| 00:18.14 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 00:24.02 | p3nguin | crcinau_: Are you using SIP, MGCP, or SCCP on the 7970? |
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| 01:49.48 | Dovid | morninng all |
| 01:50.25 | p3nguin | 172.16.197.243 |
| 01:51.53 | WIMPy | !N |
| 02:12.36 | p3nguin | Any known problems using VoIP with a Netgear FVS336G? |
| 02:13.11 | SunTsu | WIMPy: I just completed my dialplan, turned out to be much smaller than expected. Thanks again |
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| 03:22.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 04:57.06 | KNERD | Is there an issue with asterisk-addons-1.6.2.2? "./configure" functions as expected, but "make" fails |
| 04:57.54 | Kobaz | pastebin is your friend |
| 04:59.07 | KNERD | http://pastebin.com/HCDvNvsA |
| 05:04.06 | p3nguin | I wouldn't even bother looking since you aren't using the newest version available. |
| 05:04.54 | KNERD | p3nguin: let me look...I did wget 1.6-CURRENT, or something to that effect |
| 05:05.27 | p3nguin | I would have done "curl -O http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.2.3.tar.gz", but that's just me. |
| 05:09.28 | KNERD | p3nguin: thanks |
| 05:09.32 | KNERD | new pastebin http://pastebin.com/rv3tnGX3 |
| 05:10.13 | KNERD | 46. chan_ooh323.h:53:26: error: asterisk/rtp.h: No such file or directory |
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| 05:18.10 | p3nguin | Did you already install Asterisk from source? |
| 05:18.52 | KNERD | yes, and install went fine |
| 05:19.40 | KNERD | I used 1.8.3 if that make a difference |
| 05:19.56 | p3nguin | That makes a huge difference. |
| 05:20.31 | p3nguin | Two problems. 1) You've decided to try to install a 1.6 branch of asterisk-addons for asterisk branch 1.8. |
| 05:20.47 | p3nguin | 2) Asterisk 1.8 does not use asterisk-addons package from source. |
| 05:21.26 | p3nguin | Asterisk 1.8 branch builds addons with Asterisk; you have to enable it in menuselect. |
| 05:22.08 | KNERD | oh....I see thanks...back to recompiling |
| 05:22.26 | p3nguin | rm -r asterisk-addons-1.6* |
| 05:22.40 | p3nguin | Go back to Asterisk source dir and make menuselect again. |
| 05:23.42 | KNERD | doing now |
| 05:25.07 | KNERD | nice...everything is in one package |
| 05:28.45 | p3nguin | I take it you didn't have any trouble finding addons in menuselect. |
| 05:30.22 | KNERD | well, "addons" in that wording per-say was not there, but addons in previous versions were in there |
| 05:36.24 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 05:37.25 | KNERD | had to redo it yet again...I did not see the "contrib/scripts/get_mp3_source.sh" |
| 05:38.30 | p3nguin | per se |
| 05:39.17 | KNERD | ahh..thanks |
| 05:40.51 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 05:40.52 | p3nguin | So you didn't see the very top item on the menuselect menu? It says Add-ons (See README-addons.txt). |
| 05:41.12 | p3nguin | that's Asterisk 1.8 branch |
| 05:41.44 | KNERD | after the fact, yes |
| 05:44.40 | p3nguin | (2330.22) <KNERD> well, "addons" in that wording per-say was not there, but addons in previous versions were in there |
| 05:45.03 | p3nguin | I have absolutely no idea what you meant if it wasn't that you didn't see the addons on the menu. |
| 05:46.47 | *** join/#asterisk psilikon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
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| 05:47.52 | KNERD | my bad...I was distracted by something else around me...yes I selected that |
| 05:48.17 | psilikon | p3nguin: I fixed the no ringing/dahdi issue I was in here with last night. Simply added 'callprogress=no' to chan_dahdi.conf. |
| 05:48.31 | psilikon | hopefully that won't screw up cdr |
| 05:52.53 | p3nguin | I don't use any dahdi channels, so I'm not familiar with that parameter. What does it do? |
| 05:54.07 | p3nguin | My thought is that setting it to no would be backward for what you wanted. |
| 05:56.59 | psilikon | p3nguin: I know, I thought that too. I also think that setting it to 'no' would be the default and effectively the same as not specifying it at all, but it worked. |
| 06:18.03 | *** join/#asterisk _omer (~omer@182.178.159.108) |
| 06:18.06 | _omer | hello |
| 06:19.02 | _omer | chan_ooh323.so is listed in "show modules" .... but I cant see the commands in "help" ..... any help please? |
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| 08:14.15 | benngard | when u pause a queue member in the dialplan, is there some way u can auto unpause the member after say 120 seconds? |
| 08:17.20 | benngard | core show help ooh323 <- works for me |
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| 12:50.23 | z4nD4R | Q: somebody help with teardown sessions attack? ( studying purpose )... i have very interessant result ... somebody help to understand me? |
| 13:00.21 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 13:01.06 | z4nD4R | Q: somebody help with teardown sessions attack? ( studying purpose )... i have very interessant result ... somebody help to understand me? |
| 13:01.08 | z4nD4R | nobody? |
| 13:28.06 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 13:28.44 | EmleyMoor | Is the current context available as a variable I could pass as an arg in a Gosub? |
| 13:30.21 | SunTsu | EmleyMoor: ${CONTEXT}? |
| 13:30.39 | SunTsu | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 13:31.21 | EmleyMoor | Ah, yes, thanks |
| 13:31.48 | EmleyMoor | is doing some modernisation work on his dialplan - hoping to see the end of macros |
| 13:32.57 | z4nD4R | Q: somebody help with teardown sessions attack? ( studying purpose )... i have very interessant result ... somebody help to understand me? |
| 13:34.02 | SunTsu | z4nD4R: repeating something every few minutes will not help you in getting attention, rather with getting on ignore lists |
| 13:34.53 | z4nD4R | hmm....ok... |
| 13:39.11 | EmleyMoor | Is it advisable, or even necessary, to "pop" before Goto-ing out of a Gosub? |
| 13:45.48 | SunTsu | EmleyMoor: http://www.voip-info.org/wiki/view/Asterisk+cmd+Gosub Notes section says "yes"# |
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| 14:05.19 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 14:06.04 | Romeo- | hello, wich packages i do neef on a debian squeezy to run asterisk server? |
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| 14:34.49 | tzafrir_laptop | Romeo-, asterisk |
| 14:35.23 | Romeo- | ok |
| 14:35.33 | Romeo- | i install them anyway |
| 14:35.59 | Romeo- | i start the service and i go to the http://ip:8088/asterisk/ and got theis |
| 14:36.10 | Romeo- | Not Implemented |
| 14:36.51 | *** join/#asterisk puzzled (~patrick@535335AA.cm-6-4a.dynamic.ziggo.nl) |
| 14:39.08 | SunTsu | EmleyMoor: thanks for reminding me of GoSub, it just made my dialplan much shorter and much easier to maintain |
| 14:39.35 | iprouteth0 | Romeo: was the asterisk GUI included in packages?? I've always have to install and build from svn |
| 14:40.53 | iprouteth0 | Romeo-: several modifications are needed from base configs as well for this, however the GUI can check the configs after building |
| 14:41.28 | Romeo- | ok |
| 14:41.56 | iprouteth0 | http://www.asterisk.org/asterisknow/install-related |
| 14:42.30 | Romeo- | tky, but i's a small box, i will try to apt-get them;) |
| 14:43.00 | iprouteth0 | it's always easier |
| 14:43.20 | iprouteth0 | I use gentoo, and the GUI is not in our repo |
| 14:43.26 | Romeo- | asterisk-gui_2 bla is installed, get the same error |
| 14:43.36 | Romeo- | asterisk-gui 2.0.4.9.svn.4991-1 |
| 14:43.53 | iprouteth0 | you may want to confirm configs are good |
| 14:43.58 | iprouteth0 | in http.conf and manager.conf |
| 14:44.08 | Romeo- | i have a alixboard here with voyage linux (debian squezze) |
| 14:44.14 | Romeo- | i check that |
| 14:45.15 | Romeo- | ls /etc/asterisk/manager.d/ |
| 14:45.16 | Romeo- | README.conf admin.con |
| 14:46.37 | Romeo- | same |
| 14:47.46 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:50.53 | Romeo- | on [general] enable = yes or enabled=yes |
| 14:55.41 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:00.51 | tzafrir_laptop | Romeo-, you don't have the httpd running. |
| 15:01.00 | tzafrir_laptop | But do you actually want to use it? |
| 15:01.04 | Romeo- | lol |
| 15:01.06 | Romeo- | ok |
| 15:01.26 | Romeo- | <PROTECTED> |
| 15:01.26 | Romeo- | 1 |
| 15:02.24 | tzafrir_laptop | Romeo-, actually I run a plain Debian on such an Alix box |
| 15:02.36 | tzafrir_laptop | it's my home PBX |
| 15:02.50 | Romeo- | nice |
| 15:02.57 | Romeo- | wich debian, |
| 15:03.20 | Romeo- | i mean i don't have any vga port and stuff here:) |
| 15:05.10 | tzafrir_laptop | Romeo-, recently upgraded to Squeeze |
| 15:05.24 | Romeo- | hm get some ACL issues |
| 15:05.28 | tzafrir_laptop | I use it through ssh and/or a serial console |
| 15:06.00 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 15:06.08 | Romeo- | ssh here |
| 15:07.17 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 15:12.04 | Romeo- | i can'tget in with user and pass tzafrir_laptop :)) |
| 15:12.29 | tzafrir_laptop | Romeo-, into what? |
| 15:13.01 | Romeo- | asterisk_gui |
| 15:13.04 | Romeo- | Please enable manager access. |
| 15:13.04 | Romeo- | Most often you should be able to do that by setting 'enable = yes' in manager.conf under the [general] context, and reloading asterisk. |
| 15:13.17 | Romeo- | enable or enabled :) |
| 15:16.21 | tzafrir_laptop | Romeo-, did you enable webaccess in manager.conf ? |
| 15:16.27 | Romeo- | yes |
| 15:16.38 | Romeo- | what is this /etc/asterisk/extconfig.conf |
| 15:17.00 | tzafrir_laptop | There should be a script to run the trivial tests in the package. Look under /usr/share/doc/asterisk-gui/examaples , IIRC |
| 15:17.19 | Romeo- | tky |
| 15:25.45 | Romeo- | tzafrir_laptop, issuess :) RewriteEngine on |
| 15:26.18 | benngard | any1 thoght about adding a "timeout" paraeter to PauseQueueMember, like PauseQueueMember(,SIP/12345,,,300) after 5 minutes the member will be unpaused |
| 15:27.23 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:37.56 | Romeo- | tzafrir_laptop, there no user loaded :) |
| 15:38.03 | Romeo- | <PROTECTED> |
| 15:38.03 | Romeo- | There are no manager users |
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| 16:11.39 | benngard | Romeo-: u have to add the user(s) in manager.conf |
| 16:13.48 | Romeo- | yes, solved, thank you |
| 16:14.10 | Romeo- | i delete the file and write new one, then works |
| 16:23.26 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 16:23.47 | Dovid | what is the correct way to wget a patch file. like: https://issues.asterisk.org/file_download.php?file_id=28850&type=bug ? |
| 16:27.47 | Dovid | does this look OK ? http://pastebin.com/mTW9rjsH |
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| 16:34.27 | *** join/#asterisk matt2s (~mattis@hurfmydurf.com) |
| 16:36.54 | matt2s | I am trying to teach myself the basics of SIP, and thought I'd experiment a bit with Asterisk. I was wondering if I could use my Android phone as a "landline"? I have no idea what the correct terminology is |
| 16:37.29 | EmleyMoor | The U(x) option to dial - can someone please let me see an example of it? |
| 16:38.47 | EmleyMoor | matt2s: If you have a SIP client on your phone, and an Asterisk box with incoming numbers (DAHDI, SIP, IAX or indeed any other suitable technology), the Android phone can be connected and you can make and receive calls on it via Asterisk |
| 16:40.55 | EmleyMoor | (can/should the x be a context,exten,priority triplet?) |
| 16:41.29 | kaldemar | EmleyMoor: it can't. it is a context. |
| 16:41.32 | matt2s | EmleyMoor, I was hoping I could use a softphone (?) that through Asterisk used my Android phone as the connection to the PSTN ? (Sorry if I mess up terminology, but I have just ten minutes ago got a VM with Ubuntu running and starting to install Asterisk) |
| 16:41.57 | kaldemar | EmleyMoor: it will go to x,s,1. |
| 16:42.05 | EmleyMoor | kaldemar: Ah, OK |
| 16:42.18 | EmleyMoor | (exactly the other thing I would expect) |
| 16:42.37 | *** join/#asterisk Poincare (~jefffnode@2001:5c0:150f:1704::2) |
| 16:43.00 | kaldemar | EmleyMoor: it's a bit misleading imo. |
| 16:43.04 | EmleyMoor | My modernisation is nearly complete |
| 16:43.58 | EmleyMoor | kaldemar: Presumably there should not be a Return() reached in that Gosub? |
| 16:46.02 | kaldemar | EmleyMoor: i'd assume no since it seems to work exactly like a macro, but test to be sure. |
| 16:46.59 | EmleyMoor | kaldemar: That should be easy to test - it will trigger when a call is received from outside |
| 16:49.14 | EmleyMoor | Hmmm... what's the equivalent of MacroExit in a Gosub? |
| 16:52.43 | matt2s | EmleyMoor, thing is, I'm going on a job interview tomorrow that is for a trainee position for a telephony company. I know networking (CCNA), I know Linux, and I have worked some years as a programmer. They want people they can train on the job, but I want an edge and try out some of the technology beforehand, so I thought it would be fun if I could set up Asterisk so that I can phone from a windows machine, through the VM running asterisk, through my And |
| 16:52.43 | matt2s | roid phone and connect to another cellular phone we have in our house. |
| 16:54.32 | matt2s | But I can't find any resources how to use my Android phone as the link between Asterisk and the public telephone system. Was just wondering if it's possible |
| 16:54.59 | SunTsu | matt2s: you'd need a call forwarding ob your android phone |
| 16:55.15 | SunTsu | "on" even |
| 16:55.26 | SunTsu | matt2s: I only know sip clients for android |
| 16:55.42 | Romeo- | sip client for phone |
| 16:55.47 | Romeo- | will do that |
| 16:56.41 | matt2s | Okay, so I install a SIP client on my Android, and somehow register it to Asterisk, then I can use a softphone on any computer to call the outside world? |
| 16:57.00 | Romeo- | yeah, somehow :) |
| 16:57.07 | SunTsu | Romeo-: I don't think that can be used as a gateway to public phone system |
| 16:57.08 | Romeo- | afk |
| 16:57.51 | Romeo- | SunTsu, with asterisk and sip trunk, works here |
| 16:58.33 | SunTsu | Romeo-: you use a android sip client as an uplink to public telephone system? How? |
| 16:58.59 | Romeo- | i use iphone |
| 16:59.03 | Romeo- | sip app |
| 16:59.18 | Romeo- | connectet to a asterisk server with trunk |
| 17:00.09 | SunTsu | Romeo-: yeah, you use asterisk as a gateway to the phone system. matt2s wants softphone -> Asterisk -> android sip -> public phone system |
| 17:00.30 | SunTsu | at least that's what I understood |
| 17:00.36 | matt2s | SunTsu, that's correct |
| 17:01.10 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.137.11.dsl.dyn.forthnet.gr) |
| 17:01.38 | Romeo- | ok, my foult |
| 17:02.43 | SunTsu | matt2s: I don't think that there's android apps that do that. But you can get a simple sip account from lots of sip providers and do it that way |
| 17:03.02 | SunTsu | matt2s: most are free and only charge for calls |
| 17:03.45 | matt2s | SunTsu, okay, it's just for exploring SIP/Asterisk anyway. I was thinking plan B is to install a SIP client on my Android, use Asterisk to connect to Skype and call a Skype windows machine |
| 17:03.58 | matt2s | SunTsu, maybe that is more doable? |
| 17:04.48 | SunTsu | matt2s: it is, but AFAIK the skype addon for asterisk is not free |
| 17:05.59 | matt2s | SunTsu, ah, okay. Then I can try your suggestion of getting a SIP account. Do you have any recommendations? |
| 17:06.53 | SunTsu | matt2s: I personally use sipgate, but there are lots more, asterisk wiki has a list, I think |
| 17:07.45 | matt2s | SunTsu, okay, thanks a lot for your help! |
| 17:08.18 | *** join/#asterisk JParr (~JParr@24.244.133.126) |
| 17:10.03 | JParr | i just installed asterisk and dahdi from repositories on ubuntu 10.04, and dahdi_genconf/dahdi_cfg don't report any spans with a t1 card |
| 17:10.18 | JParr | it is a te210p |
| 17:10.28 | JParr | does this require manual configuration, or should the tool see the card? |
| 17:21.59 | EmleyMoor | One thing I recently found out about: same => |
| 17:23.17 | *** join/#asterisk jan_bangna (~jandetlef@ppp-124-122-77-167.revip2.asianet.co.th) |
| 17:23.30 | WIMPy | JParr: You need to modprobe wct4xxp first. |
| 17:24.32 | jan_bangna | hello. Can asterisk to auto dialing? like i would send an array of numbers to the system via api and it would dial one after another as soon as the previous hangs up. possible? |
| 17:25.36 | JParr | WIMPy: ah, that looks better, i see the module detecting the card in dmesg, and dahdi_cfg -v shows channels |
| 17:25.37 | JParr | thanks |
| 17:33.53 | EmleyMoor | q \'digits/at\' IMp - I used to use that time format as part of my "call return" function. It now claims it can't find digits/at\ - the \ seems to have become significant |
| 17:34.56 | EmleyMoor | Ah, I see |
| 17:36.39 | *** join/#asterisk philippel_mac (~p_lindhei@50.46.123.25) |
| 17:37.53 | EmleyMoor | Hmmm... or do I? |
| 17:39.43 | p3nguin | emleymoor: What have you changed recently? |
| 17:43.24 | JParr | is there a preferred echocan? |
| 17:43.36 | EmleyMoor | p3nguin: Found it. In 1.6 SayUnixTime, you must *not* use backslashes before the single quotes |
| 17:44.06 | p3nguin | emleymoor: Interesting. I certainly have no trouble with them in 1.4. |
| 17:44.12 | EmleyMoor | Only just discovered it to be the case - been on 1.6 about 4 months |
| 17:44.25 | EmleyMoor | p3nguin: Yes, in 1.4 you *must* use them |
| 17:44.42 | p3nguin | SayUnixTime(,,IMp \'silence/1\' ABdY) <-- This is what I use. |
| 17:45.06 | EmleyMoor | Needs to be SayUnixTime(,,IMp 'silence/1' ABdY) in 1.6 |
| 17:45.32 | p3nguin | I'll never use 1.6.anything, so we're good there. |
| 17:46.03 | EmleyMoor | I've been modernising my dialplan today - looks as though it's all successful |
| 17:46.20 | p3nguin | Now you just need to upgrade to 1.8.3. |
| 17:46.43 | EmleyMoor | p3nguin: Not a chance - unless it ends up being in wheezy |
| 17:47.03 | p3nguin | Oh, you don't compile your own packages from source? |
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| 17:47.30 | EmleyMoor | p3nguin: Not at present, no - apart from a recompile to add a patch |
| 17:47.55 | SunTsu | p3nguin: I'm currently pondering if I should update - just for calendar.conf |
| 17:47.58 | p3nguin | What tools do you use to repackage after you compile in a patch? |
| 17:48.25 | p3nguin | I use checkinstall because it's the only thing I know of that does its job so easily. |
| 17:48.28 | EmleyMoor | p3nguin: The Debian standard ones |
| 17:49.05 | EmleyMoor | I used to need two patches, but the one I used in zaptel made it into dahdi |
| 17:49.20 | p3nguin | I'm not a Debian user, so I just use checkinstall. No idea what other packaging tools there are. |
| 17:50.04 | p3nguin | And since it creates debs as well as rpms, I can use it on RH-based systems as well. |
| 17:51.03 | EmleyMoor | Not 100% sure my UK caller ID fix is working completely - but few people persist in calling my DAHDI line |
| 17:51.41 | p3nguin | The surefire way to keep people from calling that line would be to cancel it (or at least disconnect it from asterisk). |
| 17:52.06 | EmleyMoor | p3nguin: But then I lose my free weekend and evening calls |
| 17:52.14 | p3nguin | oh |
| 17:52.49 | p3nguin | Then I would come up with an appropriate solution based on the actual needs. |
| 17:53.10 | EmleyMoor | The IVR advises that it's deprecated |
| 17:53.31 | p3nguin | :) |
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| 17:53.46 | EmleyMoor | Besides, it's ex-directory now, and my IAX2 number is in-directory |
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| 19:43.59 | snif | hi all |
| 19:44.41 | snif | what Queue Weight means ?? |
| 19:44.53 | snif | for agent |
| 19:44.58 | snif | on asterisk platform |
| 19:45.05 | snif | i have 20 on weight |
| 19:45.13 | snif | what is the meaning of this 20 ? |
| 19:47.55 | jamko | snif: It allows you to set higher priority queues, so those calls will get handled over queues with lower priorities. |
| 19:50.12 | p3nguin | Is the weight on the queue or the queue member? |
| 19:50.29 | snif | the queue member |
| 19:50.39 | snif | <PROTECTED> |
| 19:52.04 | p3nguin | The penalty on members defines which members get called first. |
| 19:52.21 | snif | it helps on what exactly |
| 19:52.28 | snif | what is the adventage of the weight |
| 19:52.49 | snif | it should be hight or low ? |
| 19:53.04 | p3nguin | If two members have a penalty of 1 and one member has a penalty of 2, both of the members with penalty 1 would have to be busy before the one with a 2 will ever be called. |
| 19:53.45 | p3nguin | If you have another member with a penalty of 3, all three of the members with 1s and 2 will have to be busy before the one with a 3 will be called. |
| 19:54.11 | snif | oh i see |
| 19:54.16 | snif | thanks |
| 19:54.27 | p3nguin | It's just a way of making sure calls go to certain queue members first. |
| 19:54.46 | snif | k cool |
| 19:55.21 | p3nguin | A senior sales agent might have a priority of 5, where everyone else has a lower number. That way the senior sales agent only takes calls when all others are busy. |
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| 19:56.18 | snif | oh i see |
| 19:56.34 | p3nguin | Any other questions about it, or do you think you understand it good enough now? |
| 19:56.53 | snif | i can hold only 15 or 16 calls using any extention can i enlarge that number of holding ? |
| 19:57.06 | snif | i understand it enoght thanks alot |
| 19:57.20 | p3nguin | What does the extension do? |
| 19:57.25 | p3nguin | dial to a phone? |
| 19:57.37 | snif | yes |
| 19:57.40 | snif | to a phone |
| 19:58.04 | p3nguin | And the phone can only put on hold 16 calls? Which version of Asterisk are you using? |
| 20:00.05 | p3nguin | You really shouldn't be answering 16 calls and putting them all on hold anyway. That's what queue is for. If you want to answer all the calls, transfer them into queue instead of putting on hold; or put the calls into the queue first instead of answering. |
| 20:00.44 | snif | even if i transfer to other numbers |
| 20:00.55 | snif | i mean if i transfer a number to another extention |
| 20:00.59 | snif | it can only answer 16 max |
| 20:01.14 | p3nguin | Which Asterisk version do you have? |
| 20:01.56 | snif | 2.6 |
| 20:02.05 | p3nguin | There is no Asterisk 2.6. Try again. |
| 20:02.19 | benngard | he is from the future ;) |
| 20:02.23 | p3nguin | Must be. |
| 20:02.40 | snif | sorry 2.6 for the linux version |
| 20:02.40 | snif | lol |
| 20:02.41 | snif | Asterisk 1.8.3 |
| 20:03.17 | snif | its 1.8.3 |
| 20:03.47 | p3nguin | Just a second, please, I have food cooking in the kitchen that I have to run to attend. |
| 20:04.18 | snif | oh ok |
| 20:04.19 | snif | tyt |
| 20:05.04 | p3nguin | I'm back. The pizza was almost burning. |
| 20:05.49 | snif | :D |
| 20:05.52 | snif | sorry |
| 20:05.59 | snif | so any idea abou that ? |
| 20:06.02 | snif | is it configurable ? |
| 20:06.07 | p3nguin | Try changing the call-limit setting for your phones in sip.conf. |
| 20:06.52 | snif | where can i find it on the plateform? |
| 20:07.20 | p3nguin | I guess 1.8 branch recommends using groupcount instead of call-limit, but call-limit probably still works. |
| 20:08.18 | snif | where to find it |
| 20:08.19 | p3nguin | The setting goes in sip.conf. |
| 20:08.29 | snif | i cant access the files |
| 20:08.30 | snif | :s |
| 20:08.47 | p3nguin | You are the administrator? |
| 20:08.59 | snif | yes |
| 20:09.03 | snif | working only on the platform |
| 20:09.07 | snif | from the browzer |
| 20:09.07 | p3nguin | Why can't you access the file? |
| 20:09.13 | snif | i lost my ssh pass |
| 20:09.14 | snif | :s |
| 20:09.20 | p3nguin | That's going to be a problem. |
| 20:09.25 | p3nguin | Where is the server computer? |
| 20:09.47 | snif | from another city |
| 20:09.56 | snif | in* |
| 20:10.21 | snif | i cant access the server physicaly |
| 20:10.29 | p3nguin | Can you have the facility manager send someone to the server to reset your password for you? |
| 20:11.35 | snif | no i dont think so |
| 20:14.03 | snif | :( |
| 20:14.08 | snif | so cant be edited from the panel? |
| 20:14.47 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 20:15.30 | p3nguin | What panel are you talking about? |
| 20:17.43 | snif | i mean the platform |
| 20:17.51 | snif | using it from the browser |
| 20:18.14 | WIMPy | If you tell us what platform, we can tell you in which channel to ask. |
| 20:18.33 | snif | asterisk |
| 20:18.38 | snif | :s |
| 20:18.45 | WIMPy | Or you start to do some secutiry research. |
| 20:19.11 | snif | yea |
| 20:19.17 | WIMPy | Asterisk does not have a web interface. |
| 20:19.31 | snif | why mine have! |
| 20:19.40 | snif | pbx |
| 20:19.57 | WIMPy | Because you installed one onte web configuration things. |
| 20:20.12 | WIMPy | But there are several of them. |
| 20:21.45 | snif | let me show u |
| 20:23.38 | WIMPy | If you don't know what you installed, I'd expect it to tell you somewhere. |
| 20:24.06 | snif | http://smartdns.co.za/ |
| 20:24.12 | snif | thats it |
| 20:24.50 | p3nguin | I think we saw this recently. No one knows the name. |
| 20:25.45 | WIMPy | It doesn't seem to know itself. |
| 20:25.56 | WIMPy | Bad situation. |
| 20:27.13 | WIMPy | Completely unhelpful. |
| 20:27.55 | snif | :s |
| 20:59.19 | *** join/#asterisk groogs (482631f9@gateway/web/freenode/ip.72.38.49.249) |
| 21:07.00 | groogs | I just switched two channels from my provider from SIP to IAX2 to try and avoid some NAT issues. Now I am getting a weird problem I've never seen before: the call works well, but randomly, the inbound audio cuts off for ~20 seconds, then comes back.. and 30-60 later does it again (just guessing at times, it may be more consistent than that). the really weird part is while the inbound audio is not working, inbound DTMF does work (and |
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| 21:08.19 | groogs | any clues on what causes that? and yes, 4569/udp is forwarded. happens on both call termination and origination |
| 21:11.03 | IsUp | groogs: its sounds like RTP issue. but IAX2 using 4569 for RTP too as far i know |
| 21:12.28 | groogs | IsUp: yes, that's what I thought too. I haven't really used IAX2 much, but from what i know it's supposed to be much more NAT-friendly |
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| 21:15.01 | IsUp | groogs: enable debugging and place a call. |
| 21:16.59 | groogs | just looking at the IAX page: http://www.voip-info.org/wiki/view/IAX .. sounds like my problem may be NAT timeout |
| 21:18.22 | IsUp | groogs: yeah just enable debugging with 'iax2 set debug on' |
| 21:18.59 | WIMPy | IAX is not using RTP. |
| 21:19.55 | IsUp | then sorry for confusion |
| 21:23.13 | groogs | hm ,yeah ... pretty sure it's the NAT timeout |
| 21:23.22 | groogs | i changed it to 90 seconds, and couldn't get it to drop the call |
| 21:23.30 | groogs | changed it to 20, and now it's dropping again |
| 21:23.54 | groogs | and i think it's re-registering after <60 seconds, which is the asterisk IAX2 registration setting |
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| 21:24.24 | SunTsu | Damn, outgoing calls work for sipgate but not for qsc. Incoming calls work with both, but with qsc I get "Failed to authenticate on INVITE" |
| 21:26.01 | SunTsu | I already have insecure=invite and type=friend, but that does not seem to be all |
| 21:26.25 | WIMPy | That only helps for incomming. |
| 21:27.02 | WIMPy | Maybe you need to set 'defaultuser'? |
| 21:27.12 | SunTsu | I did. I set fromuser, too |
| 21:27.29 | IsUp | SunTsu: and if your provider requires registration, check with 'sip show registry' |
| 21:27.35 | WIMPy | Hmm. that sounds special then. |
| 21:27.42 | SunTsu | IsUp: it's registered alright |
| 21:30.07 | groogs | seems it didn't fix the problem. though iax2 debug shows me a flood of messages like this; Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: VNAK whenever the audio stops |
| 21:32.16 | SunTsu | I've read that some Huawei Softswitches are having Problems with a=silenceSupp:off - I know that qsc uses huawei, do you think it worth a try patching this header out? |
| 21:33.56 | WIMPy | UI is also using so huawei stuff and that seems to work with Asterisk. |
| 21:34.11 | WIMPy | But they have it combined with OpenSER, I think. |
| 21:35.41 | SunTsu | QSC has ser, shouldn't be too much of a difference, right? |
| 21:35.59 | WIMPy | Maybe it was without "Open". |
| 21:36.26 | SunTsu | I'll try patching, if it doesn't work I'll revert |
| 21:37.37 | WIMPy | Maybe it expects a certan caller ID? |
| 21:37.55 | p3nguin | Is it possible to get calls directly into Asterisk from Google Voice on 1.4 branch? I don't care about outbound calling, just inbound direct from Google Voice. Using a forwarding number (such as through sipgate or ipkall) to get GV calls adds unwanted latency. |
| 21:39.06 | SunTsu | I tried setting callerid in sip.conf and CALLERID(num) in my dialplan, didn't change anything |
| 21:43.13 | SunTsu | OK, it's not silenceSupp |
| 21:44.19 | WIMPy | Have you tried calling out using something else? |
| 21:45.40 | SunTsu | WIMPy: when I use that account on my snom phone it works flawlessly |
| 21:46.07 | WIMPy | Ok, that should be something to work on. |
| 21:46.21 | WIMPy | Have you also tried comparing traffic? |
| 21:46.55 | SunTsu | no, not yet. I'll try setting CALLERID(all) first |
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| 22:01.49 | SunTsu | OMFG! They expect Callerid userid <userid> - who in his right mind? |
| 22:03.16 | WIMPy | But the Huawei equipment is the best available on the market! |
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| 22:13.03 | high-rez | Any of you know the accuracy of the clock on digium t1/e1 cards? |
| 22:13.39 | WIMPy | You have no network connection? |
| 22:13.57 | high-rez | WIMPy: Correct |
| 22:14.06 | high-rez | I'm attempting to provide clock, but I need to be accurate within 50hz |
| 22:19.05 | Kobaz | hmm |
| 22:19.24 | Kobaz | so the new dahdi 1.4.x doesn't need a dahdi_dummy anymore, right? |
| 22:19.58 | Kobaz | er i mean, the 2.4 |
| 22:20.19 | WIMPy | It's included in the main module, yes. |
| 22:20.46 | Kobaz | when i do a dahdi show status it doesn't show the dummy channel |
| 22:20.52 | Kobaz | or is that expected |
| 22:21.18 | Kobaz | on dahdi older than 2.4, i get |
| 22:21.20 | Kobaz | DAHDI_DUMMY/1 (source: HRtimer) 1 UNCONFI 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 22:22.05 | WIMPy | Hmm. I think it should still show up. |
| 22:22.13 | Kobaz | yeah, it probably should |
| 22:23.05 | WIMPy | has switched to timerfd. |
| 22:23.24 | Kobaz | yeah |
| 22:23.32 | Kobaz | i want to, but i'm on 1.6.0.x still |
| 22:28.37 | p3nguin | Is it possible to get calls directly into Asterisk from Google Voice on 1.4 branch? I don't care about outbound calling, just inbound direct from Google Voice. Using a forwarding number (such as through sipgate or ipkall) to get GV calls adds unwanted latency. |
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