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| 00:37.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:41.46 | *** join/#asterisk doneir (var@gateway/shell/xzibition.com/x-ltlnngnigofgvpxy) |
| 00:42.32 | doneir | hey all, I came in yesterday talking about a silence detection truncation problem. I've created a patch, tested and applied it to the associated bug ticket: https://issues.asterisk.org/view.php?id=18937 |
| 00:43.25 | doneir | i have also filled out and submitted the license |
| 00:46.05 | doneir | testing sample dialplan code is also attached for testing purposes |
| 00:59.35 | p3nguin | oh |
| 01:06.17 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 01:08.09 | KingDavidNYC | can anybody please help make skip-docbook2man in the kamailio installation pleeeeaaaaseeee |
| 01:08.54 | WIMPy | Not really. Have you tried making it a NOP? |
| 01:09.33 | KingDavidNYC | WimPy: sorry, you talking to me? |
| 01:09.45 | WIMPy | yes |
| 01:09.59 | KingDavidNYC | WIMPy: thanks, what is a NOP? |
| 01:10.24 | WIMPy | No Operation |
| 01:10.42 | KingDavidNYC | WIMPy: sorry, what is that? |
| 01:10.55 | WIMPy | a do nothing |
| 01:11.39 | KingDavidNYC | the error is on the Makefile, I dont know how to make it skip the part where it wants to run something related to docbook2man |
| 01:12.23 | KingDavidNYC | I commented the lines, but it still gives me an error |
| 01:12.41 | KingDavidNYC | I really hate docbook2man, don't know what it is, and can not care less |
| 01:13.48 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 01:13.58 | KingDavidNYC | ...and kamailio does a very poor work on its documentation not referring to it |
| 01:14.09 | CRCinAU_ | ok |
| 01:14.19 | CRCinAU_ | so this Cisco 7970 is annoying as hell. |
| 01:14.39 | KingDavidNYC | kamailio is annoying as hell |
| 01:16.20 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 01:18.53 | p3nguin | NOP? You mean NoOp() probably. |
| 01:19.48 | WIMPy | more like true |
| 01:19.57 | WIMPy | /bin/true that is |
| 01:21.27 | KingDavidNYC | p3nguin: I am lost...Noop() is in the dialplan |
| 01:27.12 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 01:27.43 | CRCinAU_ | p3nguin: I think I'm going to give chan_sccp-b a go ;) |
| 01:28.14 | CRCinAU_ | p3nguin: I've tried just about every version of SIP firmware I can find for the 7970, and just no luck at all. |
| 01:35.13 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 01:38.46 | CRCinAU_ | p3nguin: ping? |
| 01:41.41 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
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| 01:46.36 | carrar | I've never had a issue with cisco phones |
| 01:46.44 | carrar | you have to make sure everything is correct |
| 01:46.49 | carrar | otherwise stuff fails |
| 01:47.07 | carrar | but thats true for anything |
| 01:48.10 | *** part/#asterisk VoipForces (~Adium@67.55.25.217) |
| 01:48.22 | CRCinAU_ | this is the 7970 - it registers a line now, and gets an OK from asterisk, then hangs. |
| 01:48.37 | CRCinAU_ | now its telling me I have voicemail, but still saying line 1 is dead |
| 01:48.55 | CRCinAU_ | won't dial, don't update the display |
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| 02:11.13 | hardwire | any reason why 1.6.2.6 and 'sip reload' doesn't.. reload correctly? |
| 02:11.50 | hardwire | I removed a host.. added it back after renaming it.. and even though it doesn't appear in sip show peers it still thinks it exists and maps that IP to that peer for inbound insecure=port,invite calls |
| 02:11.56 | hardwire | like.. it doesn't exist. |
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| 02:14.16 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 02:47.23 | p3nguin | crcinau_: yep |
| 02:50.18 | CRCinAU_ | p3nguin: do you happen to have an sccp.conf and a SEP for the 7970 you can send me? |
| 02:50.37 | CRCinAU_ | I'm struggling a bit trying to understand this... |
| 02:50.51 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 02:52.58 | p3nguin | crcinau_: chan_sccp-b provides three sample files. It should be easy to figure out once you have it installed. I'll paste an example device for you, and... how many lines do you want to configure on it? |
| 02:53.21 | CRCinAU_ | well, I'm trying to figure that out... |
| 02:53.41 | CRCinAU_ | in sccp terms, what is a line? |
| 02:53.49 | CRCinAU_ | ie, each phone has its own extension |
| 02:54.03 | CRCinAU_ | but I also have 3 x SIP trunks to different vsps |
| 02:54.24 | CRCinAU_ | so I guess each device has 4 'lines'? |
| 02:54.35 | p3nguin | You don't have to have any extensions just because you have lines. |
| 02:54.38 | CRCinAU_ | not sure if the 3 x vsp ones are shared with sccp vs only one? |
| 02:54.52 | CRCinAU_ | thats where the sharing comes in? |
| 02:55.19 | p3nguin | No. It just doesn't have any relationship. |
| 02:55.45 | CRCinAU_ | see, historically, I've had my vsps mapped to buttons on the 7961 |
| 02:56.05 | CRCinAU_ | meaning I have a line button for the local extensions, then one for each vsp |
| 02:56.43 | p3nguin | That doesn't make any sense to me. There is no relationship between extensions and lines until someone creates it in extensions.conf. |
| 02:56.57 | CRCinAU_ | thats what I've done. |
| 02:57.15 | CRCinAU_ | so in sip.conf, I register with the SIP providers |
| 02:57.16 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 02:57.24 | CRCinAU_ | then I also have a definition for each 'button' on the phone |
| 02:57.40 | p3nguin | You can have 6 lines on the phone and 0 extensions relating to the phone... but that makes it hard for anyone to make a call to the phone. |
| 02:58.04 | CRCinAU_ | no no no |
| 02:58.09 | CRCinAU_ | the phone registers to asterisk 4 times |
| 02:58.17 | p3nguin | Not with sccp. |
| 02:58.24 | CRCinAU_ | hmmm ok |
| 02:58.31 | CRCinAU_ | maybe thats where I'm getting confused. |
| 02:58.51 | p3nguin | There's almost no configuration on the phone in sccp. It's all done in sccp.conf. |
| 02:59.02 | CRCinAU_ | hmmmm |
| 02:59.04 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 02:59.30 | p3nguin | I'll put up some examples in the pastebin. Give me a minute. |
| 02:59.57 | LemensTS | Can you have a register string when you use IP auth for SIP? |
| 03:00.12 | CRCinAU_ | at the moment, I have the following in sccp.conf - thats my start: http://fpaste.org/D4TO/ |
| 03:01.53 | CRCinAU_ | and btw, I'm using: Chan_SCCP-3.0_RC3.1 |
| 03:06.55 | CRCinAU_ | ok - I *think* I have a working SEP xml |
| 03:07.16 | p3nguin | crcinau_: Here's an example using templates. It has one device (phone) and two lines configured on it, one regular and one shared line: http://pastebin.com/CBpv3Xy6 |
| 03:07.45 | p3nguin | crcinau_: If you'll follow along, I'll explain it all. |
| 03:07.50 | CRCinAU_ | how does this look for the SEP: http://fpaste.org/o2WO/ |
| 03:08.17 | p3nguin | crcinau_: Line 1 in the pastebin creates a device template by the name of "defaultdevice." |
| 03:08.56 | CRCinAU_ | so thats anything that isn't specifically defined? |
| 03:09.18 | CRCinAU_ | or jsut the default options that the device uses if not changed on a per-device basis? |
| 03:09.28 | p3nguin | No, it's a template. It will be used as a template when real devices are added. |
| 03:09.34 | CRCinAU_ | m'kay |
| 03:10.16 | p3nguin | crcinau_: Line 35 creates a device template by the name of "7960" which uses everything from the defaultdevice template and adds its own additional settings. |
| 03:11.01 | CRCinAU_ | ahhhhh ok |
| 03:11.24 | p3nguin | crcinau_: Line 41 creates a device template by the name of "7970" which uses everything that 7960 added (which already included defaultdevice). |
| 03:11.31 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 03:12.04 | p3nguin | crcinau_: Line 50 adds a real device by the name of SEP002244668800, and it is using the 7970 template. |
| 03:12.35 | p3nguin | In addition to using everything in the 7970 template, it also adds a few of its own settings, including programming some buttons. |
| 03:12.58 | CRCinAU_ | so line 50 is the first actual phone, everything else a template? |
| 03:13.14 | p3nguin | Lines 52 and 53 in the pastebin set two line keys on the phone to do something useful. |
| 03:13.49 | CRCinAU_ | can you choose which line keys? or just from top to bottom? |
| 03:14.07 | p3nguin | Line 52 programs "line 98021" to the top button. |
| 03:14.47 | p3nguin | As you add button entries, it goes from top to bottom. If you want to skip one or more, use "button = empty" to leave a line key empty. |
| 03:15.23 | CRCinAU_ | ahhh sweet. |
| 03:15.45 | p3nguin | Moving on... Line 62 in the paste creates a template for a phone line by the name of "defaultline". |
| 03:16.55 | p3nguin | Line 86 creates a line by the name of 98021, which inherits the settings that are in defaultline, plus it adds its own settings. Line 98021 was previously set on the first button of the phone. |
| 03:17.37 | p3nguin | Line 100 of the paste creates a phone line by the name of 98099, which was assigned to the second line button on the phone. 98099 is a shared line. |
| 03:18.05 | p3nguin | That's the gist of it. |
| 03:18.47 | CRCinAU_ | ok - so I'm copying this over bit by bit to my setup to try and get my head around it ;) |
| 03:19.22 | p3nguin | You don't have to use templates; you can add every setting you want to use for each device under each device's entry if you want. Same for each line entry. The templates are used to make the configs shorter and reduce duplications. |
| 03:20.17 | CRCinAU_ | yeah - If this works well, I'll change my 7961 over to SCCP as well ;) |
| 03:20.27 | CRCinAU_ | then eventually, a 7940 thats offsite |
| 03:21.35 | p3nguin | So if your phone's MAC address is 002244668800, it will be known as SEP002244668800 in SCCP. You can use SEP002244668800 as the device name in sccp.conf. I then create line entries such as 002244668800-a 002244668800-b 002244668800-c and 002244668800-d for four lines that belong on SEP002244668800. |
| 03:22.54 | p3nguin | Then to call those lines, you can create extensions using those line names. |
| 03:23.22 | CRCinAU_ | http://fpaste.org/Qj4U/ |
| 03:23.41 | CRCinAU_ | then 101 and 0383380299 are in extensions.conf |
| 03:23.53 | p3nguin | For example, extension 543 calls line 1 on the phone. exten => 543,1,Dial(SCCP/002244668800-a,60) |
| 03:24.42 | *** join/#asterisk jetlag (~jetlag@pool-173-61-212-250.cmdnnj.east.verizon.net) |
| 03:25.16 | p3nguin | So you added a device on line 102 of the paste, and you added line buttons, but you didn't add your lines yet. |
| 03:25.46 | CRCinAU_ | hmmm |
| 03:25.57 | CRCinAU_ | none of my buttons have anything next to them |
| 03:26.09 | p3nguin | Did you already flash the phone with sccp? |
| 03:26.14 | CRCinAU_ | [Mar 8 14:25:16] WARNING[11701]: sccp_actions.c:449 sccp_make_button_template: SEP001A2F63D529: line 0383380299 does not exists |
| 03:26.17 | CRCinAU_ | hmmm |
| 03:26.23 | CRCinAU_ | so I assume everything has to be in the same context? |
| 03:26.37 | p3nguin | Create a line by the name of 0383380299 to make that message go away. |
| 03:26.51 | CRCinAU_ | ohhh yes! |
| 03:26.57 | CRCinAU_ | I haven't got that far yet ;) LOL |
| 03:26.58 | p3nguin | Everything has to be in what context? We're not talking about contexts yet. |
| 03:27.35 | CRCinAU_ | I keep thinking it'll get them from sip.conf or something LOL |
| 03:29.05 | p3nguin | Speaking of context, I noticed that you didn't assign your phone a context. |
| 03:29.48 | p3nguin | Well, you also didn't create any lines, so I guess that's why I didn't see any context setting yet. |
| 03:29.56 | p3nguin | The context goes in the line entry. |
| 03:31.02 | CRCinAU_ | m'kay... getting there :p |
| 03:35.00 | CRCinAU_ | ok - so now I'm up to: http://fpaste.org/hDwf/ |
| 03:35.54 | p3nguin | Looking pretty good, but now you need to add more useful settings to each line... line context = phones or something. |
| 03:35.57 | CRCinAU_ | so I have 101 displayed on the 1st button. |
| 03:36.05 | CRCinAU_ | and that relates to extensions.conf? |
| 03:36.12 | CRCinAU_ | ie the same as the sip stuff? |
| 03:36.23 | p3nguin | context = phones would mean that calls coming from that line start in [phones] in extensions.conf. |
| 03:36.32 | CRCinAU_ | sweet. |
| 03:36.43 | p3nguin | context= will be the same as it is in sip.conf. |
| 03:38.20 | *** part/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 03:38.36 | CRCinAU_ | hell yes |
| 03:38.57 | p3nguin | You can set the label for the button, you can set callerid, mailbox, extension to transfer to voicemail, etc. |
| 03:39.12 | p3nguin | accountcode |
| 03:39.25 | CRCinAU_ | I just made a call from 101 -> 102 (sccp -> sip) and it rang and had audio. |
| 03:39.26 | CRCinAU_ | thats a win. |
| 03:40.59 | CRCinAU_ | w00t |
| 03:41.03 | CRCinAU_ | label = blah set |
| 03:42.44 | CRCinAU_ | damn this is easier than SIP :p |
| 03:42.48 | p3nguin | yeah |
| 03:43.01 | CRCinAU_ | I'm kinda glad its all supported now too |
| 03:43.14 | CRCinAU_ | as the last time I looked at SCCP, there was no hold, no transfer, no moh etc etc etc |
| 03:43.22 | CRCinAU_ | that being said, that was a long time ago |
| 03:43.32 | p3nguin | Once you figure out the sample configs, you'll enjoy setting up softkeys, too. |
| 03:43.56 | CRCinAU_ | what does the rigth hand softkey do on the 7970? |
| 03:44.04 | p3nguin | That was also the provided chan_skinny, not this 3rd party chan_sccp. |
| 03:44.07 | CRCinAU_ | that looks like the resotre button on a Windows desktop |
| 03:44.14 | p3nguin | It'll do what you program it to do, I suppose. |
| 03:44.20 | CRCinAU_ | hmmm |
| 03:44.39 | p3nguin | Look at the supplied sample confs. |
| 03:45.15 | CRCinAU_ | yeah - I think it'll take me a day or so to get my head around all these :p |
| 03:48.34 | p3nguin | One thing I like about sccp is that you can reset the phone from the console. sccp reset SEP001A2F63D529 |
| 03:48.44 | CRCinAU_ | yeah - I noticed that... major w00t |
| 03:48.49 | p3nguin | reset or restart, even |
| 03:49.11 | CRCinAU_ | ok - so |
| 03:49.24 | CRCinAU_ | http://fpaste.org/A4b2/ |
| 03:49.39 | CRCinAU_ | say I want to have the bottom 3 buttons set as shared lines. |
| 03:49.45 | CRCinAU_ | ie they show busy on other phones as well. |
| 03:50.01 | CRCinAU_ | or even so I can put a line on hold, then answer it on another phone or something |
| 03:50.18 | p3nguin | You've got 8 line keys? |
| 03:50.24 | CRCinAU_ | on the 7970, yeah |
| 03:50.33 | CRCinAU_ | the 7961 has 6 |
| 03:50.41 | p3nguin | Oh, I thought it was 6 buttons line the 7960. |
| 03:50.53 | p3nguin | I use a 7960. |
| 03:51.15 | CRCinAU_ | I've got a 7940, a 7961 and a 7970 |
| 03:51.27 | CRCinAU_ | I've just managed to be in the right place and the right time and got them cheap cheap ;) |
| 03:51.33 | CRCinAU_ | the 7970 cost me $100AUD :P |
| 03:51.41 | CRCinAU_ | thats about $105USD ;) |
| 03:51.58 | p3nguin | That's a good price. They normally run $250. |
| 03:52.43 | CRCinAU_ | indeed. |
| 03:52.54 | CRCinAU_ | the 7961 I got for $50 about a year ago |
| 03:53.01 | CRCinAU_ | and the 7940 I got for free :P |
| 03:53.46 | p3nguin | I think I want to get a 7961. Other than the lighted line buttons, what else does it have over the 7960? |
| 03:54.01 | CRCinAU_ | a shitloads better screen |
| 03:54.12 | CRCinAU_ | ie orders of magnitude better |
| 03:55.21 | CRCinAU_ | hmmm |
| 03:55.26 | *** join/#asterisk f2Knight (~ben@c-71-236-165-148.hsd1.or.comcast.net) |
| 03:55.30 | CRCinAU_ | the messages button doesn't seem to auto-login anymore |
| 03:55.49 | CRCinAU_ | probably caller ID issues. |
| 03:56.19 | p3nguin | If I press my messages key, it says "password?" |
| 03:56.51 | f2Knight | Q: A few days ago I asked if anyone could help with phantom calls from 'asterisk', I have a pcap from tcpdump that I caught is anyone could help me trace it back to a source??? |
| 03:57.18 | CRCinAU_ | ah. found the issue :p |
| 03:57.56 | CRCinAU_ | i think |
| 03:58.03 | CRCinAU_ | that's better. |
| 03:58.16 | CRCinAU_ | exten => 999,1,Ringing() |
| 03:58.16 | CRCinAU_ | exten => 999,2,VoicemailMain(s${CALLERID(num)}) |
| 03:58.17 | CRCinAU_ | exten => 999,3,Hangup() |
| 03:58.29 | CRCinAU_ | that way, it auto-logs in from the extension called. |
| 03:58.50 | p3nguin | That sentence doesn't make any sense to me. |
| 03:59.00 | f2Knight | CRCinAU_, be cautious using CallerID to auto log a user in . |
| 03:59.45 | p3nguin | I have a similar extension associated with my messages key: exten => *86,n,VoiceMailMain(${CALLERID(num)}@default); |
| 03:59.50 | CRCinAU_ | I understand if someone hacks it and gets a call happening, it'll be annoying. |
| 04:00.08 | CRCinAU_ | however I only allow SIP messages to my box from the IPs of my vsp |
| 04:00.10 | f2Knight | I for example have on my system a special extension that I can call to fudge my callerID and then dial any number I wish, I use this on friends and clients as a demo of an often overlooked item |
| 04:00.22 | CRCinAU_ | anything else gets dropped as well ;) |
| 04:00.36 | CRCinAU_ | plus, this is only at home - so yeah - security on my LAN isn't a huge issue |
| 04:01.07 | nestAr | i don't see the risk, they still have to enter a pin. |
| 04:01.07 | f2Knight | CRCinAU_, okay .. still :) |
| 04:01.07 | p3nguin | I don't see any problem with using the callerid number as the voicemail mailbox number... you still have to enter the password. |
| 04:01.41 | CRCinAU_ | I have it set so I don't need a pin... |
| 04:01.45 | nestAr | that and i seperate all the voicemail stuff into a seperate context, only accessible from internal phones |
| 04:01.49 | CRCinAU_ | it confuses my g/f too much ;) |
| 04:01.52 | f2Knight | nestAr, not with the option he just showed |
| 04:02.07 | CRCinAU_ | now. |
| 04:02.11 | CRCinAU_ | p3nguin: the mwi :) |
| 04:02.13 | f2Knight | nestAr, the 's' tells asterisk to skip the passcode |
| 04:02.19 | CRCinAU_ | we talked about it briefly the other day.... |
| 04:02.27 | p3nguin | yep |
| 04:02.33 | nestAr | pins are hard! |
| 04:02.50 | f2Knight | no pins are not hard. |
| 04:03.00 | nestAr | f2Knight: as long as it's context limited, shouldn't be a big issue |
| 04:03.07 | CRCinAU_ | my g/f has enough issues remembering the pin from her ATM cards etc...... |
| 04:03.08 | p3nguin | pins are pointy |
| 04:03.08 | f2Knight | I have a script that will pound the system for it rather quickly |
| 04:03.22 | f2Knight | p3nguin, true |
| 04:04.11 | CRCinAU_ | p3nguin: so, I have 1 new voicemail waiting in the system, but no MWI lit up. |
| 04:04.15 | f2Knight | CRCinAU_, I understand they are issues with memory of them, however if you are allowing outside access to the system in anyway I caution you against it as a mater of practice |
| 04:04.24 | p3nguin | crcinau_: Did you configure it? |
| 04:04.39 | CRCinAU_ | p3nguin: I believe so. |
| 04:04.41 | f2Knight | CRCinAU_, what kinda phone you using? |
| 04:04.48 | CRCinAU_ | mwilamp = blink ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink |
| 04:04.51 | CRCinAU_ | mwioncall = on ; Set the MWI on call. |
| 04:04.54 | CRCinAU_ | f2Knight: a cisco 7970 |
| 04:05.12 | CRCinAU_ | maybe I have to leave a new message. |
| 04:05.24 | f2Knight | ugh I hate cisco configs such a nightmare lol well the 7940/60 are have not used anyothers |
| 04:05.42 | p3nguin | crcinau_: Under the line entry, did you set mailbox = <number>@<context> ? |
| 04:05.46 | CRCinAU_ | f2Knight: it doesn't change much ;) |
| 04:06.09 | *** join/#asterisk rectifiertechnol (~rtp.aster@s1936.static.pacific.net.au) |
| 04:06.10 | CRCinAU_ | [101](defaultline) |
| 04:06.10 | CRCinAU_ | mailbox = 101 |
| 04:06.11 | CRCinAU_ | context = 101 |
| 04:06.22 | p3nguin | 7940 and 7960 are basically the easiest there is. If you have trouble with them, I'd hate to see you work on anything else. |
| 04:06.32 | p3nguin | crcinau_: Both of those are failures. |
| 04:06.34 | CRCinAU_ | do I need to set it with *both* mailbox and context? or is it smart enough to figure it out? |
| 04:06.45 | p3nguin | context = phones; This is the DIAL PLAN CONTEXT |
| 04:06.57 | p3nguin | mailbox = NUMBER@VMCONTEXT |
| 04:07.00 | f2Knight | p3nguin, then you have not worked with real SIP IP phones, snom, aastra, grandstream, yealink |
| 04:07.03 | CRCinAU_ | in my extensions.conf, I do have a context for each phone |
| 04:07.11 | CRCinAU_ | then I include what I want in each phones context |
| 04:07.17 | p3nguin | NUMBER = your voicemail box number, CONTEXT = voicemailcontext. |
| 04:07.24 | f2Knight | all allow either tftp config OR individual webservices to config them but not a 7940/60 |
| 04:07.27 | p3nguin | crcinau_: That's retarded. |
| 04:07.36 | CRCinAU_ | yes and no |
| 04:07.44 | CRCinAU_ | as I have some devices that are only line only |
| 04:07.55 | CRCinAU_ | and some should be able to dial out via a VSP, and some not |
| 04:08.18 | CRCinAU_ | so I have a context for each phone, and something like: include => local; include => vsp1 etc |
| 04:08.23 | f2Knight | and 7940/60 are only easy if your using callcenter from cisco, if your using SIP on them its a nightmare as the config options change from minor version to minor version, |
| 04:08.29 | p3nguin | So fix your mailbox setting to the correct box number and context, and the MWI will probably work. |
| 04:09.29 | f2Knight | CRCinAU_, have you pastbin your config? |
| 04:10.07 | p3nguin | If you fix that and it still doesn't work, show me what you have changed and I'll see if I have any suggestions. |
| 04:10.40 | CRCinAU_ | standby... tinkering... |
| 04:10.40 | f2Knight | ~pastebin |
| 04:10.40 | infobot | [~pastebin] 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. |
| 04:11.23 | CRCinAU_ | hmm |
| 04:11.26 | CRCinAU_ | does this matter? |
| 04:11.27 | CRCinAU_ | asterisk*CLI> sccp show mwi subscriptions |
| 04:11.28 | CRCinAU_ | Command has not been fully implemented yet! |
| 04:11.28 | CRCinAU_ | Command 'sccp show mwi subscriptions ' failed. |
| 04:11.41 | f2Knight | your using sccp? |
| 04:11.50 | CRCinAU_ | nods. |
| 04:11.57 | CRCinAU_ | SIP on a 7970 has been a waste of 2 days |
| 04:12.00 | rectifiertechnol | hi all - i have an inbound route that sends calls to a sip extension which connects via a pstn gateway to our ananlogue pabx, then i want to automatically dial an extension on the an pabx such that calls are router automatically from the voip provider to our reception number - any ideas ? ie : voip call -> inbound route - > sends to sip extension - > connects to analogue pabx - > dial 599 to |
| 04:12.00 | rectifiertechnol | call reception analogue extension |
| 04:12.20 | p3nguin | I set mwilamp = blink and my MWI no longer works. Just set it to on. |
| 04:12.24 | f2Knight | CRCinAU_ has found that CISCO phones are not worth the time unless your using cisco callcenter |
| 04:12.36 | CRCinAU_ | hahah yeah ;) |
| 04:13.45 | CRCinAU_ | the 7940/60 and 7941/61 are fine |
| 04:14.13 | f2Knight | CRCinAU_, I have a box of them anyone wants to buy them. |
| 04:14.39 | f2Knight | on the flip side,,, in extensions.conf something like ... |
| 04:15.15 | nestAr | s/your/you're/ |
| 04:15.40 | nestAr | i've got some of the new SPA525G2's and they are pretty snazzy |
| 04:15.50 | CRCinAU_ | f2Knight: where abouts are you? I might know a buyer soon :) |
| 04:15.50 | p3nguin | I could use a box of phones. You can send them to me if you'd like. |
| 04:16.00 | CRCinAU_ | yay |
| 04:16.01 | f2Knight | exten => *97,1,VoiceMailMain(s${CALLERID(num)}@default) |
| 04:16.08 | CRCinAU_ | 'you have voicemail 1/0' |
| 04:16.17 | shapr | ! |
| 04:16.37 | CRCinAU_ | the MWI is on, and the lovely envelope is fading in and out next to the button :) |
| 04:16.45 | f2Knight | nestAr, the SPA's are rebranded Linksys phones, and yes use a webinterface and config of them is MUCH easier |
| 04:17.17 | CRCinAU_ | f2Knight: unless you have to do 20+ of them |
| 04:17.21 | CRCinAU_ | then it's URGH |
| 04:17.59 | CRCinAU_ | I did a voip rollout for a pharmacy network using SPA2102s I think... I built an auto-provisioning system for ~860 devices. |
| 04:18.07 | CRCinAU_ | THANK GOD for scripting. |
| 04:18.14 | f2Knight | CRCinAU_, and if thats the case you use tftp and copy the config file after you get one. , but no good tools out there to rapidly provision them that I am aware of |
| 04:18.27 | CRCinAU_ | nah - I had to roll my own tools |
| 04:18.37 | nestAr | f2Knight: and? |
| 04:18.38 | nestAr | ;) |
| 04:18.49 | CRCinAU_ | and then the devices inital config was via TFTP in the office, then all the configs were stored on a HTTP server |
| 04:18.59 | CRCinAU_ | as too many modems break TFTP |
| 04:19.09 | f2Knight | yea I like working with HTTP better, too. |
| 04:19.57 | nestAr | i've not used the HTTP interface, i've just used the TFTP xml |
| 04:20.11 | f2Knight | CRCinAU_, are you able to dial your extension to voicemail and get your voicemail? |
| 04:21.37 | CRCinAU_ | f2Knight: yeah :) it works ok now :) |
| 04:22.00 | CRCinAU_ | but am I right in thinking theres no security to sccp? ie username / password auth for sessions? |
| 04:22.06 | f2Knight | what was your option, to fix it? |
| 04:22.24 | *** part/#asterisk lanning (~lanning@208.87.233.137) |
| 04:22.37 | f2Knight | sccp, from my understanding is not intended to be used on anything but a lan. |
| 04:44.31 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 04:44.31 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 04:56.45 | p3nguin | crcinau_: It seems somewhat secure to me. The device doesn't control any of its own settings. The only thing the device has to do is get an IP address and accept the configuration that the call control device (asterisk) has for it. |
| 05:18.03 | f2Knight | Q: Call-ID: 1822268171-5060-1@HB.CDG.BGF.BEI was found in a tcpdump.. any idea what this is? |
| 05:21.16 | *** join/#asterisk radic (~radic@dslb-178-002-215-207.pools.arcor-ip.net) |
| 05:22.02 | CRCinAU_ | p3nguin: true. |
| 05:22.41 | CRCinAU_ | p3nguin: so if I want a shared line, I simply add another device, with a button = blah? |
| 05:22.47 | CRCinAU_ | and it'll do the rest itself? |
| 05:23.27 | p3nguin | You can have a shared line with only a single configured device. |
| 05:24.05 | p3nguin | You just have to define a shared line and assign it to a button on one phone or more. |
| 05:24.54 | CRCinAU_ | yeah - so say I have my 7961 as SCCP too, I can just have the same button = blah line in the config, and done. |
| 05:25.59 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 05:27.55 | p3nguin | Basically, you just add the line to a button on both phones. |
| 05:28.02 | CRCinAU_ | thats sweet :D |
| 05:28.11 | p3nguin | But there is more to it if you are interested. |
| 05:28.18 | CRCinAU_ | how does it go if one is busy, one is available? just ring the available phone? |
| 05:28.29 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 05:28.34 | p3nguin | They'll both ring, since you have callwaiting. |
| 05:28.44 | CRCinAU_ | hmmmm |
| 05:28.49 | p3nguin | If you don't have call waiting, then only one will ring. |
| 05:28.59 | CRCinAU_ | in the same scenario |
| 05:29.06 | CRCinAU_ | if phone A is busy, B rings. |
| 05:29.29 | p3nguin | If you Dial() to the shared line, both phones will ring. |
| 05:29.33 | CRCinAU_ | A sees it as call waiting. |
| 05:29.37 | CRCinAU_ | lets say B answers |
| 05:29.43 | CRCinAU_ | does it still come up as a missed call on A? |
| 05:29.48 | p3nguin | yes |
| 05:29.55 | CRCinAU_ | fooey. |
| 05:30.08 | p3nguin | The phone got a call that it didn't answer, so it is missed. |
| 05:31.21 | CRCinAU_ | thats the one thing that bugs me about these setups |
| 05:31.26 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 05:31.27 | CRCinAU_ | missed is not always 'missed' |
| 05:32.37 | p3nguin | From the standpoint of the phone, yes it is. If there was a call that the phone got, but it wasn't answered, it's a missed call. |
| 05:33.36 | CRCinAU_ | yeah I know... but I wish there was a way to tell it 'hey! someone else answered it!; |
| 05:33.49 | p3nguin | Don't use shared lines in place of another mechanism. If you're having a problem with a phone missing a call, maybe you don't want a shared line. |
| 05:34.44 | CRCinAU_ | well, lets say we have 4 phones. |
| 05:34.52 | p3nguin | Having an SCCP shared line is going to be similar to dialing two devices using & in the Dial() command. |
| 05:34.52 | CRCinAU_ | we want all of them to ring when someone calls us from outside |
| 05:35.12 | CRCinAU_ | then its a 'first person to answer, wins' scenario |
| 05:35.58 | CRCinAU_ | then it doesn't really mean a missed call on all other phones |
| 05:36.07 | CRCinAU_ | just that you didnt 'win' |
| 05:36.26 | p3nguin | I understand what you're saying, but I don't know of any way to get around this problem. It happens on every brand of phone using every channel technology. |
| 05:36.45 | CRCinAU_ | :( |
| 05:38.46 | CRCinAU_ | anyhow - I've just been informed by the Minister of War and Finances that its time to head out.... |
| 05:38.49 | CRCinAU_ | bbk. |
| 05:39.49 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:40.58 | p3nguin | Minister of War and Finances? Is that secret code for wife/girlfriend/significant other? |
| 05:43.21 | f2Knight | CRCinAU[a], to do what you want is best done I think with simple dialplan logic |
| 05:43.39 | f2Knight | CRCinAU[a], assign each phone its own-id |
| 05:44.27 | f2Knight | CRCinAU[a], then in your dial plan do a dial(sip/101,20) |
| 05:44.58 | f2Knight | CRCinAU[a], to ring the line on device 101 for 20 seconds, at the end of that time the next rule in your dialplan would work. |
| 05:45.10 | f2Knight | CRCinAU[a], a dial(sip/102,20) |
| 05:45.29 | p3nguin | But that's annoying. |
| 05:45.30 | f2Knight | CRCinAU[a], you can get fancy with using a ${CHANSTATUS} |
| 05:45.53 | p3nguin | 'Cause then you get one phone ringing, then it stops and another phone starts ringing. |
| 05:46.22 | f2Knight | CRCinAU[a], and put it all in a generic macro so all your doing is macro(mydialer,${EXTEN},${OTHERNUM}) |
| 05:46.52 | rectifiertechnol | :( |
| 05:46.56 | f2Knight | if you really WANT both phones to ring dial(SIP/101&SIP/102,20) |
| 05:47.20 | f2Knight | that will ring them both for 20 seconds, and then you can check the status and send it to voicemail |
| 05:47.24 | p3nguin | But with a shared line, you just Dial(SCCP/number-of-shared-line,40) |
| 05:47.34 | f2Knight | ofcourse set the 20seconds to what ever you want to hear the ring |
| 05:47.42 | p3nguin | And that rings the lines on all the phones which have the shared line on them. |
| 05:48.50 | f2Knight | there is no difference then of making an extension say 600, that is a ringgroup |
| 05:49.03 | f2Knight | and dial(local/600,40) |
| 05:49.23 | p3nguin | Except that a shared line does more. |
| 05:49.45 | f2Knight | I have not seen real shared line support function in asterisk |
| 05:49.53 | p3nguin | With a shared line, you can answer it on one phone, press HOLD, and go pick up that call on another phone with the same shared line. Without parking. |
| 05:50.05 | f2Knight | i agree parking is a pain |
| 05:50.19 | *** join/#asterisk blee_ (~blee@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 05:50.23 | p3nguin | Shared line takes care of that. Almost like a key system. |
| 05:50.48 | f2Knight | I have not seen a working example of it. for that matter, I have not seen any real working examples of call pickup |
| 05:51.07 | p3nguin | I just gave you an example of the shared line. |
| 05:51.17 | f2Knight | i am supposed to be able to use call pickups to do the same thing, however, never seen it work no mater how many times i have tired |
| 05:51.36 | f2Knight | no you gave me an EXPLANATION of shared lines |
| 05:51.40 | f2Knight | not an working example. |
| 05:51.55 | f2Knight | or in other words, show me the codes |
| 05:51.59 | f2Knight | :) |
| 05:52.23 | p3nguin | Pickup is pretty simple. My co-worker's phone is ringing, I'm in the same pickupgroup/callgroup, I pick up my phone and dial *8 and hijack the call that was ringing his phone. |
| 05:53.35 | f2Knight | I have not gotten it to work, I use grandstream phones and setting a blf on the phone when you press it when ringing its supposed to do pickups |
| 05:54.22 | p3nguin | http://pastebin.com/Q5KzGKq6 |
| 05:55.11 | f2Knight | what config file is that ? |
| 05:55.15 | f2Knight | sip.conf? |
| 05:55.46 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:56.05 | p3nguin | It's part of sccp.conf |
| 05:56.14 | p3nguin | The only callgroup/pickupgroup stuff I used was from res_features. pickupexten = *8 |
| 05:57.40 | f2Knight | umm.. okay I have never looked at a sccp.conf file so i guess thats why it looks odd, I am going to try the call group, pickupgroup settings on a demo phone later tomorrow, see if I get it working , thanks for the example |
| 05:57.45 | p3nguin | The part of sccp.conf I showed you shows the device (phone) entry and it has two lines assigned to two line buttons. One line is a regular line, the other is a shared line. If I added another device and wanted to use the shared line, I would add the same shared line to a line button on the next device. |
| 05:59.19 | p3nguin | SIP callgroup/pickupgroup is basically as easy as putting callgroup=1 and pickupgroup=1 in two sip phone definitions. |
| 05:59.21 | f2Knight | is there anything special with asterisk doing it or is it a phone thing? |
| 05:59.53 | p3nguin | Now both of those devices are in callgroup 1 and pickupgroup 1. A call going to a phone in callgroup 1 can be picked up by any phone in pickupgroup 1. |
| 05:59.59 | f2Knight | i know the 794x/6x have a shared line settings |
| 06:00.50 | f2Knight | so on the group thing.. the pickupgroup = number is related to the callgroup = number? |
| 06:01.05 | p3nguin | right |
| 06:01.13 | p3nguin | A call going to a phone in callgroup 1 can be picked up by any phone in pickupgroup 1. |
| 06:01.51 | f2Knight | e.g. callgroup=2,20,40 on one device another decvice with pickupgroup = 20 would be able to pick up the first devices call? |
| 06:02.09 | p3nguin | I think you said that correctly, yes. |
| 06:02.18 | f2Knight | that might have been my big blundering |
| 06:02.40 | f2Knight | I was always thinking the pickupgroup was a group of phones to share pickups |
| 06:02.58 | p3nguin | If one phone is in three callgroups, and three other phones are in each of the groups, all three phones could dial *8 to hijack the call. |
| 06:03.39 | p3nguin | Let me make a very basic sip.conf to demonstrate. |
| 06:05.24 | f2Knight | i think i understand what you mean.. |
| 06:06.02 | f2Knight | its still not a shared line, (say for a real sip phone) |
| 06:06.21 | f2Knight | i think the shared lines your using on the ciscos are implimented in the phone. |
| 06:06.28 | p3nguin | http://pastebin.com/gCgzZrXu |
| 06:07.15 | p3nguin | In this paste, a call going to SIP/john could be picked up by jack or jane by dialing *8. |
| 06:07.52 | p3nguin | If you hear the phone ringing from the other room, grab your phone and dial *8 and see if it answers the call. |
| 06:07.54 | *** join/#asterisk fIorz (nobody@2001:1a50:503c::1) |
| 06:08.01 | f2Knight | and jack could pick up john and jane |
| 06:08.22 | p3nguin | but no one can pick up a call to SIP/jack |
| 06:08.38 | f2Knight | because no on eis in callgroup 1? |
| 06:08.48 | f2Knight | except jack |
| 06:08.58 | p3nguin | because jack is in callgroup 1 but no one has a pickupgroup 1. |
| 06:09.11 | f2Knight | ahh right got it |
| 06:09.46 | f2Knight | so you can pick it up but still hold is left to parking lots right.? |
| 06:09.48 | p3nguin | The *8 pickupexten is configured in features.conf. |
| 06:10.39 | p3nguin | Once you pick up the call that was ringing some other phone, you can put it on regular hold and then only your phone can resume it... or you can park it and then any phone can retrieve it out of parking. |
| 06:11.42 | p3nguin | In Cisco using chan_sccp/sccp.conf, a shared line allows a call to the shared line to be placed on hold and then any other phone sharing the line can resume that call. |
| 06:12.13 | p3nguin | They sort of do similar things, but they are really completely different. |
| 06:13.44 | f2Knight | okay so the sharedline is actually just the sccp/phone doing it not actually asterisk, or more exact, its not available on sip/iax channels :? |
| 06:14.07 | p3nguin | It's the sccp channel driver doing it. |
| 06:14.14 | p3nguin | chan_sccp |
| 06:14.42 | f2Knight | ah okay |
| 06:14.49 | f2Knight | that would make sense then., |
| 06:14.50 | p3nguin | If other phone brands can use SCCP, you could use those phone brands on Asterisk and chan_sccp. |
| 06:15.05 | p3nguin | But as far as I know, Cisco is the only one with SCCP. |
| 06:15.31 | f2Knight | hey, I get random phantom calls from 'asterisk' the box has no hardware in it its a pure sip box. |
| 06:16.13 | p3nguin | Okay... do you have any cron tasks originating calls? or any .call files? |
| 06:16.14 | f2Knight | I happened to capture a tcpdump of port 5060 after it did it and caught it doing it again then stopped it think you could take a look at it and see what is going on? |
| 06:16.25 | f2Knight | crons empty |
| 06:16.54 | p3nguin | ls -al /var/spool/asterisk/outgoing/ |
| 06:17.19 | f2Knight | yea nothing in the outgoing spool either |
| 06:17.25 | f2Knight | and nothting at the time of the call |
| 06:17.32 | f2Knight | its really random when it happens |
| 06:17.44 | f2Knight | and it seems to only be calling my extension, |
| 06:17.50 | f2Knight | when i pick up its dead air |
| 06:18.20 | p3nguin | Do you leave your verbose high enough to see the call on the cli? |
| 06:18.42 | f2Knight | 5? |
| 06:19.07 | f2Knight | but no this never hits the cdr records |
| 06:19.27 | f2Knight | and even when the call is happing nothing ever hits the cli i always have it open monitoring it |
| 06:20.56 | f2Knight | i have a pcap of the dump of port 5060 that has the phantom call in it |
| 06:23.23 | f2Knight | if that could help |
| 06:24.11 | p3nguin | verbose level 5 is more than enough. It should show you the call when it happens. |
| 06:25.05 | p3nguin | I doubt it will help me figure anything out, but it might help someone if other people will check it out. I'll look, too. |
| 06:30.48 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 06:32.21 | f2Knight | http://pastebin.com/bNm8NWrT |
| 06:33.00 | f2Knight | that is the UDP stream as presented by wireshark |
| 06:35.15 | *** join/#asterisk thenk (cb3b1a74@gateway/web/freenode/ip.203.59.26.116) |
| 06:35.25 | thenk | hello from australia, anyone around? |
| 06:35.33 | wdoekes2 | sure |
| 06:35.50 | thenk | :D |
| 06:36.06 | wdoekes2 | ~ask |
| 06:36.07 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 06:36.09 | thenk | so tell me, how hard is it to get video working with asterisk? |
| 06:36.33 | thenk | personally I hate joining channels just to throw a question, I'd kinda like to introduce myself first |
| 06:36.40 | wdoekes2 | I have no idea :) never done video.. but I haven't heard any complaints here, so that might be considered a good thing |
| 06:36.43 | thenk | but you know :D |
| 06:37.26 | thenk | well i've spent some time trying to get a simple dialplan together which just plays some video, but was unable to |
| 06:37.53 | wdoekes2 | perhaps it's easier to get client-asterisk-client video first |
| 06:37.59 | wdoekes2 | if you have two devices that do video |
| 06:38.04 | thenk | well that's just it, I haven't really been able to do that even |
| 06:38.18 | thenk | fresh asterisk, configured sip.conf with videosupport=yes and the codecs in [general] |
| 06:38.38 | thenk | the clients negotiate and the channel gets bridged, sip show channel on the bridge shows that it's running h263 |
| 06:38.42 | thenk | but I get no video on either end |
| 06:38.55 | thenk | same result with 1.4 and 1.6, perhaps i need to try another softphone |
| 06:39.19 | thenk | was using Ekiga |
| 06:39.54 | wdoekes2 | hm.. can't help you here. apart from telling you to 'core set verbose 10' and 'sip set debug on'.. which will yield lots of output |
| 06:39.58 | thenk | what I'm trying to do is get Asterisk to play back an RTSP stream |
| 06:40.30 | thenk | the rtsp stream encoded in h263@qcif |
| 06:40.48 | thenk | app_rtsp doesn't seem to work at all |
| 06:41.03 | thenk | but if i use app_transcoder, it complains about bad header data |
| 06:41.16 | thenk | perhaps that's the issue - but other RTSP players all work fine, no issue |
| 06:41.24 | thenk | perhaps I am not linking something correctly when I build? |
| 06:41.30 | thenk | sighs |
| 06:43.09 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 06:43.25 | thenk | i'm hoping that someone knows how to use app_rtsp and app_transcoder |
| 06:43.34 | thenk | there's pretty much nil documentation out there |
| 06:50.32 | f2Knight | thenk, have not tried video support but I am up and in the US, if you need another client I could help out |
| 06:51.28 | f2Knight | thenk, I use linux desktops so could set up a linux client if you would like. |
| 06:52.00 | thenk | hang on a sec.. |
| 06:53.42 | thenk | try to give me a call, 601@203.59.26.116:9987 |
| 06:55.42 | thenk | ah hang on.. that doesn't seem to work |
| 06:55.45 | f2Knight | installing ekiga ... |
| 06:57.30 | thenk | for some reason my modem's not port forwarding correctly :/ |
| 06:57.53 | f2Knight | have you mapped a correct nat settings? |
| 06:58.17 | thenk | the settings are identical to my mail |
| 06:58.26 | thenk | which is working fine |
| 06:58.33 | thenk | well not identical - but yeah |
| 06:58.41 | f2Knight | is your mail server your sip server? |
| 06:58.53 | f2Knight | tarting Nmap 5.21 ( http://nmap.org ) at 2011-03-07 22:58 PST |
| 06:58.53 | f2Knight | Nmap scan report for 203-59-26-116.perm.iinet.net.au (203.59.26.116) |
| 06:58.53 | f2Knight | Host is up (0.28s latency). |
| 06:58.53 | f2Knight | Not shown: 996 filtered ports |
| 06:58.53 | f2Knight | PORT STATE SERVICE |
| 06:58.54 | f2Knight | 25/tcp open smtp |
| 06:58.56 | f2Knight | 80/tcp open http |
| 06:58.58 | f2Knight | 443/tcp open https |
| 06:59.00 | f2Knight | 1723/tcp open pptp |
| 06:59.02 | f2Knight | Nmap done: 1 IP address (1 host up) scanned in 20.33 seconds |
| 06:59.04 | f2Knight | root@VGN-FW351J:~# |
| 06:59.06 | f2Knight | I am not seeing any open ports |
| 07:00.10 | thenk | nah |
| 07:00.45 | thenk | okay let's just try straight 5060 then :S |
| 07:00.50 | f2Knight | I see your landing page :) |
| 07:01.51 | thenk | hehe |
| 07:02.00 | thenk | which one |
| 07:02.01 | thenk | :P |
| 07:02.03 | f2Knight | 5060 is still not showing open |
| 07:02.11 | thenk | it's configured to be so now |
| 07:02.18 | thenk | I'll bet there's some fancy thing in this modem stopping SIP |
| 07:02.19 | f2Knight | it says : |
| 07:02.20 | f2Knight | Welcome to the home of digitalmarketing.com.au |
| 07:02.21 | f2Knight | To change this page, upload your website into the public_html directory |
| 07:02.21 | f2Knight | Date Created: Mon Aug 24 23:55:47 2009 |
| 07:03.07 | f2Knight | been a landing page for quite some time :) |
| 07:03.15 | thenk | hehe yep |
| 07:03.24 | thenk | it's not meant to have anything there, really |
| 07:03.26 | thenk | I should just remove that :D |
| 07:04.27 | thenk | ther we go |
| 07:04.33 | thenk | grr.. |
| 07:04.37 | thenk | let's try another modem then.. |
| 07:04.59 | f2Knight | I am wondering if we both have to connect through a server supporting video. |
| 07:05.19 | thenk | according to the configuration, this asterisk server does |
| 07:06.58 | f2Knight | thenk, what I mean is I am not sure if BOTH endpoint have to connect through a supporting server. I am trying to connect directly with out an account right now. |
| 07:07.41 | thenk | well, we can dial directly to each other if need be |
| 07:08.23 | f2Knight | thats what I am trying just directly dialing you |
| 07:08.55 | thenk | well that's an asterisk server |
| 07:08.58 | thenk | or meant to be |
| 07:09.05 | thenk | I'm about to throw these modems at the wall eh |
| 07:09.16 | thenk | One of them just doesn't seem to work properly for the SIP forward |
| 07:09.24 | thenk | and the other one is refusing to accept the address of the asterisk server as an internal address |
| 07:09.35 | thenk | even though there's an entry in there already for the same server using tcp |
| 07:09.42 | thenk | but i can't add a tcp entry either |
| 07:09.47 | thenk | I think today's one of those days...... |
| 07:11.06 | f2Knight | i have seen routers only support a single port setting per ip |
| 07:11.12 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 07:11.14 | f2Knight | netgears for example. |
| 07:11.26 | thenk | this one's like, won't forward to ip addresses not on it's subnet |
| 07:11.29 | f2Knight | but also you need to forward UDP, |
| 07:11.29 | thenk | even though it has routes |
| 07:11.51 | thenk | the thing is though - i used this same mode to test some other projects |
| 07:11.54 | thenk | and those forwards worked fine |
| 07:12.36 | thenk | gotta put this box on the voice lan I guess |
| 07:12.40 | thenk | hold on :P |
| 07:12.46 | f2Knight | k |
| 07:13.13 | thenk | LOL |
| 07:13.13 | thenk | gerat |
| 07:13.15 | thenk | great |
| 07:13.25 | thenk | voice lan not on the vm host |
| 07:13.34 | thenk | buhbowww |
| 07:14.09 | thenk | ugh, i don't understand it |
| 07:14.10 | thenk | oh well |
| 07:14.15 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 07:14.20 | thenk | makes some other tweaks |
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| 07:17.05 | f2Knight | thenk, i would say if you gave me an account i could register, but well i cant see the ports open at all |
| 07:18.42 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:b4fc:83f4:af47:77b9) |
| 07:20.36 | thenk | stabs shit |
| 07:21.00 | *** join/#asterisk awk (~awk@gw1.security.web.za) |
| 07:21.46 | awk | hi guys, please can somebody help me with this... sox -V vm-intro.wav -r 8000 -c 1 -t ul -w vm-intro.ulaw that works however i'm trying to do multiple files so this is my string, for a in *.wav; do sox "$a" -r 8000 -c 1 -t ul `echo $a|sed "s/.wav/.ulaw/"` resample -ql; done |
| 07:22.31 | awk | I get this error, sox: Input and Output rates must be different to use resample effect |
| 07:22.54 | thenk | ....? |
| 07:22.57 | xbp | hi |
| 07:23.07 | thenk | input and output rates must be different to use the resample effect |
| 07:23.13 | thenk | is vm-intro.wav 8000Hz? |
| 07:23.52 | thenk | you're not using the resample effect in your original command |
| 07:24.22 | thenk | i dare say it would work fine without the resample effect. |
| 07:25.54 | awk | thenk: yes, the first command works. so the -r is correct ? |
| 07:26.02 | awk | I just don't know how to structure it without the resample |
| 07:26.54 | thenk | you mean you don't know the syntax for sox? |
| 07:27.57 | awk | I mean I don't know the correct string to use without the resample -ql added, this is what I get.. |
| 07:28.31 | awk | ahhh |
| 07:28.32 | awk | it works |
| 07:28.32 | awk | :D |
| 07:28.37 | awk | woopa |
| 07:28.44 | thenk | let me guess, you deleted resample -ql ? :P |
| 07:29.58 | awk | ja, just wasn't sure how to let the loop know when to exit. but just a ' missing, tx |
| 07:31.59 | thenk | of course the loop will exit after running out of iterations of $a |
| 07:39.15 | awk | that was my problem, it was sitting, I should verbose the output, so I can at least know it was running and it wasn't because it was missing a cappie. |
| 07:39.23 | awk | thanks thenk |
| 07:40.50 | thenk | you know |
| 07:40.53 | thenk | whenever I do that stuff |
| 07:41.02 | thenk | i put 'echo' |
| 07:41.06 | thenk | in front of whatever command it is |
| 07:41.18 | thenk | that way, when you run the first test run, it just echoes what it would be executing |
| 07:41.33 | thenk | rather than executing something that could be harmful |
| 07:41.39 | awk | good point |
| 07:41.58 | thenk | and as an added bonus, you can see exactly the command it will be giving |
| 07:42.04 | thenk | hence pick up any typos or missing spaces etc |
| 07:51.00 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
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| 08:15.29 | schmidts | good morning |
| 08:31.07 | thenk | morning |
| 08:34.58 | puzzled | morning |
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| 08:46.18 | thenk | anyone able to point me in the right direction for getting app_rtsp and app_transcoder to work? |
| 08:46.49 | wdoekes2 | did you check the source? |
| 08:47.09 | wdoekes2 | it might provide clues without you needing to understand the whole thing |
| 08:49.04 | thenk | yeah, I've checked out the source |
| 08:49.29 | thenk | it's commented, but honestly doesn't seem to make much sense to me |
| 08:50.42 | thenk | all i get out of app_rtsp even with debug on 10+, is ">rtsp_play" and "<rtsp_play" |
| 08:51.08 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 08:51.33 | thenk | ast_log(LOG_DEBUG,"xxx") |
| 08:51.39 | thenk | shouldn't taht produce debug output? |
| 08:53.05 | thenk | it makes warnings, but not debugs |
| 08:53.26 | thenk | if I replace the source with all LOG_WARNING logger level, I get a lot more output |
| 08:53.30 | thenk | but shouldn't LOG_DEBUG suffice? |
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| 09:02.58 | wdoekes2 | you need the core set debug to see debug messages |
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| 09:58.45 | creativx | meh |
| 09:59.43 | creativx | man I am fed up of choppy audio |
| 10:00.16 | creativx | I wish I knew wtf is wrong with our setup |
| 10:00.42 | shapr | What have you tried? |
| 10:01.05 | creativx | setting guaranteed bw for the asterisk box on the router |
| 10:02.30 | creativx | then I ran out of ideas |
| 10:24.59 | CRCinAU_ | yawns |
| 10:25.08 | CRCinAU_ | p3nguin: well, it all seems to be working :) |
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| 10:41.22 | KNERD | have dial patterns in 1.8 changed? |
| 10:42.48 | schmidts | KNERD not really AFAIK why do you think? |
| 10:43.24 | KNERD | i am trying 9|NXXNXXXXXX ..not functioning |
| 10:44.04 | kaldemar | what's the pipe doing in there? |
| 10:45.14 | KNERD | sorry..wrong channel |
| 10:46.41 | CRCinAU_ | btw - can someone check while I an banned under the nick CRCinAU in here? |
| 10:51.25 | kaldemar | CRCinAU_: you were annoying on November 30th 2009. |
| 11:00.26 | *** join/#asterisk as001 (~uros@194.106.175.113) |
| 11:00.41 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 11:00.43 | as001 | hi how can I check packet loss to my asterisk server with hping3 |
| 11:01.08 | as001 | I know with hping but my ubuntu has just hping3 and it seems -i -u20000 does not work well |
| 11:01.27 | no1peanut | Can I use an asterisk variable inside the shell function ? |
| 11:01.46 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 11:02.00 | as001 | yes you can no1 System(script.sh ${VARIABLE}) |
| 11:02.31 | as001 | that should pass var to script as far as I know |
| 11:03.52 | kaldemar | no1peanut: sure. |
| 11:03.57 | no1peanut | as001, not with something like Set(x=${SHELL(test -f /path/to/file/part_${va}.txt && echo -n 1 || echo -n 0}) ? |
| 11:05.29 | CRCinAU_ | kaldemar: wow - 18+ months ago? |
| 11:05.30 | no1peanut | sry , missing ) in there |
| 11:08.37 | no1peanut | kaldemar, ? |
| 11:12.40 | as001 | I don't know no1peanut. |
| 11:13.59 | as001 | but you might try to do that and do NoOp(x) as next command and see what x is after that line. |
| 11:14.12 | as001 | ${x} of course |
| 11:14.32 | *** part/#asterisk as001 (~uros@194.106.175.113) |
| 11:15.10 | no1peanut | well I get 0 even though the file exist |
| 11:17.24 | kaldemar | no1peanut: that should work fine. |
| 11:18.01 | kaldemar | show the complete extension if you want another pair of eyes looking at it. |
| 11:22.14 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ybmjnuusdehkclik) |
| 11:25.03 | no1peanut | kaldemar, thx ... Seems like my problem was in an earlier line though ! .. So here is another question .. I have this Set(ARRAY(sw1,sw2,sw3)=1\,1\,1) and a NoOp(${sw1}) in next line gives this result : 1,1,1 |
| 11:25.16 | no1peanut | ? |
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| 11:34.31 | kaldemar | no1peanut: where's the question? |
| 11:36.01 | no1peanut | kaldemar, Set(ARRAY(sw1,sw2,sw3)=1\,1\,1) and a NoOp(${sw1}) in next line gives this result : 1,1,1 -- Is there something wrong with this and why dont I just get 1 assigned to sw1? |
| 11:36.48 | no1peanut | kaldemar, I might have misunderstood the way array works |
| 11:42.29 | kaldemar | no1peanut: you're escaping commas for some reason. that causes 1,1,1 to be a single argument and it all goes to sw1. |
| 11:44.29 | no1peanut | kaldemar, I do this because Set takes its on arguments : http://www.voip-info.org/wiki/view/Asterisk+func+array |
| 11:44.37 | no1peanut | *own |
| 11:45.19 | *** join/#asterisk OldMonk (~raju@117.239.12.18) |
| 11:45.22 | OldMonk | hi |
| 11:46.22 | OldMonk | i've got sip extensions of the form xx_nnnnn where xx are alpha and nnnnn are digits |
| 11:46.40 | OldMonk | now when i cycle in chanspy the extension name isn't spoken out |
| 11:46.47 | OldMonk | any workaround/fix for that? |
| 11:47.11 | no1peanut | kaldemar, but yes - it seems to work when not backslashing the commas .. thx |
| 11:47.45 | OldMonk | asterisk 1.6.2.9 |
| 11:56.12 | kaldemar | no1peanut: that example is wrong. avoid wiki as documentation. |
| 12:01.01 | no1peanut | kaldemar, thx |
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| 12:29.03 | thenk_ | hey all |
| 12:32.45 | thenk_ | anyone able to point me in the right direction of getting app_rtsp and app_transcoder to work? |
| 12:33.03 | thenk_ | i'm trying to get it to stream some video to a caller, but it's not working properly |
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| 12:52.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:55.47 | thenk_ | early morning over in the Us still? |
| 12:56.57 | leifmadsen | well, the US ranges from -0400 GMT to -0800 GMT |
| 12:57.19 | leifmadsen | so yes, it is 7:57am -0500 GMT |
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| 12:59.17 | thenk_ | ah, might be why there's not much conversation |
| 12:59.22 | thenk_ | everyone on the way to work :P |
| 12:59.34 | thenk_ | unless of course you're stuck in traffic with an ipad |
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| 13:16.21 | thenk_ | so anyone know how to work with app_rtsp and app_transcoder? |
| 13:17.49 | lathama | exit |
| 13:18.32 | lathama | quit |
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| 13:28.15 | LemensTS | register => myuser:mypassword@host_ip_address |
| 13:28.20 | LemensTS | Is that acceptable? |
| 13:29.23 | kaldemar | LemensTS: yes |
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| 13:29.45 | LemensTS | kaldemar: When i do 'sip show registry', it is not registered. |
| 13:32.55 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:36.58 | thenk_ | are you putting it in sip.conf? |
| 13:37.28 | kaldemar | LemensTS: what is the state then? |
| 13:37.47 | kaldemar | LemensTS: is it listed at all? |
| 13:40.14 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:40.17 | LemensTS | thenk/kaldemar: Finally figured it out, it has to be in [general] section, not the context of the sip provider... |
| 13:40.56 | thenk_ | yeah |
| 13:41.07 | *** join/#asterisk weinerk (~irc@bzq-79-181-210-72.red.bezeqint.net) |
| 13:42.48 | thenk_ | so does anyone have any experience with app_rtsp / app_transcoder or video playback with Asterisk? |
| 13:44.10 | thenk_ | i'm trying to make a bit of a sip tv channel, you call and tune in to a video stream |
| 13:44.18 | thenk_ | however it's not working as intended =[ |
| 13:52.14 | no1peanut | is it possible to have nested while's in the dialplan ? |
| 13:52.47 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:53.25 | kaldemar | no1peanut: yes |
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| 14:08.06 | Sheeplet | lo all |
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| 14:13.25 | *** join/#asterisk b0ot (~comm@147.177.40.47) |
| 14:13.26 | b0ot | I used to be able to install asterisk for the ubuntu software center but it doesn't look like it is there anymore... where did it go? |
| 14:16.05 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:17.24 | SuPrSluG | b0ot:that question should be asked in the Ubuntu channel |
| 14:17.32 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 14:18.13 | kaldemar | b0ot: it should still be there in the universe repo. |
| 14:19.03 | b0ot | Well I just did a clean install of Ubuntu 10.04.2 LTS and I used to be able to just do an update then select asterisk from the Ubuntu Software Center... but this no longer works kaldemar |
| 14:19.31 | russellb | check what repos you have enabled |
| 14:20.14 | Katty | morning |
| 14:20.44 | b0ot | Well I found it in synaptic... it is 1.6 but it should work for what I need it to do |
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| 14:32.42 | n3hxs | Morning Katty |
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| 14:34.02 | *** join/#asterisk datalay (~datalay@unaffiliated/datalay) |
| 14:35.15 | datalay | i search for a telemarketing product. dou know any good alternative of mor billing? |
| 14:35.41 | dmz | what do you want it to do? |
| 14:36.36 | thenk_ | is it possible to use app_rtsp and app_transcoder together with VLC? or use app_rtsp with vlc transcoding? |
| 14:36.45 | thenk_ | to display video stream to a sip caller |
| 14:37.23 | datalay | dmz |
| 14:37.50 | *** join/#asterisk wesphillips (~wphillips@host65-16-184-66.birch.net) |
| 14:37.55 | datalay | i want to use a robot. it will call a number and will play a mp3. and it will ask press 1 for talking. if user press 1 on the phone. |
| 14:38.04 | datalay | it will redirect user an agent |
| 14:38.14 | JerJer | thenk_: sounds like a fun project... given the right situation, one should be able to pull it off |
| 14:38.25 | thenk_ | is that legal? |
| 14:38.38 | JerJer | i have done RTSP streaming with vlc, but not hooked it in to anything asterisk |
| 14:39.10 | datalay | dmz what is the key term for my problem? |
| 14:39.32 | datalay | it s like a remote IVR |
| 14:40.12 | datalay | outbound telemarketing ivr |
| 14:40.15 | *** join/#asterisk wesphillips (~wphillips@host65-16-184-66.birch.net) |
| 14:41.19 | dmz | datalay you can do that with a very simple script |
| 14:41.34 | dmz | datalay you create a .call file and drop it into the outgoing folder |
| 14:41.47 | dmz | datalay the file is created by a perl script (or whatever) that pulls from your database |
| 14:42.40 | dmz | datalay the call script will call the person then based on their input (the .call file tells it which context to drop the caller into) you program the dialplan for that context to act as you need; if they hit 1 it forwards them to a support queue, otherwise hangs up and you should probably update your database with anothe rscritp |
| 14:42.57 | *** join/#asterisk wesphillips (~wphillips@host65-16-184-66.birch.net) |
| 14:45.18 | datalay | dmz i want to do it with php |
| 14:45.21 | datalay | it s easier than perl for me. |
| 14:45.31 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:45.52 | dmz | sure you can do that, php-cli works great |
| 14:45.55 | datalay | are you using asterisk based linux dist? or core asterisk application? |
| 14:45.58 | *** join/#asterisk sourcode (~code@ppp-115-87-200-165.revip4.asianet.co.th) |
| 14:46.14 | dmz | i use debian or ubuntu w/package |
| 14:46.31 | dmz | or i'll install from source if i need newer version (which i do for our core pbx so we can tie into skype) |
| 14:46.41 | datalay | which version do you use? |
| 14:46.59 | datalay | are you using apt-get for asterisk installation? |
| 14:47.02 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:47.11 | datalay | why do you need to compile? |
| 14:47.36 | datalay | is there any important parameter for compiling from source? |
| 14:47.51 | dmz | no but the skype stuff requires a very recent version |
| 14:47.56 | dmz | and most distros are not on the latest source versoin |
| 14:48.07 | dmz | for 99.999% of the things just use a package |
| 14:48.19 | dmz | there should be php-agi libraries, google will help you here |
| 14:48.29 | dmz | search for asterisk agi php outgoing call |
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| 15:11.34 | E-bola | do anybody know how i can set a variable with space in the string via agi? |
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| 15:35.15 | SunTsu | strange, INVITE auth looks nearly the same done from as it looks done from my snom phone, except that the snom phone sends an additional "user=phone" with it |
| 15:35.52 | SunTsu | still, the snom phone is able to place outgoing calls, while asterisk is not |
| 15:36.00 | *** join/#asterisk wesphillips (~wphillips@host65-16-184-66.birch.net) |
| 15:36.43 | WIMPy | IIRC there is an option to add 'user=phone'. |
| 15:37.46 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:38.31 | SunTsu | ah, usereqphone |
| 15:38.35 | SunTsu | I'll try that |
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| 15:41.15 | SunTsu | no difference, damn it |
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| 15:46.21 | Lantizia | Hey are the prompts that asterisk installs in the asterisk tarball or are they in this "asterisk-sounds-1.2.1.tar.gz" file I can see online (and I presume the asterisk install just wget's it) |
| 15:46.43 | WIMPy | The wget part. |
| 15:47.17 | Lantizia | do you know what it wget's? |
| 15:47.41 | WIMPy | What you select in 'make menuconfig'. |
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| 15:48.09 | Lantizia | WIMPy, the menuconfig lists URL's !? |
| 15:48.33 | WIMPy | No, the packages. |
| 15:48.48 | Lantizia | right but obviously it downloads a file... what file |
| 15:49.29 | kaldemar | Lantizia: http://downloads.asterisk.org/pub/telephony/sounds/releases/ |
| 15:50.10 | Lantizia | aha! thought I was looking in the wrong place... so what is that asterisk-sounds file for in the asterisk/releases dir? |
| 15:50.20 | Lantizia | or is it just not used not (given it's dated 2006) |
| 15:50.24 | kaldemar | Lantizia: see sounds/Makefile for what your asterisk downloads. |
| 15:50.25 | Lantizia | *now |
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| 16:00.16 | SunTsu | er, wait. the "Failed to authenticate on INVITE" does not come from QSC. It's from asterisk. wth? |
| 16:01.20 | SunTsu | asterisk tells qsc "SIP/2.0 503 Service Unavailable" - er, something is going really really wrong |
| 16:04.01 | *** join/#asterisk zooz (~zooz@host86-161-164-92.range86-161.btcentralplus.com) |
| 16:05.30 | zooz | I have a context [int-phones], I have a number of local extensions configured for internal phones etc, some of those phones need to be able to make outgoing calls, how can I specify which extensions can make outgoing calls via certain outside numbers? |
| 16:06.02 | WIMPy | Put them in different contexts. |
| 16:06.22 | zooz | but then they will not be able to call each other |
| 16:06.43 | WIMPy | You can include contexts in to others. |
| 16:08.32 | zooz | if I include [outgoing-calls-context] to say [priv-ext-context] and then include [priv-ext-context] to [int-phones], will it not get nested? |
| 16:09.14 | zooz | question: will [int-phones] not be able to use [outgoing-calls-context] ? |
| 16:09.14 | WIMPy | yes |
| 16:09.55 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 16:10.05 | WIMPy | You need to get your contexts sorted correctly. |
| 16:10.05 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 16:10.34 | WIMPy | That's probably the most importantn part of Asterisk configuration. |
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| 16:14.54 | SunTsu | WIMPy: I think I understood what's going on. It's not asterisk's fault. calls are requested as number@sip.qsc.de but INVITE returns as number@qsc.de - which the phone turns down |
| 16:15.33 | zooz | WIMPy, http://pastebin.com/NDuNS0c9 - will this work? |
| 16:15.46 | SunTsu | normally "broken registrar" setting should fix that, but somehow it does not |
| 16:16.10 | zooz | as far as I understand, [int-phones] will be able to make outgoing calls, right? |
| 16:16.26 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 16:19.34 | WIMPy | zooz: They will be able to use both priv-extension-context and out-calls, yes. |
| 16:20.19 | zooz | WIMPy, so how would it be correct? |
| 16:22.36 | WIMPy | zooz: What do you want to do? |
| 16:23.14 | WIMPy | And if you can describe it, I guess you should be able to arrange your includes accordingly :-) |
| 16:23.41 | WIMPy | As a hint: Always include contexts with less rights in to those with more. |
| 16:23.42 | *** join/#asterisk Tim_Toady (~moi@178.128.4.246.dsl.dyn.forthnet.gr) |
| 16:23.54 | zooz | I want to allow specific extensions to make outgoing calls from certain numbers and also want to make sure every internal extension can calls each other |
| 16:24.53 | zooz | WIMPy, I think I got it now :-) |
| 16:24.55 | no1peanut | Having a situation I cant really think of a solution to: I need a menu system where users can press 1-9 to play different wavefiles, * to leave to top menu and # to proceed in menu. I have Read(key,${var},1) doing this right now but might not be right approach. My problem is I want to ignore if 0 is pressed...(i.e I dont want the current wav to be interrupted) - Any ideas ? |
| 16:26.03 | WIMPy | no1peanut: Read and # doesn't work. Maybe you should go the WaitExten way. |
| 16:26.41 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 16:27.03 | no1peanut | WIMPy, I actually got that part working ($ [ "${LEN($key)}" == "0"]) |
| 16:27.48 | no1peanut | My problem is just that I cant ignore certain keys (in this case 0) |
| 16:27.54 | WIMPy | It could also have been a timeout. |
| 16:29.53 | WIMPy | I think I remember reading a feature request or maybe even a proposed patch to give a set of valid digits to Read. |
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| 16:35.14 | no1peanut | WIMPy, that would be nice. I will go looking |
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| 16:57.46 | gpled | does it seem easy to learn asterisk with out the web interface at first? |
| 16:58.57 | krion | gpled: you have too anyway |
| 16:59.06 | leifmadsen | no web interfaces existed when I learned Asterisk :) |
| 16:59.18 | leifmadsen | it's easier if you follow some documentation that steps you through how to do things... |
| 16:59.38 | neurosys | read his ^^^ book! It is the perfect intorduction |
| 16:59.45 | neurosys | atleast IMHO. |
| 16:59.46 | leifmadsen | no1peanut: Don't use == for comparisons in Asterisk |
| 17:00.08 | leifmadsen | no1peanut: you want $["${LEN(${key})}" = "0"] |
| 17:01.37 | gpled | so i installed with the 1 option. FreePBX. can i just shut down a service to make it back to pure asterisk? if it would be better to re-install, that is an option at this time. |
| 17:01.37 | no1peanut | leifmadsen, Ok ..thx |
| 17:02.04 | leifmadsen | with Read() you're setting a variable named 'key' which needs to be accessed via ${key} |
| 17:02.20 | leifmadsen | also the == would not work |
| 17:02.24 | leifmadsen | goes to get lunch |
| 17:02.43 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 17:02.53 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f2-69-94-238-26.totalprocess.net) |
| 17:03.26 | Letoric | leifmadsen are you available? I was hoping you might be able to offer some help on an issue you identified as an OS issue, but the CentOS guys aren't offering me much help on. |
| 17:04.10 | Letoric | When launching the asterisk service on CentOS, I receive an error: Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted |
| 17:04.24 | Letoric | I saw a bugtrack on it, and you closed it saying it is an OS issue. |
| 17:04.45 | Letoric | This is with version 1.6.2.13 of Asterisk, but I believe the bugtrack was for 1.8.0 |
| 17:05.04 | *** part/#asterisk gpled (~gpled@69.1.105.42) |
| 17:05.47 | Letoric | If anybody else knows how to resolve this, it would be appreciated, in case leif is busy/unavailable |
| 17:07.03 | pabelanger | Letoric: are you running the script as root? |
| 17:07.18 | mihai__ | tried this: http://gnufreakz.wordpress.com/2009/08/12/increase-ulimit-in-centos/? |
| 17:07.18 | Letoric | yes |
| 17:07.29 | _Corey_ | Letoric: Do you have SELinux enabled? |
| 17:07.33 | Letoric | nosir |
| 17:07.41 | Letoric | it is currently disabled |
| 17:08.06 | pabelanger | Letoric: you're best to google it, since it is an OS issue, not Asterisk |
| 17:08.21 | Letoric | I did, the google searches mostly returned bugtrack's |
| 17:08.22 | Letoric | :/ |
| 17:08.39 | Letoric | I'll continue researching it, just hoped somebody here had also run into it and could help |
| 17:08.48 | Letoric | it seems to be more than just me, seeing it |
| 17:09.23 | pabelanger | http://www.chrissearle.org/node/419 |
| 17:09.46 | pabelanger | It's Debian, but you get the idea |
| 17:11.00 | Letoric | ah hah, thank you |
| 17:11.02 | Letoric | I will give that a try |
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| 17:18.02 | drift- | f |
| 17:18.04 | drift- | fo0 |
| 17:19.42 | *** join/#asterisk Blackthorn (Blackthorn@76.77.160.10) |
| 17:21.03 | Blackthorn | if the DID number is dialed and I do not have it defined in my extention.conf file I'm creating a loop to my voip provider. is there a way to drop/ignor/out of service any undefned DID number so that I don't have to define all 100 DID numbers? |
| 17:21.43 | leifmadsen | Blackthorn: it's called "pattern matching" |
| 17:22.48 | WIMPy | Blackthorn: You should get your contexts sorted. Might be a bad idea to allow outgoing calls from external. |
| 17:23.17 | Blackthorn | umm |
| 17:23.22 | Blackthorn | ok can do that |
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| 17:28.01 | Letoric | pabelanger: I tried that fix, and it doesn't resolve the issue. I still receive the same error |
| 17:28.12 | Letoric | pabelanger: Is there a suggested quantity that asterisk needs? |
| 17:28.27 | *** join/#asterisk Deetz (~PaulDittr@174.4.128.244) |
| 17:29.09 | pabelanger | unlimited? |
| 17:29.22 | pabelanger | mostly used when generating a coredump |
| 17:29.28 | Letoric | I see |
| 17:29.58 | Letoric | how can you determine the max files open on a linux system? I looked at the safe_asterisk script, and it says on linux systems, it tries to set it to max / 2 |
| 17:30.20 | Letoric | I know ulimit -n tells me for my user, but is there an easy way to determine the max? |
| 17:30.24 | *** join/#asterisk ghchowdhury (cdd3a902@gateway/web/freenode/ip.205.211.169.2) |
| 17:32.21 | Letoric | nevermind |
| 17:32.31 | Letoric | I see the script section that checks for it, I'll see if I can work through it |
| 17:33.39 | ghchowdhury | I am going to send out automated reminder to our club members. I would like to know which Digium card will be best. I need a pstn Digium card that will relay message only after some one answers the call. I am new so please point me to the right direction. |
| 17:34.44 | leifmadsen | ghchowdhury: analog or digital circuit? North America or Europe? |
| 17:34.54 | leifmadsen | that's not really a functionality of the card that you're describing |
| 17:35.09 | leifmadsen | your dialplan will determine what to do at answer |
| 17:35.12 | ghchowdhury | leifmadsen: Analog on North America |
| 17:35.24 | drift- | anyone have a suggestion for this problem... i have this phone rebooting non stop poycom saying error loading blablabla.cfg |
| 17:35.48 | leifmadsen | ghchowdhury: without signalling on the line it won't be possible to know when the other end picks up the phone |
| 17:35.57 | leifmadsen | (which is what digital circuits are for) |
| 17:36.35 | _Corey_ | drift-: Sounds like a bad/incompatible .cfg file |
| 17:36.38 | ghchowdhury | leifmadsen: when you say digital circuits do you mean broadband line? |
| 17:36.48 | _Corey_ | drift-: You might try a fresh config |
| 17:36.59 | drift- | well... how would i do this? |
| 17:37.03 | Defraz | I have been playing with an Adtran 900e and trying to have an incoming PRI into it then SIP trunk that to my ASterisk server. Does anyone have any example config files to accomplish this? |
| 17:37.08 | drift- | this phone just keeps rebooting :( |
| 17:38.09 | _Corey_ | drift-: Yeah, they'll do that if the config is really messed up or incompatible with the loader on the phone |
| 17:38.36 | drift- | so whats esiest way to get it to be blank |
| 17:38.39 | drift- | and start fresh |
| 17:39.39 | _Corey_ | Download a fresh firmware from Polycom and build a new config from the included files |
| 17:39.57 | _Corey_ | (keep in mind the compatibility matrix if you're using older handsets) |
| 17:39.57 | ghchowdhury | leifmadsen: can you recommand any Digium card for digital circuit in North Ameria |
| 17:41.03 | leifmadsen | ghchowdhury: I mean a circuit provided via a T1 |
| 17:41.12 | leifmadsen | (typically configured for PRI signalling) |
| 17:41.26 | leifmadsen | otherwise, use a SIP provider |
| 17:48.10 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 18:14.34 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 18:14.45 | Romeo- | hellp |
| 18:14.48 | Romeo- | hello |
| 18:15.16 | Romeo- | i have install asterisk on deb squezze, but i can't get on the gui |
| 18:15.53 | Romeo- | i have to activate them on /etc/asterisk/modules.conf |
| 18:16.43 | Romeo- | manager.conf |
| 18:20.48 | Romeo- | i get this on my browser |
| 18:20.49 | Romeo- | <PROTECTED> |
| 18:20.49 | Romeo- | <PROTECTED> |
| 18:20.58 | Romeo- | Asterisk Call Manager/1.1 |
| 18:20.59 | Romeo- | Response: Error |
| 18:20.59 | Romeo- | Message: Missing action in request |
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| 18:24.40 | no1peanut | this just ticked in on my twitter: might be of interest to you all (Hope I dont get banned for spamming or anything like that!): http://www.backtrack-linux.org/wiki/index.php/Pentesting_VOIP |
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| 18:25.48 | *** mode/#asterisk [+o BMJ] by ChanServ |
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| 18:47.46 | hardwire | I'm guessing that asterisk doesn't expire sip peers from memory during a reload if there is a chance some dialogs are still open (as to not have a problem later on) |
| 18:47.52 | hardwire | but then never does garbage collection. |
| 18:48.03 | *** join/#asterisk wesphillips (~wphillips@host65-16-184-66.birch.net) |
| 18:49.06 | hardwire | and the config parser attempts to match up peers in memory based on host in order to avoid problems with "who's on first" |
| 18:49.13 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 18:49.18 | hardwire | other than restarting asterisk.. I'm not sure how to resolve this problem. |
| 18:50.02 | Letoric | pabelanger: FYI, regardless of what I set the limit to, safe_asterisk still generates the error because it is unable to set the limit itself. Shall I assume this is a non critical issue and not worry about it, since I've set the limit manually? |
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| 18:52.19 | leafartn | hi |
| 18:57.17 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 18:57.34 | leafartn | hi, i'm in troubles with the connection to mysql in asterisk 1.8, i enabled the queue_log but in CLI shows me an error |
| 18:57.53 | leafartn | WARNING[2837]: config.c:2045 find_engine: Realtime mapping for 'queue_log' found to engine 'mysql', but the engine is not available |
| 18:58.29 | leafartn | maybe i'm missing something in the configuration? thanks! |
| 19:07.41 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 19:07.44 | pabelanger | leafartn: drop mysql and use ODBC :) |
| 19:16.42 | JerJer | yawns |
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| 19:33.50 | leafartn | anyone? |
| 19:36.41 | WIMPy | leafartn: What pabelanger said |
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| 20:12.09 | kaldemar | leafartn: you dont have the msql config module from the addons |
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| 20:26.13 | jake1905 | i have an unknown person connecting to my asterisk server via SIP but i do not see the registration |
| 20:27.22 | jake1905 | the address doesnt show in sip show peers |
| 20:27.22 | jake1905 | and the sip channel is an IP address which isn't defined in my sip.conf |
| 20:27.22 | jake1905 | [Mar 7 16:15:41] VERBOSE[19821] netsock.c: == Using UDPTL CoS mark 5 |
| 20:27.22 | jake1905 | [Mar 7 16:15:41] VERBOSE[19821] netsock2.c: == Using SIP RTP CoS mark 5 |
| 20:27.22 | jake1905 | [Mar 7 16:15:41] VERBOSE[19873] pbx.c: -- Executing [011442073479999@intern:1] NoOp("SIP/91.220.62.75-0000000f", "") in new stack |
| 20:27.22 | jake1905 | that's what I am seeing |
| 20:27.22 | jake1905 | can anybody help? |
| 20:27.23 | *** join/#asterisk tvc123 (~tcameron@host-148.pl1071220.fiber.net) |
| 20:29.57 | tvc123 | Stupid question I have done a set core debug 9 but when I have a call same => n,Verbose(3,Test) nothing shows up int the CLI |
| 20:30.01 | tvc123 | any ideas? |
| 20:30.23 | tvc123 | I know it is going through it because the things happen that are below it |
| 20:31.25 | *** join/#asterisk efra (~euribe@189.143.37.44) |
| 20:32.32 | efra | Does anyone knows if it is posible to connect a nbx v5000 to an asterisk box? |
| 20:32.59 | efra | Wow, such manners |
| 20:33.02 | efra | Sorry |
| 20:33.14 | efra | Hello to all |
| 20:33.22 | tvc123 | me? sorry if I offended |
| 20:33.23 | leifmadsen | if you can make it connect via SIP or some other method, sure |
| 20:33.48 | leifmadsen | tvc123: lets see the whole snippet of dialplan in a pastebin |
| 20:33.59 | _Corey_ | tvc123: debug and verbose are different |
| 20:34.00 | leifmadsen | suspects a typo or inappropriate use |
| 20:34.11 | leifmadsen | oh ya.. that :) |
| 20:34.19 | _Corey_ | :) |
| 20:34.22 | leifmadsen | I didn't even read 'core show debug' -- I read it as 'core show version' |
| 20:34.25 | leifmadsen | verbose even |
| 20:34.34 | leifmadsen | so ya -- inappropriate use :D |
| 20:36.15 | tvc123 | _Corey_: thanks! that fixes things :) |
| 20:37.48 | efra | I'm afraid is not posible via SIP. I've tried. But I heard someone could connect them so i tought i could ask |
| 20:38.26 | leifmadsen | well you can connect via any method that the other end supports (cross-over T1 cable, analog ports, etc...) |
| 20:38.36 | leifmadsen | I don't know what methods of connection that system supports |
| 20:38.57 | efra | right, just a question of trying |
| 20:39.35 | jake1905 | anybody have an idea? |
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| 21:00.08 | benngard | is it a stupid idea to have a whish for a timer/timeout parameter to PauseQueueMember, u pass say 300 to the dialplan application and after 300 sec the member will be unpaused |
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| 21:24.22 | Schreiber1337 | When incoming callers are in the IRV and they dial an extension number, often the number is getting detected twice... IE- 3160 is getting 31160 in the CLI... anyone have an idea where this threshold is set? |
| 21:24.52 | xdvst8x | Can someone help me with my ADA config... i have it working, but when i dial a number, my phone rings.. as soon as i pick it up.. it hangs up |
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| 21:47.25 | LemensTS | whats it called when incoming calls can call in unauthenticated |
| 21:47.37 | LemensTS | Im trying to disable it, i forget what its called |
| 21:47.50 | WIMPy | allowguests |
| 21:48.23 | LemensTS | thanks |
| 21:51.50 | ccr_ | I just upgraded from 1.4 to 1.8 for gvoice support. the dialplan included a System() call which checked a file to see if i was home or not, and if not home, don't ring those home phones. was there a change in how the n+101 stuff worked in 1.8? priorityjumping is enabled |
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| 22:07.23 | leafartn | how can i see or enable the statistics of a queue in asterisk 1.8? i would like to know the actual service level of my queue |
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| 22:13.29 | xdvst8x | how can i have the phone "ring" while using Asterisk Desktop Assistant while its connecting to an outside line |
| 22:17.56 | tvc123 | anyone know off hand if "NV" line detect still exists |
| 22:17.57 | tvc123 | ? |
| 22:25.49 | CRCinAU_ | p3nguin: I just converted my 7961 to sccp as well :) |
| 22:28.30 | leafartn | anyone knows how can i see or enable the statistics of a queue in asterisk 1.8? i would like to know the actual service level of my queue... |
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| 22:52.57 | rectifiertechnol | Hi all, I have a question about asterisk inbound routes - is anyone able to help ? |
| 22:53.58 | WIMPy | ~ask |
| 22:53.58 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:54.10 | WIMPy | and try to use comprehensible terminology |
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| 22:54.35 | rectifiertechnol | ok well i asked my question yesterday and i was ignored so i will try again |
| 22:56.16 | rectifiertechnol | can i get asterisk to call a SIP extension, then call another number, after a call is received via an inbound route ? |
| 22:57.39 | WIMPy | Sure. You can do as much as you like. |
| 22:58.10 | WIMPy | However there's no such thing as a 'sip extension'. |
| 22:58.54 | rectifiertechnol | really - in freepbx there is an option to add a Sip Extension |
| 22:59.18 | WIMPy | You should take that to #freepbx then. |
| 22:59.22 | IsUp | if you are using freepbx, better to check #freepbx |
| 22:59.32 | rectifiertechnol | ok thanks |
| 22:59.56 | IsUp | hey WIMPy |
| 23:00.11 | WIMPy | jo |
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| 23:02.07 | rectifiertechnol | hugs the entire room |
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