30.20 | paulc | p3nguin: From the UK to US it's 00 1 NPA NXX XXXX |
| 21:30.21 | Cadey | 001 to call america i think mate |
| 21:30.31 | paulc | 00 = international, 1 = NANPA country code |
| 21:30.52 | p3nguin | cadey: I'm just trying to figure out what I need to change so that you get the correct caller ID number from me. I want you to be able to call me back using the caller ID number that shows up on your phone. |
| 21:31.09 | paulc | p3nguin: 001 + your national 10 digit number |
| 21:31.10 | Cadey | yeah i understand p3nguine :) |
| 21:31.20 | Cadey | 001 then your number |
| 21:31.27 | p3nguin | paulc: I currently send +1 and my 10-digit number. |
| 21:31.52 | p3nguin | What should I be sending? |
| 21:32.06 | p3nguin | In the US, I send just my 10 digits. |
| 21:32.15 | paulc | p3nguin: +1 should be enough, but it's how that gets translated along the way |
| 21:32.22 | paulc | in an ideal world, we'd all talk E.164 and be happy *sigh* |
| 21:32.51 | p3nguin | I'm just wondering if I should send 1 and 10-digits, or +1 and 10-digits, or 001 and 10-digits... |
| 21:32.53 | WIMPy | + isn't exactely correct anywhere but the only form with a defined meaning. |
| 21:32.57 | paulc | If you're sending "+1" and 10 digits, technically you're right and its down to the receiving end to mangle as needed |
| 21:33.45 | Cadey | im really new the telecoms industry tbh |
| 21:33.53 | paulc | the trouble with sending 001 + 10 digits is that it'll work in a lot of places, but not everywhere... (like countries that don't use 00 for international access for example) |
| 21:33.53 | *** part/#asterisk PeterEs (~PeterEs@78-27-57-250.dsl.alice.nl) |
| 21:33.57 | Cadey | im a programer by trade |
| 21:34.08 | paulc | Cadey: what kinda stuff? |
| 21:34.12 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:34.15 | Cadey | business systems |
| 21:34.38 | Cadey | .net mostly |
| 21:35.18 | paulc | gotcha.. I hack a bunch of stuff in PHP.. MySQL.. business systems mostly.. |
| 21:35.28 | paulc | and program telephone chat/datelines amongst other things during the day *smirks* |
| 21:35.30 | jkroon | WIMPy, yes, that is a legit concern. |
| 21:35.40 | Cadey | im currently making a proxy for asterisk AMI |
| 21:35.49 | jkroon | however, not nearly as easy to pull off as nailing the signalling protocols. |
| 21:36.10 | Cadey | and also building some client apps for the proxy for windows baised systems to control your clients calls |
| 21:36.12 | WIMPy | jkroon: You can gat the video of it being done, if you like. |
| 21:36.25 | jkroon | i think i'm going to do that. |
| 21:36.29 | WIMPy | But I think it's not directly linked there. |
| 21:36.39 | jkroon | stealing the voice data and listening in is about the most serious one there. |
| 21:36.53 | WIMPy | no |
| 21:37.03 | raden_work | I guess im SOL for music on hold |
| 21:37.10 | WIMPy | Transfering calls to premium rate numbers is the one. |
| 21:37.27 | jkroon | I highly doubt the DTMF injection (unless someone is being really stupid) will be really exploitable in the wild. |
| 21:37.29 | Cadey | paulc> : do you do much programing? |
| 21:37.42 | jkroon | probably possible on some systems though. |
| 21:37.51 | paulc | Cadey: A fair bit, depending on the day.. lately it's been more higher level design and stuff.. |
| 21:38.05 | WIMPy | jkroon: If you use Asterik and have features enabled, it works very well. |
| 21:38.18 | Cadey | paulc : Cool, I have a 3 year project comming to an end in April |
| 21:38.38 | jkroon | WIMPy, features disabled on my publicly accessible systems. |
| 21:38.39 | p3nguin | Okay, so should I leave my international caller ID number to +1 and 10 digits, or should I change it to a different format? |
| 21:38.46 | Cadey | its a rather speical CRM made in c# |
| 21:38.54 | Cadey | winForms |
| 21:40.07 | paulc | Cadey: Gotcha.. the CRM I did is old school PHP (back when register_globals was on by default, *shudder*). 10 years on, I still maintain and enhance it a bit here and there.. |
| 21:40.12 | jkroon | and even on all the systems I've installed I only allow transfers via SIP transfers, not via the ## feature codes. |
| 21:40.23 | paulc | bit of a pain, but good enough money.. especially being paid in GBP when I'm over here in CAD land |
| 21:40.31 | jkroon | except one client that specifically requested it be enabled because he's too cheap ass to buy proper sip phones and insists on using x-lite. |
| 21:41.10 | Cadey | paulc : lol yeah GBP rocks when your being paid in it :) |
| 21:41.20 | jkroon | goes to watch the video. |
| 21:41.42 | paulc | Cadey: That said, the exchange rate isn't as good as when I first came here.. around 1.5 CAD to the pound now, compared with 2.2ish when I first moved here |
| 21:41.47 | Cadey | I used to spend a lot of time in our Dallas office (texas) and the months i was there where like I was on tripple pay |
| 21:41.50 | paulc | ah well.. pays the bills.. keeps me in beer :) |
| 21:41.57 | Cadey | it was 1.8 then and everythign in the US is cheaper anyway :D |
| 21:42.03 | paulc | haha true that |
| 21:42.29 | Cadey | paulc : fule here is a record high :( |
| 21:42.55 | p3nguin | paulc: If I'm going to call you in area code 604, I need to send standard 10-digit caller ID to you because you are still in NANP with me, right? It's not an international call, even though you are in a different nation. |
| 21:42.57 | Cadey | its now £1.35 a leter which is about $8.1 a gallon @ 1.5 |
| 21:42.59 | jkroon | ok, so using strictrtp ... what problems does it cause (ie, why isn't it on by default?) |
| 21:43.42 | paulc | p3nguin: Yes, you and I are friends in terms of numbering plans, but I'm seen as inferior for a lot of products and services that see Canada as "those scum up north" hehe |
| 21:43.43 | Cadey | paulc : Also im a PHP hater :) I cant stand it lol |
| 21:43.56 | WIMPy | jkroon: The only possible problem I see is if the IP of the peer changes mid-converstaion. |
| 21:44.15 | Cadey | paulc : but then i cant stand perl or phython and its really really stupid which space syntax.... |
| 21:44.17 | paulc | Cadey: Don't be a hater - everything's got its place.. I just never got to grips with C#/.NET etc (plus kinda don't like "their way" of doing stuff, sometimes) |
| 21:44.26 | Cadey | bracing with tabs.... i mean come o |
| 21:44.55 | paulc | tabs should die? ;) |
| 21:44.55 | Cadey | paulc : heheh i was only diggin about the "hater" lol |
| 21:45.40 | Cadey | paulc : Ive used it for some things but I just prefer .net because its closer related to the ways C and Cpp work |
| 21:45.51 | Cadey | paulc : in "relative" terms anyway |
| 21:47.32 | p3nguin | I guess I'm going to leave my outgoing international CID as +1 and 10 digits until someone says something to the tune of, "Please fix your caller ID number so I can call you conveniently from my received calls list on my phone." |
| 21:48.23 | Cadey | i must be honest though, the asterisk chaps are the most forgiving techies ive ever spoken to. Most coder IRC rooms are full of idiots that rather pick apart your questions and attack you over spelling than answer any questions because they are too "stupid" |
| 21:48.58 | WIMPy | p3nguin: Leave it as it is. |
| 21:49.32 | p3nguin | cadey: I'll let you know if you're using the wrong words. than/then, it's/its, your/you're, etc. |
| 21:49.48 | Cadey | lol |
| 21:49.58 | p3nguin | But I'll still do my best to answer the questions once you learn how to ask them. :) |
| 21:49.59 | Cadey | gramer hero? |
| 21:51.00 | Cadey | im as dyslexic as they come. good with numbers, cant work with ilogical writen languages |
| 21:51.02 | p3nguin | Hero? Because I insist on using decent grammar? That's ridiculous. |
| 21:51.16 | jkroon | WIMPy, there seems to have been a discussion on the asterisk-dev list in 2007: http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg45769.html |
| 21:51.38 | jkroon | it gives some insight. and screws me completely, too much NAT'd devices in my case. |
| 21:51.53 | jkroon | and I do have to interact with at least to SBC type devices. |
| 21:52.08 | Cadey | in a world that says Live and Live sound 100% diffrent depending on context im glad im good with numbers and not words :) |
| 21:52.11 | *** join/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 21:54.30 | paulc | Cadey: hehe yeah, friendly bunch in here |
| 21:54.37 | jkroon | ok, i need stuff to sink in, bed time, night all. WIMPy thanks for the discussion. was interesting. |
| 21:54.46 | WIMPy | jkroon: That does not to me as if enabling it would make things worse. |
| 21:54.51 | Cadey | the worst place in the world is #c# on efnet |
| 21:54.53 | paulc | p3nguin: your plan is sound I think.. network edge devices should translate the + and do magic as needed |
| 21:55.13 | Cadey | there is a guy called halo3 and he does know his stuff, but hes a real fool |
| 21:55.22 | p3nguin | paulc: I hope so. That's the way I have had it for, well, forever. |
| 21:55.26 | Cadey | never leaves his PC, scared to leave the house type person |
| 21:55.42 | WIMPy | was pretty shocked after watching that talk and immediateyl switched it on. |
| 21:56.22 | jkroon | WIMPy, i'll have to go rtfs to see what exactly it does, make packet captures for a week and confirm that i won't kill anybodies voice. |
| 21:56.47 | jkroon | WIMPy, i couldn't locate the video actually. |
| 21:57.17 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:57.21 | paulc | Cadey: Familiar story. Some of the cleverest tech people often have a few social quirks. |
| 21:57.30 | paulc | Not me though - I like going out and having beers :-) |
| 21:57.37 | Cadey | lol yeah dito |
| 21:57.56 | Cadey | beers and stipers is always a good night haha |
| 21:58.01 | Cadey | *strippers |
| 21:58.04 | WIMPy | jkroon: Are you ok with a torrent? |
| 22:02.00 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 22:02.57 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 22:03.35 | *** join/#asterisk eugeneoden (~goden@southeast-invest.oneringnetworks.net) |
| 22:05.48 | jkroon | WIMPy, yea, should be. |
| 22:06.08 | WIMPy | Ok, comming |
| 22:06.19 | jkroon | permitting there are seeders I can connect to (NAT is a pain) |
| 22:07.20 | jkroon | bitsnoop, got it :) |
| 22:11.15 | *** join/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 22:22.19 | *** join/#asterisk eerie (~mime@gateway/shell/bshellz.net/x-msubdspnjecfxcie) |
| 22:28.22 | *** join/#asterisk eMBee (~eMBee@foresight/developer/pike/programmer) |
| 22:38.41 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 22:41.23 | p3nguin | That's weird. I'm behind a NAT and I run torrents without issue. |
| 22:41.39 | p3nguin | Why do so many people have so many problems? |
| 22:42.35 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 22:52.04 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-lcqomorvfhciqdbg) |
| 22:54.44 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-buwjlcmqnadnpynn) |
| 22:54.53 | Tozz_ | because lots of soho routers just suck |
| 23:01.30 | raden_work | blah 1.8.3 MOH dont work either |
| 23:02.31 | raden_work | can i go back to 1.6 from 1.8 easily ? |
| 23:04.11 | p3nguin | Actually, I'm behind a NAT and I run *everything* without issue. |
| 23:05.21 | chazzam | uses a linux box as a router/NAT |
| 23:06.09 | chazzam | my only issue is that the dhcp server likes to fail to give out DNS entries every other time it gives out a lease |
| 23:08.31 | *** join/#asterisk aut (~aut@c-174-48-60-12.hsd1.fl.comcast.net) |
| 23:10.03 | leifmadsen | likes dnsmasq for that kind of thing |
| 23:10.05 | leifmadsen | runs off |
| 23:11.02 | chazzam | yeah, my replacement box is using that instead |
| 23:16.51 | *** join/#asterisk gpled (~gpled@69.1.105.42) |
| 23:17.50 | JerJer | anyone know what the user=phone thing in an invite actually signifies / does? |
| 23:18.06 | JerJer | INVITE sip:+CC NSN@X.X.X.X;user=phone SIP/2.0 |
| 23:18.46 | gpled | how can i see incoming DID numbers? |
| 23:21.12 | p3nguin | gpled: See them in what sense? |
| 23:22.01 | p3nguin | gpled: When a call comes in from your ITSP, the extension where the call is sent is usually the DID number. Is that what you want? |
| 23:26.55 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 23:29.06 | gpled | p3nguin: would like to watch them real time |
| 23:31.46 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 23:39.18 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:41.55 | *** part/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 23:42.49 | p3nguin | gpled: What does that mean? You want to watch the call flow? |
| 23:47.49 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 23:48.18 | shmaltz | anyone know where I can get an example config file for spa5xx to be provisioned over http? |
| 23:54.40 | p3nguin | gpled: If that's what you mean... |
| 23:54.42 | p3nguin | gpled: core set verbose 4 |
| 23:54.44 | p3nguin | gpled: ... and have a day. |
| 23:56.59 | gpled | found /var/log/asterisk/full |
| 23:58.07 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
| 23:58.46 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:58.55 | *** part/#asterisk gpled (~gpled@69.1.105.42) |