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| 00:07.34 | mazpe | anyone experimenting with a2billing 1.4? |
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| 00:27.22 | Nasra | hello all ! |
| 00:27.23 | *** part/#asterisk `paul (n=kutimoy@121.97.99.151) |
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| 00:35.44 | orkid | hmm, this is weird. I set CALLERID(all)=name <5551234> but it does not get sent! |
| 00:36.09 | orkid | it does get set it seems, since NoOp(${CALLERID(all}) displays it properly |
| 00:36.12 | orkid | what the heck? |
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| 00:53.51 | orkid | huh. looks like a sipura thing! when i call from a softphone callerid changing works, but not when I call from sipura spa3102. wow, weird |
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| 01:07.54 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:10.23 | seb- | How make X-Lite softphone try to register w/ your asterisk server after you configure it? |
| 01:10.32 | orkid | it does it automatically |
| 01:10.52 | orkid | make sure it's enabled in x-lite's settings |
| 01:11.01 | seb- | orkid: that's what the o'reilly book said |
| 01:11.55 | seb- | orkid: is NAT and different networks a problem? |
| 01:12.15 | orkid | make sure sip.conf has nat=yes for the account x-lite is registering with |
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| 01:15.15 | seb- | orkid: when does X-Lite try to automatically register? |
| 01:15.31 | seb- | orkid: be nice if it said something like "Trying..." (btw..yes i enabled it) |
| 01:17.37 | seb- | orkid: anyways..thanks for the help..i got a little farther :) |
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| 01:47.21 | NMR_1122 | Is there a "mailbox show all" command? |
| 01:47.49 | orkid | seb-: when you close the window i think. |
| 01:48.04 | orkid | seb-: as soon as it gets the account info and you close the window |
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| 01:57.11 | tzafrir_laptop | NMR_1122, voicemail show users |
| 02:04.42 | NMR_1122 | ok cool |
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| 02:40.07 | raden_work | anyone seen jeff_phillips latley ? |
| 02:42.26 | drmessano | ~drmessano |
| 02:42.27 | infobot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
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| 03:00.03 | jblack | yah engrish. |
| 03:00.25 | jblack | my kindle 2 died today. :( |
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| 03:18.40 | jaytee | ~seen ManxPower |
| 03:18.42 | infobot | manxpower <n=manxpowe@user-24-236-95-236.knology.net> was last seen on IRC in channel #asterisk, 7d 10m 42s ago, saying: 'do you have a PRI?'. |
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| 03:33.10 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 03:33.33 | RoPBX | hi all, anybody awake? |
| 03:35.50 | *** join/#asterisk pcrack (n=pcrack@125.5.159.62) |
| 03:36.26 | pcrack | can anyone tell me whats the difference between diguim and switchvox? |
| 03:39.43 | tzafrir_laptop | switchvox is a product of Digium |
| 03:39.59 | tzafrir_laptop | (was originally an independent company) |
| 03:40.16 | tzafrir_laptop | Digium does quite a few other things, though |
| 03:45.44 | *** part/#asterisk jcoffi1 (n=jcoffi@75.147.155.89) |
| 03:48.55 | RoPBX | please, anybody can tell me how to register a dummy sip trunk in asterisk? |
| 03:49.07 | RoPBX | to register to itself? |
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| 03:51.13 | jblack | You want * on localhost to register with sip, on * on localhost... ? |
| 03:52.07 | jblack | a snake, eating it's own tail? |
| 03:52.24 | RoPBX | i want a dummy sip trunk to try to deal with the problem when internet goes down |
| 03:52.28 | tzafrir_laptop | wouldn't asterisk detect the loop? |
| 03:52.28 | drmessano | I saw that once. |
| 03:52.45 | tzafrir_laptop | RoPBX, do you need to test registering? |
| 03:52.59 | tzafrir_laptop | ah, ok |
| 03:53.18 | jblack | ropbx: Well, let's look at that. What do you consider "successful dealing" ? |
| 03:53.26 | jblack | i.e., what would the user experience? |
| 03:54.12 | RoPBX | ok, i have a tdm card to make local calls on the pbx, and a sip trunk, when internet goes down asterisk locks tdm lines too |
| 03:54.13 | jblack | You pick up the phone, dial a number, your sip "trunk" is down, and what do you hear? |
| 03:54.20 | RoPBX | http://forums.whirlpool.net.au/forum-replies-archive.cfm/1058622.html |
| 03:55.00 | RoPBX | well....the dummy idea is in this link: http://forums.whirlpool.net.au/forum-replies-archive.cfm/960451.html |
| 03:55.31 | jblack | ok, so, let me rephrase that to see if I have it right. |
| 03:55.49 | RoPBX | ok |
| 03:55.53 | X-Rob | whirlpool can blow me |
| 03:56.02 | jblack | Normally, you work over sip. internet goes down, sip fails. You're _supposed_ to fail on sip and fallback to the tdm, but that's not working. |
| 03:56.06 | jblack | Is that correct? |
| 03:56.13 | RoPBX | yeap |
| 03:56.29 | jblack | Ok. There's probably easier ways to get what you want. |
| 03:56.34 | jblack | Dialplan on pastebin.com, pleae |
| 03:57.22 | [TK]D-Fender | RoPBX>please, anybody can tell me how to register a dummy sip trunk in asterisk? <- WTF? |
| 03:57.23 | X-Rob | jblack, I think he's using freepbx in the guise of elastix |
| 03:57.27 | jblack | Also, if you pastebin a "sip show peers" from asterisk -r, that could be useful. |
| 03:57.54 | jblack | If he's using freepbx, that'll become apparent, and all that's left is the "you're pbx bigots!" irc fight. |
| 03:58.00 | [TK]D-Fender | X-Rob: member of #asterisk, #elastix, and @#freepbx <-------- |
| 03:58.07 | RoPBX | anything special on my dialplan, SIP phones got unregistered |
| 03:58.13 | X-Rob | [TK]D-Fender, indeed. |
| 03:58.14 | RoPBX | nop, i-m not using freepbx |
| 03:58.20 | jblack | I want the whole thing, if you don't mind. |
| 03:58.24 | pcrack | sorry my wrong questions...what is the difference of diguim and openvox? |
| 03:58.39 | X-Rob | I've been dragged back into doing voip delvopment by large sums of money waved at me |
| 03:58.51 | jblack | Sub me out for small sums of money. :) |
| 03:59.06 | [TK]D-Fender | pcrack: Openvox ripped off Digiums old designs and still suck. They also have crappy support |
| 03:59.07 | jblack | needs a new laptop. |
| 03:59.08 | MooingLemur | The waving is delicious |
| 03:59.13 | X-Rob | jblack, I've almost finished it actually. Outbound call restrictions based on source extension |
| 03:59.30 | jblack | You got large sums for _THAT_ ? |
| 03:59.43 | jblack | large.. zimbabwe sums? |
| 04:00.00 | X-Rob | for a freepbx module |
| 04:00.00 | X-Rob | rather than doing it by hand |
| 04:00.03 | X-Rob | Heh |
| 04:00.09 | X-Rob | 23 billion dollars! |
| 04:00.14 | jblack | wtf do I have money problems?!? |
| 04:00.19 | X-Rob | about AUD$0.04 |
| 04:00.41 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 04:01.00 | jblack | RoPBX: Making progress, I hope. |
| 04:02.16 | RoPBX | sorry, dialplan is in another machine... give me a sec |
| 04:03.43 | jblack | ok. while you're at it. check for me whether or not you have qualify=yes in your sip.conf, either in general, or for your..."trunk" |
| 04:05.16 | RoPBX | nop, i don-t have it |
| 04:05.34 | jblack | That might help you out a little bit in proactively detecting it's down. |
| 04:05.40 | X-Rob | I bit the bullet and tried to figure out what they were meaning about things locking up |
| 04:05.46 | jblack | Shove a qualify=yes in there while you're at it. |
| 04:05.58 | X-Rob | RoPBX, what exactly breaks. Be as specific as possible. |
| 04:06.17 | jblack | if we get desperate, we'll make a cludge to take advantage of that to reload sip.conf (after I've asked you a couple more questions) |
| 04:06.37 | jblack | x-rob: I mostly know what's up with him. It's ok. |
| 04:06.44 | X-Rob | jblack, what's the problem? |
| 04:07.01 | X-Rob | because they're hinting around DNS issues in those whirlpool posts. |
| 04:07.02 | RoPBX | sorry, i dont speak english very well... |
| 04:07.18 | jblack | his symptom is that the sip account to the provider is down. I suspect that he's using a dyndns client. |
| 04:07.34 | X-Rob | jblack, fair enough. qualify=yes. |
| 04:07.35 | jblack | but I haven't gotten there yet. |
| 04:07.36 | X-Rob | hands jblack a beer |
| 04:07.38 | X-Rob | enjoy |
| 04:07.47 | jblack | qualify=yes should help, yeah. :) |
| 04:07.56 | jblack | But before we truly fix that, I want to fix his redundancy. |
| 04:08.13 | RoPBX | i have a pbx with sip trunks and tdm lines, when internet goes down i can't use tdm lines |
| 04:08.21 | RoPBX | its a well known problem |
| 04:08.30 | X-Rob | bites his tongue |
| 04:08.47 | jblack | x-rob: For which comment, one of mine? |
| 04:08.52 | X-Rob | No no. |
| 04:08.53 | X-Rob | his |
| 04:08.57 | jblack | yeah. :) |
| 04:08.58 | X-Rob | 'it's a well known problem' |
| 04:09.04 | [TK]D-Fender | [00:08]<RoPBX>i have a pbx with sip trunks and tdm lines, when internet goes down i can't use tdm lines <- pardon? |
| 04:09.06 | X-Rob | I'll let you beat him into shape |
| 04:09.06 | jblack | robpx: That's why I'm waiting for your dialplan. :) |
| 04:09.13 | [TK]D-Fender | that makes no sense |
| 04:09.31 | jblack | [TK]D-Fender: I think I know what's going on with that too. Just waiting for his frigen extensions.conf to verify. |
| 04:09.31 | [TK]D-Fender | Somebody is smoking some good stuff there |
| 04:09.42 | jblack | lart 'em after I'm done? :) |
| 04:10.17 | [TK]D-Fender | jblack: we'll see |
| 04:10.48 | jblack | as necessary, of course, though imho, in your good days, you'll beat someone for "sip trunk". ;) |
| 04:10.50 | X-Rob | I still think he's running freepbx. If he wasn't, he'd understand dialplans and failover and stuff. |
| 04:11.04 | Talkradio | i'm smoking some kind here :)) |
| 04:11.07 | jblack | That's probably why he hasn't presented his extensions.conf yet. |
| 04:11.15 | X-Rob | jblack, yup |
| 04:11.25 | jblack | talkradio: Please keep your sex life to yourself if you don't mind. |
| 04:11.27 | RoPBX | http://pastebin.com/m27e9f337 |
| 04:11.50 | X-Rob | ahha |
| 04:11.50 | jblack | um.. where's the rest of your dialplan? |
| 04:11.58 | X-Rob | I see the problem there |
| 04:12.08 | jblack | No, that can't be the problem. |
| 04:12.23 | RoPBX | the dialplan is not the problem, i-m reproducing the problem in a local pbx |
| 04:12.30 | jblack | He only pasted part of it. I hope. I really hope. If there's a dog, I hope. |
| 04:12.36 | jblack | ropbx: Hey dude, you want help, right? |
| 04:12.38 | [TK]D-Fender | GARBAGE |
| 04:12.48 | [TK]D-Fender | there is no failover ther, |
| 04:12.55 | jblack | ropbx: So, just play ball, and you'll get exactly what you want. |
| 04:12.58 | [TK]D-Fender | exten = _XXXXX.,1,Dial(SIP/trunk_1}/${EXTEN}) <- broken variable reference |
| 04:12.58 | X-Rob | cheers on [TK]D-Fender |
| 04:13.21 | jblack | RoPBX: Now, may we have have the entire dialplan, please? |
| 04:13.35 | X-Rob | Aaw, this is why I never get any work done |
| 04:13.44 | X-Rob | goes back to doing productive stuff. |
| 04:14.21 | jblack | [TK]D-Fender: Think he typed it in by hand, rather than using rsync, or cut-n-paste? |
| 04:14.23 | RoPBX | http://pastebin.com/m263077ff |
| 04:14.36 | RoPBX | no, im not using failover between sip and tdm lines |
| 04:14.40 | jblack | THANK you. |
| 04:15.15 | jblack | Ok, now, you want calls to go out over the TDM if they fail on sip, right? |
| 04:15.21 | RoPBX | nop |
| 04:15.36 | X-Rob | hahahahah |
| 04:15.41 | X-Rob | *rofl* |
| 04:15.45 | X-Rob | sorry. I looked |
| 04:16.00 | [TK]D-Fender | RoPBX: What are you trying to do. |
| 04:16.01 | [TK]D-Fender | ? |
| 04:16.02 | RoPBX | i want to be able to use tdm line even if i cant use SIP trunks because internet is down |
| 04:16.18 | jblack | ?!? |
| 04:16.25 | jblack | How is that different than what I said? |
| 04:16.42 | X-Rob | jblack, may I point out - that's freepbx. |
| 04:16.58 | [TK]D-Fender | X-Rob: This is not FreePBX. |
| 04:17.00 | jblack | Yeah. It is. |
| 04:17.18 | X-Rob | [TK]D-Fender, tis. Honest. |
| 04:17.27 | [TK]D-Fender | X-Rob: No, this is Asterisk-GUI |
| 04:17.48 | X-Rob | [TK]D-Fender, aah. OK. Must be based on freepbx. I've been out of the loop for too long. |
| 04:18.01 | jblack | Point being, it's not vanilla *. |
| 04:18.04 | [TK]D-Fender | X-Rob: No, it isn't. It is completely differenct |
| 04:18.27 | RoPBX | jblack, if i dont have internet i want to be able to make LOCAL calls, i don't mind SIP calls |
| 04:18.31 | jblack | So, first he lies about what pbx he's using, then he hides his diaplan to hide the lie, then he pastes the whole thing. |
| 04:18.42 | X-Rob | [TK]D-Fender, ok, cool. Sorry. Looked like it, but yeah, it's NOT a freepbx dialplan. |
| 04:18.51 | jblack | ropbx: That's what we call "fallback". "Try it this way, if it fails, fall back to this other way". |
| 04:19.01 | [TK]D-Fender | RoPBX: See where you dial out your SIP provider? check the DIALSTATUS to see if it failed and then go dial out your zap channels instead |
| 04:19.24 | jblack | ropbx: Anyways, you're in the wrong place. Despite what you said, this isn't #asterisk. You should be in #asterisk-now , or #freepbx, or wherever else it is for what you're actually running. |
| 04:19.32 | RoPBX | when internet is down i dont even get anything on asterisk CLI |
| 04:19.39 | jblack | Short of more money than you have, there's no way I'm going to detangle all those macros. |
| 04:20.04 | X-Rob | jblack, it LOOKS like his extension's context is DLPN_DialPlan1 |
| 04:20.40 | X-Rob | and there's a macro-trunkdial-failover, and it is checking the dialstatus.. |
| 04:20.41 | NovceGuru | spaghettiplan() |
| 04:20.50 | RoPBX | you ask me the entire dialplan! |
| 04:20.56 | [TK]D-Fender | RoPBX: What are you calling in from? |
| 04:21.01 | jblack | becuase you said it was _asterisk_, which it's _NOT_ |
| 04:21.14 | X-Rob | RoPBX, because that complete dialplan is a hell of a lot different from what you posted originally. |
| 04:21.45 | RoPBX | you havent understand the problem :( |
| 04:21.53 | X-Rob | RoPBX, yes, we do. |
| 04:21.59 | X-Rob | You don't understand the solution. |
| 04:22.04 | jblack | I'll give you this much. You need to fix up that trunkdial-failover-0.3 macro so that it also takes a dest technology as an argument. |
| 04:22.18 | X-Rob | the problem is: you don't have any failover in your dialplan |
| 04:22.26 | X-Rob | the solution is: put failover in your dialplan |
| 04:22.33 | jblack | then, add another rule under CallingRule_todas and callinRule_cantv with the extra dial line. |
| 04:22.40 | RoPBX | noooooo i don't want to failover! |
| 04:22.50 | X-Rob | ok |
| 04:22.51 | X-Rob | actually |
| 04:22.51 | X-Rob | sorry |
| 04:22.55 | X-Rob | I am getting sidetracked her |
| 04:22.59 | jblack | In asterisk, without all that sludge, it's a 30 second thing. With all that sludge.. shrug. |
| 04:23.06 | X-Rob | he's saying that when he doesn't have internet connectivity, his zaptel stuff isn't working |
| 04:23.35 | X-Rob | RoPBX, right. So when you say 'no internet connectivity' can you ping the phone from the asterisk box? |
| 04:23.46 | RoPBX | i have two nics, one to a LAN with local SIP phones, and the other to the internet |
| 04:23.54 | jblack | That Dial(${ARG1}) seems odd. |
| 04:24.07 | RoPBX | when internet goes down, i cant call even between local SIP phones |
| 04:24.15 | X-Rob | RoPBX, right. So when you say 'no internet connectivity' can you ping the phone from the asterisk box? |
| 04:24.22 | RoPBX | yeap |
| 04:24.30 | RoPBX | but i can' make calls |
| 04:24.40 | X-Rob | and nothing appears on the asterisk console |
| 04:24.47 | jblack | x-rob: He wants to use TDM when sip isn't available.. but he doesn't want failover. Why can't you get it? |
| 04:24.50 | jblack | snickers |
| 04:24.57 | [TK]D-Fender | ..... |
| 04:25.02 | X-Rob | jblack, he's saying that when his internet connection is down, NOTHING works. |
| 04:25.09 | [TK]D-Fender | grabs his ClueBat (tm) |
| 04:25.11 | RoPBX | nothing appears on the asterisk console |
| 04:25.26 | jblack | You mean the "fork of asterisk" console? |
| 04:25.39 | [TK]D-Fender | RoPBX: Can you SSH into the machine when this happens? |
| 04:25.47 | [TK]D-Fender | RoPBX: How about ping? |
| 04:25.48 | X-Rob | At a guess - his SIP phones are connecting to the IP address of the interface that's down? |
| 04:26.01 | RoPBX | yes, i can ssh |
| 04:26.03 | [TK]D-Fender | RoPBX: When you DO get that far can you get to * CLI? |
| 04:26.04 | X-Rob | oooh |
| 04:26.12 | [TK]D-Fender | X-Rob: Excellent guess |
| 04:26.28 | [TK]D-Fender | RoPBX: Indeed, what IP are your phones pointed to?> |
| 04:26.33 | jblack | yeah, you beat me. ssh'ing in on the down interface. |
| 04:26.43 | RoPBX | of course the other interface, |
| 04:26.53 | [TK]D-Fender | RoPBX: Show us |
| 04:27.00 | X-Rob | It could also be DNS issues - if he's got a hostname as his SIP peer, asterisk locks up while it's doing DNS lookups (..well, it used to - is DNS threaded now?) |
| 04:27.12 | jblack | watches [TK]D-Fender Help with a decidedly-not-straight-asterisk problem |
| 04:27.24 | X-Rob | jblack, this _is_ an asterisk problem |
| 04:27.34 | X-Rob | if it's not making it to the dialplan, it's asterisk's problem. |
| 04:27.55 | jblack | You could only assert that if A) you knew which "*" he was actually using and B) You knew what code had diverged. |
| 04:28.18 | [TK]D-Fender | X-Rob: The internet is down! Must be *'s fault! |
| 04:28.39 | X-Rob | RoPBX, in your sip.conf - do you have an IP address on the host= line, or is it an IP address? |
| 04:29.10 | RoPBX | TKD-fender, what do you want me to show? |
| 04:29.14 | [TK]D-Fender | X-Rob: Phones should always be dynamic |
| 04:29.19 | jblack | x-rob: incidentally, * here doesn't lock up when ppp0 (the interface that * sits on here) is down. |
| 04:29.36 | [TK]D-Fender | RoPBX: Show us your phone's configs and their current registration status |
| 04:29.54 | [TK]D-Fender | jblack: Same here |
| 04:30.08 | *** join/#asterisk CrashSys (n=james@azrael.crashsys.com) |
| 04:30.17 | X-Rob | [TK]D-Fender, for the sip trunks |
| 04:30.24 | X-Rob | not the phone |
| 04:30.30 | X-Rob | jblack, it doesn't for me, either. |
| 04:30.41 | [TK]D-Fender | X-Rob: Who cares about the trunks.. no internet = failure for those. Whats the poitn? |
| 04:30.46 | X-Rob | but it's about the only thing I can think of that DOES rely on the net being up, and can cause delays |
| 04:30.52 | RoPBX | 6001/6001 192.168.0.171 D 5060 Unmonitored |
| 04:30.52 | RoPBX | 6000/6000 192.168.0.180 D 5070 Unmonitored |
| 04:30.58 | jblack | So, despite knowing better yourself, you state "asterisk -r for him" is an asterisk problem? |
| 04:31.12 | jblack | when we already know he's not running vanilla asterisk? |
| 04:31.16 | RoPBX | thats my 2 phones status |
| 04:32.16 | jblack | hrmmm. I threw some whisky into my wendy's cup. |
| 04:32.26 | jblack | Is it wrong to drink $250 whiskey with a straw? |
| 04:33.04 | *** join/#asterisk dandate2 (n=dandate2@173.11.82.122) |
| 04:33.25 | [TK]D-Fender | RoPBX: So the internet goes down. You enabe; SIP debug at CLI, and you're saying that your phones can't contact * period? |
| 04:33.31 | jblack | considers writing a minesweeper for * |
| 04:33.35 | dandate2 | is it recommended to use a server machine over a desktop machine to host a pbx that is required to run 25 channels? |
| 04:34.18 | X-Rob | [TK]D-Fender, Hah. Get him to show you his sip.conf. |
| 04:34.24 | jblack | dandate2: Is it ok to use a desktop as a * server for 25 channels? |
| 04:34.25 | X-Rob | has figured out what the problem is |
| 04:34.39 | [TK]D-Fender | jblack: I wrote a minesweeper game myself once about 15 years ago |
| 04:34.53 | jblack | Yeah, but I'm talking about for * ! |
| 04:35.07 | RoPBX | when internet is down i see this: |
| 04:35.27 | RoPBX | chan_sip.c:7487 sip_reregister: |
| 04:35.40 | dandate2 | right my agents have been getting chop and i can't believe it might be bandwidth issues |
| 04:35.41 | jblack | how could you see that if "asterisk -r is frozen" ? |
| 04:35.53 | RoPBX | --Re-registration for xxxxxx@xx.x..... |
| 04:36.06 | jblack | dandate2: Could be latency, dropped packets, comcast deciding they don't like competition.... |
| 04:36.13 | X-Rob | has /msg'ed [TK]D-Fender the solution. |
| 04:36.19 | NovceGuru | damn I probably shouldnt upgrade this box since I closed the closet door before I left |
| 04:36.19 | dandate2 | we just instilled g729 and i was afraid that my processor wasn't handling it |
| 04:36.21 | [TK]D-Fender | RoPBX: Yes, your registration to your ITSP fails. WE FUCKING GET THAT. No internet = things over the internet = UNREACHABLE |
| 04:36.30 | jblack | dandate2: it could possibly be cpu latency too, if the machine is doing other things. |
| 04:36.38 | [TK]D-Fender | RoPBX: No reason your LOCAL phones can't contact * |
| 04:36.48 | *** join/#asterisk hyerba (n=abc@cm83.sigma224.maxonline.com.sg) |
| 04:37.07 | jblack | dandate2: But cpu latency is usually less likely than bandwidth unless you've multi-purposed the box. |
| 04:37.11 | [TK]D-Fender | dandate2: Decent P4 will do fine |
| 04:37.17 | NovceGuru | dandate2: did you get your call center up and running off your home connection |
| 04:37.18 | hyerba | i'm getting a red alarm on my te121p digium card. the line is ok so i'm guessing its my config. pls help |
| 04:37.21 | RoPBX | i know, but that is the message i see when i try to make a local call |
| 04:37.34 | jblack | dandate2: At which time I'd have to say "wtf are you doing multi-purposing a box with a real-time process like *" |
| 04:37.35 | dandate2 | yes novceguru but i'm using business class docsis now |
| 04:38.33 | dandate2 | oh not i'm not multi-purposing it is has a dedicated machine on an old computer |
| 04:38.33 | jblack | Ok. |
| 04:38.33 | jblack | What's your load average? |
| 04:38.33 | dandate2 | 25 channels with g729 |
| 04:38.33 | dandate2 | is the max |
| 04:38.33 | [TK]D-Fender | RobertLaptop: Show us your attempt. BEcause right now your config does not CHOOSE what to dial out of. |
| 04:38.33 | NovceGuru | dandate2: are you using bare asterisk? |
| 04:38.33 | dandate2 | average prolly 10-15 |
| 04:38.33 | dandate2 | its freepbx |
| 04:38.33 | hyerba | hi ppl |
| 04:38.33 | NovceGuru | define: old computer |
| 04:38.35 | jblack | Then yeah, probably not enough cpu. |
| 04:38.43 | dandate2 | how do i tell my machines speed from the console? |
| 04:38.50 | NovceGuru | what OS |
| 04:38.55 | hyerba | is there anyone who can help me with zaptel using digium cards? |
| 04:38.56 | jblack | Why did you come to #asterisk for a freepbx question. Did it occur to you to go to... I dunnoo... #freepbx? |
| 04:38.59 | dandate2 | centOS i believe |
| 04:39.11 | NovceGuru | probbaly cat /proc/cpuinfo or something |
| 04:39.24 | dandate2 | well it was a question about the requirements of phone servers in general heh |
| 04:39.26 | RoPBX | well, i came here asking for a dummy sip trunk, because i found that it could be a solution, because asterisk tries to find a registered sip trunk |
| 04:39.26 | NovceGuru | gives answer jbot style |
| 04:39.35 | X-Rob | RoPBX, no, it doesn't. |
| 04:39.36 | RoPBX | is there a way to do that? |
| 04:39.44 | X-Rob | it doesn't try to find a sip trunk |
| 04:40.03 | X-Rob | the problem is that somethign is wrong with your SIP configuration and it's breaking when your internet connection is down |
| 04:40.06 | hyerba | hello ppl. hope some1 can help me |
| 04:40.14 | [TK]D-Fender | RoPBX: Your probelm is that your dialplan is DUMB. It hits a pattern match and only tries to go out your SIP provider. It does not look for alternatives. |
| 04:40.36 | NovceGuru | hyerba: are you ircing from a t9 phone |
| 04:40.42 | [TK]D-Fender | hyerba: Asks a specific question and you might get a specific answer. |
| 04:40.49 | X-Rob | [TK]D-Fender, nono. That's not his problem. Well. That's ONE of his problems. He's saying that when eth0 (internet) isn't working for whatever reason, he can't make calls between two phones on eth1 |
| 04:41.05 | hyerba | i'm getting a red alarm on my te121p digium card. the line is ok so i'm guessing its my config. pls help |
| 04:41.11 | tzafrir_laptop | NovceGuru, it's actually infobot |
| 04:41.15 | [TK]D-Fender | X-Rob: O RLY? Show me SIP debug |
| 04:41.18 | X-Rob | hyerba, digium offer free support for that |
| 04:41.24 | X-Rob | [TK]D-Fender, that's what he's saying. |
| 04:41.29 | [TK]D-Fender | X-Rob: If he can SSH, the F-ing packets are making it to * |
| 04:41.32 | RoPBX | No, my dialplan works fine, the only problem is when internet is down |
| 04:41.36 | jblack | hyerba: I've done zaptel all of once. |
| 04:41.41 | X-Rob | [TK]D-Fender, I msged you two reasons why he would be having that problem. |
| 04:41.49 | hyerba | and where do i get that free support? |
| 04:41.49 | dandate2 | ok i went to /proc/cpuinfo |
| 04:41.54 | X-Rob | hyerba, call digium |
| 04:42.03 | X-Rob | www.digium.com |
| 04:42.04 | tzafrir_laptop | at this time of day? |
| 04:42.13 | RoPBX | that problem is not happening only to me |
| 04:42.14 | jblack | I don't have the configuration available any more. verify with the phone provider the signalling type and such. |
| 04:42.27 | dandate2 | says i have 1 processor authenticAMD 64 x2 dual core processor 4400+ cpu Mhz 2310.616 |
| 04:42.33 | X-Rob | well. go to bed, and call them tomorrow then 8) |
| 04:42.42 | hyerba | yeah i did. the connection is ok and config is also ok |
| 04:42.45 | [TK]D-Fender | dandate2: Considerably more than fine |
| 04:42.58 | tzafrir_laptop | hyerba, you can also get a red alarm if your line settings are incorrect |
| 04:42.59 | jblack | tzafrir_laptop: Well, gotta keep those lazy types busy, no? :P |
| 04:43.02 | hyerba | basically i'm looking at zaptel.conf and running zttool |
| 04:43.11 | tzafrir_laptop | what do you have in zaptel.conf ? |
| 04:43.14 | X-Rob | [TK]D-Fender, to be specific, the packets are making it to _the machine_. |
| 04:43.22 | NovceGuru | dandate2: should be able to handle 25 g729 calls, but not well edumacated in that area |
| 04:43.25 | hyerba | span=1,1,0,ccs,hdb3,crc4 |
| 04:43.25 | hyerba | bchan=1-15,17-31 |
| 04:43.25 | hyerba | dchan=16 |
| 04:43.26 | X-Rob | stupid configs will break that |
| 04:43.41 | X-Rob | hyerba, that's E1. Did you change the jumper on the card from T1 to E1? |
| 04:43.45 | dandate2 | ok so the computer is fine, i am thinking it is possible that my router is not up to par, how do I check the maximum send/receive speed through the LAN? |
| 04:43.46 | jblack | dandate2: If you have the bandwidth, try dropping to something not cpu intensive. perhaps ulaw. |
| 04:43.48 | hyerba | yes i did |
| 04:44.04 | tzafrir_laptop | hyerba, was ztcfg run successfully? |
| 04:44.08 | [TK]D-Fender | dandate2: LAN is not an issue |
| 04:44.09 | dandate2 | ehh i don't have the bandwidth for ulaw though, all calls are made over the WAN and not the LAN |
| 04:44.24 | NovceGuru | speetest.net |
| 04:44.27 | X-Rob | hyerba, 'pri intense debug span 1' |
| 04:44.33 | dandate2 | i have reps in the phillipines talking to epople in washington dc through my pbx |
| 04:44.44 | NovceGuru | what speed is your cnxn |
| 04:44.50 | jblack | so... you don't have enough CPU, and you're pushing close on bandwidth too.... ? |
| 04:44.54 | X-Rob | see if you're receiving anything from the remote end. Unlikely, being red. |
| 04:44.58 | dandate2 | 16 mb down 5 up or something like that |
| 04:45.05 | NovceGuru | that will do 25 calls @ ulaw |
| 04:45.06 | tzafrir_laptop | X-Rob, that's layer 2. RED is layer 1 |
| 04:45.08 | jblack | If you're hitting more than 50% load on your network, you'll start dropping packets. |
| 04:45.21 | hyerba | well i did run ztcfg but it didnt output anything |
| 04:45.34 | tzafrir_laptop | which means: ok |
| 04:45.41 | X-Rob | tzafrir_laptop, good point.. |
| 04:45.44 | X-Rob | Cabling issue then. |
| 04:45.44 | NovceGuru | *cough*colo*cough |
| 04:45.45 | hyerba | anyway i've checked with my telco and my data centre and they're saying the line is ok |
| 04:46.03 | X-Rob | hyerba, make a crossover cable. See if you need the polarity the other way round |
| 04:46.14 | X-Rob | google pri crossover cable |
| 04:46.15 | hyerba | yeah i thot that too. i have another system with dialogic cards and the cable works fine with that |
| 04:46.21 | hackeron | I have a couple of wavs for my music on hold and I use mode=files -- the rate of the wav is 8000 but some reason the files play very slow and quiet |
| 04:46.25 | hackeron | any ideas? |
| 04:46.26 | X-Rob | dead card then |
| 04:46.40 | X-Rob | hackeron, you in .au? |
| 04:47.01 | hackeron | X-Rob: just a 8000 rate wav |
| 04:47.09 | X-Rob | wups, sorry |
| 04:47.13 | hyerba | it's a brand new card |
| 04:47.13 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 04:47.19 | X-Rob | I mean hyerba - are you in .au? |
| 04:47.30 | X-Rob | hackeron, you have to make them in a specific way |
| 04:47.41 | X-Rob | the documentation is on voip-info.org somewhere |
| 04:47.43 | hackeron | X-Rob: is there a sox command? |
| 04:47.47 | hyerba | i'm in sg |
| 04:47.51 | hyerba | singapore |
| 04:48.00 | X-Rob | hyerba, hmm. I'd contact your local reseller |
| 04:48.04 | X-Rob | they should have support |
| 04:48.10 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 04:48.38 | jblack | googles for when 729 expires |
| 04:49.08 | tzafrir_laptop | the patent? |
| 04:49.20 | tzafrir_laptop | (patents?) |
| 04:49.25 | jblack | patens. |
| 04:49.38 | jblack | actually. THe last patent. :) |
| 04:51.51 | jblack | wtf? g.729 was granted in '07 ? |
| 04:51.52 | hackeron | X-Rob: sox input.wav -t ul -r 8000 -b -c 1 output.pcm worked |
| 04:52.04 | X-Rob | there you go |
| 04:52.13 | hackeron | :) |
| 04:52.39 | jblack | By 2027, who's gonna have a phone? We'll all be in rocketships, doing telepathy |
| 04:53.09 | tzafrir_laptop | http://www.itu.int/ipr/IPRSearch.aspx?iprtype=PS |
| 04:53.14 | [TK]D-Fender | jblack: Oh? Where's the flying cars & robot maid I've been promised already? HUH!? |
| 04:53.44 | hackeron | btw, asterisk used to come with some lovely music on hold a couple of years ago, what happened to that music? :( |
| 04:53.52 | X-Rob | www.realdoll.com, [TK]D-Fender |
| 04:53.54 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 04:53.57 | X-Rob | dunno about the flying cars. |
| 04:54.06 | drmessano | digs up some "welcome to the nineteen hundred and ninety-one.. the year of the spacecloudman" video |
| 04:54.08 | [TK]D-Fender | jblack: We're alsmost 2010 and cars haven't changed worth shit. Computers haven't really changed that much either. Sure a bit bigger and faster but nothing life changing |
| 04:54.28 | tzafrir_laptop | hackeron, it's still there |
| 04:54.35 | drmessano | Jsut wait until 1980! |
| 04:54.42 | hackeron | tzafrir_laptop: hmm? - the ubuntu package just strips it out? |
| 04:54.43 | RoPBX | hi again, i-m making test with my internet-down problem, and if i took the registration of the sip trunks the problem is resolved, but i can't get incoming calls to the sip trunks |
| 04:54.46 | drmessano | Rockets for everyone |
| 04:54.47 | tzafrir_laptop | in a separate tarball now |
| 04:54.52 | [TK]D-Fender | 'Cause tonight we're going to party like its 1699! |
| 04:55.00 | tzafrir_laptop | hackeron, right. its license is limiting |
| 04:55.04 | drmessano | Well, that was kinda true, if you were libyan |
| 04:55.13 | tzafrir_laptop | but you can get it in a separate tarball anyway |
| 04:55.20 | X-Rob | RoPBX, change the hostname to an IP address |
| 04:55.24 | hackeron | tzafrir_laptop: I see - but can it be used as hold music in a production environment, or should I buy my own really? |
| 04:55.41 | RoPBX | im using ip address, not hostname |
| 04:55.44 | tzafrir_laptop | http://downloads.digium.com/pub/telephony/sounds/ |
| 04:55.52 | X-Rob | RoPBX, well. That's interesting. |
| 04:56.04 | tzafrir_laptop | The asterisk-moh-freeplay tarballs . Get the format you prefer |
| 04:56.09 | RoPBX | i saw that maybe this problem is resolved in asterisk 1.6 |
| 04:56.19 | X-Rob | RoPBX, upgrade to the latest version of asterisk. |
| 04:56.28 | X-Rob | compile a clean build yourself |
| 04:56.36 | hackeron | tzafrir_laptop: thanks, but is it legal to use that in a production environment? |
| 04:56.54 | tzafrir_laptop | hackeron, I'm not completely positively sure about the license of those files for commercial use |
| 04:57.01 | RoPBX | Ok X-Rob |
| 04:57.15 | X-Rob | tzafrir_laptop, I had a pile of moh stuff on aussievoip that was pure public domain |
| 04:57.23 | tzafrir_laptop | anyway, anything that is either in the public domain or is CC-by-sa, CC-by should be ok |
| 04:57.43 | X-Rob | http://aussievoip.com/wiki/index.php?page=MOH |
| 04:58.29 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:58.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 05:03.31 | hackeron | tzafrir_laptop: ok, I'll look at the license, thanks :) |
| 05:03.59 | hackeron | tzafrir_laptop: I like how the first line is you cannot use this with any other PBX - promising start ;) |
| 05:08.32 | X-Rob | ok |
| 05:08.41 | X-Rob | I just FIXED those sound files which had fallen off the server |
| 05:08.46 | X-Rob | they're back up and in place |
| 05:08.48 | X-Rob | http://aussievoip.com/wiki/index.php?page=MOH |
| 05:10.31 | seb- | how delete user that registered (sip) ? |
| 05:11.09 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 05:14.46 | p1mrx | can someone explain how this could happen?: |
| 05:14.48 | p1mrx | <PROTECTED> |
| 05:14.48 | p1mrx | <PROTECTED> |
| 05:15.10 | X-Rob | p1mrx, incorrect codec? check /var/log/asterisk/full and set debug to 9 |
| 05:15.11 | p1mrx | I'm trying to make asterisk answer an incoming gizmo call, but it seems to die as soon as it hits Answer() |
| 05:15.12 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 05:15.36 | seb- | X-Rob: how just kick every off your asterisk and reset? |
| 05:15.50 | X-Rob | seb-, type 'reset' in the asterisk console? |
| 05:15.56 | X-Rob | uh |
| 05:15.57 | seb- | X-Rob: rebooting * doesn't remove stuff from "sip show peers" |
| 05:15.58 | X-Rob | 'reload' even |
| 05:16.10 | X-Rob | seb-, uh. Well take them out of your sip.conf then |
| 05:16.25 | seb- | *CLI> reset |
| 05:16.25 | seb- | No such command 'reset' (type 'help' for help) |
| 05:16.34 | seb- | X-Rob: that's what reset did |
| 05:16.48 | X-Rob | seb-, I corrected myself. I then said 'take them out of your sip.conf' |
| 05:16.54 | X-Rob | which is actually the correct answer |
| 05:17.02 | leifmadsen | 'reload' |
| 05:17.17 | leifmadsen | 'module reload chan_sip.so' to reload just chan_sip |
| 05:17.34 | seb- | X-Rob: how come killing asterisk process doesn't remove crud from "sip show peers" list? |
| 05:17.48 | X-Rob | seb-, that's not crud. That's your configuration. |
| 05:18.27 | X-Rob | Well. I s'pose if you put crud into your sip.conf, then yes, it is crud. |
| 05:18.40 | seb- | X-Rob: it shows a client's IP address as if a client is connected |
| 05:18.55 | X-Rob | seb-, let's try this again. |
| 05:18.56 | seb- | CLI> sip show peers |
| 05:18.57 | seb- | Name/username Host Dyn Nat ACL Port Status |
| 05:18.59 | seb- | 1000/1000 174.144.230.147 D N 5060 Unmonitored |
| 05:19.01 | X-Rob | seb-, remove the lines from your sip.conf |
| 05:19.03 | seb- | 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] |
| 05:19.15 | seb- | X-Rob: ok |
| 05:20.37 | seb- | X-Rob: ok yes that worked |
| 05:20.52 | seb- | X-Rob: fine you win...but that seems a bit heavy to boot people off your asterisk box no? |
| 05:21.00 | X-Rob | Um. |
| 05:21.17 | *** part/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:21.18 | seb- | X-Rob: i mean...needing to nuke sip.conf entries |
| 05:21.18 | X-Rob | If you don't want them to connect to your asterisk machine, then you can't leave them there. |
| 05:21.30 | X-Rob | you could change the password, I guess, if it's just temporary |
| 05:21.44 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128744483.dsl.bell.ca) |
| 05:21.55 | seb- | X-Rob: oh hehe...i'm a newb doing *testing*....i was kinda hoping to register a client...kick him off....test again ,etc. |
| 05:22.03 | X-Rob | No, it doesn't work like that |
| 05:22.06 | X-Rob | this is a phone system |
| 05:22.11 | X-Rob | it wants to be connected ALL THE TIME |
| 05:22.21 | seb- | X-Rob: hmm |
| 05:22.21 | X-Rob | if it's not connected, it'll keep trying to connect until it does |
| 05:22.24 | seb- | thinks |
| 05:22.37 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 05:22.45 | X-Rob | but why are you testing registering? it either works, or it doesn't. |
| 05:22.47 | seb- | X-Rob: i think i understand...it isn't like TCP/IP where connections are created and torn down all the time |
| 05:23.11 | seb- | X-Rob: a phone "registers" and then happily works for months |
| 05:23.15 | X-Rob | yes |
| 05:23.26 | seb- | X-Rob: ok, thanks i learned something |
| 05:23.28 | X-Rob | registration is actually how your phone tells asterisk where it is, and where to send calls to |
| 05:23.33 | X-Rob | that's all |
| 05:23.54 | seb- | very nice |
| 05:24.06 | p1mrx | X-Rob: I have no /var/log/messages/full. "core set debug 999" and "core set verbose 999" didn't tell me anything extra. |
| 05:24.07 | seb- | my first baby steps into the world of voip |
| 05:24.28 | p1mrx | er, /var/log/asterisk/full * |
| 05:24.29 | X-Rob | p1mrx, check /etc/asterisk/logger.conf |
| 05:24.42 | X-Rob | maybe you need to turn logging on |
| 05:26.07 | *** join/#asterisk apeiron_ (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 05:28.05 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 05:31.33 | tzafrir_laptop | p1mrx, to see what's logged where: logger show channels |
| 05:31.48 | *** join/#asterisk Phurl (n=mdupont@81.210.230.234) |
| 05:31.52 | Phurl | Hi all |
| 05:31.53 | p1mrx | I've managed to enable the "full" log |
| 05:32.06 | p1mrx | I'm trying to interpret everything now. |
| 05:33.12 | Phurl | anyone who can recommend a linux video chat client with a firewire capture card? I would like to use dvgrab/ffmpeg or vlan to capture the video and pipe that to a video conference |
| 05:33.36 | Phurl | (for asterisk) to connect to a polycom video conference |
| 05:37.12 | p1mrx | bizarre... it seems that google voice hangs up on me if I answer too quickly. |
| 05:39.03 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:40.16 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:42.28 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 05:50.12 | drmessano | p1mrx: Noticed a weird issue with that too |
| 05:57.05 | *** join/#asterisk octothorp (n=vandebo@abydos.nerdbox.net) |
| 05:57.53 | octothorp | Can anyone tell me the difference between the Ring and Ringing state? And how can I transition a channel between them? |
| 05:59.38 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:06.02 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 06:07.48 | jplank | if I drop a call from the asterisk side on a tdm2400 with a FXO card, the line drops, then comes back up a few seconds later and * routes the call like it was ringing, any idea what could be causing this? |
| 06:07.59 | jplank | if I drop the call on the far end side, it disconnected cleanly |
| 06:09.11 | octothorp | nevermind my previous question... is there anyway to force a channel allocation so that I can steal it (StealChan)? dialing a LOCAL/ extension doesn't seem to allocate a channel |
| 06:10.05 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:10.26 | octothorp | Console/dsp also doesn't allocate a channel... |
| 06:10.37 | *** part/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 06:11.11 | tzafrir_laptop | jplank, a few seconds? or shorter than that? |
| 06:11.27 | jplank | about a few seconds |
| 06:11.34 | jplank | the far end stays up for about 4 seconds |
| 06:11.47 | jplank | then I see the line get picked up a second later |
| 06:12.24 | jplank | it almost seems like * is flashhooking the call, not dropping it, but that doesn't make sense |
| 06:14.01 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.13) |
| 06:22.18 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 06:23.45 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.184) |
| 06:26.14 | *** join/#asterisk saftsack (n=saftsack@p57924807.dip.t-dialin.net) |
| 06:29.03 | octothorp | sorry, nevermind me... Local does too allocate a channel |
| 06:32.26 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 06:33.28 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 06:35.04 | jplank | if the call originates from the PSTN and the asterisk side hangs up first, it works, but if the call originates from asterisk, and the asterisk side hangs up, the far end doesn't disconnect, and within a couple seconds, asterisk picks back up the call |
| 06:35.42 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 06:35.56 | jplank | if it was the other way around, I would think it was the disconnect supervision, but since if the PSTN hangs up, the call ends cleanly, I doubt thats the issue |
| 06:40.06 | jplank | I also hear a loud click when I answer the call |
| 06:46.33 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 07:04.03 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:17.11 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 07:18.16 | brunner | Do VoIP providers like VoicePulse have POPs in dozens of countries, or what? How do they make money on SIP trunking? |
| 07:18.38 | octothorp | AFAIK voicepulse has POPs in NY & SF |
| 07:22.17 | p1mrx | why would Echo() only work if I put a Playback() before it? If I Answer/Echo, I get silence. If I Playback/Echo, the echo works. |
| 07:23.06 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-47f3c10c4343ee6c) |
| 07:24.11 | octothorp | maybe you have a config problem and haven't actually answered. see if it works even with the noanswer option to playback |
| 07:25.21 | p1mrx | I know the call was answered, because if I don't answer, the other end keeps ringing |
| 07:25.51 | p1mrx | the call's arriving through google voice, so it would route me to voicemail if nobody answered |
| 07:25.53 | octothorp | could try my suggestion and eliminate the possibility |
| 07:27.07 | p1mrx | are you suggesting that I do "Playback(beep, noanswer)", followed by "Echo()" ? |
| 07:27.13 | p1mrx | without answering? |
| 07:27.23 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:28.08 | octothorp | what works? answer/playback/echo ? or just playback/echo? |
| 07:28.32 | p1mrx | I've tried Answer/Echo and Playback/Echo. I don't think I've tried Answer/Playback/Echo |
| 07:28.43 | p1mrx | Playback/Echo works. Answer/Echo doesn't |
| 07:29.06 | octothorp | I'd try answer/playback/echo & answer/playback,noanswer/echo |
| 07:31.32 | *** join/#asterisk shtoom (n=shtoom@124.123.18.143) |
| 07:31.56 | p1mrx | I just tried Answer/Echo again. silence. Answer/Playback(beep,noanswer)/Echo works. |
| 07:32.36 | p1mrx | Answer/Playback(beep)/Echo also works |
| 07:33.05 | octothorp | is there anything interesting in the verbose output with answer/echo? |
| 07:33.18 | octothorp | you might also try answer/wait(1)/echo |
| 07:33.25 | p1mrx | <PROTECTED> |
| 07:33.25 | p1mrx | <PROTECTED> |
| 07:33.25 | p1mrx | <PROTECTED> |
| 07:33.26 | p1mrx | <PROTECTED> |
| 07:33.36 | p1mrx | the final line was after I hung up, several seconds later |
| 07:34.20 | p1mrx | oh, all these have begun with Wait(6), but that's probably not relevant |
| 07:35.26 | p1mrx | Answer/Wait(1)/Echo -- still silence |
| 07:36.15 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
| 07:36.22 | octothorp | not sure... |
| 07:36.35 | octothorp | are you gizmo'ing in from GV? |
| 07:36.40 | p1mrx | yes |
| 07:36.54 | flohack | Can someone tell me how I can comment on closed bugs in the digium bugtracker? |
| 07:37.32 | octothorp | been fiddling with GV & gizmo for the last few days myself, haven't had any problem... |
| 07:37.56 | flohack | Someone closed a bug and I would like to get more info from him, as he mentioned a fix but did not provide detailled info. |
| 07:38.59 | p1mrx | octothorp: have you figured out outbound dialing through GV? |
| 07:39.21 | octothorp | p1mrx: what do you mean exactly? |
| 07:39.32 | p1mrx | http://www.pmarks.net/posted_links/google-voice-dialout.agi |
| 07:39.45 | p1mrx | that script lets you dial out through your GV account, using Asterisk 1.6 |
| 07:39.57 | p1mrx | as in, pick up an IP phone, dial a number, place a call. |
| 07:40.27 | octothorp | interesting.... just finished implementing the same concept myself |
| 07:41.09 | tzafrir_laptop | p1mrx, what's Google Voice? That service that no longer works with SIP? |
| 07:41.46 | p1mrx | it works with SIP, for incoming calls at least. |
| 07:42.11 | p1mrx | the script dials out using an incoming call, and some HTTP magic. |
| 07:49.47 | *** join/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 07:53.17 | *** join/#asterisk Chris-NB (n=chris@212.27.68.98) |
| 07:53.45 | *** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67) |
| 07:54.57 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 08:05.57 | flohack | Ok, reopening the bug lets you comment on it. |
| 08:06.44 | *** part/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 08:07.14 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 08:09.09 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
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| 08:21.17 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 08:26.27 | *** join/#asterisk mikkel (n=mikkel@130.226.37.136) |
| 08:31.40 | *** join/#asterisk Marquel (n=Marquel@port-1156.pppoe.wtnet.de) |
| 08:31.47 | *** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) |
| 08:31.48 | kippi | hey |
| 08:32.00 | Marquel | morning |
| 08:32.09 | kippi | need some help with linksys spa942 and transfers |
| 08:32.17 | kippi | where is the best place to ask? |
| 08:32.33 | kippi | it might be something that needs a change in asterisk |
| 08:34.29 | mosty | ask here |
| 08:36.26 | kippi | if I dial in from my mobile to my desk phone, then do a attended transfer, the phone that I transfer the call to is losing my mobile number |
| 08:36.28 | h-idrisi | any idea about how to activate FOP popup ??? |
| 08:39.04 | Marquel | kippi: probably a configuration issue with transfers. since pbx'es can not decide whether you're attemting attended transfer or are just making a second call, the caller-id of the phone you transfer from is transmitted. |
| 08:39.36 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-16-168.w90-56.abo.wanadoo.fr) |
| 08:39.40 | Marquel | kippi: maybe you can find some parameter to change this to change to the original caller-id upon transfer, but i'm not sure (and can as yet not test it) ;) |
| 08:40.47 | merlin8282 | Hi. I have an ISDN QuadBRI and bristuff-0.4.0-RC4-xr5 installed. I don't know why, the ZapCD isn't installed/available. How can I fix this ? |
| 08:41.11 | kippi | Marquel: So this is something that might be coming out in 1.6? |
| 08:41.59 | Marquel | kippi: it's more "something that might not work at all". but i think i remember something about this already working in 1.2 ;) |
| 08:43.06 | kippi | hmm |
| 08:43.08 | kippi | ok |
| 08:43.56 | Marquel | kippi: look at the options of Dial(), especially 'o' ;) |
| 08:47.03 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 08:47.57 | kippi | but it's the handset that is doing the transfer, not calling a dial plan |
| 08:49.05 | kippi | or is there away to change the default so that it sends callers ID |
| 08:49.13 | Marquel | mom, bbl. |
| 08:51.00 | mosty | kippi: perhaps a blind transfer? |
| 08:51.17 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 08:51.20 | sah-work | hello! |
| 08:51.29 | sah-work | i am trying to cut over from any old system to a new one |
| 08:51.33 | sah-work | having some issues |
| 08:51.39 | kippi | mosty: nope needs to be attended |
| 08:51.44 | sah-work | how do i see in asterisk if it sees my pri |
| 08:52.25 | andrebarbosa | pri show span 1? |
| 08:52.30 | sah-work | cool |
| 08:52.57 | sah-work | hum, no love |
| 08:53.00 | sah-work | dang |
| 08:53.15 | jermey_g | are there any known issues in receiving dtmf digits if you are behind a nat. voice calls come in and out fine but the incoming dtmf digits after answering that call are not received at all.what could be wrong? |
| 08:53.59 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:55.46 | sah-work | nice. it is up ;) |
| 08:58.01 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 09:03.37 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 09:04.37 | Chris-NB | hi |
| 09:04.40 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 09:04.51 | *** join/#asterisk Marquel (n=Marquel@port-1156.pppoe.wtnet.de) |
| 09:04.54 | Marquel | re |
| 09:05.36 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 09:05.42 | Chris-NB | I want my users to be able to exit a dial command by hitting a 1-digit extension to exit to a specified application |
| 09:05.51 | Chris-NB | I've configured an exitcontext and pass the option d to the Dial command |
| 09:06.02 | Chris-NB | but when I hit my extension, nothing happens |
| 09:07.22 | mosty | jermey_g: have you tried the various sip dtmf options? |
| 09:09.03 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 09:11.59 | jermey_g | mosty:yes.its out of band that works if I take the same configuration on to internet server. |
| 09:12.20 | jermey_g | but when i put that configuration on the server behind the nat, i don't receive any rtp |
| 09:12.26 | jermey_g | any dtmf |
| 09:12.46 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 09:14.15 | mosty | jermey_g, it's generally unwise to put a sip server behind nat, can you organise a public ip address for the machine? |
| 09:17.05 | *** join/#asterisk mort___ (n=mort@user-54405220.wfd76b.dsl.pol.co.uk) |
| 09:22.09 | Chris-NB | I can send dtmf tones (voicemail) but the dial command does not exit to the dialed extension |
| 09:22.15 | Chris-NB | anyone seen this ehavior? |
| 09:22.22 | Chris-NB | behavior |
| 09:23.35 | mosty | Chris-NB, when inside the Voicemail command? |
| 09:25.24 | Chris-NB | mosty, I can send dtmf 'numbers' to voicemail, so I know, my entered numbers are correctly transmitted/interpreted to/by asterisk |
| 09:25.40 | Chris-NB | but what I want: I want my users to be able to exit a dial command by hitting a 1-digit extension to exit to a specified application |
| 09:25.55 | Chris-NB | I've configured an exitcontext and pass the option d to the Dial command |
| 09:25.59 | jermey_g | mosty:i have been successfully doing this unwise thing for last 3 years. |
| 09:26.48 | jermey_g | mosty:its just that now i have an additional requirement on this sip server behind nat to read the dtmf digits with dtmfmode=rfcxxxx from the incoming call, which it can't as long as its behind the nat |
| 09:26.50 | Chris-NB | but when I hit my extension, nothing happens |
| 09:26.53 | jermey_g | what could be wrong? |
| 09:26.55 | mosty | jermey_g, sip is a hell of a lot simpler when the server isn't behind nat, that's all |
| 09:27.12 | jermey_g | does the dtmf mode = rfcxxxx send digits with the rtp |
| 09:27.35 | mosty | Chris-NB, oh i see- voicemail works as you want but you want to be able to cancel dial? |
| 09:27.40 | jermey_g | my server answers the call, and reads the digits and says read nothing. |
| 09:28.03 | jermey_g | if i dial the incoming user to a local extension, the call is setup correctly and we have two way audio |
| 09:28.24 | Chris-NB | mosty, jep and exit to the entered extension |
| 09:30.06 | Chris-NB | mosty, but nothing happens when I hit 5 (configured extension in my exit context) |
| 09:30.59 | mosty | Chris-NB, i have never used that dial option before, are there any logs when you attempt this? |
| 09:35.03 | Chris-NB | mosty, noop, I don't see anything. I'm using this on another machine without problems. but here I cant't get it to work |
| 09:42.11 | *** join/#asterisk hi365 (n=hi365@94.159.162.103) |
| 09:43.32 | *** join/#asterisk dandate2 (n=dandate2@173.11.82.122) |
| 09:43.39 | dandate2 | how do I configure QoS in my router for pbx? |
| 09:46.14 | mosty | dandate2, depends on the router- consult the manual? |
| 09:48.35 | *** join/#asterisk suma (n=chinnapa@cpe-76-168-177-23.socal.res.rr.com) |
| 09:49.32 | suma | How could I play file simultaneous to the callee and caller ? |
| 09:49.48 | hi365 | agi |
| 09:50.06 | suma | hi365: when they are on call ? |
| 09:50.16 | mosty | suma: meetme? |
| 09:50.20 | hi365 | nods |
| 09:50.28 | hi365 | that *should* do it |
| 09:50.48 | suma | hi365: without meetme ? |
| 09:51.10 | merlin8282 | Nobody know for my problem ? -> ISDN QuadBRI and bristuff-0.4.0-RC4-xr5 installed. I don't know why, the ZapCD isn't installed/available. How can I fix this ? |
| 09:51.12 | suma | I use the L option in the Dial command to play the file |
| 09:51.13 | hi365 | heck, if you know the channel name, you shoould be able to do it with dialplan/ .call files |
| 09:51.29 | hi365 | whats the problem then? |
| 09:52.14 | suma | hi365: let me google again, if still stuck will get back here |
| 09:56.55 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 09:59.36 | suma | hi365: that is an awesome solution, will give a try to get it to work |
| 10:00.01 | hi365 | :) |
| 10:01.03 | tzafrir_laptop | merlin8282, hmm.... was it included there? |
| 10:02.16 | merlin8282 | what do you mean tzafrir_laptop ? |
| 10:02.42 | merlin8282 | oh, i read wrong, sorry |
| 10:02.49 | merlin8282 | I don't know, how can I see it ? |
| 10:03.09 | merlin8282 | Is it part of libpri ? |
| 10:03.51 | tzafrir_laptop | hmm... indeed it was left out |
| 10:05.02 | tzafrir_laptop | anybody feels like providing it in a separate patch? I'm a bit busy with some other stuff right now :-( |
| 10:05.22 | merlin8282 | Ok. And what can I now do to include it to my installed server ? |
| 10:05.32 | merlin8282 | ok |
| 10:08.56 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 10:10.05 | Kernel_Core | hi all |
| 10:10.57 | tzafrir_laptop | ZapCD is part of chan_zap / chan_dahdi, right? |
| 10:12.14 | merlin8282 | No idea, I only have this error in the console: WARNING[8226]: pbx.c:1832 pbx_extension_helper: No application 'ZapCD' for extension (from-sip, 666, 4) |
| 10:12.51 | tzafrir_laptop | hmm... right. it is there. I looked for ZapCD, but it is in fact zapCD |
| 10:13.28 | tzafrir_laptop | try using the patch bristuff-part-dahdi-all instead bristuff-part-dahdi |
| 10:14.04 | tzafrir_laptop | (edit patches/asterisk/series and re-run ./download.sh . It won't actually re-download. But it will repatch) |
| 10:14.43 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 10:18.10 | Kernel_Core | many days age when I used ZAPTEL IO had some issues with my lines ( Hook Detection) I used this patch http://bugs.digium.com/view.php?id=8290 and everything worked perfectly recently I just upgraded to Dahdi ( I am useing chinies TDM400P board openvox ) I have the same Problem ! and I cannot patch dahdi files ! any suggestion ? |
| 10:18.18 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 10:19.09 | merlin8282 | ok tzafrir_laptop thanks. |
| 10:19.48 | Kernel_Core | when I reboot my box , I have to replug my lines to get them detected in dahdi-tools |
| 10:20.29 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:21.08 | tzafrir_laptop | Kernel_Core, version of Asterisk? |
| 10:21.59 | Kernel_Core | tzafrir_laptop: doesn't matter , both 1.4 1.2 and 1.6 ! |
| 10:22.20 | tzafrir_laptop | What exact versions? |
| 10:22.39 | Kernel_Core | 1.4.21.2 is my production version |
| 10:23.16 | Kernel_Core | tzafrir_laptop: but I think it is related to WCTDM not asterisk ! |
| 10:23.51 | tzafrir_laptop | what do you mean by "detected"? |
| 10:24.58 | Kernel_Core | tzafrir_laptop: imagine you have TDM400P with 4 FXO card |
| 10:25.12 | Kernel_Core | and you connect 4 lines to the card |
| 10:25.37 | Kernel_Core | when you I restart my box in (dahdi_tool) dialog I get 4/4/0 |
| 10:25.48 | Kernel_Core | when I replug the lines then it changes to 4/4/4 |
| 10:25.49 | tzafrir_laptop | imagines all the people have them |
| 10:25.55 | Kernel_Core | :D |
| 10:27.00 | Kernel_Core | it is my problem ! Dahdi doesn't detect the state of the lines when I reboot the box . |
| 10:27.13 | Kernel_Core | hook state I mean |
| 10:28.29 | Kernel_Core | in past I used Zaptel , and Patched my Zaptel with http://bugs.digium.com/view.php?id=8290 , and my problem solved , but for dahdi I cannot patch it ! I changed everything from ZT to DAHDI but it doesn't work :( |
| 10:33.30 | tzafrir_laptop | This patch should already be included in dahdi . It was fixed in Zaptel before DAHDI was split from it |
| 10:34.04 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 10:34.22 | tzafrir_laptop | dahdi incorrectly reports the line as in alarm? |
| 10:34.40 | Kernel_Core | tzafrir_laptop: yes |
| 10:35.02 | brunner | how many concurrent channels can IAX support before it falls apart? |
| 10:35.04 | Kernel_Core | tzafrir_laptop: http://bugs.digium.com/file_download.php?file_id=17339&type=bug it was the exact patch that I used |
| 10:35.11 | tzafrir_laptop | what version of dahdi-linux (or zaptel) it is? |
| 10:35.11 | Kernel_Core | brunner |
| 10:35.20 | Kernel_Core | brunner: depends on your linux configuration |
| 10:35.44 | Kernel_Core | tzafrir_laptop: dahdi-linux-2.0.0 |
| 10:36.04 | Kernel_Core | and Zaptel Version: SVN-branch-1.4-r4503M |
| 10:36.33 | mort_gib | Will there be potin-to-multipoint support for BRI in Dahdi soon?? |
| 10:36.34 | brunner | Kernel_Core: can you give me a little bit more information? |
| 10:37.38 | Kernel_Core | brunner: how many concurrent channels do you need ? do you have any TRANSCODE ? |
| 10:38.28 | brunner | ~55, no transcode |
| 10:38.42 | Kernel_Core | brunner: check here http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 10:38.57 | Chris-NB | anyone usint the d option from dial command? dial a 1 digit extension and jump to that context? |
| 10:39.15 | Kernel_Core | brunner: any P4 should be fine with 1GB Ram at least ! |
| 10:39.28 | brunner | oh |
| 10:39.37 | brunner | the Freeswitch people seem to think that IAX isn't scalable |
| 10:39.50 | Kernel_Core | brunner: no it isnot right |
| 10:40.05 | brunner | what? |
| 10:40.28 | Kernel_Core | IAX2 is scalable like SIP ! |
| 10:40.44 | brunner | okay, thanks |
| 10:41.08 | brunner | it's crazy how much more efficient it is |
| 10:41.12 | Kernel_Core | tzafrir_laptop: I couldn't find any line similar to this patch in dahdi ( wctdm.c ) file. |
| 10:42.29 | Kernel_Core | brunner: it has less CPU usage than SIP if configured in Trunk mode , because you decrese the received packets hence less CPU cycle . |
| 10:43.17 | brunner | nice |
| 10:44.14 | Kernel_Core | 55-60 concurrent channels isnot very big deal for asterisk now |
| 10:44.46 | beherit | whats the possible cause of a warning rtcp read too short? |
| 10:45.48 | hi365 | anyone good with audiocodes' products? |
| 10:48.23 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:51.01 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 10:51.54 | k-man | for some reason, when my asterisk box rebooted today, after the reboot i was unable to make any calls, i did a reload and after that it worked - i know its vague but any idea what might have caused that? |
| 10:54.41 | beherit | whats the possible cause of a warning rtcp "read too short"? |
| 10:58.20 | Dovid | for the US any way to see who owns a specific number ? |
| 10:58.40 | Dovid | locallcalling guide.com wont help cause it shows who owned it originally |
| 11:00.12 | mosty | k-man: consult your asterisk logs |
| 11:00.43 | k-man | mosty: ah, good idea |
| 11:03.18 | k-man | mosty: it seems as though it was unable to do the dns lookups - perhaps networking wasn't quite up when asterisk started |
| 11:03.45 | mosty | possibly |
| 11:05.41 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
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| 11:39.20 | *** join/#asterisk smellynoser (n=ashley@87-194-183-38.bethere.co.uk) |
| 11:39.22 | smellynoser | Hola |
| 11:39.37 | smellynoser | Can I use Python to create IVRs with Asterisk? |
| 11:42.49 | *** join/#asterisk propellerhead (n=yogurt2u@host110.190-136-239.telecom.net.ar) |
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| 11:52.20 | beek | smellynoser: You could via AGI. |
| 11:54.54 | smellynoser | beek: Do you imagine it would be easy? |
| 11:56.32 | mosty | smellynoser, is there a particular reason you want to use python? |
| 11:56.56 | beek | I've written a couple in Perl but for the most part I just use dialplan logic. Is there some particular need you have that is not addressed by dialplan? |
| 11:57.14 | smellynoser | mosty: Because all employees know it. What other languages could I use? |
| 11:57.42 | mosty | smellynoser, in that case, you're better off not using python since you will have to learn how to dialplan works anyway |
| 11:57.56 | smellynoser | beek: I haven't looked at the dialplan logic tbh. I would like to use a language because it is a lot easier for everybody on the team to change the IVR then |
| 11:58.57 | beek | smellynoser: Well, look at it this way. You can shoehorn a language into doing dialplan logic or you can write dialplan logic in a language made for it. Besides, if you're team is worth anything dialplan logic will be trivial. |
| 11:59.07 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:59.29 | *** join/#asterisk czindy (n=czindy@91.120.28.147) |
| 11:59.32 | mosty | smellynoser, you will need to know dialplan logic if you use python- so it won't really help your situation |
| 12:00.18 | mosty | you can always call small python AGI's for complicated stuff, but an IVR dialplan is what the asterisk syntax is designed for |
| 12:00.35 | beek | mosty: agreed. |
| 12:00.57 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:01.21 | beek | smellynoser: The IVR I wrote in Perl was to do a complicated subroutine within dialplan logic. |
| 12:01.28 | czindy | Hello, I need your help again :). I would like to configure a grandstream GXW4104 device but when I receive a call I got the following message in asterisk: chan_sip.c:6730 process_sdp: Unsupported SDP media type in offer: audio 8058 RTP/SAVP 0 8 4 18 3 101 why it is happening? |
| 12:02.23 | smellynoser | Does the dialplan support hitting URLs and DB queries? |
| 12:04.25 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 12:04.44 | beek | Yes: Curl() and func_odbc |
| 12:05.45 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:05.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:16.03 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.50 | czindy | Ok I solved my last question: SRTP Mode has to be disabled in grandstream config |
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| 12:20.31 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 12:20.33 | tzafrir_laptop | smellynoser, the native dialplan language is more limited than AGI, but much easier to debug and more efficient |
| 12:20.49 | tzafrir_laptop | You can also write dialplans in Lua (in 1.6.x) |
| 12:21.31 | tzafrir_laptop | If you have complex logic, taking it out to an external AGI in your familiar language can help |
| 12:22.57 | *** join/#asterisk erbse (n=Andre@voxgate.winet.ch) |
| 12:23.40 | [TK]D-Fender | tzafrir_laptop: Native dialplan more limited than AGI? Can't see how thats possible. Its more like the reverse |
| 12:25.04 | tzafrir_laptop | [TK]D-Fender, in AGI it is much easier to apply smarter control structures, data structures, and such |
| 12:25.50 | [TK]D-Fender | tzafrir_laptop: Well certainly more familiar for those who've done structured programming. But it is more limited than AGI given that's what it "compiles" down to. |
| 12:25.53 | *** join/#asterisk arpu (n=arpu@91-115-168-63.adsl.highway.telekom.at) |
| 12:25.57 | tzafrir_laptop | Though merely moving to AGI adds some inherent complexity, and thus it better be worth it |
| 12:26.06 | [TK]D-Fender | s/AGI/dstd dialplan/ |
| 12:26.44 | [TK]D-Fender | tzafrir_laptop: Sorry, juxtaposed there. |
| 12:28.37 | smellynoser | How easy it Asterisk to setup with G729? |
| 12:29.03 | khronos | Not hard at all. |
| 12:29.11 | [TK]D-Fender | smellynoser: Fairly easy. Go to Digium's site and look at the reg process |
| 12:29.17 | smellynoser | Also, we're receiving calls via SIP from a company that receives them over PSTN on DDIs (DIDs). They don't authenticate, they just send them across as: ddi@ourip.com |
| 12:29.23 | smellynoser | Is that a problem? |
| 12:29.59 | czindy | My next question is regarding the following: http://pastebin.com/m541561fc (outbound calls over GXW4101 FXO gateway are not accepted). Any suggestion would be appriciated. I looked after it on google but with no success. Thanks |
| 12:31.15 | beherit | what's the possible cause of a warning rtcp "read too short"? |
| 12:31.43 | [TK]D-Fender | smellynoser: Well it does make it easier for them to become hijacked |
| 12:32.34 | [TK]D-Fender | czindy: Try another call with SIP DEBUG enabled |
| 12:32.52 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:33.17 | czindy | [TK]D-Fender: okay thank you I'm trying... |
| 12:33.17 | X-Rob | ah fuc |
| 12:33.19 | X-Rob | k |
| 12:33.20 | X-Rob | I'm a tard. |
| 12:33.36 | [TK]D-Fender | X-Rob: 11 steps to go! |
| 12:35.10 | X-Rob | [TK]D-Fender, if I may have your invaluable advice.. |
| 12:35.10 | X-Rob | http://pastebin.ca/1377853 |
| 12:35.28 | X-Rob | that won't work - you can't have two extens includied and expect it to fail from one to the other, correct? |
| 12:35.47 | X-Rob | uh, ignore the typos. |
| 12:35.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:37.02 | [TK]D-Fender | X-Rob: Correct, that is not a failover. Standard include precedence rules says which will get executed |
| 12:37.05 | X-Rob | what I was _thinking_ that after Macro(dalout-trunk) does a MacroExit, it would then match on the other context. But I think I'm wrong. |
| 12:37.51 | X-Rob | so really, a sensible thing to do would be to increase the priority numbers of eact context so they don't clash |
| 12:39.20 | X-Rob | http://pastebin.ca/1377857 |
| 12:39.20 | X-Rob | eg |
| 12:39.21 | [TK]D-Fender | X-Rob: Doesn't work that way. |
| 12:39.21 | leifmadsen | you're still using priority numbers and pipes to separate? Need to update to priority labels and commas. |
| 12:39.40 | [TK]D-Fender | X-Rob: it will look in the appropriate context first. Do not attempt to SPAN |
| 12:39.55 | X-Rob | [TK]D-Fender, suggestions instead of? |
| 12:39.57 | leifmadsen | Dial() will continue on in the dialplan if it doesn't connect somewhere, or if you use the option to Dial() to force it to continue even after a call has connected (which you almost never want) |
| 12:40.10 | [TK]D-Fender | X-Rob: 1 context that does what you want |
| 12:40.28 | X-Rob | Bugger. |
| 12:40.40 | X-Rob | I need to rewrite outbound routes, it sucks anyway. |
| 12:40.49 | X-Rob | was hoping not to have to do it tonight, that's all 8) |
| 12:41.35 | czindy | D-Fender do you mean to debug with: core set debug channel all? If yes please check the output: http://pastebin.com/d7d1573dd |
| 12:42.23 | *** part/#asterisk khronos (n=none@c-76-110-120-247.hsd1.fl.comcast.net) |
| 12:42.24 | [TK]D-Fender | czindy: No, I said SIP DEBUG where you can see the PACKETS. |
| 12:43.14 | X-Rob | the problem is, people DO want to be able to set precedence on their dialplans. 1-212-555-xxxx via SIP/foo and 1-212-XXX-XXXX via sip/wibble. |
| 12:43.21 | X-Rob | goes rtfming on precedence in dialplans |
| 12:43.37 | X-Rob | the only way we _could_ do that was altering the order of includes |
| 12:43.43 | [TK]D-Fender | X-Rob: You will NOT span contexts. |
| 12:43.55 | X-Rob | [TK]D-Fender, uh. ok. |
| 12:44.05 | czindy | [TK]D-Fender please give a little detailed info how can I do this? |
| 12:44.07 | X-Rob | How can I do what I want to do? |
| 12:44.16 | X-Rob | Because I don't think I can 8-\ |
| 12:44.22 | [TK]D-Fender | X-Rob: Get your sorry ass out of that retard GUI |
| 12:44.39 | [TK]D-Fender | czindy: "sip set debug on" |
| 12:44.50 | leifmadsen | You have to control the order with includes, but using the same extension number in one included context and the other will not make it failover to the other context. |
| 12:45.36 | X-Rob | [TK]D-Fender, yes, ok, blah blah, you don't like freepbx. Whatever. I'm not asking a freepbx question, I'm asking someone who is a lot more up to date on dialplan programming than I am. |
| 12:46.08 | X-Rob | If your answer is 'I don't know', then that's fine. |
| 12:46.15 | beherit | what's the possible cause of a warning rtcp "read too short"? |
| 12:46.50 | [TK]D-Fender | X-Rob: FreePBX's **USE** of the dialplan is your problem. |
| 12:47.02 | X-Rob | [TK]D-Fender, I'm quite happy to rewrite it |
| 12:47.18 | X-Rob | <PROTECTED> |
| 12:47.24 | [TK]D-Fender | X-Rob: X-Rob Well you want it to do 2 things, jsut make 1 exten with everything in it all in one context. |
| 12:47.28 | X-Rob | ^^ how can I do that _without_ adjusting the order of includes |
| 12:47.41 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 12:47.49 | [TK]D-Fender | X-Rob: Want to know? go read the book and docs on how dialplan sorting works. |
| 12:47.59 | X-Rob | eg, you don't know. |
| 12:48.01 | [TK]D-Fender | X-Rob: Includes are only part of it |
| 12:48.04 | X-Rob | All you had to do was say. |
| 12:48.24 | [TK]D-Fender | X-Rob: No, I'm not simply going to hand everything to you. |
| 12:48.46 | X-Rob | boggles slightly. |
| 12:49.19 | *** join/#asterisk clintc (n=clintc@n128-227-7-200.xlate.ufl.edu) |
| 12:49.45 | czindy | [TK]D-Fender: http://pastebin.com/d352db2b9... I see that the device answered Service Unavailable, but maybe you better know why it is happening. |
| 12:51.10 | [TK]D-Fender | czindy: Unfortunately the SIP debug wasn't much more informative, but the best guess at the start is your device is not configured to process the number you are passing it. |
| 12:51.16 | X-Rob | Nup, doesn't look like it can be done. |
| 12:51.21 | [TK]D-Fender | czindy: Especially suspect because of the # & * in there |
| 12:52.03 | czindy | ok I'm trying to call a simple number |
| 12:52.07 | [TK]D-Fender | X-Rob: Plenty of smarter global patterns you can do with actual logic in them. |
| 12:52.33 | [TK]D-Fender | czindy: To: <sip:*31#06202952911@192.168.5.20:5062>;tag=as1264797b <- not so simple with *31# in front |
| 12:52.58 | [TK]D-Fender | czindy: I might very quickly suspect this part |
| 12:53.10 | X-Rob | [TK]D-Fender, not really. Users want to be able to type in 1-212-555-xxxx to sip/foo and everything else to sip/wibble. Asterisk precedence will match the one with the most wildcards first, I think. |
| 12:53.20 | X-Rob | rummages |
| 12:53.40 | [TK]D-Fender | X-Rob: and as I jsut told you feel free to use a larger wildcard and actually put LOGIC in them |
| 12:53.59 | X-Rob | my solution is an AGI |
| 12:54.01 | *** join/#asterisk canapa (i=canapa@213.162.70.112) |
| 12:54.08 | [TK]D-Fender | X-Rob: Totally unnecessary |
| 12:54.13 | X-Rob | because I don't want to scrabble around trying to write that in AGI |
| 12:54.21 | X-Rob | uh |
| 12:54.23 | X-Rob | Extensions |
| 12:54.25 | canapa | NOTICE[9532]: chan_sip.c:15839 handle_request_register: Registration from '"100"<sip:100@pulver.gutsche.at>' failed for '213.162.70.112' - Peer is not supposed to register |
| 12:54.26 | canapa | Scheduling destruction of SIP dialog 'MGI3M2ZhNmM4YjZmZThlNWE2NzM0YjI3ZTY3YWQ3YzI.' in 32000 ms (Method: REGISTER) |
| 12:54.36 | canapa | any advice what my error is ? |
| 12:54.40 | [TK]D-Fender | X-Rob: Retarded... its what... a line or TWO of dialplan code? |
| 12:54.50 | X-Rob | [TK]D-Fender, Umm. |
| 12:54.54 | X-Rob | I think it's a hell of a lot more than that. |
| 12:55.05 | X-Rob | let me try to breadboard it |
| 12:55.07 | [TK]D-Fender | X-Rob: it isn |
| 12:55.09 | [TK]D-Fender | 'tt |
| 12:55.38 | czindy | [TK]D-Fender: You are a GOD :) .... It was working before the firmware upgrade but now it is not. So I'm trying to find out how can I dial out this string for the telco provider. *31# means at T-C.M that show my number. |
| 12:55.56 | X-Rob | I s'pose I could match on _., but I hate that. It's yucky. |
| 12:56.11 | [TK]D-Fender | canapa: You specified a "host=" that isn't "dynamic" and the device is trying to register against that peer entry |
| 12:56.25 | [TK]D-Fender | X-Rob: I would never suggest that pattern. |
| 12:56.30 | [TK]D-Fender | czindy: You're welcome |
| 12:57.04 | X-Rob | so I have to match _X. and duplicate it for _[*a-z]... Can asterisk match on pattern groups yet? |
| 12:57.14 | czindy | [TK]D-Fender Thank you Thank you Thank you! If you like beer you'll be my quest for some round.. :))) |
| 12:57.21 | [TK]D-Fender | X-Rob: "groups"? |
| 12:57.33 | X-Rob | _[a-z*1-0]. |
| 12:57.54 | [TK]D-Fender | X-Rob: Taht would grab jsut about anything |
| 12:58.10 | X-Rob | which is what you need |
| 12:58.17 | X-Rob | oh |
| 12:58.20 | [TK]D-Fender | X-Rob: and the reason for an alpha-numeric exten is...? |
| 12:58.22 | X-Rob | except for i, s, h, n etc |
| 12:58.34 | X-Rob | ..because people use alphanumeric extensions? |
| 12:58.55 | X-Rob | I don't know _why_, but some people like to. |
| 12:58.58 | [TK]D-Fender | X-Rob: Like who? For what? Who dials JOHN from their phone? |
| 12:58.59 | X-Rob | it's fucking annoying, honestly. |
| 12:59.03 | *** join/#asterisk xchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 12:59.26 | X-Rob | a couple of sites that I know of. Common factor is SCCP, actually. |
| 12:59.40 | [TK]D-Fender | :| |
| 12:59.55 | X-Rob | I think I'll just give up on this idea |
| 13:00.12 | [TK]D-Fender | sees a growing pattern of ridiculous planning... |
| 13:00.12 | X-Rob | not supported, won't work, don't bother trying. |
| 13:00.14 | canapa | thx [TK]D-Fender |
| 13:00.30 | X-Rob | I'll rewrite outgoing at some stage to fix this |
| 13:00.43 | [TK]D-Fender | X-Rob: there is a safer way to do this |
| 13:01.11 | [TK]D-Fender | X-Rob: You can do _., but then you'd jump out of that context immediately. |
| 13:01.34 | [TK]D-Fender | X-Rob: and you've have to ensure it has the lowest priority. |
| 13:01.38 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:02.27 | X-Rob | Yeah, that's what I do now for something else.. it's got big comments around it saying 'Don't do this. You don't want to do this. This is a special circumstance, and you should never try to do this yourself' or something like that. |
| 13:02.31 | X-Rob | can't remember what it was for |
| 13:02.33 | X-Rob | but I hated having to do it |
| 13:03.36 | X-Rob | ah yeah. in from-sip-external... no idea what people would be sending through as a destination so I had to do that and then do a goto straight away |
| 13:05.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:05.54 | X-Rob | OK. The sensible way to do this doesn't require an AGI, it just requires more logic, as you said. At the moment I don't keep track of prefixes, I just create a route and then include them in the order the user wants. .. Ah. Um. Damn. |
| 13:06.14 | X-Rob | there's got to be a better way. |
| 13:06.57 | X-Rob | ooh. |
| 13:07.22 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 13:08.58 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 13:10.32 | viraptor | do you know any good sip tester oriented for load? |
| 13:10.45 | viraptor | (sipp is not good enough) |
| 13:10.51 | [TK]D-Fender | viraptor: "sipps" |
| 13:11.03 | [TK]D-Fender | viraptor: Ok, thats the only common one i knew |
| 13:11.27 | leifmadsen | why isn't sipp good enough? |
| 13:12.00 | viraptor | unfortunately it can't handle more than 700 calls (with media, threads are killing scheduler) |
| 13:12.17 | leifmadsen | sounds like you need another box |
| 13:12.27 | leifmadsen | 700 calls with media is a LOT for anything |
| 13:12.33 | viraptor | I am using another box |
| 13:12.43 | leifmadsen | I meant use multiple boxes for the sipp side |
| 13:12.49 | leifmadsen | that would get you to 1400 |
| 13:12.54 | viraptor | well - I mean a sparate box, sorry :) |
| 13:12.59 | leifmadsen | so do I |
| 13:13.13 | leifmadsen | sipp -------> asterisk |
| 13:13.13 | leifmadsen | sipp / |
| 13:13.39 | leifmadsen | sipp_| |
| 13:13.48 | leifmadsen | etc._| |
| 13:14.06 | viraptor | I'd like to have 1 host that can do such testing from time to time... I know asterisk can handle more than 500 calls here, but I can't test it from one source now |
| 13:14.39 | viraptor | it's kind of funny when sipp can't manage to send that much traffic from a machine with the same spec |
| 13:14.53 | leifmadsen | sipp is the best tester, that handles audio, that I've found. If it can't handle more than 700 simultaneous calls, then you may just be out of luck |
| 13:15.06 | leifmadsen | there is miTester, but it doesn't do audio (RTP) |
| 13:15.17 | viraptor | thanks |
| 13:16.03 | coppice | sipp is intriguing. its sooooo small |
| 13:17.20 | leifmadsen | sipp is teh hawt |
| 13:17.31 | X-Rob | Heya coppice |
| 13:17.49 | coppice | hi |
| 13:18.21 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:19.16 | X-Rob | nup. can't be done. |
| 13:19.31 | X-Rob | needs an agi |
| 13:19.34 | X-Rob | and basically, fuck that. |
| 13:19.51 | *** join/#asterisk Chris-NB (n=chris@213162066145.public.t-mobile.at) |
| 13:19.53 | [TK]D-Fender | X-Rob: Nope... GotoIf <- Ain't Raw-Cat Science |
| 13:20.11 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:20.59 | [TK]D-Fender | X-Rob: Mind you properly ordered include with that more specific pattern should work as well, but you're duplicating code |
| 13:21.10 | X-Rob | gotoif with regexps, that'll work, I s'pose |
| 13:21.53 | X-Rob | re-implementing exten pattern matching in dialplan would be a hell of a lot slower than just doing an AGI |
| 13:23.50 | X-Rob | I see _why_ it's doing it that way, and it makes sence, too. Fast and easy. But it's crap for this one outlying situation that I want. |
| 13:24.17 | [TK]D-Fender | X-Rob: 1 problem you had is you tried using the same pattern in both. you didn't ahve the area-code |
| 13:25.06 | X-Rob | yes, and I want one to fail and then continue on to the other. |
| 13:25.16 | X-Rob | I can do that _or_ I can have ordering of precedence. |
| 13:25.32 | [TK]D-Fender | X-Rob: No, there is no "failing over. |
| 13:25.35 | X-Rob | (using #include to specify precedence) |
| 13:25.59 | X-Rob | so I have to lose one of them, as as the failing over is a new thing, and ordering of precedence is an old thing, the new thing gets canned. |
| 13:26.25 | X-Rob | or I rewrite it all as an AGI |
| 13:28.17 | [TK]D-Fender | X-Rob: So far we're talking about a single prefix. this is 1 or 2 lines of dialplan. |
| 13:28.59 | X-Rob | [TK]D-Fender, no. I think you've lost track of where I was. S'okay. Asterisk can't do what I want it to do, the way I want it, so I just won't do it. |
| 13:29.06 | X-Rob | I'm too lazy to rewrite it all as an AGI |
| 13:29.36 | [TK]D-Fender | X-Rob: and you don't NEED AGI |
| 13:30.03 | X-Rob | Uh. I do. I ether have to re-implement pattern matching in dialplan, or do it as an agi |
| 13:31.00 | [TK]D-Fender | X-Rob: Keep building up that mountain....I see 2 lines tops needed for this. |
| 13:31.20 | X-Rob | I honestly don't KNOW why the standard pattern matching isn't sufficient. Currently it orders by number of matches, so _1555222XXXX. will be hit before_1555XXXXXXX, but people want to do stupid things. |
| 13:31.28 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-219-152.lns10.mel4.internode.on.net) |
| 13:31.33 | X-Rob | [TK]D-Fender, if you see two lines, then you dont' see the problem. |
| 13:32.12 | [TK]D-Fender | X-Rob: Or you can't see the code. |
| 13:32.36 | [TK]D-Fender | X-Rob: and yes I can accomodate for that sample |
| 13:32.51 | X-Rob | Not for 1212XXX1234 |
| 13:33.12 | [TK]D-Fender | X-Rob: 1 line of dialplan gets that case solved |
| 13:33.22 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 13:33.33 | X-Rob | or, for a _real_ example, 0[2378]23322332 |
| 13:33.41 | *** join/#asterisk nicko_ast (n=nick84@124.158.139.170) |
| 13:34.23 | [TK]D-Fender | X-Rob>or, for a _real_ example, 0[2378]23322332 ,- might take 2 for this |
| 13:34.26 | [TK]D-Fender | maybe |
| 13:34.28 | X-Rob | if you can re-implement asterisk exten pattern matching in 2 lines, you're a magician. Please produce your code and I'll attribute you in the code |
| 13:35.09 | [TK]D-Fender | X-Rob: you fail to listen to what I said earlier. use a MORE GLOBAL PATTERN tos tart and GotoIF's solve the rest |
| 13:35.18 | X-Rob | ... |
| 13:35.24 | X-Rob | I listened |
| 13:35.25 | [TK]D-Fender | X-Rob: _0. <- |
| 13:35.31 | X-Rob | and I need to pattern match in the GotoIfs |
| 13:35.36 | nicko_ast | hi all |
| 13:35.39 | [TK]D-Fender | X-Rob: 1 fucking DIALPLAN pattern, the rest is dialplan LOGIC |
| 13:36.07 | timeshell | [TK]D-Fender Dude. This is public channel. Please watch the language. |
| 13:36.18 | X-Rob | GotoIf(${EXT} = 0[2378]2332332] ?? |
| 13:36.28 | timeshell | If you don't have the patience to help, then don't. |
| 13:36.37 | X-Rob | as I said, I -could_ do it with regexps, but that would be a pain too |
| 13:36.40 | nicko_ast | is there anybody have a problem like "retrans_pkt: Maximum retries exceeded on transmission" |
| 13:36.51 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 13:36.52 | X-Rob | nicko_ast, it means the other host isn't responding to your packets |
| 13:37.04 | nicko_ast | do you know how to fix it ? |
| 13:37.05 | X-Rob | it's retried it n times, and given up |
| 13:37.16 | X-Rob | nicko_ast, normally, fix the NAT issue. |
| 13:37.18 | [TK]D-Fender | X-Rob: Well you can do it with about 1 line per exception type. Feel free to whine about it all you want. |
| 13:37.43 | elred | Hello. I need to compile chan_misdn but it's "XXX 10. chan_misdn" because it "Depends on: isdnnet(E), misdn(E), suppserv(E)". This is a crosscompilation, I just want to compile chan_misdn and then copy the .so into the right place where I have mISDN/mISDN-user already installed. Anyone got idee how to force the compilation of this module ? Thanks. |
| 13:37.44 | X-Rob | I'm asking. |
| 13:37.57 | nicko_ast | i dont have an idea how to fix NAT issue |
| 13:38.00 | X-Rob | Not whining. There's a difference to me. |
| 13:38.24 | X-Rob | in that I'm trying to find a way to fix my problem, and you seem to believe there's one but won't tell me. |
| 13:38.38 | [TK]D-Fender | X-Rob: What problem? it can be done with minimal dialplan. |
| 13:38.46 | [TK]D-Fender | X-Rob: you dont' HAVE a problem short of just going and doing it |
| 13:39.06 | defsdoor | I think that the DAHDI hardware you have on your system is: |
| 13:39.06 | defsdoor | pci:0000:03:04.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 13:39.11 | X-Rob | How do I match 0[2378]2332332 in dialplan logic? |
| 13:39.13 | defsdoor | that's a good start :) |
| 13:39.29 | X-Rob | without using regexps, I should say |
| 13:39.48 | defsdoor | it's not an A200 |
| 13:39.50 | [TK]D-Fender | X-Rob: Guess you never read the ChANNELVARIABLES docs on expressions & substrings |
| 13:41.31 | X-Rob | re-reads it to see what he's missed |
| 13:41.41 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:41.43 | [TK]D-Fender | $[${EXTEN} = 022332332 || ${EXTEN} = 032332332 || ${EXTEN} = 072332332 || ${EXTEN} = 082332332] <- there's one sample down |
| 13:41.56 | X-Rob | ...right. |
| 13:42.12 | X-Rob | so I have to re-implement the _entire_ dialplan |
| 13:42.19 | nicko_ast | how to fix NAT issue ? is there any documentation about this ? |
| 13:42.29 | X-Rob | and how is this _easier_ than doing an AGI or doing it with regexps? |
| 13:42.36 | [TK]D-Fender | X-Rob: No, 1 stupid line in a GLOBAL PATTERN |
| 13:42.44 | X-Rob | [TK]D-Fender, yes. I'm aware of that. |
| 13:42.55 | X-Rob | let me write what I think you're trying to say |
| 13:42.55 | [TK]D-Fender | X-Rob: you were already doing _0. so whats your probelm? |
| 13:42.57 | X-Rob | will pastebin it |
| 13:43.06 | X-Rob | _0 is just an example |
| 13:43.09 | X-Rob | ignore that |
| 13:43.54 | [TK]D-Fender | [09:32]<X-Rob>Not for 1212XXX1234 <- $[${EXTEN:0:4} = 1212 || ${EXTEN:7:4} = 1234] <- again, 1 line |
| 13:44.03 | [TK]D-Fender | X-Rob: How many exceptions do you ahve? |
| 13:44.09 | X-Rob | an infinite number |
| 13:44.15 | X-Rob | people will type in whatever the hell they want |
| 13:44.40 | [TK]D-Fender | X-Rob: infintie patterns? How are you going to put infinite UNIQUE possiblities intoa dialplan? |
| 13:44.49 | X-Rob | ..they way we do it now? |
| 13:44.52 | [TK]D-Fender | X-Rob: youare using PATTERN to just X nober of cases here |
| 13:45.04 | [TK]D-Fender | number* |
| 13:45.24 | X-Rob | using includes, people type in the dialplan they want to match on, and then set the order they want to match them in |
| 13:45.40 | [TK]D-Fender | X-Rob: a million UNQIUE choices in your dialplan? What are you getting at here? |
| 13:45.49 | X-Rob | no, an infinite number |
| 13:45.57 | [TK]D-Fender | .... |
| 13:45.58 | X-Rob | We don't restrict people in what they can put in pattern matching |
| 13:45.59 | X-Rob | well |
| 13:46.07 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:46.08 | X-Rob | I s'pose there's a SQL table size that will stop them eventually |
| 13:46.39 | [TK]D-Fender | Trying to reinvent an infinitely flexible dialplan on a user level? |
| 13:46.48 | X-Rob | Which currently works |
| 13:46.49 | X-Rob | perfectly |
| 13:46.50 | [TK]D-Fender | X-Rob: Taht will require serious coding. |
| 13:47.01 | X-Rob | it's been working for um, geez, 5 years now? |
| 13:47.02 | elred | blah. I got a problem, using --with-misdn in the ./configure is broken. Because it use it to 1) check the library 2) check the headers, and library and headers are not in the same directory, and it's impossible to give multiple --with-misdn. Any idee ? |
| 13:47.13 | X-Rob | and yes, it was serious coding, thanks. |
| 13:47.45 | X-Rob | but getting back to the START of the conversation, I can't fail from one macro to the other, and I can't think of a better way of doing it. |
| 13:48.22 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 13:48.26 | [TK]D-Fender | X-Rob: I don't see where your actual chain of patterns starts in there. |
| 13:48.31 | [TK]D-Fender | X-Rob: Let alone able to scale. |
| 13:48.32 | X-Rob | hang on |
| 13:48.51 | X-Rob | Have you ever _looked_ at a freepbx dialplan? |
| 13:49.02 | [TK]D-Fender | X-Rob: Yes |
| 13:49.15 | X-Rob | right, do you understand how outbound routes work? |
| 13:49.34 | [TK]D-Fender | X-Rob: marginally between the dialparies, etc |
| 13:49.51 | X-Rob | no, nothign to do with dialparties. That's just for checking for call waiting being disabled and stuff |
| 13:49.54 | X-Rob | on extensions |
| 13:50.13 | [TK]D-Fender | X-Rob: well if its basic dialplan there is a rought limit on what can be done there. |
| 13:50.37 | [TK]D-Fender | X-Rob: How simple you want to be able to add aptterns etc is limited by * config size, etc |
| 13:51.10 | X-Rob | http://pastebin.ca/1377905 outbound routes, to it's bare minimum |
| 13:51.15 | [TK]D-Fender | X-Rob: throwing words like "inifinite" around doesn't help |
| 13:51.24 | X-Rob | ..but it _is_ infinite. |
| 13:51.43 | X-Rob | mathematically infinite, in that we don't limit it. |
| 13:51.50 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 13:51.59 | X-Rob | it might be physically inifinite due to hardware limitations, but there's no soft limit. |
| 13:52.04 | X-Rob | uh |
| 13:52.10 | X-Rob | it might be physically finite due to hardware limitations, but there's no soft limit. |
| 13:52.33 | X-Rob | so there's a basic thing. They dial 1 for one trunk, and 2 for another trunk |
| 13:52.43 | [TK]D-Fender | X-Rob: * can't have an infinitely huge text file for a dialplan, there's a line limit, etc |
| 13:53.00 | [TK]D-Fender | X-Rob: You mean 0 & 1 there, right? |
| 13:53.01 | X-Rob | you don't NEED to have a line for every possible number combination. That's what pattern matching is for |
| 13:53.10 | X-Rob | wups, sorry, yeah. 0 and 1. |
| 13:53.58 | X-Rob | so anyway, the background is that people want to disable trunks on a per user basis, so there's a hook in macro-dialout-trunk to see if they actually have permission to use that trunk. |
| 13:54.20 | X-Rob | anyway |
| 13:54.25 | X-Rob | That outbound route sucks |
| 13:54.30 | X-Rob | the implementation is crap |
| 13:54.32 | [TK]D-Fender | X-Rob: Which puts us more firmly into "maing it cooperate with FreePBX" realm |
| 13:55.07 | X-Rob | but I can't think of a better way of _preserving the ability to set the precedence_, and the ability to fail from one trunk to the other |
| 13:55.09 | [TK]D-Fender | X-Rob: Its got more than its fair share of limitations including multi-tennent, etc |
| 13:55.32 | [TK]D-Fender | X-Rob: well you can do this yourself. |
| 13:55.32 | X-Rob | multi-tennant won't happen without a major rewrite. |
| 13:56.26 | X-Rob | that "_0." is what users have typed in. They're typing in asterisk pattern matching. |
| 13:56.47 | X-Rob | they can match on whatever they want, that asterisk supports |
| 13:57.01 | X-Rob | my point being, there's no way to do pattern matching of extensions _except_ in the actual extension itself |
| 13:57.30 | X-Rob | s/no way/no easy way/ |
| 13:57.33 | tzafrir_laptop | if you have an over-generic dialplan such as freepbx |
| 13:57.48 | canapa | connection is alive, but no sound, this is a common problem right ? |
| 13:58.11 | X-Rob | tzafrir_laptop, it does 99% of what 95% of people want. Anything else, they need someone with a clue. |
| 13:59.20 | X-Rob | canapa, that's a NAT issue. |
| 13:59.27 | *** join/#asterisk path_ (n=path_@3-106-21-190.adsl.terra.cl) |
| 13:59.48 | canapa | X-Rob any advise to get arround this ? |
| 13:59.49 | *** part/#asterisk lbt (n=david@78.32.229.233) |
| 13:59.51 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:59.58 | X-Rob | canapa, google asterisk nat |
| 14:00.14 | tzafrir_laptop | X-Rob, for starters, you can't just add an extension to it without adding a magical db value |
| 14:00.35 | X-Rob | tzafrir_laptop, the device/user one? yeah. |
| 14:01.03 | tzafrir_laptop | And it uses an AGI to resolve things |
| 14:01.03 | X-Rob | that's done transparently if you use the bulkimport module, or do it via the gui |
| 14:01.12 | X-Rob | tzafrir_laptop, like, eg, enum? |
| 14:01.15 | tzafrir_laptop | and tons of macros on top of that |
| 14:01.26 | X-Rob | I had to write that because asterisks Enum lookup is _severely_ broken. |
| 14:01.34 | beherit | what's the possible cause of a warning rtcp "read too short"? |
| 14:02.04 | tzafrir_laptop | the result is a dialplan that is not fun to debug. To say the least |
| 14:02.29 | tzafrir_laptop | Not to mention keeping everything in one context |
| 14:02.41 | X-Rob | tzafrir_laptop, it's reasonably easy, actually. It's well documented, and there are noops everywhere explaing what's going on. It's just LONG, and I agree that sucks. |
| 14:03.05 | X-Rob | there was discussion about having it all in an AGI |
| 14:03.14 | X-Rob | but I vetoed it at the time |
| 14:03.44 | X-Rob | rather selfishly, looking back. It's much more fun to do stuff in dialplan than it is in an agi. |
| 14:05.38 | X-Rob | it would have simplitied it a hell of a lot |
| 14:05.41 | X-Rob | bleh |
| 14:05.46 | X-Rob | typing is sucking tonight |
| 14:07.39 | jermey_g | hates vaguie idiots who say 'bad things will happen behind nat'. all my nat problems are solved u ...$## |
| 14:08.07 | X-Rob | jermey_g, and until you solved them, bad things happened, right? |
| 14:08.14 | jplank | NAT isn't so bad as long as only the IP phone is behind it :) |
| 14:08.25 | *** join/#asterisk neurosys (n=vinix@67.32.132.60) |
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| 14:09.04 | beherit | x-rob, do you happen to know the possible cause of rtcp read too short warning in the asterisk? |
| 14:09.46 | [TK]D-Fender | jplank: both behind NAT has never been a problem for me or my clients |
| 14:10.23 | X-Rob | beherit, not really. |
| 14:12.20 | beherit | i've been looking at the web but still no answer for that warning message |
| 14:12.24 | jermey_g | X-Rob:for every problem, there exist a valid reason. people blame the problem. |
| 14:12.30 | jermey_g | is dancing with joy |
| 14:13.10 | X-Rob | beherit, apparently that error was taken out of asterisk at svn revision r87394 |
| 14:13.18 | X-Rob | so it sounds like 'upgrade your asterisk' |
| 14:13.53 | beherit | x-rob i would like to know what version of asterisk affected by this? |
| 14:14.02 | X-Rob | Author: file |
| 14:14.03 | X-Rob | Date: Mon Oct 29 15:02:31 2007 |
| 14:14.03 | X-Rob | New Revision: 87394 |
| 14:14.03 | X-Rob | URL: http://svn.digium.com/view/asterisk?view=rev&rev=87394 |
| 14:14.03 | X-Rob | Log: |
| 14:14.03 | X-Rob | Drop the RTCP Read too short message to debug. There are some phones out there that send a sort of keep alive packet in the RTCP that trigger this every 5 seconds. |
| 14:14.06 | X-Rob | Modified: |
| 14:14.08 | X-Rob | trunk/main/rtp.c |
| 14:15.02 | beherit | that was long ago |
| 14:15.11 | beherit | im usin using asterisk 1.4.22.2 |
| 14:15.33 | X-Rob | beherit, well upgrade the client then, because that error only occurs when the packet is wrong |
| 14:16.48 | *** part/#asterisk nicko_ast (n=nick84@124.158.139.170) |
| 14:16.52 | beherit | X-Rob: im using eyebeam, by the way this only happen last night two times but not its working fine. I just want to know the cause so i can avoid it in the future. because when i encounter this error all active calls are being drop |
| 14:18.00 | beherit | X-Rob: im using eyebeam, by the way this only happen last night two times but NOW its working fine. I just want to know the cause so i can avoid it in the future. because when i encounter this error all active calls are being drop |
| 14:18.56 | *** join/#asterisk path_ (n=path_@82-106-21-190.adsl.terra.cl) |
| 14:19.10 | X-Rob | beherit, dunno. sounds odd. That error only occurs when the contents of the packet say it's a certan length, but the packet, as seen by asterisk, isn't that lengh. If it's happening spuriously, I'd be checking for hardware issues. |
| 14:19.27 | X-Rob | eg, change the network card, cable, switch |
| 14:20.34 | beherit | We already change the switch 3 hours ago and we will observe if the error will show again |
| 14:21.29 | beherit | X-rob thank you for your support, so this has nothing to do with the asterisk config or asterisk version. |
| 14:22.05 | elred | I have that in config.log : /usr/bin/ld: cannot find -lisdnnet |
| 14:22.05 | elred | collect2: ld returned 1 exit status |
| 14:22.05 | elred | configure:20265: $? = 1 |
| 14:22.20 | elred | that means if found "misnd" and "suppserv" but not libsdnnet |
| 14:22.42 | elred | anyone know where to find it ? None of the doc I am reading to enable chan_misdn mentioned it. |
| 14:25.14 | elred | ok I found the solution |
| 14:28.10 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
| 14:28.28 | tzafrir_laptop | How have you installed misdnuser? From where? What version? |
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| 14:53.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:02.09 | Carlos_PHX | Anyone know if China blocks VoIP traffic in/out of the country? |
| 15:02.32 | KavanS | Carlos_PHX, vpn |
| 15:02.36 | deadpigeon | no it doesn't. |
| 15:03.08 | Carlos_PHX | We're trying to avoid the need for VPN. |
| 15:06.05 | Talkradio | speaking of vpn's any recommendations for a vpn firewall that works well with remote clients? |
| 15:08.00 | defsdoor | Talkradio: linux |
| 15:08.11 | defsdoor | Talkradio: + openvpn |
| 15:08.22 | *** join/#asterisk avrPhreaque (i=Default@dsl-dynamic-209-50-30-164.inebraska.com) |
| 15:08.32 | defsdoor | I have 100% uptimes for openvpn vpns over a couple of years period |
| 15:09.10 | avrPhreaque | Just as an FYI: I found what I was asking about yesterday: Silicon Labs makes a pair of chips, the Si3201 and Si3216, which provide all the services necessary to appear to a telephone instrument as a CO. |
| 15:09.28 | jjshoe | defsdoor how many vpn connections? |
| 15:09.41 | avrPhreaque | the 3216 is the actual SLIC, then 3201 provides the physical interfae the the telephone instrtument |
| 15:09.54 | defsdoor | jjshoe: one site I look after has a few dozen (netmodules) |
| 15:11.48 | defsdoor | http://www.netmodule.com/en/products/nb/ |
| 15:12.27 | defsdoor | embedded linux with openvpn support via GPRS/UMTS - we used them for remote CCTV deployment |
| 15:14.11 | avrPhreaque | Just wanted yo let you folks know I found what I was looking for, and thanks for your assistance yesterday |
| 15:15.13 | coppice | The Si3201 is the high voltage chip |
| 15:15.21 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 15:15.34 | avrPhreaque | Coppice: right |
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| 15:19.27 | Gh0sty | anyone here have experience with asterisk .NET ? |
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| 15:30.28 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 15:30.32 | drfreeze | Hello |
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| 15:31.11 | drfreeze | I seem to be having some problems getting a call to timeout for a Zap channel |
| 15:31.16 | drfreeze | http://pastie.textmate.org/private/d8oj3hibc8v6fesyut9vgg |
| 15:31.23 | *** part/#asterisk avrPhreaque (i=Default@dsl-dynamic-209-50-30-164.inebraska.com) |
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| 15:33.16 | tzafrir_laptop | drfreeze, exten => s,800,Dial(Sip/715,25,tr) ? |
| 15:33.30 | tzafrir_laptop | 715 is just a dialplan extension, right? |
| 15:33.32 | [TK]D-Fender | drfreeze: -- Zap/6-1 answered Zap/1-1 <------- ANSWERED |
| 15:33.35 | defsdoor | sangoma ./Setup dahdi errors :( |
| 15:33.51 | [TK]D-Fender | drfreeze: welcome to the wonderful world of "OMG I have no call progress" |
| 15:36.40 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
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| 15:37.33 | bmoraca | dial the sip channel to reach a local extension? sounds like a neat trick! |
| 15:37.43 | bijit | asterisk can handle more than 500 exten and about 150 simultanous call without a problem depending on the server right? |
| 15:39.38 | bmoraca | bijit: depends on the server and what the calls are doing. if you're playing MP3 files to all of them, it's far more costly than just bridging 150 calls of the same codec |
| 15:39.41 | [TK]D-Fender | bijit: Sure |
| 15:41.15 | bijit | Thanks Guys, I heard a comment that asterisk after 400 its starts acting wierd, But to me it sounded like a server issue instead of asterisk.. |
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| 15:50.44 | drfreeze | tzafrir_laptop: yes |
| 15:50.55 | *** join/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 15:51.03 | c0rnoTa | hello guys |
| 15:51.06 | workdraft | anybody tried asterisk behind DLink DI-LB604, a load balancing router? |
| 15:51.22 | c0rnoTa | a had asterisk crash today. it was started with -g option |
| 15:51.27 | c0rnoTa | where i can fined coredump? |
| 15:52.06 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 15:52.22 | drfreeze | [TK]D-Fender: you are talking in code - I do not understand |
| 15:52.41 | [TK]D-Fender | drfreeze: the call has been ANSWERED. there is no timeout <- |
| 15:53.00 | [TK]D-Fender | drfreeze: This is not "code" , it is not considered as being "ringing" |
| 15:53.29 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 15:55.32 | drfreeze | [TK]D-Fender: hmm, is there a way to timeout on the 'ringing'? |
| 15:56.14 | [TK]D-Fender | drmit isn't "ringing" that's your problem. |
| 15:56.21 | [TK]D-Fender | drfreeze: it isn't "ringing" that's your problem. |
| 15:56.28 | [TK]D-Fender | drfreeze: What are you calling out on? |
| 15:56.33 | *** part/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 15:56.58 | xrmx__ | hi, which is the preferred asterisk tapi implementation to dial from outlook? |
| 15:57.28 | drfreeze | calling out on zap channel |
| 15:58.04 | [TK]D-Fender | drfreeze: DUH, what KIND? What CARD? |
| 15:58.08 | *** part/#asterisk Gh0sty (n=ghosty@ip-81-11-174-100.dsl.scarlet.be) |
| 15:58.14 | drfreeze | TDM card |
| 15:58.46 | drfreeze | Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) |
| 15:58.56 | [TK]D-Fender | drfreeze: Your only choice for this is to set "callprogress=yes" in zapata. This also has a habit of occasionally dropping calls for no solid reason |
| 15:59.05 | elred | I got funny bug with chan_misdn.so : 1) a segfault (http://pastebin.com/m103b549) 2) a memleak (if you specify a misdn port 144442) if will leak what is at port[144442] |
| 15:59.15 | drfreeze | lovely |
| 15:59.20 | [TK]D-Fender | drfreeze: because by default the call is considered answered the moment the Zap chanel is seized to place the call |
| 15:59.31 | *** join/#asterisk Great_Anta_Baka (i=c4249986@gateway/web/ajax/mibbit.com/x-8ebd68a28ee2c381) |
| 16:00.27 | drfreeze | any other side affects? |
| 16:00.41 | drfreeze | if I switch to PRI, will I have this problem? |
| 16:01.39 | [TK]D-Fender | drfreeze: Generally no, PRI supports call progress |
| 16:02.04 | [TK]D-Fender | drfreeze: certain switches you pass through may be a probel, but that'll be typically very rare |
| 16:02.11 | [TK]D-Fender | problem |
| 16:02.25 | Great_Anta_Baka | Hi all. I have an asterisk box setup for this residential area of 700 homes. Each extension has its own sip trunk to our voice provider who does the billing for us |
| 16:03.12 | Great_Anta_Baka | this has resulted in each extension having to dial throgh its own trunk |
| 16:03.31 | Great_Anta_Baka | I am using templates for setting up extensions.conf |
| 16:04.08 | Great_Anta_Baka | how can i get it so that I can create a generic template with each dialing context having its own variables, like |
| 16:04.20 | Great_Anta_Baka | ;[dialerall](!) ;exten => _XXXXXXXXXXX,1,Dial(SIP/SIPTRUNK) |
| 16:04.35 | Great_Anta_Baka | and then in the context |
| 16:04.38 | *** join/#asterisk RoadKill (n=Miranda@gate.ipvision.dk) |
| 16:04.42 | ajohnson | in sip.conf for each of the 700 homes do something like this |
| 16:04.57 | Great_Anta_Baka | k |
| 16:04.59 | ajohnson | setvar=trunk=trunk655 |
| 16:05.16 | ajohnson | _XXXXXXXXXX,1,Dial(SIP/${trunk}) |
| 16:05.26 | ajohnson | so... |
| 16:05.37 | ajohnson | You have 1400 entries in sip.conf for 700 homes? |
| 16:05.53 | ajohnson | one for the home device and one for your provider? |
| 16:06.05 | Great_Anta_Baka | yup basically something like that |
| 16:06.13 | ajohnson | that should work just fine |
| 16:06.14 | canapa | [TK]D-Fender: i've got, nat=yes, localnet and externalip set in sip.conf ? any other advises than google ? |
| 16:06.28 | [TK]D-Fender | canapa: read : |
| 16:06.30 | [TK]D-Fender | ~sipnat |
| 16:06.31 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:06.41 | canapa | k |
| 16:07.21 | Great_Anta_Baka | so ajohnson how will each trunk know which trunk to dial through? |
| 16:07.34 | Great_Anta_Baka | I have to create 700 setvar rules? |
| 16:07.48 | ajohnson | You have already had to create 1400 entries... |
| 16:07.52 | Great_Anta_Baka | unique* |
| 16:08.11 | [TK]D-Fender | 700 entries SOMEWHERE |
| 16:08.16 | Great_Anta_Baka | k |
| 16:08.33 | ajohnson | Asterisk != magic |
| 16:08.56 | Great_Anta_Baka | i know that :P |
| 16:08.59 | Great_Anta_Baka | hence the using templates |
| 16:09.02 | ajohnson | I'm sure you can create a script to create everything very quickly |
| 16:09.25 | ajohnson | The great thing about this is that you only need one context in extensions.conf to handle the dialing |
| 16:09.27 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 16:09.48 | Great_Anta_Baka | i see |
| 16:10.18 | [TK]D-Fender | :/ |
| 16:11.20 | drfreeze | [TK]D-Fender: all my important calls are with patients over zap. Before I turn on 'callprogress=yes', just how bad is this call dropping problem you mention? |
| 16:11.25 | kippi | hey |
| 16:11.34 | kippi | having a problem with music on hold |
| 16:11.43 | kippi | it starts playing, then it will stop |
| 16:11.52 | kippi | but then sometimes it will play |
| 16:11.59 | kippi | is there away to debug this? |
| 16:14.10 | [TK]D-Fender | drfreeze: YMMV |
| 16:14.16 | [TK]D-Fender | drfreeze: Try and see |
| 16:15.59 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:18.20 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 16:18.42 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:21.15 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:23.22 | jameswf | they see him trollin.... haten.... |
| 16:24.02 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 16:27.21 | Qwell | jameswf: what? |
| 16:27.23 | Qwell | jameswf: :p |
| 16:27.37 | Qwell | I have every right to ask that |
| 16:28.09 | Qwell | (seeing how they're kinda violating the GPL and *my* copyright at the moment) |
| 16:28.42 | Qwell | also, is a little credit too much to ask for? |
| 16:29.34 | X-Rob | Qwell, who's stealing your stuff? |
| 16:29.39 | X-Rob | 's ears perk up |
| 16:29.41 | Qwell | not stealing |
| 16:30.01 | X-Rob | gpl violation == stealing |
| 16:30.09 | Qwell | X-Rob: trixbox is using my AsteriskNOW RPMs in the latest trixbox beta though. I asked where the yum repo is that contains the source |
| 16:30.19 | cesar_CR | hello guys, with my ISP SIP user I can recieve calls with xlite, but with * not, this is the capture http://pastebin.ca/1378031 its an invite and a 401 unauthorised response from *, thanks a lot if there is anybody willing to see it |
| 16:30.21 | X-Rob | Qwell, grr. |
| 16:30.22 | jameswf | creatively re-allocating |
| 16:30.28 | Qwell | jameswf: something like that |
| 16:30.38 | X-Rob | Seriously, I'm going to fucking make isopbx one day |
| 16:30.42 | Qwell | jameswf had asked about a week ago, and got no answer. |
| 16:30.43 | X-Rob | and I'll spit on fonality and trixbox |
| 16:30.51 | Qwell | X-Rob: meh. just help me with AsteriskNOW and be done with it |
| 16:31.00 | *** join/#asterisk romenquettadil (n=romen@93.88.112.186) |
| 16:31.16 | jameswf | X-Rob, you are going to fork andrews work |
| 16:31.28 | outtolunc | still using rpath? |
| 16:31.35 | Qwell | outtolunc: no, CentOS 5.latest |
| 16:31.40 | outtolunc | sweet |
| 16:31.40 | romenquettadil | hi anyone with mISDNv2 + LCR + asterisk experience? |
| 16:31.45 | jameswf | andrew owns the iso foremula everyone is stealing |
| 16:31.48 | X-Rob | jameswf, dunno. I was thinking of a debian based system, but I _am_ comfortable with centos |
| 16:31.52 | X-Rob | Qwell, what needs doing? |
| 16:32.02 | Qwell | X-Rob: testing from somebody who can fix the bugs in freepbx, heh |
| 16:32.10 | X-Rob | Qwell, ah. That would be me then, I guess 8) |
| 16:32.20 | jameswf | I wanna see debian or even buntu system |
| 16:32.41 | Qwell | jameswf: I suspect that if anything, it would be based on an Ubuntu LTS release |
| 16:32.44 | X-Rob | jameswf, there's no 'formula' for rolling your own centos CD with additional stuffo nit. |
| 16:32.54 | Qwell | where "it" is a yet-to-be-imagined distro |
| 16:33.01 | X-Rob | Qwell, that's what I was thinking about - Ubuntu LTS, but CentOS is just as good for long term support. |
| 16:33.03 | *** join/#asterisk surfdue (n=anthonyu@unaffiliated/surfdue) |
| 16:33.06 | X-Rob | just that it's so damn ancient. |
| 16:33.08 | surfdue | ^noob :} |
| 16:33.22 | surfdue | not you rob. |
| 16:33.31 | X-Rob | I so am. |
| 16:33.40 | surfdue | haha you are? |
| 16:33.44 | X-Rob | Well. No. |
| 16:33.50 | surfdue | oh. <<< noooob |
| 16:33.52 | X-Rob | Qwell, stick with CentOS, and tell me what's broken. |
| 16:33.52 | surfdue | boob |
| 16:33.54 | surfdue | :P |
| 16:34.07 | surfdue | the b is next to the n! |
| 16:34.14 | Qwell | X-Rob: There's an issue Ethan brought up that I figured out. Recordings for an IVR don't have proper permissions when copied.. |
| 16:34.15 | jameswf | X-Rob, http://trixbox.org/forums/trixbox-forums/open-discussion/hello-world from andrew: I'll check it out. I always like to see another fork of my original asterisk@home. It's funny to see the twist each company puts on my original project to make it their own. |
| 16:34.22 | Qwell | amp_conf/htdocs/admin/modules/recordings/page.recordings.php, line ~93 |
| 16:34.26 | Qwell | That whole case statement is busted |
| 16:34.34 | X-Rob | jameswf, *bahahahaha* |
| 16:34.37 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 16:34.50 | *** part/#asterisk surfdue (n=anthonyu@unaffiliated/surfdue) |
| 16:35.13 | Qwell | X-Rob: basically, the files need to be chmod'ed after the copy |
| 16:35.22 | X-Rob | yes, they do - and they should. |
| 16:35.34 | X-Rob | The version I'm looking at -- this is the switch($action) one, right? |
| 16:35.39 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-22.hsd1.fl.comcast.net) |
| 16:36.08 | Qwell | X-Rob: yeah, but the problem is case "recorded": |
| 16:36.31 | Qwell | pretty much that entire case is wrong. starting from the if (!file_exists($astsnd."custom")) { |
| 16:36.48 | X-Rob | Um |
| 16:37.06 | Qwell | (it never hits the else, even though the if does the mkdir...) |
| 16:37.06 | X-Rob | I think you have an old version |
| 16:37.06 | Qwell | 2.5.1 |
| 16:37.06 | X-Rob | <PROTECTED> |
| 16:37.06 | X-Rob | <PROTECTED> |
| 16:37.06 | X-Rob | <PROTECTED> |
| 16:37.10 | X-Rob | oh |
| 16:37.12 | X-Rob | sorry |
| 16:37.15 | X-Rob | wrong bit |
| 16:37.17 | X-Rob | there it is |
| 16:37.34 | Qwell | that else can be removed, and unindent the whole section |
| 16:37.47 | Qwell | if it doesn't exist, mkdir it, and continue where the else was |
| 16:38.00 | X-Rob | yup. That makes sence |
| 16:38.05 | X-Rob | that's actually my code, and there was extra bits there |
| 16:38.06 | Qwell | but, the real issue is that the cp doesn't have correct permissions, so it needs to be chmod'ed after |
| 16:38.10 | X-Rob | but I can't remember what it was |
| 16:38.11 | romenquettadil | I'm trying to get asterisk and LCR working togheter to be able to use mISDN drivers? there's anyone that can help me? |
| 16:38.24 | X-Rob | Qwell, you're trying to run apache as httpd, right? |
| 16:38.25 | jameswf | could you do cp -p |
| 16:38.32 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:39.34 | Qwell | X-Rob: apache runs as apache:apache, asterisk runs as asterisk:asterisk. user apache is in group asterisk |
| 16:40.04 | X-Rob | ooh |
| 16:40.07 | X-Rob | I remember why I did it that way |
| 16:40.07 | Qwell | jameswf: nope, it's a different user. |
| 16:40.16 | Qwell | X-Rob: cross-filesystem copies |
| 16:40.22 | Qwell | erm, moves |
| 16:40.27 | X-Rob | Yeah |
| 16:40.35 | Qwell | mv would potentially fail or something. |
| 16:40.53 | jameswf | I like to cheat sometimes and use install rather than cp |
| 16:41.08 | X-Rob | The reason why the ELSE is there, is if that mkdir fails you'll lose your file. |
| 16:41.13 | Qwell | Philippe didn't like the idea of just recording to /var/lib/asterisk/sounds/tmp/ and moving from there |
| 16:41.15 | X-Rob | I could probably do it better than that |
| 16:41.25 | jameswf | install -o asterisk -m 644 foo bar |
| 16:41.30 | Qwell | (that would be "best" though, since file owner would be correct) |
| 16:41.42 | Qwell | jameswf: can't chown from another user |
| 16:41.48 | jplank | on a tdm2400 with 4 FXO modules, whenever a call originates from the PSTN, and the asterisk side hang ups the call, within a couple seconds asterisk picks back up the line and routes it like a new call was coming in. This ONLY happens when the call originates from the PSTN and the asterisk side hangs up, with any other combo, it doesn't happen. any ideas? |
| 16:41.51 | Qwell | only root can chown to a user that isn't themself |
| 16:41.53 | X-Rob | Anyway |
| 16:41.55 | X-Rob | it's 2:30am |
| 16:41.59 | X-Rob | I should go to bed. |
| 16:42.03 | X-Rob | That's not just a simple fix |
| 16:42.06 | jameswf | lives as root :( |
| 16:42.10 | X-Rob | it needs to be redone so it's a bit more sensible |
| 16:42.29 | X-Rob | That 'if !mkdir' is a bugfix, which is why it's just wedged in there at the top |
| 16:42.48 | Qwell | X-Rob: well, like I said, that's all secondary. the real issue is the lack of a chmod after copy |
| 16:43.19 | X-Rob | I'm happy to stick it in there, as that's definately a bug |
| 16:43.29 | X-Rob | but I just don't want to tread on anyone's toes by doing so. |
| 16:43.37 | X-Rob | and phillipe has buggered off for a fortnight on a boat. |
| 16:47.32 | jplank | what causes zaptel to pick up a ringing FXO line, the voltage increase? |
| 16:48.19 | [TK]D-Fender | jplank: Answer() <- |
| 16:48.49 | [TK]D-Fender | jplank: It'll notice it for the voltage cadence. |
| 16:49.36 | jplank | I'm trying to figure out what is cause asterisk to pick back up the line thinking there is someone calling after it hangs up a call on the same channel |
| 16:50.24 | jplank | ztmonitor shows the Rx start signaling, but that doesn't make any sense |
| 16:51.31 | [TK]D-Fender | jplank: Sounds like it might be a CDS hiccup |
| 16:51.57 | jplank | CDS = call disconnect signalling? |
| 16:52.13 | elred | can someone point me a doc or explain me the difference between TE and NT mode for an ISDN card ? (B100P) |
| 16:53.53 | [TK]D-Fender | ~cds |
| 16:53.54 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 16:54.35 | jplank | thats what weird though, if the PSTN side disconnects the call, everything works, but if asterisk disconnects the call, thats when the issue happens |
| 16:54.43 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:54.45 | jplank | and only happens if the PSTN originates the call |
| 16:54.45 | *** join/#asterisk jaybuffet (n=jaybuffe@user-12lc518.cable.mindspring.com) |
| 16:54.46 | elred | jplank : are you in France ? |
| 16:54.50 | jplank | no US |
| 16:54.53 | elred | ok |
| 16:55.21 | defsdoor | jplank: I'm in uk and had to get BT to turn on disconnect supervision - they call it something different mind |
| 16:55.55 | jplank | if it was the disconnect supervision, wouldn't it happen when the PSTN side hangs up the call? |
| 16:55.59 | defsdoor | that was on 6 analog lines - decided never to touch analog again after that install |
| 16:56.33 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 16:56.33 | jplank | like if someone calls in from the PSTN, got VM, and hangs up, the call disconnects |
| 16:57.20 | *** part/#asterisk jaybuffet (n=jaybuffe@user-12lc518.cable.mindspring.com) |
| 16:57.27 | jplank | but if someone calls in from the PSTN, gets a users at exten 123, then exten 123 hangs up, the far end caller shows the call up for a couple seconds, then it drops, then asterisk picks back up the line as if someone was calling in |
| 16:57.53 | jplank | almost like a reverse flashhook |
| 16:58.25 | [TK]D-Fender | jplank: I've heard several cases like this before. |
| 16:58.54 | jplank | do you know what they did to resolve it? |
| 16:59.12 | [TK]D-Fender | jplank: regrettably no |
| 16:59.18 | jplank | I have my tech onsite making sure theres no battery being introduced from somewhere else or something like that |
| 16:59.41 | *** join/#asterisk SparFux (n=raoul@e182029069.adsl.alicedsl.de) |
| 16:59.42 | jplank | and I'm sure this has something to do with me not receiving caller ID |
| 16:59.55 | jplank | I HATE TDM |
| 17:00.20 | SparFux | Hello! Is there an application for asterisk which implements a music player on a telephone, like with play, pause, rewind, and song selection and so son. |
| 17:01.33 | *** join/#asterisk mvanbaak_ (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:01.59 | *** join/#asterisk saftsack (n=saftsack@p57924807.dip.t-dialin.net) |
| 17:02.01 | [TK]D-Fender | SparFux: "show application controlplayback |
| 17:03.07 | SparFux | Fender: But it will not allow brosing a music database, right? |
| 17:03.26 | [TK]D-Fender | SparFux: taht deals with control of the given song, the rest is up to YOU |
| 17:04.39 | beek | Nothing quite like listening to music across 8KHz bandwidth, is there? |
| 17:05.00 | beek | Put it on speakerphone and you could have a party. |
| 17:06.55 | SparFux | beek: I even want to be able to control output channel and then I can play on cable-less speakers in the garden and control everything from the telephone. |
| 17:07.29 | defsdoor | or buy an ipod |
| 17:07.47 | SparFux | just buying some stuff is lame :-) |
| 17:08.01 | defsdoor | sometimes the easiest solution is the best |
| 17:08.15 | SparFux | I want to exten my absolute multi-tool, the pc with asterisk on it ;-) |
| 17:08.36 | defsdoor | get a squeezebox and use the web interface to remote control it |
| 17:08.42 | [TK]D-Fender | SparFux: you can control several music apps VIA CLI. Just do system calls to do that. |
| 17:09.06 | [TK]D-Fender | SparFux: And from there get proper audio out |
| 17:09.27 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 17:09.53 | defsdoor | my home * box is also my movie filestore and slimserver and nagios server |
| 17:10.21 | SparFux | Fender: Yes, I might control xmms2d and use it for asterisk. |
| 17:11.13 | defsdoor | uses a nokia n810 to control everything |
| 17:11.53 | [TK]D-Fender | defsdoor: Only your filestore? Mine is my full HTPC and used to make me coffee, host my sites, FTP, etc :) |
| 17:12.11 | defsdoor | I have hosted server for that |
| 17:12.17 | defsdoor | apart from the coffee bit |
| 17:12.53 | defsdoor | I have 12 external USB drives attached that I can power on/off selectively |
| 17:12.58 | [TK]D-Fender | probably has about the biggest screen of any * server out there... |
| 17:13.00 | defsdoor | 5TB+ |
| 17:14.03 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:14.51 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 17:16.50 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 17:18.31 | guax | whats wrong with this CLI commmand? originate SIP/8037 9157000 1000@orig |
| 17:18.43 | defsdoor | http://www.quasarelectronics.com/velleman/k8056-8-channel-remote-relay-card-kit.htm < I have 2 of those that power the drives on/off |
| 17:19.57 | guax | got it |
| 17:19.59 | guax | =) |
| 17:21.06 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 17:24.05 | *** join/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net) |
| 17:24.41 | rpm | anyone have an example configuration file for a mediatrix 1104? i cannot figure out how to make one - and the configuration which i download off the device in unit manager does not work on a http server for provisioning |
| 17:25.10 | jplank | fender: any idea how I can trouble shoot this? |
| 17:25.30 | *** join/#asterisk path_ (n=path_@190.196.17.106) |
| 17:26.17 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 17:31.15 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 17:33.32 | [TK]D-Fender | jplank: Not so much.... |
| 17:34.03 | jplank | checking the voltage on it, I would LOVE for it to be a bell problem |
| 17:38.33 | *** join/#asterisk Mark21 (n=mark@freenode/sponsor/mark17) |
| 17:38.50 | *** join/#asterisk DavidR2008 (n=chatzill@c-68-34-116-180.hsd1.md.comcast.net) |
| 17:39.02 | Mark21 | hello, is there a way to reload sip.conf and extensions.conf without restarting asterisk? (i use asterisk 1.6.0.6) |
| 17:40.41 | jaytee | sip reload and then dialplan reload |
| 17:41.13 | [TK]D-Fender | Mark21: "reload" |
| 17:41.22 | jaytee | yeah, that works too |
| 17:42.34 | *** part/#asterisk gultig (n=dbriggs@70.96.32.114) |
| 17:43.37 | DavidR2008 | is there a way to tell if transcoding is working? I've got a softphone that only does gsm and a hard phone that does G.711u and GSM. the softphone can't hear the hardphone if the hardphone initiates the call |
| 17:47.26 | [TK]D-Fender | DavidR2008: Sounds more like a typical NAT issue |
| 17:47.39 | [TK]D-Fender | DaviTranscoding would not fail to produce sounds |
| 17:47.51 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-190-12.telkomadsl.co.za) |
| 17:48.16 | RoadKill | you might want to try set the the codec in asterisk to gsm only for both of them |
| 17:48.46 | DavidR2008 | I'm not using NAT it's all private on the local lan |
| 17:49.02 | RoadKill | you can also type cors show channels verbose |
| 17:50.55 | [TK]D-Fender | DavidR2008: Go take a very close look at the SIP debug of the failed call. |
| 17:51.05 | [TK]D-Fender | DavidR2008: And check your firewalls |
| 17:54.13 | DavidR2008 | [TK}D-Fender: what am I looking for in the SIP debug? |
| 17:54.44 | defsdoor | could reinvite be an issue also ? |
| 17:54.54 | *** join/#asterisk _brent_ (n=_brent_@jive.fttp.xmission.com) |
| 17:54.59 | DavidR2008 | I have canreinvite=no in sip.conf |
| 17:55.00 | [TK]D-Fender | DaviConfirm the codecs selected, odd ports, mismatched NAT claims, etc. |
| 17:56.14 | DavidR2008 | 'k thx! |
| 17:56.57 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:00.57 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:03.50 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 18:04.26 | *** join/#asterisk Ritzerisk (n=Ritztech@65.105.209.226.ptr.us.xo.net) |
| 18:09.28 | *** join/#asterisk _BBV_ (n=buklov@213.138.71.254) |
| 18:10.20 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 18:14.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:15.51 | *** join/#asterisk Ralcala (n=demo@201.127.95.200) |
| 18:16.18 | Ralcala | hi |
| 18:17.00 | Ralcala | i need make some questions where chan you recommend |
| 18:17.19 | Mark21 | Ralcala: if it is about asterisk: just ask |
| 18:17.23 | Ralcala | ok |
| 18:17.25 | Ralcala | thanks |
| 18:18.39 | Ralcala | i just installed 1.6 version with FC9 and client side X-Lite but registering its always time out |
| 18:19.08 | Ralcala | i already registering but with FC3 and * 1.1 with suceed |
| 18:19.41 | Mark21 | Ralcala: login to the asterisk cli and set sip debug on |
| 18:19.44 | [TK]D-Fender | Ralcala: Sounds like a routing/firewall issue |
| 18:20.17 | _brent_ | Ralcala: did you tell iptables to allow traffic on UDP port 5060? |
| 18:20.19 | Ralcala | server its on NAT but client its local and with no NAT.... |
| 18:20.22 | Mark21 | or a config issue if the configs aren't updated |
| 18:21.03 | [TK]D-Fender | Ralcala: check the FW on the server itself |
| 18:21.18 | Ralcala | i got a FC8 firewall with already checked with * 1.0 |
| 18:22.05 | Ralcala | and yep its allowed udp packets for port 5060.... |
| 18:22.45 | _brent_ | then do like Mark21 said and watch the CLI |
| 18:22.47 | Ralcala | BTW i loaded the new modules for conntrack_sip and nat_sip from the fw script... |
| 18:23.15 | [TK]D-Fender | Ralcala: trash your FW and testr |
| 18:23.24 | Ralcala | k |
| 18:23.52 | *** join/#asterisk j3g (n=andrer@200.130.18.1) |
| 18:24.31 | j3g | is there a easy way to monitor a specific trunk in asterisk? i have a prodution server with a lot of phones in use... so reading the logs is very hard... (mixed logs) |
| 18:24.43 | j3g | does asterisk provide a better way to monitor for something specific? |
| 18:25.07 | [TK]D-Fender | j3g: "something? care to narrow that down? |
| 18:25.11 | j3g | a trunk |
| 18:25.13 | j3g | or a phone |
| 18:25.19 | j3g | something like that |
| 18:26.00 | [TK]D-Fender | j3g: You can selectively SIP debug an IP/peer. Thats it |
| 18:26.22 | j3g | not a trunk? |
| 18:26.34 | j3g | its a iax trunk |
| 18:26.46 | *** join/#asterisk kink0 (n=xchat@212.170.176.86) |
| 18:26.50 | kink0 | hello |
| 18:27.22 | [TK]D-Fender | j3g: Same thing |
| 18:27.52 | kink0 | one question about SS7 config in Asterisk, is possible to have a separate bearer trunk ? i.e. voice channels on one Cisco, and signalling T1 on the asterisk box ? |
| 18:30.42 | Ralcala | same error fellas |
| 18:31.11 | *** join/#asterisk dieguito84 (n=diego@host19-194-dynamic.12-79-r.retail.telecomitalia.it) |
| 18:31.17 | Ralcala | can you send and sip.conf and extensions.conf files that worked for you, please..... |
| 18:31.42 | j3g | [TK]D-Fender: i have "sip debug" but no iax2 debug |
| 18:32.07 | [TK]D-Fender | j3g: "help iax2" <- look a little harder |
| 18:32.32 | _brent_ | j3g: in newer versions you have to do {module} set debug {on|off} |
| 18:33.55 | [TK]D-Fender | Ralcala: If you have no PACKETS then you have a networking issue. |
| 18:34.09 | [TK]D-Fender | Ralcala: this is not an * config issue |
| 18:34.45 | Ralcala | no sip debug info on CLI console.... |
| 18:35.34 | [TK]D-Fender | Ralcala: Which means either you didn't enable it properly or you have anetworking issue |
| 18:36.15 | j3g | [TK]D-Fender: iax2 set debug trunk seems to monitor all trunks... there's no way to specify a given trunk |
| 18:36.15 | jplank | what should a POTS line read at onhook? |
| 18:36.39 | jjshoe | jplank can't recall off the top of my head, 50 some volts i think |
| 18:36.46 | jjshoe | jplank 200 or so ringer |
| 18:36.49 | jjshoe | jplank less then 10 off hook |
| 18:37.09 | jjshoe | 55 for on hook i think |
| 18:37.20 | jplank | thanks |
| 18:38.05 | [TK]D-Fender | should be the reverse |
| 18:38.07 | hackeron | I downloded the asterisk-moh-free play in ulaw format but it's playing too loud - is there anyway to make it play quiter? |
| 18:38.16 | jplank | I'm reading about 111 on hook |
| 18:38.37 | [TK]D-Fender | hackeron: Its your endpoints problem. *'s sounds are already normalized |
| 18:39.11 | hackeron | [TK]D-Fender: they are normalised, but I want the music to be quieter than the person speaking - currently it's the same volume |
| 18:39.13 | jjshoe | jplank line, not phone, correct? |
| 18:39.24 | jplank | yea |
| 18:39.32 | jjshoe | jplank seems way too high |
| 18:39.40 | [TK]D-Fender | hackeron: Feel free to SOX them to your hearts content |
| 18:39.41 | jplank | I'm getting weird readings I'm getting about 16 offhoook |
| 18:39.45 | jplank | 168 ringing |
| 18:39.50 | jplank | and about 111 onhook |
| 18:40.06 | jjshoe | hot line |
| 18:40.24 | jplank | grrrrrr I'm connected directly to verizon's block getting it |
| 18:40.35 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 18:41.42 | hackeron | [TK]D-Fender: heh, ok, thanks :) |
| 18:42.25 | Ralcala | how can i test udp packets to 5060 port (telnet or ssh)?.... |
| 18:42.42 | *** join/#asterisk M07w (n=M07w@216.234.116.122) |
| 18:42.49 | _brent_ | Ralcala: nope, those are tcp |
| 18:43.00 | _brent_ | oddly enough, though, tcpdump will pick up udp traffic |
| 18:43.13 | _brent_ | tcpdump -A -s 0 -i eth0 port 5060 |
| 18:43.26 | M07w | if all I have is an asterisk box and some softphones, is that enough to run an isolated phone network? |
| 18:43.29 | _brent_ | that won't generate traffic, just sniff it |
| 18:43.50 | jjshoe | M07w yes |
| 18:44.06 | jjshoe | M07w you could do it without asterisk if you did it by ip. *shrug* |
| 18:44.16 | *** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 18:44.54 | M07w | I don't trust these employees to understand IPs |
| 18:45.19 | M07w | also, the softphones I have in mind need an SIP manager |
| 18:45.31 | jjshoe | well then you have the answer to your question :) |
| 18:45.38 | M07w | great |
| 18:45.39 | jjshoe | heh i think skype even offers a pbx product |
| 18:45.46 | jjshoe | I can never remember the name of it though |
| 18:46.23 | M07w | would you|anybody happen to be familiar with the siemens optipoint 410? |
| 18:46.42 | jjshoe | not I |
| 18:46.51 | jjshoe | not even the slightest clue what it is. |
| 18:47.05 | M07w | a phone |
| 18:47.42 | jjshoe | http://www.prettymay.net/ is what I was thinking of, lol |
| 18:47.56 | [TK]D-Fender | M07w: General concenses on Seimens is "they're OK" |
| 18:48.59 | jplank | your saying I should be at about 55 onhook and jump to about 200 ringing? |
| 18:49.03 | jjshoe | jplank yes |
| 18:49.08 | M07w | we have about 50 of them, and I've had no luck getting them to work with the asterisk box we had before |
| 18:49.31 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 18:50.17 | jplank | I'm staying stead at 111 onhook and only going to 168 for ringing |
| 18:50.39 | jplank | Im testing a fax line, and a Linksys ATA I have to see those settings |
| 18:51.56 | jjshoe | what's a voltmeter say? |
| 18:52.16 | Ralcala | one final question guys its too differences between * 1.0 and 1.6 really so many that sip.conf and extensions.conf working with * 1.0 doesnt work with * 1.6?................. |
| 18:52.43 | jplank | thats what its saying |
| 18:52.57 | jplank | 16 offhook, 168 ringing, 111 onhook |
| 18:53.17 | jjshoe | jplank call the telco |
| 18:53.24 | jjshoe | jplank look at tip and ring in the asterisk book. |
| 18:53.29 | jjshoe | ~book |
| 18:53.29 | 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 |
| 18:53.36 | jjshoe | page 48 in my edition |
| 18:54.12 | jplank | what edition do you have? |
| 18:54.38 | jjshoe | no idea |
| 18:54.44 | jjshoe | like I said, look up tip and ring in the book |
| 18:54.49 | jjshoe | I'm sure the online version has an index |
| 18:54.58 | jplank | grrr doesn't matter I let someone borrow the book, and they left it at home |
| 18:55.07 | jplank | I'm looking at the online version |
| 18:55.27 | Ralcala | and the answer is?....... |
| 18:56.37 | [TK]D-Fender | Ralcala: a LOT has changed, its been what, 5 years? |
| 18:57.06 | Ralcala | now i see thanks TK |
| 18:57.07 | [TK]D-Fender | Ralcala: disable firewall. enable SIP debug. Look for packets. no packets = networking error and nothing to do with *. |
| 18:57.28 | [TK]D-Fender | Ralcala: Got packets? |
| 18:57.34 | *** join/#asterisk luca`gervasi (n=ashura@87.18.169.248) |
| 18:57.39 | luca`gervasi | Hallooo |
| 18:58.00 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 18:58.35 | Ralcala | FW= 3 nics with packet foward * = 1 nic |
| 18:58.58 | Ralcala | if I disable the FW i cant fw packet from localnet to * server.... |
| 18:59.20 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 18:59.29 | luca`gervasi | i read some kind of asterisk hardening guide. it was told to put "allowguest=no" on asterisk.conf, but when i put it on the config, my phone can no longer make calls (asterisk says: " Failed to authenticate on INVITE to"). what do you suggest? |
| 18:59.58 | [TK]D-Fender | Ralcala: Well you've tied your own hands. Nothing we can do for you. |
| 19:00.27 | [TK]D-Fender | luca`gervasi: Fix your phone's auth |
| 19:00.59 | luca`gervasi | uhm... it should be on... |
| 19:01.28 | Ralcala | thanks a lot anyway... |
| 19:01.30 | Ralcala | bye |
| 19:02.40 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:02.53 | Qwell | luca`gervasi: You don't put it in asterisk.conf |
| 19:03.27 | luca`gervasi | Qwell, well... actually i don't... but i'm a newbie, i think i should follow some basic hardening guide |
| 19:05.55 | seb- | Does asterisk use some sort of database to store client IP addresses? (I noticed it can remember them ever AFTER a power cycle) |
| 19:06.01 | mazpe | grrrr |
| 19:06.08 | Qwell | seb-: yes, astdb |
| 19:06.28 | seb- | Qwell: how wipe that DB clean? |
| 19:06.44 | seb- | Qwell: e.g. if you were testing and screwing around and added junk to it? |
| 19:06.47 | Qwell | seb-: help db |
| 19:06.56 | Qwell | (from the Asterisk CLI) |
| 19:07.27 | seb- | Qwell: thanky |
| 19:09.06 | Ritzerisk | could you use asterisk as a proxy server to convert nat to a public address |
| 19:09.51 | [TK]D-Fender | Ritzerisk: No. |
| 19:09.58 | [TK]D-Fender | Ritzerisk: * is not a proxy. |
| 19:10.57 | _brent_ | Ritzerisk: are your phones nat'ed or is asterisk? |
| 19:11.28 | Ritzerisk | well i need to look for a proxy software like a sip aware firewall for the sip trunk |
| 19:11.49 | _brent_ | opensips.org |
| 19:11.56 | [TK]D-Fender | Ritzerisk: ... "for the sip trunk"? WTF? |
| 19:11.56 | _brent_ | not a firewall, but a good sip proxy |
| 19:12.30 | Ritzerisk | i messed with teksips but gave me no luck |
| 19:12.35 | Ritzerisk | kind of like an ingate |
| 19:20.36 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:21.30 | bmoraca | Ritzerisk: it might help if you told us exactly what you're trying to do...you're throwing around a lot of buzz words and what not that don't really say anything |
| 19:25.41 | *** join/#asterisk Ralcala (n=demo@201.127.95.200) |
| 19:25.47 | Ralcala | hi again |
| 19:25.50 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:26.02 | Ralcala | i got news for you |
| 19:26.46 | Ralcala | i got packets with tcpdump....the * server is listening.... |
| 19:26.54 | Ralcala | now what? |
| 19:27.24 | _brent_ | Ralcala: check that asterisk is configured to listen on that interface or all interfaces |
| 19:27.35 | Ralcala | 0.0.0.0 |
| 19:28.09 | _brent_ | do tcp dump again, with host {remote IP} instead of port 5060 |
| 19:28.23 | _brent_ | see if anything is getting sent back to that remote host |
| 19:28.37 | _brent_ | i.e., ICMP messages saying destination unreachable |
| 19:28.42 | [TK]D-Fender | Ralcala: Enable SIP debug at * CLI and watch for a connection attempt |
| 19:28.43 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:29.09 | _brent_ | if you see dest unreachable going back to the remote host, adjust iptables |
| 19:29.38 | bmoraca | lol...Sheffield released by the Tigers...lol |
| 19:29.59 | Ralcala | mmmm k but brent iam working on lan i am not working on wan, yet... |
| 19:30.15 | _brent_ | shouldn't make a difference |
| 19:30.28 | Ralcala | k let me check.... |
| 19:32.55 | Ralcala | brent can i paste the packet on pm? |
| 19:33.04 | _brent_ | sure |
| 19:33.09 | Ralcala | thanks.. |
| 19:33.31 | [TK]D-Fender | Ralcala: PASTEBIN |
| 19:33.33 | [TK]D-Fender | ~pb |
| 19:33.34 | 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://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:33.55 | [TK]D-Fender | Ralcala: and inslcude the entire conversation, from beginning to end |
| 19:41.18 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 19:42.34 | *** join/#asterisk spenzenpoek (n=pimuri@85.4.90.96) |
| 19:43.55 | spenzenpoek | hi folks. I have a question about callfiles/callerids ... (maybe I didn't think this through. ) ... i try to make automate outbound calls using the provider voipgate.com ... the problem is... the calls are always anonymous .. no number is shown. I set the CallerId in the callfile and activated it in voipgate webinterface. |
| 19:44.06 | spenzenpoek | does anybody in here have an idea what I could check for? |
| 19:51.11 | *** join/#asterisk dieguito84 (n=diego@host5-197-dynamic.12-79-r.retail.telecomitalia.it) |
| 19:51.56 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 19:52.31 | spenzenpoek | it doesn't seem to matter how I write callerid: in the callfile .. didn't work with CallerId: bla<1234....> nor with CallerID: bla<1234....> |
| 19:53.20 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:59.32 | *** join/#asterisk DavidR2008 (n=chatzill@c-68-34-116-180.hsd1.md.comcast.net) |
| 20:00.38 | DavidR2008 | ok, I've done some packet tracing and I see a weird problem: the SIP packets have the correct IP address, but the RTP packets don't. what should I check? |
| 20:05.36 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 20:17.06 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 20:17.10 | _pepo_ | hi friends |
| 20:18.30 | _pepo_ | how do I test my asterisk-voicemail? I want to send automatically audio (with a lot of calls) so it will be stored. |
| 20:18.56 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 20:19.15 | [TK]D-Fender | _pepo_Send audio? Huh? "so it will be stored? DITTO. Please completely rephrase that |
| 20:19.48 | jjshoe | _pepo_ you'd have to write something to call in and play audio to it |
| 20:19.54 | jjshoe | _pepo_ I don't know of anything that will do this for you |
| 20:20.19 | _pepo_ | I was trying with sipp but I receive error messages about codecs |
| 20:20.29 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:21.35 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:22.44 | ghenry | Hi all |
| 20:23.25 | ghenry | If I want to inform a group of users that a voicemail has been left on a central vm number, can I just send out a MWI ? |
| 20:23.43 | ghenry | I want to subscribe serveral extensions to one voicemail box |
| 20:23.45 | ghenry | Is that easy? |
| 20:23.59 | _brent_ | yeah, you can put the same mailbox in the def for each of them |
| 20:24.01 | ghenry | I could use the IMAP backend for this |
| 20:24.18 | ghenry | They still need their own voicemail though |
| 20:24.27 | ghenry | It's for the "office voicemail" number ;-) |
| 20:24.42 | _brent_ | the same extension won't be able to monitor two vm boxes (that i know of) |
| 20:25.20 | ghenry | Maybe IMAP is better then |
| 20:25.29 | ghenry | and a group e-mail to say one is left |
| 20:25.41 | ghenry | or can I craft the MWI sip header? |
| 20:25.44 | _brent_ | does anybody know of a way to make a single extension monitor multiple vm boxes? that could be useful... |
| 20:27.37 | [TK]D-Fender | ghenry: Just add that box to their device entry |
| 20:27.45 | [TK]D-Fender | ghenry: No need of IMAP etc |
| 20:28.00 | ghenry | So they can have two? |
| 20:28.05 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 20:28.27 | [TK]D-Fender | ghenry: Yes |
| 20:28.37 | ghenry | OK, cool. |
| 20:28.48 | [TK]D-Fender | checkout time, BBL |
| 20:29.43 | ghenry | _brent_: in sip.conf sample there is: ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator |
| 20:29.55 | ghenry | I note the "Mailbox(-es)" ;-) Cool! |
| 20:30.08 | _brent_ | awesome, thanks [TK]D-Fender |
| 20:30.38 | ghenry | That can be handy then, plus a button programmed for that mailbox on the ip phone "Office Voicemail" |
| 20:31.06 | ghenry | They want a reception ring group and a ring group voicemail with MWI |
| 20:31.09 | ghenry | easy with *! |
| 20:31.28 | bmoraca | hrm...that's quite handy |
| 20:33.53 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:33.58 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:34.17 | _brent_ | and thanks ghenry, too :-) |
| 20:34.24 | ghenry | heh |
| 20:34.34 | sah-work | hello all. if i am doing a new system from the start what is the recommend linux distro? |
| 20:34.34 | *** join/#asterisk Steve_J-obs (n=Chris123@pool-70-19-173-91.bos.east.verizon.net) |
| 20:34.50 | Steve_J-obs | hello everybody!!!! |
| 20:34.56 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 20:35.32 | *** join/#asterisk __jeff_o__ (n=silas@ool-18b9f833.dyn.optonline.net) |
| 20:35.39 | Steve_J-obs | Free Apple Stocks to everybody, till I count to 3 |
| 20:35.42 | Steve_J-obs | 1 |
| 20:35.48 | Steve_J-obs | 2 |
| 20:35.49 | Qwell | Steve_J-obs: enough |
| 20:35.49 | Steve_J-obs | 3 |
| 20:37.04 | Steve_J-obs | Qwell: you dont like Apple stocks? |
| 20:37.47 | __jeff_o__ | hey, I think there's an option for this but I'm having trouble finding it. Here's what I need to do: I have an external asterisk server and I'm connecting to it from my home pc. I'm calling another computer which the asterisk server also connects to through the internet. I need a way, after that connection is made, to send rtp traffic directly, cutting out the asterisk server |
| 20:38.33 | __jeff_o__ | canreinvite? |
| 20:39.39 | *** join/#asterisk DavidR2008 (n=chatzill@c-68-34-116-180.hsd1.md.comcast.net) |
| 20:40.06 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 20:40.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:41.22 | bmoraca | that only works when no dial options are used and typically only when the endpoints are on the same LAN |
| 20:41.37 | bmoraca | if they're not on the same LAN, there could be problems, and i doubt it would work. |
| 20:41.40 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 20:43.09 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 20:43.53 | *** part/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 20:49.06 | __jeff_o__ | bmoraca: oh |
| 20:50.45 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 20:51.27 | cesar_CR | guys in a registe statement is ok to have a '+' character like: +659999999 |
| 20:51.32 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 20:52.10 | *** part/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 20:54.33 | *** part/#asterisk _brent_ (n=_brent_@jive.fttp.xmission.com) |
| 20:56.41 | *** join/#asterisk freh (n=freh@kotnet-160.kulnet.kuleuven.be) |
| 20:57.23 | DavidR2008 | I got disconnected earlier so I don't know if anyone answered my question or not. I have a softphone that can't hear when it get's called. it's across a VPN, but there is no firewall rules blocking anything. I packet sniffed the connection and the * to phone SIP packets have the phone address, but the * to phone RTP packets have the address of the VPN server. any idea why? |
| 20:58.34 | freh | DavidR2008, are you using NAT? |
| 20:58.43 | DavidR2008 | no |
| 20:58.54 | DavidR2008 | just routing |
| 20:59.37 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 20:59.45 | freh | Sorry, don't know then |
| 20:59.53 | *** part/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 21:00.44 | freh | Can the softphone make a call? |
| 21:00.59 | DavidR2008 | I was hoping [TK]D would know, but it looks like he signed off already |
| 21:01.55 | knarfly | was the wcfxo.ko module removed from the latest zaptel release? |
| 21:02.20 | freh | DavidR2008, Can you make a call with the softphone? |
| 21:03.06 | defsdoor | DavidR2008: packet sniff both ends |
| 21:03.24 | defsdoor | DavidR2008: looks like your router is doing some packet mangling |
| 21:03.48 | DavidR2008 | yep, that works just fine. if the softphone initiates the call all is well, if it doesn't the RTP packets to the softphone don't get the correct address |
| 21:04.00 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 21:04.16 | defsdoor | so one of your vpn routers is mucking about with outgoing connections |
| 21:04.27 | defsdoor | and masquerading them when it shouldnt |
| 21:05.02 | DavidR2008 | I sniffed from the * server and all packets from the softphone have the correct address |
| 21:05.34 | DavidR2008 | outbound SIP packets have the correct address, outbound RTP packets have the wrong address |
| 21:05.52 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:07.16 | defsdoor | DavidR2008: if you are routing only then * knows nothing of your vpn addresses |
| 21:07.28 | defsdoor | DavidR2008: and your vpn should be transparent |
| 21:08.13 | defsdoor | DavidR2008: so still suggests your VPN is mangling your packets |
| 21:08.24 | *** part/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 21:08.36 | keith4 | I see no output on the CLI, even when I set verbosity to 10. wtf? |
| 21:08.53 | DavidR2008 | that's what I would expect: x.x.0.* is the lan, x.x.5.* is the vpn. x.x.0.211 is the router. |
| 21:09.14 | *** join/#asterisk telecos (n=sergio@70.167.219.87.dynamic.jazztel.es) |
| 21:09.39 | DavidR2008 | all packets will be addressed to x.x.5.34 except the RTP packets which get sent to x.x.0.211 |
| 21:10.35 | defsdoor | are they SYN or replies ? |
| 21:10.45 | freh | DavidR2008, a SIP show peers gives the correct address of the softphone? It should be correct because the SIP packets have the right address, but just to be sure |
| 21:11.00 | DavidR2008 | yes it does |
| 21:11.15 | defsdoor | what is your VPN ? |
| 21:11.40 | defsdoor | (I route only sip over openvpn fine) |
| 21:12.00 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 21:12.06 | DavidR2008 | openvpn |
| 21:12.38 | defsdoor | you have some firewall rules on your vpn then that is natting when it should be |
| 21:12.50 | defsdoor | shouldnt* |
| 21:12.57 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 21:13.35 | freh | I'm not sure about that since the wrong address is used before the packet arrives at the router |
| 21:14.13 | defsdoor | freh: I'm not sure it is - hence asking if he was looking at replies |
| 21:14.51 | defsdoor | * doesnt know anything about his 0.211 address so it can't be initiating the use of it |
| 21:15.24 | defsdoor | so my guess is the rtp connection from the remote end is being SNAT'd on the VPN fw |
| 21:16.04 | defsdoor | simplest way to test would be to flush fw and set default to allow on all chains |
| 21:16.28 | freh | sounds very likely |
| 21:16.39 | defsdoor | DavidR2008: is the vpn also an internet gateway for other stuff ? |
| 21:16.42 | `paul | i put an entry to asterisk1 sip.conf how come i still recieve a "Failed to authenticate on INVITE" on asterisk2 when i do a DIal(sip/1234@ipaddressofasterisk1); ? |
| 21:17.09 | freh | are RTP connections initiated bij the sip clients? |
| 21:17.16 | freh | by* |
| 21:18.23 | DavidR2008 | sorry, I got behind here |
| 21:19.02 | Ritzerisk | so i need like a sip aware firewall or a local sip proxy server .... im not sure on how to setup opensips |
| 21:19.06 | kink0 | one question about SS7 config in Asterisk, is possible to have a separate bearer trunk ? i.e. voice channels on one Cisco, and signalling T1 on the asterisk box ? |
| 21:19.11 | *** join/#asterisk path_ (n=path_@190.196.17.106) |
| 21:19.35 | DavidR2008 | the vpn server is a dedicated linux box running openvpn, it has no firewall rules and accept on all chains |
| 21:21.19 | DavidR2008 | the RTP packets are initiated from * There will be an RTP packet from the hardphone to * then from * to the softphone (but to the wrong address) |
| 21:21.32 | defsdoor | DavidR2008: you've done iptables -L -t nat ; iptables -L -t mangle to make sure ? |
| 21:21.37 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 21:21.52 | DavidR2008 | ok I forgot to check mangle, I'll be right back |
| 21:22.19 | freh | is the VPN server the default gateway for your asterisk server? |
| 21:24.43 | DavidR2008 | ah that could be the problem! no it's not, the firewall is the default gateway and it has a static route for the VPN through the VPN server |
| 21:25.09 | defsdoor | the firewall has the route or the * box ? |
| 21:25.32 | DavidR2008 | the firewall, which means it could be mangling the packets |
| 21:25.51 | defsdoor | add the route on the * box |
| 21:25.55 | defsdoor | skip the firewall |
| 21:26.10 | DavidR2008 | ok, hang on |
| 21:26.14 | defsdoor | if the vpn on is same subnet anyway |
| 21:26.17 | `paul | how do i allow all calls coming from a specific IP address |
| 21:26.20 | DavidR2008 | it is |
| 21:26.46 | defsdoor | skip the firewall then for deffo - no point running the same packets across the network twice |
| 21:27.40 | defsdoor | I use my nokia at work over openvpn to my home box so I can get home calls at work etc.. |
| 21:28.00 | defsdoor | works flawlessly |
| 21:30.19 | freh | quick question, I have voip/sip router at home and I'd like to install asterisk and make it authenticate as a "softphone" to the router so I can use it as a gateway over vpn connections |
| 21:30.21 | freh | can it be done? |
| 21:32.44 | DavidR2008 | freh: it should be possible. I think there is a register statement that can go in the sip.conf file to cause asterisk to register with a sip provider |
| 21:33.19 | DavidR2008 | I'm done for today, I'll pick this back up in the next day or so |
| 21:33.29 | defsdoor | freh: is that any different to using an SPA? |
| 21:33.32 | DavidR2008 | thanks everyone for the help! |
| 21:35.10 | freh | Maybe a stupid question, but what's a SPA? |
| 21:35.26 | defsdoor | ATA - linksys spa3102 thing |
| 21:35.47 | defsdoor | I used to use one here before I got a sangoma fxo/fxs |
| 21:36.26 | *** join/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 21:36.38 | NMR_1122 | Hi, |
| 21:36.51 | NMR_1122 | I'm having a problem getting voicemail users to show up |
| 21:37.55 | NMR_1122 | I've got three mailboxes defined in voicemail.conf, but "voicemail show users" only shows context default, mailbox general |
| 21:37.55 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:38.53 | Ritzerisk | basically i have a NAT Traversal Issue that im Trying To Fix |
| 21:39.44 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 21:40.44 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:40.45 | freh | I basically want to use a softphone on my laptop so I can receive calls from home. I don't want to use analog phones or buy any additional hardware |
| 21:41.34 | defsdoor | you have * at home ? |
| 21:41.51 | freh | yup :p |
| 21:42.06 | defsdoor | with PSTN ? |
| 21:42.09 | DavidR2008 | NMR_1122: have you reloaded * since you modified your voicemail.conf file? |
| 21:42.18 | freh | I have a voip provider |
| 21:42.35 | freh | over ADSL actually |
| 21:42.42 | defsdoor | so just connect softphone to * |
| 21:42.55 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:43.04 | freh | indeed |
| 21:43.22 | *** join/#asterisk jstoker (n=jstoker@88-107-251-135.dynamic.dsl.as9105.com) |
| 21:43.47 | NMR_1122 | yes, in fact i did a restart now with no effect, and then in desperation a reboot of the entire computer |
| 21:44.00 | jblack | Yes, because linux is just like windows. |
| 21:44.13 | NMR_1122 | lol |
| 21:44.16 | *** join/#asterisk jstoker (n=jstoker@88-107-251-135.dynamic.dsl.as9105.com) |
| 21:48.46 | DavidR2008 | do you mind sharing your voicemail.conf? munge the private information if any |
| 21:49.15 | *** join/#asterisk Ralcala (n=demo@201.127.95.200) |
| 21:49.17 | NMR_1122 | ok, hold on |
| 21:49.54 | Ralcala | and where is brent? |
| 21:50.14 | Ralcala | anybody here know where he goes? |
| 21:50.52 | Ralcala | ok |
| 21:51.32 | Ralcala | here is the question ICMP myserver.com.mx unreachable - admin prohibited, length 56.... |
| 21:51.44 | NMR_1122 | This is it: |
| 21:51.47 | NMR_1122 | [General] |
| 21:51.47 | NMR_1122 | 1 => 0000,Some User,user@domain.com |
| 21:51.47 | NMR_1122 | 2 => 0000,Another User,user@domain.com |
| 21:51.47 | NMR_1122 | 3 => 0000,The President,thepres@domain.com |
| 21:51.47 | NMR_1122 | 4 => 0000,Customer Service,info@domain.com |
| 21:52.48 | defsdoor | that looks wrong |
| 21:53.11 | Ralcala | its a ipv6 or named or FW or XLite issue, please help iam on * server behind FW (Fedoras Core 8 an 9) |
| 21:53.41 | defsdoor | NMR_1122: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 21:53.55 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 21:55.26 | *** part/#asterisk jstoker (n=jstoker@88-107-251-135.dynamic.dsl.as9105.com) |
| 21:55.28 | NMR_1122 | That's the page I'm looking at.... Is a general and default context required? |
| 21:56.07 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 21:56.42 | [TK]D-Fender | NMR_1122: the HEADER is |
| 21:57.13 | NMR_1122 | what header? |
| 21:57.13 | [TK]D-Fender | NMR_1122: you need a [general] SECTION, and you don't need to use [default] but you do need some sort of section. |
| 21:57.27 | [TK]D-Fender | [general] , [default], etc |
| 21:58.01 | NMR_1122 | ok, so I can do [general] and [myboxes] and be ok without [default]? |
| 21:58.33 | [TK]D-Fender | NMR_1122: Should |
| 21:59.07 | NMR_1122 | do I need "cidinternalcontexts" under general, like the example? |
| 22:00.14 | [TK]D-Fender | NMR_1122: NMR_1122 You appear to to be nit-picking every single thing in the sample and your setup is broken. Stop trying to work your way back to normal one painful step after another |
| 22:00.42 | drmessano | Conficker is ALIVE |
| 22:00.50 | NMR_1122 | Isn't that how you learn? |
| 22:01.07 | jblack | already? |
| 22:01.27 | [TK]D-Fender | NMR_1122: No, thats how we get tired of 50 million questions you could have figured out by reading about them AFTER getting a basic working system |
| 22:01.29 | jblack | I tought we had a few more hours before skynet. |
| 22:02.03 | [TK]D-Fender | NMR_1122: Learn by tweaking some basic things that DO work. Don't waste everyones time asking how to build a car 1 friggen bolt at a time. |
| 22:03.10 | freh | :q |
| 22:03.27 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 22:04.23 | [TK]D-Fender | Off for a while, back later |
| 22:04.35 | jblack | Oh, man, he's one of those types? |
| 22:05.14 | jblack | "Durhh. I understand by doing myself. From scratch. Without reading books. Without looking at working examples. The true way to knowledge is via /dev/random" |
| 22:06.40 | Qwell | drmessano: source? |
| 22:08.29 | drmessano | It came from Harper Reed, so its 110% true. |
| 22:08.30 | drmessano | http://www.f-secure.com/weblog/archives/00001643.html |
| 22:09.03 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-19ac3ad881dcae90) |
| 22:13.52 | Steve_J-obs | anyone here can give me their opinion on how to send a call down a list of carriers using the dialplan??..I am doing one, but I dont think is the professional way to do it |
| 22:14.44 | defsdoor | has finished his mickey mouse endpoint manager! |
| 22:15.58 | drmessano | Does it have a mickey mouse pci? |
| 22:16.05 | drmessano | pic |
| 22:16.21 | defsdoor | nah - nothing fancy - just a ruby script |
| 22:18.19 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:18.19 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:18.29 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 22:19.09 | Steve_J-obs | I am dialing a number, and if {DIALSTATUS}=CONGESTED, and trying another route, problem is, I cant see this working for a list of several carriers, anyone has anythoughts on this? |
| 22:21.05 | defsdoor | cool- it works! |
| 22:21.16 | *** join/#asterisk LuisHenrique (n=LuisHenr@200-195-13-11.isimples.com.br) |
| 22:27.22 | *** join/#asterisk alerios (n=alerios@190.144.75.22) |
| 22:36.28 | *** join/#asterisk mandd (n=moo@67.204.24.118) |
| 22:41.46 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:53.58 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 22:54.23 | Dovid | Hi room |
| 22:54.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:54.53 | Dovid | i am looking over a wire shark trace. i am seeing a lot "UDP Checksum Incorrect". i am unable to figure out the "exact" meaning of it. can anyone help ? |
| 22:56.59 | carrar | http://tinyurl.com/d5uazz |
| 22:58.48 | jaytee | hehehe |
| 22:59.25 | drmessano | Its a TRAP.. UDP has NO CHECKSUMS |
| 22:59.43 | drmessano | The world is burning. |
| 22:59.45 | drmessano | Run. |
| 23:01.13 | freh | LOL |
| 23:01.40 | freh | never saw that "let me google that for you" before |
| 23:01.47 | jaytee | "Don't bother running.....you'll just die tired" |
| 23:02.10 | Dovid | freh: I tried google. to complicated to figure it out |
| 23:02.54 | X-Rob | if you're looking at wireshark, and you don't understand that.. uh. Why are you looking at wireshark? |
| 23:03.09 | carrar | haah |
| 23:03.49 | jaytee | if Google is too complicated for you than maybe computers aren't really your thing. Ever thought of macrame? becoming a professional bowler? |
| 23:04.10 | carrar | I did Macrame once |
| 23:04.16 | carrar | that can be hard too |
| 23:04.34 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 23:04.43 | Dovid | jaytee: Sorry. I meant to say that I was not able to understand that results |
| 23:04.46 | carrar | some complex multi level hangings you can make |
| 23:05.04 | carrar | David, then the answer is: It's normal |
| 23:05.18 | carrar | pending your machine setup |
| 23:05.19 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 23:05.28 | carrar | and wireshark hook into your NIC |
| 23:05.32 | Dovid | carrcar: I rately see it |
| 23:05.38 | drmessano | You just discovered the U in DP.. Bet you feel like less of a DA now? |
| 23:05.45 | Dovid | very much so ;) |
| 23:05.52 | drmessano | VFG |
| 23:05.57 | drmessano | YMMV |
| 23:05.59 | drmessano | HAND |
| 23:06.05 | drmessano | CYA |
| 23:06.14 | Steve_J-obs | can anybody here advice me on how to beat the "we already have a VAD frame" error, other than telling me to tell the other side to suppress VAD? |
| 23:06.19 | Dovid | carrar: I am asking because the customer is complaining that the RTP stream craps out in mid call. they just dont hear the other person and vice versa. thought it may have t do with it |
| 23:06.51 | carrar | Dovid, do you have a jitter buffer enabled? |
| 23:06.56 | Dovid | carrar: no |
| 23:07.13 | Dovid | i was thinking that it would it be a NAT issue |
| 23:07.17 | carrar | Whats the error you see on console? |
| 23:07.19 | Dovid | it randomly happens in calls. no set time etc. |
| 23:07.27 | Dovid | no errors in the console |
| 23:07.52 | carrar | is it replicatable? |
| 23:07.57 | drmessano | I wish I had heard of memes years ago when the local fire dept had just started going on calls to check on elderly folks who maybe had kids would couldnt reach them and worried they were in distress.. they called them R U OK calls |
| 23:08.03 | carrar | on the same LAN as your * box |
| 23:08.04 | drmessano | My reply: NO I R NOT |
| 23:08.26 | drmessano | I R NOT OK |
| 23:08.36 | drmessano | I R NEED HALP |
| 23:08.39 | carrar | Wed Apr 1 08:08:38 JST 2009 |
| 23:08.46 | carrar | conflickr hard at work in Japan |
| 23:08.48 | carrar | and moving this way! |
| 23:08.50 | carrar | WATCH OUT |
| 23:08.56 | X-Rob | it's 9am here |
| 23:08.59 | X-Rob | and still nuffin. |
| 23:09.04 | Dovid | carrar: any idea what it is doing |
| 23:09.04 | drmessano | OMMMMMMMMGASH |
| 23:09.04 | carrar | heh |
| 23:09.13 | drmessano | ITS LIKE WAITING FOR SANTA TO COME |
| 23:09.25 | drmessano | GO TO BED KIDS, HES OVER EUROPE NOW |
| 23:09.27 | carrar | Dovid, sounds like a internet connectivity issue |
| 23:09.33 | Dovid | carrar: Thanks |
| 23:09.40 | drmessano | I R XCITED |
| 23:10.03 | carrar | Dovid, put a phone on the same LAN as the pbx and see if it happens |
| 23:10.16 | carrar | LAN == Logical Network |
| 23:10.21 | drmessano | Wake me up when conficker is almost here.. I left out Linux CDs and sugar cookies for him |
| 23:10.22 | Dovid | carrar: no issues with anyone else, locally or remotely. |
| 23:10.23 | carrar | same subnet |
| 23:10.34 | carrar | yeah probably just the internet |
| 23:10.47 | carrar | It's conflickr's fault |
| 23:10.49 | drmessano | No, I CANT SLEEP... tooooo excited |
| 23:10.54 | carrar | let the user know they are infected ;) |
| 23:11.18 | drmessano | OH OHHHH |
| 23:11.23 | drmessano | INFECT ME, INFECT ME |
| 23:11.27 | drmessano | \o/ |
| 23:11.34 | drmessano | PLEEEEEEEEEASE |
| 23:11.43 | carrar | walks down to 1st ave to hook up drmessano |
| 23:11.48 | drmessano | hahaha |
| 23:11.58 | drmessano | Thats 5th and Broad in Augusta |
| 23:12.01 | carrar | She'll love you long time |
| 23:12.05 | carrar | haha |
| 23:12.26 | carrar | Here is Seattle 1st Ave seems to be in |
| 23:12.27 | carrar | it |
| 23:12.32 | jaytee | until you get them home or to the hotel and find out she's a he/she |
| 23:12.38 | carrar | hahah |
| 23:12.50 | carrar | Sorry to hear you having those issues jaytee |
| 23:12.51 | drmessano | God i hope so.. I paid for one! |
| 23:12.53 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 23:12.55 | carrar | blame conflickr |
| 23:13.14 | jaytee | it's confiker not conflickr |
| 23:13.22 | carrar | cnfklr |
| 23:13.29 | drmessano | Damn, my ram didnt get here today --------> conficker |
| 23:13.30 | X-Rob | jaytee, does that actually REALLY happen? I mean what does the tranny get out of it? OK, you look great as a chick, sure. But you have a penis, and the person who would have paid you money is now upset and NOT paying you money |
| 23:13.38 | carrar | same person who wrote that virus works for flickr |
| 23:13.44 | X-Rob | I kinda think you're either stupid, or they're stupid. |
| 23:13.46 | carrar | heh |
| 23:14.16 | jaytee | X-Rob, I think they're hope that all that rage will be released as anal rape which is probably what they're lookin for |
| 23:14.32 | drmessano | carrar: If youre implying that Yahoo is nothing but an old, useless virus and would be best served by being bought by google and blow to pieces in a drunked backyard barbecue, then yes, I am with you |
| 23:14.47 | drmessano | Down with Yahoo |
| 23:14.55 | jaytee | but then what will happen to babelfish? |
| 23:15.02 | X-Rob | jaytee, I think it's a myth. |
| 23:15.05 | drmessano | Google Translate |
| 23:15.22 | jaytee | X-Rob, could be, only one who knows for sure is Eddie Murphy |
| 23:15.30 | drmessano | Google has the cure for the Yahoo virus |
| 23:15.55 | drmessano | Yahoo IM = Lots of soccer moms, 419 scammers, and bots |
| 23:16.06 | drmessano | Google IM = No one, really |
| 23:16.12 | X-Rob | I use gtalk |
| 23:16.12 | drmessano | Problem solved |
| 23:16.18 | X-Rob | because no-one uses it |
| 23:16.19 | jaytee | did you read about Microsoft wanting 11 million in fed money to offset the cost of building a bridge between two sides of their campus? |
| 23:16.45 | drmessano | X-Rob: If the world would switch to XMPP, I would hate just a little less.. thank you for the effort |
| 23:17.40 | drmessano | I want to see the world united under XMPP.. I want to buy.. the world a Coke |
| 23:17.51 | drmessano | Down with Pepsi! |
| 23:20.02 | jaytee | I think the Mormon church still has a controlling share of Pepsi |
| 23:20.23 | jaytee | me, I don't drink it just cuz it sucks compared to Coke. |
| 23:20.55 | X-Rob | oooh |
| 23:21.02 | X-Rob | I think I've got someone with conficker |
| 23:21.52 | Qwell | X-Rob: congrats! |
| 23:22.29 | Qwell | so...what does it do? |
| 23:23.41 | drmessano | Kill it !!! |
| 23:23.42 | drmessano | Kill it !!! |
| 23:23.47 | X-Rob | Nope |
| 23:23.49 | X-Rob | it wasn't |
| 23:23.56 | Qwell | sorry for your loss |
| 23:23.58 | drmessano | Just Yahoo toolbar? |
| 23:24.00 | Dovid | jaytee: Let me know if its passed midnight where u r at so i know to ignore those comments ;) |
| 23:24.03 | drmessano | Sounds about right |
| 23:24.07 | X-Rob | They called and said that theyd' got a message saying virtual memory low and that they had to reboot |
| 23:24.13 | drmessano | I can understand the resemblence |
| 23:24.15 | denon | eh, the whole thing with conficker is that it doesn't do anything exciting/known until the creators upload a payload :) |
| 23:24.15 | *** join/#asterisk jstoker (n=jstoker@88-107-157-111.dynamic.dsl.as9105.com) |
| 23:24.30 | X-Rob | reboot? That's not right. Anyway, I got them to try to go to microsoft.com and it worked, so nup, not conficker. |
| 23:25.18 | drmessano | conficker is like Debian.. no one really knows what the hell it does, but its all over the damn place and may or may not destroy the world |
| 23:25.47 | denon | and it's Free As In Beer(tm) |
| 23:26.16 | jaytee | heheeheee |
| 23:26.25 | jaytee | debian |
| 23:26.34 | Dovid | http://news.zdnet.co.uk/security/0,1000000189,39599480,00.htm |
| 23:26.41 | drmessano | I installed debian on an old box here once |
| 23:26.49 | jaytee | why don't they just rename to ian since he and deb divorced like 4 years ago |
| 23:26.56 | drmessano | I unplugged it.. and it appeared to be generating its own power |
| 23:27.15 | denon | I dont get how so many people may get conficker ... it's exploit has been patched since late 2008 |
| 23:27.22 | denon | don't these people ever do any maintenance at all? |
| 23:27.38 | drmessano | I installed Ubuntu on the same box, unplugged it, and it cried for days |
| 23:28.01 | Dovid | dennon: Lots of dummies out there |
| 23:28.29 | Qwell | jaytee: and he's not with the project anymore.. he's with Sun now, IIRC |
| 23:28.59 | jaytee | Qwell, yup he is. Working on the OS version of Solaris, code named Project Indiana |
| 23:29.26 | jaytee | which is funny since he lived about 10 miles from here in Fishers, IN |
| 23:29.29 | drmessano | I hate Sun.. they bought MySQL and turned it into.... Well, they made it much...... Yeah, well, before Sun came along it was.... Ok, nm.. |
| 23:29.39 | jaytee | hahahahaaaaaaaaa |
| 23:30.30 | X-Rob | drmessano, oddly enough, I think that sun have blown their money on MySQL. It's forked so many times, and a lot of the forks are significantly better. |
| 23:30.58 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 23:30.58 | X-Rob | Drizzle apparently is the winner. |
| 23:31.59 | drmessano | X-Rob: Exactly.. The should have just pulled a Fonality, forked it, and rubbed out the first half of the name.. FreePBX >>> PBX, MySQL >>> SuckyDB |
| 23:32.05 | drmessano | Wait, thats not right |
| 23:32.50 | X-Rob | Heh |
| 23:33.12 | X-Rob | I don't mind if people fork FreePBX - I encouraged it. But I -do- mind if they don't abide by the GPL. Then I get my crank on. |
| 23:33.30 | *** part/#asterisk Mark21 (n=mark@freenode/sponsor/mark17) |
| 23:34.17 | drmessano | Seeing all Fonality has done for Asterisk development, I welcome them forking FreePBX, MySQL, Apache or whatever |
| 23:34.31 | drmessano | Their contribution to Asterisk has been huge and very welcome |
| 23:34.46 | X-Rob | Yeah. Well. Fonality is probably a _bad_ example. |
| 23:34.56 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 23:35.51 | drmessano | Everytime I make a call using a LAMPFA stack PBX, I can just feel the Fonality coursing through the dialplan |
| 23:37.46 | drmessano | How about that one time they helped patch.. or when they optimized.. oh, and then they did all that work on.. oh, but who can forget when they... |
| 23:38.03 | drmessano | Sun, buy Fonality |
| 23:38.13 | drmessano | ------> $1 |
| 23:38.17 | drmessano | I will halp |
| 23:38.41 | X-Rob | $1? That's a bit overpriced, isn't it? |
| 23:38.55 | drmessano | I EXPECT CHANGE |
| 23:38.58 | X-Rob | Ah. |
| 23:39.09 | drmessano | Hey! Where are you going with that??? |
| 23:39.27 | X-Rob | Um. Just over ... here? |
| 23:39.42 | X-Rob | I need to write a cheque. I keep it in the glovebox of the car |
| 23:40.02 | X-Rob | So I'll just go over to the car, and write that cheque, and be rigth back, OK? |
| 23:40.09 | drmessano | If I had a kid.. and he gave someone a whole dollar for a Fonality, I would ground him for a month and give a stern lecture on economics |
| 23:40.34 | X-Rob | Maybe for 3 or 4 fonalities. |
| 23:40.50 | X-Rob | You could maybe go through their offices looking for cookies. |
| 23:41.03 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 23:41.05 | drmessano | Im sure Kerry left cases of cookie crumbs in his desk |
| 23:41.12 | drmessano | They've probably mutated |
| 23:41.49 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 23:41.57 | drmessano | Wait, Kerry wouldnt have left them |
| 23:42.06 | drmessano | GPLLaw maybe |
| 23:42.20 | drmessano | Cause' that guy, let me tell ya |
| 23:43.10 | jaytee | and it's only been a few short months but everyone has forgotten Engineer Tim |
| 23:43.14 | drmessano | GPLLaw may have the same IP address traceable to a home internet connection at a house that looks a lot like Kerrys, with Kerrys car in the driveway, and a bunch of cameras all over the place, but man.. he is DEFINITELY.. NOT.. KERRY |
| 23:43.29 | X-Rob | Engineer Tim was amusing. I like they way he quit. |
| 23:43.43 | drmessano | My ribbon is still up |
| 23:43.46 | X-Rob | I sent him an email,but he never responded 8-( |
| 23:45.20 | X-Rob | drmessano, actually, it was a random h4xx0r who h4xx04ed a random un-secured wifi connection that JUST HAPPENED to be Kerry's. Pure fluke, one in a million chance. |
| 23:46.36 | drmessano | I hope Kerry has moved on to better things.. like working behind the counter at the local 7-11, telling the weird, crazy nighttime customers that the hot dogs really are all beef, even if they've got chicken feathers poking through the skin. |
| 23:46.40 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 23:47.05 | *** part/#asterisk alerios (n=alerios@190.144.75.22) |
| 23:47.10 | drmessano | WE ARE NOT SPYING ON YOU.. NOTHING TO SEE HERE |
| 23:47.10 | jaytee | lol |
| 23:47.29 | drmessano | Oh, you better get that |
| 23:47.32 | drmessano | Get what? |
| 23:47.34 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 23:47.35 | drmessano | Umm.. nm |
| 23:47.48 | root52 | Am I right in saying that if I have two endpoints connected to the same asterisk box I need only buy on G.729 lic? |
| 23:47.59 | root52 | *one |
| 23:48.41 | X-Rob | root52, Um. That depends if the two endpoints are both using g729? |
| 23:49.18 | drmessano | X-Rob: I love how Philippe called him out on it.. While my ego often doesnt allow me to give someone great kudos for coming up with something funnier than I did, that was hilarious. |
| 23:49.28 | X-Rob | Heheh |
| 23:49.38 | root52 | ahhh I see If I want to provide the transcoding in asterisk then I would need to buy two |
| 23:50.09 | drmessano | <Kerry> Um, but umm.. I umm... I gotta go.. .. I have a umm..... food.. in the oven. |
| 23:50.17 | jaytee | OMG! Firefox just grey screened and crashed!!! It must be the linux port of confiker!!! ...... oh, nm. it's just Firefox |
| 23:50.25 | drmessano | HAHAHHA |
| 23:50.49 | drmessano | I should start a Trixbox/conficker threat on trixbox.org |
| 23:51.01 | X-Rob | My wife has just gone to the place that called (she's an accountant, and she does their books, too) and turns out it's just stupid userism. |
| 23:51.13 | X-Rob | How amazing. |
| 23:51.28 | drmessano | "conficker may or may be conficker, and that may or may not be a rumor and/or untrue, which in itself may or may not be provable" |
| 23:51.41 | drmessano | Errrr |
| 23:51.50 | drmessano | "trixbox may or may be conficker, and that may or may not be a rumor and/or untrue, which in itself may or may not be provable" |
| 23:52.01 | drmessano | "discuss" |
| 23:52.01 | X-Rob | OK. That makes more sense 8) |
| 23:52.16 | drmessano | Yes yes.. Brain <> Keyboard problem |
| 23:52.22 | X-Rob | <GPLLaw> Trixbox is NOT conficker, because conficker doesn't abide by the rules of the GPL |
| 23:52.33 | drmessano | !!!! HAHAH |
| 23:52.35 | X-Rob | <KerryG> Yeah, what GPLLaw said |
| 23:53.14 | drmessano | <KerryG> FreePBX, get in mah BELLY! |
| 23:53.20 | X-Rob | <andrew> you guys are stealing my design of how to make a centos bootable cd! |
| 23:53.27 | *** join/#asterisk jstoker (n=jstoker@88-107-177-38.dynamic.dsl.as9105.com) |
| 23:53.38 | drmessano | ahah |
| 23:53.51 | drmessano | <andrew> Stop stuffing me in lockers... its old |
| 23:54.12 | X-Rob | hah |
| 23:54.25 | X-Rob | Mmm, I love ssh port forwarding |
| 23:54.34 | X-Rob | so damn handy to get into supposedly secure networks. |
| 23:55.26 | drmessano | I remember when I trolled Kerry on the VoIP users conference about their supposed 4 employees that support Trixbox.. like Chris Lyman has them all in some locked sandbox |
| 23:56.05 | jaytee | wish I could hack the Trixbox website and have it stream the mp3 of "It's not easy being green" by the Muppets |
| 23:56.21 | drmessano | Corner office at the Fonality bunker in Berlin, with its own door so they dont have to walk through the lobby and be seen by the real Fonality employees |
| 23:56.45 | X-Rob | heh |
| 23:56.55 | drmessano | Live, eat, and die by the contributions from vendors and donations to keep it alive |
| 23:57.06 | drmessano | Riiiiiiight |
| 23:57.08 | X-Rob | Well. There was andrew, kerryg, engee tim, and ??? |
| 23:57.53 | X-Rob | Hmm. What twonk didn't create a GUI to bulk add speed dials in freepbx? |
| 23:57.58 | X-Rob | hey? Who's fault is THAT, I ask? |
| 23:58.00 | drmessano | I can see the magazine cover with a certain AsteriskNOW dev holding an AsteriskNOW CD in his teeth with a gun to his head.. "DONATE TO THIS PROJECT OR WE'LL SHOOT THIS DEVELOPER" |
| 23:58.03 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 23:58.03 | X-Rob | oh, hang on. |
| 23:58.16 | X-Rob | drmessano, *laugh* |
| 23:58.45 | *** part/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 23:58.46 | root52 | Ok sorry guys I am a bit confused. I am looking to realize the bandwidth savings of the g.729 codec. I have one endpoint in a far off county (x-lite) that only has access to 100kbs of symmetrical bandwidth. The other endpoint (cisco 7940g) is located in the states that has access to all kinds of bandwidth. I want to make sure that the x-lite endpoint will only use the 37 or so kbs that is offered by the g.729 compression. |
| 23:58.51 | drmessano | Thats pretty much the image I got from Kerry.. What a wanker |
| 23:58.54 | X-Rob | oh, hang on. I _did_ create a gui. Don't mind me. |
| 23:58.58 | drmessano | LOL |
| 23:59.23 | X-Rob | root52, google 'free g729 codec' |