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| 00:55.00 | justdave_ | is there a way to have a dundi server return two responses for the same lookup, with different weights? |
| 00:55.33 | justdave_ | I tried putting the mapping line twice, with the higher-weighted backup route for the second one, but it seems to only use the last one defined |
| 00:59.34 | Linuturk | justdave: you have to take it out to dinner tonight |
| 00:59.42 | Linuturk | dinner first* |
| 01:05.29 | justdave | heh |
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| 01:12.02 | De_Mon | how do I reload extensions.conf in 1.6.0? |
| 01:12.32 | justdave | "extensions reload" is what 1.4 uses. 1.6 doesn't like that? |
| 01:12.41 | justdave | hasn't looked at 1.6 yet |
| 01:15.05 | WimpMan | No. That was 1.2 |
| 01:15.06 | Qwell | What happened when you typed it in 1.4? |
| 01:15.15 | Qwell | (tip: it told you what to use) |
| 01:16.13 | De_Mon | if it did it's not any more |
| 01:16.35 | De_Mon | osprey*CLI> extensions reload |
| 01:16.35 | De_Mon | No such command 'extensions reload' (type 'help extensions reload' for other possible commands) |
| 01:16.54 | De_Mon | am I misspelling something? not that it matters if i'm supposed to be using something else anyways |
| 01:17.13 | Qwell | It is gone, just like 1.4 told you it would be. |
| 01:17.22 | De_Mon | i'm a slow learner |
| 01:17.27 | Qwell | and UPGRADE.txt |
| 01:17.28 | De_Mon | bad speller too |
| 01:17.35 | seanbright | dialplan reload |
| 01:17.36 | Qwell | and hundreds of emails on various lists |
| 01:18.19 | jaytee | try dialplan reload |
| 01:18.32 | seanbright | right, like i just said. |
| 01:18.36 | jaytee | ooops, missed seanbright's post |
| 01:18.41 | seanbright | heh |
| 01:18.44 | Qwell | no offense, but if people aren't going to bother to read the deprecation notices and documentation that we write... |
| 01:18.56 | seanbright | we should make them feel bad about it and not help them |
| 01:18.57 | De_Mon | we read them, just don't *remember* them |
| 01:18.57 | seanbright | i agree |
| 01:19.12 | seanbright | the user does indeed come 2nd, after all. |
| 01:19.15 | seanbright | heh |
| 01:19.19 | Qwell | 3rd at best |
| 01:19.21 | De_Mon | i'm looking at the upgrade doc again |
| 01:19.22 | Qwell | :p |
| 01:19.31 | Qwell | De_Mon: You could have started using the new command immediately |
| 01:19.55 | seanbright | and what new command was that? that's right, it was *dialplan reload* (with the *s) |
| 01:20.03 | seanbright | err |
| 01:20.09 | seanbright | /without/ the *s |
| 01:22.06 | De_Mon | I did for most stuff, extensions.conf -> extensions reload was just too damn easy to remember |
| 01:22.36 | De_Mon | I'll alias dialplan.conf to extensions.conf, maybe that will help ^_^ |
| 01:26.14 | justdave | nobody can read the deprecation notice in 1.4 because it's printed first before the command executes, and the output of the reload is verbose enough that it scrolls off before you see it. |
| 01:26.22 | justdave | for the extensions reload |
| 01:27.08 | justdave | I pay really close attention to the deprecation notices in 1.4, and try to train myself to use the new versions, and that's the first I've heard of it, and that's my excuse for not knowing about it. :) |
| 01:27.17 | De_Mon | um |
| 01:27.33 | De_Mon | so I'm looking at UPGRADE.txt, and I don't see a secontion on depriciated CLI commands |
| 01:27.38 | De_Mon | section |
| 01:28.12 | justdave | if I go back in my scrollback and look, it's indeed there. |
| 01:28.13 | justdave | ringring*CLI> extensions reload |
| 01:28.13 | justdave | Dialplan reloaded. |
| 01:28.13 | justdave | The 'extensions reload' command is deprecated and will be removed in a future release. Please use 'dialplan reload' instead. |
| 01:28.35 | jaytee | what, you mean the documentation is not setup in an orderly manner? that's unusual |
| 01:28.43 | justdave | followed by a couple thousand lines of "Loaded config 'fooo'" and "Added extension 'xxx' priority 1 to xxxxx" |
| 01:29.45 | justdave | s/Loaded config/Parsing/ |
| 01:29.46 | De_Mon | I looked at the screen buffer and didn't see the warning in 1.6 that the command was disabled it was just gone (as expected?) |
| 01:29.46 | seanbright | yeah, it's unfortunate, but the way that CLI output is displayed and the way that the deprecation notices are printed aren't the same |
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| 01:30.07 | seanbright | so there is no way to guarantee (right now) when stuff is printed out relative to other stuff |
| 01:30.15 | justdave | De_Mon: yeah that would be expected, since 1.4 told you it was deprecated |
| 01:30.43 | De_Mon | i suppose I could always jump back to 1.4 to see what the depriciation notice says ;p |
| 01:31.11 | De_Mon | this is probably going to be the only one I didn't remember, but I woln't know for sure till I've used 1.6 a while hehe |
| 01:31.37 | jaytee | I think they should modify the next minor release of 1.6 so whenever you type a deprecated command it just prints "What you talkin bout, Willis?" and doesn't do anything else. |
| 01:31.54 | seanbright | CLI aliases are being added |
| 01:32.11 | jaytee | that's just pandering to lazy people who won't read the docs |
| 01:32.11 | De_Mon | oh great |
| 01:32.39 | De_Mon | which doc should I read to know which commands were depriciated, i'm grepping the 1.6 docs and upgrade file and haven't found what I'm looking for yet |
| 01:33.04 | jaytee | they should make it as vague as possible and then in the upgrade.txt file where they mention the deprecated command put a little note: "Ah, so you found this finally! Don't you wish now that you'd read this first?" |
| 01:34.03 | justdave | What I was trying to accomplish with DUNDi is I have a few servers in separate geographic locations using a shared pool of extension numbers, because the offices all used the same server initially, and then the remote offices got their own servers when they started getting lots of people in them to get the resources local to them |
| 01:34.07 | De_Mon | I duno how other projects do it, but a nice 'this command is depriciated, use 'new command'' and forcing the change would be nice ;) |
| 01:34.22 | justdave | trying to put extension-by-extension routing between the servers is a pain |
| 01:34.30 | De_Mon | I wonder if I'll be able to alias the old command to something like that |
| 01:34.39 | De_Mon | 'no stupid it's dialplan reload!' |
| 01:34.45 | justdave | DUNDi works great for this (and already have it working) to go directly to the destination server for that extension |
| 01:35.12 | justdave | but we do have network glitches on occasion, and all of the offices can direct-connect to each other |
| 01:35.33 | justdave | and I've been able to successfully route around network issues by bouncing off a different office on the way to the destination one in the past |
| 01:35.53 | justdave | so I'm attempting to get it to serve a backup route to that extension in addition to the direct one |
| 01:38.57 | WimpMan | justdave: That's what the priorites must be there for. |
| 01:39.07 | justdave | WimpMan: yep, exactly. |
| 01:39.19 | justdave | except I can't figure out how to get it to give more than one response |
| 01:39.32 | justdave | whichever one I define last seems to be the one it uses |
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| 01:40.07 | WimpMan | last? |
| 01:40.24 | WimpMan | You do have different contexts, don't you? |
| 01:42.07 | justdave | no... guess that'd be the way to do it, put two separate DUNDi lookups in and have the second one explicitly look for the backup route after it fails on the primary |
| 01:42.33 | salzh | Upon answering a call, how can play some audio to the calling party automatically? |
| 01:42.38 | justdave | I was thinking because the responses had weights I could just put the less-priority weight on the backup route |
| 01:43.36 | WimpMan | So how do you try to give multiple answers without using multiple contexts? |
| 01:43.49 | seanbright | salzh: in the asterisk CLI (asterisk -r) type the command: `core show applications` |
| 01:44.14 | seanbright | salzh: from there you will see a list of all of asterisk's built-in apps and a description of what they do |
| 01:44.36 | ManxPower | as well as "core show functions" |
| 01:45.01 | justdave | what I tried (which obviously didn't work) : (only two lines, not worth a pastebin) |
| 01:45.05 | justdave | interoffice => local-extensions,0,IAX2,mountainview/${NUMBER},nopartial |
| 01:45.07 | justdave | interoffice => local-extensions,10,IAX2,copenhagen/722${NUMBER},nopartial |
| 01:45.17 | seanbright | of course nothing within "core show functions" will help with playing back audio... BUT... |
| 01:45.20 | seanbright | heh |
| 01:45.36 | justdave | all I ended up getting in responses was the copenhagen one, which makes sense, since the context is probably a hash so it overwrote the first one when it loaded |
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| 01:48.09 | WimpMan | justdave: Now how could that work if you give the same list to both priorities? |
| 01:49.08 | justdave | because one is weighted 0 and one is weighted 10, so it would try the 0-weighted one first and go to the 10-weighted one if the first one didn't work |
| 01:49.50 | justdave | but I guess that's only intended for figuring out which one to try if you asked multiple servers and got a response from all of them |
| 01:51.55 | WimpMan | Indeed. But That is what you want, I thought. |
| 01:54.20 | justdave | So I suppose what I really need is for copenhagen to answer "Yeah, I can get you to extension xxx in Mountainview" and give you the lower-priority weight on that response, instead of mountainview saying "you can get to me via copenhagen also" |
| 01:55.00 | WimpMan | Makes sense anyway. |
| 01:55.26 | WimpMan | You wouldn't want to use the intermediate if you could reach te destination directely. |
| 01:55.35 | justdave | right |
| 01:55.57 | WimpMan | So you wouldn't get dundi responses that way, either. |
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| 02:10.54 | seanbright | salzh: don't PM me |
| 02:11.19 | seanbright | wanders off |
| 02:11.58 | coil | can i |
| 02:12.09 | jaytee | can you what? |
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| 02:18.57 | jaytee | wb |
| 02:19.38 | [gnubie] | waves.. |
| 02:20.07 | [gnubie] | anyone cares to follow this thread? => http://lists.digium.com/pipermail/asterisk-users/2008-October/220054.html |
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| 03:07.01 | booray | Is there any documentation or specific usage examples of the T38 in 1.6? I've looked around as well as downloaded 1.6.0.1 and have yet to find anything other than a mention. Thanks |
| 03:09.08 | jaytee | I'm pretty sure the USAF has a jet figher trainer called a T-38 but other than that I've got nutthin |
| 03:11.11 | booray | lol |
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| 03:12.29 | booray | earlier when I asked, ManxPower said to download the source and find it there.. but a few greps and pouring through pdfs reveal little |
| 03:13.34 | drmessano | Friend of mine was just asking about an upgrade |
| 03:13.39 | drmessano | I told him I had no free time anymore |
| 03:13.52 | drmessano | He said "You should switch to 4 10's, it's great" |
| 03:14.03 | drmessano | He's on the 4day, 10 hour week thing |
| 03:14.16 | drmessano | I said "naah, 357 would do more damage than a 410" |
| 03:14.19 | jaytee | I'd love that |
| 03:14.31 | Qwell | so, how are you going to make up those other 20 hours a week? |
| 03:14.39 | Qwell | 5 12s > 4 10s |
| 03:14.44 | jaytee | rather than the 60-80 hour weeks at low salary paid for 40 hrs |
| 03:14.57 | drmessano | Hes govt.. He works 40 |
| 03:15.06 | Qwell | no, he "works" 40 |
| 03:15.10 | Qwell | he *works* 2. |
| 03:15.13 | drmessano | ROFL |
| 03:15.22 | drmessano | He works 40, 2 of which is billable |
| 03:16.17 | *** join/#asterisk SanityIO (n=SanityIO@77.242.103.223) |
| 03:18.43 | drmessano | I decided to get really into provisioning my Linksys phones |
| 03:18.58 | Qwell | You know what else you could get really into? |
| 03:18.58 | drmessano | Somehow I have it locked so I cant redial or check missed calls without a password |
| 03:19.00 | drmessano | OOPS |
| 03:19.00 | Qwell | testing. |
| 03:19.05 | drmessano | Try to |
| 03:19.08 | drmessano | Trying to |
| 03:19.23 | Qwell | announcement probably going out tomorrow :D |
| 03:19.41 | Qwell | I hope that server holds up.. |
| 03:19.53 | drmessano | I unplugged my test box at 10AM this morning and said "NO MATTER WHAT, IM LOADING THIS TODAY" |
| 03:19.57 | drmessano | and then I worked 11 hours |
| 03:20.02 | booray | go dodgers |
| 03:20.25 | booray | ducks and goes back to compiling 1.6.0.1 |
| 03:21.14 | drmessano | 1.6.0.1 is out? |
| 03:21.16 | drmessano | jesus |
| 03:21.35 | drmessano | I need to upgrade |
| 03:23.38 | Qwell | drmessano: 1.6.0 to 1.6.0.1 was a sample config fix |
| 03:23.50 | drmessano | No |
| 03:23.55 | drmessano | I am still on 1.4 |
| 03:23.57 | drmessano | lol |
| 03:24.05 | drmessano | I am falling behind |
| 03:24.59 | drmessano | What is the SVN tag for the Dahdi full? |
| 03:25.12 | Qwell | complete? |
| 03:25.21 | drmessano | yeah, that too |
| 03:25.25 | Qwell | svn/dahdi/linux-complete/tags/2.0.0+2.0.0 I think |
| 03:26.20 | Qwell | I actually got it right. sweet |
| 03:26.37 | drmessano | svn/dahdi/linux-complete/tags/2.0.0+2.0.0/ |
| 03:26.43 | drmessano | I just pasted that |
| 03:26.53 | drmessano | coolness |
| 03:32.03 | fakhir | anyone know when Asterisk TheFuture of Telephony book will be updated for 1.6? |
| 03:32.30 | Qwell | fakhir: when they find time |
| 03:32.37 | fakhir | :) ok |
| 03:32.42 | Qwell | I think the other day Leif summed it up as (paraphrasing)... |
| 03:32.53 | Qwell | "Being an author doesn't pay the bills" |
| 03:33.11 | fakhir | hehe yeah |
| 03:34.05 | drmessano | I hear theres a new Trixbox book coming out |
| 03:34.06 | drmessano | Wait no |
| 03:34.16 | drmessano | They are redesigning the Trix cereal box |
| 03:34.20 | drmessano | Ehh sorry |
| 03:34.22 | jaytee | lol |
| 03:34.43 | jaytee | There's already a Trixbox book out by Packt Publishing but it's old |
| 03:35.23 | Qwell | Packt is a joke, heh |
| 03:36.42 | jaytee | seems like it |
| 03:37.01 | Qwell | ha, Packt did the trixbox book... |
| 03:37.06 | Qwell | That makes so much more sense now |
| 03:37.09 | jaytee | and yet if you look at the examples in the O'Reilly book, *, TFOT there are lots of mistakes and bad advice |
| 03:38.03 | Qwell | show me one bit of advice that's bad O.o |
| 03:38.05 | jaytee | if you follow the examples on IAX2 trunking in The Book, you'll never get IAX2 trunking working |
| 03:39.15 | jaytee | and some of the dialplan examples have you setting the SIP account to match the extension. ManxPower and [TK]D-Fender made some very solid arguments against doing it that way. |
| 03:40.56 | jaytee | I was so psyched waiting for O'Reilly's Asterisk Cookbook but it never made it to press. Pity, I like to look at other people's examples. Ones that work, not the fucked up ones that get pastebinned in here by noobs. |
| 03:42.15 | drmessano | So I have NO password set for the SPA941 other than user Password |
| 03:42.21 | drmessano | and no additional security I can find |
| 03:42.35 | drmessano | But yet I am prompted for redial and to view missed calls |
| 03:42.48 | jaytee | instead all I get is "Hey, can someone tell me what's wrong with this line in my extensions.conf. It's from the [duh!] context. exten => 101,1,Goto(duh!,1) |
| 03:44.04 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 03:44.57 | jaytee | so I tell them, "It's missing the extension and a comma seperator". they say "Thanks" and leave and then they're back 2 minutes later going, "It's looping over and over!" and I say, "Well, isn't that what you wanted it to do?" |
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| 03:52.43 | jaytee | drmessano, don't know about that problem but you might look on voxilla.org. they've got a ton of Linksys specific info and good forums |
| 03:53.19 | drmessano | I hope its not looking for the user password |
| 03:53.30 | drmessano | Because its 3089nf084n90324hrn |
| 03:53.34 | drmessano | and I cant do that on a keypad |
| 03:53.46 | jaytee | since it's wireless could it be looking for a WEP key or WPA passphrase? |
| 03:54.07 | drmessano | Not wireless... SPA941 4-line desk phone |
| 03:54.23 | drmessano | Im baffled its asking for anything AT ALL |
| 03:54.35 | jaytee | does it register with *? |
| 03:54.40 | drmessano | Oh yes |
| 03:54.42 | drmessano | It works fine |
| 03:55.09 | drmessano | Only issue is it asking me for a password to redial or view missed calls |
| 03:55.38 | jaytee | that is weird |
| 03:55.42 | drmessano | I do have Protect_IVR_Factory_Reset enabled |
| 03:55.47 | drmessano | and this is LINKSYS |
| 03:56.15 | drmessano | You know what |
| 03:56.20 | drmessano | I bet... |
| 03:56.26 | drmessano | I bet I can disable that |
| 03:57.07 | jaytee | their built in dialpan crap in their ATA's are a pain in the ass and the docs suck |
| 03:57.12 | drmessano | The passwords are alphanumeric, so keeping someone from factory resetting from the keypad is pointless |
| 03:57.42 | drmessano | Well |
| 03:57.43 | drmessano | Hmm |
| 03:57.44 | drmessano | no |
| 03:57.57 | drmessano | Ok, I guess I need to google |
| 03:58.06 | jaytee | try voxilla |
| 04:00.02 | drmessano | AH |
| 04:00.04 | jaytee | throws his arms in the air and shouts "SERENITY NOW!!!" |
| 04:00.07 | drmessano | I figured it out |
| 04:00.12 | jaytee | ? |
| 04:00.27 | drmessano | Well, it's asking for the "user password" |
| 04:00.49 | drmessano | I changed it to something reasonable, and its allowing me to manipulate with that password |
| 04:01.26 | jaytee | but you can't set it so it doesn't need a password for redial or view missed calls? |
| 04:03.51 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 04:04.32 | drmessano | I dont see anything thats spelled out in English for that.. |
| 04:04.34 | drmessano | heres the thing |
| 04:05.18 | drmessano | I started using it from Factory defaults.. I dont remember having to do this before.. I pull the XML from the phone, it becomes my template for other phones |
| 04:05.34 | drmessano | So now i've reloaded the same config back via XML |
| 04:05.37 | drmessano | and something is off |
| 04:08.45 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-12-39.phlapa.east.verizon.net) |
| 04:09.50 | jaytee | like maybe the XML got corrupted somehow? |
| 04:10.55 | jaytee | I've seen it where the .cfg files for Polycoms that are actually XML code can get screwed up and cause crap to break. |
| 04:10.58 | drmessano | The editor I use has basic XML syntax checking |
| 04:11.29 | drmessano | Basically "does tag blah have a /blah"? |
| 04:11.56 | drmessano | I blank the user password, and I am not prompted |
| 04:12.00 | drmessano | Which is fascinating |
| 04:12.32 | jaytee | so it's a security feature. if the password is set it'll prompt, if not it just does works. |
| 04:13.16 | jaytee | or do you have other SPA941's that have passwords set but don't prompt? |
| 04:14.38 | drmessano | Well, I still think theres another feature here.. user password is used to ACL the web interface, as I am sure you are very aware with the ATAs.. it's not much different.. Seems like whatever this "feature" is that blocks redial and viewing missed calls would be a different toggle |
| 04:14.38 | drmessano | Like |
| 04:15.28 | drmessano | Hmm |
| 04:17.48 | jaytee | ok, time for me to get some sleep |
| 04:17.49 | drmessano | Im going through the Web UI as a USER to see if I can set that in the interface.. Logic here is that if I am a user, and I am given user rights to admin my address book, DHCP, etc.. I should be able to protect/unprotect the display |
| 04:17.49 | jaytee | nite |
| 04:17.52 | drmessano | Nite |
| 04:19.40 | slingr | i'm trying to connect to an spa3102 out of the box and the ip is responding but i can't get to the web page |
| 04:20.31 | drmessano | **** |
| 04:20.34 | drmessano | 7932# |
| 04:20.50 | drmessano | Then dial 1 # when prompted |
| 04:23.06 | slingr | without the LINE connected? |
| 04:23.38 | drmessano | facepalms |
| 04:23.49 | drmessano | Plug a phone in the FXS port |
| 04:23.54 | drmessano | Since you need one to dial stuff |
| 04:23.59 | slingr | yes |
| 04:24.10 | drmessano | Then dial follow what I typed |
| 04:24.14 | drmessano | -dial |
| 04:25.26 | slingr | the unit is just constantly resetting |
| 04:25.42 | slingr | and as i asked |
| 04:25.48 | slingr | should LINE be connected |
| 04:25.54 | drmessano | Doesnt matter |
| 04:25.57 | slingr | a phone is already connected to the phone port |
| 04:26.00 | drmessano | Youre using the IVR, not making a call |
| 04:26.32 | drmessano | But if its in a boot loop, you have other problems |
| 04:26.38 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 04:26.47 | slingr | nevermind |
| 04:26.49 | slingr | just got IVR |
| 04:27.35 | slingr | k, did what you said |
| 04:27.49 | slingr | got the web interface |
| 04:27.49 | slingr | thx |
| 04:28.00 | drmessano | Its disabled by default |
| 04:28.20 | drmessano | For whatever reason |
| 04:29.24 | slingr | weird |
| 04:29.43 | slingr | ok.. so now i can change it from NAT to Bridge (since i already have a router in place)? |
| 04:31.25 | slingr | nevermind, already did |
| 04:31.28 | drmessano | I suppose |
| 04:31.38 | drmessano | I actually never touch that |
| 04:32.18 | drmessano | The FXS and FXO ports use the WAN IP, youre only changing whats presented to the LAN port.. and since I never use one for a router... |
| 04:32.20 | slingr | do you use your spa3102 behind a router? |
| 04:32.28 | drmessano | yep |
| 04:32.47 | drmessano | Again, the SIP interfaces are on the WAN side |
| 04:33.14 | drmessano | So unless you're gonna use the LAN port with a device attached.. you're just telling it to bridge that useless port |
| 04:34.15 | slingr | in an asterisk+spa3102 |
| 04:34.22 | slingr | asterisk box is on the wan port/ |
| 04:35.02 | drmessano | Ok, let me make this simple |
| 04:35.15 | drmessano | Do you know what a basic ATA is, like a PAP2? |
| 04:35.17 | slingr | :/ |
| 04:35.26 | slingr | i've used asterisk before |
| 04:35.29 | drmessano | No |
| 04:35.32 | drmessano | Not what ive asked |
| 04:35.35 | drmessano | Do you know what a basic ATA is, like a PAP2? |
| 04:35.44 | slingr | i know what SIP is |
| 04:35.49 | slingr | i have no clue what PAP2 is |
| 04:36.04 | drmessano | have you ever used a SIP device ? |
| 04:36.08 | slingr | yes |
| 04:36.09 | drmessano | Like a PHONE? |
| 04:36.10 | slingr | like i said |
| 04:36.11 | drmessano | Ok |
| 04:36.13 | slingr | i've used asterisk before |
| 04:36.24 | slingr | i've just never used a device |
| 04:36.29 | slingr | just softphones using sip... |
| 04:36.34 | drmessano | Youre putting too much into this router business |
| 04:36.38 | slingr | and a wifi phone using sip |
| 04:36.50 | drmessano | Plug the WAN port into your switch, it will DHCP an address |
| 04:37.01 | slingr | k |
| 04:37.05 | drmessano | Done, finished.. forget the LAN port, bridging, and any of the router nonsense |
| 04:37.10 | drmessano | Youre not using it as a router |
| 04:38.06 | slingr | aye |
| 04:38.20 | slingr | thanks... i was trying to follow a convoluted howto on the net |
| 04:38.26 | slingr | you simplified it greatly for me |
| 04:38.45 | drmessano | wiki.2l2o.com <-- Go to the SPA-3102 guide |
| 04:39.01 | drmessano | Thats my wiki.. and my instructions work |
| 04:39.08 | drmessano | Most of the ones on the web are shit |
| 04:39.10 | slingr | thx doc |
| 04:40.23 | slingr | i'm just finishing up install of PIAF on a box here |
| 04:40.40 | drmessano | Most of the guides are based on the premise of slingshotting calls to the FXO port as a dumb proxy, and then blindly taking all incoming calls and slingshotting them to a fixed IP address |
| 04:41.03 | drmessano | I actually take into account that the FXO port on the SPA-3102 is a real SIP client, and have it register to the PBX |
| 04:41.13 | drmessano | Which means you can put it behind a NAT at some remote location, etc |
| 04:41.20 | drmessano | Which is how it SHOULD be done. |
| 04:41.44 | slingr | i see |
| 04:42.18 | slingr | so right now, while i'm configging the unit, i have wan port plugged into my switch and an analog phone plugged into PHONE port |
| 04:42.38 | drmessano | Ok |
| 04:42.41 | slingr | should I plug line into the wall jack? |
| 04:42.57 | drmessano | If thats what you're looking to do, yes |
| 04:43.22 | slingr | done |
| 04:45.14 | slingr | essentially |
| 04:45.44 | slingr | could LINE be plugged directly into the demarkation point so that all traffic hits the 3102 before going into the rest of the building |
| 04:45.45 | slingr | ? |
| 04:46.10 | drmessano | Yep |
| 04:47.13 | slingr | doc> you use a self-built asterisk setup or a prebuild (trix, elastix, piaf, etc)? |
| 04:47.58 | drmessano | Self-built.. Straight Asterisk or Asterisk+FreePBX depending on the need. |
| 04:50.04 | drmessano | I've been looking at the new AsteriskNOW though.. It wont replace my DIY for personal use, but it's a bloatless way of rolling out a no-crap Asterisk+FreePBX setup |
| 04:50.35 | drmessano | I'm actually working on building one for my Parents house |
| 04:51.12 | slingr | nice |
| 04:51.45 | slingr | i tried trix a long time ago, along with the very first initial versions of elastix, |
| 04:52.09 | slingr | i was told to try piaf if i wanted a no bullshit asterisk+freepbx-and-some-goodies |
| 04:52.29 | slingr | but eventually i would like to do my own asterisk+freepbx with debian |
| 04:52.34 | slingr | i just don't like cent0s |
| 04:52.52 | drmessano | Well, if you're talking about full-release current production "ZOMG ISO PBX", PIAF is the current fav.. but AsteriskNOW is the new hotness |
| 04:53.48 | slingr | asterisknow was a baby when i was last playing with pbx stuff |
| 04:53.58 | slingr | too immature to use at the time |
| 04:54.53 | drmessano | Its been rebuilt from scratch, now uses FreePBX |
| 04:58.41 | drmessano | hmm |
| 05:05.43 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:01.22 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 06:09.49 | ta^3 | I've an Asterisk 1.4.22 in a deadlock condition, where lots of AgentCallBackLogin() get stuck trying to lock because another thread do not release their p->lock at hangup. |
| 06:10.51 | ta^3 | also I have corresponding ast_show_channels ast_show_locks ast_show_threads a gdb thread apply al bt and the bt full of the conflicting threads. |
| 06:13.00 | ta^3 | I'm not pretty sure what the problem is and how to summary/describe this in order to report it. |
| 06:30.12 | *** join/#asterisk surajvs (n=su_raj_i@203.200.19.164) |
| 06:34.40 | mvanbaak | ta^3: then create a bugreport on http://bugs.digium.com |
| 06:34.48 | mvanbaak | <--- work |
| 06:34.49 | mvanbaak | latero |
| 06:35.02 | ta^3 | mvanbaak: that's my intention, just i'm not sure how to summarize it. :) |
| 06:50.47 | jameswf-home | the humor from development never goes public :( |
| 06:58.14 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 07:10.57 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:17.30 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:19.03 | *** join/#asterisk mateo_au (n=chatzill@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 07:20.38 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 07:34.57 | surajvs | hi; would anyone know as to how I could extend the extensions.conf ... |
| 07:35.28 | surajvs | I would like to call a different file from within extensions.conf |
| 07:37.45 | ta^3 | surajvs: just #include "anotherfile.conf" |
| 07:38.16 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:43.08 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 07:44.13 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 07:50.21 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:51.29 | surajvs | thanks ta^3 |
| 07:54.30 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 07:54.38 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:08.22 | *** join/#asterisk easycrypt (n=savek@ip-174.emscb.ruhr-uni-bochum.de) |
| 08:13.09 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 08:14.08 | *** join/#asterisk jameswf (n=james@ip72-200-94-120.tc.ph.cox.net) |
| 08:15.36 | *** join/#asterisk dch24 (n=dhubbard@65.105.155.98) |
| 08:16.57 | *** join/#asterisk mateo_au_ (n=chatzill@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 08:17.39 | dch24 | I'm running a new install of asterisk-1.2.27 with chan_alsa.so, but I get no audio when I dial 1000 (runs [demo]). chan_alsa.so initializes fine. does chan_alsa.so have a volume setting? |
| 08:17.59 | *** join/#asterisk coolthreads (n=shane@203-97-238-71.cable.telstraclear.net) |
| 08:18.02 | justdave | any particular reason you're using 1.2.x if it's a new install? |
| 08:18.31 | dch24 | no, I could install 1.4.x but my distro (gentoo) has 1.2.x still so it would be more difficult |
| 08:18.50 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 08:18.51 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 08:18.51 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 08:19.13 | justdave | ah. are you dialing that number from the console or something? |
| 08:19.23 | dch24 | yes, dial 1000 from the console |
| 08:19.55 | dch24 | would a pastebin of all the output help? |
| 08:20.11 | justdave | I never did much with console dialing, short of testing dialplans |
| 08:20.34 | justdave | I'm remote from all of my servers anyway, so whether the sound worked on the console would have been irrelevant for me :) |
| 08:20.42 | dch24 | :) |
| 08:20.58 | tzafrir_laptop | dch24, bang on the head of gentoo developers to start looking into 1.4.22 |
| 08:21.09 | tzafrir_laptop | (not to mention 1.6.0) |
| 08:21.24 | dch24 | I believe they have 1.4.x in an overlay and will be introducing it as a masked package soon |
| 08:21.25 | tzafrir_laptop | or make an ebuild on your own |
| 08:21.31 | justdave | kinda makes me wonder if the gentoo package is unmaintained |
| 08:21.35 | dch24 | (I just came from #gentoo-voip but they're all asleep I think) |
| 08:21.43 | justdave | gentoo typically has the latest and greatest of most stuff |
| 08:21.44 | tzafrir_laptop | dch24, I think I heard that over a year ago |
| 08:22.33 | tzafrir_laptop | #gentoo-voip asleep <--- sounds all too familiar as well |
| 08:22.51 | dch24 | oh |
| 08:23.04 | dch24 | I guess I will make an overlay for 1.4.x then :) |
| 08:23.16 | tzafrir_laptop | please post it somewhere |
| 08:23.33 | tzafrir_laptop | or do whatever is needed to start get it into gentoo |
| 08:23.37 | dch24 | ok, the easiest place is forums.gentoo.org (I'm dch24 there O.o) |
| 08:24.05 | dch24 | I will get back on here and figure out the alsa stuff if it doesn't work in 1.4.x |
| 08:24.18 | justdave | is an RPM zealot (as in wanting everything installed via RPM if at all possible, and avoiding installing stuff from source) |
| 08:24.26 | justdave | asterisk is one of my few exceptions to that |
| 08:24.40 | dch24 | here is the pastebin fwiw, http://pastebin.ca/1225741 |
| 08:24.50 | justdave | nobody packages it with any reliable frequency, it seems |
| 08:25.07 | justdave | and it doesn't take all that long to compile, so installing from source isn't that painful |
| 08:25.33 | dch24 | it was a nice and clean install... but I can do that again for 1.4.x |
| 08:26.00 | dch24 | appreciates the help |
| 08:26.12 | *** part/#asterisk dch24 (n=dhubbard@65.105.155.98) |
| 08:27.53 | *** join/#asterisk sluxor (n=sluxor@d58-110-244-109.per6.wa.optusnet.com.au) |
| 08:28.42 | sluxor | Is it possible to setup Asterisk as a Video VOIP PBX with MCU? |
| 08:29.28 | *** join/#asterisk virtexPro (n=virtex5@213.150.163.105) |
| 08:30.17 | flohack | Has someone ever experienced a starving asterisk when using realtime queues? My box responds very slowly to queue applications (addqueuemember, remove, pause, unpause) as soon as a few callers are active in a queue. Messages on the AMI lag as much as half a MINUTE! |
| 08:30.27 | flohack | I'm on 1.4.17 |
| 08:30.33 | sluxor | and if not. Can I purchase hardware or software for asterisk that would enable it to do so? |
| 08:32.38 | flohack | dynamic realtime that is |
| 08:33.32 | *** join/#asterisk samborambo_ (n=anja@60.234.186.247) |
| 08:35.29 | *** join/#asterisk darkskiez (n=mbryars@62-50-207-144.client.stsn.net) |
| 08:36.25 | samborambo_ | hi I'm having problems registering with a SIP peer. Getting ICMP errors (dest unrch in iptraf) but the packets are getting through. Behind a NAT with 5060 forwarded to my asterisk box. |
| 08:36.39 | samborambo_ | any ideas? |
| 08:39.44 | samborambo_ | anyone? |
| 08:42.38 | subdolus | samborambo_: you need to allow incoming AND outgoing SIP ports |
| 08:42.51 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:44.14 | samborambo_ | subdolus: thanks for replying........I don't have iptables turned on on the asterisk box.....just sitting behind a nat firewall |
| 08:45.39 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 08:45.42 | *** join/#asterisk emer08 (i=emerzone@122.55.66.9) |
| 08:46.36 | subdolus | yep.. shouldn't need to. it's the NAT box / firewall that needs to allow the incoming and outgoing ports |
| 08:47.21 | samborambo_ | but I can see the packets coming back in iptraf and ngrep |
| 08:48.00 | *** join/#asterisk kannan (n=kannan@123.201.136.118) |
| 08:48.15 | samborambo_ | so I'm thinking they're being passed through NAT OK |
| 08:48.27 | subdolus | ah ok |
| 08:48.30 | subdolus | Hmm |
| 08:48.47 | kannan | hello al. I have a problem, channel zap not loading. I get the error zt_open nable to specify channel 13. Its a Sangoma 24port FXS A400 card. It was working fine upto now |
| 08:49.14 | subdolus | samborambo_: have you checked in forums related to your provider as to the sip.conf settings? |
| 08:49.38 | subdolus | if possible, could you pastebin the extension in sip.conf for the provider you're trying to register? |
| 08:49.51 | subdolus | (minus the login deets ;)) |
| 08:50.45 | tzafrir_laptop | kannan, what happened "now" that changeed things? |
| 08:50.55 | samborambo_ | yeah, I used their standard sip setup. I started with that and have been having problems....can I msg paste to you? |
| 08:51.19 | subdolus | pastebin is easier, but if you must |
| 08:51.54 | kannan | tzafrir_laptop, i havent changed a thing |
| 08:52.14 | kannan | i powered off the machine last night, now after start up, it does not work |
| 08:52.40 | samborambo_ | never used pastebin......will give it a go..... |
| 08:54.04 | samborambo_ | http://pastebin.com/d96d8e78 |
| 08:57.42 | samborambo_ | I've got an old USB ADSL modem out in the shed that I could fire up as a direct connection from the box to see if I can eliminate the NAT router. |
| 09:00.07 | samborambo_ | subdolus: any thoughts? |
| 09:01.20 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 09:04.04 | kannan | tzafrir_laptop, i am not able to rmmod the modules manually, the machine just freezes. In the asterisk CLI, module load chan_zap.so, show channels 1-12 fine, then it aborts with error. |
| 09:04.08 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:05.39 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 09:12.31 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:14.13 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:10) |
| 09:15.58 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 09:21.15 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:21.35 | *** join/#asterisk ghostknife (n=black@196-210-142-208-wbs-esr-2.dynamic.isadsl.co.za) |
| 09:22.40 | ghostknife | Is it possible to make a dial pattern, when someone dials "71xxxxxxx" (7 followed by a 1, followed by 7 digits), to have it send a "012", followed by the 1 and the 7 digits? |
| 09:22.57 | viraptor | could someone tell me what load ratio (user/sys/wait) do they have on a box with a busy asterisk? |
| 09:22.58 | ghostknife | so, someone dials, 713334444, it sends, 01213334444 ? |
| 09:24.58 | viraptor | ghostknife: 71xxxxxxx => Dial(... 012${EXTEN:1} ...) |
| 09:26.51 | ghostknife | viraptor: where do I add this? (sorry, I'm "still in the packet" new to this) |
| 09:27.24 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 09:27.40 | viraptor | ghostknife: extensions.conf - add something like |
| 09:28.28 | viraptor | exten => _71XXXXXXX,1,Dial(SIP/012${EXTEN:1}) |
| 09:31.42 | ghostknife | OK, for the sake of learning, I also added an extra trunk which matches exactly the working one, but when dialing out on it, it simply says "all circuits are busy now" |
| 09:31.55 | slingr | hey all |
| 09:32.15 | slingr | i'm following this tut on the spa3102 setup: http://wiki.2l2o.com/index.php/Linksys_SPA-3102 |
| 09:32.22 | slingr | at the end it references setting up an asterisk peer |
| 09:32.26 | slingr | what does that mean :/ |
| 09:32.27 | slingr | ? |
| 09:34.54 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 09:35.28 | Zeeek | just missed steliosk |
| 09:35.29 | *** join/#asterisk steliosk (n=Stelios@athedsl-4421802.home.otenet.gr) |
| 09:35.36 | Zeeek | steliosk: how are you? |
| 09:35.51 | Zeeek | Did you get any interest at Astricon from the Voip Users COnference? |
| 09:37.37 | slingr | does asterisk peer = extension? |
| 09:38.17 | Zeeek | peer is something that connects to asterisk |
| 09:38.35 | Zeeek | usually with login credentials |
| 09:39.47 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 09:39.57 | steliosk | Zeeek : Hi ! |
| 09:40.05 | Zeeek | people often set the extension and peer to the same name, causezing confusion |
| 09:40.17 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 09:40.24 | steliosk | Zeeek : Yeah i got a few of them at the stand |
| 09:40.40 | Zeeek | I recommend setting the peer for phones to their name or even MAC address |
| 09:40.48 | Zeeek | steliosk: cool. That's what it's all about |
| 09:41.11 | Zeeek | slingr: still there? |
| 09:41.44 | steliosk | Zeeek : Astricon was good this year : http://digital-opsis.com/astricon08 |
| 09:41.45 | Zeeek | slingr: so one refers usually to the "peer entry" |
| 09:42.23 | steliosk | Zeeek : The winner of the ruffle for the free Astricon pass come from Belgium also. He was very happy :) |
| 09:42.55 | slingr | sorry i;m back |
| 09:42.59 | Zeeek | If you have only one phone, the peer entry could be called [sipura] or something |
| 09:43.01 | slingr | yeah Zeeek> sup? |
| 09:43.22 | Zeeek | under the peer heading you'd put the codecs, the host and secret info etc |
| 09:43.29 | slingr | Zeeek> can you take a look at the end of this page: |
| 09:43.30 | slingr | http://wiki.2l2o.com/index.php/Linksys_SPA-3102 |
| 09:43.44 | slingr | i'm not sure where i'm supposed to set that asterisk information |
| 09:43.56 | slingr | under "Asterisk Peer Setup" |
| 09:44.06 | Zeeek | in sip.conf |
| 09:44.09 | jstocks | Question: My agi scripts never seem to get my digits that I press on my phone, but if I do any built in thing like confrence calls or the dication they all read the digits just fine. What am I missing? |
| 09:44.09 | slingr | unless thats just what i set for a peer extention |
| 09:44.12 | slingr | thx |
| 09:44.37 | kannan | hello al. I have a problem, channel zap not loading. I get the error zt_open nable to specify channel 13. Its a Sangoma 24port FXS A400 card. I re-installed the wanrouter, but still same error |
| 09:44.41 | Zeeek | slingr: don't forget, as they say at the end: Create a [from-SOMEUSERNAME] context in extensions.conf with an appropriate destination. |
| 09:45.03 | kannan | i get channel 13 : no such device or address when I do a mdule load in the asterisk CLI |
| 09:45.19 | slingr | ok, now using FreePBX, is there a place i can put that info... should it go in sip-custom.conf? |
| 09:48.06 | Zeeek | there was freepbx info just above asterisk. Go to #freepbx |
| 09:48.10 | slingr | doh |
| 09:48.21 | slingr | lol i already inputted all the infor |
| 09:48.27 | slingr | i was just getting confused |
| 09:48.28 | tzafrir_laptop | kannan, what do you see on /proc/zaptel/* |
| 09:48.29 | slingr | thz Zeeek :) |
| 09:48.34 | Zeeek | np |
| 09:48.34 | tzafrir_laptop | cat /proc/zaptel/* |
| 09:51.04 | *** join/#asterisk colulu (n=jg@61.141.158.178) |
| 09:51.32 | slingr | can someone recommend a good, free, sip softphone? |
| 09:51.34 | colulu | I am connecting 2 ss7 link with one Asterisk on a loopback mode. Could someone tell me how to set the point code? |
| 09:51.47 | slingr | x-lite? |
| 09:51.48 | colulu | I need help in setting up a dpc and opc |
| 09:52.54 | kannan | tzafrir_laptop, kindly gibe me a few min, brb |
| 09:53.02 | kannan | not gibe |
| 09:53.05 | kannan | lol, give |
| 09:55.15 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:55.27 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 10:01.17 | *** join/#asterisk setkeh (n=setkeh@CPE-124-180-146-148.vic.bigpond.net.au) |
| 10:05.41 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 10:08.45 | colulu | hi |
| 10:08.51 | colulu | could somone help me with a ss7 problem? |
| 10:09.33 | *** join/#asterisk jes-o-ma1 (i=jesusch@irc.82110clan.de) |
| 10:09.41 | jes-o-ma1 | Hi |
| 10:10.46 | jes-o-ma1 | I have several pcap files and each file includes a single sip-conversation (including RTP traffic) |
| 10:11.16 | jes-o-ma1 | I'd like to covert these files into wav - anyone knows how to do this? |
| 10:12.34 | *** join/#asterisk masingerz (n=sueter@201.200.64.254) |
| 10:12.38 | masingerz | hello |
| 10:18.26 | ta^3 | jes-o-ma1: http://www.enderunix.org/voipong/ |
| 10:19.46 | colulu | hi does anyone know how to set the opc parameter for a loopback ss7 asterisk connection? |
| 10:20.20 | *** join/#asterisk mateo_au (n=chatzill@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 10:26.42 | jes-o-ma1 | ta^3: I already tried voipong, but when I use tcpreplay to get the session on a dummy interface it somehow is not recognized as a voip session |
| 10:27.27 | jes-o-ma1 | maybe it's only working with H.323? |
| 10:27.40 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 10:28.48 | ta^3 | jes-o-ma1: Can't remember if wireshark is able to save the RTP stream (but I'm pretty sure it's able to play them) as a RAW file, then you can transcode it with Audacity. |
| 10:32.11 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 10:38.56 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 10:39.29 | XnOSX | how i can to know what kind or model of digium card is inside a server? |
| 10:42.13 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 10:42.41 | obruT | XnOSX: lspci ? |
| 10:43.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:43.53 | Frogzoo | people have seen this I assume? http://share.skype.com/sites/en/2008/09/skype_for_asterisk_beta.html |
| 10:48.47 | tzafrir_laptop | XnOSX, zaptel_hardware |
| 10:50.42 | kannan | tzafrir_laptop, hmm now I am no loger able to install the Sangoma card, it says not detected , so I will start from scratch again, re-install the OS |
| 10:51.42 | coolthreads | umm I find that when I play audio files its sounds slow and choppy also noisy when using either playback() or background(). Not a problem when it comes to actual voice calls though. |
| 10:52.13 | coolthreads | any cheap pointers?? |
| 10:53.48 | tzafrir_laptop | kannan, re-install is not the solution |
| 10:53.58 | tzafrir_laptop | How about some trouble-shooting ? |
| 10:54.11 | tzafrir_laptop | I asked you a simple question |
| 10:57.21 | *** join/#asterisk fdz (n=francois@c2cpc3.camptocamp.com) |
| 10:58.16 | kannan | tzafrir_laptop, ok |
| 10:58.31 | kannan | tzafrir_laptop, i have another system, the card is working on that one |
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| 11:00.06 | kannan | tzafrir_laptop, how to start trouble shooting? I can start with the card in the machine only after 8 hours though? If you will be kind enough to let me know where to sart , i will read up by then |
| 11:00.56 | tzafrir_laptop | what do you try to do? What do you expect it to do? What actually happens? |
| 11:01.12 | tzafrir_laptop | Also: what system is it? What versoins of the relevant compoents? |
| 11:01.41 | tzafrir_laptop | kannan, note that the two lines above are generic troubleshooting questions |
| 11:02.59 | kannan | ok , I ave a Intel Core 2 Duo, 4 GB AM and Seagate SATA HDD.I installed Slack 12.1 . Without zap all the rest of asterisk dialplans work fine. When I go to the wanpipe src directory and do a ./Setup install, at the end it says no compatible Sangoma voice card found |
| 11:03.20 | kannan | though I just re-installed the card once , and then it was fine |
| 11:03.33 | kannan | i re-installed just an hour back |
| 11:04.12 | kannan | at that time it came up with an eror , that channel 13 zt_open error , and the zap module never loaded |
| 11:05.26 | kannan | Google has quite a few results on this one, I am reading on it |
| 11:05.32 | jes-o-ma1 | ta^3: but I'd need sometinh scriptable and I haven't figured out if that might be possible using tshark |
| 11:05.33 | ta^3 | kannan: have you connected the 12V power molex to the A400 card? |
| 11:06.38 | kannan | NI, but the asterisk bx was untouched when it stopped working, except that it was re-started |
| 11:09.52 | tzafrir_laptop | kannan, does the card show up in lspci? |
| 11:13.57 | kannan | tzafrir_laptop, i have not checked |
| 11:14.16 | kannan | i read that one, that it can be a h/w problem |
| 11:14.24 | kannan | but then it works on h redhat server |
| 11:14.25 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 11:16.35 | kannan | i will check it now |
| 11:16.45 | kannan | switching the card back now . |
| 11:21.39 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 11:23.44 | ta^3 | kannan: don't forget to connect the molex connector into the card. |
| 11:23.52 | kannan | ta^3 , ok sure |
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| 11:29.36 | kannan | ok, lspci -v shows the card |
| 11:29.43 | kannan | i wll try to build again |
| 11:30.02 | *** join/#asterisk setkeh (n=setkeh@CPE-124-180-146-148.vic.bigpond.net.au) |
| 11:30.25 | kannan | ./Setup install |
| 11:30.36 | kannan | its a Sangoma a400 FXS |
| 11:32.22 | *** join/#asterisk marc7 (n=marc@S010600195bd55571.vc.shawcable.net) |
| 11:32.51 | marc7 | there's no ./configure or make menuselect in dahdi (something I was used to seeing with zaptel) |
| 11:33.15 | marc7 | i'm just trying to get the dahdi_dummy timing module put into place |
| 11:33.48 | marc7 | i'm reading the UPGRADE.txt as part of dahdi-2.0, I suppose that's fine... i don't need to menuselect my way to opt out of the other modules |
| 11:34.12 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:34.53 | ta^3 | marc7: no, there is not, all modules are built in dahdi-linux. |
| 11:35.36 | kannan | ok it read the card!! |
| 11:36.13 | kannan | how long ds it take to sart the wanpipe? the output reads starting device wanpipe1 , its about 2 minutes now |
| 11:37.01 | ta^3 | marc7: the configure script is for dahdi-tools, just install both and edit /etc/dahdi/modules adding a new line with dahdi_dummy (remove/comment everything else) |
| 11:37.10 | kannan | should i give a ctrl+Z or somthing |
| 11:37.42 | ta^3 | kannan: what does /var/log/wanrouter says? |
| 11:37.51 | marc7 | ta^3: nice. just going through the usual `make install; make config` on dahdi-tools has worked better than zaptel's treated me in the past |
| 11:38.03 | marc7 | autodetected the right modules to load into the kernel |
| 11:38.32 | kannan | hooray |
| 11:38.37 | kannan | it is ok |
| 11:38.44 | kannan | and zttool shows ok |
| 11:39.20 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 11:39.50 | kannan | i really dont know what had happened, I dd not do anything different AFAIK |
| 11:40.19 | kannan | why the zap module did not lad in the first place all of a suden, we used this card for more than 8 onths now |
| 11:40.22 | kannan | month |
| 11:40.52 | ta^3 | marc7: default dahdi config loads wct4xxp, wcte12xp, wct1xxp, wcte11xp, wctdm24xxp, wcfxo, wctdm and xpp_usb and if there is no card detected it also loads dahdi_dummy; i prefer to -in your case- to just load dahdi_dummy. |
| 11:41.16 | marc7 | ta^3: like you said, clear out /etc/dahdi/modules to only include the dahdi_dummy line |
| 11:41.28 | ta^3 | marc7: :) |
| 11:41.46 | marc7 | *heart warms with moving 1.6 deployment along* |
| 11:42.24 | marc7 | we want to play around with this a bit more in our staging environment. eventually we'll throw 1.6 debs together for distribution on our servers internally |
| 11:42.32 | ta^3 | kannan: perhaps you forgot to plug something meanwhile switching the card between servers. |
| 11:42.46 | kannan | tzafrir_laptop, ta^3 -> thank you |
| 11:43.15 | jes-o-ma1 | ta^3: fyi - it seems that pcapsidump is filtering RTCP packets - that is why voipong does not capture any calls |
| 11:43.20 | kannan | ta^3, nope, the card stopped working , (the server was untouched), |
| 11:43.31 | kannan | i built wanpipe again, same ror |
| 11:43.41 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 11:43.42 | kannan | then i rebuilt the wanpipe, and it refused to build |
| 11:43.53 | kannan | so then after only i switched the card |
| 11:44.02 | kannan | as there is a conference in an hour |
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| 11:59.04 | marc7 | can anybody shed some light on what /var/lib/asterisk/phoneprov/ is all about? I understand that they're polycom sip phone configuration files, but how does that overlap with asterisk? |
| 12:00.03 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 12:00.26 | marc7 | *digs and starts reading on res_phoneprov |
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| 12:08.22 | write_erase | Hi ... How cay you define "4 to 10 numbers prefixed by 0 ?" in an exten ? |
| 12:08.46 | write_erase | numbers->digits sorry |
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| 12:34.25 | *** join/#asterisk DogWater (i=DogWater@dhcp91.cmh.ee.net) |
| 12:34.51 | DogWater | Hi, this is slightly unrelated to Asterisk, but is there any general VoIP channel available on Freenode that anyone is aware of? |
| 12:35.08 | Zeeek | #voip-users-conference |
| 12:35.19 | Zeeek | but don't pee on the trees |
| 12:36.13 | DogWater | Ah, well we just ended up with a UC520 setup (cisco stuff) and we're having some funky problems with it and finding any answers from the big C is just about as easy as cold fusion (not the programming language). |
| 12:36.32 | DogWater | thanks for the tip Zeeek |
| 12:36.47 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:36.49 | Zeeek | come by and ask the question, who knows, maybe someone can help |
| 12:36.59 | Zeeek | DogWater ^^^^^^^ |
| 12:37.39 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 12:43.57 | beek | Morning Zeeek (or would that be afternoon for you?) |
| 12:44.38 | Zeeek | yeah, it's after lunch even :) |
| 12:45.28 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:47.17 | flohack | Has someone ever experienced a starving asterisk when using realtime queues? My box responds very slowly to queue applications (addqueuemember, remove, pause, unpause) as soon as a few callers are active in a queue. Messages on the AMI lag as much as half a MINUTE! |
| 12:47.20 | flohack | dynamic realtime that is |
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| 13:01.11 | kannan | tzafrir_laptop, ta^3 -> regarding the sangoma cd, it doesnt work on particular PCI slot thats all |
| 13:01.15 | kannan | card |
| 13:01.28 | kannan | some defect in that one thnk |
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| 13:04.58 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-133.dyn.embarqhsd.net) |
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| 13:07.27 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 13:08.35 | Katty | yawns |
| 13:09.08 | jaytee | mornin Katty, how's Riddick? |
| 13:09.15 | Zeeek | belches and rubs stomach |
| 13:09.20 | Katty | he's... |
| 13:09.22 | Katty | Zeeek: )= |
| 13:09.27 | Katty | he's in his kennel, hopefully asleep |
| 13:09.37 | Zeeek | I wish I was |
| 13:09.43 | *** join/#asterisk Carlos_PHX (n=Carlos@vpn.pensus.com) |
| 13:09.44 | seanbright | in his kennel? |
| 13:09.47 | seanbright | pervert. |
| 13:09.55 | Zeeek | naw, I gave that up a long time ago |
| 13:10.05 | Katty | his kennel is on a toddler matress |
| 13:10.13 | Zeeek | I'm down for that |
| 13:10.14 | Katty | and in the kennel, there are pillows... |
| 13:10.20 | Zeeek | I like pillows |
| 13:10.21 | Katty | and on top of the pillows are 3 fleece baby blankets |
| 13:10.29 | Zeeek | I like those blankets |
| 13:10.33 | Katty | and he has a handful of toys in there |
| 13:10.34 | seanbright | jeebus |
| 13:10.36 | Zeeek | any kibbles around? |
| 13:10.38 | Katty | including a kong stuffed with treats. |
| 13:10.42 | seanbright | a dog has it better than me |
| 13:10.44 | Zeeek | YES |
| 13:11.05 | Katty | seanbright: i imagine riddick has it better than some children do. |
| 13:11.11 | Zeeek | my mistress just came home from grocery shopping. I'll go beg some treats now |
| 13:11.14 | Katty | seanbright: as sad as that sounds :/ |
| 13:12.49 | *** join/#asterisk jameswf (n=james@ip72-200-94-120.tc.ph.cox.net) |
| 13:13.45 | Zeeek | just fruit :( |
| 13:17.48 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 13:20.18 | nikko | is there an asterisk business channel? |
| 13:21.14 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
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| 13:33.38 | jaytee | wow, over 72 hours with no sign of [TK]D-Fender. Is he on vacation? |
| 13:36.27 | Katty | pamples things |
| 13:36.39 | Katty | i don't recall fender saying anything about vacation |
| 13:36.43 | Zeeek | incredible, but maybe it's because of Thanksgiving? |
| 13:36.52 | Katty | thanksgiving? |
| 13:36.57 | Katty | you're a month ahead sweety |
| 13:37.05 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 13:37.09 | Katty | we've not even done halloweens yet |
| 13:37.19 | Katty | riddick has a cute lil costume :> |
| 13:39.40 | Katty | nothing on his facebook, tho that doesn't surprise me |
| 13:41.28 | Zeeek | second Monday in October is ThxGvg in Can |
| 13:41.36 | *** join/#asterisk tamberlo (n=tamberlo@net-93-144-169-217.t2.dsl.vodafone.it) |
| 13:41.47 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:41.48 | Zeeek | the thought of [TK having family is astounding though |
| 13:42.16 | Zeeek | for those who need learning: http://www.crewsnest.vispa.com/thanksgivingcanada.htm |
| 13:43.51 | *** join/#asterisk marc7 (n=marc@S010600195bd55571.vc.shawcable.net) |
| 13:45.57 | tamberlo | hi, my first time in this cool channel... |
| 13:46.53 | tamberlo | i've finished right now to read ... "how to politely use the asterisk irc channel |
| 13:47.15 | Katty | Zeeek: oh right. |
| 13:47.21 | Katty | Zeeek: forgot about that canada bit |
| 13:47.36 | Katty | Zeeek: also |
| 13:47.42 | Katty | Zeeek: wanna see riddick's halloween costume? :> |
| 13:47.51 | Zeeek | sure |
| 13:48.14 | tamberlo | :P so... i'll try to make a question : |
| 13:48.33 | Katty | Zeeek: http://www.facebook.com/photo.php?pid=34132117&l=06cc5&id=37617946 |
| 13:49.00 | tamberlo | i've compiled succesful the wrapper astxx for c++ |
| 13:49.13 | tamberlo | but it won't work... |
| 13:49.35 | tamberlo | how exactly can I use that wrapper with AGI ? |
| 13:49.36 | Zeeek | awwwww |
| 13:49.50 | Katty | Zeeek: http://www.facebook.com/photo.php?pid=34132110&l=fe91f&id=37617946 <- that's his kennel, but he has more pillows and blankets in there now. |
| 13:50.26 | Katty | and yes, that is a toddler mattress. |
| 13:50.30 | Zeeek | he's blocking the bathroom door! |
| 13:50.40 | *** join/#asterisk jameswf (n=james@ip72-200-94-120.tc.ph.cox.net) |
| 13:50.42 | Katty | that's a closet ;) |
| 13:50.46 | Katty | we don't ever get into... |
| 13:50.58 | Katty | bathroom is the door to the right of that one :P bedroom to the left. |
| 13:52.10 | tamberlo | just for couriosity.... any italian in teh forum ??? |
| 13:53.11 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 13:53.26 | Katty | Zeeek: he's so spoiled :> |
| 13:53.49 | Zeeek | UnixDawg: howdy |
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| 13:54.04 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 13:54.08 | UnixDawg | morning |
| 13:54.09 | Dovid | hi all |
| 13:54.24 | Dovid | are the granstream ATA's as bad as their phones ? |
| 13:55.42 | UnixDawg | so is the new asterisk now iso out ? |
| 13:56.06 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
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| 14:01.18 | jaytee | Dovid, they're about the same. I've used the HandyTone-286 and compared a Panasonic cordless plugged into it and then plugged into a Linksys SPA2102 and the Linksys is has much cleaner audio. |
| 14:01.43 | Dovid | thanks |
| 14:01.53 | jaytee | UnixDawg, I think it's still in alpha |
| 14:01.55 | Dovid | I have used liksys but too many NAT issues |
| 14:02.10 | Dovid | ~grandstream |
| 14:02.11 | jbot | [grandstream] the Yugo of VoIP hardware. Run. Run away now. |
| 14:02.14 | *** join/#asterisk AndyML (n=AndyML@pool-96-227-91-204.phlapa.fios.verizon.net) |
| 14:02.16 | UnixDawg | jaytee is there a url to get the alpha for testing |
| 14:02.17 | jaytee | Dovid, if you're talkin about the built in NAT you can disable that and just run bridged. |
| 14:02.32 | Dovid | jaytee: behind a router |
| 14:02.37 | Dovid | with multiple devices |
| 14:02.39 | jaytee | UnixDawg, nope not that I'm aware of. You'd have to ask someone at Digium |
| 14:02.54 | Dovid | have u used Planet ATA's ? I heard they are crap but i am looking for low budget |
| 14:03.06 | UnixDawg | all the digium people seem to be asleep at the keyboards |
| 14:03.09 | jaytee | nope, never used them |
| 14:03.10 | Dovid | lol |
| 14:03.39 | jaytee | UnixDawg, they must have been all hired on from Napster.....(ducks and hides) |
| 14:04.11 | UnixDawg | lol |
| 14:04.19 | *** join/#asterisk scrash08 (n=scrash08@unaffiliated/scrash08) |
| 14:05.24 | scrash08 | The 'handbook' @ "https://www.digium.com/en/supportcenter/documentation/viewdocs/asterisk_handbook" dates back to 2006. |
| 14:05.25 | scrash08 | Are there more current docs available? |
| 14:06.14 | UnixDawg | nope thats the best |
| 14:06.17 | UnixDawg | use it |
| 14:06.24 | UnixDawg | its only 2 years old |
| 14:07.26 | Qwell | ~book |
| 14:07.26 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:07.51 | Madkiss | does ${EXTEN:1} remove the first or the last digit from a called number? |
| 14:08.22 | *** join/#asterisk tuxfoo21 (n=tmmarini@198.183.217.179) |
| 14:08.36 | WimpMan | Madkiss: First |
| 14:09.24 | Madkiss | Okay. I am seeing a strange effect. while some clients can call in just nicely, others complain that they can't dial specific phones from employees directly. Indeed I see they are sent into the "invalid"-context |
| 14:11.46 | scrash08 | Qwell: UnixDawg Thanks. So, nothing with for v1.6? I'd read that there are lots of changes ... I.e., are 1.4 still relevant for a 1.6 deployment? |
| 14:12.31 | UnixDawg | 1.6 just came out |
| 14:13.16 | UnixDawg | there is alot to be done to update documnets and the book for it. There is alot more to it and you can always look at the change log |
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| 14:15.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 14:15.47 | XnOSX | hello friends i have a warning in my CLI Asterisk, anybody know about what is these error? |
| 14:15.48 | XnOSX | ast_rtp_read: RTP Read too short |
| 14:15.56 | ibm2 | hello, can anyone tell me how i can make instant message between 2 extension |
| 14:16.50 | scrash08 | UnixDawg: Understood. I'm simply trying to avoid following docs that'll lead me completely astray (e.g., config files, etc) cuz of the different version. As a rang asterisk begineer, I have NO sense of the magnitude of the difference, other than the online "rants" about 1.6 API breaking dialplans, etc. Hence my question abt relevance ... |
| 14:16.53 | Madkiss | "_s. => goto ${EXTEN}|1;" -- what would that thing do in asterisk? |
| 14:22.14 | Katty | go to the extension,s,1? |
| 14:22.55 | Katty | that'd more more like foo => Goto(${EXTEN},s,1) in the dialplan tho |
| 14:25.03 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.114) |
| 14:25.36 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:25.52 | Katty | seanmh: hai |
| 14:25.57 | Katty | hugs seanmh |
| 14:26.26 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:30.09 | Madkiss | okay, the problem looks like something related to overlap dialing |
| 14:31.31 | *** join/#asterisk tali81 (i=434e2716@gateway/web/ajax/mibbit.com/x-a45becd7606a966c) |
| 14:31.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 14:37.35 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
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| 14:45.22 | jaytee | Katty, didn't you do a How-To on setting up FOP? |
| 14:45.45 | Assimilate | Is there a command line way to check a Wildcard TE120p card? I am getting a fast busy tone when calling its number |
| 14:47.19 | *** join/#asterisk mog (n=mog@nat/digium/x-9556009266cecd5f) |
| 14:47.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:51.34 | *** join/#asterisk sack (n=sack@106.Red-88-24-156.staticIP.rima-tde.net) |
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| 14:56.05 | ibm2 | hello, can anyone tell me how i can make instant message between 2 extension |
| 14:57.08 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 14:57.08 | *** mode/#asterisk [+o twisted] by ChanServ |
| 14:58.16 | *** join/#asterisk bkw__ (n=brian@freeswitch/developer/bkw) |
| 15:02.42 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-164-37.dyn.embarqhsd.net) |
| 15:02.46 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:03.03 | *** join/#asterisk mocker (n=kyle@198.247.173.227) |
| 15:03.51 | mocker | Anyone have a recommendation on where to buy some Polycom IP330's w/ AC adapters? I looked on voip supply but it doesn't look like they sell the AC adapters. |
| 15:04.02 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 15:04.18 | bkw__ | mocker: they sell the adapters |
| 15:04.22 | bkw__ | just select it from the drop down list |
| 15:04.50 | mocker | checks again. |
| 15:04.55 | Qwell | Are the polycom adapters just PoE injectors? |
| 15:05.04 | jaytee | mocker, the prices at www.telephonydepot.com are usually better. we buy our Polycoms from CDW and buy the AC adapters in boxes of 5. |
| 15:05.05 | Qwell | and would they work with other PoE devices of the same class? |
| 15:05.15 | Qwell | (the newer Polycom adapters..) |
| 15:05.18 | coppice | they'll sell you power over ethernet? will they sell you power over hot women, too? |
| 15:05.33 | jaytee | Qwell, the AC adapter is just AC. there |
| 15:05.41 | Qwell | jaytee: gee, thanks |
| 15:05.51 | jaytee | 's another adapter they sell for PoE |
| 15:06.12 | Qwell | oh.. was it the older ones that were just PoE adapters then? |
| 15:06.30 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-913f5da83f2af2ed) |
| 15:06.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:08.02 | *** join/#asterisk tompaw (n=tompaw@slave12.tesserakt.eu) |
| 15:08.12 | tompaw | hello |
| 15:08.14 | jaytee | Qwell, not sure. I'm using on Linksys PoE adapter with a Polycom 320 but I haven't bought any of their PoE adapters. |
| 15:08.15 | tompaw | how do I express "a_string" + ${VARIABLE} in asterisk? |
| 15:08.25 | tompaw | for exmaple: Macro(trunkdial,SIP/trunk_blah/48${EXTEN:2}) |
| 15:08.41 | jaytee | I think it was for the 501 model required an adapter/injector |
| 15:08.51 | *** join/#asterisk wonderworld (n=ww@ip-62-143-38-55.unitymediagroup.de) |
| 15:10.09 | wonderworld | hey, i upgraded my * to 1.6.0.1 last night. now i am unable to run my agi-scripts from it. i always get: WARNING[30775]: pbx.c:3082 pbx_extension_helper: No application 'agi,clean/0.agi' for extension (soft-in, 1, 1) |
| 15:10.37 | wonderworld | asterisk agi dir is /var/lib/asterisk/agi-bin . scripts are in /var/lib/asterisk/agi-bin/clean |
| 15:11.52 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-167ee2a187bf24c0) |
| 15:11.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:11.59 | jaytee | tompaw, that example you posted would strip the first two digits of the dialed extension and the result would be 48 and whatever digits remained in ${EXTEN} |
| 15:12.42 | flohack | Has someone ever experienced a starving asterisk when using realtime queues? My box responds very slowly to queue applications (addqueuemember, remove, pause, unpause) as soon as a few callers are active in a queue. Messages on the AMI lag as much as half a MINUTE! |
| 15:12.44 | flohack | dynamic realtime that is |
| 15:12.45 | jaytee | so if you passed 5555 as an extension to that macro you'd get 4855 as a result. |
| 15:13.11 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-96-39.w86-215.abo.wanadoo.fr) |
| 15:14.38 | Katty | jaytee: indeed i did. |
| 15:14.45 | Katty | jaytee: blogged the whole kitnkaboodle. |
| 15:14.50 | Katty | jaytee: sadly, my blog's down )= |
| 15:14.57 | jaytee | :-( |
| 15:15.01 | Katty | but i might have it on the older blog... |
| 15:15.12 | Katty | dig |
| 15:15.46 | tompaw | jaytee: actually, it's working fine, I had an error somewhere else and I blame that string concatenation :-) |
| 15:15.50 | tompaw | thanks |
| 15:15.57 | jaytee | np |
| 15:16.32 | Katty | jaytee: hmm, no |
| 15:16.53 | jaytee | oh, well. |
| 15:16.55 | Katty | maybe a google cached version! |
| 15:18.32 | Katty | hmm no :< |
| 15:18.49 | Katty | but what did you wanna know about it? (= |
| 15:23.49 | *** part/#asterisk LARefugee (n=chatzill@c-76-104-191-194.hsd1.wa.comcast.net) |
| 15:24.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:26.13 | jaytee | Katty, I was just thinking of setting it up for a test drive and just wanted any real world examples I could find. I remembered seeing yours awhile back but I foolishly didn't bookmark the page. |
| 15:26.52 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 15:29.32 | PSU_Boss | hello, I can't seem to get asterisk started up again |
| 15:30.04 | PSU_Boss | it runs as root, but now won't start as the user asterisk |
| 15:30.06 | jaytee | is there gas in the tank? |
| 15:30.25 | PSU_Boss | yeah, it's 3/4 full |
| 15:30.34 | PSU_Boss | lol |
| 15:30.53 | jaytee | runs as root but not as asterisk. have you already set all the permission to allow that? |
| 15:31.09 | PSU_Boss | yeah |
| 15:31.35 | PSU_Boss | it was just running, and I was trying to get cdr_mysql working, and it didn't work after reloading asterisk |
| 15:31.50 | PSU_Boss | so i tried restarting the whole process, and it just won't start |
| 15:31.59 | jaytee | cdr_mysql didn't work or asterisk? |
| 15:32.07 | PSU_Boss | cdr_mysql |
| 15:32.31 | jaytee | did you try stopping and restarting the service for mysql before restarting asterisk? |
| 15:32.51 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 15:32.55 | PSU_Boss | yes |
| 15:33.40 | jaytee | are you running asterisk as a service? |
| 15:35.06 | jaytee | if you are try stopping the service and starting asterisk manually, asterisk -vvvvvvc and watch for errors when it loads it's modules. |
| 15:35.06 | PSU_Boss | it's running on gentoo, and i'm trying to start it using the initscripts |
| 15:35.36 | *** join/#asterisk sp00k3y (n=chatzill@wsip-98-190-136-194.ph.ph.cox.net) |
| 15:35.40 | jaytee | sorry, not a ricer so I run RHEL 5 64 |
| 15:36.04 | PSU_Boss | lol |
| 15:36.17 | PSU_Boss | well, it says permission denied opening log file |
| 15:36.27 | PSU_Boss | but it's chmodded to the asterisk user and group |
| 15:36.36 | PSU_Boss | *chowned |
| 15:36.41 | *** join/#asterisk bkw__ (n=brian@freeswitch/developer/bkw) |
| 15:36.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 15:37.30 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-178-7.phil.east.verizon.net) |
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| 15:38.02 | *** join/#asterisk bkw___ (n=brian@freeswitch/developer/bkw) |
| 15:39.41 | flohack | putnopvut: Have you got a minute for a channel deadlock issue on 1.4? |
| 15:40.00 | putnopvut | flohack: sure, what version of 1.4? |
| 15:40.10 | phpboy | How can I convert MP3 to asterisk format for music files? |
| 15:40.26 | flohack | putnopvut: 1.4.17, but from looking at the source, it seems to be present in the current version as well |
| 15:40.46 | flohack | putnopvut: I'll pastebin the details, hang on a seconds, please |
| 15:40.52 | putnopvut | flohack: all right. |
| 15:41.51 | flohack | putnopvut: http://pastebin.com/m2a80811e |
| 15:42.58 | De_Mon | do I need to restart the pbx if I make changes to asterisk.conf? |
| 15:43.43 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:43.49 | putnopvut | flohack: that's weird because the backtrace doesn't show any threads waiting on locks. |
| 15:43.51 | flohack | putnopvut: Thread 3 entered the poll with -1 for the timeout (infinite) and the set of FD contain UDP sockets as well as alertpipe and timing? pipe. This happens when both SIP endpoints disappear. The patch makes sure the timeout is never set to -1 (more of a workaround, as I'm not sure what the pipes are for, maybe they should make sure that it wakes up again) |
| 15:44.26 | flohack | putnopvut: It's not a deadlock per definition, more an infinite sleep :-) |
| 15:44.33 | flohack | putnopvut: sorry for the misnomer |
| 15:44.42 | putnopvut | Ah, gotcha. |
| 15:44.58 | De_Mon | oh, err the setting isn't where I thought it was, ignore that question |
| 15:46.11 | jaytee | PSU_Boss, who is the owner of the asterisk directory under /var/logs ? did you only set permission on the file itself? |
| 15:46.40 | jaytee | PSU_Boss, or is it referring to a mysql log? |
| 15:47.34 | jaytee | gotta grab some lunch. bbiab |
| 15:48.22 | Katty | oh, lunch |
| 15:48.24 | Katty | there's a thought |
| 15:48.34 | flohack | putnopvut: To me it looks as if ast_bridge_config->timelimit should be there to make sure the bridge does not wait forever, however I have not found any code which sets it to something != 0 |
| 15:48.49 | *** join/#asterisk oej_ (n=olle@90-84-254-204.ip.sipit.net) |
| 15:48.53 | flohack | putnopvut: or a configurable value |
| 15:48.59 | *** part/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 15:49.34 | putnopvut | flohack: if memory serves me correctly, the L option to Dial is one way to actually set the timelimit. |
| 15:50.55 | putnopvut | flohack: I think this issue you're reporting is a bit more involved than a fix-it-over-IRC thing. If you don't mind, could you please open a bug on bugs.digium.com relating to this? |
| 15:51.54 | flohack | putnopvut: sure |
| 15:52.34 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 15:53.40 | flohack | putnopvut: On an unrelated case: Have you ever seen asterisk starving on dynamic realtime queue application (add, remove, pause, unpausequeuemember) under heavy load? My system takes sevel seconds, up to half a minute to respond. AMI messages are delayed as well. However dialing voicemain for examples works like a charm. |
| 15:54.24 | putnopvut | flohack: I haven't seen that, no. |
| 15:55.35 | flohack | putnopvut: Thanks! |
| 15:56.03 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 15:56.32 | *** join/#asterisk sp00k3y (n=chatzill@wsip-98-190-136-194.ph.ph.cox.net) |
| 15:57.36 | *** join/#asterisk ayrjola (n=ayrjola@gw.aller.fi) |
| 15:59.55 | ayrjola | Hello, need some help. I have problem transfering call from deskphone out to sip trunk |
| 16:00.01 | *** join/#asterisk rigid (n=dude@port-83-236-2-93.dynamic.qsc.de) |
| 16:00.02 | ayrjola | WARNING[17865]: chan_sip.c:12389 handle_response_invite: Received response: "Forbidden" from '"73" <sip:73@123.123.123.123>;tag=as5f047536' |
| 16:00.06 | rigid | hey |
| 16:00.29 | dandre | hello, |
| 16:00.47 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 16:00.53 | rigid | can someone give me the name or a keyword for a configuration that rings multiple phones/extensions when one sip-number is called? until one of the extensions answers... |
| 16:00.59 | dandre | is there in asterisk 1.6 a manager command to put a channel on hold ? |
| 16:01.12 | rigid | i thought this would be called "ringdown" but google tells i'm wrong :) |
| 16:02.28 | _ShrikE | rigid: Dial(SIP/phone1&SIP/phone2&SIP/phone3......) |
| 16:02.34 | ayrjola | rigid, if Dial() |
| 16:02.40 | dandre | rigid: you can just dial all your extensions separated with & |
| 16:02.55 | ayrjola | sorry typo just Dial() :) |
| 16:03.55 | rigid | _ShrikE: ayrjola dandre: ah, simply by script... tnx... i'll try to weazle it out |
| 16:04.34 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:05.31 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:06.14 | dandre | I don't understand why I can't find (in docs, google, ;..) a working solution to put a channel on hold thru the manager interface as if it were from a standard sip phone. This is a basic feature in all callcenter application or crm integration. |
| 16:08.34 | PSU_Boss | jaytee: the owner and group of the /var/log/asterisk directory is asterisk:asterisk |
| 16:08.45 | PSU_Boss | and i even set the permissions of the files to 777 |
| 16:10.46 | ayrjola | dandre: if you find solution to make hold through AMI I would like to know :) |
| 16:11.08 | dandre | I haven't found |
| 16:11.23 | dandre | this must existe somewhere |
| 16:11.46 | ayrjola | I hope :) |
| 16:11.48 | dandre | someone must have worked on it |
| 16:12.05 | dandre | I don't beleive this is not true |
| 16:12.47 | dandre | in the last extend I will hav to patch asterisk manager.c code :-( |
| 16:13.06 | dandre | but this will take long long time |
| 16:13.53 | ayrjola | I gave up, my C skills are too rusted for that :) |
| 16:14.26 | dandre | you tried something? |
| 16:14.51 | ayrjola | just tried to read the code |
| 16:15.14 | dandre | ok |
| 16:15.47 | ayrjola | I think you can write your own module that ami loads, so maybe you dont need to stab manager.c |
| 16:16.55 | dandre | I don't kwon |
| 16:19.02 | ayrjola | Any help for my transfer problem? |
| 16:20.58 | *** join/#asterisk sp00k3y (n=chatzill@wsip-98-190-136-194.ph.ph.cox.net) |
| 16:28.45 | *** join/#asterisk feeds (n=feeds@85-135-230-80.adsl.slovanet.sk) |
| 16:30.03 | *** join/#asterisk angryuser (n=Miranda@199.208.20.81.dynamic.adsl.abo.nordnet.fr) |
| 16:30.04 | nikko | is there an asterisk based channel centered on running a voip business? |
| 16:30.12 | *** join/#asterisk Rambaldi (n=rambaldi@cl-1188.ams-05.nl.sixxs.net) |
| 16:30.28 | Qwell | nikko: "voip business" is a bit broad |
| 16:30.48 | angryuser | nikko : you have mailing list's asterisk-biz |
| 16:31.09 | flohack | putnopvut: Shall I report it in private? Could be considered a DoS... |
| 16:31.15 | nikko | yeah, maybe so. mostly asterisk configuration for multi-customer usage, like dialplan, security, etc |
| 16:31.34 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 16:31.53 | Ritzerisk | anyknow know of a good sip provider to get a local number |
| 16:32.21 | nikko | angryuser - thanks - I'l hang here for tech and monitor asterisk-biz for business related stuff |
| 16:33.11 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:35.42 | *** join/#asterisk ManxPower (n=manxpowe@86.sub-75-203-203.myvzw.com) |
| 16:36.17 | *** join/#asterisk sp00k3y (n=chatzill@wsip-98-190-136-194.ph.ph.cox.net) |
| 16:36.43 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 16:41.07 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-15-2.revip2.asianet.co.th) |
| 16:41.09 | flohack | putnopvut: The bug is here: http://bugs.digium.com/view.php?id=13681 |
| 16:45.29 | jaytee | anyone ever see an incoming call over PRI come in with a callerid of 10 zeros? |
| 16:48.03 | *** join/#asterisk MrNaz (n=mrnaz@58.185.105.2) |
| 16:49.15 | ta^3 | I've an Asterisk 1.4.22/SVN-r148257 in a deadlock condition, where lots of AgentCallBackLogin() get stuck trying to lock because another thread do not release their p->lock at hangup. |
| 16:50.27 | booray | exit |
| 16:50.29 | booray | ah, shit |
| 16:51.53 | *** join/#asterisk steliosk (n=Stelios@79.107.27.111) |
| 16:54.10 | carrar | jaytee, google it |
| 16:58.08 | jaytee | carrar, did. looks like telemarketers trying to get past 800 number filtering or callerid blocking |
| 16:58.33 | jaytee | I'll just route all calls like that to my "queue from Hell" |
| 17:01.15 | ManxPower | You could always just Congestion those calls or run Zapatateller |
| 17:05.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:06.48 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 17:08.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.27 | ta^3 | I've submited bug #13676 regard that, I'm able to reproduce the deadlock condition over and over again. |
| 17:12.12 | wonderworld | i want to initiate calls via the /var/spool/asterisk/outgoing directory. the problem is, that the script that creates the callfiles is not run by user asterisk. is there a smart way to change the owner of the files to asterisk without actually setting chmod suid root? |
| 17:13.34 | Qwell | just chown it to asterisk |
| 17:13.49 | Qwell | oh, you mean the dir.. add it to a group with access |
| 17:14.59 | wonderworld | i tried to chown it to asterisk. gives me permission denied |
| 17:15.08 | wonderworld | i created the call-file in /tmp |
| 17:17.44 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.35.45) |
| 17:19.47 | mocker | tries to troubleshoot why his asterisk server is failing twice a week. |
| 17:20.21 | mocker | Seeing 'Failed to write frame' and 'Unable to forward voice frame' quite a bit in the logs. |
| 17:20.53 | ManxPower | mocker: that is almost always "far end hungup" |
| 17:21.01 | mocker | Hmm. |
| 17:21.04 | mocker | Both of them? |
| 17:21.23 | mocker | I wish I could duplicate the problem, it seems to happen randomly. |
| 17:21.36 | ManxPower | mocker: there can only be one far end, dude. |
| 17:21.45 | XnOSX | hello friends one question! asterisk and zaptel last version its compatible with kernel 2.6.24? |
| 17:21.47 | mocker | ManxPower: No, both the errors. :) |
| 17:21.55 | ManxPower | mocker: call someone, tell them to hang up, test complete |
| 17:25.14 | mocker | ManxPower: Hmm, did that several times and no error. |
| 17:25.27 | ManxPower | mocker: are people complaining? |
| 17:26.08 | *** join/#asterisk CrazyTux (n=brandon@nmd.sbx08607.gardeca.wayport.net) |
| 17:26.27 | mocker | ManxPower: Yeah, when it dies I can no longer make outbound calls. |
| 17:26.43 | mocker | Asterisk is still up, but outbound calls just hang. |
| 17:26.59 | ManxPower | I doubt those messages are the cause of the issue. |
| 17:27.16 | ManxPower | what version of Asterisk, mocker? |
| 17:27.40 | mocker | Asterisk 1.4.21 |
| 17:28.35 | ManxPower | mocker: that is not the latest |
| 17:29.25 | mocker | .22 is.. |
| 17:29.28 | mocker | I'll check the changelog |
| 17:33.06 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
| 17:33.52 | FruitBasket | I have a bunch of phones (aastrax3, grandstreamx1) that just _will_not_ register with asterisk. I'm watching the packets go across the router, and they're frequent.. but nothing whatsoever shows up in the asterisk console.. |
| 17:34.17 | FruitBasket | I don't really wanna restart asterisk, but it seems like it _must_ be the server. any thoughts? |
| 17:35.02 | mocker | ManxPower: Can't hurt to try upgrading. |
| 17:35.37 | mocker | Especially before spending time trying to find a bug that may already be fixed. :) |
| 17:36.11 | Lorax | FruitBasket: nothing in the logs? |
| 17:38.21 | FruitBasket | lorax: nothing. The only thing I can think of is I changed the route to the phone server last night (i.e. 10 hours ago), and the sip box might be responding to the wrong host... though the sip ID's are unregistered. |
| 17:40.09 | Lorax | try flushing the arp tables |
| 17:40.21 | Lorax | (if the gateway has changed) |
| 17:40.32 | FruitBasket | not for the server, only the phones |
| 17:40.36 | FruitBasket | gateway is the same |
| 17:43.34 | *** join/#asterisk scrash08 (n=scrash08@unaffiliated/scrash08) |
| 17:44.21 | scrash08 | Hi. Following available v14 docs, I've built & installed release branches of Asterisk, Asterisk-GUI, etc on OpenSuse. Asterisk starts, and I've registered a Sip trunk -- which works for in/outbound calls! |
| 17:44.29 | scrash08 | The Asterisk-GUI's available, and everything seems to work -- except whtn I click "System Info" tab, I get the webpage displayer with 4 tabs -- General, Network, Disk Usage, Memory Usage -- but no info displayed at all. Just blank. |
| 17:44.32 | mocker | also increases logging in logger.conf. |
| 17:44.47 | scrash08 | Is there some 'magic' I've missed to enable it? |
| 17:45.28 | ManxPower | scrash08: try the Asterisk GUI channel. We don't generally use GUIs here. |
| 17:45.54 | scrash08 | ManxPower: Ah, sorry. Didn't even realize there was one! Thx. |
| 17:46.19 | ManxPower | scrash08: look at the /topic when you join channels. It may contain important information for you |
| 17:46.20 | FruitBasket | oddly enough, I find that by unplugging the phone for 5 minutes and plugging it back in, it starts working just fine. |
| 17:46.37 | *** part/#asterisk scrash08 (n=scrash08@unaffiliated/scrash08) |
| 17:47.01 | FruitBasket | It's as though Asterisk doesn't immediately pick up on the change in IP address of the phone -- the phone acting like it's still registered. Still, it seems like the frequent registration attempts prevent the new IP from being seen... |
| 17:47.44 | Ritzerisk | anyknow know of a good sip provider to get a local number |
| 17:48.14 | FruitBasket | vitelity isn't bad. Our colo facility has problems with them, though -- high ping for 5-10 minutes every other day or so. |
| 17:51.55 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:52.54 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 17:53.53 | AlexTO | someone familiar with autodial? |
| 17:54.21 | ManxPower | AlexTO: it's not really a term we use here. |
| 17:55.26 | FruitBasket | it is.... Asterisk is sending data to one of our two IP's. But that route hasn't been active for 10 hours.... so I don't know what Asterisk is responding to. |
| 17:55.28 | AlexTO | sorry, ManxPower i have some questions about autodial that maybe around here can help me to understand |
| 17:55.43 | ManxPower | AlexTO: do you mean .spool files, the Redial application or something else? |
| 17:55.55 | ManxPower | AlexTO: have you read the Asterisk book yet? |
| 17:55.57 | AlexTO | yes, |
| 17:56.08 | ManxPower | What term does the book use for this feature? |
| 17:56.11 | AlexTO | yes, i did, |
| 17:57.09 | mocker | Ritzerisk: I've had great luck w/ vitelity |
| 17:57.16 | mocker | ,itsp |
| 17:57.26 | ManxPower | AlexTO: what are you trying to accomplish? |
| 17:57.28 | mocker | itsp? |
| 17:57.42 | mocker | fails to get jbot to respond |
| 17:57.47 | ManxPower | I also recommend Vitelity and Teliax |
| 17:57.50 | ManxPower | ~itsp |
| 17:57.51 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:57.52 | AlexTO | basically my question is about CDRs when you make outgoing call |
| 17:58.22 | AlexTO | i could make it dial te file |
| 17:58.25 | ManxPower | AlexTO: Ah. I can't help you with that. That's a CDR rather than a dial issue. |
| 17:59.30 | AlexTO | my question is wich variable should i use catch the original number that call |
| 18:00.00 | AlexTO | manxPower can you point me in the right direction to find more info about it? |
| 18:00.11 | ManxPower | AlexTO: I don't even understand what you want. Perhaps someone else can help you. |
| 18:00.51 | *** part/#asterisk Rambaldi (n=rambaldi@cl-1188.ams-05.nl.sixxs.net) |
| 18:00.56 | ManxPower | AlexTO: as always the official Asterisk docs are in the "doc" directory in the Asterisk source code. |
| 18:01.48 | *** join/#asterisk flohack (n=fhackenb@91-115-126-173.adsl.highway.telekom.at) |
| 18:02.34 | AlexTO | OKi, basically what happend is the call is made and then it is send to the context that you choose, but the original number is not storage an any variable |
| 18:02.47 | AlexTO | i'll check the code... |
| 18:04.25 | *** join/#asterisk RAiDENZ (n=raiden@205-200-66-136.static.mts.net) |
| 18:05.10 | RAiDENZ | Hi guys |
| 18:06.29 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 18:06.42 | RAiDENZ | How do you replace single characters in a string in the asterisk extensions.conf file. I can GET single characters from a string(substring) but I dont know how to replace a single character in a string. Any ideas? |
| 18:12.54 | *** join/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 18:13.34 | *** join/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 18:19.40 | *** join/#asterisk metfan2007 (n=jc@189.135.127.11) |
| 18:20.39 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 18:21.06 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 18:21.24 | metfan2007 | Hi all! I'm doing a link between Asterisk and a Cisco SIP gateway, I can establish the calls, but the problem is that the time between Asterisk sends de Dial to Cisco, and Cisco answers is too long, about 20 seconds, the ping is Ok, any help? |
| 18:21.56 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-127-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:22.22 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:22.48 | metfan2007 | I hav a tcpdump file for reference... |
| 18:23.05 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:25.44 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:26.16 | metfan2007 | any SIP guy around here? xD |
| 18:28.13 | DarKnesS_WolF | metfan2007: both are in the same lan ? |
| 18:28.33 | jes-o-ma1 | metfan2007: not at all |
| 18:28.55 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:29.03 | metfan2007 | DarKnesS_WolF: same lan, different segments |
| 18:30.18 | DarKnesS_WolF | metfan2007: VLAN ? |
| 18:31.53 | metfan2007 | DarKnesS_WolF: the issue here is that only one step over the SIP negociation is taking a lot of time, the other steps are normal |
| 18:32.09 | metfan2007 | DarKnesS_WolF: do you want to see the cap file? |
| 18:33.41 | DarKnesS_WolF | metfan2007: sure show me i am not that good at sip but i might help |
| 18:33.42 | DarKnesS_WolF | show me |
| 18:34.08 | *** join/#asterisk flohack (n=fhackenb@212-183-82-189.adsl.highway.telekom.at) |
| 18:35.43 | metfan2007 | DarKnesS_WolF: you can see it here: http://pastebin.ca/1226079 |
| 18:35.55 | metfan2007 | DarKnesS_WolF: check the times in the left side |
| 18:37.31 | RAiDENZ | \quit |
| 18:37.57 | DarKnesS_WolF | metfan2007: yes steop 7 and 9 |
| 18:37.59 | DarKnesS_WolF | and 8 |
| 18:38.00 | DarKnesS_WolF | mmmm |
| 18:38.08 | DarKnesS_WolF | did u try both in same range of IP ?? |
| 18:38.17 | DarKnesS_WolF | maybe routing issue some how ? |
| 18:38.37 | _ShrikE | metfan2007: It would be good to see sip debug from asterisk as well |
| 18:40.10 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 18:44.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:45.21 | *** join/#asterisk itiliti (n=itiliti@75.150.198.1) |
| 18:45.54 | metfan2007 | _ShrikE: you can see it here: http://pastebin.ca/1226095 |
| 18:47.10 | itiliti | afternoon all. I am using exten => 900,1,Dial(console/dsp,12,A(ring-raining-12)) to play a sound file out of our Soundcard. We just upgraded to 1.4, and now the 900 extension is answering, and broadcasting the calling party over the3 speaker as if it is a page. THE CLI says that it is answered bu 900. How can I make the 900 extension not answer the call. |
| 18:48.18 | itiliti | I have even tried it using exten => 900,1,Dial(console/dsp,,A(ring-raining-12)) |
| 18:48.31 | itiliti | but I have it as part of a ring group so that phones will ring as well. |
| 18:49.33 | _ShrikE | metfan2007: Looks like your cisco does not know what to do with 4587901134913667449# |
| 18:50.23 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 18:59.09 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:59.42 | *** join/#asterisk steliosk (n=Stelios@79.107.27.111) |
| 19:02.35 | GlobeTrotter | hello,, i need some help with dahdi.. i have it installed on 1.6, i have all the settings in system.conf and init.conf.. but what setting do i need to get my channels on my 4 port pstn card working? |
| 19:06.13 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 19:07.11 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 19:07.58 | justdave | hmm, looks like if you encrypt an IAX trunk, the receiving system can't tell when the sending system has terminated the call. |
| 19:08.21 | justdave | get about 30 seconds of "Packet Decrypt Failed!" before it finally gives up and hangs up the call. |
| 19:08.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:08.25 | *** join/#asterisk sp00k3y (n=chatzill@ip68-3-117-115.ph.ph.cox.net) |
| 19:11.42 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 19:13.15 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 19:13.48 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:15.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:15.14 | thansen | is it possible to have asterisk receive faxes? |
| 19:16.07 | thehar | sure is |
| 19:16.41 | thansen | is it possible to have someone call into a number (with fax machine) enter some numbers, then have it receive the fax data? |
| 19:18.41 | thehar | http://www.voip-info.org/wiki-Asterisk+fax |
| 19:19.16 | *** join/#asterisk zydoon (n=zydoon@41.225.155.64) |
| 19:19.35 | *** part/#asterisk zydoon (n=zydoon@41.225.155.64) |
| 19:22.28 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 19:23.06 | Blackvel | anyone with the snom 370 big display? |
| 19:23.18 | Blackvel | how can I get into the missed call detail view? |
| 19:23.36 | Blackvel | i am pressing "listen" btn, missed, and get a list |
| 19:23.42 | Blackvel | but the text is too small |
| 19:24.39 | apocn | http://pastebin.com/d307047b8 -> having 6 calls in the queue and 2 agents available, all clients are getting stucked in the queue. Any hints? |
| 19:24.50 | Blackvel | when I press the ok button (right besides big scroll button), i am not getting into the detail view (which hopefully displays the callerid(name) text much bigger and does not cut off...but just dails the callerid(num) |
| 19:25.16 | Blackvel | but I can remember that I pressed around and got days ago into the missed call detail view |
| 19:26.12 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:26.40 | tuxfoo21 | Does any know how to get asterisk to voicemail to integrate with exchange in order to delete listen to email from exchange without having to manually delete messages? Is this even possible? |
| 19:26.41 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:27.47 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:28.36 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:29.44 | _ShrikE | http://www.pomegranatephone.com/ |
| 19:30.31 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:30.40 | creativx | tuxfoo21: possible, but you probably have to write the outlook addin / folder event sinks yourself and use AMI -> asterisk. |
| 19:32.16 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 19:33.02 | khronos | Hi, how can I set the codec for an outgoing call to whatever is listed in sip.conf for a peer? |
| 19:33.09 | khronos | Until now I've been setting the codec like: |
| 19:33.59 | khronos | Set(SIP_codec=codec_name), but what I went to do is somehow refer to the sip.conf setting for each cleint ot have the system change the codec on the fly so that the outgoing call has the same codec as what the client talks to my server with. |
| 19:34.53 | khronos | This way if I have a client that uses gsm any calls the client makes / receives will also be in gsm instead of having to convert to ulaw or some other codec. |
| 19:35.33 | khronos | I've got the inbound dialing codec working by setting the codec to what I want before the dial statement to the peer. |
| 19:39.49 | jaytee | tuxfoo21, not sure exactly what your asking to do. If you want to call Exchange and have it read your emails to you then you need to try something like this which is what I'm using. http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html |
| 19:44.38 | thansen | thehar: I can't really tell if it's possible to send dtmf data before sending the fax data...is this possible? |
| 19:46.31 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:54.32 | *** join/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 19:55.20 | *** join/#asterisk rabbit7 (i=rabbit7@stat.siff.org) |
| 19:56.08 | *** part/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 19:58.36 | rabbit7 | hey there i have a smartnode connected to my asterisk, there are lots of telephone number terminated on my smartnode. Do i need to setup a trunk for each number which should be passed to my asterisk or is there another way to do that setup ? |
| 19:59.08 | Qwell | jameswf: |
| 19:59.12 | Qwell | http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ |
| 19:59.27 | Qwell | spread teh love |
| 20:01.41 | *** join/#asterisk ManxPower (n=manxpowe@86.sub-75-203-203.myvzw.com) |
| 20:02.14 | hardwire | rabbit7: smartnode through who? |
| 20:02.30 | *** part/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 20:02.31 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:03.10 | creativx | jaytee: interesting article.. how have you implemented it with asterisk? |
| 20:03.25 | jaytee | creativx, yes |
| 20:03.45 | jaytee | all our users voicemail boxes are on Exchange 2007 UM. |
| 20:04.44 | jaytee | I'm still running * 1.4 though so I have to use sipX as a upd/tcp proxy but once I've tested 1.6 and feel comfortable I'll upgrade to that and configure a direct connection to Exchange from Asterisk |
| 20:04.46 | creativx | jaytee: i might have to look into this i understand.. hehe |
| 20:06.22 | mvanbaak | lesouvage: didn't see you in Phoenix ... |
| 20:06.25 | jaytee | Exchange UM will allow you to call in from inside or outside and play your voicemail and read your email as well as check your calendar, reschedule meetings and it also has an autoattendant feature for dial by name with Voice Recognition to dial other users. It'll do a refer to redirect the call |
| 20:08.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:09.35 | hardwire | http://www.patton.com/products/pe_products.asp?category=358 |
| 20:09.38 | hardwire | who still uses isdn? |
| 20:09.44 | hardwire | like.. 128kbps isdn. |
| 20:10.47 | ManxPower | hardwire: you mean ISDN BRI. Most of Europe uses it (for voice) |
| 20:10.55 | Qwell | jameswf: woot |
| 20:11.19 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 20:12.33 | mwalling | anyone want to read a sip debug dump and tell me why i feel like a total idoit (aside from the fact that i am |
| 20:12.36 | mwalling | ) |
| 20:12.44 | mwalling | damn enter key.... anyway: http://files.markwalling.org/sipdump.txt |
| 20:13.56 | mwalling | near the end is where i try calling the house... it seems like the ATA isnt answering the incoming sip messages |
| 20:15.54 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:19.40 | ManxPower | BTW, anyone want some Blonder Tongue channel converters or modulators just /msg me |
| 20:19.48 | DogWater | Anyone know of any software that will generate somewhat decent voice prompts for an IVR? |
| 20:20.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:20.33 | _ShrikE | DogWater: I have used a windows app called NaturalReader that is not too bad. |
| 20:21.24 | *** join/#asterisk sp00k3y (n=ikkyu@wsip-98-190-136-194.ph.ph.cox.net) |
| 20:22.12 | DogWater | Thanks. |
| 20:22.17 | rabbit7 | hardwire: the smartnode is a patton which has bri to my NT box on one side and a sip gateway on the other side |
| 20:27.20 | tuxfoo21 | Does any know how to get asterisk to voicemail to integrate with exchange in order have exchange/outlook delete it after has been listened to? I do not want to have to manually delete messages via the ip phone? Is this even possible? |
| 20:31.49 | mocker | tuxfoo21: Maybe imap voicemail? |
| 20:34.23 | tuxfoo21 | Hmm |
| 20:34.30 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 20:37.07 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:38.59 | shriven | Hello, I'm having a bit of trouble with voicemail configuration. I am using IMAP_STORAGE, fyi. |
| 20:39.39 | shriven | I have a voicemail line set up like so: 6000 => 1234,Brendan's Mailbox,,,imapuser=brendanmartens@crosscomm.net|imappassword=vasferas!1983 |
| 20:39.40 | shriven | 3216000 => d,Brendan Martens |
| 20:39.51 | shriven | hmmm.... wanted a new line there.... |
| 20:40.18 | shriven | and my voicemail estension is set like this: |
| 20:40.41 | shriven | exten => 700,1,VoiceMailMain() |
| 20:40.56 | shriven | calling it seems to work fine, but it doesn't authenticate me properly, I get this in the console |
| 20:41.12 | ManxPower | shriven: I suuggest Voicemailmain(@thevmconfcontext) |
| 20:41.23 | shriven | yeah, did that too |
| 20:41.26 | shriven | exact same issues |
| 20:41.40 | shriven | -- Incorrect password '1234' for user '6000' (context = default) |
| 20:41.41 | ManxPower | maybe the ! is the problem? |
| 20:41.48 | shriven | no, that's just for email |
| 20:41.54 | shriven | and as far as storing it it works fine |
| 20:41.56 | shriven | it's just accessing it |
| 20:42.19 | shriven | I mean to say, that password is just for the imap user storage, not for accessing the voicemail I have via phone |
| 20:43.57 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 20:44.25 | devilsoulblack | hi any one know how rsync E1 ISDN PRI |
| 20:44.28 | thansen | if I want to receive faxes with 1.6.0 do I just spandsp installed? |
| 20:45.18 | ManxPower | devilsoulblack: rsync does not work with PRI |
| 20:46.17 | devilsoulblack | i recieve to much this msg "iming source auto card 0!" and minutes later kernel panic, the telco carrier tell me this ist about un sync from the isdn pri over asterisk |
| 20:46.19 | ManxPower | see http://freshmeat.net/redir/rsync/9147/url_homepage/rsync.samba.org |
| 20:47.11 | ManxPower | devilsoulblack: that has NOTHING to do with rsync. Make sure you have a "1" in the 2nd field of the span= line that the telco is connected to in zapata.conf |
| 20:47.25 | mwalling | r*e*sync? |
| 20:47.27 | ManxPower | that tells Asterisk to take it's line sync from the telco |
| 20:47.48 | ManxPower | mwalling: maybe that's what he meant but I'm not a psychic. |
| 20:48.21 | mwalling | neither am i... until you pasted the rsync(1) link, i thought rsync was some isdn foopala |
| 20:48.46 | ManxPower | Since he is saying the kernel is panicking I strongly doubt any config changes will fix it. |
| 20:50.18 | ManxPower | devilsoulblack: make sure you have the latest version of Asterisk and Zaptel and libpri |
| 20:50.24 | devilsoulblack | ManxPower, this ist my zapata http://pastebin.ca/1226225 and this my zaptel http://pastebin.ca/1226226 |
| 20:51.10 | ManxPower | devilsoulblack: Your config is correct. |
| 20:51.29 | devilsoulblack | but still get that msg "timing source auto card 0!" |
| 20:51.57 | ManxPower | devilsoulblack: Correct. That message is not being caused by a config problem. It is caused by some OTHER problem. |
| 20:53.00 | *** join/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 20:53.24 | ManxPower | devilsoulblack: what version of asterisk/libpri/zaptel are you using? |
| 20:53.55 | devilsoulblack | the telco carrier tell me this ist problem of asterisk because need sync isdn pri over asterisk or asterisk over isdn pri |
| 20:54.38 | ManxPower | devilsoulblack: I know it is shocking, but your telco is wrong. What version of asterisk/libpri/zaptel are you using? |
| 20:55.23 | ManxPower | devilsoulblack: Sync problems do not cause kernel panics. |
| 20:56.41 | devilsoulblack | asterisk-1.4.21.1 / libpri-1.4.4 / zaptel-1.4.11 |
| 20:56.59 | devilsoulblack | ManxPower, you know how sync over asteisk and isdn pri ? |
| 20:57.06 | ManxPower | devilsoulblack: before you do anything else, upgrade to the latest versions of all those packages. |
| 20:57.16 | *** join/#asterisk sp00k3y (n=billmcme@wsip-98-190-136-194.ph.ph.cox.net) |
| 20:57.17 | rigid | i'm getting "chan_sip.c:11746 do_monitor: Disconnecting call 'SIP/sip.fubar-09e7e338' for lack of RTP activity in 11 seconds" |
| 20:57.38 | rigid | is that related to the codecs? i have no clue what could be wrong/how to debug :-/ |
| 20:57.48 | ManxPower | devilsoulblack: I cannot help you further. |
| 20:58.10 | ManxPower | rigid: sounds like SIP/sip.fubar is using silence supression |
| 20:58.13 | devilsoulblack | that box ist been ok about 16 weeks |
| 20:58.27 | ManxPower | <PROTECTED> |
| 20:58.39 | Katty | ahieeeeee |
| 20:58.47 | jaytee | it's a pity that the linux kernel doesn't come with a libxanax.so module that just kicks in when things go "tits up" and just calms the kernel down. |
| 20:59.03 | rigid | ManxPower: but i and the caller talk... so there should be no silence... or is that a known problem w/silence supr.? |
| 20:59.04 | Katty | asplodes |
| 20:59.29 | ManxPower | rigid: RTP is VoIP audio. If you did not receive RTP for 11 seconds than you did not receive audio for 11 seconds. |
| 20:59.50 | *** join/#asterisk mazpe (n=mazpe@adsl-065-006-163-191.sip.mia.bellsouth.net) |
| 21:00.09 | ManxPower | and not sending audio in the middle of a call us almost always a silence supression / voice activity detection |
| 21:00.27 | mazpe | anyone recommends a service provider like voicepulse.com that works well with asterisk? |
| 21:00.38 | mazpe | more like connect.voicepulse.com |
| 21:00.40 | ManxPower | ~itsp |
| 21:00.41 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:01.06 | mazpe | ~itsplist-us |
| 21:01.07 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 21:01.24 | jaytee | Grandstreams with earlier versions of the firmware would not turn off silence suppression even though it indicated it was. That drove me crazy till I got it fixed. Thank Cthulu I now use Polycoms |
| 21:01.51 | mazpe | voicepulse wont even answer their phones and international calls have very poor quality... odd. |
| 21:01.53 | ManxPower | grandstream products are a piece of shit |
| 21:02.04 | mocker | ~grandstream |
| 21:02.04 | jbot | from memory, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 21:02.21 | jaytee | ManxPower, yes and I finally won that arguement with my boss |
| 21:02.50 | jaytee | so we don't buy anything but Polycom phones and Linksys ATAs. |
| 21:03.13 | itiliti | anyone know the best way to play a sound file out of the soundcard to a paging system?1.4 is handling ou existing dial plan diffrently then 1.2 did. |
| 21:03.36 | itiliti | exten => 900,1,Dial(console/dsp,,A(ring-raining),i) |
| 21:04.05 | mazpe | grandstream ata used to be pretty good a coulpe of years ago... not the case anymore? |
| 21:04.07 | mazpe | and cheap |
| 21:04.27 | jaytee | mazpe, they were cheap a couple years ago but they've never been good |
| 21:04.54 | *** join/#asterisk javb (n=javb@190.166.108.29) |
| 21:05.20 | jaytee | unless you're a big fan of jitter, echo, net connection dropping in and out, etc. |
| 21:06.08 | mazpe | not quite ;) |
| 21:06.35 | jaytee | The use of Grandstream equipement should be reserved solely for child molesters and people that talk at the theater. |
| 21:06.57 | javb | Hello guys, i've two PBXs (pbx1 and pbx2), Two IP-Phones (Polycom 330), first, the two phones were registrared to the pbx1; no problem. Now, i change the sip outbount proxy in both phone to be pbx2, and, STILL registrating to pbx1, and they are even able to call eachother. (Notes: sip.conf checked; ip address cheked) ANY IDEA ? |
| 21:07.16 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:08.20 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 21:08.27 | javb | Now i have one phone in the pbx2 and another in pbx1, but BOTH appear registrared in pbx1, the one in pbx1 can call the other "poiting" to pbx2. but the one "registrared" to pbax2, CANT place a call |
| 21:08.46 | jaytee | that sounds right |
| 21:09.16 | javb | jaytee? |
| 21:09.20 | apocn | Trying to register to my sip proxy I get the error "423 Interval Too Brief". I tried applying this patch http://bugs.digium.com/view.php?id=7254 but it doesnt compile (using Asterisk 1.4.22). |
| 21:09.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:09.36 | jaytee | the outbound proxy settings don't determine where the phone registers to |
| 21:11.33 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:11.47 | jaytee | and if you've changed the registration server for one of the phones so it registers to pbx2 and the other is registered to pbx1 you can only call from the phone on pbx1 to pbx2 because it's your phone's outbound proxy. |
| 21:12.34 | lesouvage | Does any of you have good or bad experiences with FreeTDS for Asterisk CDR Storage to MS Sql Server? |
| 21:14.10 | seanbright | lesouvage: we use it in production in asterisk 1.4 and don't have a problem |
| 21:16.30 | seanbright | but asterisk 1.4 will only work with FreeTDS 0.64 and under |
| 21:16.42 | seanbright | if you need FreeTDS 0.82 or higher support you need to use 1.6.0 |
| 21:18.49 | lesouvage | seanbright: the costumer runs 1.4.18.1 so I guess FreeTDS 0.64 is the choice to go. |
| 21:19.06 | seanbright | lesouvage: indeed. |
| 21:19.16 | seanbright | lesouvage: that is what we run and it works well. |
| 21:19.40 | jaytee | quittin time, be back later |
| 21:22.41 | lesouvage | I'm reading http://www.voip-info.org/wiki/view/FreeTDS . Is it really just this and it is up and running? (the TDS cdr_tds.c scenario) |
| 21:23.22 | mazpe | anyone using broadvoice? i'm trying to make up my mind about a provider.. |
| 21:23.42 | mazpe | voicepulse just giving horrible international audio quality.. and support wont even pick up the phone now. |
| 21:23.43 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 21:24.19 | javb | I've found that the pbx2 is NOT listening to SIP ... I ping the phones from the central, but the phone CANT register to the central, SIP DEBUG doesnt show anything, any idea? |
| 21:24.22 | seanbright | lesouvage: basically. that page has a lot of unnecessary crap on it. |
| 21:24.34 | seanbright | lesouvage: you don't need unixodbc to use freetds with asterisk |
| 21:26.10 | seanbright | you install freetds, re-run asterisk's ./configure, make sure cdr_tds is selected in menuselect, rebuild |
| 21:26.26 | seanbright | take a look at /usr/etc/freetds.conf, setup your connection there |
| 21:26.49 | seanbright | then update /etc/asterisk/cdr_tds.conf to reflect the stuff in /usr/etc/freetds.conf |
| 21:26.54 | seanbright | bing bam boom, you're done. |
| 21:27.39 | *** join/#asterisk DarkRift (n=dark@65.92.169.223) |
| 21:30.05 | mocker | seanbright: Mind if I quote you on that wiki page? |
| 21:30.14 | mocker | :) |
| 21:30.23 | lesouvage | seanbright: thanks a lot. I will ask them to do add the Asterisk cdr MS SQL table/database to the Windows server and then I can fix the Asterisk server. |
| 21:31.24 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 21:31.24 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 21:31.51 | codefreeze-lap | seanbright: about your freetds usage: how many cdrs/day do you log? |
| 21:33.13 | seanbright | codefreeze-lap: a few thousand |
| 21:33.42 | codefreeze-lap | seanbright: are ya using the odbc stuff? |
| 21:33.47 | seanbright | codefreeze-lap: i tested the trunk version (the one that uses db-lib internally instead of libtds) and got 50,000 or so logged with 5-10 minutes. |
| 21:33.51 | seanbright | codefreeze-lap: no sir |
| 21:34.29 | mocker | seanbright: Are you ok w/ me pasting that stuff in so others can know what you said about the FreeTDS install? |
| 21:34.55 | seanbright | mocker: sure, i guess. it's by no means comprehensive. |
| 21:35.18 | mocker | Yeah, but if that page is crappy it at least lets people know. :) |
| 21:35.47 | seanbright | mocker: the page implies that you *need* odbc for freetds, which you do not |
| 21:35.58 | mocker | right. |
| 21:36.03 | seanbright | unixodbc can /use/ freetds, and you can in turn use it with cdr_odbc |
| 21:36.24 | seanbright | but it's not required for getting CDRs into mssql |
| 21:37.23 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 21:39.15 | lesouvage | seanbright: Does it matter/makes a different what name and password will be used for the ms sql database? Can i ask to use the name "asterisk_cdr" and a proper password? |
| 21:39.23 | Madkiss | oida! |
| 21:39.34 | seanbright | lesouvage: all of that is settable in cdr_tds.conf |
| 21:40.23 | lesouvage | seabright: thanks, I haven't done anthing windows like in years but you gave me confidence ;-) |
| 21:40.43 | seanbright | lesouvage: i do what i can |
| 21:41.20 | seanbright | and on that note, i am going to take a nap |
| 21:41.21 | *** join/#asterisk tuxfoo21 (n=tmmarini@pool-72-65-135-149.chrlwv.east.verizon.net) |
| 21:42.30 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-5.dyn.embarqhsd.net) |
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| 21:49.40 | *** join/#asterisk surfdue (n=Anthony@unaffiliated/surfdue) |
| 21:49.52 | surfdue | anyone know a very affordable hosted pbx solution? |
| 21:50.14 | surfdue | Unlimited local, and paid 800 number would be great but I cant find any. |
| 21:58.36 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 22:03.43 | rigid | i get lots of "Got RTP packet from xx.xx.xx.xx:7082 (type 0, seq 93, ts 22320, len 240) Sent RTP packet to xx.xx.xx.xx:27432 (type 0, seq 23406, ts 22160, len 160)" but i still can't hear anything |
| 22:03.54 | rigid | is there a way to debug what codecs are currently used? |
| 22:05.08 | rigid | everything works fine with my regular setup... but if i "natively bridge two SIP/connections", there's no sound after the call is received |
| 22:05.15 | *** part/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 22:05.42 | rigid | i'm using "Dial(SIP/fubar/foo&SIP/fubar/baz)" |
| 22:07.44 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 22:08.04 | *** join/#asterisk coil (n=some@unaffiliated/coil) |
| 22:08.29 | javb | I've an IAX trunk between two PBX. If i dont use username and secret option, call from each other go great. If use username and password, i've "No authotity found" problem, and i solve this adding context and type of my own pbx on each pbx.. any idea why this? |
| 22:09.40 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 22:10.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 22:10.57 | *** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233) |
| 22:12.04 | edibrac | for a block of DIDs I got a few weeks ago, I'm now getting "we're sorry you call cannot be completed as dialed. Please check the number and dial again. 000 000" |
| 22:12.41 | jasonwoot | conference call recording is not making it from the spool directory to the monitor directory when mixmonitor ends. Any suggestions? |
| 22:13.12 | Carlos_PHX | I have a project calling for a VoIP appliance, preferably Asterisk based, and have never used one. Thought someone might have suggestions. I need 3 FXO ports, and will have at most 3 concurrent calls. It will have 3 ATAs connecting to it from remote. |
| 22:14.23 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 22:14.26 | C4away | good morning |
| 22:14.51 | edibrac | the XO rep says 000 000, is an "internal code" |
| 22:15.15 | C4away | I need to migrate to SQL CDR but retain a month or two where both billing systems will remain functional |
| 22:15.17 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 22:15.20 | edibrac | is there a technical term for this sort of code? or this is just vendor (XO) specific? |
| 22:15.42 | C4away | if I add the pgsql module my csv files will still be generated right? |
| 22:19.08 | jstocks | Question: I have a asterisk setup that I have been playing with, and I have setup confrence, and the dication and they all work fine. But when try to do a AGI script, I can't seem to get any DTMF/digits from my calls? any idea what I may need to check? |
| 22:19.15 | *** join/#asterisk Ivan74 (n=1040B2DE@ip-21-184.sn1.eutelia.it) |
| 22:19.33 | Ivan74 | hi to all |
| 22:19.46 | Ivan74 | someone can help me? |
| 22:21.02 | Qwell | ~help |
| 22:21.06 | Qwell | ...stupid bot |
| 22:21.07 | Qwell | ~ask |
| 22:21.08 | jbot | methinks ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:22.02 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 22:22.22 | De_Mon | jbot can help you ask him your question if you don't believe me |
| 22:22.43 | Ivan74 | pvt? |
| 22:25.09 | Ivan74 | wow what a channel! |
| 22:25.21 | Qwell | Ivan74: Nobody can answer your question if you don't ask it. |
| 22:25.36 | Ivan74 | ok let's see |
| 22:26.19 | Ivan74 | I have an asterisk server connected to isdn phones through b400P card configured with 2 ports in TE mode and the other 2 in NT |
| 22:26.43 | Ivan74 | TE ports connected to ISDN lines |
| 22:26.47 | Ivan74 | NT to phones |
| 22:26.57 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:26.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:28.17 | Ivan74 | after that my head it's exploded I success configured the server so that the voip phones calls voip lines, isdn lines and isdn phones |
| 22:28.47 | Ivan74 | isdn phones can call voip lines, isdn lines and voip phones |
| 22:28.54 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-5-248.revip2.asianet.co.th) |
| 22:29.03 | Ivan74 | but one problem still remain.... |
| 22:30.18 | Ivan74 | when I receive a call from voip or isdn lines all the phone starts to ring except the isdn, the isdn phones starts to ring after about 15 second! |
| 22:30.20 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:30.51 | Ivan74 | and I have this problem even if Ii call an isdn phone from a voip phone (internal calls) |
| 22:31.54 | Ivan74 | so anyone can help me now? |
| 22:32.47 | *** join/#asterisk cj (n=cjac@pdpc/supporter/monthlybronze/cj) |
| 22:33.54 | cj | what would the bottleneck be if one wanted to use asterisk as a PSTN switch? |
| 22:34.42 | edibrac | cj: the # of lines you have coming in? |
| 22:34.47 | cj | de-muxing the t1 and processing the ss7 signaling? |
| 22:34.49 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 22:35.21 | Ivan74 | why my isdn phone starts to ring after 15 seconds? |
| 22:35.35 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 22:35.41 | cj | edibrac: okay... how many lines do you think a 2G dual core 2GHz server could handle? |
| 22:36.39 | edibrac | i have no clue, not had to deal with this problem |
| 22:36.45 | jaytee | lebenty-leven? |
| 22:37.15 | De_Mon | cj depends on your codec and what those calls are doing (briding to other media/codecs) |
| 22:37.18 | cj | jaytee: I was thinking that number was just about right |
| 22:37.18 | l2trace99 | cj: http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 22:38.02 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:38.16 | edibrac | is there script out there that I can run against .. my cdr file that can tell me my average concurrent line usage? |
| 22:38.31 | cj | De_Mon: oh? would we need to decode the data on those channels? I would think we'd just pass the signal down the stream, not look at the content of the line |
| 22:38.32 | edibrac | or something that i can just run and bammm instant gratification? |
| 22:38.50 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279641049.dsl.bell.ca) |
| 22:39.20 | jaytee | it also depends on how many concurrent calls. I have a PRI circuit with 2 T1 spans which gives me 46 channels but most of the time less than 10% are active and my 2ghz Quad Xeon rarely registers a bump in cpu utilization above 5%. |
| 22:40.01 | edibrac | jaytee: what are you looking at to get those stats? |
| 22:40.05 | *** join/#asterisk oilinki7 (n=oil@ppp-124-120-6-223.revip2.asianet.co.th) |
| 22:40.06 | edibrac | for usage |
| 22:41.16 | cj | jaytee: are you terminating the calls or translating the bearer channel to a different format? |
| 22:41.39 | cj | (inclusive) |
| 22:41.52 | jaytee | cj, mostly bridging calls between the 2 spans and I'm not doing any transcoding |
| 22:42.33 | cj | jaytee: cool. can you point me to the bridge setup/teardown code? |
| 22:43.13 | Ivan74 | how to set internal misdn users? |
| 22:44.20 | jameswf | I dont know why i could have swore asterisknow had 1.6,,, eh |
| 22:44.24 | jaytee | cj, not sure what you're asking for. I have calls coming in from or going out to 1 span that's connected to a Nortel PBX and the other span connects to my telco directly from Asterisk. Outbound calls from Asterisk use the outbound span as well as bridging outbound calls from Nortel from the other span. Calls from Asterisk to Nortel use the same span inbound. |
| 22:45.24 | cj | jaytee: I'm still not up on all of the jargon yet... I did read that telco 101 pdf, though :) |
| 22:45.26 | jaytee | jameswf, nope, the difference is CentOS instead of rPath and FreePBX instead of asterisk-gui |
| 22:45.40 | cj | what I'm curious about is... where in the asterisk code does the bridging code live |
| 22:46.08 | jameswf | just did yum remove asterisk*; yum install asterisk16 |
| 22:46.10 | jameswf | :) |
| 22:46.26 | cj | when a call comes in from your telco and you need to route it to the PBX, it passes through asterisk at least for a short period of time, no? |
| 22:48.25 | jaytee | cj, ok. lemme tell you how my calls get bridged. I have a nortel user who wants to make an outbound call. My Option 11 is setup to route all calls out my "span 2" on that pbx which connects to Asterisk. In the context defined in zapata.conf for that span I put a filter that determines whether the dialed number is an Asterisk extension or an outbound call. If the latter it uses the dial statement to dial out the other span to my telco. The bridgin |
| 22:48.25 | jaytee | g is accomplished at the zaptel driver layer when the incoming call from Nortel needs to connect to a channel on the other span. |
| 22:50.26 | cj | okay. so it sounds like most of the work is handled by the t1 card, and a very small portion of the work is handled in software by the zaptel driver. |
| 22:53.05 | *** join/#asterisk outtolunc (n=me@c-24-130-75-122.hsd1.ca.comcast.net) |
| 22:53.39 | jaytee | I'm using some tricks to route the calls between the two pbxs. Nortel has a feature called phantom TNs (terminal numbers) that are like virtual lines instead of actual analog or digital lines on a line card. I program a phantom TN as an extension that gets externally forwarded to an "outside number". The outside number has a NXX that isn't a valid NXX in my area code. I match by callerid so if any call coming in from the Nortel system has that 3 |
| 22:53.39 | jaytee | digit NXX I route it to the internal extensions context in Asterisk after stripping the NXX digits out leaving only the 4 digit extension, if the NXX is anything else or a 10 digit number it routes it to my outbound context that uses pattern matches for local or long distance to dial. |
| 22:55.08 | cj | was zaptel re-named dahdi ? |
| 22:55.10 | jaytee | and to dial the Nortel from an Asterisk extension the user just dials the 4 digit number and it tries to find a match in the internal extensions context. the last possible match is _XXXX which dials the nortel span. |
| 22:55.17 | jaytee | cj, in 1.6 it was |
| 22:56.10 | UnixDawg | I would have just set a pin lik _34xxxx |
| 22:56.34 | UnixDawg | that way you can still haev 4 digit exten on the pbx |
| 22:56.49 | jaytee | once my Nortel pbx is decommisioned the dialplan will become much simpler and I'll be able to use a GUI |
| 22:57.19 | *** join/#asterisk anthm][ (n=anthm@68-31-185-234.area4.spcsdns.net) |
| 22:57.48 | *** join/#asterisk anthm][ (n=anthm@68-31-185-234.area4.spcsdns.net) |
| 22:58.43 | jaytee | UnixDawg, yeah I could have done that but I felt it was better to not have to retrain users to dial extra digits. |
| 22:59.24 | anthm | cj, my net died right when I answered you, did it go through? |
| 22:59.50 | cj | anthm: doesn't look like it... let me check lastlog |
| 23:00.00 | cj | nopers |
| 23:00.03 | anthm | cj, it might be moved from the last time i checked but app_dial creates the channel then sends it to bridge via a function in res_features who then dips down into channel.c and sometimes dips again into the channel driver's specific code if both are the same type, then when you dial dtmf it exits all the way back up to res_features to parse it and starts digging its way down again .... |
| 23:00.06 | jaytee | and it was done this way so we could avoid the cost of upgrading the Option 11C with a SIP Gateway which would have cost as much as we've spent on the * server, 30 Polycom phones, etc. |
| 23:00.08 | anthm | there =D |
| 23:00.26 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 23:01.53 | *** part/#asterisk Ivan74 (n=1040B2DE@ip-21-184.sn1.eutelia.it) |
| 23:09.55 | cj | thanks, anthm |
| 23:10.35 | cj | anthm: so, if that's all the asterisk box is doing, you think it will scale pretty well? Are there any contention points? |
| 23:11.31 | cj | I'm looking to have a Tormenta II card built... my brother's taking an EE course at the local university, and I figured he might enjoy the project |
| 23:12.01 | jaytee | building your own card? Tormentas aren't supported anymore if I'm not mistaken. |
| 23:12.35 | cj | oh? that's too bad. :( |
| 23:12.43 | jaytee | and that just sounds like begging for trouble |
| 23:12.59 | cj | well... he *is* a EE student... |
| 23:13.03 | cj | an |
| 23:13.40 | cj | "EE student begging for trouble" sounds redundant to me :) |
| 23:14.05 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 23:14.12 | jaytee | so that means any student studying nuclear engineering should try building a breeder reactor in their garage? |
| 23:14.23 | cj | jaytee: and why not!? :) |
| 23:14.27 | drmessano | Oh god |
| 23:14.28 | cj | aside from the obvious.... |
| 23:14.29 | drmessano | Not this convo |
| 23:14.44 | drmessano | Every forum and IRC channel must have this same convo every 3 months like clockwork |
| 23:14.59 | cj | drmessano: hmm? which? |
| 23:15.00 | jaytee | about what? building your own card? |
| 23:15.02 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 23:15.14 | cj | maybe I should take it to -dev ? |
| 23:15.14 | drmessano | One I like to call: "What do you mean I can't have nuclear fission in my basement, FBI agent?" |
| 23:15.52 | cj | drmessano's had a reactor in his basement for years and it hasn't affected him! |
| 23:16.05 | drmessano | No |
| 23:16.08 | drmessano | I am not a lunatic |
| 23:16.26 | cj | (all proof to the contrary notwithstanding) |
| 23:16.29 | cj | ducks |
| 23:16.29 | outtolunc | waits for drmessano to argue with himself <G> |
| 23:16.45 | jaytee | One of the stupidest things I've seen in any book on VOIP is the part in the O'Reilly book, Voip Hacks that shows how to modify the zaptel code to use a winmodem as an FXO. Even though the author includes a caveat he should still be shot for that or at least forced to have sex with a really fat, ugly woman with bad breath. |
| 23:16.56 | cj | drmessano: we got off on the wrong foot. I apologize for giving you a hard time. |
| 23:17.40 | drmessano | cj: No need to apologize, I have no less desire to see you rot in hell than the next guy.. |
| 23:18.00 | drmessano | So in fact, there was no "wrong foot" |
| 23:18.04 | cj | jaytee: heh. on that subject, why isn't there a combined FXO/FXS module? |
| 23:18.30 | jaytee | cj, because they have different functions? |
| 23:18.33 | drmessano | Because there no technological basic for it? |
| 23:18.40 | drmessano | errr |
| 23:18.46 | drmessano | Because theres no technological basis for it? |
| 23:18.53 | cj | jaytee: but if you've got only one analog line, wouldn't you want to both receive and send calls using it? |
| 23:18.58 | jaytee | and you can get a card with an FXS and and FXO on it, or 3 of 1 and 1 of another, 7 of 1 etc. etc. |
| 23:19.08 | jaytee | cj, DUH!!!! |
| 23:19.19 | cj | jaytee: hmm? what am I missing? |
| 23:19.28 | jaytee | what makes you think you can't use an FXO to both recieve and send calls? |
| 23:19.54 | cj | jaytee: maybe I didn't read the docs thoroughly enough |
| 23:20.00 | jaytee | ya think? |
| 23:21.49 | jaytee | hmmmm, this doesn't look good! I just found [TK]D-Fender's picture on this milk carton. |
| 23:22.21 | lesouvage | jaytee: he is missed, and there is a reward for those who brings him back? |
| 23:23.09 | *** join/#asterisk StephenF (n=StephenF@198.144.197.28) |
| 23:23.30 | jaytee | carton doesn't say anything about a reward |
| 23:23.57 | lesouvage | Or is he just advertising his asterisk related services, milk cartons is certainly a original channel to get your message through. |
| 23:24.13 | cj | so an FXO is an FXS with the addition of being able to call out? |
| 23:24.20 | jaytee | but he hasn't been in here for 3 1/2 days which is kind of a record for him. |
| 23:24.26 | jaytee | cj, no |
| 23:24.43 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:24.49 | *** join/#asterisk anthm][ (n=anthm@68-31-185-234.area4.spcsdns.net) |
| 23:24.51 | lesouvage | jaytee: yes. |
| 23:25.08 | jaytee | cj, an FXO is for connecting to your telco over a POTS line (analog). An FXS is to connect to a phone. |
| 23:25.25 | jaytee | FXS supplies ringing voltage to the phone, FXO doesn't. |
| 23:25.55 | cj | ah. thanks for clearing that up. |
| 23:26.23 | jaytee | it well covered in the first few chapters of the book that deal with zaptel |
| 23:26.27 | jaytee | and zapata |
| 23:26.33 | cj | ah. which book is that? |
| 23:26.39 | jaytee | ~book |
| 23:26.39 | jbot | book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:27.01 | jaytee | that book, it should become your bible |
| 23:27.49 | cj | is there anything that replaces the Tormenta II card that I can build using off-the-shelf pieces? |
| 23:28.02 | jaytee | not that I'm aware of |
| 23:28.09 | lesouvage | jaytee: check this http://www.100factsabout.com/ircuser/%5BTK%5DD-Fender |
| 23:29.15 | anthm | I like the first edition better |
| 23:29.29 | jaytee | lesouvage, hahaha |
| 23:29.40 | jaytee | script kiddies are so much fun :-) |
| 23:35.22 | drmessano | http://www.100factsabout.com/My/Penis |
| 23:35.25 | drmessano | Thats kinda funny |
| 23:36.31 | jaytee | drmessano, you're sick! I think that's why we get along so well :-) |
| 23:36.31 | De_Mon | hrm... I just did a sip reload in 1.6.0 and it says Reloading SIP but no users were loaded |
| 23:37.24 | jaytee | drmessano, did you see the "leaked Palin sex tape" on Digg? |
| 23:37.59 | De_Mon | ponders opened and saved the file and now it sees everyone |
| 23:38.43 | *** join/#asterisk seanmh (n=seanmh@c-68-35-21-64.hsd1.nm.comcast.net) |
| 23:38.50 | De_Mon | yay, it's allive |
| 23:39.20 | drmessano | no lol |
| 23:40.54 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 23:48.47 | ManxPower | De_Mon: long sip reload times indicate a DNS issue. |
| 23:49.02 | ManxPower | Usually reverse lookup of the IPs on the system |
| 23:54.46 | CGMChris | Hello ManxPower. I have a problem (probably trivial), but maybe you can help. I can make outgoing calls w/ gizmo5 sip on asterisk. The gizmo5 software receives incoming calls, but asterisk does not. asterisk -rvvv does not show ANYTHING on incoming calls. Thoughts? |
| 23:55.03 | ManxPower | CGMChris: I don't use "Gizmo" |
| 23:55.13 | CGMChris | Gizmo5 is a SIP trunk provider. |
| 23:55.21 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:55.39 | ManxPower | CGMChris: then anyone here can help you. |
| 23:55.58 | CGMChris | ManxPower: Ok, anyone? |
| 23:56.26 | RMod | anyone use mediatrix ata's? |
| 23:57.03 | CGMChris | I guess this could take awhile... :-\ |
| 23:57.07 | CGMChris | anyone now? |