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| 00:39.12 | CRCinAU_ | So... ANyone from Digium here that I can point an email to? :| |
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| 00:44.39 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 00:46.27 | Xedecimal | consider me fairly a noob, we've got two sipgate sip accounts running great in asterisk (no freepbx, using the asterisk-gui thingey) but gizmo will not register, not even if I add a manual account, am I an idiot or is gizmo5 really just not liking asterisk anymore ? |
| 00:48.40 | jaskew | I managed to get gizmo registered. I remember that I needed to mess with it a bit. can you share your config (e.g. pastebin) |
| 00:49.05 | Xedecimal | I'm using a gui for configuration actually, that'd probably be my problem right? I'll need more than just the register => string right ? |
| 00:49.31 | Xedecimal | I'm guessing the gui stores it all in extensions.conf ? |
| 00:49.36 | *** join/#asterisk cesar_CR (i=cesar@201.201.41.242) |
| 00:49.55 | CRCinAU_ | register lines go in sip.conf |
| 00:50.02 | voipmonk | has nothing nice to say.... |
| 00:50.04 | jaskew | I'm kinda new myself. I used freePBX. I have a register statement as well as a [gizmo] section in sip.conf. |
| 00:50.05 | voipmonk | ... |
| 00:50.24 | Xedecimal | oh, I was afraid that I'd end up with problems with both, I added both |
| 00:50.50 | Xedecimal | I did something like register => gizmonumber:password@proxy01.sipphone.com/gizmonumber |
| 00:51.25 | Xedecimal | it attempts apparently to do the same thing as the gui tries to do, except in the gui I can set 'insecure' to 'very' which I can't ever seem to pull off in the configs cause I don't know where I'd have my little [gizmo] area, contexts are pretty confusing |
| 00:52.10 | jaskew | my register is basically the same, except I didn't put the "/gizmonumber" at the end. |
| 00:52.30 | Xedecimal | I'll yank that and yank my gui settings and pop into asterisk to see what it says on a rehash |
| 00:52.34 | jaskew | If you want to make calls, I'd think you would need the [gizmo] section though |
| 00:52.49 | CRCinAU_ | jaskew: correct. |
| 00:53.52 | jaskew | and some kind of routing... |
| 00:54.09 | Xedecimal | in my display it just says 'Unregistered' all the time |
| 00:54.21 | Xedecimal | added my register => line to my sip.conf, trying to remember how to restart this sucker |
| 00:54.33 | Xedecimal | "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 00:54.39 | Xedecimal | I'm guessing not, lol |
| 00:55.04 | jaskew | service asterisk start?? |
| 00:55.12 | Xedecimal | wow, that's waesome |
| 00:55.20 | Xedecimal | asterisk ended with code 1 |
| 00:55.24 | Xedecimal | lovely |
| 00:55.24 | Xedecimal | ! |
| 00:55.39 | Xedecimal | that was due to it already running, that's no biggy |
| 00:55.48 | Xedecimal | restarted better now! |
| 00:55.50 | jaskew | oh - I should have said: service asterisk restart |
| 00:56.07 | Xedecimal | gui sees 2 trunks, checking logs... |
| 00:56.19 | jaskew | although that is an "unelgant" shutdown. CAlls could be cut off |
| 00:56.40 | Xedecimal | we don't have some corporate office or anything, just an at home hobby |
| 00:56.48 | Xedecimal | we tear it apart every night trying to figure it all out |
| 00:56.56 | Xedecimal | well, friday to saturday |
| 00:57.12 | Xedecimal | it suggests that it tried to register |
| 00:57.51 | Xedecimal | it never seems ot tell me if it actually did register though |
| 00:57.59 | Xedecimal | isn't there an asterisk console command I can use? |
| 00:58.11 | Xedecimal | like asterisk -r, then show me my trunks mofo! |
| 00:58.50 | jaskew | sip show registry |
| 00:59.04 | Xedecimal | wow, it's registered!!! |
| 00:59.20 | *** join/#asterisk TJNII (n=TJNII@207.189.199.62) |
| 00:59.41 | Xedecimal | lets see what google does when trying to call it |
| 00:59.52 | jaskew | so if someone calls you, gizmo is probably sending you the call |
| 01:00.09 | jaskew | but since you don;t have a peer defined, it'll get rejected (I think) |
| 01:00.29 | jaskew | Or maybe a "THe number you have dialed..." message |
| 01:00.46 | CRCinAU_ | jaskew: afaik, if you don't have a peer setup, you can't make outgoing calls. |
| 01:00.59 | jaskew | yeah - that too! |
| 01:01.05 | CRCinAU_ | however if you set up the dialplan and have it register etc, you can receive cals |
| 01:01.21 | Xedecimal | I have 's' incoming bound to user 6000 which links to the phone that I have in this room, as well as the house |
| 01:01.35 | CRCinAU_ | assuming say you have all your incoming lines going to one context in extensions.conf and you have extensions.conf set up :) |
| 01:01.40 | Xedecimal | but, when it gets rejected, when I'm sitting in the asterisk console, it'll say like "call to non-existent extension blah blah" right ? |
| 01:01.49 | jaskew | Right - on mine it comes into the sip_external context. |
| 01:01.56 | Xedecimal | awesome |
| 01:02.05 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 01:02.07 | jaskew | sorry - was responding to an earlier post |
| 01:02.11 | CRCinAU_ | use something like: register => gizmonumber:password@proxy01.sipphone.com/s |
| 01:02.19 | jaskew | Xdecimal: I don't know the answer to your question |
| 01:02.29 | CRCinAU_ | then as long as it's in the right context, it should be happy |
| 01:02.47 | Xedecimal | oh, if I add the /s it'll go to the locally defined extension as that name ? |
| 01:02.52 | CRCinAU_ | I can't remember what it uses as the default if there is no /anything at the end of a register lol |
| 01:02.58 | Xedecimal | and if I added /Gizmo5 and I defined an incoming context called 'Gizmo5' it'd come to that ? |
| 01:03.06 | CRCinAU_ | Xedecimal: as long as the context is correct, yes. |
| 01:03.06 | jaskew | If there is no "/" then it does go to "s" |
| 01:03.18 | jaskew | If you put "/101" then it will go to extn 101 |
| 01:03.20 | Xedecimal | wow, learning tons of crap from you guys already |
| 01:04.01 | jaskew | Xdedecimal: contexts are different than extensions |
| 01:04.14 | Xedecimal | contexts are ini sections inside the extenions.conf right ? |
| 01:04.19 | jaskew | yes |
| 01:04.28 | Xedecimal | like a pseudo box for storing configuration settings that can be referenced to later? |
| 01:04.44 | jaskew | uh... |
| 01:04.45 | CRCinAU_ | but you need to set the default context in sip.conf to have a clue on where it's going to go. |
| 01:05.12 | Xedecimal | I just made a call to my sipgate account and it got picked up with some errors here, so it's listening on that for sure, but incoming on gizmo seems quiet |
| 01:06.01 | Xedecimal | I know gizmo gets it though... Cause I get 2 friend requests on my gmail account for a VOIP2Gizmo weird thing and it says the call lasted 0 minutes |
| 01:06.07 | CRCinAU_ | have a look at sip.conf. |
| 01:06.12 | Xedecimal | checking |
| 01:06.15 | CRCinAU_ | a line kinda like: context=incoming ; Default context for incoming calls |
| 01:06.19 | *** join/#asterisk manxpower (n=ewieling@60.sub-70-223-125.myvzw.com) |
| 01:06.30 | jaskew | Xdecimal: Look in /etc/adterisk/sip.conf and see if you have a "context=" line. |
| 01:06.43 | jaskew | what CRCinAU said... |
| 01:06.48 | ruben23 | hi |
| 01:06.55 | jaskew | look in the [general] section |
| 01:07.24 | Xedecimal | I don't even have a context for gizmo |
| 01:07.29 | CRCinAU_ | whatever that context is, is where you need to put your exten => s,1,NoOp(rest of stuff here) |
| 01:07.37 | CRCinAU_ | Xedecimal: you want the DEFAULT context |
| 01:07.44 | CRCinAU_ | not the context for Gizmo. |
| 01:07.49 | Xedecimal | ok, hunting for that |
| 01:10.07 | Xedecimal | I try a grep '^\[' sip.conf and I get the most basic is [general], but [authentication], [basic-options](!), [natted-phone](!,basic-options) and some more, from what it looks like, a 'natted-phone' context is meant for firewall name address translation and the weird exclamation point after the section header suggests extending an existing context |
| 01:10.27 | Xedecimal | but nothing looks like my 'default' context |
| 01:10.38 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 01:10.42 | jaskew | It might be in an included file... |
| 01:10.45 | Xedecimal | I'm guessing this crazy gui setup is what defined it somewhere else, would it literally be called [default] ? |
| 01:11.04 | CRCinAU_ | or whatever is generating your configs is 'elegant' but not practical ;) |
| 01:11.08 | Xedecimal | extenions.conf has [default] among like 10 other files |
| 01:11.27 | jaskew | what is underneath [general] in sip.conf? |
| 01:11.33 | Xedecimal | musiconhold, osp, voicemail, followme |
| 01:11.40 | Xedecimal | context=default! |
| 01:11.47 | Xedecimal | that's under [general] |
| 01:11.56 | CRCinAU_ | ok, so that's your default context. |
| 01:11.58 | Xedecimal | but I don't have a [default] in there! |
| 01:12.15 | Xedecimal | I do in a lot of other files... |
| 01:12.29 | CRCinAU_ | so in extensions.conf, you'll need [default] \n exten => s,1,........ |
| 01:12.54 | CRCinAU_ | btw: the \n just means new line in IRC because I don't want to spam multiple lines..... |
| 01:13.10 | Xedecimal | I was guessing that |
| 01:13.30 | Xedecimal | ok, everything in extenions.conf is commented out under [default] so we have absolutely no default extenions |
| 01:14.23 | Xedecimal | ok, s,1 (is this my user extenion to local phone? (6000) ?),NoOp(name_of_my_ext_here_right?) |
| 01:15.25 | CRCinAU_ | something like: exten => s,1,NoOp(We got a call!) \n exten => s,n,Dial(SIP/someone) \n exten => s,n,Hangup |
| 01:16.25 | Xedecimal | wow, I'm gonna screw that up for sure, is there just simply a better gui than this 'digium branded asterisk-gui' ? |
| 01:16.46 | CRCinAU_ | vim? ;) |
| 01:16.49 | Xedecimal | lol |
| 01:17.01 | Xedecimal | roomate loaded a ton of demo stuff on here too |
| 01:17.18 | Xedecimal | there's probably near 500 extenions that aren't commented out in extensions.conf |
| 01:17.40 | Xedecimal | all pointing to contexts that go here and support a service that we don't have there and spagetti off everywhere |
| 01:18.05 | Xedecimal | I'm guessing it'd be a bad idea to just delete the configs to start from scratch, lol |
| 01:18.30 | Xedecimal | there's even errors popping up in the asterisk gui suggesting that some weird experimental module is taking incoming sipgate calls and can't handle it with strange urls |
| 01:18.46 | Xedecimal | actually, I'm gonna wait for him to get home and talk to him about just re-installing the whole thing with a fresh blank slate and no demo crap |
| 01:18.51 | jaskew | FWIW, I loaded up PBX in a Flash. It has freePBX bundled. Someday I'll do a bare metal install, but I needed this working fast. |
| 01:19.07 | Xedecimal | FreePBX you mean ? |
| 01:19.14 | Xedecimal | or was it called 'PBX in a Flash' ? |
| 01:19.20 | jaskew | www.pbxinaflash.com |
| 01:19.43 | Xedecimal | ahh, no can do, would love to, but this is our router box, we run clearos and crap on it too |
| 01:20.00 | jaskew | haha - yeah - don;t fdo that. |
| 01:20.27 | *** join/#asterisk aidinb (n=Aidin@166.134.255.41) |
| 01:21.41 | jaskew | did you complile from source? |
| 01:21.50 | Xedecimal | I'm not sure how he installed it, I'm pretty sure he built it |
| 01:21.58 | Xedecimal | I was off working on our nas and media server |
| 01:22.06 | Xedecimal | cause pretty pictures are more fun |
| 01:22.26 | jaskew | I think he could wipe all of the installed files and then do a "make install" to get everything back to the beginning |
| 01:22.41 | Xedecimal | I'm not sure how much work he's put in it either, or I could just wipe that sucker out |
| 01:22.51 | Xedecimal | he might have put in some hacks or something that he wants to save |
| 01:23.09 | jaskew | I thin kall the fancy samples come in when you "make samples" or something like that (maybe exten.conf.sample...) |
| 01:23.59 | Xedecimal | samples suck, lol |
| 01:24.43 | jaskew | just wait till you find the monkeys |
| 01:25.04 | CRCinAU_ | lol |
| 01:25.18 | CRCinAU_ | the monkeys have invaded Digium I think. |
| 01:26.44 | CRCinAU_ | you know - whenever I see this 'service blah restart' I still end up doing /etc/init.d/blah restart |
| 01:26.53 | jaskew | I wonder what was going through Allison's mind when she got the request for those recordings |
| 01:26.57 | CRCinAU_ | from before this 'service' command was around lol |
| 01:27.13 | CRCinAU_ | jaskew: paycheck ;) |
| 01:27.35 | jaskew | in advance... |
| 01:28.47 | jaskew | I gotta hand it to Mark though. It's not everyone who can call up a girl and ask her to make monkey sounds - adn get her to do it... |
| 01:28.52 | *** join/#asterisk ManxPower-work (n=EWieling@60.sub-70-223-125.myvzw.com) |
| 01:29.33 | *** part/#asterisk ManxPower-work (n=EWieling@60.sub-70-223-125.myvzw.com) |
| 01:30.08 | *** join/#asterisk devmod (n=devmod@76.100.208.204) |
| 01:31.09 | Xedecimal | oh my gosh |
| 01:31.11 | Xedecimal | I found my problem |
| 01:31.25 | Xedecimal | in gizmo themselves, I did not have 'SIP' as my forward, I had 'Google' |
| 01:31.46 | Xedecimal | so it was forwarding over to google voice, rather than to the sip box, but it was also coming from google, sounds like a loop |
| 01:32.25 | CRCinAU_ | jaskew: many girls would make monkey sounds if the check was good enough |
| 01:33.59 | Xedecimal | calls are now coming from gizmo in to asterisk!!! |
| 01:34.12 | *** join/#asterisk ManxPower-work (n=EWieling@60.sub-70-223-125.myvzw.com) |
| 01:34.26 | Xedecimal | I'm going to scream like a girl, maybe named 'Allison' |
| 01:34.36 | CRCinAU_ | record it! ;) |
| 01:37.12 | CRCinAU_ | yay - just updated to 1.6.2.1 |
| 01:38.12 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 01:42.37 | jaskew | CRC: ha! Xdecimal: glad you got it working! |
| 01:42.46 | jaskew | gtg - dinner call here. |
| 01:42.49 | Xedecimal | I just need to get the phone to ring, lol |
| 01:43.20 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 01:43.33 | *** part/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 01:54.35 | *** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 01:55.34 | titter | Anyone familar with this awful Shoretel crap ... I got my Asterisk box to register with Shoretel ... but the trunk always comes back as congested. |
| 01:56.26 | *** part/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 01:57.55 | Xedecimal | outgoing even works now, except incoming still fails due to "Call from '17472729388' to extension '6000' rejected because extension not found." |
| 01:58.00 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 01:58.24 | Xedecimal | wayyy down at the bottom of my extensions.conf I have 'exten = 17472729388,1,Goto(default,6000,1)' |
| 01:59.01 | Xedecimal | changed first arg to 's' |
| 01:59.19 | Xedecimal | it's as if it's suggesting that 6000 doesn't exist, but it doooooes! |
| 02:04.56 | *** part/#asterisk lanning (n=lanning@208.87.235.224) |
| 02:04.58 | CRCinAU_ | Xedecimal: probably in a different context - therefore it doesn't exist :) |
| 02:05.23 | Xedecimal | ok, let me see if I can walk through thsi whole thing |
| 02:06.08 | CRCinAU_ | Goto(default,6000,1) means: GO to the context "default" and in extension "6000" enter at step 1. |
| 02:06.52 | Xedecimal | and Goto(default|6000|1) is the same just with pipes instead of commas ? |
| 02:06.53 | *** join/#asterisk Echostar (n=echostar@142.167.143.29) |
| 02:07.14 | CRCinAU_ | yeah - but I'm pretty sure you shouldn't use pipes anymore in favour of commas |
| 02:07.30 | Xedecimal | this gui does... I'm writing by hand now though mind you! |
| 02:07.45 | Xedecimal | ok, I do have [default] in exteions.conf but it's alllll commented out, I'm guessing that's my dead end |
| 02:08.11 | CRCinAU_ | yup - so if 6000 doesn't exist under [default] then it doesn't exist for the goto. |
| 02:08.49 | Xedecimal | gotcha, so now in the default, I want to add exten => 6000,1,Dial(SIP/WhichPhoneIsMine?) |
| 02:09.00 | Xedecimal | wait, I don't want to dial back out to sip |
| 02:09.01 | CRCinAU_ | yup |
| 02:09.07 | CRCinAU_ | sure you do. |
| 02:09.12 | CRCinAU_ | your phone is a SIP phone, right? |
| 02:09.19 | Xedecimal | ohhhh, back out through the sip server to the sip router that my phone is hooked up to |
| 02:09.36 | Xedecimal | there are two phones hooked up to it though |
| 02:09.53 | Xedecimal | with 2 users created |
| 02:09.55 | CRCinAU_ | well, not quite... it'll be SIP/<what you called your [phone] in sip.conf |
| 02:10.15 | *** join/#asterisk pawz (n=pawz@ppp118-208-61-62.lns20.bne1.internode.on.net) |
| 02:10.16 | Xedecimal | gotcha, I call the trunk my gizmo number, then an entry in there for the phone called [6000] |
| 02:10.27 | Xedecimal | but under [6000] I'm guessing the context is screwed |
| 02:10.37 | CRCinAU_ | doesn't really matter. |
| 02:10.45 | CRCinAU_ | that's only really used for outgoing stuff |
| 02:10.50 | Xedecimal | it's in /etc/asterisk/users.conf |
| 02:11.03 | Xedecimal | let me make sure it's registered |
| 02:11.06 | CRCinAU_ | users.conf eh? never even touched that ;) |
| 02:11.12 | pawz | hi. I have speex-1.2 and asterisk 1.6 and i'm trying to initiate calls via speex/1600 but the server appears to only be supporting speex/800. does anyone know what I need to do to support speex/1600 wideband ? |
| 02:11.19 | CRCinAU_ | if you do a 'sip show peers' you should see your phones |
| 02:11.44 | Xedecimal | in the router, line1 and line2 are registered... Checking sip show peers |
| 02:12.00 | Xedecimal | 6000/6000 192.168.1.17 D N 5060 Unmonitored |
| 02:12.08 | Xedecimal | it's running, not monitored though, don't know why |
| 02:12.19 | CRCinAU_ | do you're do Dial(SIP/6000) |
| 02:12.36 | CRCinAU_ | or if you have two phones you want to ring: Dial(SIP/6000;SIP/6001) etc |
| 02:12.44 | Xedecimal | wow, let me try that |
| 02:13.00 | CRCinAU_ | lol @ my english fail just before LOL |
| 02:13.20 | Xedecimal | 'are you using' I took it as |
| 02:13.31 | Xedecimal | now I have exten = s,1,Dial(SIP/6000) |
| 02:13.39 | Xedecimal | reloaded, calling... |
| 02:13.55 | CRCinAU_ | it was supposed to be: so you'd do Dial(SIP/6000) - but I was reading something else and my brain went into free-type ;) |
| 02:14.01 | Xedecimal | Call from '17472729388' to extension '6000' rejected because extension not found. |
| 02:14.02 | Xedecimal | grrr |
| 02:14.30 | CRCinAU_ | hmmm |
| 02:15.34 | CRCinAU_ | oh - and it's & not ; to concatinate calls... so for two it'd be SIP/6001&SIP/6002 etc |
| 02:15.45 | Xedecimal | that's ok, I just want to get one ringing |
| 02:16.03 | pawz | any idea what i have to configure to get speex wideband working ? i have speex narrowband working ok, but not wideband |
| 02:16.16 | Xedecimal | pawz: I have no clue what that is, much less get it working |
| 02:16.26 | pawz | it's a codec |
| 02:19.49 | Xedecimal | incoming calls can be completely configured from start to finish in just /etc/asterisk/extensions.conf right ? |
| 02:19.58 | Xedecimal | (aside from the sip.conf configuration of the trunk) |
| 02:27.49 | [TK]D-Fender | Xedecimal: All call processing = extensions.conf |
| 02:29.14 | [TK]D-Fender | Xedecimal: Call comes in. If its SIP, its authed against sip.conf and based on the peer it matches goes to the appropriate context in sip.conf looking for the target they dialed. In your case "6000". You'll need an exten that can match that in the context your call is looking in. |
| 02:32.31 | Xedecimal | ok, then call comes in from google voice to gizmo5, gizmo5 redirects to my asterisk box. In my sip.conf I have register => 17472729388:mypass@proxy01.sipphone.com and that matches... There however is no context from here out, how does it know where to go from here? |
| 02:34.47 | Xedecimal | also in extensions.conf, what context name do I make to accept a call on this number ? |
| 02:35.06 | Xedecimal | is the 'username' portion of that uri the actual name of the context to use ? |
| 02:35.20 | Xedecimal | err, url even |
| 02:39.07 | [TK]D-Fender | Xedecimal: REGISTER tells THEM what IP to send calls to. when it comes in it is matched againsg an ENTRY in sip.conf [whatever] |
| 02:39.24 | [TK]D-Fender | Xedecimal: REGISTER has nothing to do with processing a call they actually send you |
| 02:39.39 | *** join/#asterisk memph (n=memph@sd1438.sivit.org) |
| 02:39.49 | carrar | TK, isn't there a book that talks about that? |
| 02:40.16 | carrar | if only there was a book |
| 02:40.26 | [TK]D-Fender | Xedecimal: Your sip.conf [whatever] should have a "context=acontextinextensions.conf" telling it where to send the call its accepted |
| 02:40.39 | [TK]D-Fender | carrar: that's a NOVEL idea! |
| 02:40.46 | [TK]D-Fender | PUNishes carrar some more |
| 02:41.08 | carrar | maybe the book could include other topics too |
| 02:41.17 | carrar | man that would be sweeeet |
| 02:41.49 | carrar | <PROTECTED> |
| 02:42.41 | carrar | pokes at Xedecimal |
| 02:43.01 | CRCinAU_ | The Dummies Guide To Asterisk... and how to get screwed by Digium? :P |
| 02:43.13 | carrar | ~book |
| 02:43.14 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:43.43 | Xedecimal | ok, in my /etc/asterisk/sip.conf I've added [proxy01.sipphone.com] entries include context=fromgizmo... Extensions.conf I have [fromgzimo] containing exten = s,1,Dial(SIP/6000) and in /etc/asterisk/users.conf I have a [6000] but in it's context it's using 'DLPN_PlanNick' which I believe is an outbound dial plan, which is not really meant for incoming right ? |
| 02:44.52 | [TK]D-Fender | Xedecimal: "s" is not a magical exten. They are looking for a MATCH to 6000 and you offer them "s". This is your misunderstanding |
| 02:45.00 | Xedecimal | proxy01.sipphone.com/1747 198.65.166.131 N 5060 OK (115 ms) ... This sip peer says OK! and proxy01.sipphone.com:5060 N 17472729388 105 Registered Fri, 15 Jan 2010 18:44:23 shows that we're registered and ready to rock and roll right ? |
| 02:45.06 | Xedecimal | ahhhhhhhhhhhh! |
| 02:45.24 | Xedecimal | s should actually be the number that's always going to be the source... 17472729388 right ? |
| 02:45.41 | carrar | AKA DNIS |
| 02:45.48 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 02:45.50 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 02:46.00 | carrar | not source, destination |
| 02:46.08 | [TK]D-Fender | Xedecimal: the fact you have a SIP peer called "6000" is irrelevent. Call comes in and hits extensions.conf where you do WHATEVER you want to it. You need not have ANY other SIP peer, or other kind of device setup. You could use * as a pure VM box. Or a poll-taking system. Or as a jukebox. |
| 02:46.29 | [TK]D-Fender | Xedecimal: No.. you simply shouldn't HAVE an "s" there unless you have some other reason to |
| 02:47.10 | Xedecimal | how do I direct from an incoming call at 17472729388, to a sip client connected to this server listed as SIP/6000 then ? |
| 02:49.05 | Xedecimal | and I was under the impression that the source of 's' was meant to be 'all', which is what the default is supposed to translate to, but doesn't |
| 02:49.15 | Xedecimal | I need a fresh configuration, this is retarded |
| 02:49.18 | [TK]D-Fender | Xedecimal: exten => 17472729388,1,Dial(SIP/6000) |
| 02:49.34 | Xedecimal | tried that, same result, it's probably not the extension, it's probably one of these other stupid things screwing it up |
| 02:49.49 | [TK]D-Fender | Xedecimal: match 17472729388. Setp 1. Dial device SIP/6000 |
| 02:49.51 | Xedecimal | actually, that's what it's set to right now, saying extension not found, as always |
| 02:50.13 | [TK]D-Fender | Xedecimal: Now the question is... is it LOOKING where you THINK its looking. |
| 02:50.20 | Xedecimal | I wouldn't have a clue |
| 02:50.26 | Xedecimal | I don't know where the hell this thing is looking |
| 02:50.34 | Xedecimal | I don't even know if this thing has any form of eyes to look with |
| 02:50.42 | [TK]D-Fender | Xedecimal: enable SIP DEBUG at CLI "sip set debug" and PASETBIN your call attempt |
| 02:50.45 | [TK]D-Fender | ~pb |
| 02:50.46 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 02:50.48 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 02:50.54 | Xedecimal | ok!!! Now this is new! |
| 02:51.19 | Xedecimal | lol, right off the bat |
| 02:51.24 | Xedecimal | No such command 'sip set debug' |
| 02:51.32 | Xedecimal | I wonder what version of asterisk this sucker is |
| 02:51.36 | Xedecimal | 1.6.2.0 |
| 02:51.48 | Xedecimal | that's new right ? |
| 02:52.21 | [TK]D-Fender | Xedecimal: "sip set debug on" |
| 02:52.21 | Xedecimal | got debugging tunred on |
| 02:52.35 | [TK]D-Fender | Xedecimal: Barring that... try "help sip". They may have changed some syntax around again |
| 02:52.49 | [TK]D-Fender | Xedecimal: Good. No place a call and grab ALL of the output and pastebin it |
| 02:56.21 | [TK]D-Fender | Xedecimal: Looking for 6000 in DID_17472729388 (domain xedecimal.net) |
| 02:56.25 | Xedecimal | see anything crazy ? |
| 02:56.45 | [TK]D-Fender | Xedecimal: [DID_17472729388] <- this is the extensions.conf context it's looking in |
| 02:57.00 | [TK]D-Fender | Xedecimal: SIP/2.0 404 Not Found <-- where it fails to find a match |
| 02:57.21 | [TK]D-Fender | Xedecimal: this is the part where we tell you that the GUI is not supported (here, or much at all). |
| 02:57.25 | Xedecimal | Looking for 6000 in DID_17472729388 (domain xedecimal.net) |
| 02:57.26 | Xedecimal | ahhhhhh! |
| 02:57.47 | Xedecimal | I've been writing by hand, there's leftover stuff from the previous attempts at the gui |
| 02:57.56 | [TK]D-Fender | Xedecimal: You would be better served to start fresh on your own |
| 02:57.59 | Xedecimal | I went through the gui and tried to delete the trunk and all that |
| 02:58.01 | CRCinAU_ | the GUI is like app_fax and digiums offerings.... not supported ;) |
| 02:58.11 | Xedecimal | I get what you're saying though |
| 02:58.23 | [TK]D-Fender | CRCinAU_: Be nice... they support things they charge you for! |
| 02:58.26 | CRCinAU_ | Xedecimal: yeah - start with a blank set of configs :) |
| 02:58.32 | CRCinAU_ | [TK]D-Fender: if only! |
| 02:58.33 | Xedecimal | yeah, I don't even have the DID_##### part, deleted that too |
| 02:59.05 | [TK]D-Fender | Xedecimal: port any users.conf bits you want to keep back into the appropriate configs and then trash it. |
| 02:59.17 | Xedecimal | now I do! |
| 02:59.22 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 02:59.29 | [TK]D-Fender | Xedecimal: Go change your context names around and strip out just about everything you did not create rather explicitly |
| 02:59.37 | *** join/#asterisk minotaur01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 02:59.41 | CRCinAU_ | I think Digium are having internal issues atm... as I can't even lodge a bug report on their commercial fax offering :\ |
| 03:00.51 | CRCinAU_ | the only suggestion customer support had was to call them in the US :\ |
| 03:01.05 | CRCinAU_ | and I'm not that interested to be up at 3am to do that lol |
| 03:01.11 | Xedecimal | ok, for my [6000] user, that is coming from this RTP300 router, where should I be putting this? |
| 03:01.36 | [TK]D-Fender | Xedecimal: What protocol is it using? |
| 03:01.54 | Xedecimal | oh, sip, it goes into sip.conf then, otherwise iax in iax.conf |
| 03:02.01 | [TK]D-Fender | Xedecimal: Xedecimal As appropriate |
| 03:02.03 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 03:02.47 | Xedecimal | ok, then my context for local user 6000 is relating to outbound calls right ? |
| 03:03.20 | [TK]D-Fender | Xedecimal: No that is where calls FROM that device go in extensions.conf |
| 03:03.26 | Xedecimal | gotcha! |
| 03:03.29 | Xedecimal | wait, no I don't |
| 03:04.04 | Xedecimal | I'm only going to get incoming calls from 17472729388... Calls _from_ 6000 would be going out to the world right ? |
| 03:04.21 | [TK]D-Fender | Xedecimal: Only works one way. Call comes from source X based on the device config. Points to extensions.conf. Then ANYTHING happens based on your patterns there. There is no "outside number". Extensions don't have to follow anyone else's logic. |
| 03:04.46 | [TK]D-Fender | Xedecimal: right now I have no idea where you pointed [6000], or whats in that context. |
| 03:05.13 | Xedecimal | 6000 is a local server I guess, I don't know what you'd call that, it's for this little rtp router to login to it as a sip server |
| 03:06.33 | Xedecimal | rtp300 vonage router that is, logging into the asterisk box as a sip server |
| 03:07.28 | [TK]D-Fender | Xedecimal: What isxeNo, as a SIP CLIENT |
| 03:07.34 | [TK]D-Fender | Xedecimal: It is an ATA. |
| 03:07.37 | [TK]D-Fender | ~ata |
| 03:07.38 | infobot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 03:08.09 | [TK]D-Fender | Xedecimal: It registers to *. |
| 03:09.27 | Xedecimal | [Jan 15 19:09:14] WARNING[21292]: chan_sip.c:24936 reload_config: Section '6000' lacks type |
| 03:09.37 | Xedecimal | this is just going from bad to worse |
| 03:10.12 | Xedecimal | moving this from users to sip made this show up, so I'm guessing there's some default configuration in users.conf that specifies a stupid type ? |
| 03:10.29 | Xedecimal | but let me guess, there's a catch, it's not called 'type' huh ? |
| 03:10.37 | Xedecimal | this thing feels as though it was purposefully complicated, like sendmail |
| 03:13.44 | [TK]D-Fender | Xedecimal: type=peer |
| 03:14.17 | [TK]D-Fender | Xedecimal: You should seriously sit down with THE BOOK.... |
| 03:14.18 | [TK]D-Fender | ~book |
| 03:14.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 03:17.22 | jaytee | [TK]D-Fender, you wouldn't happen to know of a good way to make all Polycom phones reregister to a server quickly would you? |
| 03:17.39 | jaytee | in case I need to switch to a backup server? |
| 03:17.42 | [TK]D-Fender | jaytee: there is a frequency field... |
| 03:17.55 | jaytee | 3600 seconds default? |
| 03:17.59 | [TK]D-Fender | jaytee: it'll switch on failed call automatically as well.. |
| 03:18.04 | [TK]D-Fender | jaytee: I think so.... |
| 03:19.02 | jaytee | so if I pickup a handset and try to make a call the phone will attempt to reregister? I thought about trying to set the register timeout interval lower but was concerned about excess traffic if I set it too low |
| 03:19.29 | jaytee | I'm thinking 15 minutes might be acceptable as all the phones are in their own VLAN |
| 03:19.52 | [TK]D-Fender | jaytee: Where are the target servers relative? |
| 03:20.20 | jaytee | they are both in the same rack if that's what you're asking |
| 03:20.32 | [TK]D-Fender | jaytee: Remember many ITSP's have their CPE ATA's REG every minute both for the reg and as a keep-alive |
| 03:20.53 | [TK]D-Fender | jaytee: Set your reg high like that and disable qualify if you like |
| 03:21.11 | [TK]D-Fender | jaytee: Reg freq on local lan? Kinda sad. Run RRDNS or something then |
| 03:21.38 | jaytee | [TK]D-Fender, sorry, not familiar with RRDNS, is that a round robin type scheme? |
| 03:22.05 | jaytee | ah, just googled it |
| 03:22.19 | [TK]D-Fender | jaytee: Round Robin DNS. let them reg to a hostname and have a DNS server flip as needed |
| 03:23.41 | jaytee | they do register to a hostname, the FQDN of the server is set in our DNS but our DNS isn't setup as RRDNS I don't think, it runs on our Windows server domain controllers. |
| 03:24.11 | jaytee | and the phones are all setup to register to our asterisk server by FQDN instead of explicit ip |
| 03:24.41 | [TK]D-Fender | jaytee: Then point to a smarter router that is more reliable and have that forward then |
| 03:26.45 | jaytee | we're using a Cisco Catalyst 4507 as our main switch in our datacenter and our buildings mostly have Cisco 2905 switches. I'd have to have our network engineer do any reconfiguring. |
| 03:27.07 | jaytee | [TK]D-Fender, know any good links for an example of what you've described? |
| 03:28.49 | *** join/#asterisk etfonhomey (n=etfonhom@166.203.141.104) |
| 03:31.13 | jaytee | [TK]D-Fender, my current strategy in case of our primary server failing is to move my T1 cables to the standby server which already has the current asterisk configs loaded but it's sitting idle. Then I run a script that changes the IP address to the one in DNS that all the phones register to by hostname and the script then restarts the network service with the new IP configuration. |
| 03:32.28 | jaytee | so I'm thinking that if I can set the register timeout interval on the phones to 15 minutes or maybe even 5 and they'd all reregister within that period or if someone tries to use their phone and doing that makes it reregister my downtime would be significantly reduced. |
| 03:35.53 | *** join/#asterisk uluatu (n=uluatu@189.114.176.219.dynamic.adsl.gvt.net.br) |
| 03:36.41 | *** join/#asterisk lordmortis (n=lordmort@dsl-58-7-231-96.wa.westnet.com.au) |
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| 03:43.44 | *** join/#asterisk rizwank (n=rizwank@76.89.131.47) |
| 03:45.01 | rizwank | I'm having trouble with Asterisk kicking back a InvalidCommand error to an AGI script whenever I do a playback -- -I can't seem to figure out why |
| 03:51.13 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-12.static.o1.com) |
| 03:51.43 | *** join/#asterisk smooth_penguin (n=smoove@59.95.19.122) |
| 04:00.39 | lordmortis | Answered the channel? |
| 04:00.55 | pawz | i have no /var/db/asterisk/astdb file ? is there some way i can create it ? |
| 04:01.35 | Tim_Toady | it should be in /var/lib/asterisk/ |
| 04:02.08 | pawz | that directory doesn't exist on freebsd |
| 04:02.15 | pawz | can i create the astdb ? |
| 04:02.17 | Tim_Toady | ah freebsd ok |
| 04:02.30 | Tim_Toady | i think asterisk creates this by itself if its not there |
| 04:02.45 | Tim_Toady | if not try touch astdb |
| 04:05.22 | pawz | yeah that appears to have fixed it |
| 04:14.44 | *** join/#asterisk coppice (n=chatzill@106.202.17.210.dyn.pacific.net.hk) |
| 04:16.00 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:16.11 | rizwank | whenever I do a command via agi like playback or read with a filename, it braeks the pipe |
| 04:16.14 | rizwank | and doesn't play the file |
| 04:16.34 | [TK]D-Fender | ~oslec |
| 04:16.35 | infobot | [~oslec] OSLEC is the Open Source Line Echo Canceler. It is an superior alternative to the native SWEC routines in Zaptel/DHADHI and debatably a small notch below that of Digium's HPEC and Sangoma's SoftEcho in effectiveness. Web site : http://www.rowetel.com/ucasterisk/oslec.html , Mailing list : https://lists.sourceforge.net/lists/listinfo/freetel-oslec |
| 04:20.16 | coppice | some people end up using OSLEC to replace HPEC or SoftEcho, although its more often used to replace the troublesome echo cancellation hardware |
| 04:23.53 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 04:23.53 | *** mode/#asterisk [+o file] by ChanServ |
| 04:29.12 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:29.24 | joelsolanki | Hi all |
| 04:30.48 | [TK]D-Fender | coppice: That was a very well timed and worded assessment. Thanks |
| 04:31.50 | afink | lol |
| 04:34.29 | coppice | Both Digium and Sangoma really need to get their acts together with hardware echo cancellation. They charge plenty for it |
| 04:36.39 | *** part/#asterisk grnmtn (n=grnmtn@65.183.149.153) |
| 04:40.57 | [TK]D-Fender | coppice: Somehow I hear a lot more problems with Digium's, where both use Otasic chips with seemingly identical specs. Are you aware if they are different revisions, etc? |
| 04:41.39 | *** join/#asterisk p3nguin (i=gpz5GvdF@9xf6gbZ.a2infotech.com) |
| 04:41.56 | coppice | well, they seem to change the level of problems through firmware updates, so the problems may relate to how the Octasic stuff is applied. |
| 04:42.42 | coppice | people who have used them all say the only boards which never cause trouble are the Digium ISDN boards. Those are the only cards which don't use Octasic |
| 04:43.17 | p3nguin | Is there any way to accept DTMF during MusicOnHold()? |
| 04:43.48 | p3nguin | For, say, dialing a different exten. |
| 04:44.17 | [TK]D-Fender | p3nguin: that is a dead-end app |
| 04:44.42 | p3nguin | That's what I was afraid of. |
| 04:45.59 | p3nguin | To me, this translates to "don't use it if you can keep from it." |
| 04:49.54 | [TK]D-Fender | p3nguin: what use is it? Its there to test MoH, not to use a background for an IVR |
| 04:50.32 | [TK]D-Fender | p3nguin: You can have SIP phones call it for ambient MoH and have * kill that call before sending it a new call, etc to simulate some key system functionality |
| 04:51.46 | p3nguin | I was thinking about using it to pad an IVR with a few seconds of music, but there is a chance that a person might dial an exten during those seconds... so it's not going to work for that. |
| 04:52.20 | p3nguin | It was going to be only for "comfort" purposes. |
| 04:53.01 | p3nguin | Is there a way to delay ringing phones after a call goes into a queue? |
| 04:53.10 | *** join/#asterisk nprice (n=nprice@cpe-76-184-72-104.tx.res.rr.com) |
| 04:53.31 | p3nguin | I know there is memberdelay, but that still rings the phones right away. |
| 04:53.37 | [TK]D-Fender | p3nguin: Not really. |
| 04:53.38 | p3nguin | so that's not going to work. |
| 04:53.55 | [TK]D-Fender | p3nguin: You could delay BEFORE going to queue |
| 04:54.08 | [TK]D-Fender | p3nguin: if you stream MoH it could seem pretty seamless |
| 04:54.32 | p3nguin | I was going to delay it with MoH for a few seconds, but during that time, the caller could try to dial an exten. |
| 04:54.47 | p3nguin | Once they are in the queue, context will allow them to get back out. |
| 04:54.55 | p3nguin | context = qoutcon |
| 04:55.26 | p3nguin | But if they dial digits during the MusicOnHold(3), the keys will get ignored. |
| 04:55.54 | p3nguin | I can punch in about 7 keys in a 3-second period. |
| 04:56.43 | p3nguin | So if the IVR says "you may dial the extension at any time," that three seconds of MoH would cause a problem. |
| 04:58.33 | [TK]D-Fender | p3nguin: I'd prepare to live with it ringing members immediately then |
| 04:58.41 | p3nguin | yep |
| 04:58.51 | p3nguin | You've talked me into that several minutes ago. :) |
| 05:00.07 | *** join/#asterisk PaulR_ (n=PaulR@nat66.mia.three.co.uk) |
| 05:00.21 | PaulR_ | hi all |
| 05:00.47 | PaulR_ | any of u guys use ribbit? |
| 05:00.53 | *** join/#asterisk RypPn (n=TuMbL@rosscom.co.uk) |
| 05:10.05 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
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| 05:19.50 | *** join/#asterisk CLeron (n=asdas@d206-116-30-244.bchsia.telus.net) |
| 05:20.11 | CLeron | hi can anyone help me out, i have just installed AsteriskNow with CentOS but for some reason im not getting an internet connection. Is there anything like ipconfig in CentOS where i can play with the configurations? |
| 05:22.01 | CLeron | anyone awake in here? |
| 05:24.39 | p3nguin | cleron: CentOS is no different than any other Linux-based operating system. |
| 05:24.57 | p3nguin | With that said, you have full control over all the interconnectivity settings. |
| 05:25.55 | PaulR_ | CLeron: system-network-config or system-config-network |
| 05:26.00 | PaulR_ | i always get em mised up |
| 05:26.04 | PaulR_ | *mixed |
| 05:26.52 | PaulR_ | and that writes config files in /etc/sysconfig/eth0-ifcg etc |
| 05:26.57 | CLeron | thanx |
| 05:27.02 | PaulR_ | np |
| 05:27.05 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 05:27.23 | PaulR_ | i guess most people are asleep |
| 05:29.02 | ChannelZ | just got home |
| 05:29.45 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
| 05:30.46 | PaulR_ | anyway if anyone uses ribbit gimme a shout |
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| 07:20.58 | *** join/#asterisk viz (i=viz@122.170.31.144) |
| 07:21.08 | viz | hi |
| 07:21.17 | viz | anyone familiar with func_odbc and hash? |
| 07:23.51 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 07:28.54 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 07:33.06 | viz | guess everyone is asleep |
| 08:08.01 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 08:15.45 | sun28 | moin |
| 08:18.17 | ChannelZ | honks |
| 08:32.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 08:49.45 | *** join/#asterisk DarkRift (n=dark@65.92.167.83) |
| 08:52.17 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:55.25 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-12.static.o1.com) |
| 09:03.03 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.13.dyn.user.ono.com) |
| 09:04.18 | Godfather_ | o/ |
| 09:10.10 | *** join/#asterisk supercappy (n=captain@76.91.206.32) |
| 09:10.21 | supercappy | yay... 171 users |
| 09:11.19 | supercappy | hey guys... is there something *special* about the environment in which AGI scripts are run?? i've set write permissions so that the 'asterisk' user can write to an area in my web root, but my agi script can't |
| 09:11.55 | ChannelZ | Not that I know of |
| 09:12.06 | ChannelZ | It should run as the user * runs as |
| 09:12.07 | supercappy | but the agi script is running as 'asterisk' ... so what's missing? |
| 09:12.26 | supercappy | ChannelZ, come again? |
| 09:12.28 | supercappy | user *? |
| 09:12.41 | ChannelZ | asterisk |
| 09:12.46 | ChannelZ | * |
| 09:12.58 | ChannelZ | lazy typer |
| 09:13.02 | supercappy | oh hehe |
| 09:14.43 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 09:14.44 | supercappy | hmmm... it's so weird |
| 09:15.27 | supercappy | i "su -mc asterisk 'ls /path/to/where/i/wanna/write'" and it shows the files |
| 09:15.55 | ChannelZ | well seeing and writing are not the same thing |
| 09:15.56 | supercappy | but within the agi script i do the ls and the permissions come out as ????'s |
| 09:16.12 | supercappy | ls -l rather |
| 09:16.19 | supercappy | ChannelZ, yes, but seeing is believing :P |
| 09:16.25 | supercappy | touch works too |
| 09:16.34 | supercappy | while debugging i stepped it back |
| 09:16.46 | supercappy | and even ls gives different results |
| 09:19.30 | supercappy | i thought it was some kind of magical jail environment or something... but i can reach the parent directory |
| 09:19.56 | ChannelZ | what language is this |
| 09:21.23 | supercappy | perl |
| 09:22.45 | supercappy | one thing... the permissions it SHOULD be granted are through a secondary group (www-user) which i've added asterisk to |
| 09:23.38 | supercappy | but that shouldn't matter right? |
| 09:24.23 | ChannelZ | generally not |
| 09:24.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 09:24.29 | supercappy | hmmmm... i haven't restarted that asterisk server since i added asterisk to that group... |
| 09:24.31 | ChannelZ | Are you using some wierd perl module for file IO? |
| 09:24.48 | supercappy | i remember in an old sunos, this mattered, but i didn't think so in linux... |
| 09:24.54 | supercappy | lemme try that just for the heck of it |
| 09:25.18 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 09:25.23 | supercappy | nope... the ls -l is through backticks |
| 09:29.23 | *** join/#asterisk ycas (n=yann@85-127-84-208.dynamic.xdsl-line.inode.at) |
| 09:30.44 | supercappy | ahhhhhhhhhhh that was it |
| 09:30.50 | supercappy | i had to restart the asterisk server |
| 09:30.59 | supercappy | wow i feel dumb :( |
| 09:32.35 | supercappy | so i guess i learned that: 1. the group-derived permissions that a process has are determined by the groups the user was a member of AT THE TIME THE PROCESS IS FIRED |
| 09:32.50 | benngard | is playing around with the ami |
| 09:33.20 | supercappy | 2. AGI scripts inherit asterisk's permissions |
| 09:33.48 | benngard | howto make a space? Action: Originate ... CallerID: <Froken Ur>, but on the display it shows FrokenUr without the "space" :( |
| 09:33.59 | supercappy | well thanks for the help ChannelZ... helped me not bark up wrong trees |
| 09:34.30 | ChannelZ | I live to partially serve |
| 09:36.53 | benngard | CallerID: <Froken_Ur> shows Froken_Ur at the display, but i want it to show Froken Ur, what am i missing? |
| 09:37.21 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 09:37.37 | supercappy | ChannelZ, lol! |
| 09:38.51 | benngard | will try to originate to another sip device |
| 09:40.10 | *** join/#asterisk dzup (i=dzup@support.team.at.shellium.org) |
| 09:40.51 | dzup | hello, whats the most user-friendly and better asterisk distro out there? |
| 09:41.04 | dzup | and with more support |
| 09:41.32 | dzup | been looking at astlinux, freebsd, asteriskNOW asterCC, am kind a lost here |
| 09:41.58 | benngard | debian seems to be a good base to build on |
| 09:42.56 | dzup | but i like to be on CentOS |
| 09:43.08 | dzup | seems like there is more support for asterisk there |
| 09:43.15 | supercappy | i <3 DEBIAN! |
| 09:43.46 | benngard | but what the heck, an avaya sip phone and a siemens dect sip phone shows very different on the display |
| 09:43.52 | supercappy | i was actually surprised at how quick and easy i got asterisk running on debian |
| 09:43.55 | dzup | asteriskNOW is opensource or went private? |
| 09:44.15 | supercappy | having moved from trixbox... and having been an avid debianer for many years i was happy :D |
| 09:44.16 | benngard | no problem att all compiling asterisk on a debian |
| 09:44.25 | supercappy | no compiling! |
| 09:44.29 | supercappy | apt-get install asterisk |
| 09:44.34 | supercappy | bam! boom! :D |
| 09:44.36 | benngard | u can do that 2 |
| 09:44.49 | supercappy | i'm simplifying slightly ;) there are a few other packages too |
| 09:45.11 | supercappy | one difference i found though (and i'm not sure if this was compile options or what) was the location of agi-bin |
| 09:45.31 | benngard | but "may" and me are playing around with the ooh323 driver so i do compile very often ;) |
| 09:45.33 | dzup | benngard: can you point me in th right direction? i need install asterisk but havent decide yet on astlinux, asteriskNOW, asterCC , what do you use? |
| 09:45.46 | *** join/#asterisk toppe (n=root@cs181001225.pp.htv.fi) |
| 09:45.51 | ChannelZ | benngard: Well for one <Froken Ur> isn't valid caller ID |
| 09:46.05 | benngard | i saw that |
| 09:46.27 | dzup | or ChannelZ |
| 09:46.33 | ChannelZ | runs on Ubuntu |
| 09:46.36 | dzup | or anybody :( |
| 09:46.41 | ChannelZ | but builds * from scratch, not packaged |
| 09:46.42 | benngard | dzup: if u know linux, i would go fro debian and aptitude install asterisk |
| 09:46.52 | benngard | for* |
| 09:47.02 | supercappy | dzup, my vote doesn't count... but it's debian ;) |
| 09:47.04 | dzup | i know linux very well, but i want centos based |
| 09:47.20 | ChannelZ | so run it then |
| 09:47.35 | supercappy | why not trixbox ce? |
| 09:47.38 | benngard | ChannelZ: did change to CallerID: Froken Ur<90510>, that gave me correct at gigaset display but just number at avaya display |
| 09:47.43 | dzup | but what distro? am between astlinux, asteriskNOW or asterCC |
| 09:47.47 | drmessano | LOL |
| 09:47.48 | supercappy | i didn't hate it... just dislike rpm based distros |
| 09:47.51 | drmessano | ~trixbox |
| 09:47.52 | infobot | rumour has it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 09:48.06 | drmessano | ~trashbox |
| 09:48.07 | supercappy | lol |
| 09:48.12 | drmessano | Trixbox sucks |
| 09:48.16 | dzup | yeah, no trixbox here |
| 09:48.22 | supercappy | runs and hides |
| 09:48.32 | ChannelZ | asteriskNOW is CentOS purpose-built with asterisk+freepbx in it |
| 09:48.35 | supercappy | is glad he moved to debian |
| 09:48.36 | drmessano | trixbox is the windows vista of asterisk packaged distros |
| 09:48.42 | ChannelZ | if you are super lazy run that |
| 09:49.01 | *** join/#asterisk d00gster (n=doughant@77.30.116.159) |
| 09:49.02 | ChannelZ | if you want to learn something, run CentOS and put up * yourself |
| 09:49.07 | *** join/#asterisk chendy (n=chatzill@116.30.141.100) |
| 09:49.11 | dzup | soo then asteriskNOW it is? |
| 09:49.24 | drmessano | If you want to put up a badass Fonality honeypot, download Trixbox |
| 09:49.29 | dzup | better then astlinux? |
| 09:50.02 | drmessano | Yes, use AsteriskNOW and move along |
| 09:50.04 | benngard | why not gor for Centos and install asterisk 1.6.2.1 and start learning |
| 09:50.13 | supercappy | haha wow... put the crucifixes away guys! i moved OFF of trixbox :P |
| 09:50.44 | ChannelZ | Everytime I hear trixbox I think prostitutes |
| 09:50.51 | drmessano | supercappy: Don't flatter yourself, wasn't addressing you specifically |
| 09:50.55 | supercappy | loves prostitutes |
| 09:51.08 | *** join/#asterisk Borai (i=MS@S0106001c109e98db.no.shawcable.net) |
| 09:51.15 | Borai | Hi |
| 09:51.26 | drmessano | I am a the Borat? |
| 09:51.34 | supercappy | drmessano, i know i was kidding |
| 09:51.50 | drmessano | Oh, Borai. i c |
| 09:52.03 | drmessano | Jagshemash anyway? |
| 09:52.05 | ChannelZ | Maybe that's plural for Borat |
| 09:52.10 | Borai | Haha |
| 09:52.17 | Bora | better now? |
| 09:52.35 | drmessano | JAGSHEMASH |
| 09:52.35 | ChannelZ | it's all subjective |
| 09:52.45 | ChannelZ | Waa-waa-wee-waa |
| 09:53.21 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-12.static.o1.com) |
| 09:53.32 | drmessano | My sister is bestest trixbox in all Khazakstan |
| 09:53.33 | Bora- | Can anyone suggest me a decent provider for US/canada dids and voip termination |
| 09:53.40 | *** join/#asterisk yang (i=yang@freenode/sponsor/cacert.assurer.yang) |
| 09:53.41 | supercappy | JENDOBRE! ;) |
| 09:54.14 | Bora- | possibly a company that doesnt charge you to have an account with them, I was happy with les.net had a few numbers but they started charging 2.50 a month just to have an account with them |
| 09:54.55 | drmessano | They did? |
| 09:55.25 | supercappy | speaking of which... anyone have any comments on broadvoice? |
| 09:55.33 | Bora- | yeah |
| 09:55.35 | drmessano | $1.99.. Interesting |
| 09:55.39 | drmessano | Flowroute is pretty decent |
| 09:55.52 | Bora- | 1.99 is their tollfree did price per month |
| 09:55.55 | Bora- | I have no problem paying that |
| 09:56.03 | Bora- | but 2.50 an account charge per month |
| 09:56.14 | Bora- | just to have an account with them. |
| 09:56.25 | drmessano | LES.NET Account |
| 09:56.26 | drmessano | Subscription$1.99/Month |
| 09:56.33 | Bora- | Like I would understand if I was just using it for termination that they charge me but, im already paying a did fee right |
| 09:56.37 | drmessano | I _was_ referring to the account |
| 09:56.39 | Bora- | oh they changed it to 1.99 then |
| 09:56.40 | supercappy | ok past my bedtime... night guys! thanks again |
| 09:56.43 | Bora- | sorry. |
| 09:56.53 | Bora- | their did pricing is 1.99 too |
| 09:56.54 | *** join/#asterisk AaronM2282 (n=aaronm22@203.206.167.153) |
| 09:57.19 | drmessano | Nope |
| 09:57.35 | drmessano | 99 cents |
| 09:57.48 | drmessano | US anyway |
| 09:58.21 | Bora- | hmm |
| 09:58.26 | Bora- | lets see |
| 09:58.41 | Bora- | I got tollfree numbers with them |
| 09:58.49 | Bora- | Monthly$2.50 |
| 10:00.22 | drmessano | Flowroutes termination is much cheaper |
| 10:00.47 | drmessano | Voip.ms comes highly recommended as well, but I havent used them |
| 10:01.50 | ChannelZ | can you port existing numbers to these services or is that pretty much up to your local telco? |
| 10:01.56 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 10:02.37 | drmessano | You can port all you want.. law says no one can deny a port |
| 10:02.39 | benngard | i dont know how porting numbers work except for sweden |
| 10:03.08 | Bora- | wait |
| 10:03.20 | Bora- | so I can simply get a local Canada number and get it ported to a VOIP provider? |
| 10:03.23 | benngard | in seden it is pretty easy, takes 10 working days and then the number is transferred |
| 10:03.25 | drmessano | http://www.fcc.gov/cib/consumerfacts/numbport.html |
| 10:03.31 | benngard | in sweden |
| 10:03.45 | drmessano | Bora-: Why would you do that? |
| 10:03.58 | drmessano | Just sign up with the provider and get the number through them |
| 10:04.02 | drmessano | No porting anywhere |
| 10:04.33 | benngard | maybe we has a well-known number that he would like to reuse? |
| 10:04.39 | Bora- | well I have looked but no operator was giving a number in my area |
| 10:04.51 | drmessano | benngard: thats not what he said |
| 10:05.07 | benngard | sorry missread |
| 10:05.12 | drmessano | Bora-: If they dont offer a number in your area, chances are they can't service it |
| 10:05.25 | Bora- | True however. |
| 10:05.29 | drmessano | Bora-: My area is like that as well.. Only a few providers service here |
| 10:05.35 | Bora- | I live on Vancouver Island, on a small town called NAnaimo |
| 10:05.52 | Bora- | 16k people live in this city |
| 10:06.14 | Bora- | I can get Vancouver numbers, which are on mainland, or Victoria numbers which are on the island |
| 10:06.21 | Bora- | *is on the island |
| 10:07.01 | drmessano | 191,000 in my city, and a lot of the ITSPs don't service here |
| 10:08.30 | Bora- | HMM |
| 10:08.47 | Bora- | So the best way would be getting an ATA and using an analog line? |
| 10:09.36 | *** join/#asterisk etfonhomey (n=etfonhom@166.203.141.104) |
| 10:09.50 | drmessano | No, the best way is get a nice IP phone hooked to your Asterisk box |
| 10:10.19 | benngard | and a sip trunk to a telco provider ;) |
| 10:10.35 | Bora- | Well I have a Polcyom 550 on my desk |
| 10:10.55 | Bora- | In my terms it is a "nice IP Phone" and it is hooked up to my * box |
| 10:11.14 | Bora- | but I need a local number in this town for incoming calls |
| 10:12.14 | drmessano | What about Vancouver or Victoria.. are they long distance? |
| 10:13.38 | Bora- | yes |
| 10:14.34 | Bora- | what is a toll-free dip? |
| 10:15.48 | benngard | is preparing to throw an avaya 9650 out of the window! |
| 10:16.39 | benngard | but what stops me is the perfect sound it has |
| 10:17.03 | benngard | and the fact that i have like 200 of them |
| 10:18.46 | Bora- | erm nice i cant email @gmail.com |
| 10:22.21 | *** join/#asterisk DelphiWorld (n=Miranda@41.104.68.129) |
| 10:22.24 | DelphiWorld | hi |
| 10:22.30 | DelphiWorld | anyone know an srtp enabled softphone, |
| 10:22.33 | DelphiWorld | for free |
| 10:23.02 | *** join/#asterisk chendy_ (n=chatzill@116.30.141.100) |
| 10:28.04 | *** join/#asterisk Takapa (i=vegard@svanberg.no) |
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| 10:59.29 | *** part/#asterisk DelphiWorld (n=Miranda@41.104.68.129) |
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| 12:01.37 | *** join/#asterisk aidinb (n=Aidin@adsl-71-159-228-75.dsl.sndg02.sbcglobal.net) |
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| 12:44.53 | *** join/#asterisk TheNoOne (n=thenoone@84-119-63-216.dynamic.xdsl-line.inode.at) |
| 12:46.42 | TheNoOne | hi .. i'm using asterisk 1.4 on debian 5 .. i'm trying to get cdr logging to mysql working ... but mysql is not showing up as a registered backend |
| 12:46.47 | TheNoOne | http://pastebin.com/m6d537e7d |
| 12:53.16 | TheNoOne | i also tried to sort my music on hold files alphabetically ... but sort=alpha doesn't do anything |
| 13:07.37 | *** join/#asterisk klochan (n=klochan@95-27-110-105.broadband.corbina.ru) |
| 13:30.33 | benngard | is still playing around wth the ami |
| 13:30.46 | benngard | likes it better and better! |
| 13:32.46 | benngard | just did a VERY SIMPLE html/php app that originates a call to my sip phone and dials from a pull down menu :) |
| 13:40.19 | *** join/#asterisk proute (n=none@ARouen-153-1-72-243.w90-17.abo.wanadoo.fr) |
| 13:40.33 | proute | Hello |
| 13:41.27 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 13:41.37 | ArtemMakhutov | Hello |
| 13:42.04 | ArtemMakhutov | what could this error message mean? |
| 13:42.04 | ArtemMakhutov | [Jan 16 14:40:42] WARNING[32447]: chan_sip.c:6219 sip_write: Asked to transmit frame type 4096, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
| 13:43.06 | proute | I use asterisk 1.4 with B410P card. So, is it possible de configure Linux, Asterisk and B410P card to connect on my system via T0/SO such as Internet link. I wish to connect on my system via T0/S0 to do remote service |
| 13:43.24 | proute | Can you tell me if it's possible and how can I do? |
| 13:43.29 | proute | thanks for your help |
| 13:48.48 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 13:51.12 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 14:03.50 | *** part/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) |
| 14:04.51 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:09.14 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 14:21.06 | *** join/#asterisk smooth_penguin (n=smoove@59.96.89.57) |
| 14:29.28 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 14:39.49 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:52.20 | benngard | yay the shit is working :) |
| 14:55.10 | benngard | damn that i forgot my headset at the office, cant test everything :( |
| 15:11.13 | dlynes | ArtemMakhutov, that's because you've got one end configured for ulaw, and the other end for speex/g722 |
| 15:11.34 | dlynes | ArtemMakhutov, erm g722...ignore the speex |
| 15:29.19 | *** join/#asterisk klochan (n=klochan@95-25-166-179.broadband.corbina.ru) |
| 15:34.53 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 15:39.35 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:40.28 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
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| 16:09.08 | *** join/#asterisk manxpower (n=ewieling@60.sub-70-223-125.myvzw.com) |
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| 16:51.28 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 16:51.51 | sun28 | moin \o/ |
| 16:58.15 | LemensTS | thoughts on magicjack? |
| 17:00.57 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 17:02.57 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 17:13.01 | *** join/#asterisk jawbrkr (n=jawbrkr@stargazer.lavu.net) |
| 17:13.48 | jawbrkr | does it make sense to run asterisk just as a SIP server so I can use voice with openfire or is another alternative ? |
| 17:13.57 | jawbrkr | or is there another |
| 17:21.01 | *** join/#asterisk seather (n=seather@tyrmida.com) |
| 17:21.06 | seather | does anyone here know where in the asterisk configuration I can reduce the volume of my call recordings? |
| 17:21.27 | seather | it's distorting I presume because of the volume being too high |
| 17:31.44 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:33.30 | [TK]D-Fender | LemensTS: A constant uphill struggle to get working with * assuming it isn't still dead even now |
| 17:33.45 | [TK]D-Fender | LemensTS: they actively work to block use with any other software |
| 17:34.10 | [TK]D-Fender | seather: maybe you could actually describe your call and what its arriving on. |
| 17:43.53 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 17:45.43 | LemensTS | TK: You ever use the service, is it good? Kinda sucks they are just giving service away because they have advertisers paying their bills...kinda hard for competitors to compete... |
| 17:46.44 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:56.05 | *** join/#asterisk Baylink (n=jra@65.34.94.81) |
| 17:56.31 | *** join/#asterisk BillyFreddy (n=smithbd@67.199.182.115) |
| 17:57.25 | Baylink | 'dialplan show', in 1.4, will not a) chase includes or b) show *only* the configured extensions. Is there any good way to accomplish "what things can I dial in a given context" without writing my own parser? Am I going to have to extend 'dialplan show'? |
| 17:58.02 | carrar | look at the dialplan? :) |
| 18:05.00 | *** join/#asterisk BuSyAnToS (n=31749@93-44-17-172.ip95.fastwebnet.it) |
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| 18:27.18 | BillyFreddy | <PROTECTED> |
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| 18:50.07 | KAMAZ | Hi |
| 18:51.49 | KAMAZ | Can asterisk to support clients with different authname and username?? |
| 18:54.24 | paulc | KAMAZ: How do you mean? Like end users? |
| 18:54.35 | KAMAZ | Ex.: username=928, authname (Digest) = ghbdtn |
| 18:55.27 | KAMAZ | I have NGN router (DSLAM+VoIP) IES1248 |
| 18:55.40 | KAMAZ | and Asterisk 1.6.0.19 |
| 18:55.40 | paulc | I'm still not sure I understand the question? SIP end points? |
| 18:56.11 | paulc | whatever authname you use is the one that will be compared with the entry in sip.conf? |
| 18:56.18 | KAMAZ | IES= 50 FXS (VoIP ports) |
| 18:56.25 | KAMAZ | 1 sec |
| 18:56.53 | *** join/#asterisk TheDavidFactor-H (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 18:57.27 | *** join/#asterisk haryv (i=lanny@174.1.114.16) |
| 18:58.06 | KAMAZ | On IES each port has its own username = phone number |
| 18:58.26 | KAMAZ | but password is 1 on DSLAM |
| 18:58.51 | paulc | So multiple usernames, one for each port, but they all use the same password? |
| 18:59.22 | KAMAZ | YEs |
| 18:59.28 | KAMAZ | :( |
| 18:59.56 | KAMAZ | On registry my Asterisk (fine working) say: |
| 19:00.48 | KAMAZ | <PROTECTED> |
| 19:01.09 | KAMAZ | On each port |
| 19:01.48 | KAMAZ | ...have <5256XX>, digest has.... |
| 19:02.45 | paulc | Hmm it rings a bell because the usernames are numeric.. maybe.. |
| 19:02.51 | paulc | I'm still waking up this morning :-s |
| 19:03.50 | paulc | do you have username=525628 in your [525628] peer definition |
| 19:04.15 | KAMAZ | I also modeled this situation, putting in softphone Xlite different "Username" and "Authorization name" |
| 19:04.26 | KAMAZ | Yes |
| 19:04.32 | KAMAZ | not working |
| 19:04.44 | paulc | and what result did you get in that case? Did username end up being the caller ID name and authorization name was the actual one used for auth |
| 19:04.47 | KAMAZ | also when username=aster_server |
| 19:05.32 | paulc | I guess the question is - how does your device work? Is it a single registration to the Asterisk server, then you can address the ports separately? or does each port have it's own registration to Asterisk? |
| 19:07.30 | *** join/#asterisk Baylink (n=jra@cerberus.vicimarketing.com) |
| 19:07.35 | KAMAZ | I want to register was on the port, ie Each port has a separate registration for Asterisk |
| 19:08.19 | Baylink | carrar: Looking at the dialplan isn't all that easy, when it was written by FreePBX |
| 19:08.27 | KAMAZ | РеÑÑÑ ÑÑÑÑкоговоÑÑÑие ÑпеÑÑ Ð½Ð° канале? |
| 19:10.03 | paulc | KAMAZ: I'm a bit lost. All I can suggest is check the documentation for sip.conf, think about your peer [definition] name, as well as authuser within each peer. |
| 19:10.50 | KAMAZ | Hm... |
| 19:11.06 | KAMAZ | each port= PSTN FXS |
| 19:11.15 | KAMAZ | each port = PSTN number |
| 19:11.42 | carrar | Baylink, stop using freepbx then |
| 19:11.44 | carrar | sounds messy |
| 19:12.01 | *** join/#asterisk voipmonk (n=shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 19:12.02 | carrar | write your own dialplan so that it makes sense |
| 19:12.26 | KAMAZ | but DSLAM can provide 1 authname/password pair for each port |
| 19:13.33 | *** join/#asterisk ruied (n=ruied@bl7-213-7.dsl.telepac.pt) |
| 19:14.34 | KAMAZ | <--- SIP read from UDP://192.168.29.1:5060 ---> |
| 19:14.34 | KAMAZ | REGISTER sip:192.168.28.80:5060 SIP/2.0 |
| 19:14.34 | KAMAZ | Via: SIP/2.0/UDP 192.168.29.1:5060;branch=z9hG4bK76464e21da219407 |
| 19:14.34 | KAMAZ | Max-Forwards: 70 |
| 19:14.34 | KAMAZ | To: <sip:525628@192.168.28.80> |
| 19:14.36 | KAMAZ | From: <sip:525628@192.168.28.80>;tag=2IkM0gDMxUE |
| 19:14.36 | KAMAZ | Call-ID: 2612D1B92E12BCCC@192.168.29.1 |
| 19:14.38 | KAMAZ | CSeq: 573 REGISTER |
| 19:14.43 | KAMAZ | Contact: <sip:525628@192.168.29.1:5060> |
| 19:14.44 | KAMAZ | Authorization: Digest username="aster_server", realm="asterisk", nonce="50e8883d", uri="sip:192.168.28.80:5060", response="5097f9eab0b5f9aca8177ea074be7354", algorithm=MD5 |
| 19:14.47 | KAMAZ | User-Agent: IES-1248-51V V3.53(ARY.0) |
| 19:14.49 | KAMAZ | Content-Length: 0 |
| 19:14.51 | KAMAZ | Expires: 3600 |
| 19:14.54 | KAMAZ | on registration |
| 19:14.57 | voipmonk | now wait a minute KAMAZ |
| 19:15.05 | paulc | Please don't post like that... www.pastebin.ca is your friend |
| 19:15.06 | voipmonk | thats what pastebin.ca is for |
| 19:15.07 | KAMAZ | OK |
| 19:15.08 | voipmonk | or pasteies |
| 19:15.11 | voipmonk | pasties |
| 19:15.15 | KAMAZ | TNX |
| 19:15.19 | paulc | Cornish pasties? ;) |
| 19:15.19 | KAMAZ | sorry |
| 19:15.28 | voipmonk | just had shwarma.... |
| 19:15.38 | voipmonk | wonder what cornish shwarma would taste like |
| 19:15.44 | voipmonk | or kfc shwarma, lol |
| 19:15.54 | paulc | KAMAZ: Try not using a numeric peer name. Set up a port called "port1" and create a [port1] peer. |
| 19:15.59 | paulc | KFC shwarmer = all sorts of wrong |
| 19:16.00 | voipmonk | im sure im missing an "a" |
| 19:16.11 | paulc | I can't spell it either today, apparently |
| 19:16.26 | voipmonk | hehe |
| 19:16.32 | voipmonk | i know how to eat it... |
| 19:16.41 | voipmonk | im gonna leave it at that |
| 19:16.43 | voipmonk | ... |
| 19:16.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.16 | paulc | Call file question: If I set Channel to Local/xxx@outbound-context, and it gets busy or no answer, does the call pass to the h extension? or failed? docs say failed, 1.6 behaviour seems to be h - just want to confirm I'm doing the right thing here... |
| 19:18.22 | voipmonk | well |
| 19:18.26 | voipmonk | give it a whirl |
| 19:18.30 | voipmonk | stick an h in the context |
| 19:18.34 | voipmonk | and make it do a no op |
| 19:18.41 | voipmonk | noop(HAHAHAA IT WORKED) |
| 19:18.44 | voipmonk | or something |
| 19:18.46 | paulc | yeah, the h seems to trigger, the failed doesn't... docs = wrong, me = happy |
| 19:18.51 | voipmonk | and then let us know :) |
| 19:18.57 | voipmonk | cool |
| 19:19.05 | voipmonk | happy is where you want to me :) |
| 19:19.09 | voipmonk | be? |
| 19:19.18 | paulc | with 2 NoOp's in there, one on failed, one on h, only the h triggers.. bin the h off, leaving just a failed.. nothing happens.. |
| 19:19.22 | paulc | failed = old, apparently |
| 19:19.22 | voipmonk | these keys dont last long... :) ( hpmini) |
| 19:19.57 | paulc | netbook keyboard replacement time? (or are you just mashing it too hard?) |
| 19:20.34 | KAMAZ | paulc: I can't. When I assign a phone number to port on the device, it will automatically sign in to the asterisk. |
| 19:21.24 | KAMAZ | In SIP URI presents phone number |
| 19:21.29 | paulc | KAMAZ: So are you saying you can only enter digits, not alphanumeric? (you can then assign digits to that peer right?) |
| 19:21.40 | KAMAZ | yes |
| 19:21.46 | paulc | KAMAZ: eg: my phone is called phone1 but in Asterisk I say exten 1234 => Dial(SIP/phone1) |
| 19:22.56 | paulc | I definitely had issues with numeric peers in certain cases.. I think I had a phone set up as a 10 digit number, but calls into me from a provider, destined for that number, had auth problems.. because the To: address was the same as a local peer |
| 19:23.23 | KAMAZ | trouble in registration on asterisk with different username = left side on SIP URI and authname = Authorization: Digest username |
| 19:26.50 | *** join/#asterisk klochan (n=klochan@95-25-158-18.broadband.corbina.ru) |
| 19:27.11 | Baylink | So, voipmonk, maybe you have the answer to my question... |
| 19:28.31 | Alagar | hi all |
| 19:28.41 | KAMAZ | paulc: http://pastebin.com/d6b68ed6e |
| 19:28.50 | Alagar | iam new to asterisk. any one can help me with basic guide |
| 19:29.03 | paulc | KAMAZ: Give me a sec, I'll take a look |
| 19:29.12 | paulc | Alagar: What's up? Ask a question and maybe I can help |
| 19:29.13 | Baylink | Hey, alagar. You just playing around, or do you have a goal? |
| 19:31.00 | paulc | KAMAZ: What happens if you name your peer [test] instead of [baba].... or add authuser=test to your [baba] definition? |
| 19:33.07 | KAMAZ | testing model |
| 19:33.26 | KAMAZ | baba only alfa without number |
| 19:34.32 | KAMAZ | added authuser=test |
| 19:34.36 | KAMAZ | not working |
| 19:35.23 | paulc | ok - so how about changing [baba] to [test] (and not having authuser in your peer definition) |
| 19:38.10 | Alagar | paulc: sorry iam come back |
| 19:38.19 | ChannelZ | ~book |
| 19:38.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:38.50 | paulc | Alagar: Welcome back - was there a specific question you had, or did you just want to know where to find more information about Asterisk? |
| 19:39.37 | Alagar | paulc: how to integrate my asterisk linux box to my epbx |
| 19:39.56 | paulc | What is your epbx? |
| 19:42.57 | Alagar | paulc: simens |
| 19:46.17 | paulc | Alagar: Are you looking to connect Asterisk to your Siemens PBX via T1/E1? or via SIP? |
| 19:53.29 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:53.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:03.05 | *** join/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 20:03.29 | *** part/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 20:04.12 | TheNoOne | hi .. i'm using asterisk 1.4 on debian 5 .. i'm trying to get cdr logging to mysql working ... but mysql is not showing up as a registered backend |
| 20:04.19 | TheNoOne | http://pastebin.com/m6d537e7d |
| 20:05.39 | paulc | TheNoOne: If you do "make menuconfig" in your asterisk source directory, is the mysql CDR piece available to be built? |
| 20:06.00 | TheNoOne | i didn't compile anything |
| 20:06.04 | TheNoOne | just used apt-get packages |
| 20:06.11 | TheNoOne | from debian 5 |
| 20:06.33 | paulc | Ah.. hmm.. gut feel says you may need to compile from source (and may also require MySQL client source code on the box too) |
| 20:07.45 | TheNoOne | but there is exactly a asterisk-mysql package ... |
| 20:08.17 | paulc | What does that package provide you, functionality wise? |
| 20:10.27 | TheNoOne | cdr_addon_mysql.so, app_addon_sql_mysql.so and res_config_mysql.so |
| 20:11.55 | TheNoOne | http://packages.debian.org/lenny/comm/asterisk-mysql |
| 20:14.17 | paulc | Have you restarted asterisk since installing the package? |
| 20:14.39 | paulc | also, does /etc/asterisk/cdr.conf help you any (in terms of configuration etc?) |
| 20:14.56 | paulc | clutches at straws |
| 20:17.11 | TheNoOne | yes |
| 20:17.38 | TheNoOne | in cdr.conf i have |
| 20:17.38 | TheNoOne | [mysql] |
| 20:17.39 | TheNoOne | enable=yes |
| 20:18.08 | leifmadsen | TheNoOne: you installed the cdr_mysql driver from -addons? |
| 20:18.27 | leifmadsen | reads the scrollback and realizes that's already been mentioned :) |
| 20:19.15 | TheNoOne | i have cdr_addon_mysql.so loaded correctly |
| 20:21.56 | TheNoOne | mysql should be appearing as registered backend in cdr ... right? |
| 20:24.20 | drmessano | likes ODBC better |
| 20:24.25 | TheNoOne | i should do mysql via unixodbc? |
| 20:24.53 | drmessano | I've been told the ODBC in core is better than the mysql connector in addons |
| 20:24.57 | drmessano | and of course, more flexible |
| 20:25.27 | TheNoOne | do you also use it for realtime configuration? |
| 20:25.42 | drmessano | No, CDR |
| 20:32.45 | *** join/#asterisk niekniek (n=somebody@5ED25657.cable.ziggo.nl) |
| 20:34.01 | niekniek | Hello! |
| 20:34.10 | niekniek | I've got a question |
| 20:34.52 | niekniek | anyone knows the CID update when doing attended transfer problem? |
| 20:41.04 | drmessano | huh? |
| 20:42.02 | ChannelZ | it's not a problem per se, it's just how it works |
| 20:42.44 | ChannelZ | Because it's attended, person A really IS the one calling person B, not person C (the transferee) |
| 20:42.47 | *** join/#asterisk neurosys (n=neurosys@wireless-gw.cs.fiu.edu) |
| 20:48.46 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
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| 21:26.35 | *** part/#asterisk ruied (n=ruied@bl7-213-7.dsl.telepac.pt) |
| 21:35.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:47.08 | niekniek | Sorry, was away |
| 21:47.43 | niekniek | Channelz: what you're saying is correct behaviour, but in real life everyone wants it |
| 21:48.09 | niekniek | so we're trying to work around it... |
| 21:48.15 | niekniek | aastra phones have a feature for it |
| 21:48.22 | niekniek | only asterisk is not supporting it |
| 21:48.34 | niekniek | it's said it will be there in 1.6.3 |
| 21:52.11 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 22:03.33 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 22:06.48 | ChannelZ | I just wish my SPAs would let me move attended transfer off the main screen and put blind transfer there instead |
| 22:13.44 | *** join/#asterisk catojo (n=catojo@189.24.67.88) |
| 22:13.52 | paulc | Are the newer SPAs better than the 941's etc? or are those what you're using? |
| 22:14.18 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 22:16.18 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 22:18.49 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:3817:3376:a077:acde) |
| 22:18.53 | cusco | hi |
| 22:19.13 | cusco | is there a default prefix that telecom would use to dial as private number? |
| 22:20.32 | *** join/#asterisk hluesea (n=hulusika@88.239.38.225) |
| 22:24.50 | ChannelZ | the 922 is what I'm using |
| 22:25.10 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
| 22:33.35 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 22:34.24 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 22:34.40 | paulc | cusco: Do you mean a default prefix for withholding your caller ID? so it shows as private to the person you're calling? |
| 22:35.00 | paulc | ChannelZ: Not bad little phones, but I love my Polycoms :) |
| 22:35.09 | paulc | playing with a couple of Aastra's too, they're not bad |
| 22:35.15 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 22:36.14 | *** join/#asterisk bamsefar (i=emj@core.serv.emj.se) |
| 22:36.15 | *** part/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 22:36.20 | cusco | paulc: yes |
| 22:36.22 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 22:36.30 | bamsefar | Hi, I get this error [Jan 16 23:35:46] WARNING[6002]: app_meetme.c:866 build_conf: Unable to open pseudo device |
| 22:36.52 | bamsefar | AFter a kernel upgrade, /proc/zap/pseudo exists. |
| 22:37.02 | bamsefar | I don't know what is wrong. |
| 22:38.04 | paulc | cusco: Depends where you are. In North America, *67 is the standard code. In the UK you dial 141 before the number. The GSM standard is #31# and this works on lots of european landlines too. |
| 22:39.29 | cusco | I know about the gsm standart |
| 22:39.34 | cusco | im in portugal... |
| 22:39.39 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.13.dyn.user.ono.com) |
| 22:39.40 | cusco | ok I will try those |
| 22:39.56 | Godfather_ | o/ |
| 22:40.01 | bamsefar | Any suggestions for getting asterisk to read my pseudo device? |
| 22:40.23 | paulc | In VoIP it's less about the codes (although your provider may support them) and more about flags set in SIP headers (which may be controlled by your ATA, phone, or dialplan configuration) |
| 22:41.00 | paulc | bamsefar: I don't know - maybe rebuild zaptel (did you build from source originally/previously?) |
| 22:41.09 | bamsefar | paulc: Yeah |
| 22:42.33 | paulc | try a rebuild from source, because the kernel change may have messed up modules etc? |
| 22:42.46 | bamsefar | I did try rebuild |
| 22:44.05 | cusco | paulc: none of them seem to work |
| 22:44.05 | cusco | :/ |
| 22:44.42 | bamsefar | paulc: I get a [Jan 16 23:43:20] WARNING[7754] file.c: Failed to write frame |
| 22:44.50 | bamsefar | after the unable to open pseudo device. |
| 22:47.07 | paulc | cusco: what device are you using, with which provider? |
| 22:47.20 | paulc | bamsefar: I've had some of those before, on a box with no timing hardware in |
| 22:47.34 | paulc | bamsefar: would the latest version of Asterisk + ConfBridge (intead of MeetMe) be of any use? |
| 22:48.34 | bamsefar | paulc: Uhm, we have a working setup and I just rebooted this box for updates. Now conferencing is dead. |
| 22:50.02 | cusco | paulc: what do you mean device, asterisk? |
| 22:50.07 | cusco | ONI portugal |
| 22:50.43 | paulc | cusco: no, what phone or ATA are you using? |
| 22:51.17 | paulc | bamsefar: hmm.. I don't know what to suggest :( a recompile and make clean of the zaptel drivers didn't work, no? |
| 22:51.51 | cusco | paulc: sorry.. what do you mean by ata? |
| 22:52.08 | cusco | we have a digium pri card, and primary services from Oni |
| 22:52.30 | bamsefar | paulc: Doesn't seem so. |
| 22:52.30 | cusco | I will contact them in the future asking them, just they don't take calls on saturday/sunday |
| 22:52.33 | cusco | :p |
| 22:53.27 | cusco | wallops from freenode |
| 22:54.16 | paulc | cusco: Ah, ok - PRI |
| 22:54.19 | cusco | oops |
| 22:54.23 | cusco | yes, pri |
| 22:54.28 | paulc | look at the asterisk privacy screening settings (in the dialplan) |
| 22:54.34 | paulc | with ISDN you should be able to set the bits yourself |
| 22:54.47 | cusco | where can I read more about it? |
| 22:54.56 | paulc | or if you're using dialplan/numbering scheme as unknown/unknown, you shoudl be able to dial whatever they let normal landline customers use |
| 22:55.01 | paulc | *31* or #31# etc |
| 22:55.20 | cusco | that works on mobile |
| 22:55.23 | cusco | not on fixed lines tho |
| 22:55.32 | cusco | at least I just tried with asterisk |
| 22:55.38 | paulc | So if you're at home, normal analog landline - what do you dial to block your number? |
| 22:55.49 | cusco | I wouldn't know |
| 22:55.51 | cusco | :p |
| 22:56.12 | cusco | let me read aboud the dial privacy screenings |
| 22:57.10 | cusco | is it just the p? |
| 22:57.36 | paulc | cusco: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerPres |
| 22:57.52 | paulc | cusco: or maybe http://www.voip-info.org/wiki/view/Asterisk+cmd+CallingPres |
| 22:58.03 | cusco | ah right |
| 22:58.13 | cusco | thanks |
| 22:58.22 | cusco | I don't know the bits tho |
| 23:00.38 | cusco | Set(CALLERPRES()=unavailable); didn't work :( |
| 23:00.59 | cusco | ok I will phone up telco |
| 23:01.14 | paulc | cusco: try the prohib options |
| 23:01.20 | cusco | ok |
| 23:01.47 | paulc | When I was in the UK, you could only pass caller ID that was part of your DDI range, otherwise it would default to your pilot number |
| 23:01.59 | cusco | nope |
| 23:02.00 | paulc | the prohib options will work better than unavailable - that's more internal to the telco, I think |
| 23:02.24 | cusco | ah... |
| 23:06.43 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 23:07.24 | cusco | well it doesn't work |
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| 23:16.50 | paulc | cusco: none of the prohib options work? yeah - time for a call to the telco.. they may not allow you to do it right now, need to change your programming to permit the feature |
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| 23:41.54 | ghuru | hello all |
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| 23:42.21 | ghuru | i was just wondering if there was anyone that is using MajicJack on his/her asterisk |
| 23:43.08 | ghuru | i have been trying like crazy to obtain the SIP credentials from my majicjack but unfortunately havent been able to |
| 23:43.34 | paulc | network sniffs don't show it? |
| 23:43.43 | paulc | I guess not for the auth etc |
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